From 452a256898e7ca88115aa02d3851e67994ce3e19 Mon Sep 17 00:00:00 2001 From: anish kumar Date: Sun, 23 Oct 2016 21:03:53 -0700 Subject: ASoC: Codec to codec dai link description Signed-off-by: anish kumar Acked-by: Charles Keepax Signed-off-by: Mark Brown --- Documentation/sound/alsa/soc/codec_to_codec.txt | 103 ++++++++++++++++++++++++ 1 file changed, 103 insertions(+) create mode 100644 Documentation/sound/alsa/soc/codec_to_codec.txt (limited to 'Documentation/sound') diff --git a/Documentation/sound/alsa/soc/codec_to_codec.txt b/Documentation/sound/alsa/soc/codec_to_codec.txt new file mode 100644 index 000000000000..704a6483652c --- /dev/null +++ b/Documentation/sound/alsa/soc/codec_to_codec.txt @@ -0,0 +1,103 @@ +Creating codec to codec dai link for ALSA dapm +=================================================== + +Mostly the flow of audio is always from CPU to codec so your system +will look as below: + + --------- --------- +| | dai | | + CPU -------> codec +| | | | + --------- --------- + +In case your system looks as below: + --------- + | | + codec-2 + | | + --------- + | + dai-2 + | + ---------- --------- +| | dai-1 | | + CPU -------> codec-1 +| | | | + ---------- --------- + | + dai-3 + | + --------- + | | + codec-3 + | | + --------- + +Suppose codec-2 is a bluetooth chip and codec-3 is connected to +a speaker and you have a below scenario: +codec-2 will receive the audio data and the user wants to play that +audio through codec-3 without involving the CPU.This +aforementioned case is the ideal case when codec to codec +connection should be used. + +Your dai_link should appear as below in your machine +file: + +/* + * this pcm stream only supports 24 bit, 2 channel and + * 48k sampling rate. + */ +static const struct snd_soc_pcm_stream dsp_codec_params = { + .formats = SNDRV_PCM_FMTBIT_S24_LE, + .rate_min = 48000, + .rate_max = 48000, + .channels_min = 2, + .channels_max = 2, +}; + +{ + .name = "CPU-DSP", + .stream_name = "CPU-DSP", + .cpu_dai_name = "samsung-i2s.0", + .codec_name = "codec-2, + .codec_dai_name = "codec-2-dai_name", + .platform_name = "samsung-i2s.0", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBM_CFM, + .ignore_suspend = 1, + .params = &dsp_codec_params, +}, +{ + .name = "DSP-CODEC", + .stream_name = "DSP-CODEC", + .cpu_dai_name = "wm0010-sdi2", + .codec_name = "codec-3, + .codec_dai_name = "codec-3-dai_name", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBM_CFM, + .ignore_suspend = 1, + .params = &dsp_codec_params, +}, + +Above code snippet is motivated from sound/soc/samsung/speyside.c. + +Note the "params" callback which lets the dapm know that this +dai_link is a codec to codec connection. + +In dapm core a route is created between cpu_dai playback widget +and codec_dai capture widget for playback path and vice-versa is +true for capture path. In order for this aforementioned route to get +triggered, DAPM needs to find a valid endpoint which could be either +a sink or source widget corresponding to playback and capture path +respectively. + +In order to trigger this dai_link widget, a thin codec driver for +the speaker amp can be created as demonstrated in wm8727.c file, it +sets appropriate constraints for the device even if it needs no control. + +Make sure to name your corresponding cpu and codec playback and capture +dai names ending with "Playback" and "Capture" respectively as dapm core +will link and power those dais based on the name. + +Note that in current device tree there is no way to mark a dai_link +as codec to codec. However, it may change in future. -- cgit v1.2.3-58-ga151 From 3a5182c04a020e5ba5339c3b393cae0d17b74ef5 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 10 Nov 2016 17:37:15 +0100 Subject: ALSA: doc: Remove alsa-parameters.txt This is a really obsoleted information, effectively just listing the module names. Let's get rid of it for avoiding confusions. Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/alsa-parameters.txt | 135 --------------------------- 1 file changed, 135 deletions(-) delete mode 100644 Documentation/sound/alsa/alsa-parameters.txt (limited to 'Documentation/sound') diff --git a/Documentation/sound/alsa/alsa-parameters.txt b/Documentation/sound/alsa/alsa-parameters.txt deleted file mode 100644 index 0fa40679b080..000000000000 --- a/Documentation/sound/alsa/alsa-parameters.txt +++ /dev/null @@ -1,135 +0,0 @@ - ALSA Kernel Parameters - ~~~~~~~~~~~~~~~~~~~~~~ - -See Documentation/kernel-parameters.txt for general information on -specifying module parameters. - -This document may not be entirely up to date and comprehensive. The command -"modinfo -p ${modulename}" shows a current list of all parameters of a loadable -module. Loadable modules, after being loaded into the running kernel, also -reveal their parameters in /sys/module/${modulename}/parameters/. Some of these -parameters may be changed at runtime by the command -"echo -n ${value} > /sys/module/${modulename}/parameters/${parm}". - - - snd-ad1816a= [HW,ALSA] - - snd-ad1848= [HW,ALSA] - - snd-ali5451= [HW,ALSA] - - snd-als100= [HW,ALSA] - - snd-als4000= [HW,ALSA] - - snd-azt2320= [HW,ALSA] - - snd-cmi8330= [HW,ALSA] - - snd-cmipci= [HW,ALSA] - - snd-cs4231= [HW,ALSA] - - snd-cs4232= [HW,ALSA] - - snd-cs4236= [HW,ALSA] - - snd-cs4281= [HW,ALSA] - - snd-cs46xx= [HW,ALSA] - - snd-dt019x= [HW,ALSA] - - snd-dummy= [HW,ALSA] - - snd-emu10k1= [HW,ALSA] - - snd-ens1370= [HW,ALSA] - - snd-ens1371= [HW,ALSA] - - snd-es968= [HW,ALSA] - - snd-es1688= [HW,ALSA] - - snd-es18xx= [HW,ALSA] - - snd-es1938= [HW,ALSA] - - snd-es1968= [HW,ALSA] - - snd-fm801= [HW,ALSA] - - snd-gusclassic= [HW,ALSA] - - snd-gusextreme= [HW,ALSA] - - snd-gusmax= [HW,ALSA] - - snd-hdsp= [HW,ALSA] - - snd-ice1712= [HW,ALSA] - - snd-intel8x0= [HW,ALSA] - - snd-interwave= [HW,ALSA] - - snd-interwave-stb= - [HW,ALSA] - - snd-korg1212= [HW,ALSA] - - snd-maestro3= [HW,ALSA] - - snd-mpu401= [HW,ALSA] - - snd-mtpav= [HW,ALSA] - - snd-nm256= [HW,ALSA] - - snd-opl3sa2= [HW,ALSA] - - snd-opti92x-ad1848= - [HW,ALSA] - - snd-opti92x-cs4231= - [HW,ALSA] - - snd-opti93x= [HW,ALSA] - - snd-pmac= [HW,ALSA] - - snd-rme32= [HW,ALSA] - - snd-rme96= [HW,ALSA] - - snd-rme9652= [HW,ALSA] - - snd-sb8= [HW,ALSA] - - snd-sb16= [HW,ALSA] - - snd-sbawe= [HW,ALSA] - - snd-serial= [HW,ALSA] - - snd-sgalaxy= [HW,ALSA] - - snd-sonicvibes= [HW,ALSA] - - snd-sun-amd7930= - [HW,ALSA] - - snd-sun-cs4231= [HW,ALSA] - - snd-trident= [HW,ALSA] - - snd-usb-audio= [HW,ALSA,USB] - - snd-via82xx= [HW,ALSA] - - snd-virmidi= [HW,ALSA] - - snd-wavefront= [HW,ALSA] - - snd-ymfpci= [HW,ALSA] -- cgit v1.2.3-58-ga151 From 8551914a5e19094255a0e2aadb24f70736f7ba7d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 2 Nov 2016 21:30:39 +0100 Subject: ALSA: doc: ReSTize alsa-driver-api document A simple conversion of alsa-driver-api document from DocBook to ReST. It's moved to the new Documentation/sound/kernel-api subdirectory that will contain other ALSA kernel API documents. The GPL legal note was removed, as it's superfluous (and doesn't fit with ReST kernel docs pretty well). Signed-off-by: Takashi Iwai --- Documentation/DocBook/Makefile | 2 +- Documentation/DocBook/alsa-driver-api.tmpl | 142 --------------------- Documentation/index.rst | 1 + Documentation/sound/index.rst | 15 +++ Documentation/sound/kernel-api/alsa-driver-api.rst | 134 +++++++++++++++++++ Documentation/sound/kernel-api/index.rst | 7 + 6 files changed, 158 insertions(+), 143 deletions(-) delete mode 100644 Documentation/DocBook/alsa-driver-api.tmpl create mode 100644 Documentation/sound/index.rst create mode 100644 Documentation/sound/kernel-api/alsa-driver-api.rst create mode 100644 Documentation/sound/kernel-api/index.rst (limited to 'Documentation/sound') diff --git a/Documentation/DocBook/Makefile b/Documentation/DocBook/Makefile index fdf8232d0eeb..e173497959fa 100644 --- a/Documentation/DocBook/Makefile +++ b/Documentation/DocBook/Makefile @@ -13,7 +13,7 @@ DOCBOOKS := z8530book.xml \ gadget.xml libata.xml mtdnand.xml librs.xml rapidio.xml \ genericirq.xml s390-drivers.xml uio-howto.xml scsi.xml \ debugobjects.xml sh.xml regulator.xml \ - alsa-driver-api.xml writing-an-alsa-driver.xml \ + writing-an-alsa-driver.xml \ tracepoint.xml w1.xml \ writing_musb_glue_layer.xml crypto-API.xml iio.xml diff --git a/Documentation/DocBook/alsa-driver-api.tmpl b/Documentation/DocBook/alsa-driver-api.tmpl deleted file mode 100644 index 53f439dcc94b..000000000000 --- a/Documentation/DocBook/alsa-driver-api.tmpl +++ /dev/null @@ -1,142 +0,0 @@ - - - - - - - - - The ALSA Driver API - - - - This document is free; you can redistribute it and/or modify it - under the terms of the GNU General Public License as published by - the Free Software Foundation; either version 2 of the License, or - (at your option) any later version. - - - - This document is distributed in the hope that it will be useful, - but WITHOUT ANY WARRANTY; without even the - implied warranty of MERCHANTABILITY or FITNESS FOR A - PARTICULAR PURPOSE. See the GNU General Public License - for more details. - - - - You should have received a copy of the GNU General Public - License along with this program; if not, write to the Free - Software Foundation, Inc., 59 Temple Place, Suite 330, Boston, - MA 02111-1307 USA - - - - - - - - Management of Cards and Devices - Card Management -!Esound/core/init.c - - Device Components -!Esound/core/device.c - - Module requests and Device File Entries -!Esound/core/sound.c - - Memory Management Helpers -!Esound/core/memory.c -!Esound/core/memalloc.c - - - PCM API - PCM Core -!Esound/core/pcm.c -!Esound/core/pcm_lib.c -!Esound/core/pcm_native.c -!Iinclude/sound/pcm.h - - PCM Format Helpers -!Esound/core/pcm_misc.c - - PCM Memory Management -!Esound/core/pcm_memory.c - - PCM DMA Engine API -!Esound/core/pcm_dmaengine.c -!Iinclude/sound/dmaengine_pcm.h - - - Control/Mixer API - General Control Interface -!Esound/core/control.c - - AC97 Codec API -!Esound/pci/ac97/ac97_codec.c -!Esound/pci/ac97/ac97_pcm.c - - Virtual Master Control API -!Esound/core/vmaster.c -!Iinclude/sound/control.h - - - MIDI API - Raw MIDI API -!Esound/core/rawmidi.c - - MPU401-UART API -!Esound/drivers/mpu401/mpu401_uart.c - - - Proc Info API - Proc Info Interface -!Esound/core/info.c - - - Compress Offload - Compress Offload API -!Esound/core/compress_offload.c -!Iinclude/uapi/sound/compress_offload.h -!Iinclude/uapi/sound/compress_params.h -!Iinclude/sound/compress_driver.h - - - ASoC - ASoC Core API -!Iinclude/sound/soc.h -!Esound/soc/soc-core.c - -!Esound/soc/soc-devres.c -!Esound/soc/soc-io.c -!Esound/soc/soc-pcm.c -!Esound/soc/soc-ops.c -!Esound/soc/soc-compress.c - - ASoC DAPM API -!Esound/soc/soc-dapm.c - - ASoC DMA Engine API -!Esound/soc/soc-generic-dmaengine-pcm.c - - - Miscellaneous Functions - Hardware-Dependent Devices API -!Esound/core/hwdep.c - - Jack Abstraction Layer API -!Iinclude/sound/jack.h -!Esound/core/jack.c -!Esound/soc/soc-jack.c - - ISA DMA Helpers -!Esound/core/isadma.c - - Other Helper Macros -!Iinclude/sound/core.h - - - - diff --git a/Documentation/index.rst b/Documentation/index.rst index c53d089455a4..115c551da5f5 100644 --- a/Documentation/index.rst +++ b/Documentation/index.rst @@ -18,6 +18,7 @@ Contents: media/index gpu/index 80211/index + sound/index Indices and tables ================== diff --git a/Documentation/sound/index.rst b/Documentation/sound/index.rst new file mode 100644 index 000000000000..280a57115f00 --- /dev/null +++ b/Documentation/sound/index.rst @@ -0,0 +1,15 @@ +=================================== +Linux Sound Subsystem Documentation +=================================== + +.. toctree:: + :maxdepth: 2 + + kernel-api/index + +.. only:: subproject + + Indices + ======= + + * :ref:`genindex` diff --git a/Documentation/sound/kernel-api/alsa-driver-api.rst b/Documentation/sound/kernel-api/alsa-driver-api.rst new file mode 100644 index 000000000000..14cd138989e3 --- /dev/null +++ b/Documentation/sound/kernel-api/alsa-driver-api.rst @@ -0,0 +1,134 @@ +=================== +The ALSA Driver API +=================== + +Management of Cards and Devices +=============================== + +Card Management +--------------- +.. kernel-doc:: sound/core/init.c + +Device Components +----------------- +.. kernel-doc:: sound/core/device.c + +Module requests and Device File Entries +--------------------------------------- +.. kernel-doc:: sound/core/sound.c + +Memory Management Helpers +------------------------- +.. kernel-doc:: sound/core/memory.c +.. kernel-doc:: sound/core/memalloc.c + + +PCM API +======= + +PCM Core +-------- +.. kernel-doc:: sound/core/pcm.c +.. kernel-doc:: sound/core/pcm_lib.c +.. kernel-doc:: sound/core/pcm_native.c +.. kernel-doc:: include/sound/pcm.h + +PCM Format Helpers +------------------ +.. kernel-doc:: sound/core/pcm_misc.c + +PCM Memory Management +--------------------- +.. kernel-doc:: sound/core/pcm_memory.c + +PCM DMA Engine API +------------------ +.. kernel-doc:: sound/core/pcm_dmaengine.c +.. kernel-doc:: include/sound/dmaengine_pcm.h + +Control/Mixer API +================= + +General Control Interface +------------------------- +.. kernel-doc:: sound/core/control.c + +AC97 Codec API +-------------- +.. kernel-doc:: sound/pci/ac97/ac97_codec.c +.. kernel-doc:: sound/pci/ac97/ac97_pcm.c + +Virtual Master Control API +-------------------------- +.. kernel-doc:: sound/core/vmaster.c +.. kernel-doc:: include/sound/control.h + +MIDI API +======== + +Raw MIDI API +------------ +.. kernel-doc:: sound/core/rawmidi.c + +MPU401-UART API +--------------- +.. kernel-doc:: sound/drivers/mpu401/mpu401_uart.c + +Proc Info API +============= + +Proc Info Interface +------------------- +.. kernel-doc:: sound/core/info.c + +Compress Offload +================ + +Compress Offload API +-------------------- +.. kernel-doc:: sound/core/compress_offload.c +.. kernel-doc:: include/uapi/sound/compress_offload.h +.. kernel-doc:: include/uapi/sound/compress_params.h +.. kernel-doc:: include/sound/compress_driver.h + +ASoC +==== + +ASoC Core API +------------- +.. kernel-doc:: include/sound/soc.h +.. kernel-doc:: sound/soc/soc-core.c +.. kernel-doc:: sound/soc/soc-devres.c +.. kernel-doc:: sound/soc/soc-io.c +.. kernel-doc:: sound/soc/soc-pcm.c +.. kernel-doc:: sound/soc/soc-ops.c +.. kernel-doc:: sound/soc/soc-compress.c + +ASoC DAPM API +------------- +.. kernel-doc:: sound/soc/soc-dapm.c + +ASoC DMA Engine API +------------------- +.. kernel-doc:: sound/soc/soc-generic-dmaengine-pcm.c + +Miscellaneous Functions +======================= + +Hardware-Dependent Devices API +------------------------------ +.. kernel-doc:: sound/core/hwdep.c + +Jack Abstraction Layer API +-------------------------- +.. kernel-doc:: include/sound/jack.h +.. kernel-doc:: sound/core/jack.c +.. kernel-doc:: sound/soc/soc-jack.c + +ISA DMA Helpers +--------------- +.. kernel-doc:: sound/core/isadma.c + +Other Helper Macros +------------------- +.. kernel-doc:: include/sound/core.h diff --git a/Documentation/sound/kernel-api/index.rst b/Documentation/sound/kernel-api/index.rst new file mode 100644 index 000000000000..73c13497dec7 --- /dev/null +++ b/Documentation/sound/kernel-api/index.rst @@ -0,0 +1,7 @@ +ALSA Kernel API Documentation +============================= + +.. toctree:: + :maxdepth: 2 + + alsa-driver-api -- cgit v1.2.3-58-ga151 From 7ddedebb03b7ec030c528ebacdd43e45373476e3 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 29 Sep 2016 18:21:46 +0200 Subject: ALSA: doc: ReSTize writing-an-alsa-driver document Another simple conversion from DocBook to ReST. This required a few manual fixups and reformats, but the most of contents are kept as is. Signed-off-by: Takashi Iwai --- Documentation/DocBook/Makefile | 3 +- Documentation/DocBook/writing-an-alsa-driver.tmpl | 6206 -------------------- Documentation/sound/kernel-api/index.rst | 1 + .../sound/kernel-api/writing-an-alsa-driver.rst | 4219 +++++++++++++ 4 files changed, 4221 insertions(+), 6208 deletions(-) delete mode 100644 Documentation/DocBook/writing-an-alsa-driver.tmpl create mode 100644 Documentation/sound/kernel-api/writing-an-alsa-driver.rst (limited to 'Documentation/sound') diff --git a/Documentation/DocBook/Makefile b/Documentation/DocBook/Makefile index e173497959fa..72f78ae46c10 100644 --- a/Documentation/DocBook/Makefile +++ b/Documentation/DocBook/Makefile @@ -12,8 +12,7 @@ DOCBOOKS := z8530book.xml \ kernel-api.xml filesystems.xml lsm.xml usb.xml kgdb.xml \ gadget.xml libata.xml mtdnand.xml librs.xml rapidio.xml \ genericirq.xml s390-drivers.xml uio-howto.xml scsi.xml \ - debugobjects.xml sh.xml regulator.xml \ - writing-an-alsa-driver.xml \ + 80211.xml debugobjects.xml sh.xml regulator.xml \ tracepoint.xml w1.xml \ writing_musb_glue_layer.xml crypto-API.xml iio.xml diff --git a/Documentation/DocBook/writing-an-alsa-driver.tmpl b/Documentation/DocBook/writing-an-alsa-driver.tmpl deleted file mode 100644 index a27ab9f53fb6..000000000000 --- a/Documentation/DocBook/writing-an-alsa-driver.tmpl +++ /dev/null @@ -1,6206 +0,0 @@ - - - - - - - - - Writing an ALSA Driver - - Takashi - Iwai - -
- tiwai@suse.de -
- - - - Oct 15, 2007 - 0.3.7 - - - - This document describes how to write an ALSA (Advanced Linux - Sound Architecture) driver. - - - - - - Copyright (c) 2002-2005 Takashi Iwai tiwai@suse.de - - - - This document is free; you can redistribute it and/or modify it - under the terms of the GNU General Public License as published by - the Free Software Foundation; either version 2 of the License, or - (at your option) any later version. - - - - This document is distributed in the hope that it will be useful, - but WITHOUT ANY WARRANTY; without even the - implied warranty of MERCHANTABILITY or FITNESS FOR A - PARTICULAR PURPOSE. See the GNU General Public License - for more details. - - - - You should have received a copy of the GNU General Public - License along with this program; if not, write to the Free - Software Foundation, Inc., 59 Temple Place, Suite 330, Boston, - MA 02111-1307 USA - - - - - - - - - - Preface - - This document describes how to write an - - ALSA (Advanced Linux Sound Architecture) - driver. The document focuses mainly on PCI soundcards. - In the case of other device types, the API might - be different, too. However, at least the ALSA kernel API is - consistent, and therefore it would be still a bit help for - writing them. - - - - This document targets people who already have enough - C language skills and have basic linux kernel programming - knowledge. This document doesn't explain the general - topic of linux kernel coding and doesn't cover low-level - driver implementation details. It only describes - the standard way to write a PCI sound driver on ALSA. - - - - If you are already familiar with the older ALSA ver.0.5.x API, you - can check the drivers such as sound/pci/es1938.c or - sound/pci/maestro3.c which have also almost the same - code-base in the ALSA 0.5.x tree, so you can compare the differences. - - - - This document is still a draft version. Any feedback and - corrections, please!! - - - - - - - - - File Tree Structure - -
- General - - The ALSA drivers are provided in two ways. - - - - One is the trees provided as a tarball or via cvs from the - ALSA's ftp site, and another is the 2.6 (or later) Linux kernel - tree. To synchronize both, the ALSA driver tree is split into - two different trees: alsa-kernel and alsa-driver. The former - contains purely the source code for the Linux 2.6 (or later) - tree. This tree is designed only for compilation on 2.6 or - later environment. The latter, alsa-driver, contains many subtle - files for compiling ALSA drivers outside of the Linux kernel tree, - wrapper functions for older 2.2 and 2.4 kernels, to adapt the latest kernel API, - and additional drivers which are still in development or in - tests. The drivers in alsa-driver tree will be moved to - alsa-kernel (and eventually to the 2.6 kernel tree) when they are - finished and confirmed to work fine. - - - - The file tree structure of ALSA driver is depicted below. Both - alsa-kernel and alsa-driver have almost the same file - structure, except for core directory. It's - named as acore in alsa-driver tree. - - - ALSA File Tree Structure - - sound - /core - /oss - /seq - /oss - /instr - /ioctl32 - /include - /drivers - /mpu401 - /opl3 - /i2c - /l3 - /synth - /emux - /pci - /(cards) - /isa - /(cards) - /arm - /ppc - /sparc - /usb - /pcmcia /(cards) - /oss - - - -
- -
- core directory - - This directory contains the middle layer which is the heart - of ALSA drivers. In this directory, the native ALSA modules are - stored. The sub-directories contain different modules and are - dependent upon the kernel config. - - -
- core/oss - - - The codes for PCM and mixer OSS emulation modules are stored - in this directory. The rawmidi OSS emulation is included in - the ALSA rawmidi code since it's quite small. The sequencer - code is stored in core/seq/oss directory (see - - below). - -
- -
- core/ioctl32 - - - This directory contains the 32bit-ioctl wrappers for 64bit - architectures such like x86-64, ppc64 and sparc64. For 32bit - and alpha architectures, these are not compiled. - -
- -
- core/seq - - This directory and its sub-directories are for the ALSA - sequencer. This directory contains the sequencer core and - primary sequencer modules such like snd-seq-midi, - snd-seq-virmidi, etc. They are compiled only when - CONFIG_SND_SEQUENCER is set in the kernel - config. - -
- -
- core/seq/oss - - This contains the OSS sequencer emulation codes. - -
- -
- core/seq/instr - - This directory contains the modules for the sequencer - instrument layer. - -
-
- -
- include directory - - This is the place for the public header files of ALSA drivers, - which are to be exported to user-space, or included by - several files at different directories. Basically, the private - header files should not be placed in this directory, but you may - still find files there, due to historical reasons :) - -
- -
- drivers directory - - This directory contains code shared among different drivers - on different architectures. They are hence supposed not to be - architecture-specific. - For example, the dummy pcm driver and the serial MIDI - driver are found in this directory. In the sub-directories, - there is code for components which are independent from - bus and cpu architectures. - - -
- drivers/mpu401 - - The MPU401 and MPU401-UART modules are stored here. - -
- -
- drivers/opl3 and opl4 - - The OPL3 and OPL4 FM-synth stuff is found here. - -
-
- -
- i2c directory - - This contains the ALSA i2c components. - - - - Although there is a standard i2c layer on Linux, ALSA has its - own i2c code for some cards, because the soundcard needs only a - simple operation and the standard i2c API is too complicated for - such a purpose. - - -
- i2c/l3 - - This is a sub-directory for ARM L3 i2c. - -
-
- -
- synth directory - - This contains the synth middle-level modules. - - - - So far, there is only Emu8000/Emu10k1 synth driver under - the synth/emux sub-directory. - -
- -
- pci directory - - This directory and its sub-directories hold the top-level card modules - for PCI soundcards and the code specific to the PCI BUS. - - - - The drivers compiled from a single file are stored directly - in the pci directory, while the drivers with several source files are - stored on their own sub-directory (e.g. emu10k1, ice1712). - -
- -
- isa directory - - This directory and its sub-directories hold the top-level card modules - for ISA soundcards. - -
- -
- arm, ppc, and sparc directories - - They are used for top-level card modules which are - specific to one of these architectures. - -
- -
- usb directory - - This directory contains the USB-audio driver. In the latest version, the - USB MIDI driver is integrated in the usb-audio driver. - -
- -
- pcmcia directory - - The PCMCIA, especially PCCard drivers will go here. CardBus - drivers will be in the pci directory, because their API is identical - to that of standard PCI cards. - -
- -
- oss directory - - The OSS/Lite source files are stored here in Linux 2.6 (or - later) tree. In the ALSA driver tarball, this directory is empty, - of course :) - -
- - - - - - - - Basic Flow for PCI Drivers - -
- Outline - - The minimum flow for PCI soundcards is as follows: - - - define the PCI ID table (see the section - PCI Entries - ). - create probe() callback. - create remove() callback. - create a pci_driver structure - containing the three pointers above. - create an init() function just calling - the pci_register_driver() to register the pci_driver table - defined above. - create an exit() function to call - the pci_unregister_driver() function. - - -
- -
- Full Code Example - - The code example is shown below. Some parts are kept - unimplemented at this moment but will be filled in the - next sections. The numbers in the comment lines of the - snd_mychip_probe() function - refer to details explained in the following section. - - - Basic Flow for PCI Drivers - Example - - - #include - #include - #include - #include - - /* module parameters (see "Module Parameters") */ - /* SNDRV_CARDS: maximum number of cards supported by this module */ - static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; - static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; - static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; - - /* definition of the chip-specific record */ - struct mychip { - struct snd_card *card; - /* the rest of the implementation will be in section - * "PCI Resource Management" - */ - }; - - /* chip-specific destructor - * (see "PCI Resource Management") - */ - static int snd_mychip_free(struct mychip *chip) - { - .... /* will be implemented later... */ - } - - /* component-destructor - * (see "Management of Cards and Components") - */ - static int snd_mychip_dev_free(struct snd_device *device) - { - return snd_mychip_free(device->device_data); - } - - /* chip-specific constructor - * (see "Management of Cards and Components") - */ - static int snd_mychip_create(struct snd_card *card, - struct pci_dev *pci, - struct mychip **rchip) - { - struct mychip *chip; - int err; - static struct snd_device_ops ops = { - .dev_free = snd_mychip_dev_free, - }; - - *rchip = NULL; - - /* check PCI availability here - * (see "PCI Resource Management") - */ - .... - - /* allocate a chip-specific data with zero filled */ - chip = kzalloc(sizeof(*chip), GFP_KERNEL); - if (chip == NULL) - return -ENOMEM; - - chip->card = card; - - /* rest of initialization here; will be implemented - * later, see "PCI Resource Management" - */ - .... - - err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops); - if (err < 0) { - snd_mychip_free(chip); - return err; - } - - *rchip = chip; - return 0; - } - - /* constructor -- see "Constructor" sub-section */ - static int snd_mychip_probe(struct pci_dev *pci, - const struct pci_device_id *pci_id) - { - static int dev; - struct snd_card *card; - struct mychip *chip; - int err; - - /* (1) */ - if (dev >= SNDRV_CARDS) - return -ENODEV; - if (!enable[dev]) { - dev++; - return -ENOENT; - } - - /* (2) */ - err = snd_card_new(&pci->dev, index[dev], id[dev], THIS_MODULE, - 0, &card); - if (err < 0) - return err; - - /* (3) */ - err = snd_mychip_create(card, pci, &chip); - if (err < 0) { - snd_card_free(card); - return err; - } - - /* (4) */ - strcpy(card->driver, "My Chip"); - strcpy(card->shortname, "My Own Chip 123"); - sprintf(card->longname, "%s at 0x%lx irq %i", - card->shortname, chip->ioport, chip->irq); - - /* (5) */ - .... /* implemented later */ - - /* (6) */ - err = snd_card_register(card); - if (err < 0) { - snd_card_free(card); - return err; - } - - /* (7) */ - pci_set_drvdata(pci, card); - dev++; - return 0; - } - - /* destructor -- see the "Destructor" sub-section */ - static void snd_mychip_remove(struct pci_dev *pci) - { - snd_card_free(pci_get_drvdata(pci)); - pci_set_drvdata(pci, NULL); - } -]]> - - - -
- -
- Constructor - - The real constructor of PCI drivers is the probe callback. - The probe callback and other component-constructors which are called - from the probe callback cannot be used with - the __init prefix - because any PCI device could be a hotplug device. - - - - In the probe callback, the following scheme is often used. - - -
- 1) Check and increment the device index. - - - -= SNDRV_CARDS) - return -ENODEV; - if (!enable[dev]) { - dev++; - return -ENOENT; - } -]]> - - - - where enable[dev] is the module option. - - - - Each time the probe callback is called, check the - availability of the device. If not available, simply increment - the device index and returns. dev will be incremented also - later (step - 7). - -
- -
- 2) Create a card instance - - - -dev, index[dev], id[dev], THIS_MODULE, - 0, &card); -]]> - - - - - - The details will be explained in the section - - Management of Cards and Components. - -
- -
- 3) Create a main component - - In this part, the PCI resources are allocated. - - - - - - - - The details will be explained in the section PCI Resource - Management. - -
- -
- 4) Set the driver ID and name strings. - - - -driver, "My Chip"); - strcpy(card->shortname, "My Own Chip 123"); - sprintf(card->longname, "%s at 0x%lx irq %i", - card->shortname, chip->ioport, chip->irq); -]]> - - - - The driver field holds the minimal ID string of the - chip. This is used by alsa-lib's configurator, so keep it - simple but unique. - Even the same driver can have different driver IDs to - distinguish the functionality of each chip type. - - - - The shortname field is a string shown as more verbose - name. The longname field contains the information - shown in /proc/asound/cards. - -
- -
- 5) Create other components, such as mixer, MIDI, etc. - - Here you define the basic components such as - PCM, - mixer (e.g. AC97), - MIDI (e.g. MPU-401), - and other interfaces. - Also, if you want a proc - file, define it here, too. - -
- -
- 6) Register the card instance. - - - - - - - - - - Will be explained in the section Management - of Cards and Components, too. - -
- -
- 7) Set the PCI driver data and return zero. - - - - - - - - In the above, the card record is stored. This pointer is - used in the remove callback and power-management - callbacks, too. - -
-
- -
- Destructor - - The destructor, remove callback, simply releases the card - instance. Then the ALSA middle layer will release all the - attached components automatically. - - - - It would be typically like the following: - - - - - - - - The above code assumes that the card pointer is set to the PCI - driver data. - -
- -
- Header Files - - For the above example, at least the following include files - are necessary. - - - - - #include - #include - #include - #include -]]> - - - - where the last one is necessary only when module options are - defined in the source file. If the code is split into several - files, the files without module options don't need them. - - - - In addition to these headers, you'll need - <linux/interrupt.h> for interrupt - handling, and <asm/io.h> for I/O - access. If you use the mdelay() or - udelay() functions, you'll need to include - <linux/delay.h> too. - - - - The ALSA interfaces like the PCM and control APIs are defined in other - <sound/xxx.h> header files. - They have to be included after - <sound/core.h>. - - -
- - - - - - - - Management of Cards and Components - -
- Card Instance - - For each soundcard, a card record must be allocated. - - - - A card record is the headquarters of the soundcard. It manages - the whole list of devices (components) on the soundcard, such as - PCM, mixers, MIDI, synthesizer, and so on. Also, the card - record holds the ID and the name strings of the card, manages - the root of proc files, and controls the power-management states - and hotplug disconnections. The component list on the card - record is used to manage the correct release of resources at - destruction. - - - - As mentioned above, to create a card instance, call - snd_card_new(). - - - -dev, index, id, module, extra_size, &card); -]]> - - - - - - The function takes six arguments: the parent device pointer, - the card-index number, the id string, the module pointer (usually - THIS_MODULE), - the size of extra-data space, and the pointer to return the - card instance. The extra_size argument is used to - allocate card->private_data for the - chip-specific data. Note that these data - are allocated by snd_card_new(). - - - - The first argument, the pointer of struct - device, specifies the parent device. - For PCI devices, typically &pci-> is passed there. - -
- -
- Components - - After the card is created, you can attach the components - (devices) to the card instance. In an ALSA driver, a component is - represented as a struct snd_device object. - A component can be a PCM instance, a control interface, a raw - MIDI interface, etc. Each such instance has one component - entry. - - - - A component can be created via - snd_device_new() function. - - - - - - - - - - This takes the card pointer, the device-level - (SNDRV_DEV_XXX), the data pointer, and the - callback pointers (&ops). The - device-level defines the type of components and the order of - registration and de-registration. For most components, the - device-level is already defined. For a user-defined component, - you can use SNDRV_DEV_LOWLEVEL. - - - - This function itself doesn't allocate the data space. The data - must be allocated manually beforehand, and its pointer is passed - as the argument. This pointer (chip in the - above example) is used as the identifier for the instance. - - - - Each pre-defined ALSA component such as ac97 and pcm calls - snd_device_new() inside its - constructor. The destructor for each component is defined in the - callback pointers. Hence, you don't need to take care of - calling a destructor for such a component. - - - - If you wish to create your own component, you need to - set the destructor function to the dev_free callback in - the ops, so that it can be released - automatically via snd_card_free(). - The next example will show an implementation of chip-specific - data. - -
- -
- Chip-Specific Data - - Chip-specific information, e.g. the I/O port address, its - resource pointer, or the irq number, is stored in the - chip-specific record. - - - - - - - - - - In general, there are two ways of allocating the chip record. - - -
- 1. Allocating via <function>snd_card_new()</function>. - - As mentioned above, you can pass the extra-data-length - to the 5th argument of snd_card_new(), i.e. - - - -dev, index[dev], id[dev], THIS_MODULE, - sizeof(struct mychip), &card); -]]> - - - - struct mychip is the type of the chip record. - - - - In return, the allocated record can be accessed as - - - -private_data; -]]> - - - - With this method, you don't have to allocate twice. - The record is released together with the card instance. - -
- -
- 2. Allocating an extra device. - - - After allocating a card instance via - snd_card_new() (with - 0 on the 4th arg), call - kzalloc(). - - - -dev, index[dev], id[dev], THIS_MODULE, - 0, &card); - ..... - chip = kzalloc(sizeof(*chip), GFP_KERNEL); -]]> - - - - - - The chip record should have the field to hold the card - pointer at least, - - - - - - - - - - Then, set the card pointer in the returned chip instance. - - - -card = card; -]]> - - - - - - Next, initialize the fields, and register this chip - record as a low-level device with a specified - ops, - - - - - - - - snd_mychip_dev_free() is the - device-destructor function, which will call the real - destructor. - - - - - -device_data); - } -]]> - - - - where snd_mychip_free() is the real destructor. - -
-
- -
- Registration and Release - - After all components are assigned, register the card instance - by calling snd_card_register(). Access - to the device files is enabled at this point. That is, before - snd_card_register() is called, the - components are safely inaccessible from external side. If this - call fails, exit the probe function after releasing the card via - snd_card_free(). - - - - For releasing the card instance, you can call simply - snd_card_free(). As mentioned earlier, all - components are released automatically by this call. - - - - For a device which allows hotplugging, you can use - snd_card_free_when_closed. This one will - postpone the destruction until all devices are closed. - - -
- - - - - - - - - PCI Resource Management - -
- Full Code Example - - In this section, we'll complete the chip-specific constructor, - destructor and PCI entries. Example code is shown first, - below. - - - PCI Resource Management Example - -irq >= 0) - free_irq(chip->irq, chip); - /* release the I/O ports & memory */ - pci_release_regions(chip->pci); - /* disable the PCI entry */ - pci_disable_device(chip->pci); - /* release the data */ - kfree(chip); - return 0; - } - - /* chip-specific constructor */ - static int snd_mychip_create(struct snd_card *card, - struct pci_dev *pci, - struct mychip **rchip) - { - struct mychip *chip; - int err; - static struct snd_device_ops ops = { - .dev_free = snd_mychip_dev_free, - }; - - *rchip = NULL; - - /* initialize the PCI entry */ - err = pci_enable_device(pci); - if (err < 0) - return err; - /* check PCI availability (28bit DMA) */ - if (pci_set_dma_mask(pci, DMA_BIT_MASK(28)) < 0 || - pci_set_consistent_dma_mask(pci, DMA_BIT_MASK(28)) < 0) { - printk(KERN_ERR "error to set 28bit mask DMA\n"); - pci_disable_device(pci); - return -ENXIO; - } - - chip = kzalloc(sizeof(*chip), GFP_KERNEL); - if (chip == NULL) { - pci_disable_device(pci); - return -ENOMEM; - } - - /* initialize the stuff */ - chip->card = card; - chip->pci = pci; - chip->irq = -1; - - /* (1) PCI resource allocation */ - err = pci_request_regions(pci, "My Chip"); - if (err < 0) { - kfree(chip); - pci_disable_device(pci); - return err; - } - chip->port = pci_resource_start(pci, 0); - if (request_irq(pci->irq, snd_mychip_interrupt, - IRQF_SHARED, KBUILD_MODNAME, chip)) { - printk(KERN_ERR "cannot grab irq %d\n", pci->irq); - snd_mychip_free(chip); - return -EBUSY; - } - chip->irq = pci->irq; - - /* (2) initialization of the chip hardware */ - .... /* (not implemented in this document) */ - - err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops); - if (err < 0) { - snd_mychip_free(chip); - return err; - } - - *rchip = chip; - return 0; - } - - /* PCI IDs */ - static struct pci_device_id snd_mychip_ids[] = { - { PCI_VENDOR_ID_FOO, PCI_DEVICE_ID_BAR, - PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, }, - .... - { 0, } - }; - MODULE_DEVICE_TABLE(pci, snd_mychip_ids); - - /* pci_driver definition */ - static struct pci_driver driver = { - .name = KBUILD_MODNAME, - .id_table = snd_mychip_ids, - .probe = snd_mychip_probe, - .remove = snd_mychip_remove, - }; - - /* module initialization */ - static int __init alsa_card_mychip_init(void) - { - return pci_register_driver(&driver); - } - - /* module clean up */ - static void __exit alsa_card_mychip_exit(void) - { - pci_unregister_driver(&driver); - } - - module_init(alsa_card_mychip_init) - module_exit(alsa_card_mychip_exit) - - EXPORT_NO_SYMBOLS; /* for old kernels only */ -]]> - - - -
- -
- Some Hafta's - - The allocation of PCI resources is done in the - probe() function, and usually an extra - xxx_create() function is written for this - purpose. - - - - In the case of PCI devices, you first have to call - the pci_enable_device() function before - allocating resources. Also, you need to set the proper PCI DMA - mask to limit the accessed I/O range. In some cases, you might - need to call pci_set_master() function, - too. - - - - Suppose the 28bit mask, and the code to be added would be like: - - - - - - - -
- -
- Resource Allocation - - The allocation of I/O ports and irqs is done via standard kernel - functions. Unlike ALSA ver.0.5.x., there are no helpers for - that. And these resources must be released in the destructor - function (see below). Also, on ALSA 0.9.x, you don't need to - allocate (pseudo-)DMA for PCI like in ALSA 0.5.x. - - - - Now assume that the PCI device has an I/O port with 8 bytes - and an interrupt. Then struct mychip will have the - following fields: - - - - - - - - - - For an I/O port (and also a memory region), you need to have - the resource pointer for the standard resource management. For - an irq, you have to keep only the irq number (integer). But you - need to initialize this number as -1 before actual allocation, - since irq 0 is valid. The port address and its resource pointer - can be initialized as null by - kzalloc() automatically, so you - don't have to take care of resetting them. - - - - The allocation of an I/O port is done like this: - - - -port = pci_resource_start(pci, 0); -]]> - - - - - - - It will reserve the I/O port region of 8 bytes of the given - PCI device. The returned value, chip->res_port, is allocated - via kmalloc() by - request_region(). The pointer must be - released via kfree(), but there is a - problem with this. This issue will be explained later. - - - - The allocation of an interrupt source is done like this: - - - -irq, snd_mychip_interrupt, - IRQF_SHARED, KBUILD_MODNAME, chip)) { - printk(KERN_ERR "cannot grab irq %d\n", pci->irq); - snd_mychip_free(chip); - return -EBUSY; - } - chip->irq = pci->irq; -]]> - - - - where snd_mychip_interrupt() is the - interrupt handler defined later. - Note that chip->irq should be defined - only when request_irq() succeeded. - - - - On the PCI bus, interrupts can be shared. Thus, - IRQF_SHARED is used as the interrupt flag of - request_irq(). - - - - The last argument of request_irq() is the - data pointer passed to the interrupt handler. Usually, the - chip-specific record is used for that, but you can use what you - like, too. - - - - I won't give details about the interrupt handler at this - point, but at least its appearance can be explained now. The - interrupt handler looks usually like the following: - - - - - - - - - - Now let's write the corresponding destructor for the resources - above. The role of destructor is simple: disable the hardware - (if already activated) and release the resources. So far, we - have no hardware part, so the disabling code is not written here. - - - - To release the resources, the check-and-release - method is a safer way. For the interrupt, do like this: - - - -irq >= 0) - free_irq(chip->irq, chip); -]]> - - - - Since the irq number can start from 0, you should initialize - chip->irq with a negative value (e.g. -1), so that you can - check the validity of the irq number as above. - - - - When you requested I/O ports or memory regions via - pci_request_region() or - pci_request_regions() like in this example, - release the resource(s) using the corresponding function, - pci_release_region() or - pci_release_regions(). - - - -pci); -]]> - - - - - - When you requested manually via request_region() - or request_mem_region, you can release it via - release_resource(). Suppose that you keep - the resource pointer returned from request_region() - in chip->res_port, the release procedure looks like: - - - -res_port); -]]> - - - - - - Don't forget to call pci_disable_device() - before the end. - - - - And finally, release the chip-specific record. - - - - - - - - - - We didn't implement the hardware disabling part in the above. - If you need to do this, please note that the destructor may be - called even before the initialization of the chip is completed. - It would be better to have a flag to skip hardware disabling - if the hardware was not initialized yet. - - - - When the chip-data is assigned to the card using - snd_device_new() with - SNDRV_DEV_LOWLELVEL , its destructor is - called at the last. That is, it is assured that all other - components like PCMs and controls have already been released. - You don't have to stop PCMs, etc. explicitly, but just - call low-level hardware stopping. - - - - The management of a memory-mapped region is almost as same as - the management of an I/O port. You'll need three fields like - the following: - - - - - - - - and the allocation would be like below: - - - -iobase_phys = pci_resource_start(pci, 0); - chip->iobase_virt = ioremap_nocache(chip->iobase_phys, - pci_resource_len(pci, 0)); -]]> - - - - and the corresponding destructor would be: - - - -iobase_virt) - iounmap(chip->iobase_virt); - .... - pci_release_regions(chip->pci); - .... - } -]]> - - - - -
- -
- PCI Entries - - So far, so good. Let's finish the missing PCI - stuff. At first, we need a - pci_device_id table for this - chipset. It's a table of PCI vendor/device ID number, and some - masks. - - - - For example, - - - - - - - - - - The first and second fields of - the pci_device_id structure are the vendor and - device IDs. If you have no reason to filter the matching - devices, you can leave the remaining fields as above. The last - field of the pci_device_id struct contains - private data for this entry. You can specify any value here, for - example, to define specific operations for supported device IDs. - Such an example is found in the intel8x0 driver. - - - - The last entry of this list is the terminator. You must - specify this all-zero entry. - - - - Then, prepare the pci_driver record: - - - - - - - - - - The probe and - remove functions have already - been defined in the previous sections. - The name - field is the name string of this device. Note that you must not - use a slash / in this string. - - - - And at last, the module entries: - - - - - - - - - - Note that these module entries are tagged with - __init and - __exit prefixes. - - - - Oh, one thing was forgotten. If you have no exported symbols, - you need to declare it in 2.2 or 2.4 kernels (it's not necessary in 2.6 kernels). - - - - - - - - That's all! - -
- - - - - - - - PCM Interface - -
- General - - The PCM middle layer of ALSA is quite powerful and it is only - necessary for each driver to implement the low-level functions - to access its hardware. - - - - For accessing to the PCM layer, you need to include - <sound/pcm.h> first. In addition, - <sound/pcm_params.h> might be needed - if you access to some functions related with hw_param. - - - - Each card device can have up to four pcm instances. A pcm - instance corresponds to a pcm device file. The limitation of - number of instances comes only from the available bit size of - the Linux's device numbers. Once when 64bit device number is - used, we'll have more pcm instances available. - - - - A pcm instance consists of pcm playback and capture streams, - and each pcm stream consists of one or more pcm substreams. Some - soundcards support multiple playback functions. For example, - emu10k1 has a PCM playback of 32 stereo substreams. In this case, at - each open, a free substream is (usually) automatically chosen - and opened. Meanwhile, when only one substream exists and it was - already opened, the successful open will either block - or error with EAGAIN according to the - file open mode. But you don't have to care about such details in your - driver. The PCM middle layer will take care of such work. - -
- -
- Full Code Example - - The example code below does not include any hardware access - routines but shows only the skeleton, how to build up the PCM - interfaces. - - - PCM Example Code - - - .... - - /* hardware definition */ - static struct snd_pcm_hardware snd_mychip_playback_hw = { - .info = (SNDRV_PCM_INFO_MMAP | - SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_BLOCK_TRANSFER | - SNDRV_PCM_INFO_MMAP_VALID), - .formats = SNDRV_PCM_FMTBIT_S16_LE, - .rates = SNDRV_PCM_RATE_8000_48000, - .rate_min = 8000, - .rate_max = 48000, - .channels_min = 2, - .channels_max = 2, - .buffer_bytes_max = 32768, - .period_bytes_min = 4096, - .period_bytes_max = 32768, - .periods_min = 1, - .periods_max = 1024, - }; - - /* hardware definition */ - static struct snd_pcm_hardware snd_mychip_capture_hw = { - .info = (SNDRV_PCM_INFO_MMAP | - SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_BLOCK_TRANSFER | - SNDRV_PCM_INFO_MMAP_VALID), - .formats = SNDRV_PCM_FMTBIT_S16_LE, - .rates = SNDRV_PCM_RATE_8000_48000, - .rate_min = 8000, - .rate_max = 48000, - .channels_min = 2, - .channels_max = 2, - .buffer_bytes_max = 32768, - .period_bytes_min = 4096, - .period_bytes_max = 32768, - .periods_min = 1, - .periods_max = 1024, - }; - - /* open callback */ - static int snd_mychip_playback_open(struct snd_pcm_substream *substream) - { - struct mychip *chip = snd_pcm_substream_chip(substream); - struct snd_pcm_runtime *runtime = substream->runtime; - - runtime->hw = snd_mychip_playback_hw; - /* more hardware-initialization will be done here */ - .... - return 0; - } - - /* close callback */ - static int snd_mychip_playback_close(struct snd_pcm_substream *substream) - { - struct mychip *chip = snd_pcm_substream_chip(substream); - /* the hardware-specific codes will be here */ - .... - return 0; - - } - - /* open callback */ - static int snd_mychip_capture_open(struct snd_pcm_substream *substream) - { - struct mychip *chip = snd_pcm_substream_chip(substream); - struct snd_pcm_runtime *runtime = substream->runtime; - - runtime->hw = snd_mychip_capture_hw; - /* more hardware-initialization will be done here */ - .... - return 0; - } - - /* close callback */ - static int snd_mychip_capture_close(struct snd_pcm_substream *substream) - { - struct mychip *chip = snd_pcm_substream_chip(substream); - /* the hardware-specific codes will be here */ - .... - return 0; - - } - - /* hw_params callback */ - static int snd_mychip_pcm_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *hw_params) - { - return snd_pcm_lib_malloc_pages(substream, - params_buffer_bytes(hw_params)); - } - - /* hw_free callback */ - static int snd_mychip_pcm_hw_free(struct snd_pcm_substream *substream) - { - return snd_pcm_lib_free_pages(substream); - } - - /* prepare callback */ - static int snd_mychip_pcm_prepare(struct snd_pcm_substream *substream) - { - struct mychip *chip = snd_pcm_substream_chip(substream); - struct snd_pcm_runtime *runtime = substream->runtime; - - /* set up the hardware with the current configuration - * for example... - */ - mychip_set_sample_format(chip, runtime->format); - mychip_set_sample_rate(chip, runtime->rate); - mychip_set_channels(chip, runtime->channels); - mychip_set_dma_setup(chip, runtime->dma_addr, - chip->buffer_size, - chip->period_size); - return 0; - } - - /* trigger callback */ - static int snd_mychip_pcm_trigger(struct snd_pcm_substream *substream, - int cmd) - { - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - /* do something to start the PCM engine */ - .... - break; - case SNDRV_PCM_TRIGGER_STOP: - /* do something to stop the PCM engine */ - .... - break; - default: - return -EINVAL; - } - } - - /* pointer callback */ - static snd_pcm_uframes_t - snd_mychip_pcm_pointer(struct snd_pcm_substream *substream) - { - struct mychip *chip = snd_pcm_substream_chip(substream); - unsigned int current_ptr; - - /* get the current hardware pointer */ - current_ptr = mychip_get_hw_pointer(chip); - return current_ptr; - } - - /* operators */ - static struct snd_pcm_ops snd_mychip_playback_ops = { - .open = snd_mychip_playback_open, - .close = snd_mychip_playback_close, - .ioctl = snd_pcm_lib_ioctl, - .hw_params = snd_mychip_pcm_hw_params, - .hw_free = snd_mychip_pcm_hw_free, - .prepare = snd_mychip_pcm_prepare, - .trigger = snd_mychip_pcm_trigger, - .pointer = snd_mychip_pcm_pointer, - }; - - /* operators */ - static struct snd_pcm_ops snd_mychip_capture_ops = { - .open = snd_mychip_capture_open, - .close = snd_mychip_capture_close, - .ioctl = snd_pcm_lib_ioctl, - .hw_params = snd_mychip_pcm_hw_params, - .hw_free = snd_mychip_pcm_hw_free, - .prepare = snd_mychip_pcm_prepare, - .trigger = snd_mychip_pcm_trigger, - .pointer = snd_mychip_pcm_pointer, - }; - - /* - * definitions of capture are omitted here... - */ - - /* create a pcm device */ - static int snd_mychip_new_pcm(struct mychip *chip) - { - struct snd_pcm *pcm; - int err; - - err = snd_pcm_new(chip->card, "My Chip", 0, 1, 1, &pcm); - if (err < 0) - return err; - pcm->private_data = chip; - strcpy(pcm->name, "My Chip"); - chip->pcm = pcm; - /* set operators */ - snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, - &snd_mychip_playback_ops); - snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, - &snd_mychip_capture_ops); - /* pre-allocation of buffers */ - /* NOTE: this may fail */ - snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, - snd_dma_pci_data(chip->pci), - 64*1024, 64*1024); - return 0; - } -]]> - - - -
- -
- Constructor - - A pcm instance is allocated by the snd_pcm_new() - function. It would be better to create a constructor for pcm, - namely, - - - -card, "My Chip", 0, 1, 1, &pcm); - if (err < 0) - return err; - pcm->private_data = chip; - strcpy(pcm->name, "My Chip"); - chip->pcm = pcm; - .... - return 0; - } -]]> - - - - - - The snd_pcm_new() function takes four - arguments. The first argument is the card pointer to which this - pcm is assigned, and the second is the ID string. - - - - The third argument (index, 0 in the - above) is the index of this new pcm. It begins from zero. If - you create more than one pcm instances, specify the - different numbers in this argument. For example, - index = 1 for the second PCM device. - - - - The fourth and fifth arguments are the number of substreams - for playback and capture, respectively. Here 1 is used for - both arguments. When no playback or capture substreams are available, - pass 0 to the corresponding argument. - - - - If a chip supports multiple playbacks or captures, you can - specify more numbers, but they must be handled properly in - open/close, etc. callbacks. When you need to know which - substream you are referring to, then it can be obtained from - struct snd_pcm_substream data passed to each callback - as follows: - - - -number; -]]> - - - - - - After the pcm is created, you need to set operators for each - pcm stream. - - - - - - - - - - The operators are defined typically like this: - - - - - - - - All the callbacks are described in the - - Operators subsection. - - - - After setting the operators, you probably will want to - pre-allocate the buffer. For the pre-allocation, simply call - the following: - - - -pci), - 64*1024, 64*1024); -]]> - - - - It will allocate a buffer up to 64kB as default. - Buffer management details will be described in the later section Buffer and Memory - Management. - - - - Additionally, you can set some extra information for this pcm - in pcm->info_flags. - The available values are defined as - SNDRV_PCM_INFO_XXX in - <sound/asound.h>, which is used for - the hardware definition (described later). When your soundchip - supports only half-duplex, specify like this: - - - -info_flags = SNDRV_PCM_INFO_HALF_DUPLEX; -]]> - - - -
- -
- ... And the Destructor? - - The destructor for a pcm instance is not always - necessary. Since the pcm device will be released by the middle - layer code automatically, you don't have to call the destructor - explicitly. - - - - The destructor would be necessary if you created - special records internally and needed to release them. In such a - case, set the destructor function to - pcm->private_free: - - - PCM Instance with a Destructor - -my_private_pcm_data); - /* do what you like else */ - .... - } - - static int snd_mychip_new_pcm(struct mychip *chip) - { - struct snd_pcm *pcm; - .... - /* allocate your own data */ - chip->my_private_pcm_data = kmalloc(...); - /* set the destructor */ - pcm->private_data = chip; - pcm->private_free = mychip_pcm_free; - .... - } -]]> - - - -
- -
- Runtime Pointer - The Chest of PCM Information - - When the PCM substream is opened, a PCM runtime instance is - allocated and assigned to the substream. This pointer is - accessible via substream->runtime. - This runtime pointer holds most information you need - to control the PCM: the copy of hw_params and sw_params configurations, the buffer - pointers, mmap records, spinlocks, etc. - - - - The definition of runtime instance is found in - <sound/pcm.h>. Here are - the contents of this file: - - - - - - - - - For the operators (callbacks) of each sound driver, most of - these records are supposed to be read-only. Only the PCM - middle-layer changes / updates them. The exceptions are - the hardware description (hw) DMA buffer information and the - private data. Besides, if you use the standard buffer allocation - method via snd_pcm_lib_malloc_pages(), - you don't need to set the DMA buffer information by yourself. - - - - In the sections below, important records are explained. - - -
- Hardware Description - - The hardware descriptor (struct snd_pcm_hardware) - contains the definitions of the fundamental hardware - configuration. Above all, you'll need to define this in - - the open callback. - Note that the runtime instance holds the copy of the - descriptor, not the pointer to the existing descriptor. That - is, in the open callback, you can modify the copied descriptor - (runtime->hw) as you need. For example, if the maximum - number of channels is 1 only on some chip models, you can - still use the same hardware descriptor and change the - channels_max later: - - -runtime; - ... - runtime->hw = snd_mychip_playback_hw; /* common definition */ - if (chip->model == VERY_OLD_ONE) - runtime->hw.channels_max = 1; -]]> - - - - - - Typically, you'll have a hardware descriptor as below: - - - - - - - - - - - The info field contains the type and - capabilities of this pcm. The bit flags are defined in - <sound/asound.h> as - SNDRV_PCM_INFO_XXX. Here, at least, you - have to specify whether the mmap is supported and which - interleaved format is supported. - When the hardware supports mmap, add the - SNDRV_PCM_INFO_MMAP flag here. When the - hardware supports the interleaved or the non-interleaved - formats, SNDRV_PCM_INFO_INTERLEAVED or - SNDRV_PCM_INFO_NONINTERLEAVED flag must - be set, respectively. If both are supported, you can set both, - too. - - - - In the above example, MMAP_VALID and - BLOCK_TRANSFER are specified for the OSS mmap - mode. Usually both are set. Of course, - MMAP_VALID is set only if the mmap is - really supported. - - - - The other possible flags are - SNDRV_PCM_INFO_PAUSE and - SNDRV_PCM_INFO_RESUME. The - PAUSE bit means that the pcm supports the - pause operation, while the - RESUME bit means that the pcm supports - the full suspend/resume operation. - If the PAUSE flag is set, - the trigger callback below - must handle the corresponding (pause push/release) commands. - The suspend/resume trigger commands can be defined even without - the RESUME flag. See - Power Management section for details. - - - - When the PCM substreams can be synchronized (typically, - synchronized start/stop of a playback and a capture streams), - you can give SNDRV_PCM_INFO_SYNC_START, - too. In this case, you'll need to check the linked-list of - PCM substreams in the trigger callback. This will be - described in the later section. - - - - - - formats field contains the bit-flags - of supported formats (SNDRV_PCM_FMTBIT_XXX). - If the hardware supports more than one format, give all or'ed - bits. In the example above, the signed 16bit little-endian - format is specified. - - - - - - rates field contains the bit-flags of - supported rates (SNDRV_PCM_RATE_XXX). - When the chip supports continuous rates, pass - CONTINUOUS bit additionally. - The pre-defined rate bits are provided only for typical - rates. If your chip supports unconventional rates, you need to add - the KNOT bit and set up the hardware - constraint manually (explained later). - - - - - - rate_min and - rate_max define the minimum and - maximum sample rate. This should correspond somehow to - rates bits. - - - - - - channel_min and - channel_max - define, as you might already expected, the minimum and maximum - number of channels. - - - - - - buffer_bytes_max defines the - maximum buffer size in bytes. There is no - buffer_bytes_min field, since - it can be calculated from the minimum period size and the - minimum number of periods. - Meanwhile, period_bytes_min and - define the minimum and maximum size of the period in bytes. - periods_max and - periods_min define the maximum and - minimum number of periods in the buffer. - - - - The period is a term that corresponds to - a fragment in the OSS world. The period defines the size at - which a PCM interrupt is generated. This size strongly - depends on the hardware. - Generally, the smaller period size will give you more - interrupts, that is, more controls. - In the case of capture, this size defines the input latency. - On the other hand, the whole buffer size defines the - output latency for the playback direction. - - - - - - There is also a field fifo_size. - This specifies the size of the hardware FIFO, but currently it - is neither used in the driver nor in the alsa-lib. So, you - can ignore this field. - - - - -
- -
- PCM Configurations - - Ok, let's go back again to the PCM runtime records. - The most frequently referred records in the runtime instance are - the PCM configurations. - The PCM configurations are stored in the runtime instance - after the application sends hw_params data via - alsa-lib. There are many fields copied from hw_params and - sw_params structs. For example, - format holds the format type - chosen by the application. This field contains the enum value - SNDRV_PCM_FORMAT_XXX. - - - - One thing to be noted is that the configured buffer and period - sizes are stored in frames in the runtime. - In the ALSA world, 1 frame = channels * samples-size. - For conversion between frames and bytes, you can use the - frames_to_bytes() and - bytes_to_frames() helper functions. - - -period_size); -]]> - - - - - - Also, many software parameters (sw_params) are - stored in frames, too. Please check the type of the field. - snd_pcm_uframes_t is for the frames as unsigned - integer while snd_pcm_sframes_t is for the frames - as signed integer. - -
- -
- DMA Buffer Information - - The DMA buffer is defined by the following four fields, - dma_area, - dma_addr, - dma_bytes and - dma_private. - The dma_area holds the buffer - pointer (the logical address). You can call - memcpy from/to - this pointer. Meanwhile, dma_addr - holds the physical address of the buffer. This field is - specified only when the buffer is a linear buffer. - dma_bytes holds the size of buffer - in bytes. dma_private is used for - the ALSA DMA allocator. - - - - If you use a standard ALSA function, - snd_pcm_lib_malloc_pages(), for - allocating the buffer, these fields are set by the ALSA middle - layer, and you should not change them by - yourself. You can read them but not write them. - On the other hand, if you want to allocate the buffer by - yourself, you'll need to manage it in hw_params callback. - At least, dma_bytes is mandatory. - dma_area is necessary when the - buffer is mmapped. If your driver doesn't support mmap, this - field is not necessary. dma_addr - is also optional. You can use - dma_private as you like, too. - -
- -
- Running Status - - The running status can be referred via runtime->status. - This is the pointer to the struct snd_pcm_mmap_status - record. For example, you can get the current DMA hardware - pointer via runtime->status->hw_ptr. - - - - The DMA application pointer can be referred via - runtime->control, which points to the - struct snd_pcm_mmap_control record. - However, accessing directly to this value is not recommended. - -
- -
- Private Data - - You can allocate a record for the substream and store it in - runtime->private_data. Usually, this - is done in - - the open callback. - Don't mix this with pcm->private_data. - The pcm->private_data usually points to the - chip instance assigned statically at the creation of PCM, while the - runtime->private_data points to a dynamic - data structure created at the PCM open callback. - - - -runtime->private_data = data; - .... - } -]]> - - - - - - The allocated object must be released in - - the close callback. - -
- -
- -
- Operators - - OK, now let me give details about each pcm callback - (ops). In general, every callback must - return 0 if successful, or a negative error number - such as -EINVAL. To choose an appropriate - error number, it is advised to check what value other parts of - the kernel return when the same kind of request fails. - - - - The callback function takes at least the argument with - snd_pcm_substream pointer. To retrieve - the chip record from the given substream instance, you can use the - following macro. - - - - - - - - The macro reads substream->private_data, - which is a copy of pcm->private_data. - You can override the former if you need to assign different data - records per PCM substream. For example, the cmi8330 driver assigns - different private_data for playback and capture directions, - because it uses two different codecs (SB- and AD-compatible) for - different directions. - - -
- open callback - - - - - - - - This is called when a pcm substream is opened. - - - - At least, here you have to initialize the runtime->hw - record. Typically, this is done by like this: - - - -runtime; - - runtime->hw = snd_mychip_playback_hw; - return 0; - } -]]> - - - - where snd_mychip_playback_hw is the - pre-defined hardware description. - - - - You can allocate a private data in this callback, as described - in - Private Data section. - - - - If the hardware configuration needs more constraints, set the - hardware constraints here, too. - See - Constraints for more details. - -
- -
- close callback - - - - - - - - Obviously, this is called when a pcm substream is closed. - - - - Any private instance for a pcm substream allocated in the - open callback will be released here. - - - -runtime->private_data); - .... - } -]]> - - - -
- -
- ioctl callback - - This is used for any special call to pcm ioctls. But - usually you can pass a generic ioctl callback, - snd_pcm_lib_ioctl. - -
- -
- hw_params callback - - - - - - - - - - This is called when the hardware parameter - (hw_params) is set - up by the application, - that is, once when the buffer size, the period size, the - format, etc. are defined for the pcm substream. - - - - Many hardware setups should be done in this callback, - including the allocation of buffers. - - - - Parameters to be initialized are retrieved by - params_xxx() macros. To allocate - buffer, you can call a helper function, - - - - - - - - snd_pcm_lib_malloc_pages() is available - only when the DMA buffers have been pre-allocated. - See the section - Buffer Types for more details. - - - - Note that this and prepare callbacks - may be called multiple times per initialization. - For example, the OSS emulation may - call these callbacks at each change via its ioctl. - - - - Thus, you need to be careful not to allocate the same buffers - many times, which will lead to memory leaks! Calling the - helper function above many times is OK. It will release the - previous buffer automatically when it was already allocated. - - - - Another note is that this callback is non-atomic - (schedulable) as default, i.e. when no - nonatomic flag set. - This is important, because the - trigger callback - is atomic (non-schedulable). That is, mutexes or any - schedule-related functions are not available in - trigger callback. - Please see the subsection - - Atomicity for details. - -
- -
- hw_free callback - - - - - - - - - - This is called to release the resources allocated via - hw_params. For example, releasing the - buffer via - snd_pcm_lib_malloc_pages() is done by - calling the following: - - - - - - - - - - This function is always called before the close callback is called. - Also, the callback may be called multiple times, too. - Keep track whether the resource was already released. - -
- -
- prepare callback - - - - - - - - - - This callback is called when the pcm is - prepared. You can set the format type, sample - rate, etc. here. The difference from - hw_params is that the - prepare callback will be called each - time - snd_pcm_prepare() is called, i.e. when - recovering after underruns, etc. - - - - Note that this callback is now non-atomic. - You can use schedule-related functions safely in this callback. - - - - In this and the following callbacks, you can refer to the - values via the runtime record, - substream->runtime. - For example, to get the current - rate, format or channels, access to - runtime->rate, - runtime->format or - runtime->channels, respectively. - The physical address of the allocated buffer is set to - runtime->dma_area. The buffer and period sizes are - in runtime->buffer_size and runtime->period_size, - respectively. - - - - Be careful that this callback will be called many times at - each setup, too. - -
- -
- trigger callback - - - - - - - - This is called when the pcm is started, stopped or paused. - - - - Which action is specified in the second argument, - SNDRV_PCM_TRIGGER_XXX in - <sound/pcm.h>. At least, - the START and STOP - commands must be defined in this callback. - - - - - - - - - - When the pcm supports the pause operation (given in the info - field of the hardware table), the PAUSE_PUSH - and PAUSE_RELEASE commands must be - handled here, too. The former is the command to pause the pcm, - and the latter to restart the pcm again. - - - - When the pcm supports the suspend/resume operation, - regardless of full or partial suspend/resume support, - the SUSPEND and RESUME - commands must be handled, too. - These commands are issued when the power-management status is - changed. Obviously, the SUSPEND and - RESUME commands - suspend and resume the pcm substream, and usually, they - are identical to the STOP and - START commands, respectively. - See the - Power Management section for details. - - - - As mentioned, this callback is atomic as default unless - nonatomic flag set, and - you cannot call functions which may sleep. - The trigger callback should be as minimal as possible, - just really triggering the DMA. The other stuff should be - initialized hw_params and prepare callbacks properly - beforehand. - -
- -
- pointer callback - - - - - - - - This callback is called when the PCM middle layer inquires - the current hardware position on the buffer. The position must - be returned in frames, - ranging from 0 to buffer_size - 1. - - - - This is called usually from the buffer-update routine in the - pcm middle layer, which is invoked when - snd_pcm_period_elapsed() is called in the - interrupt routine. Then the pcm middle layer updates the - position and calculates the available space, and wakes up the - sleeping poll threads, etc. - - - - This callback is also atomic as default. - -
- -
- copy and silence callbacks - - These callbacks are not mandatory, and can be omitted in - most cases. These callbacks are used when the hardware buffer - cannot be in the normal memory space. Some chips have their - own buffer on the hardware which is not mappable. In such a - case, you have to transfer the data manually from the memory - buffer to the hardware buffer. Or, if the buffer is - non-contiguous on both physical and virtual memory spaces, - these callbacks must be defined, too. - - - - If these two callbacks are defined, copy and set-silence - operations are done by them. The detailed will be described in - the later section Buffer and Memory - Management. - -
- -
- ack callback - - This callback is also not mandatory. This callback is called - when the appl_ptr is updated in read or write operations. - Some drivers like emu10k1-fx and cs46xx need to track the - current appl_ptr for the internal buffer, and this callback - is useful only for such a purpose. - - - This callback is atomic as default. - -
- -
- page callback - - - This callback is optional too. This callback is used - mainly for non-contiguous buffers. The mmap calls this - callback to get the page address. Some examples will be - explained in the later section Buffer and Memory - Management, too. - -
-
- -
- Interrupt Handler - - The rest of pcm stuff is the PCM interrupt handler. The - role of PCM interrupt handler in the sound driver is to update - the buffer position and to tell the PCM middle layer when the - buffer position goes across the prescribed period size. To - inform this, call the snd_pcm_period_elapsed() - function. - - - - There are several types of sound chips to generate the interrupts. - - -
- Interrupts at the period (fragment) boundary - - This is the most frequently found type: the hardware - generates an interrupt at each period boundary. - In this case, you can call - snd_pcm_period_elapsed() at each - interrupt. - - - - snd_pcm_period_elapsed() takes the - substream pointer as its argument. Thus, you need to keep the - substream pointer accessible from the chip instance. For - example, define substream field in the chip record to hold the - current running substream pointer, and set the pointer value - at open callback (and reset at close callback). - - - - If you acquire a spinlock in the interrupt handler, and the - lock is used in other pcm callbacks, too, then you have to - release the lock before calling - snd_pcm_period_elapsed(), because - snd_pcm_period_elapsed() calls other pcm - callbacks inside. - - - - Typical code would be like: - - - Interrupt Handler Case #1 - -lock); - .... - if (pcm_irq_invoked(chip)) { - /* call updater, unlock before it */ - spin_unlock(&chip->lock); - snd_pcm_period_elapsed(chip->substream); - spin_lock(&chip->lock); - /* acknowledge the interrupt if necessary */ - } - .... - spin_unlock(&chip->lock); - return IRQ_HANDLED; - } -]]> - - - -
- -
- High frequency timer interrupts - - This happens when the hardware doesn't generate interrupts - at the period boundary but issues timer interrupts at a fixed - timer rate (e.g. es1968 or ymfpci drivers). - In this case, you need to check the current hardware - position and accumulate the processed sample length at each - interrupt. When the accumulated size exceeds the period - size, call - snd_pcm_period_elapsed() and reset the - accumulator. - - - - Typical code would be like the following. - - - Interrupt Handler Case #2 - -lock); - .... - if (pcm_irq_invoked(chip)) { - unsigned int last_ptr, size; - /* get the current hardware pointer (in frames) */ - last_ptr = get_hw_ptr(chip); - /* calculate the processed frames since the - * last update - */ - if (last_ptr < chip->last_ptr) - size = runtime->buffer_size + last_ptr - - chip->last_ptr; - else - size = last_ptr - chip->last_ptr; - /* remember the last updated point */ - chip->last_ptr = last_ptr; - /* accumulate the size */ - chip->size += size; - /* over the period boundary? */ - if (chip->size >= runtime->period_size) { - /* reset the accumulator */ - chip->size %= runtime->period_size; - /* call updater */ - spin_unlock(&chip->lock); - snd_pcm_period_elapsed(substream); - spin_lock(&chip->lock); - } - /* acknowledge the interrupt if necessary */ - } - .... - spin_unlock(&chip->lock); - return IRQ_HANDLED; - } -]]> - - - -
- -
- On calling <function>snd_pcm_period_elapsed()</function> - - In both cases, even if more than one period are elapsed, you - don't have to call - snd_pcm_period_elapsed() many times. Call - only once. And the pcm layer will check the current hardware - pointer and update to the latest status. - -
-
- -
- Atomicity - - One of the most important (and thus difficult to debug) problems - in kernel programming are race conditions. - In the Linux kernel, they are usually avoided via spin-locks, mutexes - or semaphores. In general, if a race condition can happen - in an interrupt handler, it has to be managed atomically, and you - have to use a spinlock to protect the critical session. If the - critical section is not in interrupt handler code and - if taking a relatively long time to execute is acceptable, you - should use mutexes or semaphores instead. - - - - As already seen, some pcm callbacks are atomic and some are - not. For example, the hw_params callback is - non-atomic, while trigger callback is - atomic. This means, the latter is called already in a spinlock - held by the PCM middle layer. Please take this atomicity into - account when you choose a locking scheme in the callbacks. - - - - In the atomic callbacks, you cannot use functions which may call - schedule or go to - sleep. Semaphores and mutexes can sleep, - and hence they cannot be used inside the atomic callbacks - (e.g. trigger callback). - To implement some delay in such a callback, please use - udelay() or mdelay(). - - - - All three atomic callbacks (trigger, pointer, and ack) are - called with local interrupts disabled. - - - - The recent changes in PCM core code, however, allow all PCM - operations to be non-atomic. This assumes that the all caller - sides are in non-atomic contexts. For example, the function - snd_pcm_period_elapsed() is called - typically from the interrupt handler. But, if you set up the - driver to use a threaded interrupt handler, this call can be in - non-atomic context, too. In such a case, you can set - nonatomic filed of - snd_pcm object after creating it. - When this flag is set, mutex and rwsem are used internally in - the PCM core instead of spin and rwlocks, so that you can call - all PCM functions safely in a non-atomic context. - - -
-
- Constraints - - If your chip supports unconventional sample rates, or only the - limited samples, you need to set a constraint for the - condition. - - - - For example, in order to restrict the sample rates in the some - supported values, use - snd_pcm_hw_constraint_list(). - You need to call this function in the open callback. - - - Example of Hardware Constraints - -runtime, 0, - SNDRV_PCM_HW_PARAM_RATE, - &constraints_rates); - if (err < 0) - return err; - .... - } -]]> - - - - - - There are many different constraints. - Look at sound/pcm.h for a complete list. - You can even define your own constraint rules. - For example, let's suppose my_chip can manage a substream of 1 channel - if and only if the format is S16_LE, otherwise it supports any format - specified in the snd_pcm_hardware structure (or in any - other constraint_list). You can build a rule like this: - - - Example of Hardware Constraints for Channels - -bits[0] == SNDRV_PCM_FMTBIT_S16_LE) { - ch.min = ch.max = 1; - ch.integer = 1; - return snd_interval_refine(c, &ch); - } - return 0; - } -]]> - - - - - - Then you need to call this function to add your rule: - - - -runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, - hw_rule_channels_by_format, NULL, - SNDRV_PCM_HW_PARAM_FORMAT, -1); -]]> - - - - - - The rule function is called when an application sets the PCM - format, and it refines the number of channels accordingly. - But an application may set the number of channels before - setting the format. Thus you also need to define the inverse rule: - - - Example of Hardware Constraints for Formats - -min < 2) { - fmt.bits[0] &= SNDRV_PCM_FMTBIT_S16_LE; - return snd_mask_refine(f, &fmt); - } - return 0; - } -]]> - - - - - - ...and in the open callback: - - -runtime, 0, SNDRV_PCM_HW_PARAM_FORMAT, - hw_rule_format_by_channels, NULL, - SNDRV_PCM_HW_PARAM_CHANNELS, -1); -]]> - - - - - - I won't give more details here, rather I - would like to say, Luke, use the source. - -
- - - - - - - - - Control Interface - -
- General - - The control interface is used widely for many switches, - sliders, etc. which are accessed from user-space. Its most - important use is the mixer interface. In other words, since ALSA - 0.9.x, all the mixer stuff is implemented on the control kernel API. - - - - ALSA has a well-defined AC97 control module. If your chip - supports only the AC97 and nothing else, you can skip this - section. - - - - The control API is defined in - <sound/control.h>. - Include this file if you want to add your own controls. - -
- -
- Definition of Controls - - To create a new control, you need to define the - following three - callbacks: info, - get and - put. Then, define a - struct snd_kcontrol_new record, such as: - - - Definition of a Control - - - - - - - - The iface field specifies the control - type, SNDRV_CTL_ELEM_IFACE_XXX, which - is usually MIXER. - Use CARD for global controls that are not - logically part of the mixer. - If the control is closely associated with some specific device on - the sound card, use HWDEP, - PCM, RAWMIDI, - TIMER, or SEQUENCER, and - specify the device number with the - device and - subdevice fields. - - - - The name is the name identifier - string. Since ALSA 0.9.x, the control name is very important, - because its role is classified from its name. There are - pre-defined standard control names. The details are described in - the - Control Names subsection. - - - - The index field holds the index number - of this control. If there are several different controls with - the same name, they can be distinguished by the index - number. This is the case when - several codecs exist on the card. If the index is zero, you can - omit the definition above. - - - - The access field contains the access - type of this control. Give the combination of bit masks, - SNDRV_CTL_ELEM_ACCESS_XXX, there. - The details will be explained in - the - Access Flags subsection. - - - - The private_value field contains - an arbitrary long integer value for this record. When using - the generic info, - get and - put callbacks, you can pass a value - through this field. If several small numbers are necessary, you can - combine them in bitwise. Or, it's possible to give a pointer - (casted to unsigned long) of some record to this field, too. - - - - The tlv field can be used to provide - metadata about the control; see the - - Metadata subsection. - - - - The other three are - - callback functions. - -
- -
- Control Names - - There are some standards to define the control names. A - control is usually defined from the three parts as - SOURCE DIRECTION FUNCTION. - - - - The first, SOURCE, specifies the source - of the control, and is a string such as Master, - PCM, CD and - Line. There are many pre-defined sources. - - - - The second, DIRECTION, is one of the - following strings according to the direction of the control: - Playback, Capture, Bypass - Playback and Bypass Capture. Or, it can - be omitted, meaning both playback and capture directions. - - - - The third, FUNCTION, is one of the - following strings according to the function of the control: - Switch, Volume and - Route. - - - - The example of control names are, thus, Master Capture - Switch or PCM Playback Volume. - - - - There are some exceptions: - - -
- Global capture and playback - - Capture Source, Capture Switch - and Capture Volume are used for the global - capture (input) source, switch and volume. Similarly, - Playback Switch and Playback - Volume are used for the global output gain switch and - volume. - -
- -
- Tone-controls - - tone-control switch and volumes are specified like - Tone Control - XXX, e.g. Tone Control - - Switch, Tone Control - Bass, - Tone Control - Center. - -
- -
- 3D controls - - 3D-control switches and volumes are specified like 3D - Control - XXX, e.g. 3D Control - - Switch, 3D Control - Center, 3D - Control - Space. - -
- -
- Mic boost - - Mic-boost switch is set as Mic Boost or - Mic Boost (6dB). - - - - More precise information can be found in - Documentation/sound/alsa/ControlNames.txt. - -
-
- -
- Access Flags - - - The access flag is the bitmask which specifies the access type - of the given control. The default access type is - SNDRV_CTL_ELEM_ACCESS_READWRITE, - which means both read and write are allowed to this control. - When the access flag is omitted (i.e. = 0), it is - considered as READWRITE access as default. - - - - When the control is read-only, pass - SNDRV_CTL_ELEM_ACCESS_READ instead. - In this case, you don't have to define - the put callback. - Similarly, when the control is write-only (although it's a rare - case), you can use the WRITE flag instead, and - you don't need the get callback. - - - - If the control value changes frequently (e.g. the VU meter), - VOLATILE flag should be given. This means - that the control may be changed without - - notification. Applications should poll such - a control constantly. - - - - When the control is inactive, set - the INACTIVE flag, too. - There are LOCK and - OWNER flags to change the write - permissions. - - -
- -
- Callbacks - -
- info callback - - The info callback is used to get - detailed information on this control. This must store the - values of the given struct snd_ctl_elem_info - object. For example, for a boolean control with a single - element: - - - Example of info callback - -type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; - uinfo->count = 1; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = 1; - return 0; - } -]]> - - - - - - The type field specifies the type - of the control. There are BOOLEAN, - INTEGER, ENUMERATED, - BYTES, IEC958 and - INTEGER64. The - count field specifies the - number of elements in this control. For example, a stereo - volume would have count = 2. The - value field is a union, and - the values stored are depending on the type. The boolean and - integer types are identical. - - - - The enumerated type is a bit different from others. You'll - need to set the string for the currently given item index. - - - -type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = 4; - if (uinfo->value.enumerated.item > 3) - uinfo->value.enumerated.item = 3; - strcpy(uinfo->value.enumerated.name, - texts[uinfo->value.enumerated.item]); - return 0; - } -]]> - - - - - - The above callback can be simplified with a helper function, - snd_ctl_enum_info. The final code - looks like below. - (You can pass ARRAY_SIZE(texts) instead of 4 in the third - argument; it's a matter of taste.) - - - - - - - - - - Some common info callbacks are available for your convenience: - snd_ctl_boolean_mono_info() and - snd_ctl_boolean_stereo_info(). - Obviously, the former is an info callback for a mono channel - boolean item, just like snd_myctl_mono_info - above, and the latter is for a stereo channel boolean item. - - -
- -
- get callback - - - This callback is used to read the current value of the - control and to return to user-space. - - - - For example, - - - Example of get callback - -value.integer.value[0] = get_some_value(chip); - return 0; - } -]]> - - - - - - The value field depends on - the type of control as well as on the info callback. For example, - the sb driver uses this field to store the register offset, - the bit-shift and the bit-mask. The - private_value field is set as follows: - - - - - - and is retrieved in callbacks like - - -private_value & 0xff; - int shift = (kcontrol->private_value >> 16) & 0xff; - int mask = (kcontrol->private_value >> 24) & 0xff; - .... - } -]]> - - - - - - In the get callback, - you have to fill all the elements if the - control has more than one elements, - i.e. count > 1. - In the example above, we filled only one element - (value.integer.value[0]) since it's - assumed as count = 1. - -
- -
- put callback - - - This callback is used to write a value from user-space. - - - - For example, - - - Example of put callback - -current_value != - ucontrol->value.integer.value[0]) { - change_current_value(chip, - ucontrol->value.integer.value[0]); - changed = 1; - } - return changed; - } -]]> - - - - As seen above, you have to return 1 if the value is - changed. If the value is not changed, return 0 instead. - If any fatal error happens, return a negative error code as - usual. - - - - As in the get callback, - when the control has more than one elements, - all elements must be evaluated in this callback, too. - -
- -
- Callbacks are not atomic - - All these three callbacks are basically not atomic. - -
-
- -
- Constructor - - When everything is ready, finally we can create a new - control. To create a control, there are two functions to be - called, snd_ctl_new1() and - snd_ctl_add(). - - - - In the simplest way, you can do like this: - - - - - - - - where my_control is the - struct snd_kcontrol_new object defined above, and chip - is the object pointer to be passed to - kcontrol->private_data - which can be referred to in callbacks. - - - - snd_ctl_new1() allocates a new - snd_kcontrol instance, - and snd_ctl_add assigns the given - control component to the card. - -
- -
- Change Notification - - If you need to change and update a control in the interrupt - routine, you can call snd_ctl_notify(). For - example, - - - - - - - - This function takes the card pointer, the event-mask, and the - control id pointer for the notification. The event-mask - specifies the types of notification, for example, in the above - example, the change of control values is notified. - The id pointer is the pointer of struct snd_ctl_elem_id - to be notified. - You can find some examples in es1938.c or - es1968.c for hardware volume interrupts. - -
- -
- Metadata - - To provide information about the dB values of a mixer control, use - on of the DECLARE_TLV_xxx macros from - <sound/tlv.h> to define a variable - containing this information, set thetlv.p - field to point to this variable, and include the - SNDRV_CTL_ELEM_ACCESS_TLV_READ flag in the - access field; like this: - - - - - - - - - The DECLARE_TLV_DB_SCALE macro defines - information about a mixer control where each step in the control's - value changes the dB value by a constant dB amount. - The first parameter is the name of the variable to be defined. - The second parameter is the minimum value, in units of 0.01 dB. - The third parameter is the step size, in units of 0.01 dB. - Set the fourth parameter to 1 if the minimum value actually mutes - the control. - - - - The DECLARE_TLV_DB_LINEAR macro defines - information about a mixer control where the control's value affects - the output linearly. - The first parameter is the name of the variable to be defined. - The second parameter is the minimum value, in units of 0.01 dB. - The third parameter is the maximum value, in units of 0.01 dB. - If the minimum value mutes the control, set the second parameter to - TLV_DB_GAIN_MUTE. - -
- - - - - - - - - API for AC97 Codec - -
- General - - The ALSA AC97 codec layer is a well-defined one, and you don't - have to write much code to control it. Only low-level control - routines are necessary. The AC97 codec API is defined in - <sound/ac97_codec.h>. - -
- -
- Full Code Example - - - Example of AC97 Interface - -private_data; - .... - /* read a register value here from the codec */ - return the_register_value; - } - - static void snd_mychip_ac97_write(struct snd_ac97 *ac97, - unsigned short reg, unsigned short val) - { - struct mychip *chip = ac97->private_data; - .... - /* write the given register value to the codec */ - } - - static int snd_mychip_ac97(struct mychip *chip) - { - struct snd_ac97_bus *bus; - struct snd_ac97_template ac97; - int err; - static struct snd_ac97_bus_ops ops = { - .write = snd_mychip_ac97_write, - .read = snd_mychip_ac97_read, - }; - - err = snd_ac97_bus(chip->card, 0, &ops, NULL, &bus); - if (err < 0) - return err; - memset(&ac97, 0, sizeof(ac97)); - ac97.private_data = chip; - return snd_ac97_mixer(bus, &ac97, &chip->ac97); - } - -]]> - - - -
- -
- Constructor - - To create an ac97 instance, first call snd_ac97_bus - with an ac97_bus_ops_t record with callback functions. - - - - - - - - The bus record is shared among all belonging ac97 instances. - - - - And then call snd_ac97_mixer() with an - struct snd_ac97_template - record together with the bus pointer created above. - - - -ac97); -]]> - - - - where chip->ac97 is a pointer to a newly created - ac97_t instance. - In this case, the chip pointer is set as the private data, so that - the read/write callback functions can refer to this chip instance. - This instance is not necessarily stored in the chip - record. If you need to change the register values from the - driver, or need the suspend/resume of ac97 codecs, keep this - pointer to pass to the corresponding functions. - -
- -
- Callbacks - - The standard callbacks are read and - write. Obviously they - correspond to the functions for read and write accesses to the - hardware low-level codes. - - - - The read callback returns the - register value specified in the argument. - - - -private_data; - .... - return the_register_value; - } -]]> - - - - Here, the chip can be cast from ac97->private_data. - - - - Meanwhile, the write callback is - used to set the register value. - - - - - - - - - - These callbacks are non-atomic like the control API callbacks. - - - - There are also other callbacks: - reset, - wait and - init. - - - - The reset callback is used to reset - the codec. If the chip requires a special kind of reset, you can - define this callback. - - - - The wait callback is used to - add some waiting time in the standard initialization of the codec. If the - chip requires the extra waiting time, define this callback. - - - - The init callback is used for - additional initialization of the codec. - -
- -
- Updating Registers in The Driver - - If you need to access to the codec from the driver, you can - call the following functions: - snd_ac97_write(), - snd_ac97_read(), - snd_ac97_update() and - snd_ac97_update_bits(). - - - - Both snd_ac97_write() and - snd_ac97_update() functions are used to - set a value to the given register - (AC97_XXX). The difference between them is - that snd_ac97_update() doesn't write a - value if the given value has been already set, while - snd_ac97_write() always rewrites the - value. - - - - - - - - - - snd_ac97_read() is used to read the value - of the given register. For example, - - - - - - - - - - snd_ac97_update_bits() is used to update - some bits in the given register. - - - - - - - - - - Also, there is a function to change the sample rate (of a - given register such as - AC97_PCM_FRONT_DAC_RATE) when VRA or - DRA is supported by the codec: - snd_ac97_set_rate(). - - - - - - - - - - The following registers are available to set the rate: - AC97_PCM_MIC_ADC_RATE, - AC97_PCM_FRONT_DAC_RATE, - AC97_PCM_LR_ADC_RATE, - AC97_SPDIF. When - AC97_SPDIF is specified, the register is - not really changed but the corresponding IEC958 status bits will - be updated. - -
- -
- Clock Adjustment - - In some chips, the clock of the codec isn't 48000 but using a - PCI clock (to save a quartz!). In this case, change the field - bus->clock to the corresponding - value. For example, intel8x0 - and es1968 drivers have their own function to read from the clock. - -
- -
- Proc Files - - The ALSA AC97 interface will create a proc file such as - /proc/asound/card0/codec97#0/ac97#0-0 and - ac97#0-0+regs. You can refer to these files to - see the current status and registers of the codec. - -
- -
- Multiple Codecs - - When there are several codecs on the same card, you need to - call snd_ac97_mixer() multiple times with - ac97.num=1 or greater. The num field - specifies the codec number. - - - - If you set up multiple codecs, you either need to write - different callbacks for each codec or check - ac97->num in the callback routines. - -
- - - - - - - - - MIDI (MPU401-UART) Interface - -
- General - - Many soundcards have built-in MIDI (MPU401-UART) - interfaces. When the soundcard supports the standard MPU401-UART - interface, most likely you can use the ALSA MPU401-UART API. The - MPU401-UART API is defined in - <sound/mpu401.h>. - - - - Some soundchips have a similar but slightly different - implementation of mpu401 stuff. For example, emu10k1 has its own - mpu401 routines. - -
- -
- Constructor - - To create a rawmidi object, call - snd_mpu401_uart_new(). - - - - - - - - - - The first argument is the card pointer, and the second is the - index of this component. You can create up to 8 rawmidi - devices. - - - - The third argument is the type of the hardware, - MPU401_HW_XXX. If it's not a special one, - you can use MPU401_HW_MPU401. - - - - The 4th argument is the I/O port address. Many - backward-compatible MPU401 have an I/O port such as 0x330. Or, it - might be a part of its own PCI I/O region. It depends on the - chip design. - - - - The 5th argument is a bitflag for additional information. - When the I/O port address above is part of the PCI I/O - region, the MPU401 I/O port might have been already allocated - (reserved) by the driver itself. In such a case, pass a bit flag - MPU401_INFO_INTEGRATED, - and the mpu401-uart layer will allocate the I/O ports by itself. - - - - When the controller supports only the input or output MIDI stream, - pass the MPU401_INFO_INPUT or - MPU401_INFO_OUTPUT bitflag, respectively. - Then the rawmidi instance is created as a single stream. - - - - MPU401_INFO_MMIO bitflag is used to change - the access method to MMIO (via readb and writeb) instead of - iob and outb. In this case, you have to pass the iomapped address - to snd_mpu401_uart_new(). - - - - When MPU401_INFO_TX_IRQ is set, the output - stream isn't checked in the default interrupt handler. The driver - needs to call snd_mpu401_uart_interrupt_tx() - by itself to start processing the output stream in the irq handler. - - - - If the MPU-401 interface shares its interrupt with the other logical - devices on the card, set MPU401_INFO_IRQ_HOOK - (see - below). - - - - Usually, the port address corresponds to the command port and - port + 1 corresponds to the data port. If not, you may change - the cport field of - struct snd_mpu401 manually - afterward. However, snd_mpu401 pointer is not - returned explicitly by - snd_mpu401_uart_new(). You need to cast - rmidi->private_data to - snd_mpu401 explicitly, - - - -private_data; -]]> - - - - and reset the cport as you like: - - - -cport = my_own_control_port; -]]> - - - - - - The 6th argument specifies the ISA irq number that will be - allocated. If no interrupt is to be allocated (because your - code is already allocating a shared interrupt, or because the - device does not use interrupts), pass -1 instead. - For a MPU-401 device without an interrupt, a polling timer - will be used instead. - -
- -
- Interrupt Handler - - When the interrupt is allocated in - snd_mpu401_uart_new(), an exclusive ISA - interrupt handler is automatically used, hence you don't have - anything else to do than creating the mpu401 stuff. Otherwise, you - have to set MPU401_INFO_IRQ_HOOK, and call - snd_mpu401_uart_interrupt() explicitly from your - own interrupt handler when it has determined that a UART interrupt - has occurred. - - - - In this case, you need to pass the private_data of the - returned rawmidi object from - snd_mpu401_uart_new() as the second - argument of snd_mpu401_uart_interrupt(). - - - -private_data, regs); -]]> - - - -
- - - - - - - - - RawMIDI Interface - -
- Overview - - - The raw MIDI interface is used for hardware MIDI ports that can - be accessed as a byte stream. It is not used for synthesizer - chips that do not directly understand MIDI. - - - - ALSA handles file and buffer management. All you have to do is - to write some code to move data between the buffer and the - hardware. - - - - The rawmidi API is defined in - <sound/rawmidi.h>. - -
- -
- Constructor - - - To create a rawmidi device, call the - snd_rawmidi_new function: - - -card, "MyMIDI", 0, outs, ins, &rmidi); - if (err < 0) - return err; - rmidi->private_data = chip; - strcpy(rmidi->name, "My MIDI"); - rmidi->info_flags = SNDRV_RAWMIDI_INFO_OUTPUT | - SNDRV_RAWMIDI_INFO_INPUT | - SNDRV_RAWMIDI_INFO_DUPLEX; -]]> - - - - - - The first argument is the card pointer, the second argument is - the ID string. - - - - The third argument is the index of this component. You can - create up to 8 rawmidi devices. - - - - The fourth and fifth arguments are the number of output and - input substreams, respectively, of this device (a substream is - the equivalent of a MIDI port). - - - - Set the info_flags field to specify - the capabilities of the device. - Set SNDRV_RAWMIDI_INFO_OUTPUT if there is - at least one output port, - SNDRV_RAWMIDI_INFO_INPUT if there is at - least one input port, - and SNDRV_RAWMIDI_INFO_DUPLEX if the device - can handle output and input at the same time. - - - - After the rawmidi device is created, you need to set the - operators (callbacks) for each substream. There are helper - functions to set the operators for all the substreams of a device: - - - - - - - - - The operators are usually defined like this: - - - - - - These callbacks are explained in the Callbacks - section. - - - - If there are more than one substream, you should give a - unique name to each of them: - - -streams[SNDRV_RAWMIDI_STREAM_OUTPUT].substreams, - list { - sprintf(substream->name, "My MIDI Port %d", substream->number + 1); - } - /* same for SNDRV_RAWMIDI_STREAM_INPUT */ -]]> - - - -
- -
- Callbacks - - - In all the callbacks, the private data that you've set for the - rawmidi device can be accessed as - substream->rmidi->private_data. - - - - - If there is more than one port, your callbacks can determine the - port index from the struct snd_rawmidi_substream data passed to each - callback: - - -number; -]]> - - - - -
- <function>open</function> callback - - - - - - - - - This is called when a substream is opened. - You can initialize the hardware here, but you shouldn't - start transmitting/receiving data yet. - -
- -
- <function>close</function> callback - - - - - - - - - Guess what. - - - - The open and close - callbacks of a rawmidi device are serialized with a mutex, - and can sleep. - -
- -
- <function>trigger</function> callback for output - substreams - - - - - - - - - This is called with a nonzero up - parameter when there is some data in the substream buffer that - must be transmitted. - - - - To read data from the buffer, call - snd_rawmidi_transmit_peek. It will - return the number of bytes that have been read; this will be - less than the number of bytes requested when there are no more - data in the buffer. - After the data have been transmitted successfully, call - snd_rawmidi_transmit_ack to remove the - data from the substream buffer: - - - - - - - - - If you know beforehand that the hardware will accept data, you - can use the snd_rawmidi_transmit function - which reads some data and removes them from the buffer at once: - - - - - - - - - If you know beforehand how many bytes you can accept, you can - use a buffer size greater than one with the - snd_rawmidi_transmit* functions. - - - - The trigger callback must not sleep. If - the hardware FIFO is full before the substream buffer has been - emptied, you have to continue transmitting data later, either - in an interrupt handler, or with a timer if the hardware - doesn't have a MIDI transmit interrupt. - - - - The trigger callback is called with a - zero up parameter when the transmission - of data should be aborted. - -
- -
- <function>trigger</function> callback for input - substreams - - - - - - - - - This is called with a nonzero up - parameter to enable receiving data, or with a zero - up parameter do disable receiving data. - - - - The trigger callback must not sleep; the - actual reading of data from the device is usually done in an - interrupt handler. - - - - When data reception is enabled, your interrupt handler should - call snd_rawmidi_receive for all received - data: - - - - - - -
- -
- <function>drain</function> callback - - - - - - - - - This is only used with output substreams. This function should wait - until all data read from the substream buffer have been transmitted. - This ensures that the device can be closed and the driver unloaded - without losing data. - - - - This callback is optional. If you do not set - drain in the struct snd_rawmidi_ops - structure, ALSA will simply wait for 50 milliseconds - instead. - -
-
- - - - - - - - - Miscellaneous Devices - -
- FM OPL3 - - The FM OPL3 is still used in many chips (mainly for backward - compatibility). ALSA has a nice OPL3 FM control layer, too. The - OPL3 API is defined in - <sound/opl3.h>. - - - - FM registers can be directly accessed through the direct-FM API, - defined in <sound/asound_fm.h>. In - ALSA native mode, FM registers are accessed through - the Hardware-Dependent Device direct-FM extension API, whereas in - OSS compatible mode, FM registers can be accessed with the OSS - direct-FM compatible API in /dev/dmfmX device. - - - - To create the OPL3 component, you have two functions to - call. The first one is a constructor for the opl3_t - instance. - - - - - - - - - - The first argument is the card pointer, the second one is the - left port address, and the third is the right port address. In - most cases, the right port is placed at the left port + 2. - - - - The fourth argument is the hardware type. - - - - When the left and right ports have been already allocated by - the card driver, pass non-zero to the fifth argument - (integrated). Otherwise, the opl3 module will - allocate the specified ports by itself. - - - - When the accessing the hardware requires special method - instead of the standard I/O access, you can create opl3 instance - separately with snd_opl3_new(). - - - - - - - - - - Then set command, - private_data and - private_free for the private - access function, the private data and the destructor. - The l_port and r_port are not necessarily set. Only the - command must be set properly. You can retrieve the data - from the opl3->private_data field. - - - - After creating the opl3 instance via snd_opl3_new(), - call snd_opl3_init() to initialize the chip to the - proper state. Note that snd_opl3_create() always - calls it internally. - - - - If the opl3 instance is created successfully, then create a - hwdep device for this opl3. - - - - - - - - - - The first argument is the opl3_t instance you - created, and the second is the index number, usually 0. - - - - The third argument is the index-offset for the sequencer - client assigned to the OPL3 port. When there is an MPU401-UART, - give 1 for here (UART always takes 0). - -
- -
- Hardware-Dependent Devices - - Some chips need user-space access for special - controls or for loading the micro code. In such a case, you can - create a hwdep (hardware-dependent) device. The hwdep API is - defined in <sound/hwdep.h>. You can - find examples in opl3 driver or - isa/sb/sb16_csp.c. - - - - The creation of the hwdep instance is done via - snd_hwdep_new(). - - - - - - - - where the third argument is the index number. - - - - You can then pass any pointer value to the - private_data. - If you assign a private data, you should define the - destructor, too. The destructor function is set in - the private_free field. - - - -private_data = p; - hw->private_free = mydata_free; -]]> - - - - and the implementation of the destructor would be: - - - -private_data; - kfree(p); - } -]]> - - - - - - The arbitrary file operations can be defined for this - instance. The file operators are defined in - the ops table. For example, assume that - this chip needs an ioctl. - - - -ops.open = mydata_open; - hw->ops.ioctl = mydata_ioctl; - hw->ops.release = mydata_release; -]]> - - - - And implement the callback functions as you like. - -
- -
- IEC958 (S/PDIF) - - Usually the controls for IEC958 devices are implemented via - the control interface. There is a macro to compose a name string for - IEC958 controls, SNDRV_CTL_NAME_IEC958() - defined in <include/asound.h>. - - - - There are some standard controls for IEC958 status bits. These - controls use the type SNDRV_CTL_ELEM_TYPE_IEC958, - and the size of element is fixed as 4 bytes array - (value.iec958.status[x]). For the info - callback, you don't specify - the value field for this type (the count field must be set, - though). - - - - IEC958 Playback Con Mask is used to return the - bit-mask for the IEC958 status bits of consumer mode. Similarly, - IEC958 Playback Pro Mask returns the bitmask for - professional mode. They are read-only controls, and are defined - as MIXER controls (iface = - SNDRV_CTL_ELEM_IFACE_MIXER). - - - - Meanwhile, IEC958 Playback Default control is - defined for getting and setting the current default IEC958 - bits. Note that this one is usually defined as a PCM control - (iface = SNDRV_CTL_ELEM_IFACE_PCM), - although in some places it's defined as a MIXER control. - - - - In addition, you can define the control switches to - enable/disable or to set the raw bit mode. The implementation - will depend on the chip, but the control should be named as - IEC958 xxx, preferably using - the SNDRV_CTL_NAME_IEC958() macro. - - - - You can find several cases, for example, - pci/emu10k1, - pci/ice1712, or - pci/cmipci.c. - -
- - - - - - - - - Buffer and Memory Management - -
- Buffer Types - - ALSA provides several different buffer allocation functions - depending on the bus and the architecture. All these have a - consistent API. The allocation of physically-contiguous pages is - done via - snd_malloc_xxx_pages() function, where xxx - is the bus type. - - - - The allocation of pages with fallback is - snd_malloc_xxx_pages_fallback(). This - function tries to allocate the specified pages but if the pages - are not available, it tries to reduce the page sizes until - enough space is found. - - - - The release the pages, call - snd_free_xxx_pages() function. - - - - Usually, ALSA drivers try to allocate and reserve - a large contiguous physical space - at the time the module is loaded for the later use. - This is called pre-allocation. - As already written, you can call the following function at - pcm instance construction time (in the case of PCI bus). - - - - - - - - where size is the byte size to be - pre-allocated and the max is the maximum - size to be changed via the prealloc proc file. - The allocator will try to get an area as large as possible - within the given size. - - - - The second argument (type) and the third argument (device pointer) - are dependent on the bus. - In the case of the ISA bus, pass snd_dma_isa_data() - as the third argument with SNDRV_DMA_TYPE_DEV type. - For the continuous buffer unrelated to the bus can be pre-allocated - with SNDRV_DMA_TYPE_CONTINUOUS type and the - snd_dma_continuous_data(GFP_KERNEL) device pointer, - where GFP_KERNEL is the kernel allocation flag to - use. - For the PCI scatter-gather buffers, use - SNDRV_DMA_TYPE_DEV_SG with - snd_dma_pci_data(pci) - (see the - Non-Contiguous Buffers - section). - - - - Once the buffer is pre-allocated, you can use the - allocator in the hw_params callback: - - - - - - - - Note that you have to pre-allocate to use this function. - -
- -
- External Hardware Buffers - - Some chips have their own hardware buffers and the DMA - transfer from the host memory is not available. In such a case, - you need to either 1) copy/set the audio data directly to the - external hardware buffer, or 2) make an intermediate buffer and - copy/set the data from it to the external hardware buffer in - interrupts (or in tasklets, preferably). - - - - The first case works fine if the external hardware buffer is large - enough. This method doesn't need any extra buffers and thus is - more effective. You need to define the - copy and - silence callbacks for - the data transfer. However, there is a drawback: it cannot - be mmapped. The examples are GUS's GF1 PCM or emu8000's - wavetable PCM. - - - - The second case allows for mmap on the buffer, although you have - to handle an interrupt or a tasklet to transfer the data - from the intermediate buffer to the hardware buffer. You can find an - example in the vxpocket driver. - - - - Another case is when the chip uses a PCI memory-map - region for the buffer instead of the host memory. In this case, - mmap is available only on certain architectures like the Intel one. - In non-mmap mode, the data cannot be transferred as in the normal - way. Thus you need to define the copy and - silence callbacks as well, - as in the cases above. The examples are found in - rme32.c and rme96.c. - - - - The implementation of the copy and - silence callbacks depends upon - whether the hardware supports interleaved or non-interleaved - samples. The copy callback is - defined like below, a bit - differently depending whether the direction is playback or - capture: - - - - - - - - - - In the case of interleaved samples, the second argument - (channel) is not used. The third argument - (pos) points the - current position offset in frames. - - - - The meaning of the fourth argument is different between - playback and capture. For playback, it holds the source data - pointer, and for capture, it's the destination data pointer. - - - - The last argument is the number of frames to be copied. - - - - What you have to do in this callback is again different - between playback and capture directions. In the - playback case, you copy the given amount of data - (count) at the specified pointer - (src) to the specified offset - (pos) on the hardware buffer. When - coded like memcpy-like way, the copy would be like: - - - - - - - - - - For the capture direction, you copy the given amount of - data (count) at the specified offset - (pos) on the hardware buffer to the - specified pointer (dst). - - - - - - - - Note that both the position and the amount of data are given - in frames. - - - - In the case of non-interleaved samples, the implementation - will be a bit more complicated. - - - - You need to check the channel argument, and if it's -1, copy - the whole channels. Otherwise, you have to copy only the - specified channel. Please check - isa/gus/gus_pcm.c as an example. - - - - The silence callback is also - implemented in a similar way. - - - - - - - - - - The meanings of arguments are the same as in the - copy - callback, although there is no src/dst - argument. In the case of interleaved samples, the channel - argument has no meaning, as well as on - copy callback. - - - - The role of silence callback is to - set the given amount - (count) of silence data at the - specified offset (pos) on the hardware - buffer. Suppose that the data format is signed (that is, the - silent-data is 0), and the implementation using a memset-like - function would be like: - - - - - - - - - - In the case of non-interleaved samples, again, the - implementation becomes a bit more complicated. See, for example, - isa/gus/gus_pcm.c. - -
- -
- Non-Contiguous Buffers - - If your hardware supports the page table as in emu10k1 or the - buffer descriptors as in via82xx, you can use the scatter-gather - (SG) DMA. ALSA provides an interface for handling SG-buffers. - The API is provided in <sound/pcm.h>. - - - - For creating the SG-buffer handler, call - snd_pcm_lib_preallocate_pages() or - snd_pcm_lib_preallocate_pages_for_all() - with SNDRV_DMA_TYPE_DEV_SG - in the PCM constructor like other PCI pre-allocator. - You need to pass snd_dma_pci_data(pci), - where pci is the struct pci_dev pointer - of the chip as well. - The struct snd_sg_buf instance is created as - substream->dma_private. You can cast - the pointer like: - - - -dma_private; -]]> - - - - - - Then call snd_pcm_lib_malloc_pages() - in the hw_params callback - as well as in the case of normal PCI buffer. - The SG-buffer handler will allocate the non-contiguous kernel - pages of the given size and map them onto the virtually contiguous - memory. The virtual pointer is addressed in runtime->dma_area. - The physical address (runtime->dma_addr) is set to zero, - because the buffer is physically non-contiguous. - The physical address table is set up in sgbuf->table. - You can get the physical address at a certain offset via - snd_pcm_sgbuf_get_addr(). - - - - When a SG-handler is used, you need to set - snd_pcm_sgbuf_ops_page as - the page callback. - (See - page callback section.) - - - - To release the data, call - snd_pcm_lib_free_pages() in the - hw_free callback as usual. - -
- -
- Vmalloc'ed Buffers - - It's possible to use a buffer allocated via - vmalloc, for example, for an intermediate - buffer. Since the allocated pages are not contiguous, you need - to set the page callback to obtain - the physical address at every offset. - - - - The implementation of page callback - would be like this: - - - - - - /* get the physical page pointer on the given offset */ - static struct page *mychip_page(struct snd_pcm_substream *substream, - unsigned long offset) - { - void *pageptr = substream->runtime->dma_area + offset; - return vmalloc_to_page(pageptr); - } -]]> - - - -
- - - - - - - - - Proc Interface - - ALSA provides an easy interface for procfs. The proc files are - very useful for debugging. I recommend you set up proc files if - you write a driver and want to get a running status or register - dumps. The API is found in - <sound/info.h>. - - - - To create a proc file, call - snd_card_proc_new(). - - - - - - - - where the second argument specifies the name of the proc file to be - created. The above example will create a file - my-file under the card directory, - e.g. /proc/asound/card0/my-file. - - - - Like other components, the proc entry created via - snd_card_proc_new() will be registered and - released automatically in the card registration and release - functions. - - - - When the creation is successful, the function stores a new - instance in the pointer given in the third argument. - It is initialized as a text proc file for read only. To use - this proc file as a read-only text file as it is, set the read - callback with a private data via - snd_info_set_text_ops(). - - - - - - - - where the second argument (chip) is the - private data to be used in the callbacks. The third parameter - specifies the read buffer size and the fourth - (my_proc_read) is the callback function, which - is defined like - - - - - - - - - - - In the read callback, use snd_iprintf() for - output strings, which works just like normal - printf(). For example, - - - -private_data; - - snd_iprintf(buffer, "This is my chip!\n"); - snd_iprintf(buffer, "Port = %ld\n", chip->port); - } -]]> - - - - - - The file permissions can be changed afterwards. As default, it's - set as read only for all users. If you want to add write - permission for the user (root as default), do as follows: - - - -mode = S_IFREG | S_IRUGO | S_IWUSR; -]]> - - - - and set the write buffer size and the callback - - - -c.text.write = my_proc_write; -]]> - - - - - - For the write callback, you can use - snd_info_get_line() to get a text line, and - snd_info_get_str() to retrieve a string from - the line. Some examples are found in - core/oss/mixer_oss.c, core/oss/and - pcm_oss.c. - - - - For a raw-data proc-file, set the attributes as follows: - - - -content = SNDRV_INFO_CONTENT_DATA; - entry->private_data = chip; - entry->c.ops = &my_file_io_ops; - entry->size = 4096; - entry->mode = S_IFREG | S_IRUGO; -]]> - - - - For the raw data, size field must be - set properly. This specifies the maximum size of the proc file access. - - - - The read/write callbacks of raw mode are more direct than the text mode. - You need to use a low-level I/O functions such as - copy_from/to_user() to transfer the - data. - - - - - - - - If the size of the info entry has been set up properly, - count and pos are - guaranteed to fit within 0 and the given size. - You don't have to check the range in the callbacks unless any - other condition is required. - - - - - - - - - - - Power Management - - If the chip is supposed to work with suspend/resume - functions, you need to add power-management code to the - driver. The additional code for power-management should be - ifdef'ed with - CONFIG_PM. - - - - If the driver fully supports suspend/resume - that is, the device can be - properly resumed to its state when suspend was called, - you can set the SNDRV_PCM_INFO_RESUME flag - in the pcm info field. Usually, this is possible when the - registers of the chip can be safely saved and restored to - RAM. If this is set, the trigger callback is called with - SNDRV_PCM_TRIGGER_RESUME after the resume - callback completes. - - - - Even if the driver doesn't support PM fully but - partial suspend/resume is still possible, it's still worthy to - implement suspend/resume callbacks. In such a case, applications - would reset the status by calling - snd_pcm_prepare() and restart the stream - appropriately. Hence, you can define suspend/resume callbacks - below but don't set SNDRV_PCM_INFO_RESUME - info flag to the PCM. - - - - Note that the trigger with SUSPEND can always be called when - snd_pcm_suspend_all is called, - regardless of the SNDRV_PCM_INFO_RESUME flag. - The RESUME flag affects only the behavior - of snd_pcm_resume(). - (Thus, in theory, - SNDRV_PCM_TRIGGER_RESUME isn't needed - to be handled in the trigger callback when no - SNDRV_PCM_INFO_RESUME flag is set. But, - it's better to keep it for compatibility reasons.) - - - In the earlier version of ALSA drivers, a common - power-management layer was provided, but it has been removed. - The driver needs to define the suspend/resume hooks according to - the bus the device is connected to. In the case of PCI drivers, the - callbacks look like below: - - - - - - - - - - The scheme of the real suspend job is as follows. - - - Retrieve the card and the chip data. - Call snd_power_change_state() with - SNDRV_CTL_POWER_D3hot to change the - power status. - Call snd_pcm_suspend_all() to suspend the running PCM streams. - If AC97 codecs are used, call - snd_ac97_suspend() for each codec. - Save the register values if necessary. - Stop the hardware if necessary. - Disable the PCI device by calling - pci_disable_device(). Then, call - pci_save_state() at last. - - - - - A typical code would be like: - - - -private_data; - /* (2) */ - snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); - /* (3) */ - snd_pcm_suspend_all(chip->pcm); - /* (4) */ - snd_ac97_suspend(chip->ac97); - /* (5) */ - snd_mychip_save_registers(chip); - /* (6) */ - snd_mychip_stop_hardware(chip); - /* (7) */ - pci_disable_device(pci); - pci_save_state(pci); - return 0; - } -]]> - - - - - - The scheme of the real resume job is as follows. - - - Retrieve the card and the chip data. - Set up PCI. First, call pci_restore_state(). - Then enable the pci device again by calling pci_enable_device(). - Call pci_set_master() if necessary, too. - Re-initialize the chip. - Restore the saved registers if necessary. - Resume the mixer, e.g. calling - snd_ac97_resume(). - Restart the hardware (if any). - Call snd_power_change_state() with - SNDRV_CTL_POWER_D0 to notify the processes. - - - - - A typical code would be like: - - - -private_data; - /* (2) */ - pci_restore_state(pci); - pci_enable_device(pci); - pci_set_master(pci); - /* (3) */ - snd_mychip_reinit_chip(chip); - /* (4) */ - snd_mychip_restore_registers(chip); - /* (5) */ - snd_ac97_resume(chip->ac97); - /* (6) */ - snd_mychip_restart_chip(chip); - /* (7) */ - snd_power_change_state(card, SNDRV_CTL_POWER_D0); - return 0; - } -]]> - - - - - - As shown in the above, it's better to save registers after - suspending the PCM operations via - snd_pcm_suspend_all() or - snd_pcm_suspend(). It means that the PCM - streams are already stopped when the register snapshot is - taken. But, remember that you don't have to restart the PCM - stream in the resume callback. It'll be restarted via - trigger call with SNDRV_PCM_TRIGGER_RESUME - when necessary. - - - - OK, we have all callbacks now. Let's set them up. In the - initialization of the card, make sure that you can get the chip - data from the card instance, typically via - private_data field, in case you - created the chip data individually. - - - -dev, index[dev], id[dev], THIS_MODULE, - 0, &card); - .... - chip = kzalloc(sizeof(*chip), GFP_KERNEL); - .... - card->private_data = chip; - .... - } -]]> - - - - When you created the chip data with - snd_card_new(), it's anyway accessible - via private_data field. - - - -dev, index[dev], id[dev], THIS_MODULE, - sizeof(struct mychip), &card); - .... - chip = card->private_data; - .... - } -]]> - - - - - - - If you need a space to save the registers, allocate the - buffer for it here, too, since it would be fatal - if you cannot allocate a memory in the suspend phase. - The allocated buffer should be released in the corresponding - destructor. - - - - And next, set suspend/resume callbacks to the pci_driver. - - - - - - - - - - - - - - - - Module Parameters - - There are standard module options for ALSA. At least, each - module should have the index, - id and enable - options. - - - - If the module supports multiple cards (usually up to - 8 = SNDRV_CARDS cards), they should be - arrays. The default initial values are defined already as - constants for easier programming: - - - - - - - - - - If the module supports only a single card, they could be single - variables, instead. enable option is not - always necessary in this case, but it would be better to have a - dummy option for compatibility. - - - - The module parameters must be declared with the standard - module_param()(), - module_param_array()() and - MODULE_PARM_DESC() macros. - - - - The typical coding would be like below: - - - - - - - - - - Also, don't forget to define the module description, classes, - license and devices. Especially, the recent modprobe requires to - define the module license as GPL, etc., otherwise the system is - shown as tainted. - - - - - - - - - - - - - - - - How To Put Your Driver Into ALSA Tree -
- General - - So far, you've learned how to write the driver codes. - And you might have a question now: how to put my own - driver into the ALSA driver tree? - Here (finally :) the standard procedure is described briefly. - - - - Suppose that you create a new PCI driver for the card - xyz. The card module name would be - snd-xyz. The new driver is usually put into the alsa-driver - tree, alsa-driver/pci directory in - the case of PCI cards. - Then the driver is evaluated, audited and tested - by developers and users. After a certain time, the driver - will go to the alsa-kernel tree (to the corresponding directory, - such as alsa-kernel/pci) and eventually - will be integrated into the Linux 2.6 tree (the directory would be - linux/sound/pci). - - - - In the following sections, the driver code is supposed - to be put into alsa-driver tree. The two cases are covered: - a driver consisting of a single source file and one consisting - of several source files. - -
- -
- Driver with A Single Source File - - - - - Modify alsa-driver/pci/Makefile - - - - Suppose you have a file xyz.c. Add the following - two lines - - - - - - - - - - - Create the Kconfig entry - - - - Add the new entry of Kconfig for your xyz driver. - - - - - - - the line, select SND_PCM, specifies that the driver xyz supports - PCM. In addition to SND_PCM, the following components are - supported for select command: - SND_RAWMIDI, SND_TIMER, SND_HWDEP, SND_MPU401_UART, - SND_OPL3_LIB, SND_OPL4_LIB, SND_VX_LIB, SND_AC97_CODEC. - Add the select command for each supported component. - - - - Note that some selections imply the lowlevel selections. - For example, PCM includes TIMER, MPU401_UART includes RAWMIDI, - AC97_CODEC includes PCM, and OPL3_LIB includes HWDEP. - You don't need to give the lowlevel selections again. - - - - For the details of Kconfig script, refer to the kbuild - documentation. - - - - - - - Run cvscompile script to re-generate the configure script and - build the whole stuff again. - - - - -
- -
- Drivers with Several Source Files - - Suppose that the driver snd-xyz have several source files. - They are located in the new subdirectory, - pci/xyz. - - - - - Add a new directory (xyz) in - alsa-driver/pci/Makefile as below - - - - - - - - - - - - Under the directory xyz, create a Makefile - - - Sample Makefile for a driver xyz - - - - - - - - - - Create the Kconfig entry - - - - This procedure is as same as in the last section. - - - - - - Run cvscompile script to re-generate the configure script and - build the whole stuff again. - - - - -
- - - - - - - - Useful Functions - -
- <function>snd_printk()</function> and friends - - ALSA provides a verbose version of the - printk() function. If a kernel config - CONFIG_SND_VERBOSE_PRINTK is set, this - function prints the given message together with the file name - and the line of the caller. The KERN_XXX - prefix is processed as - well as the original printk() does, so it's - recommended to add this prefix, e.g. - - - - - - - - - - There are also printk()'s for - debugging. snd_printd() can be used for - general debugging purposes. If - CONFIG_SND_DEBUG is set, this function is - compiled, and works just like - snd_printk(). If the ALSA is compiled - without the debugging flag, it's ignored. - - - - snd_printdd() is compiled in only when - CONFIG_SND_DEBUG_VERBOSE is set. Please note - that CONFIG_SND_DEBUG_VERBOSE is not set as default - even if you configure the alsa-driver with - option. You need to give - explicitly option instead. - -
- -
- <function>snd_BUG()</function> - - It shows the BUG? message and - stack trace as well as snd_BUG_ON at the point. - It's useful to show that a fatal error happens there. - - - When no debug flag is set, this macro is ignored. - -
- -
- <function>snd_BUG_ON()</function> - - snd_BUG_ON() macro is similar with - WARN_ON() macro. For example, - - - - - - - - or it can be used as the condition, - - - - - - - - - - The macro takes an conditional expression to evaluate. - When CONFIG_SND_DEBUG, is set, if the - expression is non-zero, it shows the warning message such as - BUG? (xxx) - normally followed by stack trace. - - In both cases it returns the evaluated value. - - -
- - - - - - - - - Acknowledgments - - I would like to thank Phil Kerr for his help for improvement and - corrections of this document. - - - Kevin Conder reformatted the original plain-text to the - DocBook format. - - - Giuliano Pochini corrected typos and contributed the example codes - in the hardware constraints section. - - - diff --git a/Documentation/sound/kernel-api/index.rst b/Documentation/sound/kernel-api/index.rst index 73c13497dec7..d0e6df35b4b4 100644 --- a/Documentation/sound/kernel-api/index.rst +++ b/Documentation/sound/kernel-api/index.rst @@ -5,3 +5,4 @@ ALSA Kernel API Documentation :maxdepth: 2 alsa-driver-api + writing-an-alsa-driver diff --git a/Documentation/sound/kernel-api/writing-an-alsa-driver.rst b/Documentation/sound/kernel-api/writing-an-alsa-driver.rst new file mode 100644 index 000000000000..95c5443eff38 --- /dev/null +++ b/Documentation/sound/kernel-api/writing-an-alsa-driver.rst @@ -0,0 +1,4219 @@ +====================== +Writing an ALSA Driver +====================== + +:Author: Takashi Iwai +:Date: Oct 15, 2007 +:Edition: 0.3.7 + +Preface +======= + +This document describes how to write an `ALSA (Advanced Linux Sound +Architecture) `__ driver. The document +focuses mainly on PCI soundcards. In the case of other device types, the +API might be different, too. However, at least the ALSA kernel API is +consistent, and therefore it would be still a bit help for writing them. + +This document targets people who already have enough C language skills +and have basic linux kernel programming knowledge. This document doesn't +explain the general topic of linux kernel coding and doesn't cover +low-level driver implementation details. It only describes the standard +way to write a PCI sound driver on ALSA. + +If you are already familiar with the older ALSA ver.0.5.x API, you can +check the drivers such as ``sound/pci/es1938.c`` or +``sound/pci/maestro3.c`` which have also almost the same code-base in +the ALSA 0.5.x tree, so you can compare the differences. + +This document is still a draft version. Any feedback and corrections, +please!! + +File Tree Structure +=================== + +General +------- + +The ALSA drivers are provided in two ways. + +One is the trees provided as a tarball or via cvs from the ALSA's ftp +site, and another is the 2.6 (or later) Linux kernel tree. To +synchronize both, the ALSA driver tree is split into two different +trees: alsa-kernel and alsa-driver. The former contains purely the +source code for the Linux 2.6 (or later) tree. This tree is designed +only for compilation on 2.6 or later environment. The latter, +alsa-driver, contains many subtle files for compiling ALSA drivers +outside of the Linux kernel tree, wrapper functions for older 2.2 and +2.4 kernels, to adapt the latest kernel API, and additional drivers +which are still in development or in tests. The drivers in alsa-driver +tree will be moved to alsa-kernel (and eventually to the 2.6 kernel +tree) when they are finished and confirmed to work fine. + +The file tree structure of ALSA driver is depicted below. Both +alsa-kernel and alsa-driver have almost the same file structure, except +for “core” directory. It's named as “acore” in alsa-driver tree. + +:: + + sound + /core + /oss + /seq + /oss + /instr + /ioctl32 + /include + /drivers + /mpu401 + /opl3 + /i2c + /l3 + /synth + /emux + /pci + /(cards) + /isa + /(cards) + /arm + /ppc + /sparc + /usb + /pcmcia /(cards) + /oss + + +core directory +-------------- + +This directory contains the middle layer which is the heart of ALSA +drivers. In this directory, the native ALSA modules are stored. The +sub-directories contain different modules and are dependent upon the +kernel config. + +core/oss +~~~~~~~~ + +The codes for PCM and mixer OSS emulation modules are stored in this +directory. The rawmidi OSS emulation is included in the ALSA rawmidi +code since it's quite small. The sequencer code is stored in +``core/seq/oss`` directory (see `below <#core-seq-oss>`__). + +core/ioctl32 +~~~~~~~~~~~~ + +This directory contains the 32bit-ioctl wrappers for 64bit architectures +such like x86-64, ppc64 and sparc64. For 32bit and alpha architectures, +these are not compiled. + +core/seq +~~~~~~~~ + +This directory and its sub-directories are for the ALSA sequencer. This +directory contains the sequencer core and primary sequencer modules such +like snd-seq-midi, snd-seq-virmidi, etc. They are compiled only when +``CONFIG_SND_SEQUENCER`` is set in the kernel config. + +core/seq/oss +~~~~~~~~~~~~ + +This contains the OSS sequencer emulation codes. + +core/seq/instr +~~~~~~~~~~~~~~ + +This directory contains the modules for the sequencer instrument layer. + +include directory +----------------- + +This is the place for the public header files of ALSA drivers, which are +to be exported to user-space, or included by several files at different +directories. Basically, the private header files should not be placed in +this directory, but you may still find files there, due to historical +reasons :) + +drivers directory +----------------- + +This directory contains code shared among different drivers on different +architectures. They are hence supposed not to be architecture-specific. +For example, the dummy pcm driver and the serial MIDI driver are found +in this directory. In the sub-directories, there is code for components +which are independent from bus and cpu architectures. + +drivers/mpu401 +~~~~~~~~~~~~~~ + +The MPU401 and MPU401-UART modules are stored here. + +drivers/opl3 and opl4 +~~~~~~~~~~~~~~~~~~~~~ + +The OPL3 and OPL4 FM-synth stuff is found here. + +i2c directory +------------- + +This contains the ALSA i2c components. + +Although there is a standard i2c layer on Linux, ALSA has its own i2c +code for some cards, because the soundcard needs only a simple operation +and the standard i2c API is too complicated for such a purpose. + +i2c/l3 +~~~~~~ + +This is a sub-directory for ARM L3 i2c. + +synth directory +--------------- + +This contains the synth middle-level modules. + +So far, there is only Emu8000/Emu10k1 synth driver under the +``synth/emux`` sub-directory. + +pci directory +------------- + +This directory and its sub-directories hold the top-level card modules +for PCI soundcards and the code specific to the PCI BUS. + +The drivers compiled from a single file are stored directly in the pci +directory, while the drivers with several source files are stored on +their own sub-directory (e.g. emu10k1, ice1712). + +isa directory +------------- + +This directory and its sub-directories hold the top-level card modules +for ISA soundcards. + +arm, ppc, and sparc directories +------------------------------- + +They are used for top-level card modules which are specific to one of +these architectures. + +usb directory +------------- + +This directory contains the USB-audio driver. In the latest version, the +USB MIDI driver is integrated in the usb-audio driver. + +pcmcia directory +---------------- + +The PCMCIA, especially PCCard drivers will go here. CardBus drivers will +be in the pci directory, because their API is identical to that of +standard PCI cards. + +oss directory +------------- + +The OSS/Lite source files are stored here in Linux 2.6 (or later) tree. +In the ALSA driver tarball, this directory is empty, of course :) + +Basic Flow for PCI Drivers +========================== + +Outline +------- + +The minimum flow for PCI soundcards is as follows: + +- define the PCI ID table (see the section `PCI Entries`_). + +- create ``probe`` callback. + +- create ``remove`` callback. + +- create a :c:type:`struct pci_driver ` structure + containing the three pointers above. + +- create an ``init`` function just calling the + :c:func:`pci_register_driver()` to register the pci_driver + table defined above. + +- create an ``exit`` function to call the + :c:func:`pci_unregister_driver()` function. + +Full Code Example +----------------- + +The code example is shown below. Some parts are kept unimplemented at +this moment but will be filled in the next sections. The numbers in the +comment lines of the :c:func:`snd_mychip_probe()` function refer +to details explained in the following section. + +:: + + #include + #include + #include + #include + #include + + /* module parameters (see "Module Parameters") */ + /* SNDRV_CARDS: maximum number of cards supported by this module */ + static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; + static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; + static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; + + /* definition of the chip-specific record */ + struct mychip { + struct snd_card *card; + /* the rest of the implementation will be in section + * "PCI Resource Management" + */ + }; + + /* chip-specific destructor + * (see "PCI Resource Management") + */ + static int snd_mychip_free(struct mychip *chip) + { + .... /* will be implemented later... */ + } + + /* component-destructor + * (see "Management of Cards and Components") + */ + static int snd_mychip_dev_free(struct snd_device *device) + { + return snd_mychip_free(device->device_data); + } + + /* chip-specific constructor + * (see "Management of Cards and Components") + */ + static int snd_mychip_create(struct snd_card *card, + struct pci_dev *pci, + struct mychip **rchip) + { + struct mychip *chip; + int err; + static struct snd_device_ops ops = { + .dev_free = snd_mychip_dev_free, + }; + + *rchip = NULL; + + /* check PCI availability here + * (see "PCI Resource Management") + */ + .... + + /* allocate a chip-specific data with zero filled */ + chip = kzalloc(sizeof(*chip), GFP_KERNEL); + if (chip == NULL) + return -ENOMEM; + + chip->card = card; + + /* rest of initialization here; will be implemented + * later, see "PCI Resource Management" + */ + .... + + err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops); + if (err < 0) { + snd_mychip_free(chip); + return err; + } + + *rchip = chip; + return 0; + } + + /* constructor -- see "Driver Constructor" sub-section */ + static int snd_mychip_probe(struct pci_dev *pci, + const struct pci_device_id *pci_id) + { + static int dev; + struct snd_card *card; + struct mychip *chip; + int err; + + /* (1) */ + if (dev >= SNDRV_CARDS) + return -ENODEV; + if (!enable[dev]) { + dev++; + return -ENOENT; + } + + /* (2) */ + err = snd_card_new(&pci->dev, index[dev], id[dev], THIS_MODULE, + 0, &card); + if (err < 0) + return err; + + /* (3) */ + err = snd_mychip_create(card, pci, &chip); + if (err < 0) { + snd_card_free(card); + return err; + } + + /* (4) */ + strcpy(card->driver, "My Chip"); + strcpy(card->shortname, "My Own Chip 123"); + sprintf(card->longname, "%s at 0x%lx irq %i", + card->shortname, chip->ioport, chip->irq); + + /* (5) */ + .... /* implemented later */ + + /* (6) */ + err = snd_card_register(card); + if (err < 0) { + snd_card_free(card); + return err; + } + + /* (7) */ + pci_set_drvdata(pci, card); + dev++; + return 0; + } + + /* destructor -- see the "Destructor" sub-section */ + static void snd_mychip_remove(struct pci_dev *pci) + { + snd_card_free(pci_get_drvdata(pci)); + pci_set_drvdata(pci, NULL); + } + + + +Driver Constructor +------------------ + +The real constructor of PCI drivers is the ``probe`` callback. The +``probe`` callback and other component-constructors which are called +from the ``probe`` callback cannot be used with the ``__init`` prefix +because any PCI device could be a hotplug device. + +In the ``probe`` callback, the following scheme is often used. + +1) Check and increment the device index. +~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + +:: + + static int dev; + .... + if (dev >= SNDRV_CARDS) + return -ENODEV; + if (!enable[dev]) { + dev++; + return -ENOENT; + } + + +where ``enable[dev]`` is the module option. + +Each time the ``probe`` callback is called, check the availability of +the device. If not available, simply increment the device index and +returns. dev will be incremented also later (`step 7 +<#set-the-pci-driver-data-and-return-zero>`__). + +2) Create a card instance +~~~~~~~~~~~~~~~~~~~~~~~~~ + +:: + + struct snd_card *card; + int err; + .... + err = snd_card_new(&pci->dev, index[dev], id[dev], THIS_MODULE, + 0, &card); + + +The details will be explained in the section `Management of Cards and +Components`_. + +3) Create a main component +~~~~~~~~~~~~~~~~~~~~~~~~~~ + +In this part, the PCI resources are allocated. + +:: + + struct mychip *chip; + .... + err = snd_mychip_create(card, pci, &chip); + if (err < 0) { + snd_card_free(card); + return err; + } + +The details will be explained in the section `PCI Resource +Management`_. + +4) Set the driver ID and name strings. +~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + +:: + + strcpy(card->driver, "My Chip"); + strcpy(card->shortname, "My Own Chip 123"); + sprintf(card->longname, "%s at 0x%lx irq %i", + card->shortname, chip->ioport, chip->irq); + +The driver field holds the minimal ID string of the chip. This is used +by alsa-lib's configurator, so keep it simple but unique. Even the +same driver can have different driver IDs to distinguish the +functionality of each chip type. + +The shortname field is a string shown as more verbose name. The longname +field contains the information shown in ``/proc/asound/cards``. + +5) Create other components, such as mixer, MIDI, etc. +~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + +Here you define the basic components such as `PCM <#PCM-Interface>`__, +mixer (e.g. `AC97 <#API-for-AC97-Codec>`__), MIDI (e.g. +`MPU-401 <#MIDI-MPU401-UART-Interface>`__), and other interfaces. +Also, if you want a `proc file <#Proc-Interface>`__, define it here, +too. + +6) Register the card instance. +~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + +:: + + err = snd_card_register(card); + if (err < 0) { + snd_card_free(card); + return err; + } + +Will be explained in the section `Management of Cards and +Components`_, too. + +7) Set the PCI driver data and return zero. +~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + +:: + + pci_set_drvdata(pci, card); + dev++; + return 0; + +In the above, the card record is stored. This pointer is used in the +remove callback and power-management callbacks, too. + +Destructor +---------- + +The destructor, remove callback, simply releases the card instance. Then +the ALSA middle layer will release all the attached components +automatically. + +It would be typically like the following: + +:: + + static void snd_mychip_remove(struct pci_dev *pci) + { + snd_card_free(pci_get_drvdata(pci)); + pci_set_drvdata(pci, NULL); + } + + +The above code assumes that the card pointer is set to the PCI driver +data. + +Header Files +------------ + +For the above example, at least the following include files are +necessary. + +:: + + #include + #include + #include + #include + #include + +where the last one is necessary only when module options are defined +in the source file. If the code is split into several files, the files +without module options don't need them. + +In addition to these headers, you'll need ```` for +interrupt handling, and ```` for I/O access. If you use the +:c:func:`mdelay()` or :c:func:`udelay()` functions, you'll need +to include ```` too. + +The ALSA interfaces like the PCM and control APIs are defined in other +```` header files. They have to be included after +````. + +Management of Cards and Components +================================== + +Card Instance +------------- + +For each soundcard, a “card” record must be allocated. + +A card record is the headquarters of the soundcard. It manages the whole +list of devices (components) on the soundcard, such as PCM, mixers, +MIDI, synthesizer, and so on. Also, the card record holds the ID and the +name strings of the card, manages the root of proc files, and controls +the power-management states and hotplug disconnections. The component +list on the card record is used to manage the correct release of +resources at destruction. + +As mentioned above, to create a card instance, call +:c:func:`snd_card_new()`. + +:: + + struct snd_card *card; + int err; + err = snd_card_new(&pci->dev, index, id, module, extra_size, &card); + + +The function takes six arguments: the parent device pointer, the +card-index number, the id string, the module pointer (usually +``THIS_MODULE``), the size of extra-data space, and the pointer to +return the card instance. The extra_size argument is used to allocate +card->private_data for the chip-specific data. Note that these data are +allocated by :c:func:`snd_card_new()`. + +The first argument, the pointer of struct :c:type:`struct device +`, specifies the parent device. For PCI devices, typically +``&pci->`` is passed there. + +Components +---------- + +After the card is created, you can attach the components (devices) to +the card instance. In an ALSA driver, a component is represented as a +:c:type:`struct snd_device ` object. A component +can be a PCM instance, a control interface, a raw MIDI interface, etc. +Each such instance has one component entry. + +A component can be created via :c:func:`snd_device_new()` +function. + +:: + + snd_device_new(card, SNDRV_DEV_XXX, chip, &ops); + +This takes the card pointer, the device-level (``SNDRV_DEV_XXX``), the +data pointer, and the callback pointers (``&ops``). The device-level +defines the type of components and the order of registration and +de-registration. For most components, the device-level is already +defined. For a user-defined component, you can use +``SNDRV_DEV_LOWLEVEL``. + +This function itself doesn't allocate the data space. The data must be +allocated manually beforehand, and its pointer is passed as the +argument. This pointer (``chip`` in the above example) is used as the +identifier for the instance. + +Each pre-defined ALSA component such as ac97 and pcm calls +:c:func:`snd_device_new()` inside its constructor. The destructor +for each component is defined in the callback pointers. Hence, you don't +need to take care of calling a destructor for such a component. + +If you wish to create your own component, you need to set the destructor +function to the dev_free callback in the ``ops``, so that it can be +released automatically via :c:func:`snd_card_free()`. The next +example will show an implementation of chip-specific data. + +Chip-Specific Data +------------------ + +Chip-specific information, e.g. the I/O port address, its resource +pointer, or the irq number, is stored in the chip-specific record. + +:: + + struct mychip { + .... + }; + + +In general, there are two ways of allocating the chip record. + +1. Allocating via :c:func:`snd_card_new()`. +~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + +As mentioned above, you can pass the extra-data-length to the 5th +argument of :c:func:`snd_card_new()`, i.e. + +:: + + err = snd_card_new(&pci->dev, index[dev], id[dev], THIS_MODULE, + sizeof(struct mychip), &card); + +:c:type:`struct mychip ` is the type of the chip record. + +In return, the allocated record can be accessed as + +:: + + struct mychip *chip = card->private_data; + +With this method, you don't have to allocate twice. The record is +released together with the card instance. + +2. Allocating an extra device. +~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + +After allocating a card instance via :c:func:`snd_card_new()` +(with ``0`` on the 4th arg), call :c:func:`kzalloc()`. + +:: + + struct snd_card *card; + struct mychip *chip; + err = snd_card_new(&pci->dev, index[dev], id[dev], THIS_MODULE, + 0, &card); + ..... + chip = kzalloc(sizeof(*chip), GFP_KERNEL); + +The chip record should have the field to hold the card pointer at least, + +:: + + struct mychip { + struct snd_card *card; + .... + }; + + +Then, set the card pointer in the returned chip instance. + +:: + + chip->card = card; + +Next, initialize the fields, and register this chip record as a +low-level device with a specified ``ops``, + +:: + + static struct snd_device_ops ops = { + .dev_free = snd_mychip_dev_free, + }; + .... + snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops); + +:c:func:`snd_mychip_dev_free()` is the device-destructor +function, which will call the real destructor. + +:: + + static int snd_mychip_dev_free(struct snd_device *device) + { + return snd_mychip_free(device->device_data); + } + +where :c:func:`snd_mychip_free()` is the real destructor. + +Registration and Release +------------------------ + +After all components are assigned, register the card instance by calling +:c:func:`snd_card_register()`. Access to the device files is +enabled at this point. That is, before +:c:func:`snd_card_register()` is called, the components are safely +inaccessible from external side. If this call fails, exit the probe +function after releasing the card via :c:func:`snd_card_free()`. + +For releasing the card instance, you can call simply +:c:func:`snd_card_free()`. As mentioned earlier, all components +are released automatically by this call. + +For a device which allows hotplugging, you can use +:c:func:`snd_card_free_when_closed()`. This one will postpone +the destruction until all devices are closed. + +PCI Resource Management +======================= + +Full Code Example +----------------- + +In this section, we'll complete the chip-specific constructor, +destructor and PCI entries. Example code is shown first, below. + +:: + + struct mychip { + struct snd_card *card; + struct pci_dev *pci; + + unsigned long port; + int irq; + }; + + static int snd_mychip_free(struct mychip *chip) + { + /* disable hardware here if any */ + .... /* (not implemented in this document) */ + + /* release the irq */ + if (chip->irq >= 0) + free_irq(chip->irq, chip); + /* release the I/O ports & memory */ + pci_release_regions(chip->pci); + /* disable the PCI entry */ + pci_disable_device(chip->pci); + /* release the data */ + kfree(chip); + return 0; + } + + /* chip-specific constructor */ + static int snd_mychip_create(struct snd_card *card, + struct pci_dev *pci, + struct mychip **rchip) + { + struct mychip *chip; + int err; + static struct snd_device_ops ops = { + .dev_free = snd_mychip_dev_free, + }; + + *rchip = NULL; + + /* initialize the PCI entry */ + err = pci_enable_device(pci); + if (err < 0) + return err; + /* check PCI availability (28bit DMA) */ + if (pci_set_dma_mask(pci, DMA_BIT_MASK(28)) < 0 || + pci_set_consistent_dma_mask(pci, DMA_BIT_MASK(28)) < 0) { + printk(KERN_ERR "error to set 28bit mask DMA\n"); + pci_disable_device(pci); + return -ENXIO; + } + + chip = kzalloc(sizeof(*chip), GFP_KERNEL); + if (chip == NULL) { + pci_disable_device(pci); + return -ENOMEM; + } + + /* initialize the stuff */ + chip->card = card; + chip->pci = pci; + chip->irq = -1; + + /* (1) PCI resource allocation */ + err = pci_request_regions(pci, "My Chip"); + if (err < 0) { + kfree(chip); + pci_disable_device(pci); + return err; + } + chip->port = pci_resource_start(pci, 0); + if (request_irq(pci->irq, snd_mychip_interrupt, + IRQF_SHARED, KBUILD_MODNAME, chip)) { + printk(KERN_ERR "cannot grab irq %d\n", pci->irq); + snd_mychip_free(chip); + return -EBUSY; + } + chip->irq = pci->irq; + + /* (2) initialization of the chip hardware */ + .... /* (not implemented in this document) */ + + err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops); + if (err < 0) { + snd_mychip_free(chip); + return err; + } + + *rchip = chip; + return 0; + } + + /* PCI IDs */ + static struct pci_device_id snd_mychip_ids[] = { + { PCI_VENDOR_ID_FOO, PCI_DEVICE_ID_BAR, + PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, }, + .... + { 0, } + }; + MODULE_DEVICE_TABLE(pci, snd_mychip_ids); + + /* pci_driver definition */ + static struct pci_driver driver = { + .name = KBUILD_MODNAME, + .id_table = snd_mychip_ids, + .probe = snd_mychip_probe, + .remove = snd_mychip_remove, + }; + + /* module initialization */ + static int __init alsa_card_mychip_init(void) + { + return pci_register_driver(&driver); + } + + /* module clean up */ + static void __exit alsa_card_mychip_exit(void) + { + pci_unregister_driver(&driver); + } + + module_init(alsa_card_mychip_init) + module_exit(alsa_card_mychip_exit) + + EXPORT_NO_SYMBOLS; /* for old kernels only */ + +Some Hafta's +------------ + +The allocation of PCI resources is done in the ``probe`` function, and +usually an extra :c:func:`xxx_create()` function is written for this +purpose. + +In the case of PCI devices, you first have to call the +:c:func:`pci_enable_device()` function before allocating +resources. Also, you need to set the proper PCI DMA mask to limit the +accessed I/O range. In some cases, you might need to call +:c:func:`pci_set_master()` function, too. + +Suppose the 28bit mask, and the code to be added would be like: + +:: + + err = pci_enable_device(pci); + if (err < 0) + return err; + if (pci_set_dma_mask(pci, DMA_BIT_MASK(28)) < 0 || + pci_set_consistent_dma_mask(pci, DMA_BIT_MASK(28)) < 0) { + printk(KERN_ERR "error to set 28bit mask DMA\n"); + pci_disable_device(pci); + return -ENXIO; + } + + +Resource Allocation +------------------- + +The allocation of I/O ports and irqs is done via standard kernel +functions. Unlike ALSA ver.0.5.x., there are no helpers for that. And +these resources must be released in the destructor function (see below). +Also, on ALSA 0.9.x, you don't need to allocate (pseudo-)DMA for PCI +like in ALSA 0.5.x. + +Now assume that the PCI device has an I/O port with 8 bytes and an +interrupt. Then :c:type:`struct mychip ` will have the +following fields: + +:: + + struct mychip { + struct snd_card *card; + + unsigned long port; + int irq; + }; + + +For an I/O port (and also a memory region), you need to have the +resource pointer for the standard resource management. For an irq, you +have to keep only the irq number (integer). But you need to initialize +this number as -1 before actual allocation, since irq 0 is valid. The +port address and its resource pointer can be initialized as null by +:c:func:`kzalloc()` automatically, so you don't have to take care of +resetting them. + +The allocation of an I/O port is done like this: + +:: + + err = pci_request_regions(pci, "My Chip"); + if (err < 0) { + kfree(chip); + pci_disable_device(pci); + return err; + } + chip->port = pci_resource_start(pci, 0); + +It will reserve the I/O port region of 8 bytes of the given PCI device. +The returned value, ``chip->res_port``, is allocated via +:c:func:`kmalloc()` by :c:func:`request_region()`. The pointer +must be released via :c:func:`kfree()`, but there is a problem with +this. This issue will be explained later. + +The allocation of an interrupt source is done like this: + +:: + + if (request_irq(pci->irq, snd_mychip_interrupt, + IRQF_SHARED, KBUILD_MODNAME, chip)) { + printk(KERN_ERR "cannot grab irq %d\n", pci->irq); + snd_mychip_free(chip); + return -EBUSY; + } + chip->irq = pci->irq; + +where :c:func:`snd_mychip_interrupt()` is the interrupt handler +defined `later <#pcm-interface-interrupt-handler>`__. Note that +``chip->irq`` should be defined only when :c:func:`request_irq()` +succeeded. + +On the PCI bus, interrupts can be shared. Thus, ``IRQF_SHARED`` is used +as the interrupt flag of :c:func:`request_irq()`. + +The last argument of :c:func:`request_irq()` is the data pointer +passed to the interrupt handler. Usually, the chip-specific record is +used for that, but you can use what you like, too. + +I won't give details about the interrupt handler at this point, but at +least its appearance can be explained now. The interrupt handler looks +usually like the following: + +:: + + static irqreturn_t snd_mychip_interrupt(int irq, void *dev_id) + { + struct mychip *chip = dev_id; + .... + return IRQ_HANDLED; + } + + +Now let's write the corresponding destructor for the resources above. +The role of destructor is simple: disable the hardware (if already +activated) and release the resources. So far, we have no hardware part, +so the disabling code is not written here. + +To release the resources, the “check-and-release” method is a safer way. +For the interrupt, do like this: + +:: + + if (chip->irq >= 0) + free_irq(chip->irq, chip); + +Since the irq number can start from 0, you should initialize +``chip->irq`` with a negative value (e.g. -1), so that you can check +the validity of the irq number as above. + +When you requested I/O ports or memory regions via +:c:func:`pci_request_region()` or +:c:func:`pci_request_regions()` like in this example, release the +resource(s) using the corresponding function, +:c:func:`pci_release_region()` or +:c:func:`pci_release_regions()`. + +:: + + pci_release_regions(chip->pci); + +When you requested manually via :c:func:`request_region()` or +:c:func:`request_mem_region()`, you can release it via +:c:func:`release_resource()`. Suppose that you keep the resource +pointer returned from :c:func:`request_region()` in +chip->res_port, the release procedure looks like: + +:: + + release_and_free_resource(chip->res_port); + +Don't forget to call :c:func:`pci_disable_device()` before the +end. + +And finally, release the chip-specific record. + +:: + + kfree(chip); + +We didn't implement the hardware disabling part in the above. If you +need to do this, please note that the destructor may be called even +before the initialization of the chip is completed. It would be better +to have a flag to skip hardware disabling if the hardware was not +initialized yet. + +When the chip-data is assigned to the card using +:c:func:`snd_device_new()` with ``SNDRV_DEV_LOWLELVEL`` , its +destructor is called at the last. That is, it is assured that all other +components like PCMs and controls have already been released. You don't +have to stop PCMs, etc. explicitly, but just call low-level hardware +stopping. + +The management of a memory-mapped region is almost as same as the +management of an I/O port. You'll need three fields like the +following: + +:: + + struct mychip { + .... + unsigned long iobase_phys; + void __iomem *iobase_virt; + }; + +and the allocation would be like below: + +:: + + if ((err = pci_request_regions(pci, "My Chip")) < 0) { + kfree(chip); + return err; + } + chip->iobase_phys = pci_resource_start(pci, 0); + chip->iobase_virt = ioremap_nocache(chip->iobase_phys, + pci_resource_len(pci, 0)); + +and the corresponding destructor would be: + +:: + + static int snd_mychip_free(struct mychip *chip) + { + .... + if (chip->iobase_virt) + iounmap(chip->iobase_virt); + .... + pci_release_regions(chip->pci); + .... + } + +PCI Entries +----------- + +So far, so good. Let's finish the missing PCI stuff. At first, we need a +:c:type:`struct pci_device_id ` table for +this chipset. It's a table of PCI vendor/device ID number, and some +masks. + +For example, + +:: + + static struct pci_device_id snd_mychip_ids[] = { + { PCI_VENDOR_ID_FOO, PCI_DEVICE_ID_BAR, + PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, }, + .... + { 0, } + }; + MODULE_DEVICE_TABLE(pci, snd_mychip_ids); + +The first and second fields of the :c:type:`struct pci_device_id +` structure are the vendor and device IDs. If you +have no reason to filter the matching devices, you can leave the +remaining fields as above. The last field of the :c:type:`struct +pci_device_id ` struct contains private data +for this entry. You can specify any value here, for example, to define +specific operations for supported device IDs. Such an example is found +in the intel8x0 driver. + +The last entry of this list is the terminator. You must specify this +all-zero entry. + +Then, prepare the :c:type:`struct pci_driver ` +record: + +:: + + static struct pci_driver driver = { + .name = KBUILD_MODNAME, + .id_table = snd_mychip_ids, + .probe = snd_mychip_probe, + .remove = snd_mychip_remove, + }; + +The ``probe`` and ``remove`` functions have already been defined in +the previous sections. The ``name`` field is the name string of this +device. Note that you must not use a slash “/” in this string. + +And at last, the module entries: + +:: + + static int __init alsa_card_mychip_init(void) + { + return pci_register_driver(&driver); + } + + static void __exit alsa_card_mychip_exit(void) + { + pci_unregister_driver(&driver); + } + + module_init(alsa_card_mychip_init) + module_exit(alsa_card_mychip_exit) + +Note that these module entries are tagged with ``__init`` and ``__exit`` +prefixes. + +Oh, one thing was forgotten. If you have no exported symbols, you need +to declare it in 2.2 or 2.4 kernels (it's not necessary in 2.6 kernels). + +:: + + EXPORT_NO_SYMBOLS; + +That's all! + +PCM Interface +============= + +General +------- + +The PCM middle layer of ALSA is quite powerful and it is only necessary +for each driver to implement the low-level functions to access its +hardware. + +For accessing to the PCM layer, you need to include ```` +first. In addition, ```` might be needed if you +access to some functions related with hw_param. + +Each card device can have up to four pcm instances. A pcm instance +corresponds to a pcm device file. The limitation of number of instances +comes only from the available bit size of the Linux's device numbers. +Once when 64bit device number is used, we'll have more pcm instances +available. + +A pcm instance consists of pcm playback and capture streams, and each +pcm stream consists of one or more pcm substreams. Some soundcards +support multiple playback functions. For example, emu10k1 has a PCM +playback of 32 stereo substreams. In this case, at each open, a free +substream is (usually) automatically chosen and opened. Meanwhile, when +only one substream exists and it was already opened, the successful open +will either block or error with ``EAGAIN`` according to the file open +mode. But you don't have to care about such details in your driver. The +PCM middle layer will take care of such work. + +Full Code Example +----------------- + +The example code below does not include any hardware access routines but +shows only the skeleton, how to build up the PCM interfaces. + +:: + + #include + .... + + /* hardware definition */ + static struct snd_pcm_hardware snd_mychip_playback_hw = { + .info = (SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_MMAP_VALID), + .formats = SNDRV_PCM_FMTBIT_S16_LE, + .rates = SNDRV_PCM_RATE_8000_48000, + .rate_min = 8000, + .rate_max = 48000, + .channels_min = 2, + .channels_max = 2, + .buffer_bytes_max = 32768, + .period_bytes_min = 4096, + .period_bytes_max = 32768, + .periods_min = 1, + .periods_max = 1024, + }; + + /* hardware definition */ + static struct snd_pcm_hardware snd_mychip_capture_hw = { + .info = (SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_MMAP_VALID), + .formats = SNDRV_PCM_FMTBIT_S16_LE, + .rates = SNDRV_PCM_RATE_8000_48000, + .rate_min = 8000, + .rate_max = 48000, + .channels_min = 2, + .channels_max = 2, + .buffer_bytes_max = 32768, + .period_bytes_min = 4096, + .period_bytes_max = 32768, + .periods_min = 1, + .periods_max = 1024, + }; + + /* open callback */ + static int snd_mychip_playback_open(struct snd_pcm_substream *substream) + { + struct mychip *chip = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + + runtime->hw = snd_mychip_playback_hw; + /* more hardware-initialization will be done here */ + .... + return 0; + } + + /* close callback */ + static int snd_mychip_playback_close(struct snd_pcm_substream *substream) + { + struct mychip *chip = snd_pcm_substream_chip(substream); + /* the hardware-specific codes will be here */ + .... + return 0; + + } + + /* open callback */ + static int snd_mychip_capture_open(struct snd_pcm_substream *substream) + { + struct mychip *chip = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + + runtime->hw = snd_mychip_capture_hw; + /* more hardware-initialization will be done here */ + .... + return 0; + } + + /* close callback */ + static int snd_mychip_capture_close(struct snd_pcm_substream *substream) + { + struct mychip *chip = snd_pcm_substream_chip(substream); + /* the hardware-specific codes will be here */ + .... + return 0; + + } + + /* hw_params callback */ + static int snd_mychip_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) + { + return snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(hw_params)); + } + + /* hw_free callback */ + static int snd_mychip_pcm_hw_free(struct snd_pcm_substream *substream) + { + return snd_pcm_lib_free_pages(substream); + } + + /* prepare callback */ + static int snd_mychip_pcm_prepare(struct snd_pcm_substream *substream) + { + struct mychip *chip = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + + /* set up the hardware with the current configuration + * for example... + */ + mychip_set_sample_format(chip, runtime->format); + mychip_set_sample_rate(chip, runtime->rate); + mychip_set_channels(chip, runtime->channels); + mychip_set_dma_setup(chip, runtime->dma_addr, + chip->buffer_size, + chip->period_size); + return 0; + } + + /* trigger callback */ + static int snd_mychip_pcm_trigger(struct snd_pcm_substream *substream, + int cmd) + { + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + /* do something to start the PCM engine */ + .... + break; + case SNDRV_PCM_TRIGGER_STOP: + /* do something to stop the PCM engine */ + .... + break; + default: + return -EINVAL; + } + } + + /* pointer callback */ + static snd_pcm_uframes_t + snd_mychip_pcm_pointer(struct snd_pcm_substream *substream) + { + struct mychip *chip = snd_pcm_substream_chip(substream); + unsigned int current_ptr; + + /* get the current hardware pointer */ + current_ptr = mychip_get_hw_pointer(chip); + return current_ptr; + } + + /* operators */ + static struct snd_pcm_ops snd_mychip_playback_ops = { + .open = snd_mychip_playback_open, + .close = snd_mychip_playback_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = snd_mychip_pcm_hw_params, + .hw_free = snd_mychip_pcm_hw_free, + .prepare = snd_mychip_pcm_prepare, + .trigger = snd_mychip_pcm_trigger, + .pointer = snd_mychip_pcm_pointer, + }; + + /* operators */ + static struct snd_pcm_ops snd_mychip_capture_ops = { + .open = snd_mychip_capture_open, + .close = snd_mychip_capture_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = snd_mychip_pcm_hw_params, + .hw_free = snd_mychip_pcm_hw_free, + .prepare = snd_mychip_pcm_prepare, + .trigger = snd_mychip_pcm_trigger, + .pointer = snd_mychip_pcm_pointer, + }; + + /* + * definitions of capture are omitted here... + */ + + /* create a pcm device */ + static int snd_mychip_new_pcm(struct mychip *chip) + { + struct snd_pcm *pcm; + int err; + + err = snd_pcm_new(chip->card, "My Chip", 0, 1, 1, &pcm); + if (err < 0) + return err; + pcm->private_data = chip; + strcpy(pcm->name, "My Chip"); + chip->pcm = pcm; + /* set operators */ + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, + &snd_mychip_playback_ops); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, + &snd_mychip_capture_ops); + /* pre-allocation of buffers */ + /* NOTE: this may fail */ + snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, + snd_dma_pci_data(chip->pci), + 64*1024, 64*1024); + return 0; + } + + +PCM Constructor +--------------- + +A pcm instance is allocated by the :c:func:`snd_pcm_new()` +function. It would be better to create a constructor for pcm, namely, + +:: + + static int snd_mychip_new_pcm(struct mychip *chip) + { + struct snd_pcm *pcm; + int err; + + err = snd_pcm_new(chip->card, "My Chip", 0, 1, 1, &pcm); + if (err < 0) + return err; + pcm->private_data = chip; + strcpy(pcm->name, "My Chip"); + chip->pcm = pcm; + .... + return 0; + } + +The :c:func:`snd_pcm_new()` function takes four arguments. The +first argument is the card pointer to which this pcm is assigned, and +the second is the ID string. + +The third argument (``index``, 0 in the above) is the index of this new +pcm. It begins from zero. If you create more than one pcm instances, +specify the different numbers in this argument. For example, ``index = +1`` for the second PCM device. + +The fourth and fifth arguments are the number of substreams for playback +and capture, respectively. Here 1 is used for both arguments. When no +playback or capture substreams are available, pass 0 to the +corresponding argument. + +If a chip supports multiple playbacks or captures, you can specify more +numbers, but they must be handled properly in open/close, etc. +callbacks. When you need to know which substream you are referring to, +then it can be obtained from :c:type:`struct snd_pcm_substream +` data passed to each callback as follows: + +:: + + struct snd_pcm_substream *substream; + int index = substream->number; + + +After the pcm is created, you need to set operators for each pcm stream. + +:: + + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, + &snd_mychip_playback_ops); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, + &snd_mychip_capture_ops); + +The operators are defined typically like this: + +:: + + static struct snd_pcm_ops snd_mychip_playback_ops = { + .open = snd_mychip_pcm_open, + .close = snd_mychip_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = snd_mychip_pcm_hw_params, + .hw_free = snd_mychip_pcm_hw_free, + .prepare = snd_mychip_pcm_prepare, + .trigger = snd_mychip_pcm_trigger, + .pointer = snd_mychip_pcm_pointer, + }; + +All the callbacks are described in the Operators_ subsection. + +After setting the operators, you probably will want to pre-allocate the +buffer. For the pre-allocation, simply call the following: + +:: + + snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, + snd_dma_pci_data(chip->pci), + 64*1024, 64*1024); + +It will allocate a buffer up to 64kB as default. Buffer management +details will be described in the later section `Buffer and Memory +Management`_. + +Additionally, you can set some extra information for this pcm in +``pcm->info_flags``. The available values are defined as +``SNDRV_PCM_INFO_XXX`` in ````, which is used for the +hardware definition (described later). When your soundchip supports only +half-duplex, specify like this: + +:: + + pcm->info_flags = SNDRV_PCM_INFO_HALF_DUPLEX; + + +... And the Destructor? +----------------------- + +The destructor for a pcm instance is not always necessary. Since the pcm +device will be released by the middle layer code automatically, you +don't have to call the destructor explicitly. + +The destructor would be necessary if you created special records +internally and needed to release them. In such a case, set the +destructor function to ``pcm->private_free``: + +:: + + static void mychip_pcm_free(struct snd_pcm *pcm) + { + struct mychip *chip = snd_pcm_chip(pcm); + /* free your own data */ + kfree(chip->my_private_pcm_data); + /* do what you like else */ + .... + } + + static int snd_mychip_new_pcm(struct mychip *chip) + { + struct snd_pcm *pcm; + .... + /* allocate your own data */ + chip->my_private_pcm_data = kmalloc(...); + /* set the destructor */ + pcm->private_data = chip; + pcm->private_free = mychip_pcm_free; + .... + } + + + +Runtime Pointer - The Chest of PCM Information +---------------------------------------------- + +When the PCM substream is opened, a PCM runtime instance is allocated +and assigned to the substream. This pointer is accessible via +``substream->runtime``. This runtime pointer holds most information you +need to control the PCM: the copy of hw_params and sw_params +configurations, the buffer pointers, mmap records, spinlocks, etc. + +The definition of runtime instance is found in ````. Here +are the contents of this file: + +:: + + struct _snd_pcm_runtime { + /* -- Status -- */ + struct snd_pcm_substream *trigger_master; + snd_timestamp_t trigger_tstamp; /* trigger timestamp */ + int overrange; + snd_pcm_uframes_t avail_max; + snd_pcm_uframes_t hw_ptr_base; /* Position at buffer restart */ + snd_pcm_uframes_t hw_ptr_interrupt; /* Position at interrupt time*/ + + /* -- HW params -- */ + snd_pcm_access_t access; /* access mode */ + snd_pcm_format_t format; /* SNDRV_PCM_FORMAT_* */ + snd_pcm_subformat_t subformat; /* subformat */ + unsigned int rate; /* rate in Hz */ + unsigned int channels; /* channels */ + snd_pcm_uframes_t period_size; /* period size */ + unsigned int periods; /* periods */ + snd_pcm_uframes_t buffer_size; /* buffer size */ + unsigned int tick_time; /* tick time */ + snd_pcm_uframes_t min_align; /* Min alignment for the format */ + size_t byte_align; + unsigned int frame_bits; + unsigned int sample_bits; + unsigned int info; + unsigned int rate_num; + unsigned int rate_den; + + /* -- SW params -- */ + struct timespec tstamp_mode; /* mmap timestamp is updated */ + unsigned int period_step; + unsigned int sleep_min; /* min ticks to sleep */ + snd_pcm_uframes_t start_threshold; + snd_pcm_uframes_t stop_threshold; + snd_pcm_uframes_t silence_threshold; /* Silence filling happens when + noise is nearest than this */ + snd_pcm_uframes_t silence_size; /* Silence filling size */ + snd_pcm_uframes_t boundary; /* pointers wrap point */ + + snd_pcm_uframes_t silenced_start; + snd_pcm_uframes_t silenced_size; + + snd_pcm_sync_id_t sync; /* hardware synchronization ID */ + + /* -- mmap -- */ + volatile struct snd_pcm_mmap_status *status; + volatile struct snd_pcm_mmap_control *control; + atomic_t mmap_count; + + /* -- locking / scheduling -- */ + spinlock_t lock; + wait_queue_head_t sleep; + struct timer_list tick_timer; + struct fasync_struct *fasync; + + /* -- private section -- */ + void *private_data; + void (*private_free)(struct snd_pcm_runtime *runtime); + + /* -- hardware description -- */ + struct snd_pcm_hardware hw; + struct snd_pcm_hw_constraints hw_constraints; + + /* -- timer -- */ + unsigned int timer_resolution; /* timer resolution */ + + /* -- DMA -- */ + unsigned char *dma_area; /* DMA area */ + dma_addr_t dma_addr; /* physical bus address (not accessible from main CPU) */ + size_t dma_bytes; /* size of DMA area */ + + struct snd_dma_buffer *dma_buffer_p; /* allocated buffer */ + + #if defined(CONFIG_SND_PCM_OSS) || defined(CONFIG_SND_PCM_OSS_MODULE) + /* -- OSS things -- */ + struct snd_pcm_oss_runtime oss; + #endif + }; + + +For the operators (callbacks) of each sound driver, most of these +records are supposed to be read-only. Only the PCM middle-layer changes +/ updates them. The exceptions are the hardware description (hw) DMA +buffer information and the private data. Besides, if you use the +standard buffer allocation method via +:c:func:`snd_pcm_lib_malloc_pages()`, you don't need to set the +DMA buffer information by yourself. + +In the sections below, important records are explained. + +Hardware Description +~~~~~~~~~~~~~~~~~~~~ + +The hardware descriptor (:c:type:`struct snd_pcm_hardware +`) contains the definitions of the fundamental +hardware configuration. Above all, you'll need to define this in the +`PCM open callback`_. Note that the runtime instance holds the copy of +the descriptor, not the pointer to the existing descriptor. That is, +in the open callback, you can modify the copied descriptor +(``runtime->hw``) as you need. For example, if the maximum number of +channels is 1 only on some chip models, you can still use the same +hardware descriptor and change the channels_max later: + +:: + + struct snd_pcm_runtime *runtime = substream->runtime; + ... + runtime->hw = snd_mychip_playback_hw; /* common definition */ + if (chip->model == VERY_OLD_ONE) + runtime->hw.channels_max = 1; + +Typically, you'll have a hardware descriptor as below: + +:: + + static struct snd_pcm_hardware snd_mychip_playback_hw = { + .info = (SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_MMAP_VALID), + .formats = SNDRV_PCM_FMTBIT_S16_LE, + .rates = SNDRV_PCM_RATE_8000_48000, + .rate_min = 8000, + .rate_max = 48000, + .channels_min = 2, + .channels_max = 2, + .buffer_bytes_max = 32768, + .period_bytes_min = 4096, + .period_bytes_max = 32768, + .periods_min = 1, + .periods_max = 1024, + }; + +- The ``info`` field contains the type and capabilities of this + pcm. The bit flags are defined in ```` as + ``SNDRV_PCM_INFO_XXX``. Here, at least, you have to specify whether + the mmap is supported and which interleaved format is + supported. When the hardware supports mmap, add the + ``SNDRV_PCM_INFO_MMAP`` flag here. When the hardware supports the + interleaved or the non-interleaved formats, + ``SNDRV_PCM_INFO_INTERLEAVED`` or ``SNDRV_PCM_INFO_NONINTERLEAVED`` + flag must be set, respectively. If both are supported, you can set + both, too. + + In the above example, ``MMAP_VALID`` and ``BLOCK_TRANSFER`` are + specified for the OSS mmap mode. Usually both are set. Of course, + ``MMAP_VALID`` is set only if the mmap is really supported. + + The other possible flags are ``SNDRV_PCM_INFO_PAUSE`` and + ``SNDRV_PCM_INFO_RESUME``. The ``PAUSE`` bit means that the pcm + supports the “pause” operation, while the ``RESUME`` bit means that + the pcm supports the full “suspend/resume” operation. If the + ``PAUSE`` flag is set, the ``trigger`` callback below must handle + the corresponding (pause push/release) commands. The suspend/resume + trigger commands can be defined even without the ``RESUME`` + flag. See `Power Management`_ section for details. + + When the PCM substreams can be synchronized (typically, + synchronized start/stop of a playback and a capture streams), you + can give ``SNDRV_PCM_INFO_SYNC_START``, too. In this case, you'll + need to check the linked-list of PCM substreams in the trigger + callback. This will be described in the later section. + +- ``formats`` field contains the bit-flags of supported formats + (``SNDRV_PCM_FMTBIT_XXX``). If the hardware supports more than one + format, give all or'ed bits. In the example above, the signed 16bit + little-endian format is specified. + +- ``rates`` field contains the bit-flags of supported rates + (``SNDRV_PCM_RATE_XXX``). When the chip supports continuous rates, + pass ``CONTINUOUS`` bit additionally. The pre-defined rate bits are + provided only for typical rates. If your chip supports + unconventional rates, you need to add the ``KNOT`` bit and set up + the hardware constraint manually (explained later). + +- ``rate_min`` and ``rate_max`` define the minimum and maximum sample + rate. This should correspond somehow to ``rates`` bits. + +- ``channel_min`` and ``channel_max`` define, as you might already + expected, the minimum and maximum number of channels. + +- ``buffer_bytes_max`` defines the maximum buffer size in + bytes. There is no ``buffer_bytes_min`` field, since it can be + calculated from the minimum period size and the minimum number of + periods. Meanwhile, ``period_bytes_min`` and define the minimum and + maximum size of the period in bytes. ``periods_max`` and + ``periods_min`` define the maximum and minimum number of periods in + the buffer. + + The “period” is a term that corresponds to a fragment in the OSS + world. The period defines the size at which a PCM interrupt is + generated. This size strongly depends on the hardware. Generally, + the smaller period size will give you more interrupts, that is, + more controls. In the case of capture, this size defines the input + latency. On the other hand, the whole buffer size defines the + output latency for the playback direction. + +- There is also a field ``fifo_size``. This specifies the size of the + hardware FIFO, but currently it is neither used in the driver nor + in the alsa-lib. So, you can ignore this field. + +PCM Configurations +~~~~~~~~~~~~~~~~~~ + +Ok, let's go back again to the PCM runtime records. The most +frequently referred records in the runtime instance are the PCM +configurations. The PCM configurations are stored in the runtime +instance after the application sends ``hw_params`` data via +alsa-lib. There are many fields copied from hw_params and sw_params +structs. For example, ``format`` holds the format type chosen by the +application. This field contains the enum value +``SNDRV_PCM_FORMAT_XXX``. + +One thing to be noted is that the configured buffer and period sizes +are stored in “frames” in the runtime. In the ALSA world, ``1 frame = +channels \* samples-size``. For conversion between frames and bytes, +you can use the :c:func:`frames_to_bytes()` and +:c:func:`bytes_to_frames()` helper functions. + +:: + + period_bytes = frames_to_bytes(runtime, runtime->period_size); + +Also, many software parameters (sw_params) are stored in frames, too. +Please check the type of the field. ``snd_pcm_uframes_t`` is for the +frames as unsigned integer while ``snd_pcm_sframes_t`` is for the +frames as signed integer. + +DMA Buffer Information +~~~~~~~~~~~~~~~~~~~~~~ + +The DMA buffer is defined by the following four fields, ``dma_area``, +``dma_addr``, ``dma_bytes`` and ``dma_private``. The ``dma_area`` +holds the buffer pointer (the logical address). You can call +:c:func:`memcpy()` from/to this pointer. Meanwhile, ``dma_addr`` holds +the physical address of the buffer. This field is specified only when +the buffer is a linear buffer. ``dma_bytes`` holds the size of buffer +in bytes. ``dma_private`` is used for the ALSA DMA allocator. + +If you use a standard ALSA function, +:c:func:`snd_pcm_lib_malloc_pages()`, for allocating the buffer, +these fields are set by the ALSA middle layer, and you should *not* +change them by yourself. You can read them but not write them. On the +other hand, if you want to allocate the buffer by yourself, you'll +need to manage it in hw_params callback. At least, ``dma_bytes`` is +mandatory. ``dma_area`` is necessary when the buffer is mmapped. If +your driver doesn't support mmap, this field is not +necessary. ``dma_addr`` is also optional. You can use dma_private as +you like, too. + +Running Status +~~~~~~~~~~~~~~ + +The running status can be referred via ``runtime->status``. This is +the pointer to the :c:type:`struct snd_pcm_mmap_status +` record. For example, you can get the current +DMA hardware pointer via ``runtime->status->hw_ptr``. + +The DMA application pointer can be referred via ``runtime->control``, +which points to the :c:type:`struct snd_pcm_mmap_control +` record. However, accessing directly to +this value is not recommended. + +Private Data +~~~~~~~~~~~~ + +You can allocate a record for the substream and store it in +``runtime->private_data``. Usually, this is done in the `PCM open +callback`_. Don't mix this with ``pcm->private_data``. The +``pcm->private_data`` usually points to the chip instance assigned +statically at the creation of PCM, while the ``runtime->private_data`` +points to a dynamic data structure created at the PCM open +callback. + +:: + + static int snd_xxx_open(struct snd_pcm_substream *substream) + { + struct my_pcm_data *data; + .... + data = kmalloc(sizeof(*data), GFP_KERNEL); + substream->runtime->private_data = data; + .... + } + + +The allocated object must be released in the `close callback`_. + +Operators +--------- + +OK, now let me give details about each pcm callback (``ops``). In +general, every callback must return 0 if successful, or a negative +error number such as ``-EINVAL``. To choose an appropriate error +number, it is advised to check what value other parts of the kernel +return when the same kind of request fails. + +The callback function takes at least the argument with :c:type:`struct +snd_pcm_substream ` pointer. To retrieve the chip +record from the given substream instance, you can use the following +macro. + +:: + + int xxx() { + struct mychip *chip = snd_pcm_substream_chip(substream); + .... + } + +The macro reads ``substream->private_data``, which is a copy of +``pcm->private_data``. You can override the former if you need to +assign different data records per PCM substream. For example, the +cmi8330 driver assigns different ``private_data`` for playback and +capture directions, because it uses two different codecs (SB- and +AD-compatible) for different directions. + +PCM open callback +~~~~~~~~~~~~~~~~~ + +:: + + static int snd_xxx_open(struct snd_pcm_substream *substream); + +This is called when a pcm substream is opened. + +At least, here you have to initialize the ``runtime->hw`` +record. Typically, this is done by like this: + +:: + + static int snd_xxx_open(struct snd_pcm_substream *substream) + { + struct mychip *chip = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + + runtime->hw = snd_mychip_playback_hw; + return 0; + } + +where ``snd_mychip_playback_hw`` is the pre-defined hardware +description. + +You can allocate a private data in this callback, as described in +`Private Data`_ section. + +If the hardware configuration needs more constraints, set the hardware +constraints here, too. See Constraints_ for more details. + +close callback +~~~~~~~~~~~~~~ + +:: + + static int snd_xxx_close(struct snd_pcm_substream *substream); + + +Obviously, this is called when a pcm substream is closed. + +Any private instance for a pcm substream allocated in the ``open`` +callback will be released here. + +:: + + static int snd_xxx_close(struct snd_pcm_substream *substream) + { + .... + kfree(substream->runtime->private_data); + .... + } + +ioctl callback +~~~~~~~~~~~~~~ + +This is used for any special call to pcm ioctls. But usually you can +pass a generic ioctl callback, :c:func:`snd_pcm_lib_ioctl()`. + +hw_params callback +~~~~~~~~~~~~~~~~~~~ + +:: + + static int snd_xxx_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params); + +This is called when the hardware parameter (``hw_params``) is set up +by the application, that is, once when the buffer size, the period +size, the format, etc. are defined for the pcm substream. + +Many hardware setups should be done in this callback, including the +allocation of buffers. + +Parameters to be initialized are retrieved by +:c:func:`params_xxx()` macros. To allocate buffer, you can call a +helper function, + +:: + + snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params)); + +:c:func:`snd_pcm_lib_malloc_pages()` is available only when the +DMA buffers have been pre-allocated. See the section `Buffer Types`_ +for more details. + +Note that this and ``prepare`` callbacks may be called multiple times +per initialization. For example, the OSS emulation may call these +callbacks at each change via its ioctl. + +Thus, you need to be careful not to allocate the same buffers many +times, which will lead to memory leaks! Calling the helper function +above many times is OK. It will release the previous buffer +automatically when it was already allocated. + +Another note is that this callback is non-atomic (schedulable) as +default, i.e. when no ``nonatomic`` flag set. This is important, +because the ``trigger`` callback is atomic (non-schedulable). That is, +mutexes or any schedule-related functions are not available in +``trigger`` callback. Please see the subsection Atomicity_ for +details. + +hw_free callback +~~~~~~~~~~~~~~~~~ + +:: + + static int snd_xxx_hw_free(struct snd_pcm_substream *substream); + +This is called to release the resources allocated via +``hw_params``. For example, releasing the buffer via +:c:func:`snd_pcm_lib_malloc_pages()` is done by calling the +following: + +:: + + snd_pcm_lib_free_pages(substream); + +This function is always called before the close callback is called. +Also, the callback may be called multiple times, too. Keep track +whether the resource was already released. + +prepare callback +~~~~~~~~~~~~~~~~ + +:: + + static int snd_xxx_prepare(struct snd_pcm_substream *substream); + +This callback is called when the pcm is “prepared”. You can set the +format type, sample rate, etc. here. The difference from ``hw_params`` +is that the ``prepare`` callback will be called each time +:c:func:`snd_pcm_prepare()` is called, i.e. when recovering after +underruns, etc. + +Note that this callback is now non-atomic. You can use +schedule-related functions safely in this callback. + +In this and the following callbacks, you can refer to the values via +the runtime record, ``substream->runtime``. For example, to get the +current rate, format or channels, access to ``runtime->rate``, +``runtime->format`` or ``runtime->channels``, respectively. The +physical address of the allocated buffer is set to +``runtime->dma_area``. The buffer and period sizes are in +``runtime->buffer_size`` and ``runtime->period_size``, respectively. + +Be careful that this callback will be called many times at each setup, +too. + +trigger callback +~~~~~~~~~~~~~~~~ + +:: + + static int snd_xxx_trigger(struct snd_pcm_substream *substream, int cmd); + +This is called when the pcm is started, stopped or paused. + +Which action is specified in the second argument, +``SNDRV_PCM_TRIGGER_XXX`` in ````. At least, the ``START`` +and ``STOP`` commands must be defined in this callback. + +:: + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + /* do something to start the PCM engine */ + break; + case SNDRV_PCM_TRIGGER_STOP: + /* do something to stop the PCM engine */ + break; + default: + return -EINVAL; + } + +When the pcm supports the pause operation (given in the info field of +the hardware table), the ``PAUSE_PUSH`` and ``PAUSE_RELEASE`` commands +must be handled here, too. The former is the command to pause the pcm, +and the latter to restart the pcm again. + +When the pcm supports the suspend/resume operation, regardless of full +or partial suspend/resume support, the ``SUSPEND`` and ``RESUME`` +commands must be handled, too. These commands are issued when the +power-management status is changed. Obviously, the ``SUSPEND`` and +``RESUME`` commands suspend and resume the pcm substream, and usually, +they are identical to the ``STOP`` and ``START`` commands, respectively. +See the `Power Management`_ section for details. + +As mentioned, this callback is atomic as default unless ``nonatomic`` +flag set, and you cannot call functions which may sleep. The +``trigger`` callback should be as minimal as possible, just really +triggering the DMA. The other stuff should be initialized +``hw_params`` and ``prepare`` callbacks properly beforehand. + +pointer callback +~~~~~~~~~~~~~~~~ + +:: + + static snd_pcm_uframes_t snd_xxx_pointer(struct snd_pcm_substream *substream) + +This callback is called when the PCM middle layer inquires the current +hardware position on the buffer. The position must be returned in +frames, ranging from 0 to ``buffer_size - 1``. + +This is called usually from the buffer-update routine in the pcm +middle layer, which is invoked when :c:func:`snd_pcm_period_elapsed()` +is called in the interrupt routine. Then the pcm middle layer updates +the position and calculates the available space, and wakes up the +sleeping poll threads, etc. + +This callback is also atomic as default. + +copy and silence callbacks +~~~~~~~~~~~~~~~~~~~~~~~~~~ + +These callbacks are not mandatory, and can be omitted in most cases. +These callbacks are used when the hardware buffer cannot be in the +normal memory space. Some chips have their own buffer on the hardware +which is not mappable. In such a case, you have to transfer the data +manually from the memory buffer to the hardware buffer. Or, if the +buffer is non-contiguous on both physical and virtual memory spaces, +these callbacks must be defined, too. + +If these two callbacks are defined, copy and set-silence operations +are done by them. The detailed will be described in the later section +`Buffer and Memory Management`_. + +ack callback +~~~~~~~~~~~~ + +This callback is also not mandatory. This callback is called when the +``appl_ptr`` is updated in read or write operations. Some drivers like +emu10k1-fx and cs46xx need to track the current ``appl_ptr`` for the +internal buffer, and this callback is useful only for such a purpose. + +This callback is atomic as default. + +page callback +~~~~~~~~~~~~~ + +This callback is optional too. This callback is used mainly for +non-contiguous buffers. The mmap calls this callback to get the page +address. Some examples will be explained in the later section `Buffer +and Memory Management`_, too. + +PCM Interrupt Handler +--------------------- + +The rest of pcm stuff is the PCM interrupt handler. The role of PCM +interrupt handler in the sound driver is to update the buffer position +and to tell the PCM middle layer when the buffer position goes across +the prescribed period size. To inform this, call the +:c:func:`snd_pcm_period_elapsed()` function. + +There are several types of sound chips to generate the interrupts. + +Interrupts at the period (fragment) boundary +~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + +This is the most frequently found type: the hardware generates an +interrupt at each period boundary. In this case, you can call +:c:func:`snd_pcm_period_elapsed()` at each interrupt. + +:c:func:`snd_pcm_period_elapsed()` takes the substream pointer as +its argument. Thus, you need to keep the substream pointer accessible +from the chip instance. For example, define ``substream`` field in the +chip record to hold the current running substream pointer, and set the +pointer value at ``open`` callback (and reset at ``close`` callback). + +If you acquire a spinlock in the interrupt handler, and the lock is used +in other pcm callbacks, too, then you have to release the lock before +calling :c:func:`snd_pcm_period_elapsed()`, because +:c:func:`snd_pcm_period_elapsed()` calls other pcm callbacks +inside. + +Typical code would be like: + +:: + + + static irqreturn_t snd_mychip_interrupt(int irq, void *dev_id) + { + struct mychip *chip = dev_id; + spin_lock(&chip->lock); + .... + if (pcm_irq_invoked(chip)) { + /* call updater, unlock before it */ + spin_unlock(&chip->lock); + snd_pcm_period_elapsed(chip->substream); + spin_lock(&chip->lock); + /* acknowledge the interrupt if necessary */ + } + .... + spin_unlock(&chip->lock); + return IRQ_HANDLED; + } + + + +High frequency timer interrupts +~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + +This happens when the hardware doesn't generate interrupts at the period +boundary but issues timer interrupts at a fixed timer rate (e.g. es1968 +or ymfpci drivers). In this case, you need to check the current hardware +position and accumulate the processed sample length at each interrupt. +When the accumulated size exceeds the period size, call +:c:func:`snd_pcm_period_elapsed()` and reset the accumulator. + +Typical code would be like the following. + +:: + + + static irqreturn_t snd_mychip_interrupt(int irq, void *dev_id) + { + struct mychip *chip = dev_id; + spin_lock(&chip->lock); + .... + if (pcm_irq_invoked(chip)) { + unsigned int last_ptr, size; + /* get the current hardware pointer (in frames) */ + last_ptr = get_hw_ptr(chip); + /* calculate the processed frames since the + * last update + */ + if (last_ptr < chip->last_ptr) + size = runtime->buffer_size + last_ptr + - chip->last_ptr; + else + size = last_ptr - chip->last_ptr; + /* remember the last updated point */ + chip->last_ptr = last_ptr; + /* accumulate the size */ + chip->size += size; + /* over the period boundary? */ + if (chip->size >= runtime->period_size) { + /* reset the accumulator */ + chip->size %= runtime->period_size; + /* call updater */ + spin_unlock(&chip->lock); + snd_pcm_period_elapsed(substream); + spin_lock(&chip->lock); + } + /* acknowledge the interrupt if necessary */ + } + .... + spin_unlock(&chip->lock); + return IRQ_HANDLED; + } + + + +On calling :c:func:`snd_pcm_period_elapsed()` +~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + +In both cases, even if more than one period are elapsed, you don't have +to call :c:func:`snd_pcm_period_elapsed()` many times. Call only +once. And the pcm layer will check the current hardware pointer and +update to the latest status. + +Atomicity +--------- + +One of the most important (and thus difficult to debug) problems in +kernel programming are race conditions. In the Linux kernel, they are +usually avoided via spin-locks, mutexes or semaphores. In general, if a +race condition can happen in an interrupt handler, it has to be managed +atomically, and you have to use a spinlock to protect the critical +session. If the critical section is not in interrupt handler code and if +taking a relatively long time to execute is acceptable, you should use +mutexes or semaphores instead. + +As already seen, some pcm callbacks are atomic and some are not. For +example, the ``hw_params`` callback is non-atomic, while ``trigger`` +callback is atomic. This means, the latter is called already in a +spinlock held by the PCM middle layer. Please take this atomicity into +account when you choose a locking scheme in the callbacks. + +In the atomic callbacks, you cannot use functions which may call +:c:func:`schedule()` or go to :c:func:`sleep()`. Semaphores and +mutexes can sleep, and hence they cannot be used inside the atomic +callbacks (e.g. ``trigger`` callback). To implement some delay in such a +callback, please use :c:func:`udelay()` or :c:func:`mdelay()`. + +All three atomic callbacks (trigger, pointer, and ack) are called with +local interrupts disabled. + +The recent changes in PCM core code, however, allow all PCM operations +to be non-atomic. This assumes that the all caller sides are in +non-atomic contexts. For example, the function +:c:func:`snd_pcm_period_elapsed()` is called typically from the +interrupt handler. But, if you set up the driver to use a threaded +interrupt handler, this call can be in non-atomic context, too. In such +a case, you can set ``nonatomic`` filed of :c:type:`struct snd_pcm +` object after creating it. When this flag is set, mutex +and rwsem are used internally in the PCM core instead of spin and +rwlocks, so that you can call all PCM functions safely in a non-atomic +context. + +Constraints +----------- + +If your chip supports unconventional sample rates, or only the limited +samples, you need to set a constraint for the condition. + +For example, in order to restrict the sample rates in the some supported +values, use :c:func:`snd_pcm_hw_constraint_list()`. You need to +call this function in the open callback. + +:: + + static unsigned int rates[] = + {4000, 10000, 22050, 44100}; + static struct snd_pcm_hw_constraint_list constraints_rates = { + .count = ARRAY_SIZE(rates), + .list = rates, + .mask = 0, + }; + + static int snd_mychip_pcm_open(struct snd_pcm_substream *substream) + { + int err; + .... + err = snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &constraints_rates); + if (err < 0) + return err; + .... + } + + + +There are many different constraints. Look at ``sound/pcm.h`` for a +complete list. You can even define your own constraint rules. For +example, let's suppose my_chip can manage a substream of 1 channel if +and only if the format is ``S16_LE``, otherwise it supports any format +specified in the :c:type:`struct snd_pcm_hardware +` structure (or in any other +constraint_list). You can build a rule like this: + +:: + + static int hw_rule_channels_by_format(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) + { + struct snd_interval *c = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_CHANNELS); + struct snd_mask *f = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); + struct snd_interval ch; + + snd_interval_any(&ch); + if (f->bits[0] == SNDRV_PCM_FMTBIT_S16_LE) { + ch.min = ch.max = 1; + ch.integer = 1; + return snd_interval_refine(c, &ch); + } + return 0; + } + + +Then you need to call this function to add your rule: + +:: + + snd_pcm_hw_rule_add(substream->runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + hw_rule_channels_by_format, NULL, + SNDRV_PCM_HW_PARAM_FORMAT, -1); + +The rule function is called when an application sets the PCM format, and +it refines the number of channels accordingly. But an application may +set the number of channels before setting the format. Thus you also need +to define the inverse rule: + +:: + + static int hw_rule_format_by_channels(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) + { + struct snd_interval *c = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_CHANNELS); + struct snd_mask *f = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); + struct snd_mask fmt; + + snd_mask_any(&fmt); /* Init the struct */ + if (c->min < 2) { + fmt.bits[0] &= SNDRV_PCM_FMTBIT_S16_LE; + return snd_mask_refine(f, &fmt); + } + return 0; + } + + +... and in the open callback: + +:: + + snd_pcm_hw_rule_add(substream->runtime, 0, SNDRV_PCM_HW_PARAM_FORMAT, + hw_rule_format_by_channels, NULL, + SNDRV_PCM_HW_PARAM_CHANNELS, -1); + +I won't give more details here, rather I would like to say, “Luke, use +the source.” + +Control Interface +================= + +General +------- + +The control interface is used widely for many switches, sliders, etc. +which are accessed from user-space. Its most important use is the mixer +interface. In other words, since ALSA 0.9.x, all the mixer stuff is +implemented on the control kernel API. + +ALSA has a well-defined AC97 control module. If your chip supports only +the AC97 and nothing else, you can skip this section. + +The control API is defined in ````. Include this file +if you want to add your own controls. + +Definition of Controls +---------------------- + +To create a new control, you need to define the following three +callbacks: ``info``, ``get`` and ``put``. Then, define a +:c:type:`struct snd_kcontrol_new ` record, such as: + +:: + + + static struct snd_kcontrol_new my_control = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "PCM Playback Switch", + .index = 0, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .private_value = 0xffff, + .info = my_control_info, + .get = my_control_get, + .put = my_control_put + }; + + +The ``iface`` field specifies the control type, +``SNDRV_CTL_ELEM_IFACE_XXX``, which is usually ``MIXER``. Use ``CARD`` +for global controls that are not logically part of the mixer. If the +control is closely associated with some specific device on the sound +card, use ``HWDEP``, ``PCM``, ``RAWMIDI``, ``TIMER``, or ``SEQUENCER``, +and specify the device number with the ``device`` and ``subdevice`` +fields. + +The ``name`` is the name identifier string. Since ALSA 0.9.x, the +control name is very important, because its role is classified from +its name. There are pre-defined standard control names. The details +are described in the `Control Names`_ subsection. + +The ``index`` field holds the index number of this control. If there +are several different controls with the same name, they can be +distinguished by the index number. This is the case when several +codecs exist on the card. If the index is zero, you can omit the +definition above. + +The ``access`` field contains the access type of this control. Give +the combination of bit masks, ``SNDRV_CTL_ELEM_ACCESS_XXX``, +there. The details will be explained in the `Access Flags`_ +subsection. + +The ``private_value`` field contains an arbitrary long integer value +for this record. When using the generic ``info``, ``get`` and ``put`` +callbacks, you can pass a value through this field. If several small +numbers are necessary, you can combine them in bitwise. Or, it's +possible to give a pointer (casted to unsigned long) of some record to +this field, too. + +The ``tlv`` field can be used to provide metadata about the control; +see the `Metadata`_ subsection. + +The other three are `Control Callbacks`_. + +Control Names +------------- + +There are some standards to define the control names. A control is +usually defined from the three parts as “SOURCE DIRECTION FUNCTION”. + +The first, ``SOURCE``, specifies the source of the control, and is a +string such as “Master”, “PCM”, “CD” and “Line”. There are many +pre-defined sources. + +The second, ``DIRECTION``, is one of the following strings according to +the direction of the control: “Playback”, “Capture”, “Bypass Playback” +and “Bypass Capture”. Or, it can be omitted, meaning both playback and +capture directions. + +The third, ``FUNCTION``, is one of the following strings according to +the function of the control: “Switch”, “Volume” and “Route”. + +The example of control names are, thus, “Master Capture Switch” or “PCM +Playback Volume”. + +There are some exceptions: + +Global capture and playback +~~~~~~~~~~~~~~~~~~~~~~~~~~~ + +“Capture Source”, “Capture Switch” and “Capture Volume” are used for the +global capture (input) source, switch and volume. Similarly, “Playback +Switch” and “Playback Volume” are used for the global output gain switch +and volume. + +Tone-controls +~~~~~~~~~~~~~ + +tone-control switch and volumes are specified like “Tone Control - XXX”, +e.g. “Tone Control - Switch”, “Tone Control - Bass”, “Tone Control - +Center”. + +3D controls +~~~~~~~~~~~ + +3D-control switches and volumes are specified like “3D Control - XXX”, +e.g. “3D Control - Switch”, “3D Control - Center”, “3D Control - Space”. + +Mic boost +~~~~~~~~~ + +Mic-boost switch is set as “Mic Boost” or “Mic Boost (6dB)”. + +More precise information can be found in +``Documentation/sound/alsa/ControlNames.txt``. + +Access Flags +------------ + +The access flag is the bitmask which specifies the access type of the +given control. The default access type is +``SNDRV_CTL_ELEM_ACCESS_READWRITE``, which means both read and write are +allowed to this control. When the access flag is omitted (i.e. = 0), it +is considered as ``READWRITE`` access as default. + +When the control is read-only, pass ``SNDRV_CTL_ELEM_ACCESS_READ`` +instead. In this case, you don't have to define the ``put`` callback. +Similarly, when the control is write-only (although it's a rare case), +you can use the ``WRITE`` flag instead, and you don't need the ``get`` +callback. + +If the control value changes frequently (e.g. the VU meter), +``VOLATILE`` flag should be given. This means that the control may be +changed without `Change notification`_. Applications should poll such +a control constantly. + +When the control is inactive, set the ``INACTIVE`` flag, too. There are +``LOCK`` and ``OWNER`` flags to change the write permissions. + +Control Callbacks +----------------- + +info callback +~~~~~~~~~~~~~ + +The ``info`` callback is used to get detailed information on this +control. This must store the values of the given :c:type:`struct +snd_ctl_elem_info ` object. For example, +for a boolean control with a single element: + +:: + + + static int snd_myctl_mono_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) + { + uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 1; + return 0; + } + + + +The ``type`` field specifies the type of the control. There are +``BOOLEAN``, ``INTEGER``, ``ENUMERATED``, ``BYTES``, ``IEC958`` and +``INTEGER64``. The ``count`` field specifies the number of elements in +this control. For example, a stereo volume would have count = 2. The +``value`` field is a union, and the values stored are depending on the +type. The boolean and integer types are identical. + +The enumerated type is a bit different from others. You'll need to set +the string for the currently given item index. + +:: + + static int snd_myctl_enum_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) + { + static char *texts[4] = { + "First", "Second", "Third", "Fourth" + }; + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + uinfo->count = 1; + uinfo->value.enumerated.items = 4; + if (uinfo->value.enumerated.item > 3) + uinfo->value.enumerated.item = 3; + strcpy(uinfo->value.enumerated.name, + texts[uinfo->value.enumerated.item]); + return 0; + } + +The above callback can be simplified with a helper function, +:c:func:`snd_ctl_enum_info()`. The final code looks like below. +(You can pass ``ARRAY_SIZE(texts)`` instead of 4 in the third argument; +it's a matter of taste.) + +:: + + static int snd_myctl_enum_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) + { + static char *texts[4] = { + "First", "Second", "Third", "Fourth" + }; + return snd_ctl_enum_info(uinfo, 1, 4, texts); + } + + +Some common info callbacks are available for your convenience: +:c:func:`snd_ctl_boolean_mono_info()` and +:c:func:`snd_ctl_boolean_stereo_info()`. Obviously, the former +is an info callback for a mono channel boolean item, just like +:c:func:`snd_myctl_mono_info()` above, and the latter is for a +stereo channel boolean item. + +get callback +~~~~~~~~~~~~ + +This callback is used to read the current value of the control and to +return to user-space. + +For example, + +:: + + + static int snd_myctl_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) + { + struct mychip *chip = snd_kcontrol_chip(kcontrol); + ucontrol->value.integer.value[0] = get_some_value(chip); + return 0; + } + + + +The ``value`` field depends on the type of control as well as on the +info callback. For example, the sb driver uses this field to store the +register offset, the bit-shift and the bit-mask. The ``private_value`` +field is set as follows: + +:: + + .private_value = reg | (shift << 16) | (mask << 24) + +and is retrieved in callbacks like + +:: + + static int snd_sbmixer_get_single(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) + { + int reg = kcontrol->private_value & 0xff; + int shift = (kcontrol->private_value >> 16) & 0xff; + int mask = (kcontrol->private_value >> 24) & 0xff; + .... + } + +In the ``get`` callback, you have to fill all the elements if the +control has more than one elements, i.e. ``count > 1``. In the example +above, we filled only one element (``value.integer.value[0]``) since +it's assumed as ``count = 1``. + +put callback +~~~~~~~~~~~~ + +This callback is used to write a value from user-space. + +For example, + +:: + + + static int snd_myctl_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) + { + struct mychip *chip = snd_kcontrol_chip(kcontrol); + int changed = 0; + if (chip->current_value != + ucontrol->value.integer.value[0]) { + change_current_value(chip, + ucontrol->value.integer.value[0]); + changed = 1; + } + return changed; + } + + + +As seen above, you have to return 1 if the value is changed. If the +value is not changed, return 0 instead. If any fatal error happens, +return a negative error code as usual. + +As in the ``get`` callback, when the control has more than one +elements, all elements must be evaluated in this callback, too. + +Callbacks are not atomic +~~~~~~~~~~~~~~~~~~~~~~~~ + +All these three callbacks are basically not atomic. + +Control Constructor +------------------- + +When everything is ready, finally we can create a new control. To create +a control, there are two functions to be called, +:c:func:`snd_ctl_new1()` and :c:func:`snd_ctl_add()`. + +In the simplest way, you can do like this: + +:: + + err = snd_ctl_add(card, snd_ctl_new1(&my_control, chip)); + if (err < 0) + return err; + +where ``my_control`` is the :c:type:`struct snd_kcontrol_new +` object defined above, and chip is the object +pointer to be passed to kcontrol->private_data which can be referred +to in callbacks. + +:c:func:`snd_ctl_new1()` allocates a new :c:type:`struct +snd_kcontrol ` instance, and +:c:func:`snd_ctl_add()` assigns the given control component to the +card. + +Change Notification +------------------- + +If you need to change and update a control in the interrupt routine, you +can call :c:func:`snd_ctl_notify()`. For example, + +:: + + snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, id_pointer); + +This function takes the card pointer, the event-mask, and the control id +pointer for the notification. The event-mask specifies the types of +notification, for example, in the above example, the change of control +values is notified. The id pointer is the pointer of :c:type:`struct +snd_ctl_elem_id ` to be notified. You can +find some examples in ``es1938.c`` or ``es1968.c`` for hardware volume +interrupts. + +Metadata +-------- + +To provide information about the dB values of a mixer control, use on of +the ``DECLARE_TLV_xxx`` macros from ```` to define a +variable containing this information, set the ``tlv.p`` field to point to +this variable, and include the ``SNDRV_CTL_ELEM_ACCESS_TLV_READ`` flag +in the ``access`` field; like this: + +:: + + static DECLARE_TLV_DB_SCALE(db_scale_my_control, -4050, 150, 0); + + static struct snd_kcontrol_new my_control = { + ... + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ, + ... + .tlv.p = db_scale_my_control, + }; + + +The :c:func:`DECLARE_TLV_DB_SCALE()` macro defines information +about a mixer control where each step in the control's value changes the +dB value by a constant dB amount. The first parameter is the name of the +variable to be defined. The second parameter is the minimum value, in +units of 0.01 dB. The third parameter is the step size, in units of 0.01 +dB. Set the fourth parameter to 1 if the minimum value actually mutes +the control. + +The :c:func:`DECLARE_TLV_DB_LINEAR()` macro defines information +about a mixer control where the control's value affects the output +linearly. The first parameter is the name of the variable to be defined. +The second parameter is the minimum value, in units of 0.01 dB. The +third parameter is the maximum value, in units of 0.01 dB. If the +minimum value mutes the control, set the second parameter to +``TLV_DB_GAIN_MUTE``. + +API for AC97 Codec +================== + +General +------- + +The ALSA AC97 codec layer is a well-defined one, and you don't have to +write much code to control it. Only low-level control routines are +necessary. The AC97 codec API is defined in ````. + +Full Code Example +----------------- + +:: + + struct mychip { + .... + struct snd_ac97 *ac97; + .... + }; + + static unsigned short snd_mychip_ac97_read(struct snd_ac97 *ac97, + unsigned short reg) + { + struct mychip *chip = ac97->private_data; + .... + /* read a register value here from the codec */ + return the_register_value; + } + + static void snd_mychip_ac97_write(struct snd_ac97 *ac97, + unsigned short reg, unsigned short val) + { + struct mychip *chip = ac97->private_data; + .... + /* write the given register value to the codec */ + } + + static int snd_mychip_ac97(struct mychip *chip) + { + struct snd_ac97_bus *bus; + struct snd_ac97_template ac97; + int err; + static struct snd_ac97_bus_ops ops = { + .write = snd_mychip_ac97_write, + .read = snd_mychip_ac97_read, + }; + + err = snd_ac97_bus(chip->card, 0, &ops, NULL, &bus); + if (err < 0) + return err; + memset(&ac97, 0, sizeof(ac97)); + ac97.private_data = chip; + return snd_ac97_mixer(bus, &ac97, &chip->ac97); + } + + +AC97 Constructor +---------------- + +To create an ac97 instance, first call :c:func:`snd_ac97_bus()` +with an ``ac97_bus_ops_t`` record with callback functions. + +:: + + struct snd_ac97_bus *bus; + static struct snd_ac97_bus_ops ops = { + .write = snd_mychip_ac97_write, + .read = snd_mychip_ac97_read, + }; + + snd_ac97_bus(card, 0, &ops, NULL, &pbus); + +The bus record is shared among all belonging ac97 instances. + +And then call :c:func:`snd_ac97_mixer()` with an :c:type:`struct +snd_ac97_template ` record together with +the bus pointer created above. + +:: + + struct snd_ac97_template ac97; + int err; + + memset(&ac97, 0, sizeof(ac97)); + ac97.private_data = chip; + snd_ac97_mixer(bus, &ac97, &chip->ac97); + +where chip->ac97 is a pointer to a newly created ``ac97_t`` +instance. In this case, the chip pointer is set as the private data, +so that the read/write callback functions can refer to this chip +instance. This instance is not necessarily stored in the chip +record. If you need to change the register values from the driver, or +need the suspend/resume of ac97 codecs, keep this pointer to pass to +the corresponding functions. + +AC97 Callbacks +-------------- + +The standard callbacks are ``read`` and ``write``. Obviously they +correspond to the functions for read and write accesses to the +hardware low-level codes. + +The ``read`` callback returns the register value specified in the +argument. + +:: + + static unsigned short snd_mychip_ac97_read(struct snd_ac97 *ac97, + unsigned short reg) + { + struct mychip *chip = ac97->private_data; + .... + return the_register_value; + } + +Here, the chip can be cast from ``ac97->private_data``. + +Meanwhile, the ``write`` callback is used to set the register +value + +:: + + static void snd_mychip_ac97_write(struct snd_ac97 *ac97, + unsigned short reg, unsigned short val) + + +These callbacks are non-atomic like the control API callbacks. + +There are also other callbacks: ``reset``, ``wait`` and ``init``. + +The ``reset`` callback is used to reset the codec. If the chip +requires a special kind of reset, you can define this callback. + +The ``wait`` callback is used to add some waiting time in the standard +initialization of the codec. If the chip requires the extra waiting +time, define this callback. + +The ``init`` callback is used for additional initialization of the +codec. + +Updating Registers in The Driver +-------------------------------- + +If you need to access to the codec from the driver, you can call the +following functions: :c:func:`snd_ac97_write()`, +:c:func:`snd_ac97_read()`, :c:func:`snd_ac97_update()` and +:c:func:`snd_ac97_update_bits()`. + +Both :c:func:`snd_ac97_write()` and +:c:func:`snd_ac97_update()` functions are used to set a value to +the given register (``AC97_XXX``). The difference between them is that +:c:func:`snd_ac97_update()` doesn't write a value if the given +value has been already set, while :c:func:`snd_ac97_write()` +always rewrites the value. + +:: + + snd_ac97_write(ac97, AC97_MASTER, 0x8080); + snd_ac97_update(ac97, AC97_MASTER, 0x8080); + +:c:func:`snd_ac97_read()` is used to read the value of the given +register. For example, + +:: + + value = snd_ac97_read(ac97, AC97_MASTER); + +:c:func:`snd_ac97_update_bits()` is used to update some bits in +the given register. + +:: + + snd_ac97_update_bits(ac97, reg, mask, value); + +Also, there is a function to change the sample rate (of a given register +such as ``AC97_PCM_FRONT_DAC_RATE``) when VRA or DRA is supported by the +codec: :c:func:`snd_ac97_set_rate()`. + +:: + + snd_ac97_set_rate(ac97, AC97_PCM_FRONT_DAC_RATE, 44100); + + +The following registers are available to set the rate: +``AC97_PCM_MIC_ADC_RATE``, ``AC97_PCM_FRONT_DAC_RATE``, +``AC97_PCM_LR_ADC_RATE``, ``AC97_SPDIF``. When ``AC97_SPDIF`` is +specified, the register is not really changed but the corresponding +IEC958 status bits will be updated. + +Clock Adjustment +---------------- + +In some chips, the clock of the codec isn't 48000 but using a PCI clock +(to save a quartz!). In this case, change the field ``bus->clock`` to +the corresponding value. For example, intel8x0 and es1968 drivers have +their own function to read from the clock. + +Proc Files +---------- + +The ALSA AC97 interface will create a proc file such as +``/proc/asound/card0/codec97#0/ac97#0-0`` and ``ac97#0-0+regs``. You +can refer to these files to see the current status and registers of +the codec. + +Multiple Codecs +--------------- + +When there are several codecs on the same card, you need to call +:c:func:`snd_ac97_mixer()` multiple times with ``ac97.num=1`` or +greater. The ``num`` field specifies the codec number. + +If you set up multiple codecs, you either need to write different +callbacks for each codec or check ``ac97->num`` in the callback +routines. + +MIDI (MPU401-UART) Interface +============================ + +General +------- + +Many soundcards have built-in MIDI (MPU401-UART) interfaces. When the +soundcard supports the standard MPU401-UART interface, most likely you +can use the ALSA MPU401-UART API. The MPU401-UART API is defined in +````. + +Some soundchips have a similar but slightly different implementation of +mpu401 stuff. For example, emu10k1 has its own mpu401 routines. + +MIDI Constructor +---------------- + +To create a rawmidi object, call :c:func:`snd_mpu401_uart_new()`. + +:: + + struct snd_rawmidi *rmidi; + snd_mpu401_uart_new(card, 0, MPU401_HW_MPU401, port, info_flags, + irq, &rmidi); + + +The first argument is the card pointer, and the second is the index of +this component. You can create up to 8 rawmidi devices. + +The third argument is the type of the hardware, ``MPU401_HW_XXX``. If +it's not a special one, you can use ``MPU401_HW_MPU401``. + +The 4th argument is the I/O port address. Many backward-compatible +MPU401 have an I/O port such as 0x330. Or, it might be a part of its own +PCI I/O region. It depends on the chip design. + +The 5th argument is a bitflag for additional information. When the I/O +port address above is part of the PCI I/O region, the MPU401 I/O port +might have been already allocated (reserved) by the driver itself. In +such a case, pass a bit flag ``MPU401_INFO_INTEGRATED``, and the +mpu401-uart layer will allocate the I/O ports by itself. + +When the controller supports only the input or output MIDI stream, pass +the ``MPU401_INFO_INPUT`` or ``MPU401_INFO_OUTPUT`` bitflag, +respectively. Then the rawmidi instance is created as a single stream. + +``MPU401_INFO_MMIO`` bitflag is used to change the access method to MMIO +(via readb and writeb) instead of iob and outb. In this case, you have +to pass the iomapped address to :c:func:`snd_mpu401_uart_new()`. + +When ``MPU401_INFO_TX_IRQ`` is set, the output stream isn't checked in +the default interrupt handler. The driver needs to call +:c:func:`snd_mpu401_uart_interrupt_tx()` by itself to start +processing the output stream in the irq handler. + +If the MPU-401 interface shares its interrupt with the other logical +devices on the card, set ``MPU401_INFO_IRQ_HOOK`` (see +`below <#MIDI-Interrupt-Handler>`__). + +Usually, the port address corresponds to the command port and port + 1 +corresponds to the data port. If not, you may change the ``cport`` +field of :c:type:`struct snd_mpu401 ` manually afterward. +However, :c:type:`struct snd_mpu401 ` pointer is +not returned explicitly by :c:func:`snd_mpu401_uart_new()`. You +need to cast ``rmidi->private_data`` to :c:type:`struct snd_mpu401 +` explicitly, + +:: + + struct snd_mpu401 *mpu; + mpu = rmidi->private_data; + +and reset the ``cport`` as you like: + +:: + + mpu->cport = my_own_control_port; + +The 6th argument specifies the ISA irq number that will be allocated. If +no interrupt is to be allocated (because your code is already allocating +a shared interrupt, or because the device does not use interrupts), pass +-1 instead. For a MPU-401 device without an interrupt, a polling timer +will be used instead. + +MIDI Interrupt Handler +---------------------- + +When the interrupt is allocated in +:c:func:`snd_mpu401_uart_new()`, an exclusive ISA interrupt +handler is automatically used, hence you don't have anything else to do +than creating the mpu401 stuff. Otherwise, you have to set +``MPU401_INFO_IRQ_HOOK``, and call +:c:func:`snd_mpu401_uart_interrupt()` explicitly from your own +interrupt handler when it has determined that a UART interrupt has +occurred. + +In this case, you need to pass the private_data of the returned rawmidi +object from :c:func:`snd_mpu401_uart_new()` as the second +argument of :c:func:`snd_mpu401_uart_interrupt()`. + +:: + + snd_mpu401_uart_interrupt(irq, rmidi->private_data, regs); + + +RawMIDI Interface +================= + +Overview +-------- + +The raw MIDI interface is used for hardware MIDI ports that can be +accessed as a byte stream. It is not used for synthesizer chips that do +not directly understand MIDI. + +ALSA handles file and buffer management. All you have to do is to write +some code to move data between the buffer and the hardware. + +The rawmidi API is defined in ````. + +RawMIDI Constructor +------------------- + +To create a rawmidi device, call the :c:func:`snd_rawmidi_new()` +function: + +:: + + struct snd_rawmidi *rmidi; + err = snd_rawmidi_new(chip->card, "MyMIDI", 0, outs, ins, &rmidi); + if (err < 0) + return err; + rmidi->private_data = chip; + strcpy(rmidi->name, "My MIDI"); + rmidi->info_flags = SNDRV_RAWMIDI_INFO_OUTPUT | + SNDRV_RAWMIDI_INFO_INPUT | + SNDRV_RAWMIDI_INFO_DUPLEX; + +The first argument is the card pointer, the second argument is the ID +string. + +The third argument is the index of this component. You can create up to +8 rawmidi devices. + +The fourth and fifth arguments are the number of output and input +substreams, respectively, of this device (a substream is the equivalent +of a MIDI port). + +Set the ``info_flags`` field to specify the capabilities of the +device. Set ``SNDRV_RAWMIDI_INFO_OUTPUT`` if there is at least one +output port, ``SNDRV_RAWMIDI_INFO_INPUT`` if there is at least one +input port, and ``SNDRV_RAWMIDI_INFO_DUPLEX`` if the device can handle +output and input at the same time. + +After the rawmidi device is created, you need to set the operators +(callbacks) for each substream. There are helper functions to set the +operators for all the substreams of a device: + +:: + + snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_OUTPUT, &snd_mymidi_output_ops); + snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_INPUT, &snd_mymidi_input_ops); + +The operators are usually defined like this: + +:: + + static struct snd_rawmidi_ops snd_mymidi_output_ops = { + .open = snd_mymidi_output_open, + .close = snd_mymidi_output_close, + .trigger = snd_mymidi_output_trigger, + }; + +These callbacks are explained in the `RawMIDI Callbacks`_ section. + +If there are more than one substream, you should give a unique name to +each of them: + +:: + + struct snd_rawmidi_substream *substream; + list_for_each_entry(substream, + &rmidi->streams[SNDRV_RAWMIDI_STREAM_OUTPUT].substreams, + list { + sprintf(substream->name, "My MIDI Port %d", substream->number + 1); + } + /* same for SNDRV_RAWMIDI_STREAM_INPUT */ + +RawMIDI Callbacks +----------------- + +In all the callbacks, the private data that you've set for the rawmidi +device can be accessed as ``substream->rmidi->private_data``. + +If there is more than one port, your callbacks can determine the port +index from the struct snd_rawmidi_substream data passed to each +callback: + +:: + + struct snd_rawmidi_substream *substream; + int index = substream->number; + +RawMIDI open callback +~~~~~~~~~~~~~~~~~~~~~ + +:: + + static int snd_xxx_open(struct snd_rawmidi_substream *substream); + + +This is called when a substream is opened. You can initialize the +hardware here, but you shouldn't start transmitting/receiving data yet. + +RawMIDI close callback +~~~~~~~~~~~~~~~~~~~~~~ + +:: + + static int snd_xxx_close(struct snd_rawmidi_substream *substream); + +Guess what. + +The ``open`` and ``close`` callbacks of a rawmidi device are +serialized with a mutex, and can sleep. + +Rawmidi trigger callback for output substreams +~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + +:: + + static void snd_xxx_output_trigger(struct snd_rawmidi_substream *substream, int up); + + +This is called with a nonzero ``up`` parameter when there is some data +in the substream buffer that must be transmitted. + +To read data from the buffer, call +:c:func:`snd_rawmidi_transmit_peek()`. It will return the number +of bytes that have been read; this will be less than the number of bytes +requested when there are no more data in the buffer. After the data have +been transmitted successfully, call +:c:func:`snd_rawmidi_transmit_ack()` to remove the data from the +substream buffer: + +:: + + unsigned char data; + while (snd_rawmidi_transmit_peek(substream, &data, 1) == 1) { + if (snd_mychip_try_to_transmit(data)) + snd_rawmidi_transmit_ack(substream, 1); + else + break; /* hardware FIFO full */ + } + +If you know beforehand that the hardware will accept data, you can use +the :c:func:`snd_rawmidi_transmit()` function which reads some +data and removes them from the buffer at once: + +:: + + while (snd_mychip_transmit_possible()) { + unsigned char data; + if (snd_rawmidi_transmit(substream, &data, 1) != 1) + break; /* no more data */ + snd_mychip_transmit(data); + } + +If you know beforehand how many bytes you can accept, you can use a +buffer size greater than one with the +:c:func:`snd_rawmidi_transmit\*()` functions. + +The ``trigger`` callback must not sleep. If the hardware FIFO is full +before the substream buffer has been emptied, you have to continue +transmitting data later, either in an interrupt handler, or with a +timer if the hardware doesn't have a MIDI transmit interrupt. + +The ``trigger`` callback is called with a zero ``up`` parameter when +the transmission of data should be aborted. + +RawMIDI trigger callback for input substreams +~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + +:: + + static void snd_xxx_input_trigger(struct snd_rawmidi_substream *substream, int up); + + +This is called with a nonzero ``up`` parameter to enable receiving data, +or with a zero ``up`` parameter do disable receiving data. + +The ``trigger`` callback must not sleep; the actual reading of data +from the device is usually done in an interrupt handler. + +When data reception is enabled, your interrupt handler should call +:c:func:`snd_rawmidi_receive()` for all received data: + +:: + + void snd_mychip_midi_interrupt(...) + { + while (mychip_midi_available()) { + unsigned char data; + data = mychip_midi_read(); + snd_rawmidi_receive(substream, &data, 1); + } + } + + +drain callback +~~~~~~~~~~~~~~ + +:: + + static void snd_xxx_drain(struct snd_rawmidi_substream *substream); + + +This is only used with output substreams. This function should wait +until all data read from the substream buffer have been transmitted. +This ensures that the device can be closed and the driver unloaded +without losing data. + +This callback is optional. If you do not set ``drain`` in the struct +snd_rawmidi_ops structure, ALSA will simply wait for 50 milliseconds +instead. + +Miscellaneous Devices +===================== + +FM OPL3 +------- + +The FM OPL3 is still used in many chips (mainly for backward +compatibility). ALSA has a nice OPL3 FM control layer, too. The OPL3 API +is defined in ````. + +FM registers can be directly accessed through the direct-FM API, defined +in ````. In ALSA native mode, FM registers are +accessed through the Hardware-Dependent Device direct-FM extension API, +whereas in OSS compatible mode, FM registers can be accessed with the +OSS direct-FM compatible API in ``/dev/dmfmX`` device. + +To create the OPL3 component, you have two functions to call. The first +one is a constructor for the ``opl3_t`` instance. + +:: + + struct snd_opl3 *opl3; + snd_opl3_create(card, lport, rport, OPL3_HW_OPL3_XXX, + integrated, &opl3); + +The first argument is the card pointer, the second one is the left port +address, and the third is the right port address. In most cases, the +right port is placed at the left port + 2. + +The fourth argument is the hardware type. + +When the left and right ports have been already allocated by the card +driver, pass non-zero to the fifth argument (``integrated``). Otherwise, +the opl3 module will allocate the specified ports by itself. + +When the accessing the hardware requires special method instead of the +standard I/O access, you can create opl3 instance separately with +:c:func:`snd_opl3_new()`. + +:: + + struct snd_opl3 *opl3; + snd_opl3_new(card, OPL3_HW_OPL3_XXX, &opl3); + +Then set ``command``, ``private_data`` and ``private_free`` for the +private access function, the private data and the destructor. The +``l_port`` and ``r_port`` are not necessarily set. Only the command +must be set properly. You can retrieve the data from the +``opl3->private_data`` field. + +After creating the opl3 instance via :c:func:`snd_opl3_new()`, +call :c:func:`snd_opl3_init()` to initialize the chip to the +proper state. Note that :c:func:`snd_opl3_create()` always calls +it internally. + +If the opl3 instance is created successfully, then create a hwdep device +for this opl3. + +:: + + struct snd_hwdep *opl3hwdep; + snd_opl3_hwdep_new(opl3, 0, 1, &opl3hwdep); + +The first argument is the ``opl3_t`` instance you created, and the +second is the index number, usually 0. + +The third argument is the index-offset for the sequencer client assigned +to the OPL3 port. When there is an MPU401-UART, give 1 for here (UART +always takes 0). + +Hardware-Dependent Devices +-------------------------- + +Some chips need user-space access for special controls or for loading +the micro code. In such a case, you can create a hwdep +(hardware-dependent) device. The hwdep API is defined in +````. You can find examples in opl3 driver or +``isa/sb/sb16_csp.c``. + +The creation of the ``hwdep`` instance is done via +:c:func:`snd_hwdep_new()`. + +:: + + struct snd_hwdep *hw; + snd_hwdep_new(card, "My HWDEP", 0, &hw); + +where the third argument is the index number. + +You can then pass any pointer value to the ``private_data``. If you +assign a private data, you should define the destructor, too. The +destructor function is set in the ``private_free`` field. + +:: + + struct mydata *p = kmalloc(sizeof(*p), GFP_KERNEL); + hw->private_data = p; + hw->private_free = mydata_free; + +and the implementation of the destructor would be: + +:: + + static void mydata_free(struct snd_hwdep *hw) + { + struct mydata *p = hw->private_data; + kfree(p); + } + +The arbitrary file operations can be defined for this instance. The file +operators are defined in the ``ops`` table. For example, assume that +this chip needs an ioctl. + +:: + + hw->ops.open = mydata_open; + hw->ops.ioctl = mydata_ioctl; + hw->ops.release = mydata_release; + +And implement the callback functions as you like. + +IEC958 (S/PDIF) +--------------- + +Usually the controls for IEC958 devices are implemented via the control +interface. There is a macro to compose a name string for IEC958 +controls, :c:func:`SNDRV_CTL_NAME_IEC958()` defined in +````. + +There are some standard controls for IEC958 status bits. These controls +use the type ``SNDRV_CTL_ELEM_TYPE_IEC958``, and the size of element is +fixed as 4 bytes array (value.iec958.status[x]). For the ``info`` +callback, you don't specify the value field for this type (the count +field must be set, though). + +“IEC958 Playback Con Mask” is used to return the bit-mask for the IEC958 +status bits of consumer mode. Similarly, “IEC958 Playback Pro Mask” +returns the bitmask for professional mode. They are read-only controls, +and are defined as MIXER controls (iface = +``SNDRV_CTL_ELEM_IFACE_MIXER``). + +Meanwhile, “IEC958 Playback Default” control is defined for getting and +setting the current default IEC958 bits. Note that this one is usually +defined as a PCM control (iface = ``SNDRV_CTL_ELEM_IFACE_PCM``), +although in some places it's defined as a MIXER control. + +In addition, you can define the control switches to enable/disable or to +set the raw bit mode. The implementation will depend on the chip, but +the control should be named as “IEC958 xxx”, preferably using the +:c:func:`SNDRV_CTL_NAME_IEC958()` macro. + +You can find several cases, for example, ``pci/emu10k1``, +``pci/ice1712``, or ``pci/cmipci.c``. + +Buffer and Memory Management +============================ + +Buffer Types +------------ + +ALSA provides several different buffer allocation functions depending on +the bus and the architecture. All these have a consistent API. The +allocation of physically-contiguous pages is done via +:c:func:`snd_malloc_xxx_pages()` function, where xxx is the bus +type. + +The allocation of pages with fallback is +:c:func:`snd_malloc_xxx_pages_fallback()`. This function tries +to allocate the specified pages but if the pages are not available, it +tries to reduce the page sizes until enough space is found. + +The release the pages, call :c:func:`snd_free_xxx_pages()` +function. + +Usually, ALSA drivers try to allocate and reserve a large contiguous +physical space at the time the module is loaded for the later use. This +is called “pre-allocation”. As already written, you can call the +following function at pcm instance construction time (in the case of PCI +bus). + +:: + + snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, + snd_dma_pci_data(pci), size, max); + +where ``size`` is the byte size to be pre-allocated and the ``max`` is +the maximum size to be changed via the ``prealloc`` proc file. The +allocator will try to get an area as large as possible within the +given size. + +The second argument (type) and the third argument (device pointer) are +dependent on the bus. In the case of the ISA bus, pass +:c:func:`snd_dma_isa_data()` as the third argument with +``SNDRV_DMA_TYPE_DEV`` type. For the continuous buffer unrelated to the +bus can be pre-allocated with ``SNDRV_DMA_TYPE_CONTINUOUS`` type and the +``snd_dma_continuous_data(GFP_KERNEL)`` device pointer, where +``GFP_KERNEL`` is the kernel allocation flag to use. For the PCI +scatter-gather buffers, use ``SNDRV_DMA_TYPE_DEV_SG`` with +``snd_dma_pci_data(pci)`` (see the `Non-Contiguous Buffers`_ +section). + +Once the buffer is pre-allocated, you can use the allocator in the +``hw_params`` callback: + +:: + + snd_pcm_lib_malloc_pages(substream, size); + +Note that you have to pre-allocate to use this function. + +External Hardware Buffers +------------------------- + +Some chips have their own hardware buffers and the DMA transfer from the +host memory is not available. In such a case, you need to either 1) +copy/set the audio data directly to the external hardware buffer, or 2) +make an intermediate buffer and copy/set the data from it to the +external hardware buffer in interrupts (or in tasklets, preferably). + +The first case works fine if the external hardware buffer is large +enough. This method doesn't need any extra buffers and thus is more +effective. You need to define the ``copy`` and ``silence`` callbacks +for the data transfer. However, there is a drawback: it cannot be +mmapped. The examples are GUS's GF1 PCM or emu8000's wavetable PCM. + +The second case allows for mmap on the buffer, although you have to +handle an interrupt or a tasklet to transfer the data from the +intermediate buffer to the hardware buffer. You can find an example in +the vxpocket driver. + +Another case is when the chip uses a PCI memory-map region for the +buffer instead of the host memory. In this case, mmap is available only +on certain architectures like the Intel one. In non-mmap mode, the data +cannot be transferred as in the normal way. Thus you need to define the +``copy`` and ``silence`` callbacks as well, as in the cases above. The +examples are found in ``rme32.c`` and ``rme96.c``. + +The implementation of the ``copy`` and ``silence`` callbacks depends +upon whether the hardware supports interleaved or non-interleaved +samples. The ``copy`` callback is defined like below, a bit +differently depending whether the direction is playback or capture: + +:: + + static int playback_copy(struct snd_pcm_substream *substream, int channel, + snd_pcm_uframes_t pos, void *src, snd_pcm_uframes_t count); + static int capture_copy(struct snd_pcm_substream *substream, int channel, + snd_pcm_uframes_t pos, void *dst, snd_pcm_uframes_t count); + +In the case of interleaved samples, the second argument (``channel``) is +not used. The third argument (``pos``) points the current position +offset in frames. + +The meaning of the fourth argument is different between playback and +capture. For playback, it holds the source data pointer, and for +capture, it's the destination data pointer. + +The last argument is the number of frames to be copied. + +What you have to do in this callback is again different between playback +and capture directions. In the playback case, you copy the given amount +of data (``count``) at the specified pointer (``src``) to the specified +offset (``pos``) on the hardware buffer. When coded like memcpy-like +way, the copy would be like: + +:: + + my_memcpy(my_buffer + frames_to_bytes(runtime, pos), src, + frames_to_bytes(runtime, count)); + +For the capture direction, you copy the given amount of data (``count``) +at the specified offset (``pos``) on the hardware buffer to the +specified pointer (``dst``). + +:: + + my_memcpy(dst, my_buffer + frames_to_bytes(runtime, pos), + frames_to_bytes(runtime, count)); + +Note that both the position and the amount of data are given in frames. + +In the case of non-interleaved samples, the implementation will be a bit +more complicated. + +You need to check the channel argument, and if it's -1, copy the whole +channels. Otherwise, you have to copy only the specified channel. Please +check ``isa/gus/gus_pcm.c`` as an example. + +The ``silence`` callback is also implemented in a similar way + +:: + + static int silence(struct snd_pcm_substream *substream, int channel, + snd_pcm_uframes_t pos, snd_pcm_uframes_t count); + +The meanings of arguments are the same as in the ``copy`` callback, +although there is no ``src/dst`` argument. In the case of interleaved +samples, the channel argument has no meaning, as well as on ``copy`` +callback. + +The role of ``silence`` callback is to set the given amount +(``count``) of silence data at the specified offset (``pos``) on the +hardware buffer. Suppose that the data format is signed (that is, the +silent-data is 0), and the implementation using a memset-like function +would be like: + +:: + + my_memcpy(my_buffer + frames_to_bytes(runtime, pos), 0, + frames_to_bytes(runtime, count)); + +In the case of non-interleaved samples, again, the implementation +becomes a bit more complicated. See, for example, ``isa/gus/gus_pcm.c``. + +Non-Contiguous Buffers +---------------------- + +If your hardware supports the page table as in emu10k1 or the buffer +descriptors as in via82xx, you can use the scatter-gather (SG) DMA. ALSA +provides an interface for handling SG-buffers. The API is provided in +````. + +For creating the SG-buffer handler, call +:c:func:`snd_pcm_lib_preallocate_pages()` or +:c:func:`snd_pcm_lib_preallocate_pages_for_all()` with +``SNDRV_DMA_TYPE_DEV_SG`` in the PCM constructor like other PCI +pre-allocator. You need to pass ``snd_dma_pci_data(pci)``, where pci is +the :c:type:`struct pci_dev ` pointer of the chip as +well. The ``struct snd_sg_buf`` instance is created as +``substream->dma_private``. You can cast the pointer like: + +:: + + struct snd_sg_buf *sgbuf = (struct snd_sg_buf *)substream->dma_private; + +Then call :c:func:`snd_pcm_lib_malloc_pages()` in the ``hw_params`` +callback as well as in the case of normal PCI buffer. The SG-buffer +handler will allocate the non-contiguous kernel pages of the given size +and map them onto the virtually contiguous memory. The virtual pointer +is addressed in runtime->dma_area. The physical address +(``runtime->dma_addr``) is set to zero, because the buffer is +physically non-contiguous. The physical address table is set up in +``sgbuf->table``. You can get the physical address at a certain offset +via :c:func:`snd_pcm_sgbuf_get_addr()`. + +When a SG-handler is used, you need to set +:c:func:`snd_pcm_sgbuf_ops_page()` as the ``page`` callback. (See +`page callback`_ section.) + +To release the data, call :c:func:`snd_pcm_lib_free_pages()` in +the ``hw_free`` callback as usual. + +Vmalloc'ed Buffers +------------------ + +It's possible to use a buffer allocated via :c:func:`vmalloc()`, for +example, for an intermediate buffer. Since the allocated pages are not +contiguous, you need to set the ``page`` callback to obtain the physical +address at every offset. + +The implementation of ``page`` callback would be like this: + +:: + + #include + + /* get the physical page pointer on the given offset */ + static struct page *mychip_page(struct snd_pcm_substream *substream, + unsigned long offset) + { + void *pageptr = substream->runtime->dma_area + offset; + return vmalloc_to_page(pageptr); + } + +Proc Interface +============== + +ALSA provides an easy interface for procfs. The proc files are very +useful for debugging. I recommend you set up proc files if you write a +driver and want to get a running status or register dumps. The API is +found in ````. + +To create a proc file, call :c:func:`snd_card_proc_new()`. + +:: + + struct snd_info_entry *entry; + int err = snd_card_proc_new(card, "my-file", &entry); + +where the second argument specifies the name of the proc file to be +created. The above example will create a file ``my-file`` under the +card directory, e.g. ``/proc/asound/card0/my-file``. + +Like other components, the proc entry created via +:c:func:`snd_card_proc_new()` will be registered and released +automatically in the card registration and release functions. + +When the creation is successful, the function stores a new instance in +the pointer given in the third argument. It is initialized as a text +proc file for read only. To use this proc file as a read-only text file +as it is, set the read callback with a private data via +:c:func:`snd_info_set_text_ops()`. + +:: + + snd_info_set_text_ops(entry, chip, my_proc_read); + +where the second argument (``chip``) is the private data to be used in +the callbacks. The third parameter specifies the read buffer size and +the fourth (``my_proc_read``) is the callback function, which is +defined like + +:: + + static void my_proc_read(struct snd_info_entry *entry, + struct snd_info_buffer *buffer); + +In the read callback, use :c:func:`snd_iprintf()` for output +strings, which works just like normal :c:func:`printf()`. For +example, + +:: + + static void my_proc_read(struct snd_info_entry *entry, + struct snd_info_buffer *buffer) + { + struct my_chip *chip = entry->private_data; + + snd_iprintf(buffer, "This is my chip!\n"); + snd_iprintf(buffer, "Port = %ld\n", chip->port); + } + +The file permissions can be changed afterwards. As default, it's set as +read only for all users. If you want to add write permission for the +user (root as default), do as follows: + +:: + + entry->mode = S_IFREG | S_IRUGO | S_IWUSR; + +and set the write buffer size and the callback + +:: + + entry->c.text.write = my_proc_write; + +For the write callback, you can use :c:func:`snd_info_get_line()` +to get a text line, and :c:func:`snd_info_get_str()` to retrieve +a string from the line. Some examples are found in +``core/oss/mixer_oss.c``, core/oss/and ``pcm_oss.c``. + +For a raw-data proc-file, set the attributes as follows: + +:: + + static struct snd_info_entry_ops my_file_io_ops = { + .read = my_file_io_read, + }; + + entry->content = SNDRV_INFO_CONTENT_DATA; + entry->private_data = chip; + entry->c.ops = &my_file_io_ops; + entry->size = 4096; + entry->mode = S_IFREG | S_IRUGO; + +For the raw data, ``size`` field must be set properly. This specifies +the maximum size of the proc file access. + +The read/write callbacks of raw mode are more direct than the text mode. +You need to use a low-level I/O functions such as +:c:func:`copy_from/to_user()` to transfer the data. + +:: + + static ssize_t my_file_io_read(struct snd_info_entry *entry, + void *file_private_data, + struct file *file, + char *buf, + size_t count, + loff_t pos) + { + if (copy_to_user(buf, local_data + pos, count)) + return -EFAULT; + return count; + } + +If the size of the info entry has been set up properly, ``count`` and +``pos`` are guaranteed to fit within 0 and the given size. You don't +have to check the range in the callbacks unless any other condition is +required. + +Power Management +================ + +If the chip is supposed to work with suspend/resume functions, you need +to add power-management code to the driver. The additional code for +power-management should be ifdef-ed with ``CONFIG_PM``. + +If the driver *fully* supports suspend/resume that is, the device can be +properly resumed to its state when suspend was called, you can set the +``SNDRV_PCM_INFO_RESUME`` flag in the pcm info field. Usually, this is +possible when the registers of the chip can be safely saved and restored +to RAM. If this is set, the trigger callback is called with +``SNDRV_PCM_TRIGGER_RESUME`` after the resume callback completes. + +Even if the driver doesn't support PM fully but partial suspend/resume +is still possible, it's still worthy to implement suspend/resume +callbacks. In such a case, applications would reset the status by +calling :c:func:`snd_pcm_prepare()` and restart the stream +appropriately. Hence, you can define suspend/resume callbacks below but +don't set ``SNDRV_PCM_INFO_RESUME`` info flag to the PCM. + +Note that the trigger with SUSPEND can always be called when +:c:func:`snd_pcm_suspend_all()` is called, regardless of the +``SNDRV_PCM_INFO_RESUME`` flag. The ``RESUME`` flag affects only the +behavior of :c:func:`snd_pcm_resume()`. (Thus, in theory, +``SNDRV_PCM_TRIGGER_RESUME`` isn't needed to be handled in the trigger +callback when no ``SNDRV_PCM_INFO_RESUME`` flag is set. But, it's better +to keep it for compatibility reasons.) + +In the earlier version of ALSA drivers, a common power-management layer +was provided, but it has been removed. The driver needs to define the +suspend/resume hooks according to the bus the device is connected to. In +the case of PCI drivers, the callbacks look like below: + +:: + + #ifdef CONFIG_PM + static int snd_my_suspend(struct pci_dev *pci, pm_message_t state) + { + .... /* do things for suspend */ + return 0; + } + static int snd_my_resume(struct pci_dev *pci) + { + .... /* do things for suspend */ + return 0; + } + #endif + +The scheme of the real suspend job is as follows. + +1. Retrieve the card and the chip data. + +2. Call :c:func:`snd_power_change_state()` with + ``SNDRV_CTL_POWER_D3hot`` to change the power status. + +3. Call :c:func:`snd_pcm_suspend_all()` to suspend the running + PCM streams. + +4. If AC97 codecs are used, call :c:func:`snd_ac97_suspend()` for + each codec. + +5. Save the register values if necessary. + +6. Stop the hardware if necessary. + +7. Disable the PCI device by calling + :c:func:`pci_disable_device()`. Then, call + :c:func:`pci_save_state()` at last. + +A typical code would be like: + +:: + + static int mychip_suspend(struct pci_dev *pci, pm_message_t state) + { + /* (1) */ + struct snd_card *card = pci_get_drvdata(pci); + struct mychip *chip = card->private_data; + /* (2) */ + snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); + /* (3) */ + snd_pcm_suspend_all(chip->pcm); + /* (4) */ + snd_ac97_suspend(chip->ac97); + /* (5) */ + snd_mychip_save_registers(chip); + /* (6) */ + snd_mychip_stop_hardware(chip); + /* (7) */ + pci_disable_device(pci); + pci_save_state(pci); + return 0; + } + + +The scheme of the real resume job is as follows. + +1. Retrieve the card and the chip data. + +2. Set up PCI. First, call :c:func:`pci_restore_state()`. Then + enable the pci device again by calling + :c:func:`pci_enable_device()`. Call + :c:func:`pci_set_master()` if necessary, too. + +3. Re-initialize the chip. + +4. Restore the saved registers if necessary. + +5. Resume the mixer, e.g. calling :c:func:`snd_ac97_resume()`. + +6. Restart the hardware (if any). + +7. Call :c:func:`snd_power_change_state()` with + ``SNDRV_CTL_POWER_D0`` to notify the processes. + +A typical code would be like: + +:: + + static int mychip_resume(struct pci_dev *pci) + { + /* (1) */ + struct snd_card *card = pci_get_drvdata(pci); + struct mychip *chip = card->private_data; + /* (2) */ + pci_restore_state(pci); + pci_enable_device(pci); + pci_set_master(pci); + /* (3) */ + snd_mychip_reinit_chip(chip); + /* (4) */ + snd_mychip_restore_registers(chip); + /* (5) */ + snd_ac97_resume(chip->ac97); + /* (6) */ + snd_mychip_restart_chip(chip); + /* (7) */ + snd_power_change_state(card, SNDRV_CTL_POWER_D0); + return 0; + } + +As shown in the above, it's better to save registers after suspending +the PCM operations via :c:func:`snd_pcm_suspend_all()` or +:c:func:`snd_pcm_suspend()`. It means that the PCM streams are +already stopped when the register snapshot is taken. But, remember that +you don't have to restart the PCM stream in the resume callback. It'll +be restarted via trigger call with ``SNDRV_PCM_TRIGGER_RESUME`` when +necessary. + +OK, we have all callbacks now. Let's set them up. In the initialization +of the card, make sure that you can get the chip data from the card +instance, typically via ``private_data`` field, in case you created the +chip data individually. + +:: + + static int snd_mychip_probe(struct pci_dev *pci, + const struct pci_device_id *pci_id) + { + .... + struct snd_card *card; + struct mychip *chip; + int err; + .... + err = snd_card_new(&pci->dev, index[dev], id[dev], THIS_MODULE, + 0, &card); + .... + chip = kzalloc(sizeof(*chip), GFP_KERNEL); + .... + card->private_data = chip; + .... + } + +When you created the chip data with :c:func:`snd_card_new()`, it's +anyway accessible via ``private_data`` field. + +:: + + static int snd_mychip_probe(struct pci_dev *pci, + const struct pci_device_id *pci_id) + { + .... + struct snd_card *card; + struct mychip *chip; + int err; + .... + err = snd_card_new(&pci->dev, index[dev], id[dev], THIS_MODULE, + sizeof(struct mychip), &card); + .... + chip = card->private_data; + .... + } + +If you need a space to save the registers, allocate the buffer for it +here, too, since it would be fatal if you cannot allocate a memory in +the suspend phase. The allocated buffer should be released in the +corresponding destructor. + +And next, set suspend/resume callbacks to the pci_driver. + +:: + + static struct pci_driver driver = { + .name = KBUILD_MODNAME, + .id_table = snd_my_ids, + .probe = snd_my_probe, + .remove = snd_my_remove, + #ifdef CONFIG_PM + .suspend = snd_my_suspend, + .resume = snd_my_resume, + #endif + }; + +Module Parameters +================= + +There are standard module options for ALSA. At least, each module should +have the ``index``, ``id`` and ``enable`` options. + +If the module supports multiple cards (usually up to 8 = ``SNDRV_CARDS`` +cards), they should be arrays. The default initial values are defined +already as constants for easier programming: + +:: + + static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; + static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; + static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; + +If the module supports only a single card, they could be single +variables, instead. ``enable`` option is not always necessary in this +case, but it would be better to have a dummy option for compatibility. + +The module parameters must be declared with the standard +``module_param()()``, ``module_param_array()()`` and +:c:func:`MODULE_PARM_DESC()` macros. + +The typical coding would be like below: + +:: + + #define CARD_NAME "My Chip" + + module_param_array(index, int, NULL, 0444); + MODULE_PARM_DESC(index, "Index value for " CARD_NAME " soundcard."); + module_param_array(id, charp, NULL, 0444); + MODULE_PARM_DESC(id, "ID string for " CARD_NAME " soundcard."); + module_param_array(enable, bool, NULL, 0444); + MODULE_PARM_DESC(enable, "Enable " CARD_NAME " soundcard."); + +Also, don't forget to define the module description, classes, license +and devices. Especially, the recent modprobe requires to define the +module license as GPL, etc., otherwise the system is shown as “tainted”. + +:: + + MODULE_DESCRIPTION("My Chip"); + MODULE_LICENSE("GPL"); + MODULE_SUPPORTED_DEVICE("{{Vendor,My Chip Name}}"); + + +How To Put Your Driver Into ALSA Tree +===================================== + +General +------- + +So far, you've learned how to write the driver codes. And you might have +a question now: how to put my own driver into the ALSA driver tree? Here +(finally :) the standard procedure is described briefly. + +Suppose that you create a new PCI driver for the card “xyz”. The card +module name would be snd-xyz. The new driver is usually put into the +alsa-driver tree, ``alsa-driver/pci`` directory in the case of PCI +cards. Then the driver is evaluated, audited and tested by developers +and users. After a certain time, the driver will go to the alsa-kernel +tree (to the corresponding directory, such as ``alsa-kernel/pci``) and +eventually will be integrated into the Linux 2.6 tree (the directory +would be ``linux/sound/pci``). + +In the following sections, the driver code is supposed to be put into +alsa-driver tree. The two cases are covered: a driver consisting of a +single source file and one consisting of several source files. + +Driver with A Single Source File +-------------------------------- + +1. Modify alsa-driver/pci/Makefile + + Suppose you have a file xyz.c. Add the following two lines + +:: + + snd-xyz-objs := xyz.o + obj-$(CONFIG_SND_XYZ) += snd-xyz.o + +2. Create the Kconfig entry + + Add the new entry of Kconfig for your xyz driver. config SND_XYZ + tristate "Foobar XYZ" depends on SND select SND_PCM help Say Y here + to include support for Foobar XYZ soundcard. To compile this driver + as a module, choose M here: the module will be called snd-xyz. the + line, select SND_PCM, specifies that the driver xyz supports PCM. In + addition to SND_PCM, the following components are supported for + select command: SND_RAWMIDI, SND_TIMER, SND_HWDEP, + SND_MPU401_UART, SND_OPL3_LIB, SND_OPL4_LIB, SND_VX_LIB, + SND_AC97_CODEC. Add the select command for each supported + component. + + Note that some selections imply the lowlevel selections. For example, + PCM includes TIMER, MPU401_UART includes RAWMIDI, AC97_CODEC + includes PCM, and OPL3_LIB includes HWDEP. You don't need to give + the lowlevel selections again. + + For the details of Kconfig script, refer to the kbuild documentation. + +3. Run cvscompile script to re-generate the configure script and build + the whole stuff again. + +Drivers with Several Source Files +--------------------------------- + +Suppose that the driver snd-xyz have several source files. They are +located in the new subdirectory, pci/xyz. + +1. Add a new directory (``xyz``) in ``alsa-driver/pci/Makefile`` as + below + +:: + + obj-$(CONFIG_SND) += xyz/ + + +2. Under the directory ``xyz``, create a Makefile + +:: + + ifndef SND_TOPDIR + SND_TOPDIR=../.. + endif + + include $(SND_TOPDIR)/toplevel.config + include $(SND_TOPDIR)/Makefile.conf + + snd-xyz-objs := xyz.o abc.o def.o + + obj-$(CONFIG_SND_XYZ) += snd-xyz.o + + include $(SND_TOPDIR)/Rules.make + +3. Create the Kconfig entry + + This procedure is as same as in the last section. + +4. Run cvscompile script to re-generate the configure script and build + the whole stuff again. + +Useful Functions +================ + +:c:func:`snd_printk()` and friends +--------------------------------------- + +ALSA provides a verbose version of the :c:func:`printk()` function. +If a kernel config ``CONFIG_SND_VERBOSE_PRINTK`` is set, this function +prints the given message together with the file name and the line of the +caller. The ``KERN_XXX`` prefix is processed as well as the original +:c:func:`printk()` does, so it's recommended to add this prefix, +e.g. snd_printk(KERN_ERR "Oh my, sorry, it's extremely bad!\\n"); + +There are also :c:func:`printk()`'s for debugging. +:c:func:`snd_printd()` can be used for general debugging purposes. +If ``CONFIG_SND_DEBUG`` is set, this function is compiled, and works +just like :c:func:`snd_printk()`. If the ALSA is compiled without +the debugging flag, it's ignored. + +:c:func:`snd_printdd()` is compiled in only when +``CONFIG_SND_DEBUG_VERBOSE`` is set. Please note that +``CONFIG_SND_DEBUG_VERBOSE`` is not set as default even if you configure +the alsa-driver with ``--with-debug=full`` option. You need to give +explicitly ``--with-debug=detect`` option instead. + +:c:func:`snd_BUG()` +------------------------ + +It shows the ``BUG?`` message and stack trace as well as +:c:func:`snd_BUG_ON()` at the point. It's useful to show that a +fatal error happens there. + +When no debug flag is set, this macro is ignored. + +:c:func:`snd_BUG_ON()` +---------------------------- + +:c:func:`snd_BUG_ON()` macro is similar with +:c:func:`WARN_ON()` macro. For example, snd_BUG_ON(!pointer); or +it can be used as the condition, if (snd_BUG_ON(non_zero_is_bug)) +return -EINVAL; + +The macro takes an conditional expression to evaluate. When +``CONFIG_SND_DEBUG``, is set, if the expression is non-zero, it shows +the warning message such as ``BUG? (xxx)`` normally followed by stack +trace. In both cases it returns the evaluated value. + +Acknowledgments +=============== + +I would like to thank Phil Kerr for his help for improvement and +corrections of this document. + +Kevin Conder reformatted the original plain-text to the DocBook format. + +Giuliano Pochini corrected typos and contributed the example codes in +the hardware constraints section. -- cgit v1.2.3-58-ga151 From 9000d69925ac45dcb346b5ea68e71b83cd897d3d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 9 Nov 2016 13:01:05 +0100 Subject: ALSA: doc: ReSTize HD-Audio document The original HD-Audio.txt was already in asciidoc format, so it's a simple conversion in the end. A new subdirectory, Documentation/sound/hd-audio, is created and the document is moved there with another file name to match better with the recent Documentation tree structure. Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/HD-Audio.txt | 853 -------------------------------- Documentation/sound/hd-audio/index.rst | 7 + Documentation/sound/hd-audio/notes.rst | 880 +++++++++++++++++++++++++++++++++ Documentation/sound/index.rst | 1 + 4 files changed, 888 insertions(+), 853 deletions(-) delete mode 100644 Documentation/sound/alsa/HD-Audio.txt create mode 100644 Documentation/sound/hd-audio/index.rst create mode 100644 Documentation/sound/hd-audio/notes.rst (limited to 'Documentation/sound') diff --git a/Documentation/sound/alsa/HD-Audio.txt b/Documentation/sound/alsa/HD-Audio.txt deleted file mode 100644 index d4510ebf2e8c..000000000000 --- a/Documentation/sound/alsa/HD-Audio.txt +++ /dev/null @@ -1,853 +0,0 @@ -MORE NOTES ON HD-AUDIO DRIVER -============================= - Takashi Iwai - - -GENERAL -------- - -HD-audio is the new standard on-board audio component on modern PCs -after AC97. Although Linux has been supporting HD-audio since long -time ago, there are often problems with new machines. A part of the -problem is broken BIOS, and the rest is the driver implementation. -This document explains the brief trouble-shooting and debugging -methods for the HD-audio hardware. - -The HD-audio component consists of two parts: the controller chip and -the codec chips on the HD-audio bus. Linux provides a single driver -for all controllers, snd-hda-intel. Although the driver name contains -a word of a well-known hardware vendor, it's not specific to it but for -all controller chips by other companies. Since the HD-audio -controllers are supposed to be compatible, the single snd-hda-driver -should work in most cases. But, not surprisingly, there are known -bugs and issues specific to each controller type. The snd-hda-intel -driver has a bunch of workarounds for these as described below. - -A controller may have multiple codecs. Usually you have one audio -codec and optionally one modem codec. In theory, there might be -multiple audio codecs, e.g. for analog and digital outputs, and the -driver might not work properly because of conflict of mixer elements. -This should be fixed in future if such hardware really exists. - -The snd-hda-intel driver has several different codec parsers depending -on the codec. It has a generic parser as a fallback, but this -functionality is fairly limited until now. Instead of the generic -parser, usually the codec-specific parser (coded in patch_*.c) is used -for the codec-specific implementations. The details about the -codec-specific problems are explained in the later sections. - -If you are interested in the deep debugging of HD-audio, read the -HD-audio specification at first. The specification is found on -Intel's web page, for example: - -- http://www.intel.com/standards/hdaudio/ - - -HD-AUDIO CONTROLLER -------------------- - -DMA-Position Problem -~~~~~~~~~~~~~~~~~~~~ -The most common problem of the controller is the inaccurate DMA -pointer reporting. The DMA pointer for playback and capture can be -read in two ways, either via a LPIB register or via a position-buffer -map. As default the driver tries to read from the io-mapped -position-buffer, and falls back to LPIB if the position-buffer appears -dead. However, this detection isn't perfect on some devices. In such -a case, you can change the default method via `position_fix` option. - -`position_fix=1` means to use LPIB method explicitly. -`position_fix=2` means to use the position-buffer. -`position_fix=3` means to use a combination of both methods, needed -for some VIA controllers. The capture stream position is corrected -by comparing both LPIB and position-buffer values. -`position_fix=4` is another combination available for all controllers, -and uses LPIB for the playback and the position-buffer for the capture -streams. -0 is the default value for all other -controllers, the automatic check and fallback to LPIB as described in -the above. If you get a problem of repeated sounds, this option might -help. - -In addition to that, every controller is known to be broken regarding -the wake-up timing. It wakes up a few samples before actually -processing the data on the buffer. This caused a lot of problems, for -example, with ALSA dmix or JACK. Since 2.6.27 kernel, the driver puts -an artificial delay to the wake up timing. This delay is controlled -via `bdl_pos_adj` option. - -When `bdl_pos_adj` is a negative value (as default), it's assigned to -an appropriate value depending on the controller chip. For Intel -chips, it'd be 1 while it'd be 32 for others. Usually this works. -Only in case it doesn't work and you get warning messages, you should -change this parameter to other values. - - -Codec-Probing Problem -~~~~~~~~~~~~~~~~~~~~~ -A less often but a more severe problem is the codec probing. When -BIOS reports the available codec slots wrongly, the driver gets -confused and tries to access the non-existing codec slot. This often -results in the total screw-up, and destructs the further communication -with the codec chips. The symptom appears usually as error messages -like: ------------------------------------------------------------------------- - hda_intel: azx_get_response timeout, switching to polling mode: - last cmd=0x12345678 - hda_intel: azx_get_response timeout, switching to single_cmd mode: - last cmd=0x12345678 ------------------------------------------------------------------------- - -The first line is a warning, and this is usually relatively harmless. -It means that the codec response isn't notified via an IRQ. The -driver uses explicit polling method to read the response. It gives -very slight CPU overhead, but you'd unlikely notice it. - -The second line is, however, a fatal error. If this happens, usually -it means that something is really wrong. Most likely you are -accessing a non-existing codec slot. - -Thus, if the second error message appears, try to narrow the probed -codec slots via `probe_mask` option. It's a bitmask, and each bit -corresponds to the codec slot. For example, to probe only the first -slot, pass `probe_mask=1`. For the first and the third slots, pass -`probe_mask=5` (where 5 = 1 | 4), and so on. - -Since 2.6.29 kernel, the driver has a more robust probing method, so -this error might happen rarely, though. - -On a machine with a broken BIOS, sometimes you need to force the -driver to probe the codec slots the hardware doesn't report for use. -In such a case, turn the bit 8 (0x100) of `probe_mask` option on. -Then the rest 8 bits are passed as the codec slots to probe -unconditionally. For example, `probe_mask=0x103` will force to probe -the codec slots 0 and 1 no matter what the hardware reports. - - -Interrupt Handling -~~~~~~~~~~~~~~~~~~ -HD-audio driver uses MSI as default (if available) since 2.6.33 -kernel as MSI works better on some machines, and in general, it's -better for performance. However, Nvidia controllers showed bad -regressions with MSI (especially in a combination with AMD chipset), -thus we disabled MSI for them. - -There seem also still other devices that don't work with MSI. If you -see a regression wrt the sound quality (stuttering, etc) or a lock-up -in the recent kernel, try to pass `enable_msi=0` option to disable -MSI. If it works, you can add the known bad device to the blacklist -defined in hda_intel.c. In such a case, please report and give the -patch back to the upstream developer. - - -HD-AUDIO CODEC --------------- - -Model Option -~~~~~~~~~~~~ -The most common problem regarding the HD-audio driver is the -unsupported codec features or the mismatched device configuration. -Most of codec-specific code has several preset models, either to -override the BIOS setup or to provide more comprehensive features. - -The driver checks PCI SSID and looks through the static configuration -table until any matching entry is found. If you have a new machine, -you may see a message like below: ------------------------------------------------------------------------- - hda_codec: ALC880: BIOS auto-probing. ------------------------------------------------------------------------- -Meanwhile, in the earlier versions, you would see a message like: ------------------------------------------------------------------------- - hda_codec: Unknown model for ALC880, trying auto-probe from BIOS... ------------------------------------------------------------------------- -Even if you see such a message, DON'T PANIC. Take a deep breath and -keep your towel. First of all, it's an informational message, no -warning, no error. This means that the PCI SSID of your device isn't -listed in the known preset model (white-)list. But, this doesn't mean -that the driver is broken. Many codec-drivers provide the automatic -configuration mechanism based on the BIOS setup. - -The HD-audio codec has usually "pin" widgets, and BIOS sets the default -configuration of each pin, which indicates the location, the -connection type, the jack color, etc. The HD-audio driver can guess -the right connection judging from these default configuration values. -However -- some codec-support codes, such as patch_analog.c, don't -support the automatic probing (yet as of 2.6.28). And, BIOS is often, -yes, pretty often broken. It sets up wrong values and screws up the -driver. - -The preset model (or recently called as "fix-up") is provided -basically to overcome such a situation. When the matching preset -model is found in the white-list, the driver assumes the static -configuration of that preset with the correct pin setup, etc. -Thus, if you have a newer machine with a slightly different PCI SSID -(or codec SSID) from the existing one, you may have a good chance to -re-use the same model. You can pass the `model` option to specify the -preset model instead of PCI (and codec-) SSID look-up. - -What `model` option values are available depends on the codec chip. -Check your codec chip from the codec proc file (see "Codec Proc-File" -section below). It will show the vendor/product name of your codec -chip. Then, see Documentation/sound/alsa/HD-Audio-Models.txt file, -the section of HD-audio driver. You can find a list of codecs -and `model` options belonging to each codec. For example, for Realtek -ALC262 codec chip, pass `model=ultra` for devices that are compatible -with Samsung Q1 Ultra. - -Thus, the first thing you can do for any brand-new, unsupported and -non-working HD-audio hardware is to check HD-audio codec and several -different `model` option values. If you have any luck, some of them -might suit with your device well. - -There are a few special model option values: -- when 'nofixup' is passed, the device-specific fixups in the codec - parser are skipped. -- when `generic` is passed, the codec-specific parser is skipped and - only the generic parser is used. - - -Speaker and Headphone Output -~~~~~~~~~~~~~~~~~~~~~~~~~~~~ -One of the most frequent (and obvious) bugs with HD-audio is the -silent output from either or both of a built-in speaker and a -headphone jack. In general, you should try a headphone output at -first. A speaker output often requires more additional controls like -the external amplifier bits. Thus a headphone output has a slightly -better chance. - -Before making a bug report, double-check whether the mixer is set up -correctly. The recent version of snd-hda-intel driver provides mostly -"Master" volume control as well as "Front" volume (where Front -indicates the front-channels). In addition, there can be individual -"Headphone" and "Speaker" controls. - -Ditto for the speaker output. There can be "External Amplifier" -switch on some codecs. Turn on this if present. - -Another related problem is the automatic mute of speaker output by -headphone plugging. This feature is implemented in most cases, but -not on every preset model or codec-support code. - -In anyway, try a different model option if you have such a problem. -Some other models may match better and give you more matching -functionality. If none of the available models works, send a bug -report. See the bug report section for details. - -If you are masochistic enough to debug the driver problem, note the -following: - -- The speaker (and the headphone, too) output often requires the - external amplifier. This can be set usually via EAPD verb or a - certain GPIO. If the codec pin supports EAPD, you have a better - chance via SET_EAPD_BTL verb (0x70c). On others, GPIO pin (mostly - it's either GPIO0 or GPIO1) may turn on/off EAPD. -- Some Realtek codecs require special vendor-specific coefficients to - turn on the amplifier. See patch_realtek.c. -- IDT codecs may have extra power-enable/disable controls on each - analog pin. See patch_sigmatel.c. -- Very rare but some devices don't accept the pin-detection verb until - triggered. Issuing GET_PIN_SENSE verb (0xf09) may result in the - codec-communication stall. Some examples are found in - patch_realtek.c. - - -Capture Problems -~~~~~~~~~~~~~~~~ -The capture problems are often because of missing setups of mixers. -Thus, before submitting a bug report, make sure that you set up the -mixer correctly. For example, both "Capture Volume" and "Capture -Switch" have to be set properly in addition to the right "Capture -Source" or "Input Source" selection. Some devices have "Mic Boost" -volume or switch. - -When the PCM device is opened via "default" PCM (without pulse-audio -plugin), you'll likely have "Digital Capture Volume" control as well. -This is provided for the extra gain/attenuation of the signal in -software, especially for the inputs without the hardware volume -control such as digital microphones. Unless really needed, this -should be set to exactly 50%, corresponding to 0dB -- neither extra -gain nor attenuation. When you use "hw" PCM, i.e., a raw access PCM, -this control will have no influence, though. - -It's known that some codecs / devices have fairly bad analog circuits, -and the recorded sound contains a certain DC-offset. This is no bug -of the driver. - -Most of modern laptops have no analog CD-input connection. Thus, the -recording from CD input won't work in many cases although the driver -provides it as the capture source. Use CDDA instead. - -The automatic switching of the built-in and external mic per plugging -is implemented on some codec models but not on every model. Partly -because of my laziness but mostly lack of testers. Feel free to -submit the improvement patch to the author. - - -Direct Debugging -~~~~~~~~~~~~~~~~ -If no model option gives you a better result, and you are a tough guy -to fight against evil, try debugging via hitting the raw HD-audio -codec verbs to the device. Some tools are available: hda-emu and -hda-analyzer. The detailed description is found in the sections -below. You'd need to enable hwdep for using these tools. See "Kernel -Configuration" section. - - -OTHER ISSUES ------------- - -Kernel Configuration -~~~~~~~~~~~~~~~~~~~~ -In general, I recommend you to enable the sound debug option, -`CONFIG_SND_DEBUG=y`, no matter whether you are debugging or not. -This enables snd_printd() macro and others, and you'll get additional -kernel messages at probing. - -In addition, you can enable `CONFIG_SND_DEBUG_VERBOSE=y`. But this -will give you far more messages. Thus turn this on only when you are -sure to want it. - -Don't forget to turn on the appropriate `CONFIG_SND_HDA_CODEC_*` -options. Note that each of them corresponds to the codec chip, not -the controller chip. Thus, even if lspci shows the Nvidia controller, -you may need to choose the option for other vendors. If you are -unsure, just select all yes. - -`CONFIG_SND_HDA_HWDEP` is a useful option for debugging the driver. -When this is enabled, the driver creates hardware-dependent devices -(one per each codec), and you have a raw access to the device via -these device files. For example, `hwC0D2` will be created for the -codec slot #2 of the first card (#0). For debug-tools such as -hda-verb and hda-analyzer, the hwdep device has to be enabled. -Thus, it'd be better to turn this on always. - -`CONFIG_SND_HDA_RECONFIG` is a new option, and this depends on the -hwdep option above. When enabled, you'll have some sysfs files under -the corresponding hwdep directory. See "HD-audio reconfiguration" -section below. - -`CONFIG_SND_HDA_POWER_SAVE` option enables the power-saving feature. -See "Power-saving" section below. - - -Codec Proc-File -~~~~~~~~~~~~~~~ -The codec proc-file is a treasure-chest for debugging HD-audio. -It shows most of useful information of each codec widget. - -The proc file is located in /proc/asound/card*/codec#*, one file per -each codec slot. You can know the codec vendor, product id and -names, the type of each widget, capabilities and so on. -This file, however, doesn't show the jack sensing state, so far. This -is because the jack-sensing might be depending on the trigger state. - -This file will be picked up by the debug tools, and also it can be fed -to the emulator as the primary codec information. See the debug tools -section below. - -This proc file can be also used to check whether the generic parser is -used. When the generic parser is used, the vendor/product ID name -will appear as "Realtek ID 0262", instead of "Realtek ALC262". - - -HD-Audio Reconfiguration -~~~~~~~~~~~~~~~~~~~~~~~~ -This is an experimental feature to allow you re-configure the HD-audio -codec dynamically without reloading the driver. The following sysfs -files are available under each codec-hwdep device directory (e.g. -/sys/class/sound/hwC0D0): - -vendor_id:: - Shows the 32bit codec vendor-id hex number. You can change the - vendor-id value by writing to this file. -subsystem_id:: - Shows the 32bit codec subsystem-id hex number. You can change the - subsystem-id value by writing to this file. -revision_id:: - Shows the 32bit codec revision-id hex number. You can change the - revision-id value by writing to this file. -afg:: - Shows the AFG ID. This is read-only. -mfg:: - Shows the MFG ID. This is read-only. -name:: - Shows the codec name string. Can be changed by writing to this - file. -modelname:: - Shows the currently set `model` option. Can be changed by writing - to this file. -init_verbs:: - The extra verbs to execute at initialization. You can add a verb by - writing to this file. Pass three numbers: nid, verb and parameter - (separated with a space). -hints:: - Shows / stores hint strings for codec parsers for any use. - Its format is `key = value`. For example, passing `jack_detect = no` - will disable the jack detection of the machine completely. -init_pin_configs:: - Shows the initial pin default config values set by BIOS. -driver_pin_configs:: - Shows the pin default values set by the codec parser explicitly. - This doesn't show all pin values but only the changed values by - the parser. That is, if the parser doesn't change the pin default - config values by itself, this will contain nothing. -user_pin_configs:: - Shows the pin default config values to override the BIOS setup. - Writing this (with two numbers, NID and value) appends the new - value. The given will be used instead of the initial BIOS value at - the next reconfiguration time. Note that this config will override - even the driver pin configs, too. -reconfig:: - Triggers the codec re-configuration. When any value is written to - this file, the driver re-initialize and parses the codec tree - again. All the changes done by the sysfs entries above are taken - into account. -clear:: - Resets the codec, removes the mixer elements and PCM stuff of the - specified codec, and clear all init verbs and hints. - -For example, when you want to change the pin default configuration -value of the pin widget 0x14 to 0x9993013f, and let the driver -re-configure based on that state, run like below: ------------------------------------------------------------------------- - # echo 0x14 0x9993013f > /sys/class/sound/hwC0D0/user_pin_configs - # echo 1 > /sys/class/sound/hwC0D0/reconfig ------------------------------------------------------------------------- - - -Hint Strings -~~~~~~~~~~~~ -The codec parser have several switches and adjustment knobs for -matching better with the actual codec or device behavior. Many of -them can be adjusted dynamically via "hints" strings as mentioned in -the section above. For example, by passing `jack_detect = no` string -via sysfs or a patch file, you can disable the jack detection, thus -the codec parser will skip the features like auto-mute or mic -auto-switch. As a boolean value, either `yes`, `no`, `true`, `false`, -`1` or `0` can be passed. - -The generic parser supports the following hints: - -- jack_detect (bool): specify whether the jack detection is available - at all on this machine; default true -- inv_jack_detect (bool): indicates that the jack detection logic is - inverted -- trigger_sense (bool): indicates that the jack detection needs the - explicit call of AC_VERB_SET_PIN_SENSE verb -- inv_eapd (bool): indicates that the EAPD is implemented in the - inverted logic -- pcm_format_first (bool): sets the PCM format before the stream tag - and channel ID -- sticky_stream (bool): keep the PCM format, stream tag and ID as long - as possible; default true -- spdif_status_reset (bool): reset the SPDIF status bits at each time - the SPDIF stream is set up -- pin_amp_workaround (bool): the output pin may have multiple amp - values -- single_adc_amp (bool): ADCs can have only single input amps -- auto_mute (bool): enable/disable the headphone auto-mute feature; - default true -- auto_mic (bool): enable/disable the mic auto-switch feature; default - true -- line_in_auto_switch (bool): enable/disable the line-in auto-switch - feature; default false -- need_dac_fix (bool): limits the DACs depending on the channel count -- primary_hp (bool): probe headphone jacks as the primary outputs; - default true -- multi_io (bool): try probing multi-I/O config (e.g. shared - line-in/surround, mic/clfe jacks) -- multi_cap_vol (bool): provide multiple capture volumes -- inv_dmic_split (bool): provide split internal mic volume/switch for - phase-inverted digital mics -- indep_hp (bool): provide the independent headphone PCM stream and - the corresponding mixer control, if available -- add_stereo_mix_input (bool): add the stereo mix (analog-loopback - mix) to the input mux if available -- add_jack_modes (bool): add "xxx Jack Mode" enum controls to each - I/O jack for allowing to change the headphone amp and mic bias VREF - capabilities -- power_save_node (bool): advanced power management for each widget, - controlling the power sate (D0/D3) of each widget node depending on - the actual pin and stream states -- power_down_unused (bool): power down the unused widgets, a subset of - power_save_node, and will be dropped in future -- add_hp_mic (bool): add the headphone to capture source if possible -- hp_mic_detect (bool): enable/disable the hp/mic shared input for a - single built-in mic case; default true -- mixer_nid (int): specifies the widget NID of the analog-loopback - mixer - - -Early Patching -~~~~~~~~~~~~~~ -When CONFIG_SND_HDA_PATCH_LOADER=y is set, you can pass a "patch" as a -firmware file for modifying the HD-audio setup before initializing the -codec. This can work basically like the reconfiguration via sysfs in -the above, but it does it before the first codec configuration. - -A patch file is a plain text file which looks like below: - ------------------------------------------------------------------------- - [codec] - 0x12345678 0xabcd1234 2 - - [model] - auto - - [pincfg] - 0x12 0x411111f0 - - [verb] - 0x20 0x500 0x03 - 0x20 0x400 0xff - - [hint] - jack_detect = no ------------------------------------------------------------------------- - -The file needs to have a line `[codec]`. The next line should contain -three numbers indicating the codec vendor-id (0x12345678 in the -example), the codec subsystem-id (0xabcd1234) and the address (2) of -the codec. The rest patch entries are applied to this specified codec -until another codec entry is given. Passing 0 or a negative number to -the first or the second value will make the check of the corresponding -field be skipped. It'll be useful for really broken devices that don't -initialize SSID properly. - -The `[model]` line allows to change the model name of the each codec. -In the example above, it will be changed to model=auto. -Note that this overrides the module option. - -After the `[pincfg]` line, the contents are parsed as the initial -default pin-configurations just like `user_pin_configs` sysfs above. -The values can be shown in user_pin_configs sysfs file, too. - -Similarly, the lines after `[verb]` are parsed as `init_verbs` -sysfs entries, and the lines after `[hint]` are parsed as `hints` -sysfs entries, respectively. - -Another example to override the codec vendor id from 0x12345678 to -0xdeadbeef is like below: ------------------------------------------------------------------------- - [codec] - 0x12345678 0xabcd1234 2 - - [vendor_id] - 0xdeadbeef ------------------------------------------------------------------------- - -In the similar way, you can override the codec subsystem_id via -`[subsystem_id]`, the revision id via `[revision_id]` line. -Also, the codec chip name can be rewritten via `[chip_name]` line. ------------------------------------------------------------------------- - [codec] - 0x12345678 0xabcd1234 2 - - [subsystem_id] - 0xffff1111 - - [revision_id] - 0x10 - - [chip_name] - My-own NEWS-0002 ------------------------------------------------------------------------- - -The hd-audio driver reads the file via request_firmware(). Thus, -a patch file has to be located on the appropriate firmware path, -typically, /lib/firmware. For example, when you pass the option -`patch=hda-init.fw`, the file /lib/firmware/hda-init.fw must be -present. - -The patch module option is specific to each card instance, and you -need to give one file name for each instance, separated by commas. -For example, if you have two cards, one for an on-board analog and one -for an HDMI video board, you may pass patch option like below: ------------------------------------------------------------------------- - options snd-hda-intel patch=on-board-patch,hdmi-patch ------------------------------------------------------------------------- - - -Power-Saving -~~~~~~~~~~~~ -The power-saving is a kind of auto-suspend of the device. When the -device is inactive for a certain time, the device is automatically -turned off to save the power. The time to go down is specified via -`power_save` module option, and this option can be changed dynamically -via sysfs. - -The power-saving won't work when the analog loopback is enabled on -some codecs. Make sure that you mute all unneeded signal routes when -you want the power-saving. - -The power-saving feature might cause audible click noises at each -power-down/up depending on the device. Some of them might be -solvable, but some are hard, I'm afraid. Some distros such as -openSUSE enables the power-saving feature automatically when the power -cable is unplugged. Thus, if you hear noises, suspect first the -power-saving. See /sys/module/snd_hda_intel/parameters/power_save to -check the current value. If it's non-zero, the feature is turned on. - -The recent kernel supports the runtime PM for the HD-audio controller -chip, too. It means that the HD-audio controller is also powered up / -down dynamically. The feature is enabled only for certain controller -chips like Intel LynxPoint. You can enable/disable this feature -forcibly by setting `power_save_controller` option, which is also -available at /sys/module/snd_hda_intel/parameters directory. - - -Tracepoints -~~~~~~~~~~~ -The hd-audio driver gives a few basic tracepoints. -`hda:hda_send_cmd` traces each CORB write while `hda:hda_get_response` -traces the response from RIRB (only when read from the codec driver). -`hda:hda_bus_reset` traces the bus-reset due to fatal error, etc, -`hda:hda_unsol_event` traces the unsolicited events, and -`hda:hda_power_down` and `hda:hda_power_up` trace the power down/up -via power-saving behavior. - -Enabling all tracepoints can be done like ------------------------------------------------------------------------- - # echo 1 > /sys/kernel/debug/tracing/events/hda/enable ------------------------------------------------------------------------- -then after some commands, you can traces from -/sys/kernel/debug/tracing/trace file. For example, when you want to -trace what codec command is sent, enable the tracepoint like: ------------------------------------------------------------------------- - # cat /sys/kernel/debug/tracing/trace - # tracer: nop - # - # TASK-PID CPU# TIMESTAMP FUNCTION - # | | | | | - <...>-7807 [002] 105147.774889: hda_send_cmd: [0:0] val=e3a019 - <...>-7807 [002] 105147.774893: hda_send_cmd: [0:0] val=e39019 - <...>-7807 [002] 105147.999542: hda_send_cmd: [0:0] val=e3a01a - <...>-7807 [002] 105147.999543: hda_send_cmd: [0:0] val=e3901a - <...>-26764 [001] 349222.837143: hda_send_cmd: [0:0] val=e3a019 - <...>-26764 [001] 349222.837148: hda_send_cmd: [0:0] val=e39019 - <...>-26764 [001] 349223.058539: hda_send_cmd: [0:0] val=e3a01a - <...>-26764 [001] 349223.058541: hda_send_cmd: [0:0] val=e3901a ------------------------------------------------------------------------- -Here `[0:0]` indicates the card number and the codec address, and -`val` shows the value sent to the codec, respectively. The value is -a packed value, and you can decode it via hda-decode-verb program -included in hda-emu package below. For example, the value e3a019 is -to set the left output-amp value to 25. ------------------------------------------------------------------------- - % hda-decode-verb 0xe3a019 - raw value = 0x00e3a019 - cid = 0, nid = 0x0e, verb = 0x3a0, parm = 0x19 - raw value: verb = 0x3a0, parm = 0x19 - verbname = set_amp_gain_mute - amp raw val = 0xa019 - output, left, idx=0, mute=0, val=25 ------------------------------------------------------------------------- - - -Development Tree -~~~~~~~~~~~~~~~~ -The latest development codes for HD-audio are found on sound git tree: - -- git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git - -The master branch or for-next branches can be used as the main -development branches in general while the development for the current -and next kernels are found in for-linus and for-next branches, -respectively. - - -Sending a Bug Report -~~~~~~~~~~~~~~~~~~~~ -If any model or module options don't work for your device, it's time -to send a bug report to the developers. Give the following in your -bug report: - -- Hardware vendor, product and model names -- Kernel version (and ALSA-driver version if you built externally) -- `alsa-info.sh` output; run with `--no-upload` option. See the - section below about alsa-info - -If it's a regression, at best, send alsa-info outputs of both working -and non-working kernels. This is really helpful because we can -compare the codec registers directly. - -Send a bug report either the followings: - -kernel-bugzilla:: - https://bugzilla.kernel.org/ -alsa-devel ML:: - alsa-devel@alsa-project.org - - -DEBUG TOOLS ------------ - -This section describes some tools available for debugging HD-audio -problems. - -alsa-info -~~~~~~~~~ -The script `alsa-info.sh` is a very useful tool to gather the audio -device information. It's included in alsa-utils package. The latest -version can be found on git repository: - -- git://git.alsa-project.org/alsa-utils.git - -The script can be fetched directly from the following URL, too: - -- http://www.alsa-project.org/alsa-info.sh - -Run this script as root, and it will gather the important information -such as the module lists, module parameters, proc file contents -including the codec proc files, mixer outputs and the control -elements. As default, it will store the information onto a web server -on alsa-project.org. But, if you send a bug report, it'd be better to -run with `--no-upload` option, and attach the generated file. - -There are some other useful options. See `--help` option output for -details. - -When a probe error occurs or when the driver obviously assigns a -mismatched model, it'd be helpful to load the driver with -`probe_only=1` option (at best after the cold reboot) and run -alsa-info at this state. With this option, the driver won't configure -the mixer and PCM but just tries to probe the codec slot. After -probing, the proc file is available, so you can get the raw codec -information before modified by the driver. Of course, the driver -isn't usable with `probe_only=1`. But you can continue the -configuration via hwdep sysfs file if hda-reconfig option is enabled. -Using `probe_only` mask 2 skips the reset of HDA codecs (use -`probe_only=3` as module option). The hwdep interface can be used -to determine the BIOS codec initialization. - - -hda-verb -~~~~~~~~ -hda-verb is a tiny program that allows you to access the HD-audio -codec directly. You can execute a raw HD-audio codec verb with this. -This program accesses the hwdep device, thus you need to enable the -kernel config `CONFIG_SND_HDA_HWDEP=y` beforehand. - -The hda-verb program takes four arguments: the hwdep device file, the -widget NID, the verb and the parameter. When you access to the codec -on the slot 2 of the card 0, pass /dev/snd/hwC0D2 to the first -argument, typically. (However, the real path name depends on the -system.) - -The second parameter is the widget number-id to access. The third -parameter can be either a hex/digit number or a string corresponding -to a verb. Similarly, the last parameter is the value to write, or -can be a string for the parameter type. - ------------------------------------------------------------------------- - % hda-verb /dev/snd/hwC0D0 0x12 0x701 2 - nid = 0x12, verb = 0x701, param = 0x2 - value = 0x0 - - % hda-verb /dev/snd/hwC0D0 0x0 PARAMETERS VENDOR_ID - nid = 0x0, verb = 0xf00, param = 0x0 - value = 0x10ec0262 - - % hda-verb /dev/snd/hwC0D0 2 set_a 0xb080 - nid = 0x2, verb = 0x300, param = 0xb080 - value = 0x0 ------------------------------------------------------------------------- - -Although you can issue any verbs with this program, the driver state -won't be always updated. For example, the volume values are usually -cached in the driver, and thus changing the widget amp value directly -via hda-verb won't change the mixer value. - -The hda-verb program is included now in alsa-tools: - -- git://git.alsa-project.org/alsa-tools.git - -Also, the old stand-alone package is found in the ftp directory: - -- ftp://ftp.suse.com/pub/people/tiwai/misc/ - -Also a git repository is available: - -- git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/hda-verb.git - -See README file in the tarball for more details about hda-verb -program. - - -hda-analyzer -~~~~~~~~~~~~ -hda-analyzer provides a graphical interface to access the raw HD-audio -control, based on pyGTK2 binding. It's a more powerful version of -hda-verb. The program gives you an easy-to-use GUI stuff for showing -the widget information and adjusting the amp values, as well as the -proc-compatible output. - -The hda-analyzer: - -- http://git.alsa-project.org/?p=alsa.git;a=tree;f=hda-analyzer - -is a part of alsa.git repository in alsa-project.org: - -- git://git.alsa-project.org/alsa.git - -Codecgraph -~~~~~~~~~~ -Codecgraph is a utility program to generate a graph and visualizes the -codec-node connection of a codec chip. It's especially useful when -you analyze or debug a codec without a proper datasheet. The program -parses the given codec proc file and converts to SVG via graphiz -program. - -The tarball and GIT trees are found in the web page at: - -- http://helllabs.org/codecgraph/ - - -hda-emu -~~~~~~~ -hda-emu is an HD-audio emulator. The main purpose of this program is -to debug an HD-audio codec without the real hardware. Thus, it -doesn't emulate the behavior with the real audio I/O, but it just -dumps the codec register changes and the ALSA-driver internal changes -at probing and operating the HD-audio driver. - -The program requires a codec proc-file to simulate. Get a proc file -for the target codec beforehand, or pick up an example codec from the -codec proc collections in the tarball. Then, run the program with the -proc file, and the hda-emu program will start parsing the codec file -and simulates the HD-audio driver: - ------------------------------------------------------------------------- - % hda-emu codecs/stac9200-dell-d820-laptop - # Parsing.. - hda_codec: Unknown model for STAC9200, using BIOS defaults - hda_codec: pin nid 08 bios pin config 40c003fa - .... ------------------------------------------------------------------------- - -The program gives you only a very dumb command-line interface. You -can get a proc-file dump at the current state, get a list of control -(mixer) elements, set/get the control element value, simulate the PCM -operation, the jack plugging simulation, etc. - -The program is found in the git repository below: - -- git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/hda-emu.git - -See README file in the repository for more details about hda-emu -program. - - -hda-jack-retask -~~~~~~~~~~~~~~~ -hda-jack-retask is a user-friendly GUI program to manipulate the -HD-audio pin control for jack retasking. If you have a problem about -the jack assignment, try this program and check whether you can get -useful results. Once when you figure out the proper pin assignment, -it can be fixed either in the driver code statically or via passing a -firmware patch file (see "Early Patching" section). - -The program is included in alsa-tools now: - -- git://git.alsa-project.org/alsa-tools.git - diff --git a/Documentation/sound/hd-audio/index.rst b/Documentation/sound/hd-audio/index.rst new file mode 100644 index 000000000000..f2dc29019f12 --- /dev/null +++ b/Documentation/sound/hd-audio/index.rst @@ -0,0 +1,7 @@ +HD-Audio +======== + +.. toctree:: + :maxdepth: 2 + + notes diff --git a/Documentation/sound/hd-audio/notes.rst b/Documentation/sound/hd-audio/notes.rst new file mode 100644 index 000000000000..168d0cfab1ce --- /dev/null +++ b/Documentation/sound/hd-audio/notes.rst @@ -0,0 +1,880 @@ +============================= +More Notes on HD-Audio Driver +============================= + +Takashi Iwai + + +General +======= + +HD-audio is the new standard on-board audio component on modern PCs +after AC97. Although Linux has been supporting HD-audio since long +time ago, there are often problems with new machines. A part of the +problem is broken BIOS, and the rest is the driver implementation. +This document explains the brief trouble-shooting and debugging +methods for the HD-audio hardware. + +The HD-audio component consists of two parts: the controller chip and +the codec chips on the HD-audio bus. Linux provides a single driver +for all controllers, snd-hda-intel. Although the driver name contains +a word of a well-known hardware vendor, it's not specific to it but for +all controller chips by other companies. Since the HD-audio +controllers are supposed to be compatible, the single snd-hda-driver +should work in most cases. But, not surprisingly, there are known +bugs and issues specific to each controller type. The snd-hda-intel +driver has a bunch of workarounds for these as described below. + +A controller may have multiple codecs. Usually you have one audio +codec and optionally one modem codec. In theory, there might be +multiple audio codecs, e.g. for analog and digital outputs, and the +driver might not work properly because of conflict of mixer elements. +This should be fixed in future if such hardware really exists. + +The snd-hda-intel driver has several different codec parsers depending +on the codec. It has a generic parser as a fallback, but this +functionality is fairly limited until now. Instead of the generic +parser, usually the codec-specific parser (coded in patch_*.c) is used +for the codec-specific implementations. The details about the +codec-specific problems are explained in the later sections. + +If you are interested in the deep debugging of HD-audio, read the +HD-audio specification at first. The specification is found on +Intel's web page, for example: + +* http://www.intel.com/standards/hdaudio/ + + +HD-Audio Controller +=================== + +DMA-Position Problem +-------------------- +The most common problem of the controller is the inaccurate DMA +pointer reporting. The DMA pointer for playback and capture can be +read in two ways, either via a LPIB register or via a position-buffer +map. As default the driver tries to read from the io-mapped +position-buffer, and falls back to LPIB if the position-buffer appears +dead. However, this detection isn't perfect on some devices. In such +a case, you can change the default method via ``position_fix`` option. + +``position_fix=1`` means to use LPIB method explicitly. +``position_fix=2`` means to use the position-buffer. +``position_fix=3`` means to use a combination of both methods, needed +for some VIA controllers. The capture stream position is corrected +by comparing both LPIB and position-buffer values. +``position_fix=4`` is another combination available for all controllers, +and uses LPIB for the playback and the position-buffer for the capture +streams. +0 is the default value for all other +controllers, the automatic check and fallback to LPIB as described in +the above. If you get a problem of repeated sounds, this option might +help. + +In addition to that, every controller is known to be broken regarding +the wake-up timing. It wakes up a few samples before actually +processing the data on the buffer. This caused a lot of problems, for +example, with ALSA dmix or JACK. Since 2.6.27 kernel, the driver puts +an artificial delay to the wake up timing. This delay is controlled +via ``bdl_pos_adj`` option. + +When ``bdl_pos_adj`` is a negative value (as default), it's assigned to +an appropriate value depending on the controller chip. For Intel +chips, it'd be 1 while it'd be 32 for others. Usually this works. +Only in case it doesn't work and you get warning messages, you should +change this parameter to other values. + + +Codec-Probing Problem +--------------------- +A less often but a more severe problem is the codec probing. When +BIOS reports the available codec slots wrongly, the driver gets +confused and tries to access the non-existing codec slot. This often +results in the total screw-up, and destructs the further communication +with the codec chips. The symptom appears usually as error messages +like: +:: + + hda_intel: azx_get_response timeout, switching to polling mode: + last cmd=0x12345678 + hda_intel: azx_get_response timeout, switching to single_cmd mode: + last cmd=0x12345678 + +The first line is a warning, and this is usually relatively harmless. +It means that the codec response isn't notified via an IRQ. The +driver uses explicit polling method to read the response. It gives +very slight CPU overhead, but you'd unlikely notice it. + +The second line is, however, a fatal error. If this happens, usually +it means that something is really wrong. Most likely you are +accessing a non-existing codec slot. + +Thus, if the second error message appears, try to narrow the probed +codec slots via ``probe_mask`` option. It's a bitmask, and each bit +corresponds to the codec slot. For example, to probe only the first +slot, pass ``probe_mask=1``. For the first and the third slots, pass +``probe_mask=5`` (where 5 = 1 | 4), and so on. + +Since 2.6.29 kernel, the driver has a more robust probing method, so +this error might happen rarely, though. + +On a machine with a broken BIOS, sometimes you need to force the +driver to probe the codec slots the hardware doesn't report for use. +In such a case, turn the bit 8 (0x100) of ``probe_mask`` option on. +Then the rest 8 bits are passed as the codec slots to probe +unconditionally. For example, ``probe_mask=0x103`` will force to probe +the codec slots 0 and 1 no matter what the hardware reports. + + +Interrupt Handling +------------------ +HD-audio driver uses MSI as default (if available) since 2.6.33 +kernel as MSI works better on some machines, and in general, it's +better for performance. However, Nvidia controllers showed bad +regressions with MSI (especially in a combination with AMD chipset), +thus we disabled MSI for them. + +There seem also still other devices that don't work with MSI. If you +see a regression wrt the sound quality (stuttering, etc) or a lock-up +in the recent kernel, try to pass ``enable_msi=0`` option to disable +MSI. If it works, you can add the known bad device to the blacklist +defined in hda_intel.c. In such a case, please report and give the +patch back to the upstream developer. + + +HD-Audio Codec +============== + +Model Option +------------ +The most common problem regarding the HD-audio driver is the +unsupported codec features or the mismatched device configuration. +Most of codec-specific code has several preset models, either to +override the BIOS setup or to provide more comprehensive features. + +The driver checks PCI SSID and looks through the static configuration +table until any matching entry is found. If you have a new machine, +you may see a message like below: +:: + + hda_codec: ALC880: BIOS auto-probing. + +Meanwhile, in the earlier versions, you would see a message like: +:: + + hda_codec: Unknown model for ALC880, trying auto-probe from BIOS... + +Even if you see such a message, DON'T PANIC. Take a deep breath and +keep your towel. First of all, it's an informational message, no +warning, no error. This means that the PCI SSID of your device isn't +listed in the known preset model (white-)list. But, this doesn't mean +that the driver is broken. Many codec-drivers provide the automatic +configuration mechanism based on the BIOS setup. + +The HD-audio codec has usually "pin" widgets, and BIOS sets the default +configuration of each pin, which indicates the location, the +connection type, the jack color, etc. The HD-audio driver can guess +the right connection judging from these default configuration values. +However -- some codec-support codes, such as patch_analog.c, don't +support the automatic probing (yet as of 2.6.28). And, BIOS is often, +yes, pretty often broken. It sets up wrong values and screws up the +driver. + +The preset model (or recently called as "fix-up") is provided +basically to overcome such a situation. When the matching preset +model is found in the white-list, the driver assumes the static +configuration of that preset with the correct pin setup, etc. +Thus, if you have a newer machine with a slightly different PCI SSID +(or codec SSID) from the existing one, you may have a good chance to +re-use the same model. You can pass the ``model`` option to specify the +preset model instead of PCI (and codec-) SSID look-up. + +What ``model`` option values are available depends on the codec chip. +Check your codec chip from the codec proc file (see "Codec Proc-File" +section below). It will show the vendor/product name of your codec +chip. Then, see Documentation/sound/HD-Audio-Models.rst file, +the section of HD-audio driver. You can find a list of codecs +and ``model`` options belonging to each codec. For example, for Realtek +ALC262 codec chip, pass ``model=ultra`` for devices that are compatible +with Samsung Q1 Ultra. + +Thus, the first thing you can do for any brand-new, unsupported and +non-working HD-audio hardware is to check HD-audio codec and several +different ``model`` option values. If you have any luck, some of them +might suit with your device well. + +There are a few special model option values: + +* when 'nofixup' is passed, the device-specific fixups in the codec + parser are skipped. +* when ``generic`` is passed, the codec-specific parser is skipped and + only the generic parser is used. + + +Speaker and Headphone Output +---------------------------- +One of the most frequent (and obvious) bugs with HD-audio is the +silent output from either or both of a built-in speaker and a +headphone jack. In general, you should try a headphone output at +first. A speaker output often requires more additional controls like +the external amplifier bits. Thus a headphone output has a slightly +better chance. + +Before making a bug report, double-check whether the mixer is set up +correctly. The recent version of snd-hda-intel driver provides mostly +"Master" volume control as well as "Front" volume (where Front +indicates the front-channels). In addition, there can be individual +"Headphone" and "Speaker" controls. + +Ditto for the speaker output. There can be "External Amplifier" +switch on some codecs. Turn on this if present. + +Another related problem is the automatic mute of speaker output by +headphone plugging. This feature is implemented in most cases, but +not on every preset model or codec-support code. + +In anyway, try a different model option if you have such a problem. +Some other models may match better and give you more matching +functionality. If none of the available models works, send a bug +report. See the bug report section for details. + +If you are masochistic enough to debug the driver problem, note the +following: + +* The speaker (and the headphone, too) output often requires the + external amplifier. This can be set usually via EAPD verb or a + certain GPIO. If the codec pin supports EAPD, you have a better + chance via SET_EAPD_BTL verb (0x70c). On others, GPIO pin (mostly + it's either GPIO0 or GPIO1) may turn on/off EAPD. +* Some Realtek codecs require special vendor-specific coefficients to + turn on the amplifier. See patch_realtek.c. +* IDT codecs may have extra power-enable/disable controls on each + analog pin. See patch_sigmatel.c. +* Very rare but some devices don't accept the pin-detection verb until + triggered. Issuing GET_PIN_SENSE verb (0xf09) may result in the + codec-communication stall. Some examples are found in + patch_realtek.c. + + +Capture Problems +---------------- +The capture problems are often because of missing setups of mixers. +Thus, before submitting a bug report, make sure that you set up the +mixer correctly. For example, both "Capture Volume" and "Capture +Switch" have to be set properly in addition to the right "Capture +Source" or "Input Source" selection. Some devices have "Mic Boost" +volume or switch. + +When the PCM device is opened via "default" PCM (without pulse-audio +plugin), you'll likely have "Digital Capture Volume" control as well. +This is provided for the extra gain/attenuation of the signal in +software, especially for the inputs without the hardware volume +control such as digital microphones. Unless really needed, this +should be set to exactly 50%, corresponding to 0dB -- neither extra +gain nor attenuation. When you use "hw" PCM, i.e., a raw access PCM, +this control will have no influence, though. + +It's known that some codecs / devices have fairly bad analog circuits, +and the recorded sound contains a certain DC-offset. This is no bug +of the driver. + +Most of modern laptops have no analog CD-input connection. Thus, the +recording from CD input won't work in many cases although the driver +provides it as the capture source. Use CDDA instead. + +The automatic switching of the built-in and external mic per plugging +is implemented on some codec models but not on every model. Partly +because of my laziness but mostly lack of testers. Feel free to +submit the improvement patch to the author. + + +Direct Debugging +---------------- +If no model option gives you a better result, and you are a tough guy +to fight against evil, try debugging via hitting the raw HD-audio +codec verbs to the device. Some tools are available: hda-emu and +hda-analyzer. The detailed description is found in the sections +below. You'd need to enable hwdep for using these tools. See "Kernel +Configuration" section. + + +Other Issues +============ + +Kernel Configuration +-------------------- +In general, I recommend you to enable the sound debug option, +``CONFIG_SND_DEBUG=y``, no matter whether you are debugging or not. +This enables snd_printd() macro and others, and you'll get additional +kernel messages at probing. + +In addition, you can enable ``CONFIG_SND_DEBUG_VERBOSE=y``. But this +will give you far more messages. Thus turn this on only when you are +sure to want it. + +Don't forget to turn on the appropriate ``CONFIG_SND_HDA_CODEC_*`` +options. Note that each of them corresponds to the codec chip, not +the controller chip. Thus, even if lspci shows the Nvidia controller, +you may need to choose the option for other vendors. If you are +unsure, just select all yes. + +``CONFIG_SND_HDA_HWDEP`` is a useful option for debugging the driver. +When this is enabled, the driver creates hardware-dependent devices +(one per each codec), and you have a raw access to the device via +these device files. For example, ``hwC0D2`` will be created for the +codec slot #2 of the first card (#0). For debug-tools such as +hda-verb and hda-analyzer, the hwdep device has to be enabled. +Thus, it'd be better to turn this on always. + +``CONFIG_SND_HDA_RECONFIG`` is a new option, and this depends on the +hwdep option above. When enabled, you'll have some sysfs files under +the corresponding hwdep directory. See "HD-audio reconfiguration" +section below. + +``CONFIG_SND_HDA_POWER_SAVE`` option enables the power-saving feature. +See "Power-saving" section below. + + +Codec Proc-File +--------------- +The codec proc-file is a treasure-chest for debugging HD-audio. +It shows most of useful information of each codec widget. + +The proc file is located in /proc/asound/card*/codec#*, one file per +each codec slot. You can know the codec vendor, product id and +names, the type of each widget, capabilities and so on. +This file, however, doesn't show the jack sensing state, so far. This +is because the jack-sensing might be depending on the trigger state. + +This file will be picked up by the debug tools, and also it can be fed +to the emulator as the primary codec information. See the debug tools +section below. + +This proc file can be also used to check whether the generic parser is +used. When the generic parser is used, the vendor/product ID name +will appear as "Realtek ID 0262", instead of "Realtek ALC262". + + +HD-Audio Reconfiguration +------------------------ +This is an experimental feature to allow you re-configure the HD-audio +codec dynamically without reloading the driver. The following sysfs +files are available under each codec-hwdep device directory (e.g. +/sys/class/sound/hwC0D0): + +vendor_id + Shows the 32bit codec vendor-id hex number. You can change the + vendor-id value by writing to this file. +subsystem_id + Shows the 32bit codec subsystem-id hex number. You can change the + subsystem-id value by writing to this file. +revision_id + Shows the 32bit codec revision-id hex number. You can change the + revision-id value by writing to this file. +afg + Shows the AFG ID. This is read-only. +mfg + Shows the MFG ID. This is read-only. +name + Shows the codec name string. Can be changed by writing to this + file. +modelname + Shows the currently set ``model`` option. Can be changed by writing + to this file. +init_verbs + The extra verbs to execute at initialization. You can add a verb by + writing to this file. Pass three numbers: nid, verb and parameter + (separated with a space). +hints + Shows / stores hint strings for codec parsers for any use. + Its format is ``key = value``. For example, passing ``jack_detect = no`` + will disable the jack detection of the machine completely. +init_pin_configs + Shows the initial pin default config values set by BIOS. +driver_pin_configs + Shows the pin default values set by the codec parser explicitly. + This doesn't show all pin values but only the changed values by + the parser. That is, if the parser doesn't change the pin default + config values by itself, this will contain nothing. +user_pin_configs + Shows the pin default config values to override the BIOS setup. + Writing this (with two numbers, NID and value) appends the new + value. The given will be used instead of the initial BIOS value at + the next reconfiguration time. Note that this config will override + even the driver pin configs, too. +reconfig + Triggers the codec re-configuration. When any value is written to + this file, the driver re-initialize and parses the codec tree + again. All the changes done by the sysfs entries above are taken + into account. +clear + Resets the codec, removes the mixer elements and PCM stuff of the + specified codec, and clear all init verbs and hints. + +For example, when you want to change the pin default configuration +value of the pin widget 0x14 to 0x9993013f, and let the driver +re-configure based on that state, run like below: +:: + + # echo 0x14 0x9993013f > /sys/class/sound/hwC0D0/user_pin_configs + # echo 1 > /sys/class/sound/hwC0D0/reconfig + + +Hint Strings +------------ +The codec parser have several switches and adjustment knobs for +matching better with the actual codec or device behavior. Many of +them can be adjusted dynamically via "hints" strings as mentioned in +the section above. For example, by passing ``jack_detect = no`` string +via sysfs or a patch file, you can disable the jack detection, thus +the codec parser will skip the features like auto-mute or mic +auto-switch. As a boolean value, either ``yes``, ``no``, ``true``, ``false``, +``1`` or ``0`` can be passed. + +The generic parser supports the following hints: + +jack_detect (bool) + specify whether the jack detection is available at all on this + machine; default true +inv_jack_detect (bool) + indicates that the jack detection logic is inverted +trigger_sense (bool) + indicates that the jack detection needs the explicit call of + AC_VERB_SET_PIN_SENSE verb +inv_eapd (bool) + indicates that the EAPD is implemented in the inverted logic +pcm_format_first (bool) + sets the PCM format before the stream tag and channel ID +sticky_stream (bool) + keep the PCM format, stream tag and ID as long as possible; + default true +spdif_status_reset (bool) + reset the SPDIF status bits at each time the SPDIF stream is set + up +pin_amp_workaround (bool) + the output pin may have multiple amp values +single_adc_amp (bool) + ADCs can have only single input amps +auto_mute (bool) + enable/disable the headphone auto-mute feature; default true +auto_mic (bool) + enable/disable the mic auto-switch feature; default true +line_in_auto_switch (bool) + enable/disable the line-in auto-switch feature; default false +need_dac_fix (bool) + limits the DACs depending on the channel count +primary_hp (bool) + probe headphone jacks as the primary outputs; default true +multi_io (bool) + try probing multi-I/O config (e.g. shared line-in/surround, + mic/clfe jacks) +multi_cap_vol (bool) + provide multiple capture volumes +inv_dmic_split (bool) + provide split internal mic volume/switch for phase-inverted + digital mics +indep_hp (bool) + provide the independent headphone PCM stream and the corresponding + mixer control, if available +add_stereo_mix_input (bool) + add the stereo mix (analog-loopback mix) to the input mux if + available +add_jack_modes (bool) + add "xxx Jack Mode" enum controls to each I/O jack for allowing to + change the headphone amp and mic bias VREF capabilities +power_save_node (bool) + advanced power management for each widget, controlling the power + sate (D0/D3) of each widget node depending on the actual pin and + stream states +power_down_unused (bool) + power down the unused widgets, a subset of power_save_node, and + will be dropped in future +add_hp_mic (bool) + add the headphone to capture source if possible +hp_mic_detect (bool) + enable/disable the hp/mic shared input for a single built-in mic + case; default true +mixer_nid (int) + specifies the widget NID of the analog-loopback mixer + + +Early Patching +-------------- +When ``CONFIG_SND_HDA_PATCH_LOADER=y`` is set, you can pass a "patch" +as a firmware file for modifying the HD-audio setup before +initializing the codec. This can work basically like the +reconfiguration via sysfs in the above, but it does it before the +first codec configuration. + +A patch file is a plain text file which looks like below: + +:: + + [codec] + 0x12345678 0xabcd1234 2 + + [model] + auto + + [pincfg] + 0x12 0x411111f0 + + [verb] + 0x20 0x500 0x03 + 0x20 0x400 0xff + + [hint] + jack_detect = no + + +The file needs to have a line ``[codec]``. The next line should contain +three numbers indicating the codec vendor-id (0x12345678 in the +example), the codec subsystem-id (0xabcd1234) and the address (2) of +the codec. The rest patch entries are applied to this specified codec +until another codec entry is given. Passing 0 or a negative number to +the first or the second value will make the check of the corresponding +field be skipped. It'll be useful for really broken devices that don't +initialize SSID properly. + +The ``[model]`` line allows to change the model name of the each codec. +In the example above, it will be changed to model=auto. +Note that this overrides the module option. + +After the ``[pincfg]`` line, the contents are parsed as the initial +default pin-configurations just like ``user_pin_configs`` sysfs above. +The values can be shown in user_pin_configs sysfs file, too. + +Similarly, the lines after ``[verb]`` are parsed as ``init_verbs`` +sysfs entries, and the lines after ``[hint]`` are parsed as ``hints`` +sysfs entries, respectively. + +Another example to override the codec vendor id from 0x12345678 to +0xdeadbeef is like below: +:: + + [codec] + 0x12345678 0xabcd1234 2 + + [vendor_id] + 0xdeadbeef + + +In the similar way, you can override the codec subsystem_id via +``[subsystem_id]``, the revision id via ``[revision_id]`` line. +Also, the codec chip name can be rewritten via ``[chip_name]`` line. +:: + + [codec] + 0x12345678 0xabcd1234 2 + + [subsystem_id] + 0xffff1111 + + [revision_id] + 0x10 + + [chip_name] + My-own NEWS-0002 + + +The hd-audio driver reads the file via request_firmware(). Thus, +a patch file has to be located on the appropriate firmware path, +typically, /lib/firmware. For example, when you pass the option +``patch=hda-init.fw``, the file /lib/firmware/hda-init.fw must be +present. + +The patch module option is specific to each card instance, and you +need to give one file name for each instance, separated by commas. +For example, if you have two cards, one for an on-board analog and one +for an HDMI video board, you may pass patch option like below: +:: + + options snd-hda-intel patch=on-board-patch,hdmi-patch + + +Power-Saving +------------ +The power-saving is a kind of auto-suspend of the device. When the +device is inactive for a certain time, the device is automatically +turned off to save the power. The time to go down is specified via +``power_save`` module option, and this option can be changed dynamically +via sysfs. + +The power-saving won't work when the analog loopback is enabled on +some codecs. Make sure that you mute all unneeded signal routes when +you want the power-saving. + +The power-saving feature might cause audible click noises at each +power-down/up depending on the device. Some of them might be +solvable, but some are hard, I'm afraid. Some distros such as +openSUSE enables the power-saving feature automatically when the power +cable is unplugged. Thus, if you hear noises, suspect first the +power-saving. See /sys/module/snd_hda_intel/parameters/power_save to +check the current value. If it's non-zero, the feature is turned on. + +The recent kernel supports the runtime PM for the HD-audio controller +chip, too. It means that the HD-audio controller is also powered up / +down dynamically. The feature is enabled only for certain controller +chips like Intel LynxPoint. You can enable/disable this feature +forcibly by setting ``power_save_controller`` option, which is also +available at /sys/module/snd_hda_intel/parameters directory. + + +Tracepoints +----------- +The hd-audio driver gives a few basic tracepoints. +``hda:hda_send_cmd`` traces each CORB write while ``hda:hda_get_response`` +traces the response from RIRB (only when read from the codec driver). +``hda:hda_bus_reset`` traces the bus-reset due to fatal error, etc, +``hda:hda_unsol_event`` traces the unsolicited events, and +``hda:hda_power_down`` and ``hda:hda_power_up`` trace the power down/up +via power-saving behavior. + +Enabling all tracepoints can be done like +:: + + # echo 1 > /sys/kernel/debug/tracing/events/hda/enable + +then after some commands, you can traces from +/sys/kernel/debug/tracing/trace file. For example, when you want to +trace what codec command is sent, enable the tracepoint like: +:: + + # cat /sys/kernel/debug/tracing/trace + # tracer: nop + # + # TASK-PID CPU# TIMESTAMP FUNCTION + # | | | | | + <...>-7807 [002] 105147.774889: hda_send_cmd: [0:0] val=e3a019 + <...>-7807 [002] 105147.774893: hda_send_cmd: [0:0] val=e39019 + <...>-7807 [002] 105147.999542: hda_send_cmd: [0:0] val=e3a01a + <...>-7807 [002] 105147.999543: hda_send_cmd: [0:0] val=e3901a + <...>-26764 [001] 349222.837143: hda_send_cmd: [0:0] val=e3a019 + <...>-26764 [001] 349222.837148: hda_send_cmd: [0:0] val=e39019 + <...>-26764 [001] 349223.058539: hda_send_cmd: [0:0] val=e3a01a + <...>-26764 [001] 349223.058541: hda_send_cmd: [0:0] val=e3901a + +Here ``[0:0]`` indicates the card number and the codec address, and +``val`` shows the value sent to the codec, respectively. The value is +a packed value, and you can decode it via hda-decode-verb program +included in hda-emu package below. For example, the value e3a019 is +to set the left output-amp value to 25. +:: + + % hda-decode-verb 0xe3a019 + raw value = 0x00e3a019 + cid = 0, nid = 0x0e, verb = 0x3a0, parm = 0x19 + raw value: verb = 0x3a0, parm = 0x19 + verbname = set_amp_gain_mute + amp raw val = 0xa019 + output, left, idx=0, mute=0, val=25 + + +Development Tree +---------------- +The latest development codes for HD-audio are found on sound git tree: + +* git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git + +The master branch or for-next branches can be used as the main +development branches in general while the development for the current +and next kernels are found in for-linus and for-next branches, +respectively. + + +Sending a Bug Report +-------------------- +If any model or module options don't work for your device, it's time +to send a bug report to the developers. Give the following in your +bug report: + +* Hardware vendor, product and model names +* Kernel version (and ALSA-driver version if you built externally) +* ``alsa-info.sh`` output; run with ``--no-upload`` option. See the + section below about alsa-info + +If it's a regression, at best, send alsa-info outputs of both working +and non-working kernels. This is really helpful because we can +compare the codec registers directly. + +Send a bug report either the followings: + +kernel-bugzilla + https://bugzilla.kernel.org/ +alsa-devel ML + alsa-devel@alsa-project.org + + +Debug Tools +=========== + +This section describes some tools available for debugging HD-audio +problems. + +alsa-info +--------- +The script ``alsa-info.sh`` is a very useful tool to gather the audio +device information. It's included in alsa-utils package. The latest +version can be found on git repository: + +* git://git.alsa-project.org/alsa-utils.git + +The script can be fetched directly from the following URL, too: + +* http://www.alsa-project.org/alsa-info.sh + +Run this script as root, and it will gather the important information +such as the module lists, module parameters, proc file contents +including the codec proc files, mixer outputs and the control +elements. As default, it will store the information onto a web server +on alsa-project.org. But, if you send a bug report, it'd be better to +run with ``--no-upload`` option, and attach the generated file. + +There are some other useful options. See ``--help`` option output for +details. + +When a probe error occurs or when the driver obviously assigns a +mismatched model, it'd be helpful to load the driver with +``probe_only=1`` option (at best after the cold reboot) and run +alsa-info at this state. With this option, the driver won't configure +the mixer and PCM but just tries to probe the codec slot. After +probing, the proc file is available, so you can get the raw codec +information before modified by the driver. Of course, the driver +isn't usable with ``probe_only=1``. But you can continue the +configuration via hwdep sysfs file if hda-reconfig option is enabled. +Using ``probe_only`` mask 2 skips the reset of HDA codecs (use +``probe_only=3`` as module option). The hwdep interface can be used +to determine the BIOS codec initialization. + + +hda-verb +-------- +hda-verb is a tiny program that allows you to access the HD-audio +codec directly. You can execute a raw HD-audio codec verb with this. +This program accesses the hwdep device, thus you need to enable the +kernel config ``CONFIG_SND_HDA_HWDEP=y`` beforehand. + +The hda-verb program takes four arguments: the hwdep device file, the +widget NID, the verb and the parameter. When you access to the codec +on the slot 2 of the card 0, pass /dev/snd/hwC0D2 to the first +argument, typically. (However, the real path name depends on the +system.) + +The second parameter is the widget number-id to access. The third +parameter can be either a hex/digit number or a string corresponding +to a verb. Similarly, the last parameter is the value to write, or +can be a string for the parameter type. + +:: + + % hda-verb /dev/snd/hwC0D0 0x12 0x701 2 + nid = 0x12, verb = 0x701, param = 0x2 + value = 0x0 + + % hda-verb /dev/snd/hwC0D0 0x0 PARAMETERS VENDOR_ID + nid = 0x0, verb = 0xf00, param = 0x0 + value = 0x10ec0262 + + % hda-verb /dev/snd/hwC0D0 2 set_a 0xb080 + nid = 0x2, verb = 0x300, param = 0xb080 + value = 0x0 + + +Although you can issue any verbs with this program, the driver state +won't be always updated. For example, the volume values are usually +cached in the driver, and thus changing the widget amp value directly +via hda-verb won't change the mixer value. + +The hda-verb program is included now in alsa-tools: + +* git://git.alsa-project.org/alsa-tools.git + +Also, the old stand-alone package is found in the ftp directory: + +* ftp://ftp.suse.com/pub/people/tiwai/misc/ + +Also a git repository is available: + +* git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/hda-verb.git + +See README file in the tarball for more details about hda-verb +program. + + +hda-analyzer +------------ +hda-analyzer provides a graphical interface to access the raw HD-audio +control, based on pyGTK2 binding. It's a more powerful version of +hda-verb. The program gives you an easy-to-use GUI stuff for showing +the widget information and adjusting the amp values, as well as the +proc-compatible output. + +The hda-analyzer: + +* http://git.alsa-project.org/?p=alsa.git;a=tree;f=hda-analyzer + +is a part of alsa.git repository in alsa-project.org: + +* git://git.alsa-project.org/alsa.git + +Codecgraph +---------- +Codecgraph is a utility program to generate a graph and visualizes the +codec-node connection of a codec chip. It's especially useful when +you analyze or debug a codec without a proper datasheet. The program +parses the given codec proc file and converts to SVG via graphiz +program. + +The tarball and GIT trees are found in the web page at: + +* http://helllabs.org/codecgraph/ + + +hda-emu +------- +hda-emu is an HD-audio emulator. The main purpose of this program is +to debug an HD-audio codec without the real hardware. Thus, it +doesn't emulate the behavior with the real audio I/O, but it just +dumps the codec register changes and the ALSA-driver internal changes +at probing and operating the HD-audio driver. + +The program requires a codec proc-file to simulate. Get a proc file +for the target codec beforehand, or pick up an example codec from the +codec proc collections in the tarball. Then, run the program with the +proc file, and the hda-emu program will start parsing the codec file +and simulates the HD-audio driver: + +:: + + % hda-emu codecs/stac9200-dell-d820-laptop + # Parsing.. + hda_codec: Unknown model for STAC9200, using BIOS defaults + hda_codec: pin nid 08 bios pin config 40c003fa + .... + + +The program gives you only a very dumb command-line interface. You +can get a proc-file dump at the current state, get a list of control +(mixer) elements, set/get the control element value, simulate the PCM +operation, the jack plugging simulation, etc. + +The program is found in the git repository below: + +* git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/hda-emu.git + +See README file in the repository for more details about hda-emu +program. + + +hda-jack-retask +--------------- +hda-jack-retask is a user-friendly GUI program to manipulate the +HD-audio pin control for jack retasking. If you have a problem about +the jack assignment, try this program and check whether you can get +useful results. Once when you figure out the proper pin assignment, +it can be fixed either in the driver code statically or via passing a +firmware patch file (see "Early Patching" section). + +The program is included in alsa-tools now: + +* git://git.alsa-project.org/alsa-tools.git diff --git a/Documentation/sound/index.rst b/Documentation/sound/index.rst index 280a57115f00..e9bac7c8b893 100644 --- a/Documentation/sound/index.rst +++ b/Documentation/sound/index.rst @@ -6,6 +6,7 @@ Linux Sound Subsystem Documentation :maxdepth: 2 kernel-api/index + hd-audio/index .. only:: subproject -- cgit v1.2.3-58-ga151 From a4caad753f0ccc201482bee4bdaa1875524bf2ab Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 9 Nov 2016 14:43:16 +0100 Subject: ALSA: doc: ReSTize HD-Audio-Models document A simple reformat with the description list of ReST, and the content was kept as is, but renamed as Documentation/sound/hd-audio/models.rst. Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/HD-Audio-Models.txt | 324 ----------------- Documentation/sound/hd-audio/index.rst | 1 + Documentation/sound/hd-audio/models.rst | 518 +++++++++++++++++++++++++++ 3 files changed, 519 insertions(+), 324 deletions(-) delete mode 100644 Documentation/sound/alsa/HD-Audio-Models.txt create mode 100644 Documentation/sound/hd-audio/models.rst (limited to 'Documentation/sound') diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt deleted file mode 100644 index ec099d4343f2..000000000000 --- a/Documentation/sound/alsa/HD-Audio-Models.txt +++ /dev/null @@ -1,324 +0,0 @@ - Model name Description - ---------- ----------- -ALC880 -====== - 3stack 3-jack in back and a headphone out - 3stack-digout 3-jack in back, a HP out and a SPDIF out - 5stack 5-jack in back, 2-jack in front - 5stack-digout 5-jack in back, 2-jack in front, a SPDIF out - 6stack 6-jack in back, 2-jack in front - 6stack-digout 6-jack with a SPDIF out - -ALC260 -====== - gpio1 Enable GPIO1 - coef Enable EAPD via COEF table - fujitsu Quirk for FSC S7020 - fujitsu-jwse Quirk for FSC S7020 with jack modes and HP mic support - -ALC262 -====== - inv-dmic Inverted internal mic workaround - -ALC267/268 -========== - inv-dmic Inverted internal mic workaround - hp-eapd Disable HP EAPD on NID 0x15 - -ALC22x/23x/25x/269/27x/28x/29x (and vendor-specific ALC3xxx models) -====== - laptop-amic Laptops with analog-mic input - laptop-dmic Laptops with digital-mic input - alc269-dmic Enable ALC269(VA) digital mic workaround - alc271-dmic Enable ALC271X digital mic workaround - inv-dmic Inverted internal mic workaround - headset-mic Indicates a combined headset (headphone+mic) jack - headset-mode More comprehensive headset support for ALC269 & co - headset-mode-no-hp-mic Headset mode support without headphone mic - lenovo-dock Enables docking station I/O for some Lenovos - hp-gpio-led GPIO LED support on HP laptops - dell-headset-multi Headset jack, which can also be used as mic-in - dell-headset-dock Headset jack (without mic-in), and also dock I/O - alc283-dac-wcaps Fixups for Chromebook with ALC283 - alc283-sense-combo Combo jack sensing on ALC283 - tpt440-dock Pin configs for Lenovo Thinkpad Dock support - -ALC66x/67x/892 -============== - mario Chromebook mario model fixup - asus-mode1 ASUS - asus-mode2 ASUS - asus-mode3 ASUS - asus-mode4 ASUS - asus-mode5 ASUS - asus-mode6 ASUS - asus-mode7 ASUS - asus-mode8 ASUS - inv-dmic Inverted internal mic workaround - dell-headset-multi Headset jack, which can also be used as mic-in - -ALC680 -====== - N/A - -ALC88x/898/1150 -====================== - acer-aspire-4930g Acer Aspire 4930G/5930G/6530G/6930G/7730G - acer-aspire-8930g Acer Aspire 8330G/6935G - acer-aspire Acer Aspire others - inv-dmic Inverted internal mic workaround - no-primary-hp VAIO Z/VGC-LN51JGB workaround (for fixed speaker DAC) - -ALC861/660 -========== - N/A - -ALC861VD/660VD -============== - N/A - -CMI9880 -======= - minimal 3-jack in back - min_fp 3-jack in back, 2-jack in front - full 6-jack in back, 2-jack in front - full_dig 6-jack in back, 2-jack in front, SPDIF I/O - allout 5-jack in back, 2-jack in front, SPDIF out - auto auto-config reading BIOS (default) - -AD1882 / AD1882A -================ - 3stack 3-stack mode - 3stack-automute 3-stack with automute front HP (default) - 6stack 6-stack mode - -AD1884A / AD1883 / AD1984A / AD1984B -==================================== - desktop 3-stack desktop (default) - laptop laptop with HP jack sensing - mobile mobile devices with HP jack sensing - thinkpad Lenovo Thinkpad X300 - touchsmart HP Touchsmart - -AD1884 -====== - N/A - -AD1981 -====== - basic 3-jack (default) - hp HP nx6320 - thinkpad Lenovo Thinkpad T60/X60/Z60 - toshiba Toshiba U205 - -AD1983 -====== - N/A - -AD1984 -====== - basic default configuration - thinkpad Lenovo Thinkpad T61/X61 - dell_desktop Dell T3400 - -AD1986A -======= - 3stack 3-stack, shared surrounds - laptop 2-channel only (FSC V2060, Samsung M50) - laptop-imic 2-channel with built-in mic - eapd Turn on EAPD constantly - -AD1988/AD1988B/AD1989A/AD1989B -============================== - 6stack 6-jack - 6stack-dig ditto with SPDIF - 3stack 3-jack - 3stack-dig ditto with SPDIF - laptop 3-jack with hp-jack automute - laptop-dig ditto with SPDIF - auto auto-config reading BIOS (default) - -Conexant 5045 -============= - laptop-hpsense Laptop with HP sense (old model laptop) - laptop-micsense Laptop with Mic sense (old model fujitsu) - laptop-hpmicsense Laptop with HP and Mic senses - benq Benq R55E - laptop-hp530 HP 530 laptop - test for testing/debugging purpose, almost all controls - can be adjusted. Appearing only when compiled with - $CONFIG_SND_DEBUG=y - -Conexant 5047 -============= - laptop Basic Laptop config - laptop-hp Laptop config for some HP models (subdevice 30A5) - laptop-eapd Laptop config with EAPD support - test for testing/debugging purpose, almost all controls - can be adjusted. Appearing only when compiled with - $CONFIG_SND_DEBUG=y - -Conexant 5051 -============= - laptop Basic Laptop config (default) - hp HP Spartan laptop - hp-dv6736 HP dv6736 - hp-f700 HP Compaq Presario F700 - ideapad Lenovo IdeaPad laptop - toshiba Toshiba Satellite M300 - -Conexant 5066 -============= - laptop Basic Laptop config (default) - hp-laptop HP laptops, e g G60 - asus Asus K52JU, Lenovo G560 - dell-laptop Dell laptops - dell-vostro Dell Vostro - olpc-xo-1_5 OLPC XO 1.5 - ideapad Lenovo IdeaPad U150 - thinkpad Lenovo Thinkpad - -STAC9200 -======== - ref Reference board - oqo OQO Model 2 - dell-d21 Dell (unknown) - dell-d22 Dell (unknown) - dell-d23 Dell (unknown) - dell-m21 Dell Inspiron 630m, Dell Inspiron 640m - dell-m22 Dell Latitude D620, Dell Latitude D820 - dell-m23 Dell XPS M1710, Dell Precision M90 - dell-m24 Dell Latitude 120L - dell-m25 Dell Inspiron E1505n - dell-m26 Dell Inspiron 1501 - dell-m27 Dell Inspiron E1705/9400 - gateway-m4 Gateway laptops with EAPD control - gateway-m4-2 Gateway laptops with EAPD control - panasonic Panasonic CF-74 - auto BIOS setup (default) - -STAC9205/9254 -============= - ref Reference board - dell-m42 Dell (unknown) - dell-m43 Dell Precision - dell-m44 Dell Inspiron - eapd Keep EAPD on (e.g. Gateway T1616) - auto BIOS setup (default) - -STAC9220/9221 -============= - ref Reference board - 3stack D945 3stack - 5stack D945 5stack + SPDIF - intel-mac-v1 Intel Mac Type 1 - intel-mac-v2 Intel Mac Type 2 - intel-mac-v3 Intel Mac Type 3 - intel-mac-v4 Intel Mac Type 4 - intel-mac-v5 Intel Mac Type 5 - intel-mac-auto Intel Mac (detect type according to subsystem id) - macmini Intel Mac Mini (equivalent with type 3) - macbook Intel Mac Book (eq. type 5) - macbook-pro-v1 Intel Mac Book Pro 1st generation (eq. type 3) - macbook-pro Intel Mac Book Pro 2nd generation (eq. type 3) - imac-intel Intel iMac (eq. type 2) - imac-intel-20 Intel iMac (newer version) (eq. type 3) - ecs202 ECS/PC chips - dell-d81 Dell (unknown) - dell-d82 Dell (unknown) - dell-m81 Dell (unknown) - dell-m82 Dell XPS M1210 - auto BIOS setup (default) - -STAC9202/9250/9251 -================== - ref Reference board, base config - m1 Some Gateway MX series laptops (NX560XL) - m1-2 Some Gateway MX series laptops (MX6453) - m2 Some Gateway MX series laptops (M255) - m2-2 Some Gateway MX series laptops - m3 Some Gateway MX series laptops - m5 Some Gateway MX series laptops (MP6954) - m6 Some Gateway NX series laptops - auto BIOS setup (default) - -STAC9227/9228/9229/927x -======================= - ref Reference board - ref-no-jd Reference board without HP/Mic jack detection - 3stack D965 3stack - 5stack D965 5stack + SPDIF - 5stack-no-fp D965 5stack without front panel - dell-3stack Dell Dimension E520 - dell-bios Fixes with Dell BIOS setup - dell-bios-amic Fixes with Dell BIOS setup including analog mic - volknob Fixes with volume-knob widget 0x24 - auto BIOS setup (default) - -STAC92HD71B* -============ - ref Reference board - dell-m4-1 Dell desktops - dell-m4-2 Dell desktops - dell-m4-3 Dell desktops - hp-m4 HP mini 1000 - hp-dv5 HP dv series - hp-hdx HP HDX series - hp-dv4-1222nr HP dv4-1222nr (with LED support) - auto BIOS setup (default) - -STAC92HD73* -=========== - ref Reference board - no-jd BIOS setup but without jack-detection - intel Intel DG45* mobos - dell-m6-amic Dell desktops/laptops with analog mics - dell-m6-dmic Dell desktops/laptops with digital mics - dell-m6 Dell desktops/laptops with both type of mics - dell-eq Dell desktops/laptops - alienware Alienware M17x - auto BIOS setup (default) - -STAC92HD83* -=========== - ref Reference board - mic-ref Reference board with power management for ports - dell-s14 Dell laptop - dell-vostro-3500 Dell Vostro 3500 laptop - hp-dv7-4000 HP dv-7 4000 - hp_cNB11_intquad HP CNB models with 4 speakers - hp-zephyr HP Zephyr - hp-led HP with broken BIOS for mute LED - hp-inv-led HP with broken BIOS for inverted mute LED - hp-mic-led HP with mic-mute LED - headset-jack Dell Latitude with a 4-pin headset jack - hp-envy-bass Pin fixup for HP Envy bass speaker (NID 0x0f) - hp-envy-ts-bass Pin fixup for HP Envy TS bass speaker (NID 0x10) - hp-bnb13-eq Hardware equalizer setup for HP laptops - auto BIOS setup (default) - -STAC92HD95 -========== - hp-led LED support for HP laptops - hp-bass Bass HPF setup for HP Spectre 13 - -STAC9872 -======== - vaio VAIO laptop without SPDIF - auto BIOS setup (default) - -Cirrus Logic CS4206/4207 -======================== - mbp55 MacBook Pro 5,5 - imac27 IMac 27 Inch - auto BIOS setup (default) - -Cirrus Logic CS4208 -=================== - mba6 MacBook Air 6,1 and 6,2 - gpio0 Enable GPIO 0 amp - auto BIOS setup (default) - -VIA VT17xx/VT18xx/VT20xx -======================== - auto BIOS setup (default) diff --git a/Documentation/sound/hd-audio/index.rst b/Documentation/sound/hd-audio/index.rst index f2dc29019f12..1e8376b0066c 100644 --- a/Documentation/sound/hd-audio/index.rst +++ b/Documentation/sound/hd-audio/index.rst @@ -5,3 +5,4 @@ HD-Audio :maxdepth: 2 notes + models diff --git a/Documentation/sound/hd-audio/models.rst b/Documentation/sound/hd-audio/models.rst new file mode 100644 index 000000000000..5338673c88d9 --- /dev/null +++ b/Documentation/sound/hd-audio/models.rst @@ -0,0 +1,518 @@ +============================== +HD-Audio Codec-Specific Models +============================== + +ALC880 +====== +3stack + 3-jack in back and a headphone out +3stack-digout + 3-jack in back, a HP out and a SPDIF out +5stack + 5-jack in back, 2-jack in front +5stack-digout + 5-jack in back, 2-jack in front, a SPDIF out +6stack + 6-jack in back, 2-jack in front +6stack-digout + 6-jack with a SPDIF out + +ALC260 +====== +gpio1 + Enable GPIO1 +coef + Enable EAPD via COEF table +fujitsu + Quirk for FSC S7020 +fujitsu-jwse + Quirk for FSC S7020 with jack modes and HP mic support + +ALC262 +====== +inv-dmic + Inverted internal mic workaround + +ALC267/268 +========== +inv-dmic + Inverted internal mic workaround +hp-eapd + Disable HP EAPD on NID 0x15 + +ALC22x/23x/25x/269/27x/28x/29x (and vendor-specific ALC3xxx models) +=================================================================== +laptop-amic + Laptops with analog-mic input +laptop-dmic + Laptops with digital-mic input +alc269-dmic + Enable ALC269(VA) digital mic workaround +alc271-dmic + Enable ALC271X digital mic workaround +inv-dmic + Inverted internal mic workaround +headset-mic + Indicates a combined headset (headphone+mic) jack +headset-mode + More comprehensive headset support for ALC269 & co +headset-mode-no-hp-mic + Headset mode support without headphone mic +lenovo-dock + Enables docking station I/O for some Lenovos +hp-gpio-led + GPIO LED support on HP laptops +dell-headset-multi + Headset jack, which can also be used as mic-in +dell-headset-dock + Headset jack (without mic-in), and also dock I/O +alc283-dac-wcaps + Fixups for Chromebook with ALC283 +alc283-sense-combo + Combo jack sensing on ALC283 +tpt440-dock + Pin configs for Lenovo Thinkpad Dock support + +ALC66x/67x/892 +============== +mario + Chromebook mario model fixup +asus-mode1 + ASUS +asus-mode2 + ASUS +asus-mode3 + ASUS +asus-mode4 + ASUS +asus-mode5 + ASUS +asus-mode6 + ASUS +asus-mode7 + ASUS +asus-mode8 + ASUS +inv-dmic + Inverted internal mic workaround +dell-headset-multi + Headset jack, which can also be used as mic-in + +ALC680 +====== +N/A + +ALC88x/898/1150 +====================== +acer-aspire-4930g + Acer Aspire 4930G/5930G/6530G/6930G/7730G +acer-aspire-8930g + Acer Aspire 8330G/6935G +acer-aspire + Acer Aspire others +inv-dmic + Inverted internal mic workaround +no-primary-hp + VAIO Z/VGC-LN51JGB workaround (for fixed speaker DAC) + +ALC861/660 +========== +N/A + +ALC861VD/660VD +============== +N/A + +CMI9880 +======= +minimal + 3-jack in back +min_fp + 3-jack in back, 2-jack in front +full + 6-jack in back, 2-jack in front +full_dig + 6-jack in back, 2-jack in front, SPDIF I/O +allout + 5-jack in back, 2-jack in front, SPDIF out +auto + auto-config reading BIOS (default) + +AD1882 / AD1882A +================ +3stack + 3-stack mode +3stack-automute + 3-stack with automute front HP (default) +6stack + 6-stack mode + +AD1884A / AD1883 / AD1984A / AD1984B +==================================== +desktop 3-stack desktop (default) +laptop laptop with HP jack sensing +mobile mobile devices with HP jack sensing +thinkpad Lenovo Thinkpad X300 +touchsmart HP Touchsmart + +AD1884 +====== +N/A + +AD1981 +====== +basic 3-jack (default) +hp HP nx6320 +thinkpad Lenovo Thinkpad T60/X60/Z60 +toshiba Toshiba U205 + +AD1983 +====== +N/A + +AD1984 +====== +basic default configuration +thinkpad Lenovo Thinkpad T61/X61 +dell_desktop Dell T3400 + +AD1986A +======= +3stack + 3-stack, shared surrounds +laptop + 2-channel only (FSC V2060, Samsung M50) +laptop-imic + 2-channel with built-in mic +eapd + Turn on EAPD constantly + +AD1988/AD1988B/AD1989A/AD1989B +============================== +6stack + 6-jack +6stack-dig + ditto with SPDIF +3stack + 3-jack +3stack-dig + ditto with SPDIF +laptop + 3-jack with hp-jack automute +laptop-dig + ditto with SPDIF +auto + auto-config reading BIOS (default) + +Conexant 5045 +============= +laptop-hpsense + Laptop with HP sense (old model laptop) +laptop-micsense + Laptop with Mic sense (old model fujitsu) +laptop-hpmicsense + Laptop with HP and Mic senses +benq + Benq R55E +laptop-hp530 + HP 530 laptop +test + for testing/debugging purpose, almost all controls can be + adjusted. Appearing only when compiled with $CONFIG_SND_DEBUG=y + +Conexant 5047 +============= +laptop + Basic Laptop config +laptop-hp + Laptop config for some HP models (subdevice 30A5) +laptop-eapd + Laptop config with EAPD support +test + for testing/debugging purpose, almost all controls can be + adjusted. Appearing only when compiled with $CONFIG_SND_DEBUG=y + +Conexant 5051 +============= +laptop + Basic Laptop config (default) +hp + HP Spartan laptop +hp-dv6736 + HP dv6736 +hp-f700 + HP Compaq Presario F700 +ideapad + Lenovo IdeaPad laptop +toshiba + Toshiba Satellite M300 + +Conexant 5066 +============= +laptop + Basic Laptop config (default) +hp-laptop + HP laptops, e g G60 +asus + Asus K52JU, Lenovo G560 +dell-laptop + Dell laptops +dell-vostro + Dell Vostro +olpc-xo-1_5 + OLPC XO 1.5 +ideapad + Lenovo IdeaPad U150 +thinkpad + Lenovo Thinkpad + +STAC9200 +======== +ref + Reference board +oqo + OQO Model 2 +dell-d21 + Dell (unknown) +dell-d22 + Dell (unknown) +dell-d23 + Dell (unknown) +dell-m21 + Dell Inspiron 630m, Dell Inspiron 640m +dell-m22 + Dell Latitude D620, Dell Latitude D820 +dell-m23 + Dell XPS M1710, Dell Precision M90 +dell-m24 + Dell Latitude 120L +dell-m25 + Dell Inspiron E1505n +dell-m26 + Dell Inspiron 1501 +dell-m27 + Dell Inspiron E1705/9400 +gateway-m4 + Gateway laptops with EAPD control +gateway-m4-2 + Gateway laptops with EAPD control +panasonic + Panasonic CF-74 +auto + BIOS setup (default) + +STAC9205/9254 +============= +ref + Reference board +dell-m42 + Dell (unknown) +dell-m43 + Dell Precision +dell-m44 + Dell Inspiron +eapd + Keep EAPD on (e.g. Gateway T1616) +auto + BIOS setup (default) + +STAC9220/9221 +============= +ref + Reference board +3stack + D945 3stack +5stack + D945 5stack + SPDIF +intel-mac-v1 + Intel Mac Type 1 +intel-mac-v2 + Intel Mac Type 2 +intel-mac-v3 + Intel Mac Type 3 +intel-mac-v4 + Intel Mac Type 4 +intel-mac-v5 + Intel Mac Type 5 +intel-mac-auto + Intel Mac (detect type according to subsystem id) +macmini + Intel Mac Mini (equivalent with type 3) +macbook + Intel Mac Book (eq. type 5) +macbook-pro-v1 + Intel Mac Book Pro 1st generation (eq. type 3) +macbook-pro + Intel Mac Book Pro 2nd generation (eq. type 3) +imac-intel + Intel iMac (eq. type 2) +imac-intel-20 + Intel iMac (newer version) (eq. type 3) +ecs202 + ECS/PC chips +dell-d81 + Dell (unknown) +dell-d82 + Dell (unknown) +dell-m81 + Dell (unknown) +dell-m82 + Dell XPS M1210 +auto + BIOS setup (default) + +STAC9202/9250/9251 +================== +ref + Reference board, base config +m1 + Some Gateway MX series laptops (NX560XL) +m1-2 + Some Gateway MX series laptops (MX6453) +m2 + Some Gateway MX series laptops (M255) +m2-2 + Some Gateway MX series laptops +m3 + Some Gateway MX series laptops +m5 + Some Gateway MX series laptops (MP6954) +m6 + Some Gateway NX series laptops +auto + BIOS setup (default) + +STAC9227/9228/9229/927x +======================= +ref + Reference board +ref-no-jd + Reference board without HP/Mic jack detection +3stack + D965 3stack +5stack + D965 5stack + SPDIF +5stack-no-fp + D965 5stack without front panel +dell-3stack + Dell Dimension E520 +dell-bios + Fixes with Dell BIOS setup +dell-bios-amic + Fixes with Dell BIOS setup including analog mic +volknob + Fixes with volume-knob widget 0x24 +auto + BIOS setup (default) + +STAC92HD71B* +============ +ref + Reference board +dell-m4-1 + Dell desktops +dell-m4-2 + Dell desktops +dell-m4-3 + Dell desktops +hp-m4 + HP mini 1000 +hp-dv5 + HP dv series +hp-hdx + HP HDX series +hp-dv4-1222nr + HP dv4-1222nr (with LED support) +auto + BIOS setup (default) + +STAC92HD73* +=========== +ref + Reference board +no-jd + BIOS setup but without jack-detection +intel + Intel DG45* mobos +dell-m6-amic + Dell desktops/laptops with analog mics +dell-m6-dmic + Dell desktops/laptops with digital mics +dell-m6 + Dell desktops/laptops with both type of mics +dell-eq + Dell desktops/laptops +alienware + Alienware M17x +auto + BIOS setup (default) + +STAC92HD83* +=========== +ref + Reference board +mic-ref + Reference board with power management for ports +dell-s14 + Dell laptop +dell-vostro-3500 + Dell Vostro 3500 laptop +hp-dv7-4000 + HP dv-7 4000 +hp_cNB11_intquad + HP CNB models with 4 speakers +hp-zephyr + HP Zephyr +hp-led + HP with broken BIOS for mute LED +hp-inv-led + HP with broken BIOS for inverted mute LED +hp-mic-led + HP with mic-mute LED +headset-jack + Dell Latitude with a 4-pin headset jack +hp-envy-bass + Pin fixup for HP Envy bass speaker (NID 0x0f) +hp-envy-ts-bass + Pin fixup for HP Envy TS bass speaker (NID 0x10) +hp-bnb13-eq + Hardware equalizer setup for HP laptops +auto + BIOS setup (default) + +STAC92HD95 +========== +hp-led + LED support for HP laptops +hp-bass + Bass HPF setup for HP Spectre 13 + +STAC9872 +======== +vaio + VAIO laptop without SPDIF +auto + BIOS setup (default) + +Cirrus Logic CS4206/4207 +======================== +mbp55 + MacBook Pro 5,5 +imac27 + IMac 27 Inch +auto + BIOS setup (default) + +Cirrus Logic CS4208 +=================== +mba6 + MacBook Air 6,1 and 6,2 +gpio0 + Enable GPIO 0 amp +auto + BIOS setup (default) + +VIA VT17xx/VT18xx/VT20xx +======================== +auto + BIOS setup (default) -- cgit v1.2.3-58-ga151 From fe0abd18e1ef3cb258b0a5e41ba26ed0c4b88dab Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 9 Nov 2016 16:56:01 +0100 Subject: ALSA: doc: ReSTize HD-Audio-Controls document A conversion from a simple text file. Put to hd-audio subdirectory with a rename. Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/HD-Audio-Controls.txt | 116 ------------------------ Documentation/sound/hd-audio/controls.rst | 121 +++++++++++++++++++++++++ Documentation/sound/hd-audio/index.rst | 1 + 3 files changed, 122 insertions(+), 116 deletions(-) delete mode 100644 Documentation/sound/alsa/HD-Audio-Controls.txt create mode 100644 Documentation/sound/hd-audio/controls.rst (limited to 'Documentation/sound') diff --git a/Documentation/sound/alsa/HD-Audio-Controls.txt b/Documentation/sound/alsa/HD-Audio-Controls.txt deleted file mode 100644 index e9621e349e17..000000000000 --- a/Documentation/sound/alsa/HD-Audio-Controls.txt +++ /dev/null @@ -1,116 +0,0 @@ -This file explains the codec-specific mixer controls. - -Realtek codecs --------------- - -* Channel Mode - This is an enum control to change the surround-channel setup, - appears only when the surround channels are available. - It gives the number of channels to be used, "2ch", "4ch", "6ch", - and "8ch". According to the configuration, this also controls the - jack-retasking of multi-I/O jacks. - -* Auto-Mute Mode - This is an enum control to change the auto-mute behavior of the - headphone and line-out jacks. If built-in speakers and headphone - and/or line-out jacks are available on a machine, this controls - appears. - When there are only either headphones or line-out jacks, it gives - "Disabled" and "Enabled" state. When enabled, the speaker is muted - automatically when a jack is plugged. - - When both headphone and line-out jacks are present, it gives - "Disabled", "Speaker Only" and "Line-Out+Speaker". When - speaker-only is chosen, plugging into a headphone or a line-out jack - mutes the speakers, but not line-outs. When line-out+speaker is - selected, plugging to a headphone jack mutes both speakers and - line-outs. - - -IDT/Sigmatel codecs -------------------- - -* Analog Loopback - This control enables/disables the analog-loopback circuit. This - appears only when "loopback" is set to true in a codec hint - (see HD-Audio.txt). Note that on some codecs the analog-loopback - and the normal PCM playback are exclusive, i.e. when this is on, you - won't hear any PCM stream. - -* Swap Center/LFE - Swaps the center and LFE channel order. Normally, the left - corresponds to the center and the right to the LFE. When this is - ON, the left to the LFE and the right to the center. - -* Headphone as Line Out - When this control is ON, treat the headphone jacks as line-out - jacks. That is, the headphone won't auto-mute the other line-outs, - and no HP-amp is set to the pins. - -* Mic Jack Mode, Line Jack Mode, etc - These enum controls the direction and the bias of the input jack - pins. Depending on the jack type, it can set as "Mic In" and "Line - In", for determining the input bias, or it can be set to "Line Out" - when the pin is a multi-I/O jack for surround channels. - - -VIA codecs ----------- - -* Smart 5.1 - An enum control to re-task the multi-I/O jacks for surround outputs. - When it's ON, the corresponding input jacks (usually a line-in and a - mic-in) are switched as the surround and the CLFE output jacks. - -* Independent HP - When this enum control is enabled, the headphone output is routed - from an individual stream (the third PCM such as hw:0,2) instead of - the primary stream. In the case the headphone DAC is shared with a - side or a CLFE-channel DAC, the DAC is switched to the headphone - automatically. - -* Loopback Mixing - An enum control to determine whether the analog-loopback route is - enabled or not. When it's enabled, the analog-loopback is mixed to - the front-channel. Also, the same route is used for the headphone - and speaker outputs. As a side-effect, when this mode is set, the - individual volume controls will be no longer available for - headphones and speakers because there is only one DAC connected to a - mixer widget. - -* Dynamic Power-Control - This control determines whether the dynamic power-control per jack - detection is enabled or not. When enabled, the widgets power state - (D0/D3) are changed dynamically depending on the jack plugging - state for saving power consumptions. However, if your system - doesn't provide a proper jack-detection, this won't work; in such a - case, turn this control OFF. - -* Jack Detect - This control is provided only for VT1708 codec which gives no proper - unsolicited event per jack plug. When this is on, the driver polls - the jack detection so that the headphone auto-mute can work, while - turning this off would reduce the power consumption. - - -Conexant codecs ---------------- - -* Auto-Mute Mode - See Reatek codecs. - - -Analog codecs --------------- - -* Channel Mode - This is an enum control to change the surround-channel setup, - appears only when the surround channels are available. - It gives the number of channels to be used, "2ch", "4ch" and "6ch". - According to the configuration, this also controls the - jack-retasking of multi-I/O jacks. - -* Independent HP - When this enum control is enabled, the headphone output is routed - from an individual stream (the third PCM such as hw:0,2) instead of - the primary stream. diff --git a/Documentation/sound/hd-audio/controls.rst b/Documentation/sound/hd-audio/controls.rst new file mode 100644 index 000000000000..f2ebc4f79b44 --- /dev/null +++ b/Documentation/sound/hd-audio/controls.rst @@ -0,0 +1,121 @@ +====================================== +HD-Audio Codec-Specific Mixer Controls +====================================== + + +This file explains the codec-specific mixer controls. + +Realtek codecs +-------------- + +Channel Mode + This is an enum control to change the surround-channel setup, + appears only when the surround channels are available. + It gives the number of channels to be used, "2ch", "4ch", "6ch", + and "8ch". According to the configuration, this also controls the + jack-retasking of multi-I/O jacks. + +Auto-Mute Mode + This is an enum control to change the auto-mute behavior of the + headphone and line-out jacks. If built-in speakers and headphone + and/or line-out jacks are available on a machine, this controls + appears. + When there are only either headphones or line-out jacks, it gives + "Disabled" and "Enabled" state. When enabled, the speaker is muted + automatically when a jack is plugged. + + When both headphone and line-out jacks are present, it gives + "Disabled", "Speaker Only" and "Line-Out+Speaker". When + speaker-only is chosen, plugging into a headphone or a line-out jack + mutes the speakers, but not line-outs. When line-out+speaker is + selected, plugging to a headphone jack mutes both speakers and + line-outs. + + +IDT/Sigmatel codecs +------------------- + +Analog Loopback + This control enables/disables the analog-loopback circuit. This + appears only when "loopback" is set to true in a codec hint + (see HD-Audio.txt). Note that on some codecs the analog-loopback + and the normal PCM playback are exclusive, i.e. when this is on, you + won't hear any PCM stream. + +Swap Center/LFE + Swaps the center and LFE channel order. Normally, the left + corresponds to the center and the right to the LFE. When this is + ON, the left to the LFE and the right to the center. + +Headphone as Line Out + When this control is ON, treat the headphone jacks as line-out + jacks. That is, the headphone won't auto-mute the other line-outs, + and no HP-amp is set to the pins. + +Mic Jack Mode, Line Jack Mode, etc + These enum controls the direction and the bias of the input jack + pins. Depending on the jack type, it can set as "Mic In" and "Line + In", for determining the input bias, or it can be set to "Line Out" + when the pin is a multi-I/O jack for surround channels. + + +VIA codecs +---------- + +Smart 5.1 + An enum control to re-task the multi-I/O jacks for surround outputs. + When it's ON, the corresponding input jacks (usually a line-in and a + mic-in) are switched as the surround and the CLFE output jacks. + +Independent HP + When this enum control is enabled, the headphone output is routed + from an individual stream (the third PCM such as hw:0,2) instead of + the primary stream. In the case the headphone DAC is shared with a + side or a CLFE-channel DAC, the DAC is switched to the headphone + automatically. + +Loopback Mixing + An enum control to determine whether the analog-loopback route is + enabled or not. When it's enabled, the analog-loopback is mixed to + the front-channel. Also, the same route is used for the headphone + and speaker outputs. As a side-effect, when this mode is set, the + individual volume controls will be no longer available for + headphones and speakers because there is only one DAC connected to a + mixer widget. + +Dynamic Power-Control + This control determines whether the dynamic power-control per jack + detection is enabled or not. When enabled, the widgets power state + (D0/D3) are changed dynamically depending on the jack plugging + state for saving power consumptions. However, if your system + doesn't provide a proper jack-detection, this won't work; in such a + case, turn this control OFF. + +Jack Detect + This control is provided only for VT1708 codec which gives no proper + unsolicited event per jack plug. When this is on, the driver polls + the jack detection so that the headphone auto-mute can work, while + turning this off would reduce the power consumption. + + +Conexant codecs +--------------- + +Auto-Mute Mode + See Reatek codecs. + + +Analog codecs +-------------- + +Channel Mode + This is an enum control to change the surround-channel setup, + appears only when the surround channels are available. + It gives the number of channels to be used, "2ch", "4ch" and "6ch". + According to the configuration, this also controls the + jack-retasking of multi-I/O jacks. + +Independent HP + When this enum control is enabled, the headphone output is routed + from an individual stream (the third PCM such as hw:0,2) instead of + the primary stream. diff --git a/Documentation/sound/hd-audio/index.rst b/Documentation/sound/hd-audio/index.rst index 1e8376b0066c..c6efd55a219f 100644 --- a/Documentation/sound/hd-audio/index.rst +++ b/Documentation/sound/hd-audio/index.rst @@ -6,3 +6,4 @@ HD-Audio notes models + controls -- cgit v1.2.3-58-ga151 From 76ab4e15158c677141e8b8ff5f0295166f474553 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 10 Nov 2016 17:41:45 +0100 Subject: ALSA: doc: ReSTize HD-Audio-DP-MST-audio.txt A simple conversion from a plain text file. Put to hd-audio subdirectory. Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/HD-Audio-DP-MST-audio.txt | 74 ------------------- Documentation/sound/hd-audio/dp-mst.rst | 84 ++++++++++++++++++++++ Documentation/sound/hd-audio/index.rst | 1 + 3 files changed, 85 insertions(+), 74 deletions(-) delete mode 100644 Documentation/sound/alsa/HD-Audio-DP-MST-audio.txt create mode 100644 Documentation/sound/hd-audio/dp-mst.rst (limited to 'Documentation/sound') diff --git a/Documentation/sound/alsa/HD-Audio-DP-MST-audio.txt b/Documentation/sound/alsa/HD-Audio-DP-MST-audio.txt deleted file mode 100644 index 82744ac3513d..000000000000 --- a/Documentation/sound/alsa/HD-Audio-DP-MST-audio.txt +++ /dev/null @@ -1,74 +0,0 @@ -To support DP MST audio, HD Audio hdmi codec driver introduces virtual pin -and dynamic pcm assignment. - -Virtual pin is an extension of per_pin. The most difference of DP MST -from legacy is that DP MST introduces device entry. Each pin can contain -several device entries. Each device entry behaves as a pin. - -As each pin may contain several device entries and each codec may contain -several pins, if we use one pcm per per_pin, there will be many PCMs. -The new solution is to create a few PCMs and to dynamically bind pcm to -per_pin. Driver uses spec->dyn_pcm_assign flag to indicate whether to use -the new solution. - -PCM -=== -To be added - - -Jack -==== - -Presume: - - MST must be dyn_pcm_assign, and it is acomp (for Intel scenario); - - NON-MST may or may not be dyn_pcm_assign, it can be acomp or !acomp; - -So there are the following scenarios: - a. MST (&& dyn_pcm_assign && acomp) - b. NON-MST && dyn_pcm_assign && acomp - c. NON-MST && !dyn_pcm_assign && !acomp - -Below discussion will ignore MST and NON-MST difference as it doesn't -impact on jack handling too much. - -Driver uses struct hdmi_pcm pcm[] array in hdmi_spec and snd_jack is -a member of hdmi_pcm. Each pin has one struct hdmi_pcm * pcm pointer. - -For !dyn_pcm_assign, per_pin->pcm will assigned to spec->pcm[n] statically. - -For dyn_pcm_assign, per_pin->pcm will assigned to spec->pcm[n] -when monitor is hotplugged. - - -Build Jack ----------- - -- dyn_pcm_assign -Will not use hda_jack but use snd_jack in spec->pcm_rec[pcm_idx].jack directly. - -- !dyn_pcm_assign -Use hda_jack and assign spec->pcm_rec[pcm_idx].jack = jack->jack statically. - - -Unsolicited Event Enabling --------------------------- -Enable unsolicited event if !acomp. - - -Monitor Hotplug Event Handling ------------------------------- -- acomp -pin_eld_notify() -> check_presence_and_report() -> hdmi_present_sense() -> -sync_eld_via_acomp(). -Use directly snd_jack_report() on spec->pcm_rec[pcm_idx].jack for -both dyn_pcm_assign and !dyn_pcm_assign - -- !acomp -Hdmi_unsol_event() -> hdmi_intrinsic_event() -> check_presence_and_report() -> -hdmi_present_sense() -> hdmi_prepsent_sense_via_verbs() -Use directly snd_jack_report() on spec->pcm_rec[pcm_idx].jack for dyn_pcm_assign. -Use hda_jack mechanism to handle jack events. - - -Others to be added later -======================== diff --git a/Documentation/sound/hd-audio/dp-mst.rst b/Documentation/sound/hd-audio/dp-mst.rst new file mode 100644 index 000000000000..58b72437e6c3 --- /dev/null +++ b/Documentation/sound/hd-audio/dp-mst.rst @@ -0,0 +1,84 @@ +======================= +HD-Audio DP-MST Support +======================= + +To support DP MST audio, HD Audio hdmi codec driver introduces virtual pin +and dynamic pcm assignment. + +Virtual pin is an extension of per_pin. The most difference of DP MST +from legacy is that DP MST introduces device entry. Each pin can contain +several device entries. Each device entry behaves as a pin. + +As each pin may contain several device entries and each codec may contain +several pins, if we use one pcm per per_pin, there will be many PCMs. +The new solution is to create a few PCMs and to dynamically bind pcm to +per_pin. Driver uses spec->dyn_pcm_assign flag to indicate whether to use +the new solution. + +PCM +=== +To be added + + +Jack +==== + +Presume: + - MST must be dyn_pcm_assign, and it is acomp (for Intel scenario); + - NON-MST may or may not be dyn_pcm_assign, it can be acomp or !acomp; + +So there are the following scenarios: + a. MST (&& dyn_pcm_assign && acomp) + b. NON-MST && dyn_pcm_assign && acomp + c. NON-MST && !dyn_pcm_assign && !acomp + +Below discussion will ignore MST and NON-MST difference as it doesn't +impact on jack handling too much. + +Driver uses struct hdmi_pcm pcm[] array in hdmi_spec and snd_jack is +a member of hdmi_pcm. Each pin has one struct hdmi_pcm * pcm pointer. + +For !dyn_pcm_assign, per_pin->pcm will assigned to spec->pcm[n] statically. + +For dyn_pcm_assign, per_pin->pcm will assigned to spec->pcm[n] +when monitor is hotplugged. + + +Build Jack +---------- + +- dyn_pcm_assign + + Will not use hda_jack but use snd_jack in spec->pcm_rec[pcm_idx].jack directly. + +- !dyn_pcm_assign + + Use hda_jack and assign spec->pcm_rec[pcm_idx].jack = jack->jack statically. + + +Unsolicited Event Enabling +-------------------------- +Enable unsolicited event if !acomp. + + +Monitor Hotplug Event Handling +------------------------------ +- acomp + + pin_eld_notify() -> check_presence_and_report() -> hdmi_present_sense() -> + sync_eld_via_acomp(). + + Use directly snd_jack_report() on spec->pcm_rec[pcm_idx].jack for + both dyn_pcm_assign and !dyn_pcm_assign + +- !acomp + + hdmi_unsol_event() -> hdmi_intrinsic_event() -> check_presence_and_report() -> + hdmi_present_sense() -> hdmi_prepsent_sense_via_verbs() + + Use directly snd_jack_report() on spec->pcm_rec[pcm_idx].jack for dyn_pcm_assign. + Use hda_jack mechanism to handle jack events. + + +Others to be added later +======================== diff --git a/Documentation/sound/hd-audio/index.rst b/Documentation/sound/hd-audio/index.rst index c6efd55a219f..f8a72ffffe66 100644 --- a/Documentation/sound/hd-audio/index.rst +++ b/Documentation/sound/hd-audio/index.rst @@ -7,3 +7,4 @@ HD-Audio notes models controls + dp-mst -- cgit v1.2.3-58-ga151 From f6d23df5cac135e7375f14557b2da959405480b6 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 9 Nov 2016 15:40:00 +0100 Subject: ALSA: doc: ReSTize ALSA-Configuration document A simple conversion from the text file. Since this is the only document specific to the configurations, it's put to the root sound subdirectory. A section describing the obsoleted configure stuff of old alsa-driver tarball got removed. Signed-off-by: Takashi Iwai --- Documentation/sound/alsa-configuration.rst | 2683 +++++++++++++++++++++++ Documentation/sound/alsa/ALSA-Configuration.txt | 2330 -------------------- Documentation/sound/index.rst | 1 + 3 files changed, 2684 insertions(+), 2330 deletions(-) create mode 100644 Documentation/sound/alsa-configuration.rst delete mode 100644 Documentation/sound/alsa/ALSA-Configuration.txt (limited to 'Documentation/sound') diff --git a/Documentation/sound/alsa-configuration.rst b/Documentation/sound/alsa-configuration.rst new file mode 100644 index 000000000000..aed6b4fb8e46 --- /dev/null +++ b/Documentation/sound/alsa-configuration.rst @@ -0,0 +1,2683 @@ +============================================================== +Advanced Linux Sound Architecture - Driver Configuration guide +============================================================== + + +Kernel Configuration +==================== + +To enable ALSA support you need at least to build the kernel with +primary sound card support (``CONFIG_SOUND``). Since ALSA can emulate +OSS, you don't have to choose any of the OSS modules. + +Enable "OSS API emulation" (``CONFIG_SND_OSSEMUL``) and both OSS mixer +and PCM supports if you want to run OSS applications with ALSA. + +If you want to support the WaveTable functionality on cards such as +SB Live! then you need to enable "Sequencer support" +(``CONFIG_SND_SEQUENCER``). + +To make ALSA debug messages more verbose, enable the "Verbose printk" +and "Debug" options. To check for memory leaks, turn on "Debug memory" +too. "Debug detection" will add checks for the detection of cards. + +Please note that all the ALSA ISA drivers support the Linux isapnp API +(if the card supports ISA PnP). You don't need to configure the cards +using isapnptools. + + +Module parameters +================= + +The user can load modules with options. If the module supports more than +one card and you have more than one card of the same type then you can +specify multiple values for the option separated by commas. + + +Module snd +---------- + +The core ALSA module. It is used by all ALSA card drivers. +It takes the following options which have global effects. + +major + major number for sound driver; + Default: 116 +cards_limit + limiting card index for auto-loading (1-8); + Default: 1; + For auto-loading more than one card, specify this option + together with snd-card-X aliases. +slots + Reserve the slot index for the given driver; + This option takes multiple strings. + See `Module Autoloading Support`_ section for details. +debug + Specifies the debug message level; + (0 = disable debug prints, 1 = normal debug messages, + 2 = verbose debug messages); + This option appears only when ``CONFIG_SND_DEBUG=y``. + This option can be dynamically changed via sysfs + /sys/modules/snd/parameters/debug file. + +Module snd-pcm-oss +------------------ + +The PCM OSS emulation module. +This module takes options which change the mapping of devices. + +dsp_map + PCM device number maps assigned to the 1st OSS device; + Default: 0 +adsp_map + PCM device number maps assigned to the 2st OSS device; + Default: 1 +nonblock_open + Don't block opening busy PCM devices; + Default: 1 + +For example, when ``dsp_map=2``, /dev/dsp will be mapped to PCM #2 of +the card #0. Similarly, when ``adsp_map=0``, /dev/adsp will be mapped +to PCM #0 of the card #0. +For changing the second or later card, specify the option with +commas, such like ``dsp_map=0,1``. + +``nonblock_open`` option is used to change the behavior of the PCM +regarding opening the device. When this option is non-zero, +opening a busy OSS PCM device won't be blocked but return +immediately with EAGAIN (just like O_NONBLOCK flag). + +Module snd-rawmidi +------------------ + +This module takes options which change the mapping of devices. +similar to those of the snd-pcm-oss module. + +midi_map + MIDI device number maps assigned to the 1st OSS device; + Default: 0 +amidi_map + MIDI device number maps assigned to the 2st OSS device; + Default: 1 + +Common parameters for top sound card modules +-------------------------------------------- + +Each of top level sound card module takes the following options. + +index + index (slot #) of sound card; + Values: 0 through 31 or negative; + If nonnegative, assign that index number; + if negative, interpret as a bitmask of permissible indices; + the first free permitted index is assigned; + Default: -1 +id + card ID (identifier or name); + Can be up to 15 characters long; + Default: the card type; + A directory by this name is created under /proc/asound/ + containing information about the card; + This ID can be used instead of the index number in + identifying the card +enable + enable card; + Default: enabled, for PCI and ISA PnP cards + +Module snd-adlib +---------------- + +Module for AdLib FM cards. + +port + port # for OPL chip + +This module supports multiple cards. It does not support autoprobe, so +the port must be specified. For actual AdLib FM cards it will be 0x388. +Note that this card does not have PCM support and no mixer; only FM +synthesis. + +Make sure you have ``sbiload`` from the alsa-tools package available and, +after loading the module, find out the assigned ALSA sequencer port +number through ``sbiload -l``. + +Example output: +:: + + Port Client name Port name + 64:0 OPL2 FM synth OPL2 FM Port + +Load the ``std.sb`` and ``drums.sb`` patches also supplied by ``sbiload``: +:: + + sbiload -p 64:0 std.sb drums.sb + +If you use this driver to drive an OPL3, you can use ``std.o3`` and ``drums.o3`` +instead. To have the card produce sound, use ``aplaymidi`` from alsa-utils: +:: + + aplaymidi -p 64:0 foo.mid + +Module snd-ad1816a +------------------ + +Module for sound cards based on Analog Devices AD1816A/AD1815 ISA chips. + +clockfreq + Clock frequency for AD1816A chip (default = 0, 33000Hz) + +This module supports multiple cards, autoprobe and PnP. + +Module snd-ad1848 +----------------- + +Module for sound cards based on AD1848/AD1847/CS4248 ISA chips. + +port + port # for AD1848 chip +irq + IRQ # for AD1848 chip +dma1 + DMA # for AD1848 chip (0,1,3) + +This module supports multiple cards. It does not support autoprobe +thus main port must be specified!!! Other ports are optional. + +The power-management is supported. + +Module snd-ad1889 +----------------- + +Module for Analog Devices AD1889 chips. + +ac97_quirk + AC'97 workaround for strange hardware; + See the description of intel8x0 module for details. + +This module supports multiple cards. + +Module snd-ali5451 +------------------ + +Module for ALi M5451 PCI chip. + +pcm_channels + Number of hardware channels assigned for PCM +spdif + Support SPDIF I/O; + Default: disabled + +This module supports one chip and autoprobe. + +The power-management is supported. + +Module snd-als100 +----------------- + +Module for sound cards based on Avance Logic ALS100/ALS120 ISA chips. + +This module supports multiple cards, autoprobe and PnP. + +The power-management is supported. + +Module snd-als300 +----------------- + +Module for Avance Logic ALS300 and ALS300+ + +This module supports multiple cards. + +The power-management is supported. + +Module snd-als4000 +------------------ + +Module for sound cards based on Avance Logic ALS4000 PCI chip. + +joystick_port + port # for legacy joystick support; + 0 = disabled (default), 1 = auto-detect + +This module supports multiple cards, autoprobe and PnP. + +The power-management is supported. + +Module snd-asihpi +----------------- + +Module for AudioScience ASI soundcards + +enable_hpi_hwdep + enable HPI hwdep for AudioScience soundcard + +This module supports multiple cards. +The driver requires the firmware loader support on kernel. + +Module snd-atiixp +----------------- + +Module for ATI IXP 150/200/250/400 AC97 controllers. + +ac97_clock + AC'97 clock (default = 48000) +ac97_quirk + AC'97 workaround for strange hardware; + See `AC97 Quirk Option`_ section below. +ac97_codec + Workaround to specify which AC'97 codec instead of probing. + If this works for you file a bug with your `lspci -vn` output. + (-2 = Force probing, -1 = Default behavior, 0-2 = Use the + specified codec.) +spdif_aclink + S/PDIF transfer over AC-link (default = 1) + +This module supports one card and autoprobe. + +ATI IXP has two different methods to control SPDIF output. One is +over AC-link and another is over the "direct" SPDIF output. The +implementation depends on the motherboard, and you'll need to +choose the correct one via spdif_aclink module option. + +The power-management is supported. + +Module snd-atiixp-modem +----------------------- + +Module for ATI IXP 150/200/250 AC97 modem controllers. + +This module supports one card and autoprobe. + +Note: The default index value of this module is -2, i.e. the first +slot is excluded. + +The power-management is supported. + +Module snd-au8810, snd-au8820, snd-au8830 +----------------------------------------- + +Module for Aureal Vortex, Vortex2 and Advantage device. + +pcifix + Control PCI workarounds; + 0 = Disable all workarounds, + 1 = Force the PCI latency of the Aureal card to 0xff, + 2 = Force the Extend PCI#2 Internal Master for Efficient + Handling of Dummy Requests on the VIA KT133 AGP Bridge, + 3 = Force both settings, + 255 = Autodetect what is required (default) + +This module supports all ADB PCM channels, ac97 mixer, SPDIF, hardware +EQ, mpu401, gameport. A3D and wavetable support are still in development. +Development and reverse engineering work is being coordinated at +http://savannah.nongnu.org/projects/openvortex/ +SPDIF output has a copy of the AC97 codec output, unless you use the +``spdif`` pcm device, which allows raw data passthru. +The hardware EQ hardware and SPDIF is only present in the Vortex2 and +Advantage. + +Note: Some ALSA mixer applications don't handle the SPDIF sample rate +control correctly. If you have problems regarding this, try +another ALSA compliant mixer (alsamixer works). + +Module snd-azt1605 +------------------ + +Module for Aztech Sound Galaxy soundcards based on the Aztech AZT1605 +chipset. + +port + port # for BASE (0x220,0x240,0x260,0x280) +wss_port + port # for WSS (0x530,0x604,0xe80,0xf40) +irq + IRQ # for WSS (7,9,10,11) +dma1 + DMA # for WSS playback (0,1,3) +dma2 + DMA # for WSS capture (0,1), -1 = disabled (default) +mpu_port + port # for MPU-401 UART (0x300,0x330), -1 = disabled (default) +mpu_irq + IRQ # for MPU-401 UART (3,5,7,9), -1 = disabled (default) +fm_port + port # for OPL3 (0x388), -1 = disabled (default) + +This module supports multiple cards. It does not support autoprobe: +``port``, ``wss_port``, ``irq`` and ``dma1`` have to be specified. +The other values are optional. + +``port`` needs to match the BASE ADDRESS jumper on the card (0x220 or 0x240) +or the value stored in the card's EEPROM for cards that have an EEPROM and +their "CONFIG MODE" jumper set to "EEPROM SETTING". The other values can +be chosen freely from the options enumerated above. + +If ``dma2`` is specified and different from ``dma1``, the card will operate in +full-duplex mode. When ``dma1=3``, only ``dma2=0`` is valid and the only way to +enable capture since only channels 0 and 1 are available for capture. + +Generic settings are ``port=0x220 wss_port=0x530 irq=10 dma1=1 dma2=0 +mpu_port=0x330 mpu_irq=9 fm_port=0x388``. + +Whatever IRQ and DMA channels you pick, be sure to reserve them for +legacy ISA in your BIOS. + +Module snd-azt2316 +------------------ + +Module for Aztech Sound Galaxy soundcards based on the Aztech AZT2316 +chipset. + +port + port # for BASE (0x220,0x240,0x260,0x280) +wss_port + port # for WSS (0x530,0x604,0xe80,0xf40) +irq + IRQ # for WSS (7,9,10,11) +dma1 + DMA # for WSS playback (0,1,3) +dma2 + DMA # for WSS capture (0,1), -1 = disabled (default) +mpu_port + port # for MPU-401 UART (0x300,0x330), -1 = disabled (default) +mpu_irq + IRQ # for MPU-401 UART (5,7,9,10), -1 = disabled (default) +fm_port + port # for OPL3 (0x388), -1 = disabled (default) + +This module supports multiple cards. It does not support autoprobe: +``port``, ``wss_port``, ``irq`` and ``dma1`` have to be specified. +The other values are optional. + +``port`` needs to match the BASE ADDRESS jumper on the card (0x220 or 0x240) +or the value stored in the card's EEPROM for cards that have an EEPROM and +their "CONFIG MODE" jumper set to "EEPROM SETTING". The other values can +be chosen freely from the options enumerated above. + +If ``dma2`` is specified and different from ``dma1``, the card will operate in +full-duplex mode. When ``dma1=3``, only ``dma2=0`` is valid and the only way to +enable capture since only channels 0 and 1 are available for capture. + +Generic settings are ``port=0x220 wss_port=0x530 irq=10 dma1=1 dma2=0 +mpu_port=0x330 mpu_irq=9 fm_port=0x388``. + +Whatever IRQ and DMA channels you pick, be sure to reserve them for +legacy ISA in your BIOS. + +Module snd-aw2 +-------------- + +Module for Audiowerk2 sound card + +This module supports multiple cards. + +Module snd-azt2320 +------------------ + +Module for sound cards based on Aztech System AZT2320 ISA chip (PnP only). + +This module supports multiple cards, PnP and autoprobe. + +The power-management is supported. + +Module snd-azt3328 +------------------ + +Module for sound cards based on Aztech AZF3328 PCI chip. + +joystick + Enable joystick (default off) + +This module supports multiple cards. + +Module snd-bt87x +---------------- + +Module for video cards based on Bt87x chips. + +digital_rate + Override the default digital rate (Hz) +load_all + Load the driver even if the card model isn't known + +This module supports multiple cards. + +Note: The default index value of this module is -2, i.e. the first +slot is excluded. + +Module snd-ca0106 +----------------- + +Module for Creative Audigy LS and SB Live 24bit + +This module supports multiple cards. + + +Module snd-cmi8330 +------------------ + +Module for sound cards based on C-Media CMI8330 ISA chips. + +isapnp + ISA PnP detection - 0 = disable, 1 = enable (default) + +with ``isapnp=0``, the following options are available: + +wssport + port # for CMI8330 chip (WSS) +wssirq + IRQ # for CMI8330 chip (WSS) +wssdma + first DMA # for CMI8330 chip (WSS) +sbport + port # for CMI8330 chip (SB16) +sbirq + IRQ # for CMI8330 chip (SB16) +sbdma8 + 8bit DMA # for CMI8330 chip (SB16) +sbdma16 + 16bit DMA # for CMI8330 chip (SB16) +fmport + (optional) OPL3 I/O port +mpuport + (optional) MPU401 I/O port +mpuirq + (optional) MPU401 irq # + +This module supports multiple cards and autoprobe. + +The power-management is supported. + +Module snd-cmipci +----------------- + +Module for C-Media CMI8338/8738/8768/8770 PCI sound cards. + +mpu_port + port address of MIDI interface (8338 only): + 0x300,0x310,0x320,0x330 = legacy port, + 0 = disable (default) +fm_port + port address of OPL-3 FM synthesizer (8x38 only): + 0x388 = legacy port, + 1 = integrated PCI port (default on 8738), + 0 = disable +soft_ac3 + Software-conversion of raw SPDIF packets (model 033 only) (default = 1) +joystick_port + Joystick port address (0 = disable, 1 = auto-detect) + +This module supports autoprobe and multiple cards. + +The power-management is supported. + +Module snd-cs4231 +----------------- + +Module for sound cards based on CS4231 ISA chips. + +port + port # for CS4231 chip +mpu_port + port # for MPU-401 UART (optional), -1 = disable +irq + IRQ # for CS4231 chip +mpu_irq + IRQ # for MPU-401 UART +dma1 + first DMA # for CS4231 chip +dma2 + second DMA # for CS4231 chip + +This module supports multiple cards. This module does not support autoprobe +thus main port must be specified!!! Other ports are optional. + +The power-management is supported. + +Module snd-cs4236 +----------------- + +Module for sound cards based on CS4232/CS4232A, +CS4235/CS4236/CS4236B/CS4237B/CS4238B/CS4239 ISA chips. + +isapnp + ISA PnP detection - 0 = disable, 1 = enable (default) + +with ``isapnp=0``, the following options are available: + +port + port # for CS4236 chip (PnP setup - 0x534) +cport + control port # for CS4236 chip (PnP setup - 0x120,0x210,0xf00) +mpu_port + port # for MPU-401 UART (PnP setup - 0x300), -1 = disable +fm_port + FM port # for CS4236 chip (PnP setup - 0x388), -1 = disable +irq + IRQ # for CS4236 chip (5,7,9,11,12,15) +mpu_irq + IRQ # for MPU-401 UART (9,11,12,15) +dma1 + first DMA # for CS4236 chip (0,1,3) +dma2 + second DMA # for CS4236 chip (0,1,3), -1 = disable + +This module supports multiple cards. This module does not support autoprobe +(if ISA PnP is not used) thus main port and control port must be +specified!!! Other ports are optional. + +The power-management is supported. + +This module is aliased as snd-cs4232 since it provides the old +snd-cs4232 functionality, too. + +Module snd-cs4281 +----------------- + +Module for Cirrus Logic CS4281 soundchip. + +dual_codec + Secondary codec ID (0 = disable, default) + +This module supports multiple cards. + +The power-management is supported. + +Module snd-cs46xx +----------------- + +Module for PCI sound cards based on CS4610/CS4612/CS4614/CS4615/CS4622/ +CS4624/CS4630/CS4280 PCI chips. + +external_amp + Force to enable external amplifier. +thinkpad + Force to enable Thinkpad's CLKRUN control. +mmap_valid + Support OSS mmap mode (default = 0). + +This module supports multiple cards and autoprobe. +Usually external amp and CLKRUN controls are detected automatically +from PCI sub vendor/device ids. If they don't work, give the options +above explicitly. + +The power-management is supported. + +Module snd-cs5530 +----------------- + +Module for Cyrix/NatSemi Geode 5530 chip. + +Module snd-cs5535audio +---------------------- + +Module for multifunction CS5535 companion PCI device + +The power-management is supported. + +Module snd-ctxfi +---------------- + +Module for Creative Sound Blaster X-Fi boards (20k1 / 20k2 chips) + +* Creative Sound Blaster X-Fi Titanium Fatal1ty Champion Series +* Creative Sound Blaster X-Fi Titanium Fatal1ty Professional Series +* Creative Sound Blaster X-Fi Titanium Professional Audio +* Creative Sound Blaster X-Fi Titanium +* Creative Sound Blaster X-Fi Elite Pro +* Creative Sound Blaster X-Fi Platinum +* Creative Sound Blaster X-Fi Fatal1ty +* Creative Sound Blaster X-Fi XtremeGamer +* Creative Sound Blaster X-Fi XtremeMusic + +reference_rate + reference sample rate, 44100 or 48000 (default) +multiple + multiple to ref. sample rate, 1 or 2 (default) +subsystem + override the PCI SSID for probing; + the value consists of SSVID << 16 | SSDID. + The default is zero, which means no override. + +This module supports multiple cards. + +Module snd-darla20 +------------------ + +Module for Echoaudio Darla20 + +This module supports multiple cards. +The driver requires the firmware loader support on kernel. + +Module snd-darla24 +------------------ + +Module for Echoaudio Darla24 + +This module supports multiple cards. +The driver requires the firmware loader support on kernel. + +Module snd-dt019x +----------------- + +Module for Diamond Technologies DT-019X / Avance Logic ALS-007 (PnP +only) + +This module supports multiple cards. This module is enabled only with +ISA PnP support. + +The power-management is supported. + +Module snd-dummy +---------------- + +Module for the dummy sound card. This "card" doesn't do any output +or input, but you may use this module for any application which +requires a sound card (like RealPlayer). + +pcm_devs + Number of PCM devices assigned to each card (default = 1, up to 4) +pcm_substreams + Number of PCM substreams assigned to each PCM (default = 8, up to 128) +hrtimer + Use hrtimer (=1, default) or system timer (=0) +fake_buffer + Fake buffer allocations (default = 1) + +When multiple PCM devices are created, snd-dummy gives different +behavior to each PCM device: +* 0 = interleaved with mmap support +* 1 = non-interleaved with mmap support +* 2 = interleaved without mmap +* 3 = non-interleaved without mmap + +As default, snd-dummy drivers doesn't allocate the real buffers +but either ignores read/write or mmap a single dummy page to all +buffer pages, in order to save the resources. If your apps need +the read/ written buffer data to be consistent, pass fake_buffer=0 +option. + +The power-management is supported. + +Module snd-echo3g +----------------- + +Module for Echoaudio 3G cards (Gina3G/Layla3G) + +This module supports multiple cards. +The driver requires the firmware loader support on kernel. + +Module snd-emu10k1 +------------------ + +Module for EMU10K1/EMU10k2 based PCI sound cards. + +* Sound Blaster Live! +* Sound Blaster PCI 512 +* Emu APS (partially supported) +* Sound Blaster Audigy + +extin + bitmap of available external inputs for FX8010 (see bellow) +extout + bitmap of available external outputs for FX8010 (see bellow) +seq_ports + allocated sequencer ports (4 by default) +max_synth_voices + limit of voices used for wavetable (64 by default) +max_buffer_size + specifies the maximum size of wavetable/pcm buffers given in MB + unit. Default value is 128. +enable_ir + enable IR + +This module supports multiple cards and autoprobe. + +Input & Output configurations [extin/extout] +* Creative Card wo/Digital out [0x0003/0x1f03] +* Creative Card w/Digital out [0x0003/0x1f0f] +* Creative Card w/Digital CD in [0x000f/0x1f0f] +* Creative Card wo/Digital out + LiveDrive [0x3fc3/0x1fc3] +* Creative Card w/Digital out + LiveDrive [0x3fc3/0x1fcf] +* Creative Card w/Digital CD in + LiveDrive [0x3fcf/0x1fcf] +* Creative Card wo/Digital out + Digital I/O 2 [0x0fc3/0x1f0f] +* Creative Card w/Digital out + Digital I/O 2 [0x0fc3/0x1f0f] +* Creative Card w/Digital CD in + Digital I/O 2 [0x0fcf/0x1f0f] +* Creative Card 5.1/w Digital out + LiveDrive [0x3fc3/0x1fff] +* Creative Card 5.1 (c) 2003 [0x3fc3/0x7cff] +* Creative Card all ins and outs [0x3fff/0x7fff] + +The power-management is supported. + +Module snd-emu10k1x +------------------- + +Module for Creative Emu10k1X (SB Live Dell OEM version) + +This module supports multiple cards. + +Module snd-ens1370 +------------------ + +Module for Ensoniq AudioPCI ES1370 PCI sound cards. + +* SoundBlaster PCI 64 +* SoundBlaster PCI 128 + +joystick + Enable joystick (default off) + +This module supports multiple cards and autoprobe. + +The power-management is supported. + +Module snd-ens1371 +------------------ + +Module for Ensoniq AudioPCI ES1371 PCI sound cards. + +* SoundBlaster PCI 64 +* SoundBlaster PCI 128 +* SoundBlaster Vibra PCI + +joystick_port + port # for joystick (0x200,0x208,0x210,0x218), 0 = disable + (default), 1 = auto-detect + +This module supports multiple cards and autoprobe. + +The power-management is supported. + +Module snd-es1688 +----------------- + +Module for ESS AudioDrive ES-1688 and ES-688 sound cards. + +isapnp + ISA PnP detection - 0 = disable, 1 = enable (default) +mpu_port + port # for MPU-401 port (0x300,0x310,0x320,0x330), -1 = disable (default) +mpu_irq + IRQ # for MPU-401 port (5,7,9,10) +fm_port + port # for OPL3 (option; share the same port as default) + +with ``isapnp=0``, the following additional options are available: + +port + port # for ES-1688 chip (0x220,0x240,0x260) +irq + IRQ # for ES-1688 chip (5,7,9,10) +dma8 + DMA # for ES-1688 chip (0,1,3) + +This module supports multiple cards and autoprobe (without MPU-401 port) +and PnP with the ES968 chip. + +Module snd-es18xx +----------------- + +Module for ESS AudioDrive ES-18xx sound cards. + +isapnp + ISA PnP detection - 0 = disable, 1 = enable (default) + +with ``isapnp=0``, the following options are available: + +port + port # for ES-18xx chip (0x220,0x240,0x260) +mpu_port + port # for MPU-401 port (0x300,0x310,0x320,0x330), -1 = disable (default) +fm_port + port # for FM (optional, not used) +irq + IRQ # for ES-18xx chip (5,7,9,10) +dma1 + first DMA # for ES-18xx chip (0,1,3) +dma2 + first DMA # for ES-18xx chip (0,1,3) + +This module supports multiple cards, ISA PnP and autoprobe (without MPU-401 +port if native ISA PnP routines are not used). +When ``dma2`` is equal with ``dma1``, the driver works as half-duplex. + +The power-management is supported. + +Module snd-es1938 +----------------- + +Module for sound cards based on ESS Solo-1 (ES1938,ES1946) chips. + +This module supports multiple cards and autoprobe. + +The power-management is supported. + +Module snd-es1968 +----------------- + +Module for sound cards based on ESS Maestro-1/2/2E (ES1968/ES1978) chips. + +total_bufsize + total buffer size in kB (1-4096kB) +pcm_substreams_p + playback channels (1-8, default=2) +pcm_substreams_c + capture channels (1-8, default=0) +clock + clock (0 = auto-detection) +use_pm + support the power-management (0 = off, 1 = on, 2 = auto (default)) +enable_mpu + enable MPU401 (0 = off, 1 = on, 2 = auto (default)) +joystick + enable joystick (default off) + +This module supports multiple cards and autoprobe. + +The power-management is supported. + +Module snd-fm801 +---------------- + +Module for ForteMedia FM801 based PCI sound cards. + +tea575x_tuner + Enable TEA575x tuner; + 1 = MediaForte 256-PCS, + 2 = MediaForte 256-PCPR, + 3 = MediaForte 64-PCR + High 16-bits are video (radio) device number + 1; + example: 0x10002 (MediaForte 256-PCPR, device 1) + +This module supports multiple cards and autoprobe. + +The power-management is supported. + +Module snd-gina20 +----------------- + +Module for Echoaudio Gina20 + +This module supports multiple cards. +The driver requires the firmware loader support on kernel. + +Module snd-gina24 +----------------- + +Module for Echoaudio Gina24 + +This module supports multiple cards. +The driver requires the firmware loader support on kernel. + +Module snd-gusclassic +--------------------- + +Module for Gravis UltraSound Classic sound card. + +port + port # for GF1 chip (0x220,0x230,0x240,0x250,0x260) +irq + IRQ # for GF1 chip (3,5,9,11,12,15) +dma1 + DMA # for GF1 chip (1,3,5,6,7) +dma2 + DMA # for GF1 chip (1,3,5,6,7,-1=disable) +joystick_dac + 0 to 31, (0.59V-4.52V or 0.389V-2.98V) +voices + GF1 voices limit (14-32) +pcm_voices + reserved PCM voices + +This module supports multiple cards and autoprobe. + +Module snd-gusextreme +--------------------- + +Module for Gravis UltraSound Extreme (Synergy ViperMax) sound card. + +port + port # for ES-1688 chip (0x220,0x230,0x240,0x250,0x260) +gf1_port + port # for GF1 chip (0x210,0x220,0x230,0x240,0x250,0x260,0x270) +mpu_port + port # for MPU-401 port (0x300,0x310,0x320,0x330), -1 = disable +irq + IRQ # for ES-1688 chip (5,7,9,10) +gf1_irq + IRQ # for GF1 chip (3,5,9,11,12,15) +mpu_irq + IRQ # for MPU-401 port (5,7,9,10) +dma8 + DMA # for ES-1688 chip (0,1,3) +dma1 + DMA # for GF1 chip (1,3,5,6,7) +joystick_dac + 0 to 31, (0.59V-4.52V or 0.389V-2.98V) +voices + GF1 voices limit (14-32) +pcm_voices + reserved PCM voices + +This module supports multiple cards and autoprobe (without MPU-401 port). + +Module snd-gusmax +----------------- + +Module for Gravis UltraSound MAX sound card. + +port + port # for GF1 chip (0x220,0x230,0x240,0x250,0x260) +irq + IRQ # for GF1 chip (3,5,9,11,12,15) +dma1 + DMA # for GF1 chip (1,3,5,6,7) +dma2 + DMA # for GF1 chip (1,3,5,6,7,-1=disable) +joystick_dac + 0 to 31, (0.59V-4.52V or 0.389V-2.98V) +voices + GF1 voices limit (14-32) +pcm_voices + reserved PCM voices + +This module supports multiple cards and autoprobe. + +Module snd-hda-intel +-------------------- + +Module for Intel HD Audio (ICH6, ICH6M, ESB2, ICH7, ICH8, ICH9, ICH10, +PCH, SCH), ATI SB450, SB600, R600, RS600, RS690, RS780, RV610, RV620, +RV630, RV635, RV670, RV770, VIA VT8251/VT8237A, SIS966, ULI M5461 + +[Multiple options for each card instance] + +model + force the model name +position_fix + Fix DMA pointer; + -1 = system default: choose appropriate one per controller hardware, + 0 = auto: falls back to LPIB when POSBUF doesn't work, + 1 = use LPIB, + 2 = POSBUF: use position buffer, + 3 = VIACOMBO: VIA-specific workaround for capture, + 4 = COMBO: use LPIB for playback, auto for capture stream +probe_mask + Bitmask to probe codecs (default = -1, meaning all slots); + When the bit 8 (0x100) is set, the lower 8 bits are used + as the "fixed" codec slots; i.e. the driver probes the + slots regardless what hardware reports back +probe_only + Only probing and no codec initialization (default=off); + Useful to check the initial codec status for debugging +bdl_pos_adj + Specifies the DMA IRQ timing delay in samples. + Passing -1 will make the driver to choose the appropriate + value based on the controller chip. +patch + Specifies the early "patch" files to modify the HD-audio setup + before initializing the codecs. + This option is available only when ``CONFIG_SND_HDA_PATCH_LOADER=y`` + is set. See hd-audio/notes.rst for details. +beep_mode + Selects the beep registration mode (0=off, 1=on); + default value is set via ``CONFIG_SND_HDA_INPUT_BEEP_MODE`` kconfig. + +[Single (global) options] + +single_cmd + Use single immediate commands to communicate with codecs + (for debugging only) +enable_msi + Enable Message Signaled Interrupt (MSI) (default = off) +power_save + Automatic power-saving timeout (in second, 0 = disable) +power_save_controller + Reset HD-audio controller in power-saving mode (default = on) +align_buffer_size + Force rounding of buffer/period sizes to multiples of 128 bytes. + This is more efficient in terms of memory access but isn't + required by the HDA spec and prevents users from specifying + exact period/buffer sizes. (default = on) +snoop + Enable/disable snooping (default = on) + +This module supports multiple cards and autoprobe. + +See hd-audio/notes.rst for more details about HD-audio driver. + +Each codec may have a model table for different configurations. +If your machine isn't listed there, the default (usually minimal) +configuration is set up. You can pass ``model=`` option to +specify a certain model in such a case. There are different +models depending on the codec chip. The list of available models +is found in hd-audio/models.rst. + +The model name ``generic`` is treated as a special case. When this +model is given, the driver uses the generic codec parser without +"codec-patch". It's sometimes good for testing and debugging. + +If the default configuration doesn't work and one of the above +matches with your device, report it together with alsa-info.sh +output (with ``--no-upload`` option) to kernel bugzilla or alsa-devel +ML (see the section `Links and Addresses`_). + +``power_save`` and ``power_save_controller`` options are for power-saving +mode. See powersave.txt for details. + +Note 2: If you get click noises on output, try the module option +``position_fix=1`` or ``2``. ``position_fix=1`` will use the SD_LPIB +register value without FIFO size correction as the current +DMA pointer. ``position_fix=2`` will make the driver to use +the position buffer instead of reading SD_LPIB register. +(Usually SD_LPIB register is more accurate than the +position buffer.) + +``position_fix=3`` is specific to VIA devices. The position +of the capture stream is checked from both LPIB and POSBUF +values. ``position_fix=4`` is a combination mode, using LPIB +for playback and POSBUF for capture. + +NB: If you get many ``azx_get_response timeout`` messages at +loading, it's likely a problem of interrupts (e.g. ACPI irq +routing). Try to boot with options like ``pci=noacpi``. Also, you +can try ``single_cmd=1`` module option. This will switch the +communication method between HDA controller and codecs to the +single immediate commands instead of CORB/RIRB. Basically, the +single command mode is provided only for BIOS, and you won't get +unsolicited events, too. But, at least, this works independently +from the irq. Remember this is a last resort, and should be +avoided as much as possible... + +MORE NOTES ON ``azx_get_response timeout`` PROBLEMS: +On some hardware, you may need to add a proper probe_mask option +to avoid the ``azx_get_response timeout`` problem above, instead. +This occurs when the access to non-existing or non-working codec slot +(likely a modem one) causes a stall of the communication via HD-audio +bus. You can see which codec slots are probed by enabling +``CONFIG_SND_DEBUG_VERBOSE``, or simply from the file name of the codec +proc files. Then limit the slots to probe by probe_mask option. +For example, ``probe_mask=1`` means to probe only the first slot, and +``probe_mask=4`` means only the third slot. + +The power-management is supported. + +Module snd-hdsp +--------------- + +Module for RME Hammerfall DSP audio interface(s) + +This module supports multiple cards. + +Note: The firmware data can be automatically loaded via hotplug +when ``CONFIG_FW_LOADER`` is set. Otherwise, you need to load +the firmware via hdsploader utility included in alsa-tools +package. +The firmware data is found in alsa-firmware package. + +Note: snd-page-alloc module does the job which snd-hammerfall-mem +module did formerly. It will allocate the buffers in advance +when any HDSP cards are found. To make the buffer +allocation sure, load snd-page-alloc module in the early +stage of boot sequence. See `Early Buffer Allocation`_ +section. + +Module snd-hdspm +---------------- + +Module for RME HDSP MADI board. + +precise_ptr + Enable precise pointer, or disable. +line_outs_monitor + Send playback streams to analog outs by default. +enable_monitor + Enable Analog Out on Channel 63/64 by default. + +See hdspm.txt for details. + +Module snd-ice1712 +------------------ + +Module for Envy24 (ICE1712) based PCI sound cards. + +* MidiMan M Audio Delta 1010 +* MidiMan M Audio Delta 1010LT +* MidiMan M Audio Delta DiO 2496 +* MidiMan M Audio Delta 66 +* MidiMan M Audio Delta 44 +* MidiMan M Audio Delta 410 +* MidiMan M Audio Audiophile 2496 +* TerraTec EWS 88MT +* TerraTec EWS 88D +* TerraTec EWX 24/96 +* TerraTec DMX 6Fire +* TerraTec Phase 88 +* Hoontech SoundTrack DSP 24 +* Hoontech SoundTrack DSP 24 Value +* Hoontech SoundTrack DSP 24 Media 7.1 +* Event Electronics, EZ8 +* Digigram VX442 +* Lionstracs, Mediastaton +* Terrasoniq TS 88 + +model + Use the given board model, one of the following: + delta1010, dio2496, delta66, delta44, audiophile, delta410, + delta1010lt, vx442, ewx2496, ews88mt, ews88mt_new, ews88d, + dmx6fire, dsp24, dsp24_value, dsp24_71, ez8, + phase88, mediastation +omni + Omni I/O support for MidiMan M-Audio Delta44/66 +cs8427_timeout + reset timeout for the CS8427 chip (S/PDIF transceiver) in msec + resolution, default value is 500 (0.5 sec) + +This module supports multiple cards and autoprobe. +Note: The consumer part is not used with all Envy24 based cards (for +example in the MidiMan Delta siree). + +Note: The supported board is detected by reading EEPROM or PCI +SSID (if EEPROM isn't available). You can override the +model by passing ``model`` module option in case that the +driver isn't configured properly or you want to try another +type for testing. + +Module snd-ice1724 +------------------ + +Module for Envy24HT (VT/ICE1724), Envy24PT (VT1720) based PCI sound cards. + +* MidiMan M Audio Revolution 5.1 +* MidiMan M Audio Revolution 7.1 +* MidiMan M Audio Audiophile 192 +* AMP Ltd AUDIO2000 +* TerraTec Aureon 5.1 Sky +* TerraTec Aureon 7.1 Space +* TerraTec Aureon 7.1 Universe +* TerraTec Phase 22 +* TerraTec Phase 28 +* AudioTrak Prodigy 7.1 +* AudioTrak Prodigy 7.1 LT +* AudioTrak Prodigy 7.1 XT +* AudioTrak Prodigy 7.1 HIFI +* AudioTrak Prodigy 7.1 HD2 +* AudioTrak Prodigy 192 +* Pontis MS300 +* Albatron K8X800 Pro II +* Chaintech ZNF3-150 +* Chaintech ZNF3-250 +* Chaintech 9CJS +* Chaintech AV-710 +* Shuttle SN25P +* Onkyo SE-90PCI +* Onkyo SE-200PCI +* ESI Juli@ +* ESI Maya44 +* Hercules Fortissimo IV +* EGO-SYS WaveTerminal 192M + +model + Use the given board model, one of the following: + revo51, revo71, amp2000, prodigy71, prodigy71lt, + prodigy71xt, prodigy71hifi, prodigyhd2, prodigy192, + juli, aureon51, aureon71, universe, ap192, k8x800, + phase22, phase28, ms300, av710, se200pci, se90pci, + fortissimo4, sn25p, WT192M, maya44 + +This module supports multiple cards and autoprobe. + +Note: The supported board is detected by reading EEPROM or PCI +SSID (if EEPROM isn't available). You can override the +model by passing ``model`` module option in case that the +driver isn't configured properly or you want to try another +type for testing. + +Module snd-indigo +----------------- + +Module for Echoaudio Indigo + +This module supports multiple cards. +The driver requires the firmware loader support on kernel. + +Module snd-indigodj +------------------- + +Module for Echoaudio Indigo DJ + +This module supports multiple cards. +The driver requires the firmware loader support on kernel. + +Module snd-indigoio +------------------- + +Module for Echoaudio Indigo IO + +This module supports multiple cards. +The driver requires the firmware loader support on kernel. + +Module snd-intel8x0 +------------------- + +Module for AC'97 motherboards from Intel and compatibles. + +* Intel i810/810E, i815, i820, i830, i84x, MX440 ICH5, ICH6, ICH7, + 6300ESB, ESB2 +* SiS 7012 (SiS 735) +* NVidia NForce, NForce2, NForce3, MCP04, CK804 CK8, CK8S, MCP501 +* AMD AMD768, AMD8111 +* ALi m5455 + +ac97_clock + AC'97 codec clock base (0 = auto-detect) +ac97_quirk + AC'97 workaround for strange hardware; + See `AC97 Quirk Option`_ section below. +buggy_irq + Enable workaround for buggy interrupts on some motherboards + (default yes on nForce chips, otherwise off) +buggy_semaphore + Enable workaround for hardware with buggy semaphores (e.g. on some + ASUS laptops) (default off) +spdif_aclink + Use S/PDIF over AC-link instead of direct connection from the + controller chip (0 = off, 1 = on, -1 = default) + +This module supports one chip and autoprobe. + +Note: the latest driver supports auto-detection of chip clock. +if you still encounter too fast playback, specify the clock +explicitly via the module option ``ac97_clock=41194``. + +Joystick/MIDI ports are not supported by this driver. If your +motherboard has these devices, use the ns558 or snd-mpu401 +modules, respectively. + +The power-management is supported. + +Module snd-intel8x0m +-------------------- + +Module for Intel ICH (i8x0) chipset MC97 modems. + +* Intel i810/810E, i815, i820, i830, i84x, MX440 ICH5, ICH6, ICH7 +* SiS 7013 (SiS 735) +* NVidia NForce, NForce2, NForce2s, NForce3 +* AMD AMD8111 +* ALi m5455 + +ac97_clock + AC'97 codec clock base (0 = auto-detect) + +This module supports one card and autoprobe. + +Note: The default index value of this module is -2, i.e. the first +slot is excluded. + +The power-management is supported. + +Module snd-interwave +-------------------- + +Module for Gravis UltraSound PnP, Dynasonic 3-D/Pro, STB Sound Rage 32 +and other sound cards based on AMD InterWave (tm) chip. + +joystick_dac + 0 to 31, (0.59V-4.52V or 0.389V-2.98V) +midi + 1 = MIDI UART enable, 0 = MIDI UART disable (default) +pcm_voices + reserved PCM voices for the synthesizer (default 2) +effect + 1 = InterWave effects enable (default 0); requires 8 voices +isapnp + ISA PnP detection - 0 = disable, 1 = enable (default) + +with ``isapnp=0``, the following options are available: + +port + port # for InterWave chip (0x210,0x220,0x230,0x240,0x250,0x260) +irq + IRQ # for InterWave chip (3,5,9,11,12,15) +dma1 + DMA # for InterWave chip (0,1,3,5,6,7) +dma2 + DMA # for InterWave chip (0,1,3,5,6,7,-1=disable) + +This module supports multiple cards, autoprobe and ISA PnP. + +Module snd-interwave-stb +------------------------ + +Module for UltraSound 32-Pro (sound card from STB used by Compaq) +and other sound cards based on AMD InterWave (tm) chip with TEA6330T +circuit for extended control of bass, treble and master volume. + +joystick_dac + 0 to 31, (0.59V-4.52V or 0.389V-2.98V) +midi + 1 = MIDI UART enable, 0 = MIDI UART disable (default) +pcm_voices + reserved PCM voices for the synthesizer (default 2) +effect + 1 = InterWave effects enable (default 0); requires 8 voices +isapnp + ISA PnP detection - 0 = disable, 1 = enable (default) + +with ``isapnp=0``, the following options are available: + +port + port # for InterWave chip (0x210,0x220,0x230,0x240,0x250,0x260) +port_tc + tone control (i2c bus) port # for TEA6330T chip (0x350,0x360,0x370,0x380) +irq + IRQ # for InterWave chip (3,5,9,11,12,15) +dma1 + DMA # for InterWave chip (0,1,3,5,6,7) +dma2 + DMA # for InterWave chip (0,1,3,5,6,7,-1=disable) + +This module supports multiple cards, autoprobe and ISA PnP. + +Module snd-jazz16 +------------------- + +Module for Media Vision Jazz16 chipset. The chipset consists of 3 chips: +MVD1216 + MVA416 + MVA514. + +port + port # for SB DSP chip (0x210,0x220,0x230,0x240,0x250,0x260) +irq + IRQ # for SB DSP chip (3,5,7,9,10,15) +dma8 + DMA # for SB DSP chip (1,3) +dma16 + DMA # for SB DSP chip (5,7) +mpu_port + MPU-401 port # (0x300,0x310,0x320,0x330) +mpu_irq + MPU-401 irq # (2,3,5,7) + +This module supports multiple cards. + +Module snd-korg1212 +------------------- + +Module for Korg 1212 IO PCI card + +This module supports multiple cards. + +Module snd-layla20 +------------------ + +Module for Echoaudio Layla20 + +This module supports multiple cards. +The driver requires the firmware loader support on kernel. + +Module snd-layla24 +------------------ + +Module for Echoaudio Layla24 + +This module supports multiple cards. +The driver requires the firmware loader support on kernel. + +Module snd-lola +--------------- + +Module for Digigram Lola PCI-e boards + +This module supports multiple cards. + +Module snd-lx6464es +------------------- + +Module for Digigram LX6464ES boards + +This module supports multiple cards. + +Module snd-maestro3 +------------------- + +Module for Allegro/Maestro3 chips + +external_amp + enable external amp (enabled by default) +amp_gpio + GPIO pin number for external amp (0-15) or -1 for default pin (8 + for allegro, 1 for others) + +This module supports autoprobe and multiple chips. + +Note: the binding of amplifier is dependent on hardware. +If there is no sound even though all channels are unmuted, try to +specify other gpio connection via amp_gpio option. +For example, a Panasonic notebook might need ``amp_gpio=0x0d`` +option. + +The power-management is supported. + +Module snd-mia +--------------- + +Module for Echoaudio Mia + +This module supports multiple cards. +The driver requires the firmware loader support on kernel. + +Module snd-miro +--------------- + +Module for Miro soundcards: miroSOUND PCM 1 pro, miroSOUND PCM 12, +miroSOUND PCM 20 Radio. + +port + Port # (0x530,0x604,0xe80,0xf40) +irq + IRQ # (5,7,9,10,11) +dma1 + 1st dma # (0,1,3) +dma2 + 2nd dma # (0,1) +mpu_port + MPU-401 port # (0x300,0x310,0x320,0x330) +mpu_irq + MPU-401 irq # (5,7,9,10) +fm_port + FM Port # (0x388) +wss + enable WSS mode +ide + enable onboard ide support + +Module snd-mixart +----------------- + +Module for Digigram miXart8 sound cards. + +This module supports multiple cards. +Note: One miXart8 board will be represented as 4 alsa cards. +See MIXART.txt for details. + +When the driver is compiled as a module and the hotplug firmware +is supported, the firmware data is loaded via hotplug automatically. +Install the necessary firmware files in alsa-firmware package. +When no hotplug fw loader is available, you need to load the +firmware via mixartloader utility in alsa-tools package. + +Module snd-mona +--------------- + +Module for Echoaudio Mona + +This module supports multiple cards. +The driver requires the firmware loader support on kernel. + +Module snd-mpu401 +----------------- + +Module for MPU-401 UART devices. + +port + port number or -1 (disable) +irq + IRQ number or -1 (disable) +pnp + PnP detection - 0 = disable, 1 = enable (default) + +This module supports multiple devices and PnP. + +Module snd-msnd-classic +----------------------- + +Module for Turtle Beach MultiSound Classic, Tahiti or Monterey +soundcards. + +io + Port # for msnd-classic card +irq + IRQ # for msnd-classic card +mem + Memory address (0xb0000, 0xc8000, 0xd0000, 0xd8000, 0xe0000 or 0xe8000) +write_ndelay + enable write ndelay (default = 1) +calibrate_signal + calibrate signal (default = 0) +isapnp + ISA PnP detection - 0 = disable, 1 = enable (default) +digital + Digital daughterboard present (default = 0) +cfg + Config port (0x250, 0x260 or 0x270) default = PnP +reset + Reset all devices +mpu_io + MPU401 I/O port +mpu_irq + MPU401 irq# +ide_io0 + IDE port #0 +ide_io1 + IDE port #1 +ide_irq + IDE irq# +joystick_io + Joystick I/O port + +The driver requires firmware files ``turtlebeach/msndinit.bin`` and +``turtlebeach/msndperm.bin`` in the proper firmware directory. + +See Documentation/sound/oss/MultiSound for important information +about this driver. Note that it has been discontinued, but the +Voyetra Turtle Beach knowledge base entry for it is still available +at +http://www.turtlebeach.com + +Module snd-msnd-pinnacle +------------------------ + +Module for Turtle Beach MultiSound Pinnacle/Fiji soundcards. + +io + Port # for pinnacle/fiji card +irq + IRQ # for pinnalce/fiji card +mem + Memory address (0xb0000, 0xc8000, 0xd0000, 0xd8000, 0xe0000 or 0xe8000) +write_ndelay + enable write ndelay (default = 1) +calibrate_signal + calibrate signal (default = 0) +isapnp + ISA PnP detection - 0 = disable, 1 = enable (default) + +The driver requires firmware files ``turtlebeach/pndspini.bin`` and +``turtlebeach/pndsperm.bin`` in the proper firmware directory. + +Module snd-mtpav +---------------- + +Module for MOTU MidiTimePiece AV multiport MIDI (on the parallel +port). + +port + I/O port # for MTPAV (0x378,0x278, default=0x378) +irq + IRQ # for MTPAV (7,5, default=7) +hwports + number of supported hardware ports, default=8. + +Module supports only 1 card. This module has no enable option. + +Module snd-mts64 +---------------- + +Module for Ego Systems (ESI) Miditerminal 4140 + +This module supports multiple devices. +Requires parport (``CONFIG_PARPORT``). + +Module snd-nm256 +---------------- + +Module for NeoMagic NM256AV/ZX chips + +playback_bufsize + max playback frame size in kB (4-128kB) +capture_bufsize + max capture frame size in kB (4-128kB) +force_ac97 + 0 or 1 (disabled by default) +buffer_top + specify buffer top address +use_cache + 0 or 1 (disabled by default) +vaio_hack + alias buffer_top=0x25a800 +reset_workaround + enable AC97 RESET workaround for some laptops +reset_workaround2 + enable extended AC97 RESET workaround for some other laptops + +This module supports one chip and autoprobe. + +The power-management is supported. + +Note: on some notebooks the buffer address cannot be detected +automatically, or causes hang-up during initialization. +In such a case, specify the buffer top address explicitly via +the buffer_top option. +For example, +Sony F250: buffer_top=0x25a800 +Sony F270: buffer_top=0x272800 +The driver supports only ac97 codec. It's possible to force +to initialize/use ac97 although it's not detected. In such a +case, use ``force_ac97=1`` option - but *NO* guarantee whether it +works! + +Note: The NM256 chip can be linked internally with non-AC97 +codecs. This driver supports only the AC97 codec, and won't work +with machines with other (most likely CS423x or OPL3SAx) chips, +even though the device is detected in lspci. In such a case, try +other drivers, e.g. snd-cs4232 or snd-opl3sa2. Some has ISA-PnP +but some doesn't have ISA PnP. You'll need to specify ``isapnp=0`` +and proper hardware parameters in the case without ISA PnP. + +Note: some laptops need a workaround for AC97 RESET. For the +known hardware like Dell Latitude LS and Sony PCG-F305, this +workaround is enabled automatically. For other laptops with a +hard freeze, you can try ``reset_workaround=1`` option. + +Note: Dell Latitude CSx laptops have another problem regarding +AC97 RESET. On these laptops, reset_workaround2 option is +turned on as default. This option is worth to try if the +previous reset_workaround option doesn't help. + +Note: This driver is really crappy. It's a porting from the +OSS driver, which is a result of black-magic reverse engineering. +The detection of codec will fail if the driver is loaded *after* +X-server as described above. You might be able to force to load +the module, but it may result in hang-up. Hence, make sure that +you load this module *before* X if you encounter this kind of +problem. + +Module snd-opl3sa2 +------------------ + +Module for Yamaha OPL3-SA2/SA3 sound cards. + +isapnp + ISA PnP detection - 0 = disable, 1 = enable (default) + +with ``isapnp=0``, the following options are available: + +port + control port # for OPL3-SA chip (0x370) +sb_port + SB port # for OPL3-SA chip (0x220,0x240) +wss_port + WSS port # for OPL3-SA chip (0x530,0xe80,0xf40,0x604) +midi_port + port # for MPU-401 UART (0x300,0x330), -1 = disable +fm_port + FM port # for OPL3-SA chip (0x388), -1 = disable +irq + IRQ # for OPL3-SA chip (5,7,9,10) +dma1 + first DMA # for Yamaha OPL3-SA chip (0,1,3) +dma2 + second DMA # for Yamaha OPL3-SA chip (0,1,3), -1 = disable + +This module supports multiple cards and ISA PnP. It does not support +autoprobe (if ISA PnP is not used) thus all ports must be specified!!! + +The power-management is supported. + +Module snd-opti92x-ad1848 +------------------------- + +Module for sound cards based on OPTi 82c92x and Analog Devices AD1848 chips. +Module works with OAK Mozart cards as well. + +isapnp + ISA PnP detection - 0 = disable, 1 = enable (default) + +with ``isapnp=0``, the following options are available: + +port + port # for WSS chip (0x530,0xe80,0xf40,0x604) +mpu_port + port # for MPU-401 UART (0x300,0x310,0x320,0x330) +fm_port + port # for OPL3 device (0x388) +irq + IRQ # for WSS chip (5,7,9,10,11) +mpu_irq + IRQ # for MPU-401 UART (5,7,9,10) +dma1 + first DMA # for WSS chip (0,1,3) + +This module supports only one card, autoprobe and PnP. + +Module snd-opti92x-cs4231 +------------------------- + +Module for sound cards based on OPTi 82c92x and Crystal CS4231 chips. + +isapnp + ISA PnP detection - 0 = disable, 1 = enable (default) + +with ``isapnp=0``, the following options are available: + +port + port # for WSS chip (0x530,0xe80,0xf40,0x604) +mpu_port + port # for MPU-401 UART (0x300,0x310,0x320,0x330) +fm_port + port # for OPL3 device (0x388) +irq + IRQ # for WSS chip (5,7,9,10,11) +mpu_irq + IRQ # for MPU-401 UART (5,7,9,10) +dma1 + first DMA # for WSS chip (0,1,3) +dma2 + second DMA # for WSS chip (0,1,3) + +This module supports only one card, autoprobe and PnP. + +Module snd-opti93x +------------------ + +Module for sound cards based on OPTi 82c93x chips. + +isapnp + ISA PnP detection - 0 = disable, 1 = enable (default) + +with ``isapnp=0``, the following options are available: + +port + port # for WSS chip (0x530,0xe80,0xf40,0x604) +mpu_port + port # for MPU-401 UART (0x300,0x310,0x320,0x330) +fm_port + port # for OPL3 device (0x388) +irq + IRQ # for WSS chip (5,7,9,10,11) +mpu_irq + IRQ # for MPU-401 UART (5,7,9,10) +dma1 + first DMA # for WSS chip (0,1,3) +dma2 + second DMA # for WSS chip (0,1,3) + +This module supports only one card, autoprobe and PnP. + +Module snd-oxygen +----------------- + +Module for sound cards based on the C-Media CMI8786/8787/8788 chip: + +* Asound A-8788 +* Asus Xonar DG/DGX +* AuzenTech X-Meridian +* AuzenTech X-Meridian 2G +* Bgears b-Enspirer +* Club3D Theatron DTS +* HT-Omega Claro (plus) +* HT-Omega Claro halo (XT) +* Kuroutoshikou CMI8787-HG2PCI +* Razer Barracuda AC-1 +* Sondigo Inferno +* TempoTec HiFier Fantasia +* TempoTec HiFier Serenade + +This module supports autoprobe and multiple cards. + +Module snd-pcsp +--------------- + +Module for internal PC-Speaker. + +nopcm + Disable PC-Speaker PCM sound. Only beeps remain. +nforce_wa + enable NForce chipset workaround. Expect bad sound. + +This module supports system beeps, some kind of PCM playback and +even a few mixer controls. + +Module snd-pcxhr +---------------- + +Module for Digigram PCXHR boards + +This module supports multiple cards. + +Module snd-portman2x4 +--------------------- + +Module for Midiman Portman 2x4 parallel port MIDI interface + +This module supports multiple cards. + +Module snd-powermac (on ppc only) +--------------------------------- + +Module for PowerMac, iMac and iBook on-board soundchips + +enable_beep + enable beep using PCM (enabled as default) + +Module supports autoprobe a chip. + +Note: the driver may have problems regarding endianness. + +The power-management is supported. + +Module snd-pxa2xx-ac97 (on arm only) +------------------------------------ + +Module for AC97 driver for the Intel PXA2xx chip + +For ARM architecture only. + +The power-management is supported. + +Module snd-riptide +------------------ + +Module for Conexant Riptide chip + +joystick_port + Joystick port # (default: 0x200) +mpu_port + MPU401 port # (default: 0x330) +opl3_port + OPL3 port # (default: 0x388) + +This module supports multiple cards. +The driver requires the firmware loader support on kernel. +You need to install the firmware file ``riptide.hex`` to the standard +firmware path (e.g. /lib/firmware). + +Module snd-rme32 +---------------- + +Module for RME Digi32, Digi32 Pro and Digi32/8 (Sek'd Prodif32, +Prodif96 and Prodif Gold) sound cards. + +This module supports multiple cards. + +Module snd-rme96 +---------------- + +Module for RME Digi96, Digi96/8 and Digi96/8 PRO/PAD/PST sound cards. + +This module supports multiple cards. + +Module snd-rme9652 +------------------ + +Module for RME Digi9652 (Hammerfall, Hammerfall-Light) sound cards. + +precise_ptr + Enable precise pointer (doesn't work reliably). (default = 0) + +This module supports multiple cards. + +Note: snd-page-alloc module does the job which snd-hammerfall-mem +module did formerly. It will allocate the buffers in advance +when any RME9652 cards are found. To make the buffer +allocation sure, load snd-page-alloc module in the early +stage of boot sequence. See `Early Buffer Allocation`_ +section. + +Module snd-sa11xx-uda1341 (on arm only) +--------------------------------------- + +Module for Philips UDA1341TS on Compaq iPAQ H3600 sound card. + +Module supports only one card. +Module has no enable and index options. + +The power-management is supported. + +Module snd-sb8 +-------------- + +Module for 8-bit SoundBlaster cards: SoundBlaster 1.0, SoundBlaster 2.0, +SoundBlaster Pro + +port + port # for SB DSP chip (0x220,0x240,0x260) +irq + IRQ # for SB DSP chip (5,7,9,10) +dma8 + DMA # for SB DSP chip (1,3) + +This module supports multiple cards and autoprobe. + +The power-management is supported. + +Module snd-sb16 and snd-sbawe +----------------------------- + +Module for 16-bit SoundBlaster cards: SoundBlaster 16 (PnP), +SoundBlaster AWE 32 (PnP), SoundBlaster AWE 64 PnP + +mic_agc + Mic Auto-Gain-Control - 0 = disable, 1 = enable (default) +csp + ASP/CSP chip support - 0 = disable (default), 1 = enable +isapnp + ISA PnP detection - 0 = disable, 1 = enable (default) + +with isapnp=0, the following options are available: + +port + port # for SB DSP 4.x chip (0x220,0x240,0x260) +mpu_port + port # for MPU-401 UART (0x300,0x330), -1 = disable +awe_port + base port # for EMU8000 synthesizer (0x620,0x640,0x660) (snd-sbawe + module only) +irq + IRQ # for SB DSP 4.x chip (5,7,9,10) +dma8 + 8-bit DMA # for SB DSP 4.x chip (0,1,3) +dma16 + 16-bit DMA # for SB DSP 4.x chip (5,6,7) + +This module supports multiple cards, autoprobe and ISA PnP. + +Note: To use Vibra16X cards in 16-bit half duplex mode, you must +disable 16bit DMA with dma16 = -1 module parameter. +Also, all Sound Blaster 16 type cards can operate in 16-bit +half duplex mode through 8-bit DMA channel by disabling their +16-bit DMA channel. + +The power-management is supported. + +Module snd-sc6000 +----------------- + +Module for Gallant SC-6000 soundcard and later models: SC-6600 and +SC-7000. + +port + Port # (0x220 or 0x240) +mss_port + MSS Port # (0x530 or 0xe80) +irq + IRQ # (5,7,9,10,11) +mpu_irq + MPU-401 IRQ # (5,7,9,10) ,0 - no MPU-401 irq +dma + DMA # (1,3,0) +joystick + Enable gameport - 0 = disable (default), 1 = enable + +This module supports multiple cards. + +This card is also known as Audio Excel DSP 16 or Zoltrix AV302. + +Module snd-sscape +----------------- + +Module for ENSONIQ SoundScape cards. + +port + Port # (PnP setup) +wss_port + WSS Port # (PnP setup) +irq + IRQ # (PnP setup) +mpu_irq + MPU-401 IRQ # (PnP setup) +dma + DMA # (PnP setup) +dma2 + 2nd DMA # (PnP setup, -1 to disable) +joystick + Enable gameport - 0 = disable (default), 1 = enable + +This module supports multiple cards. + +The driver requires the firmware loader support on kernel. + +Module snd-sun-amd7930 (on sparc only) +-------------------------------------- + +Module for AMD7930 sound chips found on Sparcs. + +This module supports multiple cards. + +Module snd-sun-cs4231 (on sparc only) +------------------------------------- + +Module for CS4231 sound chips found on Sparcs. + +This module supports multiple cards. + +Module snd-sun-dbri (on sparc only) +----------------------------------- + +Module for DBRI sound chips found on Sparcs. + +This module supports multiple cards. + +Module snd-wavefront +-------------------- + +Module for Turtle Beach Maui, Tropez and Tropez+ sound cards. + +use_cs4232_midi + Use CS4232 MPU-401 interface + (inaccessibly located inside your computer) +isapnp + ISA PnP detection - 0 = disable, 1 = enable (default) + +with isapnp=0, the following options are available: + +cs4232_pcm_port + Port # for CS4232 PCM interface. +cs4232_pcm_irq + IRQ # for CS4232 PCM interface (5,7,9,11,12,15). +cs4232_mpu_port + Port # for CS4232 MPU-401 interface. +cs4232_mpu_irq + IRQ # for CS4232 MPU-401 interface (9,11,12,15). +ics2115_port + Port # for ICS2115 +ics2115_irq + IRQ # for ICS2115 +fm_port + FM OPL-3 Port # +dma1 + DMA1 # for CS4232 PCM interface. +dma2 + DMA2 # for CS4232 PCM interface. + +The below are options for wavefront_synth features: + +wf_raw + Assume that we need to boot the OS (default:no); + If yes, then during driver loading, the state of the board is + ignored, and we reset the board and load the firmware anyway. +fx_raw + Assume that the FX process needs help (default:yes); + If false, we'll leave the FX processor in whatever state it is + when the driver is loaded. The default is to download the + microprogram and associated coefficients to set it up for + "default" operation, whatever that means. +debug_default + Debug parameters for card initialization +wait_usecs + How long to wait without sleeping, usecs (default:150); + This magic number seems to give pretty optimal throughput + based on my limited experimentation. + If you want to play around with it and find a better value, be + my guest. Remember, the idea is to get a number that causes us + to just busy wait for as many WaveFront commands as possible, + without coming up with a number so large that we hog the whole + CPU. + Specifically, with this number, out of about 134,000 status + waits, only about 250 result in a sleep. +sleep_interval + How long to sleep when waiting for reply (default: 100) +sleep_tries + How many times to try sleeping during a wait (default: 50) +ospath + Pathname to processed ICS2115 OS firmware (default:wavefront.os); + The path name of the ISC2115 OS firmware. In the recent + version, it's handled via firmware loader framework, so it + must be installed in the proper path, typically, + /lib/firmware. +reset_time + How long to wait for a reset to take effect (default:2) +ramcheck_time + How many seconds to wait for the RAM test (default:20) +osrun_time + How many seconds to wait for the ICS2115 OS (default:10) + +This module supports multiple cards and ISA PnP. + +Note: the firmware file ``wavefront.os`` was located in the earlier +version in /etc. Now it's loaded via firmware loader, and +must be in the proper firmware path, such as /lib/firmware. +Copy (or symlink) the file appropriately if you get an error +regarding firmware downloading after upgrading the kernel. + +Module snd-sonicvibes +--------------------- + +Module for S3 SonicVibes PCI sound cards. +* PINE Schubert 32 PCI + +reverb + Reverb Enable - 1 = enable, 0 = disable (default); + SoundCard must have onboard SRAM for this. +mge + Mic Gain Enable - 1 = enable, 0 = disable (default) + +This module supports multiple cards and autoprobe. + +Module snd-serial-u16550 +------------------------ + +Module for UART16550A serial MIDI ports. + +port + port # for UART16550A chip +irq + IRQ # for UART16550A chip, -1 = poll mode +speed + speed in bauds (9600,19200,38400,57600,115200) + 38400 = default +base + base for divisor in bauds (57600,115200,230400,460800) + 115200 = default +outs + number of MIDI ports in a serial port (1-4) + 1 = default +adaptor + Type of adaptor. + 0 = Soundcanvas, 1 = MS-124T, 2 = MS-124W S/A, + 3 = MS-124W M/B, 4 = Generic + +This module supports multiple cards. This module does not support autoprobe +thus the main port must be specified!!! Other options are optional. + +Module snd-trident +------------------ + +Module for Trident 4DWave DX/NX sound cards. +* Best Union Miss Melody 4DWave PCI +* HIS 4DWave PCI +* Warpspeed ONSpeed 4DWave PCI +* AzTech PCI 64-Q3D +* Addonics SV 750 +* CHIC True Sound 4Dwave +* Shark Predator4D-PCI +* Jaton SonicWave 4D +* SiS SI7018 PCI Audio +* Hoontech SoundTrack Digital 4DWave NX + +pcm_channels + max channels (voices) reserved for PCM +wavetable_size + max wavetable size in kB (4-?kb) + +This module supports multiple cards and autoprobe. + +The power-management is supported. + +Module snd-ua101 +---------------- + +Module for the Edirol UA-101/UA-1000 audio/MIDI interfaces. + +This module supports multiple devices, autoprobe and hotplugging. + +Module snd-usb-audio +-------------------- + +Module for USB audio and USB MIDI devices. + +vid + Vendor ID for the device (optional) +pid + Product ID for the device (optional) +nrpacks + Max. number of packets per URB (default: 8) +device_setup + Device specific magic number (optional); + Influence depends on the device + Default: 0x0000 +ignore_ctl_error + Ignore any USB-controller regarding mixer interface (default: no) +autoclock + Enable auto-clock selection for UAC2 devices (default: yes) +quirk_alias + Quirk alias list, pass strings like ``0123abcd:5678beef``, which + applies the existing quirk for the device 5678:beef to a new + device 0123:abcd. + +This module supports multiple devices, autoprobe and hotplugging. + +NB: ``nrpacks`` parameter can be modified dynamically via sysfs. +Don't put the value over 20. Changing via sysfs has no sanity +check. + +NB: ``ignore_ctl_error=1`` may help when you get an error at accessing +the mixer element such as URB error -22. This happens on some +buggy USB device or the controller. + +NB: quirk_alias option is provided only for testing / development. +If you want to have a proper support, contact to upstream for +adding the matching quirk in the driver code statically. + +Module snd-usb-caiaq +-------------------- + +Module for caiaq UB audio interfaces, + +* Native Instruments RigKontrol2 +* Native Instruments Kore Controller +* Native Instruments Audio Kontrol 1 +* Native Instruments Audio 8 DJ + +This module supports multiple devices, autoprobe and hotplugging. + +Module snd-usb-usx2y +-------------------- + +Module for Tascam USB US-122, US-224 and US-428 devices. + +This module supports multiple devices, autoprobe and hotplugging. + +Note: you need to load the firmware via ``usx2yloader`` utility included +in alsa-tools and alsa-firmware packages. + +Module snd-via82xx +------------------ + +Module for AC'97 motherboards based on VIA 82C686A/686B, 8233, 8233A, +8233C, 8235, 8237 (south) bridge. + +mpu_port + 0x300,0x310,0x320,0x330, otherwise obtain BIOS setup + [VIA686A/686B only] +joystick + Enable joystick (default off) [VIA686A/686B only] +ac97_clock + AC'97 codec clock base (default 48000Hz) +dxs_support + support DXS channels, 0 = auto (default), 1 = enable, 2 = disable, + 3 = 48k only, 4 = no VRA, 5 = enable any sample rate and different + sample rates on different channels [VIA8233/C, 8235, 8237 only] +ac97_quirk + AC'97 workaround for strange hardware; + See `AC97 Quirk Option`_ section below. + +This module supports one chip and autoprobe. + +Note: on some SMP motherboards like MSI 694D the interrupts might +not be generated properly. In such a case, please try to +set the SMP (or MPS) version on BIOS to 1.1 instead of +default value 1.4. Then the interrupt number will be +assigned under 15. You might also upgrade your BIOS. + +Note: VIA8233/5/7 (not VIA8233A) can support DXS (direct sound) +channels as the first PCM. On these channels, up to 4 +streams can be played at the same time, and the controller +can perform sample rate conversion with separate rates for +each channel. +As default (``dxs_support = 0``), 48k fixed rate is chosen +except for the known devices since the output is often +noisy except for 48k on some mother boards due to the +bug of BIOS. +Please try once ``dxs_support=5`` and if it works on other +sample rates (e.g. 44.1kHz of mp3 playback), please let us +know the PCI subsystem vendor/device id's (output of +``lspci -nv``). +If ``dxs_support=5`` does not work, try ``dxs_support=4``; if it +doesn't work too, try dxs_support=1. (dxs_support=1 is +usually for old motherboards. The correct implemented +board should work with 4 or 5.) If it still doesn't +work and the default setting is ok, ``dxs_support=3`` is the +right choice. If the default setting doesn't work at all, +try ``dxs_support=2`` to disable the DXS channels. +In any cases, please let us know the result and the +subsystem vendor/device ids. See `Links and Addresses`_ +below. + +Note: for the MPU401 on VIA823x, use snd-mpu401 driver +additionally. The mpu_port option is for VIA686 chips only. + +The power-management is supported. + +Module snd-via82xx-modem +------------------------ + +Module for VIA82xx AC97 modem + +ac97_clock + AC'97 codec clock base (default 48000Hz) + +This module supports one card and autoprobe. + +Note: The default index value of this module is -2, i.e. the first +slot is excluded. + +The power-management is supported. + +Module snd-virmidi +------------------ + +Module for virtual rawmidi devices. +This module creates virtual rawmidi devices which communicate +to the corresponding ALSA sequencer ports. + +midi_devs + MIDI devices # (1-4, default=4) + +This module supports multiple cards. + +Module snd-virtuoso +------------------- + +Module for sound cards based on the Asus AV66/AV100/AV200 chips, +i.e., Xonar D1, DX, D2, D2X, DS, DSX, Essence ST (Deluxe), +Essence STX (II), HDAV1.3 (Deluxe), and HDAV1.3 Slim. + +This module supports autoprobe and multiple cards. + +Module snd-vx222 +---------------- + +Module for Digigram VX-Pocket VX222, V222 v2 and Mic cards. + +mic + Enable Microphone on V222 Mic (NYI) +ibl + Capture IBL size. (default = 0, minimum size) + +This module supports multiple cards. + +When the driver is compiled as a module and the hotplug firmware +is supported, the firmware data is loaded via hotplug automatically. +Install the necessary firmware files in alsa-firmware package. +When no hotplug fw loader is available, you need to load the +firmware via vxloader utility in alsa-tools package. To invoke +vxloader automatically, add the following to /etc/modprobe.d/alsa.conf + +:: + + install snd-vx222 /sbin/modprobe --first-time -i snd-vx222\ + && /usr/bin/vxloader + +(for 2.2/2.4 kernels, add ``post-install /usr/bin/vxloader`` to +/etc/modules.conf, instead.) +IBL size defines the interrupts period for PCM. The smaller size +gives smaller latency but leads to more CPU consumption, too. +The size is usually aligned to 126. As default (=0), the smallest +size is chosen. The possible IBL values can be found in +/proc/asound/cardX/vx-status proc file. + +The power-management is supported. + +Module snd-vxpocket +------------------- + +Module for Digigram VX-Pocket VX2 and 440 PCMCIA cards. + +ibl + Capture IBL size. (default = 0, minimum size) + +This module supports multiple cards. The module is compiled only when +PCMCIA is supported on kernel. + +With the older 2.6.x kernel, to activate the driver via the card +manager, you'll need to set up /etc/pcmcia/vxpocket.conf. See the +sound/pcmcia/vx/vxpocket.c. 2.6.13 or later kernel requires no +longer require a config file. + +When the driver is compiled as a module and the hotplug firmware +is supported, the firmware data is loaded via hotplug automatically. +Install the necessary firmware files in alsa-firmware package. +When no hotplug fw loader is available, you need to load the +firmware via vxloader utility in alsa-tools package. + +About capture IBL, see the description of snd-vx222 module. + +Note: snd-vxp440 driver is merged to snd-vxpocket driver since +ALSA 1.0.10. + +The power-management is supported. + +Module snd-ymfpci +----------------- + +Module for Yamaha PCI chips (YMF72x, YMF74x & YMF75x). + +mpu_port + 0x300,0x330,0x332,0x334, 0 (disable) by default, + 1 (auto-detect for YMF744/754 only) +fm_port + 0x388,0x398,0x3a0,0x3a8, 0 (disable) by default + 1 (auto-detect for YMF744/754 only) +joystick_port + 0x201,0x202,0x204,0x205, 0 (disable) by default, + 1 (auto-detect) +rear_switch + enable shared rear/line-in switch (bool) + +This module supports autoprobe and multiple chips. + +The power-management is supported. + +Module snd-pdaudiocf +-------------------- + +Module for Sound Core PDAudioCF sound card. + +The power-management is supported. + + +AC97 Quirk Option +================= + +The ac97_quirk option is used to enable/override the workaround for +specific devices on drivers for on-board AC'97 controllers like +snd-intel8x0. Some hardware have swapped output pins between Master +and Headphone, or Surround (thanks to confusion of AC'97 +specifications from version to version :-) + +The driver provides the auto-detection of known problematic devices, +but some might be unknown or wrongly detected. In such a case, pass +the proper value with this option. + +The following strings are accepted: + +default + Don't override the default setting +none + Disable the quirk +hp_only + Bind Master and Headphone controls as a single control +swap_hp + Swap headphone and master controls +swap_surround + Swap master and surround controls +ad_sharing + For AD1985, turn on OMS bit and use headphone +alc_jack + For ALC65x, turn on the jack sense mode +inv_eapd + Inverted EAPD implementation +mute_led + Bind EAPD bit for turning on/off mute LED + +For backward compatibility, the corresponding integer value -1, 0, ... +are accepted, too. + +For example, if ``Master`` volume control has no effect on your device +but only ``Headphone`` does, pass ac97_quirk=hp_only module option. + + +Configuring Non-ISAPNP Cards +============================ + +When the kernel is configured with ISA-PnP support, the modules +supporting the isapnp cards will have module options ``isapnp``. +If this option is set, *only* the ISA-PnP devices will be probed. +For probing the non ISA-PnP cards, you have to pass ``isapnp=0`` option +together with the proper i/o and irq configuration. + +When the kernel is configured without ISA-PnP support, isapnp option +will be not built in. + + +Module Autoloading Support +========================== + +The ALSA drivers can be loaded automatically on demand by defining +module aliases. The string ``snd-card-%1`` is requested for ALSA native +devices where ``%i`` is sound card number from zero to seven. + +To auto-load an ALSA driver for OSS services, define the string +``sound-slot-%i`` where ``%i`` means the slot number for OSS, which +corresponds to the card index of ALSA. Usually, define this +as the same card module. + +An example configuration for a single emu10k1 card is like below: +:: + + ----- /etc/modprobe.d/alsa.conf + alias snd-card-0 snd-emu10k1 + alias sound-slot-0 snd-emu10k1 + ----- /etc/modprobe.d/alsa.conf + +The available number of auto-loaded sound cards depends on the module +option ``cards_limit`` of snd module. As default it's set to 1. +To enable the auto-loading of multiple cards, specify the number of +sound cards in that option. + +When multiple cards are available, it'd better to specify the index +number for each card via module option, too, so that the order of +cards is kept consistent. + +An example configuration for two sound cards is like below: +:: + + ----- /etc/modprobe.d/alsa.conf + # ALSA portion + options snd cards_limit=2 + alias snd-card-0 snd-interwave + alias snd-card-1 snd-ens1371 + options snd-interwave index=0 + options snd-ens1371 index=1 + # OSS/Free portion + alias sound-slot-0 snd-interwave + alias sound-slot-1 snd-ens1371 + ----- /etc/modprobe.d/alsa.conf + +In this example, the interwave card is always loaded as the first card +(index 0) and ens1371 as the second (index 1). + +Alternative (and new) way to fixate the slot assignment is to use +``slots`` option of snd module. In the case above, specify like the +following: +:: + + options snd slots=snd-interwave,snd-ens1371 + +Then, the first slot (#0) is reserved for snd-interwave driver, and +the second (#1) for snd-ens1371. You can omit index option in each +driver if slots option is used (although you can still have them at +the same time as long as they don't conflict). + +The slots option is especially useful for avoiding the possible +hot-plugging and the resultant slot conflict. For example, in the +case above again, the first two slots are already reserved. If any +other driver (e.g. snd-usb-audio) is loaded before snd-interwave or +snd-ens1371, it will be assigned to the third or later slot. + +When a module name is given with '!', the slot will be given for any +modules but that name. For example, ``slots=!snd-pcsp`` will reserve +the first slot for any modules but snd-pcsp. + + +ALSA PCM devices to OSS devices mapping +======================================= +:: + + /dev/snd/pcmC0D0[c|p] -> /dev/audio0 (/dev/audio) -> minor 4 + /dev/snd/pcmC0D0[c|p] -> /dev/dsp0 (/dev/dsp) -> minor 3 + /dev/snd/pcmC0D1[c|p] -> /dev/adsp0 (/dev/adsp) -> minor 12 + /dev/snd/pcmC1D0[c|p] -> /dev/audio1 -> minor 4+16 = 20 + /dev/snd/pcmC1D0[c|p] -> /dev/dsp1 -> minor 3+16 = 19 + /dev/snd/pcmC1D1[c|p] -> /dev/adsp1 -> minor 12+16 = 28 + /dev/snd/pcmC2D0[c|p] -> /dev/audio2 -> minor 4+32 = 36 + /dev/snd/pcmC2D0[c|p] -> /dev/dsp2 -> minor 3+32 = 39 + /dev/snd/pcmC2D1[c|p] -> /dev/adsp2 -> minor 12+32 = 44 + +The first number from ``/dev/snd/pcmC{X}D{Y}[c|p]`` expression means +sound card number and second means device number. The ALSA devices +have either ``c`` or ``p`` suffix indicating the direction, capture and +playback, respectively. + +Please note that the device mapping above may be varied via the module +options of snd-pcm-oss module. + + +Proc interfaces (/proc/asound) +============================== + +/proc/asound/card#/pcm#[cp]/oss +------------------------------- +erase + erase all additional information about OSS applications + + [] + + name of application with (higher priority) or without path + + number of fragments or zero if auto + + size of fragment in bytes or zero if auto + + optional parameters + + disable + the application tries to open a pcm device for + this channel but does not want to use it. + (Cause a bug or mmap needs) + It's good for Quake etc... + direct + don't use plugins + block + force block mode (rvplayer) + non-block + force non-block mode + whole-frag + write only whole fragments (optimization affecting + playback only) + no-silence + do not fill silence ahead to avoid clicks + buggy-ptr + Returns the whitespace blocks in GETOPTR ioctl + instead of filled blocks + +Example: +:: + + echo "x11amp 128 16384" > /proc/asound/card0/pcm0p/oss + echo "squake 0 0 disable" > /proc/asound/card0/pcm0c/oss + echo "rvplayer 0 0 block" > /proc/asound/card0/pcm0p/oss + + +Early Buffer Allocation +======================= + +Some drivers (e.g. hdsp) require the large contiguous buffers, and +sometimes it's too late to find such spaces when the driver module is +actually loaded due to memory fragmentation. You can pre-allocate the +PCM buffers by loading snd-page-alloc module and write commands to its +proc file in prior, for example, in the early boot stage like +``/etc/init.d/*.local`` scripts. + +Reading the proc file /proc/drivers/snd-page-alloc shows the current +usage of page allocation. In writing, you can send the following +commands to the snd-page-alloc driver: + +* add VENDOR DEVICE MASK SIZE BUFFERS + +VENDOR and DEVICE are PCI vendor and device IDs. They take +integer numbers (0x prefix is needed for the hex). +MASK is the PCI DMA mask. Pass 0 if not restricted. +SIZE is the size of each buffer to allocate. You can pass +k and m suffix for KB and MB. The max number is 16MB. +BUFFERS is the number of buffers to allocate. It must be greater +than 0. The max number is 4. + +* erase + +This will erase the all pre-allocated buffers which are not in +use. + + +Links and Addresses +=================== + +ALSA project homepage + http://www.alsa-project.org +Kernel Bugzilla + http://bugzilla.kernel.org/ +ALSA Developers ML + mailto:alsa-devel@alsa-project.org +alsa-info.sh script + http://www.alsa-project.org/alsa-info.sh diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt deleted file mode 100644 index fc53ccd9a629..000000000000 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ /dev/null @@ -1,2330 +0,0 @@ - - Advanced Linux Sound Architecture - Driver - ========================================== - Configuration guide - - -Kernel Configuration -==================== - -To enable ALSA support you need at least to build the kernel with -primary sound card support (CONFIG_SOUND). Since ALSA can emulate OSS, -you don't have to choose any of the OSS modules. - -Enable "OSS API emulation" (CONFIG_SND_OSSEMUL) and both OSS mixer and -PCM supports if you want to run OSS applications with ALSA. - -If you want to support the WaveTable functionality on cards such as -SB Live! then you need to enable "Sequencer support" -(CONFIG_SND_SEQUENCER). - -To make ALSA debug messages more verbose, enable the "Verbose printk" -and "Debug" options. To check for memory leaks, turn on "Debug memory" -too. "Debug detection" will add checks for the detection of cards. - -Please note that all the ALSA ISA drivers support the Linux isapnp API -(if the card supports ISA PnP). You don't need to configure the cards -using isapnptools. - - -Creating ALSA devices -===================== - -This depends on your distribution, but normally you use the /dev/MAKEDEV -script to create the necessary device nodes. On some systems you use a -script named 'snddevices'. - - -Module parameters -================= - -The user can load modules with options. If the module supports more than -one card and you have more than one card of the same type then you can -specify multiple values for the option separated by commas. - -Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. - - Module snd - ---------- - - The core ALSA module. It is used by all ALSA card drivers. - It takes the following options which have global effects. - - major - major number for sound driver - - Default: 116 - cards_limit - - limiting card index for auto-loading (1-8) - - Default: 1 - - For auto-loading more than one card, specify this - option together with snd-card-X aliases. - slots - Reserve the slot index for the given driver. - This option takes multiple strings. - See "Module Autoloading Support" section for details. - debug - Specifies the debug message level - (0 = disable debug prints, 1 = normal debug messages, - 2 = verbose debug messages) - This option appears only when CONFIG_SND_DEBUG=y. - This option can be dynamically changed via sysfs - /sys/modules/snd/parameters/debug file. - - Module snd-pcm-oss - ------------------ - - The PCM OSS emulation module. - This module takes options which change the mapping of devices. - - dsp_map - PCM device number maps assigned to the 1st OSS device. - - Default: 0 - adsp_map - PCM device number maps assigned to the 2st OSS device. - - Default: 1 - nonblock_open - - Don't block opening busy PCM devices. Default: 1 - - For example, when dsp_map=2, /dev/dsp will be mapped to PCM #2 of - the card #0. Similarly, when adsp_map=0, /dev/adsp will be mapped - to PCM #0 of the card #0. - For changing the second or later card, specify the option with - commas, such like "dsp_map=0,1". - - nonblock_open option is used to change the behavior of the PCM - regarding opening the device. When this option is non-zero, - opening a busy OSS PCM device won't be blocked but return - immediately with EAGAIN (just like O_NONBLOCK flag). - - Module snd-rawmidi - ------------------ - - This module takes options which change the mapping of devices. - similar to those of the snd-pcm-oss module. - - midi_map - MIDI device number maps assigned to the 1st OSS device. - - Default: 0 - amidi_map - MIDI device number maps assigned to the 2st OSS device. - - Default: 1 - - Common parameters for top sound card modules - -------------------------------------------- - - Each of top level sound card module takes the following options. - - index - index (slot #) of sound card - - Values: 0 through 31 or negative - - If nonnegative, assign that index number - - if negative, interpret as a bitmask of permissible - indices; the first free permitted index is assigned - - Default: -1 - id - card ID (identifier or name) - - Can be up to 15 characters long - - Default: the card type - - A directory by this name is created under /proc/asound/ - containing information about the card - - This ID can be used instead of the index number in - identifying the card - enable - enable card - - Default: enabled, for PCI and ISA PnP cards - - Module snd-adlib - ---------------- - - Module for AdLib FM cards. - - port - port # for OPL chip - - This module supports multiple cards. It does not support autoprobe, so - the port must be specified. For actual AdLib FM cards it will be 0x388. - Note that this card does not have PCM support and no mixer; only FM - synthesis. - - Make sure you have "sbiload" from the alsa-tools package available and, - after loading the module, find out the assigned ALSA sequencer port - number through "sbiload -l". Example output: - - Port Client name Port name - 64:0 OPL2 FM synth OPL2 FM Port - - Load the std.sb and drums.sb patches also supplied by sbiload: - - sbiload -p 64:0 std.sb drums.sb - - If you use this driver to drive an OPL3, you can use std.o3 and drums.o3 - instead. To have the card produce sound, use aplaymidi from alsa-utils: - - aplaymidi -p 64:0 foo.mid - - Module snd-ad1816a - ------------------ - - Module for sound cards based on Analog Devices AD1816A/AD1815 ISA chips. - - clockfreq - Clock frequency for AD1816A chip (default = 0, 33000Hz) - - This module supports multiple cards, autoprobe and PnP. - - Module snd-ad1848 - ----------------- - - Module for sound cards based on AD1848/AD1847/CS4248 ISA chips. - - port - port # for AD1848 chip - irq - IRQ # for AD1848 chip - dma1 - DMA # for AD1848 chip (0,1,3) - - This module supports multiple cards. It does not support autoprobe - thus main port must be specified!!! Other ports are optional. - - The power-management is supported. - - Module snd-ad1889 - ----------------- - - Module for Analog Devices AD1889 chips. - - ac97_quirk - AC'97 workaround for strange hardware - See the description of intel8x0 module for details. - - This module supports multiple cards. - - Module snd-ali5451 - ------------------ - - Module for ALi M5451 PCI chip. - - pcm_channels - Number of hardware channels assigned for PCM - spdif - Support SPDIF I/O - - Default: disabled - - This module supports one chip and autoprobe. - - The power-management is supported. - - Module snd-als100 - ----------------- - - Module for sound cards based on Avance Logic ALS100/ALS120 ISA chips. - - This module supports multiple cards, autoprobe and PnP. - - The power-management is supported. - - Module snd-als300 - ----------------- - - Module for Avance Logic ALS300 and ALS300+ - - This module supports multiple cards. - - The power-management is supported. - - Module snd-als4000 - ------------------ - - Module for sound cards based on Avance Logic ALS4000 PCI chip. - - joystick_port - port # for legacy joystick support. - 0 = disabled (default), 1 = auto-detect - - This module supports multiple cards, autoprobe and PnP. - - The power-management is supported. - - Module snd-asihpi - ----------------- - - Module for AudioScience ASI soundcards - - enable_hpi_hwdep - enable HPI hwdep for AudioScience soundcard - - This module supports multiple cards. - The driver requires the firmware loader support on kernel. - - Module snd-atiixp - ----------------- - - Module for ATI IXP 150/200/250/400 AC97 controllers. - - ac97_clock - AC'97 clock (default = 48000) - ac97_quirk - AC'97 workaround for strange hardware - See "AC97 Quirk Option" section below. - ac97_codec - Workaround to specify which AC'97 codec - instead of probing. If this works for you - file a bug with your `lspci -vn` output. - -2 -- Force probing. - -1 -- Default behavior. - 0-2 -- Use the specified codec. - spdif_aclink - S/PDIF transfer over AC-link (default = 1) - - This module supports one card and autoprobe. - - ATI IXP has two different methods to control SPDIF output. One is - over AC-link and another is over the "direct" SPDIF output. The - implementation depends on the motherboard, and you'll need to - choose the correct one via spdif_aclink module option. - - The power-management is supported. - - Module snd-atiixp-modem - ----------------------- - - Module for ATI IXP 150/200/250 AC97 modem controllers. - - This module supports one card and autoprobe. - - Note: The default index value of this module is -2, i.e. the first - slot is excluded. - - The power-management is supported. - - Module snd-au8810, snd-au8820, snd-au8830 - ----------------------------------------- - - Module for Aureal Vortex, Vortex2 and Advantage device. - - pcifix - Control PCI workarounds - 0 = Disable all workarounds - 1 = Force the PCI latency of the Aureal card to 0xff - 2 = Force the Extend PCI#2 Internal Master for Efficient - Handling of Dummy Requests on the VIA KT133 AGP Bridge - 3 = Force both settings - 255 = Autodetect what is required (default) - - This module supports all ADB PCM channels, ac97 mixer, SPDIF, hardware - EQ, mpu401, gameport. A3D and wavetable support are still in development. - Development and reverse engineering work is being coordinated at - http://savannah.nongnu.org/projects/openvortex/ - SPDIF output has a copy of the AC97 codec output, unless you use the - "spdif" pcm device, which allows raw data passthru. - The hardware EQ hardware and SPDIF is only present in the Vortex2 and - Advantage. - - Note: Some ALSA mixer applications don't handle the SPDIF sample rate - control correctly. If you have problems regarding this, try - another ALSA compliant mixer (alsamixer works). - - Module snd-azt1605 - ------------------ - - Module for Aztech Sound Galaxy soundcards based on the Aztech AZT1605 - chipset. - - port - port # for BASE (0x220,0x240,0x260,0x280) - wss_port - port # for WSS (0x530,0x604,0xe80,0xf40) - irq - IRQ # for WSS (7,9,10,11) - dma1 - DMA # for WSS playback (0,1,3) - dma2 - DMA # for WSS capture (0,1), -1 = disabled (default) - mpu_port - port # for MPU-401 UART (0x300,0x330), -1 = disabled (default) - mpu_irq - IRQ # for MPU-401 UART (3,5,7,9), -1 = disabled (default) - fm_port - port # for OPL3 (0x388), -1 = disabled (default) - - This module supports multiple cards. It does not support autoprobe: port, - wss_port, irq and dma1 have to be specified. The other values are - optional. - - "port" needs to match the BASE ADDRESS jumper on the card (0x220 or 0x240) - or the value stored in the card's EEPROM for cards that have an EEPROM and - their "CONFIG MODE" jumper set to "EEPROM SETTING". The other values can - be chosen freely from the options enumerated above. - - If dma2 is specified and different from dma1, the card will operate in - full-duplex mode. When dma1=3, only dma2=0 is valid and the only way to - enable capture since only channels 0 and 1 are available for capture. - - Generic settings are "port=0x220 wss_port=0x530 irq=10 dma1=1 dma2=0 - mpu_port=0x330 mpu_irq=9 fm_port=0x388". - - Whatever IRQ and DMA channels you pick, be sure to reserve them for - legacy ISA in your BIOS. - - Module snd-azt2316 - ------------------ - - Module for Aztech Sound Galaxy soundcards based on the Aztech AZT2316 - chipset. - - port - port # for BASE (0x220,0x240,0x260,0x280) - wss_port - port # for WSS (0x530,0x604,0xe80,0xf40) - irq - IRQ # for WSS (7,9,10,11) - dma1 - DMA # for WSS playback (0,1,3) - dma2 - DMA # for WSS capture (0,1), -1 = disabled (default) - mpu_port - port # for MPU-401 UART (0x300,0x330), -1 = disabled (default) - mpu_irq - IRQ # for MPU-401 UART (5,7,9,10), -1 = disabled (default) - fm_port - port # for OPL3 (0x388), -1 = disabled (default) - - This module supports multiple cards. It does not support autoprobe: port, - wss_port, irq and dma1 have to be specified. The other values are - optional. - - "port" needs to match the BASE ADDRESS jumper on the card (0x220 or 0x240) - or the value stored in the card's EEPROM for cards that have an EEPROM and - their "CONFIG MODE" jumper set to "EEPROM SETTING". The other values can - be chosen freely from the options enumerated above. - - If dma2 is specified and different from dma1, the card will operate in - full-duplex mode. When dma1=3, only dma2=0 is valid and the only way to - enable capture since only channels 0 and 1 are available for capture. - - Generic settings are "port=0x220 wss_port=0x530 irq=10 dma1=1 dma2=0 - mpu_port=0x330 mpu_irq=9 fm_port=0x388". - - Whatever IRQ and DMA channels you pick, be sure to reserve them for - legacy ISA in your BIOS. - - Module snd-aw2 - -------------- - - Module for Audiowerk2 sound card - - This module supports multiple cards. - - Module snd-azt2320 - ------------------ - - Module for sound cards based on Aztech System AZT2320 ISA chip (PnP only). - - This module supports multiple cards, PnP and autoprobe. - - The power-management is supported. - - Module snd-azt3328 - ------------------ - - Module for sound cards based on Aztech AZF3328 PCI chip. - - joystick - Enable joystick (default off) - - This module supports multiple cards. - - Module snd-bt87x - ---------------- - - Module for video cards based on Bt87x chips. - - digital_rate - Override the default digital rate (Hz) - load_all - Load the driver even if the card model isn't known - - This module supports multiple cards. - - Note: The default index value of this module is -2, i.e. the first - slot is excluded. - - Module snd-ca0106 - ----------------- - - Module for Creative Audigy LS and SB Live 24bit - - This module supports multiple cards. - - - Module snd-cmi8330 - ------------------ - - Module for sound cards based on C-Media CMI8330 ISA chips. - - isapnp - ISA PnP detection - 0 = disable, 1 = enable (default) - - with isapnp=0, the following options are available: - - wssport - port # for CMI8330 chip (WSS) - wssirq - IRQ # for CMI8330 chip (WSS) - wssdma - first DMA # for CMI8330 chip (WSS) - sbport - port # for CMI8330 chip (SB16) - sbirq - IRQ # for CMI8330 chip (SB16) - sbdma8 - 8bit DMA # for CMI8330 chip (SB16) - sbdma16 - 16bit DMA # for CMI8330 chip (SB16) - fmport - (optional) OPL3 I/O port - mpuport - (optional) MPU401 I/O port - mpuirq - (optional) MPU401 irq # - - This module supports multiple cards and autoprobe. - - The power-management is supported. - - Module snd-cmipci - ----------------- - - Module for C-Media CMI8338/8738/8768/8770 PCI sound cards. - - mpu_port - port address of MIDI interface (8338 only): - 0x300,0x310,0x320,0x330 = legacy port, - 0 = disable (default) - fm_port - port address of OPL-3 FM synthesizer (8x38 only): - 0x388 = legacy port, - 1 = integrated PCI port (default on 8738), - 0 = disable - soft_ac3 - Software-conversion of raw SPDIF packets (model 033 only) - (default = 1) - joystick_port - Joystick port address (0 = disable, 1 = auto-detect) - - This module supports autoprobe and multiple cards. - - The power-management is supported. - - Module snd-cs4231 - ----------------- - - Module for sound cards based on CS4231 ISA chips. - - port - port # for CS4231 chip - mpu_port - port # for MPU-401 UART (optional), -1 = disable - irq - IRQ # for CS4231 chip - mpu_irq - IRQ # for MPU-401 UART - dma1 - first DMA # for CS4231 chip - dma2 - second DMA # for CS4231 chip - - This module supports multiple cards. This module does not support autoprobe - thus main port must be specified!!! Other ports are optional. - - The power-management is supported. - - Module snd-cs4236 - ----------------- - - Module for sound cards based on CS4232/CS4232A, - CS4235/CS4236/CS4236B/CS4237B/ - CS4238B/CS4239 ISA chips. - - isapnp - ISA PnP detection - 0 = disable, 1 = enable (default) - - with isapnp=0, the following options are available: - - port - port # for CS4236 chip (PnP setup - 0x534) - cport - control port # for CS4236 chip (PnP setup - 0x120,0x210,0xf00) - mpu_port - port # for MPU-401 UART (PnP setup - 0x300), -1 = disable - fm_port - FM port # for CS4236 chip (PnP setup - 0x388), -1 = disable - irq - IRQ # for CS4236 chip (5,7,9,11,12,15) - mpu_irq - IRQ # for MPU-401 UART (9,11,12,15) - dma1 - first DMA # for CS4236 chip (0,1,3) - dma2 - second DMA # for CS4236 chip (0,1,3), -1 = disable - - This module supports multiple cards. This module does not support autoprobe - (if ISA PnP is not used) thus main port and control port must be - specified!!! Other ports are optional. - - The power-management is supported. - - This module is aliased as snd-cs4232 since it provides the old - snd-cs4232 functionality, too. - - Module snd-cs4281 - ----------------- - - Module for Cirrus Logic CS4281 soundchip. - - dual_codec - Secondary codec ID (0 = disable, default) - - This module supports multiple cards. - - The power-management is supported. - - Module snd-cs46xx - ----------------- - - Module for PCI sound cards based on CS4610/CS4612/CS4614/CS4615/CS4622/ - CS4624/CS4630/CS4280 PCI chips. - - external_amp - Force to enable external amplifier. - thinkpad - Force to enable Thinkpad's CLKRUN control. - mmap_valid - Support OSS mmap mode (default = 0). - - This module supports multiple cards and autoprobe. - Usually external amp and CLKRUN controls are detected automatically - from PCI sub vendor/device ids. If they don't work, give the options - above explicitly. - - The power-management is supported. - - Module snd-cs5530 - _________________ - - Module for Cyrix/NatSemi Geode 5530 chip. - - Module snd-cs5535audio - ---------------------- - - Module for multifunction CS5535 companion PCI device - - The power-management is supported. - - Module snd-ctxfi - ---------------- - - Module for Creative Sound Blaster X-Fi boards (20k1 / 20k2 chips) - * Creative Sound Blaster X-Fi Titanium Fatal1ty Champion Series - * Creative Sound Blaster X-Fi Titanium Fatal1ty Professional Series - * Creative Sound Blaster X-Fi Titanium Professional Audio - * Creative Sound Blaster X-Fi Titanium - * Creative Sound Blaster X-Fi Elite Pro - * Creative Sound Blaster X-Fi Platinum - * Creative Sound Blaster X-Fi Fatal1ty - * Creative Sound Blaster X-Fi XtremeGamer - * Creative Sound Blaster X-Fi XtremeMusic - - reference_rate - reference sample rate, 44100 or 48000 (default) - multiple - multiple to ref. sample rate, 1 or 2 (default) - subsystem - override the PCI SSID for probing; the value - consists of SSVID << 16 | SSDID. The default is - zero, which means no override. - - This module supports multiple cards. - - Module snd-darla20 - ------------------ - - Module for Echoaudio Darla20 - - This module supports multiple cards. - The driver requires the firmware loader support on kernel. - - Module snd-darla24 - ------------------ - - Module for Echoaudio Darla24 - - This module supports multiple cards. - The driver requires the firmware loader support on kernel. - - Module snd-dt019x - ----------------- - - Module for Diamond Technologies DT-019X / Avance Logic ALS-007 (PnP - only) - - This module supports multiple cards. This module is enabled only with - ISA PnP support. - - The power-management is supported. - - Module snd-dummy - ---------------- - - Module for the dummy sound card. This "card" doesn't do any output - or input, but you may use this module for any application which - requires a sound card (like RealPlayer). - - pcm_devs - Number of PCM devices assigned to each card - (default = 1, up to 4) - pcm_substreams - Number of PCM substreams assigned to each PCM - (default = 8, up to 128) - hrtimer - Use hrtimer (=1, default) or system timer (=0) - fake_buffer - Fake buffer allocations (default = 1) - - When multiple PCM devices are created, snd-dummy gives different - behavior to each PCM device: - 0 = interleaved with mmap support - 1 = non-interleaved with mmap support - 2 = interleaved without mmap - 3 = non-interleaved without mmap - - As default, snd-dummy drivers doesn't allocate the real buffers - but either ignores read/write or mmap a single dummy page to all - buffer pages, in order to save the resources. If your apps need - the read/ written buffer data to be consistent, pass fake_buffer=0 - option. - - The power-management is supported. - - Module snd-echo3g - ----------------- - - Module for Echoaudio 3G cards (Gina3G/Layla3G) - - This module supports multiple cards. - The driver requires the firmware loader support on kernel. - - Module snd-emu10k1 - ------------------ - - Module for EMU10K1/EMU10k2 based PCI sound cards. - * Sound Blaster Live! - * Sound Blaster PCI 512 - * Emu APS (partially supported) - * Sound Blaster Audigy - - extin - bitmap of available external inputs for FX8010 (see bellow) - extout - bitmap of available external outputs for FX8010 (see bellow) - seq_ports - allocated sequencer ports (4 by default) - max_synth_voices - limit of voices used for wavetable (64 by default) - max_buffer_size - specifies the maximum size of wavetable/pcm buffers - given in MB unit. Default value is 128. - enable_ir - enable IR - - This module supports multiple cards and autoprobe. - - Input & Output configurations [extin/extout] - * Creative Card wo/Digital out [0x0003/0x1f03] - * Creative Card w/Digital out [0x0003/0x1f0f] - * Creative Card w/Digital CD in [0x000f/0x1f0f] - * Creative Card wo/Digital out + LiveDrive [0x3fc3/0x1fc3] - * Creative Card w/Digital out + LiveDrive [0x3fc3/0x1fcf] - * Creative Card w/Digital CD in + LiveDrive [0x3fcf/0x1fcf] - * Creative Card wo/Digital out + Digital I/O 2 [0x0fc3/0x1f0f] - * Creative Card w/Digital out + Digital I/O 2 [0x0fc3/0x1f0f] - * Creative Card w/Digital CD in + Digital I/O 2 [0x0fcf/0x1f0f] - * Creative Card 5.1/w Digital out + LiveDrive [0x3fc3/0x1fff] - * Creative Card 5.1 (c) 2003 [0x3fc3/0x7cff] - * Creative Card all ins and outs [0x3fff/0x7fff] - - The power-management is supported. - - Module snd-emu10k1x - ------------------- - - Module for Creative Emu10k1X (SB Live Dell OEM version) - - This module supports multiple cards. - - Module snd-ens1370 - ------------------ - - Module for Ensoniq AudioPCI ES1370 PCI sound cards. - * SoundBlaster PCI 64 - * SoundBlaster PCI 128 - - joystick - Enable joystick (default off) - - This module supports multiple cards and autoprobe. - - The power-management is supported. - - Module snd-ens1371 - ------------------ - - Module for Ensoniq AudioPCI ES1371 PCI sound cards. - * SoundBlaster PCI 64 - * SoundBlaster PCI 128 - * SoundBlaster Vibra PCI - - joystick_port - port # for joystick (0x200,0x208,0x210,0x218), - 0 = disable (default), 1 = auto-detect - - This module supports multiple cards and autoprobe. - - The power-management is supported. - - Module snd-es1688 - ----------------- - - Module for ESS AudioDrive ES-1688 and ES-688 sound cards. - - isapnp - ISA PnP detection - 0 = disable, 1 = enable (default) - mpu_port - port # for MPU-401 port (0x300,0x310,0x320,0x330), -1 = disable (default) - mpu_irq - IRQ # for MPU-401 port (5,7,9,10) - fm_port - port # for OPL3 (option; share the same port as default) - - with isapnp=0, the following additional options are available: - port - port # for ES-1688 chip (0x220,0x240,0x260) - irq - IRQ # for ES-1688 chip (5,7,9,10) - dma8 - DMA # for ES-1688 chip (0,1,3) - - This module supports multiple cards and autoprobe (without MPU-401 port) - and PnP with the ES968 chip. - - Module snd-es18xx - ----------------- - - Module for ESS AudioDrive ES-18xx sound cards. - - isapnp - ISA PnP detection - 0 = disable, 1 = enable (default) - - with isapnp=0, the following options are available: - - port - port # for ES-18xx chip (0x220,0x240,0x260) - mpu_port - port # for MPU-401 port (0x300,0x310,0x320,0x330), -1 = disable (default) - fm_port - port # for FM (optional, not used) - irq - IRQ # for ES-18xx chip (5,7,9,10) - dma1 - first DMA # for ES-18xx chip (0,1,3) - dma2 - first DMA # for ES-18xx chip (0,1,3) - - This module supports multiple cards, ISA PnP and autoprobe (without MPU-401 - port if native ISA PnP routines are not used). - When dma2 is equal with dma1, the driver works as half-duplex. - - The power-management is supported. - - Module snd-es1938 - ----------------- - - Module for sound cards based on ESS Solo-1 (ES1938,ES1946) chips. - - This module supports multiple cards and autoprobe. - - The power-management is supported. - - Module snd-es1968 - ----------------- - - Module for sound cards based on ESS Maestro-1/2/2E (ES1968/ES1978) chips. - - total_bufsize - total buffer size in kB (1-4096kB) - pcm_substreams_p - playback channels (1-8, default=2) - pcm_substreams_c - capture channels (1-8, default=0) - clock - clock (0 = auto-detection) - use_pm - support the power-management (0 = off, 1 = on, - 2 = auto (default)) - enable_mpu - enable MPU401 (0 = off, 1 = on, 2 = auto (default)) - joystick - enable joystick (default off) - - This module supports multiple cards and autoprobe. - - The power-management is supported. - - Module snd-fm801 - ---------------- - - Module for ForteMedia FM801 based PCI sound cards. - - tea575x_tuner - Enable TEA575x tuner - - 1 = MediaForte 256-PCS - - 2 = MediaForte 256-PCPR - - 3 = MediaForte 64-PCR - - High 16-bits are video (radio) device number + 1 - - example: 0x10002 (MediaForte 256-PCPR, device 1) - - This module supports multiple cards and autoprobe. - - The power-management is supported. - - Module snd-gina20 - ----------------- - - Module for Echoaudio Gina20 - - This module supports multiple cards. - The driver requires the firmware loader support on kernel. - - Module snd-gina24 - ----------------- - - Module for Echoaudio Gina24 - - This module supports multiple cards. - The driver requires the firmware loader support on kernel. - - Module snd-gusclassic - --------------------- - - Module for Gravis UltraSound Classic sound card. - - port - port # for GF1 chip (0x220,0x230,0x240,0x250,0x260) - irq - IRQ # for GF1 chip (3,5,9,11,12,15) - dma1 - DMA # for GF1 chip (1,3,5,6,7) - dma2 - DMA # for GF1 chip (1,3,5,6,7,-1=disable) - joystick_dac - 0 to 31, (0.59V-4.52V or 0.389V-2.98V) - voices - GF1 voices limit (14-32) - pcm_voices - reserved PCM voices - - This module supports multiple cards and autoprobe. - - Module snd-gusextreme - --------------------- - - Module for Gravis UltraSound Extreme (Synergy ViperMax) sound card. - - port - port # for ES-1688 chip (0x220,0x230,0x240,0x250,0x260) - gf1_port - port # for GF1 chip (0x210,0x220,0x230,0x240,0x250,0x260,0x270) - mpu_port - port # for MPU-401 port (0x300,0x310,0x320,0x330), -1 = disable - irq - IRQ # for ES-1688 chip (5,7,9,10) - gf1_irq - IRQ # for GF1 chip (3,5,9,11,12,15) - mpu_irq - IRQ # for MPU-401 port (5,7,9,10) - dma8 - DMA # for ES-1688 chip (0,1,3) - dma1 - DMA # for GF1 chip (1,3,5,6,7) - joystick_dac - 0 to 31, (0.59V-4.52V or 0.389V-2.98V) - voices - GF1 voices limit (14-32) - pcm_voices - reserved PCM voices - - This module supports multiple cards and autoprobe (without MPU-401 port). - - Module snd-gusmax - ----------------- - - Module for Gravis UltraSound MAX sound card. - - port - port # for GF1 chip (0x220,0x230,0x240,0x250,0x260) - irq - IRQ # for GF1 chip (3,5,9,11,12,15) - dma1 - DMA # for GF1 chip (1,3,5,6,7) - dma2 - DMA # for GF1 chip (1,3,5,6,7,-1=disable) - joystick_dac - 0 to 31, (0.59V-4.52V or 0.389V-2.98V) - voices - GF1 voices limit (14-32) - pcm_voices - reserved PCM voices - - This module supports multiple cards and autoprobe. - - Module snd-hda-intel - -------------------- - - Module for Intel HD Audio (ICH6, ICH6M, ESB2, ICH7, ICH8, ICH9, ICH10, - PCH, SCH), - ATI SB450, SB600, R600, RS600, RS690, RS780, RV610, RV620, - RV630, RV635, RV670, RV770, - VIA VT8251/VT8237A, - SIS966, ULI M5461 - - [Multiple options for each card instance] - model - force the model name - position_fix - Fix DMA pointer - -1 = system default: choose appropriate one per controller - hardware - 0 = auto: falls back to LPIB when POSBUF doesn't work - 1 = use LPIB - 2 = POSBUF: use position buffer - 3 = VIACOMBO: VIA-specific workaround for capture - 4 = COMBO: use LPIB for playback, auto for capture stream - probe_mask - Bitmask to probe codecs (default = -1, meaning all slots) - When the bit 8 (0x100) is set, the lower 8 bits are used - as the "fixed" codec slots; i.e. the driver probes the - slots regardless what hardware reports back - probe_only - Only probing and no codec initialization (default=off); - Useful to check the initial codec status for debugging - bdl_pos_adj - Specifies the DMA IRQ timing delay in samples. - Passing -1 will make the driver to choose the appropriate - value based on the controller chip. - patch - Specifies the early "patch" files to modify the HD-audio - setup before initializing the codecs. This option is - available only when CONFIG_SND_HDA_PATCH_LOADER=y is set. - See HD-Audio.txt for details. - beep_mode - Selects the beep registration mode (0=off, 1=on); default - value is set via CONFIG_SND_HDA_INPUT_BEEP_MODE kconfig. - - [Single (global) options] - single_cmd - Use single immediate commands to communicate with - codecs (for debugging only) - enable_msi - Enable Message Signaled Interrupt (MSI) (default = off) - power_save - Automatic power-saving timeout (in second, 0 = - disable) - power_save_controller - Reset HD-audio controller in power-saving mode - (default = on) - align_buffer_size - Force rounding of buffer/period sizes to multiples - of 128 bytes. This is more efficient in terms of memory - access but isn't required by the HDA spec and prevents - users from specifying exact period/buffer sizes. - (default = on) - snoop - Enable/disable snooping (default = on) - - This module supports multiple cards and autoprobe. - - See Documentation/sound/alsa/HD-Audio.txt for more details about - HD-audio driver. - - Each codec may have a model table for different configurations. - If your machine isn't listed there, the default (usually minimal) - configuration is set up. You can pass "model=" option to - specify a certain model in such a case. There are different - models depending on the codec chip. The list of available models - is found in HD-Audio-Models.txt - - The model name "generic" is treated as a special case. When this - model is given, the driver uses the generic codec parser without - "codec-patch". It's sometimes good for testing and debugging. - - If the default configuration doesn't work and one of the above - matches with your device, report it together with alsa-info.sh - output (with --no-upload option) to kernel bugzilla or alsa-devel - ML (see the section "Links and Addresses"). - - power_save and power_save_controller options are for power-saving - mode. See powersave.txt for details. - - Note 2: If you get click noises on output, try the module option - position_fix=1 or 2. position_fix=1 will use the SD_LPIB - register value without FIFO size correction as the current - DMA pointer. position_fix=2 will make the driver to use - the position buffer instead of reading SD_LPIB register. - (Usually SD_LPIB register is more accurate than the - position buffer.) - - position_fix=3 is specific to VIA devices. The position - of the capture stream is checked from both LPIB and POSBUF - values. position_fix=4 is a combination mode, using LPIB - for playback and POSBUF for capture. - - NB: If you get many "azx_get_response timeout" messages at - loading, it's likely a problem of interrupts (e.g. ACPI irq - routing). Try to boot with options like "pci=noacpi". Also, you - can try "single_cmd=1" module option. This will switch the - communication method between HDA controller and codecs to the - single immediate commands instead of CORB/RIRB. Basically, the - single command mode is provided only for BIOS, and you won't get - unsolicited events, too. But, at least, this works independently - from the irq. Remember this is a last resort, and should be - avoided as much as possible... - - MORE NOTES ON "azx_get_response timeout" PROBLEMS: - On some hardware, you may need to add a proper probe_mask option - to avoid the "azx_get_response timeout" problem above, instead. - This occurs when the access to non-existing or non-working codec slot - (likely a modem one) causes a stall of the communication via HD-audio - bus. You can see which codec slots are probed by enabling - CONFIG_SND_DEBUG_VERBOSE, or simply from the file name of the codec - proc files. Then limit the slots to probe by probe_mask option. - For example, probe_mask=1 means to probe only the first slot, and - probe_mask=4 means only the third slot. - - The power-management is supported. - - Module snd-hdsp - --------------- - - Module for RME Hammerfall DSP audio interface(s) - - This module supports multiple cards. - - Note: The firmware data can be automatically loaded via hotplug - when CONFIG_FW_LOADER is set. Otherwise, you need to load - the firmware via hdsploader utility included in alsa-tools - package. - The firmware data is found in alsa-firmware package. - - Note: snd-page-alloc module does the job which snd-hammerfall-mem - module did formerly. It will allocate the buffers in advance - when any HDSP cards are found. To make the buffer - allocation sure, load snd-page-alloc module in the early - stage of boot sequence. See "Early Buffer Allocation" - section. - - Module snd-hdspm - ---------------- - - Module for RME HDSP MADI board. - - precise_ptr - Enable precise pointer, or disable. - line_outs_monitor - Send playback streams to analog outs by default. - enable_monitor - Enable Analog Out on Channel 63/64 by default. - - See hdspm.txt for details. - - Module snd-ice1712 - ------------------ - - Module for Envy24 (ICE1712) based PCI sound cards. - * MidiMan M Audio Delta 1010 - * MidiMan M Audio Delta 1010LT - * MidiMan M Audio Delta DiO 2496 - * MidiMan M Audio Delta 66 - * MidiMan M Audio Delta 44 - * MidiMan M Audio Delta 410 - * MidiMan M Audio Audiophile 2496 - * TerraTec EWS 88MT - * TerraTec EWS 88D - * TerraTec EWX 24/96 - * TerraTec DMX 6Fire - * TerraTec Phase 88 - * Hoontech SoundTrack DSP 24 - * Hoontech SoundTrack DSP 24 Value - * Hoontech SoundTrack DSP 24 Media 7.1 - * Event Electronics, EZ8 - * Digigram VX442 - * Lionstracs, Mediastaton - * Terrasoniq TS 88 - - model - Use the given board model, one of the following: - delta1010, dio2496, delta66, delta44, audiophile, delta410, - delta1010lt, vx442, ewx2496, ews88mt, ews88mt_new, ews88d, - dmx6fire, dsp24, dsp24_value, dsp24_71, ez8, - phase88, mediastation - omni - Omni I/O support for MidiMan M-Audio Delta44/66 - cs8427_timeout - reset timeout for the CS8427 chip (S/PDIF transceiver) - in msec resolution, default value is 500 (0.5 sec) - - This module supports multiple cards and autoprobe. Note: The consumer part - is not used with all Envy24 based cards (for example in the MidiMan Delta - serie). - - Note: The supported board is detected by reading EEPROM or PCI - SSID (if EEPROM isn't available). You can override the - model by passing "model" module option in case that the - driver isn't configured properly or you want to try another - type for testing. - - Module snd-ice1724 - ------------------ - - Module for Envy24HT (VT/ICE1724), Envy24PT (VT1720) based PCI sound cards. - * MidiMan M Audio Revolution 5.1 - * MidiMan M Audio Revolution 7.1 - * MidiMan M Audio Audiophile 192 - * AMP Ltd AUDIO2000 - * TerraTec Aureon 5.1 Sky - * TerraTec Aureon 7.1 Space - * TerraTec Aureon 7.1 Universe - * TerraTec Phase 22 - * TerraTec Phase 28 - * AudioTrak Prodigy 7.1 - * AudioTrak Prodigy 7.1 LT - * AudioTrak Prodigy 7.1 XT - * AudioTrak Prodigy 7.1 HIFI - * AudioTrak Prodigy 7.1 HD2 - * AudioTrak Prodigy 192 - * Pontis MS300 - * Albatron K8X800 Pro II - * Chaintech ZNF3-150 - * Chaintech ZNF3-250 - * Chaintech 9CJS - * Chaintech AV-710 - * Shuttle SN25P - * Onkyo SE-90PCI - * Onkyo SE-200PCI - * ESI Juli@ - * ESI Maya44 - * Hercules Fortissimo IV - * EGO-SYS WaveTerminal 192M - - model - Use the given board model, one of the following: - revo51, revo71, amp2000, prodigy71, prodigy71lt, - prodigy71xt, prodigy71hifi, prodigyhd2, prodigy192, - juli, aureon51, aureon71, universe, ap192, k8x800, - phase22, phase28, ms300, av710, se200pci, se90pci, - fortissimo4, sn25p, WT192M, maya44 - - This module supports multiple cards and autoprobe. - - Note: The supported board is detected by reading EEPROM or PCI - SSID (if EEPROM isn't available). You can override the - model by passing "model" module option in case that the - driver isn't configured properly or you want to try another - type for testing. - - Module snd-indigo - ----------------- - - Module for Echoaudio Indigo - - This module supports multiple cards. - The driver requires the firmware loader support on kernel. - - Module snd-indigodj - ------------------- - - Module for Echoaudio Indigo DJ - - This module supports multiple cards. - The driver requires the firmware loader support on kernel. - - Module snd-indigoio - ------------------- - - Module for Echoaudio Indigo IO - - This module supports multiple cards. - The driver requires the firmware loader support on kernel. - - Module snd-intel8x0 - ------------------- - - Module for AC'97 motherboards from Intel and compatibles. - * Intel i810/810E, i815, i820, i830, i84x, MX440 - ICH5, ICH6, ICH7, 6300ESB, ESB2 - * SiS 7012 (SiS 735) - * NVidia NForce, NForce2, NForce3, MCP04, CK804 - CK8, CK8S, MCP501 - * AMD AMD768, AMD8111 - * ALi m5455 - - ac97_clock - AC'97 codec clock base (0 = auto-detect) - ac97_quirk - AC'97 workaround for strange hardware - See "AC97 Quirk Option" section below. - buggy_irq - Enable workaround for buggy interrupts on some - motherboards (default yes on nForce chips, - otherwise off) - buggy_semaphore - Enable workaround for hardware with buggy - semaphores (e.g. on some ASUS laptops) - (default off) - spdif_aclink - Use S/PDIF over AC-link instead of direct connection - from the controller chip - (0 = off, 1 = on, -1 = default) - - This module supports one chip and autoprobe. - - Note: the latest driver supports auto-detection of chip clock. - if you still encounter too fast playback, specify the clock - explicitly via the module option "ac97_clock=41194". - - Joystick/MIDI ports are not supported by this driver. If your - motherboard has these devices, use the ns558 or snd-mpu401 - modules, respectively. - - The power-management is supported. - - Module snd-intel8x0m - -------------------- - - Module for Intel ICH (i8x0) chipset MC97 modems. - * Intel i810/810E, i815, i820, i830, i84x, MX440 - ICH5, ICH6, ICH7 - * SiS 7013 (SiS 735) - * NVidia NForce, NForce2, NForce2s, NForce3 - * AMD AMD8111 - * ALi m5455 - - ac97_clock - AC'97 codec clock base (0 = auto-detect) - - This module supports one card and autoprobe. - - Note: The default index value of this module is -2, i.e. the first - slot is excluded. - - The power-management is supported. - - Module snd-interwave - -------------------- - - Module for Gravis UltraSound PnP, Dynasonic 3-D/Pro, STB Sound Rage 32 - and other sound cards based on AMD InterWave (tm) chip. - - joystick_dac - 0 to 31, (0.59V-4.52V or 0.389V-2.98V) - midi - 1 = MIDI UART enable, 0 = MIDI UART disable (default) - pcm_voices - reserved PCM voices for the synthesizer (default 2) - effect - 1 = InterWave effects enable (default 0); - requires 8 voices - isapnp - ISA PnP detection - 0 = disable, 1 = enable (default) - - with isapnp=0, the following options are available: - - port - port # for InterWave chip (0x210,0x220,0x230,0x240,0x250,0x260) - irq - IRQ # for InterWave chip (3,5,9,11,12,15) - dma1 - DMA # for InterWave chip (0,1,3,5,6,7) - dma2 - DMA # for InterWave chip (0,1,3,5,6,7,-1=disable) - - This module supports multiple cards, autoprobe and ISA PnP. - - Module snd-interwave-stb - ------------------------ - - Module for UltraSound 32-Pro (sound card from STB used by Compaq) - and other sound cards based on AMD InterWave (tm) chip with TEA6330T - circuit for extended control of bass, treble and master volume. - - joystick_dac - 0 to 31, (0.59V-4.52V or 0.389V-2.98V) - midi - 1 = MIDI UART enable, 0 = MIDI UART disable (default) - pcm_voices - reserved PCM voices for the synthesizer (default 2) - effect - 1 = InterWave effects enable (default 0); - requires 8 voices - isapnp - ISA PnP detection - 0 = disable, 1 = enable (default) - - with isapnp=0, the following options are available: - - port - port # for InterWave chip (0x210,0x220,0x230,0x240,0x250,0x260) - port_tc - tone control (i2c bus) port # for TEA6330T chip (0x350,0x360,0x370,0x380) - irq - IRQ # for InterWave chip (3,5,9,11,12,15) - dma1 - DMA # for InterWave chip (0,1,3,5,6,7) - dma2 - DMA # for InterWave chip (0,1,3,5,6,7,-1=disable) - - This module supports multiple cards, autoprobe and ISA PnP. - - Module snd-jazz16 - ------------------- - - Module for Media Vision Jazz16 chipset. The chipset consists of 3 chips: - MVD1216 + MVA416 + MVA514. - - port - port # for SB DSP chip (0x210,0x220,0x230,0x240,0x250,0x260) - irq - IRQ # for SB DSP chip (3,5,7,9,10,15) - dma8 - DMA # for SB DSP chip (1,3) - dma16 - DMA # for SB DSP chip (5,7) - mpu_port - MPU-401 port # (0x300,0x310,0x320,0x330) - mpu_irq - MPU-401 irq # (2,3,5,7) - - This module supports multiple cards. - - Module snd-korg1212 - ------------------- - - Module for Korg 1212 IO PCI card - - This module supports multiple cards. - - Module snd-layla20 - ------------------ - - Module for Echoaudio Layla20 - - This module supports multiple cards. - The driver requires the firmware loader support on kernel. - - Module snd-layla24 - ------------------ - - Module for Echoaudio Layla24 - - This module supports multiple cards. - The driver requires the firmware loader support on kernel. - - Module snd-lola - --------------- - - Module for Digigram Lola PCI-e boards - - This module supports multiple cards. - - Module snd-lx6464es - ------------------- - - Module for Digigram LX6464ES boards - - This module supports multiple cards. - - Module snd-maestro3 - ------------------- - - Module for Allegro/Maestro3 chips - - external_amp - enable external amp (enabled by default) - amp_gpio - GPIO pin number for external amp (0-15) or - -1 for default pin (8 for allegro, 1 for - others) - - This module supports autoprobe and multiple chips. - - Note: the binding of amplifier is dependent on hardware. - If there is no sound even though all channels are unmuted, try to - specify other gpio connection via amp_gpio option. - For example, a Panasonic notebook might need "amp_gpio=0x0d" - option. - - The power-management is supported. - - Module snd-mia - --------------- - - Module for Echoaudio Mia - - This module supports multiple cards. - The driver requires the firmware loader support on kernel. - - Module snd-miro - --------------- - - Module for Miro soundcards: miroSOUND PCM 1 pro, - miroSOUND PCM 12, - miroSOUND PCM 20 Radio. - - port - Port # (0x530,0x604,0xe80,0xf40) - irq - IRQ # (5,7,9,10,11) - dma1 - 1st dma # (0,1,3) - dma2 - 2nd dma # (0,1) - mpu_port - MPU-401 port # (0x300,0x310,0x320,0x330) - mpu_irq - MPU-401 irq # (5,7,9,10) - fm_port - FM Port # (0x388) - wss - enable WSS mode - ide - enable onboard ide support - - Module snd-mixart - ----------------- - - Module for Digigram miXart8 sound cards. - - This module supports multiple cards. - Note: One miXart8 board will be represented as 4 alsa cards. - See MIXART.txt for details. - - When the driver is compiled as a module and the hotplug firmware - is supported, the firmware data is loaded via hotplug automatically. - Install the necessary firmware files in alsa-firmware package. - When no hotplug fw loader is available, you need to load the - firmware via mixartloader utility in alsa-tools package. - - Module snd-mona - --------------- - - Module for Echoaudio Mona - - This module supports multiple cards. - The driver requires the firmware loader support on kernel. - - Module snd-mpu401 - ----------------- - - Module for MPU-401 UART devices. - - port - port number or -1 (disable) - irq - IRQ number or -1 (disable) - pnp - PnP detection - 0 = disable, 1 = enable (default) - - This module supports multiple devices and PnP. - - Module snd-msnd-classic - ----------------------- - - Module for Turtle Beach MultiSound Classic, Tahiti or Monterey - soundcards. - - io - Port # for msnd-classic card - irq - IRQ # for msnd-classic card - mem - Memory address (0xb0000, 0xc8000, 0xd0000, 0xd8000, - 0xe0000 or 0xe8000) - write_ndelay - enable write ndelay (default = 1) - calibrate_signal - calibrate signal (default = 0) - isapnp - ISA PnP detection - 0 = disable, 1 = enable (default) - digital - Digital daughterboard present (default = 0) - cfg - Config port (0x250, 0x260 or 0x270) default = PnP - reset - Reset all devices - mpu_io - MPU401 I/O port - mpu_irq - MPU401 irq# - ide_io0 - IDE port #0 - ide_io1 - IDE port #1 - ide_irq - IDE irq# - joystick_io - Joystick I/O port - - The driver requires firmware files "turtlebeach/msndinit.bin" and - "turtlebeach/msndperm.bin" in the proper firmware directory. - - See Documentation/sound/oss/MultiSound for important information - about this driver. Note that it has been discontinued, but the - Voyetra Turtle Beach knowledge base entry for it is still available - at - http://www.turtlebeach.com - - Module snd-msnd-pinnacle - ------------------------ - - Module for Turtle Beach MultiSound Pinnacle/Fiji soundcards. - - io - Port # for pinnacle/fiji card - irq - IRQ # for pinnalce/fiji card - mem - Memory address (0xb0000, 0xc8000, 0xd0000, 0xd8000, - 0xe0000 or 0xe8000) - write_ndelay - enable write ndelay (default = 1) - calibrate_signal - calibrate signal (default = 0) - isapnp - ISA PnP detection - 0 = disable, 1 = enable (default) - - The driver requires firmware files "turtlebeach/pndspini.bin" and - "turtlebeach/pndsperm.bin" in the proper firmware directory. - - Module snd-mtpav - ---------------- - - Module for MOTU MidiTimePiece AV multiport MIDI (on the parallel - port). - - port - I/O port # for MTPAV (0x378,0x278, default=0x378) - irq - IRQ # for MTPAV (7,5, default=7) - hwports - number of supported hardware ports, default=8. - - Module supports only 1 card. This module has no enable option. - - Module snd-mts64 - ---------------- - - Module for Ego Systems (ESI) Miditerminal 4140 - - This module supports multiple devices. - Requires parport (CONFIG_PARPORT). - - Module snd-nm256 - ---------------- - - Module for NeoMagic NM256AV/ZX chips - - playback_bufsize - max playback frame size in kB (4-128kB) - capture_bufsize - max capture frame size in kB (4-128kB) - force_ac97 - 0 or 1 (disabled by default) - buffer_top - specify buffer top address - use_cache - 0 or 1 (disabled by default) - vaio_hack - alias buffer_top=0x25a800 - reset_workaround - enable AC97 RESET workaround for some laptops - reset_workaround2 - enable extended AC97 RESET workaround for some - other laptops - - This module supports one chip and autoprobe. - - The power-management is supported. - - Note: on some notebooks the buffer address cannot be detected - automatically, or causes hang-up during initialization. - In such a case, specify the buffer top address explicitly via - the buffer_top option. - For example, - Sony F250: buffer_top=0x25a800 - Sony F270: buffer_top=0x272800 - The driver supports only ac97 codec. It's possible to force - to initialize/use ac97 although it's not detected. In such a - case, use force_ac97=1 option - but *NO* guarantee whether it - works! - - Note: The NM256 chip can be linked internally with non-AC97 - codecs. This driver supports only the AC97 codec, and won't work - with machines with other (most likely CS423x or OPL3SAx) chips, - even though the device is detected in lspci. In such a case, try - other drivers, e.g. snd-cs4232 or snd-opl3sa2. Some has ISA-PnP - but some doesn't have ISA PnP. You'll need to specify isapnp=0 - and proper hardware parameters in the case without ISA PnP. - - Note: some laptops need a workaround for AC97 RESET. For the - known hardware like Dell Latitude LS and Sony PCG-F305, this - workaround is enabled automatically. For other laptops with a - hard freeze, you can try reset_workaround=1 option. - - Note: Dell Latitude CSx laptops have another problem regarding - AC97 RESET. On these laptops, reset_workaround2 option is - turned on as default. This option is worth to try if the - previous reset_workaround option doesn't help. - - Note: This driver is really crappy. It's a porting from the - OSS driver, which is a result of black-magic reverse engineering. - The detection of codec will fail if the driver is loaded *after* - X-server as described above. You might be able to force to load - the module, but it may result in hang-up. Hence, make sure that - you load this module *before* X if you encounter this kind of - problem. - - Module snd-opl3sa2 - ------------------ - - Module for Yamaha OPL3-SA2/SA3 sound cards. - - isapnp - ISA PnP detection - 0 = disable, 1 = enable (default) - - with isapnp=0, the following options are available: - - port - control port # for OPL3-SA chip (0x370) - sb_port - SB port # for OPL3-SA chip (0x220,0x240) - wss_port - WSS port # for OPL3-SA chip (0x530,0xe80,0xf40,0x604) - midi_port - port # for MPU-401 UART (0x300,0x330), -1 = disable - fm_port - FM port # for OPL3-SA chip (0x388), -1 = disable - irq - IRQ # for OPL3-SA chip (5,7,9,10) - dma1 - first DMA # for Yamaha OPL3-SA chip (0,1,3) - dma2 - second DMA # for Yamaha OPL3-SA chip (0,1,3), -1 = disable - - This module supports multiple cards and ISA PnP. It does not support - autoprobe (if ISA PnP is not used) thus all ports must be specified!!! - - The power-management is supported. - - Module snd-opti92x-ad1848 - ------------------------- - - Module for sound cards based on OPTi 82c92x and Analog Devices AD1848 chips. - Module works with OAK Mozart cards as well. - - isapnp - ISA PnP detection - 0 = disable, 1 = enable (default) - - with isapnp=0, the following options are available: - - port - port # for WSS chip (0x530,0xe80,0xf40,0x604) - mpu_port - port # for MPU-401 UART (0x300,0x310,0x320,0x330) - fm_port - port # for OPL3 device (0x388) - irq - IRQ # for WSS chip (5,7,9,10,11) - mpu_irq - IRQ # for MPU-401 UART (5,7,9,10) - dma1 - first DMA # for WSS chip (0,1,3) - - This module supports only one card, autoprobe and PnP. - - Module snd-opti92x-cs4231 - ------------------------- - - Module for sound cards based on OPTi 82c92x and Crystal CS4231 chips. - - isapnp - ISA PnP detection - 0 = disable, 1 = enable (default) - - with isapnp=0, the following options are available: - - port - port # for WSS chip (0x530,0xe80,0xf40,0x604) - mpu_port - port # for MPU-401 UART (0x300,0x310,0x320,0x330) - fm_port - port # for OPL3 device (0x388) - irq - IRQ # for WSS chip (5,7,9,10,11) - mpu_irq - IRQ # for MPU-401 UART (5,7,9,10) - dma1 - first DMA # for WSS chip (0,1,3) - dma2 - second DMA # for WSS chip (0,1,3) - - This module supports only one card, autoprobe and PnP. - - Module snd-opti93x - ------------------ - - Module for sound cards based on OPTi 82c93x chips. - - isapnp - ISA PnP detection - 0 = disable, 1 = enable (default) - - with isapnp=0, the following options are available: - - port - port # for WSS chip (0x530,0xe80,0xf40,0x604) - mpu_port - port # for MPU-401 UART (0x300,0x310,0x320,0x330) - fm_port - port # for OPL3 device (0x388) - irq - IRQ # for WSS chip (5,7,9,10,11) - mpu_irq - IRQ # for MPU-401 UART (5,7,9,10) - dma1 - first DMA # for WSS chip (0,1,3) - dma2 - second DMA # for WSS chip (0,1,3) - - This module supports only one card, autoprobe and PnP. - - Module snd-oxygen - ----------------- - - Module for sound cards based on the C-Media CMI8786/8787/8788 chip: - * Asound A-8788 - * Asus Xonar DG/DGX - * AuzenTech X-Meridian - * AuzenTech X-Meridian 2G - * Bgears b-Enspirer - * Club3D Theatron DTS - * HT-Omega Claro (plus) - * HT-Omega Claro halo (XT) - * Kuroutoshikou CMI8787-HG2PCI - * Razer Barracuda AC-1 - * Sondigo Inferno - * TempoTec HiFier Fantasia - * TempoTec HiFier Serenade - - This module supports autoprobe and multiple cards. - - Module snd-pcsp - ----------------- - - Module for internal PC-Speaker. - - nopcm - Disable PC-Speaker PCM sound. Only beeps remain. - nforce_wa - enable NForce chipset workaround. Expect bad sound. - - This module supports system beeps, some kind of PCM playback and - even a few mixer controls. - - Module snd-pcxhr - ---------------- - - Module for Digigram PCXHR boards - - This module supports multiple cards. - - Module snd-portman2x4 - --------------------- - - Module for Midiman Portman 2x4 parallel port MIDI interface - - This module supports multiple cards. - - Module snd-powermac (on ppc only) - --------------------------------- - - Module for PowerMac, iMac and iBook on-board soundchips - - enable_beep - enable beep using PCM (enabled as default) - - Module supports autoprobe a chip. - - Note: the driver may have problems regarding endianness. - - The power-management is supported. - - Module snd-pxa2xx-ac97 (on arm only) - ------------------------------------ - - Module for AC97 driver for the Intel PXA2xx chip - - For ARM architecture only. - - The power-management is supported. - - Module snd-riptide - ------------------ - - Module for Conexant Riptide chip - - joystick_port - Joystick port # (default: 0x200) - mpu_port - MPU401 port # (default: 0x330) - opl3_port - OPL3 port # (default: 0x388) - - This module supports multiple cards. - The driver requires the firmware loader support on kernel. - You need to install the firmware file "riptide.hex" to the standard - firmware path (e.g. /lib/firmware). - - Module snd-rme32 - ---------------- - - Module for RME Digi32, Digi32 Pro and Digi32/8 (Sek'd Prodif32, - Prodif96 and Prodif Gold) sound cards. - - This module supports multiple cards. - - Module snd-rme96 - ---------------- - - Module for RME Digi96, Digi96/8 and Digi96/8 PRO/PAD/PST sound cards. - - This module supports multiple cards. - - Module snd-rme9652 - ------------------ - - Module for RME Digi9652 (Hammerfall, Hammerfall-Light) sound cards. - - precise_ptr - Enable precise pointer (doesn't work reliably). - (default = 0) - - This module supports multiple cards. - - Note: snd-page-alloc module does the job which snd-hammerfall-mem - module did formerly. It will allocate the buffers in advance - when any RME9652 cards are found. To make the buffer - allocation sure, load snd-page-alloc module in the early - stage of boot sequence. See "Early Buffer Allocation" - section. - - Module snd-sa11xx-uda1341 (on arm only) - --------------------------------------- - - Module for Philips UDA1341TS on Compaq iPAQ H3600 sound card. - - Module supports only one card. - Module has no enable and index options. - - The power-management is supported. - - Module snd-sb8 - -------------- - - Module for 8-bit SoundBlaster cards: SoundBlaster 1.0, - SoundBlaster 2.0, - SoundBlaster Pro - - port - port # for SB DSP chip (0x220,0x240,0x260) - irq - IRQ # for SB DSP chip (5,7,9,10) - dma8 - DMA # for SB DSP chip (1,3) - - This module supports multiple cards and autoprobe. - - The power-management is supported. - - Module snd-sb16 and snd-sbawe - ----------------------------- - - Module for 16-bit SoundBlaster cards: SoundBlaster 16 (PnP), - SoundBlaster AWE 32 (PnP), - SoundBlaster AWE 64 PnP - - mic_agc - Mic Auto-Gain-Control - 0 = disable, 1 = enable (default) - csp - ASP/CSP chip support - 0 = disable (default), 1 = enable - isapnp - ISA PnP detection - 0 = disable, 1 = enable (default) - - with isapnp=0, the following options are available: - - port - port # for SB DSP 4.x chip (0x220,0x240,0x260) - mpu_port - port # for MPU-401 UART (0x300,0x330), -1 = disable - awe_port - base port # for EMU8000 synthesizer (0x620,0x640,0x660) - (snd-sbawe module only) - irq - IRQ # for SB DSP 4.x chip (5,7,9,10) - dma8 - 8-bit DMA # for SB DSP 4.x chip (0,1,3) - dma16 - 16-bit DMA # for SB DSP 4.x chip (5,6,7) - - This module supports multiple cards, autoprobe and ISA PnP. - - Note: To use Vibra16X cards in 16-bit half duplex mode, you must - disable 16bit DMA with dma16 = -1 module parameter. - Also, all Sound Blaster 16 type cards can operate in 16-bit - half duplex mode through 8-bit DMA channel by disabling their - 16-bit DMA channel. - - The power-management is supported. - - Module snd-sc6000 - ----------------- - - Module for Gallant SC-6000 soundcard and later models: SC-6600 - and SC-7000. - - port - Port # (0x220 or 0x240) - mss_port - MSS Port # (0x530 or 0xe80) - irq - IRQ # (5,7,9,10,11) - mpu_irq - MPU-401 IRQ # (5,7,9,10) ,0 - no MPU-401 irq - dma - DMA # (1,3,0) - joystick - Enable gameport - 0 = disable (default), 1 = enable - - This module supports multiple cards. - - This card is also known as Audio Excel DSP 16 or Zoltrix AV302. - - Module snd-sscape - ----------------- - - Module for ENSONIQ SoundScape cards. - - port - Port # (PnP setup) - wss_port - WSS Port # (PnP setup) - irq - IRQ # (PnP setup) - mpu_irq - MPU-401 IRQ # (PnP setup) - dma - DMA # (PnP setup) - dma2 - 2nd DMA # (PnP setup, -1 to disable) - joystick - Enable gameport - 0 = disable (default), 1 = enable - - This module supports multiple cards. - - The driver requires the firmware loader support on kernel. - - Module snd-sun-amd7930 (on sparc only) - -------------------------------------- - - Module for AMD7930 sound chips found on Sparcs. - - This module supports multiple cards. - - Module snd-sun-cs4231 (on sparc only) - ------------------------------------- - - Module for CS4231 sound chips found on Sparcs. - - This module supports multiple cards. - - Module snd-sun-dbri (on sparc only) - ----------------------------------- - - Module for DBRI sound chips found on Sparcs. - - This module supports multiple cards. - - Module snd-wavefront - -------------------- - - Module for Turtle Beach Maui, Tropez and Tropez+ sound cards. - - use_cs4232_midi - Use CS4232 MPU-401 interface - (inaccessibly located inside your computer) - isapnp - ISA PnP detection - 0 = disable, 1 = enable (default) - - with isapnp=0, the following options are available: - - cs4232_pcm_port - Port # for CS4232 PCM interface. - cs4232_pcm_irq - IRQ # for CS4232 PCM interface (5,7,9,11,12,15). - cs4232_mpu_port - Port # for CS4232 MPU-401 interface. - cs4232_mpu_irq - IRQ # for CS4232 MPU-401 interface (9,11,12,15). - ics2115_port - Port # for ICS2115 - ics2115_irq - IRQ # for ICS2115 - fm_port - FM OPL-3 Port # - dma1 - DMA1 # for CS4232 PCM interface. - dma2 - DMA2 # for CS4232 PCM interface. - - The below are options for wavefront_synth features: - wf_raw - Assume that we need to boot the OS (default:no) - If yes, then during driver loading, the state of the board is - ignored, and we reset the board and load the firmware anyway. - fx_raw - Assume that the FX process needs help (default:yes) - If false, we'll leave the FX processor in whatever state it is - when the driver is loaded. The default is to download the - microprogram and associated coefficients to set it up for - "default" operation, whatever that means. - debug_default - Debug parameters for card initialization - wait_usecs - How long to wait without sleeping, usecs - (default:150) - This magic number seems to give pretty optimal throughput - based on my limited experimentation. - If you want to play around with it and find a better value, be - my guest. Remember, the idea is to get a number that causes us - to just busy wait for as many WaveFront commands as possible, - without coming up with a number so large that we hog the whole - CPU. - Specifically, with this number, out of about 134,000 status - waits, only about 250 result in a sleep. - sleep_interval - How long to sleep when waiting for reply - (default: 100) - sleep_tries - How many times to try sleeping during a wait - (default: 50) - ospath - Pathname to processed ICS2115 OS firmware - (default:wavefront.os) - The path name of the ISC2115 OS firmware. In the recent - version, it's handled via firmware loader framework, so it - must be installed in the proper path, typically, - /lib/firmware. - reset_time - How long to wait for a reset to take effect - (default:2) - ramcheck_time - How many seconds to wait for the RAM test - (default:20) - osrun_time - How many seconds to wait for the ICS2115 OS - (default:10) - - This module supports multiple cards and ISA PnP. - - Note: the firmware file "wavefront.os" was located in the earlier - version in /etc. Now it's loaded via firmware loader, and - must be in the proper firmware path, such as /lib/firmware. - Copy (or symlink) the file appropriately if you get an error - regarding firmware downloading after upgrading the kernel. - - Module snd-sonicvibes - --------------------- - - Module for S3 SonicVibes PCI sound cards. - * PINE Schubert 32 PCI - - reverb - Reverb Enable - 1 = enable, 0 = disable (default) - - SoundCard must have onboard SRAM for this. - mge - Mic Gain Enable - 1 = enable, 0 = disable (default) - - This module supports multiple cards and autoprobe. - - Module snd-serial-u16550 - ------------------------ - - Module for UART16550A serial MIDI ports. - - port - port # for UART16550A chip - irq - IRQ # for UART16550A chip, -1 = poll mode - speed - speed in bauds (9600,19200,38400,57600,115200) - 38400 = default - base - base for divisor in bauds (57600,115200,230400,460800) - 115200 = default - outs - number of MIDI ports in a serial port (1-4) - 1 = default - adaptor - Type of adaptor. - 0 = Soundcanvas, 1 = MS-124T, 2 = MS-124W S/A, - 3 = MS-124W M/B, 4 = Generic - - This module supports multiple cards. This module does not support autoprobe - thus the main port must be specified!!! Other options are optional. - - Module snd-trident - ------------------ - - Module for Trident 4DWave DX/NX sound cards. - * Best Union Miss Melody 4DWave PCI - * HIS 4DWave PCI - * Warpspeed ONSpeed 4DWave PCI - * AzTech PCI 64-Q3D - * Addonics SV 750 - * CHIC True Sound 4Dwave - * Shark Predator4D-PCI - * Jaton SonicWave 4D - * SiS SI7018 PCI Audio - * Hoontech SoundTrack Digital 4DWave NX - - pcm_channels - max channels (voices) reserved for PCM - wavetable_size - max wavetable size in kB (4-?kb) - - This module supports multiple cards and autoprobe. - - The power-management is supported. - - Module snd-ua101 - ---------------- - - Module for the Edirol UA-101/UA-1000 audio/MIDI interfaces. - - This module supports multiple devices, autoprobe and hotplugging. - - Module snd-usb-audio - -------------------- - - Module for USB audio and USB MIDI devices. - - vid - Vendor ID for the device (optional) - pid - Product ID for the device (optional) - nrpacks - Max. number of packets per URB (default: 8) - device_setup - Device specific magic number (optional) - - Influence depends on the device - - Default: 0x0000 - ignore_ctl_error - Ignore any USB-controller regarding mixer - interface (default: no) - autoclock - Enable auto-clock selection for UAC2 devices - (default: yes) - quirk_alias - Quirk alias list, pass strings like - "0123abcd:5678beef", which applies the existing - quirk for the device 5678:beef to a new device - 0123:abcd. - - This module supports multiple devices, autoprobe and hotplugging. - - NB: nrpacks parameter can be modified dynamically via sysfs. - Don't put the value over 20. Changing via sysfs has no sanity - check. - NB: ignore_ctl_error=1 may help when you get an error at accessing - the mixer element such as URB error -22. This happens on some - buggy USB device or the controller. - NB: quirk_alias option is provided only for testing / development. - If you want to have a proper support, contact to upstream for - adding the matching quirk in the driver code statically. - - Module snd-usb-caiaq - -------------------- - - Module for caiaq UB audio interfaces, - * Native Instruments RigKontrol2 - * Native Instruments Kore Controller - * Native Instruments Audio Kontrol 1 - * Native Instruments Audio 8 DJ - - This module supports multiple devices, autoprobe and hotplugging. - - Module snd-usb-usx2y - -------------------- - - Module for Tascam USB US-122, US-224 and US-428 devices. - - This module supports multiple devices, autoprobe and hotplugging. - - Note: you need to load the firmware via usx2yloader utility included - in alsa-tools and alsa-firmware packages. - - Module snd-via82xx - ------------------ - - Module for AC'97 motherboards based on VIA 82C686A/686B, 8233, - 8233A, 8233C, 8235, 8237 (south) bridge. - - mpu_port - 0x300,0x310,0x320,0x330, otherwise obtain BIOS setup - [VIA686A/686B only] - joystick - Enable joystick (default off) [VIA686A/686B only] - ac97_clock - AC'97 codec clock base (default 48000Hz) - dxs_support - support DXS channels, - 0 = auto (default), 1 = enable, 2 = disable, - 3 = 48k only, 4 = no VRA, 5 = enable any sample - rate and different sample rates on different - channels - [VIA8233/C, 8235, 8237 only] - ac97_quirk - AC'97 workaround for strange hardware - See "AC97 Quirk Option" section below. - - This module supports one chip and autoprobe. - - Note: on some SMP motherboards like MSI 694D the interrupts might - not be generated properly. In such a case, please try to - set the SMP (or MPS) version on BIOS to 1.1 instead of - default value 1.4. Then the interrupt number will be - assigned under 15. You might also upgrade your BIOS. - - Note: VIA8233/5/7 (not VIA8233A) can support DXS (direct sound) - channels as the first PCM. On these channels, up to 4 - streams can be played at the same time, and the controller - can perform sample rate conversion with separate rates for - each channel. - As default (dxs_support = 0), 48k fixed rate is chosen - except for the known devices since the output is often - noisy except for 48k on some mother boards due to the - bug of BIOS. - Please try once dxs_support=5 and if it works on other - sample rates (e.g. 44.1kHz of mp3 playback), please let us - know the PCI subsystem vendor/device id's (output of - "lspci -nv"). - If dxs_support=5 does not work, try dxs_support=4; if it - doesn't work too, try dxs_support=1. (dxs_support=1 is - usually for old motherboards. The correct implemented - board should work with 4 or 5.) If it still doesn't - work and the default setting is ok, dxs_support=3 is the - right choice. If the default setting doesn't work at all, - try dxs_support=2 to disable the DXS channels. - In any cases, please let us know the result and the - subsystem vendor/device ids. See "Links and Addresses" - below. - - Note: for the MPU401 on VIA823x, use snd-mpu401 driver - additionally. The mpu_port option is for VIA686 chips only. - - The power-management is supported. - - Module snd-via82xx-modem - ------------------------ - - Module for VIA82xx AC97 modem - - ac97_clock - AC'97 codec clock base (default 48000Hz) - - This module supports one card and autoprobe. - - Note: The default index value of this module is -2, i.e. the first - slot is excluded. - - The power-management is supported. - - Module snd-virmidi - ------------------ - - Module for virtual rawmidi devices. - This module creates virtual rawmidi devices which communicate - to the corresponding ALSA sequencer ports. - - midi_devs - MIDI devices # (1-4, default=4) - - This module supports multiple cards. - - Module snd-virtuoso - ------------------- - - Module for sound cards based on the Asus AV66/AV100/AV200 chips, - i.e., Xonar D1, DX, D2, D2X, DS, DSX, Essence ST (Deluxe), - Essence STX (II), HDAV1.3 (Deluxe), and HDAV1.3 Slim. - - This module supports autoprobe and multiple cards. - - Module snd-vx222 - ---------------- - - Module for Digigram VX-Pocket VX222, V222 v2 and Mic cards. - - mic - Enable Microphone on V222 Mic (NYI) - ibl - Capture IBL size. (default = 0, minimum size) - - This module supports multiple cards. - - When the driver is compiled as a module and the hotplug firmware - is supported, the firmware data is loaded via hotplug automatically. - Install the necessary firmware files in alsa-firmware package. - When no hotplug fw loader is available, you need to load the - firmware via vxloader utility in alsa-tools package. To invoke - vxloader automatically, add the following to /etc/modprobe.d/alsa.conf - - install snd-vx222 /sbin/modprobe --first-time -i snd-vx222 && /usr/bin/vxloader - - (for 2.2/2.4 kernels, add "post-install /usr/bin/vxloader" to - /etc/modules.conf, instead.) - IBL size defines the interrupts period for PCM. The smaller size - gives smaller latency but leads to more CPU consumption, too. - The size is usually aligned to 126. As default (=0), the smallest - size is chosen. The possible IBL values can be found in - /proc/asound/cardX/vx-status proc file. - - The power-management is supported. - - Module snd-vxpocket - ------------------- - - Module for Digigram VX-Pocket VX2 and 440 PCMCIA cards. - - ibl - Capture IBL size. (default = 0, minimum size) - - This module supports multiple cards. The module is compiled only when - PCMCIA is supported on kernel. - - With the older 2.6.x kernel, to activate the driver via the card - manager, you'll need to set up /etc/pcmcia/vxpocket.conf. See the - sound/pcmcia/vx/vxpocket.c. 2.6.13 or later kernel requires no - longer require a config file. - - When the driver is compiled as a module and the hotplug firmware - is supported, the firmware data is loaded via hotplug automatically. - Install the necessary firmware files in alsa-firmware package. - When no hotplug fw loader is available, you need to load the - firmware via vxloader utility in alsa-tools package. - - About capture IBL, see the description of snd-vx222 module. - - Note: snd-vxp440 driver is merged to snd-vxpocket driver since - ALSA 1.0.10. - - The power-management is supported. - - Module snd-ymfpci - ----------------- - - Module for Yamaha PCI chips (YMF72x, YMF74x & YMF75x). - - mpu_port - 0x300,0x330,0x332,0x334, 0 (disable) by default, - 1 (auto-detect for YMF744/754 only) - fm_port - 0x388,0x398,0x3a0,0x3a8, 0 (disable) by default - 1 (auto-detect for YMF744/754 only) - joystick_port - 0x201,0x202,0x204,0x205, 0 (disable) by default, - 1 (auto-detect) - rear_switch - enable shared rear/line-in switch (bool) - - This module supports autoprobe and multiple chips. - - The power-management is supported. - - Module snd-pdaudiocf - -------------------- - - Module for Sound Core PDAudioCF sound card. - - The power-management is supported. - - -AC97 Quirk Option -================= - -The ac97_quirk option is used to enable/override the workaround for -specific devices on drivers for on-board AC'97 controllers like -snd-intel8x0. Some hardware have swapped output pins between Master -and Headphone, or Surround (thanks to confusion of AC'97 -specifications from version to version :-) - -The driver provides the auto-detection of known problematic devices, -but some might be unknown or wrongly detected. In such a case, pass -the proper value with this option. - -The following strings are accepted: - - default Don't override the default setting - - none Disable the quirk - - hp_only Bind Master and Headphone controls as a single control - - swap_hp Swap headphone and master controls - - swap_surround Swap master and surround controls - - ad_sharing For AD1985, turn on OMS bit and use headphone - - alc_jack For ALC65x, turn on the jack sense mode - - inv_eapd Inverted EAPD implementation - - mute_led Bind EAPD bit for turning on/off mute LED - -For backward compatibility, the corresponding integer value -1, 0, -... are accepted, too. - -For example, if "Master" volume control has no effect on your device -but only "Headphone" does, pass ac97_quirk=hp_only module option. - - -Configuring Non-ISAPNP Cards -============================ - -When the kernel is configured with ISA-PnP support, the modules -supporting the isapnp cards will have module options "isapnp". -If this option is set, *only* the ISA-PnP devices will be probed. -For probing the non ISA-PnP cards, you have to pass "isapnp=0" option -together with the proper i/o and irq configuration. - -When the kernel is configured without ISA-PnP support, isapnp option -will be not built in. - - -Module Autoloading Support -========================== - -The ALSA drivers can be loaded automatically on demand by defining -module aliases. The string 'snd-card-%1' is requested for ALSA native -devices where %i is sound card number from zero to seven. - -To auto-load an ALSA driver for OSS services, define the string -'sound-slot-%i' where %i means the slot number for OSS, which -corresponds to the card index of ALSA. Usually, define this -as the same card module. - -An example configuration for a single emu10k1 card is like below: ------ /etc/modprobe.d/alsa.conf -alias snd-card-0 snd-emu10k1 -alias sound-slot-0 snd-emu10k1 ------ /etc/modprobe.d/alsa.conf - -The available number of auto-loaded sound cards depends on the module -option "cards_limit" of snd module. As default it's set to 1. -To enable the auto-loading of multiple cards, specify the number of -sound cards in that option. - -When multiple cards are available, it'd better to specify the index -number for each card via module option, too, so that the order of -cards is kept consistent. - -An example configuration for two sound cards is like below: - ------ /etc/modprobe.d/alsa.conf -# ALSA portion -options snd cards_limit=2 -alias snd-card-0 snd-interwave -alias snd-card-1 snd-ens1371 -options snd-interwave index=0 -options snd-ens1371 index=1 -# OSS/Free portion -alias sound-slot-0 snd-interwave -alias sound-slot-1 snd-ens1371 ------ /etc/modprobe.d/alsa.conf - -In this example, the interwave card is always loaded as the first card -(index 0) and ens1371 as the second (index 1). - -Alternative (and new) way to fixate the slot assignment is to use -"slots" option of snd module. In the case above, specify like the -following: - -options snd slots=snd-interwave,snd-ens1371 - -Then, the first slot (#0) is reserved for snd-interwave driver, and -the second (#1) for snd-ens1371. You can omit index option in each -driver if slots option is used (although you can still have them at -the same time as long as they don't conflict). - -The slots option is especially useful for avoiding the possible -hot-plugging and the resultant slot conflict. For example, in the -case above again, the first two slots are already reserved. If any -other driver (e.g. snd-usb-audio) is loaded before snd-interwave or -snd-ens1371, it will be assigned to the third or later slot. - -When a module name is given with '!', the slot will be given for any -modules but that name. For example, "slots=!snd-pcsp" will reserve -the first slot for any modules but snd-pcsp. - - -ALSA PCM devices to OSS devices mapping -======================================= - -/dev/snd/pcmC0D0[c|p] -> /dev/audio0 (/dev/audio) -> minor 4 -/dev/snd/pcmC0D0[c|p] -> /dev/dsp0 (/dev/dsp) -> minor 3 -/dev/snd/pcmC0D1[c|p] -> /dev/adsp0 (/dev/adsp) -> minor 12 -/dev/snd/pcmC1D0[c|p] -> /dev/audio1 -> minor 4+16 = 20 -/dev/snd/pcmC1D0[c|p] -> /dev/dsp1 -> minor 3+16 = 19 -/dev/snd/pcmC1D1[c|p] -> /dev/adsp1 -> minor 12+16 = 28 -/dev/snd/pcmC2D0[c|p] -> /dev/audio2 -> minor 4+32 = 36 -/dev/snd/pcmC2D0[c|p] -> /dev/dsp2 -> minor 3+32 = 39 -/dev/snd/pcmC2D1[c|p] -> /dev/adsp2 -> minor 12+32 = 44 - -The first number from /dev/snd/pcmC{X}D{Y}[c|p] expression means -sound card number and second means device number. The ALSA devices -have either 'c' or 'p' suffix indicating the direction, capture and -playback, respectively. - -Please note that the device mapping above may be varied via the module -options of snd-pcm-oss module. - - -Proc interfaces (/proc/asound) -============================== - -/proc/asound/card#/pcm#[cp]/oss -------------------------------- - String "erase" - erase all additional information about OSS applications - String " []" - - - name of application with (higher priority) or without path - - number of fragments or zero if auto - - size of fragment in bytes or zero if auto - - optional parameters - - disable the application tries to open a pcm device for - this channel but does not want to use it. - (Cause a bug or mmap needs) - It's good for Quake etc... - - direct don't use plugins - - block force block mode (rvplayer) - - non-block force non-block mode - - whole-frag write only whole fragments (optimization affecting - playback only) - - no-silence do not fill silence ahead to avoid clicks - - buggy-ptr Returns the whitespace blocks in GETOPTR ioctl - instead of filled blocks - - Example: echo "x11amp 128 16384" > /proc/asound/card0/pcm0p/oss - echo "squake 0 0 disable" > /proc/asound/card0/pcm0c/oss - echo "rvplayer 0 0 block" > /proc/asound/card0/pcm0p/oss - - -Early Buffer Allocation -======================= - -Some drivers (e.g. hdsp) require the large contiguous buffers, and -sometimes it's too late to find such spaces when the driver module is -actually loaded due to memory fragmentation. You can pre-allocate the -PCM buffers by loading snd-page-alloc module and write commands to its -proc file in prior, for example, in the early boot stage like -/etc/init.d/*.local scripts. - -Reading the proc file /proc/drivers/snd-page-alloc shows the current -usage of page allocation. In writing, you can send the following -commands to the snd-page-alloc driver: - - - add VENDOR DEVICE MASK SIZE BUFFERS - - VENDOR and DEVICE are PCI vendor and device IDs. They take - integer numbers (0x prefix is needed for the hex). - MASK is the PCI DMA mask. Pass 0 if not restricted. - SIZE is the size of each buffer to allocate. You can pass - k and m suffix for KB and MB. The max number is 16MB. - BUFFERS is the number of buffers to allocate. It must be greater - than 0. The max number is 4. - - - erase - - This will erase the all pre-allocated buffers which are not in - use. - - -Links and Addresses -=================== - - ALSA project homepage - http://www.alsa-project.org - - Kernel Bugzilla - http://bugzilla.kernel.org/ - - ALSA Developers ML - mailto:alsa-devel@alsa-project.org - - alsa-info.sh script - http://www.alsa-project.org/alsa-info.sh diff --git a/Documentation/sound/index.rst b/Documentation/sound/index.rst index e9bac7c8b893..64fe47af05b6 100644 --- a/Documentation/sound/index.rst +++ b/Documentation/sound/index.rst @@ -6,6 +6,7 @@ Linux Sound Subsystem Documentation :maxdepth: 2 kernel-api/index + alsa-configuration hd-audio/index .. only:: subproject -- cgit v1.2.3-58-ga151 From afb8fd3c72a9f00018f7335590c173ce4cc2a43d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 9 Nov 2016 15:52:06 +0100 Subject: ALSA: doc: ReSTize Procfile document A simple conversion from a text file. A new subidrectory, Documentation/sound/designs, was created to put this document. The other API design and implementation docuemnts will be put to that directory in later commits. Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/Procfile.txt | 234 ------------------------------ Documentation/sound/designs/index.rst | 7 + Documentation/sound/designs/procfile.rst | 238 +++++++++++++++++++++++++++++++ Documentation/sound/index.rst | 1 + 4 files changed, 246 insertions(+), 234 deletions(-) delete mode 100644 Documentation/sound/alsa/Procfile.txt create mode 100644 Documentation/sound/designs/index.rst create mode 100644 Documentation/sound/designs/procfile.rst (limited to 'Documentation/sound') diff --git a/Documentation/sound/alsa/Procfile.txt b/Documentation/sound/alsa/Procfile.txt deleted file mode 100644 index 7f8a0d325905..000000000000 --- a/Documentation/sound/alsa/Procfile.txt +++ /dev/null @@ -1,234 +0,0 @@ - Proc Files of ALSA Drivers - ========================== - Takashi Iwai - -General -------- - -ALSA has its own proc tree, /proc/asound. Many useful information are -found in this tree. When you encounter a problem and need debugging, -check the files listed in the following sections. - -Each card has its subtree cardX, where X is from 0 to 7. The -card-specific files are stored in the card* subdirectories. - - -Global Information ------------------- - -cards - Shows the list of currently configured ALSA drivers, - index, the id string, short and long descriptions. - -version - Shows the version string and compile date. - -modules - Lists the module of each card - -devices - Lists the ALSA native device mappings. - -meminfo - Shows the status of allocated pages via ALSA drivers. - Appears only when CONFIG_SND_DEBUG=y. - -hwdep - Lists the currently available hwdep devices in format of - -: - -pcm - Lists the currently available PCM devices in format of - -: : : - -timer - Lists the currently available timer devices - - -oss/devices - Lists the OSS device mappings. - -oss/sndstat - Provides the output compatible with /dev/sndstat. - You can symlink this to /dev/sndstat. - - -Card Specific Files -------------------- - -The card-specific files are found in /proc/asound/card* directories. -Some drivers (e.g. cmipci) have their own proc entries for the -register dump, etc (e.g. /proc/asound/card*/cmipci shows the register -dump). These files would be really helpful for debugging. - -When PCM devices are available on this card, you can see directories -like pcm0p or pcm1c. They hold the PCM information for each PCM -stream. The number after 'pcm' is the PCM device number from 0, and -the last 'p' or 'c' means playback or capture direction. The files in -this subtree is described later. - -The status of MIDI I/O is found in midi* files. It shows the device -name and the received/transmitted bytes through the MIDI device. - -When the card is equipped with AC97 codecs, there are codec97#* -subdirectories (described later). - -When the OSS mixer emulation is enabled (and the module is loaded), -oss_mixer file appears here, too. This shows the current mapping of -OSS mixer elements to the ALSA control elements. You can change the -mapping by writing to this device. Read OSS-Emulation.txt for -details. - - -PCM Proc Files --------------- - -card*/pcm*/info - The general information of this PCM device: card #, device #, - substreams, etc. - -card*/pcm*/xrun_debug - This file appears when CONFIG_SND_DEBUG=y and - CONFIG_PCM_XRUN_DEBUG=y. - This shows the status of xrun (= buffer overrun/xrun) and - invalid PCM position debug/check of ALSA PCM middle layer. - It takes an integer value, can be changed by writing to this - file, such as - - # echo 5 > /proc/asound/card0/pcm0p/xrun_debug - - The value consists of the following bit flags: - bit 0 = Enable XRUN/jiffies debug messages - bit 1 = Show stack trace at XRUN / jiffies check - bit 2 = Enable additional jiffies check - - When the bit 0 is set, the driver will show the messages to - kernel log when an xrun is detected. The debug message is - shown also when the invalid H/W pointer is detected at the - update of periods (usually called from the interrupt - handler). - - When the bit 1 is set, the driver will show the stack trace - additionally. This may help the debugging. - - Since 2.6.30, this option can enable the hwptr check using - jiffies. This detects spontaneous invalid pointer callback - values, but can be lead to too much corrections for a (mostly - buggy) hardware that doesn't give smooth pointer updates. - This feature is enabled via the bit 2. - -card*/pcm*/sub*/info - The general information of this PCM sub-stream. - -card*/pcm*/sub*/status - The current status of this PCM sub-stream, elapsed time, - H/W position, etc. - -card*/pcm*/sub*/hw_params - The hardware parameters set for this sub-stream. - -card*/pcm*/sub*/sw_params - The soft parameters set for this sub-stream. - -card*/pcm*/sub*/prealloc - The buffer pre-allocation information. - -card*/pcm*/sub*/xrun_injection - Triggers an XRUN to the running stream when any value is - written to this proc file. Used for fault injection. - This entry is write-only. - -AC97 Codec Information ----------------------- - -card*/codec97#*/ac97#?-? - Shows the general information of this AC97 codec chip, such as - name, capabilities, set up. - -card*/codec97#0/ac97#?-?+regs - Shows the AC97 register dump. Useful for debugging. - - When CONFIG_SND_DEBUG is enabled, you can write to this file for - changing an AC97 register directly. Pass two hex numbers. - For example, - - # echo 02 9f1f > /proc/asound/card0/codec97#0/ac97#0-0+regs - - -USB Audio Streams ------------------ - -card*/stream* - Shows the assignment and the current status of each audio stream - of the given card. This information is very useful for debugging. - - -HD-Audio Codecs ---------------- - -card*/codec#* - Shows the general codec information and the attribute of each - widget node. - -card*/eld#* - Available for HDMI or DisplayPort interfaces. - Shows ELD(EDID Like Data) info retrieved from the attached HDMI sink, - and describes its audio capabilities and configurations. - - Some ELD fields may be modified by doing `echo name hex_value > eld#*`. - Only do this if you are sure the HDMI sink provided value is wrong. - And if that makes your HDMI audio work, please report to us so that we - can fix it in future kernel releases. - - -Sequencer Information ---------------------- - -seq/drivers - Lists the currently available ALSA sequencer drivers. - -seq/clients - Shows the list of currently available sequencer clients and - ports. The connection status and the running status are shown - in this file, too. - -seq/queues - Lists the currently allocated/running sequencer queues. - -seq/timer - Lists the currently allocated/running sequencer timers. - -seq/oss - Lists the OSS-compatible sequencer stuffs. - - -Help For Debugging? -------------------- - -When the problem is related with PCM, first try to turn on xrun_debug -mode. This will give you the kernel messages when and where xrun -happened. - -If it's really a bug, report it with the following information: - - - the name of the driver/card, show in /proc/asound/cards - - the register dump, if available (e.g. card*/cmipci) - -when it's a PCM problem, - - - set-up of PCM, shown in hw_parms, sw_params, and status in the PCM - sub-stream directory - -when it's a mixer problem, - - - AC97 proc files, codec97#*/* files - -for USB audio/midi, - - - output of lsusb -v - - stream* files in card directory - - -The ALSA bug-tracking system is found at: - - https://bugtrack.alsa-project.org/alsa-bug/ diff --git a/Documentation/sound/designs/index.rst b/Documentation/sound/designs/index.rst new file mode 100644 index 000000000000..82d19fe3c380 --- /dev/null +++ b/Documentation/sound/designs/index.rst @@ -0,0 +1,7 @@ +Designs and Implementations +=========================== + +.. toctree:: + :maxdepth: 2 + + procfile diff --git a/Documentation/sound/designs/procfile.rst b/Documentation/sound/designs/procfile.rst new file mode 100644 index 000000000000..29a466851fd2 --- /dev/null +++ b/Documentation/sound/designs/procfile.rst @@ -0,0 +1,238 @@ +========================== +Proc Files of ALSA Drivers +========================== + +Takashi Iwai + +General +======= + +ALSA has its own proc tree, /proc/asound. Many useful information are +found in this tree. When you encounter a problem and need debugging, +check the files listed in the following sections. + +Each card has its subtree cardX, where X is from 0 to 7. The +card-specific files are stored in the ``card*`` subdirectories. + + +Global Information +================== + +cards + Shows the list of currently configured ALSA drivers, + index, the id string, short and long descriptions. + +version + Shows the version string and compile date. + +modules + Lists the module of each card + +devices + Lists the ALSA native device mappings. + +meminfo + Shows the status of allocated pages via ALSA drivers. + Appears only when ``CONFIG_SND_DEBUG=y``. + +hwdep + Lists the currently available hwdep devices in format of + ``-: `` + +pcm + Lists the currently available PCM devices in format of + ``-: : : `` + +timer + Lists the currently available timer devices + + +oss/devices + Lists the OSS device mappings. + +oss/sndstat + Provides the output compatible with /dev/sndstat. + You can symlink this to /dev/sndstat. + + +Card Specific Files +=================== + +The card-specific files are found in ``/proc/asound/card*`` directories. +Some drivers (e.g. cmipci) have their own proc entries for the +register dump, etc (e.g. ``/proc/asound/card*/cmipci`` shows the register +dump). These files would be really helpful for debugging. + +When PCM devices are available on this card, you can see directories +like pcm0p or pcm1c. They hold the PCM information for each PCM +stream. The number after ``pcm`` is the PCM device number from 0, and +the last ``p`` or ``c`` means playback or capture direction. The files in +this subtree is described later. + +The status of MIDI I/O is found in ``midi*`` files. It shows the device +name and the received/transmitted bytes through the MIDI device. + +When the card is equipped with AC97 codecs, there are ``codec97#*`` +subdirectories (described later). + +When the OSS mixer emulation is enabled (and the module is loaded), +oss_mixer file appears here, too. This shows the current mapping of +OSS mixer elements to the ALSA control elements. You can change the +mapping by writing to this device. Read OSS-Emulation.txt for +details. + + +PCM Proc Files +============== + +``card*/pcm*/info`` + The general information of this PCM device: card #, device #, + substreams, etc. + +``card*/pcm*/xrun_debug`` + This file appears when ``CONFIG_SND_DEBUG=y`` and + ``CONFIG_PCM_XRUN_DEBUG=y``. + This shows the status of xrun (= buffer overrun/xrun) and + invalid PCM position debug/check of ALSA PCM middle layer. + It takes an integer value, can be changed by writing to this + file, such as:: + + # echo 5 > /proc/asound/card0/pcm0p/xrun_debug + + The value consists of the following bit flags: + + * bit 0 = Enable XRUN/jiffies debug messages + * bit 1 = Show stack trace at XRUN / jiffies check + * bit 2 = Enable additional jiffies check + + When the bit 0 is set, the driver will show the messages to + kernel log when an xrun is detected. The debug message is + shown also when the invalid H/W pointer is detected at the + update of periods (usually called from the interrupt + handler). + + When the bit 1 is set, the driver will show the stack trace + additionally. This may help the debugging. + + Since 2.6.30, this option can enable the hwptr check using + jiffies. This detects spontaneous invalid pointer callback + values, but can be lead to too much corrections for a (mostly + buggy) hardware that doesn't give smooth pointer updates. + This feature is enabled via the bit 2. + +``card*/pcm*/sub*/info`` + The general information of this PCM sub-stream. + +``card*/pcm*/sub*/status`` + The current status of this PCM sub-stream, elapsed time, + H/W position, etc. + +``card*/pcm*/sub*/hw_params`` + The hardware parameters set for this sub-stream. + +``card*/pcm*/sub*/sw_params`` + The soft parameters set for this sub-stream. + +``card*/pcm*/sub*/prealloc`` + The buffer pre-allocation information. + +``card*/pcm*/sub*/xrun_injection`` + Triggers an XRUN to the running stream when any value is + written to this proc file. Used for fault injection. + This entry is write-only. + +AC97 Codec Information +====================== + +``card*/codec97#*/ac97#?-?`` + Shows the general information of this AC97 codec chip, such as + name, capabilities, set up. + +``card*/codec97#0/ac97#?-?+regs`` + Shows the AC97 register dump. Useful for debugging. + + When CONFIG_SND_DEBUG is enabled, you can write to this file for + changing an AC97 register directly. Pass two hex numbers. + For example, + +:: + + # echo 02 9f1f > /proc/asound/card0/codec97#0/ac97#0-0+regs + + +USB Audio Streams +================= + +``card*/stream*`` + Shows the assignment and the current status of each audio stream + of the given card. This information is very useful for debugging. + + +HD-Audio Codecs +=============== + +``card*/codec#*`` + Shows the general codec information and the attribute of each + widget node. + +``card*/eld#*`` + Available for HDMI or DisplayPort interfaces. + Shows ELD(EDID Like Data) info retrieved from the attached HDMI sink, + and describes its audio capabilities and configurations. + + Some ELD fields may be modified by doing ``echo name hex_value > eld#*``. + Only do this if you are sure the HDMI sink provided value is wrong. + And if that makes your HDMI audio work, please report to us so that we + can fix it in future kernel releases. + + +Sequencer Information +===================== + +seq/drivers + Lists the currently available ALSA sequencer drivers. + +seq/clients + Shows the list of currently available sequencer clients and + ports. The connection status and the running status are shown + in this file, too. + +seq/queues + Lists the currently allocated/running sequencer queues. + +seq/timer + Lists the currently allocated/running sequencer timers. + +seq/oss + Lists the OSS-compatible sequencer stuffs. + + +Help For Debugging? +=================== + +When the problem is related with PCM, first try to turn on xrun_debug +mode. This will give you the kernel messages when and where xrun +happened. + +If it's really a bug, report it with the following information: + +- the name of the driver/card, show in ``/proc/asound/cards`` +- the register dump, if available (e.g. ``card*/cmipci``) + +when it's a PCM problem, + +- set-up of PCM, shown in hw_parms, sw_params, and status in the PCM + sub-stream directory + +when it's a mixer problem, + +- AC97 proc files, ``codec97#*/*`` files + +for USB audio/midi, + +- output of ``lsusb -v`` +- ``stream*`` files in card directory + + +The ALSA bug-tracking system is found at: +https://bugtrack.alsa-project.org/alsa-bug/ diff --git a/Documentation/sound/index.rst b/Documentation/sound/index.rst index 64fe47af05b6..e9fbbfff9d0d 100644 --- a/Documentation/sound/index.rst +++ b/Documentation/sound/index.rst @@ -6,6 +6,7 @@ Linux Sound Subsystem Documentation :maxdepth: 2 kernel-api/index + designs/index alsa-configuration hd-audio/index -- cgit v1.2.3-58-ga151 From 48e92b488d3c419e11bbf03c56ceb43399ac1901 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 9 Nov 2016 15:54:47 +0100 Subject: ALSA: doc: ReSTize powersave document A simple conversion from a text file. Put into designs subdirectory, although it's mostly relevant with HD-audio. Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/powersave.txt | 41 ----------------------------- Documentation/sound/designs/index.rst | 1 + Documentation/sound/designs/powersave.rst | 43 +++++++++++++++++++++++++++++++ 3 files changed, 44 insertions(+), 41 deletions(-) delete mode 100644 Documentation/sound/alsa/powersave.txt create mode 100644 Documentation/sound/designs/powersave.rst (limited to 'Documentation/sound') diff --git a/Documentation/sound/alsa/powersave.txt b/Documentation/sound/alsa/powersave.txt deleted file mode 100644 index 9657e8099228..000000000000 --- a/Documentation/sound/alsa/powersave.txt +++ /dev/null @@ -1,41 +0,0 @@ -Notes on Power-Saving Mode -========================== - -AC97 and HD-audio drivers have the automatic power-saving mode. -This feature is enabled via Kconfig CONFIG_SND_AC97_POWER_SAVE -and CONFIG_SND_HDA_POWER_SAVE options, respectively. - -With the automatic power-saving, the driver turns off the codec power -appropriately when no operation is required. When no applications use -the device and/or no analog loopback is set, the power disablement is -done fully or partially. It'll save a certain power consumption, thus -good for laptops (even for desktops). - -The time-out for automatic power-off can be specified via power_save -module option of snd-ac97-codec and snd-hda-intel modules. Specify -the time-out value in seconds. 0 means to disable the automatic -power-saving. The default value of timeout is given via -CONFIG_SND_AC97_POWER_SAVE_DEFAULT and -CONFIG_SND_HDA_POWER_SAVE_DEFAULT Kconfig options. Setting this to 1 -(the minimum value) isn't recommended because many applications try to -reopen the device frequently. 10 would be a good choice for normal -operations. - -The power_save option is exported as writable. This means you can -adjust the value via sysfs on the fly. For example, to turn on the -automatic power-save mode with 10 seconds, write to -/sys/modules/snd_ac97_codec/parameters/power_save (usually as root): - - # echo 10 > /sys/modules/snd_ac97_codec/parameters/power_save - - -Note that you might hear click noise/pop when changing the power -state. Also, it often takes certain time to wake up from the -power-down to the active state. These are often hardly to fix, so -don't report extra bug reports unless you have a fix patch ;-) - -For HD-audio interface, there is another module option, -power_save_controller. This enables/disables the power-save mode of -the controller side. Setting this on may reduce a bit more power -consumption, but might result in longer wake-up time and click noise. -Try to turn it off when you experience such a thing too often. diff --git a/Documentation/sound/designs/index.rst b/Documentation/sound/designs/index.rst index 82d19fe3c380..362e1c23d51f 100644 --- a/Documentation/sound/designs/index.rst +++ b/Documentation/sound/designs/index.rst @@ -5,3 +5,4 @@ Designs and Implementations :maxdepth: 2 procfile + powersave diff --git a/Documentation/sound/designs/powersave.rst b/Documentation/sound/designs/powersave.rst new file mode 100644 index 000000000000..138157452eb9 --- /dev/null +++ b/Documentation/sound/designs/powersave.rst @@ -0,0 +1,43 @@ +========================== +Notes on Power-Saving Mode +========================== + +AC97 and HD-audio drivers have the automatic power-saving mode. +This feature is enabled via Kconfig ``CONFIG_SND_AC97_POWER_SAVE`` +and ``CONFIG_SND_HDA_POWER_SAVE`` options, respectively. + +With the automatic power-saving, the driver turns off the codec power +appropriately when no operation is required. When no applications use +the device and/or no analog loopback is set, the power disablement is +done fully or partially. It'll save a certain power consumption, thus +good for laptops (even for desktops). + +The time-out for automatic power-off can be specified via ``power_save`` +module option of snd-ac97-codec and snd-hda-intel modules. Specify +the time-out value in seconds. 0 means to disable the automatic +power-saving. The default value of timeout is given via +``CONFIG_SND_AC97_POWER_SAVE_DEFAULT`` and +``CONFIG_SND_HDA_POWER_SAVE_DEFAULT`` Kconfig options. Setting this to 1 +(the minimum value) isn't recommended because many applications try to +reopen the device frequently. 10 would be a good choice for normal +operations. + +The ``power_save`` option is exported as writable. This means you can +adjust the value via sysfs on the fly. For example, to turn on the +automatic power-save mode with 10 seconds, write to +``/sys/modules/snd_ac97_codec/parameters/power_save`` (usually as root): +:: + + # echo 10 > /sys/modules/snd_ac97_codec/parameters/power_save + + +Note that you might hear click noise/pop when changing the power +state. Also, it often takes certain time to wake up from the +power-down to the active state. These are often hardly to fix, so +don't report extra bug reports unless you have a fix patch ;-) + +For HD-audio interface, there is another module option, +power_save_controller. This enables/disables the power-save mode of +the controller side. Setting this on may reduce a bit more power +consumption, but might result in longer wake-up time and click noise. +Try to turn it off when you experience such a thing too often. -- cgit v1.2.3-58-ga151 From efe541c230e41106ce912e096e19518630e810db Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 9 Nov 2016 16:05:18 +0100 Subject: ALSA: doc: ReSTize Channel-Mapping-API document A simple conversion from a text file. Put to designs subdirectory. Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/Channel-Mapping-API.txt | 153 ------------------- .../sound/designs/channel-mapping-api.rst | 164 +++++++++++++++++++++ Documentation/sound/designs/index.rst | 1 + 3 files changed, 165 insertions(+), 153 deletions(-) delete mode 100644 Documentation/sound/alsa/Channel-Mapping-API.txt create mode 100644 Documentation/sound/designs/channel-mapping-api.rst (limited to 'Documentation/sound') diff --git a/Documentation/sound/alsa/Channel-Mapping-API.txt b/Documentation/sound/alsa/Channel-Mapping-API.txt deleted file mode 100644 index 3c43d1a4ca0e..000000000000 --- a/Documentation/sound/alsa/Channel-Mapping-API.txt +++ /dev/null @@ -1,153 +0,0 @@ -ALSA PCM channel-mapping API -============================ - Takashi Iwai - -GENERAL -------- - -The channel mapping API allows user to query the possible channel maps -and the current channel map, also optionally to modify the channel map -of the current stream. - -A channel map is an array of position for each PCM channel. -Typically, a stereo PCM stream has a channel map of - { front_left, front_right } -while a 4.0 surround PCM stream has a channel map of - { front left, front right, rear left, rear right }. - -The problem, so far, was that we had no standard channel map -explicitly, and applications had no way to know which channel -corresponds to which (speaker) position. Thus, applications applied -wrong channels for 5.1 outputs, and you hear suddenly strange sound -from rear. Or, some devices secretly assume that center/LFE is the -third/fourth channels while others that C/LFE as 5th/6th channels. - -Also, some devices such as HDMI are configurable for different speaker -positions even with the same number of total channels. However, there -was no way to specify this because of lack of channel map -specification. These are the main motivations for the new channel -mapping API. - - -DESIGN ------- - -Actually, "the channel mapping API" doesn't introduce anything new in -the kernel/user-space ABI perspective. It uses only the existing -control element features. - -As a ground design, each PCM substream may contain a control element -providing the channel mapping information and configuration. This -element is specified by: - iface = SNDRV_CTL_ELEM_IFACE_PCM - name = "Playback Channel Map" or "Capture Channel Map" - device = the same device number for the assigned PCM substream - index = the same index number for the assigned PCM substream - -Note the name is different depending on the PCM substream direction. - -Each control element provides at least the TLV read operation and the -read operation. Optionally, the write operation can be provided to -allow user to change the channel map dynamically. - -* TLV - -The TLV operation gives the list of available channel -maps. A list item of a channel map is usually a TLV of - type data-bytes ch0 ch1 ch2... -where type is the TLV type value, the second argument is the total -bytes (not the numbers) of channel values, and the rest are the -position value for each channel. - -As a TLV type, either SNDRV_CTL_TLVT_CHMAP_FIXED, -SNDRV_CTL_TLV_CHMAP_VAR or SNDRV_CTL_TLVT_CHMAP_PAIRED can be used. -The _FIXED type is for a channel map with the fixed channel position -while the latter two are for flexible channel positions. _VAR type is -for a channel map where all channels are freely swappable and _PAIRED -type is where pair-wise channels are swappable. For example, when you -have {FL/FR/RL/RR} channel map, _PAIRED type would allow you to swap -only {RL/RR/FL/FR} while _VAR type would allow even swapping FL and -RR. - -These new TLV types are defined in sound/tlv.h. - -The available channel position values are defined in sound/asound.h, -here is a cut: - -/* channel positions */ -enum { - SNDRV_CHMAP_UNKNOWN = 0, - SNDRV_CHMAP_NA, /* N/A, silent */ - SNDRV_CHMAP_MONO, /* mono stream */ - /* this follows the alsa-lib mixer channel value + 3 */ - SNDRV_CHMAP_FL, /* front left */ - SNDRV_CHMAP_FR, /* front right */ - SNDRV_CHMAP_RL, /* rear left */ - SNDRV_CHMAP_RR, /* rear right */ - SNDRV_CHMAP_FC, /* front center */ - SNDRV_CHMAP_LFE, /* LFE */ - SNDRV_CHMAP_SL, /* side left */ - SNDRV_CHMAP_SR, /* side right */ - SNDRV_CHMAP_RC, /* rear center */ - /* new definitions */ - SNDRV_CHMAP_FLC, /* front left center */ - SNDRV_CHMAP_FRC, /* front right center */ - SNDRV_CHMAP_RLC, /* rear left center */ - SNDRV_CHMAP_RRC, /* rear right center */ - SNDRV_CHMAP_FLW, /* front left wide */ - SNDRV_CHMAP_FRW, /* front right wide */ - SNDRV_CHMAP_FLH, /* front left high */ - SNDRV_CHMAP_FCH, /* front center high */ - SNDRV_CHMAP_FRH, /* front right high */ - SNDRV_CHMAP_TC, /* top center */ - SNDRV_CHMAP_TFL, /* top front left */ - SNDRV_CHMAP_TFR, /* top front right */ - SNDRV_CHMAP_TFC, /* top front center */ - SNDRV_CHMAP_TRL, /* top rear left */ - SNDRV_CHMAP_TRR, /* top rear right */ - SNDRV_CHMAP_TRC, /* top rear center */ - SNDRV_CHMAP_LAST = SNDRV_CHMAP_TRC, -}; - -When a PCM stream can provide more than one channel map, you can -provide multiple channel maps in a TLV container type. The TLV data -to be returned will contain such as: - SNDRV_CTL_TLVT_CONTAINER 96 - SNDRV_CTL_TLVT_CHMAP_FIXED 4 SNDRV_CHMAP_FC - SNDRV_CTL_TLVT_CHMAP_FIXED 8 SNDRV_CHMAP_FL SNDRV_CHMAP_FR - SNDRV_CTL_TLVT_CHMAP_FIXED 16 NDRV_CHMAP_FL SNDRV_CHMAP_FR \ - SNDRV_CHMAP_RL SNDRV_CHMAP_RR - -The channel position is provided in LSB 16bits. The upper bits are -used for bit flags. - -#define SNDRV_CHMAP_POSITION_MASK 0xffff -#define SNDRV_CHMAP_PHASE_INVERSE (0x01 << 16) -#define SNDRV_CHMAP_DRIVER_SPEC (0x02 << 16) - -SNDRV_CHMAP_PHASE_INVERSE indicates the channel is phase inverted, -(thus summing left and right channels would result in almost silence). -Some digital mic devices have this. - -When SNDRV_CHMAP_DRIVER_SPEC is set, all the channel position values -don't follow the standard definition above but driver-specific. - -* READ OPERATION - -The control read operation is for providing the current channel map of -the given stream. The control element returns an integer array -containing the position of each channel. - -When this is performed before the number of the channel is specified -(i.e. hw_params is set), it should return all channels set to -UNKNOWN. - -* WRITE OPERATION - -The control write operation is optional, and only for devices that can -change the channel configuration on the fly, such as HDMI. User needs -to pass an integer value containing the valid channel positions for -all channels of the assigned PCM substream. - -This operation is allowed only at PCM PREPARED state. When called in -other states, it shall return an error. diff --git a/Documentation/sound/designs/channel-mapping-api.rst b/Documentation/sound/designs/channel-mapping-api.rst new file mode 100644 index 000000000000..58e6312a43c0 --- /dev/null +++ b/Documentation/sound/designs/channel-mapping-api.rst @@ -0,0 +1,164 @@ +============================ +ALSA PCM channel-mapping API +============================ + +Takashi Iwai + +General +======= + +The channel mapping API allows user to query the possible channel maps +and the current channel map, also optionally to modify the channel map +of the current stream. + +A channel map is an array of position for each PCM channel. +Typically, a stereo PCM stream has a channel map of +``{ front_left, front_right }`` +while a 4.0 surround PCM stream has a channel map of +``{ front left, front right, rear left, rear right }.`` + +The problem, so far, was that we had no standard channel map +explicitly, and applications had no way to know which channel +corresponds to which (speaker) position. Thus, applications applied +wrong channels for 5.1 outputs, and you hear suddenly strange sound +from rear. Or, some devices secretly assume that center/LFE is the +third/fourth channels while others that C/LFE as 5th/6th channels. + +Also, some devices such as HDMI are configurable for different speaker +positions even with the same number of total channels. However, there +was no way to specify this because of lack of channel map +specification. These are the main motivations for the new channel +mapping API. + + +Design +====== + +Actually, "the channel mapping API" doesn't introduce anything new in +the kernel/user-space ABI perspective. It uses only the existing +control element features. + +As a ground design, each PCM substream may contain a control element +providing the channel mapping information and configuration. This +element is specified by: + +* iface = SNDRV_CTL_ELEM_IFACE_PCM +* name = "Playback Channel Map" or "Capture Channel Map" +* device = the same device number for the assigned PCM substream +* index = the same index number for the assigned PCM substream + +Note the name is different depending on the PCM substream direction. + +Each control element provides at least the TLV read operation and the +read operation. Optionally, the write operation can be provided to +allow user to change the channel map dynamically. + +TLV +--- + +The TLV operation gives the list of available channel +maps. A list item of a channel map is usually a TLV of +``type data-bytes ch0 ch1 ch2...`` +where type is the TLV type value, the second argument is the total +bytes (not the numbers) of channel values, and the rest are the +position value for each channel. + +As a TLV type, either ``SNDRV_CTL_TLVT_CHMAP_FIXED``, +``SNDRV_CTL_TLV_CHMAP_VAR`` or ``SNDRV_CTL_TLVT_CHMAP_PAIRED`` can be used. +The ``_FIXED`` type is for a channel map with the fixed channel position +while the latter two are for flexible channel positions. ``_VAR`` type is +for a channel map where all channels are freely swappable and ``_PAIRED`` +type is where pair-wise channels are swappable. For example, when you +have {FL/FR/RL/RR} channel map, ``_PAIRED`` type would allow you to swap +only {RL/RR/FL/FR} while ``_VAR`` type would allow even swapping FL and +RR. + +These new TLV types are defined in ``sound/tlv.h``. + +The available channel position values are defined in ``sound/asound.h``, +here is a cut: + +:: + + /* channel positions */ + enum { + SNDRV_CHMAP_UNKNOWN = 0, + SNDRV_CHMAP_NA, /* N/A, silent */ + SNDRV_CHMAP_MONO, /* mono stream */ + /* this follows the alsa-lib mixer channel value + 3 */ + SNDRV_CHMAP_FL, /* front left */ + SNDRV_CHMAP_FR, /* front right */ + SNDRV_CHMAP_RL, /* rear left */ + SNDRV_CHMAP_RR, /* rear right */ + SNDRV_CHMAP_FC, /* front center */ + SNDRV_CHMAP_LFE, /* LFE */ + SNDRV_CHMAP_SL, /* side left */ + SNDRV_CHMAP_SR, /* side right */ + SNDRV_CHMAP_RC, /* rear center */ + /* new definitions */ + SNDRV_CHMAP_FLC, /* front left center */ + SNDRV_CHMAP_FRC, /* front right center */ + SNDRV_CHMAP_RLC, /* rear left center */ + SNDRV_CHMAP_RRC, /* rear right center */ + SNDRV_CHMAP_FLW, /* front left wide */ + SNDRV_CHMAP_FRW, /* front right wide */ + SNDRV_CHMAP_FLH, /* front left high */ + SNDRV_CHMAP_FCH, /* front center high */ + SNDRV_CHMAP_FRH, /* front right high */ + SNDRV_CHMAP_TC, /* top center */ + SNDRV_CHMAP_TFL, /* top front left */ + SNDRV_CHMAP_TFR, /* top front right */ + SNDRV_CHMAP_TFC, /* top front center */ + SNDRV_CHMAP_TRL, /* top rear left */ + SNDRV_CHMAP_TRR, /* top rear right */ + SNDRV_CHMAP_TRC, /* top rear center */ + SNDRV_CHMAP_LAST = SNDRV_CHMAP_TRC, + }; + +When a PCM stream can provide more than one channel map, you can +provide multiple channel maps in a TLV container type. The TLV data +to be returned will contain such as: +:: + + SNDRV_CTL_TLVT_CONTAINER 96 + SNDRV_CTL_TLVT_CHMAP_FIXED 4 SNDRV_CHMAP_FC + SNDRV_CTL_TLVT_CHMAP_FIXED 8 SNDRV_CHMAP_FL SNDRV_CHMAP_FR + SNDRV_CTL_TLVT_CHMAP_FIXED 16 NDRV_CHMAP_FL SNDRV_CHMAP_FR \ + SNDRV_CHMAP_RL SNDRV_CHMAP_RR + +The channel position is provided in LSB 16bits. The upper bits are +used for bit flags. +:: + + #define SNDRV_CHMAP_POSITION_MASK 0xffff + #define SNDRV_CHMAP_PHASE_INVERSE (0x01 << 16) + #define SNDRV_CHMAP_DRIVER_SPEC (0x02 << 16) + +``SNDRV_CHMAP_PHASE_INVERSE`` indicates the channel is phase inverted, +(thus summing left and right channels would result in almost silence). +Some digital mic devices have this. + +When ``SNDRV_CHMAP_DRIVER_SPEC`` is set, all the channel position values +don't follow the standard definition above but driver-specific. + +Read Operation +-------------- + +The control read operation is for providing the current channel map of +the given stream. The control element returns an integer array +containing the position of each channel. + +When this is performed before the number of the channel is specified +(i.e. hw_params is set), it should return all channels set to +``UNKNOWN``. + +Write Operation +--------------- + +The control write operation is optional, and only for devices that can +change the channel configuration on the fly, such as HDMI. User needs +to pass an integer value containing the valid channel positions for +all channels of the assigned PCM substream. + +This operation is allowed only at PCM PREPARED state. When called in +other states, it shall return an error. diff --git a/Documentation/sound/designs/index.rst b/Documentation/sound/designs/index.rst index 362e1c23d51f..dd4fcbcb2b92 100644 --- a/Documentation/sound/designs/index.rst +++ b/Documentation/sound/designs/index.rst @@ -4,5 +4,6 @@ Designs and Implementations .. toctree:: :maxdepth: 2 + channel-mapping-api procfile powersave -- cgit v1.2.3-58-ga151 From 0c266c4b707e263ac74e0bd4330b4193fa46c434 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 9 Nov 2016 16:18:14 +0100 Subject: ALSA: doc: ReSTize OSS-Emulation document A simple conversion from a text file. Put to designs subdirectory. Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/OSS-Emulation.txt | 305 ----------------------- Documentation/sound/designs/index.rst | 1 + Documentation/sound/designs/oss-emulation.rst | 336 ++++++++++++++++++++++++++ 3 files changed, 337 insertions(+), 305 deletions(-) delete mode 100644 Documentation/sound/alsa/OSS-Emulation.txt create mode 100644 Documentation/sound/designs/oss-emulation.rst (limited to 'Documentation/sound') diff --git a/Documentation/sound/alsa/OSS-Emulation.txt b/Documentation/sound/alsa/OSS-Emulation.txt deleted file mode 100644 index 152ca2a3f1bd..000000000000 --- a/Documentation/sound/alsa/OSS-Emulation.txt +++ /dev/null @@ -1,305 +0,0 @@ - NOTES ON KERNEL OSS-EMULATION - ============================= - - Jan. 22, 2004 Takashi Iwai - - -Modules -======= - -ALSA provides a powerful OSS emulation on the kernel. -The OSS emulation for PCM, mixer and sequencer devices is implemented -as add-on kernel modules, snd-pcm-oss, snd-mixer-oss and snd-seq-oss. -When you need to access the OSS PCM, mixer or sequencer devices, the -corresponding module has to be loaded. - -These modules are loaded automatically when the corresponding service -is called. The alias is defined sound-service-x-y, where x and y are -the card number and the minor unit number. Usually you don't have to -define these aliases by yourself. - -Only necessary step for auto-loading of OSS modules is to define the -card alias in /etc/modprobe.d/alsa.conf, such as - - alias sound-slot-0 snd-emu10k1 - -As the second card, define sound-slot-1 as well. -Note that you can't use the aliased name as the target name (i.e. -"alias sound-slot-0 snd-card-0" doesn't work any more like the old -modutils). - -The currently available OSS configuration is shown in -/proc/asound/oss/sndstat. This shows in the same syntax of -/dev/sndstat, which is available on the commercial OSS driver. -On ALSA, you can symlink /dev/sndstat to this proc file. - -Please note that the devices listed in this proc file appear only -after the corresponding OSS-emulation module is loaded. Don't worry -even if "NOT ENABLED IN CONFIG" is shown in it. - - -Device Mapping -============== - -ALSA supports the following OSS device files: - - PCM: - /dev/dspX - /dev/adspX - - Mixer: - /dev/mixerX - - MIDI: - /dev/midi0X - /dev/amidi0X - - Sequencer: - /dev/sequencer - /dev/sequencer2 (aka /dev/music) - -where X is the card number from 0 to 7. - -(NOTE: Some distributions have the device files like /dev/midi0 and - /dev/midi1. They are NOT for OSS but for tclmidi, which is - a totally different thing.) - -Unlike the real OSS, ALSA cannot use the device files more than the -assigned ones. For example, the first card cannot use /dev/dsp1 or -/dev/dsp2, but only /dev/dsp0 and /dev/adsp0. - -As seen above, PCM and MIDI may have two devices. Usually, the first -PCM device (hw:0,0 in ALSA) is mapped to /dev/dsp and the secondary -device (hw:0,1) to /dev/adsp (if available). For MIDI, /dev/midi and -/dev/amidi, respectively. - -You can change this device mapping via the module options of -snd-pcm-oss and snd-rawmidi. In the case of PCM, the following -options are available for snd-pcm-oss: - - dsp_map PCM device number assigned to /dev/dspX - (default = 0) - adsp_map PCM device number assigned to /dev/adspX - (default = 1) - -For example, to map the third PCM device (hw:0,2) to /dev/adsp0, -define like this: - - options snd-pcm-oss adsp_map=2 - -The options take arrays. For configuring the second card, specify -two entries separated by comma. For example, to map the third PCM -device on the second card to /dev/adsp1, define like below: - - options snd-pcm-oss adsp_map=0,2 - -To change the mapping of MIDI devices, the following options are -available for snd-rawmidi: - - midi_map MIDI device number assigned to /dev/midi0X - (default = 0) - amidi_map MIDI device number assigned to /dev/amidi0X - (default = 1) - -For example, to assign the third MIDI device on the first card to -/dev/midi00, define as follows: - - options snd-rawmidi midi_map=2 - - -PCM Mode -======== - -As default, ALSA emulates the OSS PCM with so-called plugin layer, -i.e. tries to convert the sample format, rate or channels -automatically when the card doesn't support it natively. -This will lead to some problems for some applications like quake or -wine, especially if they use the card only in the MMAP mode. - -In such a case, you can change the behavior of PCM per application by -writing a command to the proc file. There is a proc file for each PCM -stream, /proc/asound/cardX/pcmY[cp]/oss, where X is the card number -(zero-based), Y the PCM device number (zero-based), and 'p' is for -playback and 'c' for capture, respectively. Note that this proc file -exists only after snd-pcm-oss module is loaded. - -The command sequence has the following syntax: - - app_name fragments fragment_size [options] - -app_name is the name of application with (higher priority) or without -path. -fragments specifies the number of fragments or zero if no specific -number is given. -fragment_size is the size of fragment in bytes or zero if not given. -options is the optional parameters. The following options are -available: - - disable the application tries to open a pcm device for - this channel but does not want to use it. - direct don't use plugins - block force block open mode - non-block force non-block open mode - partial-frag write also partial fragments (affects playback only) - no-silence do not fill silence ahead to avoid clicks - -The disable option is useful when one stream direction (playback or -capture) is not handled correctly by the application although the -hardware itself does support both directions. -The direct option is used, as mentioned above, to bypass the automatic -conversion and useful for MMAP-applications. -For example, to playback the first PCM device without plugins for -quake, send a command via echo like the following: - - % echo "quake 0 0 direct" > /proc/asound/card0/pcm0p/oss - -While quake wants only playback, you may append the second command -to notify driver that only this direction is about to be allocated: - - % echo "quake 0 0 disable" > /proc/asound/card0/pcm0c/oss - -The permission of proc files depend on the module options of snd. -As default it's set as root, so you'll likely need to be superuser for -sending the command above. - -The block and non-block options are used to change the behavior of -opening the device file. - -As default, ALSA behaves as original OSS drivers, i.e. does not block -the file when it's busy. The -EBUSY error is returned in this case. - -This blocking behavior can be changed globally via nonblock_open -module option of snd-pcm-oss. For using the blocking mode as default -for OSS devices, define like the following: - - options snd-pcm-oss nonblock_open=0 - -The partial-frag and no-silence commands have been added recently. -Both commands are for optimization use only. The former command -specifies to invoke the write transfer only when the whole fragment is -filled. The latter stops writing the silence data ahead -automatically. Both are disabled as default. - -You can check the currently defined configuration by reading the proc -file. The read image can be sent to the proc file again, hence you -can save the current configuration - - % cat /proc/asound/card0/pcm0p/oss > /somewhere/oss-cfg - -and restore it like - - % cat /somewhere/oss-cfg > /proc/asound/card0/pcm0p/oss - -Also, for clearing all the current configuration, send "erase" command -as below: - - % echo "erase" > /proc/asound/card0/pcm0p/oss - - -Mixer Elements -============== - -Since ALSA has completely different mixer interface, the emulation of -OSS mixer is relatively complicated. ALSA builds up a mixer element -from several different ALSA (mixer) controls based on the name -string. For example, the volume element SOUND_MIXER_PCM is composed -from "PCM Playback Volume" and "PCM Playback Switch" controls for the -playback direction and from "PCM Capture Volume" and "PCM Capture -Switch" for the capture directory (if exists). When the PCM volume of -OSS is changed, all the volume and switch controls above are adjusted -automatically. - -As default, ALSA uses the following control for OSS volumes: - - OSS volume ALSA control Index - ----------------------------------------------------- - SOUND_MIXER_VOLUME Master 0 - SOUND_MIXER_BASS Tone Control - Bass 0 - SOUND_MIXER_TREBLE Tone Control - Treble 0 - SOUND_MIXER_SYNTH Synth 0 - SOUND_MIXER_PCM PCM 0 - SOUND_MIXER_SPEAKER PC Speaker 0 - SOUND_MIXER_LINE Line 0 - SOUND_MIXER_MIC Mic 0 - SOUND_MIXER_CD CD 0 - SOUND_MIXER_IMIX Monitor Mix 0 - SOUND_MIXER_ALTPCM PCM 1 - SOUND_MIXER_RECLEV (not assigned) - SOUND_MIXER_IGAIN Capture 0 - SOUND_MIXER_OGAIN Playback 0 - SOUND_MIXER_LINE1 Aux 0 - SOUND_MIXER_LINE2 Aux 1 - SOUND_MIXER_LINE3 Aux 2 - SOUND_MIXER_DIGITAL1 Digital 0 - SOUND_MIXER_DIGITAL2 Digital 1 - SOUND_MIXER_DIGITAL3 Digital 2 - SOUND_MIXER_PHONEIN Phone 0 - SOUND_MIXER_PHONEOUT Phone 1 - SOUND_MIXER_VIDEO Video 0 - SOUND_MIXER_RADIO Radio 0 - SOUND_MIXER_MONITOR Monitor 0 - -The second column is the base-string of the corresponding ALSA -control. In fact, the controls with "XXX [Playback|Capture] -[Volume|Switch]" will be checked in addition. - -The current assignment of these mixer elements is listed in the proc -file, /proc/asound/cardX/oss_mixer, which will be like the following - - VOLUME "Master" 0 - BASS "" 0 - TREBLE "" 0 - SYNTH "" 0 - PCM "PCM" 0 - ... - -where the first column is the OSS volume element, the second column -the base-string of the corresponding ALSA control, and the third the -control index. When the string is empty, it means that the -corresponding OSS control is not available. - -For changing the assignment, you can write the configuration to this -proc file. For example, to map "Wave Playback" to the PCM volume, -send the command like the following: - - % echo 'VOLUME "Wave Playback" 0' > /proc/asound/card0/oss_mixer - -The command is exactly as same as listed in the proc file. You can -change one or more elements, one volume per line. In the last -example, both "Wave Playback Volume" and "Wave Playback Switch" will -be affected when PCM volume is changed. - -Like the case of PCM proc file, the permission of proc files depend on -the module options of snd. you'll likely need to be superuser for -sending the command above. - -As well as in the case of PCM proc file, you can save and restore the -current mixer configuration by reading and writing the whole file -image. - - -Duplex Streams -============== - -Note that when attempting to use a single device file for playback and -capture, the OSS API provides no way to set the format, sample rate or -number of channels different in each direction. Thus - io_handle = open("device", O_RDWR) -will only function correctly if the values are the same in each direction. - -To use different values in the two directions, use both - input_handle = open("device", O_RDONLY) - output_handle = open("device", O_WRONLY) -and set the values for the corresponding handle. - - -Unsupported Features -==================== - -MMAP on ICE1712 driver ----------------------- -ICE1712 supports only the unconventional format, interleaved -10-channels 24bit (packed in 32bit) format. Therefore you cannot mmap -the buffer as the conventional (mono or 2-channels, 8 or 16bit) format -on OSS. - diff --git a/Documentation/sound/designs/index.rst b/Documentation/sound/designs/index.rst index dd4fcbcb2b92..5b3033c556d0 100644 --- a/Documentation/sound/designs/index.rst +++ b/Documentation/sound/designs/index.rst @@ -7,3 +7,4 @@ Designs and Implementations channel-mapping-api procfile powersave + oss-emulation diff --git a/Documentation/sound/designs/oss-emulation.rst b/Documentation/sound/designs/oss-emulation.rst new file mode 100644 index 000000000000..e8dcb9633e7b --- /dev/null +++ b/Documentation/sound/designs/oss-emulation.rst @@ -0,0 +1,336 @@ +============================= +Notes on Kernel OSS-Emulation +============================= + +Jan. 22, 2004 Takashi Iwai + + +Modules +======= + +ALSA provides a powerful OSS emulation on the kernel. +The OSS emulation for PCM, mixer and sequencer devices is implemented +as add-on kernel modules, snd-pcm-oss, snd-mixer-oss and snd-seq-oss. +When you need to access the OSS PCM, mixer or sequencer devices, the +corresponding module has to be loaded. + +These modules are loaded automatically when the corresponding service +is called. The alias is defined ``sound-service-x-y``, where x and y are +the card number and the minor unit number. Usually you don't have to +define these aliases by yourself. + +Only necessary step for auto-loading of OSS modules is to define the +card alias in ``/etc/modprobe.d/alsa.conf``, such as:: + + alias sound-slot-0 snd-emu10k1 + +As the second card, define ``sound-slot-1`` as well. +Note that you can't use the aliased name as the target name (i.e. +``alias sound-slot-0 snd-card-0`` doesn't work any more like the old +modutils). + +The currently available OSS configuration is shown in +/proc/asound/oss/sndstat. This shows in the same syntax of +/dev/sndstat, which is available on the commercial OSS driver. +On ALSA, you can symlink /dev/sndstat to this proc file. + +Please note that the devices listed in this proc file appear only +after the corresponding OSS-emulation module is loaded. Don't worry +even if "NOT ENABLED IN CONFIG" is shown in it. + + +Device Mapping +============== + +ALSA supports the following OSS device files: +:: + + PCM: + /dev/dspX + /dev/adspX + + Mixer: + /dev/mixerX + + MIDI: + /dev/midi0X + /dev/amidi0X + + Sequencer: + /dev/sequencer + /dev/sequencer2 (aka /dev/music) + +where X is the card number from 0 to 7. + +(NOTE: Some distributions have the device files like /dev/midi0 and +/dev/midi1. They are NOT for OSS but for tclmidi, which is +a totally different thing.) + +Unlike the real OSS, ALSA cannot use the device files more than the +assigned ones. For example, the first card cannot use /dev/dsp1 or +/dev/dsp2, but only /dev/dsp0 and /dev/adsp0. + +As seen above, PCM and MIDI may have two devices. Usually, the first +PCM device (``hw:0,0`` in ALSA) is mapped to /dev/dsp and the secondary +device (``hw:0,1``) to /dev/adsp (if available). For MIDI, /dev/midi and +/dev/amidi, respectively. + +You can change this device mapping via the module options of +snd-pcm-oss and snd-rawmidi. In the case of PCM, the following +options are available for snd-pcm-oss: + +dsp_map + PCM device number assigned to /dev/dspX + (default = 0) +adsp_map + PCM device number assigned to /dev/adspX + (default = 1) + +For example, to map the third PCM device (``hw:0,2``) to /dev/adsp0, +define like this: +:: + + options snd-pcm-oss adsp_map=2 + +The options take arrays. For configuring the second card, specify +two entries separated by comma. For example, to map the third PCM +device on the second card to /dev/adsp1, define like below: +:: + + options snd-pcm-oss adsp_map=0,2 + +To change the mapping of MIDI devices, the following options are +available for snd-rawmidi: + +midi_map + MIDI device number assigned to /dev/midi0X + (default = 0) +amidi_map + MIDI device number assigned to /dev/amidi0X + (default = 1) + +For example, to assign the third MIDI device on the first card to +/dev/midi00, define as follows: +:: + + options snd-rawmidi midi_map=2 + + +PCM Mode +======== + +As default, ALSA emulates the OSS PCM with so-called plugin layer, +i.e. tries to convert the sample format, rate or channels +automatically when the card doesn't support it natively. +This will lead to some problems for some applications like quake or +wine, especially if they use the card only in the MMAP mode. + +In such a case, you can change the behavior of PCM per application by +writing a command to the proc file. There is a proc file for each PCM +stream, ``/proc/asound/cardX/pcmY[cp]/oss``, where X is the card number +(zero-based), Y the PCM device number (zero-based), and ``p`` is for +playback and ``c`` for capture, respectively. Note that this proc file +exists only after snd-pcm-oss module is loaded. + +The command sequence has the following syntax: +:: + + app_name fragments fragment_size [options] + +``app_name`` is the name of application with (higher priority) or without +path. +``fragments`` specifies the number of fragments or zero if no specific +number is given. +``fragment_size`` is the size of fragment in bytes or zero if not given. +``options`` is the optional parameters. The following options are +available: + +disable + the application tries to open a pcm device for + this channel but does not want to use it. +direct + don't use plugins +block + force block open mode +non-block + force non-block open mode +partial-frag + write also partial fragments (affects playback only) +no-silence + do not fill silence ahead to avoid clicks + +The ``disable`` option is useful when one stream direction (playback or +capture) is not handled correctly by the application although the +hardware itself does support both directions. +The ``direct`` option is used, as mentioned above, to bypass the automatic +conversion and useful for MMAP-applications. +For example, to playback the first PCM device without plugins for +quake, send a command via echo like the following: +:: + + % echo "quake 0 0 direct" > /proc/asound/card0/pcm0p/oss + +While quake wants only playback, you may append the second command +to notify driver that only this direction is about to be allocated: +:: + + % echo "quake 0 0 disable" > /proc/asound/card0/pcm0c/oss + +The permission of proc files depend on the module options of snd. +As default it's set as root, so you'll likely need to be superuser for +sending the command above. + +The block and non-block options are used to change the behavior of +opening the device file. + +As default, ALSA behaves as original OSS drivers, i.e. does not block +the file when it's busy. The -EBUSY error is returned in this case. + +This blocking behavior can be changed globally via nonblock_open +module option of snd-pcm-oss. For using the blocking mode as default +for OSS devices, define like the following: +:: + + options snd-pcm-oss nonblock_open=0 + +The ``partial-frag`` and ``no-silence`` commands have been added recently. +Both commands are for optimization use only. The former command +specifies to invoke the write transfer only when the whole fragment is +filled. The latter stops writing the silence data ahead +automatically. Both are disabled as default. + +You can check the currently defined configuration by reading the proc +file. The read image can be sent to the proc file again, hence you +can save the current configuration +:: + + % cat /proc/asound/card0/pcm0p/oss > /somewhere/oss-cfg + +and restore it like +:: + + % cat /somewhere/oss-cfg > /proc/asound/card0/pcm0p/oss + +Also, for clearing all the current configuration, send ``erase`` command +as below: +:: + + % echo "erase" > /proc/asound/card0/pcm0p/oss + + +Mixer Elements +============== + +Since ALSA has completely different mixer interface, the emulation of +OSS mixer is relatively complicated. ALSA builds up a mixer element +from several different ALSA (mixer) controls based on the name +string. For example, the volume element SOUND_MIXER_PCM is composed +from "PCM Playback Volume" and "PCM Playback Switch" controls for the +playback direction and from "PCM Capture Volume" and "PCM Capture +Switch" for the capture directory (if exists). When the PCM volume of +OSS is changed, all the volume and switch controls above are adjusted +automatically. + +As default, ALSA uses the following control for OSS volumes: + +==================== ===================== ===== +OSS volume ALSA control Index +==================== ===================== ===== +SOUND_MIXER_VOLUME Master 0 +SOUND_MIXER_BASS Tone Control - Bass 0 +SOUND_MIXER_TREBLE Tone Control - Treble 0 +SOUND_MIXER_SYNTH Synth 0 +SOUND_MIXER_PCM PCM 0 +SOUND_MIXER_SPEAKER PC Speaker 0 +SOUND_MIXER_LINE Line 0 +SOUND_MIXER_MIC Mic 0 +SOUND_MIXER_CD CD 0 +SOUND_MIXER_IMIX Monitor Mix 0 +SOUND_MIXER_ALTPCM PCM 1 +SOUND_MIXER_RECLEV (not assigned) +SOUND_MIXER_IGAIN Capture 0 +SOUND_MIXER_OGAIN Playback 0 +SOUND_MIXER_LINE1 Aux 0 +SOUND_MIXER_LINE2 Aux 1 +SOUND_MIXER_LINE3 Aux 2 +SOUND_MIXER_DIGITAL1 Digital 0 +SOUND_MIXER_DIGITAL2 Digital 1 +SOUND_MIXER_DIGITAL3 Digital 2 +SOUND_MIXER_PHONEIN Phone 0 +SOUND_MIXER_PHONEOUT Phone 1 +SOUND_MIXER_VIDEO Video 0 +SOUND_MIXER_RADIO Radio 0 +SOUND_MIXER_MONITOR Monitor 0 +==================== ===================== ===== + +The second column is the base-string of the corresponding ALSA +control. In fact, the controls with ``XXX [Playback|Capture] +[Volume|Switch]`` will be checked in addition. + +The current assignment of these mixer elements is listed in the proc +file, /proc/asound/cardX/oss_mixer, which will be like the following +:: + + VOLUME "Master" 0 + BASS "" 0 + TREBLE "" 0 + SYNTH "" 0 + PCM "PCM" 0 + ... + +where the first column is the OSS volume element, the second column +the base-string of the corresponding ALSA control, and the third the +control index. When the string is empty, it means that the +corresponding OSS control is not available. + +For changing the assignment, you can write the configuration to this +proc file. For example, to map "Wave Playback" to the PCM volume, +send the command like the following: +:: + + % echo 'VOLUME "Wave Playback" 0' > /proc/asound/card0/oss_mixer + +The command is exactly as same as listed in the proc file. You can +change one or more elements, one volume per line. In the last +example, both "Wave Playback Volume" and "Wave Playback Switch" will +be affected when PCM volume is changed. + +Like the case of PCM proc file, the permission of proc files depend on +the module options of snd. you'll likely need to be superuser for +sending the command above. + +As well as in the case of PCM proc file, you can save and restore the +current mixer configuration by reading and writing the whole file +image. + + +Duplex Streams +============== + +Note that when attempting to use a single device file for playback and +capture, the OSS API provides no way to set the format, sample rate or +number of channels different in each direction. Thus +:: + + io_handle = open("device", O_RDWR) + +will only function correctly if the values are the same in each direction. + +To use different values in the two directions, use both +:: + + input_handle = open("device", O_RDONLY) + output_handle = open("device", O_WRONLY) + +and set the values for the corresponding handle. + + +Unsupported Features +==================== + +MMAP on ICE1712 driver +---------------------- +ICE1712 supports only the unconventional format, interleaved +10-channels 24bit (packed in 32bit) format. Therefore you cannot mmap +the buffer as the conventional (mono or 2-channels, 8 or 16bit) format +on OSS. -- cgit v1.2.3-58-ga151 From 07aecc06eb32ff0010dff69438806b6f200fc1fd Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 9 Nov 2016 16:46:02 +0100 Subject: ALSA: doc: ReSTize seq_oss document Converted from an ancient plain HTML document. It's much readable now! Put to designs subdirectory with a slight rename. Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/seq_oss.html | 409 -------------------------------- Documentation/sound/designs/index.rst | 1 + Documentation/sound/designs/seq-oss.rst | 371 +++++++++++++++++++++++++++++ 3 files changed, 372 insertions(+), 409 deletions(-) delete mode 100644 Documentation/sound/alsa/seq_oss.html create mode 100644 Documentation/sound/designs/seq-oss.rst (limited to 'Documentation/sound') diff --git a/Documentation/sound/alsa/seq_oss.html b/Documentation/sound/alsa/seq_oss.html deleted file mode 100644 index 9663b45f6fde..000000000000 --- a/Documentation/sound/alsa/seq_oss.html +++ /dev/null @@ -1,409 +0,0 @@ - - - - OSS Sequencer Emulation on ALSA - - - -
-

- -

- -
-

-OSS Sequencer Emulation on ALSA

- -
-

Copyright (c) 1998,1999 by Takashi Iwai -<iwai@ww.uni-erlangen.de> -

ver.0.1.8; Nov. 16, 1999 -

- -

- -

-1. Description

-This directory contains the OSS sequencer emulation driver on ALSA. Note -that this program is still in the development state. -

What this does - it provides the emulation of the OSS sequencer, access -via -/dev/sequencer and /dev/music devices. -The most of applications using OSS can run if the appropriate ALSA -sequencer is prepared. -

The following features are emulated by this driver: -

    -
  • -Normal sequencer and MIDI events:
    • - -
    They are converted to the ALSA sequencer events, and sent to the corresponding -port. -
  • -Timer events:
    • - -
    The timer is not selectable by ioctl. The control rate is fixed to -100 regardless of HZ. That is, even on Alpha system, a tick is always -1/100 second. The base rate and tempo can be changed in /dev/music. - -
  • -Patch loading:
    • - -
    It purely depends on the synth drivers whether it's supported since -the patch loading is realized by callback to the synth driver. -
  • -I/O controls:
    • - -
    Most of controls are accepted. Some controls -are dependent on the synth driver, as well as even on original OSS.
-Furthermore, you can find the following advanced features: -
    -
  • -Better queue mechanism:
    • - -
    The events are queued before processing them. -
  • -Multiple applications:
    • - -
    You can run two or more applications simultaneously (even for OSS sequencer)! -However, each MIDI device is exclusive - that is, if a MIDI device is opened -once by some application, other applications can't use it. No such a restriction -in synth devices. -
  • -Real-time event processing:
    • - -
    The events can be processed in real time without using out of bound -ioctl. To switch to real-time mode, send ABSTIME 0 event. The followed -events will be processed in real-time without queued. To switch off the -real-time mode, send RELTIME 0 event. -
  • -/proc interface:
    • - -
    The status of applications and devices can be shown via /proc/asound/seq/oss -at any time. In the later version, configuration will be changed via /proc -interface, too.
- -

-2. Installation

-Run configure script with both sequencer support (--with-sequencer=yes) -and OSS emulation (--with-oss=yes) options. A module snd-seq-oss.o -will be created. If the synth module of your sound card supports for OSS -emulation (so far, only Emu8000 driver), this module will be loaded automatically. -Otherwise, you need to load this module manually. -

At beginning, this module probes all the MIDI ports which have been -already connected to the sequencer. Once after that, the creation and deletion -of ports are watched by announcement mechanism of ALSA sequencer. -

The available synth and MIDI devices can be found in proc interface. -Run "cat /proc/asound/seq/oss", and check the devices. For example, -if you use an AWE64 card, you'll see like the following: -

        OSS sequencer emulation version 0.1.8
-        ALSA client number 63
-        ALSA receiver port 0
-
-        Number of applications: 0
-
-        Number of synth devices: 1
-
-        synth 0: [EMU8000]
-          type 0x1 : subtype 0x20 : voices 32
-          capabilties : ioctl enabled / load_patch enabled
-
-        Number of MIDI devices: 3
-
-        midi 0: [Emu8000 Port-0] ALSA port 65:0
-          capability write / opened none
-
-        midi 1: [Emu8000 Port-1] ALSA port 65:1
-          capability write / opened none
-
-        midi 2: [0: MPU-401 (UART)] ALSA port 64:0
-          capability read/write / opened none
-Note that the device number may be different from the information of -/proc/asound/oss-devices -or ones of the original OSS driver. Use the device number listed in /proc/asound/seq/oss -to play via OSS sequencer emulation. -

-3. Using Synthesizer Devices

-Run your favorite program. I've tested playmidi-2.4, awemidi-0.4.3, gmod-3.1 -and xmp-1.1.5. You can load samples via /dev/sequencer like sfxload, -too. -

If the lowlevel driver supports multiple access to synth devices (like -Emu8000 driver), two or more applications are allowed to run at the same -time. -

-4. Using MIDI Devices

-So far, only MIDI output was tested. MIDI input was not checked at all, -but hopefully it will work. Use the device number listed in /proc/asound/seq/oss. -Be aware that these numbers are mostly different from the list in -/proc/asound/oss-devices. -

-5. Module Options

-The following module options are available: -
    -
  • -maxqlen
    • - -
    specifies the maximum read/write queue length. This queue is private -for OSS sequencer, so that it is independent from the queue length of ALSA -sequencer. Default value is 1024. -
  • -seq_oss_debug
    • - -
    specifies the debug level and accepts zero (= no debug message) or -positive integer. Default value is 0.
- -

-6. Queue Mechanism

-OSS sequencer emulation uses an ALSA priority queue. The -events from /dev/sequencer are processed and put onto the queue -specified by module option. -

All the events from /dev/sequencer are parsed at beginning. -The timing events are also parsed at this moment, so that the events may -be processed in real-time. Sending an event ABSTIME 0 switches the operation -mode to real-time mode, and sending an event RELTIME 0 switches it off. -In the real-time mode, all events are dispatched immediately. -

The queued events are dispatched to the corresponding ALSA sequencer -ports after scheduled time by ALSA sequencer dispatcher. -

If the write-queue is full, the application sleeps until a certain amount -(as default one half) becomes empty in blocking mode. The synchronization -to write timing was implemented, too. -

The input from MIDI devices or echo-back events are stored on read FIFO -queue. If application reads /dev/sequencer in blocking mode, the -process will be awaked. - -

-7. Interface to Synthesizer Device

- -

-7.1. Registration

-To register an OSS synthesizer device, use snd_seq_oss_synth_register -function. -
int snd_seq_oss_synth_register(char *name, int type, int subtype, int nvoices,
-                              snd_seq_oss_callback_t *oper, void *private_data)
-The arguments name, type, subtype and -nvoices -are used for making the appropriate synth_info structure for ioctl. The -return value is an index number of this device. This index must be remembered -for unregister. If registration is failed, -errno will be returned. -

To release this device, call snd_seq_oss_synth_unregister function: -

int snd_seq_oss_synth_unregister(int index),
-where the index is the index number returned by register function. -

-7.2. Callbacks

-OSS synthesizer devices have capability for sample downloading and ioctls -like sample reset. In OSS emulation, these special features are realized -by using callbacks. The registration argument oper is used to specify these -callbacks. The following callback functions must be defined: -
snd_seq_oss_callback_t:
-        int (*open)(snd_seq_oss_arg_t *p, void *closure);
-        int (*close)(snd_seq_oss_arg_t *p);
-        int (*ioctl)(snd_seq_oss_arg_t *p, unsigned int cmd, unsigned long arg);
-        int (*load_patch)(snd_seq_oss_arg_t *p, int format, const char *buf, int offs, int count);
-        int (*reset)(snd_seq_oss_arg_t *p);
-Except for open and close callbacks, they are allowed
-to be NULL.
-
Each callback function takes the argument type snd_seq_oss_arg_t as the
-first argument.
-
struct snd_seq_oss_arg_t {
-        int app_index;
-        int file_mode;
-        int seq_mode;
-        snd_seq_addr_t addr;
-        void *private_data;
-        int event_passing;
-};

-The first three fields, app_index, file_mode and
-seq_mode
-are initialized by OSS sequencer. The app_index is the application
-index which is unique to each application opening OSS sequencer. The
-file_mode
-is bit-flags indicating the file operation mode. See
-seq_oss.h
-for its meaning. The seq_mode is sequencer operation mode. In
-the current version, only SND_OSSSEQ_MODE_SYNTH is used.
-
The next two fields, addr and private_data, must be
-filled by the synth driver at open callback. The addr contains
-the address of ALSA sequencer port which is assigned to this device. If
-the driver allocates memory for private_data, it must be released
-in close callback by itself.
-
The last field, event_passing, indicates how to translate note-on
-/ off events. In PROCESS_EVENTS mode, the note 255 is regarded
-as velocity change, and key pressure event is passed to the port. In PASS_EVENTS
-mode, all note on/off events are passed to the port without modified. PROCESS_KEYPRESS
-mode checks the note above 128 and regards it as key pressure event (mainly
-for Emu8000 driver).
-

-7.2.1. Open Callback

-The open is called at each time this device is opened by an application
-using OSS sequencer. This must not be NULL. Typically, the open callback
-does the following procedure:
-
    
-
  1. 
-Allocate private data record.
    2. 
-
-
  2. 
-Create an ALSA sequencer port.
    3. 
-
-
  3. 
-Set the new port address on arg->addr.
    4. 
-
-
  4. 
-Set the private data record pointer on arg->private_data.
    5. 
-

-Note that the type bit-flags in port_info of this synth port must NOT contain
-TYPE_MIDI_GENERIC
-bit. Instead, TYPE_SPECIFIC should be used. Also, CAP_SUBSCRIPTION
-bit should NOT be included, too. This is necessary to tell it from other
-normal MIDI devices. If the open procedure succeeded, return zero. Otherwise,
-return -errno.
-

-7.2.2 Ioctl Callback

-The ioctl callback is called when the sequencer receives device-specific
-ioctls. The following two ioctls should be processed by this callback:
-
    
-
  • 
-IOCTL_SEQ_RESET_SAMPLES
    • 
-
-
    reset all samples on memory -- return 0
-
  • 
-IOCTL_SYNTH_MEMAVL
    • 
-
-
    return the available memory size
-
  • 
-FM_4OP_ENABLE
    • 
-
-
    can be ignored usually

-The other ioctls are processed inside the sequencer without passing to
-the lowlevel driver.
-

-7.2.3 Load_Patch Callback

-The load_patch callback is used for sample-downloading. This callback
-must read the data on user-space and transfer to each device. Return 0
-if succeeded, and -errno if failed. The format argument is the patch key
-in patch_info record. The buf is user-space pointer where patch_info record
-is stored. The offs can be ignored. The count is total data size of this
-sample data.
-

-7.2.4 Close Callback

-The close callback is called when this device is closed by the
-application. If any private data was allocated in open callback, it must
-be released in the close callback. The deletion of ALSA port should be
-done here, too. This callback must not be NULL.
-

-7.2.5 Reset Callback

-The reset callback is called when sequencer device is reset or
-closed by applications. The callback should turn off the sounds on the
-relevant port immediately, and initialize the status of the port. If this
-callback is undefined, OSS seq sends a HEARTBEAT event to the
-port.
-

-7.3 Events

-Most of the events are processed by sequencer and translated to the adequate
-ALSA sequencer events, so that each synth device can receive by input_event
-callback of ALSA sequencer port. The following ALSA events should be implemented
-by the driver:
-
 
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
ALSA eventOriginal OSS events
NOTEONSEQ_NOTEON
-
MIDI_NOTEON
NOTESEQ_NOTEOFF
-
MIDI_NOTEOFF
KEYPRESSMIDI_KEY_PRESSURE
CHANPRESSSEQ_AFTERTOUCH
-
MIDI_CHN_PRESSURE
PGMCHANGESEQ_PGMCHANGE
-
MIDI_PGM_CHANGE
PITCHBENDSEQ_CONTROLLER(CTRL_PITCH_BENDER)
-
MIDI_PITCH_BEND
CONTROLLERMIDI_CTL_CHANGE
-
SEQ_BALANCE (with CTL_PAN)
CONTROL14SEQ_CONTROLLER
REGPARAMSEQ_CONTROLLER(CTRL_PITCH_BENDER_RANGE)
SYSEXSEQ_SYSEX

-
-
The most of these behavior can be realized by MIDI emulation driver
-included in the Emu8000 lowlevel driver. In the future release, this module
-will be independent.
-
Some OSS events (SEQ_PRIVATE and SEQ_VOLUME events) are passed as event
-type SND_SEQ_OSS_PRIVATE.  The OSS sequencer passes these event 8 byte
-packets without any modification. The lowlevel driver should process these
-events appropriately.
-

-8. Interface to MIDI Device

-Since the OSS emulation probes the creation and deletion of ALSA MIDI sequencer
-ports automatically by receiving announcement from ALSA sequencer, the
-MIDI devices don't need to be registered explicitly like synth devices.
-However, the MIDI port_info registered to ALSA sequencer must include a group
-name SND_SEQ_GROUP_DEVICE and a capability-bit CAP_READ or
-CAP_WRITE. Also, subscription capabilities, CAP_SUBS_READ or CAP_SUBS_WRITE,
-must be defined, too. If these conditions are not satisfied, the port is not
-registered as OSS sequencer MIDI device.
-
The events via MIDI devices are parsed in OSS sequencer and converted
-to the corresponding ALSA sequencer events. The input from MIDI sequencer
-is also converted to MIDI byte events by OSS sequencer. This works just
-a reverse way of seq_midi module.
-

-9. Known Problems / TODO's

-
-
    
-
  • 
-Patch loading via ALSA instrument layer is not implemented yet.
    • 
-

-
-
-
diff --git a/Documentation/sound/designs/index.rst b/Documentation/sound/designs/index.rst
index 5b3033c556d0..0ca3a6b0a5ce 100644
--- a/Documentation/sound/designs/index.rst
+++ b/Documentation/sound/designs/index.rst
@@ -8,3 +8,4 @@ Designs and Implementations
    procfile
    powersave
    oss-emulation
+   seq-oss
diff --git a/Documentation/sound/designs/seq-oss.rst b/Documentation/sound/designs/seq-oss.rst
new file mode 100644
index 000000000000..e82ffe0e7f43
--- /dev/null
+++ b/Documentation/sound/designs/seq-oss.rst
@@ -0,0 +1,371 @@
+===============================
+OSS Sequencer Emulation on ALSA
+===============================
+
+Copyright (c) 1998,1999 by Takashi Iwai
+
+ver.0.1.8; Nov. 16, 1999
+
+Description
+===========
+
+This directory contains the OSS sequencer emulation driver on ALSA. Note
+that this program is still in the development state.
+
+What this does - it provides the emulation of the OSS sequencer, access
+via ``/dev/sequencer`` and ``/dev/music`` devices.
+The most of applications using OSS can run if the appropriate ALSA
+sequencer is prepared.
+
+The following features are emulated by this driver:
+
+* Normal sequencer and MIDI events:
+
+    They are converted to the ALSA sequencer events, and sent to the
+    corresponding port.
+
+* Timer events:
+
+    The timer is not selectable by ioctl. The control rate is fixed to
+    100 regardless of HZ. That is, even on Alpha system, a tick is always
+    1/100 second. The base rate and tempo can be changed in ``/dev/music``.
+
+* Patch loading:
+
+    It purely depends on the synth drivers whether it's supported since
+    the patch loading is realized by callback to the synth driver.
+
+* I/O controls:
+
+    Most of controls are accepted. Some controls
+    are dependent on the synth driver, as well as even on original OSS.
+
+Furthermore, you can find the following advanced features:
+
+* Better queue mechanism:
+
+    The events are queued before processing them.
+
+* Multiple applications:
+
+    You can run two or more applications simultaneously (even for OSS
+    sequencer)!
+    However, each MIDI device is exclusive - that is, if a MIDI device
+    is opened once by some application, other applications can't use
+    it. No such a restriction in synth devices.
+
+* Real-time event processing:
+
+    The events can be processed in real time without using out of bound
+    ioctl. To switch to real-time mode, send ABSTIME 0 event. The followed
+    events will be processed in real-time without queued. To switch off the
+    real-time mode, send RELTIME 0 event.
+
+* ``/proc`` interface:
+
+    The status of applications and devices can be shown via
+    ``/proc/asound/seq/oss`` at any time. In the later version,
+    configuration will be changed via ``/proc`` interface, too.
+
+
+Installation
+============
+
+Run configure script with both sequencer support (``--with-sequencer=yes``)
+and OSS emulation (``--with-oss=yes``) options. A module ``snd-seq-oss.o``
+will be created. If the synth module of your sound card supports for OSS
+emulation (so far, only Emu8000 driver), this module will be loaded
+automatically.
+Otherwise, you need to load this module manually.
+
+At beginning, this module probes all the MIDI ports which have been
+already connected to the sequencer. Once after that, the creation and deletion
+of ports are watched by announcement mechanism of ALSA sequencer.
+
+The available synth and MIDI devices can be found in proc interface.
+Run ``cat /proc/asound/seq/oss``, and check the devices. For example,
+if you use an AWE64 card, you'll see like the following:
+::
+
+    OSS sequencer emulation version 0.1.8
+    ALSA client number 63
+    ALSA receiver port 0
+
+    Number of applications: 0
+
+    Number of synth devices: 1
+    synth 0: [EMU8000]
+      type 0x1 : subtype 0x20 : voices 32
+      capabilties : ioctl enabled / load_patch enabled
+
+    Number of MIDI devices: 3
+    midi 0: [Emu8000 Port-0] ALSA port 65:0
+      capability write / opened none
+
+    midi 1: [Emu8000 Port-1] ALSA port 65:1
+      capability write / opened none
+
+    midi 2: [0: MPU-401 (UART)] ALSA port 64:0
+      capability read/write / opened none
+
+Note that the device number may be different from the information of
+``/proc/asound/oss-devices`` or ones of the original OSS driver.
+Use the device number listed in ``/proc/asound/seq/oss``
+to play via OSS sequencer emulation.
+
+Using Synthesizer Devices
+=========================
+
+Run your favorite program. I've tested playmidi-2.4, awemidi-0.4.3, gmod-3.1
+and xmp-1.1.5. You can load samples via ``/dev/sequencer`` like sfxload,
+too.
+
+If the lowlevel driver supports multiple access to synth devices (like
+Emu8000 driver), two or more applications are allowed to run at the same
+time.
+
+Using MIDI Devices
+==================
+
+So far, only MIDI output was tested. MIDI input was not checked at all,
+but hopefully it will work. Use the device number listed in
+``/proc/asound/seq/oss``.
+Be aware that these numbers are mostly different from the list in
+``/proc/asound/oss-devices``.
+
+Module Options
+==============
+
+The following module options are available:
+
+maxqlen
+  specifies the maximum read/write queue length. This queue is private
+  for OSS sequencer, so that it is independent from the queue length of ALSA
+  sequencer. Default value is 1024.
+
+seq_oss_debug
+  specifies the debug level and accepts zero (= no debug message) or
+  positive integer. Default value is 0.
+
+Queue Mechanism
+===============
+
+OSS sequencer emulation uses an ALSA priority queue. The
+events from ``/dev/sequencer`` are processed and put onto the queue
+specified by module option.
+
+All the events from ``/dev/sequencer`` are parsed at beginning.
+The timing events are also parsed at this moment, so that the events may
+be processed in real-time. Sending an event ABSTIME 0 switches the operation
+mode to real-time mode, and sending an event RELTIME 0 switches it off.
+In the real-time mode, all events are dispatched immediately.
+
+The queued events are dispatched to the corresponding ALSA sequencer
+ports after scheduled time by ALSA sequencer dispatcher.
+
+If the write-queue is full, the application sleeps until a certain amount
+(as default one half) becomes empty in blocking mode. The synchronization
+to write timing was implemented, too.
+
+The input from MIDI devices or echo-back events are stored on read FIFO
+queue. If application reads ``/dev/sequencer`` in blocking mode, the
+process will be awaked.
+
+Interface to Synthesizer Device
+===============================
+
+Registration
+------------
+
+To register an OSS synthesizer device, use snd_seq_oss_synth_register()
+function:
+::
+
+  int snd_seq_oss_synth_register(char *name, int type, int subtype, int nvoices,
+          snd_seq_oss_callback_t *oper, void *private_data)
+
+The arguments ``name``, ``type``, ``subtype`` and ``nvoices``
+are used for making the appropriate synth_info structure for ioctl. The
+return value is an index number of this device. This index must be remembered
+for unregister. If registration is failed, -errno will be returned.
+
+To release this device, call snd_seq_oss_synth_unregister() function:
+::
+
+  int snd_seq_oss_synth_unregister(int index)
+
+where the ``index`` is the index number returned by register function.
+
+Callbacks
+---------
+
+OSS synthesizer devices have capability for sample downloading and ioctls
+like sample reset. In OSS emulation, these special features are realized
+by using callbacks. The registration argument oper is used to specify these
+callbacks. The following callback functions must be defined:
+::
+
+  snd_seq_oss_callback_t:
+   int (*open)(snd_seq_oss_arg_t *p, void *closure);
+   int (*close)(snd_seq_oss_arg_t *p);
+   int (*ioctl)(snd_seq_oss_arg_t *p, unsigned int cmd, unsigned long arg);
+   int (*load_patch)(snd_seq_oss_arg_t *p, int format, const char *buf, int offs, int count);
+   int (*reset)(snd_seq_oss_arg_t *p);
+
+Except for ``open`` and ``close`` callbacks, they are allowed to be NULL.
+
+Each callback function takes the argument type ``snd_seq_oss_arg_t`` as the
+first argument.
+::
+
+  struct snd_seq_oss_arg_t {
+      int app_index;
+      int file_mode;
+      int seq_mode;
+      snd_seq_addr_t addr;
+      void *private_data;
+      int event_passing;
+  };
+
+The first three fields, ``app_index``, ``file_mode`` and ``seq_mode``
+are initialized by OSS sequencer. The ``app_index`` is the application
+index which is unique to each application opening OSS sequencer. The
+``file_mode`` is bit-flags indicating the file operation mode. See
+``seq_oss.h`` for its meaning. The ``seq_mode`` is sequencer operation
+mode. In the current version, only ``SND_OSSSEQ_MODE_SYNTH`` is used.
+
+The next two fields, ``addr`` and ``private_data``, must be
+filled by the synth driver at open callback. The ``addr`` contains
+the address of ALSA sequencer port which is assigned to this device. If
+the driver allocates memory for ``private_data``, it must be released
+in close callback by itself.
+
+The last field, ``event_passing``, indicates how to translate note-on
+/ off events. In ``PROCESS_EVENTS`` mode, the note 255 is regarded
+as velocity change, and key pressure event is passed to the port. In
+``PASS_EVENTS`` mode, all note on/off events are passed to the port
+without modified. ``PROCESS_KEYPRESS`` mode checks the note above 128
+and regards it as key pressure event (mainly for Emu8000 driver).
+
+Open Callback
+-------------
+
+The ``open`` is called at each time this device is opened by an application
+using OSS sequencer. This must not be NULL. Typically, the open callback
+does the following procedure:
+
+#. Allocate private data record.
+#. Create an ALSA sequencer port.
+#. Set the new port address on ``arg->addr``.
+#. Set the private data record pointer on ``arg->private_data``.
+
+Note that the type bit-flags in port_info of this synth port must NOT contain
+``TYPE_MIDI_GENERIC``
+bit. Instead, ``TYPE_SPECIFIC`` should be used. Also, ``CAP_SUBSCRIPTION``
+bit should NOT be included, too. This is necessary to tell it from other
+normal MIDI devices. If the open procedure succeeded, return zero. Otherwise,
+return -errno.
+
+Ioctl Callback
+--------------
+
+The ``ioctl`` callback is called when the sequencer receives device-specific
+ioctls. The following two ioctls should be processed by this callback:
+
+IOCTL_SEQ_RESET_SAMPLES
+    reset all samples on memory -- return 0
+
+IOCTL_SYNTH_MEMAVL
+    return the available memory size
+
+FM_4OP_ENABLE
+    can be ignored usually
+
+The other ioctls are processed inside the sequencer without passing to
+the lowlevel driver.
+
+Load_Patch Callback
+-------------------
+
+The ``load_patch`` callback is used for sample-downloading. This callback
+must read the data on user-space and transfer to each device. Return 0
+if succeeded, and -errno if failed. The format argument is the patch key
+in patch_info record. The buf is user-space pointer where patch_info record
+is stored. The offs can be ignored. The count is total data size of this
+sample data.
+
+Close Callback
+--------------
+
+The ``close`` callback is called when this device is closed by the
+application. If any private data was allocated in open callback, it must
+be released in the close callback. The deletion of ALSA port should be
+done here, too. This callback must not be NULL.
+
+Reset Callback
+--------------
+
+The ``reset`` callback is called when sequencer device is reset or
+closed by applications. The callback should turn off the sounds on the
+relevant port immediately, and initialize the status of the port. If this
+callback is undefined, OSS seq sends a ``HEARTBEAT`` event to the
+port.
+
+Events
+======
+
+Most of the events are processed by sequencer and translated to the adequate
+ALSA sequencer events, so that each synth device can receive by input_event
+callback of ALSA sequencer port. The following ALSA events should be
+implemented by the driver:
+
+=============	===================
+ALSA event	Original OSS events
+=============	===================
+NOTEON		SEQ_NOTEON, MIDI_NOTEON
+NOTE		SEQ_NOTEOFF, MIDI_NOTEOFF
+KEYPRESS	MIDI_KEY_PRESSURE
+CHANPRESS	SEQ_AFTERTOUCH, MIDI_CHN_PRESSURE
+PGMCHANGE	SEQ_PGMCHANGE, MIDI_PGM_CHANGE
+PITCHBEND	SEQ_CONTROLLER(CTRL_PITCH_BENDER),
+		MIDI_PITCH_BEND
+CONTROLLER	MIDI_CTL_CHANGE,
+		SEQ_BALANCE (with CTL_PAN)
+CONTROL14	SEQ_CONTROLLER
+REGPARAM	SEQ_CONTROLLER(CTRL_PITCH_BENDER_RANGE)
+SYSEX		SEQ_SYSEX
+=============	===================
+
+The most of these behavior can be realized by MIDI emulation driver
+included in the Emu8000 lowlevel driver. In the future release, this module
+will be independent.
+
+Some OSS events (``SEQ_PRIVATE`` and ``SEQ_VOLUME`` events) are passed as event
+type SND_SEQ_OSS_PRIVATE.  The OSS sequencer passes these event 8 byte
+packets without any modification. The lowlevel driver should process these
+events appropriately.
+
+Interface to MIDI Device
+========================
+
+Since the OSS emulation probes the creation and deletion of ALSA MIDI
+sequencer ports automatically by receiving announcement from ALSA
+sequencer, the MIDI devices don't need to be registered explicitly
+like synth devices.
+However, the MIDI port_info registered to ALSA sequencer must include
+a group name ``SND_SEQ_GROUP_DEVICE`` and a capability-bit
+``CAP_READ`` or ``CAP_WRITE``. Also, subscription capabilities,
+``CAP_SUBS_READ`` or ``CAP_SUBS_WRITE``, must be defined, too. If
+these conditions are not satisfied, the port is not registered as OSS
+sequencer MIDI device.
+
+The events via MIDI devices are parsed in OSS sequencer and converted
+to the corresponding ALSA sequencer events. The input from MIDI sequencer
+is also converted to MIDI byte events by OSS sequencer. This works just
+a reverse way of seq_midi module.
+
+Known Problems / TODO's
+=======================
+
+* Patch loading via ALSA instrument layer is not implemented yet.
+
-- 
cgit v1.2.3-58-ga151


From 5a481fe309e79df2e713d7ea32175f121b251069 Mon Sep 17 00:00:00 2001
From: Takashi Iwai 
Date: Wed, 9 Nov 2016 17:55:28 +0100
Subject: ALSA: doc: ReSTize ControlNames.txt

A simple conversion from a plain text file.
Put to designs subdirectory.

Signed-off-by: Takashi Iwai 
---
 Documentation/sound/alsa/ControlNames.txt     | 107 -------------------
 Documentation/sound/designs/control-names.rst | 142 ++++++++++++++++++++++++++
 Documentation/sound/designs/index.rst         |   1 +
 3 files changed, 143 insertions(+), 107 deletions(-)
 delete mode 100644 Documentation/sound/alsa/ControlNames.txt
 create mode 100644 Documentation/sound/designs/control-names.rst

(limited to 'Documentation/sound')

diff --git a/Documentation/sound/alsa/ControlNames.txt b/Documentation/sound/alsa/ControlNames.txt
deleted file mode 100644
index 3fc1cf50d28e..000000000000
--- a/Documentation/sound/alsa/ControlNames.txt
+++ /dev/null
@@ -1,107 +0,0 @@
-This document describes standard names of mixer controls.
-
-Syntax: [LOCATION] SOURCE [CHANNEL] [DIRECTION] FUNCTION
-
-DIRECTION:
-  	(both directions)
-  Playback
-  Capture
-  Bypass Playback
-  Bypass Capture
-
-FUNCTION:
-  Switch	(on/off switch)
-  Volume
-  Route		(route control, hardware specific)
-
-CHANNEL:
-       (channel independent, or applies to all channels)
-  Front
-  Surround      (rear left/right in 4.0/5.1 surround)
-  CLFE
-  Center
-  LFE
-  Side          (side left/right for 7.1 surround)
-
-LOCATION:       (physical location of source)
-  Front
-  Rear
-  Dock          (docking station)
-  Internal
-
-SOURCE:
-  Master
-  Master Mono
-  Hardware Master
-  Speaker	(internal speaker)
-  Bass Speaker	(internal LFE speaker)
-  Headphone
-  Line Out
-  Beep		(beep generator)
-  Phone
-  Phone Input
-  Phone Output
-  Synth
-  FM
-  Mic
-  Headset Mic	(mic part of combined headset jack - 4-pin headphone + mic)
-  Headphone Mic	(mic part of either/or - 3-pin headphone or mic)
-  Line		(input only, use "Line Out" for output)
-  CD
-  Video
-  Zoom Video
-  Aux
-  PCM
-  PCM Pan
-  Loopback
-  Analog Loopback	(D/A -> A/D loopback)
-  Digital Loopback	(playback -> capture loopback - without analog path)
-  Mono
-  Mono Output
-  Multi
-  ADC
-  Wave
-  Music
-  I2S
-  IEC958
-  HDMI
-  SPDIF		(output only)
-  SPDIF In
-  Digital In
-  HDMI/DP	(either HDMI or DisplayPort)
-
-Exceptions (deprecated):
-  [Analogue|Digital] Capture Source
-  [Analogue|Digital] Capture Switch	(aka input gain switch)
-  [Analogue|Digital] Capture Volume	(aka input gain volume)
-  [Analogue|Digital] Playback Switch	(aka output gain switch)
-  [Analogue|Digital] Playback Volume	(aka output gain volume)
-  Tone Control - Switch
-  Tone Control - Bass
-  Tone Control - Treble
-  3D Control - Switch
-  3D Control - Center
-  3D Control - Depth
-  3D Control - Wide
-  3D Control - Space
-  3D Control - Level
-  Mic Boost [(?dB)]
-
-PCM interface:
-
-  Sample Clock Source	{ "Word", "Internal", "AutoSync" }
-  Clock Sync Status	{ "Lock", "Sync", "No Lock" }
-  External Rate		/* external capture rate */
-  Capture Rate		/* capture rate taken from external source */
-
-IEC958 (S/PDIF) interface:
-
-  IEC958 [...] [Playback|Capture] Switch	/* turn on/off the IEC958 interface */
-  IEC958 [...] [Playback|Capture] Volume	/* digital volume control */
-  IEC958 [...] [Playback|Capture] Default	/* default or global value - read/write */
-  IEC958 [...] [Playback|Capture] Mask		/* consumer and professional mask */
-  IEC958 [...] [Playback|Capture] Con Mask	/* consumer mask */
-  IEC958 [...] [Playback|Capture] Pro Mask	/* professional mask */
-  IEC958 [...] [Playback|Capture] PCM Stream	/* the settings assigned to a PCM stream */
-  IEC958 Q-subcode [Playback|Capture] Default	/* Q-subcode bits */
-  IEC958 Preamble [Playback|Capture] Default	/* burst preamble words (4*16bits) */
diff --git a/Documentation/sound/designs/control-names.rst b/Documentation/sound/designs/control-names.rst
new file mode 100644
index 000000000000..7fedd0f33cd9
--- /dev/null
+++ b/Documentation/sound/designs/control-names.rst
@@ -0,0 +1,142 @@
+===========================
+Standard ALSA Control Names
+===========================
+
+This document describes standard names of mixer controls.
+
+Standard Syntax
+---------------
+Syntax: [LOCATION] SOURCE [CHANNEL] [DIRECTION] FUNCTION
+
+
+DIRECTION
+~~~~~~~~~
+================	===============
+		both directions
+Playback		one direction
+Capture			one direction
+Bypass Playback		one direction
+Bypass Capture		one direction
+================	===============
+
+FUNCTION
+~~~~~~~~
+========	=================================
+Switch		on/off switch
+Volume		amplifier
+Route		route control, hardware specific
+========	=================================
+
+CHANNEL
+~~~~~~~
+============	==================================================
+	channel independent, or applies to all channels
+Front		front left/right channels
+Surround	rear left/right in 4.0/5.1 surround
+CLFE		C/LFE channels
+Center		center cannel
+LFE		LFE channel
+Side		side left/right for 7.1 surround
+============	==================================================
+
+LOCATION (Physical location of source)
+~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+============	=====================
+Front		front position
+Rear		rear position
+Dock		on docking station
+Internal	internal
+============	=====================
+
+SOURCE
+~~~~~~
+===================	=================================================
+Master
+Master Mono
+Hardware Master
+Speaker			internal speaker
+Bass Speaker		internal LFE speaker
+Headphone
+Line Out
+Beep			beep generator
+Phone
+Phone Input
+Phone Output
+Synth
+FM
+Mic
+Headset Mic		mic part of combined headset jack - 4-pin
+			headphone + mic
+Headphone Mic		mic part of either/or - 3-pin headphone or mic
+Line			input only, use "Line Out" for output
+CD
+Video
+Zoom Video
+Aux
+PCM
+PCM Pan
+Loopback
+Analog Loopback		D/A -> A/D loopback
+Digital Loopback	playback -> capture loopback -
+			without analog path
+Mono
+Mono Output
+Multi
+ADC
+Wave
+Music
+I2S
+IEC958
+HDMI
+SPDIF			output only
+SPDIF In
+Digital In
+HDMI/DP			either HDMI or DisplayPort
+===================	=================================================
+
+Exceptions (deprecated)
+-----------------------
+
+=====================================	=======================
+[Analogue|Digital] Capture Source
+[Analogue|Digital] Capture Switch	aka input gain switch
+[Analogue|Digital] Capture Volume	aka input gain volume
+[Analogue|Digital] Playback Switch	aka output gain switch
+[Analogue|Digital] Playback Volume	aka output gain volume
+Tone Control - Switch
+Tone Control - Bass
+Tone Control - Treble
+3D Control - Switch
+3D Control - Center
+3D Control - Depth
+3D Control - Wide
+3D Control - Space
+3D Control - Level
+Mic Boost [(?dB)]
+=====================================	=======================
+
+PCM interface
+-------------
+
+===================	========================================
+Sample Clock Source	{ "Word", "Internal", "AutoSync" }
+Clock Sync Status	{ "Lock", "Sync", "No Lock" }
+External Rate		external capture rate
+Capture Rate		capture rate taken from external source
+===================	========================================
+
+IEC958 (S/PDIF) interface
+-------------------------
+
+============================================	======================================
+IEC958 [...] [Playback|Capture] Switch		turn on/off the IEC958 interface
+IEC958 [...] [Playback|Capture] Volume		digital volume control
+IEC958 [...] [Playback|Capture] Default		default or global value - read/write
+IEC958 [...] [Playback|Capture] Mask		consumer and professional mask
+IEC958 [...] [Playback|Capture] Con Mask	consumer mask
+IEC958 [...] [Playback|Capture] Pro Mask	professional mask
+IEC958 [...] [Playback|Capture] PCM Stream	the settings assigned to a PCM stream
+IEC958 Q-subcode [Playback|Capture] Default	Q-subcode bits
+
+IEC958 Preamble [Playback|Capture] Default	burst preamble words (4*16bits)
+============================================	======================================
diff --git a/Documentation/sound/designs/index.rst b/Documentation/sound/designs/index.rst
index 0ca3a6b0a5ce..e53a5fac0acf 100644
--- a/Documentation/sound/designs/index.rst
+++ b/Documentation/sound/designs/index.rst
@@ -4,6 +4,7 @@ Designs and Implementations
 .. toctree::
    :maxdepth: 2
 
+   control-names
    channel-mapping-api
    procfile
    powersave
-- 
cgit v1.2.3-58-ga151


From e9df12c3bac69e2cdfd76d7717bed92dee7cd617 Mon Sep 17 00:00:00 2001
From: Takashi Iwai 
Date: Thu, 10 Nov 2016 10:58:05 +0100
Subject: ALSA: doc: ReSTize compress-offload document

A simple conversion from a plain text file.
Put to designs subdirectory.

Signed-off-by: Takashi Iwai 
---
 Documentation/sound/alsa/compress_offload.txt    | 234 ----------------------
 Documentation/sound/designs/compress-offload.rst | 245 +++++++++++++++++++++++
 Documentation/sound/designs/index.rst            |   1 +
 3 files changed, 246 insertions(+), 234 deletions(-)
 delete mode 100644 Documentation/sound/alsa/compress_offload.txt
 create mode 100644 Documentation/sound/designs/compress-offload.rst

(limited to 'Documentation/sound')

diff --git a/Documentation/sound/alsa/compress_offload.txt b/Documentation/sound/alsa/compress_offload.txt
deleted file mode 100644
index 8ba556a131c3..000000000000
--- a/Documentation/sound/alsa/compress_offload.txt
+++ /dev/null
@@ -1,234 +0,0 @@
-		compress_offload.txt
-		=====================
-	Pierre-Louis.Bossart 
-		Vinod Koul 
-
-Overview
-
-Since its early days, the ALSA API was defined with PCM support or
-constant bitrates payloads such as IEC61937 in mind. Arguments and
-returned values in frames are the norm, making it a challenge to
-extend the existing API to compressed data streams.
-
-In recent years, audio digital signal processors (DSP) were integrated
-in system-on-chip designs, and DSPs are also integrated in audio
-codecs. Processing compressed data on such DSPs results in a dramatic
-reduction of power consumption compared to host-based
-processing. Support for such hardware has not been very good in Linux,
-mostly because of a lack of a generic API available in the mainline
-kernel.
-
-Rather than requiring a compatibility break with an API change of the
-ALSA PCM interface, a new 'Compressed Data' API is introduced to
-provide a control and data-streaming interface for audio DSPs.
-
-The design of this API was inspired by the 2-year experience with the
-Intel Moorestown SOC, with many corrections required to upstream the
-API in the mainline kernel instead of the staging tree and make it
-usable by others.
-
-Requirements
-
-The main requirements are:
-
-- separation between byte counts and time. Compressed formats may have
-  a header per file, per frame, or no header at all. The payload size
-  may vary from frame-to-frame. As a result, it is not possible to
-  estimate reliably the duration of audio buffers when handling
-  compressed data. Dedicated mechanisms are required to allow for
-  reliable audio-video synchronization, which requires precise
-  reporting of the number of samples rendered at any given time.
-
-- Handling of multiple formats. PCM data only requires a specification
-  of the sampling rate, number of channels and bits per sample. In
-  contrast, compressed data comes in a variety of formats. Audio DSPs
-  may also provide support for a limited number of audio encoders and
-  decoders embedded in firmware, or may support more choices through
-  dynamic download of libraries.
-
-- Focus on main formats. This API provides support for the most
-  popular formats used for audio and video capture and playback. It is
-  likely that as audio compression technology advances, new formats
-  will be added.
-
-- Handling of multiple configurations. Even for a given format like
-  AAC, some implementations may support AAC multichannel but HE-AAC
-  stereo. Likewise WMA10 level M3 may require too much memory and cpu
-  cycles. The new API needs to provide a generic way of listing these
-  formats.
-
-- Rendering/Grabbing only. This API does not provide any means of
-  hardware acceleration, where PCM samples are provided back to
-  user-space for additional processing. This API focuses instead on
-  streaming compressed data to a DSP, with the assumption that the
-  decoded samples are routed to a physical output or logical back-end.
-
- - Complexity hiding. Existing user-space multimedia frameworks all
-  have existing enums/structures for each compressed format. This new
-  API assumes the existence of a platform-specific compatibility layer
-  to expose, translate and make use of the capabilities of the audio
-  DSP, eg. Android HAL or PulseAudio sinks. By construction, regular
-  applications are not supposed to make use of this API.
-
-
-Design
-
-The new API shares a number of concepts with the PCM API for flow
-control. Start, pause, resume, drain and stop commands have the same
-semantics no matter what the content is.
-
-The concept of memory ring buffer divided in a set of fragments is
-borrowed from the ALSA PCM API. However, only sizes in bytes can be
-specified.
-
-Seeks/trick modes are assumed to be handled by the host.
-
-The notion of rewinds/forwards is not supported. Data committed to the
-ring buffer cannot be invalidated, except when dropping all buffers.
-
-The Compressed Data API does not make any assumptions on how the data
-is transmitted to the audio DSP. DMA transfers from main memory to an
-embedded audio cluster or to a SPI interface for external DSPs are
-possible. As in the ALSA PCM case, a core set of routines is exposed;
-each driver implementer will have to write support for a set of
-mandatory routines and possibly make use of optional ones.
-
-The main additions are
-
-- get_caps
-This routine returns the list of audio formats supported. Querying the
-codecs on a capture stream will return encoders, decoders will be
-listed for playback streams.
-
-- get_codec_caps For each codec, this routine returns a list of
-capabilities. The intent is to make sure all the capabilities
-correspond to valid settings, and to minimize the risks of
-configuration failures. For example, for a complex codec such as AAC,
-the number of channels supported may depend on a specific profile. If
-the capabilities were exposed with a single descriptor, it may happen
-that a specific combination of profiles/channels/formats may not be
-supported. Likewise, embedded DSPs have limited memory and cpu cycles,
-it is likely that some implementations make the list of capabilities
-dynamic and dependent on existing workloads. In addition to codec
-settings, this routine returns the minimum buffer size handled by the
-implementation. This information can be a function of the DMA buffer
-sizes, the number of bytes required to synchronize, etc, and can be
-used by userspace to define how much needs to be written in the ring
-buffer before playback can start.
-
-- set_params
-This routine sets the configuration chosen for a specific codec. The
-most important field in the parameters is the codec type; in most
-cases decoders will ignore other fields, while encoders will strictly
-comply to the settings
-
-- get_params
-This routines returns the actual settings used by the DSP. Changes to
-the settings should remain the exception.
-
-- get_timestamp
-The timestamp becomes a multiple field structure. It lists the number
-of bytes transferred, the number of samples processed and the number
-of samples rendered/grabbed. All these values can be used to determine
-the average bitrate, figure out if the ring buffer needs to be
-refilled or the delay due to decoding/encoding/io on the DSP.
-
-Note that the list of codecs/profiles/modes was derived from the
-OpenMAX AL specification instead of reinventing the wheel.
-Modifications include:
-- Addition of FLAC and IEC formats
-- Merge of encoder/decoder capabilities
-- Profiles/modes listed as bitmasks to make descriptors more compact
-- Addition of set_params for decoders (missing in OpenMAX AL)
-- Addition of AMR/AMR-WB encoding modes (missing in OpenMAX AL)
-- Addition of format information for WMA
-- Addition of encoding options when required (derived from OpenMAX IL)
-- Addition of rateControlSupported (missing in OpenMAX AL)
-
-Gapless Playback
-================
-When playing thru an album, the decoders have the ability to skip the encoder
-delay and padding and directly move from one track content to another. The end
-user can perceive this as gapless playback as we don't have silence while
-switching from one track to another
-
-Also, there might be low-intensity noises due to encoding. Perfect gapless is
-difficult to reach with all types of compressed data, but works fine with most
-music content. The decoder needs to know the encoder delay and encoder padding.
-So we need to pass this to DSP. This metadata is extracted from ID3/MP4 headers
-and are not present by default in the bitstream, hence the need for a new
-interface to pass this information to the DSP. Also DSP and userspace needs to
-switch from one track to another and start using data for second track.
-
-The main additions are:
-
-- set_metadata
-This routine sets the encoder delay and encoder padding. This can be used by
-decoder to strip the silence. This needs to be set before the data in the track
-is written.
-
-- set_next_track
-This routine tells DSP that metadata and write operation sent after this would
-correspond to subsequent track
-
-- partial drain
-This is called when end of file is reached. The userspace can inform DSP that
-EOF is reached and now DSP can start skipping padding delay. Also next write
-data would belong to next track
-
-Sequence flow for gapless would be:
-- Open
-- Get caps / codec caps
-- Set params
-- Set metadata of the first track
-- Fill data of the first track
-- Trigger start
-- User-space finished sending all,
-- Indicate next track data by sending set_next_track
-- Set metadata of the next track
-- then call partial_drain to flush most of buffer in DSP
-- Fill data of the next track
-- DSP switches to second track
-(note: order for partial_drain and write for next track can be reversed as well)
-
-Not supported:
-
-- Support for VoIP/circuit-switched calls is not the target of this
-  API. Support for dynamic bit-rate changes would require a tight
-  coupling between the DSP and the host stack, limiting power savings.
-
-- Packet-loss concealment is not supported. This would require an
-  additional interface to let the decoder synthesize data when frames
-  are lost during transmission. This may be added in the future.
-
-- Volume control/routing is not handled by this API. Devices exposing a
-  compressed data interface will be considered as regular ALSA devices;
-  volume changes and routing information will be provided with regular
-  ALSA kcontrols.
-
-- Embedded audio effects. Such effects should be enabled in the same
-  manner, no matter if the input was PCM or compressed.
-
-- multichannel IEC encoding. Unclear if this is required.
-
-- Encoding/decoding acceleration is not supported as mentioned
-  above. It is possible to route the output of a decoder to a capture
-  stream, or even implement transcoding capabilities. This routing
-  would be enabled with ALSA kcontrols.
-
-- Audio policy/resource management. This API does not provide any
-  hooks to query the utilization of the audio DSP, nor any preemption
-  mechanisms.
-
-- No notion of underrun/overrun. Since the bytes written are compressed
-  in nature and data written/read doesn't translate directly to
-  rendered output in time, this does not deal with underrun/overrun and
-  maybe dealt in user-library
-
-Credits:
-- Mark Brown and Liam Girdwood for discussions on the need for this API
-- Harsha Priya for her work on intel_sst compressed API
-- Rakesh Ughreja for valuable feedback
-- Sing Nallasellan, Sikkandar Madar and Prasanna Samaga for
-  demonstrating and quantifying the benefits of audio offload on a
-  real platform.
diff --git a/Documentation/sound/designs/compress-offload.rst b/Documentation/sound/designs/compress-offload.rst
new file mode 100644
index 000000000000..ad4bfbdacc83
--- /dev/null
+++ b/Documentation/sound/designs/compress-offload.rst
@@ -0,0 +1,245 @@
+=========================
+ALSA Compress-Offload API
+=========================
+
+Pierre-Louis.Bossart 
+
+Vinod Koul 
+
+
+Overview
+========
+Since its early days, the ALSA API was defined with PCM support or
+constant bitrates payloads such as IEC61937 in mind. Arguments and
+returned values in frames are the norm, making it a challenge to
+extend the existing API to compressed data streams.
+
+In recent years, audio digital signal processors (DSP) were integrated
+in system-on-chip designs, and DSPs are also integrated in audio
+codecs. Processing compressed data on such DSPs results in a dramatic
+reduction of power consumption compared to host-based
+processing. Support for such hardware has not been very good in Linux,
+mostly because of a lack of a generic API available in the mainline
+kernel.
+
+Rather than requiring a compatibility break with an API change of the
+ALSA PCM interface, a new 'Compressed Data' API is introduced to
+provide a control and data-streaming interface for audio DSPs.
+
+The design of this API was inspired by the 2-year experience with the
+Intel Moorestown SOC, with many corrections required to upstream the
+API in the mainline kernel instead of the staging tree and make it
+usable by others.
+
+
+Requirements
+============
+The main requirements are:
+
+- separation between byte counts and time. Compressed formats may have
+  a header per file, per frame, or no header at all. The payload size
+  may vary from frame-to-frame. As a result, it is not possible to
+  estimate reliably the duration of audio buffers when handling
+  compressed data. Dedicated mechanisms are required to allow for
+  reliable audio-video synchronization, which requires precise
+  reporting of the number of samples rendered at any given time.
+
+- Handling of multiple formats. PCM data only requires a specification
+  of the sampling rate, number of channels and bits per sample. In
+  contrast, compressed data comes in a variety of formats. Audio DSPs
+  may also provide support for a limited number of audio encoders and
+  decoders embedded in firmware, or may support more choices through
+  dynamic download of libraries.
+
+- Focus on main formats. This API provides support for the most
+  popular formats used for audio and video capture and playback. It is
+  likely that as audio compression technology advances, new formats
+  will be added.
+
+- Handling of multiple configurations. Even for a given format like
+  AAC, some implementations may support AAC multichannel but HE-AAC
+  stereo. Likewise WMA10 level M3 may require too much memory and cpu
+  cycles. The new API needs to provide a generic way of listing these
+  formats.
+
+- Rendering/Grabbing only. This API does not provide any means of
+  hardware acceleration, where PCM samples are provided back to
+  user-space for additional processing. This API focuses instead on
+  streaming compressed data to a DSP, with the assumption that the
+  decoded samples are routed to a physical output or logical back-end.
+
+- Complexity hiding. Existing user-space multimedia frameworks all
+  have existing enums/structures for each compressed format. This new
+  API assumes the existence of a platform-specific compatibility layer
+  to expose, translate and make use of the capabilities of the audio
+  DSP, eg. Android HAL or PulseAudio sinks. By construction, regular
+  applications are not supposed to make use of this API.
+
+
+Design
+======
+The new API shares a number of concepts with the PCM API for flow
+control. Start, pause, resume, drain and stop commands have the same
+semantics no matter what the content is.
+
+The concept of memory ring buffer divided in a set of fragments is
+borrowed from the ALSA PCM API. However, only sizes in bytes can be
+specified.
+
+Seeks/trick modes are assumed to be handled by the host.
+
+The notion of rewinds/forwards is not supported. Data committed to the
+ring buffer cannot be invalidated, except when dropping all buffers.
+
+The Compressed Data API does not make any assumptions on how the data
+is transmitted to the audio DSP. DMA transfers from main memory to an
+embedded audio cluster or to a SPI interface for external DSPs are
+possible. As in the ALSA PCM case, a core set of routines is exposed;
+each driver implementer will have to write support for a set of
+mandatory routines and possibly make use of optional ones.
+
+The main additions are
+
+get_caps
+  This routine returns the list of audio formats supported. Querying the
+  codecs on a capture stream will return encoders, decoders will be
+  listed for playback streams.
+
+get_codec_caps
+  For each codec, this routine returns a list of
+  capabilities. The intent is to make sure all the capabilities
+  correspond to valid settings, and to minimize the risks of
+  configuration failures. For example, for a complex codec such as AAC,
+  the number of channels supported may depend on a specific profile. If
+  the capabilities were exposed with a single descriptor, it may happen
+  that a specific combination of profiles/channels/formats may not be
+  supported. Likewise, embedded DSPs have limited memory and cpu cycles,
+  it is likely that some implementations make the list of capabilities
+  dynamic and dependent on existing workloads. In addition to codec
+  settings, this routine returns the minimum buffer size handled by the
+  implementation. This information can be a function of the DMA buffer
+  sizes, the number of bytes required to synchronize, etc, and can be
+  used by userspace to define how much needs to be written in the ring
+  buffer before playback can start.
+
+set_params
+  This routine sets the configuration chosen for a specific codec. The
+  most important field in the parameters is the codec type; in most
+  cases decoders will ignore other fields, while encoders will strictly
+  comply to the settings
+
+get_params
+  This routines returns the actual settings used by the DSP. Changes to
+  the settings should remain the exception.
+
+get_timestamp
+  The timestamp becomes a multiple field structure. It lists the number
+  of bytes transferred, the number of samples processed and the number
+  of samples rendered/grabbed. All these values can be used to determine
+  the average bitrate, figure out if the ring buffer needs to be
+  refilled or the delay due to decoding/encoding/io on the DSP.
+
+Note that the list of codecs/profiles/modes was derived from the
+OpenMAX AL specification instead of reinventing the wheel.
+Modifications include:
+- Addition of FLAC and IEC formats
+- Merge of encoder/decoder capabilities
+- Profiles/modes listed as bitmasks to make descriptors more compact
+- Addition of set_params for decoders (missing in OpenMAX AL)
+- Addition of AMR/AMR-WB encoding modes (missing in OpenMAX AL)
+- Addition of format information for WMA
+- Addition of encoding options when required (derived from OpenMAX IL)
+- Addition of rateControlSupported (missing in OpenMAX AL)
+
+
+Gapless Playback
+================
+When playing thru an album, the decoders have the ability to skip the encoder
+delay and padding and directly move from one track content to another. The end
+user can perceive this as gapless playback as we don't have silence while
+switching from one track to another
+
+Also, there might be low-intensity noises due to encoding. Perfect gapless is
+difficult to reach with all types of compressed data, but works fine with most
+music content. The decoder needs to know the encoder delay and encoder padding.
+So we need to pass this to DSP. This metadata is extracted from ID3/MP4 headers
+and are not present by default in the bitstream, hence the need for a new
+interface to pass this information to the DSP. Also DSP and userspace needs to
+switch from one track to another and start using data for second track.
+
+The main additions are:
+
+set_metadata
+  This routine sets the encoder delay and encoder padding. This can be used by
+  decoder to strip the silence. This needs to be set before the data in the track
+  is written.
+
+set_next_track
+  This routine tells DSP that metadata and write operation sent after this would
+  correspond to subsequent track
+
+partial drain
+  This is called when end of file is reached. The userspace can inform DSP that
+  EOF is reached and now DSP can start skipping padding delay. Also next write
+  data would belong to next track
+
+Sequence flow for gapless would be:
+- Open
+- Get caps / codec caps
+- Set params
+- Set metadata of the first track
+- Fill data of the first track
+- Trigger start
+- User-space finished sending all,
+- Indicate next track data by sending set_next_track
+- Set metadata of the next track
+- then call partial_drain to flush most of buffer in DSP
+- Fill data of the next track
+- DSP switches to second track
+
+(note: order for partial_drain and write for next track can be reversed as well)
+
+
+Not supported
+=============
+- Support for VoIP/circuit-switched calls is not the target of this
+  API. Support for dynamic bit-rate changes would require a tight
+  coupling between the DSP and the host stack, limiting power savings.
+
+- Packet-loss concealment is not supported. This would require an
+  additional interface to let the decoder synthesize data when frames
+  are lost during transmission. This may be added in the future.
+
+- Volume control/routing is not handled by this API. Devices exposing a
+  compressed data interface will be considered as regular ALSA devices;
+  volume changes and routing information will be provided with regular
+  ALSA kcontrols.
+
+- Embedded audio effects. Such effects should be enabled in the same
+  manner, no matter if the input was PCM or compressed.
+
+- multichannel IEC encoding. Unclear if this is required.
+
+- Encoding/decoding acceleration is not supported as mentioned
+  above. It is possible to route the output of a decoder to a capture
+  stream, or even implement transcoding capabilities. This routing
+  would be enabled with ALSA kcontrols.
+
+- Audio policy/resource management. This API does not provide any
+  hooks to query the utilization of the audio DSP, nor any preemption
+  mechanisms.
+
+- No notion of underrun/overrun. Since the bytes written are compressed
+  in nature and data written/read doesn't translate directly to
+  rendered output in time, this does not deal with underrun/overrun and
+  maybe dealt in user-library
+
+
+Credits
+=======
+- Mark Brown and Liam Girdwood for discussions on the need for this API
+- Harsha Priya for her work on intel_sst compressed API
+- Rakesh Ughreja for valuable feedback
+- Sing Nallasellan, Sikkandar Madar and Prasanna Samaga for
+  demonstrating and quantifying the benefits of audio offload on a
+  real platform.
diff --git a/Documentation/sound/designs/index.rst b/Documentation/sound/designs/index.rst
index e53a5fac0acf..f7ca11307033 100644
--- a/Documentation/sound/designs/index.rst
+++ b/Documentation/sound/designs/index.rst
@@ -6,6 +6,7 @@ Designs and Implementations
 
    control-names
    channel-mapping-api
+   compress-offload
    procfile
    powersave
    oss-emulation
-- 
cgit v1.2.3-58-ga151


From 20a1d0f44d9447a68c387a2f561db4720302fb7c Mon Sep 17 00:00:00 2001
From: Takashi Iwai 
Date: Thu, 10 Nov 2016 11:06:55 +0100
Subject: ALSA: doc: ReSTize timestamping document

A simple conversion from a plain text file.
Put to designs subdirectory.

Signed-off-by: Takashi Iwai 
---
 Documentation/sound/alsa/timestamping.txt    | 200 -------------------------
 Documentation/sound/designs/index.rst        |   1 +
 Documentation/sound/designs/timestamping.rst | 215 +++++++++++++++++++++++++++
 3 files changed, 216 insertions(+), 200 deletions(-)
 delete mode 100644 Documentation/sound/alsa/timestamping.txt
 create mode 100644 Documentation/sound/designs/timestamping.rst

(limited to 'Documentation/sound')

diff --git a/Documentation/sound/alsa/timestamping.txt b/Documentation/sound/alsa/timestamping.txt
deleted file mode 100644
index 9d579aefbffd..000000000000
--- a/Documentation/sound/alsa/timestamping.txt
+++ /dev/null
@@ -1,200 +0,0 @@
-The ALSA API can provide two different system timestamps:
-
-- Trigger_tstamp is the system time snapshot taken when the .trigger
-callback is invoked. This snapshot is taken by the ALSA core in the
-general case, but specific hardware may have synchronization
-capabilities or conversely may only be able to provide a correct
-estimate with a delay. In the latter two cases, the low-level driver
-is responsible for updating the trigger_tstamp at the most appropriate
-and precise moment. Applications should not rely solely on the first
-trigger_tstamp but update their internal calculations if the driver
-provides a refined estimate with a delay.
-
-- tstamp is the current system timestamp updated during the last
-event or application query.
-The difference (tstamp - trigger_tstamp) defines the elapsed time.
-
-The ALSA API provides two basic pieces of information, avail
-and delay, which combined with the trigger and current system
-timestamps allow for applications to keep track of the 'fullness' of
-the ring buffer and the amount of queued samples.
-
-The use of these different pointers and time information depends on
-the application needs:
-
-- 'avail' reports how much can be written in the ring buffer
-- 'delay' reports the time it will take to hear a new sample after all
-queued samples have been played out.
-
-When timestamps are enabled, the avail/delay information is reported
-along with a snapshot of system time. Applications can select from
-CLOCK_REALTIME (NTP corrections including going backwards),
-CLOCK_MONOTONIC (NTP corrections but never going backwards),
-CLOCK_MONOTIC_RAW (without NTP corrections) and change the mode
-dynamically with sw_params
-
-
-The ALSA API also provide an audio_tstamp which reflects the passage
-of time as measured by different components of audio hardware.  In
-ascii-art, this could be represented as follows (for the playback
-case):
-
-
---------------------------------------------------------------> time
-  ^               ^              ^                ^           ^
-  |               |              |                |           |
- analog         link            dma              app       FullBuffer
- time           time           time              time        time
-  |               |              |                |           |
-  |< codec delay >|<--hw delay-->||<---avail->|
-  |<----------------- delay---------------------->|           |
-			         |<----ring buffer length---->|
-
-The analog time is taken at the last stage of the playback, as close
-as possible to the actual transducer
-
-The link time is taken at the output of the SoC/chipset as the samples
-are pushed on a link. The link time can be directly measured if
-supported in hardware by sample counters or wallclocks (e.g. with
-HDAudio 24MHz or PTP clock for networked solutions) or indirectly
-estimated (e.g. with the frame counter in USB).
-
-The DMA time is measured using counters - typically the least reliable
-of all measurements due to the bursty nature of DMA transfers.
-
-The app time corresponds to the time tracked by an application after
-writing in the ring buffer.
-
-The application can query the hardware capabilities, define which
-audio time it wants reported by selecting the relevant settings in
-audio_tstamp_config fields, thus get an estimate of the timestamp
-accuracy. It can also request the delay-to-analog be included in the
-measurement. Direct access to the link time is very interesting on
-platforms that provide an embedded DSP; measuring directly the link
-time with dedicated hardware, possibly synchronized with system time,
-removes the need to keep track of internal DSP processing times and
-latency.
-
-In case the application requests an audio tstamp that is not supported
-in hardware/low-level driver, the type is overridden as DEFAULT and the
-timestamp will report the DMA time based on the hw_pointer value.
-
-For backwards compatibility with previous implementations that did not
-provide timestamp selection, with a zero-valued COMPAT timestamp type
-the results will default to the HDAudio wall clock for playback
-streams and to the DMA time (hw_ptr) in all other cases.
-
-The audio timestamp accuracy can be returned to user-space, so that
-appropriate decisions are made:
-
-- for dma time (default), the granularity of the transfers can be
-  inferred from the steps between updates and in turn provide
-  information on how much the application pointer can be rewound
-  safely.
-
-- the link time can be used to track long-term drifts between audio
-  and system time using the (tstamp-trigger_tstamp)/audio_tstamp
-  ratio, the precision helps define how much smoothing/low-pass
-  filtering is required. The link time can be either reset on startup
-  or reported as is (the latter being useful to compare progress of
-  different streams - but may require the wallclock to be always
-  running and not wrap-around during idle periods). If supported in
-  hardware, the absolute link time could also be used to define a
-  precise start time (patches WIP)
-
-- including the delay in the audio timestamp may
-  counter-intuitively not increase the precision of timestamps, e.g. if a
-  codec includes variable-latency DSP processing or a chain of
-  hardware components the delay is typically not known with precision.
-
-The accuracy is reported in nanosecond units (using an unsigned 32-bit
-word), which gives a max precision of 4.29s, more than enough for
-audio applications...
-
-Due to the varied nature of timestamping needs, even for a single
-application, the audio_tstamp_config can be changed dynamically. In
-the STATUS ioctl, the parameters are read-only and do not allow for
-any application selection. To work around this limitation without
-impacting legacy applications, a new STATUS_EXT ioctl is introduced
-with read/write parameters. ALSA-lib will be modified to make use of
-STATUS_EXT and effectively deprecate STATUS.
-
-The ALSA API only allows for a single audio timestamp to be reported
-at a time. This is a conscious design decision, reading the audio
-timestamps from hardware registers or from IPC takes time, the more
-timestamps are read the more imprecise the combined measurements
-are. To avoid any interpretation issues, a single (system, audio)
-timestamp is reported. Applications that need different timestamps
-will be required to issue multiple queries and perform an
-interpolation of the results
-
-In some hardware-specific configuration, the system timestamp is
-latched by a low-level audio subsystem, and the information provided
-back to the driver. Due to potential delays in the communication with
-the hardware, there is a risk of misalignment with the avail and delay
-information. To make sure applications are not confused, a
-driver_timestamp field is added in the snd_pcm_status structure; this
-timestamp shows when the information is put together by the driver
-before returning from the STATUS and STATUS_EXT ioctl. in most cases
-this driver_timestamp will be identical to the regular system tstamp.
-
-Examples of typestamping with HDaudio:
-
-1. DMA timestamp, no compensation for DMA+analog delay
-$ ./audio_time  -p --ts_type=1
-playback: systime: 341121338 nsec, audio time 342000000 nsec, 	systime delta -878662
-playback: systime: 426236663 nsec, audio time 427187500 nsec, 	systime delta -950837
-playback: systime: 597080580 nsec, audio time 598000000 nsec, 	systime delta -919420
-playback: systime: 682059782 nsec, audio time 683020833 nsec, 	systime delta -961051
-playback: systime: 852896415 nsec, audio time 853854166 nsec, 	systime delta -957751
-playback: systime: 937903344 nsec, audio time 938854166 nsec, 	systime delta -950822
-
-2. DMA timestamp, compensation for DMA+analog delay
-$ ./audio_time  -p --ts_type=1 -d
-playback: systime: 341053347 nsec, audio time 341062500 nsec, 	systime delta -9153
-playback: systime: 426072447 nsec, audio time 426062500 nsec, 	systime delta 9947
-playback: systime: 596899518 nsec, audio time 596895833 nsec, 	systime delta 3685
-playback: systime: 681915317 nsec, audio time 681916666 nsec, 	systime delta -1349
-playback: systime: 852741306 nsec, audio time 852750000 nsec, 	systime delta -8694
-
-3. link timestamp, compensation for DMA+analog delay
-$ ./audio_time  -p --ts_type=2 -d
-playback: systime: 341060004 nsec, audio time 341062791 nsec, 	systime delta -2787
-playback: systime: 426242074 nsec, audio time 426244875 nsec, 	systime delta -2801
-playback: systime: 597080992 nsec, audio time 597084583 nsec, 	systime delta -3591
-playback: systime: 682084512 nsec, audio time 682088291 nsec, 	systime delta -3779
-playback: systime: 852936229 nsec, audio time 852940916 nsec, 	systime delta -4687
-playback: systime: 938107562 nsec, audio time 938112708 nsec, 	systime delta -5146
-
-Example 1 shows that the timestamp at the DMA level is close to 1ms
-ahead of the actual playback time (as a side time this sort of
-measurement can help define rewind safeguards). Compensating for the
-DMA-link delay in example 2 helps remove the hardware buffering but
-the information is still very jittery, with up to one sample of
-error. In example 3 where the timestamps are measured with the link
-wallclock, the timestamps show a monotonic behavior and a lower
-dispersion.
-
-Example 3 and 4 are with USB audio class. Example 3 shows a high
-offset between audio time and system time due to buffering. Example 4
-shows how compensating for the delay exposes a 1ms accuracy (due to
-the use of the frame counter by the driver)
-
-Example 3: DMA timestamp, no compensation for delay, delta of ~5ms
-$ ./audio_time -p -Dhw:1 -t1
-playback: systime: 120174019 nsec, audio time 125000000 nsec, 	systime delta -4825981
-playback: systime: 245041136 nsec, audio time 250000000 nsec, 	systime delta -4958864
-playback: systime: 370106088 nsec, audio time 375000000 nsec, 	systime delta -4893912
-playback: systime: 495040065 nsec, audio time 500000000 nsec, 	systime delta -4959935
-playback: systime: 620038179 nsec, audio time 625000000 nsec, 	systime delta -4961821
-playback: systime: 745087741 nsec, audio time 750000000 nsec, 	systime delta -4912259
-playback: systime: 870037336 nsec, audio time 875000000 nsec, 	systime delta -4962664
-
-Example 4: DMA timestamp, compensation for delay, delay of ~1ms
-$ ./audio_time -p -Dhw:1 -t1 -d
-playback: systime: 120190520 nsec, audio time 120000000 nsec, 	systime delta 190520
-playback: systime: 245036740 nsec, audio time 244000000 nsec, 	systime delta 1036740
-playback: systime: 370034081 nsec, audio time 369000000 nsec, 	systime delta 1034081
-playback: systime: 495159907 nsec, audio time 494000000 nsec, 	systime delta 1159907
-playback: systime: 620098824 nsec, audio time 619000000 nsec, 	systime delta 1098824
-playback: systime: 745031847 nsec, audio time 744000000 nsec, 	systime delta 1031847
diff --git a/Documentation/sound/designs/index.rst b/Documentation/sound/designs/index.rst
index f7ca11307033..798b1a44bbbd 100644
--- a/Documentation/sound/designs/index.rst
+++ b/Documentation/sound/designs/index.rst
@@ -7,6 +7,7 @@ Designs and Implementations
    control-names
    channel-mapping-api
    compress-offload
+   timestamping
    procfile
    powersave
    oss-emulation
diff --git a/Documentation/sound/designs/timestamping.rst b/Documentation/sound/designs/timestamping.rst
new file mode 100644
index 000000000000..2b0fff503415
--- /dev/null
+++ b/Documentation/sound/designs/timestamping.rst
@@ -0,0 +1,215 @@
+=====================
+ALSA PCM Timestamping
+=====================
+
+The ALSA API can provide two different system timestamps:
+
+- Trigger_tstamp is the system time snapshot taken when the .trigger
+  callback is invoked. This snapshot is taken by the ALSA core in the
+  general case, but specific hardware may have synchronization
+  capabilities or conversely may only be able to provide a correct
+  estimate with a delay. In the latter two cases, the low-level driver
+  is responsible for updating the trigger_tstamp at the most appropriate
+  and precise moment. Applications should not rely solely on the first
+  trigger_tstamp but update their internal calculations if the driver
+  provides a refined estimate with a delay.
+
+- tstamp is the current system timestamp updated during the last
+  event or application query.
+  The difference (tstamp - trigger_tstamp) defines the elapsed time.
+
+The ALSA API provides two basic pieces of information, avail
+and delay, which combined with the trigger and current system
+timestamps allow for applications to keep track of the 'fullness' of
+the ring buffer and the amount of queued samples.
+
+The use of these different pointers and time information depends on
+the application needs:
+
+- ``avail`` reports how much can be written in the ring buffer
+- ``delay`` reports the time it will take to hear a new sample after all
+  queued samples have been played out.
+
+When timestamps are enabled, the avail/delay information is reported
+along with a snapshot of system time. Applications can select from
+``CLOCK_REALTIME`` (NTP corrections including going backwards),
+``CLOCK_MONOTONIC`` (NTP corrections but never going backwards),
+``CLOCK_MONOTIC_RAW`` (without NTP corrections) and change the mode
+dynamically with sw_params
+
+
+The ALSA API also provide an audio_tstamp which reflects the passage
+of time as measured by different components of audio hardware.  In
+ascii-art, this could be represented as follows (for the playback
+case):
+::
+
+  --------------------------------------------------------------> time
+    ^               ^              ^                ^           ^
+    |               |              |                |           |
+   analog         link            dma              app       FullBuffer
+   time           time           time              time        time
+    |               |              |                |           |
+    |< codec delay >|<--hw delay-->||<---avail->|
+    |<----------------- delay---------------------->|           |
+                                   |<----ring buffer length---->|
+
+
+The analog time is taken at the last stage of the playback, as close
+as possible to the actual transducer
+
+The link time is taken at the output of the SoC/chipset as the samples
+are pushed on a link. The link time can be directly measured if
+supported in hardware by sample counters or wallclocks (e.g. with
+HDAudio 24MHz or PTP clock for networked solutions) or indirectly
+estimated (e.g. with the frame counter in USB).
+
+The DMA time is measured using counters - typically the least reliable
+of all measurements due to the bursty nature of DMA transfers.
+
+The app time corresponds to the time tracked by an application after
+writing in the ring buffer.
+
+The application can query the hardware capabilities, define which
+audio time it wants reported by selecting the relevant settings in
+audio_tstamp_config fields, thus get an estimate of the timestamp
+accuracy. It can also request the delay-to-analog be included in the
+measurement. Direct access to the link time is very interesting on
+platforms that provide an embedded DSP; measuring directly the link
+time with dedicated hardware, possibly synchronized with system time,
+removes the need to keep track of internal DSP processing times and
+latency.
+
+In case the application requests an audio tstamp that is not supported
+in hardware/low-level driver, the type is overridden as DEFAULT and the
+timestamp will report the DMA time based on the hw_pointer value.
+
+For backwards compatibility with previous implementations that did not
+provide timestamp selection, with a zero-valued COMPAT timestamp type
+the results will default to the HDAudio wall clock for playback
+streams and to the DMA time (hw_ptr) in all other cases.
+
+The audio timestamp accuracy can be returned to user-space, so that
+appropriate decisions are made:
+
+- for dma time (default), the granularity of the transfers can be
+  inferred from the steps between updates and in turn provide
+  information on how much the application pointer can be rewound
+  safely.
+
+- the link time can be used to track long-term drifts between audio
+  and system time using the (tstamp-trigger_tstamp)/audio_tstamp
+  ratio, the precision helps define how much smoothing/low-pass
+  filtering is required. The link time can be either reset on startup
+  or reported as is (the latter being useful to compare progress of
+  different streams - but may require the wallclock to be always
+  running and not wrap-around during idle periods). If supported in
+  hardware, the absolute link time could also be used to define a
+  precise start time (patches WIP)
+
+- including the delay in the audio timestamp may
+  counter-intuitively not increase the precision of timestamps, e.g. if a
+  codec includes variable-latency DSP processing or a chain of
+  hardware components the delay is typically not known with precision.
+
+The accuracy is reported in nanosecond units (using an unsigned 32-bit
+word), which gives a max precision of 4.29s, more than enough for
+audio applications...
+
+Due to the varied nature of timestamping needs, even for a single
+application, the audio_tstamp_config can be changed dynamically. In
+the ``STATUS`` ioctl, the parameters are read-only and do not allow for
+any application selection. To work around this limitation without
+impacting legacy applications, a new ``STATUS_EXT`` ioctl is introduced
+with read/write parameters. ALSA-lib will be modified to make use of
+``STATUS_EXT`` and effectively deprecate ``STATUS``.
+
+The ALSA API only allows for a single audio timestamp to be reported
+at a time. This is a conscious design decision, reading the audio
+timestamps from hardware registers or from IPC takes time, the more
+timestamps are read the more imprecise the combined measurements
+are. To avoid any interpretation issues, a single (system, audio)
+timestamp is reported. Applications that need different timestamps
+will be required to issue multiple queries and perform an
+interpolation of the results
+
+In some hardware-specific configuration, the system timestamp is
+latched by a low-level audio subsystem, and the information provided
+back to the driver. Due to potential delays in the communication with
+the hardware, there is a risk of misalignment with the avail and delay
+information. To make sure applications are not confused, a
+driver_timestamp field is added in the snd_pcm_status structure; this
+timestamp shows when the information is put together by the driver
+before returning from the ``STATUS`` and ``STATUS_EXT`` ioctl. in most cases
+this driver_timestamp will be identical to the regular system tstamp.
+
+Examples of typestamping with HDaudio:
+
+1. DMA timestamp, no compensation for DMA+analog delay
+::
+
+  $ ./audio_time  -p --ts_type=1
+  playback: systime: 341121338 nsec, audio time 342000000 nsec, 	systime delta -878662
+  playback: systime: 426236663 nsec, audio time 427187500 nsec, 	systime delta -950837
+  playback: systime: 597080580 nsec, audio time 598000000 nsec, 	systime delta -919420
+  playback: systime: 682059782 nsec, audio time 683020833 nsec, 	systime delta -961051
+  playback: systime: 852896415 nsec, audio time 853854166 nsec, 	systime delta -957751
+  playback: systime: 937903344 nsec, audio time 938854166 nsec, 	systime delta -950822
+
+2. DMA timestamp, compensation for DMA+analog delay
+::
+
+  $ ./audio_time  -p --ts_type=1 -d
+  playback: systime: 341053347 nsec, audio time 341062500 nsec, 	systime delta -9153
+  playback: systime: 426072447 nsec, audio time 426062500 nsec, 	systime delta 9947
+  playback: systime: 596899518 nsec, audio time 596895833 nsec, 	systime delta 3685
+  playback: systime: 681915317 nsec, audio time 681916666 nsec, 	systime delta -1349
+  playback: systime: 852741306 nsec, audio time 852750000 nsec, 	systime delta -8694
+
+3. link timestamp, compensation for DMA+analog delay
+::
+
+  $ ./audio_time  -p --ts_type=2 -d
+  playback: systime: 341060004 nsec, audio time 341062791 nsec, 	systime delta -2787
+  playback: systime: 426242074 nsec, audio time 426244875 nsec, 	systime delta -2801
+  playback: systime: 597080992 nsec, audio time 597084583 nsec, 	systime delta -3591
+  playback: systime: 682084512 nsec, audio time 682088291 nsec, 	systime delta -3779
+  playback: systime: 852936229 nsec, audio time 852940916 nsec, 	systime delta -4687
+  playback: systime: 938107562 nsec, audio time 938112708 nsec, 	systime delta -5146
+
+Example 1 shows that the timestamp at the DMA level is close to 1ms
+ahead of the actual playback time (as a side time this sort of
+measurement can help define rewind safeguards). Compensating for the
+DMA-link delay in example 2 helps remove the hardware buffering but
+the information is still very jittery, with up to one sample of
+error. In example 3 where the timestamps are measured with the link
+wallclock, the timestamps show a monotonic behavior and a lower
+dispersion.
+
+Example 3 and 4 are with USB audio class. Example 3 shows a high
+offset between audio time and system time due to buffering. Example 4
+shows how compensating for the delay exposes a 1ms accuracy (due to
+the use of the frame counter by the driver)
+
+Example 3: DMA timestamp, no compensation for delay, delta of ~5ms
+::
+
+  $ ./audio_time -p -Dhw:1 -t1
+  playback: systime: 120174019 nsec, audio time 125000000 nsec, 	systime delta -4825981
+  playback: systime: 245041136 nsec, audio time 250000000 nsec, 	systime delta -4958864
+  playback: systime: 370106088 nsec, audio time 375000000 nsec, 	systime delta -4893912
+  playback: systime: 495040065 nsec, audio time 500000000 nsec, 	systime delta -4959935
+  playback: systime: 620038179 nsec, audio time 625000000 nsec, 	systime delta -4961821
+  playback: systime: 745087741 nsec, audio time 750000000 nsec, 	systime delta -4912259
+  playback: systime: 870037336 nsec, audio time 875000000 nsec, 	systime delta -4962664
+
+Example 4: DMA timestamp, compensation for delay, delay of ~1ms
+::
+
+  $ ./audio_time -p -Dhw:1 -t1 -d
+  playback: systime: 120190520 nsec, audio time 120000000 nsec, 	systime delta 190520
+  playback: systime: 245036740 nsec, audio time 244000000 nsec, 	systime delta 1036740
+  playback: systime: 370034081 nsec, audio time 369000000 nsec, 	systime delta 1034081
+  playback: systime: 495159907 nsec, audio time 494000000 nsec, 	systime delta 1159907
+  playback: systime: 620098824 nsec, audio time 619000000 nsec, 	systime delta 1098824
+  playback: systime: 745031847 nsec, audio time 744000000 nsec, 	systime delta 1031847
-- 
cgit v1.2.3-58-ga151


From df3a57105c04a7bfe0874316a9e6ec968c35a6f5 Mon Sep 17 00:00:00 2001
From: Takashi Iwai 
Date: Thu, 10 Nov 2016 11:10:35 +0100
Subject: ALSA: doc: ReSTize Jack-Controls.txt

A simple conversion from a plain text file.
Put to designs subdirectory.

Signed-off-by: Takashi Iwai 
---
 Documentation/sound/alsa/Jack-Controls.txt    | 43 ------------------------
 Documentation/sound/designs/index.rst         |  1 +
 Documentation/sound/designs/jack-controls.rst | 48 +++++++++++++++++++++++++++
 3 files changed, 49 insertions(+), 43 deletions(-)
 delete mode 100644 Documentation/sound/alsa/Jack-Controls.txt
 create mode 100644 Documentation/sound/designs/jack-controls.rst

(limited to 'Documentation/sound')

diff --git a/Documentation/sound/alsa/Jack-Controls.txt b/Documentation/sound/alsa/Jack-Controls.txt
deleted file mode 100644
index fe1c5e0c8555..000000000000
--- a/Documentation/sound/alsa/Jack-Controls.txt
+++ /dev/null
@@ -1,43 +0,0 @@
-Why we need Jack kcontrols
-==========================
-
-ALSA uses kcontrols to export audio controls(switch, volume, Mux, ...)
-to user space. This means userspace applications like pulseaudio can
-switch off headphones and switch on speakers when no headphones are
-pluged in.
-
-The old ALSA jack code only created input devices for each registered
-jack. These jack input devices are not readable by userspace devices
-that run as non root.
-
-The new jack code creates embedded jack kcontrols for each jack that
-can be read by any process.
-
-This can be combined with UCM to allow userspace to route audio more
-intelligently based on jack insertion or removal events.
-
-Jack Kcontrol Internals
-=======================
-
-Each jack will have a kcontrol list, so that we can create a kcontrol
-and attach it to the jack, at jack creation stage. We can also add a
-kcontrol to an existing jack, at anytime when required.
-
-Those kcontrols will be freed automatically when the Jack is freed.
-
-How to use jack kcontrols
-=========================
-
-In order to keep compatibility, snd_jack_new() has been modified by
-adding two params :-
-
- - @initial_kctl: if true, create a kcontrol and add it to the jack
-	list.
- - @phantom_jack: Don't create a input device for phantom jacks.
-
-HDA jacks can set phantom_jack to true in order to create a phantom
-jack and set initial_kctl to true to create an initial kcontrol with
-the correct id.
-
-ASoC jacks should set initial_kctl as false. The pin name will be
-assigned as the jack kcontrol name.
diff --git a/Documentation/sound/designs/index.rst b/Documentation/sound/designs/index.rst
index 798b1a44bbbd..04dcdae3e4f2 100644
--- a/Documentation/sound/designs/index.rst
+++ b/Documentation/sound/designs/index.rst
@@ -8,6 +8,7 @@ Designs and Implementations
    channel-mapping-api
    compress-offload
    timestamping
+   jack-controls
    procfile
    powersave
    oss-emulation
diff --git a/Documentation/sound/designs/jack-controls.rst b/Documentation/sound/designs/jack-controls.rst
new file mode 100644
index 000000000000..ae25b1531bb0
--- /dev/null
+++ b/Documentation/sound/designs/jack-controls.rst
@@ -0,0 +1,48 @@
+==================
+ALSA Jack Controls
+==================
+
+Why we need Jack kcontrols
+==========================
+
+ALSA uses kcontrols to export audio controls(switch, volume, Mux, ...)
+to user space. This means userspace applications like pulseaudio can
+switch off headphones and switch on speakers when no headphones are
+pluged in.
+
+The old ALSA jack code only created input devices for each registered
+jack. These jack input devices are not readable by userspace devices
+that run as non root.
+
+The new jack code creates embedded jack kcontrols for each jack that
+can be read by any process.
+
+This can be combined with UCM to allow userspace to route audio more
+intelligently based on jack insertion or removal events.
+
+Jack Kcontrol Internals
+=======================
+
+Each jack will have a kcontrol list, so that we can create a kcontrol
+and attach it to the jack, at jack creation stage. We can also add a
+kcontrol to an existing jack, at anytime when required.
+
+Those kcontrols will be freed automatically when the Jack is freed.
+
+How to use jack kcontrols
+=========================
+
+In order to keep compatibility, snd_jack_new() has been modified by
+adding two params:
+
+initial_kctl
+  if true, create a kcontrol and add it to the jack list.
+phantom_jack
+  Don't create a input device for phantom jacks.
+
+HDA jacks can set phantom_jack to true in order to create a phantom
+jack and set initial_kctl to true to create an initial kcontrol with
+the correct id.
+
+ASoC jacks should set initial_kctl as false. The pin name will be
+assigned as the jack kcontrol name.
-- 
cgit v1.2.3-58-ga151


From f59c3c6d87d6f7e1593ca874b755265cf08f8714 Mon Sep 17 00:00:00 2001
From: Takashi Iwai 
Date: Wed, 9 Nov 2016 17:04:22 +0100
Subject: ALSA: doc: ReSTize Joystick document

A conversion from a simple text file.
A new subdirectory, cards, was created to contain the card-specific
information like this one.

Signed-off-by: Takashi Iwai 
---
 Documentation/sound/alsa/Joystick.txt  | 86 --------------------------------
 Documentation/sound/cards/index.rst    |  7 +++
 Documentation/sound/cards/joystick.rst | 91 ++++++++++++++++++++++++++++++++++
 Documentation/sound/index.rst          |  1 +
 4 files changed, 99 insertions(+), 86 deletions(-)
 delete mode 100644 Documentation/sound/alsa/Joystick.txt
 create mode 100644 Documentation/sound/cards/index.rst
 create mode 100644 Documentation/sound/cards/joystick.rst

(limited to 'Documentation/sound')

diff --git a/Documentation/sound/alsa/Joystick.txt b/Documentation/sound/alsa/Joystick.txt
deleted file mode 100644
index ccda41b10f8a..000000000000
--- a/Documentation/sound/alsa/Joystick.txt
+++ /dev/null
@@ -1,86 +0,0 @@
-Analog Joystick Support on ALSA Drivers
-=======================================
-                          Oct. 14, 2003
-           Takashi Iwai 
-
-General
--------
-
-First of all, you need to enable GAMEPORT support on Linux kernel for
-using a joystick with the ALSA driver.  For the details of gameport
-support, refer to Documentation/input/joystick.txt.
-
-The joystick support of ALSA drivers is different between ISA and PCI
-cards.  In the case of ISA (PnP) cards, it's usually handled by the
-independent module (ns558).  Meanwhile, the ALSA PCI drivers have the
-built-in gameport support.  Hence, when the ALSA PCI driver is built
-in the kernel, CONFIG_GAMEPORT must be 'y', too.  Otherwise, the
-gameport support on that card will be (silently) disabled.
-
-Some adapter modules probe the physical connection of the device at
-the load time.  It'd be safer to plug in the joystick device before
-loading the module.
-
-
-PCI Cards
----------
-
-For PCI cards, the joystick is enabled when the appropriate module
-option is specified.  Some drivers don't need options, and the
-joystick support is always enabled.  In the former ALSA version, there
-was a dynamic control API for the joystick activation.  It was
-changed, however, to the static module options because of the system
-stability and the resource management.
-
-The following PCI drivers support the joystick natively.
-
-    Driver	Module Option	Available Values
-    ---------------------------------------------------------------------------
-    als4000	joystick_port	0 = disable (default), 1 = auto-detect,
-                                manual: any address (e.g. 0x200)
-    au88x0	N/A		N/A
-    azf3328	joystick	0 = disable, 1 = enable, -1 = auto (default)
-    ens1370	joystick	0 = disable (default), 1 = enable
-    ens1371	joystick_port	0 = disable (default), 1 = auto-detect,
-                                manual: 0x200, 0x208, 0x210, 0x218
-    cmipci	joystick_port	0 = disable (default), 1 = auto-detect,
-                                manual: any address (e.g. 0x200)
-    cs4281	N/A		N/A
-    cs46xx	N/A		N/A
-    es1938	N/A		N/A
-    es1968	joystick	0 = disable (default), 1 = enable
-    sonicvibes	N/A		N/A
-    trident	N/A		N/A
-    via82xx(*1)	joystick	0 = disable (default), 1 = enable
-    ymfpci	joystick_port	0 = disable (default), 1 = auto-detect,
-                                manual: 0x201, 0x202, 0x204, 0x205(*2)
-    ---------------------------------------------------------------------------
-
-    *1)  VIA686A/B only
-    *2)  With YMF744/754 chips, the port address can be chosen arbitrarily
-
-The following drivers don't support gameport natively, but there are
-additional modules.  Load the corresponding module to add the gameport
-support.
-
-    Driver	Additional Module
-    -----------------------------
-    emu10k1	emu10k1-gp
-    fm801	fm801-gp
-    -----------------------------
-
-Note: the "pcigame" and "cs461x" modules are for the OSS drivers only.
-      These ALSA drivers (cs46xx, trident and au88x0) have the
-      built-in gameport support.
-
-As mentioned above, ALSA PCI drivers have the built-in gameport
-support, so you don't have to load ns558 module.  Just load "joydev"
-and the appropriate adapter module (e.g. "analog").
-
-
-ISA Cards
----------
-
-ALSA ISA drivers don't have the built-in gameport support.
-Instead, you need to load "ns558" module in addition to "joydev" and
-the adapter module (e.g. "analog").
diff --git a/Documentation/sound/cards/index.rst b/Documentation/sound/cards/index.rst
new file mode 100644
index 000000000000..e1f4b78c75d6
--- /dev/null
+++ b/Documentation/sound/cards/index.rst
@@ -0,0 +1,7 @@
+Card-Specific Information
+=========================
+
+.. toctree::
+   :maxdepth: 2
+
+   joystick
diff --git a/Documentation/sound/cards/joystick.rst b/Documentation/sound/cards/joystick.rst
new file mode 100644
index 000000000000..a6e468c81d02
--- /dev/null
+++ b/Documentation/sound/cards/joystick.rst
@@ -0,0 +1,91 @@
+=======================================
+Analog Joystick Support on ALSA Drivers
+=======================================
+
+Oct. 14, 2003
+
+Takashi Iwai 
+
+General
+-------
+
+First of all, you need to enable GAMEPORT support on Linux kernel for
+using a joystick with the ALSA driver.  For the details of gameport
+support, refer to Documentation/input/joystick.txt.
+
+The joystick support of ALSA drivers is different between ISA and PCI
+cards.  In the case of ISA (PnP) cards, it's usually handled by the
+independent module (ns558).  Meanwhile, the ALSA PCI drivers have the
+built-in gameport support.  Hence, when the ALSA PCI driver is built
+in the kernel, CONFIG_GAMEPORT must be 'y', too.  Otherwise, the
+gameport support on that card will be (silently) disabled.
+
+Some adapter modules probe the physical connection of the device at
+the load time.  It'd be safer to plug in the joystick device before
+loading the module.
+
+
+PCI Cards
+---------
+
+For PCI cards, the joystick is enabled when the appropriate module
+option is specified.  Some drivers don't need options, and the
+joystick support is always enabled.  In the former ALSA version, there
+was a dynamic control API for the joystick activation.  It was
+changed, however, to the static module options because of the system
+stability and the resource management.
+
+The following PCI drivers support the joystick natively.
+
+==============	=============	============================================
+Driver		Module Option	Available Values
+==============	=============	============================================
+als4000		joystick_port	0 = disable (default), 1 = auto-detect,
+	                        manual: any address (e.g. 0x200)
+au88x0		N/A		N/A
+azf3328		joystick	0 = disable, 1 = enable, -1 = auto (default)
+ens1370		joystick	0 = disable (default), 1 = enable
+ens1371		joystick_port	0 = disable (default), 1 = auto-detect,
+	                        manual: 0x200, 0x208, 0x210, 0x218
+cmipci		joystick_port	0 = disable (default), 1 = auto-detect,
+	                        manual: any address (e.g. 0x200)
+cs4281		N/A		N/A
+cs46xx		N/A		N/A
+es1938		N/A		N/A
+es1968		joystick	0 = disable (default), 1 = enable
+sonicvibes	N/A		N/A
+trident		N/A		N/A
+via82xx [#f1]_	joystick	0 = disable (default), 1 = enable
+ymfpci		joystick_port	0 = disable (default), 1 = auto-detect,
+	                        manual: 0x201, 0x202, 0x204, 0x205 [#f2]_
+==============	=============	============================================
+
+.. [#f1] VIA686A/B only
+.. [#f2] With YMF744/754 chips, the port address can be chosen arbitrarily
+
+The following drivers don't support gameport natively, but there are
+additional modules.  Load the corresponding module to add the gameport
+support.
+
+=======	=================
+Driver	Additional Module
+=======	=================
+emu10k1	emu10k1-gp
+fm801	fm801-gp
+=======	=================
+
+Note: the "pcigame" and "cs461x" modules are for the OSS drivers only.
+These ALSA drivers (cs46xx, trident and au88x0) have the
+built-in gameport support.
+
+As mentioned above, ALSA PCI drivers have the built-in gameport
+support, so you don't have to load ns558 module.  Just load "joydev"
+and the appropriate adapter module (e.g. "analog").
+
+
+ISA Cards
+---------
+
+ALSA ISA drivers don't have the built-in gameport support.
+Instead, you need to load "ns558" module in addition to "joydev" and
+the adapter module (e.g. "analog").
diff --git a/Documentation/sound/index.rst b/Documentation/sound/index.rst
index e9fbbfff9d0d..1f5d166f81c4 100644
--- a/Documentation/sound/index.rst
+++ b/Documentation/sound/index.rst
@@ -9,6 +9,7 @@ Linux Sound Subsystem Documentation
    designs/index
    alsa-configuration
    hd-audio/index
+   cards/index
 
 .. only::  subproject
 
-- 
cgit v1.2.3-58-ga151


From 95ee717a8937d1a15ad825e5a6fe8cf0befde290 Mon Sep 17 00:00:00 2001
From: Takashi Iwai 
Date: Wed, 9 Nov 2016 17:12:34 +0100
Subject: ALSA: doc: ReSTize CMIPCI document

A simple conversion from a plain text file.
Put to cards subdirectory.

Signed-off-by: Takashi Iwai 
---
 Documentation/sound/alsa/CMIPCI.txt  | 254 --------------------------------
 Documentation/sound/cards/cmipci.rst | 272 +++++++++++++++++++++++++++++++++++
 Documentation/sound/cards/index.rst  |   1 +
 3 files changed, 273 insertions(+), 254 deletions(-)
 delete mode 100644 Documentation/sound/alsa/CMIPCI.txt
 create mode 100644 Documentation/sound/cards/cmipci.rst

(limited to 'Documentation/sound')

diff --git a/Documentation/sound/alsa/CMIPCI.txt b/Documentation/sound/alsa/CMIPCI.txt
deleted file mode 100644
index 4e36e6e809ca..000000000000
--- a/Documentation/sound/alsa/CMIPCI.txt
+++ /dev/null
@@ -1,254 +0,0 @@
-         Brief Notes on C-Media 8338/8738/8768/8770 Driver
-         =================================================
-
-                   Takashi Iwai 
-
-
-Front/Rear Multi-channel Playback
----------------------------------
-
-CM8x38 chip can use ADC as the second DAC so that two different stereo
-channels can be used for front/rear playbacks.  Since there are two
-DACs, both streams are handled independently unlike the 4/6ch multi-
-channel playbacks in the section below.
-
-As default, ALSA driver assigns the first PCM device (i.e. hw:0,0 for
-card#0) for front and 4/6ch playbacks, while the second PCM device
-(hw:0,1) is assigned to the second DAC for rear playback.
-
-There are slight differences between the two DACs:
-
-- The first DAC supports U8 and S16LE formats, while the second DAC
-  supports only S16LE.
-- The second DAC supports only two channel stereo.
-
-Please note that the CM8x38 DAC doesn't support continuous playback
-rate but only fixed rates: 5512, 8000, 11025, 16000, 22050, 32000,
-44100 and 48000 Hz.
-
-The rear output can be heard only when "Four Channel Mode" switch is
-disabled.  Otherwise no signal will be routed to the rear speakers.
-As default it's turned on.
-
-*** WARNING ***
-When "Four Channel Mode" switch is off, the output from rear speakers
-will be FULL VOLUME regardless of Master and PCM volumes.
-This might damage your audio equipment.  Please disconnect speakers
-before your turn off this switch.
-*** WARNING ***
-
-[ Well.. I once got the output with correct volume (i.e. same with the
-  front one) and was so excited.  It was even with "Four Channel" bit
-  on and "double DAC" mode.  Actually I could hear separate 4 channels
-  from front and rear speakers!  But.. after reboot, all was gone.
-  It's a very pity that I didn't save the register dump at that
-  time..  Maybe there is an unknown register to achieve this... ]
-
-If your card has an extra output jack for the rear output, the rear
-playback should be routed there as default.  If not, there is a
-control switch in the driver "Line-In As Rear", which you can change
-via alsamixer or somewhat else.  When this switch is on, line-in jack
-is used as rear output.
-
-There are two more controls regarding to the rear output.
-The "Exchange DAC" switch is used to exchange front and rear playback
-routes, i.e. the 2nd DAC is output from front output.
-
-
-4/6 Multi-Channel Playback
---------------------------
-
-The recent CM8738 chips support for the 4/6 multi-channel playback
-function.  This is useful especially for AC3 decoding.
-
-When the multi-channel is supported, the driver name has a suffix
-"-MC" such like "CMI8738-MC6".  You can check this name from
-/proc/asound/cards.
-
-When the 4/6-ch output is enabled, the second DAC accepts up to 6 (or
-4) channels.  While the dual DAC supports two different rates or
-formats, the 4/6-ch playback supports only the same condition for all
-channels.  Since the multi-channel playback mode uses both DACs, you
-cannot operate with full-duplex.
-
-The 4.0 and 5.1 modes are defined as the pcm "surround40" and "surround51"
-in alsa-lib.  For example, you can play a WAV file with 6 channels like
-
-	% aplay -Dsurround51 sixchannels.wav
-
-For programming the 4/6 channel playback, you need to specify the PCM
-channels as you like and set the format S16LE.  For example, for playback
-with 4 channels,
-
-	snd_pcm_hw_params_set_access(pcm, hw, SND_PCM_ACCESS_RW_INTERLEAVED);
-	    // or mmap if you like
-	snd_pcm_hw_params_set_format(pcm, hw, SND_PCM_FORMAT_S16_LE);
-	snd_pcm_hw_params_set_channels(pcm, hw, 4);
-
-and use the interleaved 4 channel data.
-
-There are some control switches affecting to the speaker connections:
-
-"Line-In Mode"	- an enum control to change the behavior of line-in
-	jack.  Either "Line-In", "Rear Output" or "Bass Output" can
-	be selected.  The last item is available only with model 039
-	or newer. 
-	When "Rear Output" is chosen, the surround channels 3 and 4
-	are output to line-in jack.
-"Mic-In Mode"	- an enum control to change the behavior of mic-in
-	jack.  Either "Mic-In" or "Center/LFE Output" can be
-	selected. 
-	When "Center/LFE Output" is chosen, the center and bass
-	channels (channels 5 and 6) are output to mic-in jack. 
-
-Digital I/O
------------
-
-The CM8x38 provides the excellent SPDIF capability with very cheap
-price (yes, that's the reason I bought the card :)
-
-The SPDIF playback and capture are done via the third PCM device
-(hw:0,2).  Usually this is assigned to the PCM device "spdif".
-The available rates are 44100 and 48000 Hz.
-For playback with aplay, you can run like below:
-
-	% aplay -Dhw:0,2 foo.wav
-
-or
-
-	% aplay -Dspdif foo.wav
-
-24bit format is also supported experimentally.
-
-The playback and capture over SPDIF use normal DAC and ADC,
-respectively, so you cannot playback both analog and digital streams
-simultaneously.
-
-To enable SPDIF output, you need to turn on "IEC958 Output Switch"
-control via mixer or alsactl ("IEC958" is the official name of
-so-called S/PDIF).  Then you'll see the red light on from the card so
-you know that's working obviously :)
-The SPDIF input is always enabled, so you can hear SPDIF input data
-from line-out with "IEC958 In Monitor" switch at any time (see
-below).
-
-You can play via SPDIF even with the first device (hw:0,0),
-but SPDIF is enabled only when the proper format (S16LE), sample rate
-(441100 or 48000) and channels (2) are used.  Otherwise it's turned
-off.  (Also don't forget to turn on "IEC958 Output Switch", too.)
-
-
-Additionally there are relevant control switches:
-
-"IEC958 Mix Analog" - Mix analog PCM playback and FM-OPL/3 streams and
-	output through SPDIF.  This switch appears only on old chip
-	models (CM8738 033 and 037).
-	Note: without this control you can output PCM to SPDIF.
-	This is "mixing" of streams, so e.g. it's not for AC3 output
-	(see the next section).
-
-"IEC958 In Select"  - Select SPDIF input, the internal CD-in (false)
-	and the external input (true).
-
-"IEC958 Loop"       - SPDIF input data is loop back into SPDIF
-	output (aka bypass)
-
-"IEC958 Copyright"  - Set the copyright bit.
-
-"IEC958 5V"         - Select 0.5V (coax) or 5V (optical) interface.
-	On some cards this doesn't work and you need to change the
-	configuration with hardware dip-switch.
-
-"IEC958 In Monitor" - SPDIF input is routed to DAC.
-
-"IEC958 In Phase Inverse" - Set SPDIF input format as inverse.
-	[FIXME: this doesn't work on all chips..]
-
-"IEC958 In Valid"   - Set input validity flag detection.
-
-Note: When "PCM Playback Switch" is on, you'll hear the digital output
-stream through analog line-out.
-
-
-The AC3 (RAW DIGITAL) OUTPUT
-----------------------------
-
-The driver supports raw digital (typically AC3) i/o over SPDIF.  This
-can be toggled via IEC958 playback control, but usually you need to
-access it via alsa-lib.  See alsa-lib documents for more details.
-
-On the raw digital mode, the "PCM Playback Switch" is automatically
-turned off so that non-audio data is heard from the analog line-out.
-Similarly the following switches are off: "IEC958 Mix Analog" and
-"IEC958 Loop".  The switches are resumed after closing the SPDIF PCM
-device automatically to the previous state.
-
-On the model 033, AC3 is implemented by the software conversion in
-the alsa-lib.  If you need to bypass the software conversion of IEC958
-subframes, pass the "soft_ac3=0" module option.  This doesn't matter
-on the newer models.
-
-
-ANALOG MIXER INTERFACE
-----------------------
-
-The mixer interface on CM8x38 is similar to SB16.
-There are Master, PCM, Synth, CD, Line, Mic and PC Speaker playback
-volumes.  Synth, CD, Line and Mic have playback and capture switches,
-too, as well as SB16.
-
-In addition to the standard SB mixer, CM8x38 provides more functions.
-- PCM playback switch
-- PCM capture switch (to capture the data sent to DAC)
-- Mic Boost switch
-- Mic capture volume
-- Aux playback volume/switch and capture switch
-- 3D control switch
-
-
-MIDI CONTROLLER
----------------
-
-With CMI8338 chips, the MPU401-UART interface is disabled as default.
-You need to set the module option "mpu_port" to a valid I/O port address
-to enable MIDI support.  Valid I/O ports are 0x300, 0x310, 0x320 and
-0x330.  Choose a value that doesn't conflict with other cards.
-
-With CMI8738 and newer chips, the MIDI interface is enabled by default
-and the driver automatically chooses a port address.
-
-There is _no_ hardware wavetable function on this chip (except for
-OPL3 synth below).
-What's said as MIDI synth on Windows is a software synthesizer
-emulation.  On Linux use TiMidity or other softsynth program for
-playing MIDI music.
-
-
-FM OPL/3 Synth
---------------
-
-The FM OPL/3 is also enabled as default only for the first card.
-Set "fm_port" module option for more cards.
-
-The output quality of FM OPL/3 is, however, very weird.
-I don't know why..
-
-CMI8768 and newer chips do not have the FM synth.
-
-
-Joystick and Modem
-------------------
-
-The legacy joystick is supported.  To enable the joystick support, pass
-joystick_port=1 module option.  The value 1 means the auto-detection.
-If the auto-detection fails, try to pass the exact I/O address.
-
-The modem is enabled dynamically via a card control switch "Modem".
-
-
-Debugging Information
----------------------
-
-The registers are shown in /proc/asound/cardX/cmipci.  If you have any
-problem (especially unexpected behavior of mixer), please attach the
-output of this proc file together with the bug report.
diff --git a/Documentation/sound/cards/cmipci.rst b/Documentation/sound/cards/cmipci.rst
new file mode 100644
index 000000000000..9ea1de6ec4ce
--- /dev/null
+++ b/Documentation/sound/cards/cmipci.rst
@@ -0,0 +1,272 @@
+=================================================
+Brief Notes on C-Media 8338/8738/8768/8770 Driver
+=================================================
+
+Takashi Iwai 
+
+
+Front/Rear Multi-channel Playback
+---------------------------------
+
+CM8x38 chip can use ADC as the second DAC so that two different stereo
+channels can be used for front/rear playbacks.  Since there are two
+DACs, both streams are handled independently unlike the 4/6ch multi-
+channel playbacks in the section below.
+
+As default, ALSA driver assigns the first PCM device (i.e. hw:0,0 for
+card#0) for front and 4/6ch playbacks, while the second PCM device
+(hw:0,1) is assigned to the second DAC for rear playback.
+
+There are slight differences between the two DACs:
+
+- The first DAC supports U8 and S16LE formats, while the second DAC
+  supports only S16LE.
+- The second DAC supports only two channel stereo.
+
+Please note that the CM8x38 DAC doesn't support continuous playback
+rate but only fixed rates: 5512, 8000, 11025, 16000, 22050, 32000,
+44100 and 48000 Hz.
+
+The rear output can be heard only when "Four Channel Mode" switch is
+disabled.  Otherwise no signal will be routed to the rear speakers.
+As default it's turned on.
+
+.. WARNING::
+  When "Four Channel Mode" switch is off, the output from rear speakers
+  will be FULL VOLUME regardless of Master and PCM volumes [#]_.
+  This might damage your audio equipment.  Please disconnect speakers
+  before your turn off this switch.
+
+
+.. [#]
+  Well.. I once got the output with correct volume (i.e. same with the
+  front one) and was so excited.  It was even with "Four Channel" bit
+  on and "double DAC" mode.  Actually I could hear separate 4 channels
+  from front and rear speakers!  But.. after reboot, all was gone.
+  It's a very pity that I didn't save the register dump at that
+  time..  Maybe there is an unknown register to achieve this...
+
+If your card has an extra output jack for the rear output, the rear
+playback should be routed there as default.  If not, there is a
+control switch in the driver "Line-In As Rear", which you can change
+via alsamixer or somewhat else.  When this switch is on, line-in jack
+is used as rear output.
+
+There are two more controls regarding to the rear output.
+The "Exchange DAC" switch is used to exchange front and rear playback
+routes, i.e. the 2nd DAC is output from front output.
+
+
+4/6 Multi-Channel Playback
+--------------------------
+
+The recent CM8738 chips support for the 4/6 multi-channel playback
+function.  This is useful especially for AC3 decoding.
+
+When the multi-channel is supported, the driver name has a suffix
+"-MC" such like "CMI8738-MC6".  You can check this name from
+/proc/asound/cards.
+
+When the 4/6-ch output is enabled, the second DAC accepts up to 6 (or
+4) channels.  While the dual DAC supports two different rates or
+formats, the 4/6-ch playback supports only the same condition for all
+channels.  Since the multi-channel playback mode uses both DACs, you
+cannot operate with full-duplex.
+
+The 4.0 and 5.1 modes are defined as the pcm "surround40" and "surround51"
+in alsa-lib.  For example, you can play a WAV file with 6 channels like
+::
+
+	% aplay -Dsurround51 sixchannels.wav
+
+For programming the 4/6 channel playback, you need to specify the PCM
+channels as you like and set the format S16LE.  For example, for playback
+with 4 channels,
+::
+
+	snd_pcm_hw_params_set_access(pcm, hw, SND_PCM_ACCESS_RW_INTERLEAVED);
+	    // or mmap if you like
+	snd_pcm_hw_params_set_format(pcm, hw, SND_PCM_FORMAT_S16_LE);
+	snd_pcm_hw_params_set_channels(pcm, hw, 4);
+
+and use the interleaved 4 channel data.
+
+There are some control switches affecting to the speaker connections:
+
+Line-In Mode
+	an enum control to change the behavior of line-in
+	jack.  Either "Line-In", "Rear Output" or "Bass Output" can
+	be selected.  The last item is available only with model 039
+	or newer. 
+	When "Rear Output" is chosen, the surround channels 3 and 4
+	are output to line-in jack.
+Mic-In Mode
+	an enum control to change the behavior of mic-in
+	jack.  Either "Mic-In" or "Center/LFE Output" can be
+	selected. 
+	When "Center/LFE Output" is chosen, the center and bass
+	channels (channels 5 and 6) are output to mic-in jack. 
+
+Digital I/O
+-----------
+
+The CM8x38 provides the excellent SPDIF capability with very cheap
+price (yes, that's the reason I bought the card :)
+
+The SPDIF playback and capture are done via the third PCM device
+(hw:0,2).  Usually this is assigned to the PCM device "spdif".
+The available rates are 44100 and 48000 Hz.
+For playback with aplay, you can run like below:
+::
+
+	% aplay -Dhw:0,2 foo.wav
+
+or
+
+::
+
+	% aplay -Dspdif foo.wav
+
+24bit format is also supported experimentally.
+
+The playback and capture over SPDIF use normal DAC and ADC,
+respectively, so you cannot playback both analog and digital streams
+simultaneously.
+
+To enable SPDIF output, you need to turn on "IEC958 Output Switch"
+control via mixer or alsactl ("IEC958" is the official name of
+so-called S/PDIF).  Then you'll see the red light on from the card so
+you know that's working obviously :)
+The SPDIF input is always enabled, so you can hear SPDIF input data
+from line-out with "IEC958 In Monitor" switch at any time (see
+below).
+
+You can play via SPDIF even with the first device (hw:0,0),
+but SPDIF is enabled only when the proper format (S16LE), sample rate
+(441100 or 48000) and channels (2) are used.  Otherwise it's turned
+off.  (Also don't forget to turn on "IEC958 Output Switch", too.)
+
+
+Additionally there are relevant control switches:
+
+IEC958 Mix Analog
+	Mix analog PCM playback and FM-OPL/3 streams and
+	output through SPDIF.  This switch appears only on old chip
+	models (CM8738 033 and 037).
+
+	Note: without this control you can output PCM to SPDIF.
+	This is "mixing" of streams, so e.g. it's not for AC3 output
+	(see the next section).
+
+IEC958 In Select
+	Select SPDIF input, the internal CD-in (false)
+	and the external input (true).
+
+IEC958 Loop
+	SPDIF input data is loop back into SPDIF
+	output (aka bypass)
+
+IEC958 Copyright
+	Set the copyright bit.
+
+IEC958 5V
+	Select 0.5V (coax) or 5V (optical) interface.
+	On some cards this doesn't work and you need to change the
+	configuration with hardware dip-switch.
+
+IEC958 In Monitor
+	SPDIF input is routed to DAC.
+
+IEC958 In Phase Inverse
+	Set SPDIF input format as inverse.
+	[FIXME: this doesn't work on all chips..]
+
+IEC958 In Valid
+	Set input validity flag detection.
+
+Note: When "PCM Playback Switch" is on, you'll hear the digital output
+stream through analog line-out.
+
+
+The AC3 (RAW DIGITAL) OUTPUT
+----------------------------
+
+The driver supports raw digital (typically AC3) i/o over SPDIF.  This
+can be toggled via IEC958 playback control, but usually you need to
+access it via alsa-lib.  See alsa-lib documents for more details.
+
+On the raw digital mode, the "PCM Playback Switch" is automatically
+turned off so that non-audio data is heard from the analog line-out.
+Similarly the following switches are off: "IEC958 Mix Analog" and
+"IEC958 Loop".  The switches are resumed after closing the SPDIF PCM
+device automatically to the previous state.
+
+On the model 033, AC3 is implemented by the software conversion in
+the alsa-lib.  If you need to bypass the software conversion of IEC958
+subframes, pass the "soft_ac3=0" module option.  This doesn't matter
+on the newer models.
+
+
+ANALOG MIXER INTERFACE
+----------------------
+
+The mixer interface on CM8x38 is similar to SB16.
+There are Master, PCM, Synth, CD, Line, Mic and PC Speaker playback
+volumes.  Synth, CD, Line and Mic have playback and capture switches,
+too, as well as SB16.
+
+In addition to the standard SB mixer, CM8x38 provides more functions.
+- PCM playback switch
+- PCM capture switch (to capture the data sent to DAC)
+- Mic Boost switch
+- Mic capture volume
+- Aux playback volume/switch and capture switch
+- 3D control switch
+
+
+MIDI CONTROLLER
+---------------
+
+With CMI8338 chips, the MPU401-UART interface is disabled as default.
+You need to set the module option "mpu_port" to a valid I/O port address
+to enable MIDI support.  Valid I/O ports are 0x300, 0x310, 0x320 and
+0x330.  Choose a value that doesn't conflict with other cards.
+
+With CMI8738 and newer chips, the MIDI interface is enabled by default
+and the driver automatically chooses a port address.
+
+There is *no* hardware wavetable function on this chip (except for
+OPL3 synth below).
+What's said as MIDI synth on Windows is a software synthesizer
+emulation.  On Linux use TiMidity or other softsynth program for
+playing MIDI music.
+
+
+FM OPL/3 Synth
+--------------
+
+The FM OPL/3 is also enabled as default only for the first card.
+Set "fm_port" module option for more cards.
+
+The output quality of FM OPL/3 is, however, very weird.
+I don't know why..
+
+CMI8768 and newer chips do not have the FM synth.
+
+
+Joystick and Modem
+------------------
+
+The legacy joystick is supported.  To enable the joystick support, pass
+joystick_port=1 module option.  The value 1 means the auto-detection.
+If the auto-detection fails, try to pass the exact I/O address.
+
+The modem is enabled dynamically via a card control switch "Modem".
+
+
+Debugging Information
+---------------------
+
+The registers are shown in /proc/asound/cardX/cmipci.  If you have any
+problem (especially unexpected behavior of mixer), please attach the
+output of this proc file together with the bug report.
diff --git a/Documentation/sound/cards/index.rst b/Documentation/sound/cards/index.rst
index e1f4b78c75d6..67a3073157b9 100644
--- a/Documentation/sound/cards/index.rst
+++ b/Documentation/sound/cards/index.rst
@@ -5,3 +5,4 @@ Card-Specific Information
    :maxdepth: 2
 
    joystick
+   cmipci
-- 
cgit v1.2.3-58-ga151


From ecef1481d516e004a38d9472c403205dcdd1491e Mon Sep 17 00:00:00 2001
From: Takashi Iwai 
Date: Thu, 10 Nov 2016 16:17:56 +0100
Subject: ALSA: doc: ReSTize SB-Live-mixer document

Another simple conversion from a plain text file.
Put to cards subdirectory.

Signed-off-by: Takashi Iwai 
---
 Documentation/sound/alsa/SB-Live-mixer.txt  | 356 --------------------------
 Documentation/sound/cards/index.rst         |   1 +
 Documentation/sound/cards/sb-live-mixer.rst | 373 ++++++++++++++++++++++++++++
 3 files changed, 374 insertions(+), 356 deletions(-)
 delete mode 100644 Documentation/sound/alsa/SB-Live-mixer.txt
 create mode 100644 Documentation/sound/cards/sb-live-mixer.rst

(limited to 'Documentation/sound')

diff --git a/Documentation/sound/alsa/SB-Live-mixer.txt b/Documentation/sound/alsa/SB-Live-mixer.txt
deleted file mode 100644
index f4b5988f450c..000000000000
--- a/Documentation/sound/alsa/SB-Live-mixer.txt
+++ /dev/null
@@ -1,356 +0,0 @@
-
-		Sound Blaster Live mixer / default DSP code
-		===========================================
-
-
-The EMU10K1 chips have a DSP part which can be programmed to support
-various ways of sample processing, which is described here.
-(This article does not deal with the overall functionality of the 
-EMU10K1 chips. See the manuals section for further details.)
-
-The ALSA driver programs this portion of chip by default code
-(can be altered later) which offers the following functionality:
-
-
-1) IEC958 (S/PDIF) raw PCM
---------------------------
-
-This PCM device (it's the 4th PCM device (index 3!) and first subdevice
-(index 0) for a given card) allows to forward 48kHz, stereo, 16-bit
-little endian streams without any modifications to the digital output
-(coaxial or optical). The universal interface allows the creation of up
-to 8 raw PCM devices operating at 48kHz, 16-bit little endian. It would
-be easy to add support for multichannel devices to the current code,
-but the conversion routines exist only for stereo (2-channel streams)
-at the time. 
-
-Look to tram_poke routines in lowlevel/emu10k1/emufx.c for more details.
-
-
-2) Digital mixer controls
--------------------------
-
-These controls are built using the DSP instructions. They offer extended
-functionality. Only the default build-in code in the ALSA driver is described
-here. Note that the controls work as attenuators: the maximum value is the 
-neutral position leaving the signal unchanged. Note that if the  same destination 
-is mentioned in multiple controls, the signal is accumulated and can be wrapped 
-(set to maximal or minimal value without checking of overflow).
-
-
-Explanation of used abbreviations:
-
-DAC    - digital to analog converter
-ADC    - analog to digital converter
-I2S    - one-way three wire serial bus for digital sound by Philips Semiconductors
-         (this standard is used for connecting standalone DAC and ADC converters)
-LFE    - low frequency effects (subwoofer signal)
-AC97   - a chip containing an analog mixer, DAC and ADC converters
-IEC958 - S/PDIF
-FX-bus - the EMU10K1 chip has an effect bus containing 16 accumulators.
-         Each of the synthesizer voices can feed its output to these accumulators
-         and the DSP microcontroller can operate with the resulting sum.
-
-
-name='Wave Playback Volume',index=0
-
-This control is used to attenuate samples for left and right PCM FX-bus
-accumulators. ALSA uses accumulators 0 and 1 for left and right PCM samples.
-The result samples are forwarded to the front DAC PCM slots of the AC97 codec.
-
-name='Wave Surround Playback Volume',index=0
-
-This control is used to attenuate samples for left and right PCM FX-bus
-accumulators. ALSA uses accumulators 0 and 1 for left and right PCM samples.
-The result samples are forwarded to the rear I2S DACs. These DACs operates
-separately (they are not inside the AC97 codec).
-
-name='Wave Center Playback Volume',index=0
-
-This control is used to attenuate samples for left and right PCM FX-bus
-accumulators. ALSA uses accumulators 0 and 1 for left and right PCM samples.
-The result is mixed to mono signal (single channel) and forwarded to
-the ??rear?? right DAC PCM slot of the AC97 codec.
-
-name='Wave LFE Playback Volume',index=0
-
-This control is used to attenuate samples for left and right PCM FX-bus
-accumulators. ALSA uses accumulators 0 and 1 for left and right PCM.
-The result is mixed to mono signal (single channel) and forwarded to
-the ??rear?? left DAC PCM slot of the AC97 codec.
-
-name='Wave Capture Volume',index=0
-name='Wave Capture Switch',index=0
-
-These controls are used to attenuate samples for left and right PCM FX-bus
-accumulator. ALSA uses accumulators 0 and 1 for left and right PCM.
-The result is forwarded to the ADC capture FIFO (thus to the standard capture
-PCM device).
-
-name='Synth Playback Volume',index=0
-
-This control is used to attenuate samples for left and right MIDI FX-bus
-accumulators. ALSA uses accumulators 4 and 5 for left and right MIDI samples.
-The result samples are forwarded to the front DAC PCM slots of the AC97 codec.
-
-name='Synth Capture Volume',index=0
-name='Synth Capture Switch',index=0
-
-These controls are used to attenuate samples for left and right MIDI FX-bus
-accumulator. ALSA uses accumulators 4 and 5 for left and right PCM.
-The result is forwarded to the ADC capture FIFO (thus to the standard capture
-PCM device).
-
-name='Surround Playback Volume',index=0
-
-This control is used to attenuate samples for left and right rear PCM FX-bus
-accumulators. ALSA uses accumulators 2 and 3 for left and right rear PCM samples.
-The result samples are forwarded to the rear I2S DACs. These DACs operate
-separately (they are not inside the AC97 codec).
-
-name='Surround Capture Volume',index=0
-name='Surround Capture Switch',index=0
-
-These controls are used to attenuate samples for left and right rear PCM FX-bus
-accumulators. ALSA uses accumulators 2 and 3 for left and right rear PCM samples.
-The result is forwarded to the ADC capture FIFO (thus to the standard capture
-PCM device).
-
-name='Center Playback Volume',index=0
-
-This control is used to attenuate sample for center PCM FX-bus accumulator.
-ALSA uses accumulator 6 for center PCM sample. The result sample is forwarded
-to the ??rear?? right DAC PCM slot of the AC97 codec.
-
-name='LFE Playback Volume',index=0
-
-This control is used to attenuate sample for center PCM FX-bus accumulator.
-ALSA uses accumulator 6 for center PCM sample. The result sample is forwarded
-to the ??rear?? left DAC PCM slot of the AC97 codec.
-
-name='AC97 Playback Volume',index=0
-
-This control is used to attenuate samples for left and right front ADC PCM slots
-of the AC97 codec. The result samples are forwarded to the front DAC PCM
-slots of the AC97 codec.
-********************************************************************************
-*** Note: This control should be zero for the standard operations, otherwise ***
-*** a digital loopback is activated.                                         ***
-********************************************************************************
-
-name='AC97 Capture Volume',index=0
-
-This control is used to attenuate samples for left and right front ADC PCM slots
-of the AC97 codec. The result is forwarded to the ADC capture FIFO (thus to
-the standard capture PCM device).
-********************************************************************************
-*** Note: This control should be 100 (maximal value), otherwise no analog    ***
-*** inputs of the AC97 codec can be captured (recorded).                     ***
-********************************************************************************
-
-name='IEC958 TTL Playback Volume',index=0
-
-This control is used to attenuate samples from left and right IEC958 TTL
-digital inputs (usually used by a CDROM drive). The result samples are
-forwarded to the front DAC PCM slots of the AC97 codec.
-
-name='IEC958 TTL Capture Volume',index=0
-
-This control is used to attenuate samples from left and right IEC958 TTL
-digital inputs (usually used by a CDROM drive). The result samples are
-forwarded to the ADC capture FIFO (thus to the standard capture PCM device).
-
-name='Zoom Video Playback Volume',index=0
-
-This control is used to attenuate samples from left and right zoom video
-digital inputs (usually used by a CDROM drive). The result samples are
-forwarded to the front DAC PCM slots of the AC97 codec.
-
-name='Zoom Video Capture Volume',index=0
-
-This control is used to attenuate samples from left and right zoom video
-digital inputs (usually used by a CDROM drive). The result samples are
-forwarded to the ADC capture FIFO (thus to the standard capture PCM device).
-
-name='IEC958 LiveDrive Playback Volume',index=0
-
-This control is used to attenuate samples from left and right IEC958 optical
-digital input. The result samples are forwarded to the front DAC PCM slots
-of the AC97 codec.
-
-name='IEC958 LiveDrive Capture Volume',index=0
-
-This control is used to attenuate samples from left and right IEC958 optical
-digital inputs. The result samples are forwarded to the ADC capture FIFO
-(thus to the standard capture PCM device).
-
-name='IEC958 Coaxial Playback Volume',index=0
-
-This control is used to attenuate samples from left and right IEC958 coaxial
-digital inputs. The result samples are forwarded to the front DAC PCM slots
-of the AC97 codec.
-
-name='IEC958 Coaxial Capture Volume',index=0
-
-This control is used to attenuate samples from left and right IEC958 coaxial
-digital inputs. The result samples are forwarded to the ADC capture FIFO
-(thus to the standard capture PCM device).
-
-name='Line LiveDrive Playback Volume',index=0
-name='Line LiveDrive Playback Volume',index=1
-
-This control is used to attenuate samples from left and right I2S ADC
-inputs (on the LiveDrive). The result samples are forwarded to the front
-DAC PCM slots of the AC97 codec.
-
-name='Line LiveDrive Capture Volume',index=1
-name='Line LiveDrive Capture Volume',index=1
-
-This control is used to attenuate samples from left and right I2S ADC
-inputs (on the LiveDrive). The result samples are forwarded to the ADC
-capture FIFO (thus to the standard capture PCM device).
-
-name='Tone Control - Switch',index=0
-
-This control turns the tone control on or off. The samples for front, rear
-and center / LFE outputs are affected.
-
-name='Tone Control - Bass',index=0
-
-This control sets the bass intensity. There is no neutral value!!
-When the tone control code is activated, the samples are always modified.
-The closest value to pure signal is 20.
-
-name='Tone Control - Treble',index=0
-
-This control sets the treble intensity. There is no neutral value!!
-When the tone control code is activated, the samples are always modified.
-The closest value to pure signal is 20.
-
-name='IEC958 Optical Raw Playback Switch',index=0
-
-If this switch is on, then the samples for the IEC958 (S/PDIF) digital
-output are taken only from the raw FX8010 PCM, otherwise standard front
-PCM samples are taken.
-
-name='Headphone Playback Volume',index=1
-
-This control attenuates the samples for the headphone output.
-
-name='Headphone Center Playback Switch',index=1
-
-If this switch is on, then the sample for the center PCM is put to the
-left headphone output (useful for SB Live cards without separate center/LFE
-output).
-
-name='Headphone LFE Playback Switch',index=1
-
-If this switch is on, then the sample for the center PCM is put to the
-right headphone output (useful for SB Live cards without separate center/LFE
-output).
-
-
-3) PCM stream related controls
-------------------------------
-
-name='EMU10K1 PCM Volume',index 0-31
-
-Channel volume attenuation in range 0-0xffff. The maximum value (no
-attenuation) is default. The channel mapping for three values is
-as follows:
-
-	0 - mono, default 0xffff (no attenuation)
-	1 - left, default 0xffff (no attenuation)
-	2 - right, default 0xffff (no attenuation)
-
-name='EMU10K1 PCM Send Routing',index 0-31
-
-This control specifies the destination - FX-bus accumulators. There are
-twelve values with this mapping:
-
-	 0 -  mono, A destination (FX-bus 0-15), default 0
-	 1 -  mono, B destination (FX-bus 0-15), default 1
-	 2 -  mono, C destination (FX-bus 0-15), default 2
-	 3 -  mono, D destination (FX-bus 0-15), default 3
-	 4 -  left, A destination (FX-bus 0-15), default 0
-	 5 -  left, B destination (FX-bus 0-15), default 1
-	 6 -  left, C destination (FX-bus 0-15), default 2
-	 7 -  left, D destination (FX-bus 0-15), default 3
-	 8 - right, A destination (FX-bus 0-15), default 0
-	 9 - right, B destination (FX-bus 0-15), default 1
-	10 - right, C destination (FX-bus 0-15), default 2
-	11 - right, D destination (FX-bus 0-15), default 3
-
-Don't forget that it's illegal to assign a channel to the same FX-bus accumulator 
-more than once (it means 0=0 && 1=0 is an invalid combination).
- 
-name='EMU10K1 PCM Send Volume',index 0-31
-
-It specifies the attenuation (amount) for given destination in range 0-255.
-The channel mapping is following:
-
-	 0 -  mono, A destination attn, default 255 (no attenuation)
-	 1 -  mono, B destination attn, default 255 (no attenuation)
-	 2 -  mono, C destination attn, default 0 (mute)
-	 3 -  mono, D destination attn, default 0 (mute)
-	 4 -  left, A destination attn, default 255 (no attenuation)
-	 5 -  left, B destination attn, default 0 (mute)
-	 6 -  left, C destination attn, default 0 (mute)
-	 7 -  left, D destination attn, default 0 (mute)
-	 8 - right, A destination attn, default 0 (mute)
-	 9 - right, B destination attn, default 255 (no attenuation)
-	10 - right, C destination attn, default 0 (mute)
-	11 - right, D destination attn, default 0 (mute)
-
-
-
-4) MANUALS/PATENTS:
--------------------
-
-ftp://opensource.creative.com/pub/doc
--------------------------------------
-
-        Files:
-        LM4545.pdf      AC97 Codec
-
-        m2049.pdf       The EMU10K1 Digital Audio Processor
-
-        hog63.ps        FX8010 - A DSP Chip Architecture for Audio Effects
-
-
-WIPO Patents
-------------
-        Patent numbers:
-        WO 9901813 (A1) Audio Effects Processor with multiple asynchronous (Jan. 14, 1999)
-                        streams
-
-        WO 9901814 (A1) Processor with Instruction Set for Audio Effects (Jan. 14, 1999)
-
-        WO 9901953 (A1) Audio Effects Processor having Decoupled Instruction
-                        Execution and Audio Data Sequencing (Jan. 14, 1999)
-
-
-US Patents (http://www.uspto.gov/)
-----------------------------------
-
-        US 5925841      Digital Sampling Instrument employing cache memory (Jul. 20, 1999)
-
-        US 5928342      Audio Effects Processor integrated on a single chip (Jul. 27, 1999)
-                        with a multiport memory onto which multiple asynchronous
-                        digital sound samples can be concurrently loaded
-
-        US 5930158      Processor with Instruction Set for Audio Effects (Jul. 27, 1999)
-
-        US 6032235      Memory initialization circuit (Tram) (Feb. 29, 2000)
-
-        US 6138207      Interpolation looping of audio samples in cache connected to    (Oct. 24, 2000)
-                        system bus with prioritization and modification of bus transfers
-                        in accordance with loop ends and minimum block sizes
-
-        US 6151670      Method for conserving memory storage using a (Nov. 21, 2000)
-                        pool of  short term memory registers
-
-        US 6195715      Interrupt control for multiple programs communicating with      (Feb. 27, 2001)
-                        a common interrupt by associating programs to GP registers,
-                        defining interrupt register, polling GP registers, and invoking
-                        callback routine associated with defined interrupt register
diff --git a/Documentation/sound/cards/index.rst b/Documentation/sound/cards/index.rst
index 67a3073157b9..294efdb75bc9 100644
--- a/Documentation/sound/cards/index.rst
+++ b/Documentation/sound/cards/index.rst
@@ -6,3 +6,4 @@ Card-Specific Information
 
    joystick
    cmipci
+   sb-live-mixer
diff --git a/Documentation/sound/cards/sb-live-mixer.rst b/Documentation/sound/cards/sb-live-mixer.rst
new file mode 100644
index 000000000000..bcb62fc99bbb
--- /dev/null
+++ b/Documentation/sound/cards/sb-live-mixer.rst
@@ -0,0 +1,373 @@
+===========================================
+Sound Blaster Live mixer / default DSP code
+===========================================
+
+
+The EMU10K1 chips have a DSP part which can be programmed to support
+various ways of sample processing, which is described here.
+(This article does not deal with the overall functionality of the 
+EMU10K1 chips. See the manuals section for further details.)
+
+The ALSA driver programs this portion of chip by default code
+(can be altered later) which offers the following functionality:
+
+
+IEC958 (S/PDIF) raw PCM
+=======================
+
+This PCM device (it's the 4th PCM device (index 3!) and first subdevice
+(index 0) for a given card) allows to forward 48kHz, stereo, 16-bit
+little endian streams without any modifications to the digital output
+(coaxial or optical). The universal interface allows the creation of up
+to 8 raw PCM devices operating at 48kHz, 16-bit little endian. It would
+be easy to add support for multichannel devices to the current code,
+but the conversion routines exist only for stereo (2-channel streams)
+at the time. 
+
+Look to tram_poke routines in lowlevel/emu10k1/emufx.c for more details.
+
+
+Digital mixer controls
+======================
+
+These controls are built using the DSP instructions. They offer extended
+functionality. Only the default build-in code in the ALSA driver is described
+here. Note that the controls work as attenuators: the maximum value is the 
+neutral position leaving the signal unchanged. Note that if the  same destination 
+is mentioned in multiple controls, the signal is accumulated and can be wrapped 
+(set to maximal or minimal value without checking of overflow).
+
+
+Explanation of used abbreviations:
+
+DAC
+	digital to analog converter
+ADC
+	analog to digital converter
+I2S
+	one-way three wire serial bus for digital sound by Philips Semiconductors
+        (this standard is used for connecting standalone DAC and ADC converters)
+LFE
+	low frequency effects (subwoofer signal)
+AC97
+	a chip containing an analog mixer, DAC and ADC converters
+IEC958
+	S/PDIF
+FX-bus
+	the EMU10K1 chip has an effect bus containing 16 accumulators.
+	Each of the synthesizer voices can feed its output to these accumulators
+	and the DSP microcontroller can operate with the resulting sum.
+
+
+``name='Wave Playback Volume',index=0``
+---------------------------------------
+This control is used to attenuate samples for left and right PCM FX-bus
+accumulators. ALSA uses accumulators 0 and 1 for left and right PCM samples.
+The result samples are forwarded to the front DAC PCM slots of the AC97 codec.
+
+``name='Wave Surround Playback Volume',index=0``
+------------------------------------------------
+This control is used to attenuate samples for left and right PCM FX-bus
+accumulators. ALSA uses accumulators 0 and 1 for left and right PCM samples.
+The result samples are forwarded to the rear I2S DACs. These DACs operates
+separately (they are not inside the AC97 codec).
+
+``name='Wave Center Playback Volume',index=0``
+----------------------------------------------
+This control is used to attenuate samples for left and right PCM FX-bus
+accumulators. ALSA uses accumulators 0 and 1 for left and right PCM samples.
+The result is mixed to mono signal (single channel) and forwarded to
+the ??rear?? right DAC PCM slot of the AC97 codec.
+
+``name='Wave LFE Playback Volume',index=0``
+-------------------------------------------
+This control is used to attenuate samples for left and right PCM FX-bus
+accumulators. ALSA uses accumulators 0 and 1 for left and right PCM.
+The result is mixed to mono signal (single channel) and forwarded to
+the ??rear?? left DAC PCM slot of the AC97 codec.
+
+``name='Wave Capture Volume',index=0``, ``name='Wave Capture Switch',index=0``
+------------------------------------------------------------------------------
+These controls are used to attenuate samples for left and right PCM FX-bus
+accumulator. ALSA uses accumulators 0 and 1 for left and right PCM.
+The result is forwarded to the ADC capture FIFO (thus to the standard capture
+PCM device).
+
+``name='Synth Playback Volume',index=0``
+----------------------------------------
+This control is used to attenuate samples for left and right MIDI FX-bus
+accumulators. ALSA uses accumulators 4 and 5 for left and right MIDI samples.
+The result samples are forwarded to the front DAC PCM slots of the AC97 codec.
+
+``name='Synth Capture Volume',index=0``, ``name='Synth Capture Switch',index=0``
+--------------------------------------------------------------------------------
+These controls are used to attenuate samples for left and right MIDI FX-bus
+accumulator. ALSA uses accumulators 4 and 5 for left and right PCM.
+The result is forwarded to the ADC capture FIFO (thus to the standard capture
+PCM device).
+
+``name='Surround Playback Volume',index=0``
+-------------------------------------------
+This control is used to attenuate samples for left and right rear PCM FX-bus
+accumulators. ALSA uses accumulators 2 and 3 for left and right rear PCM samples.
+The result samples are forwarded to the rear I2S DACs. These DACs operate
+separately (they are not inside the AC97 codec).
+
+``name='Surround Capture Volume',index=0``, ``name='Surround Capture Switch',index=0``
+--------------------------------------------------------------------------------------
+These controls are used to attenuate samples for left and right rear PCM FX-bus
+accumulators. ALSA uses accumulators 2 and 3 for left and right rear PCM samples.
+The result is forwarded to the ADC capture FIFO (thus to the standard capture
+PCM device).
+
+``name='Center Playback Volume',index=0``
+-----------------------------------------
+This control is used to attenuate sample for center PCM FX-bus accumulator.
+ALSA uses accumulator 6 for center PCM sample. The result sample is forwarded
+to the ??rear?? right DAC PCM slot of the AC97 codec.
+
+``name='LFE Playback Volume',index=0``
+--------------------------------------
+This control is used to attenuate sample for center PCM FX-bus accumulator.
+ALSA uses accumulator 6 for center PCM sample. The result sample is forwarded
+to the ??rear?? left DAC PCM slot of the AC97 codec.
+
+``name='AC97 Playback Volume',index=0``
+---------------------------------------
+This control is used to attenuate samples for left and right front ADC PCM slots
+of the AC97 codec. The result samples are forwarded to the front DAC PCM
+slots of the AC97 codec.
+
+.. note::
+  This control should be zero for the standard operations, otherwise
+  a digital loopback is activated.
+
+
+``name='AC97 Capture Volume',index=0``
+--------------------------------------
+This control is used to attenuate samples for left and right front ADC PCM slots
+of the AC97 codec. The result is forwarded to the ADC capture FIFO (thus to
+the standard capture PCM device).
+
+.. note::
+   This control should be 100 (maximal value), otherwise no analog
+   inputs of the AC97 codec can be captured (recorded).
+
+``name='IEC958 TTL Playback Volume',index=0``
+---------------------------------------------
+This control is used to attenuate samples from left and right IEC958 TTL
+digital inputs (usually used by a CDROM drive). The result samples are
+forwarded to the front DAC PCM slots of the AC97 codec.
+
+``name='IEC958 TTL Capture Volume',index=0``
+--------------------------------------------
+This control is used to attenuate samples from left and right IEC958 TTL
+digital inputs (usually used by a CDROM drive). The result samples are
+forwarded to the ADC capture FIFO (thus to the standard capture PCM device).
+
+``name='Zoom Video Playback Volume',index=0``
+---------------------------------------------
+This control is used to attenuate samples from left and right zoom video
+digital inputs (usually used by a CDROM drive). The result samples are
+forwarded to the front DAC PCM slots of the AC97 codec.
+
+``name='Zoom Video Capture Volume',index=0``
+--------------------------------------------
+This control is used to attenuate samples from left and right zoom video
+digital inputs (usually used by a CDROM drive). The result samples are
+forwarded to the ADC capture FIFO (thus to the standard capture PCM device).
+
+``name='IEC958 LiveDrive Playback Volume',index=0``
+---------------------------------------------------
+This control is used to attenuate samples from left and right IEC958 optical
+digital input. The result samples are forwarded to the front DAC PCM slots
+of the AC97 codec.
+
+``name='IEC958 LiveDrive Capture Volume',index=0``
+--------------------------------------------------
+This control is used to attenuate samples from left and right IEC958 optical
+digital inputs. The result samples are forwarded to the ADC capture FIFO
+(thus to the standard capture PCM device).
+
+``name='IEC958 Coaxial Playback Volume',index=0``
+-------------------------------------------------
+This control is used to attenuate samples from left and right IEC958 coaxial
+digital inputs. The result samples are forwarded to the front DAC PCM slots
+of the AC97 codec.
+
+``name='IEC958 Coaxial Capture Volume',index=0``
+------------------------------------------------
+This control is used to attenuate samples from left and right IEC958 coaxial
+digital inputs. The result samples are forwarded to the ADC capture FIFO
+(thus to the standard capture PCM device).
+
+``name='Line LiveDrive Playback Volume',index=0``, ``name='Line LiveDrive Playback Volume',index=1``
+----------------------------------------------------------------------------------------------------
+This control is used to attenuate samples from left and right I2S ADC
+inputs (on the LiveDrive). The result samples are forwarded to the front
+DAC PCM slots of the AC97 codec.
+
+``name='Line LiveDrive Capture Volume',index=1``, ``name='Line LiveDrive Capture Volume',index=1``
+--------------------------------------------------------------------------------------------------
+This control is used to attenuate samples from left and right I2S ADC
+inputs (on the LiveDrive). The result samples are forwarded to the ADC
+capture FIFO (thus to the standard capture PCM device).
+
+``name='Tone Control - Switch',index=0``
+----------------------------------------
+This control turns the tone control on or off. The samples for front, rear
+and center / LFE outputs are affected.
+
+``name='Tone Control - Bass',index=0``
+--------------------------------------
+This control sets the bass intensity. There is no neutral value!!
+When the tone control code is activated, the samples are always modified.
+The closest value to pure signal is 20.
+
+``name='Tone Control - Treble',index=0``
+----------------------------------------
+This control sets the treble intensity. There is no neutral value!!
+When the tone control code is activated, the samples are always modified.
+The closest value to pure signal is 20.
+
+``name='IEC958 Optical Raw Playback Switch',index=0``
+-----------------------------------------------------
+If this switch is on, then the samples for the IEC958 (S/PDIF) digital
+output are taken only from the raw FX8010 PCM, otherwise standard front
+PCM samples are taken.
+
+``name='Headphone Playback Volume',index=1``
+--------------------------------------------
+This control attenuates the samples for the headphone output.
+
+``name='Headphone Center Playback Switch',index=1``
+---------------------------------------------------
+If this switch is on, then the sample for the center PCM is put to the
+left headphone output (useful for SB Live cards without separate center/LFE
+output).
+
+``name='Headphone LFE Playback Switch',index=1``
+------------------------------------------------
+If this switch is on, then the sample for the center PCM is put to the
+right headphone output (useful for SB Live cards without separate center/LFE
+output).
+
+
+PCM stream related controls
+===========================
+
+``name='EMU10K1 PCM Volume',index 0-31``
+----------------------------------------
+Channel volume attenuation in range 0-0xffff. The maximum value (no
+attenuation) is default. The channel mapping for three values is
+as follows:
+
+* 0 - mono, default 0xffff (no attenuation)
+* 1 - left, default 0xffff (no attenuation)
+* 2 - right, default 0xffff (no attenuation)
+
+``name='EMU10K1 PCM Send Routing',index 0-31``
+----------------------------------------------
+This control specifies the destination - FX-bus accumulators. There are
+twelve values with this mapping:
+
+*  0 -  mono, A destination (FX-bus 0-15), default 0
+*  1 -  mono, B destination (FX-bus 0-15), default 1
+*  2 -  mono, C destination (FX-bus 0-15), default 2
+*  3 -  mono, D destination (FX-bus 0-15), default 3
+*  4 -  left, A destination (FX-bus 0-15), default 0
+*  5 -  left, B destination (FX-bus 0-15), default 1
+*  6 -  left, C destination (FX-bus 0-15), default 2
+*  7 -  left, D destination (FX-bus 0-15), default 3
+*  8 - right, A destination (FX-bus 0-15), default 0
+*  9 - right, B destination (FX-bus 0-15), default 1
+* 10 - right, C destination (FX-bus 0-15), default 2
+* 11 - right, D destination (FX-bus 0-15), default 3
+
+Don't forget that it's illegal to assign a channel to the same FX-bus accumulator 
+more than once (it means 0=0 && 1=0 is an invalid combination).
+ 
+``name='EMU10K1 PCM Send Volume',index 0-31``
+---------------------------------------------
+It specifies the attenuation (amount) for given destination in range 0-255.
+The channel mapping is following:
+
+*  0 -  mono, A destination attn, default 255 (no attenuation)
+*  1 -  mono, B destination attn, default 255 (no attenuation)
+*  2 -  mono, C destination attn, default 0 (mute)
+*  3 -  mono, D destination attn, default 0 (mute)
+*  4 -  left, A destination attn, default 255 (no attenuation)
+*  5 -  left, B destination attn, default 0 (mute)
+*  6 -  left, C destination attn, default 0 (mute)
+*  7 -  left, D destination attn, default 0 (mute)
+*  8 - right, A destination attn, default 0 (mute)
+*  9 - right, B destination attn, default 255 (no attenuation)
+* 10 - right, C destination attn, default 0 (mute)
+* 11 - right, D destination attn, default 0 (mute)
+
+
+
+MANUALS/PATENTS
+===============
+
+ftp://opensource.creative.com/pub/doc
+-------------------------------------
+
+LM4545.pdf
+	AC97 Codec
+m2049.pdf
+	The EMU10K1 Digital Audio Processor
+hog63.ps
+	FX8010 - A DSP Chip Architecture for Audio Effects
+
+
+WIPO Patents
+------------
+
+WO 9901813 (A1)
+	Audio Effects Processor with multiple asynchronous streams
+	(Jan. 14, 1999)
+
+WO 9901814 (A1)
+	Processor with Instruction Set for Audio Effects (Jan. 14, 1999)
+
+WO 9901953 (A1)
+	Audio Effects Processor having Decoupled Instruction
+        Execution and Audio Data Sequencing (Jan. 14, 1999)
+
+
+US Patents (http://www.uspto.gov/)
+----------------------------------
+
+US 5925841
+	Digital Sampling Instrument employing cache memory (Jul. 20, 1999)
+
+US 5928342
+	Audio Effects Processor integrated on a single chip
+        with a multiport memory onto which multiple asynchronous
+        digital sound samples can be concurrently loaded
+	(Jul. 27, 1999)
+
+US 5930158
+	Processor with Instruction Set for Audio Effects (Jul. 27, 1999)
+
+US 6032235
+	Memory initialization circuit (Tram) (Feb. 29, 2000)
+
+US 6138207
+	Interpolation looping of audio samples in cache connected to
+        system bus with prioritization and modification of bus transfers
+        in accordance with loop ends and minimum block sizes
+	(Oct. 24, 2000)
+
+US 6151670
+	Method for conserving memory storage using a
+        pool of  short term memory registers
+	(Nov. 21, 2000)
+
+US 6195715
+	Interrupt control for multiple programs communicating with
+        a common interrupt by associating programs to GP registers,
+        defining interrupt register, polling GP registers, and invoking
+        callback routine associated with defined interrupt register
+	(Feb. 27, 2001)
-- 
cgit v1.2.3-58-ga151


From 72e69166714bfa7bfafb7a06a8499de472299ab9 Mon Sep 17 00:00:00 2001
From: Takashi Iwai 
Date: Thu, 10 Nov 2016 16:25:42 +0100
Subject: ALSA: doc: ReSTize Audigy-mixer.txt

Another simple conversion from a plain text file.
Put to cards subdirectory.

Signed-off-by: Takashi Iwai 
---
 Documentation/sound/alsa/Audigy-mixer.txt  | 345 ---------------------------
 Documentation/sound/cards/audigy-mixer.rst | 368 +++++++++++++++++++++++++++++
 Documentation/sound/cards/index.rst        |   2 +
 3 files changed, 370 insertions(+), 345 deletions(-)
 delete mode 100644 Documentation/sound/alsa/Audigy-mixer.txt
 create mode 100644 Documentation/sound/cards/audigy-mixer.rst

(limited to 'Documentation/sound')

diff --git a/Documentation/sound/alsa/Audigy-mixer.txt b/Documentation/sound/alsa/Audigy-mixer.txt
deleted file mode 100644
index 7f10dc6ff28c..000000000000
--- a/Documentation/sound/alsa/Audigy-mixer.txt
+++ /dev/null
@@ -1,345 +0,0 @@
-
-		Sound Blaster Audigy mixer / default DSP code
-		===========================================
-
-This is based on SB-Live-mixer.txt.
-
-The EMU10K2 chips have a DSP part which can be programmed to support 
-various ways of sample processing, which is described here.
-(This article does not deal with the overall functionality of the 
-EMU10K2 chips. See the manuals section for further details.)
-
-The ALSA driver programs this portion of chip by default code
-(can be altered later) which offers the following functionality:
-
-
-1) Digital mixer controls
--------------------------
-
-These controls are built using the DSP instructions. They offer extended
-functionality. Only the default build-in code in the ALSA driver is described
-here. Note that the controls work as attenuators: the maximum value is the 
-neutral position leaving the signal unchanged. Note that if the  same destination 
-is mentioned in multiple controls, the signal is accumulated and can be wrapped 
-(set to maximal or minimal value without checking of overflow).
-
-
-Explanation of used abbreviations:
-
-DAC    - digital to analog converter
-ADC    - analog to digital converter
-I2S    - one-way three wire serial bus for digital sound by Philips Semiconductors
-         (this standard is used for connecting standalone DAC and ADC converters)
-LFE    - low frequency effects (subwoofer signal)
-AC97   - a chip containing an analog mixer, DAC and ADC converters
-IEC958 - S/PDIF
-FX-bus - the EMU10K2 chip has an effect bus containing 64 accumulators.
-         Each of the synthesizer voices can feed its output to these accumulators
-         and the DSP microcontroller can operate with the resulting sum.
-
-name='PCM Front Playback Volume',index=0
-
-This control is used to attenuate samples for left and right front PCM FX-bus
-accumulators. ALSA uses accumulators 8 and 9 for left and right front PCM 
-samples for 5.1 playback. The result samples are forwarded to the front DAC PCM 
-slots of the Philips DAC.
-
-name='PCM Surround Playback Volume',index=0
-
-This control is used to attenuate samples for left and right surround PCM FX-bus
-accumulators. ALSA uses accumulators 2 and 3 for left and right surround PCM 
-samples for 5.1 playback. The result samples are forwarded to the surround DAC PCM 
-slots of the Philips DAC.
-
-name='PCM Center Playback Volume',index=0
-
-This control is used to attenuate samples for center PCM FX-bus accumulator.
-ALSA uses accumulator 6 for center PCM sample for 5.1 playback. The result sample
-is forwarded to the center DAC PCM slot of the Philips DAC.
-
-name='PCM LFE Playback Volume',index=0
-
-This control is used to attenuate sample for LFE PCM FX-bus accumulator. 
-ALSA uses accumulator 7 for LFE PCM sample for 5.1 playback. The result sample 
-is forwarded to the LFE DAC PCM slot of the Philips DAC.
-
-name='PCM Playback Volume',index=0
-
-This control is used to attenuate samples for left and right PCM FX-bus
-accumulators. ALSA uses accumulators 0 and 1 for left and right PCM samples for
-stereo playback. The result samples are forwarded to the front DAC PCM slots 
-of the Philips DAC.
-
-name='PCM Capture Volume',index=0
-
-This control is used to attenuate samples for left and right PCM FX-bus
-accumulator. ALSA uses accumulators 0 and 1 for left and right PCM.
-The result is forwarded to the ADC capture FIFO (thus to the standard capture
-PCM device).
-
-name='Music Playback Volume',index=0
-
-This control is used to attenuate samples for left and right MIDI FX-bus
-accumulators. ALSA uses accumulators 4 and 5 for left and right MIDI samples.
-The result samples are forwarded to the front DAC PCM slots of the AC97 codec.
-
-name='Music Capture Volume',index=0
-
-These controls are used to attenuate samples for left and right MIDI FX-bus
-accumulator. ALSA uses accumulators 4 and 5 for left and right PCM.
-The result is forwarded to the ADC capture FIFO (thus to the standard capture
-PCM device).
-
-name='Mic Playback Volume',index=0
-
-This control is used to attenuate samples for left and right Mic input.
-For Mic input is used AC97 codec. The result samples are forwarded to 
-the front DAC PCM slots of the Philips DAC. Samples are forwarded to Mic
-capture FIFO (device 1 - 16bit/8KHz mono) too without volume control.
-
-name='Mic Capture Volume',index=0
-
-This control is used to attenuate samples for left and right Mic input.
-The result is forwarded to the ADC capture FIFO (thus to the standard capture
-PCM device).
-
-name='Audigy CD Playback Volume',index=0
-
-This control is used to attenuate samples from left and right IEC958 TTL
-digital inputs (usually used by a CDROM drive). The result samples are
-forwarded to the front DAC PCM slots of the Philips DAC.
-
-name='Audigy CD Capture Volume',index=0
-
-This control is used to attenuate samples from left and right IEC958 TTL
-digital inputs (usually used by a CDROM drive). The result samples are
-forwarded to the ADC capture FIFO (thus to the standard capture PCM device).
-
-name='IEC958 Optical Playback Volume',index=0
-
-This control is used to attenuate samples from left and right IEC958 optical
-digital input. The result samples are forwarded to the front DAC PCM slots
-of the Philips DAC.
-
-name='IEC958 Optical Capture Volume',index=0
-
-This control is used to attenuate samples from left and right IEC958 optical
-digital inputs. The result samples are forwarded to the ADC capture FIFO
-(thus to the standard capture PCM device).
-
-name='Line2 Playback Volume',index=0
-
-This control is used to attenuate samples from left and right I2S ADC
-inputs (on the AudigyDrive). The result samples are forwarded to the front
-DAC PCM slots of the Philips DAC.
-
-name='Line2 Capture Volume',index=1
-
-This control is used to attenuate samples from left and right I2S ADC
-inputs (on the AudigyDrive). The result samples are forwarded to the ADC
-capture FIFO (thus to the standard capture PCM device).
-
-name='Analog Mix Playback Volume',index=0
-
-This control is used to attenuate samples from left and right I2S ADC
-inputs from Philips ADC. The result samples are forwarded to the front
-DAC PCM slots of the Philips DAC. This contains mix from analog sources
-like CD, Line In, Aux, ....
-
-name='Analog Mix Capture Volume',index=1
-
-This control is used to attenuate samples from left and right I2S ADC
-inputs Philips ADC. The result samples are forwarded to the ADC
-capture FIFO (thus to the standard capture PCM device).
-
-name='Aux2 Playback Volume',index=0
-
-This control is used to attenuate samples from left and right I2S ADC
-inputs (on the AudigyDrive). The result samples are forwarded to the front
-DAC PCM slots of the Philips DAC.
-
-name='Aux2 Capture Volume',index=1
-
-This control is used to attenuate samples from left and right I2S ADC
-inputs (on the AudigyDrive). The result samples are forwarded to the ADC
-capture FIFO (thus to the standard capture PCM device).
-
-name='Front Playback Volume',index=0
-
-All stereo signals are mixed together and mirrored to surround, center and LFE.
-This control is used to attenuate samples for left and right front speakers of
-this mix.
-
-name='Surround Playback Volume',index=0
-
-All stereo signals are mixed together and mirrored to surround, center and LFE.
-This control is used to attenuate samples for left and right surround speakers of
-this mix.
-
-name='Center Playback Volume',index=0
-
-All stereo signals are mixed together and mirrored to surround, center and LFE.
-This control is used to attenuate sample for center speaker of this mix.
-
-name='LFE Playback Volume',index=0
-
-All stereo signals are mixed together and mirrored to surround, center and LFE.
-This control is used to attenuate sample for LFE speaker of this mix.
-
-name='Tone Control - Switch',index=0
-
-This control turns the tone control on or off. The samples for front, rear
-and center / LFE outputs are affected.
-
-name='Tone Control - Bass',index=0
-
-This control sets the bass intensity. There is no neutral value!!
-When the tone control code is activated, the samples are always modified.
-The closest value to pure signal is 20.
-
-name='Tone Control - Treble',index=0
-
-This control sets the treble intensity. There is no neutral value!!
-When the tone control code is activated, the samples are always modified.
-The closest value to pure signal is 20.
-
-name='Master Playback Volume',index=0
-
-This control is used to attenuate samples for front, surround, center and 
-LFE outputs.
-
-name='IEC958 Optical Raw Playback Switch',index=0
-
-If this switch is on, then the samples for the IEC958 (S/PDIF) digital
-output are taken only from the raw FX8010 PCM, otherwise standard front
-PCM samples are taken.
-
-
-2) PCM stream related controls
-------------------------------
-
-name='EMU10K1 PCM Volume',index 0-31
-
-Channel volume attenuation in range 0-0xffff. The maximum value (no
-attenuation) is default. The channel mapping for three values is
-as follows:
-
-	0 - mono, default 0xffff (no attenuation)
-	1 - left, default 0xffff (no attenuation)
-	2 - right, default 0xffff (no attenuation)
-
-name='EMU10K1 PCM Send Routing',index 0-31
-
-This control specifies the destination - FX-bus accumulators. There 24
-values with this mapping:
-
-	 0 -  mono, A destination (FX-bus 0-63), default 0
-	 1 -  mono, B destination (FX-bus 0-63), default 1
-	 2 -  mono, C destination (FX-bus 0-63), default 2
-	 3 -  mono, D destination (FX-bus 0-63), default 3
-	 4 -  mono, E destination (FX-bus 0-63), default 0
-	 5 -  mono, F destination (FX-bus 0-63), default 0
-	 6 -  mono, G destination (FX-bus 0-63), default 0
-	 7 -  mono, H destination (FX-bus 0-63), default 0
-	 8 -  left, A destination (FX-bus 0-63), default 0
-	 9 -  left, B destination (FX-bus 0-63), default 1
-	10 -  left, C destination (FX-bus 0-63), default 2
-	11 -  left, D destination (FX-bus 0-63), default 3
-	12 -  left, E destination (FX-bus 0-63), default 0
-	13 -  left, F destination (FX-bus 0-63), default 0
-	14 -  left, G destination (FX-bus 0-63), default 0
-	15 -  left, H destination (FX-bus 0-63), default 0
-	16 - right, A destination (FX-bus 0-63), default 0
-	17 - right, B destination (FX-bus 0-63), default 1
-	18 - right, C destination (FX-bus 0-63), default 2
-	19 - right, D destination (FX-bus 0-63), default 3
-	20 - right, E destination (FX-bus 0-63), default 0
-	21 - right, F destination (FX-bus 0-63), default 0
-	22 - right, G destination (FX-bus 0-63), default 0
-	23 - right, H destination (FX-bus 0-63), default 0
-
-Don't forget that it's illegal to assign a channel to the same FX-bus accumulator 
-more than once (it means 0=0 && 1=0 is an invalid combination).
- 
-name='EMU10K1 PCM Send Volume',index 0-31
-
-It specifies the attenuation (amount) for given destination in range 0-255.
-The channel mapping is following:
-
-	 0 -  mono, A destination attn, default 255 (no attenuation)
-	 1 -  mono, B destination attn, default 255 (no attenuation)
-	 2 -  mono, C destination attn, default 0 (mute)
-	 3 -  mono, D destination attn, default 0 (mute)
-	 4 -  mono, E destination attn, default 0 (mute)
-	 5 -  mono, F destination attn, default 0 (mute)
-	 6 -  mono, G destination attn, default 0 (mute)
-	 7 -  mono, H destination attn, default 0 (mute)
-	 8 -  left, A destination attn, default 255 (no attenuation)
-	 9 -  left, B destination attn, default 0 (mute)
-	10 -  left, C destination attn, default 0 (mute)
-	11 -  left, D destination attn, default 0 (mute)
-	12 -  left, E destination attn, default 0 (mute)
-	13 -  left, F destination attn, default 0 (mute)
-	14 -  left, G destination attn, default 0 (mute)
-	15 -  left, H destination attn, default 0 (mute)
-	16 - right, A destination attn, default 0 (mute)
-	17 - right, B destination attn, default 255 (no attenuation)
-	18 - right, C destination attn, default 0 (mute)
-	19 - right, D destination attn, default 0 (mute)
-	20 - right, E destination attn, default 0 (mute)
-	21 - right, F destination attn, default 0 (mute)
-	22 - right, G destination attn, default 0 (mute)
-	23 - right, H destination attn, default 0 (mute)
-
-
-
-4) MANUALS/PATENTS:
--------------------
-
-ftp://opensource.creative.com/pub/doc
--------------------------------------
-
-        Files:
-        LM4545.pdf      AC97 Codec
-
-        m2049.pdf       The EMU10K1 Digital Audio Processor
-
-        hog63.ps        FX8010 - A DSP Chip Architecture for Audio Effects
-
-
-WIPO Patents
-------------
-        Patent numbers:
-        WO 9901813 (A1) Audio Effects Processor with multiple asynchronous (Jan. 14, 1999)
-                        streams
-
-        WO 9901814 (A1) Processor with Instruction Set for Audio Effects (Jan. 14, 1999)
-
-        WO 9901953 (A1) Audio Effects Processor having Decoupled Instruction
-                        Execution and Audio Data Sequencing (Jan. 14, 1999)
-
-
-US Patents (http://www.uspto.gov/)
-----------------------------------
-
-        US 5925841      Digital Sampling Instrument employing cache memory (Jul. 20, 1999)
-
-        US 5928342      Audio Effects Processor integrated on a single chip (Jul. 27, 1999)
-                        with a multiport memory onto which multiple asynchronous
-                        digital sound samples can be concurrently loaded
-
-        US 5930158      Processor with Instruction Set for Audio Effects (Jul. 27, 1999)
-
-        US 6032235      Memory initialization circuit (Tram) (Feb. 29, 2000)
-
-        US 6138207      Interpolation looping of audio samples in cache connected to    (Oct. 24, 2000)
-                        system bus with prioritization and modification of bus transfers
-                        in accordance with loop ends and minimum block sizes
-
-        US 6151670      Method for conserving memory storage using a (Nov. 21, 2000)
-                        pool of  short term memory registers
-
-        US 6195715      Interrupt control for multiple programs communicating with      (Feb. 27, 2001)
-                        a common interrupt by associating programs to GP registers,
-                        defining interrupt register, polling GP registers, and invoking
-                        callback routine associated with defined interrupt register
diff --git a/Documentation/sound/cards/audigy-mixer.rst b/Documentation/sound/cards/audigy-mixer.rst
new file mode 100644
index 000000000000..86213234435f
--- /dev/null
+++ b/Documentation/sound/cards/audigy-mixer.rst
@@ -0,0 +1,368 @@
+=============================================
+Sound Blaster Audigy mixer / default DSP code
+=============================================
+
+This is based on sb-live-mixer.rst.
+
+The EMU10K2 chips have a DSP part which can be programmed to support 
+various ways of sample processing, which is described here.
+(This article does not deal with the overall functionality of the 
+EMU10K2 chips. See the manuals section for further details.)
+
+The ALSA driver programs this portion of chip by default code
+(can be altered later) which offers the following functionality:
+
+
+Digital mixer controls
+======================
+
+These controls are built using the DSP instructions. They offer extended
+functionality. Only the default build-in code in the ALSA driver is described
+here. Note that the controls work as attenuators: the maximum value is the 
+neutral position leaving the signal unchanged. Note that if the  same destination 
+is mentioned in multiple controls, the signal is accumulated and can be wrapped 
+(set to maximal or minimal value without checking of overflow).
+
+
+Explanation of used abbreviations:
+
+DAC
+	digital to analog converter
+ADC
+	analog to digital converter
+I2S
+	one-way three wire serial bus for digital sound by Philips Semiconductors
+        (this standard is used for connecting standalone DAC and ADC converters)
+LFE
+	low frequency effects (subwoofer signal)
+AC97
+	a chip containing an analog mixer, DAC and ADC converters
+IEC958
+	S/PDIF
+FX-bus
+	the EMU10K2 chip has an effect bus containing 64 accumulators.
+        Each of the synthesizer voices can feed its output to these accumulators
+        and the DSP microcontroller can operate with the resulting sum.
+
+name='PCM Front Playback Volume',index=0
+----------------------------------------
+This control is used to attenuate samples for left and right front PCM FX-bus
+accumulators. ALSA uses accumulators 8 and 9 for left and right front PCM 
+samples for 5.1 playback. The result samples are forwarded to the front DAC PCM 
+slots of the Philips DAC.
+
+name='PCM Surround Playback Volume',index=0
+-------------------------------------------
+This control is used to attenuate samples for left and right surround PCM FX-bus
+accumulators. ALSA uses accumulators 2 and 3 for left and right surround PCM 
+samples for 5.1 playback. The result samples are forwarded to the surround DAC PCM 
+slots of the Philips DAC.
+
+name='PCM Center Playback Volume',index=0
+-----------------------------------------
+This control is used to attenuate samples for center PCM FX-bus accumulator.
+ALSA uses accumulator 6 for center PCM sample for 5.1 playback. The result sample
+is forwarded to the center DAC PCM slot of the Philips DAC.
+
+name='PCM LFE Playback Volume',index=0
+--------------------------------------
+This control is used to attenuate sample for LFE PCM FX-bus accumulator. 
+ALSA uses accumulator 7 for LFE PCM sample for 5.1 playback. The result sample 
+is forwarded to the LFE DAC PCM slot of the Philips DAC.
+
+name='PCM Playback Volume',index=0
+----------------------------------
+This control is used to attenuate samples for left and right PCM FX-bus
+accumulators. ALSA uses accumulators 0 and 1 for left and right PCM samples for
+stereo playback. The result samples are forwarded to the front DAC PCM slots 
+of the Philips DAC.
+
+name='PCM Capture Volume',index=0
+---------------------------------
+This control is used to attenuate samples for left and right PCM FX-bus
+accumulator. ALSA uses accumulators 0 and 1 for left and right PCM.
+The result is forwarded to the ADC capture FIFO (thus to the standard capture
+PCM device).
+
+name='Music Playback Volume',index=0
+------------------------------------
+This control is used to attenuate samples for left and right MIDI FX-bus
+accumulators. ALSA uses accumulators 4 and 5 for left and right MIDI samples.
+The result samples are forwarded to the front DAC PCM slots of the AC97 codec.
+
+name='Music Capture Volume',index=0
+-----------------------------------
+These controls are used to attenuate samples for left and right MIDI FX-bus
+accumulator. ALSA uses accumulators 4 and 5 for left and right PCM.
+The result is forwarded to the ADC capture FIFO (thus to the standard capture
+PCM device).
+
+name='Mic Playback Volume',index=0
+----------------------------------
+This control is used to attenuate samples for left and right Mic input.
+For Mic input is used AC97 codec. The result samples are forwarded to 
+the front DAC PCM slots of the Philips DAC. Samples are forwarded to Mic
+capture FIFO (device 1 - 16bit/8KHz mono) too without volume control.
+
+name='Mic Capture Volume',index=0
+---------------------------------
+This control is used to attenuate samples for left and right Mic input.
+The result is forwarded to the ADC capture FIFO (thus to the standard capture
+PCM device).
+
+name='Audigy CD Playback Volume',index=0
+----------------------------------------
+This control is used to attenuate samples from left and right IEC958 TTL
+digital inputs (usually used by a CDROM drive). The result samples are
+forwarded to the front DAC PCM slots of the Philips DAC.
+
+name='Audigy CD Capture Volume',index=0
+---------------------------------------
+This control is used to attenuate samples from left and right IEC958 TTL
+digital inputs (usually used by a CDROM drive). The result samples are
+forwarded to the ADC capture FIFO (thus to the standard capture PCM device).
+
+name='IEC958 Optical Playback Volume',index=0
+---------------------------------------------
+This control is used to attenuate samples from left and right IEC958 optical
+digital input. The result samples are forwarded to the front DAC PCM slots
+of the Philips DAC.
+
+name='IEC958 Optical Capture Volume',index=0
+--------------------------------------------
+This control is used to attenuate samples from left and right IEC958 optical
+digital inputs. The result samples are forwarded to the ADC capture FIFO
+(thus to the standard capture PCM device).
+
+name='Line2 Playback Volume',index=0
+------------------------------------
+This control is used to attenuate samples from left and right I2S ADC
+inputs (on the AudigyDrive). The result samples are forwarded to the front
+DAC PCM slots of the Philips DAC.
+
+name='Line2 Capture Volume',index=1
+-----------------------------------
+This control is used to attenuate samples from left and right I2S ADC
+inputs (on the AudigyDrive). The result samples are forwarded to the ADC
+capture FIFO (thus to the standard capture PCM device).
+
+name='Analog Mix Playback Volume',index=0
+-----------------------------------------
+This control is used to attenuate samples from left and right I2S ADC
+inputs from Philips ADC. The result samples are forwarded to the front
+DAC PCM slots of the Philips DAC. This contains mix from analog sources
+like CD, Line In, Aux, ....
+
+name='Analog Mix Capture Volume',index=1
+----------------------------------------
+This control is used to attenuate samples from left and right I2S ADC
+inputs Philips ADC. The result samples are forwarded to the ADC
+capture FIFO (thus to the standard capture PCM device).
+
+name='Aux2 Playback Volume',index=0
+-----------------------------------
+This control is used to attenuate samples from left and right I2S ADC
+inputs (on the AudigyDrive). The result samples are forwarded to the front
+DAC PCM slots of the Philips DAC.
+
+name='Aux2 Capture Volume',index=1
+----------------------------------
+This control is used to attenuate samples from left and right I2S ADC
+inputs (on the AudigyDrive). The result samples are forwarded to the ADC
+capture FIFO (thus to the standard capture PCM device).
+
+name='Front Playback Volume',index=0
+------------------------------------
+All stereo signals are mixed together and mirrored to surround, center and LFE.
+This control is used to attenuate samples for left and right front speakers of
+this mix.
+
+name='Surround Playback Volume',index=0
+---------------------------------------
+All stereo signals are mixed together and mirrored to surround, center and LFE.
+This control is used to attenuate samples for left and right surround speakers of
+this mix.
+
+name='Center Playback Volume',index=0
+-------------------------------------
+All stereo signals are mixed together and mirrored to surround, center and LFE.
+This control is used to attenuate sample for center speaker of this mix.
+
+name='LFE Playback Volume',index=0
+----------------------------------
+All stereo signals are mixed together and mirrored to surround, center and LFE.
+This control is used to attenuate sample for LFE speaker of this mix.
+
+name='Tone Control - Switch',index=0
+------------------------------------
+This control turns the tone control on or off. The samples for front, rear
+and center / LFE outputs are affected.
+
+name='Tone Control - Bass',index=0
+----------------------------------
+This control sets the bass intensity. There is no neutral value!!
+When the tone control code is activated, the samples are always modified.
+The closest value to pure signal is 20.
+
+name='Tone Control - Treble',index=0
+------------------------------------
+This control sets the treble intensity. There is no neutral value!!
+When the tone control code is activated, the samples are always modified.
+The closest value to pure signal is 20.
+
+name='Master Playback Volume',index=0
+-------------------------------------
+This control is used to attenuate samples for front, surround, center and 
+LFE outputs.
+
+name='IEC958 Optical Raw Playback Switch',index=0
+-------------------------------------------------
+If this switch is on, then the samples for the IEC958 (S/PDIF) digital
+output are taken only from the raw FX8010 PCM, otherwise standard front
+PCM samples are taken.
+
+
+PCM stream related controls
+===========================
+
+name='EMU10K1 PCM Volume',index 0-31
+------------------------------------
+Channel volume attenuation in range 0-0xffff. The maximum value (no
+attenuation) is default. The channel mapping for three values is
+as follows:
+
+* 0 - mono, default 0xffff (no attenuation)
+* 1 - left, default 0xffff (no attenuation)
+* 2 - right, default 0xffff (no attenuation)
+
+name='EMU10K1 PCM Send Routing',index 0-31
+------------------------------------------
+This control specifies the destination - FX-bus accumulators. There 24
+values with this mapping:
+
+* 0 -  mono, A destination (FX-bus 0-63), default 0
+* 1 -  mono, B destination (FX-bus 0-63), default 1
+* 2 -  mono, C destination (FX-bus 0-63), default 2
+* 3 -  mono, D destination (FX-bus 0-63), default 3
+* 4 -  mono, E destination (FX-bus 0-63), default 0
+* 5 -  mono, F destination (FX-bus 0-63), default 0
+* 6 -  mono, G destination (FX-bus 0-63), default 0
+* 7 -  mono, H destination (FX-bus 0-63), default 0
+* 8 -  left, A destination (FX-bus 0-63), default 0
+* 9 -  left, B destination (FX-bus 0-63), default 1
+* 10 -  left, C destination (FX-bus 0-63), default 2
+* 11 -  left, D destination (FX-bus 0-63), default 3
+* 12 -  left, E destination (FX-bus 0-63), default 0
+* 13 -  left, F destination (FX-bus 0-63), default 0
+* 14 -  left, G destination (FX-bus 0-63), default 0
+* 15 -  left, H destination (FX-bus 0-63), default 0
+* 16 - right, A destination (FX-bus 0-63), default 0
+* 17 - right, B destination (FX-bus 0-63), default 1
+* 18 - right, C destination (FX-bus 0-63), default 2
+* 19 - right, D destination (FX-bus 0-63), default 3
+* 20 - right, E destination (FX-bus 0-63), default 0
+* 21 - right, F destination (FX-bus 0-63), default 0
+* 22 - right, G destination (FX-bus 0-63), default 0
+* 23 - right, H destination (FX-bus 0-63), default 0
+
+Don't forget that it's illegal to assign a channel to the same FX-bus accumulator 
+more than once (it means 0=0 && 1=0 is an invalid combination).
+ 
+name='EMU10K1 PCM Send Volume',index 0-31
+-----------------------------------------
+It specifies the attenuation (amount) for given destination in range 0-255.
+The channel mapping is following:
+
+*  0 -  mono, A destination attn, default 255 (no attenuation)
+*  1 -  mono, B destination attn, default 255 (no attenuation)
+*  2 -  mono, C destination attn, default 0 (mute)
+*  3 -  mono, D destination attn, default 0 (mute)
+*  4 -  mono, E destination attn, default 0 (mute)
+*  5 -  mono, F destination attn, default 0 (mute)
+*  6 -  mono, G destination attn, default 0 (mute)
+*  7 -  mono, H destination attn, default 0 (mute)
+*  8 -  left, A destination attn, default 255 (no attenuation)
+*  9 -  left, B destination attn, default 0 (mute)
+* 10 -  left, C destination attn, default 0 (mute)
+* 11 -  left, D destination attn, default 0 (mute)
+* 12 -  left, E destination attn, default 0 (mute)
+* 13 -  left, F destination attn, default 0 (mute)
+* 14 -  left, G destination attn, default 0 (mute)
+* 15 -  left, H destination attn, default 0 (mute)
+* 16 - right, A destination attn, default 0 (mute)
+* 17 - right, B destination attn, default 255 (no attenuation)
+* 18 - right, C destination attn, default 0 (mute)
+* 19 - right, D destination attn, default 0 (mute)
+* 20 - right, E destination attn, default 0 (mute)
+* 21 - right, F destination attn, default 0 (mute)
+* 22 - right, G destination attn, default 0 (mute)
+* 23 - right, H destination attn, default 0 (mute)
+
+
+
+MANUALS/PATENTS
+===============
+
+ftp://opensource.creative.com/pub/doc
+-------------------------------------
+
+LM4545.pdf
+	AC97 Codec
+
+m2049.pdf
+	The EMU10K1 Digital Audio Processor
+
+hog63.ps
+	FX8010 - A DSP Chip Architecture for Audio Effects
+
+
+WIPO Patents
+------------
+
+WO 9901813 (A1)
+	Audio Effects Processor with multiple asynchronous streams
+	(Jan. 14, 1999)
+
+WO 9901814 (A1)
+	Processor with Instruction Set for Audio Effects (Jan. 14, 1999)
+
+WO 9901953 (A1)
+	Audio Effects Processor having Decoupled Instruction
+        Execution and Audio Data Sequencing (Jan. 14, 1999)
+
+
+US Patents (http://www.uspto.gov/)
+----------------------------------
+
+US 5925841
+	Digital Sampling Instrument employing cache memory (Jul. 20, 1999)
+
+US 5928342
+	Audio Effects Processor integrated on a single chip
+        with a multiport memory onto which multiple asynchronous
+        digital sound samples can be concurrently loaded
+	(Jul. 27, 1999)
+
+US 5930158
+	Processor with Instruction Set for Audio Effects (Jul. 27, 1999)
+
+US 6032235
+	Memory initialization circuit (Tram) (Feb. 29, 2000)
+
+US 6138207
+	Interpolation looping of audio samples in cache connected to
+        system bus with prioritization and modification of bus transfers
+        in accordance with loop ends and minimum block sizes
+	(Oct. 24, 2000)
+
+US 6151670
+	Method for conserving memory storage using a
+        pool of  short term memory registers
+	(Nov. 21, 2000)
+
+US 6195715
+	Interrupt control for multiple programs communicating with
+        a common interrupt by associating programs to GP registers,
+        defining interrupt register, polling GP registers, and invoking
+        callback routine associated with defined interrupt register
+	(Feb. 27, 2001)
diff --git a/Documentation/sound/cards/index.rst b/Documentation/sound/cards/index.rst
index 294efdb75bc9..c9d7ce4286b2 100644
--- a/Documentation/sound/cards/index.rst
+++ b/Documentation/sound/cards/index.rst
@@ -7,3 +7,5 @@ Card-Specific Information
    joystick
    cmipci
    sb-live-mixer
+   audigy-mixer
+  
-- 
cgit v1.2.3-58-ga151


From e7030c96fc0e08b9d1c4fb1cbedb326a3f46dad3 Mon Sep 17 00:00:00 2001
From: Takashi Iwai 
Date: Thu, 10 Nov 2016 16:31:07 +0100
Subject: ALSA: doc: ReSTize emu10k1-jack.txt

Another simple conversion from a plain text file.
Put to cards directory.

Signed-off-by: Takashi Iwai 
---
 Documentation/sound/alsa/emu10k1-jack.txt  | 74 ----------------------------
 Documentation/sound/cards/emu10k1-jack.rst | 78 ++++++++++++++++++++++++++++++
 Documentation/sound/cards/index.rst        |  2 +-
 3 files changed, 79 insertions(+), 75 deletions(-)
 delete mode 100644 Documentation/sound/alsa/emu10k1-jack.txt
 create mode 100644 Documentation/sound/cards/emu10k1-jack.rst

(limited to 'Documentation/sound')

diff --git a/Documentation/sound/alsa/emu10k1-jack.txt b/Documentation/sound/alsa/emu10k1-jack.txt
deleted file mode 100644
index 751d45036a05..000000000000
--- a/Documentation/sound/alsa/emu10k1-jack.txt
+++ /dev/null
@@ -1,74 +0,0 @@
-This document is a guide to using the emu10k1 based devices with JACK for low
-latency, multichannel recording functionality.  All of my recent work to allow
-Linux users to use the full capabilities of their hardware has been inspired 
-by the kX Project.  Without their work I never would have discovered the true
-power of this hardware.
-
-	http://www.kxproject.com
-						- Lee Revell, 2005.03.30
-
-Low latency, multichannel audio with JACK and the emu10k1/emu10k2
------------------------------------------------------------------
-
-Until recently, emu10k1 users on Linux did not have access to the same low
-latency, multichannel features offered by the "kX ASIO" feature of their
-Windows driver.  As of ALSA 1.0.9 this is no more!
-
-For those unfamiliar with kX ASIO, this consists of 16 capture and 16 playback
-channels.  With a post 2.6.9 Linux kernel, latencies down to 64 (1.33 ms) or
-even 32 (0.66ms) frames should work well.
-
-The configuration is slightly more involved than on Windows, as you have to
-select the correct device for JACK to use.  Actually, for qjackctl users it's
-fairly self explanatory - select Duplex, then for capture and playback select
-the multichannel devices, set the in and out channels to 16, and the sample
-rate to 48000Hz.  The command line looks like this:
-
-/usr/local/bin/jackd -R -dalsa -r48000 -p64 -n2 -D -Chw:0,2 -Phw:0,3 -S
-
-This will give you 16 input ports and 16 output ports.
-
-The 16 output ports map onto the 16 FX buses (or the first 16 of 64, for the
-Audigy).  The mapping from FX bus to physical output is described in
-SB-Live-mixer.txt (or Audigy-mixer.txt).
-
-The 16 input ports are connected to the 16 physical inputs.  Contrary to
-popular belief, all emu10k1 cards are multichannel cards.  Which of these
-input channels have physical inputs connected to them depends on the card
-model.  Trial and error is highly recommended; the pinout diagrams
-for the card have been reverse engineered by some enterprising kX users and are 
-available on the internet.  Meterbridge is helpful here, and the kX forums are
-packed with useful information.
-
-Each input port will either correspond to a digital (SPDIF) input, an analog
-input, or nothing.  The one exception is the SBLive! 5.1.  On these devices,
-the second and third input ports are wired to the center/LFE output.  You will
-still see 16 capture channels, but only 14 are available for recording inputs.
-
-This chart, borrowed from kxfxlib/da_asio51.cpp, describes the mapping of JACK
-ports to FXBUS2 (multitrack recording input) and EXTOUT (physical output)
-channels.
-
-/*JACK (& ASIO) mappings on 10k1 5.1 SBLive cards:
---------------------------------------------
-JACK		Epilog		FXBUS2(nr)
---------------------------------------------
-capture_1	asio14		FXBUS2(0xe)
-capture_2	asio15		FXBUS2(0xf)
-capture_3	asio0		FXBUS2(0x0)	
-~capture_4	Center		EXTOUT(0x11)	// mapped to by Center
-~capture_5	LFE		EXTOUT(0x12)	// mapped to by LFE
-capture_6	asio3		FXBUS2(0x3)
-capture_7	asio4		FXBUS2(0x4)
-capture_8	asio5		FXBUS2(0x5)
-capture_9	asio6		FXBUS2(0x6)
-capture_10	asio7		FXBUS2(0x7)
-capture_11	asio8		FXBUS2(0x8)
-capture_12	asio9		FXBUS2(0x9)
-capture_13	asio10		FXBUS2(0xa)
-capture_14	asio11		FXBUS2(0xb)
-capture_15	asio12		FXBUS2(0xc)
-capture_16	asio13		FXBUS2(0xd)
-*/
-
-TODO: describe use of ld10k1/qlo10k1 in conjunction with JACK
diff --git a/Documentation/sound/cards/emu10k1-jack.rst b/Documentation/sound/cards/emu10k1-jack.rst
new file mode 100644
index 000000000000..6597f1ea83f0
--- /dev/null
+++ b/Documentation/sound/cards/emu10k1-jack.rst
@@ -0,0 +1,78 @@
+=================================================================
+Low latency, multichannel audio with JACK and the emu10k1/emu10k2
+=================================================================
+
+This document is a guide to using the emu10k1 based devices with JACK for low
+latency, multichannel recording functionality.  All of my recent work to allow
+Linux users to use the full capabilities of their hardware has been inspired 
+by the kX Project.  Without their work I never would have discovered the true
+power of this hardware.
+
+	http://www.kxproject.com
+						- Lee Revell, 2005.03.30
+
+
+Until recently, emu10k1 users on Linux did not have access to the same low
+latency, multichannel features offered by the "kX ASIO" feature of their
+Windows driver.  As of ALSA 1.0.9 this is no more!
+
+For those unfamiliar with kX ASIO, this consists of 16 capture and 16 playback
+channels.  With a post 2.6.9 Linux kernel, latencies down to 64 (1.33 ms) or
+even 32 (0.66ms) frames should work well.
+
+The configuration is slightly more involved than on Windows, as you have to
+select the correct device for JACK to use.  Actually, for qjackctl users it's
+fairly self explanatory - select Duplex, then for capture and playback select
+the multichannel devices, set the in and out channels to 16, and the sample
+rate to 48000Hz.  The command line looks like this:
+::
+
+  /usr/local/bin/jackd -R -dalsa -r48000 -p64 -n2 -D -Chw:0,2 -Phw:0,3 -S
+
+This will give you 16 input ports and 16 output ports.
+
+The 16 output ports map onto the 16 FX buses (or the first 16 of 64, for the
+Audigy).  The mapping from FX bus to physical output is described in
+sb-live-mixer.rst (or audigy-mixer.rst).
+
+The 16 input ports are connected to the 16 physical inputs.  Contrary to
+popular belief, all emu10k1 cards are multichannel cards.  Which of these
+input channels have physical inputs connected to them depends on the card
+model.  Trial and error is highly recommended; the pinout diagrams
+for the card have been reverse engineered by some enterprising kX users and are 
+available on the internet.  Meterbridge is helpful here, and the kX forums are
+packed with useful information.
+
+Each input port will either correspond to a digital (SPDIF) input, an analog
+input, or nothing.  The one exception is the SBLive! 5.1.  On these devices,
+the second and third input ports are wired to the center/LFE output.  You will
+still see 16 capture channels, but only 14 are available for recording inputs.
+
+This chart, borrowed from kxfxlib/da_asio51.cpp, describes the mapping of JACK
+ports to FXBUS2 (multitrack recording input) and EXTOUT (physical output)
+channels.
+
+JACK (& ASIO) mappings on 10k1 5.1 SBLive cards:
+
+==============  ========        ============
+JACK		Epilog		FXBUS2(nr)
+==============  ========        ============
+capture_1	asio14		FXBUS2(0xe)
+capture_2	asio15		FXBUS2(0xf)
+capture_3	asio0		FXBUS2(0x0)	
+~capture_4	Center		EXTOUT(0x11)	// mapped to by Center
+~capture_5	LFE		EXTOUT(0x12)	// mapped to by LFE
+capture_6	asio3		FXBUS2(0x3)
+capture_7	asio4		FXBUS2(0x4)
+capture_8	asio5		FXBUS2(0x5)
+capture_9	asio6		FXBUS2(0x6)
+capture_10	asio7		FXBUS2(0x7)
+capture_11	asio8		FXBUS2(0x8)
+capture_12	asio9		FXBUS2(0x9)
+capture_13	asio10		FXBUS2(0xa)
+capture_14	asio11		FXBUS2(0xb)
+capture_15	asio12		FXBUS2(0xc)
+capture_16	asio13		FXBUS2(0xd)
+==============  ========        ============
+
+TODO: describe use of ld10k1/qlo10k1 in conjunction with JACK
diff --git a/Documentation/sound/cards/index.rst b/Documentation/sound/cards/index.rst
index c9d7ce4286b2..251f1d7675f7 100644
--- a/Documentation/sound/cards/index.rst
+++ b/Documentation/sound/cards/index.rst
@@ -8,4 +8,4 @@ Card-Specific Information
    cmipci
    sb-live-mixer
    audigy-mixer
-  
+   emu10k1-jack
-- 
cgit v1.2.3-58-ga151


From 312c01b1736e81fe0d23217fe537d415785256d2 Mon Sep 17 00:00:00 2001
From: Takashi Iwai 
Date: Thu, 10 Nov 2016 16:32:49 +0100
Subject: ALSA: doc: ReSTize VIA82xx-mixer.txt

Another simple conversion from a plain text file.
Put to cards directory.

Signed-off-by: Takashi Iwai 
---
 Documentation/sound/alsa/VIA82xx-mixer.txt  | 8 --------
 Documentation/sound/cards/index.rst         | 1 +
 Documentation/sound/cards/via82xx-mixer.rst | 8 ++++++++
 3 files changed, 9 insertions(+), 8 deletions(-)
 delete mode 100644 Documentation/sound/alsa/VIA82xx-mixer.txt
 create mode 100644 Documentation/sound/cards/via82xx-mixer.rst

(limited to 'Documentation/sound')

diff --git a/Documentation/sound/alsa/VIA82xx-mixer.txt b/Documentation/sound/alsa/VIA82xx-mixer.txt
deleted file mode 100644
index 1b0ac06ba95d..000000000000
--- a/Documentation/sound/alsa/VIA82xx-mixer.txt
+++ /dev/null
@@ -1,8 +0,0 @@
-
-				VIA82xx mixer
-				=============
-
-On many VIA82xx boards, the 'Input Source Select' mixer control does not work.
-Setting it to 'Input2' on such boards will cause recording to hang, or fail
-with EIO (input/output error) via OSS emulation.  This control should be left
-at 'Input1' for such cards.
diff --git a/Documentation/sound/cards/index.rst b/Documentation/sound/cards/index.rst
index 251f1d7675f7..4fcb88049ccc 100644
--- a/Documentation/sound/cards/index.rst
+++ b/Documentation/sound/cards/index.rst
@@ -9,3 +9,4 @@ Card-Specific Information
    sb-live-mixer
    audigy-mixer
    emu10k1-jack
+   via82xx-mixer
diff --git a/Documentation/sound/cards/via82xx-mixer.rst b/Documentation/sound/cards/via82xx-mixer.rst
new file mode 100644
index 000000000000..6ee993d4535b
--- /dev/null
+++ b/Documentation/sound/cards/via82xx-mixer.rst
@@ -0,0 +1,8 @@
+=============
+VIA82xx mixer
+=============
+
+On many VIA82xx boards, the ``Input Source Select`` mixer control does not work.
+Setting it to ``Input2`` on such boards will cause recording to hang, or fail
+with EIO (input/output error) via OSS emulation.  This control should be left
+at ``Input1`` for such cards.
-- 
cgit v1.2.3-58-ga151


From 4e47556e345c8c0bd86911a99f28299e3d962140 Mon Sep 17 00:00:00 2001
From: Takashi Iwai 
Date: Thu, 10 Nov 2016 16:52:57 +0100
Subject: ALSA: doc: ReSTize Audiophile-USB.txt

Another simple conversion from a plain text file.
Put to cards directory.

Signed-off-by: Takashi Iwai 
---
 Documentation/sound/alsa/Audiophile-Usb.txt  | 442 ---------------------
 Documentation/sound/cards/audiophile-usb.rst | 550 +++++++++++++++++++++++++++
 Documentation/sound/cards/index.rst          |   1 +
 3 files changed, 551 insertions(+), 442 deletions(-)
 delete mode 100644 Documentation/sound/alsa/Audiophile-Usb.txt
 create mode 100644 Documentation/sound/cards/audiophile-usb.rst

(limited to 'Documentation/sound')

diff --git a/Documentation/sound/alsa/Audiophile-Usb.txt b/Documentation/sound/alsa/Audiophile-Usb.txt
deleted file mode 100644
index e7a5ed4dcae8..000000000000
--- a/Documentation/sound/alsa/Audiophile-Usb.txt
+++ /dev/null
@@ -1,442 +0,0 @@
-	Guide to using M-Audio Audiophile USB with ALSA and Jack	v1.5
-	========================================================
-
-	    Thibault Le Meur 
-
-This document is a guide to using the M-Audio Audiophile USB (tm) device with 
-ALSA and JACK.
-
-History
-=======
-* v1.4 - Thibault Le Meur (2007-07-11)
- - Added Low Endianness nature of 16bits-modes
-   found by Hakan Lennestal 
- - Modifying document structure
-* v1.5 - Thibault Le Meur (2007-07-12)
- - Added AC3/DTS passthru info
-
-
-1 - Audiophile USB Specs and correct usage
-==========================================
-
-This part is a reminder of important facts about the functions and limitations 
-of the device.
-
-The device has 4 audio interfaces, and 2 MIDI ports:
- * Analog Stereo Input (Ai)
-   - This port supports 2 pairs of line-level audio inputs (1/4" TS and RCA) 
-   - When the 1/4" TS (jack) connectors are connected, the RCA connectors
-     are disabled
- * Analog Stereo Output (Ao)
- * Digital Stereo Input (Di)
- * Digital Stereo Output (Do)
- * Midi In (Mi)
- * Midi Out (Mo)
-
-The internal DAC/ADC has the following characteristics:
-* sample depth of 16 or 24 bits
-* sample rate from 8kHz to 96kHz
-* Two interfaces can't use different sample depths at the same time.
-Moreover, the Audiophile USB documentation gives the following Warning:
-"Please exit any audio application running before switching between bit depths"
-
-Due to the USB 1.1 bandwidth limitation, a limited number of interfaces can be 
-activated at the same time depending on the audio mode selected:
- * 16-bit/48kHz ==> 4 channels in + 4 channels out
-   - Ai+Ao+Di+Do
- * 24-bit/48kHz ==> 4 channels in + 2 channels out, 
-                    or 2 channels in + 4 channels out
-   - Ai+Ao+Do or Ai+Di+Ao or Ai+Di+Do or Di+Ao+Do
- * 24-bit/96kHz ==> 2 channels in _or_ 2 channels out (half duplex only)
-   - Ai or Ao or Di or Do
-
-Important facts about the Digital interface:
---------------------------------------------
- * The Do port additionally supports surround-encoded AC-3 and DTS passthrough, 
-though I haven't tested it under Linux
-   - Note that in this setup only the Do interface can be enabled
- * Apart from recording an audio digital stream, enabling the Di port is a way 
-to synchronize the device to an external sample clock
-   - As a consequence, the Di port must be enable only if an active Digital 
-source is connected
-   - Enabling Di when no digital source is connected can result in a 
-synchronization error (for instance sound played at an odd sample rate)
-
-
-2 - Audiophile USB MIDI support in ALSA
-=======================================
-
-The Audiophile USB MIDI ports will be automatically supported once the
-following modules have been loaded:
- * snd-usb-audio
- * snd-seq-midi
-
-No additional setting is required.
-
-
-3 - Audiophile USB Audio support in ALSA
-========================================
-
-Audio functions of the Audiophile USB device are handled by the snd-usb-audio 
-module. This module can work in a default mode (without any device-specific 
-parameter), or in an "advanced" mode with the device-specific parameter called 
-"device_setup".
-
-3.1 - Default Alsa driver mode
-------------------------------
-
-The default behavior of the snd-usb-audio driver is to list the device 
-capabilities at startup and activate the required mode when required 
-by the applications: for instance if the user is recording in a 
-24bit-depth-mode and immediately after wants to switch to a 16bit-depth mode,
-the snd-usb-audio module will reconfigure the device on the fly.
-
-This approach has the advantage to let the driver automatically switch from sample 
-rates/depths automatically according to the user's needs. However, those who 
-are using the device under windows know that this is not how the device is meant to
-work: under windows applications must be closed before using the m-audio control
-panel to switch the device working mode. Thus as we'll see in next section, this 
-Default Alsa driver mode can lead to device misconfigurations.
-
-Let's get back to the Default Alsa driver mode for now.  In this case the 
-Audiophile interfaces are mapped to alsa pcm devices in the following 
-way (I suppose the device's index is 1):
- * hw:1,0 is Ao in playback and Di in capture
- * hw:1,1 is Do in playback and Ai in capture
- * hw:1,2 is Do in AC3/DTS passthrough mode
-
-In this mode, the device uses Big Endian byte-encoding so that 
-supported audio format are S16_BE for 16-bit depth modes and S24_3BE for 
-24-bits depth mode.
-
-One exception is the hw:1,2 port which was reported to be Little Endian 
-compliant (supposedly supporting S16_LE) but processes in fact only S16_BE streams.
-This has been fixed in kernel 2.6.23 and above and now the hw:1,2 interface 
-is reported to be big endian in this default driver mode.
-
-Examples:
- * playing a S24_3BE encoded raw file to the Ao port
-   % aplay -D hw:1,0 -c2 -t raw -r48000 -fS24_3BE test.raw
- * recording a  S24_3BE encoded raw file from the Ai port
-   % arecord -D hw:1,1 -c2  -t raw -r48000 -fS24_3BE test.raw
- * playing a S16_BE encoded raw file to the Do port
-   % aplay -D hw:1,1 -c2 -t raw -r48000 -fS16_BE test.raw
- * playing an ac3 sample file to the Do port
-   % aplay -D hw:1,2 --channels=6 ac3_S16_BE_encoded_file.raw
-
-If you're happy with the default Alsa driver mode and don't experience any 
-issue with this mode, then you can skip the following chapter.
-
-3.2 - Advanced module setup
----------------------------
-
-Due to the hardware constraints described above, the device initialization made 
-by the Alsa driver in default mode may result in a corrupted state of the 
-device. For instance, a particularly annoying issue is that the sound captured 
-from the Ai interface sounds distorted (as if boosted with an excessive high
-volume gain).
-
-For people having this problem, the snd-usb-audio module has a new module 
-parameter called "device_setup" (this parameter was introduced in kernel
-release 2.6.17)
-
-3.2.1 - Initializing the working mode of the Audiophile USB
-
-As far as the Audiophile USB device is concerned, this value let the user 
-specify:
- * the sample depth
- * the sample rate
- * whether the Di port is used or not 
-
-When initialized with "device_setup=0x00", the snd-usb-audio module has
-the same behaviour as when the parameter is omitted (see paragraph "Default 
-Alsa driver mode" above)
-
-Others modes are described in the following subsections.
-
-3.2.1.1 - 16-bit modes
-
-The two supported modes are:
-
- * device_setup=0x01
-   - 16bits 48kHz mode with Di disabled
-   - Ai,Ao,Do can be used at the same time
-   - hw:1,0 is not available in capture mode
-   - hw:1,2 is not available
-
- * device_setup=0x11
-   - 16bits 48kHz mode with Di enabled
-   - Ai,Ao,Di,Do can be used at the same time
-   - hw:1,0 is available in capture mode
-   - hw:1,2 is not available
-
-In this modes the device operates only at 16bits-modes. Before kernel 2.6.23,
-the devices where reported to be Big-Endian when in fact they were Little-Endian
-so that playing a file was a matter of using:
-   % aplay -D hw:1,1 -c2 -t raw -r48000 -fS16_BE test_S16_LE.raw
-where "test_S16_LE.raw" was in fact a little-endian sample file.
-
-Thanks to Hakan Lennestal (who discovered the Little-Endiannes of the device in
-these modes) a fix has been committed (expected in kernel 2.6.23) and
-Alsa now reports Little-Endian interfaces. Thus playing a file now is as simple as
-using:
-   % aplay -D hw:1,1 -c2 -t raw -r48000 -fS16_LE test_S16_LE.raw
-
-3.2.1.2 - 24-bit modes
-
-The three supported modes are:
-
- * device_setup=0x09
-   - 24bits 48kHz mode with Di disabled
-   - Ai,Ao,Do can be used at the same time
-   - hw:1,0 is not available in capture mode
-   - hw:1,2 is not available
-
- * device_setup=0x19
-   - 24bits 48kHz mode with Di enabled
-   - 3 ports from {Ai,Ao,Di,Do} can be used at the same time
-   - hw:1,0 is available in capture mode and an active digital source must be 
-     connected to Di
-   - hw:1,2 is not available
-
- * device_setup=0x0D or 0x10
-   - 24bits 96kHz mode
-   - Di is enabled by default for this mode but does not need to be connected 
-     to an active source
-   - Only 1 port from {Ai,Ao,Di,Do} can be used at the same time
-   - hw:1,0 is available in captured mode
-   - hw:1,2 is not available
-
-In these modes the device is only Big-Endian compliant (see "Default Alsa driver 
-mode" above for an aplay command example)
-
-3.2.1.3 - AC3 w/ DTS passthru mode
-
-Thanks to Hakan Lennestal, I now have a report saying that this mode works.
-
- * device_setup=0x03
-   - 16bits 48kHz mode with only the Do port enabled 
-   - AC3 with DTS passthru
-   - Caution with this setup the Do port is mapped to the pcm device hw:1,0
-
-The command line used to playback the AC3/DTS encoded .wav-files in this mode:
-   % aplay -D hw:1,0 --channels=6 ac3_S16_LE_encoded_file.raw
-
-3.2.2 - How to use the device_setup parameter
-----------------------------------------------
-
-The parameter can be given:
-
- * By manually probing the device (as root):
-   # modprobe -r snd-usb-audio
-   # modprobe snd-usb-audio index=1 device_setup=0x09
-
- * Or while configuring the modules options in your modules configuration file
-   (typically a .conf file in /etc/modprobe.d/ directory:
-       alias snd-card-1 snd-usb-audio
-       options snd-usb-audio index=1 device_setup=0x09
-
-CAUTION when initializing the device
--------------------------------------
-
- * Correct initialization on the device requires that device_setup is given to
-   the module BEFORE the device is turned on. So, if you use the "manual probing"
-   method described above, take care to power-on the device AFTER this initialization.
-
- * Failing to respect this will lead to a misconfiguration of the device. In this case
-   turn off the device, unprobe the snd-usb-audio module, then probe it again with
-   correct device_setup parameter and then (and only then) turn on the device again.
-
- * If you've correctly initialized the device in a valid mode and then want to switch
-   to  another mode (possibly with another sample-depth), please use also the following 
-   procedure:
-   - first turn off the device
-   - de-register the snd-usb-audio module (modprobe -r)
-   - change the device_setup parameter by changing the device_setup
-     option in /etc/modprobe.d/*.conf
-   - turn on the device
- * A workaround for this last issue has been applied to kernel 2.6.23, but it may not
-   be enough to ensure the 'stability' of the device initialization.
-
-3.2.3 - Technical details for hackers
--------------------------------------
-This section is for hackers, wanting to understand details about the device
-internals and how Alsa supports it.
-
-3.2.3.1 - Audiophile USB's device_setup structure
-
-If you want to understand the device_setup magic numbers for the Audiophile 
-USB, you need some very basic understanding of binary computation. However, 
-this is not required to use the parameter and you may skip this section.
-
-The device_setup is one byte long and its structure is the following:
-
-       +---+---+---+---+---+---+---+---+
-       | b7| b6| b5| b4| b3| b2| b1| b0|
-       +---+---+---+---+---+---+---+---+
-       | 0 | 0 | 0 | Di|24B|96K|DTS|SET|
-       +---+---+---+---+---+---+---+---+
-
-Where:
- * b0 is the "SET" bit
-   - it MUST be set if device_setup is initialized 
- * b1 is the "DTS" bit
-   - it is set only for Digital output with DTS/AC3
-   - this setup is not tested
- * b2 is the Rate selection flag
-   - When set to "1" the rate range is 48.1-96kHz
-   - Otherwise the sample rate range is 8-48kHz
- * b3 is the bit depth selection flag
-   - When set to "1" samples are 24bits long
-   - Otherwise they are 16bits long
-   - Note that b2 implies b3 as the 96kHz mode is only supported for 24 bits 
-     samples
- * b4 is the Digital input flag
-   - When set to "1" the device assumes that an active digital source is 
-     connected 
-   - You shouldn't enable Di if no source is seen on the port (this leads to 
-     synchronization issues)
-   - b4 is implied by b2 (since only one port is enabled at a time no synch 
-     error can occur) 
- * b5 to b7 are reserved for future uses, and must be set to "0"
-   - might become Ao, Do, Ai, for b7, b6, b4 respectively
-
-Caution:
- * there is no check on the value you will give to device_setup
-   - for instance choosing 0x05 (16bits 96kHz) will fail back to 0x09 since 
-     b2 implies b3. But _there_will_be_no_warning_ in /var/log/messages
- * Hardware constraints due to the USB bus limitation aren't checked
-   - choosing b2 will prepare all interfaces for 24bits/96kHz but you'll
-     only be able to use one at the same time
-
-3.2.3.2 -  USB implementation details for this device
-
-You may safely skip this section if you're not interested in driver 
-hacking.
-
-This section describes some internal aspects of the device and summarizes the 
-data I got by usb-snooping the windows and Linux drivers.
-
-The M-Audio Audiophile USB has 7 USB Interfaces:
-a "USB interface":
- * USB Interface nb.0
- * USB Interface nb.1
-   - Audio Control function
- * USB Interface nb.2
-   - Analog Output
- * USB Interface nb.3
-   - Digital Output
- * USB Interface nb.4
-   - Analog Input
- * USB Interface nb.5
-   - Digital Input
- * USB Interface nb.6
-   - MIDI interface compliant with the MIDIMAN quirk 
-
-Each interface has 5 altsettings (AltSet 1,2,3,4,5) except:
- * Interface 3 (Digital Out) has an extra Alset nb.6 
- * Interface 5 (Digital In) does not have Alset nb.3 and 5 
-
-Here is a short description of the AltSettings capabilities:
- * AltSettings 1 corresponds to
-  - 24-bit depth, 48.1-96kHz sample mode
-  - Adaptive playback (Ao and Do), Synch capture (Ai), or Asynch capture (Di)
- * AltSettings 2 corresponds to
-  - 24-bit depth, 8-48kHz sample mode
-  - Asynch capture and playback  (Ao,Ai,Do,Di)
- * AltSettings 3 corresponds to
-  - 24-bit depth, 8-48kHz sample mode
-  - Synch capture (Ai) and Adaptive playback (Ao,Do)
- * AltSettings 4 corresponds to
-  - 16-bit depth, 8-48kHz sample mode
-  - Asynch capture and playback  (Ao,Ai,Do,Di)
- * AltSettings 5 corresponds to
-  - 16-bit depth, 8-48kHz sample mode
-  - Synch capture (Ai) and Adaptive playback (Ao,Do)
- * AltSettings 6 corresponds to
-  - 16-bit depth, 8-48kHz sample mode
-  - Synch playback (Do), audio format type III IEC1937_AC-3
-
-In order to ensure a correct initialization of the device, the driver 
-_must_know_ how the device will be used:
- * if DTS is chosen, only Interface 2 with AltSet nb.6 must be
-   registered
- * if 96KHz only AltSets nb.1 of each interface must be selected
- * if samples are using 24bits/48KHz then AltSet 2 must me used if
-   Digital input is connected, and only AltSet nb.3 if Digital input
-   is not connected
- * if samples are using 16bits/48KHz then AltSet 4 must me used if
-   Digital input is connected, and only AltSet nb.5 if Digital input
-   is not connected
-
-When device_setup is given as a parameter to the snd-usb-audio module, the 
-parse_audio_endpoints function uses a quirk called 
-"audiophile_skip_setting_quirk" in order to prevent AltSettings not 
-corresponding to device_setup from being registered in the driver.
-
-4 - Audiophile USB and Jack support
-===================================
-
-This section deals with support of the Audiophile USB device in Jack.
-
-There are 2 main potential issues when using Jackd with the device:
-* support for Big-Endian devices in 24-bit modes
-* support for 4-in / 4-out channels
-
-4.1 - Direct support in Jackd
------------------------------
-
-Jack supports big endian devices only in recent versions (thanks to
-Andreas Steinmetz for his first big-endian patch). I can't remember 
-exactly when this support was released into jackd, let's just say that
-with jackd version 0.103.0 it's almost ok (just a small bug is affecting 
-16bits Big-Endian devices, but since you've read carefully the above
-paragraphs, you're now using kernel >= 2.6.23 and your 16bits devices 
-are now Little Endians ;-) ).
-
-You can run jackd with the following command for playback with Ao and
-record with Ai:
-  % jackd -R -dalsa -Phw:1,0 -r48000 -p128 -n2 -D -Chw:1,1
-
-4.2 - Using Alsa plughw
------------------------
-If you don't have a recent Jackd installed, you can downgrade to using
-the Alsa "plug" converter.
-
-For instance here is one way to run Jack with 2 playback channels on Ao and 2 
-capture channels from Ai:
-  % jackd -R -dalsa -dplughw:1 -r48000 -p256 -n2 -D -Cplughw:1,1
-
-However you may see the following warning message:
-"You appear to be using the ALSA software "plug" layer, probably a result of 
-using the "default" ALSA device. This is less efficient than it could be. 
-Consider using a hardware device instead rather than using the plug layer."
-
-4.3 - Getting 2 input and/or output interfaces in Jack
-------------------------------------------------------
-
-As you can see, starting the Jack server this way will only enable 1 stereo
-input (Di or Ai) and 1 stereo output (Ao or Do).
-
-This is due to the following restrictions:
-* Jack can only open one capture device and one playback device at a time
-* The Audiophile USB is seen as 2 (or three) Alsa devices: hw:1,0, hw:1,1
-  (and optionally hw:1,2)
-
-If you want to get Ai+Di and/or Ao+Do support with Jack, you would need to
-combine the Alsa devices into one logical "complex" device.
-
-If you want to give it a try, I recommend reading the information from
-this page: http://www.sound-man.co.uk/linuxaudio/ice1712multi.html
-It is related to another device (ice1712) but can be adapted to suit
-the Audiophile USB.
-
-Enabling multiple Audiophile USB interfaces for Jackd will certainly require:
-* Making sure your Jackd version has the MMAP_COMPLEX patch (see the ice1712 page)
-* (maybe) patching the alsa-lib/src/pcm/pcm_multi.c file (see the ice1712 page)
-* define a multi device (combination of hw:1,0 and hw:1,1) in your .asoundrc
-  file 
-* start jackd with this device
-
-I had no success in testing this for now, if you have any success with this kind 
-of setup, please drop me an email.
diff --git a/Documentation/sound/cards/audiophile-usb.rst b/Documentation/sound/cards/audiophile-usb.rst
new file mode 100644
index 000000000000..a7bb5648331f
--- /dev/null
+++ b/Documentation/sound/cards/audiophile-usb.rst
@@ -0,0 +1,550 @@
+========================================================
+Guide to using M-Audio Audiophile USB with ALSA and Jack
+========================================================
+
+v1.5
+
+Thibault Le Meur 
+
+This document is a guide to using the M-Audio Audiophile USB (tm) device with 
+ALSA and JACK.
+
+History
+=======
+
+* v1.4 - Thibault Le Meur (2007-07-11)
+
+  - Added Low Endianness nature of 16bits-modes
+    found by Hakan Lennestal 
+  - Modifying document structure
+
+* v1.5 - Thibault Le Meur (2007-07-12)
+  - Added AC3/DTS passthru info
+
+
+Audiophile USB Specs and correct usage
+======================================
+
+This part is a reminder of important facts about the functions and limitations 
+of the device.
+
+The device has 4 audio interfaces, and 2 MIDI ports:
+
+ * Analog Stereo Input (Ai)
+
+   - This port supports 2 pairs of line-level audio inputs (1/4" TS and RCA) 
+   - When the 1/4" TS (jack) connectors are connected, the RCA connectors
+     are disabled
+
+ * Analog Stereo Output (Ao)
+ * Digital Stereo Input (Di)
+ * Digital Stereo Output (Do)
+ * Midi In (Mi)
+ * Midi Out (Mo)
+
+The internal DAC/ADC has the following characteristics:
+
+* sample depth of 16 or 24 bits
+* sample rate from 8kHz to 96kHz
+* Two interfaces can't use different sample depths at the same time.
+
+Moreover, the Audiophile USB documentation gives the following Warning:
+  Please exit any audio application running before switching between bit depths
+
+Due to the USB 1.1 bandwidth limitation, a limited number of interfaces can be 
+activated at the same time depending on the audio mode selected:
+
+ * 16-bit/48kHz ==> 4 channels in + 4 channels out
+
+   - Ai+Ao+Di+Do
+
+ * 24-bit/48kHz ==> 4 channels in + 2 channels out, 
+   or 2 channels in + 4 channels out
+
+   - Ai+Ao+Do or Ai+Di+Ao or Ai+Di+Do or Di+Ao+Do
+
+ * 24-bit/96kHz ==> 2 channels in _or_ 2 channels out (half duplex only)
+
+   - Ai or Ao or Di or Do
+
+Important facts about the Digital interface:
+--------------------------------------------
+
+ * The Do port additionally supports surround-encoded AC-3 and DTS passthrough, 
+   though I haven't tested it under Linux
+
+   - Note that in this setup only the Do interface can be enabled
+
+ * Apart from recording an audio digital stream, enabling the Di port is a way 
+   to synchronize the device to an external sample clock
+
+   - As a consequence, the Di port must be enable only if an active Digital 
+     source is connected
+   - Enabling Di when no digital source is connected can result in a 
+     synchronization error (for instance sound played at an odd sample rate)
+
+
+Audiophile USB MIDI support in ALSA
+===================================
+
+The Audiophile USB MIDI ports will be automatically supported once the
+following modules have been loaded:
+
+ * snd-usb-audio
+ * snd-seq-midi
+
+No additional setting is required.
+
+
+Audiophile USB Audio support in ALSA
+====================================
+
+Audio functions of the Audiophile USB device are handled by the snd-usb-audio 
+module. This module can work in a default mode (without any device-specific 
+parameter), or in an "advanced" mode with the device-specific parameter called 
+``device_setup``.
+
+Default Alsa driver mode
+------------------------
+
+The default behavior of the snd-usb-audio driver is to list the device 
+capabilities at startup and activate the required mode when required 
+by the applications: for instance if the user is recording in a 
+24bit-depth-mode and immediately after wants to switch to a 16bit-depth mode,
+the snd-usb-audio module will reconfigure the device on the fly.
+
+This approach has the advantage to let the driver automatically switch from sample 
+rates/depths automatically according to the user's needs. However, those who 
+are using the device under windows know that this is not how the device is meant to
+work: under windows applications must be closed before using the m-audio control
+panel to switch the device working mode. Thus as we'll see in next section, this 
+Default Alsa driver mode can lead to device misconfigurations.
+
+Let's get back to the Default Alsa driver mode for now.  In this case the 
+Audiophile interfaces are mapped to alsa pcm devices in the following 
+way (I suppose the device's index is 1):
+
+ * hw:1,0 is Ao in playback and Di in capture
+ * hw:1,1 is Do in playback and Ai in capture
+ * hw:1,2 is Do in AC3/DTS passthrough mode
+
+In this mode, the device uses Big Endian byte-encoding so that 
+supported audio format are S16_BE for 16-bit depth modes and S24_3BE for 
+24-bits depth mode.
+
+One exception is the hw:1,2 port which was reported to be Little Endian 
+compliant (supposedly supporting S16_LE) but processes in fact only S16_BE streams.
+This has been fixed in kernel 2.6.23 and above and now the hw:1,2 interface 
+is reported to be big endian in this default driver mode.
+
+Examples:
+
+ * playing a S24_3BE encoded raw file to the Ao port::
+
+   % aplay -D hw:1,0 -c2 -t raw -r48000 -fS24_3BE test.raw
+
+ * recording a  S24_3BE encoded raw file from the Ai port::
+
+   % arecord -D hw:1,1 -c2  -t raw -r48000 -fS24_3BE test.raw
+
+ * playing a S16_BE encoded raw file to the Do port::
+
+   % aplay -D hw:1,1 -c2 -t raw -r48000 -fS16_BE test.raw
+
+ * playing an ac3 sample file to the Do port::
+
+   % aplay -D hw:1,2 --channels=6 ac3_S16_BE_encoded_file.raw
+
+If you're happy with the default Alsa driver mode and don't experience any 
+issue with this mode, then you can skip the following chapter.
+
+Advanced module setup
+---------------------
+
+Due to the hardware constraints described above, the device initialization made 
+by the Alsa driver in default mode may result in a corrupted state of the 
+device. For instance, a particularly annoying issue is that the sound captured 
+from the Ai interface sounds distorted (as if boosted with an excessive high
+volume gain).
+
+For people having this problem, the snd-usb-audio module has a new module 
+parameter called ``device_setup`` (this parameter was introduced in kernel
+release 2.6.17)
+
+Initializing the working mode of the Audiophile USB
+~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+
+As far as the Audiophile USB device is concerned, this value let the user 
+specify:
+
+ * the sample depth
+ * the sample rate
+ * whether the Di port is used or not 
+
+When initialized with ``device_setup=0x00``, the snd-usb-audio module has
+the same behaviour as when the parameter is omitted (see paragraph "Default 
+Alsa driver mode" above)
+
+Others modes are described in the following subsections.
+
+16-bit modes
+~~~~~~~~~~~~
+
+The two supported modes are:
+
+ * ``device_setup=0x01``
+
+   - 16bits 48kHz mode with Di disabled
+   - Ai,Ao,Do can be used at the same time
+   - hw:1,0 is not available in capture mode
+   - hw:1,2 is not available
+
+ * ``device_setup=0x11``
+
+   - 16bits 48kHz mode with Di enabled
+   - Ai,Ao,Di,Do can be used at the same time
+   - hw:1,0 is available in capture mode
+   - hw:1,2 is not available
+
+In this modes the device operates only at 16bits-modes. Before kernel 2.6.23,
+the devices where reported to be Big-Endian when in fact they were Little-Endian
+so that playing a file was a matter of using:
+::
+
+   % aplay -D hw:1,1 -c2 -t raw -r48000 -fS16_BE test_S16_LE.raw
+
+where "test_S16_LE.raw" was in fact a little-endian sample file.
+
+Thanks to Hakan Lennestal (who discovered the Little-Endiannes of the device in
+these modes) a fix has been committed (expected in kernel 2.6.23) and
+Alsa now reports Little-Endian interfaces. Thus playing a file now is as simple as
+using:
+::
+
+   % aplay -D hw:1,1 -c2 -t raw -r48000 -fS16_LE test_S16_LE.raw
+
+
+24-bit modes
+~~~~~~~~~~~~
+
+The three supported modes are:
+
+ * ``device_setup=0x09``
+
+   - 24bits 48kHz mode with Di disabled
+   - Ai,Ao,Do can be used at the same time
+   - hw:1,0 is not available in capture mode
+   - hw:1,2 is not available
+
+ * ``device_setup=0x19``
+
+   - 24bits 48kHz mode with Di enabled
+   - 3 ports from {Ai,Ao,Di,Do} can be used at the same time
+   - hw:1,0 is available in capture mode and an active digital source must be 
+     connected to Di
+   - hw:1,2 is not available
+
+ * ``device_setup=0x0D`` or ``0x10``
+
+   - 24bits 96kHz mode
+   - Di is enabled by default for this mode but does not need to be connected 
+     to an active source
+   - Only 1 port from {Ai,Ao,Di,Do} can be used at the same time
+   - hw:1,0 is available in captured mode
+   - hw:1,2 is not available
+
+In these modes the device is only Big-Endian compliant (see "Default Alsa driver 
+mode" above for an aplay command example)
+
+AC3 w/ DTS passthru mode
+~~~~~~~~~~~~~~~~~~~~~~~~
+
+Thanks to Hakan Lennestal, I now have a report saying that this mode works.
+
+ * ``device_setup=0x03``
+
+   - 16bits 48kHz mode with only the Do port enabled 
+   - AC3 with DTS passthru
+   - Caution with this setup the Do port is mapped to the pcm device hw:1,0
+
+The command line used to playback the AC3/DTS encoded .wav-files in this mode:
+::
+
+   % aplay -D hw:1,0 --channels=6 ac3_S16_LE_encoded_file.raw
+
+How to use the ``device_setup`` parameter
+~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+
+The parameter can be given:
+
+ * By manually probing the device (as root):::
+
+   # modprobe -r snd-usb-audio
+   # modprobe snd-usb-audio index=1 device_setup=0x09
+
+ * Or while configuring the modules options in your modules configuration file
+   (typically a .conf file in /etc/modprobe.d/ directory:::
+
+       alias snd-card-1 snd-usb-audio
+       options snd-usb-audio index=1 device_setup=0x09
+
+CAUTION when initializing the device
+-------------------------------------
+
+ * Correct initialization on the device requires that device_setup is given to
+   the module BEFORE the device is turned on. So, if you use the "manual probing"
+   method described above, take care to power-on the device AFTER this initialization.
+
+ * Failing to respect this will lead to a misconfiguration of the device. In this case
+   turn off the device, unprobe the snd-usb-audio module, then probe it again with
+   correct device_setup parameter and then (and only then) turn on the device again.
+
+ * If you've correctly initialized the device in a valid mode and then want to switch
+   to  another mode (possibly with another sample-depth), please use also the following 
+   procedure:
+
+   - first turn off the device
+   - de-register the snd-usb-audio module (modprobe -r)
+   - change the device_setup parameter by changing the device_setup
+     option in ``/etc/modprobe.d/*.conf``
+   - turn on the device
+
+ * A workaround for this last issue has been applied to kernel 2.6.23, but it may not
+   be enough to ensure the 'stability' of the device initialization.
+
+Technical details for hackers
+-----------------------------
+
+This section is for hackers, wanting to understand details about the device
+internals and how Alsa supports it.
+
+Audiophile USB's ``device_setup`` structure
+~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+
+If you want to understand the device_setup magic numbers for the Audiophile 
+USB, you need some very basic understanding of binary computation. However, 
+this is not required to use the parameter and you may skip this section.
+
+The device_setup is one byte long and its structure is the following:
+::
+
+       +---+---+---+---+---+---+---+---+
+       | b7| b6| b5| b4| b3| b2| b1| b0|
+       +---+---+---+---+---+---+---+---+
+       | 0 | 0 | 0 | Di|24B|96K|DTS|SET|
+       +---+---+---+---+---+---+---+---+
+
+Where:
+
+ * b0 is the ``SET`` bit
+
+   - it MUST be set if device_setup is initialized 
+
+ * b1 is the ``DTS`` bit
+
+   - it is set only for Digital output with DTS/AC3
+   - this setup is not tested
+
+ * b2 is the Rate selection flag
+
+   - When set to ``1`` the rate range is 48.1-96kHz
+   - Otherwise the sample rate range is 8-48kHz
+
+ * b3 is the bit depth selection flag
+
+   - When set to ``1`` samples are 24bits long
+   - Otherwise they are 16bits long
+   - Note that b2 implies b3 as the 96kHz mode is only supported for 24 bits 
+     samples
+
+ * b4 is the Digital input flag
+
+   - When set to ``1`` the device assumes that an active digital source is 
+     connected 
+   - You shouldn't enable Di if no source is seen on the port (this leads to 
+     synchronization issues)
+   - b4 is implied by b2 (since only one port is enabled at a time no synch 
+     error can occur) 
+
+ * b5 to b7 are reserved for future uses, and must be set to ``0``
+
+   - might become Ao, Do, Ai, for b7, b6, b4 respectively
+
+Caution:
+
+ * there is no check on the value you will give to device_setup
+
+   - for instance choosing 0x05 (16bits 96kHz) will fail back to 0x09 since 
+     b2 implies b3. But _there_will_be_no_warning_ in /var/log/messages
+
+ * Hardware constraints due to the USB bus limitation aren't checked
+
+   - choosing b2 will prepare all interfaces for 24bits/96kHz but you'll
+     only be able to use one at the same time
+
+USB implementation details for this device
+~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+
+You may safely skip this section if you're not interested in driver 
+hacking.
+
+This section describes some internal aspects of the device and summarizes the 
+data I got by usb-snooping the windows and Linux drivers.
+
+The M-Audio Audiophile USB has 7 USB Interfaces:
+a "USB interface":
+
+ * USB Interface nb.0
+ * USB Interface nb.1
+
+   - Audio Control function
+
+ * USB Interface nb.2
+
+   - Analog Output
+
+ * USB Interface nb.3
+
+   - Digital Output
+
+ * USB Interface nb.4
+
+   - Analog Input
+
+ * USB Interface nb.5
+
+   - Digital Input
+
+ * USB Interface nb.6
+
+   - MIDI interface compliant with the MIDIMAN quirk 
+
+Each interface has 5 altsettings (AltSet 1,2,3,4,5) except:
+
+ * Interface 3 (Digital Out) has an extra Alset nb.6 
+ * Interface 5 (Digital In) does not have Alset nb.3 and 5 
+
+Here is a short description of the AltSettings capabilities:
+
+* AltSettings 1 corresponds to
+
+  - 24-bit depth, 48.1-96kHz sample mode
+  - Adaptive playback (Ao and Do), Synch capture (Ai), or Asynch capture (Di)
+
+* AltSettings 2 corresponds to
+
+  - 24-bit depth, 8-48kHz sample mode
+  - Asynch capture and playback  (Ao,Ai,Do,Di)
+
+* AltSettings 3 corresponds to
+
+  - 24-bit depth, 8-48kHz sample mode
+  - Synch capture (Ai) and Adaptive playback (Ao,Do)
+
+* AltSettings 4 corresponds to
+
+  - 16-bit depth, 8-48kHz sample mode
+  - Asynch capture and playback  (Ao,Ai,Do,Di)
+
+* AltSettings 5 corresponds to
+
+  - 16-bit depth, 8-48kHz sample mode
+  - Synch capture (Ai) and Adaptive playback (Ao,Do)
+
+* AltSettings 6 corresponds to
+
+  - 16-bit depth, 8-48kHz sample mode
+  - Synch playback (Do), audio format type III IEC1937_AC-3
+
+In order to ensure a correct initialization of the device, the driver 
+*must* *know* how the device will be used:
+
+ * if DTS is chosen, only Interface 2 with AltSet nb.6 must be
+   registered
+ * if 96KHz only AltSets nb.1 of each interface must be selected
+ * if samples are using 24bits/48KHz then AltSet 2 must me used if
+   Digital input is connected, and only AltSet nb.3 if Digital input
+   is not connected
+ * if samples are using 16bits/48KHz then AltSet 4 must me used if
+   Digital input is connected, and only AltSet nb.5 if Digital input
+   is not connected
+
+When device_setup is given as a parameter to the snd-usb-audio module, the 
+parse_audio_endpoints function uses a quirk called 
+``audiophile_skip_setting_quirk`` in order to prevent AltSettings not 
+corresponding to device_setup from being registered in the driver.
+
+Audiophile USB and Jack support
+===============================
+
+This section deals with support of the Audiophile USB device in Jack.
+
+There are 2 main potential issues when using Jackd with the device:
+
+* support for Big-Endian devices in 24-bit modes
+* support for 4-in / 4-out channels
+
+Direct support in Jackd
+-----------------------
+
+Jack supports big endian devices only in recent versions (thanks to
+Andreas Steinmetz for his first big-endian patch). I can't remember 
+exactly when this support was released into jackd, let's just say that
+with jackd version 0.103.0 it's almost ok (just a small bug is affecting 
+16bits Big-Endian devices, but since you've read carefully the above
+paragraphs, you're now using kernel >= 2.6.23 and your 16bits devices 
+are now Little Endians ;-) ).
+
+You can run jackd with the following command for playback with Ao and
+record with Ai:
+::
+
+  % jackd -R -dalsa -Phw:1,0 -r48000 -p128 -n2 -D -Chw:1,1
+
+Using Alsa plughw
+-----------------
+
+If you don't have a recent Jackd installed, you can downgrade to using
+the Alsa ``plug`` converter.
+
+For instance here is one way to run Jack with 2 playback channels on Ao and 2 
+capture channels from Ai:
+::
+
+  % jackd -R -dalsa -dplughw:1 -r48000 -p256 -n2 -D -Cplughw:1,1
+
+However you may see the following warning message:
+  You appear to be using the ALSA software "plug" layer, probably a result of 
+  using the "default" ALSA device. This is less efficient than it could be. 
+  Consider using a hardware device instead rather than using the plug layer.
+
+Getting 2 input and/or output interfaces in Jack
+------------------------------------------------
+
+As you can see, starting the Jack server this way will only enable 1 stereo
+input (Di or Ai) and 1 stereo output (Ao or Do).
+
+This is due to the following restrictions:
+
+* Jack can only open one capture device and one playback device at a time
+* The Audiophile USB is seen as 2 (or three) Alsa devices: hw:1,0, hw:1,1
+  (and optionally hw:1,2)
+
+If you want to get Ai+Di and/or Ao+Do support with Jack, you would need to
+combine the Alsa devices into one logical "complex" device.
+
+If you want to give it a try, I recommend reading the information from
+this page: http://www.sound-man.co.uk/linuxaudio/ice1712multi.html
+It is related to another device (ice1712) but can be adapted to suit
+the Audiophile USB.
+
+Enabling multiple Audiophile USB interfaces for Jackd will certainly require:
+
+* Making sure your Jackd version has the MMAP_COMPLEX patch (see the ice1712 page)
+* (maybe) patching the alsa-lib/src/pcm/pcm_multi.c file (see the ice1712 page)
+* define a multi device (combination of hw:1,0 and hw:1,1) in your .asoundrc
+  file 
+* start jackd with this device
+
+I had no success in testing this for now, if you have any success with this kind 
+of setup, please drop me an email.
diff --git a/Documentation/sound/cards/index.rst b/Documentation/sound/cards/index.rst
index 4fcb88049ccc..e92a4f7e86fc 100644
--- a/Documentation/sound/cards/index.rst
+++ b/Documentation/sound/cards/index.rst
@@ -10,3 +10,4 @@ Card-Specific Information
    audigy-mixer
    emu10k1-jack
    via82xx-mixer
+   audiophile-usb
-- 
cgit v1.2.3-58-ga151


From 3d8e81862ce404d861b75bcb1e7d986e4febb025 Mon Sep 17 00:00:00 2001
From: Takashi Iwai 
Date: Thu, 10 Nov 2016 16:56:09 +0100
Subject: ALSA: doc: ReSTize MIXART.txt

Another simple conversion from a plain text file.
Put to cards directory.

Signed-off-by: Takashi Iwai 
---
 Documentation/sound/alsa/MIXART.txt  | 100 -------------------------------
 Documentation/sound/cards/index.rst  |   1 +
 Documentation/sound/cards/mixart.rst | 110 +++++++++++++++++++++++++++++++++++
 3 files changed, 111 insertions(+), 100 deletions(-)
 delete mode 100644 Documentation/sound/alsa/MIXART.txt
 create mode 100644 Documentation/sound/cards/mixart.rst

(limited to 'Documentation/sound')

diff --git a/Documentation/sound/alsa/MIXART.txt b/Documentation/sound/alsa/MIXART.txt
deleted file mode 100644
index 4ee35b4fbe4a..000000000000
--- a/Documentation/sound/alsa/MIXART.txt
+++ /dev/null
@@ -1,100 +0,0 @@
-    Alsa driver for Digigram miXart8 and miXart8AES/EBU soundcards
-	    Digigram 
-
-
-GENERAL
-=======
-
-The miXart8 is a multichannel audio processing and mixing soundcard
-that has 4 stereo audio inputs and 4 stereo audio outputs.
-The miXart8AES/EBU is the same with a add-on card that offers further
-4 digital stereo audio inputs and outputs.
-Furthermore the add-on card offers external clock synchronisation
-(AES/EBU, Word Clock, Time Code and Video Synchro)
-
-The mainboard has a PowerPC that offers onboard mpeg encoding and
-decoding, samplerate conversions and various effects.
-
-The driver don't work properly at all until the certain firmwares
-are loaded, i.e. no PCM nor mixer devices will appear.
-Use the mixartloader that can be found in the alsa-tools package.
-
-
-VERSION 0.1.0
-=============
-
-One miXart8 board will be represented as 4 alsa cards, each with 1
-stereo analog capture 'pcm0c' and 1 stereo analog playback 'pcm0p' device.
-With a miXart8AES/EBU there is in addition 1 stereo digital input
-'pcm1c' and 1 stereo digital output 'pcm1p' per card.
-
-Formats
--------
-U8, S16_LE, S16_BE, S24_3LE, S24_3BE, FLOAT_LE, FLOAT_BE
-Sample rates : 8000 - 48000 Hz continuously
-
-Playback
---------
-For instance the playback devices are configured to have max. 4
-substreams performing hardware mixing. This could be changed to a
-maximum of 24 substreams if wished.
-Mono files will be played on the left and right channel. Each channel
-can be muted for each stream to use 8 analog/digital outputs separately.
-
-Capture
--------
-There is one substream per capture device. For instance only stereo
-formats are supported.
-
-Mixer
------
- and  : analog volume control of playback and capture PCM.
- and  : digital volume control of each analog substream.
- and  : digital volume control of each AES/EBU substream.
- : Loopback from 'pcm0c' to 'pcm0p' with digital volume
-and mute control.
-
-Rem : for best audio quality try to keep a 0 attenuation on the PCM
-and AES volume controls which is set by 219 in the range from 0 to 255
-(about 86% with alsamixer)
-
-
-NOT YET IMPLEMENTED
-===================
-
-- external clock support (AES/EBU, Word Clock, Time Code, Video Sync)
-- MPEG audio formats
-- mono record
-- on-board effects and samplerate conversions
-- linked streams
-
-
-FIRMWARE
-========
-
-[As of 2.6.11, the firmware can be loaded automatically with hotplug
- when CONFIG_FW_LOADER is set.  The mixartloader is necessary only
- for older versions or when you build the driver into kernel.]
- 
-For loading the firmware automatically after the module is loaded, use a
-install command.  For example, add the following entry to
-/etc/modprobe.d/mixart.conf for miXart driver:
-
-	install snd-mixart /sbin/modprobe --first-time -i snd-mixart && \
-			   /usr/bin/mixartloader
-(for 2.2/2.4 kernels, add "post-install snd-mixart /usr/bin/vxloader" to
- /etc/modules.conf, instead.)
-
-The firmware binaries are installed on /usr/share/alsa/firmware
-(or /usr/local/share/alsa/firmware, depending to the prefix option of
-configure).  There will be a miXart.conf file, which define the dsp image
-files.
-
-The firmware files are copyright by Digigram SA
-
-
-COPYRIGHT
-=========
-
-Copyright (c) 2003 Digigram SA 
-Distributable under GPL.
diff --git a/Documentation/sound/cards/index.rst b/Documentation/sound/cards/index.rst
index e92a4f7e86fc..0006bc547bb8 100644
--- a/Documentation/sound/cards/index.rst
+++ b/Documentation/sound/cards/index.rst
@@ -11,3 +11,4 @@ Card-Specific Information
    emu10k1-jack
    via82xx-mixer
    audiophile-usb
+   mixart
diff --git a/Documentation/sound/cards/mixart.rst b/Documentation/sound/cards/mixart.rst
new file mode 100644
index 000000000000..48aba98b088f
--- /dev/null
+++ b/Documentation/sound/cards/mixart.rst
@@ -0,0 +1,110 @@
+==============================================================
+Alsa driver for Digigram miXart8 and miXart8AES/EBU soundcards
+==============================================================
+
+Digigram 
+
+
+GENERAL
+=======
+
+The miXart8 is a multichannel audio processing and mixing soundcard
+that has 4 stereo audio inputs and 4 stereo audio outputs.
+The miXart8AES/EBU is the same with a add-on card that offers further
+4 digital stereo audio inputs and outputs.
+Furthermore the add-on card offers external clock synchronisation
+(AES/EBU, Word Clock, Time Code and Video Synchro)
+
+The mainboard has a PowerPC that offers onboard mpeg encoding and
+decoding, samplerate conversions and various effects.
+
+The driver don't work properly at all until the certain firmwares
+are loaded, i.e. no PCM nor mixer devices will appear.
+Use the mixartloader that can be found in the alsa-tools package.
+
+
+VERSION 0.1.0
+=============
+
+One miXart8 board will be represented as 4 alsa cards, each with 1
+stereo analog capture 'pcm0c' and 1 stereo analog playback 'pcm0p' device.
+With a miXart8AES/EBU there is in addition 1 stereo digital input
+'pcm1c' and 1 stereo digital output 'pcm1p' per card.
+
+Formats
+-------
+U8, S16_LE, S16_BE, S24_3LE, S24_3BE, FLOAT_LE, FLOAT_BE
+Sample rates : 8000 - 48000 Hz continuously
+
+Playback
+--------
+For instance the playback devices are configured to have max. 4
+substreams performing hardware mixing. This could be changed to a
+maximum of 24 substreams if wished.
+Mono files will be played on the left and right channel. Each channel
+can be muted for each stream to use 8 analog/digital outputs separately.
+
+Capture
+-------
+There is one substream per capture device. For instance only stereo
+formats are supported.
+
+Mixer
+-----
+ and 
+	analog volume control of playback and capture PCM.
+ and 
+	digital volume control of each analog substream.
+ and 
+	digital volume control of each AES/EBU substream.
+
+	Loopback from 'pcm0c' to 'pcm0p' with digital volume
+	and mute control.
+
+Rem : for best audio quality try to keep a 0 attenuation on the PCM
+and AES volume controls which is set by 219 in the range from 0 to 255
+(about 86% with alsamixer)
+
+
+NOT YET IMPLEMENTED
+===================
+
+- external clock support (AES/EBU, Word Clock, Time Code, Video Sync)
+- MPEG audio formats
+- mono record
+- on-board effects and samplerate conversions
+- linked streams
+
+
+FIRMWARE
+========
+
+[As of 2.6.11, the firmware can be loaded automatically with hotplug
+ when CONFIG_FW_LOADER is set.  The mixartloader is necessary only
+ for older versions or when you build the driver into kernel.]
+ 
+For loading the firmware automatically after the module is loaded, use a
+install command.  For example, add the following entry to
+/etc/modprobe.d/mixart.conf for miXart driver:
+::
+
+	install snd-mixart /sbin/modprobe --first-time -i snd-mixart && \
+			   /usr/bin/mixartloader
+
+
+(for 2.2/2.4 kernels, add "post-install snd-mixart /usr/bin/vxloader" to
+/etc/modules.conf, instead.)
+
+The firmware binaries are installed on /usr/share/alsa/firmware
+(or /usr/local/share/alsa/firmware, depending to the prefix option of
+configure).  There will be a miXart.conf file, which define the dsp image
+files.
+
+The firmware files are copyright by Digigram SA
+
+
+COPYRIGHT
+=========
+
+Copyright (c) 2003 Digigram SA 
+Distributable under GPL.
-- 
cgit v1.2.3-58-ga151


From 7bb97dfdcaca0963d59f986e0a88d5bcd0783552 Mon Sep 17 00:00:00 2001
From: Takashi Iwai 
Date: Thu, 10 Nov 2016 16:59:08 +0100
Subject: ALSA: doc: ReSTize Bt87x.txt

Another	simple conversion from a plain text file.
Put to cards directory.

Signed-off-by: Takashi Iwai 
---
 Documentation/sound/alsa/Bt87x.txt  | 78 ----------------------------------
 Documentation/sound/cards/bt87x.rst | 83 +++++++++++++++++++++++++++++++++++++
 Documentation/sound/cards/index.rst |  1 +
 3 files changed, 84 insertions(+), 78 deletions(-)
 delete mode 100644 Documentation/sound/alsa/Bt87x.txt
 create mode 100644 Documentation/sound/cards/bt87x.rst

(limited to 'Documentation/sound')

diff --git a/Documentation/sound/alsa/Bt87x.txt b/Documentation/sound/alsa/Bt87x.txt
deleted file mode 100644
index f158cde8b065..000000000000
--- a/Documentation/sound/alsa/Bt87x.txt
+++ /dev/null
@@ -1,78 +0,0 @@
-Intro
-=====
-
-You might have noticed that the bt878 grabber cards have actually
-_two_ PCI functions:
-
-$ lspci
-[ ... ]
-00:0a.0 Multimedia video controller: Brooktree Corporation Bt878 (rev 02)
-00:0a.1 Multimedia controller: Brooktree Corporation Bt878 (rev 02)
-[ ... ]
-
-The first does video, it is backward compatible to the bt848.  The second
-does audio.  snd-bt87x is a driver for the second function.  It's a sound
-driver which can be used for recording sound (and _only_ recording, no
-playback).  As most TV cards come with a short cable which can be plugged
-into your sound card's line-in you probably don't need this driver if all
-you want to do is just watching TV...
-
-Some cards do not bother to connect anything to the audio input pins of
-the chip, and some other cards use the audio function to transport MPEG
-video data, so it's quite possible that audio recording may not work
-with your card.
-
-
-Driver Status
-=============
-
-The driver is now stable.  However, it doesn't know about many TV cards,
-and it refuses to load for cards it doesn't know.
-
-If the driver complains ("Unknown TV card found, the audio driver will
-not load"), you can specify the load_all=1 option to force the driver to
-try to use the audio capture function of your card.  If the frequency of
-recorded data is not right, try to specify the digital_rate option with
-other values than the default 32000 (often it's 44100 or 64000).
-
-If you have an unknown card, please mail the ID and board name to
-, regardless of whether audio capture works
-or not, so that future versions of this driver know about your card.
-
-
-Audio modes
-===========
-
-The chip knows two different modes (digital/analog).  snd-bt87x
-registers two PCM devices, one for each mode.  They cannot be used at
-the same time.
-
-
-Digital audio mode
-==================
-
-The first device (hw:X,0) gives you 16 bit stereo sound.  The sample
-rate depends on the external source which feeds the Bt87x with digital
-sound via I2S interface.
-
-
-Analog audio mode (A/D)
-=======================
-
-The second device (hw:X,1) gives you 8 or 16 bit mono sound.  Supported
-sample rates are between 119466 and 448000 Hz (yes, these numbers are
-that high).  If you've set the CONFIG_SND_BT87X_OVERCLOCK option, the
-maximum sample rate is 1792000 Hz, but audio data becomes unusable
-beyond 896000 Hz on my card.
-
-The chip has three analog inputs.  Consequently you'll get a mixer
-device to control these.
-
-
-Have fun,
-
-  Clemens
-
-
-Written by Clemens Ladisch 
-big parts copied from btaudio.txt by Gerd Knorr 
diff --git a/Documentation/sound/cards/bt87x.rst b/Documentation/sound/cards/bt87x.rst
new file mode 100644
index 000000000000..912732d3ef9e
--- /dev/null
+++ b/Documentation/sound/cards/bt87x.rst
@@ -0,0 +1,83 @@
+=================
+ALSA BT87x Driver
+=================
+
+Intro
+=====
+
+You might have noticed that the bt878 grabber cards have actually
+*two* PCI functions:
+::
+
+  $ lspci
+  [ ... ]
+  00:0a.0 Multimedia video controller: Brooktree Corporation Bt878 (rev 02)
+  00:0a.1 Multimedia controller: Brooktree Corporation Bt878 (rev 02)
+  [ ... ]
+
+The first does video, it is backward compatible to the bt848.  The second
+does audio.  snd-bt87x is a driver for the second function.  It's a sound
+driver which can be used for recording sound (and *only* recording, no
+playback).  As most TV cards come with a short cable which can be plugged
+into your sound card's line-in you probably don't need this driver if all
+you want to do is just watching TV...
+
+Some cards do not bother to connect anything to the audio input pins of
+the chip, and some other cards use the audio function to transport MPEG
+video data, so it's quite possible that audio recording may not work
+with your card.
+
+
+Driver Status
+=============
+
+The driver is now stable.  However, it doesn't know about many TV cards,
+and it refuses to load for cards it doesn't know.
+
+If the driver complains ("Unknown TV card found, the audio driver will
+not load"), you can specify the ``load_all=1`` option to force the driver to
+try to use the audio capture function of your card.  If the frequency of
+recorded data is not right, try to specify the ``digital_rate`` option with
+other values than the default 32000 (often it's 44100 or 64000).
+
+If you have an unknown card, please mail the ID and board name to
+, regardless of whether audio capture works
+or not, so that future versions of this driver know about your card.
+
+
+Audio modes
+===========
+
+The chip knows two different modes (digital/analog).  snd-bt87x
+registers two PCM devices, one for each mode.  They cannot be used at
+the same time.
+
+
+Digital audio mode
+==================
+
+The first device (hw:X,0) gives you 16 bit stereo sound.  The sample
+rate depends on the external source which feeds the Bt87x with digital
+sound via I2S interface.
+
+
+Analog audio mode (A/D)
+=======================
+
+The second device (hw:X,1) gives you 8 or 16 bit mono sound.  Supported
+sample rates are between 119466 and 448000 Hz (yes, these numbers are
+that high).  If you've set the CONFIG_SND_BT87X_OVERCLOCK option, the
+maximum sample rate is 1792000 Hz, but audio data becomes unusable
+beyond 896000 Hz on my card.
+
+The chip has three analog inputs.  Consequently you'll get a mixer
+device to control these.
+
+
+Have fun,
+
+  Clemens
+
+
+Written by Clemens Ladisch 
+big parts copied from btaudio.txt by Gerd Knorr 
diff --git a/Documentation/sound/cards/index.rst b/Documentation/sound/cards/index.rst
index 0006bc547bb8..157372faae06 100644
--- a/Documentation/sound/cards/index.rst
+++ b/Documentation/sound/cards/index.rst
@@ -12,3 +12,4 @@ Card-Specific Information
    via82xx-mixer
    audiophile-usb
    mixart
+   bt87x
-- 
cgit v1.2.3-58-ga151


From a02f5895eee72fe365cdcbd26699a5b5091c9f1e Mon Sep 17 00:00:00 2001
From: Takashi Iwai 
Date: Thu, 10 Nov 2016 17:07:49 +0100
Subject: ALSA: doc: ReSTize README.maya44

Another	simple conversion from a plain text file.
Put to cards directory.

Signed-off-by: Takashi Iwai 
---
 Documentation/sound/alsa/README.maya44 | 163 -----------------------------
 Documentation/sound/cards/index.rst    |   1 +
 Documentation/sound/cards/maya44.rst   | 186 +++++++++++++++++++++++++++++++++
 3 files changed, 187 insertions(+), 163 deletions(-)
 delete mode 100644 Documentation/sound/alsa/README.maya44
 create mode 100644 Documentation/sound/cards/maya44.rst

(limited to 'Documentation/sound')

diff --git a/Documentation/sound/alsa/README.maya44 b/Documentation/sound/alsa/README.maya44
deleted file mode 100644
index 67b2ea1cc31d..000000000000
--- a/Documentation/sound/alsa/README.maya44
+++ /dev/null
@@ -1,163 +0,0 @@
-NOTE: The following is the original document of Rainer's patch that the
-current maya44 code based on.  Some contents might be obsoleted, but I
-keep here as reference -- tiwai
-
-----------------------------------------------------------------
- 
-STATE OF DEVELOPMENT:
-
-This driver is being developed on the initiative of Piotr Makowski (oponek@gmail.com) and financed by Lars Bergmann.
-Development is carried out by Rainer Zimmermann (mail@lightshed.de).
-
-ESI provided a sample Maya44 card for the development work.
-
-However, unfortunately it has turned out difficult to get detailed programming information, so I (Rainer Zimmermann) had to find out some card-specific information by experiment and conjecture. Some information (in particular, several GPIO bits) is still missing.
-
-This is the first testing version of the Maya44 driver released to the alsa-devel mailing list (Feb 5, 2008).
-
-
-The following functions work, as tested by Rainer Zimmermann and Piotr Makowski:
-
-- playback and capture at all sampling rates
-- input/output level
-- crossmixing
-- line/mic switch
-- phantom power switch
-- analogue monitor a.k.a bypass
-
-
-The following functions *should* work, but are not fully tested:
-
-- Channel 3+4 analogue - S/PDIF input switching
-- S/PDIF output
-- all inputs/outputs on the M/IO/DIO extension card
-- internal/external clock selection
-
-
-*In particular, we would appreciate testing of these functions by anyone who has access to an M/IO/DIO extension card.*
-
-
-Things that do not seem to work:
-
-- The level meters ("multi track") in 'alsamixer' do not seem to react to signals in (if this is a bug, it would probably be in the existing ICE1724 code).
-
-- Ardour 2.1 seems to work only via JACK, not using ALSA directly or via OSS. This still needs to be tracked down.
-
-
-DRIVER DETAILS:
-
-the following files were added:
-
-pci/ice1724/maya44.c        - Maya44 specific code
-pci/ice1724/maya44.h
-pci/ice1724/ice1724.patch
-pci/ice1724/ice1724.h.patch - PROPOSED patch to ice1724.h (see SAMPLING RATES)
-i2c/other/wm8776.c  - low-level access routines for Wolfson WM8776 codecs 
-include/wm8776.h
-
-
-Note that the wm8776.c code is meant to be card-independent and does not actually register the codec with the ALSA infrastructure.
-This is done in maya44.c, mainly because some of the WM8776 controls are used in Maya44-specific ways, and should be named appropriately.
-
-
-the following files were created in pci/ice1724, simply #including the corresponding file from the alsa-kernel tree:
-
-wtm.h
-vt1720_mobo.h
-revo.h
-prodigy192.h
-pontis.h
-phase.h
-maya44.h
-juli.h
-aureon.h
-amp.h
-envy24ht.h
-se.h
-prodigy_hifi.h
-
-
-*I hope this is the correct way to do things.*
-
-
-SAMPLING RATES:
-
-The Maya44 card (or more exactly, the Wolfson WM8776 codecs) allow a maximum sampling rate of 192 kHz for playback and 92 kHz for capture.
-
-As the ICE1724 chip only allows one global sampling rate, this is handled as follows:
-
-* setting the sampling rate on any open PCM device on the maya44 card will always set the *global* sampling rate for all playback and capture channels.
-
-* In the current state of the driver, setting rates of up to 192 kHz is permitted even for capture devices.
-
-*AVOID CAPTURING AT RATES ABOVE 96kHz*, even though it may appear to work. The codec cannot actually capture at such rates, meaning poor quality.
-
-
-I propose some additional code for limiting the sampling rate when setting on a capture pcm device. However because of the global sampling rate, this logic would be somewhat problematic.
-
-The proposed code (currently deactivated) is in ice1712.h.patch, ice1724.c and maya44.c (in pci/ice1712).
-
-
-SOUND DEVICES:
-
-PCM devices correspond to inputs/outputs as follows (assuming Maya44 is card #0):
-
-hw:0,0 input - stereo, analog input 1+2
-hw:0,0 output - stereo, analog output 1+2
-hw:0,1 input - stereo, analog input 3+4 OR S/PDIF input
-hw:0,1 output - stereo, analog output 3+4 (and SPDIF out)
-
-
-NAMING OF MIXER CONTROLS:
-
-(for more information about the signal flow, please refer to the block diagram on p.24 of the ESI Maya44 manual, or in the ESI windows software).
-
-
-PCM: (digital) output level for channel 1+2
-PCM 1: same for channel 3+4
-
-Mic Phantom+48V: switch for +48V phantom power for electrostatic microphones on input 1/2.
-    Make sure this is not turned on while any other source is connected to input 1/2.
-    It might damage the source and/or the maya44 card.
-
-Mic/Line input: if switch is on, input jack 1/2 is microphone input (mono), otherwise line input (stereo).
-
-Bypass: analogue bypass from ADC input to output for channel 1+2. Same as "Monitor" in the windows driver.
-Bypass 1: same for channel 3+4.
-
-Crossmix: cross-mixer from channels 1+2 to channels 3+4
-Crossmix 1: cross-mixer from channels 3+4 to channels 1+2
-
-IEC958 Output: switch for S/PDIF output.
-    This is not supported by the ESI windows driver.
-    S/PDIF should output the same signal as channel 3+4. [untested!]
-
-
-Digitial output selectors:
-
-    These switches allow a direct digital routing from the ADCs to the DACs.
-    Each switch determines where the digital input data to one of the DACs comes from.
-    They are not supported by the ESI windows driver.
-    For normal operation, they should all be set to "PCM out".
-
-H/W: Output source channel 1
-H/W 1: Output source channel 2
-H/W 2: Output source channel 3
-H/W 3: Output source channel 4
-
-H/W 4 ... H/W 9: unknown function, left in to enable testing.
-    Possibly some of these control S/PDIF output(s).
-    If these turn out to be unused, they will go away in later driver versions.
-
-Selectable values for each of the digital output selectors are:
-   "PCM out" -> DAC output of the corresponding channel (default setting)
-   "Input 1"...
-   "Input 4" -> direct routing from ADC output of the selected input channel
-
-
---------
-
-Feb 14, 2008
-Rainer Zimmermann
-mail@lightshed.de
-
diff --git a/Documentation/sound/cards/index.rst b/Documentation/sound/cards/index.rst
index 157372faae06..6c8e4141df9a 100644
--- a/Documentation/sound/cards/index.rst
+++ b/Documentation/sound/cards/index.rst
@@ -13,3 +13,4 @@ Card-Specific Information
    audiophile-usb
    mixart
    bt87x
+   maya44
diff --git a/Documentation/sound/cards/maya44.rst b/Documentation/sound/cards/maya44.rst
new file mode 100644
index 000000000000..bf09a584b443
--- /dev/null
+++ b/Documentation/sound/cards/maya44.rst
@@ -0,0 +1,186 @@
+=================================
+Notes on Maya44 USB Audio Support
+=================================
+
+.. note::
+   The following is the original document of Rainer's patch that the
+   current maya44 code based on.  Some contents might be obsoleted, but I
+   keep here as reference -- tiwai
+
+Feb 14, 2008
+
+Rainer Zimmermann 
+ 
+STATE OF DEVELOPMENT
+====================
+
+This driver is being developed on the initiative of Piotr Makowski (oponek@gmail.com) and financed by Lars Bergmann.
+Development is carried out by Rainer Zimmermann (mail@lightshed.de).
+
+ESI provided a sample Maya44 card for the development work.
+
+However, unfortunately it has turned out difficult to get detailed programming information, so I (Rainer Zimmermann) had to find out some card-specific information by experiment and conjecture. Some information (in particular, several GPIO bits) is still missing.
+
+This is the first testing version of the Maya44 driver released to the alsa-devel mailing list (Feb 5, 2008).
+
+
+The following functions work, as tested by Rainer Zimmermann and Piotr Makowski:
+
+- playback and capture at all sampling rates
+- input/output level
+- crossmixing
+- line/mic switch
+- phantom power switch
+- analogue monitor a.k.a bypass
+
+
+The following functions *should* work, but are not fully tested:
+
+- Channel 3+4 analogue - S/PDIF input switching
+- S/PDIF output
+- all inputs/outputs on the M/IO/DIO extension card
+- internal/external clock selection
+
+
+*In particular, we would appreciate testing of these functions by anyone who has access to an M/IO/DIO extension card.*
+
+
+Things that do not seem to work:
+
+- The level meters ("multi track") in 'alsamixer' do not seem to react to signals in (if this is a bug, it would probably be in the existing ICE1724 code).
+
+- Ardour 2.1 seems to work only via JACK, not using ALSA directly or via OSS. This still needs to be tracked down.
+
+
+DRIVER DETAILS
+==============
+
+the following files were added:
+
+* pci/ice1724/maya44.c - Maya44 specific code
+* pci/ice1724/maya44.h
+* pci/ice1724/ice1724.patch
+* pci/ice1724/ice1724.h.patch - PROPOSED patch to ice1724.h (see SAMPLING RATES)
+* i2c/other/wm8776.c - low-level access routines for Wolfson WM8776 codecs 
+* include/wm8776.h
+
+
+Note that the wm8776.c code is meant to be card-independent and does not actually register the codec with the ALSA infrastructure.
+This is done in maya44.c, mainly because some of the WM8776 controls are used in Maya44-specific ways, and should be named appropriately.
+
+
+the following files were created in pci/ice1724, simply #including the corresponding file from the alsa-kernel tree:
+
+* wtm.h
+* vt1720_mobo.h
+* revo.h
+* prodigy192.h
+* pontis.h
+* phase.h
+* maya44.h
+* juli.h
+* aureon.h
+* amp.h
+* envy24ht.h
+* se.h
+* prodigy_hifi.h
+
+
+*I hope this is the correct way to do things.*
+
+
+SAMPLING RATES
+==============
+
+The Maya44 card (or more exactly, the Wolfson WM8776 codecs) allow a maximum sampling rate of 192 kHz for playback and 92 kHz for capture.
+
+As the ICE1724 chip only allows one global sampling rate, this is handled as follows:
+
+* setting the sampling rate on any open PCM device on the maya44 card will always set the *global* sampling rate for all playback and capture channels.
+
+* In the current state of the driver, setting rates of up to 192 kHz is permitted even for capture devices.
+
+*AVOID CAPTURING AT RATES ABOVE 96kHz*, even though it may appear to work. The codec cannot actually capture at such rates, meaning poor quality.
+
+
+I propose some additional code for limiting the sampling rate when setting on a capture pcm device. However because of the global sampling rate, this logic would be somewhat problematic.
+
+The proposed code (currently deactivated) is in ice1712.h.patch, ice1724.c and maya44.c (in pci/ice1712).
+
+
+SOUND DEVICES
+=============
+
+PCM devices correspond to inputs/outputs as follows (assuming Maya44 is card #0):
+
+* hw:0,0 input - stereo, analog input 1+2
+* hw:0,0 output - stereo, analog output 1+2
+* hw:0,1 input - stereo, analog input 3+4 OR S/PDIF input
+* hw:0,1 output - stereo, analog output 3+4 (and SPDIF out)
+
+
+NAMING OF MIXER CONTROLS
+========================
+
+(for more information about the signal flow, please refer to the block diagram on p.24 of the ESI Maya44 manual, or in the ESI windows software).
+
+
+PCM
+    (digital) output level for channel 1+2
+PCM 1
+    same for channel 3+4
+
+Mic Phantom+48V
+    switch for +48V phantom power for electrostatic microphones on input 1/2.
+
+    Make sure this is not turned on while any other source is connected to input 1/2.
+    It might damage the source and/or the maya44 card.
+
+Mic/Line input
+    if switch is on, input jack 1/2 is microphone input (mono), otherwise line input (stereo).
+
+Bypass
+    analogue bypass from ADC input to output for channel 1+2. Same as "Monitor" in the windows driver.
+Bypass 1
+    same for channel 3+4.
+
+Crossmix
+    cross-mixer from channels 1+2 to channels 3+4
+Crossmix 1
+    cross-mixer from channels 3+4 to channels 1+2
+
+IEC958 Output
+    switch for S/PDIF output.
+
+    This is not supported by the ESI windows driver.
+    S/PDIF should output the same signal as channel 3+4. [untested!]
+
+
+Digitial output selectors
+    These switches allow a direct digital routing from the ADCs to the DACs.
+    Each switch determines where the digital input data to one of the DACs comes from.
+    They are not supported by the ESI windows driver.
+    For normal operation, they should all be set to "PCM out".
+
+H/W
+    Output source channel 1
+H/W 1
+    Output source channel 2
+H/W 2
+    Output source channel 3
+H/W 3
+    Output source channel 4
+
+H/W 4 ... H/W 9
+    unknown function, left in to enable testing.
+
+    Possibly some of these control S/PDIF output(s).
+    If these turn out to be unused, they will go away in later driver versions.
+
+Selectable values for each of the digital output selectors are:
+
+PCM out
+	DAC output of the corresponding channel (default setting)
+Input 1 ... Input 4
+	direct routing from ADC output of the selected input channel
+
-- 
cgit v1.2.3-58-ga151


From c79b5bb0a36f6290d38d9a4dc126cff5c7f44f2a Mon Sep 17 00:00:00 2001
From: Takashi Iwai 
Date: Thu, 10 Nov 2016 17:20:32 +0100
Subject: ALSA: doc: ReSTize hdspm.txt

A simple conversion from a plain text file.  Quite a few reformatting
in the end due to the style of the original document.

Put to cards directory.

Signed-off-by: Takashi Iwai 
---
 Documentation/sound/alsa/hdspm.txt  | 362 ----------------------------------
 Documentation/sound/cards/hdspm.rst | 379 ++++++++++++++++++++++++++++++++++++
 Documentation/sound/cards/index.rst |   1 +
 3 files changed, 380 insertions(+), 362 deletions(-)
 delete mode 100644 Documentation/sound/alsa/hdspm.txt
 create mode 100644 Documentation/sound/cards/hdspm.rst

(limited to 'Documentation/sound')

diff --git a/Documentation/sound/alsa/hdspm.txt b/Documentation/sound/alsa/hdspm.txt
deleted file mode 100644
index 7ba31948dea7..000000000000
--- a/Documentation/sound/alsa/hdspm.txt
+++ /dev/null
@@ -1,362 +0,0 @@
-Software Interface ALSA-DSP MADI Driver 
-
-(translated from German, so no good English ;-), 
-2004 - winfried ritsch
-
-
-
- Full functionality has been added to the driver. Since some of
- the Controls and startup-options  are ALSA-Standard and only the
- special Controls are described and discussed below.
-
-
- hardware functionality:
-
-   
-   Audio transmission:
-
-     number of channels --  depends on transmission mode
-
-		The number of channels chosen is from 1..Nmax. The reason to
-		use for a lower number of channels is only resource allocation,
-		since unused DMA channels are disabled and less memory is
-		allocated. So also the throughput of the PCI system can be
-		scaled. (Only important for low performance boards).
-
-       Single Speed -- 1..64 channels 
-
-		 (Note: Choosing the 56channel mode for transmission or as
-		 receiver, only 56 are transmitted/received over the MADI, but
-		 all 64 channels are available for the mixer, so channel count
-		 for the driver)
-
-       Double Speed -- 1..32 channels
-
-		 Note: Choosing the 56-channel mode for
-		 transmission/receive-mode , only 28 are transmitted/received
-		 over the MADI, but all 32 channels are available for the mixer,
-		 so channel count for the driver
-
-
-       Quad Speed -- 1..16 channels 
-
-		 Note: Choosing the 56-channel mode for
-		 transmission/receive-mode , only 14 are transmitted/received
-		 over the MADI, but all 16 channels are available for the mixer,
-		 so channel count for the driver
-
-     Format -- signed 32 Bit Little Endian (SNDRV_PCM_FMTBIT_S32_LE)
-
-     Sample Rates --
-
-       Single Speed -- 32000, 44100, 48000
-
-       Double Speed -- 64000, 88200, 96000 (untested)
-
-       Quad Speed -- 128000, 176400, 192000 (untested)
-
-     access-mode -- MMAP (memory mapped), Not interleaved
-     (PCM_NON-INTERLEAVED)
-
-     buffer-sizes -- 64,128,256,512,1024,2048,8192 Samples
-
-     fragments -- 2
-
-     Hardware-pointer -- 2 Modi
-
-
-		 The Card supports the readout of the actual Buffer-pointer,
-		 where DMA reads/writes. Since of the bulk mode of PCI it is only
-		 64 Byte accurate. SO it is not really usable for the
-		 ALSA-mid-level functions (here the buffer-ID gives a better
-		 result), but if MMAP is used by the application. Therefore it
-		 can be configured at load-time with the parameter
-		 precise-pointer.
-
-
-		 (Hint: Experimenting I found that the pointer is maximum 64 to
-		 large never to small. So if you subtract 64 you always have a
-		 safe pointer for writing, which is used on this mode inside
-		 ALSA. In theory now you can get now a latency as low as 16
-		 Samples, which is a quarter of the interrupt possibilities.)
-
-       Precise Pointer -- off
-					interrupt used for pointer-calculation
-
-       Precise Pointer -- on
-					hardware pointer used.
-
-   Controller:
-
-
-	  Since DSP-MADI-Mixer has 8152 Fader, it does not make sense to
-	  use the standard mixer-controls, since this would break most of
-	  (especially graphic) ALSA-Mixer GUIs. So Mixer control has be
-	  provided by a 2-dimensional controller using the
-	  hwdep-interface. 
-
-     Also all 128+256 Peak and RMS-Meter can be accessed via the
-     hwdep-interface. Since it could be a performance problem always
-     copying and converting Peak and RMS-Levels even if you just need
-     one, I decided to export the hardware structure, so that of
-     needed some driver-guru can implement a memory-mapping of mixer
-     or peak-meters over ioctl, or also to do only copying and no
-     conversion. A test-application shows the usage of the controller.
-
-    Latency Controls --- not implemented !!!
-
-
-	   Note: Within the windows-driver the latency is accessible of a
-	   control-panel, but buffer-sizes are controlled with ALSA from
-	   hwparams-calls and should not be changed in run-state, I did not
-	   implement it here.
-
-
-    System Clock -- suspended !!!!
-
-        Name -- "System Clock Mode"
-
-        Access -- Read Write
-
-        Values -- "Master" "Slave"
-
-
-		  !!!! This is a hardware-function but is in conflict with the
-		  Clock-source controller, which is a kind of ALSA-standard. I
-		  makes sense to set the card to a special mode (master at some
-		  frequency or slave), since even not using an Audio-application
-		  a studio should have working synchronisations setup. So use
-		  Clock-source-controller instead !!!!
-
-    Clock Source  
-
-       Name -- "Sample Clock Source"
-
-       Access -- Read Write
-
-       Values -- "AutoSync", "Internal 32.0 kHz", "Internal 44.1 kHz",
-       "Internal 48.0 kHz", "Internal 64.0 kHz", "Internal 88.2 kHz",
-       "Internal 96.0 kHz"
-
-		 Choose between Master at a specific Frequency and so also the
-		 Speed-mode or Slave (Autosync). Also see  "Preferred Sync Ref"
-
-
-       !!!! This is no pure hardware function but was implemented by
-       ALSA by some ALSA-drivers before, so I use it also. !!!
-
-
-    Preferred Sync Ref
-
-       Name -- "Preferred Sync Reference"
-
-       Access -- Read Write
-
-       Values -- "Word" "MADI"
-
-
-		 Within the Auto-sync-Mode the preferred Sync Source can be
-		 chosen. If it is not available another is used if possible.
-
-		 Note: Since MADI has a much higher bit-rate than word-clock, the
-		 card should synchronise better in MADI Mode. But since the
-		 RME-PLL is very good, there are almost no problems with
-		 word-clock too. I never found a difference.
-
-
-    TX 64 channel --- 
-
-       Name -- "TX 64 channels mode"
-
-       Access -- Read Write
-
-       Values -- 0 1
-
-		 Using 64-channel-modus (1) or 56-channel-modus for
-		 MADI-transmission (0).
-
-
-		 Note: This control is for output only. Input-mode is detected
-		 automatically from hardware sending MADI.
-
-
-    Clear TMS ---
-
-       Name -- "Clear Track Marker"
-
-       Access -- Read Write
-
-       Values -- 0 1
-
-
-		 Don't use to lower 5 Audio-bits on AES as additional Bits.
-        
-
-    Safe Mode oder Auto Input --- 
-
-       Name -- "Safe Mode"
-
-       Access -- Read Write
-
-       Values -- 0 1
-
-       (default on)
-
-		 If on (1), then if either the optical or coaxial connection
-		 has a failure, there is a takeover to the working one, with no
-		 sample failure. Its only useful if you use the second as a
-		 backup connection.
-
-    Input --- 
-
-       Name -- "Input Select"
-
-       Access -- Read Write
-
-       Values -- optical coaxial
-
-
-		 Choosing the Input, optical or coaxial. If Safe-mode is active,
-		 this is the preferred Input.
-
--------------- Mixer ----------------------
-
-    Mixer
-
-       Name -- "Mixer"
-
-       Access -- Read Write
-
-       Values -  
-
-
-		 Here as a first value the channel-index is taken to get/set the
-		 corresponding mixer channel, where 0-63 are the input to output
-		 fader and 64-127 the playback to outputs fader. Value 0
-		 is channel muted 0 and 32768 an amplification of  1.
-
-    Chn 1-64
-
-       fast mixer for the ALSA-mixer utils. The diagonal of the
-       mixer-matrix is implemented from playback to output.
-       
-
-    Line Out
-
-       Name  -- "Line Out"
-
-       Access -- Read Write
-
-       Values -- 0 1
-
-		 Switching on and off the analog out, which has nothing to do
-		 with mixing or routing. the analog outs reflects channel 63,64.
-
-
---- information (only read access):
- 
-    Sample Rate
-
-       Name -- "System Sample Rate"
-
-       Access -- Read-only
-
-		 getting the sample rate.
-
-
-    External Rate measured
-
-       Name -- "External Rate"
-
-       Access -- Read only
-
-
-		 Should be "Autosync Rate", but Name used is
-		 ALSA-Scheme. External Sample frequency liked used on Autosync is
-		 reported.
-
-
-    MADI Sync Status
-
-       Name -- "MADI Sync Lock Status"
-
-       Access -- Read
-
-       Values -- 0,1,2
-
-       MADI-Input is 0=Unlocked, 1=Locked, or 2=Synced.
-
-
-    Word Clock Sync Status
-
-       Name -- "Word Clock Lock Status"
-
-       Access -- Read
-
-       Values -- 0,1,2
-
-       Word Clock Input is 0=Unlocked, 1=Locked, or 2=Synced.
-
-    AutoSync
-
-       Name -- "AutoSync Reference"
-
-       Access -- Read
-
-       Values -- "WordClock", "MADI", "None"
-
-		 Sync-Reference is either "WordClock", "MADI" or none.
-
-   RX 64ch --- noch nicht implementiert
-
-       MADI-Receiver is in 64 channel mode oder 56 channel mode.
-
-
-   AB_inp   --- not tested 
-
-		 Used input for Auto-Input.
-
-
-   actual Buffer Position --- not implemented
-
-	   !!! this is a ALSA internal function, so no control is used !!!
-
-
-
-Calling Parameter:
-
-   index int array (min = 1, max = 8), 
-     "Index value for RME HDSPM interface." card-index within ALSA
-
-     note: ALSA-standard
-
-   id string array (min = 1, max = 8), 
-     "ID string for RME HDSPM interface."
-
-     note: ALSA-standard
-
-   enable int array (min = 1, max = 8), 
-     "Enable/disable specific HDSPM sound-cards."
-
-     note: ALSA-standard
-
-   precise_ptr int array (min = 1, max = 8), 
-     "Enable precise pointer, or disable."
-
-     note: Use only when the application supports this (which is a special case).
-
-   line_outs_monitor int array (min = 1, max = 8), 
-     "Send playback streams to analog outs by default."
-
-
-	  note: each playback channel is mixed to the same numbered output
-	  channel (routed). This is against the ALSA-convention, where all
-	  channels have to be muted on after loading the driver, but was
-	  used before on other cards, so i historically use it again)
-
-
-
-   enable_monitor int array (min = 1, max = 8), 
-     "Enable Analog Out on Channel 63/64 by default."
-
-      note: here the analog output is enabled (but not routed).
diff --git a/Documentation/sound/cards/hdspm.rst b/Documentation/sound/cards/hdspm.rst
new file mode 100644
index 000000000000..5373e51ed076
--- /dev/null
+++ b/Documentation/sound/cards/hdspm.rst
@@ -0,0 +1,379 @@
+=======================================
+Software Interface ALSA-DSP MADI Driver 
+=======================================
+
+(translated from German, so no good English ;-), 
+
+2004 - winfried ritsch
+
+
+Full functionality has been added to the driver. Since some of
+the Controls and startup-options  are ALSA-Standard and only the
+special Controls are described and discussed below.
+
+
+Hardware functionality
+======================
+   
+Audio transmission
+------------------
+
+* number of channels --  depends on transmission mode
+
+		The number of channels chosen is from 1..Nmax. The reason to
+		use for a lower number of channels is only resource allocation,
+		since unused DMA channels are disabled and less memory is
+		allocated. So also the throughput of the PCI system can be
+		scaled. (Only important for low performance boards).
+
+* Single Speed -- 1..64 channels 
+
+.. note::
+		 (Note: Choosing the 56channel mode for transmission or as
+		 receiver, only 56 are transmitted/received over the MADI, but
+		 all 64 channels are available for the mixer, so channel count
+		 for the driver)
+
+* Double Speed -- 1..32 channels
+
+.. note::
+		 Note: Choosing the 56-channel mode for
+		 transmission/receive-mode , only 28 are transmitted/received
+		 over the MADI, but all 32 channels are available for the mixer,
+		 so channel count for the driver
+
+
+* Quad Speed -- 1..16 channels 
+
+.. note::
+		 Choosing the 56-channel mode for
+		 transmission/receive-mode , only 14 are transmitted/received
+		 over the MADI, but all 16 channels are available for the mixer,
+		 so channel count for the driver
+
+* Format -- signed 32 Bit Little Endian (SNDRV_PCM_FMTBIT_S32_LE)
+
+* Sample Rates --
+
+       Single Speed -- 32000, 44100, 48000
+
+       Double Speed -- 64000, 88200, 96000 (untested)
+
+       Quad Speed -- 128000, 176400, 192000 (untested)
+
+* access-mode -- MMAP (memory mapped), Not interleaved (PCM_NON-INTERLEAVED)
+
+* buffer-sizes -- 64,128,256,512,1024,2048,8192 Samples
+
+* fragments -- 2
+
+* Hardware-pointer -- 2 Modi
+
+
+		 The Card supports the readout of the actual Buffer-pointer,
+		 where DMA reads/writes. Since of the bulk mode of PCI it is only
+		 64 Byte accurate. SO it is not really usable for the
+		 ALSA-mid-level functions (here the buffer-ID gives a better
+		 result), but if MMAP is used by the application. Therefore it
+		 can be configured at load-time with the parameter
+		 precise-pointer.
+
+
+.. hint::
+		 (Hint: Experimenting I found that the pointer is maximum 64 to
+		 large never to small. So if you subtract 64 you always have a
+		 safe pointer for writing, which is used on this mode inside
+		 ALSA. In theory now you can get now a latency as low as 16
+		 Samples, which is a quarter of the interrupt possibilities.)
+
+   * Precise Pointer -- off
+					interrupt used for pointer-calculation
+				
+   * Precise Pointer -- on
+					hardware pointer used.
+
+Controller
+----------
+
+Since DSP-MADI-Mixer has 8152 Fader, it does not make sense to
+use the standard mixer-controls, since this would break most of
+(especially graphic) ALSA-Mixer GUIs. So Mixer control has be
+provided by a 2-dimensional controller using the
+hwdep-interface. 
+
+Also all 128+256 Peak and RMS-Meter can be accessed via the
+hwdep-interface. Since it could be a performance problem always
+copying and converting Peak and RMS-Levels even if you just need
+one, I decided to export the hardware structure, so that of
+needed some driver-guru can implement a memory-mapping of mixer
+or peak-meters over ioctl, or also to do only copying and no
+conversion. A test-application shows the usage of the controller.
+
+* Latency Controls --- not implemented !!!
+
+.. note::
+	   Note: Within the windows-driver the latency is accessible of a
+	   control-panel, but buffer-sizes are controlled with ALSA from
+	   hwparams-calls and should not be changed in run-state, I did not
+	   implement it here.
+
+
+* System Clock -- suspended !!!!
+
+  * Name -- "System Clock Mode"
+
+  * Access -- Read Write
+    
+  * Values -- "Master" "Slave"
+
+.. note::
+		  !!!! This is a hardware-function but is in conflict with the
+		  Clock-source controller, which is a kind of ALSA-standard. I
+		  makes sense to set the card to a special mode (master at some
+		  frequency or slave), since even not using an Audio-application
+		  a studio should have working synchronisations setup. So use
+		  Clock-source-controller instead !!!!
+
+* Clock Source  
+
+  * Name -- "Sample Clock Source"
+
+  * Access -- Read Write
+
+  * Values -- "AutoSync", "Internal 32.0 kHz", "Internal 44.1 kHz",
+    "Internal 48.0 kHz", "Internal 64.0 kHz", "Internal 88.2 kHz",
+    "Internal 96.0 kHz"
+
+		 Choose between Master at a specific Frequency and so also the
+		 Speed-mode or Slave (Autosync). Also see  "Preferred Sync Ref"
+
+.. warning::
+       !!!! This is no pure hardware function but was implemented by
+       ALSA by some ALSA-drivers before, so I use it also. !!!
+
+
+* Preferred Sync Ref
+
+  * Name -- "Preferred Sync Reference"
+
+  * Access -- Read Write
+
+  * Values -- "Word" "MADI"
+
+
+		 Within the Auto-sync-Mode the preferred Sync Source can be
+		 chosen. If it is not available another is used if possible.
+
+.. note::
+		 Note: Since MADI has a much higher bit-rate than word-clock, the
+		 card should synchronise better in MADI Mode. But since the
+		 RME-PLL is very good, there are almost no problems with
+		 word-clock too. I never found a difference.
+
+
+* TX 64 channel
+
+  * Name -- "TX 64 channels mode"
+
+  * Access -- Read Write
+
+  * Values -- 0 1
+
+		 Using 64-channel-modus (1) or 56-channel-modus for
+		 MADI-transmission (0).
+
+
+.. note::
+		 Note: This control is for output only. Input-mode is detected
+		 automatically from hardware sending MADI.
+
+
+* Clear TMS
+
+  * Name -- "Clear Track Marker"
+
+  * Access -- Read Write
+
+  * Values -- 0 1
+
+
+		 Don't use to lower 5 Audio-bits on AES as additional Bits.
+        
+
+* Safe Mode oder Auto Input
+
+  * Name -- "Safe Mode"
+
+  * Access -- Read Write
+
+  * Values -- 0 1 (default on)
+
+		 If on (1), then if either the optical or coaxial connection
+		 has a failure, there is a takeover to the working one, with no
+		 sample failure. Its only useful if you use the second as a
+		 backup connection.
+
+* Input
+
+  * Name -- "Input Select"
+
+  * Access -- Read Write
+
+  * Values -- optical coaxial
+
+
+		 Choosing the Input, optical or coaxial. If Safe-mode is active,
+		 this is the preferred Input.
+
+Mixer
+-----
+
+* Mixer
+
+  * Name -- "Mixer"
+
+  * Access -- Read Write
+
+  * Values -  
+
+
+		 Here as a first value the channel-index is taken to get/set the
+		 corresponding mixer channel, where 0-63 are the input to output
+		 fader and 64-127 the playback to outputs fader. Value 0
+		 is channel muted 0 and 32768 an amplification of  1.
+
+* Chn 1-64
+
+       fast mixer for the ALSA-mixer utils. The diagonal of the
+       mixer-matrix is implemented from playback to output.
+       
+
+* Line Out
+
+  * Name  -- "Line Out"
+
+  * Access -- Read Write
+
+  * Values -- 0 1
+
+		 Switching on and off the analog out, which has nothing to do
+		 with mixing or routing. the analog outs reflects channel 63,64.
+
+
+Information (only read access)
+------------------------------
+ 
+* Sample Rate
+
+  * Name -- "System Sample Rate"
+
+  * Access -- Read-only
+
+		 getting the sample rate.
+
+
+* External Rate measured
+
+  * Name -- "External Rate"
+
+  * Access -- Read only
+
+
+		 Should be "Autosync Rate", but Name used is
+		 ALSA-Scheme. External Sample frequency liked used on Autosync is
+		 reported.
+
+
+* MADI Sync Status
+
+  * Name -- "MADI Sync Lock Status"
+
+  * Access -- Read
+
+  * Values -- 0,1,2
+
+       MADI-Input is 0=Unlocked, 1=Locked, or 2=Synced.
+
+
+* Word Clock Sync Status
+
+  * Name -- "Word Clock Lock Status"
+
+  * Access -- Read
+
+  * Values -- 0,1,2
+
+       Word Clock Input is 0=Unlocked, 1=Locked, or 2=Synced.
+
+* AutoSync
+
+  * Name -- "AutoSync Reference"
+
+  * Access -- Read
+
+  * Values -- "WordClock", "MADI", "None"
+
+		 Sync-Reference is either "WordClock", "MADI" or none.
+
+* RX 64ch --- noch nicht implementiert
+
+       MADI-Receiver is in 64 channel mode oder 56 channel mode.
+
+
+* AB_inp   --- not tested 
+
+		 Used input for Auto-Input.
+
+
+* actual Buffer Position --- not implemented
+
+	   !!! this is a ALSA internal function, so no control is used !!!
+
+
+
+Calling Parameter
+=================
+
+* index int array (min = 1, max = 8) 
+
+     Index value for RME HDSPM interface. card-index within ALSA
+
+     note: ALSA-standard
+
+* id string array (min = 1, max = 8) 
+
+     ID string for RME HDSPM interface.
+
+     note: ALSA-standard
+
+* enable int array (min = 1, max = 8)
+
+     Enable/disable specific HDSPM sound-cards.
+
+     note: ALSA-standard
+
+* precise_ptr int array (min = 1, max = 8)
+
+     Enable precise pointer, or disable.
+
+.. note::
+     note: Use only when the application supports this (which is a special case).
+
+* line_outs_monitor int array (min = 1, max = 8)
+
+     Send playback streams to analog outs by default.
+
+.. note::
+	  note: each playback channel is mixed to the same numbered output
+	  channel (routed). This is against the ALSA-convention, where all
+	  channels have to be muted on after loading the driver, but was
+	  used before on other cards, so i historically use it again)
+
+
+
+* enable_monitor int array (min = 1, max = 8)
+
+     Enable Analog Out on Channel 63/64 by default.
+
+.. note ::
+      note: here the analog output is enabled (but not routed).
diff --git a/Documentation/sound/cards/index.rst b/Documentation/sound/cards/index.rst
index 6c8e4141df9a..b143eff93c87 100644
--- a/Documentation/sound/cards/index.rst
+++ b/Documentation/sound/cards/index.rst
@@ -14,3 +14,4 @@ Card-Specific Information
    mixart
    bt87x
    maya44
+   hdspm
-- 
cgit v1.2.3-58-ga151


From bb02859cd3448f31e00c47c1277e27be0cee7d2a Mon Sep 17 00:00:00 2001
From: Takashi Iwai 
Date: Thu, 10 Nov 2016 17:33:12 +0100
Subject: ALSA: doc: ReSTize serial-u16550.txt

Yet another simple conversion from a plain text file.
Put to cards directory.

Signed-off-by: Takashi Iwai 
---
 Documentation/sound/alsa/serial-u16550.txt  | 88 ---------------------------
 Documentation/sound/cards/index.rst         |  1 +
 Documentation/sound/cards/serial-u16550.rst | 93 +++++++++++++++++++++++++++++
 3 files changed, 94 insertions(+), 88 deletions(-)
 delete mode 100644 Documentation/sound/alsa/serial-u16550.txt
 create mode 100644 Documentation/sound/cards/serial-u16550.rst

(limited to 'Documentation/sound')

diff --git a/Documentation/sound/alsa/serial-u16550.txt b/Documentation/sound/alsa/serial-u16550.txt
deleted file mode 100644
index c1919559d509..000000000000
--- a/Documentation/sound/alsa/serial-u16550.txt
+++ /dev/null
@@ -1,88 +0,0 @@
-
-			Serial UART 16450/16550 MIDI driver
-			===================================
-
-The adaptor module parameter allows you to select either:
-
-  0 - Roland Soundcanvas support (default)
-  1 - Midiator MS-124T support (1)
-  2 - Midiator MS-124W S/A mode (2)
-  3 - MS-124W M/B mode support (3)
-  4 - Generic device with multiple input support (4)
-
-For the Midiator MS-124W, you must set the physical M-S and A-B
-switches on the Midiator to match the driver mode you select.
-
-In Roland Soundcanvas mode, multiple ALSA raw MIDI substreams are supported
-(midiCnD0-midiCnD15).  Whenever you write to a different substream, the driver
-sends the nonstandard MIDI command sequence F5 NN, where NN is the substream
-number plus 1.  Roland modules use this command to switch between different
-"parts", so this feature lets you treat each part as a distinct raw MIDI
-substream. The driver provides no way to send F5 00 (no selection) or to not
-send the F5 NN command sequence at all; perhaps it ought to.
-
-Usage example for simple serial converter:
-
-	/sbin/setserial /dev/ttyS0 uart none
-	/sbin/modprobe snd-serial-u16550 port=0x3f8 irq=4 speed=115200
-
-Usage example for Roland SoundCanvas with 4 MIDI ports:
-
-	/sbin/setserial /dev/ttyS0 uart none
-	/sbin/modprobe snd-serial-u16550 port=0x3f8 irq=4 outs=4
-
-In MS-124T mode, one raw MIDI substream is supported (midiCnD0); the outs
-module parameter is automatically set to 1. The driver sends the same data to
-all four MIDI Out connectors.  Set the A-B switch and the speed module
-parameter to match (A=19200, B=9600).
-
-Usage example for MS-124T, with A-B switch in A position:
-
-	/sbin/setserial /dev/ttyS0 uart none
-	/sbin/modprobe snd-serial-u16550 port=0x3f8 irq=4 adaptor=1 \
-			speed=19200
-
-In MS-124W S/A mode, one raw MIDI substream is supported (midiCnD0);
-the outs module parameter is automatically set to 1. The driver sends
-the same data to all four MIDI Out connectors at full MIDI speed.
-
-Usage example for S/A mode:
-
-	/sbin/setserial /dev/ttyS0 uart none
-	/sbin/modprobe snd-serial-u16550 port=0x3f8 irq=4 adaptor=2
-
-In MS-124W M/B mode, the driver supports 16 ALSA raw MIDI substreams;
-the outs module parameter is automatically set to 16.  The substream
-number gives a bitmask of which MIDI Out connectors the data should be
-sent to, with midiCnD1 sending to Out 1, midiCnD2 to Out 2, midiCnD4 to
-Out 3, and midiCnD8 to Out 4.  Thus midiCnD15 sends the data to all 4 ports.
-As a special case, midiCnD0 also sends to all ports, since it is not useful
-to send the data to no ports.  M/B mode has extra overhead to select the MIDI
-Out for each byte, so the aggregate data rate across all four MIDI Outs is
-at most one byte every 520 us, as compared with the full MIDI data rate of
-one byte every 320 us per port.
-
-Usage example for M/B mode:
-
-	/sbin/setserial /dev/ttyS0 uart none
-	/sbin/modprobe snd-serial-u16550 port=0x3f8 irq=4 adaptor=3
-
-The MS-124W hardware's M/A mode is currently not supported. This mode allows
-the MIDI Outs to act independently at double the aggregate throughput of M/B,
-but does not allow sending the same byte simultaneously to multiple MIDI Outs. 
-The M/A protocol requires the driver to twiddle the modem control lines under
-timing constraints, so it would be a bit more complicated to implement than
-the other modes.
-
-Midiator models other than MS-124W and MS-124T are currently not supported. 
-Note that the suffix letter is significant; the MS-124 and MS-124B are not
-compatible, nor are the other known models MS-101, MS-101B, MS-103, and MS-114.
-I do have documentation (tim.mann@compaq.com) that partially covers these models,
-but no units to experiment with.  The MS-124W support is tested with a real unit.
-The MS-124T support is untested, but should work.
-
-The Generic driver supports multiple input and output substreams over a single
-serial port.  Similar to Roland Soundcanvas mode, F5 NN is used to select the
-appropriate input or output stream (depending on the data direction).
-Additionally, the CTS signal is used to regulate the data flow.  The number of
-inputs is specified by the ins parameter.
diff --git a/Documentation/sound/cards/index.rst b/Documentation/sound/cards/index.rst
index b143eff93c87..976ef5eb8f89 100644
--- a/Documentation/sound/cards/index.rst
+++ b/Documentation/sound/cards/index.rst
@@ -15,3 +15,4 @@ Card-Specific Information
    bt87x
    maya44
    hdspm
+   serial-u16550
diff --git a/Documentation/sound/cards/serial-u16550.rst b/Documentation/sound/cards/serial-u16550.rst
new file mode 100644
index 000000000000..197aeacea3da
--- /dev/null
+++ b/Documentation/sound/cards/serial-u16550.rst
@@ -0,0 +1,93 @@
+===================================
+Serial UART 16450/16550 MIDI driver
+===================================
+
+The adaptor module parameter allows you to select either:
+
+* 0 - Roland Soundcanvas support (default)
+* 1 - Midiator MS-124T support (1)
+* 2 - Midiator MS-124W S/A mode (2)
+* 3 - MS-124W M/B mode support (3)
+* 4 - Generic device with multiple input support (4)
+
+For the Midiator MS-124W, you must set the physical M-S and A-B
+switches on the Midiator to match the driver mode you select.
+
+In Roland Soundcanvas mode, multiple ALSA raw MIDI substreams are supported
+(midiCnD0-midiCnD15).  Whenever you write to a different substream, the driver
+sends the nonstandard MIDI command sequence F5 NN, where NN is the substream
+number plus 1.  Roland modules use this command to switch between different
+"parts", so this feature lets you treat each part as a distinct raw MIDI
+substream. The driver provides no way to send F5 00 (no selection) or to not
+send the F5 NN command sequence at all; perhaps it ought to.
+
+Usage example for simple serial converter:
+::
+
+	/sbin/setserial /dev/ttyS0 uart none
+	/sbin/modprobe snd-serial-u16550 port=0x3f8 irq=4 speed=115200
+
+Usage example for Roland SoundCanvas with 4 MIDI ports:
+::
+
+	/sbin/setserial /dev/ttyS0 uart none
+	/sbin/modprobe snd-serial-u16550 port=0x3f8 irq=4 outs=4
+
+In MS-124T mode, one raw MIDI substream is supported (midiCnD0); the outs
+module parameter is automatically set to 1. The driver sends the same data to
+all four MIDI Out connectors.  Set the A-B switch and the speed module
+parameter to match (A=19200, B=9600).
+
+Usage example for MS-124T, with A-B switch in A position:
+::
+
+	/sbin/setserial /dev/ttyS0 uart none
+	/sbin/modprobe snd-serial-u16550 port=0x3f8 irq=4 adaptor=1 \
+			speed=19200
+
+In MS-124W S/A mode, one raw MIDI substream is supported (midiCnD0);
+the outs module parameter is automatically set to 1. The driver sends
+the same data to all four MIDI Out connectors at full MIDI speed.
+
+Usage example for S/A mode:
+::
+
+	/sbin/setserial /dev/ttyS0 uart none
+	/sbin/modprobe snd-serial-u16550 port=0x3f8 irq=4 adaptor=2
+
+In MS-124W M/B mode, the driver supports 16 ALSA raw MIDI substreams;
+the outs module parameter is automatically set to 16.  The substream
+number gives a bitmask of which MIDI Out connectors the data should be
+sent to, with midiCnD1 sending to Out 1, midiCnD2 to Out 2, midiCnD4 to
+Out 3, and midiCnD8 to Out 4.  Thus midiCnD15 sends the data to all 4 ports.
+As a special case, midiCnD0 also sends to all ports, since it is not useful
+to send the data to no ports.  M/B mode has extra overhead to select the MIDI
+Out for each byte, so the aggregate data rate across all four MIDI Outs is
+at most one byte every 520 us, as compared with the full MIDI data rate of
+one byte every 320 us per port.
+
+Usage example for M/B mode:
+::
+
+	/sbin/setserial /dev/ttyS0 uart none
+	/sbin/modprobe snd-serial-u16550 port=0x3f8 irq=4 adaptor=3
+
+The MS-124W hardware's M/A mode is currently not supported. This mode allows
+the MIDI Outs to act independently at double the aggregate throughput of M/B,
+but does not allow sending the same byte simultaneously to multiple MIDI Outs. 
+The M/A protocol requires the driver to twiddle the modem control lines under
+timing constraints, so it would be a bit more complicated to implement than
+the other modes.
+
+Midiator models other than MS-124W and MS-124T are currently not supported. 
+Note that the suffix letter is significant; the MS-124 and MS-124B are not
+compatible, nor are the other known models MS-101, MS-101B, MS-103, and MS-114.
+I do have documentation (tim.mann@compaq.com) that partially covers these models,
+but no units to experiment with.  The MS-124W support is tested with a real unit.
+The MS-124T support is untested, but should work.
+
+The Generic driver supports multiple input and output substreams over a single
+serial port.  Similar to Roland Soundcanvas mode, F5 NN is used to select the
+appropriate input or output stream (depending on the data direction).
+Additionally, the CTS signal is used to regulate the data flow.  The number of
+inputs is specified by the ins parameter.
-- 
cgit v1.2.3-58-ga151


From f336c3f072216a16187b22069681d014dcb43db6 Mon Sep 17 00:00:00 2001
From: Takashi Iwai 
Date: Thu, 10 Nov 2016 17:36:09 +0100
Subject: ALSA: doc: ReSTize img,spdif-in.txt

Yet another simple conversion from a plain text file.
Put to cards directory.

Signed-off-by: Takashi Iwai 
---
 Documentation/sound/alsa/img,spdif-in.txt  | 49 ---------------------------
 Documentation/sound/cards/img-spdif-in.rst | 53 ++++++++++++++++++++++++++++++
 Documentation/sound/cards/index.rst        |  1 +
 3 files changed, 54 insertions(+), 49 deletions(-)
 delete mode 100644 Documentation/sound/alsa/img,spdif-in.txt
 create mode 100644 Documentation/sound/cards/img-spdif-in.rst

(limited to 'Documentation/sound')

diff --git a/Documentation/sound/alsa/img,spdif-in.txt b/Documentation/sound/alsa/img,spdif-in.txt
deleted file mode 100644
index 8b7505785fa6..000000000000
--- a/Documentation/sound/alsa/img,spdif-in.txt
+++ /dev/null
@@ -1,49 +0,0 @@
-The Imagination Technologies SPDIF Input controller contains the following
-controls:
-
-name='IEC958 Capture Mask',index=0
-
-This control returns a mask that shows which of the IEC958 status bits
-can be read using the 'IEC958 Capture Default' control.
-
-name='IEC958 Capture Default',index=0
-
-This control returns the status bits contained within the SPDIF stream that
-is being received. The 'IEC958 Capture Mask' shows which bits can be read
-from this control.
-
-name='SPDIF In Multi Frequency Acquire',index=0
-name='SPDIF In Multi Frequency Acquire',index=1
-name='SPDIF In Multi Frequency Acquire',index=2
-name='SPDIF In Multi Frequency Acquire',index=3
-
-This control is used to attempt acquisition of up to four different sample
-rates. The active rate can be obtained by reading the 'SPDIF In Lock Frequency'
-control.
-
-When the value of this control is set to {0,0,0,0}, the rate given to hw_params
-will determine the single rate the block will capture. Else, the rate given to
-hw_params will be ignored, and the block will attempt capture for each of the
-four sample rates set here.
-
-If less than four rates are required, the same rate can be specified more than
-once
-
-name='SPDIF In Lock Frequency',index=0
-
-This control returns the active capture rate, or 0 if a lock has not been
-acquired
-
-name='SPDIF In Lock TRK',index=0
-
-This control is used to modify the locking/jitter rejection characteristics
-of the block. Larger values increase the locking range, but reduce jitter
-rejection.
-
-name='SPDIF In Lock Acquire Threshold',index=0
-
-This control is used to change the threshold at which a lock is acquired.
-
-name='SPDIF In Lock Release Threshold',index=0
-
-This control is used to change the threshold at which a lock is released.
diff --git a/Documentation/sound/cards/img-spdif-in.rst b/Documentation/sound/cards/img-spdif-in.rst
new file mode 100644
index 000000000000..7df9f5ae2609
--- /dev/null
+++ b/Documentation/sound/cards/img-spdif-in.rst
@@ -0,0 +1,53 @@
+================================================
+Imagination Technologies SPDIF Input Controllers
+================================================
+
+The Imagination Technologies SPDIF Input controller contains the following
+controls:
+
+* name='IEC958 Capture Mask',index=0
+
+This control returns a mask that shows which of the IEC958 status bits
+can be read using the 'IEC958 Capture Default' control.
+
+* name='IEC958 Capture Default',index=0
+
+This control returns the status bits contained within the SPDIF stream that
+is being received. The 'IEC958 Capture Mask' shows which bits can be read
+from this control.
+
+* name='SPDIF In Multi Frequency Acquire',index=0
+* name='SPDIF In Multi Frequency Acquire',index=1
+* name='SPDIF In Multi Frequency Acquire',index=2
+* name='SPDIF In Multi Frequency Acquire',index=3
+
+This control is used to attempt acquisition of up to four different sample
+rates. The active rate can be obtained by reading the 'SPDIF In Lock Frequency'
+control.
+
+When the value of this control is set to {0,0,0,0}, the rate given to hw_params
+will determine the single rate the block will capture. Else, the rate given to
+hw_params will be ignored, and the block will attempt capture for each of the
+four sample rates set here.
+
+If less than four rates are required, the same rate can be specified more than
+once
+
+* name='SPDIF In Lock Frequency',index=0
+
+This control returns the active capture rate, or 0 if a lock has not been
+acquired
+
+* name='SPDIF In Lock TRK',index=0
+
+This control is used to modify the locking/jitter rejection characteristics
+of the block. Larger values increase the locking range, but reduce jitter
+rejection.
+
+* name='SPDIF In Lock Acquire Threshold',index=0
+
+This control is used to change the threshold at which a lock is acquired.
+
+* name='SPDIF In Lock Release Threshold',index=0
+
+This control is used to change the threshold at which a lock is released.
diff --git a/Documentation/sound/cards/index.rst b/Documentation/sound/cards/index.rst
index 976ef5eb8f89..c016f8c3b88b 100644
--- a/Documentation/sound/cards/index.rst
+++ b/Documentation/sound/cards/index.rst
@@ -16,3 +16,4 @@ Card-Specific Information
    maya44
    hdspm
    serial-u16550
+   img-spdif-in
-- 
cgit v1.2.3-58-ga151


From 8e5336a14476e7350b0cf78a99541de6ed51655c Mon Sep 17 00:00:00 2001
From: Takashi Iwai 
Date: Thu, 10 Nov 2016 22:11:21 +0100
Subject: ASoC: doc: ReSTize overview.txt

A simple conversion from a plain text file.
Created a new subdirectory, Documentation/sound/soc, for this and
other ASoC documents.

Since the index page contains the TOC, so "Documentation" section got
removed from overview.

Acked-by: Mark Brown 
Signed-off-by: Takashi Iwai 
---
 Documentation/sound/alsa/soc/overview.txt | 95 -------------------------------
 Documentation/sound/index.rst             |  1 +
 Documentation/sound/soc/index.rst         | 10 ++++
 Documentation/sound/soc/overview.rst      | 69 ++++++++++++++++++++++
 4 files changed, 80 insertions(+), 95 deletions(-)
 delete mode 100644 Documentation/sound/alsa/soc/overview.txt
 create mode 100644 Documentation/sound/soc/index.rst
 create mode 100644 Documentation/sound/soc/overview.rst

(limited to 'Documentation/sound')

diff --git a/Documentation/sound/alsa/soc/overview.txt b/Documentation/sound/alsa/soc/overview.txt
deleted file mode 100644
index f3f28b7ae242..000000000000
--- a/Documentation/sound/alsa/soc/overview.txt
+++ /dev/null
@@ -1,95 +0,0 @@
-ALSA SoC Layer
-==============
-
-The overall project goal of the ALSA System on Chip (ASoC) layer is to
-provide better ALSA support for embedded system-on-chip processors (e.g.
-pxa2xx, au1x00, iMX, etc) and portable audio codecs.  Prior to the ASoC
-subsystem there was some support in the kernel for SoC audio, however it
-had some limitations:-
-
-  * Codec drivers were often tightly coupled to the underlying SoC
-    CPU. This is not ideal and leads to code duplication - for example,
-    Linux had different wm8731 drivers for 4 different SoC platforms.
-
-  * There was no standard method to signal user initiated audio events (e.g.
-    Headphone/Mic insertion, Headphone/Mic detection after an insertion
-    event). These are quite common events on portable devices and often require
-    machine specific code to re-route audio, enable amps, etc., after such an
-    event.
-
-  * Drivers tended to power up the entire codec when playing (or
-    recording) audio. This is fine for a PC, but tends to waste a lot of
-    power on portable devices. There was also no support for saving
-    power via changing codec oversampling rates, bias currents, etc.
-
-
-ASoC Design
-===========
-
-The ASoC layer is designed to address these issues and provide the following
-features :-
-
-  * Codec independence. Allows reuse of codec drivers on other platforms
-    and machines.
-
-  * Easy I2S/PCM audio interface setup between codec and SoC. Each SoC
-    interface and codec registers its audio interface capabilities with the
-    core and are subsequently matched and configured when the application
-    hardware parameters are known.
-
-  * Dynamic Audio Power Management (DAPM). DAPM automatically sets the codec to
-    its minimum power state at all times. This includes powering up/down
-    internal power blocks depending on the internal codec audio routing and any
-    active streams.
-
-  * Pop and click reduction. Pops and clicks can be reduced by powering the
-    codec up/down in the correct sequence (including using digital mute). ASoC
-    signals the codec when to change power states.
-
-  * Machine specific controls: Allow machines to add controls to the sound card
-    (e.g. volume control for speaker amplifier).
-
-To achieve all this, ASoC basically splits an embedded audio system into
-multiple re-usable component drivers :-
-
-  * Codec class drivers: The codec class driver is platform independent and
-    contains audio controls, audio interface capabilities, codec DAPM
-    definition and codec IO functions. This class extends to BT, FM and MODEM
-    ICs if required. Codec class drivers should be generic code that can run
-    on any architecture and machine.
-
-  * Platform class drivers: The platform class driver includes the audio DMA
-    engine driver, digital audio interface (DAI) drivers (e.g. I2S, AC97, PCM)
-    and any audio DSP drivers for that platform.
-
-  * Machine class driver: The machine driver class acts as the glue that
-    describes and binds the other component drivers together to form an ALSA
-    "sound card device". It handles any machine specific controls and
-    machine level audio events (e.g. turning on an amp at start of playback).
-
-
-Documentation
-=============
-
-The documentation is spilt into the following sections:-
-
-overview.txt: This file.
-
-codec.txt: Codec driver internals.
-
-DAI.txt: Description of Digital Audio Interface standards and how to configure
-a DAI within your codec and CPU DAI drivers.
-
-dapm.txt: Dynamic Audio Power Management
-
-platform.txt: Platform audio DMA and DAI.
-
-machine.txt: Machine driver internals.
-
-pop_clicks.txt: How to minimise audio artifacts.
-
-clocking.txt: ASoC clocking for best power performance.
-
-jack.txt: ASoC jack detection.
-
-DPCM.txt: Dynamic PCM - Describes DPCM with DSP examples.
diff --git a/Documentation/sound/index.rst b/Documentation/sound/index.rst
index 1f5d166f81c4..47b89f014e69 100644
--- a/Documentation/sound/index.rst
+++ b/Documentation/sound/index.rst
@@ -7,6 +7,7 @@ Linux Sound Subsystem Documentation
 
    kernel-api/index
    designs/index
+   soc/index
    alsa-configuration
    hd-audio/index
    cards/index
diff --git a/Documentation/sound/soc/index.rst b/Documentation/sound/soc/index.rst
new file mode 100644
index 000000000000..e974fd9f38a3
--- /dev/null
+++ b/Documentation/sound/soc/index.rst
@@ -0,0 +1,10 @@
+==============
+ALSA SoC Layer
+==============
+
+The documentation is spilt into the following sections:-
+
+.. toctree::
+   :maxdepth: 2
+
+   overview
diff --git a/Documentation/sound/soc/overview.rst b/Documentation/sound/soc/overview.rst
new file mode 100644
index 000000000000..dc8370bbfff6
--- /dev/null
+++ b/Documentation/sound/soc/overview.rst
@@ -0,0 +1,69 @@
+=======================
+ALSA SoC Layer Overview
+=======================
+
+The overall project goal of the ALSA System on Chip (ASoC) layer is to
+provide better ALSA support for embedded system-on-chip processors (e.g.
+pxa2xx, au1x00, iMX, etc) and portable audio codecs.  Prior to the ASoC
+subsystem there was some support in the kernel for SoC audio, however it
+had some limitations:-
+
+  * Codec drivers were often tightly coupled to the underlying SoC
+    CPU. This is not ideal and leads to code duplication - for example,
+    Linux had different wm8731 drivers for 4 different SoC platforms.
+
+  * There was no standard method to signal user initiated audio events (e.g.
+    Headphone/Mic insertion, Headphone/Mic detection after an insertion
+    event). These are quite common events on portable devices and often require
+    machine specific code to re-route audio, enable amps, etc., after such an
+    event.
+
+  * Drivers tended to power up the entire codec when playing (or
+    recording) audio. This is fine for a PC, but tends to waste a lot of
+    power on portable devices. There was also no support for saving
+    power via changing codec oversampling rates, bias currents, etc.
+
+
+ASoC Design
+===========
+
+The ASoC layer is designed to address these issues and provide the following
+features :-
+
+  * Codec independence. Allows reuse of codec drivers on other platforms
+    and machines.
+
+  * Easy I2S/PCM audio interface setup between codec and SoC. Each SoC
+    interface and codec registers its audio interface capabilities with the
+    core and are subsequently matched and configured when the application
+    hardware parameters are known.
+
+  * Dynamic Audio Power Management (DAPM). DAPM automatically sets the codec to
+    its minimum power state at all times. This includes powering up/down
+    internal power blocks depending on the internal codec audio routing and any
+    active streams.
+
+  * Pop and click reduction. Pops and clicks can be reduced by powering the
+    codec up/down in the correct sequence (including using digital mute). ASoC
+    signals the codec when to change power states.
+
+  * Machine specific controls: Allow machines to add controls to the sound card
+    (e.g. volume control for speaker amplifier).
+
+To achieve all this, ASoC basically splits an embedded audio system into
+multiple re-usable component drivers :-
+
+  * Codec class drivers: The codec class driver is platform independent and
+    contains audio controls, audio interface capabilities, codec DAPM
+    definition and codec IO functions. This class extends to BT, FM and MODEM
+    ICs if required. Codec class drivers should be generic code that can run
+    on any architecture and machine.
+
+  * Platform class drivers: The platform class driver includes the audio DMA
+    engine driver, digital audio interface (DAI) drivers (e.g. I2S, AC97, PCM)
+    and any audio DSP drivers for that platform.
+
+  * Machine class driver: The machine driver class acts as the glue that
+    describes and binds the other component drivers together to form an ALSA
+    "sound card device". It handles any machine specific controls and
+    machine level audio events (e.g. turning on an amp at start of playback).
-- 
cgit v1.2.3-58-ga151


From 693ba474a39a2c22e1576995139c9bfdd8b554c8 Mon Sep 17 00:00:00 2001
From: Takashi Iwai 
Date: Thu, 10 Nov 2016 22:16:04 +0100
Subject: ASoC: doc: ReSTize codec.txt

A simple conversion from a plain text file.
The section numbers are dropped to align with other documents.

Acked-by: Mark Brown 
Signed-off-by: Takashi Iwai 
---
 Documentation/sound/alsa/soc/codec.txt | 179 -------------------------------
 Documentation/sound/soc/codec.rst      | 190 +++++++++++++++++++++++++++++++++
 Documentation/sound/soc/index.rst      |   1 +
 3 files changed, 191 insertions(+), 179 deletions(-)
 delete mode 100644 Documentation/sound/alsa/soc/codec.txt
 create mode 100644 Documentation/sound/soc/codec.rst

(limited to 'Documentation/sound')

diff --git a/Documentation/sound/alsa/soc/codec.txt b/Documentation/sound/alsa/soc/codec.txt
deleted file mode 100644
index db5f9c9ae149..000000000000
--- a/Documentation/sound/alsa/soc/codec.txt
+++ /dev/null
@@ -1,179 +0,0 @@
-ASoC Codec Class Driver
-=======================
-
-The codec class driver is generic and hardware independent code that configures
-the codec, FM, MODEM, BT or external DSP to provide audio capture and playback.
-It should contain no code that is specific to the target platform or machine.
-All platform and machine specific code should be added to the platform and
-machine drivers respectively.
-
-Each codec class driver *must* provide the following features:-
-
- 1) Codec DAI and PCM configuration
- 2) Codec control IO - using RegMap API
- 3) Mixers and audio controls
- 4) Codec audio operations
- 5) DAPM description.
- 6) DAPM event handler.
-
-Optionally, codec drivers can also provide:-
-
- 7) DAC Digital mute control.
-
-Its probably best to use this guide in conjunction with the existing codec
-driver code in sound/soc/codecs/
-
-ASoC Codec driver breakdown
-===========================
-
-1 - Codec DAI and PCM configuration
------------------------------------
-Each codec driver must have a struct snd_soc_dai_driver to define its DAI and
-PCM capabilities and operations. This struct is exported so that it can be
-registered with the core by your machine driver.
-
-e.g.
-
-static struct snd_soc_dai_ops wm8731_dai_ops = {
-	.prepare	= wm8731_pcm_prepare,
-	.hw_params	= wm8731_hw_params,
-	.shutdown	= wm8731_shutdown,
-	.digital_mute	= wm8731_mute,
-	.set_sysclk	= wm8731_set_dai_sysclk,
-	.set_fmt	= wm8731_set_dai_fmt,
-};
-
-struct snd_soc_dai_driver wm8731_dai = {
-	.name = "wm8731-hifi",
-	.playback = {
-		.stream_name = "Playback",
-		.channels_min = 1,
-		.channels_max = 2,
-		.rates = WM8731_RATES,
-		.formats = WM8731_FORMATS,},
-	.capture = {
-		.stream_name = "Capture",
-		.channels_min = 1,
-		.channels_max = 2,
-		.rates = WM8731_RATES,
-		.formats = WM8731_FORMATS,},
-	.ops = &wm8731_dai_ops,
-	.symmetric_rates = 1,
-};
-
-
-2 - Codec control IO
---------------------
-The codec can usually be controlled via an I2C or SPI style interface
-(AC97 combines control with data in the DAI). The codec driver should use the
-Regmap API for all codec IO. Please see include/linux/regmap.h and existing
-codec drivers for example regmap usage.
-
-
-3 - Mixers and audio controls
------------------------------
-All the codec mixers and audio controls can be defined using the convenience
-macros defined in soc.h.
-
-    #define SOC_SINGLE(xname, reg, shift, mask, invert)
-
-Defines a single control as follows:-
-
-  xname = Control name e.g. "Playback Volume"
-  reg = codec register
-  shift = control bit(s) offset in register
-  mask = control bit size(s) e.g. mask of 7 = 3 bits
-  invert = the control is inverted
-
-Other macros include:-
-
-    #define SOC_DOUBLE(xname, reg, shift_left, shift_right, mask, invert)
-
-A stereo control
-
-    #define SOC_DOUBLE_R(xname, reg_left, reg_right, shift, mask, invert)
-
-A stereo control spanning 2 registers
-
-    #define SOC_ENUM_SINGLE(xreg, xshift, xmask, xtexts)
-
-Defines an single enumerated control as follows:-
-
-   xreg = register
-   xshift = control bit(s) offset in register
-   xmask = control bit(s) size
-   xtexts = pointer to array of strings that describe each setting
-
-   #define SOC_ENUM_DOUBLE(xreg, xshift_l, xshift_r, xmask, xtexts)
-
-Defines a stereo enumerated control
-
-
-4 - Codec Audio Operations
---------------------------
-The codec driver also supports the following ALSA PCM operations:-
-
-/* SoC audio ops */
-struct snd_soc_ops {
-	int (*startup)(struct snd_pcm_substream *);
-	void (*shutdown)(struct snd_pcm_substream *);
-	int (*hw_params)(struct snd_pcm_substream *, struct snd_pcm_hw_params *);
-	int (*hw_free)(struct snd_pcm_substream *);
-	int (*prepare)(struct snd_pcm_substream *);
-};
-
-Please refer to the ALSA driver PCM documentation for details.
-http://www.alsa-project.org/~iwai/writing-an-alsa-driver/
-
-
-5 - DAPM description.
----------------------
-The Dynamic Audio Power Management description describes the codec power
-components and their relationships and registers to the ASoC core.
-Please read dapm.txt for details of building the description.
-
-Please also see the examples in other codec drivers.
-
-
-6 - DAPM event handler
-----------------------
-This function is a callback that handles codec domain PM calls and system
-domain PM calls (e.g. suspend and resume). It is used to put the codec
-to sleep when not in use.
-
-Power states:-
-
-	SNDRV_CTL_POWER_D0: /* full On */
-	/* vref/mid, clk and osc on, active */
-
-	SNDRV_CTL_POWER_D1: /* partial On */
-	SNDRV_CTL_POWER_D2: /* partial On */
-
-	SNDRV_CTL_POWER_D3hot: /* Off, with power */
-	/* everything off except vref/vmid, inactive */
-
-	SNDRV_CTL_POWER_D3cold: /* Everything Off, without power */
-
-
-7 - Codec DAC digital mute control
-----------------------------------
-Most codecs have a digital mute before the DACs that can be used to
-minimise any system noise.  The mute stops any digital data from
-entering the DAC.
-
-A callback can be created that is called by the core for each codec DAI
-when the mute is applied or freed.
-
-i.e.
-
-static int wm8974_mute(struct snd_soc_dai *dai, int mute)
-{
-	struct snd_soc_codec *codec = dai->codec;
-	u16 mute_reg = snd_soc_read(codec, WM8974_DAC) & 0xffbf;
-
-	if (mute)
-		snd_soc_write(codec, WM8974_DAC, mute_reg | 0x40);
-	else
-		snd_soc_write(codec, WM8974_DAC, mute_reg);
-	return 0;
-}
diff --git a/Documentation/sound/soc/codec.rst b/Documentation/sound/soc/codec.rst
new file mode 100644
index 000000000000..f87612b94812
--- /dev/null
+++ b/Documentation/sound/soc/codec.rst
@@ -0,0 +1,190 @@
+=======================
+ASoC Codec Class Driver
+=======================
+
+The codec class driver is generic and hardware independent code that configures
+the codec, FM, MODEM, BT or external DSP to provide audio capture and playback.
+It should contain no code that is specific to the target platform or machine.
+All platform and machine specific code should be added to the platform and
+machine drivers respectively.
+
+Each codec class driver *must* provide the following features:-
+
+1. Codec DAI and PCM configuration
+2. Codec control IO - using RegMap API
+3. Mixers and audio controls
+4. Codec audio operations
+5. DAPM description.
+6. DAPM event handler.
+
+Optionally, codec drivers can also provide:-
+
+7. DAC Digital mute control.
+
+Its probably best to use this guide in conjunction with the existing codec
+driver code in sound/soc/codecs/
+
+ASoC Codec driver breakdown
+===========================
+
+Codec DAI and PCM configuration
+-------------------------------
+Each codec driver must have a struct snd_soc_dai_driver to define its DAI and
+PCM capabilities and operations. This struct is exported so that it can be
+registered with the core by your machine driver.
+
+e.g.
+::
+
+  static struct snd_soc_dai_ops wm8731_dai_ops = {
+	.prepare	= wm8731_pcm_prepare,
+	.hw_params	= wm8731_hw_params,
+	.shutdown	= wm8731_shutdown,
+	.digital_mute	= wm8731_mute,
+	.set_sysclk	= wm8731_set_dai_sysclk,
+	.set_fmt	= wm8731_set_dai_fmt,
+  };
+  
+  struct snd_soc_dai_driver wm8731_dai = {
+	.name = "wm8731-hifi",
+	.playback = {
+		.stream_name = "Playback",
+		.channels_min = 1,
+		.channels_max = 2,
+		.rates = WM8731_RATES,
+		.formats = WM8731_FORMATS,},
+	.capture = {
+		.stream_name = "Capture",
+		.channels_min = 1,
+		.channels_max = 2,
+		.rates = WM8731_RATES,
+		.formats = WM8731_FORMATS,},
+	.ops = &wm8731_dai_ops,
+	.symmetric_rates = 1,
+  };
+
+
+Codec control IO
+----------------
+The codec can usually be controlled via an I2C or SPI style interface
+(AC97 combines control with data in the DAI). The codec driver should use the
+Regmap API for all codec IO. Please see include/linux/regmap.h and existing
+codec drivers for example regmap usage.
+
+
+Mixers and audio controls
+-------------------------
+All the codec mixers and audio controls can be defined using the convenience
+macros defined in soc.h.
+::
+
+    #define SOC_SINGLE(xname, reg, shift, mask, invert)
+
+Defines a single control as follows:-
+::
+
+  xname = Control name e.g. "Playback Volume"
+  reg = codec register
+  shift = control bit(s) offset in register
+  mask = control bit size(s) e.g. mask of 7 = 3 bits
+  invert = the control is inverted
+
+Other macros include:-
+::
+
+    #define SOC_DOUBLE(xname, reg, shift_left, shift_right, mask, invert)
+
+A stereo control
+::
+
+    #define SOC_DOUBLE_R(xname, reg_left, reg_right, shift, mask, invert)
+
+A stereo control spanning 2 registers
+::
+
+    #define SOC_ENUM_SINGLE(xreg, xshift, xmask, xtexts)
+
+Defines an single enumerated control as follows:-
+::
+
+   xreg = register
+   xshift = control bit(s) offset in register
+   xmask = control bit(s) size
+   xtexts = pointer to array of strings that describe each setting
+
+   #define SOC_ENUM_DOUBLE(xreg, xshift_l, xshift_r, xmask, xtexts)
+
+Defines a stereo enumerated control
+
+
+Codec Audio Operations
+----------------------
+The codec driver also supports the following ALSA PCM operations:-
+::
+
+  /* SoC audio ops */
+  struct snd_soc_ops {
+	int (*startup)(struct snd_pcm_substream *);
+	void (*shutdown)(struct snd_pcm_substream *);
+	int (*hw_params)(struct snd_pcm_substream *, struct snd_pcm_hw_params *);
+	int (*hw_free)(struct snd_pcm_substream *);
+	int (*prepare)(struct snd_pcm_substream *);
+  };
+
+Please refer to the ALSA driver PCM documentation for details.
+http://www.alsa-project.org/~iwai/writing-an-alsa-driver/
+
+
+DAPM description
+----------------
+The Dynamic Audio Power Management description describes the codec power
+components and their relationships and registers to the ASoC core.
+Please read dapm.txt for details of building the description.
+
+Please also see the examples in other codec drivers.
+
+
+DAPM event handler
+------------------
+This function is a callback that handles codec domain PM calls and system
+domain PM calls (e.g. suspend and resume). It is used to put the codec
+to sleep when not in use.
+
+Power states:-
+::
+
+	SNDRV_CTL_POWER_D0: /* full On */
+	/* vref/mid, clk and osc on, active */
+
+	SNDRV_CTL_POWER_D1: /* partial On */
+	SNDRV_CTL_POWER_D2: /* partial On */
+
+	SNDRV_CTL_POWER_D3hot: /* Off, with power */
+	/* everything off except vref/vmid, inactive */
+
+	SNDRV_CTL_POWER_D3cold: /* Everything Off, without power */
+
+
+Codec DAC digital mute control
+------------------------------
+Most codecs have a digital mute before the DACs that can be used to
+minimise any system noise.  The mute stops any digital data from
+entering the DAC.
+
+A callback can be created that is called by the core for each codec DAI
+when the mute is applied or freed.
+
+i.e.
+::
+
+  static int wm8974_mute(struct snd_soc_dai *dai, int mute)
+  {
+	struct snd_soc_codec *codec = dai->codec;
+	u16 mute_reg = snd_soc_read(codec, WM8974_DAC) & 0xffbf;
+
+	if (mute)
+		snd_soc_write(codec, WM8974_DAC, mute_reg | 0x40);
+	else
+		snd_soc_write(codec, WM8974_DAC, mute_reg);
+	return 0;
+  }
diff --git a/Documentation/sound/soc/index.rst b/Documentation/sound/soc/index.rst
index e974fd9f38a3..a2e023c91df2 100644
--- a/Documentation/sound/soc/index.rst
+++ b/Documentation/sound/soc/index.rst
@@ -8,3 +8,4 @@ The documentation is spilt into the following sections:-
    :maxdepth: 2
 
    overview
+   codec
-- 
cgit v1.2.3-58-ga151


From e732d1bcd452a040a18242d555996703465c1ca7 Mon Sep 17 00:00:00 2001
From: Takashi Iwai 
Date: Thu, 10 Nov 2016 22:17:40 +0100
Subject: ASoC: doc: ReSTize DAI.txt

A simple conversion from a plain text file with slight reformatting /
corrections.

The file name was changed to lower letters to align with others.

Acked-by: Mark Brown 
Signed-off-by: Takashi Iwai 
---
 Documentation/sound/alsa/soc/DAI.txt | 56 -------------------------------
 Documentation/sound/soc/dai.rst      | 64 ++++++++++++++++++++++++++++++++++++
 Documentation/sound/soc/index.rst    |  1 +
 3 files changed, 65 insertions(+), 56 deletions(-)
 delete mode 100644 Documentation/sound/alsa/soc/DAI.txt
 create mode 100644 Documentation/sound/soc/dai.rst

(limited to 'Documentation/sound')

diff --git a/Documentation/sound/alsa/soc/DAI.txt b/Documentation/sound/alsa/soc/DAI.txt
deleted file mode 100644
index c9679264c559..000000000000
--- a/Documentation/sound/alsa/soc/DAI.txt
+++ /dev/null
@@ -1,56 +0,0 @@
-ASoC currently supports the three main Digital Audio Interfaces (DAI) found on
-SoC controllers and portable audio CODECs today, namely AC97, I2S and PCM.
-
-
-AC97
-====
-
-  AC97 is a five wire interface commonly found on many PC sound cards. It is
-now also popular in many portable devices. This DAI has a reset line and time
-multiplexes its data on its SDATA_OUT (playback) and SDATA_IN (capture) lines.
-The bit clock (BCLK) is always driven by the CODEC (usually 12.288MHz) and the
-frame (FRAME) (usually 48kHz) is always driven by the controller. Each AC97
-frame is 21uS long and is divided into 13 time slots.
-
-The AC97 specification can be found at :-
-http://www.intel.com/p/en_US/business/design
-
-
-I2S
-===
-
- I2S is a common 4 wire DAI used in HiFi, STB and portable devices. The Tx and
-Rx lines are used for audio transmission, whilst the bit clock (BCLK) and
-left/right clock (LRC) synchronise the link. I2S is flexible in that either the
-controller or CODEC can drive (master) the BCLK and LRC clock lines. Bit clock
-usually varies depending on the sample rate and the master system clock
-(SYSCLK). LRCLK is the same as the sample rate. A few devices support separate
-ADC and DAC LRCLKs, this allows for simultaneous capture and playback at
-different sample rates.
-
-I2S has several different operating modes:-
-
- o I2S - MSB is transmitted on the falling edge of the first BCLK after LRC
-         transition.
-
- o Left Justified - MSB is transmitted on transition of LRC.
-
- o Right Justified - MSB is transmitted sample size BCLKs before LRC
-                     transition.
-
-PCM
-===
-
-PCM is another 4 wire interface, very similar to I2S, which can support a more
-flexible protocol. It has bit clock (BCLK) and sync (SYNC) lines that are used
-to synchronise the link whilst the Tx and Rx lines are used to transmit and
-receive the audio data. Bit clock usually varies depending on sample rate
-whilst sync runs at the sample rate. PCM also supports Time Division
-Multiplexing (TDM) in that several devices can use the bus simultaneously (this
-is sometimes referred to as network mode).
-
-Common PCM operating modes:-
-
- o Mode A - MSB is transmitted on falling edge of first BCLK after FRAME/SYNC.
-
- o Mode B - MSB is transmitted on rising edge of FRAME/SYNC.
diff --git a/Documentation/sound/soc/dai.rst b/Documentation/sound/soc/dai.rst
new file mode 100644
index 000000000000..55820e51708f
--- /dev/null
+++ b/Documentation/sound/soc/dai.rst
@@ -0,0 +1,64 @@
+==================================
+ASoC Digital Audio Interface (DAI)
+==================================
+
+ASoC currently supports the three main Digital Audio Interfaces (DAI) found on
+SoC controllers and portable audio CODECs today, namely AC97, I2S and PCM.
+
+
+AC97
+====
+
+AC97 is a five wire interface commonly found on many PC sound cards. It is
+now also popular in many portable devices. This DAI has a reset line and time
+multiplexes its data on its SDATA_OUT (playback) and SDATA_IN (capture) lines.
+The bit clock (BCLK) is always driven by the CODEC (usually 12.288MHz) and the
+frame (FRAME) (usually 48kHz) is always driven by the controller. Each AC97
+frame is 21uS long and is divided into 13 time slots.
+
+The AC97 specification can be found at :
+http://www.intel.com/p/en_US/business/design
+
+
+I2S
+===
+
+I2S is a common 4 wire DAI used in HiFi, STB and portable devices. The Tx and
+Rx lines are used for audio transmission, whilst the bit clock (BCLK) and
+left/right clock (LRC) synchronise the link. I2S is flexible in that either the
+controller or CODEC can drive (master) the BCLK and LRC clock lines. Bit clock
+usually varies depending on the sample rate and the master system clock
+(SYSCLK). LRCLK is the same as the sample rate. A few devices support separate
+ADC and DAC LRCLKs, this allows for simultaneous capture and playback at
+different sample rates.
+
+I2S has several different operating modes:-
+
+I2S
+  MSB is transmitted on the falling edge of the first BCLK after LRC
+  transition.
+
+Left Justified
+  MSB is transmitted on transition of LRC.
+
+Right Justified
+  MSB is transmitted sample size BCLKs before LRC transition.
+
+PCM
+===
+
+PCM is another 4 wire interface, very similar to I2S, which can support a more
+flexible protocol. It has bit clock (BCLK) and sync (SYNC) lines that are used
+to synchronise the link whilst the Tx and Rx lines are used to transmit and
+receive the audio data. Bit clock usually varies depending on sample rate
+whilst sync runs at the sample rate. PCM also supports Time Division
+Multiplexing (TDM) in that several devices can use the bus simultaneously (this
+is sometimes referred to as network mode).
+
+Common PCM operating modes:-
+
+Mode A
+  MSB is transmitted on falling edge of first BCLK after FRAME/SYNC.
+
+Mode B
+  MSB is transmitted on rising edge of FRAME/SYNC.
diff --git a/Documentation/sound/soc/index.rst b/Documentation/sound/soc/index.rst
index a2e023c91df2..aea7ae7e5aad 100644
--- a/Documentation/sound/soc/index.rst
+++ b/Documentation/sound/soc/index.rst
@@ -9,3 +9,4 @@ The documentation is spilt into the following sections:-
 
    overview
    codec
+   dai
-- 
cgit v1.2.3-58-ga151


From 77190f033377634bd494b67ca5a2dbcf685bd462 Mon Sep 17 00:00:00 2001
From: Takashi Iwai 
Date: Thu, 10 Nov 2016 22:19:33 +0100
Subject: ASoC: doc: ReSTize dapm.txt

A simple conversion from a plain text file.
The section numbers and the item numbers are dropped to align with the
ReST format.  Some lists are converted to description lists to be
clearer.

Acked-by: Mark Brown 
Signed-off-by: Takashi Iwai 
---
 Documentation/sound/alsa/soc/dapm.txt | 305 ------------------------------
 Documentation/sound/soc/dapm.rst      | 342 ++++++++++++++++++++++++++++++++++
 Documentation/sound/soc/index.rst     |   1 +
 3 files changed, 343 insertions(+), 305 deletions(-)
 delete mode 100644 Documentation/sound/alsa/soc/dapm.txt
 create mode 100644 Documentation/sound/soc/dapm.rst

(limited to 'Documentation/sound')

diff --git a/Documentation/sound/alsa/soc/dapm.txt b/Documentation/sound/alsa/soc/dapm.txt
deleted file mode 100644
index c45bd79f291e..000000000000
--- a/Documentation/sound/alsa/soc/dapm.txt
+++ /dev/null
@@ -1,305 +0,0 @@
-Dynamic Audio Power Management for Portable Devices
-===================================================
-
-1. Description
-==============
-
-Dynamic Audio Power Management (DAPM) is designed to allow portable
-Linux devices to use the minimum amount of power within the audio
-subsystem at all times. It is independent of other kernel PM and as
-such, can easily co-exist with the other PM systems.
-
-DAPM is also completely transparent to all user space applications as
-all power switching is done within the ASoC core. No code changes or
-recompiling are required for user space applications. DAPM makes power
-switching decisions based upon any audio stream (capture/playback)
-activity and audio mixer settings within the device.
-
-DAPM spans the whole machine. It covers power control within the entire
-audio subsystem, this includes internal codec power blocks and machine
-level power systems.
-
-There are 4 power domains within DAPM
-
-   1. Codec bias domain - VREF, VMID (core codec and audio power)
-      Usually controlled at codec probe/remove and suspend/resume, although
-      can be set at stream time if power is not needed for sidetone, etc.
-
-   2. Platform/Machine domain - physically connected inputs and outputs
-      Is platform/machine and user action specific, is configured by the
-      machine driver and responds to asynchronous events e.g when HP
-      are inserted
-
-   3. Path domain - audio subsystem signal paths
-      Automatically set when mixer and mux settings are changed by the user.
-      e.g. alsamixer, amixer.
-
-   4. Stream domain - DACs and ADCs.
-      Enabled and disabled when stream playback/capture is started and
-      stopped respectively. e.g. aplay, arecord.
-
-All DAPM power switching decisions are made automatically by consulting an audio
-routing map of the whole machine. This map is specific to each machine and
-consists of the interconnections between every audio component (including
-internal codec components). All audio components that effect power are called
-widgets hereafter.
-
-
-2. DAPM Widgets
-===============
-
-Audio DAPM widgets fall into a number of types:-
-
- o Mixer      - Mixes several analog signals into a single analog signal.
- o Mux        - An analog switch that outputs only one of many inputs.
- o PGA        - A programmable gain amplifier or attenuation widget.
- o ADC        - Analog to Digital Converter
- o DAC        - Digital to Analog Converter
- o Switch     - An analog switch
- o Input      - A codec input pin
- o Output     - A codec output pin
- o Headphone  - Headphone (and optional Jack)
- o Mic        - Mic (and optional Jack)
- o Line       - Line Input/Output (and optional Jack)
- o Speaker    - Speaker
- o Supply     - Power or clock supply widget used by other widgets.
- o Regulator  - External regulator that supplies power to audio components.
- o Clock      -	External clock that supplies clock to audio components.
- o AIF IN     - Audio Interface Input (with TDM slot mask).
- o AIF OUT    - Audio Interface Output (with TDM slot mask).
- o Siggen     - Signal Generator.
- o DAI IN     - Digital Audio Interface Input.
- o DAI OUT    - Digital Audio Interface Output.
- o DAI Link   - DAI Link between two DAI structures */
- o Pre        - Special PRE widget (exec before all others)
- o Post       - Special POST widget (exec after all others)
-
-(Widgets are defined in include/sound/soc-dapm.h)
-
-Widgets can be added to the sound card by any of the component driver types.
-There are convenience macros defined in soc-dapm.h that can be used to quickly
-build a list of widgets of the codecs and machines DAPM widgets.
-
-Most widgets have a name, register, shift and invert. Some widgets have extra
-parameters for stream name and kcontrols.
-
-
-2.1 Stream Domain Widgets
--------------------------
-
-Stream Widgets relate to the stream power domain and only consist of ADCs
-(analog to digital converters), DACs (digital to analog converters),
-AIF IN and AIF OUT.
-
-Stream widgets have the following format:-
-
-SND_SOC_DAPM_DAC(name, stream name, reg, shift, invert),
-SND_SOC_DAPM_AIF_IN(name, stream, slot, reg, shift, invert)
-
-NOTE: the stream name must match the corresponding stream name in your codec
-snd_soc_codec_dai.
-
-e.g. stream widgets for HiFi playback and capture
-
-SND_SOC_DAPM_DAC("HiFi DAC", "HiFi Playback", REG, 3, 1),
-SND_SOC_DAPM_ADC("HiFi ADC", "HiFi Capture", REG, 2, 1),
-
-e.g. stream widgets for AIF
-
-SND_SOC_DAPM_AIF_IN("AIF1RX", "AIF1 Playback", 0, SND_SOC_NOPM, 0, 0),
-SND_SOC_DAPM_AIF_OUT("AIF1TX", "AIF1 Capture", 0, SND_SOC_NOPM, 0, 0),
-
-
-2.2 Path Domain Widgets
------------------------
-
-Path domain widgets have a ability to control or affect the audio signal or
-audio paths within the audio subsystem. They have the following form:-
-
-SND_SOC_DAPM_PGA(name, reg, shift, invert, controls, num_controls)
-
-Any widget kcontrols can be set using the controls and num_controls members.
-
-e.g. Mixer widget (the kcontrols are declared first)
-
-/* Output Mixer */
-static const snd_kcontrol_new_t wm8731_output_mixer_controls[] = {
-SOC_DAPM_SINGLE("Line Bypass Switch", WM8731_APANA, 3, 1, 0),
-SOC_DAPM_SINGLE("Mic Sidetone Switch", WM8731_APANA, 5, 1, 0),
-SOC_DAPM_SINGLE("HiFi Playback Switch", WM8731_APANA, 4, 1, 0),
-};
-
-SND_SOC_DAPM_MIXER("Output Mixer", WM8731_PWR, 4, 1, wm8731_output_mixer_controls,
-	ARRAY_SIZE(wm8731_output_mixer_controls)),
-
-If you don't want the mixer elements prefixed with the name of the mixer widget,
-you can use SND_SOC_DAPM_MIXER_NAMED_CTL instead. the parameters are the same
-as for SND_SOC_DAPM_MIXER.
-
-
-2.3 Machine domain Widgets
---------------------------
-
-Machine widgets are different from codec widgets in that they don't have a
-codec register bit associated with them. A machine widget is assigned to each
-machine audio component (non codec or DSP) that can be independently
-powered. e.g.
-
- o Speaker Amp
- o Microphone Bias
- o Jack connectors
-
-A machine widget can have an optional call back.
-
-e.g. Jack connector widget for an external Mic that enables Mic Bias
-when the Mic is inserted:-
-
-static int spitz_mic_bias(struct snd_soc_dapm_widget* w, int event)
-{
-	gpio_set_value(SPITZ_GPIO_MIC_BIAS, SND_SOC_DAPM_EVENT_ON(event));
-	return 0;
-}
-
-SND_SOC_DAPM_MIC("Mic Jack", spitz_mic_bias),
-
-
-2.4 Codec (BIAS) Domain
------------------------
-
-The codec bias power domain has no widgets and is handled by the codecs DAPM
-event handler. This handler is called when the codec powerstate is changed wrt
-to any stream event or by kernel PM events.
-
-
-2.5 Virtual Widgets
--------------------
-
-Sometimes widgets exist in the codec or machine audio map that don't have any
-corresponding soft power control. In this case it is necessary to create
-a virtual widget - a widget with no control bits e.g.
-
-SND_SOC_DAPM_MIXER("AC97 Mixer", SND_SOC_DAPM_NOPM, 0, 0, NULL, 0),
-
-This can be used to merge to signal paths together in software.
-
-After all the widgets have been defined, they can then be added to the DAPM
-subsystem individually with a call to snd_soc_dapm_new_control().
-
-
-3. Codec/DSP Widget Interconnections
-====================================
-
-Widgets are connected to each other within the codec, platform and machine by
-audio paths (called interconnections). Each interconnection must be defined in
-order to create a map of all audio paths between widgets.
-
-This is easiest with a diagram of the codec or DSP (and schematic of the machine
-audio system), as it requires joining widgets together via their audio signal
-paths.
-
-e.g., from the WM8731 output mixer (wm8731.c)
-
-The WM8731 output mixer has 3 inputs (sources)
-
- 1. Line Bypass Input
- 2. DAC (HiFi playback)
- 3. Mic Sidetone Input
-
-Each input in this example has a kcontrol associated with it (defined in example
-above) and is connected to the output mixer via its kcontrol name. We can now
-connect the destination widget (wrt audio signal) with its source widgets.
-
-	/* output mixer */
-	{"Output Mixer", "Line Bypass Switch", "Line Input"},
-	{"Output Mixer", "HiFi Playback Switch", "DAC"},
-	{"Output Mixer", "Mic Sidetone Switch", "Mic Bias"},
-
-So we have :-
-
-	Destination Widget  <=== Path Name <=== Source Widget
-
-Or:-
-
-	Sink, Path, Source
-
-Or :-
-
-	"Output Mixer" is connected to the "DAC" via the "HiFi Playback Switch".
-
-When there is no path name connecting widgets (e.g. a direct connection) we
-pass NULL for the path name.
-
-Interconnections are created with a call to:-
-
-snd_soc_dapm_connect_input(codec, sink, path, source);
-
-Finally, snd_soc_dapm_new_widgets(codec) must be called after all widgets and
-interconnections have been registered with the core. This causes the core to
-scan the codec and machine so that the internal DAPM state matches the
-physical state of the machine.
-
-
-3.1 Machine Widget Interconnections
------------------------------------
-Machine widget interconnections are created in the same way as codec ones and
-directly connect the codec pins to machine level widgets.
-
-e.g. connects the speaker out codec pins to the internal speaker.
-
-	/* ext speaker connected to codec pins LOUT2, ROUT2  */
-	{"Ext Spk", NULL , "ROUT2"},
-	{"Ext Spk", NULL , "LOUT2"},
-
-This allows the DAPM to power on and off pins that are connected (and in use)
-and pins that are NC respectively.
-
-
-4 Endpoint Widgets
-===================
-An endpoint is a start or end point (widget) of an audio signal within the
-machine and includes the codec. e.g.
-
- o Headphone Jack
- o Internal Speaker
- o Internal Mic
- o Mic Jack
- o Codec Pins
-
-Endpoints are added to the DAPM graph so that their usage can be determined in
-order to save power. e.g. NC codecs pins will be switched OFF, unconnected
-jacks can also be switched OFF.
-
-
-5 DAPM Widget Events
-====================
-
-Some widgets can register their interest with the DAPM core in PM events.
-e.g. A Speaker with an amplifier registers a widget so the amplifier can be
-powered only when the spk is in use.
-
-/* turn speaker amplifier on/off depending on use */
-static int corgi_amp_event(struct snd_soc_dapm_widget *w, int event)
-{
-	gpio_set_value(CORGI_GPIO_APM_ON, SND_SOC_DAPM_EVENT_ON(event));
-	return 0;
-}
-
-/* corgi machine dapm widgets */
-static const struct snd_soc_dapm_widget wm8731_dapm_widgets =
-	SND_SOC_DAPM_SPK("Ext Spk", corgi_amp_event);
-
-Please see soc-dapm.h for all other widgets that support events.
-
-
-5.1 Event types
----------------
-
-The following event types are supported by event widgets.
-
-/* dapm event types */
-#define SND_SOC_DAPM_PRE_PMU	0x1 	/* before widget power up */
-#define SND_SOC_DAPM_POST_PMU	0x2		/* after widget power up */
-#define SND_SOC_DAPM_PRE_PMD	0x4 	/* before widget power down */
-#define SND_SOC_DAPM_POST_PMD	0x8		/* after widget power down */
-#define SND_SOC_DAPM_PRE_REG	0x10	/* before audio path setup */
-#define SND_SOC_DAPM_POST_REG	0x20	/* after audio path setup */
diff --git a/Documentation/sound/soc/dapm.rst b/Documentation/sound/soc/dapm.rst
new file mode 100644
index 000000000000..a27f42befa4d
--- /dev/null
+++ b/Documentation/sound/soc/dapm.rst
@@ -0,0 +1,342 @@
+===================================================
+Dynamic Audio Power Management for Portable Devices
+===================================================
+
+Description
+===========
+
+Dynamic Audio Power Management (DAPM) is designed to allow portable
+Linux devices to use the minimum amount of power within the audio
+subsystem at all times. It is independent of other kernel PM and as
+such, can easily co-exist with the other PM systems.
+
+DAPM is also completely transparent to all user space applications as
+all power switching is done within the ASoC core. No code changes or
+recompiling are required for user space applications. DAPM makes power
+switching decisions based upon any audio stream (capture/playback)
+activity and audio mixer settings within the device.
+
+DAPM spans the whole machine. It covers power control within the entire
+audio subsystem, this includes internal codec power blocks and machine
+level power systems.
+
+There are 4 power domains within DAPM
+
+Codec bias domain
+      VREF, VMID (core codec and audio power)
+
+      Usually controlled at codec probe/remove and suspend/resume, although
+      can be set at stream time if power is not needed for sidetone, etc.
+
+Platform/Machine domain
+      physically connected inputs and outputs
+
+      Is platform/machine and user action specific, is configured by the
+      machine driver and responds to asynchronous events e.g when HP
+      are inserted
+
+Path domain
+      audio subsystem signal paths
+
+      Automatically set when mixer and mux settings are changed by the user.
+      e.g. alsamixer, amixer.
+
+Stream domain
+      DACs and ADCs.
+
+      Enabled and disabled when stream playback/capture is started and
+      stopped respectively. e.g. aplay, arecord.
+
+All DAPM power switching decisions are made automatically by consulting an audio
+routing map of the whole machine. This map is specific to each machine and
+consists of the interconnections between every audio component (including
+internal codec components). All audio components that effect power are called
+widgets hereafter.
+
+
+DAPM Widgets
+============
+
+Audio DAPM widgets fall into a number of types:-
+
+Mixer
+	Mixes several analog signals into a single analog signal.
+Mux
+	An analog switch that outputs only one of many inputs.
+PGA
+	A programmable gain amplifier or attenuation widget.
+ADC
+	Analog to Digital Converter
+DAC
+	Digital to Analog Converter
+Switch
+	An analog switch
+Input
+	A codec input pin
+Output
+	A codec output pin
+Headphone
+	Headphone (and optional Jack)
+Mic
+	Mic (and optional Jack)
+Line
+	Line Input/Output (and optional Jack)
+Speaker
+	Speaker
+Supply
+	Power or clock supply widget used by other widgets.
+Regulator
+	External regulator that supplies power to audio components.
+Clock
+	External clock that supplies clock to audio components.
+AIF IN
+	Audio Interface Input (with TDM slot mask).
+AIF OUT
+	Audio Interface Output (with TDM slot mask).
+Siggen
+	Signal Generator.
+DAI IN
+	Digital Audio Interface Input.
+DAI OUT
+	Digital Audio Interface Output.
+DAI Link
+	DAI Link between two DAI structures
+Pre
+	Special PRE widget (exec before all others)
+Post
+	Special POST widget (exec after all others)
+
+(Widgets are defined in include/sound/soc-dapm.h)
+
+Widgets can be added to the sound card by any of the component driver types.
+There are convenience macros defined in soc-dapm.h that can be used to quickly
+build a list of widgets of the codecs and machines DAPM widgets.
+
+Most widgets have a name, register, shift and invert. Some widgets have extra
+parameters for stream name and kcontrols.
+
+
+Stream Domain Widgets
+---------------------
+
+Stream Widgets relate to the stream power domain and only consist of ADCs
+(analog to digital converters), DACs (digital to analog converters),
+AIF IN and AIF OUT.
+
+Stream widgets have the following format:-
+::
+
+  SND_SOC_DAPM_DAC(name, stream name, reg, shift, invert),
+  SND_SOC_DAPM_AIF_IN(name, stream, slot, reg, shift, invert)
+
+NOTE: the stream name must match the corresponding stream name in your codec
+snd_soc_codec_dai.
+
+e.g. stream widgets for HiFi playback and capture
+::
+
+  SND_SOC_DAPM_DAC("HiFi DAC", "HiFi Playback", REG, 3, 1),
+  SND_SOC_DAPM_ADC("HiFi ADC", "HiFi Capture", REG, 2, 1),
+
+e.g. stream widgets for AIF
+::
+
+  SND_SOC_DAPM_AIF_IN("AIF1RX", "AIF1 Playback", 0, SND_SOC_NOPM, 0, 0),
+  SND_SOC_DAPM_AIF_OUT("AIF1TX", "AIF1 Capture", 0, SND_SOC_NOPM, 0, 0),
+
+
+Path Domain Widgets
+-------------------
+
+Path domain widgets have a ability to control or affect the audio signal or
+audio paths within the audio subsystem. They have the following form:-
+::
+
+  SND_SOC_DAPM_PGA(name, reg, shift, invert, controls, num_controls)
+
+Any widget kcontrols can be set using the controls and num_controls members.
+
+e.g. Mixer widget (the kcontrols are declared first)
+::
+
+  /* Output Mixer */
+  static const snd_kcontrol_new_t wm8731_output_mixer_controls[] = {
+  SOC_DAPM_SINGLE("Line Bypass Switch", WM8731_APANA, 3, 1, 0),
+  SOC_DAPM_SINGLE("Mic Sidetone Switch", WM8731_APANA, 5, 1, 0),
+  SOC_DAPM_SINGLE("HiFi Playback Switch", WM8731_APANA, 4, 1, 0),
+  };
+
+  SND_SOC_DAPM_MIXER("Output Mixer", WM8731_PWR, 4, 1, wm8731_output_mixer_controls,
+	ARRAY_SIZE(wm8731_output_mixer_controls)),
+
+If you don't want the mixer elements prefixed with the name of the mixer widget,
+you can use SND_SOC_DAPM_MIXER_NAMED_CTL instead. the parameters are the same
+as for SND_SOC_DAPM_MIXER.
+
+
+Machine domain Widgets
+----------------------
+
+Machine widgets are different from codec widgets in that they don't have a
+codec register bit associated with them. A machine widget is assigned to each
+machine audio component (non codec or DSP) that can be independently
+powered. e.g.
+
+* Speaker Amp
+* Microphone Bias
+* Jack connectors
+
+A machine widget can have an optional call back.
+
+e.g. Jack connector widget for an external Mic that enables Mic Bias
+when the Mic is inserted:-::
+
+  static int spitz_mic_bias(struct snd_soc_dapm_widget* w, int event)
+  {
+	gpio_set_value(SPITZ_GPIO_MIC_BIAS, SND_SOC_DAPM_EVENT_ON(event));
+	return 0;
+  }
+
+  SND_SOC_DAPM_MIC("Mic Jack", spitz_mic_bias),
+
+
+Codec (BIAS) Domain
+-------------------
+
+The codec bias power domain has no widgets and is handled by the codecs DAPM
+event handler. This handler is called when the codec powerstate is changed wrt
+to any stream event or by kernel PM events.
+
+
+Virtual Widgets
+---------------
+
+Sometimes widgets exist in the codec or machine audio map that don't have any
+corresponding soft power control. In this case it is necessary to create
+a virtual widget - a widget with no control bits e.g.
+::
+
+  SND_SOC_DAPM_MIXER("AC97 Mixer", SND_SOC_DAPM_NOPM, 0, 0, NULL, 0),
+
+This can be used to merge to signal paths together in software.
+
+After all the widgets have been defined, they can then be added to the DAPM
+subsystem individually with a call to snd_soc_dapm_new_control().
+
+
+Codec/DSP Widget Interconnections
+=================================
+
+Widgets are connected to each other within the codec, platform and machine by
+audio paths (called interconnections). Each interconnection must be defined in
+order to create a map of all audio paths between widgets.
+
+This is easiest with a diagram of the codec or DSP (and schematic of the machine
+audio system), as it requires joining widgets together via their audio signal
+paths.
+
+e.g., from the WM8731 output mixer (wm8731.c)
+
+The WM8731 output mixer has 3 inputs (sources)
+
+1. Line Bypass Input
+2. DAC (HiFi playback)
+3. Mic Sidetone Input
+
+Each input in this example has a kcontrol associated with it (defined in example
+above) and is connected to the output mixer via its kcontrol name. We can now
+connect the destination widget (wrt audio signal) with its source widgets.
+::
+
+	/* output mixer */
+	{"Output Mixer", "Line Bypass Switch", "Line Input"},
+	{"Output Mixer", "HiFi Playback Switch", "DAC"},
+	{"Output Mixer", "Mic Sidetone Switch", "Mic Bias"},
+
+So we have :-
+
+* Destination Widget  <=== Path Name <=== Source Widget, or
+* Sink, Path, Source, or
+* ``Output Mixer`` is connected to the ``DAC`` via the ``HiFi Playback Switch``.
+
+When there is no path name connecting widgets (e.g. a direct connection) we
+pass NULL for the path name.
+
+Interconnections are created with a call to:-
+::
+
+  snd_soc_dapm_connect_input(codec, sink, path, source);
+
+Finally, snd_soc_dapm_new_widgets(codec) must be called after all widgets and
+interconnections have been registered with the core. This causes the core to
+scan the codec and machine so that the internal DAPM state matches the
+physical state of the machine.
+
+
+Machine Widget Interconnections
+-------------------------------
+Machine widget interconnections are created in the same way as codec ones and
+directly connect the codec pins to machine level widgets.
+
+e.g. connects the speaker out codec pins to the internal speaker.
+::
+
+	/* ext speaker connected to codec pins LOUT2, ROUT2  */
+	{"Ext Spk", NULL , "ROUT2"},
+	{"Ext Spk", NULL , "LOUT2"},
+
+This allows the DAPM to power on and off pins that are connected (and in use)
+and pins that are NC respectively.
+
+
+Endpoint Widgets
+================
+An endpoint is a start or end point (widget) of an audio signal within the
+machine and includes the codec. e.g.
+
+* Headphone Jack
+* Internal Speaker
+* Internal Mic
+* Mic Jack
+* Codec Pins
+
+Endpoints are added to the DAPM graph so that their usage can be determined in
+order to save power. e.g. NC codecs pins will be switched OFF, unconnected
+jacks can also be switched OFF.
+
+
+DAPM Widget Events
+==================
+
+Some widgets can register their interest with the DAPM core in PM events.
+e.g. A Speaker with an amplifier registers a widget so the amplifier can be
+powered only when the spk is in use.
+::
+
+  /* turn speaker amplifier on/off depending on use */
+  static int corgi_amp_event(struct snd_soc_dapm_widget *w, int event)
+  {
+	gpio_set_value(CORGI_GPIO_APM_ON, SND_SOC_DAPM_EVENT_ON(event));
+	return 0;
+  }
+
+  /* corgi machine dapm widgets */
+  static const struct snd_soc_dapm_widget wm8731_dapm_widgets =
+	SND_SOC_DAPM_SPK("Ext Spk", corgi_amp_event);
+
+Please see soc-dapm.h for all other widgets that support events.
+
+
+Event types
+-----------
+
+The following event types are supported by event widgets.
+::
+
+  /* dapm event types */
+  #define SND_SOC_DAPM_PRE_PMU	0x1 	/* before widget power up */
+  #define SND_SOC_DAPM_POST_PMU	0x2		/* after widget power up */
+  #define SND_SOC_DAPM_PRE_PMD	0x4 	/* before widget power down */
+  #define SND_SOC_DAPM_POST_PMD	0x8		/* after widget power down */
+  #define SND_SOC_DAPM_PRE_REG	0x10	/* before audio path setup */
+  #define SND_SOC_DAPM_POST_REG	0x20	/* after audio path setup */
diff --git a/Documentation/sound/soc/index.rst b/Documentation/sound/soc/index.rst
index aea7ae7e5aad..0c0c38e582b4 100644
--- a/Documentation/sound/soc/index.rst
+++ b/Documentation/sound/soc/index.rst
@@ -10,3 +10,4 @@ The documentation is spilt into the following sections:-
    overview
    codec
    dai
+   dapm
-- 
cgit v1.2.3-58-ga151


From d8a5d624cc39c416f008e52b940f47cf619dbd9e Mon Sep 17 00:00:00 2001
From: Takashi Iwai 
Date: Thu, 10 Nov 2016 22:21:56 +0100
Subject: ASoC: doc: ReSTize platform.txt

A simple conversion from a plain text file.

Acked-by: Mark Brown 
Signed-off-by: Takashi Iwai 
---
 Documentation/sound/alsa/soc/platform.txt | 79 -----------------------------
 Documentation/sound/soc/index.rst         |  1 +
 Documentation/sound/soc/platform.rst      | 82 +++++++++++++++++++++++++++++++
 3 files changed, 83 insertions(+), 79 deletions(-)
 delete mode 100644 Documentation/sound/alsa/soc/platform.txt
 create mode 100644 Documentation/sound/soc/platform.rst

(limited to 'Documentation/sound')

diff --git a/Documentation/sound/alsa/soc/platform.txt b/Documentation/sound/alsa/soc/platform.txt
deleted file mode 100644
index 3a08a2c9150c..000000000000
--- a/Documentation/sound/alsa/soc/platform.txt
+++ /dev/null
@@ -1,79 +0,0 @@
-ASoC Platform Driver
-====================
-
-An ASoC platform driver class can be divided into audio DMA drivers, SoC DAI
-drivers and DSP drivers. The platform drivers only target the SoC CPU and must
-have no board specific code.
-
-Audio DMA
-=========
-
-The platform DMA driver optionally supports the following ALSA operations:-
-
-/* SoC audio ops */
-struct snd_soc_ops {
-	int (*startup)(struct snd_pcm_substream *);
-	void (*shutdown)(struct snd_pcm_substream *);
-	int (*hw_params)(struct snd_pcm_substream *, struct snd_pcm_hw_params *);
-	int (*hw_free)(struct snd_pcm_substream *);
-	int (*prepare)(struct snd_pcm_substream *);
-	int (*trigger)(struct snd_pcm_substream *, int);
-};
-
-The platform driver exports its DMA functionality via struct
-snd_soc_platform_driver:-
-
-struct snd_soc_platform_driver {
-	char *name;
-
-	int (*probe)(struct platform_device *pdev);
-	int (*remove)(struct platform_device *pdev);
-	int (*suspend)(struct platform_device *pdev, struct snd_soc_cpu_dai *cpu_dai);
-	int (*resume)(struct platform_device *pdev, struct snd_soc_cpu_dai *cpu_dai);
-
-	/* pcm creation and destruction */
-	int (*pcm_new)(struct snd_card *, struct snd_soc_codec_dai *, struct snd_pcm *);
-	void (*pcm_free)(struct snd_pcm *);
-
-	/*
-	 * For platform caused delay reporting.
-	 * Optional.
-	 */
-	snd_pcm_sframes_t (*delay)(struct snd_pcm_substream *,
-		struct snd_soc_dai *);
-
-	/* platform stream ops */
-	struct snd_pcm_ops *pcm_ops;
-};
-
-Please refer to the ALSA driver documentation for details of audio DMA.
-http://www.alsa-project.org/~iwai/writing-an-alsa-driver/
-
-An example DMA driver is soc/pxa/pxa2xx-pcm.c
-
-
-SoC DAI Drivers
-===============
-
-Each SoC DAI driver must provide the following features:-
-
- 1) Digital audio interface (DAI) description
- 2) Digital audio interface configuration
- 3) PCM's description
- 4) SYSCLK configuration
- 5) Suspend and resume (optional)
-
-Please see codec.txt for a description of items 1 - 4.
-
-
-SoC DSP Drivers
-===============
-
-Each SoC DSP driver usually supplies the following features :-
-
- 1) DAPM graph
- 2) Mixer controls
- 3) DMA IO to/from DSP buffers (if applicable)
- 4) Definition of DSP front end (FE) PCM devices.
-
-Please see DPCM.txt for a description of item 4.
diff --git a/Documentation/sound/soc/index.rst b/Documentation/sound/soc/index.rst
index 0c0c38e582b4..c5a55195bf4d 100644
--- a/Documentation/sound/soc/index.rst
+++ b/Documentation/sound/soc/index.rst
@@ -11,3 +11,4 @@ The documentation is spilt into the following sections:-
    codec
    dai
    dapm
+   platform
diff --git a/Documentation/sound/soc/platform.rst b/Documentation/sound/soc/platform.rst
new file mode 100644
index 000000000000..d5574904d981
--- /dev/null
+++ b/Documentation/sound/soc/platform.rst
@@ -0,0 +1,82 @@
+====================
+ASoC Platform Driver
+====================
+
+An ASoC platform driver class can be divided into audio DMA drivers, SoC DAI
+drivers and DSP drivers. The platform drivers only target the SoC CPU and must
+have no board specific code.
+
+Audio DMA
+=========
+
+The platform DMA driver optionally supports the following ALSA operations:-
+::
+
+  /* SoC audio ops */
+  struct snd_soc_ops {
+	int (*startup)(struct snd_pcm_substream *);
+	void (*shutdown)(struct snd_pcm_substream *);
+	int (*hw_params)(struct snd_pcm_substream *, struct snd_pcm_hw_params *);
+	int (*hw_free)(struct snd_pcm_substream *);
+	int (*prepare)(struct snd_pcm_substream *);
+	int (*trigger)(struct snd_pcm_substream *, int);
+  };
+
+The platform driver exports its DMA functionality via struct
+snd_soc_platform_driver:-
+::
+
+  struct snd_soc_platform_driver {
+	char *name;
+
+	int (*probe)(struct platform_device *pdev);
+	int (*remove)(struct platform_device *pdev);
+	int (*suspend)(struct platform_device *pdev, struct snd_soc_cpu_dai *cpu_dai);
+	int (*resume)(struct platform_device *pdev, struct snd_soc_cpu_dai *cpu_dai);
+
+	/* pcm creation and destruction */
+	int (*pcm_new)(struct snd_card *, struct snd_soc_codec_dai *, struct snd_pcm *);
+	void (*pcm_free)(struct snd_pcm *);
+
+	/*
+	 * For platform caused delay reporting.
+	 * Optional.
+	 */
+	snd_pcm_sframes_t (*delay)(struct snd_pcm_substream *,
+		struct snd_soc_dai *);
+
+	/* platform stream ops */
+	struct snd_pcm_ops *pcm_ops;
+  };
+
+Please refer to the ALSA driver documentation for details of audio DMA.
+http://www.alsa-project.org/~iwai/writing-an-alsa-driver/
+
+An example DMA driver is soc/pxa/pxa2xx-pcm.c
+
+
+SoC DAI Drivers
+===============
+
+Each SoC DAI driver must provide the following features:-
+
+1. Digital audio interface (DAI) description
+2. Digital audio interface configuration
+3. PCM's description
+4. SYSCLK configuration
+5. Suspend and resume (optional)
+
+Please see codec.txt for a description of items 1 - 4.
+
+
+SoC DSP Drivers
+===============
+
+Each SoC DSP driver usually supplies the following features :-
+
+1. DAPM graph
+2. Mixer controls
+3. DMA IO to/from DSP buffers (if applicable)
+4. Definition of DSP front end (FE) PCM devices.
+
+Please see DPCM.txt for a description of item 4.
-- 
cgit v1.2.3-58-ga151


From d9641c9d63909ab452405f0a6a3fd6177a4a6b6d Mon Sep 17 00:00:00 2001
From: Takashi Iwai 
Date: Thu, 10 Nov 2016 22:23:02 +0100
Subject: ASoC: doc: ReSTize machine.txt

A simple conversion from a plain text file.

Acked-by: Mark Brown 
Signed-off-by: Takashi Iwai 
---
 Documentation/sound/alsa/soc/machine.txt | 93 ------------------------------
 Documentation/sound/soc/index.rst        |  1 +
 Documentation/sound/soc/machine.rst      | 97 ++++++++++++++++++++++++++++++++
 3 files changed, 98 insertions(+), 93 deletions(-)
 delete mode 100644 Documentation/sound/alsa/soc/machine.txt
 create mode 100644 Documentation/sound/soc/machine.rst

(limited to 'Documentation/sound')

diff --git a/Documentation/sound/alsa/soc/machine.txt b/Documentation/sound/alsa/soc/machine.txt
deleted file mode 100644
index 6bf2d2063b52..000000000000
--- a/Documentation/sound/alsa/soc/machine.txt
+++ /dev/null
@@ -1,93 +0,0 @@
-ASoC Machine Driver
-===================
-
-The ASoC machine (or board) driver is the code that glues together all the
-component drivers (e.g. codecs, platforms and DAIs). It also describes the
-relationships between each component which include audio paths, GPIOs,
-interrupts, clocking, jacks and voltage regulators.
-
-The machine driver can contain codec and platform specific code. It registers
-the audio subsystem with the kernel as a platform device and is represented by
-the following struct:-
-
-/* SoC machine */
-struct snd_soc_card {
-	char *name;
-
-	...
-
-	int (*probe)(struct platform_device *pdev);
-	int (*remove)(struct platform_device *pdev);
-
-	/* the pre and post PM functions are used to do any PM work before and
-	 * after the codec and DAIs do any PM work. */
-	int (*suspend_pre)(struct platform_device *pdev, pm_message_t state);
-	int (*suspend_post)(struct platform_device *pdev, pm_message_t state);
-	int (*resume_pre)(struct platform_device *pdev);
-	int (*resume_post)(struct platform_device *pdev);
-
-	...
-
-	/* CPU <--> Codec DAI links  */
-	struct snd_soc_dai_link *dai_link;
-	int num_links;
-
-	...
-};
-
-probe()/remove()
-----------------
-probe/remove are optional. Do any machine specific probe here.
-
-
-suspend()/resume()
-------------------
-The machine driver has pre and post versions of suspend and resume to take care
-of any machine audio tasks that have to be done before or after the codec, DAIs
-and DMA is suspended and resumed. Optional.
-
-
-Machine DAI Configuration
--------------------------
-The machine DAI configuration glues all the codec and CPU DAIs together. It can
-also be used to set up the DAI system clock and for any machine related DAI
-initialisation e.g. the machine audio map can be connected to the codec audio
-map, unconnected codec pins can be set as such.
-
-struct snd_soc_dai_link is used to set up each DAI in your machine. e.g.
-
-/* corgi digital audio interface glue - connects codec <--> CPU */
-static struct snd_soc_dai_link corgi_dai = {
-	.name = "WM8731",
-	.stream_name = "WM8731",
-	.cpu_dai_name = "pxa-is2-dai",
-	.codec_dai_name = "wm8731-hifi",
-	.platform_name = "pxa-pcm-audio",
-	.codec_name = "wm8713-codec.0-001a",
-	.init = corgi_wm8731_init,
-	.ops = &corgi_ops,
-};
-
-struct snd_soc_card then sets up the machine with its DAIs. e.g.
-
-/* corgi audio machine driver */
-static struct snd_soc_card snd_soc_corgi = {
-	.name = "Corgi",
-	.dai_link = &corgi_dai,
-	.num_links = 1,
-};
-
-
-Machine Power Map
------------------
-
-The machine driver can optionally extend the codec power map and to become an
-audio power map of the audio subsystem. This allows for automatic power up/down
-of speaker/HP amplifiers, etc. Codec pins can be connected to the machines jack
-sockets in the machine init function.
-
-
-Machine Controls
-----------------
-
-Machine specific audio mixer controls can be added in the DAI init function.
diff --git a/Documentation/sound/soc/index.rst b/Documentation/sound/soc/index.rst
index c5a55195bf4d..4ac3585e7dd1 100644
--- a/Documentation/sound/soc/index.rst
+++ b/Documentation/sound/soc/index.rst
@@ -12,3 +12,4 @@ The documentation is spilt into the following sections:-
    dai
    dapm
    platform
+   machine
diff --git a/Documentation/sound/soc/machine.rst b/Documentation/sound/soc/machine.rst
new file mode 100644
index 000000000000..515c9444deaf
--- /dev/null
+++ b/Documentation/sound/soc/machine.rst
@@ -0,0 +1,97 @@
+===================
+ASoC Machine Driver
+===================
+
+The ASoC machine (or board) driver is the code that glues together all the
+component drivers (e.g. codecs, platforms and DAIs). It also describes the
+relationships between each component which include audio paths, GPIOs,
+interrupts, clocking, jacks and voltage regulators.
+
+The machine driver can contain codec and platform specific code. It registers
+the audio subsystem with the kernel as a platform device and is represented by
+the following struct:-
+::
+
+  /* SoC machine */
+  struct snd_soc_card {
+	char *name;
+
+	...
+
+	int (*probe)(struct platform_device *pdev);
+	int (*remove)(struct platform_device *pdev);
+
+	/* the pre and post PM functions are used to do any PM work before and
+	 * after the codec and DAIs do any PM work. */
+	int (*suspend_pre)(struct platform_device *pdev, pm_message_t state);
+	int (*suspend_post)(struct platform_device *pdev, pm_message_t state);
+	int (*resume_pre)(struct platform_device *pdev);
+	int (*resume_post)(struct platform_device *pdev);
+
+	...
+
+	/* CPU <--> Codec DAI links  */
+	struct snd_soc_dai_link *dai_link;
+	int num_links;
+
+	...
+  };
+
+probe()/remove()
+----------------
+probe/remove are optional. Do any machine specific probe here.
+
+
+suspend()/resume()
+------------------
+The machine driver has pre and post versions of suspend and resume to take care
+of any machine audio tasks that have to be done before or after the codec, DAIs
+and DMA is suspended and resumed. Optional.
+
+
+Machine DAI Configuration
+-------------------------
+The machine DAI configuration glues all the codec and CPU DAIs together. It can
+also be used to set up the DAI system clock and for any machine related DAI
+initialisation e.g. the machine audio map can be connected to the codec audio
+map, unconnected codec pins can be set as such.
+
+struct snd_soc_dai_link is used to set up each DAI in your machine. e.g.
+::
+
+  /* corgi digital audio interface glue - connects codec <--> CPU */
+  static struct snd_soc_dai_link corgi_dai = {
+	.name = "WM8731",
+	.stream_name = "WM8731",
+	.cpu_dai_name = "pxa-is2-dai",
+	.codec_dai_name = "wm8731-hifi",
+	.platform_name = "pxa-pcm-audio",
+	.codec_name = "wm8713-codec.0-001a",
+	.init = corgi_wm8731_init,
+	.ops = &corgi_ops,
+  };
+
+struct snd_soc_card then sets up the machine with its DAIs. e.g.
+::
+
+  /* corgi audio machine driver */
+  static struct snd_soc_card snd_soc_corgi = {
+	.name = "Corgi",
+	.dai_link = &corgi_dai,
+	.num_links = 1,
+  };
+
+
+Machine Power Map
+-----------------
+
+The machine driver can optionally extend the codec power map and to become an
+audio power map of the audio subsystem. This allows for automatic power up/down
+of speaker/HP amplifiers, etc. Codec pins can be connected to the machines jack
+sockets in the machine init function.
+
+
+Machine Controls
+----------------
+
+Machine specific audio mixer controls can be added in the DAI init function.
-- 
cgit v1.2.3-58-ga151


From c9746eeafca3b5286a0355b895eb48af19d54d2d Mon Sep 17 00:00:00 2001
From: Takashi Iwai 
Date: Thu, 10 Nov 2016 22:23:58 +0100
Subject: ASoC: doc: ReSTize pops_clicks.txt

A simple conversion from a plain text file.
The file name was changed from "pops_clicks" to "pops-clicks" to align
with others.

Acked-by: Mark Brown 
Signed-off-by: Takashi Iwai 
---
 Documentation/sound/alsa/soc/pops_clicks.txt | 52 --------------------------
 Documentation/sound/soc/index.rst            |  1 +
 Documentation/sound/soc/pops-clicks.rst      | 55 ++++++++++++++++++++++++++++
 3 files changed, 56 insertions(+), 52 deletions(-)
 delete mode 100644 Documentation/sound/alsa/soc/pops_clicks.txt
 create mode 100644 Documentation/sound/soc/pops-clicks.rst

(limited to 'Documentation/sound')

diff --git a/Documentation/sound/alsa/soc/pops_clicks.txt b/Documentation/sound/alsa/soc/pops_clicks.txt
deleted file mode 100644
index e1e74daa4497..000000000000
--- a/Documentation/sound/alsa/soc/pops_clicks.txt
+++ /dev/null
@@ -1,52 +0,0 @@
-Audio Pops and Clicks
-=====================
-
-Pops and clicks are unwanted audio artifacts caused by the powering up and down
-of components within the audio subsystem. This is noticeable on PCs when an
-audio module is either loaded or unloaded (at module load time the sound card is
-powered up and causes a popping noise on the speakers).
-
-Pops and clicks can be more frequent on portable systems with DAPM. This is
-because the components within the subsystem are being dynamically powered
-depending on the audio usage and this can subsequently cause a small pop or
-click every time a component power state is changed.
-
-
-Minimising Playback Pops and Clicks
-===================================
-
-Playback pops in portable audio subsystems cannot be completely eliminated
-currently, however future audio codec hardware will have better pop and click
-suppression.  Pops can be reduced within playback by powering the audio
-components in a specific order. This order is different for startup and
-shutdown and follows some basic rules:-
-
- Startup Order :- DAC --> Mixers --> Output PGA --> Digital Unmute
-
- Shutdown Order :- Digital Mute --> Output PGA --> Mixers --> DAC
-
-This assumes that the codec PCM output path from the DAC is via a mixer and then
-a PGA (programmable gain amplifier) before being output to the speakers.
-
-
-Minimising Capture Pops and Clicks
-==================================
-
-Capture artifacts are somewhat easier to get rid as we can delay activating the
-ADC until all the pops have occurred. This follows similar power rules to
-playback in that components are powered in a sequence depending upon stream
-startup or shutdown.
-
- Startup Order - Input PGA --> Mixers --> ADC
-
- Shutdown Order - ADC --> Mixers --> Input PGA
-
-
-Zipper Noise
-============
-An unwanted zipper noise can occur within the audio playback or capture stream
-when a volume control is changed near its maximum gain value. The zipper noise
-is heard when the gain increase or decrease changes the mean audio signal
-amplitude too quickly. It can be minimised by enabling the zero cross setting
-for each volume control. The ZC forces the gain change to occur when the signal
-crosses the zero amplitude line.
diff --git a/Documentation/sound/soc/index.rst b/Documentation/sound/soc/index.rst
index 4ac3585e7dd1..4dd679927a91 100644
--- a/Documentation/sound/soc/index.rst
+++ b/Documentation/sound/soc/index.rst
@@ -13,3 +13,4 @@ The documentation is spilt into the following sections:-
    dapm
    platform
    machine
+   pops-clicks
diff --git a/Documentation/sound/soc/pops-clicks.rst b/Documentation/sound/soc/pops-clicks.rst
new file mode 100644
index 000000000000..de7eb2a6604a
--- /dev/null
+++ b/Documentation/sound/soc/pops-clicks.rst
@@ -0,0 +1,55 @@
+=====================
+Audio Pops and Clicks
+=====================
+
+Pops and clicks are unwanted audio artifacts caused by the powering up and down
+of components within the audio subsystem. This is noticeable on PCs when an
+audio module is either loaded or unloaded (at module load time the sound card is
+powered up and causes a popping noise on the speakers).
+
+Pops and clicks can be more frequent on portable systems with DAPM. This is
+because the components within the subsystem are being dynamically powered
+depending on the audio usage and this can subsequently cause a small pop or
+click every time a component power state is changed.
+
+
+Minimising Playback Pops and Clicks
+===================================
+
+Playback pops in portable audio subsystems cannot be completely eliminated
+currently, however future audio codec hardware will have better pop and click
+suppression.  Pops can be reduced within playback by powering the audio
+components in a specific order. This order is different for startup and
+shutdown and follows some basic rules:-
+::
+
+   Startup Order :- DAC --> Mixers --> Output PGA --> Digital Unmute
+  
+   Shutdown Order :- Digital Mute --> Output PGA --> Mixers --> DAC
+
+This assumes that the codec PCM output path from the DAC is via a mixer and then
+a PGA (programmable gain amplifier) before being output to the speakers.
+
+
+Minimising Capture Pops and Clicks
+==================================
+
+Capture artifacts are somewhat easier to get rid as we can delay activating the
+ADC until all the pops have occurred. This follows similar power rules to
+playback in that components are powered in a sequence depending upon stream
+startup or shutdown.
+::
+
+   Startup Order - Input PGA --> Mixers --> ADC
+  
+   Shutdown Order - ADC --> Mixers --> Input PGA
+
+
+Zipper Noise
+============
+An unwanted zipper noise can occur within the audio playback or capture stream
+when a volume control is changed near its maximum gain value. The zipper noise
+is heard when the gain increase or decrease changes the mean audio signal
+amplitude too quickly. It can be minimised by enabling the zero cross setting
+for each volume control. The ZC forces the gain change to occur when the signal
+crosses the zero amplitude line.
-- 
cgit v1.2.3-58-ga151


From fb3df950833c1e8c39a82313942e6375f3f498c6 Mon Sep 17 00:00:00 2001
From: Takashi Iwai 
Date: Thu, 10 Nov 2016 22:25:28 +0100
Subject: ASoC: doc: ReSTize clocking.txt

A simple conversion from a plain text file.

Acked-by: Mark Brown 
Signed-off-by: Takashi Iwai 
---
 Documentation/sound/alsa/soc/clocking.txt | 51 -------------------------------
 Documentation/sound/soc/clocking.rst      | 46 ++++++++++++++++++++++++++++
 Documentation/sound/soc/index.rst         |  1 +
 3 files changed, 47 insertions(+), 51 deletions(-)
 delete mode 100644 Documentation/sound/alsa/soc/clocking.txt
 create mode 100644 Documentation/sound/soc/clocking.rst

(limited to 'Documentation/sound')

diff --git a/Documentation/sound/alsa/soc/clocking.txt b/Documentation/sound/alsa/soc/clocking.txt
deleted file mode 100644
index b1300162e01c..000000000000
--- a/Documentation/sound/alsa/soc/clocking.txt
+++ /dev/null
@@ -1,51 +0,0 @@
-Audio Clocking
-==============
-
-This text describes the audio clocking terms in ASoC and digital audio in
-general. Note: Audio clocking can be complex!
-
-
-Master Clock
-------------
-
-Every audio subsystem is driven by a master clock (sometimes referred to as MCLK
-or SYSCLK). This audio master clock can be derived from a number of sources
-(e.g. crystal, PLL, CPU clock) and is responsible for producing the correct
-audio playback and capture sample rates.
-
-Some master clocks (e.g. PLLs and CPU based clocks) are configurable in that
-their speed can be altered by software (depending on the system use and to save
-power). Other master clocks are fixed at a set frequency (i.e. crystals).
-
-
-DAI Clocks
-----------
-The Digital Audio Interface is usually driven by a Bit Clock (often referred to
-as BCLK). This clock is used to drive the digital audio data across the link
-between the codec and CPU.
-
-The DAI also has a frame clock to signal the start of each audio frame. This
-clock is sometimes referred to as LRC (left right clock) or FRAME. This clock
-runs at exactly the sample rate (LRC = Rate).
-
-Bit Clock can be generated as follows:-
-
-BCLK = MCLK / x
-
- or
-
-BCLK = LRC * x
-
- or
-
-BCLK = LRC * Channels * Word Size
-
-This relationship depends on the codec or SoC CPU in particular. In general
-it is best to configure BCLK to the lowest possible speed (depending on your
-rate, number of channels and word size) to save on power.
-
-It is also desirable to use the codec (if possible) to drive (or master) the
-audio clocks as it usually gives more accurate sample rates than the CPU.
-
-
-
diff --git a/Documentation/sound/soc/clocking.rst b/Documentation/sound/soc/clocking.rst
new file mode 100644
index 000000000000..32122d6877a3
--- /dev/null
+++ b/Documentation/sound/soc/clocking.rst
@@ -0,0 +1,46 @@
+==============
+Audio Clocking
+==============
+
+This text describes the audio clocking terms in ASoC and digital audio in
+general. Note: Audio clocking can be complex!
+
+
+Master Clock
+------------
+
+Every audio subsystem is driven by a master clock (sometimes referred to as MCLK
+or SYSCLK). This audio master clock can be derived from a number of sources
+(e.g. crystal, PLL, CPU clock) and is responsible for producing the correct
+audio playback and capture sample rates.
+
+Some master clocks (e.g. PLLs and CPU based clocks) are configurable in that
+their speed can be altered by software (depending on the system use and to save
+power). Other master clocks are fixed at a set frequency (i.e. crystals).
+
+
+DAI Clocks
+----------
+The Digital Audio Interface is usually driven by a Bit Clock (often referred to
+as BCLK). This clock is used to drive the digital audio data across the link
+between the codec and CPU.
+
+The DAI also has a frame clock to signal the start of each audio frame. This
+clock is sometimes referred to as LRC (left right clock) or FRAME. This clock
+runs at exactly the sample rate (LRC = Rate).
+
+Bit Clock can be generated as follows:-
+
+- BCLK = MCLK / x, or
+- BCLK = LRC * x, or
+- BCLK = LRC * Channels * Word Size
+
+This relationship depends on the codec or SoC CPU in particular. In general
+it is best to configure BCLK to the lowest possible speed (depending on your
+rate, number of channels and word size) to save on power.
+
+It is also desirable to use the codec (if possible) to drive (or master) the
+audio clocks as it usually gives more accurate sample rates than the CPU.
+
+
+
diff --git a/Documentation/sound/soc/index.rst b/Documentation/sound/soc/index.rst
index 4dd679927a91..0055abe16e7e 100644
--- a/Documentation/sound/soc/index.rst
+++ b/Documentation/sound/soc/index.rst
@@ -14,3 +14,4 @@ The documentation is spilt into the following sections:-
    platform
    machine
    pops-clicks
+   clocking
-- 
cgit v1.2.3-58-ga151


From 8155258a7d7600e7b92c6193cf23a11e281a9b0b Mon Sep 17 00:00:00 2001
From: Takashi Iwai 
Date: Thu, 10 Nov 2016 22:26:58 +0100
Subject: ASoC: doc: ReSTize jack.txt

A simple conversion from a plain text file.

Acked-by: Mark Brown 
Signed-off-by: Takashi Iwai 
---
 Documentation/sound/alsa/soc/jack.txt | 71 ----------------------------------
 Documentation/sound/soc/index.rst     |  1 +
 Documentation/sound/soc/jack.rst      | 72 +++++++++++++++++++++++++++++++++++
 3 files changed, 73 insertions(+), 71 deletions(-)
 delete mode 100644 Documentation/sound/alsa/soc/jack.txt
 create mode 100644 Documentation/sound/soc/jack.rst

(limited to 'Documentation/sound')

diff --git a/Documentation/sound/alsa/soc/jack.txt b/Documentation/sound/alsa/soc/jack.txt
deleted file mode 100644
index fcf82a417293..000000000000
--- a/Documentation/sound/alsa/soc/jack.txt
+++ /dev/null
@@ -1,71 +0,0 @@
-ASoC jack detection
-===================
-
-ALSA has a standard API for representing physical jacks to user space,
-the kernel side of which can be seen in include/sound/jack.h.  ASoC
-provides a version of this API adding two additional features:
-
- - It allows more than one jack detection method to work together on one
-   user visible jack.  In embedded systems it is common for multiple
-   to be present on a single jack but handled by separate bits of
-   hardware.
-
- - Integration with DAPM, allowing DAPM endpoints to be updated
-   automatically based on the detected jack status (eg, turning off the
-   headphone outputs if no headphones are present).
-
-This is done by splitting the jacks up into three things working
-together: the jack itself represented by a struct snd_soc_jack, sets of
-snd_soc_jack_pins representing DAPM endpoints to update and blocks of
-code providing jack reporting mechanisms.
-
-For example, a system may have a stereo headset jack with two reporting
-mechanisms, one for the headphone and one for the microphone.  Some
-systems won't be able to use their speaker output while a headphone is
-connected and so will want to make sure to update both speaker and
-headphone when the headphone jack status changes.
-
-The jack - struct snd_soc_jack
-==============================
-
-This represents a physical jack on the system and is what is visible to
-user space.  The jack itself is completely passive, it is set up by the
-machine driver and updated by jack detection methods.
-
-Jacks are created by the machine driver calling snd_soc_jack_new().
-
-snd_soc_jack_pin
-================
-
-These represent a DAPM pin to update depending on some of the status
-bits supported by the jack.  Each snd_soc_jack has zero or more of these
-which are updated automatically.  They are created by the machine driver
-and associated with the jack using snd_soc_jack_add_pins().  The status
-of the endpoint may configured to be the opposite of the jack status if
-required (eg, enabling a built in microphone if a microphone is not
-connected via a jack).
-
-Jack detection methods
-======================
-
-Actual jack detection is done by code which is able to monitor some
-input to the system and update a jack by calling snd_soc_jack_report(),
-specifying a subset of bits to update.  The jack detection code should
-be set up by the machine driver, taking configuration for the jack to
-update and the set of things to report when the jack is connected.
-
-Often this is done based on the status of a GPIO - a handler for this is
-provided by the snd_soc_jack_add_gpio() function.  Other methods are
-also available, for example integrated into CODECs.  One example of
-CODEC integrated jack detection can be see in the WM8350 driver.
-
-Each jack may have multiple reporting mechanisms, though it will need at
-least one to be useful.
-
-Machine drivers
-===============
-
-These are all hooked together by the machine driver depending on the
-system hardware.  The machine driver will set up the snd_soc_jack and
-the list of pins to update then set up one or more jack detection
-mechanisms to update that jack based on their current status.
diff --git a/Documentation/sound/soc/index.rst b/Documentation/sound/soc/index.rst
index 0055abe16e7e..85ec51764e83 100644
--- a/Documentation/sound/soc/index.rst
+++ b/Documentation/sound/soc/index.rst
@@ -15,3 +15,4 @@ The documentation is spilt into the following sections:-
    machine
    pops-clicks
    clocking
+   jack
diff --git a/Documentation/sound/soc/jack.rst b/Documentation/sound/soc/jack.rst
new file mode 100644
index 000000000000..644b99ecba35
--- /dev/null
+++ b/Documentation/sound/soc/jack.rst
@@ -0,0 +1,72 @@
+===================
+ASoC jack detection
+===================
+
+ALSA has a standard API for representing physical jacks to user space,
+the kernel side of which can be seen in include/sound/jack.h.  ASoC
+provides a version of this API adding two additional features:
+
+ - It allows more than one jack detection method to work together on one
+   user visible jack.  In embedded systems it is common for multiple
+   to be present on a single jack but handled by separate bits of
+   hardware.
+
+ - Integration with DAPM, allowing DAPM endpoints to be updated
+   automatically based on the detected jack status (eg, turning off the
+   headphone outputs if no headphones are present).
+
+This is done by splitting the jacks up into three things working
+together: the jack itself represented by a struct snd_soc_jack, sets of
+snd_soc_jack_pins representing DAPM endpoints to update and blocks of
+code providing jack reporting mechanisms.
+
+For example, a system may have a stereo headset jack with two reporting
+mechanisms, one for the headphone and one for the microphone.  Some
+systems won't be able to use their speaker output while a headphone is
+connected and so will want to make sure to update both speaker and
+headphone when the headphone jack status changes.
+
+The jack - struct snd_soc_jack
+==============================
+
+This represents a physical jack on the system and is what is visible to
+user space.  The jack itself is completely passive, it is set up by the
+machine driver and updated by jack detection methods.
+
+Jacks are created by the machine driver calling snd_soc_jack_new().
+
+snd_soc_jack_pin
+================
+
+These represent a DAPM pin to update depending on some of the status
+bits supported by the jack.  Each snd_soc_jack has zero or more of these
+which are updated automatically.  They are created by the machine driver
+and associated with the jack using snd_soc_jack_add_pins().  The status
+of the endpoint may configured to be the opposite of the jack status if
+required (eg, enabling a built in microphone if a microphone is not
+connected via a jack).
+
+Jack detection methods
+======================
+
+Actual jack detection is done by code which is able to monitor some
+input to the system and update a jack by calling snd_soc_jack_report(),
+specifying a subset of bits to update.  The jack detection code should
+be set up by the machine driver, taking configuration for the jack to
+update and the set of things to report when the jack is connected.
+
+Often this is done based on the status of a GPIO - a handler for this is
+provided by the snd_soc_jack_add_gpio() function.  Other methods are
+also available, for example integrated into CODECs.  One example of
+CODEC integrated jack detection can be see in the WM8350 driver.
+
+Each jack may have multiple reporting mechanisms, though it will need at
+least one to be useful.
+
+Machine drivers
+===============
+
+These are all hooked together by the machine driver depending on the
+system hardware.  The machine driver will set up the snd_soc_jack and
+the list of pins to update then set up one or more jack detection
+mechanisms to update that jack based on their current status.
-- 
cgit v1.2.3-58-ga151


From 40433cd34e280bd1a56f54a3898e86863814c824 Mon Sep 17 00:00:00 2001
From: Takashi Iwai 
Date: Thu, 10 Nov 2016 22:29:49 +0100
Subject: ASoC: doc: ReSTize DPCM.txt

A simple conversion from a plain text file.
The file name was renamed to lower letters to align with others.

Acked-by: Mark Brown 
Signed-off-by: Takashi Iwai 
---
 Documentation/sound/alsa/soc/DPCM.txt | 380 --------------------------------
 Documentation/sound/soc/dpcm.rst      | 392 ++++++++++++++++++++++++++++++++++
 Documentation/sound/soc/index.rst     |   1 +
 3 files changed, 393 insertions(+), 380 deletions(-)
 delete mode 100644 Documentation/sound/alsa/soc/DPCM.txt
 create mode 100644 Documentation/sound/soc/dpcm.rst

(limited to 'Documentation/sound')

diff --git a/Documentation/sound/alsa/soc/DPCM.txt b/Documentation/sound/alsa/soc/DPCM.txt
deleted file mode 100644
index 0110180b7ac6..000000000000
--- a/Documentation/sound/alsa/soc/DPCM.txt
+++ /dev/null
@@ -1,380 +0,0 @@
-Dynamic PCM
-===========
-
-1. Description
-==============
-
-Dynamic PCM allows an ALSA PCM device to digitally route its PCM audio to
-various digital endpoints during the PCM stream runtime. e.g. PCM0 can route
-digital audio to I2S DAI0, I2S DAI1 or PDM DAI2. This is useful for on SoC DSP
-drivers that expose several ALSA PCMs and can route to multiple DAIs.
-
-The DPCM runtime routing is determined by the ALSA mixer settings in the same
-way as the analog signal is routed in an ASoC codec driver. DPCM uses a DAPM
-graph representing the DSP internal audio paths and uses the mixer settings to
-determine the patch used by each ALSA PCM.
-
-DPCM re-uses all the existing component codec, platform and DAI drivers without
-any modifications.
-
-
-Phone Audio System with SoC based DSP
--------------------------------------
-
-Consider the following phone audio subsystem. This will be used in this
-document for all examples :-
-
-| Front End PCMs    |  SoC DSP  | Back End DAIs | Audio devices |
-
-                    *************
-PCM0 <------------> *           * <----DAI0-----> Codec Headset
-                    *           *
-PCM1 <------------> *           * <----DAI1-----> Codec Speakers
-                    *   DSP     *
-PCM2 <------------> *           * <----DAI2-----> MODEM
-                    *           *
-PCM3 <------------> *           * <----DAI3-----> BT
-                    *           *
-                    *           * <----DAI4-----> DMIC
-                    *           *
-                    *           * <----DAI5-----> FM
-                    *************
-
-This diagram shows a simple smart phone audio subsystem. It supports Bluetooth,
-FM digital radio, Speakers, Headset Jack, digital microphones and cellular
-modem. This sound card exposes 4 DSP front end (FE) ALSA PCM devices and
-supports 6 back end (BE) DAIs. Each FE PCM can digitally route audio data to any
-of the BE DAIs. The FE PCM devices can also route audio to more than 1 BE DAI.
-
-
-
-Example - DPCM Switching playback from DAI0 to DAI1
----------------------------------------------------
-
-Audio is being played to the Headset. After a while the user removes the headset
-and audio continues playing on the speakers.
-
-Playback on PCM0 to Headset would look like :-
-
-                    *************
-PCM0 <============> *           * <====DAI0=====> Codec Headset
-                    *           *
-PCM1 <------------> *           * <----DAI1-----> Codec Speakers
-                    *   DSP     *
-PCM2 <------------> *           * <----DAI2-----> MODEM
-                    *           *
-PCM3 <------------> *           * <----DAI3-----> BT
-                    *           *
-                    *           * <----DAI4-----> DMIC
-                    *           *
-                    *           * <----DAI5-----> FM
-                    *************
-
-The headset is removed from the jack by user so the speakers must now be used :-
-
-                    *************
-PCM0 <============> *           * <----DAI0-----> Codec Headset
-                    *           *
-PCM1 <------------> *           * <====DAI1=====> Codec Speakers
-                    *   DSP     *
-PCM2 <------------> *           * <----DAI2-----> MODEM
-                    *           *
-PCM3 <------------> *           * <----DAI3-----> BT
-                    *           *
-                    *           * <----DAI4-----> DMIC
-                    *           *
-                    *           * <----DAI5-----> FM
-                    *************
-
-The audio driver processes this as follows :-
-
- 1) Machine driver receives Jack removal event.
-
- 2) Machine driver OR audio HAL disables the Headset path.
-
- 3) DPCM runs the PCM trigger(stop), hw_free(), shutdown() operations on DAI0
-    for headset since the path is now disabled.
-
- 4) Machine driver or audio HAL enables the speaker path.
-
- 5) DPCM runs the PCM ops for startup(), hw_params(), prepapre() and
-    trigger(start) for DAI1 Speakers since the path is enabled.
-
-In this example, the machine driver or userspace audio HAL can alter the routing
-and then DPCM will take care of managing the DAI PCM operations to either bring
-the link up or down. Audio playback does not stop during this transition.
-
-
-
-DPCM machine driver
-===================
-
-The DPCM enabled ASoC machine driver is similar to normal machine drivers
-except that we also have to :-
-
- 1) Define the FE and BE DAI links.
-
- 2) Define any FE/BE PCM operations.
-
- 3) Define widget graph connections.
-
-
-1 FE and BE DAI links
----------------------
-
-| Front End PCMs    |  SoC DSP  | Back End DAIs | Audio devices |
-
-                    *************
-PCM0 <------------> *           * <----DAI0-----> Codec Headset
-                    *           *
-PCM1 <------------> *           * <----DAI1-----> Codec Speakers
-                    *   DSP     *
-PCM2 <------------> *           * <----DAI2-----> MODEM
-                    *           *
-PCM3 <------------> *           * <----DAI3-----> BT
-                    *           *
-                    *           * <----DAI4-----> DMIC
-                    *           *
-                    *           * <----DAI5-----> FM
-                    *************
-
-For the example above we have to define 4 FE DAI links and 6 BE DAI links. The
-FE DAI links are defined as follows :-
-
-static struct snd_soc_dai_link machine_dais[] = {
-	{
-		.name = "PCM0 System",
-		.stream_name = "System Playback",
-		.cpu_dai_name = "System Pin",
-		.platform_name = "dsp-audio",
-		.codec_name = "snd-soc-dummy",
-		.codec_dai_name = "snd-soc-dummy-dai",
-		.dynamic = 1,
-		.trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
-		.dpcm_playback = 1,
-	},
-	.....< other FE and BE DAI links here >
-};
-
-This FE DAI link is pretty similar to a regular DAI link except that we also
-set the DAI link to a DPCM FE with the "dynamic = 1". The supported FE stream
-directions should also be set with the "dpcm_playback" and "dpcm_capture"
-flags. There is also an option to specify the ordering of the trigger call for
-each FE. This allows the ASoC core to trigger the DSP before or after the other
-components (as some DSPs have strong requirements for the ordering DAI/DSP
-start and stop sequences).
-
-The FE DAI above sets the codec and code DAIs to dummy devices since the BE is
-dynamic and will change depending on runtime config.
-
-The BE DAIs are configured as follows :-
-
-static struct snd_soc_dai_link machine_dais[] = {
-	.....< FE DAI links here >
-	{
-		.name = "Codec Headset",
-		.cpu_dai_name = "ssp-dai.0",
-		.platform_name = "snd-soc-dummy",
-		.no_pcm = 1,
-		.codec_name = "rt5640.0-001c",
-		.codec_dai_name = "rt5640-aif1",
-		.ignore_suspend = 1,
-		.ignore_pmdown_time = 1,
-		.be_hw_params_fixup = hswult_ssp0_fixup,
-		.ops = &haswell_ops,
-		.dpcm_playback = 1,
-		.dpcm_capture = 1,
-	},
-	.....< other BE DAI links here >
-};
-
-This BE DAI link connects DAI0 to the codec (in this case RT5460 AIF1). It sets
-the "no_pcm" flag to mark it has a BE and sets flags for supported stream
-directions using "dpcm_playback" and "dpcm_capture" above.
-
-The BE has also flags set for ignoring suspend and PM down time. This allows
-the BE to work in a hostless mode where the host CPU is not transferring data
-like a BT phone call :-
-
-                    *************
-PCM0 <------------> *           * <----DAI0-----> Codec Headset
-                    *           *
-PCM1 <------------> *           * <----DAI1-----> Codec Speakers
-                    *   DSP     *
-PCM2 <------------> *           * <====DAI2=====> MODEM
-                    *           *
-PCM3 <------------> *           * <====DAI3=====> BT
-                    *           *
-                    *           * <----DAI4-----> DMIC
-                    *           *
-                    *           * <----DAI5-----> FM
-                    *************
-
-This allows the host CPU to sleep whilst the DSP, MODEM DAI and the BT DAI are
-still in operation.
-
-A BE DAI link can also set the codec to a dummy device if the code is a device
-that is managed externally.
-
-Likewise a BE DAI can also set a dummy cpu DAI if the CPU DAI is managed by the
-DSP firmware.
-
-
-2 FE/BE PCM operations
-----------------------
-
-The BE above also exports some PCM operations and a "fixup" callback. The fixup
-callback is used by the machine driver to (re)configure the DAI based upon the
-FE hw params. i.e. the DSP may perform SRC or ASRC from the FE to BE.
-
-e.g. DSP converts all FE hw params to run at fixed rate of 48k, 16bit, stereo for
-DAI0. This means all FE hw_params have to be fixed in the machine driver for
-DAI0 so that the DAI is running at desired configuration regardless of the FE
-configuration.
-
-static int dai0_fixup(struct snd_soc_pcm_runtime *rtd,
-			struct snd_pcm_hw_params *params)
-{
-	struct snd_interval *rate = hw_param_interval(params,
-			SNDRV_PCM_HW_PARAM_RATE);
-	struct snd_interval *channels = hw_param_interval(params,
-						SNDRV_PCM_HW_PARAM_CHANNELS);
-
-	/* The DSP will covert the FE rate to 48k, stereo */
-	rate->min = rate->max = 48000;
-	channels->min = channels->max = 2;
-
-	/* set DAI0 to 16 bit */
-	snd_mask_set(¶ms->masks[SNDRV_PCM_HW_PARAM_FORMAT -
-				    SNDRV_PCM_HW_PARAM_FIRST_MASK],
-				    SNDRV_PCM_FORMAT_S16_LE);
-	return 0;
-}
-
-The other PCM operation are the same as for regular DAI links. Use as necessary.
-
-
-3 Widget graph connections
---------------------------
-
-The BE DAI links will normally be connected to the graph at initialisation time
-by the ASoC DAPM core. However, if the BE codec or BE DAI is a dummy then this
-has to be set explicitly in the driver :-
-
-/* BE for codec Headset -  DAI0 is dummy and managed by DSP FW */
-{"DAI0 CODEC IN", NULL, "AIF1 Capture"},
-{"AIF1 Playback", NULL, "DAI0 CODEC OUT"},
-
-
-Writing a DPCM DSP driver
-=========================
-
-The DPCM DSP driver looks much like a standard platform class ASoC driver
-combined with elements from a codec class driver. A DSP platform driver must
-implement :-
-
- 1) Front End PCM DAIs - i.e. struct snd_soc_dai_driver.
-
- 2) DAPM graph showing DSP audio routing from FE DAIs to BEs.
-
- 3) DAPM widgets from DSP graph.
-
- 4) Mixers for gains, routing, etc.
-
- 5) DMA configuration.
-
- 6) BE AIF widgets.
-
-Items 6 is important for routing the audio outside of the DSP. AIF need to be
-defined for each BE and each stream direction. e.g for BE DAI0 above we would
-have :-
-
-SND_SOC_DAPM_AIF_IN("DAI0 RX", NULL, 0, SND_SOC_NOPM, 0, 0),
-SND_SOC_DAPM_AIF_OUT("DAI0 TX", NULL, 0, SND_SOC_NOPM, 0, 0),
-
-The BE AIF are used to connect the DSP graph to the graphs for the other
-component drivers (e.g. codec graph).
-
-
-Hostless PCM streams
-====================
-
-A hostless PCM stream is a stream that is not routed through the host CPU. An
-example of this would be a phone call from handset to modem.
-
-
-                    *************
-PCM0 <------------> *           * <----DAI0-----> Codec Headset
-                    *           *
-PCM1 <------------> *           * <====DAI1=====> Codec Speakers/Mic
-                    *   DSP     *
-PCM2 <------------> *           * <====DAI2=====> MODEM
-                    *           *
-PCM3 <------------> *           * <----DAI3-----> BT
-                    *           *
-                    *           * <----DAI4-----> DMIC
-                    *           *
-                    *           * <----DAI5-----> FM
-                    *************
-
-In this case the PCM data is routed via the DSP. The host CPU in this use case
-is only used for control and can sleep during the runtime of the stream.
-
-The host can control the hostless link either by :-
-
- 1) Configuring the link as a CODEC <-> CODEC style link. In this case the link
-    is enabled or disabled by the state of the DAPM graph. This usually means
-    there is a mixer control that can be used to connect or disconnect the path
-    between both DAIs.
-
- 2) Hostless FE. This FE has a virtual connection to the BE DAI links on the DAPM
-    graph. Control is then carried out by the FE as regular PCM operations.
-    This method gives more control over the DAI links, but requires much more
-    userspace code to control the link. Its recommended to use CODEC<->CODEC
-    unless your HW needs more fine grained sequencing of the PCM ops.
-
-
-CODEC <-> CODEC link
---------------------
-
-This DAI link is enabled when DAPM detects a valid path within the DAPM graph.
-The machine driver sets some additional parameters to the DAI link i.e.
-
-static const struct snd_soc_pcm_stream dai_params = {
-	.formats = SNDRV_PCM_FMTBIT_S32_LE,
-	.rate_min = 8000,
-	.rate_max = 8000,
-	.channels_min = 2,
-	.channels_max = 2,
-};
-
-static struct snd_soc_dai_link dais[] = {
-	< ... more DAI links above ... >
-	{
-		.name = "MODEM",
-		.stream_name = "MODEM",
-		.cpu_dai_name = "dai2",
-		.codec_dai_name = "modem-aif1",
-		.codec_name = "modem",
-		.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
-				| SND_SOC_DAIFMT_CBM_CFM,
-		.params = &dai_params,
-	}
-	< ... more DAI links here ... >
-
-These parameters are used to configure the DAI hw_params() when DAPM detects a
-valid path and then calls the PCM operations to start the link. DAPM will also
-call the appropriate PCM operations to disable the DAI when the path is no
-longer valid.
-
-
-Hostless FE
------------
-
-The DAI link(s) are enabled by a FE that does not read or write any PCM data.
-This means creating a new FE that is connected with a virtual path to both
-DAI links. The DAI links will be started when the FE PCM is started and stopped
-when the FE PCM is stopped. Note that the FE PCM cannot read or write data in
-this configuration.
-
-
diff --git a/Documentation/sound/soc/dpcm.rst b/Documentation/sound/soc/dpcm.rst
new file mode 100644
index 000000000000..395e5a516282
--- /dev/null
+++ b/Documentation/sound/soc/dpcm.rst
@@ -0,0 +1,392 @@
+===========
+Dynamic PCM
+===========
+
+Description
+===========
+
+Dynamic PCM allows an ALSA PCM device to digitally route its PCM audio to
+various digital endpoints during the PCM stream runtime. e.g. PCM0 can route
+digital audio to I2S DAI0, I2S DAI1 or PDM DAI2. This is useful for on SoC DSP
+drivers that expose several ALSA PCMs and can route to multiple DAIs.
+
+The DPCM runtime routing is determined by the ALSA mixer settings in the same
+way as the analog signal is routed in an ASoC codec driver. DPCM uses a DAPM
+graph representing the DSP internal audio paths and uses the mixer settings to
+determine the patch used by each ALSA PCM.
+
+DPCM re-uses all the existing component codec, platform and DAI drivers without
+any modifications.
+
+
+Phone Audio System with SoC based DSP
+-------------------------------------
+
+Consider the following phone audio subsystem. This will be used in this
+document for all examples :-
+::
+
+  | Front End PCMs    |  SoC DSP  | Back End DAIs | Audio devices |
+  
+                      *************
+  PCM0 <------------> *           * <----DAI0-----> Codec Headset
+                      *           *
+  PCM1 <------------> *           * <----DAI1-----> Codec Speakers
+                      *   DSP     *
+  PCM2 <------------> *           * <----DAI2-----> MODEM
+                      *           *
+  PCM3 <------------> *           * <----DAI3-----> BT
+                      *           *
+                      *           * <----DAI4-----> DMIC
+                      *           *
+                      *           * <----DAI5-----> FM
+                      *************
+
+This diagram shows a simple smart phone audio subsystem. It supports Bluetooth,
+FM digital radio, Speakers, Headset Jack, digital microphones and cellular
+modem. This sound card exposes 4 DSP front end (FE) ALSA PCM devices and
+supports 6 back end (BE) DAIs. Each FE PCM can digitally route audio data to any
+of the BE DAIs. The FE PCM devices can also route audio to more than 1 BE DAI.
+
+
+
+Example - DPCM Switching playback from DAI0 to DAI1
+---------------------------------------------------
+
+Audio is being played to the Headset. After a while the user removes the headset
+and audio continues playing on the speakers.
+
+Playback on PCM0 to Headset would look like :-
+::
+
+                      *************
+  PCM0 <============> *           * <====DAI0=====> Codec Headset
+                      *           *
+  PCM1 <------------> *           * <----DAI1-----> Codec Speakers
+                      *   DSP     *
+  PCM2 <------------> *           * <----DAI2-----> MODEM
+                      *           *
+  PCM3 <------------> *           * <----DAI3-----> BT
+                      *           *
+                      *           * <----DAI4-----> DMIC
+                      *           *
+                      *           * <----DAI5-----> FM
+                      *************
+
+The headset is removed from the jack by user so the speakers must now be used :-
+::
+
+                      *************
+  PCM0 <============> *           * <----DAI0-----> Codec Headset
+                      *           *
+  PCM1 <------------> *           * <====DAI1=====> Codec Speakers
+                      *   DSP     *
+  PCM2 <------------> *           * <----DAI2-----> MODEM
+                      *           *
+  PCM3 <------------> *           * <----DAI3-----> BT
+                      *           *
+                      *           * <----DAI4-----> DMIC
+                      *           *
+                      *           * <----DAI5-----> FM
+                      *************
+
+The audio driver processes this as follows :-
+
+1. Machine driver receives Jack removal event.
+
+2. Machine driver OR audio HAL disables the Headset path.
+
+3. DPCM runs the PCM trigger(stop), hw_free(), shutdown() operations on DAI0
+   for headset since the path is now disabled.
+
+4. Machine driver or audio HAL enables the speaker path.
+
+5. DPCM runs the PCM ops for startup(), hw_params(), prepapre() and
+   trigger(start) for DAI1 Speakers since the path is enabled.
+
+In this example, the machine driver or userspace audio HAL can alter the routing
+and then DPCM will take care of managing the DAI PCM operations to either bring
+the link up or down. Audio playback does not stop during this transition.
+
+
+
+DPCM machine driver
+===================
+
+The DPCM enabled ASoC machine driver is similar to normal machine drivers
+except that we also have to :-
+
+1. Define the FE and BE DAI links.
+
+2. Define any FE/BE PCM operations.
+
+3. Define widget graph connections.
+
+
+FE and BE DAI links
+-------------------
+::
+
+  | Front End PCMs    |  SoC DSP  | Back End DAIs | Audio devices |
+  
+                      *************
+  PCM0 <------------> *           * <----DAI0-----> Codec Headset
+                      *           *
+  PCM1 <------------> *           * <----DAI1-----> Codec Speakers
+                      *   DSP     *
+  PCM2 <------------> *           * <----DAI2-----> MODEM
+                      *           *
+  PCM3 <------------> *           * <----DAI3-----> BT
+                      *           *
+                      *           * <----DAI4-----> DMIC
+                      *           *
+                      *           * <----DAI5-----> FM
+                      *************
+
+For the example above we have to define 4 FE DAI links and 6 BE DAI links. The
+FE DAI links are defined as follows :-
+::
+
+  static struct snd_soc_dai_link machine_dais[] = {
+	{
+		.name = "PCM0 System",
+		.stream_name = "System Playback",
+		.cpu_dai_name = "System Pin",
+		.platform_name = "dsp-audio",
+		.codec_name = "snd-soc-dummy",
+		.codec_dai_name = "snd-soc-dummy-dai",
+		.dynamic = 1,
+		.trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+		.dpcm_playback = 1,
+	},
+	.....< other FE and BE DAI links here >
+  };
+
+This FE DAI link is pretty similar to a regular DAI link except that we also
+set the DAI link to a DPCM FE with the ``dynamic = 1``. The supported FE stream
+directions should also be set with the ``dpcm_playback`` and ``dpcm_capture``
+flags. There is also an option to specify the ordering of the trigger call for
+each FE. This allows the ASoC core to trigger the DSP before or after the other
+components (as some DSPs have strong requirements for the ordering DAI/DSP
+start and stop sequences).
+
+The FE DAI above sets the codec and code DAIs to dummy devices since the BE is
+dynamic and will change depending on runtime config.
+
+The BE DAIs are configured as follows :-
+::
+
+  static struct snd_soc_dai_link machine_dais[] = {
+	.....< FE DAI links here >
+	{
+		.name = "Codec Headset",
+		.cpu_dai_name = "ssp-dai.0",
+		.platform_name = "snd-soc-dummy",
+		.no_pcm = 1,
+		.codec_name = "rt5640.0-001c",
+		.codec_dai_name = "rt5640-aif1",
+		.ignore_suspend = 1,
+		.ignore_pmdown_time = 1,
+		.be_hw_params_fixup = hswult_ssp0_fixup,
+		.ops = &haswell_ops,
+		.dpcm_playback = 1,
+		.dpcm_capture = 1,
+	},
+	.....< other BE DAI links here >
+  };
+
+This BE DAI link connects DAI0 to the codec (in this case RT5460 AIF1). It sets
+the ``no_pcm`` flag to mark it has a BE and sets flags for supported stream
+directions using ``dpcm_playback`` and ``dpcm_capture`` above.
+
+The BE has also flags set for ignoring suspend and PM down time. This allows
+the BE to work in a hostless mode where the host CPU is not transferring data
+like a BT phone call :-
+::
+
+                      *************
+  PCM0 <------------> *           * <----DAI0-----> Codec Headset
+                      *           *
+  PCM1 <------------> *           * <----DAI1-----> Codec Speakers
+                      *   DSP     *
+  PCM2 <------------> *           * <====DAI2=====> MODEM
+                      *           *
+  PCM3 <------------> *           * <====DAI3=====> BT
+                      *           *
+                      *           * <----DAI4-----> DMIC
+                      *           *
+                      *           * <----DAI5-----> FM
+                      *************
+
+This allows the host CPU to sleep whilst the DSP, MODEM DAI and the BT DAI are
+still in operation.
+
+A BE DAI link can also set the codec to a dummy device if the code is a device
+that is managed externally.
+
+Likewise a BE DAI can also set a dummy cpu DAI if the CPU DAI is managed by the
+DSP firmware.
+
+
+FE/BE PCM operations
+--------------------
+
+The BE above also exports some PCM operations and a ``fixup`` callback. The fixup
+callback is used by the machine driver to (re)configure the DAI based upon the
+FE hw params. i.e. the DSP may perform SRC or ASRC from the FE to BE.
+
+e.g. DSP converts all FE hw params to run at fixed rate of 48k, 16bit, stereo for
+DAI0. This means all FE hw_params have to be fixed in the machine driver for
+DAI0 so that the DAI is running at desired configuration regardless of the FE
+configuration.
+::
+
+  static int dai0_fixup(struct snd_soc_pcm_runtime *rtd,
+			struct snd_pcm_hw_params *params)
+  {
+	struct snd_interval *rate = hw_param_interval(params,
+			SNDRV_PCM_HW_PARAM_RATE);
+	struct snd_interval *channels = hw_param_interval(params,
+						SNDRV_PCM_HW_PARAM_CHANNELS);
+
+	/* The DSP will covert the FE rate to 48k, stereo */
+	rate->min = rate->max = 48000;
+	channels->min = channels->max = 2;
+
+	/* set DAI0 to 16 bit */
+	snd_mask_set(¶ms->masks[SNDRV_PCM_HW_PARAM_FORMAT -
+				    SNDRV_PCM_HW_PARAM_FIRST_MASK],
+				    SNDRV_PCM_FORMAT_S16_LE);
+	return 0;
+  }
+
+The other PCM operation are the same as for regular DAI links. Use as necessary.
+
+
+Widget graph connections
+------------------------
+
+The BE DAI links will normally be connected to the graph at initialisation time
+by the ASoC DAPM core. However, if the BE codec or BE DAI is a dummy then this
+has to be set explicitly in the driver :-
+::
+
+  /* BE for codec Headset -  DAI0 is dummy and managed by DSP FW */
+  {"DAI0 CODEC IN", NULL, "AIF1 Capture"},
+  {"AIF1 Playback", NULL, "DAI0 CODEC OUT"},
+
+
+Writing a DPCM DSP driver
+=========================
+
+The DPCM DSP driver looks much like a standard platform class ASoC driver
+combined with elements from a codec class driver. A DSP platform driver must
+implement :-
+
+1. Front End PCM DAIs - i.e. struct snd_soc_dai_driver.
+
+2. DAPM graph showing DSP audio routing from FE DAIs to BEs.
+
+3. DAPM widgets from DSP graph.
+
+4. Mixers for gains, routing, etc.
+
+5. DMA configuration.
+
+6. BE AIF widgets.
+
+Items 6 is important for routing the audio outside of the DSP. AIF need to be
+defined for each BE and each stream direction. e.g for BE DAI0 above we would
+have :-
+::
+
+  SND_SOC_DAPM_AIF_IN("DAI0 RX", NULL, 0, SND_SOC_NOPM, 0, 0),
+  SND_SOC_DAPM_AIF_OUT("DAI0 TX", NULL, 0, SND_SOC_NOPM, 0, 0),
+
+The BE AIF are used to connect the DSP graph to the graphs for the other
+component drivers (e.g. codec graph).
+
+
+Hostless PCM streams
+====================
+
+A hostless PCM stream is a stream that is not routed through the host CPU. An
+example of this would be a phone call from handset to modem.
+::
+
+                      *************
+  PCM0 <------------> *           * <----DAI0-----> Codec Headset
+                      *           *
+  PCM1 <------------> *           * <====DAI1=====> Codec Speakers/Mic
+                      *   DSP     *
+  PCM2 <------------> *           * <====DAI2=====> MODEM
+                      *           *
+  PCM3 <------------> *           * <----DAI3-----> BT
+                      *           *
+                      *           * <----DAI4-----> DMIC
+                      *           *
+                      *           * <----DAI5-----> FM
+                      *************
+
+In this case the PCM data is routed via the DSP. The host CPU in this use case
+is only used for control and can sleep during the runtime of the stream.
+
+The host can control the hostless link either by :-
+
+ 1. Configuring the link as a CODEC <-> CODEC style link. In this case the link
+    is enabled or disabled by the state of the DAPM graph. This usually means
+    there is a mixer control that can be used to connect or disconnect the path
+    between both DAIs.
+
+ 2. Hostless FE. This FE has a virtual connection to the BE DAI links on the DAPM
+    graph. Control is then carried out by the FE as regular PCM operations.
+    This method gives more control over the DAI links, but requires much more
+    userspace code to control the link. Its recommended to use CODEC<->CODEC
+    unless your HW needs more fine grained sequencing of the PCM ops.
+
+
+CODEC <-> CODEC link
+--------------------
+
+This DAI link is enabled when DAPM detects a valid path within the DAPM graph.
+The machine driver sets some additional parameters to the DAI link i.e.
+::
+
+  static const struct snd_soc_pcm_stream dai_params = {
+	.formats = SNDRV_PCM_FMTBIT_S32_LE,
+	.rate_min = 8000,
+	.rate_max = 8000,
+	.channels_min = 2,
+	.channels_max = 2,
+  };
+
+  static struct snd_soc_dai_link dais[] = {
+	< ... more DAI links above ... >
+	{
+		.name = "MODEM",
+		.stream_name = "MODEM",
+		.cpu_dai_name = "dai2",
+		.codec_dai_name = "modem-aif1",
+		.codec_name = "modem",
+		.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
+				| SND_SOC_DAIFMT_CBM_CFM,
+		.params = &dai_params,
+	}
+	< ... more DAI links here ... >
+
+These parameters are used to configure the DAI hw_params() when DAPM detects a
+valid path and then calls the PCM operations to start the link. DAPM will also
+call the appropriate PCM operations to disable the DAI when the path is no
+longer valid.
+
+
+Hostless FE
+-----------
+
+The DAI link(s) are enabled by a FE that does not read or write any PCM data.
+This means creating a new FE that is connected with a virtual path to both
+DAI links. The DAI links will be started when the FE PCM is started and stopped
+when the FE PCM is stopped. Note that the FE PCM cannot read or write data in
+this configuration.
+
+
diff --git a/Documentation/sound/soc/index.rst b/Documentation/sound/soc/index.rst
index 85ec51764e83..e142a0f25c5b 100644
--- a/Documentation/sound/soc/index.rst
+++ b/Documentation/sound/soc/index.rst
@@ -16,3 +16,4 @@ The documentation is spilt into the following sections:-
    pops-clicks
    clocking
    jack
+   dpcm
-- 
cgit v1.2.3-58-ga151


From c6ab9e57e84ee015bb9c5de213d9f85e5fd4e085 Mon Sep 17 00:00:00 2001
From: Takashi Iwai 
Date: Fri, 11 Nov 2016 16:55:29 +0100
Subject: ASoC: doc: ReSTize codec_to_codec.txt

Yet another simple conversion from a plain text file.
Renamed to codec-to-codec.rst to align with others.

Acked-by: Mark Brown 
Signed-off-by: Takashi Iwai 
---
 Documentation/sound/alsa/soc/codec_to_codec.txt | 103 ----------------------
 Documentation/sound/soc/codec-to-codec.rst      | 108 ++++++++++++++++++++++++
 Documentation/sound/soc/index.rst               |   1 +
 3 files changed, 109 insertions(+), 103 deletions(-)
 delete mode 100644 Documentation/sound/alsa/soc/codec_to_codec.txt
 create mode 100644 Documentation/sound/soc/codec-to-codec.rst

(limited to 'Documentation/sound')

diff --git a/Documentation/sound/alsa/soc/codec_to_codec.txt b/Documentation/sound/alsa/soc/codec_to_codec.txt
deleted file mode 100644
index 704a6483652c..000000000000
--- a/Documentation/sound/alsa/soc/codec_to_codec.txt
+++ /dev/null
@@ -1,103 +0,0 @@
-Creating codec to codec dai link for ALSA dapm
-===================================================
-
-Mostly the flow of audio is always from CPU to codec so your system
-will look as below:
-
- ---------          ---------
-|         |  dai   |         |
-    CPU    ------->    codec
-|         |        |         |
- ---------          ---------
-
-In case your system looks as below:
-                     ---------
-                    |         |
-                      codec-2
-                    |         |
-                     ---------
-                         |
-                       dai-2
-                         |
- ----------          ---------
-|          |  dai-1 |         |
-    CPU     ------->  codec-1
-|          |        |         |
- ----------          ---------
-                         |
-                       dai-3
-                         |
-                     ---------
-                    |         |
-                      codec-3
-                    |         |
-                     ---------
-
-Suppose codec-2 is a bluetooth chip and codec-3 is connected to
-a speaker and you have a below scenario:
-codec-2 will receive the audio data and the user wants to play that
-audio through codec-3 without involving the CPU.This
-aforementioned case is the ideal case when codec to codec
-connection should be used.
-
-Your dai_link should appear as below in your machine
-file:
-
-/*
- * this pcm stream only supports 24 bit, 2 channel and
- * 48k sampling rate.
- */
-static const struct snd_soc_pcm_stream dsp_codec_params = {
-        .formats = SNDRV_PCM_FMTBIT_S24_LE,
-        .rate_min = 48000,
-        .rate_max = 48000,
-        .channels_min = 2,
-        .channels_max = 2,
-};
-
-{
-    .name = "CPU-DSP",
-    .stream_name = "CPU-DSP",
-    .cpu_dai_name = "samsung-i2s.0",
-    .codec_name = "codec-2,
-    .codec_dai_name = "codec-2-dai_name",
-    .platform_name = "samsung-i2s.0",
-    .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
-            | SND_SOC_DAIFMT_CBM_CFM,
-    .ignore_suspend = 1,
-    .params = &dsp_codec_params,
-},
-{
-    .name = "DSP-CODEC",
-    .stream_name = "DSP-CODEC",
-    .cpu_dai_name = "wm0010-sdi2",
-    .codec_name = "codec-3,
-    .codec_dai_name = "codec-3-dai_name",
-    .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
-            | SND_SOC_DAIFMT_CBM_CFM,
-    .ignore_suspend = 1,
-    .params = &dsp_codec_params,
-},
-
-Above code snippet is motivated from sound/soc/samsung/speyside.c.
-
-Note the "params" callback which lets the dapm know that this
-dai_link is a codec to codec connection.
-
-In dapm core a route is created between cpu_dai playback widget
-and codec_dai capture widget for playback path and vice-versa is
-true for capture path. In order for this aforementioned route to get
-triggered, DAPM needs to find a valid endpoint which could be either
-a sink or source widget corresponding to playback and capture path
-respectively.
-
-In order to trigger this dai_link widget, a thin codec driver for
-the speaker amp can be created as demonstrated in wm8727.c file, it
-sets appropriate constraints for the device even if it needs no control.
-
-Make sure to name your corresponding cpu and codec playback and capture
-dai names ending with "Playback" and "Capture" respectively as dapm core
-will link and power those dais based on the name.
-
-Note that in current device tree there is no way to mark a dai_link
-as codec to codec. However, it may change in future.
diff --git a/Documentation/sound/soc/codec-to-codec.rst b/Documentation/sound/soc/codec-to-codec.rst
new file mode 100644
index 000000000000..810109d7500d
--- /dev/null
+++ b/Documentation/sound/soc/codec-to-codec.rst
@@ -0,0 +1,108 @@
+==============================================
+Creating codec to codec dai link for ALSA dapm
+==============================================
+
+Mostly the flow of audio is always from CPU to codec so your system
+will look as below:
+::
+
+   ---------          ---------
+  |         |  dai   |         |
+      CPU    ------->    codec
+  |         |        |         |
+   ---------          ---------
+
+In case your system looks as below:
+::
+
+                       ---------
+                      |         |
+                        codec-2
+                      |         |
+                      ---------
+                           |
+                         dai-2
+                           |
+   ----------          ---------
+  |          |  dai-1 |         |
+      CPU     ------->  codec-1
+  |          |        |         |
+   ----------          ---------
+                           |
+                         dai-3
+                           |
+                       ---------
+                      |         |
+                        codec-3
+                      |         |
+                       ---------
+
+Suppose codec-2 is a bluetooth chip and codec-3 is connected to
+a speaker and you have a below scenario:
+codec-2 will receive the audio data and the user wants to play that
+audio through codec-3 without involving the CPU.This
+aforementioned case is the ideal case when codec to codec
+connection should be used.
+
+Your dai_link should appear as below in your machine
+file:
+::
+
+ /*
+  * this pcm stream only supports 24 bit, 2 channel and
+  * 48k sampling rate.
+  */
+ static const struct snd_soc_pcm_stream dsp_codec_params = {
+        .formats = SNDRV_PCM_FMTBIT_S24_LE,
+        .rate_min = 48000,
+        .rate_max = 48000,
+        .channels_min = 2,
+        .channels_max = 2,
+ };
+
+ {
+    .name = "CPU-DSP",
+    .stream_name = "CPU-DSP",
+    .cpu_dai_name = "samsung-i2s.0",
+    .codec_name = "codec-2,
+    .codec_dai_name = "codec-2-dai_name",
+    .platform_name = "samsung-i2s.0",
+    .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
+            | SND_SOC_DAIFMT_CBM_CFM,
+    .ignore_suspend = 1,
+    .params = &dsp_codec_params,
+ },
+ {
+    .name = "DSP-CODEC",
+    .stream_name = "DSP-CODEC",
+    .cpu_dai_name = "wm0010-sdi2",
+    .codec_name = "codec-3,
+    .codec_dai_name = "codec-3-dai_name",
+    .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
+            | SND_SOC_DAIFMT_CBM_CFM,
+    .ignore_suspend = 1,
+    .params = &dsp_codec_params,
+ },
+
+Above code snippet is motivated from sound/soc/samsung/speyside.c.
+
+Note the "params" callback which lets the dapm know that this
+dai_link is a codec to codec connection.
+
+In dapm core a route is created between cpu_dai playback widget
+and codec_dai capture widget for playback path and vice-versa is
+true for capture path. In order for this aforementioned route to get
+triggered, DAPM needs to find a valid endpoint which could be either
+a sink or source widget corresponding to playback and capture path
+respectively.
+
+In order to trigger this dai_link widget, a thin codec driver for
+the speaker amp can be created as demonstrated in wm8727.c file, it
+sets appropriate constraints for the device even if it needs no control.
+
+Make sure to name your corresponding cpu and codec playback and capture
+dai names ending with "Playback" and "Capture" respectively as dapm core
+will link and power those dais based on the name.
+
+Note that in current device tree there is no way to mark a dai_link
+as codec to codec. However, it may change in future.
diff --git a/Documentation/sound/soc/index.rst b/Documentation/sound/soc/index.rst
index e142a0f25c5b..e57df2dab2fd 100644
--- a/Documentation/sound/soc/index.rst
+++ b/Documentation/sound/soc/index.rst
@@ -17,3 +17,4 @@ The documentation is spilt into the following sections:-
    clocking
    jack
    dpcm
+   codec-to-codec
-- 
cgit v1.2.3-58-ga151