From 5ef75e710b4950439f953c4897e4a871c2f9dc8f Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Wed, 25 Jul 2012 16:09:11 +0300 Subject: ASoC: omap-abe-twl6040: Add device tree support When the board boots with device tree the driver will receive the name of the card, DAPM routing map, phandle for the audio components described in the dts file, mclk speed, and the possibility of detecting the jack detection. The card will be set up based on this information. Since the routing is provided via DT we can mark the card fully routed so core can take care of disconnecting the unused pins. Signed-off-by: Peter Ujfalusi Reviwed-by: Mark Brown Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/omap-abe-twl6040.txt | 91 ++++++++++++++++++++++ 1 file changed, 91 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/omap-abe-twl6040.txt (limited to 'Documentation') diff --git a/Documentation/devicetree/bindings/sound/omap-abe-twl6040.txt b/Documentation/devicetree/bindings/sound/omap-abe-twl6040.txt new file mode 100644 index 000000000000..65dec876cb2d --- /dev/null +++ b/Documentation/devicetree/bindings/sound/omap-abe-twl6040.txt @@ -0,0 +1,91 @@ +* Texas Instruments OMAP4+ and twl6040 based audio setups + +Required properties: +- compatible: "ti,abe-twl6040" +- ti,model: Name of the sound card ( for example "SDP4430") +- ti,mclk-freq: MCLK frequency for HPPLL operation +- ti,mcpdm: phandle for the McPDM node +- ti,twl6040: phandle for the twl6040 core node +- ti,audio-routing: List of connections between audio components. + Each entry is a pair of strings, the first being the connection's sink, + the second being the connection's source. + +Optional properties: +- ti,dmic: phandle for the OMAP dmic node if the machine have it connected +- ti,jack_detection: Need to be set to <1> if the board capable to detect jack + insertion, removal. + +Available audio endpoints for the audio-routing table: + +Board connectors: + * Headset Stereophone + * Earphone Spk + * Ext Spk + * Line Out + * Vibrator + * Headset Mic + * Main Handset Mic + * Sub Handset Mic + * Line In + * Digital Mic + +twl6040 pins: + * HSOL + * HSOR + * EP + * HFL + * HFR + * AUXL + * AUXR + * VIBRAL + * VIBRAR + * HSMIC + * MAINMIC + * SUBMIC + * AFML + * AFMR + + * Headset Mic Bias + * Main Mic Bias + * Digital Mic1 Bias + * Digital Mic2 Bias + +Digital mic pins: + * DMic + +Example: + +sound { + compatible = "ti,abe-twl6040"; + ti,model = "SDP4430"; + + ti,jack-detection = <1>; + ti,mclk-freq = <38400000>; + + ti,mcpdm = <&mcpdm>; + ti,dmic = <&dmic>; + + ti,twl6040 = <&twl6040>; + + /* Audio routing */ + ti,audio-routing = + "Headset Stereophone", "HSOL", + "Headset Stereophone", "HSOR", + "Earphone Spk", "EP", + "Ext Spk", "HFL", + "Ext Spk", "HFR", + "Line Out", "AUXL", + "Line Out", "AUXR", + "Vibrator", "VIBRAL", + "Vibrator", "VIBRAR", + "HSMIC", "Headset Mic", + "Headset Mic", "Headset Mic Bias", + "MAINMIC", "Main Handset Mic", + "Main Handset Mic", "Main Mic Bias", + "SUBMIC", "Sub Handset Mic", + "Sub Handset Mic", "Main Mic Bias", + "AFML", "Line In", + "AFMR", "Line In", + "DMic", "Digital Mic", + "Digital Mic", "Digital Mic1 Bias"; +}; -- cgit v1.2.3-58-ga151 From 85d07e4d625d6511934799f7df93e9111ac2c88b Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Wed, 25 Jul 2012 15:28:34 +0200 Subject: ASoC: add DT bindings for cs4270 Signed-off-by: Daniel Mack Acked-by: Timur Tabi Signed-off-by: Mark Brown --- Documentation/devicetree/bindings/sound/cs4270.txt | 16 ++++++++++++++++ sound/soc/codecs/cs4270.c | 11 +++++++++++ 2 files changed, 27 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/cs4270.txt (limited to 'Documentation') diff --git a/Documentation/devicetree/bindings/sound/cs4270.txt b/Documentation/devicetree/bindings/sound/cs4270.txt new file mode 100644 index 000000000000..7f0bfd84d3fc --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cs4270.txt @@ -0,0 +1,16 @@ +CS4270 audio CODEC + +The driver for this device currently only supports I2C. + +Required properties: + + - compatible : "cirrus,cs4270" + + - reg : the I2C address of the device for I2C + +Example: + +codec: cs4270@48 { + compatible = "cirrus,cs4270"; + reg = <0x48>; +}; diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index 047917f0b8ae..4b71b01ecbcd 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -29,6 +29,7 @@ #include #include #include +#include /* * The codec isn't really big-endian or little-endian, since the I2S @@ -639,6 +640,15 @@ static const struct snd_soc_codec_driver soc_codec_device_cs4270 = { .reg_cache_default = cs4270_default_reg_cache, }; +/* + * cs4270_of_match - the device tree bindings + */ +static const struct of_device_id cs4270_of_match[] = { + { .compatible = "cirrus,cs4270", }, + { } +}; +MODULE_DEVICE_TABLE(of, cs4270_of_match); + /** * cs4270_i2c_probe - initialize the I2C interface of the CS4270 * @i2c_client: the I2C client object @@ -718,6 +728,7 @@ static struct i2c_driver cs4270_i2c_driver = { .driver = { .name = "cs4270", .owner = THIS_MODULE, + .of_match_table = cs4270_of_match, }, .id_table = cs4270_id, .probe = cs4270_i2c_probe, -- cgit v1.2.3-58-ga151 From 02286190f3ec86f03025a60c4d3f747ff1047248 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Wed, 25 Jul 2012 15:28:35 +0200 Subject: ASoC: Add reset-gpio DT property to cs4270 driver In the process of moving over from static board files to the device tree, reset pins of peripheral reset pins should be handled by their corresponding drivers. Add a reset-gpio DT property to the cs4270 driver, and de-assert it before probing the chip. The logic could be augmented some day to re-assert it when codec is put to suspend. Signed-off-by: Daniel Mack Acked-by: Timur Tabi Signed-off-by: Mark Brown --- Documentation/devicetree/bindings/sound/cs4270.txt | 5 +++++ sound/soc/codecs/cs4270.c | 17 +++++++++++++++++ 2 files changed, 22 insertions(+) (limited to 'Documentation') diff --git a/Documentation/devicetree/bindings/sound/cs4270.txt b/Documentation/devicetree/bindings/sound/cs4270.txt index 7f0bfd84d3fc..6b222f9b8ef5 100644 --- a/Documentation/devicetree/bindings/sound/cs4270.txt +++ b/Documentation/devicetree/bindings/sound/cs4270.txt @@ -8,6 +8,11 @@ Required properties: - reg : the I2C address of the device for I2C +Optional properties: + + - reset-gpio : a GPIO spec for the reset pin. If specified, it will be + deasserted before communication to the codec starts. + Example: codec: cs4270@48 { diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index 4b71b01ecbcd..fd11bb646d40 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -30,6 +30,7 @@ #include #include #include +#include /* * The codec isn't really big-endian or little-endian, since the I2S @@ -660,9 +661,25 @@ MODULE_DEVICE_TABLE(of, cs4270_of_match); static int cs4270_i2c_probe(struct i2c_client *i2c_client, const struct i2c_device_id *id) { + struct device_node *np = i2c_client->dev.of_node; struct cs4270_private *cs4270; int ret; + /* See if we have a way to bring the codec out of reset */ + if (np) { + enum of_gpio_flags flags; + int gpio = of_get_named_gpio_flags(np, "reset-gpio", 0, &flags); + + if (gpio_is_valid(gpio)) { + ret = devm_gpio_request_one(&i2c_client->dev, gpio, + flags & OF_GPIO_ACTIVE_LOW ? + GPIOF_OUT_INIT_LOW : GPIOF_OUT_INIT_HIGH, + "cs4270 reset"); + if (ret < 0) + return ret; + } + } + /* Verify that we have a CS4270 */ ret = i2c_smbus_read_byte_data(i2c_client, CS4270_CHIPID); -- cgit v1.2.3-58-ga151 From 1c86845268dc91fa6a53de9a4479b407cd4ee903 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 13 Aug 2012 11:09:35 +0200 Subject: ALSA: hda - Add 3stack-automute model to AD1882 codec Added a simple support of automute for the front HP jack to AD1882 stack model. Such an addition is basically an exception -- we really want to avoid the static quirk codes, but AD1882 parser isn't still ready for moving to the BIOS auto-parser yet. So, as a quick fix, I merged it for now. In near future, we really need the big clean up of patch_analog.c to move on to the auto-parser... Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/HD-Audio-Models.txt | 3 ++- sound/pci/hda/patch_analog.c | 40 ++++++++++++++++++++++++++++ 2 files changed, 42 insertions(+), 1 deletion(-) (limited to 'Documentation') diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt index a92bba816843..16dfe57f1731 100644 --- a/Documentation/sound/alsa/HD-Audio-Models.txt +++ b/Documentation/sound/alsa/HD-Audio-Models.txt @@ -74,7 +74,8 @@ CMI9880 AD1882 / AD1882A ================ - 3stack 3-stack mode (default) + 3stack 3-stack mode + 3stack-automute 3-stack with automute front HP (default) 6stack 6-stack mode AD1884A / AD1883 / AD1984A / AD1984B diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 0208fa121e5a..21218853366d 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -4814,6 +4814,32 @@ static const struct snd_kcontrol_new ad1882_3stack_mixers[] = { { } /* end */ }; +/* simple auto-mute control for AD1882 3-stack board */ +#define AD1882_HP_EVENT 0x01 + +static void ad1882_3stack_automute(struct hda_codec *codec) +{ + bool mute = snd_hda_jack_detect(codec, 0x11); + snd_hda_codec_write(codec, 0x12, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, + mute ? 0 : PIN_OUT); +} + +static int ad1882_3stack_automute_init(struct hda_codec *codec) +{ + ad198x_init(codec); + ad1882_3stack_automute(codec); + return 0; +} + +static void ad1882_3stack_unsol_event(struct hda_codec *codec, unsigned int res) +{ + switch (res >> 26) { + case AD1882_HP_EVENT: + ad1882_3stack_automute(codec); + break; + } +} + static const struct snd_kcontrol_new ad1882_6stack_mixers[] = { HDA_CODEC_MUTE("Surround Playback Switch", 0x16, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x24, 1, 0x0, HDA_OUTPUT), @@ -4928,6 +4954,11 @@ static const struct hda_verb ad1882_init_verbs[] = { { } /* end */ }; +static const struct hda_verb ad1882_3stack_automute_verbs[] = { + {0x11, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1882_HP_EVENT}, + { } /* end */ +}; + #ifdef CONFIG_SND_HDA_POWER_SAVE static const struct hda_amp_list ad1882_loopbacks[] = { { 0x20, HDA_INPUT, 0 }, /* Front Mic */ @@ -4942,12 +4973,14 @@ static const struct hda_amp_list ad1882_loopbacks[] = { enum { AD1882_3STACK, AD1882_6STACK, + AD1882_3STACK_AUTOMUTE, AD1882_MODELS }; static const char * const ad1882_models[AD1986A_MODELS] = { [AD1882_3STACK] = "3stack", [AD1882_6STACK] = "6stack", + [AD1882_3STACK_AUTOMUTE] = "3stack-automute", }; @@ -5002,6 +5035,7 @@ static int patch_ad1882(struct hda_codec *codec) switch (board_config) { default: case AD1882_3STACK: + case AD1882_3STACK_AUTOMUTE: spec->num_mixers = 3; spec->mixers[2] = ad1882_3stack_mixers; spec->channel_mode = ad1882_modes; @@ -5009,6 +5043,12 @@ static int patch_ad1882(struct hda_codec *codec) spec->need_dac_fix = 1; spec->multiout.max_channels = 2; spec->multiout.num_dacs = 1; + if (board_config != AD1882_3STACK) { + spec->init_verbs[spec->num_init_verbs++] = + ad1882_3stack_automute_verbs; + codec->patch_ops.unsol_event = ad1882_3stack_unsol_event; + codec->patch_ops.init = ad1882_3stack_automute_init; + } break; case AD1882_6STACK: spec->num_mixers = 3; -- cgit v1.2.3-58-ga151 From fff8491c8b8cce5fc9190e025d1a665f2ee71a4f Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Tue, 14 Aug 2012 12:07:56 +0300 Subject: ASoC: omap-twl4030: Simple machine driver for TI SoC with twl4030 codec Machine driver to handle simple devices using twl4030 as audio codec. The driver supports the following boards: - Beagleboard or Devkit8000 - Gumstix Overo or CompuLab CM-T35/CM-T3730 - IGEP v2 - OMAP3EVM All of these boards can be switched to use this driver since their setup is identical. Devicetree support for the omap-twl4030 machine driver also implemented. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/omap-twl4030.txt | 17 ++ include/linux/platform_data/omap-twl4030.h | 32 ++++ sound/soc/omap/Kconfig | 13 ++ sound/soc/omap/Makefile | 2 + sound/soc/omap/omap-twl4030.c | 188 +++++++++++++++++++++ 5 files changed, 252 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/omap-twl4030.txt create mode 100644 include/linux/platform_data/omap-twl4030.h create mode 100644 sound/soc/omap/omap-twl4030.c (limited to 'Documentation') diff --git a/Documentation/devicetree/bindings/sound/omap-twl4030.txt b/Documentation/devicetree/bindings/sound/omap-twl4030.txt new file mode 100644 index 000000000000..6fae51c7f766 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/omap-twl4030.txt @@ -0,0 +1,17 @@ +* Texas Instruments SoC with twl4030 based audio setups + +Required properties: +- compatible: "ti,omap-twl4030" +- ti,model: Name of the sound card (for example "omap3beagle") +- ti,mcbsp: phandle for the McBSP node +- ti,codec: phandle for the twl4030 audio node + +Example: + +sound { + compatible = "ti,omap-twl4030"; + ti,model = "omap3beagle"; + + ti,mcbsp = <&mcbsp2>; + ti,codec = <&twl_audio>; +}; diff --git a/include/linux/platform_data/omap-twl4030.h b/include/linux/platform_data/omap-twl4030.h new file mode 100644 index 000000000000..c7bef788daab --- /dev/null +++ b/include/linux/platform_data/omap-twl4030.h @@ -0,0 +1,32 @@ +/** + * omap-twl4030.h - ASoC machine driver for TI SoC based boards with twl4030 + * codec, header. + * + * Copyright (C) 2012 Texas Instruments Incorporated - http://www.ti.com + * All rights reserved. + * + * Author: Peter Ujfalusi + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program 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., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + */ + +#ifndef _OMAP_TWL4030_H_ +#define _OMAP_TWL4030_H_ + +struct omap_tw4030_pdata { + const char *card_name; +}; + +#endif /* _OMAP_TWL4030_H_ */ diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index 57a2fa751085..fc83d748625f 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -95,6 +95,19 @@ config SND_OMAP_SOC_SDP3430 Say Y if you want to add support for SoC audio on Texas Instruments SDP3430. +config SND_OMAP_SOC_OMAP_TWL4030 + tristate "SoC Audio support for TI SoC based boards with twl4030 codec" + depends on TWL4030_CORE && SND_OMAP_SOC + select SND_OMAP_SOC_MCBSP + select SND_SOC_TWL4030 + help + Say Y if you want to add support for SoC audio on TI SoC based boards + using twl4030 as c codec. This driver currently supports: + - Beagleboard or Devkit8000 + - Gumstix Overo or CompuLab CM-T35/CM-T3730 + - IGEP v2 + - OMAP3EVM + config SND_OMAP_SOC_OMAP_ABE_TWL6040 tristate "SoC Audio support for OMAP boards using ABE and twl6040 codec" depends on TWL6040_CORE && SND_OMAP_SOC && ARCH_OMAP4 diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile index 0e14dd322565..861e640e2be9 100644 --- a/sound/soc/omap/Makefile +++ b/sound/soc/omap/Makefile @@ -21,6 +21,7 @@ snd-soc-omap3evm-objs := omap3evm.o snd-soc-am3517evm-objs := am3517evm.o snd-soc-sdp3430-objs := sdp3430.o snd-soc-omap-abe-twl6040-objs := omap-abe-twl6040.o +snd-soc-omap-twl4030-objs := omap-twl4030.o snd-soc-omap3pandora-objs := omap3pandora.o snd-soc-omap3beagle-objs := omap3beagle.o snd-soc-zoom2-objs := zoom2.o @@ -37,6 +38,7 @@ obj-$(CONFIG_SND_OMAP_SOC_OMAP3EVM) += snd-soc-omap3evm.o obj-$(CONFIG_SND_OMAP_SOC_AM3517EVM) += snd-soc-am3517evm.o obj-$(CONFIG_SND_OMAP_SOC_SDP3430) += snd-soc-sdp3430.o obj-$(CONFIG_SND_OMAP_SOC_OMAP_ABE_TWL6040) += snd-soc-omap-abe-twl6040.o +obj-$(CONFIG_SND_OMAP_SOC_OMAP_TWL4030) += snd-soc-omap-twl4030.o obj-$(CONFIG_SND_OMAP_SOC_OMAP3_PANDORA) += snd-soc-omap3pandora.o obj-$(CONFIG_SND_OMAP_SOC_OMAP3_BEAGLE) += snd-soc-omap3beagle.o obj-$(CONFIG_SND_OMAP_SOC_ZOOM2) += snd-soc-zoom2.o diff --git a/sound/soc/omap/omap-twl4030.c b/sound/soc/omap/omap-twl4030.c new file mode 100644 index 000000000000..3b97b87971f5 --- /dev/null +++ b/sound/soc/omap/omap-twl4030.c @@ -0,0 +1,188 @@ +/* + * omap-twl4030.c -- SoC audio for TI SoC based boards with twl4030 codec + * + * Copyright (C) 2012 Texas Instruments Incorporated - http://www.ti.com + * All rights reserved. + * + * Author: Peter Ujfalusi + * + * This driver replaces the following machine drivers: + * omap3beagle (Author: Steve Sakoman ) + * omap3evm (Author: Anuj Aggarwal ) + * overo (Author: Steve Sakoman ) + * igep0020 (Author: Enric Balletbo i Serra ) + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program 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., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#include +#include +#include +#include + +#include +#include +#include + +#include "omap-mcbsp.h" +#include "omap-pcm.h" + +static int omap_twl4030_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_card *card = codec->card; + unsigned int fmt; + int ret; + + switch (params_channels(params)) { + case 2: /* Stereo I2S mode */ + fmt = SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM; + break; + case 4: /* Four channel TDM mode */ + fmt = SND_SOC_DAIFMT_DSP_A | + SND_SOC_DAIFMT_IB_NF | + SND_SOC_DAIFMT_CBM_CFM; + break; + default: + return -EINVAL; + } + + /* Set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, fmt); + if (ret < 0) { + dev_err(card->dev, "can't set codec DAI configuration\n"); + return ret; + } + + /* Set cpu DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, fmt); + if (ret < 0) { + dev_err(card->dev, "can't set cpu DAI configuration\n"); + return ret; + } + + return 0; +} + +static struct snd_soc_ops omap_twl4030_ops = { + .hw_params = omap_twl4030_hw_params, +}; + +/* Digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link omap_twl4030_dai_links[] = { + { + .name = "TWL4030", + .stream_name = "TWL4030", + .cpu_dai_name = "omap-mcbsp.2", + .codec_dai_name = "twl4030-hifi", + .platform_name = "omap-pcm-audio", + .codec_name = "twl4030-codec", + .ops = &omap_twl4030_ops, + }, +}; + +/* Audio machine driver */ +static struct snd_soc_card omap_twl4030_card = { + .owner = THIS_MODULE, + .dai_link = omap_twl4030_dai_links, + .num_links = ARRAY_SIZE(omap_twl4030_dai_links), +}; + +static __devinit int omap_twl4030_probe(struct platform_device *pdev) +{ + struct omap_tw4030_pdata *pdata = dev_get_platdata(&pdev->dev); + struct device_node *node = pdev->dev.of_node; + struct snd_soc_card *card = &omap_twl4030_card; + int ret = 0; + + card->dev = &pdev->dev; + + if (node) { + struct device_node *dai_node; + + if (snd_soc_of_parse_card_name(card, "ti,model")) { + dev_err(&pdev->dev, "Card name is not provided\n"); + return -ENODEV; + } + + dai_node = of_parse_phandle(node, "ti,mcbsp", 0); + if (!dai_node) { + dev_err(&pdev->dev, "McBSP node is not provided\n"); + return -EINVAL; + } + omap_twl4030_dai_links[0].cpu_dai_name = NULL; + omap_twl4030_dai_links[0].cpu_of_node = dai_node; + + } else if (pdata) { + if (pdata->card_name) { + card->name = pdata->card_name; + } else { + dev_err(&pdev->dev, "Card name is not provided\n"); + return -ENODEV; + } + } else { + dev_err(&pdev->dev, "Missing pdata\n"); + return -ENODEV; + } + + ret = snd_soc_register_card(card); + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", + ret); + return ret; + } + + return 0; +} + +static int __devexit omap_twl4030_remove(struct platform_device *pdev) +{ + struct snd_soc_card *card = platform_get_drvdata(pdev); + + snd_soc_unregister_card(card); + + return 0; +} + +static const struct of_device_id omap_twl4030_of_match[] = { + {.compatible = "ti,omap-twl4030", }, + { }, +}; +MODULE_DEVICE_TABLE(of, omap_twl4030_of_match); + +static struct platform_driver omap_twl4030_driver = { + .driver = { + .name = "omap-twl4030", + .owner = THIS_MODULE, + .pm = &snd_soc_pm_ops, + .of_match_table = omap_twl4030_of_match, + }, + .probe = omap_twl4030_probe, + .remove = __devexit_p(omap_twl4030_remove), +}; + +module_platform_driver(omap_twl4030_driver); + +MODULE_AUTHOR("Peter Ujfalusi "); +MODULE_DESCRIPTION("ALSA SoC for TI SoC based boards with twl4030 codec"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:omap-twl4030"); -- cgit v1.2.3-58-ga151 From 11dd586421b3091007e6f084a9211f3baa66f9fc Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 16 Aug 2012 16:41:08 +0300 Subject: ASoC: omap-mcbsp: Add device tree bindings Device tree support for McBSP modules on OMAP2+ SoC. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/omap-mcbsp.txt | 45 +++++++++++++++ sound/soc/omap/omap-mcbsp.c | 66 +++++++++++++++++++++- 2 files changed, 110 insertions(+), 1 deletion(-) create mode 100644 Documentation/devicetree/bindings/sound/omap-mcbsp.txt (limited to 'Documentation') diff --git a/Documentation/devicetree/bindings/sound/omap-mcbsp.txt b/Documentation/devicetree/bindings/sound/omap-mcbsp.txt new file mode 100644 index 000000000000..447cb131e909 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/omap-mcbsp.txt @@ -0,0 +1,45 @@ +* Texas Instruments OMAP2+ McBSP module + +Required properties: +- compatible: "ti,omap2420-mcbsp" for McBSP on OMAP2420 + "ti,omap2430-mcbsp" for McBSP on OMAP2430 + "ti,omap3-mcbsp" for McBSP on OMAP3 + "ti,omap4-mcbsp" for McBSP on OMAP4 and newer SoC +- reg: Register location and size, for OMAP4+ as an array: + , + ; +- interrupts: Interrupt numbers for the McBSP port, as an array in case the + McBSP IP have more interrupt lines: + , + , + ; +- interrupt-parent: The parent interrupt controller +- ti,buffer-size: Size of the FIFO on the port (OMAP2430 and newer SoC) +- ti,hwmods: Name of the hwmod associated to the McBSP port + +Sidetone support for OMAP3 McBSP2 and 3 ports: +- sidetone { }: Within this section the following parameters are required: +- reg: Register location and size for the ST block +- interrupts: The interrupt number for the ST block +- interrupt-parent: The parent interrupt controller for the ST block + +Example: + +mcbsp2: mcbsp@49022000 { + compatible = "ti,omap3-mcbsp"; + #address-cells = <1>; + #size-cells = <1>; + reg = <0x49022000 0xff>; + interrupts = <0 17 0x4>, /* OCP compliant interrup */ + <0 62 0x4>, /* TX interrup */ + <0 63 0x4>; /* RX interrup */ + interrupt-parent = <&intc>; + ti,buffer-size = <1280>; + ti,hwmods = "mcbsp2"; + + sidetone { + reg = <0x49028000 0xff>; + interrupts = <0 4 0x4>; + interrupt-parent = <&intc>; + }; +}; diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index b9770eea28a0..2e1750e2ab31 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -26,6 +26,8 @@ #include #include #include +#include +#include #include #include #include @@ -737,13 +739,74 @@ int omap_mcbsp_st_add_controls(struct snd_soc_pcm_runtime *rtd) } EXPORT_SYMBOL_GPL(omap_mcbsp_st_add_controls); +static struct omap_mcbsp_platform_data omap2420_pdata = { + .reg_step = 4, + .reg_size = 2, +}; + +static struct omap_mcbsp_platform_data omap2430_pdata = { + .reg_step = 4, + .reg_size = 4, + .has_ccr = true, +}; + +static struct omap_mcbsp_platform_data omap3_pdata = { + .reg_step = 4, + .reg_size = 4, + .has_ccr = true, + .has_wakeup = true, +}; + +static struct omap_mcbsp_platform_data omap4_pdata = { + .reg_step = 4, + .reg_size = 4, + .has_ccr = true, + .has_wakeup = true, +}; + +static const struct of_device_id omap_mcbsp_of_match[] = { + { + .compatible = "ti,omap2420-mcbsp", + .data = &omap2420_pdata, + }, + { + .compatible = "ti,omap2430-mcbsp", + .data = &omap2430_pdata, + }, + { + .compatible = "ti,omap3-mcbsp", + .data = &omap3_pdata, + }, + { + .compatible = "ti,omap4-mcbsp", + .data = &omap4_pdata, + }, + { }, +}; +MODULE_DEVICE_TABLE(of, omap_mcbsp_of_match); + static __devinit int asoc_mcbsp_probe(struct platform_device *pdev) { struct omap_mcbsp_platform_data *pdata = dev_get_platdata(&pdev->dev); struct omap_mcbsp *mcbsp; + const struct of_device_id *match; int ret; - if (!pdata) { + match = of_match_device(omap_mcbsp_of_match, &pdev->dev); + if (match) { + struct device_node *node = pdev->dev.of_node; + int buffer_size; + + pdata = devm_kzalloc(&pdev->dev, + sizeof(struct omap_mcbsp_platform_data), + GFP_KERNEL); + if (!pdata) + return -ENOMEM; + + memcpy(pdata, match->data, sizeof(*pdata)); + if (!of_property_read_u32(node, "ti,buffer-size", &buffer_size)) + pdata->buffer_size = buffer_size; + } else if (!pdata) { dev_err(&pdev->dev, "missing platform data.\n"); return -EINVAL; } @@ -785,6 +848,7 @@ static struct platform_driver asoc_mcbsp_driver = { .driver = { .name = "omap-mcbsp", .owner = THIS_MODULE, + .of_match_table = omap_mcbsp_of_match, }, .probe = asoc_mcbsp_probe, -- cgit v1.2.3-58-ga151 From b8101048f0f3cd281ed4c4901e38ae2bcfb32030 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Tue, 21 Aug 2012 17:33:56 +0300 Subject: ASoC: omap-mcbsp: Device tree binding documentation update To reflect the final devicetree node structure of McBSPs. The initial OMAP McBSP DT structure was not able to describe the IP (and it's versions) correctly. The main issue was the sidetone block of McBSP2/3 on OMAP3. With this change in the DT description the OS can get the needed information about the IP. The sidetone is still not supported when the Linux kernel is booted with DT since we still depend on hwmod to fill the resources. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/omap-mcbsp.txt | 28 ++++++++-------------- 1 file changed, 10 insertions(+), 18 deletions(-) (limited to 'Documentation') diff --git a/Documentation/devicetree/bindings/sound/omap-mcbsp.txt b/Documentation/devicetree/bindings/sound/omap-mcbsp.txt index 447cb131e909..17cce4490456 100644 --- a/Documentation/devicetree/bindings/sound/omap-mcbsp.txt +++ b/Documentation/devicetree/bindings/sound/omap-mcbsp.txt @@ -8,38 +8,30 @@ Required properties: - reg: Register location and size, for OMAP4+ as an array: , ; +- reg-names: Array of strings associated with the address space - interrupts: Interrupt numbers for the McBSP port, as an array in case the McBSP IP have more interrupt lines: , , ; +- interrupt-names: Array of strings associated with the interrupt numbers - interrupt-parent: The parent interrupt controller - ti,buffer-size: Size of the FIFO on the port (OMAP2430 and newer SoC) - ti,hwmods: Name of the hwmod associated to the McBSP port -Sidetone support for OMAP3 McBSP2 and 3 ports: -- sidetone { }: Within this section the following parameters are required: -- reg: Register location and size for the ST block -- interrupts: The interrupt number for the ST block -- interrupt-parent: The parent interrupt controller for the ST block - Example: mcbsp2: mcbsp@49022000 { compatible = "ti,omap3-mcbsp"; - #address-cells = <1>; - #size-cells = <1>; - reg = <0x49022000 0xff>; - interrupts = <0 17 0x4>, /* OCP compliant interrup */ - <0 62 0x4>, /* TX interrup */ - <0 63 0x4>; /* RX interrup */ + reg = <0x49022000 0xff>, + <0x49028000 0xff>; + reg-names = "mpu", "sidetone"; + interrupts = <0 17 0x4>, /* OCP compliant interrupt */ + <0 62 0x4>, /* TX interrupt */ + <0 63 0x4>, /* RX interrupt */ + <0 4 0x4>; /* Sidetone */ + interrupt-names = "common", "tx", "rx", "sidetone"; interrupt-parent = <&intc>; ti,buffer-size = <1280>; ti,hwmods = "mcbsp2"; - - sidetone { - reg = <0x49028000 0xff>; - interrupts = <0 4 0x4>; - interrupt-parent = <&intc>; - }; }; -- cgit v1.2.3-58-ga151 From c24fdc886fde9ce7bda8115b9c2b338818796c65 Mon Sep 17 00:00:00 2001 From: "Hebbar, Gururaja" Date: Mon, 27 Aug 2012 18:56:44 +0530 Subject: ASoC: tlv320aic3x: Add device tree bindings Device tree support for tlv320aic3x CODEC driver. Signed-off-by: Hebbar, Gururaja Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/tlv320aic3x.txt | 20 ++++++++++++++ sound/soc/codecs/tlv320aic3x.c | 31 ++++++++++++++++++++++ 2 files changed, 51 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/tlv320aic3x.txt (limited to 'Documentation') diff --git a/Documentation/devicetree/bindings/sound/tlv320aic3x.txt b/Documentation/devicetree/bindings/sound/tlv320aic3x.txt new file mode 100644 index 000000000000..e7b98f41fa5f --- /dev/null +++ b/Documentation/devicetree/bindings/sound/tlv320aic3x.txt @@ -0,0 +1,20 @@ +Texas Instruments - tlv320aic3x Codec module + +The tlv320aic3x serial control bus communicates through I2C protocols + +Required properties: +- compatible - "string" - "ti,tlv320aic3x" +- reg - - I2C slave address + + +Optional properties: + +- gpio-reset - gpio pin number used for codec reset +- ai3x-gpio-func - - AIC3X_GPIO1 & AIC3X_GPIO2 Functionality + +Example: + +tlv320aic3x: tlv320aic3x@1b { + compatible = "ti,tlv320aic3x"; + reg = <0x1b>; +}; diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 01485bd51404..5708a973a776 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -40,6 +40,7 @@ #include #include #include +#include #include #include #include @@ -1457,6 +1458,8 @@ static int aic3x_i2c_probe(struct i2c_client *i2c, { struct aic3x_pdata *pdata = i2c->dev.platform_data; struct aic3x_priv *aic3x; + struct aic3x_setup_data *ai3x_setup; + struct device_node *np = i2c->dev.of_node; int ret; aic3x = devm_kzalloc(&i2c->dev, sizeof(struct aic3x_priv), GFP_KERNEL); @@ -1471,6 +1474,25 @@ static int aic3x_i2c_probe(struct i2c_client *i2c, if (pdata) { aic3x->gpio_reset = pdata->gpio_reset; aic3x->setup = pdata->setup; + } else if (np) { + ai3x_setup = devm_kzalloc(&i2c->dev, sizeof(*ai3x_setup), + GFP_KERNEL); + if (ai3x_setup == NULL) { + dev_err(&i2c->dev, "failed to create private data\n"); + return -ENOMEM; + } + + ret = of_get_named_gpio(np, "gpio-reset", 0); + if (ret >= 0) + aic3x->gpio_reset = ret; + else + aic3x->gpio_reset = -1; + + if (of_property_read_u32_array(np, "ai3x-gpio-func", + ai3x_setup->gpio_func, 2) >= 0) { + aic3x->setup = ai3x_setup; + } + } else { aic3x->gpio_reset = -1; } @@ -1488,11 +1510,20 @@ static int aic3x_i2c_remove(struct i2c_client *client) return 0; } +#if defined(CONFIG_OF) +static const struct of_device_id tlv320aic3x_of_match[] = { + { .compatible = "ti,tlv320aic3x", }, + {}, +}; +MODULE_DEVICE_TABLE(of, tlv320aic3x_of_match); +#endif + /* machine i2c codec control layer */ static struct i2c_driver aic3x_i2c_driver = { .driver = { .name = "tlv320aic3x-codec", .owner = THIS_MODULE, + .of_match_table = of_match_ptr(tlv320aic3x_of_match), }, .probe = aic3x_i2c_probe, .remove = aic3x_i2c_remove, -- cgit v1.2.3-58-ga151 From 3e3b8c3415b15adb5a7ffcbfbeb360e7c9f5f4f7 Mon Sep 17 00:00:00 2001 From: "Hebbar, Gururaja" Date: Mon, 27 Aug 2012 18:56:42 +0530 Subject: ASoC: Davinci: McASP: add device tree support for McASP Add device tree probe for McASP driver. Note: DMA parameters are not populated from DT and will be done later. Signed-off-by: Hebbar, Gururaja Signed-off-by: Mark Brown --- .../bindings/sound/davinci-mcasp-audio.txt | 44 ++++++++ sound/soc/davinci/davinci-mcasp.c | 124 ++++++++++++++++++++- 2 files changed, 167 insertions(+), 1 deletion(-) create mode 100644 Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt (limited to 'Documentation') diff --git a/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt b/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt new file mode 100644 index 000000000000..e6148eca2942 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt @@ -0,0 +1,44 @@ +Texas Instruments McASP controller + +Required properties: +- compatible : + "ti,dm646x-mcasp-audio" : for DM646x platforms + "ti,da830-mcasp-audio" : for both DA830 & DA850 platforms + +- reg : Should contain McASP registers offset and length +- interrupts : Interrupt number for McASP +- op-mode : I2S/DIT ops mode. +- tdm-slots : Slots for TDM operation. +- num-serializer : Serializers used by McASP. +- serial-dir : A list of serializer pin mode. The list number should be equal + to "num-serializer" parameter. Each entry is a number indication + serializer pin direction. (0 - INACTIVE, 1 - TX, 2 - RX) + + +Optional properties: + +- ti,hwmods : Must be "mcasp", n is controller instance starting 0 +- tx-num-evt : FIFO levels. +- rx-num-evt : FIFO levels. +- sram-size-playback : size of sram to be allocated during playback +- sram-size-capture : size of sram to be allocated during capture + +Example: + +mcasp0: mcasp0@1d00000 { + compatible = "ti,da830-mcasp-audio"; + #address-cells = <1>; + #size-cells = <0>; + reg = <0x100000 0x3000>; + interrupts = <82 83>; + op-mode = <0>; /* MCASP_IIS_MODE */ + tdm-slots = <2>; + num-serializer = <16>; + serial-dir = < + 0 0 0 0 /* 0: INACTIVE, 1: TX, 2: RX */ + 0 0 0 0 + 0 0 0 1 + 2 0 0 0 >; + tx-num-evt = <1>; + rx-num-evt = <1>; +}; diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 8f3c5a4cf537..7ecf19dfb07c 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -22,6 +22,9 @@ #include #include #include +#include +#include +#include #include #include @@ -856,6 +859,114 @@ static struct snd_soc_dai_driver davinci_mcasp_dai[] = { }; +static const struct of_device_id mcasp_dt_ids[] = { + { + .compatible = "ti,dm646x-mcasp-audio", + .data = (void *)MCASP_VERSION_1, + }, + { + .compatible = "ti,da830-mcasp-audio", + .data = (void *)MCASP_VERSION_2, + }, + { /* sentinel */ } +}; +MODULE_DEVICE_TABLE(of, mcasp_dt_ids); + +static struct snd_platform_data *davinci_mcasp_set_pdata_from_of( + struct platform_device *pdev) +{ + struct device_node *np = pdev->dev.of_node; + struct snd_platform_data *pdata = NULL; + const struct of_device_id *match = + of_match_device(of_match_ptr(mcasp_dt_ids), &pdev->dev); + + const u32 *of_serial_dir32; + u8 *of_serial_dir; + u32 val; + int i, ret = 0; + + if (pdev->dev.platform_data) { + pdata = pdev->dev.platform_data; + return pdata; + } else if (match) { + pdata = devm_kzalloc(&pdev->dev, sizeof(*pdata), GFP_KERNEL); + if (!pdata) { + ret = -ENOMEM; + goto nodata; + } + } else { + /* control shouldn't reach here. something is wrong */ + ret = -EINVAL; + goto nodata; + } + + if (match->data) + pdata->version = (u8)((int)match->data); + + ret = of_property_read_u32(np, "op-mode", &val); + if (ret >= 0) + pdata->op_mode = val; + + ret = of_property_read_u32(np, "tdm-slots", &val); + if (ret >= 0) + pdata->tdm_slots = val; + + ret = of_property_read_u32(np, "num-serializer", &val); + if (ret >= 0) + pdata->num_serializer = val; + + of_serial_dir32 = of_get_property(np, "serial-dir", &val); + val /= sizeof(u32); + if (val != pdata->num_serializer) { + dev_err(&pdev->dev, + "num-serializer(%d) != serial-dir size(%d)\n", + pdata->num_serializer, val); + ret = -EINVAL; + goto nodata; + } + + if (of_serial_dir32) { + of_serial_dir = devm_kzalloc(&pdev->dev, + (sizeof(*of_serial_dir) * val), + GFP_KERNEL); + if (!of_serial_dir) { + ret = -ENOMEM; + goto nodata; + } + + for (i = 0; i < pdata->num_serializer; i++) + of_serial_dir[i] = be32_to_cpup(&of_serial_dir32[i]); + + pdata->serial_dir = of_serial_dir; + } + + ret = of_property_read_u32(np, "tx-num-evt", &val); + if (ret >= 0) + pdata->txnumevt = val; + + ret = of_property_read_u32(np, "rx-num-evt", &val); + if (ret >= 0) + pdata->rxnumevt = val; + + ret = of_property_read_u32(np, "sram-size-playback", &val); + if (ret >= 0) + pdata->sram_size_playback = val; + + ret = of_property_read_u32(np, "sram-size-capture", &val); + if (ret >= 0) + pdata->sram_size_capture = val; + + return pdata; + +nodata: + if (ret < 0) { + dev_err(&pdev->dev, "Error populating platform data, err %d\n", + ret); + pdata = NULL; + } + return pdata; +} + static int davinci_mcasp_probe(struct platform_device *pdev) { struct davinci_pcm_dma_params *dma_data; @@ -864,11 +975,22 @@ static int davinci_mcasp_probe(struct platform_device *pdev) struct davinci_audio_dev *dev; int ret; + if (!pdev->dev.platform_data && !pdev->dev.of_node) { + dev_err(&pdev->dev, "No platform data supplied\n"); + return -EINVAL; + } + dev = devm_kzalloc(&pdev->dev, sizeof(struct davinci_audio_dev), GFP_KERNEL); if (!dev) return -ENOMEM; + pdata = davinci_mcasp_set_pdata_from_of(pdev); + if (!pdata) { + dev_err(&pdev->dev, "no platform data\n"); + return -EINVAL; + } + mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); if (!mem) { dev_err(&pdev->dev, "no mem resource?\n"); @@ -882,7 +1004,6 @@ static int davinci_mcasp_probe(struct platform_device *pdev) return -EBUSY; } - pdata = pdev->dev.platform_data; pm_runtime_enable(&pdev->dev); ret = pm_runtime_get_sync(&pdev->dev); @@ -980,6 +1101,7 @@ static struct platform_driver davinci_mcasp_driver = { .driver = { .name = "davinci-mcasp", .owner = THIS_MODULE, + .of_match_table = of_match_ptr(mcasp_dt_ids), }, }; -- cgit v1.2.3-58-ga151 From e5ec69da24803c68f5c035662a68d367359a4132 Mon Sep 17 00:00:00 2001 From: "Hebbar, Gururaja" Date: Mon, 3 Sep 2012 13:40:40 +0530 Subject: ASoC: Davinci: McASP: add support new McASP IP Variant The OMAP2+ variant of McASP is different from Davinci variant w.r.to some register offset. Changes - Add new MCASP_VERSION_3 to identify new variant. New DT compatible "ti,omap2-mcasp-audio" to identify version 3 controller. - The register offsets are handled depending on the version. Note: DMA parameters (dma fifo offset) are not updated and will be done later. Signed-off-by: Hebbar, Gururaja Signed-off-by: Mark Brown --- .../bindings/sound/davinci-mcasp-audio.txt | 1 + include/linux/platform_data/davinci_asp.h | 1 + sound/soc/davinci/davinci-mcasp.c | 86 ++++++++++++++++++---- 3 files changed, 75 insertions(+), 13 deletions(-) (limited to 'Documentation') diff --git a/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt b/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt index e6148eca2942..374e145c2ef1 100644 --- a/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt +++ b/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt @@ -4,6 +4,7 @@ Required properties: - compatible : "ti,dm646x-mcasp-audio" : for DM646x platforms "ti,da830-mcasp-audio" : for both DA830 & DA850 platforms + "ti,omap2-mcasp-audio" : for OMAP2 platforms (TI81xx, AM33xx) - reg : Should contain McASP registers offset and length - interrupts : Interrupt number for McASP diff --git a/include/linux/platform_data/davinci_asp.h b/include/linux/platform_data/davinci_asp.h index 79c26aa11db6..d0c5825876f8 100644 --- a/include/linux/platform_data/davinci_asp.h +++ b/include/linux/platform_data/davinci_asp.h @@ -87,6 +87,7 @@ struct snd_platform_data { enum { MCASP_VERSION_1 = 0, /* DM646x */ MCASP_VERSION_2, /* DA8xx/OMAPL1x */ + MCASP_VERSION_3, /* TI81xx/AM33xx */ }; enum mcbsp_clk_input_pin { diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index c3eae1d8e077..714e51e5be5b 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -111,6 +111,10 @@ #define DAVINCI_MCASP_WFIFOSTS (0x1014) #define DAVINCI_MCASP_RFIFOCTL (0x1018) #define DAVINCI_MCASP_RFIFOSTS (0x101C) +#define MCASP_VER3_WFIFOCTL (0x1000) +#define MCASP_VER3_WFIFOSTS (0x1004) +#define MCASP_VER3_RFIFOCTL (0x1008) +#define MCASP_VER3_RFIFOSTS (0x100C) /* * DAVINCI_MCASP_PWREMUMGT_REG - Power Down and Emulation Management @@ -384,18 +388,36 @@ static void davinci_mcasp_start(struct davinci_audio_dev *dev, int stream) { if (stream == SNDRV_PCM_STREAM_PLAYBACK) { if (dev->txnumevt) { /* enable FIFO */ - mcasp_clr_bits(dev->base + DAVINCI_MCASP_WFIFOCTL, + switch (dev->version) { + case MCASP_VERSION_3: + mcasp_clr_bits(dev->base + MCASP_VER3_WFIFOCTL, FIFO_ENABLE); - mcasp_set_bits(dev->base + DAVINCI_MCASP_WFIFOCTL, + mcasp_set_bits(dev->base + MCASP_VER3_WFIFOCTL, FIFO_ENABLE); + break; + default: + mcasp_clr_bits(dev->base + + DAVINCI_MCASP_WFIFOCTL, FIFO_ENABLE); + mcasp_set_bits(dev->base + + DAVINCI_MCASP_WFIFOCTL, FIFO_ENABLE); + } } mcasp_start_tx(dev); } else { if (dev->rxnumevt) { /* enable FIFO */ - mcasp_clr_bits(dev->base + DAVINCI_MCASP_RFIFOCTL, + switch (dev->version) { + case MCASP_VERSION_3: + mcasp_clr_bits(dev->base + MCASP_VER3_RFIFOCTL, FIFO_ENABLE); - mcasp_set_bits(dev->base + DAVINCI_MCASP_RFIFOCTL, + mcasp_set_bits(dev->base + MCASP_VER3_RFIFOCTL, FIFO_ENABLE); + break; + default: + mcasp_clr_bits(dev->base + + DAVINCI_MCASP_RFIFOCTL, FIFO_ENABLE); + mcasp_set_bits(dev->base + + DAVINCI_MCASP_RFIFOCTL, FIFO_ENABLE); + } } mcasp_start_rx(dev); } @@ -416,14 +438,31 @@ static void mcasp_stop_tx(struct davinci_audio_dev *dev) static void davinci_mcasp_stop(struct davinci_audio_dev *dev, int stream) { if (stream == SNDRV_PCM_STREAM_PLAYBACK) { - if (dev->txnumevt) /* disable FIFO */ - mcasp_clr_bits(dev->base + DAVINCI_MCASP_WFIFOCTL, + if (dev->txnumevt) { /* disable FIFO */ + switch (dev->version) { + case MCASP_VERSION_3: + mcasp_clr_bits(dev->base + MCASP_VER3_WFIFOCTL, FIFO_ENABLE); + break; + default: + mcasp_clr_bits(dev->base + + DAVINCI_MCASP_WFIFOCTL, FIFO_ENABLE); + } + } mcasp_stop_tx(dev); } else { - if (dev->rxnumevt) /* disable FIFO */ - mcasp_clr_bits(dev->base + DAVINCI_MCASP_RFIFOCTL, + if (dev->rxnumevt) { /* disable FIFO */ + switch (dev->version) { + case MCASP_VERSION_3: + mcasp_clr_bits(dev->base + MCASP_VER3_RFIFOCTL, FIFO_ENABLE); + break; + + default: + mcasp_clr_bits(dev->base + + DAVINCI_MCASP_RFIFOCTL, FIFO_ENABLE); + } + } mcasp_stop_rx(dev); } } @@ -622,20 +661,37 @@ static void davinci_hw_common_param(struct davinci_audio_dev *dev, int stream) if (dev->txnumevt * tx_ser > 64) dev->txnumevt = 1; - mcasp_mod_bits(dev->base + DAVINCI_MCASP_WFIFOCTL, tx_ser, + switch (dev->version) { + case MCASP_VERSION_3: + mcasp_mod_bits(dev->base + MCASP_VER3_WFIFOCTL, tx_ser, NUMDMA_MASK); - mcasp_mod_bits(dev->base + DAVINCI_MCASP_WFIFOCTL, + mcasp_mod_bits(dev->base + MCASP_VER3_WFIFOCTL, ((dev->txnumevt * tx_ser) << 8), NUMEVT_MASK); + break; + default: + mcasp_mod_bits(dev->base + DAVINCI_MCASP_WFIFOCTL, + tx_ser, NUMDMA_MASK); + mcasp_mod_bits(dev->base + DAVINCI_MCASP_WFIFOCTL, + ((dev->txnumevt * tx_ser) << 8), NUMEVT_MASK); + } } if (dev->rxnumevt && stream == SNDRV_PCM_STREAM_CAPTURE) { if (dev->rxnumevt * rx_ser > 64) dev->rxnumevt = 1; - - mcasp_mod_bits(dev->base + DAVINCI_MCASP_RFIFOCTL, rx_ser, + switch (dev->version) { + case MCASP_VERSION_3: + mcasp_mod_bits(dev->base + MCASP_VER3_RFIFOCTL, rx_ser, NUMDMA_MASK); - mcasp_mod_bits(dev->base + DAVINCI_MCASP_RFIFOCTL, + mcasp_mod_bits(dev->base + MCASP_VER3_RFIFOCTL, + ((dev->rxnumevt * rx_ser) << 8), NUMEVT_MASK); + break; + default: + mcasp_mod_bits(dev->base + DAVINCI_MCASP_RFIFOCTL, + rx_ser, NUMDMA_MASK); + mcasp_mod_bits(dev->base + DAVINCI_MCASP_RFIFOCTL, ((dev->rxnumevt * rx_ser) << 8), NUMEVT_MASK); + } } } @@ -874,6 +930,10 @@ static const struct of_device_id mcasp_dt_ids[] = { .compatible = "ti,da830-mcasp-audio", .data = (void *)MCASP_VERSION_2, }, + { + .compatible = "ti,omap2-mcasp-audio", + .data = (void *)MCASP_VERSION_3, + }, { /* sentinel */ } }; MODULE_DEVICE_TABLE(of, mcasp_dt_ids); -- cgit v1.2.3-58-ga151 From be84bbcccc757b86449daaf924e72f95c95dc00e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 15 Aug 2012 16:06:43 +0200 Subject: ALSA: Add a documentation for channel mapping API Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/Channel-Mapping-API.txt | 144 +++++++++++++++++++++++ 1 file changed, 144 insertions(+) create mode 100644 Documentation/sound/alsa/Channel-Mapping-API.txt (limited to 'Documentation') diff --git a/Documentation/sound/alsa/Channel-Mapping-API.txt b/Documentation/sound/alsa/Channel-Mapping-API.txt new file mode 100644 index 000000000000..df930aa4f4d0 --- /dev/null +++ b/Documentation/sound/alsa/Channel-Mapping-API.txt @@ -0,0 +1,144 @@ +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_FL, /* front left */ + SNDRV_CHMAP_FC, /* front center */ + SNDRV_CHMAP_FR, /* front right */ + SNDRV_CHMAP_FLC, /* front left center */ + SNDRV_CHMAP_FRC, /* front right center */ + SNDRV_CHMAP_RL, /* rear left */ + SNDRV_CHMAP_RC, /* rear center */ + SNDRV_CHMAP_RR, /* rear right */ + SNDRV_CHMAP_RLC, /* rear left center */ + SNDRV_CHMAP_RRC, /* rear right center */ + SNDRV_CHMAP_SL, /* side left */ + SNDRV_CHMAP_SR, /* side right */ + SNDRV_CHMAP_LFE, /* LFE */ + 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_NA, /* N/A, silent */ + SNDRV_CHMAP_LAST = SNDRV_CHMAP_NA, +}; + +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. -- cgit v1.2.3-58-ga151 From 080108c4747c7378c3601b8584237484f977d8a8 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 21 Aug 2012 14:47:18 +0200 Subject: ALSA: Follow channel position definitions to alsa-lib mixer There is already a set of channel position definitions in alsa-lib mixer.h, and it'd be more practical to keep the same order for the PCM channel map, too. The value is shifted with 1 to keep zero for UNKNOWN. Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/Channel-Mapping-API.txt | 16 +++++++++------- include/sound/asound.h | 16 +++++++++------- 2 files changed, 18 insertions(+), 14 deletions(-) (limited to 'Documentation') diff --git a/Documentation/sound/alsa/Channel-Mapping-API.txt b/Documentation/sound/alsa/Channel-Mapping-API.txt index df930aa4f4d0..4bbf12d5553f 100644 --- a/Documentation/sound/alsa/Channel-Mapping-API.txt +++ b/Documentation/sound/alsa/Channel-Mapping-API.txt @@ -76,20 +76,22 @@ here is a cut: /* channel positions */ enum { + /* this follows the alsa-lib mixer channel value + 1 */ SNDRV_CHMAP_UNKNOWN = 0, SNDRV_CHMAP_FL, /* front left */ - SNDRV_CHMAP_FC, /* front center */ SNDRV_CHMAP_FR, /* front right */ - SNDRV_CHMAP_FLC, /* front left center */ - SNDRV_CHMAP_FRC, /* front right center */ SNDRV_CHMAP_RL, /* rear left */ - SNDRV_CHMAP_RC, /* rear center */ SNDRV_CHMAP_RR, /* rear right */ - SNDRV_CHMAP_RLC, /* rear left center */ - SNDRV_CHMAP_RRC, /* rear right center */ + SNDRV_CHMAP_FC, /* front center */ + SNDRV_CHMAP_LFE, /* LFE */ SNDRV_CHMAP_SL, /* side left */ SNDRV_CHMAP_SR, /* side right */ - SNDRV_CHMAP_LFE, /* LFE */ + 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 */ diff --git a/include/sound/asound.h b/include/sound/asound.h index 376e75632e07..27686da0f650 100644 --- a/include/sound/asound.h +++ b/include/sound/asound.h @@ -474,20 +474,22 @@ enum { /* channel positions */ enum { + /* this follows the alsa-lib mixer channel value + 1 */ SNDRV_CHMAP_UNKNOWN = 0, SNDRV_CHMAP_FL, /* front left */ - SNDRV_CHMAP_FC, /* front center */ SNDRV_CHMAP_FR, /* front right */ - SNDRV_CHMAP_FLC, /* front left center */ - SNDRV_CHMAP_FRC, /* front right center */ SNDRV_CHMAP_RL, /* rear left */ - SNDRV_CHMAP_RC, /* rear center */ SNDRV_CHMAP_RR, /* rear right */ - SNDRV_CHMAP_RLC, /* rear left center */ - SNDRV_CHMAP_RRC, /* rear right center */ + SNDRV_CHMAP_FC, /* front center */ + SNDRV_CHMAP_LFE, /* LFE */ SNDRV_CHMAP_SL, /* side left */ SNDRV_CHMAP_SR, /* side right */ - SNDRV_CHMAP_LFE, /* LFE */ + 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 */ -- cgit v1.2.3-58-ga151 From 7b31d0095e87221dc32c95642a2a714ea08259aa Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 12 Sep 2012 18:06:54 +0200 Subject: ALSA: Define more channel map positions For following the standard, define more channel map positions and shuffle the items a bit: - As both PulseAudio and gstreamer define MONO channel position explicitly, we should follow that, too. The mono streams point to this channel position unless they are explicitly assigned to certain channel positions. - Top-front-* and Top-rear-* positions are added, carried from PulseAudio's definitions. - Move NA and MONO definitions at the top of table right after UNKNOWN, since these are more abstract in comparison with other practical positions. Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/Channel-Mapping-API.txt | 13 ++++++++++--- include/sound/asound.h | 13 ++++++++++--- sound/pci/ctxfi/ctpcm.c | 6 +++--- 3 files changed, 23 insertions(+), 9 deletions(-) (limited to 'Documentation') diff --git a/Documentation/sound/alsa/Channel-Mapping-API.txt b/Documentation/sound/alsa/Channel-Mapping-API.txt index 4bbf12d5553f..3c43d1a4ca0e 100644 --- a/Documentation/sound/alsa/Channel-Mapping-API.txt +++ b/Documentation/sound/alsa/Channel-Mapping-API.txt @@ -76,8 +76,10 @@ here is a cut: /* channel positions */ enum { - /* this follows the alsa-lib mixer channel value + 1 */ 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 */ @@ -98,8 +100,13 @@ enum { SNDRV_CHMAP_FCH, /* front center high */ SNDRV_CHMAP_FRH, /* front right high */ SNDRV_CHMAP_TC, /* top center */ - SNDRV_CHMAP_NA, /* N/A, silent */ - SNDRV_CHMAP_LAST = SNDRV_CHMAP_NA, + 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 diff --git a/include/sound/asound.h b/include/sound/asound.h index 27686da0f650..dfe7d441748c 100644 --- a/include/sound/asound.h +++ b/include/sound/asound.h @@ -474,8 +474,10 @@ enum { /* channel positions */ enum { - /* this follows the alsa-lib mixer channel value + 1 */ 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 */ @@ -496,8 +498,13 @@ enum { SNDRV_CHMAP_FCH, /* front center high */ SNDRV_CHMAP_FRH, /* front right high */ SNDRV_CHMAP_TC, /* top center */ - SNDRV_CHMAP_NA, /* N/A, silent */ - SNDRV_CHMAP_LAST = SNDRV_CHMAP_NA, + 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, }; #define SNDRV_CHMAP_POSITION_MASK 0xffff diff --git a/sound/pci/ctxfi/ctpcm.c b/sound/pci/ctxfi/ctpcm.c index d317107d98cc..e8a4feb1ed86 100644 --- a/sound/pci/ctxfi/ctpcm.c +++ b/sound/pci/ctxfi/ctpcm.c @@ -397,7 +397,7 @@ static struct snd_pcm_ops ct_pcm_capture_ops = { static const struct snd_pcm_chmap_elem surround_map[] = { { .channels = 1, - .map = { SNDRV_CHMAP_UNKNOWN } }, + .map = { SNDRV_CHMAP_MONO } }, { .channels = 2, .map = { SNDRV_CHMAP_RL, SNDRV_CHMAP_RR } }, { } @@ -405,7 +405,7 @@ static const struct snd_pcm_chmap_elem surround_map[] = { static const struct snd_pcm_chmap_elem clfe_map[] = { { .channels = 1, - .map = { SNDRV_CHMAP_UNKNOWN } }, + .map = { SNDRV_CHMAP_MONO } }, { .channels = 2, .map = { SNDRV_CHMAP_FC, SNDRV_CHMAP_LFE } }, { } @@ -413,7 +413,7 @@ static const struct snd_pcm_chmap_elem clfe_map[] = { static const struct snd_pcm_chmap_elem side_map[] = { { .channels = 1, - .map = { SNDRV_CHMAP_UNKNOWN } }, + .map = { SNDRV_CHMAP_MONO } }, { .channels = 2, .map = { SNDRV_CHMAP_SL, SNDRV_CHMAP_SR } }, { } -- cgit v1.2.3-58-ga151 From 1dac6695c683c66d0cff10a84c6ed10dbbaabc18 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 13 Sep 2012 14:59:47 +0200 Subject: ALSA: hda - Allow to pass position_fix=0 explicitly Set the default value of position_fix -1, and allow user passing position_fix=0 explicitly to set the "auto" position-fix mode. Otherwise the auto mode may be switched to others like COMBO of VIACOMBO when the controller prefers it, thus user can't set the auto mode any longer. Also updated the documentation appropriately, too. Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/ALSA-Configuration.txt | 10 ++++++++-- sound/pci/hda/hda_intel.c | 5 +++-- 2 files changed, 11 insertions(+), 4 deletions(-) (limited to 'Documentation') diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index 4e4d0bc9816f..d90d8ec2853d 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -860,8 +860,14 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. [Multiple options for each card instance] model - force the model name - position_fix - Fix DMA pointer (0 = auto, 1 = use LPIB, 2 = POSBUF, - 3 = VIACOMBO, 4 = COMBO) + 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 diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index e1a12c754de9..195d84726187 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -64,7 +64,7 @@ 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; static char *model[SNDRV_CARDS]; -static int position_fix[SNDRV_CARDS]; +static int position_fix[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS-1)] = -1}; static int bdl_pos_adj[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS-1)] = -1}; static int probe_mask[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS-1)] = -1}; static int probe_only[SNDRV_CARDS]; @@ -88,7 +88,7 @@ module_param_array(model, charp, NULL, 0444); MODULE_PARM_DESC(model, "Use the given board model."); module_param_array(position_fix, int, NULL, 0444); MODULE_PARM_DESC(position_fix, "DMA pointer read method." - "(0 = auto, 1 = LPIB, 2 = POSBUF, 3 = VIACOMBO, 4 = COMBO)."); + "(-1 = system default, 0 = auto, 1 = LPIB, 2 = POSBUF, 3 = VIACOMBO, 4 = COMBO)."); module_param_array(bdl_pos_adj, int, NULL, 0644); MODULE_PARM_DESC(bdl_pos_adj, "BDL position adjustment offset."); module_param_array(probe_mask, int, NULL, 0444); @@ -2813,6 +2813,7 @@ static int __devinit check_position_fix(struct azx *chip, int fix) const struct snd_pci_quirk *q; switch (fix) { + case POS_FIX_AUTO: case POS_FIX_LPIB: case POS_FIX_POSBUF: case POS_FIX_VIACOMBO: -- cgit v1.2.3-58-ga151 From 7bf7ff6f57dcb30c80ad1b65cebdebf4feb3c666 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Mon, 10 Sep 2012 13:46:25 +0300 Subject: mfd: twl4030-audio: Add DT support Support for loading the twl4030 audio module via devicetree. Sub devices for codec and vibra will be created as mfd devices once the core MFD driver is loaded when the kernel is booted with a DT blob. Signed-off-by: Peter Ujfalusi Acked-by: Samuel Ortiz Signed-off-by: Mark Brown --- .../devicetree/bindings/mfd/twl4030-audio.txt | 46 ++++++++++++++++++ drivers/mfd/twl4030-audio.c | 54 +++++++++++++++++++--- 2 files changed, 93 insertions(+), 7 deletions(-) create mode 100644 Documentation/devicetree/bindings/mfd/twl4030-audio.txt (limited to 'Documentation') diff --git a/Documentation/devicetree/bindings/mfd/twl4030-audio.txt b/Documentation/devicetree/bindings/mfd/twl4030-audio.txt new file mode 100644 index 000000000000..414d2ae0adf6 --- /dev/null +++ b/Documentation/devicetree/bindings/mfd/twl4030-audio.txt @@ -0,0 +1,46 @@ +Texas Instruments TWL family (twl4030) audio module + +The audio module inside the TWL family consist of an audio codec and a vibra +driver. + +Required properties: +- compatible : must be "ti,twl4030-audio" + +Optional properties, nodes: + +Audio functionality: +- codec { }: Need to be present if the audio functionality is used. Within this + section the following options can be used: +- ti,digimic_delay: Delay need after enabling the digimic to reduce artifacts + from the start of the recorded sample (in ms) +-ti,ramp_delay_value: HS ramp delay configuration to reduce pop noise +-ti,hs_extmute: Use external mute for HS pop reduction +-ti,hs_extmute_gpio: Use external GPIO to control the external mute +-ti,offset_cncl_path: Offset cancellation path selection, refer to TRM for the + valid values. + +Vibra functionality +- ti,enable-vibra: Need to be set to <1> if the vibra functionality is used. if + missing or it is 0, the vibra functionality is disabled. + +Example: +&i2c1 { + clock-frequency = <2600000>; + + twl: twl@48 { + reg = <0x48>; + interrupts = <7>; /* SYS_NIRQ cascaded to intc */ + interrupt-parent = <&intc>; + + twl_audio: audio { + compatible = "ti,twl4030-audio"; + + ti,enable-vibra = <1>; + + codec { + ti,ramp_delay_value = <3>; + }; + + }; + }; +}; diff --git a/drivers/mfd/twl4030-audio.c b/drivers/mfd/twl4030-audio.c index a48bf3af1240..58e6c228653b 100644 --- a/drivers/mfd/twl4030-audio.c +++ b/drivers/mfd/twl4030-audio.c @@ -28,6 +28,8 @@ #include #include #include +#include +#include #include #include #include @@ -156,15 +158,42 @@ unsigned int twl4030_audio_get_mclk(void) } EXPORT_SYMBOL_GPL(twl4030_audio_get_mclk); +static bool twl4030_audio_has_codec(struct twl4030_audio_data *pdata, + struct device_node *node) +{ + if (pdata && pdata->codec) + return true; + + if (of_find_node_by_name(node, "codec")) + return true; + + return false; +} + +static bool twl4030_audio_has_vibra(struct twl4030_audio_data *pdata, + struct device_node *node) +{ + int vibra; + + if (pdata && pdata->vibra) + return true; + + if (!of_property_read_u32(node, "ti,enable-vibra", &vibra) && vibra) + return true; + + return false; +} + static int __devinit twl4030_audio_probe(struct platform_device *pdev) { struct twl4030_audio *audio; struct twl4030_audio_data *pdata = pdev->dev.platform_data; + struct device_node *node = pdev->dev.of_node; struct mfd_cell *cell = NULL; int ret, childs = 0; u8 val; - if (!pdata) { + if (!pdata && !node) { dev_err(&pdev->dev, "Platform data is missing\n"); return -EINVAL; } @@ -202,18 +231,22 @@ static int __devinit twl4030_audio_probe(struct platform_device *pdev) audio->resource[TWL4030_AUDIO_RES_APLL].reg = TWL4030_REG_APLL_CTL; audio->resource[TWL4030_AUDIO_RES_APLL].mask = TWL4030_APLL_EN; - if (pdata->codec) { + if (twl4030_audio_has_codec(pdata, node)) { cell = &audio->cells[childs]; cell->name = "twl4030-codec"; - cell->platform_data = pdata->codec; - cell->pdata_size = sizeof(*pdata->codec); + if (pdata) { + cell->platform_data = pdata->codec; + cell->pdata_size = sizeof(*pdata->codec); + } childs++; } - if (pdata->vibra) { + if (twl4030_audio_has_vibra(pdata, node)) { cell = &audio->cells[childs]; cell->name = "twl4030-vibra"; - cell->platform_data = pdata->vibra; - cell->pdata_size = sizeof(*pdata->vibra); + if (pdata) { + cell->platform_data = pdata->vibra; + cell->pdata_size = sizeof(*pdata->vibra); + } childs++; } @@ -245,10 +278,17 @@ static int __devexit twl4030_audio_remove(struct platform_device *pdev) return 0; } +static const struct of_device_id twl4030_audio_of_match[] = { + {.compatible = "ti,twl4030-audio", }, + { }, +}; +MODULE_DEVICE_TABLE(of, twl4030_audio_of_match); + static struct platform_driver twl4030_audio_driver = { .driver = { .owner = THIS_MODULE, .name = "twl4030-audio", + .of_match_table = twl4030_audio_of_match, }, .probe = twl4030_audio_probe, .remove = __devexit_p(twl4030_audio_remove), -- cgit v1.2.3-58-ga151 From a31ebc349dade4e6a7a27e88669f20dbc6f8a3b8 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Fri, 28 Sep 2012 01:36:44 +0200 Subject: ALSA: ASoC: add DT bindings for CS4271 Apart from pure matching, the bindings also support setting the the reset gpio line. Signed-off-by: Daniel Mack Cc: Alexander Sverdlin Signed-off-by: Mark Brown --- Documentation/devicetree/bindings/sound/cs4271.txt | 36 ++++++++++++++++++++++ sound/soc/codecs/cs4271.c | 24 +++++++++++++-- 2 files changed, 57 insertions(+), 3 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/cs4271.txt (limited to 'Documentation') diff --git a/Documentation/devicetree/bindings/sound/cs4271.txt b/Documentation/devicetree/bindings/sound/cs4271.txt new file mode 100644 index 000000000000..c81b5fd5a5bc --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cs4271.txt @@ -0,0 +1,36 @@ +Cirrus Logic CS4271 DT bindings + +This driver supports both the I2C and the SPI bus. + +Required properties: + + - compatible: "cirrus,cs4271" + +For required properties on SPI, please consult +Documentation/devicetree/bindings/spi/spi-bus.txt + +Required properties on I2C: + + - reg: the i2c address + + +Optional properties: + + - reset-gpio: a GPIO spec to define which pin is connected to the chip's + !RESET pin + +Examples: + + codec_i2c: cs4271@10 { + compatible = "cirrus,cs4271"; + reg = <0x10>; + reset-gpio = <&gpio 23 0>; + }; + + codec_spi: cs4271@0 { + compatible = "cirrus,cs4271"; + reg = <0x0>; + reset-gpio = <&gpio 23 0>; + spi-max-frequency = <6000000>; + }; + diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c index 9eb01d7d58a3..f994af34f552 100644 --- a/sound/soc/codecs/cs4271.c +++ b/sound/soc/codecs/cs4271.c @@ -22,12 +22,14 @@ #include #include #include -#include -#include -#include #include #include #include +#include +#include +#include +#include +#include #include #define CS4271_PCM_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ @@ -458,6 +460,14 @@ static int cs4271_soc_resume(struct snd_soc_codec *codec) #define cs4271_soc_resume NULL #endif /* CONFIG_PM */ +#ifdef CONFIG_OF +static const struct of_device_id cs4271_dt_ids[] = { + { .compatible = "cirrus,cs4271", }, + { } +}; +MODULE_DEVICE_TABLE(of, cs4271_dt_ids); +#endif + static int cs4271_probe(struct snd_soc_codec *codec) { struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec); @@ -465,6 +475,12 @@ static int cs4271_probe(struct snd_soc_codec *codec) int ret; int gpio_nreset = -EINVAL; +#ifdef CONFIG_OF + if (of_match_device(cs4271_dt_ids, codec->dev)) + gpio_nreset = of_get_named_gpio(codec->dev->of_node, + "reset-gpio", 0); +#endif + if (cs4271plat && gpio_is_valid(cs4271plat->gpio_nreset)) gpio_nreset = cs4271plat->gpio_nreset; @@ -569,6 +585,7 @@ static struct spi_driver cs4271_spi_driver = { .driver = { .name = "cs4271", .owner = THIS_MODULE, + .of_match_table = of_match_ptr(cs4271_dt_ids), }, .probe = cs4271_spi_probe, .remove = __devexit_p(cs4271_spi_remove), @@ -608,6 +625,7 @@ static struct i2c_driver cs4271_i2c_driver = { .driver = { .name = "cs4271", .owner = THIS_MODULE, + .of_match_table = of_match_ptr(cs4271_dt_ids), }, .id_table = cs4271_i2c_id, .probe = cs4271_i2c_probe, -- cgit v1.2.3-58-ga151