diff options
| author | Linus Torvalds <torvalds@linux-foundation.org> | 2023-08-30 13:45:05 -0700 |
|---|---|---|
| committer | Linus Torvalds <torvalds@linux-foundation.org> | 2023-08-30 13:45:05 -0700 |
| commit | 4fb0dacb78c6a041bbd38ddd998df806af5c2c69 (patch) | |
| tree | f175caac9bc11c78632959a954a6b618fbf8f5c1 /Documentation | |
| parent | 461f35f014466c4e26dca6be0f431f57297df3f2 (diff) | |
| parent | 358040e3807754944dbddf948a23c6d914297ed7 (diff) | |
Merge tag 'sound-6.6-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound updates from Takashi Iwai:
"We've received a fairly wide range of changes at this time, including
for ALSA and ASoC core, but all of them are rather small changes.
Here are some highlights:
ALSA / ASoC Core:
- Fixes of inconsistent locking around control API helpers
- A few new control API functions and cleanups
- Workarounds for potential UAFs by delayed kobj releases
- Unified PCM copy ops with iov_iter
- Continued efforts for ASoC API cleanups
ASoC:
- An adaptor to allow use of IIO DACs and ADCs in ASoC which pulls in
some IIO changes
- Create a library function for intlog10() and use it in the NAU8825
driver
- Convert drivers to use the more modern maple tree register cache
- Lots of work on the SOF framework, AMD and Intel drivers, including
a lot of cleanup and new device support
- Standardization of the presentation of jacks from drivers
- Provision of some generic sound card DT properties
- Support for AMD Van Gogh, AMD machines with MAX98388 and NAU8821,
AWInic AW88261, Cirrus Logic CS35L36 and CS42L43, various Intel
platforms including AVS machines with ES8336 and RT5663, Mediatek
MT7986, NXP i.MX93, RealTek RT1017 and StarFive JH7110
Others:
- New test coverage including ASoC and topology tests in KUnit; this
also involves enabling UML builds of ALSA since that's the default
KUnit test environment which pulls in the addition of some stubs to
the driver
- More enhancement of pcmtest driver
- A few fixes / enhancements of MIDI 2.0 UMP core
- Using PCI definitions in allover HD-audio code
- Support for Cirrus CS35L56 and TI TAS2781 HD-audio sub-codecs
- CS35L41 HD-audio sub-codec improvements
- Continued emu10k1 improvements"
* tag 'sound-6.6-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (693 commits)
ALSA: pcm: Fix missing fixup call in compat hw_refine ioctl
ASoC: dwc: i2s: Fix unused functions
ALSA: usb-audio: Don't try to submit URBs after disconnection
ALSA: emu10k1: add separate documentation for E-MU cards
ALSA: emu10k1: more documentation updates
ALSA: emu10k1: de-duplicate audigy-mixer.rst vs. sb-live-mixer.rst
ALSA: ump: Fix -Wformat-truncation warnings
ALSA: hda: Add missing dependency on CONFIG_EFI for Cirrus/TI sub-codecs
ALSA: doc: Fix missing backquote in midi-2.0.rst
ALSA: hda/realtek: Add quirk for mute LEDs on HP ENVY x360 15-eu0xxx
ALSA: hda/tas2781: Switch back to use struct i2c_driver's .probe()
ASoC: soc-core.c: Do not error if a DAI link component is not found
ASoC: codecs: Fix error code in aw88261_i2c_probe()
ASoC: audio-graph-card.c: move audio_graph_parse_of()
ASoC: cs42l43: Use new-style PM runtime macros
ALSA: documentation: Add description for USB MIDI 2.0 gadget driver
ALSA: ump: Don't create unused substreams for static blocks
ALSA: ump: Fill group names for legacy rawmidi substreams
ALSA: usb-audio: Attach legacy rawmidi after probing all UMP EPs
ALSA: ac97: Fix possible error value of *rac97
...
Diffstat (limited to 'Documentation')
40 files changed, 1626 insertions, 597 deletions
diff --git a/Documentation/core-api/kernel-api.rst b/Documentation/core-api/kernel-api.rst index f2bcc5a7ea43..ae92a2571388 100644 --- a/Documentation/core-api/kernel-api.rst +++ b/Documentation/core-api/kernel-api.rst @@ -162,8 +162,10 @@ Base 2 log and power Functions .. kernel-doc:: include/linux/log2.h :internal: -Integer power Functions ------------------------ +Integer log and power Functions +------------------------------- + +.. kernel-doc:: include/linux/int_log.h .. kernel-doc:: lib/math/int_pow.c :export: diff --git a/Documentation/devicetree/bindings/sound/amlogic,axg-sound-card.yaml b/Documentation/devicetree/bindings/sound/amlogic,axg-sound-card.yaml index bf1234550343..5db718e4d0e7 100644 --- a/Documentation/devicetree/bindings/sound/amlogic,axg-sound-card.yaml +++ b/Documentation/devicetree/bindings/sound/amlogic,axg-sound-card.yaml @@ -9,6 +9,9 @@ title: Amlogic AXG sound card maintainers: - Jerome Brunet <jbrunet@baylibre.com> +allOf: + - $ref: sound-card-common.yaml# + properties: compatible: const: amlogic,axg-sound-card @@ -17,23 +20,12 @@ properties: $ref: /schemas/types.yaml#/definitions/phandle-array description: list of auxiliary devices - audio-routing: - $ref: /schemas/types.yaml#/definitions/non-unique-string-array - description: - A list of the 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. - audio-widgets: $ref: /schemas/types.yaml#/definitions/non-unique-string-array description: A list off component DAPM widget. Each entry is a pair of strings, the first being the widget type, the second being the widget name - model: - $ref: /schemas/types.yaml#/definitions/string - description: User specified audio sound card name - patternProperties: "^dai-link-[0-9]+$": type: object @@ -108,7 +100,6 @@ patternProperties: - sound-dai required: - - model - dai-link-0 unevaluatedProperties: false diff --git a/Documentation/devicetree/bindings/sound/amlogic,gx-sound-card.yaml b/Documentation/devicetree/bindings/sound/amlogic,gx-sound-card.yaml index b358fd601ed3..d4277d342e69 100644 --- a/Documentation/devicetree/bindings/sound/amlogic,gx-sound-card.yaml +++ b/Documentation/devicetree/bindings/sound/amlogic,gx-sound-card.yaml @@ -9,6 +9,9 @@ title: Amlogic GX sound card maintainers: - Jerome Brunet <jbrunet@baylibre.com> +allOf: + - $ref: sound-card-common.yaml# + properties: compatible: items: @@ -18,14 +21,6 @@ properties: $ref: /schemas/types.yaml#/definitions/phandle-array description: list of auxiliary devices - audio-routing: - $ref: /schemas/types.yaml#/definitions/non-unique-string-array - minItems: 2 - description: |- - A list of the 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. - audio-widgets: $ref: /schemas/types.yaml#/definitions/non-unique-string-array minItems: 2 @@ -33,10 +28,6 @@ properties: A list off component DAPM widget. Each entry is a pair of strings, the first being the widget type, the second being the widget name - model: - $ref: /schemas/types.yaml#/definitions/string - description: User specified audio sound card name - patternProperties: "^dai-link-[0-9]+$": type: object @@ -86,7 +77,7 @@ required: - model - dai-link-0 -additionalProperties: false +unevaluatedProperties: false examples: - | diff --git a/Documentation/devicetree/bindings/sound/audio-iio-aux.yaml b/Documentation/devicetree/bindings/sound/audio-iio-aux.yaml new file mode 100644 index 000000000000..d3cc1ea4a175 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/audio-iio-aux.yaml @@ -0,0 +1,64 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/audio-iio-aux.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Audio IIO auxiliary + +maintainers: + - Herve Codina <herve.codina@bootlin.com> + +description: + Auxiliary device based on Industrial I/O device channels + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + const: audio-iio-aux + + io-channels: + description: + Industrial I/O device channels used + + io-channel-names: + description: + Industrial I/O channel names related to io-channels. + These names are used to provides sound controls, widgets and routes names. + + snd-control-invert-range: + $ref: /schemas/types.yaml#/definitions/uint32-array + description: | + A list of 0/1 flags defining whether or not the related channel is + inverted + items: + enum: [0, 1] + default: 0 + description: | + Invert the sound control value compared to the IIO channel raw value. + - 1: The related sound control value is inverted meaning that the + minimum sound control value correspond to the maximum IIO channel + raw value and the maximum sound control value correspond to the + minimum IIO channel raw value. + - 0: The related sound control value is not inverted meaning that the + minimum (resp maximum) sound control value correspond to the + minimum (resp maximum) IIO channel raw value. + +required: + - compatible + - io-channels + - io-channel-names + +unevaluatedProperties: false + +examples: + - | + iio-aux { + compatible = "audio-iio-aux"; + io-channels = <&iio 0>, <&iio 1>, <&iio 2>, <&iio 3>; + io-channel-names = "CH0", "CH1", "CH2", "CH3"; + /* Invert CH1 and CH2 */ + snd-control-invert-range = <0 1 1 0>; + }; diff --git a/Documentation/devicetree/bindings/sound/awinic,aw88395.yaml b/Documentation/devicetree/bindings/sound/awinic,aw88395.yaml index 35eef7d818a2..4051c2538caf 100644 --- a/Documentation/devicetree/bindings/sound/awinic,aw88395.yaml +++ b/Documentation/devicetree/bindings/sound/awinic,aw88395.yaml @@ -19,7 +19,9 @@ allOf: properties: compatible: - const: awinic,aw88395 + enum: + - awinic,aw88395 + - awinic,aw88261 reg: maxItems: 1 diff --git a/Documentation/devicetree/bindings/sound/fsl,rpmsg.yaml b/Documentation/devicetree/bindings/sound/fsl,rpmsg.yaml index e847611a85f7..188f38baddec 100644 --- a/Documentation/devicetree/bindings/sound/fsl,rpmsg.yaml +++ b/Documentation/devicetree/bindings/sound/fsl,rpmsg.yaml @@ -17,6 +17,9 @@ description: | such as SAI, MICFIL, .etc through building rpmsg channels between Cortex-A and Cortex-M. +allOf: + - $ref: sound-card-common.yaml# + properties: compatible: enum: @@ -25,10 +28,7 @@ properties: - fsl,imx8mm-rpmsg-audio - fsl,imx8mp-rpmsg-audio - fsl,imx8ulp-rpmsg-audio - - model: - $ref: /schemas/types.yaml#/definitions/string - description: User specified audio sound card name + - fsl,imx93-rpmsg-audio clocks: items: @@ -65,13 +65,6 @@ properties: $ref: /schemas/types.yaml#/definitions/phandle description: The phandle to a node of audio codec - audio-routing: - $ref: /schemas/types.yaml#/definitions/non-unique-string-array - description: | - A list of the 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. - fsl,enable-lpa: $ref: /schemas/types.yaml#/definitions/flag description: enable low power audio path. @@ -100,9 +93,8 @@ properties: required: - compatible - - model -additionalProperties: false +unevaluatedProperties: false examples: - | diff --git a/Documentation/devicetree/bindings/sound/google,sc7180-trogdor.yaml b/Documentation/devicetree/bindings/sound/google,sc7180-trogdor.yaml index ba5b7728cf33..bac940553965 100644 --- a/Documentation/devicetree/bindings/sound/google,sc7180-trogdor.yaml +++ b/Documentation/devicetree/bindings/sound/google,sc7180-trogdor.yaml @@ -13,23 +13,15 @@ maintainers: description: This binding describes the SC7180 sound card which uses LPASS for audio. +allOf: + - $ref: sound-card-common.yaml# + properties: compatible: enum: - google,sc7180-trogdor - google,sc7180-coachz - audio-routing: - $ref: /schemas/types.yaml#/definitions/non-unique-string-array - description: - A list of the 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. - - model: - $ref: /schemas/types.yaml#/definitions/string - description: User specified audio sound card name - "#address-cells": const: 1 @@ -86,11 +78,10 @@ patternProperties: required: - compatible - - model - "#address-cells" - "#size-cells" -additionalProperties: false +unevaluatedProperties: false examples: diff --git a/Documentation/devicetree/bindings/sound/google,sc7280-herobrine.yaml b/Documentation/devicetree/bindings/sound/google,sc7280-herobrine.yaml index 0b1a01a4c14e..ec4b6e547ca6 100644 --- a/Documentation/devicetree/bindings/sound/google,sc7280-herobrine.yaml +++ b/Documentation/devicetree/bindings/sound/google,sc7280-herobrine.yaml @@ -13,22 +13,14 @@ maintainers: description: This binding describes the SC7280 sound card which uses LPASS for audio. +allOf: + - $ref: sound-card-common.yaml# + properties: compatible: enum: - google,sc7280-herobrine - audio-routing: - $ref: /schemas/types.yaml#/definitions/non-unique-string-array - description: - A list of the 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. - - model: - $ref: /schemas/types.yaml#/definitions/string - description: User specified audio sound card name - "#address-cells": const: 1 @@ -97,11 +89,10 @@ patternProperties: required: - compatible - - model - "#address-cells" - "#size-cells" -additionalProperties: false +unevaluatedProperties: false examples: diff --git a/Documentation/devicetree/bindings/sound/gtm601.txt b/Documentation/devicetree/bindings/sound/gtm601.txt deleted file mode 100644 index efa32a486c4a..000000000000 --- a/Documentation/devicetree/bindings/sound/gtm601.txt +++ /dev/null @@ -1,19 +0,0 @@ -GTM601 UMTS modem audio interface CODEC - -This device has no configuration interface. The sample rate and channels are -based on the compatible string - "option,gtm601" = 8kHz mono - "broadmobi,bm818" = 48KHz stereo - -Required properties: - - - compatible : one of - "option,gtm601" - "broadmobi,bm818" - - -Example: - -codec: gtm601_codec { - compatible = "option,gtm601"; -}; diff --git a/Documentation/devicetree/bindings/sound/imx-audio-card.yaml b/Documentation/devicetree/bindings/sound/imx-audio-card.yaml index b6f5d486600e..f7ad5ea2491e 100644 --- a/Documentation/devicetree/bindings/sound/imx-audio-card.yaml +++ b/Documentation/devicetree/bindings/sound/imx-audio-card.yaml @@ -9,23 +9,14 @@ title: NXP i.MX audio sound card. maintainers: - Shengjiu Wang <shengjiu.wang@nxp.com> +allOf: + - $ref: sound-card-common.yaml# + properties: compatible: enum: - fsl,imx-audio-card - model: - $ref: /schemas/types.yaml#/definitions/string - description: User specified audio sound card name - - audio-routing: - $ref: /schemas/types.yaml#/definitions/non-unique-string-array - description: - A list of the connections between audio components. Each entry is a - pair of strings, the first being the connection's sink, the second - being the connection's source. Valid names could be power supplies, - MicBias of codec and the jacks on the board. - patternProperties: ".*-dai-link$": description: @@ -84,9 +75,8 @@ patternProperties: required: - compatible - - model -additionalProperties: false +unevaluatedProperties: false examples: - | diff --git a/Documentation/devicetree/bindings/sound/max9892x.txt b/Documentation/devicetree/bindings/sound/max9892x.txt deleted file mode 100644 index 98cb9ba5b328..000000000000 --- a/Documentation/devicetree/bindings/sound/max9892x.txt +++ /dev/null @@ -1,44 +0,0 @@ -Maxim Integrated MAX98925/MAX98926/MAX98927 Speaker Amplifier - -This device supports I2C. - -Required properties: - - - compatible : should be one of the following - - "maxim,max98925" - - "maxim,max98926" - - "maxim,max98927" - - - vmon-slot-no : slot number used to send voltage information - or in inteleave mode this will be used as - interleave slot. - MAX98925/MAX98926 slot range : 0 ~ 30, Default : 0 - MAX98927 slot range : 0 ~ 15, Default : 0 - - - imon-slot-no : slot number used to send current information - MAX98925/MAX98926 slot range : 0 ~ 30, Default : 0 - MAX98927 slot range : 0 ~ 15, Default : 0 - - - interleave-mode : When using two MAX9892X in a system it is - possible to create ADC data that that will - overflow the frame size. Digital Audio Interleave - mode provides a means to output VMON and IMON data - from two devices on a single DOUT line when running - smaller frames sizes such as 32 BCLKS per LRCLK or - 48 BCLKS per LRCLK. - Range : 0 (off), 1 (on), Default : 0 - - - reg : the I2C address of the device for I2C - -Optional properties: - - reset-gpios : GPIO to reset the device - -Example: - -codec: max98927@3a { - compatible = "maxim,max98927"; - vmon-slot-no = <0>; - imon-slot-no = <1>; - interleave-mode = <0>; - reg = <0x3a>; -}; diff --git a/Documentation/devicetree/bindings/sound/maxim,max98925.yaml b/Documentation/devicetree/bindings/sound/maxim,max98925.yaml new file mode 100644 index 000000000000..32fd86204a7a --- /dev/null +++ b/Documentation/devicetree/bindings/sound/maxim,max98925.yaml @@ -0,0 +1,98 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/maxim,max98925.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Maxim Integrated MAX98925/MAX98926/MAX98927 speaker amplifier + +maintainers: + - Ryan Lee <ryans.lee@maximintegrated.com> + +properties: + compatible: + enum: + - maxim,max98925 + - maxim,max98926 + - maxim,max98927 + + reg: + maxItems: 1 + + reset-gpios: + maxItems: 1 + + '#sound-dai-cells': + const: 0 + + vmon-slot-no: + $ref: /schemas/types.yaml#/definitions/uint32 + minimum: 0 + maximum: 30 + default: 0 + description: + Slot number used to send voltage information or in inteleave mode this + will be used as interleave slot. + + imon-slot-no: + $ref: /schemas/types.yaml#/definitions/uint32 + minimum: 0 + maximum: 30 + default: 0 + description: + Slot number used to send current information. + + maxim,interleave-mode: + type: boolean + description: + When using two MAX9892X in a system it is possible to create ADC data + that will overflow the frame size. When enabled, the Digital Audio + Interleave mode provides a means to output VMON and IMON data from two + devices on a single DOUT line when running smaller frames sizes such as + 32 BCLKS per LRCLK or 48 BCLKS per LRCLK. + +required: + - compatible + - reg + +allOf: + - $ref: dai-common.yaml# + - if: + properties: + compatible: + contains: + enum: + - maxim,max98927 + then: + properties: + vmon-slot-no: + minimum: 0 + maximum: 15 + + imon-slot-no: + minimum: 0 + maximum: 15 + +additionalProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + + #include <dt-bindings/gpio/gpio.h> + audio-codec@3a { + compatible = "maxim,max98927"; + reg = <0x3a>; + #sound-dai-cells = <0>; + + pinctrl-0 = <&speaker_default>; + pinctrl-names = "default"; + + reset-gpios = <&tlmm 69 GPIO_ACTIVE_LOW>; + + vmon-slot-no = <1>; + imon-slot-no = <0>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/mediatek,mt7986-afe.yaml b/Documentation/devicetree/bindings/sound/mediatek,mt7986-afe.yaml new file mode 100644 index 000000000000..398efdfe00f5 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/mediatek,mt7986-afe.yaml @@ -0,0 +1,160 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/mediatek,mt7986-afe.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: MediaTek AFE PCM controller for MT7986 + +maintainers: + - Maso Huang <maso.huang@mediatek.com> + +properties: + compatible: + oneOf: + - const: mediatek,mt7986-afe + - items: + - enum: + - mediatek,mt7981-afe + - mediatek,mt7988-afe + - const: mediatek,mt7986-afe + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + clocks: + minItems: 5 + items: + - description: audio bus clock + - description: audio 26M clock + - description: audio intbus clock + - description: audio hopping clock + - description: audio pll clock + - description: mux for pcm_mck + - description: audio i2s/pcm mck + + clock-names: + minItems: 5 + items: + - const: bus_ck + - const: 26m_ck + - const: l_ck + - const: aud_ck + - const: eg2_ck + - const: sel + - const: i2s_m + +required: + - compatible + - reg + - interrupts + - clocks + - clock-names + +allOf: + - if: + properties: + compatible: + contains: + const: mediatek,mt7986-afe + then: + properties: + clocks: + items: + - description: audio bus clock + - description: audio 26M clock + - description: audio intbus clock + - description: audio hopping clock + - description: audio pll clock + clock-names: + items: + - const: bus_ck + - const: 26m_ck + - const: l_ck + - const: aud_ck + - const: eg2_ck + + - if: + properties: + compatible: + contains: + const: mediatek,mt7981-afe + then: + properties: + clocks: + items: + - description: audio bus clock + - description: audio 26M clock + - description: audio intbus clock + - description: audio hopping clock + - description: audio pll clock + - description: mux for pcm_mck + clock-names: + items: + - const: bus_ck + - const: 26m_ck + - const: l_ck + - const: aud_ck + - const: eg2_ck + - const: sel + + - if: + properties: + compatible: + contains: + const: mediatek,mt7988-afe + then: + properties: + clocks: + items: + - description: audio bus clock + - description: audio 26M clock + - description: audio intbus clock + - description: audio hopping clock + - description: audio pll clock + - description: mux for pcm_mck + - description: audio i2s/pcm mck + clock-names: + items: + - const: bus_ck + - const: 26m_ck + - const: l_ck + - const: aud_ck + - const: eg2_ck + - const: sel + - const: i2s_m + +additionalProperties: false + +examples: + - | + #include <dt-bindings/interrupt-controller/arm-gic.h> + #include <dt-bindings/interrupt-controller/irq.h> + #include <dt-bindings/clock/mt7986-clk.h> + + afe@11210000 { + compatible = "mediatek,mt7986-afe"; + reg = <0x11210000 0x9000>; + interrupts = <GIC_SPI 106 IRQ_TYPE_LEVEL_HIGH>; + clocks = <&infracfg_ao CLK_INFRA_AUD_BUS_CK>, + <&infracfg_ao CLK_INFRA_AUD_26M_CK>, + <&infracfg_ao CLK_INFRA_AUD_L_CK>, + <&infracfg_ao CLK_INFRA_AUD_AUD_CK>, + <&infracfg_ao CLK_INFRA_AUD_EG2_CK>; + clock-names = "bus_ck", + "26m_ck", + "l_ck", + "aud_ck", + "eg2_ck"; + assigned-clocks = <&topckgen CLK_TOP_A1SYS_SEL>, + <&topckgen CLK_TOP_AUD_L_SEL>, + <&topckgen CLK_TOP_A_TUNER_SEL>; + assigned-clock-parents = <&topckgen CLK_TOP_APLL2_D4>, + <&apmixedsys CLK_APMIXED_APLL2>, + <&topckgen CLK_TOP_APLL2_D4>; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/mediatek,mt7986-wm8960.yaml b/Documentation/devicetree/bindings/sound/mediatek,mt7986-wm8960.yaml new file mode 100644 index 000000000000..09247ceea3f7 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/mediatek,mt7986-wm8960.yaml @@ -0,0 +1,67 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/mediatek,mt7986-wm8960.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: MediaTek MT7986 sound card with WM8960 codec + +maintainers: + - Maso Huang <maso.huang@mediatek.com> + +allOf: + - $ref: sound-card-common.yaml# + +properties: + compatible: + const: mediatek,mt7986-wm8960-sound + + platform: + type: object + additionalProperties: false + properties: + sound-dai: + description: The phandle of MT7986 platform. + maxItems: 1 + required: + - sound-dai + + codec: + type: object + additionalProperties: false + properties: + sound-dai: + description: The phandle of wm8960 codec. + maxItems: 1 + required: + - sound-dai + +unevaluatedProperties: false + +required: + - compatible + - audio-routing + - platform + - codec + +examples: + - | + sound { + compatible = "mediatek,mt7986-wm8960-sound"; + model = "mt7986-wm8960"; + audio-routing = + "Headphone", "HP_L", + "Headphone", "HP_R", + "LINPUT1", "AMIC", + "RINPUT1", "AMIC"; + + platform { + sound-dai = <&afe>; + }; + + codec { + sound-dai = <&wm8960>; + }; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/mediatek,mt8188-afe.yaml b/Documentation/devicetree/bindings/sound/mediatek,mt8188-afe.yaml index e6cb711ece77..0b92c71d8779 100644 --- a/Documentation/devicetree/bindings/sound/mediatek,mt8188-afe.yaml +++ b/Documentation/devicetree/bindings/sound/mediatek,mt8188-afe.yaml @@ -25,6 +25,12 @@ properties: reset-names: const: audiosys + memory-region: + maxItems: 1 + description: | + Shared memory region for AFE memif. A "shared-dma-pool". + See ../reserved-memory/reserved-memory.yaml for details. + mediatek,topckgen: $ref: /schemas/types.yaml#/definitions/phandle description: The phandle of the mediatek topckgen controller @@ -176,6 +182,7 @@ examples: interrupts = <GIC_SPI 822 IRQ_TYPE_LEVEL_HIGH 0>; resets = <&watchdog 14>; reset-names = "audiosys"; + memory-region = <&snd_dma_mem_reserved>; mediatek,topckgen = <&topckgen>; mediatek,infracfg = <&infracfg_ao>; power-domains = <&spm 13>; //MT8188_POWER_DOMAIN_AUDIO diff --git a/Documentation/devicetree/bindings/sound/mediatek,mt8188-mt6359.yaml b/Documentation/devicetree/bindings/sound/mediatek,mt8188-mt6359.yaml index 05e532b5d50a..43b3b67bdf3b 100644 --- a/Documentation/devicetree/bindings/sound/mediatek,mt8188-mt6359.yaml +++ b/Documentation/devicetree/bindings/sound/mediatek,mt8188-mt6359.yaml @@ -9,23 +9,19 @@ title: MediaTek MT8188 ASoC sound card maintainers: - Trevor Wu <trevor.wu@mediatek.com> +allOf: + - $ref: sound-card-common.yaml# + properties: compatible: enum: - mediatek,mt8188-mt6359-evb - mediatek,mt8188-nau8825 - model: - $ref: /schemas/types.yaml#/definitions/string - description: User specified audio sound card name - audio-routing: - $ref: /schemas/types.yaml#/definitions/non-unique-string-array description: - A list of the connections between audio components. Each entry is a - sink/source pair of strings. Valid names could be the input or output - widgets of audio components, power supplies, MicBias of codec and the - software switch. + Valid names could be the input or output widgets of audio components, + power supplies, MicBias of codec and the software switch. mediatek,platform: $ref: /schemas/types.yaml#/definitions/phandle @@ -86,7 +82,7 @@ patternProperties: required: - link-name -additionalProperties: false +unevaluatedProperties: false required: - compatible @@ -96,6 +92,7 @@ examples: - | sound { compatible = "mediatek,mt8188-mt6359-evb"; + model = "MT6359-EVB"; mediatek,platform = <&afe>; pinctrl-names = "default"; pinctrl-0 = <&aud_pins_default>; diff --git a/Documentation/devicetree/bindings/sound/nau8821.txt b/Documentation/devicetree/bindings/sound/nau8821.txt deleted file mode 100644 index 7c84e7c7327a..000000000000 --- a/Documentation/devicetree/bindings/sound/nau8821.txt +++ /dev/null @@ -1,55 +0,0 @@ -Nuvoton NAU88L21 audio codec - -This device supports I2C only. - -Required properties: - - compatible : Must be "nuvoton,nau8821" - - - reg : the I2C address of the device. This is either 0x1B (CSB=0) or 0x54 (CSB=1). - -Optional properties: - - nuvoton,jkdet-enable: Enable jack detection via JKDET pin. - - nuvoton,jkdet-pull-enable: Enable JKDET pin pull. If set - pin pull enabled, - otherwise pin in high impedance state. - - nuvoton,jkdet-pull-up: Pull-up JKDET pin. If set then JKDET pin is pull up, otherwise pull down. - - nuvoton,jkdet-polarity: JKDET pin polarity. 0 - active high, 1 - active low. - - - nuvoton,vref-impedance: VREF Impedance selection - 0 - Open - 1 - 25 kOhm - 2 - 125 kOhm - 3 - 2.5 kOhm - - - nuvoton,micbias-voltage: Micbias voltage level. - 0 - VDDA - 1 - VDDA - 2 - VDDA * 1.1 - 3 - VDDA * 1.2 - 4 - VDDA * 1.3 - 5 - VDDA * 1.4 - 6 - VDDA * 1.53 - 7 - VDDA * 1.53 - - - nuvoton,jack-insert-debounce: number from 0 to 7 that sets debounce time to 2^(n+2) ms - - nuvoton,jack-eject-debounce: number from 0 to 7 that sets debounce time to 2^(n+2) ms - - - nuvoton,dmic-clk-threshold: the ADC threshold of DMIC clock. - - nuvoton,key_enable: Headset button detection switch. - -Example: - - headset: nau8821@1b { - compatible = "nuvoton,nau8821"; - reg = <0x1b>; - interrupt-parent = <&gpio>; - interrupts = <23 IRQ_TYPE_LEVEL_LOW>; - nuvoton,jkdet-enable; - nuvoton,jkdet-pull-enable; - nuvoton,jkdet-pull-up; - nuvoton,jkdet-polarity = <GPIO_ACTIVE_LOW>; - nuvoton,vref-impedance = <2>; - nuvoton,micbias-voltage = <6>; - nuvoton,jack-insert-debounce = <7>; - nuvoton,jack-eject-debounce = <7>; - nuvoton,dmic-clk-threshold = 3072000; - }; diff --git a/Documentation/devicetree/bindings/sound/nuvoton,nau8821.yaml b/Documentation/devicetree/bindings/sound/nuvoton,nau8821.yaml new file mode 100644 index 000000000000..3e54abd4ca74 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nuvoton,nau8821.yaml @@ -0,0 +1,132 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/nuvoton,nau8821.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: NAU88L21 audio codec + +maintainers: + - Seven Lee <wtli@nuvoton.com> + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + const: nuvoton,nau8821 + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + nuvoton,jkdet-enable: + description: Enable jack detection via JKDET pin. + type: boolean + + nuvoton,jkdet-pull-enable: + description: Enable JKDET pin pull. If set - pin pull enabled, + otherwise pin in high impedance state. + type: boolean + + nuvoton,jkdet-pull-up: + description: Pull-up JKDET pin. If set then JKDET pin is pull up, + otherwise pull down. + type: boolean + + nuvoton,key-enable: + description: handles key press detection. + type: boolean + + nuvoton,jkdet-polarity: + description: JKDET pin polarity. + $ref: /schemas/types.yaml#/definitions/uint32 + enum: + - 0 # active high + - 1 # active low + default: 1 + + nuvoton,micbias-voltage: + description: MICBIAS output level select. + $ref: /schemas/types.yaml#/definitions/uint32 + enum: + - 0 # VDDA + - 1 # VDDA * 1 + - 2 # VDDA * 1.1 + - 3 # VDDA * 1.2 + - 4 # VDDA * 1.3 + - 5 # VDDA * 1.4 + - 6 # VDDA * 1.53 + - 7 # VDDA * 1.53 + default: 6 + + nuvoton,vref-impedance: + description: VMID Tie-off impedance select. + $ref: /schemas/types.yaml#/definitions/uint32 + enum: + - 0 # open + - 1 # 25KOhms + - 2 # 125KOhms + - 3 # 2.5KOhms + default: 2 + + nuvoton,jack-insert-debounce: + description: number from 0 to 7 that sets debounce time to 2^(n+2)ms. + $ref: /schemas/types.yaml#/definitions/uint32 + maximum: 7 + default: 7 + + nuvoton,jack-eject-debounce: + description: number from 0 to 7 that sets debounce time to 2^(n+2)ms. + $ref: /schemas/types.yaml#/definitions/uint32 + maximum: 7 + default: 0 + + nuvoton,dmic-clk-threshold: + description: DMIC clock speed expected value. Unit is Hz. + $ref: /schemas/types.yaml#/definitions/uint32 + default: 3072000 + + nuvoton,left-input-single-end: + description: Enable left input with single-ended settings if set. + For the headset mic application, the single-ended control is + just limited to the left adc for design demand. + type: boolean + + '#sound-dai-cells': + const: 0 + +required: + - compatible + - reg + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + #include <dt-bindings/interrupt-controller/irq.h> + i2c { + #address-cells = <1>; + #size-cells = <0>; + codec@1b { + compatible = "nuvoton,nau8821"; + reg = <0x1b>; + interrupt-parent = <&gpio>; + interrupts = <23 IRQ_TYPE_LEVEL_LOW>; + nuvoton,jkdet-enable; + nuvoton,jkdet-pull-enable; + nuvoton,jkdet-pull-up; + nuvoton,key-enable; + nuvoton,left-input-single-end; + nuvoton,jkdet-polarity = <GPIO_ACTIVE_LOW>; + nuvoton,micbias-voltage = <6>; + nuvoton,vref-impedance = <2>; + nuvoton,jack-insert-debounce = <7>; + nuvoton,jack-eject-debounce = <0>; + nuvoton,dmic-clk-threshold = <3072000>; + #sound-dai-cells = <0>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/nuvoton,nau8822.yaml b/Documentation/devicetree/bindings/sound/nuvoton,nau8822.yaml index 65105402a53d..cb8182bbc491 100644 --- a/Documentation/devicetree/bindings/sound/nuvoton,nau8822.yaml +++ b/Documentation/devicetree/bindings/sound/nuvoton,nau8822.yaml @@ -21,6 +21,15 @@ properties: reg: maxItems: 1 + "#sound-dai-cells": + const: 0 + + clocks: + maxItems: 1 + + clock-names: + const: mclk + nuvoton,spk-btl: description: If set, configure the two loudspeaker outputs as a Bridge Tied Load output @@ -31,6 +40,9 @@ required: - compatible - reg +allOf: + - $ref: dai-common.yaml# + additionalProperties: false examples: diff --git a/Documentation/devicetree/bindings/sound/option,gtm601.yaml b/Documentation/devicetree/bindings/sound/option,gtm601.yaml new file mode 100644 index 000000000000..ff813d97fc59 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/option,gtm601.yaml @@ -0,0 +1,42 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/option,gtm601.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: GTM601 UMTS modem audio interface CODEC + +maintainers: + - kernel@puri.sm + +description: > + This device has no configuration interface. The sample rate and channels are + based on the compatible string + +properties: + compatible: + oneOf: + - description: Broadmobi BM818 (48Khz stereo) + items: + - const: broadmobi,bm818 + - const: option,gtm601 + - description: GTM601 (8kHz mono) + const: option,gtm601 + + '#sound-dai-cells': + const: 0 + +required: + - compatible + +allOf: + - $ref: dai-common.yaml# + +additionalProperties: false + +examples: + - | + codec { + compatible = "option,gtm601"; + #sound-dai-cells = <0>; + }; diff --git a/Documentation/devicetree/bindings/sound/qcom,msm8916-wcd-analog.txt b/Documentation/devicetree/bindings/sound/qcom,msm8916-wcd-analog.txt deleted file mode 100644 index e7d17dda55db..000000000000 --- a/Documentation/devicetree/bindings/sound/qcom,msm8916-wcd-analog.txt +++ /dev/null @@ -1,101 +0,0 @@ -msm8916 analog audio CODEC - -Bindings for codec Analog IP which is integrated in pmic pm8916, - -## Bindings for codec core on pmic: - -Required properties - - compatible = "qcom,pm8916-wcd-analog-codec"; - - reg: represents the slave base address provided to the peripheral. - - interrupts: List of interrupts in given SPMI peripheral. - - interrupt-names: Names specified to above list of interrupts in same - order. List of supported interrupt names are: - "cdc_spk_cnp_int" - Speaker click and pop interrupt. - "cdc_spk_clip_int" - Speaker clip interrupt. - "cdc_spk_ocp_int" - Speaker over current protect interrupt. - "mbhc_ins_rem_det1" - jack insert removal detect interrupt 1. - "mbhc_but_rel_det" - button release interrupt. - "mbhc_but_press_det" - button press event - "mbhc_ins_rem_det" - jack insert removal detect interrupt. - "mbhc_switch_int" - multi button headset interrupt. - "cdc_ear_ocp_int" - Earphone over current protect interrupt. - "cdc_hphr_ocp_int" - Headphone R over current protect interrupt. - "cdc_hphl_ocp_det" - Headphone L over current protect interrupt. - "cdc_ear_cnp_int" - earphone cnp interrupt. - "cdc_hphr_cnp_int" - hphr click and pop interrupt. - "cdc_hphl_cnp_int" - hphl click and pop interrupt. - - - clocks: Handle to mclk. - - clock-names: should be "mclk" - - vdd-cdc-io-supply: phandle to VDD_CDC_IO regulator DT node. - - vdd-cdc-tx-rx-cx-supply: phandle to VDD_CDC_TX/RX/CX regulator DT node. - - vdd-micbias-supply: phandle of VDD_MICBIAS supply's regulator DT node. - -Optional Properties: - - qcom,mbhc-vthreshold-low: Array of 5 threshold voltages in mV for 5 buttons - detection on headset when the mbhc is powered up - by internal current source, this is a low power. - - qcom,mbhc-vthreshold-high: Array of 5 thresold voltages in mV for 5 buttons - detection on headset when mbhc is powered up - from micbias. -- qcom,micbias-lvl: Voltage (mV) for Mic Bias -- qcom,hphl-jack-type-normally-open: boolean, present if hphl pin on jack is a - NO (Normally Open). If not specified, then - its assumed that hphl pin on jack is NC - (Normally Closed). -- qcom,gnd-jack-type-normally-open: boolean, present if gnd pin on jack is - NO (Normally Open). If not specified, then - its assumed that gnd pin on jack is NC - (Normally Closed). -- qcom,micbias1-ext-cap: boolean, present if micbias1 has external capacitor - connected. -- qcom,micbias2-ext-cap: boolean, present if micbias2 has external capacitor - connected. - -Example: - -spmi_bus { - ... - audio-codec@f000{ - compatible = "qcom,pm8916-wcd-analog-codec"; - reg = <0xf000 0x200>; - reg-names = "pmic-codec-core"; - clocks = <&gcc GCC_CODEC_DIGCODEC_CLK>; - clock-names = "mclk"; - qcom,mbhc-vthreshold-low = <75 150 237 450 500>; - qcom,mbhc-vthreshold-high = <75 150 237 450 500>; - interrupt-parent = <&spmi_bus>; - interrupts = <0x1 0xf0 0x0 IRQ_TYPE_NONE>, - <0x1 0xf0 0x1 IRQ_TYPE_NONE>, - <0x1 0xf0 0x2 IRQ_TYPE_NONE>, - <0x1 0xf0 0x3 IRQ_TYPE_NONE>, - <0x1 0xf0 0x4 IRQ_TYPE_NONE>, - <0x1 0xf0 0x5 IRQ_TYPE_NONE>, - <0x1 0xf0 0x6 IRQ_TYPE_NONE>, - <0x1 0xf0 0x7 IRQ_TYPE_NONE>, - <0x1 0xf1 0x0 IRQ_TYPE_NONE>, - <0x1 0xf1 0x1 IRQ_TYPE_NONE>, - <0x1 0xf1 0x2 IRQ_TYPE_NONE>, - <0x1 0xf1 0x3 IRQ_TYPE_NONE>, - <0x1 0xf1 0x4 IRQ_TYPE_NONE>, - <0x1 0xf1 0x5 IRQ_TYPE_NONE>; - interrupt-names = "cdc_spk_cnp_int", - "cdc_spk_clip_int", - "cdc_spk_ocp_int", - "mbhc_ins_rem_det1", - "mbhc_but_rel_det", - "mbhc_but_press_det", - "mbhc_ins_rem_det", - "mbhc_switch_int", - "cdc_ear_ocp_int", - "cdc_hphr_ocp_int", - "cdc_hphl_ocp_det", - "cdc_ear_cnp_int", - "cdc_hphr_cnp_int", - "cdc_hphl_cnp_int"; - vdd-cdc-io-supply = <&pm8916_l5>; - vdd-cdc-tx-rx-cx-supply = <&pm8916_l5>; - vdd-micbias-supply = <&pm8916_l13>; - #sound-dai-cells = <1>; - }; -}; diff --git a/Documentation/devicetree/bindings/sound/qcom,pm8916-wcd-analog-codec.yaml b/Documentation/devicetree/bindings/sound/qcom,pm8916-wcd-analog-codec.yaml new file mode 100644 index 000000000000..94e7a1860977 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/qcom,pm8916-wcd-analog-codec.yaml @@ -0,0 +1,153 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/qcom,pm8916-wcd-analog-codec.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Qualcomm PM8916 WCD Analog Audio Codec + +maintainers: + - Konrad Dybcio <konradybcio@kernel.org> + +description: + The analog WCD audio codec found on Qualcomm PM8916 PMIC. + +properties: + compatible: + const: qcom,pm8916-wcd-analog-codec + + reg: + maxItems: 1 + + interrupts: + maxItems: 14 + + interrupt-names: + items: + - const: cdc_spk_cnp_int + - const: cdc_spk_clip_int + - const: cdc_spk_ocp_int + - const: mbhc_ins_rem_det1 + - const: mbhc_but_rel_det + - const: mbhc_but_press_det + - const: mbhc_ins_rem_det + - const: mbhc_switch_int + - const: cdc_ear_ocp_int + - const: cdc_hphr_ocp_int + - const: cdc_hphl_ocp_det + - const: cdc_ear_cnp_int + - const: cdc_hphr_cnp_int + - const: cdc_hphl_cnp_int + + vdd-cdc-io-supply: + description: 1.8V buck supply + + vdd-cdc-tx-rx-cx-supply: + description: 1.8V SIDO buck supply + + vdd-micbias-supply: + description: micbias supply + + qcom,mbhc-vthreshold-low: + $ref: /schemas/types.yaml#/definitions/uint32-array + description: + Array of 5 threshold voltages in mV for 5-button detection on + headset when MBHC is powered by an internal current source. + minItems: 5 + maxItems: 5 + + qcom,mbhc-vthreshold-high: + $ref: /schemas/types.yaml#/definitions/uint32-array + description: + Array of 5 threshold voltages in mV for 5-button detection on + headset when MBHC is powered from micbias. + minItems: 5 + maxItems: 5 + + qcom,micbias-lvl: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + Voltage (mV) for Mic Bias + + qcom,hphl-jack-type-normally-open: + type: boolean + description: + True if the HPHL pin on the jack is NO (Normally Open), false if it's + NC (Normally Closed). + + qcom,gnd-jack-type-normally-open: + type: boolean + description: + True if the GND pin on the jack is NO (Normally Open), false if it's + NC (Normally Closed). + + qcom,micbias1-ext-cap: + type: boolean + description: + True if micbias1 has an external capacitor. + + qcom,micbias2-ext-cap: + type: boolean + description: + True if micbias2 has an external capacitor. + + "#sound-dai-cells": + const: 1 + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + #include <dt-bindings/interrupt-controller/irq.h> + #include <dt-bindings/spmi/spmi.h> + + pmic@1 { + compatible = "qcom,pm8916", "qcom,spmi-pmic"; + reg = <0x1 SPMI_USID>; + #address-cells = <1>; + #size-cells = <0>; + + audio-codec@f000 { + compatible = "qcom,pm8916-wcd-analog-codec"; + reg = <0xf000>; + qcom,mbhc-vthreshold-low = <75 150 237 450 500>; + qcom,mbhc-vthreshold-high = <75 150 237 450 500>; + interrupt-parent = <&spmi_bus>; + interrupts = <0x1 0xf0 0x0 IRQ_TYPE_NONE>, + <0x1 0xf0 0x1 IRQ_TYPE_NONE>, + <0x1 0xf0 0x2 IRQ_TYPE_NONE>, + <0x1 0xf0 0x3 IRQ_TYPE_NONE>, + <0x1 0xf0 0x4 IRQ_TYPE_NONE>, + <0x1 0xf0 0x5 IRQ_TYPE_NONE>, + <0x1 0xf0 0x6 IRQ_TYPE_NONE>, + <0x1 0xf0 0x7 IRQ_TYPE_NONE>, + <0x1 0xf1 0x0 IRQ_TYPE_NONE>, + <0x1 0xf1 0x1 IRQ_TYPE_NONE>, + <0x1 0xf1 0x2 IRQ_TYPE_NONE>, + <0x1 0xf1 0x3 IRQ_TYPE_NONE>, + <0x1 0xf1 0x4 IRQ_TYPE_NONE>, + <0x1 0xf1 0x5 IRQ_TYPE_NONE>; + interrupt-names = "cdc_spk_cnp_int", + "cdc_spk_clip_int", + "cdc_spk_ocp_int", + "mbhc_ins_rem_det1", + "mbhc_but_rel_det", + "mbhc_but_press_det", + "mbhc_ins_rem_det", + "mbhc_switch_int", + "cdc_ear_ocp_int", + "cdc_hphr_ocp_int", + "cdc_hphl_ocp_det", + "cdc_ear_cnp_int", + "cdc_hphr_cnp_int", + "cdc_hphl_cnp_int"; + vdd-cdc-io-supply = <&pm8916_l5>; + vdd-cdc-tx-rx-cx-supply = <&pm8916_l5>; + vdd-micbias-supply = <&pm8916_l13>; + #sound-dai-cells = <1>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml b/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml index 8a821dec9526..13a5a0a10fe6 100644 --- a/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml +++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml @@ -9,6 +9,20 @@ title: Renesas R-Car Sound Driver maintainers: - Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> +definitions: + port-def: + $ref: audio-graph-port.yaml#/definitions/port-base + unevaluatedProperties: false + patternProperties: + "^endpoint(@[0-9a-f]+)?": + $ref: audio-graph-port.yaml#/definitions/endpoint-base + properties: + playback: + $ref: /schemas/types.yaml#/definitions/phandle-array + capture: + $ref: /schemas/types.yaml#/definitions/phandle-array + unevaluatedProperties: false + properties: compatible: @@ -77,6 +91,12 @@ properties: it must be 1 if your system has audio_clkout0/1/2/3 enum: [0, 1] + "#address-cells": + const: 1 + + "#size-cells": + const: 0 + clock-frequency: description: for audio_clkout0/1/2/3 @@ -103,35 +123,9 @@ properties: description: List of necessary clock names. # details are defined below - ports: - $ref: audio-graph-port.yaml#/definitions/port-base - unevaluatedProperties: false - patternProperties: - '^port(@[0-9a-f]+)?$': - $ref: audio-graph-port.yaml#/definitions/port-base - unevaluatedProperties: false - patternProperties: - "^endpoint(@[0-9a-f]+)?": - $ref: audio-graph-port.yaml#/definitions/endpoint-base - properties: - playback: - $ref: /schemas/types.yaml#/definitions/phandle-array - capture: - $ref: /schemas/types.yaml#/definitions/phandle-array - unevaluatedProperties: false - + # ports is below port: - $ref: audio-graph-port.yaml#/definitions/port-base - unevaluatedProperties: false - patternProperties: - "^endpoint(@[0-9a-f]+)?": - $ref: audio-graph-port.yaml#/definitions/endpoint-base - properties: - playback: - $ref: /schemas/types.yaml#/definitions/phandle-array - capture: - $ref: /schemas/types.yaml#/definitions/phandle-array - unevaluatedProperties: false + $ref: "#/definitions/port-def" rcar_sound,dvc: description: DVC subnode. @@ -248,8 +242,9 @@ properties: - interrupts additionalProperties: false +patternProperties: # For DAI base - rcar_sound,dai: + 'rcar_sound,dai(@[0-9a-f]+)?$': description: DAI subnode. type: object patternProperties: @@ -269,6 +264,13 @@ properties: - capture additionalProperties: false + 'ports(@[0-9a-f]+)?$': + $ref: audio-graph-port.yaml#/definitions/port-base + unevaluatedProperties: false + patternProperties: + '^port(@[0-9a-f]+)?$': + $ref: "#/definitions/port-def" + required: - compatible - reg diff --git a/Documentation/devicetree/bindings/sound/samsung,aries-wm8994.yaml b/Documentation/devicetree/bindings/sound/samsung,aries-wm8994.yaml index 447e013f6e17..5ea0819a261a 100644 --- a/Documentation/devicetree/bindings/sound/samsung,aries-wm8994.yaml +++ b/Documentation/devicetree/bindings/sound/samsung,aries-wm8994.yaml @@ -9,6 +9,9 @@ title: Samsung Aries audio complex with WM8994 codec maintainers: - Jonathan Bakker <xc-racer2@live.ca> +allOf: + - $ref: sound-card-common.yaml# + properties: compatible: enum: @@ -17,10 +20,6 @@ properties: # Without FM radio and modem slave - samsung,fascinate4g-wm8994 - model: - $ref: /schemas/types.yaml#/definitions/string - description: The user-visible name of this sound complex. - cpu: type: object additionalProperties: false @@ -46,6 +45,7 @@ properties: samsung,audio-routing: $ref: /schemas/types.yaml#/definitions/non-unique-string-array + deprecated: true description: | List of the connections between audio components; each entry is a pair of strings, the first being the @@ -56,6 +56,7 @@ properties: or FM In For samsung,fascinate4g-wm8994: HP, SPK, RCV, LINE, Main Mic, or HeadsetMic + Deprecated, use audio-routing. extcon: description: Extcon phandle for dock detection @@ -87,10 +88,9 @@ properties: required: - compatible - - model - cpu - codec - - samsung,audio-routing + - audio-routing - extcon - main-micbias-supply - headset-micbias-supply @@ -98,7 +98,7 @@ required: - headset-detect-gpios - headset-key-gpios -additionalProperties: false +unevaluatedProperties: false examples: - | @@ -121,7 +121,7 @@ examples: headset-detect-gpios = <&gph0 6 GPIO_ACTIVE_HIGH>; headset-key-gpios = <&gph3 6 GPIO_ACTIVE_HIGH>; - samsung,audio-routing = + audio-routing = "HP", "HPOUT1L", "HP", "HPOUT1R", diff --git a/Documentation/devicetree/bindings/sound/samsung,midas-audio.yaml b/Documentation/devicetree/bindings/sound/samsung,midas-audio.yaml index 31095913e330..6ec80f529d84 100644 --- a/Documentation/devicetree/bindings/sound/samsung,midas-audio.yaml +++ b/Documentation/devicetree/bindings/sound/samsung,midas-audio.yaml @@ -9,14 +9,13 @@ title: Samsung Midas audio complex with WM1811 codec maintainers: - Sylwester Nawrocki <s.nawrocki@samsung.com> +allOf: + - $ref: sound-card-common.yaml# + properties: compatible: const: samsung,midas-audio - model: - $ref: /schemas/types.yaml#/definitions/string - description: The user-visible name of this sound complex. - cpu: type: object additionalProperties: false @@ -38,6 +37,7 @@ properties: - sound-dai samsung,audio-routing: + deprecated: true $ref: /schemas/types.yaml#/definitions/non-unique-string-array description: | List of the connections between audio components; each entry is @@ -45,6 +45,7 @@ properties: being the connection's source; valid names for sources and sinks are the WM1811's pins (as documented in its binding), and the jacks on the board: HP, SPK, Main Mic, Sub Mic, Headset Mic. + Deprecated, use audio-routing. mic-bias-supply: description: Supply for the micbias on the Main microphone @@ -62,14 +63,13 @@ properties: required: - compatible - - model - cpu - codec - - samsung,audio-routing + - audio-routing - mic-bias-supply - submic-bias-supply -additionalProperties: false +unevaluatedProperties: false examples: - | @@ -84,7 +84,7 @@ examples: mic-bias-supply = <&mic_bias_reg>; submic-bias-supply = <&submic_bias_reg>; - samsung,audio-routing = + audio-routing = "HP", "HPOUT1L", "HP", "HPOUT1R", diff --git a/Documentation/devicetree/bindings/sound/samsung,odroid.yaml b/Documentation/devicetree/bindings/sound/samsung,odroid.yaml index c6751c40e63f..b77284e3e26a 100644 --- a/Documentation/devicetree/bindings/sound/samsung,odroid.yaml +++ b/Documentation/devicetree/bindings/sound/samsung,odroid.yaml @@ -10,6 +10,9 @@ maintainers: - Krzysztof Kozlowski <krzk@kernel.org> - Sylwester Nawrocki <s.nawrocki@samsung.com> +allOf: + - $ref: sound-card-common.yaml# + properties: compatible: oneOf: @@ -24,10 +27,6 @@ properties: - const: samsung,odroid-xu4-audio deprecated: true - model: - $ref: /schemas/types.yaml#/definitions/string - description: The user-visible name of this sound complex. - assigned-clock-parents: true assigned-clock-rates: true assigned-clocks: true @@ -52,6 +51,7 @@ properties: samsung,audio-routing: $ref: /schemas/types.yaml#/definitions/non-unique-string-array + deprecated: true description: | List of the connections between audio components; each entry is a pair of strings, the first being the @@ -61,6 +61,7 @@ properties: For Odroid X2: "Headphone Jack", "Mic Jack", "DMIC" For Odroid U3, XU3: "Headphone Jack", "Speakers" For Odroid XU4: no entries + Deprecated, use audio-routing. samsung,audio-widgets: $ref: /schemas/types.yaml#/definitions/non-unique-string-array @@ -70,18 +71,17 @@ properties: required: - compatible - - model - cpu - codec -additionalProperties: false +unevaluatedProperties: false examples: - | sound { compatible = "hardkernel,odroid-xu3-audio"; model = "Odroid-XU3"; - samsung,audio-routing = + audio-routing = "Headphone Jack", "HPL", "Headphone Jack", "HPR", "IN1", "Mic Jack", diff --git a/Documentation/devicetree/bindings/sound/samsung,tm2.yaml b/Documentation/devicetree/bindings/sound/samsung,tm2.yaml index 491e08019c04..760592599143 100644 --- a/Documentation/devicetree/bindings/sound/samsung,tm2.yaml +++ b/Documentation/devicetree/bindings/sound/samsung,tm2.yaml @@ -10,6 +10,9 @@ maintainers: - Krzysztof Kozlowski <krzk@kernel.org> - Sylwester Nawrocki <s.nawrocki@samsung.com> +allOf: + - $ref: sound-card-common.yaml# + properties: compatible: const: samsung,tm2-audio @@ -32,6 +35,8 @@ properties: being the connection's source; valid names for sources and sinks are the WM5110's and MAX98504's pins and the jacks on the board: HP, SPK, Main Mic, Sub Mic, Third Mic, Headset Mic. + Deprecated, use audio-routing. + deprecated: true $ref: /schemas/types.yaml#/definitions/non-unique-string-array i2s-controller: @@ -44,20 +49,15 @@ properties: mic-bias-gpios: description: GPIO pin that enables the Main Mic bias regulator. - model: - description: The user-visible name of this sound complex. - $ref: /schemas/types.yaml#/definitions/string - required: - compatible - audio-amplifier - audio-codec - - samsung,audio-routing + - audio-routing - i2s-controller - mic-bias-gpios - - model -additionalProperties: false +unevaluatedProperties: false examples: - | @@ -70,7 +70,7 @@ examples: audio-amplifier = <&max98504>; mic-bias-gpios = <&gpr3 2 GPIO_ACTIVE_HIGH>; model = "wm5110"; - samsung,audio-routing = "HP", "HPOUT1L", + audio-routing = "HP", "HPOUT1L", "HP", "HPOUT1R", "SPK", "SPKOUT", "SPKOUT", "HPOUT2L", diff --git a/Documentation/devicetree/bindings/sound/simple-card.yaml b/Documentation/devicetree/bindings/sound/simple-card.yaml index b05e05c81cc4..59ac2d1d1ccf 100644 --- a/Documentation/devicetree/bindings/sound/simple-card.yaml +++ b/Documentation/devicetree/bindings/sound/simple-card.yaml @@ -148,6 +148,15 @@ definitions: required: - sound-dai + additional-devs: + type: object + description: + Additional devices used by the simple audio card. + patternProperties: + '^iio-aux(-.+)?$': + type: object + $ref: audio-iio-aux.yaml# + properties: compatible: contains: @@ -187,6 +196,8 @@ properties: $ref: "#/definitions/mclk-fs" simple-audio-card,aux-devs: $ref: "#/definitions/aux-devs" + simple-audio-card,additional-devs: + $ref: "#/definitions/additional-devs" simple-audio-card,convert-rate: $ref: "#/definitions/convert-rate" simple-audio-card,convert-channels: @@ -360,6 +371,48 @@ examples: }; # -------------------- +# route audio to/from a codec through an amplifier +# designed with a potentiometer driven by IIO: +# -------------------- + - | + sound { + compatible = "simple-audio-card"; + + simple-audio-card,aux-devs = <&_in>, <&_out>; + simple-audio-card,routing = + "CODEC LEFTIN", "AMP_IN LEFT OUT", + "CODEC RIGHTIN", "AMP_IN RIGHT OUT", + "AMP_OUT LEFT IN", "CODEC LEFTOUT", + "AMP_OUT RIGHT IN", "CODEC RIGHTOUT"; + + simple-audio-card,additional-devs { + amp_out: iio-aux-out { + compatible = "audio-iio-aux"; + io-channels = <&pot_out 0>, <&pot_out 1>; + io-channel-names = "LEFT", "RIGHT"; + snd-control-invert-range = <1 1>; + sound-name-prefix = "AMP_OUT"; + }; + + amp_in: iio_aux-in { + compatible = "audio-iio-aux"; + io-channels = <&pot_in 0>, <&pot_in 1>; + io-channel-names = "LEFT", "RIGHT"; + sound-name-prefix = "AMP_IN"; + }; + }; + + simple-audio-card,cpu { + sound-dai = <&cpu>; + }; + + simple-audio-card,codec { + sound-dai = <&codec>; + clocks = <&clocks>; + }; + }; + +# -------------------- # Sampling Rate Conversion # -------------------- - | diff --git a/Documentation/devicetree/bindings/sound/snps,designware-i2s.yaml b/Documentation/devicetree/bindings/sound/snps,designware-i2s.yaml index a970fd264b21..a48d040b0a4f 100644 --- a/Documentation/devicetree/bindings/sound/snps,designware-i2s.yaml +++ b/Documentation/devicetree/bindings/sound/snps,designware-i2s.yaml @@ -17,6 +17,9 @@ properties: - const: snps,designware-i2s - enum: - snps,designware-i2s + - starfive,jh7110-i2stx0 + - starfive,jh7110-i2stx1 + - starfive,jh7110-i2srx reg: maxItems: 1 @@ -29,15 +32,36 @@ properties: maxItems: 1 clocks: - description: Sampling rate reference clock - maxItems: 1 + items: + - description: Sampling rate reference clock + - description: APB clock + - description: Audio master clock + - description: Inner audio master clock source + - description: External audio master clock source + - description: Bit clock + - description: Left/right channel clock + - description: External bit clock + - description: External left/right channel clock + minItems: 1 clock-names: - const: i2sclk + items: + - const: i2sclk + - const: apb + - const: mclk + - const: mclk_inner + - const: mclk_ext + - const: bclk + - const: lrck + - const: bclk_ext + - const: lrck_ext + minItems: 1 resets: items: - description: Optional controller resets + - description: controller reset of Sampling rate + minItems: 1 dmas: items: @@ -51,6 +75,17 @@ properties: - const: rx minItems: 1 + starfive,syscon: + $ref: /schemas/types.yaml#/definitions/phandle-array + items: + - items: + - description: phandle to System Register Controller sys_syscon node. + - description: I2S-rx enabled control offset of SYS_SYSCONSAIF__SYSCFG register. + - description: I2S-rx enabled control mask + description: + The phandle to System Register Controller syscon node and the I2S-rx(ADC) + enabled control offset and mask of SYS_SYSCONSAIF__SYSCFG register. + allOf: - $ref: dai-common.yaml# - if: @@ -66,6 +101,73 @@ allOf: properties: "#sound-dai-cells": const: 0 + - if: + properties: + compatible: + contains: + const: snps,designware-i2s + then: + properties: + clocks: + maxItems: 1 + clock-names: + maxItems: 1 + resets: + maxItems: 1 + else: + properties: + resets: + minItems: 2 + maxItems: 2 + - if: + properties: + compatible: + contains: + const: starfive,jh7110-i2stx0 + then: + properties: + clocks: + minItems: 5 + maxItems: 5 + clock-names: + minItems: 5 + maxItems: 5 + required: + - resets + - if: + properties: + compatible: + contains: + const: starfive,jh7110-i2stx1 + then: + properties: + clocks: + minItems: 9 + maxItems: 9 + clock-names: + minItems: 9 + maxItems: 9 + required: + - resets + - if: + properties: + compatible: + contains: + const: starfive,jh7110-i2srx + then: + properties: + clocks: + minItems: 9 + maxItems: 9 + clock-names: + minItems: 9 + maxItems: 9 + required: + - resets + - starfive,syscon + else: + properties: + starfive,syscon: false required: - compatible diff --git a/Documentation/devicetree/bindings/sound/sound-card-common.yaml b/Documentation/devicetree/bindings/sound/sound-card-common.yaml new file mode 100644 index 000000000000..3a941177f684 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/sound-card-common.yaml @@ -0,0 +1,27 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/sound-card-common.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Board Sound Card Common Properties + +maintainers: + - Mark Brown <broonie@kernel.org> + +properties: + audio-routing: + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + description: | + A list of the 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. + + model: + $ref: /schemas/types.yaml#/definitions/string + description: User specified audio sound card name + +required: + - model + +additionalProperties: true diff --git a/Documentation/devicetree/bindings/sound/wlf,wm8904.yaml b/Documentation/devicetree/bindings/sound/wlf,wm8904.yaml new file mode 100644 index 000000000000..329260cf0fa0 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/wlf,wm8904.yaml @@ -0,0 +1,74 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/wlf,wm8904.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Wolfson WM8904/WM8912 audio codecs + +maintainers: + - patches@opensource.cirrus.com + +description: | + Pins on the device (for linking into audio routes): + IN1L, IN1R, IN2L, IN2R, IN3L, IN3R, HPOUTL, HPOUTR, LINEOUTL, LINEOUTR, + MICBIAS + +properties: + compatible: + enum: + - wlf,wm8904 + - wlf,wm8912 + + reg: + maxItems: 1 + + "#sound-dai-cells": + const: 0 + + clocks: + maxItems: 1 + + clock-names: + const: mclk + + AVDD-supply: true + CPVDD-supply: true + DBVDD-supply: true + DCVDD-supply: true + MICVDD-supply: true + +required: + - compatible + - reg + - clocks + - clock-names + - AVDD-supply + - CPVDD-supply + - DBVDD-supply + - DCVDD-supply + - MICVDD-supply + +allOf: + - $ref: dai-common.yaml# + +unevaluatedProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + + codec@1a { + compatible = "wlf,wm8904"; + reg = <0x1a>; + clocks = <&pck0>; + clock-names = "mclk"; + AVDD-supply = <®_1p8v>; + CPVDD-supply = <®_1p8v>; + DBVDD-supply = <®_1p8v>; + DCVDD-supply = <®_1p8v>; + MICVDD-supply = <®_1p8v>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/wlf,wm8960.yaml b/Documentation/devicetree/bindings/sound/wlf,wm8960.yaml index ee8eba7f0104..62e62c335d07 100644 --- a/Documentation/devicetree/bindings/sound/wlf,wm8960.yaml +++ b/Documentation/devicetree/bindings/sound/wlf,wm8960.yaml @@ -26,6 +26,21 @@ properties: '#sound-dai-cells': const: 0 + AVDD-supply: + description: Analogue supply. + + DBVDD-supply: + description: Digital Buffer Supply. + + DCVDD-supply: + description: Digital Core Supply. + + SPKVDD1-supply: + description: Supply for speaker drivers 1. + + SPKVDD2-supply: + description: Supply for speaker drivers 2. + wlf,capless: type: boolean description: @@ -84,5 +99,10 @@ examples: wlf,hp-cfg = <3 2 3>; wlf,gpio-cfg = <1 3>; wlf,shared-lrclk; + DCVDD-supply = <®_audio>; + DBVDD-supply = <®_audio>; + AVDD-supply = <®_audio>; + SPKVDD1-supply = <®_audio>; + SPKVDD2-supply = <®_audio>; }; }; diff --git a/Documentation/devicetree/bindings/sound/wm8904.txt b/Documentation/devicetree/bindings/sound/wm8904.txt deleted file mode 100644 index 66bf261423b9..000000000000 --- a/Documentation/devicetree/bindings/sound/wm8904.txt +++ /dev/null @@ -1,33 +0,0 @@ -WM8904 audio CODEC - -This device supports I2C only. - -Required properties: - - compatible: "wlf,wm8904" or "wlf,wm8912" - - reg: the I2C address of the device. - - clock-names: "mclk" - - clocks: reference to - <Documentation/devicetree/bindings/clock/clock-bindings.txt> - -Pins on the device (for linking into audio routes): - - * IN1L - * IN1R - * IN2L - * IN2R - * IN3L - * IN3R - * HPOUTL - * HPOUTR - * LINEOUTL - * LINEOUTR - * MICBIAS - -Examples: - -codec: wm8904@1a { - compatible = "wlf,wm8904"; - reg = <0x1a>; - clocks = <&pck0>; - clock-names = "mclk"; -}; diff --git a/Documentation/driver-api/media/dtv-common.rst b/Documentation/driver-api/media/dtv-common.rst index f8b2c4dc8170..207a22bcaf4a 100644 --- a/Documentation/driver-api/media/dtv-common.rst +++ b/Documentation/driver-api/media/dtv-common.rst @@ -3,15 +3,6 @@ Digital TV Common functions --------------------------- -Math functions -~~~~~~~~~~~~~~ - -Provide some commonly-used math functions, usually required in order to -estimate signal strength and signal to noise measurements in dB. - -.. kernel-doc:: include/media/dvb_math.h - - DVB devices ~~~~~~~~~~~ diff --git a/Documentation/sound/cards/audigy-mixer.rst b/Documentation/sound/cards/audigy-mixer.rst index ea66b50a2b03..7ebaacb6df3d 100644 --- a/Documentation/sound/cards/audigy-mixer.rst +++ b/Documentation/sound/cards/audigy-mixer.rst @@ -46,157 +46,158 @@ FX-bus name='PCM Front Playback Volume',index=0 ---------------------------------------- -This control is used to attenuate samples for left and right front PCM FX-bus +This control is used to attenuate samples from 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. +samples for 5.1 playback. The result samples are forwarded to the front speakers. name='PCM Surround Playback Volume',index=0 ------------------------------------------- -This control is used to attenuate samples for left and right surround PCM FX-bus +This control is used to attenuate samples from 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. +samples for 5.1 playback. The result samples are forwarded to the surround (rear) +speakers. + +name='PCM Side Playback Volume',index=0 +--------------------------------------- +This control is used to attenuate samples from left and right side PCM FX-bus +accumulators. ALSA uses accumulators 14 and 15 for left and right side PCM +samples for 7.1 playback. The result samples are forwarded to the side speakers. 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. +This control is used to attenuate samples from center PCM FX-bus accumulator. +ALSA uses accumulator 6 for center PCM samples for 5.1 playback. The result +samples are forwarded to the center speaker. 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. +ALSA uses accumulator 7 for LFE PCM samples for 5.1 playback. The result +samples are forwarded to the subwoofer. name='PCM Playback Volume',index=0 ---------------------------------- -This control is used to attenuate samples for left and right PCM FX-bus +This control is used to attenuate samples from 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. +stereo playback. The result samples are forwarded to the front speakers. 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). +This control is used to attenuate samples from left and right PCM FX-bus +accumulators. ALSA uses accumulators 0 and 1 for left and right PCM samples for +stereo playback. The result is forwarded 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 +This control is used to attenuate samples from 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. +The result samples are forwarded to the virtual stereo mixer. 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). +These controls are used to attenuate samples from left and right MIDI FX-bus +accumulator. ALSA uses accumulators 4 and 5 for left and right MIDI samples. +The result is forwarded 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. +This control is used to attenuate samples from left and right Mic input of +the AC97 codec. The result samples are forwarded to the virtual stereo mixer. 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). +This control is used to attenuate samples from left and right Mic input of +the AC97 codec. The result is forwarded to the standard capture PCM device. + +The original samples are also forwarded to the Mic capture PCM device (device 1; +16bit/8KHz mono) without volume control. 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. +forwarded to the virtual stereo mixer. 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). +digital inputs (usually used by a CDROM drive). The result is forwarded +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. +digital input. The result samples are forwarded to the virtual stereo mixer. 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). +digital inputs. The result is forwarded 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. +inputs (on the AudigyDrive). The result samples are forwarded to the virtual +stereo mixer. 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). +inputs (on the AudigyDrive). The result is forwarded 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, .... +inputs from Philips ADC. The result samples are forwarded to the virtual +stereo mixer. 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). +inputs Philips ADC. The result is forwarded 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. +inputs (on the AudigyDrive). The result samples are forwarded to the virtual +stereo mixer. 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). +inputs (on the AudigyDrive). The result is forwarded 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. +This control is used to attenuate samples from the virtual stereo mixer. +The result samples are forwarded to the front speakers. 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. +This control is used to attenuate samples from the virtual stereo mixer. +The result samples are forwarded to the surround (rear) speakers. + +name='Side Playback Volume',index=0 +----------------------------------- +This control is used to attenuate samples from the virtual stereo mixer. +The result samples are forwarded to the side speakers. 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. +This control is used to attenuate samples from the virtual stereo mixer. +The result samples are forwarded to the center speaker. 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. +This control is used to attenuate samples from the virtual stereo mixer. +The result samples are forwarded to the subwoofer. 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. +This control turns the tone control on or off. The samples forwarded to +the speaker outputs are affected. name='Tone Control - Bass',index=0 ---------------------------------- @@ -212,8 +213,7 @@ 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. +This control is used to attenuate samples forwarded to the speaker outputs. name='IEC958 Optical Raw Playback Switch',index=0 ------------------------------------------------- @@ -303,69 +303,4 @@ The channel mapping is following: MANUALS/PATENTS =============== -ftp://opensource.creative.com/pub/doc -------------------------------------- - -Note that the site is defunct, but the documents are available -from various other locations. - -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 (https://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) +See sb-live-mixer.rst. diff --git a/Documentation/sound/cards/emu-mixer.rst b/Documentation/sound/cards/emu-mixer.rst new file mode 100644 index 000000000000..d87a6338d3d8 --- /dev/null +++ b/Documentation/sound/cards/emu-mixer.rst @@ -0,0 +1,226 @@ +================================================== +E-MU Digital Audio System mixer / default DSP code +================================================== + +This document covers the E-MU 0404/1010/1212/1616/1820 PCI/PCI-e/CardBus +cards. + +These cards use regular EMU10K2 (SoundBlaster Audigy) chips, but with an +alternative front-end geared towards semi-professional studio recording. + +This document is based on audigy-mixer.rst. + + +Hardware compatibility +====================== + +The EMU10K2 chips have a very short capture FIFO, which makes recording +unreliable if the card's PCI bus requests are not handled with the +appropriate priority. +This is the case on more modern motherboards, where the PCI bus is only a +secondary peripheral, rather than the actual arbiter of device access. +In particular, I got recording glitches during simultaneous playback on an +Intel DP55 board (memory controller in the CPU), but had success with an +Intel DP45 board (memory controller in the north bridge). + +The PCI Express variants of these cards (which have a PCI bridge on board, +but are otherwise identical) may be less problematic. + + +Driver capabilities +=================== + +This driver supports only 16-bit 44.1/48 kHz operation. The multi-channel +device (see emu10k1-jack.rst) additionally supports 24-bit capture. + +A patchset to enhance the driver is available from `a GitHub repository +<https://github.com/ossilator/linux/tree/ossis-emu10k1>`_. +Its multi-channel device supports 24-bit for both playback and capture, +and also supports full 88.2/96/176.4/192 kHz operation. +It is not going to be upstreamed due to a fundamental disagreement about +what constitutes a good user experience. + + +Digital mixer controls +====================== + +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 clipped +(set to maximal or minimal value without checking for overflow). + +Explanation of used abbreviations: + +DAC + digital to analog converter +ADC + analog to digital converter +LFE + low frequency effects (used as subwoofer signal) +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='Clock Source',index=0 +--------------------------- +This control allows switching the word clock between interally generated +44.1 or 48 kHz, or a number of external sources. + +Note: the sources for the 1616 CardBus card are unclear. Please report your +findings. + +name='Clock Fallback',index=0 +----------------------------- +This control determines the internal clock which the card switches to when +the selected external clock source is/becomes invalid. + +name='DAC1 0202 14dB PAD',index=0, etc. +--------------------------------------- +Output attenuation controls. Not available on 0404 cards. + +name='ADC1 14dB PAD 0202',index=0, etc. +--------------------------------------- +Input attenuation controls. Not available on 0404 cards. + +name='Optical Output Mode',index=0 +---------------------------------- +Switches the TOSLINK output port between S/PDIF and ADAT. +Not available on 0404 cards (fixed to S/PDIF). + +name='Optical Input Mode',index=0 +--------------------------------- +Switches the TOSLINK input port between S/PDIF and ADAT. +Not available on 0404 cards (fixed to S/PDIF). + +name='PCM Front Playback Volume',index=0 +---------------------------------------- +This control is used to attenuate samples from 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 DSP 0 & 1 +playback channels. + +name='PCM Surround Playback Volume',index=0 +------------------------------------------- +This control is used to attenuate samples from 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 DSP 2 & 3 +playback channels. + +name='PCM Side Playback Volume',index=0 +--------------------------------------- +This control is used to attenuate samples from left and right side PCM FX-bus +accumulators. ALSA uses accumulators 14 and 15 for left and right side PCM +samples for 7.1 playback. The result samples are forwarded to the DSP 6 & 7 +playback channels. + +name='PCM Center Playback Volume',index=0 +----------------------------------------- +This control is used to attenuate samples from the center PCM FX-bus accumulator. +ALSA uses accumulator 6 for center PCM samples for 5.1 playback. The result samples +are forwarded to the DSP 4 playback channel. + +name='PCM LFE Playback Volume',index=0 +-------------------------------------- +This control is used to attenuate samples from the LFE PCM FX-bus accumulator. +ALSA uses accumulator 7 for LFE PCM samples for 5.1 playback. The result samples +are forwarded to the DSP 5 playback channel. + +name='PCM Playback Volume',index=0 +---------------------------------- +This control is used to attenuate samples from 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 virtual stereo mixer. + +name='PCM Capture Volume',index=0 +--------------------------------- +This control is used to attenuate samples from left and right PCM FX-bus +accumulators. ALSA uses accumulators 0 and 1 for left and right PCM. +The result is forwarded to the standard capture PCM device. + +name='Music Playback Volume',index=0 +------------------------------------ +This control is used to attenuate samples from 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 virtual stereo mixer. + +name='Music Capture Volume',index=0 +----------------------------------- +These controls are used to attenuate samples from left and right MIDI FX-bus +accumulator. ALSA uses accumulators 4 and 5 for left and right MIDI samples. +The result is forwarded to the standard capture PCM device. + +name='Front Playback Volume',index=0 +------------------------------------ +This control is used to attenuate samples from the virtual stereo mixer. +The result samples are forwarded to the DSP 0 & 1 playback channels. + +name='Surround Playback Volume',index=0 +--------------------------------------- +This control is used to attenuate samples from the virtual stereo mixer. +The result samples are forwarded to the DSP 2 & 3 playback channels. + +name='Side Playback Volume',index=0 +----------------------------------- +This control is used to attenuate samples from the virtual stereo mixer. +The result samples are forwarded to the DSP 6 & 7 playback channels. + +name='Center Playback Volume',index=0 +------------------------------------- +This control is used to attenuate samples from the virtual stereo mixer. +The result samples are forwarded to the DSP 4 playback channel. + +name='LFE Playback Volume',index=0 +---------------------------------- +This control is used to attenuate samples from the virtual stereo mixer. +The result samples are forwarded to the DSP 5 playback channel. + +name='Tone Control - Switch',index=0 +------------------------------------ +This control turns the tone control on or off. The samples forwarded to +the DSP playback channels 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 all DSP playback channels. + +name='EMU Capture Volume',index=0 +---------------------------------- +This control is used to attenuate samples from the DSP 0 & 1 capture channels. +The result is forwarded to the standard capture PCM device. + +name='DAC Left',index=0, etc. +----------------------------- +Select the source for the given physical audio output. These may be physical +inputs, playback channels (DSP xx, specified as a decimal number), or silence. + +name='DSP x',index=0 +-------------------- +Select the source for the given capture channel (specified as a hexadecimal +digit). Same options as for the physical audio outputs. + + +PCM stream related controls +=========================== + +These controls are described in audigy-mixer.rst. + + +MANUALS/PATENTS +=============== + +See sb-live-mixer.rst. diff --git a/Documentation/sound/cards/index.rst b/Documentation/sound/cards/index.rst index 49c1f2f688f8..e68bbb13c384 100644 --- a/Documentation/sound/cards/index.rst +++ b/Documentation/sound/cards/index.rst @@ -8,6 +8,7 @@ Card-Specific Information cmipci sb-live-mixer audigy-mixer + emu-mixer emu10k1-jack via82xx-mixer audiophile-usb diff --git a/Documentation/sound/cards/sb-live-mixer.rst b/Documentation/sound/cards/sb-live-mixer.rst index 4dd9bfe01bd8..27667f58aae1 100644 --- a/Documentation/sound/cards/sb-live-mixer.rst +++ b/Documentation/sound/cards/sb-live-mixer.rst @@ -61,61 +61,61 @@ FX-bus ``name='Wave Playback Volume',index=0`` --------------------------------------- -This control is used to attenuate samples for left and right PCM FX-bus +This control is used to attenuate samples from 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 +This control is used to attenuate samples from 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 +This control is used to attenuate samples from 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 +This control is used to attenuate samples from 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 +These controls are used to attenuate samples from 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 +This control is used to attenuate samples from 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. +These controls are used to attenuate samples from left and right MIDI FX-bus +accumulator. ALSA uses accumulators 4 and 5 for left and right MIDI samples. 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 +This control is used to attenuate samples from 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 +These controls are used to attenuate samples from 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). @@ -134,7 +134,7 @@ 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 +This control is used to attenuate samples from 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. @@ -145,7 +145,7 @@ slots of the AC97 codec. ``name='AC97 Capture Volume',index=0`` -------------------------------------- -This control is used to attenuate samples for left and right front ADC PCM slots +This control is used to attenuate samples from 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). diff --git a/Documentation/sound/designs/midi-2.0.rst b/Documentation/sound/designs/midi-2.0.rst index 27d0d3dea1b0..45987f256b97 100644 --- a/Documentation/sound/designs/midi-2.0.rst +++ b/Documentation/sound/designs/midi-2.0.rst @@ -376,3 +376,191 @@ Sequencer API Extensions name and attributes accordingly, and notifies the changes via the announcement to the ALSA sequencer system port, similarly like the normal port change notification. + + +MIDI2 USB Gadget Function Driver +================================ + +The latest kernel contains the support for USB MIDI 2.0 gadget +function driver, which can be used for prototyping and debugging MIDI +2.0 features. + +`CONFIG_USB_GADGET`, `CONFIG_USB_CONFIGFS` and +`CONFIG_USB_CONFIGFS_F_MIDI2` need to be enabled for the MIDI2 gadget +driver. + +In addition, for using a gadget driver, you need a working UDC driver. +In the example below, we use `dummy_hcd` driver (enabled via +`CONFIG_USB_DUMMY_HCD`) that is available on PC and VM for debugging +purpose. There are other UDC drivers depending on the platform, and +those can be used for a real device, instead, too. + +At first, on a system to run the gadget, load `libcomposite` module:: + + % modprobe libcomposite + +and you'll have `usb_gadget` subdirectory under configfs space +(typically `/sys/kernel/config` on modern OS). Then create a gadget +instance and add configurations there, for example:: + + % cd /sys/kernel/config + % mkdir usb_gadget/g1 + + % cd usb_gadget/g1 + % mkdir configs/c.1 + % mkdir functions/midi2.usb0 + + % echo 0x0004 > idProduct + % echo 0x17b3 > idVendor + % mkdir strings/0x409 + % echo "ACME Enterprises" > strings/0x409/manufacturer + % echo "ACMESynth" > strings/0x409/product + % echo "ABCD12345" > strings/0x409/serialnumber + + % mkdir configs/c.1/strings/0x409 + % echo "Monosynth" > configs/c.1/strings/0x409/configuration + % echo 120 > configs/c.1/MaxPower + +At this point, there must be a subdirectory `ep.0`, and that is the +configuration for a UMP Endpoint. You can fill the Endpoint +information like:: + + % echo "ACMESynth" > functions/midi2.usb0/iface_name + % echo "ACMESynth" > functions/midi2.usb0/ep.0/ep_name + % echo "ABCD12345" > functions/midi2.usb0/ep.0/product_id + % echo 0x0123 > functions/midi2.usb0/ep.0/family + % echo 0x4567 > functions/midi2.usb0/ep.0/model + % echo 0x123456 > functions/midi2.usb0/ep.0/manufacturer + % echo 0x12345678 > functions/midi2.usb0/ep.0/sw_revision + +The default MIDI protocol can be set either 1 or 2:: + + % echo 2 > functions/midi2.usb0/ep.0/protocol + +And, you can find a subdirectory `block.0` under this Endpoint +subdirectory. This defines the Function Block information:: + + % echo "Monosynth" > functions/midi2.usb0/ep.0/block.0/name + % echo 0 > functions/midi2.usb0/ep.0/block.0/first_group + % echo 1 > functions/midi2.usb0/ep.0/block.0/num_groups + +Finally, link the configuration and enable it:: + + % ln -s functions/midi2.usb0 configs/c.1 + % echo dummy_udc.0 > UDC + +where `dummy_udc.0` is an example case and it differs depending on the +system. You can find the UDC instances in `/sys/class/udc` and pass +the found name instead:: + + % ls /sys/class/udc + dummy_udc.0 + +Now, the MIDI 2.0 gadget device is enabled, and the gadget host +creates a new sound card instance containing a UMP rawmidi device by +`f_midi2` driver:: + + % cat /proc/asound/cards + .... + 1 [Gadget ]: f_midi2 - MIDI 2.0 Gadget + MIDI 2.0 Gadget + +And on the connected host, a similar card should appear, too, but with +the card and device names given in the configfs above:: + + % cat /proc/asound/cards + .... + 2 [ACMESynth ]: USB-Audio - ACMESynth + ACME Enterprises ACMESynth at usb-dummy_hcd.0-1, high speed + +You can play a MIDI file on the gadget side:: + + % aplaymidi -p 20:1 to_host.mid + +and this will appear as an input from a MIDI device on the connected +host:: + + % aseqdump -p 20:0 -u 2 + +Vice versa, a playback on the connected host will work as an input on +the gadget, too. + +Each Function Block may have different direction and UI-hint, +specified via `direction` and `ui_hint` attributes. +Passing `1` is for input-only, `2` for out-only and `3` for +bidirectional (the default value). For example:: + + % echo 2 > functions/midi2.usb0/ep.0/block.0/direction + % echo 2 > functions/midi2.usb0/ep.0/block.0/ui_hint + +When you need more than one Function Blocks, you can create +subdirectories `block.1`, `block.2`, etc dynamically, and configure +them in the configuration procedure above before linking. +For example, to create a second Function Block for a keyboard:: + + % mkdir functions/midi2.usb0/ep.0/block.1 + % echo "Keyboard" > functions/midi2.usb0/ep.0/block.1/name + % echo 1 > functions/midi2.usb0/ep.0/block.1/first_group + % echo 1 > functions/midi2.usb0/ep.0/block.1/num_groups + % echo 1 > functions/midi2.usb0/ep.0/block.1/direction + % echo 1 > functions/midi2.usb0/ep.0/block.1/ui_hint + +The `block.*` subdirectories can be removed dynamically, too (except +for `block.0` which is persistent). + +For assigning a Function Block for MIDI 1.0 I/O, set up in `is_midi1` +attribute. 1 is for MIDI 1.0, and 2 is for MIDI 1.0 with low speed +connection:: + + % echo 2 > functions/midi2.usb0/ep.0/block.1/is_midi1 + +For disabling the processing of UMP Stream messages in the gadget +driver, pass `0` to `process_ump` attribute in the top-level config:: + + % echo 0 > functions/midi2.usb0/process_ump + +The MIDI 1.0 interface at altset 0 is supported by the gadget driver, +too. When MIDI 1.0 interface is selected by the connected host, the +UMP I/O on the gadget is translated from/to USB MIDI 1.0 packets +accordingly while the gadget driver keeps communicating with the +user-space over UMP rawmidi. + +MIDI 1.0 ports are set up from the config in each Function Block. +For example:: + + % echo 0 > functions/midi2.usb0/ep.0/block.0/midi1_first_group + % echo 1 > functions/midi2.usb0/ep.0/block.0/midi1_num_groups + +The configuration above will enable the Group 1 (the index 0) for MIDI +1.0 interface. Note that those groups must be in the groups defined +for the Function Block itself. + +The gadget driver supports more than one UMP Endpoints, too. +Similarly like the Function Blocks, you can create a new subdirectory +`ep.1` (but under the card top-level config) to enable a new Endpoint:: + + % mkdir functions/midi2.usb0/ep.1 + +and create a new Function Block there. For example, to create 4 +Groups for the Function Block of this new Endpoint:: + + % mkdir functions/midi2.usb0/ep.1/block.0 + % echo 4 > functions/midi2.usb0/ep.1/block.0/num_groups + +Now, you'll have 4 rawmidi devices in total: the first two are UMP +rawmidi devices for Endpoint 0 and Endpoint 1, and other two for the +legacy MIDI 1.0 rawmidi devices corresponding to both EP 0 and EP 1. + +The current altsetting on the gadget can be informed via a control +element "Operation Mode" with `RAWMIDI` iface. e.g. you can read it +via `amixer` program running on the gadget host like:: + + % amixer -c1 cget iface=RAWMIDI,name='Operation Mode' + ; type=INTEGER,access=r--v----,values=1,min=0,max=2,step=0 + : values=2 + +The value (shown in the second returned line with `: values=`) +indicates 1 for MIDI 1.0 (altset 0), 2 for MIDI 2.0 (altset 1) and 0 +for unset. + +As of now, the configurations can't be changed after binding. diff --git a/Documentation/sound/kernel-api/writing-an-alsa-driver.rst b/Documentation/sound/kernel-api/writing-an-alsa-driver.rst index 4335c98b3d82..cd421856409e 100644 --- a/Documentation/sound/kernel-api/writing-an-alsa-driver.rst +++ b/Documentation/sound/kernel-api/writing-an-alsa-driver.rst @@ -2018,8 +2018,8 @@ sleeping poll threads, etc. This callback is also atomic by default. -copy_user, copy_kernel and fill_silence ops -~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ +copy and fill_silence ops +~~~~~~~~~~~~~~~~~~~~~~~~~ These callbacks are not mandatory, and can be omitted in most cases. These callbacks are used when the hardware buffer cannot be in the @@ -3444,8 +3444,8 @@ 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 -efficient. You need to define the ``copy_user`` and ``copy_kernel`` -callbacks for the data transfer, in addition to the ``fill_silence`` +efficient. You need to define the ``copy`` callback +for the data transfer, in addition to the ``fill_silence`` callback for playback. However, there is a drawback: it cannot be mmapped. The examples are GUS's GF1 PCM or emu8000's wavetable PCM. @@ -3458,22 +3458,22 @@ 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_user``, ``copy_kernel`` and ``fill_silence`` callbacks as well, +``copy`` and ``fill_silence`` callbacks as well, as in the cases above. Examples are found in ``rme32.c`` and ``rme96.c``. -The implementation of the ``copy_user``, ``copy_kernel`` and +The implementation of the ``copy`` and ``silence`` callbacks depends upon whether the hardware supports -interleaved or non-interleaved samples. The ``copy_user`` callback is +interleaved or non-interleaved samples. The ``copy`` callback is defined like below, a bit differently depending on whether the direction is playback or capture:: - static int playback_copy_user(struct snd_pcm_substream *substream, + static int playback_copy(struct snd_pcm_substream *substream, int channel, unsigned long pos, - void __user *src, unsigned long count); - static int capture_copy_user(struct snd_pcm_substream *substream, + struct iov_iter *src, unsigned long count); + static int capture_copy(struct snd_pcm_substream *substream, int channel, unsigned long pos, - void __user *dst, unsigned long count); + struct iov_iter *dst, unsigned long count); In the case of interleaved samples, the second argument (``channel``) is not used. The third argument (``pos``) specifies the position in bytes. @@ -3490,18 +3490,17 @@ of data (``count``) at the specified pointer (``src``) to the specified offset (``pos``) in the hardware buffer. When coded like memcpy-like way, the copy would look like:: - my_memcpy_from_user(my_buffer + pos, src, count); + my_memcpy_from_iter(my_buffer + pos, src, count); For the capture direction, you copy the given amount of data (``count``) at the specified offset (``pos``) in the hardware buffer to the specified pointer (``dst``):: - my_memcpy_to_user(dst, my_buffer + pos, count); + my_memcpy_to_iter(dst, my_buffer + pos, count); -Here the functions are named ``from_user`` and ``to_user`` because -it's the user-space buffer that is passed to these callbacks. That -is, the callback is supposed to copy data from/to the user-space -directly to/from the hardware buffer. +The given ``src`` or ``dst`` a struct iov_iter pointer containing the +pointer and the size. Use the existing helpers to copy or access the +data as defined in ``linux/uio.h``. Careful readers might notice that these callbacks receive the arguments in bytes, not in frames like other callbacks. It's because @@ -3519,25 +3518,6 @@ the given user-space buffer, but only for the given channel. For details, please check ``isa/gus/gus_pcm.c`` or ``pci/rme9652/rme9652.c`` as examples. -The above callbacks are the copies from/to the user-space buffer. There -are some cases where we want to copy from/to the kernel-space buffer -instead. In such a case, the ``copy_kernel`` callback is called. It'd -look like:: - - static int playback_copy_kernel(struct snd_pcm_substream *substream, - int channel, unsigned long pos, - void *src, unsigned long count); - static int capture_copy_kernel(struct snd_pcm_substream *substream, - int channel, unsigned long pos, - void *dst, unsigned long count); - -As found easily, the only difference is that the buffer pointer is -without a ``__user`` prefix; that is, a kernel-buffer pointer is passed -in the fourth argument. Correspondingly, the implementation would be -a version without the user-copy, such as:: - - my_memcpy(my_buffer + pos, src, count); - Usually for the playback, another callback ``fill_silence`` is defined. It's implemented in a similar way as the copy callbacks above:: @@ -3545,10 +3525,10 @@ above:: static int silence(struct snd_pcm_substream *substream, int channel, unsigned long pos, unsigned long count); -The meanings of arguments are the same as in the ``copy_user`` and -``copy_kernel`` callbacks, although there is no buffer pointer +The meanings of arguments are the same as in the ``copy`` callback, +although there is no buffer pointer argument. In the case of interleaved samples, the channel argument has -no meaning, as for the ``copy_*`` callbacks. +no meaning, as for the ``copy`` callback. The role of the ``fill_silence`` callback is to set the given amount (``count``) of silence data at the specified offset (``pos``) in the |