January 2020 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: dt-bindings: stm32: convert sai to json-schema
by Olivier Moysan 11 Feb '20

11 Feb '20

Convert the STM32 SAI bindings to DT schema format using json-schema. Signed-off-by: Olivier Moysan <olivier.moysan(a)st.com> --- .../bindings/sound/st,stm32-sai.txt | 107 ---------- .../bindings/sound/st,stm32-sai.yaml | 193 ++++++++++++++++++ 2 files changed, 193 insertions(+), 107 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/st,stm32-sai.txt create mode 100644 Documentation/devicetree/bindings/sound/st,stm32-sai.yaml diff --git a/Documentation/devicetree/bindings/sound/st,stm32-sai.txt b/Documentation/devicetree/bindings/sound/st,stm32-sai.txt deleted file mode 100644 index 944743dd9212..000000000000 --- a/Documentation/devicetree/bindings/sound/st,stm32-sai.txt +++ /dev/null @@ -1,107 +0,0 @@ -STMicroelectronics STM32 Serial Audio Interface (SAI). - -The SAI interface (Serial Audio Interface) offers a wide set of audio protocols -as I2S standards, LSB or MSB-justified, PCM/DSP, TDM, and AC'97. -The SAI contains two independent audio sub-blocks. Each sub-block has -its own clock generator and I/O lines controller. - -Required properties: - - compatible: Should be "st,stm32f4-sai" or "st,stm32h7-sai" - - reg: Base address and size of SAI common register set. - - clocks: Must contain phandle and clock specifier pairs for each entry - in clock-names. - - clock-names: Must contain "pclk" "x8k" and "x11k" - "pclk": Clock which feeds the peripheral bus interface. - Mandatory for "st,stm32h7-sai" compatible. - Not used for "st,stm32f4-sai" compatible. - "x8k": SAI parent clock for sampling rates multiple of 8kHz. - "x11k": SAI parent clock for sampling rates multiple of 11.025kHz. - - interrupts: cpu DAI interrupt line shared by SAI sub-blocks - -Optional properties: - - resets: Reference to a reset controller asserting the SAI - -SAI subnodes: -Two subnodes corresponding to SAI sub-block instances A et B can be defined. -Subnode can be omitted for unsused sub-block. - -SAI subnodes required properties: - - compatible: Should be "st,stm32-sai-sub-a" or "st,stm32-sai-sub-b" - for SAI sub-block A or B respectively. - - reg: Base address and size of SAI sub-block register set. - - clocks: Must contain one phandle and clock specifier pair - for sai_ck which feeds the internal clock generator. - If the SAI shares a master clock, with another SAI set as MCLK - clock provider, SAI provider phandle must be specified here. - - clock-names: Must contain "sai_ck". - Must also contain "MCLK", if SAI shares a master clock, - with a SAI set as MCLK clock provider. - - dmas: see Documentation/devicetree/bindings/dma/stm32-dma.txt - - dma-names: identifier string for each DMA request line - "tx": if sai sub-block is configured as playback DAI - "rx": if sai sub-block is configured as capture DAI - - pinctrl-names: should contain only value "default" - - pinctrl-0: see Documentation/devicetree/bindings/pinctrl/st,stm32-pinctrl.yaml - -SAI subnodes Optional properties: - - st,sync: specify synchronization mode. - By default SAI sub-block is in asynchronous mode. - This property sets SAI sub-block as slave of another SAI sub-block. - Must contain the phandle and index of the sai sub-block providing - the synchronization. - - st,iec60958: support S/PDIF IEC6958 protocol for playback - IEC60958 protocol is not available for capture. - By default, custom protocol is assumed, meaning that protocol is - configured according to protocol defined in related DAI link node, - such as i2s, left justified, right justified, dsp and pdm protocols. - Note: ac97 protocol is not supported by SAI driver - - #clock-cells: should be 0. This property must be present if the SAI device - is a master clock provider, according to clocks bindings, described in - Documentation/devicetree/bindings/clock/clock-bindings.txt. - -The device node should contain one 'port' child node with one child 'endpoint' -node, according to the bindings defined in Documentation/devicetree/bindings/ -graph.txt. - -Example: -sound_card { - compatible = "audio-graph-card"; - dais = <&sai1b_port>; -}; - -sai1: sai1@40015800 { - compatible = "st,stm32h7-sai"; - #address-cells = <1>; - #size-cells = <1>; - ranges = <0 0x40015800 0x400>; - reg = <0x40015800 0x4>; - clocks = <&rcc SAI1_CK>, <&rcc PLL1_Q>, <&rcc PLL2_P>; - clock-names = "pclk", "x8k", "x11k"; - interrupts = <87>; - - sai1a: audio-controller@40015804 { - compatible = "st,stm32-sai-sub-a"; - reg = <0x4 0x1C>; - clocks = <&rcc SAI1_CK>; - clock-names = "sai_ck"; - dmas = <&dmamux1 1 87 0x400 0x0>; - dma-names = "tx"; - pinctrl-names = "default"; - pinctrl-0 = <&pinctrl_sai1a>; - - sai1b_port: port { - cpu_endpoint: endpoint { - remote-endpoint = <&codec_endpoint>; - format = "i2s"; - }; - }; - }; -}; - -audio-codec { - codec_port: port { - codec_endpoint: endpoint { - remote-endpoint = <&cpu_endpoint>; - }; - }; -}; diff --git a/Documentation/devicetree/bindings/sound/st,stm32-sai.yaml b/Documentation/devicetree/bindings/sound/st,stm32-sai.yaml new file mode 100644 index 000000000000..33dca007fc86 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/st,stm32-sai.yaml @@ -0,0 +1,193 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/st,stm32-sai.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: STMicroelectronics STM32 Serial Audio Interface (SAI) + +maintainers: + - Olivier Moysan <olivier.moysan(a)st.com> + +description: + The SAI interface (Serial Audio Interface) offers a wide set of audio + protocols as I2S standards, LSB or MSB-justified, PCM/DSP, TDM, and AC'97. + The SAI contains two independent audio sub-blocks. Each sub-block has + its own clock generator and I/O lines controller. + +properties: + compatible: + enum: + - st,stm32f4-sai + - st,stm32h7-sai + + reg: + items: + - description: Base address and size of SAI common register set. + - description: Base address and size of SAI identification register set. + minItems: 1 + maxItems: 2 + + ranges: + maxItems: 1 + + interrupts: + maxItems: 1 + + resets: + maxItems: 1 + + "#address-cells": + const: 1 + + "#size-cells": + const: 1 + + clocks: + items: + - description: pclk feeds the peripheral bus interface. + - description: x8k, SAI parent clock for sampling rates multiple of 8kHz. + - description: x11k, SAI parent clock for sampling rates multiple of 11.025kHz. + + clock-names: + items: + enum: [ pclk, x8k, x11k ] + minItems: 3 + maxItems: 3 + +required: + - compatible + - reg + - ranges + - "#address-cells" + - "#size-cells" + - clocks + - clock-names + +patternProperties: + "^audio-controller@[0-9a-f]+$": + type: object + description: + Two subnodes corresponding to SAI sub-block instances A et B + can be defined. Subnode can be omitted for unsused sub-block. + + properties: + compatible: + description: Compatible for SAI sub-block A or B. + enum: + - st,stm32-sai-sub-a + - st,stm32-sai-sub-b + + "#sound-dai-cells": + const: 0 + + reg: + maxItems: 1 + + clocks: + items: + - description: sai_ck clock feeding the internal clock generator. + - description: MCLK clock from a SAI set as master clock provider. + minItems: 1 + maxItems: 2 + + clock-names: + items: + - const: sai_ck + - const: MCLK + minItems: 1 + maxItems: 2 + + dmas: + items: + - description: SAI sub-block is configured as a capture DAI. + - description: SAI sub-block is configured as a playback DAI. + minItems: 1 + maxItems: 1 + + dma-names: + items: + - enum: [ rx, tx ] + + st,sync: + description: + Configure the SAI sub-block as slave of another SAI sub-block. + By default SAI sub-block is in asynchronous mode. + Must contain the phandle and index of the SAI sub-block providing + the synchronization. + allOf: + - $ref: /schemas/types.yaml#definitions/phandle-array + - maxItems: 1 + + st,iec60958: + description: + If set, support S/PDIF IEC6958 protocol for playback. + IEC60958 protocol is not available for capture. + By default, custom protocol is assumed, meaning that protocol is + configured according to protocol defined in related DAI link node, + such as i2s, left justified, right justified, dsp and pdm protocols. + allOf: + - $ref: /schemas/types.yaml#definitions/flag + + "#clock-cells": + description: Configure the SAI device as master clock provider. + const: 0 + + required: + - compatible + - "#sound-dai-cells" + - reg + - clocks + - clock-names + - dmas + - dma-names + +allOf: + - if: + properties: + compatible: + contains: + const: st,stm32f4-sai + + - then: + properties: + clocks: + minItems: 2 + maxItems: 2 + + clock-names: + items: + enum: [ x8k, x11k ] + minItems: 2 + maxItems: 2 + +additionalProperties: false + +examples: + - | + #include <dt-bindings/interrupt-controller/arm-gic.h> + #include <dt-bindings/clock/stm32mp1-clks.h> + #include <dt-bindings/reset/stm32mp1-resets.h> + sai1: sai@4400a000 { + compatible = "st,stm32h7-sai"; + #address-cells = <1>; + #size-cells = <1>; + ranges = <0 0x4400a000 0x400>; + reg = <0x4400a000 0x4>, <0x4400a3f0 0x10>; + interrupts = <GIC_SPI 87 IRQ_TYPE_LEVEL_HIGH>; + clocks = <&rcc SAI1>, <&rcc PLL1_Q>, <&rcc PLL2_P>; + clock-names = "pclk", "x8k", "x11k"; + resets = <&rcc SAI1_R>; + + sai1a: audio-controller@4400a004 { + compatible = "st,stm32-sai-sub-a"; + #sound-dai-cells = <0>; + reg = <0x4 0x1c>; + clocks = <&rcc SAI1_K>; + clock-names = "sai_ck"; + dmas = <&dmamux1 87 0x400 0x01>; + dma-names = "tx"; + }; + }; + +... -- 2.17.1

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[alsa-devel] [PATCH alsa-ucm-conf 0/7] DB410c/msm8916-wcd UCM fixes/additions
by Stephan Gerhold 11 Feb '20

11 Feb '20

This is a collection of fixes/refactoring/additions for the DB410c UCM files. My goal is to also make them usable for other MSM8916 devices, e.g. the three MSM8916 smartphones which can run mainline Linux since https://lore.kernel.org/linux-arm-msm/20190722092211.100586-1-stephan@gerho… For now this is just preparation - this patch series refactors the DB410c UCM configuration into re-usable device fragments that can be included when needed. By the way: it seems like the UCM2 conversion (specifically, commit 14facf88b56 ("DB410c: adapt to ucm2") breaks PulseAudio because the Playback/CaptureChannels were moved from all the "SectionDevice"s to "SectionVerb": I: [pulseaudio] alsa-ucm.c: UCM available for card msm8916 I: [pulseaudio] alsa-ucm.c: Set UCM verb to HiFi W: [pulseaudio] alsa-ucm.c: UCM file does not specify 'PlaybackChannels' or 'CaptureChannels'for device DigitalMic, assuming stereo duplex. W: [pulseaudio] alsa-ucm.c: UCM file does not specify 'PlaybackChannels' or 'CaptureChannels'for device SecondaryMic, assuming stereo duplex. W: [pulseaudio] alsa-ucm.c: UCM file does not specify 'PlaybackChannels' or 'CaptureChannels'for device PrimaryMic, assuming stereo duplex. W: [pulseaudio] alsa-ucm.c: UCM file does not specify 'PlaybackChannels' or 'CaptureChannels'for device Handset, assuming stereo duplex. W: [pulseaudio] alsa-ucm.c: UCM file does not specify 'PlaybackChannels' or 'CaptureChannels'for device Earpiece, assuming stereo duplex. W: [pulseaudio] alsa-ucm.c: UCM file does not specify 'PlaybackChannels' or 'CaptureChannels'for device Headphones, assuming stereo duplex. W: [pulseaudio] alsa-ucm.c: UCM file does not specify 'PlaybackChannels' or 'CaptureChannels'for device Speaker, assuming stereo duplex. I: [pulseaudio] module-alsa-card.c: Found UCM profiles E: [pulseaudio] alsa-ucm.c: No sink and source at HiFi: DigitalMic E: [pulseaudio] alsa-ucm.c: No sink and source at HiFi: SecondaryMic E: [pulseaudio] alsa-ucm.c: No sink and source at HiFi: PrimaryMic E: [pulseaudio] alsa-ucm.c: No sink and source at HiFi: Handset E: [pulseaudio] alsa-ucm.c: No sink and source at HiFi: Earpiece E: [pulseaudio] alsa-ucm.c: No sink and source at HiFi: Headphones E: [pulseaudio] alsa-ucm.c: No sink and source at HiFi: Speaker I: [pulseaudio] alsa-ucm.c: Set ucm verb to HiFi It seems like PulseAudio does not inherit them from "SectionVerb", but I'm not sure if this is a problem in PulseAudio or alsa-lib. Restoring Playback/CaptureChannels for all the "SectionDevice"s fixes the problem. Stephan Gerhold (7): DB410c: Rename Handset to HeadsetMic DB410c: Split devices into re-usable fragments in codecs/msm8916-wcd codecs/msm8916-wcd: Speaker: Reset volume on disable codecs/msm8916-wcd: Headphones: Drop invalid RDAC2 MUX value codecs/msm8916-wcd: Earpiece: Add missing Enable/DisableSequence codecs/msm8916-wcd: SecondaryMic: Select correct input codecs/msm8916-wcd: Add ConflictingDevice for capture devices ucm2/DB410c/HiFi.conf | 128 +--------------------- ucm2/codecs/msm8916-wcd/DigitalMic.conf | 18 +++ ucm2/codecs/msm8916-wcd/Earpiece.conf | 22 ++++ ucm2/codecs/msm8916-wcd/Headphones.conf | 29 +++++ ucm2/codecs/msm8916-wcd/HeadsetMic.conf | 22 ++++ ucm2/codecs/msm8916-wcd/PrimaryMic.conf | 20 ++++ ucm2/codecs/msm8916-wcd/SecondaryMic.conf | 22 ++++ ucm2/codecs/msm8916-wcd/Speaker.conf | 21 ++++ 8 files changed, 160 insertions(+), 122 deletions(-) create mode 100644 ucm2/codecs/msm8916-wcd/DigitalMic.conf create mode 100644 ucm2/codecs/msm8916-wcd/Earpiece.conf create mode 100644 ucm2/codecs/msm8916-wcd/Headphones.conf create mode 100644 ucm2/codecs/msm8916-wcd/HeadsetMic.conf create mode 100644 ucm2/codecs/msm8916-wcd/PrimaryMic.conf create mode 100644 ucm2/codecs/msm8916-wcd/SecondaryMic.conf create mode 100644 ucm2/codecs/msm8916-wcd/Speaker.conf -- 2.24.1

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[alsa-devel] [PATCH 0/9] ASoC: SOF: update S0ix/D0ix support
by Pierre-Louis Bossart 11 Feb '20

11 Feb '20

The initial support for S0ix had limitations that were identified during testing, and opportunitic transitions such as S0/D0ix was not a case that was handled. Ranjani suggested a major rework reviewed at length on GitHub [1][2]. So far all known issues are solved and no regressions are identified on 5 different platforms, so it's time to share upstream. This is not intended for 5.6 and can wait after the merge window closes. But since the code isn't simple we want to allow plenty of time for further reviews. Thanks Ranjani for this important update! [1] https://github.com/thesofproject/linux/pull/1631 [2] https://github.com/thesofproject/linux/pull/1702 Ranjani Sridharan (9): ASoC: SOF: Do not reset hw_params for streams that ignored suspend ASoC: SOF: pm: Unify suspend/resume routines ASoC: SOF: Add system_suspend_target field to struct snd_sof_dev ASoC: SOF: pm: Introduce DSP power states ASoC: SOF: Move DSP power state transitions to platform-specific ops ASoC: SOF: audio: Add helper to check if only D0i3 streams are active ASoC: SOF: Intel: hda: Amend the DSP state transition diagram ASoC: SOF: Intel: cnl: Implement feature to support DSP D0i3 in S0 ASoC: SOF: Intel: hda: Allow trace DMA in S0 when DSP is in D0I3 for debug sound/soc/sof/core.c | 4 +- sound/soc/sof/intel/cnl.c | 37 ++++- sound/soc/sof/intel/hda-dsp.c | 289 +++++++++++++++++++++++++++++++--- sound/soc/sof/intel/hda.c | 5 + sound/soc/sof/intel/hda.h | 21 ++- sound/soc/sof/ipc.c | 29 +++- sound/soc/sof/ops.h | 16 +- sound/soc/sof/pcm.c | 2 +- sound/soc/sof/pm.c | 176 ++++++++------------- sound/soc/sof/sof-audio.c | 42 ++++- sound/soc/sof/sof-audio.h | 3 +- sound/soc/sof/sof-priv.h | 43 +++-- 12 files changed, 502 insertions(+), 165 deletions(-) -- 2.20.1

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[alsa-devel] [PATCH] ALSA: hda - fixup for the bass speaker on Lenovo Carbon X1 7th gen
by Jaroslav Kysela 11 Feb '20

11 Feb '20

The auto-parser assigns the bass speaker to DAC3 (NID 0x06) which is without the volume control. I do not see a reason to use DAC2, because the shared output to all speakers produces the sufficient and well balanced sound. The stereo support is enough for this purpose (laptop). Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> --- sound/pci/hda/patch_realtek.c | 17 +++++++++++++++++ 1 file changed, 17 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index d2bf70a1d2fd..a596790d4245 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5544,6 +5544,16 @@ static void alc295_fixup_disable_dac3(struct hda_codec *codec, } } +/* force NID 0x17 (Bass Speaker) to DAC1 to share it with the main speaker */ +static void alc285_fixup_speaker2_to_dac1(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + if (action == HDA_FIXUP_ACT_PRE_PROBE) { + hda_nid_t conn[1] = { 0x02 }; + snd_hda_override_conn_list(codec, 0x17, 1, conn); + } +} + /* Hook to update amp GPIO4 for automute */ static void alc280_hp_gpio4_automute_hook(struct hda_codec *codec, struct hda_jack_callback *jack) @@ -5846,6 +5856,7 @@ enum { ALC225_FIXUP_DISABLE_MIC_VREF, ALC225_FIXUP_DELL1_MIC_NO_PRESENCE, ALC295_FIXUP_DISABLE_DAC3, + ALC285_FIXUP_SPEAKER2_TO_DAC1, ALC280_FIXUP_HP_HEADSET_MIC, ALC221_FIXUP_HP_FRONT_MIC, ALC292_FIXUP_TPT460, @@ -6646,6 +6657,10 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc295_fixup_disable_dac3, }, + [ALC285_FIXUP_SPEAKER2_TO_DAC1] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc285_fixup_speaker2_to_dac1, + }, [ALC256_FIXUP_DELL_INSPIRON_7559_SUBWOOFER] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { @@ -7221,6 +7236,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x224c, "Thinkpad", ALC298_FIXUP_TPT470_DOCK), SND_PCI_QUIRK(0x17aa, 0x224d, "Thinkpad", ALC298_FIXUP_TPT470_DOCK), SND_PCI_QUIRK(0x17aa, 0x225d, "Thinkpad T480", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), + SND_PCI_QUIRK(0x17aa, 0x2293, "Thinkpad X1 Carbon 7th", ALC285_FIXUP_SPEAKER2_TO_DAC1), SND_PCI_QUIRK(0x17aa, 0x30bb, "ThinkCentre AIO", ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY), SND_PCI_QUIRK(0x17aa, 0x30e2, "ThinkCentre AIO", ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY), SND_PCI_QUIRK(0x17aa, 0x310c, "ThinkCentre Station", ALC294_FIXUP_LENOVO_MIC_LOCATION), @@ -7405,6 +7421,7 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {.id = ALC255_FIXUP_DELL_SPK_NOISE, .name = "dell-spk-noise"}, {.id = ALC225_FIXUP_DELL1_MIC_NO_PRESENCE, .name = "alc225-dell1"}, {.id = ALC295_FIXUP_DISABLE_DAC3, .name = "alc295-disable-dac3"}, + {.id = ALC285_FIXUP_SPEAKER2_TO_DAC1, .name = "alc285-speaker2-to-dac1"}, {.id = ALC280_FIXUP_HP_HEADSET_MIC, .name = "alc280-hp-headset"}, {.id = ALC221_FIXUP_HP_FRONT_MIC, .name = "alc221-hp-mic"}, {.id = ALC298_FIXUP_SPK_VOLUME, .name = "alc298-spk-volume"}, -- 2.20.1

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[alsa-devel] [PATCH 0/3] ASoC: SOF: VirtIO: preparatory patches
by Guennadi Liakhovetski 10 Feb '20

10 Feb '20

This is the first set of patches for the SOF virtualisation work. We send these patches first because they touch the ASoC core. 2 of them are mostly cosmetic with no functional changes, but patch 2/3 might cause some discussions. Please review and comment. Thanks Guennadi Guennadi Liakhovetski (3): ASoC: (cosmetic) simplify dpcm_prune_paths() ASoC: add function parameters to enable forced path pruning ASoC: export DPCM runtime update functions include/sound/soc-dpcm.h | 30 +++++++++-- sound/soc/soc-compress.c | 2 +- sound/soc/soc-dapm.c | 8 +-- sound/soc/soc-pcm.c | 126 ++++++++++++++++++++++++++++++----------------- 4 files changed, 112 insertions(+), 54 deletions(-) -- 1.9.3

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[alsa-devel] [RESEND PATCH] ASoC: atmel: fix atmel_ssc_set_audio link failure
by Codrin Ciubotariu 10 Feb '20

10 Feb '20

From: Arnd Bergmann <arnd(a)arndb.de> The ssc audio driver can call into both pdc and dma backends. With the latest rework, the logic to do this in a safe way avoiding link errors was removed, bringing back link errors that were fixed long ago in commit 061981ff8cc8 ("ASoC: atmel: properly select dma driver state") such as sound/soc/atmel/atmel_ssc_dai.o: In function `atmel_ssc_set_audio': atmel_ssc_dai.c:(.text+0xac): undefined reference to `atmel_pcm_pdc_platform_register' Fix it this time using Makefile hacks and a comment to prevent this from accidentally getting removed again rather than Kconfig hacks. Fixes: 18291410557f ("ASoC: atmel: enable SOC_SSC_PDC and SOC_SSC_DMA in Kconfig") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> Signed-off-by: Codrin Ciubotariu <codrin.ciubotariu(a)microchip.com> --- Added my signed-off because I sent the patch, no other changes. sound/soc/atmel/Kconfig | 4 ++-- sound/soc/atmel/Makefile | 10 ++++++++-- 2 files changed, 10 insertions(+), 4 deletions(-) diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig index d1dc8e6366dc..71f2d42188c4 100644 --- a/sound/soc/atmel/Kconfig +++ b/sound/soc/atmel/Kconfig @@ -10,11 +10,11 @@ config SND_ATMEL_SOC if SND_ATMEL_SOC config SND_ATMEL_SOC_PDC - tristate + bool depends on HAS_DMA config SND_ATMEL_SOC_DMA - tristate + bool select SND_SOC_GENERIC_DMAENGINE_PCM config SND_ATMEL_SOC_SSC diff --git a/sound/soc/atmel/Makefile b/sound/soc/atmel/Makefile index 1f6890ed3738..c7d2989791be 100644 --- a/sound/soc/atmel/Makefile +++ b/sound/soc/atmel/Makefile @@ -6,8 +6,14 @@ snd-soc-atmel_ssc_dai-objs := atmel_ssc_dai.o snd-soc-atmel-i2s-objs := atmel-i2s.o snd-soc-mchp-i2s-mcc-objs := mchp-i2s-mcc.o -obj-$(CONFIG_SND_ATMEL_SOC_PDC) += snd-soc-atmel-pcm-pdc.o -obj-$(CONFIG_SND_ATMEL_SOC_DMA) += snd-soc-atmel-pcm-dma.o +# pdc and dma need to both be built-in if any user of +# ssc is built-in. +ifdef CONFIG_SND_ATMEL_SOC_PDC +obj-$(CONFIG_SND_ATMEL_SOC_SSC) += snd-soc-atmel-pcm-pdc.o +endif +ifdef CONFIG_SND_ATMEL_SOC_DMA +obj-$(CONFIG_SND_ATMEL_SOC_SSC) += snd-soc-atmel-pcm-dma.o +endif obj-$(CONFIG_SND_ATMEL_SOC_SSC) += snd-soc-atmel_ssc_dai.o obj-$(CONFIG_SND_ATMEL_SOC_I2S) += snd-soc-atmel-i2s.o obj-$(CONFIG_SND_MCHP_SOC_I2S_MCC) += snd-soc-mchp-i2s-mcc.o -- 2.20.1

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[alsa-devel] [PATCH v2 0/2] Update FE/BE trigger order based on the command
by Ranjani Sridharan 09 Feb '20

09 Feb '20

Currently, the trigger orders SND_SOC_DPCM_TRIGGER_PRE/POST determine the order in which FE DAI and BE DAI are triggered. In the case of SND_SOC_DPCM_TRIGGER_PRE, the FE DAI is triggered before the BE DAI and in the case of SND_SOC_DPCM_TRIGGER_POST, the BE DAI is triggered before the FE DAI. And this order remains the same irrespective of the trigger command. In the case of the SOF driver, during playback, the FW expects the BE DAI to be triggered before the FE DAI during the START trigger. The BE DAI trigger handles the starting of Link DMA and so it must be started before the FE DAI is started to prevent xruns during pause/release. This can be addressed by setting the trigger order for the FE dai link to SND_SOC_DPCM_TRIGGER_POST. But during the STOP trigger, the FW expects the FE DAI to be triggered before the BE DAI. Retaining the same order during the START and STOP commands, results in FW error as the DAI component in the FW is still active. The issue can be fixed by mirroring the trigger order of FE and BE DAI's during the START and STOP trigger. So, with the trigger order set to SND_SOC_DPCM_TRIGGER_PRE, the FE DAI will be trigger first during SNDRV_PCM_TRIGGER_START/STOP/RESUME and the BE DAI will be triggered first during the STOP/SUSPEND/PAUSE commands. Conversely, with the trigger order set to SND_SOC_DPCM_TRIGGER_POST, the BE DAI will be triggered first during the SNDRV_PCM_TRIGGER_START/STOP/RESUME commands and the FE DAI will be triggered first during the SNDRV_PCM_TRIGGER_STOP/SUSPEND/PAUSE commands. The above change has been tested with the Skylake driver on the Kabylake based chromebook and legacy HDA driver on Kabylake based Dell XPS13. It has also been tested on all platforms with SOF CI without any regressions for the past 3 weeks. More details on the issue can be found here: https://github.com/thesofproject/linux/issues/1160 Changes in v2: Fixed comments on function name and consolidating the switch cases. Ranjani Sridharan (2): ASoC: pcm: update FE/BE trigger order based on the command ASoC: SOF: topology: set trigger order for FE DAI link sound/soc/soc-pcm.c | 95 ++++++++++++++++++++++++++++------------ sound/soc/sof/topology.c | 4 ++ 2 files changed, 72 insertions(+), 27 deletions(-) -- 2.17.1

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[alsa-devel] USB Audio Interface / Denon MC7000 and MC8000 controller
by Tobias 07 Feb '20

07 Feb '20

Hello dear ALSA developers. I have purchased a MC7000 controller in order to control MIXXX on Ubuntu 16.04. According to the Denon specification the controller should have been class compliant but there is an issue with the Audio interface to work properly giving following message ... $ dmesg ... |[ 74.522831] usb 1-1.3: new high-speed USB device number 6 using xhci_hcd [ 74.623784] usb 1-1.3: New USB device found, idVendor=15e4, idProduct=8004 [ 74.623789] usb 1-1.3: New USB device strings: Mfr=1, Product=2, SerialNumber=3 [ 74.623793] usb 1-1.3: Product: DENON DJ MC7000 [ 74.623796] usb 1-1.3: Manufacturer: DENON DJ [ 74.623798] usb 1-1.3: SerialNumber: 201603 [ 74.625134] usb 1-1.3: clock source 65 is not valid, cannot use "|||clock source 65 is not valid, cannot use" is repeated uncountable times then |I found a discussion here for the MC8000 Audio interface which was not working and showing the same error message: https://alsa-user.narkive.com/2tDAO87f/troubleshooting-new-usb-audio-device… The solution was to change sound/usb/clock.c and recompile the kernel. This is not practical for daily use case so I would like to ask if the ALSA team could possibly find a permanent fix for ordinary users. To get my outputs for aplay -l aplay -L |||aplay -D plughw:CARD=MC7000,DEV=0 Grimmaldika-MakeBelieve.wav |lsusb -v mixxx --controllerDebug|| JackServer error message| you may have a look at the Mixxx Community Forums here(just on page, you don't need to find and read on multiple pages) https://www.mixxx.org/forums/viewtopic.php?f=7&t=12962&start=10 If you need any more information then please let me know. Thanks a lot for your support on this matter. Cheers! OsZ ||

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[alsa-devel] [PATCH] ASoC: dt-bindings: stm32: convert i2s to json-schema
by Olivier Moysan 05 Feb '20

05 Feb '20

Convert the STM32 I2S bindings to DT schema format using json-schema. Signed-off-by: Olivier Moysan <olivier.moysan(a)st.com> --- The items for dma-names and clock-names properties are all mandatory, but may be provided in any order. The syntax used for these properties allows to avoid order constraint. --- .../bindings/sound/st,stm32-i2s.txt | 62 ------------- .../bindings/sound/st,stm32-i2s.yaml | 91 +++++++++++++++++++ 2 files changed, 91 insertions(+), 62 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/st,stm32-i2s.txt create mode 100644 Documentation/devicetree/bindings/sound/st,stm32-i2s.yaml diff --git a/Documentation/devicetree/bindings/sound/st,stm32-i2s.txt b/Documentation/devicetree/bindings/sound/st,stm32-i2s.txt deleted file mode 100644 index cbf24bcd1b8d..000000000000 --- a/Documentation/devicetree/bindings/sound/st,stm32-i2s.txt +++ /dev/null @@ -1,62 +0,0 @@ -STMicroelectronics STM32 SPI/I2S Controller - -The SPI/I2S block supports I2S/PCM protocols when configured on I2S mode. -Only some SPI instances support I2S. - -Required properties: - - compatible: Must be "st,stm32h7-i2s" - - reg: Offset and length of the device's register set. - - interrupts: Must contain the interrupt line id. - - clocks: Must contain phandle and clock specifier pairs for each entry - in clock-names. - - clock-names: Must contain "i2sclk", "pclk", "x8k" and "x11k". - "i2sclk": clock which feeds the internal clock generator - "pclk": clock which feeds the peripheral bus interface - "x8k": I2S parent clock for sampling rates multiple of 8kHz. - "x11k": I2S parent clock for sampling rates multiple of 11.025kHz. - - dmas: DMA specifiers for tx and rx dma. - See Documentation/devicetree/bindings/dma/stm32-dma.txt. - - dma-names: Identifier for each DMA request line. Must be "tx" and "rx". - - pinctrl-names: should contain only value "default" - - pinctrl-0: see Documentation/devicetree/bindings/pinctrl/st,stm32-pinctrl.yaml - -Optional properties: - - resets: Reference to a reset controller asserting the reset controller - -The device node should contain one 'port' child node with one child 'endpoint' -node, according to the bindings defined in Documentation/devicetree/bindings/ -graph.txt. - -Example: -sound_card { - compatible = "audio-graph-card"; - dais = <&i2s2_port>; -}; - -i2s2: audio-controller@40003800 { - compatible = "st,stm32h7-i2s"; - reg = <0x40003800 0x400>; - interrupts = <36>; - clocks = <&rcc PCLK1>, <&rcc SPI2_CK>, <&rcc PLL1_Q>, <&rcc PLL2_P>; - clock-names = "pclk", "i2sclk", "x8k", "x11k"; - dmas = <&dmamux2 2 39 0x400 0x1>, - <&dmamux2 3 40 0x400 0x1>; - dma-names = "rx", "tx"; - pinctrl-names = "default"; - pinctrl-0 = <&pinctrl_i2s2>; - - i2s2_port: port@0 { - cpu_endpoint: endpoint { - remote-endpoint = <&codec_endpoint>; - format = "i2s"; - }; - }; -}; - -audio-codec { - codec_port: port@0 { - codec_endpoint: endpoint { - remote-endpoint = <&cpu_endpoint>; - }; - }; -}; diff --git a/Documentation/devicetree/bindings/sound/st,stm32-i2s.yaml b/Documentation/devicetree/bindings/sound/st,stm32-i2s.yaml new file mode 100644 index 000000000000..cdfb375c7a14 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/st,stm32-i2s.yaml @@ -0,0 +1,91 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/st,stm32-i2s.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: STMicroelectronics STM32 SPI/I2S Controller + +maintainers: + - Olivier Moysan <olivier.moysan(a)st.com> + +description: + The SPI/I2S block supports I2S/PCM protocols when configured on I2S mode. + Only some SPI instances support I2S. + +properties: + compatible: + enum: + - st,stm32h7-i2s + + "#sound-dai-cells": + const: 0 + + reg: + maxItems: 1 + + clocks: + items: + - description: clock feeding the peripheral bus interface. + - description: clock feeding the internal clock generator. + - description: I2S parent clock for sampling rates multiple of 8kHz. + - description: I2S parent clock for sampling rates multiple of 11.025kHz. + + clock-names: + items: + - enum: [ pclk, i2sclk, x8k, x11k ] + - enum: [ pclk, i2sclk, x8k, x11k ] + - enum: [ pclk, i2sclk, x8k, x11k ] + - enum: [ pclk, i2sclk, x8k, x11k ] + allOf: + - $ref: /schemas/types.yaml#/definitions/string-array + + interrupts: + maxItems: 1 + + dmas: + items: + - description: audio capture DMA. + - description: audio playback DMA. + + dma-names: + items: + - enum: [ rx, tx ] + - enum: [ rx, tx ] + allOf: + - $ref: /schemas/types.yaml#/definitions/string-array + + resets: + maxItems: 1 + +required: + - compatible + - "#sound-dai-cells" + - reg + - clocks + - clock-names + - interrupts + - dmas + - dma-names + +additionalProperties: false + +examples: + - | + #include <dt-bindings/interrupt-controller/arm-gic.h> + #include <dt-bindings/clock/stm32mp1-clks.h> + i2s2: audio-controller@4000b000 { + compatible = "st,stm32h7-i2s"; + #sound-dai-cells = <0>; + reg = <0x4000b000 0x400>; + clocks = <&rcc SPI2>, <&rcc SPI2_K>, <&rcc PLL3_Q>, <&rcc PLL3_R>; + clock-names = "pclk", "i2sclk", "x8k", "x11k"; + interrupts = <GIC_SPI 35 IRQ_TYPE_LEVEL_HIGH>; + dmas = <&dmamux1 39 0x400 0x01>, + <&dmamux1 40 0x400 0x01>; + dma-names = "rx", "tx"; + pinctrl-names = "default"; + pinctrl-0 = <&i2s2_pins_a>; + }; + +... -- 2.17.1

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Re: [alsa-devel] [FFADO-user] Toneweal / Trigaudio FW66 device
by Takashi Sakamoto 04 Feb '20

04 Feb '20

Hi, I'm an author of ALSA bebob driver (snd-bebob). On Sun, Jan 19, 2020 at 03:01:00PM +0100, Daniel Jozsef wrote: > It's a Toneweal FW66, a 6x6 in/out 24bit 96kHz audio interface with MIDI. I > kinda like it for its build quality and the time I spent with it. :D The > largest chip in the device is labeled Trigaudio MNP-TA110. I searched for > information on this device, and found very little - it's a Taiwanese > company. There seems evidence that a "Trigaudio FW66" also exists, so the > device may have been sold under different brands in different markets. In my opinion, FW66 is an application of ArchWave BeBoB solution for audio and music units on IEEE 1394 bus (ArchWave is formerly known as BridgeCo) since I can find file structure for the solution in driver package shipped by the vendor. I guess that you can see large ASIC labelled with 'BridgeCo' (or 'ArchWave') DM1000/1100/1500 inner the device. If so, the device is possibly handled by implementation of FFADO. But as a quick glance to your log, the implementation of FFADO fails to detect it: > daniel@gibbonmoon:~$ sudo ffado-test Discover > ... > 02878614308: Warning (bebob_avdevice.cpp)[ 228] discover: Using generic BeBoB support for unsupported device 'ToneWeal FW66' > 02878620133: Debug (bebob_avdevice_subunit.cpp)[ 83] discover: Discovering BeBoB::AudioSubunit... > 02878620175: Debug (avc_audiosubunit.cpp)[ 56] discover: Discovering BeBoB::AudioSubunit... > 02878620196: Debug (avc_subunit.cpp)[ 108] discoverPlugs: Discovering plugs... > 02878778208: Error (bebob_avplug.cpp)[ 237] discoverPlugType: Plug does not implement extended plug info plug type info command > 02878778236: Error (bebob_avplug.cpp)[ 120] discover: discover: Could not discover plug type (1,1,0,0,1) > 02878778272: Error (avc_subunit.cpp)[ 189] discoverPlugs: plug discover failed > 02878778283: Error (avc_subunit.cpp)[ 131] discoverPlugs: destination plug discovering failed > 02878778301: Error (avc_subunit.cpp)[ 99] discover: plug discovery failed > 02878778312: Error (avc_unit.cpp)[ 283] enumerateSubUnits: enumerateSubUnits: Could not discover subunit_id = 0, subunit_type = 1 (Audio) > 02878778335: Error (avc_unit.cpp)[ 177] discover: Could not enumerate sub units > 02878778347: Error (bebob_avdevice.cpp)[ 232] discover: Could not discover unit > 02878778376: Error (devicemanager.cpp)[ 628] discover: could not discover device > 02878778419: Debug (devicemanager.cpp)[ 661] discover: Discovery finished... ... The reason is the unit returns NOT_IMPLEMENTED response against vendor specific AV/C command (Extended Plug Information command defined by BridgeCo.) for Audio subunit. I guess that the device has no Audio subunit but the implementation performs to use it without checking available subunits. For my information, would you please clone linux-firewire-utils[1] into your system and build it, then run below command to dump device information: $ ./firewire-request /dev/fw1 read 0xffffc8020000 60 result: 000: 62 72 69 64 67 65 43 6f 03 00 00 00 00 00 00 00 bridgeCo........ result: 010: 00 96 14 00 22 03 00 00 19 00 00 00 00 00 00 00 ...."........... result: 020: 32 30 30 39 30 36 30 39 31 36 32 39 34 30 20 20 20090609162940 result: 030: 00 00 00 19 07 1f 00 00 80 00 0c 40 70 35 14 00 ...........@p5.. result: 040: 32 30 30 39 30 36 30 39 31 36 32 32 31 39 20 20 20090609162219 result: 050: 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 ................ The above is a sample of Phonic Firefly 202. I expect the first line includes bytes represent 'bridgeCo'. [1] https://github.com/cladisch/linux-firewire-utils Regards Takashi Sakamoto

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