December 2021 - Alsa-devel - mailman.alsa-project.org

Does alsa-lib call this kernel api?
by GitHub issues - opened 14 Dec '21

14 Dec '21

alsa-project/alsa-lib issue #204 was opened from Nick0412: Where in the alsa library does it reference the alsa kernel api: https://www.kernel.org/doc/html/latest/sound/kernel-api/alsa-driver-api.htm… Do these two apis interact at all? Issue URL : https://github.com/alsa-project/alsa-lib/issues/204 Repository URL: https://github.com/alsa-project/alsa-lib

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[PATCH 0/4] fsl-asoc-card: Add optional dt property for setting mclk-id
by Ariel D'Alessandro 13 Dec '21

13 Dec '21

This is a follow up of patchset: [RFC patch 0/5] Support BCLK input clock in tlv320aic31xx Sound cards may allow using different main clock inputs. In the generic fsl-asoc-card driver, these values are hardcoded for each specific card configuration. Let's make it more flexible, allowing setting mclk-id from the device-tree node. Ariel D'Alessandro (4): dt-bindings: sound: Rename tlv320aic31xx-micbias as tlv320aic31xx dt-bindings: tlv320aic31xx: Define PLL clock inputs ASoC: fsl-asoc-card: Add optional dt property for setting mclk-id ASoC: fsl-asoc-card: Remove BCLK default value for tlv320aic31xx card .../devicetree/bindings/sound/fsl-asoc-card.txt | 1 + .../devicetree/bindings/sound/tlv320aic31xx.txt | 2 +- arch/arm/boot/dts/am43x-epos-evm.dts | 2 +- include/dt-bindings/sound/tlv320aic31xx-micbias.h | 9 --------- include/dt-bindings/sound/tlv320aic31xx.h | 14 ++++++++++++++ sound/soc/codecs/tlv320aic31xx.c | 2 +- sound/soc/fsl/fsl-asoc-card.c | 7 ++++++- 7 files changed, 24 insertions(+), 13 deletions(-) delete mode 100644 include/dt-bindings/sound/tlv320aic31xx-micbias.h create mode 100644 include/dt-bindings/sound/tlv320aic31xx.h -- 2.30.2

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[PATCH 0/7] ASoC: Intel: SOF: clarifications on hardware support
by Pierre-Louis Bossart 13 Dec '21

13 Dec '21

This patchset revisits the Intel hardware support in SOF. The HDAudio DMA position information was not following hardware recommended programming sequences (similar changes are already part of the HDaudio legacy driver), and the stream assignment applied a work-around that was only needed on specific versions of hardware. These changes are not tagged as 'Fixes' and don't need to be applied to -stable versions. While we're at it, DPIB handling is improved, useless fields removed, a comment added on JasperLake support, and IceLake-specific routines are isolated. Pierre-Louis Bossart (6): ASoC: SOF: Intel: hda-stream: limit PROCEN workaround ASoC: SOF: Intel: hda-ctrl: apply symmetry for DPIB ASoC: SOF: hda-stream: only enable DPIB if needed ASoC: SOF: Intel: hda: add quirks for HDAudio DMA position information ASoC: SOF: Intel: hda-dai: remove unused fields ASoC: SOF: Intel: add comment on JasperLake support Ranjani Sridharan (1): ASoC: SOF: Intel: ICL: move ICL-specific ops to icl.c sound/soc/sof/intel/apl.c | 1 + sound/soc/sof/intel/cnl.c | 7 +++ sound/soc/sof/intel/hda-ctrl.c | 2 +- sound/soc/sof/intel/hda-dai.c | 4 -- sound/soc/sof/intel/hda-loader.c | 64 ------------------------ sound/soc/sof/intel/hda-pcm.c | 86 ++++++++++++++++++++++---------- sound/soc/sof/intel/hda-stream.c | 25 ++++++---- sound/soc/sof/intel/hda.c | 9 +++- sound/soc/sof/intel/hda.h | 8 ++- sound/soc/sof/intel/icl.c | 67 ++++++++++++++++++++++++- sound/soc/sof/intel/shim.h | 4 ++ 11 files changed, 169 insertions(+), 108 deletions(-) -- 2.25.1

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[PATCH] ASoC: dt-bindings: tegra: Document interconnects property
by Thierry Reding 13 Dec '21

13 Dec '21

From: Thierry Reding <treding(a)nvidia.com> Add the interconnects and interconnect-names properties to the bindings for the sound card on various NVIDIA Tegra based boards. These are used to describe the device's memory paths to and from memory. Signed-off-by: Thierry Reding <treding(a)nvidia.com> --- .../bindings/sound/nvidia,tegra-audio-graph-card.yaml | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-graph-card.yaml b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-graph-card.yaml index 5bdd30a8a404..b4bee466d67a 100644 --- a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-graph-card.yaml +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-graph-card.yaml @@ -44,6 +44,16 @@ properties: minItems: 1 maxItems: 3 + interconnects: + items: + - description: APE read memory client + - description: APE write memory client + + interconnect-names: + items: + - const: dma-mem # read + - const: write + iommus: maxItems: 1 -- 2.33.1

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[PATCH v1 1/2] ASoC: tegra: Add DAPM switches for headphones and mic jack
by Dmitry Osipenko 13 Dec '21

13 Dec '21

UCM of Acer Chromebook (Nyan) uses DAPM switches of headphones and mic jack. These switches were lost by accident during unification of the machine drivers, restore them. Cc: <stable(a)vger.kernel.org> Fixes: cc8f70f ("ASoC: tegra: Unify ASoC machine drivers") Reported-by: Thomas Graichen <thomas.graichen(a)gmail.com> # T124 Nyan Big Tested-by: Thomas Graichen <thomas.graichen(a)gmail.com> # T124 Nyan Big Signed-off-by: Dmitry Osipenko <digetx(a)gmail.com> --- sound/soc/tegra/tegra_asoc_machine.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/tegra/tegra_asoc_machine.c b/sound/soc/tegra/tegra_asoc_machine.c index b95438c3dbf7..f3e86bd714b4 100644 --- a/sound/soc/tegra/tegra_asoc_machine.c +++ b/sound/soc/tegra/tegra_asoc_machine.c @@ -116,6 +116,8 @@ static const struct snd_kcontrol_new tegra_machine_controls[] = { SOC_DAPM_PIN_SWITCH("Headset Mic"), SOC_DAPM_PIN_SWITCH("Internal Mic 1"), SOC_DAPM_PIN_SWITCH("Internal Mic 2"), + SOC_DAPM_PIN_SWITCH("Headphones"), + SOC_DAPM_PIN_SWITCH("Mic Jack"), }; int tegra_asoc_machine_init(struct snd_soc_pcm_runtime *rtd) -- 2.33.1

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[PATCH] SoC: qcom: Distinguish headset codec by codec_dai->name
by Judy Hsiao 13 Dec '21

13 Dec '21

Distinguish which headset codec is on the board by codec_dai->name instead of card->name. Fixes: 425c5fce8a03 ("ASoC: qcom: Add support for ALC5682I-VS codec") Signed-off-by: Judy Hsiao <judyhsiao(a)chromium.org> --- sound/soc/qcom/sc7180.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/qcom/sc7180.c b/sound/soc/qcom/sc7180.c index 2fff764a00a7..37225ef2563a 100644 --- a/sound/soc/qcom/sc7180.c +++ b/sound/soc/qcom/sc7180.c @@ -131,13 +131,13 @@ static int sc7180_snd_startup(struct snd_pcm_substream *substream) struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); int pll_id, pll_source, pll_in, pll_out, clk_id, ret; - if (!(strcmp(card->name, "sc7180-rt5682-max98357a-1mic"))) { + if (!strcmp(codec_dai->name, "rt5682-aif1")) { pll_source = RT5682_PLL1_S_MCLK; pll_id = 0; clk_id = RT5682_SCLK_S_PLL1; pll_out = RT5682_PLL1_FREQ; pll_in = DEFAULT_MCLK_RATE; - } else if (!(strcmp(card->name, "sc7180-rt5682s-max98357a-1mic"))) { + } else if (!strcmp(codec_dai->name, "rt5682s-aif1")) { pll_source = RT5682S_PLL_S_MCLK; pll_id = RT5682S_PLL2; clk_id = RT5682S_SCLK_S_PLL2; -- 2.34.1.173.g76aa8bc2d0-goog

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[PATCH] ASoC: dt-bindings: add missing pins
by David Heidelberg 13 Dec '21

13 Dec '21

Add pins missing in documentation, but present in the hardware. Fixes: 0e826e867264 ("ASoC: add RT5677 CODEC driver") Signed-off-by: David Heidelberg <david(a)ixit.cz> --- Documentation/devicetree/bindings/sound/rt5677.txt | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/rt5677.txt b/Documentation/devicetree/bindings/sound/rt5677.txt index da2430099181..74b6359b802d 100644 --- a/Documentation/devicetree/bindings/sound/rt5677.txt +++ b/Documentation/devicetree/bindings/sound/rt5677.txt @@ -54,9 +54,21 @@ Pins on the device (for linking into audio routes): * DMIC2 * DMIC3 * DMIC4 + * DMIC L1 + * DMIC L2 + * DMIC L3 + * DMIC L4 + * DMIC R1 + * DMIC R2 + * DMIC R3 + * DMIC R4 * LOUT1 * LOUT2 * LOUT3 + * PDM1L + * PDM1R + * PDM2L + * PDM2R Example: -- 2.33.0

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Re: [PATCH v2 1/4] ASoC: dt-bindings: stm32: i2s: add audio-graph-card port
by Rob Herring 13 Dec '21

13 Dec '21

On Tue, Dec 7, 2021 at 7:52 AM Alexandre TORGUE <alexandre.torgue(a)foss.st.com> wrote: > > Hi Rob > > On 12/1/21 11:34 PM, Rob Herring wrote: > > On Fri, Nov 26, 2021 at 11:25:27AM +0100, Olivier MOYSAN wrote: > >> Hi Rob, > >> > >> On 11/25/21 10:26 PM, Rob Herring wrote: > >>> On Thu, 25 Nov 2021 15:40:50 +0100, Olivier Moysan wrote: > >>>> The STM2 I2S DAI can be connected via the audio-graph-card. > >>>> Add port entry into the bindings. > >>>> > >>>> Signed-off-by: Olivier Moysan <olivier.moysan(a)foss.st.com> > >>>> --- > >>>> Documentation/devicetree/bindings/sound/st,stm32-i2s.yaml | 5 +++++ > >>>> 1 file changed, 5 insertions(+) > >>>> > >>> > >>> Running 'make dtbs_check' with the schema in this patch gives the > >>> following warnings. Consider if they are expected or the schema is > >>> incorrect. These may not be new warnings. > >>> > >>> Note that it is not yet a requirement to have 0 warnings for dtbs_check. > >>> This will change in the future. > >>> > >>> Full log is available here: https://patchwork.ozlabs.org/patch/1559750 > >>> > >>> > >>> audio-controller@4000b000: 'port' does not match any of the regexes: '^port@[0-9]', 'pinctrl-[0-9]+' > >>> arch/arm/boot/dts/stm32mp157a-dk1.dt.yaml > >>> arch/arm/boot/dts/stm32mp157c-dk2.dt.yaml > >>> > >> > >> This warning is not a new one. > >> > >> The i2s2 node in stm32mp15xx-dkx.dtsi would require the following binding: > >> port: > >> $ref: audio-graph-port.yaml# > >> unevaluatedProperties: false > >> > >> However the spi binding requires to introduce a unit address: > >> patternProperties: > >> '^port@[0-9]': > >> $ref: audio-graph-port.yaml# > >> unevaluatedProperties: false > >> > >> The warning can be removed by re-ordering the bindings patches in the serie, > >> as "additionalProperties: true" makes the check more tolerant on extra > >> properties. > > > > That's never right. > > > >> The patch "ASoC: dt-bindings: stm32: i2s: add audio-graph-card port" can > >> even be merely dropped. > >> So, I suggest to resend the serie without audio-graph-card patch. > > > > Only if you aren't using audio-graph-card. > > > >> > >> Does it sound too permissive to you ? > > > > I think perhaps you need to combine the schemas into 1. Or you need to > > restructure your dtsi files such that you only add spi specific > > properties when spi mode is enabled and only add i2s specific properties > > when i2s mode is enabled. Or use the /delete-property/ directive. > > Initially the aim of this series was to fix a "make W=1" warnings seen > on spi and i2s nodes (duplicate unit-address). Moving both nodes in a > common node + using a different compatible depending on SPI or I2S usage > sounded good) but it is not enough. In this series the common node is > named as following: "spi2s2: spi@4000b000". It is fine for a spi usage > but if we want to use this "common node" with I2S compatible and > specific bindings, the node name remains spi@... and then specific spi > checks are done. For this with this series applied we got this issue > reported by spi-controller.yaml: > > spi@4000b000: port@0: 'compatible' is a required property > > So, if we use two separates nodes we got W=1 warning and if we use a > common node we got yaml check issue. One possibility would be to use a > common node with a new node name (for example i2spi@...) but I think it > is not acceptable. It is acceptable, see this thread[1]. Rob [1] https://lore.kernel.org/all/20211203183517.11390-1-semen.protsenko@linaro.o…

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[v2] SoC: qcom: Distinguish headset codec by codec_dai->name
by Judy Hsiao 13 Dec '21

13 Dec '21

Distinguish which headset codec is on the board by codec_dai->name instead of card->name. It fixes the crash of being unable to handle kernel paging requests at virtual address ADDR by initializing the correct audio codec on the board. Call stack of the crash: ``` Unable to handle kernel paging request at virtual address ffffffbfe7bba9ce Mem abort info: ESR = 0x96000005 EC = 0x25: DABT (current EL), IL = 32 bits SET = 0, FnV = 0 EA = 0, S1PTW = 0 Data abort info: ISV = 0, ISS = 0x00000005 CM = 0, WnR = 0 swapper pgtable: 4k pages, 39-bit VAs, pgdp=000000008379a000 [ffffffbfe7bba9ce] pgd=0000000000000000, pud=0000000000000000 Internal error: Oops: 96000005 [#1] PREEMPT SMP Modules linked in: ip6table_nat fuse uvcvideo videobuf2_vmalloc [trimmed] CPU: 7 PID: 2168 Comm: cras Not tainted 5.4.163-lockdep-17364-gfe3d4f499cf1 #1 Hardware name: Google Pompom (rev2) with LTE (DT) pstate: 00400009 (nzcv daif +PAN -UAO) pc : rt5682_set_component_pll+0xcc/0xb78 [snd_soc_rt5682] lr : rt5682_set_component_pll+0xbc/0xb78 [snd_soc_rt5682] sp : ffffff808ed7f320 x29: ffffff808ed7f390 x28: dfffffd000000000 x27: ffffff80b1409550 x26: 00000000aaaaaaaa x25: ffffff80d0a0b820 x24: ffffff80bc1f7098 x23: ffffff809332a080 x22: 00000000aaaaaaaa x21: 1ffffff01783ee13 x20: 00000000aaaaaaaa x19: 00000000aaaaaaaa x18: 1ffffff011dafe18 x17: 0000000000000000 x16: 0000000000000201 x15: 0000000000000000 x14: 0000000062f77d15 x13: dfffffd000000000 x12: ffffffd01302ed7c x11: 0000000000000000 x10: ffffff7f3ddd4e74 x9 : 0000000000000000 x8 : 1fffffefe7bba9ce x7 : aaaaaaaaaaaaaaaa x6 : 0000000000000000 x5 : 0000000000000000 x4 : 0000000000000008 x3 : 0000000000000000 x2 : 0000000000000008 x1 : 00000000000000aa x0 : ffffff808ed7f358 Call trace: rt5682_set_component_pll+0xcc/0xb78 [snd_soc_rt5682] snd_soc_component_set_pll+0x90/0x154 snd_soc_dai_set_pll+0xf4/0x1ac sc7180_snd_startup+0x268/0x3c0 [snd_soc_sc7180] snd_soc_link_startup+0xa4/0x180 soc_pcm_open+0x35c/0x15c8 snd_pcm_open_substream+0xa90/0x13b0 snd_pcm_open+0x1a4/0x55c snd_pcm_capture_open+0x7c/0xe8 snd_open+0x2b8/0x2e4 chrdev_open+0x364/0x3d4 do_dentry_open+0x66c/0xc58 vfs_open+0x7c/0x8c path_openat+0x108c/0x2bbc do_filp_open+0x15c/0x258 do_sys_open+0x278/0x62c __arm64_compat_sys_openat+0x9c/0xb0 el0_svc_common+0x1c0/0x3dc el0_svc_compat_handler+0x88/0xd4 el0_svc_compat+0x8/0x2c Code: 8b3acae8 910d310a d343fd48 a9012be8 (38fc6908) ``` Fixes: 425c5fce8a03 ("ASoC: qcom: Add support for ALC5682I-VS codec") Signed-off-by: Judy Hsiao <judyhsiao(a)chromium.org> --- sound/soc/qcom/sc7180.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/qcom/sc7180.c b/sound/soc/qcom/sc7180.c index 2fff764a00a7..37225ef2563a 100644 --- a/sound/soc/qcom/sc7180.c +++ b/sound/soc/qcom/sc7180.c @@ -131,13 +131,13 @@ static int sc7180_snd_startup(struct snd_pcm_substream *substream) struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); int pll_id, pll_source, pll_in, pll_out, clk_id, ret; - if (!(strcmp(card->name, "sc7180-rt5682-max98357a-1mic"))) { + if (!strcmp(codec_dai->name, "rt5682-aif1")) { pll_source = RT5682_PLL1_S_MCLK; pll_id = 0; clk_id = RT5682_SCLK_S_PLL1; pll_out = RT5682_PLL1_FREQ; pll_in = DEFAULT_MCLK_RATE; - } else if (!(strcmp(card->name, "sc7180-rt5682s-max98357a-1mic"))) { + } else if (!strcmp(codec_dai->name, "rt5682s-aif1")) { pll_source = RT5682S_PLL_S_MCLK; pll_id = RT5682S_PLL2; clk_id = RT5682S_SCLK_S_PLL2; -- 2.31.0

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[PATCH v2] ALSA: gus: Fix memory leaks at memory allocator error paths
by Takashi Iwai 13 Dec '21

13 Dec '21

When snd_gf1_mem_xalloc() returns NULL, the current code still leaves the formerly allocated block.name string but returns an error immediately. This patch does code-refactoring to move the kstrdup() call itself into snd_gf1_mem_xalloc() and deals with the resource free in the helper code by itself for fixing those memory leaks. Suggested-by: Jaroslav Kysela <perex(a)perex.cz> Link: https://lore.kernel.org/r/20211213132444.22385-2-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/isa/gus/gus_mem.c | 24 ++++++++++++------------ 1 file changed, 12 insertions(+), 12 deletions(-) diff --git a/sound/isa/gus/gus_mem.c b/sound/isa/gus/gus_mem.c index 5e3ff3137dd7..3e56c01c4544 100644 --- a/sound/isa/gus/gus_mem.c +++ b/sound/isa/gus/gus_mem.c @@ -24,8 +24,9 @@ void snd_gf1_mem_lock(struct snd_gf1_mem * alloc, int xup) } } -static struct snd_gf1_mem_block *snd_gf1_mem_xalloc(struct snd_gf1_mem * alloc, - struct snd_gf1_mem_block * block) +static struct snd_gf1_mem_block * +snd_gf1_mem_xalloc(struct snd_gf1_mem *alloc, struct snd_gf1_mem_block *block, + const char *name) { struct snd_gf1_mem_block *pblock, *nblock; @@ -33,6 +34,12 @@ static struct snd_gf1_mem_block *snd_gf1_mem_xalloc(struct snd_gf1_mem * alloc, if (nblock == NULL) return NULL; *nblock = *block; + nblock->name = kstrdup(name, GFP_KERNEL); + if (!nblock->name) { + kfree(nblock); + return NULL; + } + pblock = alloc->first; while (pblock) { if (pblock->ptr > nblock->ptr) { @@ -198,12 +205,7 @@ struct snd_gf1_mem_block *snd_gf1_mem_alloc(struct snd_gf1_mem * alloc, int owne if (share_id != NULL) memcpy(&block.share_id, share_id, sizeof(block.share_id)); block.owner = owner; - block.name = kstrdup(name, GFP_KERNEL); - if (block.name == NULL) { - snd_gf1_mem_lock(alloc, 1); - return NULL; - } - nblock = snd_gf1_mem_xalloc(alloc, &block); + nblock = snd_gf1_mem_xalloc(alloc, &block, name); snd_gf1_mem_lock(alloc, 1); return nblock; } @@ -240,14 +242,12 @@ int snd_gf1_mem_init(struct snd_gus_card * gus) if (gus->gf1.enh_mode) { block.ptr = 0; block.size = 1024; - block.name = kstrdup("InterWave LFOs", GFP_KERNEL); - if (block.name == NULL || snd_gf1_mem_xalloc(alloc, &block) == NULL) + if (!snd_gf1_mem_xalloc(alloc, &block, "InterWave LFOs")) return -ENOMEM; } block.ptr = gus->gf1.default_voice_address; block.size = 4; - block.name = kstrdup("Voice default (NULL's)", GFP_KERNEL); - if (block.name == NULL || snd_gf1_mem_xalloc(alloc, &block) == NULL) + if (!snd_gf1_mem_xalloc(alloc, &block, "Voice default (NULL's)")) return -ENOMEM; #ifdef CONFIG_SND_DEBUG snd_card_ro_proc_new(gus->card, "gusmem", gus, snd_gf1_mem_info_read); -- 2.31.1

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