November 2019 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] dt-bindings: sound: adau7118: Fix example warning
by Maxime Ripard 04 Nov '19

04 Nov '19

The ADAU7118 has an example where the codec has an i2c address of 14, and the unit address set to 14 as well. However, while the address is expressed in decimal, the unit-address is supposed to be in hexadecimal, which ends up with two different addresses that trigger a DTC warning. Fix this by setting the unit address to (0x)e. Cc: Nuno Sá <nuno.sa(a)analog.com> Fixes: 969d49b2cdc8 ("dt-bindings: asoc: Add ADAU7118 documentation") Signed-off-by: Maxime Ripard <maxime(a)cerno.tech> --- Documentation/devicetree/bindings/sound/adi,adau7118.yaml | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/adi,adau7118.yaml b/Documentation/devicetree/bindings/sound/adi,adau7118.yaml index c3f10afbdd6f..65f6844a0c6d 100644 --- a/Documentation/devicetree/bindings/sound/adi,adau7118.yaml +++ b/Documentation/devicetree/bindings/sound/adi,adau7118.yaml @@ -65,7 +65,7 @@ examples: /* example with i2c support */ #address-cells = <1>; #size-cells = <0>; - adau7118_codec: audio-codec@14 { + adau7118_codec: audio-codec@e { compatible = "adi,adau7118"; reg = <14>; #sound-dai-cells = <0>; -- 2.23.0

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[alsa-devel] [PATCH] ASoC: wm8524: Add support S32_LE
by S.j. Wang 04 Nov '19

04 Nov '19

Allow 32bit sample with this codec. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/codecs/wm8524.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8524.c b/sound/soc/codecs/wm8524.c index 91e3d1570c45..4e9ab542f648 100644 --- a/sound/soc/codecs/wm8524.c +++ b/sound/soc/codecs/wm8524.c @@ -159,7 +159,9 @@ static int wm8524_mute_stream(struct snd_soc_dai *dai, int mute, int stream) #define WM8524_RATES SNDRV_PCM_RATE_8000_192000 -#define WM8524_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE) +#define WM8524_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\ + SNDRV_PCM_FMTBIT_S24_LE |\ + SNDRV_PCM_FMTBIT_S32_LE) static const struct snd_soc_dai_ops wm8524_dai_ops = { .startup = wm8524_startup, -- 2.21.0

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[alsa-devel] [PATCH] ALSA: pcm: Fix missing check of the new non-cached buffer type
by Takashi Iwai 04 Nov '19

04 Nov '19

The check for the mmap support via hw_support_mmap() function misses the case where the device is with SNDRV_DMA_TYPE_DEV_UC, which should have been treated equally as SNDRV_DMA_TYPE_DEV. Let's fix it. Note that this bug doesn't hit any practical problem, because SNDRV_DMA_TYPE_DEV_UC is used only for x86-specific drivers (snd-hda-intel and snd-intel8x0) for the specific platforms that need the non-cached buffers. And, on such platforms, hw_support_mmap() already returns true in anyway. That's the reason I didn't put Cc-to-stable mark here. This is only for any theoretical future extension. Fixes: 425da159707b ("ALSA: pcm: use dma_can_mmap() to check if a device supports dma_mmap_*") Fixes: 42e748a0b325 ("ALSA: memalloc: Add non-cached buffer type") Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/pcm_native.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 91c6ad58729f..c3a139436ac2 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -222,7 +222,8 @@ static bool hw_support_mmap(struct snd_pcm_substream *substream) return false; if (substream->ops->mmap || - substream->dma_buffer.dev.type != SNDRV_DMA_TYPE_DEV) + (substream->dma_buffer.dev.type != SNDRV_DMA_TYPE_DEV && + substream->dma_buffer.dev.type != SNDRV_DMA_TYPE_DEV_UC)) return true; return dma_can_mmap(substream->dma_buffer.dev.dev); -- 2.16.4

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[alsa-devel] [PATCH] ALSA: bebob: fix to detect configured source of sampling clock for Focusrite Saffire Pro i/o series
by Takashi Sakamoto 04 Nov '19

04 Nov '19

For Focusrite Saffire Pro i/o, the lowest 8 bits of register represents configured source of sampling clock. The next lowest 8 bits represents whether the configured source is actually detected or not just after the register is changed for the source. Current implementation evaluates whole the register to detect configured source. This results in failure due to the next lowest 8 bits when the source is connected in advance. This commit fixes the bug. Fixes: 25784ec2d034 ("ALSA: bebob: Add support for Focusrite Saffire/SaffirePro series") Cc: <stable(a)vger.kernel.org> # v3.16+ Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/bebob/bebob_focusrite.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/firewire/bebob/bebob_focusrite.c b/sound/firewire/bebob/bebob_focusrite.c index 32b864bee25f..06d6a37cd853 100644 --- a/sound/firewire/bebob/bebob_focusrite.c +++ b/sound/firewire/bebob/bebob_focusrite.c @@ -27,6 +27,8 @@ #define SAFFIRE_CLOCK_SOURCE_SPDIF 1 /* clock sources as returned from register of Saffire Pro 10 and 26 */ +#define SAFFIREPRO_CLOCK_SOURCE_SELECT_MASK 0x000000ff +#define SAFFIREPRO_CLOCK_SOURCE_DETECT_MASK 0x0000ff00 #define SAFFIREPRO_CLOCK_SOURCE_INTERNAL 0 #define SAFFIREPRO_CLOCK_SOURCE_SKIP 1 /* never used on hardware */ #define SAFFIREPRO_CLOCK_SOURCE_SPDIF 2 @@ -189,6 +191,7 @@ saffirepro_both_clk_src_get(struct snd_bebob *bebob, unsigned int *id) map = saffirepro_clk_maps[1]; /* In a case that this driver cannot handle the value of register. */ + value &= SAFFIREPRO_CLOCK_SOURCE_SELECT_MASK; if (value >= SAFFIREPRO_CLOCK_SOURCE_COUNT || map[value] < 0) { err = -EIO; goto end; -- 2.20.1

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[alsa-devel] [GIT PULL] soundwire fixes for v5.4-rc6
by Vinod Koul 02 Nov '19

02 Nov '19

Hi Greg, Please pull to receive couple of fixes for soundwire. All these changes have been sitting in linux-next as well The following changes since commit 54ecb8f7028c5eb3d740bb82b0f1d90f2df63c5c: Linux 5.4-rc1 (2019-09-30 10:35:40 -0700) are available in the Git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/vkoul/soundwire.git tags/soundwire-5.4-rc6 for you to fetch changes up to 7b47ad3322d114736e08cc244a84fe7af9b861f6: soundwire: slave: fix scanf format (2019-10-24 16:55:45 +0530) ---------------------------------------------------------------- soundwire fixes for v5.4-rc6 - Kconfig fixes to ensure soundwire is built only for ACPI and DT platform - fix for intel PDI offsets and numbers - slave scanf format fix ---------------------------------------------------------------- Bard Liao (1): soundwire: intel: fix intel_register_dai PDI offsets and numbers Michal Suchanek (2): soundwire: depend on ACPI soundwire: depend on ACPI || OF Pierre-Louis Bossart (1): soundwire: slave: fix scanf format drivers/soundwire/Kconfig | 1 + drivers/soundwire/intel.c | 4 ++-- drivers/soundwire/slave.c | 3 ++- 3 files changed, 5 insertions(+), 3 deletions(-) Thanks -- ~Vinod

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[alsa-devel] Question about FE & BE DPCM
by Daniel Baluta 01 Nov '19

01 Nov '19

Hi Morimoto-san, I'm really confused on what exactly is Front End and Back End. Is FE a DAI or a DAI link or part of a DAI link? In simple-card.c there is simple_dai_link_of_dpcm function: struct snd_soc_dai_link_component *codecs = dai_link->codecs; » /* BE is dummy */ » » codecs->of_node»» = NULL; » » codecs->dai_name» = "snd-soc-dummy-dai"; » » codecs->name» » = "snd-soc-dummy"; » » /* FE settings */ » » dai_link->dynamic» » = 1; » » dai_link->dpcm_merged_format» = 1; So, from here it looks like BE/FE are components being part of the same link. Another thing is: asoc_simple_canonicalize_platform function where it does say: » * DPCM BE can be no platform. » * Alloced memory will be waste, but not leak. » */ » if (!dai_link->platforms->of_node) » » dai_link->num_platforms = 0; But I do want to create a BE with a platform. So, should I try to revert the following patch: commit 794fcee8da3c0c8a01b08ecad1c181cb0a622868 Author: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Wed Jul 10 17:01:12 2019 +0900 ASoC: simple-card-utils: care no Platform for DPCM commit 34614739988ad ("ASoC: soc-core: support dai_link with platforms_num != 1") supports multi Platform, and commit 9f3eb91753451 ("ASoC: simple-card-utils: consider CPU-Platform possibility") removed no Platform from simple-card. Multi Platform is now checking both Platform name/of_node are NULL case. But in normal case, DPCM be doesn't have Platform. Basically, what I want to do is: after the DAI link is created to call: snd_soc_fixup_dai_links_platform_name function to set a proper name for BE platform. The problem is that above mentioned patch sets dai_link->num_platforms = 0; so even the name is set the platform component is not probed. thanks, Daniel.

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