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

[alsa-devel] Applied "ASoC: ux500: Remove redundant variable "status"" to the asoc tree
by Mark Brown 04 Nov '19

04 Nov '19

The patch ASoC: ux500: Remove redundant variable "status" has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-5.5 All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From e0859710516c98b189879966b48ea1c77e0cd979 Mon Sep 17 00:00:00 2001 From: zhong jiang <zhongjiang(a)huawei.com> Date: Thu, 31 Oct 2019 21:34:15 +0800 Subject: [PATCH] ASoC: ux500: Remove redundant variable "status" local variable "status" is not used. hence it is safe to remove and just return 0. Signed-off-by: zhong jiang <zhongjiang(a)huawei.com> Link: https://lore.kernel.org/r/1572528855-25990-1-git-send-email-zhongjiang@huaw… Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/ux500/ux500_msp_i2s.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/ux500/ux500_msp_i2s.c b/sound/soc/ux500/ux500_msp_i2s.c index a90e0d7f0b73..394d8b2a4a16 100644 --- a/sound/soc/ux500/ux500_msp_i2s.c +++ b/sound/soc/ux500/ux500_msp_i2s.c @@ -533,7 +533,6 @@ static void disable_msp_tx(struct ux500_msp *msp) static int disable_msp(struct ux500_msp *msp, unsigned int dir) { u32 reg_val_GCR; - int status = 0; unsigned int disable_tx, disable_rx; reg_val_GCR = readl(msp->registers + MSP_GCR); @@ -566,7 +565,7 @@ static int disable_msp(struct ux500_msp *msp, unsigned int dir) else if (disable_rx) disable_msp_rx(msp); - return status; + return 0; } int ux500_msp_i2s_trigger(struct ux500_msp *msp, int cmd, int direction) -- 2.20.1

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[alsa-devel] [PATCH] ASoC: hdac_hda: fix race in device removal
by Pierre-Louis Bossart 04 Nov '19

04 Nov '19

From: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> When ASoC card instance is removed containing a HDA codec, hdac_hda_codec_remove() may run in parallel with codec resume. This will cause problems if the HDA link is freed with snd_hdac_ext_bus_link_put() while the codec is still in middle of its resume process. To fix this, change the order such that pm_runtime_disable() is called before the link is freed. This will ensure any pending runtime PM action is completed before proceeding to free the link. This issue can be easily hit with e.g. SOF driver by loading and unloading the drivers. Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/codecs/hdac_hda.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/hdac_hda.c b/sound/soc/codecs/hdac_hda.c index 298761a26180..6803d39e09a5 100644 --- a/sound/soc/codecs/hdac_hda.c +++ b/sound/soc/codecs/hdac_hda.c @@ -500,8 +500,8 @@ static void hdac_hda_codec_remove(struct snd_soc_component *component) return; } - snd_hdac_ext_bus_link_put(hdev->bus, hlink); pm_runtime_disable(&hdev->dev); + snd_hdac_ext_bus_link_put(hdev->bus, hlink); } static const struct snd_soc_dapm_route hdac_hda_dapm_routes[] = { -- 2.20.1

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[alsa-devel] [PATCH v5 0/2] ASoC: Intel: boards: Add CML M/C for RT5682 and RT1011
by Pierre-Louis Bossart 04 Nov '19

04 Nov '19

The series adds CML m/c driver with RT5682 on SSP0 and RT1011 on SSP1. The RT1011 is smart AMP which supports 2 Whoofers and 2 tweeter support. RT5682 is headset codec supporting playback and record. The m/c supports HDMI playback and DMIC record. Changes since v4 (from Kai Vehmanen) added prefix for series, reformatted changes fixed bad filename in Makefile Support for hdmi common codec Changes since v3 Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> repost with more recipients and edits to commit messages Changes since v2: re ordered match table to ensure non i2c device matches are at the end Changes since V1: Addressed comments on ASoC: Intel: boards: Add CML m/c using RT1011 and RT5682 Naveen Manohar (2): ASoC: Intel: Add acpi match for rt1011 based m/c driver ASoC: Intel: boards: Add CML m/c using RT1011 and RT5682 sound/soc/intel/boards/Kconfig | 14 + sound/soc/intel/boards/Makefile | 2 + sound/soc/intel/boards/cml_rt1011_rt5682.c | 487 ++++++++++++++++++ .../intel/common/soc-acpi-intel-cnl-match.c | 7 + 4 files changed, 510 insertions(+) create mode 100644 sound/soc/intel/boards/cml_rt1011_rt5682.c -- 2.20.1

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[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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