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

[PATCH v2] Asoc: qcom: lpass-cpu: Fix clock disable failure
by Srinivasa Rao Mandadapu 29 Oct '20

29 Oct '20

From: V Sujith Kumar Reddy <vsujithk(a)codeaurora.org> Disable MI2S bit clock from PAUSE/STOP/SUSPEND usecase instead of shutdown time. Acheive this by invoking clk_disable API from cpu daiops trigger instead of cpu daiops shutdown. Change non-atomic API "clk_prepare_enable" to atomic API "clk_enable" in trigger, as trigger is being called from atomic context. Fixes: commit 7e6799d8f87d ("ASoC: qcom: lpass-cpu: Enable MI2S BCLK and LRCLK together") Signed-off-by: V Sujith Kumar Reddy <vsujithk(a)codeaurora.org> Signed-off-by: Srinivasa Rao Mandadapu <srivasam(a)codeaurora.org> --- sound/soc/qcom/lpass-cpu.c | 14 ++++++++++---- 1 file changed, 10 insertions(+), 4 deletions(-) diff --git a/sound/soc/qcom/lpass-cpu.c b/sound/soc/qcom/lpass-cpu.c index 78de888..6001049 100644 --- a/sound/soc/qcom/lpass-cpu.c +++ b/sound/soc/qcom/lpass-cpu.c @@ -80,6 +80,12 @@ static int lpass_cpu_daiops_startup(struct snd_pcm_substream *substream, dev_err(dai->dev, "error in enabling mi2s osr clk: %d\n", ret); return ret; } + ret = clk_prepare(drvdata->mi2s_bit_clk[dai->driver->id]); + if (ret) { + dev_err(dai->dev, "error in enabling mi2s bit clk: %d\n", ret); + clk_disable_unprepare(drvdata->mi2s_osr_clk[dai->driver->id]); + return ret; + } return 0; } @@ -88,9 +94,8 @@ static void lpass_cpu_daiops_shutdown(struct snd_pcm_substream *substream, { struct lpass_data *drvdata = snd_soc_dai_get_drvdata(dai); - clk_disable_unprepare(drvdata->mi2s_bit_clk[dai->driver->id]); - clk_disable_unprepare(drvdata->mi2s_osr_clk[dai->driver->id]); + clk_unprepare(drvdata->mi2s_bit_clk[dai->driver->id]); } static int lpass_cpu_daiops_hw_params(struct snd_pcm_substream *substream, @@ -303,10 +308,10 @@ static int lpass_cpu_daiops_trigger(struct snd_pcm_substream *substream, dev_err(dai->dev, "error writing to i2sctl reg: %d\n", ret); - ret = clk_prepare_enable(drvdata->mi2s_bit_clk[id]); + ret = clk_enable(drvdata->mi2s_bit_clk[id]); if (ret) { dev_err(dai->dev, "error in enabling mi2s bit clk: %d\n", ret); - clk_disable_unprepare(drvdata->mi2s_osr_clk[id]); + clk_disable(drvdata->mi2s_osr_clk[id]); return ret; } @@ -324,6 +329,7 @@ static int lpass_cpu_daiops_trigger(struct snd_pcm_substream *substream, if (ret) dev_err(dai->dev, "error writing to i2sctl reg: %d\n", ret); + clk_disable(drvdata->mi2s_bit_clk[dai->driver->id]); break; } -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

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[PATCH][next] ASoC: qcom: fix unsigned int bitwidth compared to less than zero
by Colin King 29 Oct '20

29 Oct '20

From: Colin Ian King <colin.king(a)canonical.com> The check for an error return from the call to snd_pcm_format_width is never true as the unsigned int bitwidth can never be less than zero. Fix this by making bitwidth an int. Addresses-Coverity: ("Unsigned compared against 0") Fixes: 7cb37b7bd0d3 ("ASoC: qcom: Add support for lpass hdmi driver") Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/qcom/lpass-hdmi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/qcom/lpass-hdmi.c b/sound/soc/qcom/lpass-hdmi.c index 172952d3a5d6..abfb8737a89f 100644 --- a/sound/soc/qcom/lpass-hdmi.c +++ b/sound/soc/qcom/lpass-hdmi.c @@ -24,7 +24,7 @@ static int lpass_hdmi_daiops_hw_params(struct snd_pcm_substream *substream, unsigned int rate = params_rate(params); unsigned int channels = params_channels(params); unsigned int ret; - unsigned int bitwidth; + int bitwidth; unsigned int word_length; unsigned int ch_sts_buf0; unsigned int ch_sts_buf1; -- 2.27.0

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[PATCH v1 0/2] ASoC: qcom: add support for QRB5165 RB5 machine
by Srinivas Kandagatla 29 Oct '20

29 Oct '20

This patchset adds support to Qualcomm Robotics RB5 Development Kit based on QRB5165 Robotics SoC. This board has 2 WSA881X smart speakers with onboard DMIC connected to internal LPASS codec via WSA and VA macros respectively. Srinivas Kandagatla (2): ASoC: qcom: dt-bindings: Add SM8250 sound card bindings ASoC: qcom: sm8250: add sound card qrb5165-rb5 support .../bindings/sound/qcom,sm8250.yaml | 161 +++++++++++++ sound/soc/qcom/Kconfig | 11 + sound/soc/qcom/Makefile | 2 + sound/soc/qcom/sm8250.c | 228 ++++++++++++++++++ 4 files changed, 402 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/qcom,sm8250.yaml create mode 100644 sound/soc/qcom/sm8250.c -- 2.21.0

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ucm: return error if boot_list is empty
by GitHub pull_request - opened 29 Oct '20

29 Oct '20

alsa-project/alsa-lib pull request #94 was opened from jason77-wang: If bootlist is empty, that means there is no BootSequence defined in the ucm for this sound card, let set_boot_user() return error, then alsactl could have chance to use non-ucm ways to initialize the sound card. Signed-off-by: Hui Wang <hui.wang(a)canonical.com> Request URL : https://github.com/alsa-project/alsa-lib/pull/94 Patch URL : https://github.com/alsa-project/alsa-lib/pull/94.patch Repository URL: https://github.com/alsa-project/alsa-lib

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Fix initsetting
by GitHub pull_request - opened 29 Oct '20

29 Oct '20

alsa-project/alsa-ucm-conf pull request #57 was opened from jason77-wang: This issue was found when we prepared ubuntu 20.10. When users install the OS to a machine with HDA codec and use sof driver, the audio output is muted after installation. And even we set the master playback switch to on, the output volume from speaker is too low. This issue could be easily reproduced by: sudo rm /var/lib/asla/* rm ~/.config/pulse/* (optional) sudo /usr/sbin/alsactl -E HOME=/run/alsa -E XDG_RUNTIME_DIR=/run/alsa/runtime restore Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/57 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/57.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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[PATCH v4 0/3] ASoC: dt-bindings: audio-graph-card: switch to yaml base Documentation
by Kuninori Morimoto 29 Oct '20

29 Oct '20

Hi Rob, Mark Cc: Sameer These are v4 of switch to yaml base Documentation for audio-graph-card and renesas,rsnd which uses audio-graph-card. v3 -> v4 - fixuped yamllint warnings/errors v2 -> v3 - fixuped Rob's pointed points v1 -> v2 - upgrade dt-schema, and fixup errors for [2/2] patch Kuninori Morimoto (3): dt-bindings: ASoC: audio-graph-card: switch to yaml base Documentation dt-bindings: ASoC: renesas,rsnd: switch to yaml base Documentation dt-bindings: ASoC: renesas,rsnd: Add r8a77961 support .../bindings/sound/audio-graph-card.txt | 337 ------------ .../bindings/sound/audio-graph-card.yaml | 165 ++++++ .../bindings/sound/renesas,rsnd.txt | 520 ------------------ .../bindings/sound/renesas,rsnd.yaml | 457 +++++++++++++++ .../devicetree/bindings/vendor-prefixes.yaml | 2 +- 5 files changed, 623 insertions(+), 858 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/audio-graph-card.txt create mode 100644 Documentation/devicetree/bindings/sound/audio-graph-card.yaml create mode 100644 Documentation/devicetree/bindings/sound/renesas,rsnd.yaml -- 2.25.1

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[PATCH v3 0/3] ASoC: dt-bindings: audio-graph-card: switch to yaml base Documentation
by Kuninori Morimoto 29 Oct '20

29 Oct '20

Hi Rob, Mark Cc: Sameer These are v3 of switch to yaml base Documentation for audio-graph-card and renesas,rsnd which uses audio-graph-card. v2 -> v3 - fixuped Rob's pointed points v1 -> v2 - upgrade dt-schema, and fixup errors for [2/2] patch Kuninori Morimoto (3): ASoC: dt-bindings: audio-graph-card: switch to yaml base Documentation ASoC: dt-bindings: renesas,rsnd: switch to yaml base Documentation dt-bindings: ASoC: renesas,rsnd: Add r8a77961 support .../bindings/sound/audio-graph-card.txt | 337 ------------ .../bindings/sound/audio-graph-card.yaml | 165 ++++++ .../bindings/sound/renesas,rsnd.txt | 520 ------------------ .../bindings/sound/renesas,rsnd.yaml | 457 +++++++++++++++ .../devicetree/bindings/vendor-prefixes.yaml | 2 +- 5 files changed, 623 insertions(+), 858 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/audio-graph-card.txt create mode 100644 Documentation/devicetree/bindings/sound/audio-graph-card.yaml create mode 100644 Documentation/devicetree/bindings/sound/renesas,rsnd.yaml -- 2.25.1

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[PATCH] ASoC: qcom: sm8250: Fix array out of bounds access
by Srinivas Kandagatla 28 Oct '20

28 Oct '20

Static analysis Coverity had detected a potential array out-of-bounds write issue due to the fact that MAX AFE port Id was set to 16 instead of using AFE_PORT_MAX macro. Fix this by properly using AFE_PORT_MAX macro. Fixes: aa2e2785545a ("ASoC: qcom: sm8250: add sound card qrb5165-rb5 support") Reported-by: Colin Ian King <colin.king(a)canonical.com> Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- sound/soc/qcom/sm8250.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/soc/qcom/sm8250.c b/sound/soc/qcom/sm8250.c index 7d43de6d909f..52c40512102f 100644 --- a/sound/soc/qcom/sm8250.c +++ b/sound/soc/qcom/sm8250.c @@ -13,12 +13,11 @@ #define DRIVER_NAME "sm8250" #define MI2S_BCLK_RATE 1536000 -#define MAX_SDW_STREAMS 16 struct sm8250_snd_data { - bool stream_prepared[MAX_SDW_STREAMS]; + bool stream_prepared[AFE_PORT_MAX]; struct snd_soc_card *card; - struct sdw_stream_runtime *sruntime[MAX_SDW_STREAMS]; + struct sdw_stream_runtime *sruntime[AFE_PORT_MAX]; }; static int sm8250_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, -- 2.21.0

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[PATCH 1/2] ASoC: codecs: wcd934x: Set digital gain range correctly
by Srinivas Kandagatla 28 Oct '20

28 Oct '20

digital gain range is -84dB min to 40dB max, however this was not correctly specified in the range. Fix this by with correct range! Fixes: 1cde8b822332 ("ASoC: wcd934x: add basic controls") Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- sound/soc/codecs/wcd934x.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wcd934x.c b/sound/soc/codecs/wcd934x.c index 35697b072367..40f682f5dab8 100644 --- a/sound/soc/codecs/wcd934x.c +++ b/sound/soc/codecs/wcd934x.c @@ -551,7 +551,7 @@ struct wcd_iir_filter_ctl { struct soc_bytes_ext bytes_ext; }; -static const DECLARE_TLV_DB_SCALE(digital_gain, 0, 1, 0); +static const DECLARE_TLV_DB_SCALE(digital_gain, -8400, 100, -8400); static const DECLARE_TLV_DB_SCALE(line_gain, 0, 7, 1); static const DECLARE_TLV_DB_SCALE(analog_gain, 0, 25, 1); static const DECLARE_TLV_DB_SCALE(ear_pa_gain, 0, 150, 0); -- 2.21.0

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[PATCH v3 0/3] HDA controller PM and codec PM cleanups
by Kai-Heng Feng 28 Oct '20

28 Oct '20

While working on the issue "ALSA: hda: fix jack detection with Realtek codecs when in D3", I've found using pm_runtime_force_{suspend,resume}() have surprising behavior, specifically, pm_runtime_need_not_resume() uses pm_runtime_need_not_resume() to avoid calling resume callback, so jackpoll was used to really power up the codec. We can use direct-complete to do the keep the codec suspended throughout the system PM flow, namely, keep the codec suspended all the way, unless the codec needs to be woken up after resume. For HDA controller, PCI core may enable direct-complete for it if conditions are met. So make runtime and system PM distinctive to always apply correct wake up setting. At least point, hopefully all runtime PM issues are solved, let's enable runtime PM by default again. v3: - Drop "ALSA: hda: Stop mangling PCI IRQ" Kai-Heng Feng (3): ALSA: hda: Refactor codec PM to use direct-complete optimization ALSA: hda: Separate runtime and system suspend ALSA: hda: Reinstate runtime_allow() for all hda controllers sound/pci/hda/hda_codec.c | 45 +++++++++++++++--------- sound/pci/hda/hda_controller.h | 3 +- sound/pci/hda/hda_intel.c | 63 +++++++++++++++++++--------------- 3 files changed, 66 insertions(+), 45 deletions(-) -- 2.17.1

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