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

[v2 1/3] ASoC: rt5682: Add a property for DMIC clock rate setting
by Oder Chiou 19 Mar '20

19 Mar '20

The patch adds a property for DMIC clock rate setting and changes the default clock rate to the common optimize DMIC clock rate. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- include/sound/rt5682.h | 1 + sound/soc/codecs/rt5682.c | 9 +++++++-- 2 files changed, 8 insertions(+), 2 deletions(-) diff --git a/include/sound/rt5682.h b/include/sound/rt5682.h index 6bf0e3581056..96b268ac96bd 100644 --- a/include/sound/rt5682.h +++ b/include/sound/rt5682.h @@ -38,6 +38,7 @@ struct rt5682_platform_data { enum rt5682_dmic1_clk_pin dmic1_clk_pin; enum rt5682_jd_src jd_src; unsigned int btndet_delay; + unsigned int dmic_clk_rate; const char *dai_clk_names[RT5682_DAI_NUM_CLKS]; }; diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index 2ee01e782fe0..c9e6f287424b 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -1231,10 +1231,13 @@ static int set_dmic_clk(struct snd_soc_dapm_widget *w, struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component); - int idx = -EINVAL; + int idx = -EINVAL, dmic_clk_rate = 3072000; static const int div[] = {2, 4, 6, 8, 12, 16, 24, 32, 48, 64, 96, 128}; - idx = rt5682_div_sel(rt5682, 1500000, div, ARRAY_SIZE(div)); + if (rt5682->pdata.dmic_clk_rate) + dmic_clk_rate = rt5682->pdata.dmic_clk_rate; + + idx = rt5682_div_sel(rt5682, dmic_clk_rate, div, ARRAY_SIZE(div)); snd_soc_component_update_bits(component, RT5682_DMIC_CTRL_1, RT5682_DMIC_CLK_MASK, idx << RT5682_DMIC_CLK_SFT); @@ -3231,6 +3234,8 @@ static int rt5682_parse_dt(struct rt5682_priv *rt5682, struct device *dev) &rt5682->pdata.jd_src); device_property_read_u32(dev, "realtek,btndet-delay", &rt5682->pdata.btndet_delay); + device_property_read_u32(dev, "realtek,dmic-clk-rate", + &rt5682->pdata.dmic_clk_rate); rt5682->pdata.ldo1_en = of_get_named_gpio(dev->of_node, "realtek,ldo1-en-gpios", 0); -- 2.25.1

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[PATCH v2 00/17] SoundWire: cadence: add clock stop and fix programming sequences
by Pierre-Louis Bossart 19 Mar '20

19 Mar '20

To make progress with SoundWire support, this patchset provides the missing support for clock stop modes, and revisits all Cadence Master register settings. The current code is for some reason not aligned with internal documentation and hardware recommended flows, specifically for multi-link operation. Changes since v1: Removed log in is_clock_stop(), use the helper in the main clock_stop() and change return 1->return 0. Fixed squash issue in patch5 to remove irrelevant udelay() change Added Patch17 to clear FIFOs and avoid pop noise Pierre-Louis Bossart (12): soundwire: cadence: s/update_config/config_update soundwire: cadence: handle error cases with CONFIG_UPDATE soundwire: cadence: mask Slave interrupt before stopping clock soundwire: cadence: merge routines to clear/set bits soundwire: cadence: move clock/SSP related inits to dedicated function soundwire: cadence: make SSP interval programmable soundwire: cadence: reorder MCP_CONFIG settings soundwire: cadence: enable NORMAL operation in cdns_init() soundwire: cadence: remove PREQ_DELAY assignment soundwire: cadence: remove automatic command retries soundwire: cadence: commit changes in the exit_reset() sequence soundwire: cadence: multi-link support Rander Wang (4): soundwire: cadence: simplifiy cdns_init() soundwire: cadence: add interface to check clock status soundwire: cadence: add clock_stop/restart routines soundwire: cadence: fix a io timeout issue in S3 test randerwang (1): soundwire: cadence: clear FIFO to avoid pop noise issue on playback start drivers/soundwire/cadence_master.c | 282 ++++++++++++++++++++++++----- drivers/soundwire/cadence_master.h | 9 +- drivers/soundwire/intel.c | 2 +- 3 files changed, 249 insertions(+), 44 deletions(-) base-commit: 0b43fef979b4664d51a09dc7e0c430ebb2d18267 -- 2.20.1

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Corrections for pcm_a52 to work with pulseaudio
by GitHub pull_request - edited 19 Mar '20

19 Mar '20

alsa-project/alsa-plugins pull request #8 was edited from amotzkau: Hi there, First this is my first contact with ALSA programming, so please forgive me for any obvious mistakes I made. On my computer Pulseaudio didn't work with the pcm_a52 plugin (Digital Surround 5.1 profile in Pulseaudio), everything was played with 30x speed (approximately). So I looked into the cause of this. Pulseaudio uses the zero-copy mmap interface of ALSA and expects that snd_pcm_mmap_commit() will accept all frames that snd_pcm_mmap_begin() allowed to send. If snd_pcm_mmap_commit() accepts less frames or none (-EAGAIN) then Pulseaudio didn't remember that or even saved those frames. They were just discarded. I thought that it would be a sound contract to guarantee that snd_pcm_mmap_commit() will always take those frames. Therefore I looked into pcm_a52, why it doesn't promise that, and changed the a52_pointer() function to offer such a guarantee. These are my patches to make it work. Best regards, Alex Request URL : https://github.com/alsa-project/alsa-plugins/pull/8 Patch URL : https://github.com/alsa-project/alsa-plugins/pull/8.patch Repository URL: https://github.com/alsa-project/alsa-plugins

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[PATCH v2 0/2] Add generic FSL CPU DAI driver
by Daniel Baluta 19 Mar '20

19 Mar '20

From: Daniel Baluta <daniel.baluta(a)nxp.com> On platforms with a DSP we need to split the resource handling between Application Processor (AP) and DSP. On platforms where the DSP doesn't have an easy access to resources, the AP will take care of configuring them. Resources handled by this generic driver are: clocks, power domains, pinctrl. Changes since v1: - added dt-bindings patch - add missing signed-off-by - do not hardcode the name of DAI driver but derive it from newly introduced dai_index property. Daniel Baluta (2): dt-bindings: sound: Add FSL CPU DAI bindings ASoC: fsl: Add generic CPU DAI driver .../devicetree/bindings/sound/fsl,dai.yaml | 97 ++++++ sound/soc/fsl/Kconfig | 8 + sound/soc/fsl/Makefile | 2 + sound/soc/fsl/fsl_dai.c | 288 ++++++++++++++++++ 4 files changed, 395 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/fsl,dai.yaml create mode 100644 sound/soc/fsl/fsl_dai.c -- 2.17.1

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Corrections for pcm_a52 to work with pulseaudio
by GitHub pull_request - opened 18 Mar '20

18 Mar '20

alsa-project/alsa-plugins pull request #8 was opened from amotzkau: Hi there, First this is my first contact with ALSA programming, so please excuse any obvious mistakes. On my computer Pulseaudio didn't work with the pcm_a52 plugin (Digital Surround 5.1 profile in Pulseaudio), everything was played with 30x speed (approximately). So I looked into the cause of this. Pulseaudio uses the zero-copy mmap interface of ALSA and expects that snd_pcm_mmap_commit() will except all frames that snd_pcm_mmap_begin() allowed to send. If snd_pcm_mmap_commit() accepts less frames or none (-EAGAIN) then Pulseaudio didn't remember that or even saved those frames. They were just discarded. I thought that it would be a sound contract to guarantee that snd_pcm_mmap_commit() will always take those frames. Therefore I looked into pcm_a52, why it doesn't promise that, and changed the a52_pointer() function to offer such a guarantee. These are my patches to make it work. Best regards, Alex Request URL : https://github.com/alsa-project/alsa-plugins/pull/8 Patch URL : https://github.com/alsa-project/alsa-plugins/pull/8.patch Repository URL: https://github.com/alsa-project/alsa-plugins

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[PATCH 0/3] ASoC: stm32: manage rebind issue
by Olivier Moysan 18 Mar '20

18 Mar '20

This patchset corrects a rebind issue on STM32 SPDIFRX and I2S drivers. The same correction has already been applied for SAI driver: 0d6defc7e0e4 ("ASoC: stm32: sai: manage rebind issue") The commit e894efef9ac7 ("ASoC: core: add support to card rebind") allows to rebind the sound card after a rebind of one of its component. With this commit, the sound card is actually rebound, but may be no more functional. The following problems have been seen on STM32 drivers. 1) DMA channel is not requested: With the sound card rebind the simplified call sequence is: probe snd_soc_register_component snd_soc_try_rebind_card snd_soc_instantiate_card devm_snd_dmaengine_pcm_register The problem occurs because the pcm must be registered, before snd_soc_instantiate_card() is called. Modify the driver, to change the call sequence as follows: probe devm_snd_dmaengine_pcm_register snd_soc_register_component snd_soc_try_rebind_card 2) DMA channel is not released: dma_release_channel() is not called when devm_dmaengine_pcm_release() is executed. This occurs because SND_DMAENGINE_PCM_DRV_NAME component, has already been released through devm_component_release(). devm_dmaengine_pcm_release() should be called before devm_component_release() to avoid this problem. Call snd_dmaengine_pcm_unregister() and snd_soc_unregister_component() explicitly from the driver, to have the right sequence. Olivier Moysan (3): ASoC: stm32: spdifrx: fix regmap status check ASoC: stm32: spdifrx: manage rebind issue ASoC: stm32: i2s: manage rebind issue sound/soc/stm/stm32_i2s.c | 40 ++++++++++++++++------ sound/soc/stm/stm32_spdifrx.c | 64 +++++++++++++++++++---------------- 2 files changed, 63 insertions(+), 41 deletions(-) -- 2.17.1

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Applied "ASoC: codecs: wsa881x: remove soundwire stream handling" to the asoc tree
by Mark Brown 18 Mar '20

18 Mar '20

The patch ASoC: codecs: wsa881x: remove soundwire stream handling has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git 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 16252a8f3af77f69c2193fdc7b2f595b30845a44 Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Date: Tue, 17 Mar 2020 15:12:33 +0000 Subject: [PATCH] ASoC: codecs: wsa881x: remove soundwire stream handling There could be multiple instances of this codec on any platform, so handling stream directly in this codec driver can lead to multiple calls to prepare/enable/disable on the same SoundWire stream. Move this stream handling to machine driver to fix this issue. Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Link: https://lore.kernel.org/r/20200317151233.8763-3-srinivas.kandagatla@linaro.… Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/wsa881x.c | 44 +------------------------------------- 1 file changed, 1 insertion(+), 43 deletions(-) diff --git a/sound/soc/codecs/wsa881x.c b/sound/soc/codecs/wsa881x.c index b59f1d0e7f84..3d525297eac9 100644 --- a/sound/soc/codecs/wsa881x.c +++ b/sound/soc/codecs/wsa881x.c @@ -676,7 +676,6 @@ struct wsa881x_priv { int active_ports; bool port_prepared[WSA881X_MAX_SWR_PORTS]; bool port_enable[WSA881X_MAX_SWR_PORTS]; - bool stream_prepared; }; static void wsa881x_init(struct wsa881x_priv *wsa881x) @@ -954,41 +953,6 @@ static const struct snd_soc_dapm_widget wsa881x_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("SPKR"), }; -static int wsa881x_prepare(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct wsa881x_priv *wsa881x = dev_get_drvdata(dai->dev); - int ret; - - if (wsa881x->stream_prepared) { - sdw_disable_stream(wsa881x->sruntime); - sdw_deprepare_stream(wsa881x->sruntime); - wsa881x->stream_prepared = false; - } - - - ret = sdw_prepare_stream(wsa881x->sruntime); - if (ret) - return ret; - - /** - * NOTE: there is a strict hw requirement about the ordering of port - * enables and actual PA enable. PA enable should only happen after - * soundwire ports are enabled if not DC on the line is accumulated - * resulting in Click/Pop Noise - * PA enable/mute are handled as part of DAPM and digital mute. - */ - - ret = sdw_enable_stream(wsa881x->sruntime); - if (ret) { - sdw_deprepare_stream(wsa881x->sruntime); - return ret; - } - wsa881x->stream_prepared = true; - - return ret; -} - static int wsa881x_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) @@ -1016,12 +980,7 @@ static int wsa881x_hw_free(struct snd_pcm_substream *substream, { struct wsa881x_priv *wsa881x = dev_get_drvdata(dai->dev); - if (wsa881x->stream_prepared) { - sdw_disable_stream(wsa881x->sruntime); - sdw_deprepare_stream(wsa881x->sruntime); - sdw_stream_remove_slave(wsa881x->slave, wsa881x->sruntime); - wsa881x->stream_prepared = false; - } + sdw_stream_remove_slave(wsa881x->slave, wsa881x->sruntime); return 0; } @@ -1052,7 +1011,6 @@ static int wsa881x_digital_mute(struct snd_soc_dai *dai, int mute, int stream) static struct snd_soc_dai_ops wsa881x_dai_ops = { .hw_params = wsa881x_hw_params, - .prepare = wsa881x_prepare, .hw_free = wsa881x_hw_free, .mute_stream = wsa881x_digital_mute, .set_sdw_stream = wsa881x_set_sdw_stream, -- 2.20.1

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[PATCH v2 0/2] ASoC: sdm845: fix soundwire stream handling
by Srinivas Kandagatla 18 Mar '20

18 Mar '20

Recent addition of SoundWire stream state-machine checks in linux-next have shown an existing issue with handling soundwire streams in codec drivers. In general soundwire stream prepare/enable/disable can be called from either codec/machine/controller driver. However calling it in codec driver means that if multiple instances(Left/Right speakers) of the same codec is connected to the same stream then it will endup calling stream prepare/enable/disable more than once. This will mess up the stream state-machine checks in the soundwire core. Moving this stream handling to machine driver would fix this issue and also allow board/platform specfic power sequencing. Changes since v1: - removed false error check while setting sruntime. Srinivas Kandagatla (2): ASoC: qcom: sdm845: handle soundwire stream ASoC: codecs: wsa881x: remove soundwire stream handling sound/soc/codecs/wsa881x.c | 44 +------------------------ sound/soc/qcom/Kconfig | 2 +- sound/soc/qcom/sdm845.c | 67 ++++++++++++++++++++++++++++++++++++++ 3 files changed, 69 insertions(+), 44 deletions(-) -- 2.21.0

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[PATCH 2/2] ASoC: rt5682: remove noisy debug messages
by shumingf＠realtek.com 18 Mar '20

18 Mar '20

From: Shuming Fan <shumingf(a)realtek.com> Some debug messages are too noisy. This patch removes it. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/rt5682.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index 023bc672274b..2ee01e782fe0 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -1197,11 +1197,11 @@ static int rt5682_div_sel(struct rt5682_priv *rt5682, } for (i = 0; i < size - 1; i++) { - pr_info("div[%d]=%d\n", i, div[i]); + dev_dbg(rt5682->component->dev, "div[%d]=%d\n", i, div[i]); if (target * div[i] == rt5682->sysclk) return i; if (target * div[i + 1] > rt5682->sysclk) { - pr_err("can't find div for sysclk %d\n", + dev_dbg(rt5682->component->dev, "can't find div for sysclk %d\n", rt5682->sysclk); return i; } -- 2.25.1

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[PATCH 1/2] ASoC: rt5682: fix the random recording noise of headset
by shumingf＠realtek.com 18 Mar '20

18 Mar '20

From: Shuming Fan <shumingf(a)realtek.com> The cycle time of FIFO clock should increase 2 times to avoid the random recording noise issue. This setting could apply to all known situations in i2s mode. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/rt5682.c | 2 ++ sound/soc/codecs/rt5682.h | 2 ++ 2 files changed, 4 insertions(+) diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index 7ca02a5e52e9..023bc672274b 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -3643,6 +3643,8 @@ static int rt5682_i2c_probe(struct i2c_client *i2c, RT5682_CP_CLK_HP_MASK, RT5682_CP_CLK_HP_300KHZ); regmap_update_bits(rt5682->regmap, RT5682_HP_CHARGE_PUMP_1, RT5682_PM_HP_MASK, RT5682_PM_HP_HV); + regmap_update_bits(rt5682->regmap, RT5682_DMIC_CTRL_1, + RT5682_FIFO_CLK_DIV_MASK, RT5682_FIFO_CLK_DIV_2); INIT_DELAYED_WORK(&rt5682->jack_detect_work, rt5682_jack_detect_handler); diff --git a/sound/soc/codecs/rt5682.h b/sound/soc/codecs/rt5682.h index 43de6e802309..0baeece84ec4 100644 --- a/sound/soc/codecs/rt5682.h +++ b/sound/soc/codecs/rt5682.h @@ -657,6 +657,8 @@ #define RT5682_DMIC_1_EN_SFT 15 #define RT5682_DMIC_1_DIS (0x0 << 15) #define RT5682_DMIC_1_EN (0x1 << 15) +#define RT5682_FIFO_CLK_DIV_MASK (0x7 << 12) +#define RT5682_FIFO_CLK_DIV_2 (0x1 << 12) #define RT5682_DMIC_1_DP_MASK (0x3 << 4) #define RT5682_DMIC_1_DP_SFT 4 #define RT5682_DMIC_1_DP_GPIO2 (0x0 << 4) -- 2.25.1

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