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

RE: Enable Headset Mic for Dell desktop
by Kailang 05 Mar '20

05 Mar '20

Sorry! Forgot attach file. > -----Original Message----- > From: Kailang > Sent: Thursday, February 20, 2020 4:30 PM > To: Takashi Iwai (tiwai(a)suse.de) <tiwai(a)suse.de> > Cc: (alsa-devel(a)alsa-project.org) <alsa-devel(a)alsa-project.org> > Subject: Enable Headset Mic for Dell desktop > > Hi Takashi, > > Attach patch was supported headset Mic. > > BR, > Kailang

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[PATCH v2 0/8] ASoC: soc-pcm cleanup step5
by Kuninori Morimoto 05 Mar '20

05 Mar '20

Hi Mark These are v2 of soc-pcm cleanup step5. These are based on latest for-5.7 branch. Mainly v2 fixed the points reviewed in v1. [7/8] patch might exchange the behavior Kuninori Morimoto (8): 1) ASoC: soc-pcm: use defined stream 2) ASoC: soc-pcm: remove duplicate be check from dpcm_add_paths() 3) ASoC: soc-pcm: move dpcm_fe_dai_close() 4) ASoC: soc-pcm: add dpcm_fe_dai_cleanup() 5) ASoC: soc-pcm: use snd_soc_dai_get_pcm_stream() at dpcm_set_fe_runtime() 6) ASoC: soc-pcm: check DAI's activity more simply 7) ASoC: soc-pcm: Do Digital Mute for both CPU/Codec in same timing. 8) ASoC: soc-pcm: tidyup dulicate handing at dpcm_fe_dai_startup() sound/soc/soc-pcm.c | 114 +++++++++++++++++++------------------------- 1 file changed, 50 insertions(+), 64 deletions(-) -- 2.17.1

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[PATCH] ASoC: Fix SND_SOC_ALL_CODECS imply ac97 fallout
by Geert Uytterhoeven 05 Mar '20

05 Mar '20

On i386 randconfig: sound/soc/codecs/wm9705.o: In function `wm9705_soc_resume': wm9705.c:(.text+0x128): undefined reference to `snd_ac97_reset' sound/soc/codecs/wm9712.o: In function `wm9712_soc_resume': wm9712.c:(.text+0x2d1): undefined reference to `snd_ac97_reset' sound/soc/codecs/wm9713.o: In function `wm9713_soc_resume': wm9713.c:(.text+0x820): undefined reference to `snd_ac97_reset' Fix this by adding the missing dependencies on SND_SOC_AC97_BUS. Reported-by: Randy Dunlap <rdunlap(a)infradead.org> Signed-off-by: Geert Uytterhoeven <geert(a)linux-m68k.org> --- Before commit ea00d95200d02ece ("ASoC: Use imply for SND_SOC_ALL_CODECS"), SND_SOC_ALL_CODECS used: select SND_SOC_WM9705 if (SND_SOC_AC97_BUS || SND_SOC_AC97_BUS_NEW) select SND_SOC_WM9712 if (SND_SOC_AC97_BUS || SND_SOC_AC97_BUS_NEW) select SND_SOC_WM9713 if (SND_SOC_AC97_BUS || SND_SOC_AC97_BUS_NEW) but SND_SOC_AC97_BUS_NEW never existed in upstream. Should there be another dependency> See also "non-existent SND_SOC_AC97_BUS_NEW (was: Re: [PATCH v9] ASoC: pxa: switch to new ac97 bus support)" http://lore.kernel.org/r/CAMuHMdU3uxfBwKd8SkOtZSDV5Ai3CKc3CWRhDy0Cz94T1Hn0i… --- sound/soc/codecs/Kconfig | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 9e9d54e4576ce5ba..a7e89567edbe8b47 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -1610,16 +1610,19 @@ config SND_SOC_WM9090 config SND_SOC_WM9705 tristate + depends on SND_SOC_AC97_BUS select REGMAP_AC97 select AC97_BUS_COMPAT if AC97_BUS_NEW config SND_SOC_WM9712 tristate + depends on SND_SOC_AC97_BUS select REGMAP_AC97 select AC97_BUS_COMPAT if AC97_BUS_NEW config SND_SOC_WM9713 tristate + depends on SND_SOC_AC97_BUS select REGMAP_AC97 select AC97_BUS_COMPAT if AC97_BUS_NEW -- 2.17.1

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[PATCH] ASoC: soc-pcm: Revert "call snd_soc_component_open/close() once"
by Kai Vehmanen 04 Mar '20

04 Mar '20

ASoC component open/close and snd_soc_component_module_get/put are called independently for each component-substream pair, so the logic in the reverted patch was not sufficient and led to PCM playback and module unload errors. Fixes: dd03907bf129 ("ASoC: soc-pcm: call snd_soc_component_open/close() once") Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> --- include/sound/soc-component.h | 7 ++----- sound/soc/soc-component.c | 35 +++++++---------------------------- sound/soc/soc-pcm.c | 19 +++++++++++++------ 3 files changed, 22 insertions(+), 39 deletions(-) diff --git a/include/sound/soc-component.h b/include/sound/soc-component.h index 1866ecc8e94b..154d02fbbfed 100644 --- a/include/sound/soc-component.h +++ b/include/sound/soc-component.h @@ -147,6 +147,8 @@ struct snd_soc_component { unsigned int active; + unsigned int suspended:1; /* is in suspend PM state */ + struct list_head list; struct list_head card_aux_list; /* for auxiliary bound components */ struct list_head card_list; @@ -180,11 +182,6 @@ struct snd_soc_component { struct dentry *debugfs_root; const char *debugfs_prefix; #endif - - /* bit field */ - unsigned int suspended:1; /* is in suspend PM state */ - unsigned int opened:1; - unsigned int module:1; }; #define for_each_component_dais(component, dai)\ diff --git a/sound/soc/soc-component.c b/sound/soc/soc-component.c index ee00c09df5e7..14e175cdeeb8 100644 --- a/sound/soc/soc-component.c +++ b/sound/soc/soc-component.c @@ -297,55 +297,34 @@ EXPORT_SYMBOL_GPL(snd_soc_component_set_jack); int snd_soc_component_module_get(struct snd_soc_component *component, int upon_open) { - if (component->module) - return 0; - if (component->driver->module_get_upon_open == !!upon_open && !try_module_get(component->dev->driver->owner)) return -ENODEV; - component->module = 1; - return 0; } void snd_soc_component_module_put(struct snd_soc_component *component, int upon_open) { - if (component->module && - component->driver->module_get_upon_open == !!upon_open) + if (component->driver->module_get_upon_open == !!upon_open) module_put(component->dev->driver->owner); - - component->module = 0; } int snd_soc_component_open(struct snd_soc_component *component, struct snd_pcm_substream *substream) { - int ret = 0; - - if (!component->opened && - component->driver->open) - ret = component->driver->open(component, substream); - - if (ret == 0) - component->opened = 1; - - return ret; + if (component->driver->open) + return component->driver->open(component, substream); + return 0; } int snd_soc_component_close(struct snd_soc_component *component, struct snd_pcm_substream *substream) { - int ret = 0; - - if (component->opened && - component->driver->close) - ret = component->driver->close(component, substream); - - component->opened = 0; - - return ret; + if (component->driver->close) + return component->driver->close(component, substream); + return 0; } int snd_soc_component_prepare(struct snd_soc_component *component, diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 63f67eb7c077..0c3a12d6290a 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -466,13 +466,16 @@ static void soc_pcm_init_runtime_hw(struct snd_pcm_substream *substream) hw->rate_max = min_not_zero(hw->rate_max, rate_max); } -static int soc_pcm_components_open(struct snd_pcm_substream *substream) +static int soc_pcm_components_open(struct snd_pcm_substream *substream, + struct snd_soc_component **last) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_component *component; int i, ret = 0; for_each_rtd_components(rtd, i, component) { + *last = component; + ret = snd_soc_component_module_get_when_open(component); if (ret < 0) { dev_err(component->dev, @@ -489,17 +492,21 @@ static int soc_pcm_components_open(struct snd_pcm_substream *substream) return ret; } } - + *last = NULL; return 0; } -static int soc_pcm_components_close(struct snd_pcm_substream *substream) +static int soc_pcm_components_close(struct snd_pcm_substream *substream, + struct snd_soc_component *last) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_component *component; int i, r, ret = 0; for_each_rtd_components(rtd, i, component) { + if (component == last) + break; + r = snd_soc_component_close(component, substream); if (r < 0) ret = r; /* use last ret */ @@ -536,7 +543,7 @@ static int soc_pcm_close(struct snd_pcm_substream *substream) soc_rtd_shutdown(rtd, substream); - soc_pcm_components_close(substream); + soc_pcm_components_close(substream, NULL); snd_soc_dapm_stream_stop(rtd, substream->stream); @@ -577,7 +584,7 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass); - ret = soc_pcm_components_open(substream); + ret = soc_pcm_components_open(substream, &component); if (ret < 0) goto component_err; @@ -682,7 +689,7 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) soc_rtd_shutdown(rtd, substream); component_err: - soc_pcm_components_close(substream); + soc_pcm_components_close(substream, component); mutex_unlock(&rtd->card->pcm_mutex); -- 2.17.1

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[PATCH v3 0/3] simple-audio-card codec2codec support
by Samuel Holland 03 Mar '20

03 Mar '20

We are currently using simple-audio-card on the Allwinner A64 SoC. The digital audio codec there (sun8i-codec) has 3 AIFs, one each for the CPU, the modem, and Bluetooth. Adding support for the secondary AIFs requires adding codec2codec DAI links. Since the modem and bt-sco codec DAI drivers only have one set of possible PCM parameters (namely, 8kHz mono S16LE), there's no real need for a machine driver to specify the DAI link configuration. The parameters for these "simple" DAI links can be chosen automatically. This series adds codec2codec DAI link support to simple-audio-card. Codec to codec links are automatically detected when all DAIs in the link belong to codec components. I tried to reuse as much code as possible, so the first two patches refactor a couple of helper functions to be more generic. The last patch adds the new feature and its documentation. Changes in v3: - Update use of for_each_rtd_components for v5.6 Changes in v2: - Drop patch 1 as it was merged - Automatically detect codec2codec links instead of using a DT property Samuel Holland (3): ALSA: pcm: Add a standalone version of snd_pcm_limit_hw_rates ASoC: pcm: Export parameter intersection logic ASoC: simple-card: Add support for codec2codec DAI links Documentation/sound/soc/codec-to-codec.rst | 9 +++- include/sound/pcm.h | 9 +++- include/sound/soc.h | 3 ++ sound/core/pcm_misc.c | 18 +++---- sound/soc/generic/simple-card-utils.c | 49 +++++++++++++++++++ sound/soc/soc-pcm.c | 55 +++++++++++++++------- 6 files changed, 115 insertions(+), 28 deletions(-) -- 2.24.1

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[PATCH v2] soundwire: bus: provide correct return value on error
by Pierre-Louis Bossart 03 Mar '20

03 Mar '20

From: Bard Liao <yung-chuan.liao(a)linux.intel.com> It seems to be a typo. It makes more sense to return the return value of sdw_update() instead of the value we want to update. Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- v2: address Vinod's comment and use single return drivers/soundwire/bus.c | 7 ++----- 1 file changed, 2 insertions(+), 5 deletions(-) diff --git a/drivers/soundwire/bus.c b/drivers/soundwire/bus.c index ccaa590df61e..488c3c9e4947 100644 --- a/drivers/soundwire/bus.c +++ b/drivers/soundwire/bus.c @@ -1056,13 +1056,10 @@ static int sdw_initialize_slave(struct sdw_slave *slave) val |= SDW_DP0_INT_PORT_READY | SDW_DP0_INT_BRA_FAILURE; ret = sdw_update(slave, SDW_DP0_INTMASK, val, val); - if (ret < 0) { + if (ret < 0) dev_err(slave->bus->dev, "SDW_DP0_INTMASK read failed:%d\n", ret); - return val; - } - - return 0; + return ret; } static int sdw_handle_dp0_interrupt(struct sdw_slave *slave, u8 *slave_status) -- 2.20.1

2 1

[PATCH 0/9] ASoC: Intel: machine driver updates for 5.7
by Pierre-Louis Bossart 02 Mar '20

02 Mar '20

Fix HDaudio/HDMI Kconfig dependencies, cleanup code, and add support for 2 new configs (DA7219+Maxim 98360A and RT5692+RT1015) Guennadi Liakhovetski (3): ASoC: codecs: hdac_hdmi: (cosmetic) remove redundant variable initialisations ASoC: Intel: skylake: (cosmetic) remove redundant variable initialisations ASoC: Intel: (cosmetic) simplify structure member access Kai Vehmanen (4): ASoC: Intel: boards: drop reverse deps for SND_HDA_CODEC_HDMI ASoC: Intel: sof_pcm512x: drop reverse deps for SND_HDA_CODEC_HDMI ASoC: Intel: sof_pcm512x: make HDMI optional for all platforms ASoC: SOF: Intel: hda: remove SND_SOC_SOF_HDA_COMMON_HDMI_CODEC Yong Zhi (2): ASoC: Intel: sof_da7219_max98373: Add support for max98360a speaker amp ASoC: Intel: sof_rt5682: Add rt1015 speaker amp support sound/soc/codecs/hdac_hdmi.c | 6 +- sound/soc/intel/boards/Kconfig | 20 ++-- sound/soc/intel/boards/bdw-rt5650.c | 2 +- sound/soc/intel/boards/bdw-rt5677.c | 2 +- sound/soc/intel/boards/broadwell.c | 2 +- sound/soc/intel/boards/bxt_da7219_max98357a.c | 2 +- sound/soc/intel/boards/bxt_rt298.c | 2 +- sound/soc/intel/boards/bytcht_da7213.c | 2 +- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 2 +- sound/soc/intel/boards/cht_bsw_nau8824.c | 2 +- sound/soc/intel/boards/cht_bsw_rt5645.c | 2 +- sound/soc/intel/boards/cml_rt1011_rt5682.c | 2 +- sound/soc/intel/boards/glk_rt5682_max98357a.c | 2 +- sound/soc/intel/boards/haswell.c | 2 +- sound/soc/intel/boards/kbl_rt5663_max98927.c | 2 +- .../intel/boards/kbl_rt5663_rt5514_max98927.c | 2 +- sound/soc/intel/boards/skl_hda_dsp_generic.c | 2 +- .../soc/intel/boards/skl_nau88l25_max98357a.c | 2 +- sound/soc/intel/boards/skl_nau88l25_ssm4567.c | 2 +- sound/soc/intel/boards/sof_da7219_max98373.c | 76 ++++++++++-- sound/soc/intel/boards/sof_pcm512x.c | 38 ++++-- sound/soc/intel/boards/sof_rt5682.c | 110 +++++++++++++++++- .../intel/common/soc-acpi-intel-jsl-match.c | 34 +++++- sound/soc/intel/skylake/skl.c | 4 +- sound/soc/sof/intel/Kconfig | 11 -- sound/soc/sof/intel/hda.c | 3 +- 26 files changed, 267 insertions(+), 69 deletions(-) base-commit: 6941b0b5f919e9839e8c25efaeb53854efee14e5 -- 2.20.1

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[PATCH 0/4] ASoC: SOF: updates for 5.7
by Pierre-Louis Bossart 02 Mar '20

02 Mar '20

4 unrelated improvements grouped in one bundle. Jaska Uimonen (1): ASoC: SOF: ipc: check ipc return value before data copy Keyon Jie (2): ASoC: SOF: pcm: skip DMA buffer pre-allocation ASoC: SOF: Intel: hda-loader: clear the IPC ack bit after FW_PURGE done Tomasz Lauda (1): ASoC: SOF: add core id to sof_ipc_comp include/sound/sof/topology.h | 3 ++- include/uapi/sound/sof/abi.h | 2 +- sound/soc/sof/intel/hda-loader.c | 6 ++++++ sound/soc/sof/ipc.c | 12 +++++++----- sound/soc/sof/pcm.c | 6 ++---- 5 files changed, 18 insertions(+), 11 deletions(-) base-commit: 6941b0b5f919e9839e8c25efaeb53854efee14e5 -- 2.20.1

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[PATCH] ASoC: SOF: Intel: hda: use snd_sof_dsp_set_power_state() op
by Ranjani Sridharan 02 Mar '20

02 Mar '20

Replace the calls to hda_dsp_set_power_state() with the top-level SOF op snd_sof_set_power_state(). Along with this, modify the hda_dsp_resume() function to return the value of snd_sof_set_power_state() directly. Signed-off-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Jaska Uimonen <jaska.uimonen(a)linux.intel.com> --- sound/soc/sof/intel/hda-dsp.c | 15 +++++++-------- 1 file changed, 7 insertions(+), 8 deletions(-) diff --git a/sound/soc/sof/intel/hda-dsp.c b/sound/soc/sof/intel/hda-dsp.c index 0e61c27785a3..79ce52c32ef1 100644 --- a/sound/soc/sof/intel/hda-dsp.c +++ b/sound/soc/sof/intel/hda-dsp.c @@ -662,7 +662,7 @@ int hda_dsp_resume(struct snd_sof_dev *sdev) hda_codec_i915_display_power(sdev, true); /* Set DSP power state */ - ret = hda_dsp_set_power_state(sdev, &target_state); + ret = snd_sof_dsp_set_power_state(sdev, &target_state); if (ret < 0) { dev_err(sdev->dev, "error: setting dsp state %d substate %d\n", target_state.state, target_state.substate); @@ -686,8 +686,7 @@ int hda_dsp_resume(struct snd_sof_dev *sdev) if (ret < 0) return ret; - hda_dsp_set_power_state(sdev, &target_state); - return ret; + return snd_sof_dsp_set_power_state(sdev, &target_state); } int hda_dsp_runtime_resume(struct snd_sof_dev *sdev) @@ -702,7 +701,7 @@ int hda_dsp_runtime_resume(struct snd_sof_dev *sdev) if (ret < 0) return ret; - return hda_dsp_set_power_state(sdev, &target_state); + return snd_sof_dsp_set_power_state(sdev, &target_state); } int hda_dsp_runtime_idle(struct snd_sof_dev *sdev) @@ -730,7 +729,7 @@ int hda_dsp_runtime_suspend(struct snd_sof_dev *sdev) if (ret < 0) return ret; - return hda_dsp_set_power_state(sdev, &target_state); + return snd_sof_dsp_set_power_state(sdev, &target_state); } int hda_dsp_suspend(struct snd_sof_dev *sdev, u32 target_state) @@ -753,7 +752,7 @@ int hda_dsp_suspend(struct snd_sof_dev *sdev, u32 target_state) hda_codec_i915_display_power(sdev, false); /* Set DSP power state */ - ret = hda_dsp_set_power_state(sdev, &target_dsp_state); + ret = snd_sof_dsp_set_power_state(sdev, &target_dsp_state); if (ret < 0) { dev_err(sdev->dev, "error: setting dsp state %d substate %d\n", target_dsp_state.state, @@ -781,7 +780,7 @@ int hda_dsp_suspend(struct snd_sof_dev *sdev, u32 target_state) return ret; } - return hda_dsp_set_power_state(sdev, &target_dsp_state); + return snd_sof_dsp_set_power_state(sdev, &target_dsp_state); } int hda_dsp_set_hw_params_upon_resume(struct snd_sof_dev *sdev) @@ -849,7 +848,7 @@ void hda_dsp_d0i3_work(struct work_struct *work) return; /* This can fail but error cannot be propagated */ - ret = hda_dsp_set_power_state(sdev, &target_state); + ret = snd_sof_dsp_set_power_state(sdev, &target_state); if (ret < 0) dev_err_ratelimited(sdev->dev, "error: failed to set DSP state %d substate %d\n", -- 2.17.1

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[PATCH v2 0/3] Clean up RK3328 audio codec GPIO control
by Robin Murphy 01 Mar '20

01 Mar '20

Hi all, Since nobody said anything resembling "that looks like a terrible idea!" to what I propsed on top of v1, resending as a full v2 per Mark's request. Robin. Robin Murphy (3): ASoC: dt-bindings: Make RK3328 codec GPIO explicit ASoC: rockchip: Make RK3328 GPIO_MUTE control explicit arm64: dts: rockchip: Describe RK3328 GPIO_MUTE users .../bindings/sound/rockchip,rk3328-codec.txt | 7 ++++- arch/arm64/boot/dts/rockchip/rk3328-a1.dts | 1 + .../arm64/boot/dts/rockchip/rk3328-rock64.dts | 1 + sound/soc/codecs/rk3328_codec.c | 31 ++++++++++--------- 4 files changed, 24 insertions(+), 16 deletions(-) -- 2.17.1

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