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

[PATCH] ALSA: hda/realtek - Enable the headset of ASUS B9450FA with ALC294
by Jian-Hong Pan 05 Mar '20

05 Mar '20

A headset on the laptop like ASUS B9450FA does not work, until quirk ALC294_FIXUP_ASUS_HPE is applied. Signed-off-by: Jian-Hong Pan <jian-hong(a)endlessm.com> Singed-off-by: Kailang Yang <kailang(a)realtek.com> --- sound/pci/hda/patch_realtek.c | 15 +++++++++++++-- 1 file changed, 13 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 477589e7ec1d..a47f6404aea9 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5920,7 +5920,7 @@ enum { ALC289_FIXUP_DUAL_SPK, ALC294_FIXUP_SPK2_TO_DAC1, ALC294_FIXUP_ASUS_DUAL_SPK, - + ALC294_FIXUP_ASUS_HPE, }; static const struct hda_fixup alc269_fixups[] = { @@ -7040,7 +7040,17 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC294_FIXUP_SPK2_TO_DAC1 }, - + [ALC294_FIXUP_ASUS_HPE] = { + .type = HDA_FIXUP_VERBS, + .v.verbs = (const struct hda_verb[]) { + /* Set EAPD high */ + { 0x20, AC_VERB_SET_COEF_INDEX, 0x0f }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x7774 }, + { } + }, + .chained = true, + .chain_id = ALC294_FIXUP_ASUS_HEADSET_MIC + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -7204,6 +7214,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x16e3, "ASUS UX50", ALC269_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x1043, 0x17d1, "ASUS UX431FL", ALC294_FIXUP_ASUS_DUAL_SPK), SND_PCI_QUIRK(0x1043, 0x18b1, "Asus MJ401TA", ALC256_FIXUP_ASUS_HEADSET_MIC), + SND_PCI_QUIRK(0x1043, 0x19ce, "ASUS B9450FA", ALC294_FIXUP_ASUS_HPE), SND_PCI_QUIRK(0x1043, 0x1a13, "Asus G73Jw", ALC269_FIXUP_ASUS_G73JW), SND_PCI_QUIRK(0x1043, 0x1a30, "ASUS X705UD", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x1b13, "Asus U41SV", ALC269_FIXUP_INV_DMIC), -- 2.25.1

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Enable Headset Button for ThinkPad X1
by Kailang 05 Mar '20

05 Mar '20

Hi Takashi, Lenovo ThinkPad X1 want to support Headset Button. Attach patch will enable it. BR, Kailang

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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

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[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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