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[PATCH] ASoC: da7219: Fix I/O voltage range configuration during probe
by Adam Thomson 21 Aug '20

21 Aug '20

Previous improvements around handling device and codec level probe functionality added the possibility of the voltage level being undefined for the scenario where the IO voltage retrieved from the regulator supply was below 1.2V, whereas previously the code defaulted to the 2.5V to 3.6V range in that case. This commit restores the default value to avoid this happening. Fixes: aa5b18d1c290 ("ASoC: da7219: Move soft reset handling to codec level probe") Reported-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Adam Thomson <Adam.Thomson.Opensource(a)diasemi.com> --- sound/soc/codecs/da7219.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/da7219.c b/sound/soc/codecs/da7219.c index ddd422c..045aeb6 100644 --- a/sound/soc/codecs/da7219.c +++ b/sound/soc/codecs/da7219.c @@ -1862,6 +1862,9 @@ static int da7219_handle_supplies(struct snd_soc_component *component, return ret; } + /* Default to upper range */ + *io_voltage_lvl = DA7219_IO_VOLTAGE_LEVEL_2_5V_3_6V; + /* Determine VDDIO voltage provided */ vddio = da7219->supplies[DA7219_SUPPLY_VDDIO].consumer; ret = regulator_get_voltage(vddio); @@ -1869,8 +1872,6 @@ static int da7219_handle_supplies(struct snd_soc_component *component, dev_warn(component->dev, "Invalid VDDIO voltage\n"); else if (ret < 2800000) *io_voltage_lvl = DA7219_IO_VOLTAGE_LEVEL_1_2V_2_8V; - else - *io_voltage_lvl = DA7219_IO_VOLTAGE_LEVEL_2_5V_3_6V; /* Enable main supplies */ ret = regulator_bulk_enable(DA7219_NUM_SUPPLIES, da7219->supplies); -- 1.9.1

2 1

[PATCH] ASoC: qcom: Set card->owner to avoid warnings
by Stephan Gerhold 21 Aug '20

21 Aug '20

On Linux 5.9-rc1 I get the following warning with apq8016-sbc: WARNING: CPU: 2 PID: 69 at sound/core/init.c:207 snd_card_new+0x36c/0x3b0 [snd] CPU: 2 PID: 69 Comm: kworker/2:1 Not tainted 5.9.0-rc1 #1 Workqueue: events deferred_probe_work_func pc : snd_card_new+0x36c/0x3b0 [snd] lr : snd_card_new+0xf4/0x3b0 [snd] Call trace: snd_card_new+0x36c/0x3b0 [snd] snd_soc_bind_card+0x340/0x9a0 [snd_soc_core] snd_soc_register_card+0xf4/0x110 [snd_soc_core] devm_snd_soc_register_card+0x44/0xa0 [snd_soc_core] apq8016_sbc_platform_probe+0x11c/0x140 [snd_soc_apq8016_sbc] This warning was introduced in commit 81033c6b584b ("ALSA: core: Warn on empty module"). It looks like we are supposed to set card->owner to THIS_MODULE. Fix this for all the qcom ASoC drivers. Cc: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Fixes: 79119c798649 ("ASoC: qcom: Add Storm machine driver") Fixes: bdb052e81f62 ("ASoC: qcom: add apq8016 sound card support") Fixes: a6f933f63f2f ("ASoC: qcom: apq8096: Add db820c machine driver") Fixes: 6b1687bf76ef ("ASoC: qcom: add sdm845 sound card support") Signed-off-by: Stephan Gerhold <stephan(a)gerhold.net> --- sound/soc/qcom/apq8016_sbc.c | 1 + sound/soc/qcom/apq8096.c | 1 + sound/soc/qcom/sdm845.c | 1 + sound/soc/qcom/storm.c | 1 + 4 files changed, 4 insertions(+) diff --git a/sound/soc/qcom/apq8016_sbc.c b/sound/soc/qcom/apq8016_sbc.c index 083413abc2f6..575e2aefefe3 100644 --- a/sound/soc/qcom/apq8016_sbc.c +++ b/sound/soc/qcom/apq8016_sbc.c @@ -143,6 +143,7 @@ static int apq8016_sbc_platform_probe(struct platform_device *pdev) card = &data->card; card->dev = dev; + card->owner = THIS_MODULE; card->dapm_widgets = apq8016_sbc_dapm_widgets; card->num_dapm_widgets = ARRAY_SIZE(apq8016_sbc_dapm_widgets); diff --git a/sound/soc/qcom/apq8096.c b/sound/soc/qcom/apq8096.c index 253549600c5a..1a69baefc5ce 100644 --- a/sound/soc/qcom/apq8096.c +++ b/sound/soc/qcom/apq8096.c @@ -114,6 +114,7 @@ static int apq8096_platform_probe(struct platform_device *pdev) return -ENOMEM; card->dev = dev; + card->owner = THIS_MODULE; dev_set_drvdata(dev, card); ret = qcom_snd_parse_of(card); if (ret) diff --git a/sound/soc/qcom/sdm845.c b/sound/soc/qcom/sdm845.c index 0d10fba53945..ab1bf23c21a6 100644 --- a/sound/soc/qcom/sdm845.c +++ b/sound/soc/qcom/sdm845.c @@ -555,6 +555,7 @@ static int sdm845_snd_platform_probe(struct platform_device *pdev) card->dapm_widgets = sdm845_snd_widgets; card->num_dapm_widgets = ARRAY_SIZE(sdm845_snd_widgets); card->dev = dev; + card->owner = THIS_MODULE; dev_set_drvdata(dev, card); ret = qcom_snd_parse_of(card); if (ret) diff --git a/sound/soc/qcom/storm.c b/sound/soc/qcom/storm.c index c0c388d4db82..80c9cf2f254a 100644 --- a/sound/soc/qcom/storm.c +++ b/sound/soc/qcom/storm.c @@ -96,6 +96,7 @@ static int storm_platform_probe(struct platform_device *pdev) return -ENOMEM; card->dev = &pdev->dev; + card->owner = THIS_MODULE; ret = snd_soc_of_parse_card_name(card, "qcom,model"); if (ret) { -- 2.28.0

2 1

[PATCH 1/2] ASoC: bindings: fsl-asoc-card: add compatible string for TLV320AIC32x4 codec
by Matthias Schiffer 21 Aug '20

21 Aug '20

The TLV320AIC32x4 is commonly used on TQ-Systems starterkit mainboards for i.MX-based SoMs (i.MX6Q/DL, i.MX6UL, i.MX7) and LS1021A. Signed-off-by: Matthias Schiffer <matthias.schiffer(a)ew.tq-group.com> --- Documentation/devicetree/bindings/sound/fsl-asoc-card.txt | 2 ++ 1 file changed, 2 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt b/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt index 63ebf52b43e8..f339be62e7e4 100644 --- a/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt +++ b/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt @@ -38,6 +38,8 @@ The compatible list for this generic sound card currently: "fsl,imx-audio-wm8524" + "fsl,imx-audio-tlv320aic32x4" + Required properties: - compatible : Contains one of entries in the compatible list. -- 2.17.1

3 4

[PATCH] ASoC: intel: sof_sdw: add .exit callback function
by Bard Liao 21 Aug '20

21 Aug '20

We may allocate some resources in sof_sdw_codec_info .init function. Adding a corresponding .exit function can help to release these resources. Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> --- sound/soc/intel/boards/sof_sdw.c | 31 +++++++++++++++++++++++++ sound/soc/intel/boards/sof_sdw_common.h | 1 + 2 files changed, 32 insertions(+) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index 2463d432bf4d..4bc1ed757009 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -1032,12 +1032,43 @@ static int mc_probe(struct platform_device *pdev) return ret; } +static int mc_remove(struct platform_device *pdev) +{ + struct snd_soc_card *card = platform_get_drvdata(pdev); + struct snd_soc_dai_link *link; + int ret; + int i, j; + + for (i = 0; i < ARRAY_SIZE(codec_info_list); i++) { + if (!codec_info_list[i].exit) + continue; + /* + * We don't need to call .exit function if there is no matched + * dai link found. + */ + for_each_card_prelinks(card, j, link) { + if (!strcmp(link->codecs[0].dai_name, + codec_info_list[i].dai_name)) { + ret = codec_info_list[i].exit(&pdev->dev, link); + if (ret) + dev_warn(&pdev->dev, + "codec exit failed %d\n", + ret); + break; + } + } + } + + return 0; +} + static struct platform_driver sof_sdw_driver = { .driver = { .name = "sof_sdw", .pm = &snd_soc_pm_ops, }, .probe = mc_probe, + .remove = mc_remove, }; module_platform_driver(sof_sdw_driver); diff --git a/sound/soc/intel/boards/sof_sdw_common.h b/sound/soc/intel/boards/sof_sdw_common.h index 12e32439ba46..76d6c0c3839d 100644 --- a/sound/soc/intel/boards/sof_sdw_common.h +++ b/sound/soc/intel/boards/sof_sdw_common.h @@ -63,6 +63,7 @@ struct sof_sdw_codec_info { struct sof_sdw_codec_info *info, bool playback); + int (*exit)(struct device *dev, struct snd_soc_dai_link *dai_link); bool late_probe; int (*codec_card_late_probe)(struct snd_soc_card *card); }; -- 2.17.1

2 1

[PATCH AUTOSEL 4.4 01/22] ALSA: pci: delete repeated words in comments
by Sasha Levin 21 Aug '20

21 Aug '20

From: Randy Dunlap <rdunlap(a)infradead.org> [ Upstream commit c7fabbc51352f50cc58242a6dc3b9c1a3599849b ] Drop duplicated words in sound/pci/. {and, the, at} Signed-off-by: Randy Dunlap <rdunlap(a)infradead.org> Link: https://lore.kernel.org/r/20200806021926.32418-1-rdunlap@infradead.org Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/cs46xx/cs46xx_lib.c | 2 +- sound/pci/cs46xx/dsp_spos_scb_lib.c | 2 +- sound/pci/hda/hda_codec.c | 2 +- sound/pci/hda/hda_generic.c | 2 +- sound/pci/hda/patch_sigmatel.c | 2 +- sound/pci/ice1712/prodigy192.c | 2 +- sound/pci/oxygen/xonar_dg.c | 2 +- 7 files changed, 7 insertions(+), 7 deletions(-) diff --git a/sound/pci/cs46xx/cs46xx_lib.c b/sound/pci/cs46xx/cs46xx_lib.c index 2706f271a83b0..8a174c170e0aa 100644 --- a/sound/pci/cs46xx/cs46xx_lib.c +++ b/sound/pci/cs46xx/cs46xx_lib.c @@ -780,7 +780,7 @@ static void snd_cs46xx_set_capture_sample_rate(struct snd_cs46xx *chip, unsigned rate = 48000 / 9; /* - * We can not capture at at rate greater than the Input Rate (48000). + * We can not capture at a rate greater than the Input Rate (48000). * Return an error if an attempt is made to stray outside that limit. */ if (rate > 48000) diff --git a/sound/pci/cs46xx/dsp_spos_scb_lib.c b/sound/pci/cs46xx/dsp_spos_scb_lib.c index 7488e1b7a7707..4e726d39b05d1 100644 --- a/sound/pci/cs46xx/dsp_spos_scb_lib.c +++ b/sound/pci/cs46xx/dsp_spos_scb_lib.c @@ -1742,7 +1742,7 @@ int cs46xx_iec958_pre_open (struct snd_cs46xx *chip) struct dsp_spos_instance * ins = chip->dsp_spos_instance; if ( ins->spdif_status_out & DSP_SPDIF_STATUS_OUTPUT_ENABLED ) { - /* remove AsynchFGTxSCB and and PCMSerialInput_II */ + /* remove AsynchFGTxSCB and PCMSerialInput_II */ cs46xx_dsp_disable_spdif_out (chip); /* save state */ diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 825d9b27dbe12..4962a9d8a572b 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -3496,7 +3496,7 @@ EXPORT_SYMBOL_GPL(snd_hda_set_power_save); * @nid: NID to check / update * * Check whether the given NID is in the amp list. If it's in the list, - * check the current AMP status, and update the the power-status according + * check the current AMP status, and update the power-status according * to the mute status. * * This function is supposed to be set or called from the check_power_status diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 869c322ddae31..7cd1047a4edf3 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -837,7 +837,7 @@ static void activate_amp_in(struct hda_codec *codec, struct nid_path *path, } } -/* sync power of each widget in the the given path */ +/* sync power of each widget in the given path */ static hda_nid_t path_power_update(struct hda_codec *codec, struct nid_path *path, bool allow_powerdown) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index d1a6d20ace0da..80b72d0702c5e 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -862,7 +862,7 @@ static int stac_auto_create_beep_ctls(struct hda_codec *codec, static struct snd_kcontrol_new beep_vol_ctl = HDA_CODEC_VOLUME(NULL, 0, 0, 0); - /* check for mute support for the the amp */ + /* check for mute support for the amp */ if ((caps & AC_AMPCAP_MUTE) >> AC_AMPCAP_MUTE_SHIFT) { const struct snd_kcontrol_new *temp; if (spec->anabeep_nid == nid) diff --git a/sound/pci/ice1712/prodigy192.c b/sound/pci/ice1712/prodigy192.c index 3919aed39ca03..5e52086d7b986 100644 --- a/sound/pci/ice1712/prodigy192.c +++ b/sound/pci/ice1712/prodigy192.c @@ -31,7 +31,7 @@ * Experimentally I found out that only a combination of * OCKS0=1, OCKS1=1 (128fs, 64fs output) and ice1724 - * VT1724_MT_I2S_MCLK_128X=0 (256fs input) yields correct - * sampling rate. That means the the FPGA doubles the + * sampling rate. That means that the FPGA doubles the * MCK01 rate. * * Copyright (c) 2003 Takashi Iwai <tiwai(a)suse.de> diff --git a/sound/pci/oxygen/xonar_dg.c b/sound/pci/oxygen/xonar_dg.c index 4cf3200e988b0..df44135e1b0c9 100644 --- a/sound/pci/oxygen/xonar_dg.c +++ b/sound/pci/oxygen/xonar_dg.c @@ -39,7 +39,7 @@ * GPIO 4 <- headphone detect * GPIO 5 -> enable ADC analog circuit for the left channel * GPIO 6 -> enable ADC analog circuit for the right channel - * GPIO 7 -> switch green rear output jack between CS4245 and and the first + * GPIO 7 -> switch green rear output jack between CS4245 and the first * channel of CS4361 (mechanical relay) * GPIO 8 -> enable output to speakers * -- 2.25.1

1 1

[PATCH AUTOSEL 4.9 01/26] ALSA: pci: delete repeated words in comments
by Sasha Levin 21 Aug '20

21 Aug '20

From: Randy Dunlap <rdunlap(a)infradead.org> [ Upstream commit c7fabbc51352f50cc58242a6dc3b9c1a3599849b ] Drop duplicated words in sound/pci/. {and, the, at} Signed-off-by: Randy Dunlap <rdunlap(a)infradead.org> Link: https://lore.kernel.org/r/20200806021926.32418-1-rdunlap@infradead.org Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/cs46xx/cs46xx_lib.c | 2 +- sound/pci/cs46xx/dsp_spos_scb_lib.c | 2 +- sound/pci/hda/hda_codec.c | 2 +- sound/pci/hda/hda_generic.c | 2 +- sound/pci/hda/patch_sigmatel.c | 2 +- sound/pci/ice1712/prodigy192.c | 2 +- sound/pci/oxygen/xonar_dg.c | 2 +- 7 files changed, 7 insertions(+), 7 deletions(-) diff --git a/sound/pci/cs46xx/cs46xx_lib.c b/sound/pci/cs46xx/cs46xx_lib.c index 528102cc2d5d0..d824ff4ae3e3b 100644 --- a/sound/pci/cs46xx/cs46xx_lib.c +++ b/sound/pci/cs46xx/cs46xx_lib.c @@ -780,7 +780,7 @@ static void snd_cs46xx_set_capture_sample_rate(struct snd_cs46xx *chip, unsigned rate = 48000 / 9; /* - * We can not capture at at rate greater than the Input Rate (48000). + * We can not capture at a rate greater than the Input Rate (48000). * Return an error if an attempt is made to stray outside that limit. */ if (rate > 48000) diff --git a/sound/pci/cs46xx/dsp_spos_scb_lib.c b/sound/pci/cs46xx/dsp_spos_scb_lib.c index 7488e1b7a7707..4e726d39b05d1 100644 --- a/sound/pci/cs46xx/dsp_spos_scb_lib.c +++ b/sound/pci/cs46xx/dsp_spos_scb_lib.c @@ -1742,7 +1742,7 @@ int cs46xx_iec958_pre_open (struct snd_cs46xx *chip) struct dsp_spos_instance * ins = chip->dsp_spos_instance; if ( ins->spdif_status_out & DSP_SPDIF_STATUS_OUTPUT_ENABLED ) { - /* remove AsynchFGTxSCB and and PCMSerialInput_II */ + /* remove AsynchFGTxSCB and PCMSerialInput_II */ cs46xx_dsp_disable_spdif_out (chip); /* save state */ diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index cbe0248225c1c..4e67614f15f8e 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -3496,7 +3496,7 @@ EXPORT_SYMBOL_GPL(snd_hda_set_power_save); * @nid: NID to check / update * * Check whether the given NID is in the amp list. If it's in the list, - * check the current AMP status, and update the the power-status according + * check the current AMP status, and update the power-status according * to the mute status. * * This function is supposed to be set or called from the check_power_status diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 949c90a859fab..184089c5e8cbc 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -820,7 +820,7 @@ static void activate_amp_in(struct hda_codec *codec, struct nid_path *path, } } -/* sync power of each widget in the the given path */ +/* sync power of each widget in the given path */ static hda_nid_t path_power_update(struct hda_codec *codec, struct nid_path *path, bool allow_powerdown) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index d1a6d20ace0da..80b72d0702c5e 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -862,7 +862,7 @@ static int stac_auto_create_beep_ctls(struct hda_codec *codec, static struct snd_kcontrol_new beep_vol_ctl = HDA_CODEC_VOLUME(NULL, 0, 0, 0); - /* check for mute support for the the amp */ + /* check for mute support for the amp */ if ((caps & AC_AMPCAP_MUTE) >> AC_AMPCAP_MUTE_SHIFT) { const struct snd_kcontrol_new *temp; if (spec->anabeep_nid == nid) diff --git a/sound/pci/ice1712/prodigy192.c b/sound/pci/ice1712/prodigy192.c index 3919aed39ca03..5e52086d7b986 100644 --- a/sound/pci/ice1712/prodigy192.c +++ b/sound/pci/ice1712/prodigy192.c @@ -31,7 +31,7 @@ * Experimentally I found out that only a combination of * OCKS0=1, OCKS1=1 (128fs, 64fs output) and ice1724 - * VT1724_MT_I2S_MCLK_128X=0 (256fs input) yields correct - * sampling rate. That means the the FPGA doubles the + * sampling rate. That means that the FPGA doubles the * MCK01 rate. * * Copyright (c) 2003 Takashi Iwai <tiwai(a)suse.de> diff --git a/sound/pci/oxygen/xonar_dg.c b/sound/pci/oxygen/xonar_dg.c index 4cf3200e988b0..df44135e1b0c9 100644 --- a/sound/pci/oxygen/xonar_dg.c +++ b/sound/pci/oxygen/xonar_dg.c @@ -39,7 +39,7 @@ * GPIO 4 <- headphone detect * GPIO 5 -> enable ADC analog circuit for the left channel * GPIO 6 -> enable ADC analog circuit for the right channel - * GPIO 7 -> switch green rear output jack between CS4245 and and the first + * GPIO 7 -> switch green rear output jack between CS4245 and the first * channel of CS4361 (mechanical relay) * GPIO 8 -> enable output to speakers * -- 2.25.1

1 1

[PATCH AUTOSEL 4.14 01/30] ALSA: pci: delete repeated words in comments
by Sasha Levin 21 Aug '20

21 Aug '20

From: Randy Dunlap <rdunlap(a)infradead.org> [ Upstream commit c7fabbc51352f50cc58242a6dc3b9c1a3599849b ] Drop duplicated words in sound/pci/. {and, the, at} Signed-off-by: Randy Dunlap <rdunlap(a)infradead.org> Link: https://lore.kernel.org/r/20200806021926.32418-1-rdunlap@infradead.org Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/cs46xx/cs46xx_lib.c | 2 +- sound/pci/cs46xx/dsp_spos_scb_lib.c | 2 +- sound/pci/hda/hda_codec.c | 2 +- sound/pci/hda/hda_generic.c | 2 +- sound/pci/hda/patch_sigmatel.c | 2 +- sound/pci/ice1712/prodigy192.c | 2 +- sound/pci/oxygen/xonar_dg.c | 2 +- 7 files changed, 7 insertions(+), 7 deletions(-) diff --git a/sound/pci/cs46xx/cs46xx_lib.c b/sound/pci/cs46xx/cs46xx_lib.c index 0020fd0efc466..09c547f4cc186 100644 --- a/sound/pci/cs46xx/cs46xx_lib.c +++ b/sound/pci/cs46xx/cs46xx_lib.c @@ -780,7 +780,7 @@ static void snd_cs46xx_set_capture_sample_rate(struct snd_cs46xx *chip, unsigned rate = 48000 / 9; /* - * We can not capture at at rate greater than the Input Rate (48000). + * We can not capture at a rate greater than the Input Rate (48000). * Return an error if an attempt is made to stray outside that limit. */ if (rate > 48000) diff --git a/sound/pci/cs46xx/dsp_spos_scb_lib.c b/sound/pci/cs46xx/dsp_spos_scb_lib.c index 7488e1b7a7707..4e726d39b05d1 100644 --- a/sound/pci/cs46xx/dsp_spos_scb_lib.c +++ b/sound/pci/cs46xx/dsp_spos_scb_lib.c @@ -1742,7 +1742,7 @@ int cs46xx_iec958_pre_open (struct snd_cs46xx *chip) struct dsp_spos_instance * ins = chip->dsp_spos_instance; if ( ins->spdif_status_out & DSP_SPDIF_STATUS_OUTPUT_ENABLED ) { - /* remove AsynchFGTxSCB and and PCMSerialInput_II */ + /* remove AsynchFGTxSCB and PCMSerialInput_II */ cs46xx_dsp_disable_spdif_out (chip); /* save state */ diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 7d65fe31c8257..a56f018d586f5 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -3394,7 +3394,7 @@ EXPORT_SYMBOL_GPL(snd_hda_set_power_save); * @nid: NID to check / update * * Check whether the given NID is in the amp list. If it's in the list, - * check the current AMP status, and update the the power-status according + * check the current AMP status, and update the power-status according * to the mute status. * * This function is supposed to be set or called from the check_power_status diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 28ef409a9e6ae..9dee657ce9e27 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -823,7 +823,7 @@ static void activate_amp_in(struct hda_codec *codec, struct nid_path *path, } } -/* sync power of each widget in the the given path */ +/* sync power of each widget in the given path */ static hda_nid_t path_power_update(struct hda_codec *codec, struct nid_path *path, bool allow_powerdown) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 7cd147411b22d..f7896a9ae3d65 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -863,7 +863,7 @@ static int stac_auto_create_beep_ctls(struct hda_codec *codec, static struct snd_kcontrol_new beep_vol_ctl = HDA_CODEC_VOLUME(NULL, 0, 0, 0); - /* check for mute support for the the amp */ + /* check for mute support for the amp */ if ((caps & AC_AMPCAP_MUTE) >> AC_AMPCAP_MUTE_SHIFT) { const struct snd_kcontrol_new *temp; if (spec->anabeep_nid == nid) diff --git a/sound/pci/ice1712/prodigy192.c b/sound/pci/ice1712/prodigy192.c index 3919aed39ca03..5e52086d7b986 100644 --- a/sound/pci/ice1712/prodigy192.c +++ b/sound/pci/ice1712/prodigy192.c @@ -31,7 +31,7 @@ * Experimentally I found out that only a combination of * OCKS0=1, OCKS1=1 (128fs, 64fs output) and ice1724 - * VT1724_MT_I2S_MCLK_128X=0 (256fs input) yields correct - * sampling rate. That means the the FPGA doubles the + * sampling rate. That means that the FPGA doubles the * MCK01 rate. * * Copyright (c) 2003 Takashi Iwai <tiwai(a)suse.de> diff --git a/sound/pci/oxygen/xonar_dg.c b/sound/pci/oxygen/xonar_dg.c index 4cf3200e988b0..df44135e1b0c9 100644 --- a/sound/pci/oxygen/xonar_dg.c +++ b/sound/pci/oxygen/xonar_dg.c @@ -39,7 +39,7 @@ * GPIO 4 <- headphone detect * GPIO 5 -> enable ADC analog circuit for the left channel * GPIO 6 -> enable ADC analog circuit for the right channel - * GPIO 7 -> switch green rear output jack between CS4245 and and the first + * GPIO 7 -> switch green rear output jack between CS4245 and the first * channel of CS4361 (mechanical relay) * GPIO 8 -> enable output to speakers * -- 2.25.1

1 1

[PATCH AUTOSEL 4.19 01/38] ALSA: pci: delete repeated words in comments
by Sasha Levin 21 Aug '20

21 Aug '20

From: Randy Dunlap <rdunlap(a)infradead.org> [ Upstream commit c7fabbc51352f50cc58242a6dc3b9c1a3599849b ] Drop duplicated words in sound/pci/. {and, the, at} Signed-off-by: Randy Dunlap <rdunlap(a)infradead.org> Link: https://lore.kernel.org/r/20200806021926.32418-1-rdunlap@infradead.org Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/cs46xx/cs46xx_lib.c | 2 +- sound/pci/cs46xx/dsp_spos_scb_lib.c | 2 +- sound/pci/hda/hda_codec.c | 2 +- sound/pci/hda/hda_generic.c | 2 +- sound/pci/hda/patch_sigmatel.c | 2 +- sound/pci/ice1712/prodigy192.c | 2 +- sound/pci/oxygen/xonar_dg.c | 2 +- 7 files changed, 7 insertions(+), 7 deletions(-) diff --git a/sound/pci/cs46xx/cs46xx_lib.c b/sound/pci/cs46xx/cs46xx_lib.c index 146e1a3498c73..419da70cd942a 100644 --- a/sound/pci/cs46xx/cs46xx_lib.c +++ b/sound/pci/cs46xx/cs46xx_lib.c @@ -780,7 +780,7 @@ static void snd_cs46xx_set_capture_sample_rate(struct snd_cs46xx *chip, unsigned rate = 48000 / 9; /* - * We can not capture at at rate greater than the Input Rate (48000). + * We can not capture at a rate greater than the Input Rate (48000). * Return an error if an attempt is made to stray outside that limit. */ if (rate > 48000) diff --git a/sound/pci/cs46xx/dsp_spos_scb_lib.c b/sound/pci/cs46xx/dsp_spos_scb_lib.c index 8d0a3d3573457..8ef51a29380af 100644 --- a/sound/pci/cs46xx/dsp_spos_scb_lib.c +++ b/sound/pci/cs46xx/dsp_spos_scb_lib.c @@ -1739,7 +1739,7 @@ int cs46xx_iec958_pre_open (struct snd_cs46xx *chip) struct dsp_spos_instance * ins = chip->dsp_spos_instance; if ( ins->spdif_status_out & DSP_SPDIF_STATUS_OUTPUT_ENABLED ) { - /* remove AsynchFGTxSCB and and PCMSerialInput_II */ + /* remove AsynchFGTxSCB and PCMSerialInput_II */ cs46xx_dsp_disable_spdif_out (chip); /* save state */ diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index f3a6b1d869d8a..dbeb62362f1c3 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -3410,7 +3410,7 @@ EXPORT_SYMBOL_GPL(snd_hda_set_power_save); * @nid: NID to check / update * * Check whether the given NID is in the amp list. If it's in the list, - * check the current AMP status, and update the the power-status according + * check the current AMP status, and update the power-status according * to the mute status. * * This function is supposed to be set or called from the check_power_status diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 2609161707a41..97adb7e340f99 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -825,7 +825,7 @@ static void activate_amp_in(struct hda_codec *codec, struct nid_path *path, } } -/* sync power of each widget in the the given path */ +/* sync power of each widget in the given path */ static hda_nid_t path_power_update(struct hda_codec *codec, struct nid_path *path, bool allow_powerdown) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index d8168aa2cef38..85c33f528d7b3 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -845,7 +845,7 @@ static int stac_auto_create_beep_ctls(struct hda_codec *codec, static struct snd_kcontrol_new beep_vol_ctl = HDA_CODEC_VOLUME(NULL, 0, 0, 0); - /* check for mute support for the the amp */ + /* check for mute support for the amp */ if ((caps & AC_AMPCAP_MUTE) >> AC_AMPCAP_MUTE_SHIFT) { const struct snd_kcontrol_new *temp; if (spec->anabeep_nid == nid) diff --git a/sound/pci/ice1712/prodigy192.c b/sound/pci/ice1712/prodigy192.c index 3919aed39ca03..5e52086d7b986 100644 --- a/sound/pci/ice1712/prodigy192.c +++ b/sound/pci/ice1712/prodigy192.c @@ -31,7 +31,7 @@ * Experimentally I found out that only a combination of * OCKS0=1, OCKS1=1 (128fs, 64fs output) and ice1724 - * VT1724_MT_I2S_MCLK_128X=0 (256fs input) yields correct - * sampling rate. That means the the FPGA doubles the + * sampling rate. That means that the FPGA doubles the * MCK01 rate. * * Copyright (c) 2003 Takashi Iwai <tiwai(a)suse.de> diff --git a/sound/pci/oxygen/xonar_dg.c b/sound/pci/oxygen/xonar_dg.c index 4cf3200e988b0..df44135e1b0c9 100644 --- a/sound/pci/oxygen/xonar_dg.c +++ b/sound/pci/oxygen/xonar_dg.c @@ -39,7 +39,7 @@ * GPIO 4 <- headphone detect * GPIO 5 -> enable ADC analog circuit for the left channel * GPIO 6 -> enable ADC analog circuit for the right channel - * GPIO 7 -> switch green rear output jack between CS4245 and and the first + * GPIO 7 -> switch green rear output jack between CS4245 and the first * channel of CS4361 (mechanical relay) * GPIO 8 -> enable output to speakers * -- 2.25.1

1 3

[PATCH AUTOSEL 5.4 01/48] ALSA: hda/hdmi: Add quirk to force connectivity
by Sasha Levin 21 Aug '20

21 Aug '20

From: Kai-Heng Feng <kai.heng.feng(a)canonical.com> [ Upstream commit cd72c317a0a11f64225b9a3f1fe503bb8c7327b5 ] HDMI on some platforms doesn't enable audio support because its Port Connectivity [31:30] is set to AC_JACK_PORT_NONE: Node 0x05 [Pin Complex] wcaps 0x40778d: 8-Channels Digital Amp-Out CP Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0b000094: OUT Detect HBR HDMI DP Pin Default 0x58560010: [N/A] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0x1, Sequence = 0x0 Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Power states: D0 D3 EPSS Power: setting=D0, actual=D0 Devices: 0 Connection: 3 0x02 0x03* 0x04 For now, use a quirk to force connectivity based on SSID. If there are more platforms affected by the same issue, we can eye for a more generic solution. Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> Link: https://lore.kernel.org/r/20200804155836.16252-1-kai.heng.feng@canonical.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/patch_hdmi.c | 16 +++++++++++++++- 1 file changed, 15 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 908b68fda24c9..a9559fb29e209 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -176,6 +176,7 @@ struct hdmi_spec { bool use_jack_detect; /* jack detection enabled */ bool use_acomp_notifier; /* use eld_notify callback for hotplug */ bool acomp_registered; /* audio component registered in this driver */ + bool force_connect; /* force connectivity */ struct drm_audio_component_audio_ops drm_audio_ops; int (*port2pin)(struct hda_codec *, int); /* reverse port/pin mapping */ @@ -1711,7 +1712,8 @@ static int hdmi_add_pin(struct hda_codec *codec, hda_nid_t pin_nid) * all device entries on the same pin */ config = snd_hda_codec_get_pincfg(codec, pin_nid); - if (get_defcfg_connect(config) == AC_JACK_PORT_NONE) + if (get_defcfg_connect(config) == AC_JACK_PORT_NONE && + !spec->force_connect) return 0; /* @@ -1815,11 +1817,18 @@ static int hdmi_add_cvt(struct hda_codec *codec, hda_nid_t cvt_nid) return 0; } +static const struct snd_pci_quirk force_connect_list[] = { + SND_PCI_QUIRK(0x103c, 0x871a, "HP", 1), + {} +}; + static int hdmi_parse_codec(struct hda_codec *codec) { + struct hdmi_spec *spec = codec->spec; hda_nid_t start_nid; unsigned int caps; int i, nodes; + const struct snd_pci_quirk *q; nodes = snd_hda_get_sub_nodes(codec, codec->core.afg, &start_nid); if (!start_nid || nodes < 0) { @@ -1827,6 +1836,11 @@ static int hdmi_parse_codec(struct hda_codec *codec) return -EINVAL; } + q = snd_pci_quirk_lookup(codec->bus->pci, force_connect_list); + + if (q && q->value) + spec->force_connect = true; + /* * hdmi_add_pin() assumes total amount of converters to * be known, so first discover all converters -- 2.25.1

1 7

[PATCH AUTOSEL 5.7 01/61] ALSA: hda/hdmi: Add quirk to force connectivity
by Sasha Levin 21 Aug '20

21 Aug '20

From: Kai-Heng Feng <kai.heng.feng(a)canonical.com> [ Upstream commit cd72c317a0a11f64225b9a3f1fe503bb8c7327b5 ] HDMI on some platforms doesn't enable audio support because its Port Connectivity [31:30] is set to AC_JACK_PORT_NONE: Node 0x05 [Pin Complex] wcaps 0x40778d: 8-Channels Digital Amp-Out CP Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0b000094: OUT Detect HBR HDMI DP Pin Default 0x58560010: [N/A] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0x1, Sequence = 0x0 Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Power states: D0 D3 EPSS Power: setting=D0, actual=D0 Devices: 0 Connection: 3 0x02 0x03* 0x04 For now, use a quirk to force connectivity based on SSID. If there are more platforms affected by the same issue, we can eye for a more generic solution. Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> Link: https://lore.kernel.org/r/20200804155836.16252-1-kai.heng.feng@canonical.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/patch_hdmi.c | 16 +++++++++++++++- 1 file changed, 15 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 37391c3d2f47c..de9570032e07c 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -160,6 +160,7 @@ struct hdmi_spec { bool use_acomp_notifier; /* use eld_notify callback for hotplug */ bool acomp_registered; /* audio component registered in this driver */ + bool force_connect; /* force connectivity */ struct drm_audio_component_audio_ops drm_audio_ops; int (*port2pin)(struct hda_codec *, int); /* reverse port/pin mapping */ @@ -1700,7 +1701,8 @@ static int hdmi_add_pin(struct hda_codec *codec, hda_nid_t pin_nid) * all device entries on the same pin */ config = snd_hda_codec_get_pincfg(codec, pin_nid); - if (get_defcfg_connect(config) == AC_JACK_PORT_NONE) + if (get_defcfg_connect(config) == AC_JACK_PORT_NONE && + !spec->force_connect) return 0; /* @@ -1802,11 +1804,18 @@ static int hdmi_add_cvt(struct hda_codec *codec, hda_nid_t cvt_nid) return 0; } +static const struct snd_pci_quirk force_connect_list[] = { + SND_PCI_QUIRK(0x103c, 0x871a, "HP", 1), + {} +}; + static int hdmi_parse_codec(struct hda_codec *codec) { + struct hdmi_spec *spec = codec->spec; hda_nid_t start_nid; unsigned int caps; int i, nodes; + const struct snd_pci_quirk *q; nodes = snd_hda_get_sub_nodes(codec, codec->core.afg, &start_nid); if (!start_nid || nodes < 0) { @@ -1814,6 +1823,11 @@ static int hdmi_parse_codec(struct hda_codec *codec) return -EINVAL; } + q = snd_pci_quirk_lookup(codec->bus->pci, force_connect_list); + + if (q && q->value) + spec->force_connect = true; + /* * hdmi_add_pin() assumes total amount of converters to * be known, so first discover all converters -- 2.25.1

1 8

[PATCH AUTOSEL 5.8 01/62] ALSA: hda/hdmi: Add quirk to force connectivity
by Sasha Levin 21 Aug '20

21 Aug '20

From: Kai-Heng Feng <kai.heng.feng(a)canonical.com> [ Upstream commit cd72c317a0a11f64225b9a3f1fe503bb8c7327b5 ] HDMI on some platforms doesn't enable audio support because its Port Connectivity [31:30] is set to AC_JACK_PORT_NONE: Node 0x05 [Pin Complex] wcaps 0x40778d: 8-Channels Digital Amp-Out CP Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0b000094: OUT Detect HBR HDMI DP Pin Default 0x58560010: [N/A] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0x1, Sequence = 0x0 Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Power states: D0 D3 EPSS Power: setting=D0, actual=D0 Devices: 0 Connection: 3 0x02 0x03* 0x04 For now, use a quirk to force connectivity based on SSID. If there are more platforms affected by the same issue, we can eye for a more generic solution. Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> Link: https://lore.kernel.org/r/20200804155836.16252-1-kai.heng.feng@canonical.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/patch_hdmi.c | 16 +++++++++++++++- 1 file changed, 15 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index cd46247988e4d..b62cd3abb8273 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -160,6 +160,7 @@ struct hdmi_spec { bool use_acomp_notifier; /* use eld_notify callback for hotplug */ bool acomp_registered; /* audio component registered in this driver */ + bool force_connect; /* force connectivity */ struct drm_audio_component_audio_ops drm_audio_ops; int (*port2pin)(struct hda_codec *, int); /* reverse port/pin mapping */ @@ -1701,7 +1702,8 @@ static int hdmi_add_pin(struct hda_codec *codec, hda_nid_t pin_nid) * all device entries on the same pin */ config = snd_hda_codec_get_pincfg(codec, pin_nid); - if (get_defcfg_connect(config) == AC_JACK_PORT_NONE) + if (get_defcfg_connect(config) == AC_JACK_PORT_NONE && + !spec->force_connect) return 0; /* @@ -1803,11 +1805,18 @@ static int hdmi_add_cvt(struct hda_codec *codec, hda_nid_t cvt_nid) return 0; } +static const struct snd_pci_quirk force_connect_list[] = { + SND_PCI_QUIRK(0x103c, 0x871a, "HP", 1), + {} +}; + static int hdmi_parse_codec(struct hda_codec *codec) { + struct hdmi_spec *spec = codec->spec; hda_nid_t start_nid; unsigned int caps; int i, nodes; + const struct snd_pci_quirk *q; nodes = snd_hda_get_sub_nodes(codec, codec->core.afg, &start_nid); if (!start_nid || nodes < 0) { @@ -1815,6 +1824,11 @@ static int hdmi_parse_codec(struct hda_codec *codec) return -EINVAL; } + q = snd_pci_quirk_lookup(codec->bus->pci, force_connect_list); + + if (q && q->value) + spec->force_connect = true; + /* * hdmi_add_pin() assumes total amount of converters to * be known, so first discover all converters -- 2.25.1

1 8

[PATCH] soundwire: intel: fix CONFIG_PM and CONFIG_PM_SLEEP confusion
by Bard Liao 21 Aug '20

21 Aug '20

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> When CONFIG_PM_SLEEP is not defined, GCC throws compilation warnings: drivers/soundwire/intel.c:1816:12: warning: ‘intel_resume’ defined but not used [-Wunused-function] 1816 | static int intel_resume(struct device *dev) | ^~~~~~~~~~~~ drivers/soundwire/intel.c:1697:12: warning: ‘intel_suspend’ defined but not used [-Wunused-function] 1697 | static int intel_suspend(struct device *dev) Fix by adding the missing CONFIG_PM_SLEEP. Note that we could move code around and use only 2 ifdefs, but this will generate conflicts so let's do this when all the pm handling is merged. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> --- drivers/soundwire/intel.c | 12 ++++++++++-- 1 file changed, 10 insertions(+), 2 deletions(-) diff --git a/drivers/soundwire/intel.c b/drivers/soundwire/intel.c index dbcbe2708563..a2f0026cb2c1 100644 --- a/drivers/soundwire/intel.c +++ b/drivers/soundwire/intel.c @@ -1525,7 +1525,7 @@ int intel_master_process_wakeen_event(struct platform_device *pdev) * PM calls */ -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP static int intel_suspend(struct device *dev) { @@ -1562,6 +1562,9 @@ static int intel_suspend(struct device *dev) return 0; } +#endif + +#ifdef CONFIG_PM static int intel_suspend_runtime(struct device *dev) { @@ -1624,6 +1627,9 @@ static int intel_suspend_runtime(struct device *dev) return ret; } +#endif + +#ifdef CONFIG_PM_SLEEP static int intel_resume(struct device *dev) { @@ -1691,6 +1697,9 @@ static int intel_resume(struct device *dev) return ret; } +#endif + +#ifdef CONFIG_PM static int intel_resume_runtime(struct device *dev) { @@ -1797,7 +1806,6 @@ static int intel_resume_runtime(struct device *dev) return ret; } - #endif static const struct dev_pm_ops intel_pm = { -- 2.17.1

3 2

[alsa-devel] Attempting to understand odd snd-usb-audio code and behavior
by Keith A. Milner 21 Aug '20

21 Aug '20

Over the last few weeks I've been trying to dig through the snd-usb-audio code to try to get an understanding of it. I've used ftrace and also a local version which I have heavily polluted with kprintf(KERN_INFO statements so I can track what it's doing in practice. I've been playing with this with some USB audio interfaces I have some which work, and some which don't, and trying to understand the differences and, ultimately, what I can do to make the non-functioning ones work. I've got to the point where I have a fairly good understanding of some parts, but I'm struggling to understand why the code is behaving as it does in practice. My observation when plugging the USB device in is that, as expected, it tries to build the audio device, and add components to it including attaching capture and playback streams. It seems to go through this phase without too many surprises, and we get to the standard dmesg output: usbcore: registered new interface driver snd-usb-audio After this, it gets confusing. It seems to open and close the PCMs (e.g, snd_usb_capture_open) several times in a row. Each time it sets up the hardware info, checks and sets the format and then closes the PCM. I don't see anything obviously different between most of these cycles. Here's an example my logging output (hopefully this will make sense): Aug 22 20:04:16 kerneldev kernel: [ 923.140789] Called: pcm.c - snd_usb_capture_open for pcm substream 0 Aug 22 20:04:16 kerneldev kernel: [ 923.140789] Called: pcm.c - snd_usb_pcm_open for pcm substream 0 Aug 22 20:04:16 kerneldev kernel: [ 923.140789] Setting up hw info Aug 22 20:04:16 kerneldev kernel: [ 923.140789] Checking format: Aug 22 20:04:16 kerneldev kernel: [ 923.140790] Interface number = 2 Aug 22 20:04:16 kerneldev kernel: [ 923.140790] alt setting = 1 Aug 22 20:04:16 kerneldev kernel: [ 923.140790] alt setting index = 1 Aug 22 20:04:16 kerneldev kernel: [ 923.140790] format type = 1 Aug 22 20:04:16 kerneldev kernel: [ 923.140790] frame size = 0 Aug 22 20:04:16 kerneldev kernel: [ 923.140791] Setting pt=125 ptmin=125 Aug 22 20:04:16 kerneldev kernel: [ 923.140792] Checking for unusual bit rates Aug 22 20:04:16 kerneldev kernel: [ 923.140794] Called: init.c - snd_card_file_remove Aug 22 20:04:16 kerneldev kernel: [ 923.140921] Called: pcm.c - snd_usb_hw_params for pcm substream 0 Aug 22 20:04:16 kerneldev kernel: [ 923.140945] Setting format: Aug 22 20:04:16 kerneldev kernel: [ 923.140945] Bytes per period = 8816 Aug 22 20:04:16 kerneldev kernel: [ 923.140945] Frames per period = 1102 Aug 22 20:04:16 kerneldev kernel: [ 923.140946] Periods per buffer = 4 Aug 22 20:04:16 kerneldev kernel: [ 923.140946] Channels = 2 Aug 22 20:04:16 kerneldev kernel: [ 923.140946] Rate = 44100 Aug 22 20:04:16 kerneldev kernel: [ 923.141279] Called: endpoint.c - snd_usb_add_endpoint of type 0 with direction 1 Aug 22 20:04:16 kerneldev kernel: [ 923.141292] Called: pcm.c - snd_usb_pcm_prepare for pcm substream 0 Aug 22 20:04:16 kerneldev kernel: [ 923.141293] Called: pcm.c - configure_endpoint Aug 22 20:04:16 kerneldev kernel: [ 923.141294] Called: endpoint.c - snd_usb_endpoint_set_params for endpoint 142, type 0 Aug 22 20:04:16 kerneldev kernel: [ 923.141294] Setting endpoint params for endpoint num 142 Aug 22 20:04:16 kerneldev kernel: [ 923.141294] channels=2 Aug 22 20:04:16 kerneldev kernel: [ 923.141294] period bytes=8816 Aug 22 20:04:16 kerneldev kernel: [ 923.141295] frames per period=1102 Aug 22 20:04:16 kerneldev kernel: [ 923.141295] periods per buffer=4 Aug 22 20:04:16 kerneldev kernel: [ 923.141296] Called: endpoint.c - data_ep_set_params for endpoint 142, type 0 Aug 22 20:04:16 kerneldev kernel: [ 923.141301] Called: pcm.c - snd_usb_pcm_prepare for pcm substream 0 Aug 22 20:04:16 kerneldev kernel: [ 923.142768] Called: init.c - snd_card_file_add Aug 22 20:04:16 kerneldev kernel: [ 923.142772] Called: init.c - snd_card_file_remove Aug 22 20:04:16 kerneldev kernel: [ 923.142774] Called: init.c - snd_card_file_add Aug 22 20:04:16 kerneldev kernel: [ 923.142830] Called: init.c - snd_card_file_remove Aug 22 20:04:16 kerneldev kernel: [ 923.142836] Called: pcm.c - snd_usb_hw_free for pcm substream 0 Aug 22 20:04:16 kerneldev kernel: [ 923.142837] Called: endpoint.c - snd_usb_endpoint_deactivate for endpoint 142, type 0 Aug 22 20:04:16 kerneldev kernel: [ 923.142846] Called: pcm.c - snd_usb_capture_close for pcm substream 0 Aug 22 20:04:16 kerneldev kernel: [ 923.146760] Called: init.c - snd_card_file_remove Aug 22 20:04:16 kerneldev kernel: [ 923.146834] Called: init.c - snd_card_file_add Aug 22 20:04:16 kerneldev kernel: [ 923.146838] Called: init.c - snd_card_file_remove Aug 22 20:04:16 kerneldev kernel: [ 923.146841] Called: init.c - snd_card_file_add Aug 22 20:04:16 kerneldev kernel: [ 923.146845] Called: init.c - snd_card_file_add Aug 22 20:04:16 kerneldev kernel: [ 923.146848] Called: pcm.c - snd_usb_capture_open for pcm substream 0 Aug 22 20:04:16 kerneldev kernel: [ 923.146849] Called: pcm.c - snd_usb_pcm_open for pcm substream 0 Aug 22 20:04:16 kerneldev kernel: [ 923.146849] Setting up hw info It seems to do this several times, before the behavior changing slightly, as it now starts to try to set the endpoint parameters and seems to leave the stream open. However, it then tries to open the stream and set the endpoint parameters again resulting in errors: Aug 22 20:22:00 kerneldev kernel: [ 1987.452945] Called: endpoint.c - snd_usb_endpoint_set_params for endpoint 142, type 0 Aug 22 20:22:00 kerneldev kernel: [ 1987.452945] Setting endpoint params for endpoint num 142 Aug 22 20:22:00 kerneldev kernel: [ 1987.452945] channels=2 Aug 22 20:22:00 kerneldev kernel: [ 1987.452945] period bytes=1768 Aug 22 20:22:00 kerneldev kernel: [ 1987.452946] frames per period=221 Aug 22 20:22:00 kerneldev kernel: [ 1987.452946] periods per buffer=593 Aug 22 20:22:00 kerneldev kernel: [ 1987.452947] usb 1-1: Unable to change format on ep #8e: already in use Maybe this is driven by hotplug or something, but it seems to be performing a lot of pointless activities. Eventually it seems to try to open the PCM playback or capture stream (with some devices it opens playback, with others capture) and then checks the pointer: Aug 22 20:22:00 kerneldev kernel: [ 1987.470194] Called: pcm.c - snd_usb_substream_playback_trigger for pcm substream 0 Aug 22 20:22:00 kerneldev kernel: [ 1987.470198] Called: pcm.c - snd_usb_pcm_pointer for pcm substream 0 Aug 22 20:22:00 kerneldev kernel: [ 1987.470198] Pointer is 0 Aug 22 20:22:00 kerneldev kernel: [ 1987.470656] Called: pcm.c - snd_usb_pcm_pointer for pcm substream 0 Aug 22 20:22:00 kerneldev kernel: [ 1987.470657] Pointer is 0 Aug 22 20:22:00 kerneldev kernel: [ 1987.470660] Called: pcm.c - snd_usb_pcm_pointer for pcm substream 0 Aug 22 20:22:00 kerneldev kernel: [ 1987.470660] Pointer is 0 Aug 22 20:22:05 kerneldev kernel: [ 1992.473567] Called: pcm.c - snd_usb_pcm_pointer for pcm substream 0 Aug 22 20:22:05 kerneldev kernel: [ 1992.473569] Pointer is 0 The above is from a non-working device. For a working device the pointer increments. Eventually this activity ends: Aug 22 20:22:05 kerneldev kernel: [ 1992.473575] Called: pcm.c - snd_usb_substream_playback_trigger for pcm substream 0 Aug 22 20:22:05 kerneldev kernel: [ 1992.473576] Called: endpoint.c - snd_usb_endpoint_stop for endpoint 142, type 0 Aug 22 20:22:05 kerneldev kernel: [ 1992.473578] Called: endpoint.c - snd_usb_endpoint_stop for endpoint 13, type 0 Aug 22 20:22:05 kerneldev kernel: [ 1992.473602] Called: pcm.c - snd_usb_hw_free for pcm substream 0 Aug 22 20:22:05 kerneldev kernel: [ 1992.503287] Called: endpoint.c - snd_usb_endpoint_deactivate for endpoint 142, type 0 Aug 22 20:22:05 kerneldev kernel: [ 1992.503288] Called: endpoint.c - snd_usb_endpoint_deactivate for endpoint 13, type 0 Aug 22 20:22:05 kerneldev kernel: [ 1992.503323] Called: pcm.c - snd_usb_playback_close for pcm substream 0 Aug 22 20:22:05 kerneldev kernel: [ 1992.503656] Called: init.c - snd_card_file_remove This seems as if it might be some sort of test of the PCMs, to make sure they are working. Certainly in non-working devices the pcm_pointer stays at zero and in working devices it increments. It also seems to try harder with some devices than others, even though the device descriptors seem very similar. Can anyone explain any of this to me, particularly why it's exhibiting this behaviour. I've uploaded one of the logs from a non-working device to Pastebin if anyone wants to look at a full example: https://pastebin.com/uBZEuQ57 Also, is this the best way to approach this, or are there other tools/techniques that can help? I've experimented with ftrace and that was useful, but didn't give the level of detail I wanted. Cheers, Keith

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[PATCH 0/2] soundwire: fix port_ready[] dynamic allocation in
by Bard Liao 21 Aug '20

21 Aug '20

The existing code allocates memory for the total number of ports. This only works if the ports are contiguous, but will break if e.g. a Devices uses port0, 1, and 14. The port_ready[] array would contain 3 elements, which would lead to an out-of-bounds access. Conversely in other cases, the wrong port index would be used leading to timeouts on prepare. This can be fixed by allocating for the worst-case of 15 ports (DP0..DP14). In addition since the number is now fixed, we can use an array instead of a dynamic allocation. Pierre-Louis Bossart (2): soundwire: add definition for maximum number of ports soundwire: fix port_ready[] dynamic allocation in mipi_disco and ASoC codecs drivers/soundwire/mipi_disco.c | 18 +----------------- drivers/soundwire/slave.c | 4 ++++ include/linux/soundwire/sdw.h | 5 +++-- sound/soc/codecs/max98373-sdw.c | 15 +-------------- sound/soc/codecs/rt1308-sdw.c | 14 +------------- sound/soc/codecs/rt5682-sdw.c | 15 +-------------- sound/soc/codecs/rt700-sdw.c | 15 +-------------- sound/soc/codecs/rt711-sdw.c | 15 +-------------- sound/soc/codecs/rt715-sdw.c | 33 +-------------------------------- 9 files changed, 14 insertions(+), 120 deletions(-) -- 2.17.1

4 9

[PATCH v2] soundwire: SDCA: add helper macro to access controls
by Bard Liao 21 Aug '20

21 Aug '20

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> The upcoming SDCA (SoundWire Device Class Audio) specification defines a hiearchical encoding to interface with Class-defined capabilities, based on which audio function, entity, control and channel being used. The specification is not yet accessible to the general public but this information is released with explicit permission from the MIPI Board to avoid delays with SDCA support on Linux platforms. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Rander Wang <rander.wang(a)linux.intel.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> --- Changelog: v2: - add SDW_SDCA_MBQ_CTL --- include/linux/soundwire/sdw_registers.h | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/include/linux/soundwire/sdw_registers.h b/include/linux/soundwire/sdw_registers.h index 5d3c271af7d1..4517c0f65b4f 100644 --- a/include/linux/soundwire/sdw_registers.h +++ b/include/linux/soundwire/sdw_registers.h @@ -305,4 +305,16 @@ #define SDW_CASC_PORT_MASK_INTSTAT3 1 #define SDW_CASC_PORT_REG_OFFSET_INTSTAT3 2 +/* v1.2 device - SDCA address mapping */ +#define SDW_SDCA_CTL(fun, ent, ctl, ch) (BIT(30) | \ + (((fun) & 0x7) << 22) | \ + (((ent) & 0x40) << 15) | \ + (((ent) & 0x3f) << 7) | \ + (((ctl) & 0x30) << 15) | \ + (((ctl) & 0x0f) << 3) | \ + (((ch) & 0x38) << 12) | \ + ((ch) & 0x07)) + +#define SDW_SDCA_MBQ_CTL(reg) ((reg) | BIT(13)) + #endif /* __SDW_REGISTERS_H */ -- 2.17.1

3 3

[RESEND] ASoC: SOF: Intel: add build support for SoundWire
by Bard Liao 20 Aug '20

20 Aug '20

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Select SoundWire capabilities on newer Intel platforms, starting with CannonLake/CoffeeLake/CometLake. As done for HDaudio, the SoundWire link is an opt-in capability. We explicitly test for ACPI to avoid warnings on unmet dependencies on the SoundWire side. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> --- sound/soc/sof/intel/Kconfig | 29 +++++++++++++++++++++++++++++ 1 file changed, 29 insertions(+) diff --git a/sound/soc/sof/intel/Kconfig b/sound/soc/sof/intel/Kconfig index 3aaf25e4f766..f644010cc568 100644 --- a/sound/soc/sof/intel/Kconfig +++ b/sound/soc/sof/intel/Kconfig @@ -166,6 +166,7 @@ config SND_SOC_SOF_CANNONLAKE_SUPPORT config SND_SOC_SOF_CANNONLAKE tristate select SND_SOC_SOF_HDA_COMMON + select SND_SOC_SOF_INTEL_SOUNDWIRE_LINK_BASELINE help This option is not user-selectable but automagically handled by 'select' statements at a higher level @@ -181,6 +182,7 @@ config SND_SOC_SOF_COFFEELAKE_SUPPORT config SND_SOC_SOF_COFFEELAKE tristate select SND_SOC_SOF_HDA_COMMON + select SND_SOC_SOF_INTEL_SOUNDWIRE_LINK_BASELINE help This option is not user-selectable but automagically handled by 'select' statements at a higher level @@ -196,6 +198,7 @@ config SND_SOC_SOF_ICELAKE_SUPPORT config SND_SOC_SOF_ICELAKE tristate select SND_SOC_SOF_HDA_COMMON + select SND_SOC_SOF_INTEL_SOUNDWIRE_LINK_BASELINE help This option is not user-selectable but automagically handled by 'select' statements at a higher level @@ -203,6 +206,7 @@ config SND_SOC_SOF_ICELAKE config SND_SOC_SOF_COMETLAKE tristate select SND_SOC_SOF_HDA_COMMON + select SND_SOC_SOF_INTEL_SOUNDWIRE_LINK_BASELINE help This option is not user-selectable but automagically handled by 'select' statements at a higher level @@ -229,6 +233,7 @@ config SND_SOC_SOF_TIGERLAKE_SUPPORT config SND_SOC_SOF_TIGERLAKE tristate select SND_SOC_SOF_HDA_COMMON + select SND_SOC_SOF_INTEL_SOUNDWIRE_LINK_BASELINE help This option is not user-selectable but automagically handled by 'select' statements at a higher level @@ -244,6 +249,7 @@ config SND_SOC_SOF_ELKHARTLAKE_SUPPORT config SND_SOC_SOF_ELKHARTLAKE tristate select SND_SOC_SOF_HDA_COMMON + select SND_SOC_SOF_INTEL_SOUNDWIRE_LINK_BASELINE help This option is not user-selectable but automagically handled by 'select' statements at a higher level @@ -329,6 +335,29 @@ config SND_SOC_SOF_HDA This option is not user-selectable but automagically handled by 'select' statements at a higher level +config SND_SOC_SOF_INTEL_SOUNDWIRE_LINK + bool "SOF support for SoundWire" + depends on SOUNDWIRE && ACPI + help + This adds support for SoundWire with Sound Open Firmware + for Intel(R) platforms. + Say Y if you want to enable SoundWire links with SOF. + If unsure select "N". + +config SND_SOC_SOF_INTEL_SOUNDWIRE_LINK_BASELINE + tristate + select SND_SOC_SOF_INTEL_SOUNDWIRE if SND_SOC_SOF_INTEL_SOUNDWIRE_LINK + help + This option is not user-selectable but automagically handled by + 'select' statements at a higher level + +config SND_SOC_SOF_INTEL_SOUNDWIRE + tristate + select SOUNDWIRE_INTEL + help + This option is not user-selectable but automagically handled by + 'select' statements at a higher level + endif ## SND_SOC_SOF_INTEL_PCI endif ## SND_SOC_SOF_INTEL_TOPLEVEL -- 2.17.1

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[RESEND] ASoC: SOF: Intel: hda: import SOUNDWIRE_INIT namespace
by Bard Liao 20 Aug '20

20 Aug '20

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Make sure the SoundWire driver can be loaded Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> --- sound/soc/sof/intel/hda.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c index 63ca920c8e6e..c0b75d682750 100644 --- a/sound/soc/sof/intel/hda.c +++ b/sound/soc/sof/intel/hda.c @@ -1207,3 +1207,4 @@ MODULE_LICENSE("Dual BSD/GPL"); MODULE_IMPORT_NS(SND_SOC_SOF_HDA_AUDIO_CODEC); MODULE_IMPORT_NS(SND_SOC_SOF_HDA_AUDIO_CODEC_I915); MODULE_IMPORT_NS(SND_SOC_SOF_XTENSA); +MODULE_IMPORT_NS(SOUNDWIRE_INTEL_INIT); -- 2.17.1

2 1

[PATCH][next] ASoC: qcom: add missing out of memory check on drvdata->clks allocation
by Colin King 20 Aug '20

20 Aug '20

From: Colin Ian King <colin.king(a)canonical.com> Currently drvdata->clks is not being checked for an allocation failure, leading to potential null pointer dereferencing. Fix this by adding a check and returning -ENOMEM if an error occurred. Addresses-Coverity: ("Dereference null return value") Fixes: 1220f6a76e77 ("ASoC: qcom: Add common array to initialize soc based core clocks") Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/qcom/lpass-apq8016.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/qcom/lpass-apq8016.c b/sound/soc/qcom/lpass-apq8016.c index dd9e3dd014f6..5c8ae225cd5d 100644 --- a/sound/soc/qcom/lpass-apq8016.c +++ b/sound/soc/qcom/lpass-apq8016.c @@ -168,6 +168,8 @@ static int apq8016_lpass_init(struct platform_device *pdev) drvdata->clks = devm_kcalloc(dev, variant->num_clks, sizeof(*drvdata->clks), GFP_KERNEL); + if (!drvdata->clks) + return -ENOMEM; drvdata->num_clks = variant->num_clks; for (i = 0; i < drvdata->num_clks; i++) -- 2.27.0

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Re: [PATCH v2 00/41] spi / fbdev / cpufreq / usb / mmc / hwmon / ARM: Prepare for multiplatform S3C
by Krzysztof Kozlowski 20 Aug '20

20 Aug '20

On Thu, Aug 06, 2020 at 08:19:32PM +0200, Krzysztof Kozlowski wrote: > Hi All, > > Shortly > ======= > This is a continuation of Arnd's work from 2019 [1]. The goal is to > cleanup, merge and finally make the Samsung S3C24xx and S3C64xx > architectures multiplatform. The multiplatform did not happen yet > here... just cleaning up and merging into one mach-s3c. > > I intend to take it through Samsung SoC tree so all Acks are welcomed. > > Changes since v1 > ================ > 1. Rebased, > 2. Addressed comments (including mine), > 3. Few new patches. > > Please see individual changelogs (per patch). > > [1] https://patchwork.kernel.org/project/linux-samsung-soc/list/?series=185855&… > Hi All, I applied second part of the set which finishes this step of S3C cleanup. Thanks to Arnd for the work! Best regards, Krzysztof

1 0

JamVOX usb soundcard works only after windows reboot
by Krabador 20 Aug '20

20 Aug '20

Hi alsa-devel team. This device from linux from 5.2 (my first plug) to now with 5.8.1, the only way to use it properly is to run linux after windows reboot. If linux is run at poweron, the device is correctly seen by the kernel but muted. Here the results of alsa-info.sh when card is working , not working, a diff, dmesg and lsusb -v https://paste.ee/p/YrJSP#pRmvQ58T7sCeulHr1ZSluw2cTQt7exsh Waiting to give more information, Best Regards.

1 1

[PATCH v6 0/2] Add mediatek codec mt6359 driver
by Jiaxin Yu 20 Aug '20

20 Aug '20

Add mediatek codec (MT6359) driver MT6359 support playback and capture feature. On downlink path, it includes three DACs for handset, headset, and lineout path. On unlink path, it includeds three ADCs for main mic, second mic, 3rd mic, and headset mic. By scenario, select *_MUX widget to create damp path. And by select mic_type_mux to choose DMIC/AMIC/.... For example, select these MUX widget to create headset path (1) DAC In Mux --> "Normal Path" (2) HP Mux --> "Audio Playback" v6 changes: 1. Remove the compatible string in mt6359codec because MFD should be registering the platform device. v5 changes: 1. Don't need to unregister the component whic is already relegated to devm. 2. patchwork link: https://patchwork.kernel.org/cover/11716387/ https://patchwork.kernel.org/patch/11717757/ https://patchwork.kernel.org/patch/11716491/ v4 changes: 1. Add a remove() function to undo regulator_enable(). 2. Removed unnecessary logs. 3. patchwork link: https://patchwork.kernel.org/cover/11715553/ https://patchwork.kernel.org/patch/11716015/ https://patchwork.kernel.org/patch/11715557/ v3 changes: 1. patchwork link: https://patchwork.kernel.org/cover/11714291/ https://patchwork.kernel.org/patch/11714295/ https://patchwork.kernel.org/patch/11714339/ v2 changes: 1. patchwork link: https://patchwork.kernel.org/cover/11706935/ https://patchwork.kernel.org/patch/11708865/ https://patchwork.kernel.org/patch/11706937/ v1 changes: 1.lkml link: https://lkml.org/lkml/2020/3/5/1257 Jiaxin Yu (2): ASoC: mediatek: mt6359: add codec driver dt-bindings: mediatek: mt6359: add codec document .../devicetree/bindings/sound/mt6359.yaml | 61 + sound/soc/codecs/Kconfig | 8 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/mt6359.c | 2753 +++++++++++++++++ sound/soc/codecs/mt6359.h | 2640 ++++++++++++++++ 5 files changed, 5464 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/mt6359.yaml create mode 100644 sound/soc/codecs/mt6359.c create mode 100644 sound/soc/codecs/mt6359.h -- 2.18.0

2 3

Request to pick up couple NVIDIA Tegra ASoC patches into 5.7 kernel
by Dmitry Osipenko 20 Aug '20

20 Aug '20

Hello, stable-kernel maintainers! Could you please cherry-pick these commits into the v5.7.x kernel? commit 0de6db30ef79b391cedd749801a49c485d2daf4b Author: Sowjanya Komatineni <skomatineni(a)nvidia.com> Date: Mon Jan 13 23:24:17 2020 -0800 ASoC: tegra: Use device managed resource APIs to get the clock commit 1e4e0bf136aa4b4aa59c1e6af19844bd6d807794 Author: Sowjanya Komatineni <skomatineni(a)nvidia.com> Date: Mon Jan 13 23:24:23 2020 -0800 ASoC: tegra: Add audio mclk parent configuration commit ff5d18cb04f4ecccbcf05b7f83ab6df2a0d95c16 Author: Sowjanya Komatineni <skomatineni(a)nvidia.com> Date: Mon Jan 13 23:24:24 2020 -0800 ASoC: tegra: Enable audio mclk during tegra_asoc_utils_init() It will fix a huge warnings splat during of kernel boot on NVIDIA Tegra SoCs. For some reason these patches haven't made into 5.7 when it was released and several people complained about the warnings. Thanks in advance!

3 3

[PATCH] ALSA: usb-audio: Add prevent wakeup from s3 state trig by Lenovo ThinkCentre TI024Gen3 USB-audio
by penghao 20 Aug '20

20 Aug '20

TI024Gen3 USB-audio is controlled by TI024Gen3,when TI024Gens enter sleep mode, USB-audio will disconnect from USB bus port, wakup form s3 state Signed-off-by: penghao <penghao(a)uniontech.com> --- sound/usb/card.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/usb/card.c b/sound/usb/card.c index 696e788c5d31..6bdbb34009b3 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -658,6 +658,12 @@ static int usb_audio_probe(struct usb_interface *intf, } dev_set_drvdata(&dev->dev, chip); + /* + *ALSA: usb-audio: Add prevent wakeup from s3 state trig by Lenovo + *ThinkCentre TI024Gen3 usb-audio + */ + if ((usb_id->idVendor == 0x17ef) && (usb_id->idProduct == 0xa012)) + device_set_wakeup_enable(＆dev->dev, 0); /* * For devices with more than one control interface, we assume the -- 2.11.0

6 7

[PATCH v2] ALSA: hda: avoid reset of sdo_limit
by Sameer Pujar 19 Aug '20

19 Aug '20

By default 'sdo_limit' is initialized with a default value of '8' as per spec. This is overridden in cases where a different value is required. However this is getting reset when snd_hdac_bus_init_chip() is called again, which happens during runtime PM cycle. Avoid this reset by moving 'sdo_limit' setup to 'snd_hdac_bus_init()' function which would be called only once. Fixes: 67ae482a59e9 ("ALSA: hda: add member to store ratio for stripe control") Cc: <stable(a)vger.kernel.org> Signed-off-by: Sameer Pujar <spujar(a)nvidia.com> --- sound/hda/hdac_bus.c | 12 ++++++++++++ sound/hda/hdac_controller.c | 11 ----------- 2 files changed, 12 insertions(+), 11 deletions(-) diff --git a/sound/hda/hdac_bus.c b/sound/hda/hdac_bus.c index 09ddab5..9766f6a 100644 --- a/sound/hda/hdac_bus.c +++ b/sound/hda/hdac_bus.c @@ -46,6 +46,18 @@ int snd_hdac_bus_init(struct hdac_bus *bus, struct device *dev, INIT_LIST_HEAD(&bus->hlink_list); init_waitqueue_head(&bus->rirb_wq); bus->irq = -1; + + /* + * Default value of '8' is as per the HD audio specification (Rev 1.0a). + * Following relation is used to derive STRIPE control value. + * For sample rate <= 48K: + * { ((num_channels * bits_per_sample) / number of SDOs) >= 8 } + * For sample rate > 48K: + * { ((num_channels * bits_per_sample * rate/48000) / + * number of SDOs) >= 8 } + */ + bus->sdo_limit = 8; + return 0; } EXPORT_SYMBOL_GPL(snd_hdac_bus_init); diff --git a/sound/hda/hdac_controller.c b/sound/hda/hdac_controller.c index 011b17c..b98449f 100644 --- a/sound/hda/hdac_controller.c +++ b/sound/hda/hdac_controller.c @@ -529,17 +529,6 @@ bool snd_hdac_bus_init_chip(struct hdac_bus *bus, bool full_reset) bus->chip_init = true; - /* - * Default value of '8' is as per the HD audio specification (Rev 1.0a). - * Following relation is used to derive STRIPE control value. - * For sample rate <= 48K: - * { ((num_channels * bits_per_sample) / number of SDOs) >= 8 } - * For sample rate > 48K: - * { ((num_channels * bits_per_sample * rate/48000) / - * number of SDOs) >= 8 } - */ - bus->sdo_limit = 8; - return true; } EXPORT_SYMBOL_GPL(snd_hdac_bus_init_chip); -- 2.7.4

2 1

[PATCH] ALSA: hda: avoid reset of sdo_limit
by Sameer Pujar 19 Aug '20

19 Aug '20

By default 'sdo_limit' is initialized with a default value of '8' as per spec. This is overridden in cases where a different value is required. However this is getting reset when snd_hdac_bus_init_chip() is called again, which happens during runtime PM cycle. Avoid reset by not initializing to default value everytime. Fixes: 67ae482a59e9 ("ALSA: hda: add member to store ratio for stripe control") Cc: <stable(a)vger.kernel.org> Signed-off-by: Sameer Pujar <spujar(a)nvidia.com> --- sound/hda/hdac_controller.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/hda/hdac_controller.c b/sound/hda/hdac_controller.c index 011b17c..0e26e96 100644 --- a/sound/hda/hdac_controller.c +++ b/sound/hda/hdac_controller.c @@ -538,7 +538,8 @@ bool snd_hdac_bus_init_chip(struct hdac_bus *bus, bool full_reset) * { ((num_channels * bits_per_sample * rate/48000) / * number of SDOs) >= 8 } */ - bus->sdo_limit = 8; + if (!bus->sdo_limit) + bus->sdo_limit = 8; return true; } -- 2.7.4

2 2

[PATCH] ASoC: codecs: rt*-sdw: use SDW_SLAVE_ENTRY_EXT
by Bard Liao 19 Aug '20

19 Aug '20

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Add version and class information explicitly to prepare for support for new devices. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Rander Wang <rander.wang(a)linux.intel.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Reviewed-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> --- sound/soc/codecs/rt1308-sdw.c | 2 +- sound/soc/codecs/rt5682-sdw.c | 2 +- sound/soc/codecs/rt700-sdw.c | 2 +- sound/soc/codecs/rt711-sdw.c | 2 +- sound/soc/codecs/rt715-sdw.c | 2 +- 5 files changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/rt1308-sdw.c b/sound/soc/codecs/rt1308-sdw.c index 81e3e2caeb2f..ecb03ad8102e 100644 --- a/sound/soc/codecs/rt1308-sdw.c +++ b/sound/soc/codecs/rt1308-sdw.c @@ -696,7 +696,7 @@ static int rt1308_sdw_probe(struct sdw_slave *slave, } static const struct sdw_device_id rt1308_id[] = { - SDW_SLAVE_ENTRY(0x025d, 0x1308, 0), + SDW_SLAVE_ENTRY_EXT(0x025d, 0x1308, 0x2, 0, 0), {}, }; MODULE_DEVICE_TABLE(sdw, rt1308_id); diff --git a/sound/soc/codecs/rt5682-sdw.c b/sound/soc/codecs/rt5682-sdw.c index 148d4cff267b..4d5bc8ac38df 100644 --- a/sound/soc/codecs/rt5682-sdw.c +++ b/sound/soc/codecs/rt5682-sdw.c @@ -722,7 +722,7 @@ static int rt5682_sdw_remove(struct sdw_slave *slave) } static const struct sdw_device_id rt5682_id[] = { - SDW_SLAVE_ENTRY(0x025d, 0x5682, 0), + SDW_SLAVE_ENTRY_EXT(0x025d, 0x5682, 0x2, 0, 0), {}, }; MODULE_DEVICE_TABLE(sdw, rt5682_id); diff --git a/sound/soc/codecs/rt700-sdw.c b/sound/soc/codecs/rt700-sdw.c index 2d475405b20d..b08ccc3da1dd 100644 --- a/sound/soc/codecs/rt700-sdw.c +++ b/sound/soc/codecs/rt700-sdw.c @@ -483,7 +483,7 @@ static int rt700_sdw_remove(struct sdw_slave *slave) } static const struct sdw_device_id rt700_id[] = { - SDW_SLAVE_ENTRY(0x025d, 0x700, 0), + SDW_SLAVE_ENTRY_EXT(0x025d, 0x700, 0x1, 0, 0), {}, }; MODULE_DEVICE_TABLE(sdw, rt700_id); diff --git a/sound/soc/codecs/rt711-sdw.c b/sound/soc/codecs/rt711-sdw.c index 7a1ae7442e75..c72cd4fb64fd 100644 --- a/sound/soc/codecs/rt711-sdw.c +++ b/sound/soc/codecs/rt711-sdw.c @@ -484,7 +484,7 @@ static int rt711_sdw_remove(struct sdw_slave *slave) } static const struct sdw_device_id rt711_id[] = { - SDW_SLAVE_ENTRY(0x025d, 0x711, 0), + SDW_SLAVE_ENTRY_EXT(0x025d, 0x711, 0x2, 0, 0), {}, }; MODULE_DEVICE_TABLE(sdw, rt711_id); diff --git a/sound/soc/codecs/rt715-sdw.c b/sound/soc/codecs/rt715-sdw.c index 761d4663e813..c87a3fc0678b 100644 --- a/sound/soc/codecs/rt715-sdw.c +++ b/sound/soc/codecs/rt715-sdw.c @@ -546,7 +546,7 @@ static int rt715_sdw_probe(struct sdw_slave *slave, } static const struct sdw_device_id rt715_id[] = { - SDW_SLAVE_ENTRY(0x025d, 0x715, 0), + SDW_SLAVE_ENTRY_EXT(0x025d, 0x715, 0x2, 0, 0), {}, }; MODULE_DEVICE_TABLE(sdw, rt715_id); -- 2.17.1

2 1

[PATCH] ASoC: Intel: skl_hda_dsp_generic: Fix NULLptr dereference in autosuspend delay
by Mateusz Gorski 19 Aug '20

19 Aug '20

Different modules for HDMI codec are used depending on the "hda_codec_use_common_hdmi" option being enabled or not. Driver private context for both of them is different. This leads to null-pointer dereference error when driver tries to set autosuspend delay for HDMI codec while the option is off (hdac_hdmi module is used for HDMI). Change the string in conditional statement to "ehdaudio0D0" to ensure that only the HDAudio codec is handled by this function. Fixes: 5bf73b1b1dec ("ASoC: intel/skl/hda - fix oops on systems without i915 audio codec") Signed-off-by: Mateusz Gorski <mateusz.gorski(a)linux.intel.com> --- sound/soc/intel/boards/skl_hda_dsp_generic.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/boards/skl_hda_dsp_generic.c b/sound/soc/intel/boards/skl_hda_dsp_generic.c index ca4900036ead..bc50eda297ab 100644 --- a/sound/soc/intel/boards/skl_hda_dsp_generic.c +++ b/sound/soc/intel/boards/skl_hda_dsp_generic.c @@ -181,7 +181,7 @@ static void skl_set_hda_codec_autosuspend_delay(struct snd_soc_card *card) struct snd_soc_dai *dai; for_each_card_rtds(card, rtd) { - if (!strstr(rtd->dai_link->codecs->name, "ehdaudio")) + if (!strstr(rtd->dai_link->codecs->name, "ehdaudio0D0")) continue; dai = asoc_rtd_to_codec(rtd, 0); hda_pvt = snd_soc_component_get_drvdata(dai->component); -- 2.17.1

6 7

[PATCH V2] dt-bindings: sound: Convert NXP spdif to json-schema
by Anson Huang 19 Aug '20

19 Aug '20

Convert the NXP SPDIF binding to DT schema format using json-schema. Signed-off-by: Anson Huang <Anson.Huang(a)nxp.com> Reviewed-by: Rob Herring <robh(a)kernel.org> --- changes since V1: - add "additionalProperties: false". --- .../devicetree/bindings/sound/fsl,spdif.txt | 68 ------------- .../devicetree/bindings/sound/fsl,spdif.yaml | 110 +++++++++++++++++++++ 2 files changed, 110 insertions(+), 68 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/fsl,spdif.txt create mode 100644 Documentation/devicetree/bindings/sound/fsl,spdif.yaml diff --git a/Documentation/devicetree/bindings/sound/fsl,spdif.txt b/Documentation/devicetree/bindings/sound/fsl,spdif.txt deleted file mode 100644 index e1365b0..0000000 --- a/Documentation/devicetree/bindings/sound/fsl,spdif.txt +++ /dev/null @@ -1,68 +0,0 @@ -Freescale Sony/Philips Digital Interface Format (S/PDIF) Controller - -The Freescale S/PDIF audio block is a stereo transceiver that allows the -processor to receive and transmit digital audio via an coaxial cable or -a fibre cable. - -Required properties: - - - compatible : Compatible list, should contain one of the following - compatibles: - "fsl,imx35-spdif", - "fsl,vf610-spdif", - "fsl,imx6sx-spdif", - - - reg : Offset and length of the register set for the device. - - - interrupts : Contains the spdif interrupt. - - - dmas : Generic dma devicetree binding as described in - Documentation/devicetree/bindings/dma/dma.txt. - - - dma-names : Two dmas have to be defined, "tx" and "rx". - - - clocks : Contains an entry for each entry in clock-names. - - - clock-names : Includes the following entries: - "core" The core clock of spdif controller. - "rxtx<0-7>" Clock source list for tx and rx clock. - This clock list should be identical to the source - list connecting to the spdif clock mux in "SPDIF - Transceiver Clock Diagram" of SoC reference manual. - It can also be referred to TxClk_Source bit of - register SPDIF_STC. - "spba" The spba clock is required when SPDIF is placed as a - bus slave of the Shared Peripheral Bus and when two - or more bus masters (CPU, DMA or DSP) try to access - it. This property is optional depending on the SoC - design. - -Optional properties: - - - big-endian : If this property is absent, the native endian mode - will be in use as default, or the big endian mode - will be in use for all the device registers. - -Example: - -spdif: spdif@2004000 { - compatible = "fsl,imx35-spdif"; - reg = <0x02004000 0x4000>; - interrupts = <0 52 0x04>; - dmas = <&sdma 14 18 0>, - <&sdma 15 18 0>; - dma-names = "rx", "tx"; - - clocks = <&clks 197>, <&clks 3>, - <&clks 197>, <&clks 107>, - <&clks 0>, <&clks 118>, - <&clks 62>, <&clks 139>, - <&clks 0>; - clock-names = "core", "rxtx0", - "rxtx1", "rxtx2", - "rxtx3", "rxtx4", - "rxtx5", "rxtx6", - "rxtx7"; - - big-endian; -}; diff --git a/Documentation/devicetree/bindings/sound/fsl,spdif.yaml b/Documentation/devicetree/bindings/sound/fsl,spdif.yaml new file mode 100644 index 0000000..2ac671f --- /dev/null +++ b/Documentation/devicetree/bindings/sound/fsl,spdif.yaml @@ -0,0 +1,110 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/fsl,spdif.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Freescale Sony/Philips Digital Interface Format (S/PDIF) Controller + +maintainers: + - Shengjiu Wang <shengjiu.wang(a)nxp.com> + +description: | + The Freescale S/PDIF audio block is a stereo transceiver that allows the + processor to receive and transmit digital audio via an coaxial cable or + a fibre cable. + +properties: + compatible: + enum: + - fsl,imx35-spdif + - fsl,vf610-spdif + - fsl,imx6sx-spdif + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + dmas: + items: + - description: DMA controller phandle and request line for RX + - description: DMA controller phandle and request line for TX + + dma-names: + items: + - const: rx + - const: tx + + clocks: + items: + - description: The core clock of spdif controller. + - description: Clock for tx0 and rx0. + - description: Clock for tx1 and rx1. + - description: Clock for tx2 and rx2. + - description: Clock for tx3 and rx3. + - description: Clock for tx4 and rx4. + - description: Clock for tx5 and rx5. + - description: Clock for tx6 and rx6. + - description: Clock for tx7 and rx7. + - description: The spba clock is required when SPDIF is placed as a bus + slave of the Shared Peripheral Bus and when two or more bus masters + (CPU, DMA or DSP) try to access it. This property is optional depending + on the SoC design. + minItems: 9 + + clock-names: + items: + - const: core + - const: rxtx0 + - const: rxtx1 + - const: rxtx2 + - const: rxtx3 + - const: rxtx4 + - const: rxtx5 + - const: rxtx6 + - const: rxtx7 + - const: spba + minItems: 9 + + big-endian: + $ref: /schemas/types.yaml#/definitions/flag + description: | + If this property is absent, the native endian mode will be in use + as default, or the big endian mode will be in use for all the device + registers. Set this flag for HCDs with big endian descriptors and big + endian registers. + +required: + - compatible + - reg + - interrupts + - dmas + - dma-names + - clocks + - clock-names + +additionalProperties: false + +examples: + - | + spdif@2004000 { + compatible = "fsl,imx35-spdif"; + reg = <0x02004000 0x4000>; + interrupts = <0 52 0x04>; + dmas = <&sdma 14 18 0>, + <&sdma 15 18 0>; + dma-names = "rx", "tx"; + clocks = <&clks 197>, <&clks 3>, + <&clks 197>, <&clks 107>, + <&clks 0>, <&clks 118>, + <&clks 62>, <&clks 139>, + <&clks 0>; + clock-names = "core", "rxtx0", + "rxtx1", "rxtx2", + "rxtx3", "rxtx4", + "rxtx5", "rxtx6", + "rxtx7"; + big-endian; + }; -- 2.7.4

2 1

[PATCH] soundwire: bus: fix typo in comment on INTSTAT registers
by Bard Liao 19 Aug '20

19 Aug '20

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> s/Instat/Intstat/ Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> --- drivers/soundwire/bus.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/drivers/soundwire/bus.c b/drivers/soundwire/bus.c index e6e0fb9a81b4..da0201693c24 100644 --- a/drivers/soundwire/bus.c +++ b/drivers/soundwire/bus.c @@ -1372,7 +1372,7 @@ static int sdw_handle_slave_alerts(struct sdw_slave *slave) return ret; } - /* Read Instat 1, Instat 2 and Instat 3 registers */ + /* Read Intstat 1, Intstat 2 and Intstat 3 registers */ ret = sdw_read(slave, SDW_SCP_INT1); if (ret < 0) { dev_err(slave->bus->dev, -- 2.17.1

2 1

[PATCH] ALSA: hda/realtek: Add quirk for Samsung Galaxy Book Ion
by Mike Pozulp 19 Aug '20

19 Aug '20

The Galaxy Book Ion uses the same ALC298 codec as other Samsung laptops which have the no headphone sound bug, like my Samsung Notebook. The Galaxy Book owner confirmed that this patch fixes the bug. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207423 Signed-off-by: Mike Pozulp <pozulp.kernel(a)gmail.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 29f5878f0c50..72aa3da479c9 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7612,6 +7612,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x144d, 0xc109, "Samsung Ativ book 9 (NP900X3G)", ALC269_FIXUP_INV_DMIC), SND_PCI_QUIRK(0x144d, 0xc169, "Samsung Notebook 9 Pen (NP930SBE-K01US)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET), SND_PCI_QUIRK(0x144d, 0xc176, "Samsung Notebook 9 Pro (NP930MBE-K04US)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET), + SND_PCI_QUIRK(0x144d, 0xc18a, "Samsung Galaxy Book Ion (NT950XCJ-X716A)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET), SND_PCI_QUIRK(0x144d, 0xc740, "Samsung Ativ book 8 (NP870Z5G)", ALC269_FIXUP_ATIV_BOOK_8), SND_PCI_QUIRK(0x144d, 0xc812, "Samsung Notebook Pen S (NT950SBE-X58)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET), SND_PCI_QUIRK(0x1458, 0xfa53, "Gigabyte BXBT-2807", ALC283_FIXUP_HEADSET_MIC), -- 2.28.0

2 1

[GIT PULL] ASoC fixes for v5.9-rc1
by Mark Brown 19 Aug '20

19 Aug '20

The following changes since commit 8e34f1e867b572f1e20b5250c2897fe5f041c99f: Merge series "ASoC: core: Two step component registration" from Cezary Rojewski <cezary.rojewski(a)intel.com>: (2020-07-31 19:36:00 +0100) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v5.9-rc1 for you to fetch changes up to 062fa09f44f4fb3776a23184d5d296b0c8872eb9: ASoC: intel: Fix memleak in sst_media_open (2020-08-17 12:28:57 +0100) ---------------------------------------------------------------- ASoC: Fixes for v5.9 A bunch of fixes that came in during the merge window, mostly for issues that were uncovered by the changes to report errors on invalid register access plus one important fix in that code itself. ---------------------------------------------------------------- Dinghao Liu (1): ASoC: intel: Fix memleak in sst_media_open Fabio Estevam (1): ASoC: wm8962: Do not remove ADDITIONAL_CONTROL_4 from readable register list Hui Wang (1): ASoC: amd: renoir: restore two more registers during resume Mark Brown (1): Merge series "ASoC: tegra: Fix compile warning with CONFIG_PM=n" from Takashi Iwai <tiwai(a)suse.de>: Ravulapati Vishnu vardhan rao (1): ASoC: amd: Replacing component->name with codec_dai->name. Shengjiu Wang (1): ASoC: fsl-asoc-card: Remove fsl_asoc_card_set_bias_level function Srinivas Kandagatla (3): ASoC: q6afe-dai: mark all widgets registers as SND_SOC_NOPM ASoC: q6routing: add dummy register read/write function ASoC: msm8916-wcd-analog: fix register Interrupt offset Sylwester Nawrocki (2): ASoC: wm8994: Prevent access to invalid VU register bits on WM1811 ASoC: wm8994: Avoid attempts to read unreadable registers Takashi Iwai (7): ASoC: tegra: tegra186_dspk: Fix compile warning with CONFIG_PM=n ASoC: tegra: tegra210_admaif: Fix compile warning with CONFIG_PM=n ASoC: tegra: tegra210_ahub: Fix compile warning with CONFIG_PM=n ASoC: tegra: tegra210_dmic: Fix compile warning with CONFIG_PM=n ASoC: tegra: tegra210_i2s: Fix compile warning with CONFIG_PM=n ASoC: fsl: Fix unused variable warning ASoC: Make soc_component_read() returning an error code again sound/soc/amd/acp3x-rt5682-max9836.c | 2 +- sound/soc/amd/renoir/acp3x-pdm-dma.c | 29 ++-- sound/soc/codecs/msm8916-wcd-analog.c | 4 +- sound/soc/codecs/wm8958-dsp2.c | 4 + sound/soc/codecs/wm8962.c | 2 +- sound/soc/codecs/wm8994.c | 60 ++++++-- sound/soc/fsl/fsl-asoc-card.c | 154 +++++++++----------- sound/soc/fsl/mpc5200_dma.c | 1 - sound/soc/intel/atom/sst-mfld-platform-pcm.c | 5 +- sound/soc/qcom/qdsp6/q6afe-dai.c | 210 +++++++++++++-------------- sound/soc/qcom/qdsp6/q6routing.c | 16 ++ sound/soc/soc-component.c | 2 +- sound/soc/tegra/tegra186_dspk.c | 4 +- sound/soc/tegra/tegra210_admaif.c | 4 +- sound/soc/tegra/tegra210_ahub.c | 4 +- sound/soc/tegra/tegra210_dmic.c | 4 +- sound/soc/tegra/tegra210_i2s.c | 4 +- 17 files changed, 266 insertions(+), 243 deletions(-)

2 1

ucm: let rval_sysfs check the device/driver/module first
by GitHub pull_request - opened 19 Aug '20

19 Aug '20

alsa-project/alsa-lib pull request #79 was opened from jason77-wang: We plan to use the sound driver module's name for the top ucm's name, I checked it on 3 different ASoC machines, the device/driver links to machine driver while deice/driver/module links to the sound driver module, so change the code to readlink the device/driver/module first, if it fails then readlink the device/driver. This is the output from those 3 machines: $readlink /sys/class/sound/card0/device/driver ../../../../bus/platform/drivers/skl_hda_dsp_generic $readlink /sys/class/sound/card0/device/driver/module ../../../../module/snd_soc_skl_hda_dsp $readlink /sys/class/sound/card2/device/driver ../../../../../bus/platform/drivers/acp_pdm_mach $readlink /sys/class/sound/card2/device/driver/module ../../../../module/snd_acp3x_rn $readlink /sys/class/sound/card0/device/driver ../../../../bus/platform/drivers/sof_sdw $readlink /sys/class/sound/card0/device/driver/module ../../../../module/snd_soc_sof_sdw Signed-off-by: Hui Wang <hui.wang(a)canonical.com> Request URL : https://github.com/alsa-project/alsa-lib/pull/79 Patch URL : https://github.com/alsa-project/alsa-lib/pull/79.patch Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH] ALSA: usb-audio: Add prevent wakeup from s3 state trig by Lenovo ThinkCentre TI024Gen3 USB-audio
by penghao 19 Aug '20

19 Aug '20

TI024Gen3 USB-audio is controlled by TI024Gen3,when TI024Gens enter sleep mode, USB-audio will disconnect from USB bus port, wakup form s3 state Signed-off-by: penghao <penghao(a)uniontech.com> --- sound/usb/card.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/usb/card.c b/sound/usb/card.c index 696e788c5d31..6bdbb34009b3 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -658,6 +658,12 @@ static int usb_audio_probe(struct usb_interface *intf, } dev_set_drvdata(&dev->dev, chip); + /* + *ALSA: usb-audio: Add prevent wakeup from s3 state trig by Lenovo + *ThinkCentre TI024Gen3 usb-audio + */ + if ((usb_id->idVendor == 0x17ef) && (usb_id->idProduct == 0xa012)) + device_set_wakeup_enable(＆dev->dev, 0); /* * For devices with more than one control interface, we assume the -- 2.11.0

1 0

[PATCH v2 0/3] Fix Kconfig dependency issue with DMAENGINES selection
by Laurent Pinchart 19 Aug '20

19 Aug '20

Hello, This small series fixes a Kconfig dependency issue with the recently merged Xilixn DPSUB DRM/KMS driver. The fix is in patch 3/3, but requires a separate fixes in patches 1/3 and 2/3 to avoid circular dependencies: drivers/i2c/Kconfig:8:error: recursive dependency detected! drivers/i2c/Kconfig:8: symbol I2C is selected by FB_DDC drivers/video/fbdev/Kconfig:63: symbol FB_DDC depends on FB drivers/video/fbdev/Kconfig:12: symbol FB is selected by DRM_KMS_FB_HELPER drivers/gpu/drm/Kconfig:80: symbol DRM_KMS_FB_HELPER depends on DRM_KMS_HELPER drivers/gpu/drm/Kconfig:74: symbol DRM_KMS_HELPER is selected by DRM_ZYNQMP_DPSUB drivers/gpu/drm/xlnx/Kconfig:1: symbol DRM_ZYNQMP_DPSUB depends on DMA_ENGINE drivers/dma/Kconfig:44: symbol DMA_ENGINE depends on DMADEVICES drivers/dma/Kconfig:6: symbol DMADEVICES is selected by SND_SOC_SH4_SIU sound/soc/sh/Kconfig:30: symbol SND_SOC_SH4_SIU is selected by SND_SIU_MIGOR sound/soc/sh/Kconfig:60: symbol SND_SIU_MIGOR depends on I2C For a resolution refer to Documentation/kbuild/kconfig-language.rst subsection "Kconfig recursive dependency limitations" Due to the DPSUB driver being merged in v5.9, this is a candidate fix for v5.9 as well. 1/3 and 2/3 can be merged independently, 3/3 depends on the first two. What's the best course of action, can I merge this all in a single tree, or should the rapidio and ASoC patches be merged independently early in the -rc cycle, and the DRM patch later on top ? I don't expect conflicts (especially in 2/3 and 3/3), so merging the whole series in one go would be simpler in my opinion. Compared to v1, commit messages have been fixed to mention DMADEVICES instead of DMAENGINES. Laurent Pinchart (3): rapidio: Replace 'select' DMADEVICES 'with depends on' ASoC: sh: Replace 'select' DMADEVICES 'with depends on' drm: xlnx: dpsub: Fix DMADEVICES Kconfig dependency drivers/gpu/drm/xlnx/Kconfig | 1 + drivers/rapidio/Kconfig | 2 +- sound/soc/sh/Kconfig | 2 +- 3 files changed, 3 insertions(+), 2 deletions(-) -- Regards, Laurent Pinchart

3 10

[RFC PATCH v1 1/2] ASoC: rockchip-spdif: add description for rk3308
by Johan Jonker 18 Aug '20

18 Aug '20

A test with the command below shows that the compatible string "rockchip,rk3308-spdif", "rockchip,rk3328-spdif" is already in use, but is not added to a document. The current fallback string "rockchip,rk3328-spdif" points to a data set enum RK_SPDIF_RK3366 in rockchip_spdif.c that is not used both in the mainline as in the manufacturer kernel. (Of the enum only RK_SPDIF_RK3288 is used.) So if the properties don't change we might as well use the first SoC in line as fallback string and add the description for rk3308 as: "rockchip,rk3308-spdif", "rockchip,rk3066-spdif" make ARCH=arm64 dtbs_check DT_SCHEMA_FILES=Documentation/devicetree/bindings/sound/rockchip-spdif.yaml Signed-off-by: Johan Jonker <jbx6244(a)gmail.com> --- enum rk_spdif_type { RK_SPDIF_RK3066, RK_SPDIF_RK3188, RK_SPDIF_RK3288, RK_SPDIF_RK3366, }; static const struct of_device_id rk_spdif_match[] = { { .compatible = "rockchip,rk3066-spdif", .data = (void *)RK_SPDIF_RK3066 }, { .compatible = "rockchip,rk3188-spdif", .data = (void *)RK_SPDIF_RK3188 }, { .compatible = "rockchip,rk3228-spdif", .data = (void *)RK_SPDIF_RK3366 }, { .compatible = "rockchip,rk3288-spdif", .data = (void *)RK_SPDIF_RK3288 }, { .compatible = "rockchip,rk3328-spdif", .data = (void *)RK_SPDIF_RK3366 }, { .compatible = "rockchip,rk3366-spdif", .data = (void *)RK_SPDIF_RK3366 }, { .compatible = "rockchip,rk3368-spdif", .data = (void *)RK_SPDIF_RK3366 }, { .compatible = "rockchip,rk3399-spdif", .data = (void *)RK_SPDIF_RK3366 }, {}, }; --- Documentation/devicetree/bindings/sound/rockchip-spdif.yaml | 1 + 1 file changed, 1 insertion(+) diff --git a/Documentation/devicetree/bindings/sound/rockchip-spdif.yaml b/Documentation/devicetree/bindings/sound/rockchip-spdif.yaml index c46715265..26e986097 100644 --- a/Documentation/devicetree/bindings/sound/rockchip-spdif.yaml +++ b/Documentation/devicetree/bindings/sound/rockchip-spdif.yaml @@ -27,6 +27,7 @@ properties: - enum: - rockchip,rk3188-spdif - rockchip,rk3288-spdif + - rockchip,rk3308-spdif - const: rockchip,rk3066-spdif reg: -- 2.11.0

2 3

[PATCH] ASoC: hdac_hdmi: support 'ELD' mixer
by Brent Lu 18 Aug '20

18 Aug '20

Add an binary mixer 'ELD' to each HDMI PCM device so user space could read the ELD data of external HDMI display. Signed-off-by: Brent Lu <brent.lu(a)intel.com> --- sound/soc/codecs/hdac_hdmi.c | 139 +++++++++++++++++++++++++++++++++++ 1 file changed, 139 insertions(+) diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c index f26b77faed59..83656bbf22a7 100644 --- a/sound/soc/codecs/hdac_hdmi.c +++ b/sound/soc/codecs/hdac_hdmi.c @@ -9,6 +9,7 @@ * * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ */ + #include <linux/init.h> #include <linux/delay.h> #include <linux/module.h> @@ -107,6 +108,7 @@ struct hdac_hdmi_pcm { unsigned char chmap[8]; /* ALSA API channel-map */ struct mutex lock; int jack_event; + struct snd_kcontrol *eld_ctl; }; struct hdac_hdmi_dai_port_map { @@ -1248,6 +1250,7 @@ static void hdac_hdmi_present_sense(struct hdac_hdmi_pin *pin, struct hdac_hdmi_pcm *pcm; int size = 0; int port_id = -1; + bool eld_valid, eld_changed; if (!hdmi) return; @@ -1273,6 +1276,8 @@ static void hdac_hdmi_present_sense(struct hdac_hdmi_pin *pin, size = -EINVAL; } + eld_valid = port->eld.eld_valid; + if (size > 0) { port->eld.eld_valid = true; port->eld.eld_size = size; @@ -1281,6 +1286,8 @@ static void hdac_hdmi_present_sense(struct hdac_hdmi_pin *pin, port->eld.eld_size = 0; } + eld_changed = (eld_valid != port->eld.eld_valid); + pcm = hdac_hdmi_get_pcm(hdev, port); if (!port->eld.monitor_present || !port->eld.eld_valid) { @@ -1313,6 +1320,12 @@ static void hdac_hdmi_present_sense(struct hdac_hdmi_pin *pin, } mutex_unlock(&hdmi->pin_mutex); + + if (eld_changed && pcm) + snd_ctl_notify(hdmi->card, + SNDRV_CTL_EVENT_MASK_VALUE | + SNDRV_CTL_EVENT_MASK_INFO, + &pcm->eld_ctl->id); } static int hdac_hdmi_add_ports(struct hdac_device *hdev, @@ -1411,6 +1424,122 @@ static void hdac_hdmi_skl_enable_dp12(struct hdac_device *hdev) } +static int hdac_hdmi_eld_ctl_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct hdac_hdmi_priv *hdmi = snd_soc_component_get_drvdata(component); + struct hdac_hdmi_pcm *pcm; + struct hdac_hdmi_port *port; + struct hdac_hdmi_eld *eld; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_BYTES; + uinfo->count = 0; + + pcm = get_hdmi_pcm_from_id(hdmi, kcontrol->id.device); + if (!pcm) { + dev_dbg(component->dev, "%s: no pcm, device %d\n", __func__, + kcontrol->id.device); + return 0; + } + + if (list_empty(&pcm->port_list)) { + dev_dbg(component->dev, "%s: empty port list, device %d\n", + __func__, kcontrol->id.device); + return 0; + } + + mutex_lock(&hdmi->pin_mutex); + + list_for_each_entry(port, &pcm->port_list, head) { + eld = &port->eld; + + if (eld->eld_valid) { + uinfo->count = eld->eld_size; + break; + } + } + + mutex_unlock(&hdmi->pin_mutex); + + return 0; +} + +static int hdac_hdmi_eld_ctl_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct hdac_hdmi_priv *hdmi = snd_soc_component_get_drvdata(component); + struct hdac_hdmi_pcm *pcm; + struct hdac_hdmi_port *port; + struct hdac_hdmi_eld *eld; + + memset(ucontrol->value.bytes.data, 0, ARRAY_SIZE(ucontrol->value.bytes.data)); + + pcm = get_hdmi_pcm_from_id(hdmi, kcontrol->id.device); + if (!pcm) { + dev_dbg(component->dev, "%s: no pcm, device %d\n", __func__, + kcontrol->id.device); + return 0; + } + + if (list_empty(&pcm->port_list)) { + dev_dbg(component->dev, "%s: empty port list, device %d\n", + __func__, kcontrol->id.device); + return 0; + } + + mutex_lock(&hdmi->pin_mutex); + + list_for_each_entry(port, &pcm->port_list, head) { + eld = &port->eld; + + if (!eld->eld_valid) + continue; + + if (eld->eld_size > ARRAY_SIZE(ucontrol->value.bytes.data) || + eld->eld_size > ELD_MAX_SIZE) { + mutex_unlock(&hdmi->pin_mutex); + + dev_err(component->dev, "%s: buffer too small, device %d eld_size %d\n", + __func__, kcontrol->id.device, eld->eld_size); + snd_BUG(); + return -EINVAL; + } + + memcpy(ucontrol->value.bytes.data, eld->eld_buffer, + eld->eld_size); + break; + } + + mutex_unlock(&hdmi->pin_mutex); + + return 0; +} + +static int hdac_hdmi_create_eld_ctl(struct snd_soc_component *component, struct hdac_hdmi_pcm *pcm) +{ + struct snd_kcontrol *kctl; + struct snd_kcontrol_new hdmi_eld_ctl = { + .access = SNDRV_CTL_ELEM_ACCESS_READ | + SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .name = "ELD", + .info = hdac_hdmi_eld_ctl_info, + .get = hdac_hdmi_eld_ctl_get, + .device = pcm->pcm_id, + }; + + /* add ELD ctl with the device number corresponding to the PCM stream */ + kctl = snd_ctl_new1(&hdmi_eld_ctl, component); + if (!kctl) + return -ENOMEM; + + pcm->eld_ctl = kctl; + + return snd_ctl_add(component->card->snd_card, kctl); +} + static const struct snd_soc_dai_ops hdmi_dai_ops = { .startup = hdac_hdmi_pcm_open, .shutdown = hdac_hdmi_pcm_close, @@ -1784,6 +1913,16 @@ int hdac_hdmi_jack_init(struct snd_soc_dai *dai, int device, } } + /* add control for ELD Bytes */ + err = hdac_hdmi_create_eld_ctl(component, pcm); + if (err < 0) { + dev_err(&hdev->dev, + "eld control add failed with err: %d for pcm: %d\n", + err, device); + kfree(pcm); + return err; + } + list_add_tail(&pcm->head, &hdmi->pcm_list); return 0; -- 2.17.1

5 12

[PATCH] ASoC: wm_adsp: Update naming in error handling
by Charles Keepax 18 Aug '20

18 Aug '20

It seems the datasheet has never used the word slave for this error status bit and has always used the term address error. So update the driver to match the datasheets and also in the process align a bit better with avoiding the use of such words where possible. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- sound/soc/codecs/wm_adsp.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 410cca57da52d..4ff0c6cfa32de 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -199,7 +199,7 @@ #define ADSP2_PMEM_ERR_ADDR_XMEM_ERR_ADDR 0x7C #define ADSP2_REGION_LOCK_ERR_MASK 0x8000 -#define ADSP2_SLAVE_ERR_MASK 0x4000 +#define ADSP2_ADDR_ERR_MASK 0x4000 #define ADSP2_WDT_TIMEOUT_STS_MASK 0x2000 #define ADSP2_CTRL_ERR_PAUSE_ENA 0x0002 #define ADSP2_CTRL_ERR_EINT 0x0001 @@ -4364,9 +4364,9 @@ irqreturn_t wm_adsp2_bus_error(int irq, void *data) wm_adsp_fatal_error(dsp); } - if (val & (ADSP2_SLAVE_ERR_MASK | ADSP2_REGION_LOCK_ERR_MASK)) { - if (val & ADSP2_SLAVE_ERR_MASK) - adsp_err(dsp, "bus error: slave error\n"); + if (val & (ADSP2_ADDR_ERR_MASK | ADSP2_REGION_LOCK_ERR_MASK)) { + if (val & ADSP2_ADDR_ERR_MASK) + adsp_err(dsp, "bus error: address error\n"); else adsp_err(dsp, "bus error: region lock error\n"); -- 2.11.0

2 1

[PATCH 0/3] ASoC: da7219: Reorganise device/codec level probe/remove
by Adam Thomson 18 Aug '20

18 Aug '20

This patch set reorganises and fixes device and codec level probe/remove handling within the driver, to allow clean probe and remove at the codec level. This set relates to an issue raised by Yong Zhi where a codec level re-probe would fail due to clks still being registered from the previous instantiation. In addition some improvements around regulator handling and soft reset have also been included. Adam Thomson (3): ASoC: da7219: Move required devm_* allocations to device level code ASoC: da7219: Move soft reset handling to codec level probe ASoC: da7219: Fix clock handling around codec level probe sound/soc/codecs/da7219-aad.c | 85 +++++--- sound/soc/codecs/da7219-aad.h | 3 + sound/soc/codecs/da7219.c | 493 +++++++++++++++++++++++------------------- sound/soc/codecs/da7219.h | 1 + 4 files changed, 328 insertions(+), 254 deletions(-) -- 1.9.1

3 6

[PATCH 0/3] ASoC: rt5682: Use clk APIs better
by Stephen Boyd 18 Aug '20

18 Aug '20

This patch series drops a printk message down to dev_dbg() because it was noisy and then migrates this driver to use clk_hw based APIs instead of clk based APIs because this device is a clk provider, not a clk consumer. I've only lightly tested the last two patches but I don't have all combinations of clks for this device. Cc: Cheng-Yi Chiang <cychiang(a)chromium.org> Cc: Shuming Fan <shumingf(a)realtek.com> Stephen Boyd (3): ASoC: rt5682: Use dev_dbg() in rt5682_clk_check() ASoC: rt5682: Drop usage of __clk_get_name() ASoC: rt5682: Use clk_hw based APIs for registration sound/soc/codecs/rt5682.c | 73 ++++++++++++--------------------------- sound/soc/codecs/rt5682.h | 2 -- 2 files changed, 23 insertions(+), 52 deletions(-) Based on the last patch to this driver in linux-next. base-commit: 6301adf942a31bed65e026a554e5bd55d9e731e1 -- Sent by a computer, using git, on the internet

2 4

[PATCH RFC] ASoC: wm8994: Avoid attempts to read unreadable registers
by Sylwester Nawrocki 18 Aug '20

18 Aug '20

The driver supports WM1811, WM8994, WM8958 devices but according to documentation and the regmap definitions the WM8958_DSP2_* registers are only available on WM8958. In current code these registers are being accessed as if they were available on all the three chips. When starting playback on WM1811 CODEC multiple errors like: "wm8994-codec wm8994-codec: ASoC: error at soc_component_read_no_lock on wm8994-codec: -5" can be seen, which is caused by attempts to read an unavailable WM8958_DSP2_PROGRAM register. The issue has been uncovered by recent commit "e2329ee ASoC: soc-component: add soc_component_err()". This patch adds a check in wm8958_aif_ev() callback so the DSP2 handling is only done for WM8958. Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> --- sound/soc/codecs/wm8958-dsp2.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/codecs/wm8958-dsp2.c b/sound/soc/codecs/wm8958-dsp2.c index 68a3b48..3bce9a1 100644 --- a/sound/soc/codecs/wm8958-dsp2.c +++ b/sound/soc/codecs/wm8958-dsp2.c @@ -412,8 +412,12 @@ int wm8958_aif_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); + struct wm8994 *control = dev_get_drvdata(component->dev->parent); int i; + if (control->type != WM8958) + return 0; + switch (event) { case SND_SOC_DAPM_POST_PMU: case SND_SOC_DAPM_PRE_PMU: -- 2.7.4

3 2

[PATCH] sound: remove duplicate "the the" phrase in Kconfig text
by Colin King 18 Aug '20

18 Aug '20

From: Colin Ian King <colin.king(a)canonical.com> There are a couple of occurrences of "the the" in the Kconfig text. Fix these. Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/xilinx/Kconfig | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/xilinx/Kconfig b/sound/soc/xilinx/Kconfig index 1d3586b68db7..5bd2730aab76 100644 --- a/sound/soc/xilinx/Kconfig +++ b/sound/soc/xilinx/Kconfig @@ -9,14 +9,14 @@ config SND_SOC_XILINX_I2S encapsulates PCM in AES format and sends AES data. config SND_SOC_XILINX_AUDIO_FORMATTER - tristate "Audio support for the the Xilinx audio formatter" + tristate "Audio support for the Xilinx audio formatter" help Select this option to enable Xilinx audio formatter support. This provides DMA platform device support for audio functionality. config SND_SOC_XILINX_SPDIF - tristate "Audio support for the the Xilinx SPDIF" + tristate "Audio support for the Xilinx SPDIF" help Select this option to enable Xilinx SPDIF Audio. This provides playback and capture of SPDIF audio in -- 2.27.0

3 2

[PATCH v3 00/10] ASoC: qdsp6: add gapless compressed audio support
by Srinivas Kandagatla 18 Aug '20

18 Aug '20

This patchset adds gapless compressed audio support on q6asm. Gapless on q6asm is implemented using 2 streams in a single q6asm session. First few patches such as stream id per each command, gapless flags and silence meta data are for preparedness for adding gapless support. Last patch implements copy callback to allow finer control over buffer offsets, specially in partial drain cases. This patchset is tested on RB3 aka DB845c platform. This patchset as it is will support gapless however QDSP can also support switching decoders on a single stream. Patches to support such feature are send in different patchset which involves adding generic interfaces. Thanks, srini Changes since v2:(mostly suggested by Pierre) - removed unnessary kernel style comments, - moved TIMESTAMP flag to respective patch. - move preparatory code from gapless support patch to new one. - fix subject prefix of one patch. - add comments to clarify valid stream_ids Srinivas Kandagatla (10): ASoC: q6asm: rename misleading session id variable ASoC: q6asm: make commands specific to streams ASoC: q6asm: use flags directly from q6asm-dai ASoC: q6asm: add length to write command token ASoC: q6asm: add support to remove intial and trailing silence ASoC: q6asm: add support to gapless flag in q6asm open ASoC: q6asm-dai: add next track metadata support ASoC: q6asm-dai: prepare set params to accept profile change ASoC: q6asm-dai: add gapless support ASoC: q6asm-dai: add support to copy callback sound/soc/qcom/qdsp6/q6asm-dai.c | 414 +++++++++++++++++++++++-------- sound/soc/qcom/qdsp6/q6asm.c | 169 +++++++++---- sound/soc/qcom/qdsp6/q6asm.h | 49 ++-- 3 files changed, 469 insertions(+), 163 deletions(-) -- 2.21.0

4 13

[PATCH] sound/soc/intel: Fix spelling mistake "cant" --> "can't"
by Youling Tang 18 Aug '20

18 Aug '20

There is some spelling mistakes in a dev_err message. Fix it. Signed-off-by: Youling Tang <tangyouling(a)loongson.cn> --- sound/soc/intel/common/sst-firmware.c | 4 ++-- sound/soc/intel/haswell/sst-haswell-ipc.c | 6 +++--- 2 files changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/intel/common/sst-firmware.c b/sound/soc/intel/common/sst-firmware.c index 0594f89..1189ec3 100644 --- a/sound/soc/intel/common/sst-firmware.c +++ b/sound/soc/intel/common/sst-firmware.c @@ -130,7 +130,7 @@ static void block_list_remove(struct sst_dsp *dsp, err = block->ops->disable(block); if (err < 0) dev_err(dsp->dev, - "error: cant disable block %d:%d\n", + "error: can't disable block %d:%d\n", block->type, block->index); } } @@ -158,7 +158,7 @@ static int block_list_prepare(struct sst_dsp *dsp, ret = block->ops->enable(block); if (ret < 0) { dev_err(dsp->dev, - "error: cant disable block %d:%d\n", + "error: can't disable block %d:%d\n", block->type, block->index); goto err; } diff --git a/sound/soc/intel/haswell/sst-haswell-ipc.c b/sound/soc/intel/haswell/sst-haswell-ipc.c index 0ff89ea..773688b 100644 --- a/sound/soc/intel/haswell/sst-haswell-ipc.c +++ b/sound/soc/intel/haswell/sst-haswell-ipc.c @@ -1507,7 +1507,7 @@ static int sst_hsw_dx_state_dump(struct sst_hsw *hsw) ret = sst_dsp_dma_get_channel(sst, 0); if (ret < 0) { - dev_err(hsw->dev, "error: cant allocate dma channel %d\n", ret); + dev_err(hsw->dev, "error: can't allocate dma channel %d\n", ret); return ret; } @@ -1587,7 +1587,7 @@ int sst_hsw_dsp_load(struct sst_hsw *hsw) ret = sst_dsp_dma_get_channel(dsp, 0); if (ret < 0) { - dev_err(hsw->dev, "error: cant allocate dma channel %d\n", ret); + dev_err(hsw->dev, "error: can't allocate dma channel %d\n", ret); return ret; } @@ -1616,7 +1616,7 @@ static int sst_hsw_dsp_restore(struct sst_hsw *hsw) ret = sst_dsp_dma_get_channel(dsp, 0); if (ret < 0) { - dev_err(hsw->dev, "error: cant allocate dma channel %d\n", ret); + dev_err(hsw->dev, "error: can't allocate dma channel %d\n", ret); return ret; } -- 2.1.0

2 1

[PATCH] ASoC: wm8962: Export DAC/ADC monomix switches
by Sebastian Krzyszkowiak 18 Aug '20

18 Aug '20

This allows solutions like ALSA UCM to utilize hardware mono downmix for cases where mono output to a single speaker is desired only in specific situations (like on a mobile phone). Signed-off-by: Sebastian Krzyszkowiak <sebastian.krzyszkowiak(a)puri.sm> --- sound/soc/codecs/wm8962.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 317916cb4e27..d36aa24da7e0 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -1702,6 +1702,8 @@ SOC_DOUBLE_R_TLV("Digital Playback Volume", WM8962_LEFT_DAC_VOLUME, SOC_SINGLE("DAC High Performance Switch", WM8962_ADC_DAC_CONTROL_2, 0, 1, 0), SOC_SINGLE("DAC L/R Swap Switch", WM8962_AUDIO_INTERFACE_0, 5, 1, 0), SOC_SINGLE("ADC L/R Swap Switch", WM8962_AUDIO_INTERFACE_0, 8, 1, 0), +SOC_SINGLE("DAC Monomix Switch", WM8962_DAC_DSP_MIXING_1, WM8962_DAC_MONOMIX_SHIFT, 1, 0), +SOC_SINGLE("ADC Monomix Switch", WM8962_THREED1, WM8962_ADC_MONOMIX_SHIFT, 1, 0), SOC_SINGLE("ADC High Performance Switch", WM8962_ADDITIONAL_CONTROL_1, 5, 1, 0), -- 2.28.0

3 2

[PATCH v2 0/8] ASoC: sun50i-codec-analog: Cleanup and power management
by Samuel Holland 18 Aug '20

18 Aug '20

This series performs some minor cleanup on the driver for the analog codec in the Allwinner A64, and hooks up the existing mute switches to DAPM widgets, in order to provide improved power management. Changes since v1: - Collected Acked-by/Reviewed-by tags - Used SOC_MIXER_NAMED_CTL_ARRAY to avoid naming a widget "Earpiece" Samuel Holland (8): ASoC: sun50i-codec-analog: Fix duplicate use of ADC enable bits ASoC: sun50i-codec-analog: Gate the amplifier clock during suspend ASoC: sun50i-codec-analog: Group and sort mixer routes ASoC: sun50i-codec-analog: Make headphone routes stereo ASoC: sun50i-codec-analog: Enable DAPM for headphone switch ASoC: sun50i-codec-analog: Make line out routes stereo ASoC: sun50i-codec-analog: Enable DAPM for line out switch ASoC: sun50i-codec-analog: Enable DAPM for earpiece switch sound/soc/sunxi/sun50i-codec-analog.c | 176 ++++++++++++++++---------- 1 file changed, 111 insertions(+), 65 deletions(-) -- 2.26.2

2 9

[PATCH v3 0/3] refine and clean code for synchronous mode
by Shengjiu Wang 18 Aug '20

18 Aug '20

refine and clean code for synchronous mode Shengjiu Wang (3): ASoC: fsl_sai: Refine enable/disable TE/RE sequence in trigger() ASoC: fsl_sai: Drop TMR/RMR settings for synchronous mode ASoC: fsl_sai: Replace synchronous check with fsl_sai_dir_is_synced changes in v3: - Add reviewed-by Nicolin - refine the commit log. - Add one more patch #3 changes in v2: - Split the commit - refine the sequence in trigger stop sound/soc/fsl/fsl_sai.c | 173 +++++++++++++++++++++++----------------- 1 file changed, 102 insertions(+), 71 deletions(-) -- 2.27.0

3 5

[PATCH] ASoC: wm8994: Prevent access to invalid VU register bits on WM1811
by Sylwester Nawrocki 18 Aug '20

18 Aug '20

The ADC2 and DAC2 are not available on WM1811 device. This patch moves the ADC2, DAC2 VU bitfields to a separate array so we can skip accessing them and avoid unreadable register access on WM1811. This allows to get rid of warnings during boot like: wm8994-codec: ASoC: error at soc_component_read_no_lock on wm8994-codec: -5 Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> --- sound/soc/codecs/wm8994.c | 60 ++++++++++++++++++++++++++++----------- 1 file changed, 44 insertions(+), 16 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index a84ae879d37e..038be667c1a6 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -43,10 +43,12 @@ #define WM8994_NUM_DRC 3 #define WM8994_NUM_EQ 3 -static struct { +struct wm8994_reg_mask { unsigned int reg; unsigned int mask; -} wm8994_vu_bits[] = { +}; + +static struct wm8994_reg_mask wm8994_vu_bits[] = { { WM8994_LEFT_LINE_INPUT_1_2_VOLUME, WM8994_IN1_VU }, { WM8994_RIGHT_LINE_INPUT_1_2_VOLUME, WM8994_IN1_VU }, { WM8994_LEFT_LINE_INPUT_3_4_VOLUME, WM8994_IN2_VU }, @@ -60,14 +62,10 @@ static struct { { WM8994_AIF1_DAC1_LEFT_VOLUME, WM8994_AIF1DAC1_VU }, { WM8994_AIF1_DAC1_RIGHT_VOLUME, WM8994_AIF1DAC1_VU }, - { WM8994_AIF1_DAC2_LEFT_VOLUME, WM8994_AIF1DAC2_VU }, - { WM8994_AIF1_DAC2_RIGHT_VOLUME, WM8994_AIF1DAC2_VU }, { WM8994_AIF2_DAC_LEFT_VOLUME, WM8994_AIF2DAC_VU }, { WM8994_AIF2_DAC_RIGHT_VOLUME, WM8994_AIF2DAC_VU }, { WM8994_AIF1_ADC1_LEFT_VOLUME, WM8994_AIF1ADC1_VU }, { WM8994_AIF1_ADC1_RIGHT_VOLUME, WM8994_AIF1ADC1_VU }, - { WM8994_AIF1_ADC2_LEFT_VOLUME, WM8994_AIF1ADC2_VU }, - { WM8994_AIF1_ADC2_RIGHT_VOLUME, WM8994_AIF1ADC2_VU }, { WM8994_AIF2_ADC_LEFT_VOLUME, WM8994_AIF2ADC_VU }, { WM8994_AIF2_ADC_RIGHT_VOLUME, WM8994_AIF1ADC2_VU }, { WM8994_DAC1_LEFT_VOLUME, WM8994_DAC1_VU }, @@ -76,6 +74,14 @@ static struct { { WM8994_DAC2_RIGHT_VOLUME, WM8994_DAC2_VU }, }; +/* VU bitfields for ADC2, DAC2 not available on WM1811 */ +static struct wm8994_reg_mask wm8994_adc2_dac2_vu_bits[] = { + { WM8994_AIF1_DAC2_LEFT_VOLUME, WM8994_AIF1DAC2_VU }, + { WM8994_AIF1_DAC2_RIGHT_VOLUME, WM8994_AIF1DAC2_VU }, + { WM8994_AIF1_ADC2_LEFT_VOLUME, WM8994_AIF1ADC2_VU }, + { WM8994_AIF1_ADC2_RIGHT_VOLUME, WM8994_AIF1ADC2_VU }, +}; + static int wm8994_drc_base[] = { WM8994_AIF1_DRC1_1, WM8994_AIF1_DRC2_1, @@ -1030,6 +1036,26 @@ static bool wm8994_check_class_w_digital(struct snd_soc_component *component) return true; } +static void wm8994_update_vu_bits(struct snd_soc_component *component) +{ + struct wm8994_priv *wm8994 = snd_soc_component_get_drvdata(component); + struct wm8994 *control = wm8994->wm8994; + int i; + + for (i = 0; i < ARRAY_SIZE(wm8994_vu_bits); i++) + snd_soc_component_write(component, wm8994_vu_bits[i].reg, + snd_soc_component_read(component, + wm8994_vu_bits[i].reg)); + if (control->type == WM1811) + return; + + for (i = 0; i < ARRAY_SIZE(wm8994_adc2_dac2_vu_bits); i++) + snd_soc_component_write(component, + wm8994_adc2_dac2_vu_bits[i].reg, + snd_soc_component_read(component, + wm8994_adc2_dac2_vu_bits[i].reg)); +} + static int aif_mclk_set(struct snd_soc_component *component, int aif, bool enable) { struct wm8994_priv *wm8994 = snd_soc_component_get_drvdata(component); @@ -1076,7 +1102,7 @@ static int aif1clk_ev(struct snd_soc_dapm_widget *w, struct wm8994_priv *wm8994 = snd_soc_component_get_drvdata(component); struct wm8994 *control = wm8994->wm8994; int mask = WM8994_AIF1DAC1L_ENA | WM8994_AIF1DAC1R_ENA; - int ret, i; + int ret; int dac; int adc; int val; @@ -1144,10 +1170,7 @@ static int aif1clk_ev(struct snd_soc_dapm_widget *w, break; case SND_SOC_DAPM_POST_PMU: - for (i = 0; i < ARRAY_SIZE(wm8994_vu_bits); i++) - snd_soc_component_write(component, wm8994_vu_bits[i].reg, - snd_soc_component_read(component, - wm8994_vu_bits[i].reg)); + wm8994_update_vu_bits(component); break; case SND_SOC_DAPM_PRE_PMD: @@ -1181,7 +1204,7 @@ static int aif2clk_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); - int ret, i; + int ret; int dac; int adc; int val; @@ -1237,10 +1260,7 @@ static int aif2clk_ev(struct snd_soc_dapm_widget *w, break; case SND_SOC_DAPM_POST_PMU: - for (i = 0; i < ARRAY_SIZE(wm8994_vu_bits); i++) - snd_soc_component_write(component, wm8994_vu_bits[i].reg, - snd_soc_component_read(component, - wm8994_vu_bits[i].reg)); + wm8994_update_vu_bits(component); break; case SND_SOC_DAPM_PRE_PMD: @@ -4346,6 +4366,14 @@ static int wm8994_component_probe(struct snd_soc_component *component) wm8994_vu_bits[i].mask, wm8994_vu_bits[i].mask); + if (control->type != WM1811) { + for (i = 0; i < ARRAY_SIZE(wm8994_adc2_dac2_vu_bits); i++) + snd_soc_component_update_bits(component, + wm8994_adc2_dac2_vu_bits[i].reg, + wm8994_adc2_dac2_vu_bits[i].mask, + wm8994_adc2_dac2_vu_bits[i].mask); + } + /* Set the low bit of the 3D stereo depth so TLV matches */ snd_soc_component_update_bits(component, WM8994_AIF1_DAC1_FILTERS_2, 1 << WM8994_AIF1DAC1_3D_GAIN_SHIFT, -- 2.17.1

3 2

[PATCH v2 0/3] Codec workaround
by Michael Sit Wei Hong 18 Aug '20

18 Aug '20

This patch series enables some features on the tlv3204 codec and also fixes some issues faced while testing v2: Fixed the build error from snd_soc_component_read32 v1: initial ASoC: codec: tlv3204: Codec workaround series Michael Sit Wei Hong (3): ASoC: codec: tlv3204: Enable 24 bit audio support ASoC: codec: tlv3204: Increased maximum supported channels ASoC: codec: tlv3204: Moving GPIO reset and add ADC reset sound/soc/codecs/tlv320aic32x4.c | 60 +++++++++++++++++++++++--------- 1 file changed, 44 insertions(+), 16 deletions(-) -- 2.17.1

3 5

[PATCH] ASoC: codecs: wm0010: use DECLARE_COMPLETION_ONSTACK() macro
by Pierre-Louis Bossart 18 Aug '20

18 Aug '20

Follow recommendation in Documentation/scheduler/completion.rst and use macro to declare local 'struct completion' Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/codecs/wm0010.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm0010.c b/sound/soc/codecs/wm0010.c index 2f2b2f5d55e4..28b4656c4e14 100644 --- a/sound/soc/codecs/wm0010.c +++ b/sound/soc/codecs/wm0010.c @@ -346,7 +346,7 @@ static int wm0010_firmware_load(const char *name, struct snd_soc_component *comp struct list_head xfer_list; struct wm0010_boot_xfer *xfer; int ret; - struct completion done; + DECLARE_COMPLETION_ONSTACK(done); const struct firmware *fw; const struct dfw_binrec *rec; const struct dfw_inforec *inforec; @@ -370,7 +370,6 @@ static int wm0010_firmware_load(const char *name, struct snd_soc_component *comp wm0010->boot_failed = false; if (WARN_ON(!list_empty(&xfer_list))) return -EINVAL; - init_completion(&done); /* First record should be INFO */ if (rec->command != DFW_CMD_INFO) { -- 2.25.1

3 2

[PATCH v2 0/2] This patch series is to enable multiple features on the Keembay Platform
by Michael Sit Wei Hong 18 Aug '20

18 Aug '20

v2: Change channel-max device tree property to use new compatible string v1: Initial patch series Michael Sit Wei Hong (2): ASoC: Intel: KMB: Enable TDM audio capture dt-bindings: sound: intel,keembay-i2s: Add new compatible string .../bindings/sound/intel,keembay-i2s.yaml | 1 + sound/soc/intel/keembay/kmb_platform.c | 145 +++++++++++++----- 2 files changed, 109 insertions(+), 37 deletions(-) -- 2.17.1

3 5

[PATCH] ASoC: msm8916-wcd-analog: fix register Interrupt offset
by Srinivas Kandagatla 18 Aug '20

18 Aug '20

For some reason interrupt set and clear register offsets are not set correctly. This patch corrects them! Fixes: 585e881e5b9e ("ASoC: codecs: Add msm8916-wcd analog codec") Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- sound/soc/codecs/msm8916-wcd-analog.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/msm8916-wcd-analog.c b/sound/soc/codecs/msm8916-wcd-analog.c index 4428c62e25cf..3ddd822240e3 100644 --- a/sound/soc/codecs/msm8916-wcd-analog.c +++ b/sound/soc/codecs/msm8916-wcd-analog.c @@ -19,8 +19,8 @@ #define CDC_D_REVISION1 (0xf000) #define CDC_D_PERPH_SUBTYPE (0xf005) -#define CDC_D_INT_EN_SET (0x015) -#define CDC_D_INT_EN_CLR (0x016) +#define CDC_D_INT_EN_SET (0xf015) +#define CDC_D_INT_EN_CLR (0xf016) #define MBHC_SWITCH_INT BIT(7) #define MBHC_MIC_ELECTRICAL_INS_REM_DET BIT(6) #define MBHC_BUTTON_PRESS_DET BIT(5) -- 2.21.0

3 2

[PATCH] [v2] ASoC: intel: Fix memleak in sst_media_open
by Dinghao Liu 18 Aug '20

18 Aug '20

When power_up_sst() fails, stream needs to be freed just like when try_module_get() fails. However, current code is returning directly and ends up leaking memory. Fixes: 0121327c1a68b ("ASoC: Intel: mfld-pcm: add control for powering up/down dsp") Signed-off-by: Dinghao Liu <dinghao.liu(a)zju.edu.cn> --- Changelog: v2: - Add a new label 'out_power_up' to unify code style. --- sound/soc/intel/atom/sst-mfld-platform-pcm.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/atom/sst-mfld-platform-pcm.c b/sound/soc/intel/atom/sst-mfld-platform-pcm.c index 49b9f18472bc..b1cac7abdc0a 100644 --- a/sound/soc/intel/atom/sst-mfld-platform-pcm.c +++ b/sound/soc/intel/atom/sst-mfld-platform-pcm.c @@ -331,7 +331,7 @@ static int sst_media_open(struct snd_pcm_substream *substream, ret_val = power_up_sst(stream); if (ret_val < 0) - return ret_val; + goto out_power_up; /* Make sure, that the period size is always even */ snd_pcm_hw_constraint_step(substream->runtime, 0, @@ -340,8 +340,9 @@ static int sst_media_open(struct snd_pcm_substream *substream, return snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS); out_ops: - kfree(stream); mutex_unlock(&sst_lock); +out_power_up: + kfree(stream); return ret_val; } -- 2.17.1

3 2

[PATCH] ASoC: codecs: delete repeated words in comments
by Randy Dunlap 18 Aug '20

18 Aug '20

Drop the repeated words {start, it, the} in comments. Signed-off-by: Randy Dunlap <rdunlap(a)infradead.org> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: alsa-devel(a)alsa-project.org --- sound/soc/codecs/nau8825.c | 2 +- sound/soc/codecs/tas5086.c | 2 +- sound/soc/codecs/wm9713.c | 2 +- 3 files changed, 3 insertions(+), 3 deletions(-) --- linux-next-20200807.orig/sound/soc/codecs/nau8825.c +++ linux-next-20200807/sound/soc/codecs/nau8825.c @@ -251,7 +251,7 @@ static const unsigned short logtable[256 * * Acquires the semaphore without jiffies. Try to acquire the semaphore * atomically. Returns 0 if the semaphore has been acquired successfully - * or 1 if it it cannot be acquired. + * or 1 if it cannot be acquired. */ static int nau8825_sema_acquire(struct nau8825 *nau8825, long timeout) { --- linux-next-20200807.orig/sound/soc/codecs/tas5086.c +++ linux-next-20200807/sound/soc/codecs/tas5086.c @@ -487,7 +487,7 @@ static int tas5086_init(struct device *d /* * If any of the channels is configured to start in Mid-Z mode, * configure 'part 1' of the PWM starts to use Mid-Z, and tell - * all configured mid-z channels to start start under 'part 1'. + * all configured mid-z channels to start under 'part 1'. */ if (priv->pwm_start_mid_z) regmap_write(priv->regmap, TAS5086_PWM_START, --- linux-next-20200807.orig/sound/soc/codecs/wm9713.c +++ linux-next-20200807/sound/soc/codecs/wm9713.c @@ -755,7 +755,7 @@ static void pll_factors(struct snd_soc_c u64 Kpart; unsigned int K, Ndiv, Nmod, target; - /* The the PLL output is always 98.304MHz. */ + /* The PLL output is always 98.304MHz. */ target = 98304000; /* If the input frequency is over 14.4MHz then scale it down. */

2 1

[PATCH 00/21] ASoC: Intel: fix cppcheck warnings
by Pierre-Louis Bossart 18 Aug '20

18 Aug '20

This patchset tries to reduce the number of warnings on those drivers, so that cppcheck can become a viable tool to detect issues (currently hundreds of reports). Most of the problems are related to unnecessary/redundant variable assignments, prototypes and one nice logical mistake resulting in an always-true condition. Pierre-Louis Bossart (21): ASoC: Intel: Atom: sst-atom-controls: remove redundant assignments ASoC: Intel: Atom: compress: remove redundant assignment ASoC: Intel: Atom: platform-pcm: remove redundant assignment ASoC: Intel: Atom: sst: remove useless NULL assignment ASoC: Intel: Atom: remove redundant initialization ASoC: Intel: Atom: sst_pvt: remove redundant initialization ASoC: Intel: Atom: platform-pcm: fix redundant return ASoC: Intel: Atom: remove useless assignment ASoC: Intel: Atom: sst_loader: remove always-true condition ASoC: Intel: Atom: sst_pvt: simplify return handling ASoC: Intel: Atom: (cosmetic) align parameters ASoC: Intel: Baytrail: (cosmetic) align function parameters ASoC: Intel: common: (cosmetic) align function parameters ASoC: Intel: haswell: (cosmetic) align function parameters ASoC: Intel: haswell-ipc: remove redundant assignments ASoC: Intel: Skylake: skl-nhlt: remove redundant initialization ASoC: Intel: Skylake: cldma: remove redundant initialization ASoC: Intel: Skylake: sst-utils: remove redundant assignment ASoC: Intel: Skylake: skl-topology: remove redundant assignments ASoC: Intel: Skylake: skl-topology: remove redundant assignment ASoC: Intel: Skylake: (cosmetic) align function parameters sound/soc/intel/atom/sst-atom-controls.c | 4 +-- .../intel/atom/sst-mfld-platform-compress.c | 3 +- sound/soc/intel/atom/sst-mfld-platform-pcm.c | 4 +-- sound/soc/intel/atom/sst-mfld-platform.h | 4 +-- sound/soc/intel/atom/sst/sst.c | 5 ++- sound/soc/intel/atom/sst/sst.h | 34 +++++++++---------- sound/soc/intel/atom/sst/sst_loader.c | 3 +- sound/soc/intel/atom/sst/sst_pvt.c | 4 +-- sound/soc/intel/baytrail/sst-baytrail-ipc.h | 3 +- sound/soc/intel/common/sst-dsp-priv.h | 2 +- sound/soc/intel/common/sst-dsp.h | 15 ++++---- sound/soc/intel/haswell/sst-haswell-ipc.c | 4 +-- sound/soc/intel/haswell/sst-haswell-ipc.h | 2 +- sound/soc/intel/skylake/cnl-sst-dsp.h | 4 +-- sound/soc/intel/skylake/skl-nhlt.c | 2 +- sound/soc/intel/skylake/skl-sst-cldma.c | 2 +- sound/soc/intel/skylake/skl-sst-ipc.h | 16 ++++----- sound/soc/intel/skylake/skl-sst-utils.c | 2 +- sound/soc/intel/skylake/skl-topology.c | 8 ++--- sound/soc/intel/skylake/skl-topology.h | 8 ++--- sound/soc/intel/skylake/skl.h | 2 +- 21 files changed, 63 insertions(+), 68 deletions(-) -- 2.25.1

2 22

[PATCH 1/2] ASoC: dt-bindings: ak4458: Add power supply property
by Shengjiu Wang 18 Aug '20

18 Aug '20

AVDD-supply is for Analog power supply DVDD-supply is for Digital power supply Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- Documentation/devicetree/bindings/sound/ak4458.txt | 2 ++ 1 file changed, 2 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/ak4458.txt b/Documentation/devicetree/bindings/sound/ak4458.txt index e5820235e0d5..8f6c84f21468 100644 --- a/Documentation/devicetree/bindings/sound/ak4458.txt +++ b/Documentation/devicetree/bindings/sound/ak4458.txt @@ -10,6 +10,8 @@ Required properties: Optional properties: - reset-gpios: A GPIO specifier for the power down & reset pin - mute-gpios: A GPIO specifier for the soft mute pin +- AVDD-supply: Analog power supply +- DVDD-supply: Digital power supply Example: -- 2.27.0

2 2

[PATCH] ASoC: various vendors: delete repeated words in comments
by Randy Dunlap 18 Aug '20

18 Aug '20

Drop the repeated words {related, we, is, the} in comments. Signed-off-by: Randy Dunlap <rdunlap(a)infradead.org> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: alsa-devel(a)alsa-project.org --- sound/soc/fsl/fsl_dma.c | 2 +- sound/soc/intel/skylake/skl-sst.c | 2 +- sound/soc/meson/axg-tdm-formatter.c | 2 +- sound/soc/sprd/sprd-pcm-compress.c | 2 +- sound/soc/sunxi/sun4i-codec.c | 2 +- sound/soc/ti/davinci-mcasp.c | 2 +- 6 files changed, 6 insertions(+), 6 deletions(-) --- linux-next-20200807.orig/sound/soc/fsl/fsl_dma.c +++ linux-next-20200807/sound/soc/fsl/fsl_dma.c @@ -154,7 +154,7 @@ static void fsl_dma_abort_stream(struct /** * fsl_dma_update_pointers - update LD pointers to point to the next period * - * As each period is completed, this function changes the the link + * As each period is completed, this function changes the link * descriptor pointers for that period to point to the next period. */ static void fsl_dma_update_pointers(struct fsl_dma_private *dma_private) --- linux-next-20200807.orig/sound/soc/intel/skylake/skl-sst.c +++ linux-next-20200807/sound/soc/intel/skylake/skl-sst.c @@ -354,7 +354,7 @@ static int skl_transfer_module(struct ss /* * if bytes_left > 0 then wait for BDL complete interrupt and * copy the next chunk till bytes_left is 0. if bytes_left is - * is zero, then wait for load module IPC reply + * zero, then wait for load module IPC reply */ while (bytes_left > 0) { curr_pos = size - bytes_left; --- linux-next-20200807.orig/sound/soc/meson/axg-tdm-formatter.c +++ linux-next-20200807/sound/soc/meson/axg-tdm-formatter.c @@ -398,7 +398,7 @@ void axg_tdm_stream_free(struct axg_tdm_ /* * If the list is not empty, it would mean that one of the formatter * widget is still powered and attached to the interface while we - * we are removing the TDM DAI. It should not be possible + * are removing the TDM DAI. It should not be possible */ WARN_ON(!list_empty(&ts->formatter_list)); mutex_destroy(&ts->lock); --- linux-next-20200807.orig/sound/soc/sprd/sprd-pcm-compress.c +++ linux-next-20200807/sound/soc/sprd/sprd-pcm-compress.c @@ -559,7 +559,7 @@ static int sprd_platform_compr_copy(stru } else { /* * If the data count is larger than the available spaces - * of the the stage 0 IRAM buffer, we should copy one + * of the stage 0 IRAM buffer, we should copy one * partial data to the stage 0 IRAM buffer, and copy * the left to the stage 1 DDR buffer. */ --- linux-next-20200807.orig/sound/soc/sunxi/sun4i-codec.c +++ linux-next-20200807/sound/soc/sunxi/sun4i-codec.c @@ -335,7 +335,7 @@ static int sun4i_codec_prepare_capture(s /* * FIXME: Undocumented in the datasheet, but - * Allwinner's code mentions that it is related + * Allwinner's code mentions that it is * related to microphone gain */ if (of_device_is_compatible(scodec->dev->of_node, --- linux-next-20200807.orig/sound/soc/ti/davinci-mcasp.c +++ linux-next-20200807/sound/soc/ti/davinci-mcasp.c @@ -633,7 +633,7 @@ static int __davinci_mcasp_set_clkdiv(st * right channels), so it has to be divided by number * of tdm-slots (for I2S - divided by 2). * Instead of storing this ratio, we calculate a new - * tdm_slot width by dividing the the ratio by the + * tdm_slot width by dividing the ratio by the * number of configured tdm slots. */ mcasp->slot_width = div / mcasp->tdm_slots;

2 1

[PATCH v2] ASoC: fsl-asoc-card: Get "extal" clock rate by clk_get_rate
by Shengjiu Wang 18 Aug '20

18 Aug '20

On some platform(.e.g. i.MX8QM MEK), the "extal" clock is different with the mclk of codec, then the clock rate is also different. So it is better to get clock rate of "extal" rate by clk_get_rate, don't reuse the clock rate of mclk. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- changes in v2 - add defer probe handler sound/soc/fsl/fsl-asoc-card.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index 52adedc03245..32f8f756e6bb 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -696,6 +696,17 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) goto asrc_fail; } } else if (of_node_name_eq(cpu_np, "esai")) { + struct clk *esai_clk = clk_get(&cpu_pdev->dev, "extal"); + + if (!IS_ERR(esai_clk)) { + priv->cpu_priv.sysclk_freq[TX] = clk_get_rate(esai_clk); + priv->cpu_priv.sysclk_freq[RX] = clk_get_rate(esai_clk); + clk_put(esai_clk); + } else if (PTR_ERR(esai_clk) == -EPROBE_DEFER) { + ret = -EPROBE_DEFER; + goto asrc_fail; + } + priv->cpu_priv.sysclk_id[1] = ESAI_HCKT_EXTAL; priv->cpu_priv.sysclk_id[0] = ESAI_HCKR_EXTAL; } else if (of_node_name_eq(cpu_np, "sai")) { -- 2.27.0

4 4

[PATCH] ASoC: SOF: delete repeated words in comments
by Randy Dunlap 18 Aug '20

18 Aug '20

Drop the repeated words {that, the} in comments. Signed-off-by: Randy Dunlap <rdunlap(a)infradead.org> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: alsa-devel(a)alsa-project.org --- sound/soc/sof/intel/cnl.c | 2 +- sound/soc/sof/sof-audio.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) --- linux-next-20200807.orig/sound/soc/sof/intel/cnl.c +++ linux-next-20200807/sound/soc/sof/intel/cnl.c @@ -202,7 +202,7 @@ static int cnl_ipc_send_msg(struct snd_s * IPCs are sent at a high-rate. mod_delayed_work() * modifies the timer if the work is pending. * Also, a new delayed work should not be queued after the - * the CTX_SAVE IPC, which is sent before the DSP enters D3. + * CTX_SAVE IPC, which is sent before the DSP enters D3. */ if (hdr->cmd != (SOF_IPC_GLB_PM_MSG | SOF_IPC_PM_CTX_SAVE)) mod_delayed_work(system_wq, &hdev->d0i3_work, --- linux-next-20200807.orig/sound/soc/sof/sof-audio.c +++ linux-next-20200807/sound/soc/sof/sof-audio.c @@ -29,7 +29,7 @@ bool snd_sof_dsp_only_d0i3_compatible_st continue; /* - * substream->runtime being not NULL indicates that + * substream->runtime being not NULL indicates * that the stream is open. No need to check the * stream state. */

3 2

[PATCH 1/3] ASoC: core: Add common helper to parse aux devs from device tree
by Stephan Gerhold 18 Aug '20

18 Aug '20

simple-card.c and meson-card-utils.c use pretty much the same helper function to parse auxiliary devices from the device tree. Make it easier for other drivers to parse these from the device tree as well by adding a shared helper function to soc-core.c. snd_soc_of_parse_aux_devs() is pretty much a copy of meson_card_add_aux_devices() from meson-card-utils.c with two minor changes: - Make property name configurable as parameter - Change dev_err() message slightly for consistency with other error messages in soc-core.c Signed-off-by: Stephan Gerhold <stephan(a)gerhold.net> --- I have another patch set that I will submit separately which makes use of this function to parse aux devs from the device tree within qcom_snd_parse_of(). This is preparation for that patch set. --- include/sound/soc.h | 1 + sound/soc/soc-core.c | 31 +++++++++++++++++++++++++++++++ 2 files changed, 32 insertions(+) diff --git a/include/sound/soc.h b/include/sound/soc.h index 5e3919ffb00c..a0918d159fd3 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1331,6 +1331,7 @@ void snd_soc_of_parse_audio_prefix(struct snd_soc_card *card, int snd_soc_of_parse_audio_routing(struct snd_soc_card *card, const char *propname); +int snd_soc_of_parse_aux_devs(struct snd_soc_card *card, const char *propname); unsigned int snd_soc_of_parse_daifmt(struct device_node *np, const char *prefix, struct device_node **bitclkmaster, diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 2fe1b2ec7c8f..bf46f410c8c6 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2827,6 +2827,37 @@ int snd_soc_of_parse_audio_routing(struct snd_soc_card *card, } EXPORT_SYMBOL_GPL(snd_soc_of_parse_audio_routing); +int snd_soc_of_parse_aux_devs(struct snd_soc_card *card, const char *propname) +{ + struct device_node *node = card->dev->of_node; + struct snd_soc_aux_dev *aux; + int num, i; + + num = of_count_phandle_with_args(node, propname, NULL); + if (num == -ENOENT) { + return 0; + } else if (num < 0) { + dev_err(card->dev, "ASOC: Property '%s' could not be read: %d\n", + propname, num); + return num; + } + + aux = devm_kcalloc(card->dev, num, sizeof(*aux), GFP_KERNEL); + if (!aux) + return -ENOMEM; + card->aux_dev = aux; + card->num_aux_devs = num; + + for_each_card_pre_auxs(card, i, aux) { + aux->dlc.of_node = of_parse_phandle(node, propname, i); + if (!aux->dlc.of_node) + return -EINVAL; + } + + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_of_parse_aux_devs); + unsigned int snd_soc_of_parse_daifmt(struct device_node *np, const char *prefix, struct device_node **bitclkmaster, -- 2.27.0

3 5

[PATCH 1/2] dt-bindings: tas2562: Remove tas2562 text file
by Dan Murphy 18 Aug '20

18 Aug '20

Remove the tas2562 text file as the tas2562.yaml is now available. Signed-off-by: Dan Murphy <dmurphy(a)ti.com> --- .../devicetree/bindings/sound/tas2562.txt | 37 ------------------- 1 file changed, 37 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/tas2562.txt diff --git a/Documentation/devicetree/bindings/sound/tas2562.txt b/Documentation/devicetree/bindings/sound/tas2562.txt deleted file mode 100644 index dc6d7362ded7..000000000000 --- a/Documentation/devicetree/bindings/sound/tas2562.txt +++ /dev/null @@ -1,37 +0,0 @@ -Texas Instruments TAS2562 Smart PA - -The TAS2562 is a mono, digital input Class-D audio amplifier optimized for -efficiently driving high peak power into small loudspeakers. -Integrated speaker voltage and current sense provides for -real time monitoring of loudspeaker behavior. - -Required properties: - - #address-cells - Should be <1>. - - #size-cells - Should be <0>. - - compatible: - Should contain "ti,tas2562", "ti,tas2563". - - reg: - The i2c address. Should be 0x4c, 0x4d, 0x4e or 0x4f. - - ti,imon-slot-no:- TDM TX current sense time slot. - - ti,vmon-slot-no:- TDM TX voltage sense time slot. This slot must always be - greater then ti,imon-slot-no. - -Optional properties: -- interrupt-parent: phandle to the interrupt controller which provides - the interrupt. -- interrupts: (GPIO) interrupt to which the chip is connected. -- shut-down-gpio: GPIO used to control the state of the device. - -Examples: -tas2562@4c { - #address-cells = <1>; - #size-cells = <0>; - compatible = "ti,tas2562"; - reg = <0x4c>; - - interrupt-parent = <&gpio1>; - interrupts = <14>; - - shut-down-gpio = <&gpio1 15 0>; - ti,imon-slot-no = <0>; - ti,vmon-slot-no = <1>; -}; - -- 2.28.0

2 2

[PATCH] ASoC: qcom: add a dedicated menuconfig
by Srinivas Kandagatla 18 Aug '20

18 Aug '20

Currently list of Qualcomm drivers is growing, so put them in to a proper menu so that it does not mix up with other ASOC configs in menuconfig. Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- sound/soc/qcom/Kconfig | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) diff --git a/sound/soc/qcom/Kconfig b/sound/soc/qcom/Kconfig index 5d6b2466a2f2..be6b8d0e2f70 100644 --- a/sound/soc/qcom/Kconfig +++ b/sound/soc/qcom/Kconfig @@ -1,11 +1,13 @@ # SPDX-License-Identifier: GPL-2.0-only -config SND_SOC_QCOM +menuconfig SND_SOC_QCOM tristate "ASoC support for QCOM platforms" depends on ARCH_QCOM || COMPILE_TEST help Say Y or M if you want to add support to use audio devices in Qualcomm Technologies SOC-based platforms. +if SND_SOC_QCOM + config SND_SOC_LPASS_CPU tristate select REGMAP_MMIO @@ -26,7 +28,6 @@ config SND_SOC_LPASS_APQ8016 config SND_SOC_STORM tristate "ASoC I2S support for Storm boards" - depends on SND_SOC_QCOM select SND_SOC_LPASS_IPQ806X select SND_SOC_MAX98357A help @@ -35,7 +36,6 @@ config SND_SOC_STORM config SND_SOC_APQ8016_SBC tristate "SoC Audio support for APQ8016 SBC platforms" - depends on SND_SOC_QCOM select SND_SOC_LPASS_APQ8016 select SND_SOC_QCOM_COMMON help @@ -110,3 +110,5 @@ config SND_SOC_SDM845 To add support for audio on Qualcomm Technologies Inc. SDM845 SoC-based systems. Say Y if you want to use audio device on this SoCs. + +endif #SND_SOC_QCOM -- 2.21.0

2 1

[PATCH v6 00/12] ASoC: qcom: Add support for SC7180 lpass variant
by Rohit kumar 18 Aug '20

18 Aug '20

This patch chain add audio support for SC7180 soc by doing the required modification in existing common lpass-cpu/lpass-platform driver. This also fixes some concurrency issue. This patch series is already tested by Srinivas on Dragon Board 410c. Changes since v5: - Fixed remove api in lpass-sc7180.c - Addressed comments by Rob in yaml Documentation. Ajit Pandey (4): ASoC: qcom: Add common array to initialize soc based core clocks ASoC: qcom: lpass-platform: Replace card->dev with component->dev include: dt-bindings: sound: Add sc7180-lpass bindings header ASoC: qcom: lpass-sc7180: Add platform driver for lpass audio Rohit kumar (8): ASoC: qcom: lpass-cpu: Move ahbix clk to platform specific function ASoC: qcom: lpass-platform: fix memory leak ASoC: qcom: lpass: Use regmap_field for i2sctl and dmactl registers ASoC: qcom: lpass-cpu: fix concurrency issue dt-bindings: sound: lpass-cpu: Add sc7180 lpass cpu node ASoC: qcom: lpass-cpu: Use platform_get_resource ASoC: qcom: lpass-platform: Use platform_get_irq dt-bindings: sound: lpass-cpu: Move to yaml format .../devicetree/bindings/sound/qcom,lpass-cpu.txt | 79 -------- .../devicetree/bindings/sound/qcom,lpass-cpu.yaml | 189 ++++++++++++++++++ include/dt-bindings/sound/sc7180-lpass.h | 10 + sound/soc/qcom/Kconfig | 5 + sound/soc/qcom/Makefile | 2 + sound/soc/qcom/lpass-apq8016.c | 86 ++++++-- sound/soc/qcom/lpass-cpu.c | 204 ++++++++++--------- sound/soc/qcom/lpass-ipq806x.c | 67 +++++++ sound/soc/qcom/lpass-lpaif-reg.h | 157 ++++++++------- sound/soc/qcom/lpass-platform.c | 155 +++++++++++---- sound/soc/qcom/lpass-sc7180.c | 216 +++++++++++++++++++++ sound/soc/qcom/lpass.h | 63 +++++- 12 files changed, 934 insertions(+), 299 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/qcom,lpass-cpu.txt create mode 100644 Documentation/devicetree/bindings/sound/qcom,lpass-cpu.yaml create mode 100644 include/dt-bindings/sound/sc7180-lpass.h create mode 100644 sound/soc/qcom/lpass-sc7180.c -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

3 15

[PATCH] Add documentation for snd_ctl_elem_list_*
by tanjeff 18 Aug '20

18 Aug '20

From 731895b67fcae743551989a5dd04382003e66560 Mon Sep 17 00:00:00 2001 From: "Tanjeff-N. Moos" <tanjeff(a)cccmz.de> Date: Fri, 14 Aug 2020 08:40:28 +0200 Subject: [PATCH] Add documentation for snd_ctl_elem_list_*. Signed-off-by: Tanjeff-N. Moos <tanjeff(a)cccmz.de> --- include/control.h | 81 +++++++++++++++++++++++++++++++++++++++++-- src/control/control.c | 80 ++++++++++++++++++++++++++++++++++++------ 2 files changed, 148 insertions(+), 13 deletions(-) diff --git a/include/control.h b/include/control.h index 02db72d4..9deec6f3 100644 --- a/include/control.h +++ b/include/control.h @@ -56,7 +56,75 @@ typedef struct _snd_ctl_card_info snd_ctl_card_info_t; /** CTL element identifier container */ typedef struct _snd_ctl_elem_id snd_ctl_elem_id_t; -/** CTL element identifier list container */ +/** CTL element list container + * + * This is a list of CTL elements. The list contains management + * information (e.g. how many elements the sound card has) as well as + * the element identifiers. All functions which operate on the list + * are named snd_ctl_elem_list_*(). + * + * \par Memory management + * + * There are two memory areas to deal with: The list container itself + * and the memory for the element identifiers. + * + * To manage the area for the list container, the following functions + * are used: + * + * - snd_ctl_elem_list_malloc() / snd_ctl_elem_list_free() to allocate + * and free memory on the heap, or + * - snd_ctl_elem_list_alloca() to allocate the memory on the + * stack. This memory is auto-released when the stack is unwound. + * + * To manage the space for the element identifiers, the + * snd_ctl_elem_list_alloc_space() and snd_ctl_elem_list_free_space() + * are used. Allocating the right amount of space can be achieved by + * first obtaining the number of elements and then calling + * snd_ctl_elem_list_alloc_space(): + * + * \code + * snd_ctl_elem_list_t* list; + * int count; + * + * // Initialise list + * snd_ctl_elem_list_malloc(&list); + * + * // Get number of elements + * snd_ctl_elem_list(ctl, list); + * count = snd_ctl_elem_list_get_count(list); + * + * // Allocate space for identifiers + * snd_ctl_elem_list_alloc_space(list, count); + * + * // Get identifiers + * snd_ctl_elem_list(ctl, list); // yes, this is same as above :) + * + * // Do something useful with the list... + * + * // Cleanup + * snd_ctl_elem_list_free_space(list); + * snd_ctl_elem_list_free(list); + * \endcode + * + * + * \par The Elements + * + * The elements in the list are accessed using an index. This index is + * the location in the list; Don't confuse it with the 'index' of the + * element identifier. For example: + * + * \code + * snd_ctl_elem_list_t list; + * unsigned int element_index; + * + * // Allocate space, fill list ... + * + * element_index = snd_ctl_elem_list_get_index(&list, 2); + * \endcode + * + * This will access the 3rd element in the list (index=2) and get the + * elements index from the driver (which might be 13, for example). + */ typedef struct _snd_ctl_elem_list snd_ctl_elem_list_t; /** CTL element info container */ @@ -354,11 +422,18 @@ void snd_ctl_event_copy(snd_ctl_event_t *dst, const snd_ctl_event_t *src); snd_ctl_event_type_t snd_ctl_event_get_type(const snd_ctl_event_t *obj); size_t snd_ctl_elem_list_sizeof(void); + /** \hideinitializer - * \brief allocate an invalid #snd_ctl_elem_list_t using standard alloca - * \param ptr returned pointer + * + * \brief Allocate a #snd_ctl_elem_list_t using standard alloca. + * + * The memory is allocated on the stack and will automatically be + * released when the stack unwinds (i.e. no free() is needed). + * + * \param ptr Pointer to allocated memory. */ #define snd_ctl_elem_list_alloca(ptr) __snd_alloca(ptr, snd_ctl_elem_list) + int snd_ctl_elem_list_malloc(snd_ctl_elem_list_t **ptr); void snd_ctl_elem_list_free(snd_ctl_elem_list_t *obj); void snd_ctl_elem_list_clear(snd_ctl_elem_list_t *obj); diff --git a/src/control/control.c b/src/control/control.c index e21e8f1d..1bcf1ab2 100644 --- a/src/control/control.c +++ b/src/control/control.c @@ -280,6 +280,21 @@ int snd_ctl_card_info(snd_ctl_t *ctl, snd_ctl_card_info_t *info) /** * \brief Get a list of element identifiers + * + * Before calling this function, memoru must be allocated using + * snd_ctl_elem_list_malloc(). + * + * This function obtains data from the sound card driver and puts it + * into the list. + * + * If there was space allocated for the element identifiers (using + * snd_ctl_elem_list_alloc_space()), information will be filled in. If + * too little space was allocated, only a part of the elements will be + * queried. If there was too much space allocated, some of it remains + * unused. Use snd_ctl_elem_list_get_count() and + * snd_ctl_elem_list_get_used() to obtain information about space + * usage. See #snd_ctl_elem_list_t to learn more. + * * \param ctl CTL handle * \param list CTL element identifiers list pointer * \return 0 on success otherwise a negative error code @@ -1508,9 +1523,14 @@ const char *snd_ctl_event_type_name(snd_ctl_event_type_t type) /** * \brief allocate space for CTL element identifiers list - * \param obj CTL element identifiers list - * \param entries Entries to allocate - * \return 0 on success otherwise a negative error code + * + * The space can be released with snd_ctl_elem_list_free_space(). + * + * \param obj CTL element identifiers list. + * \param entries How many entries to allocate. See + * #snd_ctl_elem_list_t to learn how to obtain + * this number in advance. + * \return 0 on success otherwise a negative error code. */ int snd_ctl_elem_list_alloc_space(snd_ctl_elem_list_t *obj, unsigned int entries) { @@ -1526,6 +1546,10 @@ int snd_ctl_elem_list_alloc_space(snd_ctl_elem_list_t *obj, unsigned int entries /** * \brief free previously allocated space for CTL element identifiers list + * + * Releases space previously allocated using + * snd_ctl_elem_list_alloc_space(). + * * \param obj CTL element identifiers list */ void snd_ctl_elem_list_free_space(snd_ctl_elem_list_t *obj) @@ -2016,7 +2040,7 @@ snd_ctl_event_type_t snd_ctl_event_get_type(const snd_ctl_event_t *obj) } /** - * \brief get size of #snd_ctl_elem_list_t + * \brief get size of #snd_ctl_elem_list_t. * \return size in bytes */ size_t snd_ctl_elem_list_sizeof() @@ -2025,7 +2049,10 @@ size_t snd_ctl_elem_list_sizeof() } /** - * \brief allocate an invalid #snd_ctl_elem_list_t using standard malloc + * \brief allocate a #snd_ctl_elem_list_t using standard malloc. + * + * The memory can be released using snd_ctl_elem_list_free(). + * * \param ptr returned pointer * \return 0 on success otherwise negative error code */ @@ -2039,7 +2066,15 @@ int snd_ctl_elem_list_malloc(snd_ctl_elem_list_t **ptr) } /** - * \brief frees a previously allocated #snd_ctl_elem_list_t + * \brief frees a previously allocated #snd_ctl_elem_list_t. + * + * Release memory previously allocated using + * snd_ctl_elem_list_malloc(). + * + * If you used snd_ctl_elem_list_alloc_space() on the list, you must + * use snd_ctl_elem_list_free_space() \em before calling this + * function. + * * \param obj pointer to object to free */ void snd_ctl_elem_list_free(snd_ctl_elem_list_t *obj) @@ -2048,7 +2083,15 @@ void snd_ctl_elem_list_free(snd_ctl_elem_list_t *obj) } /** - * \brief clear given #snd_ctl_elem_list_t object + * \brief Clear given #snd_ctl_elem_list_t object. + * + * This will make the stored identifiers inaccessible without freeing + * their space. + * + * \warning The element identifier space cannot be freed after calling + * this function. Therefore, snd_ctl_elem_list_free_space() + * must be called in advance. + * * \param obj pointer to object to clear */ void snd_ctl_elem_list_clear(snd_ctl_elem_list_t *obj) @@ -2057,7 +2100,11 @@ void snd_ctl_elem_list_clear(snd_ctl_elem_list_t *obj) } /** - * \brief copy one #snd_ctl_elem_list_t to another + * \brief copy one #snd_ctl_elem_list_t to another. + * + * This performs a shallow copy. That means the both lists will share + * the same space for the elements. The elements will not be copied. + * * \param dst pointer to destination * \param src pointer to source */ @@ -2080,6 +2127,12 @@ void snd_ctl_elem_list_set_offset(snd_ctl_elem_list_t *obj, unsigned int val) /** * \brief Get number of used entries in CTL element identifiers list + * + * This function returns how many entries are actually filled with + * useful information. + * + * See also snd_ctl_elem_list_get_count(). + * * \param obj CTL element identifier list * \return number of used entries */ @@ -2090,7 +2143,14 @@ unsigned int snd_ctl_elem_list_get_used(const snd_ctl_elem_list_t *obj) } /** - * \brief Get total count of elements present in CTL device (information present in every filled CTL element identifiers list) + * \brief Get total count of elements present in CTL device + * + * This function returns how many entries were allocated using + * snd_ctl_elem_list_alloc_space(). This information is present after + * snd_ctl_elem_list() was called. + * + * See also snd_ctl_elem_list_get_used(). + * * \param obj CTL element identifier list * \return total number of elements */ @@ -2140,7 +2200,7 @@ snd_ctl_elem_iface_t snd_ctl_elem_list_get_interface(const snd_ctl_elem_list_t * } /** - * \brief Get device part of CTL element identifier for an entry of a CTL element identifiers list + * \brief Get the device part of CTL element identifier for an entry of a CTL element identifiers list * \param obj CTL element identifier list * \param idx Index of entry * \return CTL element related device -- 2.17.1

2 1

[PATCH v2 00/12] soundwire: intel: add power management support
by Bard Liao 18 Aug '20

18 Aug '20

This series adds power management support for Intel soundwire links. Changes in v2: - Move "#include <linux/pm_runtime.h>" to the first required patch. - Fit debug log in single line. Bard Liao (1): soundwire: intel: reinitialize IP+DSP in .prepare(), but only when resuming Pierre-Louis Bossart (9): soundwire: intel: add pm_runtime support soundwire: intel: reset pm_runtime status during system resume soundwire: intel: fix race condition on system resume soundwire: intel: call helper to reset Slave states on resume soundwire: intel: pm_runtime idle scheduling soundwire: intel: add CLK_STOP_TEARDOWN for pm_runtime suspend soundwire: intel: add CLK_STOP_NOT_ALLOWED support soundwire: intel_init: handle power rail dependencies for clock stop mode soundwire: intel: support clock_stop mode without quirks Rander Wang (2): soundwire: intel: add CLK_STOP_BUS_RESET support soundwire: intel: refine runtime pm for SDW_INTEL_CLK_STOP_BUS_RESET drivers/soundwire/cadence_master.h | 4 + drivers/soundwire/intel.c | 363 ++++++++++++++++++++++++++++- drivers/soundwire/intel.h | 2 + drivers/soundwire/intel_init.c | 19 +- 4 files changed, 382 insertions(+), 6 deletions(-) -- 2.17.1

2 13

[PATCH 00/13] soundwire: intel: add power management support
by Bard Liao 18 Aug '20

18 Aug '20

This series adds power management support for Intel soundwire links. Bard Liao (1): soundwire: intel: reinitialize IP+DSP in .prepare(), but only when resuming Pierre-Louis Bossart (10): soundwire: intel: Add basic power management support soundwire: intel: add pm_runtime support soundwire: intel: reset pm_runtime status during system resume soundwire: intel: fix race condition on system resume soundwire: intel: call helper to reset Slave states on resume soundwire: intel: pm_runtime idle scheduling soundwire: intel: add CLK_STOP_TEARDOWN for pm_runtime suspend soundwire: intel: add CLK_STOP_NOT_ALLOWED support soundwire: intel_init: handle power rail dependencies for clock stop mode soundwire: intel: support clock_stop mode without quirks Rander Wang (2): soundwire: intel: add CLK_STOP_BUS_RESET support soundwire: intel: refine runtime pm for SDW_INTEL_CLK_STOP_BUS_RESET drivers/soundwire/cadence_master.h | 4 + drivers/soundwire/intel.c | 448 ++++++++++++++++++++++++++++- drivers/soundwire/intel.h | 2 + drivers/soundwire/intel_init.c | 19 +- 4 files changed, 465 insertions(+), 8 deletions(-) -- 2.17.1

3 21

[PATCH] ASoC: soc-core: use client_mutex for snd_soc_find_dai()
by Kuninori Morimoto 18 Aug '20

18 Aug '20

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> commit 25612477d20b52 ("ASoC: soc-dai: set dai_link dpcm_ flags with a helper") added snd_soc_dai_link_set_capabilities(). But it is using snd_soc_find_dai() (A) which is required client_mutex (B). And client_mutex is soc-core.c local. struct snd_soc_dai *snd_soc_find_dai(xxx) { ... (B) lockdep_assert_held(&client_mutex); ... } void snd_soc_dai_link_set_capabilities(xxx) { ... for_each_pcm_streams(direction) { ... for_each_link_cpus(dai_link, i, cpu) { (A) dai = snd_soc_find_dai(cpu); ... } ... for_each_link_codecs(dai_link, i, codec) { (A) dai = snd_soc_find_dai(codec); ... } } ... } Because of these background, we will get WARNING if .config has CONFIG_LOCKDEP. WARNING: CPU: 2 PID: 53 at sound/soc/soc-core.c:814 snd_soc_find_dai+0xf8/0x100 CPU: 2 PID: 53 Comm: kworker/2:1 Not tainted 5.7.0-rc1+ #328 Hardware name: Renesas H3ULCB Kingfisher board based on r8a77951 (DT) Workqueue: events deferred_probe_work_func pstate: 60000005 (nZCv daif -PAN -UAO) pc : snd_soc_find_dai+0xf8/0x100 lr : snd_soc_find_dai+0xf4/0x100 ... Call trace: snd_soc_find_dai+0xf8/0x100 snd_soc_dai_link_set_capabilities+0xa0/0x16c graph_dai_link_of_dpcm+0x390/0x3c0 graph_for_each_link+0x134/0x200 graph_probe+0x144/0x230 platform_drv_probe+0x5c/0xb0 really_probe+0xe4/0x430 driver_probe_device+0x60/0xf4 Many drivers are already calling snd_soc_find_dai() and/or snd_soc_dai_link_set_capabilities(). Thus, same thing can be happen on these drivers, too. > grep snd_soc_find_dai -rl sound/soc/* sound/soc/samsung/tm2_wm5110.c sound/soc/soc-core.c sound/soc/soc-dai.c sound/soc/soc-topology.c sound/soc/sof/topology.c > grep snd_soc_dai_link_set_capabilities -rl sound/soc/* sound/soc/generic/simple-card.c sound/soc/generic/audio-graph-card.c sound/soc/qcom/common.c sound/soc/soc-dai.c This patch renames current snd_soc_find_dai() to _no_mutex(), and new snd_soc_find_dai() uses client_mutex. Fixes: 25612477d20b52 ("ASoC: soc-dai: set dai_link dpcm_ flags with a helper") Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/soc-core.c | 17 ++++++++++++++--- 1 file changed, 14 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index fe23e936e2d1..6ea66b57b2c1 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -805,7 +805,7 @@ static struct snd_soc_component *soc_find_component( * * Return: pointer of DAI, or NULL if not found. */ -struct snd_soc_dai *snd_soc_find_dai( +static struct snd_soc_dai *snd_soc_find_dai_no_mutex( const struct snd_soc_dai_link_component *dlc) { struct snd_soc_component *component; @@ -829,6 +829,17 @@ struct snd_soc_dai *snd_soc_find_dai( return NULL; } + +struct snd_soc_dai *snd_soc_find_dai(const struct snd_soc_dai_link_component *dlc) +{ + struct snd_soc_dai *dai; + + mutex_lock(&client_mutex); + dai = snd_soc_find_dai_no_mutex(dlc); + mutex_unlock(&client_mutex); + + return dai; +} EXPORT_SYMBOL_GPL(snd_soc_find_dai); static int soc_dai_link_sanity_check(struct snd_soc_card *card, @@ -999,7 +1010,7 @@ int snd_soc_add_pcm_runtime(struct snd_soc_card *card, return -ENOMEM; for_each_link_cpus(dai_link, i, cpu) { - asoc_rtd_to_cpu(rtd, i) = snd_soc_find_dai(cpu); + asoc_rtd_to_cpu(rtd, i) = snd_soc_find_dai_no_mutex(cpu); if (!asoc_rtd_to_cpu(rtd, i)) { dev_info(card->dev, "ASoC: CPU DAI %s not registered\n", cpu->dai_name); @@ -1010,7 +1021,7 @@ int snd_soc_add_pcm_runtime(struct snd_soc_card *card, /* Find CODEC from registered CODECs */ for_each_link_codecs(dai_link, i, codec) { - asoc_rtd_to_codec(rtd, i) = snd_soc_find_dai(codec); + asoc_rtd_to_codec(rtd, i) = snd_soc_find_dai_no_mutex(codec); if (!asoc_rtd_to_codec(rtd, i)) { dev_info(card->dev, "ASoC: CODEC DAI %s not registered\n", codec->dai_name); -- 2.25.1

1 1

hdajackretask: Update kernel doc URL in README
by GitHub pull_request - opened 18 Aug '20

18 Aug '20

alsa-project/alsa-tools pull request #2 was opened from ferdnyc: As of THIS writing, the kernel documentation tree has been rearranged, and the "Hint strings" section now resides at the updated URL Request URL : https://github.com/alsa-project/alsa-tools/pull/2 Patch URL : https://github.com/alsa-project/alsa-tools/pull/2.patch Repository URL: https://github.com/alsa-project/alsa-tools

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[PATCH] dt-bindings: sound: Convert NXP spdif to json-schema
by Anson Huang 17 Aug '20

17 Aug '20

Convert the NXP SPDIF binding to DT schema format using json-schema. Signed-off-by: Anson Huang <Anson.Huang(a)nxp.com> --- .../devicetree/bindings/sound/fsl,spdif.txt | 68 ------------- .../devicetree/bindings/sound/fsl,spdif.yaml | 108 +++++++++++++++++++++ 2 files changed, 108 insertions(+), 68 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/fsl,spdif.txt create mode 100644 Documentation/devicetree/bindings/sound/fsl,spdif.yaml diff --git a/Documentation/devicetree/bindings/sound/fsl,spdif.txt b/Documentation/devicetree/bindings/sound/fsl,spdif.txt deleted file mode 100644 index e1365b0..0000000 --- a/Documentation/devicetree/bindings/sound/fsl,spdif.txt +++ /dev/null @@ -1,68 +0,0 @@ -Freescale Sony/Philips Digital Interface Format (S/PDIF) Controller - -The Freescale S/PDIF audio block is a stereo transceiver that allows the -processor to receive and transmit digital audio via an coaxial cable or -a fibre cable. - -Required properties: - - - compatible : Compatible list, should contain one of the following - compatibles: - "fsl,imx35-spdif", - "fsl,vf610-spdif", - "fsl,imx6sx-spdif", - - - reg : Offset and length of the register set for the device. - - - interrupts : Contains the spdif interrupt. - - - dmas : Generic dma devicetree binding as described in - Documentation/devicetree/bindings/dma/dma.txt. - - - dma-names : Two dmas have to be defined, "tx" and "rx". - - - clocks : Contains an entry for each entry in clock-names. - - - clock-names : Includes the following entries: - "core" The core clock of spdif controller. - "rxtx<0-7>" Clock source list for tx and rx clock. - This clock list should be identical to the source - list connecting to the spdif clock mux in "SPDIF - Transceiver Clock Diagram" of SoC reference manual. - It can also be referred to TxClk_Source bit of - register SPDIF_STC. - "spba" The spba clock is required when SPDIF is placed as a - bus slave of the Shared Peripheral Bus and when two - or more bus masters (CPU, DMA or DSP) try to access - it. This property is optional depending on the SoC - design. - -Optional properties: - - - big-endian : If this property is absent, the native endian mode - will be in use as default, or the big endian mode - will be in use for all the device registers. - -Example: - -spdif: spdif@2004000 { - compatible = "fsl,imx35-spdif"; - reg = <0x02004000 0x4000>; - interrupts = <0 52 0x04>; - dmas = <&sdma 14 18 0>, - <&sdma 15 18 0>; - dma-names = "rx", "tx"; - - clocks = <&clks 197>, <&clks 3>, - <&clks 197>, <&clks 107>, - <&clks 0>, <&clks 118>, - <&clks 62>, <&clks 139>, - <&clks 0>; - clock-names = "core", "rxtx0", - "rxtx1", "rxtx2", - "rxtx3", "rxtx4", - "rxtx5", "rxtx6", - "rxtx7"; - - big-endian; -}; diff --git a/Documentation/devicetree/bindings/sound/fsl,spdif.yaml b/Documentation/devicetree/bindings/sound/fsl,spdif.yaml new file mode 100644 index 0000000..819f37f --- /dev/null +++ b/Documentation/devicetree/bindings/sound/fsl,spdif.yaml @@ -0,0 +1,108 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/fsl,spdif.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Freescale Sony/Philips Digital Interface Format (S/PDIF) Controller + +maintainers: + - Shengjiu Wang <shengjiu.wang(a)nxp.com> + +description: | + The Freescale S/PDIF audio block is a stereo transceiver that allows the + processor to receive and transmit digital audio via an coaxial cable or + a fibre cable. + +properties: + compatible: + enum: + - fsl,imx35-spdif + - fsl,vf610-spdif + - fsl,imx6sx-spdif + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + dmas: + items: + - description: DMA controller phandle and request line for RX + - description: DMA controller phandle and request line for TX + + dma-names: + items: + - const: rx + - const: tx + + clocks: + items: + - description: The core clock of spdif controller. + - description: Clock for tx0 and rx0. + - description: Clock for tx1 and rx1. + - description: Clock for tx2 and rx2. + - description: Clock for tx3 and rx3. + - description: Clock for tx4 and rx4. + - description: Clock for tx5 and rx5. + - description: Clock for tx6 and rx6. + - description: Clock for tx7 and rx7. + - description: The spba clock is required when SPDIF is placed as a bus + slave of the Shared Peripheral Bus and when two or more bus masters + (CPU, DMA or DSP) try to access it. This property is optional depending + on the SoC design. + minItems: 9 + + clock-names: + items: + - const: core + - const: rxtx0 + - const: rxtx1 + - const: rxtx2 + - const: rxtx3 + - const: rxtx4 + - const: rxtx5 + - const: rxtx6 + - const: rxtx7 + - const: spba + minItems: 9 + + big-endian: + $ref: /schemas/types.yaml#/definitions/flag + description: | + If this property is absent, the native endian mode will be in use + as default, or the big endian mode will be in use for all the device + registers. Set this flag for HCDs with big endian descriptors and big + endian registers. + +required: + - compatible + - reg + - interrupts + - dmas + - dma-names + - clocks + - clock-names + +examples: + - | + spdif@2004000 { + compatible = "fsl,imx35-spdif"; + reg = <0x02004000 0x4000>; + interrupts = <0 52 0x04>; + dmas = <&sdma 14 18 0>, + <&sdma 15 18 0>; + dma-names = "rx", "tx"; + clocks = <&clks 197>, <&clks 3>, + <&clks 197>, <&clks 107>, + <&clks 0>, <&clks 118>, + <&clks 62>, <&clks 139>, + <&clks 0>; + clock-names = "core", "rxtx0", + "rxtx1", "rxtx2", + "rxtx3", "rxtx4", + "rxtx5", "rxtx6", + "rxtx7"; + big-endian; + }; -- 2.7.4

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Enabling DAPM for Dummy DAIs
by Sridharan, Ranjani 17 Aug '20

17 Aug '20

Hi Mark/Takashi, We're using the Dummy DAI and dummy component for the nocodec mode in the SOF driver and we noticed that there are no DAPM power status updates for the widgets because there are no output/input widgets in the Dummy component. Adding these widgets in the dummy component wont help either because we explicitly skip adding widgets for the Dummy component in the core. But it would be very useful for us to debug PM issue if we could have the DAPM power updates for the widgets in topology. Reverting the change that implemented the check for skipping the dummy component probe (ASoC: core: Don't probe the component which is dummy) would not be advisable because it will be used by multiple cards and will lead to the same issue again. More details about the issue can be found here: https://github.com/thesofproject/linux/issues/1987 What would your recommendation be to get around this problem in SOF? Thanks for your help. Thanks, Ranjani

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[PATCH v3 1/3] ASoC: samsung: Document DT bindings for Midas sound subsystem
by Sylwester Nawrocki 17 Aug '20

17 Aug '20

This patch adds documentation of DT biding for the Midas sound complex. Partially based on the *txt version by Simon Shields <simon(a)lineageos.org>. Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> --- Changes for v3: - change the sound-dai property type in cpu, codec nodes from phandle-array to phandle Changes for v2: - fix wrong *-gpios entries in the example, - mark sound-dai properties as required. --- .../bindings/sound/samsung,midas-audio.yaml | 108 +++++++++++++++++++++ 1 file changed, 108 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/samsung,midas-audio.yaml diff --git a/Documentation/devicetree/bindings/sound/samsung,midas-audio.yaml b/Documentation/devicetree/bindings/sound/samsung,midas-audio.yaml new file mode 100644 index 0000000..1c755de --- /dev/null +++ b/Documentation/devicetree/bindings/sound/samsung,midas-audio.yaml @@ -0,0 +1,108 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/samsung,midas-audio.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Samsung Midas audio complex with WM1811 codec + +maintainers: + - Sylwester Nawrocki <s.nawrocki(a)samsung.com> + +properties: + compatible: + const: samsung,midas-audio + + model: + $ref: /schemas/types.yaml#/definitions/string + description: The user-visible name of this sound complex. + + cpu: + type: object + properties: + sound-dai: + $ref: /schemas/types.yaml#/definitions/phandle + description: phandle to the I2S controller + required: + - sound-dai + + codec: + type: object + properties: + sound-dai: + $ref: /schemas/types.yaml#/definitions/phandle + description: phandle to the WM1811 CODEC + required: + - sound-dai + + samsung,audio-routing: + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + description: | + List of the connections between audio components; each entry is + a pair of strings, the first being the connection's sink, the second + being the connection's source; valid names for sources and sinks are + the WM1811's pins (as documented in its binding), and the jacks + on the board: HP, SPK, Main Mic, Sub Mic, Headset Mic. + + mic-bias-supply: + description: Supply for the micbias on the Main microphone + + submic-bias-supply: + description: Supply for the micbias on the Sub microphone + + fm-sel-gpios: + description: GPIO pin for FM selection + + lineout-sel-gpios: + description: GPIO pin for line out selection + +required: + - compatible + - model + - cpu + - codec + - samsung,audio-routing + - mic-bias-supply + - submic-bias-supply + +additionalProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + + sound { + compatible = "samsung,midas-audio"; + model = "Midas"; + + fm-sel-gpios = <&gpaa0 3 GPIO_ACTIVE_HIGH>; + + mic-bias-supply = <&mic_bias_reg>; + submic-bias-supply = <&submic_bias_reg>; + + samsung,audio-routing = + "HP", "HPOUT1L", + "HP", "HPOUT1R", + + "SPK", "SPKOUTLN", + "SPK", "SPKOUTLP", + "SPK", "SPKOUTRN", + "SPK", "SPKOUTRP", + + "RCV", "HPOUT2N", + "RCV", "HPOUT2P", + + "IN1LP", "Main Mic", + "IN1LN", "Main Mic", + "IN1RP", "Sub Mic", + "IN1LP", "Sub Mic"; + + cpu { + sound-dai = <&i2s0>; + }; + + codec { + sound-dai = <&wm1811>; + }; + + }; -- 2.7.4

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[PATCH] Add documentation for snd_ctl_elem_list_*
by Tanjeff Moos 17 Aug '20

17 Aug '20

I resend this patch, this time adding Jaroslav Kysela and Takashi Iwai to CC. I added doxygen documentation to alsa-lib. The patch was created using git format-patch. This is my first patch. Please let me know if I need to improve it. >From 731895b67fcae743551989a5dd04382003e66560 Mon Sep 17 00:00:00 2001 From: "Tanjeff-N. Moos" <tanjeff(a)cccmz.de> Date: Fri, 14 Aug 2020 08:40:28 +0200 Subject: [PATCH] Add documentation for snd_ctl_elem_list_*. --- include/control.h | 81 +++++++++++++++++++++++++++++++++++++++++-- src/control/control.c | 80 ++++++++++++++++++++++++++++++++++++------ 2 files changed, 148 insertions(+), 13 deletions(-) diff --git a/include/control.h b/include/control.h index 02db72d4..9deec6f3 100644 --- a/include/control.h +++ b/include/control.h @@ -56,7 +56,75 @@ typedef struct _snd_ctl_card_info snd_ctl_card_info_t; /** CTL element identifier container */ typedef struct _snd_ctl_elem_id snd_ctl_elem_id_t; -/** CTL element identifier list container */ +/** CTL element list container + * + * This is a list of CTL elements. The list contains management + * information (e.g. how many elements the sound card has) as well as + * the element identifiers. All functions which operate on the list + * are named snd_ctl_elem_list_*(). + * + * \par Memory management + * + * There are two memory areas to deal with: The list container itself + * and the memory for the element identifiers. + * + * To manage the area for the list container, the following functions + * are used: + * + * - snd_ctl_elem_list_malloc() / snd_ctl_elem_list_free() to allocate + * and free memory on the heap, or + * - snd_ctl_elem_list_alloca() to allocate the memory on the + * stack. This memory is auto-released when the stack is unwound. + * + * To manage the space for the element identifiers, the + * snd_ctl_elem_list_alloc_space() and snd_ctl_elem_list_free_space() + * are used. Allocating the right amount of space can be achieved by + * first obtaining the number of elements and then calling + * snd_ctl_elem_list_alloc_space(): + * + * \code + * snd_ctl_elem_list_t* list; + * int count; + * + * // Initialise list + * snd_ctl_elem_list_malloc(&list); + * + * // Get number of elements + * snd_ctl_elem_list(ctl, list); + * count = snd_ctl_elem_list_get_count(list); + * + * // Allocate space for identifiers + * snd_ctl_elem_list_alloc_space(list, count); + * + * // Get identifiers + * snd_ctl_elem_list(ctl, list); // yes, this is same as above + * + * // Do something useful with the list... + * + * // Cleanup + * snd_ctl_elem_list_free_space(list); + * snd_ctl_elem_list_free(list); + * \endcode + * + * + * \par The Elements + * + * The elements in the list are accessed using an index. This index is + * the location in the list; Don't confuse it with the 'index' of the + * element identifier. For example: + * + * \code + * snd_ctl_elem_list_t list; + * unsigned int element_index; + * + * // Allocate space, fill list ... + * + * element_index = snd_ctl_elem_list_get_index(&list, 2); + * \endcode + * + * This will access the 3rd element in the list (index=2) and get the + * elements index from the driver (which might be 13, for example). + */ typedef struct _snd_ctl_elem_list snd_ctl_elem_list_t; /** CTL element info container */ @@ -354,11 +422,18 @@ void snd_ctl_event_copy(snd_ctl_event_t *dst, const snd_ctl_event_t *src); snd_ctl_event_type_t snd_ctl_event_get_type(const snd_ctl_event_t *obj); size_t snd_ctl_elem_list_sizeof(void); + /** \hideinitializer - * \brief allocate an invalid #snd_ctl_elem_list_t using standard alloca - * \param ptr returned pointer + * + * \brief Allocate a #snd_ctl_elem_list_t using standard alloca. + * + * The memory is allocated on the stack and will automatically be + * released when the stack unwinds (i.e. no free() is needed). + * + * \param ptr Pointer to allocated memory. */ #define snd_ctl_elem_list_alloca(ptr) __snd_alloca(ptr, snd_ctl_elem_list) + int snd_ctl_elem_list_malloc(snd_ctl_elem_list_t **ptr); void snd_ctl_elem_list_free(snd_ctl_elem_list_t *obj); void snd_ctl_elem_list_clear(snd_ctl_elem_list_t *obj); diff --git a/src/control/control.c b/src/control/control.c index e21e8f1d..1bcf1ab2 100644 --- a/src/control/control.c +++ b/src/control/control.c @@ -280,6 +280,21 @@ int snd_ctl_card_info(snd_ctl_t *ctl, snd_ctl_card_info_t *info) /** * \brief Get a list of element identifiers + * + * Before calling this function, memoru must be allocated using + * snd_ctl_elem_list_malloc(). + * + * This function obtains data from the sound card driver and puts it + * into the list. + * + * If there was space allocated for the element identifiers (using + * snd_ctl_elem_list_alloc_space()), information will be filled in. If + * too little space was allocated, only a part of the elements will be + * queried. If there was too much space allocated, some of it remains + * unused. Use snd_ctl_elem_list_get_count() and + * snd_ctl_elem_list_get_used() to obtain information about space + * usage. See #snd_ctl_elem_list_t to learn more. + * * \param ctl CTL handle * \param list CTL element identifiers list pointer * \return 0 on success otherwise a negative error code @@ -1508,9 +1523,14 @@ const char *snd_ctl_event_type_name(snd_ctl_event_type_t type) /** * \brief allocate space for CTL element identifiers list - * \param obj CTL element identifiers list - * \param entries Entries to allocate - * \return 0 on success otherwise a negative error code + * + * The space can be released with snd_ctl_elem_list_free_space(). + * + * \param obj CTL element identifiers list. + * \param entries How many entries to allocate. See + * #snd_ctl_elem_list_t to learn how to obtain + * this number in advance. + * \return 0 on success otherwise a negative error code. */ int snd_ctl_elem_list_alloc_space(snd_ctl_elem_list_t *obj, unsigned int entries) { @@ -1526,6 +1546,10 @@ int snd_ctl_elem_list_alloc_space(snd_ctl_elem_list_t *obj, unsigned int entries /** * \brief free previously allocated space for CTL element identifiers list + * + * Releases space previously allocated using + * snd_ctl_elem_list_alloc_space(). + * * \param obj CTL element identifiers list */ void snd_ctl_elem_list_free_space(snd_ctl_elem_list_t *obj) @@ -2016,7 +2040,7 @@ snd_ctl_event_type_t snd_ctl_event_get_type(const snd_ctl_event_t *obj) } /** - * \brief get size of #snd_ctl_elem_list_t + * \brief get size of #snd_ctl_elem_list_t. * \return size in bytes */ size_t snd_ctl_elem_list_sizeof() @@ -2025,7 +2049,10 @@ size_t snd_ctl_elem_list_sizeof() } /** - * \brief allocate an invalid #snd_ctl_elem_list_t using standard malloc + * \brief allocate a #snd_ctl_elem_list_t using standard malloc. + * + * The memory can be released using snd_ctl_elem_list_free(). + * * \param ptr returned pointer * \return 0 on success otherwise negative error code */ @@ -2039,7 +2066,15 @@ int snd_ctl_elem_list_malloc(snd_ctl_elem_list_t **ptr) } /** - * \brief frees a previously allocated #snd_ctl_elem_list_t + * \brief frees a previously allocated #snd_ctl_elem_list_t. + * + * Release memory previously allocated using + * snd_ctl_elem_list_malloc(). + * + * If you used snd_ctl_elem_list_alloc_space() on the list, you must + * use snd_ctl_elem_list_free_space() \em before calling this + * function. + * * \param obj pointer to object to free */ void snd_ctl_elem_list_free(snd_ctl_elem_list_t *obj) @@ -2048,7 +2083,15 @@ void snd_ctl_elem_list_free(snd_ctl_elem_list_t *obj) } /** - * \brief clear given #snd_ctl_elem_list_t object + * \brief Clear given #snd_ctl_elem_list_t object. + * + * This will make the stored identifiers inaccessible without freeing + * their space. + * + * \warning The element identifier space cannot be freed after calling + * this function. Therefore, snd_ctl_elem_list_free_space() + * must be called in advance. + * * \param obj pointer to object to clear */ void snd_ctl_elem_list_clear(snd_ctl_elem_list_t *obj) @@ -2057,7 +2100,11 @@ void snd_ctl_elem_list_clear(snd_ctl_elem_list_t *obj) } /** - * \brief copy one #snd_ctl_elem_list_t to another + * \brief copy one #snd_ctl_elem_list_t to another. + * + * This performs a shallow copy. That means the both lists will share + * the same space for the elements. The elements will not be copied. + * * \param dst pointer to destination * \param src pointer to source */ @@ -2080,6 +2127,12 @@ void snd_ctl_elem_list_set_offset(snd_ctl_elem_list_t *obj, unsigned int val) /** * \brief Get number of used entries in CTL element identifiers list + * + * This function returns how many entries are actually filled with + * useful information. + * + * See also snd_ctl_elem_list_get_count(). + * * \param obj CTL element identifier list * \return number of used entries */ @@ -2090,7 +2143,14 @@ unsigned int snd_ctl_elem_list_get_used(const snd_ctl_elem_list_t *obj) } /** - * \brief Get total count of elements present in CTL device (information present in every filled CTL element identifiers list) + * \brief Get total count of elements present in CTL device + * + * This function returns how many entries were allocated using + * snd_ctl_elem_list_alloc_space(). This information is present after + * snd_ctl_elem_list() was called. + * + * See also snd_ctl_elem_list_get_used(). + * * \param obj CTL element identifier list * \return total number of elements */ @@ -2140,7 +2200,7 @@ snd_ctl_elem_iface_t snd_ctl_elem_list_get_interface(const snd_ctl_elem_list_t * } /** - * \brief Get device part of CTL element identifier for an entry of a CTL element identifiers list + * \brief Get the device part of CTL element identifier for an entry of a CTL element identifiers list * \param obj CTL element identifier list * \param idx Index of entry * \return CTL element related device -- 2.17.1

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[PATCH] ALSA: hda/realtek: Add model alc298-samsung-headphone
by Mike Pozulp 17 Aug '20

17 Aug '20

The very quiet and distorted headphone output bug that afflicted my Samsung Notebook 9 is appearing in many other Samsung laptops. Expose the quirk which fixed my laptop as a model so other users can try it. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207423 Signed-off-by: Mike Pozulp <pozulp.kernel(a)gmail.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 29f5878f0c50..722d01a06422 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7867,6 +7867,7 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {.id = ALC299_FIXUP_PREDATOR_SPK, .name = "predator-spk"}, {.id = ALC298_FIXUP_HUAWEI_MBX_STEREO, .name = "huawei-mbx-stereo"}, {.id = ALC256_FIXUP_MEDION_HEADSET_NO_PRESENCE, .name = "alc256-medion-headset"}, + {.id = ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET, .name = "alc298-samsung-headphone"}, {} }; #define ALC225_STANDARD_PINS \ -- 2.28.0

2 1

[PATCH] usb-audio: Update documentation comment for MS2109 quirk
by Hector Martin 17 Aug '20

17 Aug '20

Signed-off-by: Hector Martin <marcan(a)marcan.st> --- sound/usb/quirks-table.h | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index d79e3ddc5690..72cc03e7aed1 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -3714,8 +3714,8 @@ ALC1220_VB_DESKTOP(0x26ce, 0x0a01), /* Asrock TRX40 Creator */ * they pretend to be 96kHz mono as a workaround for stereo being broken * by that... * - * They also have swapped L-R channels, but that's for userspace to deal - * with. + * They also have an issue with initial stream alignment that causes the + * channels to be swapped and out of phase, which is dealt with in quirks.c. */ { .match_flags = USB_DEVICE_ID_MATCH_DEVICE | -- 2.27.0

2 1

[PATCH v2] soundwire: master: enable pm runtime
by Bard Liao 17 Aug '20

17 Aug '20

The hierarchy of soundwire devices is platform device -> M device -> S device. A S device is physically attached on the platform device. So the platform device should be resumed when a S device is resumed. As the bridge of platform device and S device, we have to implement runtime pm on M driver. We have set runtime pm ops in M driver already, but still need to enable runtime pm. Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> --- Changes in v2: - Edit the commit message to explain why we need the patch. --- drivers/soundwire/master.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/drivers/soundwire/master.c b/drivers/soundwire/master.c index 5f0b2189defe..3488bb824e84 100644 --- a/drivers/soundwire/master.c +++ b/drivers/soundwire/master.c @@ -154,6 +154,7 @@ int sdw_master_device_add(struct sdw_bus *bus, struct device *parent, bus->dev = &md->dev; bus->md = md; + pm_runtime_enable(&bus->md->dev); device_register_err: return ret; } @@ -166,6 +167,7 @@ int sdw_master_device_add(struct sdw_bus *bus, struct device *parent, */ int sdw_master_device_del(struct sdw_bus *bus) { + pm_runtime_disable(&bus->md->dev); device_unregister(bus->dev); return 0; -- 2.17.1

2 1

[PATCH] ALSA: usb-audio: ignore broken processing/extension unit
by tom.ty89＠gmail.com 17 Aug '20

17 Aug '20

From: Tom Yan <tom.ty89(a)gmail.com> Some devices have broken extension unit where getting current value doesn't work. Attempt that once when creating mixer control for it. If it fails, just ignore it, so that it won't cripple the device entirely (and/or make the error floods). --- sound/usb/mixer.c | 8 +++++++- 1 file changed, 7 insertions(+), 1 deletion(-) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index eab0fd4fd7c3..e0b7174c1043 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -2367,7 +2367,7 @@ static int build_audio_procunit(struct mixer_build *state, int unitid, int num_ins; struct usb_mixer_elem_info *cval; struct snd_kcontrol *kctl; - int i, err, nameid, type, len; + int i, err, nameid, type, len, val; const struct procunit_info *info; const struct procunit_value_info *valinfo; const struct usbmix_name_map *map; @@ -2470,6 +2470,12 @@ static int build_audio_procunit(struct mixer_build *state, int unitid, break; } + err = get_cur_ctl_value(cval, cval->control << 8, &val); + if (err < 0) { + usb_mixer_elem_info_free(cval); + return -EINVAL; + } + kctl = snd_ctl_new1(&mixer_procunit_ctl, cval); if (!kctl) { usb_mixer_elem_info_free(cval); -- 2.28.0

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Re: pcm|dmaengine|imx-sdma race condition on i.MX6
by Richard Leitner 17 Aug '20

17 Aug '20

On Fri, Aug 14, 2020 at 11:18:10AM +0200, Robin Gong wrote: > > On 2020/08/13 19:23: Richard Leitner <richard.leitner(a)skidata.com> wrote: > > Hi, > > we've found a race condition with the PCM on the i.MX6 which results > > in an -EIO for the SNDRV_PCM_IOCTL_READI_FRAMES ioctl after an -EPIPE (XRUN). > > > > A possible reproduction may look like the following reduced call graph > > during a PCM capture: > > > > us -> ioctl(SNDRV_PCM_IOCTL_READI_FRAMES) > > - wait_for_avail() > > - schedule_timeout() > > -> snd_pcm_update_hw_ptr0() > > - snd_pcm_update_state: EPIPE (XRUN) > > - sdma_disable_channel_async() # get's scheduled away due to sleep us > > <- ioctl(SNDRV_PCM_IOCTL_READI_FRAMES) returns -EPIPE > > us -> ioctl(SNDRV_PCM_IOCTL_PREPARE) # as reaction to the EPIPE (XRUN) > > us -> ioctl(SNDRV_PCM_IOCTL_READI_FRAMES) # next try to capture frames > > - sdma_prep_dma_cyclic() > > - sdma_load_context() # not loaded as context_loaded is 1 > > - wait_for_avail() > > - schedule_timeout() > > # now the sdma_channel_terminate_work() comes back and sets # > > context_loaded = false and frees in vchan_dma_desc_free_list(). > > us <- ioctl returns -EIO (capture write error (DMA or IRQ trouble?)) > > Seems the write error caused by context_loaded not set to false before > next transfer start? If yes, please have a try with the 03/04 of the below > patch set, anyway, could you post your failure log? > https://lkml.org/lkml/2020/8/11/111 Thanks for the pointer to those patches. I've tested them on top of the v5.8 tag during the weekend. It seems those patches are mitigiating the described EIO issue. Nonetheless IMHO this patches are not fixing the root cause of the problem (unsynchronized sdma_disable_channel_async / sdma_prep_dma_cyclic). Do you (or somebody else) have any suggestions/comments/objections on that? regards;rl > > > > > > > What we have found out, based on our understanding: > > The dmaengine docu states that a dmaengine_terminate_async() must be > > followed by a dmaengine_synchronize(). > > However, in the pcm_dmaengine.c, only dmaengine_terminate_async() is > > called (for performance reasons and because it might be called from an > > interrupt handler). > > > > In our tests, we saw that the user-space immediately calls > > ioctl(SNDRV_PCM_IOCTL_PREPARE) as a handler for the happened xrun > > (previous ioctl(SNDRV_PCM_IOCTL_READI_FRAMES) returns with -EPIPE). In > > our case (imx-sdma.c), the terminate really happens asynchronously > > with a worker thread which is not awaited/synchronized by the > > ioctl(SNDRV_PCM_IOCTL_PREPARE) call. > > > > Since the syscall immediately enters an atomic context > > (snd_pcm_stream_lock_irq()), we are not able to flush the work of the > > termination worker from within the DMA context. This leads to an > > unterminated DMA getting re-initialized and then terminated. > > > > On the i.MX6 platform the problem is (if I got it correctly) that the > > sdma_channel_terminate_work() called after the -EPIPE gets scheduled > > away (for the 1-2ms sleep [1]). During that time the userspace already > > sends in the > > ioctl(SNDRV_PCM_IOCTL_PREPARE) and > > ioctl(SNDRV_PCM_IOCTL_READI_FRAMES). > > As none of them are anyhow synchronized to the terminate_worker the > > vchan_dma_desc_free_list() [2] and "sdmac->context_loaded = false;" > > [3] are executed during the wait_for_avail() [4] of the > > ioctl(SNDRV_PCM_IOCTL_READI_FRAMES). > > > > To make sure we identified the problem correctly we've tested to add a > > "dmaengine_synchronize()" before the snd_pcm_prepare() in [5]. This > > fixed the race condition in all our tests. (Before we were able to > > reproduce it in 100% of the test runs). > > > > Based on our understanding, there are two different points to ensure > > the > > termination: > > Either ensure that the termination is finished within the previous > > SNDRV_PCM_IOCTL_READI_FRAMES call (inside the DMA context) or > > finishing it in the SNDRV_PCM_IOCTL_PREPARE call (and all other > > applicable ioclts) before entering the atomic context (from the PCM context). > > > > We initially thought about implementing the first approach, basically > > splitting up the dma_device terminate_all operation into a sync > > (busy-wait) and a async one. This would align the operations with the > > DMAengine interface and would enable a sync termination variant from > > atomic contexts. > > However, we saw that the dma_free_attrs() function has a WARN_ON on > > irqs disabled, which would be the case for the sync variant. > > Side note: We found this issue on the current v5.4.y LTS branch, but > > it also affects v5.8.y. > > > > Any feedback or pointers how we may fix the problem are warmly welcome! > > If anything is unclear please just ask :-) > > > > regards; > > Richard Leitner > > Benjamin Bara

1 0

[PATCH v4 0/2] Add mediatek codec mt6359 driver
by Jiaxin Yu 17 Aug '20

17 Aug '20

Add mediatek codec (MT6359) driver MT6359 support playback and capture feature. On downlink path, it includes three DACs for handset, headset, and lineout path. On unlink path, it includeds three ADCs for main mic, second mic, 3rd mic, and headset mic. By scenario, select *_MUX widget to create damp path. And by select mic_type_mux to choose DMIC/AMIC/.... For example, select these MUX widget to create headset path (1) DAC In Mux --> "Normal Path" (2) HP Mux --> "Audio Playback" v4 changes: 1. Add a remove() function to undo regulator_enable(). 2. Removed unnecessary logs. v3 changes: 1. patchwork link: https://patchwork.kernel.org/cover/11714291/ https://patchwork.kernel.org/patch/11714295/ https://patchwork.kernel.org/patch/11714339/ v2 changes: 1. patchwork link: https://patchwork.kernel.org/cover/11706935/ https://patchwork.kernel.org/patch/11708865/ https://patchwork.kernel.org/patch/11706937/ v1 changes: 1.lkml link: https://lkml.org/lkml/2020/3/5/1257 Jiaxin Yu (2): ASoC: mediatek: mt6359: add codec driver dt-bindings: mediatek: mt6359: add codec document .../devicetree/bindings/sound/mt6359.yaml | 68 + sound/soc/codecs/Kconfig | 8 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/mt6359.c | 2755 +++++++++++++++++ sound/soc/codecs/mt6359.h | 2640 ++++++++++++++++ 5 files changed, 5473 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/mt6359.yaml create mode 100644 sound/soc/codecs/mt6359.c create mode 100644 sound/soc/codecs/mt6359.h -- 2.18.0

2 3

USB-Audio: Dell-WD15-Dock: make input and output volume adjustable
by GitHub pull_request - opened 17 Aug '20

17 Aug '20

alsa-project/alsa-ucm-conf pull request #47 was opened from jason77-wang: Recently we found the input volume is too low for some specific headset-mic, even we adjust the input volume to max from PA, we still can't record the sound from that headset-mic. That is because we change the input or output volume from PA, but the mixer's volume is not changed, only PA's soft-volume is changed. Checking the amixer controls for this sound card, it supports MixerElem, Volume and Switch, So adding them in the ucm. Signed-off-by: Hui Wang <hui.wang(a)canonical.com> Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/47 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/47.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

HDA-Intel: HiFi-dual: Fix the Rear Mic's Jack name
by GitHub pull_request - opened 17 Aug '20

17 Aug '20

alsa-project/alsa-ucm-conf pull request #46 was opened from jason77-wang: On the LENOVO P520 (dual codecs machine), the Jack name of Rear Mic is "Rear Mic Jack" instead of "Mic Jack". Below is picked from alsa-info.txt on the LENOVO p520 machine: control.18 { iface CARD name 'Rear Mic Jack' value true comment { access read type BOOLEAN count 1 } } Signed-off-by: Hui Wang <hui.wang(a)canonical.com> Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/46 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/46.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH v2] soundwire: SDCA: detect sdca_cascade interrupt
by Bard Liao 16 Aug '20

16 Aug '20

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> The SoundWire 1.2 specification defines an "SDCA cascade" bit which handles a logical OR of all SDCA interrupt sources (up to 30 defined). Due to limitations of the addressing space, this bit is located in the SDW_DP0_INT register when DP0 is used, or alternatively in the DP0_SDCA_Support_INTSTAT register when DP0 is not used. To allow for both cases to be handled, this bit will be checked in the main device-level interrupt handling code. This will result in the register being read twice if DP0 is enabled, but it's not clear how to optimize this case. It's also more logical to deal with this interrupt at the device than the port level, this bit is really not DP0 specific and its location in the DP0_INTSTAT bit is only due to the lack of free space in SCP_INTSTAT_1. The SDCA_Cascade bit cannot be masked or cleared, so the interrupt handling only forwards the detection to the Slave driver, which will deal with reading the relevant SDCA status bits and clearing them. The bus driver only signals the detection. The communication with the Slave driver is based on the same interrupt callback, with only an extension to provide the status of the sdca_cascade bit. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Rander Wang <rander.wang(a)linux.intel.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> --- Changelog: v2: - Re-read SDW_DP0_INT while reading status again. --- drivers/soundwire/bus.c | 28 +++++++++++++++++++++++++++- include/linux/soundwire/sdw.h | 4 ++++ 2 files changed, 31 insertions(+), 1 deletion(-) diff --git a/drivers/soundwire/bus.c b/drivers/soundwire/bus.c index e6e0fb9a81b4..d254c23201b9 100644 --- a/drivers/soundwire/bus.c +++ b/drivers/soundwire/bus.c @@ -1361,6 +1361,7 @@ static int sdw_handle_slave_alerts(struct sdw_slave *slave) int port_num, stat, ret, count = 0; unsigned long port; bool slave_notify = false; + u8 sdca_cascade = 0; u8 buf, buf2[2], _buf, _buf2[2]; sdw_modify_slave_status(slave, SDW_SLAVE_ALERT); @@ -1388,6 +1389,16 @@ static int sdw_handle_slave_alerts(struct sdw_slave *slave) goto io_err; } + if (slave->prop.is_sdca) { + ret = sdw_read(slave, SDW_DP0_INT); + if (ret < 0) { + dev_err(slave->bus->dev, + "SDW_DP0_INT read failed:%d\n", ret); + goto io_err; + } + sdca_cascade = ret & SDW_DP0_SDCA_CASCADE; + } + do { /* * Check parity, bus clash and Slave (impl defined) @@ -1416,6 +1427,10 @@ static int sdw_handle_slave_alerts(struct sdw_slave *slave) slave_notify = true; } + /* the SDCA interrupts are cleared in the codec driver .interrupt_callback() */ + if (sdca_cascade) + slave_notify = true; + /* Check port 0 - 3 interrupts */ port = buf & SDW_SCP_INT1_PORT0_3; @@ -1453,6 +1468,7 @@ static int sdw_handle_slave_alerts(struct sdw_slave *slave) /* Update the Slave driver */ if (slave_notify && slave->ops && slave->ops->interrupt_callback) { + slave_intr.sdca_cascade = sdca_cascade; slave_intr.control_port = clear; memcpy(slave_intr.port, &port_status, sizeof(slave_intr.port)); @@ -1487,11 +1503,21 @@ static int sdw_handle_slave_alerts(struct sdw_slave *slave) goto io_err; } + if (slave->prop.is_sdca) { + ret = sdw_read(slave, SDW_DP0_INT); + if (ret < 0) { + dev_err(slave->bus->dev, + "SDW_DP0_INT read failed:%d\n", ret); + goto io_err; + } + sdca_cascade = ret & SDW_DP0_SDCA_CASCADE; + } + /* Make sure no interrupts are pending */ buf &= _buf; buf2[0] &= _buf2[0]; buf2[1] &= _buf2[1]; - stat = buf || buf2[0] || buf2[1]; + stat = buf || buf2[0] || buf2[1] || sdca_cascade; /* * Exit loop if Slave is continuously in ALERT state even diff --git a/include/linux/soundwire/sdw.h b/include/linux/soundwire/sdw.h index 76052f12c9f7..a51494646a2c 100644 --- a/include/linux/soundwire/sdw.h +++ b/include/linux/soundwire/sdw.h @@ -355,6 +355,7 @@ struct sdw_dpn_prop { * @dp0_prop: Data Port 0 properties * @src_dpn_prop: Source Data Port N properties * @sink_dpn_prop: Sink Data Port N properties + * @is_sdca: the Slave supports the SDCA specification */ struct sdw_slave_prop { u32 mipi_revision; @@ -376,6 +377,7 @@ struct sdw_slave_prop { struct sdw_dp0_prop *dp0_prop; struct sdw_dpn_prop *src_dpn_prop; struct sdw_dpn_prop *sink_dpn_prop; + bool is_sdca; }; /** @@ -465,10 +467,12 @@ struct sdw_slave_id { /** * struct sdw_slave_intr_status - Slave interrupt status + * @sdca_cascade: set if the Slave device reports an SDCA interrupt * @control_port: control port status * @port: data port status */ struct sdw_slave_intr_status { + bool sdca_cascade; u8 control_port; u8 port[15]; }; -- 2.17.1

1 0

Re: [PATCH] dt-bindings: Whitespace clean-ups in schema files
by Sam Ravnborg 16 Aug '20

16 Aug '20

Hi Rob. On Wed, Aug 12, 2020 at 02:36:18PM -0600, Rob Herring wrote: > Clean-up incorrect indentation, extra spaces, long lines, and missing > EOF newline in schema files. Most of the clean-ups are for list > indentation which should always be 2 spaces more than the preceding > keyword. > > Found with yamllint (which I plan to integrate into the checks). I have browsed through the patch - and there was only a few things that jumped at me. With these points considered: Acked-by: Sam Ravnborg <sam(a)ravnborg.org> I expect only some (few) of my points to actually results in any updates. I look forward to have the lint functionality as part of the built-in tools so we catch these things early. Sam > diff --git a/Documentation/devicetree/bindings/arm/fsl.yaml b/Documentation/devicetree/bindings/arm/fsl.yaml > index f63895c8ce2d..88814a2a14a5 100644 > --- a/Documentation/devicetree/bindings/arm/fsl.yaml > +++ b/Documentation/devicetree/bindings/arm/fsl.yaml > @@ -273,8 +273,8 @@ properties: > - fsl,imx6ull-14x14-evk # i.MX6 UltraLiteLite 14x14 EVK Board > - kontron,imx6ull-n6411-som # Kontron N6411 SOM > - myir,imx6ull-mys-6ulx-eval # MYiR Tech iMX6ULL Evaluation Board > - - toradex,colibri-imx6ull-eval # Colibri iMX6ULL Module on Colibri Evaluation Board > - - toradex,colibri-imx6ull-wifi-eval # Colibri iMX6ULL Wi-Fi / Bluetooth Module on Colibri Evaluation Board > + - toradex,colibri-imx6ull-eval # Colibri iMX6ULL Module on Colibri Eval Board > + - toradex,colibri-imx6ull-wifi-eval # Colibri iMX6ULL Wi-Fi / BT Module on Colibri Eval Board > - const: fsl,imx6ull This change looks bad as it drops the alignment with the comments below. See following patch chunck: > > - description: Kontron N6411 S Board > @@ -312,9 +312,12 @@ properties: > - toradex,colibri-imx7d # Colibri iMX7 Dual Module > - toradex,colibri-imx7d-aster # Colibri iMX7 Dual Module on Aster Carrier Board > - toradex,colibri-imx7d-emmc # Colibri iMX7 Dual 1GB (eMMC) Module > - - toradex,colibri-imx7d-emmc-aster # Colibri iMX7 Dual 1GB (eMMC) Module on Aster Carrier Board > - - toradex,colibri-imx7d-emmc-eval-v3 # Colibri iMX7 Dual 1GB (eMMC) Module on Colibri Evaluation Board V3 > - - toradex,colibri-imx7d-eval-v3 # Colibri iMX7 Dual Module on Colibri Evaluation Board V3 > + - toradex,colibri-imx7d-emmc-aster # Colibri iMX7 Dual 1GB (eMMC) Module on > + # Aster Carrier Board > diff --git a/Documentation/devicetree/bindings/display/panel/ilitek,ili9322.yaml b/Documentation/devicetree/bindings/display/panel/ilitek,ili9322.yaml > index 177d48c5bd97..e89c1ea62ffa 100644 > --- a/Documentation/devicetree/bindings/display/panel/ilitek,ili9322.yaml > +++ b/Documentation/devicetree/bindings/display/panel/ilitek,ili9322.yaml > @@ -25,8 +25,7 @@ properties: > compatible: > items: > - enum: > - - dlink,dir-685-panel > - > + - dlink,dir-685-panel > - const: ilitek,ili9322 > > reset-gpios: true > diff --git a/Documentation/devicetree/bindings/display/panel/ilitek,ili9881c.yaml b/Documentation/devicetree/bindings/display/panel/ilitek,ili9881c.yaml > index a39332276bab..76a9068a85dd 100644 > --- a/Documentation/devicetree/bindings/display/panel/ilitek,ili9881c.yaml > +++ b/Documentation/devicetree/bindings/display/panel/ilitek,ili9881c.yaml > @@ -13,8 +13,7 @@ properties: > compatible: > items: > - enum: > - - bananapi,lhr050h41 > - > + - bananapi,lhr050h41 > - const: ilitek,ili9881c > The extra lines is a simple way to indicate that here shall be added more in the future. So I like the empty line. > diff --git a/Documentation/devicetree/bindings/leds/backlight/qcom-wled.yaml b/Documentation/devicetree/bindings/leds/backlight/qcom-wled.yaml > index 32e0896c6bc1..47938e372987 100644 > --- a/Documentation/devicetree/bindings/leds/backlight/qcom-wled.yaml > +++ b/Documentation/devicetree/bindings/leds/backlight/qcom-wled.yaml > @@ -79,7 +79,8 @@ properties: > description: | > kHz; switching frequency. > $ref: /schemas/types.yaml#/definitions/uint32 > - enum: [ 600, 640, 685, 738, 800, 872, 960, 1066, 1200, 1371, 1600, 1920, 2400, 3200, 4800, 9600 ] > + enum: [ 600, 640, 685, 738, 800, 872, 960, 1066, 1200, 1371, 1600, 1920, > + 2400, 3200, 4800, 9600 ] > > qcom,ovp: > description: | In the modern world we are living in now line length of 100 chars are OK. checkpatch and coding_style is updated to reflected this. > diff --git a/Documentation/devicetree/bindings/spi/mikrotik,rb4xx-spi.yaml b/Documentation/devicetree/bindings/spi/mikrotik,rb4xx-spi.yaml > index 4ddb42a4ae05..9102feae90a2 100644 > --- a/Documentation/devicetree/bindings/spi/mikrotik,rb4xx-spi.yaml > +++ b/Documentation/devicetree/bindings/spi/mikrotik,rb4xx-spi.yaml > @@ -33,4 +33,5 @@ examples: > reg = <0x1f000000 0x10>; > }; > > -... > \ No newline at end of file > +... > + Added one line too much? diff --git a/Documentation/devicetree/bindings/spi/spi-mux.yaml b/Documentation/devicetree/bindings/spi/spi-mux.yaml > index 0ae692dc28b5..3d3fed63409b 100644 > --- a/Documentation/devicetree/bindings/spi/spi-mux.yaml > +++ b/Documentation/devicetree/bindings/spi/spi-mux.yaml > @@ -43,47 +43,47 @@ properties: > maxItems: 1 > > required: > - - compatible > - - reg > - - spi-max-frequency > - - mux-controls > + - compatible > + - reg > + - spi-max-frequency > + - mux-controls > > examples: > - - | > - #include <dt-bindings/gpio/gpio.h> > - mux: mux-controller { > - compatible = "gpio-mux"; > - #mux-control-cells = <0>; > + - | > + #include <dt-bindings/gpio/gpio.h> > + mux: mux-controller { > + compatible = "gpio-mux"; > + #mux-control-cells = <0>; > > - mux-gpios = <&gpio0 3 GPIO_ACTIVE_HIGH>; > - }; > + mux-gpios = <&gpio0 3 GPIO_ACTIVE_HIGH>; > + }; Example is updated to use 4-space indent. I like. But many other examples are left untouched. So I wonder if updating all examples to the same indent should be left for another mega-patch? > diff --git a/Documentation/devicetree/bindings/vendor-prefixes.yaml b/Documentation/devicetree/bindings/vendor-prefixes.yaml > index f3d847832fdc..2baee2c817c1 100644 > --- a/Documentation/devicetree/bindings/vendor-prefixes.yaml > +++ b/Documentation/devicetree/bindings/vendor-prefixes.yaml > @@ -993,7 +993,8 @@ patternProperties: > "^sst,.*": > description: Silicon Storage Technology, Inc. > "^sstar,.*": > - description: Xiamen Xingchen(SigmaStar) Technology Co., Ltd. (formerly part of MStar Semiconductor, Inc.) > + description: Xiamen Xingchen(SigmaStar) Technology Co., Ltd. > + (formerly part of MStar Semiconductor, Inc.) > "^st,.*": > description: STMicroelectronics > "^starry,.*": Did you check that they are all in alphabetical order? I would be suprised if this is the only issue in this file.

3 2

[PATCH] sound/isa:fix spelling mistakes in the comments
by Liang Wang 16 Aug '20

16 Aug '20

Fix spelling mistakes in the comments: initailise ==> initialise tranfer ==> transfer Initialse ==> Initialise Signed-off-by: Liang Wang <wangliang101(a)huawei.com> --- sound/isa/sscape.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/isa/sscape.c b/sound/isa/sscape.c index 5363d88cc4b9..2e5a5c5279e8 100644 --- a/sound/isa/sscape.c +++ b/sound/isa/sscape.c @@ -308,7 +308,7 @@ static inline int verify_mpu401(const struct snd_mpu401 *mpu) } /* - * This is apparently the standard way to initailise an MPU-401 + * This is apparently the standard way to initialise an MPU-401 */ static inline void initialise_mpu401(const struct snd_mpu401 *mpu) { @@ -339,7 +339,7 @@ static void soundscape_free(struct snd_card *c) } /* - * Tell the SoundScape to begin a DMA tranfer using the given channel. + * Tell the SoundScape to begin a DMA transfer using the given channel. * All locking issues are left to the caller. */ static void sscape_start_dma_unsafe(unsigned io_base, enum GA_REG reg) @@ -803,7 +803,7 @@ static int mpu401_open(struct snd_mpu401 *mpu) } /* - * Initialse an MPU-401 subdevice for MIDI support on the SoundScape. + * Initialise an MPU-401 subdevice for MIDI support on the SoundScape. */ static int create_mpu401(struct snd_card *card, int devnum, unsigned long port, int irq) -- 2.27.0

2 1

[PATCH v2 alsa-lib] README: add patch submission instructions
by Tanu Kaskinen 16 Aug '20

16 Aug '20

Signed-off-by: Tanu Kaskinen <tanuk(a)iki.fi> --- v2: - fixed Markdown syntax: changed ' to ` - added Signed-off-by README.md | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/README.md b/README.md index 3cd122d5..56af78d0 100644 --- a/README.md +++ b/README.md @@ -12,3 +12,14 @@ http://www.alsa-project.org/alsa-doc/alsa-lib/ You may give a look for more information about the ALSA project to URL http://www.alsa-project.org. + +### Submitting patches + +The preferred way to submit patches is by sending them by email to the +alsa-devel mailing list. Sending mail to the list requires subscription, +subscribe here: https://mailman.alsa-project.org/mailman/listinfo/alsa-devel + +Add Takashi Iwai `<tiwai(a)suse.de>` and/or Jaroslav Kysela `<perex(a)perex.cz>` to +Cc so that your patch won't be missed. + +Patches are also accepted as GitHub pull requests. -- 2.20.1

2 1

[PATCH v3 0/2] Add mediatek codec mt6359 driver
by Jiaxin Yu 15 Aug '20

15 Aug '20

Add mediatek codec (MT6359) driver MT6359 support playback and capture feature. On downlink path, it includes three DACs for handset, headset, and lineout path. On unlink path, it includeds three ADCs for main mic, second mic, 3rd mic, and headset mic. By scenario, select *_MUX widget to create damp path. And by select mic_type_mux to choose DMIC/AMIC/.... For example, select these MUX widget to create headset path (1) DAC In Mux --> "Normal Path" (2) HPL Mux --> "Audio Playback" (3) HPR Mux --> "Audio Playback" v3 changes: 1. Remove calibration related functions. 2. Conbined 'HPL Mux' and 'HPR Mux' to 'HP Mux'. 3. Move the regulator_enable of 'vaud18' to the component probe. 4. Fix some incorrect coding style. v2 changes: 1. patchwork link: https://patchwork.kernel.org/cover/11706935/ https://patchwork.kernel.org/patch/11708865/ https://patchwork.kernel.org/patch/11706937/ v1 changes: 1.lkml link: https://lkml.org/lkml/2020/3/5/1257 Jiaxin Yu (2): WIP: ASoC: mediatek: mt6359: add codec driver WIP: dt-bindings: mediatek: mt6359: add codec document .../devicetree/bindings/sound/mt6359.yaml | 68 + sound/soc/codecs/Kconfig | 8 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/mt6359.c | 2737 +++++++++++++++++ sound/soc/codecs/mt6359.h | 2640 ++++++++++++++++ 5 files changed, 5455 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/mt6359.yaml create mode 100644 sound/soc/codecs/mt6359.c create mode 100644 sound/soc/codecs/mt6359.h -- 2.18.0

3 5

[PATCH alsa-lib] README: add patch submission instructions
by Tanu Kaskinen 15 Aug '20

15 Aug '20

--- README.md | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/README.md b/README.md index 3cd122d5..57fd353e 100644 --- a/README.md +++ b/README.md @@ -12,3 +12,14 @@ http://www.alsa-project.org/alsa-doc/alsa-lib/ You may give a look for more information about the ALSA project to URL http://www.alsa-project.org. + +### Submitting patches + +The preferred way to submit patches is by sending them by email to the +alsa-devel mailing list. Sending mail to the list requires subscription, +subscribe here: https://mailman.alsa-project.org/mailman/listinfo/alsa-devel + +Add Takashi Iwai `<tiwai(a)suse.de>` and/or Jaroslav Kysela `<perex(a)perex.cz>' to +Cc so that your patch won't be missed. + +Patches are also accepted as GitHub pull requests. -- 2.20.1

1 0

How to send in patches?
by Tanjeff Moos 15 Aug '20

15 Aug '20

Hi folks, I'd like to send in my first patch for alsa-lib. Is it okay if I simply send it to this list? Or is there a process how to do this? Kind regards, Tanjeff

2 1

[PATCH] ALSA: usb-audio: Add capture support for Saffire 6 (USB 1.1)
by Alexander Tsoy 15 Aug '20

15 Aug '20

Capture and playback endpoints on Saffire 6 (USB 1.1) resides on the same interface. This was not supported by the composite quirk back in the day when initial support for this device was added, thus only playback was enabled until now. Signed-off-by: Alexander Tsoy <alexander(a)tsoy.me> --- sound/usb/quirks-table.h | 30 ++++++++++++++++++++++++++++++ 1 file changed, 30 insertions(+) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index d79e3ddc5690..e6202608e043 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -2673,28 +2673,58 @@ YAMAHA_DEVICE(0x7010, "UB99"), { USB_DEVICE(0x1235, 0x0010), .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { .vendor_name = "Focusrite", .product_name = "Saffire 6 USB", .ifnum = QUIRK_ANY_INTERFACE, .type = QUIRK_COMPOSITE, .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 0, + .type = QUIRK_AUDIO_STANDARD_MIXER, + }, { .ifnum = 0, .type = QUIRK_AUDIO_FIXED_ENDPOINT, .data = &(const struct audioformat) { .formats = SNDRV_PCM_FMTBIT_S24_3LE, .channels = 4, .iface = 0, .altsetting = 1, .altset_idx = 1, .attributes = UAC_EP_CS_ATTR_SAMPLE_RATE, .endpoint = 0x01, .ep_attr = USB_ENDPOINT_XFER_ISOC, + .datainterval = 1, + .maxpacksize = 0x024c, + .rates = SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000, + .rate_min = 44100, + .rate_max = 48000, + .nr_rates = 2, + .rate_table = (unsigned int[]) { + 44100, 48000 + } + } + }, + { + .ifnum = 0, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 2, + .iface = 0, + .altsetting = 1, + .altset_idx = 1, + .attributes = 0, + .endpoint = 0x82, + .ep_attr = USB_ENDPOINT_XFER_ISOC, + .datainterval = 1, + .maxpacksize = 0x0126, .rates = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000, .rate_min = 44100, .rate_max = 48000, .nr_rates = 2, .rate_table = (unsigned int[]) { 44100, 48000 } -- 2.26.2

2 1

[PATCH] Add documentation for snd_ctl_elem_list_*
by Tanjeff Moos 15 Aug '20

15 Aug '20

I added doxygen documentation to alsa-lib. The patch was created using git format-patch. This is my first patch. Please let me know if I need to improve it. >From 731895b67fcae743551989a5dd04382003e66560 Mon Sep 17 00:00:00 2001 From: "Tanjeff-N. Moos" <tanjeff(a)cccmz.de> Date: Fri, 14 Aug 2020 08:40:28 +0200 Subject: [PATCH] Add documentation for snd_ctl_elem_list_*. --- include/control.h | 81 +++++++++++++++++++++++++++++++++++++++++-- src/control/control.c | 80 ++++++++++++++++++++++++++++++++++++------ 2 files changed, 148 insertions(+), 13 deletions(-) diff --git a/include/control.h b/include/control.h index 02db72d4..9deec6f3 100644 --- a/include/control.h +++ b/include/control.h @@ -56,7 +56,75 @@ typedef struct _snd_ctl_card_info snd_ctl_card_info_t; /** CTL element identifier container */ typedef struct _snd_ctl_elem_id snd_ctl_elem_id_t; -/** CTL element identifier list container */ +/** CTL element list container + * + * This is a list of CTL elements. The list contains management + * information (e.g. how many elements the sound card has) as well as + * the element identifiers. All functions which operate on the list + * are named snd_ctl_elem_list_*(). + * + * \par Memory management + * + * There are two memory areas to deal with: The list container itself + * and the memory for the element identifiers. + * + * To manage the area for the list container, the following functions + * are used: + * + * - snd_ctl_elem_list_malloc() / snd_ctl_elem_list_free() to allocate + * and free memory on the heap, or + * - snd_ctl_elem_list_alloca() to allocate the memory on the + * stack. This memory is auto-released when the stack is unwound. + * + * To manage the space for the element identifiers, the + * snd_ctl_elem_list_alloc_space() and snd_ctl_elem_list_free_space() + * are used. Allocating the right amount of space can be achieved by + * first obtaining the number of elements and then calling + * snd_ctl_elem_list_alloc_space(): + * + * \code + * snd_ctl_elem_list_t* list; + * int count; + * + * // Initialise list + * snd_ctl_elem_list_malloc(&list); + * + * // Get number of elements + * snd_ctl_elem_list(ctl, list); + * count = snd_ctl_elem_list_get_count(list); + * + * // Allocate space for identifiers + * snd_ctl_elem_list_alloc_space(list, count); + * + * // Get identifiers + * snd_ctl_elem_list(ctl, list); // yes, this is same as above :) + * + * // Do something useful with the list... + * + * // Cleanup + * snd_ctl_elem_list_free_space(list); + * snd_ctl_elem_list_free(list); + * \endcode + * + * + * \par The Elements + * + * The elements in the list are accessed using an index. This index is + * the location in the list; Don't confuse it with the 'index' of the + * element identifier. For example: + * + * \code + * snd_ctl_elem_list_t list; + * unsigned int element_index; + * + * // Allocate space, fill list ... + * + * element_index = snd_ctl_elem_list_get_index(&list, 2); + * \endcode + * + * This will access the 3rd element in the list (index=2) and get the + * elements index from the driver (which might be 13, for example). + */ typedef struct _snd_ctl_elem_list snd_ctl_elem_list_t; /** CTL element info container */ @@ -354,11 +422,18 @@ void snd_ctl_event_copy(snd_ctl_event_t *dst, const snd_ctl_event_t *src); snd_ctl_event_type_t snd_ctl_event_get_type(const snd_ctl_event_t *obj); size_t snd_ctl_elem_list_sizeof(void); + /** \hideinitializer - * \brief allocate an invalid #snd_ctl_elem_list_t using standard alloca - * \param ptr returned pointer + * + * \brief Allocate a #snd_ctl_elem_list_t using standard alloca. + * + * The memory is allocated on the stack and will automatically be + * released when the stack unwinds (i.e. no free() is needed). + * + * \param ptr Pointer to allocated memory. */ #define snd_ctl_elem_list_alloca(ptr) __snd_alloca(ptr, snd_ctl_elem_list) + int snd_ctl_elem_list_malloc(snd_ctl_elem_list_t **ptr); void snd_ctl_elem_list_free(snd_ctl_elem_list_t *obj); void snd_ctl_elem_list_clear(snd_ctl_elem_list_t *obj); diff --git a/src/control/control.c b/src/control/control.c index e21e8f1d..1bcf1ab2 100644 --- a/src/control/control.c +++ b/src/control/control.c @@ -280,6 +280,21 @@ int snd_ctl_card_info(snd_ctl_t *ctl, snd_ctl_card_info_t *info) /** * \brief Get a list of element identifiers + * + * Before calling this function, memoru must be allocated using + * snd_ctl_elem_list_malloc(). + * + * This function obtains data from the sound card driver and puts it + * into the list. + * + * If there was space allocated for the element identifiers (using + * snd_ctl_elem_list_alloc_space()), information will be filled in. If + * too little space was allocated, only a part of the elements will be + * queried. If there was too much space allocated, some of it remains + * unused. Use snd_ctl_elem_list_get_count() and + * snd_ctl_elem_list_get_used() to obtain information about space + * usage. See #snd_ctl_elem_list_t to learn more. + * * \param ctl CTL handle * \param list CTL element identifiers list pointer * \return 0 on success otherwise a negative error code @@ -1508,9 +1523,14 @@ const char *snd_ctl_event_type_name(snd_ctl_event_type_t type) /** * \brief allocate space for CTL element identifiers list - * \param obj CTL element identifiers list - * \param entries Entries to allocate - * \return 0 on success otherwise a negative error code + * + * The space can be released with snd_ctl_elem_list_free_space(). + * + * \param obj CTL element identifiers list. + * \param entries How many entries to allocate. See + * #snd_ctl_elem_list_t to learn how to obtain + * this number in advance. + * \return 0 on success otherwise a negative error code. */ int snd_ctl_elem_list_alloc_space(snd_ctl_elem_list_t *obj, unsigned int entries) { @@ -1526,6 +1546,10 @@ int snd_ctl_elem_list_alloc_space(snd_ctl_elem_list_t *obj, unsigned int entries /** * \brief free previously allocated space for CTL element identifiers list + * + * Releases space previously allocated using + * snd_ctl_elem_list_alloc_space(). + * * \param obj CTL element identifiers list */ void snd_ctl_elem_list_free_space(snd_ctl_elem_list_t *obj) @@ -2016,7 +2040,7 @@ snd_ctl_event_type_t snd_ctl_event_get_type(const snd_ctl_event_t *obj) } /** - * \brief get size of #snd_ctl_elem_list_t + * \brief get size of #snd_ctl_elem_list_t. * \return size in bytes */ size_t snd_ctl_elem_list_sizeof() @@ -2025,7 +2049,10 @@ size_t snd_ctl_elem_list_sizeof() } /** - * \brief allocate an invalid #snd_ctl_elem_list_t using standard malloc + * \brief allocate a #snd_ctl_elem_list_t using standard malloc. + * + * The memory can be released using snd_ctl_elem_list_free(). + * * \param ptr returned pointer * \return 0 on success otherwise negative error code */ @@ -2039,7 +2066,15 @@ int snd_ctl_elem_list_malloc(snd_ctl_elem_list_t **ptr) } /** - * \brief frees a previously allocated #snd_ctl_elem_list_t + * \brief frees a previously allocated #snd_ctl_elem_list_t. + * + * Release memory previously allocated using + * snd_ctl_elem_list_malloc(). + * + * If you used snd_ctl_elem_list_alloc_space() on the list, you must + * use snd_ctl_elem_list_free_space() \em before calling this + * function. + * * \param obj pointer to object to free */ void snd_ctl_elem_list_free(snd_ctl_elem_list_t *obj) @@ -2048,7 +2083,15 @@ void snd_ctl_elem_list_free(snd_ctl_elem_list_t *obj) } /** - * \brief clear given #snd_ctl_elem_list_t object + * \brief Clear given #snd_ctl_elem_list_t object. + * + * This will make the stored identifiers inaccessible without freeing + * their space. + * + * \warning The element identifier space cannot be freed after calling + * this function. Therefore, snd_ctl_elem_list_free_space() + * must be called in advance. + * * \param obj pointer to object to clear */ void snd_ctl_elem_list_clear(snd_ctl_elem_list_t *obj) @@ -2057,7 +2100,11 @@ void snd_ctl_elem_list_clear(snd_ctl_elem_list_t *obj) } /** - * \brief copy one #snd_ctl_elem_list_t to another + * \brief copy one #snd_ctl_elem_list_t to another. + * + * This performs a shallow copy. That means the both lists will share + * the same space for the elements. The elements will not be copied. + * * \param dst pointer to destination * \param src pointer to source */ @@ -2080,6 +2127,12 @@ void snd_ctl_elem_list_set_offset(snd_ctl_elem_list_t *obj, unsigned int val) /** * \brief Get number of used entries in CTL element identifiers list + * + * This function returns how many entries are actually filled with + * useful information. + * + * See also snd_ctl_elem_list_get_count(). + * * \param obj CTL element identifier list * \return number of used entries */ @@ -2090,7 +2143,14 @@ unsigned int snd_ctl_elem_list_get_used(const snd_ctl_elem_list_t *obj) } /** - * \brief Get total count of elements present in CTL device (information present in every filled CTL element identifiers list) + * \brief Get total count of elements present in CTL device + * + * This function returns how many entries were allocated using + * snd_ctl_elem_list_alloc_space(). This information is present after + * snd_ctl_elem_list() was called. + * + * See also snd_ctl_elem_list_get_used(). + * * \param obj CTL element identifier list * \return total number of elements */ @@ -2140,7 +2200,7 @@ snd_ctl_elem_iface_t snd_ctl_elem_list_get_interface(const snd_ctl_elem_list_t * } /** - * \brief Get device part of CTL element identifier for an entry of a CTL element identifiers list + * \brief Get the device part of CTL element identifier for an entry of a CTL element identifiers list * \param obj CTL element identifier list * \param idx Index of entry * \return CTL element related device -- 2.17.1

1 0

Re: [PATCH] dt-bindings: Whitespace clean-ups in schema files
by Luca Ceresoli 14 Aug '20

14 Aug '20

Hi Rob, On 12/08/20 22:36, Rob Herring wrote: > Clean-up incorrect indentation, extra spaces, long lines, and missing > EOF newline in schema files. Most of the clean-ups are for list > indentation which should always be 2 spaces more than the preceding > keyword. > > Found with yamllint (which I plan to integrate into the checks). [...] > diff --git a/Documentation/devicetree/bindings/clock/idt,versaclock5.yaml b/Documentation/devicetree/bindings/clock/idt,versaclock5.yaml > index 3d4e1685cc55..28c6461b9a9a 100644 > --- a/Documentation/devicetree/bindings/clock/idt,versaclock5.yaml > +++ b/Documentation/devicetree/bindings/clock/idt,versaclock5.yaml > @@ -95,10 +95,10 @@ allOf: > # Devices without builtin crystal > properties: > clock-names: > - minItems: 1 > - maxItems: 2 > - items: > - enum: [ xin, clkin ] > + minItems: 1 > + maxItems: 2 > + items: > + enum: [ xin, clkin ] > clocks: > minItems: 1 > maxItems: 2 Thanks for noticing, LGTM. [...] > diff --git a/Documentation/devicetree/bindings/input/touchscreen/touchscreen.yaml b/Documentation/devicetree/bindings/input/touchscreen/touchscreen.yaml > index d7dac16a3960..36dc7b56a453 100644 > --- a/Documentation/devicetree/bindings/input/touchscreen/touchscreen.yaml > +++ b/Documentation/devicetree/bindings/input/touchscreen/touchscreen.yaml > @@ -33,8 +33,8 @@ properties: > $ref: /schemas/types.yaml#/definitions/uint32 > > touchscreen-min-pressure: > - description: minimum pressure on the touchscreen to be achieved in order for the > - touchscreen driver to report a touch event. > + description: minimum pressure on the touchscreen to be achieved in order > + for the touchscreen driver to report a touch event. Out of personal taste, I find the original layout more pleasant and readable. This third option is also good, especially for long descriptions: description: minimum pressure on the touchscreen to be achieved in order for the touchscreen driver to report a touch event. At first glance yamllint seems to support exactly these two by default: > With indentation: {spaces: 4, check-multi-line-strings: true} > > the following code snippet would PASS: > > Blaise Pascal: > Je vous écris une longue lettre parce que > je n'ai pas le temps d'en écrire une courte. > > the following code snippet would PASS: > > Blaise Pascal: Je vous écris une longue lettre parce que > je n'ai pas le temps d'en écrire une courte. > > the following code snippet would FAIL: > > Blaise Pascal: Je vous écris une longue lettre parce que > je n'ai pas le temps d'en écrire une courte. > (https://yamllint.readthedocs.io/en/stable/rules.html#module-yamllint.rules.…) -- Luca

2 2

[PATCH v2 0/2] Add mediatek codec mt6359 driver
by Jiaxin Yu 14 Aug '20

14 Aug '20

Add mediatek codec (MT6359) driver MT6359 support playback and capture feature. On downlink path, it includes three DACs for handset, headset, and lineout path. On unlink path, it includeds three ADCs for main mic, second mic, 3rd mic, and headset mic. By scenario, select *_MUX widget to create damp path. And by select mic_type_mux to choose DMIC/AMIC/.... For example, select these MUX widget to create headset path (1) DAC In Mux --> "Normal Path" (2) HPL Mux --> "Audio Playback" (3) HPR Mux --> "Audio Playback" v2 changes: 1. Move playback_gpio/capture_gpio to the machine driver. 2. Fix mic_type(ACC/DMIC/DCC/DCC_*) at init stage. 3. Move devm_regulor_get to mt6359_platform_driver_probe. 4. Add relulator_disable in remove function. 5. Use stereo controls to the volume control. 6. Use SOC_ENUM_SINGLE_DECL instead of SOC_VALUE_ENUM_SINGLE_DECL. 7. Cleanup unused code. v1 changes: 1.lkml link: https://lkml.org/lkml/2020/3/5/1257 Jiaxin Yu (2): ASoC: mediatek: mt6359: add codec driver dt-bindings: mediatek: mt6359: add codec document .../devicetree/bindings/sound/mt6359.yaml | 68 + sound/soc/codecs/Kconfig | 8 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/mt6359.c | 2966 ++++++++++++++++++++ sound/soc/codecs/mt6359.h | 2653 +++++++++++++++++ 5 files changed, 5697 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/mt6359.yaml create mode 100644 sound/soc/codecs/mt6359.c create mode 100644 sound/soc/codecs/mt6359.h -- 1.8.1.1.dirty

3 9

[PATCH] ALSA: hda/realtek: Add quirk for Samsung Galaxy Flex Book
by Mike Pozulp 14 Aug '20

14 Aug '20

The Flex Book uses the same ALC298 codec as other Samsung laptops which have the no headphone sound bug, like my Samsung Notebook. The Flex Book owner used Early Patching to confirm that this quirk fixes the bug. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207423 Signed-off-by: Mike Pozulp <pozulp.kernel(a)gmail.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 2477f3ed7237..449ea64919ec 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7688,6 +7688,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x144d, 0xc109, "Samsung Ativ book 9 (NP900X3G)", ALC269_FIXUP_INV_DMIC), SND_PCI_QUIRK(0x144d, 0xc169, "Samsung Notebook 9 Pen (NP930SBE-K01US)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET), SND_PCI_QUIRK(0x144d, 0xc176, "Samsung Notebook 9 Pro (NP930MBE-K04US)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET), + SND_PCI_QUIRK(0x144d, 0xc189, "Samsung Galaxy Flex Book (NT950QCG-X716)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET), SND_PCI_QUIRK(0x144d, 0xc740, "Samsung Ativ book 8 (NP870Z5G)", ALC269_FIXUP_ATIV_BOOK_8), SND_PCI_QUIRK(0x144d, 0xc812, "Samsung Notebook Pen S (NT950SBE-X58)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET), SND_PCI_QUIRK(0x1458, 0xfa53, "Gigabyte BXBT-2807", ALC283_FIXUP_HEADSET_MIC), -- 2.26.2

2 1

[PATCH] soundwire: SDCA: detect sdca_cascade interrupt
by Bard Liao 14 Aug '20

14 Aug '20

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> The SoundWire 1.2 specification defines an "SDCA cascade" bit which handles a logical OR of all SDCA interrupt sources (up to 30 defined). Due to limitations of the addressing space, this bit is located in the SDW_DP0_INT register when DP0 is used, or alternatively in the DP0_SDCA_Support_INTSTAT register when DP0 is not used. To allow for both cases to be handled, this bit will be checked in the main device-level interrupt handling code. This will result in the register being read twice if DP0 is enabled, but it's not clear how to optimize this case. It's also more logical to deal with this interrupt at the device than the port level, this bit is really not DP0 specific and its location in the DP0_INTSTAT bit is only due to the lack of free space in SCP_INTSTAT_1. The SDCA_Cascade bit cannot be masked or cleared, so the interrupt handling only forwards the detection to the Slave driver, which will deal with reading the relevant SDCA status bits and clearing them. The bus driver only signals the detection. The communication with the Slave driver is based on the same interrupt callback, with only an extension to provide the status of the sdca_cascade bit. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Rander Wang <rander.wang(a)linux.intel.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> --- drivers/soundwire/bus.c | 14 ++++++++++++++ include/linux/soundwire/sdw.h | 4 ++++ 2 files changed, 18 insertions(+) diff --git a/drivers/soundwire/bus.c b/drivers/soundwire/bus.c index e6e0fb9a81b4..17c697f5ad78 100644 --- a/drivers/soundwire/bus.c +++ b/drivers/soundwire/bus.c @@ -1361,6 +1361,7 @@ static int sdw_handle_slave_alerts(struct sdw_slave *slave) int port_num, stat, ret, count = 0; unsigned long port; bool slave_notify = false; + bool sdca_cascade = false; u8 buf, buf2[2], _buf, _buf2[2]; sdw_modify_slave_status(slave, SDW_SLAVE_ALERT); @@ -1388,6 +1389,18 @@ static int sdw_handle_slave_alerts(struct sdw_slave *slave) goto io_err; } + if (slave->prop.is_sdca) { + ret = sdw_read(slave, SDW_DP0_INT); + if (ret < 0) { + dev_err(slave->bus->dev, + "SDW_DP0_INT read failed:%d\n", ret); + goto io_err; + } + sdca_cascade = ret & SDW_DP0_SDCA_CASCADE; + if (sdca_cascade) + slave_notify = true; + } + do { /* * Check parity, bus clash and Slave (impl defined) @@ -1453,6 +1466,7 @@ static int sdw_handle_slave_alerts(struct sdw_slave *slave) /* Update the Slave driver */ if (slave_notify && slave->ops && slave->ops->interrupt_callback) { + slave_intr.sdca_cascade = sdca_cascade; slave_intr.control_port = clear; memcpy(slave_intr.port, &port_status, sizeof(slave_intr.port)); diff --git a/include/linux/soundwire/sdw.h b/include/linux/soundwire/sdw.h index 76052f12c9f7..a51494646a2c 100644 --- a/include/linux/soundwire/sdw.h +++ b/include/linux/soundwire/sdw.h @@ -355,6 +355,7 @@ struct sdw_dpn_prop { * @dp0_prop: Data Port 0 properties * @src_dpn_prop: Source Data Port N properties * @sink_dpn_prop: Sink Data Port N properties + * @is_sdca: the Slave supports the SDCA specification */ struct sdw_slave_prop { u32 mipi_revision; @@ -376,6 +377,7 @@ struct sdw_slave_prop { struct sdw_dp0_prop *dp0_prop; struct sdw_dpn_prop *src_dpn_prop; struct sdw_dpn_prop *sink_dpn_prop; + bool is_sdca; }; /** @@ -465,10 +467,12 @@ struct sdw_slave_id { /** * struct sdw_slave_intr_status - Slave interrupt status + * @sdca_cascade: set if the Slave device reports an SDCA interrupt * @control_port: control port status * @port: data port status */ struct sdw_slave_intr_status { + bool sdca_cascade; u8 control_port; u8 port[15]; }; -- 2.17.1

1 0

[PATCH] soundwire: SDCA: add helper macro to access controls
by Bard Liao 14 Aug '20

14 Aug '20

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> The upcoming SDCA (SoundWire Device Class Audio) specification defines a hiearchical encoding to interface with Class-defined capabilities, based on which audio function, entity, control and channel being used. The specification is not yet accessible to the general public but this information is released with explicit permission from the MIPI Board to avoid delays with SDCA support on Linux platforms. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Rander Wang <rander.wang(a)linux.intel.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> --- include/linux/soundwire/sdw_registers.h | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/include/linux/soundwire/sdw_registers.h b/include/linux/soundwire/sdw_registers.h index 5d3c271af7d1..3ca09a02b9fc 100644 --- a/include/linux/soundwire/sdw_registers.h +++ b/include/linux/soundwire/sdw_registers.h @@ -305,4 +305,14 @@ #define SDW_CASC_PORT_MASK_INTSTAT3 1 #define SDW_CASC_PORT_REG_OFFSET_INTSTAT3 2 +/* v1.2 device - SDCA address mapping */ +#define SDW_SDCA_CTL(fun, ent, ctl, ch) (BIT(30) | \ + (((fun) & 0x7) << 22) | \ + (((ent) & 0x40) << 15) | \ + (((ent) & 0x3f) << 7) | \ + (((ctl) & 0x30) << 15) | \ + (((ctl) & 0x0f) << 3) | \ + (((ch) & 0x38) << 12) | \ + ((ch) & 0x07)) + #endif /* __SDW_REGISTERS_H */ -- 2.17.1

1 0

[PATCH v2] ASoC: Intel: boards: eve: Fix DMIC records zero
by Harsha Priya 13 Aug '20

13 Aug '20

This patch adds a dapm route to provide early mclk/sclk for for DMIC on SSP0(rt5514) and Headset on SSP1(rt5663). The sclk rate for both codecs is different which is taken care of in the platform_clock_control().The platform has one mclk and one sclk. Two variables sclk0 and sclk1 are used for the two codecs on differnet ssp that use different clock rate. This change ensures the DMIC PCM port will return valid data Signed-off-by: Brent Lu <brent.lu(a)intel.com> Signed-off-by: Vamshi Krishna Gopal <vamshi.krishna.gopal(a)intel.com> Signed-off-by: Harsha Priya <harshapriya.n(a)intel.com> --- v1 -> v2: - Only one mclk with same rate is used, so changed to using one variable - dropping ssp_ prefix from sclk variable names to make them sound right. - removing a return statment that was not required - fixed commit message accordingly .../soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 146 ++++++++++++++------- 1 file changed, 97 insertions(+), 49 deletions(-) diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c index b34cf6c..155f2b4 100644 --- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c @@ -54,7 +54,8 @@ struct kbl_codec_private { struct list_head hdmi_pcm_list; struct snd_soc_jack kabylake_hdmi[2]; struct clk *mclk; - struct clk *sclk; + struct clk *sclk0; + struct clk *sclk1; }; enum { @@ -77,13 +78,29 @@ static const struct snd_kcontrol_new kabylake_controls[] = { }; static int platform_clock_control(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *k, int event) + struct snd_kcontrol *k, int event, int ssp_num) { struct snd_soc_dapm_context *dapm = w->dapm; struct snd_soc_card *card = dapm->card; struct kbl_codec_private *priv = snd_soc_card_get_drvdata(card); + struct clk *sclk; + unsigned long sclk_rate; int ret = 0; + switch (ssp_num) { + case 0: + sclk = priv->sclk0; + sclk_rate = 6144000; + break; + case 1: + sclk = priv->sclk1; + sclk_rate = 3072000; + break; + default: + dev_err(card->dev, "Invalid ssp_num %d\n", ssp_num); + return -EINVAL; + } + /* * MCLK/SCLK need to be ON early for a successful synchronization of * codec internal clock. And the clocks are turned off during @@ -91,38 +108,48 @@ static int platform_clock_control(struct snd_soc_dapm_widget *w, */ switch (event) { case SND_SOC_DAPM_PRE_PMU: - /* Enable MCLK */ ret = clk_set_rate(priv->mclk, 24000000); if (ret < 0) { - dev_err(card->dev, "Can't set rate for mclk, err: %d\n", - ret); - return ret; + dev_err(card->dev, "Can't set rate for mclk for ssp%d, err: %d\n", + ssp_num, ret); + return ret; } - ret = clk_prepare_enable(priv->mclk); - if (ret < 0) { - dev_err(card->dev, "Can't enable mclk, err: %d\n", ret); - return ret; + if (!__clk_is_enabled(priv->mclk)) { + /* Enable MCLK */ + ret = clk_prepare_enable(priv->mclk); + if (ret < 0) { + dev_err(card->dev, "Can't enable mclk for ssp%d, err: %d\n", + ssp_num, ret); + return ret; + } } - /* Enable SCLK */ - ret = clk_set_rate(priv->sclk, 3072000); + ret = clk_set_rate(sclk, sclk_rate); if (ret < 0) { - dev_err(card->dev, "Can't set rate for sclk, err: %d\n", - ret); + dev_err(card->dev, "Can't set rate for sclk for ssp%d, err: %d\n", + ssp_num, ret); clk_disable_unprepare(priv->mclk); return ret; } - ret = clk_prepare_enable(priv->sclk); - if (ret < 0) { - dev_err(card->dev, "Can't enable sclk, err: %d\n", ret); - clk_disable_unprepare(priv->mclk); + if (!__clk_is_enabled(sclk)) { + /* Enable SCLK */ + ret = clk_prepare_enable(sclk); + if (ret < 0) { + dev_err(card->dev, "Can't enable sclk for ssp%d, err: %d\n", + ssp_num, ret); + clk_disable_unprepare(priv->mclk); + return ret; + } } break; case SND_SOC_DAPM_POST_PMD: - clk_disable_unprepare(priv->mclk); - clk_disable_unprepare(priv->sclk); + if (__clk_is_enabled(priv->mclk)) + clk_disable_unprepare(priv->mclk); + + if (__clk_is_enabled(sclk)) + clk_disable_unprepare(sclk); break; default: return 0; @@ -131,6 +158,18 @@ static int platform_clock_control(struct snd_soc_dapm_widget *w, return 0; } +static int platform_clock_control_ssp0(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + return platform_clock_control(w, k, event, 0); +} + +static int platform_clock_control_ssp1(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + return platform_clock_control(w, k, event, 1); +} + static const struct snd_soc_dapm_widget kabylake_widgets[] = { SND_SOC_DAPM_HP("Headphone Jack", NULL), SND_SOC_DAPM_MIC("Headset Mic", NULL), @@ -139,15 +178,17 @@ static const struct snd_soc_dapm_widget kabylake_widgets[] = { SND_SOC_DAPM_MIC("DMIC", NULL), SND_SOC_DAPM_SPK("HDMI1", NULL), SND_SOC_DAPM_SPK("HDMI2", NULL), - SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0, - platform_clock_control, SND_SOC_DAPM_PRE_PMU | + SND_SOC_DAPM_SUPPLY("Platform Clock SSP0", SND_SOC_NOPM, 0, 0, + platform_clock_control_ssp0, SND_SOC_DAPM_PRE_PMU | + SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_SUPPLY("Platform Clock SSP1", SND_SOC_NOPM, 0, 0, + platform_clock_control_ssp1, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), - }; static const struct snd_soc_dapm_route kabylake_map[] = { /* Headphones */ - { "Headphone Jack", NULL, "Platform Clock" }, + { "Headphone Jack", NULL, "Platform Clock SSP1" }, { "Headphone Jack", NULL, "HPOL" }, { "Headphone Jack", NULL, "HPOR" }, @@ -156,7 +197,7 @@ static const struct snd_soc_dapm_route kabylake_map[] = { { "Right Spk", NULL, "Right BE_OUT" }, /* other jacks */ - { "Headset Mic", NULL, "Platform Clock" }, + { "Headset Mic", NULL, "Platform Clock SSP1" }, { "IN1P", NULL, "Headset Mic" }, { "IN1N", NULL, "Headset Mic" }, @@ -180,6 +221,7 @@ static const struct snd_soc_dapm_route kabylake_map[] = { { "ssp0 Rx", NULL, "Right HiFi Capture" }, /* DMIC */ + { "DMIC", NULL, "Platform Clock SSP0" }, { "DMIC1L", NULL, "DMIC" }, { "DMIC1R", NULL, "DMIC" }, { "DMIC2L", NULL, "DMIC" }, @@ -666,7 +708,7 @@ static int kabylake_set_bias_level(struct snd_soc_card *card, if (!component || strcmp(component->name, RT5514_DEV_NAME)) return 0; - if (IS_ERR(priv->mclk)) + if (IS_ERR(priv->mclk0)) return 0; /* @@ -757,6 +799,28 @@ static struct snd_soc_card kabylake_audio_card = { .late_probe = kabylake_card_late_probe, }; +static int kabylake_audio_clk_get(struct device *dev, const char *id, + struct clk **clk) +{ + int ret = 0; + + if (!clk) + return -EINVAL; + + *clk = devm_clk_get(dev, id); + if (IS_ERR(*clk)) { + ret = PTR_ERR(*clk); + if (ret == -ENOENT) { + dev_info(dev, "Failed to get %s, defer probe\n", id); + return -EPROBE_DEFER; + } + + dev_err(dev, "Failed to get %s with err:%d\n", id, ret); + } + + return ret; +} + static int kabylake_audio_probe(struct platform_device *pdev) { struct kbl_codec_private *ctx; @@ -777,33 +841,17 @@ static int kabylake_audio_probe(struct platform_device *pdev) dmic_constraints = mach->mach_params.dmic_num == 2 ? &constraints_dmic_2ch : &constraints_dmic_channels; - ctx->mclk = devm_clk_get(&pdev->dev, "ssp1_mclk"); - if (IS_ERR(ctx->mclk)) { - ret = PTR_ERR(ctx->mclk); - if (ret == -ENOENT) { - dev_info(&pdev->dev, - "Failed to get ssp1_mclk, defer probe\n"); - return -EPROBE_DEFER; - } - - dev_err(&pdev->dev, "Failed to get ssp1_mclk with err:%d\n", - ret); + ret = kabylake_audio_clk_get(&pdev->dev, "ssp0_sclk", &ctx->sclk0); + if (ret != 0) return ret; - } - ctx->sclk = devm_clk_get(&pdev->dev, "ssp1_sclk"); - if (IS_ERR(ctx->sclk)) { - ret = PTR_ERR(ctx->sclk); - if (ret == -ENOENT) { - dev_info(&pdev->dev, - "Failed to get ssp1_sclk, defer probe\n"); - return -EPROBE_DEFER; - } + ret = kabylake_audio_clk_get(&pdev->dev, "ssp1_mclk", &ctx->mclk); + if (ret != 0) + return ret; - dev_err(&pdev->dev, "Failed to get ssp1_sclk with err:%d\n", - ret); + ret = kabylake_audio_clk_get(&pdev->dev, "ssp1_sclk", &ctx->sclk1); + if (ret != 0) return ret; - } return devm_snd_soc_register_card(&pdev->dev, &kabylake_audio_card); } -- 2.7.4

5 9

[PATCH] pcm: dmix: fix access to sum-buffer in non-interleaved mixing mode
by Vijay Palaniswamy (RBEI/ECF3) 13 Aug '20

13 Aug '20

Hi All, When dmix uses non-interleaved mixing mode the offset and step width to sum_buffer was calculated by using the dmix channels instead of the slave channels. This leads to audio distortions due to frame corruption. Issue happens while doing channel wise mixing of src and destination buffers. Attached the patch to fix the offset and step width calculation based on slave channels for sum buffer. So mixed data will be saved in respective channels of sum buffer. Steps to reproduce the issue: With below configuratio, Do aplay on both device in parallel for audio distortion pcm.dmix_2_channels { type dmix ipc_key 5678293 ipc_perm 0660 ipc_gid audio bindings [0 1] slave { pcm "hardware" channels 2 periods 4 period_time 40000 } } pcm.dmix_1_channels { type dmix ipc_key 5678293 ipc_perm 0660 ipc_gid audio bindings [0] slave { pcm "hardware" channels 1 periods 4 period_time 40000 } } pcm.hardware { type hw card 0 channels 2 rate 48000 format S16_LE } Best regards Palaniswamy Vijay

2 1

pcm: dmix: fix access to sum-buffer in non-interleaved mixing mode
by GitHub pull_request - opened 13 Aug '20

13 Aug '20

alsa-project/alsa-lib pull request #78 was opened from vijaypalaniswamy: When dmix uses non-interleaved mixing mode the offset and step width to sum_buffer was calculated by using the dmix channels instead of the slave channels. This leads to audio distortions due to frame corruption. example: - With below configuratio, Do aplay on both device in parallel for audio distortion pcm.dmix_2_channels { type dmix ipc_key 5678293 ipc_perm 0660 ipc_gid audio bindings [0 1] slave { pcm "hardware" channels 2 periods 4 period_time 40000 } } pcm.dmix_1_channels { type dmix ipc_key 5678293 ipc_perm 0660 ipc_gid audio bindings [0] slave { pcm "hardware" channels 1 periods 4 period_time 40000 } } pcm.hardware { type hw card 0 channels 2 rate 48000 format S16_LE } Signed-off-by: Vijay Palaniswamy <vijay.palaniswamy(a)in.bosch.com> Request URL : https://github.com/alsa-project/alsa-lib/pull/78 Patch URL : https://github.com/alsa-project/alsa-lib/pull/78.patch Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH] ALSA: usb-audio: ignore broken processing/extension unit
by tom.ty89＠gmail.com 13 Aug '20

13 Aug '20

From: Tom Yan <tom.ty89(a)gmail.com> Some devices have broken extension unit where getting current value doesn't work. Attempt that once when creating mixer control for it. If it fails, just ignore it, so that it won't cripple the device entirely (and/or make the error floods). --- sound/usb/mixer.c | 8 +++++++- 1 file changed, 7 insertions(+), 1 deletion(-) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index eab0fd4fd7c3..e0b7174c1043 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -2367,7 +2367,7 @@ static int build_audio_procunit(struct mixer_build *state, int unitid, int num_ins; struct usb_mixer_elem_info *cval; struct snd_kcontrol *kctl; - int i, err, nameid, type, len; + int i, err, nameid, type, len, val; const struct procunit_info *info; const struct procunit_value_info *valinfo; const struct usbmix_name_map *map; @@ -2470,6 +2470,12 @@ static int build_audio_procunit(struct mixer_build *state, int unitid, break; } + err = get_cur_ctl_value(cval, cval->control << 8, &val); + if (err < 0) { + usb_mixer_elem_info_free(cval); + return -EINVAL; + } + kctl = snd_ctl_new1(&mixer_procunit_ctl, cval); if (!kctl) { usb_mixer_elem_info_free(cval); -- 2.28.0

1 0

[PATCH] [v2] ALSA: echoaudio: Fix potential Oops in snd_echo_resume()
by Dinghao Liu 13 Aug '20

13 Aug '20

Freeing chip on error may lead to an Oops at the next time the system goes to resume. Fix this by removing all snd_echo_free() calls on error. Fixes: 47b5d028fdce8 ("ALSA: Echoaudio - Add suspend support #2") Signed-off-by: Dinghao Liu <dinghao.liu(a)zju.edu.cn> --- Changelog: v2: - Remove snd_echo_free() calls in all error paths. --- sound/pci/echoaudio/echoaudio.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c index 6aeb99aa2414..868f1dcd27d6 100644 --- a/sound/pci/echoaudio/echoaudio.c +++ b/sound/pci/echoaudio/echoaudio.c @@ -2213,7 +2213,6 @@ static int snd_echo_resume(struct device *dev) if (err < 0) { kfree(commpage_bak); dev_err(dev, "resume init_hw err=%d\n", err); - snd_echo_free(chip); return err; } @@ -2240,7 +2239,6 @@ static int snd_echo_resume(struct device *dev) if (request_irq(pci->irq, snd_echo_interrupt, IRQF_SHARED, KBUILD_MODNAME, chip)) { dev_err(chip->card->dev, "cannot grab irq\n"); - snd_echo_free(chip); return -EBUSY; } chip->irq = pci->irq; -- 2.17.1

2 1

[PATCH] ASoC: intel: Fix memleak in sst_media_open
by Dinghao Liu 13 Aug '20

13 Aug '20

When power_up_sst() fails, stream needs to be freed just like when try_module_get() fails. However, current code is returning directly and ends up leaking memory. Fixes: 0121327c1a68b ("ASoC: Intel: mfld-pcm: add control for powering up/down dsp") Signed-off-by: Dinghao Liu <dinghao.liu(a)zju.edu.cn> --- sound/soc/intel/atom/sst-mfld-platform-pcm.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/intel/atom/sst-mfld-platform-pcm.c b/sound/soc/intel/atom/sst-mfld-platform-pcm.c index 49b9f18472bc..79fedf9e3da1 100644 --- a/sound/soc/intel/atom/sst-mfld-platform-pcm.c +++ b/sound/soc/intel/atom/sst-mfld-platform-pcm.c @@ -330,8 +330,10 @@ static int sst_media_open(struct snd_pcm_substream *substream, runtime->private_data = stream; ret_val = power_up_sst(stream); - if (ret_val < 0) + if (ret_val < 0) { + kfree(stream); return ret_val; + } /* Make sure, that the period size is always even */ snd_pcm_hw_constraint_step(substream->runtime, 0, -- 2.17.1

3 2

Re: [PATCH] dt-bindings: Whitespace clean-ups in schema files
by Geert Uytterhoeven 13 Aug '20

13 Aug '20

Hi Rob, On Wed, Aug 12, 2020 at 10:36 PM Rob Herring <robh(a)kernel.org> wrote: > Clean-up incorrect indentation, extra spaces, long lines, and missing > EOF newline in schema files. Most of the clean-ups are for list > indentation which should always be 2 spaces more than the preceding > keyword. > > Found with yamllint (which I plan to integrate into the checks). > Signed-off-by: Rob Herring <robh(a)kernel.org> Thanks for your patch! > --- a/Documentation/devicetree/bindings/clock/renesas,cpg-clocks.yaml > +++ b/Documentation/devicetree/bindings/clock/renesas,cpg-clocks.yaml > @@ -24,9 +24,9 @@ properties: > - const: renesas,r8a7778-cpg-clocks # R-Car M1 > - const: renesas,r8a7779-cpg-clocks # R-Car H1 > - items: > - - enum: > - - renesas,r7s72100-cpg-clocks # RZ/A1H > - - const: renesas,rz-cpg-clocks # RZ/A1 > + - enum: > + - renesas,r7s72100-cpg-clocks # RZ/A1H > + - const: renesas,rz-cpg-clocks # RZ/A1 This change breaks alignment of the comments at the end of each line. > - const: renesas,sh73a0-cpg-clocks # SH-Mobile AG5 (I only checked the files I care about) If you don't update commit e0fe7fc6f2ca0781 ("dt-bindings: Whitespace clean-ups in schema files"), I can send a patch after v5.9-rc1. Gr{oetje,eeting}s, Geert -- Geert Uytterhoeven -- There's lots of Linux beyond ia32 -- geert(a)linux-m68k.org In personal conversations with technical people, I call myself a hacker. But when I'm talking to journalists I just say "programmer" or something like that. -- Linus Torvalds

1 0

[PATCH] ALSA: echoaudio: Fix memory leak in snd_echo_resume()
by Dinghao Liu 13 Aug '20

13 Aug '20

When restore_dsp_rettings() fails, chip should be freed just like when init_hw() and request_irq() fails. Fixes: 47b5d028fdce8 ("ALSA: Echoaudio - Add suspend support #2") Signed-off-by: Dinghao Liu <dinghao.liu(a)zju.edu.cn> --- sound/pci/echoaudio/echoaudio.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c index 6aeb99aa2414..2aa183fe5dc1 100644 --- a/sound/pci/echoaudio/echoaudio.c +++ b/sound/pci/echoaudio/echoaudio.c @@ -2226,6 +2226,7 @@ static int snd_echo_resume(struct device *dev) chip->pipe_alloc_mask = pipe_alloc_mask; if (err < 0) { kfree(commpage_bak); + snd_echo_free(chip); return err; } -- 2.17.1

3 2

[PATCH] ALSA: pci/asihpi: Fix memory leak in snd_card_asihpi_capture_open()
by Dinghao Liu 13 Aug '20

13 Aug '20

When snd_pcm_hw_constraint_pow2() fails, dpcm should be freed just like when hpi_instream_open() fails. Fixes: 719f82d3987aa ("ALSA: Add support of AudioScience ASI boards") Signed-off-by: Dinghao Liu <dinghao.liu(a)zju.edu.cn> --- sound/pci/asihpi/asihpi.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/pci/asihpi/asihpi.c b/sound/pci/asihpi/asihpi.c index 023c35a2a951..736eceacd3d0 100644 --- a/sound/pci/asihpi/asihpi.c +++ b/sound/pci/asihpi/asihpi.c @@ -1244,8 +1244,10 @@ static int snd_card_asihpi_capture_open(struct snd_pcm_substream *substream) if (card->can_dma) err = snd_pcm_hw_constraint_pow2(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_BYTES); - if (err < 0) + if (err < 0) { + kfree(dpcm); return err; + } snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, card->update_interval_frames); -- 2.17.1

3 2

Re: [PATCH] dt-bindings: Whitespace clean-ups in schema files
by Joe Perches 13 Aug '20

13 Aug '20

On Wed, 2020-08-12 at 14:36 -0600, Rob Herring wrote: > Clean-up incorrect indentation, extra spaces, long lines, and missing > EOF newline in schema files. Most of the clean-ups are for list > indentation which should always be 2 spaces more than the preceding ^ > keyword. [] > diff --git a/Documentation/devicetree/bindings/arm/arm,integrator.yaml b/Documentation/devicetree/bindings/arm/arm,integrator.yaml > index 192ded470e32..f0daf990e077 100644 > --- a/Documentation/devicetree/bindings/arm/arm,integrator.yaml > +++ b/Documentation/devicetree/bindings/arm/arm,integrator.yaml > @@ -67,9 +67,9 @@ patternProperties: > compatible: > items: > - enum: > - - arm,integrator-ap-syscon > - - arm,integrator-cp-syscon > - - arm,integrator-sp-syscon > + - arm,integrator-ap-syscon > + - arm,integrator-cp-syscon > + - arm,integrator-sp-syscon Confused a bit here. - enum: 10 spaces to dash old line: - arm,integrator-ap-syscon 12 spaces to dash new line: - arm,integrator-ap-syscon 14 spaces to dash Is it supposed to be 2 spaces more than the preceding line or 4 more?

2 1

[PATCH v3 1/2] dt-bindings: sound: add DT bindings for Microchip S/PDIF TX Controller
by Codrin Ciubotariu 12 Aug '20

12 Aug '20

This patch adds DT bindings for the new Microchip S/PDIF TX Controller embedded inside sama7g5 SoCs. Signed-off-by: Codrin Ciubotariu <codrin.ciubotariu(a)microchip.com> --- Changes in v3: - removed 'oneOf' from 'compatible' property; - added 'maxItems: 1' to 'dmas' property; - removed pinctrl related properties; Changes in v2: - replaced https with http; - reworked example, included bindings; .../bindings/sound/mchp,spdiftx.yaml | 75 +++++++++++++++++++ 1 file changed, 75 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/mchp,spdiftx.yaml diff --git a/Documentation/devicetree/bindings/sound/mchp,spdiftx.yaml b/Documentation/devicetree/bindings/sound/mchp,spdiftx.yaml new file mode 100644 index 000000000000..a03b0b871fc9 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/mchp,spdiftx.yaml @@ -0,0 +1,75 @@ +# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/mchp,spdiftx.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Microchip S/PDIF Tx Controller Device Tree Bindings + +maintainers: + - Codrin Ciubotariu <codrin.ciubotariu(a)microchip.com> + +description: + The Microchip Sony/Philips Digital Interface Transmitter is a + serial port compliant with the IEC-60958 standard. + +properties: + "#sound-dai-cells": + const: 0 + + compatible: + const: microchip,sama7g5-spdiftx + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + clocks: + items: + - description: Peripheral Bus Clock + - description: Generic Clock + + clock-names: + items: + - const: pclk + - const: gclk + + dmas: + description: TX DMA Channel + maxItems: 1 + + dma-names: + const: tx + +required: + - "#sound-dai-cells" + - compatible + - reg + - interrupts + - clocks + - clock-names + - dmas + - dma-names + +additionalProperties: false + +examples: + - | + #include <dt-bindings/clock/at91.h> + #include <dt-bindings/dma/at91.h> + #include <dt-bindings/interrupt-controller/arm-gic.h> + + spdiftx@e1618000 { + #sound-dai-cells = <0>; + compatible = "microchip,sama7g5-spdiftx"; + reg = <0xe1618000 0x4000>; + interrupts = <GIC_SPI 85 IRQ_TYPE_LEVEL_HIGH>; + dmas = <&dma0 AT91_XDMAC_DT_PERID(50)>; + dma-names = "tx"; + clocks = <&pmc PMC_TYPE_PERIPHERAL 85>, <&pmc PMC_TYPE_GCK 85>; + clock-names = "pclk", "gclk"; + pinctrl-names = "default"; + pinctrl-0 = <&pinctrl_spdiftx_default>; + }; -- 2.25.1

5 8

[PATCH] ALSA: hda/hdmi: Use force connectivity quirk on another HP desktop
by Kai-Heng Feng 12 Aug '20

12 Aug '20

There's another HP desktop has buggy BIOS which flags the Port Connectivity bit as no connection. Apply force connectivity quirk to enable DP/HDMI audio. Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> --- sound/pci/hda/patch_hdmi.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 4bbd12d3b1b5..b8c8490e568b 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1863,6 +1863,7 @@ static int hdmi_add_cvt(struct hda_codec *codec, hda_nid_t cvt_nid) } static const struct snd_pci_quirk force_connect_list[] = { + SND_PCI_QUIRK(0x103c, 0x870f, "HP", 1), SND_PCI_QUIRK(0x103c, 0x871a, "HP", 1), {} }; -- 2.17.1

2 3

[PATCH 0/3] Modernize tasklet callback API
by Kees Cook 12 Aug '20

12 Aug '20

Hi, This is the infrastructure changes to prepare the tasklet API for conversion to passing the tasklet struct as the callback argument instead of an arbitrary unsigned long. The first patch details why this is useful (it's the same rationale as the timer_struct changes from a bit ago: less abuse during memory corruption attacks, more in line with existing ways of doing things in the kernel, save a little space in struct, etc). Notably, the existing tasklet API use is much less messy, so there is less to clean up. It's not clear to me which tree this should go through... Greg since it starts with a USB clean-up, -tip for timer or interrupt, or if I should just carry it. I'm open to suggestions, but if I don't hear otherwise, I'll just carry it. My goal is to have this merged for v5.9-rc1 so that during the v5.10 development cycle the new API will be available. The entire tree of changes is here[1] currently, but to split it up by maintainer the infrastructure changes need to be landed first. Review and Acks appreciated! :) Thanks, -Kees [1] https://github.com/allenpais/tasklets/commits/tasklets_V2 Kees Cook (2): usb: gadget: udc: Avoid tasklet passing a global treewide: Replace DECLARE_TASKLET() with DECLARE_TASKLET_OLD() Romain Perier (1): tasklet: Introduce new initialization API drivers/input/keyboard/omap-keypad.c | 2 +- drivers/input/serio/hil_mlc.c | 2 +- drivers/net/wan/farsync.c | 4 +-- drivers/s390/crypto/ap_bus.c | 2 +- drivers/staging/most/dim2/dim2.c | 2 +- drivers/staging/octeon/ethernet-tx.c | 2 +- drivers/tty/vt/keyboard.c | 2 +- drivers/usb/gadget/udc/snps_udc_core.c | 6 ++--- drivers/usb/host/fhci-sched.c | 2 +- include/linux/interrupt.h | 37 ++++++++++++++++++++++---- kernel/backtracetest.c | 2 +- kernel/debug/debug_core.c | 2 +- kernel/irq/resend.c | 2 +- kernel/softirq.c | 18 ++++++++++++- net/atm/pppoatm.c | 2 +- net/iucv/iucv.c | 2 +- sound/drivers/pcsp/pcsp_lib.c | 2 +- 17 files changed, 66 insertions(+), 25 deletions(-) -- 2.25.1

10 26

[PATCH v2 1/2] ASoC: q6afe-dai: mark all widgets registers as SND_SOC_NOPM
by Srinivas Kandagatla 12 Aug '20

12 Aug '20

Looks like the q6afe-dai dapm widget registers are set as "0", which is a not correct. As this registers will be read by ASoC core during startup which will throw up errors, Fix this by making the registers as SND_SOC_NOPM as these should be never used. With recent changes to ASoC core, every register read/write failures are reported very verbosely. Prior to this fails to reads are totally ignored, so we never saw any error messages. Reported-by: John Stultz <john.stultz(a)linaro.org> Fixes: 24c4cbcfac09 ("ASoC: qdsp6: q6afe: Add q6afe dai driver") Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- Changes since v1: - replaced reg 0 with SND_SOC_NOPM and removed read callback sound/soc/qcom/qdsp6/q6afe-dai.c | 210 +++++++++++++++---------------- 1 file changed, 105 insertions(+), 105 deletions(-) diff --git a/sound/soc/qcom/qdsp6/q6afe-dai.c b/sound/soc/qcom/qdsp6/q6afe-dai.c index 2a5302f1db98..0168af849272 100644 --- a/sound/soc/qcom/qdsp6/q6afe-dai.c +++ b/sound/soc/qcom/qdsp6/q6afe-dai.c @@ -1150,206 +1150,206 @@ static int q6afe_of_xlate_dai_name(struct snd_soc_component *component, } static const struct snd_soc_dapm_widget q6afe_dai_widgets[] = { - SND_SOC_DAPM_AIF_IN("HDMI_RX", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_IN("SLIMBUS_0_RX", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_IN("SLIMBUS_1_RX", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_IN("SLIMBUS_2_RX", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_IN("SLIMBUS_3_RX", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_IN("SLIMBUS_4_RX", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_IN("SLIMBUS_5_RX", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_IN("SLIMBUS_6_RX", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_OUT("SLIMBUS_0_TX", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_OUT("SLIMBUS_1_TX", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_OUT("SLIMBUS_2_TX", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_OUT("SLIMBUS_3_TX", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_OUT("SLIMBUS_4_TX", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_OUT("SLIMBUS_5_TX", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_OUT("SLIMBUS_6_TX", NULL, 0, 0, 0, 0), + SND_SOC_DAPM_AIF_IN("HDMI_RX", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("SLIMBUS_0_RX", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("SLIMBUS_1_RX", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("SLIMBUS_2_RX", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("SLIMBUS_3_RX", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("SLIMBUS_4_RX", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("SLIMBUS_5_RX", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("SLIMBUS_6_RX", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("SLIMBUS_0_TX", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("SLIMBUS_1_TX", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("SLIMBUS_2_TX", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("SLIMBUS_3_TX", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("SLIMBUS_4_TX", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("SLIMBUS_5_TX", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("SLIMBUS_6_TX", NULL, 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_IN("QUAT_MI2S_RX", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_OUT("QUAT_MI2S_TX", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_IN("TERT_MI2S_RX", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_OUT("TERT_MI2S_TX", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_IN("SEC_MI2S_RX", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_OUT("SEC_MI2S_TX", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_IN("SEC_MI2S_RX_SD1", "Secondary MI2S Playback SD1", - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_IN("PRI_MI2S_RX", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_OUT("PRI_MI2S_TX", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_IN("PRIMARY_TDM_RX_0", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_IN("PRIMARY_TDM_RX_1", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_IN("PRIMARY_TDM_RX_2", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_IN("PRIMARY_TDM_RX_3", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_IN("PRIMARY_TDM_RX_4", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_IN("PRIMARY_TDM_RX_5", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_IN("PRIMARY_TDM_RX_6", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_IN("PRIMARY_TDM_RX_7", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_OUT("PRIMARY_TDM_TX_0", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_OUT("PRIMARY_TDM_TX_1", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_OUT("PRIMARY_TDM_TX_2", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_OUT("PRIMARY_TDM_TX_3", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_OUT("PRIMARY_TDM_TX_4", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_OUT("PRIMARY_TDM_TX_5", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_OUT("PRIMARY_TDM_TX_6", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_OUT("PRIMARY_TDM_TX_7", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_IN("SEC_TDM_RX_0", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_IN("SEC_TDM_RX_1", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_IN("SEC_TDM_RX_2", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_IN("SEC_TDM_RX_3", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_IN("SEC_TDM_RX_4", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_IN("SEC_TDM_RX_5", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_IN("SEC_TDM_RX_6", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_IN("SEC_TDM_RX_7", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_OUT("SEC_TDM_TX_0", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_OUT("SEC_TDM_TX_1", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_OUT("SEC_TDM_TX_2", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_OUT("SEC_TDM_TX_3", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_OUT("SEC_TDM_TX_4", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_OUT("SEC_TDM_TX_5", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_OUT("SEC_TDM_TX_6", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_OUT("SEC_TDM_TX_7", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_IN("TERT_TDM_RX_0", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_IN("TERT_TDM_RX_1", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_IN("TERT_TDM_RX_2", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_IN("TERT_TDM_RX_3", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_IN("TERT_TDM_RX_4", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_IN("TERT_TDM_RX_5", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_IN("TERT_TDM_RX_6", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_IN("TERT_TDM_RX_7", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_OUT("TERT_TDM_TX_0", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_OUT("TERT_TDM_TX_1", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_OUT("TERT_TDM_TX_2", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_OUT("TERT_TDM_TX_3", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_OUT("TERT_TDM_TX_4", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_OUT("TERT_TDM_TX_5", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_OUT("TERT_TDM_TX_6", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_OUT("TERT_TDM_TX_7", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_IN("QUAT_TDM_RX_0", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_IN("QUAT_TDM_RX_1", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_IN("QUAT_TDM_RX_2", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_IN("QUAT_TDM_RX_3", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_IN("QUAT_TDM_RX_4", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_IN("QUAT_TDM_RX_5", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_IN("QUAT_TDM_RX_6", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_IN("QUAT_TDM_RX_7", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_OUT("QUAT_TDM_TX_0", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_OUT("QUAT_TDM_TX_1", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_OUT("QUAT_TDM_TX_2", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_OUT("QUAT_TDM_TX_3", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_OUT("QUAT_TDM_TX_4", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_OUT("QUAT_TDM_TX_5", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_OUT("QUAT_TDM_TX_6", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_OUT("QUAT_TDM_TX_7", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_IN("QUIN_TDM_RX_0", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_IN("QUIN_TDM_RX_1", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_IN("QUIN_TDM_RX_2", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_IN("QUIN_TDM_RX_3", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_IN("QUIN_TDM_RX_4", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_IN("QUIN_TDM_RX_5", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_IN("QUIN_TDM_RX_6", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_IN("QUIN_TDM_RX_7", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_OUT("QUIN_TDM_TX_0", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_OUT("QUIN_TDM_TX_1", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_OUT("QUIN_TDM_TX_2", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_OUT("QUIN_TDM_TX_3", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_OUT("QUIN_TDM_TX_4", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_OUT("QUIN_TDM_TX_5", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_OUT("QUIN_TDM_TX_6", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_OUT("QUIN_TDM_TX_7", NULL, - 0, 0, 0, 0), - SND_SOC_DAPM_AIF_OUT("DISPLAY_PORT_RX", "NULL", 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("DISPLAY_PORT_RX", "NULL", 0, SND_SOC_NOPM, 0, 0), }; static const struct snd_soc_component_driver q6afe_dai_component = { -- 2.21.0

3 3

[PATCH] ASoC: Make soc_component_read() returning an error code again
by Takashi Iwai 12 Aug '20

12 Aug '20

Along with the recent unification of snd_soc_component_read*() functions, the behavior of snd_soc_component_read() was changed slightly; namely it returns the register read value directly, and even if an error happens, it returns zero (but it prints an error message). That said, the caller side can't know whether it's an error or not any longer. Ideally this shouldn't matter much, but in practice this seems causing a regression, as John reported. And, grepping the tree revealed that there are still plenty of callers that do check the error code, so we'll need to deal with them in anyway. As a quick band-aid over the regression, this patch changes the return value of snd_soc_component_read() again to the negative error code. It can't work, obviously, for 32bit register values, but it should be enough for the known regressions, so far. Fixes: cf6e26c71bfd ("ASoC: soc-component: merge snd_soc_component_read() and snd_soc_component_read32()") Reported-by: John Stultz <john.stultz(a)linaro.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/soc/soc-component.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-component.c b/sound/soc/soc-component.c index f0b4f4bc44a4..5504b92946e3 100644 --- a/sound/soc/soc-component.c +++ b/sound/soc/soc-component.c @@ -406,7 +406,7 @@ static unsigned int soc_component_read_no_lock( ret = -EIO; if (ret < 0) - soc_component_ret(component, ret); + return soc_component_ret(component, ret); return val; } -- 2.16.4

3 3

[PATCH] ALSA: hda/realtek - Fix unused variable warning
by Takashi Iwai 12 Aug '20

12 Aug '20

The previous fix forgot to remove the unused variable that triggers a compile warning now: sound/pci/hda/patch_realtek.c: In function 'alc285_fixup_hp_gpio_led': sound/pci/hda/patch_realtek.c:4163:19: warning: unused variable 'spec' [-Wunused-variable] Fix it. Fixes: 404690649e6a ("ALSA: hda - reverse the setting value in the micmute_led_set") Reported-by: Stephen Rothwell <sfr(a)canb.auug.org.au> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 073029aeaf3c..7f9d35273734 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4160,8 +4160,6 @@ static void alc269_fixup_hp_gpio_led(struct hda_codec *codec, static void alc285_fixup_hp_gpio_led(struct hda_codec *codec, const struct hda_fixup *fix, int action) { - struct alc_spec *spec = codec->spec; - alc_fixup_hp_gpio_led(codec, action, 0x04, 0x01); } -- 2.16.4

1 0

[PATCH] conf: USB-Audio: Disable IEC958 on Lenovo ThinkStation P620
by Kai-Heng Feng 12 Aug '20

12 Aug '20

Both USB audio cards on Lenovo ThinkStation P620 don't support IEC958, so disable IEC958 accordingly. Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> --- src/conf/cards/USB-Audio.conf | 2 ++ 1 file changed, 2 insertions(+) diff --git a/src/conf/cards/USB-Audio.conf b/src/conf/cards/USB-Audio.conf index 297629f3..9b64af3c 100644 --- a/src/conf/cards/USB-Audio.conf +++ b/src/conf/cards/USB-Audio.conf @@ -63,6 +63,8 @@ USB-Audio.pcm.iec958_device { "Scarlett 2i4 USB" 999 "Sennheiser USB headset" 999 "SWTOR Gaming Headset by Razer" 999 + "ThinkStation P620 Main" 999 + "ThinkStation P620 Rear" 999 "Thunderbolt Dock Audio Headset" 999 "Thunderbolt Dock Audio Module" 999 "USB Device 0x46d_0x821" 999 -- 2.17.1

2 4

[PATCH v3 00/13] ASoC: Intel: Catpt - Lynx and Wildcat point
by Cezary Rojewski 11 Aug '20

11 Aug '20

Implement support for Lynxpoint and Wildcat Point AudioDSP. Catpt solution deprecates existing sound/soc/intel/haswell which is removed in the following series. This cover-letter is followed by 'Developer's deep dive' message schedding light on catpt's key concepts and areas addressed. Due to high range of errors and desynchronization from recommendations set by Windows solution, re-write came as a lower-cost solution compared to refactoring /haswell/ with several series of patches. Special thanks go to Marcin Barlik and Piotr Papierkowski for sharing their LPT/WPT AudioDSP architecture expertise as well as helping backtrack its historical background. My thanks go to Amadeusz Slawinski for reviews and improvements proposed on and off the internal list. Most of internal diff below is his contribution. Krzysztof Hejmowski helped me setup my own Xtensa environment and recompile LPT/WPT FW binary sources what sped up the development greatly. This would not have been possible without help from these champions, especially considering how quickly the catpt was written: 2 weeks features, 3 weeks optimizations. Thank you. Userspace-exposed members are compatible with what is exposed by deprecated solution as well as FW binary being re-used thus no harm is done. The only visible differences are: the newly added 'Loopback Mute' kcontrol and volume support extending to quad from stereo. On top of fixing erros and design flows, catpt also adds module reload, dynamic SRAM memory allocation during PCM runtime and exposes missing userspace API: 'Loopback Mute' kcontrol, quad volume controls and sysfs fw-version entries. Event tracing is provided to easy solution debugging. Following are not included in this update and are scheduled as later addition: - fw logging - module (library) support Note: LPT power up/down sequences might get aligned with WPT once enough testing is done as capabilities are shared for both DSPs. Note #2: Both LPT and WPT power up/down sequences may get optimized in future updates as thanks to help from the Windows team, most of nuances behind why/what/when in regard to hw registers have been backtracked and reviewed again. Link to developer's deep dive message: https://www.spinics.net/lists/alsa-devel/msg113563.html Changes in v3: - fixed IRAM mask usage in lpt_dsp_power_up (dsp.c) - updated dbg message formatting in catpt_restore_fwimage as suggested by Andy - fixed alignment for struct catpt_ssp_device_format - catpt_set_ctlvol now verifies all-equal scenario based on all channels rather than just first two as requested by Amadeo - fixed SPDX for registers.h Changes in v2: - fixed SPDX formatting for all header files as well as pcm.c - fixed size used when memcping fs.c::fw_build_read as reported by Mark - renamed struct catpt_pdata to struct catpt_spec (cosmetic) - fixed erroneous path in catpt_load_block: region is properly released - trace.h events for updating registers have been removed and usages replaced by dev_dbg (SRAMPGE/ LPCS) - as requested by Andy, struct resource has replaced struct catpt_mbank and struct catpt_mregion. This change cascaded into: - catpt_mbank_size and catpt_mregion_size replaced by resource_size - catpt_mregion_intersects replaced by resource_overlaps - all catpt_mbank_ and catpt_mregion_ handlers found in loader.c (_request, _reserve, _release, _extract, _split, _join) have been removed - __request_region and __release_region have been enlisted in their place - catpt_mregion_intersecting renamed to catpt_resource_overlapping - catpt_request_region helper has been provided to deal with -size based requests o haven't found direct replacements in resource.c/ ioport.h for both functions - catpt_mbank_create and catpt_mbank_remove renamed to catpt_sram_init and catpt_sram_free respectively - catpt_sram_init now returns void instead of int and has been converted to simple initialized. This change ultimately cascaded into: o both SRAM banks initialization being moved to catpt_dev_init from catpt_acpi_probe (device.c) o catpt_dev::spec is now initialized first, with catpt_dev_init following it soon after o catpt_acpi_probe erroneous path has been simplified as SRAM banks no longer need to be freed - catpt_sram_free now frees all resources via child -> sibling enumeration rather than region_list iteration - catpt_dsp_update_srampge and catpt_dsp_set_srampge now accept new argument: unsigned long mask. Caused by removal of catpt_mbank - mask is taken directly from catpt_dev::spec::d/iram_mask - trace.h events for catpt_mbank and catpt_mregion have been removed Diff against last drop on internal list: - replaced spinlock with mutex for mregion allocation and release to address sleeping in atomic context warnings - fixed coredump fw_hash dumping - kcontrol values are now always stored regardless of stream of interest is running or not - kcontrol values are now applied after stream is prepared instead of ignoring what has been set by user initially - catpt_pdata instances have been renamed from hsw_ and bdw_ to lpt_ and wpt_ respectively - reordered Makefile .o(s) (cosmetic) Cezary Rojewski (13): ASoC: Intel: Add catpt device ASoC: Intel: catpt: Define DSP operations ASoC: Intel: catpt: Firmware loading and context restore ASoC: Intel: catpt: Implement IPC protocol ASoC: Intel: catpt: Add IPC messages ASoC: Intel: catpt: PCM operations ASoC: Intel: catpt: Event tracing ASoC: Intel: catpt: Simple sysfs attributes ASoC: Intel: Select catpt and deprecate haswell ASoC: Intel: haswell: Remove haswell-solution specific code ASoC: Intel: broadwell: Remove haswell-solution specific code ASoC: Intel: bdw-5650: Remove haswell-solution specific code ASoC: Intel: bdw-5677: Remove haswell-solution specific code sound/soc/intel/Kconfig | 22 +- sound/soc/intel/Makefile | 2 +- sound/soc/intel/boards/Kconfig | 8 +- sound/soc/intel/boards/bdw-rt5650.c | 36 - sound/soc/intel/boards/bdw-rt5677.c | 33 - sound/soc/intel/boards/broadwell.c | 33 - sound/soc/intel/boards/haswell.c | 28 +- sound/soc/intel/catpt/Makefile | 6 + sound/soc/intel/catpt/core.h | 187 +++++ sound/soc/intel/catpt/device.c | 376 +++++++++ sound/soc/intel/catpt/dsp.c | 596 +++++++++++++ sound/soc/intel/catpt/fs.c | 79 ++ sound/soc/intel/catpt/ipc.c | 298 +++++++ sound/soc/intel/catpt/loader.c | 673 +++++++++++++++ sound/soc/intel/catpt/messages.c | 312 +++++++ sound/soc/intel/catpt/messages.h | 401 +++++++++ sound/soc/intel/catpt/pcm.c | 1198 +++++++++++++++++++++++++++ sound/soc/intel/catpt/registers.h | 191 +++++ sound/soc/intel/catpt/trace.h | 83 ++ 19 files changed, 4420 insertions(+), 142 deletions(-) create mode 100644 sound/soc/intel/catpt/Makefile create mode 100644 sound/soc/intel/catpt/core.h create mode 100644 sound/soc/intel/catpt/device.c create mode 100644 sound/soc/intel/catpt/dsp.c create mode 100644 sound/soc/intel/catpt/fs.c create mode 100644 sound/soc/intel/catpt/ipc.c create mode 100644 sound/soc/intel/catpt/loader.c create mode 100644 sound/soc/intel/catpt/messages.c create mode 100644 sound/soc/intel/catpt/messages.h create mode 100644 sound/soc/intel/catpt/pcm.c create mode 100644 sound/soc/intel/catpt/registers.h create mode 100644 sound/soc/intel/catpt/trace.h -- 2.17.1

2 14

[PATCH v2 00/13] ASoC: Intel: Catpt - Lynx and Wildcat point
by Cezary Rojewski 11 Aug '20

11 Aug '20

Implement support for Lynxpoint and Wildcat Point AudioDSP. Catpt solution deprecates existing sound/soc/intel/haswell which is removed in the following series. This cover-letter is followed by 'Developer's deep dive' message schedding light on catpt's key concepts and areas addressed. Due to high range of errors and desynchronization from recommendations set by Windows solution, re-write came as a lower-cost solution compared to refactoring /haswell/ with several series of patches. Special thanks go to Marcin Barlik and Piotr Papierkowski for sharing their LPT/WPT AudioDSP architecture expertise as well as helping backtrack its historical background. My thanks go to Amadeusz Slawinski for reviews and improvements proposed on and off the internal list. Most of internal diff below is his contribution. Krzysztof Hejmowski helped me setup my own Xtensa environment and recompile LPT/WPT FW binary sources what sped up the development greatly. This would not have been possible without help from these champions, especially considering how quickly the catpt was written: 2 weeks features, 3 weeks optimizations. Thank you. Userspace-exposed members are compatible with what is exposed by deprecated solution as well as FW binary being re-used thus no harm is done. The only visible differences are: the newly added 'Loopback Mute' kcontrol and volume support extending to quad from stereo. On top of fixing erros and design flows, catpt also adds module reload, dynamic SRAM memory allocation during PCM runtime and exposes missing userspace API: 'Loopback Mute' kcontrol, quad volume controls and sysfs fw-version entries. Event tracing is provided to easy solution debugging. Following are not included in this update and are scheduled as later addition: - fw logging - module (library) support Note: LPT power up/down sequences might get aligned with WPT once enough testing is done as capabilities are shared for both DSPs. Note #2: Both LPT and WPT power up/down sequences may get optimized in future updates as thanks to help from the Windows team, most of nuances behind why/what/when in regard to hw registers have been backtracked and reviewed again. Link to developer's deep dive message: https://www.spinics.net/lists/alsa-devel/msg113563.html Changes in v2: - fixed SPDX formatting for all header files as well as pcm.c - fixed size used when memcping fs.c::fw_build_read as reported by Mark - renamed struct catpt_pdata to struct catpt_spec (cosmetic) - fixed erroneous patch in catpt_load_block: region if properly released - trace.h events for updating registers have been removed and usages replaced by dev_dbg (SRAMPGE/ LPCS) - as requested by Andy, struct resource has replaced struct catpt_mbank and struct catpt_mregion. This change cascaded into: - catpt_mbank_size and catpt_mregion_size replaced by resource_size - catpt_mregion_intersects replaced by resource_overlaps - all catpt_mbank_ and catpt_mregion_ handlers found in loader.c (_request, _reserve, _release, _extract, _split, _join) have been removed - __request_region and __release_region have been enlisted in their place - catpt_mregion_intersecting renamed to catpt_resource_overlapping - catpt_request_region helper has been provided to deal with -size based requests o haven't found direct replacements in resource.c/ ioport.h for both functions - catpt_mbank_create and catpt_mbank_remove renamed to catpt_sram_init and catpt_sram_free respectively - catpt_sram_init now returns void instead of int and has been converted to simple initialized. This change ultimately cascaded into: o both SRAM banks initialization being moved to catpt_dev_init from catpt_acpi_probe (device.c) o catpt_dev::spec is now initialized first, with catpt_dev_init following it soon after o catpt_acpi_probe erroneous path has been simplified as SRAM banks no longer need to be freed - catpt_sram_free now frees all resources via child -> sibling enumeration rather than region_list iteration - catpt_dsp_update_srampge and catpt_dsp_set_srampge now accept new argument: unsigned long mask. Caused by removal of catpt_mbank - mask is taken directly from catpt_dev::spec::d/iram_mask - trace.h events for catpt_mbank and catpt_mregion have been removed Diff against last drop on internal list: - replaced spinlock with mutex for mregion allocation and release to address sleeping in atomic context warnings - fixed coredump fw_hash dumping - kcontrol values are now always stored regardless of stream of interest is running or not - kcontrol values are now applied after stream is prepared instead of ignoring what has been set by user initially - catpt_pdata instances have been renamed from hsw_ and bdw_ to lpt_ and wpt_ respectively - reordered Makefile .o(s) (cosmetic) Cezary Rojewski (13): ASoC: Intel: Add catpt device ASoC: Intel: catpt: Define DSP operations ASoC: Intel: catpt: Firmware loading and context restore ASoC: Intel: catpt: Implement IPC protocol ASoC: Intel: catpt: Add IPC messages ASoC: Intel: catpt: PCM operations ASoC: Intel: catpt: Event tracing ASoC: Intel: catpt: Simple sysfs attributes ASoC: Intel: Select catpt and deprecate haswell ASoC: Intel: haswell: Remove haswell-solution specific code ASoC: Intel: broadwell: Remove haswell-solution specific code ASoC: Intel: bdw-5650: Remove haswell-solution specific code ASoC: Intel: bdw-5677: Remove haswell-solution specific code sound/soc/intel/Kconfig | 22 +- sound/soc/intel/Makefile | 2 +- sound/soc/intel/boards/Kconfig | 8 +- sound/soc/intel/boards/bdw-rt5650.c | 36 - sound/soc/intel/boards/bdw-rt5677.c | 33 - sound/soc/intel/boards/broadwell.c | 33 - sound/soc/intel/boards/haswell.c | 28 +- sound/soc/intel/catpt/Makefile | 6 + sound/soc/intel/catpt/core.h | 187 +++++ sound/soc/intel/catpt/device.c | 376 +++++++++ sound/soc/intel/catpt/dsp.c | 596 +++++++++++++ sound/soc/intel/catpt/fs.c | 79 ++ sound/soc/intel/catpt/ipc.c | 298 +++++++ sound/soc/intel/catpt/loader.c | 676 +++++++++++++++ sound/soc/intel/catpt/messages.c | 312 +++++++ sound/soc/intel/catpt/messages.h | 401 +++++++++ sound/soc/intel/catpt/pcm.c | 1194 +++++++++++++++++++++++++++ sound/soc/intel/catpt/registers.h | 191 +++++ sound/soc/intel/catpt/trace.h | 83 ++ 19 files changed, 4419 insertions(+), 142 deletions(-) create mode 100644 sound/soc/intel/catpt/Makefile create mode 100644 sound/soc/intel/catpt/core.h create mode 100644 sound/soc/intel/catpt/device.c create mode 100644 sound/soc/intel/catpt/dsp.c create mode 100644 sound/soc/intel/catpt/fs.c create mode 100644 sound/soc/intel/catpt/ipc.c create mode 100644 sound/soc/intel/catpt/loader.c create mode 100644 sound/soc/intel/catpt/messages.c create mode 100644 sound/soc/intel/catpt/messages.h create mode 100644 sound/soc/intel/catpt/pcm.c create mode 100644 sound/soc/intel/catpt/registers.h create mode 100644 sound/soc/intel/catpt/trace.h -- 2.17.1

5 24

[PATCH] ALSA: hda - reverse the setting value in the micmute_led_set
by Hui Wang 11 Aug '20

11 Aug '20

Before the micmute_led_set() is introduced, the function of alc_gpio_micmute_update() will set the gpio value with the !micmute_led.led_value, and the machines have the correct micmute led status. After the micmute_led_set() is introduced, it sets the gpio value with !!micmute_led.led_value, so the led status is not correct anymore, we need to set micmute_led_polarity = 1 to workaround it. Now we fix the micmute_led_set() and remove micmute_led_polarity = 1. Fixes: 87dc36482cab ("ALSA: hda/realtek - Add LED class support for micmute LED") Reported-and-suggested-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> Cc: <stable(a)vger.kernel.org> Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 5 +---- 1 file changed, 1 insertion(+), 4 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 09d93dd88713..073029aeaf3c 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4125,7 +4125,7 @@ static int micmute_led_set(struct led_classdev *led_cdev, struct alc_spec *spec = codec->spec; alc_update_gpio_led(codec, spec->gpio_mic_led_mask, - spec->micmute_led_polarity, !!brightness); + spec->micmute_led_polarity, !brightness); return 0; } @@ -4162,8 +4162,6 @@ static void alc285_fixup_hp_gpio_led(struct hda_codec *codec, { struct alc_spec *spec = codec->spec; - spec->micmute_led_polarity = 1; - alc_fixup_hp_gpio_led(codec, action, 0x04, 0x01); } @@ -4414,7 +4412,6 @@ static void alc233_fixup_lenovo_line2_mic_hotkey(struct hda_codec *codec, { struct alc_spec *spec = codec->spec; - spec->micmute_led_polarity = 1; alc_fixup_hp_gpio_led(codec, action, 0, 0x04); if (action == HDA_FIXUP_ACT_PRE_PROBE) { spec->init_amp = ALC_INIT_DEFAULT; -- 2.17.1

2 1

[PATCH] ALSA: hda - fix the micmute led status for Lenovo ThinkCentre AIO
by Hui Wang 11 Aug '20

11 Aug '20

After installing the Ubuntu Linux, the micmute led status is not correct. Users expect that the led is on if the capture is disabled, but with the current kernel, the led is off with the capture disabled. We tried the old linux kernel like linux-4.15, there is no this issue. It looks like we introduced this issue when switching to the led_cdev. Cc: <stable(a)vger.kernel.org> Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index daedcc0adc21..09d93dd88713 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4414,6 +4414,7 @@ static void alc233_fixup_lenovo_line2_mic_hotkey(struct hda_codec *codec, { struct alc_spec *spec = codec->spec; + spec->micmute_led_polarity = 1; alc_fixup_hp_gpio_led(codec, action, 0, 0x04); if (action == HDA_FIXUP_ACT_PRE_PROBE) { spec->init_amp = ALC_INIT_DEFAULT; -- 2.17.1

3 8

[PATCH 1/2] ASoC: q6afe-dai: add dummy register read function
by Srinivas Kandagatla 11 Aug '20

11 Aug '20

Most of the DAPM widgets for DSP ASoC components reuse reg field of the widgets for its internal calculations, however these are not real registers. So read/writes to these numbers are not really valid. However ASoC core will read these registers to get default state during startup. With recent changes to ASoC core, every register read/write failures are reported very verbosely. Prior to this fails to reads are totally ignored, so we never saw any error messages. To fix this add dummy read function to return default value. Reported-by: John Stultz <john.stultz(a)linaro.org> Fixes: 24c4cbcfac09 ("ASoC: qdsp6: q6afe: Add q6afe dai driver") Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- sound/soc/qcom/qdsp6/q6afe-dai.c | 9 ++++++++- 1 file changed, 8 insertions(+), 1 deletion(-) diff --git a/sound/soc/qcom/qdsp6/q6afe-dai.c b/sound/soc/qcom/qdsp6/q6afe-dai.c index 2a5302f1db98..492bdbaf13c4 100644 --- a/sound/soc/qcom/qdsp6/q6afe-dai.c +++ b/sound/soc/qcom/qdsp6/q6afe-dai.c @@ -1352,6 +1352,13 @@ static const struct snd_soc_dapm_widget q6afe_dai_widgets[] = { SND_SOC_DAPM_AIF_OUT("DISPLAY_PORT_RX", "NULL", 0, 0, 0, 0), }; +static unsigned int q6afe_reg_read(struct snd_soc_component *component, + unsigned int reg) +{ + /* default value */ + return 0; +} + static const struct snd_soc_component_driver q6afe_dai_component = { .name = "q6afe-dai-component", .dapm_widgets = q6afe_dai_widgets, @@ -1359,7 +1366,7 @@ static const struct snd_soc_component_driver q6afe_dai_component = { .dapm_routes = q6afe_dapm_routes, .num_dapm_routes = ARRAY_SIZE(q6afe_dapm_routes), .of_xlate_dai_name = q6afe_of_xlate_dai_name, - + .read = q6afe_reg_read, }; static void of_q6afe_parse_dai_data(struct device *dev, -- 2.21.0

3 4

[PATCH v5 00/12] ASoC: qcom: Add support for SC7180 lpass variant
by Rohit kumar 11 Aug '20

11 Aug '20

This patch chain add audio support for SC7180 soc by doing the required modification in existing common lpass-cpu/lpass-platform driver. This also fixes some concurrency issue. This patch series is already tested by Srinivas on Dragon Board 410c. Changes since v4: - Updated compatible string for sc7180 lpass cpu as suggested by Rob - Addressed comments by Rob in yaml Documentation. Ajit Pandey (4): ASoC: qcom: Add common array to initialize soc based core clocks ASoC: qcom: lpass-platform: Replace card->dev with component->dev include: dt-bindings: sound: Add sc7180-lpass bindings header ASoC: qcom: lpass-sc7180: Add platform driver for lpass audio Rohit kumar (8): ASoC: qcom: lpass-cpu: Move ahbix clk to platform specific function ASoC: qcom: lpass-platform: fix memory leak ASoC: qcom: lpass: Use regmap_field for i2sctl and dmactl registers ASoC: qcom: lpass-cpu: fix concurrency issue dt-bindings: sound: lpass-cpu: Add sc7180 lpass cpu node ASoC: qcom: lpass-cpu: Use platform_get_resource ASoC: qcom: lpass-platform: Use platform_get_irq dt-bindings: sound: lpass-cpu: Move to yaml format .../devicetree/bindings/sound/qcom,lpass-cpu.txt | 79 -------- .../devicetree/bindings/sound/qcom,lpass-cpu.yaml | 179 +++++++++++++++++ include/dt-bindings/sound/sc7180-lpass.h | 10 + sound/soc/qcom/Kconfig | 5 + sound/soc/qcom/Makefile | 2 + sound/soc/qcom/lpass-apq8016.c | 86 ++++++-- sound/soc/qcom/lpass-cpu.c | 204 ++++++++++--------- sound/soc/qcom/lpass-ipq806x.c | 67 +++++++ sound/soc/qcom/lpass-lpaif-reg.h | 157 ++++++++------- sound/soc/qcom/lpass-platform.c | 155 +++++++++++---- sound/soc/qcom/lpass-sc7180.c | 216 +++++++++++++++++++++ sound/soc/qcom/lpass.h | 63 +++++- 12 files changed, 924 insertions(+), 299 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/qcom,lpass-cpu.txt create mode 100644 Documentation/devicetree/bindings/sound/qcom,lpass-cpu.yaml create mode 100644 include/dt-bindings/sound/sc7180-lpass.h create mode 100644 sound/soc/qcom/lpass-sc7180.c -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

3 17

[PATCH 00/13] ASoC: Intel: Catpt - Lynx and Wildcat point
by Cezary Rojewski 11 Aug '20

11 Aug '20

Implement support for Lynxpoint and Wildcat Point AudioDSP. Catpt solution deprecates existing sound/soc/intel/haswell which is removed in the following series. This cover-letter is followed by 'Developer's deep dive' message schedding light on catpt's key concepts and areas addressed. Due to high range of errors and desynchronization from recommendations set by Windows solution, re-write came as a lower-cost solution compared to refactoring /haswell/ with several series of patches. Special thanks go to Marcin Barlik and Piotr Papierkowski for sharing their LPT/WPT AudioDSP architecture expertise as well as helping backtrack its historical background. My thanks go to Amadeusz Slawinski for reviews and improvements proposed on and off the internal list. Most of diff below is his contribution. Krzysztof Hejmowski helped me setup my own Xtensa environment and recompile LPT/WPT FW binary sources what sped up the development greatly. This would not have been possible without help from these champions, especially considering how quickly the catpt was written: 2 weeks features, 3 weeks optimizations. Thank you. Diff against last drop on internal list: - replaced spinlock with mutex for mregion allocation and release to address sleeping in atomic context warnings - fixed coredump fw_hash dumping - kcontrol values are now always stored regardless of stream of interest is running or not - kcontrol values are now applied after stream is prepared instead of ignoring what has been set by user initially - catpt_pdata instances have been renamed from hsw_ and bdw_ to lpt_ and wpt_ respectively - reordered Makefile .o(s) (cosmetic) Userspace-exposed members are compatible with what is exposed by deprecated solution as well as FW binary being re-used thus no harm is done. The only visible differences are: the newly added 'Loopback Mute' kcontrol and volume support extending to quad from stereo. On top of fixing erros and design flows, catpt also adds module reload, dynamic SRAM memory allocation during PCM runtime and exposes missing userspace API: 'Loopback Mute' kcontrol, quad volume controls and sysfs fw-version entries. Event tracing is provided to easy solution debugging. Following are not included in this update and are scheduled as later addition: - fw logging - module (library) support Note: LPT power up/down sequences might get aligned with WPT once enough testing is done as capabilities are shared for both DSPs. Note #2: Both LPT and WPT power up/down sequences may get optimized in future updates as thanks to help from the Windows team, most of nuances behind why/what/when in regard to hw registers have been backtracked and reviewed again. Cezary Rojewski (13): ASoC: Intel: Add catpt device ASoC: Intel: catpt: Define DSP operations ASoC: Intel: catpt: Firmware loading and context restore ASoC: Intel: catpt: Implement IPC protocol ASoC: Intel: catpt: Add IPC messages ASoC: Intel: catpt: PCM operations ASoC: Intel: catpt: Event tracing ASoC: Intel: catpt: Simple sysfs attributes ASoC: Intel: Select catpt and deprecate haswell ASoC: Intel: haswell: Remove haswell-solution specific code ASoC: Intel: broadwell: Remove haswell-solution specific code ASoC: Intel: bdw-5650: Remove haswell-solution specific code ASoC: Intel: bdw-5677: Remove haswell-solution specific code sound/soc/intel/Kconfig | 22 +- sound/soc/intel/Makefile | 2 +- sound/soc/intel/boards/Kconfig | 8 +- sound/soc/intel/boards/bdw-rt5650.c | 36 - sound/soc/intel/boards/bdw-rt5677.c | 33 - sound/soc/intel/boards/broadwell.c | 33 - sound/soc/intel/boards/haswell.c | 28 +- sound/soc/intel/catpt/Makefile | 6 + sound/soc/intel/catpt/core.h | 224 +++++ sound/soc/intel/catpt/device.c | 389 +++++++++ sound/soc/intel/catpt/dsp.c | 590 +++++++++++++ sound/soc/intel/catpt/fs.c | 79 ++ sound/soc/intel/catpt/ipc.c | 298 +++++++ sound/soc/intel/catpt/loader.c | 837 +++++++++++++++++++ sound/soc/intel/catpt/messages.c | 312 +++++++ sound/soc/intel/catpt/messages.h | 402 +++++++++ sound/soc/intel/catpt/pcm.c | 1194 +++++++++++++++++++++++++++ sound/soc/intel/catpt/registers.h | 191 +++++ sound/soc/intel/catpt/trace.h | 169 ++++ 19 files changed, 4711 insertions(+), 142 deletions(-) create mode 100644 sound/soc/intel/catpt/Makefile create mode 100644 sound/soc/intel/catpt/core.h create mode 100644 sound/soc/intel/catpt/device.c create mode 100644 sound/soc/intel/catpt/dsp.c create mode 100644 sound/soc/intel/catpt/fs.c create mode 100644 sound/soc/intel/catpt/ipc.c create mode 100644 sound/soc/intel/catpt/loader.c create mode 100644 sound/soc/intel/catpt/messages.c create mode 100644 sound/soc/intel/catpt/messages.h create mode 100644 sound/soc/intel/catpt/pcm.c create mode 100644 sound/soc/intel/catpt/registers.h create mode 100644 sound/soc/intel/catpt/trace.h -- 2.17.1

4 23

USB audio cuts 2-3 secs at start
by Stuart Naylor 11 Aug '20

11 Aug '20

http://alsa-project.org/db/?f=6b107178c9939f235a390c07cb5c1f71d6a6e0c0 Its an Anker power conf and for some reason the first couple of seconds of aplay do not play via USB. Alsa-info above anyone and tips or ideas? Stuart

2 2

[PATCH] ASoC: fsl_sai: Add -EPROBE_DEFER check for regmap init
by Shengjiu Wang 11 Aug '20

11 Aug '20

Regmap initialization may return -EPROBE_DEFER for clock may not be ready, so check -EPROBE_DEFER error type before start another Regmap initialization. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/fsl_sai.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index a22562f2df47..eb933fe9b6d1 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -927,7 +927,7 @@ static int fsl_sai_probe(struct platform_device *pdev) "bus", base, &fsl_sai_regmap_config); /* Compatible with old DTB cases */ - if (IS_ERR(sai->regmap)) + if (IS_ERR(sai->regmap) && PTR_ERR(sai->regmap) != -EPROBE_DEFER) sai->regmap = devm_regmap_init_mmio_clk(&pdev->dev, "sai", base, &fsl_sai_regmap_config); if (IS_ERR(sai->regmap)) { -- 2.27.0

3 3

[PATCH] ASoC: amd: Replacing component->name with codec_dai->name
by Ravulapati Vishnu vardhan rao 10 Aug '20

10 Aug '20

Replacing string compare with codec_dai->name instead of comparing with codec_dai->component->name in hw_params. Signed-off-by: Ravulapati Vishnu vardhan rao <Vishnuvardhanrao.Ravulapati(a)amd.com> --- sound/soc/amd/acp3x-rt5682-max9836.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/amd/acp3x-rt5682-max9836.c b/sound/soc/amd/acp3x-rt5682-max9836.c index 55815fdaa1aa..406526e79af3 100644 --- a/sound/soc/amd/acp3x-rt5682-max9836.c +++ b/sound/soc/amd/acp3x-rt5682-max9836.c @@ -138,7 +138,7 @@ static int acp3x_1015_hw_params(struct snd_pcm_substream *substream, srate = params_rate(params); for_each_rtd_codec_dais(rtd, i, codec_dai) { - if (strcmp(codec_dai->component->name, "rt1015-aif")) + if (strcmp(codec_dai->name, "rt1015-aif")) continue; ret = snd_soc_dai_set_bclk_ratio(codec_dai, 64); if (ret < 0) -- 2.17.1

3 5

[PATCH] ASoC: amd: Replacing component->name with codec_dai->name.
by Ravulapati Vishnu vardhan rao 10 Aug '20

10 Aug '20

Replacing string compare with "codec_dai->name" instead of comparing with "codec_dai->component->name" in hw_params because, Here the component name for codec RT1015 is "i2c-10EC5682:00" and will never be "rt1015-aif1" as it is codec-dai->name. So, strcmp() always compares and fails to set the sysclk,pll,bratio for expected codec-dai="rt1015-aif1". Signed-off-by: Ravulapati Vishnu vardhan rao <Vishnuvardhanrao.Ravulapati(a)amd.com> --- sound/soc/amd/acp3x-rt5682-max9836.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/amd/acp3x-rt5682-max9836.c b/sound/soc/amd/acp3x-rt5682-max9836.c index 55815fdaa1aa..406526e79af3 100644 --- a/sound/soc/amd/acp3x-rt5682-max9836.c +++ b/sound/soc/amd/acp3x-rt5682-max9836.c @@ -138,7 +138,7 @@ static int acp3x_1015_hw_params(struct snd_pcm_substream *substream, srate = params_rate(params); for_each_rtd_codec_dais(rtd, i, codec_dai) { - if (strcmp(codec_dai->component->name, "rt1015-aif")) + if (strcmp(codec_dai->name, "rt1015-aif")) continue; ret = snd_soc_dai_set_bclk_ratio(codec_dai, 64); if (ret < 0) -- 2.17.1

2 1

[PATCH] alsaseq: fix constant objects in Python 3
by Hector Martin 10 Aug '20

10 Aug '20

In Py3 everything is a long, so we can't get away with duplicating the integer object layout any more. Properly subclass PyLong_Type instead, with a bit of a hack to add an extra name field after the variable-sized data section. Also get rid of the enum type stuff, which seems to be unused cruft, and the numprotocol stuff, because the base type can take care of that (and we can't map arbitrary arithmetic results back to constants anyway). Signed-off-by: Hector Martin <marcan(a)marcan.st> --- pyalsa/alsaseq.c | 149 +++++++++++++++++------------------------------ 1 file changed, 52 insertions(+), 97 deletions(-) diff --git a/pyalsa/alsaseq.c b/pyalsa/alsaseq.c index 0e7cc35..9d9b1a9 100644 --- a/pyalsa/alsaseq.c +++ b/pyalsa/alsaseq.c @@ -51,9 +51,6 @@ /* the C variable of a constant dict */ #define TDICT(subtype) _dictPYALSASEQ_CONST_##subtype -/* the C enumeration of a constant dict */ -#define TTYPE(subtype) PYALSASEQ_CONST_##subtype - /* defines constant dict by type */ #define TCONSTDICT(subtype) \ static PyObject * TDICT(subtype); @@ -71,7 +68,7 @@ /* creates a typed constant and add it to the module and dictionary */ #define TCONSTADD(module, subtype, name) { \ PyObject *tmp = \ - Constant_create(#name, SND_##name, TTYPE(subtype)); \ + Constant_create(#name, SND_##name); \ if (tmp == NULL) { \ return MOD_ERROR_VAL; \ } \ @@ -109,46 +106,6 @@ } \ } - -/* num protocol support for binary Constant operations */ -#define NUMPROTOCOL2(name, oper) \ - static PyObject * \ - Constant_##name (PyObject *v, PyObject *w) { \ - int type = 0; \ - long val1, val2, val; \ - if (get_long1(v, &val1) || get_long1(w, &val2)) { \ - Py_INCREF(Py_NotImplemented); \ - return Py_NotImplemented; \ - } \ - val = val1 oper val2; \ - /* always asume left will be the type */ \ - if (PyObject_TypeCheck(v, &ConstantType)) { \ - type = ((ConstantObject *) v)->type; \ - } else if (PyObject_TypeCheck(w, &ConstantType)) { \ - type = ((ConstantObject *) w)->type; \ - } \ - PyObject *self = Constant_create(#oper, val, type); \ - return self; \ - } - -/* num protocol support for unary Constant operations */ -#define NUMPROTOCOL1(name, oper) \ - static PyObject * \ - Constant_##name (PyObject *v) { \ - int type = 0; \ - long val1, val; \ - if (get_long1(v, &val1)) { \ - Py_INCREF(Py_NotImplemented); \ - return Py_NotImplemented; \ - } \ - val = oper val1; \ - if (PyObject_TypeCheck(v, &ConstantType)) { \ - type = ((ConstantObject *) v)->type; \ - } \ - PyObject *self = Constant_create(#oper, val, type); \ - return self; \ - } - /* sets the object into the dict */ #define SETDICTOBJ(name, object) \ PyDict_SetItemString(dict, name, object) @@ -328,21 +285,6 @@ // alsaseq.Constant implementation ////////////////////////////////////////////////////////////////////////////// -/* alsaseq.Constant->type enumeration */ -enum { - PYALSASEQ_CONST_STREAMS, - PYALSASEQ_CONST_MODE, - PYALSASEQ_CONST_QUEUE, - PYALSASEQ_CONST_CLIENT_TYPE, - PYALSASEQ_CONST_PORT_CAP, - PYALSASEQ_CONST_PORT_TYPE, - PYALSASEQ_CONST_EVENT_TYPE, - PYALSASEQ_CONST_EVENT_TIMESTAMP, - PYALSASEQ_CONST_EVENT_TIMEMODE, - PYALSASEQ_CONST_ADDR_CLIENT, - PYALSASEQ_CONST_ADDR_PORT, -}; - // constants dictionaries TCONSTDICT(STREAMS); @@ -390,34 +332,66 @@ PyDoc_STRVAR(Constant__doc__, "Python number protocol." ); -/** alsaseq.Constant object structure type */ +/** alsaseq.Constant additional fields */ +/* This follows the variable length portion of the Long type */ typedef struct { - PyObject_HEAD - ; - - /* value of constant */ - unsigned long int value; /* name of constant */ const char *name; - /* type of constant */ - int type; +} ConstantExtraFields; + +/** alsaseq.Constant object structure type */ +typedef struct { +#if PY_MAJOR_VERSION < 3 + PyIntObject base; +#else + /* NOTE: this only works if the value fits in one digit */ + PyLongObject base; +#endif + /* This field is actually offset by the base type's variable size portion */ + ConstantExtraFields extra; } ConstantObject; +#if PY_MAJOR_VERSION < 3 +/* PyInt is fixed size in Python 2 */ +# define CONST_VALUE(x) PyInt_AsLong((PyObject *)x) +# define CONST_EXTRA(x) (&(x->extra)) +#else +/* PyLong is variable size in Python 3 */ +# define CONST_VALUE(x) PyLong_AsLong((PyObject *)x) +# define CONST_EXTRA(x) \ + ((ConstantExtraFields *)( \ + ((intptr_t)(&x->extra)) \ + + abs(Py_SIZE(&x->base)) * Py_TYPE(x)->tp_itemsize \ + )) +#endif + /** alsaseq.Constant type (initialized later...) */ static PyTypeObject ConstantType; /** alsaseq.Constant internal create */ static PyObject * -Constant_create(const char *name, long value, int type) { - ConstantObject *self = PyObject_New(ConstantObject, &ConstantType); +Constant_create(const char *name, long value) { +#if PY_MAJOR_VERSION < 3 + PyObject *val = PyInt_FromLong(value); +#else + PyObject *val = PyLong_FromLong(value); +#endif + + PyObject *args = PyTuple_Pack(1, val); + Py_DECREF(val); + +#if PY_MAJOR_VERSION < 3 + ConstantObject *self = (ConstantObject *)PyInt_Type.tp_new(&ConstantType, args, NULL); +#else + ConstantObject *self = (ConstantObject *)PyLong_Type.tp_new(&ConstantType, args, NULL); +#endif + Py_DECREF(args); if (self == NULL) { return NULL; } - self->value = value; - self->name = name; - self->type = type; + CONST_EXTRA(self)->name = name; return (PyObject *)self; } @@ -426,34 +400,16 @@ Constant_create(const char *name, long value, int type) { static PyObject * Constant_repr(ConstantObject *self) { return PyUnicode_FromFormat("%s(0x%x)", - self->name, - (unsigned int)self->value); + CONST_EXTRA(self)->name, + (unsigned int)CONST_VALUE(self)); } /** alsaseq.Constant tp_str */ static PyObject * Constant_str(ConstantObject *self) { return PyUnicode_FromFormat("%s", - self->name); -} - -/** alsaseq.Constant Number protocol support (note: not all ops supported) */ -NUMPROTOCOL2(Add, +) -NUMPROTOCOL2(Subtract, -) -NUMPROTOCOL2(Xor, ^) -NUMPROTOCOL2(Or, |) -NUMPROTOCOL2(And, &) -NUMPROTOCOL1(Invert, ~) - -/** alsaseq.Constant number protocol methods */ -static PyNumberMethods Constant_as_number = { - nb_add: (binaryfunc)Constant_Add, - nb_subtract: (binaryfunc)Constant_Subtract, - nb_xor: (binaryfunc)Constant_Xor, - nb_or: (binaryfunc)Constant_Or, - nb_and: (binaryfunc)Constant_And, - nb_invert: (unaryfunc)Constant_Invert -}; + CONST_EXTRA(self)->name); +} /** alsaseq.Constant type */ static PyTypeObject ConstantType = { @@ -464,7 +420,7 @@ static PyTypeObject ConstantType = { #else tp_base: &PyLong_Type, #endif - tp_basicsize: sizeof(ConstantObject), + tp_basicsize: sizeof(ConstantObject) + sizeof(ConstantExtraFields), tp_flags: #if PY_MAJOR_VERSION < 3 Py_TPFLAGS_HAVE_GETCHARBUFFER @@ -474,7 +430,6 @@ static PyTypeObject ConstantType = { 0, #endif tp_doc: Constant__doc__, - tp_as_number: &Constant_as_number, tp_free: PyObject_Del, tp_str: (reprfunc)Constant_str, tp_repr: (reprfunc)Constant_repr @@ -1730,7 +1685,7 @@ SeqEvent_repr(SeqEventObject *self) { unsigned int ntime = 0; Py_DECREF(key); if (constObject != NULL) { - typestr = constObject->name; + typestr = constObject->extra.name; } if (snd_seq_ev_is_real(self->event)) { -- 2.27.0

2 1

[PATCH] ALSA: usb-audio: Disable Lenovo P620 Rear line-in volume control
by Kai-Heng Feng 10 Aug '20

10 Aug '20

The USB device (0x17aa:0x1046) that support Lenovo P620 rear panel line-in claim to support volume control, but it doens't seem to have an AMP, so when line-in volume lowers below 80, nothing gets recorded anymore. Disable the volume control to workaround the issue. Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> --- sound/usb/mixer_maps.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/mixer_maps.c b/sound/usb/mixer_maps.c index c369c81e74c4..5b43e9e40e49 100644 --- a/sound/usb/mixer_maps.c +++ b/sound/usb/mixer_maps.c @@ -371,6 +371,7 @@ static const struct usbmix_name_map asus_rog_map[] = { }; static const struct usbmix_name_map lenovo_p620_rear_map[] = { + { 19, NULL, 2 }, /* FU, Volume */ { 19, NULL, 12 }, /* FU, Input Gain Pad */ {} }; -- 2.17.1

2 1

[PATCH 0/4] ASoC: Intel: KMB: TDM Enablement patches
by Michael Sit Wei Hong 10 Aug '20

10 Aug '20

This patch series is to enable multiple features on the Keembay Platform Michael Sit Wei Hong (4): ASoC: Intel: KMB: Add 8kHz audio support ASoC: Intel: KMB: Rework disable channel function ASoC: Intel: KMB: Enable TDM audio capture dt-bindings: sound: intel,keembay-i2s: Add channel-max property .../bindings/sound/intel,keembay-i2s.yaml | 8 + sound/soc/intel/keembay/kmb_platform.c | 137 +++++++++++++----- sound/soc/intel/keembay/kmb_platform.h | 1 + 3 files changed, 112 insertions(+), 34 deletions(-) -- 2.17.1

3 14

[PATCH] ALSA: usb-audio: work around streaming quirk for MacroSilicon MS2109
by Hector Martin 10 Aug '20

10 Aug '20

Further investigation of the L-R swap problem on the MS2109 reveals that the problem isn't that the channels are swapped, but rather that they are swapped and also out of phase by one sample. In other words, the issue is actually that the very first frame that comes from the hardware is a half-frame containing only the right channel, and after that everything becomes offset. So introduce a new quirk field to drop the very first 2 bytes that come in after the format is configured and a capture stream starts. This puts the channels in phase and in the correct order. Cc: stable(a)vger.kernel.org Signed-off-by: Hector Martin <marcan(a)marcan.st> --- sound/usb/card.h | 1 + sound/usb/pcm.c | 6 ++++++ sound/usb/quirks.c | 3 +++ sound/usb/stream.c | 1 + 4 files changed, 11 insertions(+) diff --git a/sound/usb/card.h b/sound/usb/card.h index de43267b9c8a..5351d7183b1b 100644 --- a/sound/usb/card.h +++ b/sound/usb/card.h @@ -137,6 +137,7 @@ struct snd_usb_substream { unsigned int tx_length_quirk:1; /* add length specifier to transfers */ unsigned int fmt_type; /* USB audio format type (1-3) */ unsigned int pkt_offset_adj; /* Bytes to drop from beginning of packets (for non-compliant devices) */ + unsigned int stream_offset_adj; /* Bytes to drop from beginning of stream (for non-compliant devices) */ unsigned int running: 1; /* running status */ diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 415bfec49a01..5600751803cf 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -1420,6 +1420,12 @@ static void retire_capture_urb(struct snd_usb_substream *subs, // continue; } bytes = urb->iso_frame_desc[i].actual_length; + if (subs->stream_offset_adj > 0) { + unsigned int adj = min(subs->stream_offset_adj, bytes); + cp += adj; + bytes -= adj; + subs->stream_offset_adj -= adj; + } frames = bytes / stride; if (!subs->txfr_quirk) bytes = frames * stride; diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index c551141f337e..abf99b814a0f 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1495,6 +1495,9 @@ void snd_usb_set_format_quirk(struct snd_usb_substream *subs, case USB_ID(0x2b73, 0x000a): /* Pioneer DJ DJM-900NXS2 */ pioneer_djm_set_format_quirk(subs); break; + case USB_ID(0x534d, 0x2109): /* MacroSilicon MS2109 */ + subs->stream_offset_adj = 2; + break; } } diff --git a/sound/usb/stream.c b/sound/usb/stream.c index 4d1e6579e54d..ca76ba5b5c0b 100644 --- a/sound/usb/stream.c +++ b/sound/usb/stream.c @@ -94,6 +94,7 @@ static void snd_usb_init_substream(struct snd_usb_stream *as, subs->tx_length_quirk = as->chip->tx_length_quirk; subs->speed = snd_usb_get_speed(subs->dev); subs->pkt_offset_adj = 0; + subs->stream_offset_adj = 0; snd_usb_set_pcm_ops(as->pcm, stream); -- 2.27.0

3 2

[PATCH] ALSA: usb-audio: add quirk for Pioneer DDJ-RB
by Hector Martin 10 Aug '20

10 Aug '20

This is just another Pioneer device with fixed endpoints. Input is dummy but used as feedback (it always returns silence). Cc: stable(a)vger.kernel.org Signed-off-by: Hector Martin <marcan(a)marcan.st> --- sound/usb/quirks-table.h | 56 ++++++++++++++++++++++++++++++++++++++++ 1 file changed, 56 insertions(+) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 9c3c03dc96d3..d79e3ddc5690 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -3558,6 +3558,62 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"), } } }, +{ + /* + * PIONEER DJ DDJ-RB + * PCM is 4 channels out, 2 dummy channels in @ 44.1 fixed + * The feedback for the output is the dummy input. + */ + USB_DEVICE_VENDOR_SPEC(0x2b73, 0x000e), + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 0, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 4, + .iface = 0, + .altsetting = 1, + .altset_idx = 1, + .endpoint = 0x01, + .ep_attr = USB_ENDPOINT_XFER_ISOC| + USB_ENDPOINT_SYNC_ASYNC, + .rates = SNDRV_PCM_RATE_44100, + .rate_min = 44100, + .rate_max = 44100, + .nr_rates = 1, + .rate_table = (unsigned int[]) { 44100 } + } + }, + { + .ifnum = 0, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 2, + .iface = 0, + .altsetting = 1, + .altset_idx = 1, + .endpoint = 0x82, + .ep_attr = USB_ENDPOINT_XFER_ISOC| + USB_ENDPOINT_SYNC_ASYNC| + USB_ENDPOINT_USAGE_IMPLICIT_FB, + .rates = SNDRV_PCM_RATE_44100, + .rate_min = 44100, + .rate_max = 44100, + .nr_rates = 1, + .rate_table = (unsigned int[]) { 44100 } + } + }, + { + .ifnum = -1 + } + } + } +}, #define ALC1220_VB_DESKTOP(vend, prod) { \ USB_DEVICE(vend, prod), \ -- 2.27.0

2 1

Accidently deleted alsa folder
by GitHub issues - opened 10 Aug '20

10 Aug '20

alsa-project/alsa-lib issue #77 was opened from themagicalmammal: I deleted alsa folder accidently and not mic doesnt work. Even after using alsa command. I searched and tried to apply many stack-overflow solutions but nothing helped when I try to use mic on pycharm it doesnt work it yields this error ``` ALSA lib conf.c:3916:(snd_config_update_r) Cannot access file /usr/share/alsa/alsa.conf ALSA lib control.c:1373:(snd_ctl_open_noupdate) Invalid CTL hw:0 python3.6: src/hostapi/alsa/pa_linux_alsa.c:1454: BuildDeviceList: Assertion `devIdx < numDeviceNames' failed. ``` While External recorder works Issue URL : https://github.com/alsa-project/alsa-lib/issues/77 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH v2] ALSA: usb-audio: fix overeager device match for MacroSilicon MS2109
by Hector Martin 10 Aug '20

10 Aug '20

Matching by device matches all interfaces, which breaks the video/HID portions of the device depending on module load order. Cc: stable(a)vger.kernel.org Signed-off-by: Hector Martin <marcan(a)marcan.st> --- sound/usb/quirks-table.h | 8 +++++++- 1 file changed, 7 insertions(+), 1 deletion(-) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 9092cc0aa807..7cbb3d591a7f 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -3645,7 +3645,13 @@ ALC1220_VB_DESKTOP(0x26ce, 0x0a01), /* Asrock TRX40 Creator */ * with. */ { - USB_DEVICE(0x534d, 0x2109), + .match_flags = USB_DEVICE_ID_MATCH_DEVICE | + USB_DEVICE_ID_MATCH_INT_CLASS | + USB_DEVICE_ID_MATCH_INT_SUBCLASS, + .idVendor = 0x534d, + .idProduct = 0x2109, + .bInterfaceClass = USB_CLASS_AUDIO, + .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL, .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { .vendor_name = "MacroSilicon", .product_name = "MS2109", -- 2.27.0

2 1

[PATCH] ALSA: usb-audio: fix overeager device match for MacroSilicon MS2109
by Hector Martin 10 Aug '20

10 Aug '20

Matching by device matches all interfaces, which breaks the video/HID portions of the device depending on module load order. Signed-off-by: Hector Martin <marcan(a)marcan.st> --- sound/usb/quirks-table.h | 8 +++++++- 1 file changed, 7 insertions(+), 1 deletion(-) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 9092cc0aa807..7cbb3d591a7f 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -3645,7 +3645,13 @@ ALC1220_VB_DESKTOP(0x26ce, 0x0a01), /* Asrock TRX40 Creator */ * with. */ { - USB_DEVICE(0x534d, 0x2109), + .match_flags = USB_DEVICE_ID_MATCH_DEVICE | + USB_DEVICE_ID_MATCH_INT_CLASS | + USB_DEVICE_ID_MATCH_INT_SUBCLASS, + .idVendor = 0x534d, + .idProduct = 0x2109, + .bInterfaceClass = USB_CLASS_AUDIO, + .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL, .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { .vendor_name = "MacroSilicon", .product_name = "MS2109", -- 2.27.0

1 0

Low volume with chtmax98090 (Toshiba Chromebook 2/Swanky)
by GitHub issues - edited 10 Aug '20

10 Aug '20

alsa-project/alsa-ucm-conf issue #45 was edited from reanimus: It looks like the UCM configuration for this board is the Chromebook Asus C300 one, but the volume levels set by default are far too low. It looks like when Pulseaudio loads on my system, it follows the initialization steps and results in a really low volume, as the Speaker is set to 27/100 and one must manually set the alsamixer levels to fix it. I could have a periodic task reload the volume state, but I find it better to override the UCM configuration manually and tell my package manager to ignore changes to that file during upgrade. FWIW, my fix is to modify ucm2/codecs/max98090/EnableSeq.conf and change ``` cset "name='Headphone Volume' 10" cset "name='Speaker Volume' 10" ``` to ``` cset "name='Headphone Volume' 80%" cset "name='Speaker Volume' 80%" ``` Naturally, this is a bit of a nasty solution. It'd be nice to have an upstream fix in case other stuff is changed for this. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/45 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Low volume with chtmax98090 (Toshiba Chromebook 2/Swanky)
by GitHub issues - opened 10 Aug '20

10 Aug '20

alsa-project/alsa-ucm-conf issue #45 was opened from reanimus: It looks like the UCM configuration for this board is the Chromebook Asus C300 one, but the volume levels set by default are far too low. It looks like when Pulseaudio loads on my system, it follows the initialization steps and results in a really low volume, as the Speaker is set to 27/100 and one must manually set the alsamixer levels to fix it. I could have a periodic task reload the volume state, but I find it better to override the UCM configuration manually and tell my package manager to ignore changes to that file during upgrade. Naturally, this is a bit of a nasty solution. It'd be nice to have an upstream fix in case other stuff is changed for this. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/45 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

pcm_meter.c issue at s16_update
by Pavel Hofman 09 Aug '20

09 Aug '20

Hi, I am debugging the following problem with the PCM METER API: An application with meter/scope configured in .asoundrc hits a high CPU load sometimes when a new playback stream is opened. E.g. in MPD when opening a new radio stream. The 100% CPU core load takes tens of seconds. Debugging the issue revealed this: TL;DR: at decrease of [status.appl_ptr - status.delay] between consequent runs the size of buffer to convert by the built-in s16 scope is set from usual 100s to huge value of pcm->boundary (1.5G), causing a very long processing at 100% core load. Details: Every scope setup uses a default s16 scope defined in alsa-lib, which converts samples to s16 so that the user-provided scopes can access data via snd_pcm_scope_s16_get_channel_buffer buffers with converted samples. The s16 scope converts samples in a loop, in frames chunks, decrementing size variable by frames until zero. The value of size is hundreds samples max at each s16_update call, calculated in https://github.com/alsa-project/alsa-lib/blob/master/src/pcm/pcm_meter.c#L1… . A debug added after that line shows this: @@ -1098,14 +1103,18 @@ static void s16_update(snd_pcm_scope_t *scope) snd_pcm_sframes_t size; snd_pcm_uframes_t offset; size = meter->now - s16->old; + fprintf(stderr, "s16_update 1: meter->now %ld, s16->old %ld, size %ld\n", meter->now, s16->old, size); if (size < 0) size += spcm->boundary; offset = s16->old % meter->buf_size; s16_update 1: meter->now 2154673, s16->old 2153771, size 902 s16_update 1: meter->now 2155579, s16->old 2154673, size 906 s16_update 1: meter->now 2156487, s16->old 2155579, size 908 s16_update 1: meter->now 2157391, s16->old 2156487, size 904 s16_update 1: meter->now 2158298, s16->old 2157391, size 907 s16_update 1: meter->now 2159203, s16->old 2158298, size 905 s16_update 1: meter->now 2160091, s16->old 2159203, size 888 s16_update 1: meter->now 2160976, s16->old 2160091, size 885 s16_update 1: meter->now 2161924, s16->old 2160976, size 948 s16_update 1: meter->now 2162829, s16->old 2161924, size 905 s16_update 1: meter->now 2163733, s16->old 2162829, size 904 s16_update 1: meter->now 2164594, s16->old 2163733, size 861 s16_update 1: meter->now 2165542, s16->old 2164594, size 948 s16_update 1: meter->now 2166448, s16->old 2165542, size 906 s16_update 1: meter->now 2167352, s16->old 2166448, size 904 s16_update 1: meter->now 2168259, s16->old 2167352, size 907 s16_update 1: meter->now 2169165, s16->old 2168259, size 906 s16_update 1: meter->now 2170050, s16->old 2169165, size 885 s16_update 1: meter->now 2170936, s16->old 2170050, size 886 s16_update 1: meter->now 2171877, s16->old 2170936, size 941 s16_update 1: meter->now 2172783, s16->old 2171877, size 906 s16_update 1: meter->now 2173690, s16->old 2172783, size 907 s16_update 1: meter->now 2174596, s16->old 2173690, size 906 s16_update 1: meter->now 2175502, s16->old 2174596, size 906 s16_update 1: meter->now 2176407, s16->old 2175502, size 905 s16_update 1: meter->now 2177311, s16->old 2176407, size 904 s16_update 1: meter->now 2178215, s16->old 2177311, size 904 However, sometimes meter->now < s16->old, causing size < 0, for which this line applies https://github.com/alsa-project/alsa-lib/blob/master/src/pcm/pcm_meter.c#L1… size += spcm->boundary and size becomes a huge number, because hw:0 pcm -> boundary is 1445068800. The loop at https://github.com/alsa-project/alsa-lib/blob/master/src/pcm/pcm_meter.c#L1… takes tens of seconds, untill the huge buffer is converted to s16. Usually the negative difference is quite small: s16_update 1: meter->now 988, s16->old 21139, size -20151 s16_update 1: meter->now 870, s16->old 18865, size -17995 s16_update 1: meter->now 1065, s16->old 20786, size -19721 s16_update 1: meter->now 701, s16->old 24839, size -24138 s16_update 1: meter->now 782, s16->old 869, size -87 s16_update 1: meter->now 763, s16->old 841, size -78 s16_update 1: meter->now 963, s16->old 22525, size -21562 s16_update 1: meter->now 665, s16->old 832, size -167 s16_update 1: meter->now 745, s16->old 865, size -120 s16_update 1: meter->now 859, s16->old 1002, size -143 s16_update 1: meter->now 935, s16->old 117452, size -116517 Yet in all these cases the size variable is incremented by 1445068800, which clearly does not make sense. Now the question is how it happens that meter->now < s16->old, when meter->now is calculated by now = status.appl_ptr - status.delay https://github.com/alsa-project/alsa-lib/blob/master/src/pcm/pcm_meter.c#L2… and s16->old is set to meter->now at the end of each s16_update call https://github.com/alsa-project/alsa-lib/blob/master/src/pcm/pcm_meter.c#L1… . The debug around the event looks like this, suddenly meter->now drops down: s16_update 1: meter->now 2567498, s16->old 2566593, size 905 s16_update 1: meter->now 2568401, s16->old 2567498, size 903 s16_update 1: meter->now 20786, s16->old 20786, size 0 s16_update 1: meter->now 1065, s16->old 20786, size -19721 s16_update 1: meter->now 24839, s16->old 24838, size 1 s16_update 1: meter->now 701, s16->old 24839, size -24138 s16_update 1: meter->now 1253162, s16->old 701, size 1252461 s16_update 1: meter->now 1255148, s16->old 1253162, size 1986 .......... s16_update 1: meter->now 11136, s16->old 10261, size 875 s16_update 1: meter->now 22525, s16->old 22524, size 1 s16_update 1: meter->now 963, s16->old 22525, size -21562 s16_update 1: meter->now 1270914, s16->old 963, size 1269951 s16_update 1: meter->now 1272917, s16->old 1270914, size 2003 Could it perhaps be an XRUN, causing the difference (status.appl_ptr - status.delay) to actually decrease compared to the previous run? I really do not know how/when such decrease can happen. But if it is a standard situation, than IMO the size of to-be converted buffer by the s16 scope should not be set to the huge pcm->boundary. Thanks a lot for help. Best regards, Pavel.

3 11

Re: [PATCH] ASoC: intel: Fix memory leak in sst_media_open()
by Markus Elfring 09 Aug '20

09 Aug '20

> When power_up_sst() fails, stream needs to be freed > just like when try_module_get() fails. However, current > code is returning directly and ends up leaking memory. Would an imperative wording be preferred for the change description? https://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux.git/tree/Doc… … > +++ b/sound/soc/intel/atom/sst-mfld-platform-pcm.c > @@ -330,8 +330,10 @@ static int sst_media_open(struct snd_pcm_substream *substream, > runtime->private_data = stream; > > ret_val = power_up_sst(stream); > - if (ret_val < 0) > + if (ret_val < 0) { > + kfree(stream); > return ret_val; > + } … I propose to add a jump target so that a bit of common exception handling code can be better reused at the end of this function implementation. Regards, Markus

1 0

[PATCH v4 00/12] ASoC: qcom: Add support for SC7180 lpass variant
by Rohit kumar 07 Aug '20

07 Aug '20

This patch chain add audio support for SC7180 soc by doing the required modification in existing common lpass-cpu/lpass-platform driver. This also fixes some concurrency issue. Changes since v3: - Fixed yaml documentation comments and make dt_binding_check issues. - Moved general fixes out of sc7180 specific patches as suggested by Srinivas. - Update clock-names to make it same as existing platforms. Ajit Pandey (4): ASoC: qcom: Add common array to initialize soc based core clocks ASoC: qcom: lpass-platform: Replace card->dev with component->dev include: dt-bindings: sound: Add sc7180-lpass bindings header ASoC: qcom: lpass-sc7180: Add platform driver for lpass audio Rohit kumar (8): ASoC: qcom: lpass-cpu: Move ahbix clk to platform specific function ASoC: qcom: lpass-platform: fix memory leak ASoC: qcom: lpass: Use regmap_field for i2sctl and dmactl registers ASoC: qcom: lpass-cpu: fix concurrency issue dt-bindings: sound: lpass-cpu: Add sc7180 lpass cpu node ASoC: qcom: lpass-cpu: Use platform_get_resource ASoC: qcom: lpass-platform: Use platform_get_irq dt-bindings: sound: lpass-cpu: Move to yaml format .../devicetree/bindings/sound/qcom,lpass-cpu.txt | 79 -------- .../devicetree/bindings/sound/qcom,lpass-cpu.yaml | 185 ++++++++++++++++++ include/dt-bindings/sound/sc7180-lpass.h | 10 + sound/soc/qcom/Kconfig | 5 + sound/soc/qcom/Makefile | 2 + sound/soc/qcom/lpass-apq8016.c | 86 ++++++-- sound/soc/qcom/lpass-cpu.c | 204 ++++++++++--------- sound/soc/qcom/lpass-ipq806x.c | 67 +++++++ sound/soc/qcom/lpass-lpaif-reg.h | 157 ++++++++------- sound/soc/qcom/lpass-platform.c | 155 +++++++++++---- sound/soc/qcom/lpass-sc7180.c | 216 +++++++++++++++++++++ sound/soc/qcom/lpass.h | 63 +++++- 12 files changed, 930 insertions(+), 299 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/qcom,lpass-cpu.txt create mode 100644 Documentation/devicetree/bindings/sound/qcom,lpass-cpu.yaml create mode 100644 include/dt-bindings/sound/sc7180-lpass.h create mode 100644 sound/soc/qcom/lpass-sc7180.c -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

4 27

aplay: fix for aborted capture streams
by GitHub pull_request - opened 07 Aug '20

07 Aug '20

alsa-project/alsa-utils pull request #51 was opened from mrajwa: This patch set fixes the issue with capture streams which were aborted by external signal like ctrl+c. Currently functions pcm_read() and pcm_readv() when aborted (in_aborting flag set) return the amount of requested frames instead of those actually read prior to interrupt. The consequence of this is repetition of recent X frames where X stands for amount of frames in one period. This problem is barely visible or rather *audible* when the period is small like few milliseconds because repetition of 1 [ms] of data is not-noticeable however if we use buffer and period sizes in seconds then the problem becomes apparent. Example issue -> https://github.com/thesofproject/sof/issues/3189 Request URL : https://github.com/alsa-project/alsa-utils/pull/51 Patch URL : https://github.com/alsa-project/alsa-utils/pull/51.patch Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[HDAudio][ALC285] Native speaker support for HP Spectre x360 15-eb0xxx
by Alexander Giese 07 Aug '20

07 Aug '20

Hello alsa-devel-team, I have a problem with the built-in speakers on the 2020 model of the hp-spectre. From my unexperienced point of view, the audio sinks are correctly identified and therefore I tried tweaking the GPIO pins with success. Information from alsa-info is here: http://alsa-project.org/db/?f=c8ee13e52fd6b2377573a7123c2887fe4d59c917 When I use hda-verb with the following parameters 2 of 4 of my built-in speakers start working: hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DATA 0x02 (or 0x04) hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DIR 0x01 hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_MASK 0x01 In addition, my mute-mic-led on the keyboard glows when: hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DATA 0x01 (or 0x02) hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DIR 0x04 hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_MASK 0x04 Since I do not understand how the GPIO-settings are integrated and applied in the upstream software for corresponding models, I would very much appreciate any help to get this to work out of the box in future releases so I do not have tweak my system all the time and others can benefit as well. Any suggestions on getting all 4 speakers to work or the mute-volume-led to glow are very much welcome alike! Thank you very much! Alexander

2 2

[PATCH] aplay: capture: write only the amount of read frames
by Rajwa, Marcin 07 Aug '20

07 Aug '20

This patch changes the way aborted stream is being saved. Currently when abort signal happens the write back of read samples is skipped but there is no reason to not save them. Also, we need to know how much frames have been read and write only those. Signed-off-by: Marcin Rajwa <marcin.rajwa(a)linux.intel.com> diff --git a/aplay/aplay.c b/aplay/aplay.c index 6836558..f5010dc 100644 --- a/aplay/aplay.c +++ b/aplay/aplay.c @@ -3204,11 +3204,12 @@ static void capture(char *orig_name) size_t c = (rest <= (off64_t)chunk_bytes) ? (size_t)rest : chunk_bytes; size_t f = c * 8 / bits_per_frame; - if (pcm_read(audiobuf, f) != f) { + size_t read = pcm_read(audiobuf, f); + size_t save; + if (read != f) in_aborting = 1; - break; - } - if (xwrite(fd, audiobuf, c) != c) { + save = read / 8 * bits_per_frame; + if (xwrite(fd, audiobuf, save) != save) { perror(name); in_aborting = 1; break;

1 0

[PATCH] Aplay: pcm_readv(): return read samples instead of requested upon abort
by Rajwa, Marcin 07 Aug '20

07 Aug '20

aplay: pcm_readv(): return read samples instead of requested upon abort This patch changes the logic of pcm_readv() when abort signal has been detected. During such condition we should return the amount of frames actually read instead of the size requested by caller. Signed-off-by: Marcin Rajwa <marcin.rajwa(a)linux.intel.com> diff --git a/aplay/aplay.c b/aplay/aplay.c index e8714d8..5d4b56e 100644 --- a/aplay/aplay.c +++ b/aplay/aplay.c @@ -2174,7 +2174,9 @@ static ssize_t pcm_readv(u_char **data, unsigned int channels, size_t rcount) count = chunk_size; } - while (count > 0 && !in_aborting) { + while (count > 0) { + if (in_aborting) + return result; unsigned int channel; void *bufs[channels]; size_t offset = result;

1 1

Correct conflicting mic in max98090
by GitHub pull_request - reopened 07 Aug '20

07 Aug '20

alsa-project/alsa-ucm-conf pull request #43 was reopened from jacobopantoja: 'mic' was conflicting 'mic' instead of 'headset', thus preventing correct configuration regarding microphones. With this corrected, introducing a headset enables 'headset' mic and disables internal 'mic', whereas releasing the headset disables 'headset' mic and enables internal 'mic' Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/43 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/43.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

request for missing access permission to libhinawa repository
by Takashi Sakamoto 07 Aug '20

07 Aug '20

Hi Iwai-san, As Jaroslav is absent so long[1] (summer vacation?), I would like you to arrange some team/repository settings for libhinawa. I lost my write permission to libhinawa repository[2] since Jaroslav's arrangement for github team. You are an owner of 'alsa-project' team in github.com, and I guess you have enough permission to configure below items: 1. setting enough permission for my account to libhinawa repository 2. add libhinawa repository to 'GObject Introspection' team[2] Would I request you the above? [1] https://github.com/orgs/alsa-project/teams/gobject-introspection/discussion… [2] https://github.com/alsa-project/libhinawa [3] https://github.com/orgs/alsa-project/teams/gobject-introspection/repositori… Thanks in advance Takashi Sakamoto

3 3

[PATCH v2 28/41] ARM: s3c24xx: move iis pinctrl config into boards
by Krzysztof Kozlowski 07 Aug '20

07 Aug '20

From: Arnd Bergmann <arnd(a)arndb.de> The s3c_gpio_cfgall_range() function is an internal interface of the samsung gpio driver and should not be called directly by drivers, so move the iis pin initialization into the boards. This means the pin configuration is only run once at early boot, rather than each time the driver binds, but the effect should be the same. Note that the s3c2412-i2s driver has no boards using it in mainline linux, the driver gets selected for the jive machine but is never instantiated. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> Acked-by: Linus Walleij <linus.walleij(a)linaro.org> Signed-off-by: Krzysztof Kozlowski <krzk(a)kernel.org> --- arch/arm/mach-s3c24xx/mach-gta02.c | 4 ++++ arch/arm/mach-s3c24xx/mach-h1940.c | 3 +++ arch/arm/mach-s3c24xx/mach-mini2440.c | 5 +++++ arch/arm/mach-s3c24xx/mach-n30.c | 5 +++++ arch/arm/mach-s3c24xx/mach-nexcoder.c | 5 +++++ arch/arm/mach-s3c24xx/mach-otom.c | 6 ++++++ arch/arm/mach-s3c24xx/mach-qt2410.c | 3 +++ arch/arm/mach-s3c24xx/mach-rx1950.c | 3 +++ arch/arm/mach-s3c24xx/mach-rx3715.c | 4 ++++ arch/arm/mach-s3c24xx/mach-smdk2410.c | 5 +++++ arch/arm/mach-s3c24xx/mach-smdk2413.c | 4 ++++ arch/arm/mach-s3c24xx/mach-smdk2440.c | 6 +++++- arch/arm/mach-s3c24xx/mach-vstms.c | 6 +++++- arch/arm/mach-s3c24xx/simtec-audio.c | 6 ++++++ sound/soc/samsung/s3c2412-i2s.c | 7 ------- sound/soc/samsung/s3c24xx-i2s.c | 7 ------- 16 files changed, 63 insertions(+), 16 deletions(-) diff --git a/arch/arm/mach-s3c24xx/mach-gta02.c b/arch/arm/mach-s3c24xx/mach-gta02.c index 46a80fae7056..3c7f2a3d00a5 100644 --- a/arch/arm/mach-s3c24xx/mach-gta02.c +++ b/arch/arm/mach-s3c24xx/mach-gta02.c @@ -538,6 +538,10 @@ static void __init gta02_machine_init(void) i2c_register_board_info(0, gta02_i2c_devs, ARRAY_SIZE(gta02_i2c_devs)); + /* Configure the I2S pins (GPE0...GPE4) in correct mode */ + s3c_gpio_cfgall_range(S3C2410_GPE(0), 5, S3C_GPIO_SFN(2), + S3C_GPIO_PULL_NONE); + gpiod_add_lookup_table(&gta02_audio_gpio_table); platform_add_devices(gta02_devices, ARRAY_SIZE(gta02_devices)); pm_power_off = gta02_poweroff; diff --git a/arch/arm/mach-s3c24xx/mach-h1940.c b/arch/arm/mach-s3c24xx/mach-h1940.c index c7269cd26ba8..1f18ac4e84b2 100644 --- a/arch/arm/mach-s3c24xx/mach-h1940.c +++ b/arch/arm/mach-s3c24xx/mach-h1940.c @@ -707,6 +707,9 @@ static void __init h1940_init(void) s3c24xx_fb_set_platdata(&h1940_fb_info); gpiod_add_lookup_table(&h1940_mmc_gpio_table); gpiod_add_lookup_table(&h1940_audio_gpio_table); + /* Configure the I2S pins (GPE0...GPE4) in correct mode */ + s3c_gpio_cfgall_range(S3C2410_GPE(0), 5, S3C_GPIO_SFN(2), + S3C_GPIO_PULL_NONE); s3c24xx_mci_set_platdata(&h1940_mmc_cfg); s3c24xx_udc_set_platdata(&h1940_udc_cfg); s3c24xx_ts_set_platdata(&h1940_ts_cfg); diff --git a/arch/arm/mach-s3c24xx/mach-mini2440.c b/arch/arm/mach-s3c24xx/mach-mini2440.c index 65f48094114e..0bd2746f19a6 100644 --- a/arch/arm/mach-s3c24xx/mach-mini2440.c +++ b/arch/arm/mach-s3c24xx/mach-mini2440.c @@ -714,6 +714,11 @@ static void __init mini2440_init(void) s3c_gpio_setpull(mini2440_buttons[i].gpio, S3C_GPIO_PULL_UP); s3c_gpio_cfgpin(mini2440_buttons[i].gpio, S3C2410_GPIO_INPUT); } + + /* Configure the I2S pins (GPE0...GPE4) in correct mode */ + s3c_gpio_cfgall_range(S3C2410_GPE(0), 5, S3C_GPIO_SFN(2), + S3C_GPIO_PULL_NONE); + if (features.lcd_index != -1) { int li; diff --git a/arch/arm/mach-s3c24xx/mach-n30.c b/arch/arm/mach-s3c24xx/mach-n30.c index 152f09242579..9410fcb82340 100644 --- a/arch/arm/mach-s3c24xx/mach-n30.c +++ b/arch/arm/mach-s3c24xx/mach-n30.c @@ -36,6 +36,7 @@ #include <mach/regs-gpio.h> #include <mach/regs-lcd.h> #include <mach/gpio-samsung.h> +#include <plat/gpio-cfg.h> #include <asm/mach/arch.h> #include <asm/mach/irq.h> @@ -599,6 +600,10 @@ static void __init n30_init(void) S3C2410_MISCCR_USBSUSPND0 | S3C2410_MISCCR_USBSUSPND1, 0x0); + /* Configure the I2S pins (GPE0...GPE4) in correct mode */ + s3c_gpio_cfgall_range(S3C2410_GPE(0), 5, S3C_GPIO_SFN(2), + S3C_GPIO_PULL_NONE); + if (machine_is_n30()) { /* Turn off suspend on both USB ports, and switch the * selectable USB port to USB device mode. */ diff --git a/arch/arm/mach-s3c24xx/mach-nexcoder.c b/arch/arm/mach-s3c24xx/mach-nexcoder.c index 5385437ef464..ec23b115261e 100644 --- a/arch/arm/mach-s3c24xx/mach-nexcoder.c +++ b/arch/arm/mach-s3c24xx/mach-nexcoder.c @@ -143,6 +143,11 @@ static void __init nexcoder_init_time(void) static void __init nexcoder_init(void) { s3c_i2c0_set_platdata(NULL); + + /* Configure the I2S pins (GPE0...GPE4) in correct mode */ + s3c_gpio_cfgall_range(S3C2410_GPE(0), 5, S3C_GPIO_SFN(2), + S3C_GPIO_PULL_NONE); + platform_add_devices(nexcoder_devices, ARRAY_SIZE(nexcoder_devices)); }; diff --git a/arch/arm/mach-s3c24xx/mach-otom.c b/arch/arm/mach-s3c24xx/mach-otom.c index e7eec5ccddb3..1f777ab4ced7 100644 --- a/arch/arm/mach-s3c24xx/mach-otom.c +++ b/arch/arm/mach-s3c24xx/mach-otom.c @@ -23,6 +23,8 @@ #include <asm/mach/irq.h> #include <mach/regs-gpio.h> +#include <mach/gpio-samsung.h> +#include <plat/gpio-cfg.h> #include <plat/cpu.h> #include <plat/devs.h> @@ -105,6 +107,10 @@ static void __init otom11_init_time(void) static void __init otom11_init(void) { s3c_i2c0_set_platdata(NULL); + + /* Configure the I2S pins (GPE0...GPE4) in correct mode */ + s3c_gpio_cfgall_range(S3C2410_GPE(0), 5, S3C_GPIO_SFN(2), + S3C_GPIO_PULL_NONE); platform_add_devices(otom11_devices, ARRAY_SIZE(otom11_devices)); } diff --git a/arch/arm/mach-s3c24xx/mach-qt2410.c b/arch/arm/mach-s3c24xx/mach-qt2410.c index 4ac3d0531992..5e33f3da629e 100644 --- a/arch/arm/mach-s3c24xx/mach-qt2410.c +++ b/arch/arm/mach-s3c24xx/mach-qt2410.c @@ -341,6 +341,9 @@ static void __init qt2410_machine_init(void) s3c24xx_udc_set_platdata(&qt2410_udc_cfg); s3c_i2c0_set_platdata(NULL); + /* Configure the I2S pins (GPE0...GPE4) in correct mode */ + s3c_gpio_cfgall_range(S3C2410_GPE(0), 5, S3C_GPIO_SFN(2), + S3C_GPIO_PULL_NONE); gpiod_add_lookup_table(&qt2410_spi_gpiod_table); s3c_gpio_setpull(S3C2410_GPB(0), S3C_GPIO_PULL_NONE); gpiod_add_lookup_table(&qt2410_led_gpio_table); diff --git a/arch/arm/mach-s3c24xx/mach-rx1950.c b/arch/arm/mach-s3c24xx/mach-rx1950.c index a5461b2227bd..86d348f33972 100644 --- a/arch/arm/mach-s3c24xx/mach-rx1950.c +++ b/arch/arm/mach-s3c24xx/mach-rx1950.c @@ -828,6 +828,9 @@ static void __init rx1950_init_machine(void) pwm_add_table(rx1950_pwm_lookup, ARRAY_SIZE(rx1950_pwm_lookup)); gpiod_add_lookup_table(&rx1950_audio_gpio_table); + /* Configure the I2S pins (GPE0...GPE4) in correct mode */ + s3c_gpio_cfgall_range(S3C2410_GPE(0), 5, S3C_GPIO_SFN(2), + S3C_GPIO_PULL_NONE); platform_add_devices(rx1950_devices, ARRAY_SIZE(rx1950_devices)); i2c_register_board_info(0, rx1950_i2c_devices, diff --git a/arch/arm/mach-s3c24xx/mach-rx3715.c b/arch/arm/mach-s3c24xx/mach-rx3715.c index 617392de01ac..fc197cee77a0 100644 --- a/arch/arm/mach-s3c24xx/mach-rx3715.c +++ b/arch/arm/mach-s3c24xx/mach-rx3715.c @@ -38,6 +38,7 @@ #include <mach/regs-gpio.h> #include <mach/regs-lcd.h> #include <mach/gpio-samsung.h> +#include <plat/gpio-cfg.h> #include <plat/cpu.h> #include <plat/devs.h> @@ -197,6 +198,9 @@ static void __init rx3715_init_machine(void) s3c_nand_set_platdata(&rx3715_nand_info); s3c24xx_fb_set_platdata(&rx3715_fb_info); + /* Configure the I2S pins (GPE0...GPE4) in correct mode */ + s3c_gpio_cfgall_range(S3C2410_GPE(0), 5, S3C_GPIO_SFN(2), + S3C_GPIO_PULL_NONE); platform_add_devices(rx3715_devices, ARRAY_SIZE(rx3715_devices)); } diff --git a/arch/arm/mach-s3c24xx/mach-smdk2410.c b/arch/arm/mach-s3c24xx/mach-smdk2410.c index 1d4d8735f283..c78fe1d8802c 100644 --- a/arch/arm/mach-s3c24xx/mach-smdk2410.c +++ b/arch/arm/mach-s3c24xx/mach-smdk2410.c @@ -19,6 +19,8 @@ #include <linux/serial_s3c.h> #include <linux/platform_device.h> #include <linux/io.h> +#include <mach/gpio-samsung.h> +#include <plat/gpio-cfg.h> #include <asm/mach/arch.h> #include <asm/mach/map.h> @@ -92,6 +94,9 @@ static void __init smdk2410_init(void) { s3c_i2c0_set_platdata(NULL); platform_add_devices(smdk2410_devices, ARRAY_SIZE(smdk2410_devices)); + /* Configure the I2S pins (GPE0...GPE4) in correct mode */ + s3c_gpio_cfgall_range(S3C2410_GPE(0), 5, S3C_GPIO_SFN(2), + S3C_GPIO_PULL_NONE); smdk_machine_init(); } diff --git a/arch/arm/mach-s3c24xx/mach-smdk2413.c b/arch/arm/mach-s3c24xx/mach-smdk2413.c index 9782cc3e698c..287bd502a030 100644 --- a/arch/arm/mach-s3c24xx/mach-smdk2413.c +++ b/arch/arm/mach-s3c24xx/mach-smdk2413.c @@ -37,6 +37,7 @@ #include <linux/platform_data/i2c-s3c2410.h> #include <mach/gpio-samsung.h> #include <mach/fb.h> +#include <plat/gpio-cfg.h> #include <plat/devs.h> #include <plat/cpu.h> @@ -118,6 +119,9 @@ static void __init smdk2413_machine_init(void) s3c24xx_udc_set_platdata(&smdk2413_udc_cfg); s3c_i2c0_set_platdata(NULL); + /* Configure the I2S pins (GPE0...GPE4) in correct mode */ + s3c_gpio_cfgall_range(S3C2410_GPE(0), 5, S3C_GPIO_SFN(2), + S3C_GPIO_PULL_NONE); platform_add_devices(smdk2413_devices, ARRAY_SIZE(smdk2413_devices)); smdk_machine_init(); diff --git a/arch/arm/mach-s3c24xx/mach-smdk2440.c b/arch/arm/mach-s3c24xx/mach-smdk2440.c index bd6a35105518..5939372ecec2 100644 --- a/arch/arm/mach-s3c24xx/mach-smdk2440.c +++ b/arch/arm/mach-s3c24xx/mach-smdk2440.c @@ -28,6 +28,8 @@ #include <mach/regs-gpio.h> #include <mach/regs-lcd.h> +#include <mach/gpio-samsung.h> +#include <plat/gpio-cfg.h> #include <mach/fb.h> #include <linux/platform_data/i2c-s3c2410.h> @@ -165,7 +167,9 @@ static void __init smdk2440_machine_init(void) { s3c24xx_fb_set_platdata(&smdk2440_fb_info); s3c_i2c0_set_platdata(NULL); - + /* Configure the I2S pins (GPE0...GPE4) in correct mode */ + s3c_gpio_cfgall_range(S3C2410_GPE(0), 5, S3C_GPIO_SFN(2), + S3C_GPIO_PULL_NONE); platform_add_devices(smdk2440_devices, ARRAY_SIZE(smdk2440_devices)); smdk_machine_init(); } diff --git a/arch/arm/mach-s3c24xx/mach-vstms.c b/arch/arm/mach-s3c24xx/mach-vstms.c index f0acbaa6dfd6..c6e777aab24e 100644 --- a/arch/arm/mach-s3c24xx/mach-vstms.c +++ b/arch/arm/mach-s3c24xx/mach-vstms.c @@ -30,6 +30,8 @@ #include <mach/regs-gpio.h> #include <mach/regs-lcd.h> +#include <mach/gpio-samsung.h> +#include <plat/gpio-cfg.h> #include <mach/fb.h> @@ -147,7 +149,9 @@ static void __init vstms_init(void) { s3c_i2c0_set_platdata(NULL); s3c_nand_set_platdata(&vstms_nand_info); - + /* Configure the I2S pins (GPE0...GPE4) in correct mode */ + s3c_gpio_cfgall_range(S3C2410_GPE(0), 5, S3C_GPIO_SFN(2), + S3C_GPIO_PULL_NONE); platform_add_devices(vstms_devices, ARRAY_SIZE(vstms_devices)); } diff --git a/arch/arm/mach-s3c24xx/simtec-audio.c b/arch/arm/mach-s3c24xx/simtec-audio.c index cfe78bf8cc9b..af87261e6a59 100644 --- a/arch/arm/mach-s3c24xx/simtec-audio.c +++ b/arch/arm/mach-s3c24xx/simtec-audio.c @@ -13,6 +13,8 @@ #include <linux/io.h> #include <mach/regs-gpio.h> +#include <mach/gpio-samsung.h> +#include <plat/gpio-cfg.h> #include <linux/platform_data/asoc-s3c24xx_simtec.h> #include <plat/devs.h> @@ -64,6 +66,10 @@ int __init simtec_audio_add(const char *name, bool has_lr_routing, if (has_lr_routing) simtec_audio_platdata.startup = simtec_audio_startup_lrroute; + /* Configure the I2S pins (GPE0...GPE4) in correct mode */ + s3c_gpio_cfgall_range(S3C2410_GPE(0), 5, S3C_GPIO_SFN(2), + S3C_GPIO_PULL_NONE); + platform_device_register(&s3c_device_iis); platform_device_register(&simtec_audio_dev); return 0; diff --git a/sound/soc/samsung/s3c2412-i2s.c b/sound/soc/samsung/s3c2412-i2s.c index edfa9d11d2e5..81f416ac457e 100644 --- a/sound/soc/samsung/s3c2412-i2s.c +++ b/sound/soc/samsung/s3c2412-i2s.c @@ -19,9 +19,6 @@ #include <sound/soc.h> #include <sound/pcm_params.h> -#include <mach/gpio-samsung.h> -#include <plat/gpio-cfg.h> - #include "dma.h" #include "regs-i2s-v2.h" #include "s3c2412-i2s.h" @@ -70,10 +67,6 @@ static int s3c2412_i2s_probe(struct snd_soc_dai *dai) if (ret) goto err; - /* Configure the I2S pins (GPE0...GPE4) in correct mode */ - s3c_gpio_cfgall_range(S3C2410_GPE(0), 5, S3C_GPIO_SFN(2), - S3C_GPIO_PULL_NONE); - return 0; err: diff --git a/sound/soc/samsung/s3c24xx-i2s.c b/sound/soc/samsung/s3c24xx-i2s.c index 60bfaed5f7a6..50c08008aacb 100644 --- a/sound/soc/samsung/s3c24xx-i2s.c +++ b/sound/soc/samsung/s3c24xx-i2s.c @@ -18,10 +18,7 @@ #include <sound/soc.h> #include <sound/pcm_params.h> -#include <mach/gpio-samsung.h> -#include <plat/gpio-cfg.h> #include "regs-iis.h" - #include "dma.h" #include "s3c24xx-i2s.h" @@ -348,10 +345,6 @@ static int s3c24xx_i2s_probe(struct snd_soc_dai *dai) if (ret) return ret; - /* Configure the I2S pins (GPE0...GPE4) in correct mode */ - s3c_gpio_cfgall_range(S3C2410_GPE(0), 5, S3C_GPIO_SFN(2), - S3C_GPIO_PULL_NONE); - writel(S3C2410_IISCON_IISEN, s3c24xx_i2s.regs + S3C2410_IISCON); s3c24xx_snd_txctrl(0); -- 2.17.1

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Re: [PATCH v2 00/41] spi / fbdev / cpufreq / usb / mmc / hwmon / ARM: Prepare for multiplatform S3C
by Mark Brown 07 Aug '20

07 Aug '20

On Thu, Aug 06, 2020 at 08:19:32PM +0200, Krzysztof Kozlowski wrote: > I intend to take it through Samsung SoC tree so all Acks are welcomed. Any issue with me applying the ASoC bits and sending you a branch? This seems like it might run into some of the subsystem wide cleanups we've got going on.

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[PATCH v2 18/41] ARM: s5pv210: don't imply CONFIG_PLAT_SAMSUNG
by Krzysztof Kozlowski 07 Aug '20

07 Aug '20

From: Arnd Bergmann <arnd(a)arndb.de> The plat-samsung directory and mach-s5pv210 can be build completely independently, so split the two Kconfig symbols CONFIG_PLAT_SAMSUNG and CONFIG_ARCH_S5PV210. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> Acked-by: Ulf Hansson <ulf.hansson(a)linaro.org> Acked-by: Thierry Reding <thierry.reding(a)gmail.com> Signed-off-by: Krzysztof Kozlowski <krzk(a)kernel.org> --- arch/arm/Kconfig.debug | 6 +++--- arch/arm/Makefile | 1 - arch/arm/plat-samsung/Kconfig | 2 +- drivers/mmc/host/Kconfig | 2 +- drivers/pwm/Kconfig | 2 +- drivers/spi/Kconfig | 2 +- drivers/tty/serial/Kconfig | 2 +- sound/soc/samsung/Kconfig | 2 +- 8 files changed, 9 insertions(+), 10 deletions(-) diff --git a/arch/arm/Kconfig.debug b/arch/arm/Kconfig.debug index 153a9a35dd20..b4b41f9d246b 100644 --- a/arch/arm/Kconfig.debug +++ b/arch/arm/Kconfig.debug @@ -1005,7 +1005,7 @@ choice via SCIFA4 on Renesas SH-Mobile AG5 (SH73A0). config DEBUG_S3C_UART0 - depends on PLAT_SAMSUNG || ARCH_EXYNOS + depends on PLAT_SAMSUNG || ARCH_S5PV210 || ARCH_EXYNOS select DEBUG_EXYNOS_UART if ARCH_EXYNOS select DEBUG_S3C24XX_UART if ARCH_S3C24XX select DEBUG_S3C64XX_UART if ARCH_S3C64XX @@ -1017,7 +1017,7 @@ choice by the boot-loader before use. config DEBUG_S3C_UART1 - depends on PLAT_SAMSUNG || ARCH_EXYNOS + depends on PLAT_SAMSUNG || ARCH_S5PV210 || ARCH_EXYNOS select DEBUG_EXYNOS_UART if ARCH_EXYNOS select DEBUG_S3C24XX_UART if ARCH_S3C24XX select DEBUG_S3C64XX_UART if ARCH_S3C64XX @@ -1029,7 +1029,7 @@ choice by the boot-loader before use. config DEBUG_S3C_UART2 - depends on PLAT_SAMSUNG || ARCH_EXYNOS + depends on PLAT_SAMSUNG || ARCH_S5PV210 || ARCH_EXYNOS select DEBUG_EXYNOS_UART if ARCH_EXYNOS select DEBUG_S3C24XX_UART if ARCH_S3C24XX select DEBUG_S3C64XX_UART if ARCH_S3C64XX diff --git a/arch/arm/Makefile b/arch/arm/Makefile index 826613a20b7d..3997cec8848e 100644 --- a/arch/arm/Makefile +++ b/arch/arm/Makefile @@ -234,7 +234,6 @@ machine-$(CONFIG_PLAT_SPEAR) += spear # by CONFIG_* macro name. plat-$(CONFIG_ARCH_OMAP) += omap plat-$(CONFIG_ARCH_S3C64XX) += samsung -plat-$(CONFIG_ARCH_S5PV210) += samsung plat-$(CONFIG_PLAT_ORION) += orion plat-$(CONFIG_PLAT_PXA) += pxa plat-$(CONFIG_PLAT_S3C24XX) += samsung diff --git a/arch/arm/plat-samsung/Kconfig b/arch/arm/plat-samsung/Kconfig index a7cb0db1d5cc..13b39684b582 100644 --- a/arch/arm/plat-samsung/Kconfig +++ b/arch/arm/plat-samsung/Kconfig @@ -4,7 +4,7 @@ config PLAT_SAMSUNG bool - depends on PLAT_S3C24XX || ARCH_S3C64XX || ARCH_S5PV210 + depends on PLAT_S3C24XX || ARCH_S3C64XX default y select GENERIC_IRQ_CHIP select NO_IOPORT_MAP diff --git a/drivers/mmc/host/Kconfig b/drivers/mmc/host/Kconfig index ddce8e62280c..9f387daa764a 100644 --- a/drivers/mmc/host/Kconfig +++ b/drivers/mmc/host/Kconfig @@ -289,7 +289,7 @@ config MMC_SDHCI_TEGRA config MMC_SDHCI_S3C tristate "SDHCI support on Samsung S3C SoC" - depends on MMC_SDHCI && (PLAT_SAMSUNG || ARCH_EXYNOS) + depends on MMC_SDHCI && (PLAT_SAMSUNG || ARCH_S5PV210 || ARCH_EXYNOS) help This selects the Secure Digital Host Controller Interface (SDHCI) often referrered to as the HSMMC block in some of the Samsung S3C diff --git a/drivers/pwm/Kconfig b/drivers/pwm/Kconfig index 7dbcf6973d33..9448e4ca8c73 100644 --- a/drivers/pwm/Kconfig +++ b/drivers/pwm/Kconfig @@ -410,7 +410,7 @@ config PWM_ROCKCHIP config PWM_SAMSUNG tristate "Samsung PWM support" - depends on PLAT_SAMSUNG || ARCH_EXYNOS || COMPILE_TEST + depends on PLAT_SAMSUNG || ARCH_S5PV210 || ARCH_EXYNOS || COMPILE_TEST help Generic PWM framework driver for Samsung. diff --git a/drivers/spi/Kconfig b/drivers/spi/Kconfig index c3008e423f59..85619ab64156 100644 --- a/drivers/spi/Kconfig +++ b/drivers/spi/Kconfig @@ -680,7 +680,7 @@ config SPI_S3C24XX_FIQ config SPI_S3C64XX tristate "Samsung S3C64XX series type SPI" - depends on (PLAT_SAMSUNG || ARCH_EXYNOS || COMPILE_TEST) + depends on (PLAT_SAMSUNG || ARCH_S5PV210 || ARCH_EXYNOS || COMPILE_TEST) help SPI driver for Samsung S3C64XX and newer SoCs. diff --git a/drivers/tty/serial/Kconfig b/drivers/tty/serial/Kconfig index 8a0352eb337c..10de494e808a 100644 --- a/drivers/tty/serial/Kconfig +++ b/drivers/tty/serial/Kconfig @@ -235,7 +235,7 @@ config SERIAL_CLPS711X_CONSOLE config SERIAL_SAMSUNG tristate "Samsung SoC serial support" - depends on PLAT_SAMSUNG || ARCH_EXYNOS || COMPILE_TEST + depends on PLAT_SAMSUNG || ARCH_S5PV210 || ARCH_EXYNOS || COMPILE_TEST select SERIAL_CORE help Support for the on-chip UARTs on the Samsung S3C24XX series CPUs, diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index 1431be4ed054..a2221ebb1b6a 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -1,7 +1,7 @@ # SPDX-License-Identifier: GPL-2.0-only menuconfig SND_SOC_SAMSUNG tristate "ASoC support for Samsung" - depends on PLAT_SAMSUNG || ARCH_EXYNOS || COMPILE_TEST + depends on PLAT_SAMSUNG || ARCH_S5PV210 || ARCH_EXYNOS || COMPILE_TEST depends on COMMON_CLK select SND_SOC_GENERIC_DMAENGINE_PCM help -- 2.17.1

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[PATCH v5 00/11] Add ASoC AHUB components for Tegra210 and later
by Sameer Pujar 07 Aug '20

07 Aug '20

Overview ======== Audio Processing Engine (APE) comprises of Audio DMA (ADMA) and Audio Hub (AHUB) unit. AHUB is a collection of hardware accelerators for audio pre-processing and post-processing. It also includes a programmable full crossbar for routing audio data across these accelerators. This series exposes some of these below mentioned HW devices as ASoC components for Tegra platforms from Tegra210 onwards. * ADMAIF : The interface between ADMA and AHUB * XBAR : Crossbar for routing audio samples across various modules * I2S : Inter-IC Sound Controller * DMIC : Digital Microphone * DSPK : Digital Speaker Following is the summary of current series. * Add YAML DT binding documentation for above mentioned modules. * Helper function for ACIF programming is exposed for Tegra210 and later. * Add ASoC driver components for each of the above modules. * Build ACONNECT and ADMA drivers which are essential to realize audio use case. * Add DT entries for above components for Tegra210, Tegra186 and Tegra194. As per the suggestion in [0] audio graph based sound card support is pushed in a separate series. [0] https://lkml.org/lkml/2020/6/27/4 Changelog ========= v4 -> v5 -------- * Common changes - simple-card driver changes are dropped. Changes are migrated to audio graph card and are moved to a separate series as suggested. - '#sound-dai-cells' property is not needed for planned audio graph card Hence dropped from documentation and related DT binding of component drivers. - CIF and DAP DAIs are added for I/O drivers (DMIC, DSPK, I2S) to represent DAI links using audio graph card. Similary DAIs are added in AHUB driver to describe endpoints in audio crossbar. Routing is updated to reflect the same in drivers. v3 -> v4 -------- * [1/23] "ASoC: dt-bindings: tegra: Add DT bindings for Tegra210" - Removed multiple examples and retained one example per doc - Fixed as per inputs on the previous series - Tested bindings with 'make dt_binding_check/dtbs_check' * [2/23] "ASoC: tegra: Add support for CIF programming" - No change * Common changes (for patch [3/10] to [7/10]) - Mixer control overrides, for PCM parameters (rate, channel, bits), in each driver are dropped. - Updated routing as per DPCM usage - Minor changes related to formatting * New changes (patch [8/23] to [18/23] and patch [23/23]) - Based on discussions in following threads DPCM is used for Tegra Audio. https://lkml.org/lkml/2020/2/20/91 https://lkml.org/lkml/2020/4/30/519 - The simple-card driver is used for Tegra Audio and accordingly some enhancements are made in simple-card and core drivers. - Patch [8/23] to [18/23] are related to simple-card and core changes. - Patch [23/23] adds sound card support to realize complete audio path. This is based on simple-card driver with proposed enhancements. - Re-ordered patches depending on above v2 -> v3 -------- * [1/10] "dt-bindings: sound: tegra: add DT binding for AHUB - Updated licence - Removed redundancy w.r.t items/const/enum - Added constraints wherever needed with "pattern" property * [2/10] "ASoC: tegra: add support for CIF programming" - Removed tegra_cif.c - Instead added inline helper function in tegra_cif.h * common changes (for patch [3/10] to [7/10]) - Replace LATE system calls with Normal sleep - Remove explicit RPM suspend in driver remove() call - Use devm_kzalloc() instead of devm_kcalloc() for single element - Replace 'ret' with 'err' for better reading - Consistent error printing style across drivers - Minor formating fixes * [8/10] "arm64: tegra: add AHUB components for few Tegra chips" - no change * [9/10] "arm64: tegra: enable AHUB modules for few Tegra chips" - no change * [10/10] "arm64: defconfig: enable AHUB components for Tegra210 and later" (New patch) - Enables ACONNECT and AHUB components. With this AHUB and components are registered with ASoC core. v1 -> v2 -------- * [1/9] "dt-bindings: sound: tegra: add DT binding for AHUB" - no changes * [2/9] "ASoC: tegra: add support for CIF programming" - removed CIF programming changes for legacy chips. - this patch now exposes helper function for CIF programming, which can be used on Tegra210 later. - later tegra_cif.c can be extended for legacy chips as well. - updated commit message accordingly * [3/9] "ASoC: tegra: add Tegra210 based DMIC driver" - removed unnecessary initialization of 'ret' in probe() * [4/9] "ASoC: tegra: add Tegra210 based I2S driver" - removed unnecessary initialization of 'ret' in probe() - fixed indentation - added consistent bracing for if-else clauses - updated 'rx_fifo_th' type to 'unsigned int' - used BIT() macro for defines like '1 << {x}' in tegra210_i2s.h * [5/9] "ASoC: tegra: add Tegra210 based AHUB driver" - used of_device_get_match_data() to get 'soc_data' and removed explicit of_match_device() - used devm_platform_ioremap_resource() and removed explicit platform_get_resource() - fixed indentation for devm_snd_soc_register_component() - updated commit message - updated commit message to reflect compatible binding for Tegra186 and Tegra194. * [6/9] "ASoC: tegra: add Tegra186 based DSPK driver" - removed unnecessary initialization of 'ret' in probe() - updated 'max_th' to 'unsigned int' - shortened lengthy macro names to avoid wrapping in tegra186_dspk_wr_reg() and to be consistent * [7/9] "ASoC: tegra: add Tegra210 based ADMAIF driver" - used of_device_get_match_data() and removed explicit of_match_device() - used BIT() macro for defines like '1 << {x}' in tegra210_admaif.h - updated commit message to reflect compatible binding for Tegra186 and Tegra194. * [8/9] "arm64: tegra: add AHUB components for few Tegra chips" - no change * [9/9] "arm64: tegra: enable AHUB modules for few Tegra chips" - no change * common changes for patch [3/9] to [7/9] - sorted headers in alphabetical order - moved MODULE_DEVICE_TABLE() right below *_of_match table - removed macro DRV_NAME - removed explicit 'owner' field from platform_driver structure - added 'const' to snd_soc_dai_ops structure Sameer Pujar (11): ASoC: dt-bindings: tegra: Add DT bindings for Tegra210 ASoC: tegra: Add support for CIF programming ASoC: tegra: Add Tegra210 based DMIC driver ASoC: tegra: Add Tegra210 based I2S driver ASoC: tegra: Add Tegra210 based AHUB driver ASoC: tegra: Add Tegra186 based DSPK driver ASoC: tegra: Add Tegra210 based ADMAIF driver arm64: defconfig: Build AHUB component drivers arm64: defconfig: Build ADMA and ACONNECT driver arm64: tegra: Enable ACONNECT, ADMA and AGIC on Jetson Nano arm64: tegra: Add DT binding for AHUB components .../bindings/sound/nvidia,tegra186-dspk.yaml | 83 +++ .../bindings/sound/nvidia,tegra210-admaif.yaml | 111 +++ .../bindings/sound/nvidia,tegra210-ahub.yaml | 136 ++++ .../bindings/sound/nvidia,tegra210-dmic.yaml | 83 +++ .../bindings/sound/nvidia,tegra210-i2s.yaml | 101 +++ arch/arm64/boot/dts/nvidia/tegra186.dtsi | 217 +++++- arch/arm64/boot/dts/nvidia/tegra194.dtsi | 225 +++++- arch/arm64/boot/dts/nvidia/tegra210-p3450-0000.dts | 12 + arch/arm64/boot/dts/nvidia/tegra210.dtsi | 140 ++++ arch/arm64/configs/defconfig | 8 + sound/soc/tegra/Kconfig | 56 ++ sound/soc/tegra/Makefile | 10 + sound/soc/tegra/tegra186_dspk.c | 442 +++++++++++ sound/soc/tegra/tegra186_dspk.h | 70 ++ sound/soc/tegra/tegra210_admaif.c | 800 ++++++++++++++++++++ sound/soc/tegra/tegra210_admaif.h | 162 ++++ sound/soc/tegra/tegra210_ahub.c | 676 +++++++++++++++++ sound/soc/tegra/tegra210_ahub.h | 127 ++++ sound/soc/tegra/tegra210_dmic.c | 455 ++++++++++++ sound/soc/tegra/tegra210_dmic.h | 82 +++ sound/soc/tegra/tegra210_i2s.c | 812 +++++++++++++++++++++ sound/soc/tegra/tegra210_i2s.h | 126 ++++ sound/soc/tegra/tegra_cif.h | 65 ++ sound/soc/tegra/tegra_pcm.c | 235 +++++- sound/soc/tegra/tegra_pcm.h | 21 +- 25 files changed, 5251 insertions(+), 4 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra186-dspk.yaml create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-admaif.yaml create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-ahub.yaml create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-dmic.yaml create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-i2s.yaml create mode 100644 sound/soc/tegra/tegra186_dspk.c create mode 100644 sound/soc/tegra/tegra186_dspk.h create mode 100644 sound/soc/tegra/tegra210_admaif.c create mode 100644 sound/soc/tegra/tegra210_admaif.h create mode 100644 sound/soc/tegra/tegra210_ahub.c create mode 100644 sound/soc/tegra/tegra210_ahub.h create mode 100644 sound/soc/tegra/tegra210_dmic.c create mode 100644 sound/soc/tegra/tegra210_dmic.h create mode 100644 sound/soc/tegra/tegra210_i2s.c create mode 100644 sound/soc/tegra/tegra210_i2s.h create mode 100644 sound/soc/tegra/tegra_cif.h -- 2.7.4

4 21

[PATCH] ALSA: usb-audio: Fix memleak in scarlett2_add_new_ctl
by Dinghao Liu 07 Aug '20

07 Aug '20

When snd_usb_mixer_add_control() fails, elem needs to be freed just like when snd_ctl_new1() fails. However, current code is returning directly and ends up leaking memory. Fixes: 9e4d5c1be21f0 ("ALSA: usb-audio: Scarlett Gen 2 mixer interface") Signed-off-by: Dinghao Liu <dinghao.liu(a)zju.edu.cn> --- sound/usb/mixer_scarlett_gen2.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/usb/mixer_scarlett_gen2.c b/sound/usb/mixer_scarlett_gen2.c index 74c00c905d24..4b2da0866cdc 100644 --- a/sound/usb/mixer_scarlett_gen2.c +++ b/sound/usb/mixer_scarlett_gen2.c @@ -964,8 +964,10 @@ static int scarlett2_add_new_ctl(struct usb_mixer_interface *mixer, strlcpy(kctl->id.name, name, sizeof(kctl->id.name)); err = snd_usb_mixer_add_control(&elem->head, kctl); - if (err < 0) + if (err < 0) { + kfree(elem); return err; + } if (kctl_return) *kctl_return = kctl; -- 2.17.1

3 2

Why doesn't mixer control (values) have some kind of locking mechanism? (mutex?)
by Tom Yan 07 Aug '20

07 Aug '20

Hi all, I just wonder if it's a "no one cares" or a "no one was aware of it" issue (or maybe both?). When you change (integer) values (e.g. volume) of a mixer control, it usually (if not always) involves calling two functions/methods of a snd_kcontrol_new, which are get and put, in order to do relative volume adjustments. (Apparently it is often done relatively even if we have absolute values, for reasons.) While these two "actions" can be and probably are mostly "atomic" (with the help of mutex) in the kernel drivers *respectively*, they are not and cannot be atomic as a whole. This won't really be an issue when the actions (either for one or multiple channels) are done "synchronously" in *one* program run (e.g. amixer -c STX set Master 1+). However, if such a program run is issued multiple times "asynchronously" (e.g. binding it to some XF86Audio{Raise,Lower}Volume scroll wheel), volume adjustment becomes a total mess / failure. If it isn't obvious enough. it could happen like the following: get1(100 100) set1(101 100) get2(101 100) set2(102 100) ... Or worse: get1(100 100) get2(100 100) set1(101 100) set2(100 101) ... Not only that it may/will not finish the first set of adjustments for all channels before the second, get() from the second set could happen before set() from the first, reverting the effect of the earlier one(s). Certainly one can use something like `flock` with amixer to make sure the atomicity of each issue/run, but not only that it looks silly and primitive, we don't always manipulate the mixer control with an "executable". For example, this weird issue in pulseaudio is probably related: https://bugs.freedesktop.org/show_bug.cgi?id=92717 So I wonder, is there a particular reason that mixer control doesn't possess some form of lock, which allows any form of userspace manipulation to lock it until what should be / is considered atomic is finished?

4 14

aplay: change if statement to else if in xrun function
by GitHub pull_request - edited 07 Aug '20

07 Aug '20

alsa-project/alsa-utils pull request #50 was edited from JSFamily: Honestly, It does not affect behavior in runtime. But I made this patch to help understanding the source code. Request URL : https://github.com/alsa-project/alsa-utils/pull/50 Patch URL : https://github.com/alsa-project/alsa-utils/pull/50.patch Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[PATCH] ALSA: hda/realtek: Fix pin default on Intel NUC 8 Rugged
by Kai-Heng Feng 07 Aug '20

07 Aug '20

The jack on Intel NUC 8 Rugged rear panel doesn't work. The spec [1] states that the jack supports both headphone and microphone, so override a Pin Complex which has both Amp-In and Amp-Out to make the jack work. Node 0x1b fits the requirement, and user confirmed the jack now works with new pin config. [1] https://www.intel.com/content/dam/support/us/en/documents/mini-pcs/NUC8CCH_… BugLink: https://bugs.launchpad.net/bugs/1875199 Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 2477f3ed7237..daedcc0adc21 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6159,6 +6159,7 @@ enum { ALC269_FIXUP_CZC_L101, ALC269_FIXUP_LEMOTE_A1802, ALC269_FIXUP_LEMOTE_A190X, + ALC256_FIXUP_INTEL_NUC8_RUGGED, }; static const struct hda_fixup alc269_fixups[] = { @@ -7480,6 +7481,15 @@ static const struct hda_fixup alc269_fixups[] = { }, .chain_id = ALC269_FIXUP_DMIC, }, + [ALC256_FIXUP_INTEL_NUC8_RUGGED] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x1b, 0x01a1913c }, /* use as headset mic, without its own jack detect */ + { } + }, + .chained = true, + .chain_id = ALC269_FIXUP_HEADSET_MODE + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -7777,6 +7787,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x10ec, 0x118c, "Medion EE4254 MD62100", ALC256_FIXUP_MEDION_HEADSET_NO_PRESENCE), SND_PCI_QUIRK(0x1c06, 0x2013, "Lemote A1802", ALC269_FIXUP_LEMOTE_A1802), SND_PCI_QUIRK(0x1c06, 0x2015, "Lemote A190X", ALC269_FIXUP_LEMOTE_A190X), + SND_PCI_QUIRK(0x8086, 0x2080, "Intel NUC 8 Rugged", ALC256_FIXUP_INTEL_NUC8_RUGGED), #if 0 /* Below is a quirk table taken from the old code. -- 2.17.1

2 1

aplay: change if statement to else if in xrun function
by GitHub pull_request - opened 07 Aug '20

07 Aug '20

alsa-project/alsa-utils pull request #50 was opened from JSFamily: Request URL : https://github.com/alsa-project/alsa-utils/pull/50 Patch URL : https://github.com/alsa-project/alsa-utils/pull/50.patch Repository URL: https://github.com/alsa-project/alsa-utils

1 0

Re: [PATCH] ALSA: usb-audio: Fix memory leak in scarlett2_add_new_ctl()
by Markus Elfring 07 Aug '20

07 Aug '20

> When snd_usb_mixer_add_control() fails, elem needs to be > freed just like when snd_ctl_new1() fails. However, current > code is returning directly and ends up leaking memory. Can an imperative wording be preferred for the change description? … > +++ b/sound/usb/mixer_scarlett_gen2.c > @@ -964,8 +964,10 @@ static int scarlett2_add_new_ctl(struct usb_mixer_interface *mixer, > strlcpy(kctl->id.name, name, sizeof(kctl->id.name)); > > err = snd_usb_mixer_add_control(&elem->head, kctl); > - if (err < 0) > + if (err < 0) { > + kfree(elem); > return err; > + } … I propose to add a jump target so that a bit of common exception handling code can be better reused in this function implementation. Regards, Markus

1 0

[PATCH] ASoC: fsl-asoc-card: Get "extal" clock rate by clk_get_rate
by Shengjiu Wang 07 Aug '20

07 Aug '20

On some platform(.e.g. i.MX8QM MEK), the "extal" clock is different with the mclk of codec, then the clock rate is also different. So it is better to get clock rate of "extal" rate by clk_get_rate, don't reuse the clock rate of mclk. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/fsl-asoc-card.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index 52adedc03245..2c92a5efad61 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -696,6 +696,13 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) goto asrc_fail; } } else if (of_node_name_eq(cpu_np, "esai")) { + struct clk *esai_clk = clk_get(&cpu_pdev->dev, "extal"); + + if (!IS_ERR(esai_clk)) { + priv->cpu_priv.sysclk_freq[TX] = clk_get_rate(esai_clk); + priv->cpu_priv.sysclk_freq[RX] = clk_get_rate(esai_clk); + clk_put(esai_clk); + } priv->cpu_priv.sysclk_id[1] = ESAI_HCKT_EXTAL; priv->cpu_priv.sysclk_id[0] = ESAI_HCKR_EXTAL; } else if (of_node_name_eq(cpu_np, "sai")) { -- 2.27.0

3 2

[PATCH v2 27/41] ASoC: samsung: s3c2412-i2s: avoid hardcoded S3C2410_PA_IIS
by Krzysztof Kozlowski 06 Aug '20

06 Aug '20

From: Arnd Bergmann <arnd(a)arndb.de> The constant requires indirectly including a machine header file, but it's not actually used any more since commit 87b132bc0315 ("ASoC: samsung: s3c24{xx,12}-i2s: port to use generic dmaengine API"), so remove it completely. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> Acked-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Krzysztof Kozlowski <krzk(a)kernel.org> --- sound/soc/samsung/s3c-i2s-v2.c | 3 +-- sound/soc/samsung/s3c-i2s-v2.h | 3 +-- sound/soc/samsung/s3c2412-i2s.c | 2 +- 3 files changed, 3 insertions(+), 5 deletions(-) diff --git a/sound/soc/samsung/s3c-i2s-v2.c b/sound/soc/samsung/s3c-i2s-v2.c index ed21786104a1..e9481187a08c 100644 --- a/sound/soc/samsung/s3c-i2s-v2.c +++ b/sound/soc/samsung/s3c-i2s-v2.c @@ -616,8 +616,7 @@ int s3c_i2sv2_iis_calc_rate(struct s3c_i2sv2_rate_calc *info, EXPORT_SYMBOL_GPL(s3c_i2sv2_iis_calc_rate); int s3c_i2sv2_probe(struct snd_soc_dai *dai, - struct s3c_i2sv2_info *i2s, - unsigned long base) + struct s3c_i2sv2_info *i2s) { struct device *dev = dai->dev; unsigned int iismod; diff --git a/sound/soc/samsung/s3c-i2s-v2.h b/sound/soc/samsung/s3c-i2s-v2.h index fe42b77999fd..8c6fc0d3d77e 100644 --- a/sound/soc/samsung/s3c-i2s-v2.h +++ b/sound/soc/samsung/s3c-i2s-v2.h @@ -83,8 +83,7 @@ extern int s3c_i2sv2_iis_calc_rate(struct s3c_i2sv2_rate_calc *info, * @base: The base address for the registers. */ extern int s3c_i2sv2_probe(struct snd_soc_dai *dai, - struct s3c_i2sv2_info *i2s, - unsigned long base); + struct s3c_i2sv2_info *i2s); /** * s3c_i2sv2_cleanup - cleanup resources allocated in s3c_i2sv2_probe diff --git a/sound/soc/samsung/s3c2412-i2s.c b/sound/soc/samsung/s3c2412-i2s.c index b35d828c1cfe..edfa9d11d2e5 100644 --- a/sound/soc/samsung/s3c2412-i2s.c +++ b/sound/soc/samsung/s3c2412-i2s.c @@ -49,7 +49,7 @@ static int s3c2412_i2s_probe(struct snd_soc_dai *dai) snd_soc_dai_init_dma_data(dai, &s3c2412_i2s_pcm_stereo_out, &s3c2412_i2s_pcm_stereo_in); - ret = s3c_i2sv2_probe(dai, &s3c2412_i2s, S3C2410_PA_IIS); + ret = s3c_i2sv2_probe(dai, &s3c2412_i2s); if (ret) return ret; -- 2.17.1

1 0

[PATCH v2 26/41] ASoC: samsung: rx1950: turn into platform driver
by Krzysztof Kozlowski 06 Aug '20

06 Aug '20

From: Arnd Bergmann <arnd(a)arndb.de> Avoid machine specific headers by using a gpio lookup table combined with a platform_driver for this board. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> Signed-off-by: Krzysztof Kozlowski <krzk(a)kernel.org> --- arch/arm/mach-s3c24xx/mach-rx1950.c | 16 +++++++ sound/soc/samsung/rx1950_uda1380.c | 72 +++++++++-------------------- 2 files changed, 39 insertions(+), 49 deletions(-) diff --git a/arch/arm/mach-s3c24xx/mach-rx1950.c b/arch/arm/mach-s3c24xx/mach-rx1950.c index 65440c39500e..a5461b2227bd 100644 --- a/arch/arm/mach-s3c24xx/mach-rx1950.c +++ b/arch/arm/mach-s3c24xx/mach-rx1950.c @@ -727,6 +727,20 @@ static struct i2c_board_info rx1950_i2c_devices[] = { }, }; +static struct gpiod_lookup_table rx1950_audio_gpio_table = { + .dev_id = "rx1950-audio", + .table = { + GPIO_LOOKUP("GPIOG", 12, "hp-gpio", GPIO_ACTIVE_HIGH), + GPIO_LOOKUP("GPIOA", 1, "speaker-power", GPIO_ACTIVE_HIGH), + { }, + }, +}; + +static struct platform_device rx1950_audio = { + .name = "rx1950-audio", + .id = -1, +}; + static struct platform_device *rx1950_devices[] __initdata = { &s3c2410_device_dclk, &s3c_device_lcd, @@ -745,6 +759,7 @@ static struct platform_device *rx1950_devices[] __initdata = { &power_supply, &rx1950_battery, &rx1950_leds, + &rx1950_audio, }; static void __init rx1950_map_io(void) @@ -812,6 +827,7 @@ static void __init rx1950_init_machine(void) gpio_direction_output(S3C2410_GPJ(6), 0); pwm_add_table(rx1950_pwm_lookup, ARRAY_SIZE(rx1950_pwm_lookup)); + gpiod_add_lookup_table(&rx1950_audio_gpio_table); platform_add_devices(rx1950_devices, ARRAY_SIZE(rx1950_devices)); i2c_register_board_info(0, rx1950_i2c_devices, diff --git a/sound/soc/samsung/rx1950_uda1380.c b/sound/soc/samsung/rx1950_uda1380.c index 08f7c82aedb6..400a7f77c711 100644 --- a/sound/soc/samsung/rx1950_uda1380.c +++ b/sound/soc/samsung/rx1950_uda1380.c @@ -12,16 +12,13 @@ // Vasily Khoruzhick <anarsoul(a)gmail.com> #include <linux/types.h> -#include <linux/gpio.h> +#include <linux/gpio/consumer.h> #include <linux/module.h> #include <sound/soc.h> #include <sound/jack.h> -#include <mach/gpio-samsung.h> #include "regs-iis.h" -#include <asm/mach-types.h> - #include "s3c24xx-i2s.h" static int rx1950_uda1380_init(struct snd_soc_pcm_runtime *rtd); @@ -58,7 +55,6 @@ static struct snd_soc_jack_pin hp_jack_pins[] = { static struct snd_soc_jack_gpio hp_jack_gpios[] = { [0] = { - .gpio = S3C2410_GPG(12), .name = "hp-gpio", .report = SND_JACK_HEADPHONE, .invert = 1, @@ -123,8 +119,6 @@ static struct snd_soc_card rx1950_asoc = { .num_dapm_routes = ARRAY_SIZE(audio_map), }; -static struct platform_device *s3c24xx_snd_device; - static int rx1950_startup(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; @@ -134,13 +128,15 @@ static int rx1950_startup(struct snd_pcm_substream *substream) &hw_rates); } +struct gpio_desc *gpiod_speaker_power; + static int rx1950_spk_power(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { if (SND_SOC_DAPM_EVENT_ON(event)) - gpio_set_value(S3C2410_GPA(1), 1); + gpiod_set_value(gpiod_speaker_power, 1); else - gpio_set_value(S3C2410_GPA(1), 0); + gpiod_set_value(gpiod_speaker_power, 0); return 0; } @@ -214,57 +210,35 @@ static int rx1950_uda1380_init(struct snd_soc_pcm_runtime *rtd) return 0; } -static int __init rx1950_init(void) +static int rx1950_probe(struct platform_device *pdev) { - int ret; - - if (!machine_is_rx1950()) - return -ENODEV; + struct device *dev = &pdev->dev; /* configure some gpios */ - ret = gpio_request(S3C2410_GPA(1), "speaker-power"); - if (ret) - goto err_gpio; - - ret = gpio_direction_output(S3C2410_GPA(1), 0); - if (ret) - goto err_gpio_conf; - - s3c24xx_snd_device = platform_device_alloc("soc-audio", -1); - if (!s3c24xx_snd_device) { - ret = -ENOMEM; - goto err_plat_alloc; - } - - platform_set_drvdata(s3c24xx_snd_device, &rx1950_asoc); - ret = platform_device_add(s3c24xx_snd_device); - - if (ret) { - platform_device_put(s3c24xx_snd_device); - goto err_plat_add; + gpiod_speaker_power = devm_gpiod_get(dev, "speaker-power", GPIOD_OUT_LOW); + if (IS_ERR(gpiod_speaker_power)) { + dev_err(dev, "cannot get gpio\n"); + return PTR_ERR(gpiod_speaker_power); } - return 0; - -err_plat_add: -err_plat_alloc: -err_gpio_conf: - gpio_free(S3C2410_GPA(1)); + hp_jack_gpios[0].gpiod_dev = dev; + rx1950_asoc.dev = dev; -err_gpio: - return ret; + return devm_snd_soc_register_card(dev, &rx1950_asoc); } -static void __exit rx1950_exit(void) -{ - platform_device_unregister(s3c24xx_snd_device); - gpio_free(S3C2410_GPA(1)); -} +struct platform_driver rx1950_audio = { + .driver = { + .name = "rx1950-audio", + .pm = &snd_soc_pm_ops, + }, + .probe = rx1950_probe, +}; -module_init(rx1950_init); -module_exit(rx1950_exit); +module_platform_driver(rx1950_audio); /* Module information */ MODULE_AUTHOR("Vasily Khoruzhick"); MODULE_DESCRIPTION("ALSA SoC RX1950"); MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:rx1950-audio"); -- 2.17.1

1 0

[PATCH v2 25/41] ASoC: samsung: neo1973: turn into platform driver
by Krzysztof Kozlowski 06 Aug '20

06 Aug '20

From: Arnd Bergmann <arnd(a)arndb.de> Avoid machine specific headers by using a gpio lookup table combined with a platform_driver for this board. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> Signed-off-by: Krzysztof Kozlowski <krzk(a)kernel.org> --- arch/arm/mach-s3c24xx/mach-gta02.c | 17 ++++++ sound/soc/samsung/neo1973_wm8753.c | 85 +++++++++--------------------- 2 files changed, 43 insertions(+), 59 deletions(-) diff --git a/arch/arm/mach-s3c24xx/mach-gta02.c b/arch/arm/mach-s3c24xx/mach-gta02.c index 346a158f9960..46a80fae7056 100644 --- a/arch/arm/mach-s3c24xx/mach-gta02.c +++ b/arch/arm/mach-s3c24xx/mach-gta02.c @@ -15,6 +15,7 @@ #include <linux/delay.h> #include <linux/timer.h> #include <linux/init.h> +#include <linux/gpio/machine.h> #include <linux/gpio.h> #include <linux/gpio_keys.h> #include <linux/workqueue.h> @@ -472,6 +473,20 @@ static struct platform_device gta02_buttons_device = { }, }; +static struct gpiod_lookup_table gta02_audio_gpio_table = { + .dev_id = "neo1973-audio", + .table = { + GPIO_LOOKUP("GPIOJ", 2, "amp-shut", GPIO_ACTIVE_HIGH), + GPIO_LOOKUP("GPIOJ", 1, "hp", GPIO_ACTIVE_HIGH), + { }, + }, +}; + +static struct platform_device gta02_audio = { + .name = "neo1973-audio", + .id = -1, +}; + static void __init gta02_map_io(void) { s3c24xx_init_io(gta02_iodesc, ARRAY_SIZE(gta02_iodesc)); @@ -496,6 +511,7 @@ static struct platform_device *gta02_devices[] __initdata = { &gta02_buttons_device, &s3c_device_adc, &s3c_device_ts, + &gta02_audio, }; static void gta02_poweroff(void) @@ -522,6 +538,7 @@ static void __init gta02_machine_init(void) i2c_register_board_info(0, gta02_i2c_devs, ARRAY_SIZE(gta02_i2c_devs)); + gpiod_add_lookup_table(&gta02_audio_gpio_table); platform_add_devices(gta02_devices, ARRAY_SIZE(gta02_devices)); pm_power_off = gta02_poweroff; diff --git a/sound/soc/samsung/neo1973_wm8753.c b/sound/soc/samsung/neo1973_wm8753.c index 54317e0f68f8..9266070e0181 100644 --- a/sound/soc/samsung/neo1973_wm8753.c +++ b/sound/soc/samsung/neo1973_wm8753.c @@ -11,14 +11,11 @@ #include <linux/module.h> #include <linux/platform_device.h> -#include <linux/gpio.h> +#include <linux/gpio/consumer.h> #include <sound/soc.h> -#include <mach/gpio-samsung.h> -#include <asm/mach-types.h> #include "regs-iis.h" - #include "../codecs/wm8753.h" #include "s3c24xx-i2s.h" @@ -166,6 +163,7 @@ static struct snd_soc_ops neo1973_voice_ops = { .hw_free = neo1973_voice_hw_free, }; +static struct gpio_desc *gpiod_hp_in, *gpiod_amp_shut; static int gta02_speaker_enabled; static int lm4853_set_spk(struct snd_kcontrol *kcontrol, @@ -173,7 +171,7 @@ static int lm4853_set_spk(struct snd_kcontrol *kcontrol, { gta02_speaker_enabled = ucontrol->value.integer.value[0]; - gpio_set_value(S3C2410_GPJ(2), !gta02_speaker_enabled); + gpiod_set_value(gpiod_hp_in, !gta02_speaker_enabled); return 0; } @@ -188,7 +186,7 @@ static int lm4853_get_spk(struct snd_kcontrol *kcontrol, static int lm4853_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *k, int event) { - gpio_set_value(S3C2410_GPJ(1), SND_SOC_DAPM_EVENT_OFF(event)); + gpiod_set_value(gpiod_amp_shut, SND_SOC_DAPM_EVENT_OFF(event)); return 0; } @@ -308,13 +306,8 @@ static struct snd_soc_codec_conf neo1973_codec_conf[] = { }, }; -static const struct gpio neo1973_gta02_gpios[] = { - { S3C2410_GPJ(2), GPIOF_OUT_INIT_HIGH, "GTA02_HP_IN" }, - { S3C2410_GPJ(1), GPIOF_OUT_INIT_HIGH, "GTA02_AMP_SHUT" }, -}; - static struct snd_soc_card neo1973 = { - .name = "neo1973", + .name = "neo1973gta02", .owner = THIS_MODULE, .dai_link = neo1973_dai, .num_links = ARRAY_SIZE(neo1973_dai), @@ -332,62 +325,36 @@ static struct snd_soc_card neo1973 = { .fully_routed = true, }; -static struct platform_device *neo1973_snd_device; - -static int __init neo1973_init(void) +static int neo1973_probe(struct platform_device *pdev) { - int ret; - - if (!machine_is_neo1973_gta02()) - return -ENODEV; + struct device *dev = &pdev->dev; - if (machine_is_neo1973_gta02()) { - neo1973.name = "neo1973gta02"; - neo1973.num_aux_devs = 1; - - ret = gpio_request_array(neo1973_gta02_gpios, - ARRAY_SIZE(neo1973_gta02_gpios)); - if (ret) - return ret; + gpiod_hp_in = devm_gpiod_get(dev, "hp", GPIOD_OUT_HIGH); + if (IS_ERR(gpiod_hp_in)) { + dev_err(dev, "missing gpio %s\n", "hp"); + return PTR_ERR(gpiod_hp_in); } - - neo1973_snd_device = platform_device_alloc("soc-audio", -1); - if (!neo1973_snd_device) { - ret = -ENOMEM; - goto err_gpio_free; + gpiod_amp_shut = devm_gpiod_get(dev, "amp-shut", GPIOD_OUT_HIGH); + if (IS_ERR(gpiod_amp_shut)) { + dev_err(dev, "missing gpio %s\n", "amp-shut"); + return PTR_ERR(gpiod_amp_shut); } - platform_set_drvdata(neo1973_snd_device, &neo1973); - ret = platform_device_add(neo1973_snd_device); - - if (ret) - goto err_put_device; - - return 0; - -err_put_device: - platform_device_put(neo1973_snd_device); -err_gpio_free: - if (machine_is_neo1973_gta02()) { - gpio_free_array(neo1973_gta02_gpios, - ARRAY_SIZE(neo1973_gta02_gpios)); - } - return ret; + neo1973.dev = dev; + return devm_snd_soc_register_card(dev, &neo1973); } -module_init(neo1973_init); - -static void __exit neo1973_exit(void) -{ - platform_device_unregister(neo1973_snd_device); - if (machine_is_neo1973_gta02()) { - gpio_free_array(neo1973_gta02_gpios, - ARRAY_SIZE(neo1973_gta02_gpios)); - } -} -module_exit(neo1973_exit); +struct platform_driver neo1973_audio = { + .driver = { + .name = "neo1973-audio", + .pm = &snd_soc_pm_ops, + }, + .probe = neo1973_probe, +}; +module_platform_driver(neo1973_audio); /* Module information */ MODULE_AUTHOR("Graeme Gregory, graeme(a)openmoko.org, www.openmoko.org"); MODULE_DESCRIPTION("ALSA SoC WM8753 Neo1973 and Frerunner"); MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:neo1973-audio"); -- 2.17.1

1 0

[PATCH v2 24/41] ASoC: samsung: h1940: turn into platform driver
by Krzysztof Kozlowski 06 Aug '20

06 Aug '20

From: Arnd Bergmann <arnd(a)arndb.de> Avoid machine specific headers by using a gpio lookup table combined with a platform_driver for this board. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> Signed-off-by: Krzysztof Kozlowski <krzk(a)kernel.org> --- arch/arm/mach-s3c24xx/mach-h1940.c | 18 ++++++++ sound/soc/samsung/h1940_uda1380.c | 71 ++++++++++-------------------- 2 files changed, 41 insertions(+), 48 deletions(-) diff --git a/arch/arm/mach-s3c24xx/mach-h1940.c b/arch/arm/mach-s3c24xx/mach-h1940.c index 9c71472c939f..c7269cd26ba8 100644 --- a/arch/arm/mach-s3c24xx/mach-h1940.c +++ b/arch/arm/mach-s3c24xx/mach-h1940.c @@ -474,6 +474,22 @@ static struct gpiod_lookup_table h1940_mmc_gpio_table = { }, }; +static struct gpiod_lookup_table h1940_audio_gpio_table = { + .dev_id = "h1940-audio", + .table = { + GPIO_LOOKUP("H1940_LATCH", + H1940_LATCH_AUDIO_POWER - H1940_LATCH_GPIO(0), + "speaker-power", GPIO_ACTIVE_HIGH), + GPIO_LOOKUP("GPIOG", 4, "hp", GPIO_ACTIVE_HIGH), + { }, + }, +}; + +static struct platform_device h1940_audio = { + .name = "h1940-audio", + .id = -1, +}; + static struct pwm_lookup h1940_pwm_lookup[] = { PWM_LOOKUP("samsung-pwm", 0, "pwm-backlight", NULL, 36296, PWM_POLARITY_NORMAL), @@ -650,6 +666,7 @@ static struct platform_device *h1940_devices[] __initdata = { &s3c_device_ts, &power_supply, &h1940_battery, + &h1940_audio, }; static void __init h1940_map_io(void) @@ -689,6 +706,7 @@ static void __init h1940_init(void) s3c24xx_fb_set_platdata(&h1940_fb_info); gpiod_add_lookup_table(&h1940_mmc_gpio_table); + gpiod_add_lookup_table(&h1940_audio_gpio_table); s3c24xx_mci_set_platdata(&h1940_mmc_cfg); s3c24xx_udc_set_platdata(&h1940_udc_cfg); s3c24xx_ts_set_platdata(&h1940_ts_cfg); diff --git a/sound/soc/samsung/h1940_uda1380.c b/sound/soc/samsung/h1940_uda1380.c index b8f0057a0510..8aa78ff640f5 100644 --- a/sound/soc/samsung/h1940_uda1380.c +++ b/sound/soc/samsung/h1940_uda1380.c @@ -15,9 +15,6 @@ #include <sound/jack.h> #include "regs-iis.h" -#include <asm/mach-types.h> - -#include <mach/gpio-samsung.h> #include "s3c24xx-i2s.h" static const unsigned int rates[] = { @@ -31,6 +28,8 @@ static const struct snd_pcm_hw_constraint_list hw_rates = { .list = rates, }; +static struct gpio_desc *gpiod_speaker_power; + static struct snd_soc_jack hp_jack; static struct snd_soc_jack_pin hp_jack_pins[] = { @@ -47,7 +46,6 @@ static struct snd_soc_jack_pin hp_jack_pins[] = { static struct snd_soc_jack_gpio hp_jack_gpios[] = { { - .gpio = S3C2410_GPG(4), .name = "hp-gpio", .report = SND_JACK_HEADPHONE, .invert = 1, @@ -123,9 +121,9 @@ static int h1940_spk_power(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { if (SND_SOC_DAPM_EVENT_ON(event)) - gpio_set_value(S3C_GPIO_END + 9, 1); + gpiod_set_value(gpiod_speaker_power, 1); else - gpio_set_value(S3C_GPIO_END + 9, 0); + gpiod_set_value(gpiod_speaker_power, 0); return 0; } @@ -151,8 +149,6 @@ static const struct snd_soc_dapm_route audio_map[] = { {"VINM", NULL, "Mic Jack"}, }; -static struct platform_device *s3c24xx_snd_device; - static int h1940_uda1380_init(struct snd_soc_pcm_runtime *rtd) { snd_soc_card_jack_new(rtd->card, "Headphone Jack", SND_JACK_HEADPHONE, @@ -194,55 +190,34 @@ static struct snd_soc_card h1940_asoc = { .num_dapm_routes = ARRAY_SIZE(audio_map), }; -static int __init h1940_init(void) +static int h1940_probe(struct platform_device *pdev) { - int ret; + struct device *dev = &pdev->dev; - if (!machine_is_h1940()) - return -ENODEV; + h1940_asoc.dev = dev; + hp_jack_gpios[0].gpiod_dev = dev; + gpiod_speaker_power = devm_gpiod_get(&pdev->dev, "speaker-power", + GPIOD_OUT_LOW); - /* configure some gpios */ - ret = gpio_request(S3C_GPIO_END + 9, "speaker-power"); - if (ret) - goto err_out; - - ret = gpio_direction_output(S3C_GPIO_END + 9, 0); - if (ret) - goto err_gpio; - - s3c24xx_snd_device = platform_device_alloc("soc-audio", -1); - if (!s3c24xx_snd_device) { - ret = -ENOMEM; - goto err_gpio; + if (IS_ERR(gpiod_speaker_power)) { + dev_err(dev, "Could not get gpio\n"); + return PTR_ERR(gpiod_speaker_power); } - platform_set_drvdata(s3c24xx_snd_device, &h1940_asoc); - ret = platform_device_add(s3c24xx_snd_device); - - if (ret) - goto err_plat; - - return 0; - -err_plat: - platform_device_put(s3c24xx_snd_device); -err_gpio: - gpio_free(S3C_GPIO_END + 9); - -err_out: - return ret; -} - -static void __exit h1940_exit(void) -{ - platform_device_unregister(s3c24xx_snd_device); - gpio_free(S3C_GPIO_END + 9); + return devm_snd_soc_register_card(dev, &h1940_asoc); } -module_init(h1940_init); -module_exit(h1940_exit); +static struct platform_driver h1940_audio_driver = { + .driver = { + .name = "h1940-audio", + .pm = &snd_soc_pm_ops, + }, + .probe = h1940_probe, +}; +module_platform_driver(h1940_audio_driver); /* Module information */ MODULE_AUTHOR("Arnaud Patard, Vasily Khoruzhick"); MODULE_DESCRIPTION("ALSA SoC H1940"); MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:h1940-audio"); -- 2.17.1

1 0

[PATCH] ALSA: usb - Creative USB X-Fi Pro SB1095 volume knob support
by buzz＠l4m1.de 06 Aug '20

06 Aug '20

From: Mirko Dietrich <buzz(a)l4m1.de> Adds an entry for Creative USB X-Fi to the rc_config array in mixer_quirks.c to allow use of volume knob on the device. Adds support for newer X-Fi Pro card, known as "Model No. SB1095" with USB ID "041e:3263" Signed-off-by: Mirko Dietrich <buzz(a)l4m1.de> --- sound/usb/mixer_quirks.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index cec1cfd7edb7..199cdbfdc761 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -185,6 +185,7 @@ static const struct rc_config { { USB_ID(0x041e, 0x3042), 0, 1, 1, 1, 1, 0x000d }, /* Usb X-Fi S51 */ { USB_ID(0x041e, 0x30df), 0, 1, 1, 1, 1, 0x000d }, /* Usb X-Fi S51 Pro */ { USB_ID(0x041e, 0x3237), 0, 1, 1, 1, 1, 0x000d }, /* Usb X-Fi S51 Pro */ + { USB_ID(0x041e, 0x3263), 0, 1, 1, 1, 1, 0x000d }, /* Usb X-Fi S51 Pro */ { USB_ID(0x041e, 0x3048), 2, 2, 6, 6, 2, 0x6e91 }, /* Toshiba SB0500 */ }; -- 2.28.0

2 1

[RFC PATCH v2 0/6] ALSA: compress: add support to change codec profile in gapless playback
by Srinivas Kandagatla 06 Aug '20

06 Aug '20

For gapless playback it is possible that each track can have different codec profile with same decoder, for example we have WMA album, we may have different tracks as WMA v9, WMA v10 and so on Or if DSP's like QDSP have abililty to switch decoders on single stream for each track, then this call could be used to set new codec parameters. Existing code does not allow to change this profile while doing gapless playback. This patchset adds new SNDRV_COMPRESS_SET_CODEC_PARAMS IOCTL along with flags in capablity structure to allow userspace to set this new parameters required which switching codec profile, either for gapless or cross fade usecase. thanks, srini Changes since v1: - split patch into smaller chuncks, - bump up the version - added flags in compress capablity structure - added user for this new functionality. - add this new call in Documentation. Srinivas Kandagatla (6): ALSA: compress: move codec parameter check to a function ALSA: compress: add new ioctl for setting codec parameters ALSA: compress: add flags to snd_compr_caps to expose dsp caps ASoC: compress: add snd_soc_dai_compr_set_codec_params() ALSA: compress: bump the version ASoC: q6asm-dai: add support to set_codec_params .../sound/designs/compress-offload.rst | 6 ++ include/sound/compress_driver.h | 5 ++ include/sound/soc-component.h | 3 + include/sound/soc-dai.h | 5 ++ include/uapi/sound/compress_offload.h | 10 ++- sound/core/compress_offload.c | 72 +++++++++++++++++-- sound/soc/qcom/qdsp6/q6asm-dai.c | 33 +++++++++ sound/soc/soc-compress.c | 30 ++++++++ sound/soc/soc-dai.c | 14 ++++ 9 files changed, 170 insertions(+), 8 deletions(-) -- 2.21.0

5 22

[PATCH] ALSA: usb: fix spelling mistake "buss" -> "bus"
by Colin King 06 Aug '20

06 Aug '20

From: Colin Ian King <colin.king(a)canonical.com> There is a spelling mistake in a usb_audio_dbg debug message. Also replace "param" with "parameter". Fix these. Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/usb/mixer_us16x08.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/mixer_us16x08.c b/sound/usb/mixer_us16x08.c index 986145fd2ce0..a4d4d71db55b 100644 --- a/sound/usb/mixer_us16x08.c +++ b/sound/usb/mixer_us16x08.c @@ -329,7 +329,7 @@ static int snd_us16x08_bus_put(struct snd_kcontrol *kcontrol, elem->cached |= 1; elem->cache_val[0] = val; } else { - usb_audio_dbg(chip, "Failed to set buss param, err:%d\n", err); + usb_audio_dbg(chip, "Failed to set bus parameter, err:%d\n", err); } return err > 0 ? 1 : 0; -- 2.27.0

2 1

asoc : snd_soc_simple_card connection to AKM4556 codec (i2s) issue on arm64
by Markus Bollinger 06 Aug '20

06 Aug '20

Hello, I try to get working an AKM4556 codec on the arm64 Rockchip RK3399 board under Linux 4.4.179 The codec is a simple one (without i2c connection), similar to AKM4554 (but 24 bit instead of 16 bit audio) and hard configured to use i2s. I implemented the driver soc/codecs/ak4556.c which has only few differences to ak4554.c. But when I try to get it working with the soc/general/simple-card.c, the connection to the codec fails Both drivers (snd_soc_simple_card and snd_soc_ak4556) get loaded, the call to asoc_simple_card_dai_link_of -> asoc_simple_card_sub_parse_of(cpu, ..) is OK, but the call to asoc_simple_card_dai_link_of -> asoc_simple_card_sub_parse_of(codec, ..) -> snd_soc_of_get_dai_name fails with -EPROBE_DEFER. Now, i am not sure if it is a driver issue or rather a device-tree one. I declared dt like this : / { ak4556_card: ak4556-adc-dac { status = "okay"; compatible = "simple-audio-card"; simple-audio-card,name = "ak4556"; simple-audio-card,format = "i2s"; simple-audio-card,mclk-fs = <256>; simple-audio-card,widgets = "Line", "Line In", "Line", "Line Out"; simple-audio-card,routing = "LIN", "Line In", "RIN", "Line In", "Line Out", "LOUT", "Line Out", "ROUT"; simple-audio-card,cpu { sound-dai = <&i2s0>; }; simple-audio-card,codec { sound-dai = <&ak4556_codec>; }; }; ak4556_codec: ak4556-codec { status = "okay"; compatible = "asahi-kasei,ak4556"; #sound-dai-cells = <0>; }; i2s0: i2s@ff880000 { compatible = "rockchip,rk3399-i2s", "rockchip,rk3066-i2s"; reg = <0x0 0xff880000 0x0 0x1000>; rockchip,grf = <&grf>; interrupts = <GIC_SPI 39 IRQ_TYPE_LEVEL_HIGH 0>; dmas = <&dmac_bus 0>, <&dmac_bus 1>; dma-names = "tx", "rx"; clock-names = "i2s_clk", "i2s_hclk"; clocks = <&cru SCLK_I2S0_8CH>, <&cru HCLK_I2S0_8CH>; resets = <&cru SRST_I2S0_8CH>, <&cru SRST_H_I2S0_8CH>; reset-names = "reset-m", "reset-h"; pinctrl-names = "default"; pinctrl-0 = <&i2s0_8ch_bus>; power-domains = <&power RK3399_PD_SDIOAUDIO>; status = "disabled"; }; &i2s0 { #sound-dai-cells = <0>; assigned-clocks = <&cru SCLK_I2S0_DIV>; assigned-clock-parents = <&cru PLL_GPLL>; status = "okay"; }; Thank you in advance for your suggestions, Markus

1 1

[PATCH v4 0/2] Add documentation and machine driver for SC7180 sound card
by Cheng-Yi Chiang 06 Aug '20

06 Aug '20

Note: - The machine driver patch depends on LPASS patch series so it is not ready to be merged now. ASoC: qcom: Add support for SC7180 lpass variant https://patchwork.kernel.org/cover/11678133/ - The machine driver patch is made by the collaboration of Cheng-Yi Chiang <cychiang(a)chromium.org> Rohit kumar <rohitkr(a)codeaurora.org> Ajit Pandey <ajitp(a)codeaurora.org> But Ajit has left codeaurora. Changes from v1 to v2: - Ducumentation: Addressed all suggestions from Doug. - Machine driver: - Fix comment style for license. - Sort includes. - Remove sc7180_snd_hw_params. - Remove sc7180_dai_init and use aux device instead for headset jack registration. - Statically define format for Primary MI2S. - Atomic is not a concern because there is mutex in card to make sure startup and shutdown happen sequentially. - Fix missing return -EINVAL in startup. - Use static sound card. - Use devm_kzalloc to avoid kfree. Changes from v2 to v3: - Ducumentation: Addressed suggestions from Srini. - Machine driver: - Reuse qcom_snd_parse_of to parse properties. - Remove playback-only and capture-only. - Misc fixes to address comments. Changes from v3 to v4: - Ducumentation: Addressed suggestions from Rob. - Remove definition of dai. - Use 'sound-dai: true' for sound-dai schema. - Add reg property to pass 'make dt_binding_check' check although reg is not used in the driver. - Machine driver: - Add Reviewed-by: Tzung-Bi Shih <tzungbi(a)google.com> Ajit Pandey (1): ASoC: qcom: sc7180: Add machine driver for sound card registration Cheng-Yi Chiang (1): ASoC: qcom: dt-bindings: Add sc7180 machine bindings .../bindings/sound/qcom,sc7180.yaml | 113 ++++++++ sound/soc/qcom/Kconfig | 12 + sound/soc/qcom/Makefile | 2 + sound/soc/qcom/sc7180.c | 244 ++++++++++++++++++ 4 files changed, 371 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/qcom,sc7180.yaml create mode 100644 sound/soc/qcom/sc7180.c -- 2.28.0.163.g6104cc2f0b6-goog

2 3

Microphone selection logic not working for max98090
by GitHub issues - edited 06 Aug '20

06 Aug '20

alsa-project/alsa-ucm-conf issue #44 was edited from jacobopantoja: An incorrect conflicting device is set for internal mic. See PR #43 and commit 73fac5b1 for proposed solution. Adding this issue for traceability purposes Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/44 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Microphone selection logic not working for max98090
by GitHub issues - opened 06 Aug '20

06 Aug '20

alsa-project/alsa-ucm-conf issue #44 was opened from jacobopantoja: An incorrect conflicting device is set for internal mic. See PR #43 and commit #73 and commit 73fac5b1 for proposed solution. Adding this issue for traceability purposes Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/44 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Correct conflicting mic in max98090
by GitHub pull_request - opened 06 Aug '20

06 Aug '20

alsa-project/alsa-ucm-conf pull request #43 was opened from jacobopantoja: 'mic' was conflicting 'mic' instead of 'headset', thus preventing correct configuration regarding microphones. With this corrected, introducing a headset enables 'headset' mic and disables internal 'mic', whereas releasing the headset disables 'headset' mic and enables internal 'mic' Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/43 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/43.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Creative USB X-Fi Pro V3 volume knob and IR remote control
by _ 06 Aug '20

06 Aug '20

Hi, I hope I'm in the right place here. I bought a Creative USB X-Fi Pro (external USB DAC) which plays/records audio just fine out of the box. It features a volume knob and also an IR remote control (RM-820). These controls didn't work. I followed a guide which describes how to get the controls to work on Linux using LIRC alsa_usb driver (https://alsa.opensrc.org/Usb-audio under section "Creative USB X-Fi Surround 5.1"). It turned out the method didn't work for my device (LIRC wouldn't detect the IR device) so I did some digging and found this patch (https://github.com/torvalds/linux/commit/3dc8523fa7412e731441c01fb33f003eb3…) which enables the volume knob for the device. I noticed the USB ID is 041e:3237, while my device has 041e:3263. My device is marketed as "CREATIVE Sound Blaster X-Fi Surround 5.1 Pro v3" and only available since the beginning of 2019. So, I assume it to be another revision of the same hardware. The device model is SB1095. I recompiled my kernel with the attached simple patch and the controls including the IR remote are now detected by LIRC and working. I'd be happy if the change could be picked up if possible. Thank you Mirko --- sound/usb/mixer_quirks.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index cec1cfd7edb7..199cdbfdc761 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -185,6 +185,7 @@ static const struct rc_config { { USB_ID(0x041e, 0x3042), 0, 1, 1, 1, 1, 0x000d }, /* Usb X-Fi S51 */ { USB_ID(0x041e, 0x30df), 0, 1, 1, 1, 1, 0x000d }, /* Usb X-Fi S51 Pro */ { USB_ID(0x041e, 0x3237), 0, 1, 1, 1, 1, 0x000d }, /* Usb X-Fi S51 Pro */ + { USB_ID(0x041e, 0x3263), 0, 1, 1, 1, 1, 0x000d }, /* Usb X-Fi S51 Pro */ { USB_ID(0x041e, 0x3048), 2, 2, 6, 6, 2, 0x6e91 }, /* Toshiba SB0500 */ }; -- 2.28.0

1 0

Add support for Lenovo ThinkStation P620 Main Audio
by GitHub pull_request - opened 06 Aug '20

06 Aug '20

alsa-project/alsa-ucm-conf pull request #42 was opened from khfeng: This add support for internal speaker and front headset. We need two separate configs to let PulseAudio understands they are two different profiles, so the headset's port availability won't affect speaker's profile availability. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/42 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/42.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH] sound: pci: delete repeated words in comments
by Randy Dunlap 06 Aug '20

06 Aug '20

Drop duplicated words in sound/pci/. {and, the, at} Signed-off-by: Randy Dunlap <rdunlap(a)infradead.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: alsa-devel(a)alsa-project.org Cc: Clemens Ladisch <clemens(a)ladisch.de> --- sound/pci/cs46xx/cs46xx_lib.c | 2 +- sound/pci/cs46xx/dsp_spos_scb_lib.c | 2 +- sound/pci/hda/hda_codec.c | 2 +- sound/pci/hda/hda_generic.c | 2 +- sound/pci/hda/patch_sigmatel.c | 2 +- sound/pci/ice1712/prodigy192.c | 2 +- sound/pci/oxygen/xonar_dg.c | 2 +- 7 files changed, 7 insertions(+), 7 deletions(-) --- linux-next-20200805.orig/sound/pci/cs46xx/cs46xx_lib.c +++ linux-next-20200805/sound/pci/cs46xx/cs46xx_lib.c @@ -766,7 +766,7 @@ static void snd_cs46xx_set_capture_sampl rate = 48000 / 9; /* - * We can not capture at at rate greater than the Input Rate (48000). + * We can not capture at a rate greater than the Input Rate (48000). * Return an error if an attempt is made to stray outside that limit. */ if (rate > 48000) --- linux-next-20200805.orig/sound/pci/cs46xx/dsp_spos_scb_lib.c +++ linux-next-20200805/sound/pci/cs46xx/dsp_spos_scb_lib.c @@ -1716,7 +1716,7 @@ int cs46xx_iec958_pre_open (struct snd_c struct dsp_spos_instance * ins = chip->dsp_spos_instance; if ( ins->spdif_status_out & DSP_SPDIF_STATUS_OUTPUT_ENABLED ) { - /* remove AsynchFGTxSCB and and PCMSerialInput_II */ + /* remove AsynchFGTxSCB and PCMSerialInput_II */ cs46xx_dsp_disable_spdif_out (chip); /* save state */ --- linux-next-20200805.orig/sound/pci/hda/hda_codec.c +++ linux-next-20200805/sound/pci/hda/hda_codec.c @@ -3428,7 +3428,7 @@ EXPORT_SYMBOL_GPL(snd_hda_set_power_save * @nid: NID to check / update * * Check whether the given NID is in the amp list. If it's in the list, - * check the current AMP status, and update the the power-status according + * check the current AMP status, and update the power-status according * to the mute status. * * This function is supposed to be set or called from the check_power_status --- linux-next-20200805.orig/sound/pci/hda/hda_generic.c +++ linux-next-20200805/sound/pci/hda/hda_generic.c @@ -813,7 +813,7 @@ static void activate_amp_in(struct hda_c } } -/* sync power of each widget in the the given path */ +/* sync power of each widget in the given path */ static hda_nid_t path_power_update(struct hda_codec *codec, struct nid_path *path, bool allow_powerdown) --- linux-next-20200805.orig/sound/pci/hda/patch_sigmatel.c +++ linux-next-20200805/sound/pci/hda/patch_sigmatel.c @@ -838,7 +838,7 @@ static int stac_auto_create_beep_ctls(st static const struct snd_kcontrol_new beep_vol_ctl = HDA_CODEC_VOLUME(NULL, 0, 0, 0); - /* check for mute support for the the amp */ + /* check for mute support for the amp */ if ((caps & AC_AMPCAP_MUTE) >> AC_AMPCAP_MUTE_SHIFT) { const struct snd_kcontrol_new *temp; if (spec->anabeep_nid == nid) --- linux-next-20200805.orig/sound/pci/ice1712/prodigy192.c +++ linux-next-20200805/sound/pci/ice1712/prodigy192.c @@ -32,7 +32,7 @@ * Experimentally I found out that only a combination of * OCKS0=1, OCKS1=1 (128fs, 64fs output) and ice1724 - * VT1724_MT_I2S_MCLK_128X=0 (256fs input) yields correct - * sampling rate. That means the the FPGA doubles the + * sampling rate. That means that the FPGA doubles the * MCK01 rate. * * Copyright (c) 2003 Takashi Iwai <tiwai(a)suse.de> --- linux-next-20200805.orig/sound/pci/oxygen/xonar_dg.c +++ linux-next-20200805/sound/pci/oxygen/xonar_dg.c @@ -29,7 +29,7 @@ * GPIO 4 <- headphone detect * GPIO 5 -> enable ADC analog circuit for the left channel * GPIO 6 -> enable ADC analog circuit for the right channel - * GPIO 7 -> switch green rear output jack between CS4245 and and the first + * GPIO 7 -> switch green rear output jack between CS4245 and the first * channel of CS4361 (mechanical relay) * GPIO 8 -> enable output to speakers *

2 1

[PATCH] sound: isa: delete repeated words in comments
by Randy Dunlap 06 Aug '20

06 Aug '20

Drop duplicated words in sound/isa/. {be, bit} Signed-off-by: Randy Dunlap <rdunlap(a)infradead.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: alsa-devel(a)alsa-project.org --- sound/isa/cs423x/cs4236_lib.c | 2 +- sound/isa/es18xx.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) --- linux-next-20200805.orig/sound/isa/cs423x/cs4236_lib.c +++ linux-next-20200805/sound/isa/cs423x/cs4236_lib.c @@ -39,7 +39,7 @@ * D7: consumer serial port enable (CS4237B,CS4238B) * D6: channels status block reset (CS4237B,CS4238B) * D5: user bit in sub-frame of digital audio data (CS4237B,CS4238B) - * D4: validity bit bit in sub-frame of digital audio data (CS4237B,CS4238B) + * D4: validity bit in sub-frame of digital audio data (CS4237B,CS4238B) * * C5 lower channel status (digital serial data description) (CS4237B,CS4238B) * D7-D6: first two bits of category code --- linux-next-20200805.orig/sound/isa/es18xx.c +++ linux-next-20200805/sound/isa/es18xx.c @@ -955,7 +955,7 @@ static int snd_es18xx_info_mux(struct sn case 0x1887: case 0x1888: return snd_ctl_enum_info(uinfo, 1, 5, texts5Source); - case 0x1869: /* DS somewhat contradictory for 1869: could be be 5 or 8 */ + case 0x1869: /* DS somewhat contradictory for 1869: could be 5 or 8 */ case 0x1879: return snd_ctl_enum_info(uinfo, 1, 8, texts8Source); default:

2 1

[PATCH] soc: qmi: allow user to set handle wq to hiprio
by 王文虎 05 Aug '20

05 Aug '20

Currently the qmi_handle is initialized single threaded and strictly ordered with the active set to 1. This is pretty simple and safe but sometimes ineffency. So it is better to allow user to decide whether a high priority workqueue should be used. Signed-off-by: Wang Wenhu <wenhu.wang(a)vivo.com> --- drivers/net/ipa/ipa_qmi.c | 4 ++-- drivers/net/wireless/ath/ath10k/qmi.c | 2 +- drivers/net/wireless/ath/ath11k/qmi.c | 2 +- drivers/remoteproc/qcom_sysmon.c | 2 +- drivers/slimbus/qcom-ngd-ctrl.c | 4 ++-- drivers/soc/qcom/pdr_interface.c | 4 ++-- drivers/soc/qcom/qmi_interface.c | 9 +++++++-- include/linux/soc/qcom/qmi.h | 3 ++- samples/qmi/qmi_sample_client.c | 4 ++-- 9 files changed, 20 insertions(+), 14 deletions(-) diff --git a/drivers/net/ipa/ipa_qmi.c b/drivers/net/ipa/ipa_qmi.c index 5090f0f923ad..d78b0fe6bd83 100644 --- a/drivers/net/ipa/ipa_qmi.c +++ b/drivers/net/ipa/ipa_qmi.c @@ -486,7 +486,7 @@ int ipa_qmi_setup(struct ipa *ipa) */ ret = qmi_handle_init(&ipa_qmi->server_handle, IPA_QMI_SERVER_MAX_RCV_SZ, &ipa_server_ops, - ipa_server_msg_handlers); + ipa_server_msg_handlers, 0); if (ret) return ret; @@ -500,7 +500,7 @@ int ipa_qmi_setup(struct ipa *ipa) */ ret = qmi_handle_init(&ipa_qmi->client_handle, IPA_QMI_CLIENT_MAX_RCV_SZ, &ipa_client_ops, - ipa_client_msg_handlers); + ipa_client_msg_handlers, 0); if (ret) goto err_server_handle_release; diff --git a/drivers/net/wireless/ath/ath10k/qmi.c b/drivers/net/wireless/ath/ath10k/qmi.c index 5468a41e928e..02881882b4d9 100644 --- a/drivers/net/wireless/ath/ath10k/qmi.c +++ b/drivers/net/wireless/ath/ath10k/qmi.c @@ -1034,7 +1034,7 @@ int ath10k_qmi_init(struct ath10k *ar, u32 msa_size) ret = qmi_handle_init(&qmi->qmi_hdl, WLFW_BDF_DOWNLOAD_REQ_MSG_V01_MAX_MSG_LEN, - &ath10k_qmi_ops, qmi_msg_handler); + &ath10k_qmi_ops, qmi_msg_handler, 0); if (ret) goto err; diff --git a/drivers/net/wireless/ath/ath11k/qmi.c b/drivers/net/wireless/ath/ath11k/qmi.c index c00a99ad8dbc..91394d58d36e 100644 --- a/drivers/net/wireless/ath/ath11k/qmi.c +++ b/drivers/net/wireless/ath/ath11k/qmi.c @@ -2397,7 +2397,7 @@ int ath11k_qmi_init_service(struct ath11k_base *ab) ab->qmi.target_mem_mode = ATH11K_QMI_TARGET_MEM_MODE_DEFAULT; ret = qmi_handle_init(&ab->qmi.handle, ATH11K_QMI_RESP_LEN_MAX, - &ath11k_qmi_ops, ath11k_qmi_msg_handlers); + &ath11k_qmi_ops, ath11k_qmi_msg_handlers, 0); if (ret < 0) { ath11k_warn(ab, "failed to initialize qmi handle\n"); return ret; diff --git a/drivers/remoteproc/qcom_sysmon.c b/drivers/remoteproc/qcom_sysmon.c index 8d8996d714f0..4ec470e424ef 100644 --- a/drivers/remoteproc/qcom_sysmon.c +++ b/drivers/remoteproc/qcom_sysmon.c @@ -614,7 +614,7 @@ struct qcom_sysmon *qcom_add_sysmon_subdev(struct rproc *rproc, } ret = qmi_handle_init(&sysmon->qmi, SSCTL_MAX_MSG_LEN, &ssctl_ops, - qmi_indication_handler); + qmi_indication_handler, 0); if (ret < 0) { dev_err(sysmon->dev, "failed to initialize qmi handle\n"); kfree(sysmon); diff --git a/drivers/slimbus/qcom-ngd-ctrl.c b/drivers/slimbus/qcom-ngd-ctrl.c index 743ee7b4e63f..ba76691fc5a5 100644 --- a/drivers/slimbus/qcom-ngd-ctrl.c +++ b/drivers/slimbus/qcom-ngd-ctrl.c @@ -446,7 +446,7 @@ static int qcom_slim_qmi_init(struct qcom_slim_ngd_ctrl *ctrl, return -ENOMEM; rc = qmi_handle_init(handle, SLIMBUS_QMI_POWER_REQ_MAX_MSG_LEN, - NULL, qcom_slim_qmi_msg_handlers); + NULL, qcom_slim_qmi_msg_handlers, 0); if (rc < 0) { dev_err(ctrl->dev, "QMI client init failed: %d\n", rc); goto qmi_handle_init_failed; @@ -1293,7 +1293,7 @@ static int qcom_slim_ngd_qmi_svc_event_init(struct qcom_slim_ngd_ctrl *ctrl) int ret; ret = qmi_handle_init(&qmi->svc_event_hdl, 0, - &qcom_slim_ngd_qmi_svc_event_ops, NULL); + &qcom_slim_ngd_qmi_svc_event_ops, NULL, 0); if (ret < 0) { dev_err(ctrl->dev, "qmi_handle_init failed: %d\n", ret); return ret; diff --git a/drivers/soc/qcom/pdr_interface.c b/drivers/soc/qcom/pdr_interface.c index bdcf16f88a97..cc1cb90c1968 100644 --- a/drivers/soc/qcom/pdr_interface.c +++ b/drivers/soc/qcom/pdr_interface.c @@ -685,7 +685,7 @@ struct pdr_handle *pdr_handle_alloc(void (*status)(int state, ret = qmi_handle_init(&pdr->locator_hdl, SERVREG_GET_DOMAIN_LIST_RESP_MAX_LEN, - &pdr_locator_ops, NULL); + &pdr_locator_ops, NULL, 0); if (ret < 0) goto destroy_indack; @@ -696,7 +696,7 @@ struct pdr_handle *pdr_handle_alloc(void (*status)(int state, ret = qmi_handle_init(&pdr->notifier_hdl, SERVREG_STATE_UPDATED_IND_MAX_LEN, &pdr_notifier_ops, - qmi_indication_handler); + qmi_indication_handler, 0); if (ret < 0) goto release_qmi_handle; diff --git a/drivers/soc/qcom/qmi_interface.c b/drivers/soc/qcom/qmi_interface.c index 1a03eaa38c46..01160dbfc4d0 100644 --- a/drivers/soc/qcom/qmi_interface.c +++ b/drivers/soc/qcom/qmi_interface.c @@ -609,6 +609,7 @@ static struct socket *qmi_sock_create(struct qmi_handle *qmi, * @recv_buf_size: maximum size of incoming message * @ops: reference to callbacks for QRTR notifications * @handlers: NULL-terminated list of QMI message handlers + * @hiprio: whether high priority worker is used for workqueue * * This initializes the QMI client handle to allow sending and receiving QMI * messages. As messages are received the appropriate handler will be invoked. @@ -617,9 +618,11 @@ static struct socket *qmi_sock_create(struct qmi_handle *qmi, */ int qmi_handle_init(struct qmi_handle *qmi, size_t recv_buf_size, const struct qmi_ops *ops, - const struct qmi_msg_handler *handlers) + const struct qmi_msg_handler *handlers, + unsigned int hiprio) { int ret; + unsigned int flags = WQ_UNBOUND; mutex_init(&qmi->txn_lock); mutex_init(&qmi->sock_lock); @@ -647,7 +650,9 @@ int qmi_handle_init(struct qmi_handle *qmi, size_t recv_buf_size, if (!qmi->recv_buf) return -ENOMEM; - qmi->wq = alloc_workqueue("qmi_msg_handler", WQ_UNBOUND, 1); + if (hiprio) + flags |= WQ_HIGHPRI; + qmi->wq = alloc_workqueue("qmi_msg_handler", flags, 1); if (!qmi->wq) { ret = -ENOMEM; goto err_free_recv_buf; diff --git a/include/linux/soc/qcom/qmi.h b/include/linux/soc/qcom/qmi.h index e712f94b89fc..24062fd7163d 100644 --- a/include/linux/soc/qcom/qmi.h +++ b/include/linux/soc/qcom/qmi.h @@ -244,7 +244,8 @@ int qmi_add_server(struct qmi_handle *qmi, unsigned int service, int qmi_handle_init(struct qmi_handle *qmi, size_t max_msg_len, const struct qmi_ops *ops, - const struct qmi_msg_handler *handlers); + const struct qmi_msg_handler *handlers, + unsigned int hiprio); void qmi_handle_release(struct qmi_handle *qmi); ssize_t qmi_send_request(struct qmi_handle *qmi, struct sockaddr_qrtr *sq, diff --git a/samples/qmi/qmi_sample_client.c b/samples/qmi/qmi_sample_client.c index c9e7276c3d83..a91d1633ea38 100644 --- a/samples/qmi/qmi_sample_client.c +++ b/samples/qmi/qmi_sample_client.c @@ -463,7 +463,7 @@ static int qmi_sample_probe(struct platform_device *pdev) ret = qmi_handle_init(&sample->qmi, TEST_DATA_REQ_MAX_MSG_LEN_V01, NULL, - qmi_sample_handlers); + qmi_sample_handlers, 0); if (ret < 0) return ret; @@ -590,7 +590,7 @@ static int qmi_sample_init(void) if (ret) goto err_remove_debug_dir; - ret = qmi_handle_init(&lookup_client, 0, &lookup_ops, NULL); + ret = qmi_handle_init(&lookup_client, 0, &lookup_ops, NULL, 0); if (ret < 0) goto err_unregister_driver; -- 2.17.1

3 3

[PATCH v2 0/3] Tegra Specific fixes
by Mohan Kumar 05 Aug '20

05 Aug '20

This series proposes following fixes to Tegra HDA driver. * Align the buffer to 128 bytes for Tegra. * Add new behavior flag dma_stop_delay in hdac_bus structure. * Add 100us as dma stop delay for Tegra. Mohan Kumar (3): ASoC: hda/tegra: Set buffer alignment to 128 bytes ALSA: hda: Add dma stop delay variable ALSA: hda/tegra: Add 100us dma stop delay include/sound/hdaudio.h | 3 +++ sound/hda/hdac_stream.c | 7 +++++++ sound/pci/hda/hda_tegra.c | 3 +++ 3 files changed, 13 insertions(+) -- 2.17.1

2 4

[PATCH 0/3] Tegra Specific fixes
by Mohan Kumar 05 Aug '20

05 Aug '20

This series proposes following fixes to Tegra HDA driver. * Align the buffer to 128 bytes for Tegra. * Add new behavior flag dma_stop_delay in hdac_bus structure. * Add 100us as dma stop delay for Tegra. Mohan Kumar (3): ASoC: hda/tegra: Set buffer alignment to 128 bytes ALSA: hda: Add behaviour flag for dma stop delay ALSA: hda/tegra: Add 100us dma stop delay include/sound/hdaudio.h | 2 ++ sound/hda/hdac_stream.c | 7 +++++++ sound/pci/hda/hda_tegra.c | 3 +++ 3 files changed, 12 insertions(+) -- 2.17.1

2 4

[PATCH v2 0/2] refine and clean code for synchronous mode
by Shengjiu Wang 05 Aug '20

05 Aug '20

refine and clean code for synchronous mode Shengjiu Wang (2): ASoC: fsl_sai: Clean code for synchronous mode ASoC: fsl_sai: Refine enable and disable sequence for synchronous mode changes in v2: - Split the commit - refine the sequence in trigger stop sound/soc/fsl/fsl_sai.c | 133 ++++++++++++++++++++++++++-------------- 1 file changed, 86 insertions(+), 47 deletions(-) -- 2.27.0

3 6

[PATCH] ALSA: seq: oss: Serialize ioctls
by Takashi Iwai 04 Aug '20

04 Aug '20

Some ioctls via OSS sequencer API may race and lead to UAF when the port create and delete are performed concurrently, as spotted by a couple of syzkaller cases. This patch is an attempt to address it by serializing the ioctls with the existing register_mutex. Basically OSS sequencer API is an obsoleted interface and was designed without much consideration of the concurrency. There are very few applications with it, and the concurrent performance isn't asked, hence this "big hammer" approach should be good enough. Reported-by: syzbot+1a54a94bd32716796edd(a)syzkaller.appspotmail.com Reported-by: syzbot+9d2abfef257f3e2d4713(a)syzkaller.appspotmail.com Suggested-by: Hillf Danton <hdanton(a)sina.com> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/seq/oss/seq_oss.c | 8 +++++++- 1 file changed, 7 insertions(+), 1 deletion(-) diff --git a/sound/core/seq/oss/seq_oss.c b/sound/core/seq/oss/seq_oss.c index 17f913657304..c8b9c0b315d8 100644 --- a/sound/core/seq/oss/seq_oss.c +++ b/sound/core/seq/oss/seq_oss.c @@ -168,10 +168,16 @@ static long odev_ioctl(struct file *file, unsigned int cmd, unsigned long arg) { struct seq_oss_devinfo *dp; + long rc; + dp = file->private_data; if (snd_BUG_ON(!dp)) return -ENXIO; - return snd_seq_oss_ioctl(dp, cmd, arg); + + mutex_lock(&register_mutex); + rc = snd_seq_oss_ioctl(dp, cmd, arg); + mutex_unlock(&register_mutex); + return rc; } #ifdef CONFIG_COMPAT -- 2.16.4

1 0

[PATCH v3] ALSA: hda/hdmi: Add quirk to force connectivity
by Kai-Heng Feng 04 Aug '20

04 Aug '20

HDMI on some platforms doesn't enable audio support because its Port Connectivity [31:30] is set to AC_JACK_PORT_NONE: Node 0x05 [Pin Complex] wcaps 0x40778d: 8-Channels Digital Amp-Out CP Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0b000094: OUT Detect HBR HDMI DP Pin Default 0x58560010: [N/A] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0x1, Sequence = 0x0 Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Power states: D0 D3 EPSS Power: setting=D0, actual=D0 Devices: 0 Connection: 3 0x02 0x03* 0x04 For now, use a quirk to force connectivity based on SSID. If there are more platforms affected by the same issue, we can eye for a more generic solution. Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> --- v3: - Move the flag into hdmi_spec. v2: - Use a quirk list instead. sound/pci/hda/patch_hdmi.c | 16 +++++++++++++++- 1 file changed, 15 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index cd46247988e4..b62cd3abb827 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -160,6 +160,7 @@ struct hdmi_spec { bool use_acomp_notifier; /* use eld_notify callback for hotplug */ bool acomp_registered; /* audio component registered in this driver */ + bool force_connect; /* force connectivity */ struct drm_audio_component_audio_ops drm_audio_ops; int (*port2pin)(struct hda_codec *, int); /* reverse port/pin mapping */ @@ -1701,7 +1702,8 @@ static int hdmi_add_pin(struct hda_codec *codec, hda_nid_t pin_nid) * all device entries on the same pin */ config = snd_hda_codec_get_pincfg(codec, pin_nid); - if (get_defcfg_connect(config) == AC_JACK_PORT_NONE) + if (get_defcfg_connect(config) == AC_JACK_PORT_NONE && + !spec->force_connect) return 0; /* @@ -1803,11 +1805,18 @@ static int hdmi_add_cvt(struct hda_codec *codec, hda_nid_t cvt_nid) return 0; } +static const struct snd_pci_quirk force_connect_list[] = { + SND_PCI_QUIRK(0x103c, 0x871a, "HP", 1), + {} +}; + static int hdmi_parse_codec(struct hda_codec *codec) { + struct hdmi_spec *spec = codec->spec; hda_nid_t start_nid; unsigned int caps; int i, nodes; + const struct snd_pci_quirk *q; nodes = snd_hda_get_sub_nodes(codec, codec->core.afg, &start_nid); if (!start_nid || nodes < 0) { @@ -1815,6 +1824,11 @@ static int hdmi_parse_codec(struct hda_codec *codec) return -EINVAL; } + q = snd_pci_quirk_lookup(codec->bus->pci, force_connect_list); + + if (q && q->value) + spec->force_connect = true; + /* * hdmi_add_pin() assumes total amount of converters to * be known, so first discover all converters -- 2.17.1

2 1

Lenovo m720s combined audio jack not working (ALC233)
by Bastien Nocera 04 Aug '20

04 Aug '20

Hey, The front audio jack on my newly purchased (but not new) Lenovo m720s doesn't make a microphone output appear in PulseAudio when a headset is plugged in. The headphone part works correctly though. The "sof-hda-dsp Front Headphone" evdev device will switch the state of SW_HEADPHONE_INSERT as expected. Does anyone know whether the headphones/headset switching is supposed to be automatic, or would that trigger a "what did you plug in" dialogue as on some Dell machines I've worked with? I couldn't find any quirks in sound/pci/hda that would be directly related to this problem (eg. Lenovo, ALC233 and mic presence), so I'm a bit stumped. I've attached the pinout information from alsa-info. Any ideas? Cheers

2 5

[PATCH v2] ALSA: hda/hdmi: Add quirk to force connectivity
by Kai-Heng Feng 04 Aug '20

04 Aug '20

HDMI on some platforms doesn't enable audio support because its Port Connectivity [31:30] is set to AC_JACK_PORT_NONE: Node 0x05 [Pin Complex] wcaps 0x40778d: 8-Channels Digital Amp-Out CP Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0b000094: OUT Detect HBR HDMI DP Pin Default 0x58560010: [N/A] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0x1, Sequence = 0x0 Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Power states: D0 D3 EPSS Power: setting=D0, actual=D0 Devices: 0 Connection: 3 0x02 0x03* 0x04 For now, use a quirk to force connectivity based on SSID. If there are more platforms affected by the same issue, we can eye for a more generic solution. Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> --- v2: - Use a quirk list instead. include/sound/hda_codec.h | 1 + sound/pci/hda/patch_hdmi.c | 14 +++++++++++++- 2 files changed, 14 insertions(+), 1 deletion(-) diff --git a/include/sound/hda_codec.h b/include/sound/hda_codec.h index d16a4229209b..947cba68d9fe 100644 --- a/include/sound/hda_codec.h +++ b/include/sound/hda_codec.h @@ -254,6 +254,7 @@ struct hda_codec { unsigned int link_down_at_suspend:1; /* link down at runtime suspend */ unsigned int relaxed_resume:1; /* don't resume forcibly for jack */ unsigned int mst_no_extra_pcms:1; /* no backup PCMs for DP-MST */ + unsigned int force_connect:1; /* force connectivity */ #ifdef CONFIG_PM unsigned long power_on_acct; diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index cd46247988e4..40d4f0e0e3d7 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1701,7 +1701,8 @@ static int hdmi_add_pin(struct hda_codec *codec, hda_nid_t pin_nid) * all device entries on the same pin */ config = snd_hda_codec_get_pincfg(codec, pin_nid); - if (get_defcfg_connect(config) == AC_JACK_PORT_NONE) + if (get_defcfg_connect(config) == AC_JACK_PORT_NONE && + !codec->force_connect) return 0; /* @@ -1803,11 +1804,17 @@ static int hdmi_add_cvt(struct hda_codec *codec, hda_nid_t cvt_nid) return 0; } +static const struct snd_pci_quirk force_connect_list[] = { + SND_PCI_QUIRK(0x103c, 0x871a, "HP", 1), + {} +}; + static int hdmi_parse_codec(struct hda_codec *codec) { hda_nid_t start_nid; unsigned int caps; int i, nodes; + const struct snd_pci_quirk *q; nodes = snd_hda_get_sub_nodes(codec, codec->core.afg, &start_nid); if (!start_nid || nodes < 0) { @@ -1815,6 +1822,11 @@ static int hdmi_parse_codec(struct hda_codec *codec) return -EINVAL; } + q = snd_pci_quirk_lookup(codec->bus->pci, force_connect_list); + + if (q && q->value) + codec->force_connect = 1; + /* * hdmi_add_pin() assumes total amount of converters to * be known, so first discover all converters -- 2.17.1

2 1

[PATCH] ALSA: hda/hdmi: Add pins with jack detection support
by Kai-Heng Feng 04 Aug '20

04 Aug '20

HDMI on some platforms doesn't enable audio support because its Port Connectivity [31:30] is set to AC_JACK_PORT_NONE: Node 0x05 [Pin Complex] wcaps 0x40778d: 8-Channels Digital Amp-Out CP Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0b000094: OUT Detect HBR HDMI DP Pin Default 0x58560010: [N/A] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0x1, Sequence = 0x0 Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Power states: D0 D3 EPSS Power: setting=D0, actual=D0 Devices: 0 Connection: 3 0x02 0x03* 0x04 Those pins were filtered out by commit 116dcde63806 ("ALSA: HDA: Remove unconnected PCM devices for Intel HDMI"). However, jacks that support detection won't have the issues the commit addresses. So still add the pin if it supports jack detection. Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> --- sound/pci/hda/patch_hdmi.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index cd46247988e4..db3a5148bd40 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1701,7 +1701,8 @@ static int hdmi_add_pin(struct hda_codec *codec, hda_nid_t pin_nid) * all device entries on the same pin */ config = snd_hda_codec_get_pincfg(codec, pin_nid); - if (get_defcfg_connect(config) == AC_JACK_PORT_NONE) + if ((get_defcfg_connect(config) == AC_JACK_PORT_NONE) && + !(caps & AC_PINCAP_PRES_DETECT)) return 0; /* -- 2.17.1

2 5

[PATCH 0/3] ASoC: codec: tlv3204: Codec workaround
by Michael Sit Wei Hong 04 Aug '20

04 Aug '20

This patch series enables some features on the tlv3204 codec and also fixes some issues faced while testing v1: initial ASoC: codec: tlv3204: Codec workaround series Michael Sit Wei Hong (3): ASoC: codec: tlv3204: Enable 24 bit audio support ASoC: codec: tlv3204: Increased maximum supported channels ASoC: codec: tlv3204: Moving GPIO reset and add ADC reset sound/soc/codecs/tlv320aic32x4.c | 60 +++++++++++++++++++++++--------- 1 file changed, 44 insertions(+), 16 deletions(-) -- 2.17.1

5 10

[PATCH] ASoC: fsl_sai: Clean code for synchronize mode
by Shengjiu Wang 04 Aug '20

04 Aug '20

TX synchronous with RX: The RMR is no need to be changed when Tx is enabled, the other configuration in hw_params() is enough for clock generation. The TCSR.TE is no need to enabled when only RX is enabled. RX synchronous with TX: The TMR is no need to be changed when Rx is enabled, the other configuration in hw_params() is enough for clock generation. The RCSR.RE is no need to enabled when only TX is enabled. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/fsl_sai.c | 17 ++++++++++------- 1 file changed, 10 insertions(+), 7 deletions(-) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index cdff739924e2..a210c9836a9a 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -482,8 +482,6 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, regmap_update_bits(sai->regmap, FSL_SAI_TCR5(ofs), FSL_SAI_CR5_WNW_MASK | FSL_SAI_CR5_W0W_MASK | FSL_SAI_CR5_FBT_MASK, val_cr5); - regmap_write(sai->regmap, FSL_SAI_TMR, - ~0UL - ((1 << channels) - 1)); } else if (!sai->synchronous[RX] && sai->synchronous[TX] && tx) { regmap_update_bits(sai->regmap, FSL_SAI_RCR4(ofs), FSL_SAI_CR4_SYWD_MASK | FSL_SAI_CR4_FRSZ_MASK, @@ -491,8 +489,6 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, regmap_update_bits(sai->regmap, FSL_SAI_RCR5(ofs), FSL_SAI_CR5_WNW_MASK | FSL_SAI_CR5_W0W_MASK | FSL_SAI_CR5_FBT_MASK, val_cr5); - regmap_write(sai->regmap, FSL_SAI_RMR, - ~0UL - ((1 << channels) - 1)); } } @@ -553,11 +549,18 @@ static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd, regmap_update_bits(sai->regmap, FSL_SAI_xCSR(tx, ofs), FSL_SAI_CSR_FRDE, FSL_SAI_CSR_FRDE); - regmap_update_bits(sai->regmap, FSL_SAI_RCSR(ofs), - FSL_SAI_CSR_TERE, FSL_SAI_CSR_TERE); - regmap_update_bits(sai->regmap, FSL_SAI_TCSR(ofs), + regmap_update_bits(sai->regmap, FSL_SAI_xCSR(tx, ofs), FSL_SAI_CSR_TERE, FSL_SAI_CSR_TERE); + /* Enable opposite direction when necessarily */ + if (!sai->synchronous[TX] && sai->synchronous[RX] && !tx) { + regmap_update_bits(sai->regmap, FSL_SAI_xCSR((!tx), ofs), + FSL_SAI_CSR_TERE, FSL_SAI_CSR_TERE); + } else if (!sai->synchronous[RX] && sai->synchronous[TX] && tx) { + regmap_update_bits(sai->regmap, FSL_SAI_xCSR((!tx), ofs), + FSL_SAI_CSR_TERE, FSL_SAI_CSR_TERE); + } + regmap_update_bits(sai->regmap, FSL_SAI_xCSR(tx, ofs), FSL_SAI_CSR_xIE_MASK, FSL_SAI_FLAGS); break; -- 2.27.0

3 13

[PATCH] snd: usb: add startech usb audio dock name
by Curtis Malainey 04 Aug '20

04 Aug '20

The dock sold from startech (PID: ICUSBAUDIO7D) has no friendly name and shows up currently as "USB Sound Device" in ALSA. Signed-off-by: Curtis Malainey <cujomalainey(a)chromium.org> --- sound/usb/quirks-table.h | 12 ++++++++---- 1 file changed, 8 insertions(+), 4 deletions(-) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 9092cc0aa8072..fef06836115c1 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -3252,11 +3252,15 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"), } }, +/* + * The original product_name is "USB Sound Device", however this name + * is also used by the CM106 based cards, so make it unique. + */ +{ + USB_DEVICE(0x0d8c, 0x0102), + QUIRK_RENAME_DEVICE(NULL, "ICUSBAUDIO7D") +}, { - /* - * The original product_name is "USB Sound Device", however this name - * is also used by the CM106 based cards, so make it unique. - */ USB_DEVICE(0x0d8c, 0x0103), QUIRK_RENAME_DEVICE(NULL, "Audio Advantage MicroII") }, -- 2.28.0.163.g6104cc2f0b6-goog

2 1

[PATCH] Revert "ALSA: hda: call runtime_allow() for all hda controllers"
by Hui Wang 04 Aug '20

04 Aug '20

This reverts commit 9a6418487b56 ("ALSA: hda: call runtime_allow() for all hda controllers"). The reverted patch already introduced some regressions on some machines: - on gemini-lake machines, the error of "azx_get_response timeout" happens in the hda driver. - on the machines with alc662 codec, the audio jack detection doesn't work anymore. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=208511 Cc: <stable(a)vger.kernel.org> Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/hda_intel.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index e699873c8293..e34a4d5d047c 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2352,7 +2352,6 @@ static int azx_probe_continue(struct azx *chip) if (azx_has_pm_runtime(chip)) { pm_runtime_use_autosuspend(&pci->dev); - pm_runtime_allow(&pci->dev); pm_runtime_put_autosuspend(&pci->dev); } -- 2.17.1

3 7

[PATCH 0/5] ASoC: tegra: Fix compile warning with CONFIG_PM=n
by Takashi Iwai 03 Aug '20

03 Aug '20

Hi, this is a trivial patch set to add the missing __maybe_unused for covering the compile warnings with CONFIG_PM=n. Takashi === Takashi Iwai (5): ASoC: tegra: tegra186_dspk: Fix compile warning with CONFIG_PM=n ASoC: tegra: tegra210_admaif: Fix compile warning with CONFIG_PM=n ASoC: tegra: tegra210_ahub: Fix compile warning with CONFIG_PM=n ASoC: tegra: tegra210_dmic: Fix compile warning with CONFIG_PM=n ASoC: tegra: tegra210_i2s: Fix compile warning with CONFIG_PM=n sound/soc/tegra/tegra186_dspk.c | 4 ++-- sound/soc/tegra/tegra210_admaif.c | 4 ++-- sound/soc/tegra/tegra210_ahub.c | 4 ++-- sound/soc/tegra/tegra210_dmic.c | 4 ++-- sound/soc/tegra/tegra210_i2s.c | 4 ++-- 5 files changed, 10 insertions(+), 10 deletions(-) -- 2.16.4

3 7

[PATCH v3] ASoC: fsl-asoc-card: Remove fsl_asoc_card_set_bias_level function
by Shengjiu Wang 03 Aug '20

03 Aug '20

With this case: aplay -Dhw:x 16khz.wav 24khz.wav There is sound distortion for 24khz.wav. The reason is that setting PLL of WM8962 with set_bias_level function, the bias level is not changed when 24khz.wav is played, then the PLL won't be reset, the clock is not correct, so distortion happens. The resolution of this issue is to remove fsl_asoc_card_set_bias_level. Move PLL configuration to hw_params and hw_free. After removing fsl_asoc_card_set_bias_level, also test WM8960 case, it can work. Fixes: 708b4351f08c ("ASoC: fsl: Add Freescale Generic ASoC Sound Card with ASRC support") Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- changes in v3 - replace "out" with "fail" changes in v2 - replace is_stream_in_use with streams - add "out" error handler in hw_params() sound/soc/fsl/fsl-asoc-card.c | 154 ++++++++++++++++------------------ 1 file changed, 70 insertions(+), 84 deletions(-) diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index ee80d02b56c6..52a9fc73ceb8 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -73,6 +73,7 @@ struct cpu_priv { * @codec_priv: CODEC private data * @cpu_priv: CPU private data * @card: ASoC card structure + * @streams: Mask of current active streams * @sample_rate: Current sample rate * @sample_format: Current sample format * @asrc_rate: ASRC sample rate used by Back-Ends @@ -89,6 +90,7 @@ struct fsl_asoc_card_priv { struct codec_priv codec_priv; struct cpu_priv cpu_priv; struct snd_soc_card card; + u8 streams; u32 sample_rate; snd_pcm_format_t sample_format; u32 asrc_rate; @@ -151,21 +153,17 @@ static int fsl_asoc_card_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(rtd->card); bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + struct codec_priv *codec_priv = &priv->codec_priv; struct cpu_priv *cpu_priv = &priv->cpu_priv; struct device *dev = rtd->card->dev; + unsigned int pll_out; int ret; priv->sample_rate = params_rate(params); priv->sample_format = params_format(params); + priv->streams |= BIT(substream->stream); - /* - * If codec-dai is DAI Master and all configurations are already in the - * set_bias_level(), bypass the remaining settings in hw_params(). - * Note: (dai_fmt & CBM_CFM) includes CBM_CFM and CBM_CFS. - */ - if ((priv->card.set_bias_level && - priv->dai_fmt & SND_SOC_DAIFMT_CBM_CFM) || - fsl_asoc_card_is_ac97(priv)) + if (fsl_asoc_card_is_ac97(priv)) return 0; /* Specific configurations of DAIs starts from here */ @@ -174,7 +172,7 @@ static int fsl_asoc_card_hw_params(struct snd_pcm_substream *substream, cpu_priv->sysclk_dir[tx]); if (ret && ret != -ENOTSUPP) { dev_err(dev, "failed to set sysclk for cpu dai\n"); - return ret; + goto fail; } if (cpu_priv->slot_width) { @@ -182,6 +180,68 @@ static int fsl_asoc_card_hw_params(struct snd_pcm_substream *substream, cpu_priv->slot_width); if (ret && ret != -ENOTSUPP) { dev_err(dev, "failed to set TDM slot for cpu dai\n"); + goto fail; + } + } + + /* Specific configuration for PLL */ + if (codec_priv->pll_id && codec_priv->fll_id) { + if (priv->sample_format == SNDRV_PCM_FORMAT_S24_LE) + pll_out = priv->sample_rate * 384; + else + pll_out = priv->sample_rate * 256; + + ret = snd_soc_dai_set_pll(asoc_rtd_to_codec(rtd, 0), + codec_priv->pll_id, + codec_priv->mclk_id, + codec_priv->mclk_freq, pll_out); + if (ret) { + dev_err(dev, "failed to start FLL: %d\n", ret); + goto fail; + } + + ret = snd_soc_dai_set_sysclk(asoc_rtd_to_codec(rtd, 0), + codec_priv->fll_id, + pll_out, SND_SOC_CLOCK_IN); + + if (ret && ret != -ENOTSUPP) { + dev_err(dev, "failed to set SYSCLK: %d\n", ret); + goto fail; + } + } + + return 0; + +fail: + priv->streams &= ~BIT(substream->stream); + return ret; +} + +static int fsl_asoc_card_hw_free(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(rtd->card); + struct codec_priv *codec_priv = &priv->codec_priv; + struct device *dev = rtd->card->dev; + int ret; + + priv->streams &= ~BIT(substream->stream); + + if (!priv->streams && codec_priv->pll_id && codec_priv->fll_id) { + /* Force freq to be 0 to avoid error message in codec */ + ret = snd_soc_dai_set_sysclk(asoc_rtd_to_codec(rtd, 0), + codec_priv->mclk_id, + 0, + SND_SOC_CLOCK_IN); + if (ret) { + dev_err(dev, "failed to switch away from FLL: %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_pll(asoc_rtd_to_codec(rtd, 0), + codec_priv->pll_id, 0, 0, 0); + if (ret && ret != -ENOTSUPP) { + dev_err(dev, "failed to stop FLL: %d\n", ret); return ret; } } @@ -191,6 +251,7 @@ static int fsl_asoc_card_hw_params(struct snd_pcm_substream *substream, static const struct snd_soc_ops fsl_asoc_card_ops = { .hw_params = fsl_asoc_card_hw_params, + .hw_free = fsl_asoc_card_hw_free, }; static int be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, @@ -254,75 +315,6 @@ static struct snd_soc_dai_link fsl_asoc_card_dai[] = { }, }; -static int fsl_asoc_card_set_bias_level(struct snd_soc_card *card, - struct snd_soc_dapm_context *dapm, - enum snd_soc_bias_level level) -{ - struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(card); - struct snd_soc_pcm_runtime *rtd; - struct snd_soc_dai *codec_dai; - struct codec_priv *codec_priv = &priv->codec_priv; - struct device *dev = card->dev; - unsigned int pll_out; - int ret; - - rtd = snd_soc_get_pcm_runtime(card, &card->dai_link[0]); - codec_dai = asoc_rtd_to_codec(rtd, 0); - if (dapm->dev != codec_dai->dev) - return 0; - - switch (level) { - case SND_SOC_BIAS_PREPARE: - if (dapm->bias_level != SND_SOC_BIAS_STANDBY) - break; - - if (priv->sample_format == SNDRV_PCM_FORMAT_S24_LE) - pll_out = priv->sample_rate * 384; - else - pll_out = priv->sample_rate * 256; - - ret = snd_soc_dai_set_pll(codec_dai, codec_priv->pll_id, - codec_priv->mclk_id, - codec_priv->mclk_freq, pll_out); - if (ret) { - dev_err(dev, "failed to start FLL: %d\n", ret); - return ret; - } - - ret = snd_soc_dai_set_sysclk(codec_dai, codec_priv->fll_id, - pll_out, SND_SOC_CLOCK_IN); - if (ret && ret != -ENOTSUPP) { - dev_err(dev, "failed to set SYSCLK: %d\n", ret); - return ret; - } - break; - - case SND_SOC_BIAS_STANDBY: - if (dapm->bias_level != SND_SOC_BIAS_PREPARE) - break; - - ret = snd_soc_dai_set_sysclk(codec_dai, codec_priv->mclk_id, - codec_priv->mclk_freq, - SND_SOC_CLOCK_IN); - if (ret && ret != -ENOTSUPP) { - dev_err(dev, "failed to switch away from FLL: %d\n", ret); - return ret; - } - - ret = snd_soc_dai_set_pll(codec_dai, codec_priv->pll_id, 0, 0, 0); - if (ret) { - dev_err(dev, "failed to stop FLL: %d\n", ret); - return ret; - } - break; - - default: - break; - } - - return 0; -} - static int fsl_asoc_card_audmux_init(struct device_node *np, struct fsl_asoc_card_priv *priv) { @@ -608,7 +600,6 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) /* Diversify the card configurations */ if (of_device_is_compatible(np, "fsl,imx-audio-cs42888")) { codec_dai_name = "cs42888"; - priv->card.set_bias_level = NULL; priv->cpu_priv.sysclk_freq[TX] = priv->codec_priv.mclk_freq; priv->cpu_priv.sysclk_freq[RX] = priv->codec_priv.mclk_freq; priv->cpu_priv.sysclk_dir[TX] = SND_SOC_CLOCK_OUT; @@ -625,26 +616,22 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; } else if (of_device_is_compatible(np, "fsl,imx-audio-wm8962")) { codec_dai_name = "wm8962"; - priv->card.set_bias_level = fsl_asoc_card_set_bias_level; priv->codec_priv.mclk_id = WM8962_SYSCLK_MCLK; priv->codec_priv.fll_id = WM8962_SYSCLK_FLL; priv->codec_priv.pll_id = WM8962_FLL; priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; } else if (of_device_is_compatible(np, "fsl,imx-audio-wm8960")) { codec_dai_name = "wm8960-hifi"; - priv->card.set_bias_level = fsl_asoc_card_set_bias_level; priv->codec_priv.fll_id = WM8960_SYSCLK_AUTO; priv->codec_priv.pll_id = WM8960_SYSCLK_AUTO; priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; } else if (of_device_is_compatible(np, "fsl,imx-audio-ac97")) { codec_dai_name = "ac97-hifi"; - priv->card.set_bias_level = NULL; priv->dai_fmt = SND_SOC_DAIFMT_AC97; priv->card.dapm_routes = audio_map_ac97; priv->card.num_dapm_routes = ARRAY_SIZE(audio_map_ac97); } else if (of_device_is_compatible(np, "fsl,imx-audio-mqs")) { codec_dai_name = "fsl-mqs-dai"; - priv->card.set_bias_level = NULL; priv->dai_fmt = SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_CBS_CFS | SND_SOC_DAIFMT_NB_NF; @@ -654,7 +641,6 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) priv->card.num_dapm_routes = ARRAY_SIZE(audio_map_tx); } else if (of_device_is_compatible(np, "fsl,imx-audio-wm8524")) { codec_dai_name = "wm8524-hifi"; - priv->card.set_bias_level = NULL; priv->dai_fmt |= SND_SOC_DAIFMT_CBS_CFS; priv->dai_link[1].dpcm_capture = 0; priv->dai_link[2].dpcm_capture = 0; -- 2.27.0

3 2

[PATCH] ASoC: wm8962: Do not remove ADDITIONAL_CONTROL_4 from readable register list
by Fabio Estevam 03 Aug '20

03 Aug '20

Removing ADDITIONAL_CONTROL_4 from the list of readable registers cause audio distortion. This change was sent as a comment below the --- line when submitting commit 658bb297e393 ("ASoC: wm8962: Do not access WM8962_GPIO_BASE"), so it was not supposed to get merged. Keep WM8962_ADDITIONAL_CONTROL_4 inside wm8962_readable_register() to fix the regression. Fixes: 658bb297e393 ("ASoC: wm8962: Do not access WM8962_GPIO_BASE") Reported-by: Shengjiu Wang <shengjiu.wang(a)gmail.com> Signed-off-by: Fabio Estevam <festevam(a)gmail.com> --- sound/soc/codecs/wm8962.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 317916cb4e27..0623a2251084 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -151,7 +151,6 @@ static const struct reg_default wm8962_reg[] = { { 40, 0x0000 }, /* R40 - SPKOUTL volume */ { 41, 0x0000 }, /* R41 - SPKOUTR volume */ - { 48, 0x0000 }, /* R48 - Additional control(4) */ { 49, 0x0010 }, /* R49 - Class D Control 1 */ { 51, 0x0003 }, /* R51 - Class D Control 2 */ @@ -842,6 +841,7 @@ static bool wm8962_readable_register(struct device *dev, unsigned int reg) case WM8962_SPKOUTL_VOLUME: case WM8962_SPKOUTR_VOLUME: case WM8962_THERMAL_SHUTDOWN_STATUS: + case WM8962_ADDITIONAL_CONTROL_4: case WM8962_CLASS_D_CONTROL_1: case WM8962_CLASS_D_CONTROL_2: case WM8962_CLOCKING_4: -- 2.17.1

2 2

[PATCH] ASoC: amd: renoir: restore two more registers during resume
by Hui Wang 03 Aug '20

03 Aug '20

Recently we found an issue about the suspend and resume. If dmic is recording the sound, and we run suspend and resume, after the resume, the dmic can't work well anymore. we need to close the app and reopen the app, then the dmic could record the sound again. For example, we run "arecord -D hw:CARD=acp,DEV=0 -f S32_LE -c 2 -r 48000 test.wav", then suspend and resume, after the system resume back, we speak to the dmic. then stop the arecord, use aplay to play the test.wav, we could hear the sound recorded after resume is weird, it is not what we speak to the dmic. I found two registers are set in the dai_hw_params(), if the two registers are set in the resume() too, this issue could be fixed. Cc: stable(a)vger.kernel.org Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/soc/amd/renoir/acp3x-pdm-dma.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/soc/amd/renoir/acp3x-pdm-dma.c b/sound/soc/amd/renoir/acp3x-pdm-dma.c index 623dfd3ea705..8acb0315a169 100644 --- a/sound/soc/amd/renoir/acp3x-pdm-dma.c +++ b/sound/soc/amd/renoir/acp3x-pdm-dma.c @@ -474,6 +474,11 @@ static int acp_pdm_resume(struct device *dev) rtd = runtime->private_data; period_bytes = frames_to_bytes(runtime, runtime->period_size); buffer_len = frames_to_bytes(runtime, runtime->buffer_size); + if (runtime->channels == TWO_CH) { + rn_writel(0x0 , rtd->acp_base + ACP_WOV_PDM_NO_OF_CHANNELS); + rn_writel(PDM_DECIMATION_FACTOR, rtd->acp_base + + ACP_WOV_PDM_DECIMATION_FACTOR); + } config_acp_dma(rtd, SNDRV_PCM_STREAM_CAPTURE); init_pdm_ring_buffer(MEM_WINDOW_START, buffer_len, period_bytes, adata->acp_base); -- 2.17.1

3 3

[PATCH v2] ASoC: amd: renoir: restore two more registers during resume
by Hui Wang 03 Aug '20

03 Aug '20

Recently we found an issue about the suspend and resume. If dmic is recording the sound, and we run suspend and resume, after the resume, the dmic can't work well anymore. we need to close the app and reopen the app, then the dmic could record the sound again. For example, we run "arecord -D hw:CARD=acp,DEV=0 -f S32_LE -c 2 -r 48000 test.wav", then suspend and resume, after the system resume back, we speak to the dmic. then stop the arecord, use aplay to play the test.wav, we could hear the sound recorded after resume is weird, it is not what we speak to the dmic. I found two registers are set in the dai_hw_params(), if the two registers are set during the resume, this issue could be fixed. Move the code of the dai_hw_params() into the pdm_dai_trigger(), then these two registers will be set during resume since pdm_dai_trigger() will be called during resume. And delete the empty function dai_hw_params(). Cc: stable(a)vger.kernel.org Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/soc/amd/renoir/acp3x-pdm-dma.c | 29 +++++++++------------------- 1 file changed, 9 insertions(+), 20 deletions(-) diff --git a/sound/soc/amd/renoir/acp3x-pdm-dma.c b/sound/soc/amd/renoir/acp3x-pdm-dma.c index 623dfd3ea705..7b14d9a81b97 100644 --- a/sound/soc/amd/renoir/acp3x-pdm-dma.c +++ b/sound/soc/amd/renoir/acp3x-pdm-dma.c @@ -314,40 +314,30 @@ static int acp_pdm_dma_close(struct snd_soc_component *component, return 0; } -static int acp_pdm_dai_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *dai) +static int acp_pdm_dai_trigger(struct snd_pcm_substream *substream, + int cmd, struct snd_soc_dai *dai) { struct pdm_stream_instance *rtd; + int ret; + bool pdm_status; unsigned int ch_mask; rtd = substream->runtime->private_data; - switch (params_channels(params)) { + ret = 0; + switch (substream->runtime->channels) { case TWO_CH: ch_mask = 0x00; break; default: return -EINVAL; } - rn_writel(ch_mask, rtd->acp_base + ACP_WOV_PDM_NO_OF_CHANNELS); - rn_writel(PDM_DECIMATION_FACTOR, rtd->acp_base + - ACP_WOV_PDM_DECIMATION_FACTOR); - return 0; -} - -static int acp_pdm_dai_trigger(struct snd_pcm_substream *substream, - int cmd, struct snd_soc_dai *dai) -{ - struct pdm_stream_instance *rtd; - int ret; - bool pdm_status; - - rtd = substream->runtime->private_data; - ret = 0; switch (cmd) { case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + rn_writel(ch_mask, rtd->acp_base + ACP_WOV_PDM_NO_OF_CHANNELS); + rn_writel(PDM_DECIMATION_FACTOR, rtd->acp_base + + ACP_WOV_PDM_DECIMATION_FACTOR); rtd->bytescount = acp_pdm_get_byte_count(rtd, substream->stream); pdm_status = check_pdm_dma_status(rtd->acp_base); @@ -369,7 +359,6 @@ static int acp_pdm_dai_trigger(struct snd_pcm_substream *substream, } static struct snd_soc_dai_ops acp_pdm_dai_ops = { - .hw_params = acp_pdm_dai_hw_params, .trigger = acp_pdm_dai_trigger, }; -- 2.17.1

3 2

[PATCH] ASoC: fsl: Fix unused variable warning
by Takashi Iwai 03 Aug '20

03 Aug '20

The variable rtd was left unused in psc_dma_free(), even unnoticed during conversion to a new style: sound/soc/fsl/mpc5200_dma.c:342:30: warning: unused variable 'rtd' [-Wunused-variable] Drop the superfluous one. Fixes: 6d1048bc1152 ("ASoC: fsl: mpc5200_dma: remove snd_pcm_ops") Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/soc/fsl/mpc5200_dma.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/fsl/mpc5200_dma.c b/sound/soc/fsl/mpc5200_dma.c index 9e4f66b6b92b..231984882176 100644 --- a/sound/soc/fsl/mpc5200_dma.c +++ b/sound/soc/fsl/mpc5200_dma.c @@ -339,7 +339,6 @@ static int psc_dma_new(struct snd_soc_component *component, static void psc_dma_free(struct snd_soc_component *component, struct snd_pcm *pcm) { - struct snd_soc_pcm_runtime *rtd = pcm->private_data; struct snd_pcm_substream *substream; int stream; -- 2.16.4

2 1

[PATCH] ALSA: usb-audio: Add support for Lenovo ThinkStation P620
by Kai-Heng Feng 03 Aug '20

03 Aug '20

Lenovo ThinkStation P620 is like other TRX40 boards, is equipped with two USB audio cards. USB device (17aa:104d) provides functionality for Internal Speaker and Front Headset. It's UAC v2, so it supports insertion control (jack detection). However, when trying to get the connector status of the speaker, an error occurs: [ 5.787405] usb 3-1: cannot get connectors status: req = 0x81, wValue = 0x200, wIndex = 0x1000, type = 0 Since the insertion control works perfectly for the headset, the error for speaker is probably casued by connecting internally. So let's relax the error for a bit if it's a speaker, and always reports it's connected. USB device (17aa:1046) is for rear Line-in, Line-out and Microphone. The insertion control works for all three jacks. However, there's an Function Unit that doesn't work: [ 5.905415] usb 3-6: cannot get ctl value: req = 0x83, wValue = 0xc00, wIndex = 0x1300, type = 4 [ 5.905418] usb 3-6: 19:0: cannot get min/max values for control 12 (id 19) So turn off the FU to avoid the error. Also, add specific card name for both devices, so userspace can easily indentify both cards. Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> --- sound/usb/mixer.c | 4 ++++ sound/usb/mixer_maps.c | 9 +++++++++ sound/usb/quirks-table.h | 13 +++++++++++++ 3 files changed, 26 insertions(+) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index eab0fd4fd7c3..6b0f3a8469ef 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -1463,6 +1463,10 @@ static int mixer_ctl_connector_get(struct snd_kcontrol *kcontrol, snd_usb_unlock_shutdown(chip); if (ret < 0) { + if (strstr(kcontrol->id.name, "Speaker")) { + ucontrol->value.integer.value[0] = 1; + return 0; + } error: usb_audio_err(chip, "cannot get connectors status: req = %#x, wValue = %#x, wIndex = %#x, type = %d\n", diff --git a/sound/usb/mixer_maps.c b/sound/usb/mixer_maps.c index 9af7aa93f6fa..08eb230e2bff 100644 --- a/sound/usb/mixer_maps.c +++ b/sound/usb/mixer_maps.c @@ -370,6 +370,11 @@ static const struct usbmix_name_map asus_rog_map[] = { {} }; +static const struct usbmix_name_map lenovo_p620_rear_map[] = { + { 19, NULL, 12 }, /* FU, Input Gain Pad */ + {} +}; + /* TRX40 mobos with Realtek ALC1220-VB */ static const struct usbmix_name_map trx40_mobo_map[] = { { 18, NULL }, /* OT, IEC958 - broken response, disabled */ @@ -573,6 +578,10 @@ static const struct usbmix_ctl_map usbmix_ctl_maps[] = { .map = trx40_mobo_map, .connector_map = trx40_mobo_connector_map, }, + { /* Lenovo ThinkStation P620 Rear */ + .id = USB_ID(0x17aa, 0x1046), + .map = lenovo_p620_rear_map, + }, { 0 } /* terminator */ }; diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 9092cc0aa807..09590f055a1d 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -2794,6 +2794,19 @@ YAMAHA_DEVICE(0x7010, "UB99"), QUIRK_RENAME_DEVICE("Rane", "SL-1") }, +/* Lenovo ThinkStation P620 Rear Line-in, Line-out and Microphone */ +{ + USB_DEVICE(0x17aa, 0x1046), + QUIRK_DEVICE_PROFILE("Lenovo", "ThinkStation P620 Rear", + "Lenovo-ThinkStation-P620-Rear"), +}, +/* Lenovo ThinkStation P620 Internal Speaker + Front Headset */ +{ + USB_DEVICE(0x17aa, 0x104d), + QUIRK_DEVICE_PROFILE("Lenovo", "ThinkStation P620 Main", + "Lenovo-ThinkStation-P620-Main"), +}, + /* Native Instruments MK2 series */ { /* Komplete Audio 6 */ -- 2.17.1

2 1

[PATCH] ALSA: echoaduio: Drop superfluous volatile modifier
by Takashi Iwai 03 Aug '20

03 Aug '20

The dsp_registers field of struct echoaduio has the volatile modifier, but it's basically superfluous; the field is accessed only for the base pointer of readl() and writel(), hence marking with __iomem alone should suffice. OTOH, having the volatile prefix causes a compile warning like: sound/pci/echoaudio/echoaudio.c:1878:14: warning: passing argument 1 of 'iounmap' discards 'volatile' qualifier from pointer target type [-Wdiscarded-qualifiers] So it's better to drop this superfluous modifier. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/echoaudio/echoaudio.c | 3 +-- sound/pci/echoaudio/echoaudio.h | 2 +- 2 files changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c index 6aeb99aa2414..1b7a67ccabce 100644 --- a/sound/pci/echoaudio/echoaudio.c +++ b/sound/pci/echoaudio/echoaudio.c @@ -1950,8 +1950,7 @@ static int snd_echo_create(struct snd_card *card, snd_echo_free(chip); return -EBUSY; } - chip->dsp_registers = (volatile u32 __iomem *) - ioremap(chip->dsp_registers_phys, sz); + chip->dsp_registers = ioremap(chip->dsp_registers_phys, sz); if (!chip->dsp_registers) { dev_err(chip->card->dev, "ioremap failed\n"); snd_echo_free(chip); diff --git a/sound/pci/echoaudio/echoaudio.h b/sound/pci/echoaudio/echoaudio.h index 30c640931f1e..0afe13f7b6e5 100644 --- a/sound/pci/echoaudio/echoaudio.h +++ b/sound/pci/echoaudio/echoaudio.h @@ -419,7 +419,7 @@ struct echoaudio { short asic_code; /* Current ASIC code */ u32 comm_page_phys; /* Physical address of the * memory seen by DSP */ - volatile u32 __iomem *dsp_registers; /* DSP's register base */ + u32 __iomem *dsp_registers; /* DSP's register base */ u32 active_mask; /* Chs. active mask or * punks out */ #ifdef CONFIG_PM_SLEEP -- 2.16.4

1 0

[GIT PULL] ASoC updates for v5.9
by Mark Brown 03 Aug '20

03 Aug '20

The following changes since commit 92ed301919932f777713b9172e525674157e983d: Linux 5.8-rc7 (2020-07-26 14:14:06 -0700) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v5.9 for you to fetch changes up to 84569f329f7fcb40b7b1860f273b2909dabf2a2b: Merge remote-tracking branch 'asoc/for-5.9' into asoc-next (2020-07-31 19:54:03 +0100) ---------------------------------------------------------------- ASoC: Updates for v5.9 The biggest changes here one again come from Mormioto-san who has continued his dilligent work cleaning up long standing issues in the APIs, it's particularly nice to see the transition from digital_mute() to mute_stream() finally completed. There's also been a lot of work on the x86 code again, this time a big focus has been on cleaning up some issues identified by various static tests, and on the Freescale systems. Otherwise the biggest thing has been a lot of driver additions: - Convert users of digital_mute() to mute_stream(). - Simplify I/O helper functions. - Add a helper for getting the RTD from a substream. - Many, many fixes and cleanups to the x86 code. - New drivers for Freescale MQS and i.MX6sx, Intel KeemBay I2S, Maxim MAX98360A and MAX98373 Soundwire, several Mediatek boards, nVidia Tegra 186 and 210, RealTek RL6231, Samsung Midas and Aries boards (some of the first phones I worked on!) and TI J721e EVM. ---------------------------------------------------------------- Akshu Agrawal (3): ASoC: AMD: Use mixer control to switch between DMICs ASoC: amd: Enable interrupt in dma_open ASoC: AMD: Restore PME_EN state at Power On Alexander A. Klimov (4): ASoC: Replace HTTP links with HTTPS ones ASoC: ti: Replace HTTP links with HTTPS ones ASoC: fsl: Replace HTTP links with HTTPS ones ASoC: omap: Replace HTTP links with HTTPS ones Alper Nebi Yasak (1): ASoC: rk3399_gru_sound: Add DAPM pins, kcontrols for jack detection Bard Liao (2): ASoC: intel: cml_rt1011_rt5682: use for_each_card_prelinks ASoC: Intel: sof_sdw_rt711: remove hard-coded codec name Brent Lu (1): ASoC: Intel: bxt-da7219-max98357a: support MAX98390 speaker amp Cezary Rojewski (3): ASoC: core: Relocate and expose snd_soc_component_initialize ASoC: core: Simplify snd_soc_component_initialize declaration ASoC: core: Two step component registration Charles Keepax (2): ASoC: madera: Add channel numbers to AIFs ASoC: soc-core: Fix regression causing sysfs entries to disappear Codrin Ciubotariu (2): ASoC: atmel-classd: remove codec component ASoC: atmel-pdmic: remove codec component Colin Ian King (1): ASoC: Intel: bxt-da7219-max98357a: return -EINVAL on unrecognized speaker amplifier Colton Lewis (1): snd/soc: correct trivial kernel-doc inconsistency Dan Murphy (17): ASoC: tas2562: Update shutdown GPIO property ASoC: tas2562: Fix format issue for extra space before a comma ASoC: tas2562: Fix shut-down gpio property ASoC: tas2562: Add right and left channel slot programming ASoC: tas2562: Add voltage sense slot configuration ASoC: tas2562: Add voltage sense slot property ASoC: tlv320adcx140: Add ASI enable for channel 5-8 ASoC: tlv320adcx140: Add tx offset slot programming ASoC: tas2770: Fix reset gpio property name ASoC: tas2770: Convert tas2770 binding to yaml ASoC: tas2562: Update shutdown GPIO property ASoC: tas2562: Convert the tas2562 binding to yaml ASoC: tlv320adcx140: Fix various style errors and warnings dt-bindings: tlv320adcx140: Add GPO config and drive config ASoC: tlv320adcx140: Add GPO configuration and drive output config ASoC: tlv320adcx140: Fix GPO register start address ASoC: tlv320adcx140: Move device reset to before programming Daniel Baluta (7): ASoC: SOF: define INFO_ flags in dsp_ops for imx8 ASoC: SOF: imx: Use ARRAY_SIZE instead of hardcoded value ASoC: SOF: imx8: Fix ESAI DAI driver name for i.MX8/iMX8X ASoC: SOF: imx8m: Fix SAI DAI driver for i.MX8M ASoC: SOF: imx8: Add SAI dai driver for i.MX/i.MX8X ASoC: SOF: topology: Update SAI config bclk/fsync rate ASoC: SOF: pcm: Update rate/channels for SAI/ESAI DAIs Dharageswari R (3): ASoC: Intel: Boards: tgl_max98373: add dai_trigger function ASoC: Intel: Boards: tgl_max98373: Fix the comment for max_98373_components ASoC: Intel: Boards: tgl_max98373: Update TDM configuration in hw_params Fabio Estevam (3): ASoC: dt-bindings: keembay-i2s: Fix reg descriptions ASoC: dt-bindings: simple-card: Fix 'make dt_binding_check' warnings ASoC: wm8962: Do not access WM8962_GPIO_BASE Fred Oh (6): ASoC: intel: sof_rt5682: move disabling jack to dai link's exit() ASoC: intel: cml_rt1011_rt5682: disable jack in dailink .exit() ASoC: rt1011: fix KASAN out-of-bounds bug in find_next_bit() ASoC: Intel: Boards: cml_rt1011_rt5682: reduce log level for printing quirk ASoC: Intel: Boards: cml_rt1011_rt5682: use statically define codec config ASoc: Intel: cml_rt1011_rt5682: explicitly access first codec Fuqian Huang (1): ASoC: wm0010: Use kmemdup rather than duplicating its implementation Gustavo A. R. Silva (2): ASoC: codecs: Use fallthrough pseudo-keyword ASoC: Intel: Skylake: Avoid the use of one-element array Hans de Goede (3): ASoC: rt5670: Remove struct rt5670_platform_data ASoC: rt5670: Rename dev_gpio to gpio1_is_irq ASoC: Intel: cht_bsw_rt5672: Improve dai-set-fmt comment in cht_codec_fixup() Harsha Priya (1): ASoC: Intel: kbl_rt5663_rt5514_max98927: Fix kabylake_ssp_fixup function Hauke Mehrtens (1): spi: lantiq-ssc: Fix warning by using WQ_MEM_RECLAIM Jack Yu (1): ASoC: rt1015: Flush DAC data before playback. James Schulman (1): ASoC: wm_adsp: Support new metadata block ID's Jerome Brunet (7): ASoC: meson: imply acodec glue on axg sound card ASoC: soc-component: don't report of_xlate_dai_name failures ASoC: meson: axg-tdm-interface: fix link fmt setup ASoC: meson: axg-tdmin: fix g12a skew ASoC: meson: axg-tdm-formatters: fix sclk inversion ASoC: meson: cards: remove DT_PREFIX for standard daifmt properties ASoC: meson: cards: deal dpcm flag change Jiaxin Yu (2): ASoC: mediatek: mt6358: support DMIC one-wire mode ASoC: dt-bindings: mediatek: mt6358: add dmic-mode property Jing Xiangfeng (1): ASoC: meson: fixes the missed kfree() for axg_card_add_tdm_loopback John Stultz (1): ASoC: qcom: Kconfig: Tweak dependencies on SND_SOC_SDM845 Jonathan Bakker (3): ASoC: samsung: Add driver for Aries boards ASoC: Document wm8994 endpoints ASoC: Add bindings for Samsung Aries audio complex Julia Lawall (2): ASoC: Intel: drop unnecessary list_empty ASoC: SOF: imx: use resource_size Kai Vehmanen (8): ALSA: hda: export snd_hda_codec_cleanup_for_unbind() ALSA: hda: fix snd_hda_codec_cleanup() documentation ASoC: Intel: sof_sdw: add support for systems without i915 audio ASoC: Intel: sof_sdw: avoid crash if invalid DSP topology loaded ASoC: intel/skl/hda - fix probe regression on systems without i915 ASoC: hdac_hda: call patch_ops.free() on probe error ASoC: hdac_hda: fix memleak on module unload ASoC: hdac_hda: fix deadlock after PCM open error Katsuhiro Suzuki (3): ASoC: convert rk3328 codec binding to yaml ASoC: convert ROHM BD28623 amplifier binding to yaml ASoC: convert Everest ES8316 binding to yaml Kuninori Morimoto (90): ASoC: soc-component: add soc_component_pin() and share code ASoC: soc-component: move snd_soc_component_xxx_regmap() to soc-component ASoC: soc-component: move snd_soc_component_initialize() to soc-component.c ASoC: soc-component: add soc_component_err() ASoC: soc-component: add snd_soc_pcm_component_prepare() ASoC: soc-component: add snd_soc_pcm_component_hw_params() ASoC: soc-component: add snd_soc_pcm_component_hw_free() ASoC: soc-component: add snd_soc_pcm_component_trigger() ASoC: soc-component: add snd_soc_component_init() ASoC: soc-component: merge soc-io.c into soc-component.c ASoC: soc-component: merge soc_pcm_trigger_start/stop() ASoC: soc-component: tidyup Copyright ASoC: soc-component: merge snd_soc_component_read() and snd_soc_component_read32() ASoC: soc-component: use io_mutex correctly ASoC: pxa: rename to snd_soc_component_read() ASoC: atmel: rename to snd_soc_component_read() ASoC: codecs: rename to snd_soc_component_read() ASoC: codecs: wcd*: rename to snd_soc_component_read() ASoC: codecs: tlv*: rename to snd_soc_component_read() ASoC: codecs: max*: rename to snd_soc_component_read() ASoC: codecs: msm*: rename to snd_soc_component_read() ASoC: codecs: alc*: rename to snd_soc_component_read() ASoC: codecs: wm*: rename to snd_soc_component_read() ASoC: codecs: rt*: rename to snd_soc_component_read() ASoC: codecs: da*: rename to snd_soc_component_read() ASoC: codecs: cs*: rename to snd_soc_component_read() ASoC: codecs: ak*: rename to snd_soc_component_read() ASoC: remove snd_soc_component_read32() ARM: dts: motorola-mapphone-common: remove unneeded "simple-graph-card" ASoC: rsnd: don't use discriminatory terms for function names ASoC: rsnd: don't use discriminatory terms for comment ASoC: fsi: don't use discriminatory terms for comment ASoC: siu: don't use discriminatory terms for parameter ASoC: ssi: don't use discriminatory terms for debug log ASoC: hdmi-codec: return -ENOTSUPP for digital_mute ASoC: soc-dai.c: add .no_capture_mute support ASoC: hdmi-codec: merge .digital_mute() into .mute_stream() ASoC: ti: merge .digital_mute() into .mute_stream() ASoC: spear: merge .digital_mute() into .mute_stream() ASoC: meson: merge .digital_mute() into .mute_stream() ASoC: codecs: merge .digital_mute() into .mute_stream() ASoC: codecs: tlv*: merge .digital_mute() into .mute_stream() ASoC: codecs: tas*: merge .digital_mute() into .mute_stream() ASoC: codecs: ssm*: merge .digital_mute() into .mute_stream() ASoC: codecs: pcm*: merge .digital_mute() into .mute_stream() ASoC: codecs: max*: merge .digital_mute() into .mute_stream() ASoC: codecs: alc*: merge .digital_mute() into .mute_stream() ASoC: codecs: wm*: merge .digital_mute() into .mute_stream() ASoC: codecs: es*: merge .digital_mute() into .mute_stream() ASoC: codecs: da*: merge .digital_mute() into .mute_stream() ASoC: codecs: cs*: merge .digital_mute() into .mute_stream() ASoC: codecs: ak*: merge .digital_mute() into .mute_stream() ASoC: dt-bindings: ak4642: switch to yaml base Documentation ASoC: dt-bindings: ak4613: switch to yaml base Documentation ASoC: dt-bindings: renesas, fsi: use patternProperties for FSI-A/B ASoC: atmel: merge .digital_mute() into .mute_stream() ASoC: soc-core: snd_soc_dai_digital_mute() for both CPU/Codec ASoC: soc-dai.h: don't use discriminatory terms for comment ASoC: soc-dai: remove .digital_mute ASoC: soc-xxx: add asoc_substream_to_rtd() ASoC: ux500: use asoc_substream_to_rtd() ASoC: ti: use asoc_substream_to_rtd() ASoC: tegra: use asoc_substream_to_rtd() ASoC: sunxi: use asoc_substream_to_rtd() ASoC: stm: use asoc_substream_to_rtd() ASoC: sof: use asoc_substream_to_rtd() ASoC: sh: use asoc_substream_to_rtd() ASoC: samsung: use asoc_substream_to_rtd() ASoC: pxa: use asoc_substream_to_rtd() ASoC: cirrus: use asoc_substream_to_rtd() ASoC: rockchip: use asoc_substream_to_rtd() ASoC: amd: use asoc_substream_to_rtd() ASoC: fsl: use asoc_substream_to_rtd() ASoC: mediatek: use asoc_substream_to_rtd() ASoC: atmel: use asoc_substream_to_rtd() ASoC: qcom: use asoc_substream_to_rtd() ASoC: dwc: use asoc_substream_to_rtd() ASoC: meson: use asoc_substream_to_rtd() ASoC: au1x: use asoc_substream_to_rtd() ASoC: bcm: use asoc_substream_to_rtd() ASoC: codecs: use asoc_substream_to_rtd() ASoC: generic: use asoc_substream_to_rtd() ASoC: sprd: use asoc_substream_to_rtd() ASoC: kirkwood: use asoc_substream_to_rtd() ASoC: xtensa: use asoc_substream_to_rtd() ASoC: mxs: use asoc_substream_to_rtd() ASoC: uniphier: use asoc_substream_to_rtd() ASoC: intel/boards: use asoc_substream_to_rtd() ASoC: intel: use asoc_substream_to_rtd() ASoC: dt-bindings: ak4613: switch to yaml base Documentation Lad Prabhakar (1): dt-bindings: sound: renesas, rsnd: Document r8a774e1 bindings Laurent Pinchart (1): ASoC: sh: Replace 'select' DMADEVICES 'with depends on' Lee Jones (9): ASoC: codecs: jz4770: Remove defined but never used variable 'mic_boost_tlv' ASoC: ti: omap-mcbsp-st: Remove set, but unused variable 'w' ASoC: sunxi: sun4i-spdif: Fix misspelling of 'reg_dac_txdata' in kernel-doc ASoC: pxa: pxa-ssp: Demote seemingly unintentional kerneldoc header ASoC: ux500: ux500_msp_i2s: Remove unused variables 'reg_val_DR' and 'reg_val_TSTDR' ASoC: codecs: rt5659: Remove many unused const variables ASoC: codecs: tlv320aic26: Demote seemingly unintentional kerneldoc header ASoC: codecs: ak4458: Remove set but never checked variable 'ret' ASoC: fsl: fsl-asoc-card: Trivial: Fix misspelling of 'exists' Liao Pingfang (1): ASoC: fsl: mpc8610_hpcd: Add missing of_node_put() Libin Yang (1): ASoC: Intel: common: change match table ehl-rt5660 Marcin Rajwa (2): ASoC: SOF: Intel: fix the suspend procedure to support s0ix entry ASoC: SOF: Intel: disable traces when switching to S0Ix D0I3 Mark Brown (51): Merge existing fixes from asoc/for-5.8 Merge series "ASoC: soc-component: collect component functions" from Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com>: Merge series "ASoC: mediatek: mt6358: support DMIC one-wire mode" from Jiaxin Yu <jiaxin.yu(a)mediatek.com>: Merge series "ASoC: max98357a: support MAX98360A in OF" from Tzung-Bi Shih <tzungbi(a)google.com>: Merge series "ASoC: improve core dmesg logs and verbosity" from Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com>: Merge series "ASoC: Intel: Add KeemBay ASoC platform driver" from Sia Jee Heng <jee.heng.sia(a)intel.com>: Merge series "ASoC: fsl_easrc: Fix several warnings" from Shengjiu Wang <shengjiu.wang(a)nxp.com>: Merge series "ASoC: merge snd_soc_component_read() and snd_soc_component_read32()" from Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com>: Merge series "ASoC: add dailink .exit() callback" from Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com>: Merge series "ASoC: Intel: machine driver updates for 5.9" from Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com>: Merge branch 'for-5.8' of https://git.kernel.org/.../broonie/sound into asoc-5.9 Merge series "ASoC: ti: Add support for audio on J721e EVM" from Peter Ujfalusi <peter.ujfalusi(a)ti.com>: Merge series "ASoC: Intel: atom: fix kernel-doc and W=1 warnings" from Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com>: Merge series "ASoC: codecs: wm/cs: fix kernel-doc and W=1 warnings" from Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com>: Merge series "ASoC: rt5670: 2 small cleanups" from Hans de Goede <hdegoede(a)redhat.com>: Merge series "ASoC: fsl: fix kernel-doc" from Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com>: Merge series "ASoC: amd: remove warnings with make W=1" from Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com>: Merge series "ASoC: qdsp6: add gapless compressed audio support" from Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org>: Merge series "ASoC: topology: fix error handling flow" from Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com>: Merge series "ASoC: mediatek: mt8183: support machine driver for rt1015" from Tzung-Bi Shih <tzungbi(a)google.com>: Merge series "ASoC: Clean-up W=1 build warnings - part1" from Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com>: Merge series "ASoC: codecs: add MAX98373 Soundwire driver" from Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com>: Merge series "ASoC: mediatek: mt8183-da7219: support machine driver for rt1015" from Tzung-Bi Shih <tzungbi(a)google.com>: Merge series "ASoC: fsl_spdif: Clear the validity bit for TX" from Shengjiu Wang <shengjiu.wang(a)nxp.com>: Merge series "ASoC: Clean-up W=1 build warnings - part2" from Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com>: Merge series "ASoC: mediatek: mt8183: support DP audio" from Tzung-Bi Shih <tzungbi(a)google.com>: Merge series "ASoC: sh: remove discriminatory terms" from Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com>: Merge series "ASoC: merge .digital_mute() into .mute_stream()" from Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com>: Merge series "ASoC: fsl-asoc-card: Support hp and mic detection" from Shengjiu Wang <shengjiu.wang(a)nxp.com>: Merge series "ALSA: hda: export snd_hda_codec_cleanup()" from Kai Vehmanen <kai.vehmanen(a)linux.intel.com>: Merge series "Add support for SATA/PCIe/USB2[3]/VIN/CSI on R8A774E1" from Lad Prabhakar <prabhakar.mahadev-lad.rj(a)bp.renesas.com>: Merge series "ASoC: merge .digital_mute() into .mute_stream()" from Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com>: Merge series "ASoC: Intel: machine driver updates for 5.9" from Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com>: Merge series "ASoC: Intel: machine driver updates for 5.9" from Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com>: Merge series "Add ASoC AHUB components for Tegra210 and later" from Sameer Pujar <spujar(a)nvidia.com>: Merge series "ASoC: Intel: machine driver updates for 5.9" from Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com>: Merge series "Add ASoC AHUB components for Tegra210 and later" from Sameer Pujar <spujar(a)nvidia.com>: Merge series "SOF IMX fixes" from Daniel Baluta <daniel.baluta(a)oss.nxp.com> Daniel Baluta <daniel.baluta(a)nxp.com>: Merge series "ASoC: mediatek: mt8183: support machine driver for max98357b" from Tzung-Bi Shih <tzungbi(a)google.com>: Merge series "ASoC: add asoc_substream_to_rtd() macro" from Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com>: Merge series "ASoC: qcom: Use qcom_snd_parse_of() for apq8016_sbc" from Stephan Gerhold <stephan(a)gerhold.net>: Merge series "ASoC: intel: use asoc_substream_to_rtd()" from Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com>: Merge series "SOF Fixes for S0iX suspend/resume sequence" from Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com>: ASoC: ak4613: Undo conversion to YAML bindings Merge series "ASoC: meson: tdm fixes" from Jerome Brunet <jbrunet(a)baylibre.com>: Merge series "ASoC: meson: tdm fixes" from Jerome Brunet <jbrunet(a)baylibre.com>: Merge series "ASoC: Intel: KMB: TDM Enablement patches" from Michael Sit Wei Hong <michael.wei.hong.sit(a)intel.com>: Merge series "drop unnecessary list_empty" from Julia Lawall <Julia.Lawall(a)inria.fr>: Merge series "ASoC: core: Two step component registration" from Cezary Rojewski <cezary.rojewski(a)intel.com>: Merge remote-tracking branch 'asoc/for-5.8' into asoc-linus Merge remote-tracking branch 'asoc/for-5.9' into asoc-next Masahiro Yamada (2): ASoC: Convert UniPhier EVEA codec to json-schema ASoC: Convert UniPhier AIO audio system to json-schema Masanari Iida (1): ASoC: samsung: Fix a typo in Kconfig Michael Sit Wei Hong (2): ASoC: Intel: KMB: Add 8kHz audio support ASoC: Intel: KMB: Rework disable channel function Nathan Chancellor (1): ASoC: Intel: KeemBay: Fix header guard Naveen Manohar (2): ASoC: Intel: common: add match table for TGL MAX98373 + RT5682 SoundWire driver ASoC: Intel: sof_sdw: Add MAX98373 support Peter Ujfalusi (3): ASoC: ti: davinci-mcasp: Specify stream_name for playback/capture ASoC: dt-bindings: Add documentation for TI j721e EVM (CPB and IVI) ASoC: ti: Add custom machine driver for j721e EVM (CPB and IVI) Pierre-Louis Bossart (62): ASoC: soc-core: reduce verbosity of BE override message ASoC: soc-pcm: improve error messages in soc_pcm_new() ASoC: soc-pcm/compress: reduce verbosity on mapping ok messages ASoC: Intel: byt*: simplify card names for SOF uses ASoC: Intel: cht*: simplify card names for SOF uses ASoC: Intel: broadwell: simplify card names for SOF uses ASoC: soc-link: introduce exit() callback ASoC: Intel: bdw-rt5677: fix module load/unload issues ASoC: Intel: kbl-rt5660: use .exit() dailink callback to release gpiod ASoC: SOF: nocodec: add missing .owner field ASoC: Intel: cml_rt1011_rt5682: add missing .owner field ASoC: Intel: sof_sdw: add missing .owner field ASoC: Intel: bxt_rt298: add missing .owner field ASoC: Intel: sof_sdw: add quirk override with kernel parameter ASoC: Intel: boards: byt*.c: remove cast in dev_info quirk log ASoC: Intel: atom: fix kernel-doc ASoC: Intel: atom: fix 'defined but not used' warning ASoC: SOF: sof-acpi-dev: fix 'defined but unused' warning ASoC: codecs: cs4270: fix kernel-doc ASoC: codecs: cs42l42: remove always-true comparisons ASoC: codecs: wm8986: fix missing kernel-doc arguments ASoC: codecs: wm8960: fix kernel-doc ASoC: codecs: wm9713: remove spurious kernel-doc comment start ASoC: codecs: wm8994: fix kernel-doc ASoC: fsl: fsl_ssi_dbg: remove spurious kernel-doc comment start ASoC: fsl: fsl_ssi: fix kernel-doc ASoC: fsl: fsl-asoc-card: fix kernel-doc ASoC: fsl: fsl_spdif: fix kernel-doc ASoC: fsl: fsl_asrc: fix kernel-doc ASoC: fsl: fsl_esai: fix kernel-doc ASoC: soc-ac97: fix kernel-doc ASoC: fsl: fsl_ssi: fix kernel-doc ASoC: fsl: fsl_asrc: fix kernel-doc ASoC: codecs: wm8400: remove unused variables ASoC: amd: acp-da7219-max98357a: fix 'defined but not used' warning ASoC: amd: acp-rt5645: fix 'defined but not used' warning ASoC: amd: raven: acp3x-pcm-dma: fix 'set but not used' warning ASoC: amd: raven: acp3x-i2s: fix 'set but not used' warning ASoC: amd: raven: acp3x-pcm-dma: remove unused-but-set variable ASoC: amd: raven: acp3x-i2s: remove unused-but-set variable ASoC: topology: use break on errors, not continue ASoC: topology: factor kfree(se) in error handling ASoC: topology: add more logs when topology load fails. ASoC: codecs: max98373: split I2C and common parts ASoC: Intel: sof-sdw: add MAX98373 I2C dependencies ASoC: atmel: fix kernel-doc ASoC: samsung: spdif: fix kernel-doc ASoC: samsung: pcm: fix kernel-doc ASoC: tegra: tegra20_das: remove always-true comparison ASoC: uniphier: aio-core: fix kernel-doc ASoC: codecs: da7210: fix kernel-doc ASoC: codecs: da7219: fix 'defined but not used' warning ASoC: codecs: cros_ec_codec: fix 'defined but not used' warning ASoC: codecs: es8316: fix 'defined but not used' warning ASoC: codecs: max98390: fix 'defined but not used' warning ASoC: codecs: rt*: fix 'defined but not used' warning ASoC: sti: uniperif: fix 'defined by not used' warning ASoC: qcom: qdsp6: q6asm: Provide documentation for 'codec_profile' ASoC: sunxi: sun4i-i2s: add missing clock and format arguments in kernel-doc ASoC: codecs: rt5631: fix spurious kernel-doc start and missing arguments ASoC: Intel: sof_sdw_rt711: remove properties in card remove ASoC: core: use less strict tests for dailink capabilities Qiushi Wu (3): ASoC: img: Fix a reference count leak in img_i2s_in_set_fmt ASoC: img-parallel-out: Fix a reference count leak ASoC: tegra: Fix reference count leaks. Randy Dunlap (5): ASoC: Documentation: fix reference to renamed source file ASoC: tegra20_das.h: delete duplicated words ASoC: wcd9335.h: fix duplicated word ASoC: soc-dai.h: drop a duplicated word ASoC: ti: fix SND_SOC_J721E_EVM warnings & errors Ravulapati Vishnu vardhan rao (6): ASoC: amd: Removing unnecessary instance initialization ASoC: amd: Renaming snd-soc-card structure and fields ASoC: amd: Passing card structure based on codec ASoC: amd: Adding support for ALC1015 codec in machine driver ASoC: amd: Adding DAI LINK for rt1015 codec ASoC: amd: Added hw_params support for ALC1015 Rohit kumar (1): asoc: Update supported rate and format for dummy dai Ryan Lee (1): ASoC: codecs: max98373: add SoundWire support Sameer Pujar (8): ASoC: dt-bindings: tegra: Add DT bindings for Tegra210 ASoC: tegra: Add support for CIF programming ASoC: tegra: Add Tegra210 based DMIC driver ASoC: tegra: Add Tegra210 based I2S driver ASoC: tegra: Add Tegra210 based AHUB driver ASoC: tegra: Add Tegra186 based DSPK driver ASoC: tegra: Add Tegra210 based ADMAIF driver ASoC: tegra: Fix build error due to 64-by-32 division Sebastian Reichel (1): ASoC: da7213: add default clock handling Seung-Woo Kim (1): ASoC: bcm2835: Silence clk_get() error on -EPROBE_DEFER Shengjiu Wang (25): ASoC: fsl_easrc: Fix -Wmissing-prototypes warning ASoC: fsl_easrc: Fix -Wunused-but-set-variable ASoC: fsl_easrc: Fix "Function parameter not described" warnings ASoC: bindings: wm8960: Add property for headphone detection ASoC: wm8960: Support headphone jack detection function ASoC: bindings: fsl-asoc-card: Add compatible string for MQS ASoC: fsl-asoc-card: Add MQS support ASoC: bindings: fsl_spdif: Add new compatible string for imx6sx ASoC: fsl_spdif: Add support for imx6sx platform ASoC: fsl_spdif: Add pm runtime function ASoC: fsl_easrc: Fix uninitialized scalar variable in fsl_easrc_set_ctx_format ASoC: fsl-asoc-card: Add WM8524 support ASoC: bindings: fsl-asoc-card: Add compatible string for wm8524 ASoC: fsl_sai: Refine regcache usage with pm runtime ASoC: fsl_asrc: Add an option to select internal ratio mode ASoC: fsl_spdif: Clear the validity bit for TX ASoC: fsl_spdif: Add kctl for configuring TX validity bit ASoC: simple-card-utils: Support configure pin_name for asoc_simple_init_jack ASoC: bindings: fsl-asoc-card: Support hp-det-gpio and mic-det-gpio ASoC: fsl-asoc-card: Support Headphone and Microphone Jack detection ASoC: soc-component: Add missed return for calling soc_component_ret ASoC: fsl_esai: add IRQF_SHARED for devm_request_irq ASoC: fsl-asoc-card: Support configuring dai fmt from DT ASoC: bindings: fsl-asoc-card: Support properties for configuring dai fmt ASoC: fsl_sai: Fix value of FSL_SAI_CR1_RFW_MASK Shuming Fan (2): ASoC: rt5682: disable MICBIAS and Vref2 widget in default ASoC: rt5682: optimize the power consumption Sia Jee Heng (3): ASoC: Intel: Add KeemBay platform driver ASoC: Intel: Add makefiles and kconfig changes for KeemBay ASoC: Add documentation for KeemBay i2s Simon Shields (1): ASoC: samsung: Add sound support for Midas boards Srinivas Kandagatla (2): ASoC: q6asm: add command opcode to timeout error report ASoC: qdsp6: use dev_err instead of pr_err Stephan Gerhold (8): ASoC: qcom: Use devm for resource management ASoC: qcom: common: Use snd_soc_dai_link_set_capabilities() ASoC: q6afe: Remove unused q6afe_is_rx_port() function ASoC: qcom: common: Support parsing links without DPCM ASoC: qcom: common: Parse properties with "qcom," prefix ASoC: qcom: apq8016_sbc: Use qcom_snd_parse_of() ASoC: qcom: common: Avoid printing errors for -EPROBE_DEFER ASoC: dt-bindings: q6asm: Add Q6ASM_DAI_{TX_RX, TX, RX} defines Steve Lee (5): dt-bindings: Added device tree binding for max98390 ASoC: max98390: Add Amp init common setting func. ASoC: max98390: Fix dac event dapm mixer. ASoC: max98390: Update dsm init sequence and condition. ASoC: max98390: update dsm param bin max size Sugar Zhang (1): ASoC: rockchip: spdif: Handle clk by pm runtime Sylwester Nawrocki (1): ASoC: samsung: Document DT bindings for Midas sound subsystem Tang Bin (2): ASoC: qcom: qdsp6: Use the defined variable to clean code ASoC: qcom: qdsp6: Use IS_ERR() instead of IS_ERR_OR_NULL() Thierry Reding (1): ASoC: sgtl5000: Convert to json-schema Tzung-Bi Shih (25): ASoC: max98357a: add compatible string for MAX98360A ASoC: dt-bindings: add compatible string for MAX98360A ASoC: mediatek: mt8183-da7219: set playback and capture constraints ASoC: core: move definition of enum snd_soc_bias_level ASoC: dapm: declare missing structure prototypes ASoC: rt1015: add missing header inclusion ASoC: mediatek: mt8183: sort header inclusions in alphabetical ASoC: mediatek: mt8183: support machine driver with rt1015 ASoC: mt8183: add compatible string for using rt1015 ASoC: mediatek: mt8183-da7219: sort header inclusions in alphabetical ASoC: mediatek: mt8183-da7219: remove forward declaration of headset_init ASoC: mediatek: mt8183-da7219: extract codec and DAI names ASoC: mediatek: mt8183-da7219: add compatible string for using rt1015 ASoC: mediatek: mt8183-da7219: support machine driver with rt1015 ASoC: dt-bindings: mt8183: add a property "mediatek, hdmi-codec" ASoC: mediatek: mt8183: use hdmi-codec ASoC: mediatek: mt8183: support HDMI jack reporting ASoC: dt-bindings: mt8183-da7219: add a property "mediatek, hdmi-codec" ASoC: mediatek: mt8183-da7219: use hdmi-codec ASoC: mediatek: mt8183-da7219: support HDMI jack reporting ASoC: max98357a: move control of SD_MODE back to DAI ops ASoC: mediatek: mt8183: support left justified format for I2S ASoC: dt-bindings: mt8183: add compatible string for using max98357b ASoC: mediatek: mt8183: support machine driver with max98357b ASoC: dapm: don't call pm_runtime_* on card device Vijendar Mukunda (3): ASoC: amd: add logic to check dmic hardware runtime ASoC: amd: add ACPI dependency check ASoC: amd: fixed kernel warnings Vlad Karpovich (1): ASoC: wm_adsp: Add controls for calibration and diagnostic FW Wei Yongjun (1): ASoC: ti: j721e-evm: Fix missing unlock on error in j721e_audio_hw_params() Yong Zhi (2): ASoC: intel: sof_rt5682: Add support for jsl-max98360a-rt5682 ASoC: intel: board: sof_rt5682: Update rt1015 pll input clk freq derek.fang (3): ASoC: rl6231: Add new supports on rl6231 ASoC: rt5682: Let PLL2 support the freq conversion for 44100Hz sample rate ASoC: rt5682: DAI wclk supports 44100 Hz output randerwang (1): ASoC: Intel: sdw_max98373: add card_late_probe support .../devicetree/bindings/sound/adi,adau1977.txt | 6 +- Documentation/devicetree/bindings/sound/ak4613.txt | 27 - .../devicetree/bindings/sound/ak4613.yaml | 49 ++ Documentation/devicetree/bindings/sound/ak4642.txt | 37 - .../devicetree/bindings/sound/ak4642.yaml | 58 ++ .../devicetree/bindings/sound/everest,es8316.txt | 23 - .../devicetree/bindings/sound/everest,es8316.yaml | 50 ++ .../devicetree/bindings/sound/fsl,spdif.txt | 6 +- .../devicetree/bindings/sound/fsl-asoc-card.txt | 20 +- .../bindings/sound/intel,keembay-i2s.yaml | 70 ++ .../devicetree/bindings/sound/max98357a.txt | 12 +- .../devicetree/bindings/sound/maxim,max98390.yaml | 51 ++ Documentation/devicetree/bindings/sound/mt6358.txt | 6 + .../bindings/sound/mt8183-da7219-max98357.txt | 9 +- .../sound/mt8183-mt6358-ts3a227-max98357.txt | 8 +- .../bindings/sound/nvidia,tegra186-dspk.yaml | 83 ++ .../bindings/sound/nvidia,tegra210-admaif.yaml | 111 +++ .../bindings/sound/nvidia,tegra210-ahub.yaml | 136 ++++ .../bindings/sound/nvidia,tegra210-dmic.yaml | 83 ++ .../bindings/sound/nvidia,tegra210-i2s.yaml | 101 +++ .../devicetree/bindings/sound/qcom,q6asm.txt | 9 +- .../devicetree/bindings/sound/renesas,fsi.yaml | 19 +- .../devicetree/bindings/sound/renesas,rsnd.txt | 1 + .../bindings/sound/rockchip,rk3328-codec.txt | 28 - .../bindings/sound/rockchip,rk3328-codec.yaml | 69 ++ .../devicetree/bindings/sound/rohm,bd28623.txt | 29 - .../devicetree/bindings/sound/rohm,bd28623.yaml | 67 ++ .../bindings/sound/samsung,aries-wm8994.yaml | 147 ++++ .../bindings/sound/samsung,midas-audio.yaml | 108 +++ .../devicetree/bindings/sound/sgtl5000.txt | 60 -- .../devicetree/bindings/sound/sgtl5000.yaml | 103 +++ .../bindings/sound/socionext,uniphier-aio.yaml | 81 ++ .../bindings/sound/socionext,uniphier-evea.yaml | 70 ++ .../devicetree/bindings/sound/tas2552.txt | 2 +- .../devicetree/bindings/sound/tas2562.txt | 7 +- .../devicetree/bindings/sound/tas2562.yaml | 69 ++ .../devicetree/bindings/sound/tas2770.txt | 37 - .../devicetree/bindings/sound/tas2770.yaml | 76 ++ .../devicetree/bindings/sound/tas5720.txt | 6 +- .../bindings/sound/ti,j721e-cpb-audio.yaml | 95 +++ .../bindings/sound/ti,j721e-cpb-ivi-audio.yaml | 150 ++++ .../devicetree/bindings/sound/ti,tas6424.txt | 2 +- .../devicetree/bindings/sound/tlv320adcx140.yaml | 34 +- .../devicetree/bindings/sound/uniphier,aio.txt | 45 -- .../devicetree/bindings/sound/uniphier,evea.txt | 26 - Documentation/devicetree/bindings/sound/wm8960.txt | 11 + Documentation/devicetree/bindings/sound/wm8994.txt | 23 + .../devicetree/bindings/trivial-devices.yaml | 2 - .../devicetree/bindings/vendor-prefixes.yaml | 2 +- Documentation/sound/kernel-api/alsa-driver-api.rst | 2 +- Documentation/sound/soc/dai.rst | 2 +- arch/arm/boot/dts/motorola-mapphone-common.dtsi | 4 +- drivers/gpu/drm/bridge/sii902x.c | 7 +- drivers/gpu/drm/exynos/exynos_hdmi.c | 6 +- drivers/gpu/drm/i2c/tda998x_drv.c | 7 +- drivers/gpu/drm/mediatek/mtk_hdmi.c | 6 +- drivers/gpu/drm/rockchip/cdn-dp-core.c | 7 +- drivers/gpu/drm/sti/sti_hdmi.c | 6 +- drivers/gpu/drm/zte/zx_hdmi.c | 7 +- drivers/spi/spi-lantiq-ssc.c | 2 +- include/dt-bindings/sound/qcom,q6asm.h | 4 + include/sound/hda_codec.h | 2 + include/sound/hdmi-codec.h | 6 +- include/sound/rt5670.h | 26 - include/sound/simple_card_utils.h | 6 +- include/sound/soc-component.h | 30 +- include/sound/soc-dai.h | 14 +- include/sound/soc-dapm.h | 20 + include/sound/soc-link.h | 1 + include/sound/soc.h | 34 +- include/sound/wm8960.h | 17 + sound/pci/hda/hda_codec.c | 3 +- sound/soc/Makefile | 2 +- sound/soc/amd/Kconfig | 1 + sound/soc/amd/acp-da7219-max98357a.c | 14 +- sound/soc/amd/acp-pcm-dma.c | 2 +- sound/soc/amd/acp-rt5645.c | 4 +- sound/soc/amd/acp3x-rt5682-max9836.c | 236 ++++-- sound/soc/amd/raven/acp3x-i2s.c | 14 +- sound/soc/amd/raven/acp3x-pcm-dma.c | 12 +- sound/soc/amd/raven/pci-acp3x.c | 21 +- sound/soc/amd/renoir/rn-pci-acp3x.c | 33 + sound/soc/amd/renoir/rn_acp3x.h | 2 + sound/soc/atmel/atmel-classd.c | 141 ++-- sound/soc/atmel/atmel-pcm-dma.c | 6 +- sound/soc/atmel/atmel-pcm-pdc.c | 2 +- sound/soc/atmel/atmel-pdmic.c | 124 ++- sound/soc/atmel/atmel_ssc_dai.c | 1 + sound/soc/atmel/atmel_wm8904.c | 2 +- sound/soc/au1x/db1200.c | 2 +- sound/soc/au1x/dbdma2.c | 2 +- sound/soc/au1x/dma.c | 2 +- sound/soc/bcm/bcm2835-i2s.c | 9 +- sound/soc/bcm/bcm63xx-pcm-whistler.c | 12 +- sound/soc/bcm/cygnus-pcm.c | 16 +- sound/soc/cirrus/edb93xx.c | 2 +- sound/soc/cirrus/ep93xx-ac97.c | 2 +- sound/soc/cirrus/snappercl15.c | 2 +- sound/soc/codecs/88pm860x-codec.c | 22 +- sound/soc/codecs/Kconfig | 20 +- sound/soc/codecs/Makefile | 4 + sound/soc/codecs/ab8500-codec.c | 10 +- sound/soc/codecs/ad193x.c | 5 +- sound/soc/codecs/ad1980.c | 4 +- sound/soc/codecs/adau1701.c | 5 +- sound/soc/codecs/adau1761.c | 4 +- sound/soc/codecs/adau17x1.c | 4 +- sound/soc/codecs/adav80x.c | 2 +- sound/soc/codecs/ak4458.c | 13 +- sound/soc/codecs/ak4535.c | 10 +- sound/soc/codecs/ak4613.c | 10 +- sound/soc/codecs/ak4641.c | 8 +- sound/soc/codecs/ak4671.c | 8 +- sound/soc/codecs/alc5623.c | 11 +- sound/soc/codecs/alc5632.c | 11 +- sound/soc/codecs/arizona.c | 18 +- sound/soc/codecs/cpcap.c | 11 +- sound/soc/codecs/cq93vc.c | 5 +- sound/soc/codecs/cros_ec_codec.c | 2 + sound/soc/codecs/cs4265.c | 5 +- sound/soc/codecs/cs4270.c | 19 +- sound/soc/codecs/cs42l42.c | 16 +- sound/soc/codecs/cs42l51.c | 13 +- sound/soc/codecs/cs42l52.c | 5 +- sound/soc/codecs/cs42l56.c | 5 +- sound/soc/codecs/cs42l73.c | 4 +- sound/soc/codecs/cs42xx8.c | 5 +- sound/soc/codecs/cs4341.c | 5 +- sound/soc/codecs/cs4349.c | 5 +- sound/soc/codecs/cs47l15.c | 36 +- sound/soc/codecs/cs47l35.c | 58 +- sound/soc/codecs/cs47l85.c | 102 ++- sound/soc/codecs/cs47l90.c | 92 +-- sound/soc/codecs/cs47l92.c | 96 +-- sound/soc/codecs/da7210.c | 41 +- sound/soc/codecs/da7213.c | 112 ++- sound/soc/codecs/da7213.h | 2 + sound/soc/codecs/da7218.c | 34 +- sound/soc/codecs/da7219-aad.c | 16 +- sound/soc/codecs/da7219.c | 22 +- sound/soc/codecs/da732x.c | 18 +- sound/soc/codecs/da9055.c | 19 +- sound/soc/codecs/es8316.c | 7 +- sound/soc/codecs/es8328.c | 9 +- sound/soc/codecs/hdac_hda.c | 30 +- sound/soc/codecs/hdmi-codec.c | 27 +- sound/soc/codecs/inno_rk3036.c | 6 +- sound/soc/codecs/isabelle.c | 15 +- sound/soc/codecs/jz4770.c | 6 +- sound/soc/codecs/lm49453.c | 25 +- sound/soc/codecs/madera.c | 49 +- sound/soc/codecs/max98088.c | 24 +- sound/soc/codecs/max98090.c | 26 +- sound/soc/codecs/max98095.c | 16 +- sound/soc/codecs/max98357a.c | 51 +- sound/soc/codecs/max98373-i2c.c | 612 ++++++++++++++ sound/soc/codecs/max98373-sdw.c | 887 ++++++++++++++++++++ sound/soc/codecs/max98373-sdw.h | 72 ++ sound/soc/codecs/max98373.c | 611 +------------- sound/soc/codecs/max98373.h | 17 +- sound/soc/codecs/max98390.c | 40 +- sound/soc/codecs/max98390.h | 2 +- sound/soc/codecs/max9850.c | 4 +- sound/soc/codecs/max9860.c | 2 +- sound/soc/codecs/max9867.c | 5 +- sound/soc/codecs/mc13783.c | 2 +- sound/soc/codecs/ml26124.c | 5 +- sound/soc/codecs/msm8916-wcd-analog.c | 16 +- sound/soc/codecs/msm8916-wcd-digital.c | 16 +- sound/soc/codecs/mt6358.c | 23 +- sound/soc/codecs/nau8822.c | 11 +- sound/soc/codecs/pcm1681.c | 5 +- sound/soc/codecs/pcm1789.c | 5 +- sound/soc/codecs/pcm179x.c | 5 +- sound/soc/codecs/pcm186x-i2c.c | 2 +- sound/soc/codecs/pcm186x-spi.c | 2 +- sound/soc/codecs/pcm186x.c | 2 +- sound/soc/codecs/pcm186x.h | 2 +- sound/soc/codecs/pcm3168a.c | 5 +- sound/soc/codecs/pcm512x.c | 5 +- sound/soc/codecs/rk3328_codec.c | 5 +- sound/soc/codecs/rl6231.c | 2 + sound/soc/codecs/rt1011.c | 20 +- sound/soc/codecs/rt1015.c | 35 +- sound/soc/codecs/rt1015.h | 5 + sound/soc/codecs/rt1305.c | 2 +- sound/soc/codecs/rt274.c | 6 +- sound/soc/codecs/rt286.c | 2 + sound/soc/codecs/rt298.c | 4 +- sound/soc/codecs/rt5616.c | 2 +- sound/soc/codecs/rt5631.c | 40 +- sound/soc/codecs/rt5640.c | 14 +- sound/soc/codecs/rt5645.c | 16 +- sound/soc/codecs/rt5651.c | 6 +- sound/soc/codecs/rt5659.c | 51 +- sound/soc/codecs/rt5660.c | 4 +- sound/soc/codecs/rt5663.c | 34 +- sound/soc/codecs/rt5665.c | 16 +- sound/soc/codecs/rt5668.c | 16 +- sound/soc/codecs/rt5670.c | 93 +-- sound/soc/codecs/rt5670.h | 16 +- sound/soc/codecs/rt5677-spi.c | 6 +- sound/soc/codecs/rt5677.c | 2 +- sound/soc/codecs/rt5682-i2c.c | 4 +- sound/soc/codecs/rt5682-sdw.c | 2 +- sound/soc/codecs/rt5682.c | 93 ++- sound/soc/codecs/rt5682.h | 4 + sound/soc/codecs/sgtl5000.c | 21 +- sound/soc/codecs/ssm2518.c | 5 +- sound/soc/codecs/ssm2602.c | 5 +- sound/soc/codecs/ssm4567.c | 5 +- sound/soc/codecs/sta32x.c | 6 +- sound/soc/codecs/sta350.c | 2 +- sound/soc/codecs/sta529.c | 5 +- sound/soc/codecs/tas2552.c | 13 +- sound/soc/codecs/tas2552.h | 2 +- sound/soc/codecs/tas2562.c | 166 +++- sound/soc/codecs/tas2562.h | 7 +- sound/soc/codecs/tas2770.c | 10 +- sound/soc/codecs/tas2770.h | 2 +- sound/soc/codecs/tas571x.c | 5 +- sound/soc/codecs/tas5720.c | 11 +- sound/soc/codecs/tas5720.h | 2 +- sound/soc/codecs/tas6424.c | 7 +- sound/soc/codecs/tas6424.h | 2 +- sound/soc/codecs/tda7419.c | 9 +- sound/soc/codecs/tfa9879.c | 5 +- sound/soc/codecs/tlv320adcx140.c | 124 ++- sound/soc/codecs/tlv320adcx140.h | 16 +- sound/soc/codecs/tlv320aic23.c | 21 +- sound/soc/codecs/tlv320aic26.c | 11 +- sound/soc/codecs/tlv320aic31xx.c | 13 +- sound/soc/codecs/tlv320aic31xx.h | 2 +- sound/soc/codecs/tlv320aic32x4.c | 21 +- sound/soc/codecs/tlv320aic3x.c | 19 +- sound/soc/codecs/tpa6130a2.c | 2 +- sound/soc/codecs/tscs42xx.c | 4 +- sound/soc/codecs/tscs454.c | 24 +- sound/soc/codecs/twl6040.c | 5 +- sound/soc/codecs/uda134x.c | 5 +- sound/soc/codecs/wcd-clsh-v2.c | 2 +- sound/soc/codecs/wcd9335.c | 48 +- sound/soc/codecs/wcd9335.h | 6 +- sound/soc/codecs/wcd934x.c | 52 +- sound/soc/codecs/wm0010.c | 4 +- sound/soc/codecs/wm2200.c | 4 +- sound/soc/codecs/wm5100.c | 18 +- sound/soc/codecs/wm5110.c | 6 +- sound/soc/codecs/wm8350.c | 37 +- sound/soc/codecs/wm8400.c | 67 +- sound/soc/codecs/wm8510.c | 33 +- sound/soc/codecs/wm8523.c | 6 +- sound/soc/codecs/wm8580.c | 17 +- sound/soc/codecs/wm8711.c | 13 +- sound/soc/codecs/wm8728.c | 15 +- sound/soc/codecs/wm8731.c | 11 +- sound/soc/codecs/wm8741.c | 5 +- sound/soc/codecs/wm8750.c | 13 +- sound/soc/codecs/wm8753.c | 56 +- sound/soc/codecs/wm8770.c | 7 +- sound/soc/codecs/wm8776.c | 7 +- sound/soc/codecs/wm8804.c | 2 +- sound/soc/codecs/wm8900.c | 27 +- sound/soc/codecs/wm8903.c | 27 +- sound/soc/codecs/wm8904.c | 25 +- sound/soc/codecs/wm8940.c | 37 +- sound/soc/codecs/wm8955.c | 9 +- sound/soc/codecs/wm8958-dsp2.c | 18 +- sound/soc/codecs/wm8960.c | 49 +- sound/soc/codecs/wm8961.c | 65 +- sound/soc/codecs/wm8962.c | 49 +- sound/soc/codecs/wm8971.c | 13 +- sound/soc/codecs/wm8974.c | 29 +- sound/soc/codecs/wm8978.c | 17 +- sound/soc/codecs/wm8983.c | 15 +- sound/soc/codecs/wm8985.c | 15 +- sound/soc/codecs/wm8988.c | 17 +- sound/soc/codecs/wm8990.c | 23 +- sound/soc/codecs/wm8991.c | 45 +- sound/soc/codecs/wm8993.c | 37 +- sound/soc/codecs/wm8994.c | 77 +- sound/soc/codecs/wm8995.c | 26 +- sound/soc/codecs/wm8996.c | 35 +- sound/soc/codecs/wm8998.c | 8 +- sound/soc/codecs/wm9081.c | 43 +- sound/soc/codecs/wm9090.c | 4 +- sound/soc/codecs/wm9713.c | 4 +- sound/soc/codecs/wm_adsp.c | 11 +- sound/soc/codecs/wm_hubs.c | 30 +- sound/soc/codecs/wmfw.h | 1 + sound/soc/dwc/dwc-pcm.c | 2 +- sound/soc/fsl/Kconfig | 1 + sound/soc/fsl/eukrea-tlv320.c | 2 +- sound/soc/fsl/fsl-asoc-card.c | 216 ++++- sound/soc/fsl/fsl_asrc.c | 103 ++- sound/soc/fsl/fsl_asrc_dma.c | 4 +- sound/soc/fsl/fsl_audmix.c | 10 +- sound/soc/fsl/fsl_dma.c | 2 +- sound/soc/fsl/fsl_easrc.c | 49 +- sound/soc/fsl/fsl_esai.c | 34 +- sound/soc/fsl/fsl_sai.c | 8 +- sound/soc/fsl/fsl_sai.h | 2 +- sound/soc/fsl/fsl_spdif.c | 233 ++++-- sound/soc/fsl/fsl_ssi.c | 78 +- sound/soc/fsl/fsl_ssi_dbg.c | 4 +- sound/soc/fsl/imx-audmix.c | 10 +- sound/soc/fsl/imx-audmux.c | 2 +- sound/soc/fsl/imx-mc13783.c | 2 +- sound/soc/fsl/mpc5200_dma.c | 8 +- sound/soc/fsl/mpc5200_psc_i2s.c | 2 +- sound/soc/fsl/mpc8610_hpcd.c | 4 +- sound/soc/fsl/mx27vis-aic32x4.c | 2 +- sound/soc/fsl/p1022_ds.c | 2 +- sound/soc/fsl/p1022_rdk.c | 2 +- sound/soc/fsl/wm1133-ev1.c | 2 +- sound/soc/generic/simple-card-utils.c | 13 +- sound/soc/img/img-i2s-in.c | 4 +- sound/soc/img/img-parallel-out.c | 4 +- sound/soc/intel/Kconfig | 7 + sound/soc/intel/Makefile | 1 + sound/soc/intel/atom/sst-atom-controls.c | 65 +- sound/soc/intel/atom/sst-mfld-platform-pcm.c | 6 +- sound/soc/intel/atom/sst/sst_loader.c | 14 +- sound/soc/intel/atom/sst/sst_stream.c | 43 +- sound/soc/intel/baytrail/sst-baytrail-pcm.c | 16 +- sound/soc/intel/boards/Kconfig | 15 +- sound/soc/intel/boards/Makefile | 2 + sound/soc/intel/boards/bdw-rt5650.c | 14 +- sound/soc/intel/boards/bdw-rt5677.c | 34 +- sound/soc/intel/boards/broadwell.c | 14 +- sound/soc/intel/boards/bxt_da7219_max98357a.c | 117 ++- sound/soc/intel/boards/bxt_rt298.c | 4 +- sound/soc/intel/boards/byt-rt5640.c | 2 +- sound/soc/intel/boards/bytcht_cx2072x.c | 12 +- sound/soc/intel/boards/bytcht_da7213.c | 16 +- sound/soc/intel/boards/bytcht_es8316.c | 17 +- sound/soc/intel/boards/bytcr_rt5640.c | 18 +- sound/soc/intel/boards/bytcr_rt5651.c | 18 +- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 14 +- sound/soc/intel/boards/cht_bsw_nau8824.c | 14 +- sound/soc/intel/boards/cht_bsw_rt5645.c | 19 +- sound/soc/intel/boards/cht_bsw_rt5672.c | 32 +- sound/soc/intel/boards/cml_rt1011_rt5682.c | 102 +-- sound/soc/intel/boards/ehl_rt5660.c | 2 +- sound/soc/intel/boards/glk_rt5682_max98357a.c | 2 +- sound/soc/intel/boards/haswell.c | 2 +- sound/soc/intel/boards/kbl_da7219_max98927.c | 8 +- sound/soc/intel/boards/kbl_rt5660.c | 19 +- sound/soc/intel/boards/kbl_rt5663_max98927.c | 4 +- .../soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 45 +- sound/soc/intel/boards/skl_hda_dsp_common.h | 1 + sound/soc/intel/boards/skl_hda_dsp_generic.c | 17 +- sound/soc/intel/boards/skl_nau88l25_max98357a.c | 2 +- sound/soc/intel/boards/skl_nau88l25_ssm4567.c | 2 +- sound/soc/intel/boards/skl_rt286.c | 2 +- sound/soc/intel/boards/sof_da7219_max98373.c | 2 +- sound/soc/intel/boards/sof_maxim_common.c | 57 +- sound/soc/intel/boards/sof_maxim_common.h | 3 + sound/soc/intel/boards/sof_pcm512x.c | 4 +- sound/soc/intel/boards/sof_rt5682.c | 69 +- sound/soc/intel/boards/sof_sdw.c | 103 ++- sound/soc/intel/boards/sof_sdw_common.h | 17 + sound/soc/intel/boards/sof_sdw_hdmi.c | 6 + sound/soc/intel/boards/sof_sdw_max98373.c | 86 ++ sound/soc/intel/boards/sof_sdw_rt1308.c | 2 +- sound/soc/intel/boards/sof_sdw_rt711.c | 17 +- sound/soc/intel/boards/sof_wm8804.c | 2 +- sound/soc/intel/common/soc-acpi-intel-cml-match.c | 13 + sound/soc/intel/common/soc-acpi-intel-ehl-match.c | 2 +- sound/soc/intel/common/soc-acpi-intel-jsl-match.c | 13 + sound/soc/intel/common/soc-acpi-intel-tgl-match.c | 25 + sound/soc/intel/haswell/sst-haswell-pcm.c | 12 +- sound/soc/intel/keembay/Makefile | 4 + sound/soc/intel/keembay/kmb_platform.c | 668 +++++++++++++++ sound/soc/intel/keembay/kmb_platform.h | 146 ++++ sound/soc/intel/skylake/skl-pcm.c | 16 +- sound/soc/intel/skylake/skl-topology.c | 5 +- sound/soc/intel/skylake/skl-topology.h | 2 +- sound/soc/kirkwood/armada-370-db.c | 2 +- sound/soc/mediatek/Kconfig | 12 +- sound/soc/mediatek/common/mtk-afe-fe-dai.c | 12 +- .../soc/mediatek/common/mtk-afe-platform-driver.c | 2 +- sound/soc/mediatek/mt2701/mt2701-afe-pcm.c | 2 +- sound/soc/mediatek/mt2701/mt2701-cs42448.c | 2 +- sound/soc/mediatek/mt2701/mt2701-wm8960.c | 2 +- sound/soc/mediatek/mt6797/mt6797-afe-pcm.c | 4 +- sound/soc/mediatek/mt8173/mt8173-afe-pcm.c | 2 +- sound/soc/mediatek/mt8173/mt8173-max98090.c | 2 +- sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c | 2 +- sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c | 2 +- sound/soc/mediatek/mt8173/mt8173-rt5650.c | 2 +- sound/soc/mediatek/mt8183/mt8183-afe-pcm.c | 4 +- sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c | 321 ++++++-- sound/soc/mediatek/mt8183/mt8183-dai-i2s.c | 59 +- .../mt8183/mt8183-mt6358-ts3a227-max98357.c | 230 +++++- sound/soc/meson/Kconfig | 1 + sound/soc/meson/aiu-encoder-i2s.c | 3 +- sound/soc/meson/aiu-fifo-i2s.c | 3 +- sound/soc/meson/aiu-fifo.c | 3 +- sound/soc/meson/axg-card.c | 22 +- sound/soc/meson/axg-spdifout.c | 5 +- sound/soc/meson/axg-tdm-formatter.c | 11 +- sound/soc/meson/axg-tdm-formatter.h | 1 - sound/soc/meson/axg-tdm-interface.c | 26 +- sound/soc/meson/axg-tdmin.c | 16 +- sound/soc/meson/axg-tdmout.c | 3 - sound/soc/meson/gx-card.c | 20 +- sound/soc/meson/meson-card-utils.c | 8 +- sound/soc/meson/meson-codec-glue.c | 2 +- sound/soc/mxs/mxs-sgtl5000.c | 2 +- sound/soc/pxa/brownstone.c | 2 +- sound/soc/pxa/corgi.c | 4 +- sound/soc/pxa/hx4700.c | 2 +- sound/soc/pxa/imote2.c | 2 +- sound/soc/pxa/magician.c | 6 +- sound/soc/pxa/mioa701_wm9713.c | 8 +- sound/soc/pxa/mmp-pcm.c | 2 +- sound/soc/pxa/poodle.c | 4 +- sound/soc/pxa/pxa-ssp.c | 2 +- sound/soc/pxa/pxa2xx-i2s.c | 2 +- sound/soc/pxa/spitz.c | 4 +- sound/soc/pxa/tosa.c | 2 +- sound/soc/pxa/z2.c | 2 +- sound/soc/pxa/zylonite.c | 2 +- sound/soc/qcom/Kconfig | 5 +- sound/soc/qcom/apq8016_sbc.c | 120 +-- sound/soc/qcom/apq8096.c | 30 +- sound/soc/qcom/common.c | 58 +- sound/soc/qcom/lpass-platform.c | 14 +- sound/soc/qcom/qdsp6/q6adm.c | 7 +- sound/soc/qcom/qdsp6/q6afe.c | 8 - sound/soc/qcom/qdsp6/q6afe.h | 1 - sound/soc/qcom/qdsp6/q6asm-dai.c | 36 +- sound/soc/qcom/qdsp6/q6asm.c | 6 +- sound/soc/qcom/qdsp6/q6routing.c | 2 +- sound/soc/qcom/sdm845.c | 54 +- sound/soc/qcom/storm.c | 2 +- sound/soc/rockchip/rk3288_hdmi_analog.c | 2 +- sound/soc/rockchip/rk3399_gru_sound.c | 25 +- sound/soc/rockchip/rockchip_i2s.c | 2 +- sound/soc/rockchip/rockchip_max98090.c | 2 +- sound/soc/rockchip/rockchip_rt5645.c | 2 +- sound/soc/rockchip/rockchip_spdif.c | 59 +- sound/soc/samsung/Kconfig | 23 +- sound/soc/samsung/Makefile | 4 + sound/soc/samsung/aries_wm8994.c | 695 ++++++++++++++++ sound/soc/samsung/arndale.c | 4 +- sound/soc/samsung/h1940_uda1380.c | 2 +- sound/soc/samsung/i2s.c | 2 +- sound/soc/samsung/jive_wm8750.c | 2 +- sound/soc/samsung/littlemill.c | 2 +- sound/soc/samsung/midas_wm1811.c | 543 +++++++++++++ sound/soc/samsung/neo1973_wm8753.c | 8 +- sound/soc/samsung/odroid.c | 6 +- sound/soc/samsung/pcm.c | 9 +- sound/soc/samsung/rx1950_uda1380.c | 2 +- sound/soc/samsung/s3c-i2s-v2.c | 2 +- sound/soc/samsung/s3c24xx_simtec.c | 2 +- sound/soc/samsung/s3c24xx_uda134x.c | 6 +- sound/soc/samsung/smartq_wm8987.c | 2 +- sound/soc/samsung/smdk_spdif.c | 2 +- sound/soc/samsung/smdk_wm8580.c | 2 +- sound/soc/samsung/smdk_wm8994.c | 2 +- sound/soc/samsung/smdk_wm8994pcm.c | 2 +- sound/soc/samsung/snow.c | 2 +- sound/soc/samsung/spdif.c | 12 +- sound/soc/samsung/tm2_wm5110.c | 8 +- sound/soc/sh/Kconfig | 2 +- sound/soc/sh/dma-sh7760.c | 12 +- sound/soc/sh/fsi.c | 6 +- sound/soc/sh/migor.c | 4 +- sound/soc/sh/rcar/core.c | 8 +- sound/soc/sh/rcar/rsnd.h | 2 +- sound/soc/sh/rcar/ssi.c | 28 +- sound/soc/sh/rcar/ssiu.c | 6 +- sound/soc/sh/siu_pcm.c | 6 +- sound/soc/sh/ssi.c | 2 +- sound/soc/soc-ac97.c | 9 +- sound/soc/soc-component.c | 670 ++++++++++----- sound/soc/soc-compress.c | 4 +- sound/soc/soc-core.c | 158 ++-- sound/soc/soc-dai.c | 36 +- sound/soc/soc-dapm.c | 41 +- sound/soc/soc-generic-dmaengine-pcm.c | 18 +- sound/soc/soc-io.c | 202 ----- sound/soc/soc-link.c | 18 +- sound/soc/soc-ops.c | 43 +- sound/soc/soc-pcm.c | 232 +++--- sound/soc/soc-topology.c | 73 +- sound/soc/soc-utils.c | 5 +- sound/soc/sof/imx/imx8.c | 24 +- sound/soc/sof/imx/imx8m.c | 7 +- sound/soc/sof/intel/hda-dai.c | 10 +- sound/soc/sof/intel/hda-dsp.c | 50 +- sound/soc/sof/intel/hda-pcm.c | 2 +- sound/soc/sof/nocodec.c | 1 + sound/soc/sof/pcm.c | 26 +- sound/soc/sof/sof-acpi-dev.c | 8 +- sound/soc/sof/topology.c | 2 + sound/soc/spear/spdif_out.c | 8 +- sound/soc/sprd/sprd-pcm-dma.c | 2 +- sound/soc/sti/uniperif.h | 2 +- sound/soc/stm/stm32_adfsdm.c | 21 +- sound/soc/stm/stm32_sai_sub.c | 2 +- sound/soc/sunxi/sun4i-codec.c | 12 +- sound/soc/sunxi/sun4i-i2s.c | 10 +- sound/soc/sunxi/sun4i-spdif.c | 4 +- sound/soc/tegra/Kconfig | 56 ++ sound/soc/tegra/Makefile | 10 + sound/soc/tegra/tegra186_dspk.c | 442 ++++++++++ sound/soc/tegra/tegra186_dspk.h | 70 ++ sound/soc/tegra/tegra20_das.c | 3 +- sound/soc/tegra/tegra20_das.h | 4 +- sound/soc/tegra/tegra210_admaif.c | 800 ++++++++++++++++++ sound/soc/tegra/tegra210_admaif.h | 162 ++++ sound/soc/tegra/tegra210_ahub.c | 676 ++++++++++++++++ sound/soc/tegra/tegra210_ahub.h | 127 +++ sound/soc/tegra/tegra210_dmic.c | 456 +++++++++++ sound/soc/tegra/tegra210_dmic.h | 82 ++ sound/soc/tegra/tegra210_i2s.c | 812 +++++++++++++++++++ sound/soc/tegra/tegra210_i2s.h | 126 +++ sound/soc/tegra/tegra30_ahub.c | 4 +- sound/soc/tegra/tegra30_i2s.c | 4 +- sound/soc/tegra/tegra_alc5632.c | 2 +- sound/soc/tegra/tegra_cif.h | 65 ++ sound/soc/tegra/tegra_max98090.c | 2 +- sound/soc/tegra/tegra_pcm.c | 235 +++++- sound/soc/tegra/tegra_pcm.h | 21 +- sound/soc/tegra/tegra_rt5640.c | 2 +- sound/soc/tegra/tegra_rt5677.c | 2 +- sound/soc/tegra/tegra_sgtl5000.c | 2 +- sound/soc/tegra/tegra_wm8753.c | 2 +- sound/soc/tegra/tegra_wm8903.c | 2 +- sound/soc/tegra/trimslice.c | 2 +- sound/soc/ti/Kconfig | 9 + sound/soc/ti/Makefile | 2 + sound/soc/ti/ams-delta.c | 9 +- sound/soc/ti/davinci-evm.c | 6 +- sound/soc/ti/davinci-mcasp.c | 3 + sound/soc/ti/davinci-vcif.c | 4 +- sound/soc/ti/j721e-evm.c | 896 +++++++++++++++++++++ sound/soc/ti/n810.c | 4 +- sound/soc/ti/omap-abe-twl6040.c | 4 +- sound/soc/ti/omap-hdmi.c | 2 +- sound/soc/ti/omap-mcbsp-st.c | 3 - sound/soc/ti/omap-mcbsp.c | 4 +- sound/soc/ti/omap-twl4030.c | 4 +- sound/soc/ti/omap3pandora.c | 2 +- sound/soc/ti/osk5912.c | 2 +- sound/soc/ti/rx51.c | 4 +- sound/soc/ti/sdma-pcm.c | 2 +- sound/soc/ti/sdma-pcm.h | 2 +- sound/soc/ti/udma-pcm.c | 2 +- sound/soc/ti/udma-pcm.h | 2 +- sound/soc/uniphier/aio-core.c | 7 +- sound/soc/uniphier/aio-dma.c | 6 +- sound/soc/ux500/mop500_ab8500.c | 8 +- sound/soc/ux500/ux500_msp_i2s.c | 8 +- sound/soc/ux500/ux500_pcm.c | 2 +- sound/soc/xtensa/xtfpga-i2s.c | 2 +- 560 files changed, 16469 insertions(+), 4906 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/ak4613.txt create mode 100644 Documentation/devicetree/bindings/sound/ak4613.yaml delete mode 100644 Documentation/devicetree/bindings/sound/ak4642.txt create mode 100644 Documentation/devicetree/bindings/sound/ak4642.yaml delete mode 100644 Documentation/devicetree/bindings/sound/everest,es8316.txt create mode 100644 Documentation/devicetree/bindings/sound/everest,es8316.yaml create mode 100644 Documentation/devicetree/bindings/sound/intel,keembay-i2s.yaml create mode 100644 Documentation/devicetree/bindings/sound/maxim,max98390.yaml create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra186-dspk.yaml create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-admaif.yaml create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-ahub.yaml create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-dmic.yaml create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-i2s.yaml delete mode 100644 Documentation/devicetree/bindings/sound/rockchip,rk3328-codec.txt create mode 100644 Documentation/devicetree/bindings/sound/rockchip,rk3328-codec.yaml delete mode 100644 Documentation/devicetree/bindings/sound/rohm,bd28623.txt create mode 100644 Documentation/devicetree/bindings/sound/rohm,bd28623.yaml create mode 100644 Documentation/devicetree/bindings/sound/samsung,aries-wm8994.yaml create mode 100644 Documentation/devicetree/bindings/sound/samsung,midas-audio.yaml delete mode 100644 Documentation/devicetree/bindings/sound/sgtl5000.txt create mode 100644 Documentation/devicetree/bindings/sound/sgtl5000.yaml create mode 100644 Documentation/devicetree/bindings/sound/socionext,uniphier-aio.yaml create mode 100644 Documentation/devicetree/bindings/sound/socionext,uniphier-evea.yaml create mode 100644 Documentation/devicetree/bindings/sound/tas2562.yaml delete mode 100644 Documentation/devicetree/bindings/sound/tas2770.txt create mode 100644 Documentation/devicetree/bindings/sound/tas2770.yaml create mode 100644 Documentation/devicetree/bindings/sound/ti,j721e-cpb-audio.yaml create mode 100644 Documentation/devicetree/bindings/sound/ti,j721e-cpb-ivi-audio.yaml delete mode 100644 Documentation/devicetree/bindings/sound/uniphier,aio.txt delete mode 100644 Documentation/devicetree/bindings/sound/uniphier,evea.txt delete mode 100644 include/sound/rt5670.h create mode 100644 sound/soc/codecs/max98373-i2c.c create mode 100644 sound/soc/codecs/max98373-sdw.c create mode 100644 sound/soc/codecs/max98373-sdw.h create mode 100644 sound/soc/intel/boards/sof_sdw_max98373.c create mode 100644 sound/soc/intel/keembay/Makefile create mode 100644 sound/soc/intel/keembay/kmb_platform.c create mode 100644 sound/soc/intel/keembay/kmb_platform.h create mode 100644 sound/soc/samsung/aries_wm8994.c create mode 100644 sound/soc/samsung/midas_wm1811.c delete mode 100644 sound/soc/soc-io.c create mode 100644 sound/soc/tegra/tegra186_dspk.c create mode 100644 sound/soc/tegra/tegra186_dspk.h create mode 100644 sound/soc/tegra/tegra210_admaif.c create mode 100644 sound/soc/tegra/tegra210_admaif.h create mode 100644 sound/soc/tegra/tegra210_ahub.c create mode 100644 sound/soc/tegra/tegra210_ahub.h create mode 100644 sound/soc/tegra/tegra210_dmic.c create mode 100644 sound/soc/tegra/tegra210_dmic.h create mode 100644 sound/soc/tegra/tegra210_i2s.c create mode 100644 sound/soc/tegra/tegra210_i2s.h create mode 100644 sound/soc/tegra/tegra_cif.h create mode 100644 sound/soc/ti/j721e-evm.c

2 1

[PATCH 00/10] Audio graph card updates and usage with Tegra210 audio
by Sameer Pujar 03 Aug '20

03 Aug '20

This series proposes following enhancements to audio-graph card driver. * Support multiple instances of a component. * Support Codec port with multiple endpoints. * Support open platforms with empty Codec endpoint. * Identify no-pcm DPCM DAI links which can be used in BE<->BE connections. * Add new compatible to support DPCM based DAI chaining. This pushes DT support for Tegra210 based platforms which uses audio-graph card and above enhancements. The series is based on following references where DPCM usgae for Tegra Audio and simple-card driver proposal were discussed. * https://lkml.org/lkml/2020/4/30/519 (DPCM for Tegra) * https://lkml.org/lkml/2020/6/27/4 (simple-card driver) Sameer Pujar (10): ASoC: soc-core: Fix component name_prefix parsing ASoC: audio-graph: Use of_node and DAI for DPCM DAI link names ASoC: audio-graph: Support Codec with multiple endpoints ASoC: audio-graph: Support for empty end point ASoC: audio-graph: Identify 'no_pcm' DAI links for DPCM ASoC: soc-pcm: Get all BEs along DAPM path ASoC: dt-bindings: audio-graph-card: Support for component chaining ASoC: audio-graph: Add support for component chaining arm64: tegra: Audio graph header for Tegra210 arm64: tegra: Audio graph sound card for Jetson Nano and TX1 .../devicetree/bindings/sound/audio-graph-card.txt | 1 + .../boot/dts/nvidia/tegra210-audio-graph.dtsi | 120 +++++++++++++ arch/arm64/boot/dts/nvidia/tegra210-p2371-2180.dts | 196 +++++++++++++++++++++ arch/arm64/boot/dts/nvidia/tegra210-p3450-0000.dts | 110 ++++++++++++ include/sound/soc.h | 1 + sound/soc/generic/audio-graph-card.c | 85 ++++++--- sound/soc/soc-core.c | 3 +- sound/soc/soc-pcm.c | 3 +- 8 files changed, 495 insertions(+), 24 deletions(-) create mode 100644 arch/arm64/boot/dts/nvidia/tegra210-audio-graph.dtsi -- 2.7.4

2 18

[PATCH] ASoC: wm8962: Do not access WM8962_GPIO_BASE
by Fabio Estevam 03 Aug '20

03 Aug '20

According to the WM8962 datasheet, there is no register at address 0x200. WM8962_GPIO_BASE is just a base address for the GPIO registers and not a real register, so remove it from wm8962_readable_register(). Also, Register 515 (WM8962_GPIO_BASE + 3) does not exist, so skip its access. This fixes the following errors: wm8962 0-001a: ASoC: error at soc_component_read_no_lock on wm8962.0-001a: -16 wm8962 0-001a: ASoC: error at soc_component_read_no_lock on wm8962.0-001a: -16 Signed-off-by: Fabio Estevam <festevam(a)gmail.com> --- Hi, There is still one more soc_component_read_no_lock error left on register 48. I can get rid of it with the below change: --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -151,6 +151,7 @@ static const struct reg_default wm8962_reg[] = { { 40, 0x0000 }, /* R40 - SPKOUTL volume */ { 41, 0x0000 }, /* R41 - SPKOUTR volume */ + { 48, 0x0000 }, /* R48 - Additional control(4) */ { 49, 0x0010 }, /* R49 - Class D Control 1 */ { 51, 0x0003 }, /* R51 - Class D Control 2 */ @@ -841,7 +842,6 @@ static bool wm8962_readable_register(struct device *dev, unsigned int reg) case WM8962_SPKOUTL_VOLUME: case WM8962_SPKOUTR_VOLUME: case WM8962_THERMAL_SHUTDOWN_STATUS: - case WM8962_ADDITIONAL_CONTROL_4: case WM8962_CLASS_D_CONTROL_1: case WM8962_CLASS_D_CONTROL_2: case WM8962_CLOCKING_4: I haven't submitted it yet because I don't know if this is the correct approach. Please advise. Thanks sound/soc/codecs/wm8962.c | 9 +++++++-- 1 file changed, 7 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index df8cdc71357d..8159a3866cde 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -956,7 +956,6 @@ static bool wm8962_readable_register(struct device *dev, unsigned int reg) case WM8962_EQ39: case WM8962_EQ40: case WM8962_EQ41: - case WM8962_GPIO_BASE: case WM8962_GPIO_2: case WM8962_GPIO_3: case WM8962_GPIO_5: @@ -3437,7 +3436,13 @@ static int wm8962_probe(struct snd_soc_component *component) /* Save boards having to disable DMIC when not in use */ dmicclk = false; dmicdat = false; - for (i = 0; i < WM8962_MAX_GPIO; i++) { + for (i = 1; i < WM8962_MAX_GPIO; i++) { + /* + * Register 515 (WM8962_GPIO_BASE + 3) does not exist, + * so skip its access + */ + if (i == 3) + continue; switch (snd_soc_component_read(component, WM8962_GPIO_BASE + i) & WM8962_GP2_FN_MASK) { case WM8962_GPIO_FN_DMICCLK: -- 2.17.1

4 9

[PATCH v3 0/2] Add documentation and machine driver for SC7180 sound card
by Cheng-Yi Chiang 03 Aug '20

03 Aug '20

Note: - The machine driver patch depends on LPASS patch series so it is not ready to be merged now. ASoC: qcom: Add support for SC7180 lpass variant https://patchwork.kernel.org/cover/11678133/ - The machine driver patch is made by the collaboration of Cheng-Yi Chiang <cychiang(a)chromium.org> Rohit kumar <rohitkr(a)codeaurora.org> Ajit Pandey <ajitp(a)codeaurora.org> But Ajit has left codeaurora. Changes from v1 to v2: - Ducumentation: Addressed all suggestions from Doug. - Machine driver: - Fix comment style for license. - Sort includes. - Remove sc7180_snd_hw_params. - Remove sc7180_dai_init and use aux device instead for headset jack registration. - Statically define format for Primary MI2S. - Atomic is not a concern because there is mutex in card to make sure startup and shutdown happen sequentially. - Fix missing return -EINVAL in startup. - Use static sound card. - Use devm_kzalloc to avoid kfree. Changes from v2 to v3: - Ducumentation: Addressed suggestions from Srini. - Machine driver: - Reuse qcom_snd_parse_of to parse properties. - Remove playback-only and capture-only. - Misc fixes to address comments. Thanks for the review! Ajit Pandey (1): ASoC: qcom: sc7180: Add machine driver for sound card registration Cheng-Yi Chiang (1): ASoC: qcom: dt-bindings: Add sc7180 machine bindings .../bindings/sound/qcom,sc7180.yaml | 113 ++++++++ sound/soc/qcom/Kconfig | 12 + sound/soc/qcom/Makefile | 2 + sound/soc/qcom/sc7180.c | 244 ++++++++++++++++++ 4 files changed, 371 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/qcom,sc7180.yaml create mode 100644 sound/soc/qcom/sc7180.c -- 2.28.0.163.g6104cc2f0b6-goog

3 7

Re: 回复: [PATCH] ALSA: seq: KASAN: use-after-free Read in delete_and_unsubscribe_port
by Takashi Iwai 03 Aug '20

03 Aug '20

On Mon, 03 Aug 2020 03:35:05 +0200, Zhang, Qiang wrote: > > >Thanks for the patch. But I'm afraid that this change would break the > >existing behavior and might have a bad side-effect. > > >It's likely the same issue as reported in another syzkaller report > >("KASAN: invalid-free in snd_seq_port_disconnect"), and Hillf's patch > >below should covert this as well. Could you check whether it works? > > yes It's should same issue, add mutex lock in odev_ioctl, ensure serialization. > however, it should not be necessary to mutually exclusive with open and close. That's a big-hammer approach indeed, but it should be more reasonable in this case. It makes the patch shorter and simpler, while the OSS sequencer is an ancient interface that wasn't considered much for the concurrency, and this might also cover the case where the access to another sequencer object that is being to be closed. So, it'd be great if you can confirm that the patch actually works. Then we can brush up and merge it for 5.9-rc1. thanks, Takashi > > > > >thanks, > > >Takashi > > >--- > >--- a/sound/core/seq/oss/seq_oss.c > >+++ b/sound/core/seq/oss/seq_oss.c > >@@ -167,11 +167,17 @@ odev_write(struct file *file, const char > >static long > >odev_ioctl(struct file *file, unsigned int cmd, unsigned long arg) > >{ > >+ long rc; > > struct seq_oss_devinfo *dp; > >+ > >+ mutex_lock(&register_mutex); > > dp = file->private_data; > > if (snd_BUG_ON(!dp)) > >- return -ENXIO; > >- return snd_seq_oss_ioctl(dp, cmd, arg); > >+ rc = -ENXIO; > >+ else > >+ rc = snd_seq_oss_ioctl(dp, cmd, arg); > >+ mutex_unlock(&register_mutex); > >+ return rc; > >} > > >#ifdef CONFIG_COMPAT

1 0

[PATCH 1/3] ALSA: hda/ca0132 - Fix ZxR Headphone gain control get value.
by Connor McAdams 03 Aug '20

03 Aug '20

When the ZxR headphone gain control was added, the ca0132_switch_get function was not updated, which meant that the changes to the control state were not saved when entering/exiting alsamixer. Signed-off-by: Connor McAdams <conmanx360(a)gmail.com> --- sound/pci/hda/patch_ca0132.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index 19c575fd28a1..2cbe01d59c16 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -5748,6 +5748,11 @@ static int ca0132_switch_get(struct snd_kcontrol *kcontrol, return 0; } + if (nid == ZXR_HEADPHONE_GAIN) { + *valp = spec->zxr_gain_set; + return 0; + } + return 0; } -- 2.20.1

2 5

[PATCH v2] ASoC: fsl-asoc-card: Remove fsl_asoc_card_set_bias_level function
by Shengjiu Wang 03 Aug '20

03 Aug '20

With this case: aplay -Dhw:x 16khz.wav 24khz.wav There is sound distortion for 24khz.wav. The reason is that setting PLL of WM8962 with set_bias_level function, the bias level is not changed when 24khz.wav is played, then the PLL won't be reset, the clock is not correct, so distortion happens. The resolution of this issue is to remove fsl_asoc_card_set_bias_level. Move PLL configuration to hw_params and hw_free. After removing fsl_asoc_card_set_bias_level, also test WM8960 case, it can work. Fixes: 708b4351f08c ("ASoC: fsl: Add Freescale Generic ASoC Sound Card with ASRC support") Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- changes in v2 - replace is_stream_in_use with streams - add "out" error handler in hw_params() sound/soc/fsl/fsl-asoc-card.c | 155 ++++++++++++++++------------------ 1 file changed, 71 insertions(+), 84 deletions(-) diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index 4848ba61d083..b6a2a527dc04 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -73,6 +73,7 @@ struct cpu_priv { * @codec_priv: CODEC private data * @cpu_priv: CPU private data * @card: ASoC card structure + * @streams: Mask of current active streams * @sample_rate: Current sample rate * @sample_format: Current sample format * @asrc_rate: ASRC sample rate used by Back-Ends @@ -89,6 +90,7 @@ struct fsl_asoc_card_priv { struct codec_priv codec_priv; struct cpu_priv cpu_priv; struct snd_soc_card card; + u8 streams; u32 sample_rate; snd_pcm_format_t sample_format; u32 asrc_rate; @@ -151,21 +153,17 @@ static int fsl_asoc_card_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(rtd->card); bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + struct codec_priv *codec_priv = &priv->codec_priv; struct cpu_priv *cpu_priv = &priv->cpu_priv; struct device *dev = rtd->card->dev; + unsigned int pll_out; int ret; priv->sample_rate = params_rate(params); priv->sample_format = params_format(params); + priv->streams |= BIT(substream->stream); - /* - * If codec-dai is DAI Master and all configurations are already in the - * set_bias_level(), bypass the remaining settings in hw_params(). - * Note: (dai_fmt & CBM_CFM) includes CBM_CFM and CBM_CFS. - */ - if ((priv->card.set_bias_level && - priv->dai_fmt & SND_SOC_DAIFMT_CBM_CFM) || - fsl_asoc_card_is_ac97(priv)) + if (fsl_asoc_card_is_ac97(priv)) return 0; /* Specific configurations of DAIs starts from here */ @@ -174,7 +172,7 @@ static int fsl_asoc_card_hw_params(struct snd_pcm_substream *substream, cpu_priv->sysclk_dir[tx]); if (ret && ret != -ENOTSUPP) { dev_err(dev, "failed to set sysclk for cpu dai\n"); - return ret; + goto out; } if (cpu_priv->slot_width) { @@ -182,6 +180,69 @@ static int fsl_asoc_card_hw_params(struct snd_pcm_substream *substream, cpu_priv->slot_width); if (ret && ret != -ENOTSUPP) { dev_err(dev, "failed to set TDM slot for cpu dai\n"); + goto out; + } + } + + /* Specific configuration for PLL */ + if (codec_priv->pll_id && codec_priv->fll_id) { + if (priv->sample_format == SNDRV_PCM_FORMAT_S24_LE) + pll_out = priv->sample_rate * 384; + else + pll_out = priv->sample_rate * 256; + + ret = snd_soc_dai_set_pll(asoc_rtd_to_codec(rtd, 0), + codec_priv->pll_id, + codec_priv->mclk_id, + codec_priv->mclk_freq, pll_out); + if (ret) { + dev_err(dev, "failed to start FLL: %d\n", ret); + goto out; + } + + ret = snd_soc_dai_set_sysclk(asoc_rtd_to_codec(rtd, 0), + codec_priv->fll_id, + pll_out, SND_SOC_CLOCK_IN); + + if (ret && ret != -ENOTSUPP) { + dev_err(dev, "failed to set SYSCLK: %d\n", ret); + goto out; + } + } + + return 0; + +out: + priv->streams &= ~BIT(substream->stream); + return ret; +} + +static int fsl_asoc_card_hw_free(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(rtd->card); + struct codec_priv *codec_priv = &priv->codec_priv; + struct device *dev = rtd->card->dev; + int ret; + + priv->streams &= ~BIT(substream->stream); + + if (!priv->streams && codec_priv->pll_id && + codec_priv->fll_id) { + /* Force freq to be 0 to avoid error message in codec */ + ret = snd_soc_dai_set_sysclk(asoc_rtd_to_codec(rtd, 0), + codec_priv->mclk_id, + 0, + SND_SOC_CLOCK_IN); + if (ret) { + dev_err(dev, "failed to switch away from FLL: %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_pll(asoc_rtd_to_codec(rtd, 0), + codec_priv->pll_id, 0, 0, 0); + if (ret && ret != -ENOTSUPP) { + dev_err(dev, "failed to stop FLL: %d\n", ret); return ret; } } @@ -191,6 +252,7 @@ static int fsl_asoc_card_hw_params(struct snd_pcm_substream *substream, static const struct snd_soc_ops fsl_asoc_card_ops = { .hw_params = fsl_asoc_card_hw_params, + .hw_free = fsl_asoc_card_hw_free, }; static int be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, @@ -254,75 +316,6 @@ static struct snd_soc_dai_link fsl_asoc_card_dai[] = { }, }; -static int fsl_asoc_card_set_bias_level(struct snd_soc_card *card, - struct snd_soc_dapm_context *dapm, - enum snd_soc_bias_level level) -{ - struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(card); - struct snd_soc_pcm_runtime *rtd; - struct snd_soc_dai *codec_dai; - struct codec_priv *codec_priv = &priv->codec_priv; - struct device *dev = card->dev; - unsigned int pll_out; - int ret; - - rtd = snd_soc_get_pcm_runtime(card, &card->dai_link[0]); - codec_dai = asoc_rtd_to_codec(rtd, 0); - if (dapm->dev != codec_dai->dev) - return 0; - - switch (level) { - case SND_SOC_BIAS_PREPARE: - if (dapm->bias_level != SND_SOC_BIAS_STANDBY) - break; - - if (priv->sample_format == SNDRV_PCM_FORMAT_S24_LE) - pll_out = priv->sample_rate * 384; - else - pll_out = priv->sample_rate * 256; - - ret = snd_soc_dai_set_pll(codec_dai, codec_priv->pll_id, - codec_priv->mclk_id, - codec_priv->mclk_freq, pll_out); - if (ret) { - dev_err(dev, "failed to start FLL: %d\n", ret); - return ret; - } - - ret = snd_soc_dai_set_sysclk(codec_dai, codec_priv->fll_id, - pll_out, SND_SOC_CLOCK_IN); - if (ret && ret != -ENOTSUPP) { - dev_err(dev, "failed to set SYSCLK: %d\n", ret); - return ret; - } - break; - - case SND_SOC_BIAS_STANDBY: - if (dapm->bias_level != SND_SOC_BIAS_PREPARE) - break; - - ret = snd_soc_dai_set_sysclk(codec_dai, codec_priv->mclk_id, - codec_priv->mclk_freq, - SND_SOC_CLOCK_IN); - if (ret && ret != -ENOTSUPP) { - dev_err(dev, "failed to switch away from FLL: %d\n", ret); - return ret; - } - - ret = snd_soc_dai_set_pll(codec_dai, codec_priv->pll_id, 0, 0, 0); - if (ret) { - dev_err(dev, "failed to stop FLL: %d\n", ret); - return ret; - } - break; - - default: - break; - } - - return 0; -} - static int fsl_asoc_card_audmux_init(struct device_node *np, struct fsl_asoc_card_priv *priv) { @@ -611,7 +604,6 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) /* Diversify the card configurations */ if (of_device_is_compatible(np, "fsl,imx-audio-cs42888")) { codec_dai_name = "cs42888"; - priv->card.set_bias_level = NULL; priv->cpu_priv.sysclk_freq[TX] = priv->codec_priv.mclk_freq; priv->cpu_priv.sysclk_freq[RX] = priv->codec_priv.mclk_freq; priv->cpu_priv.sysclk_dir[TX] = SND_SOC_CLOCK_OUT; @@ -628,26 +620,22 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; } else if (of_device_is_compatible(np, "fsl,imx-audio-wm8962")) { codec_dai_name = "wm8962"; - priv->card.set_bias_level = fsl_asoc_card_set_bias_level; priv->codec_priv.mclk_id = WM8962_SYSCLK_MCLK; priv->codec_priv.fll_id = WM8962_SYSCLK_FLL; priv->codec_priv.pll_id = WM8962_FLL; priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; } else if (of_device_is_compatible(np, "fsl,imx-audio-wm8960")) { codec_dai_name = "wm8960-hifi"; - priv->card.set_bias_level = fsl_asoc_card_set_bias_level; priv->codec_priv.fll_id = WM8960_SYSCLK_AUTO; priv->codec_priv.pll_id = WM8960_SYSCLK_AUTO; priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; } else if (of_device_is_compatible(np, "fsl,imx-audio-ac97")) { codec_dai_name = "ac97-hifi"; - priv->card.set_bias_level = NULL; priv->dai_fmt = SND_SOC_DAIFMT_AC97; priv->card.dapm_routes = audio_map_ac97; priv->card.num_dapm_routes = ARRAY_SIZE(audio_map_ac97); } else if (of_device_is_compatible(np, "fsl,imx-audio-mqs")) { codec_dai_name = "fsl-mqs-dai"; - priv->card.set_bias_level = NULL; priv->dai_fmt = SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_CBS_CFS | SND_SOC_DAIFMT_NB_NF; @@ -657,7 +645,6 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) priv->card.num_dapm_routes = ARRAY_SIZE(audio_map_tx); } else if (of_device_is_compatible(np, "fsl,imx-audio-wm8524")) { codec_dai_name = "wm8524-hifi"; - priv->card.set_bias_level = NULL; priv->dai_fmt |= SND_SOC_DAIFMT_CBS_CFS; priv->dai_link[1].dpcm_capture = 0; priv->dai_link[2].dpcm_capture = 0; -- 2.27.0

3 4

[PATCH] ALSA: hda/realtek: Add alc269/alc662 pin-tables for Loongson-3 laptops
by Huacai Chen 02 Aug '20

02 Aug '20

There are several Loongson-3 based laptops produced by CZC or Lemote, they use alc269/alc662 codecs and need specific pin-tables, this patch add their pin-tables. Signed-off-by: Huacai Chen <chenhc(a)lemote.com> --- sound/pci/hda/patch_realtek.c | 114 ++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 114 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 29f5878..65c75c6 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6166,6 +6166,11 @@ enum { ALC289_FIXUP_ASUS_GA502, ALC256_FIXUP_ACER_MIC_NO_PRESENCE, ALC285_FIXUP_HP_GPIO_AMP_INIT, + ALC269_FIXUP_CZC_B20, + ALC269_FIXUP_CZC_TMI, + ALC269_FIXUP_CZC_L101, + ALC269_FIXUP_LEMOTE_A1802, + ALC269_FIXUP_LEMOTE_A190X, }; static const struct hda_fixup alc269_fixups[] = { @@ -7404,6 +7409,89 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC285_FIXUP_HP_GPIO_LED }, + [ALC269_FIXUP_CZC_B20] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x12, 0x411111f0 }, + { 0x14, 0x90170110 }, /* speaker */ + { 0x15, 0x032f1020 }, /* HP out */ + { 0x17, 0x411111f0 }, + { 0x18, 0x03ab1040 }, /* mic */ + { 0x19, 0xb7a7013f }, + { 0x1a, 0x0181305f }, + { 0x1b, 0x411111f0 }, + { 0x1d, 0x411111f0 }, + { 0x1e, 0x411111f0 }, + { } + }, + .chain_id = ALC269_FIXUP_DMIC, + }, + [ALC269_FIXUP_CZC_TMI] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x12, 0x4000c000 }, + { 0x14, 0x90170110 }, /* speaker */ + { 0x15, 0x0421401f }, /* HP out */ + { 0x17, 0x411111f0 }, + { 0x18, 0x04a19020 }, /* mic */ + { 0x19, 0x411111f0 }, + { 0x1a, 0x411111f0 }, + { 0x1b, 0x411111f0 }, + { 0x1d, 0x40448505 }, + { 0x1e, 0x411111f0 }, + { 0x20, 0x8000ffff }, + { } + }, + .chain_id =ALC269_FIXUP_DMIC, + }, + [ALC269_FIXUP_CZC_L101] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x12, 0x40000000 }, + { 0x14, 0x01014010 }, /* speaker */ + { 0x15, 0x411111f0 }, /* HP out */ + { 0x16, 0x411111f0 }, + { 0x18, 0x01a19020 }, /* mic */ + { 0x19, 0x02a19021 }, + { 0x1a, 0x0181302f }, + { 0x1b, 0x0221401f }, + { 0x1c, 0x411111f0 }, + { 0x1d, 0x4044c601 }, + { 0x1e, 0x411111f0 }, + { } + }, + .chain_id =ALC269_FIXUP_DMIC, + }, + [ALC269_FIXUP_LEMOTE_A1802] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x12, 0x40000000 }, + { 0x14, 0x90170110 }, /* speaker */ + { 0x17, 0x411111f0 }, + { 0x18, 0x03a19040 }, /* mic1 */ + { 0x19, 0x90a70130 }, /* mic2 */ + { 0x1a, 0x411111f0 }, + { 0x1b, 0x411111f0 }, + { 0x1d, 0x40489d2d }, + { 0x1e, 0x411111f0 }, + { 0x20, 0x0003ffff }, + { 0x21, 0x03214020 }, + { } + }, + .chain_id = ALC269_FIXUP_DMIC, + }, + [ALC269_FIXUP_LEMOTE_A190X] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x14, 0x99130110 }, /* speaker */ + { 0x15, 0x0121401f }, /* HP out */ + { 0x18, 0x01a19c20 }, /* rear mic */ + { 0x19, 0x99a3092f }, /* front mic */ + { 0x1b, 0x0201401f }, /* front lineout */ + { } + }, + .chain_id = ALC269_FIXUP_DMIC, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -7693,9 +7781,14 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_PCM_44K), SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD), SND_PCI_QUIRK(0x19e5, 0x3204, "Huawei MACH-WX9", ALC256_FIXUP_HUAWEI_MACH_WX9_PINS), + SND_PCI_QUIRK(0x1b35, 0x1235, "CZC B20", ALC269_FIXUP_CZC_B20), + SND_PCI_QUIRK(0x1b35, 0x1236, "CZC TMI", ALC269_FIXUP_CZC_TMI), + SND_PCI_QUIRK(0x1b35, 0x1237, "CZC L101", ALC269_FIXUP_CZC_L101), SND_PCI_QUIRK(0x1b7d, 0xa831, "Ordissimo EVE2 ", ALC269VB_FIXUP_ORDISSIMO_EVE2), /* Also known as Malata PC-B1303 */ SND_PCI_QUIRK(0x1d72, 0x1901, "RedmiBook 14", ALC256_FIXUP_ASUS_HEADSET_MIC), SND_PCI_QUIRK(0x10ec, 0x118c, "Medion EE4254 MD62100", ALC256_FIXUP_MEDION_HEADSET_NO_PRESENCE), + SND_PCI_QUIRK(0x1c06, 0x2013, "Lemote A1802", ALC269_FIXUP_LEMOTE_A1802), + SND_PCI_QUIRK(0x1c06, 0x2015, "Lemote A190X", ALC269_FIXUP_LEMOTE_A190X), #if 0 /* Below is a quirk table taken from the old code. @@ -8951,6 +9044,7 @@ enum { ALC662_FIXUP_LED_GPIO1, ALC662_FIXUP_IDEAPAD, ALC272_FIXUP_MARIO, + ALC662_FIXUP_CZC_ET26, ALC662_FIXUP_CZC_P10T, ALC662_FIXUP_SKU_IGNORE, ALC662_FIXUP_HP_RP5800, @@ -9020,6 +9114,25 @@ static const struct hda_fixup alc662_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc272_fixup_mario, }, + [ALC662_FIXUP_CZC_ET26] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + {0x12, 0x403cc000}, + {0x14, 0x90170110}, /* speaker */ + {0x15, 0x411111f0}, + {0x16, 0x411111f0}, + {0x18, 0x01a19030}, /* mic */ + {0x19, 0x90a7013f}, /* int-mic */ + {0x1a, 0x01014020}, + {0x1b, 0x0121401f}, + {0x1c, 0x411111f0}, + {0x1d, 0x411111f0}, + {0x1e, 0x40478e35}, + {} + }, + .chained = true, + .chain_id = ALC662_FIXUP_SKU_IGNORE + }, [ALC662_FIXUP_CZC_P10T] = { .type = HDA_FIXUP_VERBS, .v.verbs = (const struct hda_verb[]) { @@ -9403,6 +9516,7 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = { SND_PCI_QUIRK(0x1849, 0x5892, "ASRock B150M", ALC892_FIXUP_ASROCK_MOBO), SND_PCI_QUIRK(0x19da, 0xa130, "Zotac Z68", ALC662_FIXUP_ZOTAC_Z68), SND_PCI_QUIRK(0x1b0a, 0x01b8, "ACER Veriton", ALC662_FIXUP_ACER_VERITON), + SND_PCI_QUIRK(0x1b35, 0x1234, "CZC ET26", ALC662_FIXUP_CZC_ET26), SND_PCI_QUIRK(0x1b35, 0x2206, "CZC P10T", ALC662_FIXUP_CZC_P10T), SND_PCI_QUIRK(0x1025, 0x0566, "Acer Aspire Ethos 8951G", ALC669_FIXUP_ACER_ASPIRE_ETHOS), -- 2.7.0

2 2

[PATCH] ALSA: seq: KASAN: use-after-free Read in delete_and_unsubscribe_port
by qiang.zhang＠windriver.com 01 Aug '20

01 Aug '20

From: Zhang Qiang <qiang.zhang(a)windriver.com> There is a potential race window,when a task acquire "src->list_mutex" write sem,traverse the linked list to find "subs" objects through parameter "info" in snd_seq_port_disconnect and then release this write sem, before this task acquire write sem again,this write sem may be acquired by another task, and get the same "subs" object through the same "info" before, it could happen "use-after-free" later, so a simple solution is to delete the object from the linked list when it is found. BUG: KASAN: use-after-free in list_empty include/linux/list.h:282 [inline] BUG: KASAN: use-after-free in delete_and_unsubscribe_port+0x8b/0x450 sound/core/seq/seq_ports.c:530 Read of size 8 at addr ffff888098523060 by task syz-executor.0/7202 Call Trace: __dump_stack lib/dump_stack.c:77 [inline] dump_stack+0x1f0/0x31e lib/dump_stack.c:118 print_address_description+0x66/0x5a0 mm/kasan/report.c:383 __kasan_report mm/kasan/report.c:513 [inline] kasan_report+0x132/0x1d0 mm/kasan/report.c:530 list_empty include/linux/list.h:282 [inline] delete_and_unsubscribe_port+0x8b/0x450 sound/core/seq/seq_ports.c:530 snd_seq_port_disconnect+0x568/0x610 sound/core/seq/seq_ports.c:612 snd_seq_ioctl_unsubscribe_port+0x349/0x6c0 sound/core/seq/seq_clientmgr.c:1525 snd_seq_oss_midi_close+0x397/0x620 sound/core/seq/oss/seq_oss_midi.c:405 snd_seq_oss_synth_reset+0x335/0x8b0 sound/core/seq/oss/seq_oss_synth.c:406 snd_seq_oss_reset+0x5b/0x250 sound/core/seq/oss/seq_oss_init.c:435 snd_seq_oss_ioctl+0x5c2/0x1090 sound/core/seq/oss/seq_oss_ioctl.c:93 odev_ioctl+0x51/0x70 sound/core/seq/oss/seq_oss.c:174 vfs_ioctl fs/ioctl.c:48 [inline] ksys_ioctl fs/ioctl.c:753 [inline] __do_sys_ioctl fs/ioctl.c:762 [inline] __se_sys_ioctl+0xf9/0x160 fs/ioctl.c:760 do_syscall_64+0x73/0xe0 arch/x86/entry/common.c:384 entry_SYSCALL_64_after_hwframe+0x44/0xa9 Allocated by task 7202: save_stack mm/kasan/common.c:48 [inline] set_track mm/kasan/common.c:56 [inline] __kasan_kmalloc+0x103/0x140 mm/kasan/common.c:494 kmem_cache_alloc_trace+0x234/0x300 mm/slab.c:3551 kmalloc include/linux/slab.h:555 [inline] kzalloc include/linux/slab.h:669 [inline] snd_seq_port_connect+0x66/0x460 sound/core/seq/seq_ports.c:553 snd_seq_ioctl_subscribe_port+0x349/0x6c0 sound/core/seq/seq_clientmgr.c:1484 snd_seq_oss_midi_open+0x4db/0x830 sound/core/seq/oss/seq_oss_midi.c:364 snd_seq_oss_synth_setup_midi+0x108/0x510 sound/core/seq/oss/seq_oss_synth.c:269 snd_seq_oss_open+0x899/0xe90 sound/core/seq/oss/seq_oss_init.c:261 odev_open+0x5e/0x90 sound/core/seq/oss/seq_oss.c:125 chrdev_open+0x498/0x580 fs/char_dev.c:414 do_dentry_open+0x813/0x1070 fs/open.c:828 do_open fs/namei.c:3243 [inline] path_openat+0x278d/0x37f0 fs/namei.c:3360 do_filp_open+0x191/0x3a0 fs/namei.c:3387 do_sys_openat2+0x463/0x770 fs/open.c:1179 do_sys_open fs/open.c:1195 [inline] __do_sys_openat fs/open.c:1209 [inline] __se_sys_openat fs/open.c:1204 [inline] __x64_sys_openat+0x1c8/0x1f0 fs/open.c:1204 do_syscall_64+0x73/0xe0 arch/x86/entry/common.c:384 entry_SYSCALL_64_after_hwframe+0x44/0xa9 Freed by task 7203: save_stack mm/kasan/common.c:48 [inline] set_track mm/kasan/common.c:56 [inline] kasan_set_free_info mm/kasan/common.c:316 [inline] __kasan_slab_free+0x114/0x170 mm/kasan/common.c:455 __cache_free mm/slab.c:3426 [inline] kfree+0x10a/0x220 mm/slab.c:3757 snd_seq_port_disconnect+0x570/0x610 sound/core/seq/seq_ports.c:614 snd_seq_ioctl_unsubscribe_port+0x349/0x6c0 sound/core/seq/seq_clientmgr.c:1525 snd_seq_oss_midi_close+0x397/0x620 sound/core/seq/oss/seq_oss_midi.c:405 snd_seq_oss_synth_reset+0x335/0x8b0 sound/core/seq/oss/seq_oss_synth.c:406 snd_seq_oss_reset+0x5b/0x250 sound/core/seq/oss/seq_oss_init.c:435 snd_seq_oss_ioctl+0x5c2/0x1090 sound/core/seq/oss/seq_oss_ioctl.c:93 odev_ioctl+0x51/0x70 sound/core/seq/oss/seq_oss.c:174 vfs_ioctl fs/ioctl.c:48 [inline] ksys_ioctl fs/ioctl.c:753 [inline] __do_sys_ioctl fs/ioctl.c:762 [inline] __se_sys_ioctl+0xf9/0x160 fs/ioctl.c:760 do_syscall_64+0x73/0xe0 arch/x86/entry/common.c:384 entry_SYSCALL_64_after_hwframe+0x44/0xa9 The buggy address belongs to the object at ffff888098523000 which belongs to the cache kmalloc-128 of size 128 The buggy address is located 96 bytes inside of 128-byte region [ffff888098523000, ffff888098523080) The buggy address belongs to the page: page:ffffea00026148c0 refcount:1 mapcount:0 mapping:0000000000000000 index:0x0 flags: 0xfffe0000000200(slab) raw: 00fffe0000000200 ffffea0002613988 ffffea000262c648 ffff8880aa400700 raw: 0000000000000000 ffff888098523000 0000000100000010 0000000000000000 page dumped because: kasan: bad access detected Memory state around the buggy address: ffff888098522f00: fc fc fc fc fc fc fc fc fc fc fc fc fc fc fc fc ffff888098522f80: fc fc fc fc fc fc fc fc fc fc fc fc fc fc fc fc >ffff888098523000: fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb ^ ffff888098523080: fc fc fc fc fc fc fc fc fc fc fc fc fc fc fc fc ffff888098523100: fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb ================================================================== Reported-by: syzbot+1a54a94bd32716796edd(a)syzkaller.appspotmail.com Signed-off-by: Zhang Qiang <qiang.zhang(a)windriver.com> --- sound/core/seq/seq_ports.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/core/seq/seq_ports.c b/sound/core/seq/seq_ports.c index 83be6b982a87..9675d3fc146e 100644 --- a/sound/core/seq/seq_ports.c +++ b/sound/core/seq/seq_ports.c @@ -533,8 +533,7 @@ static void delete_and_unsubscribe_port(struct snd_seq_client *client, grp->exclusive = 0; write_unlock_irq(&grp->list_lock); - if (!empty) - unsubscribe_port(client, port, grp, &subs->info, ack); + unsubscribe_port(client, port, grp, &subs->info, ack); up_write(&grp->list_mutex); } @@ -599,6 +598,7 @@ int snd_seq_port_disconnect(struct snd_seq_client *connector, list_for_each_entry(subs, &src->list_head, src_list) { if (match_subs_info(info, &subs->info)) { atomic_dec(&subs->ref_count); /* mark as not ready */ + list_del_init(&subs->src_list); err = 0; break; } -- 2.26.2

2 1

[PATCH] ALSA: docs: fix typo
by Julia Lawall 01 Aug '20

01 Aug '20

GFP_KRENEL -> GFP_KERNEL Signed-off-by: Julia Lawall <Julia.Lawall(a)inria.fr> --- Documentation/sound/kernel-api/writing-an-alsa-driver.rst | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/Documentation/sound/kernel-api/writing-an-alsa-driver.rst b/Documentation/sound/kernel-api/writing-an-alsa-driver.rst index fa49688..aa9d5ab 100644 --- a/Documentation/sound/kernel-api/writing-an-alsa-driver.rst +++ b/Documentation/sound/kernel-api/writing-an-alsa-driver.rst @@ -3579,7 +3579,7 @@ dependent on the bus. For normal devices, pass the device pointer ``SNDRV_DMA_TYPE_DEV`` type. For the continuous buffer unrelated to the bus can be pre-allocated with ``SNDRV_DMA_TYPE_CONTINUOUS`` type. You can pass NULL to the device pointer in that case, which is the -default mode implying to allocate with ``GFP_KRENEL`` flag. +default mode implying to allocate with ``GFP_KERNEL`` flag. If you need a different GFP flag, you can pass it by encoding the flag into the device pointer via a special macro :c:func:`snd_dma_continuous_data()`.

2 1

[PATCH] ALSA: doc: use correct config variable name
by Julia Lawall 01 Aug '20

01 Aug '20

CONFIG_PCM_XRUN_DEBUG should be CONFIG_SND_PCM_XRUN_DEBUG Signed-off-by: Julia Lawall <Julia.Lawall(a)inria.fr> --- Documentation/sound/designs/procfile.rst | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/Documentation/sound/designs/procfile.rst b/Documentation/sound/designs/procfile.rst index 29a4668..e9f7e0c 100644 --- a/Documentation/sound/designs/procfile.rst +++ b/Documentation/sound/designs/procfile.rst @@ -91,7 +91,7 @@ PCM Proc Files ``card*/pcm*/xrun_debug`` This file appears when ``CONFIG_SND_DEBUG=y`` and - ``CONFIG_PCM_XRUN_DEBUG=y``. + ``CONFIG_SND_PCM_XRUN_DEBUG=y``. This shows the status of xrun (= buffer overrun/xrun) and invalid PCM position debug/check of ALSA PCM middle layer. It takes an integer value, can be changed by writing to this

2 1

[PATCH v2 1/2] dt-bindings: sound: add DT bindings for Microchip S/PDIF TX Controller
by Codrin Ciubotariu 31 Jul '20

31 Jul '20

This patch adds DT bindings for the new Microchip S/PDIF TX Controller embedded inside sama7g5 SoCs. Signed-off-by: Codrin Ciubotariu <codrin.ciubotariu(a)microchip.com> --- Changes in v2: - replaced https with http - reworked example, included bindings; .../bindings/sound/mchp,spdiftx.yaml | 80 +++++++++++++++++++ 1 file changed, 80 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/mchp,spdiftx.yaml diff --git a/Documentation/devicetree/bindings/sound/mchp,spdiftx.yaml b/Documentation/devicetree/bindings/sound/mchp,spdiftx.yaml new file mode 100644 index 000000000000..2b7c27391f0d --- /dev/null +++ b/Documentation/devicetree/bindings/sound/mchp,spdiftx.yaml @@ -0,0 +1,80 @@ +# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/mchp,spdiftx.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Microchip S/PDIF Tx Controller Device Tree Bindings + +maintainers: + - Codrin Ciubotariu <codrin.ciubotariu(a)microchip.com> + +description: + The Microchip Sony/Philips Digital Interface Transmitter is a + serial port compliant with the IEC-60958 standard. + +properties: + "#sound-dai-cells": + const: 0 + + compatible: + oneOf: + - const: microchip,sama7g5-spdiftx + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + clocks: + items: + - description: Peripheral Bus Clock + - description: Generic Clock + + clock-names: + items: + - const: pclk + - const: gclk + + dmas: + description: TX DMA Channel + + dma-names: + const: tx + + pinctrl-0: true + + pinctrl-names: + const: default + +required: + - "#sound-dai-cells" + - compatible + - reg + - interrupts + - clocks + - clock-names + - dmas + - dma-names + +additionalProperties: false + +examples: + - | + #include <dt-bindings/clock/at91.h> + #include <dt-bindings/dma/at91.h> + #include <dt-bindings/interrupt-controller/arm-gic.h> + + spdiftx@e1618000 { + #sound-dai-cells = <0>; + compatible = "microchip,sama7g5-spdiftx"; + reg = <0xe1618000 0x4000>; + interrupts = <GIC_SPI 85 IRQ_TYPE_LEVEL_HIGH>; + dmas = <&dma0 AT91_XDMAC_DT_PERID(50)>; + dma-names = "tx"; + clocks = <&pmc PMC_TYPE_PERIPHERAL 85>, <&pmc PMC_TYPE_GCK 85>; + clock-names = "pclk", "gclk"; + pinctrl-names = "default"; + pinctrl-0 = <&pinctrl_spdiftx_default>; + }; -- 2.25.1

2 2