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

[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

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[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

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[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

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[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

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[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

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[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

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