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

[PATCH] ASoC: soc-pcm/soc-compress: don't use snd_soc_dapm_stream_stop()
by Kuninori Morimoto 25 Feb '20

25 Feb '20

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> commit b0edff42360ab4 ("ASoC: soc-pcm/soc-compress: use snd_soc_dapm_stream_stop() for SND_SOC_DAPM_STREAM_STOP") uses snd_soc_dapm_stream_stop() for soc_compr_free_fe() and dpcm_fe_dai_shutdown() because it didn't care about pmdown_time. But, it didn't need to care. This patch rollback to original code. Some system will wait unneeded timed-out without this patch. Special Thanks for reporting to Chris Gorman. ... intel_sst_acpi 808622A8:00: Wait timed-out condition:0x0, msg_id:0x1 fw_state 0x3 intel_sst_acpi 808622A8:00: fw returned err -16 sst-mfld-platform sst-mfld-platform: ASoC: PRE_PMD: pcm0_in event failed: -16 ... Fixes: commit b0edff42360ab4 ("ASoC: soc-pcm/soc-compress: use snd_soc_dapm_stream_stop() for SND_SOC_DAPM_STREAM_STOP") Reported-by: Chris Gorman <chrisjohgorman(a)gmail.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- > Mark This patch is for asoc/for-5.6 branch sound/soc/soc-compress.c | 2 +- sound/soc/soc-pcm.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index 223cd045719e..392a1c5b15d3 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -299,7 +299,7 @@ static int soc_compr_free_fe(struct snd_compr_stream *cstream) for_each_dpcm_be(fe, stream, dpcm) dpcm->state = SND_SOC_DPCM_LINK_STATE_FREE; - snd_soc_dapm_stream_stop(fe, stream); + dpcm_dapm_stream_event(fe, stream, SND_SOC_DAPM_STREAM_STOP); fe->dpcm[stream].state = SND_SOC_DPCM_STATE_CLOSE; fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_NO; diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 42fb03a99cb1..6d400bf02ca1 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2006,7 +2006,7 @@ static int dpcm_fe_dai_shutdown(struct snd_pcm_substream *substream) soc_pcm_close(substream); /* run the stream event for each BE */ - snd_soc_dapm_stream_stop(fe, stream); + dpcm_dapm_stream_event(fe, stream, SND_SOC_DAPM_STREAM_STOP); fe->dpcm[stream].state = SND_SOC_DPCM_STATE_CLOSE; dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_NO); -- 2.17.1

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[PATCH] ASoC: Intel: Skylake: Fix available clock counter incrementation
by Amadeusz Sławiński 25 Feb '20

25 Feb '20

Incrementation of avail_clk_cnt was incorrectly moved to error path. Put it back to success path. Fixes: 6ee927f2f01466 ('ASoC: Intel: Skylake: Fix NULL ptr dereference when unloading clk dev') Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> --- sound/soc/intel/skylake/skl-ssp-clk.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/intel/skylake/skl-ssp-clk.c b/sound/soc/intel/skylake/skl-ssp-clk.c index 1c0e5226cb5b..bd43885f3805 100644 --- a/sound/soc/intel/skylake/skl-ssp-clk.c +++ b/sound/soc/intel/skylake/skl-ssp-clk.c @@ -384,9 +384,11 @@ static int skl_clk_dev_probe(struct platform_device *pdev) &clks[i], clk_pdata, i); if (IS_ERR(data->clk[data->avail_clk_cnt])) { - ret = PTR_ERR(data->clk[data->avail_clk_cnt++]); + ret = PTR_ERR(data->clk[data->avail_clk_cnt]); goto err_unreg_skl_clk; } + + data->avail_clk_cnt++; } platform_set_drvdata(pdev, data); -- 2.17.1

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[PATCH 00/13] ASoC: soc-pcm cleanup step4
by Kuninori Morimoto 25 Feb '20

25 Feb '20

Hi Mark These are soc-pcm cleanup step4. [01/13] - [06/13] are posted but not yet accepted patches. To avoid conflict, I merged these into step4. These patches are already received Reviewed-by from Pierre-Louis. Subject: [PATCH 0/6] ASoC: use for_each_rtd_codec_dai() macro Date: Thu, 13 Feb 2020 13:08:07 +0900 Kuninori Morimoto (13): 1) ASoC: soundwaire: qcom: use for_each_rtd_codec_dai() macro 2) ASoC: qcom: sdm845: use for_each_rtd_codec_dai() macro 3) ASoC: qcom: apq8016_sbc: use for_each_rtd_codec_dai() macro 4) ASoC: intel: cml_rt1011_rt5682: use for_each_rtd_codec_dai() macro 5) ASoC: intel: kbl_da7219_max98927: use for_each_rtd_codec_dai() macro 6) ASoC: mediatek: mt8183-da7219-max98357: use for_each_rtd_codec_dai() macro 7) ASoC: soc-pcm: add snd_soc_dai_get_pcm_stream() 8) ASoC: soc-pcm: cleanup soc_pcm_apply_msb() 9) ASoC: soc-pcm: add snd_soc_dai_get_widget() 10) ASoC: soc-pcm: merge dpcm_run_new/old_update() into dpcm_fe_runtime_update() 11) ASoC: soc-pcm: move dpcm_path_put() to soc-pcm.c 12) ASoC: soc-pcm: move CONFIG_DEBUG_FS functions to top side 13) ASoC: soc-pcm: add dpcm_create/remove_debugfs_state() drivers/soundwire/qcom.c | 7 +- include/sound/soc-dai.h | 15 + include/sound/soc-dapm.h | 1 + include/sound/soc-dpcm.h | 7 +- sound/soc/intel/boards/cml_rt1011_rt5682.c | 3 +- sound/soc/intel/boards/kbl_da7219_max98927.c | 8 +- sound/soc/intel/skylake/skl-pcm.c | 10 +- .../mediatek/mt8183/mt8183-da7219-max98357.c | 8 +- sound/soc/qcom/apq8016_sbc.c | 7 +- sound/soc/qcom/sdm845.c | 20 +- sound/soc/soc-dai.c | 7 +- sound/soc/soc-dapm.c | 20 +- sound/soc/soc-pcm.c | 462 ++++++++---------- 13 files changed, 266 insertions(+), 309 deletions(-) -- 2.17.1 Thank you for your help !! Best regards --- Kuninori Morimoto

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[PATCH] ASoC: soc-pcm: fix state tracking error in snd_soc_component_open/close()
by Kai Vehmanen 25 Feb '20

25 Feb '20

ASoC component open/close and snd_soc_component_module_get/put are called independently for each component-substream pair, so the logic added in commit dd03907bf129 ("ASoC: soc-pcm: call snd_soc_component_open/close() once") was not sufficient and led to PCM playback and module unload errors. Implement handling of failures directly in soc_pcm_components_open(), so that any successfully opened components are closed upon error with other components. This allows to clean up error handling in soc_pcm_open() without adding more state tracking. Fixes: dd03907bf129 ("ASoC: soc-pcm: call snd_soc_component_open/close() once") Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> --- include/sound/soc-component.h | 7 ++----- sound/soc/soc-component.c | 35 +++++++---------------------------- sound/soc/soc-pcm.c | 27 +++++++++++++++++++++------ 3 files changed, 30 insertions(+), 39 deletions(-) diff --git a/include/sound/soc-component.h b/include/sound/soc-component.h index 1866ecc8e94b..154d02fbbfed 100644 --- a/include/sound/soc-component.h +++ b/include/sound/soc-component.h @@ -147,6 +147,8 @@ struct snd_soc_component { unsigned int active; + unsigned int suspended:1; /* is in suspend PM state */ + struct list_head list; struct list_head card_aux_list; /* for auxiliary bound components */ struct list_head card_list; @@ -180,11 +182,6 @@ struct snd_soc_component { struct dentry *debugfs_root; const char *debugfs_prefix; #endif - - /* bit field */ - unsigned int suspended:1; /* is in suspend PM state */ - unsigned int opened:1; - unsigned int module:1; }; #define for_each_component_dais(component, dai)\ diff --git a/sound/soc/soc-component.c b/sound/soc/soc-component.c index ee00c09df5e7..14e175cdeeb8 100644 --- a/sound/soc/soc-component.c +++ b/sound/soc/soc-component.c @@ -297,55 +297,34 @@ EXPORT_SYMBOL_GPL(snd_soc_component_set_jack); int snd_soc_component_module_get(struct snd_soc_component *component, int upon_open) { - if (component->module) - return 0; - if (component->driver->module_get_upon_open == !!upon_open && !try_module_get(component->dev->driver->owner)) return -ENODEV; - component->module = 1; - return 0; } void snd_soc_component_module_put(struct snd_soc_component *component, int upon_open) { - if (component->module && - component->driver->module_get_upon_open == !!upon_open) + if (component->driver->module_get_upon_open == !!upon_open) module_put(component->dev->driver->owner); - - component->module = 0; } int snd_soc_component_open(struct snd_soc_component *component, struct snd_pcm_substream *substream) { - int ret = 0; - - if (!component->opened && - component->driver->open) - ret = component->driver->open(component, substream); - - if (ret == 0) - component->opened = 1; - - return ret; + if (component->driver->open) + return component->driver->open(component, substream); + return 0; } int snd_soc_component_close(struct snd_soc_component *component, struct snd_pcm_substream *substream) { - int ret = 0; - - if (component->opened && - component->driver->close) - ret = component->driver->close(component, substream); - - component->opened = 0; - - return ret; + if (component->driver->close) + return component->driver->close(component, substream); + return 0; } int snd_soc_component_prepare(struct snd_soc_component *component, diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 63f67eb7c077..9e761f932887 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -469,28 +469,43 @@ static void soc_pcm_init_runtime_hw(struct snd_pcm_substream *substream) static int soc_pcm_components_open(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_component *last = NULL; struct snd_soc_component *component; int i, ret = 0; for_each_rtd_components(rtd, i, component) { + last = component; + ret = snd_soc_component_module_get_when_open(component); if (ret < 0) { dev_err(component->dev, "ASoC: can't get module %s\n", component->name); - return ret; + break; } ret = snd_soc_component_open(component, substream); if (ret < 0) { + snd_soc_component_module_put_when_close(component); dev_err(component->dev, "ASoC: can't open component %s: %d\n", component->name, ret); - return ret; + break; } } - return 0; + if (ret < 0) { + /* rollback on error */ + for_each_rtd_components(rtd, i, component) { + if (component == last) + break; + + snd_soc_component_close(component, substream); + snd_soc_component_module_put_when_close(component); + } + } + + return ret; } static int soc_pcm_components_close(struct snd_pcm_substream *substream) @@ -585,7 +600,7 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) if (ret < 0) { pr_err("ASoC: %s startup failed: %d\n", rtd->dai_link->name, ret); - goto component_err; + goto rtd_startup_err; } /* startup the audio subsystem */ @@ -681,9 +696,9 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) snd_soc_dai_shutdown(cpu_dai, substream); soc_rtd_shutdown(rtd, substream); -component_err: +rtd_startup_err: soc_pcm_components_close(substream); - +component_err: mutex_unlock(&rtd->card->pcm_mutex); for_each_rtd_components(rtd, i, component) { -- 2.17.1

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Re: [PATCH] docs: dt: fix several broken doc references
by Rob Herring 24 Feb '20

24 Feb '20

On Sun, Feb 23, 2020 at 09:59:53AM +0100, Mauro Carvalho Chehab wrote: > There are several DT doc references that require manual fixes. > I found 3 cases fixed on this patch: > > - directory named "binding/" instead of "bindings/"; > - .txt to .yaml renames; > - file renames (still on txt format); > > Signed-off-by: Mauro Carvalho Chehab <mchehab+huawei(a)kernel.org> > --- > .../devicetree/bindings/mtd/cadence-nand-controller.txt | 2 +- > .../devicetree/bindings/net/brcm,bcm7445-switch-v4.0.txt | 2 +- > Documentation/devicetree/bindings/sound/st,stm32-sai.txt | 2 +- > Documentation/devicetree/bindings/sound/st,stm32-spdifrx.txt | 2 +- > Documentation/devicetree/bindings/spi/st,stm32-spi.yaml | 2 +- > MAINTAINERS | 4 ++-- > .../devicetree/bindings/net/wireless/siliabs,wfx.txt | 2 +- > 7 files changed, 8 insertions(+), 8 deletions(-) Applied. Rob

1 0

[alsa-devel] [PATCH] ASoC: codecs: make PCM5102A codec selectable
by David Douard 24 Feb '20

24 Feb '20

The PCM5102A codec entry in the Kconfig file lacks a title/description to make it selectable in menuconfig (since it has no dependent i2s/spi sub-modules like e.g. the pcm512x one). --- sound/soc/codecs/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 7e90f5d83097..676fa62e9b7f 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -934,7 +934,7 @@ config SND_SOC_PCM3168A_SPI select REGMAP_SPI config SND_SOC_PCM5102A - tristate + tristate "Texas Instruments PCM5102A CODEC" config SND_SOC_PCM512x tristate -- 2.20.1

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[PATCH 1/2] mfd: wm8994: Fix driver operation if loaded as modules
by Marek Szyprowski 24 Feb '20

24 Feb '20

WM8994 chip has built-in regulators, which might be used for chip operation. They are controlled by a separate wm8994-regulator driver, which should be loaded before this driver calls regulator_get(), because that driver also provides consumer-supply mapping for the them. If that driver is not yet loaded, regulator core substitute them with dummy regulator, what breaks chip operation, because the built-in regulators are never enabled. Fix this by adding a comment and a call to module_request() helper. Signed-off-by: Marek Szyprowski <m.szyprowski(a)samsung.com> --- drivers/mfd/wm8994-core.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/drivers/mfd/wm8994-core.c b/drivers/mfd/wm8994-core.c index 1e9fe7d92597..f15f12d8bccc 100644 --- a/drivers/mfd/wm8994-core.c +++ b/drivers/mfd/wm8994-core.c @@ -321,6 +321,13 @@ static int wm8994_device_init(struct wm8994 *wm8994, int irq) int ret, i, patch_regs = 0; int pulls = 0; + /* + * Request a driver for built-in regulators first, because they are + * default supplies for "AVDD1" and "DCVDD". Otherwise, getting them + * fails due to lack of consumer mapping defined by that driver. + */ + request_module("wm8994_regulator"); + if (dev_get_platdata(wm8994->dev)) { pdata = dev_get_platdata(wm8994->dev); wm8994->pdata = *pdata; -- 2.17.1

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[PATCH for-next 1/2] ASoC: tlv320adcx140: Add DRE and AGC support
by Dan Murphy 24 Feb '20

24 Feb '20

The TLV320ADCx140 parts support Dynamic Range Enhancer (DRE) as defined in Section 8.3.2 of the data sheets. The DRE achieves a complete-channel dynamic range as high as 120 dB. At a system level, the DRE scheme enables far-field, high-fidelity recording of audio signals in very quiet environments and low-distortion recording in loud environments. There are 2 enables for DRE. The first is a global setting that enables the DRE engine in the device and the other enable is per channel. If the DRE is enabled globally then either DRE or AGC can be used per each configured channel. If global DRE is disabled then even setting the DRE enable bit in the channel config register will have no effect. Signed-off-by: Dan Murphy <dmurphy(a)ti.com> --- sound/soc/codecs/tlv320adcx140.c | 55 ++++++++++++++++++++++++++++++++ sound/soc/codecs/tlv320adcx140.h | 1 + 2 files changed, 56 insertions(+) diff --git a/sound/soc/codecs/tlv320adcx140.c b/sound/soc/codecs/tlv320adcx140.c index 8182c584de9c..105e51be6fe6 100644 --- a/sound/soc/codecs/tlv320adcx140.c +++ b/sound/soc/codecs/tlv320adcx140.c @@ -108,6 +108,7 @@ static const struct reg_default adcx140_reg_defaults[] = { { ADCX140_DSP_CFG0, 0x01 }, { ADCX140_DSP_CFG1, 0x40 }, { ADCX140_DRE_CFG0, 0x7b }, + { ADCX140_AGC_CFG0, 0xe7 }, { ADCX140_IN_CH_EN, 0xf0 }, { ADCX140_ASI_OUT_CH_EN, 0x00 }, { ADCX140_PWR_CFG, 0x00 }, @@ -158,6 +159,16 @@ static DECLARE_TLV_DB_SCALE(dig_vol_tlv, -10000, 50, 0); /* ADC gain. From 0 to 42 dB in 1 dB steps */ static DECLARE_TLV_DB_SCALE(adc_tlv, 0, 100, 0); +/* DRE Level. From -12 dB to -66 dB in 1 dB steps */ +static DECLARE_TLV_DB_SCALE(dre_thresh_tlv, -6600, 100, 0); +/* DRE Max Gain. From 2 dB to 26 dB in 2 dB steps */ +static DECLARE_TLV_DB_SCALE(dre_gain_tlv, 200, 200, 0); + +/* AGC Level. From -6 dB to -36 dB in 2 dB steps */ +static DECLARE_TLV_DB_SCALE(agc_thresh_tlv, -3600, 200, 0); +/* AGC Max Gain. From 3 dB to 42 dB in 3 dB steps */ +static DECLARE_TLV_DB_SCALE(agc_gain_tlv, 300, 300, 0); + static const char * const resistor_text[] = { "2.5 kOhm", "10 kOhm", "20 kOhm" }; @@ -281,6 +292,18 @@ static const struct snd_kcontrol_new adcx140_dapm_ch3_en_switch = static const struct snd_kcontrol_new adcx140_dapm_ch4_en_switch = SOC_DAPM_SINGLE("Switch", ADCX140_ASI_OUT_CH_EN, 4, 1, 0); +static const struct snd_kcontrol_new adcx140_dapm_ch1_dre_en_switch = + SOC_DAPM_SINGLE("Switch", ADCX140_CH1_CFG0, 0, 1, 0); +static const struct snd_kcontrol_new adcx140_dapm_ch2_dre_en_switch = + SOC_DAPM_SINGLE("Switch", ADCX140_CH2_CFG0, 0, 1, 0); +static const struct snd_kcontrol_new adcx140_dapm_ch3_dre_en_switch = + SOC_DAPM_SINGLE("Switch", ADCX140_CH3_CFG0, 0, 1, 0); +static const struct snd_kcontrol_new adcx140_dapm_ch4_dre_en_switch = + SOC_DAPM_SINGLE("Switch", ADCX140_CH4_CFG0, 0, 1, 0); + +static const struct snd_kcontrol_new adcx140_dapm_dre_en_switch = + SOC_DAPM_SINGLE("Switch", ADCX140_DSP_CFG1, 3, 1, 0); + /* Output Mixer */ static const struct snd_kcontrol_new adcx140_output_mixer_controls[] = { SOC_DAPM_SINGLE("Digital CH1 Switch", 0, 0, 0, 0), @@ -361,6 +384,18 @@ static const struct snd_soc_dapm_widget adcx140_dapm_widgets[] = { SND_SOC_DAPM_SWITCH("CH4_ASI_EN", SND_SOC_NOPM, 0, 0, &adcx140_dapm_ch4_en_switch), + SND_SOC_DAPM_SWITCH("DRE_ENABLE", SND_SOC_NOPM, 0, 0, + &adcx140_dapm_dre_en_switch), + + SND_SOC_DAPM_SWITCH("CH1_DRE_EN", SND_SOC_NOPM, 0, 0, + &adcx140_dapm_ch1_dre_en_switch), + SND_SOC_DAPM_SWITCH("CH2_DRE_EN", SND_SOC_NOPM, 0, 0, + &adcx140_dapm_ch2_dre_en_switch), + SND_SOC_DAPM_SWITCH("CH3_DRE_EN", SND_SOC_NOPM, 0, 0, + &adcx140_dapm_ch3_dre_en_switch), + SND_SOC_DAPM_SWITCH("CH4_DRE_EN", SND_SOC_NOPM, 0, 0, + &adcx140_dapm_ch4_dre_en_switch), + SND_SOC_DAPM_MUX("IN1 Analog Mic Resistor", SND_SOC_NOPM, 0, 0, in1_resistor_controls), SND_SOC_DAPM_MUX("IN2 Analog Mic Resistor", SND_SOC_NOPM, 0, 0, @@ -383,6 +418,16 @@ static const struct snd_soc_dapm_route adcx140_audio_map[] = { {"CH3_ASI_EN", "Switch", "CH3_ADC"}, {"CH4_ASI_EN", "Switch", "CH4_ADC"}, + {"DRE_ENABLE", "Switch", "CH1_DRE_EN"}, + {"DRE_ENABLE", "Switch", "CH2_DRE_EN"}, + {"DRE_ENABLE", "Switch", "CH3_DRE_EN"}, + {"DRE_ENABLE", "Switch", "CH4_DRE_EN"}, + + {"CH1_DRE_EN", "Switch", "CH1_ADC"}, + {"CH2_DRE_EN", "Switch", "CH2_ADC"}, + {"CH3_DRE_EN", "Switch", "CH3_ADC"}, + {"CH4_DRE_EN", "Switch", "CH4_ADC"}, + /* Mic input */ {"CH1_ADC", NULL, "MIC_GAIN_CTL_CH1"}, {"CH2_ADC", NULL, "MIC_GAIN_CTL_CH2"}, @@ -455,6 +500,16 @@ static const struct snd_kcontrol_new adcx140_snd_controls[] = { SOC_SINGLE_TLV("Analog CH4 Mic Gain Volume", ADCX140_CH1_CFG4, 2, 42, 0, adc_tlv), + SOC_SINGLE_TLV("DRE Threshold", ADCX140_DRE_CFG0, 4, 9, 0, + dre_thresh_tlv), + SOC_SINGLE_TLV("DRE Max Gain", ADCX140_DRE_CFG0, 0, 12, 0, + dre_gain_tlv), + + SOC_SINGLE_TLV("AGC Threshold", ADCX140_AGC_CFG0, 4, 15, 0, + agc_thresh_tlv), + SOC_SINGLE_TLV("AGC Max Gain", ADCX140_AGC_CFG0, 0, 13, 0, + agc_gain_tlv), + SOC_SINGLE_TLV("Digital CH1 Out Volume", ADCX140_CH1_CFG2, 0, 0xff, 0, dig_vol_tlv), SOC_SINGLE_TLV("Digital CH2 Out Volume", ADCX140_CH2_CFG2, diff --git a/sound/soc/codecs/tlv320adcx140.h b/sound/soc/codecs/tlv320adcx140.h index 66b1c3b33f1e..6d055e55909e 100644 --- a/sound/soc/codecs/tlv320adcx140.h +++ b/sound/soc/codecs/tlv320adcx140.h @@ -84,6 +84,7 @@ #define ADCX140_DSP_CFG0 0x6b #define ADCX140_DSP_CFG1 0x6c #define ADCX140_DRE_CFG0 0x6d +#define ADCX140_AGC_CFG0 0x70 #define ADCX140_IN_CH_EN 0x73 #define ADCX140_ASI_OUT_CH_EN 0x74 #define ADCX140_PWR_CFG 0x75 -- 2.25.0

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Applied "ASoC: meson: aiu: fix semicolon.cocci warnings" to the asoc tree
by Mark Brown 24 Feb '20

24 Feb '20

The patch ASoC: meson: aiu: fix semicolon.cocci warnings has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 1640c8df0bbac9e5156132e24c0f0a932c2b2865 Mon Sep 17 00:00:00 2001 From: kbuild test robot <lkp(a)intel.com> Date: Sun, 23 Feb 2020 01:01:54 +0800 Subject: [PATCH] ASoC: meson: aiu: fix semicolon.cocci warnings sound/soc/meson/aiu-encoder-i2s.c:129:2-3: Unneeded semicolon Remove unneeded semicolon. Generated by: scripts/coccinelle/misc/semicolon.cocci Fixes: 3e25c44598aa ("ASoC: meson: aiu: add support for the Meson8 and Meson8b SoC families") Signed-off-by: kbuild test robot <lkp(a)intel.com> CC: Martin Blumenstingl <martin.blumenstingl(a)googlemail.com> Link: https://lore.kernel.org/r/20200222170154.GA119396@e50d7db646c3 Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/meson/aiu-encoder-i2s.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/meson/aiu-encoder-i2s.c b/sound/soc/meson/aiu-encoder-i2s.c index cc73b5d5c2b7..832e22d275fe 100644 --- a/sound/soc/meson/aiu-encoder-i2s.c +++ b/sound/soc/meson/aiu-encoder-i2s.c @@ -126,7 +126,7 @@ static int aiu_encoder_i2s_set_legacy_div(struct snd_soc_component *component, default: dev_err(component->dev, "Unsupported i2s divider: %u\n", bs); return -EINVAL; - }; + } snd_soc_component_update_bits(component, AIU_CLK_CTRL, AIU_CLK_CTRL_I2S_DIV, -- 2.20.1

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Re: [PATCH] docs: dt: fix several broken doc references
by Mark Brown 24 Feb '20

24 Feb '20

On Sun, Feb 23, 2020 at 09:59:53AM +0100, Mauro Carvalho Chehab wrote: > There are several DT doc references that require manual fixes. > I found 3 cases fixed on this patch: > > - directory named "binding/" instead of "bindings/"; > - .txt to .yaml renames; > - file renames (still on txt format); This seems like it should've been split up a bit. :/ Acked-by: Mark Brown <broonie(a)kernel.org>

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