April 2019 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH v2 0/1] ASoC: Mediatek: MT8183: add I2S2 control path from UL2 and UL3
by Shunli Wang 02 May '19

02 May '19

ASoC: Mediatek: MT8183: add I2S2 control path from UL2 and UL3 Shunli Wang (1): ASoC: Mediatek: MT8183: add I2S2 control path from UL2 and UL3 sound/soc/mediatek/mt8183/mt8183-afe-pcm.c | 38 ++++++++++++++++++++++ 1 file changed, 38 insertions(+) -- 2.18.0

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[alsa-devel] [PATCH] ASoC: tlv320aic32x4: Fix potential uninitialized variable
by Annaliese McDermond 02 May '19

02 May '19

Fix compiler warning about uninitialized variable reported by Stephen Rothwell <sfr(a)canb.auug.org.au>. Signed-off-by: Annaliese McDermond <nh6z(a)nh6z.net> --- sound/soc/codecs/tlv320aic32x4-clk.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/tlv320aic32x4-clk.c b/sound/soc/codecs/tlv320aic32x4-clk.c index 9e4899eb1d8e..156c153c12ab 100644 --- a/sound/soc/codecs/tlv320aic32x4-clk.c +++ b/sound/soc/codecs/tlv320aic32x4-clk.c @@ -82,7 +82,7 @@ static int clk_aic32x4_pll_get_muldiv(struct clk_aic32x4 *pll, int ret; ret = regmap_read(pll->regmap, AIC32X4_PLLPR, &val); - if (ret) + if (ret < 0) return ret; settings->r = val & AIC32X4_PLL_R_MASK; settings->p = (val & AIC32X4_PLL_P_MASK) >> AIC32X4_PLL_P_SHIFT; -- 2.19.1

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[alsa-devel] [PATCH v4 0/4] ASoC: fsl: audmix: remove "model" attribute and fix ref leaks
by Viorel Suman 02 May '19

02 May '19

The latest audmix patch-set (v5) had the "model" attribute removed as requested by Nicolin Chen, but looks like (v4) version of DAI driver reached "for-next" branch - fix this by removing "model" attribute. Asside of this fix object reference leaks in machine probe reported by Julia Lawall. Viorel Suman (4): ASoC: fsl_audmix: remove "model" attribute ASoC: fsl_audmix: remove "model" attribute from DT document ASoC: imx-audmix: fix object reference leaks in probe ASoC: fsl_audmix: cache pdev->dev pointer Changes since V1: a) Removed "model" attribute from dt-bindings documentation b) Adressed Daniel's comments Changes since V2: a) Cache pdev->dev pointer in fsl_audmix probe as suggested by Nicolin Changes since V3: a) Use subject lines matching the style for the subsystem. .../devicetree/bindings/sound/fsl,audmix.txt | 4 -- sound/soc/fsl/fsl_audmix.c | 60 +++++++++++----------- sound/soc/fsl/imx-audmix.c | 4 ++ 3 files changed, 35 insertions(+), 33 deletions(-) -- 2.7.4

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[alsa-devel] [PATCH] ASoC: dapm: fix kcontrols for effect widgets
by Pierre-Louis Bossart 02 May '19

02 May '19

From: Seppo Ingalsuo <seppo.ingalsuo(a)linux.intel.com> This patch adds the handling of snd_soc_dapm_effect that was missing. Signed-off-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Signed-off-by: Seppo Ingalsuo <seppo.ingalsuo(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/soc-dapm.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 65ee0bb5dd0b..81a7a12196ff 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -883,6 +883,7 @@ static int dapm_create_or_share_kcontrol(struct snd_soc_dapm_widget *w, case snd_soc_dapm_switch: case snd_soc_dapm_mixer: case snd_soc_dapm_pga: + case snd_soc_dapm_effect: case snd_soc_dapm_out_drv: wname_in_long_name = true; kcname_in_long_name = true; @@ -2370,6 +2371,7 @@ static ssize_t dapm_widget_show_component(struct snd_soc_component *cmpnt, case snd_soc_dapm_dac: case snd_soc_dapm_adc: case snd_soc_dapm_pga: + case snd_soc_dapm_effect: case snd_soc_dapm_out_drv: case snd_soc_dapm_mixer: case snd_soc_dapm_mixer_named_ctl: @@ -3197,6 +3199,7 @@ int snd_soc_dapm_new_widgets(struct snd_soc_card *card) dapm_new_mux(w); break; case snd_soc_dapm_pga: + case snd_soc_dapm_effect: case snd_soc_dapm_out_drv: dapm_new_pga(w); break; -- 2.17.1

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[alsa-devel] [PATCH -next] ASoC: sprd: Fix to use list_for_each_entry_safe() when delete items
by Wei Yongjun 02 May '19

02 May '19

Since we will remove items off the list using list_del() we need to use a safe version of the list_for_each_entry() macro aptly named list_for_each_entry_safe(). Fixes: d7bff893e04f ("ASoC: sprd: Add Spreadtrum multi-channel data transfer support") Signed-off-by: Wei Yongjun <weiyongjun1(a)huawei.com> --- sound/soc/sprd/sprd-mcdt.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/sprd/sprd-mcdt.c b/sound/soc/sprd/sprd-mcdt.c index 28f5e649733d..df250f7f2b6f 100644 --- a/sound/soc/sprd/sprd-mcdt.c +++ b/sound/soc/sprd/sprd-mcdt.c @@ -978,12 +978,12 @@ static int sprd_mcdt_probe(struct platform_device *pdev) static int sprd_mcdt_remove(struct platform_device *pdev) { - struct sprd_mcdt_chan *temp; + struct sprd_mcdt_chan *chan, *temp; mutex_lock(&sprd_mcdt_list_mutex); - list_for_each_entry(temp, &sprd_mcdt_chan_list, list) - list_del(&temp->list); + list_for_each_entry_safe(chan, temp, &sprd_mcdt_chan_list, list) + list_del(&chan->list); mutex_unlock(&sprd_mcdt_list_mutex);

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[alsa-devel] [PATCH -next] ASoC: sprd: Fix return value check in sprd_mcdt_probe()
by Wei Yongjun 02 May '19

02 May '19

In case of error, the function devm_ioremap_resource() returns ERR_PTR() and never returns NULL. The NULL test in the return value check should be replaced with IS_ERR(). Fixes: d7bff893e04f ("ASoC: sprd: Add Spreadtrum multi-channel data transfer support") Signed-off-by: Wei Yongjun <weiyongjun1(a)huawei.com> --- sound/soc/sprd/sprd-mcdt.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/sprd/sprd-mcdt.c b/sound/soc/sprd/sprd-mcdt.c index 28f5e649733d..e9318d7a4810 100644 --- a/sound/soc/sprd/sprd-mcdt.c +++ b/sound/soc/sprd/sprd-mcdt.c @@ -951,8 +951,8 @@ static int sprd_mcdt_probe(struct platform_device *pdev) res = platform_get_resource(pdev, IORESOURCE_MEM, 0); mcdt->base = devm_ioremap_resource(&pdev->dev, res); - if (!mcdt->base) - return -ENOMEM; + if (IS_ERR(mcdt->base)) + return PTR_ERR(mcdt->base); mcdt->dev = &pdev->dev; spin_lock_init(&mcdt->lock);

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[alsa-devel] [PATCH V6] ASoC: fsl_esai: Fix missing break in switch statement
by S.j. Wang 02 May '19

02 May '19

case ESAI_HCKT_EXTAL and case ESAI_HCKR_EXTAL should be independent of each other, so replace fall-through with break. Fixes: 43d24e76b698 ("ASoC: fsl_esai: Add ESAI CPU DAI driver") Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> Acked-by: Nicolin Chen <nicoleotsuka(a)gmail.com> Cc: <stable(a)vger.kernel.org> --- Changes in V6 - resend base one for-5.2 Changes in v5 - remove new line after Fixes Changes in v4 - Add acked-by Changes in v3 - Update subject line and cc stable Changes in v2 - Fix "Fixes" tag sound/soc/fsl/fsl_esai.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index c7410bbfd2af..bad0dfed6b68 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -251,7 +251,7 @@ static int fsl_esai_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id, break; case ESAI_HCKT_EXTAL: ecr |= ESAI_ECR_ETI; - /* fall through */ + break; case ESAI_HCKR_EXTAL: ecr |= esai_priv->synchronous ? ESAI_ECR_ETI : ESAI_ECR_ERI; break; -- 1.9.1

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[alsa-devel] [PATCH v2] ASoC: tlv320aic3x: Add support for high power analog output
by Saravanan Sekar 02 May '19

02 May '19

Add support to output level control for the analog high power output drivers HPOUT and HPCOM. Signed-off-by: Saravanan Sekar <sravanhome(a)gmail.com> --- Notes: Changes in V2: - Removed power control as it is handled by DAPM - Added level control for left channel sound/soc/codecs/tlv320aic3x.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 516d17cb2182..90f53f9b5c2f 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -324,6 +324,9 @@ static DECLARE_TLV_DB_SCALE(adc_tlv, 0, 50, 0); */ static DECLARE_TLV_DB_SCALE(output_stage_tlv, -5900, 50, 1); +/* HP/HPCOM volumes. From 0 to 9 dB in 1 dB steps */ +static DECLARE_TLV_DB_SCALE(hp_tlv, 0, 100, 0); + static const struct snd_kcontrol_new aic3x_snd_controls[] = { /* Output */ SOC_DOUBLE_R_TLV("PCM Playback Volume", @@ -419,6 +422,12 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = { /* Pop reduction */ SOC_ENUM("Output Driver Power-On time", aic3x_poweron_time_enum), SOC_ENUM("Output Driver Ramp-up step", aic3x_rampup_step_enum), + + /* Analog HPOUT, HPCOM output level controls */ + SOC_DOUBLE_R_TLV("HP Playback Volume", HPLOUT_CTRL, HPROUT_CTRL, + 4, 9, 0, hp_tlv); + SOC_DOUBLE_R_TLV("HPCOM Playback Volume", HPLCOM_CTRL, HPRCOM_CTRL, + 4, 9, 0, hp_tlv); }; /* For other than tlv320aic3104 */ -- 2.17.1

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[alsa-devel] Correct git tree for ALSA development
by Ross Zwisler 02 May '19

02 May '19

Hi, I'm trying to ramp up a bit on ALSA development, and I noticed that many sound-related entries in MAINTAINERS still refer to alsa-kernel.git hosted on alsa-project.org, i.e.: BT87X AUDIO DRIVER M: Clemens Ladisch <clemens(a)ladisch.de> L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) T: git git://git.alsa-project.org/alsa-kernel.git S: Maintained F: Documentation/sound/cards/bt87x.rst F: sound/pci/bt87x.c >From the alsa-project.org pages: https://alsa-project.org/wiki/Download#Developers:_GIT_access https://alsa-project.org/wiki/GIT_Server#For_developers_-_kernel_drivers it seems like current development/maintenance actually happens somewhere else, possibly on Mark Brown's kernel.org tree? https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git/ If so, should we update those stale MAINTAINERS entries? Thanks, - Ross

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[alsa-devel] [PATCH] ASoC: da7219: Use clk_round_rate to handle enabled bclk/wclk case
by Adam Thomson 02 May '19

02 May '19

For some platforms where DA7219 is the DAI clock master, BCLK/WCLK will be set and enabled prior to the codec's hw_params() function being called. It is possible the platform requires a different BCLK/WCLK configuration than would be chosen by hw_params(), for example S16_LE format needed with a 64-bit frame to satisfy certain devices using the clocks. To handle those kinds of scenarios, the use of clk_round_rate() is now employed as part of hw_params(). If BCLK/WCLk are already enabled then this function will just return the currently set rates, so the subsequent calls to clk_set_rate() will succeed and nothing changes with regards to clocking. In addition the specific recalc_rate() implementations needed updating to always give back a real value, as those functions are called as part of the clk init code and a real value is needed for the clk_round_rate() calls to work as expected. Signed-off-by: Adam Thomson <Adam.Thomson.Opensource(a)diasemi.com> --- sound/soc/codecs/da7219.c | 21 +++++++++++++++------ 1 file changed, 15 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/da7219.c b/sound/soc/codecs/da7219.c index 5f5fa34..72268f5 100644 --- a/sound/soc/codecs/da7219.c +++ b/sound/soc/codecs/da7219.c @@ -1593,6 +1593,13 @@ static int da7219_hw_params(struct snd_pcm_substream *substream, sr = params_rate(params); if (da7219->master && wclk) { + /* + * Rounding the rate here avoids failure trying to set a new + * rate on an already enabled wclk. In that instance this will + * just set the same rate as is currently in use, and so should + * continue without problem. + */ + sr = clk_round_rate(wclk, sr); ret = clk_set_rate(wclk, sr); if (ret) { dev_err(component->dev, @@ -1621,6 +1628,14 @@ static int da7219_hw_params(struct snd_pcm_substream *substream, if (bclk) { bclk_rate = frame_size * sr; + /* + * Rounding the rate here avoids failure trying to set a + * new rate on an already enabled bclk. In that + * instance this will just set the same rate as is + * currently in use, and so should continue without + * problem. + */ + bclk_rate = clk_round_rate(bclk, bclk_rate); ret = clk_set_rate(bclk, bclk_rate); if (ret) { dev_err(component->dev, @@ -1927,9 +1942,6 @@ static unsigned long da7219_wclk_recalc_rate(struct clk_hw *hw, struct snd_soc_component *component = da7219->component; u8 fs = snd_soc_component_read32(component, DA7219_SR); - if (!da7219->master) - return 0; - switch (fs & DA7219_SR_MASK) { case DA7219_SR_8000: return 8000; @@ -2016,9 +2028,6 @@ static unsigned long da7219_bclk_recalc_rate(struct clk_hw *hw, u8 bclks_per_wclk = snd_soc_component_read32(component, DA7219_DAI_CLK_MODE); - if (!da7219->master) - return 0; - switch (bclks_per_wclk & DA7219_DAI_BCLKS_PER_WCLK_MASK) { case DA7219_DAI_BCLKS_PER_WCLK_32: return parent_rate * 32; -- 1.9.1

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