October 2018 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH 1/2] ASoC: pcm3060: Add control for differential output
by Kirill Marinushkin 14 Nov '18

14 Nov '18

DAC may be switched between differential and single-ended output. Signed-off-by: Kirill Marinushkin <kmarinushkin(a)birdec.tech> --- sound/soc/codecs/pcm3060.c | 9 +++++++++ sound/soc/codecs/pcm3060.h | 1 + 2 files changed, 10 insertions(+) diff --git a/sound/soc/codecs/pcm3060.c b/sound/soc/codecs/pcm3060.c index 494d9d662be8..f792ad637ae4 100644 --- a/sound/soc/codecs/pcm3060.c +++ b/sound/soc/codecs/pcm3060.c @@ -179,6 +179,13 @@ static struct snd_soc_dai_driver pcm3060_dai[] = { /* dapm */ +static const char * const pcm3060_dapm_out_dif_txt[] = { + "Differential", "Single-Ended" +}; + +static SOC_ENUM_SINGLE_DECL(pcm3060_dapm_out_dif_enum, PCM3060_REG64, + PCM3060_REG_SHIFT_SE, pcm3060_dapm_out_dif_txt); + static DECLARE_TLV_DB_SCALE(pcm3060_dapm_tlv, -10050, 50, 1); static const struct snd_kcontrol_new pcm3060_dapm_controls[] = { @@ -195,6 +202,8 @@ static const struct snd_kcontrol_new pcm3060_dapm_controls[] = { 0, pcm3060_dapm_tlv), SOC_DOUBLE("Master Capture Switch", PCM3060_REG73, PCM3060_REG_SHIFT_MUT11, PCM3060_REG_SHIFT_MUT12, 1, 1), + + SOC_ENUM("DAC Out Differential", pcm3060_dapm_out_dif_enum), }; static const struct snd_soc_dapm_widget pcm3060_dapm_widgets[] = { diff --git a/sound/soc/codecs/pcm3060.h b/sound/soc/codecs/pcm3060.h index fd89a68aa8a7..32080ff00166 100644 --- a/sound/soc/codecs/pcm3060.h +++ b/sound/soc/codecs/pcm3060.h @@ -38,6 +38,7 @@ int pcm3060_remove(struct device *dev); #define PCM3060_REG_ADPSV 0x20 #define PCM3060_REG_DAPSV 0x10 #define PCM3060_REG_SE 0x01 +#define PCM3060_REG_SHIFT_SE 0x00 #define PCM3060_REG65 0x41 #define PCM3060_REG66 0x42 -- 2.13.6

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[alsa-devel] [PATCH 0/3] constify snd_soc_dai_ops structures
by Julia Lawall 12 Nov '18

12 Nov '18

The snd_soc_dai_ops structures are only stored in the ops field of a snd_soc_dai_driver structure, so make the snd_soc_dai_ops structures const as well. Done with the help of Coccinelle. --- drivers/soundwire/intel.c | 4 ++-- sound/soc/codecs/ak4458.c | 2 +- sound/soc/codecs/ak5558.c | 2 +- sound/soc/codecs/hdac_hda.c | 2 +- sound/soc/codecs/tas6424.c | 2 +- sound/soc/qcom/qdsp6/q6afe-dai.c | 8 ++++---- 6 files changed, 10 insertions(+), 10 deletions(-)

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[alsa-devel] [PATCH 0/1] slimbus: simplify getting .driver_data
by Wolfram Sang 07 Nov '18

07 Nov '18

I got tired of fixing this in Renesas drivers manually, so I took the big hammer. Remove this cumbersome code pattern which got copy-pasted too much already: - struct platform_device *pdev = to_platform_device(dev); - struct ep93xx_keypad *keypad = platform_get_drvdata(pdev); + struct ep93xx_keypad *keypad = dev_get_drvdata(dev); A branch, tested by buildbot, can be found here: git://git.kernel.org/pub/scm/linux/kernel/git/wsa/linux.git coccinelle/get_drvdata I have been asked if it couldn't be done for dev_set_drvdata as well. I checked it and did not find one occasion where it could be simplified like this. Not much of a surprise because driver_data is usually set in probe() functions which access struct platform_device in many other ways. I am open for other comments, suggestions, too, of course. Here is the cocci-script I created: @@ struct device* d; identifier pdev; expression *ptr; @@ ( - struct platform_device *pdev = to_platform_device(d); | - struct platform_device *pdev; ... - pdev = to_platform_device(d); ) <... when != pdev - &pdev->dev + d ...> ptr = - platform_get_drvdata(pdev) + dev_get_drvdata(d) <... when != pdev - &pdev->dev + d ...> Kind regards, Wolfram Wolfram Sang (1): slimbus: qcom-ctrl: simplify getting .driver_data drivers/slimbus/qcom-ctrl.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) -- 2.19.0

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[alsa-devel] [PATCH] ASoC: stm32: sai: fix less than zero comparison on unsigned int
by Colin King 05 Nov '18

05 Nov '18

From: Colin Ian King <colin.king(a)canonical.com> Currently, the comparison of div < 0 is always false because div is an unsigned int. Fix this by making div an int. Detected by CoverityScan, CID#1475309 ("Unsigned compared against 0") Fixes: 8307b2afd386 "(ASoC: stm32: sai: set sai as mclk clock provider") Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/stm/stm32_sai_sub.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c index 211589b0b2ef..d4825700b63f 100644 --- a/sound/soc/stm/stm32_sai_sub.c +++ b/sound/soc/stm/stm32_sai_sub.c @@ -336,8 +336,7 @@ static int stm32_sai_mclk_set_rate(struct clk_hw *hw, unsigned long rate, { struct stm32_sai_mclk_data *mclk = to_mclk_data(hw); struct stm32_sai_sub_data *sai = mclk->sai_data; - unsigned int div; - int ret; + int div, ret; div = stm32_sai_get_clk_div(sai, parent_rate, rate); if (div < 0) -- 2.19.1

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[alsa-devel] [PATCH] ASoC: AMD: constify regulator_desc structure
by Julia Lawall 05 Nov '18

05 Nov '18

The regulator_desc structure can be const as it is only passed as the second argument of devm_regulator_register and the corresponding parameter is declared as const. Done with the help of Coccinelle. Signed-off-by: Julia Lawall <Julia.Lawall(a)lip6.fr> --- sound/soc/amd/acp-da7219-max98357a.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/amd/acp-da7219-max98357a.c b/sound/soc/amd/acp-da7219-max98357a.c index 3f813ea5210a..a5daad973ce5 100644 --- a/sound/soc/amd/acp-da7219-max98357a.c +++ b/sound/soc/amd/acp-da7219-max98357a.c @@ -403,7 +403,7 @@ static struct regulator_config acp_da7219_cfg = { static struct regulator_ops acp_da7219_ops = { }; -static struct regulator_desc acp_da7219_desc = { +static const struct regulator_desc acp_da7219_desc = { .name = "reg-fixed-1.8V", .type = REGULATOR_VOLTAGE, .owner = THIS_MODULE,

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[alsa-devel] [PATCH] ASoC: omap-mcbsp: On OMAP1 and OMAP2420 only 16bit format is supported
by Peter Ujfalusi 05 Nov '18

05 Nov '18

For samples sizes of 20/24/32 bit the DRR2/DXR2 registers must be used to receive or transmit the most significant part of the 32bit sample. We can not trick the system by using the DRR2/DXR2 register offset and use 32bit element size since these (and other McBSP) registers are 16bit ones. We would need support for port_window in cyclic mode from the sDMA driver, but it is not straight forward as we would need to have the sDMA frame to cover the 32bit (DRR2+DRR1/DXR2+DXR1), but the frames must cover the ALSA period to be able to receive periodic interrupts. Since the 32bit samples are not working (DMA timeout), just remove it in case we have McBSP with reg_size of 2. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- Hi, I have noticed that on Nokia n810 32bit samples just don't work at all for audio Regards, Peter sound/soc/omap/omap-mcbsp.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index d0ebb6b9bfac..caa24c463b3e 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -862,6 +862,11 @@ static int asoc_mcbsp_probe(struct platform_device *pdev) if (ret) return ret; + if (mcbsp->pdata->reg_size == 2) { + omap_mcbsp_dai.playback.formats = SNDRV_PCM_FMTBIT_S16_LE; + omap_mcbsp_dai.capture.formats = SNDRV_PCM_FMTBIT_S16_LE; + } + ret = devm_snd_soc_register_component(&pdev->dev, &omap_mcbsp_component, &omap_mcbsp_dai, 1); -- Peter Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki. Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki

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[alsa-devel] [PATCH] ASoC: pcm3168a: cleanup hw_constraint for LEFT_J/I2S
by Kuninori Morimoto 05 Nov '18

05 Nov '18

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> hw constraint for LEFT_J/I2S are same, but had duplicated code. This code cleanup these. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/codecs/pcm3168a.c | 15 +++++---------- 1 file changed, 5 insertions(+), 10 deletions(-) diff --git a/sound/soc/codecs/pcm3168a.c b/sound/soc/codecs/pcm3168a.c index e1c3a18..115f2165 100644 --- a/sound/soc/codecs/pcm3168a.c +++ b/sound/soc/codecs/pcm3168a.c @@ -497,6 +497,10 @@ static int pcm3168a_startup(struct snd_pcm_substream *substream, unsigned int fmt; unsigned int sample_min; unsigned int channel_max; + unsigned int channel_maxs[] = { + 6, /* rx */ + 8 /* tx */ + }; if (tx) fmt = pcm3168a->dac_fmt; @@ -521,18 +525,9 @@ static int pcm3168a_startup(struct snd_pcm_substream *substream, channel_max = 2; break; case PCM3168A_FMT_LEFT_J: - sample_min = 24; - if (tx) - channel_max = 8; - else - channel_max = 6; - break; case PCM3168A_FMT_I2S: sample_min = 24; - if (tx) - channel_max = 8; - else - channel_max = 6; + channel_max = channel_maxs[tx]; break; default: sample_min = 24; -- 2.7.4

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[alsa-devel] [PATCH 0/3] ASoC: rsnd: prepare patches for BUSIF support
by Kuninori Morimoto 05 Nov '18

05 Nov '18

Hi Mark These are prepare/cleanup for for Renesas rsnd driver BUSIF support. I will post BUSIF support patch soon, but it needs Jiada Wang <jiada_wang(a)mentor.com> 's check. If it was OK, I will focus to TDM split mode. Kuninori Morimoto (3): ASoC: rsnd: move .get_status under rsnd_mod_ops ASoC: rsnd: add .get_id/.get_id_sub ASoC: rsnd: rsnd_mod_name() handles both name and ID sound/soc/sh/rcar/adg.c | 2 +- sound/soc/sh/rcar/cmd.c | 11 ++-- sound/soc/sh/rcar/core.c | 72 +++++++++++++++++------ sound/soc/sh/rcar/ctu.c | 23 +++++++- sound/soc/sh/rcar/dma.c | 47 +++++++-------- sound/soc/sh/rcar/dvc.c | 3 +- sound/soc/sh/rcar/gen.c | 12 ++-- sound/soc/sh/rcar/mix.c | 3 +- sound/soc/sh/rcar/rsnd.h | 21 +++---- sound/soc/sh/rcar/src.c | 29 +++++----- sound/soc/sh/rcar/ssi.c | 147 +++++++++++++++++++++++------------------------ sound/soc/sh/rcar/ssiu.c | 37 ++++++------ 12 files changed, 232 insertions(+), 175 deletions(-) -- 2.7.4 Best regards --- Kuninori Morimoto

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[alsa-devel] [PATCH] ASoC: intel: cht_bsw_max98090_ti: Add quirk for boards using pmc_plt_clk_0
by Hans de Goede 05 Nov '18

05 Nov '18

Some boards such as the Swanky model Chromebooks use pmc_plt_clk_0 for the mclk instead of pmc_plt_clk_3. This commit adds a DMI based quirk for this. This fixing audio no longer working on these devices after commit 648e921888ad ("clk: x86: Stop marking clocks as CLK_IS_CRITICAL") that commit fixes us unnecessary keeping unused clocks on, but in case of the Swanky that was breaking audio support since we were not using the right clock in the cht_bsw_max98090_ti machine driver. Cc: stable(a)vger.kernel.org Fixes: 648e921888ad ("clk: x86: Stop marking clocks as CLK_IS_CRITICAL") Reported-and-tested-by: Dean Wallace <duffydack73(a)gmail.com> Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 32 ++++++++++++++++++-- 1 file changed, 29 insertions(+), 3 deletions(-) diff --git a/sound/soc/intel/boards/cht_bsw_max98090_ti.c b/sound/soc/intel/boards/cht_bsw_max98090_ti.c index db6976f4ddaa..9d9f6e41d81c 100644 --- a/sound/soc/intel/boards/cht_bsw_max98090_ti.c +++ b/sound/soc/intel/boards/cht_bsw_max98090_ti.c @@ -19,6 +19,7 @@ * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ */ +#include <linux/dmi.h> #include <linux/module.h> #include <linux/platform_device.h> #include <linux/slab.h> @@ -35,6 +36,8 @@ #define CHT_PLAT_CLK_3_HZ 19200000 #define CHT_CODEC_DAI "HiFi" +#define QUIRK_PMC_PLT_CLK_0 0x01 + struct cht_mc_private { struct clk *mclk; struct snd_soc_jack jack; @@ -385,11 +388,29 @@ static struct snd_soc_card snd_soc_card_cht = { .num_controls = ARRAY_SIZE(cht_mc_controls), }; +static const struct dmi_system_id cht_max98090_quirk_table[] = { + { + /* Swanky model Chromebook (Toshiba Chromebook 2) */ + .matches = { + DMI_MATCH(DMI_PRODUCT_NAME, "Swanky"), + }, + .driver_data = (void *)QUIRK_PMC_PLT_CLK_0, + }, + {} +}; + static int snd_cht_mc_probe(struct platform_device *pdev) { + const struct dmi_system_id *dmi_id; struct device *dev = &pdev->dev; int ret_val = 0; struct cht_mc_private *drv; + const char *mclk_name; + int quirks = 0; + + dmi_id = dmi_first_match(cht_max98090_quirk_table); + if (dmi_id) + quirks = (unsigned long)dmi_id->driver_data; drv = devm_kzalloc(&pdev->dev, sizeof(*drv), GFP_KERNEL); if (!drv) @@ -411,11 +432,16 @@ static int snd_cht_mc_probe(struct platform_device *pdev) snd_soc_card_cht.dev = &pdev->dev; snd_soc_card_set_drvdata(&snd_soc_card_cht, drv); - drv->mclk = devm_clk_get(&pdev->dev, "pmc_plt_clk_3"); + if (quirks & QUIRK_PMC_PLT_CLK_0) + mclk_name = "pmc_plt_clk_0"; + else + mclk_name = "pmc_plt_clk_3"; + + drv->mclk = devm_clk_get(&pdev->dev, mclk_name); if (IS_ERR(drv->mclk)) { dev_err(&pdev->dev, - "Failed to get MCLK from pmc_plt_clk_3: %ld\n", - PTR_ERR(drv->mclk)); + "Failed to get MCLK from %s: %ld\n", + mclk_name, PTR_ERR(drv->mclk)); return PTR_ERR(drv->mclk); } -- 2.19.0

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[alsa-devel] [PATCH 1/2] ALSA: x86: Fix runtime PM for hdmi-lpe-audio
by Ville Syrjala 02 Nov '18

02 Nov '18

From: Ville Syrjälä <ville.syrjala(a)linux.intel.com> Commit 46e831abe864 ("drm/i915/lpe: Mark LPE audio runtime pm as "no callbacks"") broke runtime PM with lpe audio. We can no longer runtime suspend the GPU since the sysfs power/control for the lpe-audio device no longer exists and the device is considered always active. We can fix this by not marking the device as active. Cc: Chris Wilson <chris(a)chris-wilson.co.uk> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Fixes: 46e831abe864 ("drm/i915/lpe: Mark LPE audio runtime pm as "no callbacks"") Signed-off-by: Ville Syrjälä <ville.syrjala(a)linux.intel.com> --- sound/x86/intel_hdmi_audio.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/x86/intel_hdmi_audio.c b/sound/x86/intel_hdmi_audio.c index 83d76c345940..bbed4acaafd1 100644 --- a/sound/x86/intel_hdmi_audio.c +++ b/sound/x86/intel_hdmi_audio.c @@ -1877,7 +1877,6 @@ static int hdmi_lpe_audio_probe(struct platform_device *pdev) pm_runtime_use_autosuspend(&pdev->dev); pm_runtime_mark_last_busy(&pdev->dev); - pm_runtime_set_active(&pdev->dev); dev_dbg(&pdev->dev, "%s: handle pending notification\n", __func__); for_each_port(card_ctx, port) { -- 2.18.1

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