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

[v3] ASoC: AMD: Use mixer control to switch between DMICs
by Akshu Agrawal 16 Jun '20

16 Jun '20

Having mixer control to switch between DMICs prevents user to initiate capture simultaneously on both the DMIcs. Earlier 2 separate devices, one for each DMIC, gave an option of using them simultaneously, which is not supported. Signed-off-by: Akshu Agrawal <akshu.agrawal(a)amd.com> --- v2: Modified "Front Mic" to "DMIC Switch" v3: Changed to using of Mux sound/soc/amd/acp3x-rt5682-max9836.c | 58 +++++++++++++++------------- 1 file changed, 31 insertions(+), 27 deletions(-) diff --git a/sound/soc/amd/acp3x-rt5682-max9836.c b/sound/soc/amd/acp3x-rt5682-max9836.c index e499c00e0c66..f745b42dfd23 100644 --- a/sound/soc/amd/acp3x-rt5682-max9836.c +++ b/sound/soc/amd/acp3x-rt5682-max9836.c @@ -188,25 +188,27 @@ static int acp3x_ec_dmic0_startup(struct snd_pcm_substream *substream) machine->cap_i2s_instance = I2S_BT_INSTANCE; snd_soc_dai_set_bclk_ratio(codec_dai, 64); - if (dmic_sel) - gpiod_set_value(dmic_sel, 0); return rt5682_clk_enable(substream); } -static int acp3x_ec_dmic1_startup(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_card *card = rtd->card; - struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); - struct acp3x_platform_info *machine = snd_soc_card_get_drvdata(card); +static int dmic_switch; - machine->cap_i2s_instance = I2S_BT_INSTANCE; - snd_soc_dai_set_bclk_ratio(codec_dai, 64); - if (dmic_sel) - gpiod_set_value(dmic_sel, 1); +static int dmic_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.integer.value[0] = dmic_switch; + return 0; +} - return rt5682_clk_enable(substream); +static int dmic_set(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + if (dmic_sel) { + dmic_switch = ucontrol->value.integer.value[0]; + gpiod_set_value(dmic_sel, dmic_switch); + } + return 0; } static void rt5682_shutdown(struct snd_pcm_substream *substream) @@ -229,11 +231,6 @@ static const struct snd_soc_ops acp3x_ec_cap0_ops = { .shutdown = rt5682_shutdown, }; -static const struct snd_soc_ops acp3x_ec_cap1_ops = { - .startup = acp3x_ec_dmic1_startup, - .shutdown = rt5682_shutdown, -}; - SND_SOC_DAILINK_DEF(acp3x_i2s, DAILINK_COMP_ARRAY(COMP_CPU("acp3x_i2s_playcap.0"))); SND_SOC_DAILINK_DEF(acp3x_bt, @@ -279,21 +276,26 @@ static struct snd_soc_dai_link acp3x_dai_5682_98357[] = { .ops = &acp3x_ec_cap0_ops, SND_SOC_DAILINK_REG(acp3x_bt, cros_ec, platform), }, - { - .name = "acp3x-ec-dmic1-capture", - .stream_name = "Capture DMIC1", - .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBS_CFS, - .dpcm_capture = 1, - .ops = &acp3x_ec_cap1_ops, - SND_SOC_DAILINK_REG(acp3x_bt, cros_ec, platform), - }, }; +static const char * const dmic_mux_text[] = { + "Front Mic", + "Rear Mic", +}; + +static SOC_ENUM_SINGLE_DECL( + acp3x_dmic_enum, SND_SOC_NOPM, 0, dmic_mux_text); + +static const struct snd_kcontrol_new acp3x_dmic_mux_control = + SOC_DAPM_ENUM_EXT("DMIC Select Mux", acp3x_dmic_enum, + dmic_get, dmic_set); + static const struct snd_soc_dapm_widget acp3x_widgets[] = { SND_SOC_DAPM_HP("Headphone Jack", NULL), SND_SOC_DAPM_SPK("Spk", NULL), SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_MUX("Dmic Mux", SND_SOC_NOPM, 0, 0, + &acp3x_dmic_mux_control), }; static const struct snd_soc_dapm_route acp3x_audio_route[] = { @@ -301,6 +303,8 @@ static const struct snd_soc_dapm_route acp3x_audio_route[] = { {"Headphone Jack", NULL, "HPOR"}, {"IN1P", NULL, "Headset Mic"}, {"Spk", NULL, "Speaker"}, + {"Dmic Mux", "Front Mic", "DMIC"}, + {"Dmic Mux", "Rear Mic", "DMIC"}, }; static const struct snd_kcontrol_new acp3x_mc_controls[] = { -- 2.20.1

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[PATCH 0/3] ASoC: improve core dmesg logs and verbosity
by Pierre-Louis Bossart 16 Jun '20

16 Jun '20

Try to both reduce useless verbosity and keep useful error reports. Pierre-Louis Bossart (3): 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 sound/soc/soc-compress.c | 4 ++-- sound/soc/soc-core.c | 4 ++-- sound/soc/soc-pcm.c | 13 +++++++------ 3 files changed, 11 insertions(+), 10 deletions(-) -- 2.20.1

2 4

[PATCH] ASoC: img: Fix a reference count leak in img_i2s_in_set_fmt
by wu000273＠umn.edu 16 Jun '20

16 Jun '20

From: Qiushi Wu <wu000273(a)umn.edu> pm_runtime_get_sync() increments the runtime PM usage counter even when it returns an error code, causing incorrect ref count if pm_runtime_put_noidle() is not called in error handling paths. Thus call pm_runtime_put_noidle() if pm_runtime_get_sync() fails. Signed-off-by: Qiushi Wu <wu000273(a)umn.edu> --- sound/soc/img/img-i2s-in.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/img/img-i2s-in.c b/sound/soc/img/img-i2s-in.c index e30b66b94bf6..0843235d73c9 100644 --- a/sound/soc/img/img-i2s-in.c +++ b/sound/soc/img/img-i2s-in.c @@ -343,8 +343,10 @@ static int img_i2s_in_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) chan_control_mask = IMG_I2S_IN_CH_CTL_CLK_TRANS_MASK; ret = pm_runtime_get_sync(i2s->dev); - if (ret < 0) + if (ret < 0) { + pm_runtime_put_noidle(i2s->dev); return ret; + } for (i = 0; i < i2s->active_channels; i++) img_i2s_in_ch_disable(i2s, i); -- 2.17.1

2 1

[PATCH v2 00/12] ASoC: soc-component: collect component functions
by Kuninori Morimoto 16 Jun '20

16 Jun '20

Hi Mark We have soc-component.c now, but still many component related functions are implemented many place. This patch-set collect these into soc-component.c. v1 -> v2 - remove soc-compress.c exchange (But I have plan to repost it) - fixup loop break issue on some functions - direct return on some functions Link: https://lore.kernel.org/r/87a71nzhy2.wl-kuninori.morimoto.gx@renesas.com Kuninori Morimoto (12): 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 include/sound/soc-component.h | 29 +- sound/soc/Makefile | 2 +- sound/soc/soc-component.c | 666 ++++++++++++++++++++++++---------- sound/soc/soc-core.c | 102 +----- sound/soc/soc-io.c | 202 ----------- sound/soc/soc-pcm.c | 114 ++---- 6 files changed, 531 insertions(+), 584 deletions(-) delete mode 100644 sound/soc/soc-io.c -- 2.17.1

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[PATCH 1/3] dt-bindings: sound: Document wm8994 endpoints
by Jonathan Bakker 16 Jun '20

16 Jun '20

The wm8994 exposes several inputs and outputs that can be used by machine drivers in their routing. Add them to the documention so they don't have been duplicated in any machine drivers bindings. Signed-off-by: Jonathan Bakker <xc-racer2(a)live.ca> --- .../devicetree/bindings/sound/wm8994.txt | 23 +++++++++++++++++++ 1 file changed, 23 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/wm8994.txt b/Documentation/devicetree/bindings/sound/wm8994.txt index 367b58ce1bb9..8fa947509c10 100644 --- a/Documentation/devicetree/bindings/sound/wm8994.txt +++ b/Documentation/devicetree/bindings/sound/wm8994.txt @@ -68,6 +68,29 @@ Optional properties: - wlf,csnaddr-pd : If present enable the internal pull-down resistor on the CS/ADDR pin. +Pins on the device (for linking into audio routes): + + * IN1LN + * IN1LP + * IN2LN + * IN2LP:VXRN + * IN1RN + * IN1RP + * IN2RN + * IN2RP:VXRP + * SPKOUTLP + * SPKOUTLN + * SPKOUTRP + * SPKOUTRN + * HPOUT1L + * HPOUT1R + * HPOUT2P + * HPOUT2N + * LINEOUT1P + * LINEOUT1N + * LINEOUT2P + * LINEOUT2N + Example: wm8994: codec@1a { -- 2.20.1

2 2

[PATCH 0/4] Fix potential issues with x86 PCM SG-buffer
by Takashi Iwai 15 Jun '20

15 Jun '20

Hi, it turned out that x86 SG-pages that are used for PCM buffers aren't safe on x86 in all cases, and we need some limitation and workarounds for non-standard configurations. This patch set is an attempt to paper over it in a minimalistic way. Takashi === Takashi Iwai (4): ALSA: pcm: Use dma_mmap_coherent() on x86, too ALSA: memalloc: Initialize all fields of snd_dma_buffer properly ALSA: memalloc: Make SG-buffer helper usable for continuous buffer, too ALSA: pcm: Use SG-buffer only when direct DMA is available include/sound/memalloc.h | 9 ++++++++- sound/core/memalloc.c | 7 +++---- sound/core/pcm_memory.c | 13 +++++++++++++ sound/core/pcm_native.c | 2 -- sound/core/sgbuf.c | 3 +++ 5 files changed, 27 insertions(+), 7 deletions(-) -- 2.16.4

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[PATCH] ASoC: soc-pcm: fix checks for multi-cpu FE dailinks
by Pierre-Louis Bossart 15 Jun '20

15 Jun '20

soc_dpcm_fe_runtime_update() is called for all dailinks, and we want to first discard all back-ends, then deal with front-ends. The existing code first reports an error with multi-cpu front-ends, and that check needs to be moved after we know that we are dealing with a front-end. BugLink: https://github.com/thesofproject/linux/issues/1970 Fixes: 6e1276a5e613d ('ASoC: Return error if the function does not support multi-cpu') Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> --- sound/soc/soc-pcm.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 2c114b4542ce..c517064f5391 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2630,15 +2630,15 @@ static int soc_dpcm_fe_runtime_update(struct snd_soc_pcm_runtime *fe, int new) int count, paths; int ret; + if (!fe->dai_link->dynamic) + return 0; + if (fe->num_cpus > 1) { dev_err(fe->dev, "%s doesn't support Multi CPU yet\n", __func__); return -EINVAL; } - if (!fe->dai_link->dynamic) - return 0; - /* only check active links */ if (!snd_soc_dai_active(asoc_rtd_to_cpu(fe, 0))) return 0; -- 2.20.1

2 1

[PATCH 1/2] ASoC: rt1015: Update rt1015 default register value according to spec modification.
by jack.yu＠realtek.com 15 Jun '20

15 Jun '20

From: Jack Yu <jack.yu(a)realtek.com> Update rt1015 default register value according to spec modification. Signed-off-by: Jack Yu <jack.yu(a)realtek.com> --- sound/soc/codecs/rt1015.c | 124 +++++++++++++++++++++++--------------- sound/soc/codecs/rt1015.h | 15 ++++- 2 files changed, 89 insertions(+), 50 deletions(-) diff --git a/sound/soc/codecs/rt1015.c b/sound/soc/codecs/rt1015.c index 67e2e944d21b..2cccb310fa96 100644 --- a/sound/soc/codecs/rt1015.c +++ b/sound/soc/codecs/rt1015.c @@ -34,30 +34,32 @@ static const struct reg_default rt1015_reg[] = { { 0x0000, 0x0000 }, { 0x0004, 0xa000 }, { 0x0006, 0x0003 }, - { 0x000a, 0x0802 }, - { 0x000c, 0x0020 }, + { 0x000a, 0x081e }, + { 0x000c, 0x0006 }, { 0x000e, 0x0000 }, { 0x0010, 0x0000 }, { 0x0012, 0x0000 }, + { 0x0014, 0x0000 }, + { 0x0016, 0x0000 }, + { 0x0018, 0x0000 }, { 0x0020, 0x8000 }, - { 0x0022, 0x471b }, - { 0x006a, 0x0000 }, - { 0x006c, 0x4020 }, + { 0x0022, 0x8043 }, { 0x0076, 0x0000 }, { 0x0078, 0x0000 }, - { 0x007a, 0x0000 }, + { 0x007a, 0x0002 }, { 0x007c, 0x10ec }, { 0x007d, 0x1015 }, { 0x00f0, 0x5000 }, - { 0x00f2, 0x0774 }, - { 0x00f3, 0x8400 }, + { 0x00f2, 0x004c }, + { 0x00f3, 0xecfe }, { 0x00f4, 0x0000 }, + { 0x00f6, 0x0400 }, { 0x0100, 0x0028 }, { 0x0102, 0xff02 }, - { 0x0104, 0x8232 }, + { 0x0104, 0xa213 }, { 0x0106, 0x200c }, - { 0x010c, 0x002f }, - { 0x010e, 0xc000 }, + { 0x010c, 0x0000 }, + { 0x010e, 0x0058 }, { 0x0111, 0x0200 }, { 0x0112, 0x0400 }, { 0x0114, 0x0022 }, @@ -65,38 +67,46 @@ static const struct reg_default rt1015_reg[] = { { 0x0118, 0x0000 }, { 0x011a, 0x0123 }, { 0x011c, 0x4567 }, - { 0x0300, 0xdddd }, - { 0x0302, 0x0000 }, - { 0x0311, 0x9330 }, - { 0x0313, 0x0000 }, - { 0x0314, 0x0000 }, + { 0x0300, 0x203d }, + { 0x0302, 0x001e }, + { 0x0311, 0x0000 }, + { 0x0313, 0x6014 }, + { 0x0314, 0x00a2 }, { 0x031a, 0x00a0 }, { 0x031c, 0x001f }, { 0x031d, 0xffff }, { 0x031e, 0x0000 }, { 0x031f, 0x0000 }, + { 0x0320, 0x0000 }, { 0x0321, 0x0000 }, - { 0x0322, 0x0000 }, - { 0x0328, 0x0000 }, - { 0x0329, 0x0000 }, - { 0x032a, 0x0000 }, - { 0x032b, 0x0000 }, - { 0x032c, 0x0000 }, - { 0x032d, 0x0000 }, - { 0x032e, 0x030e }, - { 0x0330, 0x0080 }, + { 0x0322, 0xd7df }, + { 0x0328, 0x10b2 }, + { 0x0329, 0x0175 }, + { 0x032a, 0x36ad }, + { 0x032b, 0x7e55 }, + { 0x032c, 0x0520 }, + { 0x032d, 0xaa00 }, + { 0x032e, 0x570e }, + { 0x0330, 0xe180 }, { 0x0332, 0x0034 }, - { 0x0334, 0x0000 }, - { 0x0336, 0x0000 }, + { 0x0334, 0x0001 }, + { 0x0336, 0x0010 }, + { 0x0338, 0x0000 }, + { 0x04fa, 0x0030 }, + { 0x04fc, 0x35c8 }, + { 0x04fe, 0x0800 }, + { 0x0500, 0x0400 }, + { 0x0502, 0x1000 }, + { 0x0504, 0x0000 }, { 0x0506, 0x04ff }, - { 0x0508, 0x0030 }, - { 0x050a, 0x0018 }, - { 0x0519, 0x307f }, - { 0x051a, 0xffff }, - { 0x051b, 0x4000 }, + { 0x0508, 0x0010 }, + { 0x050a, 0x001a }, + { 0x0519, 0x1c68 }, + { 0x051a, 0x0ccc }, + { 0x051b, 0x0666 }, { 0x051d, 0x0000 }, { 0x051f, 0x0000 }, - { 0x0536, 0x1000 }, + { 0x0536, 0x061c }, { 0x0538, 0x0000 }, { 0x053a, 0x0000 }, { 0x053c, 0x0000 }, @@ -110,19 +120,18 @@ static const struct reg_default rt1015_reg[] = { { 0x0544, 0x0000 }, { 0x0568, 0x0000 }, { 0x056a, 0x0000 }, - { 0x1000, 0x0000 }, - { 0x1002, 0x6505 }, + { 0x1000, 0x0040 }, + { 0x1002, 0x5405 }, { 0x1006, 0x5515 }, - { 0x1007, 0x003f }, - { 0x1009, 0x770f }, - { 0x100a, 0x01ff }, - { 0x100c, 0x0000 }, + { 0x1007, 0x05f7 }, + { 0x1009, 0x0b0a }, + { 0x100a, 0x00ef }, { 0x100d, 0x0003 }, { 0x1010, 0xa433 }, { 0x1020, 0x0000 }, - { 0x1200, 0x3d02 }, - { 0x1202, 0x0813 }, - { 0x1204, 0x0211 }, + { 0x1200, 0x5a01 }, + { 0x1202, 0x6524 }, + { 0x1204, 0x1f00 }, { 0x1206, 0x0000 }, { 0x1208, 0x0000 }, { 0x120a, 0x0000 }, @@ -130,16 +139,16 @@ static const struct reg_default rt1015_reg[] = { { 0x120e, 0x0000 }, { 0x1210, 0x0000 }, { 0x1212, 0x0000 }, - { 0x1300, 0x0701 }, - { 0x1302, 0x12f9 }, - { 0x1304, 0x3405 }, + { 0x1300, 0x10a1 }, + { 0x1302, 0x12ff }, + { 0x1304, 0x0400 }, { 0x1305, 0x0844 }, - { 0x1306, 0x1611 }, + { 0x1306, 0x4611 }, { 0x1308, 0x555e }, { 0x130a, 0x0000 }, - { 0x130c, 0x2400}, - { 0x130e, 0x7700 }, - { 0x130f, 0x0000 }, + { 0x130c, 0x2000 }, + { 0x130e, 0x0100 }, + { 0x130f, 0x0001 }, { 0x1310, 0x0000 }, { 0x1312, 0x0000 }, { 0x1314, 0x0000 }, @@ -209,6 +218,9 @@ static bool rt1015_volatile_register(struct device *dev, unsigned int reg) case RT1015_DC_CALIB_CLSD7: case RT1015_DC_CALIB_CLSD8: case RT1015_S_BST_TIMING_INTER1: + case RT1015_OSCK_STA: + case RT1015_MONO_DYNA_CTRL1: + case RT1015_MONO_DYNA_CTRL5: return true; default: @@ -224,6 +236,12 @@ static bool rt1015_readable_register(struct device *dev, unsigned int reg) case RT1015_CLK3: case RT1015_PLL1: case RT1015_PLL2: + case RT1015_DUM_RW1: + case RT1015_DUM_RW2: + case RT1015_DUM_RW3: + case RT1015_DUM_RW4: + case RT1015_DUM_RW5: + case RT1015_DUM_RW6: case RT1015_CLK_DET: case RT1015_SIL_DET: case RT1015_CUSTOMER_ID: @@ -235,6 +253,7 @@ static bool rt1015_readable_register(struct device *dev, unsigned int reg) case RT1015_PAD_DRV2: case RT1015_GAT_BOOST: case RT1015_PRO_ALT: + case RT1015_OSCK_STA: case RT1015_MAN_I2C: case RT1015_DAC1: case RT1015_DAC2: @@ -272,6 +291,13 @@ static bool rt1015_readable_register(struct device *dev, unsigned int reg) case RT1015_SMART_BST_CTRL2: case RT1015_ANA_CTRL1: case RT1015_ANA_CTRL2: + case RT1015_PWR_STATE_CTRL: + case RT1015_MONO_DYNA_CTRL: + case RT1015_MONO_DYNA_CTRL1: + case RT1015_MONO_DYNA_CTRL2: + case RT1015_MONO_DYNA_CTRL3: + case RT1015_MONO_DYNA_CTRL4: + case RT1015_MONO_DYNA_CTRL5: case RT1015_SPK_VOL: case RT1015_SHORT_DETTOP1: case RT1015_SHORT_DETTOP2: diff --git a/sound/soc/codecs/rt1015.h b/sound/soc/codecs/rt1015.h index 6fbe802082c4..8169962935a5 100644 --- a/sound/soc/codecs/rt1015.h +++ b/sound/soc/codecs/rt1015.h @@ -21,6 +21,12 @@ #define RT1015_CLK3 0x0006 #define RT1015_PLL1 0x000a #define RT1015_PLL2 0x000c +#define RT1015_DUM_RW1 0x000e +#define RT1015_DUM_RW2 0x0010 +#define RT1015_DUM_RW3 0x0012 +#define RT1015_DUM_RW4 0x0014 +#define RT1015_DUM_RW5 0x0016 +#define RT1015_DUM_RW6 0x0018 #define RT1015_CLK_DET 0x0020 #define RT1015_SIL_DET 0x0022 #define RT1015_CUSTOMER_ID 0x0076 @@ -32,6 +38,7 @@ #define RT1015_PAD_DRV2 0x00f2 #define RT1015_GAT_BOOST 0x00f3 #define RT1015_PRO_ALT 0x00f4 +#define RT1015_OSCK_STA 0x00f6 #define RT1015_MAN_I2C 0x0100 #define RT1015_DAC1 0x0102 #define RT1015_DAC2 0x0104 @@ -70,7 +77,13 @@ #define RT1015_ANA_CTRL1 0x0334 #define RT1015_ANA_CTRL2 0x0336 #define RT1015_PWR_STATE_CTRL 0x0338 -#define RT1015_SPK_VOL 0x0506 +#define RT1015_MONO_DYNA_CTRL 0x04fa +#define RT1015_MONO_DYNA_CTRL1 0x04fc +#define RT1015_MONO_DYNA_CTRL2 0x04fe +#define RT1015_MONO_DYNA_CTRL3 0x0500 +#define RT1015_MONO_DYNA_CTRL4 0x0502 +#define RT1015_MONO_DYNA_CTRL5 0x0504 +#define RT1015_SPK_VOL 0x0506 #define RT1015_SHORT_DETTOP1 0x0508 #define RT1015_SHORT_DETTOP2 0x050a #define RT1015_SPK_DC_DETECT1 0x0519 -- 2.25.1

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[PATCH v3] ASoC: SOF: Intel: hda: Clear RIRB status before reading WP
by Brent Lu 15 Jun '20

15 Jun '20

Port commit 6d011d5057ff ("ALSA: hda: Clear RIRB status before reading WP") from legacy HDA driver to fix the get response timeout issue. Current SOF driver does not suffer from this issue because sync write is enabled in hda_init. The issue will come back if the sync write is disabled for some reason. Signed-off-by: Brent Lu <brent.lu(a)intel.com> --- sound/soc/sof/intel/hda-stream.c | 9 +++++++-- 1 file changed, 7 insertions(+), 2 deletions(-) diff --git a/sound/soc/sof/intel/hda-stream.c b/sound/soc/sof/intel/hda-stream.c index 7f65dcc..1bda14c 100644 --- a/sound/soc/sof/intel/hda-stream.c +++ b/sound/soc/sof/intel/hda-stream.c @@ -653,11 +653,16 @@ irqreturn_t hda_dsp_stream_threaded_handler(int irq, void *context) if (status & AZX_INT_CTRL_EN) { rirb_status = snd_hdac_chip_readb(bus, RIRBSTS); if (rirb_status & RIRB_INT_MASK) { + /* + * Clearing the interrupt status here ensures + * that no interrupt gets masked after the RIRB + * wp is read in snd_hdac_bus_update_rirb. + */ + snd_hdac_chip_writeb(bus, RIRBSTS, + RIRB_INT_MASK); active = true; if (rirb_status & RIRB_INT_RESPONSE) snd_hdac_bus_update_rirb(bus); - snd_hdac_chip_writeb(bus, RIRBSTS, - RIRB_INT_MASK); } } #endif -- 2.7.4

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[PATCH 1/2] ASoc: q6afe: add support to get port direction
by Srinivas Kandagatla 15 Jun '20

15 Jun '20

This patch adds support to q6afe_is_rx_port() to get direction of DSP BE dai port, this is useful for setting dailink directions correctly. Fixes: c25e295cd77b (ASoC: qcom: Add support to parse common audio device nodes") Reported-by: John Stultz <john.stultz(a)linaro.org> Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- sound/soc/qcom/qdsp6/q6afe.c | 8 ++++++++ sound/soc/qcom/qdsp6/q6afe.h | 1 + 2 files changed, 9 insertions(+) diff --git a/sound/soc/qcom/qdsp6/q6afe.c b/sound/soc/qcom/qdsp6/q6afe.c index e0945f7a58c8..0ce4eb60f984 100644 --- a/sound/soc/qcom/qdsp6/q6afe.c +++ b/sound/soc/qcom/qdsp6/q6afe.c @@ -800,6 +800,14 @@ int q6afe_get_port_id(int index) } EXPORT_SYMBOL_GPL(q6afe_get_port_id); +int q6afe_is_rx_port(int index) +{ + if (index < 0 || index >= AFE_PORT_MAX) + return -EINVAL; + + return port_maps[index].is_rx; +} +EXPORT_SYMBOL_GPL(q6afe_is_rx_port); static int afe_apr_send_pkt(struct q6afe *afe, struct apr_pkt *pkt, struct q6afe_port *port) { diff --git a/sound/soc/qcom/qdsp6/q6afe.h b/sound/soc/qcom/qdsp6/q6afe.h index c7ed5422baff..1a0f80a14afe 100644 --- a/sound/soc/qcom/qdsp6/q6afe.h +++ b/sound/soc/qcom/qdsp6/q6afe.h @@ -198,6 +198,7 @@ int q6afe_port_start(struct q6afe_port *port); int q6afe_port_stop(struct q6afe_port *port); void q6afe_port_put(struct q6afe_port *port); int q6afe_get_port_id(int index); +int q6afe_is_rx_port(int index); void q6afe_hdmi_port_prepare(struct q6afe_port *port, struct q6afe_hdmi_cfg *cfg); void q6afe_slim_port_prepare(struct q6afe_port *port, -- 2.21.0

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