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[alsa-devel] [PATCH] ASoC: codecs: ad193x: make two arrays static const, makes object smaller
by Colin King 09 Sep '19

09 Sep '19

From: Colin Ian King <colin.king(a)canonical.com> Don't populate the arrays on the stack but instead make them static const. Makes the object code smaller by 37 bytes. Before: text data bss dec hex filename 16253 7200 0 23453 5b9d sound/soc/codecs/ad193x.o After: text data bss dec hex filename 16056 7360 0 23416 5b78 sound/soc/codecs/ad193x.o (gcc version 9.2.1, amd64) Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/codecs/ad193x.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/ad193x.c b/sound/soc/codecs/ad193x.c index fb04c9379b71..980e024a5720 100644 --- a/sound/soc/codecs/ad193x.c +++ b/sound/soc/codecs/ad193x.c @@ -416,7 +416,7 @@ static struct snd_soc_dai_driver ad193x_no_adc_dai = { /* codec register values to set after reset */ static void ad193x_reg_default_init(struct ad193x_priv *ad193x) { - const struct reg_sequence reg_init[] = { + static const struct reg_sequence reg_init[] = { { 0, 0x99 }, /* PLL_CLK_CTRL0: pll input: mclki/xi 12.288Mhz */ { 1, 0x04 }, /* PLL_CLK_CTRL1: no on-chip Vref */ { 2, 0x40 }, /* DAC_CTRL0: TDM mode */ @@ -432,7 +432,7 @@ static void ad193x_reg_default_init(struct ad193x_priv *ad193x) { 12, 0x00 }, /* DAC_L4_VOL: no attenuation */ { 13, 0x00 }, /* DAC_R4_VOL: no attenuation */ }; - const struct reg_sequence reg_adc_init[] = { + static const struct reg_sequence reg_adc_init[] = { { 14, 0x03 }, /* ADC_CTRL0: high-pass filter enable */ { 15, 0x43 }, /* ADC_CTRL1: sata delay=1, adc aux mode */ { 16, 0x00 }, /* ADC_CTRL2: reset */ -- 2.20.1

2 1

[alsa-devel] [PATCH] ASoC: rt1011: make array pd static const, makes object smaller
by Colin King 09 Sep '19

09 Sep '19

From: Colin Ian King <colin.king(a)canonical.com> Don't populate the array pd on the stack but instead make it static const. Makes the object code smaller by 100 bytes. Before: text data bss dec hex filename 51463 13016 128 64607 fc5f sound/soc/codecs/rt1011.o After: text data bss dec hex filename 51299 13080 128 64507 fbfb sound/soc/codecs/rt1011.o (gcc version 9.2.1, amd64) Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/codecs/rt1011.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt1011.c b/sound/soc/codecs/rt1011.c index f9235764ad32..be1e276e3631 100644 --- a/sound/soc/codecs/rt1011.c +++ b/sound/soc/codecs/rt1011.c @@ -1519,7 +1519,8 @@ static const struct snd_soc_dapm_route rt1011_dapm_routes[] = { static int rt1011_get_clk_info(int sclk, int rate) { - int i, pd[] = {1, 2, 3, 4, 6, 8, 12, 16}; + int i; + static const int pd[] = {1, 2, 3, 4, 6, 8, 12, 16}; if (sclk <= 0 || rate <= 0) return -EINVAL; -- 2.20.1

2 1

[alsa-devel] [RFC PATCH] ASoC: ams-delta: Take control over audio mute GPIO pins
by Janusz Krzysztofik 09 Sep '19

09 Sep '19

Since commit 1137ceee76ba ("ARM: OMAP1: ams-delta: Don't request unused GPIOs"), on-board audio has appeared muted. It has been discovered that believed to be unused GPIO pins "hookflash1" and "hookflash2" need to be set low for audible sound in handsfree and handset mode respectively. According to Amstrad E3 wiki, the purpose of both pins hasn't been clearly identified. Original Amstrad software used to produce a high pulse on them when the phone was taken off hook or recall was pressed. With the current findings, we can assume the pins provide a kind of audio mute function, separately for handset and handsfree operation modes. Commit 2afdb4c41d78 ("ARM: OMAP1: ams-delta: Fix audio permanently muted") attempted to fix the issue temporarily by hogging the GPIO pin "hookflash1" renamed to "audio_mute", however the fix occurred incomplete as it restored audible sound only for handsfree mode. Stop hogging that pin, rename the pins to "handsfree_mute" and "handset_mute" respectively and implement appropriate DAPM event callbacks for "Speaker" and "Earpiece" DAPM widgets. Fixes: 1137ceee76ba ("ARM: OMAP1: ams-delta: Don't request unused GPIOs") Signed-off-by: Janusz Krzysztofik <jmkrzyszt(a)gmail.com> --- arch/arm/mach-omap1/board-ams-delta.c | 10 +++++---- sound/soc/ti/ams-delta.c | 31 +++++++++++++++++++++++++-- 2 files changed, 35 insertions(+), 6 deletions(-) diff --git a/arch/arm/mach-omap1/board-ams-delta.c b/arch/arm/mach-omap1/board-ams-delta.c index e47a6fbcfd6e..a2aa7a12b374 100644 --- a/arch/arm/mach-omap1/board-ams-delta.c +++ b/arch/arm/mach-omap1/board-ams-delta.c @@ -246,8 +246,8 @@ static struct platform_device latch2_gpio_device = { #define LATCH2_PIN_SCARD_CMDVCC 11 #define LATCH2_PIN_MODEM_NRESET 12 #define LATCH2_PIN_MODEM_CODEC 13 -#define LATCH2_PIN_AUDIO_MUTE 14 -#define LATCH2_PIN_HOOKFLASH 15 +#define LATCH2_PIN_HANDSFREE_MUTE 14 +#define LATCH2_PIN_HANDSET_MUTE 15 static struct regulator_consumer_supply modem_nreset_consumers[] = { REGULATOR_SUPPLY("RESET#", "serial8250.1"), @@ -476,6 +476,10 @@ static struct gpiod_lookup_table ams_delta_audio_gpio_table = { "hook_switch", 0), GPIO_LOOKUP(LATCH2_LABEL, LATCH2_PIN_MODEM_CODEC, "modem_codec", 0), + GPIO_LOOKUP(LATCH2_LABEL, LATCH2_PIN_HANDSFREE_MUTE, + "handsfree_mute", 0), + GPIO_LOOKUP(LATCH2_LABEL, LATCH2_PIN_HANDSET_MUTE, + "handset_mute", 0), { }, }, }; @@ -590,8 +594,6 @@ static int gpiochip_match_by_label(struct gpio_chip *chip, void *data) static struct gpiod_hog ams_delta_gpio_hogs[] = { GPIO_HOG(LATCH2_LABEL, LATCH2_PIN_KEYBRD_DATAOUT, "keybrd_dataout", GPIO_ACTIVE_HIGH, GPIOD_OUT_LOW), - GPIO_HOG(LATCH2_LABEL, LATCH2_PIN_AUDIO_MUTE, "audio_mute", - GPIO_ACTIVE_HIGH, GPIOD_OUT_LOW), {}, }; diff --git a/sound/soc/ti/ams-delta.c b/sound/soc/ti/ams-delta.c index dee8fc70a64f..8e2fb81ad05c 100644 --- a/sound/soc/ti/ams-delta.c +++ b/sound/soc/ti/ams-delta.c @@ -23,14 +23,31 @@ #include "omap-mcbsp.h" #include "../codecs/cx20442.h" +static struct gpio_desc *handset_mute; +static struct gpio_desc *handsfree_mute; + +static int ams_delta_event_handset(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + gpiod_set_value_cansleep(handset_mute, !SND_SOC_DAPM_EVENT_ON(event)); + return 0; +} + +static int ams_delta_event_handsfree(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + gpiod_set_value_cansleep(handsfree_mute, !SND_SOC_DAPM_EVENT_ON(event)); + return 0; +} + /* Board specific DAPM widgets */ static const struct snd_soc_dapm_widget ams_delta_dapm_widgets[] = { /* Handset */ SND_SOC_DAPM_MIC("Mouthpiece", NULL), - SND_SOC_DAPM_HP("Earpiece", NULL), + SND_SOC_DAPM_HP("Earpiece", ams_delta_event_handset), /* Handsfree/Speakerphone */ SND_SOC_DAPM_MIC("Microphone", NULL), - SND_SOC_DAPM_SPK("Speaker", NULL), + SND_SOC_DAPM_SPK("Speaker", ams_delta_event_handsfree), }; /* How they are connected to codec pins */ @@ -542,6 +559,16 @@ static int ams_delta_probe(struct platform_device *pdev) card->dev = &pdev->dev; + handset_mute = devm_gpiod_get(card->dev, "handset_mute", + GPIOD_OUT_HIGH); + if (IS_ERR(handset_mute)) + return PTR_ERR(handset_mute); + + handsfree_mute = devm_gpiod_get(card->dev, "handsfree_mute", + GPIOD_OUT_HIGH); + if (IS_ERR(handsfree_mute)) + return PTR_ERR(handsfree_mute); + ret = snd_soc_register_card(card); if (ret) { dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); -- 2.21.0

3 2

[alsa-devel] [PATCH 1/2] ASoC: es8316: fix redundant codes of clock
by Katsuhiro Suzuki 09 Sep '19

09 Sep '19

This patch removes redundant null checks for optional MCLK clock. And fix DT binding document for changing clock property to optional from required. Signed-off-by: Katsuhiro Suzuki <katsuhiro(a)katsuster.net> --- .../bindings/sound/everest,es8316.txt | 3 ++ sound/soc/codecs/es8316.c | 31 ++++++++----------- 2 files changed, 16 insertions(+), 18 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/everest,es8316.txt b/Documentation/devicetree/bindings/sound/everest,es8316.txt index aefcff9c48a2..1bf03c5f2af4 100644 --- a/Documentation/devicetree/bindings/sound/everest,es8316.txt +++ b/Documentation/devicetree/bindings/sound/everest,es8316.txt @@ -6,6 +6,9 @@ Required properties: - compatible : should be "everest,es8316" - reg : the I2C address of the device for I2C + +Optional properties: + - clocks : a list of phandle, should contain entries for clock-names - clock-names : should include as follows: "mclk" : master clock (MCLK) of the device diff --git a/sound/soc/codecs/es8316.c b/sound/soc/codecs/es8316.c index d07d50f51b28..1424ecd6952c 100644 --- a/sound/soc/codecs/es8316.c +++ b/sound/soc/codecs/es8316.c @@ -373,11 +373,9 @@ static int es8316_set_dai_sysclk(struct snd_soc_dai *codec_dai, if (freq == 0) return 0; - if (es8316->mclk) { - ret = clk_set_rate(es8316->mclk, freq); - if (ret) - return ret; - } + ret = clk_set_rate(es8316->mclk, freq); + if (ret) + return ret; /* Limit supported sample rates to ones that can be autodetected * by the codec running in slave mode. @@ -712,20 +710,18 @@ static int es8316_probe(struct snd_soc_component *component) es8316->component = component; - es8316->mclk = devm_clk_get(component->dev, "mclk"); - if (PTR_ERR(es8316->mclk) == -EPROBE_DEFER) - return -EPROBE_DEFER; + es8316->mclk = devm_clk_get_optional(component->dev, "mclk"); if (IS_ERR(es8316->mclk)) { - dev_err(component->dev, "clock is invalid, ignored\n"); - es8316->mclk = NULL; + dev_err(component->dev, "unable to get mclk\n"); + return PTR_ERR(es8316->mclk); } + if (!es8316->mclk) + dev_warn(component->dev, "assuming static mclk\n"); - if (es8316->mclk) { - ret = clk_prepare_enable(es8316->mclk); - if (ret) { - dev_err(component->dev, "unable to enable clock\n"); - return ret; - } + ret = clk_prepare_enable(es8316->mclk); + if (ret) { + dev_err(component->dev, "unable to enable mclk\n"); + return ret; } /* Reset codec and enable current state machine */ @@ -754,8 +750,7 @@ static void es8316_remove(struct snd_soc_component *component) { struct es8316_priv *es8316 = snd_soc_component_get_drvdata(component); - if (es8316->mclk) - clk_disable_unprepare(es8316->mclk); + clk_disable_unprepare(es8316->mclk); } static const struct snd_soc_component_driver soc_component_dev_es8316 = { -- 2.23.0.rc1

2 3

[alsa-devel] [PATCH] ASoC: rockchip: ignore 0Hz sysclk
by Katsuhiro Suzuki 09 Sep '19

09 Sep '19

This patch ignores sysclk setting if it is 0Hz. Some codecs treat 0Hz sysclk as signal of applying no constraints. This driver does not have such feature but current implementation outputs 'Failed to set mclk' error message if machine driver sets 0Hz sysclk to this driver. Signed-off-by: Katsuhiro Suzuki <katsuhiro(a)katsuster.net> --- sound/soc/rockchip/rockchip_i2s.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c index 88ebaf6e1880..af2d5a6124c8 100644 --- a/sound/soc/rockchip/rockchip_i2s.c +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -419,6 +419,9 @@ static int rockchip_i2s_set_sysclk(struct snd_soc_dai *cpu_dai, int clk_id, struct rk_i2s_dev *i2s = to_info(cpu_dai); int ret; + if (freq == 0) + return 0; + ret = clk_set_rate(i2s->mclk, freq); if (ret) dev_err(i2s->dev, "Fail to set mclk %d\n", ret); -- 2.23.0.rc1

2 1

[alsa-devel] [PATCH] SoC: simple-card-utils: set 0Hz to sysclk when shutdown
by Katsuhiro Suzuki 09 Sep '19

09 Sep '19

This patch set 0Hz to sysclk when shutdown the card. Some codecs set rate constraints that derives from sysclk. This mechanism works correctly if machine drivers give fixed frequency. But simple-audio and audio-graph card set variable clock rate if 'mclk-fs' property exists. In this case, rate constraints will go bad scenario. For example a codec accepts three limited rates (mclk / 256, mclk / 384, mclk / 512). Bad scenario as follows (mclk-fs = 256): - Initialize sysclk by correct value (Ex. 12.288MHz) - Codec set constraints of PCM rate by sysclk 48kHz (1/256), 32kHz (1/384), 24kHz (1/512) - Play 48kHz sound, it's acceptable - Sysclk is not changed - Play 32kHz sound, it's acceptable - Set sysclk to 8.192MHz (= fs * mclk-fs = 32k * 256) - Codec set constraints of PCM rate by sysclk 32kHz (1/256), 21.33kHz (1/384), 16kHz (1/512) - Play 48kHz again, but it's NOT acceptable because constraints do not allow 48kHz So codecs treat 0Hz sysclk as signal of applying no constraints to avoid this problem. Signed-off-by: Katsuhiro Suzuki <katsuhiro(a)katsuster.net> --- sound/soc/generic/simple-card-utils.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index 556b1a789629..9b794775df53 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -213,10 +213,17 @@ EXPORT_SYMBOL_GPL(asoc_simple_startup); void asoc_simple_shutdown(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; struct asoc_simple_priv *priv = snd_soc_card_get_drvdata(rtd->card); struct simple_dai_props *dai_props = simple_priv_to_props(priv, rtd->num); + if (dai_props->mclk_fs) { + snd_soc_dai_set_sysclk(codec_dai, 0, 0, SND_SOC_CLOCK_IN); + snd_soc_dai_set_sysclk(cpu_dai, 0, 0, SND_SOC_CLOCK_OUT); + } + asoc_simple_clk_disable(dai_props->cpu_dai); asoc_simple_clk_disable(dai_props->codec_dai); -- 2.23.0.rc1

3 4

[alsa-devel] [PATCH resend 00/11] ASoC: soc-core cleanup
by Kuninori Morimoto 09 Sep '19

09 Sep '19

Hi Mark These are resend of soc-core cleanup patches which are not yet accepted/reviewed. These are based on latest mark/for-v5.4 branch. Kuninori Morimoto (11): ASoC: soc-core: add comment to jack at soc_remove_component() ASoC: soc-core: self contained soc_probe_link_components() ASoC: soc-core: self contained soc_remove_link_components() ASoC: soc-core: self contained soc_remove_link_dais() ASoC: soc-core: move soc_probe_dai() next to soc_remove_dai() ASoC: soc-core: add new soc_link_init() ASoC: soc-core: self contained soc_probe_link_dais() ASoC: soc-core: move soc_probe_link_dais() next to soc_remove_link_dais() ASoC: soc-core: self contained soc_bind_aux_dev() ASoC: soc-core: add soc_unbind_aux_dev() ASoC: soc-core: self contained soc_unbind_aux_dev() sound/soc/soc-core.c | 264 +++++++++++++++++++++++++++------------------------ 1 file changed, 139 insertions(+), 125 deletions(-) -- 2.7.4

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[alsa-devel] [PATCH] ASoC: rt1305: make array pd static const, makes object smaller
by Colin King 09 Sep '19

09 Sep '19

From: Colin Ian King <colin.king(a)canonical.com> Don't populate the array pd on the stack but instead make it static const. Makes the object code smaller by 93 bytes. Before: text data bss dec hex filename 38961 9784 64 48809 bea9 sound/soc/codecs/rt1305.o After: text data bss dec hex filename 38804 9848 64 48716 be4c sound/soc/codecs/rt1305.o (gcc version 9.2.1, amd64) Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/codecs/rt1305.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt1305.c b/sound/soc/codecs/rt1305.c index 9909369483f0..e27742abfa76 100644 --- a/sound/soc/codecs/rt1305.c +++ b/sound/soc/codecs/rt1305.c @@ -608,7 +608,8 @@ static const struct snd_soc_dapm_route rt1305_dapm_routes[] = { static int rt1305_get_clk_info(int sclk, int rate) { - int i, pd[] = {1, 2, 3, 4, 6, 8, 12, 16}; + int i; + static const int pd[] = {1, 2, 3, 4, 6, 8, 12, 16}; if (sclk <= 0 || rate <= 0) return -EINVAL; -- 2.20.1

2 1

[alsa-devel] [PATCH] ASoC: rt1308: make array pd static const, makes object smaller
by Colin King 09 Sep '19

09 Sep '19

From: Colin Ian King <colin.king(a)canonical.com> Don't populate the array pd on the stack but instead make it static const. Makes the object code smaller by 82 bytes. Before: text data bss dec hex filename 26548 7288 64 33900 846c sound/soc/codecs/rt1308.o After: text data bss dec hex filename 26370 7384 64 33818 841a sound/soc/codecs/rt1308.o (gcc version 9.2.1, amd64) Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/codecs/rt1308.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt1308.c b/sound/soc/codecs/rt1308.c index 6f2ee6809dbb..b75931a69a1c 100644 --- a/sound/soc/codecs/rt1308.c +++ b/sound/soc/codecs/rt1308.c @@ -436,7 +436,8 @@ static const struct snd_soc_dapm_route rt1308_dapm_routes[] = { static int rt1308_get_clk_info(int sclk, int rate) { - int i, pd[] = {1, 2, 3, 4, 6, 8, 12, 16}; + int i; + static const int pd[] = {1, 2, 3, 4, 6, 8, 12, 16}; if (sclk <= 0 || rate <= 0) return -EINVAL; -- 2.20.1

2 1

[alsa-devel] [PATCH v2] drm: bridge/dw_hdmi: add audio sample channel status setting
by Cheng-Yi Chiang 09 Sep '19

09 Sep '19

From: Yakir Yang <ykk(a)rock-chips.com> When transmitting IEC60985 linear PCM audio, we configure the Aduio Sample Channel Status information of all the channel status bits in the IEC60958 frame. Refer to 60958-3 page 10 for frequency, original frequency, and wordlength setting. This fix the issue that audio does not come out on some monitors (e.g. LG 22CV241) Note that these registers are only for interfaces: I2S audio interface, General Purpose Audio (GPA), or AHB audio DMA (AHBAUDDMA). For S/PDIF interface this information comes from the stream. Currently this function dw_hdmi_set_channel_status is only called from dw-hdmi-i2s-audio in I2S setup. Signed-off-by: Yakir Yang <ykk(a)rock-chips.com> Signed-off-by: Cheng-Yi Chiang <cychiang(a)chromium.org> --- Original patch by Yakir Yang is at https://lore.kernel.org/patchwork/patch/539653/ Change from v1 to v2: 1. Remove the version check because this will only be called by dw-hdmi-i2s-audio, and the registers are available in I2S setup. 2. Set these registers in dw_hdmi_i2s_hw_params. 3. Fix the sample width setting so it can use 16 or 24 bits. .../drm/bridge/synopsys/dw-hdmi-i2s-audio.c | 1 + drivers/gpu/drm/bridge/synopsys/dw-hdmi.c | 70 +++++++++++++++++++ drivers/gpu/drm/bridge/synopsys/dw-hdmi.h | 20 ++++++ include/drm/bridge/dw_hdmi.h | 2 + 4 files changed, 93 insertions(+) diff --git a/drivers/gpu/drm/bridge/synopsys/dw-hdmi-i2s-audio.c b/drivers/gpu/drm/bridge/synopsys/dw-hdmi-i2s-audio.c index 34d8e837555f..b801a28b0f17 100644 --- a/drivers/gpu/drm/bridge/synopsys/dw-hdmi-i2s-audio.c +++ b/drivers/gpu/drm/bridge/synopsys/dw-hdmi-i2s-audio.c @@ -102,6 +102,7 @@ static int dw_hdmi_i2s_hw_params(struct device *dev, void *data, } dw_hdmi_set_sample_rate(hdmi, hparms->sample_rate); + dw_hdmi_set_channel_status(hdmi, hparms->sample_width); dw_hdmi_set_channel_count(hdmi, hparms->channels); dw_hdmi_set_channel_allocation(hdmi, hparms->cea.channel_allocation); diff --git a/drivers/gpu/drm/bridge/synopsys/dw-hdmi.c b/drivers/gpu/drm/bridge/synopsys/dw-hdmi.c index bd65d0479683..d1daa369c8ae 100644 --- a/drivers/gpu/drm/bridge/synopsys/dw-hdmi.c +++ b/drivers/gpu/drm/bridge/synopsys/dw-hdmi.c @@ -582,6 +582,76 @@ static unsigned int hdmi_compute_n(unsigned int freq, unsigned long pixel_clk) return n; } +/* + * When transmitting IEC60958 linear PCM audio, these registers allow to + * configure the channel status information of all the channel status + * bits in the IEC60958 frame. For the moment this configuration is only + * used when the I2S audio interface, General Purpose Audio (GPA), + * or AHB audio DMA (AHBAUDDMA) interface is active + * (for S/PDIF interface this information comes from the stream). + */ +void dw_hdmi_set_channel_status(struct dw_hdmi *hdmi, + unsigned int sample_width) +{ + u8 aud_schnl_samplerate; + u8 aud_schnl_8; + u8 word_length_bits; + + switch (hdmi->sample_rate) { + case 32000: + aud_schnl_samplerate = HDMI_FC_AUDSCHNLS7_SMPRATE_32K; + break; + case 44100: + aud_schnl_samplerate = HDMI_FC_AUDSCHNLS7_SMPRATE_44K1; + break; + case 48000: + aud_schnl_samplerate = HDMI_FC_AUDSCHNLS7_SMPRATE_48K; + break; + case 88200: + aud_schnl_samplerate = HDMI_FC_AUDSCHNLS7_SMPRATE_88K2; + break; + case 96000: + aud_schnl_samplerate = HDMI_FC_AUDSCHNLS7_SMPRATE_96K; + break; + case 176400: + aud_schnl_samplerate = HDMI_FC_AUDSCHNLS7_SMPRATE_176K4; + break; + case 192000: + aud_schnl_samplerate = HDMI_FC_AUDSCHNLS7_SMPRATE_192K; + break; + case 768000: + aud_schnl_samplerate = HDMI_FC_AUDSCHNLS7_SMPRATE_768K; + break; + default: + dev_warn(hdmi->dev, "Unsupported audio sample rate (%u)\n", + hdmi->sample_rate); + return; + } + + /* set channel status register */ + hdmi_modb(hdmi, aud_schnl_samplerate, HDMI_FC_AUDSCHNLS7_SMPRATE_MASK, + HDMI_FC_AUDSCHNLS7); + + /* + * Set original frequency to be the same as frequency. + * Use one-complement value as stated in IEC60958-3 page 13. + */ + aud_schnl_8 = (~aud_schnl_samplerate) << + HDMI_FC_AUDSCHNLS8_ORIGSAMPFREQ_OFFSET; + + /* + * Refer to IEC60958-3 page 12. We can accept 16 bits or 24 bits. + * Otherwise, set the register to 0t o indicate using default value. + */ + word_length_bits = (sample_width == 16) ? 0x2 : + ((sample_width == 24) ? 0xb : 0); + + aud_schnl_8 |= word_length_bits << HDMI_FC_AUDSCHNLS8_WORDLEGNTH_OFFSET; + + hdmi_writeb(hdmi, aud_schnl_8, HDMI_FC_AUDSCHNLS8); +} +EXPORT_SYMBOL_GPL(dw_hdmi_set_channel_status); + static void hdmi_set_clk_regenerator(struct dw_hdmi *hdmi, unsigned long pixel_clk, unsigned int sample_rate) { diff --git a/drivers/gpu/drm/bridge/synopsys/dw-hdmi.h b/drivers/gpu/drm/bridge/synopsys/dw-hdmi.h index 6988f12d89d9..619ebc1c8354 100644 --- a/drivers/gpu/drm/bridge/synopsys/dw-hdmi.h +++ b/drivers/gpu/drm/bridge/synopsys/dw-hdmi.h @@ -158,6 +158,17 @@ #define HDMI_FC_SPDDEVICEINF 0x1062 #define HDMI_FC_AUDSCONF 0x1063 #define HDMI_FC_AUDSSTAT 0x1064 +#define HDMI_FC_AUDSV 0x1065 +#define HDMI_FC_AUDSU 0x1066 +#define HDMI_FC_AUDSCHNLS0 0x1067 +#define HDMI_FC_AUDSCHNLS1 0x1068 +#define HDMI_FC_AUDSCHNLS2 0x1069 +#define HDMI_FC_AUDSCHNLS3 0x106a +#define HDMI_FC_AUDSCHNLS4 0x106b +#define HDMI_FC_AUDSCHNLS5 0x106c +#define HDMI_FC_AUDSCHNLS6 0x106d +#define HDMI_FC_AUDSCHNLS7 0x106e +#define HDMI_FC_AUDSCHNLS8 0x106f #define HDMI_FC_DATACH0FILL 0x1070 #define HDMI_FC_DATACH1FILL 0x1071 #define HDMI_FC_DATACH2FILL 0x1072 @@ -706,6 +717,15 @@ enum { /* HDMI_FC_AUDSCHNLS7 field values */ HDMI_FC_AUDSCHNLS7_ACCURACY_OFFSET = 4, HDMI_FC_AUDSCHNLS7_ACCURACY_MASK = 0x30, + HDMI_FC_AUDSCHNLS7_SMPRATE_MASK = 0x0f, + HDMI_FC_AUDSCHNLS7_SMPRATE_192K = 0xe, + HDMI_FC_AUDSCHNLS7_SMPRATE_176K4 = 0xc, + HDMI_FC_AUDSCHNLS7_SMPRATE_96K = 0xa, + HDMI_FC_AUDSCHNLS7_SMPRATE_768K = 0x9, + HDMI_FC_AUDSCHNLS7_SMPRATE_88K2 = 0x8, + HDMI_FC_AUDSCHNLS7_SMPRATE_32K = 0x3, + HDMI_FC_AUDSCHNLS7_SMPRATE_48K = 0x2, + HDMI_FC_AUDSCHNLS7_SMPRATE_44K1 = 0x0, /* HDMI_FC_AUDSCHNLS8 field values */ HDMI_FC_AUDSCHNLS8_ORIGSAMPFREQ_MASK = 0xf0, diff --git a/include/drm/bridge/dw_hdmi.h b/include/drm/bridge/dw_hdmi.h index cf528c289857..12144d2f80f4 100644 --- a/include/drm/bridge/dw_hdmi.h +++ b/include/drm/bridge/dw_hdmi.h @@ -156,6 +156,8 @@ void dw_hdmi_setup_rx_sense(struct dw_hdmi *hdmi, bool hpd, bool rx_sense); void dw_hdmi_set_sample_rate(struct dw_hdmi *hdmi, unsigned int rate); void dw_hdmi_set_channel_count(struct dw_hdmi *hdmi, unsigned int cnt); +void dw_hdmi_set_channel_status(struct dw_hdmi *hdmi, + unsigned int sample_width); void dw_hdmi_set_channel_allocation(struct dw_hdmi *hdmi, unsigned int ca); void dw_hdmi_audio_enable(struct dw_hdmi *hdmi); void dw_hdmi_audio_disable(struct dw_hdmi *hdmi); -- 2.23.0.187.g17f5b7556c-goog

3 5

[alsa-devel] [PATCH v2 0/6] ] ASoC: atmel: extend SSC support
by Michał Mirosław 08 Sep '19

08 Sep '19

This series improves support for various configurations using SSC module as implemented in Atmel SAMA5Dx SoCs. Patches are: 1. enable SSC in Kconfig for audio-graph-card support 2. DRY mode setting code 3. implement left-justified data mode 4-6. enable shared FSYNC source for slave mode Patches against tiwai/sound/for-next tree. You can also pull from https://rere.qmqm.pl/git/linux branch: atmel-ssc Michał Mirosław (6): ASoC: atmel: enable SOC_SSC_PDC and SOC_SSC_DMA in Kconfig ASoC: atmel_ssc_dai: rework DAI format configuration ASoC: atmel_ssc_dai: implement left-justified data mode dt-bindings: misc: atmel-ssc: LRCLK from TF/RF pin option misc: atmel-ssc: get LRCLK pin selection from DT ASoC: atmel_ssc_dai: Enable shared FSYNC source in frame-slave mode .../devicetree/bindings/misc/atmel-ssc.txt | 5 + drivers/misc/atmel-ssc.c | 9 + include/linux/atmel-ssc.h | 2 + sound/soc/atmel/Kconfig | 30 +- sound/soc/atmel/atmel_ssc_dai.c | 305 ++++++------------ 5 files changed, 137 insertions(+), 214 deletions(-) -- 2.20.1

5 16

[alsa-devel] [PATCH] alsa: lx6464es - add support for LX6464ESe pci express variant
by Tim Blechmann 08 Sep '19

08 Sep '19

The pci express variant of the digigram lx6464es card has a different device ID, but works without changes to the driver. Thanks to Nikolas Slottke for reporting and testing. Signed-off-by: Tim Blechmann <tim(a)klingt.org> --- include/linux/pci_ids.h | 2 ++ sound/pci/lx6464es/lx6464es.c | 8 ++++++++ 2 files changed, 10 insertions(+) diff --git a/include/linux/pci_ids.h b/include/linux/pci_ids.h index 7fa3f1498b34..08e15891ed2e 100644 --- a/include/linux/pci_ids.h +++ b/include/linux/pci_ids.h @@ -1933,6 +1933,8 @@ #define PCI_VENDOR_ID_DIGIGRAM 0x1369 #define PCI_SUBDEVICE_ID_DIGIGRAM_LX6464ES_SERIAL_SUBSYSTEM 0xc001 #define PCI_SUBDEVICE_ID_DIGIGRAM_LX6464ES_CAE_SERIAL_SUBSYSTEM 0xc002 +#define PCI_SUBDEVICE_ID_DIGIGRAM_LX6464ESE_SERIAL_SUBSYSTEM 0xc021 +#define PCI_SUBDEVICE_ID_DIGIGRAM_LX6464ESE_CAE_SERIAL_SUBSYSTEM 0xc022 #define PCI_VENDOR_ID_KAWASAKI 0x136b #define PCI_DEVICE_ID_MCHIP_KL5A72002 0xff01 diff --git a/sound/pci/lx6464es/lx6464es.c b/sound/pci/lx6464es/lx6464es.c index 9655b08a1c52..3f07e346ad41 100644 --- a/sound/pci/lx6464es/lx6464es.c +++ b/sound/pci/lx6464es/lx6464es.c @@ -65,6 +65,14 @@ static const struct pci_device_id snd_lx6464es_ids[] = { PCI_VENDOR_ID_DIGIGRAM, PCI_SUBDEVICE_ID_DIGIGRAM_LX6464ES_CAE_SERIAL_SUBSYSTEM), }, /* LX6464ES-CAE */ + { PCI_DEVICE_SUB(PCI_VENDOR_ID_PLX, PCI_DEVICE_ID_PLX_LX6464ES, + PCI_VENDOR_ID_DIGIGRAM, + PCI_SUBDEVICE_ID_DIGIGRAM_LX6464ESE_SERIAL_SUBSYSTEM), + }, /* LX6464ESe */ + { PCI_DEVICE_SUB(PCI_VENDOR_ID_PLX, PCI_DEVICE_ID_PLX_LX6464ES, + PCI_VENDOR_ID_DIGIGRAM, + PCI_SUBDEVICE_ID_DIGIGRAM_LX6464ESE_CAE_SERIAL_SUBSYSTEM), + }, /* LX6464ESe-CAE */ { 0, }, }; -- 2.18.0

2 1

[alsa-devel] Low sound and no mic in Lenovo Yoga / Ideapad S940 (Realtek ALC298)
by José Mesa Orihuela 07 Sep '19

07 Sep '19

I have tested 3 differents linux distros (manjaro, ubuntu, popos) with the same results: * Low sound volume / ouput * No mic (seems detected but doesn't work) This problem looks similar to C930. I have tried all the solutions (pin remapping, changing model...) without success. https://github.com/droserasprout/lenovo-yoga-c930-linux#speaker Alsa-info.sh log: http://alsa-project.org/db/?f=701015751be757989bb93d70a9a06f7e89d64e24 Linux lnx 4.19.66-1-MANJARO#1SMP PREEMPT Fri Aug 9 18:01:53 UTC 2019 x86_64 GNU/Lin More logs: |00:1f.3 Multimedia audio controller [0401]: Intel Corporation Cannon Point-LP High Definition Audio Controller [8086:9dc8] (rev 30) | |card 0: PCH [HDA Intel PCH], device 0: ALC298 Analog [ALC298 Analog] | |snd_hda_codec_hdmi 69632 1 snd_sof_intel_hda_common 73728 1 sof_pci_dev snd_sof_intel_hda 20480 1 snd_sof_intel_hda_common snd_sof 118784 4 snd_sof_intel_hda_common,snd_sof_intel_byt,snd_sof_intel_ipc,sof_pci_dev snd_hda_codec_realtek 126976 1 snd_hda_codec_generic 94208 1 snd_hda_codec_realtek snd_soc_hdac_hda 24576 2 snd_sof_intel_hda_common,snd_soc_skl ledtrig_audio 16384 2 snd_hda_codec_generic,snd_hda_codec_realtek snd_hda_ext_core 36864 4 snd_sof_intel_hda_common,snd_soc_hdac_hda,snd_soc_skl,snd_sof_intel_hda snd_soc_acpi_intel_match 28672 3 snd_sof_intel_hda_common,sof_pci_dev,snd_soc_skl snd_soc_core 290816 4 snd_sof,snd_sof_intel_hda_common,snd_soc_hdac_hda,snd_soc_skl snd_hda_intel 49152 8 snd_hda_codec 159744 5 snd_hda_codec_generic,snd_hda_codec_hdmi,snd_hda_intel,snd_hda_codec_realtek,snd_soc_hdac_hda snd_hda_core 102400 10 snd_hda_codec_generic,snd_hda_codec_hdmi,snd_hda_intel,snd_hda_ext_core,snd_hda_codec,snd_hda_codec_realtek,snd_sof_intel_hda_common,snd_soc_hdac_hda,snd_soc_skl,snd_sof_intel_hda snd_hwdep 20480 1 snd_hda_codec snd_pcm 139264 11 snd_hda_codec_hdmi,snd_hda_intel,snd_hda_ext_core,snd_hda_codec,snd_sof,snd_sof_intel_hda_common,snd_soc_core,snd_soc_skl,snd_hda_core,snd_pcm_dmaengine snd 110592 25 snd_hda_codec_generic,snd_hda_codec_hdmi,snd_hwdep,snd_hda_intel,snd_hda_codec,snd_hda_codec_realtek,snd_timer,snd_compress,snd_soc_core,snd_pcm | Any ideas to solve this problem? Regards

1 0

[alsa-devel] [PATCH] ALSA: firewire-lib: remove WARN_ON() at destruction of AMDTP domain
by Takashi Sakamoto 07 Sep '19

07 Sep '19

The destructor of AMDTP domain has WARN_ON() for the list of associated AMDTP stream. Although this reminds a case that developers forget to program consumer drivers to stop AMDTP domain, it hits when AMDTP domain is not initialized yet. This occurs when initialization of sound card fails as well and it's superfluous. This commit removes the WARN_ON. Although the API to AMDTP domain does nothing, it's left for future usage. Fixes: 3ec3d7a3ff106 ("ALSA: firewire-lib: add AMDTP domain structure to handle several isoc contexts") Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/amdtp-stream.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c index 88270257e896..e50e28f77e74 100644 --- a/sound/firewire/amdtp-stream.c +++ b/sound/firewire/amdtp-stream.c @@ -1153,7 +1153,8 @@ EXPORT_SYMBOL_GPL(amdtp_domain_init); */ void amdtp_domain_destroy(struct amdtp_domain *d) { - WARN_ON(!list_empty(&d->streams)); + // At present nothing to do. + return; } EXPORT_SYMBOL_GPL(amdtp_domain_destroy); -- 2.20.1

2 1

[alsa-devel] [RFC PATCH v2 0/6] adapt SOF to use snd-hda-codec-hdmi
by Kai Vehmanen 06 Sep '19

06 Sep '19

Hi all, here's a second RFC round for this series that adapts SOF (and one example machine driver) to use snd-hda-codec-hdmi (patch_hdmi.c) codec driver instead of hdac_hdmi (soc/codecs/hdac_hdmi.c). The primary goal is to unify the HDMI codec implementation between DSP and non-DSP HDA configurations, offer same interface to user-space and reduce maintenance load for all. If the approach looks ok, I'll remove RFC from v3 and update the remaining machine drivers (this v2 still changes only one machine driver and thus the RFC tag). v2 changes: - Codec selection via module parameter. Concerns were raised that a Kconfig option will not be sufficient as the patch changes user-space mixer controls. Distro kernels will want to use same kernel config for all platforms and a kconfig control would apply to all drivers. To address this, a combination of a module parameter, and a default value set via kconfig, was added to the patchset. This allows distros to support different combinations with a single kernel build. To switch codecs for a specific machine, UCM file update can be deployed together with a change to modprobe parameter. This is somewhat complicated, so further feedback is very welcome whether this is enough. - In machine drivers that are shared with SST and SOF, added runtime detection for which codec is used. For SST, the old hdac-hdmi is always used. - One patch dropped from series. This patch is no longer needed (pcm handles now passed to codec driver). Feature and testing info: - Tested on multiple Intel platforms supported by SOF. - Tested with ALSA console tools as well as with Pulseaudio. - requires Pulseaudio 12.x or newer, see https://lists.freedesktop.org/archives/pulseaudio-discuss/2019-August/03135… - HDMI, DP, DP-MST with multi-monitor use-scenarios work ok. - New feature for SOF: ELD /proc fs works just like in DSP-less mode. - New feature for SOF: jack detection works out-of-the-box with Pulseaudio (no need for card specific UCM for HDMI) - Pre-reviews at: https://github.com/thesofproject/linux/pull/1155 Kai Vehmanen (6): ALSA: hda - add mst_no_extra_pcms flag to hda_codec ASoC: Intel: skl-hda-dsp-generic: use snd-hda-codec-hdmi ASoC: hdac_hda: add support for HDMI/DP as a HDA codec ALSA: hda/hdmi - implement mst_no_extra_pcms flag ALSA: hda/hdmi - allow control creation without a linked pcm ASoC: SOF: Intel: add support for snd-hda-codec-hdmi include/sound/hda_codec.h | 1 + sound/pci/hda/patch_hdmi.c | 23 +++-- sound/soc/codecs/hdac_hda.c | 95 +++++++++++++++++--- sound/soc/codecs/hdac_hda.h | 12 ++- sound/soc/intel/boards/skl_hda_dsp_common.c | 28 +++++- sound/soc/intel/boards/skl_hda_dsp_common.h | 69 ++++++++++++++ sound/soc/intel/boards/skl_hda_dsp_generic.c | 7 -- sound/soc/sof/intel/Kconfig | 10 +++ sound/soc/sof/intel/hda-codec.c | 14 ++- sound/soc/sof/intel/hda.h | 6 +- 10 files changed, 231 insertions(+), 34 deletions(-) -- 2.17.1

3 10

[alsa-devel] [PATCH] ALSA: firewire-tascam: add note for FE-8
by Takashi Sakamoto 06 Sep '19

06 Sep '19

TASCAM FE-8 is the rest of model in TASCAM FireWire series. This device has no functionality to process audio signal and MIDI messages. Instead, it transfers control messages to host system corresponding to operations for some faders, buttons and knobs on its surface. Unlike the other devices in this series, the control messages are transmitted by asynchronous transactions. Some registers of device are used for registration of destination address for the transaction. The transaction includes quadlet-aligned data up to 32 quadlets. Userspace applications can receive the transaction and parse it for control message via Linux FireWire subsystem, without any support by ALSA firewire-tascam driver. Therefore the driver gives no support for it. This commit removes placeholder for FE-8 and add some comment for its functionalities as notes. $ python2 linux-firewire-utils/src/crpp < ~/git/am-config-rom/tascam/tascam-fe8.img ROM header and bus information block ----------------------------------------------------------------- 400 040f4798 bus_info_length 4, crc_length 15, crc 18328 (should be 14256) 404 31333934 bus_name "1394" 408 20ff7002 irmc 0, cmc 0, isc 1, bmc 0, cyc_clk_acc 255, max_rec 7 (256) 40c 00022eff company_id 00022e | 410 a094dcb7 device_id ffa094dcb7 | EUI-64 00022effa094dcb7 root directory ----------------------------------------------------------------- 414 0004bccc directory_length 4, crc 48332 418 0300022e vendor 41c 0c0083c0 node capabilities per IEEE 1394 420 8d000006 --> eui-64 leaf at 438 424 d1000001 --> unit directory at 428 unit directory at 428 ----------------------------------------------------------------- 428 0003eda4 directory_length 3, crc 60836 42c 1200022e specifier id 430 13800001 version 434 d4000004 --> dependent info directory at 444 eui-64 leaf at 438 ----------------------------------------------------------------- 438 0002461e leaf_length 2, crc 17950 43c 00022eff company_id 00022e | 440 a094dcb7 device_id ffa094dcb7 | EUI-64 00022effa094dcb7 dependent info directory at 444 ----------------------------------------------------------------- 444 0002ae47 directory_length 2, crc 44615 448 81000002 --> descriptor leaf at 450 44c 82000006 --> bus dependent info leaf at 464 descriptor leaf at 450 ----------------------------------------------------------------- 450 0004a79e leaf_length 4, crc 42910 454 00000000 textual descriptor 458 00000000 minimal ASCII 45c 54415343 "TASC" 460 414d0000 "AM" bus dependent info leaf at 464 ----------------------------------------------------------------- 464 0004a7d8 leaf_length 4, crc 42968 468 00000000 46c 00000000 470 46452d38 474 00000000 Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/tascam/tascam.c | 4 +++- sound/firewire/tascam/tascam.h | 20 ++++++++++++++++++++ 2 files changed, 23 insertions(+), 1 deletion(-) diff --git a/sound/firewire/tascam/tascam.c b/sound/firewire/tascam/tascam.c index 231052db5680..addc464503bc 100644 --- a/sound/firewire/tascam/tascam.c +++ b/sound/firewire/tascam/tascam.c @@ -39,6 +39,9 @@ static const struct snd_tscm_spec model_specs[] = { .midi_capture_ports = 2, .midi_playback_ports = 4, }, + // This kernel module doesn't support FE-8 because the most of features + // can be implemented in userspace without any specific support of this + // module. }; static int identify_model(struct snd_tscm *tscm) @@ -214,7 +217,6 @@ static const struct ieee1394_device_id snd_tscm_id_table[] = { .vendor_id = 0x00022e, .specifier_id = 0x00022e, }, - /* FE-08 requires reverse-engineering because it just has faders. */ {} }; MODULE_DEVICE_TABLE(ieee1394, snd_tscm_id_table); diff --git a/sound/firewire/tascam/tascam.h b/sound/firewire/tascam/tascam.h index 64a2e4d2bbfe..15bd335fa07f 100644 --- a/sound/firewire/tascam/tascam.h +++ b/sound/firewire/tascam/tascam.h @@ -129,6 +129,26 @@ struct snd_tscm { #define TSCM_OFFSET_MIDI_RX_QUAD 0x4000 +// Although FE-8 supports the above registers, it has no I/O interfaces for +// audio samples and music messages. Otherwise it supports another notification +// for status and control message as well as LED brightening. The message +// consists of quadlet-aligned data up to 32 quadlets. The first byte of message +// is fixed to 0x40. The second byte is between 0x00 to 0x1f and represent each +// control: +// fader: 0x00-0x07 +// button: 0x0d, 0x0e +// knob: 0x14-0x1b +// sensing: 0x0b +// +// The rest two bytes represent state of the controls; e.g. current value for +// fader and knob, bitmasks for button and sensing. +// Just after turning on, 32 quadlets messages with 0x00-0x1f are immediately +// sent in one transaction. After, several quadlets are sent in one transaction. +// +// TSCM_OFFSET_FE8_CTL_TX_ON 0x0310 +// TSCM_OFFSET_FE8_CTL_TX_ADDR_HI 0x0314 +// TSCM_OFFSET_FE8_CTL_TX_ADDR_LO 0x0318 + enum snd_tscm_clock { SND_TSCM_CLOCK_INTERNAL = 0, SND_TSCM_CLOCK_WORD = 1, -- 2.20.1

2 2

[alsa-devel] [RFC PATCH 1/3] ALSA:hda: Simplify and clear calculating SDxFMT
by Pawel Harlozinski 06 Sep '19

06 Sep '19

Set SDxFMT based only on given format, due to maxbps not always being set. Split cases for formats 20,24,32 bits. For format SNDRV_PCM_FORMAT_FLOAT_LE width is equal 32 so it will end up with same mask. Signed-off-by: Pawel Harlozinski <pawel.harlozinski(a)linux.intel.com> --- sound/hda/hdac_device.c | 11 +++++------ 1 file changed, 5 insertions(+), 6 deletions(-) diff --git a/sound/hda/hdac_device.c b/sound/hda/hdac_device.c index b26cc93e7e10..add758e11b85 100644 --- a/sound/hda/hdac_device.c +++ b/sound/hda/hdac_device.c @@ -759,14 +759,13 @@ unsigned int snd_hdac_calc_stream_format(unsigned int rate, val |= AC_FMT_BITS_16; break; case 20: + val |= AC_FMT_BITS_20; + break; case 24: + val |= AC_FMT_BITS_24; + break; case 32: - if (maxbps >= 32 || format == SNDRV_PCM_FORMAT_FLOAT_LE) - val |= AC_FMT_BITS_32; - else if (maxbps >= 24) - val |= AC_FMT_BITS_24; - else - val |= AC_FMT_BITS_20; + val |= AC_FMT_BITS_32; break; default: return 0; -- 2.17.1

4 11

[alsa-devel] [PATCH] soundwire: add back ACPI dependency
by Arnd Bergmann 06 Sep '19

06 Sep '19

Soundwire gained a warning for randconfig builds without CONFIG_ACPI during the linux-5.3-rc cycle: drivers/soundwire/slave.c:16:12: error: unused function 'sdw_slave_add' [-Werror,-Wunused-function] Add the CONFIG_ACPI dependency at the top level now. Fixes: 8676b3ca4673 ("soundwire: fix regmap dependencies and align with other serial links") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- drivers/soundwire/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/drivers/soundwire/Kconfig b/drivers/soundwire/Kconfig index f518273cfbe3..c73bfbaa2659 100644 --- a/drivers/soundwire/Kconfig +++ b/drivers/soundwire/Kconfig @@ -5,6 +5,7 @@ menuconfig SOUNDWIRE tristate "SoundWire support" + depends on ACPI help SoundWire is a 2-Pin interface with data and clock line ratified by the MIPI Alliance. SoundWire is used for transporting data -- 2.20.0

2 4

[alsa-devel] alsa-devel ML archived on lore.kernel.org
by Takashi Iwai 06 Sep '19

06 Sep '19

Hi, per popular demands, now lore.kernel.org also archives the posts on alsa-devel ML: https://lore.kernel.org/alsa-devel/ This includes the posts in the past that are found on alsa-project.org ML archive, too. A good thing is that this allows us to refer to the thread via the URL like: https://lore.kernel.org/r/$MESSAGEID thanks, Takashi

1 0

[alsa-devel] [PATCH] ALSA: hda/hdmi: remove redundant assignment to variable pcm_idx
by Colin King 05 Sep '19

05 Sep '19

From: Colin Ian King <colin.king(a)canonical.com> Variable pcm_idx is being initialized with a value that is never read and is being re-assigned immediately afterwards. The assignment is redundant and hence can be removed. Addresses-Coverity: ("Unused value") Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/pci/hda/patch_hdmi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 83b8b9d27711..bca5de78e9ad 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1440,7 +1440,7 @@ static bool update_eld(struct hda_codec *codec, struct hdmi_spec *spec = codec->spec; bool old_eld_valid = pin_eld->eld_valid; bool eld_changed; - int pcm_idx = -1; + int pcm_idx; /* for monitor disconnection, save pcm_idx firstly */ pcm_idx = per_pin->pcm_idx; -- 2.20.1

2 1

[alsa-devel] [PATCH] ASoC: qcom: common: Include link-name in error messages
by Bjorn Andersson 05 Sep '19

05 Sep '19

Reading out the link-name earlier and including it in the various error messages makes it much more convenient to figure out what links have unmet dependencies. Signed-off-by: Bjorn Andersson <bjorn.andersson(a)linaro.org> --- sound/soc/qcom/common.c | 22 +++++++++++----------- 1 file changed, 11 insertions(+), 11 deletions(-) diff --git a/sound/soc/qcom/common.c b/sound/soc/qcom/common.c index 2c7348ddbbb3..6c20bdd850f3 100644 --- a/sound/soc/qcom/common.c +++ b/sound/soc/qcom/common.c @@ -53,12 +53,18 @@ int qcom_snd_parse_of(struct snd_soc_card *card) link->num_cpus = 1; link->num_platforms = 1; + ret = of_property_read_string(np, "link-name", &link->name); + if (ret) { + dev_err(card->dev, "error getting codec dai_link name\n"); + goto err; + } + cpu = of_get_child_by_name(np, "cpu"); platform = of_get_child_by_name(np, "platform"); codec = of_get_child_by_name(np, "codec"); if (!cpu) { - dev_err(dev, "Can't find cpu DT node\n"); + dev_err(dev, "%s: Can't find cpu DT node\n", link->name); ret = -EINVAL; goto err; } @@ -66,7 +72,7 @@ int qcom_snd_parse_of(struct snd_soc_card *card) ret = of_parse_phandle_with_args(cpu, "sound-dai", "#sound-dai-cells", 0, &args); if (ret) { - dev_err(card->dev, "error getting cpu phandle\n"); + dev_err(card->dev, "%s: error getting cpu phandle\n", link->name); goto err; } link->cpus->of_node = args.np; @@ -74,7 +80,7 @@ int qcom_snd_parse_of(struct snd_soc_card *card) ret = snd_soc_of_get_dai_name(cpu, &link->cpus->dai_name); if (ret) { - dev_err(card->dev, "error getting cpu dai name\n"); + dev_err(card->dev, "%s: error getting cpu dai name\n", link->name); goto err; } @@ -83,14 +89,14 @@ int qcom_snd_parse_of(struct snd_soc_card *card) "sound-dai", 0); if (!link->platforms->of_node) { - dev_err(card->dev, "platform dai not found\n"); + dev_err(card->dev, "%s: platform dai not found\n", link->name); ret = -EINVAL; goto err; } ret = snd_soc_of_get_dai_link_codecs(dev, codec, link); if (ret < 0) { - dev_err(card->dev, "codec dai not found\n"); + dev_err(card->dev, "%s: codec dai not found\n", link->name); goto err; } link->no_pcm = 1; @@ -110,12 +116,6 @@ int qcom_snd_parse_of(struct snd_soc_card *card) } link->ignore_suspend = 1; - ret = of_property_read_string(np, "link-name", &link->name); - if (ret) { - dev_err(card->dev, "error getting codec dai_link name\n"); - goto err; - } - link->nonatomic = 1; link->dpcm_playback = 1; link->dpcm_capture = 1; -- 2.18.0

2 1

[alsa-devel] [PATCH 0/8] ASoC: meson: add sm1 support
by Jerome Brunet 05 Sep '19

05 Sep '19

This patchset prepare then add the audio support on the amlogic sm1 SoC family in ASoC. This has been tested on sei610 platform. Since this platform does not have spdif, this patchset does not include the change necessary to support spdif input or output on the sm1. Jerome Brunet (8): ASoC: meson: add sm1 compatibles ASoC: meson: add reset binding ASoC: meson: axg-frddr: expose all 8 outputs ASoC: meson: axg-toddr: expose all 8 inputs ASoC: meson: tdmin: expose all 16 inputs ASoC: meson: axg-frddr: add sm1 support ASoC: meson: axg-toddr: add sm1 support ASoC: meson: tdmout: add sm1 support .../bindings/sound/amlogic,axg-fifo.txt | 9 +- .../bindings/sound/amlogic,axg-pdm.txt | 6 +- .../bindings/sound/amlogic,axg-spdifin.txt | 6 +- .../bindings/sound/amlogic,axg-spdifout.txt | 6 +- .../sound/amlogic,axg-tdm-formatters.txt | 4 +- .../bindings/sound/amlogic,g12a-tohdmitx.txt | 5 +- sound/soc/meson/axg-fifo.c | 2 +- sound/soc/meson/axg-fifo.h | 1 + sound/soc/meson/axg-frddr.c | 105 ++++++++++++++++-- sound/soc/meson/axg-tdmin.c | 47 +++++--- sound/soc/meson/axg-tdmout.c | 103 +++++++++++++---- sound/soc/meson/axg-toddr.c | 83 ++++++++++++-- 12 files changed, 321 insertions(+), 56 deletions(-) -- 2.21.0

2 16

[alsa-devel] [PATCH -next] ASoC: SOF: imx8: Fix COMPILE_TEST error
by YueHaibing 05 Sep '19

05 Sep '19

When do compile test, if SND_SOC_SOF_OF is not set, we get: sound/soc/sof/imx/imx8.o: In function `imx8_dsp_handle_request': imx8.c:(.text+0xb0): undefined reference to `snd_sof_ipc_msgs_rx' sound/soc/sof/imx/imx8.o: In function `imx8_ipc_msg_data': imx8.c:(.text+0xf4): undefined reference to `sof_mailbox_read' sound/soc/sof/imx/imx8.o: In function `imx8_dsp_handle_reply': imx8.c:(.text+0x160): undefined reference to `sof_mailbox_read' Make SND_SOC_SOF_IMX_TOPLEVEL always depends on SND_SOC_SOF_OF Reported-by: Hulk Robot <hulkci(a)huawei.com> Fixes: 202acc565a1f ("ASoC: SOF: imx: Add i.MX8 HW support") Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/soc/sof/imx/Kconfig | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/sof/imx/Kconfig b/sound/soc/sof/imx/Kconfig index fd73d84..5acae75 100644 --- a/sound/soc/sof/imx/Kconfig +++ b/sound/soc/sof/imx/Kconfig @@ -2,7 +2,8 @@ config SND_SOC_SOF_IMX_TOPLEVEL bool "SOF support for NXP i.MX audio DSPs" - depends on ARM64 && SND_SOC_SOF_OF || COMPILE_TEST + depends on ARM64|| COMPILE_TEST + depends on SND_SOC_SOF_OF help This adds support for Sound Open Firmware for NXP i.MX platforms. Say Y if you have such a device. -- 2.7.4

4 3

[alsa-devel] [PATCH]Add tas2770 driver code
by shifu0704＠thundersoft.com 05 Sep '19

05 Sep '19

From 5b539e7dbebebf7e58c38a6b54f222bc116d63f3 Mon Sep 17 00:00:00 2001 From: Shi Fu <shifu0704(a)thundersoft.com> Date: Mon, 1 Jul 2019 11:00:13 +0800 Signed-off-by: Shi Fu <shifu0704(a)thundersoft.com> Subject: [PATCH v5] dt: tas2770: Add tas2770 smart PA dt bindings. Add tas2770 smart PA dt bindings. --- Documentation/devicetree/bindings/tas2770.txt | 38 +++++++++++++++++++++++++++ 1 file changed, 38 insertions(+) create mode 100644 Documentation/devicetree/bindings/tas2770.txt diff --git a/Documentation/devicetree/bindings/tas2770.txt b/Documentation/devicetree/bindings/tas2770.txt new file mode 100644 index 0000000..f70b310 --- /dev/null +++ b/Documentation/devicetree/bindings/tas2770.txt @@ -0,0 +1,38 @@ +Texas Instruments TAS2770 Smart PA + +The TAS2770 is a mono, digital input Class-D audio amplifier optimized for +efficiently driving high peak power into small loudspeakers. +Integrated speaker voltage and current sense provides for +real time monitoring of loudspeaker behavior. + +Required properties: + + - compatible: - Should contain "ti,tas2770". + - reg: - The i2c address. Should contain <0x4c>, <0x4d>,<0x4e>, or <0x4f>. + - #address-cells - Should be <1>. + - #size-cells - Should be <0>. + - ti,asi-format: - Sets TDM RX capture edge. 0->Rising; 1->Falling. + - ti,left-slot: - Sets TDM RX left time slots. + - ti,right-slot: - Sets TDM RX right time slots. + - ti,imon-slot-no:- TDM TX current sense time slot. + - ti,vmon-slot-no:- TDM TX voltage sense time slot. + +Optional properties: + + - reset-gpio: Reset GPIO number of left device. + - irq-gpio: IRQ GPIO number of left device. + +Examples: + + tas2770@4c { + compatible = "ti,tas2770"; + reg = <0x4c>; + reset-gpio = <&gpio15 1 GPIO_ACTIVE_LOW>; + irq-gpio = <&gpio16 1 GPIO_ACTIVE_LOW>; + ti,asi-format = <0>; + ti,left-slot = <0>; + ti,right-slot = <1>; + ti,imon-slot-no = <0>; + ti,vmon-slot-no = <2>; + }; + -- 2.7.4 From 167a85aa30a41471ccddd9912b192d92800a4212 Mon Sep 17 00:00:00 2001 From: TS Shi Fu <shifu0704(a)thundersoft.com> Date: Mon, 24 Jun 2019 18:26:09 +0800 Subject: [PATCH v5] Add tas2770 driver code Add tas2770 driver code Date: Wed, 15 May 2019 12:08:42 +0800 From: Shi Fu <shifu0704(a)thundersoft.com> Signed-off-by: Shi Fu <shifu0704(a)thundersoft.com> Change-Id: Ibd5b04b4c0d0021a8057023aa6b43f97bba372b1 --- sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tas2770.c | 1103 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/tas2770.h | 173 +++++++ 4 files changed, 1283 insertions(+) create mode 100644 sound/soc/codecs/tas2770.c create mode 100644 sound/soc/codecs/tas2770.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 8f3e787..cc92da3 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -111,6 +111,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_STAC9766 if SND_SOC_AC97_BUS select SND_SOC_STI_SAS select SND_SOC_TAS2552 if I2C + select SND_SOC_TAS2770 if I2C select SND_SOC_TAS5086 if I2C select SND_SOC_TAS571X if I2C select SND_SOC_TFA9879 if I2C @@ -652,6 +653,10 @@ config SND_SOC_TAS2552 tristate "Texas Instruments TAS2552 Mono Audio amplifier" depends on I2C +config SND_SOC_TAS2770 + tristate "Texas Instruments TAS2770 speaker amplifier" + depends on I2C + config SND_SOC_TAS5086 tristate "Texas Instruments TAS5086 speaker amplifier" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 5305cc6..1b58d08 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -116,6 +116,7 @@ snd-soc-stac9766-objs := stac9766.o snd-soc-sti-sas-objs := sti-sas.o snd-soc-tas5086-objs := tas5086.o snd-soc-tas571x-objs := tas571x.o +snd-soc-tas2770-objs := tas2770.o snd-soc-tfa9879-objs := tfa9879.o snd-soc-tlv320aic23-objs := tlv320aic23.o snd-soc-tlv320aic23-i2c-objs := tlv320aic23-i2c.o @@ -332,6 +333,7 @@ obj-$(CONFIG_SND_SOC_STI_SAS) += snd-soc-sti-sas.o obj-$(CONFIG_SND_SOC_TAS2552) += snd-soc-tas2552.o obj-$(CONFIG_SND_SOC_TAS5086) += snd-soc-tas5086.o obj-$(CONFIG_SND_SOC_TAS571X) += snd-soc-tas571x.o +obj-$(CONFIG_SND_SOC_TAS2770) += snd-soc-tas2770.o obj-$(CONFIG_SND_SOC_TFA9879) += snd-soc-tfa9879.o obj-$(CONFIG_SND_SOC_TLV320AIC23) += snd-soc-tlv320aic23.o obj-$(CONFIG_SND_SOC_TLV320AIC23_I2C) += snd-soc-tlv320aic23-i2c.o diff --git a/sound/soc/codecs/tas2770.c b/sound/soc/codecs/tas2770.c new file mode 100644 index 0000000..9fc0c11 --- /dev/null +++ b/sound/soc/codecs/tas2770.c @@ -0,0 +1,1103 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * ALSA SoC Texas Instruments TAS2770 20-W Digital Input Mono Class-D + * Audio Amplifier with Speaker I/V Sense + * + * Copyright (C) 2016-2017 Texas Instruments Incorporated - http://www.ti.com/ + * Author: Tracy Yi <tracy-yi(a)ti.com> + * Shi Fu <shifu0704(a)thundersoft.com> + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/err.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/gpio.h> +#include <linux/pm_runtime.h> +#include <linux/regulator/consumer.h> +#include <linux/firmware.h> +#include <linux/regmap.h> +#include <linux/of.h> +#include <linux/of_gpio.h> +#include <linux/slab.h> +#include <sound/soc.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/initval.h> +#include <sound/tlv.h> + +#include "tas2770.h" + +#define TAS2770_MDELAY 0xFFFFFFFE +#define TAS2770_CHECK_PERIOD 5000 /* 5 second */ + +static void tas2770_hw_reset(struct tas2770_priv *p_tas2770) +{ + if (p_tas2770->mn_reset_gpio) { + gpiod_set_value_cansleep(p_tas2770->mn_reset_gpio, 0); + msleep(20); + gpiod_set_value_cansleep(p_tas2770->mn_reset_gpio, 1); + } + + p_tas2770->mn_current_book = -1; + p_tas2770->mn_current_page = -1; +} + +static void tas2770_enable_irq(struct tas2770_priv *p_tas2770, bool enable) +{ + if (enable) { + if (p_tas2770->mb_irq_enable) + return; + + if (gpio_is_valid(p_tas2770->mn_irq)) + enable_irq(p_tas2770->mn_irq); + p_tas2770->mb_irq_enable = true; + } else { + if (gpio_is_valid(p_tas2770->mn_irq)) + disable_irq_nosync(p_tas2770->mn_irq); + p_tas2770->mb_irq_enable = false; + } +} + +static int tas2770_runtime_suspend(struct tas2770_priv *p_tas2770) +{ + p_tas2770->mb_runtime_suspend = true; + + return 0; +} + +static int tas2770_runtime_resume(struct tas2770_priv *p_tas2770) +{ + + p_tas2770->mb_runtime_suspend = false; + + return 0; +} + +static int tas2770_regmap_write(struct tas2770_priv *p_tas2770, + unsigned int reg, unsigned int value) +{ + int nResult = 0; + int retry_count = TAS2770_I2C_RETRY_COUNT; + + while (retry_count--) { + nResult = snd_soc_component_write(p_tas2770->component, reg, + value); + if (!nResult) + break; + msleep(20); + } + if (retry_count == -1) + return ERROR_I2C_FAILED; + else + return 0; +} + +static int tas2770_regmap_read(struct tas2770_priv *p_tas2770, + unsigned int reg, unsigned int *value) +{ + int nResult = 0; + int retry_count = TAS2770_I2C_RETRY_COUNT; + + while (retry_count--) { + nResult = snd_soc_component_read(p_tas2770->component, reg, + value); + if (!nResult) + break; + + msleep(20); + } + if (!retry_count) + return ERROR_I2C_FAILED; + + return 0; +} + +static int tas2770_codec_suspend(struct snd_soc_component *component) +{ + struct tas2770_priv *p_tas2770 = + snd_soc_component_get_drvdata(component); + + mutex_lock(&p_tas2770->codec_lock); + tas2770_runtime_suspend(p_tas2770); + mutex_unlock(&p_tas2770->codec_lock); + + return 0; +} + +static int tas2770_codec_resume(struct snd_soc_component *component) +{ + struct tas2770_priv *p_tas2770 = + snd_soc_component_get_drvdata(component); + + mutex_lock(&p_tas2770->codec_lock); + tas2770_runtime_resume(p_tas2770); + mutex_unlock(&p_tas2770->codec_lock); + + return 0; +} + +static const char * const tas2770_ASI1_src[] = { + "I2C offset", "Left", "Right", "LeftRightDiv2", +}; + +static SOC_ENUM_SINGLE_DECL( + tas2770_ASI1_src_enum, TAS2770_TDM_CFG_REG2, + 4, tas2770_ASI1_src); + +static const struct snd_kcontrol_new tas2770_asi1_mux = + SOC_DAPM_ENUM("ASI1 Source", tas2770_ASI1_src_enum); + +static int tas2770_set_power_state(struct tas2770_priv *p_tas2770, int state) +{ + struct snd_soc_component *component = p_tas2770->component; + + switch (state) { + case TAS2770_POWER_ACTIVE: + snd_soc_component_update_bits(component, TAS2770_PWR_CTRL, + TAS2770_PWR_CTRL_MASK, + TAS2770_PWR_CTRL_ACTIVE); + p_tas2770->mn_power_state = TAS2770_POWER_ACTIVE; + tas2770_enable_irq(p_tas2770, true); + break; + + case TAS2770_POWER_MUTE: + snd_soc_component_update_bits(component, TAS2770_PWR_CTRL, + TAS2770_PWR_CTRL_MASK, + TAS2770_PWR_CTRL_MUTE); + p_tas2770->mn_power_state = TAS2770_POWER_MUTE; + break; + + case TAS2770_POWER_SHUTDOWN: + snd_soc_component_update_bits(component, TAS2770_PWR_CTRL, + TAS2770_PWR_CTRL_MASK, + TAS2770_PWR_CTRL_SHUTDOWN); + tas2770_enable_irq(p_tas2770, false); + p_tas2770->mn_power_state = TAS2770_POWER_SHUTDOWN; + break; + + default: + dev_err(p_tas2770->dev, + "wrong power state setting %d\n", state); + return -EINVAL; + } + + return 0; +} + +static int tas2770_dac_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + struct tas2770_priv *p_tas2770 = + snd_soc_component_get_drvdata(component); + int ret; + + mutex_lock(&p_tas2770->codec_lock); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + ret = tas2770_set_power_state(p_tas2770, + TAS2770_POWER_MUTE); + if (ret) + goto end; + break; + case SND_SOC_DAPM_PRE_PMD: + ret = tas2770_set_power_state(p_tas2770, + TAS2770_POWER_SHUTDOWN); + if (ret) + goto end; + break; + default: + dev_err(p_tas2770->dev, "Not supported evevt\n"); + ret = -EINVAL; + } + +end: + mutex_unlock(&p_tas2770->codec_lock); + return ret; +} + +static const struct snd_kcontrol_new isense_switch = + SOC_DAPM_SINGLE("Switch", TAS2770_PWR_CTRL, 3, 1, 1); +static const struct snd_kcontrol_new vsense_switch = + SOC_DAPM_SINGLE("Switch", TAS2770_PWR_CTRL, 2, 1, 1); + +static const struct snd_soc_dapm_widget tas2770_dapm_widgets[] = { + SND_SOC_DAPM_AIF_IN("ASI1", "ASI1 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_MUX("ASI1 Sel", SND_SOC_NOPM, 0, 0, + &tas2770_asi1_mux), + SND_SOC_DAPM_SWITCH("ISENSE", TAS2770_PWR_CTRL, 3, 1, + &isense_switch), + SND_SOC_DAPM_SWITCH("VSENSE", TAS2770_PWR_CTRL, 2, 1, + &vsense_switch), + SND_SOC_DAPM_DAC_E("DAC", NULL, SND_SOC_NOPM, 0, 0, tas2770_dac_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_OUTPUT("OUT"), + SND_SOC_DAPM_SIGGEN("VMON"), + SND_SOC_DAPM_SIGGEN("IMON") +}; + +static const struct snd_soc_dapm_route tas2770_audio_map[] = { + {"ASI1 Sel", "I2C offset", "ASI1"}, + {"ASI1 Sel", "Left", "ASI1"}, + {"ASI1 Sel", "Right", "ASI1"}, + {"ASI1 Sel", "LeftRightDiv2", "ASI1"}, + {"DAC", NULL, "ASI1 Sel"}, + {"OUT", NULL, "DAC"}, + {"ISENSE", "Switch", "IMON"}, + {"VSENSE", "Switch", "VMON"}, +}; + + +static int tas2770_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_component *component = dai->component; + struct tas2770_priv *p_tas2770 = + snd_soc_component_get_drvdata(component); + int ret; + + + mutex_lock(&p_tas2770->codec_lock); + if (mute) + ret = tas2770_set_power_state(p_tas2770, TAS2770_POWER_MUTE); + else + ret = tas2770_set_power_state(p_tas2770, TAS2770_POWER_ACTIVE); + + mutex_unlock(&p_tas2770->codec_lock); + return ret; +} + +static int tas2770_set_slot(struct snd_soc_component *component, int slot_width) +{ + int ret; + struct tas2770_priv *p_tas2770 = + snd_soc_component_get_drvdata(component); + + switch (slot_width) { + case 16: + ret = snd_soc_component_update_bits(component, + TAS2770_TDM_CFG_REG2, + TAS2770_TDM_CFG_REG2_RXS_MASK, + TAS2770_TDM_CFG_REG2_RXS_16BITS); + break; + + case 24: + ret = snd_soc_component_update_bits(component, + TAS2770_TDM_CFG_REG2, + TAS2770_TDM_CFG_REG2_RXS_MASK, + TAS2770_TDM_CFG_REG2_RXS_24BITS); + break; + + case 32: + ret = snd_soc_component_update_bits(component, + TAS2770_TDM_CFG_REG2, + TAS2770_TDM_CFG_REG2_RXS_MASK, + TAS2770_TDM_CFG_REG2_RXS_32BITS); + break; + + case 0: + /* Do not change slot width */ + ret = 0; + break; + + default: + ret = -EINVAL; + } + + if (!ret) + p_tas2770->mn_slot_width = slot_width; + + return ret; +} + +static int tas2770_set_bitwidth(struct tas2770_priv *p_tas2770, int bitwidth) +{ + int ret; + struct snd_soc_component *component = p_tas2770->component; + + switch (bitwidth) { + case SNDRV_PCM_FORMAT_S16_LE: + ret = snd_soc_component_update_bits(component, + TAS2770_TDM_CFG_REG2, + TAS2770_TDM_CFG_REG2_RXW_MASK, + TAS2770_TDM_CFG_REG2_RXW_16BITS); + /* If machine driver did not call set slot width */ + if (p_tas2770->mn_slot_width == 0) + tas2770_set_slot(component, TAS2770_SLOT_16BIT); + p_tas2770->mn_v_mon_slot_no = p_tas2770->mn_i_mon_slot_no + 2; + break; + case SNDRV_PCM_FORMAT_S24_LE: + ret = snd_soc_component_update_bits(component, + TAS2770_TDM_CFG_REG2, + TAS2770_TDM_CFG_REG2_RXW_MASK, + TAS2770_TDM_CFG_REG2_RXW_24BITS); + if (p_tas2770->mn_slot_width == 0) + tas2770_set_slot(component, TAS2770_SLOT_32BIT); + p_tas2770->mn_v_mon_slot_no = p_tas2770->mn_i_mon_slot_no + 4; + break; + case SNDRV_PCM_FORMAT_S32_LE: + ret = snd_soc_component_update_bits(component, + TAS2770_TDM_CFG_REG2, + TAS2770_TDM_CFG_REG2_RXW_MASK, + TAS2770_TDM_CFG_REG2_RXW_32BITS); + if (p_tas2770->mn_slot_width == 0) + tas2770_set_slot(component, TAS2770_SLOT_32BIT); + p_tas2770->mn_v_mon_slot_no = p_tas2770->mn_i_mon_slot_no + 4; + break; + + default: + return -EINVAL; + } + + p_tas2770->mn_ch_size = bitwidth; + + ret = snd_soc_component_update_bits(component, + TAS2770_TDM_CFG_REG5, + TAS2770_TDM_CFG_REG5_VSNS_MASK | + TAS2770_TDM_CFG_REG5_50_MASK, + TAS2770_TDM_CFG_REG5_VSNS_ENABLE | + p_tas2770->mn_v_mon_slot_no); + if (ret) + goto end; + ret = snd_soc_component_update_bits(component, + TAS2770_TDM_CFG_REG6, + TAS2770_TDM_CFG_REG6_ISNS_MASK | + TAS2770_TDM_CFG_REG6_50_MASK, + TAS2770_TDM_CFG_REG6_ISNS_ENABLE | + p_tas2770->mn_i_mon_slot_no); + +end: + return ret; +} + +static int tas2770_set_samplerate(struct tas2770_priv *p_tas2770, + int samplerate) +{ + int ret; + struct snd_soc_component *component = p_tas2770->component; + + switch (samplerate) { + case 48000: + ret = snd_soc_component_update_bits(component, + TAS2770_TDM_CFG_REG0, + TAS2770_TDM_CFG_REG0_SMP_MASK, + TAS2770_TDM_CFG_REG0_SMP_48KHZ); + if (ret) + goto end; + ret = snd_soc_component_update_bits(component, + TAS2770_TDM_CFG_REG0, + TAS2770_TDM_CFG_REG0_31_MASK, + TAS2770_TDM_CFG_REG0_31_44_1_48KHZ); + if (ret) + goto end; + break; + case 44100: + ret = snd_soc_component_update_bits(component, + TAS2770_TDM_CFG_REG0, + TAS2770_TDM_CFG_REG0_SMP_MASK, + TAS2770_TDM_CFG_REG0_SMP_44_1KHZ); + if (ret) + goto end; + ret = snd_soc_component_update_bits(component, + TAS2770_TDM_CFG_REG0, + TAS2770_TDM_CFG_REG0_31_MASK, + TAS2770_TDM_CFG_REG0_31_44_1_48KHZ); + if (ret) + goto end; + break; + case 96000: + ret = snd_soc_component_update_bits(component, + TAS2770_TDM_CFG_REG0, + TAS2770_TDM_CFG_REG0_SMP_MASK, + TAS2770_TDM_CFG_REG0_SMP_48KHZ); + if (ret) + goto end; + ret = snd_soc_component_update_bits(component, + TAS2770_TDM_CFG_REG0, + TAS2770_TDM_CFG_REG0_31_MASK, + TAS2770_TDM_CFG_REG0_31_88_2_96KHZ); + break; + case 88200: + ret = snd_soc_component_update_bits(component, + TAS2770_TDM_CFG_REG0, + TAS2770_TDM_CFG_REG0_SMP_MASK, + TAS2770_TDM_CFG_REG0_SMP_44_1KHZ); + if (ret) + goto end; + ret = snd_soc_component_update_bits(component, + TAS2770_TDM_CFG_REG0, + TAS2770_TDM_CFG_REG0_31_MASK, + TAS2770_TDM_CFG_REG0_31_88_2_96KHZ); + break; + case 19200: + ret = snd_soc_component_update_bits(component, + TAS2770_TDM_CFG_REG0, + TAS2770_TDM_CFG_REG0_SMP_MASK, + TAS2770_TDM_CFG_REG0_SMP_48KHZ); + if (ret) + goto end; + ret = snd_soc_component_update_bits(component, + TAS2770_TDM_CFG_REG0, + TAS2770_TDM_CFG_REG0_31_MASK, + TAS2770_TDM_CFG_REG0_31_176_4_192KHZ); + if (ret) + goto end; + break; + case 17640: + ret = snd_soc_component_update_bits(component, + TAS2770_TDM_CFG_REG0, + TAS2770_TDM_CFG_REG0_SMP_MASK, + TAS2770_TDM_CFG_REG0_SMP_44_1KHZ); + if (ret) + goto end; + ret = snd_soc_component_update_bits(component, + TAS2770_TDM_CFG_REG0, + TAS2770_TDM_CFG_REG0_31_MASK, + TAS2770_TDM_CFG_REG0_31_176_4_192KHZ); + break; + default: + ret = -EINVAL; + } + +end: + if (!ret) + p_tas2770->mn_sampling_rate = samplerate; + return ret; +} + +static int tas2770_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct tas2770_priv *p_tas2770 = + snd_soc_component_get_drvdata(component); + int ret; + + mutex_lock(&p_tas2770->codec_lock); + + ret = tas2770_set_bitwidth(p_tas2770, params_format(params)); + if (ret < 0) + goto end; + + + ret = tas2770_set_samplerate(p_tas2770, params_rate(params)); + +end: + mutex_unlock(&p_tas2770->codec_lock); + return ret; +} + +static int tas2770_set_fmt(struct tas2770_priv *p_tas2770, unsigned int fmt) +{ + u8 tdm_rx_start_slot = 0, asi_cfg_1 = 0; + int ret; + int value = 0; + struct snd_soc_component *component = p_tas2770->component; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + dev_err(p_tas2770->dev, "ASI format master is not found\n"); + ret = -EINVAL; + return ret; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + asi_cfg_1 |= TAS2770_TDM_CFG_REG1_RX_RSING; + break; + case SND_SOC_DAIFMT_IB_NF: + asi_cfg_1 |= TAS2770_TDM_CFG_REG1_RX_FALING; + break; + default: + dev_err(p_tas2770->dev, "ASI format Inverse is not found\n"); + ret = -EINVAL; + return ret; + } + + ret = snd_soc_component_update_bits(component, TAS2770_TDM_CFG_REG1, + TAS2770_TDM_CFG_REG1_RX_MASK, + asi_cfg_1); + if (ret) + return ret; + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case (SND_SOC_DAIFMT_I2S): + tdm_rx_start_slot = 1; + break; + case (SND_SOC_DAIFMT_DSP_A): + case (SND_SOC_DAIFMT_DSP_B): + tdm_rx_start_slot = 1; + break; + case (SND_SOC_DAIFMT_LEFT_J): + tdm_rx_start_slot = 0; + break; + default: + dev_err(p_tas2770->dev, + "DAI Format is not found, fmt=0x%x\n", fmt); + ret = -EINVAL; + return ret; + } + + ret = snd_soc_component_update_bits(component, TAS2770_TDM_CFG_REG1, + TAS2770_TDM_CFG_REG1_MASK, + (tdm_rx_start_slot << TAS2770_TDM_CFG_REG1_51_SHIFT)); + if (ret) + return ret; + ret = snd_soc_component_update_bits(component, TAS2770_TDM_CFG_REG3, + TAS2770_TDM_CFG_REG3_30_MASK, + (p_tas2770->mn_left_slot << + TAS2770_TDM_CFG_REG3_30_SHIFT)); + if (ret) + return ret; + ret = snd_soc_component_update_bits(component, TAS2770_TDM_CFG_REG3, + TAS2770_TDM_CFG_REG3_RXS_MASK, + (p_tas2770->mn_right_slot << + TAS2770_TDM_CFG_REG3_RXS_SHIFT)); + if (ret) + return ret; + + value = snd_soc_component_read32(component, TAS2770_TDM_CFG_REG3); + + p_tas2770->mn_asi_format = fmt; + + return 0; +} + +static int tas2770_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_component *component = dai->component; + struct tas2770_priv *p_tas2770 = + snd_soc_component_get_drvdata(component); + int ret; + + mutex_lock(&p_tas2770->codec_lock); + + ret = tas2770_set_fmt(p_tas2770, fmt); + + mutex_unlock(&p_tas2770->codec_lock); + return ret; +} + +static int tas2770_set_dai_tdm_slot(struct snd_soc_dai *dai, + unsigned int tx_mask, + unsigned int rx_mask, + int slots, int slot_width) +{ + int ret; + struct snd_soc_component *component = dai->component; + struct tas2770_priv *p_tas2770 = + snd_soc_component_get_drvdata(component); + + mutex_lock(&p_tas2770->codec_lock); + ret = tas2770_set_slot(component, slot_width); + mutex_unlock(&p_tas2770->codec_lock); + + return ret; +} + +static struct snd_soc_dai_ops tas2770_dai_ops = { + .digital_mute = tas2770_mute, + .hw_params = tas2770_hw_params, + .set_fmt = tas2770_set_dai_fmt, + .set_tdm_slot = tas2770_set_dai_tdm_slot, +}; + +#define TAS2770_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +#define TAS2770_RATES (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |\ + SNDRV_PCM_RATE_96000 |\ + SNDRV_PCM_RATE_192000\ + ) + +static struct snd_soc_dai_driver tas2770_dai_driver[] = { + { + .name = "tas2770 ASI1", + .id = 0, + .playback = { + .stream_name = "ASI1 Playback", + .channels_min = 2, + .channels_max = 2, + .rates = TAS2770_RATES, + .formats = TAS2770_FORMATS, + }, + .capture = { + .stream_name = "ASI1 Capture", + .channels_min = 0, + .channels_max = 2, + .rates = TAS2770_RATES, + .formats = TAS2770_FORMATS, + }, + .ops = &tas2770_dai_ops, + .symmetric_rates = 1, + }, +}; + +static int tas2770_codec_probe(struct snd_soc_component *component) +{ + struct tas2770_priv *p_tas2770 = + snd_soc_component_get_drvdata(component); + + p_tas2770->component = component; + + return 0; +} + +static void tas2770_codec_remove(struct snd_soc_component *component) +{ + pm_runtime_put(component->dev); +} + +static DECLARE_TLV_DB_SCALE(tas2770_digital_tlv, 1100, 50, 0); +static DECLARE_TLV_DB_SCALE(tas2770_playback_volume, -12750, 50, 0); + +static const struct snd_kcontrol_new tas2770_snd_controls[] = { + SOC_SINGLE_TLV("Amp Output Level", TAS2770_PLAY_CFG_REG0, + 0, 0x14, 0, + tas2770_digital_tlv), + SOC_SINGLE_TLV("Playback Volume", TAS2770_PLAY_CFG_REG2, + 0, TAS2770_PLAY_CFG_REG2_VMAX, 1, + tas2770_playback_volume), +}; + +static const struct snd_soc_component_driver soc_component_driver_tas2770 = { + .probe = tas2770_codec_probe, + .remove = tas2770_codec_remove, + .suspend = tas2770_codec_suspend, + .resume = tas2770_codec_resume, + .controls = tas2770_snd_controls, + .num_controls = ARRAY_SIZE(tas2770_snd_controls), + .dapm_widgets = tas2770_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(tas2770_dapm_widgets), + .dapm_routes = tas2770_audio_map, + .num_dapm_routes = ARRAY_SIZE(tas2770_audio_map), + .idle_bias_on = 1, + .endianness = 1, + .non_legacy_dai_naming = 1, +}; + +static int tas2770_register_codec(struct tas2770_priv *p_tas2770) +{ + return devm_snd_soc_register_component(p_tas2770->dev, + &soc_component_driver_tas2770, + tas2770_dai_driver, ARRAY_SIZE(tas2770_dai_driver)); +} + +static void tas2770_load_config(struct tas2770_priv *p_tas2770) +{ + int ret; + + tas2770_hw_reset(p_tas2770); + tas2770_regmap_write(p_tas2770, TAS2770_SW_RST, + TAS2770_RST); + + ret = tas2770_set_slot(p_tas2770->component, p_tas2770->mn_slot_width); + if (ret) + goto error_out; + + ret = tas2770_set_fmt(p_tas2770, p_tas2770->mn_asi_format); + if (ret) + goto error_out; + + ret = tas2770_set_bitwidth(p_tas2770, p_tas2770->mn_ch_size); + if (ret) + goto error_out; + + ret = tas2770_set_samplerate(p_tas2770, p_tas2770->mn_sampling_rate); + if (ret) + goto error_out; + + ret = tas2770_set_power_state(p_tas2770, p_tas2770->mn_power_state); + if (ret) + goto error_out; + +error_out: + return; +} + +static const struct reg_default tas2770_reg_defaults[] = { + { TAS2770_PAGE, 0x00 }, + { TAS2770_SW_RST, 0x00 }, + { TAS2770_PWR_CTRL, 0x0e }, + { TAS2770_PLAY_CFG_REG0, 0x10 }, + { TAS2770_PLAY_CFG_REG1, 0x01 }, + { TAS2770_PLAY_CFG_REG2, 0x00 }, + { TAS2770_MSC_CFG_REG0, 0x07 }, + { TAS2770_TDM_CFG_REG1, 0x02 }, + { TAS2770_TDM_CFG_REG2, 0x0a }, + { TAS2770_TDM_CFG_REG3, 0x10 }, + { TAS2770_INT_MASK_REG0, 0xfc }, + { TAS2770_INT_MASK_REG1, 0xb1 }, + { TAS2770_INT_CFG, 0x05 }, + { TAS2770_MISC_IRQ, 0x81 }, + { TAS2770_CLK_CGF, 0x0c }, + +}; + +static bool tas2770_volatile(struct device *dev, unsigned int reg) +{ + switch (reg) { + case TAS2770_PAGE: /* regmap implementation requires this */ + case TAS2770_SW_RST: /* always clears after write */ + case TAS2770_BO_PRV_REG0:/* has a self clearing bit */ + case TAS2770_LVE_INT_REG0: + case TAS2770_LVE_INT_REG1: + case TAS2770_LAT_INT_REG0:/* Sticky interrupt flags */ + case TAS2770_LAT_INT_REG1:/* Sticky interrupt flags */ + case TAS2770_VBAT_MSB: + case TAS2770_VBAT_LSB: + case TAS2770_TEMP_MSB: + case TAS2770_TEMP_LSB: + return true; + } + return false; +} + +static bool tas2770_writeable(struct device *dev, unsigned int reg) +{ + switch (reg) { + case TAS2770_LVE_INT_REG0: + case TAS2770_LVE_INT_REG1: + case TAS2770_LAT_INT_REG0: + case TAS2770_LAT_INT_REG1: + case TAS2770_VBAT_MSB: + case TAS2770_VBAT_LSB: + case TAS2770_TEMP_MSB: + case TAS2770_TEMP_LSB: + case TAS2770_TDM_CLK_DETC: + case TAS2770_REV_AND_GPID: + return false; + } + return true; +} + +static const struct regmap_range_cfg tas2770_regmap_ranges[] = { + { + .range_min = 0, + .range_max = 1 * 128, + .selector_reg = TAS2770_PAGE, + .selector_mask = 0xff, + .selector_shift = 0, + .window_start = 0, + .window_len = 128, + }, +}; + +static const struct regmap_config tas2770_i2c_regmap = { + .reg_bits = 8, + .val_bits = 8, + .writeable_reg = tas2770_writeable, + .volatile_reg = tas2770_volatile, + .reg_defaults = tas2770_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(tas2770_reg_defaults), + .cache_type = REGCACHE_RBTREE, + .ranges = tas2770_regmap_ranges, + .num_ranges = ARRAY_SIZE(tas2770_regmap_ranges), + .max_register = 1 * 128, +}; + +static irqreturn_t tas2770_irq_handler(int irq, void *dev_id) +{ + struct tas2770_priv *p_tas2770 = (struct tas2770_priv *)dev_id; + + unsigned int nDevInt1Status = 0, nDevInt2Status = 0; + int n_counter = 2; + int n_result; + + mutex_lock(&p_tas2770->codec_lock); + + tas2770_enable_irq(p_tas2770, false); + + if (p_tas2770->mb_runtime_suspend) + goto end; + + if (p_tas2770->mn_power_state == TAS2770_POWER_SHUTDOWN) + goto end; + + n_result = tas2770_regmap_write(p_tas2770, TAS2770_INT_MASK_REG0, + TAS2770_INT_MASK_REG0_DISABLE); + if (n_result) + goto reload; + n_result = tas2770_regmap_write(p_tas2770, TAS2770_INT_MASK_REG1, + TAS2770_INT_MASK_REG1_DISABLE); + if (n_result) + goto reload; + + n_result = tas2770_regmap_read(p_tas2770, + TAS2770_LAT_INT_REG0, &nDevInt1Status); + if (n_result >= 0) + n_result = tas2770_regmap_read(p_tas2770, + TAS2770_LAT_INT_REG1, &nDevInt2Status); + else + goto reload; + + if (((nDevInt1Status & 0x3) != 0) || ((nDevInt2Status & 0x0f) != 0)) { + /* in case of INT_OC, INT_OT, INT_OVLT, INT_UVLT, INT_BO */ + + if (nDevInt1Status & + TAS2770_LAT_INT_REG0_OCE_FLG) { + p_tas2770->mn_err_code |= ERROR_OVER_CURRENT; + dev_err(p_tas2770->dev, "SPK over current!\n"); + } else { + p_tas2770->mn_err_code &= ~ERROR_OVER_CURRENT; + } + + if (nDevInt1Status & + TAS2770_LAT_INT_REG0_OTE_FLG) { + p_tas2770->mn_err_code |= ERROR_DIE_OVERTEMP; + dev_err(p_tas2770->dev, "die over temperature!\n"); + } else { + p_tas2770->mn_err_code &= ~ERROR_DIE_OVERTEMP; + } + + if (nDevInt2Status & + TAS2770_LAT_INT_REG1_VBA_TOV) { + p_tas2770->mn_err_code |= ERROR_OVER_VOLTAGE; + dev_err(p_tas2770->dev, "SPK over voltage!\n"); + } else { + p_tas2770->mn_err_code &= ~ERROR_OVER_VOLTAGE; + } + + if (nDevInt2Status & + TAS2770_LAT_INT_REG1_VBA_TUV) { + p_tas2770->mn_err_code |= ERROR_UNDER_VOLTAGE; + dev_err(p_tas2770->dev, "SPK under voltage!\n"); + } else { + p_tas2770->mn_err_code &= ~ERROR_UNDER_VOLTAGE; + } + + if (nDevInt2Status & + TAS2770_LAT_INT_REG1_BOUT_FLG) { + p_tas2770->mn_err_code |= ERROR_BROWNOUT; + dev_err(p_tas2770->dev, "brownout!\n"); + } else { + p_tas2770->mn_err_code &= ~ERROR_BROWNOUT; + } + + goto reload; + } else { + n_counter = 2; + + while (n_counter > 0) { + n_result = tas2770_regmap_read(p_tas2770, + TAS2770_PWR_CTRL, &nDevInt1Status); + if (n_result) + goto reload; + + if ((nDevInt1Status & + TAS2770_PWR_CTRL_MASK) + != TAS2770_PWR_CTRL_SHUTDOWN) + break; + + n_counter--; + if (n_counter > 0) + msleep(20); + } + + if ((nDevInt1Status & + TAS2770_PWR_CTRL_MASK) + == TAS2770_PWR_CTRL_SHUTDOWN) { + dev_err(p_tas2770->dev, "%s, Critical ERROR REG[0x%x] = 0x%x\n", + __func__, + TAS2770_PWR_CTRL, + nDevInt1Status); + p_tas2770->mn_err_code |= ERROR_CLASSD_PWR; + goto reload; + } + p_tas2770->mn_err_code &= ~ERROR_CLASSD_PWR; + } + + n_result = tas2770_regmap_write(p_tas2770, + TAS2770_INT_MASK_REG0, 0xfc); + if (n_result) + goto reload; + + n_result = tas2770_regmap_write(p_tas2770, + TAS2770_INT_MASK_REG1, 0xb1); + if (n_result) + goto reload; + + goto end; + +reload: + /* hardware reset and reload */ + tas2770_load_config(p_tas2770); + +end: + mutex_unlock(&p_tas2770->codec_lock); + tas2770_enable_irq(p_tas2770, true); + + return IRQ_HANDLED; +} + +static int tas2770_parse_dt(struct device *dev, struct tas2770_priv *p_tas2770) +{ + int rc = 0; + + rc = fwnode_property_read_u32(dev->fwnode, "ti,asi-format", + &p_tas2770->mn_asi_format); + if (rc) { + dev_err(p_tas2770->dev, "Looking up %s property failed %d\n", + "ti,asi-format", rc); + goto end; + } + + rc = fwnode_property_read_u32(dev->fwnode, "ti,left-slot", + &p_tas2770->mn_left_slot); + if (rc) { + dev_err(p_tas2770->dev, "Looking up %s property failed %d\n", + "ti,left-slot", rc); + goto end; + } + + rc = fwnode_property_read_u32(dev->fwnode, "ti,right-slot", + &p_tas2770->mn_right_slot); + if (rc) { + dev_err(p_tas2770->dev, "Looking up %s property failed %d\n", + "ti,right-slot", rc); + goto end; + } + + rc = fwnode_property_read_u32(dev->fwnode, "ti,imon-slot-no", + &p_tas2770->mn_i_mon_slot_no); + if (rc) { + dev_err(p_tas2770->dev, "Looking up %s property failed %d\n", + "ti,imon-slot-no", rc); + goto end; + } + + rc = fwnode_property_read_u32(dev->fwnode, "ti,vmon-slot-no", + &p_tas2770->mn_v_mon_slot_no); + if (rc) { + dev_err(p_tas2770->dev, "Looking up %s property failed %d\n", + "ti,vmon-slot-no", rc); + goto end; + } + +end: + return rc; +} + +static int tas2770_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + struct tas2770_priv *p_tas2770; + int n_result; + + p_tas2770 = devm_kzalloc(&client->dev, + sizeof(struct tas2770_priv), GFP_KERNEL); + if (p_tas2770 == NULL) + return -ENOMEM; + p_tas2770->dev = &client->dev; + p_tas2770->mn_irq = client->irq; + + i2c_set_clientdata(client, p_tas2770); + dev_set_drvdata(&client->dev, p_tas2770); + p_tas2770->mn_power_state = TAS2770_POWER_SHUTDOWN; + + p_tas2770->regmap = devm_regmap_init_i2c(client, &tas2770_i2c_regmap); + if (IS_ERR(p_tas2770->regmap)) { + n_result = PTR_ERR(p_tas2770->regmap); + dev_err(&client->dev, "Failed to allocate register map: %d\n", + n_result); + goto end; + } + + if (client->dev.of_node) { + n_result = tas2770_parse_dt(&client->dev, p_tas2770); + if (n_result) { + dev_err(p_tas2770->dev, "%s: Failed to parse devicetree\n", + __func__); + goto end; + } + } + + p_tas2770->mn_reset_gpio = devm_gpiod_get_optional(p_tas2770->dev, + "reset-gpio", + GPIOD_OUT_HIGH); + if (IS_ERR(p_tas2770->mn_reset_gpio)) { + if (PTR_ERR(p_tas2770->mn_reset_gpio) == -EPROBE_DEFER) + return -EPROBE_DEFER; + p_tas2770->mn_reset_gpio = NULL; + } + + if (p_tas2770->mn_irq > 0) { + n_result = devm_request_threaded_irq(p_tas2770->dev, + p_tas2770->mn_irq, + NULL, tas2770_irq_handler, + IRQF_TRIGGER_FALLING | IRQF_ONESHOT, client->name, + p_tas2770); + if (n_result) { + dev_err(p_tas2770->dev, "Unable to request IRQ\n"); + return n_result; + } + } + + p_tas2770->mn_ch_size = 0; + p_tas2770->mn_slot_width = 0; + + tas2770_hw_reset(p_tas2770); + tas2770_regmap_write(p_tas2770, TAS2770_SW_RST, + TAS2770_RST); + + mutex_init(&p_tas2770->dev_lock); + + mutex_init(&p_tas2770->codec_lock); + n_result = tas2770_register_codec(p_tas2770); + if (n_result) + dev_err(p_tas2770->dev, "Register codec failed.\n"); + +end: + return n_result; +} + +static int tas2770_i2c_remove(struct i2c_client *client) +{ + struct tas2770_priv *p_tas2770 = i2c_get_clientdata(client); + + mutex_destroy(&p_tas2770->dev_lock); + + return 0; +} + + +static const struct i2c_device_id tas2770_i2c_id[] = { + { "tas2770", 0}, + { } +}; +MODULE_DEVICE_TABLE(i2c, tas2770_i2c_id); + +#if defined(CONFIG_OF) +static const struct of_device_id tas2770_of_match[] = { + { .compatible = "ti,tas2770" }, + {}, +}; +MODULE_DEVICE_TABLE(of, tas2770_of_match); +#endif + +static struct i2c_driver tas2770_i2c_driver = { + .driver = { + .name = "tas2770", + .owner = THIS_MODULE, +#if defined(CONFIG_OF) + .of_match_table = of_match_ptr(tas2770_of_match), +#endif + }, + .probe = tas2770_i2c_probe, + .remove = tas2770_i2c_remove, + .id_table = tas2770_i2c_id, +}; + +module_i2c_driver(tas2770_i2c_driver); + +MODULE_AUTHOR("Shi Fu <shifu0704(a)thundersoft.com>"); +MODULE_DESCRIPTION("TAS2770 I2C Smart Amplifier driver"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/tas2770.h b/sound/soc/codecs/tas2770.h new file mode 100644 index 0000000..f08e208 --- /dev/null +++ b/sound/soc/codecs/tas2770.h @@ -0,0 +1,173 @@ +/* SPDX-License-Identifier: GPL-2.0 + * + * ALSA SoC TAS2770 codec driver + * + * Copyright (C) 2016-2017 Texas Instruments Incorporated - http://www.ti.com/ + */ +#ifndef __TAS2770__ +#define __TAS2770__ + +/* Book Control Register (available in page0 of each book) */ +#define TAS2770_BOOKCTL_PAGE 0 +#define TAS2770_BOOKCTL_REG 127 +#define TAS2770_REG(page, reg) ((page * 128) + reg) + /* Page */ +#define TAS2770_PAGE TAS2770_REG(0X0, 0x00) +#define TAS2770_PAGE_PAGE_MASK 255 + /* Software Reset */ +#define TAS2770_SW_RST TAS2770_REG(0X0, 0x01) +#define TAS2770_RST BIT(0) + /* Power Control */ +#define TAS2770_PWR_CTRL TAS2770_REG(0X0, 0x02) +#define TAS2770_PWR_CTRL_MASK 0x3 +#define TAS2770_PWR_CTRL_ACTIVE 0x0 +#define TAS2770_PWR_CTRL_MUTE BIT(0) +#define TAS2770_PWR_CTRL_SHUTDOWN 0x2 + /* Playback Configuration Reg0 */ +#define TAS2770_PLAY_CFG_REG0 TAS2770_REG(0X0, 0x03) + /* Playback Configuration Reg1 */ +#define TAS2770_PLAY_CFG_REG1 TAS2770_REG(0X0, 0x04) + /* Playback Configuration Reg2 */ +#define TAS2770_PLAY_CFG_REG2 TAS2770_REG(0X0, 0x05) +#define TAS2770_PLAY_CFG_REG2_VMAX 0xc9 + /* Misc Configuration Reg0 */ +#define TAS2770_MSC_CFG_REG0 TAS2770_REG(0X0, 0x07) + /* TDM Configuration Reg0 */ +#define TAS2770_TDM_CFG_REG0 TAS2770_REG(0X0, 0x0A) +#define TAS2770_TDM_CFG_REG0_SMP_MASK BIT(5) +#define TAS2770_TDM_CFG_REG0_SMP_48KHZ 0x0 +#define TAS2770_TDM_CFG_REG0_SMP_44_1KHZ BIT(5) +#define TAS2770_TDM_CFG_REG0_31_MASK 0xe +#define TAS2770_TDM_CFG_REG0_31_44_1_48KHZ 0x6 +#define TAS2770_TDM_CFG_REG0_31_88_2_96KHZ 0x8 +#define TAS2770_TDM_CFG_REG0_31_176_4_192KHZ 0xa + /* TDM Configuration Reg1 */ +#define TAS2770_TDM_CFG_REG1 TAS2770_REG(0X0, 0x0B) +#define TAS2770_TDM_CFG_REG1_MASK 0x3e +#define TAS2770_TDM_CFG_REG1_51_SHIFT 1 +#define TAS2770_TDM_CFG_REG1_RX_MASK BIT(0) +#define TAS2770_TDM_CFG_REG1_RX_RSING 0x0 +#define TAS2770_TDM_CFG_REG1_RX_FALING BIT(0) + /* TDM Configuration Reg2 */ +#define TAS2770_TDM_CFG_REG2 TAS2770_REG(0X0, 0x0C) +#define TAS2770_TDM_CFG_REG2_RXW_MASK 0xc +#define TAS2770_TDM_CFG_REG2_RXW_16BITS 0x0 +#define TAS2770_TDM_CFG_REG2_RXW_24BITS 0x8 +#define TAS2770_TDM_CFG_REG2_RXW_32BITS 0xc +#define TAS2770_TDM_CFG_REG2_RXS_MASK 0x3 +#define TAS2770_TDM_CFG_REG2_RXS_16BITS 0x0 +#define TAS2770_TDM_CFG_REG2_RXS_24BITS BIT(0) +#define TAS2770_TDM_CFG_REG2_RXS_32BITS 0x2 + /* TDM Configuration Reg3 */ +#define TAS2770_TDM_CFG_REG3 TAS2770_REG(0X0, 0x0D) +#define TAS2770_TDM_CFG_REG3_RXS_MASK 0xf0 +#define TAS2770_TDM_CFG_REG3_RXS_SHIFT 0x4 +#define TAS2770_TDM_CFG_REG3_30_MASK 0xf +#define TAS2770_TDM_CFG_REG3_30_SHIFT 0 + /* TDM Configuration Reg5 */ +#define TAS2770_TDM_CFG_REG5 TAS2770_REG(0X0, 0x0F) +#define TAS2770_TDM_CFG_REG5_VSNS_MASK BIT(6) +#define TAS2770_TDM_CFG_REG5_VSNS_ENABLE BIT(6) +#define TAS2770_TDM_CFG_REG5_50_MASK 0x3f + /* TDM Configuration Reg6 */ +#define TAS2770_TDM_CFG_REG6 TAS2770_REG(0X0, 0x10) +#define TAS2770_TDM_CFG_REG6_ISNS_MASK BIT(6) +#define TAS2770_TDM_CFG_REG6_ISNS_ENABLE BIT(6) +#define TAS2770_TDM_CFG_REG6_50_MASK 0x3f + /* Brown Out Prevention Reg0 */ +#define TAS2770_BO_PRV_REG0 TAS2770_REG(0X0, 0x1B) + /* Interrupt MASK Reg0 */ +#define TAS2770_INT_MASK_REG0 TAS2770_REG(0X0, 0x20) +#define TAS2770_INT_MASK_REG0_DISABLE 0xff + /* Interrupt MASK Reg1 */ +#define TAS2770_INT_MASK_REG1 TAS2770_REG(0X0, 0x21) +#define TAS2770_INT_MASK_REG1_DISABLE 0xff + /* Live-Interrupt Reg0 */ +#define TAS2770_LVE_INT_REG0 TAS2770_REG(0X0, 0x22) + /* Live-Interrupt Reg1 */ +#define TAS2770_LVE_INT_REG1 TAS2770_REG(0X0, 0x23) + /* Latched-Interrupt Reg0 */ +#define TAS2770_LAT_INT_REG0 TAS2770_REG(0X0, 0x24) +#define TAS2770_LAT_INT_REG0_OCE_FLG BIT(1) +#define TAS2770_LAT_INT_REG0_OTE_FLG BIT(0) + /* Latched-Interrupt Reg1 */ +#define TAS2770_LAT_INT_REG1 TAS2770_REG(0X0, 0x25) +#define TAS2770_LAT_INT_REG1_VBA_TOV BIT(3) +#define TAS2770_LAT_INT_REG1_VBA_TUV BIT(2) +#define TAS2770_LAT_INT_REG1_BOUT_FLG BIT(1) + /* VBAT MSB */ +#define TAS2770_VBAT_MSB TAS2770_REG(0X0, 0x27) + /* VBAT LSB */ +#define TAS2770_VBAT_LSB TAS2770_REG(0X0, 0x28) + /* TEMP MSB */ +#define TAS2770_TEMP_MSB TAS2770_REG(0X0, 0x29) + /* TEMP LSB */ +#define TAS2770_TEMP_LSB TAS2770_REG(0X0, 0x2A) + /* Interrupt Configuration */ +#define TAS2770_INT_CFG TAS2770_REG(0X0, 0x30) + /* Misc IRQ */ +#define TAS2770_MISC_IRQ TAS2770_REG(0X0, 0x32) + /* Clock Configuration */ +#define TAS2770_CLK_CGF TAS2770_REG(0X0, 0x3C) + /* TDM Clock detection monitor */ +#define TAS2770_TDM_CLK_DETC TAS2770_REG(0X0, 0x77) + /* Revision and PG ID */ +#define TAS2770_REV_AND_GPID TAS2770_REG(0X0, 0x7D) + +#define TAS2770_POWER_ACTIVE 0 +#define TAS2770_POWER_MUTE 1 +#define TAS2770_POWER_SHUTDOWN 2 +#define ERROR_OVER_CURRENT 0x0000001 +#define ERROR_DIE_OVERTEMP 0x0000002 +#define ERROR_OVER_VOLTAGE 0x0000004 +#define ERROR_UNDER_VOLTAGE 0x0000008 +#define ERROR_BROWNOUT 0x0000010 +#define ERROR_CLASSD_PWR 0x0000020 +#define TAS2770_SLOT_16BIT 16 +#define TAS2770_SLOT_32BIT 32 +#define TAS2770_I2C_RETRY_COUNT 3 +#define ERROR_I2C_FAILED -2 + +struct tas2770_register { + int book; + int page; + int reg; +}; + +struct tas2770_dai_cfg { + unsigned int dai_fmt; + unsigned int tdm_delay; +}; + +struct tas2770_priv { + struct device *dev; + struct regmap *regmap; + struct snd_soc_codec *codec; + struct snd_soc_component *component; + struct mutex dev_lock; + struct hrtimer mtimer; + int mn_clkin; + int mn_clkid; + int mn_power_state; + int mn_current_book; + int mn_current_page; + int mn_load; + int mn_asi_format; + struct gpio_desc *mn_reset_gpio; + int mn_irq; + bool mb_irq_enable; + int mn_sampling_rate; + int mn_frame_size; + int mn_pll; + int mn_left_slot; + int mn_right_slot; + int mn_ch_size; + int mn_slot_width; + int mn_v_mon_slot_no; + int mn_i_mon_slot_no; + bool mb_runtime_suspend; + unsigned int mn_err_code; + struct mutex codec_lock; +}; + +#endif /* __TAS2770__ */ -- 2.7.4 Frank Shi 石扶 Thunder Software Technology Co., Ltd. 中科创达软件股份有限公司 Mobile: +86-13002957678 | Email: shifu0704(a)thundersoft.com Add: 10F, #1, Digital China Technology Park, No.20, Zhangba Rd. 4th, High-tech Zone, Xi'an City, Shaanxi Province, 710075, P.R. China 西安市高新区丈八四路20号神州数码科技园1号楼10层，710075 www.thundersoft.com CONFIDENTIALITY. This e-mail and its attachments are intended only for the person or entity whose address is listed above and may contain confidential and/or privileged material. Any retransmission, dissemination or other use of this information by persons or entities other than the intended recipient is prohibited. If you received this e-mail in error, please do not disclose the contents to anyone, but notify the sender by email immediately, and delete this e-mail and any attachments from your system. Thank you. 保密提示。本电子邮件及其附件仅适用于上述地址的个人或实体，并可能包含机密和/或特许资料。禁止对目标收件人以外的个人或实体进行此类信息的任何转播，传播或其他使用。如果您误收了此电子邮件，请不要向任何人透露这些内容，而应立即以邮件形式通知发件人，并从您的系统中删除本邮件和任何附件。谢谢。

2 1

[alsa-devel] [PATCH -next] ASoC: amd: use devm_platform_ioremap_resource() to simplify code
by YueHaibing 04 Sep '19

04 Sep '19

Use devm_platform_ioremap_resource() to simplify the code a bit. This is detected by coccinelle. Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/soc/amd/acp-pcm-dma.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c index d26653f..52225b4 100644 --- a/sound/soc/amd/acp-pcm-dma.c +++ b/sound/soc/amd/acp-pcm-dma.c @@ -1251,8 +1251,7 @@ static int acp_audio_probe(struct platform_device *pdev) if (!audio_drv_data) return -ENOMEM; - res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - audio_drv_data->acp_mmio = devm_ioremap_resource(&pdev->dev, res); + audio_drv_data->acp_mmio = devm_platform_ioremap_resource(pdev, 0); if (IS_ERR(audio_drv_data->acp_mmio)) return PTR_ERR(audio_drv_data->acp_mmio); -- 2.7.4

2 1

[alsa-devel] [PATCH -next] ASoC: ep93xx: use devm_platform_ioremap_resource() to simplify code
by YueHaibing 04 Sep '19

04 Sep '19

Use devm_platform_ioremap_resource() to simplify the code a bit. This is detected by coccinelle. Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/soc/cirrus/ep93xx-ac97.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/cirrus/ep93xx-ac97.c b/sound/soc/cirrus/ep93xx-ac97.c index 84c967f..e21eaa1 100644 --- a/sound/soc/cirrus/ep93xx-ac97.c +++ b/sound/soc/cirrus/ep93xx-ac97.c @@ -362,7 +362,6 @@ static const struct snd_soc_component_driver ep93xx_ac97_component = { static int ep93xx_ac97_probe(struct platform_device *pdev) { struct ep93xx_ac97_info *info; - struct resource *res; int irq; int ret; @@ -370,8 +369,7 @@ static int ep93xx_ac97_probe(struct platform_device *pdev) if (!info) return -ENOMEM; - res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - info->regs = devm_ioremap_resource(&pdev->dev, res); + info->regs = devm_platform_ioremap_resource(pdev, 0); if (IS_ERR(info->regs)) return PTR_ERR(info->regs); -- 2.7.4

2 1

[alsa-devel] [PATCH -next] ASoC: sirf-audio: use devm_platform_ioremap_resource() to simplify code
by YueHaibing 04 Sep '19

04 Sep '19

Use devm_platform_ioremap_resource() to simplify the code a bit. This is detected by coccinelle. Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/soc/codecs/sirf-audio-codec.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/codecs/sirf-audio-codec.c b/sound/soc/codecs/sirf-audio-codec.c index 9009a74..a061d78 100644 --- a/sound/soc/codecs/sirf-audio-codec.c +++ b/sound/soc/codecs/sirf-audio-codec.c @@ -459,7 +459,6 @@ static int sirf_audio_codec_driver_probe(struct platform_device *pdev) int ret; struct sirf_audio_codec *sirf_audio_codec; void __iomem *base; - struct resource *mem_res; sirf_audio_codec = devm_kzalloc(&pdev->dev, sizeof(struct sirf_audio_codec), GFP_KERNEL); @@ -468,8 +467,7 @@ static int sirf_audio_codec_driver_probe(struct platform_device *pdev) platform_set_drvdata(pdev, sirf_audio_codec); - mem_res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - base = devm_ioremap_resource(&pdev->dev, mem_res); + base = devm_platform_ioremap_resource(pdev, 0); if (IS_ERR(base)) return PTR_ERR(base); -- 2.7.4

2 1

[alsa-devel] [PATCH -next] ASoC: tegra: use devm_platform_ioremap_resource() to simplify code
by YueHaibing 04 Sep '19

04 Sep '19

Use devm_platform_ioremap_resource() to simplify the code a bit. This is detected by coccinelle. Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/soc/tegra/tegra30_ahub.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/soc/tegra/tegra30_ahub.c b/sound/soc/tegra/tegra30_ahub.c index 9523812..635eacb 100644 --- a/sound/soc/tegra/tegra30_ahub.c +++ b/sound/soc/tegra/tegra30_ahub.c @@ -511,7 +511,7 @@ static int tegra30_ahub_probe(struct platform_device *pdev) const struct tegra30_ahub_soc_data *soc_data; struct reset_control *rst; int i; - struct resource *res0, *res1; + struct resource *res0; void __iomem *regs_apbif, *regs_ahub; int ret = 0; @@ -587,8 +587,7 @@ static int tegra30_ahub_probe(struct platform_device *pdev) } regcache_cache_only(ahub->regmap_apbif, true); - res1 = platform_get_resource(pdev, IORESOURCE_MEM, 1); - regs_ahub = devm_ioremap_resource(&pdev->dev, res1); + regs_ahub = devm_platform_ioremap_resource(pdev, 1); if (IS_ERR(regs_ahub)) return PTR_ERR(regs_ahub); -- 2.7.4

2 1

[alsa-devel] [RFC PATCH 0/5] ASoC: SOF: Intel: SoundWire initial integration
by Pierre-Louis Bossart 04 Sep '19

04 Sep '19

This RFC is the companion of the other RFC 'soundwire: intel: simplify DAI/PDI handling'. Our purpose at this point is to gather feedback on the interfaces between the Intel SOF parts and the SoundWire code. The suggested solution is a simple init/release inserted at probe/remove and resume/suspend, as well as two callbacks for the SOF driver to generate IPC configurations with the firmware. That level of separation completely hides the details of the SoundWire DAIs and will allow for 'transparent' multi-cpu DAI support, which will be handled in the machine driver and the soundwire DAIs. This solution was tested on IceLake and CometLake, and captures the feedback from SOF contributors on an initial integration that was deemed too complicated (and rightly so). Pierre-Louis Bossart (5): ASoC: SOF: IPC: dai-intel: move ALH declarations in header file ASoC: SOF: Intel: hda: add helper to initialize SoundWire IP ASoC: SOF: Intel: hda: add SoundWire IP support ASoC: SOF: Intel: hda: add SoundWire stream config/free callbacks ASoC: SOF: Intel: add support for SoundWire suspend/resume include/sound/sof/dai-intel.h | 18 ++-- sound/soc/sof/intel/hda-dsp.c | 11 +++ sound/soc/sof/intel/hda.c | 157 ++++++++++++++++++++++++++++++++++ sound/soc/sof/intel/hda.h | 11 +++ 4 files changed, 188 insertions(+), 9 deletions(-) base-commit: 3b3aaa017e8072b1bfddda92be296b3463d870be -- 2.20.1

3 19

[alsa-devel] [PATCH v3 1/4] ASoC: es8316: judge PCM rate at later timing
by Katsuhiro Suzuki 04 Sep '19

04 Sep '19

This patch change the judge timing about playing/capturing PCM rate. Original code set constraints list of PCM rate limits at set_sysclk. This strategy works well if system is using fixed rate clock. But some boards and SoC (such as RockPro64 and RockChip I2S) has connected SoC MCLK out to ES8316 MCLK in. In this case, SoC side I2S will choose suitable frequency of MCLK such as fs * mclk-fs when user starts playing or capturing. Bad scenario as follows (mclk-fs = 256): - Initialize sysclk by correct value (Ex. 12.288MHz) - ES8316 set constraints of PCM rate by sysclk 48kHz (1/256), 32kHz (1/384), 30.720kHz (1/400), 24kHz (1/512), 16kHz (1/768), 12kHz (1/1024) - Play 48kHz sound, it's acceptable - Sysclk is not changed - Play 32kHz sound, it's acceptable - Set sysclk by 8.192MHz (= fs * mclk-fs = 32k * 256) - ES8316 set constraints of PCM rate by sysclk 32kHz (1/256), 21.33kHz (1/384), 20.48kHz (1/400), 16kHz (1/512), 10.66kHz (1/768), 8kHz (1/1024) - Play 48kHz again, but it's NOT acceptable because constraints list does not allow 48kHz Root cause of this strange behavior is changing constraints list at set_sysclk timing. It seems that is too early to determine. So this patch does not use constraints list and check PCM rate limit more later timing at hw_params. Signed-off-by: Katsuhiro Suzuki <katsuhiro(a)katsuster.net> --- sound/soc/codecs/es8316.c | 37 +++++++++++++++---------------------- 1 file changed, 15 insertions(+), 22 deletions(-) diff --git a/sound/soc/codecs/es8316.c b/sound/soc/codecs/es8316.c index ed2959dbe1fb..229808fa627c 100644 --- a/sound/soc/codecs/es8316.c +++ b/sound/soc/codecs/es8316.c @@ -363,27 +363,12 @@ static int es8316_set_dai_sysclk(struct snd_soc_dai *codec_dai, { struct snd_soc_component *component = codec_dai->component; struct es8316_priv *es8316 = snd_soc_component_get_drvdata(component); - int i; - int count = 0; es8316->sysclk = freq; if (freq == 0) return 0; - /* Limit supported sample rates to ones that can be autodetected - * by the codec running in slave mode. - */ - for (i = 0; i < NR_SUPPORTED_MCLK_LRCK_RATIOS; i++) { - const unsigned int ratio = supported_mclk_lrck_ratios[i]; - - if (freq % ratio == 0) - es8316->allowed_rates[count++] = freq / ratio; - } - - es8316->sysclk_constraints.list = es8316->allowed_rates; - es8316->sysclk_constraints.count = count; - return 0; } @@ -449,13 +434,6 @@ static int es8316_pcm_startup(struct snd_pcm_substream *substream, return -EINVAL; } - /* The set of sample rates that can be supported depends on the - * MCLK supplied to the CODEC. - */ - snd_pcm_hw_constraint_list(substream->runtime, 0, - SNDRV_PCM_HW_PARAM_RATE, - &es8316->sysclk_constraints); - return 0; } @@ -466,12 +444,27 @@ static int es8316_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_soc_component *component = dai->component; struct es8316_priv *es8316 = snd_soc_component_get_drvdata(component); u8 wordlen = 0; + int i; if (!es8316->sysclk) { dev_err(component->dev, "No MCLK configured\n"); return -EINVAL; } + /* Limit supported sample rates to ones that can be autodetected + * by the codec running in slave mode. + */ + for (i = 0; i < NR_SUPPORTED_MCLK_LRCK_RATIOS; i++) { + const unsigned int ratio = supported_mclk_lrck_ratios[i]; + + if (es8316->sysclk % ratio != 0) + continue; + if (es8316->sysclk / ratio == params_rate(params)) + break; + } + if (i == NR_SUPPORTED_MCLK_LRCK_RATIOS) + return -EINVAL; + switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: wordlen = ES8316_SERDATA2_LEN_16; -- 2.23.0.rc1

3 12

[alsa-devel] [PATCH] ALSA: hda/realtek - Fix the problem of two front mics on a ThinkCentre
by Hui Wang 04 Sep '19

04 Sep '19

This ThinkCentre machine has a new realtek codec alc222, it is not in the support list, we add it in the realtek.c then this machine can apply FIXUPs for the realtek codec. And this machine has two front mics which can't be handled by PA so far, it uses the pin 0x18 and 0x19 as the front mics, as a result the existing FIXUP ALC294_FIXUP_LENOVO_MIC_LOCATION doesn't work on this machine. Fortunately another FIXUP ALC283_FIXUP_HEADSET_MIC also can change the location for one of the two mics on this machine. Cc: <stable(a)vger.kernel.org> Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 3b9583aa63c1..74697c1dd1d5 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7151,6 +7151,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x312a, "ThinkCentre Station", ALC294_FIXUP_LENOVO_MIC_LOCATION), SND_PCI_QUIRK(0x17aa, 0x312f, "ThinkCentre Station", ALC294_FIXUP_LENOVO_MIC_LOCATION), SND_PCI_QUIRK(0x17aa, 0x313c, "ThinkCentre Station", ALC294_FIXUP_LENOVO_MIC_LOCATION), + SND_PCI_QUIRK(0x17aa, 0x3151, "ThinkCentre Station", ALC283_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x17aa, 0x3902, "Lenovo E50-80", ALC269_FIXUP_DMIC_THINKPAD_ACPI), SND_PCI_QUIRK(0x17aa, 0x3977, "IdeaPad S210", ALC283_FIXUP_INT_MIC), SND_PCI_QUIRK(0x17aa, 0x3978, "Lenovo B50-70", ALC269_FIXUP_DMIC_THINKPAD_ACPI), @@ -9038,6 +9039,7 @@ static int patch_alc680(struct hda_codec *codec) static const struct hda_device_id snd_hda_id_realtek[] = { HDA_CODEC_ENTRY(0x10ec0215, "ALC215", patch_alc269), HDA_CODEC_ENTRY(0x10ec0221, "ALC221", patch_alc269), + HDA_CODEC_ENTRY(0x10ec0222, "ALC222", patch_alc269), HDA_CODEC_ENTRY(0x10ec0225, "ALC225", patch_alc269), HDA_CODEC_ENTRY(0x10ec0231, "ALC231", patch_alc269), HDA_CODEC_ENTRY(0x10ec0233, "ALC233", patch_alc269), -- 2.17.1

3 3

[alsa-devel] [PATCH] drm: bridge/dw_hdmi: add audio sample channel status setting
by Cheng-Yi Chiang 04 Sep '19

04 Sep '19

From: Yakir Yang <ykk(a)rock-chips.com> When transmitting IEC60985 linear PCM audio, we configure the Audio Sample Channel Status information of all the channel status bits in the IEC60958 frame. Refer to 60958-3 page 10 for frequency, original frequency, and wordlength setting. This fix the issue that audio does not come out on some monitors (e.g. LG 22CV241) Signed-off-by: Yakir Yang <ykk(a)rock-chips.com> Signed-off-by: Cheng-Yi Chiang <cychiang(a)chromium.org> --- drivers/gpu/drm/bridge/synopsys/dw-hdmi.c | 59 +++++++++++++++++++++++ drivers/gpu/drm/bridge/synopsys/dw-hdmi.h | 20 ++++++++ 2 files changed, 79 insertions(+) diff --git a/drivers/gpu/drm/bridge/synopsys/dw-hdmi.c b/drivers/gpu/drm/bridge/synopsys/dw-hdmi.c index bd65d0479683..34d46e25d610 100644 --- a/drivers/gpu/drm/bridge/synopsys/dw-hdmi.c +++ b/drivers/gpu/drm/bridge/synopsys/dw-hdmi.c @@ -582,6 +582,63 @@ static unsigned int hdmi_compute_n(unsigned int freq, unsigned long pixel_clk) return n; } +static void hdmi_set_schnl(struct dw_hdmi *hdmi) +{ + u8 aud_schnl_samplerate; + u8 aud_schnl_8; + + /* These registers are on RK3288 using version 2.0a. */ + if (hdmi->version != 0x200a) + return; + + switch (hdmi->sample_rate) { + case 32000: + aud_schnl_samplerate = HDMI_FC_AUDSCHNLS7_SMPRATE_32K; + break; + case 44100: + aud_schnl_samplerate = HDMI_FC_AUDSCHNLS7_SMPRATE_44K1; + break; + case 48000: + aud_schnl_samplerate = HDMI_FC_AUDSCHNLS7_SMPRATE_48K; + break; + case 88200: + aud_schnl_samplerate = HDMI_FC_AUDSCHNLS7_SMPRATE_88K2; + break; + case 96000: + aud_schnl_samplerate = HDMI_FC_AUDSCHNLS7_SMPRATE_96K; + break; + case 176400: + aud_schnl_samplerate = HDMI_FC_AUDSCHNLS7_SMPRATE_176K4; + break; + case 192000: + aud_schnl_samplerate = HDMI_FC_AUDSCHNLS7_SMPRATE_192K; + break; + case 768000: + aud_schnl_samplerate = HDMI_FC_AUDSCHNLS7_SMPRATE_768K; + break; + default: + dev_warn(hdmi->dev, "Unsupported audio sample rate (%u)\n", + hdmi->sample_rate); + return; + } + + /* set channel status register */ + hdmi_modb(hdmi, aud_schnl_samplerate, HDMI_FC_AUDSCHNLS7_SMPRATE_MASK, + HDMI_FC_AUDSCHNLS7); + + /* + * Set original frequency to be the same as frequency. + * Use one-complement value as stated in IEC60958-3 page 13. + */ + aud_schnl_8 = (~aud_schnl_samplerate) << + HDMI_FC_AUDSCHNLS8_ORIGSAMPFREQ_OFFSET; + + /* This means word length is 16 bit. Refer to IEC60958-3 page 12. */ + aud_schnl_8 |= 2 << HDMI_FC_AUDSCHNLS8_WORDLEGNTH_OFFSET; + + hdmi_writeb(hdmi, aud_schnl_8, HDMI_FC_AUDSCHNLS8); +} + static void hdmi_set_clk_regenerator(struct dw_hdmi *hdmi, unsigned long pixel_clk, unsigned int sample_rate) { @@ -620,6 +677,8 @@ static void hdmi_set_clk_regenerator(struct dw_hdmi *hdmi, hdmi->audio_cts = cts; hdmi_set_cts_n(hdmi, cts, hdmi->audio_enable ? n : 0); spin_unlock_irq(&hdmi->audio_lock); + + hdmi_set_schnl(hdmi); } static void hdmi_init_clk_regenerator(struct dw_hdmi *hdmi) diff --git a/drivers/gpu/drm/bridge/synopsys/dw-hdmi.h b/drivers/gpu/drm/bridge/synopsys/dw-hdmi.h index 6988f12d89d9..619ebc1c8354 100644 --- a/drivers/gpu/drm/bridge/synopsys/dw-hdmi.h +++ b/drivers/gpu/drm/bridge/synopsys/dw-hdmi.h @@ -158,6 +158,17 @@ #define HDMI_FC_SPDDEVICEINF 0x1062 #define HDMI_FC_AUDSCONF 0x1063 #define HDMI_FC_AUDSSTAT 0x1064 +#define HDMI_FC_AUDSV 0x1065 +#define HDMI_FC_AUDSU 0x1066 +#define HDMI_FC_AUDSCHNLS0 0x1067 +#define HDMI_FC_AUDSCHNLS1 0x1068 +#define HDMI_FC_AUDSCHNLS2 0x1069 +#define HDMI_FC_AUDSCHNLS3 0x106a +#define HDMI_FC_AUDSCHNLS4 0x106b +#define HDMI_FC_AUDSCHNLS5 0x106c +#define HDMI_FC_AUDSCHNLS6 0x106d +#define HDMI_FC_AUDSCHNLS7 0x106e +#define HDMI_FC_AUDSCHNLS8 0x106f #define HDMI_FC_DATACH0FILL 0x1070 #define HDMI_FC_DATACH1FILL 0x1071 #define HDMI_FC_DATACH2FILL 0x1072 @@ -706,6 +717,15 @@ enum { /* HDMI_FC_AUDSCHNLS7 field values */ HDMI_FC_AUDSCHNLS7_ACCURACY_OFFSET = 4, HDMI_FC_AUDSCHNLS7_ACCURACY_MASK = 0x30, + HDMI_FC_AUDSCHNLS7_SMPRATE_MASK = 0x0f, + HDMI_FC_AUDSCHNLS7_SMPRATE_192K = 0xe, + HDMI_FC_AUDSCHNLS7_SMPRATE_176K4 = 0xc, + HDMI_FC_AUDSCHNLS7_SMPRATE_96K = 0xa, + HDMI_FC_AUDSCHNLS7_SMPRATE_768K = 0x9, + HDMI_FC_AUDSCHNLS7_SMPRATE_88K2 = 0x8, + HDMI_FC_AUDSCHNLS7_SMPRATE_32K = 0x3, + HDMI_FC_AUDSCHNLS7_SMPRATE_48K = 0x2, + HDMI_FC_AUDSCHNLS7_SMPRATE_44K1 = 0x0, /* HDMI_FC_AUDSCHNLS8 field values */ HDMI_FC_AUDSCHNLS8_ORIGSAMPFREQ_MASK = 0xf0, -- 2.23.0.187.g17f5b7556c-goog

5 13

[alsa-devel] [PATCH] soundwire: bus: set initial value to port_status
by Bard liao 04 Sep '19

04 Sep '19

From: Bard Liao <yung-chuan.liao(a)linux.intel.com> port_status[port_num] are assigned for each port_num in some if conditions. So some of the port_status may not be initialized. Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> --- drivers/soundwire/bus.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/drivers/soundwire/bus.c b/drivers/soundwire/bus.c index f6a1e4b4813d..33f41b3e642e 100644 --- a/drivers/soundwire/bus.c +++ b/drivers/soundwire/bus.c @@ -835,7 +835,7 @@ static int sdw_handle_port_interrupt(struct sdw_slave *slave, static int sdw_handle_slave_alerts(struct sdw_slave *slave) { struct sdw_slave_intr_status slave_intr; - u8 clear = 0, bit, port_status[15]; + u8 clear = 0, bit, port_status[15] = {0}; int port_num, stat, ret, count = 0; unsigned long port; bool slave_notify = false; -- 2.17.1

3 2

[alsa-devel] some lock needed for flush_delayed_work ?
by Kuninori Morimoto 04 Sep '19

04 Sep '19

Hi ALSA ML I got below error when I unbind/bind CPU DAI. It seems we need to have some kind of lock ? I think => message came from linux/kernel/locking/lockdep.c :: assign_lock_key ... [ 6.392640] asoc-audio-graph-card sound: ASoC: CPU DAI not registered => [ 6.399430] INFO: trying to register non-static key. => [ 6.404393] the code is fine but needs lockdep annotation. => [ 6.409875] turning off the locking correctness validator. [ 6.415362] CPU: 2 PID: 193 Comm: test-bind-unbin Not tainted 5.3.0-rc1+ #625 [ 6.422495] Hardware name: Renesas H3ULCB Kingfisher board based on r8a7795 ES2.0+ (DT) [ 6.430497] Call trace: [ 6.432949] dump_backtrace+0x0/0x140 [ 6.436609] show_stack+0x24/0x30 [ 6.439927] dump_stack+0xc4/0x10c [ 6.443328] register_lock_class+0x4e0/0x618 [ 6.447594] __lock_acquire+0x7c/0x1de8 [ 6.451426] lock_acquire+0xdc/0x1e0 [ 6.455000] del_timer_sync+0x60/0xe8 [ 6.458660] flush_delayed_work+0x2c/0x70 [ 6.462671] soc_cleanup_card_resources+0x50/0x348 [ 6.467459] snd_soc_bind_card+0x1e4/0x990 [ 6.471551] snd_soc_add_component+0x248/0x310 [ 6.475992] snd_soc_register_component+0x54/0x70 [ 6.480696] devm_snd_soc_register_component+0x60/0xb8 [ 6.485833] rsnd_probe+0x164/0x448 [ 6.489321] platform_drv_probe+0x58/0xa8 [ 6.493330] really_probe+0xec/0x2a0 [ 6.496902] driver_probe_device+0x5c/0xf0 [ 6.500996] device_driver_attach+0x74/0x80 [ 6.505177] bind_store+0xdc/0x120 [ 6.508575] drv_attr_store+0x40/0x58 [ 6.512237] sysfs_kf_write+0x50/0x78 [ 6.515896] kernfs_fop_write+0xf0/0x1e8 [ 6.519819] __vfs_write+0x48/0x90 [ 6.523218] vfs_write+0xac/0x1b8 [ 6.526529] ksys_write+0x74/0xf8 [ 6.529840] __arm64_sys_write+0x24/0x30 [ 6.533762] el0_svc_common.constprop.0+0x98/0x170 [ 6.538550] el0_svc_compat_handler+0x2c/0x38 [ 6.542904] el0_svc_compat+0x8/0x10 Thank you for your help !! Best regards --- Kuninori Morimoto

2 2

[alsa-devel] snd_usb_audio - ALSA woke us up to write...was actually nothing to write
by Nathan Royce 03 Sep '19

03 Sep '19

***** ... Sep 03 14:32:58 username kernel: perf: interrupt took too long (4174 > 4156), lowering kernel.perf_event_max_sample_rate to 47700 Sep 03 14:33:03 username plasmashell[1057]: qml: temp unit: 0 Sep 03 14:33:14 username org_kde_powerdevil[1089]: powerdevil: Releasing inhibition with cookie 14 Sep 03 14:33:14 username org_kde_powerdevil[1089]: powerdevil: Releasing inhibition with cookie 15 Sep 03 14:33:14 username org_kde_powerdevil[1089]: powerdevil: Restoring DPMS features after inhibition release Sep 03 14:33:14 username org_kde_powerdevil[1089]: powerdevil: Can't contact ck Sep 03 14:33:14 username org_kde_powerdevil[1089]: powerdevil: Releasing inhibition with cookie 16 Sep 03 14:33:14 username org_kde_powerdevil[1089]: powerdevil: Can't contact ck Sep 03 14:33:15 username plasmashell[1057]: qml: temp unit: 0 ... Sep 03 14:33:23 username pulseaudio[1081]: E: [alsa-sink-USB Audio] alsa-sink.c: ALSA woke us up to write new data to the device, but there was actually nothing to write. Sep 03 14:33:23 username pulseaudio[1081]: E: [alsa-sink-USB Audio] alsa-sink.c: Most likely this is a bug in the ALSA driver 'snd_usb_audio'. Please report this issue to the ALSA developers. Sep 03 14:33:23 username pulseaudio[1081]: E: [alsa-sink-USB Audio] alsa-sink.c: We were woken up with POLLOUT set -- however a subsequent snd_pcm_avail() returned 0 or another value < min_avail. Sep 03 14:33:24 username plasmashell[1057]: qml: temp unit: 0 ... ***** During this time period, I do believe I was starting an android emulator and also watching a twitter video that had audio.

1 0

[alsa-devel] [PATCH] ASoC: soc-core: add comment to jack at soc_remove_component()
by Kuninori Morimoto 03 Sep '19

03 Sep '19

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Basically, driver which setups snd_soc_component_set_jack() need to release it by themselves. But, as framework level robustness, soc_remove_component() also releases it. To avoid code reader confuse, this patch makes it clarify. This patch makes it clarify. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/soc-core.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 659b90a..60f7e8b 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1171,7 +1171,9 @@ static void soc_remove_component(struct snd_soc_component *component) if (component->probed) snd_soc_component_remove(component); + /* For framework level robustness */ snd_soc_component_set_jack(component, NULL, NULL); + list_del(&component->card_list); snd_soc_dapm_free(snd_soc_component_get_dapm(component)); soc_cleanup_component_debugfs(component); -- 2.7.4

2 1

[alsa-devel] sound/usb kernel modules as a DKMS package.
by Hin-Tak Leung 03 Sep '19

03 Sep '19

Hi all, I (We... this is kind of my 9-5 job) have been having some reliability issue with a Focusrite Scarlett Gen 2 18i20, and I was almost tempted to start looking at the windows driver, when I came upon Geoffrey's work. So I made the bleeding-edge sound/usb modules (for the 5.4 time frame) from Takashi's tree built as dkms modules against main line 5.2 . I hope this is useful for other people using USB sound devices. https://github.com/HinTak/sound-usb-dkms This make it easier / faster to try out new code, without building the whole kernel. I would appreciate tips and tricks about the Scarlett 18i20, as well as general recommendation for new professional gear of the same sort to buy. Incidentally, Tobias Hoffmann (who wrote the first gen driver about 5 years ago) was my GSoC student working on printer-related matters more than a decade ago; I did some wifi/USB related work in the kernel (am listed in linux/MAINTAINERS) so you can assume that I am relatively happy with driver-related matters. Cheers, Hin-Tak

2 3

[alsa-devel] [PATCH 1/2] conf/ucm: Add UCM profile for cht-bsw-rt5672 boards
by Hans de Goede 03 Sep '19

03 Sep '19

Add an UCM profile for Intel boards with a RT5672 codec. Re-use the existing platform enable and disable sequences for BYT/CHT SST support and add a codecs/rt5672 dir with codec specific enable / disable sequences for the various inputs and outputs. This is partly based on earlier work done here: https://github.com/plbossart/UCM/tree/master/cht-bsw-rt5672 Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- configure.ac | 2 + src/conf/ucm/Makefile.am | 1 + src/conf/ucm/cht-bsw-rt5672/HiFi.conf | 27 +++++++++++ src/conf/ucm/cht-bsw-rt5672/Makefile.am | 4 ++ .../ucm/cht-bsw-rt5672/cht-bsw-rt5672.conf | 6 +++ src/conf/ucm/codecs/Makefile.am | 1 + src/conf/ucm/codecs/rt5672/DMIC1.conf | 28 +++++++++++ src/conf/ucm/codecs/rt5672/DMIC2.conf | 28 +++++++++++ src/conf/ucm/codecs/rt5672/EnableSeq.conf | 48 +++++++++++++++++++ src/conf/ucm/codecs/rt5672/HeadPhones.conf | 23 +++++++++ src/conf/ucm/codecs/rt5672/HeadsetMic.conf | 38 +++++++++++++++ src/conf/ucm/codecs/rt5672/Makefile.am | 5 ++ src/conf/ucm/codecs/rt5672/MonoSpeaker.conf | 26 ++++++++++ src/conf/ucm/codecs/rt5672/Speaker.conf | 22 +++++++++ 14 files changed, 259 insertions(+) create mode 100644 src/conf/ucm/cht-bsw-rt5672/HiFi.conf create mode 100644 src/conf/ucm/cht-bsw-rt5672/Makefile.am create mode 100644 src/conf/ucm/cht-bsw-rt5672/cht-bsw-rt5672.conf create mode 100644 src/conf/ucm/codecs/rt5672/DMIC1.conf create mode 100644 src/conf/ucm/codecs/rt5672/DMIC2.conf create mode 100644 src/conf/ucm/codecs/rt5672/EnableSeq.conf create mode 100644 src/conf/ucm/codecs/rt5672/HeadPhones.conf create mode 100644 src/conf/ucm/codecs/rt5672/HeadsetMic.conf create mode 100644 src/conf/ucm/codecs/rt5672/Makefile.am create mode 100644 src/conf/ucm/codecs/rt5672/MonoSpeaker.conf create mode 100644 src/conf/ucm/codecs/rt5672/Speaker.conf diff --git a/configure.ac b/configure.ac index 2e955760..9ccca7aa 100644 --- a/configure.ac +++ b/configure.ac @@ -741,6 +741,7 @@ AC_OUTPUT(Makefile doc/Makefile doc/pictures/Makefile doc/doxygen.cfg \ src/conf/ucm/bytcr-rt5651-stereo-spk-in1-mic/Makefile \ src/conf/ucm/bytcr-rt5651-stereo-spk-in2-mic/Makefile \ src/conf/ucm/bytcr-rt5651-stereo-spk-in12-mic/Makefile \ + src/conf/ucm/cht-bsw-rt5672/Makefile \ src/conf/ucm/chtnau8824/Makefile \ src/conf/ucm/chtrt5645/Makefile \ src/conf/ucm/chtrt5645-mono-speaker-analog-mic/Makefile \ @@ -772,6 +773,7 @@ AC_OUTPUT(Makefile doc/Makefile doc/pictures/Makefile doc/doxygen.cfg \ src/conf/ucm/codecs/rt5640/Makefile \ src/conf/ucm/codecs/rt5645/Makefile \ src/conf/ucm/codecs/rt5651/Makefile \ + src/conf/ucm/codecs/rt5672/Makefile \ src/conf/ucm/platforms/Makefile \ src/conf/ucm/platforms/bytcr/Makefile \ src/conf/topology/Makefile \ diff --git a/src/conf/ucm/Makefile.am b/src/conf/ucm/Makefile.am index 02257048..2ed4e1a3 100644 --- a/src/conf/ucm/Makefile.am +++ b/src/conf/ucm/Makefile.am @@ -25,6 +25,7 @@ bytcr-rt5651-stereo-spk-dmic-mic \ bytcr-rt5651-stereo-spk-in1-mic \ bytcr-rt5651-stereo-spk-in2-mic \ bytcr-rt5651-stereo-spk-in12-mic \ +cht-bsw-rt5672 \ chtnau8824 \ chtrt5645 \ chtrt5645-mono-speaker-analog-mic \ diff --git a/src/conf/ucm/cht-bsw-rt5672/HiFi.conf b/src/conf/ucm/cht-bsw-rt5672/HiFi.conf new file mode 100644 index 00000000..b83099d8 --- /dev/null +++ b/src/conf/ucm/cht-bsw-rt5672/HiFi.conf @@ -0,0 +1,27 @@ +# Adapted from https://github.com/plbossart/UCM/tree/master/cht-bsw-rt5672 + +SectionVerb { + EnableSequence [ + cdev "hw:chtbswrt5672" + <platforms/bytcr/PlatformEnableSeq.conf> + <codecs/rt5672/EnableSeq.conf> + ] + + DisableSequence [ + cdev "hw:chtbswrt5672" + <platforms/bytcr/PlatformDisableSeq.conf> + ] + + Value { + PlaybackPCM "hw:chtbswrt5672" + CapturePCM "hw:chtbswrt5672" + } +} + +<codecs/rt5672/Speaker.conf> +<codecs/rt5672/MonoSpeaker.conf> +<codecs/rt5672/HeadPhones.conf> + +<codecs/rt5672/DMIC1.conf> +<codecs/rt5672/DMIC2.conf> +<codecs/rt5672/HeadsetMic.conf> diff --git a/src/conf/ucm/cht-bsw-rt5672/Makefile.am b/src/conf/ucm/cht-bsw-rt5672/Makefile.am new file mode 100644 index 00000000..c3d557fa --- /dev/null +++ b/src/conf/ucm/cht-bsw-rt5672/Makefile.am @@ -0,0 +1,4 @@ +alsaconfigdir = @ALSA_CONFIG_DIR@ +ucmdir = $(alsaconfigdir)/ucm/cht-bsw-rt5672 +ucm_DATA = cht-bsw-rt5672.conf HiFi.conf +EXTRA_DIST = $(ucm_DATA) diff --git a/src/conf/ucm/cht-bsw-rt5672/cht-bsw-rt5672.conf b/src/conf/ucm/cht-bsw-rt5672/cht-bsw-rt5672.conf new file mode 100644 index 00000000..c79024d9 --- /dev/null +++ b/src/conf/ucm/cht-bsw-rt5672/cht-bsw-rt5672.conf @@ -0,0 +1,6 @@ +# Adapted from https://github.com/plbossart/UCM/tree/master/cht-bsw-rt5672 + +SectionUseCase."HiFi" { + File "HiFi.conf" + Comment "Play HiFi quality Music" +} diff --git a/src/conf/ucm/codecs/Makefile.am b/src/conf/ucm/codecs/Makefile.am index 5987b9cf..17c4ebf9 100644 --- a/src/conf/ucm/codecs/Makefile.am +++ b/src/conf/ucm/codecs/Makefile.am @@ -4,4 +4,5 @@ es8316 \ rt5640 \ rt5645 \ rt5651 \ +rt5672 \ nau8824 diff --git a/src/conf/ucm/codecs/rt5672/DMIC1.conf b/src/conf/ucm/codecs/rt5672/DMIC1.conf new file mode 100644 index 00000000..9a8db1a5 --- /dev/null +++ b/src/conf/ucm/codecs/rt5672/DMIC1.conf @@ -0,0 +1,28 @@ +SectionDevice."DigitalMic-DMIC1" { + Comment "Internal Digital Microphone on DMIC1" + + ConflictingDevice [ + "DigitalMic-DMIC2" + "HeadsetMic" + ] + + EnableSequence [ + cdev "hw:chtbswrt5672" + cset "name='Int Mic Switch' on" + cset "name='Stereo1 DMIC Mux' DMIC1" + cset "name='Stereo1 ADC 2 Mux' DMIC" + cset "name='Sto1 ADC MIXL ADC2 Switch' on" + cset "name='Sto1 ADC MIXR ADC2 Switch' on" + ] + + DisableSequence [ + cdev "hw:chtbswrt5672" + cset "name='Int Mic Switch' off" + cset "name='Sto1 ADC MIXL ADC2 Switch' off" + cset "name='Sto1 ADC MIXR ADC2 Switch' off" + ] + + Value { + CaptureChannels 2 + } +} diff --git a/src/conf/ucm/codecs/rt5672/DMIC2.conf b/src/conf/ucm/codecs/rt5672/DMIC2.conf new file mode 100644 index 00000000..b960436e --- /dev/null +++ b/src/conf/ucm/codecs/rt5672/DMIC2.conf @@ -0,0 +1,28 @@ +SectionDevice."DigitalMic-DMIC2" { + Comment "Internal Digital Microphone on DMIC2" + + ConflictingDevice [ + "DigitalMic-DMIC1" + "HeadsetMic" + ] + + EnableSequence [ + cdev "hw:chtbswrt5672" + cset "name='Int Mic Switch' on" + cset "name='Stereo1 DMIC Mux' DMIC2" + cset "name='Stereo1 ADC 2 Mux' DMIC" + cset "name='Sto1 ADC MIXL ADC2 Switch' on" + cset "name='Sto1 ADC MIXR ADC2 Switch' on" + ] + + DisableSequence [ + cdev "hw:chtbswrt5672" + cset "name='Int Mic Switch' off" + cset "name='Sto1 ADC MIXL ADC2 Switch' off" + cset "name='Sto1 ADC MIXR ADC2 Switch' off" + ] + + Value { + CaptureChannels 2 + } +} diff --git a/src/conf/ucm/codecs/rt5672/EnableSeq.conf b/src/conf/ucm/codecs/rt5672/EnableSeq.conf new file mode 100644 index 00000000..bd4f357c --- /dev/null +++ b/src/conf/ucm/codecs/rt5672/EnableSeq.conf @@ -0,0 +1,48 @@ +# Playback (I2S1) + +# common +cset "name='DAC1 L Mux' IF1 DAC" +cset "name='DAC1 R Mux' IF1 DAC" +cset "name='DAC1 MIXL DAC1 Switch' on" +cset "name='DAC1 MIXR DAC1 Switch' on" +cset "name='Stereo DAC MIXL DAC L1 Switch' on" +cset "name='Stereo DAC MIXL DAC R1 Switch' off" +cset "name='Stereo DAC MIXR DAC R1 Switch' on" +cset "name='Stereo DAC MIXR DAC L1 Switch' off" + +# speaker +cset "name='PDM1 L Mux' Stereo DAC" +cset "name='PDM1 R Mux' Stereo DAC" + +# headphone +cset "name='HPOVOL MIXL DAC1 Switch' on" +cset "name='HPOVOL MIXR DAC1 Switch' on" +cset "name='HPO MIX HPVOL Switch' on" + +# line out +# cset "name='OUT MIXL DAC L1 Switch' on" +# cset "name='OUT MIXR DAC R1 Switch' on" +# cset "name='LOUT MIX OUTMIX L Switch' on" +# cset "name='LOUT MIX OUTMIX R Switch' on" + +# 47=0dB, 0.375 dB/step +cset "name='ADC Capture Volume' 47" +cset "name='ADC Capture Switch' on" + +# Headset mic is quite soft, boost it a bit, 1 = 20dB which is the first +# available boost step +cset "name='IN1 Boost Volume' 1" + +# Start with all outputs / inputs disabled +cset "name='Ext Spk Switch' off" +cset "name='Headphone Switch' off" +cset "name='Int Mic Switch' off" +cset "name='Headset Mic Switch' off" +cset "name='Sto1 ADC MIXL ADC1 Switch' off" +cset "name='Sto1 ADC MIXR ADC1 Switch' off" +cset "name='Sto1 ADC MIXL ADC2 Switch' off" +cset "name='Sto1 ADC MIXR ADC2 Switch' off" +cset "name='RECMIXL BST1 Switch' off" +cset "name='RECMIXR BST1 Switch' off" +cset "name='RECMIXL BST2 Switch' off" +cset "name='RECMIXR BST2 Switch' off" diff --git a/src/conf/ucm/codecs/rt5672/HeadPhones.conf b/src/conf/ucm/codecs/rt5672/HeadPhones.conf new file mode 100644 index 00000000..cb8aad2d --- /dev/null +++ b/src/conf/ucm/codecs/rt5672/HeadPhones.conf @@ -0,0 +1,23 @@ +SectionDevice."Headphones" { + Comment "Headphones" + + ConflictingDevice [ + "Speaker" + "MonoSpeaker" + ] + + EnableSequence [ + cdev "hw:chtbswrt5672" + cset "name='Headphone Switch' on" + ] + + DisableSequence [ + cdev "hw:chtbswrt5672" + cset "name='Headphone Switch' off" + ] + + Value { + PlaybackChannels "2" + JackControl "Headphone Jack" + } +} diff --git a/src/conf/ucm/codecs/rt5672/HeadsetMic.conf b/src/conf/ucm/codecs/rt5672/HeadsetMic.conf new file mode 100644 index 00000000..cc6d8d56 --- /dev/null +++ b/src/conf/ucm/codecs/rt5672/HeadsetMic.conf @@ -0,0 +1,38 @@ +SectionDevice."HeadsetMic" { + Comment "Headset Microphone" + + ConflictingDevice [ + "DigitalMic-DMIC1" + "DigitalMic-DMIC2" + ] + + EnableSequence [ + cdev "hw:chtbswrt5672" + cset "name='Headset Mic Switch' on" + cset "name='RECMIXL BST1 Switch' on" + cset "name='RECMIXR BST1 Switch' on" + cset "name='Sto1 ADC MIXL ADC1 Switch' on" + cset "name='Sto1 ADC MIXR ADC1 Switch' on" + # For unknown reasons the headset mic sound is only recorded + # on the left channel, we tweak the data-stream reception to + # use the left samples for both slots, so that we get a stereo + # stream with the mic sound on both channels + cset "name='codec_in rx deinterleaver codec_in0_1' 1" + ] + + DisableSequence [ + cdev "hw:chtbswrt5672" + cset "name='Headset Mic Switch' off" + cset "name='RECMIXL BST1 Switch' off" + cset "name='RECMIXR BST1 Switch' off" + cset "name='Sto1 ADC MIXL ADC1 Switch' off" + cset "name='Sto1 ADC MIXR ADC1 Switch' off" + # Undo data-stream reception mono-mix workaround + cset "name='codec_in rx deinterleaver codec_in0_1' 2" + ] + + Value { + CaptureChannels "2" + JackControl "Headset Mic Jack" + } +} diff --git a/src/conf/ucm/codecs/rt5672/Makefile.am b/src/conf/ucm/codecs/rt5672/Makefile.am new file mode 100644 index 00000000..feba0b76 --- /dev/null +++ b/src/conf/ucm/codecs/rt5672/Makefile.am @@ -0,0 +1,5 @@ +alsaconfigdir = @ALSA_CONFIG_DIR@ +ucmdir = $(alsaconfigdir)/ucm/codecs/rt5672 +ucm_DATA = DMIC1.conf DMIC2.conf EnableSeq.conf HeadPhones.conf \ + HeadsetMic.conf MonoSpeaker.conf Speaker.conf +EXTRA_DIST = $(ucm_DATA) diff --git a/src/conf/ucm/codecs/rt5672/MonoSpeaker.conf b/src/conf/ucm/codecs/rt5672/MonoSpeaker.conf new file mode 100644 index 00000000..828a1154 --- /dev/null +++ b/src/conf/ucm/codecs/rt5672/MonoSpeaker.conf @@ -0,0 +1,26 @@ +SectionDevice."MonoSpeaker" { + Comment "Mono Speaker" + + ConflictingDevice [ + "Speaker" + "Headphones" + ] + + EnableSequence [ + cdev "hw:chtbswrt5672" + cset "name='Stereo DAC MIXR DAC R1 Switch' off" + cset "name='Stereo DAC MIXL DAC R1 Switch' on" + cset "name='Ext Spk Switch' on" + ] + + DisableSequence [ + cdev "hw:chtbswrt5672" + cset "name='Ext Spk Switch' off" + cset "name='Stereo DAC MIXL DAC R1 Switch' off" + cset "name='Stereo DAC MIXR DAC R1 Switch' on" + ] + + Value { + PlaybackChannels "2" + } +} diff --git a/src/conf/ucm/codecs/rt5672/Speaker.conf b/src/conf/ucm/codecs/rt5672/Speaker.conf new file mode 100644 index 00000000..f9291e53 --- /dev/null +++ b/src/conf/ucm/codecs/rt5672/Speaker.conf @@ -0,0 +1,22 @@ +SectionDevice."Speaker" { + Comment "Speakers" + + ConflictingDevice [ + "MonoSpeaker" + "Headphones" + ] + + EnableSequence [ + cdev "hw:chtbswrt5672" + cset "name='Ext Spk Switch' on" + ] + + DisableSequence [ + cdev "hw:chtbswrt5672" + cset "name='Ext Spk Switch' off" + ] + + Value { + PlaybackChannels "2" + } +} -- 2.23.0

4 9

[alsa-devel] [RFC PATCH 0/7] adapt SOF to use snd-hda-codec-hdmi
by Kai Vehmanen 03 Sep '19

03 Sep '19

Hi all, here's a RFC patch series that adapts SOF (and one example machine driver) to use snd-hda-codec-hdmi (patch_hdmi.c) codec driver instead of hdac_hdmi (soc/codecs/hdac_hdmi.c). The primary goal is unify the HDMI codec implementation between DSP and non-DSP HDA configurations, offer same interface to user-space and reduce maintenance load for all. Main points I'd like your input on: 1) Is the high-level approach ok? - SOF already uses pci/hda/ for all others codecs, so HDMI has been the sole exception where code has been duplicated. - I've tried to keep changes to hda/hdmi minimal. - This series implements logic to parse the PCM topology in a dynamic fashion, so we do not have to change all existing (and future) DSP topologis to use fixed PCM device numbers for HDMI, and we avoid need to hardcode PCM device numbers in machine driver code. 2) Can we drop hdac_hdmi and its support from machine drivers, or do we need to make it optional and keep it around? - Current series does not add any Kconfig options, but simply switches SOF to use HDA codecs for all, including HDMI/DP. This means hdac_hdmi is never used with SOF and could be dropped (if SST is ok as well). - This may break some usage with SST (input is welcome!) - The change is visible to applications. The ALSA mixer interface is different (OTOH with the new driver, playback works out-of-the-box while with hdac_hdmi you needed to set the multiple controls first, to have any audio out). - Alternatively I could add a KConfig option and we could have a deprecation period for hdac_hdmi, allowing people to compile with the old driver during transition time. This will require #ifdef'ing in all the machine drivers. .. once these are addressed, I can proceed to extend the patchset with all affected machine drivers. Feature and testing info: - Tested on multiple Intel platforms supported by SOF. - Tested with ALSA console tools as well as with Pulseaudio. - requires Pulseaudio 12.x or newer, see https://lists.freedesktop.org/archives/pulseaudio-discuss/2019-August/03135… - HDMI, DP, DP-MST with multi-monitor use-scenarios work ok. - New feature for SOF: ELD /proc fs works just like in DSP-less mode. - New feature for SOF: jack detection works out-of-the-box with Pulseaudio (no need for card specific UCM for HDMI) - Pre-reviewed at: https://github.com/thesofproject/linux/pull/1155 Kai Vehmanen (7): ALSA: hda - add mst_no_extra_pcms flag to hda_codec ASoC: Intel: skl-hda-dsp-generic: use snd-hda-codec-hdmi ASoC: hdac_hda: add support for HDMI/DP as a HDA codec ALSA: hda/hdmi - allow control creation without a linked pcm ALSA: hda/hdmi - implement mst_no_extra_pcms flag ALSA: hda/hdmi - complete pcm_setup_pin without snd_pcm link ASoC: SOF: Intel: load hda codec module also for HDMI/DP include/sound/hda_codec.h | 1 + sound/pci/hda/patch_hdmi.c | 31 ++++--- sound/soc/codecs/hdac_hda.c | 95 +++++++++++++++++--- sound/soc/codecs/hdac_hda.h | 10 ++- sound/soc/intel/boards/skl_hda_dsp_common.c | 27 ++---- sound/soc/intel/boards/skl_hda_dsp_common.h | 61 +++++++++++++ sound/soc/intel/boards/skl_hda_dsp_generic.c | 7 -- sound/soc/sof/intel/hda-codec.c | 11 +-- sound/soc/sof/intel/hda.h | 5 +- 9 files changed, 189 insertions(+), 59 deletions(-) -- 2.17.1

2 14

[alsa-devel] [PATCH v2 1/3] ASoC: es8316: judge PCM rate at later timing
by Katsuhiro Suzuki 03 Sep '19

03 Sep '19

This patch change the judge timing about playing/capturing PCM rate. Original code set constraints list of PCM rate limits at set_sysclk. This strategy works well if system is using fixed rate clock. But some boards and SoC (such as RockPro64 and RockChip I2S) has connected SoC MCLK out to ES8316 MCLK in. In this case, SoC side I2S will choose suitable frequency of MCLK such as fs * mclk-fs when user starts playing or capturing. Bad scenario as follows (mclk-fs = 256): - Initialize sysclk by correct value (Ex. 12.288MHz) - ES8316 set constraints of PCM rate by sysclk 48kHz (1/256), 32kHz (1/384), 30.720kHz (1/400), 24kHz (1/512), 16kHz (1/768), 12kHz (1/1024) - Play 48kHz sound, it's acceptable - Sysclk is not changed - Play 32kHz sound, it's acceptable - Set sysclk by 8.192MHz (= fs * mclk-fs = 32k * 256) - ES8316 set constraints of PCM rate by sysclk 32kHz (1/256), 21.33kHz (1/384), 20.48kHz (1/400), 16kHz (1/512), 10.66kHz (1/768), 8kHz (1/1024) - Play 48kHz again, but it's NOT acceptable because constraints list does not allow 48kHz Root cause of this strange behavior is changing constraints list at set_sysclk timing. It seems that is too early to determine. So this patch does not use constraints list and check PCM rate limit more later timing at hw_params. Signed-off-by: Katsuhiro Suzuki <katsuhiro(a)katsuster.net> --- sound/soc/codecs/es8316.c | 37 +++++++++++++++---------------------- 1 file changed, 15 insertions(+), 22 deletions(-) diff --git a/sound/soc/codecs/es8316.c b/sound/soc/codecs/es8316.c index ed2959dbe1fb..229808fa627c 100644 --- a/sound/soc/codecs/es8316.c +++ b/sound/soc/codecs/es8316.c @@ -363,27 +363,12 @@ static int es8316_set_dai_sysclk(struct snd_soc_dai *codec_dai, { struct snd_soc_component *component = codec_dai->component; struct es8316_priv *es8316 = snd_soc_component_get_drvdata(component); - int i; - int count = 0; es8316->sysclk = freq; if (freq == 0) return 0; - /* Limit supported sample rates to ones that can be autodetected - * by the codec running in slave mode. - */ - for (i = 0; i < NR_SUPPORTED_MCLK_LRCK_RATIOS; i++) { - const unsigned int ratio = supported_mclk_lrck_ratios[i]; - - if (freq % ratio == 0) - es8316->allowed_rates[count++] = freq / ratio; - } - - es8316->sysclk_constraints.list = es8316->allowed_rates; - es8316->sysclk_constraints.count = count; - return 0; } @@ -449,13 +434,6 @@ static int es8316_pcm_startup(struct snd_pcm_substream *substream, return -EINVAL; } - /* The set of sample rates that can be supported depends on the - * MCLK supplied to the CODEC. - */ - snd_pcm_hw_constraint_list(substream->runtime, 0, - SNDRV_PCM_HW_PARAM_RATE, - &es8316->sysclk_constraints); - return 0; } @@ -466,12 +444,27 @@ static int es8316_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_soc_component *component = dai->component; struct es8316_priv *es8316 = snd_soc_component_get_drvdata(component); u8 wordlen = 0; + int i; if (!es8316->sysclk) { dev_err(component->dev, "No MCLK configured\n"); return -EINVAL; } + /* Limit supported sample rates to ones that can be autodetected + * by the codec running in slave mode. + */ + for (i = 0; i < NR_SUPPORTED_MCLK_LRCK_RATIOS; i++) { + const unsigned int ratio = supported_mclk_lrck_ratios[i]; + + if (es8316->sysclk % ratio != 0) + continue; + if (es8316->sysclk / ratio == params_rate(params)) + break; + } + if (i == NR_SUPPORTED_MCLK_LRCK_RATIOS) + return -EINVAL; + switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: wordlen = ES8316_SERDATA2_LEN_16; -- 2.23.0.rc1

2 6

[alsa-devel] [PATCH] MAINTAINERS: update entry for firewire audio drivers with UAPI header
by Takashi Sakamoto 03 Sep '19

03 Sep '19

The most of drivers in ALSA firewire stack supports common ioctl commands to enable/disable packet streaming as well as some ioctl commands for model-specific features. An UAPI header is exported to userspace. This commit adds supplement for entry of ALSA firewire stack with a path of the UAPI header. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- MAINTAINERS | 1 + 1 file changed, 1 insertion(+) diff --git a/MAINTAINERS b/MAINTAINERS index 4200194e69ea..6bd54ce6dc66 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -6271,6 +6271,7 @@ L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) T: git git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git S: Maintained F: sound/firewire/ +F: include/uapi/sound/firewire.h FIREWIRE MEDIA DRIVERS (firedtv) M: Stefan Richter <stefanr(a)s5r6.in-berlin.de> -- 2.20.1

2 1

[alsa-devel] [PATCH] ASoC: simple-card: indicate rebind issue
by Kuninori Morimoto 03 Sep '19

03 Sep '19

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> ALSA SoC try to rebind Sound Card if Card/CPU/Codec/Platform were unbinded and re-binded again. But, Simple Card might can't rebind again if user do for example unbind CPU or Codec driver bind CPU or Codec driver Because Simple Card is still pointing old/unbinded CPU or Codec driver's DAI name at dlc->dai_name. To avoid this issue, it needs to alloc memory and keep DAI name even though if CPU or Codec driver was unbinded. Or, always do unbind/bind at Sound Card. For now, this patch indicates this issue as FIXME. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/generic/simple-card.c | 20 +++++++++++++++++++- 1 file changed, 19 insertions(+), 1 deletion(-) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 0eac3bc..fc9c753 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -46,7 +46,25 @@ static int asoc_simple_parse_dai(struct device_node *node, if (ret) return ret; - /* Get dai->name */ + /* + * FIXME + * + * Here, dlc->dai_name is pointer to CPU/Codec DAI name. + * If user unbinded CPU or Codec driver, but not for Sound Card, + * dlc->dai_name is keeping unbinded CPU or Codec + * driver's pointer. + * + * If user re-bind CPU or Codec driver again, ALSA SoC will try + * to rebind Card via snd_soc_try_rebind_card(), but because of + * above reason, it might can't bind Sound Card. + * Because Sound Card is pointing to released dai_name pointer. + * + * To avoid this rebind Card issue, + * 1) It needs to alloc memory to keep dai_name eventhough + * CPU or Codec driver was unbinded, or + * 2) user need to rebind Sound Card everytime + * if he unbinded CPU or Codec. + */ ret = snd_soc_of_get_dai_name(node, &dlc->dai_name); if (ret < 0) return ret; -- 2.7.4

2 1

[alsa-devel] [PATCH] ASoC: audio-graph: indicate rebind issue
by Kuninori Morimoto 03 Sep '19

03 Sep '19

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> ALSA SoC try to rebind Sound Card if Card/CPU/Codec/Platform were unbinded and re-binded again. But, Audio Graph Card might can't rebind again if user do for example unbind CPU or Codec driver bind CPU or Codec driver Because Audio Graph Card is still pointing old/unbinded CPU or Codec driver's DAI name at dlc->dai_name. To avoid this issue, it needs to alloc memory and keep DAI name even though if CPU or Codec driver was unbinded. Or, always do unbind/bind at Sound Card. For now, this patch indicates this issue as FIXME. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/generic/audio-graph-card.c | 19 +++++++++++++++++++ 1 file changed, 19 insertions(+) diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c index 288df24..6007e63 100644 --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -129,6 +129,25 @@ static int asoc_simple_parse_dai(struct device_node *ep, args.args[0] = graph_get_dai_id(ep); args.args_count = (of_graph_get_endpoint_count(node) > 1); + /* + * FIXME + * + * Here, dlc->dai_name is pointer to CPU/Codec DAI name. + * If user unbinded CPU or Codec driver, but not for Sound Card, + * dlc->dai_name is keeping unbinded CPU or Codec + * driver's pointer. + * + * If user re-bind CPU or Codec driver again, ALSA SoC will try + * to rebind Card via snd_soc_try_rebind_card(), but because of + * above reason, it might can't bind Sound Card. + * Because Sound Card is pointing to released dai_name pointer. + * + * To avoid this rebind Card issue, + * 1) It needs to alloc memory to keep dai_name eventhough + * CPU or Codec driver was unbinded, or + * 2) user need to rebind Sound Card everytime + * if he unbinded CPU or Codec. + */ ret = snd_soc_get_dai_name(&args, &dlc->dai_name); if (ret < 0) return ret; -- 2.7.4

2 1

[alsa-devel] [PATCH] alsamixer: Remove exp10 usage
by Rosen Penev 03 Sep '19

03 Sep '19

exp10 is a GNU extension and not available everywhere (eg. uClibc-ng). Signed-off-by: Rosen Penev <rosenp(a)gmail.com> --- alsamixer/volume_mapping.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/alsamixer/volume_mapping.c b/alsamixer/volume_mapping.c index 48cfbe2..29ab061 100644 --- a/alsamixer/volume_mapping.c +++ b/alsamixer/volume_mapping.c @@ -108,9 +108,9 @@ static double get_normalized_volume(snd_mixer_elem_t *elem, if (use_linear_dB_scale(min, max)) return (value - min) / (double)(max - min); - normalized = exp10((value - max) / 6000.0); + normalized = pow(10, (value - max) / 6000.0); if (min != SND_CTL_TLV_DB_GAIN_MUTE) { - min_norm = exp10((min - max) / 6000.0); + min_norm = pow(10, (min - max) / 6000.0); normalized = (normalized - min_norm) / (1 - min_norm); } @@ -143,7 +143,7 @@ static int set_normalized_volume(snd_mixer_elem_t *elem, } if (min != SND_CTL_TLV_DB_GAIN_MUTE) { - min_norm = exp10((min - max) / 6000.0); + min_norm = pow(10, (min - max) / 6000.0); volume = volume * (1 - min_norm) + min_norm; } value = lrint_dir(6000.0 * log10(volume), dir) + max; -- 2.17.1

2 1

[alsa-devel] [PATCH 1/3] ASoC: xlnx: add Xilinx logicPD-I2S FPGA IP support
by Miquel Raynal 03 Sep '19

03 Sep '19

This IP is very simple so this driver manage both the DAI and the PCM streams, hence the presence of both components in this driver. There are plenty available interruptions when capturing or playing back audio that can be triggered but the only one that fits the ALSA sound system is the XFER_DONE which is used to bound sound periods. Other interrupts are masked. Please note that capture and playback are not possible at the same time though. Capture seems to work (at least it creates a file with something inside) but I have no capture mechanism on the board to actually test that it works correctly. Signed-off-by: Miquel Raynal <miquel.raynal(a)bootlin.com> --- Hello, This is my first contribution in the sound subsystem, I hope I've understood the core but I might be entirely wrong as well, so please do not hesitate to be critical on my choices. Thanks, Miquèl sound/soc/xilinx/Kconfig | 7 + sound/soc/xilinx/Makefile | 2 + sound/soc/xilinx/xlnx-logicpd-i2s.c | 468 ++++++++++++++++++++++++++++ 3 files changed, 477 insertions(+) create mode 100644 sound/soc/xilinx/xlnx-logicpd-i2s.c diff --git a/sound/soc/xilinx/Kconfig b/sound/soc/xilinx/Kconfig index 47f606b924e4..b62cae6750b9 100644 --- a/sound/soc/xilinx/Kconfig +++ b/sound/soc/xilinx/Kconfig @@ -7,6 +7,13 @@ config SND_SOC_XILINX_I2S PCM data. In receiver mode, IP receives PCM audio and encapsulates PCM in AES format and sends AES data. +config SND_SOC_XILINX_LOGICPD_I2S + tristate "Audio support for the Xilinx logicPD I2S" + help + Select this option to enable Xilinx logicPD I2S slave + transceiver. This enables I2S playback and capture using + Xilinx/logicPD IP. + config SND_SOC_XILINX_AUDIO_FORMATTER tristate "Audio support for the the Xilinx audio formatter" help diff --git a/sound/soc/xilinx/Makefile b/sound/soc/xilinx/Makefile index d79fd38b094b..d127c30f8fe2 100644 --- a/sound/soc/xilinx/Makefile +++ b/sound/soc/xilinx/Makefile @@ -1,5 +1,7 @@ snd-soc-xlnx-i2s-objs := xlnx_i2s.o obj-$(CONFIG_SND_SOC_XILINX_I2S) += snd-soc-xlnx-i2s.o +snd-soc-xlnx-logicpd-i2s-objs := xlnx-logicpd-i2s.o +obj-$(CONFIG_SND_SOC_XILINX_LOGICPD_I2S) += snd-soc-xlnx-logicpd-i2s.o snd-soc-xlnx-formatter-pcm-objs := xlnx_formatter_pcm.o obj-$(CONFIG_SND_SOC_XILINX_AUDIO_FORMATTER) += snd-soc-xlnx-formatter-pcm.o snd-soc-xlnx-spdif-objs := xlnx_spdif.o diff --git a/sound/soc/xilinx/xlnx-logicpd-i2s.c b/sound/soc/xilinx/xlnx-logicpd-i2s.c new file mode 100644 index 000000000000..325a5bb6978a --- /dev/null +++ b/sound/soc/xilinx/xlnx-logicpd-i2s.c @@ -0,0 +1,468 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * Xilinx logicPD logiI2S - I2S slave transceiver v2 support + * + * Copyright (C) 2019 Bootlin + * + * Author: Miquel Raynal <miquel.raynal(a)bootlin.com> + */ + +#include <linux/dma-mapping.h> +#include <linux/io.h> +#include <linux/module.h> +#include <linux/of.h> +#include <linux/of_platform.h> +#include <linux/platform_device.h> +#include <sound/dmaengine_pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#define DRV_NAME "xlnx_logicpd_i2s" + +#define IP_VERSION 0x0 +#define PATCH_LEVEL(reg) (((reg) & GENMASK(4, 0)) + 'a') +#define MINOR_REV(reg) (((reg) & GENMASK(10, 5)) >> 5) +#define MAJOR_REV(reg) (((reg) & GENMASK(16, 11)) >> 11) +#define LICENSE_TYPE(reg) (((reg) & GENMASK(18, 17)) >> 17) +#define CONTROL_REG(s) ((s) == SNDRV_PCM_STREAM_PLAYBACK ? 0x4 : 0x24) +#define ENGINE_EN BIT(0) +#define XFER_DONE BIT(1) +#define BUFF_BASE_ADDR_REG(s) ((s) == SNDRV_PCM_STREAM_PLAYBACK ? 0x8 : 0x28) +#define BUFF_LEN_REG(s) ((s) == SNDRV_PCM_STREAM_PLAYBACK ? 0xC : 0x2C) +#define FIFO_STAT_REG(s) ((s) == SNDRV_PCM_STREAM_PLAYBACK ? 0x10 : 0x30) +#define INTR_MASK_REG(s) ((s) == SNDRV_PCM_STREAM_PLAYBACK ? 0x14 : 0x34) +#define XFER_DONE_INTR BIT(31) +#define INTR_STAT_REG(s) ((s) == SNDRV_PCM_STREAM_PLAYBACK ? 0x18 : 0x38) +#define FIFO_COUNT(reg) ((reg) >> 20) + +#define BYTES_TO_WORDS(n) ((n) / 4) + +/* Arbitrarily chosen period size */ +#define PCM_PERIOD_WORDS SZ_8K +#define PCM_PERIOD_BYTES (PCM_PERIOD_WORDS * 4) +/* This is the actual maximum size that can actually be moved in one chunk */ +#define PCM_BUF_WORDS (SZ_64K - 1) +#define PCM_BUF_BYTES (PCM_BUF_WORDS * 4) + +struct xlnx_logicpd_i2s; + +/** + * struct xlnx_logicpd_stream - Internal stream representation + * + * @i2s: Chip data + * @substream: Core substream structure + * @period_idx: Index of the period within the circular buffer + */ +struct xlnx_logicpd_stream { + struct xlnx_logicpd_i2s *i2s; + struct snd_pcm_substream *substream; + unsigned int period_idx; +}; + +/** + * struct xlnx_logicpd_i2s - Chip structure + * + * @base: Registers base address + * @streams: Playback and capture streams in an array + */ +struct xlnx_logicpd_i2s { + void __iomem *base; + struct xlnx_logicpd_stream streams[2]; +}; + +static struct xlnx_logicpd_i2s *substream_to_cpu_dai_chip(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream); + + return snd_soc_dai_get_drvdata(rtd->cpu_dai); +} + +/* PCM methods */ + +static const struct snd_pcm_hardware xlnx_logicpd_pcm_hardware = { + .info = SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_HALF_DUPLEX, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + .rates = SNDRV_PCM_RATE_8000_192000, + .rate_min = 8000, + .rate_max = 192000, + .channels_min = 2, + .channels_max = 2, + .period_bytes_min = 0, + .period_bytes_max = PCM_PERIOD_BYTES, + .periods_min = 0, + .periods_max = -1, + .buffer_bytes_max = PCM_BUF_BYTES, +}; + +static int xlnx_logicpd_pcm_open(struct snd_pcm_substream *substream) +{ + struct xlnx_logicpd_i2s *i2s = substream_to_cpu_dai_chip(substream); + unsigned int dir = substream->stream; + + snd_soc_set_runtime_hwparams(substream, &xlnx_logicpd_pcm_hardware); + + i2s->streams[dir].substream = substream; + + return 0; +} + +static int xlnx_logicpd_pcm_close(struct snd_pcm_substream *substream) +{ + struct xlnx_logicpd_i2s *i2s = substream_to_cpu_dai_chip(substream); + unsigned int dir = substream->stream; + + i2s->streams[dir].substream = NULL; + + return 0; +} + +static int xlnx_logicpd_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + + snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); + runtime->dma_bytes = params_buffer_bytes(params); + + return 0; +} + +static snd_pcm_uframes_t xlnx_logicpd_pcm_pointer(struct snd_pcm_substream *substream) +{ + struct xlnx_logicpd_i2s *i2s = substream_to_cpu_dai_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + unsigned int period_sz = snd_pcm_lib_period_bytes(substream); + unsigned int dir = substream->stream; + + return bytes_to_frames(runtime, + i2s->streams[dir].period_idx * period_sz); +} + +static int xlnx_logicpd_pcm_mmap(struct snd_pcm_substream *substream, + struct vm_area_struct *vma) +{ + return remap_pfn_range(vma, vma->vm_start, + substream->dma_buffer.addr >> PAGE_SHIFT, + vma->vm_end - vma->vm_start, vma->vm_page_prot); +} + +static const struct snd_pcm_ops xlnx_logicpd_pcm_ops = { + .open = xlnx_logicpd_pcm_open, + .close = xlnx_logicpd_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = xlnx_logicpd_pcm_hw_params, + .pointer = xlnx_logicpd_pcm_pointer, + .mmap = xlnx_logicpd_pcm_mmap, +}; + +static int xlnx_logicpd_pcm_new(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_pcm *pcm = rtd->pcm; + struct snd_pcm_substream *substream; + struct snd_dma_buffer *buf; + int dir; + + for (dir = SNDRV_PCM_STREAM_PLAYBACK; + dir <= SNDRV_PCM_STREAM_CAPTURE; dir++) { + substream = pcm->streams[dir].substream; + if (!substream) + continue; + + buf = &substream->dma_buffer; + buf->area = dma_alloc_coherent(pcm->card->dev, PCM_BUF_BYTES, + &buf->addr, GFP_KERNEL); + buf->bytes = PCM_BUF_BYTES; + if (!buf->area) + return -ENOMEM; + } + + return 0; +} + +static void xlnx_logicpd_pcm_free(struct snd_pcm *pcm) +{ + struct snd_pcm_substream *substream; + struct snd_dma_buffer *buf; + int dir; + + for (dir = SNDRV_PCM_STREAM_PLAYBACK; + dir <= SNDRV_PCM_STREAM_CAPTURE; dir++) { + substream = pcm->streams[dir].substream; + if (!substream) + continue; + + buf = &substream->dma_buffer; + if (!buf->area) + continue; + + dma_free_coherent(pcm->card->dev, buf->bytes, + buf->area, buf->addr); + buf->area = NULL; + } +} + +static const struct snd_soc_component_driver xlnx_logicpd_pcm_component = { + .name = "xlnx-logicp-pcm", + .ops = &xlnx_logicpd_pcm_ops, + .pcm_new = xlnx_logicpd_pcm_new, + .pcm_free = xlnx_logicpd_pcm_free, +}; + +/* DAI methods */ + +static void xlnx_logicpd_dai_int_en(struct xlnx_logicpd_i2s *i2s, int dir) +{ + u32 reg; + + reg = readl_relaxed(i2s->base + INTR_MASK_REG(dir)); + reg &= ~XFER_DONE_INTR; + writel(reg, i2s->base + INTR_MASK_REG(dir)); +} + +static void xlnx_logicpd_dai_int_dis(struct xlnx_logicpd_i2s *i2s, int dir) +{ + u32 reg; + + reg = readl_relaxed(i2s->base + INTR_MASK_REG(dir)); + reg |= XFER_DONE_INTR; + writel_relaxed(reg, i2s->base + INTR_MASK_REG(dir)); +} + +static irqreturn_t xlnx_logicpd_dai_isr(int irq, void *dev_id) +{ + struct xlnx_logicpd_stream *stream = dev_id; + struct xlnx_logicpd_i2s *i2s = stream->i2s; + struct snd_pcm_substream *substream = stream->substream; + unsigned int period_sz = snd_pcm_lib_period_bytes(substream); + unsigned int buf_sz = snd_pcm_lib_buffer_bytes(substream); + dma_addr_t buf_addr = substream->dma_buffer.addr; + unsigned int dir = substream->stream; + u32 reg; + + /* Reading INTR_STAT deasserts the host interrupt */ + reg = readl_relaxed(i2s->base + INTR_STAT_REG(dir)); + + /* + * When the XFER_DONE interrupt is triggered, it means the period has + * been entirely shifted into the FIFO. At this point, we can move the + * buffer pointer to the next period and ask to transfer another chunk + * of data. Whenever the FIFO will be at its "almost full" state (4096 + * words minus the threshold of 100 words) the internal DMA engine will + * automatically restart shifting data to the FIFO until its full state. + * Hence, the host has up to 3996 words (in our case, 3996 frames) to + * serve the interrupt before an underrun that would happen, at eg. + * 44100Hz, after 90ms. + */ + if (reg & XFER_DONE_INTR) { + unsigned int offset_in_buf = ++stream->period_idx * period_sz; + + if (offset_in_buf >= buf_sz) { + stream->period_idx = 0; + offset_in_buf = stream->period_idx * period_sz; + } + + /* Move on to the next period in the overall buffer */ + writel_relaxed(buf_addr + offset_in_buf, + i2s->base + BUFF_BASE_ADDR_REG(dir)); + /* The last period might be smaller, update length if needed */ + period_sz = min(period_sz, buf_sz - offset_in_buf); + writel_relaxed(BYTES_TO_WORDS(period_sz), + i2s->base + BUFF_LEN_REG(dir)); + + /* Inform the PCM middle-layer */ + snd_pcm_period_elapsed(substream); + } + + return IRQ_HANDLED; +} + +static int xlnx_logicpd_dai_trigger(struct snd_pcm_substream *substream, + int cmd, struct snd_soc_dai *dai) +{ + struct xlnx_logicpd_i2s *i2s = snd_soc_dai_get_drvdata(dai); + unsigned int period_sz = snd_pcm_lib_period_bytes(substream); + dma_addr_t buf_addr = substream->dma_buffer.addr; + unsigned int dir = substream->stream; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + i2s->streams[dir].period_idx = 0; + /* Disable the other engine if enabled */ + if (readl(i2s->base + CONTROL_REG(!dir)) & ENGINE_EN) + writel(0, i2s->base + CONTROL_REG(!dir)); + /* Enable the desired engine */ + writel_relaxed(ENGINE_EN, i2s->base + CONTROL_REG(dir)); + /* Set the buffer start address */ + writel_relaxed(buf_addr, i2s->base + BUFF_BASE_ADDR_REG(dir)); + /* Enable the XFER_DONE IRQ, signaling the end of the period */ + xlnx_logicpd_dai_int_en(i2s, dir); + /* Actually start the internal DMA engine */ + writel(BYTES_TO_WORDS(period_sz), + i2s->base + BUFF_LEN_REG(dir)); + break; + case SNDRV_PCM_TRIGGER_STOP: + /* Disable the interrupts */ + xlnx_logicpd_dai_int_dis(i2s, dir); + /* Ensure the host IRQ is deasserted */ + readl_relaxed(i2s->base + INTR_STAT_REG(dir)); + break; + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + default: + return -EINVAL; + } + + return 0; +} + +static const struct snd_soc_dai_ops xlnx_logicpd_dai_ops = { + .trigger = xlnx_logicpd_dai_trigger, +}; + +static int xlnx_logicpd_dai_probe(struct snd_soc_dai *dai) +{ + struct xlnx_logicpd_i2s *i2s = snd_soc_dai_get_drvdata(dai); + unsigned int dir; + + for (dir = SNDRV_PCM_STREAM_PLAYBACK; + dir <= SNDRV_PCM_STREAM_CAPTURE; dir++) { + i2s->streams[dir].i2s = i2s; + + /* Reset the transmitter/receiver engine */ + writel_relaxed(0, i2s->base + CONTROL_REG(dir)); + /* Mask all interrupts */ + writel_relaxed(GENMASK(31, 0), i2s->base + INTR_MASK_REG(dir)); + } + + return 0; +} + +struct snd_soc_dai_driver xlnx_logicpd_dai = { + .name = "xylinx-logicpd-dai", + .probe = xlnx_logicpd_dai_probe, + .capture = { + .stream_name = "Capture", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_192000, + .rate_min = 8000, + .rate_max = 192000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_192000, + .rate_min = 8000, + .rate_max = 192000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .symmetric_rates = 1, + .ops = &xlnx_logicpd_dai_ops, +}; + +static const struct snd_soc_component_driver xlnx_logicpd_i2s_component = { + .name = DRV_NAME, + .ops = &xlnx_logicpd_pcm_ops, +}; + +static const struct of_device_id xlnx_logicpd_i2s_of_match[] = { + { + .compatible = "xlnx,logicpd-i2s-dai", + }, + {}, +}; +MODULE_DEVICE_TABLE(of, xlnx_logicpd_i2s_of_match); + +static int xlnx_logicpd_i2s_probe(struct platform_device *pdev) +{ + struct device *dev = &pdev->dev; + struct xlnx_logicpd_i2s *i2s; + struct xlnx_logicpd_stream *stream; + int tx_irq, rx_irq, ret; + u32 reg; + + i2s = devm_kzalloc(dev, sizeof(*i2s), GFP_KERNEL); + if (!i2s) + return -ENOMEM; + + dev_set_drvdata(dev, i2s); + + i2s->base = devm_platform_ioremap_resource(pdev, 0); + if (IS_ERR(i2s->base)) + return PTR_ERR(i2s->base); + + stream = &i2s->streams[SNDRV_PCM_STREAM_PLAYBACK]; + tx_irq = platform_get_irq_byname(pdev, "tx"); + if (tx_irq > 0) { + ret = devm_request_irq(dev, tx_irq, xlnx_logicpd_dai_isr, + 0, "logicpd-i2s-tx", stream); + if (ret) + return ret; + } else { + dev_err(dev, "TX IRQ not available (%d), disabling playback\n", + tx_irq); + tx_irq = 0; + } + + stream = &i2s->streams[SNDRV_PCM_STREAM_CAPTURE]; + rx_irq = platform_get_irq_byname(pdev, "rx"); + if (rx_irq > 0) { + ret = devm_request_irq(dev, rx_irq, xlnx_logicpd_dai_isr, + 0, "logicpd-i2s-rx", stream); + if (ret) + return ret; + } else { + dev_err(dev, "RX IRQ not available (%d), disabling capture\n", + rx_irq); + rx_irq = 0; + } + + if (!tx_irq && !rx_irq) + return -EINVAL; + + ret = devm_snd_soc_register_component(dev, &xlnx_logicpd_pcm_component, + NULL, 0); + if (ret) { + dev_err(dev, "cannot register PCM component (%d)\n", ret); + return ret; + } + + ret = devm_snd_soc_register_component(dev, &xlnx_logicpd_i2s_component, + &xlnx_logicpd_dai, 1); + if (ret) { + dev_err(dev, "cannot register I2S component (%d)\n", ret); + return ret; + } + + reg = readl_relaxed(i2s->base + IP_VERSION); + dev_info(dev, "%s DAI version %u.%u.%c (license: %s) registered\n", + xlnx_logicpd_dai.name, + (unsigned int)MAJOR_REV(reg), + (unsigned int)MINOR_REV(reg), + (char)PATCH_LEVEL(reg), + LICENSE_TYPE(reg) == 0 ? "source" : + (LICENSE_TYPE(reg) == 1 ? "eval" : "release")); + + return ret; +} + +static struct platform_driver xlnx_logicpd_i2s_driver = { + .driver = { + .name = DRV_NAME, + .of_match_table = xlnx_logicpd_i2s_of_match, + }, + .probe = xlnx_logicpd_i2s_probe, +}; + +module_platform_driver(xlnx_logicpd_i2s_driver); + +MODULE_AUTHOR("Miquel Raynal <miquel.raynal(a)bootlin.com>"); +MODULE_DESCRIPTION("Xilinx logicPD I2S module"); +MODULE_LICENSE("GPL v2"); -- 2.20.1

3 8

[alsa-devel] [PATCH] ALSA: hda/realtek - Enable internal speaker & headset mic of ASUS UX431FL
by Jian-Hong Pan 03 Sep '19

03 Sep '19

Original pin node values of ASUS UX431FL with ALC294: 0x12 0xb7a60140 0x13 0x40000000 0x14 0x90170110 0x15 0x411111f0 0x16 0x411111f0 0x17 0x90170111 0x18 0x411111f0 0x19 0x411111f0 0x1a 0x411111f0 0x1b 0x411111f0 0x1d 0x4066852d 0x1e 0x411111f0 0x1f 0x411111f0 0x21 0x04211020 1. Has duplicated internal speakers (0x14 & 0x17) which makes the output route become confused. So, the output volume cannot be changed by setting. 2. Misses the headset mic pin node. This patch disables the confusing speaker (NID 0x14) and enables the headset mic (NID 0x19). Signed-off-by: Jian-Hong Pan <jian-hong(a)endlessm.com> --- sound/pci/hda/patch_realtek.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index e333b3e30e31..0a1fa99a6723 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5797,6 +5797,7 @@ enum { ALC286_FIXUP_ACER_AIO_HEADSET_MIC, ALC256_FIXUP_ASUS_MIC_NO_PRESENCE, ALC299_FIXUP_PREDATOR_SPK, + ALC294_FIXUP_ASUS_INTSPK_HEADSET_MIC, }; static const struct hda_fixup alc269_fixups[] = { @@ -6837,6 +6838,16 @@ static const struct hda_fixup alc269_fixups[] = { { } } }, + [ALC294_FIXUP_ASUS_INTSPK_HEADSET_MIC] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x14, 0x411111f0 }, /* disable confusing internal speaker */ + { 0x19, 0x04a11150 }, /* use as headset mic, without its own jack detect */ + { } + }, + .chained = true, + .chain_id = ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -6995,6 +7006,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1427, "Asus Zenbook UX31E", ALC269VB_FIXUP_ASUS_ZENBOOK), SND_PCI_QUIRK(0x1043, 0x1517, "Asus Zenbook UX31A", ALC269VB_FIXUP_ASUS_ZENBOOK_UX31A), SND_PCI_QUIRK(0x1043, 0x16e3, "ASUS UX50", ALC269_FIXUP_STEREO_DMIC), + SND_PCI_QUIRK(0x1043, 0x17d1, "ASUS UX431FL", ALC294_FIXUP_ASUS_INTSPK_HEADSET_MIC), SND_PCI_QUIRK(0x1043, 0x1a13, "Asus G73Jw", ALC269_FIXUP_ASUS_G73JW), SND_PCI_QUIRK(0x1043, 0x1a30, "ASUS X705UD", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x1b13, "Asus U41SV", ALC269_FIXUP_INV_DMIC), -- 2.20.1

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[alsa-devel] [PATCH 2/2] Fix a stack buffer overflow bug in check_input_term
by Hui Peng 02 Sep '19

02 Sep '19

`check_input_term` recursively calls itself with input from device side (e.g., uac_input_terminal_descriptor.bCSourceID) as argument (id). In `check_input_term`, if `check_input_term` is called with the same `id` argument as the caller, it triggers endless recursive call, resulting kernel space stack overflow. This patch fixes the bug by adding a bitmap to `struct mixer_build` to keep track of the checked ids and stop the execution if some id has been checked (similar to how parse_audio_unit handles unitid argument). CVE: CVE-2018-15118 Reported-by: Hui Peng <benquike(a)gmail.com> Reported-by: Mathias Payer <mathias.payer(a)nebelwelt.net> Signed-off-by: Hui Peng <benquike(a)gmail.com> --- sound/usb/mixer.c | 29 ++++++++++++++++++++++++----- 1 file changed, 24 insertions(+), 5 deletions(-) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 10ddec76f906..e24572fd6e30 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -81,6 +81,7 @@ struct mixer_build { unsigned char *buffer; unsigned int buflen; DECLARE_BITMAP(unitbitmap, MAX_ID_ELEMS); + DECLARE_BITMAP(termbitmap, MAX_ID_ELEMS); struct usb_audio_term oterm; const struct usbmix_name_map *map; const struct usbmix_selector_map *selector_map; @@ -709,15 +710,24 @@ static int get_term_name(struct mixer_build *state, struct usb_audio_term *iterm * parse the source unit recursively until it reaches to a terminal * or a branched unit. */ -static int check_input_term(struct mixer_build *state, int id, +static int __check_input_term(struct mixer_build *state, int id, struct usb_audio_term *term) { int err; void *p1; + unsigned char *hdr; memset(term, 0, sizeof(*term)); - while ((p1 = find_audio_control_unit(state, id)) != NULL) { - unsigned char *hdr = p1; + for (;;) { + /* a loop in the terminal chain? */ + if (test_and_set_bit(id, state->termbitmap)) + return -EINVAL; + + p1 = find_audio_control_unit(state, id); + if (!p1) + break; + + hdr = p1; term->id = id; switch (hdr[2]) { case UAC_INPUT_TERMINAL: @@ -732,7 +742,7 @@ static int check_input_term(struct mixer_build *state, int id, /* call recursively to verify that the * referenced clock entity is valid */ - err = check_input_term(state, d->bCSourceID, term); + err = __check_input_term(state, d->bCSourceID, term); if (err < 0) return err; @@ -764,7 +774,7 @@ static int check_input_term(struct mixer_build *state, int id, case UAC2_CLOCK_SELECTOR: { struct uac_selector_unit_descriptor *d = p1; /* call recursively to retrieve the channel info */ - err = check_input_term(state, d->baSourceID[0], term); + err = __check_input_term(state, d->baSourceID[0], term); if (err < 0) return err; term->type = d->bDescriptorSubtype << 16; /* virtual type */ @@ -811,6 +821,15 @@ static int check_input_term(struct mixer_build *state, int id, return -ENODEV; } + +static int check_input_term(struct mixer_build *state, int id, + struct usb_audio_term *term) +{ + memset(term, 0, sizeof(*term)); + memset(state->termbitmap, 0, sizeof(state->termbitmap)); + return __check_input_term(state, id, term); +} + /* * Feature Unit */ -- 2.17.1

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[alsa-devel] [PATCH 1/2] Fix an OOB bug in parse_audio_mixer_unit
by Hui Peng 02 Sep '19

02 Sep '19

The `uac_mixer_unit_descriptor` shown as below is read from the device side. In `parse_audio_mixer_unit`, `baSourceID` field is accessed from index 0 to `bNrInPins` - 1, the current implementation assumes that descriptor is always valid (the length of descriptor is no shorter than 5 + `bNrInPins`). If a descriptor read from the device side is invalid, it may trigger out-of-bound memory access. ``` struct uac_mixer_unit_descriptor { __u8 bLength; __u8 bDescriptorType; __u8 bDescriptorSubtype; __u8 bUnitID; __u8 bNrInPins; __u8 baSourceID[]; } ``` This patch fixes the bug by add a sanity check on the length of the descriptor. CVE: CVE-2018-15117 Reported-by: Hui Peng <benquike(a)gmail.com> Reported-by: Mathias Payer <mathias.payer(a)nebelwelt.net> Signed-off-by: Hui Peng <benquike(a)gmail.com> --- sound/usb/mixer.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 1f7eb3816cd7..10ddec76f906 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -1628,6 +1628,7 @@ static int parse_audio_mixer_unit(struct mixer_build *state, int unitid, int pin, ich, err; if (desc->bLength < 11 || !(input_pins = desc->bNrInPins) || + desc->bLength < sizeof(*desc) + desc->bNrInPins || !(num_outs = uac_mixer_unit_bNrChannels(desc))) { usb_audio_err(state->chip, "invalid MIXER UNIT descriptor %d\n", -- 2.17.1

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[alsa-devel] [PATCH] ASoC: fsl_sai: Set SAI Channel Mode to Output Mode
by Daniel Baluta 02 Sep '19

02 Sep '19

>From SAI datasheet: CHMOD, configures if transmit data pins are configured for TDM mode or Output mode. * (0) TDM mode, transmit data pins are tri-stated when slots are masked or channels are disabled. * (1) Output mode, transmit data pins are never tri-stated and will output zero when slots are masked or channels are disabled. When data pins are tri-stated, there is noise on some channels when FS clock value is high and data is read while fsclk is transitioning from high to low. Fix this by setting CHMOD to Output Mode so that pins will output zero when slots are masked or channels are disabled. Cc: NXP Linux Team <linux-imx(a)nxp.com> Signed-off-by: Cosmin-Gabriel Samoila <cosmin.samoila(a)nxp.com> Signed-off-by: Daniel Baluta <daniel.baluta(a)nxp.com> --- sound/soc/fsl/fsl_sai.c | 15 ++++++++++++--- sound/soc/fsl/fsl_sai.h | 2 ++ 2 files changed, 14 insertions(+), 3 deletions(-) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index e896b577b1f7..b9daab0eb6eb 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -467,6 +467,12 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, val_cr4 |= FSL_SAI_CR4_FRSZ(slots); + /* + * set CHMOD to Output Mode so that transmit data pins will + * output zero when slots are masked or channels are disabled + */ + val_cr4 |= FSL_SAI_CR4_CHMOD; + /* * For SAI master mode, when Tx(Rx) sync with Rx(Tx) clock, Rx(Tx) will * generate bclk and frame clock for Tx(Rx), we should set RCR4(TCR4), @@ -477,7 +483,8 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, if (!sai->is_slave_mode) { if (!sai->synchronous[TX] && sai->synchronous[RX] && !tx) { regmap_update_bits(sai->regmap, FSL_SAI_TCR4(ofs), - FSL_SAI_CR4_SYWD_MASK | FSL_SAI_CR4_FRSZ_MASK, + FSL_SAI_CR4_SYWD_MASK | FSL_SAI_CR4_FRSZ_MASK | + FSL_SAI_CR4_CHMOD_MASK, val_cr4); regmap_update_bits(sai->regmap, FSL_SAI_TCR5(ofs), FSL_SAI_CR5_WNW_MASK | FSL_SAI_CR5_W0W_MASK | @@ -486,7 +493,8 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, ~0UL - ((1 << channels) - 1)); } else if (!sai->synchronous[RX] && sai->synchronous[TX] && tx) { regmap_update_bits(sai->regmap, FSL_SAI_RCR4(ofs), - FSL_SAI_CR4_SYWD_MASK | FSL_SAI_CR4_FRSZ_MASK, + FSL_SAI_CR4_SYWD_MASK | FSL_SAI_CR4_FRSZ_MASK | + FSL_SAI_CR4_CHMOD_MASK, val_cr4); regmap_update_bits(sai->regmap, FSL_SAI_RCR5(ofs), FSL_SAI_CR5_WNW_MASK | FSL_SAI_CR5_W0W_MASK | @@ -497,7 +505,8 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, } regmap_update_bits(sai->regmap, FSL_SAI_xCR4(tx, ofs), - FSL_SAI_CR4_SYWD_MASK | FSL_SAI_CR4_FRSZ_MASK, + FSL_SAI_CR4_SYWD_MASK | FSL_SAI_CR4_FRSZ_MASK | + FSL_SAI_CR4_CHMOD_MASK, val_cr4); regmap_update_bits(sai->regmap, FSL_SAI_xCR5(tx, ofs), FSL_SAI_CR5_WNW_MASK | FSL_SAI_CR5_W0W_MASK | diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h index f96f8d97489d..1e3b4a6889a8 100644 --- a/sound/soc/fsl/fsl_sai.h +++ b/sound/soc/fsl/fsl_sai.h @@ -119,6 +119,8 @@ #define FSL_SAI_CR4_FRSZ_MASK (0x1f << 16) #define FSL_SAI_CR4_SYWD(x) (((x) - 1) << 8) #define FSL_SAI_CR4_SYWD_MASK (0x1f << 8) +#define FSL_SAI_CR4_CHMOD BIT(5) +#define FSL_SAI_CR4_CHMOD_MASK GENMASK(5, 5) #define FSL_SAI_CR4_MF BIT(4) #define FSL_SAI_CR4_FSE BIT(3) #define FSL_SAI_CR4_FSP BIT(1) -- 2.17.1

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[alsa-devel] [PATCH] ALSA: hda/realtek - Add quirk for HP Pavilion 15
by Sam Bazley 02 Sep '19

02 Sep '19

HP Pavilion 15 (AMD Ryzen-based model) with 103c:84e7 needs the same quirk like HP Envy/Spectre x360 for enabling the mute LED over Mic3 pin. Signed-off-by: Sam Bazley <sambazley(a)fastmail.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index e333b3e30e31..31ba8d68c16d 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6973,6 +6973,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x802e, "HP Z240 SFF", ALC221_FIXUP_HP_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x802f, "HP Z240", ALC221_FIXUP_HP_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x820d, "HP Pavilion 15", ALC269_FIXUP_HP_MUTE_LED_MIC3), + SND_PCI_QUIRK(0x103c, 0x84e7, "HP Pavilion 15", ALC269_FIXUP_HP_MUTE_LED_MIC3), SND_PCI_QUIRK(0x103c, 0x8256, "HP", ALC221_FIXUP_HP_FRONT_MIC), SND_PCI_QUIRK(0x103c, 0x827e, "HP x360", ALC295_FIXUP_HP_X360), SND_PCI_QUIRK(0x103c, 0x82bf, "HP G3 mini", ALC221_FIXUP_HP_MIC_NO_PRESENCE), -- 2.23.0

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Re: [alsa-devel] sound/usb kernel modules as a DKMS package. (Re: Alsa-devel Digest, Vol 151, Issue 10)
by Hin-Tak Leung 02 Sep '19

02 Sep '19

> Date: Mon, 2 Sep 2019 22:06:07 +0900 > From: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> > As a quick glance, the repository is just for deb-based distributions. > >From my experience, users who eager to use the latest source codes tend > to use Arch Linux and Gentoo Linux. It's helpful for the users to > include some instructions just to use the code with DKMS system. Then > you could get any feedback from them as well. Yes, that's correct. It appears that the "dkms mkrpm ..." functionality is not on Ubuntu. I am mostly a Fedora person privately (Slackware before Fedora). The Ubuntu thing is a work-requirement. Have never used Arch nor Gentoo. I'll see if I could get it done on Fedora too. The dkms packaging is mostly done for junior colleagues - I had removed all of work-related material before posting, so it is a bit bare at the moment. > For your information, I maintain out-of-tree codes to develop ALSA > firewire stack for a couple of years, as you start: > https://github.com/takaswie/snd-firewire-improve Yes, thanks. I came across that, and a few other repos which try to package up other part of bleeding-edge sound driver code (like HDA), but not sound/usb yet. Hence this posting.

1 0

[alsa-devel] [PATCH 00/16] ASoC: soc-core cleanup
by Kuninori Morimoto 02 Sep '19

02 Sep '19

Hi Mark I want to add multi CPU DAI support to ALSA SoC. But I noticed that we want to cleanup ALSA SoC before that. These patches will do nothing from "technical" point of view. Just for cleanup Kuninori Morimoto (16): ASoC: soc-core: rename soc_post_component_init() to soc_rtd_init() ASoC: soc-core: add soc_rtd_free() ASoC: soc-core: move soc_probe_component() position ASoC: soc-core: dapm related setup at one place ASoC: soc-core: add snd_soc_dapm_init() ASoC: soc-core: move soc_probe_link_components() position ASoC: soc-core: self contained soc_probe_link_components() ASoC: soc-core: self contained soc_remove_link_components() ASoC: soc-core: self contained soc_remove_link_dais() ASoC: soc-core: move soc_probe_dai() next to soc_remove_dai() ASoC: soc-core: add new soc_link_init() ASoC: soc-core: self contained soc_probe_link_dais() ASoC: soc-core: move soc_probe_link_dais() next to soc_remove_link_dais() ASoC: soc-core: self contained soc_bind_aux_dev() ASoC: soc-core: add soc_unbind_aux_dev() ASoC: soc-core: self contained soc_unbind_aux_dev() include/sound/soc-dapm.h | 3 + sound/soc/soc-core.c | 568 ++++++++++++++++++++++++----------------------- sound/soc/soc-dapm.c | 21 ++ 3 files changed, 310 insertions(+), 282 deletions(-) -- 2.7.4

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[alsa-devel] [PATCH v2] drm: dw-hdmi-i2s: enable audio clock in audio_startup
by Cheng-Yi Chiang 02 Sep '19

02 Sep '19

In the designware databook, the sequence of enabling audio clock and setting format is not clearly specified. Currently, audio clock is enabled in the end of hw_param ops after setting format. On some monitors, there is a possibility that audio does not come out. Fix this by enabling audio clock in audio_startup ops before hw_param ops setting format. Signed-off-by: Cheng-Yi Chiang <cychiang(a)chromium.org> Reviewed-by: Douglas Anderson <dianders(a)chromium.org> Reviewed-by: Jonas Karlman <jonas(a)kwiboo.se> Tested-by: Douglas Anderson <dianders(a)chromium.org> --- Changes from v1: 1. Move audio_startup to the front of audio_shutdown. 2. Fix the indentation of audio_startup equal sign using tab. 3. Rebase the patch on linux-next/master. 4. Add Reviewed-by and Tested-by fields from dianders(a)chromium.org. 5. Add Reviewed-by field from jonas(a)kwiboo.se. drivers/gpu/drm/bridge/synopsys/dw-hdmi-i2s-audio.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/drivers/gpu/drm/bridge/synopsys/dw-hdmi-i2s-audio.c b/drivers/gpu/drm/bridge/synopsys/dw-hdmi-i2s-audio.c index 1d15cf9b6821..34d8e837555f 100644 --- a/drivers/gpu/drm/bridge/synopsys/dw-hdmi-i2s-audio.c +++ b/drivers/gpu/drm/bridge/synopsys/dw-hdmi-i2s-audio.c @@ -109,6 +109,14 @@ static int dw_hdmi_i2s_hw_params(struct device *dev, void *data, hdmi_write(audio, conf0, HDMI_AUD_CONF0); hdmi_write(audio, conf1, HDMI_AUD_CONF1); + return 0; +} + +static int dw_hdmi_i2s_audio_startup(struct device *dev, void *data) +{ + struct dw_hdmi_i2s_audio_data *audio = data; + struct dw_hdmi *hdmi = audio->hdmi; + dw_hdmi_audio_enable(hdmi); return 0; @@ -153,6 +161,7 @@ static int dw_hdmi_i2s_get_dai_id(struct snd_soc_component *component, static struct hdmi_codec_ops dw_hdmi_i2s_ops = { .hw_params = dw_hdmi_i2s_hw_params, + .audio_startup = dw_hdmi_i2s_audio_startup, .audio_shutdown = dw_hdmi_i2s_audio_shutdown, .get_eld = dw_hdmi_i2s_get_eld, .get_dai_id = dw_hdmi_i2s_get_dai_id, -- 2.23.0.187.g17f5b7556c-goog

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[alsa-devel] [PATCH] ASoC: sun4i-i2s: incorrect regmap for A83t
by codekipper＠gmail.com 02 Sep '19

02 Sep '19

From: Marcus Cooper <codekipper(a)gmail.com> Fixes: 21faaea1343f ("ASoC: sun4i-i2s: Add support for A83T") Signed-off-by: Marcus Cooper <codekipper(a)gmail.com> --- sound/soc/sunxi/sun4i-i2s.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sunxi/sun4i-i2s.c b/sound/soc/sunxi/sun4i-i2s.c index cbc6e59aa089..056a299c03fb 100644 --- a/sound/soc/sunxi/sun4i-i2s.c +++ b/sound/soc/sunxi/sun4i-i2s.c @@ -1109,7 +1109,7 @@ static const struct sun4i_i2s_quirks sun6i_a31_i2s_quirks = { static const struct sun4i_i2s_quirks sun8i_a83t_i2s_quirks = { .has_reset = true, .reg_offset_txdata = SUN8I_I2S_FIFO_TX_REG, - .sun4i_i2s_regmap = &sun4i_i2s_regmap_config, + .sun4i_i2s_regmap = &sun8i_i2s_regmap_config, .field_clkdiv_mclk_en = REG_FIELD(SUN4I_I2S_CLK_DIV_REG, 8, 8), .field_fmt_wss = REG_FIELD(SUN4I_I2S_FMT0_REG, 0, 2), .field_fmt_sr = REG_FIELD(SUN4I_I2S_FMT0_REG, 4, 6), -- 2.23.0

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[alsa-devel] [PATCH] MAINTAINERS: update entry of firewire audio drivers
by Takashi Sakamoto 02 Sep '19

02 Sep '19

This commit adds myself as one of maintainers for firewire audio drivers and IEC 61883-1/6 packet streaming engine. I call them ALSA firewire stack as a whole. 6 years ago I joined in development for this category of drivers with heavy reverse-engineering tasks and over 100 models are now available from ALSA applications. IEEE 1394 bus itself and units on the bus are enough legacy but the development still continues. I have a plan to add drastic enhancement in kernel v5.5 and v5.6 period. This commit adds myself into MAINTAINERS so that developers and users can easily find active developer to post their issues, especially for regression. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- MAINTAINERS | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/MAINTAINERS b/MAINTAINERS index 24e29b2e53c9..8929a2ec75f7 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -6264,8 +6264,9 @@ S: Maintained F: drivers/hwmon/f75375s.c F: include/linux/f75375s.h -FIREWIRE AUDIO DRIVERS +FIREWIRE AUDIO DRIVERS and IEC 61883-1/6 PACKET STREAMING ENGINE M: Clemens Ladisch <clemens(a)ladisch.de> ++M: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) T: git git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git S: Maintained -- 2.20.1

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[alsa-devel] [PATCH] alsa-utils: Remove usleep usage
by Rosen Penev 02 Sep '19

02 Sep '19

usleep was deprecated in POSIX 2008 and is not available without deprecated APIs enabled in uClibc-ng. Signed-off-by: Rosen Penev <rosenp(a)gmail.com> --- alsactl/daemon.c | 3 ++- alsaloop/pcmjob.c | 3 ++- amidi/amidi.c | 6 ++++-- bat/bat.c | 5 +++-- 4 files changed, 11 insertions(+), 6 deletions(-) diff --git a/alsactl/daemon.c b/alsactl/daemon.c index 1d98caa..3b84cf6 100644 --- a/alsactl/daemon.c +++ b/alsactl/daemon.c @@ -348,7 +348,8 @@ static int check_another_instance(const char *pidfile) if (pid >= 0) { /* invoke new card rescan */ if (kill(pid, SIGUSR1) >= 0) { - usleep(1000); + struct timespec req = { 0, 1 * 1000 * 1000 }; + nanosleep(&req, NULL); pid = read_pid_file(pidfile); if (pid >= 0) return 1; diff --git a/alsaloop/pcmjob.c b/alsaloop/pcmjob.c index 29d1aba..d0347dc 100644 --- a/alsaloop/pcmjob.c +++ b/alsaloop/pcmjob.c @@ -662,10 +662,11 @@ static int xrun(struct loopback_handle *lhandle) static int suspend(struct loopback_handle *lhandle) { + struct timespec req = { 0, 1000}; int err; while ((err = snd_pcm_resume(lhandle->handle)) == -EAGAIN) - usleep(1); + nanosleep(&req, NULL); if (err < 0) return xrun(lhandle); return 0; diff --git a/amidi/amidi.c b/amidi/amidi.c index b6e05e1..3b2953d 100644 --- a/amidi/amidi.c +++ b/amidi/amidi.c @@ -254,10 +254,12 @@ static int send_midi_interleaved(void) snd_rawmidi_status(output, st); do { /* 320 µs per byte as noted in Page 1 of MIDI spec */ - usleep((buffer_size - snd_rawmidi_status_get_avail(st)) * 320); + struct timespec req = { 0, (buffer_size - snd_rawmidi_status_get_avail(st)) * 320 * 1000}; + nanosleep(&req, NULL); snd_rawmidi_status(output, st); } while(snd_rawmidi_status_get_avail(st) < buffer_size); - usleep(sysex_interval * 1000); + struct timespec sysex = { 0, sysex_interval * 1000 * 1000}; + nanosleep(&sysex, NULL); } /* find end of SysEx */ diff --git a/bat/bat.c b/bat/bat.c index 8645770..27e7a34 100644 --- a/bat/bat.c +++ b/bat/bat.c @@ -176,6 +176,7 @@ static void test_loopback(struct bat *bat) pthread_t capture_id, playback_id; int err; int *thread_result_capture, *thread_result_playback; + struct timespec capture_delay = { 0, CAPTURE_DELAY * 1000 * 1000 }; /* start playback */ err = pthread_create(&playback_id, NULL, @@ -188,7 +189,7 @@ static void test_loopback(struct bat *bat) /* TODO: use a pipe to signal stream start etc - i.e. to sync threads */ /* Let some time for playing something before capturing */ - usleep(CAPTURE_DELAY * 1000); + nanosleep(&capture_delay, NULL); /* start capture */ err = pthread_create(&capture_id, NULL, (void *) bat->capture.fct, bat); @@ -691,7 +692,7 @@ int main(int argc, char *argv[]) } /* Waiting 500ms and start the next round */ - usleep(CAPTURE_DELAY * 1000); + nanosleep(&capture_delay, NULL); } goto out; } -- 2.17.1

2 1

Re: [alsa-devel] [PATCH 2/2] ASoC: rt1011: ADCDAT pin config modification
by Shuming [范書銘] 02 Sep '19

02 Sep '19

> > > So really this is setting up a loopback for testing? > > > We would not like to change ADCDAT pin to input mode in normal case. > > That's why the driver enables ADCDAT pin to the output mode in default. > > The rt1011 supports the feedback signal which could be playback data or I/V > data, etc. > > If the system wants the AEC reference data, rt1011 could feedback the > playback data. > > > The product could connect 2/4/6/8 rt1011 chips on the same I2S bus. > > In a test or debug mode, we could toggle ADCDAT pin to input mode that > > also helps HW engineer check the slot of feedback signal for each rt1011. > > I think this needs more than just a straight userspace control on one device, > these use cases make sense but they'll need to be configured over multiple > chips simultaneously otherwise there's some possibility of hardware damage > (eg, if two chips try to drive the signal at the same time). If this really can be > usefully varied at runtime then the driver bit of this should probably be an API > that the machine driver can call, the machine driver can then expose a control > that sets all the chips involved up together. I understand your concerns and comments. In a design-in project, we will provide the proper control settings to arrange the slot location of ADCDAT signal for each rt1011. HW engineer will confirm the ADCDAT signal, too. So, the customer should not make two chips drive the signal at the same slot and at the same time. But, there is a chance to happen if the customer sets the wrong control settings. We will take your suggestion to make it as API call. The machine driver knows how many rt1011 chips connect and make pin config change together. Thanks. > ------Please consider the environment before printing this e-mail.

1 0

[alsa-devel] bug
by liebrecht＠grossmann-venter.com 02 Sep '19

02 Sep '19

Clearly a problem with Alsa. Alsa reports "Most likely this is a bug in the ALSA driver 'snd_usb_audio'." Alsa used to work great. howevere recently it takes 1/2-1hour for pulseaudio to convince alsa to make the 1818VSL available. Jack is not involved at all. Pulseaudio connects straight to alsa. Pulseaudio is my saving grace roight now as it perpetually tries to initialize and get alsa to present the audio interface, which it eventually after an hour does "waking up" Also sorts of decides to "wake up" and then suddenly the 1818vsl is found. How do I "wake up" alsa so that I can use soundcards. I cannot wait an hour after reboot or logout to get alsa to work again. See below What happens after starting pulseaudio is that it cycles the 1818 for about 1/2 hour to get a link to the 181vsl soundcard. That is the first paragraph (1) below which is perpetually cycled by pulseaudio. Eventually it succeeds which gives the BUG comment as below in paragraph (2) ------------------------------------------------------- 1) E: [pulseaudio] udev-util.c: Failed to get card object. E: [pulseaudio] module-alsa-card.c: Failed to find a working profile. E: [pulseaudio] module.c: Failed to load module "module-alsa-card" (argument: "device_id="3" name="usb-PreSonus_AudioBox_1818_VSL_2209-00" card_name="alsa_card.usb-PreSonus_Audio.Box_1818_VSL_2209-00" namereg_fail=false tsched=yes fixed_latency_range=no ignore_dB=no deferred_volume=yes use_ucm=yes card_properties="module-udev-detect.discovered=1""): initialization failed. (**Note this message is cycled for an hour**) 2) (**Then miraculously....Alsa wakes up and pulseaudio can connect to 1818vsl **) E: [alsa-sink-USB Audio] alsa-sink.c: ALSA woke us up to write new data to the device, but there was actually nothing to write. E: [alsa-sink-USB Audio] alsa-sink.c: Most likely this is a bug in the ALSA driver 'snd_usb_audio'. Please report this issue to the ALSA developers. E: [alsa-sink-USB Audio] alsa-sink.c: We were woken up with POLLOUT set -- however a subsequent snd_pcm_avail() returned 0 or another value < min_avail. ------------------------------------------------------ Dont know what happened top Also to behave this way. Maybe someone has an idea ? upload=true&script=true&cardinfo= !!################################ !!ALSA Information Script v 0.4.64 !!################################ !!Script ran on: Mon Sep 2 03:27:26 UTC 2019 !!Linux Distribution !!------------------ [1;36mWelcome to [1;33mMX Linux[1;36m! Powered by [1;33mDebian[1;36m.[0m PRETTY_NAME="Debian GNU/Linux 9 (stretch)" NAME="Debian GNU/Linux" ID=debian HOME_URL="https://www.debian.org/" SUPPORT_URL="https://www.debian.org/support" BUG_REPORT_URL="https://bugs.debian.org/" MX-18.3_x64 Continuum Dec 20, 2018 MX-18.3_x64 Continuum Dec 20, 2018 !!DMI Information !!--------------- Manufacturer: Supermicro Product Name: H8DGU Product Version: 1234567890 Firmware Version: 3.5c Board Vendor: Supermicro Board Name: H8DGU !!ACPI Device Status Information !!--------------- /sys/bus/acpi/devices/PNP0103:00/status 15 /sys/bus/acpi/devices/PNP0501:00/status 15 /sys/bus/acpi/devices/PNP0501:01/status 15 /sys/bus/acpi/devices/PNP0C0C:00/status 11 /sys/bus/acpi/devices/PNP0C0F:00/status 9 /sys/bus/acpi/devices/PNP0C0F:01/status 9 /sys/bus/acpi/devices/PNP0C0F:02/status 9 /sys/bus/acpi/devices/PNP0C0F:03/status 9 /sys/bus/acpi/devices/PNP0C0F:04/status 9 /sys/bus/acpi/devices/PNP0C0F:05/status 9 /sys/bus/acpi/devices/PNP0C0F:06/status 9 /sys/bus/acpi/devices/PNP0C0F:07/status 9 /sys/bus/acpi/devices/device:1e/status 15 /sys/bus/acpi/devices/device:1f/status 15 /sys/bus/acpi/devices/device:21/status 15 /sys/bus/acpi/devices/device:22/status 15 !!Kernel Information !!------------------ Kernel release: 4.19.0-5-amd64 Operating System: GNU/Linux Architecture: x86_64 Processor: unknown SMP Enabled: Yes !!ALSA Version !!------------ Driver version: k4.19.0-5-amd64 Library version: 1.1.9 Utilities version: 1.1.3 !!Loaded ALSA modules !!------------------- snd_hda_intel snd_usb_audio snd_usb_audio !!Sound Servers on this system !!---------------------------- Pulseaudio: Installed - Yes (/usr/bin/pulseaudio) Running - Yes Jack: Installed - Yes (/usr/bin/jackd) Running - No !!Soundcards recognised by ALSA !!----------------------------- 0 [HDMI ]: HDA-Intel - HDA ATI HDMI HDA ATI HDMI at 0xfebfc000 irq 46 1 [Interface ]: USB-Audio - USB Uno MIDI Interface M-Audio USB Uno MIDI Interface at usb-0000:00:12.2-1.2.3, full speed 2 [M1x1 ]: USB-Audio - MidiSport 1x1 M-Audio MidiSport 1x1 at usb-0000:00:12.2-1.2.2, full speed !!PCI Soundcards installed in the system !!-------------------------------------- 03:00.1 Audio device: Advanced Micro Devices, Inc. [AMD/ATI] Device aae0 !!Advanced information - PCI Vendor/Device/Subsystem ID's !!------------------------------------------------------- 03:00.1 0403: 1002:aae0 Subsystem: 1462:aae0 !!Modprobe options (Sound related) !!-------------------------------- snd_pcsp: index=-2 snd_usb_audio: index=-2 snd_atiixp_modem: index=-2 snd_intel8x0m: index=-2 snd_via82xx_modem: index=-2 !!Loaded sound module options !!--------------------------- !!Module: snd_hda_intel align_buffer_size : -1 bdl_pos_adj : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 beep_mode : Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y enable : Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y enable_msi : -1 id : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) index : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 jackpoll_ms : 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0 model : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) patch : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) pm_blacklist : Y position_fix : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 power_save : 0 power_save_controller : N probe_mask : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 probe_only : 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0 single_cmd : -1 snoop : -1 !!Module: snd_usb_audio autoclock : Y device_setup : 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0 enable : Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y id : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) ignore_ctl_error : N index : -2,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 pid : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 quirk_alias : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) use_vmalloc : Y vid : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 !!Module: snd_usb_audio autoclock : Y device_setup : 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0 enable : Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y id : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) ignore_ctl_error : N index : -2,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 pid : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 quirk_alias : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) use_vmalloc : Y vid : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 !!HDA-Intel Codec information !!--------------------------- --startcollapse-- Codec: ATI R6xx HDMI Address: 0 AFG Function Id: 0x1 (unsol 0) Vendor Id: 0x1002aa01 Subsystem Id: 0x00aa0100 Revision Id: 0x100700 No Modem Function Group found Default PCM: rates [0x70]: 32000 44100 48000 bits [0x2]: 16 formats [0x1]: PCM Default Amp-In caps: N/A Default Amp-Out caps: N/A State of AFG node 0x01: Power states: D0 D3 CLKSTOP EPSS Power: setting=D0, actual=D0, Clock-stop-OK GPIO: io=0, o=0, i=0, unsolicited=0, wake=0 Node 0x02 [Audio Output] wcaps 0x221: Stereo Digital Stripe Converter: stream=0, channel=0 Digital: Enabled Digital category: 0x0 IEC Coding Type: 0x0 Node 0x03 [Pin Complex] wcaps 0x400381: Stereo Digital Control: name="IEC958 Playback Con Mask", index=0, device=0 Control: name="IEC958 Playback Pro Mask", index=0, device=0 Control: name="IEC958 Playback Default", index=0, device=0 Control: name="IEC958 Playback Switch", index=0, device=0 Pincap 0x00000094: OUT Detect HDMI Pin Default 0x185600f0: [Jack] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Pin-ctls: 0x40: OUT Unsolicited: tag=01, enabled=1 Connection: 1 0x02 Node 0x04 [Audio Output] wcaps 0x221: Stereo Digital Stripe Converter: stream=0, channel=0 Digital: Enabled Digital category: 0x0 IEC Coding Type: 0x0 Node 0x05 [Pin Complex] wcaps 0x400381: Stereo Digital Control: name="IEC958 Playback Con Mask", index=1, device=0 Control: name="IEC958 Playback Pro Mask", index=1, device=0 Control: name="IEC958 Playback Default", index=1, device=0 Control: name="IEC958 Playback Switch", index=1, device=0 Pincap 0x00000094: OUT Detect HDMI Pin Default 0x185600f0: [Jack] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Pin-ctls: 0x40: OUT Unsolicited: tag=02, enabled=1 Connection: 1 0x04 Node 0x06 [Audio Output] wcaps 0x221: Stereo Digital Stripe Converter: stream=0, channel=0 Digital: Enabled Digital category: 0x0 IEC Coding Type: 0x0 Node 0x07 [Pin Complex] wcaps 0x400381: Stereo Digital Control: name="IEC958 Playback Con Mask", index=2, device=0 Control: name="IEC958 Playback Pro Mask", index=2, device=0 Control: name="IEC958 Playback Default", index=2, device=0 Control: name="IEC958 Playback Switch", index=2, device=0 Pincap 0x00000094: OUT Detect HDMI Pin Default 0x185600f0: [Jack] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Pin-ctls: 0x40: OUT Unsolicited: tag=03, enabled=1 Connection: 1 0x06 Node 0x08 [Audio Output] wcaps 0x221: Stereo Digital Stripe Converter: stream=0, channel=0 Digital: Enabled Digital category: 0x0 IEC Coding Type: 0x0 Node 0x09 [Pin Complex] wcaps 0x400381: Stereo Digital Control: name="IEC958 Playback Con Mask", index=3, device=0 Control: name="IEC958 Playback Pro Mask", index=3, device=0 Control: name="IEC958 Playback Default", index=3, device=0 Control: name="IEC958 Playback Switch", index=3, device=0 Pincap 0x00000094: OUT Detect HDMI Pin Default 0x185600f0: [Jack] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Pin-ctls: 0x40: OUT Unsolicited: tag=04, enabled=1 Connection: 1 0x08 Node 0x0a [Audio Output] wcaps 0x221: Stereo Digital Stripe Converter: stream=1, channel=0 Digital: Enabled Digital category: 0x0 IEC Coding Type: 0x0 Node 0x0b [Pin Complex] wcaps 0x400381: Stereo Digital Control: name="IEC958 Playback Con Mask", index=4, device=0 Control: name="IEC958 Playback Pro Mask", index=4, device=0 Control: name="IEC958 Playback Default", index=4, device=0 Control: name="IEC958 Playback Switch", index=4, device=0 Pincap 0x00000094: OUT Detect HDMI Pin Default 0x185600f0: [Jack] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Pin-ctls: 0x40: OUT Unsolicited: tag=05, enabled=1 Connection: 1 0x0a Node 0x0c [Audio Output] wcaps 0x221: Stereo Digital Stripe Converter: stream=0, channel=0 Digital: Digital category: 0x0 IEC Coding Type: 0x0 Node 0x0d [Pin Complex] wcaps 0x400381: Stereo Digital Pincap 0x00000094: OUT Detect HDMI Pin Default 0x585600f0: [N/A] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Connection: 1 0x0c --endcollapse-- !!USB Mixer information !!--------------------- --startcollapse-- USB Mixer: usb_id=0x07630150, ctrlif=0, ctlerr=0 Card: M-Audio USB Uno MIDI Interface at usb-0000:00:12.2-1.2.3, full speed --endcollapse-- !!ALSA Device nodes !!----------------- crw-rw---- 1 root audio 116, 8 Sep 1 04:44 /dev/snd/controlC0 crw-rw---- 1 root audio 116, 10 Sep 1 06:36 /dev/snd/controlC1 crw-rw---- 1 root audio 116, 12 Sep 1 06:36 /dev/snd/controlC2 crw-rw---- 1 root audio 116, 7 Sep 1 04:44 /dev/snd/hwC0D0 crw-rw---- 1 root audio 116, 9 Sep 1 06:36 /dev/snd/midiC1D0 crw-rw---- 1 root audio 116, 11 Sep 1 06:36 /dev/snd/midiC2D0 crw-rw---- 1 root audio 116, 6 Sep 1 23:17 /dev/snd/pcmC0D10p crw-rw---- 1 root audio 116, 2 Sep 1 23:17 /dev/snd/pcmC0D3p crw-rw---- 1 root audio 116, 3 Sep 1 23:17 /dev/snd/pcmC0D7p crw-rw---- 1 root audio 116, 4 Sep 1 23:17 /dev/snd/pcmC0D8p crw-rw---- 1 root audio 116, 5 Sep 1 23:17 /dev/snd/pcmC0D9p crw-rw---- 1 root audio 116, 1 Sep 1 05:13 /dev/snd/seq crw-rw---- 1 root audio 116, 33 Sep 1 04:44 /dev/snd/timer /dev/snd/by-id: total 0 drwxr-xr-x 2 root root 80 Sep 1 22:40 . drwxr-xr-x 4 root root 340 Sep 1 22:40 .. lrwxrwxrwx 1 root root 12 Sep 1 06:36 usb-0763_1011-00 -> ../controlC2 lrwxrwxrwx 1 root root 12 Sep 1 06:36 usb-M-Audio_USB_Uno_MIDI_Interface-00 -> ../controlC1 /dev/snd/by-path: total 0 drwxr-xr-x 2 root root 100 Sep 1 22:40 . drwxr-xr-x 4 root root 340 Sep 1 22:40 .. lrwxrwxrwx 1 root root 12 Sep 1 06:36 pci-0000:00:12.2-usb-0:1.2.2:1.0 -> ../controlC2 lrwxrwxrwx 1 root root 12 Sep 1 06:36 pci-0000:00:12.2-usb-0:1.2.3:1.0 -> ../controlC1 lrwxrwxrwx 1 root root 12 Sep 1 04:44 pci-0000:03:00.1 -> ../controlC0 !!ALSA configuration files !!------------------------ !!System wide config file (/etc/asound.conf) # v4.0 # Requires alsa-utils libasound2-plugin-equal # ***** Defaults ******************************************************* # Audio card/chip and device to use and control # In most cases these settings should remain commented (#) # When commented, card and device 0 are usually automatically selected # When uncommented, the item corresponding to the number is selected #defaults.pcm.card 0 #defaults.pcm.device 0 #defaults.ctl.card 0 # ***** Reset ********************************************************** # override the existing definition of the default device pcm.!default { type plug # point output to the equalizer device slave.pcm plugequal } # ***** Equalizer ****************************************************** # set up the control interface of the equalizer device ctl.equalizer { type equal } # set up the equalizer device pcm.plugequal { type equal # point output to device named preamp slave.pcm "plug:preamp" } # ***** Pre-Amp ******************************************************** # set up the preamp device pcm.preamp { type softvol # name of slider control to display in alsamixer interface control.name Pre-Amp # minimum dB when slider is at 0% min_dB -5.0 # maximum dB when slider is at 100% max_dB 40.0 # point output to device named duplex slave.pcm "duplex" } # ***** Playback and Capture ******************************************* # set up the playback/capture device pcm.duplex { type asym # point playback output to use dmix playback.pcm "dmix" # point capture input record to use dsnoop capture.pcm "dsnoop" } !!Aplay/Arecord output !!-------------------- APLAY **** List of PLAYBACK Hardware Devices **** card 0: HDMI [HDA ATI HDMI], device 3: HDMI 0 [HDMI 0] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: HDMI [HDA ATI HDMI], device 7: HDMI 1 [HDMI 1] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: HDMI [HDA ATI HDMI], device 8: HDMI 2 [HDMI 2] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: HDMI [HDA ATI HDMI], device 9: HDMI 3 [HDMI 3] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: HDMI [HDA ATI HDMI], device 10: HDMI 4 [HDMI 4] Subdevices: 1/1 Subdevice #0: subdevice #0 ARECORD **** List of CAPTURE Hardware Devices **** !!Amixer output !!------------- !!-------Mixer controls for card 0 [HDMI] Card hw:0 'HDMI'/'HDA ATI HDMI at 0xfebfc000 irq 46' Mixer name : 'ATI R6xx HDMI' Components : 'HDA:1002aa01,00aa0100,00100700' Controls : 35 Simple ctrls : 5 Simple mixer control 'IEC958',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [on] Simple mixer control 'IEC958',1 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [on] Simple mixer control 'IEC958',2 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [on] Simple mixer control 'IEC958',3 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [on] Simple mixer control 'IEC958',4 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [on] !!-------Mixer controls for card 1 [Interface] Card hw:1 'Interface'/'M-Audio USB Uno MIDI Interface at usb-0000:00:12.2-1.2.3, full speed' Mixer name : 'USB Mixer' Components : 'USB0763:0150' Controls : 1 Simple ctrls : 0 !!-------Mixer controls for card 2 [M1x1] Card hw:2 'M1x1'/'M-Audio MidiSport 1x1 at usb-0000:00:12.2-1.2.2, full speed' Mixer name : '' Components : 'USB0763:1011' Controls : 0 Simple ctrls : 0 !!Alsactl output !!-------------- --startcollapse-- state.HDMI { control.1 { iface CARD name 'HDMI/DP,pcm=3 Jack' value false comment { access read type BOOLEAN count 1 } } control.2 { iface MIXER name 'IEC958 Playback Con Mask' value '0fff000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access read type IEC958 count 1 } } control.3 { iface MIXER name 'IEC958 Playback Pro Mask' value '0f00000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access read type IEC958 count 1 } } control.4 { iface MIXER name 'IEC958 Playback Default' value '0400000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access 'read write' type IEC958 count 1 } } control.5 { iface MIXER name 'IEC958 Playback Switch' value true comment { access 'read write' type BOOLEAN count 1 } } control.6 { iface PCM device 3 name ELD value '' comment { access 'read volatile' type BYTES count 0 } } control.7 { iface CARD name 'HDMI/DP,pcm=7 Jack' value false comment { access read type BOOLEAN count 1 } } control.8 { iface MIXER name 'IEC958 Playback Con Mask' index 1 value '0fff000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access read type IEC958 count 1 } } control.9 { iface MIXER name 'IEC958 Playback Pro Mask' index 1 value '0f00000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access read type IEC958 count 1 } } control.10 { iface MIXER name 'IEC958 Playback Default' index 1 value '0400000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access 'read write' type IEC958 count 1 } } control.11 { iface MIXER name 'IEC958 Playback Switch' index 1 value true comment { access 'read write' type BOOLEAN count 1 } } control.12 { iface PCM device 7 name ELD value '' comment { access 'read volatile' type BYTES count 0 } } control.13 { iface CARD name 'HDMI/DP,pcm=8 Jack' value false comment { access read type BOOLEAN count 1 } } control.14 { iface MIXER name 'IEC958 Playback Con Mask' index 2 value '0fff000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access read type IEC958 count 1 } } control.15 { iface MIXER name 'IEC958 Playback Pro Mask' index 2 value '0f00000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access read type IEC958 count 1 } } control.16 { iface MIXER name 'IEC958 Playback Default' index 2 value '0400000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access 'read write' type IEC958 count 1 } } control.17 { iface MIXER name 'IEC958 Playback Switch' index 2 value true comment { access 'read write' type BOOLEAN count 1 } } control.18 { iface PCM device 8 name ELD value '' comment { access 'read volatile' type BYTES count 0 } } control.19 { iface CARD name 'HDMI/DP,pcm=9 Jack' value false comment { access read type BOOLEAN count 1 } } control.20 { iface MIXER name 'IEC958 Playback Con Mask' index 3 value '0fff000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access read type IEC958 count 1 } } control.21 { iface MIXER name 'IEC958 Playback Pro Mask' index 3 value '0f00000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access read type IEC958 count 1 } } control.22 { iface MIXER name 'IEC958 Playback Default' index 3 value '0400000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access 'read write' type IEC958 count 1 } } control.23 { iface MIXER name 'IEC958 Playback Switch' index 3 value true comment { access 'read write' type BOOLEAN count 1 } } control.24 { iface PCM device 9 name ELD value '' comment { access 'read volatile' type BYTES count 0 } } control.25 { iface CARD name 'HDMI/DP,pcm=10 Jack' value false comment { access read type BOOLEAN count 1 } } control.26 { iface MIXER name 'IEC958 Playback Con Mask' index 4 value '0fff000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access read type IEC958 count 1 } } control.27 { iface MIXER name 'IEC958 Playback Pro Mask' index 4 value '0f00000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access read type IEC958 count 1 } } control.28 { iface MIXER name 'IEC958 Playback Default' index 4 value '0400000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access 'read write' type IEC958 count 1 } } control.29 { iface MIXER name 'IEC958 Playback Switch' index 4 value true comment { access 'read write' type BOOLEAN count 1 } } control.30 { iface PCM device 10 name ELD value '' comment { access 'read volatile' type BYTES count 0 } } control.31 { iface PCM device 3 name 'Playback Channel Map' value.0 0 value.1 0 value.2 0 value.3 0 value.4 0 value.5 0 value.6 0 value.7 0 comment { access 'read write' type INTEGER count 8 range '0 - 36' } } control.32 { iface PCM device 7 name 'Playback Channel Map' value.0 0 value.1 0 value.2 0 value.3 0 value.4 0 value.5 0 value.6 0 value.7 0 comment { access 'read write' type INTEGER count 8 range '0 - 36' } } control.33 { iface PCM device 8 name 'Playback Channel Map' value.0 0 value.1 0 value.2 0 value.3 0 value.4 0 value.5 0 value.6 0 value.7 0 comment { access 'read write' type INTEGER count 8 range '0 - 36' } } control.34 { iface PCM device 9 name 'Playback Channel Map' value.0 0 value.1 0 value.2 0 value.3 0 value.4 0 value.5 0 value.6 0 value.7 0 comment { access 'read write' type INTEGER count 8 range '0 - 36' } } control.35 { iface PCM device 10 name 'Playback Channel Map' value.0 0 value.1 0 value.2 0 value.3 0 value.4 0 value.5 0 value.6 0 value.7 0 comment { access 'read write' type INTEGER count 8 range '0 - 36' } } } state.Interface { control.1 { iface CARD name 'Keep Interface' value false comment { access 'read write' type BOOLEAN count 1 } } } state.M1x1 { control { } } --endcollapse-- !!All Loaded Modules !!------------------ Module snd_seq_dummy snd_seq_midi snd_seq_midi_event snd_seq bnep rpcsec_gss_krb5 nfsv4 dns_resolver nfs fscache cpufreq_userspace cpufreq_conservative cpufreq_powersave bluetooth pci_stub vboxpci drbg ansi_cprng vboxnetadp ecdh_generic rfkill vboxnetflt vboxdrv ftdi_sio usbserial binfmt_misc snd_usb_audio snd_usbmidi_lib snd_rawmidi snd_seq_device joydev snd_hda_codec_hdmi snd_hda_intel snd_hda_codec amd64_edac_mod edac_mce_amd amdgpu snd_hda_core snd_hwdep snd_pcm_oss kvm_amd ccp mgag200 rng_core snd_mixer_oss kvm irqbypass chash snd_pcm gpu_sched ipmi_si crct10dif_pclmul evdev crc32_pclmul serio_raw ghash_clmulni_intel ipmi_devintf snd_timer ttm ipmi_msghandler drm_kms_helper k10temp sg snd fam15h_power sp5100_tco drm soundcore button nfsd auth_rpcgss nfs_acl lockd grace sunrpc loop vhba ecryptfs parport_pc ppdev lp parport ip_tables x_tables autofs4 sr_mod cdrom ext4 crc16 mbcache jbd2 fscrypto ecb uas usb_storage hid_generic usbhid hid btrfs xor zstd_decompress zstd_compress xxhash sd_mod raid6_pq libcrc32c crc32c_generic vmd crc32c_intel ohci_pci ata_generic aesni_intel aes_x86_64 ahci crypto_simd ehci_pci ohci_hcd pata_atiixp libahci cryptd glue_helper ehci_hcd psmouse libata igb usbcore i2c_piix4 i2c_algo_bit scsi_mod usb_common dca !!Sysfs Files !!----------- /sys/class/sound/hwC0D0/init_pin_configs: 0x03 0x185600f0 0x05 0x185600f0 0x07 0x185600f0 0x09 0x185600f0 0x0b 0x185600f0 0x0d 0x585600f0 /sys/class/sound/hwC0D0/driver_pin_configs: /sys/class/sound/hwC0D0/user_pin_configs: /sys/class/sound/hwC0D0/init_verbs: /sys/class/sound/hwC0D0/hints: !!ALSA/HDA dmesg !!--------------

1 0

[alsa-devel] [PATCH] drm: dw-hdmi-i2s: enable audio clock in audio_startup
by Cheng-Yi Chiang 02 Sep '19

02 Sep '19

In the designware databook, the sequence of enabling audio clock and setting format is not clearly specified. Currently, audio clock is enabled in the end of hw_param ops after setting format. On some monitors, there is a possibility that audio does not come out. Fix this by enabling audio clock in audio_startup ops before hw_param ops setting format. Signed-off-by: Cheng-Yi Chiang <cychiang(a)chromium.org> --- drivers/gpu/drm/bridge/synopsys/dw-hdmi-i2s-audio.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/drivers/gpu/drm/bridge/synopsys/dw-hdmi-i2s-audio.c b/drivers/gpu/drm/bridge/synopsys/dw-hdmi-i2s-audio.c index 5cbb71a866d5..08b4adbb1ddc 100644 --- a/drivers/gpu/drm/bridge/synopsys/dw-hdmi-i2s-audio.c +++ b/drivers/gpu/drm/bridge/synopsys/dw-hdmi-i2s-audio.c @@ -69,6 +69,14 @@ static int dw_hdmi_i2s_hw_params(struct device *dev, void *data, hdmi_write(audio, conf0, HDMI_AUD_CONF0); hdmi_write(audio, conf1, HDMI_AUD_CONF1); + return 0; +} + +static int dw_hdmi_i2s_audio_startup(struct device *dev, void *data) +{ + struct dw_hdmi_i2s_audio_data *audio = data; + struct dw_hdmi *hdmi = audio->hdmi; + dw_hdmi_audio_enable(hdmi); return 0; @@ -105,6 +113,7 @@ static int dw_hdmi_i2s_get_dai_id(struct snd_soc_component *component, } static struct hdmi_codec_ops dw_hdmi_i2s_ops = { + .audio_startup = dw_hdmi_i2s_audio_startup, .hw_params = dw_hdmi_i2s_hw_params, .audio_shutdown = dw_hdmi_i2s_audio_shutdown, .get_dai_id = dw_hdmi_i2s_get_dai_id, -- 2.23.0.187.g17f5b7556c-goog

3 3

[alsa-devel] [PATCH] alsa-lib: Remove usleep usage
by Rosen Penev 02 Sep '19

02 Sep '19

usleep was deprecated by POSIX 2008 and is unavailable without deprecated APIs enabled in uClibc-ng. Signed-off-by: Rosen Penev <rosenp(a)gmail.com> --- src/pcm/pcm_shm.c | 3 ++- src/ucm/main.c | 3 ++- test/mixtest.c | 2 +- 3 files changed, 5 insertions(+), 3 deletions(-) diff --git a/src/pcm/pcm_shm.c b/src/pcm/pcm_shm.c index 26a27a57..30a77cd4 100644 --- a/src/pcm/pcm_shm.c +++ b/src/pcm/pcm_shm.c @@ -485,12 +485,13 @@ static int snd_pcm_shm_drain(snd_pcm_t *pcm) snd_pcm_shm_t *shm = pcm->private_data; volatile snd_pcm_shm_ctrl_t *ctrl = shm->ctrl; int err; + struct timespec req = { 0, 10 * 1000 * 1000 }; do { ctrl->cmd = SNDRV_PCM_IOCTL_DRAIN; err = snd_pcm_shm_action(pcm); if (err != -EAGAIN) break; - usleep(10000); + nanosleep(&req, NULL); } while (1); if (err < 0) return err; diff --git a/src/ucm/main.c b/src/ucm/main.c index 99b1cd08..0e5236cb 100644 --- a/src/ucm/main.c +++ b/src/ucm/main.c @@ -359,6 +359,7 @@ static int execute_sequence(snd_use_case_mgr_t *uc_mgr, { struct list_head *pos; struct sequence_element *s; + struct timespec sleep = { 0, s->data.sleep * 1000 }; char *cdev = NULL; snd_ctl_t *ctl = NULL; int err = 0; @@ -440,7 +441,7 @@ static int execute_sequence(snd_use_case_mgr_t *uc_mgr, } break; case SEQUENCE_ELEMENT_TYPE_SLEEP: - usleep(s->data.sleep); + nanosleep(&sleep, NULL); break; case SEQUENCE_ELEMENT_TYPE_EXEC: err = system(s->data.exec); diff --git a/test/mixtest.c b/test/mixtest.c index 57ca5d53..f0940344 100644 --- a/test/mixtest.c +++ b/test/mixtest.c @@ -77,7 +77,7 @@ static double detect_cpu_clock() rdtscll(tsc_begin); clock_gettime(CLOCK_MONOTONIC, &tm_begin); - usleep(1000000); + sleep(1); rdtscll(tsc_end); clock_gettime(CLOCK_MONOTONIC, &tm_end); -- 2.17.1

1 0

[alsa-devel] [PATCH 1/3] ASoC: es8316: judge PCM rate at later timing
by Katsuhiro Suzuki 31 Aug '19

31 Aug '19

This patch change the judge timing about playing/capturing PCM rate. Original code set constraints list of PCM rate limits at set_sysclk. This strategy works well if system is using fixed rate clock. But some boards and SoC (such as RockPro64 and RockChip I2S) has connected SoC MCLK out to ES8316 MCLK in. In this case, SoC side I2S will choose suitable frequency of MCLK such as fs * mclk-fs when user starts playing or capturing. Bad scenario as follows (mclk-fs = 256): - Initialize sysclk by correct value (Ex. 12.288MHz) - ES8316 set constraints of PCM rate by sysclk 48kHz (1/256), 32kHz (1/384), 30.720kHz (1/400), 24kHz (1/512), 16kHz (1/768), 12kHz (1/1024) - Play 48kHz sound, it's acceptable - Sysclk is not changed - Play 32kHz sound, it's acceptable - Set sysclk by 8.192MHz (= fs * mclk-fs = 32k * 256) - ES8316 set constraints of PCM rate by sysclk 32kHz (1/256), 21.33kHz (1/384), 20.48kHz (1/400), 16kHz (1/512), 10.66kHz (1/768), 8kHz (1/1024) - Play 48kHz again, but it's NOT acceptable because constraints list does not allow 48kHz Root cause of this strange behavior is changing constraints list at set_sysclk timing. It seems that is too early to determine. So this patch does not use constraints list and check PCM rate limit more later timing at hw_params. Signed-off-by: Katsuhiro Suzuki <katsuhiro(a)katsuster.net> --- sound/soc/codecs/es8316.c | 37 +++++++++++++++---------------------- 1 file changed, 15 insertions(+), 22 deletions(-) diff --git a/sound/soc/codecs/es8316.c b/sound/soc/codecs/es8316.c index ed2959dbe1fb..229808fa627c 100644 --- a/sound/soc/codecs/es8316.c +++ b/sound/soc/codecs/es8316.c @@ -363,27 +363,12 @@ static int es8316_set_dai_sysclk(struct snd_soc_dai *codec_dai, { struct snd_soc_component *component = codec_dai->component; struct es8316_priv *es8316 = snd_soc_component_get_drvdata(component); - int i; - int count = 0; es8316->sysclk = freq; if (freq == 0) return 0; - /* Limit supported sample rates to ones that can be autodetected - * by the codec running in slave mode. - */ - for (i = 0; i < NR_SUPPORTED_MCLK_LRCK_RATIOS; i++) { - const unsigned int ratio = supported_mclk_lrck_ratios[i]; - - if (freq % ratio == 0) - es8316->allowed_rates[count++] = freq / ratio; - } - - es8316->sysclk_constraints.list = es8316->allowed_rates; - es8316->sysclk_constraints.count = count; - return 0; } @@ -449,13 +434,6 @@ static int es8316_pcm_startup(struct snd_pcm_substream *substream, return -EINVAL; } - /* The set of sample rates that can be supported depends on the - * MCLK supplied to the CODEC. - */ - snd_pcm_hw_constraint_list(substream->runtime, 0, - SNDRV_PCM_HW_PARAM_RATE, - &es8316->sysclk_constraints); - return 0; } @@ -466,12 +444,27 @@ static int es8316_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_soc_component *component = dai->component; struct es8316_priv *es8316 = snd_soc_component_get_drvdata(component); u8 wordlen = 0; + int i; if (!es8316->sysclk) { dev_err(component->dev, "No MCLK configured\n"); return -EINVAL; } + /* Limit supported sample rates to ones that can be autodetected + * by the codec running in slave mode. + */ + for (i = 0; i < NR_SUPPORTED_MCLK_LRCK_RATIOS; i++) { + const unsigned int ratio = supported_mclk_lrck_ratios[i]; + + if (es8316->sysclk % ratio != 0) + continue; + if (es8316->sysclk / ratio == params_rate(params)) + break; + } + if (i == NR_SUPPORTED_MCLK_LRCK_RATIOS) + return -EINVAL; + switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: wordlen = ES8316_SERDATA2_LEN_16; -- 2.23.0.rc1

2 4

[alsa-devel] [PATCH 0/4] ASoC: mediatek: mt8183: some cleanups
by Tzung-Bi Shih 30 Aug '19

30 Aug '19

The series contains some cleanup refactors and should not change any behavior. Tzung-Bi Shih (4): ASoC: mediatek: mt8183: move private structure ASoC: mediatek: mt8183: move headset jack to card-specific storage ASoC: mediatek: mt8183: remove forward declaration of headset_init ASoC: mediatek: mt8183: fix space issues .../mt8183/mt8183-mt6358-ts3a227-max98357.c | 55 +++++++++---------- 1 file changed, 26 insertions(+), 29 deletions(-) -- 2.23.0.187.g17f5b7556c-goog

2 8

[alsa-devel] [PATCH v2 1/5] ASoC: dt-bindings: sun4i-spdif: Fix dma-names warning
by Maxime Ripard 30 Aug '19

30 Aug '19

From: Maxime Ripard <maxime.ripard(a)bootlin.com> Even though the H6 compatible has been properly added, the exeption for the number of DMA channels hasn't been updated, leading in a validation warning. Fix this. Fixes: b20453031472 ("dt-bindings: sound: sun4i-spdif: Add Allwinner H6 compatible") Signed-off-by: Maxime Ripard <maxime.ripard(a)bootlin.com> --- Changes from v1: - Fix subject prefix --- .../devicetree/bindings/sound/allwinner,sun4i-a10-spdif.yaml | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/allwinner,sun4i-a10-spdif.yaml b/Documentation/devicetree/bindings/sound/allwinner,sun4i-a10-spdif.yaml index e0284d8c3b63..38d4cede0860 100644 --- a/Documentation/devicetree/bindings/sound/allwinner,sun4i-a10-spdif.yaml +++ b/Documentation/devicetree/bindings/sound/allwinner,sun4i-a10-spdif.yaml @@ -70,7 +70,9 @@ allOf: properties: compatible: contains: - const: allwinner,sun8i-h3-spdif + enum: + - allwinner,sun8i-h3-spdif + - allwinner,sun50i-h6-spdif then: properties: -- 2.21.0

3 12

Re: [alsa-devel] [PATCH 2/2] ASoC: rt1011: ADCDAT pin config modification
by Mark Brown 30 Aug '19

30 Aug '19

On Wed, Aug 28, 2019 at 11:01:21AM +0000, Shuming [范書銘] wrote: > > > ADCDAT pin config sets to output mode in default. > > > And the driver creates the kcontrol to control ADCDAT pin configuration. > > Why would this change at runtime? I'd expect this to be controlled by the > > machine driver or a DT property. > It will not change at runtime in the normal case. > I think we may use it to test the feedback signal in factory mode. So really this is setting up a loopback for testing?

2 2

[alsa-devel] [PATCH] ASoC: fsl_ssi: Fix clock control issue in master mode
by Shengjiu Wang 30 Aug '19

30 Aug '19

The test case is arecord -Dhw:0 -d 10 -f S16_LE -r 48000 -c 2 temp.wav & aplay -Dhw:0 -d 30 -f S16_LE -r 48000 -c 2 test.wav There will be error after end of arecord: aplay: pcm_write:2051: write error: Input/output error Capture and Playback work in parallel in master mode, one substream stops, the other substream is impacted, the reason is that clock is disabled wrongly. The clock's reference count is not increased when second substream starts, the hw_param() function returns in the beginning because first substream is enabled, then in end of first substream, the hw_free() disables the clock. This patch is to move the clock enablement to the place before checking of the device enablement in hw_param(). Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/fsl_ssi.c | 18 +++++++++--------- 1 file changed, 9 insertions(+), 9 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index b0a6fead1a6a..537dc69256f0 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -799,15 +799,6 @@ static int fsl_ssi_hw_params(struct snd_pcm_substream *substream, u32 wl = SSI_SxCCR_WL(sample_size); int ret; - /* - * SSI is properly configured if it is enabled and running in - * the synchronous mode; Note that AC97 mode is an exception - * that should set separate configurations for STCCR and SRCCR - * despite running in the synchronous mode. - */ - if (ssi->streams && ssi->synchronous) - return 0; - if (fsl_ssi_is_i2s_master(ssi)) { ret = fsl_ssi_set_bclk(substream, dai, hw_params); if (ret) @@ -823,6 +814,15 @@ static int fsl_ssi_hw_params(struct snd_pcm_substream *substream, } } + /* + * SSI is properly configured if it is enabled and running in + * the synchronous mode; Note that AC97 mode is an exception + * that should set separate configurations for STCCR and SRCCR + * despite running in the synchronous mode. + */ + if (ssi->streams && ssi->synchronous) + return 0; + if (!fsl_ssi_is_ac97(ssi)) { /* * Keep the ssi->i2s_net intact while having a local variable -- 2.21.0

3 2

[alsa-devel] [PATCH v2] ASoC: Intel: Baytrail: Fix implicit fallthrough warning
by Cezary Rojewski 30 Aug '19

30 Aug '19

Append fallthrough statement to fix warning reported during compilation. Fixes: b80d19c166c4 ("ASoC: Intel: Restore Baytrail ADSP streams only when ADSP was in reset") Signed-off-by: Cezary Rojewski <cezary.rojewski(a)intel.com> --- Changes from v1: - made use of fallthrough statement as suggested by Pierre - updated commit message sound/soc/intel/baytrail/sst-baytrail-pcm.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/intel/baytrail/sst-baytrail-pcm.c b/sound/soc/intel/baytrail/sst-baytrail-pcm.c index 9cbc982d46a9..54f2ee3010ee 100644 --- a/sound/soc/intel/baytrail/sst-baytrail-pcm.c +++ b/sound/soc/intel/baytrail/sst-baytrail-pcm.c @@ -193,6 +193,7 @@ static int sst_byt_pcm_trigger(struct snd_pcm_substream *substream, int cmd) break; case SNDRV_PCM_TRIGGER_SUSPEND: pdata->restore_stream = false; + /* fallthrough */ case SNDRV_PCM_TRIGGER_PAUSE_PUSH: sst_byt_stream_pause(byt, pcm_data->stream); break; -- 2.17.1

3 2

[alsa-devel] [PATCH 0/4] ASoC: ti: davinci: Fixes for daVinci boards with McBSP
by Peter Ujfalusi 30 Aug '19

30 Aug '19

Hi, Morimoto-san pointed out in a private thread that the davinci-i2s driver is using snd_soc_rtdcom_lookup(), which is should not really. It took a while to get access to a daVinci board which uses McBSP, but eventually I got access to a remote dm644x-evm and dma365-evm. Initially the audio was not even probing on them (fixes sent for mach-davinci), but then the driver was still crashing and not behaving correctly. With the 4 patch dm644x-evm have working audio and the offending snd_soc_rtdcom_lookup() is gone from the driver. dm365-evm still have some issues regarding to EDMA, but it looks like a system issue which can be fixed later. Regards, Peter --- Peter Ujfalusi (4): ASoC: ti: davinci-i2s: Add S32_LE as support format ASoC: ti: davinci-i2s: Move the XSYNCERR workaround to .prepare callback ASoC: ti: edma-pcm: Fix for legacy dma_slave_map based channel lookup ASoC: ti: davinci-evm: Do not fail if the dai_set_sysclk returns -ENOTSUPP sound/soc/ti/davinci-evm.c | 2 +- sound/soc/ti/davinci-i2s.c | 90 +++++++++++++++++--------------------- sound/soc/ti/edma-pcm.c | 17 ++++++- 3 files changed, 57 insertions(+), 52 deletions(-) -- Peter Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki. Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki

2 8

[alsa-devel] [PATCH 0/4] ASoC: Intel: Haswell: Adjust machine device private
by Cezary Rojewski 30 Aug '19

30 Aug '19

Apart from Haswell machines, all other devices have their private data set to snd_soc_acpi_mach instance. Changes for HSW/ BDW boards introduced with series: https://patchwork.kernel.org/cover/10782035/ added support for dai_link platform_name adjustments within card probe routines. These take for granted private_data points to snd_soc_acpi_mach whereas for Haswell, it's sst_pdata instead. Change private context of platform_device - representing machine board - to address this. Caught by recent cleanups where content of sst_pdata was moved. Currently, despite the incorrect cast, dereferenced field points happily to NULL (uninitialized field), so no panics were observed. Cezary Rojewski (4): ASoC: Intel: Haswell: Adjust machine device private context ASoC: Intel: haswell: Simplify device probe ASoC: Intel: bdw-rt5677: Simplify device probe ASoC: Intel: broadwell: Simplify device probe sound/soc/intel/boards/bdw-rt5677.c | 6 +----- sound/soc/intel/boards/broadwell.c | 6 +----- sound/soc/intel/boards/haswell.c | 6 +----- sound/soc/intel/common/sst-acpi.c | 3 ++- 4 files changed, 5 insertions(+), 16 deletions(-) -- 2.17.1

3 21

[alsa-devel] [PATCH] ASoC: SOF/Intel: fix selection of SND_INTEL_NHTL
by Pierre-Louis Bossart 30 Aug '19

30 Aug '19

We should only select SND_INTEL_NHLT when ACPI is defined. This was done for the legacy HDAudio driver but not for DSP-enabled cases, leading to compilation errors with randconfig. Fix by aligning on the same solution. For the Skylake driver this is overkill since there is a top-level dependency on ACPI, but it doesn't hurt and it's better to have consistency. Fixes: 68b953aeb50d9 ('ASoC: SOF: Intel: hda: fixup HDaudio topology name with DMIC number') Reported-by: Randy Dunlap <rdunlap(a)infradead.org> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/intel/Kconfig | 2 +- sound/soc/sof/intel/Kconfig | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index a3ec17fd63cd..01c99750212a 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -215,7 +215,7 @@ config SND_SOC_INTEL_SKYLAKE_COMMON select SND_SOC_INTEL_SST select SND_SOC_HDAC_HDA if SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC select SND_SOC_ACPI_INTEL_MATCH - select SND_INTEL_NHLT + select SND_INTEL_NHLT if ACPI help If you have a Intel Skylake/Broxton/ApolloLake/KabyLake/ GeminiLake or CannonLake platform with the DSP enabled in the BIOS diff --git a/sound/soc/sof/intel/Kconfig b/sound/soc/sof/intel/Kconfig index 889b6202d054..479ba249e219 100644 --- a/sound/soc/sof/intel/Kconfig +++ b/sound/soc/sof/intel/Kconfig @@ -286,7 +286,7 @@ config SND_SOC_SOF_HDA tristate select SND_HDA_EXT_CORE if SND_SOC_SOF_HDA_LINK select SND_SOC_HDAC_HDA if SND_SOC_SOF_HDA_AUDIO_CODEC - select SND_INTEL_NHLT + select SND_INTEL_NHLT if ACPI help This option is not user-selectable but automagically handled by 'select' statements at a higher level -- 2.20.1

2 1

[alsa-devel] [PATCH] ASoC: rsnd: fixup 6ch settings to 8ch
by Kuninori Morimoto 30 Aug '19

30 Aug '19

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> rsnd need to use 8ch clock settings for 6ch for TDM. Otherwise, it can't work correctly. This patch fixup it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/core.c | 18 +++++++++++++----- sound/soc/sh/rcar/rsnd.h | 1 + sound/soc/sh/rcar/ssi.c | 2 ++ 3 files changed, 16 insertions(+), 5 deletions(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 4fe83e61..37cb615 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -300,6 +300,18 @@ int rsnd_runtime_channel_after_ctu_with_params(struct rsnd_dai_stream *io, return chan; } +int rsnd_channel_normalization(int chan) +{ + if ((chan > 8) || (chan < 0)) + return 0; + + /* TDM Extend Mode needs 8ch */ + if (chan == 6) + chan = 8; + + return chan; +} + int rsnd_runtime_channel_for_ssi_with_params(struct rsnd_dai_stream *io, struct snd_pcm_hw_params *params) { @@ -312,11 +324,7 @@ int rsnd_runtime_channel_for_ssi_with_params(struct rsnd_dai_stream *io, if (rsnd_runtime_is_multi_ssi(io)) chan /= rsnd_rdai_ssi_lane_get(rdai); - /* TDM Extend Mode needs 8ch */ - if (chan == 6) - chan = 8; - - return chan; + return rsnd_channel_normalization(chan); } int rsnd_runtime_is_multi_ssi(struct rsnd_dai_stream *io) diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index 0e6ef4e1..7727add 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -446,6 +446,7 @@ void rsnd_parse_connect_common(struct rsnd_dai *rdai, struct device_node *playback, struct device_node *capture); +int rsnd_channel_normalization(int chan); #define rsnd_runtime_channel_original(io) \ rsnd_runtime_channel_original_with_params(io, NULL) int rsnd_runtime_channel_original_with_params(struct rsnd_dai_stream *io, diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index f5afab6..44bda21 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -303,6 +303,8 @@ static int rsnd_ssi_master_clk_start(struct rsnd_mod *mod, if (rsnd_runtime_is_tdm_split(io)) chan = rsnd_io_converted_chan(io); + chan = rsnd_channel_normalization(chan); + main_rate = rsnd_ssi_clk_query(rdai, rate, chan, &idx); if (!main_rate) { dev_err(dev, "unsupported clock rate\n"); -- 2.7.4

3 7

[alsa-devel] [PATCH] ALSA: hda/realtek - Fix overridden device-specific initialization
by Takashi Iwai 30 Aug '19

30 Aug '19

The recent change to shuffle the codec initialization procedure for Realtek via commit 607ca3bd220f ("ALSA: hda/realtek - EAPD turn on later") caused the silent output on some machines. This change was supposed to be safe, but it isn't actually; some devices have quirk setups to override the EAPD via COEF or BTL in the additional verb table, which is applied at the beginning of snd_hda_gen_init(). And this EAPD setup is again overridden in alc_auto_init_amp(). For recovering from the regression, tell snd_hda_gen_init() not to apply the verbs there by a new flag, then apply the verbs in alc_init(). BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=204727 Fixes: 607ca3bd220f ("ALSA: hda/realtek - EAPD turn on later") Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_generic.c | 3 ++- sound/pci/hda/hda_generic.h | 1 + sound/pci/hda/patch_realtek.c | 2 ++ 3 files changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 5bf24fb819d2..10d502328b76 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -6009,7 +6009,8 @@ int snd_hda_gen_init(struct hda_codec *codec) if (spec->init_hook) spec->init_hook(codec); - snd_hda_apply_verbs(codec); + if (!spec->skip_verbs) + snd_hda_apply_verbs(codec); init_multi_out(codec); init_extra_out(codec); diff --git a/sound/pci/hda/hda_generic.h b/sound/pci/hda/hda_generic.h index 5f199dcb0d18..fb9f1a90238b 100644 --- a/sound/pci/hda/hda_generic.h +++ b/sound/pci/hda/hda_generic.h @@ -243,6 +243,7 @@ struct hda_gen_spec { unsigned int indep_hp_enabled:1; /* independent HP enabled */ unsigned int have_aamix_ctl:1; unsigned int hp_mic_jack_modes:1; + unsigned int skip_verbs:1; /* don't apply verbs at snd_hda_gen_init() */ /* additional mute flags (only effective with auto_mute_via_amp=1) */ u64 mute_bits; diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index e333b3e30e31..03f903f9247d 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -837,9 +837,11 @@ static int alc_init(struct hda_codec *codec) if (spec->init_hook) spec->init_hook(codec); + spec->gen.skip_verbs = 1; /* applied in below */ snd_hda_gen_init(codec); alc_fix_pll(codec); alc_auto_init_amp(codec, spec->init_amp); + snd_hda_apply_verbs(codec); /* apply verbs here after own init */ snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_INIT); -- 2.16.4

1 0

[alsa-devel] [PATCH] [PATCH v2] ALSA: hda: Retry codec read on first time failure
by vani.ganji＠intel.com 30 Aug '19

30 Aug '19

From: Vani Ganji <vani.ganji(a)intel.com> Sometimes HDMI audio fails while fast switching between HDMI and Speaker due to codec Get Power state verb read failure. Verb is codec driver to access codec HW. Since issue scenario involves fast switching between HDMI and speaker, there might be chance to fail read from HW and same happened in the error scenario. Based on Display/Graphics architect suggestion, added a retry mechanism for failure on first attempt. Changes from v1: -Corrected missing braces. Signed-off-by: Vani Ganji <vani.ganji(a)intel.com> --- sound/hda/hdac_device.c | 15 +++++++++++---- 1 file changed, 11 insertions(+), 4 deletions(-) diff --git a/sound/hda/hdac_device.c b/sound/hda/hdac_device.c index b26cc93e7e10..3874d54aa686 100644 --- a/sound/hda/hdac_device.c +++ b/sound/hda/hdac_device.c @@ -984,10 +984,17 @@ static unsigned int codec_read(struct hdac_device *hdac, hda_nid_t nid, { unsigned int cmd = snd_hdac_make_cmd(hdac, nid, verb, parm); unsigned int res; - - if (snd_hdac_exec_verb(hdac, cmd, flags, &res)) - return -1; - + unsigned int count; + + /* Retry verb once more if it fails first time */ + /* recommended to retry since it is HW related */ + for (count = 0; count <= 1; count++) { + if (snd_hdac_exec_verb(hdac, cmd, flags, &res)) { + if (count == 1) + return -1; + } else + break; + } return res; } -- 2.17.1

2 1

[alsa-devel] [PATCH] ALSA: firewire-motu: add support for MOTU 4pre
by Takashi Sakamoto 30 Aug '19

30 Aug '19

MOTU 4pre was launched in 2012 by MOTU, Inc. This commit allows userspace applications can transmit and receive PCM frames and MIDI messages for this model via ALSA PCM interface and RawMidi/Sequencer interfaces. The device supports MOTU protocol version 3. Unlike the other devices, the device is simply designed. The size of data block is fixed to 10 quadlets during available sampling rates (44.1 - 96.0 kHz). Each data block includes 1 source packet header, 2 data chunks for messages, 8 data chunks for PCM samples and 2 data chunks for padding to quadlet alignment. The device has no MIDI, optical, BNC and AES/EBU interfaces. Like support for the other MOTU devices, the quality of playback sound is not enough good with periodical noise yet. $ python2 crpp < ~/git/am-config-rom/motu/motu-4pre.img ROM header and bus information block ----------------------------------------------------------------- 400 041078cc bus_info_length 4, crc_length 16, crc 30924 404 31333934 bus_name "1394" 408 20ff7000 irmc 0, cmc 0, isc 1, bmc 0, cyc_clk_acc 255, max_rec 7 (256) 40c 0001f200 company_id 0001f2 | 410 000a41c5 device_id 00000a41c5 | EUI-64 0001f200000a41c5 root directory ----------------------------------------------------------------- 414 0004ef04 directory_length 4, crc 61188 418 030001f2 vendor 41c 0c0083c0 node capabilities per IEEE 1394 420 d1000002 --> unit directory at 428 424 8d000005 --> eui-64 leaf at 438 unit directory at 428 ----------------------------------------------------------------- 428 0003ceda directory_length 3, crc 52954 42c 120001f2 specifier id 430 13000045 version 434 17103800 model eui-64 leaf at 438 ----------------------------------------------------------------- 438 0002d248 leaf_length 2, crc 53832 43c 0001f200 company_id 0001f2 | 440 000a41c5 device_id 00000a41c5 | EUI-64 0001f200000a41c5 Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/motu/motu.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/firewire/motu/motu.c b/sound/firewire/motu/motu.c index 03cda2166ea3..72908b4de77c 100644 --- a/sound/firewire/motu/motu.c +++ b/sound/firewire/motu/motu.c @@ -247,6 +247,17 @@ static const struct snd_motu_spec motu_audio_express = { .analog_out_ports = 4, }; +static const struct snd_motu_spec motu_4pre = { + .name = "4pre", + .protocol = &snd_motu_protocol_v3, + .flags = SND_MOTU_SPEC_SUPPORT_CLOCK_X2 | + SND_MOTU_SPEC_TX_MICINST_CHUNK | + SND_MOTU_SPEC_TX_RETURN_CHUNK | + SND_MOTU_SPEC_RX_SEPARETED_MAIN, + .analog_in_ports = 2, + .analog_out_ports = 2, +}; + #define SND_MOTU_DEV_ENTRY(model, data) \ { \ .match_flags = IEEE1394_MATCH_VENDOR_ID | \ @@ -265,6 +276,7 @@ static const struct ieee1394_device_id motu_id_table[] = { SND_MOTU_DEV_ENTRY(0x000015, &motu_828mk3), /* FireWire only. */ SND_MOTU_DEV_ENTRY(0x000035, &motu_828mk3), /* Hybrid. */ SND_MOTU_DEV_ENTRY(0x000033, &motu_audio_express), + SND_MOTU_DEV_ENTRY(0x000045, &motu_4pre), { } }; MODULE_DEVICE_TABLE(ieee1394, motu_id_table); -- 2.20.1

2 1

[alsa-devel] [PATCH v5 0/4] ASoC: codecs: Add WSA881x Smart Speaker amplifier support
by Srinivas Kandagatla 29 Aug '19

29 Aug '19

Thanks for reviewing v4 patchset, here is v5 with addressing the comments in v4 This patchset adds support to WSA8810/WSA8815 Class-D Smart Speaker Amplifier which is SoundWire interfaced. This also adds support to some missing bits in SoundWire bus layer like Device Tree support. This patchset along with DB845c machine driver and WCD934x codec driver has been tested on SDM845 SoC based DragonBoard DB845c with two WSA8810 speakers. Most of the code in this driver is rework of Qualcomm downstream drivers used in Andriod. Credits to Banajit Goswami and Patrick Lai's Team. TODO: Add thermal sensor support in WSA881x. Thanks, srini Changes since v4: - updated slave bindings according to Rob's and Pierre Suggestion. - fixup warnings from make dt_binding_check Srinivas Kandagatla (4): dt-bindings: soundwire: add slave bindings soundwire: core: add device tree support for slave devices dt-bindings: ASoC: Add WSA881x bindings ASoC: codecs: add wsa881x amplifier support .../bindings/sound/qcom,wsa881x.yaml | 41 + .../soundwire/soundwire-controller.yaml | 72 ++ drivers/soundwire/bus.c | 2 + drivers/soundwire/bus.h | 1 + drivers/soundwire/slave.c | 52 + sound/soc/codecs/Kconfig | 10 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/wsa881x.c | 1134 +++++++++++++++++ 8 files changed, 1314 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/qcom,wsa881x.yaml create mode 100644 Documentation/devicetree/bindings/soundwire/soundwire-controller.yaml create mode 100644 sound/soc/codecs/wsa881x.c -- 2.21.0

3 10

[alsa-devel] [ANNOUNCE] 2019 Linux Audio miniconference
by Mark Brown 29 Aug '19

29 Aug '19

Hi, As in previous years we're going to have an audio miniconference so we can get together and talk through issues, especially design decisions, face to face. This year's event will be held on October 31st in Lyon, France, the day after ELC-E. This will be held at the Lyon Convention Center (the ELC-E venue), generously sponsored by Intel. As with previous years let's pull together an agenda through a mailing list discussion - if people could reply to this mail with any topics they'd like to discuss we can take it from there. Of course if we can sort things out more quickly via the mailing list that's even better! If you're planning to attend please fill out the form here: https://docs.google.com/forms/d/e/1FAIpQLSd6FV-tWYZ1fHlCmyzQhG98OtYH9W9GX-1… This event will be covered by the same code of conduct as ELC-E. Thanks again to Intel for supporting this event. Thanks, Mark

3 3

Re: [alsa-devel] mmotm 2019-08-27-20-39 uploaded (sound/hda/intel-nhlt.c)
by Randy Dunlap 29 Aug '19

29 Aug '19

On 8/27/19 8:40 PM, akpm(a)linux-foundation.org wrote: > The mm-of-the-moment snapshot 2019-08-27-20-39 has been uploaded to > > http://www.ozlabs.org/~akpm/mmotm/ > > mmotm-readme.txt says > > README for mm-of-the-moment: > > http://www.ozlabs.org/~akpm/mmotm/ > > This is a snapshot of my -mm patch queue. Uploaded at random hopefully > more than once a week. > > You will need quilt to apply these patches to the latest Linus release (5.x > or 5.x-rcY). The series file is in broken-out.tar.gz and is duplicated in > http://ozlabs.org/~akpm/mmotm/series > > The file broken-out.tar.gz contains two datestamp files: .DATE and > .DATE-yyyy-mm-dd-hh-mm-ss. Both contain the string yyyy-mm-dd-hh-mm-ss, > followed by the base kernel version against which this patch series is to > be applied. (from linux-next tree, but problem found/seen in mmotm) Sorry, I don't know who is responsible for this driver. ~~~~~~~~~~~~~~~~~~~~~~ on i386: CC sound/hda/intel-nhlt.o ../sound/hda/intel-nhlt.c:14:25: error: redefinition of ‘intel_nhlt_init’ struct nhlt_acpi_table *intel_nhlt_init(struct device *dev) ^~~~~~~~~~~~~~~ In file included from ../sound/hda/intel-nhlt.c:5:0: ../include/sound/intel-nhlt.h:134:39: note: previous definition of ‘intel_nhlt_init’ was here static inline struct nhlt_acpi_table *intel_nhlt_init(struct device *dev) ^~~~~~~~~~~~~~~ ../sound/hda/intel-nhlt.c: In function ‘intel_nhlt_init’: ../sound/hda/intel-nhlt.c:39:14: error: dereferencing pointer to incomplete type ‘struct nhlt_resource_desc’ if (nhlt_ptr->length) ^~ ../sound/hda/intel-nhlt.c:41:4: error: implicit declaration of function ‘memremap’; did you mean ‘ioremap’? [-Werror=implicit-function-declaration] memremap(nhlt_ptr->min_addr, nhlt_ptr->length, ^~~~~~~~ ioremap ../sound/hda/intel-nhlt.c:42:6: error: ‘MEMREMAP_WB’ undeclared (first use in this function) MEMREMAP_WB); ^~~~~~~~~~~ ../sound/hda/intel-nhlt.c:42:6: note: each undeclared identifier is reported only once for each function it appears in ../sound/hda/intel-nhlt.c:45:25: error: dereferencing pointer to incomplete type ‘struct nhlt_acpi_table’ (strncmp(nhlt_table->header.signature, ^~ ../sound/hda/intel-nhlt.c:48:3: error: implicit declaration of function ‘memunmap’; did you mean ‘vunmap’? [-Werror=implicit-function-declaration] memunmap(nhlt_table); ^~~~~~~~ vunmap ../sound/hda/intel-nhlt.c: At top level: ../sound/hda/intel-nhlt.c:56:6: error: redefinition of ‘intel_nhlt_free’ void intel_nhlt_free(struct nhlt_acpi_table *nhlt) ^~~~~~~~~~~~~~~ In file included from ../sound/hda/intel-nhlt.c:5:0: ../include/sound/intel-nhlt.h:139:20: note: previous definition of ‘intel_nhlt_free’ was here static inline void intel_nhlt_free(struct nhlt_acpi_table *addr) ^~~~~~~~~~~~~~~ ../sound/hda/intel-nhlt.c:62:5: error: redefinition of ‘intel_nhlt_get_dmic_geo’ int intel_nhlt_get_dmic_geo(struct device *dev, struct nhlt_acpi_table *nhlt) ^~~~~~~~~~~~~~~~~~~~~~~ In file included from ../sound/hda/intel-nhlt.c:5:0: ../include/sound/intel-nhlt.h:143:19: note: previous definition of ‘intel_nhlt_get_dmic_geo’ was here static inline int intel_nhlt_get_dmic_geo(struct device *dev, ^~~~~~~~~~~~~~~~~~~~~~~ ../sound/hda/intel-nhlt.c: In function ‘intel_nhlt_get_dmic_geo’: ../sound/hda/intel-nhlt.c:76:11: error: dereferencing pointer to incomplete type ‘struct nhlt_endpoint’ if (epnt->linktype == NHLT_LINK_DMIC) { ^~ ../sound/hda/intel-nhlt.c:76:25: error: ‘NHLT_LINK_DMIC’ undeclared (first use in this function) if (epnt->linktype == NHLT_LINK_DMIC) { ^~~~~~~~~~~~~~ ../sound/hda/intel-nhlt.c:79:15: error: dereferencing pointer to incomplete type ‘struct nhlt_dmic_array_config’ switch (cfg->array_type) { ^~ ../sound/hda/intel-nhlt.c:80:9: error: ‘NHLT_MIC_ARRAY_2CH_SMALL’ undeclared (first use in this function) case NHLT_MIC_ARRAY_2CH_SMALL: ^~~~~~~~~~~~~~~~~~~~~~~~ ../sound/hda/intel-nhlt.c:81:9: error: ‘NHLT_MIC_ARRAY_2CH_BIG’ undeclared (first use in this function); did you mean ‘NHLT_MIC_ARRAY_2CH_SMALL’? case NHLT_MIC_ARRAY_2CH_BIG: ^~~~~~~~~~~~~~~~~~~~~~ NHLT_MIC_ARRAY_2CH_SMALL ../sound/hda/intel-nhlt.c:82:16: error: ‘MIC_ARRAY_2CH’ undeclared (first use in this function); did you mean ‘NHLT_MIC_ARRAY_2CH_BIG’? dmic_geo = MIC_ARRAY_2CH; ^~~~~~~~~~~~~ NHLT_MIC_ARRAY_2CH_BIG ../sound/hda/intel-nhlt.c:85:9: error: ‘NHLT_MIC_ARRAY_4CH_1ST_GEOM’ undeclared (first use in this function); did you mean ‘NHLT_MIC_ARRAY_2CH_BIG’? case NHLT_MIC_ARRAY_4CH_1ST_GEOM: ^~~~~~~~~~~~~~~~~~~~~~~~~~~ NHLT_MIC_ARRAY_2CH_BIG ../sound/hda/intel-nhlt.c:86:9: error: ‘NHLT_MIC_ARRAY_4CH_L_SHAPED’ undeclared (first use in this function); did you mean ‘NHLT_MIC_ARRAY_4CH_1ST_GEOM’? case NHLT_MIC_ARRAY_4CH_L_SHAPED: ^~~~~~~~~~~~~~~~~~~~~~~~~~~ NHLT_MIC_ARRAY_4CH_1ST_GEOM AR sound/i2c/other/built-in.a ../sound/hda/intel-nhlt.c:87:9: error: ‘NHLT_MIC_ARRAY_4CH_2ND_GEOM’ undeclared (first use in this function); did you mean ‘NHLT_MIC_ARRAY_4CH_1ST_GEOM’? case NHLT_MIC_ARRAY_4CH_2ND_GEOM: ^~~~~~~~~~~~~~~~~~~~~~~~~~~ NHLT_MIC_ARRAY_4CH_1ST_GEOM ../sound/hda/intel-nhlt.c:88:16: error: ‘MIC_ARRAY_4CH’ undeclared (first use in this function); did you mean ‘MIC_ARRAY_2CH’? dmic_geo = MIC_ARRAY_4CH; ^~~~~~~~~~~~~ MIC_ARRAY_2CH AR sound/i2c/built-in.a CC drivers/bluetooth/btmtksdio.o ../sound/hda/intel-nhlt.c:90:9: error: ‘NHLT_MIC_ARRAY_VENDOR_DEFINED’ undeclared (first use in this function); did you mean ‘NHLT_MIC_ARRAY_4CH_L_SHAPED’? case NHLT_MIC_ARRAY_VENDOR_DEFINED: ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~ NHLT_MIC_ARRAY_4CH_L_SHAPED ../sound/hda/intel-nhlt.c:92:26: error: dereferencing pointer to incomplete type ‘struct nhlt_vendor_dmic_array_config’ dmic_geo = cfg_vendor->nb_mics; ^~ ../sound/hda/intel-nhlt.c: At top level: ../sound/hda/intel-nhlt.c:106:16: error: expected declaration specifiers or ‘...’ before string constant MODULE_LICENSE("GPL v2"); ^~~~~~~~ ../sound/hda/intel-nhlt.c:107:20: error: expected declaration specifiers or ‘...’ before string constant MODULE_DESCRIPTION("Intel NHLT driver"); ^~~~~~~~~~~~~~~~~~~ cc1: some warnings being treated as errors make[3]: *** [../scripts/Makefile.build:266: sound/hda/intel-nhlt.o] Error 1 -- ~Randy

3 8

[alsa-devel] [PATCH] ALSA: firewire-lib: fix isoc cycle count to which rx packet is scheduled
by Takashi Sakamoto 29 Aug '19

29 Aug '19

When introducing the list of packet descriptor, for rx packet, the calculation of scheduled isoc cycle is omitted. This commit fixes the bug. Fixes: f4f6ae7b7c1f ("ALSA: firewire-lib: use packet descriptor for IT context") Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/amdtp-stream.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c index 1a92855c7647..88270257e896 100644 --- a/sound/firewire/amdtp-stream.c +++ b/sound/firewire/amdtp-stream.c @@ -732,7 +732,7 @@ static void generate_ideal_pkt_descs(struct amdtp_stream *s, struct pkt_desc *desc = descs + i; unsigned int index = (s->packet_index + i) % QUEUE_LENGTH; - desc->cycle = compute_cycle_count(*ctx_header); + desc->cycle = compute_it_cycle(*ctx_header); desc->syt = calculate_syt(s, desc->cycle); desc->data_blocks = calculate_data_blocks(s, desc->syt); -- 2.20.1

2 1

[alsa-devel] [PATCH] ALSA: dice: fix error path for initialization of stream structures
by Takashi Sakamoto 29 Aug '19

29 Aug '19

When introducing AMDTP domain to ALSA dice driver, error path does not handle error correctly. This commit fixes the bug. Fixes: e9f21129b8d8 ("ALSA: dice: support AMDTP domain") Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/dice/dice-stream.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/firewire/dice/dice-stream.c b/sound/firewire/dice/dice-stream.c index af8c5a2c28f3..f6a8627ae5a2 100644 --- a/sound/firewire/dice/dice-stream.c +++ b/sound/firewire/dice/dice-stream.c @@ -570,7 +570,7 @@ int snd_dice_stream_init_duplex(struct snd_dice *dice) destroy_stream(dice, AMDTP_OUT_STREAM, i); for (i = 0; i < MAX_STREAMS; i++) destroy_stream(dice, AMDTP_IN_STREAM, i); - break; + goto end; } } -- 2.20.1

2 1

[alsa-devel] mailing list outage
by Jaroslav Kysela 29 Aug '19

29 Aug '19

Hi, I am really sorry, but the e-mail delivery was paused for more than one day for this alsa-devel mailing list. The problem is fixed right now (messages from the queue are just being delivered). No messages should be lost. Jaroslav -- Jaroslav Kysela <perex(a)perex.cz> Linux Sound Maintainer; ALSA Project; Red Hat, Inc.

1 0

[alsa-devel] [PATCH v2 0/4] wm8904: adapt driver for use with audio-graph-card
by Michał Mirosław 29 Aug '19

29 Aug '19

This series allows to use WM8904 codec as audio-graph-card component. It starts with rework of FLL handling in the codec's driver, and as an example includes (untested) rework for codec with similar FLL: WM8994. Series based on tiwai/sound/for-next tree. You can also pull from: https://rere.qmqm.pl/git/linux branch: wm8904 (branch includes two fixes already sent to alsa-devel, but not merged yet). Michał Mirosław (4): ASoC: wm_fll: extract common code for Wolfson FLLs ASoC: wm8904: use common FLL code ASoC: wm8904: automatically choose clock source [RFT] ASoC: wm8994: use common FLL code sound/soc/atmel/atmel_wm8904.c | 11 +- sound/soc/codecs/Kconfig | 9 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/wm8904.c | 516 +++++++++++--------------------- sound/soc/codecs/wm8904.h | 5 - sound/soc/codecs/wm8994.c | 281 +++++------------- sound/soc/codecs/wm8994.h | 4 +- sound/soc/codecs/wm_fll.c | 518 +++++++++++++++++++++++++++++++++ sound/soc/codecs/wm_fll.h | 60 ++++ 9 files changed, 849 insertions(+), 557 deletions(-) create mode 100644 sound/soc/codecs/wm_fll.c create mode 100644 sound/soc/codecs/wm_fll.h -- 2.20.1

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[alsa-devel] [PATCH] ASoC: cs42xx8: Force suspend/resume during system suspend/resume
by Shengjiu Wang 29 Aug '19

29 Aug '19

Use force_suspend/resume to make sure clocks are disabled/enabled accordingly during system suspend/resume. Signed-off-by: Dong Aisheng <aisheng.dong(a)nxp.com> Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/codecs/cs42xx8.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/cs42xx8.c b/sound/soc/codecs/cs42xx8.c index 5b049fcdba20..94b1adb088fd 100644 --- a/sound/soc/codecs/cs42xx8.c +++ b/sound/soc/codecs/cs42xx8.c @@ -684,6 +684,8 @@ static int cs42xx8_runtime_suspend(struct device *dev) #endif const struct dev_pm_ops cs42xx8_pm = { + SET_SYSTEM_SLEEP_PM_OPS(pm_runtime_force_suspend, + pm_runtime_force_resume) SET_RUNTIME_PM_OPS(cs42xx8_runtime_suspend, cs42xx8_runtime_resume, NULL) }; EXPORT_SYMBOL_GPL(cs42xx8_pm); -- 2.21.0

4 3

[alsa-devel] [PATCH v2 00/25] ASoC: cleanup patches for soc-core
by Kuninori Morimoto 29 Aug '19

29 Aug '19

Hi Mark These are v2 of soc-core cleanup patches. I removed 3 patches which are reviewd from Ranjani / Pierre-Louis. Kuninori Morimoto (25): ASoC: soc-core: use device_register() ASoC: soc-core: set component->debugfs_root NULL ASoC: soc-core: add comment for for_each_xxx ASoC: soc-core: check return value of snd_soc_add_dai_link() ASoC: soc-core: don't use for_each_card_links_safe() at snd_soc_find_dai_link() ASoC: soc-core: reuse rtdcom at snd_soc_rtdcom_add() ASoC: soc-core: tidyup for snd_soc_dapm_new_controls() ASoC: soc-core: tidyup for snd_soc_add_component_controls() ASoC: soc-core: tidyup for snd_soc_dapm_add_routes() ASoC: soc-core: tidyup for snd_soc_add_card_controls() ASoC: soc-core: call snd_soc_dapm_debugfs_init() at soc_init_card_debugfs() ASoC: soc-core: remove unneeded list_empty() check for snd_soc_try_rebind_card() ASoC: soc-core: tidyup for card->deferred_resume_work ASoC: soc-core: define soc_dpcm_debugfs_add() for non CONFIG_DEBUG_FS ASoC: soc-core: dai_link check under soc_dpcm_debugfs_add() ASoC: soc-core: merge snd_soc_initialize_card_lists() ASoC: soc-core: remove unneeded dai_link check from snd_soc_remove_dai_link() ASoC: soc-core: add NOTE to snd_soc_rtdcom_lookup() ASoC: soc-core: don't call snd_soc_component_set_jack() ASoC: soc-core: don't alloc memory carelessly when creating debugfs ASoC: soc-core: soc_cleanup_card_resources() become void ASoC: soc-core: initialize component list ASoC: soc-core: initialize list at one place ASoC: soc-dai: use bit field for bus_control ASoC: soc-topology: use for_each_component_dais() at remove_dai() include/sound/soc-dai.h | 3 +- include/sound/soc-dpcm.h | 9 ++- include/sound/soc.h | 15 +--- sound/soc/soc-core.c | 197 ++++++++++++++++++++++++----------------------- sound/soc/soc-pcm.c | 3 + sound/soc/soc-topology.c | 2 +- 6 files changed, 118 insertions(+), 111 deletions(-) -- 2.7.4

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[alsa-devel] [PATCH] ALSA: hda - Fix potential endless loop at applying quirks
by Takashi Iwai 29 Aug '19

29 Aug '19

Since the chained quirks via chained_before flag is applied before the depth check, it may lead to the endless recursive calls, when the chain were set up incorrectly. Fix it by moving the depth check at the beginning of the loop. Fixes: 1f57825077dc ("ALSA: hda - Add chained_before flag to the fixup entry") Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_auto_parser.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/hda_auto_parser.c b/sound/pci/hda/hda_auto_parser.c index 8a134f750bae..2c6d2becfe1a 100644 --- a/sound/pci/hda/hda_auto_parser.c +++ b/sound/pci/hda/hda_auto_parser.c @@ -824,6 +824,8 @@ static void apply_fixup(struct hda_codec *codec, int id, int action, int depth) while (id >= 0) { const struct hda_fixup *fix = codec->fixup_list + id; + if (++depth > 10) + break; if (fix->chained_before) apply_fixup(codec, fix->chain_id, action, depth + 1); @@ -863,8 +865,6 @@ static void apply_fixup(struct hda_codec *codec, int id, int action, int depth) } if (!fix->chained || fix->chained_before) break; - if (++depth > 10) - break; id = fix->chain_id; } } -- 2.16.4

1 0

[alsa-devel] Options for ASYNC feedback source in USB-audio gadget (USB OTG)?
by Pavel Hofman 28 Aug '19

28 Aug '19

Hi, The existing usb-audio v.2 gadget defines its EP OUT as asynchronous https://github.com/torvalds/linux/blob/master/drivers/usb/gadget/function/f… , but does not define any feedback IN EP (which is against USB audio class 2.0 specs). Windows 10 USB audio class 2 driver requires presence of the feedback EP. By changing the EP OUT to adaptive (which in fact it is), we could fix the driver issue and windows plays through such device (RPi4) correctly now (linux/OSX already worked OK before). We will post a patch soon. RPi can now be used as an "intelligent" USB soundcard, allowing further processing in linux on RPi, before outputting to some final sound device. However, there are two clock domains involved - the host USB controller sending stream to the USB gadget, and the output soundcard of the gadget. While adaptive resampling is possible (jackd, PA do that), IMO the correct solution is going back to ASYNC EP OUT in the audio gadget and implementing a proper feedback EP. But the feedback EP needs input data. The whole chain behind the virtual capture alsa device defined by the gadget driver is clocked by the playback soundcard. May I ask for your opinions on possible options for 1) gathering the feedback data from the playback chain 2) passing the gathered samplerate-diff data to the gadget module for the feedback EP to send to the USB host A general solution would be very useful, at best without using alsa-lib defined PCM devices (e.g. Java still supports only hardware devices (hw:X), not PCM devices defined by alsa-lib). I very much appreciate your suggestions and help. Best regards, Pavel.

1 2

[alsa-devel] [PATCH] ALSA: hda - Update descriptions about new position_fix values
by Takashi Iwai 28 Aug '19

28 Aug '19

The new values for the recent Intel and AMD chips are missing in the documentation. Add the new descriptions. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- Documentation/sound/alsa-configuration.rst | 2 ++ Documentation/sound/hd-audio/notes.rst | 5 +++++ 2 files changed, 7 insertions(+) diff --git a/Documentation/sound/alsa-configuration.rst b/Documentation/sound/alsa-configuration.rst index 4a3cecc8ad38..02aacd69ab96 100644 --- a/Documentation/sound/alsa-configuration.rst +++ b/Documentation/sound/alsa-configuration.rst @@ -1001,6 +1001,8 @@ position_fix 2 = POSBUF: use position buffer, 3 = VIACOMBO: VIA-specific workaround for capture, 4 = COMBO: use LPIB for playback, auto for capture stream + 5 = SKL+: apply the delay calculation available on recent Intel chips + 6 = FIFO: correct the position with the fixed FIFO size, for recent AMD chips probe_mask Bitmask to probe codecs (default = -1, meaning all slots); When the bit 8 (0x100) is set, the lower 8 bits are used diff --git a/Documentation/sound/hd-audio/notes.rst b/Documentation/sound/hd-audio/notes.rst index 9f7347830ba4..0f3109d9abc8 100644 --- a/Documentation/sound/hd-audio/notes.rst +++ b/Documentation/sound/hd-audio/notes.rst @@ -66,6 +66,11 @@ by comparing both LPIB and position-buffer values. ``position_fix=4`` is another combination available for all controllers, and uses LPIB for the playback and the position-buffer for the capture streams. +``position_fix=5`` is specific to Intel platforms, so far, for Skylake +and onward. It applies the delay calculation for the precise position +reporting. +``position_fix=6`` is to correct the position with the fixed FIFO +size, mainly targeted for the recent AMD controllers. 0 is the default value for all other controllers, the automatic check and fallback to LPIB as described in the above. If you get a problem of repeated sounds, this option might -- 2.16.4

1 0

[alsa-devel] [PATCH v2 0/3] ASoC: mediatek: mt8183: fix mt8183-mt6358 tdm out issues
by Jiaxin Yu 28 Aug '19

28 Aug '19

v2: 1. Base on branch asoc-next that merged branch 'asoc-5.4'. 2. Split the PATCH 2/2 that in v1 into three patches. v1: 1. Base on branch asoc-next that merged branch 'asoc-5.3'. 2. Mark suggest the patch would be split up into separate patches, each fixing single issue. 3. PATCH 1/2 has been applied to the asoc tree 'asoc-5.4'. Jiaxin Yu (3): ASoC: mediatek: mt8183: fix tdm out data align issue ASoC: mediatek: mt8183: tdm hw support tdm out and 8ch i2s out ASoC: mediatek: mt8183: fix tdm out data is valid on rising edge sound/soc/mediatek/common/mtk-afe-fe-dai.c | 3 +- sound/soc/mediatek/mt8183/mt8183-dai-tdm.c | 177 ++++++++++++++---- .../mt8183/mt8183-mt6358-ts3a227-max98357.c | 3 + sound/soc/mediatek/mt8183/mt8183-reg.h | 8 +- 4 files changed, 152 insertions(+), 39 deletions(-) -- 2.18.0

2 5

[alsa-devel] [PATCH v2 0/4] ALSA: pcm: add support for 352.8KHz and 384KHz sample rate
by Srinivas Kandagatla 28 Aug '19

28 Aug '19

This patchset adds missing support to 352.8KHz and 384KHz sample rates in Qualcomm WCD9335 codec and QDSP dais. First patch adds these new rates to known list of rates in pcm core and also adds defines in pcm.h so that drivers can use it. Changes since v1: - Added wcd and qdsp users of these new rates. Srinivas Kandagatla (3): ASoC: wcd9335: Fix primary interpolator max rate ASoC: qdsp6: q6afe-dai: Update max rate for slim and tdm dais ASoC: qdsp6: q6asm-dai: fix max rates on q6asm dais Vidyakumar Athota (1): ALSA: pcm: add support for 352.8KHz and 384KHz sample rate include/sound/pcm.h | 5 ++ sound/core/pcm_native.c | 2 +- sound/soc/codecs/wcd9335.c | 20 ++++--- sound/soc/qcom/qdsp6/q6afe-dai.c | 92 +++++++++++--------------------- sound/soc/qcom/qdsp6/q6asm-dai.c | 18 +++---- 5 files changed, 59 insertions(+), 78 deletions(-) -- 2.21.0

3 10

[alsa-devel] [PATCH 1/5] dt-bindings: sound: sun4i-spdif: Fix dma-names warning
by Maxime Ripard 28 Aug '19

28 Aug '19

From: Maxime Ripard <maxime.ripard(a)bootlin.com> Even though the H6 compatible has been properly added, the exeption for the number of DMA channels hasn't been updated, leading in a validation warning. Fix this. Fixes: b20453031472 ("dt-bindings: sound: sun4i-spdif: Add Allwinner H6 compatible") Signed-off-by: Maxime Ripard <maxime.ripard(a)bootlin.com> --- .../devicetree/bindings/sound/allwinner,sun4i-a10-spdif.yaml | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/allwinner,sun4i-a10-spdif.yaml b/Documentation/devicetree/bindings/sound/allwinner,sun4i-a10-spdif.yaml index e0284d8c3b63..38d4cede0860 100644 --- a/Documentation/devicetree/bindings/sound/allwinner,sun4i-a10-spdif.yaml +++ b/Documentation/devicetree/bindings/sound/allwinner,sun4i-a10-spdif.yaml @@ -70,7 +70,9 @@ allOf: properties: compatible: contains: - const: allwinner,sun8i-h3-spdif + enum: + - allwinner,sun8i-h3-spdif + - allwinner,sun50i-h6-spdif then: properties: -- 2.21.0

3 7

[alsa-devel] [PATCH 2/2] ASoC: rt1011: ADCDAT pin config modification
by shumingf＠realtek.com 28 Aug '19

28 Aug '19

From: Shuming Fan <shumingf(a)realtek.com> ADCDAT pin config sets to output mode in default. And the driver creates the kcontrol to control ADCDAT pin configuration. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/rt1011.c | 18 +++++++++++++++--- 1 file changed, 15 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/rt1011.c b/sound/soc/codecs/rt1011.c index f9235764ad32..52c17450d5a0 100644 --- a/sound/soc/codecs/rt1011.c +++ b/sound/soc/codecs/rt1011.c @@ -1029,6 +1029,14 @@ static SOC_ENUM_SINGLE_DECL(rt1011_tdm_adc1_1_enum, RT1011_TDM1_SET_3, 6, static SOC_ENUM_SINGLE_DECL(rt1011_tdm_adc2_1_enum, RT1011_TDM1_SET_3, 4, rt1011_tdm_adc_swap_select); +static const char * const rt1011_adcdat_pin_config_select[] = { + "Output", "Input" +}; +static SOC_ENUM_SINGLE_DECL(rt1011_adcdat1_pin_config_enum, RT1011_TDM_TOTAL_SET, 4, + rt1011_adcdat_pin_config_select); +static SOC_ENUM_SINGLE_DECL(rt1011_adcdat2_pin_config_enum, RT1011_TDM_TOTAL_SET, 3, + rt1011_adcdat_pin_config_select); + static void rt1011_reset(struct regmap *regmap) { regmap_write(regmap, RT1011_RESET, 0); @@ -1343,6 +1351,8 @@ static const struct snd_kcontrol_new rt1011_snd_controls[] = { SOC_ENUM("I2S ADC DOUT Mode", rt1011_adc_dout_mode_enum), SOC_ENUM("TDM1 DOUT Length", rt1011_tdm1_dout_len_enum), SOC_ENUM("TDM2 DOUT Length", rt1011_tdm2_dout_len_enum), + SOC_ENUM("ADCDAT1 Pin Config", rt1011_adcdat1_pin_config_enum), + SOC_ENUM("ADCDAT2 Pin Config", rt1011_adcdat2_pin_config_enum), /* Speaker/Receiver Mode */ SOC_SINGLE_EXT("RECV SPK Mode", SND_SOC_NOPM, 0, 1, 0, @@ -1859,9 +1869,6 @@ static int rt1011_set_tdm_slot(struct snd_soc_dai *dai, RT1011_TDM_I2S_DOCK_EN_1_MASK, tdm_en); snd_soc_component_update_bits(component, RT1011_TDM2_SET_2, RT1011_TDM_I2S_DOCK_EN_2_MASK, tdm_en); - snd_soc_component_update_bits(component, RT1011_TDM_TOTAL_SET, - RT1011_ADCDAT1_PIN_CONFIG | RT1011_ADCDAT2_PIN_CONFIG, - RT1011_ADCDAT1_OUTPUT | RT1011_ADCDAT2_OUTPUT); snd_soc_dapm_mutex_unlock(dapm); return ret; @@ -2207,6 +2214,11 @@ static void rt1011_calibration_work(struct work_struct *work) /* initial */ rt1011_reg_init(component); + + /* set ADCDAT to output mode */ + snd_soc_component_update_bits(component, RT1011_TDM_TOTAL_SET, + RT1011_ADCDAT1_PIN_CONFIG | RT1011_ADCDAT2_PIN_CONFIG, + RT1011_ADCDAT1_OUTPUT | RT1011_ADCDAT2_OUTPUT); } static int rt1011_i2c_probe(struct i2c_client *i2c, -- 2.23.0

2 1

[alsa-devel] [PATCH] ASoC: Intel: Baytrail: Fix implicit fallthrough warning
by Cezary Rojewski 28 Aug '19

28 Aug '19

Separate _SUSPEND from _PAUSE_PUSH to ensure code is handled correctly while fixing warning reported during compilation. Fixes: b80d19c166c4 ("ASoC: Intel: Restore Baytrail ADSP streams only when ADSP was in reset") Signed-off-by: Cezary Rojewski <cezary.rojewski(a)intel.com> --- sound/soc/intel/baytrail/sst-baytrail-pcm.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/intel/baytrail/sst-baytrail-pcm.c b/sound/soc/intel/baytrail/sst-baytrail-pcm.c index 9cbc982d46a9..eddb24ab8072 100644 --- a/sound/soc/intel/baytrail/sst-baytrail-pcm.c +++ b/sound/soc/intel/baytrail/sst-baytrail-pcm.c @@ -193,6 +193,8 @@ static int sst_byt_pcm_trigger(struct snd_pcm_substream *substream, int cmd) break; case SNDRV_PCM_TRIGGER_SUSPEND: pdata->restore_stream = false; + sst_byt_stream_pause(byt, pcm_data->stream); + break; case SNDRV_PCM_TRIGGER_PAUSE_PUSH: sst_byt_stream_pause(byt, pcm_data->stream); break; -- 2.17.1

2 2

[alsa-devel] [PATCH v2 1/2] ASoC: sun4i-i2s: Revert "ASoC: sun4i-i2s: Remove duplicated quirks structure"
by Maxime Ripard 28 Aug '19

28 Aug '19

From: Maxime Ripard <maxime.ripard(a)bootlin.com> This reverts commit 3e9acd7ac6933cdc20c441bbf9a38ed9e42e1490. It turns out that while one I2S controller is described in the A83t datasheet, the driver supports another, undocumented, one that has been inherited from the older SoCs, while the documented one uses the new design. Fixes: 3e9acd7ac693 ("ASoC: sun4i-i2s: Remove duplicated quirks structure") Signed-off-by: Maxime Ripard <maxime.ripard(a)bootlin.com> --- Changes from v1: - Add the proper prefix to the commit title --- sound/soc/sunxi/sun4i-i2s.c | 25 ++++++++++++++++++++++++- 1 file changed, 24 insertions(+), 1 deletion(-) diff --git a/sound/soc/sunxi/sun4i-i2s.c b/sound/soc/sunxi/sun4i-i2s.c index 57bf2a33753e..a6a3f772fdf0 100644 --- a/sound/soc/sunxi/sun4i-i2s.c +++ b/sound/soc/sunxi/sun4i-i2s.c @@ -1097,6 +1097,11 @@ static const struct sun4i_i2s_quirks sun6i_a31_i2s_quirks = { .set_fmt = sun4i_i2s_set_soc_fmt, }; +/* + * This doesn't describe the TDM controller documented in the A83t + * datasheet, but the three undocumented I2S controller that use the + * older design. + */ static const struct sun4i_i2s_quirks sun8i_a83t_i2s_quirks = { .has_reset = true, .reg_offset_txdata = SUN8I_I2S_FIFO_TX_REG, @@ -1115,6 +1120,24 @@ static const struct sun4i_i2s_quirks sun8i_a83t_i2s_quirks = { .set_fmt = sun8i_i2s_set_soc_fmt, }; +static const struct sun4i_i2s_quirks sun8i_h3_i2s_quirks = { + .has_reset = true, + .reg_offset_txdata = SUN8I_I2S_FIFO_TX_REG, + .sun4i_i2s_regmap = &sun8i_i2s_regmap_config, + .field_clkdiv_mclk_en = REG_FIELD(SUN4I_I2S_CLK_DIV_REG, 8, 8), + .field_fmt_wss = REG_FIELD(SUN4I_I2S_FMT0_REG, 0, 2), + .field_fmt_sr = REG_FIELD(SUN4I_I2S_FMT0_REG, 4, 6), + .bclk_dividers = sun8i_i2s_clk_div, + .num_bclk_dividers = ARRAY_SIZE(sun8i_i2s_clk_div), + .mclk_dividers = sun8i_i2s_clk_div, + .num_mclk_dividers = ARRAY_SIZE(sun8i_i2s_clk_div), + .get_bclk_parent_rate = sun8i_i2s_get_bclk_parent_rate, + .get_sr = sun8i_i2s_get_sr_wss, + .get_wss = sun8i_i2s_get_sr_wss, + .set_chan_cfg = sun8i_i2s_set_chan_cfg, + .set_fmt = sun8i_i2s_set_soc_fmt, +}; + static const struct sun4i_i2s_quirks sun50i_a64_codec_i2s_quirks = { .has_reset = true, .reg_offset_txdata = SUN8I_I2S_FIFO_TX_REG, @@ -1296,7 +1319,7 @@ static const struct of_device_id sun4i_i2s_match[] = { }, { .compatible = "allwinner,sun8i-h3-i2s", - .data = &sun8i_a83t_i2s_quirks, + .data = &sun8i_h3_i2s_quirks, }, { .compatible = "allwinner,sun50i-a64-codec-i2s", -- 2.21.0

3 3

[alsa-devel] [PATCH 1/2] ALSA: usb-audio: DSD auto-detection for Playback Designs
by Jussi Laako 28 Aug '19

28 Aug '19

Add DSD support auto-detection for newer Playback Designs devices. Older device generations have a different USB interface implementation. Keep the auto-detection VID whitelist sorted. Signed-off-by: Jussi Laako <jussi(a)sonarnerd.net> --- sound/usb/quirks.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 7e9735aa7ac9..79fb24843d5b 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1566,7 +1566,8 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip, struct usb_interface *iface; /* Playback Designs */ - if (USB_ID_VENDOR(chip->usb_id) == 0x23ba) { + if (USB_ID_VENDOR(chip->usb_id) == 0x23ba && + USB_ID_PRODUCT(chip->usb_id) < 0x0110) { switch (fp->altsetting) { case 1: fp->dsd_dop = true; @@ -1654,8 +1655,9 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip, * from XMOS/Thesycon */ switch (USB_ID_VENDOR(chip->usb_id)) { - case 0x20b1: /* XMOS based devices */ case 0x152a: /* Thesycon devices */ + case 0x20b1: /* XMOS based devices */ + case 0x23ba: /* Playback Designs */ case 0x25ce: /* Mytek devices */ case 0x2ab6: /* T+A devices */ if (fp->dsd_raw) -- 2.17.1

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[alsa-devel] [BUG] bdw-rt5650 DSP boot timeout
by Jon Flatley 28 Aug '19

28 Aug '19

I've been working on upstreaming the bdw-rt5650 machine driver for the Acer Chromebase 24 (buddy). There seems to be an issue when first setting the hardware controls that appears to be crashing the DSP: [ 51.424554] haswell-pcm-audio haswell-pcm-audio: FW loaded, mailbox readback FW info: type 01, - version: 00.00, build 77, source commit id: 876ac6906f31a43b6772b23c7c983ce9dcb18a19 ... [ 84.924666] haswell-pcm-audio haswell-pcm-audio: error: audio DSP boot timeout IPCD 0x0 IPCX 0x0 [ 85.260655] haswell-pcm-audio haswell-pcm-audio: ipc: --message timeout-- ipcx 0x83000000 isr 0x00000000 ipcd 0x00000000 imrx 0x7fff0000 [ 85.273609] haswell-pcm-audio haswell-pcm-audio: error: stream commit failed [ 85.279746] System PCM: error: failed to commit stream -110 [ 85.285388] haswell-pcm-audio haswell-pcm-audio: ASoC: haswell-pcm-audio hw params failed: -110 [ 85.293963] System PCM: ASoC: hw_params FE failed -110 This happens roughly 50% of the time when first setting hardware controls after a reboot. The other 50% of the time the DSP comes up just fine and audio works fine thereafter. Adding "#define DEBUG 1" to sound/soc/intel/haswell/sst-haswell-ipc.c makes the issue occur much less frequently in my testing. Seems like a subtle timing issue. There were timing issues encountered during the bringup of the 2015 chromebook pixel (samus) which uses the bdw-rt5677 machine driver. Those were slightly different, and manifested during repeated arecords. Both devices use the same revision of the sst2 firmware. Any ideas for how to debug this? Thanks, Jon

7 20

[alsa-devel] [RESEND PATCH v4 0/4] ASoC: codecs: Add WSA881x Smart Speaker amplifier support
by Srinivas Kandagatla 28 Aug '19

28 Aug '19

Resending this series due to a typo in yaml filename! Thanks for reviewing v3 patchset, here is v4 with addressing the comments in v3 This patchset adds support to WSA8810/WSA8815 Class-D Smart Speaker Amplifier which is SoundWire interfaced. This also adds support to some missing bits in SoundWire bus layer like Device Tree support. This patchset along with DB845c machine driver and WCD934x codec driver has been tested on SDM845 SoC based DragonBoard DB845c with two WSA8810 speakers. Most of the code in this driver is rework of Qualcomm downstream drivers used in Andriod. Credits to Banajit Goswami and Patrick Lai's Team. TODO: Add thermal sensor support in WSA881x. Thanks, srini Changes since v3: - updated slave bindings according to Rob's Suggestion. - moved bindings to yaml Srinivas Kandagatla (4): dt-bindings: soundwire: add slave bindings soundwire: core: add device tree support for slave devices dt-bindings: ASoC: Add WSA881x bindings ASoC: codecs: add wsa881x amplifier support .../bindings/sound/qcom,wsa881x.yaml | 44 + .../soundwire/soundwire-controller.yaml | 75 ++ drivers/soundwire/bus.c | 2 + drivers/soundwire/bus.h | 1 + drivers/soundwire/slave.c | 52 + sound/soc/codecs/Kconfig | 10 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/wsa881x.c | 1134 +++++++++++++++++ 8 files changed, 1320 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/qcom,wsa881x.yaml create mode 100644 Documentation/devicetree/bindings/soundwire/soundwire-controller.yaml create mode 100644 sound/soc/codecs/wsa881x.c -- 2.21.0

5 17

[alsa-devel] [PATCH -next] ASoC: SOF: imx8: Make some functions static
by YueHaibing 27 Aug '19

27 Aug '19

Fix sparse warnings: sound/soc/sof/imx/imx8.c:104:6: warning: symbol 'imx8_dsp_handle_reply' was not declared. Should it be static? sound/soc/sof/imx/imx8.c:115:6: warning: symbol 'imx8_dsp_handle_request' was not declared. Should it be static? sound/soc/sof/imx/imx8.c:336:5: warning: symbol 'imx8_get_bar_index' was not declared. Should it be static? sound/soc/sof/imx/imx8.c:341:6: warning: symbol 'imx8_ipc_msg_data' was not declared. Should it be static? sound/soc/sof/imx/imx8.c:348:5: warning: symbol 'imx8_ipc_pcm_params' was not declared. Should it be static? Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/soc/sof/imx/imx8.c | 18 +++++++++--------- 1 file changed, 9 insertions(+), 9 deletions(-) diff --git a/sound/soc/sof/imx/imx8.c b/sound/soc/sof/imx/imx8.c index e502f58..6404724 100644 --- a/sound/soc/sof/imx/imx8.c +++ b/sound/soc/sof/imx/imx8.c @@ -101,7 +101,7 @@ static int imx8_get_window_offset(struct snd_sof_dev *sdev, u32 id) return MBOX_OFFSET; } -void imx8_dsp_handle_reply(struct imx_dsp_ipc *ipc) +static void imx8_dsp_handle_reply(struct imx_dsp_ipc *ipc) { struct imx8_priv *priv = imx_dsp_get_data(ipc); unsigned long flags; @@ -112,7 +112,7 @@ void imx8_dsp_handle_reply(struct imx_dsp_ipc *ipc) spin_unlock_irqrestore(&priv->sdev->ipc_lock, flags); } -void imx8_dsp_handle_request(struct imx_dsp_ipc *ipc) +static void imx8_dsp_handle_request(struct imx_dsp_ipc *ipc) { struct imx8_priv *priv = imx_dsp_get_data(ipc); @@ -333,21 +333,21 @@ static int imx8_remove(struct snd_sof_dev *sdev) } /* on i.MX8 there is 1 to 1 match between type and BAR idx */ -int imx8_get_bar_index(struct snd_sof_dev *sdev, u32 type) +static int imx8_get_bar_index(struct snd_sof_dev *sdev, u32 type) { return type; } -void imx8_ipc_msg_data(struct snd_sof_dev *sdev, - struct snd_pcm_substream *substream, - void *p, size_t sz) +static void imx8_ipc_msg_data(struct snd_sof_dev *sdev, + struct snd_pcm_substream *substream, + void *p, size_t sz) { sof_mailbox_read(sdev, sdev->dsp_box.offset, p, sz); } -int imx8_ipc_pcm_params(struct snd_sof_dev *sdev, - struct snd_pcm_substream *substream, - const struct sof_ipc_pcm_params_reply *reply) +static int imx8_ipc_pcm_params(struct snd_sof_dev *sdev, + struct snd_pcm_substream *substream, + const struct sof_ipc_pcm_params_reply *reply) { return 0; } -- 2.7.4

3 2

[alsa-devel] [PATCH] ASoC: imx-audmix: register the card on a proper dev
by Shengjiu Wang 27 Aug '19

27 Aug '19

This platform device is registered from "fsl_audmix", which is its parent device. If use pdev->dev.parent for the priv->card.dev, the value set by dev_set_drvdata in parent device will be covered by the value in child device. Fixes: b86ef5367761 ("ASoC: fsl: Add Audio Mixer machine driver") Signed-off-by: Viorel Suman <viorel.suman(a)nxp.com> Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/imx-audmix.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/imx-audmix.c b/sound/soc/fsl/imx-audmix.c index 9e1cb18859ce..71590ca6394b 100644 --- a/sound/soc/fsl/imx-audmix.c +++ b/sound/soc/fsl/imx-audmix.c @@ -325,14 +325,14 @@ static int imx_audmix_probe(struct platform_device *pdev) priv->card.num_configs = priv->num_dai_conf; priv->card.dapm_routes = priv->dapm_routes; priv->card.num_dapm_routes = priv->num_dapm_routes; - priv->card.dev = pdev->dev.parent; + priv->card.dev = &pdev->dev; priv->card.owner = THIS_MODULE; priv->card.name = "imx-audmix"; platform_set_drvdata(pdev, &priv->card); snd_soc_card_set_drvdata(&priv->card, priv); - ret = devm_snd_soc_register_card(pdev->dev.parent, &priv->card); + ret = devm_snd_soc_register_card(&pdev->dev, &priv->card); if (ret) { dev_err(&pdev->dev, "snd_soc_register_card failed\n"); return ret; -- 2.21.0

3 2

[alsa-devel] Applied "ASoC: sunxi: Revert initial A83t support" to the asoc tree
by Mark Brown 27 Aug '19

27 Aug '19

The patch ASoC: sunxi: Revert initial A83t support 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 455b1d42e82c6027b9763f0055b54e45ff6cd7fd Mon Sep 17 00:00:00 2001 From: Maxime Ripard <maxime.ripard(a)bootlin.com> Date: Tue, 27 Aug 2019 11:32:05 +0200 Subject: [PATCH] ASoC: sunxi: Revert initial A83t support This reverts commit 3e9acd7ac6933c (ASoC: sun4i-i2s: Remove duplicated quirks structure"). It turns out that while one I2S controller is described in the A83t datasheet, the driver supports another, undocumented, one that has been inherited from the older SoCs, while the documented one uses the new design. Fixes: 3e9acd7ac693 ("ASoC: sun4i-i2s: Remove duplicated quirks structure") Signed-off-by: Maxime Ripard <maxime.ripard(a)bootlin.com> Link: https://lore.kernel.org/r/20190827093206.17919-1-mripard@kernel.org Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/sunxi/sun4i-i2s.c | 25 ++++++++++++++++++++++++- 1 file changed, 24 insertions(+), 1 deletion(-) diff --git a/sound/soc/sunxi/sun4i-i2s.c b/sound/soc/sunxi/sun4i-i2s.c index 57bf2a33753e..a6a3f772fdf0 100644 --- a/sound/soc/sunxi/sun4i-i2s.c +++ b/sound/soc/sunxi/sun4i-i2s.c @@ -1097,6 +1097,11 @@ static const struct sun4i_i2s_quirks sun6i_a31_i2s_quirks = { .set_fmt = sun4i_i2s_set_soc_fmt, }; +/* + * This doesn't describe the TDM controller documented in the A83t + * datasheet, but the three undocumented I2S controller that use the + * older design. + */ static const struct sun4i_i2s_quirks sun8i_a83t_i2s_quirks = { .has_reset = true, .reg_offset_txdata = SUN8I_I2S_FIFO_TX_REG, @@ -1115,6 +1120,24 @@ static const struct sun4i_i2s_quirks sun8i_a83t_i2s_quirks = { .set_fmt = sun8i_i2s_set_soc_fmt, }; +static const struct sun4i_i2s_quirks sun8i_h3_i2s_quirks = { + .has_reset = true, + .reg_offset_txdata = SUN8I_I2S_FIFO_TX_REG, + .sun4i_i2s_regmap = &sun8i_i2s_regmap_config, + .field_clkdiv_mclk_en = REG_FIELD(SUN4I_I2S_CLK_DIV_REG, 8, 8), + .field_fmt_wss = REG_FIELD(SUN4I_I2S_FMT0_REG, 0, 2), + .field_fmt_sr = REG_FIELD(SUN4I_I2S_FMT0_REG, 4, 6), + .bclk_dividers = sun8i_i2s_clk_div, + .num_bclk_dividers = ARRAY_SIZE(sun8i_i2s_clk_div), + .mclk_dividers = sun8i_i2s_clk_div, + .num_mclk_dividers = ARRAY_SIZE(sun8i_i2s_clk_div), + .get_bclk_parent_rate = sun8i_i2s_get_bclk_parent_rate, + .get_sr = sun8i_i2s_get_sr_wss, + .get_wss = sun8i_i2s_get_sr_wss, + .set_chan_cfg = sun8i_i2s_set_chan_cfg, + .set_fmt = sun8i_i2s_set_soc_fmt, +}; + static const struct sun4i_i2s_quirks sun50i_a64_codec_i2s_quirks = { .has_reset = true, .reg_offset_txdata = SUN8I_I2S_FIFO_TX_REG, @@ -1296,7 +1319,7 @@ static const struct of_device_id sun4i_i2s_match[] = { }, { .compatible = "allwinner,sun8i-h3-i2s", - .data = &sun8i_a83t_i2s_quirks, + .data = &sun8i_h3_i2s_quirks, }, { .compatible = "allwinner,sun50i-a64-codec-i2s", -- 2.20.1

1 0

[alsa-devel] [PATCH] ASoC: mediatek: mt8183: remove unused DAPM-related variables
by Tzung-Bi Shih 27 Aug '19

27 Aug '19

Remove unused variable mt8183_mt6358_ts3a227_max98357_dapm_widgets and mt8183_mt6358_ts3a227_max98357_dapm_routes. They are accidentially included when rebasing commits. Fixes: 6191cbde5ff0 ("ASoC: mediatek: mt8183: switch tdm pins gpio function when playback on or off") Signed-off-by: Tzung-Bi Shih <tzungbi(a)google.com> --- .../mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c | 10 ---------- 1 file changed, 10 deletions(-) diff --git a/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c b/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c index fe8a7461ace5..951706fa75fc 100644 --- a/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c +++ b/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c @@ -46,16 +46,6 @@ static int mt8183_i2s_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, return 0; } -static const struct snd_soc_dapm_widget -mt8183_mt6358_ts3a227_max98357_dapm_widgets[] = { - SND_SOC_DAPM_OUTPUT("IT6505_8CH"), -}; - -static const struct snd_soc_dapm_route -mt8183_mt6358_ts3a227_max98357_dapm_routes[] = { - {"IT6505_8CH", NULL, "TDM"}, -}; - enum PINCTRL_PIN_STATE { PIN_STATE_DEFAULT = 0, PIN_TDM_OUT_ON, -- 2.23.0.187.g17f5b7556c-goog

2 1

[alsa-devel] [PATCH -next] ASoC: SOF: imx8: Fix return value check in imx8_probe()
by Wei Yongjun 27 Aug '19

27 Aug '19

In case of error, the function devm_ioremap_wc() returns NULL pointer not ERR_PTR(). The IS_ERR() test in the return value check should be replaced with NULL test. Fixes: 202acc565a1f ("ASoC: SOF: imx: Add i.MX8 HW support") Signed-off-by: Wei Yongjun <weiyongjun1(a)huawei.com> --- sound/soc/sof/imx/imx8.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/sof/imx/imx8.c b/sound/soc/sof/imx/imx8.c index e502f584207f..263d4df35fe8 100644 --- a/sound/soc/sof/imx/imx8.c +++ b/sound/soc/sof/imx/imx8.c @@ -296,10 +296,10 @@ static int imx8_probe(struct snd_sof_dev *sdev) sdev->bar[SOF_FW_BLK_TYPE_SRAM] = devm_ioremap_wc(sdev->dev, res.start, res.end - res.start + 1); - if (IS_ERR(sdev->bar[SOF_FW_BLK_TYPE_SRAM])) { + if (!sdev->bar[SOF_FW_BLK_TYPE_SRAM]) { dev_err(sdev->dev, "failed to ioremap mem 0x%x size 0x%x\n", base, size); - ret = PTR_ERR(sdev->bar[SOF_FW_BLK_TYPE_SRAM]); + ret = -ENOMEM; goto exit_pdev_unregister; } sdev->mailbox_bar = SOF_FW_BLK_TYPE_SRAM;

3 2

[alsa-devel] [PATCH] ASoC: SOF: imx8: Fix an is IS_ERR() vs NULL check
by Dan Carpenter 27 Aug '19

27 Aug '19

The device_link_add() function only returns NULL on error, it doesn't return error pointers. Fixes: 202acc565a1f ("ASoC: SOF: imx: Add i.MX8 HW support") Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- sound/soc/sof/imx/imx8.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/sof/imx/imx8.c b/sound/soc/sof/imx/imx8.c index e502f584207f..41ad3a310808 100644 --- a/sound/soc/sof/imx/imx8.c +++ b/sound/soc/sof/imx/imx8.c @@ -227,8 +227,8 @@ static int imx8_probe(struct snd_sof_dev *sdev) DL_FLAG_STATELESS | DL_FLAG_PM_RUNTIME | DL_FLAG_RPM_ACTIVE); - if (IS_ERR(priv->link[i])) { - ret = PTR_ERR(priv->link[i]); + if (!priv->link[i]) { + ret = -ENOMEM; dev_pm_domain_detach(priv->pd_dev[i], false); goto exit_unroll_pm; } -- 2.20.1

3 2

[alsa-devel] [PATCH v2 1/3] ASoC: mediatek: mt8183: fix tdm out data align issue
by Jiaxin Yu 27 Aug '19

27 Aug '19

Mt8183 tdm out support S16_LE/S24_LE/S32_LE formats. When output S32_LE, we need set hd_align so that memif can output MSB 24bits. When output S24_LE, we need reset hd_align so that memif can output LSB 24bits. Signed-off-by: Jiaxin Yu <jiaxin.yu(a)mediatek.com> --- sound/soc/mediatek/common/mtk-afe-fe-dai.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/mediatek/common/mtk-afe-fe-dai.c b/sound/soc/mediatek/common/mtk-afe-fe-dai.c index d16563408465..10ea4fdbeb1e 100644 --- a/sound/soc/mediatek/common/mtk-afe-fe-dai.c +++ b/sound/soc/mediatek/common/mtk-afe-fe-dai.c @@ -241,7 +241,7 @@ int mtk_afe_fe_prepare(struct snd_pcm_substream *substream, struct mtk_base_afe *afe = snd_soc_dai_get_drvdata(dai); struct mtk_base_afe_memif *memif = &afe->memif[rtd->cpu_dai->id]; int hd_audio = 0; - int hd_align = 1; + int hd_align = 0; /* set hd mode */ switch (substream->runtime->format) { @@ -254,7 +254,6 @@ int mtk_afe_fe_prepare(struct snd_pcm_substream *substream, break; case SNDRV_PCM_FORMAT_S24_LE: hd_audio = 1; - hd_align = 0; break; default: dev_err(afe->dev, "%s() error: unsupported format %d\n", -- 2.18.0

2 1

[alsa-devel] [PATCH 1/2] ASoC: rt1011: add mutex protection to set_fmt/set_tdm_slot
by shumingf＠realtek.com 27 Aug '19

27 Aug '19

From: Shuming Fan <shumingf(a)realtek.com> The calibration process at booting will reset registers and bypass cache to make sure the calibration is done. We add mutex protection to avoid unexpected settings while the registration process and calibration are interleaved. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/rt1011.c | 27 +++++++++++++++++++-------- 1 file changed, 19 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/rt1011.c b/sound/soc/codecs/rt1011.c index fa34565a3938..f9235764ad32 100644 --- a/sound/soc/codecs/rt1011.c +++ b/sound/soc/codecs/rt1011.c @@ -1627,14 +1627,18 @@ static int rt1011_hw_params(struct snd_pcm_substream *substream, static int rt1011_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) { struct snd_soc_component *component = dai->component; + struct snd_soc_dapm_context *dapm = + snd_soc_component_get_dapm(component); unsigned int reg_val = 0, reg_bclk_inv = 0; + int ret = 0; + snd_soc_dapm_mutex_lock(dapm); switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBS_CFS: reg_val |= RT1011_I2S_TDM_MS_S; break; default: - return -EINVAL; + ret = -EINVAL; } switch (fmt & SND_SOC_DAIFMT_INV_MASK) { @@ -1644,7 +1648,7 @@ static int rt1011_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) reg_bclk_inv |= RT1011_TDM_INV_BCLK; break; default: - return -EINVAL; + ret = -EINVAL; } switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { @@ -1660,7 +1664,7 @@ static int rt1011_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) reg_val |= RT1011_I2S_TDM_DF_PCM_B; break; default: - return -EINVAL; + ret = -EINVAL; } switch (dai->id) { @@ -1675,9 +1679,11 @@ static int rt1011_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) break; default: dev_err(component->dev, "Invalid dai->id: %d\n", dai->id); - return -EINVAL; + ret = -EINVAL; } - return 0; + + snd_soc_dapm_mutex_unlock(dapm); + return ret; } static int rt1011_set_component_sysclk(struct snd_soc_component *component, @@ -1796,8 +1802,12 @@ static int rt1011_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width) { struct snd_soc_component *component = dai->component; + struct snd_soc_dapm_context *dapm = + snd_soc_component_get_dapm(component); unsigned int val = 0, tdm_en = 0; + int ret = 0; + snd_soc_dapm_mutex_lock(dapm); if (rx_mask || tx_mask) tdm_en = RT1011_TDM_I2S_DOCK_EN_1; @@ -1817,7 +1827,7 @@ static int rt1011_set_tdm_slot(struct snd_soc_dai *dai, case 2: break; default: - return -EINVAL; + ret = -EINVAL; } switch (slot_width) { @@ -1836,7 +1846,7 @@ static int rt1011_set_tdm_slot(struct snd_soc_dai *dai, case 16: break; default: - return -EINVAL; + ret = -EINVAL; } snd_soc_component_update_bits(component, RT1011_TDM1_SET_1, @@ -1853,7 +1863,8 @@ static int rt1011_set_tdm_slot(struct snd_soc_dai *dai, RT1011_ADCDAT1_PIN_CONFIG | RT1011_ADCDAT2_PIN_CONFIG, RT1011_ADCDAT1_OUTPUT | RT1011_ADCDAT2_OUTPUT); - return 0; + snd_soc_dapm_mutex_unlock(dapm); + return ret; } static int rt1011_probe(struct snd_soc_component *component) -- 2.23.0

2 1

[alsa-devel] [PATCH v2 1/2] ASoC: es8316: fix headphone mixer volume table
by Katsuhiro Suzuki 27 Aug '19

27 Aug '19

This patch fix setting table of Headphone mixer volume. Current code uses 4 ... 7 values but these values are prohibited. Correct settings are the following: 0000 -12dB 0001 -10.5dB 0010 -9dB 0011 -7.5dB 0100 -6dB 1000 -4.5dB 1001 -3dB 1010 -1.5dB 1011 0dB Signed-off-by: Katsuhiro Suzuki <katsuhiro(a)katsuster.net> --- sound/soc/codecs/es8316.c | 7 +++++-- 1 file changed, 5 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/es8316.c b/sound/soc/codecs/es8316.c index 8dfb5dbeebbf..f97d9c5210f0 100644 --- a/sound/soc/codecs/es8316.c +++ b/sound/soc/codecs/es8316.c @@ -53,7 +53,10 @@ static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(adc_vol_tlv, -9600, 50, 1); static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(alc_max_gain_tlv, -650, 150, 0); static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(alc_min_gain_tlv, -1200, 150, 0); static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(alc_target_tlv, -1650, 150, 0); -static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(hpmixer_gain_tlv, -1200, 150, 0); +static const SNDRV_CTL_TLVD_DECLARE_DB_RANGE(hpmixer_gain_tlv, + 0, 4, TLV_DB_SCALE_ITEM(-1200, 150, 0), + 8, 11, TLV_DB_SCALE_ITEM(-450, 150, 0), +); static const SNDRV_CTL_TLVD_DECLARE_DB_RANGE(adc_pga_gain_tlv, 0, 0, TLV_DB_SCALE_ITEM(-350, 0, 0), @@ -91,7 +94,7 @@ static const struct snd_kcontrol_new es8316_snd_controls[] = { SOC_DOUBLE_TLV("Headphone Playback Volume", ES8316_CPHP_ICAL_VOL, 4, 0, 3, 1, hpout_vol_tlv), SOC_DOUBLE_TLV("Headphone Mixer Volume", ES8316_HPMIX_VOL, - 0, 4, 7, 0, hpmixer_gain_tlv), + 0, 4, 11, 0, hpmixer_gain_tlv), SOC_ENUM("Playback Polarity", dacpol), SOC_DOUBLE_R_TLV("DAC Playback Volume", ES8316_DAC_VOLL, -- 2.23.0.rc1

3 5

[alsa-devel] [PATCH 00/35] ASoC: Intel: Clenaup SST initialization
by Cezary Rojewski 27 Aug '19

27 Aug '19

This segment targets initialization procedures for all SST drivers. For legacy: - it concentrates on updating ACPI loader so generic DSP framework is not fed any platform specific data. sst-acpi module ceases to exist and is replaced by hsw-acpi and byt-acpi For cAVS (Skylake+): - Repairs what is currently an initialization mess, given the order of invocation of engaged handlers and sheer amount of them. - provides interface to offload hardware-specifics away from driver Following is the order of Skylake initialization currently: - skl_probe -> schedule skl_probe_work - skl_first_init - skl_create -> skl_init_dsp --> skl/ bxt/ cnl_sst_dsp_init ---> skl_sst_ctx_init ----> skl_dsp_ctx_init -----> sst_dsp_new ------> sst_ops::init (invoked but unimplemented!) Listing all the types engaged together with the naming chosen for the above paints an even darker picture. Code is unreadable and hides initialization for diffenret members is various handlers. Moreover, due to existing ill relationship between skl_dev, sst_dsp and sst_generic_ipc, one must be extra careful when accessing so called "dsp/ thread_context" as they all don't get initialized immediately before object is yield for further processing. However, 100/100 series is nonsense and thus cleanups have been divided into chunks and prioritized. Code seen here is part of new Skylake fundament, located at the very bottom of internal mainline. Said mainline is tested constantly on at least sigle platform from every cAVS bucket (description below). This week, BDW has been added to the CI family and was essential in validating legacy changes. Baytrail platform is still missing. Changes for BYT directly mirror HSW/ BDW but due to current lack of platform were untested. Boards engaged in testing: rt286, rt298, rt274. !!! IMPORTANT !!! Some upstream FW binaries are not compatible with existing /skylake driver while changes found here (HARDWARE_CONFIG/ FIRMWARE_CONFIG) make use of firmware ability to offload hardware-specifics away from driver. These and more are core part of any cAVS design and are to be implemented and used by host. This too is missing on Linux upstream. As explanined once, five main FW branches are available: - kbl, 1.5 cAVS (supports SKL, KBL, KBL-R, ABL, more..) - apl_auto, 1.5+ cAVS (supports APL, GLK) - cnl, 1.8 cAVS (supports CNL, CFL, WHL, CML) - icl, 2.0 cAVS (supports ICL, LKF) - master, 2.5 cAVS (supports TGL, EHL and everything above) SKL FW binary existing on upstream is a descendant of old spt branch, obsoleted for 4-5 years now. That FW is a stub, quickly replaced by kbl which is to be used on all 1.5 cAVS platforms. The same story goes with bxtn binary, a descendant of apl branch, replaced by apl_auto 2-3 years ago. All vendors, entire validation and development is located on apl_auto. Message: all FW binaries are to be updated as we cannot guarantee these are still functional. Given the fact that all vendors are fed with new binaries on regular basis from all main branches, it's highly probable some scenarios fail with existing FWs. In consequence, linux-firmware patch will be provided with fresh, updated binaries, soon. Once more, please note, we do not support, nor test platforms using obsolete FW binaries. That includes all platforms, not just SKL or APL. Amadeusz Sławiński (1): ASoC: Intel: Skylake: Put FW runtime params defs in one place Cezary Rojewski (34): ASoC: Intel: Skylake: Add FIRMWARE_CONFIG IPC request ASoC: Intel: Skylake: Add HARDWARE_CONFIG IPC request ASoC: Intel: Skylake: Unify firmware loading mechanism ASoC: Intel: Skylake: Reload libraries on D0 entry for CNL ASoC: Intel: Skylake: Unhardcode dsp cores number ASoC: Intel: Skylake: Update interrupt disabling routine ASoC: Intel: Skylake: Inline ipc free operations ASoC: Intel: Skylake: Unify driver cleanup mechanism ASoC: Intel: Relocate irq thread header to sst_ops ASoC: Intel: Merge sst_dsp_device into sst_pdata ASoC: Intel: Skylake: Reuse sst_dsp_free ASoC: Intel: Skylake: Reuse sst_dsp_new ASoC: Intel: Skylake: Remove skl_dsp_acquire_irq ASoC: Intel: Skylake: Use dsp loading functions directly ASoC: Intel: Skylake: Make dsp_ops::stream_tag obsolete ASoC: Intel: Skylake: Remove skl_dsp_loader_ops ASoC: Intel: Skylake: Remove window0 sst_addr fields ASoC: Intel: Skylake: Remove redundant W0 and W1 macros ASoC: Intel: Skylake: Remove redundant SRAM fields ASoC: Intel: Expose ACPI loading members ASoC: Intel: Haswell: Define separate ACPI loader ASoC: Intel: Baytrail: Define separate ACPI loader ASoC: Intel: Refactor probing of ACPI devices ASoC: Intel: Skylake: Simplify skl_sst_ctx_init declaration ASoC: Intel: Skylake: Simplify all sst_dsp_init declarations ASoC: Intel: Skylake: Define platform descriptors ASoC: Intel: Skylake: Update skl_ids table ASoC: Intel: Skylake: Flip SST initialization order ASoC: Intel: Reuse sst_pdata::fw_name field ASoC: Intel: Reuse sst_pdata::fw field ASoC: Intel: Skylake: Remove skl_dsp_ops ASoC: Intel: Skylake: Privatize SST init handlers ASoC: Intel: Skylake: Merge skl_sst_ctx_init into skl_init_dsp ASoC: Intel: Remove obsolete firmware fields sound/soc/intel/Kconfig | 14 +- sound/soc/intel/baytrail/Makefile | 2 + sound/soc/intel/baytrail/acpi.c | 64 +++++ sound/soc/intel/baytrail/sst-baytrail-dsp.c | 2 +- sound/soc/intel/baytrail/sst-baytrail-ipc.c | 13 +- sound/soc/intel/baytrail/sst-baytrail-ipc.h | 2 + sound/soc/intel/common/Makefile | 4 +- .../intel/common/soc-acpi-intel-bxt-match.c | 2 - .../intel/common/soc-acpi-intel-byt-match.c | 2 - .../intel/common/soc-acpi-intel-cnl-match.c | 1 - .../intel/common/soc-acpi-intel-glk-match.c | 3 - .../intel/common/soc-acpi-intel-hda-match.c | 2 - .../common/soc-acpi-intel-hsw-bdw-match.c | 4 - .../intel/common/soc-acpi-intel-icl-match.c | 1 - .../intel/common/soc-acpi-intel-kbl-match.c | 12 - .../intel/common/soc-acpi-intel-skl-match.c | 3 - sound/soc/intel/common/sst-acpi.c | 117 +-------- sound/soc/intel/common/sst-dsp-priv.h | 8 +- sound/soc/intel/common/sst-dsp.h | 37 +-- sound/soc/intel/common/sst-firmware.c | 13 +- sound/soc/intel/haswell/Makefile | 2 + sound/soc/intel/haswell/acpi.c | 78 ++++++ sound/soc/intel/haswell/sst-haswell-dsp.c | 1 + sound/soc/intel/haswell/sst-haswell-ipc.c | 13 +- sound/soc/intel/haswell/sst-haswell-ipc.h | 2 + sound/soc/intel/skylake/bxt-sst.c | 143 ++++------- sound/soc/intel/skylake/cnl-sst-dsp.c | 13 +- sound/soc/intel/skylake/cnl-sst-dsp.h | 15 +- sound/soc/intel/skylake/cnl-sst.c | 142 ++++------- sound/soc/intel/skylake/skl-debug.c | 2 +- sound/soc/intel/skylake/skl-messages.c | 194 ++------------ sound/soc/intel/skylake/skl-pcm.c | 22 +- sound/soc/intel/skylake/skl-sst-cldma.c | 10 +- sound/soc/intel/skylake/skl-sst-dsp.c | 79 ++---- sound/soc/intel/skylake/skl-sst-dsp.h | 55 ++-- sound/soc/intel/skylake/skl-sst-ipc.c | 238 ++++++++++++++++-- sound/soc/intel/skylake/skl-sst-ipc.h | 124 ++++++++- sound/soc/intel/skylake/skl-sst-utils.c | 28 +-- sound/soc/intel/skylake/skl-sst.c | 150 ++++++----- sound/soc/intel/skylake/skl.c | 65 +++-- sound/soc/intel/skylake/skl.h | 26 +- 41 files changed, 877 insertions(+), 831 deletions(-) create mode 100644 sound/soc/intel/baytrail/acpi.c create mode 100644 sound/soc/intel/haswell/acpi.c -- 2.17.1

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[alsa-devel] [PATCH 0/6] Small fixes
by Amadeusz Sławiński 27 Aug '19

27 Aug '19

Series of small fixes: * fixes few issues found during checking code with static checkers * fix few prints * define function in header, like it should be * release topology when done with it Amadeusz Sławiński (6): ASoC: Intel: Skylake: Use correct function to access iomem space ASoC: Intel: Fix use of potentially uninitialized variable ASoC: dapm: Expose snd_soc_dapm_new_control_unlocked properly ASoC: Intel: Skylake: Print module type instead of id ASoC: Intel: Skylake: Release topology when we are done with it ASoC: Intel: NHLT: Fix debug print format include/sound/soc-dapm.h | 3 +++ sound/soc/intel/common/sst-ipc.c | 2 ++ sound/soc/intel/skylake/skl-debug.c | 2 +- sound/soc/intel/skylake/skl-messages.c | 5 +++-- sound/soc/intel/skylake/skl-nhlt.c | 2 +- sound/soc/intel/skylake/skl-topology.c | 20 ++++++++++---------- sound/soc/intel/skylake/skl.h | 1 - sound/soc/soc-topology.c | 6 ------ 8 files changed, 20 insertions(+), 21 deletions(-) -- 2.17.1

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[alsa-devel] [PATCH] ALSA: hda - Allow runtime PM for controller if component notifier is used
by Takashi Iwai 27 Aug '19

27 Aug '19

Currently we disallow the runtime PM of the HD-audio controller if it's bound with HDMI/DP on Nvidia / AMD unless it's for dGPU. This is for keeping the link up to get the proper notification for ELD hotplug. As explained in the commit 37a3a98ef601 ("ALSA: hda - Enable runtime PM only for discrete GPU"), this keep-power-up behavior is rather a stop-gap solution until the ELD notification via audio component. And now we finally got the audio component for these graphics drivers via commit ade49db337a9 ("ALSA: hda/hdmi - Allow audio component for AMD/ATI and Nvidia HDMI"), so it's time to change. This patch makes HD-audio controller again runtime-suspendable when the device gets bound with audio component in HDMI codec driver. For making it easier to access from the codec driver, move the flag into the common hda_bus object instead of hda_intel flag. Also rename it to keep_power, to indicate the actual meaning. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/sound/hda_codec.h | 1 + sound/pci/hda/hda_intel.c | 11 +++++------ sound/pci/hda/hda_intel.h | 1 - sound/pci/hda/patch_hdmi.c | 4 +++- 4 files changed, 9 insertions(+), 8 deletions(-) diff --git a/include/sound/hda_codec.h b/include/sound/hda_codec.h index 871993696c5f..9a0393cf024c 100644 --- a/include/sound/hda_codec.h +++ b/include/sound/hda_codec.h @@ -59,6 +59,7 @@ struct hda_bus { unsigned int in_reset:1; /* during reset operation */ unsigned int no_response_fallback:1; /* don't fallback at RIRB error */ unsigned int bus_probing :1; /* during probing process */ + unsigned int keep_power:1; /* keep power up for notification */ int primary_dig_out_type; /* primary digital out PCM type */ unsigned int mixer_assigned; /* codec addr for mixer name */ diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 4496fce21300..91e71be42fa4 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -356,7 +356,7 @@ enum { */ #ifdef SUPPORT_VGA_SWITCHEROO #define use_vga_switcheroo(chip) ((chip)->use_vga_switcheroo) -#define needs_eld_notify_link(chip) ((chip)->need_eld_notify_link) +#define needs_eld_notify_link(chip) ((chip)->bus.keep_power) #else #define use_vga_switcheroo(chip) 0 #define needs_eld_notify_link(chip) false @@ -1145,7 +1145,7 @@ static int azx_runtime_idle(struct device *dev) return -EBUSY; /* ELD notification gets broken when HD-audio bus is off */ - if (needs_eld_notify_link(hda)) + if (needs_eld_notify_link(chip)) return -EBUSY; return 0; @@ -1256,7 +1256,7 @@ static void setup_vga_switcheroo_runtime_pm(struct azx *chip) struct hda_intel *hda = container_of(chip, struct hda_intel, chip); struct hda_codec *codec; - if (hda->use_vga_switcheroo && !hda->need_eld_notify_link) { + if (hda->use_vga_switcheroo && !needs_eld_notify_link(chip)) { list_for_each_codec(codec, &chip->bus) codec->auto_runtime_pm = 1; /* reset the power save setup */ @@ -1270,10 +1270,9 @@ static void azx_vs_gpu_bound(struct pci_dev *pci, { struct snd_card *card = pci_get_drvdata(pci); struct azx *chip = card->private_data; - struct hda_intel *hda = container_of(chip, struct hda_intel, chip); if (client_id == VGA_SWITCHEROO_DIS) - hda->need_eld_notify_link = 0; + chip->bus.keep_power = 0; setup_vga_switcheroo_runtime_pm(chip); } @@ -1285,7 +1284,7 @@ static void init_vga_switcheroo(struct azx *chip) dev_info(chip->card->dev, "Handle vga_switcheroo audio client\n"); hda->use_vga_switcheroo = 1; - hda->need_eld_notify_link = 1; /* cleared in gpu_bound op */ + chip->bus.keep_power = 1; /* cleared in either gpu_bound op or codec probe */ chip->driver_caps |= AZX_DCAPS_PM_RUNTIME; pci_dev_put(p); } diff --git a/sound/pci/hda/hda_intel.h b/sound/pci/hda/hda_intel.h index 1468865e0342..2acfff3da1a0 100644 --- a/sound/pci/hda/hda_intel.h +++ b/sound/pci/hda/hda_intel.h @@ -25,7 +25,6 @@ struct hda_intel { /* vga_switcheroo setup */ unsigned int use_vga_switcheroo:1; - unsigned int need_eld_notify_link:1; unsigned int vga_switcheroo_registered:1; unsigned int init_failed:1; /* delayed init failed */ diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 933c7bf47ef6..83b8b9d27711 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -2542,8 +2542,10 @@ static void generic_acomp_init(struct hda_codec *codec, spec->port2pin = port2pin; setup_drm_audio_ops(codec, ops); if (!snd_hdac_acomp_init(&codec->bus->core, &spec->drm_audio_ops, - match_bound_vga, 0)) + match_bound_vga, 0)) { spec->acomp_registered = true; + codec->bus->keep_power = 0; + } } /* -- 2.16.4

1 0

[alsa-devel] pulse audio problem
by John Menninger 27 Aug '19

27 Aug '19

In going over my Ubuntu 16.04 (64bit) system log I found the following: "Aug 27 00:02:22 Desktop64 pulseaudio[2726]: [alsa-sink-ALC889A Analog] alsa-util.c: snd_pcm_avail() returned a value that is exceptionally large: 2284120 bytes (12948 ms). "Aug 27 00:02:29 Desktop64 pulseaudio[2726]: [alsa-sink-ALC889A Analog] alsa-util.c: Most likely this is a bug in the ALSA driver 'snd_hda_intel'. Please report this issue to the ALSA developers." I have no idea how to manage any problem that might have caused this notification and I would appreciate any advice you may have. -- thanks, John Menninger -- John Menninger 735 George St. Iowa City, Iowa 52246

1 0

[alsa-devel] [PATCH v6 0/3] ASoC: sun4i-i2s: Updates to the driver
by codekipper＠gmail.com 27 Aug '19

27 Aug '19

From: Marcus Cooper <codekipper(a)gmail.com> Hi All, here is a patch series to add some improvements to the sun4i-i2s driver which is enough to get HDMI audio working on the A83T, A64, H3 and H5 platforms. I've dropped a lot of the functionality that was presented earlier in favour of getting initial HDMI audio delivered. H6 and multi-channel HDMI will follow shortly. My test branch for this can be found at https://github.com/codekipper/linux-sunxi/commits/upstream-i2s , I've been using a Pine64 to test with; validating the new SoC block with HDMI audio and ensuring that I've not broken the old block by making sure that the audio codec still works. BR, CK --- v6 changes compared to v5 are: - removed patches for multi-channel and H6 HDMI audio. - removed patch for 20, 24 and 32 bit (will push support for just 20 and 24bit) - ditched tdm patches as support has already been added. - added fix for A83T reg map. v5 changes compared to v4 are: - removed delivered patches. - Added more details to commit messages. - replaced some reg fields with function calls. - Added DSP_A and DSP_B support for H3 and later SoCs. - Added support for the Allwinner H6. v4 changes compared to v3 are: - Moved patches around so that the more controversial of patches are at the top of the stack. - Added more details to commit messages. - Fixed 20bit audio PCM format to use 4 bytes. - Reduced number of flags used to indicate a new SoC. v3 changes compared to v2 are: - added back slave mode changes - added back the use of tdm properties - changes to regmap and caching - removed loopback functionality - fixes to the channel offset mask v2 changes compared to v1 are: - removed slave mode changes which didn't set mclk and bclk div. - removed use of tdm and now use a dedicated property. - fix commit message to better explain reason for sign extending - add divider calculations for newer SoCs. - add support for multi-lane i2s data output. - add support for 20, 24 and 32 bit samples. - add loopback property so blocks can be tested without a codec. --- Marcus Cooper (3): ASoC: sun4i-i2s: incorrect regmap for A83T ASoC: sun4i-i2s: Add regmap field to sign extend sample ASoC: sun4i-i2s: Adjust LRCLK width sound/soc/sunxi/sun4i-i2s.c | 24 ++++++++++++++++++++++-- 1 file changed, 22 insertions(+), 2 deletions(-) -- 2.23.0

4 12

[alsa-devel] [PATCH 00/21] ASoC: sun4i-i2s: Number of fixes and TDM Support
by Maxime Ripard 27 Aug '19

27 Aug '19

From: Maxime Ripard <maxime.ripard(a)bootlin.com> Hi, This series aims at fixing a number of issues in the current i2s driver, mostly related to the i2s master support and the A83t support. It also uses that occasion to cleanup a few things and simplify the driver. Finally, it builds on those fixes and cleanups to introduce TDM and DSP formats support. Let me know what you think, Maxime Marcus Cooper (1): ASoC: sun4i-i2s: Fix the MCLK and BCLK dividers on newer SoCs Maxime Ripard (20): ASoC: sun4i-i2s: Register regmap and PCM before our component ASoC: sun4i-i2s: Switch to devm for PCM register ASoC: sun4i-i2s: Replace call to params_channels by local variable ASoC: sun4i-i2s: Move the channel configuration to a callback ASoC: sun4i-i2s: Move the format configuration to a callback ASoC: sun4i-i2s: Rework MCLK divider calculation ASoC: sun4i-i2s: Don't use the oversample to calculate BCLK ASoC: sun4i-i2s: Use module clock as BCLK parent on newer SoCs ASoC: sun4i-i2s: RX and TX counter registers are swapped ASoC: sun4i-i2s: Use the actual format width instead of an hardcoded one ASoC: sun4i-i2s: Fix LRCK and BCLK polarity offsets on newer SoCs ASoC: sun4i-i2s: Fix the LRCK polarity ASoC: sun4i-i2s: Fix WSS and SR fields for the A83t ASoC: sun4i-i2s: Fix MCLK Enable bit offset on A83t ASoC: sun4i-i2s: Fix the LRCK period on A83t ASoC: sun4i-i2s: Remove duplicated quirks structure ASoC: sun4i-i2s: Pass the channels number as an argument ASoC: sun4i-i2s: Support more channels ASoC: sun4i-i2s: Add support for TDM slots ASoC: sun4i-i2s: Add support for DSP formats sound/soc/sunxi/sun4i-i2s.c | 660 ++++++++++++++++++++----------------- 1 file changed, 372 insertions(+), 288 deletions(-) base-commit: d45331b00ddb179e291766617259261c112db872 -- git-series 0.9.1

4 50

[alsa-devel] [PATCH 1/2] Revert "ASoC: sun4i-i2s: Remove duplicated quirks structure"
by Maxime Ripard 27 Aug '19

27 Aug '19

From: Maxime Ripard <maxime.ripard(a)bootlin.com> This reverts commit 3e9acd7ac6933cdc20c441bbf9a38ed9e42e1490. It turns out that while one I2S controller is described in the A83t datasheet, the driver supports another, undocumented, one that has been inherited from the older SoCs, while the documented one uses the new design. Fixes: 3e9acd7ac693 ("ASoC: sun4i-i2s: Remove duplicated quirks structure") Signed-off-by: Maxime Ripard <maxime.ripard(a)bootlin.com> --- sound/soc/sunxi/sun4i-i2s.c | 25 ++++++++++++++++++++++++- 1 file changed, 24 insertions(+), 1 deletion(-) diff --git a/sound/soc/sunxi/sun4i-i2s.c b/sound/soc/sunxi/sun4i-i2s.c index 57bf2a33753e..a6a3f772fdf0 100644 --- a/sound/soc/sunxi/sun4i-i2s.c +++ b/sound/soc/sunxi/sun4i-i2s.c @@ -1097,6 +1097,11 @@ static const struct sun4i_i2s_quirks sun6i_a31_i2s_quirks = { .set_fmt = sun4i_i2s_set_soc_fmt, }; +/* + * This doesn't describe the TDM controller documented in the A83t + * datasheet, but the three undocumented I2S controller that use the + * older design. + */ static const struct sun4i_i2s_quirks sun8i_a83t_i2s_quirks = { .has_reset = true, .reg_offset_txdata = SUN8I_I2S_FIFO_TX_REG, @@ -1115,6 +1120,24 @@ static const struct sun4i_i2s_quirks sun8i_a83t_i2s_quirks = { .set_fmt = sun8i_i2s_set_soc_fmt, }; +static const struct sun4i_i2s_quirks sun8i_h3_i2s_quirks = { + .has_reset = true, + .reg_offset_txdata = SUN8I_I2S_FIFO_TX_REG, + .sun4i_i2s_regmap = &sun8i_i2s_regmap_config, + .field_clkdiv_mclk_en = REG_FIELD(SUN4I_I2S_CLK_DIV_REG, 8, 8), + .field_fmt_wss = REG_FIELD(SUN4I_I2S_FMT0_REG, 0, 2), + .field_fmt_sr = REG_FIELD(SUN4I_I2S_FMT0_REG, 4, 6), + .bclk_dividers = sun8i_i2s_clk_div, + .num_bclk_dividers = ARRAY_SIZE(sun8i_i2s_clk_div), + .mclk_dividers = sun8i_i2s_clk_div, + .num_mclk_dividers = ARRAY_SIZE(sun8i_i2s_clk_div), + .get_bclk_parent_rate = sun8i_i2s_get_bclk_parent_rate, + .get_sr = sun8i_i2s_get_sr_wss, + .get_wss = sun8i_i2s_get_sr_wss, + .set_chan_cfg = sun8i_i2s_set_chan_cfg, + .set_fmt = sun8i_i2s_set_soc_fmt, +}; + static const struct sun4i_i2s_quirks sun50i_a64_codec_i2s_quirks = { .has_reset = true, .reg_offset_txdata = SUN8I_I2S_FIFO_TX_REG, @@ -1296,7 +1319,7 @@ static const struct of_device_id sun4i_i2s_match[] = { }, { .compatible = "allwinner,sun8i-h3-i2s", - .data = &sun8i_a83t_i2s_quirks, + .data = &sun8i_h3_i2s_quirks, }, { .compatible = "allwinner,sun50i-a64-codec-i2s", -- 2.21.0

3 4

[alsa-devel] [PATCH 5.2 010/162] ASoC: samsung: odroid: fix a double-free issue for cpu_dai
by Greg Kroah-Hartman 27 Aug '19

27 Aug '19

[ Upstream commit 2abee12c0ab1924a69993d2c063a39a952e7d836 ] The cpu_dai variable is still being used after the of_node_put() call, which may result in double-free: of_node_put(cpu_dai); ---> released here ret = devm_snd_soc_register_card(dev, card); if (ret < 0) { ... goto err_put_clk_i2s; --> jump to err_put_clk_i2s ... err_put_clk_i2s: clk_put(priv->clk_i2s_bus); err_put_sclk: clk_put(priv->sclk_i2s); err_put_cpu_dai: of_node_put(cpu_dai); --> double-free here Fixes: d832d2b246c5 ("ASoC: samsung: odroid: Fix of_node refcount unbalance") Signed-off-by: Wen Yang <wen.yang99(a)zte.com.cn> Cc: Krzysztof Kozlowski <krzk(a)kernel.org> Cc: Sangbeom Kim <sbkim73(a)samsung.com> Cc: Sylwester Nawrocki <s.nawrocki(a)samsung.com> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: alsa-devel(a)alsa-project.org Cc: linux-kernel(a)vger.kernel.org Link: https://lore.kernel.org/r/1562989575-33785-3-git-send-email-wen.yang99@zte.… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/samsung/odroid.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/samsung/odroid.c b/sound/soc/samsung/odroid.c index 95c35e3ff3303..d606e48fe551a 100644 --- a/sound/soc/samsung/odroid.c +++ b/sound/soc/samsung/odroid.c @@ -299,7 +299,6 @@ static int odroid_audio_probe(struct platform_device *pdev) ret = PTR_ERR(priv->clk_i2s_bus); goto err_put_sclk; } - of_node_put(cpu_dai); ret = devm_snd_soc_register_card(dev, card); if (ret < 0) { @@ -307,6 +306,7 @@ static int odroid_audio_probe(struct platform_device *pdev) goto err_put_clk_i2s; } + of_node_put(cpu_dai); of_node_put(codec); return 0; -- 2.20.1

1 0

[alsa-devel] [PATCH 5.2 009/162] ASoC: samsung: odroid: fix an use-after-free issue for codec
by Greg Kroah-Hartman 27 Aug '19

27 Aug '19

[ Upstream commit 9b6d104a6b150bd4d3e5b039340e1f6b20c2e3c1 ] The codec variable is still being used after the of_node_put() call, which may result in use-after-free. Fixes: bc3cf17b575a ("ASoC: samsung: odroid: Add support for secondary CPU DAI") Signed-off-by: Wen Yang <wen.yang99(a)zte.com.cn> Cc: Krzysztof Kozlowski <krzk(a)kernel.org> Cc: Sangbeom Kim <sbkim73(a)samsung.com> Cc: Sylwester Nawrocki <s.nawrocki(a)samsung.com> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: alsa-devel(a)alsa-project.org Cc: linux-kernel(a)vger.kernel.org Link: https://lore.kernel.org/r/1562989575-33785-2-git-send-email-wen.yang99@zte.… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/samsung/odroid.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/samsung/odroid.c b/sound/soc/samsung/odroid.c index e688169ff12ab..95c35e3ff3303 100644 --- a/sound/soc/samsung/odroid.c +++ b/sound/soc/samsung/odroid.c @@ -275,9 +275,8 @@ static int odroid_audio_probe(struct platform_device *pdev) } of_node_put(cpu); - of_node_put(codec); if (ret < 0) - return ret; + goto err_put_node; ret = snd_soc_of_get_dai_link_codecs(dev, codec, codec_link); if (ret < 0) @@ -308,6 +307,7 @@ static int odroid_audio_probe(struct platform_device *pdev) goto err_put_clk_i2s; } + of_node_put(codec); return 0; err_put_clk_i2s: @@ -317,6 +317,8 @@ err_put_sclk: err_put_cpu_dai: of_node_put(cpu_dai); snd_soc_of_put_dai_link_codecs(codec_link); +err_put_node: + of_node_put(codec); return ret; } -- 2.20.1

1 0

[alsa-devel] [PATCH 1/1] ASoC: topology: Fix variable overflow in split_format
by Marcin Pietraszko 27 Aug '19

27 Aug '19

The 1 which was 32-bit long, when left-shifted by more than 31 caused an overflow, which resulted in some of the formats being not supported. Signed-off-by: Marcin Pietraszko <marcin.pietraszko(a)linux.intel.com> --- src/topology/pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/src/topology/pcm.c b/src/topology/pcm.c index 5f586dc..00a35b9 100644 --- a/src/topology/pcm.c +++ b/src/topology/pcm.c @@ -321,7 +321,7 @@ static int split_format(struct snd_soc_tplg_stream_caps *caps, char *str) return -EINVAL; } - caps->formats |= 1 << format; + caps->formats |= 1ull << format; s = strtok(NULL, ", "); i++; } -- 2.7.4

3 2

[alsa-devel] [PATCH] ALSA: usb-audio: Add Hiby R3 to quirks for native DSD support
by ilya.pshonkin＠netforce.ua 26 Aug '19

26 Aug '19

From: Sudokamikaze <sudokamikaze(a)protonmail.com> This patch adds quirk VID/PID IDs for Hiby R3 portable DSD player DSD support Signed-off-by: Sudokamikaze <sudokamikaze(a)protonmail.com> --- sound/usb/quirks.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 78858918cbc1..f90418149e4e 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1588,6 +1588,12 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip, if (fp->altsetting == 2) return SNDRV_PCM_FMTBIT_DSD_U32_BE; break; + + /* ESS Sabre based USB DACs */ + case USB_ID(0xc502, 0x0051): /* Hiby R3 */ + if (fp->altsetting == 4) + return SNDRV_PCM_FMTBIT_DSD_U32_BE; + break; case USB_ID(0x0d8c, 0x0316): /* Hegel HD12 DSD */ case USB_ID(0x10cb, 0x0103): /* The Bit Opus #3; with fp->dsd_raw */ -- 2.23.0

2 1

Re: [alsa-devel] [PATCH] ALSA: usb-audio: Add Hiby R3 to quirks for native DSD support
by Takashi Iwai 26 Aug '19

26 Aug '19

On Mon, 26 Aug 2019 18:05:05 +0200, Sudo Kamikaze wrote: > > From: Sudokamikaze <sudokamikaze(a)protonmail.com> > > This patch adds quirk VID/PID IDs for Hiby R3 portable DSD player DSD support > > Signed-off-by: Sudokamikaze <sudokamikaze(a)protonmail.com> > --- > sound/usb/quirks.c | 6 ++++++ > 1 file changed, 6 insertions(+) > > diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c > index 78858918cbc1..f90418149e4e 100644 > --- a/sound/usb/quirks.c > +++ b/sound/usb/quirks.c > @@ -1588,6 +1588,12 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip, > if (fp->altsetting == 2) > return SNDRV_PCM_FMTBIT_DSD_U32_BE; > break; > + > + /* ESS Sabre based USB DACs */ > + case USB_ID(0xc502, 0x0051): /* Hiby R3 */ > + if (fp->altsetting == 4) > + return SNDRV_PCM_FMTBIT_DSD_U32_BE; > + break; Is fp->dsd_raw set for your device? IOW, instead of adding an individual entry here, adding the vendor 0xc502 with fp->dsd_raw check like below would work? thanks, Takashi --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1658,6 +1658,7 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip, case 0x152a: /* Thesycon devices */ case 0x25ce: /* Mytek devices */ case 0x2ab6: /* T+A devices */ + case 0xc502: /* ESS Sabre */ if (fp->dsd_raw) return SNDRV_PCM_FMTBIT_DSD_U32_BE; break;

1 0

[alsa-devel] [PATCH] ASoC: es8316: limit headphone mixer volume
by Katsuhiro Suzuki 26 Aug '19

26 Aug '19

This patch limits Headphone mixer volume to 4 from 7. Because output sound suddenly becomes very loudly with many noise if set volume over 4. Signed-off-by: Katsuhiro Suzuki <katsuhiro(a)katsuster.net> --- sound/soc/codecs/es8316.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/es8316.c b/sound/soc/codecs/es8316.c index 8dfb5dbeebbf..bc4141e1eb7f 100644 --- a/sound/soc/codecs/es8316.c +++ b/sound/soc/codecs/es8316.c @@ -91,7 +91,7 @@ static const struct snd_kcontrol_new es8316_snd_controls[] = { SOC_DOUBLE_TLV("Headphone Playback Volume", ES8316_CPHP_ICAL_VOL, 4, 0, 3, 1, hpout_vol_tlv), SOC_DOUBLE_TLV("Headphone Mixer Volume", ES8316_HPMIX_VOL, - 0, 4, 7, 0, hpmixer_gain_tlv), + 0, 4, 4, 0, hpmixer_gain_tlv), SOC_ENUM("Playback Polarity", dacpol), SOC_DOUBLE_R_TLV("DAC Playback Volume", ES8316_DAC_VOLL, -- 2.23.0.rc1

3 6

[alsa-devel] [PATCH] ALSA: oxfw: fix to handle correct stream for PCM playback
by Takashi Sakamoto 26 Aug '19

26 Aug '19

When userspace application calls ioctl(2) to configure hardware for PCM playback substream, ALSA OXFW driver handles incoming AMDTP stream. In this case, outgoing AMDTP stream should be handled. This commit fixes the bug for v5.3-rc kernel. Fixes: 4f380d007052 ("ALSA: oxfw: configure packet format in pcm.hw_params callback") Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/oxfw/oxfw-pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/firewire/oxfw/oxfw-pcm.c b/sound/firewire/oxfw/oxfw-pcm.c index 9ea39348cdf5..7c6d1c277d4d 100644 --- a/sound/firewire/oxfw/oxfw-pcm.c +++ b/sound/firewire/oxfw/oxfw-pcm.c @@ -248,7 +248,7 @@ static int pcm_playback_hw_params(struct snd_pcm_substream *substream, unsigned int channels = params_channels(hw_params); mutex_lock(&oxfw->mutex); - err = snd_oxfw_stream_reserve_duplex(oxfw, &oxfw->tx_stream, + err = snd_oxfw_stream_reserve_duplex(oxfw, &oxfw->rx_stream, rate, channels); if (err >= 0) ++oxfw->substreams_count; -- 2.20.1

2 1

[alsa-devel] [PATCH] ALSA: oxfw: fix NULL pointer dereference to unused stream structure
by Takashi Sakamoto 26 Aug '19

26 Aug '19

ALSA oxfw driver supports Griffin FireWave. This device supports one isochronous stream for PCM playback. Current driver executes code to wait event for stream structure of opposite direction. This causes NULL pointer dereference. This commit fixes the bug. Fixes: ac5d77864cfc ("ALSA: oxfw: support AMDTP domain") Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/oxfw/oxfw-stream.c | 9 +++++++-- 1 file changed, 7 insertions(+), 2 deletions(-) diff --git a/sound/firewire/oxfw/oxfw-stream.c b/sound/firewire/oxfw/oxfw-stream.c index 0318dc4dcc55..3c9a796b6526 100644 --- a/sound/firewire/oxfw/oxfw-stream.c +++ b/sound/firewire/oxfw/oxfw-stream.c @@ -350,13 +350,18 @@ int snd_oxfw_stream_start_duplex(struct snd_oxfw *oxfw) // Wait first packet. if (!amdtp_stream_wait_callback(&oxfw->rx_stream, - CALLBACK_TIMEOUT) || - !amdtp_stream_wait_callback(&oxfw->tx_stream, CALLBACK_TIMEOUT)) { err = -ETIMEDOUT; goto error; } + if (oxfw->has_output) { + if (!amdtp_stream_wait_callback(&oxfw->tx_stream, + CALLBACK_TIMEOUT)) { + err = -ETIMEDOUT; + goto error; + } + } } return 0; -- 2.20.1

2 1

[alsa-devel] [PATCH] ALSA: usb-audio: remove some dead code
by Dan Carpenter 26 Aug '19

26 Aug '19

We recently cleaned up the error handling in commit 52c3e317a857 ("ALSA: usb-audio: Unify the release of usb_mixer_elem_info objects") but accidentally left this stray return. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- sound/usb/mixer.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index a0468c4a6326..a607577f0097 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -2633,7 +2633,6 @@ static int parse_audio_selector_unit(struct mixer_build *state, int unitid, usb_audio_err(state->chip, "cannot malloc kcontrol\n"); err = -ENOMEM; goto error_name; - return -ENOMEM; } kctl->private_value = (unsigned long)namelist; kctl->private_free = usb_mixer_selector_elem_free; -- 2.20.1

2 1

[alsa-devel] [bug report] ALSA: usb-audio: More validations of descriptor units
by Dan Carpenter 26 Aug '19

26 Aug '19

Hello Takashi Iwai, This is a semi-automatic email about new static checker warnings. The patch 57f8770620e9: "ALSA: usb-audio: More validations of descriptor units" from Aug 20, 2019, leads to the following Smatch complaint: sound/usb/quirks.c:254 create_yamaha_midi_quirk() warn: variable dereferenced before check 'injd' (see line 251) sound/usb/quirks.c 244 /* must have some valid jack descriptors */ 245 injd = snd_usb_find_csint_desc(alts->extra, alts->extralen, 246 NULL, USB_MS_MIDI_IN_JACK); 247 outjd = snd_usb_find_csint_desc(alts->extra, alts->extralen, 248 NULL, USB_MS_MIDI_OUT_JACK); 249 if (!injd && !outjd) ^^^^^^^^^^^^^ At least one must be valid. 250 return -ENODEV; 250 return -ENODEV; 251 if (!snd_usb_validate_midi_desc(injd) || 252 !snd_usb_validate_midi_desc(outjd)) So this should return true/valid if the pointer is NULL? 253 return -ENODEV; 254 if (injd && (injd->bLength < 5 || 255 (injd->bJackType != USB_MS_EMBEDDED && 256 injd->bJackType != USB_MS_EXTERNAL))) regards, dan carpenter

2 1

[alsa-devel] [PATCH] ALSA: firewire-lib: allocate additional entries for list of packet descriptor to avoid out-of-bounds access
by Takashi Sakamoto 25 Aug '19

25 Aug '19

This patch is for Linux v5.3-rc7. In a case of delay to execute scheduled tasklet for isoc context, it's possible to handle queued packets than 16 (=INTERRUPT_INTERVAL). In the case, out-of-bounds access occurs because the list of packet descriptor is allocated just for 16 packets. This causes any negative effects in kernel memory or software IRQ context. It's quite rare because current implementation allows user processes to flush the queued packet in process context by executing several PCM ioctl(2) commands. However, it's good to prevent. This commit is a prevention against this bug. For safe, the allocation is done for 16 plus 12 packets, equivalent to 1.5 msec plus. Furthermore, when detecting the case, packet streaming is cancelled and kernel log is printed to notice to users and developers. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/amdtp-stream.c | 27 ++++++++++++++++++++++++--- 1 file changed, 24 insertions(+), 3 deletions(-) diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c index 1a92855c7647..f03321888997 100644 --- a/sound/firewire/amdtp-stream.c +++ b/sound/firewire/amdtp-stream.c @@ -56,6 +56,11 @@ #define INTERRUPT_INTERVAL 16 #define QUEUE_LENGTH 48 +// For jitter of software IRQ execution, keep more entries for the list +// of packet descriptor equivalent to 1.5 msec to avoid out-of-bounds +// access to process queued packets. +#define DESC_COUNT (INTERRUPT_INTERVAL + 12) + // For iso header, tstamp and 2 CIP header. #define IR_CTX_HEADER_SIZE_CIP 16 // For iso header and tstamp. @@ -779,12 +784,22 @@ static void out_stream_callback(struct fw_iso_context *context, u32 tstamp, { struct amdtp_stream *s = private_data; const __be32 *ctx_header = header; - unsigned int packets = header_length / sizeof(*ctx_header); + unsigned int packets; int i; if (s->packet_index < 0) return; + // The number of packets in buffer. + packets = header_length / sizeof(*ctx_header); + if (packets > DESC_COUNT) { + cancel_stream(s); + dev_info(&s->unit->device, + "out-stream: Unexpected count of packet: %d\n", + packets); + return; + } + generate_ideal_pkt_descs(s, s->pkt_descs, ctx_header, packets); process_ctx_payloads(s, s->pkt_descs, packets); @@ -830,6 +845,13 @@ static void in_stream_callback(struct fw_iso_context *context, u32 tstamp, // The number of packets in buffer. packets = header_length / s->ctx_data.tx.ctx_header_size; + if (packets > DESC_COUNT) { + cancel_stream(s); + dev_info(&s->unit->device, + "in-stream: Unexpected count of packet: %d\n", + packets); + return; + } err = generate_device_pkt_descs(s, s->pkt_descs, ctx_header, packets); if (err < 0) { @@ -981,8 +1003,7 @@ static int amdtp_stream_start(struct amdtp_stream *s, int channel, int speed) else s->tag = TAG_CIP; - s->pkt_descs = kcalloc(INTERRUPT_INTERVAL, sizeof(*s->pkt_descs), - GFP_KERNEL); + s->pkt_descs = kcalloc(DESC_COUNT, sizeof(*s->pkt_descs), GFP_KERNEL); if (!s->pkt_descs) { err = -ENOMEM; goto err_context; -- 2.20.1

2 2

[alsa-devel] [PATCH] ALSA: usb-audio: Clean up check_input_term()
by Takashi Iwai 25 Aug '19

25 Aug '19

The primary changes in this patch are cleanups of __check_input_term() and move to a non-nested switch-case block by evaluating the pair of UAC version and the unit type, as we've done for parse_audio_unit(). Also each parser is split into the function for readability. Now, a slight behavior change by this cleanup is the handling of processing and extension units. Formerly we've dealt with them differently between UAC1/2 and UAC3; the latter returns an error if no input sources are available, while the former continues to parse. In this patch, unify the behavior in all cases: when input sources are available, it parses recursively, then override the type and the id, as well as channel information if not provided yet. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/mixer.c | 401 ++++++++++++++++++++++++++++-------------------------- 1 file changed, 209 insertions(+), 192 deletions(-) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index c9777d8a76f5..5291739cab57 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -758,224 +758,242 @@ static int uac_mixer_unit_get_channels(struct mixer_build *state, } /* - * parse the source unit recursively until it reaches to a terminal - * or a branched unit. + * Parse Input Terminal Unit */ static int __check_input_term(struct mixer_build *state, int id, - struct usb_audio_term *term) + struct usb_audio_term *term); + +static int parse_term_uac1_iterm_unit(struct mixer_build *state, + struct usb_audio_term *term, + void *p1, int id) { - int protocol = state->mixer->protocol; + struct uac_input_terminal_descriptor *d = p1; + + term->type = le16_to_cpu(d->wTerminalType); + term->channels = d->bNrChannels; + term->chconfig = le16_to_cpu(d->wChannelConfig); + term->name = d->iTerminal; + return 0; +} + +static int parse_term_uac2_iterm_unit(struct mixer_build *state, + struct usb_audio_term *term, + void *p1, int id) +{ + struct uac2_input_terminal_descriptor *d = p1; int err; - void *p1; - unsigned char *hdr; - memset(term, 0, sizeof(*term)); - for (;;) { - /* a loop in the terminal chain? */ - if (test_and_set_bit(id, state->termbitmap)) - return -EINVAL; + /* call recursively to verify the referenced clock entity */ + err = __check_input_term(state, d->bCSourceID, term); + if (err < 0) + return err; - p1 = find_audio_control_unit(state, id); - if (!p1) - break; - if (!snd_usb_validate_audio_desc(p1, protocol)) - break; /* bad descriptor */ + /* save input term properties after recursion, + * to ensure they are not overriden by the recursion calls + */ + term->id = id; + term->type = le16_to_cpu(d->wTerminalType); + term->channels = d->bNrChannels; + term->chconfig = le32_to_cpu(d->bmChannelConfig); + term->name = d->iTerminal; + return 0; +} - hdr = p1; - term->id = id; +static int parse_term_uac3_iterm_unit(struct mixer_build *state, + struct usb_audio_term *term, + void *p1, int id) +{ + struct uac3_input_terminal_descriptor *d = p1; + int err; - if (protocol == UAC_VERSION_1 || protocol == UAC_VERSION_2) { - switch (hdr[2]) { - case UAC_INPUT_TERMINAL: - if (protocol == UAC_VERSION_1) { - struct uac_input_terminal_descriptor *d = p1; - - term->type = le16_to_cpu(d->wTerminalType); - term->channels = d->bNrChannels; - term->chconfig = le16_to_cpu(d->wChannelConfig); - term->name = d->iTerminal; - } else { /* UAC_VERSION_2 */ - struct uac2_input_terminal_descriptor *d = p1; - - /* call recursively to verify that the - * referenced clock entity is valid */ - err = __check_input_term(state, d->bCSourceID, term); - if (err < 0) - return err; + /* call recursively to verify the referenced clock entity */ + err = __check_input_term(state, d->bCSourceID, term); + if (err < 0) + return err; - /* save input term properties after recursion, - * to ensure they are not overriden by the - * recursion calls */ - term->id = id; - term->type = le16_to_cpu(d->wTerminalType); - term->channels = d->bNrChannels; - term->chconfig = le32_to_cpu(d->bmChannelConfig); - term->name = d->iTerminal; - } - return 0; - case UAC_FEATURE_UNIT: { - /* the header is the same for v1 and v2 */ - struct uac_feature_unit_descriptor *d = p1; + /* save input term properties after recursion, + * to ensure they are not overriden by the recursion calls + */ + term->id = id; + term->type = le16_to_cpu(d->wTerminalType); - id = d->bSourceID; - break; /* continue to parse */ - } - case UAC_MIXER_UNIT: { - struct uac_mixer_unit_descriptor *d = p1; - - term->type = UAC3_MIXER_UNIT << 16; /* virtual type */ - term->channels = uac_mixer_unit_bNrChannels(d); - term->chconfig = uac_mixer_unit_wChannelConfig(d, protocol); - term->name = uac_mixer_unit_iMixer(d); - return 0; - } - case UAC_SELECTOR_UNIT: - case UAC2_CLOCK_SELECTOR: { - struct uac_selector_unit_descriptor *d = p1; - /* call recursively to retrieve the channel info */ - err = __check_input_term(state, d->baSourceID[0], term); - if (err < 0) - return err; - term->type = UAC3_SELECTOR_UNIT << 16; /* virtual type */ - term->id = id; - term->name = uac_selector_unit_iSelector(d); - return 0; - } - case UAC1_PROCESSING_UNIT: - /* UAC2_EFFECT_UNIT */ - if (protocol == UAC_VERSION_1) - term->type = UAC3_PROCESSING_UNIT << 16; /* virtual type */ - else /* UAC_VERSION_2 */ - term->type = UAC3_EFFECT_UNIT << 16; /* virtual type */ - /* fall through */ - case UAC1_EXTENSION_UNIT: - /* UAC2_PROCESSING_UNIT_V2 */ - if (protocol == UAC_VERSION_1 && !term->type) - term->type = UAC3_EXTENSION_UNIT << 16; /* virtual type */ - else if (protocol == UAC_VERSION_2 && !term->type) - term->type = UAC3_PROCESSING_UNIT << 16; /* virtual type */ - /* fall through */ - case UAC2_EXTENSION_UNIT_V2: { - struct uac_processing_unit_descriptor *d = p1; - - if (protocol == UAC_VERSION_2 && - hdr[2] == UAC2_EFFECT_UNIT) { - /* UAC2/UAC1 unit IDs overlap here in an - * uncompatible way. Ignore this unit for now. - */ - return 0; - } + err = get_cluster_channels_v3(state, le16_to_cpu(d->wClusterDescrID)); + if (err < 0) + return err; + term->channels = err; - if (d->bNrInPins) { - id = d->baSourceID[0]; - break; /* continue to parse */ - } - if (!term->type) - term->type = UAC3_EXTENSION_UNIT << 16; /* virtual type */ + /* REVISIT: UAC3 IT doesn't have channels cfg */ + term->chconfig = 0; - term->channels = uac_processing_unit_bNrChannels(d); - term->chconfig = uac_processing_unit_wChannelConfig(d, protocol); - term->name = uac_processing_unit_iProcessing(d, protocol); - return 0; - } - case UAC2_CLOCK_SOURCE: { - struct uac_clock_source_descriptor *d = p1; + term->name = le16_to_cpu(d->wTerminalDescrStr); + return 0; +} - term->type = UAC3_CLOCK_SOURCE << 16; /* virtual type */ - term->id = id; - term->name = d->iClockSource; - return 0; - } - default: - return -ENODEV; - } - } else { /* UAC_VERSION_3 */ - switch (hdr[2]) { - case UAC_INPUT_TERMINAL: { - struct uac3_input_terminal_descriptor *d = p1; - - /* call recursively to verify that the - * referenced clock entity is valid */ - err = __check_input_term(state, d->bCSourceID, term); - if (err < 0) - return err; +static int parse_term_mixer_unit(struct mixer_build *state, + struct usb_audio_term *term, + void *p1, int id) +{ + struct uac_mixer_unit_descriptor *d = p1; + int protocol = state->mixer->protocol; + int err; - /* save input term properties after recursion, - * to ensure they are not overriden by the - * recursion calls */ - term->id = id; - term->type = le16_to_cpu(d->wTerminalType); + err = uac_mixer_unit_get_channels(state, d); + if (err <= 0) + return err; - err = get_cluster_channels_v3(state, le16_to_cpu(d->wClusterDescrID)); - if (err < 0) - return err; - term->channels = err; + term->type = UAC3_MIXER_UNIT << 16; /* virtual type */ + term->channels = err; + if (protocol != UAC_VERSION_3) { + term->chconfig = uac_mixer_unit_wChannelConfig(d, protocol); + term->name = uac_mixer_unit_iMixer(d); + } + return 0; +} + +static int parse_term_selector_unit(struct mixer_build *state, + struct usb_audio_term *term, + void *p1, int id) +{ + struct uac_selector_unit_descriptor *d = p1; + int err; - /* REVISIT: UAC3 IT doesn't have channels cfg */ - term->chconfig = 0; + /* call recursively to retrieve the channel info */ + err = __check_input_term(state, d->baSourceID[0], term); + if (err < 0) + return err; + term->type = UAC3_SELECTOR_UNIT << 16; /* virtual type */ + term->id = id; + if (state->mixer->protocol != UAC_VERSION_3) + term->name = uac_selector_unit_iSelector(d); + return 0; +} - term->name = le16_to_cpu(d->wTerminalDescrStr); - return 0; - } - case UAC3_FEATURE_UNIT: { - struct uac3_feature_unit_descriptor *d = p1; +static int parse_term_proc_unit(struct mixer_build *state, + struct usb_audio_term *term, + void *p1, int id, int vtype) +{ + struct uac_processing_unit_descriptor *d = p1; + int protocol = state->mixer->protocol; + int err; - id = d->bSourceID; - break; /* continue to parse */ - } - case UAC3_CLOCK_SOURCE: { - struct uac3_clock_source_descriptor *d = p1; + if (d->bNrInPins) { + /* call recursively to retrieve the channel info */ + err = __check_input_term(state, d->baSourceID[0], term); + if (err < 0) + return err; + } - term->type = UAC3_CLOCK_SOURCE << 16; /* virtual type */ - term->id = id; - term->name = le16_to_cpu(d->wClockSourceStr); - return 0; - } - case UAC3_MIXER_UNIT: { - struct uac_mixer_unit_descriptor *d = p1; + term->type = vtype << 16; /* virtual type */ + term->id = id; - err = uac_mixer_unit_get_channels(state, d); - if (err <= 0) - return err; + if (protocol == UAC_VERSION_3) + return 0; - term->channels = err; - term->type = UAC3_MIXER_UNIT << 16; /* virtual type */ + if (!term->channels) { + term->channels = uac_processing_unit_bNrChannels(d); + term->chconfig = uac_processing_unit_wChannelConfig(d, protocol); + } + term->name = uac_processing_unit_iProcessing(d, protocol); + return 0; +} - return 0; - } - case UAC3_SELECTOR_UNIT: - case UAC3_CLOCK_SELECTOR: { - struct uac_selector_unit_descriptor *d = p1; - /* call recursively to retrieve the channel info */ - err = __check_input_term(state, d->baSourceID[0], term); - if (err < 0) - return err; - term->type = UAC3_SELECTOR_UNIT << 16; /* virtual type */ - term->id = id; - term->name = 0; /* TODO: UAC3 Class-specific strings */ +static int parse_term_uac2_clock_source(struct mixer_build *state, + struct usb_audio_term *term, + void *p1, int id) +{ + struct uac_clock_source_descriptor *d = p1; - return 0; - } - case UAC3_PROCESSING_UNIT: { - struct uac_processing_unit_descriptor *d = p1; + term->type = UAC3_CLOCK_SOURCE << 16; /* virtual type */ + term->id = id; + term->name = d->iClockSource; + return 0; +} - if (!d->bNrInPins) - return -EINVAL; +static int parse_term_uac3_clock_source(struct mixer_build *state, + struct usb_audio_term *term, + void *p1, int id) +{ + struct uac3_clock_source_descriptor *d = p1; - /* call recursively to retrieve the channel info */ - err = __check_input_term(state, d->baSourceID[0], term); - if (err < 0) - return err; + term->type = UAC3_CLOCK_SOURCE << 16; /* virtual type */ + term->id = id; + term->name = le16_to_cpu(d->wClockSourceStr); + return 0; +} - term->type = UAC3_PROCESSING_UNIT << 16; /* virtual type */ - term->id = id; - term->name = 0; /* TODO: UAC3 Class-specific strings */ +#define PTYPE(a, b) ((a) << 8 | (b)) - return 0; - } - default: - return -ENODEV; - } +/* + * parse the source unit recursively until it reaches to a terminal + * or a branched unit. + */ +static int __check_input_term(struct mixer_build *state, int id, + struct usb_audio_term *term) +{ + int protocol = state->mixer->protocol; + void *p1; + unsigned char *hdr; + + for (;;) { + /* a loop in the terminal chain? */ + if (test_and_set_bit(id, state->termbitmap)) + return -EINVAL; + + p1 = find_audio_control_unit(state, id); + if (!p1) + break; + if (!snd_usb_validate_audio_desc(p1, protocol)) + break; /* bad descriptor */ + + hdr = p1; + term->id = id; + + switch (PTYPE(protocol, hdr[2])) { + case PTYPE(UAC_VERSION_1, UAC_FEATURE_UNIT): + case PTYPE(UAC_VERSION_2, UAC_FEATURE_UNIT): + case PTYPE(UAC_VERSION_3, UAC3_FEATURE_UNIT): { + /* the header is the same for all versions */ + struct uac_feature_unit_descriptor *d = p1; + + id = d->bSourceID; + break; /* continue to parse */ + } + case PTYPE(UAC_VERSION_1, UAC_INPUT_TERMINAL): + return parse_term_uac1_iterm_unit(state, term, p1, id); + case PTYPE(UAC_VERSION_2, UAC_INPUT_TERMINAL): + return parse_term_uac2_iterm_unit(state, term, p1, id); + case PTYPE(UAC_VERSION_3, UAC_INPUT_TERMINAL): + return parse_term_uac3_iterm_unit(state, term, p1, id); + case PTYPE(UAC_VERSION_1, UAC_MIXER_UNIT): + case PTYPE(UAC_VERSION_2, UAC_MIXER_UNIT): + case PTYPE(UAC_VERSION_3, UAC3_MIXER_UNIT): + return parse_term_mixer_unit(state, term, p1, id); + case PTYPE(UAC_VERSION_1, UAC_SELECTOR_UNIT): + case PTYPE(UAC_VERSION_2, UAC_SELECTOR_UNIT): + case PTYPE(UAC_VERSION_2, UAC2_CLOCK_SELECTOR): + case PTYPE(UAC_VERSION_3, UAC3_SELECTOR_UNIT): + case PTYPE(UAC_VERSION_3, UAC3_CLOCK_SELECTOR): + return parse_term_selector_unit(state, term, p1, id); + case PTYPE(UAC_VERSION_1, UAC1_PROCESSING_UNIT): + case PTYPE(UAC_VERSION_2, UAC2_PROCESSING_UNIT_V2): + case PTYPE(UAC_VERSION_3, UAC3_PROCESSING_UNIT): + return parse_term_proc_unit(state, term, p1, id, + UAC3_PROCESSING_UNIT); + case PTYPE(UAC_VERSION_2, UAC2_EFFECT_UNIT): + case PTYPE(UAC_VERSION_3, UAC3_EFFECT_UNIT): + return parse_term_proc_unit(state, term, p1, id, + UAC3_EFFECT_UNIT); + case PTYPE(UAC_VERSION_1, UAC1_EXTENSION_UNIT): + case PTYPE(UAC_VERSION_2, UAC2_EXTENSION_UNIT_V2): + case PTYPE(UAC_VERSION_3, UAC3_EXTENSION_UNIT): + return parse_term_proc_unit(state, term, p1, id, + UAC3_EXTENSION_UNIT); + case PTYPE(UAC_VERSION_2, UAC2_CLOCK_SOURCE): + return parse_term_uac2_clock_source(state, term, p1, id); + case PTYPE(UAC_VERSION_3, UAC3_CLOCK_SOURCE): + return parse_term_uac3_clock_source(state, term, p1, id); + default: + return -ENODEV; } } return -ENODEV; @@ -2712,7 +2730,6 @@ static int parse_audio_unit(struct mixer_build *state, int unitid) return 0; /* skip invalid unit */ } -#define PTYPE(a, b) ((a) << 8 | (b)) switch (PTYPE(protocol, p1[2])) { case PTYPE(UAC_VERSION_1, UAC_INPUT_TERMINAL): case PTYPE(UAC_VERSION_2, UAC_INPUT_TERMINAL): -- 2.16.4

1 0

[alsa-devel] [PATCH] ALSA: seq: Fix potential concurrent access to the deleted pool
by Takashi Iwai 25 Aug '19

25 Aug '19

The input pool of a client might be deleted via the resize ioctl, the the access to it should be covered by the proper locks. Currently the only missing place is the call in snd_seq_ioctl_get_client_pool(), and this patch papers over it. Reported-by: syzbot+4a75454b9ca2777f35c7(a)syzkaller.appspotmail.com Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/seq/seq_clientmgr.c | 3 +-- sound/core/seq/seq_fifo.c | 17 +++++++++++++++++ sound/core/seq/seq_fifo.h | 2 ++ 3 files changed, 20 insertions(+), 2 deletions(-) diff --git a/sound/core/seq/seq_clientmgr.c b/sound/core/seq/seq_clientmgr.c index 7737b2670064..6d9592f0ae1d 100644 --- a/sound/core/seq/seq_clientmgr.c +++ b/sound/core/seq/seq_clientmgr.c @@ -1835,8 +1835,7 @@ static int snd_seq_ioctl_get_client_pool(struct snd_seq_client *client, if (cptr->type == USER_CLIENT) { info->input_pool = cptr->data.user.fifo_pool_size; info->input_free = info->input_pool; - if (cptr->data.user.fifo) - info->input_free = snd_seq_unused_cells(cptr->data.user.fifo->pool); + info->input_free = snd_seq_fifo_unused_cells(cptr->data.user.fifo); } else { info->input_pool = 0; info->input_free = 0; diff --git a/sound/core/seq/seq_fifo.c b/sound/core/seq/seq_fifo.c index ea69261f269a..eaaa8b5830bb 100644 --- a/sound/core/seq/seq_fifo.c +++ b/sound/core/seq/seq_fifo.c @@ -263,3 +263,20 @@ int snd_seq_fifo_resize(struct snd_seq_fifo *f, int poolsize) return 0; } + +/* get the number of unused cells safely */ +int snd_seq_fifo_unused_cells(struct snd_seq_fifo *f) +{ + unsigned long flags; + int cells; + + if (!f) + return 0; + + snd_use_lock_use(&f->use_lock); + spin_lock_irqsave(&f->lock, flags); + cells = snd_seq_unused_cells(f->pool); + spin_unlock_irqrestore(&f->lock, flags); + snd_use_lock_free(&f->use_lock); + return cells; +} diff --git a/sound/core/seq/seq_fifo.h b/sound/core/seq/seq_fifo.h index edc68743943d..b56a7b897c9c 100644 --- a/sound/core/seq/seq_fifo.h +++ b/sound/core/seq/seq_fifo.h @@ -53,5 +53,7 @@ int snd_seq_fifo_poll_wait(struct snd_seq_fifo *f, struct file *file, poll_table /* resize pool in fifo */ int snd_seq_fifo_resize(struct snd_seq_fifo *f, int poolsize); +/* get the number of unused cells safely */ +int snd_seq_fifo_unused_cells(struct snd_seq_fifo *f); #endif -- 2.16.4

1 0

[alsa-devel] [PATCH -next] soundwire: Fix -Wunused-function warning
by YueHaibing 25 Aug '19

25 Aug '19

If CONFIG_ACPI is not set, gcc warning this: drivers/soundwire/slave.c:16:12: warning: 'sdw_slave_add' defined but not used [-Wunused-function] move them to #ifdef CONFIG_ACPI block. Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- drivers/soundwire/slave.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/drivers/soundwire/slave.c b/drivers/soundwire/slave.c index f39a581..34c7e65 100644 --- a/drivers/soundwire/slave.c +++ b/drivers/soundwire/slave.c @@ -6,6 +6,7 @@ #include <linux/soundwire/sdw_type.h> #include "bus.h" +#if IS_ENABLED(CONFIG_ACPI) static void sdw_slave_release(struct device *dev) { struct sdw_slave *slave = dev_to_sdw_dev(dev); @@ -60,7 +61,6 @@ static int sdw_slave_add(struct sdw_bus *bus, return ret; } -#if IS_ENABLED(CONFIG_ACPI) /* * sdw_acpi_find_slaves() - Find Slave devices in Master ACPI node * @bus: SDW bus instance -- 2.7.4

4 5

[alsa-devel] sof/hda rework to share more of patch_hdmi.c logic
by Kai Vehmanen 23 Aug '19

23 Aug '19

Hi Takashi, I notice you are doing a lot of cleanups to HDA code. Just FYI I'm looking into modifying the SOF Intel backend to use snd-hda-codec-hdmi/patch_hdmi.c for HDMI/DP audio support, i.e. to be able to share this code between snd-hda-intel and SOF Intel (and not using hdac-hdmi). Let me know if this clashes with something you are already looking into. I have a very rough version up and running, but it still needs some work. If the general idea seems ok to you, I'll continue to work on a RFC patch and send for review. This will change how HDMI is exposed to user-space with SOF Intel drivers, so we need to be extra careful how this is introduced. But this really seems to be the best way to go to avoid the duplicated maintenance work with two drivers that we now have. PS Tracked in SOF github as https://github.com/thesofproject/linux/issues/1123 Br, Kai

2 6

[alsa-devel] [RFC PATCH 00/40] soundwire: updates for 5.4
by Pierre-Louis Bossart 23 Aug '19

23 Aug '19

The existing upstream code allows for SoundWire devices to be enumerated and managed by the bus, but streaming is not currently supported. Bard Liao, Rander Wang and I did quite a bit of integration/validation work to close this gap and we now have SoundWire streaming + basic power managemement on Intel CometLake and IceLake reference boards. These changes are still preliminary and should not be merged as is, but it's time to start reviews. While the number of patches is quite large, each of the changes is quite small. SOF driver changes will be submitted shortly as well but are still being validated. ClockStop modes and synchronized playback on multiple links are not supported for now and will likely be part of the next cycle (dependencies on codec drivers and multi-cpu DAI support). Acknowledgements: This work would not have been possible without the support of Slawomir Blauciak and Tomasz Lauda on the SOF side, currently being reviewed, see https://github.com/thesofproject/sof/pull/1638 Comments and feedback welcome! Bard liao (1): soundwire: include mod_devicetable.h to avoid compiling warnings Pierre-Louis Bossart (38): soundwire: add debugfs support soundwire: cadence_master: add debugfs register dump soundwire: cadence_master: align debugfs to 8 digits soundwire: intel: add debugfs register dump soundwire: intel: move interrupt enable after interrupt handler registration soundwire: intel: prevent possible dereference in hw_params soundwire: intel: fix channel number reported by hardware soundwire: intel: remove BIOS work-arounds soundwire: cadence_master: fix usage of CONFIG_UPDATE soundwire: cadence_master: remove useless wrapper soundwire: cadence_master: simplify bus clash interrupt clear soundwire: cadence_master: revisit interrupt settings soundwire: cadence_master: fix register definition for SLAVE_STATE soundwire: cadence_master: fix definitions for INTSTAT0/1 soundwire: cadence_master: handle multiple status reports per Slave soundwire: cadence_master: improve startup sequence with link hw_reset soundwire: bus: use runtime_pm_get_sync/pm when enabled soundwire: bus: split handling of Device0 events soundwire: bus: improve dynamic debug comments for enumeration soundwire: prototypes for suspend/resume soundwire: export helpers to find row and column values soundwire: stream: fix disable sequence soundwire: cadence_master: use BIOS defaults for frame shape soundwire: intel: use BIOS information to set clock dividers soundwire: Add Intel resource management algorithm soundwire: intel: handle disabled links soundwire: intel_init: add kernel module parameter to filter out links soundwire: cadence_master: add kernel parameter to override interrupt mask soundwire: intel: move shutdown() callback and don't export symbol soundwire: intel: add helper for initialization soundwire: intel: Add basic power management support soundwire: intel: ignore disabled links for suspend/resume soundwire: intel: export helper to exit reset soundwire: intel: disable interrupts on suspend soundwire: cadence_master: add hw_reset capability in debugfs soundwire: cadence_master: make clock stop exit configurable on init soundwire: intel: add pm_runtime support soundwire: intel: add delay on restart for enumeration Rander Wang (1): soundwire: cadence_master: fix divider setting in clock register drivers/soundwire/Makefile | 4 +- drivers/soundwire/algo_dynamic_allocation.c | 403 ++++++++++++++++++++ drivers/soundwire/bus.c | 44 ++- drivers/soundwire/bus.h | 77 +++- drivers/soundwire/bus_type.c | 3 + drivers/soundwire/cadence_master.c | 365 ++++++++++++++---- drivers/soundwire/cadence_master.h | 12 +- drivers/soundwire/debugfs.c | 156 ++++++++ drivers/soundwire/intel.c | 381 +++++++++++++++++- drivers/soundwire/intel_init.c | 14 + drivers/soundwire/slave.c | 1 + drivers/soundwire/stream.c | 53 ++- include/linux/soundwire/sdw.h | 15 + 13 files changed, 1414 insertions(+), 114 deletions(-) create mode 100644 drivers/soundwire/algo_dynamic_allocation.c create mode 100644 drivers/soundwire/debugfs.c -- 2.20.1

10 182

[alsa-devel] [PATCH v3 0/4] soundwire: debugfs support for 5.4
by Pierre-Louis Bossart 23 Aug '19

23 Aug '19

This patchset enables debugfs support and corrects all the feedback provided on an earlier RFC ('soundwire: updates for 5.4') There is one remaining hard-coded value in intel.c that will need to be fixed in a follow-up patchset not specific to debugfs: we need to remove hard-coded Intel-specific configurations from cadence_master.c (PDI offsets, etc). Changes since v2: No code change, just rebase to soundwire/next Added GKH and Sanyog's tags Also added patch4 submitted earlier in another series which depends on debugfs Changes since v1 (Feedback from GKH) Handle debugfs in a more self-contained way (no dentry as return or parameter) Used CONFIG_DEBUG_FS in structures and code to make it easier to remove if need be. No functional change for register dumps. Changes since RFC (Feedback from GKH, Vinod, Guennadi, Cezary, Sanyog): removed error checks used DEFINE_SHOW_ATTRIBUTE and seq_file fixed copyright dates fixed SPDX license info to use GPL2.0 only fixed Makefile to include debugfs only if CONFIG_DEBUG_FS is selected used static inlines for fallback compilation removed intermediate variables removed hard-coded constants in loops (used registers offsets and hardware capabilities) squashed patch 3 Pierre-Louis Bossart (4): soundwire: add debugfs support soundwire: cadence_master: add debugfs register dump soundwire: intel: add debugfs register dump soundwire: intel: handle disabled links drivers/soundwire/Makefile | 4 + drivers/soundwire/bus.c | 6 ++ drivers/soundwire/bus.h | 16 +++ drivers/soundwire/bus_type.c | 3 + drivers/soundwire/cadence_master.c | 107 ++++++++++++++++++++ drivers/soundwire/cadence_master.h | 4 + drivers/soundwire/debugfs.c | 151 +++++++++++++++++++++++++++++ drivers/soundwire/intel.c | 145 ++++++++++++++++++++++++++- drivers/soundwire/slave.c | 1 + include/linux/soundwire/sdw.h | 10 ++ 10 files changed, 444 insertions(+), 3 deletions(-) create mode 100644 drivers/soundwire/debugfs.c base-commit: 183c7687802e4132eb782808a8bf80689a9219c1 -- 2.20.1

2 6

[alsa-devel] [PATCH 0/5] ASoC: atmel: extend SSC support
by Michał Mirosław 23 Aug '19

23 Aug '19

This series improves support for various configurations using SSC module as implemented in Atmel SAMA5Dx SoCs. Patches are: 1. enable SSC in Kconfig for audio-graph-card support 2-3. implement left-justified data mode 4-5. fix shared FSYNC source for slave mode Patches against tiwai/sound/for-next tree. You can also pull from https://rere.qmqm.pl/git/linux branch: atmel-ssc Michał Mirosław (5): ASoC: atmel: enable SSC_PCM_DMA in Kconfig ASoC: atmel_ssc_dai: rework DAI format configuration ASoC: atmel_ssc_dai: implement left-justified data mode ASoC: atmel_ssc_dai: split TX/RX FS constants ASoC: atmel_ssc_dai: Enable shared FSYNC source in frame-slave mode .../devicetree/bindings/misc/atmel-ssc.txt | 4 + drivers/misc/atmel-ssc.c | 2 + include/linux/atmel-ssc.h | 1 + sound/soc/atmel/Kconfig | 2 +- sound/soc/atmel/atmel_ssc_dai.c | 306 ++++++------------ sound/soc/atmel/atmel_ssc_dai.h | 9 +- 6 files changed, 120 insertions(+), 204 deletions(-) -- 2.20.1

5 22

[alsa-devel] [PATCH] soundwire: intel: remove unused variables
by Vinod Koul 23 Aug '19

23 Aug '19

Variables 'nval' and 'i' are no longer used sdw_master_read_intel_prop() so remove them. drivers/soundwire/intel.c: In function 'sdw_master_read_intel_prop': drivers/soundwire/intel.c:829:12: warning: unused variable 'i' [-Wunused-variable] int nval, i; ^ drivers/soundwire/intel.c:829:6: warning: unused variable 'nval' [-Wunused-variable] int nval, i; ^~~~ Reported-by: Stephen Rothwell <sfr(a)canb.auug.org.au> Fixes: 085f4ace103d ("soundwire: intel: read mclk_freq property from firmware") Signed-off-by: Vinod Koul <vkoul(a)kernel.org> --- drivers/soundwire/intel.c | 1 - 1 file changed, 1 deletion(-) diff --git a/drivers/soundwire/intel.c b/drivers/soundwire/intel.c index 459cc1e6355d..cad378f741c0 100644 --- a/drivers/soundwire/intel.c +++ b/drivers/soundwire/intel.c @@ -826,7 +826,6 @@ static int sdw_master_read_intel_prop(struct sdw_bus *bus) struct sdw_master_prop *prop = &bus->prop; struct fwnode_handle *link; char name[32]; - int nval, i; /* Find master handle */ snprintf(name, sizeof(name), -- 2.20.1

2 2

[alsa-devel] [PATCH v4 0/4] ASoC: codecs: Add WSA881x Smart Speaker amplifier support
by Srinivas Kandagatla 23 Aug '19

23 Aug '19

Thanks for reviewing v3 patchset, here is v4 with addressing the comments in v3 This patchset adds support to WSA8810/WSA8815 Class-D Smart Speaker Amplifier which is SoundWire interfaced. This also adds support to some missing bits in SoundWire bus layer like Device Tree support. This patchset along with DB845c machine driver and WCD934x codec driver has been tested on SDM845 SoC based DragonBoard DB845c with two WSA8810 speakers. Most of the code in this driver is rework of Qualcomm downstream drivers used in Andriod. Credits to Banajit Goswami and Patrick Lai's Team. TODO: Add thermal sensor support in WSA881x. Thanks, srini Changes since v3: - updated slave bindings according to Rob's Suggestion. - moved bindings to yaml Srinivas Kandagatla (4): dt-bindings: soundwire: add slave bindings soundwire: core: add device tree support for slave devices dt-bindings: ASoC: Add WSA881x bindings ASoC: codecs: add wsa881x amplifier support .../bindings/sound/qcom,wsa881x.yaml | 44 + .../soundwire/soudwire-controller.yaml | 75 ++ drivers/soundwire/bus.c | 2 + drivers/soundwire/bus.h | 1 + drivers/soundwire/slave.c | 52 + sound/soc/codecs/Kconfig | 10 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/wsa881x.c | 1134 +++++++++++++++++ 8 files changed, 1320 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/qcom,wsa881x.yaml create mode 100644 Documentation/devicetree/bindings/soundwire/soudwire-controller.yaml create mode 100644 sound/soc/codecs/wsa881x.c -- 2.21.0

1 4

[alsa-devel] [PATCH 0/2] ASoC: MediaTek: fix mt8183-mt6358 tdm out issues
by Jiaxin Yu 22 Aug '19

22 Aug '19

These two patches are used to fix several issues related to TDM OUT. The patches are based on broonie tre "for-next" branch. Jiaxin Yu (2): ASoC: mediatek: mt8183: switch tdm pins gpio function when playback on or off ASoC: mediatek: mt8183: fix tdm out issue sound/soc/mediatek/common/mtk-afe-fe-dai.c | 3 +- sound/soc/mediatek/mt8183/mt8183-dai-tdm.c | 177 +++++++++++++++++---- .../mt8183/mt8183-mt6358-ts3a227-max98357.c | 118 +++++++++++++- sound/soc/mediatek/mt8183/mt8183-reg.h | 8 +- 4 files changed, 261 insertions(+), 45 deletions(-) -- 1.8.1.1.dirty

2 4

[alsa-devel] [PATCH -next] ASoC: AMD: Fix Kconfig warning without GPIOLIB
by YueHaibing 22 Aug '19

22 Aug '19

While do rand build without GPIOLIB, we get Kconfig warning:\ WARNING: unmet direct dependencies detected for SND_SOC_MAX98357A Depends on [n]: SOUND [=y] && !UML && SND [=m] && SND_SOC [=m] && GPIOLIB [=n] Selected by [m]: - SND_SOC_AMD_CZ_DA7219MX98357_MACH [=m] && SOUND [=y] && !UML && SND [=m] && SND_SOC [=m] && SND_SOC_AMD_ACP [=m] && I2C [=y] Add GPIOLIB dependency to fix this. Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/soc/amd/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/amd/Kconfig b/sound/soc/amd/Kconfig index 9ca9214..5f40517 100644 --- a/sound/soc/amd/Kconfig +++ b/sound/soc/amd/Kconfig @@ -10,7 +10,7 @@ config SND_SOC_AMD_CZ_DA7219MX98357_MACH select SND_SOC_MAX98357A select SND_SOC_ADAU7002 select REGULATOR - depends on SND_SOC_AMD_ACP && I2C + depends on SND_SOC_AMD_ACP && I2C && GPIOLIB help This option enables machine driver for DA7219 and MAX9835. -- 2.7.4

2 1

[alsa-devel] [PATCH -next] ASoC: wm8988: fix typo in wm8988_right_line_controls
by YueHaibing 22 Aug '19

22 Aug '19

sound/soc/codecs/wm8988.c:270:30: warning: wm8988_rline_enum defined but not used [-Wunused-const-variable=] wm8988_rline_enum should be used in wm8988_right_line_controls. Reported-by: Hulk Robot <hulkci(a)huawei.com> Fixes: 5409fb4e327a ("ASoC: Add WM8988 CODEC driver") Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/soc/codecs/wm8988.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8988.c b/sound/soc/codecs/wm8988.c index 25e74cf..85bfd04 100644 --- a/sound/soc/codecs/wm8988.c +++ b/sound/soc/codecs/wm8988.c @@ -273,7 +273,7 @@ static const struct soc_enum wm8988_rline_enum = wm8988_line_texts, wm8988_line_values); static const struct snd_kcontrol_new wm8988_right_line_controls = - SOC_DAPM_ENUM("Route", wm8988_lline_enum); + SOC_DAPM_ENUM("Route", wm8988_rline_enum); /* Left Mixer */ static const struct snd_kcontrol_new wm8988_left_mixer_controls[] = { -- 2.7.4

2 1

[alsa-devel] [PATCH -next] ASoC: mediatek: mt2701: Fix -Wunused-const-variable warnings
by YueHaibing 22 Aug '19

22 Aug '19

sound/soc/mediatek/mt2701/mt2701-afe-common.h:66:27: warning: mt2701_afe_backup_list defined but not used [-Wunused-const-variable=] mt2701_afe_backup_list is only used in mt2701-afe-pcm.c, so just move the definition over there. Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/soc/mediatek/mt2701/mt2701-afe-common.h | 21 --------------------- sound/soc/mediatek/mt2701/mt2701-afe-pcm.c | 21 +++++++++++++++++++++ 2 files changed, 21 insertions(+), 21 deletions(-) diff --git a/sound/soc/mediatek/mt2701/mt2701-afe-common.h b/sound/soc/mediatek/mt2701/mt2701-afe-common.h index d44faba..32bef5e 100644 --- a/sound/soc/mediatek/mt2701/mt2701-afe-common.h +++ b/sound/soc/mediatek/mt2701/mt2701-afe-common.h @@ -63,27 +63,6 @@ enum audio_base_clock { MT2701_BASE_CLK_NUM, }; -static const unsigned int mt2701_afe_backup_list[] = { - AUDIO_TOP_CON0, - AUDIO_TOP_CON4, - AUDIO_TOP_CON5, - ASYS_TOP_CON, - AFE_CONN0, - AFE_CONN1, - AFE_CONN2, - AFE_CONN3, - AFE_CONN15, - AFE_CONN16, - AFE_CONN17, - AFE_CONN18, - AFE_CONN19, - AFE_CONN20, - AFE_CONN21, - AFE_CONN22, - AFE_DAC_CON0, - AFE_MEMIF_PBUF_SIZE, -}; - struct mt2701_i2s_data { int i2s_ctrl_reg; int i2s_asrc_fs_shift; diff --git a/sound/soc/mediatek/mt2701/mt2701-afe-pcm.c b/sound/soc/mediatek/mt2701/mt2701-afe-pcm.c index d7f5def..76502ba 100644 --- a/sound/soc/mediatek/mt2701/mt2701-afe-pcm.c +++ b/sound/soc/mediatek/mt2701/mt2701-afe-pcm.c @@ -60,6 +60,27 @@ static const struct mt2701_afe_rate mt2701_afe_i2s_rates[] = { { .rate = 352800, .regvalue = 24 }, }; +static const unsigned int mt2701_afe_backup_list[] = { + AUDIO_TOP_CON0, + AUDIO_TOP_CON4, + AUDIO_TOP_CON5, + ASYS_TOP_CON, + AFE_CONN0, + AFE_CONN1, + AFE_CONN2, + AFE_CONN3, + AFE_CONN15, + AFE_CONN16, + AFE_CONN17, + AFE_CONN18, + AFE_CONN19, + AFE_CONN20, + AFE_CONN21, + AFE_CONN22, + AFE_DAC_CON0, + AFE_MEMIF_PBUF_SIZE, +}; + static int mt2701_dai_num_to_i2s(struct mtk_base_afe *afe, int num) { struct mt2701_afe_private *afe_priv = afe->platform_priv; -- 2.7.4

2 1

[alsa-devel] [PATCH resend 00/11] ASoC: soc-core cleanup repost
by Kuninori Morimoto 22 Aug '19

22 Aug '19

Hi Mark I posted soc-core cleanup patches, almost all are already accepted, and I got response/review for some patches. I adjusted to not-yet-accepted patches Kuninori Morimoto (11): ASoC: soc-core: use device_register() ASoC: soc-core: merge snd_soc_initialize_card_lists() ASoC: soc-core: remove unneeded dai_link check from snd_soc_remove_dai_link() ASoC: soc-core: add NOTE to snd_soc_rtdcom_lookup() ASoC: soc-core: don't alloc memory carelessly when creating debugfs ASoC: soc-core: soc_cleanup_card_resources() become void ASoC: soc-core: initialize component list ASoC: soc-core: initialize list at one place ASoC: soc-dai: use bit field for bus_control ASoC: soc-topology: use for_each_component_dais() at remove_dai() ASoC: soc-core: add soc_unbind_aux_dev() include/sound/soc-dai.h | 3 +- include/sound/soc.h | 10 ------- sound/soc/soc-core.c | 77 +++++++++++++++++++++++++----------------------- sound/soc/soc-topology.c | 2 +- 4 files changed, 42 insertions(+), 50 deletions(-) -- 2.7.4

2 23

[alsa-devel] [PATCH v2 0/3] ASoC: SOF: initial support for i.MX8
by Pierre-Louis Bossart 22 Aug '19

22 Aug '19

The reviews for these patches took place already on mailing lists and GitHub, and the code is already integrated in the SOF tree (along with dependencies already accepted in the i.MX tree) Changes since v1: As agreed with Daniel and Mark, this v2 series includes the DT bindings (Fixed 'compatible' typo in examples) The SAI/ESAI support was merged separately so dropped from this set. Daniel Baluta (3): dt-bindings: dsp: fsl: Add DSP core binding support ASoC: SOF: Add OF DSP device support ASoC: SOF: imx: Add i.MX8 HW support .../devicetree/bindings/dsp/fsl,dsp.yaml | 88 ++++ sound/soc/sof/Kconfig | 11 + sound/soc/sof/Makefile | 4 + sound/soc/sof/imx/Kconfig | 22 + sound/soc/sof/imx/Makefile | 4 + sound/soc/sof/imx/imx8.c | 394 ++++++++++++++++++ sound/soc/sof/sof-of-dev.c | 143 +++++++ 7 files changed, 666 insertions(+) create mode 100644 Documentation/devicetree/bindings/dsp/fsl,dsp.yaml create mode 100644 sound/soc/sof/imx/Kconfig create mode 100644 sound/soc/sof/imx/Makefile create mode 100644 sound/soc/sof/imx/imx8.c create mode 100644 sound/soc/sof/sof-of-dev.c -- 2.20.1

2 6

[alsa-devel] [PATCH v2] ASoC: SOF: topology: fix get control data return type and arguments
by Jaska Uimonen 22 Aug '19

22 Aug '19

From: Jaska Uimonen <jaska.uimonen(a)intel.com> sof_get_control_data returns negative values even though the return value is defined unsigned (size_t). So change the return value type to int and add the data size as pointer argument to sof_get_control_data to avoid ambiquity in the meaning of the return type. Fixes: cac974a51ebb ("ASoC: SOF: topology: use set_get_data in process load") Reported by: Dan Carpenter <dan.carpenter(a)oracle.com> Signed-off-by: Jaska Uimonen <jaska.uimonen(a)intel.com> --- sound/soc/sof/topology.c | 21 +++++++++++---------- 1 file changed, 11 insertions(+), 10 deletions(-) diff --git a/sound/soc/sof/topology.c b/sound/soc/sof/topology.c index 28a7a6e06a53..fc85efbad378 100644 --- a/sound/soc/sof/topology.c +++ b/sound/soc/sof/topology.c @@ -1752,17 +1752,19 @@ static int sof_widget_load_siggen(struct snd_soc_component *scomp, int index, return ret; } -static size_t sof_get_control_data(struct snd_sof_dev *sdev, - struct snd_soc_dapm_widget *widget, - struct sof_widget_data *wdata) +static int sof_get_control_data(struct snd_sof_dev *sdev, + struct snd_soc_dapm_widget *widget, + struct sof_widget_data *wdata, + size_t *size) { const struct snd_kcontrol_new *kc; struct soc_mixer_control *sm; struct soc_bytes_ext *sbe; struct soc_enum *se; - size_t size = 0; int i; + *size = 0; + for (i = 0; i < widget->num_kcontrols; i++) { kc = &widget->kcontrol_news[i]; @@ -1800,7 +1802,7 @@ static size_t sof_get_control_data(struct snd_sof_dev *sdev, if (wdata[i].pdata->magic != SOF_ABI_MAGIC) return -EINVAL; - size += wdata[i].pdata->size; + *size += wdata[i].pdata->size; /* get data type */ switch (wdata[i].control->cmd) { @@ -1819,7 +1821,7 @@ static size_t sof_get_control_data(struct snd_sof_dev *sdev, } } - return size; + return 0; } static int sof_process_load(struct snd_soc_component *scomp, int index, @@ -1855,12 +1857,11 @@ static int sof_process_load(struct snd_soc_component *scomp, int index, return -ENOMEM; /* get possible component controls and get size of all pdata */ - ipc_data_size = sof_get_control_data(sdev, widget, wdata); + ret = sof_get_control_data(sdev, widget, wdata, + &ipc_data_size); - if (ipc_data_size <= 0) { - ret = ipc_data_size; + if (ret < 0) goto out; - } } ipc_size = sizeof(struct sof_ipc_comp_process) + -- 2.13.6

2 1

[alsa-devel] [PATCH 2/3] input: Add SW_UNSUPPORT_INSERT define
by bgoswami＠codeaurora.org 22 Aug '19

22 Aug '19

From: Banajit Goswami <bgoswami(a)codeaurora.org> Some devices may not support specific type of input devices. For example, when a headset or extension cable with GND/MIC swap is plugged into a headset jack that does not support the headset/cable, it needs to be reported with a corresponding input event. Also, increase the max values for INPUT_DEVICE_ID_SW_MAX and SW_MAX, to accommodate future extension of the number of event. Signed-off-by: Gopikrishnaiah Anandan <agopik(a)codeaurora.org> Signed-off-by: Banajit Goswami <bgoswami(a)codeaurora.org> --- include/linux/mod_devicetable.h | 2 +- include/uapi/linux/input-event-codes.h | 3 ++- 2 files changed, 3 insertions(+), 2 deletions(-) diff --git a/include/linux/mod_devicetable.h b/include/linux/mod_devicetable.h index 448621c..7586099 100644 --- a/include/linux/mod_devicetable.h +++ b/include/linux/mod_devicetable.h @@ -299,7 +299,7 @@ struct pcmcia_device_id { #define INPUT_DEVICE_ID_LED_MAX 0x0f #define INPUT_DEVICE_ID_SND_MAX 0x07 #define INPUT_DEVICE_ID_FF_MAX 0x7f -#define INPUT_DEVICE_ID_SW_MAX 0x0f +#define INPUT_DEVICE_ID_SW_MAX 0x1f #define INPUT_DEVICE_ID_PROP_MAX 0x1f #define INPUT_DEVICE_ID_MATCH_BUS 1 diff --git a/include/uapi/linux/input-event-codes.h b/include/uapi/linux/input-event-codes.h index 85387c7..960fa86 100644 --- a/include/uapi/linux/input-event-codes.h +++ b/include/uapi/linux/input-event-codes.h @@ -808,7 +808,8 @@ #define SW_LINEIN_INSERT 0x0d /* set = inserted */ #define SW_MUTE_DEVICE 0x0e /* set = device disabled */ #define SW_PEN_INSERTED 0x0f /* set = pen inserted */ -#define SW_MAX 0x0f +#define SW_UNSUPPORT_INSERT 0x10 /* set = unsupported device inserted */ +#define SW_MAX 0x1f #define SW_CNT (SW_MAX+1) /* -- The Qualcomm Innovation Center, Inc. is a member of the Code Aurora Forum, a Linux Foundation Collaborative Project

2 1

[alsa-devel] [PATCH v3 0/4] ASoC: codecs: Add WSA881x Smart Speaker amplifier support
by Srinivas Kandagatla 22 Aug '19

22 Aug '19

Thanks for reviewing v2 patchset, here is v3 with addressing the comments in v2. This patchset adds support to WSA8810/WSA8815 Class-D Smart Speaker Amplifier which is SoundWire interfaced. This also adds support to some missing bits in SoundWire bus layer like Device Tree support. This patchset along with DB845c machine driver and WCD934x codec driver has been tested on SDM845 SoC based DragonBoard DB845c with two WSA8810 speakers. Most of the code in this driver is rework of Qualcomm downstream drivers used in Andriod. Credits to Banajit Goswami and Patrick Lai's Team. TODO: Add thermal sensor support in WSA881x. Thanks, srini Changes since v2: - Updated compatible string to include LinkID. - udpdated wsa driver to not register/unregister component in SoundWire status callbacks. - Updated few minor coding style review comments. Changes since v1 RFC: - bindings document renamed to slave.txt - fix error code from dt slave parsing Srinivas Kandagatla (4): dt-bindings: soundwire: add slave bindings soundwire: core: add device tree support for slave devices dt-bindings: ASoC: Add WSA881x bindings ASoC: codecs: add wsa881x amplifier support .../bindings/sound/qcom,wsa881x.txt | 24 + .../devicetree/bindings/soundwire/slave.txt | 51 + drivers/soundwire/bus.c | 2 + drivers/soundwire/bus.h | 1 + drivers/soundwire/slave.c | 44 + sound/soc/codecs/Kconfig | 10 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/wsa881x.c | 1134 +++++++++++++++++ 8 files changed, 1268 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/qcom,wsa881x.txt create mode 100644 Documentation/devicetree/bindings/soundwire/slave.txt create mode 100644 sound/soc/codecs/wsa881x.c -- 2.21.0

4 14

[alsa-devel] [PATCH -next] ALSA: line6: Fix -Wunused-const-variable warnings
by YueHaibing 22 Aug '19

22 Aug '19

sound/usb/line6/driver.h:69:18: warning: SYSEX_DATA_OFS defined but not used [-Wunused-const-variable=] sound/usb/line6/driver.h:70:18: warning: SYSEX_EXTRA_SIZE defined but not used [-Wunused-const-variable=] SYSEX_EXTRA_SIZE is only used in driver.c and SYSEX_DATA_OFS in pod.c, so move them to .c fix this warning. Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/usb/line6/driver.c | 2 ++ sound/usb/line6/driver.h | 3 --- sound/usb/line6/pod.c | 2 ++ 3 files changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/usb/line6/driver.c b/sound/usb/line6/driver.c index b5a3f75..8027da8 100644 --- a/sound/usb/line6/driver.c +++ b/sound/usb/line6/driver.c @@ -31,6 +31,8 @@ const unsigned char line6_midi_id[3] = { }; EXPORT_SYMBOL_GPL(line6_midi_id); +static const int SYSEX_EXTRA_SIZE = sizeof(line6_midi_id) + 4; + /* Code to request version of POD, Variax interface (and maybe other devices). diff --git a/sound/usb/line6/driver.h b/sound/usb/line6/driver.h index e5e572e..c57633a 100644 --- a/sound/usb/line6/driver.h +++ b/sound/usb/line6/driver.h @@ -66,9 +66,6 @@ extern const unsigned char line6_midi_id[3]; -static const int SYSEX_DATA_OFS = sizeof(line6_midi_id) + 3; -static const int SYSEX_EXTRA_SIZE = sizeof(line6_midi_id) + 4; - /* Common properties of Line 6 devices. */ diff --git a/sound/usb/line6/pod.c b/sound/usb/line6/pod.c index ee4c9d2..24ad337 100644 --- a/sound/usb/line6/pod.c +++ b/sound/usb/line6/pod.c @@ -110,6 +110,8 @@ enum { POD_BUSY_MIDISEND }; +static const int SYSEX_DATA_OFS = sizeof(line6_midi_id) + 3; + static struct snd_ratden pod_ratden = { .num_min = 78125, .num_max = 78125, -- 2.7.4

2 1

[alsa-devel] KASAN: use-after-free Read in snd_seq_ioctl_get_client_pool
by syzbot 22 Aug '19

22 Aug '19

Hello, syzbot found the following crash on: HEAD commit: d1abaeb3 Linux 5.3-rc5 git tree: upstream console output: https://syzkaller.appspot.com/x/log.txt?x=15b5add2600000 kernel config: https://syzkaller.appspot.com/x/.config?x=f7242b2607d5d53a dashboard link: https://syzkaller.appspot.com/bug?extid=4a75454b9ca2777f35c7 compiler: clang version 9.0.0 (/home/glider/llvm/clang 80fee25776c2fb61e74c1ecb1a523375c2500b69) Unfortunately, I don't have any reproducer for this crash yet. IMPORTANT: if you fix the bug, please add the following tag to the commit: Reported-by: syzbot+4a75454b9ca2777f35c7(a)syzkaller.appspotmail.com ================================================================== BUG: KASAN: use-after-free in atomic_read include/asm-generic/atomic-instrumented.h:26 [inline] BUG: KASAN: use-after-free in snd_seq_unused_cells sound/core/seq/seq_memory.h:60 [inline] BUG: KASAN: use-after-free in snd_seq_ioctl_get_client_pool+0x2e0/0x680 sound/core/seq/seq_clientmgr.c:1839 Read of size 4 at addr ffff8880a8e27b14 by task syz-executor.5/31741 CPU: 0 PID: 31741 Comm: syz-executor.5 Not tainted 5.3.0-rc5 #82 Hardware name: Google Google Compute Engine/Google Compute Engine, BIOS Google 01/01/2011 Call Trace: __dump_stack lib/dump_stack.c:77 [inline] dump_stack+0x1d8/0x2f8 lib/dump_stack.c:113 print_address_description+0x75/0x5b0 mm/kasan/report.c:351 __kasan_report+0x14b/0x1c0 mm/kasan/report.c:482 kasan_report+0x26/0x50 mm/kasan/common.c:612 check_memory_region_inline mm/kasan/generic.c:182 [inline] check_memory_region+0x2cf/0x2e0 mm/kasan/generic.c:192 __kasan_check_read+0x11/0x20 mm/kasan/common.c:92 atomic_read include/asm-generic/atomic-instrumented.h:26 [inline] snd_seq_unused_cells sound/core/seq/seq_memory.h:60 [inline] snd_seq_ioctl_get_client_pool+0x2e0/0x680 sound/core/seq/seq_clientmgr.c:1839 snd_seq_ioctl+0x1d1/0x340 sound/core/seq/seq_clientmgr.c:2158 do_vfs_ioctl+0x744/0x1730 fs/ioctl.c:46 ksys_ioctl fs/ioctl.c:713 [inline] __do_sys_ioctl fs/ioctl.c:720 [inline] __se_sys_ioctl fs/ioctl.c:718 [inline] __x64_sys_ioctl+0xe3/0x120 fs/ioctl.c:718 do_syscall_64+0xfe/0x140 arch/x86/entry/common.c:296 entry_SYSCALL_64_after_hwframe+0x49/0xbe RIP: 0033:0x459829 Code: fd b7 fb ff c3 66 2e 0f 1f 84 00 00 00 00 00 66 90 48 89 f8 48 89 f7 48 89 d6 48 89 ca 4d 89 c2 4d 89 c8 4c 8b 4c 24 08 0f 05 <48> 3d 01 f0 ff ff 0f 83 cb b7 fb ff c3 66 2e 0f 1f 84 00 00 00 00 RSP: 002b:00007faf5e0c4c78 EFLAGS: 00000246 ORIG_RAX: 0000000000000010 RAX: ffffffffffffffda RBX: 0000000000000003 RCX: 0000000000459829 RDX: 00000000200003c0 RSI: 00000000c058534b RDI: 0000000000000003 RBP: 000000000075bf20 R08: 0000000000000000 R09: 0000000000000000 R10: 0000000000000000 R11: 0000000000000246 R12: 00007faf5e0c56d4 R13: 00000000004cdc18 R14: 00000000004d7ba8 R15: 00000000ffffffff Allocated by task 31737: save_stack mm/kasan/common.c:69 [inline] set_track mm/kasan/common.c:77 [inline] __kasan_kmalloc+0x11c/0x1b0 mm/kasan/common.c:487 kasan_kmalloc+0x9/0x10 mm/kasan/common.c:501 kmem_cache_alloc_trace+0x221/0x2f0 mm/slab.c:3550 kmalloc include/linux/slab.h:552 [inline] kzalloc include/linux/slab.h:748 [inline] snd_seq_pool_new+0x53/0x1b0 sound/core/seq/seq_memory.c:459 snd_seq_fifo_new+0x72/0x1c0 sound/core/seq/seq_fifo.c:26 snd_seq_open+0x1e4/0x470 sound/core/seq/seq_clientmgr.c:350 snd_open+0x313/0x380 sound/core/sound.c:163 chrdev_open+0x509/0x590 fs/char_dev.c:414 do_dentry_open+0x73b/0xf90 fs/open.c:797 vfs_open+0x73/0x80 fs/open.c:906 do_last fs/namei.c:3416 [inline] path_openat+0x1397/0x4460 fs/namei.c:3533 do_filp_open+0x192/0x3d0 fs/namei.c:3563 do_sys_open+0x29f/0x560 fs/open.c:1089 __do_sys_open fs/open.c:1107 [inline] __se_sys_open fs/open.c:1102 [inline] __x64_sys_open+0x87/0x90 fs/open.c:1102 do_syscall_64+0xfe/0x140 arch/x86/entry/common.c:296 entry_SYSCALL_64_after_hwframe+0x49/0xbe Freed by task 31737: save_stack mm/kasan/common.c:69 [inline] set_track mm/kasan/common.c:77 [inline] __kasan_slab_free+0x12a/0x1e0 mm/kasan/common.c:449 kasan_slab_free+0xe/0x10 mm/kasan/common.c:457 __cache_free mm/slab.c:3425 [inline] kfree+0x115/0x200 mm/slab.c:3756 snd_seq_pool_delete+0x9e/0xd0 sound/core/seq/seq_memory.c:487 snd_seq_fifo_resize+0x1de/0x250 sound/core/seq/seq_fifo.c:262 snd_seq_ioctl_set_client_pool+0x3c4/0x740 sound/core/seq/seq_clientmgr.c:1880 snd_seq_ioctl+0x1d1/0x340 sound/core/seq/seq_clientmgr.c:2158 do_vfs_ioctl+0x744/0x1730 fs/ioctl.c:46 ksys_ioctl fs/ioctl.c:713 [inline] __do_sys_ioctl fs/ioctl.c:720 [inline] __se_sys_ioctl fs/ioctl.c:718 [inline] __x64_sys_ioctl+0xe3/0x120 fs/ioctl.c:718 do_syscall_64+0xfe/0x140 arch/x86/entry/common.c:296 entry_SYSCALL_64_after_hwframe+0x49/0xbe The buggy address belongs to the object at ffff8880a8e27b00 which belongs to the cache kmalloc-192 of size 192 The buggy address is located 20 bytes inside of 192-byte region [ffff8880a8e27b00, ffff8880a8e27bc0) The buggy address belongs to the page: page:ffffea0002a389c0 refcount:1 mapcount:0 mapping:ffff8880aa400000 index:0x0 flags: 0x1fffc0000000200(slab) raw: 01fffc0000000200 ffffea00026d1008 ffffea0002870048 ffff8880aa400000 raw: 0000000000000000 ffff8880a8e27000 0000000100000010 0000000000000000 page dumped because: kasan: bad access detected Memory state around the buggy address: ffff8880a8e27a00: fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb ffff8880a8e27a80: fb fb fb fb fb fb fb fb fc fc fc fc fc fc fc fc > ffff8880a8e27b00: fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb ^ ffff8880a8e27b80: fb fb fb fb fb fb fb fb fc fc fc fc fc fc fc fc ffff8880a8e27c00: fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb ================================================================== --- This bug is generated by a bot. It may contain errors. See https://goo.gl/tpsmEJ for more information about syzbot. syzbot engineers can be reached at syzkaller(a)googlegroups.com. syzbot will keep track of this bug report. See: https://goo.gl/tpsmEJ#status for how to communicate with syzbot.

1 0

[alsa-devel] [PATCH -next] ASoC: sun4i-i2s: Use PTR_ERR_OR_ZERO in sun4i_i2s_init_regmap_fields()
by YueHaibing 22 Aug '19

22 Aug '19

Use PTR_ERR_OR_ZERO rather than if(IS_ERR(...)) + PTR_ERR Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/soc/sunxi/sun4i-i2s.c | 5 +---- 1 file changed, 1 insertion(+), 4 deletions(-) diff --git a/sound/soc/sunxi/sun4i-i2s.c b/sound/soc/sunxi/sun4i-i2s.c index 9e691baee1e8..2071c54265f3 100644 --- a/sound/soc/sunxi/sun4i-i2s.c +++ b/sound/soc/sunxi/sun4i-i2s.c @@ -1095,10 +1095,7 @@ static int sun4i_i2s_init_regmap_fields(struct device *dev, i2s->field_fmt_sr = devm_regmap_field_alloc(dev, i2s->regmap, i2s->variant->field_fmt_sr); - if (IS_ERR(i2s->field_fmt_sr)) - return PTR_ERR(i2s->field_fmt_sr); - - return 0; + return PTR_ERR_OR_ZERO(i2s->field_fmt_sr); } static int sun4i_i2s_probe(struct platform_device *pdev)

3 3

[alsa-devel] What to do when a device clears the buffer in the start trigger?
by Paul Pawlowski 22 Aug '19

22 Aug '19

Hello, I have a device which clears the DMA buffer when I command it to start the playback. How can this be workarounded? I found the SNDRV_PCM_INFO_DOUBLE flag, but it doesn't seem that it changes alsa/alsa-lib behaviour. I wasn't able to find any drivers which shared this quirk. I'd like to avoid double buffering if possible, as once the stream is actually started it's not required. Thank you, Paul Pawlowski

2 3

[alsa-devel] [PATCH] ALSA: usb-audio: Fix the mixer control range limiting issue
by Srinivas Kandagatla 22 Aug '19

22 Aug '19

From: Deepa Madiregama <dmadireg(a)codeaurora.org> - mixer_ctl_set() function is limiting the volume level to particular range. This results in incorrect initial volume setting for that device. - In USB mixer while calculating the dBmin/dBmax values resolution factor is hardcoded to 256 which results in populating the wrong values for dBmin/dBmax. - Fix is to use appropriate resolution factor while calculating the dBmin/dBmax values. Signed-off-by: Deepa Madiregama <dmadireg(a)codeaurora.org> Signed-off-by: Banajit Goswami <bgoswami(a)codeaurora.org> Signed-off-by: Meng Wang <mwang(a)codeaurora.org> Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- sound/usb/mixer.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 5070a6a76ab3..a67448327d07 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -1248,8 +1248,10 @@ static int get_min_max_with_quirks(struct usb_mixer_elem_info *cval, /* USB descriptions contain the dB scale in 1/256 dB unit * while ALSA TLV contains in 1/100 dB unit */ - cval->dBmin = (convert_signed_value(cval, cval->min) * 100) / 256; - cval->dBmax = (convert_signed_value(cval, cval->max) * 100) / 256; + cval->dBmin = + (convert_signed_value(cval, cval->min) * 100) / (cval->res); + cval->dBmax = + (convert_signed_value(cval, cval->max) * 100) / (cval->res); if (cval->dBmin > cval->dBmax) { /* something is wrong; assume it's either from/to 0dB */ if (cval->dBmin < 0) -- 2.21.0

2 2

[alsa-devel] [PATCH] ALSA: pcm: add support for 352.8KHz and 384KHz sample rate
by Srinivas Kandagatla 22 Aug '19

22 Aug '19

From: Vidyakumar Athota <vathota(a)codeaurora.org> Most of the modern codecs supports 352.8KHz and 384KHz sample rates. Currently HW params fails to set 352.8Kz and 384KHz sample rate as these are not in known rates list. Add these new rates to known list to allow them. This patch also adds defines in pcm.h so that drivers can use it. Signed-off-by: Vidyakumar Athota <vathota(a)codeaurora.org> Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- include/sound/pcm.h | 5 +++++ sound/core/pcm_native.c | 2 +- 2 files changed, 6 insertions(+), 1 deletion(-) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 1e9bb1c91770..bbe6eb1ff5d2 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -117,6 +117,8 @@ struct snd_pcm_ops { #define SNDRV_PCM_RATE_96000 (1<<10) /* 96000Hz */ #define SNDRV_PCM_RATE_176400 (1<<11) /* 176400Hz */ #define SNDRV_PCM_RATE_192000 (1<<12) /* 192000Hz */ +#define SNDRV_PCM_RATE_352800 (1<<13) /* 352800Hz */ +#define SNDRV_PCM_RATE_384000 (1<<14) /* 384000Hz */ #define SNDRV_PCM_RATE_CONTINUOUS (1<<30) /* continuous range */ #define SNDRV_PCM_RATE_KNOT (1<<31) /* supports more non-continuos rates */ @@ -129,6 +131,9 @@ struct snd_pcm_ops { SNDRV_PCM_RATE_88200|SNDRV_PCM_RATE_96000) #define SNDRV_PCM_RATE_8000_192000 (SNDRV_PCM_RATE_8000_96000|SNDRV_PCM_RATE_176400|\ SNDRV_PCM_RATE_192000) +#define SNDRV_PCM_RATE_8000_384000 (SNDRV_PCM_RATE_8000_192000|\ + SNDRV_PCM_RATE_352800|\ + SNDRV_PCM_RATE_384000) #define _SNDRV_PCM_FMTBIT(fmt) (1ULL << (__force int)SNDRV_PCM_FORMAT_##fmt) #define SNDRV_PCM_FMTBIT_S8 _SNDRV_PCM_FMTBIT(S8) #define SNDRV_PCM_FMTBIT_U8 _SNDRV_PCM_FMTBIT(U8) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 703857aab00f..11e653c8aa0e 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -2170,7 +2170,7 @@ static int snd_pcm_hw_rule_sample_bits(struct snd_pcm_hw_params *params, static const unsigned int rates[] = { 5512, 8000, 11025, 16000, 22050, 32000, 44100, - 48000, 64000, 88200, 96000, 176400, 192000 + 48000, 64000, 88200, 96000, 176400, 192000, 352800, 384000 }; const struct snd_pcm_hw_constraint_list snd_pcm_known_rates = { -- 2.21.0

2 2

[alsa-devel] [PATCH 0/4] ALSA: usb-audio: More strict validation and cleanups
by Takashi Iwai 22 Aug '19

22 Aug '19

Hi, this patchset introduces a more comprehensive validation code for USB-audio descriptor units, mainly for mixer stuff, for yet more hardening against the malformed descriptors. The rest are rather cleanups. Takashi === Takashi Iwai (4): ALSA: usb-audio: More validations of descriptor units ALSA: usb-audio: Simplify parse_audio_unit() ALSA: usb-audio: Unify the release of usb_mixer_elem_info objects ALSA: usb-audio: Remove superfluous bLength checks sound/usb/Makefile | 3 +- sound/usb/clock.c | 14 +-- sound/usb/helper.h | 4 + sound/usb/mixer.c | 237 ++++++++++++------------------------ sound/usb/power.c | 2 + sound/usb/quirks.c | 3 + sound/usb/stream.c | 25 ++-- sound/usb/validate.c | 332 +++++++++++++++++++++++++++++++++++++++++++++++++++ 8 files changed, 439 insertions(+), 181 deletions(-) create mode 100644 sound/usb/validate.c -- 2.16.4

2 10

[alsa-devel] [PATCH 1/2] ALSA: hda/realtek - Check beep whitelist before assigning in all codecs
by Takashi Iwai 22 Aug '19

22 Aug '19

Some Realtek codec parsers didn't check the availability of PC beep. Add has_cdefine_beep() check appropriately. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 9 ++++++--- 1 file changed, 6 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index b6ae452a1d8f..19831521bd95 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2848,7 +2848,8 @@ static int patch_alc268(struct hda_codec *codec) return err; spec = codec->spec; - spec->gen.beep_nid = 0x01; + if (has_cdefine_beep(codec)) + spec->gen.beep_nid = 0x01; spec->shutup = alc_eapd_shutup; @@ -8102,7 +8103,8 @@ static int patch_alc861(struct hda_codec *codec) return err; spec = codec->spec; - spec->gen.beep_nid = 0x23; + if (has_cdefine_beep(codec)) + spec->gen.beep_nid = 0x23; #ifdef CONFIG_PM spec->power_hook = alc_power_eapd; @@ -8203,7 +8205,8 @@ static int patch_alc861vd(struct hda_codec *codec) return err; spec = codec->spec; - spec->gen.beep_nid = 0x23; + if (has_cdefine_beep(codec)) + spec->gen.beep_nid = 0x23; spec->shutup = alc_eapd_shutup; -- 2.16.4

1 1

[alsa-devel] [PATCH] ALSA: usb-audio: Check mixer unit bitmap yet more strictly
by Takashi Iwai 22 Aug '19

22 Aug '19

The bmControls (for UAC1) or bmMixerControls (for UAC2/3) bitmap has a variable size depending on both input and output pins. Its size is to fit with input * output bits. The problem is that the input size can't be determined simply from the unit descriptor itself but it needs to parse the whole connected sources. Although the uac_mixer_unit_get_channels() tries to check some possible overflow of this bitmap, it's incomplete due to the lack of the evaluation of input pins. For covering possible overflows, this patch adds the bitmap overflow check in the loop of input pins in parse_audio_mixer_unit(). Fixes: 0bfe5e434e66 ("ALSA: usb-audio: Check mixer unit descriptors more strictly") Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/mixer.c | 34 ++++++++++++++++++++++++++-------- 1 file changed, 26 insertions(+), 8 deletions(-) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index b5927c3d5bc0..27ee68a507a2 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -739,7 +739,6 @@ static int uac_mixer_unit_get_channels(struct mixer_build *state, struct uac_mixer_unit_descriptor *desc) { int mu_channels; - void *c; if (desc->bLength < sizeof(*desc)) return -EINVAL; @@ -762,13 +761,6 @@ static int uac_mixer_unit_get_channels(struct mixer_build *state, break; } - if (!mu_channels) - return 0; - - c = uac_mixer_unit_bmControls(desc, state->mixer->protocol); - if (c - (void *)desc + (mu_channels - 1) / 8 >= desc->bLength) - return 0; /* no bmControls -> skip */ - return mu_channels; } @@ -2009,6 +2001,29 @@ static int parse_audio_feature_unit(struct mixer_build *state, int unitid, * Mixer Unit */ +/* check whether the given in/out overflows bmMixerControls matrix */ +static bool mixer_bitmap_overflow(struct uac_mixer_unit_descriptor *desc, + int protocol, int num_ins, int num_outs) +{ + u8 *hdr = (u8 *)desc; + u8 *c = uac_mixer_unit_bmControls(desc, protocol); + size_t rest; /* remaining bytes after bmMixerControls */ + + switch (protocol) { + case UAC_VERSION_1: + default: + rest = 1; /* iMixer */ + case UAC_VERSION_2: + rest = 2; /* bmControls + iMixer */ + case UAC_VERSION_3: + rest = 6; /* bmControls + wMixerDescrStr */ + break; + } + + /* overflow? */ + return c + (num_ins * num_outs + 7) / 8 + rest > hdr + hdr[0]; +} + /* * build a mixer unit control * @@ -2137,6 +2152,9 @@ static int parse_audio_mixer_unit(struct mixer_build *state, int unitid, if (err < 0) return err; num_ins += iterm.channels; + if (mixer_bitmap_overflow(desc, state->mixer->protocol, + num_ins, num_outs)) + break; for (; ich < num_ins; ich++) { int och, ich_has_controls = 0; -- 2.16.4

1 0

[alsa-devel] [RFC PATCH 00/11] soundwire: intel: simplify DAI/PDI handling
by Pierre-Louis Bossart 21 Aug '19

21 Aug '19

In the initial SoundWire code released by Intel, the PDIs and ports on the Master interface were dynamically allocated. This wasn't a bad idea at the time and would have allowed for interesting routing. Fast-forward to 2019, with the hardware available on CometLake/IceLake, that dynamic allocation makes it complicated to deal with statically-allocated ASoC dailinks and topology-defined DAIs. In this series, we suggest a drastic simplification where the SoundWire code reuses information provided by DAIs and dailinks. We also suggest removing the dynamic allocation of ports on the master since in practice there is a 1:1 mapping between ports and PDIs. In the second part of the series, we suggest adding new callbacks to SoundWire DAIs, so that all the SoundWire stream operations are contained at the DAI level. This solution results in a very simple integration with the SOF code (which will be shared in a separate series since SOF will not apply directly on top of soundwire/next). The SOF parts only call a SoundWire init/release API, and provides 2 callbacks for hw_params and free, with all the details of the SoundWire DAIs and IP handled in drivers/soundwire. This solution has been tested on CometLake/IceLake with simple capture/playback. When ASoC supports the multi-cpu capability needed for synchronized playback/capture across multiple links, we will have to modify slightly this solution so that the stream alloc, release and trigger operations are done once. This is future work that will take place later, likely after 5.4, and which should not impact the SOF integration. The code in this patchset is the result of collaboration between Bard Liao, Rander Wang and Pierre Bossart, with ideas coming from all 3 sides. It's likely that there are still some parts in the code that can be improved, hence the RFC state. Bard Liao (3): soundwire: intel: fix intel_register_dai PDI offsets and numbers soundwire: intel: remove playback/capture stream_name soundwire: cadence_master: improve PDI allocation Pierre-Louis Bossart (3): soundwire: remove DAI_ID_RANGE definitions soundwire: cadence/intel: simplify PDI/port mapping soundwire: intel: don't filter out PDI0/1 Rander Wang (5): soundwire: intel: improve .config_stream callback, add .free_stream soundwire: intel: add prepare support in sdw dai driver soundwire: intel: add trigger support in sdw dai driver soundwire: intel: do sdw stream setup in setup function soundwire: intel: free all resources on hw_free() drivers/soundwire/cadence_master.c | 158 ++++------------ drivers/soundwire/cadence_master.h | 34 +--- drivers/soundwire/intel.c | 278 ++++++++++++++++------------ include/linux/soundwire/sdw.h | 3 - include/linux/soundwire/sdw_intel.h | 4 +- 5 files changed, 209 insertions(+), 268 deletions(-) -- 2.20.1

1 11

[alsa-devel] [PATCH] ALSA: line6: Fix memory leak at line6_init_pcm() error path
by Takashi Iwai 21 Aug '19

21 Aug '19

I forgot to release the allocated object at the early error path in line6_init_pcm(). For addressing it, slightly shuffle the code so that the PCM destructor (pcm->private_free) is assigned properly before all error paths. Fixes: 3450121997ce ("ALSA: line6: Fix write on zero-sized buffer") Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/line6/pcm.c | 18 +++++++++--------- 1 file changed, 9 insertions(+), 9 deletions(-) diff --git a/sound/usb/line6/pcm.c b/sound/usb/line6/pcm.c index 2c03e0f6bf72..f70211e6b174 100644 --- a/sound/usb/line6/pcm.c +++ b/sound/usb/line6/pcm.c @@ -550,6 +550,15 @@ int line6_init_pcm(struct usb_line6 *line6, line6pcm->volume_monitor = 255; line6pcm->line6 = line6; + spin_lock_init(&line6pcm->out.lock); + spin_lock_init(&line6pcm->in.lock); + line6pcm->impulse_period = LINE6_IMPULSE_DEFAULT_PERIOD; + + line6->line6pcm = line6pcm; + + pcm->private_data = line6pcm; + pcm->private_free = line6_cleanup_pcm; + line6pcm->max_packet_size_in = usb_maxpacket(line6->usbdev, usb_rcvisocpipe(line6->usbdev, ep_read), 0); @@ -562,15 +571,6 @@ int line6_init_pcm(struct usb_line6 *line6, return -EINVAL; } - spin_lock_init(&line6pcm->out.lock); - spin_lock_init(&line6pcm->in.lock); - line6pcm->impulse_period = LINE6_IMPULSE_DEFAULT_PERIOD; - - line6->line6pcm = line6pcm; - - pcm->private_data = line6pcm; - pcm->private_free = line6_cleanup_pcm; - err = line6_create_audio_out_urbs(line6pcm); if (err < 0) return err; -- 2.16.4

1 0

[alsa-devel] [PATCH 00/17] soundwire: fixes for 5.4
by Pierre-Louis Bossart 21 Aug '19

21 Aug '19

This series provides an update on the initial RFC. Debugfs and Intel updates will be provided in follow-up patches. The order of patches was changed since the RFC so detailed change logs are provided below. Changes since RFC: patch1: feedback from Vinod and Cezary use local variable to reduce number of de-references document that first argument is mandatory patch2: feedback from Vinod document that the work-around is required for all existing controllers patch 3: feedback from Vinod clarify comment that MCP_INT_IRQ ungates all other settings. use MCP_INT_SLAVE_MASK instead of all individual settings ORed. Patch 4: feedback from Vinod: demote dynamic debug log to dev_dbg (was dev_err) patch5: clarify commit message that the helpers will be used in the Cadence parts as well. patch 6: feedback from Vinod s/BIOS/firmware remove magic numbers, introduce macros patch 7: add kbuild warning message in commit message patch 8: no change patch 9: feedback from Guennadi remove unnecessary initializations patch 10/11/12: feedback from Vinod split initial patch in 3 (prototype, Intel, Cadence) add explanations on what mclk_freq is in commit messages remove pr_err logs missed in RFC patch 13: feedback from Bard remove unecessary reads before update patch 14: no change patch 15: update commit message patch 16: update commit message remove unnecessary dynamic debug log patch 17: no change Bard liao (1): soundwire: include mod_devicetable.h to avoid compiling warnings Pierre-Louis Bossart (15): soundwire: intel: prevent possible dereference in hw_params soundwire: intel: fix channel number reported by hardware soundwire: cadence_master: revisit interrupt settings soundwire: bus: improve dynamic debug comments for enumeration soundwire: export helpers to find row and column values soundwire: cadence_master: use firmware defaults for frame shape soundwire: stream: fix disable sequence soundwire: stream: remove unnecessary variable initializations soundwire: add new mclk_freq field for properties soundwire: intel: read mclk_freq property from firmware soundwire: cadence_master: make use of mclk_freq property soundwire: intel: handle disabled links soundwire: intel_init: add kernel module parameter to filter out links soundwire: cadence_master: add kernel parameter to override interrupt mask soundwire: intel: move shutdown() callback and don't export symbol Rander Wang (1): soundwire: cadence_master: fix divider setting in clock register drivers/soundwire/bus.c | 5 +- drivers/soundwire/bus.h | 7 +- drivers/soundwire/cadence_master.c | 102 +++++++++++++++++++--------- drivers/soundwire/cadence_master.h | 2 - drivers/soundwire/intel.c | 83 ++++++++++++++++++++-- drivers/soundwire/intel_init.c | 11 +++ drivers/soundwire/stream.c | 93 ++++++++++++++++--------- include/linux/soundwire/sdw.h | 6 ++ include/linux/soundwire/sdw_intel.h | 1 + 9 files changed, 236 insertions(+), 74 deletions(-) -- 2.20.1

3 26

[alsa-devel] [PATCH] ASoC: SOF: topology: fix get control data return type and arguments
by Jaska Uimonen 21 Aug '19

21 Aug '19

From: Jaska Uimonen <jaska.uimonen(a)intel.com> sof_get_control_data returns negative values even though the return value is defined unsigned (size_t). So change the return value type to int. Add the data size as pointer argument to sof_get_control_data to avoid ambiquity in the meaning of the return type. Also make the sof_get_control_data to return -EINVAL if data pointer is valid but the size is for some reason zero. Fixes: cac974a51ebb ("ASoC: SOF: topology: use set_get_data in process load") Reported by: Dan Carpenter <dan.carpenter(a)oracle.com> Signed-off-by: Jaska Uimonen <jaska.uimonen(a)intel.com> --- sound/soc/sof/topology.c | 23 +++++++++++++---------- 1 file changed, 13 insertions(+), 10 deletions(-) diff --git a/sound/soc/sof/topology.c b/sound/soc/sof/topology.c index 28a7a6e06a53..8288b1758542 100644 --- a/sound/soc/sof/topology.c +++ b/sound/soc/sof/topology.c @@ -1752,15 +1752,15 @@ static int sof_widget_load_siggen(struct snd_soc_component *scomp, int index, return ret; } -static size_t sof_get_control_data(struct snd_sof_dev *sdev, - struct snd_soc_dapm_widget *widget, - struct sof_widget_data *wdata) +static int sof_get_control_data(struct snd_sof_dev *sdev, + struct snd_soc_dapm_widget *widget, + struct sof_widget_data *wdata, + size_t *size) { const struct snd_kcontrol_new *kc; struct soc_mixer_control *sm; struct soc_bytes_ext *sbe; struct soc_enum *se; - size_t size = 0; int i; for (i = 0; i < widget->num_kcontrols; i++) { @@ -1800,7 +1800,11 @@ static size_t sof_get_control_data(struct snd_sof_dev *sdev, if (wdata[i].pdata->magic != SOF_ABI_MAGIC) return -EINVAL; - size += wdata[i].pdata->size; + /* don't accept 0 size for data */ + if (!wdata[i].pdata->size) + return -EINVAL; + + *size += wdata[i].pdata->size; /* get data type */ switch (wdata[i].control->cmd) { @@ -1819,7 +1823,7 @@ static size_t sof_get_control_data(struct snd_sof_dev *sdev, } } - return size; + return 0; } static int sof_process_load(struct snd_soc_component *scomp, int index, @@ -1855,12 +1859,11 @@ static int sof_process_load(struct snd_soc_component *scomp, int index, return -ENOMEM; /* get possible component controls and get size of all pdata */ - ipc_data_size = sof_get_control_data(sdev, widget, wdata); + ret = sof_get_control_data(sdev, widget, wdata, + &ipc_data_size); - if (ipc_data_size <= 0) { - ret = ipc_data_size; + if (ret < 0) goto out; - } } ipc_size = sizeof(struct sof_ipc_comp_process) + -- 2.13.6

3 2

[alsa-devel] [PATCH v2] ASoC: SOF: topology: fix get control data return type and arguments
by Jaska Uimonen 21 Aug '19

21 Aug '19

From: Jaska Uimonen <jaska.uimonen(a)intel.com> sof_get_control_data returns negative values even though the return value is defined unsigned (size_t). So change the return value type to int and add the data size as pointer argument to sof_get_control_data to avoid ambiquity in the meaning of the return type. Fixes: cac974a51ebb ("ASoC: SOF: topology: use set_get_data in process load") Reported by: Dan Carpenter <dan.carpenter(a)oracle.com> Signed-off-by: Jaska Uimonen <jaska.uimonen(a)intel.com> --- sound/soc/sof/topology.c | 21 +++++++++++---------- 1 file changed, 11 insertions(+), 10 deletions(-) diff --git a/sound/soc/sof/topology.c b/sound/soc/sof/topology.c index 28a7a6e06a53..fc85efbad378 100644 --- a/sound/soc/sof/topology.c +++ b/sound/soc/sof/topology.c @@ -1752,17 +1752,19 @@ static int sof_widget_load_siggen(struct snd_soc_component *scomp, int index, return ret; } -static size_t sof_get_control_data(struct snd_sof_dev *sdev, - struct snd_soc_dapm_widget *widget, - struct sof_widget_data *wdata) +static int sof_get_control_data(struct snd_sof_dev *sdev, + struct snd_soc_dapm_widget *widget, + struct sof_widget_data *wdata, + size_t *size) { const struct snd_kcontrol_new *kc; struct soc_mixer_control *sm; struct soc_bytes_ext *sbe; struct soc_enum *se; - size_t size = 0; int i; + *size = 0; + for (i = 0; i < widget->num_kcontrols; i++) { kc = &widget->kcontrol_news[i]; @@ -1800,7 +1802,7 @@ static size_t sof_get_control_data(struct snd_sof_dev *sdev, if (wdata[i].pdata->magic != SOF_ABI_MAGIC) return -EINVAL; - size += wdata[i].pdata->size; + *size += wdata[i].pdata->size; /* get data type */ switch (wdata[i].control->cmd) { @@ -1819,7 +1821,7 @@ static size_t sof_get_control_data(struct snd_sof_dev *sdev, } } - return size; + return 0; } static int sof_process_load(struct snd_soc_component *scomp, int index, @@ -1855,12 +1857,11 @@ static int sof_process_load(struct snd_soc_component *scomp, int index, return -ENOMEM; /* get possible component controls and get size of all pdata */ - ipc_data_size = sof_get_control_data(sdev, widget, wdata); + ret = sof_get_control_data(sdev, widget, wdata, + &ipc_data_size); - if (ipc_data_size <= 0) { - ret = ipc_data_size; + if (ret < 0) goto out; - } } ipc_size = sizeof(struct sof_ipc_comp_process) + -- 2.13.6

1 0

[alsa-devel] [PATCH v2 0/4] ASoC: sun4i-i2s: Number of fixes and TDM Support
by Maxime Ripard 21 Aug '19

21 Aug '19

From: Maxime Ripard <maxime.ripard(a)bootlin.com> Hi, This series aims at fixing a number of issues in the current i2s driver, mostly related to the i2s master support and the A83t support. It also uses that occasion to cleanup a few things and simplify the driver. Finally, it builds on those fixes and cleanups to introduce TDM and DSP formats support. Let me know what you think, Maxime Changes from v1: - Removed patches applied - Refactor a bit the call to sun4i_i2s_set_clk_rate - Fix build issue - Add an extra patch to cleanup sun4i_i2s_hw_params Maxime Ripard (4): ASoC: sun4i-i2s: Use the physical / slot width for the clocks ASoC: sun4i-i2s: Use the actual format width instead of an hardcoded one ASoC: sun4i-i2s: Replace call to params_width by local variable ASoC: sun4i-i2s: Add support for DSP formats sound/soc/sunxi/sun4i-i2s.c | 58 ++++++++++++++++++++++++++++---------- 1 file changed, 43 insertions(+), 15 deletions(-) base-commit: 137befe19f310400a8b20fd8a4ce8c4141aafde0 -- git-series 0.9.1

1 4

[alsa-devel] [PATCH 0/2] Add support for i.MX8QM
by Daniel Baluta 21 Aug '19

21 Aug '19

i.MX8QM SOC integrates 6 SAI instances. Register map is similar with i.MX6 series. Daniel Baluta (2): ASoC: fsl_sai: Add support for imx8qm ASoC: dt-bindings: Introduce compatible string for imx8qm Documentation/devicetree/bindings/sound/fsl-sai.txt | 3 ++- sound/soc/fsl/fsl_sai.c | 7 +++++++ 2 files changed, 9 insertions(+), 1 deletion(-) -- 2.17.1

2 4

[alsa-devel] [PATCH] ASoC: SOF: topology: fix get control data return type and arguments
by Jaska Uimonen 21 Aug '19

21 Aug '19

sof_get_control_data returns negative values even though the return value is defined unsigned (size_t). So change the return value type to int. Add the data size as pointer argument to sof_get_control_data to avoid ambiquity in the meaning of the return type. Also make the sof_get_control_data to return -EINVAL if data pointer is valid but the size is for some reason zero. Fixes: cac974a51ebb ("ASoC: SOF: topology: use set_get_data in process load") Reported by: Dan Carpenter <dan.carpenter(a)oracle.com> Signed-off-by: Jaska Uimonen <jaska.uimonen(a)intel.com> --- sound/soc/sof/topology.c | 23 +++++++++++++---------- 1 file changed, 13 insertions(+), 10 deletions(-) diff --git a/sound/soc/sof/topology.c b/sound/soc/sof/topology.c index 28a7a6e06a53..8288b1758542 100644 --- a/sound/soc/sof/topology.c +++ b/sound/soc/sof/topology.c @@ -1752,15 +1752,15 @@ static int sof_widget_load_siggen(struct snd_soc_component *scomp, int index, return ret; } -static size_t sof_get_control_data(struct snd_sof_dev *sdev, - struct snd_soc_dapm_widget *widget, - struct sof_widget_data *wdata) +static int sof_get_control_data(struct snd_sof_dev *sdev, + struct snd_soc_dapm_widget *widget, + struct sof_widget_data *wdata, + size_t *size) { const struct snd_kcontrol_new *kc; struct soc_mixer_control *sm; struct soc_bytes_ext *sbe; struct soc_enum *se; - size_t size = 0; int i; for (i = 0; i < widget->num_kcontrols; i++) { @@ -1800,7 +1800,11 @@ static size_t sof_get_control_data(struct snd_sof_dev *sdev, if (wdata[i].pdata->magic != SOF_ABI_MAGIC) return -EINVAL; - size += wdata[i].pdata->size; + /* don't accept 0 size for data */ + if (!wdata[i].pdata->size) + return -EINVAL; + + *size += wdata[i].pdata->size; /* get data type */ switch (wdata[i].control->cmd) { @@ -1819,7 +1823,7 @@ static size_t sof_get_control_data(struct snd_sof_dev *sdev, } } - return size; + return 0; } static int sof_process_load(struct snd_soc_component *scomp, int index, @@ -1855,12 +1859,11 @@ static int sof_process_load(struct snd_soc_component *scomp, int index, return -ENOMEM; /* get possible component controls and get size of all pdata */ - ipc_data_size = sof_get_control_data(sdev, widget, wdata); + ret = sof_get_control_data(sdev, widget, wdata, + &ipc_data_size); - if (ipc_data_size <= 0) { - ret = ipc_data_size; + if (ret < 0) goto out; - } } ipc_size = sizeof(struct sof_ipc_comp_process) + -- 2.13.6 --------------------------------------------------------------------- Intel Finland Oy Registered Address: PL 281, 00181 Helsinki Business Identity Code: 0357606 - 4 Domiciled in Helsinki This e-mail and any attachments may contain confidential material for the sole use of the intended recipient(s). Any review or distribution by others is strictly prohibited. If you are not the intended recipient, please contact the sender and delete all copies.

3 2

[alsa-devel] [PATCH v5 00/15] ASoC: sun4i-i2s: Updates to the driver
by codekipper＠gmail.com 21 Aug '19

21 Aug '19

From: Marcus Cooper <codekipper(a)gmail.com> Hi All, here is a patch series to add some improvements to the sun4i-i2s driver found whilst getting slave clocking and hdmi audio working on the newer SoCs. As the LibreELEC project is progressing extremely well then there has been some activity getting H6 SoC support with surround sound working and these changes are also included. The functionality included with the new patch set has been extended to cover more sample resolutions, multi-lane data output for HDMI audio and some bug fixes that have been discovered along the way. I have changed some of the original reg fields into function calls as this made it easier to setup for multi-channel audio especially across different SoCs. I can see more usage of the tdm property since I last attempted to push these patches and the examples currently in mainline sort of the opposite to what I'm trying to achieve. When we first started looking at the i2s driver, the codecs that we were using allowed for the frame width to be determined based on the sampling resolution but in most use cases it seems that a fixed width is required(my highest priority should be to get HDMI audio support in). We're using the tdm property to override the old way to calculate the frame width. What I've seen in what has already been mainlined is that the i2s driver has a frame width that is fixed to 32 bits and this can be overridden using the tdm property. My test branch for this can be found at https://github.com/codekipper/linux-sunxi/commits/upstream-i2s , I've been using a Pine64 to test with; validating the new SoC block with HDMI audio and ensuring that I've not broken the old block by making sure that the audio codec still works. If we able to get the first three patches delivered then that is enough for HDMI audio support on the newer SoCs(H3, A64 etc). I still need to investigate the FIFO syncing issues which i've not had a chance to change or address the concerns that broonie and wens brought up. This change has been moved to the top of the patch stack. I would also like to make the multi-channel audio and audio mapping more configurable via the device tree. Currently what is implemented suites our current needs. BR, CK --- v5 changes compared to v4 are: - removed delivered patches. - Added more details to commit messages. - replaced some reg fields with function calls. - Added DSP_A and DSP_B support for H3 and later SoCs. - Added support for the Allwinner H6. v4 changes compared to v3 are: - Moved patches around so that the more controversial of patches are at the top of the stack. - Added more details to commit messages. - Fixed 20bit audio PCM format to use 4 bytes. - Reduced number of flags used to indicate a new SoC. v3 changes compared to v2 are: - added back slave mode changes - added back the use of tdm properties - changes to regmap and caching - removed loopback functionality - fixes to the channel offset mask v2 changes compared to v1 are: - removed slave mode changes which didn't set mclk and bclk div. - removed use of tdm and now use a dedicated property. - fix commit message to better explain reason for sign extending - add divider calculations for newer SoCs. - add support for multi-lane i2s data output. - add support for 20, 24 and 32 bit samples. - add loopback property so blocks can be tested without a codec. --- Jernej Skrabec (3): clk: sunxi-ng: h6: Allow I2S to change parent rate dt-bindings: ASoC: sun4i-i2s: Add H6 compatible ASoC: sun4i-i2s: Add support for H6 I2S Marcus Cooper (12): ASoC: sun4i-i2s: Add regmap field to sign extend sample ASoC: sun4i-i2s: Add set_tdm_slot functionality ASoC: sun4i-i2s: Correct divider calculations ASoC: sun4i-i2s: Support more formats on newer SoCs ASoC: sun4i-i2s: Add functions for RX and TX channel offsets ASoC: sun4i-i2s: Add functions for RX and TX channel enables ASoC: sun4i-i2s: Add functions for RX and TX channel selects ASoC: sun4i-i2s: Add functions for channel mapping ASoC: sun4i-i2s: Add multi-lane functionality ASoC: sun4i-i2s: Add multichannel functionality ASoc: sun4i-i2s: Add 20, 24 and 32 bit support ASoC: sun4i-i2s: Adjust regmap settings .../sound/allwinner,sun4i-a10-i2s.yaml | 2 + drivers/clk/sunxi-ng/ccu-sun50i-h6.c | 8 +- sound/soc/sunxi/sun4i-i2s.c | 690 ++++++++++++++---- 3 files changed, 542 insertions(+), 158 deletions(-) -- 2.22.0

7 42

[alsa-devel] [PATCH] ALSA: usb-audio: Fix invalid NULL check in snd_emuusb_set_samplerate()
by Takashi Iwai 21 Aug '19

21 Aug '19

The quirk function snd_emuusb_set_samplerate() has a NULL check for the mixer element, but this is useless in the current code. It used to be a check against mixer->id_elems[unitid] but it was changed later to the value after mixer_eleme_list_to_info() which is always non-NULL due to the container_of() usage. This patch fixes the check before the conversion. While we're at it, correct a typo in the comment in the function, too. Fixes: 8c558076c740 ("ALSA: usb-audio: Clean up mixer element list traverse") Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/mixer_quirks.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index 199fa157a411..27dcb3743690 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -1155,17 +1155,17 @@ void snd_emuusb_set_samplerate(struct snd_usb_audio *chip, { struct usb_mixer_interface *mixer; struct usb_mixer_elem_info *cval; - int unitid = 12; /* SamleRate ExtensionUnit ID */ + int unitid = 12; /* SampleRate ExtensionUnit ID */ list_for_each_entry(mixer, &chip->mixer_list, list) { - cval = mixer_elem_list_to_info(mixer->id_elems[unitid]); - if (cval) { + if (mixer->id_elems[unitid]) { + cval = mixer_elem_list_to_info(mixer->id_elems[unitid]); snd_usb_mixer_set_ctl_value(cval, UAC_SET_CUR, cval->control << 8, samplerate_id); snd_usb_mixer_notify_id(mixer, unitid); + break; } - break; } } -- 2.16.4

1 0

[alsa-devel] Master volume not controlling applications' volume.
by Martín Martínez Rivera 21 Aug '19

21 Aug '19

Changing the master volume in pavucontrol (or alsamixer) does not have any effect. The only way to change the volume is to go into the playback tab and change it for each application. The volume only changes when I set the master volume to 0 or when I increase it over 100%. In between there aren't any changes. Running arch on a Lenovo X1 Carbon if that's of any use. I am using the arch default configuration files. I posted somewhere else and was told to email this list as it could be a kernel bug. kernel: 5.2.9-arch1-1-ARCH pactl info output: Server String: /run/user/1000/pulse/native Library Protocol Version: 32 Server Protocol Version: 32 Is Local: yes Client Index: 13 Tile Size: 65472 User Name: martinmr Host Name: martinmr-laptop Server Name: pulseaudio Server Version: 12.2 Default Sample Specification: s16le 2ch 44100Hz Default Channel Map: front-left,front-right Default Sink: alsa_output.pci-0000_00_1f.3.analog-stereo Default Source: alsa_output.pci-0000_00_1f.3.analog-stereo.monitor Cookie: 1afa:f937 Let me know if the output from some other command is needed. Best, Martin

1 0

[alsa-devel] Applied "ASoC: intel: cht_bsw_max98090_ti: Add all Chromebooks that need pmc_plt_clk_0 quirk" to the asoc tree
by Mark Brown 20 Aug '19

20 Aug '19

The patch ASoC: intel: cht_bsw_max98090_ti: Add all Chromebooks that need pmc_plt_clk_0 quirk has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-5.4 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 d5e120422db8808e1c8b1507900ca393a877c58f Mon Sep 17 00:00:00 2001 From: Daniel Stuart <daniel.stuart14(a)gmail.com> Date: Thu, 15 Aug 2019 14:12:55 -0300 Subject: [PATCH] ASoC: intel: cht_bsw_max98090_ti: Add all Chromebooks that need pmc_plt_clk_0 quirk Every single baytrail chromebook sets PMC to 0, as can be seeing below by searching through coreboot source code: $ grep -rl "PMC_PLT_CLK\[0\]" . ./rambi/variants/glimmer/devicetree.cb ./rambi/variants/clapper/devicetree.cb ./rambi/variants/swanky/devicetree.cb ./rambi/variants/enguarde/devicetree.cb ./rambi/variants/winky/devicetree.cb ./rambi/variants/kip/devicetree.cb ./rambi/variants/squawks/devicetree.cb ./rambi/variants/orco/devicetree.cb ./rambi/variants/ninja/devicetree.cb ./rambi/variants/heli/devicetree.cb ./rambi/variants/sumo/devicetree.cb ./rambi/variants/banjo/devicetree.cb ./rambi/variants/candy/devicetree.cb ./rambi/variants/gnawty/devicetree.cb ./rambi/variants/rambi/devicetree.cb ./rambi/variants/quawks/devicetree.cb Plus, Cyan (only non-baytrail chromebook with max98090) also needs this patch for audio to work. Thus, this commit adds all the missing devices to bsw_max98090 quirk table, implemented by commit a182ecd3809c ("ASoC: intel: cht_bsw_max98090_ti: Add quirk for boards using pmc_plt_clk_0"). Signed-off-by: Daniel Stuart <daniel.stuart14(a)gmail.com> Link: https://lore.kernel.org/r/20190815171300.30126-1-daniel.stuart14@gmail.com Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 98 ++++++++++++++++++++ 1 file changed, 98 insertions(+) diff --git a/sound/soc/intel/boards/cht_bsw_max98090_ti.c b/sound/soc/intel/boards/cht_bsw_max98090_ti.c index 1db9a95e6a79..eaf3e2208a06 100644 --- a/sound/soc/intel/boards/cht_bsw_max98090_ti.c +++ b/sound/soc/intel/boards/cht_bsw_max98090_ti.c @@ -398,6 +398,20 @@ static struct snd_soc_card snd_soc_card_cht = { }; static const struct dmi_system_id cht_max98090_quirk_table[] = { + { + /* Banjo model Chromebook */ + .matches = { + DMI_MATCH(DMI_PRODUCT_NAME, "Banjo"), + }, + .driver_data = (void *)QUIRK_PMC_PLT_CLK_0, + }, + { + /* Candy model Chromebook */ + .matches = { + DMI_MATCH(DMI_PRODUCT_NAME, "Candy"), + }, + .driver_data = (void *)QUIRK_PMC_PLT_CLK_0, + }, { /* Clapper model Chromebook */ .matches = { @@ -405,6 +419,27 @@ static const struct dmi_system_id cht_max98090_quirk_table[] = { }, .driver_data = (void *)QUIRK_PMC_PLT_CLK_0, }, + { + /* Cyan model Chromebook */ + .matches = { + DMI_MATCH(DMI_PRODUCT_NAME, "Cyan"), + }, + .driver_data = (void *)QUIRK_PMC_PLT_CLK_0, + }, + { + /* Enguarde model Chromebook */ + .matches = { + DMI_MATCH(DMI_PRODUCT_NAME, "Enguarde"), + }, + .driver_data = (void *)QUIRK_PMC_PLT_CLK_0, + }, + { + /* Glimmer model Chromebook */ + .matches = { + DMI_MATCH(DMI_PRODUCT_NAME, "Glimmer"), + }, + .driver_data = (void *)QUIRK_PMC_PLT_CLK_0, + }, { /* Gnawty model Chromebook (Acer Chromebook CB3-111) */ .matches = { @@ -412,6 +447,62 @@ static const struct dmi_system_id cht_max98090_quirk_table[] = { }, .driver_data = (void *)QUIRK_PMC_PLT_CLK_0, }, + { + /* Heli model Chromebook */ + .matches = { + DMI_MATCH(DMI_PRODUCT_NAME, "Heli"), + }, + .driver_data = (void *)QUIRK_PMC_PLT_CLK_0, + }, + { + /* Kip model Chromebook */ + .matches = { + DMI_MATCH(DMI_PRODUCT_NAME, "Kip"), + }, + .driver_data = (void *)QUIRK_PMC_PLT_CLK_0, + }, + { + /* Ninja model Chromebook */ + .matches = { + DMI_MATCH(DMI_PRODUCT_NAME, "Ninja"), + }, + .driver_data = (void *)QUIRK_PMC_PLT_CLK_0, + }, + { + /* Orco model Chromebook */ + .matches = { + DMI_MATCH(DMI_PRODUCT_NAME, "Orco"), + }, + .driver_data = (void *)QUIRK_PMC_PLT_CLK_0, + }, + { + /* Quawks model Chromebook */ + .matches = { + DMI_MATCH(DMI_PRODUCT_NAME, "Quawks"), + }, + .driver_data = (void *)QUIRK_PMC_PLT_CLK_0, + }, + { + /* Rambi model Chromebook */ + .matches = { + DMI_MATCH(DMI_PRODUCT_NAME, "Rambi"), + }, + .driver_data = (void *)QUIRK_PMC_PLT_CLK_0, + }, + { + /* Squawks model Chromebook */ + .matches = { + DMI_MATCH(DMI_PRODUCT_NAME, "Squawks"), + }, + .driver_data = (void *)QUIRK_PMC_PLT_CLK_0, + }, + { + /* Sumo model Chromebook */ + .matches = { + DMI_MATCH(DMI_PRODUCT_NAME, "Sumo"), + }, + .driver_data = (void *)QUIRK_PMC_PLT_CLK_0, + }, { /* Swanky model Chromebook (Toshiba Chromebook 2) */ .matches = { @@ -419,6 +510,13 @@ static const struct dmi_system_id cht_max98090_quirk_table[] = { }, .driver_data = (void *)QUIRK_PMC_PLT_CLK_0, }, + { + /* Winky model Chromebook */ + .matches = { + DMI_MATCH(DMI_PRODUCT_NAME, "Winky"), + }, + .driver_data = (void *)QUIRK_PMC_PLT_CLK_0, + }, {} }; -- 2.20.1

1 0

[alsa-devel] [PATCH 0/3] ASoC: mchp-i2s-mcc: Several fixes
by Codrin Ciubotariu 20 Aug '19

20 Aug '19

This pathset fixes some issues detected while testing some more the Microchip I2S multichannel controller. The first two patches fix some issues that appear mostly when hw_free() and hw_params() callbacks are called multiple times. The third patch fixes a problem caused when the controller is in master mode and both capture and playback are played at the same time. All three patches have a "Fixes" tag. Although they are independent, some conflicts might appear if they are not applied in the order presented in this patchset. If so, please let me know so I can rebase them. Codrin Ciubotariu (3): ASoC: mchp-i2s-mcc: Wait for RX/TX RDY only if controller is running ASoC: mchp-i2s-mcc: Fix unprepare of GCLK ASoC: mchp-i2s-mcc: Fix simultaneous capture and playback in master mode sound/soc/atmel/mchp-i2s-mcc.c | 111 +++++++++++++++++++-------------- 1 file changed, 64 insertions(+), 47 deletions(-) -- 2.20.1

2 6

[alsa-devel] [PATCH] ASoC: meson: axg-tdm-formatter: free reset on device removal
by Jerome Brunet 20 Aug '19

20 Aug '19

Use the devm variant to get the formatter reset so it is properly freed on device removal Fixes: 751bd5db5260 ("ASoC: meson: axg-tdm-formatter: add reset") Signed-off-by: Jerome Brunet <jbrunet(a)baylibre.com> --- sound/soc/meson/axg-tdm-formatter.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/meson/axg-tdm-formatter.c b/sound/soc/meson/axg-tdm-formatter.c index 21c735afab35..358c8c0d861c 100644 --- a/sound/soc/meson/axg-tdm-formatter.c +++ b/sound/soc/meson/axg-tdm-formatter.c @@ -325,7 +325,7 @@ int axg_tdm_formatter_probe(struct platform_device *pdev) } /* Formatter dedicated reset line */ - formatter->reset = reset_control_get_optional_exclusive(dev, NULL); + formatter->reset = devm_reset_control_get_optional_exclusive(dev, NULL); if (IS_ERR(formatter->reset)) { ret = PTR_ERR(formatter->reset); if (ret != -EPROBE_DEFER) -- 2.21.0

2 1

[alsa-devel] [PATCH] ASoC: meson: g12a-tohdmitx: require regmap mmio
by Jerome Brunet 20 Aug '19

20 Aug '19

The tohdmitx glue uses regmap MMIO so it should require it. Fixes: c8609f3870f7 ("ASoC: meson: add g12a tohdmitx control") Signed-off-by: Jerome Brunet <jbrunet(a)baylibre.com> --- sound/soc/meson/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/meson/Kconfig b/sound/soc/meson/Kconfig index 63b38c123103..2e3676147cea 100644 --- a/sound/soc/meson/Kconfig +++ b/sound/soc/meson/Kconfig @@ -87,6 +87,7 @@ config SND_MESON_AXG_PDM config SND_MESON_G12A_TOHDMITX tristate "Amlogic G12A To HDMI TX Control Support" + select REGMAP_MMIO imply SND_SOC_HDMI_CODEC help Select Y or M to add support for HDMI audio on the g12a SoC -- 2.21.0

2 1

[alsa-devel] Applied "ASoC: intel: cht_bsw_max98090_ti: Add all Chromebooks that need pmc_plt_clk_0 quirk" to the asoc tree
by Mark Brown 20 Aug '19

20 Aug '19

The patch ASoC: intel: cht_bsw_max98090_ti: Add all Chromebooks that need pmc_plt_clk_0 quirk 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 404be07f4ed27697f5fa69162f67a94555738595 Mon Sep 17 00:00:00 2001 From: Daniel Stuart <daniel.stuart14(a)gmail.com> Date: Thu, 15 Aug 2019 14:12:55 -0300 Subject: [PATCH] ASoC: intel: cht_bsw_max98090_ti: Add all Chromebooks that need pmc_plt_clk_0 quirk Every single baytrail chromebook sets PMC to 0, as can be seeing below by searching through coreboot source code: $ grep -rl "PMC_PLT_CLK\[0\]" . ./rambi/variants/glimmer/devicetree.cb ./rambi/variants/clapper/devicetree.cb ./rambi/variants/swanky/devicetree.cb ./rambi/variants/enguarde/devicetree.cb ./rambi/variants/winky/devicetree.cb ./rambi/variants/kip/devicetree.cb ./rambi/variants/squawks/devicetree.cb ./rambi/variants/orco/devicetree.cb ./rambi/variants/ninja/devicetree.cb ./rambi/variants/heli/devicetree.cb ./rambi/variants/sumo/devicetree.cb ./rambi/variants/banjo/devicetree.cb ./rambi/variants/candy/devicetree.cb ./rambi/variants/gnawty/devicetree.cb ./rambi/variants/rambi/devicetree.cb ./rambi/variants/quawks/devicetree.cb Plus, Cyan (only non-baytrail chromebook with max98090) also needs this patch for audio to work. Thus, this commit adds all the missing devices to bsw_max98090 quirk table, implemented by commit a182ecd3809c ("ASoC: intel: cht_bsw_max98090_ti: Add quirk for boards using pmc_plt_clk_0"). Signed-off-by: Daniel Stuart <daniel.stuart14(a)gmail.com> Link: https://lore.kernel.org/r/20190815171300.30126-1-daniel.stuart14@gmail.com Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 98 ++++++++++++++++++++ 1 file changed, 98 insertions(+) diff --git a/sound/soc/intel/boards/cht_bsw_max98090_ti.c b/sound/soc/intel/boards/cht_bsw_max98090_ti.c index 33eb72545be6..83b978e7b4c4 100644 --- a/sound/soc/intel/boards/cht_bsw_max98090_ti.c +++ b/sound/soc/intel/boards/cht_bsw_max98090_ti.c @@ -399,6 +399,20 @@ static struct snd_soc_card snd_soc_card_cht = { }; static const struct dmi_system_id cht_max98090_quirk_table[] = { + { + /* Banjo model Chromebook */ + .matches = { + DMI_MATCH(DMI_PRODUCT_NAME, "Banjo"), + }, + .driver_data = (void *)QUIRK_PMC_PLT_CLK_0, + }, + { + /* Candy model Chromebook */ + .matches = { + DMI_MATCH(DMI_PRODUCT_NAME, "Candy"), + }, + .driver_data = (void *)QUIRK_PMC_PLT_CLK_0, + }, { /* Clapper model Chromebook */ .matches = { @@ -406,6 +420,27 @@ static const struct dmi_system_id cht_max98090_quirk_table[] = { }, .driver_data = (void *)QUIRK_PMC_PLT_CLK_0, }, + { + /* Cyan model Chromebook */ + .matches = { + DMI_MATCH(DMI_PRODUCT_NAME, "Cyan"), + }, + .driver_data = (void *)QUIRK_PMC_PLT_CLK_0, + }, + { + /* Enguarde model Chromebook */ + .matches = { + DMI_MATCH(DMI_PRODUCT_NAME, "Enguarde"), + }, + .driver_data = (void *)QUIRK_PMC_PLT_CLK_0, + }, + { + /* Glimmer model Chromebook */ + .matches = { + DMI_MATCH(DMI_PRODUCT_NAME, "Glimmer"), + }, + .driver_data = (void *)QUIRK_PMC_PLT_CLK_0, + }, { /* Gnawty model Chromebook (Acer Chromebook CB3-111) */ .matches = { @@ -413,6 +448,62 @@ static const struct dmi_system_id cht_max98090_quirk_table[] = { }, .driver_data = (void *)QUIRK_PMC_PLT_CLK_0, }, + { + /* Heli model Chromebook */ + .matches = { + DMI_MATCH(DMI_PRODUCT_NAME, "Heli"), + }, + .driver_data = (void *)QUIRK_PMC_PLT_CLK_0, + }, + { + /* Kip model Chromebook */ + .matches = { + DMI_MATCH(DMI_PRODUCT_NAME, "Kip"), + }, + .driver_data = (void *)QUIRK_PMC_PLT_CLK_0, + }, + { + /* Ninja model Chromebook */ + .matches = { + DMI_MATCH(DMI_PRODUCT_NAME, "Ninja"), + }, + .driver_data = (void *)QUIRK_PMC_PLT_CLK_0, + }, + { + /* Orco model Chromebook */ + .matches = { + DMI_MATCH(DMI_PRODUCT_NAME, "Orco"), + }, + .driver_data = (void *)QUIRK_PMC_PLT_CLK_0, + }, + { + /* Quawks model Chromebook */ + .matches = { + DMI_MATCH(DMI_PRODUCT_NAME, "Quawks"), + }, + .driver_data = (void *)QUIRK_PMC_PLT_CLK_0, + }, + { + /* Rambi model Chromebook */ + .matches = { + DMI_MATCH(DMI_PRODUCT_NAME, "Rambi"), + }, + .driver_data = (void *)QUIRK_PMC_PLT_CLK_0, + }, + { + /* Squawks model Chromebook */ + .matches = { + DMI_MATCH(DMI_PRODUCT_NAME, "Squawks"), + }, + .driver_data = (void *)QUIRK_PMC_PLT_CLK_0, + }, + { + /* Sumo model Chromebook */ + .matches = { + DMI_MATCH(DMI_PRODUCT_NAME, "Sumo"), + }, + .driver_data = (void *)QUIRK_PMC_PLT_CLK_0, + }, { /* Swanky model Chromebook (Toshiba Chromebook 2) */ .matches = { @@ -420,6 +511,13 @@ static const struct dmi_system_id cht_max98090_quirk_table[] = { }, .driver_data = (void *)QUIRK_PMC_PLT_CLK_0, }, + { + /* Winky model Chromebook */ + .matches = { + DMI_MATCH(DMI_PRODUCT_NAME, "Winky"), + }, + .driver_data = (void *)QUIRK_PMC_PLT_CLK_0, + }, {} }; -- 2.20.1

1 0

[alsa-devel] [PATCH] ASoC: uniphier: Fix double reset assersion when transitioning to suspend state
by Kunihiko Hayashi 20 Aug '19

20 Aug '19

When transitioning to supend state, uniphier_aio_dai_suspend() is called and asserts reset lines and disables clocks. However, if there are two or more DAIs, uniphier_aio_dai_suspend() are called multiple times, and double reset assersion will cause. This patch defines the counter that has the number of DAIs at first, and whenever uniphier_aio_dai_suspend() are called, it decrements the counter. And only if the counter is zero, it asserts reset lines and disables clocks. In the same way, uniphier_aio_dai_resume() are called, it increments the counter after deasserting reset lines and enabling clocks. Fixes: 139a34200233 ("ASoC: uniphier: add support for UniPhier AIO CPU DAI driver") Signed-off-by: Kunihiko Hayashi <hayashi.kunihiko(a)socionext.com> --- sound/soc/uniphier/aio-cpu.c | 31 +++++++++++++++++++++---------- sound/soc/uniphier/aio.h | 1 + 2 files changed, 22 insertions(+), 10 deletions(-) diff --git a/sound/soc/uniphier/aio-cpu.c b/sound/soc/uniphier/aio-cpu.c index ee90e6c..2ae582a 100644 --- a/sound/soc/uniphier/aio-cpu.c +++ b/sound/soc/uniphier/aio-cpu.c @@ -424,8 +424,11 @@ int uniphier_aio_dai_suspend(struct snd_soc_dai *dai) { struct uniphier_aio *aio = uniphier_priv(dai); - reset_control_assert(aio->chip->rst); - clk_disable_unprepare(aio->chip->clk); + aio->chip->num_wup_aios--; + if (!aio->chip->num_wup_aios) { + reset_control_assert(aio->chip->rst); + clk_disable_unprepare(aio->chip->clk); + } return 0; } @@ -439,13 +442,15 @@ int uniphier_aio_dai_resume(struct snd_soc_dai *dai) if (!aio->chip->active) return 0; - ret = clk_prepare_enable(aio->chip->clk); - if (ret) - return ret; + if (!aio->chip->num_wup_aios) { + ret = clk_prepare_enable(aio->chip->clk); + if (ret) + return ret; - ret = reset_control_deassert(aio->chip->rst); - if (ret) - goto err_out_clock; + ret = reset_control_deassert(aio->chip->rst); + if (ret) + goto err_out_clock; + } aio_iecout_set_enable(aio->chip, true); aio_chip_init(aio->chip); @@ -458,7 +463,7 @@ int uniphier_aio_dai_resume(struct snd_soc_dai *dai) ret = aio_init(sub); if (ret) - goto err_out_clock; + goto err_out_reset; if (!sub->setting) continue; @@ -466,11 +471,16 @@ int uniphier_aio_dai_resume(struct snd_soc_dai *dai) aio_port_reset(sub); aio_src_reset(sub); } + aio->chip->num_wup_aios++; return 0; +err_out_reset: + if (!aio->chip->num_wup_aios) + reset_control_assert(aio->chip->rst); err_out_clock: - clk_disable_unprepare(aio->chip->clk); + if (!aio->chip->num_wup_aios) + clk_disable_unprepare(aio->chip->clk); return ret; } @@ -619,6 +629,7 @@ int uniphier_aio_probe(struct platform_device *pdev) return PTR_ERR(chip->rst); chip->num_aios = chip->chip_spec->num_dais; + chip->num_wup_aios = chip->num_aios; chip->aios = devm_kcalloc(dev, chip->num_aios, sizeof(struct uniphier_aio), GFP_KERNEL); diff --git a/sound/soc/uniphier/aio.h b/sound/soc/uniphier/aio.h index ca6ccba..a7ff7e5 100644 --- a/sound/soc/uniphier/aio.h +++ b/sound/soc/uniphier/aio.h @@ -285,6 +285,7 @@ struct uniphier_aio_chip { struct uniphier_aio *aios; int num_aios; + int num_wup_aios; struct uniphier_aio_pll *plls; int num_plls; -- 2.7.4

3 2

[alsa-devel] [PATCH v2 1/2] ASoC: wm8904: fix typo in DAPM kcontrol name
by Michał Mirosław 20 Aug '19

20 Aug '19

Trivial fix for typo in "Capture Inverting Mux"es' name. Signed-off-by: Michał Mirosław <mirq-linux(a)rere.qmqm.pl> Acked-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- v2: no changes --- sound/soc/codecs/wm8904.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index 5ebdd1d9afde..525e4ef654a1 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -866,7 +866,7 @@ static SOC_ENUM_SINGLE_DECL(lin_inv_enum, WM8904_ANALOGUE_LEFT_INPUT_1, 4, lin_text); static const struct snd_kcontrol_new lin_inv_mux = - SOC_DAPM_ENUM("Left Capture Inveting Mux", lin_inv_enum); + SOC_DAPM_ENUM("Left Capture Inverting Mux", lin_inv_enum); static const char *rin_text[] = { "IN1R", "IN2R", "IN3R" @@ -882,7 +882,7 @@ static SOC_ENUM_SINGLE_DECL(rin_inv_enum, WM8904_ANALOGUE_RIGHT_INPUT_1, 4, rin_text); static const struct snd_kcontrol_new rin_inv_mux = - SOC_DAPM_ENUM("Right Capture Inveting Mux", rin_inv_enum); + SOC_DAPM_ENUM("Right Capture Inverting Mux", rin_inv_enum); static const char *aif_text[] = { "Left", "Right" -- 2.20.1

1 1

[alsa-devel] [PATCH] fix odd_ptr_err.cocci warnings
by Julia Lawall 20 Aug '19

20 Aug '19

From: kbuild test robot <lkp(a)intel.com> PTR_ERR should normally access the value just tested by IS_ERR Generated by: scripts/coccinelle/tests/odd_ptr_err.cocci Fixes: 62c9c1442c8f ("j721e new machine driver wip") Signed-off-by: kbuild test robot <lkp(a)intel.com> Signed-off-by: Julia Lawall <julia.lawall(a)lip6.fr> --- tree: https://github.com/omap-audio/linux-audio peter/ti-linux-4.19.y/wip head: 62c9c1442c8f61ca93e62e1a9d8318be0abd9d9a commit: 62c9c1442c8f61ca93e62e1a9d8318be0abd9d9a [34/34] j721e new machine driver wip :::::: branch date: 20 hours ago :::::: commit date: 20 hours ago j721e-evm.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) --- a/sound/soc/ti/j721e-evm.c +++ b/sound/soc/ti/j721e-evm.c @@ -283,7 +283,7 @@ static int j721e_get_clocks(struct platf parent = devm_clk_get(&pdev->dev, clk_name); kfree(clk_name); if (IS_ERR(parent)) { - ret = PTR_ERR(clocks->parent); + ret = PTR_ERR(parent); if (ret != -EPROBE_DEFER) dev_err(&pdev->dev, "failed to acquire %s': %d\n", prefix, ret); @@ -299,7 +299,7 @@ static int j721e_get_clocks(struct platf parent = devm_clk_get(&pdev->dev, clk_name); kfree(clk_name); if (IS_ERR(parent)) { - ret = PTR_ERR(clocks->parent); + ret = PTR_ERR(parent); if (ret != -EPROBE_DEFER) dev_err(&pdev->dev, "failed to acquire %s': %d\n", prefix, ret);

4 5

[alsa-devel] [PATCH 1/2] ASoC: wm8904: fix typo in DAPM kcontrol name
by Michał Mirosław 20 Aug '19

20 Aug '19

Trivial fix for typo in "Capture Inverting Mux"es' name. Signed-off-by: Michał Mirosław <mirq-linux(a)rere.qmqm.pl> --- sound/soc/codecs/wm8904.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index 5ebdd1d9afde..525e4ef654a1 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -866,7 +866,7 @@ static SOC_ENUM_SINGLE_DECL(lin_inv_enum, WM8904_ANALOGUE_LEFT_INPUT_1, 4, lin_text); static const struct snd_kcontrol_new lin_inv_mux = - SOC_DAPM_ENUM("Left Capture Inveting Mux", lin_inv_enum); + SOC_DAPM_ENUM("Left Capture Inverting Mux", lin_inv_enum); static const char *rin_text[] = { "IN1R", "IN2R", "IN3R" @@ -882,7 +882,7 @@ static SOC_ENUM_SINGLE_DECL(rin_inv_enum, WM8904_ANALOGUE_RIGHT_INPUT_1, 4, rin_text); static const struct snd_kcontrol_new rin_inv_mux = - SOC_DAPM_ENUM("Right Capture Inveting Mux", rin_inv_enum); + SOC_DAPM_ENUM("Right Capture Inverting Mux", rin_inv_enum); static const char *aif_text[] = { "Left", "Right" -- 2.20.1

2 3

[alsa-devel] [PATCH v2] ALSA: hda/ca0132 - Add new SBZ quirk
by Paweł Rekowski 20 Aug '19

20 Aug '19

This patch adds a new PCI subsys ID for the SBZ, as found and tested by me and some reddit users. Signed-off-by: Paweł Rekowski <p.rekowski(a)gmail.com> --- sound/pci/hda/patch_ca0132.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index c3096796ee05..c41865e1222c 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -1175,6 +1175,7 @@ static const struct snd_pci_quirk ca0132_quirks[] = { SND_PCI_QUIRK(0x1028, 0x0708, "Alienware 15 R2 2016", QUIRK_ALIENWARE), SND_PCI_QUIRK(0x1102, 0x0010, "Sound Blaster Z", QUIRK_SBZ), SND_PCI_QUIRK(0x1102, 0x0023, "Sound Blaster Z", QUIRK_SBZ), + SND_PCI_QUIRK(0x1102, 0x0027, "Sound Blaster Z", QUIRK_SBZ), SND_PCI_QUIRK(0x1102, 0x0033, "Sound Blaster ZxR", QUIRK_SBZ), SND_PCI_QUIRK(0x1458, 0xA016, "Recon3Di", QUIRK_R3DI), SND_PCI_QUIRK(0x1458, 0xA026, "Gigabyte G1.Sniper Z97", QUIRK_R3DI), -- 2.23.0

2 1

[alsa-devel] [PATCH] ALSA: usb-audio: Add implicit fb quirk for Behringer UFX1604
by Takashi Iwai 20 Aug '19

20 Aug '19

Behringer UFX1604 requires the similar quirk to apply implicit fb like another Behringer model UFX1204 in order to fix the noisy playback. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=204631 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/pcm.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 75b96929f76c..e4bbf79de956 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -339,6 +339,7 @@ static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs, ep = 0x81; ifnum = 2; goto add_sync_ep_from_ifnum; + case USB_ID(0x1397, 0x0001): /* Behringer UFX1604 */ case USB_ID(0x1397, 0x0002): /* Behringer UFX1204 */ ep = 0x81; ifnum = 1; -- 2.16.4

1 0

[alsa-devel] [PATCH 00/15] ASoC: use modern style for aux_dev
by Kuninori Morimoto 20 Aug '19

20 Aug '19

Hi Mark, aux_dev users I'm sorry to bother you. I know I'm posting soc-core.c cleanup patches, and these are still under reviewing, but I want to post next patches before my summer vacation. These patches support modern style for aux_dev. I think it should work, but I can't test, because my platform doesn't use aux_dev. Please test and review these. These exchange aux_dev "style" only, do nothing from "technical" point of view. Kuninori Morimoto (15): ASoC: soc-core: support snd_soc_dai_link_component for aux_dev ASoC: simple-card: use snd_soc_dai_link_component for aux_dev ASoC: intel: cht_bsw_max98090_ti: use snd_soc_dai_link_component for aux_dev ASoC: mediatek: mt8183-da7219-max98357: use snd_soc_dai_link_component for aux_dev ASoC: mediatek: mt8183-mt6358-ts3a227-max98357: use snd_soc_dai_link_component for aux_dev ASoC: meson: axg-card: use snd_soc_dai_link_component for aux_dev ASoC: rockchip: rockchip_max98090: use snd_soc_dai_link_component for aux_dev ASoC: samsung: neo1973_wm8753: use snd_soc_dai_link_component for aux_dev ASoC: samsung: speyside: use snd_soc_dai_link_component for aux_dev ASoC: samsung: tm2_wm5110: use snd_soc_dai_link_component for aux_dev ASoC: sunxi: sun4i-codec: use snd_soc_dai_link_component for aux_dev ASoC: ti: rx51: use snd_soc_dai_link_component for aux_dev ASoC: soc-core: remove legacy style of aux_dev ASoC: soc-core: add for_each_xxx macro for aux_dev ASoC: soc-core: add soc_unbind_aux_dev() include/sound/soc.h | 16 +++++-- sound/soc/generic/simple-card.c | 2 +- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 3 +- sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c | 6 +-- .../mt8183/mt8183-mt6358-ts3a227-max98357.c | 6 +-- sound/soc/meson/axg-card.c | 11 ++--- sound/soc/rockchip/rockchip_max98090.c | 6 +-- sound/soc/samsung/neo1973_wm8753.c | 3 +- sound/soc/samsung/speyside.c | 3 +- sound/soc/samsung/tm2_wm5110.c | 10 ++--- sound/soc/soc-core.c | 52 ++++++++-------------- sound/soc/sunxi/sun4i-codec.c | 14 +++--- sound/soc/ti/rx51.c | 14 +++--- 13 files changed, 66 insertions(+), 80 deletions(-) -- 2.7.4

7 41

[alsa-devel] [PATCH] Fix an OOB bug in uac_mixer_unit_bmControls
by Hui Peng 20 Aug '19

20 Aug '19

`uac_mixer_unit_get_channels` calls `uac_mixer_unit_bmControls` to get pointer to bmControls field. The current implementation of `uac_mixer_unit_get_channels` does properly check the size of uac_mixer_unit_descriptor descriptor and may allow OOB access in `uac_mixer_unit_bmControls`. Reported-by: Hui Peng <benquike(a)gmail.com> Reported-by: Mathias Payer <mathias.payer(a)nebelwelt.net> Signed-off-by: Hui Peng <benquike(a)gmail.com> --- sound/usb/mixer.c | 25 ++++++++++++++++++------- 1 file changed, 18 insertions(+), 7 deletions(-) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index b5927c3d5bc0..00e6274a63c3 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -738,28 +738,39 @@ static int get_cluster_channels_v3(struct mixer_build *state, unsigned int clust static int uac_mixer_unit_get_channels(struct mixer_build *state, struct uac_mixer_unit_descriptor *desc) { - int mu_channels; + int mu_channels = 0; void *c; - if (desc->bLength < sizeof(*desc)) - return -EINVAL; if (!desc->bNrInPins) return -EINVAL; - if (desc->bLength < sizeof(*desc) + desc->bNrInPins) - return -EINVAL; switch (state->mixer->protocol) { case UAC_VERSION_1: + // limit derived from uac_mixer_unit_bmControls + if (desc->bLength < sizeof(*desc) + desc->bNrInPins + 4) + return 0; + + mu_channels = uac_mixer_unit_bNrChannels(desc); + break; + case UAC_VERSION_2: - default: - if (desc->bLength < sizeof(*desc) + desc->bNrInPins + 1) + // limit derived from uac_mixer_unit_bmControls + if (desc->bLength < sizeof(*desc) + desc->bNrInPins + 6) return 0; /* no bmControls -> skip */ + mu_channels = uac_mixer_unit_bNrChannels(desc); break; case UAC_VERSION_3: + // limit derived from uac_mixer_unit_bmControls + if (desc->bLength < sizeof(*desc) + desc->bNrInPins + 2) + return 0; /* no bmControls -> skip */ + mu_channels = get_cluster_channels_v3(state, uac3_mixer_unit_wClusterDescrID(desc)); break; + + default: + break; } if (!mu_channels) -- 2.22.1

1 1

[alsa-devel] [PATCH 0/3] ASoC: SOF: initial support for i.MX8
by Pierre-Louis Bossart 20 Aug '19

20 Aug '19

The reviews for these patches took place already on mailing lists and GitHub, and the code is already integrated in the SOF tree (along with dependencies already accepted in the i.MX tree) As agreed with Daniel, I am sending these patches as part of my weekly updates so that we avoid conflicts with other contributions (the next series adds Intel ALH/SoundWire suport which changes the same files) Daniel Baluta (3): ASoC: SOF: Add OF DSP device support ASoC: SOF: imx: Add i.MX8 HW support ASoC: SOF: topology: Add dummy support for i.MX8 DAIs include/sound/sof/dai.h | 2 + include/uapi/sound/sof/tokens.h | 8 + sound/soc/sof/Kconfig | 11 + sound/soc/sof/Makefile | 4 + sound/soc/sof/imx/Kconfig | 22 ++ sound/soc/sof/imx/Makefile | 4 + sound/soc/sof/imx/imx8.c | 394 ++++++++++++++++++++++++++++++++ sound/soc/sof/sof-of-dev.c | 143 ++++++++++++ sound/soc/sof/topology.c | 30 +++ 9 files changed, 618 insertions(+) create mode 100644 sound/soc/sof/imx/Kconfig create mode 100644 sound/soc/sof/imx/Makefile create mode 100644 sound/soc/sof/imx/imx8.c create mode 100644 sound/soc/sof/sof-of-dev.c -- 2.20.1

5 13

[alsa-devel] [PATCH] ALSA: hda - Fixes inverted Conexant GPIO mic mute led
by jeronimo＠borque.com.ar 19 Aug '19

19 Aug '19

From: Jeronimo Borque <jeronimo(a)borque.com.ar> "enabled" parameter historically referred to the device input or output, not to the led indicator. After the changes added with the led helper functions the mic mute led logic refers to the led and not to the mic input which caused led indicator to be negated (Mic mute led was on when the input enabled) Fixing it in the call to cxt_update_gpio_led at the cxt_gpio_micmute_update hook. Maybe more changes are required to be consistent everywhere. Signed-off-by: Jeronimo Borque <jeronimo(a)borque.com.ar> --- sound/pci/hda/patch_conexant.c | 7 +++++-- 1 file changed, 5 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index f299f137eaea..8edf0d1290b5 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -636,6 +636,10 @@ static void cxt_update_gpio_led(struct hda_codec *codec, unsigned int mask, spec->gpio_led &= ~mask; else spec->gpio_led |= mask; + + codec_dbg(codec, "mask:%d enabled:%d gpio_led:%d\n", + mask, enabled, spec->gpio_led); + if (spec->gpio_led != oldval) snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA, spec->gpio_led); @@ -656,7 +660,7 @@ static void cxt_gpio_micmute_update(struct hda_codec *codec) struct conexant_spec *spec = codec->spec; cxt_update_gpio_led(codec, spec->gpio_mic_led_mask, - spec->gen.micmute_led.led_value); + !spec->gen.micmute_led.led_value); } @@ -669,7 +673,6 @@ static void cxt_fixup_mute_led_gpio(struct hda_codec *codec, { 0x01, AC_VERB_SET_GPIO_DIRECTION, 0x03 }, {} }; - codec_info(codec, "action: %d gpio_led: %d\n", action, spec->gpio_led); if (action == HDA_FIXUP_ACT_PRE_PROBE) { spec->gen.vmaster_mute.hook = cxt_fixup_gpio_mute_hook; -- 2.21.0

3 9

[alsa-devel] [bug report] ASoC: SOF: topology: use set_get_data in process load
by Dan Carpenter 19 Aug '19

19 Aug '19

Hello Jaska Uimonen, The patch cac974a51ebb: "ASoC: SOF: topology: use set_get_data in process load" from Aug 9, 2019, leads to the following static checker warning: sound/soc/sof/topology.c:1857 sof_process_load() warn: 'ipc_data_size' unsigned <= 0 sound/soc/sof/topology.c 1825 static int sof_process_load(struct snd_soc_component *scomp, int index, 1826 struct snd_sof_widget *swidget, 1827 struct snd_soc_tplg_dapm_widget *tw, 1828 struct sof_ipc_comp_reply *r, 1829 int type) 1830 { 1831 struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(scomp); 1832 struct snd_soc_dapm_widget *widget = swidget->widget; 1833 struct snd_soc_tplg_private *private = &tw->priv; 1834 struct sof_ipc_comp_process *process = NULL; 1835 struct sof_widget_data *wdata = NULL; 1836 size_t ipc_data_size = 0; ^^^^^^^^^^^^^^^^^^^^ Unsigned. 1837 size_t ipc_size; 1838 int offset = 0; 1839 int ret = 0; 1840 int i; 1841 1842 if (type == SOF_COMP_NONE) { 1843 dev_err(sdev->dev, "error: invalid process comp type %d\n", 1844 type); 1845 return -EINVAL; 1846 } 1847 1848 /* allocate struct for widget control data sizes and types */ 1849 if (widget->num_kcontrols) { 1850 wdata = kcalloc(widget->num_kcontrols, 1851 sizeof(*wdata), 1852 GFP_KERNEL); 1853 1854 if (!wdata) 1855 return -ENOMEM; 1856 1857 /* get possible component controls and get size of all pdata */ 1858 ipc_data_size = sof_get_control_data(sdev, widget, wdata); 1859 1860 if (ipc_data_size <= 0) { The sof_get_control_data() returns negative error codes, but it's also declared as unsigned. But the main reason I'm reporting this instead of fixing it myself (besides laziness) is that I was wondering if it was really handling zero returns correctly. Should we really return success in that situation? 1861 ret = ipc_data_size; 1862 goto out; 1863 } 1864 } 1865 regards, dan carpenter

1 0

[alsa-devel] es8316 multiple issues
by H. C. Dornelas 19 Aug '19

19 Aug '19

Hi. Using only ALSA on my Positivo 2 em 1 Q432A (motherboard CHT12CP, es8316 sound card, kernel 5.2.9), speakers only work with headphones plugged in. Using Pulseaudio, speakers work without headphones, but many other problems remain: no microphones work; speakers are muted every boot and anytime headphones are unplugged, have no volume control and are only a mute/unmute switch; there are more than 160 controls in alsamixer's bytcht-es8316 card, most having no effect on sound output; and I still couldn't test hdmi audio. Here's the alsa-info file: http://alsa-project.org/db/?f=ee6dd9bdcf11a192fcccba9ab1b7abb2f3999b5d UCM files might solve these issues, but I have no experience with those, so any help would be dearly appreciated. If any other info is needed, just tell me and I'll add it. Thank you very much.

1 0

[alsa-devel] [PATCH] Fix an OOB bug in uac_mixer_unit_bmControls
by Hui Peng 17 Aug '19

17 Aug '19

`uac_mixer_unit_get_channels` calls `uac_mixer_unit_bmControls` to get pointer to bmControls field. The current implementation of `uac_mixer_unit_get_channels` does properly check the size of uac_mixer_unit_descriptor descriptor and may allow OOB access in `uac_mixer_unit_bmControls`. Reported-by: Hui Peng <benquike(a)gmail.com> Reported-by: Mathias Payer <mathias.payer(a)nebelwelt.net> Signed-off-by: Hui Peng <benquike(a)gmail.com> --- sound/usb/mixer.c | 25 ++++++++++++++++++------- 1 file changed, 18 insertions(+), 7 deletions(-) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index b5927c3d5bc0..00e6274a63c3 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -738,28 +738,39 @@ static int get_cluster_channels_v3(struct mixer_build *state, unsigned int clust static int uac_mixer_unit_get_channels(struct mixer_build *state, struct uac_mixer_unit_descriptor *desc) { - int mu_channels; + int mu_channels = 0; void *c; - if (desc->bLength < sizeof(*desc)) - return -EINVAL; if (!desc->bNrInPins) return -EINVAL; - if (desc->bLength < sizeof(*desc) + desc->bNrInPins) - return -EINVAL; switch (state->mixer->protocol) { case UAC_VERSION_1: + // limit derived from uac_mixer_unit_bmControls + if (desc->bLength < sizeof(*desc) + desc->bNrInPins + 4) + return 0; + + mu_channels = uac_mixer_unit_bNrChannels(desc); + break; + case UAC_VERSION_2: - default: - if (desc->bLength < sizeof(*desc) + desc->bNrInPins + 1) + // limit derived from uac_mixer_unit_bmControls + if (desc->bLength < sizeof(*desc) + desc->bNrInPins + 6) return 0; /* no bmControls -> skip */ + mu_channels = uac_mixer_unit_bNrChannels(desc); break; case UAC_VERSION_3: + // limit derived from uac_mixer_unit_bmControls + if (desc->bLength < sizeof(*desc) + desc->bNrInPins + 2) + return 0; /* no bmControls -> skip */ + mu_channels = get_cluster_channels_v3(state, uac3_mixer_unit_wClusterDescrID(desc)); break; + + default: + break; } if (!mu_channels) -- 2.22.1

2 5

Re: [alsa-devel] INFO: rcu detected stall in __do_softirq
by Srinivas Kandagatla 17 Aug '19

17 Aug '19

On 16/08/2019 01:10, syzbot wrote: > syzbot has bisected this bug to: > > commit 2aeac95d1a4cc85aae57ab842d5c3340df0f817f > Author: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> > Date: Tue Jun 11 10:40:41 2019 +0000 > > soundwire: add module_sdw_driver helper macro Not sure how adding a macro with no users triggers this rcu stall. BTW this was in mainline since rc1. --srini > > bisection log: https://syzkaller.appspot.com/x/bisect.txt?x=114b45ee600000 > start commit: 882e8691 Add linux-next specific files for 20190801 > git tree: linux-next > final crash: https://syzkaller.appspot.com/x/report.txt?x=134b45ee600000 > console output: https://syzkaller.appspot.com/x/log.txt?x=154b45ee600000 > kernel config: https://syzkaller.appspot.com/x/.config?x=466b331af3f34e94 > dashboard link: > https://syzkaller.appspot.com/bug?extid=6593c6b8c8b66a07cd98 > syz repro: https://syzkaller.appspot.com/x/repro.syz?x=16b216b2600000 > > Reported-by: syzbot+6593c6b8c8b66a07cd98(a)syzkaller.appspotmail.com > Fixes: 2aeac95d1a4c ("soundwire: add module_sdw_driver helper macro") > > For information about bisection process see: > https://goo.gl/tpsmEJ#bisection

2 1

[alsa-devel] ALSA HW PARAMS GET BUFFER TIME ERROR
by 李松鞠 17 Aug '19

17 Aug '19

Hello! Dear Developers, I am from HNU University (Changsha, China), a fresh man in ALSA sound developing. Recently, a alsa bug( snd_pcm_hw_params_get_buffer_time snd_pcm_hw_params_get_period_time ) occured to me which made me confused! Firstly (Step 1) I set hardware params normally(rate: 44100, channel: 2, format: S16_LE, access: raw_interleaved, period size: not set, buffer size: not set), It was OK. Laterly (Step 2) I get hw_params and print it, it was not surprising that they were the same as what I set (buffer_time == period_time * 1024, buffer_size == period_size * 1024). But! In another code ( Add Step 3)I used snd_pcm_format_name((snd_pcm_format_t)val), snd_pcm_format_description( (snd_pcm_format_t)val) to display the format information, It got wrong buffer size(buffer_time == period_time, buffer_size == period_size * 1024 ). the Attachments are code pcm_setting.cpp(With Step 1, 2), pcm_open_seeting.cpp(With Step 1, 2, 3). My alsa-lib version is about 1.1.0 . pcm_setting.cpp running result: get rate: 44100 get channles: 1 period time: 46439 buffer time is 47554467 period size: 2048 frames buffer size default: 2097152 frames periods per buffer = 1024 frames pcm_open_setting running result: PCM handle name = 'default' PCM state = PREPARED access type = RW_INTERLEAVED format(length: 2) = 'S16_LE' (Signed 16 bit Little Endian) channels = 1 rate = 44100 bps period time = 46438 us buffer time = 46438 us period size = 2048 frames buffer size = 2097152 frames periods per buffer = 1024 frames

1 0

Re: [alsa-devel] [PATCH] ASoC: imx-audmux: Add driver suspend and resume to support MEGA Fast
by S.j. Wang 17 Aug '19

17 Aug '19

Hi Mark > > On Fri, Aug 16, 2019 at 01:03:14AM -0400, Shengjiu Wang wrote: > > > + for (i = 0; i < reg_max; i++) > > + regcache[i] = readl(audmux_base + i * 4); > > If only there were some framework which provided a register cache! 😝 Yes, next step I can refine this driver to use the regmap. Best regards Wang shengjiu

1 0

[alsa-devel] [PATCH] ASoC: imx-audmux: Add driver suspend and resume to support MEGA Fast
by Shengjiu Wang 16 Aug '19

16 Aug '19

For i.MX6 SoloX, there is a mode of the SoC to shutdown all power source of modules during system suspend and resume procedure. Thus, AUDMUX needs to save all the values of registers before the system suspend and restore them after the system resume. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/imx-audmux.c | 54 +++++++++++++++++++++++++++++++++++++- 1 file changed, 53 insertions(+), 1 deletion(-) diff --git a/sound/soc/fsl/imx-audmux.c b/sound/soc/fsl/imx-audmux.c index 7595f24a006e..3ce85a43e08f 100644 --- a/sound/soc/fsl/imx-audmux.c +++ b/sound/soc/fsl/imx-audmux.c @@ -23,6 +23,8 @@ static struct clk *audmux_clk; static void __iomem *audmux_base; +static u32 *regcache; +static u32 reg_max; #define IMX_AUDMUX_V2_PTCR(x) ((x) * 8) #define IMX_AUDMUX_V2_PDCR(x) ((x) * 8 + 4) @@ -315,8 +317,23 @@ static int imx_audmux_probe(struct platform_device *pdev) if (of_id) pdev->id_entry = of_id->data; audmux_type = pdev->id_entry->driver_data; - if (audmux_type == IMX31_AUDMUX) + + switch (audmux_type) { + case IMX31_AUDMUX: audmux_debugfs_init(); + reg_max = 14; + break; + case IMX21_AUDMUX: + reg_max = 6; + break; + default: + dev_err(&pdev->dev, "unsupported version!\n"); + return -EINVAL; + } + + regcache = devm_kzalloc(&pdev->dev, sizeof(u32) * reg_max, GFP_KERNEL); + if (!regcache) + return -ENOMEM; if (of_id) imx_audmux_parse_dt_defaults(pdev, pdev->dev.of_node); @@ -332,12 +349,47 @@ static int imx_audmux_remove(struct platform_device *pdev) return 0; } +#ifdef CONFIG_PM_SLEEP +static int imx_audmux_suspend(struct device *dev) +{ + int i; + + clk_prepare_enable(audmux_clk); + + for (i = 0; i < reg_max; i++) + regcache[i] = readl(audmux_base + i * 4); + + clk_disable_unprepare(audmux_clk); + + return 0; +} + +static int imx_audmux_resume(struct device *dev) +{ + int i; + + clk_prepare_enable(audmux_clk); + + for (i = 0; i < reg_max; i++) + writel(regcache[i], audmux_base + i * 4); + + clk_disable_unprepare(audmux_clk); + + return 0; +} +#endif /* CONFIG_PM_SLEEP */ + +static const struct dev_pm_ops imx_audmux_pm = { + SET_SYSTEM_SLEEP_PM_OPS(imx_audmux_suspend, imx_audmux_resume) +}; + static struct platform_driver imx_audmux_driver = { .probe = imx_audmux_probe, .remove = imx_audmux_remove, .id_table = imx_audmux_ids, .driver = { .name = DRIVER_NAME, + .pm = &imx_audmux_pm, .of_match_table = imx_audmux_dt_ids, } }; -- 2.21.0

2 2

[alsa-devel] [PATCH 0/4] ASoC: SOF: i.MX8 SAI and Intel ALH/SoundWire DAI support
by Pierre-Louis Bossart 16 Aug '19

16 Aug '19

This patchset adds the basic functionality to extract DAI information from the topology and send the information to firmware. The i.MX8 SAI/ESAI definitions are minimal placeholders for now, they are included to keep the alignment with the firmware/topology. For the Intel ALH, there is no specific configuration in topology file, all the programming takes place based on the dai index and the related ALH stream ID. The renaming from SoundWire to ALH (Audio Link Hub) is a better representation of the hardware, the ALH acts as a proxy between the DMA transfers and the Cadence IP. Bard liao (1): ASoC: SOF: rename SOUNDWIRE to ALH Daniel Baluta (1): ASoC: SOF: topology: Add dummy support for i.MX8 DAIs Pierre-Louis Bossart (2): ASoC: SOF: topology: initial support for Intel ALH DAI type ASoC: SOF: pcm: add ALH support include/sound/sof/dai.h | 4 +- include/uapi/sound/sof/tokens.h | 8 ++++ sound/soc/sof/pcm.c | 3 ++ sound/soc/sof/topology.c | 72 ++++++++++++++++++++++++++++++++- 4 files changed, 85 insertions(+), 2 deletions(-) -- 2.20.1

2 8

[alsa-devel] [PATCH v2 0/3] soundwire: debugfs support for 5.4
by Pierre-Louis Bossart 16 Aug '19

16 Aug '19

This patchset enables debugfs support and corrects all the feedback provided on an earlier RFC ('soundwire: updates for 5.4') There is one remaining hard-coded value in intel.c that will need to be fixed in a follow-up patchset not specific to debugfs: we need to remove hard-coded Intel-specific configurations from cadence_master.c (PDI offsets, etc). Changes since v1 (Feedback from GKH) Handle debugfs in a more self-contained way (no dentry as return or parameter) Used CONFIG_DEBUG_FS in structures and code to make it easier to remove if need be. No functional change for register dumps. Changes since RFC (Feedback from GKH, Vinod, Guennadi, Cezary, Sanyog): removed error checks used DEFINE_SHOW_ATTRIBUTE and seq_file fixed copyright dates fixed SPDX license info to use GPL2.0 only fixed Makefile to include debugfs only if CONFIG_DEBUG_FS is selected used static inlines for fallback compilation removed intermediate variables removed hard-coded constants in loops (used registers offsets and hardware capabilities) squashed patch 3 Pierre-Louis Bossart (3): soundwire: add debugfs support soundwire: cadence_master: add debugfs register dump soundwire: intel: add debugfs register dump drivers/soundwire/Makefile | 4 + drivers/soundwire/bus.c | 6 ++ drivers/soundwire/bus.h | 16 +++ drivers/soundwire/bus_type.c | 3 + drivers/soundwire/cadence_master.c | 107 ++++++++++++++++++++ drivers/soundwire/cadence_master.h | 4 + drivers/soundwire/debugfs.c | 151 +++++++++++++++++++++++++++++ drivers/soundwire/intel.c | 121 +++++++++++++++++++++++ drivers/soundwire/slave.c | 1 + include/linux/soundwire/sdw.h | 8 ++ 10 files changed, 421 insertions(+) create mode 100644 drivers/soundwire/debugfs.c -- 2.20.1

4 7

[alsa-devel] [PATCH] ASoC: core: Move pcm_mutex up to card level from snd_soc_pcm_runtime
by Peter Ujfalusi 16 Aug '19

16 Aug '19

The pcm_mutex is used to prevent concurrent execution of snd_pcm_ops callbacks. This works fine most of the cases but it can not handle setups when the same DAI is used by different rtd, for example: pcm3168a have two DAIs: one for Playback and one for Capture. If the codec is connected to a single CPU DAI we need to have two dai_link to support both playback and capture. In this case the snd_pcm_ops callbacks can be executed in parallel causing unexpected races in DAI drivers. By moving the pcm_mutex up to card level this can be solved while - hopefully - not breaking other setups. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- Hi, I have tested the patch on several boards. it fixes the issue with single card with multiple dai_links where the CPU side (mcasp) is the same. However I can not test with anything which use DPCM. It would be great if the patch would gather few tested-by from folks having access to more complicated setups. The backround of this patch: https://mailman.alsa-project.org/pipermail/alsa-devel/2019-August/153864.ht… Regards, Peter include/sound/soc.h | 6 +++-- sound/soc/soc-compress.c | 48 ++++++++++++++++++++-------------------- sound/soc/soc-core.c | 2 +- sound/soc/soc-pcm.c | 36 +++++++++++++++--------------- 4 files changed, 47 insertions(+), 45 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 6ac6481b4882..ec2dbe0ac2aa 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -990,6 +990,10 @@ struct snd_soc_card { struct mutex mutex; struct mutex dapm_mutex; + /* Mutex for PCM operations */ + struct mutex pcm_mutex; + enum snd_soc_pcm_subclass pcm_subclass; + spinlock_t dpcm_lock; bool instantiated; @@ -1107,8 +1111,6 @@ struct snd_soc_pcm_runtime { struct device *dev; struct snd_soc_card *card; struct snd_soc_dai_link *dai_link; - struct mutex pcm_mutex; - enum snd_soc_pcm_subclass pcm_subclass; struct snd_pcm_ops ops; unsigned int params_select; /* currently selected param for dai link */ diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index 289211069a1e..9e54d8ae6d2c 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -80,7 +80,7 @@ static int soc_compr_open(struct snd_compr_stream *cstream) struct snd_soc_dai *cpu_dai = rtd->cpu_dai; int ret; - mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); + mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass); if (cpu_dai->driver->cops && cpu_dai->driver->cops->startup) { ret = cpu_dai->driver->cops->startup(cstream, cpu_dai); @@ -108,7 +108,7 @@ static int soc_compr_open(struct snd_compr_stream *cstream) snd_soc_runtime_activate(rtd, cstream->direction); - mutex_unlock(&rtd->pcm_mutex); + mutex_unlock(&rtd->card->pcm_mutex); return 0; @@ -118,7 +118,7 @@ static int soc_compr_open(struct snd_compr_stream *cstream) if (cpu_dai->driver->cops && cpu_dai->driver->cops->shutdown) cpu_dai->driver->cops->shutdown(cstream, cpu_dai); out: - mutex_unlock(&rtd->pcm_mutex); + mutex_unlock(&rtd->card->pcm_mutex); return ret; } @@ -224,7 +224,7 @@ static void close_delayed_work(struct work_struct *work) container_of(work, struct snd_soc_pcm_runtime, delayed_work.work); struct snd_soc_dai *codec_dai = rtd->codec_dai; - mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); + mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass); dev_dbg(rtd->dev, "Compress ASoC: pop wq checking: %s status: %s waiting: %s\n", @@ -239,7 +239,7 @@ static void close_delayed_work(struct work_struct *work) SND_SOC_DAPM_STREAM_STOP); } - mutex_unlock(&rtd->pcm_mutex); + mutex_unlock(&rtd->card->pcm_mutex); } static int soc_compr_free(struct snd_compr_stream *cstream) @@ -249,7 +249,7 @@ static int soc_compr_free(struct snd_compr_stream *cstream) struct snd_soc_dai *codec_dai = rtd->codec_dai; int stream; - mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); + mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass); if (cstream->direction == SND_COMPRESS_PLAYBACK) stream = SNDRV_PCM_STREAM_PLAYBACK; @@ -292,7 +292,7 @@ static int soc_compr_free(struct snd_compr_stream *cstream) SND_SOC_DAPM_STREAM_STOP); } - mutex_unlock(&rtd->pcm_mutex); + mutex_unlock(&rtd->card->pcm_mutex); return 0; } @@ -375,7 +375,7 @@ static int soc_compr_trigger(struct snd_compr_stream *cstream, int cmd) struct snd_soc_dai *cpu_dai = rtd->cpu_dai; int ret; - mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); + mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass); ret = soc_compr_components_trigger(cstream, cmd); if (ret < 0) @@ -394,7 +394,7 @@ static int soc_compr_trigger(struct snd_compr_stream *cstream, int cmd) } out: - mutex_unlock(&rtd->pcm_mutex); + mutex_unlock(&rtd->card->pcm_mutex); return ret; } @@ -480,7 +480,7 @@ static int soc_compr_set_params(struct snd_compr_stream *cstream, struct snd_soc_dai *cpu_dai = rtd->cpu_dai; int ret; - mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); + mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass); /* * First we call set_params for the CPU DAI, then the component @@ -514,14 +514,14 @@ static int soc_compr_set_params(struct snd_compr_stream *cstream, /* cancel any delayed stream shutdown that is pending */ rtd->pop_wait = 0; - mutex_unlock(&rtd->pcm_mutex); + mutex_unlock(&rtd->card->pcm_mutex); cancel_delayed_work_sync(&rtd->delayed_work); return 0; err: - mutex_unlock(&rtd->pcm_mutex); + mutex_unlock(&rtd->card->pcm_mutex); return ret; } @@ -593,7 +593,7 @@ static int soc_compr_get_params(struct snd_compr_stream *cstream, struct snd_soc_dai *cpu_dai = rtd->cpu_dai; int ret = 0; - mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); + mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass); if (cpu_dai->driver->cops && cpu_dai->driver->cops->get_params) { ret = cpu_dai->driver->cops->get_params(cstream, params, cpu_dai); @@ -613,7 +613,7 @@ static int soc_compr_get_params(struct snd_compr_stream *cstream, } err: - mutex_unlock(&rtd->pcm_mutex); + mutex_unlock(&rtd->card->pcm_mutex); return ret; } @@ -625,7 +625,7 @@ static int soc_compr_get_caps(struct snd_compr_stream *cstream, struct snd_soc_rtdcom_list *rtdcom; int ret = 0; - mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); + mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass); for_each_rtdcom(rtd, rtdcom) { component = rtdcom->component; @@ -638,7 +638,7 @@ static int soc_compr_get_caps(struct snd_compr_stream *cstream, break; } - mutex_unlock(&rtd->pcm_mutex); + mutex_unlock(&rtd->card->pcm_mutex); return ret; } @@ -650,7 +650,7 @@ static int soc_compr_get_codec_caps(struct snd_compr_stream *cstream, struct snd_soc_rtdcom_list *rtdcom; int ret = 0; - mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); + mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass); for_each_rtdcom(rtd, rtdcom) { component = rtdcom->component; @@ -664,7 +664,7 @@ static int soc_compr_get_codec_caps(struct snd_compr_stream *cstream, break; } - mutex_unlock(&rtd->pcm_mutex); + mutex_unlock(&rtd->card->pcm_mutex); return ret; } @@ -676,7 +676,7 @@ static int soc_compr_ack(struct snd_compr_stream *cstream, size_t bytes) struct snd_soc_dai *cpu_dai = rtd->cpu_dai; int ret = 0; - mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); + mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass); if (cpu_dai->driver->cops && cpu_dai->driver->cops->ack) { ret = cpu_dai->driver->cops->ack(cstream, bytes, cpu_dai); @@ -697,7 +697,7 @@ static int soc_compr_ack(struct snd_compr_stream *cstream, size_t bytes) } err: - mutex_unlock(&rtd->pcm_mutex); + mutex_unlock(&rtd->card->pcm_mutex); return ret; } @@ -710,7 +710,7 @@ static int soc_compr_pointer(struct snd_compr_stream *cstream, int ret = 0; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); + mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass); if (cpu_dai->driver->cops && cpu_dai->driver->cops->pointer) cpu_dai->driver->cops->pointer(cstream, tstamp, cpu_dai); @@ -726,7 +726,7 @@ static int soc_compr_pointer(struct snd_compr_stream *cstream, break; } - mutex_unlock(&rtd->pcm_mutex); + mutex_unlock(&rtd->card->pcm_mutex); return ret; } @@ -738,7 +738,7 @@ static int soc_compr_copy(struct snd_compr_stream *cstream, struct snd_soc_rtdcom_list *rtdcom; int ret = 0; - mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); + mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass); for_each_rtdcom(rtd, rtdcom) { component = rtdcom->component; @@ -751,7 +751,7 @@ static int soc_compr_copy(struct snd_compr_stream *cstream, break; } - mutex_unlock(&rtd->pcm_mutex); + mutex_unlock(&rtd->card->pcm_mutex); return ret; } diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index bb1e9e2c4ff4..7f2064889a23 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1344,7 +1344,6 @@ static int soc_post_component_init(struct snd_soc_pcm_runtime *rtd, rtd->dev->groups = soc_dev_attr_groups; dev_set_name(rtd->dev, "%s", name); dev_set_drvdata(rtd->dev, rtd); - mutex_init(&rtd->pcm_mutex); INIT_LIST_HEAD(&rtd->dpcm[SNDRV_PCM_STREAM_PLAYBACK].be_clients); INIT_LIST_HEAD(&rtd->dpcm[SNDRV_PCM_STREAM_CAPTURE].be_clients); INIT_LIST_HEAD(&rtd->dpcm[SNDRV_PCM_STREAM_PLAYBACK].fe_clients); @@ -2395,6 +2394,7 @@ int snd_soc_register_card(struct snd_soc_card *card) card->instantiated = 0; mutex_init(&card->mutex); mutex_init(&card->dapm_mutex); + mutex_init(&card->pcm_mutex); spin_lock_init(&card->dpcm_lock); return snd_soc_bind_card(card); diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 77c986fe08d0..93bb4a347779 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -36,7 +36,7 @@ * Increments the active count for all the DAIs and components attached to a PCM * runtime. Should typically be called when a stream is opened. * - * Must be called with the rtd->pcm_mutex being held + * Must be called with the rtd->card->pcm_mutex being held */ void snd_soc_runtime_activate(struct snd_soc_pcm_runtime *rtd, int stream) { @@ -44,7 +44,7 @@ void snd_soc_runtime_activate(struct snd_soc_pcm_runtime *rtd, int stream) struct snd_soc_dai *codec_dai; int i; - lockdep_assert_held(&rtd->pcm_mutex); + lockdep_assert_held(&rtd->card->pcm_mutex); if (stream == SNDRV_PCM_STREAM_PLAYBACK) { cpu_dai->playback_active++; @@ -72,7 +72,7 @@ void snd_soc_runtime_activate(struct snd_soc_pcm_runtime *rtd, int stream) * Decrements the active count for all the DAIs and components attached to a PCM * runtime. Should typically be called when a stream is closed. * - * Must be called with the rtd->pcm_mutex being held + * Must be called with the rtd->card->pcm_mutex being held */ void snd_soc_runtime_deactivate(struct snd_soc_pcm_runtime *rtd, int stream) { @@ -80,7 +80,7 @@ void snd_soc_runtime_deactivate(struct snd_soc_pcm_runtime *rtd, int stream) struct snd_soc_dai *codec_dai; int i; - lockdep_assert_held(&rtd->pcm_mutex); + lockdep_assert_held(&rtd->card->pcm_mutex); if (stream == SNDRV_PCM_STREAM_PLAYBACK) { cpu_dai->playback_active--; @@ -506,7 +506,7 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) pm_runtime_get_sync(component->dev); } - mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); + mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass); /* startup the audio subsystem */ ret = snd_soc_dai_startup(cpu_dai, substream); @@ -604,7 +604,7 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) snd_soc_runtime_activate(rtd, substream->stream); - mutex_unlock(&rtd->pcm_mutex); + mutex_unlock(&rtd->card->pcm_mutex); return 0; config_err: @@ -623,7 +623,7 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) snd_soc_dai_shutdown(cpu_dai, substream); out: - mutex_unlock(&rtd->pcm_mutex); + mutex_unlock(&rtd->card->pcm_mutex); for_each_rtdcom(rtd, rtdcom) { component = rtdcom->component; @@ -653,7 +653,7 @@ static void close_delayed_work(struct work_struct *work) container_of(work, struct snd_soc_pcm_runtime, delayed_work.work); struct snd_soc_dai *codec_dai = rtd->codec_dais[0]; - mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); + mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass); dev_dbg(rtd->dev, "ASoC: pop wq checking: %s status: %s waiting: %s\n", codec_dai->driver->playback.stream_name, @@ -667,7 +667,7 @@ static void close_delayed_work(struct work_struct *work) SND_SOC_DAPM_STREAM_STOP); } - mutex_unlock(&rtd->pcm_mutex); + mutex_unlock(&rtd->card->pcm_mutex); } static void codec2codec_close_delayed_work(struct work_struct *work) @@ -694,7 +694,7 @@ static int soc_pcm_close(struct snd_pcm_substream *substream) struct snd_soc_dai *codec_dai; int i; - mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); + mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass); snd_soc_runtime_deactivate(rtd, substream->stream); @@ -738,7 +738,7 @@ static int soc_pcm_close(struct snd_pcm_substream *substream) SND_SOC_DAPM_STREAM_STOP); } - mutex_unlock(&rtd->pcm_mutex); + mutex_unlock(&rtd->card->pcm_mutex); for_each_rtdcom(rtd, rtdcom) { component = rtdcom->component; @@ -771,7 +771,7 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) struct snd_soc_dai *codec_dai; int i, ret = 0; - mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); + mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass); if (rtd->dai_link->ops->prepare) { ret = rtd->dai_link->ops->prepare(substream); @@ -826,7 +826,7 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) snd_soc_dai_digital_mute(cpu_dai, 0, substream->stream); out: - mutex_unlock(&rtd->pcm_mutex); + mutex_unlock(&rtd->card->pcm_mutex); return ret; } @@ -876,7 +876,7 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *codec_dai; int i, ret = 0; - mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); + mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass); if (rtd->dai_link->ops->hw_params) { ret = rtd->dai_link->ops->hw_params(substream, params); if (ret < 0) { @@ -962,7 +962,7 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, if (ret) goto component_err; out: - mutex_unlock(&rtd->pcm_mutex); + mutex_unlock(&rtd->card->pcm_mutex); return ret; component_err: @@ -986,7 +986,7 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, if (rtd->dai_link->ops->hw_free) rtd->dai_link->ops->hw_free(substream); - mutex_unlock(&rtd->pcm_mutex); + mutex_unlock(&rtd->card->pcm_mutex); return ret; } @@ -1001,7 +1001,7 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream) bool playback = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; int i; - mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); + mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass); /* clear the corresponding DAIs parameters when going to be inactive */ if (cpu_dai->active == 1) { @@ -1043,7 +1043,7 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream) snd_soc_dai_hw_free(cpu_dai, substream); - mutex_unlock(&rtd->pcm_mutex); + mutex_unlock(&rtd->card->pcm_mutex); return 0; } -- Peter Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki. Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki

3 5

[alsa-devel] [GIT PULL] soundwire fixes for v5.4-rc5
by Vinod Koul 16 Aug '19

16 Aug '19

Hi Greg, We have couple of fixes queued up, please pull. Some more are in review, will send them later. These fixes are in linux-next as well. The following changes since commit 5f9e832c137075045d15cd6899ab0505cfb2ca4b: Linus 5.3-rc1 (2019-07-21 14:05:38 -0700) are available in the Git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/vkoul/soundwire.git tags/soundwire-5.3-rc5 for you to fetch changes up to 8676b3ca4673517650fd509d7fa586aff87b3c28: soundwire: fix regmap dependencies and align with other serial links (2019-08-09 10:20:40 +0530) ---------------------------------------------------------------- soundwire fixes for v5.3-rc5 Pierre sent fixes which are queued now for v5.3-rc5 are: - regmap dependecy - cadence register definitions ---------------------------------------------------------------- Pierre-Louis Bossart (3): soundwire: cadence_master: fix register definition for SLAVE_STATE soundwire: cadence_master: fix definitions for INTSTAT0/1 soundwire: fix regmap dependencies and align with other serial links drivers/base/regmap/Kconfig | 2 +- drivers/soundwire/Kconfig | 7 +------ drivers/soundwire/Makefile | 2 +- drivers/soundwire/cadence_master.c | 8 ++++---- 4 files changed, 7 insertions(+), 12 deletions(-) Thanks -- ~Vinod

2 1

[alsa-devel] [PATCH v3 1/2] ALSA: hda - Expand pin_match function to match upcoming new tbls
by Hui Wang 16 Aug '19

16 Aug '19

With the existing pintbl, we already have many entries in it. it is better to figure out a new way to reduce the size of the pintbl. We plan to define a new tbl which will match more machines with a single tbl, To do that, this function doesn't need to match all valid pins between machine and tbl, it just needs to match all pins defined in the tbl with the machine. And the plan is to move some tbls from pin_fixup_tbl to fallback_pin_fixup_tbl gradually. Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/hda_auto_parser.c | 12 ++++++++---- sound/pci/hda/hda_local.h | 3 ++- sound/pci/hda/patch_realtek.c | 4 ++-- 3 files changed, 12 insertions(+), 7 deletions(-) diff --git a/sound/pci/hda/hda_auto_parser.c b/sound/pci/hda/hda_auto_parser.c index 92390d457567..8a134f750bae 100644 --- a/sound/pci/hda/hda_auto_parser.c +++ b/sound/pci/hda/hda_auto_parser.c @@ -884,7 +884,8 @@ EXPORT_SYMBOL_GPL(snd_hda_apply_fixup); #define IGNORE_SEQ_ASSOC (~(AC_DEFCFG_SEQUENCE | AC_DEFCFG_DEF_ASSOC)) static bool pin_config_match(struct hda_codec *codec, - const struct hda_pintbl *pins) + const struct hda_pintbl *pins, + bool match_all_pins) { const struct hda_pincfg *pin; int i; @@ -908,7 +909,8 @@ static bool pin_config_match(struct hda_codec *codec, return false; } } - if (!found && (cfg & 0xf0000000) != 0x40000000) + if (match_all_pins && + !found && (cfg & 0xf0000000) != 0x40000000) return false; } @@ -920,10 +922,12 @@ static bool pin_config_match(struct hda_codec *codec, * @codec: the HDA codec * @pin_quirk: zero-terminated pin quirk list * @fixlist: the fixup list + * @match_all_pins: all valid pins must match with the table entries */ void snd_hda_pick_pin_fixup(struct hda_codec *codec, const struct snd_hda_pin_quirk *pin_quirk, - const struct hda_fixup *fixlist) + const struct hda_fixup *fixlist, + bool match_all_pins) { const struct snd_hda_pin_quirk *pq; @@ -935,7 +939,7 @@ void snd_hda_pick_pin_fixup(struct hda_codec *codec, continue; if (codec->core.vendor_id != pq->codec) continue; - if (pin_config_match(codec, pq->pins)) { + if (pin_config_match(codec, pq->pins, match_all_pins)) { codec->fixup_id = pq->value; #ifdef CONFIG_SND_DEBUG_VERBOSE codec->fixup_name = pq->name; diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index 349a8312d06a..3942e1b528d8 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -361,7 +361,8 @@ void snd_hda_pick_fixup(struct hda_codec *codec, const struct hda_fixup *fixlist); void snd_hda_pick_pin_fixup(struct hda_codec *codec, const struct snd_hda_pin_quirk *pin_quirk, - const struct hda_fixup *fixlist); + const struct hda_fixup *fixlist, + bool match_all_pins); /* helper macros to retrieve pin default-config values */ #define get_defcfg_connect(cfg) \ diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index ac7177ce67eb..2fa03fb6bd9e 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7950,7 +7950,7 @@ static int patch_alc269(struct hda_codec *codec) snd_hda_pick_fixup(codec, alc269_fixup_models, alc269_fixup_tbl, alc269_fixups); - snd_hda_pick_pin_fixup(codec, alc269_pin_fixup_tbl, alc269_fixups); + snd_hda_pick_pin_fixup(codec, alc269_pin_fixup_tbl, alc269_fixups, true); snd_hda_pick_fixup(codec, NULL, alc269_fixup_vendor_tbl, alc269_fixups); snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE); @@ -8935,7 +8935,7 @@ static int patch_alc662(struct hda_codec *codec) snd_hda_pick_fixup(codec, alc662_fixup_models, alc662_fixup_tbl, alc662_fixups); - snd_hda_pick_pin_fixup(codec, alc662_pin_fixup_tbl, alc662_fixups); + snd_hda_pick_pin_fixup(codec, alc662_pin_fixup_tbl, alc662_fixups, true); snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE); alc_auto_parse_customize_define(codec); -- 2.17.1

2 3

[alsa-devel] [PATCH] Fix a stack buffer overflow bug check_input_term
by Hui Peng 15 Aug '19

15 Aug '19

`check_input_term` recursively calls itself with input from device side (e.g., uac_input_terminal_descriptor.bCSourceID) as argument (id). In `check_input_term`, if `check_input_term` is called with the same `id` argument as the caller, it triggers endless recursive call, resulting kernel space stack overflow. This patch fixes the bug by adding a bitmap to `struct mixer_build` to keep track of the checked ids by `check_input_term` and stop the execution if some id has been checked (similar to how parse_audio_unit handles unitid argument). Reported-by: Hui Peng <benquike(a)gmail.com> Reported-by: Mathias Payer <mathias.payer(a)nebelwelt.net> Signed-off-by: Hui Peng <benquike(a)gmail.com> --- sound/usb/mixer.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index ea487378be17..1f6c8213df82 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -68,6 +68,7 @@ struct mixer_build { unsigned char *buffer; unsigned int buflen; DECLARE_BITMAP(unitbitmap, MAX_ID_ELEMS); + DECLARE_BITMAP(termbitmap, MAX_ID_ELEMS); struct usb_audio_term oterm; const struct usbmix_name_map *map; const struct usbmix_selector_map *selector_map; @@ -782,6 +783,8 @@ static int check_input_term(struct mixer_build *state, int id, int err; void *p1; + if (test_and_set_bit(id, state->termbitmap)) + return 0; memset(term, 0, sizeof(*term)); while ((p1 = find_audio_control_unit(state, id)) != NULL) { unsigned char *hdr = p1; -- 2.22.1

2 7

[alsa-devel] [PATCH] Revert "ALSA: hda: Add codec on bus address table lately"
by Takashi Iwai 15 Aug '19

15 Aug '19

This reverts commit ee5f85d9290f ("ALSA: hda: Add codec on bus address table lately"). The commit caused several regression since I've overlooked that the function doesn't manage only the caddr_tbl but also the codec linked list that is referred indirectly in the other drivers. Revert for now to make everything back to work. Fixes: ee5f85d9290f ("ALSA: hda: Add codec on bus address table lately") Reported-by: Chris Wilson <chris(a)chris-wilson.co.uk> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/hda/hdac_device.c | 21 +++++++++------------ 1 file changed, 9 insertions(+), 12 deletions(-) diff --git a/sound/hda/hdac_device.c b/sound/hda/hdac_device.c index bf83d7062ef6..9f3e37511408 100644 --- a/sound/hda/hdac_device.c +++ b/sound/hda/hdac_device.c @@ -61,6 +61,10 @@ int snd_hdac_device_init(struct hdac_device *codec, struct hdac_bus *bus, pm_runtime_get_noresume(&codec->dev); atomic_set(&codec->in_pm, 0); + err = snd_hdac_bus_add_device(bus, codec); + if (err < 0) + goto error; + /* fill parameters */ codec->vendor_id = snd_hdac_read_parm(codec, AC_NODE_ROOT, AC_PAR_VENDOR_ID); @@ -139,22 +143,15 @@ int snd_hdac_device_register(struct hdac_device *codec) err = device_add(&codec->dev); if (err < 0) return err; - err = snd_hdac_bus_add_device(codec->bus, codec); - if (err < 0) - goto error; mutex_lock(&codec->widget_lock); err = hda_widget_sysfs_init(codec); mutex_unlock(&codec->widget_lock); - if (err < 0) - goto error_remove; + if (err < 0) { + device_del(&codec->dev); + return err; + } return 0; - - error_remove: - snd_hdac_bus_remove_device(codec->bus, codec); - error: - device_del(&codec->dev); - return err; } EXPORT_SYMBOL_GPL(snd_hdac_device_register); @@ -168,8 +165,8 @@ void snd_hdac_device_unregister(struct hdac_device *codec) mutex_lock(&codec->widget_lock); hda_widget_sysfs_exit(codec); mutex_unlock(&codec->widget_lock); - snd_hdac_bus_remove_device(codec->bus, codec); device_del(&codec->dev); + snd_hdac_bus_remove_device(codec->bus, codec); } } EXPORT_SYMBOL_GPL(snd_hdac_device_unregister); -- 2.16.4

1 0

[alsa-devel] [PATCH 0/4] ASoC: SOF: initial support for Intel ALH/SoundWire DAIs
by Pierre-Louis Bossart 15 Aug '19

15 Aug '19

This patchset adds the basic functionality to extract DAI information from the topology and send the information to firmware. At the moment there is no specific configuration in topology file, all the programming takes place based on the dai index and the related ALH stream ID. The renaming from SoundWire to ALH (Audio Link Hub) is a better representation of the hardware, the ALH acts as a proxy between the DMA transfers and the Cadence IP. Bard liao (1): ASoC: SOF: rename SOUNDWIRE to ALH Pierre-Louis Bossart (3): ASoC: SOF: ipc: add ALH parameters ASoC: SOF: topology: initial support for Intel ALH DAI type ASoC: SOF: pcm: add ALH support include/sound/sof/dai-intel.h | 9 ++++++++ include/sound/sof/dai.h | 3 ++- include/uapi/sound/sof/abi.h | 2 +- sound/soc/sof/pcm.c | 3 +++ sound/soc/sof/topology.c | 42 ++++++++++++++++++++++++++++++++++- 5 files changed, 56 insertions(+), 3 deletions(-) -- 2.20.1

2 8

Re: [alsa-devel] [PATCH] ASoC: intel: cht_bsw_max98090_ti: Add all Chromebooks that need pmc_plt_clk_0 quirk
by Hans de Goede 15 Aug '19

15 Aug '19

Hi, On 15-08-19 19:12, Daniel Stuart wrote: > Every single baytrail chromebook sets PMC to 0, as can be seeing > below by searching through coreboot source code: > $ grep -rl "PMC_PLT_CLK\[0\]" . > ./rambi/variants/glimmer/devicetree.cb > ./rambi/variants/clapper/devicetree.cb > ./rambi/variants/swanky/devicetree.cb > ./rambi/variants/enguarde/devicetree.cb > ./rambi/variants/winky/devicetree.cb > ./rambi/variants/kip/devicetree.cb > ./rambi/variants/squawks/devicetree.cb > ./rambi/variants/orco/devicetree.cb > ./rambi/variants/ninja/devicetree.cb > ./rambi/variants/heli/devicetree.cb > ./rambi/variants/sumo/devicetree.cb > ./rambi/variants/banjo/devicetree.cb > ./rambi/variants/candy/devicetree.cb > ./rambi/variants/gnawty/devicetree.cb > ./rambi/variants/rambi/devicetree.cb > ./rambi/variants/quawks/devicetree.cb > > Plus, Cyan (only non-baytrail chromebook with max98090) also needs > this patch for audio to work. > > Thus, this commit adds all the missing devices to bsw_max98090 quirk > table, implemented by commit a182ecd3809c ("ASoC: intel: > cht_bsw_max98090_ti: Add quirk for boards using pmc_plt_clk_0"). > > Signed-off-by: Daniel Stuart <daniel.stuart14(a)gmail.com> Thank you for catching this, this patch looks good to me: Reviewed-by: Hans de Goede <hdegoede(a)redhat.com> Regards, Hans > --- > sound/soc/intel/boards/cht_bsw_max98090_ti.c | 98 ++++++++++++++++++++ > 1 file changed, 98 insertions(+) > > diff --git a/sound/soc/intel/boards/cht_bsw_max98090_ti.c b/sound/soc/intel/boards/cht_bsw_max98090_ti.c > index 33eb72545be6..83b978e7b4c4 100644 > --- a/sound/soc/intel/boards/cht_bsw_max98090_ti.c > +++ b/sound/soc/intel/boards/cht_bsw_max98090_ti.c > @@ -399,6 +399,20 @@ static struct snd_soc_card snd_soc_card_cht = { > }; > > static const struct dmi_system_id cht_max98090_quirk_table[] = { > + { > + /* Banjo model Chromebook */ > + .matches = { > + DMI_MATCH(DMI_PRODUCT_NAME, "Banjo"), > + }, > + .driver_data = (void *)QUIRK_PMC_PLT_CLK_0, > + }, > + { > + /* Candy model Chromebook */ > + .matches = { > + DMI_MATCH(DMI_PRODUCT_NAME, "Candy"), > + }, > + .driver_data = (void *)QUIRK_PMC_PLT_CLK_0, > + }, > { > /* Clapper model Chromebook */ > .matches = { > @@ -406,6 +420,27 @@ static const struct dmi_system_id cht_max98090_quirk_table[] = { > }, > .driver_data = (void *)QUIRK_PMC_PLT_CLK_0, > }, > + { > + /* Cyan model Chromebook */ > + .matches = { > + DMI_MATCH(DMI_PRODUCT_NAME, "Cyan"), > + }, > + .driver_data = (void *)QUIRK_PMC_PLT_CLK_0, > + }, > + { > + /* Enguarde model Chromebook */ > + .matches = { > + DMI_MATCH(DMI_PRODUCT_NAME, "Enguarde"), > + }, > + .driver_data = (void *)QUIRK_PMC_PLT_CLK_0, > + }, > + { > + /* Glimmer model Chromebook */ > + .matches = { > + DMI_MATCH(DMI_PRODUCT_NAME, "Glimmer"), > + }, > + .driver_data = (void *)QUIRK_PMC_PLT_CLK_0, > + }, > { > /* Gnawty model Chromebook (Acer Chromebook CB3-111) */ > .matches = { > @@ -413,6 +448,62 @@ static const struct dmi_system_id cht_max98090_quirk_table[] = { > }, > .driver_data = (void *)QUIRK_PMC_PLT_CLK_0, > }, > + { > + /* Heli model Chromebook */ > + .matches = { > + DMI_MATCH(DMI_PRODUCT_NAME, "Heli"), > + }, > + .driver_data = (void *)QUIRK_PMC_PLT_CLK_0, > + }, > + { > + /* Kip model Chromebook */ > + .matches = { > + DMI_MATCH(DMI_PRODUCT_NAME, "Kip"), > + }, > + .driver_data = (void *)QUIRK_PMC_PLT_CLK_0, > + }, > + { > + /* Ninja model Chromebook */ > + .matches = { > + DMI_MATCH(DMI_PRODUCT_NAME, "Ninja"), > + }, > + .driver_data = (void *)QUIRK_PMC_PLT_CLK_0, > + }, > + { > + /* Orco model Chromebook */ > + .matches = { > + DMI_MATCH(DMI_PRODUCT_NAME, "Orco"), > + }, > + .driver_data = (void *)QUIRK_PMC_PLT_CLK_0, > + }, > + { > + /* Quawks model Chromebook */ > + .matches = { > + DMI_MATCH(DMI_PRODUCT_NAME, "Quawks"), > + }, > + .driver_data = (void *)QUIRK_PMC_PLT_CLK_0, > + }, > + { > + /* Rambi model Chromebook */ > + .matches = { > + DMI_MATCH(DMI_PRODUCT_NAME, "Rambi"), > + }, > + .driver_data = (void *)QUIRK_PMC_PLT_CLK_0, > + }, > + { > + /* Squawks model Chromebook */ > + .matches = { > + DMI_MATCH(DMI_PRODUCT_NAME, "Squawks"), > + }, > + .driver_data = (void *)QUIRK_PMC_PLT_CLK_0, > + }, > + { > + /* Sumo model Chromebook */ > + .matches = { > + DMI_MATCH(DMI_PRODUCT_NAME, "Sumo"), > + }, > + .driver_data = (void *)QUIRK_PMC_PLT_CLK_0, > + }, > { > /* Swanky model Chromebook (Toshiba Chromebook 2) */ > .matches = { > @@ -420,6 +511,13 @@ static const struct dmi_system_id cht_max98090_quirk_table[] = { > }, > .driver_data = (void *)QUIRK_PMC_PLT_CLK_0, > }, > + { > + /* Winky model Chromebook */ > + .matches = { > + DMI_MATCH(DMI_PRODUCT_NAME, "Winky"), > + }, > + .driver_data = (void *)QUIRK_PMC_PLT_CLK_0, > + }, > {} > }; > >

1 0

[alsa-devel] [PATCH] ASoC: rockchip: rockchip_max98090: Set period size to 240
by Cheng-Yi Chiang 15 Aug '19

15 Aug '19

>From stress testing of arecord, we found that period size greater than ~900 will bring pl330 to DYING state and can not recover within 100 iterations. The result is that arecord will stuck and get I/O error, and issue can not be recovered until reboot. This issue does not happen when period size is small. Set constraint of period size to 240 to prevent such issue. With the constraint, there will be no issue after 2000 iterations. We can revert this patch once the root cause is found in rockchip's pl330 implementation. Signed-off-by: Cheng-Yi Chiang <cychiang(a)chromium.org> --- sound/soc/rockchip/rockchip_max98090.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/soc/rockchip/rockchip_max98090.c b/sound/soc/rockchip/rockchip_max98090.c index 7b0c21fa6dca..0097df1fae66 100644 --- a/sound/soc/rockchip/rockchip_max98090.c +++ b/sound/soc/rockchip/rockchip_max98090.c @@ -137,8 +137,19 @@ static int rk_aif1_hw_params(struct snd_pcm_substream *substream, return ret; } +static int rk_aif1_startup(struct snd_pcm_substream *substream) +{ + /* + * Set period size to 240 because pl330 has issue + * dealing with larger period in stress testing. + */ + return snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_PERIOD_SIZE, 240, 240); +} + static const struct snd_soc_ops rk_aif1_ops = { .hw_params = rk_aif1_hw_params, + .startup = rk_aif1_startup, }; SND_SOC_DAILINK_DEFS(hifi, -- 2.23.0.rc1.153.gdeed80330f-goog

2 1

[alsa-devel] [PATCH] ASoC: Intel: hsw: remove i386 build warning w/ size_t argument
by Pierre-Louis Bossart 15 Aug '19

15 Aug '19

Recent changes in the common IPC code introduced a build warning with size_t fields, use the correct %zu format. include/linux/dynamic_debug.h:82:16: warning: format '%lu' expects argument of type 'long unsigned int', but argument 4 has type 'size_t' [-Wformat=] Fixes: abf31feea26c0 ('ASoC: Intel: Update request-reply IPC model') Reported-by: kbuild test robot <lkp(a)intel.com> Cc: Cezary Rojewski <cezary.rojewski(a)intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/intel/haswell/sst-haswell-ipc.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/haswell/sst-haswell-ipc.c b/sound/soc/intel/haswell/sst-haswell-ipc.c index 5c73b11375e3..0ff89ea96ccf 100644 --- a/sound/soc/intel/haswell/sst-haswell-ipc.c +++ b/sound/soc/intel/haswell/sst-haswell-ipc.c @@ -2015,11 +2015,11 @@ int sst_hsw_module_set_param(struct sst_hsw *hsw, if (payload_size <= SST_HSW_IPC_MAX_SHORT_PARAMETER_SIZE) { /* short parameter, mailbox can contain data */ - dev_dbg(dev, "transfer parameter size : %lu\n", + dev_dbg(dev, "transfer parameter size : %zu\n", request.size); request.size = ALIGN(payload_size, 4); - dev_dbg(dev, "transfer parameter aligned size : %lu\n", + dev_dbg(dev, "transfer parameter aligned size : %zu\n", request.size); parameter = kzalloc(request.size, GFP_KERNEL); -- 2.20.1

3 2

[alsa-devel] [PATCH] ASoC: SOF: fix HDA direct MMIO access
by Pierre-Louis Bossart 15 Aug '19

15 Aug '19

The recent change to remove the bus->io_ops callbacks used an older version of the SOF code base, and when merged into Mark's for-next it invalidated changes, resulting in broken compilation identified by kbuild and reproduced during the weekly SOF rebase. Restore SOF code overridden by git merge and apply Takashi's intended change in the 'right' location. Fixes: c2f16a94a8049 ("Merge branch 'topic/hda-bus-ops-cleanup'") Reported-by: kbuild test robot <lkp(a)intel.com> Cc: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/sof/intel/hda-ctrl.c | 2 +- sound/soc/sof/intel/hda-dsp.c | 39 ---------------------------------- 2 files changed, 1 insertion(+), 40 deletions(-) diff --git a/sound/soc/sof/intel/hda-ctrl.c b/sound/soc/sof/intel/hda-ctrl.c index a7fee403cb90..bc41028a7a01 100644 --- a/sound/soc/sof/intel/hda-ctrl.c +++ b/sound/soc/sof/intel/hda-ctrl.c @@ -254,7 +254,7 @@ int hda_dsp_ctrl_init_chip(struct snd_sof_dev *sdev, bool full_reset) #if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA) /* Reset stream-to-link mapping */ list_for_each_entry(hlink, &bus->hlink_list, list) - bus->io_ops->reg_writel(0, hlink->ml_addr + AZX_REG_ML_LOSIDV); + writel(0, hlink->ml_addr + AZX_REG_ML_LOSIDV); #endif bus->chip_init = true; diff --git a/sound/soc/sof/intel/hda-dsp.c b/sound/soc/sof/intel/hda-dsp.c index 097727cda5cb..fb55a3c5afd0 100644 --- a/sound/soc/sof/intel/hda-dsp.c +++ b/sound/soc/sof/intel/hda-dsp.c @@ -354,45 +354,6 @@ static int hda_resume(struct snd_sof_dev *sdev, bool runtime_resume) return ret; } - hda_dsp_ctrl_misc_clock_gating(sdev, false); - - /* Reset stream-to-link mapping */ - list_for_each_entry(hlink, &bus->hlink_list, list) - writel(0, hlink->ml_addr + AZX_REG_ML_LOSIDV); - - hda_dsp_ctrl_misc_clock_gating(sdev, true); -#else - - hda_dsp_ctrl_misc_clock_gating(sdev, false); - - /* reset controller */ - ret = hda_dsp_ctrl_link_reset(sdev, true); - if (ret < 0) { - dev_err(sdev->dev, - "error: failed to reset controller during resume\n"); - return ret; - } - - /* take controller out of reset */ - ret = hda_dsp_ctrl_link_reset(sdev, false); - if (ret < 0) { - dev_err(sdev->dev, - "error: failed to ready controller during resume\n"); - return ret; - } - - /* enable hda bus irq */ - snd_sof_dsp_update_bits(sdev, HDA_DSP_HDA_BAR, SOF_HDA_INTCTL, - SOF_HDA_INT_CTRL_EN | SOF_HDA_INT_GLOBAL_EN, - SOF_HDA_INT_CTRL_EN | SOF_HDA_INT_GLOBAL_EN); - - hda_dsp_ctrl_misc_clock_gating(sdev, true); -#endif - - /* enable ppcap interrupt */ - hda_dsp_ctrl_ppcap_enable(sdev, true); - hda_dsp_ctrl_ppcap_int_enable(sdev, true); - #if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA) /* check jack status */ if (runtime_resume) -- 2.20.1

2 1

[alsa-devel] [PATCH] ASoC: ti: Fix typos in ti/Kconfig
by Masanari Iida 15 Aug '19

15 Aug '19

This patch fixes some spelling typo in Kconfig. Signed-off-by: Masanari Iida <standby24x7(a)gmail.com> --- sound/soc/ti/Kconfig | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/ti/Kconfig b/sound/soc/ti/Kconfig index 2197f3e1eaed..87a9b9dd4e98 100644 --- a/sound/soc/ti/Kconfig +++ b/sound/soc/ti/Kconfig @@ -12,7 +12,7 @@ config SND_SOC_TI_SDMA_PCM comment "Texas Instruments DAI support for:" config SND_SOC_DAVINCI_ASP - tristate "daVinci Audio Serial Port (ASP) or McBSP suport" + tristate "daVinci Audio Serial Port (ASP) or McBSP support" depends on ARCH_DAVINCI || COMPILE_TEST select SND_SOC_TI_EDMA_PCM help @@ -33,7 +33,7 @@ config SND_SOC_DAVINCI_MCASP - Keystone devices config SND_SOC_DAVINCI_VCIF - tristate "daVinci Voice Interface (VCIF) suport" + tristate "daVinci Voice Interface (VCIF) support" depends on ARCH_DAVINCI || COMPILE_TEST select SND_SOC_TI_EDMA_PCM help -- 2.23.0.rc2

3 2

[alsa-devel] [PATCH -next] ASoC: es8328: Fix copy-paste error in es8328_right_line_controls
by YueHaibing 15 Aug '19

15 Aug '19

It seems 'es8328_rline_enum' should be used in es8328_right_line_controls Fixes: 567e4f98922c ("ASoC: add es8328 codec driver") Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/soc/codecs/es8328.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/es8328.c b/sound/soc/codecs/es8328.c index 4a3d303..fdf64c2 100644 --- a/sound/soc/codecs/es8328.c +++ b/sound/soc/codecs/es8328.c @@ -227,7 +227,7 @@ static const struct soc_enum es8328_rline_enum = ARRAY_SIZE(es8328_line_texts), es8328_line_texts); static const struct snd_kcontrol_new es8328_right_line_controls = - SOC_DAPM_ENUM("Route", es8328_lline_enum); + SOC_DAPM_ENUM("Route", es8328_rline_enum); /* Left Mixer */ static const struct snd_kcontrol_new es8328_left_mixer_controls[] = { -- 2.7.4

2 2

[alsa-devel] [PATCH -next] ASoC: soc-core: Fix -Wunused-const-variable warning
by YueHaibing 15 Aug '19

15 Aug '19

If CONFIG_DMI is not set, gcc warns: sound/soc/soc-core.c:81:27: warning: dmi_blacklist defined but not used [-Wunused-const-variable=] Add #ifdef guard around it. Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/soc/soc-core.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index bb1e9e2..dcf39eb 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -73,6 +73,7 @@ static int pmdown_time = 5000; module_param(pmdown_time, int, 0); MODULE_PARM_DESC(pmdown_time, "DAPM stream powerdown time (msecs)"); +#ifdef CONFIG_DMI /* * If a DMI filed contain strings in this blacklist (e.g. * "Type2 - Board Manufacturer" or "Type1 - TBD by OEM"), it will be taken @@ -87,6 +88,7 @@ static const char * const dmi_blacklist[] = { "Board Product Name", NULL, /* terminator */ }; +#endif static ssize_t pmdown_time_show(struct device *dev, struct device_attribute *attr, char *buf) -- 2.7.4

2 1

[alsa-devel] [PATCH -next] ASoC: mt2701: remove unused variables
by YueHaibing 15 Aug '19

15 Aug '19

sound/soc/mediatek/mt2701/mt2701-afe-pcm.c:799:38: warning: mt2701_afe_o23_mix defined but not used [-Wunused-const-variable=] sound/soc/mediatek/mt2701/mt2701-afe-pcm.c:803:38: warning: mt2701_afe_o24_mix defined but not used [-Wunused-const-variable=] sound/soc/mediatek/mt2701/mt2701-afe-pcm.c:835:38: warning: mt2701_afe_multi_ch_out_i2s4 defined but not used [-Wunused-const-variable=] They are never used, so can be removed. Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/soc/mediatek/mt2701/mt2701-afe-pcm.c | 13 ------------- 1 file changed, 13 deletions(-) diff --git a/sound/soc/mediatek/mt2701/mt2701-afe-pcm.c b/sound/soc/mediatek/mt2701/mt2701-afe-pcm.c index 9af76ae..d7f5def 100644 --- a/sound/soc/mediatek/mt2701/mt2701-afe-pcm.c +++ b/sound/soc/mediatek/mt2701/mt2701-afe-pcm.c @@ -796,14 +796,6 @@ static const struct snd_kcontrol_new mt2701_afe_o22_mix[] = { SOC_DAPM_SINGLE_AUTODISABLE("I19 Switch", AFE_CONN22, 19, 1, 0), }; -static const struct snd_kcontrol_new mt2701_afe_o23_mix[] = { - SOC_DAPM_SINGLE_AUTODISABLE("I20 Switch", AFE_CONN23, 20, 1, 0), -}; - -static const struct snd_kcontrol_new mt2701_afe_o24_mix[] = { - SOC_DAPM_SINGLE_AUTODISABLE("I21 Switch", AFE_CONN24, 21, 1, 0), -}; - static const struct snd_kcontrol_new mt2701_afe_o31_mix[] = { SOC_DAPM_SINGLE_AUTODISABLE("I35 Switch", AFE_CONN41, 9, 1, 0), }; @@ -832,11 +824,6 @@ static const struct snd_kcontrol_new mt2701_afe_multi_ch_out_i2s3[] = { PWR2_TOP_CON, 18, 1, 0), }; -static const struct snd_kcontrol_new mt2701_afe_multi_ch_out_i2s4[] = { - SOC_DAPM_SINGLE_AUTODISABLE("Multich I2S4 Out Switch", - PWR2_TOP_CON, 19, 1, 0), -}; - static const struct snd_soc_dapm_widget mt2701_afe_pcm_widgets[] = { /* inter-connections */ SND_SOC_DAPM_MIXER("I00", SND_SOC_NOPM, 0, 0, NULL, 0), -- 2.7.4

2 1

[alsa-devel] [PATCH -next] ASoC: cs4349: Use PM ops 'cs4349_runtime_pm'
by YueHaibing 15 Aug '19

15 Aug '19

sound/soc/codecs/cs4349.c:358:32: warning: cs4349_runtime_pm defined but not used [-Wunused-const-variable=] cs4349_runtime_pm ops already defined, it seems we should enable it. Reported-by: Hulk Robot <hulkci(a)huawei.com> Fixes: e40da86 ("ASoC: cs4349: Add support for Cirrus Logic CS4349") Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/soc/codecs/cs4349.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/cs4349.c b/sound/soc/codecs/cs4349.c index 09716fa..e0d44f6 100644 --- a/sound/soc/codecs/cs4349.c +++ b/sound/soc/codecs/cs4349.c @@ -378,6 +378,7 @@ static struct i2c_driver cs4349_i2c_driver = { .driver = { .name = "cs4349", .of_match_table = cs4349_of_match, + .pm = &cs4349_runtime_pm, }, .id_table = cs4349_i2c_id, .probe = cs4349_i2c_probe, -- 2.7.4

2 2

[alsa-devel] [PATCH -next] ASoC: wm8737: Fix copy-paste error in wm8737_snd_controls
by YueHaibing 15 Aug '19

15 Aug '19

sound/soc/codecs/wm8737.c:112:29: warning: high_3d defined but not used [-Wunused-const-variable=] 'high_3d' should be used for 3D High Cut-off. Reported-by: Hulk Robot <hulkci(a)huawei.com> Fixes: 2a9ae13a2641 ("ASoC: Add initial WM8737 driver") Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/soc/codecs/wm8737.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8737.c b/sound/soc/codecs/wm8737.c index 0c246fb..7a3f9fb 100644 --- a/sound/soc/codecs/wm8737.c +++ b/sound/soc/codecs/wm8737.c @@ -167,7 +167,7 @@ SOC_DOUBLE("Polarity Invert Switch", WM8737_ADC_CONTROL, 5, 6, 1, 0), SOC_SINGLE("3D Switch", WM8737_3D_ENHANCE, 0, 1, 0), SOC_SINGLE("3D Depth", WM8737_3D_ENHANCE, 1, 15, 0), SOC_ENUM("3D Low Cut-off", low_3d), -SOC_ENUM("3D High Cut-off", low_3d), +SOC_ENUM("3D High Cut-off", high_3d), SOC_SINGLE_TLV("3D ADC Volume", WM8737_3D_ENHANCE, 7, 1, 1, adc_tlv), SOC_SINGLE("Noise Gate Switch", WM8737_NOISE_GATE, 0, 1, 0), -- 2.7.4

3 2

[alsa-devel] [PATCH] ASoC: mediatek: mt6358: add delay after dmic clock on
by Jiaxin Yu 15 Aug '19

15 Aug '19

Most dmics produce a high level when they receive clock. The difference between power-on and memory record time is about 10ms, but the dmic needs 50ms to output normal data. This commit add 100ms delay after SoC output clock so that we can cut off the pop noise at the beginning. Signed-off-by: Jiaxin Yu <jiaxin.yu(a)mediatek.com> --- Hi, This patch is based on for-next and discussed with tzungbi(a)google.com. --- --- sound/soc/codecs/mt6358.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/codecs/mt6358.c b/sound/soc/codecs/mt6358.c index c17250a..bb737fd 100644 --- a/sound/soc/codecs/mt6358.c +++ b/sound/soc/codecs/mt6358.c @@ -1730,6 +1730,10 @@ static int mt6358_dmic_enable(struct mt6358_priv *priv) /* UL turn on */ regmap_write(priv->regmap, MT6358_AFE_UL_SRC_CON0_L, 0x0003); + + /* Prevent pop noise form dmic hw */ + msleep(100); + return 0; } -- 1.8.1.1.dirty

2 1