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

[PATCH] ASoC: meson: axg-toddr: fix channel order on g12 platforms
by Jerome Brunet 28 Aug '20

28 Aug '20

On g12 and following platforms, The first channel of record with more than 2 channels ends being placed randomly on an even channel of the output. On these SoCs, a bit was added to force the first channel to be placed at the beginning of the output. Apparently the behavior if the bit is not set is not easily predictable. According to the documentation, this bit is not present on the axg series. Set the bit on g12 and fix the problem. Fixes: a3c23a8ad4dc ("ASoC: meson: axg-toddr: add g12a support") Reported-by: Nicolas Belin <nbelin(a)baylibre.com> Signed-off-by: Jerome Brunet <jbrunet(a)baylibre.com> --- sound/soc/meson/axg-toddr.c | 24 +++++++++++++++++++++++- 1 file changed, 23 insertions(+), 1 deletion(-) diff --git a/sound/soc/meson/axg-toddr.c b/sound/soc/meson/axg-toddr.c index e711abcf8c12..d6adf7edea41 100644 --- a/sound/soc/meson/axg-toddr.c +++ b/sound/soc/meson/axg-toddr.c @@ -18,6 +18,7 @@ #define CTRL0_TODDR_SEL_RESAMPLE BIT(30) #define CTRL0_TODDR_EXT_SIGNED BIT(29) #define CTRL0_TODDR_PP_MODE BIT(28) +#define CTRL0_TODDR_SYNC_CH BIT(27) #define CTRL0_TODDR_TYPE_MASK GENMASK(15, 13) #define CTRL0_TODDR_TYPE(x) ((x) << 13) #define CTRL0_TODDR_MSB_POS_MASK GENMASK(12, 8) @@ -189,10 +190,31 @@ static const struct axg_fifo_match_data axg_toddr_match_data = { .dai_drv = &axg_toddr_dai_drv }; +static int g12a_toddr_dai_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct axg_fifo *fifo = snd_soc_dai_get_drvdata(dai); + int ret; + + ret = axg_toddr_dai_startup(substream, dai); + if (ret) + return ret; + + /* + * Make sure the first channel ends up in the at beginning of the output + * As weird as it looks, without this the first channel may be misplaced + * in memory, with a random shift of 2 channels. + */ + regmap_update_bits(fifo->map, FIFO_CTRL0, CTRL0_TODDR_SYNC_CH, + CTRL0_TODDR_SYNC_CH); + + return 0; +} + static const struct snd_soc_dai_ops g12a_toddr_ops = { .prepare = g12a_toddr_dai_prepare, .hw_params = axg_toddr_dai_hw_params, - .startup = axg_toddr_dai_startup, + .startup = g12a_toddr_dai_startup, .shutdown = axg_toddr_dai_shutdown, }; -- 2.25.4

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[PATCH] ASoC: tlv320adcx140: Add digital mic channel enable routing
by Dan Murphy 28 Aug '20

28 Aug '20

Add the audio routing map to enable the digital mic paths when the analog mic paths are not enabled. Signed-off-by: Dan Murphy <dmurphy(a)ti.com> --- sound/soc/codecs/tlv320adcx140.c | 37 ++++++++++++++++++++++++++++++++ 1 file changed, 37 insertions(+) diff --git a/sound/soc/codecs/tlv320adcx140.c b/sound/soc/codecs/tlv320adcx140.c index 5cd50d841177..5d1b2a03f0ac 100644 --- a/sound/soc/codecs/tlv320adcx140.c +++ b/sound/soc/codecs/tlv320adcx140.c @@ -412,6 +412,16 @@ static const struct snd_soc_dapm_widget adcx140_dapm_widgets[] = { SND_SOC_DAPM_ADC("CH3_ADC", "CH3 Capture", ADCX140_IN_CH_EN, 5, 0), SND_SOC_DAPM_ADC("CH4_ADC", "CH4 Capture", ADCX140_IN_CH_EN, 4, 0), + SND_SOC_DAPM_ADC("CH1_DIG", "CH1 Capture", ADCX140_IN_CH_EN, 7, 0), + SND_SOC_DAPM_ADC("CH2_DIG", "CH2 Capture", ADCX140_IN_CH_EN, 6, 0), + SND_SOC_DAPM_ADC("CH3_DIG", "CH3 Capture", ADCX140_IN_CH_EN, 5, 0), + SND_SOC_DAPM_ADC("CH4_DIG", "CH4 Capture", ADCX140_IN_CH_EN, 4, 0), + SND_SOC_DAPM_ADC("CH5_DIG", "CH5 Capture", ADCX140_IN_CH_EN, 3, 0), + SND_SOC_DAPM_ADC("CH6_DIG", "CH6 Capture", ADCX140_IN_CH_EN, 2, 0), + SND_SOC_DAPM_ADC("CH7_DIG", "CH7 Capture", ADCX140_IN_CH_EN, 1, 0), + SND_SOC_DAPM_ADC("CH8_DIG", "CH8 Capture", ADCX140_IN_CH_EN, 0, 0), + + SND_SOC_DAPM_SWITCH("CH1_ASI_EN", SND_SOC_NOPM, 0, 0, &adcx140_dapm_ch1_en_switch), SND_SOC_DAPM_SWITCH("CH2_ASI_EN", SND_SOC_NOPM, 0, 0, @@ -470,6 +480,15 @@ static const struct snd_soc_dapm_route adcx140_audio_map[] = { {"CH3_ASI_EN", "Switch", "CH3_ADC"}, {"CH4_ASI_EN", "Switch", "CH4_ADC"}, + {"CH1_ASI_EN", "Switch", "CH1_DIG"}, + {"CH2_ASI_EN", "Switch", "CH2_DIG"}, + {"CH3_ASI_EN", "Switch", "CH3_DIG"}, + {"CH4_ASI_EN", "Switch", "CH4_DIG"}, + {"CH5_ASI_EN", "Switch", "CH5_DIG"}, + {"CH6_ASI_EN", "Switch", "CH6_DIG"}, + {"CH7_ASI_EN", "Switch", "CH7_DIG"}, + {"CH8_ASI_EN", "Switch", "CH8_DIG"}, + {"CH5_ASI_EN", "Switch", "CH5_OUT"}, {"CH6_ASI_EN", "Switch", "CH6_OUT"}, {"CH7_ASI_EN", "Switch", "CH7_OUT"}, @@ -541,6 +560,15 @@ static const struct snd_soc_dapm_route adcx140_audio_map[] = { {"PDM Clk Div Select", "705.6 kHz", "MIC1P Input Mux"}, {"PDM Clk Div Select", "5.6448 MHz", "MIC1P Input Mux"}, + {"MIC1P Input Mux", NULL, "CH1_DIG"}, + {"MIC1M Input Mux", NULL, "CH2_DIG"}, + {"MIC2P Input Mux", NULL, "CH3_DIG"}, + {"MIC2M Input Mux", NULL, "CH4_DIG"}, + {"MIC3P Input Mux", NULL, "CH5_DIG"}, + {"MIC3M Input Mux", NULL, "CH6_DIG"}, + {"MIC4P Input Mux", NULL, "CH7_DIG"}, + {"MIC4M Input Mux", NULL, "CH8_DIG"}, + {"MIC1 Analog Mux", "Line In", "MIC1P"}, {"MIC2 Analog Mux", "Line In", "MIC2P"}, {"MIC3 Analog Mux", "Line In", "MIC3P"}, @@ -554,6 +582,15 @@ static const struct snd_soc_dapm_route adcx140_audio_map[] = { {"MIC3M Input Mux", "Analog", "MIC3M"}, {"MIC4P Input Mux", "Analog", "MIC4P"}, {"MIC4M Input Mux", "Analog", "MIC4M"}, + + {"MIC1P Input Mux", "Digital", "MIC1P"}, + {"MIC1M Input Mux", "Digital", "MIC1M"}, + {"MIC2P Input Mux", "Digital", "MIC2P"}, + {"MIC2M Input Mux", "Digital", "MIC2M"}, + {"MIC3P Input Mux", "Digital", "MIC3P"}, + {"MIC3M Input Mux", "Digital", "MIC3M"}, + {"MIC4P Input Mux", "Digital", "MIC4P"}, + {"MIC4M Input Mux", "Digital", "MIC4M"}, }; static const struct snd_kcontrol_new adcx140_snd_controls[] = { -- 2.28.0

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[PATCH] ASoC: core: remove artificial component and DAI name constraint
by Dmitry Baryshkov 28 Aug '20

28 Aug '20

Current fmt_single_name code limits maximum name of a DAI or component to 32 bytes. On some systems corresponding device names might be longer than that (e.g. 17300000.remoteproc:glink-edge:apr:apr-service@8:routing). This will result in duplicate DAI/component names. Rewrite fmt_single_name() to remove such length limitations. Signed-off-by: Dmitry Baryshkov <dmitry.baryshkov(a)linaro.org> --- sound/soc/soc-core.c | 33 +++++++++++++++------------------ 1 file changed, 15 insertions(+), 18 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 2fe1b2ec7c8f..4f2e2270a0d3 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -44,8 +44,6 @@ #define CREATE_TRACE_POINTS #include <trace/events/asoc.h> -#define NAME_SIZE 32 - static DEFINE_MUTEX(client_mutex); static LIST_HEAD(component_list); static LIST_HEAD(unbind_card_list); @@ -2218,13 +2216,14 @@ EXPORT_SYMBOL_GPL(snd_soc_unregister_card); */ static char *fmt_single_name(struct device *dev, int *id) { - char *found, name[NAME_SIZE]; + const char *devname = dev_name(dev); + char *found, *name; int id1, id2; - if (dev_name(dev) == NULL) + if (devname == NULL) return NULL; - strlcpy(name, dev_name(dev), NAME_SIZE); + name = devm_kstrdup(dev, devname, GFP_KERNEL); /* are we a "%s.%d" name (platform and SPI components) */ found = strstr(name, dev->driver->name); @@ -2237,23 +2236,21 @@ static char *fmt_single_name(struct device *dev, int *id) found[strlen(dev->driver->name)] = '\0'; } - } else { - /* I2C component devices are named "bus-addr" */ - if (sscanf(name, "%x-%x", &id1, &id2) == 2) { - char tmp[NAME_SIZE]; + /* I2C component devices are named "bus-addr" */ + } else if (sscanf(name, "%x-%x", &id1, &id2) == 2) { - /* create unique ID number from I2C addr and bus */ - *id = ((id1 & 0xffff) << 16) + id2; + /* create unique ID number from I2C addr and bus */ + *id = ((id1 & 0xffff) << 16) + id2; - /* sanitize component name for DAI link creation */ - snprintf(tmp, NAME_SIZE, "%s.%s", dev->driver->name, - name); - strlcpy(name, tmp, NAME_SIZE); - } else - *id = 0; + devm_kfree(dev, name); + + /* sanitize component name for DAI link creation */ + name = devm_kasprintf(dev, GFP_KERNEL, "%s.%s", dev->driver->name, devname); + } else { + *id = 0; } - return devm_kstrdup(dev, name, GFP_KERNEL); + return name; } /* -- 2.28.0

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[PATCH 0/4] regmap: add SoundWire 1.2 MBQ support
by Pierre-Louis Bossart 28 Aug '20

28 Aug '20

In preparation of the upstream contribution of SDCA (SoundWire Device Class for Audio) ASoC codec drivers [1] [2], add regmap support SoundWire 1.2 MBQ support. The MBQ (Multi-Byte Quantity) registers need to be handled in a different way from regular 8-bit SoundWire registers, their main application is going to be for volume/gain controls. The 3rd patch was initially suggested for inclusion in the SoundWire tree, and was modified to add more background information in the commit message as requested by Vinod Koul. Pierre-Louis Bossart (4): regmap: sdw: move to -EOPNOTSUPP regmap: sdw: add required header files soundwire: SDCA: add helper macro to access controls regmap: sdw: add support for SoundWire 1.2 MBQ drivers/base/regmap/Kconfig | 6 +- drivers/base/regmap/Makefile | 1 + drivers/base/regmap/regmap-sdw-mbq.c | 102 ++++++++++++++++++++++++ drivers/base/regmap/regmap-sdw.c | 8 +- include/linux/regmap.h | 20 +++++ include/linux/soundwire/sdw_registers.h | 13 +++ 6 files changed, 146 insertions(+), 4 deletions(-) create mode 100644 drivers/base/regmap/regmap-sdw-mbq.c base-commit: 9123e3a74ec7b934a4a099e98af6a61c2f80bbf5 -- 2.25.1

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[PATCH v1] ASoC: core: Replace dma_request_slave_channel() with dma_request_chan()
by Andy Shevchenko 28 Aug '20

28 Aug '20

Drivers should use dma_request_chan() instead of dma_request_slave_channel(). dma_request_slave_channel() is a simple wrapper for dma_request_chan() eating up the error code for channel request failure and makes deferred probing impossible. Signed-off-by: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> --- sound/soc/soc-generic-dmaengine-pcm.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index fb95c1464e66..ab13e08cd636 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -249,8 +249,7 @@ static int dmaengine_pcm_new(struct snd_soc_component *component, continue; if (!pcm->chan[i] && config && config->chan_names[i]) - pcm->chan[i] = dma_request_slave_channel(dev, - config->chan_names[i]); + pcm->chan[i] = dma_request_chan(dev, config->chan_names[i]); if (!pcm->chan[i] && (pcm->flags & SND_DMAENGINE_PCM_FLAG_COMPAT)) { pcm->chan[i] = dmaengine_pcm_compat_request_channel( -- 2.28.0

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[PATCH 0/7] Allwinner A64 digital audio codec fixes
by Samuel Holland 28 Aug '20

28 Aug '20

This series fixes a couple of issues with the digital audio codec in the Allwinner A64 SoC: 1) Left/right channels were swapped when playing/recording audio 2) DAPM topology was wrong, breaking some kcontrols This is the minimum set of changes necessary to fix these issues in a backward-compatible way. For that reason, some DAPM widgets still have incorrect or confusing names; those and other issues will be fixed in later patch sets. Samuel Holland (7): ASoC: dt-bindings: Add a new compatible for the A64 codec ASoC: sun8i-codec: Fix DAPM to match the hardware topology ASoC: sun8i-codec: Add missing mixer routes ASoC: sun8i-codec: Add a quirk for LRCK inversion ARM: dts: sun8i: a33: Update codec widget names arm64: dts: allwinner: a64: Update codec widget names arm64: dts: allwinner: a64: Update the audio codec compatible .../sound/allwinner,sun8i-a33-codec.yaml | 6 +- arch/arm/boot/dts/sun8i-a33-olinuxino.dts | 4 +- arch/arm/boot/dts/sun8i-a33.dtsi | 4 +- .../dts/allwinner/sun50i-a64-bananapi-m64.dts | 8 +- .../dts/allwinner/sun50i-a64-orangepi-win.dts | 8 +- .../boot/dts/allwinner/sun50i-a64-pine64.dts | 8 +- .../dts/allwinner/sun50i-a64-pinebook.dts | 8 +- .../dts/allwinner/sun50i-a64-pinephone.dtsi | 8 +- .../boot/dts/allwinner/sun50i-a64-pinetab.dts | 8 +- .../allwinner/sun50i-a64-sopine-baseboard.dts | 8 +- .../boot/dts/allwinner/sun50i-a64-teres-i.dts | 8 +- arch/arm64/boot/dts/allwinner/sun50i-a64.dtsi | 11 +- sound/soc/sunxi/sun8i-codec.c | 137 ++++++++++++++---- 13 files changed, 155 insertions(+), 71 deletions(-) -- 2.26.2

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[PATCH v2 0/3] soundwire: fix port_ready[] dynamic allocation
by Bard Liao 28 Aug '20

28 Aug '20

The existing code allocates memory for the total number of ports. This only works if the ports are contiguous, but will break if e.g. a Devices uses port0, 1, and 14. The port_ready[] array would contain 3 elements, which would lead to an out-of-bounds access. Conversely in other cases, the wrong port index would be used leading to timeouts on prepare. This can be fixed by allocating for the worst-case of 15 ports (DP0..DP14). In addition since the number is now fixed, we can use an array instead of a dynamic allocation. Changes in v2: - Split patches into sdw and asoc patches. Please note that "soundwire: fix port_ready[] dynamic allocation in mipi_disco" and "ASoC: codecs: fix port_ready[] dynamic allocation in ASoC codecs" should be merged at the same time. Pierre-Louis Bossart (3): soundwire: add definition for maximum number of ports soundwire: fix port_ready[] dynamic allocation in mipi_disco ASoC: codecs: fix port_ready[] dynamic allocation in ASoC codecs drivers/soundwire/mipi_disco.c | 18 +----------------- drivers/soundwire/slave.c | 4 ++++ include/linux/soundwire/sdw.h | 5 +++-- sound/soc/codecs/max98373-sdw.c | 15 +-------------- sound/soc/codecs/rt1308-sdw.c | 14 +------------- sound/soc/codecs/rt5682-sdw.c | 15 +-------------- sound/soc/codecs/rt700-sdw.c | 15 +-------------- sound/soc/codecs/rt711-sdw.c | 15 +-------------- sound/soc/codecs/rt715-sdw.c | 33 +-------------------------------- 9 files changed, 14 insertions(+), 120 deletions(-) -- 2.17.1

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[PATCH] ASoC: soc-core: add snd_soc_find_dai_with_mutex()
by Kuninori Morimoto 28 Aug '20

28 Aug '20

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> commit 25612477d20b52 ("ASoC: soc-dai: set dai_link dpcm_ flags with a helper") added snd_soc_dai_link_set_capabilities(). But it is using snd_soc_find_dai() (A) which is required client_mutex (B). And client_mutex is soc-core.c local. struct snd_soc_dai *snd_soc_find_dai(xxx) { ... (B) lockdep_assert_held(&client_mutex); ... } void snd_soc_dai_link_set_capabilities(xxx) { ... for_each_pcm_streams(direction) { ... for_each_link_cpus(dai_link, i, cpu) { (A) dai = snd_soc_find_dai(cpu); ... } ... for_each_link_codecs(dai_link, i, codec) { (A) dai = snd_soc_find_dai(codec); ... } } ... } Because of these background, we will get WARNING if .config has CONFIG_LOCKDEP. WARNING: CPU: 2 PID: 53 at sound/soc/soc-core.c:814 snd_soc_find_dai+0xf8/0x100 CPU: 2 PID: 53 Comm: kworker/2:1 Not tainted 5.7.0-rc1+ #328 Hardware name: Renesas H3ULCB Kingfisher board based on r8a77951 (DT) Workqueue: events deferred_probe_work_func pstate: 60000005 (nZCv daif -PAN -UAO) pc : snd_soc_find_dai+0xf8/0x100 lr : snd_soc_find_dai+0xf4/0x100 ... Call trace: snd_soc_find_dai+0xf8/0x100 snd_soc_dai_link_set_capabilities+0xa0/0x16c graph_dai_link_of_dpcm+0x390/0x3c0 graph_for_each_link+0x134/0x200 graph_probe+0x144/0x230 platform_drv_probe+0x5c/0xb0 really_probe+0xe4/0x430 driver_probe_device+0x60/0xf4 snd_soc_find_dai() will be used from (X) CPU/Codec/Platform driver with mutex lock, and (Y) Card driver without mutex lock. This snd_soc_dai_link_set_capabilities() is for Card driver, this means called without mutex. This patch adds snd_soc_find_dai_with_mutex() to solve it. Fixes: 25612477d20b52 ("ASoC: soc-dai: set dai_link dpcm_ flags with a helper") Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- This patch was originally "ASoC: soc-core: move snd_soc_dai_link_set_capabilities() to soc-core.c". This is for asoc/for-5.9 branch include/sound/soc.h | 2 ++ sound/soc/soc-core.c | 13 +++++++++++++ sound/soc/soc-dai.c | 4 ++-- 3 files changed, 17 insertions(+), 2 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 5e3919ffb00c..4176071f61bf 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1361,6 +1361,8 @@ void snd_soc_unregister_dai(struct snd_soc_dai *dai); struct snd_soc_dai *snd_soc_find_dai( const struct snd_soc_dai_link_component *dlc); +struct snd_soc_dai *snd_soc_find_dai_with_mutex( + const struct snd_soc_dai_link_component *dlc); #include <sound/soc-dai.h> diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 2fe1b2ec7c8f..fe11856d7a63 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -834,6 +834,19 @@ struct snd_soc_dai *snd_soc_find_dai( } EXPORT_SYMBOL_GPL(snd_soc_find_dai); +struct snd_soc_dai *snd_soc_find_dai_with_mutex( + const struct snd_soc_dai_link_component *dlc) +{ + struct snd_soc_dai *dai; + + mutex_lock(&client_mutex); + dai = snd_soc_find_dai(dlc); + mutex_unlock(&client_mutex); + + return dai; +} +EXPORT_SYMBOL_GPL(snd_soc_find_dai_with_mutex); + static int soc_dai_link_sanity_check(struct snd_soc_card *card, struct snd_soc_dai_link *link) { diff --git a/sound/soc/soc-dai.c b/sound/soc/soc-dai.c index 91a2551e4cef..0dbd312aad08 100644 --- a/sound/soc/soc-dai.c +++ b/sound/soc/soc-dai.c @@ -412,14 +412,14 @@ void snd_soc_dai_link_set_capabilities(struct snd_soc_dai_link *dai_link) supported_codec = false; for_each_link_cpus(dai_link, i, cpu) { - dai = snd_soc_find_dai(cpu); + dai = snd_soc_find_dai_with_mutex(cpu); if (dai && snd_soc_dai_stream_valid(dai, direction)) { supported_cpu = true; break; } } for_each_link_codecs(dai_link, i, codec) { - dai = snd_soc_find_dai(codec); + dai = snd_soc_find_dai_with_mutex(codec); if (dai && snd_soc_dai_stream_valid(dai, direction)) { supported_codec = true; break; -- 2.25.1

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[PATCH 1/3] mfd: madera: Simplify with dev_err_probe()
by Krzysztof Kozlowski 28 Aug '20

28 Aug '20

Common pattern of handling deferred probe can be simplified with dev_err_probe(). Less code and also it prints the error value. Signed-off-by: Krzysztof Kozlowski <krzk(a)kernel.org> --- drivers/mfd/madera-core.c | 11 +++-------- 1 file changed, 3 insertions(+), 8 deletions(-) diff --git a/drivers/mfd/madera-core.c b/drivers/mfd/madera-core.c index 8a8d733fdce5..4ed6ad8ce002 100644 --- a/drivers/mfd/madera-core.c +++ b/drivers/mfd/madera-core.c @@ -369,19 +369,14 @@ EXPORT_SYMBOL_GPL(madera_of_match); static int madera_get_reset_gpio(struct madera *madera) { struct gpio_desc *reset; - int ret; if (madera->pdata.reset) return 0; reset = devm_gpiod_get_optional(madera->dev, "reset", GPIOD_OUT_LOW); - if (IS_ERR(reset)) { - ret = PTR_ERR(reset); - if (ret != -EPROBE_DEFER) - dev_err(madera->dev, "Failed to request /RESET: %d\n", - ret); - return ret; - } + if (IS_ERR(reset)) + return dev_err_probe(madera->dev, PTR_ERR(reset), + "Failed to request /RESET"); /* * A hard reset is needed for full reset of the chip. We allow running -- 2.17.1

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[PATCH] ALSA: Realtek: Early Forbid of runtime PM
by Harsha Priya 28 Aug '20

28 Aug '20

For Realtek codecs, pm_runtime_forbid() is called in the probe function after the build_controls(). In a stress test, occasionally the runtime PM calls are invoked before controls are built. This causes the codec to be runtime suspended before probe completes. Because of this, not all controls are enumerated correctly and audio does not work until system is rebooted. This patch calls pm_runtime_forbid() early to fix the issue. Multiple stress tests of 2000+ cycles has been done to test the fix. Signed-off-by: Harsha Priya <harshapriya.n(a)intel.com> Signed-off-by: Emmanuel Jillela <emmanuel.jillela(a)intel.com> Reviewed-by: Kailang Yang <kailang(a)realtek.com> --- sound/pci/hda/patch_realtek.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 98789691a479..30ea07fd6735 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8418,6 +8418,7 @@ static int patch_alc269(struct hda_codec *codec) { struct alc_spec *spec; int err; + struct device *dev = hda_codec_dev(codec); err = alc_alloc_spec(codec, 0x0b); if (err < 0) @@ -8430,6 +8431,8 @@ static int patch_alc269(struct hda_codec *codec) #ifdef CONFIG_PM codec->patch_ops.suspend = alc269_suspend; codec->patch_ops.resume = alc269_resume; + pm_runtime_dont_use_autosuspend(dev); + pm_runtime_forbid(dev); #endif spec->shutup = alc_default_shutup; spec->init_hook = alc_default_init; -- 2.17.1

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