From bc094709de0192a756c6946a7c89c543243ae609 Mon Sep 17 00:00:00 2001 From: Shuming Fan Date: Mon, 25 Nov 2019 17:19:40 +0800 Subject: ASoC: rt5682: fix i2c arbitration lost issue This patch modified the HW initial setting to fix i2c arbitration lost issue. Signed-off-by: Shuming Fan Link: https://lore.kernel.org/r/20191125091940.11953-1-shumingf@realtek.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt5682.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index 35dcec135c8a..9feba9a24501 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -73,6 +73,7 @@ struct rt5682_priv { static const struct reg_sequence patch_list[] = { {RT5682_HP_IMP_SENS_CTRL_19, 0x1000}, {RT5682_DAC_ADC_DIG_VOL1, 0xa020}, + {RT5682_I2C_CTRL, 0x000f}, }; static const struct reg_default rt5682_reg[] = { @@ -2496,6 +2497,7 @@ static void rt5682_calibrate(struct rt5682_priv *rt5682) mutex_lock(&rt5682->calibrate_mutex); rt5682_reset(rt5682->regmap); + regmap_write(rt5682->regmap, RT5682_I2C_CTRL, 0x000f); regmap_write(rt5682->regmap, RT5682_PWR_ANLG_1, 0xa2af); usleep_range(15000, 20000); regmap_write(rt5682->regmap, RT5682_PWR_ANLG_1, 0xf2af); -- cgit v1.2.1 From 756ae8f237e19a014a1c20ad5a51b0686463d1f7 Mon Sep 17 00:00:00 2001 From: Yu-Hsuan Hsu Date: Tue, 26 Nov 2019 15:54:24 +0800 Subject: ASoC: AMD: Enable clk in startup intead of hw_params Some usages only call startup and shutdown without setting hw_params (e.g. arecord --dump-hw-params). If we don't enable clk in startup, it will cause ref count error because the clk will be disabled in shutdown. For this reason, we should move enabling clk from hw_params to startup. In addition, the hw_params is fixed in this driver(48000 rate, 2 channels, S16_LE format) so we don't need to change the clk rate after the hw_params is set. Signed-off-by: Yu-Hsuan Hsu Acked-by: Akshu Agrawal Link: https://lore.kernel.org/r/20191126075424.80668-1-yuhsuan@chromium.org Signed-off-by: Mark Brown --- sound/soc/amd/acp-da7219-max98357a.c | 46 +++++++++++------------------------- 1 file changed, 14 insertions(+), 32 deletions(-) diff --git a/sound/soc/amd/acp-da7219-max98357a.c b/sound/soc/amd/acp-da7219-max98357a.c index f4ee6798154a..7a5621e5e233 100644 --- a/sound/soc/amd/acp-da7219-max98357a.c +++ b/sound/soc/amd/acp-da7219-max98357a.c @@ -96,14 +96,19 @@ static int cz_da7219_init(struct snd_soc_pcm_runtime *rtd) return 0; } -static int da7219_clk_enable(struct snd_pcm_substream *substream, - int wclk_rate, int bclk_rate) +static int da7219_clk_enable(struct snd_pcm_substream *substream) { int ret = 0; struct snd_soc_pcm_runtime *rtd = substream->private_data; - clk_set_rate(da7219_dai_wclk, wclk_rate); - clk_set_rate(da7219_dai_bclk, bclk_rate); + /* + * Set wclk to 48000 because the rate constraint of this driver is + * 48000. ADAU7002 spec: "The ADAU7002 requires a BCLK rate that is + * minimum of 64x the LRCLK sample rate." DA7219 is the only clk + * source so for all codecs we have to limit bclk to 64X lrclk. + */ + clk_set_rate(da7219_dai_wclk, 48000); + clk_set_rate(da7219_dai_bclk, 48000 * 64); ret = clk_prepare_enable(da7219_dai_bclk); if (ret < 0) { dev_err(rtd->dev, "can't enable master clock %d\n", ret); @@ -156,7 +161,7 @@ static int cz_da7219_play_startup(struct snd_pcm_substream *substream) &constraints_rates); machine->play_i2s_instance = I2S_SP_INSTANCE; - return 0; + return da7219_clk_enable(substream); } static int cz_da7219_cap_startup(struct snd_pcm_substream *substream) @@ -178,7 +183,7 @@ static int cz_da7219_cap_startup(struct snd_pcm_substream *substream) machine->cap_i2s_instance = I2S_SP_INSTANCE; machine->capture_channel = CAP_CHANNEL1; - return 0; + return da7219_clk_enable(substream); } static int cz_max_startup(struct snd_pcm_substream *substream) @@ -199,7 +204,7 @@ static int cz_max_startup(struct snd_pcm_substream *substream) &constraints_rates); machine->play_i2s_instance = I2S_BT_INSTANCE; - return 0; + return da7219_clk_enable(substream); } static int cz_dmic0_startup(struct snd_pcm_substream *substream) @@ -220,7 +225,7 @@ static int cz_dmic0_startup(struct snd_pcm_substream *substream) &constraints_rates); machine->cap_i2s_instance = I2S_BT_INSTANCE; - return 0; + return da7219_clk_enable(substream); } static int cz_dmic1_startup(struct snd_pcm_substream *substream) @@ -242,25 +247,7 @@ static int cz_dmic1_startup(struct snd_pcm_substream *substream) machine->cap_i2s_instance = I2S_SP_INSTANCE; machine->capture_channel = CAP_CHANNEL0; - return 0; -} - -static int cz_da7219_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - int wclk, bclk; - - wclk = params_rate(params); - bclk = wclk * params_channels(params) * - snd_pcm_format_width(params_format(params)); - /* ADAU7002 spec: "The ADAU7002 requires a BCLK rate - * that is minimum of 64x the LRCLK sample rate." - * DA7219 is the only clk source so for all codecs - * we have to limit bclk to 64X lrclk. - */ - if (bclk < (wclk * 64)) - bclk = wclk * 64; - return da7219_clk_enable(substream, wclk, bclk); + return da7219_clk_enable(substream); } static void cz_da7219_shutdown(struct snd_pcm_substream *substream) @@ -271,31 +258,26 @@ static void cz_da7219_shutdown(struct snd_pcm_substream *substream) static const struct snd_soc_ops cz_da7219_play_ops = { .startup = cz_da7219_play_startup, .shutdown = cz_da7219_shutdown, - .hw_params = cz_da7219_params, }; static const struct snd_soc_ops cz_da7219_cap_ops = { .startup = cz_da7219_cap_startup, .shutdown = cz_da7219_shutdown, - .hw_params = cz_da7219_params, }; static const struct snd_soc_ops cz_max_play_ops = { .startup = cz_max_startup, .shutdown = cz_da7219_shutdown, - .hw_params = cz_da7219_params, }; static const struct snd_soc_ops cz_dmic0_cap_ops = { .startup = cz_dmic0_startup, .shutdown = cz_da7219_shutdown, - .hw_params = cz_da7219_params, }; static const struct snd_soc_ops cz_dmic1_cap_ops = { .startup = cz_dmic1_startup, .shutdown = cz_da7219_shutdown, - .hw_params = cz_da7219_params, }; SND_SOC_DAILINK_DEF(designware1, -- cgit v1.2.1 From b81eb73be03ac736f1f8d27d64a372c62c7159e5 Mon Sep 17 00:00:00 2001 From: Keyon Jie Date: Tue, 26 Nov 2019 08:15:33 -0600 Subject: ASoC: SOF: Intel: BYT: fix a copy/paste mistake in byt_dump() The shim registers in BYT/CHT/BSW are 64bits based, correct the copy/paste (from bdw.c where the shim registers are 32bits based) error in byt_dump(). Fixes: 3a9e204d4e36 ("ASoC: SOF: Intel: Add context data to any IPC timeout") Signed-off-by: Keyon Jie Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20191126141533.21601-1-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/byt.c | 18 +++++++++--------- 1 file changed, 9 insertions(+), 9 deletions(-) diff --git a/sound/soc/sof/intel/byt.c b/sound/soc/sof/intel/byt.c index 2abf80b3eb52..b9061b79a57b 100644 --- a/sound/soc/sof/intel/byt.c +++ b/sound/soc/sof/intel/byt.c @@ -145,33 +145,33 @@ static void byt_dump(struct snd_sof_dev *sdev, u32 flags) struct sof_ipc_dsp_oops_xtensa xoops; struct sof_ipc_panic_info panic_info; u32 stack[BYT_STACK_DUMP_SIZE]; - u32 status, panic, imrd, imrx; + u64 status, panic, imrd, imrx; /* now try generic SOF status messages */ - status = snd_sof_dsp_read(sdev, BYT_DSP_BAR, SHIM_IPCD); - panic = snd_sof_dsp_read(sdev, BYT_DSP_BAR, SHIM_IPCX); + status = snd_sof_dsp_read64(sdev, BYT_DSP_BAR, SHIM_IPCD); + panic = snd_sof_dsp_read64(sdev, BYT_DSP_BAR, SHIM_IPCX); byt_get_registers(sdev, &xoops, &panic_info, stack, BYT_STACK_DUMP_SIZE); snd_sof_get_status(sdev, status, panic, &xoops, &panic_info, stack, BYT_STACK_DUMP_SIZE); /* provide some context for firmware debug */ - imrx = snd_sof_dsp_read(sdev, BYT_DSP_BAR, SHIM_IMRX); - imrd = snd_sof_dsp_read(sdev, BYT_DSP_BAR, SHIM_IMRD); + imrx = snd_sof_dsp_read64(sdev, BYT_DSP_BAR, SHIM_IMRX); + imrd = snd_sof_dsp_read64(sdev, BYT_DSP_BAR, SHIM_IMRD); dev_err(sdev->dev, - "error: ipc host -> DSP: pending %s complete %s raw 0x%8.8x\n", + "error: ipc host -> DSP: pending %s complete %s raw 0x%llx\n", (panic & SHIM_IPCX_BUSY) ? "yes" : "no", (panic & SHIM_IPCX_DONE) ? "yes" : "no", panic); dev_err(sdev->dev, - "error: mask host: pending %s complete %s raw 0x%8.8x\n", + "error: mask host: pending %s complete %s raw 0x%llx\n", (imrx & SHIM_IMRX_BUSY) ? "yes" : "no", (imrx & SHIM_IMRX_DONE) ? "yes" : "no", imrx); dev_err(sdev->dev, - "error: ipc DSP -> host: pending %s complete %s raw 0x%8.8x\n", + "error: ipc DSP -> host: pending %s complete %s raw 0x%llx\n", (status & SHIM_IPCD_BUSY) ? "yes" : "no", (status & SHIM_IPCD_DONE) ? "yes" : "no", status); dev_err(sdev->dev, - "error: mask DSP: pending %s complete %s raw 0x%8.8x\n", + "error: mask DSP: pending %s complete %s raw 0x%llx\n", (imrd & SHIM_IMRD_BUSY) ? "yes" : "no", (imrd & SHIM_IMRD_DONE) ? "yes" : "no", imrd); -- cgit v1.2.1 From 469b3ad672e27b28c5865c804426f65e69c5e41a Mon Sep 17 00:00:00 2001 From: Daniel Baluta Date: Tue, 26 Nov 2019 08:16:06 -0600 Subject: ASoC: SOF: topology: Fix unload for SAI/ESAI Link unload now fails for ESAI/SAI DAIs with: "error: invalid DAI type 6" because DAI type is not properly handled. Fix this by correctly handling cases where type is ESAI or SAI. Fixes: a4eff5f86c9c5e7 ("ASoC: SOF: imx: Read ESAI parameters and send them to DSP") Signed-off-by: Daniel Baluta Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20191126141606.21650-1-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/topology.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/sof/topology.c b/sound/soc/sof/topology.c index d82ab981e840..e20b806ec80f 100644 --- a/sound/soc/sof/topology.c +++ b/sound/soc/sof/topology.c @@ -3132,7 +3132,9 @@ found: case SOF_DAI_INTEL_SSP: case SOF_DAI_INTEL_DMIC: case SOF_DAI_INTEL_ALH: - /* no resource needs to be released for SSP, DMIC and ALH */ + case SOF_DAI_IMX_SAI: + case SOF_DAI_IMX_ESAI: + /* no resource needs to be released for all cases above */ break; case SOF_DAI_INTEL_HDA: ret = sof_link_hda_unload(sdev, link); -- cgit v1.2.1 From fb3194413d1ef79732931a40f54da21a16505a76 Mon Sep 17 00:00:00 2001 From: YueHaibing Date: Wed, 27 Nov 2019 16:21:45 +0800 Subject: ASoC: rt5677: Fix build error without CONFIG_SPI If CONFIG_SPI is n, SND_SOC_RT5677_SPI also is n, building fails: sound/soc/codecs/rt5677.o: In function `rt5677_irq': rt5677.c:(.text+0x2dbf): undefined reference to `rt5677_spi_hotword_detected' sound/soc/codecs/rt5677.o: In function `rt5677_dsp_work': rt5677.c:(.text+0x3709): undefined reference to `rt5677_spi_write' This adds stub helpers to fix this. Reported-by: Hulk Robot Fixes: 461c623270e4 ("ASoC: rt5677: Load firmware via SPI using delayed work") Signed-off-by: YueHaibing Link: https://lore.kernel.org/r/20191127082145.6100-1-yuehaibing@huawei.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt5677-spi.h | 16 ++++++++++++++++ 1 file changed, 16 insertions(+) diff --git a/sound/soc/codecs/rt5677-spi.h b/sound/soc/codecs/rt5677-spi.h index 3af36ec928e9..088b77931727 100644 --- a/sound/soc/codecs/rt5677-spi.h +++ b/sound/soc/codecs/rt5677-spi.h @@ -9,9 +9,25 @@ #ifndef __RT5677_SPI_H__ #define __RT5677_SPI_H__ +#if IS_ENABLED(CONFIG_SND_SOC_RT5677_SPI) int rt5677_spi_read(u32 addr, void *rxbuf, size_t len); int rt5677_spi_write(u32 addr, const void *txbuf, size_t len); int rt5677_spi_write_firmware(u32 addr, const struct firmware *fw); void rt5677_spi_hotword_detected(void); +#else +static inline int rt5677_spi_read(u32 addr, void *rxbuf, size_t len) +{ + return -EINVAL; +} +static inline int rt5677_spi_write(u32 addr, const void *txbuf, size_t len) +{ + return -EINVAL; +} +static inline int rt5677_spi_write_firmware(u32 addr, const struct firmware *fw) +{ + return -EINVAL; +} +static inline void rt5677_spi_hotword_detected(void){} +#endif #endif /* __RT5677_SPI_H__ */ -- cgit v1.2.1 From 2a0bda276c64212e517cc1d65cf65719a9ab1ef6 Mon Sep 17 00:00:00 2001 From: Michael Walle Date: Sat, 23 Nov 2019 00:25:32 +0100 Subject: ASoC: wm8904: fix automatic sysclk configuration The simple-card tries to signal the codec to disable rate constraints, see commit 2458adb8f92a ("SoC: simple-card-utils: set 0Hz to sysclk when shutdown"). This wasn't handled by the codec, instead it would set the FLL frequency to 0Hz which isn't working. Since we don't have any rate constraints just ignore this request. Fixes: 13409d27cb39 ("ASoC: wm8904: configure sysclk/FLL automatically") Signed-off-by: Michael Walle Acked-by: Charles Keepax Link: https://lore.kernel.org/r/20191122232532.22258-1-michael@walle.cc Signed-off-by: Mark Brown --- sound/soc/codecs/wm8904.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index 2a7d23a5daa8..d191d81850ee 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -1806,6 +1806,12 @@ static int wm8904_set_sysclk(struct snd_soc_dai *dai, int clk_id, switch (clk_id) { case WM8904_CLK_AUTO: + /* We don't have any rate constraints, so just ignore the + * request to disable constraining. + */ + if (!freq) + return 0; + mclk_freq = clk_get_rate(priv->mclk); /* enable FLL if a different sysclk is desired */ if (mclk_freq != freq) { -- cgit v1.2.1 From acb874a7c049ec49d8fc66c893170fb42c01bdf7 Mon Sep 17 00:00:00 2001 From: Tzung-Bi Shih Date: Fri, 22 Nov 2019 15:31:12 +0800 Subject: ASoC: max98090: remove msleep in PLL unlocked workaround It was observed Baytrail-based chromebooks could cause continuous PLL unlocked when using playback stream and capture stream simultaneously. Specifically, starting a capture stream after started a playback stream. As a result, the audio data could corrupt or turn completely silent. As the datasheet suggested, the maximum PLL lock time should be 7 msec. The workaround resets the codec softly by toggling SHDN off and on if PLL failed to lock for 10 msec. Notably, there is no suggested hold time for SHDN off. On Baytrail-based chromebooks, it would easily happen continuous PLL unlocked if there is a 10 msec delay between SHDN off and on. Removes the msleep(). Signed-off-by: Tzung-Bi Shih Link: https://lore.kernel.org/r/20191122073114.219945-2-tzungbi@google.com Reviewed-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/codecs/max98090.c | 8 +++++++- 1 file changed, 7 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index f6bf4cfbea23..12cb87c0d463 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -2114,10 +2114,16 @@ static void max98090_pll_work(struct work_struct *work) dev_info_ratelimited(component->dev, "PLL unlocked\n"); + /* + * As the datasheet suggested, the maximum PLL lock time should be + * 7 msec. The workaround resets the codec softly by toggling SHDN + * off and on if PLL failed to lock for 10 msec. Notably, there is + * no suggested hold time for SHDN off. + */ + /* Toggle shutdown OFF then ON */ snd_soc_component_update_bits(component, M98090_REG_DEVICE_SHUTDOWN, M98090_SHDNN_MASK, 0); - msleep(10); snd_soc_component_update_bits(component, M98090_REG_DEVICE_SHUTDOWN, M98090_SHDNN_MASK, M98090_SHDNN_MASK); -- cgit v1.2.1 From 6f49919d11690a9b5614445ba30fde18083fdd63 Mon Sep 17 00:00:00 2001 From: Tzung-Bi Shih Date: Fri, 22 Nov 2019 15:31:13 +0800 Subject: ASoC: max98090: exit workaround earlier if PLL is locked According to the datasheet, PLL lock time typically takes 2 msec and at most takes 7 msec. Check the lock status every 1 msec and exit the workaround if PLL is locked. Signed-off-by: Tzung-Bi Shih Link: https://lore.kernel.org/r/20191122073114.219945-3-tzungbi@google.com Reviewed-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/codecs/max98090.c | 14 ++++++++++++-- 1 file changed, 12 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index 12cb87c0d463..f531e5a11bdd 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -2108,6 +2108,8 @@ static void max98090_pll_work(struct work_struct *work) struct max98090_priv *max98090 = container_of(work, struct max98090_priv, pll_work); struct snd_soc_component *component = max98090->component; + unsigned int pll; + int i; if (!snd_soc_component_is_active(component)) return; @@ -2127,8 +2129,16 @@ static void max98090_pll_work(struct work_struct *work) snd_soc_component_update_bits(component, M98090_REG_DEVICE_SHUTDOWN, M98090_SHDNN_MASK, M98090_SHDNN_MASK); - /* Give PLL time to lock */ - msleep(10); + for (i = 0; i < 10; ++i) { + /* Give PLL time to lock */ + usleep_range(1000, 1200); + + /* Check lock status */ + pll = snd_soc_component_read32( + component, M98090_REG_DEVICE_STATUS); + if (!(pll & M98090_ULK_MASK)) + break; + } } static void max98090_jack_work(struct work_struct *work) -- cgit v1.2.1 From 45dfbf56975994822cce00b7475732a49f8aefed Mon Sep 17 00:00:00 2001 From: Tzung-Bi Shih Date: Fri, 22 Nov 2019 15:31:14 +0800 Subject: ASoC: max98090: fix possible race conditions max98090_interrupt() and max98090_pll_work() run in 2 different threads. There are 2 possible races: Note: M98090_REG_DEVICE_STATUS = 0x01. Note: ULK == 0, PLL is locked; ULK == 1, PLL is unlocked. max98090_interrupt max98090_pll_work ---------------------------------------------- schedule max98090_pll_work restart max98090 codec receive ULK INT assert ULK == 0 schedule max98090_pll_work (1). In the case (1), the PLL is locked but max98090_interrupt unnecessarily schedules another max98090_pll_work. max98090_interrupt max98090_pll_work max98090 codec ---------------------------------------------------------------------- ULK = 1 receive ULK INT read 0x01 ULK = 0 (clear on read) schedule max98090_pll_work restart max98090 codec ULK = 1 receive ULK INT read 0x01 ULK = 0 (clear on read) read 0x01 assert ULK == 0 (2). In the case (2), both max98090_interrupt and max98090_pll_work read the same clear-on-read register. max98090_pll_work would falsely thought PLL is locked. Note: the case (2) race is introduced by the previous commit ("ASoC: max98090: exit workaround earlier if PLL is locked") to check the status and exit the loop earlier in max98090_pll_work. There are 2 possible solution options: A. turn off ULK interrupt before scheduling max98090_pll_work; and turn on again before exiting max98090_pll_work. B. remove the second thread of execution. Option A cannot fix the case (2) race because it still has 2 threads access the same clear-on-read register simultaneously. Although we could suppose the register is volatile and read the status via I2C could be much slower than the hardware raises the bits. Option B introduces a maximum 10~12 msec penalty delay in the interrupt handler. However, it could only punish the jack detection by extra 10~12 msec. Adopts option B which is the better solution overall. Signed-off-by: Tzung-Bi Shih Link: https://lore.kernel.org/r/20191122073114.219945-4-tzungbi@google.com Reviewed-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/codecs/max98090.c | 8 ++------ sound/soc/codecs/max98090.h | 1 - 2 files changed, 2 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index f531e5a11bdd..e46b6ada13b1 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -2103,10 +2103,8 @@ static void max98090_pll_det_disable_work(struct work_struct *work) M98090_IULK_MASK, 0); } -static void max98090_pll_work(struct work_struct *work) +static void max98090_pll_work(struct max98090_priv *max98090) { - struct max98090_priv *max98090 = - container_of(work, struct max98090_priv, pll_work); struct snd_soc_component *component = max98090->component; unsigned int pll; int i; @@ -2275,7 +2273,7 @@ static irqreturn_t max98090_interrupt(int irq, void *data) if (active & M98090_ULK_MASK) { dev_dbg(component->dev, "M98090_ULK_MASK\n"); - schedule_work(&max98090->pll_work); + max98090_pll_work(max98090); } if (active & M98090_JDET_MASK) { @@ -2438,7 +2436,6 @@ static int max98090_probe(struct snd_soc_component *component) max98090_pll_det_enable_work); INIT_WORK(&max98090->pll_det_disable_work, max98090_pll_det_disable_work); - INIT_WORK(&max98090->pll_work, max98090_pll_work); /* Enable jack detection */ snd_soc_component_write(component, M98090_REG_JACK_DETECT, @@ -2491,7 +2488,6 @@ static void max98090_remove(struct snd_soc_component *component) cancel_delayed_work_sync(&max98090->jack_work); cancel_delayed_work_sync(&max98090->pll_det_enable_work); cancel_work_sync(&max98090->pll_det_disable_work); - cancel_work_sync(&max98090->pll_work); max98090->component = NULL; } diff --git a/sound/soc/codecs/max98090.h b/sound/soc/codecs/max98090.h index 57965cd678b4..a197114b0dad 100644 --- a/sound/soc/codecs/max98090.h +++ b/sound/soc/codecs/max98090.h @@ -1530,7 +1530,6 @@ struct max98090_priv { struct delayed_work jack_work; struct delayed_work pll_det_enable_work; struct work_struct pll_det_disable_work; - struct work_struct pll_work; struct snd_soc_jack *jack; unsigned int dai_fmt; int tdm_slots; -- cgit v1.2.1 From d5ee9108adacfbed140e0ac2371941ce7ca1fc54 Mon Sep 17 00:00:00 2001 From: YueHaibing Date: Thu, 28 Nov 2019 21:58:53 +0800 Subject: ASoC: Intel: sst: Add missing include Fix build error: sound/soc/intel/atom/sst/sst.c: In function intel_sst_interrupt_mrfld: sound/soc/intel/atom/sst/sst.c:93:5: error: implicit declaration of function memcpy_fromio; did you mean memcpy32_fromio? [-Werror=implicit-function-declaration] memcpy_fromio(msg->mailbox_data, ^~~~~~~~~~~~~ memcpy32_fromio Reported-by: Hulk Robot Signed-off-by: YueHaibing Acked-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20191128135853.8360-1-yuehaibing@huawei.com Signed-off-by: Mark Brown --- sound/soc/intel/atom/sst/sst.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/intel/atom/sst/sst.c b/sound/soc/intel/atom/sst/sst.c index fbecbb74350b..68bcec5241f7 100644 --- a/sound/soc/intel/atom/sst/sst.c +++ b/sound/soc/intel/atom/sst/sst.c @@ -14,6 +14,7 @@ #include #include #include +#include #include #include #include -- cgit v1.2.1 From 4bf2e385aa59c2fae5f880aa25cfd2b470109093 Mon Sep 17 00:00:00 2001 From: Curtis Malainey Date: Tue, 3 Dec 2019 09:30:07 -0800 Subject: ASoC: core: Init pcm runtime work early to avoid warnings There are cases where we fail before we reach soc_new_pcm which would init the workqueue. When we fail we attempt to flush the queue which generates warnings from the workqueue subsystem when we have not inited the queue. Solution is to use a proxy function to get around this issue. Signed-off-by: Curtis Malainey Reviewed-by: Takashi Iwai Link: https://lore.kernel.org/r/20191203173007.46504-1-cujomalainey@chromium.org Signed-off-by: Mark Brown --- include/sound/soc.h | 1 + sound/soc/soc-compress.c | 6 ++---- sound/soc/soc-core.c | 10 ++++++++++ sound/soc/soc-pcm.c | 11 ++++------- 4 files changed, 17 insertions(+), 11 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index c28a1ed5e8df..262896799826 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1150,6 +1150,7 @@ struct snd_soc_pcm_runtime { unsigned int num_codecs; struct delayed_work delayed_work; + void (*close_delayed_work_func)(struct snd_soc_pcm_runtime *rtd); #ifdef CONFIG_DEBUG_FS struct dentry *debugfs_dpcm_root; #endif diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index 61f230324164..6615ef64c7f5 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -214,10 +214,8 @@ be_err: * This is to ensure there are no pops or clicks in between any music tracks * due to DAPM power cycling. */ -static void close_delayed_work(struct work_struct *work) +static void close_delayed_work(struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_pcm_runtime *rtd = - container_of(work, struct snd_soc_pcm_runtime, delayed_work.work); struct snd_soc_dai *codec_dai = rtd->codec_dai; mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass); @@ -929,7 +927,7 @@ int snd_soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num) } /* DAPM dai link stream work */ - INIT_DELAYED_WORK(&rtd->delayed_work, close_delayed_work); + rtd->close_delayed_work_func = close_delayed_work; rtd->compr = compr; compr->private_data = rtd; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 062653ab03a3..0e2e628302f1 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -435,6 +435,15 @@ static void soc_free_pcm_runtime(struct snd_soc_pcm_runtime *rtd) device_unregister(rtd->dev); } +static void close_delayed_work(struct work_struct *work) { + struct snd_soc_pcm_runtime *rtd = + container_of(work, struct snd_soc_pcm_runtime, + delayed_work.work); + + if (rtd->close_delayed_work_func) + rtd->close_delayed_work_func(rtd); +} + static struct snd_soc_pcm_runtime *soc_new_pcm_runtime( struct snd_soc_card *card, struct snd_soc_dai_link *dai_link) { @@ -470,6 +479,7 @@ static struct snd_soc_pcm_runtime *soc_new_pcm_runtime( rtd->dev = dev; dev_set_drvdata(dev, rtd); + INIT_DELAYED_WORK(&rtd->delayed_work, close_delayed_work); /* * for rtd->codec_dais diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 01eb8700c3de..b78f6ff2b1d3 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -637,10 +637,8 @@ out: * This is to ensure there are no pops or clicks in between any music tracks * due to DAPM power cycling. */ -static void close_delayed_work(struct work_struct *work) +static void close_delayed_work(struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_pcm_runtime *rtd = - 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->card->pcm_mutex, rtd->card->pcm_subclass); @@ -660,7 +658,7 @@ static void close_delayed_work(struct work_struct *work) mutex_unlock(&rtd->card->pcm_mutex); } -static void codec2codec_close_delayed_work(struct work_struct *work) +static void codec2codec_close_delayed_work(struct snd_soc_pcm_runtime *rtd) { /* * Currently nothing to do for c2c links @@ -2974,10 +2972,9 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) /* DAPM dai link stream work */ if (rtd->dai_link->params) - INIT_DELAYED_WORK(&rtd->delayed_work, - codec2codec_close_delayed_work); + rtd->close_delayed_work_func = codec2codec_close_delayed_work; else - INIT_DELAYED_WORK(&rtd->delayed_work, close_delayed_work); + rtd->close_delayed_work_func = close_delayed_work; pcm->nonatomic = rtd->dai_link->nonatomic; rtd->pcm = pcm; -- cgit v1.2.1 From 7eccc05c7101f34cc36afe9405d15de6d4099fb4 Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Tue, 3 Dec 2019 23:14:42 +0100 Subject: ASoC: Intel: bytcr_rt5640: Update quirk for Teclast X89 When the Teclast X89 quirk was added we did not have jack-detection support yet. Note the over-current detection limit is set to 2mA instead of the usual 1.5mA because this tablet tends to give false-positive button-presses when it is set to 1.5mA. Signed-off-by: Hans de Goede Acked-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20191203221442.2657-1-hdegoede@redhat.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/bytcr_rt5640.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 9c1aa4ec9cba..cb511ea3b771 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -705,13 +705,17 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { BYT_RT5640_MCLK_EN), }, { + /* Teclast X89 */ .matches = { DMI_MATCH(DMI_BOARD_VENDOR, "TECLAST"), DMI_MATCH(DMI_BOARD_NAME, "tPAD"), }, .driver_data = (void *)(BYT_RT5640_IN3_MAP | - BYT_RT5640_MCLK_EN | - BYT_RT5640_SSP0_AIF1), + BYT_RT5640_JD_SRC_JD1_IN4P | + BYT_RT5640_OVCD_TH_2000UA | + BYT_RT5640_OVCD_SF_1P0 | + BYT_RT5640_SSP0_AIF1 | + BYT_RT5640_MCLK_EN), }, { /* Toshiba Satellite Click Mini L9W-B */ .matches = { -- cgit v1.2.1 From 9c9b65203492927cc4ae419e9601e837ecbd889e Mon Sep 17 00:00:00 2001 From: Curtis Malainey Date: Wed, 27 Nov 2019 17:13:58 -0800 Subject: ASoC: core: only flush inited work during free There are many paths to soc_free_pcm_runtime which can both have and have not yet inited the workqueue yet. When we flush the queue when we have not yet inited the queue we cause warnings to be printed. An example is soc_cleanup_card_resources which is called by snd_soc_bind_card which has multiple failure points before and after soc_link_init -> soc_new_pcm which is where the queue is inited. Signed-off-by: Curtis Malainey Link: https://lore.kernel.org/r/20191128011358.39234-1-cujomalainey@chromium.org Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 0e2e628302f1..1c84ff1a5bf9 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -419,7 +419,8 @@ static void soc_free_pcm_runtime(struct snd_soc_pcm_runtime *rtd) list_del(&rtd->list); - flush_delayed_work(&rtd->delayed_work); + if (delayed_work_pending(&rtd->delayed_work)) + flush_delayed_work(&rtd->delayed_work); snd_soc_pcm_component_free(rtd); /* -- cgit v1.2.1 From 76d2703649321c296df7ec0dafd50add96215de4 Mon Sep 17 00:00:00 2001 From: Dragos Tarcatu Date: Mon, 9 Dec 2019 18:39:38 -0600 Subject: ASoC: topology: Check return value for snd_soc_add_dai_link() snd_soc_add_dai_link() might fail. This situation occurs for instance in a very specific use case where a PCM device and a Back End DAI link are given identical names in the topology. When this happens, soc_new_pcm_runtime() fails and then snd_soc_add_dai_link() returns -ENOMEM when called from soc_tplg_fe_link_create(). Because of that, the link will not get added into the card list, so any attempt to remove it later ends up in a panic. Fix that by checking the return status and free the memory in case of an error. Reviewed-by: Ranjani Sridharan Signed-off-by: Dragos Tarcatu Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20191210003939.15752-2-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/soc-topology.c | 19 +++++++++++++------ 1 file changed, 13 insertions(+), 6 deletions(-) diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 81d2af000a5c..65c2796b6e02 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -1933,11 +1933,13 @@ static int soc_tplg_fe_link_create(struct soc_tplg *tplg, ret = soc_tplg_dai_link_load(tplg, link, NULL); if (ret < 0) { dev_err(tplg->comp->dev, "ASoC: FE link loading failed\n"); - kfree(link->name); - kfree(link->stream_name); - kfree(link->cpus->dai_name); - kfree(link); - return ret; + goto err; + } + + ret = snd_soc_add_dai_link(tplg->comp->card, link); + if (ret < 0) { + dev_err(tplg->comp->dev, "ASoC: adding FE link failed\n"); + goto err; } link->dobj.index = tplg->index; @@ -1945,8 +1947,13 @@ static int soc_tplg_fe_link_create(struct soc_tplg *tplg, link->dobj.type = SND_SOC_DOBJ_DAI_LINK; list_add(&link->dobj.list, &tplg->comp->dobj_list); - snd_soc_add_dai_link(tplg->comp->card, link); return 0; +err: + kfree(link->name); + kfree(link->stream_name); + kfree(link->cpus->dai_name); + kfree(link); + return ret; } /* create a FE DAI and DAI link from the PCM object */ -- cgit v1.2.1 From a3039aef52d9ffeb67e9211899cd3e8a2953a01f Mon Sep 17 00:00:00 2001 From: Dragos Tarcatu Date: Mon, 9 Dec 2019 18:39:39 -0600 Subject: ASoC: topology: Check return value for soc_tplg_pcm_create() The return value of soc_tplg_pcm_create() is currently not checked in soc_tplg_pcm_elems_load(). If an error is to occur there, the topology ignores it and continues loading. Fix that by checking the status and rejecting the topology on error. Reviewed-by: Ranjani Sridharan Signed-off-by: Dragos Tarcatu Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20191210003939.15752-3-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/soc-topology.c | 8 +++++++- 1 file changed, 7 insertions(+), 1 deletion(-) diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 65c2796b6e02..b28613149b0c 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -2046,6 +2046,7 @@ static int soc_tplg_pcm_elems_load(struct soc_tplg *tplg, int size; int i; bool abi_match; + int ret; count = le32_to_cpu(hdr->count); @@ -2087,7 +2088,12 @@ static int soc_tplg_pcm_elems_load(struct soc_tplg *tplg, } /* create the FE DAIs and DAI links */ - soc_tplg_pcm_create(tplg, _pcm); + ret = soc_tplg_pcm_create(tplg, _pcm); + if (ret < 0) { + if (!abi_match) + kfree(_pcm); + return ret; + } /* offset by version-specific struct size and * real priv data size -- cgit v1.2.1 From 5525cf07d15f7c7eab619707627c31aa8e39dff1 Mon Sep 17 00:00:00 2001 From: Daniel Baluta Date: Mon, 9 Dec 2019 15:53:53 +0200 Subject: ASoC: simple-card: Don't create separate link when platform is present In normal sound case all DAIs are detected as CPU-Codec. simple_dai_link_of supports the presence of a platform but it counts it as a CPU DAI resulting in the creation of an extra link. Adding a platform property to a link description like: simple-audio-card,dai-link { cpu { sound-dai = <&sai1>; }; plat { sound-dai = <&dsp>; }; codec { sound-dai = <&wm8960>; } will result in the creation of two links: * sai1 <-> wm8960 * dsp <-> wm8960 which is obviously not what we want. We just want one single link with: * sai1 <-> wm8960 (and platform set to dsp). Signed-off-by: Daniel Baluta Acked-by: Kuninori Morimoto Link: https://lore.kernel.org/r/20191209135353.17427-1-daniel.baluta@nxp.com Signed-off-by: Mark Brown --- sound/soc/generic/simple-card.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 10b82bf043d1..55e9f8800b3e 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -371,6 +371,7 @@ static int simple_for_each_link(struct asoc_simple_priv *priv, do { struct asoc_simple_data adata; struct device_node *codec; + struct device_node *plat; struct device_node *np; int num = of_get_child_count(node); @@ -381,6 +382,9 @@ static int simple_for_each_link(struct asoc_simple_priv *priv, ret = -ENODEV; goto error; } + /* get platform */ + plat = of_get_child_by_name(node, is_top ? + PREFIX "plat" : "plat"); /* get convert-xxx property */ memset(&adata, 0, sizeof(adata)); @@ -389,6 +393,8 @@ static int simple_for_each_link(struct asoc_simple_priv *priv, /* loop for all CPU/Codec node */ for_each_child_of_node(node, np) { + if (plat == np) + continue; /* * It is DPCM * if it has many CPUs, -- cgit v1.2.1 From 8edc95667646a75f0fc97e08ecb180581fdff300 Mon Sep 17 00:00:00 2001 From: Karol Trzcinski Date: Mon, 9 Dec 2019 18:48:48 -0600 Subject: ASoC: SOF: loader: snd_sof_fw_parse_ext_data log warning on unknown header Added warning log when found some unknown FW boot ext header, to improve debuggability. Signed-off-by: Karol Trzcinski Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20191210004854.16845-3-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/loader.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/sof/loader.c b/sound/soc/sof/loader.c index 9a9a381a908d..a041adf0669d 100644 --- a/sound/soc/sof/loader.c +++ b/sound/soc/sof/loader.c @@ -66,6 +66,8 @@ int snd_sof_fw_parse_ext_data(struct snd_sof_dev *sdev, u32 bar, u32 offset) ret = get_ext_windows(sdev, ext_hdr); break; default: + dev_warn(sdev->dev, "warning: unknown ext header type %d size 0x%x\n", + ext_hdr->type, ext_hdr->hdr.size); break; } -- cgit v1.2.1 From 6bb03c21e4bfee29e48e480ee4ca7cb1e12f587c Mon Sep 17 00:00:00 2001 From: Karol Trzcinski Date: Mon, 9 Dec 2019 18:48:49 -0600 Subject: ASoC: SOF: loader: fix snd_sof_fw_parse_ext_data An error occurs during parsing more than one ext_data from the mailbox, because of invalid data offset handling. Fix by removing the incorrect duplicate increment of the offset. The return value is also reset in the switch case. This does not change the behavior but improves readability - there is no longer a need to check what the return value of get_ext_windows is. Signed-off-by: Karol Trzcinski Signed-off-by: Bartosz Kokoszko Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20191210004854.16845-4-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/loader.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/sof/loader.c b/sound/soc/sof/loader.c index a041adf0669d..432d12bd4937 100644 --- a/sound/soc/sof/loader.c +++ b/sound/soc/sof/loader.c @@ -50,8 +50,7 @@ int snd_sof_fw_parse_ext_data(struct snd_sof_dev *sdev, u32 bar, u32 offset) while (ext_hdr->hdr.cmd == SOF_IPC_FW_READY) { /* read in ext structure */ - offset += sizeof(*ext_hdr); - snd_sof_dsp_block_read(sdev, bar, offset, + snd_sof_dsp_block_read(sdev, bar, offset + sizeof(*ext_hdr), (void *)((u8 *)ext_data + sizeof(*ext_hdr)), ext_hdr->hdr.size - sizeof(*ext_hdr)); @@ -61,6 +60,7 @@ int snd_sof_fw_parse_ext_data(struct snd_sof_dev *sdev, u32 bar, u32 offset) /* process structure data */ switch (ext_hdr->type) { case SOF_IPC_EXT_DMA_BUFFER: + ret = 0; break; case SOF_IPC_EXT_WINDOW: ret = get_ext_windows(sdev, ext_hdr); @@ -68,6 +68,7 @@ int snd_sof_fw_parse_ext_data(struct snd_sof_dev *sdev, u32 bar, u32 offset) default: dev_warn(sdev->dev, "warning: unknown ext header type %d size 0x%x\n", ext_hdr->type, ext_hdr->hdr.size); + ret = 0; break; } -- cgit v1.2.1 From f84337c3fb8ff4d533ccbed0d2db4e8587d0ff58 Mon Sep 17 00:00:00 2001 From: Curtis Malainey Date: Mon, 9 Dec 2019 18:48:52 -0600 Subject: ASoC: SOF: Intel: split cht and byt debug window sizes Turns out SSP 3-5 are only available on cht, to avoid dumping on undefined registers let's split the definition. Signed-off-by: Curtis Malainey Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20191210004854.16845-7-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/byt.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/soc/sof/intel/byt.c b/sound/soc/sof/intel/byt.c index b9061b79a57b..92ef6a796fd5 100644 --- a/sound/soc/sof/intel/byt.c +++ b/sound/soc/sof/intel/byt.c @@ -24,7 +24,8 @@ #define DRAM_OFFSET 0x100000 #define DRAM_SIZE (160 * 1024) #define SHIM_OFFSET 0x140000 -#define SHIM_SIZE 0x100 +#define SHIM_SIZE_BYT 0x100 +#define SHIM_SIZE_CHT 0x118 #define MBOX_OFFSET 0x144000 #define MBOX_SIZE 0x1000 #define EXCEPT_OFFSET 0x800 @@ -75,7 +76,7 @@ static const struct snd_sof_debugfs_map byt_debugfs[] = { SOF_DEBUGFS_ACCESS_D0_ONLY}, {"dram", BYT_DSP_BAR, DRAM_OFFSET, DRAM_SIZE, SOF_DEBUGFS_ACCESS_D0_ONLY}, - {"shim", BYT_DSP_BAR, SHIM_OFFSET, SHIM_SIZE, + {"shim", BYT_DSP_BAR, SHIM_OFFSET, SHIM_SIZE_BYT, SOF_DEBUGFS_ACCESS_ALWAYS}, }; @@ -102,7 +103,7 @@ static const struct snd_sof_debugfs_map cht_debugfs[] = { SOF_DEBUGFS_ACCESS_D0_ONLY}, {"dram", BYT_DSP_BAR, DRAM_OFFSET, DRAM_SIZE, SOF_DEBUGFS_ACCESS_D0_ONLY}, - {"shim", BYT_DSP_BAR, SHIM_OFFSET, SHIM_SIZE, + {"shim", BYT_DSP_BAR, SHIM_OFFSET, SHIM_SIZE_CHT, SOF_DEBUGFS_ACCESS_ALWAYS}, }; -- cgit v1.2.1 From 4e64ba3fd181b207c00d392b27ed484b89108dce Mon Sep 17 00:00:00 2001 From: Amery Song Date: Mon, 9 Dec 2019 18:48:54 -0600 Subject: ASoC: Intel: common: work-around incorrect ACPI HID for CML boards On CML boards with the RT5682 headset codec and RT1011 speaker amplifier, the platform firmware exposes three ACPI HIDs (10EC5682, 10EC1011 and MX98357A). The last HID is a mistake in DSDT tables, which causes the wrong machine driver to be loaded. This patch changes the key used to identify boards and changes the order of entries in the table to load the correct machine driver. The order does matter and should not be modified to work-around this firmware issue. Signed-off-by: Amery Song Signed-off-by: Keyon Jie Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20191210004854.16845-9-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/common/soc-acpi-intel-cml-match.c | 41 +++++++++++++---------- 1 file changed, 24 insertions(+), 17 deletions(-) diff --git a/sound/soc/intel/common/soc-acpi-intel-cml-match.c b/sound/soc/intel/common/soc-acpi-intel-cml-match.c index 5d08ae066738..fb9ba8819706 100644 --- a/sound/soc/intel/common/soc-acpi-intel-cml-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-cml-match.c @@ -9,45 +9,52 @@ #include #include -static struct snd_soc_acpi_codecs cml_codecs = { +static struct snd_soc_acpi_codecs rt1011_spk_codecs = { .num_codecs = 1, - .codecs = {"10EC5682"} + .codecs = {"10EC1011"} }; -static struct snd_soc_acpi_codecs cml_spk_codecs = { +static struct snd_soc_acpi_codecs max98357a_spk_codecs = { .num_codecs = 1, .codecs = {"MX98357A"} }; +/* + * The order of the three entries with .id = "10EC5682" matters + * here, because DSDT tables expose an ACPI HID for the MAX98357A + * speaker amplifier which is not populated on the board. + */ struct snd_soc_acpi_mach snd_soc_acpi_intel_cml_machines[] = { { - .id = "DLGS7219", - .drv_name = "cml_da7219_max98357a", - .quirk_data = &cml_spk_codecs, + .id = "10EC5682", + .drv_name = "cml_rt1011_rt5682", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &rt1011_spk_codecs, .sof_fw_filename = "sof-cml.ri", - .sof_tplg_filename = "sof-cml-da7219-max98357a.tplg", + .sof_tplg_filename = "sof-cml-rt1011-rt5682.tplg", }, { - .id = "MX98357A", + .id = "10EC5682", .drv_name = "sof_rt5682", - .quirk_data = &cml_codecs, + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &max98357a_spk_codecs, .sof_fw_filename = "sof-cml.ri", .sof_tplg_filename = "sof-cml-rt5682-max98357a.tplg", }, - { - .id = "10EC1011", - .drv_name = "cml_rt1011_rt5682", - .quirk_data = &cml_codecs, - .sof_fw_filename = "sof-cml.ri", - .sof_tplg_filename = "sof-cml-rt1011-rt5682.tplg", - }, { .id = "10EC5682", .drv_name = "sof_rt5682", .sof_fw_filename = "sof-cml.ri", .sof_tplg_filename = "sof-cml-rt5682.tplg", }, - + { + .id = "DLGS7219", + .drv_name = "cml_da7219_max98357a", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &max98357a_spk_codecs, + .sof_fw_filename = "sof-cml.ri", + .sof_tplg_filename = "sof-cml-da7219-max98357a.tplg", + }, {}, }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_cml_machines); -- cgit v1.2.1 From 5815bdfd7f54739be9abed1301d55f5e74d7ad1f Mon Sep 17 00:00:00 2001 From: Hui Wang Date: Wed, 11 Dec 2019 13:13:21 +0800 Subject: ALSA: hda/realtek - Line-out jack doesn't work on a Dell AIO After applying the fixup ALC274_FIXUP_DELL_AIO_LINEOUT_VERB, the Line-out jack works well. And instead of adding a new set of pin definition in the pin_fixup_tbl, we put a more generic matching entry in the fallback_pin_fixup_tbl. Cc: Signed-off-by: Hui Wang Link: https://lore.kernel.org/r/20191211051321.5883-1-hui.wang@canonical.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 8 +++----- 1 file changed, 3 insertions(+), 5 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 6d6e34b3b3aa..dbfafee97931 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7643,11 +7643,6 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { {0x1a, 0x90a70130}, {0x1b, 0x90170110}, {0x21, 0x03211020}), - SND_HDA_PIN_QUIRK(0x10ec0274, 0x1028, "Dell", ALC274_FIXUP_DELL_AIO_LINEOUT_VERB, - {0x12, 0xb7a60130}, - {0x13, 0xb8a61140}, - {0x16, 0x90170110}, - {0x21, 0x04211020}), SND_HDA_PIN_QUIRK(0x10ec0280, 0x103c, "HP", ALC280_FIXUP_HP_GPIO4, {0x12, 0x90a60130}, {0x14, 0x90170110}, @@ -7841,6 +7836,9 @@ static const struct snd_hda_pin_quirk alc269_fallback_pin_fixup_tbl[] = { SND_HDA_PIN_QUIRK(0x10ec0236, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, {0x19, 0x40000000}, {0x1a, 0x40000000}), + SND_HDA_PIN_QUIRK(0x10ec0274, 0x1028, "Dell", ALC274_FIXUP_DELL_AIO_LINEOUT_VERB, + {0x19, 0x40000000}, + {0x1a, 0x40000000}), {} }; -- cgit v1.2.1 From add9d56d7b3781532208afbff5509d7382fb6efe Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 11 Dec 2019 16:57:42 +0100 Subject: ALSA: pcm: Avoid possible info leaks from PCM stream buffers The current PCM code doesn't initialize explicitly the buffers allocated for PCM streams, hence it might leak some uninitialized kernel data or previous stream contents by mmapping or reading the buffer before actually starting the stream. Since this is a common problem, this patch simply adds the clearance of the buffer data at hw_params callback. Although this does only zero-clear no matter which format is used, which doesn't mean the silence for some formats, but it should be OK because the intention is just to clear the previous data on the buffer. Reported-by: Lionel Koenig Cc: Link: https://lore.kernel.org/r/20191211155742.3213-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/core/pcm_native.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 1fe581167b7b..d083225344a0 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -739,6 +739,10 @@ static int snd_pcm_hw_params(struct snd_pcm_substream *substream, while (runtime->boundary * 2 <= LONG_MAX - runtime->buffer_size) runtime->boundary *= 2; + /* clear the buffer for avoiding possible kernel info leaks */ + if (runtime->dma_area && !substream->ops->copy_user) + memset(runtime->dma_area, 0, runtime->dma_bytes); + snd_pcm_timer_resolution_change(substream); snd_pcm_set_state(substream, SNDRV_PCM_STATE_SETUP); -- cgit v1.2.1 From 377bc0cfabce0244632dada19060839ced4e6949 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 13 Dec 2019 09:51:09 +0100 Subject: ALSA: hda/ca0132 - Keep power on during processing DSP response We need to keep power on while processing the DSP response via unsol event. Each snd_hda_codec_read() call does the power management, so it should work normally, but still it's safer to keep the power up for the whole function. Fixes: a73d511c4867 ("ALSA: hda/ca0132: Add unsol handler for DSP and jack detection") Cc: Link: https://lore.kernel.org/r/20191213085111.22855-2-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_ca0132.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index b7a1abb3e231..c3d34ff3d9ec 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -7588,12 +7588,14 @@ static void ca0132_process_dsp_response(struct hda_codec *codec, struct ca0132_spec *spec = codec->spec; codec_dbg(codec, "ca0132_process_dsp_response\n"); + snd_hda_power_up_pm(codec); if (spec->wait_scp) { if (dspio_get_response_data(codec) >= 0) spec->wait_scp = 0; } dspio_clear_response_queue(codec); + snd_hda_power_down_pm(codec); } static void hp_callback(struct hda_codec *codec, struct hda_jack_callback *cb) -- cgit v1.2.1 From cb04fc3b6b076f67d228a0b7d096c69ad486c09c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 13 Dec 2019 09:51:10 +0100 Subject: ALSA: hda/ca0132 - Avoid endless loop Introduce a timeout to dspio_clear_response_queue() so that it won't be caught in an endless loop even if the hardware doesn't respond properly. Fixes: a73d511c4867 ("ALSA: hda/ca0132: Add unsol handler for DSP and jack detection") Cc: Link: https://lore.kernel.org/r/20191213085111.22855-3-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_ca0132.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index c3d34ff3d9ec..8d0209fff8f5 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -1809,13 +1809,14 @@ struct scp_msg { static void dspio_clear_response_queue(struct hda_codec *codec) { + unsigned long timeout = jiffies + msecs_to_jiffies(1000); unsigned int dummy = 0; - int status = -1; + int status; /* clear all from the response queue */ do { status = dspio_read(codec, &dummy); - } while (status == 0); + } while (status == 0 && time_before(jiffies, timeout)); } static int dspio_get_response_data(struct hda_codec *codec) -- cgit v1.2.1 From 42fb6b1d41eb5905d77c06cad2e87b70289bdb76 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 13 Dec 2019 09:51:11 +0100 Subject: ALSA: hda/ca0132 - Fix work handling in delayed HP detection CA0132 has the delayed HP jack detection code that is invoked from the unsol handler, but it does a few weird things: it contains the cancel of a work inside the work handler, and yet it misses the cancel-sync call at (runtime-)suspend. This patch addresses those issues. Fixes: 15c2b3cc09a3 ("ALSA: hda/ca0132 - Fix possible workqueue stall") Cc: Link: https://lore.kernel.org/r/20191213085111.22855-4-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_ca0132.c | 16 ++++++++++++++-- 1 file changed, 14 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index 8d0209fff8f5..32ed46464af7 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -7607,11 +7607,10 @@ static void hp_callback(struct hda_codec *codec, struct hda_jack_callback *cb) /* Delay enabling the HP amp, to let the mic-detection * state machine run. */ - cancel_delayed_work(&spec->unsol_hp_work); - schedule_delayed_work(&spec->unsol_hp_work, msecs_to_jiffies(500)); tbl = snd_hda_jack_tbl_get(codec, cb->nid); if (tbl) tbl->block_report = 1; + schedule_delayed_work(&spec->unsol_hp_work, msecs_to_jiffies(500)); } static void amic_callback(struct hda_codec *codec, struct hda_jack_callback *cb) @@ -8457,12 +8456,25 @@ static void ca0132_reboot_notify(struct hda_codec *codec) codec->patch_ops.free(codec); } +#ifdef CONFIG_PM +static int ca0132_suspend(struct hda_codec *codec) +{ + struct ca0132_spec *spec = codec->spec; + + cancel_delayed_work_sync(&spec->unsol_hp_work); + return 0; +} +#endif + static const struct hda_codec_ops ca0132_patch_ops = { .build_controls = ca0132_build_controls, .build_pcms = ca0132_build_pcms, .init = ca0132_init, .free = ca0132_free, .unsol_event = snd_hda_jack_unsol_event, +#ifdef CONFIG_PM + .suspend = ca0132_suspend, +#endif .reboot_notify = ca0132_reboot_notify, }; -- cgit v1.2.1 From 6fd739c04ffd877641b01371f9fde67901e7f9cb Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sat, 14 Dec 2019 18:52:17 +0100 Subject: ALSA: hda: Fix regression by strip mask fix The commit e38e486d66e2 ("ALSA: hda: Modify stream stripe mask only when needed") tried to address the regression by the unconditional application of the stripe mask, but this caused yet another regression for the previously working devices. Namely, the patch clears the azx_dev->stripe flag at snd_hdac_stream_clear(), but this may be called multiple times before restarting the stream, so this ended up with clearance of the flag for the whole time. This patch fixes the regression by moving the azx_dev->stripe flag clearance at the counter-part, the close callback of HDMI codec driver instead. Fixes: e38e486d66e2 ("ALSA: hda: Modify stream stripe mask only when needed") BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=205855 BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=204477 Cc: Link: https://lore.kernel.org/r/20191214175217.31852-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/hda/hdac_stream.c | 4 +--- sound/pci/hda/patch_hdmi.c | 2 ++ 2 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/hda/hdac_stream.c b/sound/hda/hdac_stream.c index f9707fb05efe..682ed39f79b0 100644 --- a/sound/hda/hdac_stream.c +++ b/sound/hda/hdac_stream.c @@ -120,10 +120,8 @@ void snd_hdac_stream_clear(struct hdac_stream *azx_dev) snd_hdac_stream_updateb(azx_dev, SD_CTL, SD_CTL_DMA_START | SD_INT_MASK, 0); snd_hdac_stream_writeb(azx_dev, SD_STS, SD_INT_MASK); /* to be sure */ - if (azx_dev->stripe) { + if (azx_dev->stripe) snd_hdac_stream_updateb(azx_dev, SD_CTL_3B, SD_CTL_STRIPE_MASK, 0); - azx_dev->stripe = 0; - } azx_dev->running = false; } EXPORT_SYMBOL_GPL(snd_hdac_stream_clear); diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 78647ee02339..630b1f5c276d 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -2021,6 +2021,8 @@ static int hdmi_pcm_close(struct hda_pcm_stream *hinfo, per_cvt->assigned = 0; hinfo->nid = 0; + azx_stream(get_azx_dev(substream))->stripe = 0; + mutex_lock(&spec->pcm_lock); snd_hda_spdif_ctls_unassign(codec, pcm_idx); clear_bit(pcm_idx, &spec->pcm_in_use); -- cgit v1.2.1 From 556672d75ff486e0b6786056da624131679e0576 Mon Sep 17 00:00:00 2001 From: Shengjiu Wang Date: Wed, 11 Dec 2019 19:57:22 +0800 Subject: ASoC: wm8962: fix lambda value According to user manual, it is required that FLL_LAMBDA > 0 in all cases (Integer and Franctional modes). Fixes: 9a76f1ff6e29 ("ASoC: Add initial WM8962 CODEC driver") Signed-off-by: Shengjiu Wang Acked-by: Charles Keepax Link: https://lore.kernel.org/r/1576065442-19763-1-git-send-email-shengjiu.wang@nxp.com Signed-off-by: Mark Brown --- sound/soc/codecs/wm8962.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 3e5c69fbc33a..d9d59f45833f 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -2788,7 +2788,7 @@ static int fll_factors(struct _fll_div *fll_div, unsigned int Fref, if (target % Fref == 0) { fll_div->theta = 0; - fll_div->lambda = 0; + fll_div->lambda = 1; } else { gcd_fll = gcd(target, fratio * Fref); @@ -2858,7 +2858,7 @@ static int wm8962_set_fll(struct snd_soc_component *component, int fll_id, int s return -EINVAL; } - if (fll_div.theta || fll_div.lambda) + if (fll_div.theta) fll1 |= WM8962_FLL_FRAC; /* Stop the FLL while we reconfigure */ -- cgit v1.2.1 From 475feec0c41ad71cb7d02f0310e56256606b57c5 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 16 Dec 2019 16:12:24 +0100 Subject: ALSA: hda - Downgrade error message for single-cmd fallback We made the error message for the CORB/RIRB communication clearer by upgrading to dev_WARN() so that user can notice better. But this struck us like a boomerang: now it caught syzbot and reported back as a fatal issue although it's not really any too serious bug that worth for stopping the whole system. OK, OK, let's be softy, downgrade it to the standard dev_err() again. Fixes: dd65f7e19c69 ("ALSA: hda - Show the fatal CORB/RIRB error more clearly") Reported-by: syzbot+b3028ac3933f5c466389@syzkaller.appspotmail.com Link: https://lore.kernel.org/r/20191216151224.30013-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_controller.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index 2f3b7a35f2d9..ba56b59b3e17 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -883,7 +883,7 @@ static int azx_rirb_get_response(struct hdac_bus *bus, unsigned int addr, return -EAGAIN; /* give a chance to retry */ } - dev_WARN(chip->card->dev, + dev_err(chip->card->dev, "azx_get_response timeout, switching to single_cmd mode: last cmd=0x%08x\n", bus->last_cmd[addr]); chip->single_cmd = 1; -- cgit v1.2.1 From 92adc96f8eecd9522a907c197cc3d62e405539fe Mon Sep 17 00:00:00 2001 From: Hui Wang Date: Wed, 18 Dec 2019 21:26:50 +0800 Subject: ALSA: usb-audio: set the interface format after resume on Dell WD19 Recently we found the headset-mic on the Dell Dock WD19 doesn't work anymore after s3 (s2i or deep), this problem could be workarounded by closing (pcm_close) the app and then reopening (pcm_open) the app, so this bug is not easy to be detected by users. When problem happens, retire_capture_urb() could still be called periodically, but the size of captured data is always 0, it could be a firmware bug on the dock. Anyway I found after resuming, the snd_usb_pcm_prepare() will be called, and if we forcibly run set_format() to set the interface and its endpoint, the capture size will be normal again. This problem and workaound also apply to playback. To fix it in the kernel, add a quirk to let set_format() run forcibly once after resume. Signed-off-by: Hui Wang Cc: Link: https://lore.kernel.org/r/20191218132650.6303-1-hui.wang@canonical.com Signed-off-by: Takashi Iwai --- sound/usb/card.h | 1 + sound/usb/pcm.c | 21 +++++++++++++++++++-- sound/usb/quirks-table.h | 3 ++- sound/usb/quirks.c | 11 +++++++++++ sound/usb/usbaudio.h | 3 ++- 5 files changed, 35 insertions(+), 4 deletions(-) diff --git a/sound/usb/card.h b/sound/usb/card.h index 2991b9986f66..395403a2d33f 100644 --- a/sound/usb/card.h +++ b/sound/usb/card.h @@ -145,6 +145,7 @@ struct snd_usb_substream { struct snd_usb_endpoint *sync_endpoint; unsigned long flags; bool need_setup_ep; /* (re)configure EP at prepare? */ + bool need_setup_fmt; /* (re)configure fmt after resume? */ unsigned int speed; /* USB_SPEED_XXX */ u64 formats; /* format bitmasks (all or'ed) */ diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 9c8930bb00c8..96298c767c76 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -510,11 +510,11 @@ static int set_format(struct snd_usb_substream *subs, struct audioformat *fmt) if (WARN_ON(altsd->bAlternateSetting != fmt->altsetting)) return -EINVAL; - if (fmt == subs->cur_audiofmt) + if (fmt == subs->cur_audiofmt && !subs->need_setup_fmt) return 0; /* close the old interface */ - if (subs->interface >= 0 && subs->interface != fmt->iface) { + if (subs->interface >= 0 && (subs->interface != fmt->iface || subs->need_setup_fmt)) { if (!subs->stream->chip->keep_iface) { err = usb_set_interface(subs->dev, subs->interface, 0); if (err < 0) { @@ -528,6 +528,9 @@ static int set_format(struct snd_usb_substream *subs, struct audioformat *fmt) subs->altset_idx = 0; } + if (subs->need_setup_fmt) + subs->need_setup_fmt = false; + /* set interface */ if (iface->cur_altsetting != alts) { err = snd_usb_select_mode_quirk(subs, fmt); @@ -1728,6 +1731,13 @@ static int snd_usb_substream_playback_trigger(struct snd_pcm_substream *substrea subs->data_endpoint->retire_data_urb = retire_playback_urb; subs->running = 0; return 0; + case SNDRV_PCM_TRIGGER_SUSPEND: + if (subs->stream->chip->setup_fmt_after_resume_quirk) { + stop_endpoints(subs, true); + subs->need_setup_fmt = true; + return 0; + } + break; } return -EINVAL; @@ -1760,6 +1770,13 @@ static int snd_usb_substream_capture_trigger(struct snd_pcm_substream *substream subs->data_endpoint->retire_data_urb = retire_capture_urb; subs->running = 1; return 0; + case SNDRV_PCM_TRIGGER_SUSPEND: + if (subs->stream->chip->setup_fmt_after_resume_quirk) { + stop_endpoints(subs, true); + subs->need_setup_fmt = true; + return 0; + } + break; } return -EINVAL; diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 70c338f3ae24..d187aa6d50db 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -3466,7 +3466,8 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"), .vendor_name = "Dell", .product_name = "WD19 Dock", .profile_name = "Dell-WD15-Dock", - .ifnum = QUIRK_NO_INTERFACE + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_SETUP_FMT_AFTER_RESUME } }, /* MOTU Microbook II */ diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 349e1e52996d..a81c2066499f 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -508,6 +508,16 @@ static int create_standard_mixer_quirk(struct snd_usb_audio *chip, return snd_usb_create_mixer(chip, quirk->ifnum, 0); } + +static int setup_fmt_after_resume_quirk(struct snd_usb_audio *chip, + struct usb_interface *iface, + struct usb_driver *driver, + const struct snd_usb_audio_quirk *quirk) +{ + chip->setup_fmt_after_resume_quirk = 1; + return 1; /* Continue with creating streams and mixer */ +} + /* * audio-interface quirks * @@ -546,6 +556,7 @@ int snd_usb_create_quirk(struct snd_usb_audio *chip, [QUIRK_AUDIO_EDIROL_UAXX] = create_uaxx_quirk, [QUIRK_AUDIO_ALIGN_TRANSFER] = create_align_transfer_quirk, [QUIRK_AUDIO_STANDARD_MIXER] = create_standard_mixer_quirk, + [QUIRK_SETUP_FMT_AFTER_RESUME] = setup_fmt_after_resume_quirk, }; if (quirk->type < QUIRK_TYPE_COUNT) { diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index ff3cbf653de8..6fe3ab582ec6 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -33,7 +33,7 @@ struct snd_usb_audio { wait_queue_head_t shutdown_wait; unsigned int txfr_quirk:1; /* Subframe boundaries on transfers */ unsigned int tx_length_quirk:1; /* Put length specifier in transfers */ - + unsigned int setup_fmt_after_resume_quirk:1; /* setup the format to interface after resume */ int num_interfaces; int num_suspended_intf; int sample_rate_read_error; @@ -98,6 +98,7 @@ enum quirk_type { QUIRK_AUDIO_EDIROL_UAXX, QUIRK_AUDIO_ALIGN_TRANSFER, QUIRK_AUDIO_STANDARD_MIXER, + QUIRK_SETUP_FMT_AFTER_RESUME, QUIRK_TYPE_COUNT }; -- cgit v1.2.1