From ea75deef1a738d25502cfbb2caa564270b271525 Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Mon, 15 Aug 2022 13:31:38 +0100 Subject: [PATCH 01/29] ASoC: cs42l42: Only report button state if there was a button interrupt Only report a button state change if the interrupt status shows that there was a button event. Previously the code would always drop into the button reporting at the end of interrupt handling if the jack was present. If neither of the button report interrupts were pending it would report all buttons released. This could then lead to a button being reported as released while it is still pressed. Fixes: c5b8ee0879bc ("ASoC: cs42l42: Report jack and button detection") Signed-off-by: Richard Fitzgerald Link: https://lore.kernel.org/r/20220815123138.3810249-1-rf@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/cs42l42.c | 13 +++++++------ 1 file changed, 7 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c index d545a593a251..daafd4251ce6 100644 --- a/sound/soc/codecs/cs42l42.c +++ b/sound/soc/codecs/cs42l42.c @@ -1617,7 +1617,6 @@ static irqreturn_t cs42l42_irq_thread(int irq, void *data) unsigned int current_plug_status; unsigned int current_button_status; unsigned int i; - int report = 0; mutex_lock(&cs42l42->irq_lock); if (cs42l42->suspended) { @@ -1711,13 +1710,15 @@ static irqreturn_t cs42l42_irq_thread(int irq, void *data) if (current_button_status & CS42L42_M_DETECT_TF_MASK) { dev_dbg(cs42l42->dev, "Button released\n"); - report = 0; + snd_soc_jack_report(cs42l42->jack, 0, + SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3); } else if (current_button_status & CS42L42_M_DETECT_FT_MASK) { - report = cs42l42_handle_button_press(cs42l42); - + snd_soc_jack_report(cs42l42->jack, + cs42l42_handle_button_press(cs42l42), + SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3); } - snd_soc_jack_report(cs42l42->jack, report, SND_JACK_BTN_0 | SND_JACK_BTN_1 | - SND_JACK_BTN_2 | SND_JACK_BTN_3); } } From dcdfa3471f9c28ee716c687d85701353e2e86fde Mon Sep 17 00:00:00 2001 From: Pieterjan Camerlynck Date: Sat, 13 Aug 2022 10:33:52 +0200 Subject: [PATCH 02/29] ASoC: fsl_sai: fix incorrect mclk number in error message In commit c3ecef21c3f26 ("ASoC: fsl_sai: add sai master mode support") the loop was changed to start iterating from 1 instead of 0. The error message however was not updated, reporting the wrong clock to the user. Signed-off-by: Pieterjan Camerlynck Acked-by: Shengjiu Wang Link: https://lore.kernel.org/r/20220813083353.8959-1-pieterjan.camerlynck@gmail.com Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_sai.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 7523bb944b21..d430eece1d6b 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -1306,7 +1306,7 @@ static int fsl_sai_probe(struct platform_device *pdev) sai->mclk_clk[i] = devm_clk_get(dev, tmp); if (IS_ERR(sai->mclk_clk[i])) { dev_err(dev, "failed to get mclk%d clock: %ld\n", - i + 1, PTR_ERR(sai->mclk_clk[i])); + i, PTR_ERR(sai->mclk_clk[i])); sai->mclk_clk[i] = NULL; } } From c6e14bb9f50df7126ca64405ae807d8bc7b39f9a Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Tue, 16 Aug 2022 17:52:29 +0100 Subject: [PATCH 03/29] ASoC: qcom: sm8250: add missing module owner Add missing module owner to able to build and load this driver as module. Fixes: aa2e2785545a ("ASoC: qcom: sm8250: add sound card qrb5165-rb5 support") Signed-off-by: Srinivas Kandagatla Link: https://lore.kernel.org/r/20220816165229.7971-1-srinivas.kandagatla@linaro.org Signed-off-by: Mark Brown --- sound/soc/qcom/sm8250.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/qcom/sm8250.c b/sound/soc/qcom/sm8250.c index ce4a5713386a..98a2fde9e004 100644 --- a/sound/soc/qcom/sm8250.c +++ b/sound/soc/qcom/sm8250.c @@ -270,6 +270,7 @@ static int sm8250_platform_probe(struct platform_device *pdev) if (!card) return -ENOMEM; + card->owner = THIS_MODULE; /* Allocate the private data */ data = devm_kzalloc(dev, sizeof(*data), GFP_KERNEL); if (!data) From ecdb10df7e0d83bfd12fb8f71e28ea4753e3716a Mon Sep 17 00:00:00 2001 From: Yang Yingliang Date: Thu, 18 Aug 2022 16:17:51 +0800 Subject: [PATCH 04/29] ASoC: SOF: ipc4-topology: fix wrong use of sizeof in sof_ipc4_widget_setup_comp_src() It should be size of the struct sof_ipc4_src, not data pointer pass to sof_update_ipc_object(). Fixes: b85f4fc40d56 ("ASoC: SOF: add ipc4 SRC module support") Signed-off-by: Yang Yingliang Acked-by: Peter Ujfalusi Link: https://lore.kernel.org/r/20220818081751.2407066-1-yangyingliang@huawei.com Signed-off-by: Mark Brown --- sound/soc/sof/ipc4-topology.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sof/ipc4-topology.c b/sound/soc/sof/ipc4-topology.c index af072b484a60..c6abfaf5d532 100644 --- a/sound/soc/sof/ipc4-topology.c +++ b/sound/soc/sof/ipc4-topology.c @@ -771,7 +771,7 @@ static int sof_ipc4_widget_setup_comp_src(struct snd_sof_widget *swidget) goto err; ret = sof_update_ipc_object(scomp, src, SOF_SRC_TOKENS, swidget->tuples, - swidget->num_tuples, sizeof(src), 1); + swidget->num_tuples, sizeof(*src), 1); if (ret) { dev_err(scomp->dev, "Parsing SRC tokens failed\n"); goto err; From 221ab1f0bf46236cf1a3fef5298ff5894acfb0c5 Mon Sep 17 00:00:00 2001 From: Jiaxin Yu Date: Sat, 20 Aug 2022 15:19:25 +0800 Subject: [PATCH 05/29] ASoC: mediatek: mt8186: fix DMIC record noise When the first DMIC recording is power down, mtkaif_dmic will be reset. This will cause configuration error in the second DMIC recording. So do not reset mtkaif_dmic except in "MTKAIF_DMIC Switch" kcontrol. Signed-off-by: Jiaxin Yu Link: https://lore.kernel.org/r/20220820071925.13557-1-jiaxin.yu@mediatek.com Signed-off-by: Mark Brown --- sound/soc/mediatek/mt8186/mt8186-dai-adda.c | 3 --- 1 file changed, 3 deletions(-) diff --git a/sound/soc/mediatek/mt8186/mt8186-dai-adda.c b/sound/soc/mediatek/mt8186/mt8186-dai-adda.c index 266704556f37..094402470dc2 100644 --- a/sound/soc/mediatek/mt8186/mt8186-dai-adda.c +++ b/sound/soc/mediatek/mt8186/mt8186-dai-adda.c @@ -271,9 +271,6 @@ static int mtk_adda_ul_event(struct snd_soc_dapm_widget *w, /* should delayed 1/fs(smallest is 8k) = 125us before afe off */ usleep_range(125, 135); mt8186_afe_gpio_request(afe->dev, false, MT8186_DAI_ADDA, 1); - - /* reset dmic */ - afe_priv->mtkaif_dmic = 0; break; default: break; From cf5071876baf995f8f98e86ef06f85a58feda63c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 23 Aug 2022 10:09:56 +0200 Subject: [PATCH 06/29] ASoC: nau8821: Implement hw constraint for rates nau8821 driver restricts the sample rate with over sampling rate, but currently it barely bails out at hw_params with -EINVAL error (with a kernel message); this doesn't help for user-space to recognize which rate can be actually used. This patch introduces the proper hw constraint for adjusting the available range of the sample rate depending on the OSR setup, as well as some code cleanup, for improving the communication with user-space. Now applications can know the valid rate beforehand and reduces the rate appropriately without errors. Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/20220823081000.2965-2-tiwai@suse.de Signed-off-by: Mark Brown --- sound/soc/codecs/nau8821.c | 66 +++++++++++++++++++++----------------- 1 file changed, 36 insertions(+), 30 deletions(-) diff --git a/sound/soc/codecs/nau8821.c b/sound/soc/codecs/nau8821.c index 2d21339932e6..4a72b94e8410 100644 --- a/sound/soc/codecs/nau8821.c +++ b/sound/soc/codecs/nau8821.c @@ -670,28 +670,40 @@ static const struct snd_soc_dapm_route nau8821_dapm_routes[] = { {"HPOR", NULL, "Class G"}, }; -static int nau8821_clock_check(struct nau8821 *nau8821, - int stream, int rate, int osr) +static const struct nau8821_osr_attr * +nau8821_get_osr(struct nau8821 *nau8821, int stream) { - int osrate = 0; + unsigned int osr; if (stream == SNDRV_PCM_STREAM_PLAYBACK) { + regmap_read(nau8821->regmap, NAU8821_R2C_DAC_CTRL1, &osr); + osr &= NAU8821_DAC_OVERSAMPLE_MASK; if (osr >= ARRAY_SIZE(osr_dac_sel)) - return -EINVAL; - osrate = osr_dac_sel[osr].osr; + return NULL; + return &osr_dac_sel[osr]; } else { + regmap_read(nau8821->regmap, NAU8821_R2B_ADC_RATE, &osr); + osr &= NAU8821_ADC_SYNC_DOWN_MASK; if (osr >= ARRAY_SIZE(osr_adc_sel)) - return -EINVAL; - osrate = osr_adc_sel[osr].osr; + return NULL; + return &osr_adc_sel[osr]; } +} - if (!osrate || rate * osrate > CLK_DA_AD_MAX) { - dev_err(nau8821->dev, - "exceed the maximum frequency of CLK_ADC or CLK_DAC"); +static int nau8821_dai_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct nau8821 *nau8821 = snd_soc_component_get_drvdata(component); + const struct nau8821_osr_attr *osr; + + osr = nau8821_get_osr(nau8821, substream->stream); + if (!osr || !osr->osr) return -EINVAL; - } - return 0; + return snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_RATE, + 0, CLK_DA_AD_MAX / osr->osr); } static int nau8821_hw_params(struct snd_pcm_substream *substream, @@ -699,7 +711,8 @@ static int nau8821_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_component *component = dai->component; struct nau8821 *nau8821 = snd_soc_component_get_drvdata(component); - unsigned int val_len = 0, osr, ctrl_val, bclk_fs, clk_div; + unsigned int val_len = 0, ctrl_val, bclk_fs, clk_div; + const struct nau8821_osr_attr *osr; nau8821->fs = params_rate(params); /* CLK_DAC or CLK_ADC = OSR * FS @@ -708,27 +721,19 @@ static int nau8821_hw_params(struct snd_pcm_substream *substream, * values must be selected such that the maximum frequency is less * than 6.144 MHz. */ - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - regmap_read(nau8821->regmap, NAU8821_R2C_DAC_CTRL1, &osr); - osr &= NAU8821_DAC_OVERSAMPLE_MASK; - if (nau8821_clock_check(nau8821, substream->stream, - nau8821->fs, osr)) { - return -EINVAL; - } + osr = nau8821_get_osr(nau8821, substream->stream); + if (!osr || !osr->osr) + return -EINVAL; + if (nau8821->fs * osr->osr > CLK_DA_AD_MAX) + return -EINVAL; + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) regmap_update_bits(nau8821->regmap, NAU8821_R03_CLK_DIVIDER, NAU8821_CLK_DAC_SRC_MASK, - osr_dac_sel[osr].clk_src << NAU8821_CLK_DAC_SRC_SFT); - } else { - regmap_read(nau8821->regmap, NAU8821_R2B_ADC_RATE, &osr); - osr &= NAU8821_ADC_SYNC_DOWN_MASK; - if (nau8821_clock_check(nau8821, substream->stream, - nau8821->fs, osr)) { - return -EINVAL; - } + osr->clk_src << NAU8821_CLK_DAC_SRC_SFT); + else regmap_update_bits(nau8821->regmap, NAU8821_R03_CLK_DIVIDER, NAU8821_CLK_ADC_SRC_MASK, - osr_adc_sel[osr].clk_src << NAU8821_CLK_ADC_SRC_SFT); - } + osr->clk_src << NAU8821_CLK_ADC_SRC_SFT); /* make BCLK and LRC divde configuration if the codec as master. */ regmap_read(nau8821->regmap, NAU8821_R1D_I2S_PCM_CTRL2, &ctrl_val); @@ -843,6 +848,7 @@ static int nau8821_digital_mute(struct snd_soc_dai *dai, int mute, } static const struct snd_soc_dai_ops nau8821_dai_ops = { + .startup = nau8821_dai_startup, .hw_params = nau8821_hw_params, .set_fmt = nau8821_set_dai_fmt, .mute_stream = nau8821_digital_mute, From 5628560e90395d3812800a8e44a01c32ffa429ec Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 23 Aug 2022 10:09:57 +0200 Subject: [PATCH 07/29] ASoC: nau8824: Fix semaphore unbalance at error paths The semaphore of nau8824 wasn't properly unlocked at some error handling code paths, hence this may result in the unbalance (and potential lock-up). Fix them to handle the semaphore up properly. Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/20220823081000.2965-3-tiwai@suse.de Signed-off-by: Mark Brown --- sound/soc/codecs/nau8824.c | 17 ++++++++++------- 1 file changed, 10 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/nau8824.c b/sound/soc/codecs/nau8824.c index ad54d70f7d8e..10bdfebe92d5 100644 --- a/sound/soc/codecs/nau8824.c +++ b/sound/soc/codecs/nau8824.c @@ -1043,6 +1043,7 @@ static int nau8824_hw_params(struct snd_pcm_substream *substream, struct snd_soc_component *component = dai->component; struct nau8824 *nau8824 = snd_soc_component_get_drvdata(component); unsigned int val_len = 0, osr, ctrl_val, bclk_fs, bclk_div; + int err = -EINVAL; nau8824_sema_acquire(nau8824, HZ); @@ -1059,7 +1060,7 @@ static int nau8824_hw_params(struct snd_pcm_substream *substream, osr &= NAU8824_DAC_OVERSAMPLE_MASK; if (nau8824_clock_check(nau8824, substream->stream, nau8824->fs, osr)) - return -EINVAL; + goto error; regmap_update_bits(nau8824->regmap, NAU8824_REG_CLK_DIVIDER, NAU8824_CLK_DAC_SRC_MASK, osr_dac_sel[osr].clk_src << NAU8824_CLK_DAC_SRC_SFT); @@ -1069,7 +1070,7 @@ static int nau8824_hw_params(struct snd_pcm_substream *substream, osr &= NAU8824_ADC_SYNC_DOWN_MASK; if (nau8824_clock_check(nau8824, substream->stream, nau8824->fs, osr)) - return -EINVAL; + goto error; regmap_update_bits(nau8824->regmap, NAU8824_REG_CLK_DIVIDER, NAU8824_CLK_ADC_SRC_MASK, osr_adc_sel[osr].clk_src << NAU8824_CLK_ADC_SRC_SFT); @@ -1090,7 +1091,7 @@ static int nau8824_hw_params(struct snd_pcm_substream *substream, else if (bclk_fs <= 256) bclk_div = 0; else - return -EINVAL; + goto error; regmap_update_bits(nau8824->regmap, NAU8824_REG_PORT0_I2S_PCM_CTRL_2, NAU8824_I2S_LRC_DIV_MASK | NAU8824_I2S_BLK_DIV_MASK, @@ -1111,15 +1112,17 @@ static int nau8824_hw_params(struct snd_pcm_substream *substream, val_len |= NAU8824_I2S_DL_32; break; default: - return -EINVAL; + goto error; } regmap_update_bits(nau8824->regmap, NAU8824_REG_PORT0_I2S_PCM_CTRL_1, NAU8824_I2S_DL_MASK, val_len); + err = 0; + error: nau8824_sema_release(nau8824); - return 0; + return err; } static int nau8824_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) @@ -1128,8 +1131,6 @@ static int nau8824_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) struct nau8824 *nau8824 = snd_soc_component_get_drvdata(component); unsigned int ctrl1_val = 0, ctrl2_val = 0; - nau8824_sema_acquire(nau8824, HZ); - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBM_CFM: ctrl2_val |= NAU8824_I2S_MS_MASTER; @@ -1171,6 +1172,8 @@ static int nau8824_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) return -EINVAL; } + nau8824_sema_acquire(nau8824, HZ); + regmap_update_bits(nau8824->regmap, NAU8824_REG_PORT0_I2S_PCM_CTRL_1, NAU8824_I2S_DF_MASK | NAU8824_I2S_BP_MASK | NAU8824_I2S_PCMB_EN, ctrl1_val); From 92283c86260d8712b55f97eada13b3c8b2f469b2 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 23 Aug 2022 10:09:58 +0200 Subject: [PATCH 08/29] ASoC: nau8824: Implement hw constraint for rates nau8824 driver restricts the sample rate with over sampling rate, but currently it barely bails out at hw_params with -EINVAL error (with a kernel message); this doesn't help for user-space to recognize which rate can be actually used. This patch introduces the proper hw constraint for adjusting the available range of the sample rate depending on the OSR setup, as well as some code cleanup, for improving the communication with user-space. Now applications can know the valid rate beforehand and reduces the rate appropriately without errors. Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/20220823081000.2965-4-tiwai@suse.de Signed-off-by: Mark Brown --- sound/soc/codecs/nau8824.c | 67 +++++++++++++++++++++----------------- 1 file changed, 38 insertions(+), 29 deletions(-) diff --git a/sound/soc/codecs/nau8824.c b/sound/soc/codecs/nau8824.c index 10bdfebe92d5..15596452ca37 100644 --- a/sound/soc/codecs/nau8824.c +++ b/sound/soc/codecs/nau8824.c @@ -1014,27 +1014,42 @@ static irqreturn_t nau8824_interrupt(int irq, void *data) return IRQ_HANDLED; } -static int nau8824_clock_check(struct nau8824 *nau8824, - int stream, int rate, int osr) +static const struct nau8824_osr_attr * +nau8824_get_osr(struct nau8824 *nau8824, int stream) { - int osrate; + unsigned int osr; if (stream == SNDRV_PCM_STREAM_PLAYBACK) { + regmap_read(nau8824->regmap, + NAU8824_REG_DAC_FILTER_CTRL_1, &osr); + osr &= NAU8824_DAC_OVERSAMPLE_MASK; if (osr >= ARRAY_SIZE(osr_dac_sel)) - return -EINVAL; - osrate = osr_dac_sel[osr].osr; + return NULL; + return &osr_dac_sel[osr]; } else { + regmap_read(nau8824->regmap, + NAU8824_REG_ADC_FILTER_CTRL, &osr); + osr &= NAU8824_ADC_SYNC_DOWN_MASK; if (osr >= ARRAY_SIZE(osr_adc_sel)) - return -EINVAL; - osrate = osr_adc_sel[osr].osr; + return NULL; + return &osr_adc_sel[osr]; } +} - if (!osrate || rate * osr > CLK_DA_AD_MAX) { - dev_err(nau8824->dev, "exceed the maximum frequency of CLK_ADC or CLK_DAC\n"); +static int nau8824_dai_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct nau8824 *nau8824 = snd_soc_component_get_drvdata(component); + const struct nau8824_osr_attr *osr; + + osr = nau8824_get_osr(nau8824, substream->stream); + if (!osr || !osr->osr) return -EINVAL; - } - return 0; + return snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_RATE, + 0, CLK_DA_AD_MAX / osr->osr); } static int nau8824_hw_params(struct snd_pcm_substream *substream, @@ -1042,7 +1057,8 @@ static int nau8824_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_component *component = dai->component; struct nau8824 *nau8824 = snd_soc_component_get_drvdata(component); - unsigned int val_len = 0, osr, ctrl_val, bclk_fs, bclk_div; + unsigned int val_len = 0, ctrl_val, bclk_fs, bclk_div; + const struct nau8824_osr_attr *osr; int err = -EINVAL; nau8824_sema_acquire(nau8824, HZ); @@ -1054,27 +1070,19 @@ static int nau8824_hw_params(struct snd_pcm_substream *substream, * than 6.144 MHz. */ nau8824->fs = params_rate(params); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - regmap_read(nau8824->regmap, - NAU8824_REG_DAC_FILTER_CTRL_1, &osr); - osr &= NAU8824_DAC_OVERSAMPLE_MASK; - if (nau8824_clock_check(nau8824, substream->stream, - nau8824->fs, osr)) - goto error; + osr = nau8824_get_osr(nau8824, substream->stream); + if (!osr || !osr->osr) + goto error; + if (nau8824->fs * osr->osr > CLK_DA_AD_MAX) + goto error; + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) regmap_update_bits(nau8824->regmap, NAU8824_REG_CLK_DIVIDER, NAU8824_CLK_DAC_SRC_MASK, - osr_dac_sel[osr].clk_src << NAU8824_CLK_DAC_SRC_SFT); - } else { - regmap_read(nau8824->regmap, - NAU8824_REG_ADC_FILTER_CTRL, &osr); - osr &= NAU8824_ADC_SYNC_DOWN_MASK; - if (nau8824_clock_check(nau8824, substream->stream, - nau8824->fs, osr)) - goto error; + osr->clk_src << NAU8824_CLK_DAC_SRC_SFT); + else regmap_update_bits(nau8824->regmap, NAU8824_REG_CLK_DIVIDER, NAU8824_CLK_ADC_SRC_MASK, - osr_adc_sel[osr].clk_src << NAU8824_CLK_ADC_SRC_SFT); - } + osr->clk_src << NAU8824_CLK_ADC_SRC_SFT); /* make BCLK and LRC divde configuration if the codec as master. */ regmap_read(nau8824->regmap, @@ -1550,6 +1558,7 @@ static const struct snd_soc_component_driver nau8824_component_driver = { }; static const struct snd_soc_dai_ops nau8824_dai_ops = { + .startup = nau8824_dai_startup, .hw_params = nau8824_hw_params, .set_fmt = nau8824_set_fmt, .set_tdm_slot = nau8824_set_tdm_slot, From bed41de0f679c516de45cfeb2c40c412bc5e0c0b Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 23 Aug 2022 10:09:59 +0200 Subject: [PATCH 09/29] ASoC: nau8825: Implement hw constraint for rates nau8825 driver restricts the sample rate with over sampling rate, but currently it barely bails out at hw_params with -EINVAL error (with a kernel message); this doesn't help for user-space to recognize which rate can be actually used. This patch introduces the proper hw constraint for adjusting the available range of the sample rate depending on the OSR setup, as well as some code cleanup, for improving the communication with user-space. Now applications can know the valid rate beforehand and reduces the rate appropriately without errors. Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/20220823081000.2965-5-tiwai@suse.de Signed-off-by: Mark Brown --- sound/soc/codecs/nau8825.c | 83 +++++++++++++++++++++----------------- 1 file changed, 45 insertions(+), 38 deletions(-) diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index 54ef7b0fa878..8213273f501e 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -1247,27 +1247,42 @@ static const struct snd_soc_dapm_route nau8825_dapm_routes[] = { {"HPOR", NULL, "Class G"}, }; -static int nau8825_clock_check(struct nau8825 *nau8825, - int stream, int rate, int osr) +static const struct nau8825_osr_attr * +nau8825_get_osr(struct nau8825 *nau8825, int stream) { - int osrate; + unsigned int osr; if (stream == SNDRV_PCM_STREAM_PLAYBACK) { + regmap_read(nau8825->regmap, + NAU8825_REG_DAC_CTRL1, &osr); + osr &= NAU8825_DAC_OVERSAMPLE_MASK; if (osr >= ARRAY_SIZE(osr_dac_sel)) - return -EINVAL; - osrate = osr_dac_sel[osr].osr; + return NULL; + return &osr_dac_sel[osr]; } else { + regmap_read(nau8825->regmap, + NAU8825_REG_ADC_RATE, &osr); + osr &= NAU8825_ADC_SYNC_DOWN_MASK; if (osr >= ARRAY_SIZE(osr_adc_sel)) - return -EINVAL; - osrate = osr_adc_sel[osr].osr; + return NULL; + return &osr_adc_sel[osr]; } +} - if (!osrate || rate * osr > CLK_DA_AD_MAX) { - dev_err(nau8825->dev, "exceed the maximum frequency of CLK_ADC or CLK_DAC\n"); +static int nau8825_dai_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct nau8825 *nau8825 = snd_soc_component_get_drvdata(component); + const struct nau8825_osr_attr *osr; + + osr = nau8825_get_osr(nau8825, substream->stream); + if (!osr || !osr->osr) return -EINVAL; - } - return 0; + return snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_RATE, + 0, CLK_DA_AD_MAX / osr->osr); } static int nau8825_hw_params(struct snd_pcm_substream *substream, @@ -1276,7 +1291,9 @@ static int nau8825_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_component *component = dai->component; struct nau8825 *nau8825 = snd_soc_component_get_drvdata(component); - unsigned int val_len = 0, osr, ctrl_val, bclk_fs, bclk_div; + unsigned int val_len = 0, ctrl_val, bclk_fs, bclk_div; + const struct nau8825_osr_attr *osr; + int err = -EINVAL; nau8825_sema_acquire(nau8825, 3 * HZ); @@ -1286,29 +1303,19 @@ static int nau8825_hw_params(struct snd_pcm_substream *substream, * values must be selected such that the maximum frequency is less * than 6.144 MHz. */ - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - regmap_read(nau8825->regmap, NAU8825_REG_DAC_CTRL1, &osr); - osr &= NAU8825_DAC_OVERSAMPLE_MASK; - if (nau8825_clock_check(nau8825, substream->stream, - params_rate(params), osr)) { - nau8825_sema_release(nau8825); - return -EINVAL; - } + osr = nau8825_get_osr(nau8825, substream->stream); + if (!osr || !osr->osr) + goto error; + if (params_rate(params) * osr->osr > CLK_DA_AD_MAX) + goto error; + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) regmap_update_bits(nau8825->regmap, NAU8825_REG_CLK_DIVIDER, NAU8825_CLK_DAC_SRC_MASK, - osr_dac_sel[osr].clk_src << NAU8825_CLK_DAC_SRC_SFT); - } else { - regmap_read(nau8825->regmap, NAU8825_REG_ADC_RATE, &osr); - osr &= NAU8825_ADC_SYNC_DOWN_MASK; - if (nau8825_clock_check(nau8825, substream->stream, - params_rate(params), osr)) { - nau8825_sema_release(nau8825); - return -EINVAL; - } + osr->clk_src << NAU8825_CLK_DAC_SRC_SFT); + else regmap_update_bits(nau8825->regmap, NAU8825_REG_CLK_DIVIDER, NAU8825_CLK_ADC_SRC_MASK, - osr_adc_sel[osr].clk_src << NAU8825_CLK_ADC_SRC_SFT); - } + osr->clk_src << NAU8825_CLK_ADC_SRC_SFT); /* make BCLK and LRC divde configuration if the codec as master. */ regmap_read(nau8825->regmap, NAU8825_REG_I2S_PCM_CTRL2, &ctrl_val); @@ -1321,10 +1328,8 @@ static int nau8825_hw_params(struct snd_pcm_substream *substream, bclk_div = 1; else if (bclk_fs <= 128) bclk_div = 0; - else { - nau8825_sema_release(nau8825); - return -EINVAL; - } + else + goto error; regmap_update_bits(nau8825->regmap, NAU8825_REG_I2S_PCM_CTRL2, NAU8825_I2S_LRC_DIV_MASK | NAU8825_I2S_BLK_DIV_MASK, ((bclk_div + 1) << NAU8825_I2S_LRC_DIV_SFT) | bclk_div); @@ -1344,17 +1349,18 @@ static int nau8825_hw_params(struct snd_pcm_substream *substream, val_len |= NAU8825_I2S_DL_32; break; default: - nau8825_sema_release(nau8825); - return -EINVAL; + goto error; } regmap_update_bits(nau8825->regmap, NAU8825_REG_I2S_PCM_CTRL1, NAU8825_I2S_DL_MASK, val_len); + err = 0; + error: /* Release the semaphore. */ nau8825_sema_release(nau8825); - return 0; + return err; } static int nau8825_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) @@ -1420,6 +1426,7 @@ static int nau8825_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) } static const struct snd_soc_dai_ops nau8825_dai_ops = { + .startup = nau8825_dai_startup, .hw_params = nau8825_hw_params, .set_fmt = nau8825_set_dai_fmt, }; From be919239fbcab19290bfd6802c7ad1dc946c515b Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 23 Aug 2022 10:10:00 +0200 Subject: [PATCH 10/29] ASoC: nau8540: Implement hw constraint for rates nau8540 driver restricts the sample rate with over sampling rate, but currently it barely bails out at hw_params with -EINVAL error (with a kernel message); this doesn't help for user-space to recognize which rate can be actually used. This patch introduces the proper hw constraint for adjusting the available range of the sample rate depending on the OSR setup, as well as some code cleanup, for improving the communication with user-space. Now applications can know the valid rate beforehand and reduces the rate appropriately without errors. Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/20220823081000.2965-6-tiwai@suse.de Signed-off-by: Mark Brown --- sound/soc/codecs/nau8540.c | 42 +++++++++++++++++++++++++++----------- 1 file changed, 30 insertions(+), 12 deletions(-) diff --git a/sound/soc/codecs/nau8540.c b/sound/soc/codecs/nau8540.c index 58f70a02f18a..0626d5694c22 100644 --- a/sound/soc/codecs/nau8540.c +++ b/sound/soc/codecs/nau8540.c @@ -357,17 +357,32 @@ static const struct snd_soc_dapm_route nau8540_dapm_routes[] = { {"AIFTX", NULL, "Digital CH4 Mux"}, }; -static int nau8540_clock_check(struct nau8540 *nau8540, int rate, int osr) +static const struct nau8540_osr_attr * +nau8540_get_osr(struct nau8540 *nau8540) { + unsigned int osr; + + regmap_read(nau8540->regmap, NAU8540_REG_ADC_SAMPLE_RATE, &osr); + osr &= NAU8540_ADC_OSR_MASK; if (osr >= ARRAY_SIZE(osr_adc_sel)) + return NULL; + return &osr_adc_sel[osr]; +} + +static int nau8540_dai_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct nau8540 *nau8540 = snd_soc_component_get_drvdata(component); + const struct nau8540_osr_attr *osr; + + osr = nau8540_get_osr(nau8540); + if (!osr || !osr->osr) return -EINVAL; - if (rate * osr > CLK_ADC_MAX) { - dev_err(nau8540->dev, "exceed the maximum frequency of CLK_ADC\n"); - return -EINVAL; - } - - return 0; + return snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_RATE, + 0, CLK_ADC_MAX / osr->osr); } static int nau8540_hw_params(struct snd_pcm_substream *substream, @@ -375,7 +390,8 @@ static int nau8540_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_component *component = dai->component; struct nau8540 *nau8540 = snd_soc_component_get_drvdata(component); - unsigned int val_len = 0, osr; + unsigned int val_len = 0; + const struct nau8540_osr_attr *osr; /* CLK_ADC = OSR * FS * ADC clock frequency is defined as Over Sampling Rate (OSR) @@ -383,13 +399,14 @@ static int nau8540_hw_params(struct snd_pcm_substream *substream, * values must be selected such that the maximum frequency is less * than 6.144 MHz. */ - regmap_read(nau8540->regmap, NAU8540_REG_ADC_SAMPLE_RATE, &osr); - osr &= NAU8540_ADC_OSR_MASK; - if (nau8540_clock_check(nau8540, params_rate(params), osr)) + osr = nau8540_get_osr(nau8540); + if (!osr || !osr->osr) + return -EINVAL; + if (params_rate(params) * osr->osr > CLK_ADC_MAX) return -EINVAL; regmap_update_bits(nau8540->regmap, NAU8540_REG_CLOCK_SRC, NAU8540_CLK_ADC_SRC_MASK, - osr_adc_sel[osr].clk_src << NAU8540_CLK_ADC_SRC_SFT); + osr->clk_src << NAU8540_CLK_ADC_SRC_SFT); switch (params_width(params)) { case 16: @@ -515,6 +532,7 @@ static int nau8540_set_tdm_slot(struct snd_soc_dai *dai, static const struct snd_soc_dai_ops nau8540_dai_ops = { + .startup = nau8540_dai_startup, .hw_params = nau8540_hw_params, .set_fmt = nau8540_set_fmt, .set_tdm_slot = nau8540_set_tdm_slot, From 1faa6f8274e2b08a38c0cca74113dfb26c6ad7b7 Mon Sep 17 00:00:00 2001 From: Shengjiu Wang Date: Tue, 23 Aug 2022 17:35:08 +0800 Subject: [PATCH 11/29] ASoC: fsl_mqs: Fix supported clock DAI format The MQS works as codec DAI, not cpu DAI. It is clock consumer, not clock privider. Fixes: 3b14c15a333b ("ASoC: fsl: Update to use set_fmt_new callback") Signed-off-by: Shengjiu Wang Link: https://lore.kernel.org/r/1661247308-2650-1-git-send-email-shengjiu.wang@nxp.com Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_mqs.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_mqs.c b/sound/soc/fsl/fsl_mqs.c index c1e2f671191b..4922e6795b73 100644 --- a/sound/soc/fsl/fsl_mqs.c +++ b/sound/soc/fsl/fsl_mqs.c @@ -122,7 +122,7 @@ static int fsl_mqs_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) } switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { - case SND_SOC_DAIFMT_BP_FP: + case SND_SOC_DAIFMT_CBC_CFC: break; default: return -EINVAL; From 3942499fba11de048c3ac1390b808e9e6ae88de5 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Tue, 23 Aug 2022 15:15:53 +0300 Subject: [PATCH 12/29] ASoC: SOF: Kconfig: Make IPC_FLOOD_TEST depend on SND_SOC_SOF Make sure that the IPC_FLOOD client can not be built in when SND_SOC_SOF is built as module. Fixes: 6e9548cdb30e5 ("ASoC: SOF: Convert the generic IPC flood test into SOF client") Reported-by: kernel test robot Signed-off-by: Peter Ujfalusi Reviewed-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20220823121554.4255-2-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/sof/Kconfig b/sound/soc/sof/Kconfig index e90f173d067c..cfb244e4e142 100644 --- a/sound/soc/sof/Kconfig +++ b/sound/soc/sof/Kconfig @@ -196,6 +196,7 @@ config SND_SOC_SOF_DEBUG_ENABLE_FIRMWARE_TRACE config SND_SOC_SOF_DEBUG_IPC_FLOOD_TEST tristate "SOF enable IPC flood test" + depends on SND_SOC_SOF select SND_SOC_SOF_CLIENT help This option enables a separate client device for IPC flood test From 2cf520ffbcbd55c0f2b4276065444d7526d9d197 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Tue, 23 Aug 2022 15:15:54 +0300 Subject: [PATCH 13/29] ASoC: SOF: Kconfig: Make IPC_MESSAGE_INJECTOR depend on SND_SOC_SOF Make sure that the IPC_MESSAGE_INJECTOR client can not be built in when SND_SOC_SOF is built as module. Fixes: cac0b0887e530 ("ASoC: SOF: Convert the generic IPC message injector into SOF client") Reported-by: kernel test robot Signed-off-by: Peter Ujfalusi Reviewed-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20220823121554.4255-3-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/sof/Kconfig b/sound/soc/sof/Kconfig index cfb244e4e142..37f7df5fde17 100644 --- a/sound/soc/sof/Kconfig +++ b/sound/soc/sof/Kconfig @@ -215,6 +215,7 @@ config SND_SOC_SOF_DEBUG_IPC_FLOOD_TEST_NUM config SND_SOC_SOF_DEBUG_IPC_MSG_INJECTOR tristate "SOF enable IPC message injector" + depends on SND_SOC_SOF select SND_SOC_SOF_CLIENT help This option enables the IPC message injector which can be used to send From 5c5c2baad2b55cc0a4b190266889959642298f79 Mon Sep 17 00:00:00 2001 From: Nathan Chancellor Date: Tue, 9 Aug 2022 18:08:09 -0700 Subject: [PATCH 14/29] ASoC: mchp-spdiftx: Fix clang -Wbitfield-constant-conversion A recent change in clang strengthened its -Wbitfield-constant-conversion to warn when 1 is assigned to a 1-bit signed integer bitfield, as it can only be 0 or -1, not 1: sound/soc/atmel/mchp-spdiftx.c:505:20: error: implicit truncation from 'int' to bit-field changes value from 1 to -1 [-Werror,-Wbitfield-constant-conversion] dev->gclk_enabled = 1; ^ ~ 1 error generated. The actual value of the field is never checked, just that it is not zero, so there is not a real bug here. However, it is simple enough to silence the warning by making the bitfield unsigned, which matches the mchp-spdifrx driver. Fixes: 06ca24e98e6b ("ASoC: mchp-spdiftx: add driver for S/PDIF TX Controller") Link: https://github.com/ClangBuiltLinux/linux/issues/1686 Link: https://github.com/llvm/llvm-project/commit/82afc9b169a67e8b8a1862fb9c41a2cd974d6691 Signed-off-by: Nathan Chancellor Reviewed-by: Nick Desaulniers Link: https://lore.kernel.org/r/20220810010809.2024482-1-nathan@kernel.org Signed-off-by: Mark Brown --- sound/soc/atmel/mchp-spdiftx.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/atmel/mchp-spdiftx.c b/sound/soc/atmel/mchp-spdiftx.c index 4850a177803d..ab2d7a791f39 100644 --- a/sound/soc/atmel/mchp-spdiftx.c +++ b/sound/soc/atmel/mchp-spdiftx.c @@ -196,7 +196,7 @@ struct mchp_spdiftx_dev { struct clk *pclk; struct clk *gclk; unsigned int fmt; - int gclk_enabled:1; + unsigned int gclk_enabled:1; }; static inline int mchp_spdiftx_is_running(struct mchp_spdiftx_dev *dev) From 4ee6fc271b59e805301371ea3862f558a23d9c7b Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Mon, 22 Aug 2022 21:02:11 +0200 Subject: [PATCH 15/29] ASoC: SOF: ipc4-topology: fix alh_group_ida max value group_id is from 0 ~ ALH_MULTI_GTW_COUNT - 1, not 0 ~ ALH_MULTI_GTW_COUNT. Fixes: a150345aa7584 ("ASoC: SOF: ipc4-topology: add SoundWire/ALH aggregation support") Reported-by: kernel test robot Reported-by: Dan Carpenter Reviewed-by: Rander Wang Signed-off-by: Bard Liao Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20220822190211.170537-1-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/ipc4-topology.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sof/ipc4-topology.c b/sound/soc/sof/ipc4-topology.c index c6abfaf5d532..64929dc9af39 100644 --- a/sound/soc/sof/ipc4-topology.c +++ b/sound/soc/sof/ipc4-topology.c @@ -1251,7 +1251,7 @@ sof_ipc4_prepare_copier_module(struct snd_sof_widget *swidget, if (blob->alh_cfg.count > 1) { int group_id; - group_id = ida_alloc_max(&alh_group_ida, ALH_MULTI_GTW_COUNT, + group_id = ida_alloc_max(&alh_group_ida, ALH_MULTI_GTW_COUNT - 1, GFP_KERNEL); if (group_id < 0) From ea532c29972df96fda20393d9bf057e898f5e965 Mon Sep 17 00:00:00 2001 From: Shengjiu Wang Date: Thu, 25 Aug 2022 20:27:39 +0800 Subject: [PATCH 16/29] ASoC: fsl_aud2htx: register platform component before registering cpu dai There is no defer probe when adding platform component to snd_soc_pcm_runtime(rtd), the code is in snd_soc_add_pcm_runtime() snd_soc_register_card() -> snd_soc_bind_card() -> snd_soc_add_pcm_runtime() -> adding cpu dai -> adding codec dai -> adding platform component. So if the platform component is not ready at that time, then the sound card still registered successfully, but platform component is empty, the sound card can't be used. As there is defer probe checking for cpu dai component, then register platform component before cpu dai to avoid such issue. And the behavior of imx_pcm_dma_init() is same as common devm_snd_dmaengine_pcm_register(), so use devm_snd_dmaengine_pcm_register() instead Signed-off-by: Shengjiu Wang Link: https://lore.kernel.org/r/1661430460-5234-1-git-send-email-shengjiu.wang@nxp.com Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_aud2htx.c | 14 ++++++++++---- 1 file changed, 10 insertions(+), 4 deletions(-) diff --git a/sound/soc/fsl/fsl_aud2htx.c b/sound/soc/fsl/fsl_aud2htx.c index 873295f59ad7..bc1b9c6df95c 100644 --- a/sound/soc/fsl/fsl_aud2htx.c +++ b/sound/soc/fsl/fsl_aud2htx.c @@ -234,6 +234,16 @@ static int fsl_aud2htx_probe(struct platform_device *pdev) regcache_cache_only(aud2htx->regmap, true); + /* + * Register platform component before registering cpu dai for there + * is not defer probe for platform component in snd_soc_add_pcm_runtime(). + */ + ret = devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0); + if (ret) { + dev_err(&pdev->dev, "failed to pcm register\n"); + return ret; + } + ret = devm_snd_soc_register_component(&pdev->dev, &fsl_aud2htx_component, &fsl_aud2htx_dai, 1); @@ -242,10 +252,6 @@ static int fsl_aud2htx_probe(struct platform_device *pdev) return ret; } - ret = imx_pcm_dma_init(pdev); - if (ret) - dev_err(&pdev->dev, "failed to init imx pcm dma: %d\n", ret); - return ret; } From b1cd3fd42db7593a2d24c06f1c53b8c886592080 Mon Sep 17 00:00:00 2001 From: Shengjiu Wang Date: Thu, 25 Aug 2022 20:27:40 +0800 Subject: [PATCH 17/29] ASoC: fsl_aud2htx: Add error handler for pm_runtime_enable Call pm_runtime_disable() when error happens in probe() Signed-off-by: Shengjiu Wang Link: https://lore.kernel.org/r/1661430460-5234-2-git-send-email-shengjiu.wang@nxp.com Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_aud2htx.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/fsl/fsl_aud2htx.c b/sound/soc/fsl/fsl_aud2htx.c index bc1b9c6df95c..1e421d9a03fb 100644 --- a/sound/soc/fsl/fsl_aud2htx.c +++ b/sound/soc/fsl/fsl_aud2htx.c @@ -241,6 +241,7 @@ static int fsl_aud2htx_probe(struct platform_device *pdev) ret = devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0); if (ret) { dev_err(&pdev->dev, "failed to pcm register\n"); + pm_runtime_disable(&pdev->dev); return ret; } @@ -249,6 +250,7 @@ static int fsl_aud2htx_probe(struct platform_device *pdev) &fsl_aud2htx_dai, 1); if (ret) { dev_err(&pdev->dev, "failed to register ASoC DAI\n"); + pm_runtime_disable(&pdev->dev); return ret; } From 7e1afce5866e02b45bf88c27dd7de1b9dfade1cc Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 31 Aug 2022 14:59:00 +0200 Subject: [PATCH 18/29] ALSA: usb-audio: Inform the delayed registration more properly The info message that was added in the commit a4aad5636c72 ("ALSA: usb-audio: Inform devices that need delayed registration") is actually useful to know the need for the delayed registration. However, it turned out that this doesn't catch the all cases; namely, this warned only when a PCM stream is attached onto the existing PCM instance, but it doesn't count for a newly created PCM instance. This made confusion as if there were no further delayed registration. This patch moves the check to the code path for either adding a stream or creating a PCM instance. Also, make it simpler by checking the card->registered flag instead of querying each snd_device state. Fixes: a4aad5636c72 ("ALSA: usb-audio: Inform devices that need delayed registration") Link: https://bugzilla.kernel.org/show_bug.cgi?id=216082 Link: https://lore.kernel.org/r/20220831125901.4660-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/usb/stream.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/usb/stream.c b/sound/usb/stream.c index ceb93d798182..40b7821c6c99 100644 --- a/sound/usb/stream.c +++ b/sound/usb/stream.c @@ -495,6 +495,10 @@ static int __snd_usb_add_audio_stream(struct snd_usb_audio *chip, return 0; } } + + if (chip->card->registered) + chip->need_delayed_register = true; + /* look for an empty stream */ list_for_each_entry(as, &chip->pcm_list, list) { if (as->fmt_type != fp->fmt_type) @@ -502,9 +506,6 @@ static int __snd_usb_add_audio_stream(struct snd_usb_audio *chip, subs = &as->substream[stream]; if (subs->ep_num) continue; - if (snd_device_get_state(chip->card, as->pcm) != - SNDRV_DEV_BUILD) - chip->need_delayed_register = true; err = snd_pcm_new_stream(as->pcm, stream, 1); if (err < 0) return err; From 2027f114686e0f3f1f39971964dfc618637c88c2 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 31 Aug 2022 14:59:01 +0200 Subject: [PATCH 19/29] ALSA: usb-audio: Register card again for iface over delayed_register option When the delayed registration is specified via either delayed_register option or the quirk, we delay the invocation of snd_card_register() until the given interface. But if a wrong value has been set there and there are more interfaces over the given interface number, snd_card_register() call would be missing for those interfaces. This patch catches up those missing calls by fixing the comparison of the interface number. Now the call is skipped only if the processed interface is less than the given interface, instead of the exact match. Fixes: b70038ef4fea ("ALSA: usb-audio: Add delayed_register option") Link: https://bugzilla.kernel.org/show_bug.cgi?id=216082 Link: https://lore.kernel.org/r/20220831125901.4660-2-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/usb/card.c | 2 +- sound/usb/quirks.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/usb/card.c b/sound/usb/card.c index d356743de2ff..706d249a9ad6 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -699,7 +699,7 @@ static bool check_delayed_register_option(struct snd_usb_audio *chip, int iface) if (delayed_register[i] && sscanf(delayed_register[i], "%x:%x", &id, &inum) == 2 && id == chip->usb_id) - return inum != iface; + return iface < inum; } return false; diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 9bfead5efc4c..5b4d8f5eade2 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1764,7 +1764,7 @@ bool snd_usb_registration_quirk(struct snd_usb_audio *chip, int iface) for (q = registration_quirks; q->usb_id; q++) if (chip->usb_id == q->usb_id) - return iface != q->interface; + return iface < q->interface; /* Register as normal */ return false; From ff878b408a03bef5d610b7e2302702e16a53636e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 1 Sep 2022 14:41:36 +0200 Subject: [PATCH 20/29] ALSA: usb-audio: Split endpoint setups for hw_params and prepare One of the former changes for the endpoint management was the more consistent setup of endpoints at hw_params. snd_usb_endpoint_configure() is a single function that does the full setup, and it's called from both PCM hw_params and prepare callbacks. Although the EP setup at the prepare phase is usually skipped (by checking need_setup flag), it may be still effective in some cases like suspend/resume that requires the interface setup again. As it's a full and single setup, the invocation of snd_usb_endpoint_configure() includes not only the USB interface setup but also the buffer release and allocation. OTOH, doing the buffer release and re-allocation at PCM prepare phase is rather superfluous, and better to be done only in the hw_params phase. For those optimizations, this patch splits the endpoint setup to two phases: snd_usb_endpoint_set_params() and snd_usb_endpoint_prepare(), to be called from hw_params and from prepare, respectively. Note that this patch changes the driver operation slightly, effectively moving the USB interface setup again to PCM prepare stage instead of hw_params stage, while the buffer allocation and such initializations are still done at hw_params stage. And, the change of the USB interface setup timing (moving to prepare) gave an interesting "fix", too: it was reported that the recent kernels caused silent output at the beginning on playbacks on some devices on Android, and this change casually fixed the regression. It seems that those devices are picky about the sample rate change (or the interface change?), and don't follow the too immediate rate changes. Meanwhile, Android operates the PCM in the following order: - open, then hw_params with the possibly highest sample rate - close without prepare - re-open, hw_params with the normal sample rate - prepare, and start streaming This procedure ended up the hw_params twice with different rates, and because the recent kernel did set up the sample rate twice one and after, it screwed up the device. OTOH, the earlier kernels didn't set up the USB interface at hw_params, hence this problem didn't appear. Now, with this patch, the USB interface setup is again back to the prepare phase, and it works around the problem automagically. Although we should address the sample rate problem in a more solid way in future, let's keep things working as before for now. Fixes: bf6313a0ff76 ("ALSA: usb-audio: Refactor endpoint management") Cc: Reported-by: chihhao chen Link: https://lore.kernel.org/r/87e6d6ae69d68dc588ac9acc8c0f24d6188375c3.camel@mediatek.com Link: https://lore.kernel.org/r/20220901124136.4984-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/usb/endpoint.c | 23 +++++++++-------------- sound/usb/endpoint.h | 6 ++++-- sound/usb/pcm.c | 14 ++++++++++---- 3 files changed, 23 insertions(+), 20 deletions(-) diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index 0d7b73bf7945..a42f2ce19455 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -758,7 +758,8 @@ bool snd_usb_endpoint_compatible(struct snd_usb_audio *chip, * The endpoint needs to be closed via snd_usb_endpoint_close() later. * * Note that this function doesn't configure the endpoint. The substream - * needs to set it up later via snd_usb_endpoint_configure(). + * needs to set it up later via snd_usb_endpoint_set_params() and + * snd_usb_endpoint_prepare(). */ struct snd_usb_endpoint * snd_usb_endpoint_open(struct snd_usb_audio *chip, @@ -1290,12 +1291,13 @@ out_of_memory: /* * snd_usb_endpoint_set_params: configure an snd_usb_endpoint * + * It's called either from hw_params callback. * Determine the number of URBs to be used on this endpoint. * An endpoint must be configured before it can be started. * An endpoint that is already running can not be reconfigured. */ -static int snd_usb_endpoint_set_params(struct snd_usb_audio *chip, - struct snd_usb_endpoint *ep) +int snd_usb_endpoint_set_params(struct snd_usb_audio *chip, + struct snd_usb_endpoint *ep) { const struct audioformat *fmt = ep->cur_audiofmt; int err; @@ -1378,18 +1380,18 @@ static int init_sample_rate(struct snd_usb_audio *chip, } /* - * snd_usb_endpoint_configure: Configure the endpoint + * snd_usb_endpoint_prepare: Prepare the endpoint * * This function sets up the EP to be fully usable state. - * It's called either from hw_params or prepare callback. + * It's called either from prepare callback. * The function checks need_setup flag, and performs nothing unless needed, * so it's safe to call this multiple times. * * This returns zero if unchanged, 1 if the configuration has changed, * or a negative error code. */ -int snd_usb_endpoint_configure(struct snd_usb_audio *chip, - struct snd_usb_endpoint *ep) +int snd_usb_endpoint_prepare(struct snd_usb_audio *chip, + struct snd_usb_endpoint *ep) { bool iface_first; int err = 0; @@ -1410,9 +1412,6 @@ int snd_usb_endpoint_configure(struct snd_usb_audio *chip, if (err < 0) goto unlock; } - err = snd_usb_endpoint_set_params(chip, ep); - if (err < 0) - goto unlock; goto done; } @@ -1440,10 +1439,6 @@ int snd_usb_endpoint_configure(struct snd_usb_audio *chip, if (err < 0) goto unlock; - err = snd_usb_endpoint_set_params(chip, ep); - if (err < 0) - goto unlock; - err = snd_usb_select_mode_quirk(chip, ep->cur_audiofmt); if (err < 0) goto unlock; diff --git a/sound/usb/endpoint.h b/sound/usb/endpoint.h index 6a9af04cf175..e67ea28faa54 100644 --- a/sound/usb/endpoint.h +++ b/sound/usb/endpoint.h @@ -17,8 +17,10 @@ snd_usb_endpoint_open(struct snd_usb_audio *chip, bool is_sync_ep); void snd_usb_endpoint_close(struct snd_usb_audio *chip, struct snd_usb_endpoint *ep); -int snd_usb_endpoint_configure(struct snd_usb_audio *chip, - struct snd_usb_endpoint *ep); +int snd_usb_endpoint_set_params(struct snd_usb_audio *chip, + struct snd_usb_endpoint *ep); +int snd_usb_endpoint_prepare(struct snd_usb_audio *chip, + struct snd_usb_endpoint *ep); int snd_usb_endpoint_get_clock_rate(struct snd_usb_audio *chip, int clock); bool snd_usb_endpoint_compatible(struct snd_usb_audio *chip, diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index d45d1d7e6664..b604f7e95e82 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -443,17 +443,17 @@ static int configure_endpoints(struct snd_usb_audio *chip, if (stop_endpoints(subs, false)) sync_pending_stops(subs); if (subs->sync_endpoint) { - err = snd_usb_endpoint_configure(chip, subs->sync_endpoint); + err = snd_usb_endpoint_prepare(chip, subs->sync_endpoint); if (err < 0) return err; } - err = snd_usb_endpoint_configure(chip, subs->data_endpoint); + err = snd_usb_endpoint_prepare(chip, subs->data_endpoint); if (err < 0) return err; snd_usb_set_format_quirk(subs, subs->cur_audiofmt); } else { if (subs->sync_endpoint) { - err = snd_usb_endpoint_configure(chip, subs->sync_endpoint); + err = snd_usb_endpoint_prepare(chip, subs->sync_endpoint); if (err < 0) return err; } @@ -551,7 +551,13 @@ static int snd_usb_hw_params(struct snd_pcm_substream *substream, subs->cur_audiofmt = fmt; mutex_unlock(&chip->mutex); - ret = configure_endpoints(chip, subs); + if (subs->sync_endpoint) { + ret = snd_usb_endpoint_set_params(chip, subs->sync_endpoint); + if (ret < 0) + goto unlock; + } + + ret = snd_usb_endpoint_set_params(chip, subs->data_endpoint); unlock: if (ret < 0) From 3e48940abee88b8dbbeeaf8a07e7b2b6be1271b3 Mon Sep 17 00:00:00 2001 From: Pattara Teerapong Date: Thu, 1 Sep 2022 14:40:36 +0000 Subject: [PATCH 21/29] ALSA: aloop: Fix random zeros in capture data when using jiffies timer In loopback_jiffies_timer_pos_update(), we are getting jiffies twice. First time for playback, second time for capture. Jiffies can be updated between these two calls and if the capture jiffies is larger, extra zeros will be filled in the capture buffer. Change to get jiffies once and use it for both playback and capture. Signed-off-by: Pattara Teerapong Cc: Link: https://lore.kernel.org/r/20220901144036.4049060-1-pteerapong@chromium.org Signed-off-by: Takashi Iwai --- sound/drivers/aloop.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/drivers/aloop.c b/sound/drivers/aloop.c index 9b4a7cdb103a..12f12a294df5 100644 --- a/sound/drivers/aloop.c +++ b/sound/drivers/aloop.c @@ -605,17 +605,18 @@ static unsigned int loopback_jiffies_timer_pos_update cable->streams[SNDRV_PCM_STREAM_PLAYBACK]; struct loopback_pcm *dpcm_capt = cable->streams[SNDRV_PCM_STREAM_CAPTURE]; - unsigned long delta_play = 0, delta_capt = 0; + unsigned long delta_play = 0, delta_capt = 0, cur_jiffies; unsigned int running, count1, count2; + cur_jiffies = jiffies; running = cable->running ^ cable->pause; if (running & (1 << SNDRV_PCM_STREAM_PLAYBACK)) { - delta_play = jiffies - dpcm_play->last_jiffies; + delta_play = cur_jiffies - dpcm_play->last_jiffies; dpcm_play->last_jiffies += delta_play; } if (running & (1 << SNDRV_PCM_STREAM_CAPTURE)) { - delta_capt = jiffies - dpcm_capt->last_jiffies; + delta_capt = cur_jiffies - dpcm_capt->last_jiffies; dpcm_capt->last_jiffies += delta_capt; } From 414d38ba871092aeac4ed097ac4ced89486646f7 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sun, 4 Sep 2022 09:27:50 +0200 Subject: [PATCH 22/29] ALSA: hda/sigmatel: Keep power up while beep is enabled It seems that the beep playback doesn't work well on IDT codec devices when the codec auto-pm is enabled. Keep the power on while the beep switch is enabled. Link: https://bugzilla.suse.com/show_bug.cgi?id=1200544 Link: https://lore.kernel.org/r/20220904072750.26164-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 22 ++++++++++++++++++++++ 1 file changed, 22 insertions(+) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 61df4d33c48f..066bfccd2587 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -209,6 +209,7 @@ struct sigmatel_spec { /* beep widgets */ hda_nid_t anabeep_nid; + bool beep_power_on; /* SPDIF-out mux */ const char * const *spdif_labels; @@ -4443,6 +4444,26 @@ static int stac_suspend(struct hda_codec *codec) return 0; } + +static int stac_check_power_status(struct hda_codec *codec, hda_nid_t nid) +{ + struct sigmatel_spec *spec = codec->spec; + int ret = snd_hda_gen_check_power_status(codec, nid); + +#ifdef CONFIG_SND_HDA_INPUT_BEEP + if (nid == spec->gen.beep_nid && codec->beep) { + if (codec->beep->enabled != spec->beep_power_on) { + spec->beep_power_on = codec->beep->enabled; + if (spec->beep_power_on) + snd_hda_power_up_pm(codec); + else + snd_hda_power_down_pm(codec); + } + ret |= spec->beep_power_on; + } +#endif + return ret; +} #else #define stac_suspend NULL #endif /* CONFIG_PM */ @@ -4455,6 +4476,7 @@ static const struct hda_codec_ops stac_patch_ops = { .unsol_event = snd_hda_jack_unsol_event, #ifdef CONFIG_PM .suspend = stac_suspend, + .check_power_status = stac_check_power_status, #endif }; From 8423f0b6d513b259fdab9c9bf4aaa6188d054c2d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 5 Sep 2022 08:07:14 +0200 Subject: [PATCH 23/29] ALSA: pcm: oss: Fix race at SNDCTL_DSP_SYNC There is a small race window at snd_pcm_oss_sync() that is called from OSS PCM SNDCTL_DSP_SYNC ioctl; namely the function calls snd_pcm_oss_make_ready() at first, then takes the params_lock mutex for the rest. When the stream is set up again by another thread between them, it leads to inconsistency, and may result in unexpected results such as NULL dereference of OSS buffer as a fuzzer spotted recently. The fix is simply to cover snd_pcm_oss_make_ready() call into the same params_lock mutex with snd_pcm_oss_make_ready_locked() variant. Reported-and-tested-by: butt3rflyh4ck Reviewed-by: Jaroslav Kysela Cc: Link: https://lore.kernel.org/r/CAFcO6XN7JDM4xSXGhtusQfS2mSBcx50VJKwQpCq=WeLt57aaZA@mail.gmail.com Link: https://lore.kernel.org/r/20220905060714.22549-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/core/oss/pcm_oss.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c index 90c3a367d7de..02df915eb3c6 100644 --- a/sound/core/oss/pcm_oss.c +++ b/sound/core/oss/pcm_oss.c @@ -1672,14 +1672,14 @@ static int snd_pcm_oss_sync(struct snd_pcm_oss_file *pcm_oss_file) runtime = substream->runtime; if (atomic_read(&substream->mmap_count)) goto __direct; - err = snd_pcm_oss_make_ready(substream); - if (err < 0) - return err; atomic_inc(&runtime->oss.rw_ref); if (mutex_lock_interruptible(&runtime->oss.params_lock)) { atomic_dec(&runtime->oss.rw_ref); return -ERESTARTSYS; } + err = snd_pcm_oss_make_ready_locked(substream); + if (err < 0) + goto unlock; format = snd_pcm_oss_format_from(runtime->oss.format); width = snd_pcm_format_physical_width(format); if (runtime->oss.buffer_used > 0) { From 51bdc8bb82525cd70feb92279c8b7660ad7948dd Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 5 Sep 2022 15:06:30 +0200 Subject: [PATCH 24/29] ALSA: hda/sigmatel: Fix unused variable warning for beep power change The newly added stac_check_power_status() caused a compile warning when CONFIG_SND_HDA_INPUT_BEEP is disabled. Fix it. Fixes: 414d38ba8710 ("ALSA: hda/sigmatel: Keep power up while beep is enabled") Reported-by: kernel test robot Link: https://lore.kernel.org/r/20220905130630.2845-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 066bfccd2587..7f340f18599c 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -4447,7 +4447,9 @@ static int stac_suspend(struct hda_codec *codec) static int stac_check_power_status(struct hda_codec *codec, hda_nid_t nid) { +#ifdef CONFIG_SND_HDA_INPUT_BEEP struct sigmatel_spec *spec = codec->spec; +#endif int ret = snd_hda_gen_check_power_status(codec, nid); #ifdef CONFIG_SND_HDA_INPUT_BEEP From 8d44e6044a0e885acdd01813768a0b27906d64fd Mon Sep 17 00:00:00 2001 From: Mohan Kumar Date: Mon, 5 Sep 2022 22:54:20 +0530 Subject: [PATCH 25/29] ALSA: hda/tegra: Align BDL entry to 4KB boundary AZA HW may send a burst read/write request crossing 4K memory boundary. The 4KB boundary is not guaranteed by Tegra HDA HW. Make SW change to include the flag AZX_DCAPS_4K_BDLE_BOUNDARY to align BDLE to 4K boundary. Signed-off-by: Mohan Kumar Link: https://lore.kernel.org/r/20220905172420.3801-1-mkumard@nvidia.com Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_tegra.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_tegra.c b/sound/pci/hda/hda_tegra.c index 7debb2c76aa6..976a112c7d00 100644 --- a/sound/pci/hda/hda_tegra.c +++ b/sound/pci/hda/hda_tegra.c @@ -474,7 +474,8 @@ MODULE_DEVICE_TABLE(of, hda_tegra_match); static int hda_tegra_probe(struct platform_device *pdev) { const unsigned int driver_flags = AZX_DCAPS_CORBRP_SELF_CLEAR | - AZX_DCAPS_PM_RUNTIME; + AZX_DCAPS_PM_RUNTIME | + AZX_DCAPS_4K_BDLE_BOUNDARY; struct snd_card *card; struct azx *chip; struct hda_tegra *hda; From e53f47f6c1a56d2af728909f1cb894da6b43d9bf Mon Sep 17 00:00:00 2001 From: Dongxiang Ke Date: Tue, 6 Sep 2022 10:49:28 +0800 Subject: [PATCH 26/29] ALSA: usb-audio: Fix an out-of-bounds bug in __snd_usb_parse_audio_interface() There may be a bad USB audio device with a USB ID of (0x04fa, 0x4201) and the number of it's interfaces less than 4, an out-of-bounds read bug occurs when parsing the interface descriptor for this device. Fix this by checking the number of interfaces. Signed-off-by: Dongxiang Ke Link: https://lore.kernel.org/r/20220906024928.10951-1-kdx.glider@gmail.com Signed-off-by: Takashi Iwai --- sound/usb/stream.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/stream.c b/sound/usb/stream.c index 40b7821c6c99..f10f4e6d3fb8 100644 --- a/sound/usb/stream.c +++ b/sound/usb/stream.c @@ -1106,7 +1106,7 @@ static int __snd_usb_parse_audio_interface(struct snd_usb_audio *chip, * Dallas DS4201 workaround: It presents 5 altsettings, but the last * one misses syncpipe, and does not produce any sound. */ - if (chip->usb_id == USB_ID(0x04fa, 0x4201)) + if (chip->usb_id == USB_ID(0x04fa, 0x4201) && num >= 4) num = 4; for (i = 0; i < num; i++) { From 37137ec26c2c03039d8064c00f6eae176841ee0d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 6 Sep 2022 11:03:19 +0200 Subject: [PATCH 27/29] ALSA: hda: Once again fix regression of page allocations with IOMMU The last fix for trying to recover the regression on AMD platforms, unfortunately, leaded to yet another regression: it turned out that IOMMUs don't like the usage of raw page allocations. This is yet another attempt for addressing the log saga; at this time, we re-use the existing buffer allocation mechanism with SG-pages although we require only single pages. The SG buffer allocation itself was confirmed to work for stream buffers, so it's relatively easy to adapt for other places. The only problem is: although the HD-audio code is accessing the address directly via dmab->address field, SG-pages don't set up it. For the ease of adaption, we now set up the dmab->addr field from the address of the first page as default, so that it can run with the HD-audio driver code as-is without the excessive call of snd_sgbuf_get_addr() multiple times; that's the only change in the memalloc helper side. The rest is nothing but a flip of the dma_type field in the HD-audio side. Fixes: a8d302a0b770 ("ALSA: memalloc: Revive x86-specific WC page allocations again") Reported-by: Mikhail Gavrilov Tested-by: Mikhail Gavrilov Cc: Link: https://lore.kernel.org/r/CABXGCsO+kB2t5QyHY-rUe76npr1m0-5JOtt8g8SiHUo34ur7Ww@mail.gmail.com Link: https://bugzilla.kernel.org/show_bug.cgi?id=216112 Link: https://bugzilla.kernel.org/show_bug.cgi?id=216363 Link: https://lore.kernel.org/r/20220906090319.23358-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/core/memalloc.c | 9 +++++++-- sound/pci/hda/hda_intel.c | 2 +- 2 files changed, 8 insertions(+), 3 deletions(-) diff --git a/sound/core/memalloc.c b/sound/core/memalloc.c index b665ac66ccbe..cfcd8eff4139 100644 --- a/sound/core/memalloc.c +++ b/sound/core/memalloc.c @@ -543,10 +543,13 @@ static void *snd_dma_noncontig_alloc(struct snd_dma_buffer *dmab, size_t size) dmab->dev.need_sync = dma_need_sync(dmab->dev.dev, sg_dma_address(sgt->sgl)); p = dma_vmap_noncontiguous(dmab->dev.dev, size, sgt); - if (p) + if (p) { dmab->private_data = sgt; - else + /* store the first page address for convenience */ + dmab->addr = snd_sgbuf_get_addr(dmab, 0); + } else { dma_free_noncontiguous(dmab->dev.dev, size, sgt, dmab->dev.dir); + } return p; } @@ -780,6 +783,8 @@ static void *snd_dma_sg_fallback_alloc(struct snd_dma_buffer *dmab, size_t size) if (!p) goto error; dmab->private_data = sgbuf; + /* store the first page address for convenience */ + dmab->addr = snd_sgbuf_get_addr(dmab, 0); return p; error: diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index a77165bd92a9..b20694fd69de 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1817,7 +1817,7 @@ static int azx_create(struct snd_card *card, struct pci_dev *pci, /* use the non-cached pages in non-snoop mode */ if (!azx_snoop(chip)) - azx_bus(chip)->dma_type = SNDRV_DMA_TYPE_DEV_WC; + azx_bus(chip)->dma_type = SNDRV_DMA_TYPE_DEV_WC_SG; if (chip->driver_type == AZX_DRIVER_NVIDIA) { dev_dbg(chip->card->dev, "Enable delay in RIRB handling\n"); From d29f59051d3a07b81281b2df2b8c9dfe4716067f Mon Sep 17 00:00:00 2001 From: Tasos Sahanidis Date: Wed, 7 Sep 2022 04:18:00 +0300 Subject: [PATCH 28/29] ALSA: emu10k1: Fix out of bounds access in snd_emu10k1_pcm_channel_alloc() The voice allocator sometimes begins allocating from near the end of the array and then wraps around, however snd_emu10k1_pcm_channel_alloc() accesses the newly allocated voices as if it never wrapped around. This results in out of bounds access if the first voice has a high enough index so that first_voice + requested_voice_count > NUM_G (64). The more voices are requested, the more likely it is for this to occur. This was initially discovered using PipeWire, however it can be reproduced by calling aplay multiple times with 16 channels: aplay -r 48000 -D plughw:CARD=Live,DEV=3 -c 16 /dev/zero UBSAN: array-index-out-of-bounds in sound/pci/emu10k1/emupcm.c:127:40 index 65 is out of range for type 'snd_emu10k1_voice [64]' CPU: 1 PID: 31977 Comm: aplay Tainted: G W IOE 6.0.0-rc2-emu10k1+ #7 Hardware name: ASUSTEK COMPUTER INC P5W DH Deluxe/P5W DH Deluxe, BIOS 3002 07/22/2010 Call Trace: dump_stack_lvl+0x49/0x63 dump_stack+0x10/0x16 ubsan_epilogue+0x9/0x3f __ubsan_handle_out_of_bounds.cold+0x44/0x49 snd_emu10k1_playback_hw_params+0x3bc/0x420 [snd_emu10k1] snd_pcm_hw_params+0x29f/0x600 [snd_pcm] snd_pcm_common_ioctl+0x188/0x1410 [snd_pcm] ? exit_to_user_mode_prepare+0x35/0x170 ? do_syscall_64+0x69/0x90 ? syscall_exit_to_user_mode+0x26/0x50 ? do_syscall_64+0x69/0x90 ? exit_to_user_mode_prepare+0x35/0x170 snd_pcm_ioctl+0x27/0x40 [snd_pcm] __x64_sys_ioctl+0x95/0xd0 do_syscall_64+0x5c/0x90 ? do_syscall_64+0x69/0x90 ? do_syscall_64+0x69/0x90 entry_SYSCALL_64_after_hwframe+0x63/0xcd Signed-off-by: Tasos Sahanidis Cc: Link: https://lore.kernel.org/r/3707dcab-320a-62ff-63c0-73fc201ef756@tasossah.com Signed-off-by: Takashi Iwai --- sound/pci/emu10k1/emupcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/emu10k1/emupcm.c b/sound/pci/emu10k1/emupcm.c index b2701a4452d8..48af77ae8020 100644 --- a/sound/pci/emu10k1/emupcm.c +++ b/sound/pci/emu10k1/emupcm.c @@ -124,7 +124,7 @@ static int snd_emu10k1_pcm_channel_alloc(struct snd_emu10k1_pcm * epcm, int voic epcm->voices[0]->epcm = epcm; if (voices > 1) { for (i = 1; i < voices; i++) { - epcm->voices[i] = &epcm->emu->voices[epcm->voices[0]->number + i]; + epcm->voices[i] = &epcm->emu->voices[(epcm->voices[0]->number + i) % NUM_G]; epcm->voices[i]->epcm = epcm; } } From 809f44a0cc5ad4b1209467a6287f8ac0eb49d393 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 7 Sep 2022 12:04:21 +0200 Subject: [PATCH 29/29] ALSA: usb-audio: Clear fixed clock rate at closing EP The recent commit c11117b634f4 ("ALSA: usb-audio: Refcount multiple accesses on the single clock") tries to manage the clock rate shared by several endpoints. This was intended for avoiding the unmatched rate by a different endpoint, but unfortunately, it introduced a regression for PulseAudio and pipewire, too; those applications try to probe the multiple possible rates (44.1k and 48kHz) and setting up the normal rate fails but only the last rate is applied. The cause is that the last sample rate is still left to the clock reference even after closing the endpoint, and this value is still used at the next open. It happens only when applications set up via PCM prepare but don't start/stop the stream; the rate is reset when the stream is stopped, but it's not cleared at close. This patch addresses the issue above, simply by clearing the rate set in the clock reference at the last close of each endpoint. Fixes: c11117b634f4 ("ALSA: usb-audio: Refcount multiple accesses on the single clock") Reported-by: Jason A. Donenfeld Tested-by: Jason A. Donenfeld Cc: Link: https://lore.kernel.org/all/YxXIWv8dYmg1tnXP@zx2c4.com/ Link: https://gitlab.freedesktop.org/pipewire/pipewire/-/issues/2620 Link: https://lore.kernel.org/r/20220907100421.6443-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/usb/endpoint.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index a42f2ce19455..8c8f9a851f89 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -925,6 +925,8 @@ void snd_usb_endpoint_close(struct snd_usb_audio *chip, endpoint_set_interface(chip, ep, false); if (!--ep->opened) { + if (ep->clock_ref && !atomic_read(&ep->clock_ref->locked)) + ep->clock_ref->rate = 0; ep->iface = 0; ep->altsetting = 0; ep->cur_audiofmt = NULL;