From fdabc3f10e774ddc86ba715b9bc0c861d7e0834c Mon Sep 17 00:00:00 2001 From: Zhang Qilong Date: Thu, 29 Sep 2022 23:06:50 +0800 Subject: [PATCH 01/50] ASoC: wm8997: Fix PM disable depth imbalance in wm8997_probe The pm_runtime_enable will increase power disable depth. Thus a pairing decrement is needed on the error handling path to keep it balanced according to context. We fix it by calling pm_runtime_disable when error returns. Fixes:40843aea5a9bd ("ASoC: wm8997: Initial CODEC driver") Signed-off-by: Zhang Qilong Acked-by: Charles Keepax Link: https://lore.kernel.org/r/20220929150653.63845-2-zhangqilong3@huawei.com Signed-off-by: Mark Brown --- sound/soc/codecs/wm8997.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/wm8997.c b/sound/soc/codecs/wm8997.c index 77136a521605..f8993176d5c0 100644 --- a/sound/soc/codecs/wm8997.c +++ b/sound/soc/codecs/wm8997.c @@ -1187,6 +1187,7 @@ static int wm8997_probe(struct platform_device *pdev) err_spk_irqs: arizona_free_spk_irqs(arizona); err_jack_codec_dev: + pm_runtime_disable(&pdev->dev); arizona_jack_codec_dev_remove(&wm8997->core); return ret; From 6ab646c985b529a32bf162de48d2d4a8bb7c9b64 Mon Sep 17 00:00:00 2001 From: Zhang Qilong Date: Thu, 29 Sep 2022 23:06:51 +0800 Subject: [PATCH 02/50] ASoC: wm5110: Fix PM disable depth imbalance in wm5110_probe The pm_runtime_enable will increase power disable depth. Thus a pairing decrement is needed on the error handling path to keep it balanced according to context. We fix it by calling pm_runtime_enable when error returns. Fixes:5c6af635fd772 ("ASoC: wm5110: Add audio CODEC driver") Signed-off-by: Zhang Qilong Acked-by: Charles Keepax Link: https://lore.kernel.org/r/20220929150653.63845-3-zhangqilong3@huawei.com Signed-off-by: Mark Brown --- sound/soc/codecs/wm5110.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index fc634c995834..8a61563eae11 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -2500,6 +2500,7 @@ err_dsp_irq: arizona_set_irq_wake(arizona, ARIZONA_IRQ_DSP_IRQ1, 0); arizona_free_irq(arizona, ARIZONA_IRQ_DSP_IRQ1, wm5110); err_jack_codec_dev: + pm_runtime_disable(&pdev->dev); arizona_jack_codec_dev_remove(&wm5110->core); return ret; From 96e4abbd35adb5582573c463ccc554a644ac2434 Mon Sep 17 00:00:00 2001 From: Zhang Qilong Date: Thu, 29 Sep 2022 23:06:52 +0800 Subject: [PATCH 03/50] ASoC: wm5102: Fix PM disable depth imbalance in wm5102_probe The pm_runtime_enable will increase power disable depth. Thus a pairing decrement is needed on the error handling path to keep it balanced according to context. We fix it by calling pm_runtime_disable when error returns. Fixes:93e8791dd34ca ("ASoC: wm5102: Initial driver") Signed-off-by: Zhang Qilong Acked-by: Charles Keepax Link: https://lore.kernel.org/r/20220929150653.63845-4-zhangqilong3@huawei.com Signed-off-by: Mark Brown --- sound/soc/codecs/wm5102.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index c09c9ac51b3e..e56e30d59760 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -2142,6 +2142,7 @@ err_dsp_irq: arizona_set_irq_wake(arizona, ARIZONA_IRQ_DSP_IRQ1, 0); arizona_free_irq(arizona, ARIZONA_IRQ_DSP_IRQ1, wm5102); err_jack_codec_dev: + pm_runtime_disable(&pdev->dev); arizona_jack_codec_dev_remove(&wm5102->core); return ret; From 0c72dbc96be870e4de8f9707c9a4c6d7a641381c Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Tue, 4 Oct 2022 14:51:21 +0300 Subject: [PATCH 04/50] Revert "ASoC: soc-component: using pm_runtime_resume_and_get instead of pm_runtime_get_sync" This reverts commit 08fc2a7448afc1660ec2f1b5c437fcd14155a7ee. The reverted commit causes the following warnigs: Runtime PM usage count underflow! This is due to the fact that the pm_runtime_resume_and_get() is calling pm_runtime_put_noidle() in case of < 0 return value of pm_runtime_get_sync() which includes the -EACCES. The change is wrong as -EACCES is returned in case of 'nested' get_sync() and it is a valid use of PM runtime. Fixes: 08fc2a7448af ("ASoC: soc-component: using pm_runtime_resume_and_get instead of pm_runtime_get_sync") Signed-off-by: Peter Ujfalusi Link: https://lore.kernel.org/r/20221004115121.26180-1-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/soc-component.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-component.c b/sound/soc/soc-component.c index 659b9ade4158..e12f8244242b 100644 --- a/sound/soc/soc-component.c +++ b/sound/soc/soc-component.c @@ -1213,9 +1213,11 @@ int snd_soc_pcm_component_pm_runtime_get(struct snd_soc_pcm_runtime *rtd, int i; for_each_rtd_components(rtd, i, component) { - int ret = pm_runtime_resume_and_get(component->dev); - if (ret < 0 && ret != -EACCES) + int ret = pm_runtime_get_sync(component->dev); + if (ret < 0 && ret != -EACCES) { + pm_runtime_put_noidle(component->dev); return soc_component_ret(component, ret); + } /* mark stream if succeeded */ soc_component_mark_push(component, stream, pm); } From de71d7567e358effd06dfc3e2a154b25f1331c10 Mon Sep 17 00:00:00 2001 From: Zhang Qilong Date: Mon, 10 Oct 2022 19:48:50 +0800 Subject: [PATCH 05/50] ASoC: wm5102: Revert "ASoC: wm5102: Fix PM disable depth imbalance in wm5102_probe" This reverts commit fcbb60820cd3008bb44334a0395e5e57ccb77329. The pm_runtime_disable is redundant when error returns in wm5102_probe, we just revert the old patch to fix it. Signed-off-by: Zhang Qilong Acked-by: Charles Keepax Link: https://lore.kernel.org/r/20221010114852.88127-2-zhangqilong3@huawei.com Signed-off-by: Mark Brown --- sound/soc/codecs/wm5102.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index e56e30d59760..adaf886b0a9d 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -2099,6 +2099,9 @@ static int wm5102_probe(struct platform_device *pdev) regmap_update_bits(arizona->regmap, wm5102_digital_vu[i], WM5102_DIG_VU, WM5102_DIG_VU); + pm_runtime_enable(&pdev->dev); + pm_runtime_idle(&pdev->dev); + ret = arizona_request_irq(arizona, ARIZONA_IRQ_DSP_IRQ1, "ADSP2 Compressed IRQ", wm5102_adsp2_irq, wm5102); @@ -2131,9 +2134,6 @@ static int wm5102_probe(struct platform_device *pdev) goto err_spk_irqs; } - pm_runtime_enable(&pdev->dev); - pm_runtime_idle(&pdev->dev); - return ret; err_spk_irqs: From 7d4e966f4cd73ff69bf06934e8e14a33fb7ef447 Mon Sep 17 00:00:00 2001 From: Zhang Qilong Date: Mon, 10 Oct 2022 19:48:51 +0800 Subject: [PATCH 06/50] ASoC: wm5110: Revert "ASoC: wm5110: Fix PM disable depth imbalance in wm5110_probe" This reverts commit 86b46bf1feb83898d89a2b4a8d08d21e9ea277a7. The pm_runtime_disable is redundant when error returns in wm5110_probe, we just revert the old patch to fix it. Signed-off-by: Zhang Qilong Acked-by: Charles Keepax Link: https://lore.kernel.org/r/20221010114852.88127-3-zhangqilong3@huawei.com Signed-off-by: Mark Brown --- sound/soc/codecs/wm5110.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index 8a61563eae11..e0b971620d0f 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -2457,6 +2457,9 @@ static int wm5110_probe(struct platform_device *pdev) regmap_update_bits(arizona->regmap, wm5110_digital_vu[i], WM5110_DIG_VU, WM5110_DIG_VU); + pm_runtime_enable(&pdev->dev); + pm_runtime_idle(&pdev->dev); + ret = arizona_request_irq(arizona, ARIZONA_IRQ_DSP_IRQ1, "ADSP2 Compressed IRQ", wm5110_adsp2_irq, wm5110); @@ -2489,9 +2492,6 @@ static int wm5110_probe(struct platform_device *pdev) goto err_spk_irqs; } - pm_runtime_enable(&pdev->dev); - pm_runtime_idle(&pdev->dev); - return ret; err_spk_irqs: From 68ce83e3bb26feba0fcdd59667fde942b3a600a1 Mon Sep 17 00:00:00 2001 From: Zhang Qilong Date: Mon, 10 Oct 2022 19:48:52 +0800 Subject: [PATCH 07/50] ASoC: wm8997: Revert "ASoC: wm8997: Fix PM disable depth imbalance in wm8997_probe" This reverts commit 41a736ac20602f64773e80f0f5b32cde1830a44a. The pm_runtime_disable is redundant when error returns in wm8997_probe, we just revert the old patch to fix it. Signed-off-by: Zhang Qilong Acked-by: Charles Keepax Link: https://lore.kernel.org/r/20221010114852.88127-4-zhangqilong3@huawei.com Signed-off-by: Mark Brown --- sound/soc/codecs/wm8997.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm8997.c b/sound/soc/codecs/wm8997.c index f8993176d5c0..c0207e9a7d53 100644 --- a/sound/soc/codecs/wm8997.c +++ b/sound/soc/codecs/wm8997.c @@ -1161,6 +1161,9 @@ static int wm8997_probe(struct platform_device *pdev) regmap_update_bits(arizona->regmap, wm8997_digital_vu[i], WM8997_DIG_VU, WM8997_DIG_VU); + pm_runtime_enable(&pdev->dev); + pm_runtime_idle(&pdev->dev); + arizona_init_common(arizona); ret = arizona_init_vol_limit(arizona); @@ -1179,9 +1182,6 @@ static int wm8997_probe(struct platform_device *pdev) goto err_spk_irqs; } - pm_runtime_enable(&pdev->dev); - pm_runtime_idle(&pdev->dev); - return ret; err_spk_irqs: From 551f2994b8ccdbe296e239278531e345d6e94d4d Mon Sep 17 00:00:00 2001 From: Randy Dunlap Date: Thu, 6 Oct 2022 16:58:22 -0700 Subject: [PATCH 08/50] ASoC: codec: tlv320adc3xxx: add GPIOLIB dependency Fix build errors when CONFIG_GPIOLIB is not enabled: ../sound/soc/codecs/tlv320adc3xxx.c: In function 'adc3xxx_i2c_probe': ../sound/soc/codecs/tlv320adc3xxx.c:1352:28: error: implicit declaration of function 'devm_gpiod_get'; did you mean 'devm_gpio_free'? [-Werror=implicit-function-declaration] 1352 | adc3xxx->rst_pin = devm_gpiod_get(dev, "reset", GPIOD_OUT_LOW); ../sound/soc/codecs/tlv320adc3xxx.c:1352:57: error: 'GPIOD_OUT_LOW' undeclared (first use in this function); did you mean 'GPIOF_INIT_LOW'? 1352 | adc3xxx->rst_pin = devm_gpiod_get(dev, "reset", GPIOD_OUT_LOW); CC lib/dynamic_debug.o ../sound/soc/codecs/tlv320adc3xxx.c:1400:9: error: implicit declaration of function 'gpiod_set_value_cansleep'; did you mean 'gpio_set_value_cansleep'? [-Werror=implicit-function-declaration] 1400 | gpiod_set_value_cansleep(adc3xxx->rst_pin, 1); Fixes: e9a3b57efd28 ("ASoC: codec: tlv320adc3xxx: New codec driver") Signed-off-by: Randy Dunlap Reported-by: kernel test robot Cc: Mark Brown Cc: Liam Girdwood Cc: Ricard Wanderlof Cc: Jaroslav Kysela Cc: Takashi Iwai Cc: alsa-devel@alsa-project.org Link: https://lore.kernel.org/r/20221006235822.30074-1-rdunlap@infradead.org Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index e3b90c425faf..7022e6286e6c 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -1629,6 +1629,7 @@ config SND_SOC_TFA989X config SND_SOC_TLV320ADC3XXX tristate "Texas Instruments TLV320ADC3001/3101 audio ADC" depends on I2C + depends on GPIOLIB help Enable support for Texas Instruments TLV320ADC3001 and TLV320ADC3101 ADCs. From c4ab29b0f3a6f1e167c5a627f7cd036c1d2b7d65 Mon Sep 17 00:00:00 2001 From: Zhang Qilong Date: Sat, 8 Oct 2022 22:05:22 +0800 Subject: [PATCH 09/50] ASoC: mt6660: Keep the pm_runtime enables before component stuff in mt6660_i2c_probe It would be better to keep the pm_runtime enables before the IRQ and component stuff. Both of those could start triggering PM runtime events. Signed-off-by: Zhang Qilong Reviewed-by: AngeloGioacchino Del Regno Link: https://lore.kernel.org/r/20221008140522.134912-1-zhangqilong3@huawei.com Signed-off-by: Mark Brown --- sound/soc/codecs/mt6660.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/mt6660.c b/sound/soc/codecs/mt6660.c index 45e0df13afb9..b8369eeccc30 100644 --- a/sound/soc/codecs/mt6660.c +++ b/sound/soc/codecs/mt6660.c @@ -503,14 +503,14 @@ static int mt6660_i2c_probe(struct i2c_client *client) dev_err(chip->dev, "read chip revision fail\n"); goto probe_fail; } + pm_runtime_set_active(chip->dev); + pm_runtime_enable(chip->dev); ret = devm_snd_soc_register_component(chip->dev, &mt6660_component_driver, &mt6660_codec_dai, 1); - if (!ret) { - pm_runtime_set_active(chip->dev); - pm_runtime_enable(chip->dev); - } + if (ret) + pm_runtime_disable(chip->dev); return ret; From 29eb79a9a6283d661ea1f70ab012809fdbf057a7 Mon Sep 17 00:00:00 2001 From: Jiangshan Yi Date: Sun, 9 Oct 2022 15:48:16 +0800 Subject: [PATCH 10/50] ASoC: cx2072x: fix spelling typo in comment Fix spelling typo in comment. Reported-by: k2ci Signed-off-by: Jiangshan Yi Link: https://lore.kernel.org/r/20221009074816.2641162-1-13667453960@163.com Signed-off-by: Mark Brown --- sound/soc/codecs/cx2072x.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/cx2072x.h b/sound/soc/codecs/cx2072x.h index ebdd567fa225..09e3a92b184f 100644 --- a/sound/soc/codecs/cx2072x.h +++ b/sound/soc/codecs/cx2072x.h @@ -177,7 +177,7 @@ #define CX2072X_PLBK_DRC_PARM_LEN 9 #define CX2072X_CLASSD_AMP_LEN 6 -/* DAI interfae type */ +/* DAI interface type */ #define CX2072X_DAI_HIFI 1 #define CX2072X_DAI_DSP 2 #define CX2072X_DAI_DSP_PWM 3 /* 4 ch, including mic and AEC */ From d94bf16e920047c9b4ea2b57f7b53b4ff5039d9f Mon Sep 17 00:00:00 2001 From: Derek Fang Date: Wed, 12 Oct 2022 11:13:20 +0800 Subject: [PATCH 11/50] ASoC: rt5682s: Fix the TDM Tx settings Complete the missing and correct the TDM Tx settings. Signed-off-by: Derek Fang Link: https://lore.kernel.org/r/20221012031320.6980-1-derek.fang@realtek.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt5682s.c | 15 +++++++++++++-- sound/soc/codecs/rt5682s.h | 1 + 2 files changed, 14 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5682s.c b/sound/soc/codecs/rt5682s.c index 3b91a3442c68..5199d3bbaf0b 100644 --- a/sound/soc/codecs/rt5682s.c +++ b/sound/soc/codecs/rt5682s.c @@ -1981,7 +1981,7 @@ static int rt5682s_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width) { struct snd_soc_component *component = dai->component; - unsigned int cl, val = 0; + unsigned int cl, val = 0, tx_slotnum; if (tx_mask || rx_mask) snd_soc_component_update_bits(component, @@ -1990,6 +1990,16 @@ static int rt5682s_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, snd_soc_component_update_bits(component, RT5682S_TDM_ADDA_CTRL_2, RT5682S_TDM_EN, 0); + /* Tx slot configuration */ + tx_slotnum = hweight_long(tx_mask); + if (tx_slotnum) { + if (tx_slotnum > slots) { + dev_err(component->dev, "Invalid or oversized Tx slots.\n"); + return -EINVAL; + } + val |= (tx_slotnum - 1) << RT5682S_TDM_ADC_DL_SFT; + } + switch (slots) { case 4: val |= RT5682S_TDM_TX_CH_4; @@ -2010,7 +2020,8 @@ static int rt5682s_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, } snd_soc_component_update_bits(component, RT5682S_TDM_CTRL, - RT5682S_TDM_TX_CH_MASK | RT5682S_TDM_RX_CH_MASK, val); + RT5682S_TDM_TX_CH_MASK | RT5682S_TDM_RX_CH_MASK | + RT5682S_TDM_ADC_DL_MASK, val); switch (slot_width) { case 8: diff --git a/sound/soc/codecs/rt5682s.h b/sound/soc/codecs/rt5682s.h index 824dc6543c18..45464a041765 100644 --- a/sound/soc/codecs/rt5682s.h +++ b/sound/soc/codecs/rt5682s.h @@ -899,6 +899,7 @@ #define RT5682S_TDM_RX_CH_8 (0x3 << 8) #define RT5682S_TDM_ADC_LCA_MASK (0x7 << 4) #define RT5682S_TDM_ADC_LCA_SFT 4 +#define RT5682S_TDM_ADC_DL_MASK (0x3 << 0) #define RT5682S_TDM_ADC_DL_SFT 0 /* TDM control 2 (0x007a) */ From f2635d45a750182c6d5de15e2d6b059e0c302d7e Mon Sep 17 00:00:00 2001 From: Derek Fang Date: Wed, 12 Oct 2022 11:01:02 +0800 Subject: [PATCH 12/50] ASoC: rt1019: Fix the TDM settings Complete the missing and correct the TDM settings. Signed-off-by: Derek Fang Link: https://lore.kernel.org/r/20221012030102.4042-1-derek.fang@realtek.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt1019.c | 20 +++++++++++--------- sound/soc/codecs/rt1019.h | 6 ++++++ 2 files changed, 17 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/rt1019.c b/sound/soc/codecs/rt1019.c index b66bfecbb879..49f527c61a7a 100644 --- a/sound/soc/codecs/rt1019.c +++ b/sound/soc/codecs/rt1019.c @@ -391,18 +391,18 @@ static int rt1019_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width) { struct snd_soc_component *component = dai->component; - unsigned int val = 0, rx_slotnum; + unsigned int cn = 0, cl = 0, rx_slotnum; int ret = 0, first_bit; switch (slots) { case 4: - val |= RT1019_I2S_TX_4CH; + cn = RT1019_I2S_TX_4CH; break; case 6: - val |= RT1019_I2S_TX_6CH; + cn = RT1019_I2S_TX_6CH; break; case 8: - val |= RT1019_I2S_TX_8CH; + cn = RT1019_I2S_TX_8CH; break; case 2: break; @@ -412,16 +412,16 @@ static int rt1019_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, switch (slot_width) { case 20: - val |= RT1019_I2S_DL_20; + cl = RT1019_TDM_CL_20; break; case 24: - val |= RT1019_I2S_DL_24; + cl = RT1019_TDM_CL_24; break; case 32: - val |= RT1019_I2S_DL_32; + cl = RT1019_TDM_CL_32; break; case 8: - val |= RT1019_I2S_DL_8; + cl = RT1019_TDM_CL_8; break; case 16: break; @@ -470,8 +470,10 @@ static int rt1019_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, goto _set_tdm_err_; } + snd_soc_component_update_bits(component, RT1019_TDM_1, + RT1019_TDM_CL_MASK, cl); snd_soc_component_update_bits(component, RT1019_TDM_2, - RT1019_I2S_CH_TX_MASK | RT1019_I2S_DF_MASK, val); + RT1019_I2S_CH_TX_MASK, cn); _set_tdm_err_: return ret; diff --git a/sound/soc/codecs/rt1019.h b/sound/soc/codecs/rt1019.h index 64df831eeb72..48ba15efb48d 100644 --- a/sound/soc/codecs/rt1019.h +++ b/sound/soc/codecs/rt1019.h @@ -95,6 +95,12 @@ #define RT1019_TDM_BCLK_MASK (0x1 << 6) #define RT1019_TDM_BCLK_NORM (0x0 << 6) #define RT1019_TDM_BCLK_INV (0x1 << 6) +#define RT1019_TDM_CL_MASK (0x7) +#define RT1019_TDM_CL_8 (0x4) +#define RT1019_TDM_CL_32 (0x3) +#define RT1019_TDM_CL_24 (0x2) +#define RT1019_TDM_CL_20 (0x1) +#define RT1019_TDM_CL_16 (0x0) /* 0x0401 TDM Control-2 */ #define RT1019_I2S_CH_TX_MASK (0x3 << 6) From ee1aa2ae3eaa96e70229fa61deee87ef4528ffdf Mon Sep 17 00:00:00 2001 From: Xiaolei Wang Date: Mon, 10 Oct 2022 17:20:14 +0800 Subject: [PATCH 13/50] ASoC: wm8962: Add an event handler for TEMP_HP and TEMP_SPK In wm8962 driver, the WM8962_ADDITIONAL_CONTROL_4 is used as a volatile register, but this register mixes a bunch of volatile status bits and a bunch of non-volatile control bits. The dapm widgets TEMP_HP and TEMP_SPK leverages the control bits in this register. After the wm8962 probe, the regmap will bet set to cache only mode, then a read error like below would be triggered when trying to read the initial power state of the dapm widgets TEMP_HP and TEMP_SPK. wm8962 0-001a: ASoC: error at soc_component_read_no_lock on wm8962.0-001a: -16 In order to fix this issue, we add event handler to actually power up/down these widgets. With this change, we also need to explicitly power off these widgets in the wm8962 probe since they are enabled by default. Signed-off-by: Xiaolei Wang Tested-by: Adam Ford Acked-by: Charles Keepax Link: https://lore.kernel.org/r/20221010092014.2229246-1-xiaolei.wang@windriver.com Signed-off-by: Mark Brown --- sound/soc/codecs/wm8962.c | 54 +++++++++++++++++++++++++++++++++++++-- 1 file changed, 52 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 398c448ea854..6df06fba4377 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -1840,6 +1840,49 @@ SOC_SINGLE_TLV("SPKOUTR Mixer DACR Volume", WM8962_SPEAKER_MIXER_5, 4, 1, 0, inmix_tlv), }; +static int tp_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + int ret, reg, val, mask; + struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); + + ret = pm_runtime_resume_and_get(component->dev); + if (ret < 0) { + dev_err(component->dev, "Failed to resume device: %d\n", ret); + return ret; + } + + reg = WM8962_ADDITIONAL_CONTROL_4; + + if (!strcmp(w->name, "TEMP_HP")) { + mask = WM8962_TEMP_ENA_HP_MASK; + val = WM8962_TEMP_ENA_HP; + } else if (!strcmp(w->name, "TEMP_SPK")) { + mask = WM8962_TEMP_ENA_SPK_MASK; + val = WM8962_TEMP_ENA_SPK; + } else { + pm_runtime_put(component->dev); + return -EINVAL; + } + + switch (event) { + case SND_SOC_DAPM_POST_PMD: + val = 0; + fallthrough; + case SND_SOC_DAPM_POST_PMU: + ret = snd_soc_component_update_bits(component, reg, mask, val); + break; + default: + WARN(1, "Invalid event %d\n", event); + pm_runtime_put(component->dev); + return -EINVAL; + } + + pm_runtime_put(component->dev); + + return 0; +} + static int cp_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -2140,8 +2183,10 @@ SND_SOC_DAPM_SUPPLY("TOCLK", WM8962_ADDITIONAL_CONTROL_1, 0, 0, NULL, 0), SND_SOC_DAPM_SUPPLY_S("DSP2", 1, WM8962_DSP2_POWER_MANAGEMENT, WM8962_DSP2_ENA_SHIFT, 0, dsp2_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), -SND_SOC_DAPM_SUPPLY("TEMP_HP", WM8962_ADDITIONAL_CONTROL_4, 2, 0, NULL, 0), -SND_SOC_DAPM_SUPPLY("TEMP_SPK", WM8962_ADDITIONAL_CONTROL_4, 1, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("TEMP_HP", SND_SOC_NOPM, 0, 0, tp_event, + SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD), +SND_SOC_DAPM_SUPPLY("TEMP_SPK", SND_SOC_NOPM, 0, 0, tp_event, + SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_MIXER("INPGAL", WM8962_LEFT_INPUT_PGA_CONTROL, 4, 0, inpgal, ARRAY_SIZE(inpgal)), @@ -3763,6 +3808,11 @@ static int wm8962_i2c_probe(struct i2c_client *i2c) if (ret < 0) goto err_pm_runtime; + regmap_update_bits(wm8962->regmap, WM8962_ADDITIONAL_CONTROL_4, + WM8962_TEMP_ENA_HP_MASK, 0); + regmap_update_bits(wm8962->regmap, WM8962_ADDITIONAL_CONTROL_4, + WM8962_TEMP_ENA_SPK_MASK, 0); + regcache_cache_only(wm8962->regmap, true); /* The drivers should power up as needed */ From c9a3545b1d771fb7b06a487796c40288c02c41c5 Mon Sep 17 00:00:00 2001 From: Srinivasa Rao Mandadapu Date: Thu, 13 Oct 2022 10:38:31 +0530 Subject: [PATCH 14/50] ASoC: qcom: lpass-cpu: mark HDMI TX registers as volatile Update HDMI volatile registers list as DMA, Channel Selection registers, vbit control registers are being reflected by hardware DP port disconnection. This update is required to fix no display and no sound issue observed after reconnecting TAMA/SANWA DP cables. Once DP cable is unplugged, DMA control registers are being reset by hardware, however at second plugin, new dma control values does not updated to the dma hardware registers since new register value and cached values at the time of first plugin are same. Fixes: 7cb37b7bd0d3 ("ASoC: qcom: Add support for lpass hdmi driver") Signed-off-by: Srinivasa Rao Mandadapu Reported-by: Kuogee Hsieh Link: https://lore.kernel.org/r/1665637711-13300-1-git-send-email-quic_srivasam@quicinc.com Signed-off-by: Mark Brown --- sound/soc/qcom/lpass-cpu.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/soc/qcom/lpass-cpu.c b/sound/soc/qcom/lpass-cpu.c index 8a56f38dc7e8..99a3b4428591 100644 --- a/sound/soc/qcom/lpass-cpu.c +++ b/sound/soc/qcom/lpass-cpu.c @@ -782,10 +782,18 @@ static bool lpass_hdmi_regmap_volatile(struct device *dev, unsigned int reg) return true; if (reg == LPASS_HDMI_TX_LEGACY_ADDR(v)) return true; + if (reg == LPASS_HDMI_TX_VBIT_CTL_ADDR(v)) + return true; for (i = 0; i < v->hdmi_rdma_channels; ++i) { if (reg == LPAIF_HDMI_RDMACURR_REG(v, i)) return true; + if (reg == LPASS_HDMI_TX_DMA_ADDR(v, i)) + return true; + if (reg == LPASS_HDMI_TX_CH_LSB_ADDR(v, i)) + return true; + if (reg == LPASS_HDMI_TX_CH_MSB_ADDR(v, i)) + return true; } return false; } From a8dfb85095dd8b884ee962e64b16ef52bc54119d Mon Sep 17 00:00:00 2001 From: Yang Yingliang Date: Fri, 14 Oct 2022 09:36:40 +0800 Subject: [PATCH 15/50] ALSA: hda/realtek: simplify the return of comp_bind() After commit 23904f7b2518 ("ALSA: hda: cs35l41: Remove suspend/resume hda hooks"), the return of comp_bind() can be simplified. No functional changed. Signed-off-by: Yang Yingliang Link: https://lore.kernel.org/r/20221014013640.1142107-1-yangyingliang@huawei.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 7 +------ 1 file changed, 1 insertion(+), 6 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index e6c4bb5fa041..7c177426bf30 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6654,13 +6654,8 @@ static int comp_bind(struct device *dev) { struct hda_codec *cdc = dev_to_hda_codec(dev); struct alc_spec *spec = cdc->spec; - int ret; - ret = component_bind_all(dev, spec->comps); - if (ret) - return ret; - - return 0; + return component_bind_all(dev, spec->comps); } static void comp_unbind(struct device *dev) From 1013999b431b4bcdc1f5ae47dd3338122751db31 Mon Sep 17 00:00:00 2001 From: Siarhei Volkau Date: Sun, 16 Oct 2022 16:26:42 +0300 Subject: [PATCH 16/50] ASoC: codecs: jz4725b: add missed Line In power control bit Line In path stayed powered off during capturing or bypass to mixer. Signed-off-by: Siarhei Volkau Link: https://lore.kernel.org/r/20221016132648.3011729-2-lis8215@gmail.com Signed-off-by: Mark Brown --- sound/soc/codecs/jz4725b.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/jz4725b.c b/sound/soc/codecs/jz4725b.c index 5201a8f6d7b6..cc7a48c96aa4 100644 --- a/sound/soc/codecs/jz4725b.c +++ b/sound/soc/codecs/jz4725b.c @@ -236,7 +236,8 @@ static const struct snd_soc_dapm_widget jz4725b_codec_dapm_widgets[] = { SND_SOC_DAPM_MIXER("DAC to Mixer", JZ4725B_CODEC_REG_CR1, REG_CR1_DACSEL_OFFSET, 0, NULL, 0), - SND_SOC_DAPM_MIXER("Line In", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("Line In", JZ4725B_CODEC_REG_PMR1, + REG_PMR1_SB_LIN_OFFSET, 1, NULL, 0), SND_SOC_DAPM_MIXER("HP Out", JZ4725B_CODEC_REG_CR1, REG_CR1_HP_DIS_OFFSET, 1, NULL, 0), From 088777bf65b98cfa4b5378119d0a7d49a58ece44 Mon Sep 17 00:00:00 2001 From: Siarhei Volkau Date: Sun, 16 Oct 2022 16:26:43 +0300 Subject: [PATCH 17/50] ASoC: codecs: jz4725b: fix reported volume for Master ctl DAC volume control is the Master Playback Volume at the moment and it reports wrong levels in alsamixer and other alsa apps. The patch fixes that, as stated in manual on the jz4725b SoC (16.6.3.4 Programmable attenuation: GOD) the ctl range varies from -22.5dB to 0dB with 1.5dB step. Signed-off-by: Siarhei Volkau Link: https://lore.kernel.org/r/20221016132648.3011729-3-lis8215@gmail.com Signed-off-by: Mark Brown --- sound/soc/codecs/jz4725b.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/jz4725b.c b/sound/soc/codecs/jz4725b.c index cc7a48c96aa4..72549ee2e789 100644 --- a/sound/soc/codecs/jz4725b.c +++ b/sound/soc/codecs/jz4725b.c @@ -142,8 +142,8 @@ struct jz_icdc { struct clk *clk; }; -static const SNDRV_CTL_TLVD_DECLARE_DB_LINEAR(jz4725b_dac_tlv, -2250, 0); static const SNDRV_CTL_TLVD_DECLARE_DB_LINEAR(jz4725b_line_tlv, -1500, 600); +static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(jz4725b_dac_tlv, -2250, 150, 0); static const struct snd_kcontrol_new jz4725b_codec_controls[] = { SOC_DOUBLE_TLV("Master Playback Volume", From 1538e2c8c9b7e7a656effcc6e4e7cfe8c1b405fd Mon Sep 17 00:00:00 2001 From: Siarhei Volkau Date: Sun, 16 Oct 2022 16:26:44 +0300 Subject: [PATCH 18/50] ASoC: codecs: jz4725b: use right control for Capture Volume Line In Bypass control is used as Master Capture at the moment this is completely incorrect. Current control routed to Mixer instead of ADC, thus can't affect Capture path. ADC control shall be used instead. ADC volume control parameters are different, so the patch fixes that as well. Manual says (16.6.3.2 Programmable input attenuation amplifier: PGATM) that gain varies in range 0dB..22.5dB with 1.5dB step. Signed-off-by: Siarhei Volkau Link: https://lore.kernel.org/r/20221016132648.3011729-4-lis8215@gmail.com Signed-off-by: Mark Brown --- sound/soc/codecs/jz4725b.c | 15 +++++++++------ 1 file changed, 9 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/jz4725b.c b/sound/soc/codecs/jz4725b.c index 72549ee2e789..4363d898a7d4 100644 --- a/sound/soc/codecs/jz4725b.c +++ b/sound/soc/codecs/jz4725b.c @@ -136,13 +136,16 @@ enum { #define REG_CGR3_GO1L_OFFSET 0 #define REG_CGR3_GO1L_MASK (0x1f << REG_CGR3_GO1L_OFFSET) +#define REG_CGR10_GIL_OFFSET 0 +#define REG_CGR10_GIR_OFFSET 4 + struct jz_icdc { struct regmap *regmap; void __iomem *base; struct clk *clk; }; -static const SNDRV_CTL_TLVD_DECLARE_DB_LINEAR(jz4725b_line_tlv, -1500, 600); +static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(jz4725b_adc_tlv, 0, 150, 0); static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(jz4725b_dac_tlv, -2250, 150, 0); static const struct snd_kcontrol_new jz4725b_codec_controls[] = { @@ -151,11 +154,11 @@ static const struct snd_kcontrol_new jz4725b_codec_controls[] = { REG_CGR1_GODL_OFFSET, REG_CGR1_GODR_OFFSET, 0xf, 1, jz4725b_dac_tlv), - SOC_DOUBLE_R_TLV("Master Capture Volume", - JZ4725B_CODEC_REG_CGR3, - JZ4725B_CODEC_REG_CGR2, - REG_CGR2_GO1R_OFFSET, - 0x1f, 1, jz4725b_line_tlv), + SOC_DOUBLE_TLV("Master Capture Volume", + JZ4725B_CODEC_REG_CGR10, + REG_CGR10_GIL_OFFSET, + REG_CGR10_GIR_OFFSET, + 0xf, 0, jz4725b_adc_tlv), SOC_SINGLE("Master Playback Switch", JZ4725B_CODEC_REG_CR1, REG_CR1_DAC_MUTE_OFFSET, 1, 1), From 80852f8268769715db335a22305e81a0c4a38a84 Mon Sep 17 00:00:00 2001 From: Siarhei Volkau Date: Sun, 16 Oct 2022 16:26:45 +0300 Subject: [PATCH 19/50] ASoC: codecs: jz4725b: fix capture selector naming At the moment Capture source selector appears on Playback tab in the alsamixer and has a senseless name. Let's fix that. Signed-off-by: Siarhei Volkau Link: https://lore.kernel.org/r/20221016132648.3011729-5-lis8215@gmail.com Signed-off-by: Mark Brown --- sound/soc/codecs/jz4725b.c | 14 +++++++------- 1 file changed, 7 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/jz4725b.c b/sound/soc/codecs/jz4725b.c index 4363d898a7d4..d57c2c6a3add 100644 --- a/sound/soc/codecs/jz4725b.c +++ b/sound/soc/codecs/jz4725b.c @@ -183,7 +183,7 @@ static SOC_VALUE_ENUM_SINGLE_DECL(jz4725b_codec_adc_src_enum, jz4725b_codec_adc_src_texts, jz4725b_codec_adc_src_values); static const struct snd_kcontrol_new jz4725b_codec_adc_src_ctrl = - SOC_DAPM_ENUM("Route", jz4725b_codec_adc_src_enum); + SOC_DAPM_ENUM("ADC Source Capture Route", jz4725b_codec_adc_src_enum); static const struct snd_kcontrol_new jz4725b_codec_mixer_controls[] = { SOC_DAPM_SINGLE("Line In Bypass", JZ4725B_CODEC_REG_CR1, @@ -228,7 +228,7 @@ static const struct snd_soc_dapm_widget jz4725b_codec_dapm_widgets[] = { SND_SOC_DAPM_ADC("ADC", "Capture", JZ4725B_CODEC_REG_PMR1, REG_PMR1_SB_ADC_OFFSET, 1), - SND_SOC_DAPM_MUX("ADC Source", SND_SOC_NOPM, 0, 0, + SND_SOC_DAPM_MUX("ADC Source Capture Route", SND_SOC_NOPM, 0, 0, &jz4725b_codec_adc_src_ctrl), /* Mixer */ @@ -287,11 +287,11 @@ static const struct snd_soc_dapm_route jz4725b_codec_dapm_routes[] = { {"Mixer", NULL, "DAC to Mixer"}, {"Mixer to ADC", NULL, "Mixer"}, - {"ADC Source", "Mixer", "Mixer to ADC"}, - {"ADC Source", "Line In", "Line In"}, - {"ADC Source", "Mic 1", "Mic 1"}, - {"ADC Source", "Mic 2", "Mic 2"}, - {"ADC", NULL, "ADC Source"}, + {"ADC Source Capture Route", "Mixer", "Mixer to ADC"}, + {"ADC Sourc Capture Routee", "Line In", "Line In"}, + {"ADC Source Capture Route", "Mic 1", "Mic 1"}, + {"ADC Source Capture Route", "Mic 2", "Mic 2"}, + {"ADC", NULL, "ADC Source Capture Route"}, {"Out Stage", NULL, "Mixer"}, {"HP Out", NULL, "Out Stage"}, From 4e8ff35878685291978b93543d6b9e9290be770a Mon Sep 17 00:00:00 2001 From: Geert Uytterhoeven Date: Sun, 16 Oct 2022 10:33:50 +0200 Subject: [PATCH 20/50] ASoC: codecs: tlv320adc3xxx: Wrap adc3xxx_i2c_remove() in __exit_p() If CONFIG_SND_SOC_TLV320ADC3XXX=y: `.exit.text' referenced in section `.data' of sound/soc/codecs/tlv320adc3xxx.o: defined in discarded section `.exit.text' of sound/soc/codecs/tlv320adc3xxx.o Fix this by wrapping the adc3xxx_i2c_remove() pointer in __exit_p(). Fixes: e9a3b57efd28fe88 ("ASoC: codec: tlv320adc3xxx: New codec driver") Signed-off-by: Geert Uytterhoeven Link: https://lore.kernel.org/r/3225ba4cfe558d9380155e75385954dd21d4e7eb.1665909132.git.geert@linux-m68k.org Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320adc3xxx.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/tlv320adc3xxx.c b/sound/soc/codecs/tlv320adc3xxx.c index 748998e48af9..8a0965cd3e66 100644 --- a/sound/soc/codecs/tlv320adc3xxx.c +++ b/sound/soc/codecs/tlv320adc3xxx.c @@ -1450,7 +1450,7 @@ static struct i2c_driver adc3xxx_i2c_driver = { .of_match_table = tlv320adc3xxx_of_match, }, .probe_new = adc3xxx_i2c_probe, - .remove = adc3xxx_i2c_remove, + .remove = __exit_p(adc3xxx_i2c_remove), .id_table = adc3xxx_i2c_id, }; From 9a7f2c9e7a19b16b4409f372cf2e16e4334cdca2 Mon Sep 17 00:00:00 2001 From: Randy Dunlap Date: Fri, 14 Oct 2022 17:12:28 -0700 Subject: [PATCH 21/50] ASoC: qcom: SND_SOC_SC7180 optionally depends on SOUNDWIRE If SOUNDWIRE is enabled, then SND_SOC_SC7180 should depend on SOUNDWIRE to prevent SOUNDWIRE=m and SND_SOC_SC7180=y, which causes build errors: s390-linux-ld: sound/soc/qcom/common.o: in function `qcom_snd_sdw_prepare': common.c:(.text+0x140): undefined reference to `sdw_disable_stream' s390-linux-ld: common.c:(.text+0x14a): undefined reference to `sdw_deprepare_stream' s390-linux-ld: common.c:(.text+0x158): undefined reference to `sdw_prepare_stream' s390-linux-ld: common.c:(.text+0x16a): undefined reference to `sdw_enable_stream' s390-linux-ld: common.c:(.text+0x17c): undefined reference to `sdw_deprepare_stream' s390-linux-ld: sound/soc/qcom/common.o: in function `qcom_snd_sdw_hw_free': common.c:(.text+0x344): undefined reference to `sdw_disable_stream' s390-linux-ld: common.c:(.text+0x34e): undefined reference to `sdw_deprepare_stream' Fixes: 3bd975f3ae0a ("ASoC: qcom: sm8250: move some code to common") Fixes: 9e3ecb5b1681 ("ASoC: qcom: sc7180: Add machine driver for sound card registration") Signed-off-by: Randy Dunlap Reported-by: kernel test robot Cc: Srinivas Kandagatla Cc: Banajit Goswami Cc: Mark Brown Cc: Liam Girdwood Cc: Ajit Pandey Cc: Cheng-Yi Chiang Cc: Jaroslav Kysela Cc: Takashi Iwai Cc: stable@vger.kernel.org Cc: alsa-devel@alsa-project.org Link: https://lore.kernel.org/r/20221015001228.18990-1-rdunlap@infradead.org Signed-off-by: Mark Brown --- sound/soc/qcom/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/qcom/Kconfig b/sound/soc/qcom/Kconfig index d0e59e07b1fc..8c7398bc1ca8 100644 --- a/sound/soc/qcom/Kconfig +++ b/sound/soc/qcom/Kconfig @@ -187,6 +187,7 @@ config SND_SOC_SC8280XP config SND_SOC_SC7180 tristate "SoC Machine driver for SC7180 boards" depends on I2C && GPIOLIB + depends on SOUNDWIRE || SOUNDWIRE=n select SND_SOC_QCOM_COMMON select SND_SOC_LPASS_SC7180 select SND_SOC_MAX98357A From 491a4ccd8a0258392900c80c6b2b622c7115fc23 Mon Sep 17 00:00:00 2001 From: Stefan Binding Date: Tue, 18 Oct 2022 13:15:06 +0100 Subject: [PATCH 22/50] ALSA: hda/realtek: Add quirk for ASUS Zenbook using CS35L41 This Asus Zenbook laptop use Realtek HDA codec combined with 2xCS35L41 Amplifiers using SPI with External Boost. Signed-off-by: Stefan Binding Cc: Link: https://lore.kernel.org/r/20221018121506.2561397-1-sbinding@opensource.cirrus.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 7c177426bf30..79acd2a2caf2 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9395,6 +9395,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1ccd, "ASUS X555UB", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x1d42, "ASUS Zephyrus G14 2022", ALC289_FIXUP_ASUS_GA401), SND_PCI_QUIRK(0x1043, 0x1d4e, "ASUS TM420", ALC256_FIXUP_ASUS_HPE), + SND_PCI_QUIRK(0x1043, 0x1e02, "ASUS UX3402", ALC245_FIXUP_CS35L41_SPI_2), SND_PCI_QUIRK(0x1043, 0x1e11, "ASUS Zephyrus G15", ALC289_FIXUP_ASUS_GA502), SND_PCI_QUIRK(0x1043, 0x1e51, "ASUS Zephyrus M15", ALC294_FIXUP_ASUS_GU502_PINS), SND_PCI_QUIRK(0x1043, 0x1e5e, "ASUS ROG Strix G513", ALC294_FIXUP_ASUS_G513_PINS), From 41deb2db64997d01110faaf763bd911d490dfde7 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Mon, 17 Oct 2022 15:40:54 -0500 Subject: [PATCH 23/50] ASoC: Intel: sof_sdw: add quirk variant for LAPBC710 NUC15 Some NUC15 LAPBC710 devices don't expose the same DMI information as the Intel reference, add additional entry in the match table. BugLink: https://github.com/thesofproject/linux/issues/3885 Signed-off-by: Pierre-Louis Bossart Reviewed-by: Ranjani Sridharan Reviewed-by: Bard Liao Link: https://lore.kernel.org/r/20221017204054.207512-1-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_sdw.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index a49bfaab6b21..5223089c3426 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -202,6 +202,17 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { SOF_SDW_PCH_DMIC | RT711_JD1), }, + { + /* NUC15 LAPBC710 skews */ + .callback = sof_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "Intel Corporation"), + DMI_MATCH(DMI_BOARD_NAME, "LAPBC710"), + }, + .driver_data = (void *)(SOF_SDW_TGL_HDMI | + SOF_SDW_PCH_DMIC | + RT711_JD1), + }, /* TigerLake-SDCA devices */ { .callback = sof_sdw_quirk_cb, From 73189c064e11137c8b78a825800a374924ebb7b7 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Mon, 17 Oct 2022 15:40:04 -0500 Subject: [PATCH 24/50] ASoC: SOF: Intel: pci-mtl: fix firmware name Initial IPC4 tests used the same conventions as previous reference closed-source firmware, but for MeteorLake the convention is the same as previous SOF releases (sof-.ri). Only the prefix changes to avoid confusions between IPC types. This change has no impact on users since the firmware has not yet been released. Fixes: 064520e8aeaa2 ("ASoC: SOF: Intel: Add support for MeteorLake (MTL)") Signed-off-by: Pierre-Louis Bossart Reviewed-by: Chao Song Reviewed-by: Kai Vehmanen Reviewed-by: Ranjani Sridharan Link: https://lore.kernel.org/r/20221017204004.207446-1-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/pci-mtl.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sof/intel/pci-mtl.c b/sound/soc/sof/intel/pci-mtl.c index 899b00d53d64..9f39da984e9f 100644 --- a/sound/soc/sof/intel/pci-mtl.c +++ b/sound/soc/sof/intel/pci-mtl.c @@ -38,7 +38,7 @@ static const struct sof_dev_desc mtl_desc = { [SOF_INTEL_IPC4] = "intel/sof-ace-tplg", }, .default_fw_filename = { - [SOF_INTEL_IPC4] = "dsp_basefw.bin", + [SOF_INTEL_IPC4] = "sof-mtl.ri", }, .nocodec_tplg_filename = "sof-mtl-nocodec.tplg", .ops = &sof_mtl_ops, From b4dd2e3758709aa8a2abd1ac34c56bd09b980039 Mon Sep 17 00:00:00 2001 From: Yong Zhi Date: Mon, 17 Oct 2022 15:57:28 -0500 Subject: [PATCH 25/50] ASoC: Intel: sof_rt5682: Add quirk for Rex board Add mtl_mx98357_rt5682 driver data for Chrome Rex board support. Reviewed-by: Bard Liao Reviewed-by: Curtis Malainey Signed-off-by: Yong Zhi Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20221017205728.210813-1-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_rt5682.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index 2d0986824b3d..2358be208c1f 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -223,6 +223,18 @@ static const struct dmi_system_id sof_rt5682_quirk_table[] = { SOF_RT5682_SSP_AMP(2) | SOF_RT5682_NUM_HDMIDEV(4)), }, + { + .callback = sof_rt5682_quirk_cb, + .matches = { + DMI_MATCH(DMI_PRODUCT_FAMILY, "Google_Rex"), + }, + .driver_data = (void *)(SOF_RT5682_MCLK_EN | + SOF_RT5682_SSP_CODEC(2) | + SOF_SPEAKER_AMP_PRESENT | + SOF_RT5682_SSP_AMP(0) | + SOF_RT5682_NUM_HDMIDEV(4) + ), + }, {} }; From af6514f2f3828dc39c96cd4686ef5c9d8368626f Mon Sep 17 00:00:00 2001 From: Kai Vehmanen Date: Tue, 18 Oct 2022 15:13:32 +0300 Subject: [PATCH 26/50] ASoC: SOF: ipc4-mtrace: protect per-core nodes against multiple open MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Add protection against multiple open of the mtrace/coreN debugfs nodes. This is not supported in the implementation, and this will show up as unexpected behaviour of the interface, and potential use of already freed memory. Fixes: f4ea22f7aa75 ("ASoC: SOF: ipc4: Add support for mtrace log extraction") Signed-off-by: Kai Vehmanen Reviewed-by: Péter Ujfalusi Reviewed-by: Pierre-Louis Bossart Reviewed-by: Ranjani Sridharan Link: https://lore.kernel.org/r/20221018121332.20802-1-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/ipc4-mtrace.c | 20 ++++++++++++++++++-- 1 file changed, 18 insertions(+), 2 deletions(-) diff --git a/sound/soc/sof/ipc4-mtrace.c b/sound/soc/sof/ipc4-mtrace.c index 9c7080041d08..70dea8ae706e 100644 --- a/sound/soc/sof/ipc4-mtrace.c +++ b/sound/soc/sof/ipc4-mtrace.c @@ -108,6 +108,7 @@ struct sof_mtrace_core_data { int id; u32 slot_offset; void *log_buffer; + struct mutex buffer_lock; /* for log_buffer alloc/free */ u32 host_read_ptr; u32 dsp_write_ptr; /* pos update IPC arrived before the slot offset is known, queried */ @@ -128,14 +129,22 @@ static int sof_ipc4_mtrace_dfs_open(struct inode *inode, struct file *file) struct sof_mtrace_core_data *core_data = inode->i_private; int ret; + mutex_lock(&core_data->buffer_lock); + + if (core_data->log_buffer) { + ret = -EBUSY; + goto out; + } + ret = debugfs_file_get(file->f_path.dentry); if (unlikely(ret)) - return ret; + goto out; core_data->log_buffer = kmalloc(SOF_MTRACE_SLOT_SIZE, GFP_KERNEL); if (!core_data->log_buffer) { debugfs_file_put(file->f_path.dentry); - return -ENOMEM; + ret = -ENOMEM; + goto out; } ret = simple_open(inode, file); @@ -144,6 +153,9 @@ static int sof_ipc4_mtrace_dfs_open(struct inode *inode, struct file *file) debugfs_file_put(file->f_path.dentry); } +out: + mutex_unlock(&core_data->buffer_lock); + return ret; } @@ -280,7 +292,10 @@ static int sof_ipc4_mtrace_dfs_release(struct inode *inode, struct file *file) debugfs_file_put(file->f_path.dentry); + mutex_lock(&core_data->buffer_lock); kfree(core_data->log_buffer); + core_data->log_buffer = NULL; + mutex_unlock(&core_data->buffer_lock); return 0; } @@ -563,6 +578,7 @@ static int ipc4_mtrace_init(struct snd_sof_dev *sdev) struct sof_mtrace_core_data *core_data = &priv->cores[i]; init_waitqueue_head(&core_data->trace_sleep); + mutex_init(&core_data->buffer_lock); core_data->sdev = sdev; core_data->id = i; } From 00aaf8bfe0ee2b807b452df806d725e080d85404 Mon Sep 17 00:00:00 2001 From: Shuming Fan Date: Wed, 19 Oct 2022 17:57:31 +0800 Subject: [PATCH 27/50] ASoC: rt1308-sdw: update the preset settings This patch updates the pad control and checks the hardware version to set the different preset settings. Signed-off-by: Shuming Fan Link: https://lore.kernel.org/r/20221019095731.31101-1-shumingf@realtek.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt1308-sdw.c | 17 ++++++++++++++--- sound/soc/codecs/rt1308-sdw.h | 1 + sound/soc/codecs/rt1308.h | 5 +++++ 3 files changed, 20 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/rt1308-sdw.c b/sound/soc/codecs/rt1308-sdw.c index 5c29416aa781..f99aed353f10 100644 --- a/sound/soc/codecs/rt1308-sdw.c +++ b/sound/soc/codecs/rt1308-sdw.c @@ -50,6 +50,7 @@ static bool rt1308_volatile_register(struct device *dev, unsigned int reg) case 0x3008: case 0x300a: case 0xc000: + case 0xc710: case 0xc860 ... 0xc863: case 0xc870 ... 0xc873: return true; @@ -200,6 +201,7 @@ static int rt1308_io_init(struct device *dev, struct sdw_slave *slave) { struct rt1308_sdw_priv *rt1308 = dev_get_drvdata(dev); int ret = 0; + unsigned int tmp; if (rt1308->hw_init) return 0; @@ -231,6 +233,10 @@ static int rt1308_io_init(struct device *dev, struct sdw_slave *slave) /* sw reset */ regmap_write(rt1308->regmap, RT1308_SDW_RESET, 0); + regmap_read(rt1308->regmap, 0xc710, &tmp); + rt1308->hw_ver = tmp; + dev_dbg(dev, "%s, hw_ver=0x%x\n", __func__, rt1308->hw_ver); + /* initial settings */ regmap_write(rt1308->regmap, 0xc103, 0xc0); regmap_write(rt1308->regmap, 0xc030, 0x17); @@ -246,8 +252,14 @@ static int rt1308_io_init(struct device *dev, struct sdw_slave *slave) regmap_write(rt1308->regmap, 0xc062, 0x05); regmap_write(rt1308->regmap, 0xc171, 0x07); regmap_write(rt1308->regmap, 0xc173, 0x0d); - regmap_write(rt1308->regmap, 0xc311, 0x7f); - regmap_write(rt1308->regmap, 0xc900, 0x90); + if (rt1308->hw_ver == RT1308_VER_C) { + regmap_write(rt1308->regmap, 0xc311, 0x7f); + regmap_write(rt1308->regmap, 0xc300, 0x09); + } else { + regmap_write(rt1308->regmap, 0xc311, 0x4f); + regmap_write(rt1308->regmap, 0xc300, 0x0b); + } + regmap_write(rt1308->regmap, 0xc900, 0x5a); regmap_write(rt1308->regmap, 0xc1a0, 0x84); regmap_write(rt1308->regmap, 0xc1a1, 0x01); regmap_write(rt1308->regmap, 0xc360, 0x78); @@ -257,7 +269,6 @@ static int rt1308_io_init(struct device *dev, struct sdw_slave *slave) regmap_write(rt1308->regmap, 0xc070, 0x00); regmap_write(rt1308->regmap, 0xc100, 0xd7); regmap_write(rt1308->regmap, 0xc101, 0xd7); - regmap_write(rt1308->regmap, 0xc300, 0x09); if (rt1308->first_hw_init) { regcache_cache_bypass(rt1308->regmap, false); diff --git a/sound/soc/codecs/rt1308-sdw.h b/sound/soc/codecs/rt1308-sdw.h index 6668e19d85d4..f88f52e8917e 100644 --- a/sound/soc/codecs/rt1308-sdw.h +++ b/sound/soc/codecs/rt1308-sdw.h @@ -163,6 +163,7 @@ struct rt1308_sdw_priv { bool first_hw_init; int rx_mask; int slots; + int hw_ver; }; struct sdw_stream_data { diff --git a/sound/soc/codecs/rt1308.h b/sound/soc/codecs/rt1308.h index ff7c423e879e..d3a0f91630ca 100644 --- a/sound/soc/codecs/rt1308.h +++ b/sound/soc/codecs/rt1308.h @@ -286,4 +286,9 @@ enum { RT1308_AIFS }; +enum rt1308_hw_ver { + RT1308_VER_C = 2, + RT1308_VER_D +}; + #endif /* end of _RT1308_H_ */ From 75d8b1662ca5c20cf8365575222abaef18ff1f50 Mon Sep 17 00:00:00 2001 From: Shuming Fan Date: Wed, 19 Oct 2022 17:57:15 +0800 Subject: [PATCH 28/50] ASoC: rt1308-sdw: add the default value of some registers The driver missed the default value of register 0xc070/0xc360. This patch adds that default value to avoid invalid register access when the device doesn't be enumerated yet. BugLink: https://github.com/thesofproject/linux/issues/3924 Signed-off-by: Shuming Fan Link: https://lore.kernel.org/r/20221019095715.31082-1-shumingf@realtek.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt1308-sdw.h | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/rt1308-sdw.h b/sound/soc/codecs/rt1308-sdw.h index f88f52e8917e..62ce27799307 100644 --- a/sound/soc/codecs/rt1308-sdw.h +++ b/sound/soc/codecs/rt1308-sdw.h @@ -139,10 +139,12 @@ static const struct reg_default rt1308_reg_defaults[] = { { 0x3005, 0x23 }, { 0x3008, 0x02 }, { 0x300a, 0x00 }, + { 0xc000 | (RT1308_DATA_PATH << 4), 0x00 }, { 0xc003 | (RT1308_DAC_SET << 4), 0x00 }, { 0xc000 | (RT1308_POWER << 4), 0x00 }, { 0xc001 | (RT1308_POWER << 4), 0x00 }, { 0xc002 | (RT1308_POWER << 4), 0x00 }, + { 0xc000 | (RT1308_POWER_STATUS << 4), 0x00 }, }; #define RT1308_SDW_OFFSET 0xc000 From 32def55d237e8507d4eb8442628fc2e59a899ea0 Mon Sep 17 00:00:00 2001 From: Aidan MacDonald Date: Wed, 19 Oct 2022 02:23:02 +0100 Subject: [PATCH 29/50] ASoC: simple-card: Fix up checks for HW param fixups The "convert-xxx" properties only have an effect for DPCM DAI links. A DAI link is only created as DPCM if the device tree requires it; part of this involves checking for the use of "convert-xxx" properties. When the convert-sample-format property was added, the checks got out of sync. A DAI link that specified only convert-sample-format but did not pass any of the other DPCM checks would not go into DPCM mode and the convert-sample-format property would be silently ignored. Fix this by adding a function to do the "convert-xxx" property checks, instead of open-coding it in simple-card and audio-graph-card. And add "convert-sample-format" to the check function so that DAI links using it will be initialized correctly. Fixes: 047a05366f4b ("ASoC: simple-card-utils: Fixup DAI sample format") Acked-by: Kuninori Morimoto Signed-off-by: Aidan MacDonald Acked-by: Sameer Pujar Link: https://lore.kernel.org/r/20221019012302.633830-1-aidanmacdonald.0x0@gmail.com Signed-off-by: Mark Brown --- include/sound/simple_card_utils.h | 1 + sound/soc/generic/audio-graph-card.c | 2 +- sound/soc/generic/simple-card-utils.c | 15 +++++++++++++++ sound/soc/generic/simple-card.c | 3 +-- 4 files changed, 18 insertions(+), 3 deletions(-) diff --git a/include/sound/simple_card_utils.h b/include/sound/simple_card_utils.h index a0b827f0c2f6..25e049f44178 100644 --- a/include/sound/simple_card_utils.h +++ b/include/sound/simple_card_utils.h @@ -177,6 +177,7 @@ void asoc_simple_convert_fixup(struct asoc_simple_data *data, struct snd_pcm_hw_params *params); void asoc_simple_parse_convert(struct device_node *np, char *prefix, struct asoc_simple_data *data); +bool asoc_simple_is_convert_required(const struct asoc_simple_data *data); int asoc_simple_parse_routing(struct snd_soc_card *card, char *prefix); diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c index b327372f2e4a..fe7cf972d44c 100644 --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -417,7 +417,7 @@ static inline bool parse_as_dpcm_link(struct asoc_simple_priv *priv, * or has convert-xxx property */ if ((of_get_child_count(codec_port) > 1) || - (adata->convert_rate || adata->convert_channels)) + asoc_simple_is_convert_required(adata)) return true; return false; diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index bef16833c487..be69bbc47f81 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -85,6 +85,21 @@ void asoc_simple_parse_convert(struct device_node *np, } EXPORT_SYMBOL_GPL(asoc_simple_parse_convert); +/** + * asoc_simple_is_convert_required() - Query if HW param conversion was requested + * @data: Link data. + * + * Returns true if any HW param conversion was requested for this DAI link with + * any "convert-xxx" properties. + */ +bool asoc_simple_is_convert_required(const struct asoc_simple_data *data) +{ + return data->convert_rate || + data->convert_channels || + data->convert_sample_format; +} +EXPORT_SYMBOL_GPL(asoc_simple_is_convert_required); + int asoc_simple_parse_daifmt(struct device *dev, struct device_node *node, struct device_node *codec, diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 78419e18717d..feb55b66239b 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -393,8 +393,7 @@ static int __simple_for_each_link(struct asoc_simple_priv *priv, * or has convert-xxx property */ if (dpcm_selectable && - (num > 2 || - adata.convert_rate || adata.convert_channels)) { + (num > 2 || asoc_simple_is_convert_required(&adata))) { /* * np * |1(CPU)|0(Codec) li->cpu From df496157a5afa1b6d1f4c46ad6549c2c346d1e59 Mon Sep 17 00:00:00 2001 From: Colin Ian King Date: Wed, 19 Oct 2022 08:16:39 +0100 Subject: [PATCH 30/50] ASoC: codecs: jz4725b: Fix spelling mistake "Sourc" -> "Source", "Routee" -> "Route" MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit There are two spelling mistakes in codec routing description. Fix it. Signed-off-by: Colin Ian King Reviewed-by: Philippe Mathieu-Daudé Acked-by: Paul Cercueil Link: https://lore.kernel.org/r/20221019071639.1003730-1-colin.i.king@gmail.com Signed-off-by: Mark Brown --- sound/soc/codecs/jz4725b.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/jz4725b.c b/sound/soc/codecs/jz4725b.c index d57c2c6a3add..71ea576f7e67 100644 --- a/sound/soc/codecs/jz4725b.c +++ b/sound/soc/codecs/jz4725b.c @@ -288,7 +288,7 @@ static const struct snd_soc_dapm_route jz4725b_codec_dapm_routes[] = { {"Mixer to ADC", NULL, "Mixer"}, {"ADC Source Capture Route", "Mixer", "Mixer to ADC"}, - {"ADC Sourc Capture Routee", "Line In", "Line In"}, + {"ADC Source Capture Route", "Line In", "Line In"}, {"ADC Source Capture Route", "Mic 1", "Mic 1"}, {"ADC Source Capture Route", "Mic 2", "Mic 2"}, {"ADC", NULL, "ADC Source Capture Route"}, From a450b5c8739248069e11f72129fca61a56125577 Mon Sep 17 00:00:00 2001 From: linkt Date: Tue, 11 Oct 2022 10:51:36 +0800 Subject: [PATCH 31/50] ASoC: amd: yc: Adding Lenovo ThinkBook 14 Gen 4+ ARA and Lenovo ThinkBook 16 Gen 4+ ARA to the Quirks List Lenovo ThinkBook 14 Gen 4+ ARA and ThinkBook 16 Gen 4+ ARA need to be added to the list of quirks for the microphone to work properly. Signed-off-by: linkt Reviewed-by: Mario Limonciello Link: https://lore.kernel.org/r/MEYPR01MB8397A3C27DE6206FA3EF834DB6239@MEYPR01MB8397.ausprd01.prod.outlook.com Signed-off-by: Mark Brown --- sound/soc/amd/yc/acp6x-mach.c | 14 ++++++++++++++ 1 file changed, 14 insertions(+) diff --git a/sound/soc/amd/yc/acp6x-mach.c b/sound/soc/amd/yc/acp6x-mach.c index 2cb50d5cf1a9..09a8aceff22f 100644 --- a/sound/soc/amd/yc/acp6x-mach.c +++ b/sound/soc/amd/yc/acp6x-mach.c @@ -45,6 +45,20 @@ static struct snd_soc_card acp6x_card = { }; static const struct dmi_system_id yc_acp_quirk_table[] = { + { + .driver_data = &acp6x_card, + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"), + DMI_MATCH(DMI_PRODUCT_NAME, "21D0"), + } + }, + { + .driver_data = &acp6x_card, + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"), + DMI_MATCH(DMI_PRODUCT_NAME, "21D1"), + } + }, { .driver_data = &acp6x_card, .matches = { From 1dd5166102e7ca91e8c5d833110333835e147ddb Mon Sep 17 00:00:00 2001 From: Srinivasa Rao Mandadapu Date: Sat, 15 Oct 2022 14:48:50 +0530 Subject: [PATCH 32/50] ASoC: qcom: lpass-cpu: Mark HDMI TX parity register as volatile Update LPASS_HDMI_TX_PARITY_ADDR register as volatile, to fix dp audio failures observed with some of external monitors. Fixes: 7cb37b7bd0d3 ("ASoC: qcom: Add support for lpass hdmi driver") Signed-off-by: Srinivasa Rao Mandadapu Reviewed-by: Stephen Boyd Link: https://lore.kernel.org/r/1665825530-7593-1-git-send-email-quic_srivasam@quicinc.com Signed-off-by: Mark Brown --- sound/soc/qcom/lpass-cpu.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/qcom/lpass-cpu.c b/sound/soc/qcom/lpass-cpu.c index 99a3b4428591..54353842dc07 100644 --- a/sound/soc/qcom/lpass-cpu.c +++ b/sound/soc/qcom/lpass-cpu.c @@ -784,6 +784,8 @@ static bool lpass_hdmi_regmap_volatile(struct device *dev, unsigned int reg) return true; if (reg == LPASS_HDMI_TX_VBIT_CTL_ADDR(v)) return true; + if (reg == LPASS_HDMI_TX_PARITY_ADDR(v)) + return true; for (i = 0; i < v->hdmi_rdma_channels; ++i) { if (reg == LPAIF_HDMI_RDMACURR_REG(v, i)) From 05de5cf6fb7d73d2bf0a0c882433f31db5c93f63 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Wed, 19 Oct 2022 10:49:26 -0500 Subject: [PATCH 33/50] ASoC: SOF: Intel: pci-tgl: fix ADL-N descriptor ADL-N uses a different signing key, which means we can't reuse the regular ADL descriptor used for ADL-P/M/S. Fixes: cd57eb3c403cb ("ASoC: SOF: Intel: pci-tgl: add ADL-N support") Signed-off-by: Pierre-Louis Bossart Reviewed-by: Ranjani Sridharan Reviewed-by: Chao Song Reviewed-by: Bard Liao Link: https://lore.kernel.org/r/20221019154926.163539-1-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/pci-tgl.c | 30 +++++++++++++++++++++++++++++- 1 file changed, 29 insertions(+), 1 deletion(-) diff --git a/sound/soc/sof/intel/pci-tgl.c b/sound/soc/sof/intel/pci-tgl.c index 2d63cc236a68..4cfe4f242fc5 100644 --- a/sound/soc/sof/intel/pci-tgl.c +++ b/sound/soc/sof/intel/pci-tgl.c @@ -159,6 +159,34 @@ static const struct sof_dev_desc adl_desc = { .ops_init = sof_tgl_ops_init, }; +static const struct sof_dev_desc adl_n_desc = { + .machines = snd_soc_acpi_intel_adl_machines, + .alt_machines = snd_soc_acpi_intel_adl_sdw_machines, + .use_acpi_target_states = true, + .resindex_lpe_base = 0, + .resindex_pcicfg_base = -1, + .resindex_imr_base = -1, + .irqindex_host_ipc = -1, + .chip_info = &tgl_chip_info, + .ipc_supported_mask = BIT(SOF_IPC) | BIT(SOF_INTEL_IPC4), + .ipc_default = SOF_IPC, + .default_fw_path = { + [SOF_IPC] = "intel/sof", + [SOF_INTEL_IPC4] = "intel/avs/adl-n", + }, + .default_tplg_path = { + [SOF_IPC] = "intel/sof-tplg", + [SOF_INTEL_IPC4] = "intel/avs-tplg", + }, + .default_fw_filename = { + [SOF_IPC] = "sof-adl-n.ri", + [SOF_INTEL_IPC4] = "dsp_basefw.bin", + }, + .nocodec_tplg_filename = "sof-adl-nocodec.tplg", + .ops = &sof_tgl_ops, + .ops_init = sof_tgl_ops_init, +}; + static const struct sof_dev_desc rpls_desc = { .machines = snd_soc_acpi_intel_rpl_machines, .alt_machines = snd_soc_acpi_intel_rpl_sdw_machines, @@ -246,7 +274,7 @@ static const struct pci_device_id sof_pci_ids[] = { { PCI_DEVICE(0x8086, 0x51cf), /* RPL-PX */ .driver_data = (unsigned long)&rpl_desc}, { PCI_DEVICE(0x8086, 0x54c8), /* ADL-N */ - .driver_data = (unsigned long)&adl_desc}, + .driver_data = (unsigned long)&adl_n_desc}, { 0, } }; MODULE_DEVICE_TABLE(pci, sof_pci_ids); From 4881bda5ea05c8c240fc8afeaa928e2bc43f61fa Mon Sep 17 00:00:00 2001 From: Yang Yingliang Date: Wed, 19 Oct 2022 17:30:25 +0800 Subject: [PATCH 34/50] ALSA: ac97: fix possible memory leak in snd_ac97_dev_register() If device_register() fails in snd_ac97_dev_register(), it should call put_device() to give up reference, or the name allocated in dev_set_name() is leaked. Fixes: 0ca06a00e206 ("[ALSA] AC97 bus interface for ad-hoc drivers") Signed-off-by: Yang Yingliang Link: https://lore.kernel.org/r/20221019093025.1179475-1-yangyingliang@huawei.com Signed-off-by: Takashi Iwai --- sound/pci/ac97/ac97_codec.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c index cb60a07d39a8..ceead55f13ab 100644 --- a/sound/pci/ac97/ac97_codec.c +++ b/sound/pci/ac97/ac97_codec.c @@ -2009,6 +2009,7 @@ static int snd_ac97_dev_register(struct snd_device *device) err = device_register(&ac97->dev); if (err < 0) { ac97_err(ac97, "Can't register ac97 bus\n"); + put_device(&ac97->dev); ac97->dev.bus = NULL; return err; } From 966f015fe4329199cc49084ee2886cfb626b34d3 Mon Sep 17 00:00:00 2001 From: "Maciej S. Szmigiero" Date: Thu, 20 Oct 2022 22:46:21 +0200 Subject: [PATCH 35/50] ALSA: control: add snd_ctl_rename() Add a snd_ctl_rename() function that takes care of updating the control hash entries for callers that already have the relevant struct snd_kcontrol at hand and hold the control write lock (or simply haven't registered the card yet). Fixes: c27e1efb61c5 ("ALSA: control: Use xarray for faster lookups") Cc: stable@vger.kernel.org Signed-off-by: Maciej S. Szmigiero Link: https://lore.kernel.org/r/4170b71117ea81357a4f7eb8410f7cde20836c70.1666296963.git.maciej.szmigiero@oracle.com Signed-off-by: Takashi Iwai --- include/sound/control.h | 1 + sound/core/control.c | 23 +++++++++++++++++++++++ 2 files changed, 24 insertions(+) diff --git a/include/sound/control.h b/include/sound/control.h index eae443ba79ba..cc3dcc6cfb0f 100644 --- a/include/sound/control.h +++ b/include/sound/control.h @@ -138,6 +138,7 @@ int snd_ctl_remove(struct snd_card * card, struct snd_kcontrol * kcontrol); int snd_ctl_replace(struct snd_card *card, struct snd_kcontrol *kcontrol, bool add_on_replace); int snd_ctl_remove_id(struct snd_card * card, struct snd_ctl_elem_id *id); int snd_ctl_rename_id(struct snd_card * card, struct snd_ctl_elem_id *src_id, struct snd_ctl_elem_id *dst_id); +void snd_ctl_rename(struct snd_card *card, struct snd_kcontrol *kctl, const char *name); int snd_ctl_activate_id(struct snd_card *card, struct snd_ctl_elem_id *id, int active); struct snd_kcontrol *snd_ctl_find_numid(struct snd_card * card, unsigned int numid); struct snd_kcontrol *snd_ctl_find_id(struct snd_card * card, struct snd_ctl_elem_id *id); diff --git a/sound/core/control.c b/sound/core/control.c index a7271927d875..50e7ba66f187 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -753,6 +753,29 @@ int snd_ctl_rename_id(struct snd_card *card, struct snd_ctl_elem_id *src_id, } EXPORT_SYMBOL(snd_ctl_rename_id); +/** + * snd_ctl_rename - rename the control on the card + * @card: the card instance + * @kctl: the control to rename + * @name: the new name + * + * Renames the specified control on the card to the new name. + * + * Make sure to take the control write lock - down_write(&card->controls_rwsem). + */ +void snd_ctl_rename(struct snd_card *card, struct snd_kcontrol *kctl, + const char *name) +{ + remove_hash_entries(card, kctl); + + if (strscpy(kctl->id.name, name, sizeof(kctl->id.name)) < 0) + pr_warn("ALSA: Renamed control new name '%s' truncated to '%s'\n", + name, kctl->id.name); + + add_hash_entries(card, kctl); +} +EXPORT_SYMBOL(snd_ctl_rename); + #ifndef CONFIG_SND_CTL_FAST_LOOKUP static struct snd_kcontrol * snd_ctl_find_numid_slow(struct snd_card *card, unsigned int numid) From 0b4f0debb34754002cee295441c9ca89ba8cdfcc Mon Sep 17 00:00:00 2001 From: "Maciej S. Szmigiero" Date: Thu, 20 Oct 2022 22:46:22 +0200 Subject: [PATCH 36/50] ALSA: usb-audio: Use snd_ctl_rename() to rename a control With the recent addition of hashed controls lookup it's not enough to just update the control name field, the hash entries for the modified control have to be updated too. snd_ctl_rename() takes care of that, so use it instead of directly modifying the control name. Fixes: c27e1efb61c5 ("ALSA: control: Use xarray for faster lookups") Cc: stable@vger.kernel.org Signed-off-by: Maciej S. Szmigiero Link: https://lore.kernel.org/r/723877882e3a56bb42a2a2214cfc85f347d36e19.1666296963.git.maciej.szmigiero@oracle.com Signed-off-by: Takashi Iwai --- sound/usb/mixer.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index a5641956ef10..9105ec623120 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -1631,7 +1631,7 @@ static void check_no_speaker_on_headset(struct snd_kcontrol *kctl, if (!found) return; - strscpy(kctl->id.name, "Headphone", sizeof(kctl->id.name)); + snd_ctl_rename(card, kctl, "Headphone"); } static const struct usb_feature_control_info *get_feature_control_info(int control) From b51c225376a684d02fb58b49cf0ce3d693b6f14b Mon Sep 17 00:00:00 2001 From: "Maciej S. Szmigiero" Date: Thu, 20 Oct 2022 22:46:23 +0200 Subject: [PATCH 37/50] ALSA: hda/realtek: Use snd_ctl_rename() to rename a control With the recent addition of hashed controls lookup it's not enough to just update the control name field, the hash entries for the modified control have to be updated too. snd_ctl_rename() takes care of that, so use it instead of directly modifying the control name. Fixes: c27e1efb61c5 ("ALSA: control: Use xarray for faster lookups") Cc: stable@vger.kernel.org Signed-off-by: Maciej S. Szmigiero Link: https://lore.kernel.org/r/37496bd80f91f373268148f877fd735917d97287.1666296963.git.maciej.szmigiero@oracle.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 79acd2a2caf2..9945861f02ef 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2142,7 +2142,7 @@ static void rename_ctl(struct hda_codec *codec, const char *oldname, kctl = snd_hda_find_mixer_ctl(codec, oldname); if (kctl) - strcpy(kctl->id.name, newname); + snd_ctl_rename(codec->card, kctl, newname); } static void alc1220_fixup_gb_dual_codecs(struct hda_codec *codec, From 36476b81b2b5db1de5adb8ced1f71b8972a9d4dd Mon Sep 17 00:00:00 2001 From: "Maciej S. Szmigiero" Date: Thu, 20 Oct 2022 22:46:24 +0200 Subject: [PATCH 38/50] ALSA: emu10k1: Use snd_ctl_rename() to rename a control With the recent addition of hashed controls lookup it's not enough to just update the control name field, the hash entries for the modified control have to be updated too. snd_ctl_rename() takes care of that, so use it instead of directly modifying the control name. Fixes: c27e1efb61c5 ("ALSA: control: Use xarray for faster lookups") Cc: stable@vger.kernel.org Signed-off-by: Maciej S. Szmigiero Link: https://lore.kernel.org/r/38b19f019f95ee78a6e4e59d39afb9e2c3379413.1666296963.git.maciej.szmigiero@oracle.com Signed-off-by: Takashi Iwai --- sound/pci/emu10k1/emumixer.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/emu10k1/emumixer.c b/sound/pci/emu10k1/emumixer.c index e9c0fe3b8446..3c115f8ab96c 100644 --- a/sound/pci/emu10k1/emumixer.c +++ b/sound/pci/emu10k1/emumixer.c @@ -1767,7 +1767,7 @@ static int rename_ctl(struct snd_card *card, const char *src, const char *dst) { struct snd_kcontrol *kctl = ctl_find(card, src); if (kctl) { - strcpy(kctl->id.name, dst); + snd_ctl_rename(card, kctl, dst); return 0; } return -ENOENT; From 957ccc434c398a88a332ae92d70790c186a18a1c Mon Sep 17 00:00:00 2001 From: "Maciej S. Szmigiero" Date: Thu, 20 Oct 2022 22:46:25 +0200 Subject: [PATCH 39/50] ALSA: ca0106: Use snd_ctl_rename() to rename a control With the recent addition of hashed controls lookup it's not enough to just update the control name field, the hash entries for the modified control have to be updated too. snd_ctl_rename() takes care of that, so use it instead of directly modifying the control name. Fixes: c27e1efb61c5 ("ALSA: control: Use xarray for faster lookups") Cc: stable@vger.kernel.org Signed-off-by: Maciej S. Szmigiero Link: https://lore.kernel.org/r/bffee980a420f9b0eee5681d2f48d34a70cec0ce.1666296963.git.maciej.szmigiero@oracle.com Signed-off-by: Takashi Iwai --- sound/pci/ca0106/ca0106_mixer.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/ca0106/ca0106_mixer.c b/sound/pci/ca0106/ca0106_mixer.c index 05f56015ddd8..f6381c098d4f 100644 --- a/sound/pci/ca0106/ca0106_mixer.c +++ b/sound/pci/ca0106/ca0106_mixer.c @@ -720,7 +720,7 @@ static int rename_ctl(struct snd_card *card, const char *src, const char *dst) { struct snd_kcontrol *kctl = ctl_find(card, src); if (kctl) { - strcpy(kctl->id.name, dst); + snd_ctl_rename(card, kctl, dst); return 0; } return -ENOENT; From 52d256cc71f546f67037100c64eb4fa3ae5e4704 Mon Sep 17 00:00:00 2001 From: "Maciej S. Szmigiero" Date: Thu, 20 Oct 2022 22:46:26 +0200 Subject: [PATCH 40/50] ALSA: ac97: Use snd_ctl_rename() to rename a control With the recent addition of hashed controls lookup it's not enough to just update the control name field, the hash entries for the modified control have to be updated too. snd_ctl_rename() takes care of that, so use it instead of directly modifying the control name. While we are at it, check also that the new control name doesn't accidentally overwrite the available buffer space. Fixes: c27e1efb61c5 ("ALSA: control: Use xarray for faster lookups") Cc: stable@vger.kernel.org Signed-off-by: Maciej S. Szmigiero Link: https://lore.kernel.org/r/adb68bfa0885ba4a2583794b828f8e20d23f67c7.1666296963.git.maciej.szmigiero@oracle.com Signed-off-by: Takashi Iwai --- sound/pci/ac97/ac97_codec.c | 32 ++++++++++++++++++++++++-------- 1 file changed, 24 insertions(+), 8 deletions(-) diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c index ceead55f13ab..ff685321f1a1 100644 --- a/sound/pci/ac97/ac97_codec.c +++ b/sound/pci/ac97/ac97_codec.c @@ -2656,11 +2656,18 @@ EXPORT_SYMBOL(snd_ac97_resume); */ static void set_ctl_name(char *dst, const char *src, const char *suffix) { - if (suffix) - sprintf(dst, "%s %s", src, suffix); - else - strcpy(dst, src); -} + const size_t msize = SNDRV_CTL_ELEM_ID_NAME_MAXLEN; + + if (suffix) { + if (snprintf(dst, msize, "%s %s", src, suffix) >= msize) + pr_warn("ALSA: AC97 control name '%s %s' truncated to '%s'\n", + src, suffix, dst); + } else { + if (strscpy(dst, src, msize) < 0) + pr_warn("ALSA: AC97 control name '%s' truncated to '%s'\n", + src, dst); + } +} /* remove the control with the given name and optional suffix */ static int snd_ac97_remove_ctl(struct snd_ac97 *ac97, const char *name, @@ -2687,8 +2694,11 @@ static int snd_ac97_rename_ctl(struct snd_ac97 *ac97, const char *src, const char *dst, const char *suffix) { struct snd_kcontrol *kctl = ctl_find(ac97, src, suffix); + char name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; + if (kctl) { - set_ctl_name(kctl->id.name, dst, suffix); + set_ctl_name(name, dst, suffix); + snd_ctl_rename(ac97->bus->card, kctl, name); return 0; } return -ENOENT; @@ -2707,11 +2717,17 @@ static int snd_ac97_swap_ctl(struct snd_ac97 *ac97, const char *s1, const char *s2, const char *suffix) { struct snd_kcontrol *kctl1, *kctl2; + char name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; + kctl1 = ctl_find(ac97, s1, suffix); kctl2 = ctl_find(ac97, s2, suffix); if (kctl1 && kctl2) { - set_ctl_name(kctl1->id.name, s2, suffix); - set_ctl_name(kctl2->id.name, s1, suffix); + set_ctl_name(name, s2, suffix); + snd_ctl_rename(ac97->bus->card, kctl1, name); + + set_ctl_name(name, s1, suffix); + snd_ctl_rename(ac97->bus->card, kctl2, name); + return 0; } return -ENOENT; From 0e213813df02da048ffd22a2c4fac041768ca327 Mon Sep 17 00:00:00 2001 From: Yang Yingliang Date: Thu, 20 Oct 2022 18:59:37 +0800 Subject: [PATCH 41/50] ASoC: Intel: Skylake: fix possible memory leak in skl_codec_device_init() If snd_hdac_device_register() fails, 'codec' and name allocated in dev_set_name() called in snd_hdac_device_init() are leaked. Fix this by calling put_device(), so they can be freed in snd_hda_codec_dev_release() and kobject_cleanup(). Fixes: e4746d94d00c ("ASoC: Intel: Skylake: Introduce HDA codec init and exit routines") Signed-off-by: Yang Yingliang Suggested-by: Cezary Rojewski Link: https://lore.kernel.org/r/20221020105937.1448951-1-yangyingliang@huawei.com Signed-off-by: Mark Brown --- sound/soc/intel/skylake/skl.c | 8 +------- 1 file changed, 1 insertion(+), 7 deletions(-) diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index bbba2df33aaf..3312b57e3c0c 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -689,11 +689,6 @@ static void load_codec_module(struct hda_codec *codec) #endif /* CONFIG_SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC */ -static void skl_codec_device_exit(struct device *dev) -{ - snd_hdac_device_exit(dev_to_hdac_dev(dev)); -} - static struct hda_codec *skl_codec_device_init(struct hdac_bus *bus, int addr) { struct hda_codec *codec; @@ -706,12 +701,11 @@ static struct hda_codec *skl_codec_device_init(struct hdac_bus *bus, int addr) } codec->core.type = HDA_DEV_ASOC; - codec->core.dev.release = skl_codec_device_exit; ret = snd_hdac_device_register(&codec->core); if (ret) { dev_err(bus->dev, "failed to register hdac device\n"); - snd_hdac_device_exit(&codec->core); + put_device(&codec->core.dev); return ERR_PTR(ret); } From a75481fa00cc06a8763e1795b93140407948c03a Mon Sep 17 00:00:00 2001 From: Leohearts Date: Fri, 21 Oct 2022 14:34:32 +0800 Subject: [PATCH 42/50] ASoC: amd: yc: Add Lenovo Thinkbook 14+ 2022 21D0 to quirks table Lenovo Thinkbook 14+ 2022 (ThinkBook 14 G4+ ARA) uses Ryzen 6000 processor, and has the same microphone problem as other ThinkPads with AMD Ryzen 6000 series CPUs, which has been listed in https://bugzilla.kernel.org/show_bug.cgi?id=216267. Adding 21D0 to quirks table solves this microphone problem for ThinkBook 14 G4+ ARA. Signed-off-by: Taroe Leohearts Link: https://lore.kernel.org/r/26B141B486BEF706+313d1732-e00c-ea41-3123-0d048d40ebb6@leohearts.com Signed-off-by: Mark Brown --- sound/soc/amd/yc/acp6x-mach.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/amd/yc/acp6x-mach.c b/sound/soc/amd/yc/acp6x-mach.c index 09a8aceff22f..6c0f1de10429 100644 --- a/sound/soc/amd/yc/acp6x-mach.c +++ b/sound/soc/amd/yc/acp6x-mach.c @@ -52,6 +52,13 @@ static const struct dmi_system_id yc_acp_quirk_table[] = { DMI_MATCH(DMI_PRODUCT_NAME, "21D0"), } }, + { + .driver_data = &acp6x_card, + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"), + DMI_MATCH(DMI_PRODUCT_NAME, "21D0"), + } + }, { .driver_data = &acp6x_card, .matches = { From e9441675edc1bb8dbfadacf68aafacca60d65a25 Mon Sep 17 00:00:00 2001 From: Yang Yingliang Date: Thu, 20 Oct 2022 19:01:57 +0800 Subject: [PATCH 43/50] ASoC: SOF: Intel: hda-codec: fix possible memory leak in hda_codec_device_init() If snd_hdac_device_register() fails, 'codec' and name allocated in dev_set_name() called in snd_hdac_device_init() are leaked. Fix this by calling put_device(), so they can be freed in snd_hda_codec_dev_release() and kobject_cleanup(). Fixes: 829c67319806 ("ASoC: SOF: Intel: Introduce HDA codec init and exit routines") Fixes: dfe66a18780d ("ALSA: hdac_ext: add extended HDA bus") Signed-off-by: Yang Yingliang Reviewed-by: Kai Vehmanen Link: https://lore.kernel.org/r/20221020110157.1450191-1-yangyingliang@huawei.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda-codec.c | 8 +------- 1 file changed, 1 insertion(+), 7 deletions(-) diff --git a/sound/soc/sof/intel/hda-codec.c b/sound/soc/sof/intel/hda-codec.c index 1e9afc48394c..f2ec2a6c2e0f 100644 --- a/sound/soc/sof/intel/hda-codec.c +++ b/sound/soc/sof/intel/hda-codec.c @@ -109,11 +109,6 @@ EXPORT_SYMBOL_NS(hda_codec_jack_check, SND_SOC_SOF_HDA_AUDIO_CODEC); #define is_generic_config(x) 0 #endif -static void hda_codec_device_exit(struct device *dev) -{ - snd_hdac_device_exit(dev_to_hdac_dev(dev)); -} - static struct hda_codec *hda_codec_device_init(struct hdac_bus *bus, int addr, int type) { struct hda_codec *codec; @@ -126,12 +121,11 @@ static struct hda_codec *hda_codec_device_init(struct hdac_bus *bus, int addr, i } codec->core.type = type; - codec->core.dev.release = hda_codec_device_exit; ret = snd_hdac_device_register(&codec->core); if (ret) { dev_err(bus->dev, "failed to register hdac device\n"); - snd_hdac_device_exit(&codec->core); + put_device(&codec->core.dev); return ERR_PTR(ret); } From 794814529384721ce8f4d34228dc599cc010353d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 21 Oct 2022 14:27:22 +0200 Subject: [PATCH 44/50] ALSA: usb-audio: Add quirks for M-Audio Fast Track C400/600 M-Audio Fast Track C400 and C600 devices (0763:2030 and 0763:2031, respectively) seem requiring the explicit setup for the implicit feedback mode. This patch adds the quirk entries for those. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=214817 Cc: Link: https://lore.kernel.org/r/20221021122722.24784-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/usb/implicit.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/usb/implicit.c b/sound/usb/implicit.c index e1bf1b5da423..f3e8484b3d9c 100644 --- a/sound/usb/implicit.c +++ b/sound/usb/implicit.c @@ -47,6 +47,8 @@ struct snd_usb_implicit_fb_match { static const struct snd_usb_implicit_fb_match playback_implicit_fb_quirks[] = { /* Fixed EP */ /* FIXME: check the availability of generic matching */ + IMPLICIT_FB_FIXED_DEV(0x0763, 0x2030, 0x81, 3), /* M-Audio Fast Track C400 */ + IMPLICIT_FB_FIXED_DEV(0x0763, 0x2031, 0x81, 3), /* M-Audio Fast Track C600 */ IMPLICIT_FB_FIXED_DEV(0x0763, 0x2080, 0x81, 2), /* M-Audio FastTrack Ultra */ IMPLICIT_FB_FIXED_DEV(0x0763, 0x2081, 0x81, 2), /* M-Audio FastTrack Ultra */ IMPLICIT_FB_FIXED_DEV(0x2466, 0x8010, 0x81, 2), /* Fractal Audio Axe-Fx III */ From f86bfeb689f2c4ebe12782ef0578ef778fb1a050 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sat, 22 Oct 2022 09:21:07 +0200 Subject: [PATCH 45/50] ALSA: hda/realtek: Add another HP ZBook G9 model quirks HP ZBook Firefly 16 G9 (103c:896d) and HP ZBook Power 15.6 G9 (103c:89c0) require the same quirk for enabling CS35L41 speaker amps. Signed-off-by: Takashi Iwai Cc: Link: https://lore.kernel.org/r/20221022072107.3401-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 9945861f02ef..701a72ec5629 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9323,6 +9323,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x8898, "HP EliteBook 845 G8 Notebook PC", ALC285_FIXUP_HP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x103c, 0x88d0, "HP Pavilion 15-eh1xxx (mainboard 88D0)", ALC287_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8902, "HP OMEN 16", ALC285_FIXUP_HP_MUTE_LED), + SND_PCI_QUIRK(0x103c, 0x896d, "HP ZBook Firefly 16 G9", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x896e, "HP EliteBook x360 830 G9", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8971, "HP EliteBook 830 G9", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8972, "HP EliteBook 840 G9", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), @@ -9341,6 +9342,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x89aa, "HP EliteBook 630 G9", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x89ac, "HP EliteBook 640 G9", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x89ae, "HP EliteBook 650 G9", ALC236_FIXUP_HP_GPIO_LED), + SND_PCI_QUIRK(0x103c, 0x89c0, "HP ZBook Power 15.6 G9", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x89c3, "Zbook Studio G9", ALC245_FIXUP_CS35L41_SPI_4_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x89c6, "Zbook Fury 17 G9", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x89ca, "HP", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), From ee03c0f200eb0d9f22dd8732d9fb7956d91019c2 Mon Sep 17 00:00:00 2001 From: "Jason A. Donenfeld" Date: Mon, 24 Oct 2022 18:29:29 +0200 Subject: [PATCH 46/50] ALSA: au88x0: use explicitly signed char With char becoming unsigned by default, and with `char` alone being ambiguous and based on architecture, signed chars need to be marked explicitly as such. This fixes warnings like: sound/pci/au88x0/au88x0_core.c:2029 vortex_adb_checkinout() warn: signedness bug returning '(-22)' sound/pci/au88x0/au88x0_core.c:2046 vortex_adb_checkinout() warn: signedness bug returning '(-12)' sound/pci/au88x0/au88x0_core.c:2125 vortex_adb_allocroute() warn: 'vortex_adb_checkinout(vortex, (0), en, 0)' is unsigned sound/pci/au88x0/au88x0_core.c:2170 vortex_adb_allocroute() warn: 'vortex_adb_checkinout(vortex, stream->resources, en, 4)' is unsigned As well, since one function returns errnos, return an `int` rather than a `signed char`. Signed-off-by: Jason A. Donenfeld Cc: Link: https://lore.kernel.org/r/20221024162929.536004-1-Jason@zx2c4.com Signed-off-by: Takashi Iwai --- sound/pci/au88x0/au88x0.h | 6 +++--- sound/pci/au88x0/au88x0_core.c | 2 +- 2 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/pci/au88x0/au88x0.h b/sound/pci/au88x0/au88x0.h index 0aa7af049b1b..6cbb2bc4a048 100644 --- a/sound/pci/au88x0/au88x0.h +++ b/sound/pci/au88x0/au88x0.h @@ -141,7 +141,7 @@ struct snd_vortex { #ifndef CHIP_AU8810 stream_t dma_wt[NR_WT]; wt_voice_t wt_voice[NR_WT]; /* WT register cache. */ - char mixwt[(NR_WT / NR_WTPB) * 6]; /* WT mixin objects */ + s8 mixwt[(NR_WT / NR_WTPB) * 6]; /* WT mixin objects */ #endif /* Global resources */ @@ -235,8 +235,8 @@ static int vortex_alsafmt_aspfmt(snd_pcm_format_t alsafmt, vortex_t *v); static void vortex_connect_default(vortex_t * vortex, int en); static int vortex_adb_allocroute(vortex_t * vortex, int dma, int nr_ch, int dir, int type, int subdev); -static char vortex_adb_checkinout(vortex_t * vortex, int resmap[], int out, - int restype); +static int vortex_adb_checkinout(vortex_t * vortex, int resmap[], int out, + int restype); #ifndef CHIP_AU8810 static int vortex_wt_allocroute(vortex_t * vortex, int dma, int nr_ch); static void vortex_wt_connect(vortex_t * vortex, int en); diff --git a/sound/pci/au88x0/au88x0_core.c b/sound/pci/au88x0/au88x0_core.c index 2ed5100b8cae..f217c02dfdfa 100644 --- a/sound/pci/au88x0/au88x0_core.c +++ b/sound/pci/au88x0/au88x0_core.c @@ -1998,7 +1998,7 @@ static const int resnum[VORTEX_RESOURCE_LAST] = out: Mean checkout if != 0. Else mean Checkin resource. restype: Indicates type of resource to be checked in or out. */ -static char +static int vortex_adb_checkinout(vortex_t * vortex, int resmap[], int out, int restype) { int i, qty = resnum[restype], resinuse = 0; From 50895a55bcfde8ac6f22a37c6bc8cff506b3c7c6 Mon Sep 17 00:00:00 2001 From: "Jason A. Donenfeld" Date: Tue, 25 Oct 2022 02:03:13 +0200 Subject: [PATCH 47/50] ALSA: rme9652: use explicitly signed char With char becoming unsigned by default, and with `char` alone being ambiguous and based on architecture, signed chars need to be marked explicitly as such. This fixes warnings like: sound/pci/rme9652/hdsp.c:3953 hdsp_channel_buffer_location() warn: 'hdsp->channel_map[channel]' is unsigned sound/pci/rme9652/hdsp.c:4153 snd_hdsp_channel_info() warn: impossible condition '(hdsp->channel_map[channel] < 0) => (0-255 < 0)' sound/pci/rme9652/rme9652.c:1833 rme9652_channel_buffer_location() warn: 'rme9652->channel_map[channel]' is unsigned Signed-off-by: Jason A. Donenfeld Cc: Link: https://lore.kernel.org/r/20221025000313.546261-1-Jason@zx2c4.com Signed-off-by: Takashi Iwai --- sound/pci/rme9652/hdsp.c | 26 +++++++++++++------------- sound/pci/rme9652/rme9652.c | 22 +++++++++++----------- 2 files changed, 24 insertions(+), 24 deletions(-) diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c index dcc43a81ae0e..65add92c88aa 100644 --- a/sound/pci/rme9652/hdsp.c +++ b/sound/pci/rme9652/hdsp.c @@ -433,7 +433,7 @@ struct hdsp_midi { struct snd_rawmidi *rmidi; struct snd_rawmidi_substream *input; struct snd_rawmidi_substream *output; - char istimer; /* timer in use */ + signed char istimer; /* timer in use */ struct timer_list timer; spinlock_t lock; int pending; @@ -480,7 +480,7 @@ struct hdsp { pid_t playback_pid; int running; int system_sample_rate; - const char *channel_map; + const signed char *channel_map; int dev; int irq; unsigned long port; @@ -502,7 +502,7 @@ struct hdsp { where the data for that channel can be read/written from/to. */ -static const char channel_map_df_ss[HDSP_MAX_CHANNELS] = { +static const signed char channel_map_df_ss[HDSP_MAX_CHANNELS] = { 0, 1, 2, 3, 4, 5, 6, 7, 8, 9, 10, 11, 12, 13, 14, 15, 16, 17, 18, 19, 20, 21, 22, 23, 24, 25 }; @@ -517,7 +517,7 @@ static const char channel_map_mf_ss[HDSP_MAX_CHANNELS] = { /* Multiface */ -1, -1, -1, -1, -1, -1, -1, -1 }; -static const char channel_map_ds[HDSP_MAX_CHANNELS] = { +static const signed char channel_map_ds[HDSP_MAX_CHANNELS] = { /* ADAT channels are remapped */ 1, 3, 5, 7, 9, 11, 13, 15, 17, 19, 21, 23, /* channels 12 and 13 are S/PDIF */ @@ -526,7 +526,7 @@ static const char channel_map_ds[HDSP_MAX_CHANNELS] = { -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1 }; -static const char channel_map_H9632_ss[HDSP_MAX_CHANNELS] = { +static const signed char channel_map_H9632_ss[HDSP_MAX_CHANNELS] = { /* ADAT channels */ 0, 1, 2, 3, 4, 5, 6, 7, /* SPDIF */ @@ -540,7 +540,7 @@ static const char channel_map_H9632_ss[HDSP_MAX_CHANNELS] = { -1, -1 }; -static const char channel_map_H9632_ds[HDSP_MAX_CHANNELS] = { +static const signed char channel_map_H9632_ds[HDSP_MAX_CHANNELS] = { /* ADAT */ 1, 3, 5, 7, /* SPDIF */ @@ -554,7 +554,7 @@ static const char channel_map_H9632_ds[HDSP_MAX_CHANNELS] = { -1, -1, -1, -1, -1, -1 }; -static const char channel_map_H9632_qs[HDSP_MAX_CHANNELS] = { +static const signed char channel_map_H9632_qs[HDSP_MAX_CHANNELS] = { /* ADAT is disabled in this mode */ /* SPDIF */ 8, 9, @@ -3939,7 +3939,7 @@ static snd_pcm_uframes_t snd_hdsp_hw_pointer(struct snd_pcm_substream *substream return hdsp_hw_pointer(hdsp); } -static char *hdsp_channel_buffer_location(struct hdsp *hdsp, +static signed char *hdsp_channel_buffer_location(struct hdsp *hdsp, int stream, int channel) @@ -3964,7 +3964,7 @@ static int snd_hdsp_playback_copy(struct snd_pcm_substream *substream, void __user *src, unsigned long count) { struct hdsp *hdsp = snd_pcm_substream_chip(substream); - char *channel_buf; + signed char *channel_buf; if (snd_BUG_ON(pos + count > HDSP_CHANNEL_BUFFER_BYTES)) return -EINVAL; @@ -3982,7 +3982,7 @@ static int snd_hdsp_playback_copy_kernel(struct snd_pcm_substream *substream, void *src, unsigned long count) { struct hdsp *hdsp = snd_pcm_substream_chip(substream); - char *channel_buf; + signed char *channel_buf; channel_buf = hdsp_channel_buffer_location(hdsp, substream->pstr->stream, channel); if (snd_BUG_ON(!channel_buf)) @@ -3996,7 +3996,7 @@ static int snd_hdsp_capture_copy(struct snd_pcm_substream *substream, void __user *dst, unsigned long count) { struct hdsp *hdsp = snd_pcm_substream_chip(substream); - char *channel_buf; + signed char *channel_buf; if (snd_BUG_ON(pos + count > HDSP_CHANNEL_BUFFER_BYTES)) return -EINVAL; @@ -4014,7 +4014,7 @@ static int snd_hdsp_capture_copy_kernel(struct snd_pcm_substream *substream, void *dst, unsigned long count) { struct hdsp *hdsp = snd_pcm_substream_chip(substream); - char *channel_buf; + signed char *channel_buf; channel_buf = hdsp_channel_buffer_location(hdsp, substream->pstr->stream, channel); if (snd_BUG_ON(!channel_buf)) @@ -4028,7 +4028,7 @@ static int snd_hdsp_hw_silence(struct snd_pcm_substream *substream, unsigned long count) { struct hdsp *hdsp = snd_pcm_substream_chip(substream); - char *channel_buf; + signed char *channel_buf; channel_buf = hdsp_channel_buffer_location (hdsp, substream->pstr->stream, channel); if (snd_BUG_ON(!channel_buf)) diff --git a/sound/pci/rme9652/rme9652.c b/sound/pci/rme9652/rme9652.c index 1d614fe89a6a..e7c320afefe8 100644 --- a/sound/pci/rme9652/rme9652.c +++ b/sound/pci/rme9652/rme9652.c @@ -230,7 +230,7 @@ struct snd_rme9652 { int last_spdif_sample_rate; /* so that we can catch externally ... */ int last_adat_sample_rate; /* ... induced rate changes */ - const char *channel_map; + const signed char *channel_map; struct snd_card *card; struct snd_pcm *pcm; @@ -247,12 +247,12 @@ struct snd_rme9652 { where the data for that channel can be read/written from/to. */ -static const char channel_map_9652_ss[26] = { +static const signed char channel_map_9652_ss[26] = { 0, 1, 2, 3, 4, 5, 6, 7, 8, 9, 10, 11, 12, 13, 14, 15, 16, 17, 18, 19, 20, 21, 22, 23, 24, 25 }; -static const char channel_map_9636_ss[26] = { +static const signed char channel_map_9636_ss[26] = { 0, 1, 2, 3, 4, 5, 6, 7, 8, 9, 10, 11, 12, 13, 14, 15, /* channels 16 and 17 are S/PDIF */ 24, 25, @@ -260,7 +260,7 @@ static const char channel_map_9636_ss[26] = { -1, -1, -1, -1, -1, -1, -1, -1 }; -static const char channel_map_9652_ds[26] = { +static const signed char channel_map_9652_ds[26] = { /* ADAT channels are remapped */ 1, 3, 5, 7, 9, 11, 13, 15, 17, 19, 21, 23, /* channels 12 and 13 are S/PDIF */ @@ -269,7 +269,7 @@ static const char channel_map_9652_ds[26] = { -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1 }; -static const char channel_map_9636_ds[26] = { +static const signed char channel_map_9636_ds[26] = { /* ADAT channels are remapped */ 1, 3, 5, 7, 9, 11, 13, 15, /* channels 8 and 9 are S/PDIF */ @@ -1819,7 +1819,7 @@ static snd_pcm_uframes_t snd_rme9652_hw_pointer(struct snd_pcm_substream *substr return rme9652_hw_pointer(rme9652); } -static char *rme9652_channel_buffer_location(struct snd_rme9652 *rme9652, +static signed char *rme9652_channel_buffer_location(struct snd_rme9652 *rme9652, int stream, int channel) @@ -1847,7 +1847,7 @@ static int snd_rme9652_playback_copy(struct snd_pcm_substream *substream, void __user *src, unsigned long count) { struct snd_rme9652 *rme9652 = snd_pcm_substream_chip(substream); - char *channel_buf; + signed char *channel_buf; if (snd_BUG_ON(pos + count > RME9652_CHANNEL_BUFFER_BYTES)) return -EINVAL; @@ -1867,7 +1867,7 @@ static int snd_rme9652_playback_copy_kernel(struct snd_pcm_substream *substream, void *src, unsigned long count) { struct snd_rme9652 *rme9652 = snd_pcm_substream_chip(substream); - char *channel_buf; + signed char *channel_buf; channel_buf = rme9652_channel_buffer_location(rme9652, substream->pstr->stream, @@ -1883,7 +1883,7 @@ static int snd_rme9652_capture_copy(struct snd_pcm_substream *substream, void __user *dst, unsigned long count) { struct snd_rme9652 *rme9652 = snd_pcm_substream_chip(substream); - char *channel_buf; + signed char *channel_buf; if (snd_BUG_ON(pos + count > RME9652_CHANNEL_BUFFER_BYTES)) return -EINVAL; @@ -1903,7 +1903,7 @@ static int snd_rme9652_capture_copy_kernel(struct snd_pcm_substream *substream, void *dst, unsigned long count) { struct snd_rme9652 *rme9652 = snd_pcm_substream_chip(substream); - char *channel_buf; + signed char *channel_buf; channel_buf = rme9652_channel_buffer_location(rme9652, substream->pstr->stream, @@ -1919,7 +1919,7 @@ static int snd_rme9652_hw_silence(struct snd_pcm_substream *substream, unsigned long count) { struct snd_rme9652 *rme9652 = snd_pcm_substream_chip(substream); - char *channel_buf; + signed char *channel_buf; channel_buf = rme9652_channel_buffer_location (rme9652, substream->pstr->stream, From 4a4c8482e370d697738a78dcd7bf2780832cb712 Mon Sep 17 00:00:00 2001 From: Yang Yingliang Date: Thu, 27 Oct 2022 09:34:38 +0800 Subject: [PATCH 48/50] ALSA: aoa: i2sbus: fix possible memory leak in i2sbus_add_dev() dev_set_name() in soundbus_add_one() allocates memory for name, it need be freed when of_device_register() fails, call soundbus_dev_put() to give up the reference that hold in device_initialize(), so that it can be freed in kobject_cleanup() when the refcount hit to 0. And other resources are also freed in i2sbus_release_dev(), so it can return 0 directly. Fixes: f3d9478b2ce4 ("[ALSA] snd-aoa: add snd-aoa") Signed-off-by: Yang Yingliang Link: https://lore.kernel.org/r/20221027013438.991920-1-yangyingliang@huawei.com Signed-off-by: Takashi Iwai --- sound/aoa/soundbus/i2sbus/core.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/aoa/soundbus/i2sbus/core.c b/sound/aoa/soundbus/i2sbus/core.c index faf6b03131ee..f6841daf9e3b 100644 --- a/sound/aoa/soundbus/i2sbus/core.c +++ b/sound/aoa/soundbus/i2sbus/core.c @@ -302,6 +302,10 @@ static int i2sbus_add_dev(struct macio_dev *macio, if (soundbus_add_one(&dev->sound)) { printk(KERN_DEBUG "i2sbus: device registration error!\n"); + if (dev->sound.ofdev.dev.kobj.state_initialized) { + soundbus_dev_put(&dev->sound); + return 0; + } goto err; } From f0a868788fcbf63cdab51f5adcf73b271ede8164 Mon Sep 17 00:00:00 2001 From: "Steven Rostedt (Google)" Date: Wed, 26 Oct 2022 23:12:36 -0400 Subject: [PATCH 49/50] ALSA: Use del_timer_sync() before freeing timer The current code for freeing the emux timer is extremely dangerous: CPU0 CPU1 ---- ---- snd_emux_timer_callback() snd_emux_free() spin_lock(&emu->voice_lock) del_timer(&emu->tlist); <-- returns immediately spin_unlock(&emu->voice_lock); [..] kfree(emu); spin_lock(&emu->voice_lock); [BOOM!] Instead just use del_timer_sync() which will wait for the timer to finish before continuing. No need to check if the timer is active or not when doing so. This doesn't fix the race of a possible re-arming of the timer, but at least it won't use the data that has just been freed. [ Fixed unused variable warning by tiwai ] Cc: stable@vger.kernel.org Fixes: 1da177e4c3f4 ("Linux-2.6.12-rc2") Signed-off-by: Steven Rostedt (Google) Reviewed-by: Guenter Roeck Link: https://lore.kernel.org/r/20221026231236.6834b551@gandalf.local.home Signed-off-by: Takashi Iwai --- sound/synth/emux/emux.c | 7 +------ 1 file changed, 1 insertion(+), 6 deletions(-) diff --git a/sound/synth/emux/emux.c b/sound/synth/emux/emux.c index 5ed8e36d2e04..a870759d179e 100644 --- a/sound/synth/emux/emux.c +++ b/sound/synth/emux/emux.c @@ -126,15 +126,10 @@ EXPORT_SYMBOL(snd_emux_register); */ int snd_emux_free(struct snd_emux *emu) { - unsigned long flags; - if (! emu) return -EINVAL; - spin_lock_irqsave(&emu->voice_lock, flags); - if (emu->timer_active) - del_timer(&emu->tlist); - spin_unlock_irqrestore(&emu->voice_lock, flags); + del_timer_sync(&emu->tlist); snd_emux_proc_free(emu); snd_emux_delete_virmidi(emu); From f1fae475f10a26b7e34da4ff2e2f19b7feb3548e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 27 Oct 2022 08:52:33 +0200 Subject: [PATCH 50/50] ALSA: aoa: Fix I2S device accounting i2sbus_add_dev() is supposed to return the number of probed devices, i.e. either 1 or 0. However, i2sbus_add_dev() has one error handling that returns -ENODEV; this will screw up the accumulation number counted in the caller, i2sbus_probe(). Fix the return value to 0 and add the comment for better understanding for readers. Fixes: f3d9478b2ce4 ("[ALSA] snd-aoa: add snd-aoa") Link: https://lore.kernel.org/r/20221027065233.13292-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/aoa/soundbus/i2sbus/core.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/aoa/soundbus/i2sbus/core.c b/sound/aoa/soundbus/i2sbus/core.c index f6841daf9e3b..51ed2f34b276 100644 --- a/sound/aoa/soundbus/i2sbus/core.c +++ b/sound/aoa/soundbus/i2sbus/core.c @@ -147,6 +147,7 @@ static int i2sbus_get_and_fixup_rsrc(struct device_node *np, int index, return rc; } +/* Returns 1 if added, 0 for otherwise; don't return a negative value! */ /* FIXME: look at device node refcounting */ static int i2sbus_add_dev(struct macio_dev *macio, struct i2sbus_control *control, @@ -213,7 +214,7 @@ static int i2sbus_add_dev(struct macio_dev *macio, * either as the second one in that case is just a modem. */ if (!ok) { kfree(dev); - return -ENODEV; + return 0; } mutex_init(&dev->lock);