From 03d0f97fdb45c99cf6f808832db8bd5534e22374 Mon Sep 17 00:00:00 2001 From: Luca Ceresoli Date: Fri, 3 Mar 2023 10:34:10 +0100 Subject: [PATCH 01/36] ASoC: clarify that SND_SOC_IMX_SGTL5000 is the old driver Both SND_SOC_IMX_SGTL5000 and SND_SOC_FSL_ASOC_CARD implement the fsl,imx-audio-sgtl5000 compatible string, which is confusing. It took a little research to find out that the latter is much newer and it is supposed to be the preferred choice since several years. Add a clarification note to avoid wasting time for future readers. Signed-off-by: Luca Ceresoli Link: https://lore.kernel.org/r/20230303093410.357621-1-luca.ceresoli@bootlin.com Signed-off-by: Mark Brown --- sound/soc/fsl/Kconfig | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 614eceda6b9e..33b67db8794e 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -294,6 +294,10 @@ config SND_SOC_IMX_SGTL5000 Say Y if you want to add support for SoC audio on an i.MX board with a sgtl5000 codec. + Note that this is an old driver. Consider enabling + SND_SOC_FSL_ASOC_CARD and SND_SOC_SGTL5000 to use the newer + driver. + config SND_SOC_IMX_SPDIF tristate "SoC Audio support for i.MX boards with S/PDIF" select SND_SOC_IMX_PCM_DMA From 65882134bc622a1e57bd5928ac588855ea2e3ddd Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Thu, 2 Mar 2023 13:29:08 +0100 Subject: [PATCH 02/36] ASoC: qcom: q6prm: fix incorrect clk_root passed to ADSP The second to last argument is clk_root (root of the clock), however the code called q6prm_request_lpass_clock() with clk_attr instead (copy-paste error). This effectively was passing value of 1 as root clock which worked on some of the SoCs (e.g. SM8450) but fails on others, depending on the ADSP. For example on SM8550 this "1" as root clock is not accepted and results in errors coming from ADSP. Fixes: 2f20640491ed ("ASoC: qdsp6: qdsp6: q6prm: handle clk disable correctly") Cc: Signed-off-by: Krzysztof Kozlowski Reviewed-by: Srinivas Kandagatla Tested-by: Srinivas Kandagatla Link: https://lore.kernel.org/r/20230302122908.221398-1-krzysztof.kozlowski@linaro.org Signed-off-by: Mark Brown --- sound/soc/qcom/qdsp6/q6prm.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/qcom/qdsp6/q6prm.c b/sound/soc/qcom/qdsp6/q6prm.c index 3aa63aac4a68..81554d202658 100644 --- a/sound/soc/qcom/qdsp6/q6prm.c +++ b/sound/soc/qcom/qdsp6/q6prm.c @@ -184,9 +184,9 @@ int q6prm_set_lpass_clock(struct device *dev, int clk_id, int clk_attr, int clk_ unsigned int freq) { if (freq) - return q6prm_request_lpass_clock(dev, clk_id, clk_attr, clk_attr, freq); + return q6prm_request_lpass_clock(dev, clk_id, clk_attr, clk_root, freq); - return q6prm_release_lpass_clock(dev, clk_id, clk_attr, clk_attr, freq); + return q6prm_release_lpass_clock(dev, clk_id, clk_attr, clk_root, freq); } EXPORT_SYMBOL_GPL(q6prm_set_lpass_clock); From e5e7e398f6bb7918dab0612eb6991f7bae95520d Mon Sep 17 00:00:00 2001 From: Ravulapati Vishnu Vardhan Rao Date: Sat, 4 Mar 2023 13:37:02 +0530 Subject: [PATCH 03/36] ASoC: codecs: tx-macro: Fix for KASAN: slab-out-of-bounds When we run syzkaller we get below Out of Bound. "KASAN: slab-out-of-bounds Read in regcache_flat_read" Below is the backtrace of the issue: dump_backtrace+0x0/0x4c8 show_stack+0x34/0x44 dump_stack_lvl+0xd8/0x118 print_address_description+0x30/0x2d8 kasan_report+0x158/0x198 __asan_report_load4_noabort+0x44/0x50 regcache_flat_read+0x10c/0x110 regcache_read+0xf4/0x180 _regmap_read+0xc4/0x278 _regmap_update_bits+0x130/0x290 regmap_update_bits_base+0xc0/0x15c snd_soc_component_update_bits+0xa8/0x22c snd_soc_component_write_field+0x68/0xd4 tx_macro_digital_mute+0xec/0x140 Actually There is no need to have decimator with 32 bits. By limiting the variable with short type u8 issue is resolved. Signed-off-by: Ravulapati Vishnu Vardhan Rao Link: https://lore.kernel.org/r/20230304080702.609-1-quic_visr@quicinc.com Signed-off-by: Mark Brown --- sound/soc/codecs/lpass-tx-macro.c | 11 ++++++----- 1 file changed, 6 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/lpass-tx-macro.c b/sound/soc/codecs/lpass-tx-macro.c index bf27bdd5be20..473d3cd39554 100644 --- a/sound/soc/codecs/lpass-tx-macro.c +++ b/sound/soc/codecs/lpass-tx-macro.c @@ -242,7 +242,7 @@ enum { struct tx_mute_work { struct tx_macro *tx; - u32 decimator; + u8 decimator; struct delayed_work dwork; }; @@ -635,7 +635,7 @@ exit: return 0; } -static bool is_amic_enabled(struct snd_soc_component *component, int decimator) +static bool is_amic_enabled(struct snd_soc_component *component, u8 decimator) { u16 adc_mux_reg, adc_reg, adc_n; @@ -849,7 +849,7 @@ static int tx_macro_enable_dec(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); - unsigned int decimator; + u8 decimator; u16 tx_vol_ctl_reg, dec_cfg_reg, hpf_gate_reg, tx_gain_ctl_reg; u8 hpf_cut_off_freq; int hpf_delay = TX_MACRO_DMIC_HPF_DELAY_MS; @@ -1064,7 +1064,8 @@ static int tx_macro_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct snd_soc_component *component = dai->component; - u32 decimator, sample_rate; + u32 sample_rate; + u8 decimator; int tx_fs_rate; struct tx_macro *tx = snd_soc_component_get_drvdata(component); @@ -1128,7 +1129,7 @@ static int tx_macro_digital_mute(struct snd_soc_dai *dai, int mute, int stream) { struct snd_soc_component *component = dai->component; struct tx_macro *tx = snd_soc_component_get_drvdata(component); - u16 decimator; + u8 decimator; /* active decimator not set yet */ if (tx->active_decimator[dai->id] == -1) From d16c893425d07ada1fdd817ec06d322efcf69480 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Amadeusz=20S=C5=82awi=C5=84ski?= Date: Fri, 3 Mar 2023 14:48:50 +0100 Subject: [PATCH 04/36] ASoC: Intel: avs: max98357a: Explicitly define codec format MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit max98357a is speaker codec configured in 48000/2/S16_LE format regardless of front end format, so force it to be so. Reviewed-by: Cezary Rojewski Signed-off-by: Amadeusz Sławiński Link: https://lore.kernel.org/r/20230303134854.2277146-2-amadeuszx.slawinski@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/avs/boards/max98357a.c | 22 ++++++++++++++++++++++ 1 file changed, 22 insertions(+) diff --git a/sound/soc/intel/avs/boards/max98357a.c b/sound/soc/intel/avs/boards/max98357a.c index 921f42caf7e0..183123d08c5a 100644 --- a/sound/soc/intel/avs/boards/max98357a.c +++ b/sound/soc/intel/avs/boards/max98357a.c @@ -8,6 +8,7 @@ #include #include +#include #include #include #include @@ -24,6 +25,26 @@ static const struct snd_soc_dapm_route card_base_routes[] = { { "Spk", NULL, "Speaker" }, }; +static int +avs_max98357a_be_fixup(struct snd_soc_pcm_runtime *runrime, struct snd_pcm_hw_params *params) +{ + struct snd_interval *rate, *channels; + struct snd_mask *fmt; + + rate = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); + channels = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); + fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); + + /* The ADSP will convert the FE rate to 48k, stereo */ + rate->min = rate->max = 48000; + channels->min = channels->max = 2; + + /* set SSP0 to 16 bit */ + snd_mask_none(fmt); + snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S16_LE); + return 0; +} + static int avs_create_dai_link(struct device *dev, const char *platform_name, int ssp_port, struct snd_soc_dai_link **dai_link) { @@ -55,6 +76,7 @@ static int avs_create_dai_link(struct device *dev, const char *platform_name, in dl->num_platforms = 1; dl->id = 0; dl->dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS; + dl->be_hw_params_fixup = avs_max98357a_be_fixup; dl->nonatomic = 1; dl->no_pcm = 1; dl->dpcm_playback = 1; From 61f368624fe4d0c25c6e9c917574b8ace51d776e Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Amadeusz=20S=C5=82awi=C5=84ski?= Date: Fri, 3 Mar 2023 14:48:51 +0100 Subject: [PATCH 05/36] ASoC: Intel: avs: da7219: Explicitly define codec format MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit da7219 is headset codec configured in 48000/2/S24_LE format regardless of front end format, so force it to be so. Reviewed-by: Cezary Rojewski Signed-off-by: Amadeusz Sławiński Link: https://lore.kernel.org/r/20230303134854.2277146-3-amadeuszx.slawinski@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/avs/boards/da7219.c | 21 +++++++++++++++++++++ 1 file changed, 21 insertions(+) diff --git a/sound/soc/intel/avs/boards/da7219.c b/sound/soc/intel/avs/boards/da7219.c index acd43b6108e9..1a1d572cc1d0 100644 --- a/sound/soc/intel/avs/boards/da7219.c +++ b/sound/soc/intel/avs/boards/da7219.c @@ -117,6 +117,26 @@ static void avs_da7219_codec_exit(struct snd_soc_pcm_runtime *rtd) snd_soc_component_set_jack(asoc_rtd_to_codec(rtd, 0)->component, NULL, NULL); } +static int +avs_da7219_be_fixup(struct snd_soc_pcm_runtime *runrime, struct snd_pcm_hw_params *params) +{ + struct snd_interval *rate, *channels; + struct snd_mask *fmt; + + rate = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); + channels = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); + fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); + + /* The ADSP will convert the FE rate to 48k, stereo */ + rate->min = rate->max = 48000; + channels->min = channels->max = 2; + + /* set SSP0 to 24 bit */ + snd_mask_none(fmt); + snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S24_LE); + return 0; +} + static int avs_create_dai_link(struct device *dev, const char *platform_name, int ssp_port, struct snd_soc_dai_link **dai_link) { @@ -148,6 +168,7 @@ static int avs_create_dai_link(struct device *dev, const char *platform_name, in dl->num_platforms = 1; dl->id = 0; dl->dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS; + dl->be_hw_params_fixup = avs_da7219_be_fixup; dl->init = avs_da7219_codec_init; dl->exit = avs_da7219_codec_exit; dl->nonatomic = 1; From d24dbc865c2bd5946bef62bb862a65df092dfc79 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Amadeusz=20S=C5=82awi=C5=84ski?= Date: Fri, 3 Mar 2023 14:48:52 +0100 Subject: [PATCH 06/36] ASoC: Intel: avs: rt5682: Explicitly define codec format MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit rt5682 is headset codec configured in 48000/2/S24_LE format regardless of front end format, so force it to be so. Reviewed-by: Cezary Rojewski Signed-off-by: Amadeusz Sławiński Link: https://lore.kernel.org/r/20230303134854.2277146-4-amadeuszx.slawinski@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/avs/boards/rt5682.c | 22 ++++++++++++++++++++++ 1 file changed, 22 insertions(+) diff --git a/sound/soc/intel/avs/boards/rt5682.c b/sound/soc/intel/avs/boards/rt5682.c index 473e9fe5d0bf..b2c2ba93dcb5 100644 --- a/sound/soc/intel/avs/boards/rt5682.c +++ b/sound/soc/intel/avs/boards/rt5682.c @@ -169,6 +169,27 @@ static const struct snd_soc_ops avs_rt5682_ops = { .hw_params = avs_rt5682_hw_params, }; +static int +avs_rt5682_be_fixup(struct snd_soc_pcm_runtime *runtime, struct snd_pcm_hw_params *params) +{ + struct snd_interval *rate, *channels; + struct snd_mask *fmt; + + rate = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); + channels = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); + fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); + + /* The ADSP will convert the FE rate to 48k, stereo */ + rate->min = rate->max = 48000; + channels->min = channels->max = 2; + + /* set SSPN to 24 bit */ + snd_mask_none(fmt); + snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S24_LE); + + return 0; +} + static int avs_create_dai_link(struct device *dev, const char *platform_name, int ssp_port, struct snd_soc_dai_link **dai_link) { @@ -201,6 +222,7 @@ static int avs_create_dai_link(struct device *dev, const char *platform_name, in dl->id = 0; dl->init = avs_rt5682_codec_init; dl->exit = avs_rt5682_codec_exit; + dl->be_hw_params_fixup = avs_rt5682_be_fixup; dl->ops = &avs_rt5682_ops; dl->nonatomic = 1; dl->no_pcm = 1; From 933de2d127281731166cf2880fa1e23c5a0f7faa Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Fri, 3 Mar 2023 14:48:53 +0100 Subject: [PATCH 07/36] ASoC: Intel: avs: ssm4567: Remove nau8825 bits MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Some of the nau8825 clock control got into the ssm4567, remove it. Signed-off-by: Cezary Rojewski Signed-off-by: Amadeusz Sławiński Link: https://lore.kernel.org/r/20230303134854.2277146-5-amadeuszx.slawinski@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/avs/boards/ssm4567.c | 31 ---------------------------- 1 file changed, 31 deletions(-) diff --git a/sound/soc/intel/avs/boards/ssm4567.c b/sound/soc/intel/avs/boards/ssm4567.c index c5db69612762..2b7f5ad92aca 100644 --- a/sound/soc/intel/avs/boards/ssm4567.c +++ b/sound/soc/intel/avs/boards/ssm4567.c @@ -15,7 +15,6 @@ #include #include "../../../codecs/nau8825.h" -#define SKL_NUVOTON_CODEC_DAI "nau8825-hifi" #define SKL_SSM_CODEC_DAI "ssm4567-hifi" static struct snd_soc_codec_conf card_codec_conf[] = { @@ -34,41 +33,11 @@ static const struct snd_kcontrol_new card_controls[] = { SOC_DAPM_PIN_SWITCH("Right Speaker"), }; -static int -platform_clock_control(struct snd_soc_dapm_widget *w, struct snd_kcontrol *control, int event) -{ - struct snd_soc_dapm_context *dapm = w->dapm; - struct snd_soc_card *card = dapm->card; - struct snd_soc_dai *codec_dai; - int ret; - - codec_dai = snd_soc_card_get_codec_dai(card, SKL_NUVOTON_CODEC_DAI); - if (!codec_dai) { - dev_err(card->dev, "Codec dai not found\n"); - return -EINVAL; - } - - if (SND_SOC_DAPM_EVENT_ON(event)) { - ret = snd_soc_dai_set_sysclk(codec_dai, NAU8825_CLK_MCLK, 24000000, - SND_SOC_CLOCK_IN); - if (ret < 0) - dev_err(card->dev, "set sysclk err = %d\n", ret); - } else { - ret = snd_soc_dai_set_sysclk(codec_dai, NAU8825_CLK_INTERNAL, 0, SND_SOC_CLOCK_IN); - if (ret < 0) - dev_err(card->dev, "set sysclk err = %d\n", ret); - } - - return ret; -} - static const struct snd_soc_dapm_widget card_widgets[] = { SND_SOC_DAPM_SPK("Left Speaker", NULL), SND_SOC_DAPM_SPK("Right Speaker", NULL), SND_SOC_DAPM_SPK("DP1", NULL), SND_SOC_DAPM_SPK("DP2", NULL), - SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0, platform_clock_control, - SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), }; static const struct snd_soc_dapm_route card_base_routes[] = { From 6206b2e787da2ed567922c37bb588a44f6fb6705 Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Fri, 3 Mar 2023 14:48:54 +0100 Subject: [PATCH 08/36] ASoC: Intel: avs: nau8825: Adjust clock control MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Internal clock shall be adjusted also in cases when DAPM event other than 'ON' is triggered. Signed-off-by: Cezary Rojewski Signed-off-by: Amadeusz Sławiński Link: https://lore.kernel.org/r/20230303134854.2277146-6-amadeuszx.slawinski@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/avs/boards/nau8825.c | 14 +++++++------- 1 file changed, 7 insertions(+), 7 deletions(-) diff --git a/sound/soc/intel/avs/boards/nau8825.c b/sound/soc/intel/avs/boards/nau8825.c index b31fa931ba8b..b69fc5567135 100644 --- a/sound/soc/intel/avs/boards/nau8825.c +++ b/sound/soc/intel/avs/boards/nau8825.c @@ -33,15 +33,15 @@ avs_nau8825_clock_control(struct snd_soc_dapm_widget *w, struct snd_kcontrol *co return -EINVAL; } - if (!SND_SOC_DAPM_EVENT_ON(event)) { + if (SND_SOC_DAPM_EVENT_ON(event)) + ret = snd_soc_dai_set_sysclk(codec_dai, NAU8825_CLK_MCLK, 24000000, + SND_SOC_CLOCK_IN); + else ret = snd_soc_dai_set_sysclk(codec_dai, NAU8825_CLK_INTERNAL, 0, SND_SOC_CLOCK_IN); - if (ret < 0) { - dev_err(card->dev, "set sysclk err = %d\n", ret); - return ret; - } - } + if (ret < 0) + dev_err(card->dev, "Set sysclk failed: %d\n", ret); - return 0; + return ret; } static const struct snd_kcontrol_new card_controls[] = { From a659e35ca0af2765f567bdfdccfa247eff0cdab8 Mon Sep 17 00:00:00 2001 From: Ranjani Sridharan Date: Tue, 7 Mar 2023 11:39:11 +0200 Subject: [PATCH 09/36] ASoC: SOF: Intel: MTL: Fix the device description Add the missing ops_free callback. Fixes: 064520e8aeaa ("ASoC: SOF: Intel: Add support for MeteorLake (MTL)") Signed-off-by: Ranjani Sridharan Reviewed-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Signed-off-by: Peter Ujfalusi Link: https://lore.kernel.org/r/20230307093914.25409-2-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/pci-mtl.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/sof/intel/pci-mtl.c b/sound/soc/sof/intel/pci-mtl.c index 6e4e6d4ef5a5..b183dc0014b4 100644 --- a/sound/soc/sof/intel/pci-mtl.c +++ b/sound/soc/sof/intel/pci-mtl.c @@ -46,6 +46,7 @@ static const struct sof_dev_desc mtl_desc = { .nocodec_tplg_filename = "sof-mtl-nocodec.tplg", .ops = &sof_mtl_ops, .ops_init = sof_mtl_ops_init, + .ops_free = hda_ops_free, }; /* PCI IDs */ From 9eb2b4cac223095d2079a6d52b8bbddc6e064288 Mon Sep 17 00:00:00 2001 From: Ranjani Sridharan Date: Tue, 7 Mar 2023 11:39:12 +0200 Subject: [PATCH 10/36] ASoC: SOF: Intel: HDA: Fix device description Add the missing ops_free callback for APL/CNL/CML/JSL/TGL/EHL platforms. Fixes: 1da51943725f ("ASoC: SOF: Intel: hda: init NHLT for IPC4") Signed-off-by: Ranjani Sridharan Reviewed-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Signed-off-by: Peter Ujfalusi Link: https://lore.kernel.org/r/20230307093914.25409-3-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/pci-apl.c | 1 + sound/soc/sof/intel/pci-cnl.c | 2 ++ sound/soc/sof/intel/pci-icl.c | 1 + sound/soc/sof/intel/pci-tgl.c | 5 +++++ 4 files changed, 9 insertions(+) diff --git a/sound/soc/sof/intel/pci-apl.c b/sound/soc/sof/intel/pci-apl.c index 69279dcc92dc..aff6cb573c27 100644 --- a/sound/soc/sof/intel/pci-apl.c +++ b/sound/soc/sof/intel/pci-apl.c @@ -78,6 +78,7 @@ static const struct sof_dev_desc glk_desc = { .nocodec_tplg_filename = "sof-glk-nocodec.tplg", .ops = &sof_apl_ops, .ops_init = sof_apl_ops_init, + .ops_free = hda_ops_free, }; /* PCI IDs */ diff --git a/sound/soc/sof/intel/pci-cnl.c b/sound/soc/sof/intel/pci-cnl.c index 8db3f8d15b55..4c0c1c369dcd 100644 --- a/sound/soc/sof/intel/pci-cnl.c +++ b/sound/soc/sof/intel/pci-cnl.c @@ -48,6 +48,7 @@ static const struct sof_dev_desc cnl_desc = { .nocodec_tplg_filename = "sof-cnl-nocodec.tplg", .ops = &sof_cnl_ops, .ops_init = sof_cnl_ops_init, + .ops_free = hda_ops_free, }; static const struct sof_dev_desc cfl_desc = { @@ -111,6 +112,7 @@ static const struct sof_dev_desc cml_desc = { .nocodec_tplg_filename = "sof-cnl-nocodec.tplg", .ops = &sof_cnl_ops, .ops_init = sof_cnl_ops_init, + .ops_free = hda_ops_free, }; /* PCI IDs */ diff --git a/sound/soc/sof/intel/pci-icl.c b/sound/soc/sof/intel/pci-icl.c index d6cf75e357db..6785669113b3 100644 --- a/sound/soc/sof/intel/pci-icl.c +++ b/sound/soc/sof/intel/pci-icl.c @@ -79,6 +79,7 @@ static const struct sof_dev_desc jsl_desc = { .nocodec_tplg_filename = "sof-jsl-nocodec.tplg", .ops = &sof_cnl_ops, .ops_init = sof_cnl_ops_init, + .ops_free = hda_ops_free, }; /* PCI IDs */ diff --git a/sound/soc/sof/intel/pci-tgl.c b/sound/soc/sof/intel/pci-tgl.c index e80c4dfef85a..adc7314a1b57 100644 --- a/sound/soc/sof/intel/pci-tgl.c +++ b/sound/soc/sof/intel/pci-tgl.c @@ -48,6 +48,7 @@ static const struct sof_dev_desc tgl_desc = { .nocodec_tplg_filename = "sof-tgl-nocodec.tplg", .ops = &sof_tgl_ops, .ops_init = sof_tgl_ops_init, + .ops_free = hda_ops_free, }; static const struct sof_dev_desc tglh_desc = { @@ -110,6 +111,7 @@ static const struct sof_dev_desc ehl_desc = { .nocodec_tplg_filename = "sof-ehl-nocodec.tplg", .ops = &sof_tgl_ops, .ops_init = sof_tgl_ops_init, + .ops_free = hda_ops_free, }; static const struct sof_dev_desc adls_desc = { @@ -141,6 +143,7 @@ static const struct sof_dev_desc adls_desc = { .nocodec_tplg_filename = "sof-adl-nocodec.tplg", .ops = &sof_tgl_ops, .ops_init = sof_tgl_ops_init, + .ops_free = hda_ops_free, }; static const struct sof_dev_desc adl_desc = { @@ -172,6 +175,7 @@ static const struct sof_dev_desc adl_desc = { .nocodec_tplg_filename = "sof-adl-nocodec.tplg", .ops = &sof_tgl_ops, .ops_init = sof_tgl_ops_init, + .ops_free = hda_ops_free, }; static const struct sof_dev_desc adl_n_desc = { @@ -203,6 +207,7 @@ static const struct sof_dev_desc adl_n_desc = { .nocodec_tplg_filename = "sof-adl-nocodec.tplg", .ops = &sof_tgl_ops, .ops_init = sof_tgl_ops_init, + .ops_free = hda_ops_free, }; static const struct sof_dev_desc rpls_desc = { From 1f320bdb29b644a2c9fb301a6fb2d6170e6417e9 Mon Sep 17 00:00:00 2001 From: Ranjani Sridharan Date: Tue, 7 Mar 2023 11:39:13 +0200 Subject: [PATCH 11/36] ASoC: SOF: Intel: SKL: Fix device description Add missing ops_free callback for SKL/KBL platforms. Fixes: 52d7939d10f2 ("ASoC: SOF: Intel: add ops for SKL/KBL") Signed-off-by: Ranjani Sridharan Reviewed-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Signed-off-by: Peter Ujfalusi Link: https://lore.kernel.org/r/20230307093914.25409-4-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/pci-skl.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/sof/intel/pci-skl.c b/sound/soc/sof/intel/pci-skl.c index 3a99dc444f92..5b4bccf81965 100644 --- a/sound/soc/sof/intel/pci-skl.c +++ b/sound/soc/sof/intel/pci-skl.c @@ -38,6 +38,7 @@ static struct sof_dev_desc skl_desc = { .nocodec_tplg_filename = "sof-skl-nocodec.tplg", .ops = &sof_skl_ops, .ops_init = sof_skl_ops_init, + .ops_free = hda_ops_free, }; static struct sof_dev_desc kbl_desc = { @@ -61,6 +62,7 @@ static struct sof_dev_desc kbl_desc = { .nocodec_tplg_filename = "sof-kbl-nocodec.tplg", .ops = &sof_skl_ops, .ops_init = sof_skl_ops_init, + .ops_free = hda_ops_free, }; /* PCI IDs */ From 376f79bbf521fc37b871b536276319951b5bef3a Mon Sep 17 00:00:00 2001 From: Ranjani Sridharan Date: Tue, 7 Mar 2023 11:39:14 +0200 Subject: [PATCH 12/36] ASOC: SOF: Intel: pci-tgl: Fix device description Add the missing ops_free callback. Fixes: 63d375b9f2a9 ("ASoC: SOF: Intel: pci-tgl: use RPL specific firmware definitions") Signed-off-by: Ranjani Sridharan Reviewed-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Signed-off-by: Peter Ujfalusi Link: https://lore.kernel.org/r/20230307093914.25409-5-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/pci-tgl.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/sof/intel/pci-tgl.c b/sound/soc/sof/intel/pci-tgl.c index adc7314a1b57..22e769e0831d 100644 --- a/sound/soc/sof/intel/pci-tgl.c +++ b/sound/soc/sof/intel/pci-tgl.c @@ -239,6 +239,7 @@ static const struct sof_dev_desc rpls_desc = { .nocodec_tplg_filename = "sof-rpl-nocodec.tplg", .ops = &sof_tgl_ops, .ops_init = sof_tgl_ops_init, + .ops_free = hda_ops_free, }; static const struct sof_dev_desc rpl_desc = { @@ -270,6 +271,7 @@ static const struct sof_dev_desc rpl_desc = { .nocodec_tplg_filename = "sof-rpl-nocodec.tplg", .ops = &sof_tgl_ops, .ops_init = sof_tgl_ops_init, + .ops_free = hda_ops_free, }; /* PCI IDs */ From 989a3e4479177d0f4afab8be1960731bc0ffbbd0 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Tue, 7 Mar 2023 13:49:17 +0200 Subject: [PATCH 13/36] ASoC: SOF: ipc3: Check for upper size limit for the received message The sof_ipc3_rx_msg() checks for minimum size of a new rx message but it is missing the check for upper limit. Corrupted or compromised firmware might be able to take advantage of this to cause out of bounds reads outside of the message area. Reported-by: Curtis Malainey Signed-off-by: Peter Ujfalusi Reviewed-by: Pierre-Louis Bossart Reviewed-by: Curtis Malainey Signed-off-by: Peter Ujfalusi Link: https://lore.kernel.org/r/20230307114917.5124-1-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/ipc3.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/sof/ipc3.c b/sound/soc/sof/ipc3.c index 3de64ea2dc9a..4493bbd7faf1 100644 --- a/sound/soc/sof/ipc3.c +++ b/sound/soc/sof/ipc3.c @@ -970,8 +970,9 @@ static void sof_ipc3_rx_msg(struct snd_sof_dev *sdev) return; } - if (hdr.size < sizeof(hdr)) { - dev_err(sdev->dev, "The received message size is invalid\n"); + if (hdr.size < sizeof(hdr) || hdr.size > SOF_IPC_MSG_MAX_SIZE) { + dev_err(sdev->dev, "The received message size is invalid: %u\n", + hdr.size); return; } From 9e269e3aa9006440de639597079ee7140ef5b5f3 Mon Sep 17 00:00:00 2001 From: Seppo Ingalsuo Date: Tue, 7 Mar 2023 13:07:51 +0200 Subject: [PATCH 14/36] ASoC: SOF: ipc4-topology: Fix incorrect sample rate print unit This patch fixes the sample rate print unit from KHz to Hz. E.g. 48000KHz becomes 48000Hz. Signed-off-by: Seppo Ingalsuo Reviewed-by: Pierre-Louis Bossart Reviewed-by: Ranjani Sridharan Signed-off-by: Peter Ujfalusi Link: https://lore.kernel.org/r/20230307110751.2053-1-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/ipc4-topology.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sof/ipc4-topology.c b/sound/soc/sof/ipc4-topology.c index 3e27c7a48ebd..dc44ba2ec71c 100644 --- a/sound/soc/sof/ipc4-topology.c +++ b/sound/soc/sof/ipc4-topology.c @@ -155,7 +155,7 @@ static void sof_ipc4_dbg_audio_format(struct device *dev, for (i = 0; i < num_format; i++, ptr = (u8 *)ptr + object_size) { fmt = ptr; dev_dbg(dev, - " #%d: %uKHz, %ubit (ch_map %#x ch_cfg %u interleaving_style %u fmt_cfg %#x)\n", + " #%d: %uHz, %ubit (ch_map %#x ch_cfg %u interleaving_style %u fmt_cfg %#x)\n", i, fmt->sampling_frequency, fmt->bit_depth, fmt->ch_map, fmt->ch_cfg, fmt->interleaving_style, fmt->fmt_cfg); } From 858a438a6cf919e5727d2a0f5f3f0e68b2d5354e Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Tue, 7 Mar 2023 12:07:33 +0200 Subject: [PATCH 15/36] ASoC: Intel: soc-acpi: fix copy-paste issue in topology names MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit For some reason the convention for topology names was not followed and the name inspired by another unrelated hardware configuration. As a result, the kernel will request a non-existent topology file. Link: https://github.com/thesofproject/sof/pull/6878 Fixes: 2ec8b081d59f ("ASoC: Intel: soc-acpi: Add entry for sof_es8336 in ADL match table") Cc: stable@vger.kernel.org Signed-off-by: Pierre-Louis Bossart Reviewed-by: Ranjani Sridharan Reviewed-by: Bard Liao Reviewed-by: Péter Ujfalusi Signed-off-by: Peter Ujfalusi Link: https://lore.kernel.org/r/20230307100733.15025-1-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/common/soc-acpi-intel-adl-match.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/common/soc-acpi-intel-adl-match.c b/sound/soc/intel/common/soc-acpi-intel-adl-match.c index 56ee5fef66a8..28dd2046e4ac 100644 --- a/sound/soc/intel/common/soc-acpi-intel-adl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-adl-match.c @@ -559,7 +559,7 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_machines[] = { { .comp_ids = &essx_83x6, .drv_name = "sof-essx8336", - .sof_tplg_filename = "sof-adl-es83x6", /* the tplg suffix is added at run time */ + .sof_tplg_filename = "sof-adl-es8336", /* the tplg suffix is added at run time */ .tplg_quirk_mask = SND_SOC_ACPI_TPLG_INTEL_SSP_NUMBER | SND_SOC_ACPI_TPLG_INTEL_SSP_MSB | SND_SOC_ACPI_TPLG_INTEL_DMIC_NUMBER, From 6ba8ddf86a3ada463e9952a19b069f978a70a748 Mon Sep 17 00:00:00 2001 From: Ranjani Sridharan Date: Tue, 7 Mar 2023 13:48:15 +0200 Subject: [PATCH 16/36] ASoC: SOF: topology: Fix error handling in sof_widget_ready() MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Fix the error paths in sof_widget_ready() to free all allocated memory and prevent memory leaks. Signed-off-by: Ranjani Sridharan Reviewed-by: Péter Ujfalusi Reviewed-by: Bard Liao Reviewed-by: Pierre-Louis Bossart Signed-off-by: Peter Ujfalusi Link: https://lore.kernel.org/r/20230307114815.4909-1-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/topology.c | 34 +++++++++++++++++----------------- 1 file changed, 17 insertions(+), 17 deletions(-) diff --git a/sound/soc/sof/topology.c b/sound/soc/sof/topology.c index 4a62ccc71fcb..9f3a038fe21a 100644 --- a/sound/soc/sof/topology.c +++ b/sound/soc/sof/topology.c @@ -1388,14 +1388,15 @@ static int sof_widget_ready(struct snd_soc_component *scomp, int index, if (ret < 0) { dev_err(scomp->dev, "failed to parse component pin tokens for %s\n", w->name); - return ret; + goto widget_free; } if (swidget->num_sink_pins > SOF_WIDGET_MAX_NUM_PINS || swidget->num_source_pins > SOF_WIDGET_MAX_NUM_PINS) { dev_err(scomp->dev, "invalid pins for %s: [sink: %d, src: %d]\n", swidget->widget->name, swidget->num_sink_pins, swidget->num_source_pins); - return -EINVAL; + ret = -EINVAL; + goto widget_free; } if (swidget->num_sink_pins > 1) { @@ -1404,7 +1405,7 @@ static int sof_widget_ready(struct snd_soc_component *scomp, int index, if (ret < 0) { dev_err(scomp->dev, "failed to parse sink pin binding for %s\n", w->name); - return ret; + goto widget_free; } } @@ -1414,7 +1415,7 @@ static int sof_widget_ready(struct snd_soc_component *scomp, int index, if (ret < 0) { dev_err(scomp->dev, "failed to parse source pin binding for %s\n", w->name); - return ret; + goto widget_free; } } @@ -1436,9 +1437,8 @@ static int sof_widget_ready(struct snd_soc_component *scomp, int index, case snd_soc_dapm_dai_out: dai = kzalloc(sizeof(*dai), GFP_KERNEL); if (!dai) { - kfree(swidget); - return -ENOMEM; - + ret = -ENOMEM; + goto widget_free; } ret = sof_widget_parse_tokens(scomp, swidget, tw, token_list, token_list_size); @@ -1496,8 +1496,7 @@ static int sof_widget_ready(struct snd_soc_component *scomp, int index, tw->shift, swidget->id, tw->name, strnlen(tw->sname, SNDRV_CTL_ELEM_ID_NAME_MAXLEN) > 0 ? tw->sname : "none"); - kfree(swidget); - return ret; + goto widget_free; } if (sof_debug_check_flag(SOF_DBG_DISABLE_MULTICORE)) { @@ -1518,10 +1517,7 @@ static int sof_widget_ready(struct snd_soc_component *scomp, int index, if (ret) { dev_err(scomp->dev, "widget event binding failed for %s\n", swidget->widget->name); - kfree(swidget->private); - kfree(swidget->tuples); - kfree(swidget); - return ret; + goto free; } } } @@ -1532,10 +1528,8 @@ static int sof_widget_ready(struct snd_soc_component *scomp, int index, spipe = kzalloc(sizeof(*spipe), GFP_KERNEL); if (!spipe) { - kfree(swidget->private); - kfree(swidget->tuples); - kfree(swidget); - return -ENOMEM; + ret = -ENOMEM; + goto free; } spipe->pipe_widget = swidget; @@ -1546,6 +1540,12 @@ static int sof_widget_ready(struct snd_soc_component *scomp, int index, w->dobj.private = swidget; list_add(&swidget->list, &sdev->widget_list); return ret; +free: + kfree(swidget->private); + kfree(swidget->tuples); +widget_free: + kfree(swidget); + return ret; } static int sof_route_unload(struct snd_soc_component *scomp, From ca09e2a351fbc7836ba9418304ff0c3e72addfe0 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Tue, 7 Mar 2023 11:53:41 +0200 Subject: [PATCH 17/36] ASoC: SOF: Intel: pci-tng: revert invalid bar size setting MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The logic for the ioremap is to find the resource index 3 (IRAM) and infer the BAR address by subtracting the IRAM offset. The BAR size defined in hardware specifications is 2MB. The commit 5947b2726beb6 ("ASoC: SOF: Intel: Check the bar size before remapping") tried to find the BAR size by querying the resource length instead of a pre-canned value, but by requesting the size for index 3 it only gets the size of the IRAM. That's obviously wrong and prevents the probe from proceeding. This commit attempted to fix an issue in a fuzzing/simulated environment but created another on actual devices, so the best course of action is to revert that change. Reported-by: Ferry Toth Tested-by: Ferry Toth (Intel Edison-Arduino) Link: https://github.com/thesofproject/linux/issues/3901 Signed-off-by: Pierre-Louis Bossart Reviewed-by: Péter Ujfalusi Reviewed-by: Ranjani Sridharan Signed-off-by: Peter Ujfalusi Link: https://lore.kernel.org/r/20230307095341.3222-1-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/pci-tng.c | 6 +----- 1 file changed, 1 insertion(+), 5 deletions(-) diff --git a/sound/soc/sof/intel/pci-tng.c b/sound/soc/sof/intel/pci-tng.c index 5b2b409752c5..8c22a00266c0 100644 --- a/sound/soc/sof/intel/pci-tng.c +++ b/sound/soc/sof/intel/pci-tng.c @@ -75,11 +75,7 @@ static int tangier_pci_probe(struct snd_sof_dev *sdev) /* LPE base */ base = pci_resource_start(pci, desc->resindex_lpe_base) - IRAM_OFFSET; - size = pci_resource_len(pci, desc->resindex_lpe_base); - if (size < PCI_BAR_SIZE) { - dev_err(sdev->dev, "error: I/O region is too small.\n"); - return -ENODEV; - } + size = PCI_BAR_SIZE; dev_dbg(sdev->dev, "LPE PHY base at 0x%x size 0x%x", base, size); sdev->bar[DSP_BAR] = devm_ioremap(sdev->dev, base, size); From b66bfc3a9810caed5d55dd8907110bdc8028b06b Mon Sep 17 00:00:00 2001 From: Ranjani Sridharan Date: Tue, 7 Mar 2023 13:46:39 +0200 Subject: [PATCH 18/36] ASoC: SOF: sof-audio: Fix broken early bclk feature for SSP MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit With the removal of widget setup during BE hw_params, the DAI config IPC is never sent with the SOF_DAI_CONFIG_FLAGS_HW_PARAMS. This means that the early bit clock feature required for certain codecs will be broken. Fix this by saving the config flags sent during BE DAI hw_params and reusing it when the DAI_CONFIG IPC is sent after the DAI widget is set up. Also, free the DAI config before the widget is freed. The DAI_CONFIG IPC sent during the sof_widget_free() does not have the DAI index information. So, save the dai_index in the config during hw_params and reuse it during hw_free. For IPC4, do not clear the node ID during hw_free. It will be needed for freeing the group_ida during unprepare. Signed-off-by: Ranjani Sridharan Reviewed-by: Pierre-Louis Bossart Reviewed-by: Rander Wang Reviewed-by: Bard Liao Reviewed-by: Péter Ujfalusi Signed-off-by: Peter Ujfalusi Link: https://lore.kernel.org/r/20230307114639.4553-1-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/ipc3-topology.c | 32 ++++++++++++++++++++++++++++++-- sound/soc/sof/ipc4-topology.c | 15 +++++++++++++-- sound/soc/sof/sof-audio.c | 28 +++++++++++++++++++++++++--- 3 files changed, 68 insertions(+), 7 deletions(-) diff --git a/sound/soc/sof/ipc3-topology.c b/sound/soc/sof/ipc3-topology.c index dceb78bfe17c..b1f425b39db9 100644 --- a/sound/soc/sof/ipc3-topology.c +++ b/sound/soc/sof/ipc3-topology.c @@ -2081,7 +2081,9 @@ static int sof_ipc3_dai_config(struct snd_sof_dev *sdev, struct snd_sof_widget * break; case SOF_DAI_INTEL_ALH: if (data) { - config->dai_index = data->dai_index; + /* save the dai_index during hw_params and reuse it for hw_free */ + if (flags & SOF_DAI_CONFIG_FLAGS_HW_PARAMS) + config->dai_index = data->dai_index; config->alh.stream_id = data->dai_data; } break; @@ -2089,7 +2091,30 @@ static int sof_ipc3_dai_config(struct snd_sof_dev *sdev, struct snd_sof_widget * break; } - config->flags = flags; + /* + * The dai_config op is invoked several times and the flags argument varies as below: + * BE DAI hw_params: When the op is invoked during the BE DAI hw_params, flags contains + * SOF_DAI_CONFIG_FLAGS_HW_PARAMS along with quirks + * FE DAI hw_params: When invoked during FE DAI hw_params after the DAI widget has + * just been set up in the DSP, flags is set to SOF_DAI_CONFIG_FLAGS_HW_PARAMS with no + * quirks + * BE DAI trigger: When invoked during the BE DAI trigger, flags is set to + * SOF_DAI_CONFIG_FLAGS_PAUSE and contains no quirks + * BE DAI hw_free: When invoked during the BE DAI hw_free, flags is set to + * SOF_DAI_CONFIG_FLAGS_HW_FREE and contains no quirks + * FE DAI hw_free: When invoked during the FE DAI hw_free, flags is set to + * SOF_DAI_CONFIG_FLAGS_HW_FREE and contains no quirks + * + * The DAI_CONFIG IPC is sent to the DSP, only after the widget is set up during the FE + * DAI hw_params. But since the BE DAI hw_params precedes the FE DAI hw_params, the quirks + * need to be preserved when assigning the flags before sending the IPC. + * For the case of PAUSE/HW_FREE, since there are no quirks, flags can be used as is. + */ + + if (flags & SOF_DAI_CONFIG_FLAGS_HW_PARAMS) + config->flags |= flags; + else + config->flags = flags; /* only send the IPC if the widget is set up in the DSP */ if (swidget->use_count > 0) { @@ -2097,6 +2122,9 @@ static int sof_ipc3_dai_config(struct snd_sof_dev *sdev, struct snd_sof_widget * &reply, sizeof(reply)); if (ret < 0) dev_err(sdev->dev, "Failed to set dai config for %s\n", dai->name); + + /* clear the flags once the IPC has been sent even if it fails */ + config->flags = SOF_DAI_CONFIG_FLAGS_NONE; } return ret; diff --git a/sound/soc/sof/ipc4-topology.c b/sound/soc/sof/ipc4-topology.c index dc44ba2ec71c..ae02cc152f87 100644 --- a/sound/soc/sof/ipc4-topology.c +++ b/sound/soc/sof/ipc4-topology.c @@ -980,6 +980,7 @@ static void sof_ipc4_unprepare_copier_module(struct snd_sof_widget *swidget) ipc4_copier = dai->private; if (ipc4_copier->dai_type == SOF_DAI_INTEL_ALH) { + struct sof_ipc4_copier_data *copier_data = &ipc4_copier->data; struct sof_ipc4_alh_configuration_blob *blob; unsigned int group_id; @@ -989,6 +990,9 @@ static void sof_ipc4_unprepare_copier_module(struct snd_sof_widget *swidget) ALH_MULTI_GTW_BASE; ida_free(&alh_group_ida, group_id); } + + /* clear the node ID */ + copier_data->gtw_cfg.node_id &= ~SOF_IPC4_NODE_INDEX_MASK; } } @@ -1940,8 +1944,15 @@ static int sof_ipc4_dai_config(struct snd_sof_dev *sdev, struct snd_sof_widget * pipeline->skip_during_fe_trigger = true; fallthrough; case SOF_DAI_INTEL_ALH: - copier_data->gtw_cfg.node_id &= ~SOF_IPC4_NODE_INDEX_MASK; - copier_data->gtw_cfg.node_id |= SOF_IPC4_NODE_INDEX(data->dai_data); + /* + * Do not clear the node ID when this op is invoked with + * SOF_DAI_CONFIG_FLAGS_HW_FREE. It is needed to free the group_ida during + * unprepare. + */ + if (flags & SOF_DAI_CONFIG_FLAGS_HW_PARAMS) { + copier_data->gtw_cfg.node_id &= ~SOF_IPC4_NODE_INDEX_MASK; + copier_data->gtw_cfg.node_id |= SOF_IPC4_NODE_INDEX(data->dai_data); + } break; case SOF_DAI_INTEL_DMIC: case SOF_DAI_INTEL_SSP: diff --git a/sound/soc/sof/sof-audio.c b/sound/soc/sof/sof-audio.c index 760621bfc802..d7df29f2ada8 100644 --- a/sound/soc/sof/sof-audio.c +++ b/sound/soc/sof/sof-audio.c @@ -50,9 +50,27 @@ static int sof_widget_free_unlocked(struct snd_sof_dev *sdev, /* reset route setup status for all routes that contain this widget */ sof_reset_route_setup_status(sdev, swidget); + /* free DAI config and continue to free widget even if it fails */ + if (WIDGET_IS_DAI(swidget->id)) { + struct snd_sof_dai_config_data data; + unsigned int flags = SOF_DAI_CONFIG_FLAGS_HW_FREE; + + data.dai_data = DMA_CHAN_INVALID; + + if (tplg_ops && tplg_ops->dai_config) { + err = tplg_ops->dai_config(sdev, swidget, flags, &data); + if (err < 0) + dev_err(sdev->dev, "failed to free config for widget %s\n", + swidget->widget->name); + } + } + /* continue to disable core even if IPC fails */ - if (tplg_ops && tplg_ops->widget_free) - err = tplg_ops->widget_free(sdev, swidget); + if (tplg_ops && tplg_ops->widget_free) { + ret = tplg_ops->widget_free(sdev, swidget); + if (ret < 0 && !err) + err = ret; + } /* * disable widget core. continue to route setup status and complete flag @@ -151,8 +169,12 @@ static int sof_widget_setup_unlocked(struct snd_sof_dev *sdev, /* send config for DAI components */ if (WIDGET_IS_DAI(swidget->id)) { - unsigned int flags = SOF_DAI_CONFIG_FLAGS_NONE; + unsigned int flags = SOF_DAI_CONFIG_FLAGS_HW_PARAMS; + /* + * The config flags saved during BE DAI hw_params will be used for IPC3. IPC4 does + * not use the flags argument. + */ if (tplg_ops && tplg_ops->dai_config) { ret = tplg_ops->dai_config(sdev, swidget, flags, NULL); if (ret < 0) From c99e48f4ce9b986ab7992ec7283a06dae875f668 Mon Sep 17 00:00:00 2001 From: Jaska Uimonen Date: Tue, 7 Mar 2023 13:07:30 +0200 Subject: [PATCH 19/36] ASoC: SOF: ipc4-topology: set dmic dai index from copier MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Dmic dai index was set incorrectly to bits 5-7, when it is actually using just the lowest 3. Fix the macro for setting the bits. Fixes: aa84ffb72158 ("ASoC: SOF: ipc4-topology: Add support for SSP/DMIC DAI's") Signed-off-by: Jaska Uimonen Reviewed-by: Adrian Bonislawski Reviewed-by: Péter Ujfalusi Reviewed-by: Pierre-Louis Bossart Reviewed-by: Ranjani Sridharan Signed-off-by: Peter Ujfalusi Link: https://lore.kernel.org/r/20230307110730.1995-1-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/ipc4-topology.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sof/ipc4-topology.h b/sound/soc/sof/ipc4-topology.h index 72529179ac22..c0e457f7f51a 100644 --- a/sound/soc/sof/ipc4-topology.h +++ b/sound/soc/sof/ipc4-topology.h @@ -46,7 +46,7 @@ #define SOF_IPC4_NODE_INDEX_INTEL_SSP(x) (((x) & 0xf) << 4) /* Node ID for DMIC type DAI copiers */ -#define SOF_IPC4_NODE_INDEX_INTEL_DMIC(x) (((x) & 0x7) << 5) +#define SOF_IPC4_NODE_INDEX_INTEL_DMIC(x) ((x) & 0x7) #define SOF_IPC4_GAIN_ALL_CHANNELS_MASK 0xffffffff #define SOF_IPC4_VOL_ZERO_DB 0x7fffffff From 52a55779ed14792a150421339664193d6eb8e036 Mon Sep 17 00:00:00 2001 From: Rander Wang Date: Tue, 7 Mar 2023 11:54:52 +0200 Subject: [PATCH 20/36] ASoC: SOF: Intel: hda-dsp: harden D0i3 programming sequence MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Add delay between set and wait command according to hardware programming sequence. Also add debug log to detect error. Signed-off-by: Rander Wang Reviewed-by: Péter Ujfalusi Reviewed-by: Pierre-Louis Bossart Reviewed-by: Péter Ujfalusi Reviewed-by: Ranjani Sridharan Signed-off-by: Peter Ujfalusi Link: https://lore.kernel.org/r/20230307095453.3719-1-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda-dsp.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/soc/sof/intel/hda-dsp.c b/sound/soc/sof/intel/hda-dsp.c index 68eb06f13a1f..a6f2822401e0 100644 --- a/sound/soc/sof/intel/hda-dsp.c +++ b/sound/soc/sof/intel/hda-dsp.c @@ -392,6 +392,12 @@ static int hda_dsp_update_d0i3c_register(struct snd_sof_dev *sdev, u8 value) snd_sof_dsp_update8(sdev, HDA_DSP_HDA_BAR, chip->d0i3_offset, SOF_HDA_VS_D0I3C_I3, value); + /* + * The value written to the D0I3C::I3 bit may not be taken into account immediately. + * A delay is recommended before checking if D0I3C::CIP is cleared + */ + usleep_range(30, 40); + /* Wait for cmd in progress to be cleared before exiting the function */ ret = hda_dsp_wait_d0i3c_done(sdev); if (ret < 0) { @@ -400,6 +406,12 @@ static int hda_dsp_update_d0i3c_register(struct snd_sof_dev *sdev, u8 value) } reg = snd_sof_dsp_read8(sdev, HDA_DSP_HDA_BAR, chip->d0i3_offset); + /* Confirm d0i3 state changed with paranoia check */ + if ((reg ^ value) & SOF_HDA_VS_D0I3C_I3) { + dev_err(sdev->dev, "failed to update D0I3C!\n"); + return -EIO; + } + trace_sof_intel_D0I3C_updated(sdev, reg); return 0; From 8bac40b8ed17ab1be9133e9620f65fae80262b7e Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Tue, 7 Mar 2023 11:54:12 +0200 Subject: [PATCH 21/36] ASoC: SOF: Intel: hda-ctrl: re-add sleep after entering and exiting reset MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit This reverts commit a09d82ce0a867 ("ASoC: SOF: Intel: hda-ctrl: remove useless sleep") It was a mistake to remove those delays, in light of comments in the HDaudio spec captured in snd_hdac_bus_reset_link() that the codec needs time for its initialization and PLL lock. Signed-off-by: Pierre-Louis Bossart Reviewed-by: Ranjani Sridharan Reviewed-by: Péter Ujfalusi Reviewed-by: Rander Wang Signed-off-by: Peter Ujfalusi Link: https://lore.kernel.org/r/20230307095412.3416-1-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda-ctrl.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/sof/intel/hda-ctrl.c b/sound/soc/sof/intel/hda-ctrl.c index 3aea36c077c9..f3bdeba28412 100644 --- a/sound/soc/sof/intel/hda-ctrl.c +++ b/sound/soc/sof/intel/hda-ctrl.c @@ -196,12 +196,15 @@ int hda_dsp_ctrl_init_chip(struct snd_sof_dev *sdev) goto err; } + usleep_range(500, 1000); + /* exit HDA controller reset */ ret = hda_dsp_ctrl_link_reset(sdev, false); if (ret < 0) { dev_err(sdev->dev, "error: failed to exit HDA controller reset\n"); goto err; } + usleep_range(1000, 1200); hda_codec_detect_mask(sdev); From c7e328f1cbf22efe23bc3cd7dd6bb14efccc28d0 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Tue, 7 Mar 2023 13:46:59 +0200 Subject: [PATCH 22/36] ASoC: SOF: sof-audio: don't squelch errors in WIDGET_SETUP phase When an IPC error happens while setting-up a widget during the FE hw_params phase, the existing logic will unwind all previous configurations but will overwrite the return status. The ALSA/ASoC logic will then proceed with the prepare and trigger phases, even though the firmware resources are not available. Fix by returning the initial error code and ignoring the code returned in the UNPREPARE phase. Signed-off-by: Pierre-Louis Bossart Reviewed-by: Ranjani Sridharan Reviewed-by: Chao Song Reviewed-by: Bard Liao Signed-off-by: Peter Ujfalusi Link: https://lore.kernel.org/r/20230307114659.4614-1-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/sof-audio.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/sof/sof-audio.c b/sound/soc/sof/sof-audio.c index d7df29f2ada8..6de388a8d0b8 100644 --- a/sound/soc/sof/sof-audio.c +++ b/sound/soc/sof/sof-audio.c @@ -610,8 +610,8 @@ int sof_widget_list_setup(struct snd_sof_dev *sdev, struct snd_sof_pcm *spcm, ret = sof_walk_widgets_in_order(sdev, spcm, fe_params, platform_params, dir, SOF_WIDGET_SETUP); if (ret < 0) { - ret = sof_walk_widgets_in_order(sdev, spcm, fe_params, platform_params, - dir, SOF_WIDGET_UNPREPARE); + sof_walk_widgets_in_order(sdev, spcm, fe_params, platform_params, + dir, SOF_WIDGET_UNPREPARE); return ret; } From e45cd86c3a78bfb9875a5eb8ab5dab459b59bbe2 Mon Sep 17 00:00:00 2001 From: Rander Wang Date: Tue, 7 Mar 2023 13:06:56 +0200 Subject: [PATCH 23/36] ASoC: SOF: IPC4: update gain ipc msg definition to align with fw MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Recent firmware changes modified the curve duration from 32 to 64 bits, which breaks volume ramps. A simple solution would be to change the definition, but unfortunately the ASoC topology framework only supports up to 32 bit tokens. This patch suggests breaking the 64 bit value in low and high parts, with only the low-part extracted from topology and high-part only zeroes. Since the curve duration is represented in hundred of nanoseconds, we can still represent a 400s ramp, which is just fine. The defacto ABI change has no effect on existing users since the IPC4 firmware has not been released just yet. Link: https://github.com/thesofproject/linux/issues/4026 Signed-off-by: Rander Wang Reviewed-by: Ranjani Sridharan Reviewed-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Reviewed-by: Péter Ujfalusi Signed-off-by: Peter Ujfalusi Link: https://lore.kernel.org/r/20230307110656.1816-1-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/ipc4-control.c | 3 ++- sound/soc/sof/ipc4-topology.c | 4 ++-- sound/soc/sof/ipc4-topology.h | 6 ++++-- 3 files changed, 8 insertions(+), 5 deletions(-) diff --git a/sound/soc/sof/ipc4-control.c b/sound/soc/sof/ipc4-control.c index 67bd2233fd9a..9a71af1a613a 100644 --- a/sound/soc/sof/ipc4-control.c +++ b/sound/soc/sof/ipc4-control.c @@ -97,7 +97,8 @@ sof_ipc4_set_volume_data(struct snd_sof_dev *sdev, struct snd_sof_widget *swidge } /* set curve type and duration from topology */ - data.curve_duration = gain->data.curve_duration; + data.curve_duration_l = gain->data.curve_duration_l; + data.curve_duration_h = gain->data.curve_duration_h; data.curve_type = gain->data.curve_type; msg->data_ptr = &data; diff --git a/sound/soc/sof/ipc4-topology.c b/sound/soc/sof/ipc4-topology.c index ae02cc152f87..a623707c8ffc 100644 --- a/sound/soc/sof/ipc4-topology.c +++ b/sound/soc/sof/ipc4-topology.c @@ -107,7 +107,7 @@ static const struct sof_topology_token gain_tokens[] = { get_token_u32, offsetof(struct sof_ipc4_gain_data, curve_type)}, {SOF_TKN_GAIN_RAMP_DURATION, SND_SOC_TPLG_TUPLE_TYPE_WORD, get_token_u32, - offsetof(struct sof_ipc4_gain_data, curve_duration)}, + offsetof(struct sof_ipc4_gain_data, curve_duration_l)}, {SOF_TKN_GAIN_VAL, SND_SOC_TPLG_TUPLE_TYPE_WORD, get_token_u32, offsetof(struct sof_ipc4_gain_data, init_val)}, }; @@ -692,7 +692,7 @@ static int sof_ipc4_widget_setup_comp_pga(struct snd_sof_widget *swidget) dev_dbg(scomp->dev, "pga widget %s: ramp type: %d, ramp duration %d, initial gain value: %#x, cpc %d\n", - swidget->widget->name, gain->data.curve_type, gain->data.curve_duration, + swidget->widget->name, gain->data.curve_type, gain->data.curve_duration_l, gain->data.init_val, gain->base_config.cpc); ret = sof_ipc4_widget_setup_msg(swidget, &gain->msg); diff --git a/sound/soc/sof/ipc4-topology.h b/sound/soc/sof/ipc4-topology.h index c0e457f7f51a..123f1096f326 100644 --- a/sound/soc/sof/ipc4-topology.h +++ b/sound/soc/sof/ipc4-topology.h @@ -277,14 +277,16 @@ struct sof_ipc4_control_data { * @init_val: Initial value * @curve_type: Curve type * @reserved: reserved for future use - * @curve_duration: Curve duration + * @curve_duration_l: Curve duration low part + * @curve_duration_h: Curve duration high part */ struct sof_ipc4_gain_data { uint32_t channels; uint32_t init_val; uint32_t curve_type; uint32_t reserved; - uint32_t curve_duration; + uint32_t curve_duration_l; + uint32_t curve_duration_h; } __aligned(8); /** From bbdf904b13a62bb8b1272d92a7dde082dff86fbb Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Mon, 6 Mar 2023 15:41:01 +0800 Subject: [PATCH 24/36] ALSA: hda: intel-dsp-config: add MTL PCI id Use SOF as default audio driver. Signed-off-by: Bard Liao Reviewed-by: Gongjun Song Reviewed-by: Kai Vehmanen Cc: Link: https://lore.kernel.org/r/20230306074101.3906707-1-yung-chuan.liao@linux.intel.com Signed-off-by: Takashi Iwai --- sound/hda/intel-dsp-config.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/hda/intel-dsp-config.c b/sound/hda/intel-dsp-config.c index ae31bb127594..317bdf6dcbef 100644 --- a/sound/hda/intel-dsp-config.c +++ b/sound/hda/intel-dsp-config.c @@ -472,6 +472,15 @@ static const struct config_entry config_table[] = { }, #endif +/* Meteor Lake */ +#if IS_ENABLED(CONFIG_SND_SOC_SOF_METEORLAKE) + /* Meteorlake-P */ + { + .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC_OR_SOUNDWIRE, + .device = 0x7e28, + }, +#endif + }; static const struct config_entry *snd_intel_dsp_find_config From 7bb62340951a9af20235a3bde8c98e2e292915df Mon Sep 17 00:00:00 2001 From: Jeremy Szu Date: Tue, 7 Mar 2023 21:53:16 +0800 Subject: [PATCH 25/36] ALSA: hda/realtek: fix speaker, mute/micmute LEDs not work on a HP platform There is a HP platform needs ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED quirk to make mic-mute/audio-mute/speaker working. Signed-off-by: Jeremy Szu Cc: Link: https://lore.kernel.org/r/20230307135317.37621-1-jeremy.szu@canonical.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 3c629f4ae080..5d530b489c48 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9447,6 +9447,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x8b8a, "HP", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8b8b, "HP", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8b8d, "HP", ALC236_FIXUP_HP_GPIO_LED), + SND_PCI_QUIRK(0x103c, 0x8b8f, "HP", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8b92, "HP", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8bf0, "HP", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x1043, 0x103e, "ASUS X540SA", ALC256_FIXUP_ASUS_MIC), From a86e79e3015f5dd8e1b01ccfa49bd5c6e41047a1 Mon Sep 17 00:00:00 2001 From: "Hamidreza H. Fard" Date: Tue, 7 Mar 2023 16:37:41 +0000 Subject: [PATCH 26/36] ALSA: hda/realtek: Fix the speaker output on Samsung Galaxy Book2 Pro Samsung Galaxy Book2 Pro (13" 2022 NP930XED-KA1DE) with codec SSID 144d:c868 requires the same workaround for enabling the speaker amp like other Samsung models with ALC298 code. Signed-off-by: Hamidreza H. Fard Cc: Link: https://lore.kernel.org/r/20230307163741.3878-1-nitocris@posteo.net 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 5d530b489c48..f09a1d7c1b18 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9540,6 +9540,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x144d, 0xc830, "Samsung Galaxy Book Ion (NT950XCJ-X716A)", ALC298_FIXUP_SAMSUNG_AMP), SND_PCI_QUIRK(0x144d, 0xc832, "Samsung Galaxy Book Flex Alpha (NP730QCJ)", ALC256_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET), SND_PCI_QUIRK(0x144d, 0xca03, "Samsung Galaxy Book2 Pro 360 (NP930QED)", ALC298_FIXUP_SAMSUNG_AMP), + SND_PCI_QUIRK(0x144d, 0xc868, "Samsung Galaxy Book2 Pro (NP930XED)", ALC298_FIXUP_SAMSUNG_AMP), SND_PCI_QUIRK(0x1458, 0xfa53, "Gigabyte BXBT-2807", ALC283_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x1462, 0xb120, "MSI Cubi MS-B120", ALC283_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x1462, 0xb171, "Cubi N 8GL (MS-B171)", ALC283_FIXUP_HEADSET_MIC), From ff447886e675979d66b2bc01810035d3baea1b3a Mon Sep 17 00:00:00 2001 From: Bjorn Helgaas Date: Tue, 7 Mar 2023 15:40:54 -0600 Subject: [PATCH 27/36] ALSA: hda: Match only Intel devices with CONTROLLER_IN_GPU() CONTROLLER_IN_GPU() is clearly intended to match only Intel devices, but previously it checked only the PCI Device ID, not the Vendor ID, so it could match devices from other vendors that happened to use the same Device ID. Update CONTROLLER_IN_GPU() so it matches only Intel devices. Fixes: 535115b5ff51 ("ALSA: hda - Abort the probe without i915 binding for HSW/B") Signed-off-by: Bjorn Helgaas Link: https://lore.kernel.org/r/20230307214054.886721-1-helgaas@kernel.org Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 81c4a45254ff..77a592f21947 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -328,14 +328,15 @@ enum { #define needs_eld_notify_link(chip) false #endif -#define CONTROLLER_IN_GPU(pci) (((pci)->device == 0x0a0c) || \ +#define CONTROLLER_IN_GPU(pci) (((pci)->vendor == 0x8086) && \ + (((pci)->device == 0x0a0c) || \ ((pci)->device == 0x0c0c) || \ ((pci)->device == 0x0d0c) || \ ((pci)->device == 0x160c) || \ ((pci)->device == 0x490d) || \ ((pci)->device == 0x4f90) || \ ((pci)->device == 0x4f91) || \ - ((pci)->device == 0x4f92)) + ((pci)->device == 0x4f92))) #define IS_BXT(pci) ((pci)->vendor == 0x8086 && (pci)->device == 0x5a98) From af0f46e5b9a462aaa1d76e82781a5316f03828eb Mon Sep 17 00:00:00 2001 From: Guenter Roeck Date: Tue, 7 Mar 2023 07:51:11 -0800 Subject: [PATCH 28/36] ASoC: da7219: Initialize jack_det_mutex The following traceback is reported if mutex debugging is enabled. DEBUG_LOCKS_WARN_ON(lock->magic != lock) WARNING: CPU: 0 PID: 17 at kernel/locking/mutex.c:950 __mutex_lock_common+0x31c/0x11d4 Modules linked in: CPU: 0 PID: 17 Comm: kworker/0:1 Not tainted 5.10.172-lockdep-21846-g849884cfca5a #1 fd2de466502012eb58bc8beb467f07d0b925611f Hardware name: MediaTek kakadu rev0/rev1 board (DT) Workqueue: events da7219_aad_jack_det_work pstate: 60400005 (nZCv daif +PAN -UAO -TCO BTYPE=--) pc : __mutex_lock_common+0x31c/0x11d4 lr : __mutex_lock_common+0x31c/0x11d4 sp : ffffff80c0317ae0 x29: ffffff80c0317b50 x28: ffffff80c0317b20 x27: 0000000000000000 x26: 0000000000000000 x25: 0000000000000000 x24: 0000000100000000 x23: ffffffd0121d296c x22: dfffffd000000000 x21: 0000000000000000 x20: 0000000000000000 x19: ffffff80c73d7190 x18: 1ffffff018050f52 x17: 0000000000000000 x16: 0000000000000000 x15: 0000000000000000 x14: 0000000000000000 x13: 0000000000000001 x12: 0000000000000001 x11: 0000000000000000 x10: 0000000000000000 x9 : 83f0d991da544b00 x8 : 83f0d991da544b00 x7 : 0000000000000000 x6 : 0000000000000001 x5 : ffffff80c03176a0 x4 : 0000000000000000 x3 : ffffffd01067fd78 x2 : 0000000100000000 x1 : ffffff80c030ba80 x0 : 0000000000000028 Call trace: __mutex_lock_common+0x31c/0x11d4 mutex_lock_nested+0x98/0xac da7219_aad_jack_det_work+0x54/0xf0 process_one_work+0x6cc/0x19dc worker_thread+0x458/0xddc kthread+0x2fc/0x370 ret_from_fork+0x10/0x30 irq event stamp: 579 hardirqs last enabled at (579): [] exit_to_kernel_mode+0x108/0x138 hardirqs last disabled at (577): [] __do_softirq+0x53c/0x125c softirqs last enabled at (578): [] __irq_exit_rcu+0x264/0x4f4 softirqs last disabled at (573): [] __irq_exit_rcu+0x264/0x4f4 ---[ end trace 26da674636181c40 ]--- Initialize the mutex to fix the problem. Cc: David Rau Fixes: 7fde88eda855 ("ASoC: da7219: Improve the IRQ process to increase the stability") Signed-off-by: Guenter Roeck Link: https://lore.kernel.org/r/20230307155111.1985522-1-linux@roeck-us.net Signed-off-by: Mark Brown --- sound/soc/codecs/da7219-aad.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/da7219-aad.c b/sound/soc/codecs/da7219-aad.c index 4a4f09f924bc..e3d398b8f54e 100644 --- a/sound/soc/codecs/da7219-aad.c +++ b/sound/soc/codecs/da7219-aad.c @@ -968,6 +968,8 @@ int da7219_aad_init(struct snd_soc_component *component) INIT_WORK(&da7219_aad->hptest_work, da7219_aad_hptest_work); INIT_WORK(&da7219_aad->jack_det_work, da7219_aad_jack_det_work); + mutex_init(&da7219_aad->jack_det_mutex); + ret = request_threaded_irq(da7219_aad->irq, da7219_aad_pre_irq_thread, da7219_aad_irq_thread, IRQF_TRIGGER_LOW | IRQF_ONESHOT, From e041a2a550582106cba6a7c862c90dfc2ad14492 Mon Sep 17 00:00:00 2001 From: Emil Abildgaard Svendsen Date: Thu, 9 Mar 2023 06:54:41 +0000 Subject: [PATCH 29/36] ASoC: hdmi-codec: only startup/shutdown on supported streams Currently only one stream is supported. This isn't usally a problem until you have a multi codec audio card. Because the audio card will run startup and shutdown on both capture and playback streams. So if your hdmi-codec only support either playback or capture. Then ALSA can't open for playback and capture. This patch will ignore if startup and shutdown are called with a non supported stream. Thus, allowing an audio card like this: +-+ cpu1 <--@-| |-> codec1 (HDMI-CODEC) | |<- codec2 (NOT HDMI-CODEC) +-+ Signed-off-by: Emil Svendsen Link: https://lore.kernel.org/r/20230309065432.4150700-2-emas@bang-olufsen.dk Signed-off-by: Mark Brown --- sound/soc/codecs/hdmi-codec.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c index 01e8ffda2a4b..6d980fbc4207 100644 --- a/sound/soc/codecs/hdmi-codec.c +++ b/sound/soc/codecs/hdmi-codec.c @@ -428,8 +428,13 @@ static int hdmi_codec_startup(struct snd_pcm_substream *substream, { struct hdmi_codec_priv *hcp = snd_soc_dai_get_drvdata(dai); bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + bool has_capture = !hcp->hcd.no_i2s_capture; + bool has_playback = !hcp->hcd.no_i2s_playback; int ret = 0; + if (!((has_playback && tx) || (has_capture && !tx))) + return 0; + mutex_lock(&hcp->lock); if (hcp->busy) { dev_err(dai->dev, "Only one simultaneous stream supported!\n"); @@ -468,6 +473,12 @@ static void hdmi_codec_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct hdmi_codec_priv *hcp = snd_soc_dai_get_drvdata(dai); + bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + bool has_capture = !hcp->hcd.no_i2s_capture; + bool has_playback = !hcp->hcd.no_i2s_playback; + + if (!((has_playback && tx) || (has_capture && !tx))) + return; hcp->chmap_idx = HDMI_CODEC_CHMAP_IDX_UNKNOWN; hcp->hcd.ops->audio_shutdown(dai->dev->parent, hcp->hcd.data); From 9026c0bf233db53b86f74f4c620715e94eb32a09 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 13 Mar 2023 00:49:24 +0000 Subject: [PATCH 30/36] ALSA: asihpi: check pao in control_message() control_message() might be called with pao = NULL. Here indicates control_message() as sample. (B) static void control_message(struct hpi_adapter_obj *pao, ...) { ^^^ struct hpi_hw_obj *phw = pao->priv; ... ^^^ } (A) void _HPI_6205(struct hpi_adapter_obj *pao, ...) { ^^^ ... case HPI_OBJ_CONTROL: (B) control_message(pao, phm, phr); break; ^^^ ... } void HPI_6205(...) { ... (A) _HPI_6205(NULL, phm, phr); ... ^^^^ } Therefore, We will get too many warning via cppcheck, like below sound/pci/asihpi/hpi6205.c:238:27: warning: Possible null pointer dereference: pao [nullPointer] struct hpi_hw_obj *phw = pao->priv; ^ sound/pci/asihpi/hpi6205.c:433:13: note: Calling function '_HPI_6205', 1st argument 'NULL' value is 0 _HPI_6205(NULL, phm, phr); ^ sound/pci/asihpi/hpi6205.c:401:20: note: Calling function 'control_message', 1st argument 'pao' value is 0 control_message(pao, phm, phr); ^ Set phr->error like many functions doing, and don't call _HPI_6205() with NULL. Signed-off-by: Kuninori Morimoto Link: https://lore.kernel.org/r/87ttypeaqz.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Takashi Iwai --- sound/pci/asihpi/hpi6205.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/asihpi/hpi6205.c b/sound/pci/asihpi/hpi6205.c index 27e11b5f70b9..c7d7eff86727 100644 --- a/sound/pci/asihpi/hpi6205.c +++ b/sound/pci/asihpi/hpi6205.c @@ -430,7 +430,7 @@ void HPI_6205(struct hpi_message *phm, struct hpi_response *phr) pao = hpi_find_adapter(phm->adapter_index); } else { /* subsys messages don't address an adapter */ - _HPI_6205(NULL, phm, phr); + phr->error = HPI_ERROR_INVALID_OBJ_INDEX; return; } From 98e5eb110095ec77cb6d775051d181edbf9cd3cf Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 13 Mar 2023 00:50:28 +0000 Subject: [PATCH 31/36] ALSA: hda/ca0132: fixup buffer overrun at tuning_ctl_set() tuning_ctl_set() might have buffer overrun at (X) if it didn't break from loop by matching (A). static int tuning_ctl_set(...) { for (i = 0; i < TUNING_CTLS_COUNT; i++) (A) if (nid == ca0132_tuning_ctls[i].nid) break; snd_hda_power_up(...); (X) dspio_set_param(..., ca0132_tuning_ctls[i].mid, ...); snd_hda_power_down(...); ^ return 1; } We will get below error by cppcheck sound/pci/hda/patch_ca0132.c:4229:2: note: After for loop, i has value 12 for (i = 0; i < TUNING_CTLS_COUNT; i++) ^ sound/pci/hda/patch_ca0132.c:4234:43: note: Array index out of bounds dspio_set_param(codec, ca0132_tuning_ctls[i].mid, 0x20, ^ This patch cares non match case. Signed-off-by: Kuninori Morimoto Link: https://lore.kernel.org/r/87sfe9eap7.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_ca0132.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index acde4cd58785..099722ebaed8 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -4228,8 +4228,10 @@ static int tuning_ctl_set(struct hda_codec *codec, hda_nid_t nid, for (i = 0; i < TUNING_CTLS_COUNT; i++) if (nid == ca0132_tuning_ctls[i].nid) - break; + goto found; + return -EINVAL; +found: snd_hda_power_up(codec); dspio_set_param(codec, ca0132_tuning_ctls[i].mid, 0x20, ca0132_tuning_ctls[i].req, From b7a5822810c4398515300d614d988cf638adecad Mon Sep 17 00:00:00 2001 From: Tim Crawford Date: Fri, 17 Mar 2023 08:18:25 -0600 Subject: [PATCH 32/36] ALSA: hda/realtek: Add quirks for some Clevo laptops Add the audio quirk for some of Clevo's latest RPL laptops: - NP50RNJS (ALC256) - NP70SNE (ALC256) - PD50SNE (ALC1220) - PE60RNE (ALC1220) Co-authored-by: Jeremy Soller Signed-off-by: Tim Crawford Cc: Link: https://lore.kernel.org/r/20230317141825.11807-1-tcrawford@system76.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index f09a1d7c1b18..0ec2c59bb8d5 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2631,6 +2631,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x1558, 0x65e5, "Clevo PC50D[PRS](?:-D|-G)?", ALC1220_FIXUP_CLEVO_PB51ED_PINS), SND_PCI_QUIRK(0x1558, 0x65f1, "Clevo PC50HS", ALC1220_FIXUP_CLEVO_PB51ED_PINS), SND_PCI_QUIRK(0x1558, 0x65f5, "Clevo PD50PN[NRT]", ALC1220_FIXUP_CLEVO_PB51ED_PINS), + SND_PCI_QUIRK(0x1558, 0x66a2, "Clevo PE60RNE", ALC1220_FIXUP_CLEVO_PB51ED_PINS), SND_PCI_QUIRK(0x1558, 0x67d1, "Clevo PB71[ER][CDF]", ALC1220_FIXUP_CLEVO_PB51ED_PINS), SND_PCI_QUIRK(0x1558, 0x67e1, "Clevo PB71[DE][CDF]", ALC1220_FIXUP_CLEVO_PB51ED_PINS), SND_PCI_QUIRK(0x1558, 0x67e5, "Clevo PC70D[PRS](?:-D|-G)?", ALC1220_FIXUP_CLEVO_PB51ED_PINS), @@ -2651,6 +2652,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x1558, 0x96e1, "Clevo P960[ER][CDFN]-K", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK(0x1558, 0x97e1, "Clevo P970[ER][CDFN]", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK(0x1558, 0x97e2, "Clevo P970RC-M", ALC1220_FIXUP_CLEVO_P950), + SND_PCI_QUIRK(0x1558, 0xd502, "Clevo PD50SNE", ALC1220_FIXUP_CLEVO_PB51ED_PINS), SND_PCI_QUIRK_VENDOR(0x1558, "Clevo laptop", ALC882_FIXUP_EAPD), SND_PCI_QUIRK(0x161f, 0x2054, "Medion laptop", ALC883_FIXUP_EAPD), SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo Y530", ALC882_FIXUP_LENOVO_Y530), @@ -9575,6 +9577,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1558, 0x5101, "Clevo S510WU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x5157, "Clevo W517GU1", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x51a1, "Clevo NS50MU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1558, 0x5630, "Clevo NP50RNJS", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x70a1, "Clevo NB70T[HJK]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x70b3, "Clevo NK70SB", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x70f2, "Clevo NH79EPY", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), @@ -9609,6 +9612,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1558, 0x971d, "Clevo N970T[CDF]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0xa500, "Clevo NL5[03]RU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0xa600, "Clevo NL50NU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1558, 0xa671, "Clevo NP70SN[CDE]", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0xb018, "Clevo NP50D[BE]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0xb019, "Clevo NH77D[BE]Q", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0xb022, "Clevo NH77D[DC][QW]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), From b871cb971c683f7f212e7ca3c9a6709a75785116 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 20 Mar 2023 15:09:54 +0100 Subject: [PATCH 33/36] ALSA: hda/conexant: Partial revert of a quirk for Lenovo The recent commit f83bb2592482 ("ALSA: hda/conexant: Add quirk for LENOVO 20149 Notebook model") introduced a quirk for the device with 17aa:3977, but this caused a regression on another model (Lenovo Ideadpad U31) with the very same PCI SSID. And, through skimming over the net, it seems that this PCI SSID is used for multiple different models, so it's no good idea to apply the quirk with the SSID. Although we may take a different ID check (e.g. the codec SSID instead of the PCI SSID), unfortunately, the original patch author couldn't identify the hardware details any longer as the machine was returned, and we can't develop the further proper fix. In this patch, instead, we partially revert the change so that the quirk won't be applied as default for addressing the regression. Meanwhile, the quirk function itself is kept, and it's now made to be applicable via the explicit model=lenovo-20149 option. Fixes: f83bb2592482 ("ALSA: hda/conexant: Add quirk for LENOVO 20149 Notebook model") Reported-by: Jetro Jormalainen Link: https://lore.kernel.org/r/20230308215009.4d3e58a6@mopti Cc: Link: https://lore.kernel.org/r/20230320140954.31154-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 75e1d00074b9..a889cccdd607 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -980,7 +980,10 @@ static const struct snd_pci_quirk cxt5066_fixups[] = { SND_PCI_QUIRK(0x17aa, 0x3905, "Lenovo G50-30", CXT_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x17aa, 0x390b, "Lenovo G50-80", CXT_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x17aa, 0x3975, "Lenovo U300s", CXT_FIXUP_STEREO_DMIC), - SND_PCI_QUIRK(0x17aa, 0x3977, "Lenovo IdeaPad U310", CXT_PINCFG_LENOVO_NOTEBOOK), + /* NOTE: we'd need to extend the quirk for 17aa:3977 as the same + * PCI SSID is used on multiple Lenovo models + */ + SND_PCI_QUIRK(0x17aa, 0x3977, "Lenovo IdeaPad U310", CXT_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x17aa, 0x3978, "Lenovo G50-70", CXT_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x17aa, 0x397b, "Lenovo S205", CXT_FIXUP_STEREO_DMIC), SND_PCI_QUIRK_VENDOR(0x17aa, "Thinkpad", CXT_FIXUP_THINKPAD_ACPI), @@ -1003,6 +1006,7 @@ static const struct hda_model_fixup cxt5066_fixup_models[] = { { .id = CXT_FIXUP_MUTE_LED_GPIO, .name = "mute-led-gpio" }, { .id = CXT_FIXUP_HP_ZBOOK_MUTE_LED, .name = "hp-zbook-mute-led" }, { .id = CXT_FIXUP_HP_MIC_NO_PRESENCE, .name = "hp-mic-fix" }, + { .id = CXT_PINCFG_LENOVO_NOTEBOOK, .name = "lenovo-20149" }, {} }; From 8c721c53dda512fdd48eb24d6d99e56deee57898 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 20 Mar 2023 15:28:38 +0100 Subject: [PATCH 34/36] ALSA: usb-audio: Fix recursive locking at XRUN during syncing The recent support of low latency playback in USB-audio driver made the snd_usb_queue_pending_output_urbs() function to be called via PCM ack ops. In the new code path, the function is performed already in the PCM stream lock. The problem is that, when an XRUN is detected, the function calls snd_pcm_xrun() to notify, but snd_pcm_xrun() is supposed to be called only outside the stream lock. As a result, it leads to a deadlock of PCM stream locking. For avoiding such a recursive locking, this patch adds an additional check to the code paths in PCM core that call the ack callback; now it checks the error code from the callback, and if it's -EPIPE, the XRUN is handled in the PCM core side gracefully. Along with it, the USB-audio driver code is changed to follow that, i.e. -EPIPE is returned instead of the explicit snd_pcm_xrun() call when the function is performed already in the stream lock. Fixes: d5f871f89e21 ("ALSA: usb-audio: Improved lowlatency playback support") Reported-and-tested-by: John Keeping Link: https://lore.kernel.org/r/20230317195128.3911155-1-john@metanate.com Reviewed-by: Jaroslav Kysela Reviewed-by; Takashi Sakamoto Link: https://lore.kernel.org/r/20230320142838.494-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/core/pcm_lib.c | 2 ++ sound/usb/endpoint.c | 22 ++++++++++++++-------- sound/usb/endpoint.h | 4 ++-- sound/usb/pcm.c | 2 +- 4 files changed, 19 insertions(+), 11 deletions(-) diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 8b6aeb8a78f7..02fd65993e7e 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -2155,6 +2155,8 @@ int pcm_lib_apply_appl_ptr(struct snd_pcm_substream *substream, ret = substream->ops->ack(substream); if (ret < 0) { runtime->control->appl_ptr = old_appl_ptr; + if (ret == -EPIPE) + __snd_pcm_xrun(substream); return ret; } } diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index 419302e2057e..647fa054d8b1 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -455,8 +455,8 @@ static void push_back_to_ready_list(struct snd_usb_endpoint *ep, * This function is used both for implicit feedback endpoints and in low- * latency playback mode. */ -void snd_usb_queue_pending_output_urbs(struct snd_usb_endpoint *ep, - bool in_stream_lock) +int snd_usb_queue_pending_output_urbs(struct snd_usb_endpoint *ep, + bool in_stream_lock) { bool implicit_fb = snd_usb_endpoint_implicit_feedback_sink(ep); @@ -480,7 +480,7 @@ void snd_usb_queue_pending_output_urbs(struct snd_usb_endpoint *ep, spin_unlock_irqrestore(&ep->lock, flags); if (ctx == NULL) - return; + break; /* copy over the length information */ if (implicit_fb) { @@ -495,11 +495,14 @@ void snd_usb_queue_pending_output_urbs(struct snd_usb_endpoint *ep, break; if (err < 0) { /* push back to ready list again for -EAGAIN */ - if (err == -EAGAIN) + if (err == -EAGAIN) { push_back_to_ready_list(ep, ctx); - else + break; + } + + if (!in_stream_lock) notify_xrun(ep); - return; + return -EPIPE; } err = usb_submit_urb(ctx->urb, GFP_ATOMIC); @@ -507,13 +510,16 @@ void snd_usb_queue_pending_output_urbs(struct snd_usb_endpoint *ep, usb_audio_err(ep->chip, "Unable to submit urb #%d: %d at %s\n", ctx->index, err, __func__); - notify_xrun(ep); - return; + if (!in_stream_lock) + notify_xrun(ep); + return -EPIPE; } set_bit(ctx->index, &ep->active_mask); atomic_inc(&ep->submitted_urbs); } + + return 0; } /* diff --git a/sound/usb/endpoint.h b/sound/usb/endpoint.h index 924f4351588c..c09f68ce08b1 100644 --- a/sound/usb/endpoint.h +++ b/sound/usb/endpoint.h @@ -52,7 +52,7 @@ int snd_usb_endpoint_implicit_feedback_sink(struct snd_usb_endpoint *ep); int snd_usb_endpoint_next_packet_size(struct snd_usb_endpoint *ep, struct snd_urb_ctx *ctx, int idx, unsigned int avail); -void snd_usb_queue_pending_output_urbs(struct snd_usb_endpoint *ep, - bool in_stream_lock); +int snd_usb_queue_pending_output_urbs(struct snd_usb_endpoint *ep, + bool in_stream_lock); #endif /* __USBAUDIO_ENDPOINT_H */ diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index d959da7a1afb..eec5232f9fb2 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -1639,7 +1639,7 @@ static int snd_usb_pcm_playback_ack(struct snd_pcm_substream *substream) * outputs here */ if (!ep->active_mask) - snd_usb_queue_pending_output_urbs(ep, true); + return snd_usb_queue_pending_output_urbs(ep, true); return 0; } From 5f4efc9dfcfd8440113057290f624ba2c893afb7 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Wed, 22 Mar 2023 16:34:04 +0100 Subject: [PATCH 35/36] ALSA: hda/realtek: Fix support for Dell Precision 3260 Unfortunately, in commit 5911d78fabbb a wrong codec patch was selected. The model=alc283-dac-wcaps is equivalent to ALC283_FIXUP_CHROME_BOOK not ALC295_FIXUP_CHROME_BOOK. Fixes: 5911d78fabbb ("ALSA: hda/realtek: Improve support for Dell Precision 3260") Signed-off-by: Jaroslav Kysela Link: https://lore.kernel.org/r/20230322153404.386473-1-perex@perex.cz 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 0ec2c59bb8d5..b501f9489fc1 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9262,7 +9262,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0a62, "Dell Precision 5560", ALC289_FIXUP_DUAL_SPK), SND_PCI_QUIRK(0x1028, 0x0a9d, "Dell Latitude 5430", ALC269_FIXUP_DELL4_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0a9e, "Dell Latitude 5430", ALC269_FIXUP_DELL4_MIC_NO_PRESENCE), - SND_PCI_QUIRK(0x1028, 0x0ac9, "Dell Precision 3260", ALC295_FIXUP_CHROME_BOOK), + SND_PCI_QUIRK(0x1028, 0x0ac9, "Dell Precision 3260", ALC283_FIXUP_CHROME_BOOK), SND_PCI_QUIRK(0x1028, 0x0b19, "Dell XPS 15 9520", ALC289_FIXUP_DUAL_SPK), SND_PCI_QUIRK(0x1028, 0x0b1a, "Dell Precision 5570", ALC289_FIXUP_DUAL_SPK), SND_PCI_QUIRK(0x1028, 0x0b37, "Dell Inspiron 16 Plus 7620 2-in-1", ALC295_FIXUP_DELL_INSPIRON_TOP_SPEAKERS), From fa4e7a6fa12b1132340785e14bd439cbe95b7a5a Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 24 Mar 2023 08:50:05 +0100 Subject: [PATCH 36/36] ALSA: usb-audio: Fix regression on detection of Roland VS-100 It's been reported that the recent kernel can't probe the PCM devices on Roland VS-100 properly, and it turned out to be a regression by the recent addition of the bit shift range check for the format bits. In the old code, we just did bit-shift and it resulted in zero, which is then corrected to the standard PCM format, while the new code explicitly returns an error in such a case. For addressing the regression, relax the check and fallback to the standard PCM type (with the info output). Fixes: 43d5ca88dfcd ("ALSA: usb-audio: Fix potential out-of-bounds shift") Cc: Link: https://bugzilla.kernel.org/show_bug.cgi?id=217084 Link: https://lore.kernel.org/r/20230324075005.19403-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/usb/format.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/usb/format.c b/sound/usb/format.c index 405dc0bf6678..4b1c5ba121f3 100644 --- a/sound/usb/format.c +++ b/sound/usb/format.c @@ -39,8 +39,12 @@ static u64 parse_audio_format_i_type(struct snd_usb_audio *chip, case UAC_VERSION_1: default: { struct uac_format_type_i_discrete_descriptor *fmt = _fmt; - if (format >= 64) - return 0; /* invalid format */ + if (format >= 64) { + usb_audio_info(chip, + "%u:%d: invalid format type 0x%llx is detected, processed as PCM\n", + fp->iface, fp->altsetting, format); + format = UAC_FORMAT_TYPE_I_PCM; + } sample_width = fmt->bBitResolution; sample_bytes = fmt->bSubframeSize; format = 1ULL << format;