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[PATCH 0/4] hwdep: define interface for hwdep device
by GitHub pull_request - edited 02 Jun '22

02 Jun '22

alsa-project/alsa-gobject pull request #73 was edited from takaswie: The protocol in ALSA HwDep character device has common feature to each device. It's suitable for userspace implementation to have GObject interface for such features. This patchset adds the interface. The implementation of interface should have four methods; open, get_protocol_version, get_device_info, and create_source. Additionally, it should emit handle-disconnection signal when detecting ENODEV error in the interation of source. It requires to notify userspace application since the application should release character device immediately. ``` Takashi Sakamoto (3): hwdep: use explicit value for iface type enumeration hwdep: device-common: add interface for common feature of ALSA HwDep device hwdep: add error domain for implementators of DeviceCommon interface src/hwdep/alsahwdep-enum-types.h | 89 +++++++++------- src/hwdep/alsahwdep.h | 2 + src/hwdep/alsahwdep.map | 10 ++ src/hwdep/device-common.c | 178 +++++++++++++++++++++++++++++++ src/hwdep/device-common.h | 106 ++++++++++++++++++ src/hwdep/meson.build | 2 + tests/alsahwdep-enums | 10 ++ 7 files changed, 356 insertions(+), 41 deletions(-) create mode 100644 src/hwdep/device-common.c create mode 100644 src/hwdep/device-common.h ``` Request URL : https://github.com/alsa-project/alsa-gobject/pull/73 Patch URL : https://github.com/alsa-project/alsa-gobject/pull/73.patch Repository URL: https://github.com/alsa-project/alsa-gobject

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[PATCH 0/3] hwdep: define interface for hwdep device
by GitHub pull_request - edited 02 Jun '22

02 Jun '22

alsa-project/alsa-gobject pull request #73 was edited from takaswie: The protocol in ALSA HwDep character device has common feature to each device. It's suitable for userspace implementation to have GObject interface for such features. This patchset adds the interface. The implementation of interface should have four methods; open, get_protocol_version, get_device_info, and create_source. Additionally, it should emit handle-disconnection signal when detecting ENODEV error in the interation of source. It requires to notify userspace application since the application should release character device immediately. ``` Takashi Sakamoto (3): hwdep: use explicit value for iface type enumeration hwdep: device-common: add interface for common feature of ALSA HwDep device hwdep: add error domain for implementators of DeviceCommon interface src/hwdep/alsahwdep-enum-types.h | 89 +++++++++------- src/hwdep/alsahwdep.h | 2 + src/hwdep/alsahwdep.map | 10 ++ src/hwdep/device-common.c | 178 +++++++++++++++++++++++++++++++ src/hwdep/device-common.h | 106 ++++++++++++++++++ src/hwdep/meson.build | 2 + tests/alsahwdep-enums | 10 ++ 7 files changed, 356 insertions(+), 41 deletions(-) create mode 100644 src/hwdep/device-common.c create mode 100644 src/hwdep/device-common.h ``` Request URL : https://github.com/alsa-project/alsa-gobject/pull/73 Patch URL : https://github.com/alsa-project/alsa-gobject/pull/73.patch Repository URL: https://github.com/alsa-project/alsa-gobject

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[PATCH 0/2] hwdep: define interface for hwdep device
by GitHub pull_request - edited 02 Jun '22

02 Jun '22

alsa-project/alsa-gobject pull request #73 was edited from takaswie: The protocol in ALSA HwDep character device has common feature to each device. It's suitable for userspace implementation to have GObject interface for such features. This patchset adds the interface. The implementation of interface should have four methods; open, get_protocol_version, get_device_info, and create_source. Additionally, it should emit handle-disconnection signal when detecting ENODEV error in the interation of source. It requires to notify userspace application since the application should release character device immediately. ``` Takashi Sakamoto (3): hwdep: use explicit value for iface type enumeration hwdep: device-common: add interface for common feature of ALSA HwDep device hwdep: add error domain for implementators of DeviceCommon interface src/hwdep/alsahwdep-enum-types.h | 89 +++++++++------- src/hwdep/alsahwdep.h | 2 + src/hwdep/alsahwdep.map | 10 ++ src/hwdep/device-common.c | 178 +++++++++++++++++++++++++++++++ src/hwdep/device-common.h | 106 ++++++++++++++++++ src/hwdep/meson.build | 2 + tests/alsahwdep-enums | 10 ++ 7 files changed, 356 insertions(+), 41 deletions(-) create mode 100644 src/hwdep/device-common.c create mode 100644 src/hwdep/device-common.h ``` Request URL : https://github.com/alsa-project/alsa-gobject/pull/73 Patch URL : https://github.com/alsa-project/alsa-gobject/pull/73.patch Repository URL: https://github.com/alsa-project/alsa-gobject

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[PATCH 0/2] hwdep: define interface for hwdep device
by GitHub pull_request - opened 02 Jun '22

02 Jun '22

alsa-project/alsa-gobject pull request #73 was opened from takaswie: The protocol in ALSA HwDep character device has common feature to each device. It's suitable for userspace implementation to have GObject interface for such features. This patchset adds the interface. The implementation of interface should have four methods; open, get_protocol_version, get_device_info, and create_source. Additionally, it should emit handle-disconnection signal when detecting ENODEV error in the interation of source. It requires to notify userspace application since the application should release character device immediately. ``` Takashi Sakamoto (2): hwdep: use explicit value for iface type enumeration hwdep: device-common: add interface for common feature of ALSA HwDep device src/hwdep/alsahwdep-enum-types.h | 67 +++++++--------- src/hwdep/alsahwdep.h | 2 + src/hwdep/alsahwdep.map | 6 ++ src/hwdep/device-common.c | 128 +++++++++++++++++++++++++++++++ src/hwdep/device-common.h | 102 ++++++++++++++++++++++++ src/hwdep/meson.build | 2 + 6 files changed, 267 insertions(+), 40 deletions(-) create mode 100644 src/hwdep/device-common.c create mode 100644 src/hwdep/device-common.h ``` Request URL : https://github.com/alsa-project/alsa-gobject/pull/73 Patch URL : https://github.com/alsa-project/alsa-gobject/pull/73.patch Repository URL: https://github.com/alsa-project/alsa-gobject

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Semantics for _SX controls
by Mark Brown 02 Jun '22

02 Jun '22

Hi, The fixes for input validation have highlighed a bunch of problems with the _SX controls. These aren't widely used, it's just some Cirrus Logic and Qualcomm CODECs plus the obsolte ab8500 driver, but it has become apparent that at least some of the drivers were relying on the lack of validation we used to have to work. For the Cirrus case 34198710f55b5f ("ASoC: Add info callback for SX_TLV controls") says that the intended semantic is Currently every CODEC implementing these controls specifies the minimum as the non-zero value for the minimum and the maximum as the number of gain settings available. which was from Charles at Cirrus so hopefully that's the semantic used by Cirrus drivers. However Tan Nayir pointed at some out of tree patches from Qualcomm which change the info callback such that max is instead the maximum register value that can be written and reports that since we started validating written values properly they can no longer set some valid values in some controls on at least some Qualcomm devices (this was a wcd9340) which clearly indicates that there's some problems that need fixing. This isn't helped by the fact that snd_soc_info_volsw_sx() is implemented by calling snd_soc_info_volsw() and then applying an offset to the maximum value which makes things harder to follow - I think we should make snd_soc_info_volsw_sx() a standalone function, and we probably need some fixes in some combination of the shared functions and the drivers as well. Can people please check how the values in the current drivers correspond to what the actual bitfields in the registers have so we can get a handle on what's going on here? I've pasted a quick grep for _SX controls below: sound/soc/codecs/ab8500-codec.c: SOC_SINGLE_XR_SX("ANC Warp Delay Shift", sound/soc/codecs/ab8500-codec.c: SOC_SINGLE_XR_SX("ANC FIR Output Shift", sound/soc/codecs/ab8500-codec.c: SOC_SINGLE_XR_SX("ANC IIR Output Shift", sound/soc/codecs/ab8500-codec.c: SOC_SINGLE_XR_SX("ANC Warp Delay", sound/soc/codecs/cs35l33.c: SOC_SINGLE_SX_TLV("DAC Volume", CS35L33_DIG_VOL_CTL, sound/soc/codecs/cs35l34.c: SOC_SINGLE_SX_TLV("Digital Volume", CS35L34_AMP_DIG_VOL, sound/soc/codecs/cs35l35.c: SOC_SINGLE_SX_TLV("Digital Audio Volume", CS35L35_AMP_DIG_VOL, sound/soc/codecs/cs35l35.c: SOC_SINGLE_SX_TLV("Digital Advisory Volume", CS35L35_ADV_DIG_VOL, sound/soc/codecs/cs35l36.c: SOC_SINGLE_SX_TLV("Digital PCM Volume", CS35L36_AMP_DIG_VOL_CTRL, sound/soc/codecs/cs35l41.c: SOC_SINGLE_SX_TLV("Digital PCM Volume", CS35L41_AMP_DIG_VOL_CTRL, sound/soc/codecs/cs4265.c: SOC_DOUBLE_R_SX_TLV("PGA Volume", CS4265_CHA_PGA_CTL, sound/soc/codecs/cs42l51.c: SOC_DOUBLE_R_SX_TLV("PCM Playback Volume", sound/soc/codecs/cs42l51.c: SOC_DOUBLE_R_SX_TLV("Analog Playback Volume", sound/soc/codecs/cs42l51.c: SOC_DOUBLE_R_SX_TLV("ADC Mixer Volume", sound/soc/codecs/cs42l51.c: SOC_DOUBLE_R_SX_TLV("ADC Attenuator Volume", sound/soc/codecs/cs42l51.c: SOC_DOUBLE_R_SX_TLV("PGA Volume", sound/soc/codecs/cs42l52.c: SOC_DOUBLE_R_SX_TLV("Master Volume", CS42L52_MASTERA_VOL, sound/soc/codecs/cs42l52.c: SOC_DOUBLE_R_SX_TLV("Headphone Volume", CS42L52_HPA_VOL, sound/soc/codecs/cs42l52.c: SOC_DOUBLE_R_SX_TLV("Speaker Volume", CS42L52_SPKA_VOL, sound/soc/codecs/cs42l52.c: SOC_DOUBLE_R_SX_TLV("Bypass Volume", CS42L52_PASSTHRUA_VOL, sound/soc/codecs/cs42l52.c: SOC_DOUBLE_R_SX_TLV("ADC Volume", CS42L52_ADCA_VOL, sound/soc/codecs/cs42l52.c: SOC_DOUBLE_R_SX_TLV("ADC Mixer Volume", sound/soc/codecs/cs42l52.c: SOC_DOUBLE_R_SX_TLV("PGA Volume", CS42L52_PGAA_CTL, sound/soc/codecs/cs42l52.c: SOC_DOUBLE_R_SX_TLV("PCM Mixer Volume", sound/soc/codecs/cs42l52.c: SOC_SINGLE_SX_TLV("Beep Volume", CS42L52_BEEP_VOL, sound/soc/codecs/cs42l56.c: SOC_DOUBLE_R_SX_TLV("Master Volume", CS42L56_MASTER_A_VOLUME, sound/soc/codecs/cs42l56.c: SOC_DOUBLE_R_SX_TLV("ADC Mixer Volume", CS42L56_ADCA_MIX_VOLUME, sound/soc/codecs/cs42l56.c: SOC_DOUBLE_R_SX_TLV("PCM Mixer Volume", CS42L56_PCMA_MIX_VOLUME, sound/soc/codecs/cs42l56.c: SOC_DOUBLE_R_SX_TLV("PGA Volume", CS42L56_PGAA_MUX_VOLUME, sound/soc/codecs/cs42l56.c: SOC_DOUBLE_R_SX_TLV("Headphone Volume", CS42L56_HPA_VOLUME, sound/soc/codecs/cs42l56.c: SOC_DOUBLE_R_SX_TLV("LineOut Volume", CS42L56_LOA_VOLUME, sound/soc/codecs/cs42l56.c: SOC_SINGLE_SX_TLV("Beep Volume", CS42L56_BEEP_FREQ_OFFTIME, sound/soc/codecs/cs42l73.c: SOC_DOUBLE_R_SX_TLV("Headphone Analog Playback Volume", sound/soc/codecs/cs42l73.c: SOC_DOUBLE_R_SX_TLV("LineOut Analog Playback Volume", CS42L73_LOAAVOL, sound/soc/codecs/cs42l73.c: SOC_DOUBLE_R_SX_TLV("Input PGA Analog Volume", CS42L73_MICAPREPGAAVOL, sound/soc/codecs/cs42l73.c: SOC_DOUBLE_R_SX_TLV("Input Path Digital Volume", CS42L73_IPADVOL, sound/soc/codecs/cs42l73.c: SOC_DOUBLE_R_SX_TLV("HL Digital Playback Volume", sound/soc/codecs/cs42l73.c: SOC_SINGLE_SX_TLV("Speakerphone Digital Volume", sound/soc/codecs/cs42l73.c: SOC_SINGLE_SX_TLV("Ear Speaker Digital Volume", sound/soc/codecs/cs53l30.c: SOC_SINGLE_SX_TLV("ADC1A PGA Volume", sound/soc/codecs/cs53l30.c: SOC_SINGLE_SX_TLV("ADC1B PGA Volume", sound/soc/codecs/cs53l30.c: SOC_SINGLE_SX_TLV("ADC2A PGA Volume", sound/soc/codecs/cs53l30.c: SOC_SINGLE_SX_TLV("ADC2B PGA Volume", sound/soc/codecs/cs53l30.c: SOC_SINGLE_SX_TLV("ADC1A Digital Volume", sound/soc/codecs/cs53l30.c: SOC_SINGLE_SX_TLV("ADC1B Digital Volume", sound/soc/codecs/cs53l30.c: SOC_SINGLE_SX_TLV("ADC2A Digital Volume", sound/soc/codecs/cs53l30.c: SOC_SINGLE_SX_TLV("ADC2B Digital Volume", sound/soc/codecs/msm8916-wcd-digital.c: SOC_SINGLE_SX_TLV("IIR1 INP1 Volume", LPASS_CDC_IIR1_GAIN_B1_CTL, sound/soc/codecs/msm8916-wcd-digital.c: SOC_SINGLE_SX_TLV("IIR1 INP2 Volume", LPASS_CDC_IIR1_GAIN_B2_CTL, sound/soc/codecs/msm8916-wcd-digital.c: SOC_SINGLE_SX_TLV("IIR1 INP3 Volume", LPASS_CDC_IIR1_GAIN_B3_CTL, sound/soc/codecs/msm8916-wcd-digital.c: SOC_SINGLE_SX_TLV("IIR1 INP4 Volume", LPASS_CDC_IIR1_GAIN_B4_CTL, sound/soc/codecs/msm8916-wcd-digital.c: SOC_SINGLE_SX_TLV("IIR2 INP1 Volume", LPASS_CDC_IIR2_GAIN_B1_CTL, sound/soc/codecs/msm8916-wcd-digital.c: SOC_SINGLE_SX_TLV("IIR2 INP2 Volume", LPASS_CDC_IIR2_GAIN_B2_CTL, sound/soc/codecs/msm8916-wcd-digital.c: SOC_SINGLE_SX_TLV("IIR2 INP3 Volume", LPASS_CDC_IIR2_GAIN_B3_CTL, sound/soc/codecs/msm8916-wcd-digital.c: SOC_SINGLE_SX_TLV("IIR2 INP4 Volume", LPASS_CDC_IIR2_GAIN_B4_CTL, sound/soc/codecs/wcd9335.c: SOC_SINGLE_SX_TLV("RX0 Digital Volume", WCD9335_CDC_RX0_RX_VOL_CTL, sound/soc/codecs/wcd9335.c: SOC_SINGLE_SX_TLV("RX1 Digital Volume", WCD9335_CDC_RX1_RX_VOL_CTL, sound/soc/codecs/wcd9335.c: SOC_SINGLE_SX_TLV("RX2 Digital Volume", WCD9335_CDC_RX2_RX_VOL_CTL, sound/soc/codecs/wcd9335.c: SOC_SINGLE_SX_TLV("RX3 Digital Volume", WCD9335_CDC_RX3_RX_VOL_CTL, sound/soc/codecs/wcd9335.c: SOC_SINGLE_SX_TLV("RX4 Digital Volume", WCD9335_CDC_RX4_RX_VOL_CTL, sound/soc/codecs/wcd9335.c: SOC_SINGLE_SX_TLV("RX5 Digital Volume", WCD9335_CDC_RX5_RX_VOL_CTL, sound/soc/codecs/wcd9335.c: SOC_SINGLE_SX_TLV("RX6 Digital Volume", WCD9335_CDC_RX6_RX_VOL_CTL, sound/soc/codecs/wcd9335.c: SOC_SINGLE_SX_TLV("RX7 Digital Volume", WCD9335_CDC_RX7_RX_VOL_CTL, sound/soc/codecs/wcd9335.c: SOC_SINGLE_SX_TLV("RX8 Digital Volume", WCD9335_CDC_RX8_RX_VOL_CTL, sound/soc/codecs/wcd9335.c: SOC_SINGLE_SX_TLV("RX0 Mix Digital Volume", sound/soc/codecs/wcd9335.c: SOC_SINGLE_SX_TLV("RX1 Mix Digital Volume", sound/soc/codecs/wcd9335.c: SOC_SINGLE_SX_TLV("RX2 Mix Digital Volume", sound/soc/codecs/wcd9335.c: SOC_SINGLE_SX_TLV("RX3 Mix Digital Volume", sound/soc/codecs/wcd9335.c: SOC_SINGLE_SX_TLV("RX4 Mix Digital Volume", sound/soc/codecs/wcd9335.c: SOC_SINGLE_SX_TLV("RX5 Mix Digital Volume", sound/soc/codecs/wcd9335.c: SOC_SINGLE_SX_TLV("RX6 Mix Digital Volume", sound/soc/codecs/wcd9335.c: SOC_SINGLE_SX_TLV("RX7 Mix Digital Volume", sound/soc/codecs/wcd9335.c: SOC_SINGLE_SX_TLV("RX8 Mix Digital Volume", Thanks, Mark

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"topology: nhlt plugin: second attempt to fix the .so linking" cause build static library failed
by GitHub issues - opened 02 Jun '22

02 Jun '22

alsa-project/alsa-utils issue #152 was opened from youling257: ./configure --prefix=/system/usr --with-alsa-inc-prefix=/system/usr/include --with-alsa-prefix=/system/usr/lib --enable-static=yes --enable-shared=no make LDFLAGS="--static" -j8 Revert "topology: nhlt plugin: second attempt to fix the .so linking" can build static library. Issue URL : https://github.com/alsa-project/alsa-utils/issues/152 Repository URL: https://github.com/alsa-project/alsa-utils

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[PATCH v3] ASoC: fsl_xcvr:Fix unbalanced pm_runtime_enable in fsl_xcvr_probe
by zhangqilong 02 Jun '22

02 Jun '22

a) Add missing pm_runtime_disable() when probe error out. It could avoid pm_runtime implementation complains when removing and probing again the driver. b) Add remove for missing pm_runtime_disable(). v2: --- Add remove to put PM usage counter. v3: --- Modify commit message. Fix:c590fa80b3928 ("ASoC: fsl_xcvr: register platform component before registering cpu dai") Signed-off-by: Zhang Qilong <zhangqilong3(a)huawei.com> --- sound/soc/fsl/fsl_xcvr.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/soc/fsl/fsl_xcvr.c b/sound/soc/fsl/fsl_xcvr.c index d0556c79fdb1..55e640cba87d 100644 --- a/sound/soc/fsl/fsl_xcvr.c +++ b/sound/soc/fsl/fsl_xcvr.c @@ -1228,6 +1228,7 @@ static int fsl_xcvr_probe(struct platform_device *pdev) */ ret = devm_snd_dmaengine_pcm_register(dev, NULL, 0); if (ret) { + pm_runtime_disable(dev); dev_err(dev, "failed to pcm register\n"); return ret; } @@ -1235,6 +1236,7 @@ static int fsl_xcvr_probe(struct platform_device *pdev) ret = devm_snd_soc_register_component(dev, &fsl_xcvr_comp, &fsl_xcvr_dai, 1); if (ret) { + pm_runtime_disable(dev); dev_err(dev, "failed to register component %s\n", fsl_xcvr_comp.name); } @@ -1242,6 +1244,12 @@ static int fsl_xcvr_probe(struct platform_device *pdev) return ret; } +static int fsl_xcvr_remove(struct platform_device *pdev) +{ + pm_runtime_disable(&pdev->dev); + return 0; +} + static __maybe_unused int fsl_xcvr_runtime_suspend(struct device *dev) { struct fsl_xcvr *xcvr = dev_get_drvdata(dev); @@ -1370,6 +1378,7 @@ static struct platform_driver fsl_xcvr_driver = { .pm = &fsl_xcvr_pm_ops, .of_match_table = fsl_xcvr_dt_ids, }, + .remove = fsl_xcvr_remove, }; module_platform_driver(fsl_xcvr_driver); -- 2.31.1

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[PATCH V2] ASoC: fsl_xcvr:Fix unbalanced pm_runtime_enable in fsl_xcvr_probe
by zhangqilong 02 Jun '22

02 Jun '22

a) Add missing pm_runtime_disable() when probe error out. It could avoid pm_runtime implementation complains when removing and probing again the driver. b) Add remove for missing pm_runtime_disable(). v2: add remove to put PM usage counter. Fix:c590fa80b3928 ("ASoC: fsl_xcvr: register platform component before registering cpu dai") Signed-off-by: Zhang Qilong <zhangqilong3(a)huawei.com> --- sound/soc/fsl/fsl_xcvr.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/soc/fsl/fsl_xcvr.c b/sound/soc/fsl/fsl_xcvr.c index d0556c79fdb1..55e640cba87d 100644 --- a/sound/soc/fsl/fsl_xcvr.c +++ b/sound/soc/fsl/fsl_xcvr.c @@ -1228,6 +1228,7 @@ static int fsl_xcvr_probe(struct platform_device *pdev) */ ret = devm_snd_dmaengine_pcm_register(dev, NULL, 0); if (ret) { + pm_runtime_disable(dev); dev_err(dev, "failed to pcm register\n"); return ret; } @@ -1235,6 +1236,7 @@ static int fsl_xcvr_probe(struct platform_device *pdev) ret = devm_snd_soc_register_component(dev, &fsl_xcvr_comp, &fsl_xcvr_dai, 1); if (ret) { + pm_runtime_disable(dev); dev_err(dev, "failed to register component %s\n", fsl_xcvr_comp.name); } @@ -1242,6 +1244,12 @@ static int fsl_xcvr_probe(struct platform_device *pdev) return ret; } +static int fsl_xcvr_remove(struct platform_device *pdev) +{ + pm_runtime_disable(&pdev->dev); + return 0; +} + static __maybe_unused int fsl_xcvr_runtime_suspend(struct device *dev) { struct fsl_xcvr *xcvr = dev_get_drvdata(dev); @@ -1370,6 +1378,7 @@ static struct platform_driver fsl_xcvr_driver = { .pm = &fsl_xcvr_pm_ops, .of_match_table = fsl_xcvr_dt_ids, }, + .remove = fsl_xcvr_remove, }; module_platform_driver(fsl_xcvr_driver); -- 2.31.1

2 1

1.2.7: missing git tag
by GitHub issues - opened 01 Jun '22

01 Jun '22

alsa-project/alsa-utils issue #151 was opened from kloczek: https://www.alsa-project.org/wiki/Main_Page says that last release is 1.2.7 but there is no git tag for that version here. Issue URL : https://github.com/alsa-project/alsa-utils/issues/151 Repository URL: https://github.com/alsa-project/alsa-utils

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1.2.7: missing git tag
by GitHub issues - opened 01 Jun '22

01 Jun '22

alsa-project/alsa-lib issue #232 was opened from kloczek: https://www.alsa-project.org/wiki/Main_Page says that last release is 1.2.7 but there is no git tag for that version here. Issue URL : https://github.com/alsa-project/alsa-lib/issues/232 Repository URL: https://github.com/alsa-project/alsa-lib

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soundwire: device drivers for PC based hardware
by Muni Sekhar 01 Jun '22

01 Jun '22

Hi all, I am looking for the soundwire device drivers which run on x86 CPU for PC based platforms. linux/drivers/soundwire at master · torvalds/linux · GitHub Can the above mentioned mainline tree drivers can be used for this purpose? config SOUNDWIRE_INTEL depends on ACPI && SND_SOC config SOUNDWIRE_QCOM depends on SND_SOC As per the Kconfig file, both the Intel and Qualcomm SoundWire Master driver depend on SND_SOC, does it mean it is only possible to run on SoC CPU(Arm chipsets)? -- Thanks, Sekhar

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Re: [syzbot] KASAN: use-after-free Read in snd_rawmidi_transmit
by syzbot 01 Jun '22

01 Jun '22

syzbot has bisected this issue to: commit d098538ed4e8a6c09f86cf243f406c1451066040 Author: Sasha Neftin <sasha.neftin(a)intel.com> Date: Mon Mar 21 06:05:10 2022 +0000 igc: Remove igc_set_spd_dplx method bisection log: https://syzkaller.appspot.com/x/bisect.txt?x=16458313f00000 start commit: 2a5699b0de4e Merge tag 'leds-5.19-rc1' of git://git.kernel.. git tree: upstream final oops: https://syzkaller.appspot.com/x/report.txt?x=15458313f00000 console output: https://syzkaller.appspot.com/x/log.txt?x=11458313f00000 kernel config: https://syzkaller.appspot.com/x/.config?x=f915857c44e13980 dashboard link: https://syzkaller.appspot.com/bug?extid=39e3268af9968f153591 syz repro: https://syzkaller.appspot.com/x/repro.syz?x=119d1fcbf00000 C reproducer: https://syzkaller.appspot.com/x/repro.c?x=17716bedf00000 Reported-by: syzbot+39e3268af9968f153591(a)syzkaller.appspotmail.com Fixes: d098538ed4e8 ("igc: Remove igc_set_spd_dplx method") For information about bisection process see: https://goo.gl/tpsmEJ#bisection

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[PATCH AUTOSEL 5.15 14/37] ALSA: usb-audio: Move generic implicit fb quirk entries into quirks.c
by Sasha Levin 01 Jun '22

01 Jun '22

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 67d64069bc0867e52e73a1e255b17462005ca9b4 ] Use the new quirk bits to manage the generic implicit fb quirk entries. This makes easier to compare with other devices. Link: https://lore.kernel.org/r/20220421064101.12456-2-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/usb/implicit.c | 5 ----- sound/usb/quirks.c | 6 ++++++ 2 files changed, 6 insertions(+), 5 deletions(-) diff --git a/sound/usb/implicit.c b/sound/usb/implicit.c index 1fd087128538..e1bf1b5da423 100644 --- a/sound/usb/implicit.c +++ b/sound/usb/implicit.c @@ -45,11 +45,6 @@ struct snd_usb_implicit_fb_match { /* Implicit feedback quirk table for playback */ static const struct snd_usb_implicit_fb_match playback_implicit_fb_quirks[] = { - /* Generic matching */ - IMPLICIT_FB_GENERIC_DEV(0x0499, 0x1509), /* Steinberg UR22 */ - IMPLICIT_FB_GENERIC_DEV(0x0763, 0x2030), /* M-Audio Fast Track C400 */ - IMPLICIT_FB_GENERIC_DEV(0x0763, 0x2031), /* M-Audio Fast Track C600 */ - /* Fixed EP */ /* FIXME: check the availability of generic matching */ IMPLICIT_FB_FIXED_DEV(0x0763, 0x2080, 0x81, 2), /* M-Audio FastTrack Ultra */ diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index ab9f3da49941..5461cdf907e2 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1793,6 +1793,8 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = { QUIRK_FLAG_CTL_MSG_DELAY_1M | QUIRK_FLAG_IGNORE_CTL_ERROR), DEVICE_FLG(0x046d, 0x09a4, /* Logitech QuickCam E 3500 */ QUIRK_FLAG_CTL_MSG_DELAY_1M | QUIRK_FLAG_IGNORE_CTL_ERROR), + DEVICE_FLG(0x0499, 0x1509, /* Steinberg UR22 */ + QUIRK_FLAG_GENERIC_IMPLICIT_FB), DEVICE_FLG(0x04d8, 0xfeea, /* Benchmark DAC1 Pre */ QUIRK_FLAG_GET_SAMPLE_RATE), DEVICE_FLG(0x04e8, 0xa051, /* Samsung USBC Headset (AKG) */ @@ -1824,6 +1826,10 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = { QUIRK_FLAG_IGNORE_CTL_ERROR), DEVICE_FLG(0x074d, 0x3553, /* Outlaw RR2150 (Micronas UAC3553B) */ QUIRK_FLAG_GET_SAMPLE_RATE), + DEVICE_FLG(0x0763, 0x2030, /* M-Audio Fast Track C400 */ + QUIRK_FLAG_GENERIC_IMPLICIT_FB), + DEVICE_FLG(0x0763, 0x2031, /* M-Audio Fast Track C600 */ + QUIRK_FLAG_GENERIC_IMPLICIT_FB), DEVICE_FLG(0x08bb, 0x2702, /* LineX FM Transmitter */ QUIRK_FLAG_IGNORE_CTL_ERROR), DEVICE_FLG(0x0951, 0x16ad, /* Kingston HyperX */ -- 2.35.1

1 0

[PATCH AUTOSEL 5.15 13/37] ALSA: usb-audio: Add quirk bits for enabling/disabling generic implicit fb
by Sasha Levin 01 Jun '22

01 Jun '22

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 0f1f7a6661394fe4a53db254c346d6aa2dd64397 ] For making easier to test, add the new quirk_flags bits 17 and 18 to enable and disable the generic implicit feedback mode. The bit 17 is equivalent with implicit_fb=1 option, applying the generic implicit feedback sync mode. OTOH, the bit 18 disables the implicit fb mode forcibly. Link: https://lore.kernel.org/r/20220421064101.12456-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- Documentation/sound/alsa-configuration.rst | 4 +++- sound/usb/implicit.c | 5 ++++- sound/usb/usbaudio.h | 6 ++++++ 3 files changed, 13 insertions(+), 2 deletions(-) diff --git a/Documentation/sound/alsa-configuration.rst b/Documentation/sound/alsa-configuration.rst index 65f61695f561..5d093fb4896b 100644 --- a/Documentation/sound/alsa-configuration.rst +++ b/Documentation/sound/alsa-configuration.rst @@ -2237,7 +2237,7 @@ implicit_fb Apply the generic implicit feedback sync mode. When this is set and the playback stream sync mode is ASYNC, the driver tries to tie an adjacent ASYNC capture stream as the implicit feedback - source. + source. This is equivalent with quirk_flags bit 17. use_vmalloc Use vmalloc() for allocations of the PCM buffers (default: yes). For architectures with non-coherent memory like ARM or MIPS, the @@ -2279,6 +2279,8 @@ quirk_flags * bit 14: Ignore errors for mixer access * bit 15: Support generic DSD raw U32_BE format * bit 16: Set up the interface at first like UAC1 + * bit 17: Apply the generic implicit feedback sync mode + * bit 18: Don't apply implicit feedback sync mode This module supports multiple devices, autoprobe and hotplugging. diff --git a/sound/usb/implicit.c b/sound/usb/implicit.c index 2d444ec74202..1fd087128538 100644 --- a/sound/usb/implicit.c +++ b/sound/usb/implicit.c @@ -350,7 +350,8 @@ static int audioformat_implicit_fb_quirk(struct snd_usb_audio *chip, } /* Try the generic implicit fb if available */ - if (chip->generic_implicit_fb) + if (chip->generic_implicit_fb || + (chip->quirk_flags & QUIRK_FLAG_GENERIC_IMPLICIT_FB)) return add_generic_implicit_fb(chip, fmt, alts); /* No quirk */ @@ -387,6 +388,8 @@ int snd_usb_parse_implicit_fb_quirk(struct snd_usb_audio *chip, struct audioformat *fmt, struct usb_host_interface *alts) { + if (chip->quirk_flags & QUIRK_FLAG_SKIP_IMPLICIT_FB) + return 0; if (fmt->endpoint & USB_DIR_IN) return audioformat_capture_quirk(chip, fmt, alts); else diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index b8359a0aa008..044cd7ab27cb 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -164,6 +164,10 @@ extern bool snd_usb_skip_validation; * Support generic DSD raw U32_BE format * QUIRK_FLAG_SET_IFACE_FIRST: * Set up the interface at first like UAC1 + * QUIRK_FLAG_GENERIC_IMPLICIT_FB + * Apply the generic implicit feedback sync mode (same as implicit_fb=1 option) + * QUIRK_FLAG_SKIP_IMPLICIT_FB + * Don't apply implicit feedback sync mode */ #define QUIRK_FLAG_GET_SAMPLE_RATE (1U << 0) @@ -183,5 +187,7 @@ extern bool snd_usb_skip_validation; #define QUIRK_FLAG_IGNORE_CTL_ERROR (1U << 14) #define QUIRK_FLAG_DSD_RAW (1U << 15) #define QUIRK_FLAG_SET_IFACE_FIRST (1U << 16) +#define QUIRK_FLAG_GENERIC_IMPLICIT_FB (1U << 17) +#define QUIRK_FLAG_SKIP_IMPLICIT_FB (1U << 18) #endif /* __USBAUDIO_H */ -- 2.35.1

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[PATCH AUTOSEL 5.17 17/48] ALSA: usb-audio: Move generic implicit fb quirk entries into quirks.c
by Sasha Levin 01 Jun '22

01 Jun '22

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 67d64069bc0867e52e73a1e255b17462005ca9b4 ] Use the new quirk bits to manage the generic implicit fb quirk entries. This makes easier to compare with other devices. Link: https://lore.kernel.org/r/20220421064101.12456-2-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/usb/implicit.c | 5 ----- sound/usb/quirks.c | 6 ++++++ 2 files changed, 6 insertions(+), 5 deletions(-) diff --git a/sound/usb/implicit.c b/sound/usb/implicit.c index 1fd087128538..e1bf1b5da423 100644 --- a/sound/usb/implicit.c +++ b/sound/usb/implicit.c @@ -45,11 +45,6 @@ struct snd_usb_implicit_fb_match { /* Implicit feedback quirk table for playback */ static const struct snd_usb_implicit_fb_match playback_implicit_fb_quirks[] = { - /* Generic matching */ - IMPLICIT_FB_GENERIC_DEV(0x0499, 0x1509), /* Steinberg UR22 */ - IMPLICIT_FB_GENERIC_DEV(0x0763, 0x2030), /* M-Audio Fast Track C400 */ - IMPLICIT_FB_GENERIC_DEV(0x0763, 0x2031), /* M-Audio Fast Track C600 */ - /* Fixed EP */ /* FIXME: check the availability of generic matching */ IMPLICIT_FB_FIXED_DEV(0x0763, 0x2080, 0x81, 2), /* M-Audio FastTrack Ultra */ diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index ab9f3da49941..5461cdf907e2 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1793,6 +1793,8 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = { QUIRK_FLAG_CTL_MSG_DELAY_1M | QUIRK_FLAG_IGNORE_CTL_ERROR), DEVICE_FLG(0x046d, 0x09a4, /* Logitech QuickCam E 3500 */ QUIRK_FLAG_CTL_MSG_DELAY_1M | QUIRK_FLAG_IGNORE_CTL_ERROR), + DEVICE_FLG(0x0499, 0x1509, /* Steinberg UR22 */ + QUIRK_FLAG_GENERIC_IMPLICIT_FB), DEVICE_FLG(0x04d8, 0xfeea, /* Benchmark DAC1 Pre */ QUIRK_FLAG_GET_SAMPLE_RATE), DEVICE_FLG(0x04e8, 0xa051, /* Samsung USBC Headset (AKG) */ @@ -1824,6 +1826,10 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = { QUIRK_FLAG_IGNORE_CTL_ERROR), DEVICE_FLG(0x074d, 0x3553, /* Outlaw RR2150 (Micronas UAC3553B) */ QUIRK_FLAG_GET_SAMPLE_RATE), + DEVICE_FLG(0x0763, 0x2030, /* M-Audio Fast Track C400 */ + QUIRK_FLAG_GENERIC_IMPLICIT_FB), + DEVICE_FLG(0x0763, 0x2031, /* M-Audio Fast Track C600 */ + QUIRK_FLAG_GENERIC_IMPLICIT_FB), DEVICE_FLG(0x08bb, 0x2702, /* LineX FM Transmitter */ QUIRK_FLAG_IGNORE_CTL_ERROR), DEVICE_FLG(0x0951, 0x16ad, /* Kingston HyperX */ -- 2.35.1

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[PATCH AUTOSEL 5.17 16/48] ALSA: usb-audio: Add quirk bits for enabling/disabling generic implicit fb
by Sasha Levin 01 Jun '22

01 Jun '22

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 0f1f7a6661394fe4a53db254c346d6aa2dd64397 ] For making easier to test, add the new quirk_flags bits 17 and 18 to enable and disable the generic implicit feedback mode. The bit 17 is equivalent with implicit_fb=1 option, applying the generic implicit feedback sync mode. OTOH, the bit 18 disables the implicit fb mode forcibly. Link: https://lore.kernel.org/r/20220421064101.12456-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- Documentation/sound/alsa-configuration.rst | 4 +++- sound/usb/implicit.c | 5 ++++- sound/usb/usbaudio.h | 6 ++++++ 3 files changed, 13 insertions(+), 2 deletions(-) diff --git a/Documentation/sound/alsa-configuration.rst b/Documentation/sound/alsa-configuration.rst index 34888d4fc4a8..21ab5e6f7062 100644 --- a/Documentation/sound/alsa-configuration.rst +++ b/Documentation/sound/alsa-configuration.rst @@ -2246,7 +2246,7 @@ implicit_fb Apply the generic implicit feedback sync mode. When this is set and the playback stream sync mode is ASYNC, the driver tries to tie an adjacent ASYNC capture stream as the implicit feedback - source. + source. This is equivalent with quirk_flags bit 17. use_vmalloc Use vmalloc() for allocations of the PCM buffers (default: yes). For architectures with non-coherent memory like ARM or MIPS, the @@ -2288,6 +2288,8 @@ quirk_flags * bit 14: Ignore errors for mixer access * bit 15: Support generic DSD raw U32_BE format * bit 16: Set up the interface at first like UAC1 + * bit 17: Apply the generic implicit feedback sync mode + * bit 18: Don't apply implicit feedback sync mode This module supports multiple devices, autoprobe and hotplugging. diff --git a/sound/usb/implicit.c b/sound/usb/implicit.c index 2d444ec74202..1fd087128538 100644 --- a/sound/usb/implicit.c +++ b/sound/usb/implicit.c @@ -350,7 +350,8 @@ static int audioformat_implicit_fb_quirk(struct snd_usb_audio *chip, } /* Try the generic implicit fb if available */ - if (chip->generic_implicit_fb) + if (chip->generic_implicit_fb || + (chip->quirk_flags & QUIRK_FLAG_GENERIC_IMPLICIT_FB)) return add_generic_implicit_fb(chip, fmt, alts); /* No quirk */ @@ -387,6 +388,8 @@ int snd_usb_parse_implicit_fb_quirk(struct snd_usb_audio *chip, struct audioformat *fmt, struct usb_host_interface *alts) { + if (chip->quirk_flags & QUIRK_FLAG_SKIP_IMPLICIT_FB) + return 0; if (fmt->endpoint & USB_DIR_IN) return audioformat_capture_quirk(chip, fmt, alts); else diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index b8359a0aa008..044cd7ab27cb 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -164,6 +164,10 @@ extern bool snd_usb_skip_validation; * Support generic DSD raw U32_BE format * QUIRK_FLAG_SET_IFACE_FIRST: * Set up the interface at first like UAC1 + * QUIRK_FLAG_GENERIC_IMPLICIT_FB + * Apply the generic implicit feedback sync mode (same as implicit_fb=1 option) + * QUIRK_FLAG_SKIP_IMPLICIT_FB + * Don't apply implicit feedback sync mode */ #define QUIRK_FLAG_GET_SAMPLE_RATE (1U << 0) @@ -183,5 +187,7 @@ extern bool snd_usb_skip_validation; #define QUIRK_FLAG_IGNORE_CTL_ERROR (1U << 14) #define QUIRK_FLAG_DSD_RAW (1U << 15) #define QUIRK_FLAG_SET_IFACE_FIRST (1U << 16) +#define QUIRK_FLAG_GENERIC_IMPLICIT_FB (1U << 17) +#define QUIRK_FLAG_SKIP_IMPLICIT_FB (1U << 18) #endif /* __USBAUDIO_H */ -- 2.35.1

1 0

[PATCH AUTOSEL 5.17 06/48] ASoC: amd: Add driver data to acp6x machine driver
by Sasha Levin 01 Jun '22

01 Jun '22

From: Mario Limonciello <mario.limonciello(a)amd.com> [ Upstream commit e521f087780d07731e8c950f2f34d08358c86bc9 ] Currently all of the quirked systems use the same card and so the DMI quirk list doesn't contain driver data. Add driver data to these quirks and then check the data was present or not. This will allow potentially setting quirks for systems with faulty firmware that claims to have a DMIC but doesn't really. Signed-off-by: Mario Limonciello <mario.limonciello(a)amd.com> Link: https://lore.kernel.org/r/20220411134532.13538-2-mario.limonciello@amd.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/amd/yc/acp6x-mach.c | 29 +++++++++++++++++++++++++---- 1 file changed, 25 insertions(+), 4 deletions(-) diff --git a/sound/soc/amd/yc/acp6x-mach.c b/sound/soc/amd/yc/acp6x-mach.c index 9a767f47b89f..959b70e8baf2 100644 --- a/sound/soc/amd/yc/acp6x-mach.c +++ b/sound/soc/amd/yc/acp6x-mach.c @@ -45,108 +45,126 @@ 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, "21D2"), } }, { + .driver_data = &acp6x_card, .matches = { DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"), DMI_MATCH(DMI_PRODUCT_NAME, "21D3"), } }, { + .driver_data = &acp6x_card, .matches = { DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"), DMI_MATCH(DMI_PRODUCT_NAME, "21D4"), } }, { + .driver_data = &acp6x_card, .matches = { DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"), DMI_MATCH(DMI_PRODUCT_NAME, "21D5"), } }, { + .driver_data = &acp6x_card, .matches = { DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"), DMI_MATCH(DMI_PRODUCT_NAME, "21CF"), } }, { + .driver_data = &acp6x_card, .matches = { DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"), DMI_MATCH(DMI_PRODUCT_NAME, "21CG"), } }, { + .driver_data = &acp6x_card, .matches = { DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"), DMI_MATCH(DMI_PRODUCT_NAME, "21CQ"), } }, { + .driver_data = &acp6x_card, .matches = { DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"), DMI_MATCH(DMI_PRODUCT_NAME, "21CR"), } }, { + .driver_data = &acp6x_card, .matches = { DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"), DMI_MATCH(DMI_PRODUCT_NAME, "21AW"), } }, { + .driver_data = &acp6x_card, .matches = { DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"), DMI_MATCH(DMI_PRODUCT_NAME, "21AX"), } }, { + .driver_data = &acp6x_card, .matches = { DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"), DMI_MATCH(DMI_PRODUCT_NAME, "21BN"), } }, { + .driver_data = &acp6x_card, .matches = { DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"), DMI_MATCH(DMI_PRODUCT_NAME, "21BQ"), } }, { + .driver_data = &acp6x_card, .matches = { DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"), DMI_MATCH(DMI_PRODUCT_NAME, "21CH"), } }, { + .driver_data = &acp6x_card, .matches = { DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"), DMI_MATCH(DMI_PRODUCT_NAME, "21CJ"), } }, { + .driver_data = &acp6x_card, .matches = { DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"), DMI_MATCH(DMI_PRODUCT_NAME, "21CK"), } }, { + .driver_data = &acp6x_card, .matches = { DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"), DMI_MATCH(DMI_PRODUCT_NAME, "21CL"), } }, { + .driver_data = &acp6x_card, .matches = { DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"), DMI_MATCH(DMI_PRODUCT_NAME, "21D8"), } }, { + .driver_data = &acp6x_card, .matches = { DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"), DMI_MATCH(DMI_PRODUCT_NAME, "21D9"), @@ -157,18 +175,21 @@ static const struct dmi_system_id yc_acp_quirk_table[] = { static int acp6x_probe(struct platform_device *pdev) { + const struct dmi_system_id *dmi_id; struct acp6x_pdm *machine = NULL; struct snd_soc_card *card; int ret; - const struct dmi_system_id *dmi_id; + /* check for any DMI overrides */ dmi_id = dmi_first_match(yc_acp_quirk_table); - if (!dmi_id) + if (dmi_id) + platform_set_drvdata(pdev, dmi_id->driver_data); + + card = platform_get_drvdata(pdev); + if (!card) return -ENODEV; - card = &acp6x_card; acp6x_card.dev = &pdev->dev; - platform_set_drvdata(pdev, card); snd_soc_card_set_drvdata(card, machine); ret = devm_snd_soc_register_card(&pdev->dev, card); if (ret) { -- 2.35.1

1 0

[PATCH AUTOSEL 5.18 17/49] ALSA: usb-audio: Move generic implicit fb quirk entries into quirks.c
by Sasha Levin 01 Jun '22

01 Jun '22

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 67d64069bc0867e52e73a1e255b17462005ca9b4 ] Use the new quirk bits to manage the generic implicit fb quirk entries. This makes easier to compare with other devices. Link: https://lore.kernel.org/r/20220421064101.12456-2-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/usb/implicit.c | 5 ----- sound/usb/quirks.c | 6 ++++++ 2 files changed, 6 insertions(+), 5 deletions(-) diff --git a/sound/usb/implicit.c b/sound/usb/implicit.c index 1fd087128538..e1bf1b5da423 100644 --- a/sound/usb/implicit.c +++ b/sound/usb/implicit.c @@ -45,11 +45,6 @@ struct snd_usb_implicit_fb_match { /* Implicit feedback quirk table for playback */ static const struct snd_usb_implicit_fb_match playback_implicit_fb_quirks[] = { - /* Generic matching */ - IMPLICIT_FB_GENERIC_DEV(0x0499, 0x1509), /* Steinberg UR22 */ - IMPLICIT_FB_GENERIC_DEV(0x0763, 0x2030), /* M-Audio Fast Track C400 */ - IMPLICIT_FB_GENERIC_DEV(0x0763, 0x2031), /* M-Audio Fast Track C600 */ - /* Fixed EP */ /* FIXME: check the availability of generic matching */ IMPLICIT_FB_FIXED_DEV(0x0763, 0x2080, 0x81, 2), /* M-Audio FastTrack Ultra */ diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index fbbe59054c3f..e8468f9b007d 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1793,6 +1793,8 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = { QUIRK_FLAG_CTL_MSG_DELAY_1M | QUIRK_FLAG_IGNORE_CTL_ERROR), DEVICE_FLG(0x046d, 0x09a4, /* Logitech QuickCam E 3500 */ QUIRK_FLAG_CTL_MSG_DELAY_1M | QUIRK_FLAG_IGNORE_CTL_ERROR), + DEVICE_FLG(0x0499, 0x1509, /* Steinberg UR22 */ + QUIRK_FLAG_GENERIC_IMPLICIT_FB), DEVICE_FLG(0x04d8, 0xfeea, /* Benchmark DAC1 Pre */ QUIRK_FLAG_GET_SAMPLE_RATE), DEVICE_FLG(0x04e8, 0xa051, /* Samsung USBC Headset (AKG) */ @@ -1826,6 +1828,10 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = { QUIRK_FLAG_GET_SAMPLE_RATE), DEVICE_FLG(0x074d, 0x3553, /* Outlaw RR2150 (Micronas UAC3553B) */ QUIRK_FLAG_GET_SAMPLE_RATE), + DEVICE_FLG(0x0763, 0x2030, /* M-Audio Fast Track C400 */ + QUIRK_FLAG_GENERIC_IMPLICIT_FB), + DEVICE_FLG(0x0763, 0x2031, /* M-Audio Fast Track C600 */ + QUIRK_FLAG_GENERIC_IMPLICIT_FB), DEVICE_FLG(0x08bb, 0x2702, /* LineX FM Transmitter */ QUIRK_FLAG_IGNORE_CTL_ERROR), DEVICE_FLG(0x0951, 0x16ad, /* Kingston HyperX */ -- 2.35.1

1 0

[PATCH AUTOSEL 5.18 16/49] ALSA: usb-audio: Add quirk bits for enabling/disabling generic implicit fb
by Sasha Levin 01 Jun '22

01 Jun '22

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 0f1f7a6661394fe4a53db254c346d6aa2dd64397 ] For making easier to test, add the new quirk_flags bits 17 and 18 to enable and disable the generic implicit feedback mode. The bit 17 is equivalent with implicit_fb=1 option, applying the generic implicit feedback sync mode. OTOH, the bit 18 disables the implicit fb mode forcibly. Link: https://lore.kernel.org/r/20220421064101.12456-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- Documentation/sound/alsa-configuration.rst | 4 +++- sound/usb/implicit.c | 5 ++++- sound/usb/usbaudio.h | 6 ++++++ 3 files changed, 13 insertions(+), 2 deletions(-) diff --git a/Documentation/sound/alsa-configuration.rst b/Documentation/sound/alsa-configuration.rst index 34888d4fc4a8..21ab5e6f7062 100644 --- a/Documentation/sound/alsa-configuration.rst +++ b/Documentation/sound/alsa-configuration.rst @@ -2246,7 +2246,7 @@ implicit_fb Apply the generic implicit feedback sync mode. When this is set and the playback stream sync mode is ASYNC, the driver tries to tie an adjacent ASYNC capture stream as the implicit feedback - source. + source. This is equivalent with quirk_flags bit 17. use_vmalloc Use vmalloc() for allocations of the PCM buffers (default: yes). For architectures with non-coherent memory like ARM or MIPS, the @@ -2288,6 +2288,8 @@ quirk_flags * bit 14: Ignore errors for mixer access * bit 15: Support generic DSD raw U32_BE format * bit 16: Set up the interface at first like UAC1 + * bit 17: Apply the generic implicit feedback sync mode + * bit 18: Don't apply implicit feedback sync mode This module supports multiple devices, autoprobe and hotplugging. diff --git a/sound/usb/implicit.c b/sound/usb/implicit.c index 2d444ec74202..1fd087128538 100644 --- a/sound/usb/implicit.c +++ b/sound/usb/implicit.c @@ -350,7 +350,8 @@ static int audioformat_implicit_fb_quirk(struct snd_usb_audio *chip, } /* Try the generic implicit fb if available */ - if (chip->generic_implicit_fb) + if (chip->generic_implicit_fb || + (chip->quirk_flags & QUIRK_FLAG_GENERIC_IMPLICIT_FB)) return add_generic_implicit_fb(chip, fmt, alts); /* No quirk */ @@ -387,6 +388,8 @@ int snd_usb_parse_implicit_fb_quirk(struct snd_usb_audio *chip, struct audioformat *fmt, struct usb_host_interface *alts) { + if (chip->quirk_flags & QUIRK_FLAG_SKIP_IMPLICIT_FB) + return 0; if (fmt->endpoint & USB_DIR_IN) return audioformat_capture_quirk(chip, fmt, alts); else diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index b8359a0aa008..044cd7ab27cb 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -164,6 +164,10 @@ extern bool snd_usb_skip_validation; * Support generic DSD raw U32_BE format * QUIRK_FLAG_SET_IFACE_FIRST: * Set up the interface at first like UAC1 + * QUIRK_FLAG_GENERIC_IMPLICIT_FB + * Apply the generic implicit feedback sync mode (same as implicit_fb=1 option) + * QUIRK_FLAG_SKIP_IMPLICIT_FB + * Don't apply implicit feedback sync mode */ #define QUIRK_FLAG_GET_SAMPLE_RATE (1U << 0) @@ -183,5 +187,7 @@ extern bool snd_usb_skip_validation; #define QUIRK_FLAG_IGNORE_CTL_ERROR (1U << 14) #define QUIRK_FLAG_DSD_RAW (1U << 15) #define QUIRK_FLAG_SET_IFACE_FIRST (1U << 16) +#define QUIRK_FLAG_GENERIC_IMPLICIT_FB (1U << 17) +#define QUIRK_FLAG_SKIP_IMPLICIT_FB (1U << 18) #endif /* __USBAUDIO_H */ -- 2.35.1

1 0

[PATCH AUTOSEL 5.18 06/49] ASoC: amd: Add driver data to acp6x machine driver
by Sasha Levin 01 Jun '22

01 Jun '22

From: Mario Limonciello <mario.limonciello(a)amd.com> [ Upstream commit e521f087780d07731e8c950f2f34d08358c86bc9 ] Currently all of the quirked systems use the same card and so the DMI quirk list doesn't contain driver data. Add driver data to these quirks and then check the data was present or not. This will allow potentially setting quirks for systems with faulty firmware that claims to have a DMIC but doesn't really. Signed-off-by: Mario Limonciello <mario.limonciello(a)amd.com> Link: https://lore.kernel.org/r/20220411134532.13538-2-mario.limonciello@amd.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/amd/yc/acp6x-mach.c | 29 +++++++++++++++++++++++++---- 1 file changed, 25 insertions(+), 4 deletions(-) diff --git a/sound/soc/amd/yc/acp6x-mach.c b/sound/soc/amd/yc/acp6x-mach.c index 9a767f47b89f..959b70e8baf2 100644 --- a/sound/soc/amd/yc/acp6x-mach.c +++ b/sound/soc/amd/yc/acp6x-mach.c @@ -45,108 +45,126 @@ 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, "21D2"), } }, { + .driver_data = &acp6x_card, .matches = { DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"), DMI_MATCH(DMI_PRODUCT_NAME, "21D3"), } }, { + .driver_data = &acp6x_card, .matches = { DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"), DMI_MATCH(DMI_PRODUCT_NAME, "21D4"), } }, { + .driver_data = &acp6x_card, .matches = { DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"), DMI_MATCH(DMI_PRODUCT_NAME, "21D5"), } }, { + .driver_data = &acp6x_card, .matches = { DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"), DMI_MATCH(DMI_PRODUCT_NAME, "21CF"), } }, { + .driver_data = &acp6x_card, .matches = { DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"), DMI_MATCH(DMI_PRODUCT_NAME, "21CG"), } }, { + .driver_data = &acp6x_card, .matches = { DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"), DMI_MATCH(DMI_PRODUCT_NAME, "21CQ"), } }, { + .driver_data = &acp6x_card, .matches = { DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"), DMI_MATCH(DMI_PRODUCT_NAME, "21CR"), } }, { + .driver_data = &acp6x_card, .matches = { DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"), DMI_MATCH(DMI_PRODUCT_NAME, "21AW"), } }, { + .driver_data = &acp6x_card, .matches = { DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"), DMI_MATCH(DMI_PRODUCT_NAME, "21AX"), } }, { + .driver_data = &acp6x_card, .matches = { DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"), DMI_MATCH(DMI_PRODUCT_NAME, "21BN"), } }, { + .driver_data = &acp6x_card, .matches = { DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"), DMI_MATCH(DMI_PRODUCT_NAME, "21BQ"), } }, { + .driver_data = &acp6x_card, .matches = { DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"), DMI_MATCH(DMI_PRODUCT_NAME, "21CH"), } }, { + .driver_data = &acp6x_card, .matches = { DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"), DMI_MATCH(DMI_PRODUCT_NAME, "21CJ"), } }, { + .driver_data = &acp6x_card, .matches = { DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"), DMI_MATCH(DMI_PRODUCT_NAME, "21CK"), } }, { + .driver_data = &acp6x_card, .matches = { DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"), DMI_MATCH(DMI_PRODUCT_NAME, "21CL"), } }, { + .driver_data = &acp6x_card, .matches = { DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"), DMI_MATCH(DMI_PRODUCT_NAME, "21D8"), } }, { + .driver_data = &acp6x_card, .matches = { DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"), DMI_MATCH(DMI_PRODUCT_NAME, "21D9"), @@ -157,18 +175,21 @@ static const struct dmi_system_id yc_acp_quirk_table[] = { static int acp6x_probe(struct platform_device *pdev) { + const struct dmi_system_id *dmi_id; struct acp6x_pdm *machine = NULL; struct snd_soc_card *card; int ret; - const struct dmi_system_id *dmi_id; + /* check for any DMI overrides */ dmi_id = dmi_first_match(yc_acp_quirk_table); - if (!dmi_id) + if (dmi_id) + platform_set_drvdata(pdev, dmi_id->driver_data); + + card = platform_get_drvdata(pdev); + if (!card) return -ENODEV; - card = &acp6x_card; acp6x_card.dev = &pdev->dev; - platform_set_drvdata(pdev, card); snd_soc_card_set_drvdata(card, machine); ret = devm_snd_soc_register_card(&pdev->dev, card); if (ret) { -- 2.35.1

1 0

[PATCH] ASoC: fsl: Check before clk_put() not needed
by Yihao Han 01 Jun '22

01 Jun '22

clk_put() already checks the clk ptr using !clk and IS_ERR() so there is no need to check it again before calling it. Signed-off-by: Yihao Han <hanyihao(a)vivo.com> --- sound/soc/fsl/imx-sgtl5000.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c index 580a0d963f0e..16a281820186 100644 --- a/sound/soc/fsl/imx-sgtl5000.c +++ b/sound/soc/fsl/imx-sgtl5000.c @@ -185,8 +185,7 @@ static int imx_sgtl5000_probe(struct platform_device *pdev) put_device: put_device(&codec_dev->dev); fail: - if (data && !IS_ERR(data->codec_clk)) - clk_put(data->codec_clk); + clk_put(data->codec_clk); of_node_put(ssi_np); of_node_put(codec_np); -- 2.17.1

2 1

[PATCH] ASoC: Intel: cirrus-common: fix incorrect channel mapping
by Brent Lu 01 Jun '22

01 Jun '22

From: xliu <xiang.liu(a)cirrus.com> The default mapping of ASPRX1 Slot is left channel. Map the slots of right amplifiers (WR and TR) to right channel. Signed-off-by: xliu <xiang.liu(a)cirrus.com> Signed-off-by: Brent Lu <brent.lu(a)intel.com> --- sound/soc/intel/boards/sof_cirrus_common.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/soc/intel/boards/sof_cirrus_common.c b/sound/soc/intel/boards/sof_cirrus_common.c index e71d74ec1b0b..64ca0e3991dc 100644 --- a/sound/soc/intel/boards/sof_cirrus_common.c +++ b/sound/soc/intel/boards/sof_cirrus_common.c @@ -107,6 +107,7 @@ static int cs35l41_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); struct snd_soc_dai *codec_dai; int clk_freq, i, ret; + int rx_ch[2] = {1, 0}; clk_freq = sof_dai_get_bclk(rtd); /* BCLK freq */ @@ -134,6 +135,17 @@ static int cs35l41_hw_params(struct snd_pcm_substream *substream, ret); return ret; } + + /* Setup for R channel Slot: WR and TR */ + if (i % 2) { + ret = snd_soc_dai_set_channel_map(codec_dai, 0, NULL, + ARRAY_SIZE(rx_ch), rx_ch); + if (ret < 0) { + dev_err(codec_dai->dev, "fail to set channel map, ret %d\n", + ret); + return ret; + } + } } return 0; -- 2.25.1

3 2

[PATCH] ASoC: qcom: lpass-platform: Update VMA access permissions in mmap callback
by Srinivasa Rao Mandadapu 01 Jun '22

01 Jun '22

Replace page protection permissions from noncashed to writecombine, in lpass codec DMA path mmp callabck, to support 64 bit chromeOS. Avoid SIGBUS error in userspace caused by noncached permissions in 64 bit chromeOS. Signed-off-by: Srinivasa Rao Mandadapu <quic_srivasam(a)quicinc.com> --- sound/soc/qcom/lpass-platform.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/qcom/lpass-platform.c b/sound/soc/qcom/lpass-platform.c index f03a7ae..b41ab7a 100644 --- a/sound/soc/qcom/lpass-platform.c +++ b/sound/soc/qcom/lpass-platform.c @@ -898,7 +898,7 @@ static int lpass_platform_cdc_dma_mmap(struct snd_pcm_substream *substream, struct snd_pcm_runtime *runtime = substream->runtime; unsigned long size, offset; - vma->vm_page_prot = pgprot_noncached(vma->vm_page_prot); + vma->vm_page_prot = pgprot_writecombine(vma->vm_page_prot); size = vma->vm_end - vma->vm_start; offset = vma->vm_pgoff << PAGE_SHIFT; return io_remap_pfn_range(vma, vma->vm_start, -- 2.7.4

2 1

[PATCH 0/2] Add pinctrl support adsp based platforms
by Srinivasa Rao Mandadapu 01 Jun '22

01 Jun '22

This patch set is to add pinctrl support adsp enabled sc7280 platforms. Srinivasa Rao Mandadapu (2): dt-bindings: pinctrl: qcom: sc7280: Add compatible string for adsp based platforms pinctrl: qcom: sc7280: Add lpi pinctrl variant data structure for adsp based targets .../bindings/pinctrl/qcom,sc7280-lpass-lpi-pinctrl.yaml | 4 +++- drivers/pinctrl/qcom/pinctrl-sc7280-lpass-lpi.c | 14 ++++++++++++++ 2 files changed, 17 insertions(+), 1 deletion(-) -- 2.7.4

3 5

Re: [syzbot] KASAN: use-after-free Read in snd_rawmidi_transmit
by syzbot 01 Jun '22

01 Jun '22

syzbot has found a reproducer for the following issue on: HEAD commit: 2a5699b0de4e Merge tag 'leds-5.19-rc1' of git://git.kernel.. git tree: upstream console+strace: https://syzkaller.appspot.com/x/log.txt?x=13df4d13f00000 kernel config: https://syzkaller.appspot.com/x/.config?x=f915857c44e13980 dashboard link: https://syzkaller.appspot.com/bug?extid=39e3268af9968f153591 compiler: Debian clang version 13.0.1-++20220126092033+75e33f71c2da-1~exp1~20220126212112.63, GNU ld (GNU Binutils for Debian) 2.35.2 syz repro: https://syzkaller.appspot.com/x/repro.syz?x=119d1fcbf00000 C reproducer: https://syzkaller.appspot.com/x/repro.c?x=17716bedf00000 IMPORTANT: if you fix the issue, please add the following tag to the commit: Reported-by: syzbot+39e3268af9968f153591(a)syzkaller.appspotmail.com ================================================================== BUG: KASAN: use-after-free in __snd_rawmidi_transmit_peek sound/core/rawmidi.c:1286 [inline] BUG: KASAN: use-after-free in snd_rawmidi_transmit+0x3f3/0x420 sound/core/rawmidi.c:1411 Read of size 1 at addr ffff888074d3c07f by task kworker/1:1H/756 CPU: 1 PID: 756 Comm: kworker/1:1H Not tainted 5.18.0-syzkaller-11650-g2a5699b0de4e #0 Hardware name: Google Google Compute Engine/Google Compute Engine, BIOS Google 01/01/2011 Workqueue: events_highpri snd_usbmidi_out_work Call Trace: <TASK> __dump_stack lib/dump_stack.c:88 [inline] dump_stack_lvl+0x1e3/0x2cb lib/dump_stack.c:106 print_address_description+0x65/0x4b0 mm/kasan/report.c:313 print_report+0xf4/0x210 mm/kasan/report.c:429 kasan_report+0xfb/0x130 mm/kasan/report.c:491 __snd_rawmidi_transmit_peek sound/core/rawmidi.c:1286 [inline] snd_rawmidi_transmit+0x3f3/0x420 sound/core/rawmidi.c:1411 snd_usbmidi_standard_output+0x202/0xf30 sound/usb/midi.c:650 snd_usbmidi_do_output+0x20e/0x530 sound/usb/midi.c:311 process_one_work+0x81c/0xd10 kernel/workqueue.c:2289 worker_thread+0xb14/0x1330 kernel/workqueue.c:2436 kthread+0x266/0x300 kernel/kthread.c:376 ret_from_fork+0x1f/0x30 </TASK> Allocated by task 3983: kasan_save_stack mm/kasan/common.c:38 [inline] kasan_set_track mm/kasan/common.c:45 [inline] set_alloc_info mm/kasan/common.c:436 [inline] ____kasan_kmalloc+0xdc/0x110 mm/kasan/common.c:515 kasan_kmalloc include/linux/kasan.h:234 [inline] __kmalloc_node+0x262/0x400 mm/slub.c:4465 kmalloc_node include/linux/slab.h:623 [inline] kvmalloc_node+0x6e/0x160 mm/util.c:613 kvmalloc include/linux/slab.h:750 [inline] kvzalloc include/linux/slab.h:758 [inline] snd_rawmidi_runtime_create sound/core/rawmidi.c:162 [inline] open_substream+0x2a1/0x6d0 sound/core/rawmidi.c:306 rawmidi_open_priv+0xc2/0x6a0 sound/core/rawmidi.c:357 snd_rawmidi_kernel_open+0x1e3/0x280 sound/core/rawmidi.c:392 midisynth_use+0xf9/0x2b0 sound/core/seq/seq_midi.c:215 subscribe_port sound/core/seq/seq_ports.c:412 [inline] check_and_subscribe_port+0x633/0xac0 sound/core/seq/seq_ports.c:495 snd_seq_port_connect+0x27e/0x480 sound/core/seq/seq_ports.c:581 snd_seq_ioctl_subscribe_port+0x394/0x750 sound/core/seq/seq_clientmgr.c:1492 snd_seq_oss_midi_open+0x44b/0x9b0 sound/core/seq/oss/seq_oss_midi.c:359 snd_seq_oss_synth_setup_midi+0x126/0x530 sound/core/seq/oss/seq_oss_synth.c:269 snd_seq_oss_open+0x96d/0x1020 sound/core/seq/oss/seq_oss_init.c:260 odev_open+0x5e/0x90 sound/core/seq/oss/seq_oss.c:128 chrdev_open+0x5fb/0x680 fs/char_dev.c:414 do_dentry_open+0x789/0x1040 fs/open.c:848 do_open fs/namei.c:3527 [inline] path_openat+0x26c0/0x2ec0 fs/namei.c:3660 do_filp_open+0x277/0x4f0 fs/namei.c:3687 do_sys_openat2+0x13b/0x500 fs/open.c:1278 do_sys_open fs/open.c:1294 [inline] __do_sys_openat fs/open.c:1310 [inline] __se_sys_openat fs/open.c:1305 [inline] __x64_sys_openat+0x243/0x290 fs/open.c:1305 do_syscall_x64 arch/x86/entry/common.c:50 [inline] do_syscall_64+0x2b/0x70 arch/x86/entry/common.c:80 entry_SYSCALL_64_after_hwframe+0x46/0xb0 Freed by task 3983: kasan_save_stack mm/kasan/common.c:38 [inline] kasan_set_track+0x4c/0x70 mm/kasan/common.c:45 kasan_set_free_info+0x1f/0x40 mm/kasan/generic.c:370 ____kasan_slab_free+0xd8/0x110 mm/kasan/common.c:366 kasan_slab_free include/linux/kasan.h:200 [inline] slab_free_hook mm/slub.c:1727 [inline] slab_free_freelist_hook+0x12e/0x1a0 mm/slub.c:1753 slab_free mm/slub.c:3507 [inline] kfree+0xc6/0x210 mm/slub.c:4555 snd_rawmidi_runtime_free sound/core/rawmidi.c:176 [inline] close_substream+0x38a/0x5f0 sound/core/rawmidi.c:528 rawmidi_release_priv+0xf1/0x180 sound/core/rawmidi.c:547 snd_rawmidi_kernel_release+0x42/0xa0 sound/core/rawmidi.c:564 unsubscribe_port sound/core/seq/seq_ports.c:437 [inline] __delete_and_unsubscribe_port+0x15c/0x330 sound/core/seq/seq_ports.c:537 snd_seq_port_disconnect+0x533/0x660 sound/core/seq/seq_ports.c:616 snd_seq_ioctl_unsubscribe_port+0x394/0x750 sound/core/seq/seq_clientmgr.c:1537 snd_seq_oss_midi_close+0x33f/0x740 sound/core/seq/oss/seq_oss_midi.c:404 snd_seq_oss_synth_reset+0x3b6/0x9d0 sound/core/seq/oss/seq_oss_synth.c:406 snd_seq_oss_reset+0x5b/0x240 sound/core/seq/oss/seq_oss_init.c:435 snd_seq_oss_release+0xe9/0x300 sound/core/seq/oss/seq_oss_init.c:412 odev_release+0x52/0x70 sound/core/seq/oss/seq_oss.c:144 __fput+0x3b9/0x820 fs/file_table.c:317 task_work_run+0x146/0x1c0 kernel/task_work.c:177 ptrace_notify+0x29f/0x350 kernel/signal.c:2372 ptrace_report_syscall include/linux/ptrace.h:427 [inline] ptrace_report_syscall_exit include/linux/ptrace.h:489 [inline] syscall_exit_work+0x8c/0xe0 kernel/entry/common.c:249 syscall_exit_to_user_mode_prepare+0x62/0x80 kernel/entry/common.c:276 __syscall_exit_to_user_mode_work kernel/entry/common.c:281 [inline] syscall_exit_to_user_mode+0xa/0x60 kernel/entry/common.c:294 entry_SYSCALL_64_after_hwframe+0x46/0xb0 The buggy address belongs to the object at ffff888074d3c000 which belongs to the cache kmalloc-4k of size 4096 The buggy address is located 127 bytes inside of 4096-byte region [ffff888074d3c000, ffff888074d3d000) The buggy address belongs to the physical page: page:ffffea0001d34e00 refcount:1 mapcount:0 mapping:0000000000000000 index:0x0 pfn:0x74d38 head:ffffea0001d34e00 order:3 compound_mapcount:0 compound_pincount:0 flags: 0xfff00000010200(slab|head|node=0|zone=1|lastcpupid=0x7ff) raw: 00fff00000010200 0000000000000000 dead000000000122 ffff888011442140 raw: 0000000000000000 0000000000040004 00000001ffffffff 0000000000000000 page dumped because: kasan: bad access detected page_owner tracks the page as allocated page last allocated via order 3, migratetype Unmovable, gfp_mask 0xd2040(__GFP_IO|__GFP_NOWARN|__GFP_NORETRY|__GFP_COMP|__GFP_NOMEMALLOC), pid 3983, tgid 3982 (syz-executor242), ts 217737880306, free_ts 217582307012 prep_new_page mm/page_alloc.c:2456 [inline] get_page_from_freelist+0x72b/0x7a0 mm/page_alloc.c:4198 __alloc_pages+0x259/0x560 mm/page_alloc.c:5426 alloc_slab_page+0x70/0xf0 mm/slub.c:1797 allocate_slab+0x5e/0x520 mm/slub.c:1942 new_slab mm/slub.c:2002 [inline] ___slab_alloc+0x41e/0xcd0 mm/slub.c:3002 __slab_alloc mm/slub.c:3089 [inline] slab_alloc_node mm/slub.c:3180 [inline] slab_alloc mm/slub.c:3222 [inline] __kmalloc+0x2ba/0x370 mm/slub.c:4413 kmalloc include/linux/slab.h:605 [inline] tomoyo_realpath_from_path+0xd8/0x5f0 security/tomoyo/realpath.c:254 tomoyo_get_realpath security/tomoyo/file.c:151 [inline] tomoyo_check_open_permission+0x22f/0x490 security/tomoyo/file.c:771 security_file_open+0x50/0x560 security/security.c:1645 do_dentry_open+0x34a/0x1040 fs/open.c:835 do_open fs/namei.c:3527 [inline] path_openat+0x26c0/0x2ec0 fs/namei.c:3660 do_filp_open+0x277/0x4f0 fs/namei.c:3687 do_sys_openat2+0x13b/0x500 fs/open.c:1278 do_sys_open fs/open.c:1294 [inline] __do_sys_openat fs/open.c:1310 [inline] __se_sys_openat fs/open.c:1305 [inline] __x64_sys_openat+0x243/0x290 fs/open.c:1305 do_syscall_x64 arch/x86/entry/common.c:50 [inline] do_syscall_64+0x2b/0x70 arch/x86/entry/common.c:80 entry_SYSCALL_64_after_hwframe+0x46/0xb0 page last free stack trace: reset_page_owner include/linux/page_owner.h:24 [inline] free_pages_prepare mm/page_alloc.c:1371 [inline] free_pcp_prepare+0x812/0x900 mm/page_alloc.c:1421 free_unref_page_prepare mm/page_alloc.c:3343 [inline] free_unref_page+0x7d/0x390 mm/page_alloc.c:3438 free_slab mm/slub.c:2041 [inline] discard_slab mm/slub.c:2047 [inline] __unfreeze_partials+0x1ab/0x200 mm/slub.c:2521 put_cpu_partial+0x116/0x180 mm/slub.c:2597 do_slab_free mm/slub.c:3495 [inline] ___cache_free+0x118/0x1a0 mm/slub.c:3514 qlist_free_all+0x2b/0x70 mm/kasan/quarantine.c:187 kasan_quarantine_reduce+0x169/0x180 mm/kasan/quarantine.c:294 __kasan_slab_alloc+0x2f/0xe0 mm/kasan/common.c:446 kasan_slab_alloc include/linux/kasan.h:224 [inline] slab_post_alloc_hook mm/slab.h:750 [inline] slab_alloc_node mm/slub.c:3214 [inline] __kmalloc_node+0x1f3/0x400 mm/slub.c:4461 kmalloc_node include/linux/slab.h:623 [inline] kvmalloc_node+0x6e/0x160 mm/util.c:613 kvmalloc include/linux/slab.h:750 [inline] seq_buf_alloc fs/seq_file.c:38 [inline] seq_read_iter+0x1f6/0xd30 fs/seq_file.c:210 call_read_iter include/linux/fs.h:2054 [inline] new_sync_read fs/read_write.c:401 [inline] vfs_read+0xa01/0xd10 fs/read_write.c:482 ksys_read+0x19b/0x2c0 fs/read_write.c:620 do_syscall_x64 arch/x86/entry/common.c:50 [inline] do_syscall_64+0x2b/0x70 arch/x86/entry/common.c:80 entry_SYSCALL_64_after_hwframe+0x46/0xb0 Memory state around the buggy address: ffff888074d3bf00: fc fc fc fc fc fc fc fc fc fc fc fc fc fc fc fc ffff888074d3bf80: fc fc fc fc fc fc fc fc fc fc fc fc fc fc fc fc >ffff888074d3c000: fa fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb ^ ffff888074d3c080: fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb ffff888074d3c100: fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb ==================================================================

1 0

[GIT PULL] ASoC fixes for v5.19-rc0
by Mark Brown 01 Jun '22

01 Jun '22

The following changes since commit e5cd20e0d6713138444cc3f3f982712cf9a36143: ASoC: SOF: Introduce generic (in)firmware tracing infrastructure (2022-05-20 14:56:35 +0100) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v5.19-rc0 for you to fetch changes up to ab0925ab728163b854ea17da09e299254a86076e: ASoC: da7219: Small fixes for jack detection and removal (2022-05-31 10:16:59 +0200) ---------------------------------------------------------------- ASoC: Fixes for v5.19 A few more fixes that came in during the merge window - nothing huge here, there is one core fix for DPCM from Pierre but mostly driver changes. ---------------------------------------------------------------- Adam Thomson (2): ASoC: da7219: Fix pole orientation detection on certain headsets ASoC: da7219: cancel AAD related work earlier for jack removal David Lin (1): ASoC: Intel: common: fix typo for tplg naming Mark Brown (1): ASoC: da7219: Small fixes for jack detection and removal Oder Chiou (1): ASoC: rt5640: Do not manipulate pin "Platform Clock" if the "Platform Clock" is not in the DAPM Pierre-Louis Bossart (1): ASoC: soc-pcm: fix BE transition for TRIGGER_START Shengjiu Wang (1): ASoC: fsl_sai: Fix FSL_SAI_xDR/xFR definition Syed Saba kareem (1): ASoC: SOF: amd: Fixed Build error Uwe Kleine-König (1): ASoC: Intel: avs: Fix build error on arc, m68k and sparc sound/soc/amd/acp/acp-pci.c | 1 + sound/soc/codecs/da7219-aad.c | 18 +++++++++++++++--- sound/soc/codecs/rt5640.c | 11 +++++++++-- sound/soc/codecs/rt5640.h | 2 ++ sound/soc/fsl/fsl_sai.h | 4 ++-- sound/soc/intel/avs/board_selection.c | 3 ++- sound/soc/intel/boards/bytcr_rt5640.c | 2 ++ sound/soc/intel/common/soc-acpi-intel-adl-match.c | 2 +- sound/soc/soc-pcm.c | 2 -- 9 files changed, 34 insertions(+), 11 deletions(-)

2 1

[PATCH 0/5] seq: add interface for queue timer
by GitHub pull_request - opened 01 Jun '22

01 Jun '22

alsa-project/alsa-gobject pull request #72 was opened from takaswie: Current implementation uses function of QueueTimer to retrieve data for variants of queue timer, however it's not enough convenient in the case of language bindings. The usage of GObject.Value in object property is expected to be supported widely instead. This patchset uses object classes derived from GObject to express the variants. The common properties are expressed by QueueTimerCommon interface. The ALSATimer variant is currently supported to implement the interface. ``` Takashi Sakamoto (5): doc: seq: fix URL to document for ALSATimer seq: queue-timer-common: add common interface for queue timer seq: queue-timer-alsa: add class for queue timer specialized to ALSA Timer seq: user-client: use QueueTimerCommon interface instead of QueueTimer seq: queue-timer/queue-timer-data-alsa: remove unused gobject classes doc/seq/alsaseq.toml.in | 2 +- doc/seq/urlmap.js | 2 +- src/seq/alsaseq.h | 5 +- src/seq/alsaseq.map | 16 +- src/seq/meson.build | 8 +- src/seq/privates.h | 21 ++- src/seq/queue-timer-alsa.c | 134 +++++++++++++++++ src/seq/queue-timer-alsa.h | 22 +++ src/seq/queue-timer-common.c | 87 +++++++++++ src/seq/queue-timer-common.h | 20 +++ src/seq/queue-timer-data-alsa.c | 76 ---------- src/seq/queue-timer-data-alsa.h | 32 ---- src/seq/queue-timer.c | 138 ------------------ src/seq/queue-timer.h | 26 ---- src/seq/user-client.c | 42 +++--- src/seq/user-client.h | 4 +- ...q-queue-timer => alsaseq-queue-timer-alsa} | 8 +- tests/meson.build | 2 +- 18 files changed, 325 insertions(+), 320 deletions(-) create mode 100644 src/seq/queue-timer-alsa.c create mode 100644 src/seq/queue-timer-alsa.h create mode 100644 src/seq/queue-timer-common.c create mode 100644 src/seq/queue-timer-common.h delete mode 100644 src/seq/queue-timer-data-alsa.c delete mode 100644 src/seq/queue-timer-data-alsa.h delete mode 100644 src/seq/queue-timer.c delete mode 100644 src/seq/queue-timer.h rename tests/{alsaseq-queue-timer => alsaseq-queue-timer-alsa} (76%) ``` Request URL : https://github.com/alsa-project/alsa-gobject/pull/72 Patch URL : https://github.com/alsa-project/alsa-gobject/pull/72.patch Repository URL: https://github.com/alsa-project/alsa-gobject

1 0

[PATCH 00/13] ctl: add interfaces for element information
by GitHub pull_request - opened 01 Jun '22

01 Jun '22

alsa-project/alsa-gobject pull request #71 was opened from takaswie: Current implementation uses function of ElemInfo to express variants of element information, however it's not enough convenient in the case of language bindings. The usage of GObject.Value in object property is expected to be supported widely instead. This patchset uses object classes derived from GObject to express the variants. The common properties are expressed by two interfaces; ElemInfoCommon and ElemInfoSingleArray. All of classes implement the former, and the latter is implemented except for IEC60958 element information. ``` Takashi Sakamoto (13): ctl: card: fix indentation fix missing usage of declaration macro ctl: card: add error path when generating name list at element addition ctl: elem-info-common: add common interface for element information ctl: elem-info-iec60958: add class for element information of IEC60958 type ctl: elem-info-single-array: add interface for single array element information ctl: elem-info-boolean: add class for element information of boolean type ctl: elem-info-bytes: add class for element information of bytes type ctl: elem-info-integer: add class for element information of integer type ctl: elem-info-integer64: add class for element information of integer64 type ctl: elem-info-enumerated: add class for element information of enumerated type ctl: card: use ALSACtl.ElemInfoCommon instead of ElemInfo ctl: elem-info: remove unused classes samples/ctl | 56 +-- src/ctl/alsactl-enum-types.h | 4 + src/ctl/alsactl.h | 10 +- src/ctl/alsactl.map | 30 +- src/ctl/card.c | 156 ++++--- src/ctl/card.h | 6 +- src/ctl/elem-info-boolean.c | 90 +++++ src/ctl/elem-info-boolean.h | 22 + src/ctl/elem-info-bytes.c | 89 ++++ src/ctl/elem-info-bytes.h | 21 + src/ctl/elem-info-common.c | 135 +++++++ src/ctl/elem-info-common.h | 20 + src/ctl/elem-info-enumerated.c | 137 +++++++ src/ctl/elem-info-enumerated.h | 22 + src/ctl/elem-info-iec60958.c | 80 ++++ src/ctl/elem-info-iec60958.h | 22 + src/ctl/elem-info-integer.c | 162 ++++++++ src/ctl/elem-info-integer.h | 22 + src/ctl/elem-info-integer64.c | 162 ++++++++ src/ctl/elem-info-integer64.h | 22 + src/ctl/elem-info-single-array.c | 68 ++++ src/ctl/elem-info-single-array.h | 20 + src/ctl/elem-info.c | 379 ------------------ src/ctl/elem-info.h | 30 -- src/ctl/meson.build | 18 +- src/ctl/privates.h | 54 ++- src/hwdep/alsahwdep-enum-types.h | 4 + src/rawmidi/alsarawmidi-enum-types.h | 4 + src/seq/alsaseq-enum-types.h | 4 + src/timer/alsatimer-enum-types.h | 4 + ...tl-elem-info => alsactl-elem-info-boolean} | 10 +- tests/alsactl-elem-info-bytes | 26 ++ tests/alsactl-elem-info-enumerated | 27 ++ tests/alsactl-elem-info-iec60958 | 25 ++ tests/alsactl-elem-info-integer | 29 ++ tests/alsactl-elem-info-integer64 | 29 ++ tests/meson.build | 7 +- 37 files changed, 1492 insertions(+), 514 deletions(-) create mode 100644 src/ctl/elem-info-boolean.c create mode 100644 src/ctl/elem-info-boolean.h create mode 100644 src/ctl/elem-info-bytes.c create mode 100644 src/ctl/elem-info-bytes.h create mode 100644 src/ctl/elem-info-common.c create mode 100644 src/ctl/elem-info-common.h create mode 100644 src/ctl/elem-info-enumerated.c create mode 100644 src/ctl/elem-info-enumerated.h create mode 100644 src/ctl/elem-info-iec60958.c create mode 100644 src/ctl/elem-info-iec60958.h create mode 100644 src/ctl/elem-info-integer.c create mode 100644 src/ctl/elem-info-integer.h create mode 100644 src/ctl/elem-info-integer64.c create mode 100644 src/ctl/elem-info-integer64.h create mode 100644 src/ctl/elem-info-single-array.c create mode 100644 src/ctl/elem-info-single-array.h delete mode 100644 src/ctl/elem-info.c delete mode 100644 src/ctl/elem-info.h rename tests/{alsactl-elem-info => alsactl-elem-info-boolean} (68%) create mode 100644 tests/alsactl-elem-info-bytes create mode 100644 tests/alsactl-elem-info-enumerated create mode 100644 tests/alsactl-elem-info-iec60958 create mode 100644 tests/alsactl-elem-info-integer create mode 100644 tests/alsactl-elem-info-integer64 ``` Request URL : https://github.com/alsa-project/alsa-gobject/pull/71 Patch URL : https://github.com/alsa-project/alsa-gobject/pull/71.patch Repository URL: https://github.com/alsa-project/alsa-gobject

1 0

ucm2: rk3399-gru-sound: Add missing symlink from conf.d tree
by GitHub pull_request - opened 31 May '22

31 May '22

alsa-project/alsa-ucm-conf pull request #162 was opened from alpernebbi: Since V2Module lookup was disabled my rk3399-gru-kevin chromebook can't find the UCM it's supposed to use. This adds a `conf.d/rk3399-gru-soun/rk3399-gru-soun.conf` symlink for it. The name is not a typo: $ cat /proc/asound/cards 0 [rk3399grusound ]: rk3399-gru-soun - rk3399-gru-sound Unknown-UnknownProduct- $ strace -e trace=file pulseaudio 2>&1 | grep -i ucm2 faccessat(AT_FDCWD, "/usr/share/alsa/ucm2/ucm.conf", R_OK) = 0 openat(AT_FDCWD, "/usr/share/alsa/ucm2/ucm.conf", O_RDONLY) = 16 openat(AT_FDCWD, "/usr/share/alsa/ucm2/lib/generic.conf", O_RDONLY) = 16 faccessat(AT_FDCWD, "/usr/share/alsa/ucm2/conf.d/rk3399-gru-soun/Unknown-UnknownProduct-.conf", R_OK) = -1 ENOENT (No such file or directory) faccessat(AT_FDCWD, "/usr/share/alsa/ucm2/conf.d/rk3399-gru-soun/rk3399-gru-soun.conf", R_OK) = -1 ENOENT (No such file or directory) faccessat(AT_FDCWD, "/usr/share/alsa/ucm2/ucm.conf", R_OK) = 0 openat(AT_FDCWD, "/usr/share/alsa/ucm2/ucm.conf", O_RDONLY) = 16 openat(AT_FDCWD, "/usr/share/alsa/ucm2/lib/generic.conf", O_RDONLY) = 16 faccessat(AT_FDCWD, "/usr/share/alsa/ucm2/conf.d/rk3399-gru-soun/Unknown-UnknownProduct-.conf", R_OK) = -1 ENOENT (No such file or directory) faccessat(AT_FDCWD, "/usr/share/alsa/ucm2/conf.d/rk3399-gru-soun/rk3399-gru-soun.conf", R_OK) = -1 ENOENT (No such file or directory) (Ignore the `Unknown-UnknownProduct-`, that's because I'm using a custom unfinished U-Boot build. For more info there's an [older alsa-info output](https://alsa-project.org/db/?f=9b4102975fa9c0f109af6ad687c856c620d7… I had uploaded) Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/162 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/162.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

ALSA 1.2.7 release
by Jaroslav Kysela 31 May '22

31 May '22

Hello all, new ALSA userspace packages were released. You may download them from the ALSA website http://www.alsa-project.org or directly: HTTP: https://www.alsa-project.org/files/pub FTP: ftp://ftp.alsa-project.org/pub Released packages: alsa-lib alsa-utils alsa-ucm-conf pyalsa Full list of changes: https://www.alsa-project.org/wiki/Changes_v1.2.6.3_v1.2.7 The fingerprint of the public signing key is: F04D F507 37AC 1A88 4C4B 3D71 8380 596D A6E5 9C91 Have fun, Jaroslav -- Jaroslav Kysela <perex(a)perex.cz> Linux Sound Maintainer; ALSA Project; Red Hat, Inc.

1 0

[PATCH] selftests: alsa: Handle pkg-config failure more gracefully
by Mark Brown 31 May '22

31 May '22

Follow the pattern used by other selftests like memfd and fall back on the standard toolchain options to build with a system installed alsa-lib if we don't get anything from pkg-config. This reduces our build dependencies a bit in the common case while still allowing use of pkg-config in case there is a need for it. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- tools/testing/selftests/alsa/Makefile | 3 +++ 1 file changed, 3 insertions(+) diff --git a/tools/testing/selftests/alsa/Makefile b/tools/testing/selftests/alsa/Makefile index f64d9090426d..fd8ddce2b1a6 100644 --- a/tools/testing/selftests/alsa/Makefile +++ b/tools/testing/selftests/alsa/Makefile @@ -3,6 +3,9 @@ CFLAGS += $(shell pkg-config --cflags alsa) LDLIBS += $(shell pkg-config --libs alsa) +ifeq ($(LDLIBS),) +LDLIBS += -lasound +endif TEST_GEN_PROGS := mixer-test -- 2.30.2

2 1

[PATCH] ALSA: usb-audio: Optimize TEAC clock quirk
by Takashi Iwai 31 May '22

31 May '22

Maris found out that the quirk for TEAC devices to work around the clock setup is needed to apply only when the base clock is changed, e.g. from 48000-based clocks (48000, 96000, 192000, 384000) to 44100-based clocks (44100, 88200, 176400, 352800), or vice versa, while switching to another clock with the same base clock doesn't need the (forcible) interface setup. This patch implements the optimization for the TEAC clock quirk to avoid the unnecessary interface re-setup. Fixes: 5ce0b06ae5e6 ("ALSA: usb-audio: Workaround for clock setup on TEAC devices") Reported-by: Maris Abele <maris7abele(a)gmail.com> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/clock.c | 12 ++++++++---- 1 file changed, 8 insertions(+), 4 deletions(-) diff --git a/sound/usb/clock.c b/sound/usb/clock.c index 3c435d379306..db429320599e 100644 --- a/sound/usb/clock.c +++ b/sound/usb/clock.c @@ -573,10 +573,14 @@ static int set_sample_rate_v2v3(struct snd_usb_audio *chip, } /* FIXME - TEAC devices require the immediate interface setup */ - if (rate != prev_rate && USB_ID_VENDOR(chip->usb_id) == 0x0644) { - usb_set_interface(chip->dev, fmt->iface, fmt->altsetting); - if (chip->quirk_flags & QUIRK_FLAG_IFACE_DELAY) - msleep(50); + if (USB_ID_VENDOR(chip->usb_id) == 0x0644) { + bool cur_base_48k = (rate % 48000 == 0); + bool prev_base_48k = (prev_rate % 48000 == 0); + if (cur_base_48k != prev_base_48k) { + usb_set_interface(chip->dev, fmt->iface, fmt->altsetting); + if (chip->quirk_flags & QUIRK_FLAG_IFACE_DELAY) + msleep(50); + } } validation: -- 2.35.3

1 0

[PATCH v3 00/15] ASoC: Intel: avs: Driver core and PCM operations
by Cezary Rojewski 31 May '22

31 May '22

Part three of main AVS driver series. This series was originally part of the initial series which was later divided [1] into smaller, easier-to-review chunks. Thus, many patches found here were already present on the list. This series consists of code typical to many drivers - PCI driver operations, trace ability, PM operations - as well as PCM handlers for all standard audio interfaces, that is, HDA, I2S and DMIC are found here. Series starts with updating firmware boot flow - libraries are no longer ignored. This change is dependent on already merged topology code [2] and because of that could not be part of the initial series [1]. PCM operations are split into four changes. First component operations alone i.e. operations which are usually agnostic towards path position (FE/BE). Then it continues with "generic" FE operations - there is no interface split here as from Intel ADSP point of view, FE, or HOST side as it's called in the specs, involves HD-Audio operations only. BE (also known as LINK) side on the other hand is divided into "non-HD-Audio" and HD-Audio part. The former represents transfer over DMIC and I2S interfaces both. While patches implementing standard PCI driver operations along (again standard) HD-Audio initialization routines followed up by power management handlers are two major ones, series covers also other important subjects such as: While patches implementing standard PCI driver operations along (again standard) HD-Audio initialization routines followed up by power management handlers are two major ones, series covers also other important subjects such as: - event tracing - preparation for firmware tracing (debugability) - coredump (debugability) - recovery flow (attempt recovery after IPC timeout or exception) - D0ix (D0 device substate, complements standard power management) Series is finalized by actual addition of supported platforms: SKL and APL-based. Platform-specific files are limited to firmware-specific bits, that is, bits that are specific to given firmware generation. Everything else is shared and is part of already upstream messaging code found in ipc.c, messages.c and messages.h files. Changes in v3: - addressed (hopefully) trace-code compilation under .configs with CONFIG_FTRACE dropped Changes in v2: - usage of avs_releast_last_firmware() dropped in error path for library loading procedure as suggested by Pierre - 'link_mask' usage replaced with 'i2s_link_mask' as requested by Pierre. Existing code addressed with new patch: "ASoC: Intel: avs: Replace link_mask usage with i2s_link_mask" - fixed possible race during recovery flow (->recovering flag is now atomic and tested in single location only). - dropped platform prefixes for basically all i2s board descriptors - 'ssp_test' renamed to 'i2s_test' to match naming convention of other boards - simplified PM implementation for current series, 'low_power' bits moved to future series - replaced SND_INTEL_DSP_DRIVER_SST with _AVS as suggested by Mark. Required changes to intel-dspcfg will be added in future series - number of typos across commit messages addressed [1]: https://lore.kernel.org/all/20220311153544.136854-1-cezary.rojewski@intel.c… [2]: https://lore.kernel.org/all/20220331135246.993089-1-cezary.rojewski@intel.c… Cezary Rojewski (14): ASoC: Intel: avs: Account for libraries when booting basefw ASoC: Intel: avs: Generic soc component driver ASoC: Intel: avs: Generic PCM FE operations ASoC: Intel: avs: non-HDA PCM BE operations ASoC: Intel: avs: HDA PCM BE operations ASoC: Intel: avs: Coredump and recovery flow ASoC: Intel: avs: Prepare for firmware tracing ASoC: Intel: avs: D0ix power state support ASoC: Intel: avs: Event tracing ASoC: Intel: avs: Machine board registration ASoC: Intel: avs: PCI driver implementation ASoC: Intel: avs: Power management ASoC: Intel: avs: SKL-based platforms support ASoC: Intel: avs: APL-based platforms support Piotr Maziarz (1): ASoC: Intel: avs: Replace link_mask usage with i2s_link_mask include/sound/intel-dsp-config.h | 3 +- include/sound/soc-acpi.h | 2 + sound/soc/intel/Kconfig | 2 + sound/soc/intel/avs/Makefile | 7 +- sound/soc/intel/avs/apl.c | 250 ++++++ sound/soc/intel/avs/avs.h | 79 ++ sound/soc/intel/avs/board_selection.c | 501 +++++++++++ sound/soc/intel/avs/core.c | 631 +++++++++++++ sound/soc/intel/avs/dsp.c | 27 +- sound/soc/intel/avs/ipc.c | 253 +++++- sound/soc/intel/avs/loader.c | 84 ++ sound/soc/intel/avs/messages.c | 35 +- sound/soc/intel/avs/messages.h | 51 ++ sound/soc/intel/avs/pcm.c | 1182 +++++++++++++++++++++++++ sound/soc/intel/avs/registers.h | 8 + sound/soc/intel/avs/skl.c | 125 +++ sound/soc/intel/avs/topology.c | 14 +- sound/soc/intel/avs/trace.c | 33 + sound/soc/intel/avs/trace.h | 154 ++++ sound/soc/intel/avs/utils.c | 23 + 20 files changed, 3443 insertions(+), 21 deletions(-) create mode 100644 sound/soc/intel/avs/apl.c create mode 100644 sound/soc/intel/avs/board_selection.c create mode 100644 sound/soc/intel/avs/pcm.c create mode 100644 sound/soc/intel/avs/skl.c create mode 100644 sound/soc/intel/avs/trace.c create mode 100644 sound/soc/intel/avs/trace.h -- 2.25.1

4 31

[PATCH 0/2] ASoC: da7219: Small fixes for jack detection and removal
by Adam Thomson 31 May '22

31 May '22

This series contains 2 small fixes around the AAD part of DA7219, particularly in relation to jack pole detection on certain active headsets, and tidy up when a jack is removed. Adam Thomson (2): ASoC: da7219: Fix pole orientation detection on certain headsets ASoC: da7219: cancel AAD related work earlier for jack removal sound/soc/codecs/da7219-aad.c | 18 +++++++++++++++--- 1 file changed, 15 insertions(+), 3 deletions(-) -- 2.17.1

2 3

[PATCH] ASoC: da7219: Drop no-op remove function
by Uwe Kleine-König 30 May '22

30 May '22

A remove callback that just returns 0 is equivalent to no callback at all as can be seen in i2c_device_remove(). So simplify accordingly. Signed-off-by: Uwe Kleine-König <u.kleine-koenig(a)pengutronix.de> --- sound/soc/codecs/da7219.c | 6 ------ 1 file changed, 6 deletions(-) diff --git a/sound/soc/codecs/da7219.c b/sound/soc/codecs/da7219.c index c7493549a9a5..7df18a2c36b9 100644 --- a/sound/soc/codecs/da7219.c +++ b/sound/soc/codecs/da7219.c @@ -2694,11 +2694,6 @@ static int da7219_i2c_probe(struct i2c_client *i2c, return ret; } -static int da7219_i2c_remove(struct i2c_client *client) -{ - return 0; -} - static const struct i2c_device_id da7219_i2c_id[] = { { "da7219", }, { } @@ -2712,7 +2707,6 @@ static struct i2c_driver da7219_i2c_driver = { .acpi_match_table = ACPI_PTR(da7219_acpi_match), }, .probe = da7219_i2c_probe, - .remove = da7219_i2c_remove, .id_table = da7219_i2c_id, }; base-commit: 4b0986a3613c92f4ec1bdc7f60ec66fea135991f -- 2.36.1

3 4

[PATCH AUTOSEL 5.18 089/159] ASoC: tscs454: Add endianness flag in snd_soc_component_driver
by Sasha Levin 30 May '22

30 May '22

From: Charles Keepax <ckeepax(a)opensource.cirrus.com> [ Upstream commit ff69ec96b87dccb3a29edef8cec5d4fefbbc2055 ] The endianness flag is used on the CODEC side to specify an ambivalence to endian, typically because it is lost over the hardware link. This device receives audio over an I2S DAI and as such should have endianness applied. A fixup is also required to use the width directly rather than relying on the format in hw_params, now both little and big endian would be supported. It is worth noting this changes the behaviour of S24_LE to use a word length of 24 rather than 32. This would appear to be a correction since the fact S24_LE is stored as 32 bits should not be presented over the bus. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220504170905.332415-26-ckeepax@opensource.cirru… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/tscs454.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/tscs454.c b/sound/soc/codecs/tscs454.c index 7e1826d6f06f..32e6fa7b0a06 100644 --- a/sound/soc/codecs/tscs454.c +++ b/sound/soc/codecs/tscs454.c @@ -3120,18 +3120,17 @@ static int set_aif_sample_format(struct snd_soc_component *component, unsigned int width; int ret; - switch (format) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (snd_pcm_format_width(format)) { + case 16: width = FV_WL_16; break; - case SNDRV_PCM_FORMAT_S20_3LE: + case 20: width = FV_WL_20; break; - case SNDRV_PCM_FORMAT_S24_3LE: + case 24: width = FV_WL_24; break; - case SNDRV_PCM_FORMAT_S24_LE: - case SNDRV_PCM_FORMAT_S32_LE: + case 32: width = FV_WL_32; break; default: @@ -3326,6 +3325,7 @@ static const struct snd_soc_component_driver soc_component_dev_tscs454 = { .num_dapm_routes = ARRAY_SIZE(tscs454_intercon), .controls = tscs454_snd_controls, .num_controls = ARRAY_SIZE(tscs454_snd_controls), + .endianness = 1, }; #define TSCS454_RATES SNDRV_PCM_RATE_8000_96000 -- 2.35.1

3 2

[PATCH AUTOSEL 4.9 20/24] ASoC: rt5645: Fix errorenous cleanup order
by Sasha Levin 30 May '22

30 May '22

From: Lin Ma <linma(a)zju.edu.cn> [ Upstream commit 2def44d3aec59e38d2701c568d65540783f90f2f ] There is a logic error when removing rt5645 device as the function rt5645_i2c_remove() first cancel the &rt5645->jack_detect_work and delete the &rt5645->btn_check_timer latter. However, since the timer handler rt5645_btn_check_callback() will re-queue the jack_detect_work, this cleanup order is buggy. That is, once the del_timer_sync in rt5645_i2c_remove is concurrently run with the rt5645_btn_check_callback, the canceled jack_detect_work will be rescheduled again, leading to possible use-after-free. This patch fix the issue by placing the del_timer_sync function before the cancel_delayed_work_sync. Signed-off-by: Lin Ma <linma(a)zju.edu.cn> Link: https://lore.kernel.org/r/20220516092035.28283-1-linma@zju.edu.cn Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt5645.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 1ac96ef9ee20..703b26ec4e15 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3878,6 +3878,12 @@ static int rt5645_i2c_remove(struct i2c_client *i2c) if (i2c->irq) free_irq(i2c->irq, rt5645); + /* + * Since the rt5645_btn_check_callback() can queue jack_detect_work, + * the timer need to be delted first + */ + del_timer_sync(&rt5645->btn_check_timer); + cancel_delayed_work_sync(&rt5645->jack_detect_work); cancel_delayed_work_sync(&rt5645->rcclock_work); -- 2.35.1

1 0

[PATCH AUTOSEL 4.9 14/24] ASoC: dapm: Don't fold register value changes into notifications
by Sasha Levin 30 May '22

30 May '22

From: Mark Brown <broonie(a)kernel.org> [ Upstream commit ad685980469b9f9b99d4d6ea05f4cb8f57cb2234 ] DAPM tracks and reports the value presented to the user from DAPM controls separately to the register value, these may diverge during initialisation or when an autodisable control is in use. When writing DAPM controls we currently report that a change has occurred if either the DAPM value or the value stored in the register has changed, meaning that if the two are out of sync we may appear to report a spurious event to userspace. Since we use this folded in value for nothing other than the value reported to userspace simply drop the folding in of the register change. Signed-off-by: Mark Brown <broonie(a)kernel.org> Link: https://lore.kernel.org/r/20220428161833.3690050-1-broonie@kernel.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/soc-dapm.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 878a4fc97f04..40bf50cd87bc 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3165,7 +3165,6 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, update.val = val; card->update = &update; } - change |= reg_change; ret = soc_dapm_mixer_update_power(card, kcontrol, connect); @@ -3270,7 +3269,6 @@ int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol, update.val = val; card->update = &update; } - change |= reg_change; ret = soc_dapm_mux_update_power(card, kcontrol, item[0], e); -- 2.35.1

1 0

[PATCH AUTOSEL 4.9 07/24] ALSA: jack: Access input_dev under mutex
by Sasha Levin 30 May '22

30 May '22

From: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> [ Upstream commit 1b6a6fc5280e97559287b61eade2d4b363e836f2 ] It is possible when using ASoC that input_dev is unregistered while calling snd_jack_report, which causes NULL pointer dereference. In order to prevent this serialize access to input_dev using mutex lock. Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> Reviewed-by: Cezary Rojewski <cezary.rojewski(a)intel.com> Link: https://lore.kernel.org/r/20220412091628.3056922-1-amadeuszx.slawinski@linu… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- include/sound/jack.h | 1 + sound/core/jack.c | 34 +++++++++++++++++++++++++++------- 2 files changed, 28 insertions(+), 7 deletions(-) diff --git a/include/sound/jack.h b/include/sound/jack.h index 1e84bfb553cf..4742f842b457 100644 --- a/include/sound/jack.h +++ b/include/sound/jack.h @@ -77,6 +77,7 @@ struct snd_jack { const char *id; #ifdef CONFIG_SND_JACK_INPUT_DEV struct input_dev *input_dev; + struct mutex input_dev_lock; int registered; int type; char name[100]; diff --git a/sound/core/jack.c b/sound/core/jack.c index 36cfe1c54109..d2f9a92453f2 100644 --- a/sound/core/jack.c +++ b/sound/core/jack.c @@ -48,8 +48,11 @@ static int snd_jack_dev_disconnect(struct snd_device *device) #ifdef CONFIG_SND_JACK_INPUT_DEV struct snd_jack *jack = device->device_data; - if (!jack->input_dev) + mutex_lock(&jack->input_dev_lock); + if (!jack->input_dev) { + mutex_unlock(&jack->input_dev_lock); return 0; + } /* If the input device is registered with the input subsystem * then we need to use a different deallocator. */ @@ -58,6 +61,7 @@ static int snd_jack_dev_disconnect(struct snd_device *device) else input_free_device(jack->input_dev); jack->input_dev = NULL; + mutex_unlock(&jack->input_dev_lock); #endif /* CONFIG_SND_JACK_INPUT_DEV */ return 0; } @@ -96,8 +100,11 @@ static int snd_jack_dev_register(struct snd_device *device) snprintf(jack->name, sizeof(jack->name), "%s %s", card->shortname, jack->id); - if (!jack->input_dev) + mutex_lock(&jack->input_dev_lock); + if (!jack->input_dev) { + mutex_unlock(&jack->input_dev_lock); return 0; + } jack->input_dev->name = jack->name; @@ -122,6 +129,7 @@ static int snd_jack_dev_register(struct snd_device *device) if (err == 0) jack->registered = 1; + mutex_unlock(&jack->input_dev_lock); return err; } #endif /* CONFIG_SND_JACK_INPUT_DEV */ @@ -242,9 +250,11 @@ int snd_jack_new(struct snd_card *card, const char *id, int type, return -ENOMEM; } - /* don't creat input device for phantom jack */ - if (!phantom_jack) { #ifdef CONFIG_SND_JACK_INPUT_DEV + mutex_init(&jack->input_dev_lock); + + /* don't create input device for phantom jack */ + if (!phantom_jack) { int i; jack->input_dev = input_allocate_device(); @@ -262,8 +272,8 @@ int snd_jack_new(struct snd_card *card, const char *id, int type, input_set_capability(jack->input_dev, EV_SW, jack_switch_types[i]); -#endif /* CONFIG_SND_JACK_INPUT_DEV */ } +#endif /* CONFIG_SND_JACK_INPUT_DEV */ err = snd_device_new(card, SNDRV_DEV_JACK, jack, &ops); if (err < 0) @@ -303,10 +313,14 @@ EXPORT_SYMBOL(snd_jack_new); void snd_jack_set_parent(struct snd_jack *jack, struct device *parent) { WARN_ON(jack->registered); - if (!jack->input_dev) + mutex_lock(&jack->input_dev_lock); + if (!jack->input_dev) { + mutex_unlock(&jack->input_dev_lock); return; + } jack->input_dev->dev.parent = parent; + mutex_unlock(&jack->input_dev_lock); } EXPORT_SYMBOL(snd_jack_set_parent); @@ -354,6 +368,8 @@ EXPORT_SYMBOL(snd_jack_set_key); /** * snd_jack_report - Report the current status of a jack + * Note: This function uses mutexes and should be called from a + * context which can sleep (such as a workqueue). * * @jack: The jack to report status for * @status: The current status of the jack @@ -373,8 +389,11 @@ void snd_jack_report(struct snd_jack *jack, int status) status & jack_kctl->mask_bits); #ifdef CONFIG_SND_JACK_INPUT_DEV - if (!jack->input_dev) + mutex_lock(&jack->input_dev_lock); + if (!jack->input_dev) { + mutex_unlock(&jack->input_dev_lock); return; + } for (i = 0; i < ARRAY_SIZE(jack->key); i++) { int testbit = SND_JACK_BTN_0 >> i; @@ -393,6 +412,7 @@ void snd_jack_report(struct snd_jack *jack, int status) } input_sync(jack->input_dev); + mutex_unlock(&jack->input_dev_lock); #endif /* CONFIG_SND_JACK_INPUT_DEV */ } EXPORT_SYMBOL(snd_jack_report); -- 2.35.1

1 0

[PATCH AUTOSEL 4.14 24/29] ASoC: rt5645: Fix errorenous cleanup order
by Sasha Levin 30 May '22

30 May '22

From: Lin Ma <linma(a)zju.edu.cn> [ Upstream commit 2def44d3aec59e38d2701c568d65540783f90f2f ] There is a logic error when removing rt5645 device as the function rt5645_i2c_remove() first cancel the &rt5645->jack_detect_work and delete the &rt5645->btn_check_timer latter. However, since the timer handler rt5645_btn_check_callback() will re-queue the jack_detect_work, this cleanup order is buggy. That is, once the del_timer_sync in rt5645_i2c_remove is concurrently run with the rt5645_btn_check_callback, the canceled jack_detect_work will be rescheduled again, leading to possible use-after-free. This patch fix the issue by placing the del_timer_sync function before the cancel_delayed_work_sync. Signed-off-by: Lin Ma <linma(a)zju.edu.cn> Link: https://lore.kernel.org/r/20220516092035.28283-1-linma@zju.edu.cn Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt5645.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index a98647ac497c..01de25813c72 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3972,9 +3972,14 @@ static int rt5645_i2c_remove(struct i2c_client *i2c) if (i2c->irq) free_irq(i2c->irq, rt5645); + /* + * Since the rt5645_btn_check_callback() can queue jack_detect_work, + * the timer need to be delted first + */ + del_timer_sync(&rt5645->btn_check_timer); + cancel_delayed_work_sync(&rt5645->jack_detect_work); cancel_delayed_work_sync(&rt5645->rcclock_work); - del_timer_sync(&rt5645->btn_check_timer); snd_soc_unregister_codec(&i2c->dev); regulator_bulk_disable(ARRAY_SIZE(rt5645->supplies), rt5645->supplies); -- 2.35.1

1 0

[PATCH AUTOSEL 4.14 16/29] ASoC: dapm: Don't fold register value changes into notifications
by Sasha Levin 30 May '22

30 May '22

From: Mark Brown <broonie(a)kernel.org> [ Upstream commit ad685980469b9f9b99d4d6ea05f4cb8f57cb2234 ] DAPM tracks and reports the value presented to the user from DAPM controls separately to the register value, these may diverge during initialisation or when an autodisable control is in use. When writing DAPM controls we currently report that a change has occurred if either the DAPM value or the value stored in the register has changed, meaning that if the two are out of sync we may appear to report a spurious event to userspace. Since we use this folded in value for nothing other than the value reported to userspace simply drop the folding in of the register change. Signed-off-by: Mark Brown <broonie(a)kernel.org> Link: https://lore.kernel.org/r/20220428161833.3690050-1-broonie@kernel.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/soc-dapm.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index dd3053c243c1..320d262c16c9 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3282,7 +3282,6 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, update.val = val; card->update = &update; } - change |= reg_change; ret = soc_dapm_mixer_update_power(card, kcontrol, connect, rconnect); @@ -3388,7 +3387,6 @@ int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol, update.val = val; card->update = &update; } - change |= reg_change; ret = soc_dapm_mux_update_power(card, kcontrol, item[0], e); -- 2.35.1

1 0

[PATCH AUTOSEL 4.14 07/29] ALSA: jack: Access input_dev under mutex
by Sasha Levin 30 May '22

30 May '22

From: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> [ Upstream commit 1b6a6fc5280e97559287b61eade2d4b363e836f2 ] It is possible when using ASoC that input_dev is unregistered while calling snd_jack_report, which causes NULL pointer dereference. In order to prevent this serialize access to input_dev using mutex lock. Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> Reviewed-by: Cezary Rojewski <cezary.rojewski(a)intel.com> Link: https://lore.kernel.org/r/20220412091628.3056922-1-amadeuszx.slawinski@linu… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- include/sound/jack.h | 1 + sound/core/jack.c | 34 +++++++++++++++++++++++++++------- 2 files changed, 28 insertions(+), 7 deletions(-) diff --git a/include/sound/jack.h b/include/sound/jack.h index 1e84bfb553cf..4742f842b457 100644 --- a/include/sound/jack.h +++ b/include/sound/jack.h @@ -77,6 +77,7 @@ struct snd_jack { const char *id; #ifdef CONFIG_SND_JACK_INPUT_DEV struct input_dev *input_dev; + struct mutex input_dev_lock; int registered; int type; char name[100]; diff --git a/sound/core/jack.c b/sound/core/jack.c index 36cfe1c54109..d2f9a92453f2 100644 --- a/sound/core/jack.c +++ b/sound/core/jack.c @@ -48,8 +48,11 @@ static int snd_jack_dev_disconnect(struct snd_device *device) #ifdef CONFIG_SND_JACK_INPUT_DEV struct snd_jack *jack = device->device_data; - if (!jack->input_dev) + mutex_lock(&jack->input_dev_lock); + if (!jack->input_dev) { + mutex_unlock(&jack->input_dev_lock); return 0; + } /* If the input device is registered with the input subsystem * then we need to use a different deallocator. */ @@ -58,6 +61,7 @@ static int snd_jack_dev_disconnect(struct snd_device *device) else input_free_device(jack->input_dev); jack->input_dev = NULL; + mutex_unlock(&jack->input_dev_lock); #endif /* CONFIG_SND_JACK_INPUT_DEV */ return 0; } @@ -96,8 +100,11 @@ static int snd_jack_dev_register(struct snd_device *device) snprintf(jack->name, sizeof(jack->name), "%s %s", card->shortname, jack->id); - if (!jack->input_dev) + mutex_lock(&jack->input_dev_lock); + if (!jack->input_dev) { + mutex_unlock(&jack->input_dev_lock); return 0; + } jack->input_dev->name = jack->name; @@ -122,6 +129,7 @@ static int snd_jack_dev_register(struct snd_device *device) if (err == 0) jack->registered = 1; + mutex_unlock(&jack->input_dev_lock); return err; } #endif /* CONFIG_SND_JACK_INPUT_DEV */ @@ -242,9 +250,11 @@ int snd_jack_new(struct snd_card *card, const char *id, int type, return -ENOMEM; } - /* don't creat input device for phantom jack */ - if (!phantom_jack) { #ifdef CONFIG_SND_JACK_INPUT_DEV + mutex_init(&jack->input_dev_lock); + + /* don't create input device for phantom jack */ + if (!phantom_jack) { int i; jack->input_dev = input_allocate_device(); @@ -262,8 +272,8 @@ int snd_jack_new(struct snd_card *card, const char *id, int type, input_set_capability(jack->input_dev, EV_SW, jack_switch_types[i]); -#endif /* CONFIG_SND_JACK_INPUT_DEV */ } +#endif /* CONFIG_SND_JACK_INPUT_DEV */ err = snd_device_new(card, SNDRV_DEV_JACK, jack, &ops); if (err < 0) @@ -303,10 +313,14 @@ EXPORT_SYMBOL(snd_jack_new); void snd_jack_set_parent(struct snd_jack *jack, struct device *parent) { WARN_ON(jack->registered); - if (!jack->input_dev) + mutex_lock(&jack->input_dev_lock); + if (!jack->input_dev) { + mutex_unlock(&jack->input_dev_lock); return; + } jack->input_dev->dev.parent = parent; + mutex_unlock(&jack->input_dev_lock); } EXPORT_SYMBOL(snd_jack_set_parent); @@ -354,6 +368,8 @@ EXPORT_SYMBOL(snd_jack_set_key); /** * snd_jack_report - Report the current status of a jack + * Note: This function uses mutexes and should be called from a + * context which can sleep (such as a workqueue). * * @jack: The jack to report status for * @status: The current status of the jack @@ -373,8 +389,11 @@ void snd_jack_report(struct snd_jack *jack, int status) status & jack_kctl->mask_bits); #ifdef CONFIG_SND_JACK_INPUT_DEV - if (!jack->input_dev) + mutex_lock(&jack->input_dev_lock); + if (!jack->input_dev) { + mutex_unlock(&jack->input_dev_lock); return; + } for (i = 0; i < ARRAY_SIZE(jack->key); i++) { int testbit = SND_JACK_BTN_0 >> i; @@ -393,6 +412,7 @@ void snd_jack_report(struct snd_jack *jack, int status) } input_sync(jack->input_dev); + mutex_unlock(&jack->input_dev_lock); #endif /* CONFIG_SND_JACK_INPUT_DEV */ } EXPORT_SYMBOL(snd_jack_report); -- 2.35.1

1 0

[PATCH AUTOSEL 4.19 32/38] ASoC: rt5645: Fix errorenous cleanup order
by Sasha Levin 30 May '22

30 May '22

From: Lin Ma <linma(a)zju.edu.cn> [ Upstream commit 2def44d3aec59e38d2701c568d65540783f90f2f ] There is a logic error when removing rt5645 device as the function rt5645_i2c_remove() first cancel the &rt5645->jack_detect_work and delete the &rt5645->btn_check_timer latter. However, since the timer handler rt5645_btn_check_callback() will re-queue the jack_detect_work, this cleanup order is buggy. That is, once the del_timer_sync in rt5645_i2c_remove is concurrently run with the rt5645_btn_check_callback, the canceled jack_detect_work will be rescheduled again, leading to possible use-after-free. This patch fix the issue by placing the del_timer_sync function before the cancel_delayed_work_sync. Signed-off-by: Lin Ma <linma(a)zju.edu.cn> Link: https://lore.kernel.org/r/20220516092035.28283-1-linma@zju.edu.cn Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt5645.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 9185bd7c5a6d..d34000182f67 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -4105,9 +4105,14 @@ static int rt5645_i2c_remove(struct i2c_client *i2c) if (i2c->irq) free_irq(i2c->irq, rt5645); + /* + * Since the rt5645_btn_check_callback() can queue jack_detect_work, + * the timer need to be delted first + */ + del_timer_sync(&rt5645->btn_check_timer); + cancel_delayed_work_sync(&rt5645->jack_detect_work); cancel_delayed_work_sync(&rt5645->rcclock_work); - del_timer_sync(&rt5645->btn_check_timer); regulator_bulk_disable(ARRAY_SIZE(rt5645->supplies), rt5645->supplies); -- 2.35.1

1 0

[PATCH AUTOSEL 4.19 23/38] ASoC: tscs454: Add endianness flag in snd_soc_component_driver
by Sasha Levin 30 May '22

30 May '22

From: Charles Keepax <ckeepax(a)opensource.cirrus.com> [ Upstream commit ff69ec96b87dccb3a29edef8cec5d4fefbbc2055 ] The endianness flag is used on the CODEC side to specify an ambivalence to endian, typically because it is lost over the hardware link. This device receives audio over an I2S DAI and as such should have endianness applied. A fixup is also required to use the width directly rather than relying on the format in hw_params, now both little and big endian would be supported. It is worth noting this changes the behaviour of S24_LE to use a word length of 24 rather than 32. This would appear to be a correction since the fact S24_LE is stored as 32 bits should not be presented over the bus. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220504170905.332415-26-ckeepax@opensource.cirru… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/tscs454.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/tscs454.c b/sound/soc/codecs/tscs454.c index ff85a0bf6170..00a90ccd6566 100644 --- a/sound/soc/codecs/tscs454.c +++ b/sound/soc/codecs/tscs454.c @@ -3129,18 +3129,17 @@ static int set_aif_sample_format(struct snd_soc_component *component, unsigned int width; int ret; - switch (format) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (snd_pcm_format_width(format)) { + case 16: width = FV_WL_16; break; - case SNDRV_PCM_FORMAT_S20_3LE: + case 20: width = FV_WL_20; break; - case SNDRV_PCM_FORMAT_S24_3LE: + case 24: width = FV_WL_24; break; - case SNDRV_PCM_FORMAT_S24_LE: - case SNDRV_PCM_FORMAT_S32_LE: + case 32: width = FV_WL_32; break; default: @@ -3338,6 +3337,7 @@ static const struct snd_soc_component_driver soc_component_dev_tscs454 = { .num_dapm_routes = ARRAY_SIZE(tscs454_intercon), .controls = tscs454_snd_controls, .num_controls = ARRAY_SIZE(tscs454_snd_controls), + .endianness = 1, }; #define TSCS454_RATES SNDRV_PCM_RATE_8000_96000 -- 2.35.1

1 0

[PATCH AUTOSEL 4.19 21/38] ASoC: dapm: Don't fold register value changes into notifications
by Sasha Levin 30 May '22

30 May '22

From: Mark Brown <broonie(a)kernel.org> [ Upstream commit ad685980469b9f9b99d4d6ea05f4cb8f57cb2234 ] DAPM tracks and reports the value presented to the user from DAPM controls separately to the register value, these may diverge during initialisation or when an autodisable control is in use. When writing DAPM controls we currently report that a change has occurred if either the DAPM value or the value stored in the register has changed, meaning that if the two are out of sync we may appear to report a spurious event to userspace. Since we use this folded in value for nothing other than the value reported to userspace simply drop the folding in of the register change. Signed-off-by: Mark Brown <broonie(a)kernel.org> Link: https://lore.kernel.org/r/20220428161833.3690050-1-broonie@kernel.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/soc-dapm.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index af9f28dd957d..4d70e6bc2c12 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3302,7 +3302,6 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, update.val = val; card->update = &update; } - change |= reg_change; ret = soc_dapm_mixer_update_power(card, kcontrol, connect, rconnect); @@ -3408,7 +3407,6 @@ int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol, update.val = val; card->update = &update; } - change |= reg_change; ret = soc_dapm_mux_update_power(card, kcontrol, item[0], e); -- 2.35.1

1 0

[PATCH AUTOSEL 4.19 09/38] ALSA: jack: Access input_dev under mutex
by Sasha Levin 30 May '22

30 May '22

From: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> [ Upstream commit 1b6a6fc5280e97559287b61eade2d4b363e836f2 ] It is possible when using ASoC that input_dev is unregistered while calling snd_jack_report, which causes NULL pointer dereference. In order to prevent this serialize access to input_dev using mutex lock. Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> Reviewed-by: Cezary Rojewski <cezary.rojewski(a)intel.com> Link: https://lore.kernel.org/r/20220412091628.3056922-1-amadeuszx.slawinski@linu… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- include/sound/jack.h | 1 + sound/core/jack.c | 34 +++++++++++++++++++++++++++------- 2 files changed, 28 insertions(+), 7 deletions(-) diff --git a/include/sound/jack.h b/include/sound/jack.h index 1e84bfb553cf..4742f842b457 100644 --- a/include/sound/jack.h +++ b/include/sound/jack.h @@ -77,6 +77,7 @@ struct snd_jack { const char *id; #ifdef CONFIG_SND_JACK_INPUT_DEV struct input_dev *input_dev; + struct mutex input_dev_lock; int registered; int type; char name[100]; diff --git a/sound/core/jack.c b/sound/core/jack.c index acb531749efb..074b15fcb0ac 100644 --- a/sound/core/jack.c +++ b/sound/core/jack.c @@ -48,8 +48,11 @@ static int snd_jack_dev_disconnect(struct snd_device *device) #ifdef CONFIG_SND_JACK_INPUT_DEV struct snd_jack *jack = device->device_data; - if (!jack->input_dev) + mutex_lock(&jack->input_dev_lock); + if (!jack->input_dev) { + mutex_unlock(&jack->input_dev_lock); return 0; + } /* If the input device is registered with the input subsystem * then we need to use a different deallocator. */ @@ -58,6 +61,7 @@ static int snd_jack_dev_disconnect(struct snd_device *device) else input_free_device(jack->input_dev); jack->input_dev = NULL; + mutex_unlock(&jack->input_dev_lock); #endif /* CONFIG_SND_JACK_INPUT_DEV */ return 0; } @@ -96,8 +100,11 @@ static int snd_jack_dev_register(struct snd_device *device) snprintf(jack->name, sizeof(jack->name), "%s %s", card->shortname, jack->id); - if (!jack->input_dev) + mutex_lock(&jack->input_dev_lock); + if (!jack->input_dev) { + mutex_unlock(&jack->input_dev_lock); return 0; + } jack->input_dev->name = jack->name; @@ -122,6 +129,7 @@ static int snd_jack_dev_register(struct snd_device *device) if (err == 0) jack->registered = 1; + mutex_unlock(&jack->input_dev_lock); return err; } #endif /* CONFIG_SND_JACK_INPUT_DEV */ @@ -242,9 +250,11 @@ int snd_jack_new(struct snd_card *card, const char *id, int type, return -ENOMEM; } - /* don't creat input device for phantom jack */ - if (!phantom_jack) { #ifdef CONFIG_SND_JACK_INPUT_DEV + mutex_init(&jack->input_dev_lock); + + /* don't create input device for phantom jack */ + if (!phantom_jack) { int i; jack->input_dev = input_allocate_device(); @@ -262,8 +272,8 @@ int snd_jack_new(struct snd_card *card, const char *id, int type, input_set_capability(jack->input_dev, EV_SW, jack_switch_types[i]); -#endif /* CONFIG_SND_JACK_INPUT_DEV */ } +#endif /* CONFIG_SND_JACK_INPUT_DEV */ err = snd_device_new(card, SNDRV_DEV_JACK, jack, &ops); if (err < 0) @@ -303,10 +313,14 @@ EXPORT_SYMBOL(snd_jack_new); void snd_jack_set_parent(struct snd_jack *jack, struct device *parent) { WARN_ON(jack->registered); - if (!jack->input_dev) + mutex_lock(&jack->input_dev_lock); + if (!jack->input_dev) { + mutex_unlock(&jack->input_dev_lock); return; + } jack->input_dev->dev.parent = parent; + mutex_unlock(&jack->input_dev_lock); } EXPORT_SYMBOL(snd_jack_set_parent); @@ -354,6 +368,8 @@ EXPORT_SYMBOL(snd_jack_set_key); /** * snd_jack_report - Report the current status of a jack + * Note: This function uses mutexes and should be called from a + * context which can sleep (such as a workqueue). * * @jack: The jack to report status for * @status: The current status of the jack @@ -373,8 +389,11 @@ void snd_jack_report(struct snd_jack *jack, int status) status & jack_kctl->mask_bits); #ifdef CONFIG_SND_JACK_INPUT_DEV - if (!jack->input_dev) + mutex_lock(&jack->input_dev_lock); + if (!jack->input_dev) { + mutex_unlock(&jack->input_dev_lock); return; + } for (i = 0; i < ARRAY_SIZE(jack->key); i++) { int testbit = SND_JACK_BTN_0 >> i; @@ -393,6 +412,7 @@ void snd_jack_report(struct snd_jack *jack, int status) } input_sync(jack->input_dev); + mutex_unlock(&jack->input_dev_lock); #endif /* CONFIG_SND_JACK_INPUT_DEV */ } EXPORT_SYMBOL(snd_jack_report); -- 2.35.1

1 0

[PATCH AUTOSEL 5.4 50/55] ASoC: max98357a: remove dependency on GPIOLIB
by Sasha Levin 30 May '22

30 May '22

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit 21ca3274333f5c1cbbf9d91e5b33f4f2463859b2 ] commit dcc2c012c7691 ("ASoC: Fix gpiolib dependencies") removed a series of unnecessary dependencies on GPIOLIB when the gpio was optional. A similar simplification seems valid for max98357a, so remove the dependency as well. This will avoid the following warning WARNING: unmet direct dependencies detected for SND_SOC_MAX98357A Depends on [n]: SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && GPIOLIB [=n] Selected by [y]: - SND_SOC_INTEL_SOF_CS42L42_MACH [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && SND_SOC_INTEL_MACH [=y] && (SND_SOC_SOF_HDA_LINK [=y] || SND_SOC_SOF_BAYTRAIL [=n]) && I2C [=y] && ACPI [=y] && SND_HDA_CODEC_HDMI [=y] && SND_SOC_SOF_HDA_AUDIO_CODEC [=y] && (MFD_INTEL_LPSS [=y] || COMPILE_TEST [=n]) Reported-by: kernel test robot <yujie.liu(a)intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Péter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Link: https://lore.kernel.org/r/20220517172647.468244-2-pierre-louis.bossart@linu… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/Kconfig | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 466dc67799f4..dfc536cd9d2f 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -759,7 +759,6 @@ config SND_SOC_MAX98095 config SND_SOC_MAX98357A tristate "Maxim MAX98357A CODEC" - depends on GPIOLIB config SND_SOC_MAX98371 tristate -- 2.35.1

1 0

[PATCH AUTOSEL 5.4 46/55] ASoC: rt5645: Fix errorenous cleanup order
by Sasha Levin 30 May '22

30 May '22

From: Lin Ma <linma(a)zju.edu.cn> [ Upstream commit 2def44d3aec59e38d2701c568d65540783f90f2f ] There is a logic error when removing rt5645 device as the function rt5645_i2c_remove() first cancel the &rt5645->jack_detect_work and delete the &rt5645->btn_check_timer latter. However, since the timer handler rt5645_btn_check_callback() will re-queue the jack_detect_work, this cleanup order is buggy. That is, once the del_timer_sync in rt5645_i2c_remove is concurrently run with the rt5645_btn_check_callback, the canceled jack_detect_work will be rescheduled again, leading to possible use-after-free. This patch fix the issue by placing the del_timer_sync function before the cancel_delayed_work_sync. Signed-off-by: Lin Ma <linma(a)zju.edu.cn> Link: https://lore.kernel.org/r/20220516092035.28283-1-linma@zju.edu.cn Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt5645.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index c83f7f5da96b..a66e93a3af74 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -4074,9 +4074,14 @@ static int rt5645_i2c_remove(struct i2c_client *i2c) if (i2c->irq) free_irq(i2c->irq, rt5645); + /* + * Since the rt5645_btn_check_callback() can queue jack_detect_work, + * the timer need to be delted first + */ + del_timer_sync(&rt5645->btn_check_timer); + cancel_delayed_work_sync(&rt5645->jack_detect_work); cancel_delayed_work_sync(&rt5645->rcclock_work); - del_timer_sync(&rt5645->btn_check_timer); regulator_bulk_disable(ARRAY_SIZE(rt5645->supplies), rt5645->supplies); -- 2.35.1

1 0

[PATCH AUTOSEL 5.4 33/55] ASoC: tscs454: Add endianness flag in snd_soc_component_driver
by Sasha Levin 30 May '22

30 May '22

From: Charles Keepax <ckeepax(a)opensource.cirrus.com> [ Upstream commit ff69ec96b87dccb3a29edef8cec5d4fefbbc2055 ] The endianness flag is used on the CODEC side to specify an ambivalence to endian, typically because it is lost over the hardware link. This device receives audio over an I2S DAI and as such should have endianness applied. A fixup is also required to use the width directly rather than relying on the format in hw_params, now both little and big endian would be supported. It is worth noting this changes the behaviour of S24_LE to use a word length of 24 rather than 32. This would appear to be a correction since the fact S24_LE is stored as 32 bits should not be presented over the bus. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220504170905.332415-26-ckeepax@opensource.cirru… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/tscs454.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/tscs454.c b/sound/soc/codecs/tscs454.c index c3587af9985c..3d981441b8d1 100644 --- a/sound/soc/codecs/tscs454.c +++ b/sound/soc/codecs/tscs454.c @@ -3128,18 +3128,17 @@ static int set_aif_sample_format(struct snd_soc_component *component, unsigned int width; int ret; - switch (format) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (snd_pcm_format_width(format)) { + case 16: width = FV_WL_16; break; - case SNDRV_PCM_FORMAT_S20_3LE: + case 20: width = FV_WL_20; break; - case SNDRV_PCM_FORMAT_S24_3LE: + case 24: width = FV_WL_24; break; - case SNDRV_PCM_FORMAT_S24_LE: - case SNDRV_PCM_FORMAT_S32_LE: + case 32: width = FV_WL_32; break; default: @@ -3337,6 +3336,7 @@ static const struct snd_soc_component_driver soc_component_dev_tscs454 = { .num_dapm_routes = ARRAY_SIZE(tscs454_intercon), .controls = tscs454_snd_controls, .num_controls = ARRAY_SIZE(tscs454_snd_controls), + .endianness = 1, }; #define TSCS454_RATES SNDRV_PCM_RATE_8000_96000 -- 2.35.1

1 0

[PATCH AUTOSEL 5.4 29/55] ASoC: dapm: Don't fold register value changes into notifications
by Sasha Levin 30 May '22

30 May '22

From: Mark Brown <broonie(a)kernel.org> [ Upstream commit ad685980469b9f9b99d4d6ea05f4cb8f57cb2234 ] DAPM tracks and reports the value presented to the user from DAPM controls separately to the register value, these may diverge during initialisation or when an autodisable control is in use. When writing DAPM controls we currently report that a change has occurred if either the DAPM value or the value stored in the register has changed, meaning that if the two are out of sync we may appear to report a spurious event to userspace. Since we use this folded in value for nothing other than the value reported to userspace simply drop the folding in of the register change. Signed-off-by: Mark Brown <broonie(a)kernel.org> Link: https://lore.kernel.org/r/20220428161833.3690050-1-broonie@kernel.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/soc-dapm.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 1c09dfb0c0f0..56c9c4189f26 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3421,7 +3421,6 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, update.val = val; card->update = &update; } - change |= reg_change; ret = soc_dapm_mixer_update_power(card, kcontrol, connect, rconnect); @@ -3527,7 +3526,6 @@ int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol, update.val = val; card->update = &update; } - change |= reg_change; ret = soc_dapm_mux_update_power(card, kcontrol, item[0], e); -- 2.35.1

1 0

[PATCH AUTOSEL 5.4 10/55] ALSA: jack: Access input_dev under mutex
by Sasha Levin 30 May '22

30 May '22

From: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> [ Upstream commit 1b6a6fc5280e97559287b61eade2d4b363e836f2 ] It is possible when using ASoC that input_dev is unregistered while calling snd_jack_report, which causes NULL pointer dereference. In order to prevent this serialize access to input_dev using mutex lock. Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> Reviewed-by: Cezary Rojewski <cezary.rojewski(a)intel.com> Link: https://lore.kernel.org/r/20220412091628.3056922-1-amadeuszx.slawinski@linu… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- include/sound/jack.h | 1 + sound/core/jack.c | 34 +++++++++++++++++++++++++++------- 2 files changed, 28 insertions(+), 7 deletions(-) diff --git a/include/sound/jack.h b/include/sound/jack.h index 9eb2b5ec1ec4..78f3619f3de9 100644 --- a/include/sound/jack.h +++ b/include/sound/jack.h @@ -62,6 +62,7 @@ struct snd_jack { const char *id; #ifdef CONFIG_SND_JACK_INPUT_DEV struct input_dev *input_dev; + struct mutex input_dev_lock; int registered; int type; char name[100]; diff --git a/sound/core/jack.c b/sound/core/jack.c index b00ae6f39f05..e7ac82d46821 100644 --- a/sound/core/jack.c +++ b/sound/core/jack.c @@ -34,8 +34,11 @@ static int snd_jack_dev_disconnect(struct snd_device *device) #ifdef CONFIG_SND_JACK_INPUT_DEV struct snd_jack *jack = device->device_data; - if (!jack->input_dev) + mutex_lock(&jack->input_dev_lock); + if (!jack->input_dev) { + mutex_unlock(&jack->input_dev_lock); return 0; + } /* If the input device is registered with the input subsystem * then we need to use a different deallocator. */ @@ -44,6 +47,7 @@ static int snd_jack_dev_disconnect(struct snd_device *device) else input_free_device(jack->input_dev); jack->input_dev = NULL; + mutex_unlock(&jack->input_dev_lock); #endif /* CONFIG_SND_JACK_INPUT_DEV */ return 0; } @@ -82,8 +86,11 @@ static int snd_jack_dev_register(struct snd_device *device) snprintf(jack->name, sizeof(jack->name), "%s %s", card->shortname, jack->id); - if (!jack->input_dev) + mutex_lock(&jack->input_dev_lock); + if (!jack->input_dev) { + mutex_unlock(&jack->input_dev_lock); return 0; + } jack->input_dev->name = jack->name; @@ -108,6 +115,7 @@ static int snd_jack_dev_register(struct snd_device *device) if (err == 0) jack->registered = 1; + mutex_unlock(&jack->input_dev_lock); return err; } #endif /* CONFIG_SND_JACK_INPUT_DEV */ @@ -228,9 +236,11 @@ int snd_jack_new(struct snd_card *card, const char *id, int type, return -ENOMEM; } - /* don't creat input device for phantom jack */ - if (!phantom_jack) { #ifdef CONFIG_SND_JACK_INPUT_DEV + mutex_init(&jack->input_dev_lock); + + /* don't create input device for phantom jack */ + if (!phantom_jack) { int i; jack->input_dev = input_allocate_device(); @@ -248,8 +258,8 @@ int snd_jack_new(struct snd_card *card, const char *id, int type, input_set_capability(jack->input_dev, EV_SW, jack_switch_types[i]); -#endif /* CONFIG_SND_JACK_INPUT_DEV */ } +#endif /* CONFIG_SND_JACK_INPUT_DEV */ err = snd_device_new(card, SNDRV_DEV_JACK, jack, &ops); if (err < 0) @@ -289,10 +299,14 @@ EXPORT_SYMBOL(snd_jack_new); void snd_jack_set_parent(struct snd_jack *jack, struct device *parent) { WARN_ON(jack->registered); - if (!jack->input_dev) + mutex_lock(&jack->input_dev_lock); + if (!jack->input_dev) { + mutex_unlock(&jack->input_dev_lock); return; + } jack->input_dev->dev.parent = parent; + mutex_unlock(&jack->input_dev_lock); } EXPORT_SYMBOL(snd_jack_set_parent); @@ -340,6 +354,8 @@ EXPORT_SYMBOL(snd_jack_set_key); /** * snd_jack_report - Report the current status of a jack + * Note: This function uses mutexes and should be called from a + * context which can sleep (such as a workqueue). * * @jack: The jack to report status for * @status: The current status of the jack @@ -359,8 +375,11 @@ void snd_jack_report(struct snd_jack *jack, int status) status & jack_kctl->mask_bits); #ifdef CONFIG_SND_JACK_INPUT_DEV - if (!jack->input_dev) + mutex_lock(&jack->input_dev_lock); + if (!jack->input_dev) { + mutex_unlock(&jack->input_dev_lock); return; + } for (i = 0; i < ARRAY_SIZE(jack->key); i++) { int testbit = SND_JACK_BTN_0 >> i; @@ -379,6 +398,7 @@ void snd_jack_report(struct snd_jack *jack, int status) } input_sync(jack->input_dev); + mutex_unlock(&jack->input_dev_lock); #endif /* CONFIG_SND_JACK_INPUT_DEV */ } EXPORT_SYMBOL(snd_jack_report); -- 2.35.1

1 0

[PATCH AUTOSEL 5.10 66/76] ASoC: rt1015p: remove dependency on GPIOLIB
by Sasha Levin 30 May '22

30 May '22

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit b390c25c6757b9d56cecdfbf6d55f15fc89a6386 ] commit dcc2c012c7691 ("ASoC: Fix gpiolib dependencies") removed a series of unnecessary dependencies on GPIOLIB when the gpio was optional. A similar simplification seems valid for rt1015p, so remove the dependency as well. This will avoid the following warning WARNING: unmet direct dependencies detected for SND_SOC_RT1015P Depends on [n]: SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && GPIOLIB [=n] Selected by [y]: - SND_SOC_INTEL_SOF_RT5682_MACH [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && SND_SOC_INTEL_MACH [=y] && (SND_SOC_SOF_HDA_LINK [=y] || SND_SOC_SOF_BAYTRAIL [=n]) && I2C [=y] && ACPI [=y] && (SND_HDA_CODEC_HDMI [=y] && SND_SOC_SOF_HDA_AUDIO_CODEC [=y] && (MFD_INTEL_LPSS [=y] || COMPILE_TEST [=y]) || SND_SOC_SOF_BAYTRAIL [=n] && (X86_INTEL_LPSS [=n] || COMPILE_TEST [=y])) Reported-by: kernel test robot <yujie.liu(a)intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Péter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Link: https://lore.kernel.org/r/20220517172647.468244-3-pierre-louis.bossart@linu… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/Kconfig | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 612fd7516666..25f331551f68 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -1098,7 +1098,6 @@ config SND_SOC_RT1015 config SND_SOC_RT1015P tristate - depends on GPIOLIB config SND_SOC_RT1305 tristate -- 2.35.1

1 0

[PATCH AUTOSEL 5.10 65/76] ASoC: max98357a: remove dependency on GPIOLIB
by Sasha Levin 30 May '22

30 May '22

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit 21ca3274333f5c1cbbf9d91e5b33f4f2463859b2 ] commit dcc2c012c7691 ("ASoC: Fix gpiolib dependencies") removed a series of unnecessary dependencies on GPIOLIB when the gpio was optional. A similar simplification seems valid for max98357a, so remove the dependency as well. This will avoid the following warning WARNING: unmet direct dependencies detected for SND_SOC_MAX98357A Depends on [n]: SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && GPIOLIB [=n] Selected by [y]: - SND_SOC_INTEL_SOF_CS42L42_MACH [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && SND_SOC_INTEL_MACH [=y] && (SND_SOC_SOF_HDA_LINK [=y] || SND_SOC_SOF_BAYTRAIL [=n]) && I2C [=y] && ACPI [=y] && SND_HDA_CODEC_HDMI [=y] && SND_SOC_SOF_HDA_AUDIO_CODEC [=y] && (MFD_INTEL_LPSS [=y] || COMPILE_TEST [=n]) Reported-by: kernel test robot <yujie.liu(a)intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Péter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Link: https://lore.kernel.org/r/20220517172647.468244-2-pierre-louis.bossart@linu… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/Kconfig | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 52c89a6f54e9..612fd7516666 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -857,7 +857,6 @@ config SND_SOC_MAX98095 config SND_SOC_MAX98357A tristate "Maxim MAX98357A CODEC" - depends on GPIOLIB config SND_SOC_MAX98371 tristate -- 2.35.1

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[PATCH AUTOSEL 5.10 61/76] ASoC: rt5645: Fix errorenous cleanup order
by Sasha Levin 30 May '22

30 May '22

From: Lin Ma <linma(a)zju.edu.cn> [ Upstream commit 2def44d3aec59e38d2701c568d65540783f90f2f ] There is a logic error when removing rt5645 device as the function rt5645_i2c_remove() first cancel the &rt5645->jack_detect_work and delete the &rt5645->btn_check_timer latter. However, since the timer handler rt5645_btn_check_callback() will re-queue the jack_detect_work, this cleanup order is buggy. That is, once the del_timer_sync in rt5645_i2c_remove is concurrently run with the rt5645_btn_check_callback, the canceled jack_detect_work will be rescheduled again, leading to possible use-after-free. This patch fix the issue by placing the del_timer_sync function before the cancel_delayed_work_sync. Signed-off-by: Lin Ma <linma(a)zju.edu.cn> Link: https://lore.kernel.org/r/20220516092035.28283-1-linma@zju.edu.cn Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt5645.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 420003d062c7..d1533e95a74f 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -4095,9 +4095,14 @@ static int rt5645_i2c_remove(struct i2c_client *i2c) if (i2c->irq) free_irq(i2c->irq, rt5645); + /* + * Since the rt5645_btn_check_callback() can queue jack_detect_work, + * the timer need to be delted first + */ + del_timer_sync(&rt5645->btn_check_timer); + cancel_delayed_work_sync(&rt5645->jack_detect_work); cancel_delayed_work_sync(&rt5645->rcclock_work); - del_timer_sync(&rt5645->btn_check_timer); regulator_bulk_disable(ARRAY_SIZE(rt5645->supplies), rt5645->supplies); -- 2.35.1

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[PATCH AUTOSEL 5.10 45/76] ASoC: tscs454: Add endianness flag in snd_soc_component_driver
by Sasha Levin 30 May '22

30 May '22

From: Charles Keepax <ckeepax(a)opensource.cirrus.com> [ Upstream commit ff69ec96b87dccb3a29edef8cec5d4fefbbc2055 ] The endianness flag is used on the CODEC side to specify an ambivalence to endian, typically because it is lost over the hardware link. This device receives audio over an I2S DAI and as such should have endianness applied. A fixup is also required to use the width directly rather than relying on the format in hw_params, now both little and big endian would be supported. It is worth noting this changes the behaviour of S24_LE to use a word length of 24 rather than 32. This would appear to be a correction since the fact S24_LE is stored as 32 bits should not be presented over the bus. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220504170905.332415-26-ckeepax@opensource.cirru… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/tscs454.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/tscs454.c b/sound/soc/codecs/tscs454.c index d0af16b4db2f..a6f339bb4771 100644 --- a/sound/soc/codecs/tscs454.c +++ b/sound/soc/codecs/tscs454.c @@ -3115,18 +3115,17 @@ static int set_aif_sample_format(struct snd_soc_component *component, unsigned int width; int ret; - switch (format) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (snd_pcm_format_width(format)) { + case 16: width = FV_WL_16; break; - case SNDRV_PCM_FORMAT_S20_3LE: + case 20: width = FV_WL_20; break; - case SNDRV_PCM_FORMAT_S24_3LE: + case 24: width = FV_WL_24; break; - case SNDRV_PCM_FORMAT_S24_LE: - case SNDRV_PCM_FORMAT_S32_LE: + case 32: width = FV_WL_32; break; default: @@ -3321,6 +3320,7 @@ static const struct snd_soc_component_driver soc_component_dev_tscs454 = { .num_dapm_routes = ARRAY_SIZE(tscs454_intercon), .controls = tscs454_snd_controls, .num_controls = ARRAY_SIZE(tscs454_snd_controls), + .endianness = 1, }; #define TSCS454_RATES SNDRV_PCM_RATE_8000_96000 -- 2.35.1

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[PATCH AUTOSEL 5.10 40/76] ASoC: dapm: Don't fold register value changes into notifications
by Sasha Levin 30 May '22

30 May '22

From: Mark Brown <broonie(a)kernel.org> [ Upstream commit ad685980469b9f9b99d4d6ea05f4cb8f57cb2234 ] DAPM tracks and reports the value presented to the user from DAPM controls separately to the register value, these may diverge during initialisation or when an autodisable control is in use. When writing DAPM controls we currently report that a change has occurred if either the DAPM value or the value stored in the register has changed, meaning that if the two are out of sync we may appear to report a spurious event to userspace. Since we use this folded in value for nothing other than the value reported to userspace simply drop the folding in of the register change. Signed-off-by: Mark Brown <broonie(a)kernel.org> Link: https://lore.kernel.org/r/20220428161833.3690050-1-broonie@kernel.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/soc-dapm.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 417732bdf286..f2f7f2dde93c 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3427,7 +3427,6 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, update.val = val; card->update = &update; } - change |= reg_change; ret = soc_dapm_mixer_update_power(card, kcontrol, connect, rconnect); @@ -3529,7 +3528,6 @@ int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol, update.val = val; card->update = &update; } - change |= reg_change; ret = soc_dapm_mux_update_power(card, kcontrol, item[0], e); -- 2.35.1

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[PATCH AUTOSEL 5.10 32/76] ASoC: Intel: bytcr_rt5640: Add quirk for the HP Pro Tablet 408
by Sasha Levin 30 May '22

30 May '22

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit ce216cfa84a4e1c23b105e652c550bdeaac9e922 ] Add a quirk for the HP Pro Tablet 408, this BYTCR tablet has no CHAN package in its ACPI tables and uses SSP0-AIF1 rather then SSP0-AIF2 which is the default for BYTCR devices. It also uses DMIC1 for the internal mic rather then the default IN3 and it uses JD2 rather then the default JD1 for jack-detect. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=211485 Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20220427134918.527381-1-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/bytcr_rt5640.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 43ee3d095a1b..3020a993f6ef 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -615,6 +615,18 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { BYT_RT5640_OVCD_SF_0P75 | BYT_RT5640_MCLK_EN), }, + { /* HP Pro Tablet 408 */ + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Hewlett-Packard"), + DMI_MATCH(DMI_PRODUCT_NAME, "HP Pro Tablet 408"), + }, + .driver_data = (void *)(BYT_RT5640_DMIC1_MAP | + BYT_RT5640_JD_SRC_JD2_IN4N | + BYT_RT5640_OVCD_TH_1500UA | + BYT_RT5640_OVCD_SF_0P75 | + BYT_RT5640_SSP0_AIF1 | + BYT_RT5640_MCLK_EN), + }, { /* HP Stream 7 */ .matches = { DMI_EXACT_MATCH(DMI_SYS_VENDOR, "Hewlett-Packard"), -- 2.35.1

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[PATCH AUTOSEL 5.10 16/76] ALSA: jack: Access input_dev under mutex
by Sasha Levin 30 May '22

30 May '22

From: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> [ Upstream commit 1b6a6fc5280e97559287b61eade2d4b363e836f2 ] It is possible when using ASoC that input_dev is unregistered while calling snd_jack_report, which causes NULL pointer dereference. In order to prevent this serialize access to input_dev using mutex lock. Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> Reviewed-by: Cezary Rojewski <cezary.rojewski(a)intel.com> Link: https://lore.kernel.org/r/20220412091628.3056922-1-amadeuszx.slawinski@linu… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- include/sound/jack.h | 1 + sound/core/jack.c | 34 +++++++++++++++++++++++++++------- 2 files changed, 28 insertions(+), 7 deletions(-) diff --git a/include/sound/jack.h b/include/sound/jack.h index 9eb2b5ec1ec4..78f3619f3de9 100644 --- a/include/sound/jack.h +++ b/include/sound/jack.h @@ -62,6 +62,7 @@ struct snd_jack { const char *id; #ifdef CONFIG_SND_JACK_INPUT_DEV struct input_dev *input_dev; + struct mutex input_dev_lock; int registered; int type; char name[100]; diff --git a/sound/core/jack.c b/sound/core/jack.c index dc2e06ae2414..45e28db6ea38 100644 --- a/sound/core/jack.c +++ b/sound/core/jack.c @@ -34,8 +34,11 @@ static int snd_jack_dev_disconnect(struct snd_device *device) #ifdef CONFIG_SND_JACK_INPUT_DEV struct snd_jack *jack = device->device_data; - if (!jack->input_dev) + mutex_lock(&jack->input_dev_lock); + if (!jack->input_dev) { + mutex_unlock(&jack->input_dev_lock); return 0; + } /* If the input device is registered with the input subsystem * then we need to use a different deallocator. */ @@ -44,6 +47,7 @@ static int snd_jack_dev_disconnect(struct snd_device *device) else input_free_device(jack->input_dev); jack->input_dev = NULL; + mutex_unlock(&jack->input_dev_lock); #endif /* CONFIG_SND_JACK_INPUT_DEV */ return 0; } @@ -82,8 +86,11 @@ static int snd_jack_dev_register(struct snd_device *device) snprintf(jack->name, sizeof(jack->name), "%s %s", card->shortname, jack->id); - if (!jack->input_dev) + mutex_lock(&jack->input_dev_lock); + if (!jack->input_dev) { + mutex_unlock(&jack->input_dev_lock); return 0; + } jack->input_dev->name = jack->name; @@ -108,6 +115,7 @@ static int snd_jack_dev_register(struct snd_device *device) if (err == 0) jack->registered = 1; + mutex_unlock(&jack->input_dev_lock); return err; } #endif /* CONFIG_SND_JACK_INPUT_DEV */ @@ -228,9 +236,11 @@ int snd_jack_new(struct snd_card *card, const char *id, int type, return -ENOMEM; } - /* don't creat input device for phantom jack */ - if (!phantom_jack) { #ifdef CONFIG_SND_JACK_INPUT_DEV + mutex_init(&jack->input_dev_lock); + + /* don't create input device for phantom jack */ + if (!phantom_jack) { int i; jack->input_dev = input_allocate_device(); @@ -248,8 +258,8 @@ int snd_jack_new(struct snd_card *card, const char *id, int type, input_set_capability(jack->input_dev, EV_SW, jack_switch_types[i]); -#endif /* CONFIG_SND_JACK_INPUT_DEV */ } +#endif /* CONFIG_SND_JACK_INPUT_DEV */ err = snd_device_new(card, SNDRV_DEV_JACK, jack, &ops); if (err < 0) @@ -289,10 +299,14 @@ EXPORT_SYMBOL(snd_jack_new); void snd_jack_set_parent(struct snd_jack *jack, struct device *parent) { WARN_ON(jack->registered); - if (!jack->input_dev) + mutex_lock(&jack->input_dev_lock); + if (!jack->input_dev) { + mutex_unlock(&jack->input_dev_lock); return; + } jack->input_dev->dev.parent = parent; + mutex_unlock(&jack->input_dev_lock); } EXPORT_SYMBOL(snd_jack_set_parent); @@ -340,6 +354,8 @@ EXPORT_SYMBOL(snd_jack_set_key); /** * snd_jack_report - Report the current status of a jack + * Note: This function uses mutexes and should be called from a + * context which can sleep (such as a workqueue). * * @jack: The jack to report status for * @status: The current status of the jack @@ -359,8 +375,11 @@ void snd_jack_report(struct snd_jack *jack, int status) status & jack_kctl->mask_bits); #ifdef CONFIG_SND_JACK_INPUT_DEV - if (!jack->input_dev) + mutex_lock(&jack->input_dev_lock); + if (!jack->input_dev) { + mutex_unlock(&jack->input_dev_lock); return; + } for (i = 0; i < ARRAY_SIZE(jack->key); i++) { int testbit = SND_JACK_BTN_0 >> i; @@ -379,6 +398,7 @@ void snd_jack_report(struct snd_jack *jack, int status) } input_sync(jack->input_dev); + mutex_unlock(&jack->input_dev_lock); #endif /* CONFIG_SND_JACK_INPUT_DEV */ } EXPORT_SYMBOL(snd_jack_report); -- 2.35.1

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[PATCH AUTOSEL 5.15 094/109] ASoC: rt1015p: remove dependency on GPIOLIB
by Sasha Levin 30 May '22

30 May '22

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit b390c25c6757b9d56cecdfbf6d55f15fc89a6386 ] commit dcc2c012c7691 ("ASoC: Fix gpiolib dependencies") removed a series of unnecessary dependencies on GPIOLIB when the gpio was optional. A similar simplification seems valid for rt1015p, so remove the dependency as well. This will avoid the following warning WARNING: unmet direct dependencies detected for SND_SOC_RT1015P Depends on [n]: SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && GPIOLIB [=n] Selected by [y]: - SND_SOC_INTEL_SOF_RT5682_MACH [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && SND_SOC_INTEL_MACH [=y] && (SND_SOC_SOF_HDA_LINK [=y] || SND_SOC_SOF_BAYTRAIL [=n]) && I2C [=y] && ACPI [=y] && (SND_HDA_CODEC_HDMI [=y] && SND_SOC_SOF_HDA_AUDIO_CODEC [=y] && (MFD_INTEL_LPSS [=y] || COMPILE_TEST [=y]) || SND_SOC_SOF_BAYTRAIL [=n] && (X86_INTEL_LPSS [=n] || COMPILE_TEST [=y])) Reported-by: kernel test robot <yujie.liu(a)intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Péter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Link: https://lore.kernel.org/r/20220517172647.468244-3-pierre-louis.bossart@linu… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/Kconfig | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 2cbf4fc0f675..d59a7e99ce42 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -1148,7 +1148,6 @@ config SND_SOC_RT1015 config SND_SOC_RT1015P tristate - depends on GPIOLIB config SND_SOC_RT1019 tristate -- 2.35.1

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[PATCH AUTOSEL 5.15 093/109] ASoC: max98357a: remove dependency on GPIOLIB
by Sasha Levin 30 May '22

30 May '22

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit 21ca3274333f5c1cbbf9d91e5b33f4f2463859b2 ] commit dcc2c012c7691 ("ASoC: Fix gpiolib dependencies") removed a series of unnecessary dependencies on GPIOLIB when the gpio was optional. A similar simplification seems valid for max98357a, so remove the dependency as well. This will avoid the following warning WARNING: unmet direct dependencies detected for SND_SOC_MAX98357A Depends on [n]: SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && GPIOLIB [=n] Selected by [y]: - SND_SOC_INTEL_SOF_CS42L42_MACH [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && SND_SOC_INTEL_MACH [=y] && (SND_SOC_SOF_HDA_LINK [=y] || SND_SOC_SOF_BAYTRAIL [=n]) && I2C [=y] && ACPI [=y] && SND_HDA_CODEC_HDMI [=y] && SND_SOC_SOF_HDA_AUDIO_CODEC [=y] && (MFD_INTEL_LPSS [=y] || COMPILE_TEST [=n]) Reported-by: kernel test robot <yujie.liu(a)intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Péter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Link: https://lore.kernel.org/r/20220517172647.468244-2-pierre-louis.bossart@linu… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/Kconfig | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 47e675e8bd00..2cbf4fc0f675 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -900,7 +900,6 @@ config SND_SOC_MAX98095 config SND_SOC_MAX98357A tristate "Maxim MAX98357A CODEC" - depends on GPIOLIB config SND_SOC_MAX98371 tristate -- 2.35.1

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[PATCH AUTOSEL 5.15 086/109] ASoC: rt5645: Fix errorenous cleanup order
by Sasha Levin 30 May '22

30 May '22

From: Lin Ma <linma(a)zju.edu.cn> [ Upstream commit 2def44d3aec59e38d2701c568d65540783f90f2f ] There is a logic error when removing rt5645 device as the function rt5645_i2c_remove() first cancel the &rt5645->jack_detect_work and delete the &rt5645->btn_check_timer latter. However, since the timer handler rt5645_btn_check_callback() will re-queue the jack_detect_work, this cleanup order is buggy. That is, once the del_timer_sync in rt5645_i2c_remove is concurrently run with the rt5645_btn_check_callback, the canceled jack_detect_work will be rescheduled again, leading to possible use-after-free. This patch fix the issue by placing the del_timer_sync function before the cancel_delayed_work_sync. Signed-off-by: Lin Ma <linma(a)zju.edu.cn> Link: https://lore.kernel.org/r/20220516092035.28283-1-linma@zju.edu.cn Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt5645.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 9408ee63cb26..8ea6d4333562 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -4154,9 +4154,14 @@ static int rt5645_i2c_remove(struct i2c_client *i2c) if (i2c->irq) free_irq(i2c->irq, rt5645); + /* + * Since the rt5645_btn_check_callback() can queue jack_detect_work, + * the timer need to be delted first + */ + del_timer_sync(&rt5645->btn_check_timer); + cancel_delayed_work_sync(&rt5645->jack_detect_work); cancel_delayed_work_sync(&rt5645->rcclock_work); - del_timer_sync(&rt5645->btn_check_timer); regulator_bulk_disable(ARRAY_SIZE(rt5645->supplies), rt5645->supplies); -- 2.35.1

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[PATCH AUTOSEL 5.15 062/109] ASoC: tscs454: Add endianness flag in snd_soc_component_driver
by Sasha Levin 30 May '22

30 May '22

From: Charles Keepax <ckeepax(a)opensource.cirrus.com> [ Upstream commit ff69ec96b87dccb3a29edef8cec5d4fefbbc2055 ] The endianness flag is used on the CODEC side to specify an ambivalence to endian, typically because it is lost over the hardware link. This device receives audio over an I2S DAI and as such should have endianness applied. A fixup is also required to use the width directly rather than relying on the format in hw_params, now both little and big endian would be supported. It is worth noting this changes the behaviour of S24_LE to use a word length of 24 rather than 32. This would appear to be a correction since the fact S24_LE is stored as 32 bits should not be presented over the bus. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220504170905.332415-26-ckeepax@opensource.cirru… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/tscs454.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/tscs454.c b/sound/soc/codecs/tscs454.c index 43220bb36701..c27ca9a273e1 100644 --- a/sound/soc/codecs/tscs454.c +++ b/sound/soc/codecs/tscs454.c @@ -3120,18 +3120,17 @@ static int set_aif_sample_format(struct snd_soc_component *component, unsigned int width; int ret; - switch (format) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (snd_pcm_format_width(format)) { + case 16: width = FV_WL_16; break; - case SNDRV_PCM_FORMAT_S20_3LE: + case 20: width = FV_WL_20; break; - case SNDRV_PCM_FORMAT_S24_3LE: + case 24: width = FV_WL_24; break; - case SNDRV_PCM_FORMAT_S24_LE: - case SNDRV_PCM_FORMAT_S32_LE: + case 32: width = FV_WL_32; break; default: @@ -3326,6 +3325,7 @@ static const struct snd_soc_component_driver soc_component_dev_tscs454 = { .num_dapm_routes = ARRAY_SIZE(tscs454_intercon), .controls = tscs454_snd_controls, .num_controls = ARRAY_SIZE(tscs454_snd_controls), + .endianness = 1, }; #define TSCS454_RATES SNDRV_PCM_RATE_8000_96000 -- 2.35.1

1 0

[PATCH AUTOSEL 5.15 054/109] ASoC: dapm: Don't fold register value changes into notifications
by Sasha Levin 30 May '22

30 May '22

From: Mark Brown <broonie(a)kernel.org> [ Upstream commit ad685980469b9f9b99d4d6ea05f4cb8f57cb2234 ] DAPM tracks and reports the value presented to the user from DAPM controls separately to the register value, these may diverge during initialisation or when an autodisable control is in use. When writing DAPM controls we currently report that a change has occurred if either the DAPM value or the value stored in the register has changed, meaning that if the two are out of sync we may appear to report a spurious event to userspace. Since we use this folded in value for nothing other than the value reported to userspace simply drop the folding in of the register change. Signed-off-by: Mark Brown <broonie(a)kernel.org> Link: https://lore.kernel.org/r/20220428161833.3690050-1-broonie@kernel.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/soc-dapm.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 0b166e074457..47b85ba5b7d6 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3428,7 +3428,6 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, update.val = val; card->update = &update; } - change |= reg_change; ret = soc_dapm_mixer_update_power(card, kcontrol, connect, rconnect); @@ -3530,7 +3529,6 @@ int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol, update.val = val; card->update = &update; } - change |= reg_change; ret = soc_dapm_mux_update_power(card, kcontrol, item[0], e); -- 2.35.1

1 0

[PATCH AUTOSEL 5.15 045/109] ASoC: Intel: bytcr_rt5640: Add quirk for the HP Pro Tablet 408
by Sasha Levin 30 May '22

30 May '22

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit ce216cfa84a4e1c23b105e652c550bdeaac9e922 ] Add a quirk for the HP Pro Tablet 408, this BYTCR tablet has no CHAN package in its ACPI tables and uses SSP0-AIF1 rather then SSP0-AIF2 which is the default for BYTCR devices. It also uses DMIC1 for the internal mic rather then the default IN3 and it uses JD2 rather then the default JD1 for jack-detect. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=211485 Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20220427134918.527381-1-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/bytcr_rt5640.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index a6e837290c7d..f9c82ebc552c 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -759,6 +759,18 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { BYT_RT5640_OVCD_SF_0P75 | BYT_RT5640_MCLK_EN), }, + { /* HP Pro Tablet 408 */ + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Hewlett-Packard"), + DMI_MATCH(DMI_PRODUCT_NAME, "HP Pro Tablet 408"), + }, + .driver_data = (void *)(BYT_RT5640_DMIC1_MAP | + BYT_RT5640_JD_SRC_JD2_IN4N | + BYT_RT5640_OVCD_TH_1500UA | + BYT_RT5640_OVCD_SF_0P75 | + BYT_RT5640_SSP0_AIF1 | + BYT_RT5640_MCLK_EN), + }, { /* HP Stream 7 */ .matches = { DMI_EXACT_MATCH(DMI_SYS_VENDOR, "Hewlett-Packard"), -- 2.35.1

1 0

[PATCH AUTOSEL 5.15 030/109] ASoC: rsnd: care return value from rsnd_node_fixed_index()
by Sasha Levin 30 May '22

30 May '22

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> [ Upstream commit d09a7db431c65aaa8303eb456439d1831ca2e6b4 ] Renesas Sound is very complex, and thus it needs to use rsnd_node_fixed_index() to know enabled pin index. It returns error if strange pin was selected, but some codes didn't check it. This patch 1) indicates error message, 2) check return value. Reported-by: kernel test robot <lkp(a)intel.com> Reported-by: Dan Carpenter <dan.carpenter(a)oracle.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Link: https://lore.kernel.org/r/87pmlbgn5t.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/sh/rcar/core.c | 15 ++++++++++----- sound/soc/sh/rcar/dma.c | 9 ++++++++- sound/soc/sh/rcar/rsnd.h | 2 +- sound/soc/sh/rcar/src.c | 7 ++++++- sound/soc/sh/rcar/ssi.c | 14 ++++++++++++-- sound/soc/sh/rcar/ssiu.c | 7 ++++++- 6 files changed, 43 insertions(+), 11 deletions(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 6a8fe0da7670..af8ef2a27d34 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -1159,6 +1159,7 @@ void rsnd_parse_connect_common(struct rsnd_dai *rdai, char *name, struct device_node *capture) { struct rsnd_priv *priv = rsnd_rdai_to_priv(rdai); + struct device *dev = rsnd_priv_to_dev(priv); struct device_node *np; int i; @@ -1169,7 +1170,11 @@ void rsnd_parse_connect_common(struct rsnd_dai *rdai, char *name, for_each_child_of_node(node, np) { struct rsnd_mod *mod; - i = rsnd_node_fixed_index(np, name, i); + i = rsnd_node_fixed_index(dev, np, name, i); + if (i < 0) { + of_node_put(np); + break; + } mod = mod_get(priv, i); @@ -1183,7 +1188,7 @@ void rsnd_parse_connect_common(struct rsnd_dai *rdai, char *name, of_node_put(node); } -int rsnd_node_fixed_index(struct device_node *node, char *name, int idx) +int rsnd_node_fixed_index(struct device *dev, struct device_node *node, char *name, int idx) { char node_name[16]; @@ -1210,6 +1215,8 @@ int rsnd_node_fixed_index(struct device_node *node, char *name, int idx) return idx; } + dev_err(dev, "strange node numbering (%s)", + of_node_full_name(node)); return -EINVAL; } @@ -1221,10 +1228,8 @@ int rsnd_node_count(struct rsnd_priv *priv, struct device_node *node, char *name i = 0; for_each_child_of_node(node, np) { - i = rsnd_node_fixed_index(np, name, i); + i = rsnd_node_fixed_index(dev, np, name, i); if (i < 0) { - dev_err(dev, "strange node numbering (%s)", - of_node_full_name(node)); of_node_put(np); return 0; } diff --git a/sound/soc/sh/rcar/dma.c b/sound/soc/sh/rcar/dma.c index 03e0d4eca781..463ab237d7bd 100644 --- a/sound/soc/sh/rcar/dma.c +++ b/sound/soc/sh/rcar/dma.c @@ -240,12 +240,19 @@ static int rsnd_dmaen_start(struct rsnd_mod *mod, struct dma_chan *rsnd_dma_request_channel(struct device_node *of_node, char *name, struct rsnd_mod *mod, char *x) { + struct rsnd_priv *priv = rsnd_mod_to_priv(mod); + struct device *dev = rsnd_priv_to_dev(priv); struct dma_chan *chan = NULL; struct device_node *np; int i = 0; for_each_child_of_node(of_node, np) { - i = rsnd_node_fixed_index(np, name, i); + i = rsnd_node_fixed_index(dev, np, name, i); + if (i < 0) { + chan = NULL; + of_node_put(np); + break; + } if (i == rsnd_mod_id_raw(mod) && (!chan)) chan = of_dma_request_slave_channel(np, x); diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index 6580bab0e229..d9cd190d7e19 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -460,7 +460,7 @@ void rsnd_parse_connect_common(struct rsnd_dai *rdai, char *name, struct device_node *playback, struct device_node *capture); int rsnd_node_count(struct rsnd_priv *priv, struct device_node *node, char *name); -int rsnd_node_fixed_index(struct device_node *node, char *name, int idx); +int rsnd_node_fixed_index(struct device *dev, struct device_node *node, char *name, int idx); int rsnd_channel_normalization(int chan); #define rsnd_runtime_channel_original(io) \ diff --git a/sound/soc/sh/rcar/src.c b/sound/soc/sh/rcar/src.c index 42a100c6303d..0ea84ae57c6a 100644 --- a/sound/soc/sh/rcar/src.c +++ b/sound/soc/sh/rcar/src.c @@ -676,7 +676,12 @@ int rsnd_src_probe(struct rsnd_priv *priv) if (!of_device_is_available(np)) goto skip; - i = rsnd_node_fixed_index(np, SRC_NAME, i); + i = rsnd_node_fixed_index(dev, np, SRC_NAME, i); + if (i < 0) { + ret = -EINVAL; + of_node_put(np); + goto rsnd_src_probe_done; + } src = rsnd_src_get(priv, i); diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 87e606f688d3..43c5e27dc5c8 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -1105,6 +1105,7 @@ void rsnd_parse_connect_ssi(struct rsnd_dai *rdai, struct device_node *capture) { struct rsnd_priv *priv = rsnd_rdai_to_priv(rdai); + struct device *dev = rsnd_priv_to_dev(priv); struct device_node *node; struct device_node *np; int i; @@ -1117,7 +1118,11 @@ void rsnd_parse_connect_ssi(struct rsnd_dai *rdai, for_each_child_of_node(node, np) { struct rsnd_mod *mod; - i = rsnd_node_fixed_index(np, SSI_NAME, i); + i = rsnd_node_fixed_index(dev, np, SSI_NAME, i); + if (i < 0) { + of_node_put(np); + break; + } mod = rsnd_ssi_mod_get(priv, i); @@ -1182,7 +1187,12 @@ int rsnd_ssi_probe(struct rsnd_priv *priv) if (!of_device_is_available(np)) goto skip; - i = rsnd_node_fixed_index(np, SSI_NAME, i); + i = rsnd_node_fixed_index(dev, np, SSI_NAME, i); + if (i < 0) { + ret = -EINVAL; + of_node_put(np); + goto rsnd_ssi_probe_done; + } ssi = rsnd_ssi_get(priv, i); diff --git a/sound/soc/sh/rcar/ssiu.c b/sound/soc/sh/rcar/ssiu.c index 138f95dd9f4a..4b8a63e336c7 100644 --- a/sound/soc/sh/rcar/ssiu.c +++ b/sound/soc/sh/rcar/ssiu.c @@ -462,6 +462,7 @@ void rsnd_parse_connect_ssiu(struct rsnd_dai *rdai, struct device_node *capture) { struct rsnd_priv *priv = rsnd_rdai_to_priv(rdai); + struct device *dev = rsnd_priv_to_dev(priv); struct device_node *node = rsnd_ssiu_of_node(priv); struct rsnd_dai_stream *io_p = &rdai->playback; struct rsnd_dai_stream *io_c = &rdai->capture; @@ -474,7 +475,11 @@ void rsnd_parse_connect_ssiu(struct rsnd_dai *rdai, for_each_child_of_node(node, np) { struct rsnd_mod *mod; - i = rsnd_node_fixed_index(np, SSIU_NAME, i); + i = rsnd_node_fixed_index(dev, np, SSIU_NAME, i); + if (i < 0) { + of_node_put(np); + break; + } mod = rsnd_ssiu_mod_get(priv, i); -- 2.35.1

1 0

[PATCH AUTOSEL 5.15 029/109] ASoC: rsnd: care default case on rsnd_ssiu_busif_err_status_clear()
by Sasha Levin 30 May '22

30 May '22

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> [ Upstream commit b1384d4c95088d01f4266237faabf165d3d605fc ] commit cfb7b8bf1e2d66 ("ASoC: rsnd: tidyup rsnd_ssiu_busif_err_status_clear()") merged duplicate code, but it didn't care about default case, and causes smatch warnings. smatch warnings: sound/soc/sh/rcar/ssiu.c:112 rsnd_ssiu_busif_err_status_clear() \ error: uninitialized symbol 'offset'. sound/soc/sh/rcar/ssiu.c:114 rsnd_ssiu_busif_err_status_clear() \ error: uninitialized symbol 'shift'. This patch cares it. Reported-by: kernel test robot <lkp(a)intel.com> Reported-by: Dan Carpenter <dan.carpenter(a)oracle.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Link: https://lore.kernel.org/r/87r15rgn6p.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/sh/rcar/ssiu.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/sh/rcar/ssiu.c b/sound/soc/sh/rcar/ssiu.c index 0d8f97633dd2..138f95dd9f4a 100644 --- a/sound/soc/sh/rcar/ssiu.c +++ b/sound/soc/sh/rcar/ssiu.c @@ -102,6 +102,8 @@ bool rsnd_ssiu_busif_err_status_clear(struct rsnd_mod *mod) shift = 1; offset = 1; break; + default: + goto out; } for (i = 0; i < 4; i++) { @@ -120,7 +122,7 @@ bool rsnd_ssiu_busif_err_status_clear(struct rsnd_mod *mod) } rsnd_mod_write(mod, reg, val); } - +out: return error; } -- 2.35.1

1 0

[PATCH AUTOSEL 5.15 020/109] ALSA: jack: Access input_dev under mutex
by Sasha Levin 30 May '22

30 May '22

From: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> [ Upstream commit 1b6a6fc5280e97559287b61eade2d4b363e836f2 ] It is possible when using ASoC that input_dev is unregistered while calling snd_jack_report, which causes NULL pointer dereference. In order to prevent this serialize access to input_dev using mutex lock. Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> Reviewed-by: Cezary Rojewski <cezary.rojewski(a)intel.com> Link: https://lore.kernel.org/r/20220412091628.3056922-1-amadeuszx.slawinski@linu… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- include/sound/jack.h | 1 + sound/core/jack.c | 34 +++++++++++++++++++++++++++------- 2 files changed, 28 insertions(+), 7 deletions(-) diff --git a/include/sound/jack.h b/include/sound/jack.h index 1181f536557e..1ed90e2109e9 100644 --- a/include/sound/jack.h +++ b/include/sound/jack.h @@ -62,6 +62,7 @@ struct snd_jack { const char *id; #ifdef CONFIG_SND_JACK_INPUT_DEV struct input_dev *input_dev; + struct mutex input_dev_lock; int registered; int type; char name[100]; diff --git a/sound/core/jack.c b/sound/core/jack.c index d1e3055f2b6a..88493cc31914 100644 --- a/sound/core/jack.c +++ b/sound/core/jack.c @@ -42,8 +42,11 @@ static int snd_jack_dev_disconnect(struct snd_device *device) #ifdef CONFIG_SND_JACK_INPUT_DEV struct snd_jack *jack = device->device_data; - if (!jack->input_dev) + mutex_lock(&jack->input_dev_lock); + if (!jack->input_dev) { + mutex_unlock(&jack->input_dev_lock); return 0; + } /* If the input device is registered with the input subsystem * then we need to use a different deallocator. */ @@ -52,6 +55,7 @@ static int snd_jack_dev_disconnect(struct snd_device *device) else input_free_device(jack->input_dev); jack->input_dev = NULL; + mutex_unlock(&jack->input_dev_lock); #endif /* CONFIG_SND_JACK_INPUT_DEV */ return 0; } @@ -90,8 +94,11 @@ static int snd_jack_dev_register(struct snd_device *device) snprintf(jack->name, sizeof(jack->name), "%s %s", card->shortname, jack->id); - if (!jack->input_dev) + mutex_lock(&jack->input_dev_lock); + if (!jack->input_dev) { + mutex_unlock(&jack->input_dev_lock); return 0; + } jack->input_dev->name = jack->name; @@ -116,6 +123,7 @@ static int snd_jack_dev_register(struct snd_device *device) if (err == 0) jack->registered = 1; + mutex_unlock(&jack->input_dev_lock); return err; } #endif /* CONFIG_SND_JACK_INPUT_DEV */ @@ -517,9 +525,11 @@ int snd_jack_new(struct snd_card *card, const char *id, int type, return -ENOMEM; } - /* don't creat input device for phantom jack */ - if (!phantom_jack) { #ifdef CONFIG_SND_JACK_INPUT_DEV + mutex_init(&jack->input_dev_lock); + + /* don't create input device for phantom jack */ + if (!phantom_jack) { int i; jack->input_dev = input_allocate_device(); @@ -537,8 +547,8 @@ int snd_jack_new(struct snd_card *card, const char *id, int type, input_set_capability(jack->input_dev, EV_SW, jack_switch_types[i]); -#endif /* CONFIG_SND_JACK_INPUT_DEV */ } +#endif /* CONFIG_SND_JACK_INPUT_DEV */ err = snd_device_new(card, SNDRV_DEV_JACK, jack, &ops); if (err < 0) @@ -578,10 +588,14 @@ EXPORT_SYMBOL(snd_jack_new); void snd_jack_set_parent(struct snd_jack *jack, struct device *parent) { WARN_ON(jack->registered); - if (!jack->input_dev) + mutex_lock(&jack->input_dev_lock); + if (!jack->input_dev) { + mutex_unlock(&jack->input_dev_lock); return; + } jack->input_dev->dev.parent = parent; + mutex_unlock(&jack->input_dev_lock); } EXPORT_SYMBOL(snd_jack_set_parent); @@ -629,6 +643,8 @@ EXPORT_SYMBOL(snd_jack_set_key); /** * snd_jack_report - Report the current status of a jack + * Note: This function uses mutexes and should be called from a + * context which can sleep (such as a workqueue). * * @jack: The jack to report status for * @status: The current status of the jack @@ -654,8 +670,11 @@ void snd_jack_report(struct snd_jack *jack, int status) status & jack_kctl->mask_bits); #ifdef CONFIG_SND_JACK_INPUT_DEV - if (!jack->input_dev) + mutex_lock(&jack->input_dev_lock); + if (!jack->input_dev) { + mutex_unlock(&jack->input_dev_lock); return; + } for (i = 0; i < ARRAY_SIZE(jack->key); i++) { int testbit = ((SND_JACK_BTN_0 >> i) & ~mask_bits); @@ -675,6 +694,7 @@ void snd_jack_report(struct snd_jack *jack, int status) } input_sync(jack->input_dev); + mutex_unlock(&jack->input_dev_lock); #endif /* CONFIG_SND_JACK_INPUT_DEV */ } EXPORT_SYMBOL(snd_jack_report); -- 2.35.1

1 0

[PATCH AUTOSEL 5.17 119/135] ASoC: rt1015p: remove dependency on GPIOLIB
by Sasha Levin 30 May '22

30 May '22

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit b390c25c6757b9d56cecdfbf6d55f15fc89a6386 ] commit dcc2c012c7691 ("ASoC: Fix gpiolib dependencies") removed a series of unnecessary dependencies on GPIOLIB when the gpio was optional. A similar simplification seems valid for rt1015p, so remove the dependency as well. This will avoid the following warning WARNING: unmet direct dependencies detected for SND_SOC_RT1015P Depends on [n]: SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && GPIOLIB [=n] Selected by [y]: - SND_SOC_INTEL_SOF_RT5682_MACH [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && SND_SOC_INTEL_MACH [=y] && (SND_SOC_SOF_HDA_LINK [=y] || SND_SOC_SOF_BAYTRAIL [=n]) && I2C [=y] && ACPI [=y] && (SND_HDA_CODEC_HDMI [=y] && SND_SOC_SOF_HDA_AUDIO_CODEC [=y] && (MFD_INTEL_LPSS [=y] || COMPILE_TEST [=y]) || SND_SOC_SOF_BAYTRAIL [=n] && (X86_INTEL_LPSS [=n] || COMPILE_TEST [=y])) Reported-by: kernel test robot <yujie.liu(a)intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Péter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Link: https://lore.kernel.org/r/20220517172647.468244-3-pierre-louis.bossart@linu… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/Kconfig | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index a8c6c2bfd5a7..3496403004ac 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -1202,7 +1202,6 @@ config SND_SOC_RT1015 config SND_SOC_RT1015P tristate - depends on GPIOLIB config SND_SOC_RT1019 tristate -- 2.35.1

1 0

[PATCH AUTOSEL 5.17 118/135] ASoC: max98357a: remove dependency on GPIOLIB
by Sasha Levin 30 May '22

30 May '22

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit 21ca3274333f5c1cbbf9d91e5b33f4f2463859b2 ] commit dcc2c012c7691 ("ASoC: Fix gpiolib dependencies") removed a series of unnecessary dependencies on GPIOLIB when the gpio was optional. A similar simplification seems valid for max98357a, so remove the dependency as well. This will avoid the following warning WARNING: unmet direct dependencies detected for SND_SOC_MAX98357A Depends on [n]: SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && GPIOLIB [=n] Selected by [y]: - SND_SOC_INTEL_SOF_CS42L42_MACH [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && SND_SOC_INTEL_MACH [=y] && (SND_SOC_SOF_HDA_LINK [=y] || SND_SOC_SOF_BAYTRAIL [=n]) && I2C [=y] && ACPI [=y] && SND_HDA_CODEC_HDMI [=y] && SND_SOC_SOF_HDA_AUDIO_CODEC [=y] && (MFD_INTEL_LPSS [=y] || COMPILE_TEST [=n]) Reported-by: kernel test robot <yujie.liu(a)intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Péter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Link: https://lore.kernel.org/r/20220517172647.468244-2-pierre-louis.bossart@linu… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/Kconfig | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 30c00380499c..a8c6c2bfd5a7 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -943,7 +943,6 @@ config SND_SOC_MAX98095 config SND_SOC_MAX98357A tristate "Maxim MAX98357A CODEC" - depends on GPIOLIB config SND_SOC_MAX98371 tristate -- 2.35.1

1 0

[PATCH AUTOSEL 5.17 106/135] ASoC: rt5645: Fix errorenous cleanup order
by Sasha Levin 30 May '22

30 May '22

From: Lin Ma <linma(a)zju.edu.cn> [ Upstream commit 2def44d3aec59e38d2701c568d65540783f90f2f ] There is a logic error when removing rt5645 device as the function rt5645_i2c_remove() first cancel the &rt5645->jack_detect_work and delete the &rt5645->btn_check_timer latter. However, since the timer handler rt5645_btn_check_callback() will re-queue the jack_detect_work, this cleanup order is buggy. That is, once the del_timer_sync in rt5645_i2c_remove is concurrently run with the rt5645_btn_check_callback, the canceled jack_detect_work will be rescheduled again, leading to possible use-after-free. This patch fix the issue by placing the del_timer_sync function before the cancel_delayed_work_sync. Signed-off-by: Lin Ma <linma(a)zju.edu.cn> Link: https://lore.kernel.org/r/20220516092035.28283-1-linma@zju.edu.cn Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt5645.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 197c56047947..4b2e027c1033 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -4154,9 +4154,14 @@ static int rt5645_i2c_remove(struct i2c_client *i2c) if (i2c->irq) free_irq(i2c->irq, rt5645); + /* + * Since the rt5645_btn_check_callback() can queue jack_detect_work, + * the timer need to be delted first + */ + del_timer_sync(&rt5645->btn_check_timer); + cancel_delayed_work_sync(&rt5645->jack_detect_work); cancel_delayed_work_sync(&rt5645->rcclock_work); - del_timer_sync(&rt5645->btn_check_timer); regulator_bulk_disable(ARRAY_SIZE(rt5645->supplies), rt5645->supplies); -- 2.35.1

1 0

[PATCH AUTOSEL 5.17 084/135] ASoC: SOF: amd: add missing platform_device_unregister in acp_pci_rn_probe
by Sasha Levin 30 May '22

30 May '22

From: Zheng Bin <zhengbin13(a)huawei.com> [ Upstream commit cbcab8cd737c74c20195c31d647e19f7cb49c9b8 ] acp_pci_rn_probe misses a call platform_device_unregister in error path, this patch fixes that. Signed-off-by: Zheng Bin <zhengbin13(a)huawei.com> Link: https://lore.kernel.org/r/20220512013728.4128903-1-zhengbin13@huawei.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/sof/amd/pci-rn.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/sof/amd/pci-rn.c b/sound/soc/sof/amd/pci-rn.c index 392ffbdf6417..d809d151a38c 100644 --- a/sound/soc/sof/amd/pci-rn.c +++ b/sound/soc/sof/amd/pci-rn.c @@ -93,6 +93,7 @@ static int acp_pci_rn_probe(struct pci_dev *pci, const struct pci_device_id *pci res = devm_kzalloc(&pci->dev, sizeof(struct resource) * ARRAY_SIZE(renoir_res), GFP_KERNEL); if (!res) { sof_pci_remove(pci); + platform_device_unregister(dmic_dev); return -ENOMEM; } -- 2.35.1

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[PATCH AUTOSEL 5.17 074/135] ASoC: tscs454: Add endianness flag in snd_soc_component_driver
by Sasha Levin 30 May '22

30 May '22

From: Charles Keepax <ckeepax(a)opensource.cirrus.com> [ Upstream commit ff69ec96b87dccb3a29edef8cec5d4fefbbc2055 ] The endianness flag is used on the CODEC side to specify an ambivalence to endian, typically because it is lost over the hardware link. This device receives audio over an I2S DAI and as such should have endianness applied. A fixup is also required to use the width directly rather than relying on the format in hw_params, now both little and big endian would be supported. It is worth noting this changes the behaviour of S24_LE to use a word length of 24 rather than 32. This would appear to be a correction since the fact S24_LE is stored as 32 bits should not be presented over the bus. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220504170905.332415-26-ckeepax@opensource.cirru… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/tscs454.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/tscs454.c b/sound/soc/codecs/tscs454.c index 43220bb36701..c27ca9a273e1 100644 --- a/sound/soc/codecs/tscs454.c +++ b/sound/soc/codecs/tscs454.c @@ -3120,18 +3120,17 @@ static int set_aif_sample_format(struct snd_soc_component *component, unsigned int width; int ret; - switch (format) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (snd_pcm_format_width(format)) { + case 16: width = FV_WL_16; break; - case SNDRV_PCM_FORMAT_S20_3LE: + case 20: width = FV_WL_20; break; - case SNDRV_PCM_FORMAT_S24_3LE: + case 24: width = FV_WL_24; break; - case SNDRV_PCM_FORMAT_S24_LE: - case SNDRV_PCM_FORMAT_S32_LE: + case 32: width = FV_WL_32; break; default: @@ -3326,6 +3325,7 @@ static const struct snd_soc_component_driver soc_component_dev_tscs454 = { .num_dapm_routes = ARRAY_SIZE(tscs454_intercon), .controls = tscs454_snd_controls, .num_controls = ARRAY_SIZE(tscs454_snd_controls), + .endianness = 1, }; #define TSCS454_RATES SNDRV_PCM_RATE_8000_96000 -- 2.35.1

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[PATCH AUTOSEL 5.17 065/135] ASoC: dapm: Don't fold register value changes into notifications
by Sasha Levin 30 May '22

30 May '22

From: Mark Brown <broonie(a)kernel.org> [ Upstream commit ad685980469b9f9b99d4d6ea05f4cb8f57cb2234 ] DAPM tracks and reports the value presented to the user from DAPM controls separately to the register value, these may diverge during initialisation or when an autodisable control is in use. When writing DAPM controls we currently report that a change has occurred if either the DAPM value or the value stored in the register has changed, meaning that if the two are out of sync we may appear to report a spurious event to userspace. Since we use this folded in value for nothing other than the value reported to userspace simply drop the folding in of the register change. Signed-off-by: Mark Brown <broonie(a)kernel.org> Link: https://lore.kernel.org/r/20220428161833.3690050-1-broonie@kernel.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/soc-dapm.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index fb43b331a36e..f62dd119639d 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3430,7 +3430,6 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, update.val = val; card->update = &update; } - change |= reg_change; ret = soc_dapm_mixer_update_power(card, kcontrol, connect, rconnect); @@ -3532,7 +3531,6 @@ int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol, update.val = val; card->update = &update; } - change |= reg_change; ret = soc_dapm_mux_update_power(card, kcontrol, item[0], e); -- 2.35.1

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[PATCH AUTOSEL 5.17 053/135] ASoC: Intel: bytcr_rt5640: Add quirk for the HP Pro Tablet 408
by Sasha Levin 30 May '22

30 May '22

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit ce216cfa84a4e1c23b105e652c550bdeaac9e922 ] Add a quirk for the HP Pro Tablet 408, this BYTCR tablet has no CHAN package in its ACPI tables and uses SSP0-AIF1 rather then SSP0-AIF2 which is the default for BYTCR devices. It also uses DMIC1 for the internal mic rather then the default IN3 and it uses JD2 rather then the default JD1 for jack-detect. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=211485 Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20220427134918.527381-1-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/bytcr_rt5640.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 2ace32c03ec9..b5ac226c59e1 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -773,6 +773,18 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { BYT_RT5640_OVCD_SF_0P75 | BYT_RT5640_MCLK_EN), }, + { /* HP Pro Tablet 408 */ + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Hewlett-Packard"), + DMI_MATCH(DMI_PRODUCT_NAME, "HP Pro Tablet 408"), + }, + .driver_data = (void *)(BYT_RT5640_DMIC1_MAP | + BYT_RT5640_JD_SRC_JD2_IN4N | + BYT_RT5640_OVCD_TH_1500UA | + BYT_RT5640_OVCD_SF_0P75 | + BYT_RT5640_SSP0_AIF1 | + BYT_RT5640_MCLK_EN), + }, { /* HP Stream 7 */ .matches = { DMI_EXACT_MATCH(DMI_SYS_VENDOR, "Hewlett-Packard"), -- 2.35.1

1 0

[PATCH AUTOSEL 5.17 037/135] ASoC: rsnd: care return value from rsnd_node_fixed_index()
by Sasha Levin 30 May '22

30 May '22

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> [ Upstream commit d09a7db431c65aaa8303eb456439d1831ca2e6b4 ] Renesas Sound is very complex, and thus it needs to use rsnd_node_fixed_index() to know enabled pin index. It returns error if strange pin was selected, but some codes didn't check it. This patch 1) indicates error message, 2) check return value. Reported-by: kernel test robot <lkp(a)intel.com> Reported-by: Dan Carpenter <dan.carpenter(a)oracle.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Link: https://lore.kernel.org/r/87pmlbgn5t.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/sh/rcar/core.c | 15 ++++++++++----- sound/soc/sh/rcar/dma.c | 9 ++++++++- sound/soc/sh/rcar/rsnd.h | 2 +- sound/soc/sh/rcar/src.c | 7 ++++++- sound/soc/sh/rcar/ssi.c | 14 ++++++++++++-- sound/soc/sh/rcar/ssiu.c | 7 ++++++- 6 files changed, 43 insertions(+), 11 deletions(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 6a8fe0da7670..af8ef2a27d34 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -1159,6 +1159,7 @@ void rsnd_parse_connect_common(struct rsnd_dai *rdai, char *name, struct device_node *capture) { struct rsnd_priv *priv = rsnd_rdai_to_priv(rdai); + struct device *dev = rsnd_priv_to_dev(priv); struct device_node *np; int i; @@ -1169,7 +1170,11 @@ void rsnd_parse_connect_common(struct rsnd_dai *rdai, char *name, for_each_child_of_node(node, np) { struct rsnd_mod *mod; - i = rsnd_node_fixed_index(np, name, i); + i = rsnd_node_fixed_index(dev, np, name, i); + if (i < 0) { + of_node_put(np); + break; + } mod = mod_get(priv, i); @@ -1183,7 +1188,7 @@ void rsnd_parse_connect_common(struct rsnd_dai *rdai, char *name, of_node_put(node); } -int rsnd_node_fixed_index(struct device_node *node, char *name, int idx) +int rsnd_node_fixed_index(struct device *dev, struct device_node *node, char *name, int idx) { char node_name[16]; @@ -1210,6 +1215,8 @@ int rsnd_node_fixed_index(struct device_node *node, char *name, int idx) return idx; } + dev_err(dev, "strange node numbering (%s)", + of_node_full_name(node)); return -EINVAL; } @@ -1221,10 +1228,8 @@ int rsnd_node_count(struct rsnd_priv *priv, struct device_node *node, char *name i = 0; for_each_child_of_node(node, np) { - i = rsnd_node_fixed_index(np, name, i); + i = rsnd_node_fixed_index(dev, np, name, i); if (i < 0) { - dev_err(dev, "strange node numbering (%s)", - of_node_full_name(node)); of_node_put(np); return 0; } diff --git a/sound/soc/sh/rcar/dma.c b/sound/soc/sh/rcar/dma.c index 03e0d4eca781..463ab237d7bd 100644 --- a/sound/soc/sh/rcar/dma.c +++ b/sound/soc/sh/rcar/dma.c @@ -240,12 +240,19 @@ static int rsnd_dmaen_start(struct rsnd_mod *mod, struct dma_chan *rsnd_dma_request_channel(struct device_node *of_node, char *name, struct rsnd_mod *mod, char *x) { + struct rsnd_priv *priv = rsnd_mod_to_priv(mod); + struct device *dev = rsnd_priv_to_dev(priv); struct dma_chan *chan = NULL; struct device_node *np; int i = 0; for_each_child_of_node(of_node, np) { - i = rsnd_node_fixed_index(np, name, i); + i = rsnd_node_fixed_index(dev, np, name, i); + if (i < 0) { + chan = NULL; + of_node_put(np); + break; + } if (i == rsnd_mod_id_raw(mod) && (!chan)) chan = of_dma_request_slave_channel(np, x); diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index 6580bab0e229..d9cd190d7e19 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -460,7 +460,7 @@ void rsnd_parse_connect_common(struct rsnd_dai *rdai, char *name, struct device_node *playback, struct device_node *capture); int rsnd_node_count(struct rsnd_priv *priv, struct device_node *node, char *name); -int rsnd_node_fixed_index(struct device_node *node, char *name, int idx); +int rsnd_node_fixed_index(struct device *dev, struct device_node *node, char *name, int idx); int rsnd_channel_normalization(int chan); #define rsnd_runtime_channel_original(io) \ diff --git a/sound/soc/sh/rcar/src.c b/sound/soc/sh/rcar/src.c index 42a100c6303d..0ea84ae57c6a 100644 --- a/sound/soc/sh/rcar/src.c +++ b/sound/soc/sh/rcar/src.c @@ -676,7 +676,12 @@ int rsnd_src_probe(struct rsnd_priv *priv) if (!of_device_is_available(np)) goto skip; - i = rsnd_node_fixed_index(np, SRC_NAME, i); + i = rsnd_node_fixed_index(dev, np, SRC_NAME, i); + if (i < 0) { + ret = -EINVAL; + of_node_put(np); + goto rsnd_src_probe_done; + } src = rsnd_src_get(priv, i); diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 87e606f688d3..43c5e27dc5c8 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -1105,6 +1105,7 @@ void rsnd_parse_connect_ssi(struct rsnd_dai *rdai, struct device_node *capture) { struct rsnd_priv *priv = rsnd_rdai_to_priv(rdai); + struct device *dev = rsnd_priv_to_dev(priv); struct device_node *node; struct device_node *np; int i; @@ -1117,7 +1118,11 @@ void rsnd_parse_connect_ssi(struct rsnd_dai *rdai, for_each_child_of_node(node, np) { struct rsnd_mod *mod; - i = rsnd_node_fixed_index(np, SSI_NAME, i); + i = rsnd_node_fixed_index(dev, np, SSI_NAME, i); + if (i < 0) { + of_node_put(np); + break; + } mod = rsnd_ssi_mod_get(priv, i); @@ -1182,7 +1187,12 @@ int rsnd_ssi_probe(struct rsnd_priv *priv) if (!of_device_is_available(np)) goto skip; - i = rsnd_node_fixed_index(np, SSI_NAME, i); + i = rsnd_node_fixed_index(dev, np, SSI_NAME, i); + if (i < 0) { + ret = -EINVAL; + of_node_put(np); + goto rsnd_ssi_probe_done; + } ssi = rsnd_ssi_get(priv, i); diff --git a/sound/soc/sh/rcar/ssiu.c b/sound/soc/sh/rcar/ssiu.c index 138f95dd9f4a..4b8a63e336c7 100644 --- a/sound/soc/sh/rcar/ssiu.c +++ b/sound/soc/sh/rcar/ssiu.c @@ -462,6 +462,7 @@ void rsnd_parse_connect_ssiu(struct rsnd_dai *rdai, struct device_node *capture) { struct rsnd_priv *priv = rsnd_rdai_to_priv(rdai); + struct device *dev = rsnd_priv_to_dev(priv); struct device_node *node = rsnd_ssiu_of_node(priv); struct rsnd_dai_stream *io_p = &rdai->playback; struct rsnd_dai_stream *io_c = &rdai->capture; @@ -474,7 +475,11 @@ void rsnd_parse_connect_ssiu(struct rsnd_dai *rdai, for_each_child_of_node(node, np) { struct rsnd_mod *mod; - i = rsnd_node_fixed_index(np, SSIU_NAME, i); + i = rsnd_node_fixed_index(dev, np, SSIU_NAME, i); + if (i < 0) { + of_node_put(np); + break; + } mod = rsnd_ssiu_mod_get(priv, i); -- 2.35.1

1 0

[PATCH AUTOSEL 5.17 036/135] ASoC: rsnd: care default case on rsnd_ssiu_busif_err_status_clear()
by Sasha Levin 30 May '22

30 May '22

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> [ Upstream commit b1384d4c95088d01f4266237faabf165d3d605fc ] commit cfb7b8bf1e2d66 ("ASoC: rsnd: tidyup rsnd_ssiu_busif_err_status_clear()") merged duplicate code, but it didn't care about default case, and causes smatch warnings. smatch warnings: sound/soc/sh/rcar/ssiu.c:112 rsnd_ssiu_busif_err_status_clear() \ error: uninitialized symbol 'offset'. sound/soc/sh/rcar/ssiu.c:114 rsnd_ssiu_busif_err_status_clear() \ error: uninitialized symbol 'shift'. This patch cares it. Reported-by: kernel test robot <lkp(a)intel.com> Reported-by: Dan Carpenter <dan.carpenter(a)oracle.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Link: https://lore.kernel.org/r/87r15rgn6p.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/sh/rcar/ssiu.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/sh/rcar/ssiu.c b/sound/soc/sh/rcar/ssiu.c index 0d8f97633dd2..138f95dd9f4a 100644 --- a/sound/soc/sh/rcar/ssiu.c +++ b/sound/soc/sh/rcar/ssiu.c @@ -102,6 +102,8 @@ bool rsnd_ssiu_busif_err_status_clear(struct rsnd_mod *mod) shift = 1; offset = 1; break; + default: + goto out; } for (i = 0; i < 4; i++) { @@ -120,7 +122,7 @@ bool rsnd_ssiu_busif_err_status_clear(struct rsnd_mod *mod) } rsnd_mod_write(mod, reg, val); } - +out: return error; } -- 2.35.1

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[PATCH AUTOSEL 5.17 024/135] ALSA: jack: Access input_dev under mutex
by Sasha Levin 30 May '22

30 May '22

From: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> [ Upstream commit 1b6a6fc5280e97559287b61eade2d4b363e836f2 ] It is possible when using ASoC that input_dev is unregistered while calling snd_jack_report, which causes NULL pointer dereference. In order to prevent this serialize access to input_dev using mutex lock. Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> Reviewed-by: Cezary Rojewski <cezary.rojewski(a)intel.com> Link: https://lore.kernel.org/r/20220412091628.3056922-1-amadeuszx.slawinski@linu… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- include/sound/jack.h | 1 + sound/core/jack.c | 34 +++++++++++++++++++++++++++------- 2 files changed, 28 insertions(+), 7 deletions(-) diff --git a/include/sound/jack.h b/include/sound/jack.h index 1181f536557e..1ed90e2109e9 100644 --- a/include/sound/jack.h +++ b/include/sound/jack.h @@ -62,6 +62,7 @@ struct snd_jack { const char *id; #ifdef CONFIG_SND_JACK_INPUT_DEV struct input_dev *input_dev; + struct mutex input_dev_lock; int registered; int type; char name[100]; diff --git a/sound/core/jack.c b/sound/core/jack.c index d1e3055f2b6a..88493cc31914 100644 --- a/sound/core/jack.c +++ b/sound/core/jack.c @@ -42,8 +42,11 @@ static int snd_jack_dev_disconnect(struct snd_device *device) #ifdef CONFIG_SND_JACK_INPUT_DEV struct snd_jack *jack = device->device_data; - if (!jack->input_dev) + mutex_lock(&jack->input_dev_lock); + if (!jack->input_dev) { + mutex_unlock(&jack->input_dev_lock); return 0; + } /* If the input device is registered with the input subsystem * then we need to use a different deallocator. */ @@ -52,6 +55,7 @@ static int snd_jack_dev_disconnect(struct snd_device *device) else input_free_device(jack->input_dev); jack->input_dev = NULL; + mutex_unlock(&jack->input_dev_lock); #endif /* CONFIG_SND_JACK_INPUT_DEV */ return 0; } @@ -90,8 +94,11 @@ static int snd_jack_dev_register(struct snd_device *device) snprintf(jack->name, sizeof(jack->name), "%s %s", card->shortname, jack->id); - if (!jack->input_dev) + mutex_lock(&jack->input_dev_lock); + if (!jack->input_dev) { + mutex_unlock(&jack->input_dev_lock); return 0; + } jack->input_dev->name = jack->name; @@ -116,6 +123,7 @@ static int snd_jack_dev_register(struct snd_device *device) if (err == 0) jack->registered = 1; + mutex_unlock(&jack->input_dev_lock); return err; } #endif /* CONFIG_SND_JACK_INPUT_DEV */ @@ -517,9 +525,11 @@ int snd_jack_new(struct snd_card *card, const char *id, int type, return -ENOMEM; } - /* don't creat input device for phantom jack */ - if (!phantom_jack) { #ifdef CONFIG_SND_JACK_INPUT_DEV + mutex_init(&jack->input_dev_lock); + + /* don't create input device for phantom jack */ + if (!phantom_jack) { int i; jack->input_dev = input_allocate_device(); @@ -537,8 +547,8 @@ int snd_jack_new(struct snd_card *card, const char *id, int type, input_set_capability(jack->input_dev, EV_SW, jack_switch_types[i]); -#endif /* CONFIG_SND_JACK_INPUT_DEV */ } +#endif /* CONFIG_SND_JACK_INPUT_DEV */ err = snd_device_new(card, SNDRV_DEV_JACK, jack, &ops); if (err < 0) @@ -578,10 +588,14 @@ EXPORT_SYMBOL(snd_jack_new); void snd_jack_set_parent(struct snd_jack *jack, struct device *parent) { WARN_ON(jack->registered); - if (!jack->input_dev) + mutex_lock(&jack->input_dev_lock); + if (!jack->input_dev) { + mutex_unlock(&jack->input_dev_lock); return; + } jack->input_dev->dev.parent = parent; + mutex_unlock(&jack->input_dev_lock); } EXPORT_SYMBOL(snd_jack_set_parent); @@ -629,6 +643,8 @@ EXPORT_SYMBOL(snd_jack_set_key); /** * snd_jack_report - Report the current status of a jack + * Note: This function uses mutexes and should be called from a + * context which can sleep (such as a workqueue). * * @jack: The jack to report status for * @status: The current status of the jack @@ -654,8 +670,11 @@ void snd_jack_report(struct snd_jack *jack, int status) status & jack_kctl->mask_bits); #ifdef CONFIG_SND_JACK_INPUT_DEV - if (!jack->input_dev) + mutex_lock(&jack->input_dev_lock); + if (!jack->input_dev) { + mutex_unlock(&jack->input_dev_lock); return; + } for (i = 0; i < ARRAY_SIZE(jack->key); i++) { int testbit = ((SND_JACK_BTN_0 >> i) & ~mask_bits); @@ -675,6 +694,7 @@ void snd_jack_report(struct snd_jack *jack, int status) } input_sync(jack->input_dev); + mutex_unlock(&jack->input_dev_lock); #endif /* CONFIG_SND_JACK_INPUT_DEV */ } EXPORT_SYMBOL(snd_jack_report); -- 2.35.1

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[PATCH AUTOSEL 5.18 141/159] ASoC: rt1015p: remove dependency on GPIOLIB
by Sasha Levin 30 May '22

30 May '22

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit b390c25c6757b9d56cecdfbf6d55f15fc89a6386 ] commit dcc2c012c7691 ("ASoC: Fix gpiolib dependencies") removed a series of unnecessary dependencies on GPIOLIB when the gpio was optional. A similar simplification seems valid for rt1015p, so remove the dependency as well. This will avoid the following warning WARNING: unmet direct dependencies detected for SND_SOC_RT1015P Depends on [n]: SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && GPIOLIB [=n] Selected by [y]: - SND_SOC_INTEL_SOF_RT5682_MACH [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && SND_SOC_INTEL_MACH [=y] && (SND_SOC_SOF_HDA_LINK [=y] || SND_SOC_SOF_BAYTRAIL [=n]) && I2C [=y] && ACPI [=y] && (SND_HDA_CODEC_HDMI [=y] && SND_SOC_SOF_HDA_AUDIO_CODEC [=y] && (MFD_INTEL_LPSS [=y] || COMPILE_TEST [=y]) || SND_SOC_SOF_BAYTRAIL [=n] && (X86_INTEL_LPSS [=n] || COMPILE_TEST [=y])) Reported-by: kernel test robot <yujie.liu(a)intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Péter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Link: https://lore.kernel.org/r/20220517172647.468244-3-pierre-louis.bossart@linu… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/Kconfig | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 156f2519459d..3dea20b2c405 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -1212,7 +1212,6 @@ config SND_SOC_RT1015 config SND_SOC_RT1015P tristate - depends on GPIOLIB config SND_SOC_RT1019 tristate -- 2.35.1

1 0

[PATCH AUTOSEL 5.18 140/159] ASoC: max98357a: remove dependency on GPIOLIB
by Sasha Levin 30 May '22

30 May '22

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit 21ca3274333f5c1cbbf9d91e5b33f4f2463859b2 ] commit dcc2c012c7691 ("ASoC: Fix gpiolib dependencies") removed a series of unnecessary dependencies on GPIOLIB when the gpio was optional. A similar simplification seems valid for max98357a, so remove the dependency as well. This will avoid the following warning WARNING: unmet direct dependencies detected for SND_SOC_MAX98357A Depends on [n]: SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && GPIOLIB [=n] Selected by [y]: - SND_SOC_INTEL_SOF_CS42L42_MACH [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && SND_SOC_INTEL_MACH [=y] && (SND_SOC_SOF_HDA_LINK [=y] || SND_SOC_SOF_BAYTRAIL [=n]) && I2C [=y] && ACPI [=y] && SND_HDA_CODEC_HDMI [=y] && SND_SOC_SOF_HDA_AUDIO_CODEC [=y] && (MFD_INTEL_LPSS [=y] || COMPILE_TEST [=n]) Reported-by: kernel test robot <yujie.liu(a)intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Péter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Link: https://lore.kernel.org/r/20220517172647.468244-2-pierre-louis.bossart@linu… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/Kconfig | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index f46a22660103..156f2519459d 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -953,7 +953,6 @@ config SND_SOC_MAX98095 config SND_SOC_MAX98357A tristate "Maxim MAX98357A CODEC" - depends on GPIOLIB config SND_SOC_MAX98371 tristate -- 2.35.1

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[PATCH AUTOSEL 5.18 127/159] ASoC: rt5645: Fix errorenous cleanup order
by Sasha Levin 30 May '22

30 May '22

From: Lin Ma <linma(a)zju.edu.cn> [ Upstream commit 2def44d3aec59e38d2701c568d65540783f90f2f ] There is a logic error when removing rt5645 device as the function rt5645_i2c_remove() first cancel the &rt5645->jack_detect_work and delete the &rt5645->btn_check_timer latter. However, since the timer handler rt5645_btn_check_callback() will re-queue the jack_detect_work, this cleanup order is buggy. That is, once the del_timer_sync in rt5645_i2c_remove is concurrently run with the rt5645_btn_check_callback, the canceled jack_detect_work will be rescheduled again, leading to possible use-after-free. This patch fix the issue by placing the del_timer_sync function before the cancel_delayed_work_sync. Signed-off-by: Lin Ma <linma(a)zju.edu.cn> Link: https://lore.kernel.org/r/20220516092035.28283-1-linma@zju.edu.cn Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt5645.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 197c56047947..4b2e027c1033 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -4154,9 +4154,14 @@ static int rt5645_i2c_remove(struct i2c_client *i2c) if (i2c->irq) free_irq(i2c->irq, rt5645); + /* + * Since the rt5645_btn_check_callback() can queue jack_detect_work, + * the timer need to be delted first + */ + del_timer_sync(&rt5645->btn_check_timer); + cancel_delayed_work_sync(&rt5645->jack_detect_work); cancel_delayed_work_sync(&rt5645->rcclock_work); - del_timer_sync(&rt5645->btn_check_timer); regulator_bulk_disable(ARRAY_SIZE(rt5645->supplies), rt5645->supplies); -- 2.35.1

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[PATCH AUTOSEL 5.18 102/159] ASoC: SOF: amd: add missing platform_device_unregister in acp_pci_rn_probe
by Sasha Levin 30 May '22

30 May '22

From: Zheng Bin <zhengbin13(a)huawei.com> [ Upstream commit cbcab8cd737c74c20195c31d647e19f7cb49c9b8 ] acp_pci_rn_probe misses a call platform_device_unregister in error path, this patch fixes that. Signed-off-by: Zheng Bin <zhengbin13(a)huawei.com> Link: https://lore.kernel.org/r/20220512013728.4128903-1-zhengbin13@huawei.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/sof/amd/pci-rn.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/sof/amd/pci-rn.c b/sound/soc/sof/amd/pci-rn.c index 392ffbdf6417..d809d151a38c 100644 --- a/sound/soc/sof/amd/pci-rn.c +++ b/sound/soc/sof/amd/pci-rn.c @@ -93,6 +93,7 @@ static int acp_pci_rn_probe(struct pci_dev *pci, const struct pci_device_id *pci res = devm_kzalloc(&pci->dev, sizeof(struct resource) * ARRAY_SIZE(renoir_res), GFP_KERNEL); if (!res) { sof_pci_remove(pci); + platform_device_unregister(dmic_dev); return -ENOMEM; } -- 2.35.1

1 0

[PATCH AUTOSEL 5.18 091/159] ASoC: Intel: sof_ssp_amp: fix no DMIC BE Link on Chromebooks
by Sasha Levin 30 May '22

30 May '22

From: Brent Lu <brent.lu(a)intel.com> [ Upstream commit d1c808765deb2bcd35d827402ed4d75d068aae18 ] The SOF topology supports 2 BE Links(dmic01 and dmic16k) and each link supports up to four DMICs. However, Chromebook does not implement ACPI NHLT table so the mach->mach_params.dmic_num is always zero. We add a quirk so machine driver knows it's running on a Chromebook and need to create BE Links for DMIC. Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Signed-off-by: Brent Lu <brent.lu(a)intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20220509170922.54868-3-pierre-louis.bossart@linux… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/sof_ssp_amp.c | 16 ++++++++++++++-- 1 file changed, 14 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/boards/sof_ssp_amp.c b/sound/soc/intel/boards/sof_ssp_amp.c index 88530e9de543..ef70c6f27fe1 100644 --- a/sound/soc/intel/boards/sof_ssp_amp.c +++ b/sound/soc/intel/boards/sof_ssp_amp.c @@ -9,6 +9,7 @@ #include <linux/acpi.h> #include <linux/delay.h> +#include <linux/dmi.h> #include <linux/module.h> #include <linux/platform_device.h> #include <sound/core.h> @@ -78,6 +79,16 @@ struct sof_card_private { bool idisp_codec; }; +static const struct dmi_system_id chromebook_platforms[] = { + { + .ident = "Google Chromebooks", + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Google"), + } + }, + {}, +}; + static const struct snd_soc_dapm_widget sof_ssp_amp_dapm_widgets[] = { SND_SOC_DAPM_MIC("SoC DMIC", NULL), }; @@ -371,7 +382,7 @@ static int sof_ssp_amp_probe(struct platform_device *pdev) struct snd_soc_dai_link *dai_links; struct snd_soc_acpi_mach *mach; struct sof_card_private *ctx; - int dmic_be_num, hdmi_num = 0; + int dmic_be_num = 0, hdmi_num = 0; int ret, ssp_codec; ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL); @@ -383,7 +394,8 @@ static int sof_ssp_amp_probe(struct platform_device *pdev) mach = pdev->dev.platform_data; - dmic_be_num = mach->mach_params.dmic_num; + if (dmi_check_system(chromebook_platforms) || mach->mach_params.dmic_num > 0) + dmic_be_num = 2; ssp_codec = sof_ssp_amp_quirk & SOF_AMPLIFIER_SSP_MASK; -- 2.35.1

1 0

[PATCH AUTOSEL 5.18 079/159] ASoC: SOF: ipc3-topology: Correct get_control_data for non bytes payload
by Sasha Levin 30 May '22

30 May '22

From: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> [ Upstream commit a962890a5a3cce903ff7c7a19fadee63ed9efdc7 ] It is possible to craft a topology where sof_get_control_data() would do out of bounds access because it expects that it is only called when the payload is bytes type. Confusingly it also handles other types of controls, but the payload parsing implementation is only valid for bytes. Fix the code to count the non bytes controls and instead of storing a pointer to sof_abi_hdr in sof_widget_data (which is only valid for bytes), store the pointer to the data itself and add a new member to save the size of the data. In case of non bytes controls we store the pointer to the chanv itself, which is just an array of values at the end. In case of bytes control, drop the wrong cdata->data (wdata[i].pdata) check against NULL since it is incorrect and invalid in this context. The data is pointing to the end of cdata struct, so it should never be null. Reported-by: Sergey Senozhatsky <senozhatsky(a)chromium.org> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-by: Sergey Senozhatsky <senozhatsky(a)chromium.org> Tested-by: Sergey Senozhatsky <senozhatsky(a)chromium.org> Link: https://lore.kernel.org/r/20220427185221.28928-1-peter.ujfalusi@linux.intel… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/sof/ipc3-topology.c | 39 +++++++++++++++++++++++------------ 1 file changed, 26 insertions(+), 13 deletions(-) diff --git a/sound/soc/sof/ipc3-topology.c b/sound/soc/sof/ipc3-topology.c index 2f8450a8c0a1..af1bbd34213c 100644 --- a/sound/soc/sof/ipc3-topology.c +++ b/sound/soc/sof/ipc3-topology.c @@ -20,7 +20,8 @@ struct sof_widget_data { int ctrl_type; int ipc_cmd; - struct sof_abi_hdr *pdata; + void *pdata; + size_t pdata_size; struct snd_sof_control *control; }; @@ -784,16 +785,26 @@ static int sof_get_control_data(struct snd_soc_component *scomp, } cdata = wdata[i].control->ipc_control_data; - wdata[i].pdata = cdata->data; - if (!wdata[i].pdata) - return -EINVAL; - /* make sure data is valid - data can be updated at runtime */ - if (widget->dobj.widget.kcontrol_type[i] == SND_SOC_TPLG_TYPE_BYTES && - wdata[i].pdata->magic != SOF_ABI_MAGIC) - return -EINVAL; + if (widget->dobj.widget.kcontrol_type[i] == SND_SOC_TPLG_TYPE_BYTES) { + /* make sure data is valid - data can be updated at runtime */ + if (cdata->data->magic != SOF_ABI_MAGIC) + return -EINVAL; + + wdata[i].pdata = cdata->data->data; + wdata[i].pdata_size = cdata->data->size; + } else { + /* points to the control data union */ + wdata[i].pdata = cdata->chanv; + /* + * wdata[i].control->size is calculated with struct_size + * and includes the size of struct sof_ipc_ctrl_data + */ + wdata[i].pdata_size = wdata[i].control->size - + sizeof(struct sof_ipc_ctrl_data); + } - *size += wdata[i].pdata->size; + *size += wdata[i].pdata_size; /* get data type */ switch (cdata->cmd) { @@ -876,10 +887,12 @@ static int sof_process_load(struct snd_soc_component *scomp, */ if (ipc_data_size) { for (i = 0; i < widget->num_kcontrols; i++) { - memcpy(&process->data[offset], - wdata[i].pdata->data, - wdata[i].pdata->size); - offset += wdata[i].pdata->size; + if (!wdata[i].pdata_size) + continue; + + memcpy(&process->data[offset], wdata[i].pdata, + wdata[i].pdata_size); + offset += wdata[i].pdata_size; } } -- 2.35.1

1 0

[PATCH AUTOSEL 5.18 078/159] ASoC: dapm: Don't fold register value changes into notifications
by Sasha Levin 30 May '22

30 May '22

From: Mark Brown <broonie(a)kernel.org> [ Upstream commit ad685980469b9f9b99d4d6ea05f4cb8f57cb2234 ] DAPM tracks and reports the value presented to the user from DAPM controls separately to the register value, these may diverge during initialisation or when an autodisable control is in use. When writing DAPM controls we currently report that a change has occurred if either the DAPM value or the value stored in the register has changed, meaning that if the two are out of sync we may appear to report a spurious event to userspace. Since we use this folded in value for nothing other than the value reported to userspace simply drop the folding in of the register change. Signed-off-by: Mark Brown <broonie(a)kernel.org> Link: https://lore.kernel.org/r/20220428161833.3690050-1-broonie@kernel.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/soc-dapm.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index ca917a849c42..869c76506b66 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3437,7 +3437,6 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, update.val = val; card->update = &update; } - change |= reg_change; ret = soc_dapm_mixer_update_power(card, kcontrol, connect, rconnect); @@ -3539,7 +3538,6 @@ int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol, update.val = val; card->update = &update; } - change |= reg_change; ret = soc_dapm_mux_update_power(card, kcontrol, item[0], e); -- 2.35.1

1 0

[PATCH AUTOSEL 5.18 064/159] ASoC: Intel: bytcr_rt5640: Add quirk for the HP Pro Tablet 408
by Sasha Levin 30 May '22

30 May '22

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit ce216cfa84a4e1c23b105e652c550bdeaac9e922 ] Add a quirk for the HP Pro Tablet 408, this BYTCR tablet has no CHAN package in its ACPI tables and uses SSP0-AIF1 rather then SSP0-AIF2 which is the default for BYTCR devices. It also uses DMIC1 for the internal mic rather then the default IN3 and it uses JD2 rather then the default JD1 for jack-detect. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=211485 Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20220427134918.527381-1-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/bytcr_rt5640.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index d76a505052fb..f81ae742faa7 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -773,6 +773,18 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { BYT_RT5640_OVCD_SF_0P75 | BYT_RT5640_MCLK_EN), }, + { /* HP Pro Tablet 408 */ + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Hewlett-Packard"), + DMI_MATCH(DMI_PRODUCT_NAME, "HP Pro Tablet 408"), + }, + .driver_data = (void *)(BYT_RT5640_DMIC1_MAP | + BYT_RT5640_JD_SRC_JD2_IN4N | + BYT_RT5640_OVCD_TH_1500UA | + BYT_RT5640_OVCD_SF_0P75 | + BYT_RT5640_SSP0_AIF1 | + BYT_RT5640_MCLK_EN), + }, { /* HP Stream 7 */ .matches = { DMI_EXACT_MATCH(DMI_SYS_VENDOR, "Hewlett-Packard"), -- 2.35.1

1 0

[PATCH AUTOSEL 5.18 045/159] ASoC: rsnd: care return value from rsnd_node_fixed_index()
by Sasha Levin 30 May '22

30 May '22

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> [ Upstream commit d09a7db431c65aaa8303eb456439d1831ca2e6b4 ] Renesas Sound is very complex, and thus it needs to use rsnd_node_fixed_index() to know enabled pin index. It returns error if strange pin was selected, but some codes didn't check it. This patch 1) indicates error message, 2) check return value. Reported-by: kernel test robot <lkp(a)intel.com> Reported-by: Dan Carpenter <dan.carpenter(a)oracle.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Link: https://lore.kernel.org/r/87pmlbgn5t.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/sh/rcar/core.c | 15 ++++++++++----- sound/soc/sh/rcar/dma.c | 9 ++++++++- sound/soc/sh/rcar/rsnd.h | 2 +- sound/soc/sh/rcar/src.c | 7 ++++++- sound/soc/sh/rcar/ssi.c | 14 ++++++++++++-- sound/soc/sh/rcar/ssiu.c | 7 ++++++- 6 files changed, 43 insertions(+), 11 deletions(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 6a8fe0da7670..af8ef2a27d34 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -1159,6 +1159,7 @@ void rsnd_parse_connect_common(struct rsnd_dai *rdai, char *name, struct device_node *capture) { struct rsnd_priv *priv = rsnd_rdai_to_priv(rdai); + struct device *dev = rsnd_priv_to_dev(priv); struct device_node *np; int i; @@ -1169,7 +1170,11 @@ void rsnd_parse_connect_common(struct rsnd_dai *rdai, char *name, for_each_child_of_node(node, np) { struct rsnd_mod *mod; - i = rsnd_node_fixed_index(np, name, i); + i = rsnd_node_fixed_index(dev, np, name, i); + if (i < 0) { + of_node_put(np); + break; + } mod = mod_get(priv, i); @@ -1183,7 +1188,7 @@ void rsnd_parse_connect_common(struct rsnd_dai *rdai, char *name, of_node_put(node); } -int rsnd_node_fixed_index(struct device_node *node, char *name, int idx) +int rsnd_node_fixed_index(struct device *dev, struct device_node *node, char *name, int idx) { char node_name[16]; @@ -1210,6 +1215,8 @@ int rsnd_node_fixed_index(struct device_node *node, char *name, int idx) return idx; } + dev_err(dev, "strange node numbering (%s)", + of_node_full_name(node)); return -EINVAL; } @@ -1221,10 +1228,8 @@ int rsnd_node_count(struct rsnd_priv *priv, struct device_node *node, char *name i = 0; for_each_child_of_node(node, np) { - i = rsnd_node_fixed_index(np, name, i); + i = rsnd_node_fixed_index(dev, np, name, i); if (i < 0) { - dev_err(dev, "strange node numbering (%s)", - of_node_full_name(node)); of_node_put(np); return 0; } diff --git a/sound/soc/sh/rcar/dma.c b/sound/soc/sh/rcar/dma.c index 03e0d4eca781..463ab237d7bd 100644 --- a/sound/soc/sh/rcar/dma.c +++ b/sound/soc/sh/rcar/dma.c @@ -240,12 +240,19 @@ static int rsnd_dmaen_start(struct rsnd_mod *mod, struct dma_chan *rsnd_dma_request_channel(struct device_node *of_node, char *name, struct rsnd_mod *mod, char *x) { + struct rsnd_priv *priv = rsnd_mod_to_priv(mod); + struct device *dev = rsnd_priv_to_dev(priv); struct dma_chan *chan = NULL; struct device_node *np; int i = 0; for_each_child_of_node(of_node, np) { - i = rsnd_node_fixed_index(np, name, i); + i = rsnd_node_fixed_index(dev, np, name, i); + if (i < 0) { + chan = NULL; + of_node_put(np); + break; + } if (i == rsnd_mod_id_raw(mod) && (!chan)) chan = of_dma_request_slave_channel(np, x); diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index 6580bab0e229..d9cd190d7e19 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -460,7 +460,7 @@ void rsnd_parse_connect_common(struct rsnd_dai *rdai, char *name, struct device_node *playback, struct device_node *capture); int rsnd_node_count(struct rsnd_priv *priv, struct device_node *node, char *name); -int rsnd_node_fixed_index(struct device_node *node, char *name, int idx); +int rsnd_node_fixed_index(struct device *dev, struct device_node *node, char *name, int idx); int rsnd_channel_normalization(int chan); #define rsnd_runtime_channel_original(io) \ diff --git a/sound/soc/sh/rcar/src.c b/sound/soc/sh/rcar/src.c index 42a100c6303d..0ea84ae57c6a 100644 --- a/sound/soc/sh/rcar/src.c +++ b/sound/soc/sh/rcar/src.c @@ -676,7 +676,12 @@ int rsnd_src_probe(struct rsnd_priv *priv) if (!of_device_is_available(np)) goto skip; - i = rsnd_node_fixed_index(np, SRC_NAME, i); + i = rsnd_node_fixed_index(dev, np, SRC_NAME, i); + if (i < 0) { + ret = -EINVAL; + of_node_put(np); + goto rsnd_src_probe_done; + } src = rsnd_src_get(priv, i); diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 87e606f688d3..43c5e27dc5c8 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -1105,6 +1105,7 @@ void rsnd_parse_connect_ssi(struct rsnd_dai *rdai, struct device_node *capture) { struct rsnd_priv *priv = rsnd_rdai_to_priv(rdai); + struct device *dev = rsnd_priv_to_dev(priv); struct device_node *node; struct device_node *np; int i; @@ -1117,7 +1118,11 @@ void rsnd_parse_connect_ssi(struct rsnd_dai *rdai, for_each_child_of_node(node, np) { struct rsnd_mod *mod; - i = rsnd_node_fixed_index(np, SSI_NAME, i); + i = rsnd_node_fixed_index(dev, np, SSI_NAME, i); + if (i < 0) { + of_node_put(np); + break; + } mod = rsnd_ssi_mod_get(priv, i); @@ -1182,7 +1187,12 @@ int rsnd_ssi_probe(struct rsnd_priv *priv) if (!of_device_is_available(np)) goto skip; - i = rsnd_node_fixed_index(np, SSI_NAME, i); + i = rsnd_node_fixed_index(dev, np, SSI_NAME, i); + if (i < 0) { + ret = -EINVAL; + of_node_put(np); + goto rsnd_ssi_probe_done; + } ssi = rsnd_ssi_get(priv, i); diff --git a/sound/soc/sh/rcar/ssiu.c b/sound/soc/sh/rcar/ssiu.c index 138f95dd9f4a..4b8a63e336c7 100644 --- a/sound/soc/sh/rcar/ssiu.c +++ b/sound/soc/sh/rcar/ssiu.c @@ -462,6 +462,7 @@ void rsnd_parse_connect_ssiu(struct rsnd_dai *rdai, struct device_node *capture) { struct rsnd_priv *priv = rsnd_rdai_to_priv(rdai); + struct device *dev = rsnd_priv_to_dev(priv); struct device_node *node = rsnd_ssiu_of_node(priv); struct rsnd_dai_stream *io_p = &rdai->playback; struct rsnd_dai_stream *io_c = &rdai->capture; @@ -474,7 +475,11 @@ void rsnd_parse_connect_ssiu(struct rsnd_dai *rdai, for_each_child_of_node(node, np) { struct rsnd_mod *mod; - i = rsnd_node_fixed_index(np, SSIU_NAME, i); + i = rsnd_node_fixed_index(dev, np, SSIU_NAME, i); + if (i < 0) { + of_node_put(np); + break; + } mod = rsnd_ssiu_mod_get(priv, i); -- 2.35.1

1 0

[PATCH AUTOSEL 5.18 044/159] ASoC: rsnd: care default case on rsnd_ssiu_busif_err_status_clear()
by Sasha Levin 30 May '22

30 May '22

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> [ Upstream commit b1384d4c95088d01f4266237faabf165d3d605fc ] commit cfb7b8bf1e2d66 ("ASoC: rsnd: tidyup rsnd_ssiu_busif_err_status_clear()") merged duplicate code, but it didn't care about default case, and causes smatch warnings. smatch warnings: sound/soc/sh/rcar/ssiu.c:112 rsnd_ssiu_busif_err_status_clear() \ error: uninitialized symbol 'offset'. sound/soc/sh/rcar/ssiu.c:114 rsnd_ssiu_busif_err_status_clear() \ error: uninitialized symbol 'shift'. This patch cares it. Reported-by: kernel test robot <lkp(a)intel.com> Reported-by: Dan Carpenter <dan.carpenter(a)oracle.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Link: https://lore.kernel.org/r/87r15rgn6p.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/sh/rcar/ssiu.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/sh/rcar/ssiu.c b/sound/soc/sh/rcar/ssiu.c index 0d8f97633dd2..138f95dd9f4a 100644 --- a/sound/soc/sh/rcar/ssiu.c +++ b/sound/soc/sh/rcar/ssiu.c @@ -102,6 +102,8 @@ bool rsnd_ssiu_busif_err_status_clear(struct rsnd_mod *mod) shift = 1; offset = 1; break; + default: + goto out; } for (i = 0; i < 4; i++) { @@ -120,7 +122,7 @@ bool rsnd_ssiu_busif_err_status_clear(struct rsnd_mod *mod) } rsnd_mod_write(mod, reg, val); } - +out: return error; } -- 2.35.1

1 0

[PATCH AUTOSEL 5.18 028/159] ALSA: jack: Access input_dev under mutex
by Sasha Levin 30 May '22

30 May '22

From: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> [ Upstream commit 1b6a6fc5280e97559287b61eade2d4b363e836f2 ] It is possible when using ASoC that input_dev is unregistered while calling snd_jack_report, which causes NULL pointer dereference. In order to prevent this serialize access to input_dev using mutex lock. Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> Reviewed-by: Cezary Rojewski <cezary.rojewski(a)intel.com> Link: https://lore.kernel.org/r/20220412091628.3056922-1-amadeuszx.slawinski@linu… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- include/sound/jack.h | 1 + sound/core/jack.c | 34 +++++++++++++++++++++++++++------- 2 files changed, 28 insertions(+), 7 deletions(-) diff --git a/include/sound/jack.h b/include/sound/jack.h index 1181f536557e..1ed90e2109e9 100644 --- a/include/sound/jack.h +++ b/include/sound/jack.h @@ -62,6 +62,7 @@ struct snd_jack { const char *id; #ifdef CONFIG_SND_JACK_INPUT_DEV struct input_dev *input_dev; + struct mutex input_dev_lock; int registered; int type; char name[100]; diff --git a/sound/core/jack.c b/sound/core/jack.c index d1e3055f2b6a..88493cc31914 100644 --- a/sound/core/jack.c +++ b/sound/core/jack.c @@ -42,8 +42,11 @@ static int snd_jack_dev_disconnect(struct snd_device *device) #ifdef CONFIG_SND_JACK_INPUT_DEV struct snd_jack *jack = device->device_data; - if (!jack->input_dev) + mutex_lock(&jack->input_dev_lock); + if (!jack->input_dev) { + mutex_unlock(&jack->input_dev_lock); return 0; + } /* If the input device is registered with the input subsystem * then we need to use a different deallocator. */ @@ -52,6 +55,7 @@ static int snd_jack_dev_disconnect(struct snd_device *device) else input_free_device(jack->input_dev); jack->input_dev = NULL; + mutex_unlock(&jack->input_dev_lock); #endif /* CONFIG_SND_JACK_INPUT_DEV */ return 0; } @@ -90,8 +94,11 @@ static int snd_jack_dev_register(struct snd_device *device) snprintf(jack->name, sizeof(jack->name), "%s %s", card->shortname, jack->id); - if (!jack->input_dev) + mutex_lock(&jack->input_dev_lock); + if (!jack->input_dev) { + mutex_unlock(&jack->input_dev_lock); return 0; + } jack->input_dev->name = jack->name; @@ -116,6 +123,7 @@ static int snd_jack_dev_register(struct snd_device *device) if (err == 0) jack->registered = 1; + mutex_unlock(&jack->input_dev_lock); return err; } #endif /* CONFIG_SND_JACK_INPUT_DEV */ @@ -517,9 +525,11 @@ int snd_jack_new(struct snd_card *card, const char *id, int type, return -ENOMEM; } - /* don't creat input device for phantom jack */ - if (!phantom_jack) { #ifdef CONFIG_SND_JACK_INPUT_DEV + mutex_init(&jack->input_dev_lock); + + /* don't create input device for phantom jack */ + if (!phantom_jack) { int i; jack->input_dev = input_allocate_device(); @@ -537,8 +547,8 @@ int snd_jack_new(struct snd_card *card, const char *id, int type, input_set_capability(jack->input_dev, EV_SW, jack_switch_types[i]); -#endif /* CONFIG_SND_JACK_INPUT_DEV */ } +#endif /* CONFIG_SND_JACK_INPUT_DEV */ err = snd_device_new(card, SNDRV_DEV_JACK, jack, &ops); if (err < 0) @@ -578,10 +588,14 @@ EXPORT_SYMBOL(snd_jack_new); void snd_jack_set_parent(struct snd_jack *jack, struct device *parent) { WARN_ON(jack->registered); - if (!jack->input_dev) + mutex_lock(&jack->input_dev_lock); + if (!jack->input_dev) { + mutex_unlock(&jack->input_dev_lock); return; + } jack->input_dev->dev.parent = parent; + mutex_unlock(&jack->input_dev_lock); } EXPORT_SYMBOL(snd_jack_set_parent); @@ -629,6 +643,8 @@ EXPORT_SYMBOL(snd_jack_set_key); /** * snd_jack_report - Report the current status of a jack + * Note: This function uses mutexes and should be called from a + * context which can sleep (such as a workqueue). * * @jack: The jack to report status for * @status: The current status of the jack @@ -654,8 +670,11 @@ void snd_jack_report(struct snd_jack *jack, int status) status & jack_kctl->mask_bits); #ifdef CONFIG_SND_JACK_INPUT_DEV - if (!jack->input_dev) + mutex_lock(&jack->input_dev_lock); + if (!jack->input_dev) { + mutex_unlock(&jack->input_dev_lock); return; + } for (i = 0; i < ARRAY_SIZE(jack->key); i++) { int testbit = ((SND_JACK_BTN_0 >> i) & ~mask_bits); @@ -675,6 +694,7 @@ void snd_jack_report(struct snd_jack *jack, int status) } input_sync(jack->input_dev); + mutex_unlock(&jack->input_dev_lock); #endif /* CONFIG_SND_JACK_INPUT_DEV */ } EXPORT_SYMBOL(snd_jack_report); -- 2.35.1

1 0

[PATCH] ALSA: hda/realtek - Fix microphone noise on ASUS TUF B550M-PLUS
by Marios Levogiannis 30 May '22

30 May '22

Set microphone pins 0x18 (rear) and 0x19 (front) to VREF_50 to fix the microphone noise on ASUS TUF B550M-PLUS which uses the ALCS1200A codec. The initial value was VREF_80. The same issue is also present on Windows using both the default Windows driver and all tested Realtek drivers before version 6.0.9049.1. Comparing Realtek driver 6.0.9049.1 (the first one without the microphone noise) to Realtek driver 6.0.9047.1 (the last one with the microphone noise) revealed that the fix is the result of setting pins 0x18 and 0x19 to VREF_50. This fix may also work for other boards that have been reported to have the same microphone issue and use the ALC1150 and ALCS1200A codecs, since these codecs are similar and the fix in the Realtek driver on Windows is common for both. However, it is currently enabled only for ASUS TUF B550M-PLUS as this is the only board that could be tested. --- sound/pci/hda/patch_realtek.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 89d610afa..f3ad454b3 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1981,6 +1981,7 @@ enum { ALC1220_FIXUP_CLEVO_PB51ED_PINS, ALC887_FIXUP_ASUS_AUDIO, ALC887_FIXUP_ASUS_HMIC, + ALCS1200A_FIXUP_MIC_VREF, }; static void alc889_fixup_coef(struct hda_codec *codec, @@ -2526,6 +2527,14 @@ static const struct hda_fixup alc882_fixups[] = { .chained = true, .chain_id = ALC887_FIXUP_ASUS_AUDIO, }, + [ALCS1200A_FIXUP_MIC_VREF] = { + .type = HDA_FIXUP_PINCTLS, + .v.pins = (const struct hda_pintbl[]) { + { 0x18, PIN_VREF50 }, /* rear mic */ + { 0x19, PIN_VREF50 }, /* front mic */ + {} + } + }, }; static const struct snd_pci_quirk alc882_fixup_tbl[] = { @@ -2563,6 +2572,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x835f, "Asus Eee 1601", ALC888_FIXUP_EEE1601), SND_PCI_QUIRK(0x1043, 0x84bc, "ASUS ET2700", ALC887_FIXUP_ASUS_BASS), SND_PCI_QUIRK(0x1043, 0x8691, "ASUS ROG Ranger VIII", ALC882_FIXUP_GPIO3), + SND_PCI_QUIRK(0x1043, 0x8797, "ASUS TUF B550M-PLUS", ALCS1200A_FIXUP_MIC_VREF), SND_PCI_QUIRK(0x104d, 0x9043, "Sony Vaio VGC-LN51JGB", ALC882_FIXUP_NO_PRIMARY_HP), SND_PCI_QUIRK(0x104d, 0x9044, "Sony VAIO AiO", ALC882_FIXUP_NO_PRIMARY_HP), SND_PCI_QUIRK(0x104d, 0x9047, "Sony Vaio TT", ALC889_FIXUP_VAIO_TT), -- 2.36.1

2 3

[PATCH 00/15] rewrite public API according to convention of GNOME project
by GitHub pull_request - opened 30 May '22

30 May '22

alsa-project/alsa-gobject pull request #70 was opened from takaswie: In GNOME convention for throw function which has an argument of GError to report failure, the function is programmed to return value of gboolean type to report whether the argument is filled with instance of GError. On the other hand, I wrote the throw functions to return nothing (void). This patchset rewrite public API according to it. Unfortunately, it brings loss of backward compatibility. ``` Takashi Sakamoto (15): ctl: remove useless exposed symbols timer: user-instance: fix closure marshaller for event signal ctl: query: rewrite public API to return gboolean according to GNOME convention ctl: card: rewrite public API to return gboolean according to GNOME convention ctl: elem_info: rewrite public API to return gboolean according to GNOME convention rawmidi: query: rewrite public API to return gboolean according to GNOME convention rawmidi: stream_pair: rewrite public API to return gboolean according to GNOME convention hwdep: query: rewrite public API to return gboolean according to GNOME convention timer: query: rewrite public API to return gboolean according to GNOME convention timer: user_instance: rewrite public API to return gboolean according to GNOME convention timer: instance-params: rewrite public API to return gboolean according to GNOME convention seq: query: rewrite public API to return gboolean according to GNOME convention seq: user_client: rewrite public API to return gboolean according to GNOME convention seq: client_info: rewrite public API to return gboolean according to GNOME convention seq: event_cntr: rewrite public API to return gboolean according to GNOME convention samples/ctl | 21 +- samples/hwdep | 8 +- samples/rawmidi | 14 +- samples/seq | 37 +- samples/timer | 9 +- src/ctl/alsactl.map | 73 ++-- src/ctl/card.c | 322 +++++++++------- src/ctl/card.h | 82 ++--- src/ctl/elem-info.c | 84 +++-- src/ctl/elem-info.h | 12 +- src/ctl/query.c | 59 ++- src/ctl/query.h | 8 +- src/hwdep/alsahwdep.map | 13 +- src/hwdep/query.c | 64 ++-- src/hwdep/query.h | 17 +- src/rawmidi/alsarawmidi.map | 37 +- src/rawmidi/privates.h | 2 +- src/rawmidi/query.c | 96 +++-- src/rawmidi/query.h | 37 +- src/rawmidi/stream-pair.c | 163 +++++---- src/rawmidi/stream-pair.h | 31 +- src/seq/alsaseq.map | 98 ++--- src/seq/client-info.c | 38 +- src/seq/client-info.h | 16 +- src/seq/event-cntr.c | 711 +++++++++++++++++++++--------------- src/seq/event-cntr.h | 203 +++++----- src/seq/query.c | 298 +++++++++------ src/seq/query.h | 46 +-- src/seq/user-client.c | 413 +++++++++++++-------- src/seq/user-client.h | 140 +++---- src/timer/alsatimer.map | 44 ++- src/timer/instance-params.c | 30 +- src/timer/instance-params.h | 4 +- src/timer/query.c | 163 ++++++--- src/timer/query.h | 23 +- src/timer/user-instance.c | 192 ++++++---- src/timer/user-instance.h | 43 +-- 37 files changed, 2117 insertions(+), 1534 deletions(-) ``` Request URL : https://github.com/alsa-project/alsa-gobject/pull/70 Patch URL : https://github.com/alsa-project/alsa-gobject/pull/70.patch Repository URL: https://github.com/alsa-project/alsa-gobject

1 0

[PATCH v2 0/6] OPE support on Tegra210 and later
by Sameer Pujar 30 May '22

30 May '22

This series adds support for Output Prcoessing Module (OPE) module on Tegra210 and later generations of SoCs. OPE is a client of AHUB and it has sub blocks of PEQ (Parametric Equalizer) and MBDRC (Multi Band Dynamic Range Compressor) for data processing. An ASoC component is registered for OPE, which includes PEQ and MBDRC functions as well. This can be plugged in audio path using ALSA mixer controls. The series adds necessary binding documentaion, driver and DT binding patches to enable OPE module on Jetson platforms. Changelog ========= v1 -> v2: --------- * Use generic node names for OPE, PEQ and MBDRC devices. Update binding doc and DT patches for this. * Remove redundant nodename rule enforcement from OPE, PEQ and MBDRC nodes. Update binding doc patch for this. * Fix spaces before binding doc examples and remove '|' from binding doc descriptions. Sameer Pujar (6): ASoC: tegra: Add binding doc for OPE module ASoC: tegra: Add Tegra210 based OPE driver ASoC: tegra: AHUB routes for OPE module arm64: defconfig: Build Tegra OPE module arm64: tegra: Add OPE device on Tegra210 and later arm64: tegra: Enable OPE on various platforms .../bindings/sound/nvidia,tegra210-ahub.yaml | 4 + .../bindings/sound/nvidia,tegra210-mbdrc.yaml | 47 + .../bindings/sound/nvidia,tegra210-ope.yaml | 87 ++ .../bindings/sound/nvidia,tegra210-peq.yaml | 48 + arch/arm64/boot/dts/nvidia/tegra186-p2771-0000.dts | 43 + arch/arm64/boot/dts/nvidia/tegra186.dtsi | 23 + arch/arm64/boot/dts/nvidia/tegra194-p2972-0000.dts | 43 + .../arm64/boot/dts/nvidia/tegra194-p3509-0000.dtsi | 43 + arch/arm64/boot/dts/nvidia/tegra194.dtsi | 23 + arch/arm64/boot/dts/nvidia/tegra210-p2371-2180.dts | 84 ++ arch/arm64/boot/dts/nvidia/tegra210-p3450-0000.dts | 84 ++ arch/arm64/boot/dts/nvidia/tegra210.dtsi | 40 + .../dts/nvidia/tegra234-p3737-0000+p3701-0000.dts | 43 + arch/arm64/boot/dts/nvidia/tegra234.dtsi | 23 + arch/arm64/configs/defconfig | 1 + sound/soc/tegra/Kconfig | 9 + sound/soc/tegra/Makefile | 2 + sound/soc/tegra/tegra210_ahub.c | 39 +- sound/soc/tegra/tegra210_mbdrc.c | 1012 ++++++++++++++++++++ sound/soc/tegra/tegra210_mbdrc.h | 215 +++++ sound/soc/tegra/tegra210_ope.c | 419 ++++++++ sound/soc/tegra/tegra210_ope.h | 90 ++ sound/soc/tegra/tegra210_peq.c | 434 +++++++++ sound/soc/tegra/tegra210_peq.h | 56 ++ 24 files changed, 2908 insertions(+), 4 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-mbdrc.yaml create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-ope.yaml create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-peq.yaml create mode 100644 sound/soc/tegra/tegra210_mbdrc.c create mode 100644 sound/soc/tegra/tegra210_mbdrc.h create mode 100644 sound/soc/tegra/tegra210_ope.c create mode 100644 sound/soc/tegra/tegra210_ope.h create mode 100644 sound/soc/tegra/tegra210_peq.c create mode 100644 sound/soc/tegra/tegra210_peq.h -- 2.7.4

2 9

ucm2 - splitting 10 channel interface into mix of mono and stereo channels
by GitHub issues - opened 29 May '22

29 May '22

alsa-project/alsa-ucm-conf issue #161 was opened from smangels: Hi, I've pushed a PR for the Behringer Flow8 digital mixer configured to be in "streaming" mode. Now that I've managed to wrap my head around the configuration, I would like to continue with the so called "recording" mode (different ID in `lsusb`). Thus I wonder whether UCM2 syntax allows me to split a 10 channel interface into a mix of mono and stereo channels (mic - mono, master, stereo)? Regards, Sebastian Mangelsen Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/161 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH v2] ASoC: dt-bindings: mediatek: mt8192: Add i2s-share properties
by Nícolas F. R. A. Prado 29 May '22

29 May '22

The Mediatek AFE PCM controller for MT8192 allows two I2S interfaces to share the same clock and act as a single interface with both input and output. Add patterns for these properties in the dt-binding. The property is split into two patterns in order to allow all valid interface pairings. Signed-off-by: Nícolas F. R. A. Prado <nfraprado(a)collabora.com> --- The series from v1 of this patch was merged although some changes were still needed in this patch, so the v1 of this patch was reverted [1] and this standalone commit addresses the feedback from v1 and readds the property. [1] https://lore.kernel.org/all/20220509185625.580811-1-nfraprado@collabora.com v1: https://lore.kernel.org/all/20220429203039.2207848-2-nfraprado@collabora.co… Changes in v2: - Added "mediatek," prefix to property - Rewrote and added more information to property description - Split into two patterns to validate that output-input pairings are done .../bindings/sound/mt8192-afe-pcm.yaml | 16 ++++++++++++++++ 1 file changed, 16 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/mt8192-afe-pcm.yaml b/Documentation/devicetree/bindings/sound/mt8192-afe-pcm.yaml index 7a25bc9b8060..2abf43c6c2c3 100644 --- a/Documentation/devicetree/bindings/sound/mt8192-afe-pcm.yaml +++ b/Documentation/devicetree/bindings/sound/mt8192-afe-pcm.yaml @@ -54,6 +54,22 @@ properties: - const: aud_infra_clk - const: aud_infra_26m_clk +patternProperties: + "^mediatek,i2s[13579]-share$": + description: + Each I2S interface has a single data line, input if its index is even or + output if the index is odd. An input and an output I2S interface can be + used together as if they were a single I2S interface with both input and + output data lines by sharing the same clock. This property represents this + pairing. The value should be the name of the interface whose clock is + used, and the property name the other interface that depends on this + clock. + pattern: "^I2S[0268]$" + + "^mediatek,i2s[0268]-share$": + description: Same as above. + pattern: "^I2S[13579]$" + required: - compatible - interrupts -- 2.36.0

3 5

ucm2 profile for Behringer Flow8
by GitHub pull_request - edited 28 May '22

28 May '22

alsa-project/alsa-ucm-conf pull request #160 was edited from smangels: - kudos to Pekka Oinas for supporting me finding the right settings - provides an UCM2 profile for Behringer's digital mixer Flow8 configured into "streaming" mode - 2 stereo outputs, 1 stereo input, when vendor ID product ID are `1397:050d` - prepare filenames for Flow8's recording mode `1397:050c` (10 chan input, 4 chan output) Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/160 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/160.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

ucm2 profile for Behringer Flow8
by GitHub pull_request - edited 28 May '22

28 May '22

alsa-project/alsa-ucm-conf pull request #160 was edited from smangels: - kudos to Pekka Oinas for supporting me finding the right settings - provides an UCM2 profile for Behringer's digital mixer Flow8 configured into "streaming" mode - 2 stereo outputs, 1 stereo input - prepare filenames for Flow8's recording mode (10 chan input, 4 chan output) Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/160 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/160.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

ucm2 profile for Behringer Flow8
by GitHub pull_request - opened 28 May '22

28 May '22

alsa-project/alsa-ucm-conf pull request #160 was opened from smangels: - kudos to Pekka Oinas for supporting me finding the right settings - provides an UCM2 profile for Behringer's digital mixer Flow configured into "streaming" mode - 2 stereo outputs, 1 stereo input - prepare filenames for Flow8's recording mode (10 chan input, 4 chan output) Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/160 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/160.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

ucm2 profile for Behringer Flow8
by GitHub pull_request - opened 28 May '22

28 May '22

alsa-project/alsa-ucm-conf pull request #159 was opened from smangels: - kudos to Pekka Oinas for supporting me finding the right settings - provides an UCM2 profile for Behringer's digital mixer Flow configured into "streaming" mode - 2 stereo outputs, 1 stereo input Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/159 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/159.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH] ALSA: hda/via: Delete does not require return
by Xiang wangx 27 May '22

27 May '22

Void function return statements are not generally useful. Signed-off-by: Xiang wangx <wangxiang(a)cdjrlc.com> --- sound/pci/hda/patch_via.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 773a136161f1..a05304f340df 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -449,8 +449,6 @@ static void vt1708_set_pinconfig_connect(struct hda_codec *codec, hda_nid_t nid) def_conf = def_conf & (~(AC_JACK_PORT_BOTH << 30)); snd_hda_codec_set_pincfg(codec, nid, def_conf); } - - return; } static int vt1708_jack_detect_get(struct snd_kcontrol *kcontrol, -- 2.36.1

2 1

[Patch] ALSA: hda/realtek: Enable 4-speaker output for Dell XPS 15 9520 laptop
by Rik van der Kemp 27 May '22

27 May '22

The 2022-model XPS 15 appears to use the same 4-speakers-on-ALC289 audio setup as the Dell XPS 15 9510, so requires the same quirk to enable woofer output. Tested on my own 9520. Signed-off-by: Rik van der Kemp <rik(a)upto11.nl> --- 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 80e4955e8c10..e64597f2675b 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8902,6 +8902,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0a38, "Dell Latitude 7520", ALC269_FIXUP_DELL4_MIC_NO_PRESENCE_QUIET), SND_PCI_QUIRK(0x1028, 0x0a58, "Dell", ALC255_FIXUP_DELL_HEADSET_MIC), SND_PCI_QUIRK(0x1028, 0x0a61, "Dell XPS 15 9510", ALC289_FIXUP_DUAL_SPK), + SND_PCI_QUIRK(0x1028, 0x0b19, "Dell XPS 15 9520", ALC289_FIXUP_DUAL_SPK), 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), BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=216035 Signed-off-by: Rik van der Kemp <rik(a)upto11.nl>

2 1

[PATCH] Enable SPDIF output on Intel Hades Canyon
by Dag B 27 May '22

27 May '22

Without the attached patch, the s/pdif output on the Hades Canyon NUC does not work. "Well known" issue, less known fix. As far as I can tell, there is no risk of any averse side-effects. But a bonus fix is enabling on-wire headset microphone, by chaining the right 'model' choice for the hardware. If I should Cc: someone directly for this patch to be picked up, please let me know. I have been running with this patch for half a year or so. Patch passes checkpatch.pl Patch is based on what 'cyber4o' posted on the insanelymac forum [1]. All the glory to this person, any error is likely mine. Some other users with issues: [2] [3] Dag Bakke [1] https://www.insanelymac.com/forum/topic/339291-guide-hac-mini-osx-mojave-on… 2] https://bbs.archlinux.org/viewtopic.php?id=270917 [3] https://www.reddit.com/r/intelnuc/comments/9ft9x8/any_linux_users_got_the_s…

3 6

[PATCH] ASoC: da732x: Drop no-op remove function
by Uwe Kleine-König 27 May '22

27 May '22

A remove callback that just returns 0 is equivalent to no callback at all as can be seen in i2c_device_remove(). So simplify accordingly. Signed-off-by: Uwe Kleine-König <u.kleine-koenig(a)pengutronix.de> --- sound/soc/codecs/da732x.c | 6 ------ 1 file changed, 6 deletions(-) diff --git a/sound/soc/codecs/da732x.c b/sound/soc/codecs/da732x.c index 42d6a3fc3af5..267936015cb8 100644 --- a/sound/soc/codecs/da732x.c +++ b/sound/soc/codecs/da732x.c @@ -1547,11 +1547,6 @@ static int da732x_i2c_probe(struct i2c_client *i2c, return ret; } -static int da732x_i2c_remove(struct i2c_client *client) -{ - return 0; -} - static const struct i2c_device_id da732x_i2c_id[] = { { "da7320", 0}, { } @@ -1563,7 +1558,6 @@ static struct i2c_driver da732x_i2c_driver = { .name = "da7320", }, .probe = da732x_i2c_probe, - .remove = da732x_i2c_remove, .id_table = da732x_i2c_id, }; base-commit: 4b0986a3613c92f4ec1bdc7f60ec66fea135991f -- 2.36.1

2 1

[PATCH] ASoC: fsl_xcvr:Fix unbalanced pm_runtime_enable in fsl_xcvr_probe
by zhangqilong 27 May '22

27 May '22

Add missing pm_runtime_disable() when probe error out. It could avoid pm_runtime implementation complains when removing and probing again the driver. Fix:c590fa80b3928 ("ASoC: fsl_xcvr: register platform component before registering cpu dai") Signed-off-by: Zhang Qilong <zhangqilong3(a)huawei.com> --- sound/soc/fsl/fsl_xcvr.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/fsl/fsl_xcvr.c b/sound/soc/fsl/fsl_xcvr.c index d0556c79fdb1..907435efe121 100644 --- a/sound/soc/fsl/fsl_xcvr.c +++ b/sound/soc/fsl/fsl_xcvr.c @@ -1228,6 +1228,7 @@ static int fsl_xcvr_probe(struct platform_device *pdev) */ ret = devm_snd_dmaengine_pcm_register(dev, NULL, 0); if (ret) { + pm_runtime_disable(dev); dev_err(dev, "failed to pcm register\n"); return ret; } @@ -1235,6 +1236,7 @@ static int fsl_xcvr_probe(struct platform_device *pdev) ret = devm_snd_soc_register_component(dev, &fsl_xcvr_comp, &fsl_xcvr_dai, 1); if (ret) { + pm_runtime_disable(dev); dev_err(dev, "failed to register component %s\n", fsl_xcvr_comp.name); } -- 2.31.1

2 1

[PATCH] ASoC: lm49453: Drop no-op remove function
by Uwe Kleine-König 27 May '22

27 May '22

A remove callback that just returns 0 is equivalent to no callback at all as can be seen in i2c_device_remove(). So simplify accordingly. Signed-off-by: Uwe Kleine-König <u.kleine-koenig(a)pengutronix.de> --- sound/soc/codecs/lm49453.c | 6 ------ 1 file changed, 6 deletions(-) diff --git a/sound/soc/codecs/lm49453.c b/sound/soc/codecs/lm49453.c index 973d781b4b6a..86cead5022c0 100644 --- a/sound/soc/codecs/lm49453.c +++ b/sound/soc/codecs/lm49453.c @@ -1443,11 +1443,6 @@ static int lm49453_i2c_probe(struct i2c_client *i2c, return ret; } -static int lm49453_i2c_remove(struct i2c_client *client) -{ - return 0; -} - static const struct i2c_device_id lm49453_i2c_id[] = { { "lm49453", 0 }, { } @@ -1459,7 +1454,6 @@ static struct i2c_driver lm49453_i2c_driver = { .name = "lm49453", }, .probe = lm49453_i2c_probe, - .remove = lm49453_i2c_remove, .id_table = lm49453_i2c_id, }; base-commit: 4b0986a3613c92f4ec1bdc7f60ec66fea135991f -- 2.36.1

1 1

[PATCH] ASoC: Intel: common: fix typo for tplg naming
by David Lin 26 May '22

26 May '22

Correct typo form sof-adl-mx98360a-nau8825.tplg to sof-adl-max98360a-nau8825.tplg. The reason is tplg naming without naming limitaion of length. It will be consistency with sof topology generation. Signed-off-by: David Lin <CTLIN0(a)nuvoton.com> --- 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 e6e52c7b6803..c1385161cdc8 100644 --- a/sound/soc/intel/common/soc-acpi-intel-adl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-adl-match.c @@ -453,7 +453,7 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_machines[] = { .drv_name = "adl_mx98360a_nau8825", .machine_quirk = snd_soc_acpi_codec_list, .quirk_data = &adl_max98360a_amp, - .sof_tplg_filename = "sof-adl-mx98360a-nau8825.tplg", + .sof_tplg_filename = "sof-adl-max98360a-nau8825.tplg", }, { .id = "RTL5682", -- 2.25.1

3 2

[PATCH] ASoC: rt5640: Do not manipulate pin "Platform Clock" if the "Platform Clock" is not in the DAPM
by Oder Chiou 26 May '22

26 May '22

The pin "Platform Clock" was only used by the Intel Byt CR platform. In the others, the error log will be informed. The patch will set the flag to avoid the pin "Platform Clock" manipulated by the other platforms. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> Reported-by: Sameer Pujar <spujar(a)nvidia.com> --- sound/soc/codecs/rt5640.c | 11 +++++++++-- sound/soc/codecs/rt5640.h | 2 ++ sound/soc/intel/boards/bytcr_rt5640.c | 2 ++ 3 files changed, 13 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 12da2bea1a7b..69c80d80ed9d 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -2094,12 +2094,14 @@ EXPORT_SYMBOL_GPL(rt5640_sel_asrc_clk_src); void rt5640_enable_micbias1_for_ovcd(struct snd_soc_component *component) { struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component); + struct rt5640_priv *rt5640 = snd_soc_component_get_drvdata(component); snd_soc_dapm_mutex_lock(dapm); snd_soc_dapm_force_enable_pin_unlocked(dapm, "LDO2"); snd_soc_dapm_force_enable_pin_unlocked(dapm, "MICBIAS1"); /* OVCD is unreliable when used with RCCLK as sysclk-source */ - snd_soc_dapm_force_enable_pin_unlocked(dapm, "Platform Clock"); + if (rt5640->use_platform_clock) + snd_soc_dapm_force_enable_pin_unlocked(dapm, "Platform Clock"); snd_soc_dapm_sync_unlocked(dapm); snd_soc_dapm_mutex_unlock(dapm); } @@ -2108,9 +2110,11 @@ EXPORT_SYMBOL_GPL(rt5640_enable_micbias1_for_ovcd); void rt5640_disable_micbias1_for_ovcd(struct snd_soc_component *component) { struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component); + struct rt5640_priv *rt5640 = snd_soc_component_get_drvdata(component); snd_soc_dapm_mutex_lock(dapm); - snd_soc_dapm_disable_pin_unlocked(dapm, "Platform Clock"); + if (rt5640->use_platform_clock) + snd_soc_dapm_disable_pin_unlocked(dapm, "Platform Clock"); snd_soc_dapm_disable_pin_unlocked(dapm, "MICBIAS1"); snd_soc_dapm_disable_pin_unlocked(dapm, "LDO2"); snd_soc_dapm_sync_unlocked(dapm); @@ -2535,6 +2539,9 @@ static void rt5640_enable_jack_detect(struct snd_soc_component *component, rt5640->jd_gpio_irq_requested = true; } + if (jack_data && jack_data->use_platform_clock) + rt5640->use_platform_clock = jack_data->use_platform_clock; + ret = request_irq(rt5640->irq, rt5640_irq, IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING | IRQF_ONESHOT, "rt5640", rt5640); diff --git a/sound/soc/codecs/rt5640.h b/sound/soc/codecs/rt5640.h index 9e49b9a0ccaa..505c93514051 100644 --- a/sound/soc/codecs/rt5640.h +++ b/sound/soc/codecs/rt5640.h @@ -2155,11 +2155,13 @@ struct rt5640_priv { bool jd_inverted; unsigned int ovcd_th; unsigned int ovcd_sf; + bool use_platform_clock; }; struct rt5640_set_jack_data { int codec_irq_override; struct gpio_desc *jd_gpio; + bool use_platform_clock; }; int rt5640_dmic_enable(struct snd_soc_component *component, diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 7b948a219177..ed9fa1728722 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -1191,12 +1191,14 @@ static int byt_rt5640_init(struct snd_soc_pcm_runtime *runtime) { struct snd_soc_card *card = runtime->card; struct byt_rt5640_private *priv = snd_soc_card_get_drvdata(card); + struct rt5640_set_jack_data *jack_data = &priv->jack_data; struct snd_soc_component *component = asoc_rtd_to_codec(runtime, 0)->component; const struct snd_soc_dapm_route *custom_map = NULL; int num_routes = 0; int ret; card->dapm.idle_bias_off = true; + jack_data->use_platform_clock = true; /* Start with RC clk for jack-detect (we disable MCLK below) */ if (byt_rt5640_quirk & BYT_RT5640_MCLK_EN) -- 2.35.3

3 2

Update mapping for the new Gigabyte Aorus Pro WiFi rev 1.2
by GitHub pull_request - edited 26 May '22

26 May '22

alsa-project/alsa-ucm-conf pull request #156 was edited from 13r0ck: Requires kernel patch from: https://bugzilla.kernel.org/attachment.cgi?id=301036&action=diff This doesn't fix the front mic for some reason (any ideas please?)~~, as well I am not sure if this will cause a regression for the 1.1 board~~ Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/156 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/156.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH] ALSA: usb-audio: Cancel pending work at closing a MIDI substream
by Takashi Iwai 25 May '22

25 May '22

At closing a USB MIDI output substream, there might be still a pending work, which would eventually access the rawmidi runtime object that is being released. For fixing the race, make sure to cancel the pending work at closing. Reported-by: syzbot+6912c9592caca7ca0e7d(a)syzkaller.appspotmail.com Cc: <stable(a)vger.kernel.org> Link: https://lore.kernel.org/r/000000000000e7e75005dfd07cf6@google.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/midi.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/usb/midi.c b/sound/usb/midi.c index 7c6ca2b433a5..344fbeadf161 100644 --- a/sound/usb/midi.c +++ b/sound/usb/midi.c @@ -1145,6 +1145,9 @@ static int snd_usbmidi_output_open(struct snd_rawmidi_substream *substream) static int snd_usbmidi_output_close(struct snd_rawmidi_substream *substream) { + struct usbmidi_out_port *port = substream->runtime->private_data; + + cancel_work_sync(&port->ep->work); return substream_open(substream, 0, 0); } -- 2.35.3

1 0

[PATCH] ASoC: stm32: sai: Use of_device_get_match_data() to simplify code
by Tang Bin 25 May '22

25 May '22

Retrieve of match data, it's better and cleaner to use 'of_device_get_match_data' over 'of_match_device'. Signed-off-by: Tang Bin <tangbin(a)cmss.chinamobile.com> --- sound/soc/stm/stm32_sai_sub.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c index dd636af81..d300605a2 100644 --- a/sound/soc/stm/stm32_sai_sub.c +++ b/sound/soc/stm/stm32_sai_sub.c @@ -1500,7 +1500,6 @@ static int stm32_sai_sub_parse_of(struct platform_device *pdev, static int stm32_sai_sub_probe(struct platform_device *pdev) { struct stm32_sai_sub_data *sai; - const struct of_device_id *of_id; const struct snd_dmaengine_pcm_config *conf = &stm32_sai_pcm_config; int ret; @@ -1508,10 +1507,9 @@ static int stm32_sai_sub_probe(struct platform_device *pdev) if (!sai) return -ENOMEM; - of_id = of_match_device(stm32_sai_sub_ids, &pdev->dev); - if (!of_id) + sai->id = (uintptr_t)of_device_get_match_data(&pdev->dev); + if (!sai->id) return -EINVAL; - sai->id = (uintptr_t)of_id->data; sai->pdev = pdev; mutex_init(&sai->ctrl_lock); -- 2.20.1.windows.1

4 6

[PATCH] ALSA: usb-audio: Add mixer mapping for Gigabyte B450/550 Mobos
by Takashi Iwai 25 May '22

25 May '22

This patch implements a static mapping for Gigabyte B450/550 Mobos so that the mixer elements appear reasonably and jack detections work properly. Reported-and-tested-by: Brock Szuszczewicz <brock(a)system76.com> BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=215988 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/mixer_maps.c | 30 ++++++++++++++++++++++++++++++ 1 file changed, 30 insertions(+) diff --git a/sound/usb/mixer_maps.c b/sound/usb/mixer_maps.c index 7ef7a8abcc2b..3c795675f048 100644 --- a/sound/usb/mixer_maps.c +++ b/sound/usb/mixer_maps.c @@ -439,6 +439,31 @@ static const struct usbmix_name_map msi_mpg_x570s_carbon_max_wifi_alc4080_map[] {} }; +/* Gigabyte B450/550 Mobo */ +static const struct usbmix_name_map gigabyte_b450_map[] = { + { 24, NULL }, /* OT, IEC958?, disabled */ + { 21, "Speaker" }, /* OT */ + { 29, "Speaker Playback" }, /* FU */ + { 22, "Headphone" }, /* OT */ + { 30, "Headphone Playback" }, /* FU */ + { 11, "Line" }, /* IT */ + { 27, "Line Capture" }, /* FU */ + { 12, "Mic" }, /* IT */ + { 28, "Mic Capture" }, /* FU */ + { 9, "Front Mic" }, /* IT */ + { 25, "Front Mic Capture" }, /* FU */ + {} +}; + +static const struct usbmix_connector_map gigabyte_b450_connector_map[] = { + { 13, 21 }, /* Speaker */ + { 14, 22 }, /* Headphone */ + { 19, 11 }, /* Line */ + { 20, 12 }, /* Mic */ + { 17, 9 }, /* Front Mic */ + {} +}; + /* * Control map entries */ @@ -581,6 +606,11 @@ static const struct usbmix_ctl_map usbmix_ctl_maps[] = { .map = trx40_mobo_map, .connector_map = trx40_mobo_connector_map, }, + { /* Gigabyte B450/550 Mobo */ + .id = USB_ID(0x0414, 0xa00d), + .map = gigabyte_b450_map, + .connector_map = gigabyte_b450_connector_map, + }, { /* ASUS ROG Zenith II */ .id = USB_ID(0x0b05, 0x1916), .map = asus_rog_map, -- 2.35.3

1 0

conf: Use ino64_t to save and compare inode numbers
by GitHub pull_request - opened 25 May '22

25 May '22

alsa-project/alsa-lib pull request #231 was opened from smcv: On 32-bit platforms when not using the large-file-support ABI, struct stat64 contains ino64_t which is 64-bit, while ino_t is only 32-bit. snd_config_update_r() checks whether a file has been replaced by saving the ino member of a struct stat64 and comparing it with a previously-saved inode number. On 32-bit platforms, assigning the 64-bit member of struct stat64 to a 32-bit member of struct finfo will truncate it modulo 1<<32, which could conceivably result in libasound not reloading configuration when it should (although the inode number space is large enough to make this failure mode highly unlikely). --- This is from source code inspection while looking at #223, I haven't seen a genuine user-observable bug result from this. cc @dos1, @perexg, @bertogg Request URL : https://github.com/alsa-project/alsa-lib/pull/231 Patch URL : https://github.com/alsa-project/alsa-lib/pull/231.patch Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH v7 0/1] firmware: mtk: add adsp ipc protocol for SOF
by Allen-KH Cheng 25 May '22

25 May '22

This patch provides mediatek adsp ipc support for SOF. ADSP IPC protocol offers (send/recv) interfaces using mediatek-mailbox APIs. This patch was tested and confirmed to with SOF fw on MT8195 cherry board. Based on matthias.bgg/linux.git, v5.18-next/soc changes since v6: - rebase to matthias.bgg/linux.git, v5.18-next/soc - Prefer "GPL" over "GPL v2" for MODULE_LICENSE changes since v5: - fix WARNING: modpost: missing MODULE_LICENSE() in drivers/mailbox /mtk-adsp-mailbox.o. Add MODULE_LICENSE in the last line. - Due to WARNING: Missing or malformed SPDX-License-Identifier tag in line 1 in checkpatch, we don't remove SPDX-License in line 1. changes since v4: - add error message for wrong mbox chan changes since v3: - rebase on v5.16-rc8 - update reviewers changes since v2: - add out tag for two memory free phases changes since v1: - add comments for mtk_adsp_ipc_send and mtk_adsp_ipc_recv - remove useless MODULE_LICENSE - change label name to out_free Allen-KH Cheng (1): firmware: mediatek: add adsp ipc protocol interface drivers/firmware/Kconfig | 1 + drivers/firmware/Makefile | 1 + drivers/firmware/mediatek/Kconfig | 9 + drivers/firmware/mediatek/Makefile | 2 + drivers/firmware/mediatek/mtk-adsp-ipc.c | 161 ++++++++++++++++++ .../linux/firmware/mediatek/mtk-adsp-ipc.h | 65 +++++++ 6 files changed, 239 insertions(+) create mode 100644 drivers/firmware/mediatek/Kconfig create mode 100644 drivers/firmware/mediatek/Makefile create mode 100644 drivers/firmware/mediatek/mtk-adsp-ipc.c create mode 100644 include/linux/firmware/mediatek/mtk-adsp-ipc.h -- 2.18.0

4 6

[syzbot] KASAN: use-after-free Read in __snd_rawmidi_transmit_ack
by syzbot 25 May '22

25 May '22

Hello, syzbot found the following issue on: HEAD commit: 3b5e1590a267 Merge tag 'gpio-fixes-for-v5.18' of git://git.. git tree: upstream console output: https://syzkaller.appspot.com/x/log.txt?x=1195135ef00000 kernel config: https://syzkaller.appspot.com/x/.config?x=902c5209311d387c dashboard link: https://syzkaller.appspot.com/bug?extid=6912c9592caca7ca0e7d compiler: gcc (Debian 10.2.1-6) 10.2.1 20210110, GNU ld (GNU Binutils for Debian) 2.35.2 Unfortunately, I don't have any reproducer for this issue yet. IMPORTANT: if you fix the issue, please add the following tag to the commit: Reported-by: syzbot+6912c9592caca7ca0e7d(a)syzkaller.appspotmail.com ================================================================== BUG: KASAN: use-after-free in __snd_rawmidi_transmit_ack+0x2cd/0x2f0 sound/core/rawmidi.c:1348 Read of size 8 at addr ffff888078b36410 by task kworker/1:1H/754 CPU: 1 PID: 754 Comm: kworker/1:1H Not tainted 5.18.0-rc7-syzkaller-00136-g3b5e1590a267 #0 Hardware name: Google Google Compute Engine/Google Compute Engine, BIOS Google 01/01/2011 Workqueue: events_highpri snd_usbmidi_out_work Call Trace: <TASK> __dump_stack lib/dump_stack.c:88 [inline] dump_stack_lvl+0xcd/0x134 lib/dump_stack.c:106 print_address_description.constprop.0.cold+0xeb/0x495 mm/kasan/report.c:313 print_report mm/kasan/report.c:429 [inline] kasan_report.cold+0xf4/0x1c6 mm/kasan/report.c:491 __snd_rawmidi_transmit_ack+0x2cd/0x2f0 sound/core/rawmidi.c:1348 snd_rawmidi_transmit+0xae/0xf0 sound/core/rawmidi.c:1415 snd_usbmidi_standard_output+0x264/0xc10 sound/usb/midi.c:650 snd_usbmidi_do_output+0x200/0x510 sound/usb/midi.c:311 process_one_work+0x996/0x1610 kernel/workqueue.c:2289 worker_thread+0x665/0x1080 kernel/workqueue.c:2436 kthread+0x2e9/0x3a0 kernel/kthread.c:376 ret_from_fork+0x1f/0x30 arch/x86/entry/entry_64.S:298 </TASK> Allocated by task 9345: kasan_save_stack+0x1e/0x40 mm/kasan/common.c:38 kasan_set_track mm/kasan/common.c:45 [inline] set_alloc_info mm/kasan/common.c:436 [inline] ____kasan_kmalloc mm/kasan/common.c:515 [inline] ____kasan_kmalloc mm/kasan/common.c:474 [inline] __kasan_kmalloc+0xa9/0xd0 mm/kasan/common.c:524 kmalloc include/linux/slab.h:581 [inline] kzalloc include/linux/slab.h:714 [inline] snd_rawmidi_runtime_create sound/core/rawmidi.c:148 [inline] open_substream+0xe9/0x8b0 sound/core/rawmidi.c:306 rawmidi_open_priv+0x591/0x6f0 sound/core/rawmidi.c:357 snd_rawmidi_kernel_open+0x1b5/0x270 sound/core/rawmidi.c:392 midisynth_use+0xee/0x270 sound/core/seq/seq_midi.c:215 subscribe_port sound/core/seq/seq_ports.c:412 [inline] check_and_subscribe_port+0x89a/0xb80 sound/core/seq/seq_ports.c:495 snd_seq_port_connect+0x382/0x540 sound/core/seq/seq_ports.c:581 snd_seq_ioctl_subscribe_port+0x1fc/0x400 sound/core/seq/seq_clientmgr.c:1492 snd_seq_kernel_client_ctl+0x102/0x1e0 sound/core/seq/seq_clientmgr.c:2369 snd_seq_oss_midi_open+0x582/0x6e0 sound/core/seq/oss/seq_oss_midi.c:359 snd_seq_oss_synth_setup_midi+0x12d/0x530 sound/core/seq/oss/seq_oss_synth.c:269 snd_seq_oss_open+0x8c3/0xa80 sound/core/seq/oss/seq_oss_init.c:260 odev_open+0x6c/0x90 sound/core/seq/oss/seq_oss.c:128 soundcore_open+0x44e/0x620 sound/sound_core.c:593 chrdev_open+0x266/0x770 fs/char_dev.c:414 do_dentry_open+0x4a1/0x11e0 fs/open.c:824 do_open fs/namei.c:3476 [inline] path_openat+0x1c71/0x2910 fs/namei.c:3609 do_filp_open+0x1aa/0x400 fs/namei.c:3636 do_sys_openat2+0x16d/0x4c0 fs/open.c:1213 do_sys_open fs/open.c:1229 [inline] __do_sys_openat fs/open.c:1245 [inline] __se_sys_openat fs/open.c:1240 [inline] __x64_sys_openat+0x13f/0x1f0 fs/open.c:1240 do_syscall_x64 arch/x86/entry/common.c:50 [inline] do_syscall_64+0x35/0xb0 arch/x86/entry/common.c:80 entry_SYSCALL_64_after_hwframe+0x44/0xae Freed by task 9345: kasan_save_stack+0x1e/0x40 mm/kasan/common.c:38 kasan_set_track+0x21/0x30 mm/kasan/common.c:45 kasan_set_free_info+0x20/0x30 mm/kasan/generic.c:370 ____kasan_slab_free mm/kasan/common.c:366 [inline] ____kasan_slab_free+0x166/0x1a0 mm/kasan/common.c:328 kasan_slab_free include/linux/kasan.h:200 [inline] slab_free_hook mm/slub.c:1728 [inline] slab_free_freelist_hook+0x8b/0x1c0 mm/slub.c:1754 slab_free mm/slub.c:3510 [inline] kfree+0xd6/0x4d0 mm/slub.c:4552 snd_rawmidi_runtime_free sound/core/rawmidi.c:177 [inline] close_substream.part.0+0x18d/0x720 sound/core/rawmidi.c:528 close_substream sound/core/rawmidi.c:507 [inline] rawmidi_release_priv+0x192/0x270 sound/core/rawmidi.c:547 snd_rawmidi_kernel_release+0x39/0xd0 sound/core/rawmidi.c:564 midisynth_unuse+0x45/0x80 sound/core/seq/seq_midi.c:244 unsubscribe_port sound/core/seq/seq_ports.c:437 [inline] __delete_and_unsubscribe_port+0x270/0x4c0 sound/core/seq/seq_ports.c:537 snd_seq_port_disconnect+0x41c/0x5d0 sound/core/seq/seq_ports.c:616 snd_seq_ioctl_unsubscribe_port+0x1fc/0x400 sound/core/seq/seq_clientmgr.c:1537 snd_seq_kernel_client_ctl+0x102/0x1e0 sound/core/seq/seq_clientmgr.c:2369 snd_seq_oss_midi_close+0x44f/0x4d0 sound/core/seq/oss/seq_oss_midi.c:404 snd_seq_oss_synth_reset+0x422/0x880 sound/core/seq/oss/seq_oss_synth.c:406 snd_seq_oss_reset+0x6f/0x290 sound/core/seq/oss/seq_oss_init.c:435 snd_seq_oss_release+0x78/0x1a0 sound/core/seq/oss/seq_oss_init.c:412 odev_release+0x4f/0x70 sound/core/seq/oss/seq_oss.c:144 __fput+0x277/0x9d0 fs/file_table.c:317 task_work_run+0xdd/0x1a0 kernel/task_work.c:164 get_signal+0x1c5/0x24c0 kernel/signal.c:2641 arch_do_signal_or_restart+0x82/0x20f0 arch/x86/kernel/signal.c:867 exit_to_user_mode_loop kernel/entry/common.c:166 [inline] exit_to_user_mode_prepare+0x15f/0x250 kernel/entry/common.c:201 __syscall_exit_to_user_mode_work kernel/entry/common.c:283 [inline] syscall_exit_to_user_mode+0x19/0x60 kernel/entry/common.c:294 do_syscall_64+0x42/0xb0 arch/x86/entry/common.c:86 entry_SYSCALL_64_after_hwframe+0x44/0xae The buggy address belongs to the object at ffff888078b36400 which belongs to the cache kmalloc-512 of size 512 The buggy address is located 16 bytes inside of 512-byte region [ffff888078b36400, ffff888078b36600) The buggy address belongs to the physical page: page:ffffea0001e2cd00 refcount:1 mapcount:0 mapping:0000000000000000 index:0x0 pfn:0x78b34 head:ffffea0001e2cd00 order:2 compound_mapcount:0 compound_pincount:0 flags: 0xfff00000010200(slab|head|node=0|zone=1|lastcpupid=0x7ff) raw: 00fff00000010200 dead000000000100 dead000000000122 ffff888010c41c80 raw: 0000000000000000 0000000000100010 00000001ffffffff 0000000000000000 page dumped because: kasan: bad access detected page_owner tracks the page as allocated page last allocated via order 2, migratetype Unmovable, gfp_mask 0xd20c0(__GFP_IO|__GFP_FS|__GFP_NOWARN|__GFP_NORETRY|__GFP_COMP|__GFP_NOMEMALLOC), pid 2967, tgid 2967 (udevadm), ts 15461279103, free_ts 12532036824 prep_new_page mm/page_alloc.c:2441 [inline] get_page_from_freelist+0xba2/0x3e00 mm/page_alloc.c:4182 __alloc_pages+0x1b2/0x500 mm/page_alloc.c:5408 alloc_pages+0x1aa/0x310 mm/mempolicy.c:2272 alloc_slab_page mm/slub.c:1799 [inline] allocate_slab+0x26c/0x3c0 mm/slub.c:1944 new_slab mm/slub.c:2004 [inline] ___slab_alloc+0x8df/0xf20 mm/slub.c:3005 __slab_alloc.constprop.0+0x4d/0xa0 mm/slub.c:3092 slab_alloc_node mm/slub.c:3183 [inline] __kmalloc_node_track_caller+0x2cb/0x360 mm/slub.c:4947 kmalloc_reserve net/core/skbuff.c:354 [inline] __alloc_skb+0xde/0x340 net/core/skbuff.c:426 alloc_skb include/linux/skbuff.h:1300 [inline] alloc_uevent_skb+0x7b/0x210 lib/kobject_uevent.c:290 uevent_net_broadcast_untagged lib/kobject_uevent.c:326 [inline] kobject_uevent_net_broadcast lib/kobject_uevent.c:409 [inline] kobject_uevent_env+0xc42/0x1660 lib/kobject_uevent.c:593 kobject_synth_uevent+0x701/0x850 lib/kobject_uevent.c:208 store_uevent+0x12/0x20 kernel/module.c:1166 module_attr_store+0x50/0x80 kernel/params.c:919 sysfs_kf_write+0x110/0x160 fs/sysfs/file.c:136 kernfs_fop_write_iter+0x3f8/0x610 fs/kernfs/file.c:291 call_write_iter include/linux/fs.h:2050 [inline] new_sync_write+0x38a/0x560 fs/read_write.c:504 page last free stack trace: reset_page_owner include/linux/page_owner.h:24 [inline] free_pages_prepare mm/page_alloc.c:1356 [inline] free_pcp_prepare+0x549/0xd20 mm/page_alloc.c:1406 free_unref_page_prepare mm/page_alloc.c:3328 [inline] free_unref_page+0x19/0x6a0 mm/page_alloc.c:3423 free_contig_range+0xb1/0x180 mm/page_alloc.c:9418 destroy_args+0xa8/0x646 mm/debug_vm_pgtable.c:1018 debug_vm_pgtable+0x2a51/0x2ae3 mm/debug_vm_pgtable.c:1332 do_one_initcall+0x103/0x650 init/main.c:1298 do_initcall_level init/main.c:1371 [inline] do_initcalls init/main.c:1387 [inline] do_basic_setup init/main.c:1406 [inline] kernel_init_freeable+0x6b1/0x73a init/main.c:1613 kernel_init+0x1a/0x1d0 init/main.c:1502 ret_from_fork+0x1f/0x30 arch/x86/entry/entry_64.S:298 Memory state around the buggy address: ffff888078b36300: fc fc fc fc fc fc fc fc fc fc fc fc fc fc fc fc ffff888078b36380: fc fc fc fc fc fc fc fc fc fc fc fc fc fc fc fc >ffff888078b36400: fa fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb ^ ffff888078b36480: fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb ffff888078b36500: fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb ================================================================== --- This report is generated by a bot. It may contain errors. See https://goo.gl/tpsmEJ for more information about syzbot. syzbot engineers can be reached at syzkaller(a)googlegroups.com. syzbot will keep track of this issue. See: https://goo.gl/tpsmEJ#status for how to communicate with syzbot.

1 0

[PATCH] ASoC: nau8822: choose the best master clock prescalar
by Hui Wang 25 May '22

25 May '22

We have an imx6sx EVB with the codec nau8822, when playing the audio with 48k sample rate, the audio sounds distorted and obviously faster than expected. The codec input MCLK is 24000000Hz on our board, if the target sample rate is 48000Hz, with the current algorithm, the computed prescalar will result in 62500Hz, with the patched algorithm, the sample rate will be 46875Hz which is nearer the target sample rate. And for other sample rate like 44100, 22050, 11025 or 8000, the patched algorithm has no difference with the current algorithm, they will generate the same prescalar and the same sample rate. Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/soc/codecs/nau8822.c | 9 ++++++--- 1 file changed, 6 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/nau8822.c b/sound/soc/codecs/nau8822.c index 66bbd8f4f1ad..b67fdf64ddab 100644 --- a/sound/soc/codecs/nau8822.c +++ b/sound/soc/codecs/nau8822.c @@ -668,21 +668,24 @@ static int nau8822_config_clkdiv(struct snd_soc_dai *dai, int div, int rate) struct snd_soc_component *component = dai->component; struct nau8822 *nau8822 = snd_soc_component_get_drvdata(component); struct nau8822_pll *pll = &nau8822->pll; + int diff = INT_MAX; int i, sclk, imclk; switch (nau8822->div_id) { case NAU8822_CLK_MCLK: /* Configure the master clock prescaler div to make system * clock to approximate the internal master clock (IMCLK); - * and large or equal to IMCLK. + * and find the nearest value to IMCLK. */ div = 0; imclk = rate * 256; for (i = 1; i < ARRAY_SIZE(nau8822_mclk_scaler); i++) { sclk = (nau8822->sysclk * 10) / nau8822_mclk_scaler[i]; - if (sclk < imclk) + if (abs(sclk - imclk) < diff) { + diff = abs(sclk - imclk); + div = i; + } else break; - div = i; } dev_dbg(component->dev, "master clock prescaler %x for fs %d\n", div, rate); -- 2.25.1

2 5

Update mapping for the new Gigabyte Aorus Pro WiFi rev 1.2
by GitHub pull_request - edited 25 May '22

25 May '22

alsa-project/alsa-ucm-conf pull request #156 was edited from 13r0ck: Requires kernel patch from: https://bugzilla.kernel.org/attachment.cgi?id=301036&action=diff This doesn't fix the front mic for some reason (any ideas please?), as well I am not sure if this will cause a regression for the 1.1 board Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/156 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/156.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Update mapping for the new Gigabyte Aorus Pro WiFi rev 1.2
by GitHub pull_request - opened 25 May '22

25 May '22

alsa-project/alsa-ucm-conf pull request #157 was opened from 13r0ck: Requires kernel patch from: https://bugzilla.kernel.org/show_bug.cgi?id=215988 Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/157 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/157.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Update mapping for the new Gigabyte Aorus Pro WiFi rev 1.2
by GitHub pull_request - edited 25 May '22

25 May '22

alsa-project/alsa-ucm-conf pull request #156 was edited from 13r0ck: Requires kernel patch from: https://bugzilla.kernel.org/attachment.cgi?id=301036&action=diff This doesn't fix the front mic for some reason, as well I am not sure if this will cause a regression for the 1.1 board Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/156 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/156.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Update mapping for the new Gigabyte Aorus Pro WiFi rev 1.2
by GitHub pull_request - opened 25 May '22

25 May '22

alsa-project/alsa-ucm-conf pull request #156 was opened from 13r0ck: Requires kernel patch from: https://bugzilla.kernel.org/show_bug.cgi?id=215988 This doesn't fix the front mic for some reason, as well I am not sure if this will cause a regression for the 1.1 board Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/156 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/156.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH] ASoC: SOF: amd: Fixed Build error
by Syed Saba kareem 24 May '22

24 May '22

Add linux/module.h in acp-pci.c to solve the below dependency All error/warnings (new ones prefixed by >>): >> sound/soc/amd/acp/acp-pci.c:148:1: warning: data definition has no type or storage class 148 | MODULE_DEVICE_TABLE(pci, acp_pci_ids); | ^~~~~~~~~~~~~~~~~~~ >> sound/soc/amd/acp/acp-pci.c:148:1: error: type defaults to 'int' in declaration of 'MODULE_DEVICE_TABLE' [-Werror=implicit-int] ... Reported-by: kernel test robot <lkp(a)intel.com> Signed-off-by: Syed Saba Kareem<ssabakar(a)amd.com> --- sound/soc/amd/acp/acp-pci.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/amd/acp/acp-pci.c b/sound/soc/amd/acp/acp-pci.c index 340e39d7f420..c893963ee2d0 100644 --- a/sound/soc/amd/acp/acp-pci.c +++ b/sound/soc/amd/acp/acp-pci.c @@ -16,6 +16,7 @@ #include <linux/pci.h> #include <linux/platform_device.h> #include <linux/pm_runtime.h> +#include <linux/module.h> #include "amd.h" #include "../mach-config.h" -- 2.25.1

2 1

[PATCH AUTOSEL 5.15 01/10] ALSA: usb-audio: Don't get sample rate for MCT Trigger 5 USB-to-HDMI
by Sasha Levin 24 May '22

24 May '22

From: Forest Crossman <cyrozap(a)gmail.com> [ Upstream commit d7be213849232a2accb219d537edf056d29186b4 ] This device doesn't support reading the sample rate, so we need to apply this quirk to avoid a 15-second delay waiting for three timeouts. Signed-off-by: Forest Crossman <cyrozap(a)gmail.com> Link: https://lore.kernel.org/r/20220504002444.114011-2-cyrozap@gmail.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/usb/quirks.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index ab9f3da49941..fbbe59054c3f 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1822,6 +1822,8 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = { QUIRK_FLAG_IGNORE_CTL_ERROR), DEVICE_FLG(0x06f8, 0xd002, /* Hercules DJ Console (Macintosh Edition) */ QUIRK_FLAG_IGNORE_CTL_ERROR), + DEVICE_FLG(0x0711, 0x5800, /* MCT Trigger 5 USB-to-HDMI */ + QUIRK_FLAG_GET_SAMPLE_RATE), DEVICE_FLG(0x074d, 0x3553, /* Outlaw RR2150 (Micronas UAC3553B) */ QUIRK_FLAG_GET_SAMPLE_RATE), DEVICE_FLG(0x08bb, 0x2702, /* LineX FM Transmitter */ -- 2.35.1

1 0

[PATCH AUTOSEL 5.17 01/12] ALSA: usb-audio: Don't get sample rate for MCT Trigger 5 USB-to-HDMI
by Sasha Levin 24 May '22

24 May '22

From: Forest Crossman <cyrozap(a)gmail.com> [ Upstream commit d7be213849232a2accb219d537edf056d29186b4 ] This device doesn't support reading the sample rate, so we need to apply this quirk to avoid a 15-second delay waiting for three timeouts. Signed-off-by: Forest Crossman <cyrozap(a)gmail.com> Link: https://lore.kernel.org/r/20220504002444.114011-2-cyrozap@gmail.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/usb/quirks.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index ab9f3da49941..fbbe59054c3f 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1822,6 +1822,8 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = { QUIRK_FLAG_IGNORE_CTL_ERROR), DEVICE_FLG(0x06f8, 0xd002, /* Hercules DJ Console (Macintosh Edition) */ QUIRK_FLAG_IGNORE_CTL_ERROR), + DEVICE_FLG(0x0711, 0x5800, /* MCT Trigger 5 USB-to-HDMI */ + QUIRK_FLAG_GET_SAMPLE_RATE), DEVICE_FLG(0x074d, 0x3553, /* Outlaw RR2150 (Micronas UAC3553B) */ QUIRK_FLAG_GET_SAMPLE_RATE), DEVICE_FLG(0x08bb, 0x2702, /* LineX FM Transmitter */ -- 2.35.1

1 2

Add missing implementation for topology ignore_suspend flag
by GitHub pull_request - opened 24 May '22

24 May '22

alsa-project/alsa-lib pull request #230 was opened from amadeuszslawinski-intel: ignore_suspend flag was ignored while parsing topology, add missing support for it. This allows use of it on DAI links and DAPM widgets. Request URL : https://github.com/alsa-project/alsa-lib/pull/230 Patch URL : https://github.com/alsa-project/alsa-lib/pull/230.patch Repository URL: https://github.com/alsa-project/alsa-lib

1 0

Fix for arecord recording ghost data
by GitHub pull_request - opened 24 May '22

24 May '22

alsa-project/alsa-utils pull request #150 was opened from amadeuszslawinski-intel: We noticed the issue in our tests, when using arecord would produce files with glitches at recording end. Apparently it was caused by writing full buffer to output file instead of the data read, when stopping recording. Following commit adjust the code to only write the amount of data that was read and making sure that file size matches internal format data. Request URL : https://github.com/alsa-project/alsa-utils/pull/150 Patch URL : https://github.com/alsa-project/alsa-utils/pull/150.patch Repository URL: https://github.com/alsa-project/alsa-utils

1 0

Add new type for ALC245
by Kailang 24 May '22

24 May '22

Hi Takashi, Attach patch will add new codec type for ALC245. BR, Kailang

2 1

[PATCH] ALSA: usb-audio: Configure sync endpoints before data
by Takashi Iwai 24 May '22

24 May '22

From: Craig McLure <craig(a)mclure.net> Devices such as the TC-Helicon GoXLR require the sync endpoint to be configured in advance of the data endpoint in order for sound output to work. This patch simply changes the ordering of EP configuration to resolve this. Fixes: bf6313a0ff76 ("ALSA: usb-audio: Refactor endpoint management") BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=215079 Signed-off-by: Craig McLure <craig(a)mclure.net> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/pcm.c | 17 +++++++++++------ 1 file changed, 11 insertions(+), 6 deletions(-) diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 6d699065e81a..b470404a5376 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -439,16 +439,21 @@ static int configure_endpoints(struct snd_usb_audio *chip, /* stop any running stream beforehand */ if (stop_endpoints(subs, false)) sync_pending_stops(subs); + if (subs->sync_endpoint) { + err = snd_usb_endpoint_configure(chip, subs->sync_endpoint); + if (err < 0) + return err; + } err = snd_usb_endpoint_configure(chip, subs->data_endpoint); if (err < 0) return err; snd_usb_set_format_quirk(subs, subs->cur_audiofmt); - } - - if (subs->sync_endpoint) { - err = snd_usb_endpoint_configure(chip, subs->sync_endpoint); - if (err < 0) - return err; + } else { + if (subs->sync_endpoint) { + err = snd_usb_endpoint_configure(chip, subs->sync_endpoint); + if (err < 0) + return err; + } } return 0; -- 2.35.3

2 1

GoXLR: Fixed sampler input configuration
by GitHub pull_request - opened 24 May '22

24 May '22

alsa-project/alsa-ucm-conf pull request #155 was opened from FrostyCoolSlug: Renamed the 'Sample' input channel to 'Sampler' to better reflect it's behaviour, and corrected channel numbers. This fixes sampler capture on the GoXLR. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/155 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/155.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

amixer returns a RC=1 when you use 'amixer -v' while other 'amixer' commands (like 'amixer -h` or `amixer --help') return with a '0'.
by GitHub issues - opened 23 May '22

23 May '22

alsa-project/alsa-utils issue #149 was opened from juggledad: amixer version 1.2.4 Raspberry Pi Zero W v1.1 Raspbian GNU/Linux 11 (bullseye) - desktop version Node-RED v2.2.2 This was discovered using an `exec` node in on a The exec node runs a command and returns its output. When running a test, I wanted to get the amixer version. There are two ways to ask `amixer` for it's version number 1 - use 'amixer -v' 2 - use 'amixer --version` Both returned the version number but had a Return Code of 1. All other 'amixer' commands (like 'amixer -h` or `amixer --help') return with a '0'. Here is a copy of the node-red flow that was used: ` [{"active":true,"complete":"true","console":false,"id":"38409c9414d4fb0b","name":"stdout","statusType":"auto","statusVal":"","targetType":"full","tosidebar":true,"tostatus":false,"type":"debug","wires":[],"x":730,"y":80,"z":"58ad38887591e1b1"},{"active":true,"complete":"true","console":false,"id":"de852fbd07407b7a","name":"stderr","statusType":"auto","statusVal":"","targetType":"full","tosidebar":true,"tostatus":false,"type":"debug","wires":[],"x":730,"y":120,"z":"58ad38887591e1b1"},{"active":true,"complete":"true","console":false,"id":"42faac21ce0190b3","name":"return code","statusType":"auto","statusVal":"","targetType":"full","tosidebar":true,"tostatus":false,"type":"debug","wires":[],"x":750,"y":160,"z":"58ad38887591e1b1"},{"crontab":"","id":"84f238ce5b65227d","name":"","once":false,"onceDelay":0.1,"payload":"amixer -v","payloadType":"str","props":[{"p":"payload"},{"p":"topic","vt":"str"}],"repeat":"","topic":"","type":"inject","wires":[["05c0a4eb46c0c1c8"]],"x":180,"y":60,"z":"58ad38887591e1b1"},{"crontab":"","id":"fabfd7b565392096","name":"","once":false,"onceDelay":0.1,"payload":"amixer -h","payloadType":"str","props":[{"p":"payload"},{"p":"topic","vt":"str"}],"repeat":"","topic":"","type":"inject","wires":[["05c0a4eb46c0c1c8"]],"x":180,"y":180,"z":"58ad38887591e1b1"},{"crontab":"","id":"c54064c5cc3b2484","name":"","once":false,"onceDelay":0.1,"payload":"amixer --version","payloadType":"str","props":[{"p":"payload"},{"p":"topic","vt":"str"}],"repeat":"","topic":"","type":"inject","wires":[["05c0a4eb46c0c1c8"]],"x":200,"y":120,"z":"58ad38887591e1b1"},{"addpay":"payload","append":"","command":"","id":"05c0a4eb46c0c1c8","name":"","oldrc":false,"timer":"","type":"exec","useSpawn":"false","winHide":false,"wires":[["38409c9414d4fb0b"],["de852fbd07407b7a"],["42faac21ce0190b3"]],"x":490,"y":120,"z":"58ad38887591e1b1"}] ` Issue URL : https://github.com/alsa-project/alsa-utils/issues/149 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[PATCH] ASoC: fsl_sai: Fix FSL_SAI_xDR/xFR definition
by Shengjiu Wang 23 May '22

23 May '22

There are multiple xDR and xFR registers, the index is from 0 to 7. FSL_SAI_xDR and FSL_SAI_xFR is abandoned, replace them with FSL_SAI_xDR0 and FSL_SAI_xFR0. Fixes: 4f7a0728b530 ("ASoC: fsl_sai: Add support for SAI new version") Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/fsl_sai.h | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h index e4965efe6102..1c8f5ca07f9d 100644 --- a/sound/soc/fsl/fsl_sai.h +++ b/sound/soc/fsl/fsl_sai.h @@ -80,8 +80,8 @@ #define FSL_SAI_xCR3(tx, ofs) (tx ? FSL_SAI_TCR3(ofs) : FSL_SAI_RCR3(ofs)) #define FSL_SAI_xCR4(tx, ofs) (tx ? FSL_SAI_TCR4(ofs) : FSL_SAI_RCR4(ofs)) #define FSL_SAI_xCR5(tx, ofs) (tx ? FSL_SAI_TCR5(ofs) : FSL_SAI_RCR5(ofs)) -#define FSL_SAI_xDR(tx, ofs) (tx ? FSL_SAI_TDR(ofs) : FSL_SAI_RDR(ofs)) -#define FSL_SAI_xFR(tx, ofs) (tx ? FSL_SAI_TFR(ofs) : FSL_SAI_RFR(ofs)) +#define FSL_SAI_xDR0(tx) (tx ? FSL_SAI_TDR0 : FSL_SAI_RDR0) +#define FSL_SAI_xFR0(tx) (tx ? FSL_SAI_TFR0 : FSL_SAI_RFR0) #define FSL_SAI_xMR(tx) (tx ? FSL_SAI_TMR : FSL_SAI_RMR) /* SAI Transmit/Receive Control Register */ -- 2.17.1

2 1

[PATCH] ASoC: soc-pcm: fix BE transition for TRIGGER_START
by Pierre-Louis Bossart 23 May '22

23 May '22

A obvious editing mistake caught with a cppcheck warning sound/soc/soc-pcm.c:2132:8: style: Variable 'ret' is reassigned a value before the old one has been used. [redundantAssignment] ret = soc_pcm_trigger(be_substream, cmd); ^ sound/soc/soc-pcm.c:2126:9: note: ret is assigned ret = soc_pcm_trigger(be_substream, ^ sound/soc/soc-pcm.c:2129:9: note: ret is assigned ret = soc_pcm_trigger(be_substream, ^ Fixes: 374b50e234a3e ('ASoC: soc-pcm: improve BE transition for TRIGGER_START') Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Reviewed-by: Péter Ujfalusi <peter.ujfalusi(a)linux.intel.com> --- sound/soc/soc-pcm.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 6f43db35a5c84..a827cc3c158ae 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2128,8 +2128,6 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, else ret = soc_pcm_trigger(be_substream, SNDRV_PCM_TRIGGER_START); - - ret = soc_pcm_trigger(be_substream, cmd); if (ret) { be->dpcm[stream].be_start--; goto next; -- 2.30.2

2 1

[v6] FROMLIST: ASoC: Intel: sof_cs42l42: adding support for ADL configuration and BT offload audio
by Terry Chen 23 May '22

23 May '22

To be able to do driver data for adl_mx98360a_cs4242 which supports two max98360a speaker amplifiers on SSP1 and cs42l42 headphone codec on SSP0 running on ADL platform. Also add the capability to machine driver of creating DAI Link for BT offload. Although BT offload always uses SSP2 port but we reserve the flexibility to assign the port number in macro. Signed-off-by: Terry Chen <terry_chen(a)wistron.corp-partner.google.com> (am from https://patchwork.kernel.org/patch/12845884/) (also found at https://lore.kernel.org/r/20220511075522.1764114-1-terry_chen@wistron.corp-…) --- sound/soc/intel/boards/sof_cs42l42.c | 92 ++++++++++++++++++- .../intel/common/soc-acpi-intel-adl-match.c | 7 ++ 2 files changed, 95 insertions(+), 4 deletions(-) diff --git a/sound/soc/intel/boards/sof_cs42l42.c b/sound/soc/intel/boards/sof_cs42l42.c index ce78c18798876..2efffc7933479 100644 --- a/sound/soc/intel/boards/sof_cs42l42.c +++ b/sound/soc/intel/boards/sof_cs42l42.c @@ -41,8 +41,13 @@ #define SOF_CS42L42_DAILINK_MASK (GENMASK(24, 10)) #define SOF_CS42L42_DAILINK(link1, link2, link3, link4, link5) \ ((((link1) | ((link2) << 3) | ((link3) << 6) | ((link4) << 9) | ((link5) << 12)) << SOF_CS42L42_DAILINK_SHIFT) & SOF_CS42L42_DAILINK_MASK) -#define SOF_MAX98357A_SPEAKER_AMP_PRESENT BIT(25) -#define SOF_MAX98360A_SPEAKER_AMP_PRESENT BIT(26) +#define SOF_BT_OFFLOAD_PRESENT BIT(25) +#define SOF_CS42L42_SSP_BT_SHIFT 26 +#define SOF_CS42L42_SSP_BT_MASK (GENMASK(28, 26)) +#define SOF_CS42L42_SSP_BT(quirk) \ + (((quirk) << SOF_CS42L42_SSP_BT_SHIFT) & SOF_CS42L42_SSP_BT_MASK) +#define SOF_MAX98357A_SPEAKER_AMP_PRESENT BIT(29) +#define SOF_MAX98360A_SPEAKER_AMP_PRESENT BIT(30) enum { LINK_NONE = 0, @@ -50,6 +55,7 @@ enum { LINK_SPK = 2, LINK_DMIC = 3, LINK_HDMI = 4, + LINK_BT = 5, }; /* Default: SSP2 */ @@ -278,6 +284,13 @@ static struct snd_soc_dai_link_component dmic_component[] = { } }; +static struct snd_soc_dai_link_component dummy_component[] = { + { + .name = "snd-soc-dummy", + .dai_name = "snd-soc-dummy-dai", + } +}; + static int create_spk_amp_dai_links(struct device *dev, struct snd_soc_dai_link *links, struct snd_soc_dai_link_component *cpus, @@ -467,9 +480,56 @@ static int create_hdmi_dai_links(struct device *dev, return -ENOMEM; } +static int create_bt_offload_dai_links(struct device *dev, + struct snd_soc_dai_link *links, + struct snd_soc_dai_link_component *cpus, + int *id, int ssp_bt) +{ + int ret = 0; + + /* bt offload */ + if (!(sof_cs42l42_quirk & SOF_BT_OFFLOAD_PRESENT)) + return 0; + + links[*id].name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d-BT", + ssp_bt); + if (!links[*id].name) { + ret = -ENOMEM; + goto devm_err; + } + + links[*id].id = *id; + links[*id].codecs = dummy_component; + links[*id].num_codecs = ARRAY_SIZE(dummy_component); + links[*id].platforms = platform_component; + links[*id].num_platforms = ARRAY_SIZE(platform_component); + + links[*id].dpcm_playback = 1; + links[*id].dpcm_capture = 1; + links[*id].no_pcm = 1; + links[*id].cpus = &cpus[*id]; + links[*id].num_cpus = 1; + + links[*id].cpus->dai_name = devm_kasprintf(dev, GFP_KERNEL, + "SSP%d Pin", + ssp_bt); + if (!links[*id].cpus->dai_name) { + ret = -ENOMEM; + goto devm_err; + } + + (*id)++; + + return 0; + +devm_err: + return ret; +} + static struct snd_soc_dai_link *sof_card_dai_links_create(struct device *dev, int ssp_codec, int ssp_amp, + int ssp_bt, int dmic_be_num, int hdmi_num) { @@ -522,6 +582,14 @@ static struct snd_soc_dai_link *sof_card_dai_links_create(struct device *dev, goto devm_err; } break; + case LINK_BT: + ret = create_bt_offload_dai_links(dev, links, cpus, &id, ssp_bt); + if (ret < 0) { + dev_err(dev, "fail to create bt offload dai links, ret %d\n", + ret); + goto devm_err; + } + break; case LINK_NONE: /* caught here if it's not used as terminator in macro */ default: @@ -543,7 +611,7 @@ static int sof_audio_probe(struct platform_device *pdev) struct snd_soc_acpi_mach *mach; struct sof_card_private *ctx; int dmic_be_num, hdmi_num; - int ret, ssp_amp, ssp_codec; + int ret, ssp_bt, ssp_amp, ssp_codec; ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL); if (!ctx) @@ -568,6 +636,9 @@ static int sof_audio_probe(struct platform_device *pdev) dev_dbg(&pdev->dev, "sof_cs42l42_quirk = %lx\n", sof_cs42l42_quirk); + ssp_bt = (sof_cs42l42_quirk & SOF_CS42L42_SSP_BT_MASK) >> + SOF_CS42L42_SSP_BT_SHIFT; + ssp_amp = (sof_cs42l42_quirk & SOF_CS42L42_SSP_AMP_MASK) >> SOF_CS42L42_SSP_AMP_SHIFT; @@ -578,9 +649,11 @@ static int sof_audio_probe(struct platform_device *pdev) if (sof_cs42l42_quirk & SOF_SPEAKER_AMP_PRESENT) sof_audio_card_cs42l42.num_links++; + if (sof_cs42l42_quirk & SOF_BT_OFFLOAD_PRESENT) + sof_audio_card_cs42l42.num_links++; dai_links = sof_card_dai_links_create(&pdev->dev, ssp_codec, ssp_amp, - dmic_be_num, hdmi_num); + ssp_bt, dmic_be_num, hdmi_num); if (!dai_links) return -ENOMEM; @@ -621,6 +694,17 @@ static const struct platform_device_id board_ids[] = { SOF_CS42L42_SSP_AMP(1)) | SOF_CS42L42_DAILINK(LINK_HP, LINK_DMIC, LINK_HDMI, LINK_SPK, LINK_NONE), }, + { + .name = "adl_mx98360a_cs4242", + .driver_data = (kernel_ulong_t)(SOF_CS42L42_SSP_CODEC(0) | + SOF_SPEAKER_AMP_PRESENT | + SOF_MAX98360A_SPEAKER_AMP_PRESENT | + SOF_CS42L42_SSP_AMP(1) | + SOF_CS42L42_NUM_HDMIDEV(4) | + SOF_BT_OFFLOAD_PRESENT | + SOF_CS42L42_SSP_BT(2)) | + SOF_CS42L42_DAILINK(LINK_HP, LINK_DMIC, LINK_HDMI, LINK_SPK, LINK_BT), + }, { } }; MODULE_DEVICE_TABLE(platform, board_ids); 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 7c8cd00457f81..311adeb3afbc4 100644 --- a/sound/soc/intel/common/soc-acpi-intel-adl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-adl-match.c @@ -384,6 +384,13 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_machines[] = { .sof_fw_filename = "sof-adl.ri", .sof_tplg_filename = "sof-adl-cs35l41.tplg", }, + { + .id = "10134242", + .drv_name = "adl_mx98360a_cs4242", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &adl_max98360a_amp, + .sof_tplg_filename = "sof-adl-max98360a-rt5682.tplg", + }, {}, }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_adl_machines); -- 2.25.1

3 3

[GIT PULL] ASoC updates for v5.19
by Mark Brown 23 May '22

23 May '22

The following changes since commit c61711c1c95791850be48dd65a1d72eb34ba719f: ASoC: SOF: Fix NULL pointer exception in sof_pci_probe callback (2022-04-27 14:15:21 +0100) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v5.19 for you to fetch changes up to e5cd20e0d6713138444cc3f3f982712cf9a36143: ASoC: SOF: Introduce generic (in)firmware tracing infrastructure (2022-05-20 14:56:35 +0100) ---------------------------------------------------------------- ASoC: Updates for v5.19 This is quite a big update, partly due to the addition of some larger drivers (more of which is to follow since at least the AVS driver is still a work in progress) and partly due to Charles' work sorting out our handling of endianness. As has been the case recently it's much more about drivers than the core. - Overhaul of endianness specification for data formats, avoiding needless restrictions due to CODECs. - Initial stages of Intel AVS driver merge. - Introduction of v4 IPC mechanism for SOF. - TDM mode support for AK4613. - Support for Analog Devices ADAU1361, Cirrus Logic CS35L45, Maxim MAX98396, MediaTek MT8186, NXP i.MX8 micfil and SAI interfaces, nVidia Tegra186 ASRC, and Texas Instruments TAS2764 and TAS2780 ---------------------------------------------------------------- Ajit Kumar Pandey (3): ASoC: amd: acp: Add pm ops callback in machine driver ASoC: SOF: amd: Add psp_mbox_ready() and psp_send_cmd() callback ASoC: SOF: amd: Use dedicated MBOX for ACP and PSP communication Akihiko Odaki (1): ASoC: soc-card: Create jack kcontrol without pins Alexey Khoroshilov (1): ASoC: max98090: Move check for invalid values before casting in max98090_put_enab_tlv() Allen-KH Cheng (1): ASoC: SOF: mediatek: Add ipc support for mt8195 Amadeusz Sławiński (10): ASoC: topology: Use for loop instead of while ASoC: topology: Remove unnecessary looping ASoC: topology: Return bool instead of int ASoC: topology: Rename SOC_TPLG_PASS_MIXER to _CONTROL ASoC: topology: Correct error message ASoC: topology: Rename soc_tplg_init_kcontrol() function ASoC: Intel: avs: Enable AVS driver only on x86 platforms ASoC: Intel: avs: Depend on SND_INTEL_DSP_CONFIG ALSA: hda: intel-nhlt: Move structs out of #define block ASoC: Intel: avs: Drop direct ACPI dependency Andy Shevchenko (1): ASoC: cs43130: Re-use generic struct u16_fract Biju Das (1): ASoC: sh: Make SND_SOC_RZ depend on ARCH_RZG2L Brent Lu (2): ASoC: Intel: sof_ssp_amp: fix no DMIC BE Link on Chromebooks ASoC: Intel: sof_ssp_amp: enable capture stream from cs35l41 Cezary Rojewski (29): ASoC: Intel: catpt: Drop redundant enum constant ASoC: Intel: avs: Declare vendor tokens ASoC: Intel: avs: Add topology parsing infrastructure ASoC: Intel: avs: Parse module-extension tuples ASoC: Intel: avs: Parse pplcfg and binding tuples ASoC: Intel: avs: Parse pipeline and module tuples ASoC: Intel: avs: Parse path and path templates tuples ASoC: Intel: avs: Add topology loading operations ASoC: Intel: avs: Support link_mask formatted string ASoC: Intel: avs: Declare path and its components ASoC: Intel: avs: Path creation and freeing ASoC: Intel: avs: Path state management ASoC: Intel: avs: Arm paths after creating them ASoC: Intel: avs: Prepare modules before bindings them ASoC: Intel: avs: Configure modules according to their type ASoC: Intel: avs: Account for libraries when booting basefw ASoC: Intel: avs: Generic soc component driver ASoC: Intel: avs: Generic PCM FE operations ASoC: Intel: avs: non-HDA PCM BE operations ASoC: Intel: avs: HDA PCM BE operations ASoC: Intel: avs: Coredump and recovery flow ASoC: Intel: avs: Prepare for firmware tracing ASoC: Intel: avs: D0ix power state support ASoC: Intel: avs: Event tracing ASoC: Intel: avs: Machine board registration ASoC: Intel: avs: PCI driver implementation ASoC: Intel: avs: Power management ASoC: Intel: avs: SKL-based platforms support ASoC: Intel: avs: APL-based platforms support Charles Keepax (43): ASoC: soc-component: Add comment for the endianness flag ASoC: atmel-pdmic: Remove endianness flag on pdmic component ASoC: atmel-classd: Remove endianness flag on class d component ASoC: cs4270: Remove redundant big endian formats ASoC: cs42l51: Remove redundant big endian formats ASoC: cs4349: Remove redundant big endian formats ASoC: hdmi-codec: Remove redundant big endian formats ASoC: sta32x: Remove redundant big endian formats ASoC: sta350: Remove redundant big endian formats ASoC: hdac_hda: Add endianness flag in snd_soc_component_driver ASoC: max98504: Add endianness flag in snd_soc_component_driver ASoC: adau1372: Add endianness flag in snd_soc_component_driver ASoC: cs4234: Add endianness flag in snd_soc_component_driver ASoC: cs35l41: Add endianness flag in snd_soc_component_driver ASoC: cx2072x: Add endianness flag in snd_soc_component_driver ASoC: lochnagar: Add endianness flag in snd_soc_component_driver ASoC: mt6351: Add endianness flag in snd_soc_component_driver ASoC: mt6358: Add endianness flag in snd_soc_component_driver ASoC: mt6359: Add endianness flag in snd_soc_component_driver ASoC: mt6660: Add endianness flag in snd_soc_component_driver ASoC: pcm3060: Add endianness flag in snd_soc_component_driver ASoC: rt1019: Add endianness flag in snd_soc_component_driver ASoC: rt9120: Add endianness flag in snd_soc_component_driver ASoC: tlv320adc3xxx: Add endianness flag in snd_soc_component_driver ASoC: tscs454: Add endianness flag in snd_soc_component_driver ASoC: cros_ec_codec: Add endianness flag in i2s_rx_component_driver ASoC: wcd934x: Add endianness flag in snd_soc_component_driver ASoC: wcd9335: Add endianness flag in snd_soc_component_driver ASoC: rt700: Add endianness flag in snd_soc_component_driver ASoC: rt711: Add endianness flag in snd_soc_component_driver ASoC: rt711-sdca: Add endianness flag in snd_soc_component_driver ASoC: rt715: Add endianness flag in snd_soc_component_driver ASoC: rt715-sdca: Add endianness flag in snd_soc_component_driver ASoC: rt1308-sdw: Add endianness flag in snd_soc_component_driver ASoC: rt1316-sdw: Add endianness flag in snd_soc_component_driver ASoC: wcd938x: Add endianness flag in snd_soc_component_driver ASoC: wsa881x: Add endianness flag in snd_soc_component_driver ASoC: sdw-mockup: Add endianness flag in snd_soc_component_driver ASoC: wm2000: Remove redundant endianness flag ASoC: wm9090: Remove redundant endianness flag ASoC: core: Correct spelling fliped -> flipped ASoC: core: Pass legacy_dai_naming flag directly ASoC: soc-component: Update handling to component delays ChiYuan Huang (2): ASoC: rt9120: Fix 3byte read, valule offset typo ASoC: rt9120: Use pm_runtime and regcache to optimize 'pwdnn' logic Christophe JAILLET (1): ASoC: SOF: topology: Avoid open coded arithmetic in memory allocation Corentin Labbe (1): ASoC: fix invalid yaml Derek Fang (1): ASoC: rt5682s: Separate the regulator consumer controls Gongjun Song (1): ASoC: SOF: Intel: pci-tgl: add RPL-P support Hans de Goede (1): ASoC: Intel: bytcr_rt5640: Add quirk for the HP Pro Tablet 408 Haowen Bai (1): ASoC: imx-hdmi: remove useless null check before call of_node_put() Hui Tang (1): ASoC: max98396: Fix build error for implicit function declaration Hui Wang (2): ASoC: cs35l41: Add one more variable in the debug log ASoC: cs35l41: Fix an out-of-bounds access in otp_packed_element_t James Schulman (1): ASoC: cs35l45: Add driver for Cirrus Logic CS35L45 Smart Amp Janusz Krzysztofik (1): ASoC: ti: osk5912: Make it CCF clk API compatible Jaroslav Kysela (1): ASoC: SOF: topology: use new sound control LED layer Jayesh Choudhary (1): ASoC: ti: davinci-mcasp: Add dma-type for bcdma Jiaxin Yu (4): ASoC: dt-bindings: mt8192-mt6359: add new compatible and new properties ASoC: mediatek: mt8192: refactor for I2S3 DAI link of speaker ASoC: mediatek: mt8192: refactor for I2S8/I2S9 DAI links of headset ASoC: mediatek: mt8192: support rt1015p_rt5682s Judy Hsiao (2): ASoC: qcom: Add driver support for ALC5682I-VS ASoC: qcom: Use MCLK as RT5682I-VS sysclk source Kuninori Morimoto (10): ASoC: rsnd: use inclusive language for DAIFMT mask ASoC: ak4613: add missing mutex_lock() ASoC: ak4613: tidyup ak4613_interface ASoC: ak4613: return error if it was setup as clock provider ASoC: ak4613: priv has ctrl1 instead of iface ASoC: ak4613: rename constraint to constraint_rates ASoC: ak4613: add TDM256 support ASoC: rsnd: care default case on rsnd_ssiu_busif_err_status_clear() ASoC: audio-graph-card2: indicate "Experimental stage" warning only when successed ASoC: rsnd: care return value from rsnd_node_fixed_index() Lad Prabhakar (3): ASoC: sh: rz-ssi: Drop SSIFSR_TDC and SSIFSR_RDC macros ASoC: sh: rz-ssi: Propagate error codes returned from platform_get_irq_byname() ASoC: sh: rz-ssi: Release the DMA channels in rz_ssi_probe() error path Libin Yang (2): ASoC: SOF: extend the interface to stop DMA trace ASoC: SOF: disable dma trace in s0ix Lin Ma (1): ASoC: rt5645: Fix errorenous cleanup order Lukasz Majewski (2): doc: dts: Add device tree information regarding wm8940 codec ASoC: wm8940: add devicetree support Lv Ruyi (1): ASoC: mediatek: mt8195: Make sure of_device_id table are NULL terminated Mario Limonciello (2): ASoC: amd: Add driver data to acp6x machine driver ASoC: amd: Add support for enabling DMIC on acp6x via _DSD Mark Brown (57): ASoC: wm8731: Update to modern DAI terminology ASoC: wm8731: Factor component init out of bus code ASoC: wm8731: Move regulator request into wm8731_init() ASoC: wm8731: Factor our MCLK and mutex initialisation ASoC: wm8731: Factor out the I2C and SPI bus code into separate modules ASoC: wm8731: Convert DT bindings to YAML format ARM: configs: Update multi_v5_defconfig for WM8731 bus refactoring ASoC: atmel: Fix error handling in at91samg20ek probe() ASoC: atmel: Don't squash error codes from atmel_ssc_set_audio() ASoC: atmel: Fixes for AT91SAM9G20-EK audio driver ASoC: mediatek: Add support for MT8195 sound card with max98390 and rt5682 ASoC: SOF: Miscellaneous fixes for IPC and trace ASoC: wm8731: Overhaul of the driver ASoC: topology: Cleanup patches ASoC: ASRC support on Tegra186 and later ASoC: Pull in fixes ASoC: Add a driver for the Cirrus Logic CS35L45 Smart Amplifier Update dt-bindings for sc7280 platform ASoC: ak4613: add TDM256 test support ASoC: Merge fixes ASoC: mediatek: mt8192: support rt1015p_rt5682s Allow detecting ACP6x DMIC via _DSD ASoC: SOF: Abstractions for top-level IPC ops ASoC: remaining simple i2c probe changes ASoC: fsl_micfil: Driver updates ASoC: Intel: avs: Topology and path management ASoC: SOF: add INTEL_IPC4 plumbing ASoC: soc-pcm: improve BE state transitions ASoC: SOF: AMD updates ASoC: SOF: pci: add quirks and PCI IDS Add support of MediaTek mt8186 to SOF ASoC: remaining i2c_match_id i2c probe changes ASoC: Add macros for signed TLV controls with callbacks ASoC: SOF: Rework the firmware ready message handling ASoC: SOF: Intel: improve HDaudio DAI support ASoC: SOF: use pm_runtime_resume_and_get() ASoC: sh: rz-ssi: Trivial fixes ASoC: SOF: IPC Abstraction for FW loading ASoC: SOF: Miscellaneous preparatory patches for IPC4 ASoC: max98090: Fix event generation for custom controls ASoC: dapm: Don't fold register value changes into notifications ASoC: rt5514: Fix event generation for "DSP Voice Wake Up" control ASoC: Pull in fixes ASoC: SOF: IPC4: Introduce message handling functionality Add missing dt-binding properties for audio components on mt8192-asurada ASoC: sam9g20_wm8731: Use dev_err_probe() for snd_soc_register_card() ASoC: SOF: sof-client: Update for different IPC versions Clean up usage of the endianness flag ASoC: Intel: boards: Chromebook configuration updates ASOC: Fix the error handling code of the probe ASoC: SOF: Add IPC4 FW loader support firmware: mtk: add adsp ipc protocol for SOF ASoC: fsl_sai: Add support for i.MX8MM, MP, ULP ASoC: Intel: avs: Driver core and PCM operations ASoC: remove two unnecessary gpiolib dependencies ASoC: SOF: mediatek: add debug dump ASoC: SOF: Introduce generic (in)firmware tracing infrastructure Martin Povišer (1): ASoC: Introduce snd_soc_of_get_dai_link_cpus Miaoqian Lin (7): ASoC: mediatek: Fix error handling in mt8173_max98090_dev_probe ASoC: mediatek: Fix missing of_node_put in mt2701_wm8960_machine_probe ASoC: fsl: Fix refcount leak in imx_sgtl5000_probe ASoC: imx-hdmi: Fix refcount leak in imx_hdmi_probe ASoC: mxs-saif: Fix refcount leak in mxs_saif_probe ASoC: samsung: Fix refcount leak in aries_audio_probe ASoC: ti: j721e-evm: Fix refcount leak in j721e_soc_probe_* Minghao Chi (11): ASoC: fsl: using pm_runtime_resume_and_get instead of pm_runtime_get_sync ASoC: codecs: wm8962: using pm_runtime_resume_and_get instead of pm_runtime_get_sync ASoC: SOF: using pm_runtime_resume_and_get to simplify the code ASoC: fsl_asrc: using pm_runtime_resume_and_get to simplify the code ASoC: img-spdif-in: using pm_runtime_resume_and_get instead of pm_runtime_get_sync ASoC: img-spdif-out: using pm_runtime_resume_and_get to simplify the code ASoC: img-parallel-out: using pm_runtime_resume_and_get ASoC: img-i2s-in: using pm_runtime_resume_and_get instead of pm_runtime_get_sync ASoC: uniphier: simplify the return expression of uniphier_aio_compr_set_params() ASoC: mediatek: simplify the return expression of mtk_dai_pcm_prepare() ASoC: pcm186x: simplify the return expression of pcm186x_power_off() Muralidhar Reddy (1): ASoC: SOF: Intel: pci-tgl: add ADL-PS support Nicola Lunghi (1): ASoC: wm8960: Add ACPI support Nicolas Frattaroli (2): ASoC: rk3328: fix disabling mclk on pclk probe failure ASoC: rk3328: fix disabling mclk on pclk probe failure Nícolas F. R. A. Prado (4): ASoC: dt-bindings: mediatek: mt8192: Add i2s-share properties ASoC: dt-bindings: rt1015p: Add #sound-dai-cells ASoC: dt-bindings: rt5682: Add #sound-dai-cells ASoC: dt-bindings: mediatek: mt8192: Drop i2s-share properties Peter Ujfalusi (66): ASoC: SOF: Intel: hda-loader: Use local snd_dma_buffer ASoC: SOF: Remove dmab and dmab_bdl from snd_sof_dev struct ASoC: SOF: ipc: Add max_payload_size field in struct snd_sof_ipc ASoC: SOF: ipc: Use ipc->max_payload_size for message/reply size checking ASoC: SOF: Drop support for mapped control data ASoC: SOF: ipc: Use msg->reply_data directly in snd_sof_ipc_get_reply() ASoC: SOF: Drop 'header' parameter from tx_message() API ASoC: SOF: Remove header from struct snd_sof_ipc_msg ASoC: SOF: ipc3-topology: Set scontrol->priv to NULL after freeing it ASoC: SOF: Add helper function to prepare and send an IPC message ASoC: SOF: Add high level IPC IO callback definitions to ipc_ops ASoC: SOF: ipc3: Implement the tx_msg IPC ops ASoC: SOF: ipc3: Use sof_ipc3_tx_msg() internally for message sending ASoC: SOF: ipc3: Implement the set_get_data IPC ops ASoC: SOF: ipc3: Implement the get_reply IPC ops ASoC: SOF: ipc3: Implement rx_msg IPC ops ASoC: SOF: ipc: Separate the ops checks by functions/topics ASoC: SOF: ipc: Add check for mandatory IPC message handling ops ASoC: SOF: ipc: Use the get_reply ops in snd_sof_ipc_get_reply() ASoC: SOF: ipc: Switch over to use the tx_msg and set_get_data ops ASoC: SOF: ipc: Switch over to use the rx_msg ops ASoC: SOF: Add widget_kcontrol_setup control ops for IPC3 ASoC: SOF: sof-audio: Use the widget_kcontrol_setup ops for kcontrol set up ASoC: SOF: ipc: Move the ipc_set_get_comp_data() local to ipc3-control ASoC: SOF: topology: Check w->sname validity once in sof_connect_dai_widget() ASoC: SOF: Intel: hda: Revisit IMR boot sequence ASoC: SOF: ipc3: Add local implementation for handling fw_ready message ASoC: SOF: Do not check for the fw_ready callback ASoC: SOF: amd: Do not set fw_ready callback ASoC: SOF: imx: Do not set fw_ready callback ASoC: SOF: Intel: Do not set fw_ready callback ASoC: SOF: loader: Remove the old fw_ready related code ASoC: SOF: Intel: Do not (wrongly) assume positive return value in IMR path ASoC: SOF: Introduce IPC dependent ops for firmware handling, loading ASoC: SOF: ipc3-loader: Implement firmware parsing and loading ASoC: SOF: ipc: Add check for fw_loader ops ASoC: SOF: loader: Switch to use the fw_loader ops ASoC: SOF: amd: renoir: Do not set the load_module ops ASoC: SOF: imx: Do not set the load_module ops ASoC: SOF: Intel: bdw/byt/pci-tng: Do not set the load_module ops ASoC: SOF: mediatek: mt8195: Do not set the load_module ops ASoC: SOF: loader: Remove snd_sof_parse_module_memcpy() as it is not used ASoC: SOF: loader: Call optional query_fw_configuration on first boot ASoC: SOF: ipc3-topology: Correct get_control_data for non bytes payload ASoC: SOF: ipc3: Remove the ipc3-ops.h header file ASoC: SOF: Add rx_data pointer to snd_sof_ipc_msg struct ASoC: SOF: Add initial header file for ipc4 ASoC: SOF: ipc4: Add support for mandatory message handling functionality ASoC: SOF: sof-client: Add API to get the maximum IPC payload size ASoC: SOF: ipc-msg-injector: Query the maximum IPC payload size ASoC: SOF: sof-client-probes: Query the maximum IPC payload size ASoC: SOF: sof-client: Add API to get the ipc_type ASoC: SOF: sof-client: Add support IPC4 message sending ASoC: SOF: ipc-msg-injector: Separate the message sending ASoC: SOF: ipc-msg-injector: Add support for IPC4 messages ASoC: SOF: sof-client: IPC flood test can only work with SOF_IPC ASoC: SOF: trace: The dtrace is only available with SOF_IPC ASoC: SOF: ipc-msg-injector: Cap the rmaining to count in IPC4 mode ASoC: SOF: Introduce IPC independent ops for firmware tracing support ASoC: SOF: Rename dtrace_is_supported flag to fw_trace_is_supported ASoC: SOF: Clone the trace code to ipc3-dtrace as fw_tracing implementation ASoC: SOF: Switch to IPC generic firmware tracing ASoC: SOF: ipc3-dtrace: Move host ops wrappers from generic header to private ASoC: SOF: Modify the host trace_init parameter list to include dmab ASoC: SOF: Introduce opaque storage of private data for firmware tracing ASoC: SOF: ipc3-dtrace: Move dtrace related variables local from sof_dev Pierre-Louis Bossart (39): ASoC: SOF: add definitions for different IPC types ASoC: SOF: add path indirection to each IPC type ASoC: SOF: add default IPC capability and file paths ASoC: SOF: add IPC-dependent file names ASoC: SOF: sof-pci-dev: add parameter to override firmware filename ASoC: SOF: sof-pci-dev: add parameter to override topology filename ASoC: SOF: sof-audio: preserve topology filename override in nocodec mode ASoC: SOF: remove const qualifier for 'struct snd_sof_dsp_ops' ASoC: SOF: Intel: hda: use common ops across platforms ASoC: SOF: Intel: hda: use BIT() macros for consistency ASoC: SOF: Intel: add IP identifier ASoC: soc-pcm: improve BE transition for PAUSE_RELEASE ASoC: soc-pcm: improve BE transition for TRIGGER_START ASoC: Intel: sof_ssp_amp: remove 'set-but-not-used' warning ASoC: SOF: sof-pci-dev: don't use the community key on APL Chromebooks ASoC: SOF: sof-audio: remove useless assignment ASoC: SOF: Intel: hda: simplify NHLT handling ASoC: soc-core: remove always-false path ASOC: SOF: Intel: hda-dai: consistent naming for HDA DAI and HDA link DMA ASoC: SOF: Intel: hda-dai: simplify hda_dai_widget_update() prototype ASoC: SOF: Intel: hda-dai: use snd_soc_dai_get_widget() helper ASoC: SOF: Intel: hda-dai: split link DMA and dai operations ASoC: SOF: Intel: hda-dai: regroup dai and link DMA operations ASoC: SOF: sof-audio: flag errors on pipeline teardown ASOC: SOF: Intel: hda-dai: add hda_dai_hw_free_ipc() helper ASoC: SOF: Intel: hda-dai: move code to deal with hda dai/dailink suspend ASoC: SOF: Intel: hda-dai: improve suspend case ASoC: SOF: Intel: hda-dai: reset dma_data and release stream ASoC: SOF: Intel: add helper for link DMA cleanups ASoC: SOF: Intel: hda-dai: protect hw_params against successive calls ASoC: SOF: control: use pm_runtime_resume_and_get() ASoC: SOF: debug: use pm_runtime_resume_and_get() ASoC: SOF: sof-client-ipc-msg-injector: use pm_runtime_resume_and_get() ASoC: Intel: atom: use pm_runtime_resume_and_get() ASoC: SOF: bump SOF_ABI_MINOR ASoC: codecs: rt715-sdca: remove useless assignment of ops ASoC: max98357a: remove dependency on GPIOLIB ASoC: rt1015p: remove dependency on GPIOLIB ASoC: SOF: sof-client-ipc-flood-test: use pm_runtime_resume_and_get() Piotr Maziarz (1): ASoC: Intel: avs: Replace link_mask usage with i2s_link_mask Rander Wang (1): ASOC: SOF: pci: add ipc_type override for Intel IPC4 tests Ranjani Sridharan (22): ASoC: SOF: Intel: move HDA_CL_STREAM_FORMAT definition to header ASoC: SOF: Intel: hda: define check_ipc_irq op ASoC: SOF: Intel: hda: Define rom_status_reg in sof_intel_dsp_desc ASoC: SOF: Intel: hda: expose some codeloader functions ASoC: SOF: remove incorrect clearing of prepared flag ASoC: SOF: Intel: Add IPC-specific dai ops for IPC3 ASoC: SOF: Add a new op to set up volume table ASoC: SOF: sof-audio: reset route status before freeing widget ASoC: SOF: sof-audio: Set up widgets from source to sink ASoC: SOF: pcm: Move the call to snd_sof_pcm_platform_hw_params() ASoC: SOF: expose a couple of functions ASoC: SOF: Add a route_free op in struct sof_ipc_tplg_ops ASoC: SOF: Add two new fields to struct snd_sof_widget ASoC: SOF: pcm: remove unnecessary function declaration ASoC: SOF: topology: Skip parsing DAI link tokens if not needed ASoC: SOF: clarify use of widget complete flag ASoC: SOF: Add a prepare op to IPC topology widget ops ASoC: SOF: Intel: HDA: Set IPC4-specific DSP ops for CaVS platforms ASoC: SOF: Add IPC4 private header ASoC: SOF: Add header for IPC4 manifest ASoC: SOF: IPC4: Add FW loader ops ASoC: SOF: Intel: HDA: Set up sof_ipc4_fw_data for IPC4 Raphael-Xu (2): ASoC: rename tas2764 to tas27xx ASoC: update to support either TAS2764 or TAS2780 Ricard Wanderlof (1): ASoC: adau1761: Add ADAU1761-as-ADAU1361 compatibility mode Richard Fitzgerald (9): ASoC: soc.h: Add SOC_SINGLE_S_TLV() macro ASoC: soc-utils: Add helper to calculate BCLK from TDM info ASoC: soc-utils: Add kunit test for snd_soc_tdm_params_to_bclk() ASoC: dt-bindings: cs35l45: Cirrus Logic CS35L45 Smart Amp ASoC: cs35l45: Make exports namespaced ASoC: SOF: Kconfig: Make SND_SOC_SOF_HDA_PROBES tristate ASoC: soc.h: Add SOC_SINGLE_S_EXT_TLV macro firmware: cirrus: cs_dsp: Avoid padding bytes in cs_dsp_coeff_ctl ASoC: cs35l45: Enable BOOST Rikard Falkeborn (1): ASoC: samsung: Constify snd_soc_dapm_{route,widget} structs Ryan Lee (2): ASoC: max98396: add amplifier driver ASoC: dt-bindings: max98396: add amplifier driver Sameer Pujar (3): ASoC: tegra: Add binding doc for ASRC module ASoC: tegra: Add Tegra186 based ASRC driver ASoC: tegra: AHUB routes for ASRC module Sascha Hauer (22): ASoC: fsl_micfil: Drop unnecessary register read ASoC: fsl_micfil: Drop unused register read ASoC: fsl_micfil: drop fsl_micfil_set_mclk_rate() ASoC: fsl_micfil: do not define SHIFT/MASK for single bits ASoC: fsl_micfil: use GENMASK to define register bit fields ASoC: fsl_micfil: use clear/set bits ASoC: fsl_micfil: drop error messages from failed register accesses ASoC: fsl_micfil: drop unused variables dmaengine: imx: Move header to include/dma/ dmaengine: imx-sdma: error out on unsupported transfer types dmaengine: imx-sdma: Add multi fifo support ASoC: fsl_micfil: add multi fifo support ASoC: fsl_micfil: use define for OSR default value ASoC: fsl_micfil: Drop get_pdm_clk() ASoC: fsl_micfil: simplify clock setting ASoC: fsl_micfil: rework quality setting ASoC: fsl_micfil: drop unused include ASoC: fsl_micfil: drop only once used defines ASoC: fsl_micfil: drop support for undocumented property ASoC: fsl_micfil: fold fsl_set_clock_params() into its only user ASoC: fsl_micfil: Remove debug message ASoC: soc-generic-dmaengine-pcm: Use default config when none is given Shengjiu Wang (8): ASoC: dmic: Add support for DSD data format ASoC: fsl_micfil: fix the naming style for mask definition ASoC: fsl_micfil: Add support for i.MX8MPlus ASoC: dt-bindings: fsl,micfil: Add compatible string for imx8mp ASoC: fsl_ssi: Add support multi fifo script ASoC: fsl_sai: Add support for i.MX8MM ASoC: fsl_sai: Add support for i.MX8M Plus ASoC: fsl_sai: Add support for i.MX8ULP Shuming Fan (1): ASoC: rt1308-sdw: add the default value of register 0xc320 Simon Trimmer (1): ASoC: soc.h: Introduce SOC_DOUBLE_R_S_EXT_TLV() macro Srinivasa Rao Mandadapu (6): ASoC: codecs: Fix error handling in power domain init and exit handlers ASoC: qcom: dt-bindings: Update bindings for clocks in lpass digital codes ASoC: dt-bindings: wcd938x: Add mic bias supply property ASoC: dt-bindings: lpass-cpu: Update clocks and power domain names for sc7280 platform ASoC: qcom: lpass-platform: Update memremap flag to MEMREMAP_WC ASoC: qcom: SC7280: Update machine driver startup, shutdown callbacks Stephen Kitt (36): ASoC: ad1*, ada*, ssm*: use i2c_match_id and simple i2c probe ASoC: da[79]*: use simple i2c probe function ASoC: da7218: use i2c_match_id and simple i2c probe ASoC: ada*, ssm*: use simple i2c probe function ASoC: wm*: use i2c_match_id and simple i2c probe ASoC: cs*: use simple i2c probe function ASoC: mt6660: use simple i2c probe function ASoC: lm49xxx: use simple i2c probe function ASoC: wm*: use simple i2c probe function ASoC: tscs*: use simple i2c probe function ASoC: rt*: use simple i2c probe function ASoC: pcm3060: use simple i2c probe function ASoC: rt1016: enable building ASoC: sgtl5000: use simple i2c probe function ASoC: wm8731: use simple i2c probe function ASoC: ak4*: use simple i2c probe function ASoC: cx2072x: use simple i2c probe function ASoC: es83*: use simple i2c probe function ASoC: lm4857: use simple i2c probe function ASoC: max9*: use simple i2c probe function ASoC: ml26124: use simple i2c probe function ASoC: nau8*: use simple i2c probe function ASoC: pcm*: use simple i2c probe function ASoC: sta*: use simple i2c probe function ASoC: tas*: use simple i2c probe function ASoC: tda7419: use simple i2c probe function ASoC: tlv320*: use simple i2c probe function ASoC: ts3a227e: use simple i2c probe function ASoC: uda1380: use simple i2c probe function ASoC: ak*: use i2c_match_id and simple i2c probe ASoC: alc56*: use i2c_match_id and simple i2c probe ASoC: max980*: use i2c_match_id and simple i2c probe ASoC: pcm186x: use i2c_match_id and simple i2c probe ASoC: tas*: use i2c_match_id and simple i2c probe ASoC: tlv320*: use i2c_match_id and simple i2c probe ASoC: tpa6130: use i2c_match_id and simple i2c probe Steve Lee (2): ASoC: max98390: Add reset gpio control ASoC: dt-bindings: max98390: add reset gpio bindings Takashi Iwai (1): ASoC: intel: atom: Remove superfluous flush_scheduled_work() TingHan Shen (1): firmware: mediatek: Add adsp ipc protocol interface Tinghan Shen (6): ASoC: SOF: mediatek: Add mt8186 hardware support ASoC: SOF: mediatek: Add mt8186 sof fw loader and dsp ops ASoC: SOF: mediatek: Add mt8186 dsp clock support ASoC: SOF: mediatek: Add DSP system PM callback for mt8186 ASoC: SOF: mediatek: Fix allyesconfig build error ASoC: SOF: mediatek: Add mt8186 ipc support Trevor Wu (6): ASoC: mediatek: mt8195: revise mt8195-mt6359-rt1019-rt5682.c ASoC: mediatek: mt8195: merge machine driver ASoC: dt-bindings: mediatek: mt8195: merge mt8195 machine yaml ASoC: mediatek: mt8195: rename card controls ASoC: mediatek: mt8195: add machine support for max98390 and rt5682 ASoC: dt-bindings: mediatek: mt8195: support mt8195-mt6359-max98390-rt5682 Uwe Kleine-König (2): ASoC: tas6424: Return zero in remove callback ASoC: pcm1789: Make pcm1789_common_exit() return void V sujith kumar Reddy (2): ASoC: amd: acp: Set Speaker enable/disable pin through rt1019 codec driver ASoC: amd: acp: Set Speaker enable/disable pin through rt1019 codec driver. Vamshi Krishna (1): ASoC: Intel: sof_rt5682: Add support for adl_rt1019_rt5682s Vijendar Mukunda (1): ASoC: amd: yc: add new YC platform varaint support Wan Jiabing (1): ASoC: SOF: mediatek: remove duplicate include in mt8195.c YC Hung (7): ASoC: SOF: OF: Add shutdown callback for SOF OF device ASoC: SOF: mediatek: mt8195 add shutdown callback ASoC: SOF: mediatek: mt8195 change ipc timeout as 1 second ASoC: dsp: mediatek: add mt8195 dsp document ASoC: SOF: ipc: introduce cont_update_posn in sof_ipc_stream_params struct ASoC: SOF: mediatek: Add mediatek common debug dump ASoC: SOF: mediatek: Add mt8195 debug dump Yang Yingliang (3): ASoC: SOF: sof-pci-dev: fix missing pci_release_regions() on error in sof_pci_probe() ASoC: samsung: spdif: remove unnecessary check of mem_res ASoC: wm2000: fix missing clk_disable_unprepare() on error in wm2000_anc_transition() Yihao Han (1): ASoC: ux500: mop500: Check before clk_put() not needed YueHaibing (4): ASoC: tegra186_asrc: mark runtime-pm functions as __maybe_unused ASoC: mediatek: mt8195: Fix build warning without CONFIG_OF ASoC: amd: vangogh: Remove duplicate include files ASoC: codecs: lpass: Fix passing zero to 'PTR_ERR' Zhen Ni (1): ASoC: Intel: boards: Use temporary variable for struct device Zheng Bin (1): ASoC: SOF: amd: add missing platform_device_unregister in acp_pci_rn_probe Zheyu Ma (9): ASoC: SOF: Intel: Check the bar size before remapping ASoC: tlv320adcx140: Register a callback to disable the regulator_disable ASoC: cs35l36: Fix the error handling of cs35l36_i2c_probe() ASoC: rt5645: Fix the error handling of rt5645_i2c_probe() ASoC: tas571x: Fix the error handling of tas571x_i2c_probe() ASoC: tas6424: Fix the error handling of tas6424_i2c_probe() ASoC: wm8903: Fix the error handling of wm8903_i2c_probe() ASoC: cs42l46: Fix the error handling of cs42l56_i2c_probe() ASoC: cs42l56: Fix the error handling of cs42l56_i2c_probe() tangmeng (1): ASoC: SOF: Remove redundant return statements .../bindings/dsp/mediatek,mt8195-dsp.yaml | 105 ++ .../devicetree/bindings/sound/adi,max98396.yaml | 79 + .../devicetree/bindings/sound/cirrus,cs35l45.yaml | 75 + .../devicetree/bindings/sound/fsl,micfil.txt | 1 + .../devicetree/bindings/sound/maxim,max98390.yaml | 5 + .../sound/mt8192-mt6359-rt1015-rt5682.yaml | 32 + .../sound/mt8195-mt6359-rt1011-rt5682.yaml | 51 - ...t6359-rt1019-rt5682.yaml => mt8195-mt6359.yaml} | 9 +- .../bindings/sound/nvidia,tegra186-asrc.yaml | 81 + .../bindings/sound/nvidia,tegra210-ahub.yaml | 4 + .../devicetree/bindings/sound/qcom,lpass-cpu.yaml | 17 +- .../bindings/sound/qcom,lpass-rx-macro.yaml | 17 +- .../bindings/sound/qcom,lpass-tx-macro.yaml | 17 +- .../bindings/sound/qcom,lpass-va-macro.yaml | 11 +- .../devicetree/bindings/sound/qcom,wcd938x.yaml | 3 + .../devicetree/bindings/sound/realtek,rt1015p.yaml | 3 + Documentation/devicetree/bindings/sound/rt5682.txt | 2 + .../bindings/sound/{tas2764.yaml => tas27xx.yaml} | 17 +- .../devicetree/bindings/sound/wlf,wm8731.yaml | 97 ++ .../devicetree/bindings/sound/wlf,wm8940.yaml | 57 + Documentation/devicetree/bindings/sound/wm8731.txt | 27 - MAINTAINERS | 2 + arch/arm/configs/multi_v5_defconfig | 3 +- drivers/dma/imx-dma.c | 2 +- drivers/dma/imx-sdma.c | 76 +- drivers/firmware/Kconfig | 9 + drivers/firmware/Makefile | 1 + drivers/firmware/mtk-adsp-ipc.c | 157 ++ drivers/mmc/host/mxcmmc.c | 2 +- drivers/spi/spi-fsl-lpspi.c | 2 +- drivers/spi/spi-imx.c | 2 +- drivers/staging/greybus/audio_codec.c | 12 +- drivers/tty/serial/imx.c | 2 +- drivers/video/fbdev/mx3fb.c | 2 +- include/dt-bindings/sound/cs35l45.h | 20 + .../{platform_data/dma-imx.h => dma/imx-dma.h} | 26 +- include/linux/firmware/cirrus/cs_dsp.h | 28 +- include/linux/firmware/mediatek/mtk-adsp-ipc.h | 65 + include/sound/cs35l41.h | 1 - include/sound/intel-dsp-config.h | 3 +- include/sound/intel-nhlt.h | 6 +- include/sound/soc-acpi.h | 2 + include/sound/soc-card.h | 7 +- include/sound/soc-component.h | 9 + include/sound/soc-dpcm.h | 2 + include/sound/soc.h | 27 +- include/sound/sof.h | 23 +- include/sound/sof/ext_manifest4.h | 119 ++ include/sound/sof/ipc4/header.h | 460 ++++++ include/sound/sof/stream.h | 3 +- include/uapi/sound/intel/avs/tokens.h | 126 ++ include/uapi/sound/sof/abi.h | 4 +- sound/soc/Kconfig | 9 +- sound/soc/Makefile | 5 + sound/soc/amd/acp-da7219-max98357a.c | 4 +- sound/soc/amd/acp-rt5645.c | 2 +- sound/soc/amd/acp/Kconfig | 6 +- sound/soc/amd/acp/acp-legacy-mach.c | 22 +- sound/soc/amd/acp/acp-mach-common.c | 29 +- sound/soc/amd/acp/acp-mach.h | 9 +- sound/soc/amd/acp/acp-sof-mach.c | 22 +- sound/soc/amd/acp3x-rt5682-max9836.c | 2 +- sound/soc/amd/vangogh/acp5x-mach.c | 10 +- sound/soc/amd/yc/acp6x-mach.c | 42 +- sound/soc/amd/yc/pci-acp6x.c | 9 +- sound/soc/atmel/Kconfig | 4 +- sound/soc/atmel/atmel-classd.c | 1 - sound/soc/atmel/atmel-pdmic.c | 1 - sound/soc/atmel/sam9g20_wm8731.c | 16 +- sound/soc/au1x/Kconfig | 2 +- sound/soc/codecs/Kconfig | 67 +- sound/soc/codecs/Makefile | 16 + sound/soc/codecs/ad193x-i2c.c | 6 +- sound/soc/codecs/adau1372-i2c.c | 4 +- sound/soc/codecs/adau1372.c | 1 + sound/soc/codecs/adau1373.c | 5 +- sound/soc/codecs/adau1701.c | 5 +- sound/soc/codecs/adau1761-i2c.c | 8 +- sound/soc/codecs/adau1761.c | 86 +- sound/soc/codecs/adau1781-i2c.c | 8 +- sound/soc/codecs/adau17x1.c | 20 +- sound/soc/codecs/adau17x1.h | 1 + sound/soc/codecs/adau1977-i2c.c | 8 +- sound/soc/codecs/adau7118-i2c.c | 5 +- sound/soc/codecs/adav803.c | 5 +- sound/soc/codecs/ak4118.c | 5 +- sound/soc/codecs/ak4535.c | 5 +- sound/soc/codecs/ak4613.c | 377 ++++- sound/soc/codecs/ak4641.c | 5 +- sound/soc/codecs/ak4642.c | 8 +- sound/soc/codecs/ak4671.c | 5 +- sound/soc/codecs/alc5623.c | 24 +- sound/soc/codecs/alc5632.c | 20 +- sound/soc/codecs/cros_ec_codec.c | 7 +- sound/soc/codecs/cs35l32.c | 5 +- sound/soc/codecs/cs35l33.c | 5 +- sound/soc/codecs/cs35l34.c | 5 +- sound/soc/codecs/cs35l35.c | 5 +- sound/soc/codecs/cs35l36.c | 7 +- sound/soc/codecs/cs35l41-i2c.c | 5 +- sound/soc/codecs/cs35l41-lib.c | 16 +- sound/soc/codecs/cs35l41.c | 2 + sound/soc/codecs/cs35l45-i2c.c | 74 + sound/soc/codecs/cs35l45-spi.c | 74 + sound/soc/codecs/cs35l45-tables.c | 202 +++ sound/soc/codecs/cs35l45.c | 690 +++++++++ sound/soc/codecs/cs35l45.h | 217 +++ sound/soc/codecs/cs4234.c | 5 +- sound/soc/codecs/cs4265.c | 5 +- sound/soc/codecs/cs4270.c | 20 +- sound/soc/codecs/cs4271-i2c.c | 5 +- sound/soc/codecs/cs42l42.c | 5 +- sound/soc/codecs/cs42l51-i2c.c | 5 +- sound/soc/codecs/cs42l51.c | 7 +- sound/soc/codecs/cs42l52.c | 5 +- sound/soc/codecs/cs42l56.c | 7 +- sound/soc/codecs/cs42l73.c | 5 +- sound/soc/codecs/cs42xx8-i2c.c | 5 +- sound/soc/codecs/cs43130.c | 23 +- sound/soc/codecs/cs43130.h | 151 +- sound/soc/codecs/cs4341.c | 5 +- sound/soc/codecs/cs4349.c | 14 +- sound/soc/codecs/cs53l30.c | 5 +- sound/soc/codecs/cx2072x.c | 6 +- sound/soc/codecs/da7210.c | 5 +- sound/soc/codecs/da7213.c | 5 +- sound/soc/codecs/da7218.c | 19 +- sound/soc/codecs/da7219.c | 5 +- sound/soc/codecs/da732x.c | 5 +- sound/soc/codecs/da9055.c | 5 +- sound/soc/codecs/dmic.c | 5 +- sound/soc/codecs/es8316.c | 5 +- sound/soc/codecs/es8328-i2c.c | 5 +- sound/soc/codecs/hdac_hda.c | 15 +- sound/soc/codecs/hdmi-codec.c | 15 +- sound/soc/codecs/isabelle.c | 5 +- sound/soc/codecs/lm4857.c | 5 +- sound/soc/codecs/lm49453.c | 5 +- sound/soc/codecs/lochnagar-sc.c | 5 +- sound/soc/codecs/lpass-macro-common.c | 35 +- sound/soc/codecs/max9768.c | 5 +- sound/soc/codecs/max98088.c | 21 +- sound/soc/codecs/max98090.c | 33 +- sound/soc/codecs/max98095.c | 19 +- sound/soc/codecs/max98371.c | 5 +- sound/soc/codecs/max98373-i2c.c | 5 +- sound/soc/codecs/max98390.c | 17 +- sound/soc/codecs/max98396.c | 1637 ++++++++++++++++++++ sound/soc/codecs/max98396.h | 305 ++++ sound/soc/codecs/max9850.c | 5 +- sound/soc/codecs/max98504.c | 6 +- sound/soc/codecs/max98520.c | 4 +- sound/soc/codecs/max9867.c | 5 +- sound/soc/codecs/max9877.c | 5 +- sound/soc/codecs/max98925.c | 5 +- sound/soc/codecs/max98926.c | 5 +- sound/soc/codecs/max98927.c | 5 +- sound/soc/codecs/ml26124.c | 5 +- sound/soc/codecs/mt6351.c | 10 +- sound/soc/codecs/mt6358.c | 10 +- sound/soc/codecs/mt6359.c | 10 +- sound/soc/codecs/mt6660.c | 6 +- sound/soc/codecs/nau8540.c | 5 +- sound/soc/codecs/nau8810.c | 5 +- sound/soc/codecs/nau8821.c | 5 +- sound/soc/codecs/nau8822.c | 5 +- sound/soc/codecs/nau8824.c | 5 +- sound/soc/codecs/nau8825.c | 5 +- sound/soc/codecs/pcm1681.c | 5 +- sound/soc/codecs/pcm1789-i2c.c | 9 +- sound/soc/codecs/pcm1789.c | 4 +- sound/soc/codecs/pcm1789.h | 2 +- sound/soc/codecs/pcm179x-i2c.c | 5 +- sound/soc/codecs/pcm186x-i2c.c | 24 +- sound/soc/codecs/pcm186x.c | 7 +- sound/soc/codecs/pcm3060-i2c.c | 5 +- sound/soc/codecs/pcm3060.c | 1 + sound/soc/codecs/pcm3168a-i2c.c | 5 +- sound/soc/codecs/pcm512x-i2c.c | 5 +- sound/soc/codecs/rk3328_codec.c | 2 +- sound/soc/codecs/rt1011.c | 5 +- sound/soc/codecs/rt1015.c | 5 +- sound/soc/codecs/rt1016.c | 5 +- sound/soc/codecs/rt1019.c | 8 +- sound/soc/codecs/rt1305.c | 5 +- sound/soc/codecs/rt1308-sdw.c | 1 + sound/soc/codecs/rt1308-sdw.h | 1 + sound/soc/codecs/rt1308.c | 5 +- sound/soc/codecs/rt1316-sdw.c | 1 + sound/soc/codecs/rt274.c | 5 +- sound/soc/codecs/rt286.c | 5 +- sound/soc/codecs/rt298.c | 5 +- sound/soc/codecs/rt5514.c | 7 +- sound/soc/codecs/rt5616.c | 5 +- sound/soc/codecs/rt5631.c | 5 +- sound/soc/codecs/rt5640.c | 5 +- sound/soc/codecs/rt5645.c | 16 +- sound/soc/codecs/rt5651.c | 5 +- sound/soc/codecs/rt5659.c | 5 +- sound/soc/codecs/rt5660.c | 5 +- sound/soc/codecs/rt5663.c | 5 +- sound/soc/codecs/rt5665.c | 5 +- sound/soc/codecs/rt5668.c | 5 +- sound/soc/codecs/rt5670.c | 5 +- sound/soc/codecs/rt5682-i2c.c | 5 +- sound/soc/codecs/rt5682s.c | 32 +- sound/soc/codecs/rt5682s.h | 6 +- sound/soc/codecs/rt700.c | 1 + sound/soc/codecs/rt711-sdca.c | 1 + sound/soc/codecs/rt711.c | 1 + sound/soc/codecs/rt715-sdca-sdw.c | 2 - sound/soc/codecs/rt715-sdca.c | 1 + sound/soc/codecs/rt715.c | 1 + sound/soc/codecs/rt9120.c | 114 +- sound/soc/codecs/sdw-mockup.c | 1 + sound/soc/codecs/sgtl5000.c | 5 +- sound/soc/codecs/ssm2518.c | 5 +- sound/soc/codecs/ssm2602-i2c.c | 8 +- sound/soc/codecs/ssm4567.c | 5 +- sound/soc/codecs/sta32x.c | 14 +- sound/soc/codecs/sta350.c | 14 +- sound/soc/codecs/sta529.c | 5 +- sound/soc/codecs/tas2552.c | 5 +- sound/soc/codecs/tas2562.c | 25 +- sound/soc/codecs/tas2764.c | 5 +- sound/soc/codecs/tas2770.c | 5 +- sound/soc/codecs/tas5086.c | 5 +- sound/soc/codecs/tas571x.c | 17 +- sound/soc/codecs/tas5720.c | 21 +- sound/soc/codecs/tas6424.c | 17 +- sound/soc/codecs/tda7419.c | 5 +- sound/soc/codecs/tlv320adc3xxx.c | 32 +- sound/soc/codecs/tlv320adcx140.c | 16 +- sound/soc/codecs/tlv320aic23-i2c.c | 5 +- sound/soc/codecs/tlv320aic31xx.c | 32 +- sound/soc/codecs/tlv320aic32x4-i2c.c | 11 +- sound/soc/codecs/tlv320aic3x-i2c.c | 25 +- sound/soc/codecs/tlv320dac33.c | 5 +- sound/soc/codecs/tpa6130a2.c | 21 +- sound/soc/codecs/ts3a227e.c | 5 +- sound/soc/codecs/tscs42xx.c | 5 +- sound/soc/codecs/tscs454.c | 17 +- sound/soc/codecs/uda1380.c | 5 +- sound/soc/codecs/wcd9335.c | 1 + sound/soc/codecs/wcd934x.c | 1 + sound/soc/codecs/wcd938x.c | 1 + sound/soc/codecs/wm1250-ev1.c | 5 +- sound/soc/codecs/wm2000.c | 12 +- sound/soc/codecs/wm2200.c | 5 +- sound/soc/codecs/wm5100.c | 5 +- sound/soc/codecs/wm8510.c | 5 +- sound/soc/codecs/wm8523.c | 5 +- sound/soc/codecs/wm8580.c | 5 +- sound/soc/codecs/wm8711.c | 5 +- sound/soc/codecs/wm8728.c | 5 +- sound/soc/codecs/wm8731-i2c.c | 68 + sound/soc/codecs/wm8731-spi.c | 59 + sound/soc/codecs/wm8731.c | 267 +--- sound/soc/codecs/wm8731.h | 27 + sound/soc/codecs/wm8737.c | 5 +- sound/soc/codecs/wm8741.c | 5 +- sound/soc/codecs/wm8750.c | 5 +- sound/soc/codecs/wm8753.c | 5 +- sound/soc/codecs/wm8776.c | 5 +- sound/soc/codecs/wm8804-i2c.c | 5 +- sound/soc/codecs/wm8900.c | 5 +- sound/soc/codecs/wm8903.c | 7 +- sound/soc/codecs/wm8904.c | 9 +- sound/soc/codecs/wm8940.c | 12 +- sound/soc/codecs/wm8955.c | 5 +- sound/soc/codecs/wm8960.c | 20 +- sound/soc/codecs/wm8961.c | 5 +- sound/soc/codecs/wm8962.c | 11 +- sound/soc/codecs/wm8971.c | 5 +- sound/soc/codecs/wm8974.c | 5 +- sound/soc/codecs/wm8978.c | 5 +- sound/soc/codecs/wm8983.c | 5 +- sound/soc/codecs/wm8985.c | 8 +- sound/soc/codecs/wm8988.c | 5 +- sound/soc/codecs/wm8990.c | 5 +- sound/soc/codecs/wm8991.c | 5 +- sound/soc/codecs/wm8993.c | 5 +- sound/soc/codecs/wm8995.c | 5 +- sound/soc/codecs/wm8996.c | 5 +- sound/soc/codecs/wm9081.c | 5 +- sound/soc/codecs/wm9090.c | 6 +- sound/soc/codecs/wsa881x.c | 1 + sound/soc/fsl/fsl_asrc.c | 8 +- sound/soc/fsl/fsl_asrc_dma.c | 2 +- sound/soc/fsl/fsl_easrc.h | 2 +- sound/soc/fsl/fsl_esai.c | 6 +- sound/soc/fsl/fsl_micfil.c | 382 ++--- sound/soc/fsl/fsl_micfil.h | 269 +--- sound/soc/fsl/fsl_sai.c | 43 +- sound/soc/fsl/fsl_sai.h | 1 + sound/soc/fsl/fsl_ssi.c | 34 +- sound/soc/fsl/imx-es8328.c | 2 +- sound/soc/fsl/imx-hdmi.c | 9 +- sound/soc/fsl/imx-pcm.h | 2 +- sound/soc/fsl/imx-sgtl5000.c | 14 +- sound/soc/fsl/imx-ssi.h | 2 +- sound/soc/generic/audio-graph-card2.c | 7 +- sound/soc/generic/simple-card-utils.c | 5 +- sound/soc/img/img-i2s-in.c | 6 +- sound/soc/img/img-parallel-out.c | 6 +- sound/soc/img/img-spdif-in.c | 6 +- sound/soc/img/img-spdif-out.c | 6 +- sound/soc/intel/Kconfig | 9 +- sound/soc/intel/atom/sst/sst.c | 1 - sound/soc/intel/atom/sst/sst_drv_interface.c | 15 +- sound/soc/intel/avs/Makefile | 8 +- sound/soc/intel/avs/apl.c | 250 +++ sound/soc/intel/avs/avs.h | 102 ++ sound/soc/intel/avs/board_selection.c | 501 ++++++ sound/soc/intel/avs/core.c | 631 ++++++++ sound/soc/intel/avs/dsp.c | 27 +- sound/soc/intel/avs/ipc.c | 253 ++- sound/soc/intel/avs/loader.c | 84 + sound/soc/intel/avs/messages.c | 35 +- sound/soc/intel/avs/messages.h | 51 + sound/soc/intel/avs/path.c | 1005 ++++++++++++ sound/soc/intel/avs/path.h | 72 + sound/soc/intel/avs/pcm.c | 1182 ++++++++++++++ sound/soc/intel/avs/registers.h | 8 + sound/soc/intel/avs/skl.c | 125 ++ sound/soc/intel/avs/topology.c | 1598 +++++++++++++++++++ sound/soc/intel/avs/topology.h | 194 +++ sound/soc/intel/avs/trace.c | 33 + sound/soc/intel/avs/trace.h | 154 ++ sound/soc/intel/avs/utils.c | 23 + sound/soc/intel/boards/bdw-rt5650.c | 6 +- sound/soc/intel/boards/bdw-rt5677.c | 4 +- sound/soc/intel/boards/broadwell.c | 2 +- sound/soc/intel/boards/bxt_da7219_max98357a.c | 5 +- sound/soc/intel/boards/bxt_rt298.c | 5 +- sound/soc/intel/boards/bytcht_cx2072x.c | 10 +- sound/soc/intel/boards/bytcht_es8316.c | 8 +- sound/soc/intel/boards/bytcr_rt5640.c | 36 +- sound/soc/intel/boards/bytcr_rt5651.c | 7 +- sound/soc/intel/boards/bytcr_wm5102.c | 6 +- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 28 +- sound/soc/intel/boards/cht_bsw_nau8824.c | 4 +- sound/soc/intel/boards/cht_bsw_rt5645.c | 6 +- sound/soc/intel/boards/cht_bsw_rt5672.c | 12 +- sound/soc/intel/boards/cml_rt1011_rt5682.c | 5 +- sound/soc/intel/boards/glk_rt5682_max98357a.c | 5 +- sound/soc/intel/boards/kbl_da7219_max98357a.c | 5 +- sound/soc/intel/boards/kbl_da7219_max98927.c | 5 +- sound/soc/intel/boards/kbl_rt5660.c | 15 +- sound/soc/intel/boards/kbl_rt5663_max98927.c | 7 +- .../soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 7 +- sound/soc/intel/boards/skl_hda_dsp_common.c | 8 +- sound/soc/intel/boards/skl_nau88l25_max98357a.c | 6 +- sound/soc/intel/boards/skl_nau88l25_ssm4567.c | 6 +- sound/soc/intel/boards/skl_rt286.c | 5 +- sound/soc/intel/boards/sof_cs42l42.c | 5 +- sound/soc/intel/boards/sof_da7219_max98373.c | 2 +- sound/soc/intel/boards/sof_es8336.c | 8 +- sound/soc/intel/boards/sof_nau8825.c | 2 +- sound/soc/intel/boards/sof_realtek_common.c | 39 + sound/soc/intel/boards/sof_realtek_common.h | 5 + sound/soc/intel/boards/sof_rt5682.c | 18 +- sound/soc/intel/boards/sof_sdw_rt5682.c | 14 +- sound/soc/intel/boards/sof_sdw_rt700.c | 14 +- sound/soc/intel/boards/sof_sdw_rt711.c | 14 +- sound/soc/intel/boards/sof_sdw_rt711_sdca.c | 14 +- sound/soc/intel/boards/sof_ssp_amp.c | 25 +- sound/soc/intel/catpt/messages.h | 4 +- sound/soc/intel/common/soc-acpi-intel-adl-match.c | 7 + sound/soc/mediatek/Kconfig | 27 +- sound/soc/mediatek/mt2701/mt2701-wm8960.c | 9 +- sound/soc/mediatek/mt8173/mt8173-max98090.c | 13 +- sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c | 2 +- sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c | 2 +- sound/soc/mediatek/mt8173/mt8173-rt5650.c | 4 +- sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c | 5 +- .../mt8183/mt8183-mt6358-ts3a227-max98357.c | 5 +- .../mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c | 203 ++- sound/soc/mediatek/mt8195/Makefile | 3 +- sound/soc/mediatek/mt8195/mt8195-dai-pcm.c | 8 +- .../mediatek/mt8195/mt8195-mt6359-rt1011-rt5682.c | 1198 -------------- ...8195-mt6359-rt1019-rt5682.c => mt8195-mt6359.c} | 871 +++++++---- sound/soc/mxs/mxs-saif.c | 1 + sound/soc/pxa/Kconfig | 4 +- sound/soc/pxa/hx4700.c | 6 +- sound/soc/pxa/palm27x.c | 7 +- sound/soc/pxa/ttc-dkb.c | 14 +- sound/soc/pxa/z2.c | 7 +- sound/soc/qcom/Kconfig | 2 + sound/soc/qcom/apq8016_sbc.c | 2 +- sound/soc/qcom/lpass-platform.c | 2 +- sound/soc/qcom/sc7180.c | 4 +- sound/soc/qcom/sc7280.c | 99 +- sound/soc/qcom/sdm845.c | 2 +- sound/soc/qcom/sm8250.c | 2 +- sound/soc/rockchip/rk3288_hdmi_analog.c | 8 +- sound/soc/rockchip/rk3399_gru_sound.c | 16 +- sound/soc/rockchip/rockchip_max98090.c | 16 +- sound/soc/rockchip/rockchip_rt5645.c | 2 +- sound/soc/samsung/aries_wm8994.c | 6 +- sound/soc/samsung/bells.c | 4 +- sound/soc/samsung/h1940_uda1380.c | 3 +- sound/soc/samsung/littlemill.c | 6 +- sound/soc/samsung/lowland.c | 13 +- sound/soc/samsung/midas_wm1811.c | 2 +- sound/soc/samsung/rx1950_uda1380.c | 3 +- sound/soc/samsung/smartq_wm8987.c | 8 +- sound/soc/samsung/spdif.c | 3 +- sound/soc/samsung/speyside.c | 14 +- sound/soc/samsung/tobermory.c | 12 +- sound/soc/sh/Kconfig | 2 +- sound/soc/sh/rcar/core.c | 17 +- sound/soc/sh/rcar/dma.c | 9 +- sound/soc/sh/rcar/rsnd.h | 2 +- sound/soc/sh/rcar/src.c | 7 +- sound/soc/sh/rcar/ssi.c | 14 +- sound/soc/sh/rcar/ssiu.c | 11 +- sound/soc/sh/rz-ssi.c | 24 +- sound/soc/soc-card.c | 56 +- sound/soc/soc-component.c | 16 +- sound/soc/soc-core.c | 98 +- sound/soc/soc-dapm.c | 2 - sound/soc/soc-generic-dmaengine-pcm.c | 57 +- sound/soc/soc-jack.c | 2 +- sound/soc/soc-pcm.c | 38 +- sound/soc/soc-topology.c | 455 +++--- sound/soc/soc-utils-test.c | 186 +++ sound/soc/soc-utils.c | 45 + sound/soc/sof/Makefile | 4 +- sound/soc/sof/amd/acp-trace.c | 4 +- sound/soc/sof/amd/acp.c | 66 +- sound/soc/sof/amd/acp.h | 10 +- sound/soc/sof/amd/pci-rn.c | 15 +- sound/soc/sof/amd/renoir.c | 6 +- sound/soc/sof/compress.c | 8 +- sound/soc/sof/control.c | 36 +- sound/soc/sof/core.c | 28 +- sound/soc/sof/debug.c | 9 +- sound/soc/sof/imx/imx8.c | 40 +- sound/soc/sof/imx/imx8m.c | 20 +- sound/soc/sof/intel/Kconfig | 2 +- sound/soc/sof/intel/Makefile | 2 +- sound/soc/sof/intel/apl.c | 143 +- sound/soc/sof/intel/bdw.c | 21 +- sound/soc/sof/intel/byt.c | 56 +- sound/soc/sof/intel/cnl.c | 212 ++- sound/soc/sof/intel/hda-common-ops.c | 106 ++ sound/soc/sof/intel/hda-dai.c | 434 ++++-- sound/soc/sof/intel/hda-dsp.c | 52 +- sound/soc/sof/intel/hda-ipc.c | 82 + sound/soc/sof/intel/hda-loader.c | 123 +- sound/soc/sof/intel/hda-trace.c | 4 +- sound/soc/sof/intel/hda.c | 40 +- sound/soc/sof/intel/hda.h | 50 +- sound/soc/sof/intel/icl.c | 121 +- sound/soc/sof/intel/pci-apl.c | 36 +- sound/soc/sof/intel/pci-cnl.c | 54 +- sound/soc/sof/intel/pci-icl.c | 36 +- sound/soc/sof/intel/pci-tgl.c | 97 +- sound/soc/sof/intel/pci-tng.c | 27 +- sound/soc/sof/intel/shim.h | 16 +- sound/soc/sof/intel/tgl.c | 160 +- sound/soc/sof/ipc.c | 969 +----------- sound/soc/sof/ipc3-control.c | 155 +- sound/soc/sof/ipc3-dtrace.c | 649 ++++++++ sound/soc/sof/ipc3-loader.c | 415 +++++ sound/soc/sof/ipc3-ops.h | 21 - sound/soc/sof/ipc3-pcm.c | 10 +- sound/soc/sof/ipc3-priv.h | 65 + sound/soc/sof/ipc3-topology.c | 79 +- sound/soc/sof/ipc3.c | 1040 ++++++++++++- sound/soc/sof/ipc4-loader.c | 210 +++ sound/soc/sof/ipc4-priv.h | 44 + sound/soc/sof/ipc4.c | 606 ++++++++ sound/soc/sof/loader.c | 698 +-------- sound/soc/sof/mediatek/Kconfig | 11 + sound/soc/sof/mediatek/Makefile | 2 + sound/soc/sof/mediatek/adsp_helper.h | 20 +- sound/soc/sof/mediatek/mt8186/Makefile | 4 + sound/soc/sof/mediatek/mt8186/mt8186-clk.c | 101 ++ sound/soc/sof/mediatek/mt8186/mt8186-clk.h | 24 + sound/soc/sof/mediatek/mt8186/mt8186-loader.c | 58 + sound/soc/sof/mediatek/mt8186/mt8186.c | 554 +++++++ sound/soc/sof/mediatek/mt8186/mt8186.h | 80 + sound/soc/sof/mediatek/mt8195/mt8195.c | 192 ++- sound/soc/sof/mediatek/mtk-adsp-common.c | 84 + sound/soc/sof/mediatek/mtk-adsp-common.h | 10 + sound/soc/sof/ops.c | 2 +- sound/soc/sof/ops.h | 34 +- sound/soc/sof/pcm.c | 30 +- sound/soc/sof/pm.c | 22 +- sound/soc/sof/sof-acpi-dev.c | 6 +- sound/soc/sof/sof-audio.c | 398 +++-- sound/soc/sof/sof-audio.h | 80 +- sound/soc/sof/sof-client-ipc-flood-test.c | 3 +- sound/soc/sof/sof-client-ipc-msg-injector.c | 191 ++- sound/soc/sof/sof-client-probes.c | 14 +- sound/soc/sof/sof-client.c | 66 +- sound/soc/sof/sof-client.h | 2 + sound/soc/sof/sof-of-dev.c | 12 +- sound/soc/sof/sof-of-dev.h | 1 + sound/soc/sof/sof-pci-dev.c | 96 +- sound/soc/sof/sof-priv.h | 130 +- sound/soc/sof/topology.c | 79 +- sound/soc/sof/trace.c | 596 +------ sound/soc/tegra/Kconfig | 12 + sound/soc/tegra/Makefile | 2 + sound/soc/tegra/tegra186_asrc.c | 1046 +++++++++++++ sound/soc/tegra/tegra186_asrc.h | 112 ++ sound/soc/tegra/tegra210_ahub.c | 82 +- sound/soc/tegra/tegra_asoc_machine.c | 30 +- sound/soc/tegra/tegra_wm8903.c | 10 +- sound/soc/ti/ams-delta.c | 4 +- sound/soc/ti/davinci-mcasp.c | 2 + sound/soc/ti/j721e-evm.c | 44 +- sound/soc/ti/omap-abe-twl6040.c | 8 +- sound/soc/ti/omap-twl4030.c | 8 +- sound/soc/ti/osk5912.c | 4 +- sound/soc/ti/rx51.c | 2 +- sound/soc/uniphier/aio-compress.c | 7 +- sound/soc/ux500/mop500_ab8500.c | 9 +- 521 files changed, 23027 insertions(+), 7603 deletions(-) create mode 100644 Documentation/devicetree/bindings/dsp/mediatek,mt8195-dsp.yaml create mode 100644 Documentation/devicetree/bindings/sound/adi,max98396.yaml create mode 100644 Documentation/devicetree/bindings/sound/cirrus,cs35l45.yaml delete mode 100644 Documentation/devicetree/bindings/sound/mt8195-mt6359-rt1011-rt5682.yaml rename Documentation/devicetree/bindings/sound/{mt8195-mt6359-rt1019-rt5682.yaml => mt8195-mt6359.yaml} (84%) create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra186-asrc.yaml rename Documentation/devicetree/bindings/sound/{tas2764.yaml => tas27xx.yaml} (73%) create mode 100644 Documentation/devicetree/bindings/sound/wlf,wm8731.yaml create mode 100644 Documentation/devicetree/bindings/sound/wlf,wm8940.yaml delete mode 100644 Documentation/devicetree/bindings/sound/wm8731.txt create mode 100644 drivers/firmware/mtk-adsp-ipc.c create mode 100644 include/dt-bindings/sound/cs35l45.h rename include/linux/{platform_data/dma-imx.h => dma/imx-dma.h} (67%) create mode 100644 include/linux/firmware/mediatek/mtk-adsp-ipc.h create mode 100644 include/sound/sof/ext_manifest4.h create mode 100644 include/sound/sof/ipc4/header.h create mode 100644 include/uapi/sound/intel/avs/tokens.h create mode 100644 sound/soc/codecs/cs35l45-i2c.c create mode 100644 sound/soc/codecs/cs35l45-spi.c create mode 100644 sound/soc/codecs/cs35l45-tables.c create mode 100644 sound/soc/codecs/cs35l45.c create mode 100644 sound/soc/codecs/cs35l45.h create mode 100644 sound/soc/codecs/max98396.c create mode 100644 sound/soc/codecs/max98396.h create mode 100644 sound/soc/codecs/wm8731-i2c.c create mode 100644 sound/soc/codecs/wm8731-spi.c create mode 100644 sound/soc/intel/avs/apl.c create mode 100644 sound/soc/intel/avs/board_selection.c create mode 100644 sound/soc/intel/avs/path.c create mode 100644 sound/soc/intel/avs/path.h create mode 100644 sound/soc/intel/avs/pcm.c create mode 100644 sound/soc/intel/avs/skl.c create mode 100644 sound/soc/intel/avs/topology.c create mode 100644 sound/soc/intel/avs/topology.h create mode 100644 sound/soc/intel/avs/trace.c create mode 100644 sound/soc/intel/avs/trace.h delete mode 100644 sound/soc/mediatek/mt8195/mt8195-mt6359-rt1011-rt5682.c rename sound/soc/mediatek/mt8195/{mt8195-mt6359-rt1019-rt5682.c => mt8195-mt6359.c} (78%) create mode 100644 sound/soc/soc-utils-test.c create mode 100644 sound/soc/sof/intel/hda-common-ops.c create mode 100644 sound/soc/sof/ipc3-dtrace.c create mode 100644 sound/soc/sof/ipc3-loader.c delete mode 100644 sound/soc/sof/ipc3-ops.h create mode 100644 sound/soc/sof/ipc3-priv.h create mode 100644 sound/soc/sof/ipc4-loader.c create mode 100644 sound/soc/sof/ipc4-priv.h create mode 100644 sound/soc/sof/ipc4.c create mode 100644 sound/soc/sof/mediatek/mt8186/Makefile create mode 100644 sound/soc/sof/mediatek/mt8186/mt8186-clk.c create mode 100644 sound/soc/sof/mediatek/mt8186/mt8186-clk.h create mode 100644 sound/soc/sof/mediatek/mt8186/mt8186-loader.c create mode 100644 sound/soc/sof/mediatek/mt8186/mt8186.c create mode 100644 sound/soc/sof/mediatek/mt8186/mt8186.h create mode 100644 sound/soc/sof/mediatek/mtk-adsp-common.c create mode 100644 sound/soc/sof/mediatek/mtk-adsp-common.h create mode 100644 sound/soc/tegra/tegra186_asrc.c create mode 100644 sound/soc/tegra/tegra186_asrc.h

2 1

[PATCH] ASoC: SOF: amd: Fixed Build error
by Syed Saba kareem 23 May '22

23 May '22

Add linux/module.h in acp-pci.c to solve the below dependency All error/warnings (new ones prefixed by >>): >> sound/soc/amd/acp/acp-pci.c:148:1: warning: data definition has no type or storage class 148 | MODULE_DEVICE_TABLE(pci, acp_pci_ids); | ^~~~~~~~~~~~~~~~~~~ >> sound/soc/amd/acp/acp-pci.c:148:1: error: type defaults to 'int' in declaration of 'MODULE_DEVICE_TABLE' [-Werror=implicit-int] ... Reported-by: kernel test robot <lkp(a)intel.com> Signed-off-by: Syed Saba Kareem<ssbakar(a)amd.com> --- sound/soc/amd/acp/acp-pci.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/amd/acp/acp-pci.c b/sound/soc/amd/acp/acp-pci.c index 340e39d7f420..c893963ee2d0 100644 --- a/sound/soc/amd/acp/acp-pci.c +++ b/sound/soc/amd/acp/acp-pci.c @@ -16,6 +16,7 @@ #include <linux/pci.h> #include <linux/platform_device.h> #include <linux/pm_runtime.h> +#include <linux/module.h> #include "amd.h" #include "../mach-config.h" -- 2.25.1

1 0

[PATCH] ALSA: ctxfi: fix typo in comment
by Julia Lawall 23 May '22

23 May '22

Spelling mistake (triple letters) in comment. Detected with the help of Coccinelle. Signed-off-by: Julia Lawall <Julia.Lawall(a)inria.fr> --- sound/pci/ctxfi/cthw20k1.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/ctxfi/cthw20k1.c b/sound/pci/ctxfi/cthw20k1.c index 0cea4982ed7d..9edbf5d8c3c7 100644 --- a/sound/pci/ctxfi/cthw20k1.c +++ b/sound/pci/ctxfi/cthw20k1.c @@ -1916,7 +1916,7 @@ static int hw_card_start(struct hw *hw) } - /* Switch to X-Fi mode from UAA mode if neeeded */ + /* Switch to X-Fi mode from UAA mode if needed */ if (hw->model == CTUAA) { err = uaa_to_xfi(pci); if (err)

2 1

[PATCH] ALSA: cs5535audio: fix typo in comment
by Julia Lawall 23 May '22

23 May '22

Spelling mistake (triple letters) in comment. Detected with the help of Coccinelle. Signed-off-by: Julia Lawall <Julia.Lawall(a)inria.fr> --- sound/pci/cs5535audio/cs5535audio_pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/cs5535audio/cs5535audio_pcm.c b/sound/pci/cs5535audio/cs5535audio_pcm.c index 5ff10fec7b90..0db24cc4d916 100644 --- a/sound/pci/cs5535audio/cs5535audio_pcm.c +++ b/sound/pci/cs5535audio/cs5535audio_pcm.c @@ -129,7 +129,7 @@ static int cs5535audio_build_dma_packets(struct cs5535audio *cs5535au, return 0; /* the u32 cast is okay because in snd*create we successfully told - pci alloc that we're only 32 bit capable so the uppper will be 0 */ + pci alloc that we're only 32 bit capable so the upper will be 0 */ addr = (u32) substream->runtime->dma_addr; desc_addr = (u32) dma->desc_buf.addr; for (i = 0; i < periods; i++) {

2 1

[PATCH] ALSA: ctxfi: Add SB046x PCI ID
by Edward Matijevic 23 May '22

23 May '22

Adds the PCI ID for X-Fi cards sold under the Platnum and XtremeMusic names Before: snd_ctxfi 0000:05:05.0: chip 20K1 model Unknown (1102:0021) is found After: snd_ctxfi 0000:05:05.0: chip 20K1 model SB046x (1102:0021) is found Signed-off-by: Edward Matijevic <motolav(a)gmail.com> --- sound/pci/ctxfi/ctatc.c | 2 ++ sound/pci/ctxfi/cthardware.h | 3 ++- 2 files changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/pci/ctxfi/ctatc.c b/sound/pci/ctxfi/ctatc.c index 78f35e88aed6..fbdb8a3d5b8e 100644 --- a/sound/pci/ctxfi/ctatc.c +++ b/sound/pci/ctxfi/ctatc.c @@ -36,6 +36,7 @@ | ((IEC958_AES3_CON_FS_48000) << 24)) static const struct snd_pci_quirk subsys_20k1_list[] = { + SND_PCI_QUIRK(PCI_VENDOR_ID_CREATIVE, 0x0021, "SB046x", CTSB046X), SND_PCI_QUIRK(PCI_VENDOR_ID_CREATIVE, 0x0022, "SB055x", CTSB055X), SND_PCI_QUIRK(PCI_VENDOR_ID_CREATIVE, 0x002f, "SB055x", CTSB055X), SND_PCI_QUIRK(PCI_VENDOR_ID_CREATIVE, 0x0029, "SB073x", CTSB073X), @@ -64,6 +65,7 @@ static const struct snd_pci_quirk subsys_20k2_list[] = { static const char *ct_subsys_name[NUM_CTCARDS] = { /* 20k1 models */ + [CTSB046X] = "SB046x", [CTSB055X] = "SB055x", [CTSB073X] = "SB073x", [CTUAA] = "UAA", diff --git a/sound/pci/ctxfi/cthardware.h b/sound/pci/ctxfi/cthardware.h index f406b626a28c..2875cec83b8f 100644 --- a/sound/pci/ctxfi/cthardware.h +++ b/sound/pci/ctxfi/cthardware.h @@ -26,8 +26,9 @@ enum CHIPTYP { enum CTCARDS { /* 20k1 models */ + CTSB046X, + CT20K1_MODEL_FIRST = CTSB046X, CTSB055X, - CT20K1_MODEL_FIRST = CTSB055X, CTSB073X, CTUAA, CT20K1_UNKNOWN, -- 2.36.1

2 3

Is it possible to set the CM6533 so it has 96 kHz sample rate on capture and 48 kHz for playback using ALSA?
by Rob Frohne 22 May '22

22 May '22

Hi All, I found this thread <https://www.alsa-project.org/pipermail/alsa-devel/2017-November/127289.html> on the list that explains why I cannot get 96 kHz sampling with the C-media CM6533. Reading the datasheet <cid:part1.SZL6HGoZ.u1f9OjPs@wallawalla.edu> it is tempting to think the mode I need for my class QSD SDR receiver project, (96 kHz sample rate on capture and 48 kHz for playback could be selected perhaps by ALSA in the HID interface). Is this possible to do with ALSA somehow? The sound cards I purchased for my class on Amazon from Syba, used to use Via VT1620a codecs, but now have been redesigned to use the C-media CM6533, and these have the limitation that will not allow 96 kHz sampling on both capture and playback. I need to change the present setting (96 kHz playback, 48 kHz capture) to the opposite. Page 7 of the datasheet has a note about this. Thanks, Rob -- Rob Frohne, PhD, PE Professor EF Cross School of Engineering Walla Walla University 100 SW 4th Street College Place, WA 99324 (509) 527-2075

1 0

Is it possible to set the CM6533 so it has 96 kHz sample rate on capture and 48 kHz for playback using ALSA?
by Rob Frohne 22 May '22

22 May '22

Hi All, I found this thread <https://www.alsa-project.org/pipermail/alsa-devel/2017-November/127289.html> on the list that explains why I cannot get 96 kHz sampling with the C-media CM6533. Reading the datasheet <cid:part1.yl7hLKYg.W0fZhG6J@wallawalla.edu> it is tempting to think the mode I need for my class QSD SDR receiver project, (96 kHz sample rate on capture and 48 kHz for playback could be selected perhaps by ALSA in the HID interface). Is this possible to do with ALSA somehow? The sound cards I purchased for my class on Amazon from Syba, used to use Via VT1620a codecs, but now have been redesigned to use the C-media CM6533, and these have the limitation that will not allow 96 kHz sampling on both capture and playback. I need to change the present setting (96 kHz playback, 48 kHz capture) to the opposite. Page 7 of the datasheet has a note about this. Thanks, Rob -- Rob Frohne, PhD, PE Professor EF Cross School of Engineering Walla Walla University 100 SW 4th Street College Place, WA 99324 (509) 527-2075

1 0

[PATCH] ALSA: usb-audio: Add missing ep_idx in fixed EP quirks
by Takashi Iwai 21 May '22

21 May '22

The quirk entry for Focusrite Saffire 6 had no proper ep_idx for the capture endpoint, and this confused the driver, resulting in the broken sound. This patch adds the missing ep_idx in the entry. While we are at it, a couple of other entries (for Digidesign MBox and MOTU MicroBook II) seem to have the same problem, and those are covered as well. Fixes: bf6313a0ff76 ("ALSA: usb-audio: Refactor endpoint management") Reported-by: André Kapelrud <a.kapelrud(a)gmail.com> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/quirks-table.h | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 40a5e3eb4ef2..78eb41b621d6 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -2672,6 +2672,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), .altset_idx = 1, .attributes = 0, .endpoint = 0x82, + .ep_idx = 1, .ep_attr = USB_ENDPOINT_XFER_ISOC, .datainterval = 1, .maxpacksize = 0x0126, @@ -2875,6 +2876,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), .altset_idx = 1, .attributes = 0x4, .endpoint = 0x81, + .ep_idx = 1, .ep_attr = USB_ENDPOINT_XFER_ISOC | USB_ENDPOINT_SYNC_ASYNC, .maxpacksize = 0x130, @@ -3391,6 +3393,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), .altset_idx = 1, .attributes = 0, .endpoint = 0x03, + .ep_idx = 1, .rates = SNDRV_PCM_RATE_96000, .ep_attr = USB_ENDPOINT_XFER_ISOC | USB_ENDPOINT_SYNC_ASYNC, -- 2.35.3

1 0

[PATCH] ALSA: lola: Bounds check loop iterator against streams array size
by Kees Cook 21 May '22

21 May '22

GCC 12 sees that it's technically possible for num_streams to be larger than ARRAY_SIZE(pcm->streams). Bounds-check the iterator. ../sound/pci/lola/lola_pcm.c: In function 'lola_pcm_update': ../sound/pci/lola/lola_pcm.c:567:64: warning: array subscript [0, 31] is outside array bounds of 'struct lola_stream[16]' [-Warray-bounds] 567 | struct lola_stream *str = &pcm->streams[i]; | ~~~~~~~~~~~~^~~ In file included from ../sound/pci/lola/lola_pcm.c:15: ../sound/pci/lola/lola.h:307:28: note: while referencing 'streams' 307 | struct lola_stream streams[MAX_STREAM_COUNT]; | ^~~~~~~ Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: alsa-devel(a)alsa-project.org Signed-off-by: Kees Cook <keescook(a)chromium.org> --- sound/pci/lola/lola_pcm.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/pci/lola/lola_pcm.c b/sound/pci/lola/lola_pcm.c index 738ec987000a..32193fae978d 100644 --- a/sound/pci/lola/lola_pcm.c +++ b/sound/pci/lola/lola_pcm.c @@ -561,8 +561,9 @@ static snd_pcm_uframes_t lola_pcm_pointer(struct snd_pcm_substream *substream) void lola_pcm_update(struct lola *chip, struct lola_pcm *pcm, unsigned int bits) { int i; + u8 num_streams = min_t(u8, pcm->num_streams, ARRAY_SIZE(pcm->streams)); - for (i = 0; bits && i < pcm->num_streams; i++) { + for (i = 0; bits && i < num_streams; i++) { if (bits & (1 << i)) { struct lola_stream *str = &pcm->streams[i]; if (str->substream && str->running) -- 2.32.0

2 1

[GIT PULL] ASoC fixes for v5.18-rc7
by Mark Brown 21 May '22

21 May '22

The following changes since commit 5e02fb590e83684f63217f93a9cdeabd6a925f9c: ASoC: cs35l41: Fix DSP mbox start command and global enable order (2022-03-07 13:12:38 +0000) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v5.18-rc7 for you to fetch changes up to 353bb6a5f2ac495f289b7c7a528c7d134c9a8ec4: ASoC: wm_adsp: Compressed stream DSP memory structs should be __packed (2022-03-09 17:50:26 +0000) ---------------------------------------------------------------- ASoC: Fix for v5.17 This is rather late and at this point I'm expecting it to get merged in the merge window rather than as a fix but if we get a -rc8 it's a small, driver specific fix which should be fine to send. ---------------------------------------------------------------- Simon Trimmer (1): ASoC: wm_adsp: Compressed stream DSP memory structs should be __packed sound/soc/codecs/wm_adsp.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-)

2 1

[PATCH 0/6] OPE support on Tegra210 and later
by Sameer Pujar 21 May '22

21 May '22

This series adds support for Output Prcoessing Module (OPE) module on Tegra210 and later generations of SoCs. OPE is a client of AHUB and it has sub blocks of PEQ (Parametric Equalizer) and MBDRC (Multi Band Dynamic Range Compressor) for data processing. An ASoC component is registered for OPE, which includes PEQ and MBDRC functions as well. This can be plugged in audio path using ALSA mixer controls. The series adds necessary binding documentaion, driver and DT binding patches to enable OPE module on Jetson platforms. Sameer Pujar (6): ASoC: tegra: Add binding doc for OPE module ASoC: tegra: Add Tegra210 based OPE driver ASoC: tegra: AHUB routes for OPE module arm64: defconfig: Build Tegra OPE module arm64: tegra: Add OPE device on Tegra210 and later arm64: tegra: Enable OPE on various platforms .../bindings/sound/nvidia,tegra210-ahub.yaml | 4 + .../bindings/sound/nvidia,tegra210-mbdrc.yaml | 51 + .../bindings/sound/nvidia,tegra210-ope.yaml | 91 ++ .../bindings/sound/nvidia,tegra210-peq.yaml | 52 + arch/arm64/boot/dts/nvidia/tegra186-p2771-0000.dts | 43 + arch/arm64/boot/dts/nvidia/tegra186.dtsi | 23 + arch/arm64/boot/dts/nvidia/tegra194-p2972-0000.dts | 43 + .../arm64/boot/dts/nvidia/tegra194-p3509-0000.dtsi | 43 + arch/arm64/boot/dts/nvidia/tegra194.dtsi | 23 + arch/arm64/boot/dts/nvidia/tegra210-p2371-2180.dts | 84 ++ arch/arm64/boot/dts/nvidia/tegra210-p3450-0000.dts | 84 ++ arch/arm64/boot/dts/nvidia/tegra210.dtsi | 40 + .../dts/nvidia/tegra234-p3737-0000+p3701-0000.dts | 43 + arch/arm64/boot/dts/nvidia/tegra234.dtsi | 23 + arch/arm64/configs/defconfig | 1 + sound/soc/tegra/Kconfig | 9 + sound/soc/tegra/Makefile | 2 + sound/soc/tegra/tegra210_ahub.c | 39 +- sound/soc/tegra/tegra210_mbdrc.c | 1012 ++++++++++++++++++++ sound/soc/tegra/tegra210_mbdrc.h | 215 +++++ sound/soc/tegra/tegra210_ope.c | 419 ++++++++ sound/soc/tegra/tegra210_ope.h | 90 ++ sound/soc/tegra/tegra210_peq.c | 434 +++++++++ sound/soc/tegra/tegra210_peq.h | 56 ++ 24 files changed, 2920 insertions(+), 4 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-mbdrc.yaml create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-ope.yaml create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-peq.yaml create mode 100644 sound/soc/tegra/tegra210_mbdrc.c create mode 100644 sound/soc/tegra/tegra210_mbdrc.h create mode 100644 sound/soc/tegra/tegra210_ope.c create mode 100644 sound/soc/tegra/tegra210_ope.h create mode 100644 sound/soc/tegra/tegra210_peq.c create mode 100644 sound/soc/tegra/tegra210_peq.h -- 2.7.4

3 11

[PATCH] ALSA: usb-audio: Workaround for clock setup on TEAC devices
by Takashi Iwai 21 May '22

21 May '22

Maris reported that TEAC UD-501 (0644:8043) doesn't work with the typical "clock source 41 is not valid, cannot use" errors on the recent kernels. The currently known workaround so far is to restore (partially) what we've done unconditionally at the clock setup; namely, re-setup the USB interface immediately after the clock is changed. This patch re-introduces the behavior conditionally for TEAC devices. Further notes: - The USB interface shall be set later in snd_usb_endpoint_configure(), but this seems to be too late. - Even calling usb_set_interface() right after sne_usb_init_sample_rate() doesn't help; so this must be related with the clock validation, too. - The device may still spew the "clock source 41 is not valid" error at the first clock setup. This seems happening at the very first try of clock setup, but it disappears at later attempts. The error is likely harmless because the driver retries the clock setup (such an error is more or less expected on some devices). Fixes: bf6313a0ff76 ("ALSA: usb-audio: Refactor endpoint management") Reported-and-tested-by: Maris Abele <maris7abele(a)gmail.com> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/clock.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/usb/clock.c b/sound/usb/clock.c index 4dfe76416794..3c435d379306 100644 --- a/sound/usb/clock.c +++ b/sound/usb/clock.c @@ -572,6 +572,13 @@ static int set_sample_rate_v2v3(struct snd_usb_audio *chip, /* continue processing */ } + /* FIXME - TEAC devices require the immediate interface setup */ + if (rate != prev_rate && USB_ID_VENDOR(chip->usb_id) == 0x0644) { + usb_set_interface(chip->dev, fmt->iface, fmt->altsetting); + if (chip->quirk_flags & QUIRK_FLAG_IFACE_DELAY) + msleep(50); + } + validation: /* validate clock after rate change */ if (!uac_clock_source_is_valid(chip, fmt, clock)) -- 2.35.3

1 0

Complete support for "Pro" USB multichannel cards
by GitHub issues - edited 20 May '22

20 May '22

alsa-project/alsa-ucm-conf issue #154 was edited from perexg: - [x] channel split using alsa-lib - [x] channel split inside app - [x] channel split hide the specific configs (alsa-lib, app side support) - [x] mixer control remap using alsa-lib - [x] simplify the mixer control remap Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/154 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Complete support for "Pro" USB multichannel cards
by GitHub issues - edited 20 May '22

20 May '22

alsa-project/alsa-ucm-conf issue #154 was edited from perexg: - [x] channel split using alsa-lib - [x] channel split inside app - [x] channel split hide the specific configs (alsa-lib, app side support) - [x] mixer control remap using alsa-lib - [x] simplify the mixer control remap Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/154 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Complete support for "Pro" USB multichannel cards
by GitHub issues - edited 20 May '22

20 May '22

alsa-project/alsa-ucm-conf issue #154 was edited from perexg: - [x] channel split using alsa-lib - [x] channel split inside app - [ ] channel split hide the specific configs (alsa-lib, app side support) - [x] mixer control remap using alsa-lib - [ ] simplify the mixer control remap Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/154 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH] ASoC: rt1308-sdw: add the default value of register 0xc320
by shumingf＠realtek.com 20 May '22

20 May '22

From: Shuming Fan <shumingf(a)realtek.com> The driver missed the default value of register 0xc320. This patch adds that default value to avoid the error messages when the driver went to suspend mode already. BugLink: https://github.com/thesofproject/linux/issues/3651 Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/rt1308-sdw.h | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/rt1308-sdw.h b/sound/soc/codecs/rt1308-sdw.h index c5ce75666dcc..6668e19d85d4 100644 --- a/sound/soc/codecs/rt1308-sdw.h +++ b/sound/soc/codecs/rt1308-sdw.h @@ -140,6 +140,7 @@ static const struct reg_default rt1308_reg_defaults[] = { { 0x3008, 0x02 }, { 0x300a, 0x00 }, { 0xc003 | (RT1308_DAC_SET << 4), 0x00 }, + { 0xc000 | (RT1308_POWER << 4), 0x00 }, { 0xc001 | (RT1308_POWER << 4), 0x00 }, { 0xc002 | (RT1308_POWER << 4), 0x00 }, }; -- 2.35.3

3 2

[PATCH] ASoC: max98090: Remove unneeded check in max98090_put_enab_tlv()
by Alexey Khoroshilov 20 May '22

20 May '22

Variable sel is of unsigned int type, so sel < 0 is not required. Found by Linux Verification Center (linuxtesting.org) with SVACE. Signed-off-by: Alexey Khoroshilov <khoroshilov(a)ispras.ru> Fixes: 2fbe467bcbfc ("ASoC: max98090: Reject invalid values in custom control put()") --- sound/soc/codecs/max98090.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index 62b41ca050a2..c535a8496bf1 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -413,7 +413,7 @@ static int max98090_put_enab_tlv(struct snd_kcontrol *kcontrol, val = (val >> mc->shift) & mask; - if (sel < 0 || sel > mc->max) + if (sel > mc->max) return -EINVAL; *select = sel; -- 2.7.4

3 10

[PATCH 0/8] ASoC: SOF: Introduce generic (in)firmware tracing infrastructure
by Peter Ujfalusi 20 May '22

20 May '22

Hi, SOF is using dma-trace (or dtrace) as a firmware tracing method, which is only supported with IPC3 and it is not applicable for IPC4. Currently the dtrace is 'open managed' regardless of IPC version (we do force disable it for IPC4, but the dtrace calls remain in place). >From the kernel point of view there are only few ops that needs to be exposed by the firmware tracing support and everything else is IPC private, should not be known by the core. This series converts the current dma-trace as ipc3 specific firmware tracing sub-component and moves all private data out from generic code. Regards, Peter --- Peter Ujfalusi (8): ASoC: SOF: Introduce IPC independent ops for firmware tracing support ASoC: SOF: Rename dtrace_is_supported flag to fw_trace_is_supported ASoC: SOF: Clone the trace code to ipc3-dtrace as fw_tracing implementation ASoC: SOF: Switch to IPC generic firmware tracing ASoC: SOF: ipc3-dtrace: Move host ops wrappers from generic header to private ASoC: SOF: Modify the host trace_init parameter list to include dmab ASoC: SOF: Introduce opaque storage of private data for firmware tracing ASoC: SOF: ipc3-dtrace: Move dtrace related variables local from sof_dev sound/soc/sof/Makefile | 1 + sound/soc/sof/amd/acp-trace.c | 4 +- sound/soc/sof/amd/acp.h | 2 +- sound/soc/sof/core.c | 13 +- sound/soc/sof/debug.c | 2 +- sound/soc/sof/intel/hda-dsp.c | 2 +- sound/soc/sof/intel/hda-trace.c | 4 +- sound/soc/sof/intel/hda.h | 2 +- sound/soc/sof/ipc.c | 6 + sound/soc/sof/ipc3-dtrace.c | 649 ++++++++++++++++++++++++++++++++ sound/soc/sof/ipc3-priv.h | 38 ++ sound/soc/sof/ipc3.c | 3 +- sound/soc/sof/ops.c | 2 +- sound/soc/sof/ops.h | 26 -- sound/soc/sof/pm.c | 8 +- sound/soc/sof/sof-priv.h | 53 +-- sound/soc/sof/trace.c | 621 ++---------------------------- 17 files changed, 767 insertions(+), 669 deletions(-) create mode 100644 sound/soc/sof/ipc3-dtrace.c -- 2.36.1

2 9

[PATCH 0/2] ASoC: rt9120: Fix 3byte read logic and optimize 'PWDN' control
by cy_huang 20 May '22

20 May '22

From: ChiYuan Huang <cy_huang(a)richtek.com> Hi, This patch series try to fix two things. - 3byte read logic bug - Use regcache and pm_runtime to optimize 'PWDN' control ChiYuan Huang (2): ASoC: rt9120: Fix 3byte read, valule offset typo ASoC: rt9120: Use pm_runtime and regcache to optimize 'pwdnn' logic sound/soc/codecs/rt9120.c | 113 +++++++++++++++++++++++++++++++++++++++++++--- 1 file changed, 106 insertions(+), 7 deletions(-) -- 2.7.4

2 3

[PATCH] ASoC: SOF: mediatek: remove duplicate include in mt8195.c
by Wan Jiabing 20 May '22

20 May '22

Fix following checkincludes.pl warning: sound/soc/sof/mediatek/mt8195/mt8195.c: linux/of_platform.h is included more than once. Signed-off-by: Wan Jiabing <wanjiabing(a)vivo.com> --- sound/soc/sof/mediatek/mt8195/mt8195.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/sof/mediatek/mt8195/mt8195.c b/sound/soc/sof/mediatek/mt8195/mt8195.c index f4b24afb6f75..7278132300e0 100644 --- a/sound/soc/sof/mediatek/mt8195/mt8195.c +++ b/sound/soc/sof/mediatek/mt8195/mt8195.c @@ -12,7 +12,6 @@ #include <linux/delay.h> #include <linux/firmware.h> #include <linux/io.h> -#include <linux/of_platform.h> #include <linux/of_address.h> #include <linux/of_irq.h> #include <linux/of_platform.h> -- 2.36.1

2 1

[PATCH -next] ASoC: wm2000: fix missing clk_disable_unprepare() on error in wm2000_anc_transition()
by Yang Yingliang 20 May '22

20 May '22

Fix the missing clk_disable_unprepare() before return from wm2000_anc_transition() in the error handling case. Fixes: 514cfd6dd725 ("ASoC: wm2000: Integrate with clock API") Signed-off-by: Yang Yingliang <yangyingliang(a)huawei.com> --- sound/soc/codecs/wm2000.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm2000.c b/sound/soc/codecs/wm2000.c index 72e165cc6443..97ece3114b3d 100644 --- a/sound/soc/codecs/wm2000.c +++ b/sound/soc/codecs/wm2000.c @@ -536,7 +536,7 @@ static int wm2000_anc_transition(struct wm2000_priv *wm2000, { struct i2c_client *i2c = wm2000->i2c; int i, j; - int ret; + int ret = 0; if (wm2000->anc_mode == mode) return 0; @@ -566,13 +566,13 @@ static int wm2000_anc_transition(struct wm2000_priv *wm2000, ret = anc_transitions[i].step[j](i2c, anc_transitions[i].analogue); if (ret != 0) - return ret; + break; } if (anc_transitions[i].dest == ANC_OFF) clk_disable_unprepare(wm2000->mclk); - return 0; + return ret; } static int wm2000_anc_set_mode(struct wm2000_priv *wm2000) -- 2.25.1

3 2

[PATCH 0/2] ASoC: SOF: mediatek: add debug dump
by Pierre-Louis Bossart 20 May '22

20 May '22

YC Hung (2): ASoC: SOF: mediatek: Add mediatek common debug dump ASoC: SOF: mediatek: Add mt8195 debug dump sound/soc/sof/mediatek/Makefile | 1 + sound/soc/sof/mediatek/mt8195/mt8195.c | 29 ++++++++ sound/soc/sof/mediatek/mtk-adsp-common.c | 84 ++++++++++++++++++++++++ sound/soc/sof/mediatek/mtk-adsp-common.h | 10 +++ 4 files changed, 124 insertions(+) create mode 100644 sound/soc/sof/mediatek/mtk-adsp-common.c create mode 100644 sound/soc/sof/mediatek/mtk-adsp-common.h -- 2.30.2

2 3

[PATCH RESEND] ASoC: SOF: sof-client-ipc-flood-test: use pm_runtime_resume_and_get()
by Pierre-Louis Bossart 20 May '22

20 May '22

Use pm_runtime_resume_and_get() to replace the pm_runtime_get_sync() and pm_runtime_put_noidle() pattern. No functional changes. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Péter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> --- This patch was sent as part of the "[PATCH 0/4] ASoC: SOF: use pm_runtime_resume_and_get()" series, but somehow not applied. The other 3 patches were selected. sound/soc/sof/sof-client-ipc-flood-test.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/sof/sof-client-ipc-flood-test.c b/sound/soc/sof/sof-client-ipc-flood-test.c index db3a052c5dd2b..4bdecd80248ac 100644 --- a/sound/soc/sof/sof-client-ipc-flood-test.c +++ b/sound/soc/sof/sof-client-ipc-flood-test.c @@ -217,10 +217,9 @@ static ssize_t sof_ipc_flood_dfs_write(struct file *file, const char __user *buf ipc_count = MAX_IPC_FLOOD_COUNT; } - ret = pm_runtime_get_sync(dev); + ret = pm_runtime_resume_and_get(dev); if (ret < 0 && ret != -EACCES) { dev_err_ratelimited(dev, "debugfs write failed to resume %d\n", ret); - pm_runtime_put_noidle(dev); goto out; } -- 2.30.2

2 1

[PATCH 0/2] ASoC: remove two unnecessary gpiolib dependencies
by Pierre-Louis Bossart 20 May '22

20 May '22

Remove two dependencies - issues reported by Intel kernel test bot. Pierre-Louis Bossart (2): ASoC: max98357a: remove dependency on GPIOLIB ASoC: rt1015p: remove dependency on GPIOLIB sound/soc/codecs/Kconfig | 2 -- 1 file changed, 2 deletions(-) -- 2.30.2

2 3

[PATCH v2 1/2] Revert "ASoC: amd: acp: Set gpio_spkr_en to None for max speaker amplifer in machine driver"
by V sujith kumar Reddy 20 May '22

20 May '22

ASoC: amd : acp : Set Speaker enable/disable pin through rt1019 codec driver. RT1019 codec has two ways of controlling the en_spkr. one way is controlling through gpio pin method the another way is through codec register update through driver. Now Speaker enable/disable is controlled through codec register updated by codec driver. This patch reverts gpio logic. This reverts commit 7fa5c33d043160eba3be9fb8e21588dff2a467c7. Signed-off-by: V sujith kumar Reddy <Vsujithkumar.Reddy(a)amd.com> --- sound/soc/amd/acp/acp-mach.h | 1 + sound/soc/amd/acp/acp-sof-mach.c | 4 ++-- 2 files changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/amd/acp/acp-mach.h b/sound/soc/amd/acp/acp-mach.h index c855f50d6b34..fd6299844ebe 100644 --- a/sound/soc/amd/acp/acp-mach.h +++ b/sound/soc/amd/acp/acp-mach.h @@ -21,6 +21,7 @@ #include <linux/gpio/consumer.h> #define EN_SPKR_GPIO_GB 0x11F +#define EN_SPKR_GPIO_NK 0x146 #define EN_SPKR_GPIO_NONE -EINVAL enum be_id { diff --git a/sound/soc/amd/acp/acp-sof-mach.c b/sound/soc/amd/acp/acp-sof-mach.c index bf61a1726f0e..8243765d490f 100644 --- a/sound/soc/amd/acp/acp-sof-mach.c +++ b/sound/soc/amd/acp/acp-sof-mach.c @@ -37,7 +37,7 @@ static struct acp_card_drvdata sof_rt5682_max_data = { .hs_codec_id = RT5682, .amp_codec_id = MAX98360A, .dmic_codec_id = DMIC, - .gpio_spkr_en = EN_SPKR_GPIO_NONE, + .gpio_spkr_en = EN_SPKR_GPIO_NK, }; static struct acp_card_drvdata sof_rt5682s_rt1019_data = { @@ -56,7 +56,7 @@ static struct acp_card_drvdata sof_rt5682s_max_data = { .hs_codec_id = RT5682S, .amp_codec_id = MAX98360A, .dmic_codec_id = DMIC, - .gpio_spkr_en = EN_SPKR_GPIO_NONE, + .gpio_spkr_en = EN_SPKR_GPIO_NK, }; static const struct snd_kcontrol_new acp_controls[] = { -- 2.25.1

2 2

[PATCH -next] ASoC: codecs: lpass: Fix passing zero to 'PTR_ERR'
by YueHaibing 20 May '22

20 May '22

sound/soc/codecs/lpass-macro-common.c:28 lpass_macro_pds_init() warn: passing zero to 'PTR_ERR' sound/soc/codecs/lpass-macro-common.c:38 lpass_macro_pds_init() warn: passing zero to 'PTR_ERR' sound/soc/codecs/lpass-macro-common.c:54 lpass_macro_pds_init() warn: passing zero to 'ERR_PTR' dev_pm_domain_attach_by_name() may return NULL, set 'ret' as -ENODATA to fix this warning. Fixes: 1a8ee4cf8418 ("ASoC: codecs: Fix error handling in power domain init and exit handlers") Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/soc/codecs/lpass-macro-common.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/lpass-macro-common.c b/sound/soc/codecs/lpass-macro-common.c index 3c661fd61173..1b9082d237c1 100644 --- a/sound/soc/codecs/lpass-macro-common.c +++ b/sound/soc/codecs/lpass-macro-common.c @@ -25,7 +25,7 @@ struct lpass_macro *lpass_macro_pds_init(struct device *dev) l_pds->macro_pd = dev_pm_domain_attach_by_name(dev, "macro"); if (IS_ERR_OR_NULL(l_pds->macro_pd)) { - ret = PTR_ERR(l_pds->macro_pd); + ret = l_pds->macro_pd ? PTR_ERR(l_pds->macro_pd) : -ENODATA; goto macro_err; } @@ -35,7 +35,7 @@ struct lpass_macro *lpass_macro_pds_init(struct device *dev) l_pds->dcodec_pd = dev_pm_domain_attach_by_name(dev, "dcodec"); if (IS_ERR_OR_NULL(l_pds->dcodec_pd)) { - ret = PTR_ERR(l_pds->dcodec_pd); + ret = l_pds->dcodec_pd ? PTR_ERR(l_pds->dcodec_pd) : -ENODATA; goto dcodec_err; } -- 2.17.1

2 1

topology: plugins: nhlt: fix intel dmic mode calculation
by GitHub pull_request - opened 20 May '22

20 May '22

alsa-project/alsa-utils pull request #148 was opened from juimonen: Dmic find_modes function was used with same sampling rate for both fifo a and b parameter calculations. This bug was found in testing two simultaneous dmic dais in topology. Fix this by introducing a sampling rate argument for find_modes. Signed-off-by: Jaska Uimonen <jaska.uimonen(a)linux.intel.com> Request URL : https://github.com/alsa-project/alsa-utils/pull/148 Patch URL : https://github.com/alsa-project/alsa-utils/pull/148.patch Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[PATCH] ASoC: ops: Shift tested values in snd_soc_put_volsw() by +min
by Marek Vasut 19 May '22

19 May '22

While the $val/$val2 values passed in from userspace are always >= 0 integers, the limits of the control can be signed integers and the $min can be non-zero and less than zero. To correctly validate $val/$val2 against platform_max, add the $min offset to val first. Fixes: 817f7c9335ec0 ("ASoC: ops: Reject out of bounds values in snd_soc_put_volsw()") Signed-off-by: Marek Vasut <marex(a)denx.de> Cc: Mark Brown <broonie(a)kernel.org> Cc: stable(a)vger.kernel.org --- sound/soc/soc-ops.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c index f24f7354f46fe..6389a512c4dc6 100644 --- a/sound/soc/soc-ops.c +++ b/sound/soc/soc-ops.c @@ -317,7 +317,7 @@ int snd_soc_put_volsw(struct snd_kcontrol *kcontrol, mask = BIT(sign_bit + 1) - 1; val = ucontrol->value.integer.value[0]; - if (mc->platform_max && val > mc->platform_max) + if (mc->platform_max && ((int)val + min) > mc->platform_max) return -EINVAL; if (val > max - min) return -EINVAL; @@ -330,7 +330,7 @@ int snd_soc_put_volsw(struct snd_kcontrol *kcontrol, val = val << shift; if (snd_soc_volsw_is_stereo(mc)) { val2 = ucontrol->value.integer.value[1]; - if (mc->platform_max && val2 > mc->platform_max) + if (mc->platform_max && ((int)val2 + min) > mc->platform_max) return -EINVAL; if (val2 > max - min) return -EINVAL; -- 2.34.1

6 14

[PATCH v1 0/2] ASOC : amd: acp :Add Machine driver changes for
by V sujith kumar Reddy 19 May '22

19 May '22

Add Machine driver support for nau8825, max98560 and rt5682s,rt1019 combination This patch set depends on: --gpio patch --https://patchwork.kernel.org/project/alsa-devel/patch/20220516160619.17832-1-Vsujithkumar.Reddy@amd.com/ --https://patchwork.kernel.org/project/alsa-devel/patch/20220516160619.17832-2-Vsujithkumar.Reddy@amd.com/ Signed-off-by: V sujith kumar Reddy <Vsujithkumar.Reddy(a)amd.com> V sujith kumar Reddy (2): ASoC: amd: acp: Add support for nau8825 and max98360 card ASoC: amd: acp: Add support for rt5682s and rt1019 card with hs instance sound/soc/amd/acp-config.c | 18 +++ sound/soc/amd/acp/Kconfig | 1 + sound/soc/amd/acp/acp-mach-common.c | 228 +++++++++++++++++++++++++--- sound/soc/amd/acp/acp-mach.h | 3 + sound/soc/amd/acp/acp-sof-mach.c | 30 ++++ 5 files changed, 263 insertions(+), 17 deletions(-) -- 2.25.1

1 2

[v7] FROMLIST: ASoC: Intel: sof_cs42l42: adding support for ADL configuration and BT offload audio
by Terry Chen 18 May '22

18 May '22

To be able to do driver data for adl_mx98360a_cs4242 which supports two max98360a speaker amplifiers on SSP1 and cs42l42 headphone codec on SSP0 running on ADL platform. Also add the capability to machine driver of creating DAI Link for BT offload. Although BT offload always uses SSP2 port but we reserve the flexibility to assign the port number in macro. Signed-off-by: Terry Chen <terry_chen(a)wistron.corp-partner.google.com> --- sound/soc/intel/boards/sof_cs42l42.c | 86 ++++++++++++++++++- .../intel/common/soc-acpi-intel-adl-match.c | 7 ++ 2 files changed, 89 insertions(+), 4 deletions(-) diff --git a/sound/soc/intel/boards/sof_cs42l42.c b/sound/soc/intel/boards/sof_cs42l42.c index ce78c18798876..56bae2185904b 100644 --- a/sound/soc/intel/boards/sof_cs42l42.c +++ b/sound/soc/intel/boards/sof_cs42l42.c @@ -41,8 +41,13 @@ #define SOF_CS42L42_DAILINK_MASK (GENMASK(24, 10)) #define SOF_CS42L42_DAILINK(link1, link2, link3, link4, link5) \ ((((link1) | ((link2) << 3) | ((link3) << 6) | ((link4) << 9) | ((link5) << 12)) << SOF_CS42L42_DAILINK_SHIFT) & SOF_CS42L42_DAILINK_MASK) -#define SOF_MAX98357A_SPEAKER_AMP_PRESENT BIT(25) -#define SOF_MAX98360A_SPEAKER_AMP_PRESENT BIT(26) +#define SOF_BT_OFFLOAD_PRESENT BIT(25) +#define SOF_CS42L42_SSP_BT_SHIFT 26 +#define SOF_CS42L42_SSP_BT_MASK (GENMASK(28, 26)) +#define SOF_CS42L42_SSP_BT(quirk) \ + (((quirk) << SOF_CS42L42_SSP_BT_SHIFT) & SOF_CS42L42_SSP_BT_MASK) +#define SOF_MAX98357A_SPEAKER_AMP_PRESENT BIT(29) +#define SOF_MAX98360A_SPEAKER_AMP_PRESENT BIT(30) enum { LINK_NONE = 0, @@ -50,6 +55,7 @@ enum { LINK_SPK = 2, LINK_DMIC = 3, LINK_HDMI = 4, + LINK_BT = 5, }; /* Default: SSP2 */ @@ -278,6 +284,13 @@ static struct snd_soc_dai_link_component dmic_component[] = { } }; +static struct snd_soc_dai_link_component dummy_component[] = { + { + .name = "snd-soc-dummy", + .dai_name = "snd-soc-dummy-dai", + } +}; + static int create_spk_amp_dai_links(struct device *dev, struct snd_soc_dai_link *links, struct snd_soc_dai_link_component *cpus, @@ -467,9 +480,50 @@ static int create_hdmi_dai_links(struct device *dev, return -ENOMEM; } +static int create_bt_offload_dai_links(struct device *dev, + struct snd_soc_dai_link *links, + struct snd_soc_dai_link_component *cpus, + int *id, int ssp_bt) +{ + /* bt offload */ + if (!(sof_cs42l42_quirk & SOF_BT_OFFLOAD_PRESENT)) + return 0; + + links[*id].name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d-BT", + ssp_bt); + if (!links[*id].name) + goto devm_err; + + links[*id].id = *id; + links[*id].codecs = dummy_component; + links[*id].num_codecs = ARRAY_SIZE(dummy_component); + links[*id].platforms = platform_component; + links[*id].num_platforms = ARRAY_SIZE(platform_component); + + links[*id].dpcm_playback = 1; + links[*id].dpcm_capture = 1; + links[*id].no_pcm = 1; + links[*id].cpus = &cpus[*id]; + links[*id].num_cpus = 1; + + links[*id].cpus->dai_name = devm_kasprintf(dev, GFP_KERNEL, + "SSP%d Pin", + ssp_bt); + if (!links[*id].cpus->dai_name) + goto devm_err; + + (*id)++; + + return 0; + +devm_err: + return -ENOMEM; +} + static struct snd_soc_dai_link *sof_card_dai_links_create(struct device *dev, int ssp_codec, int ssp_amp, + int ssp_bt, int dmic_be_num, int hdmi_num) { @@ -522,6 +576,14 @@ static struct snd_soc_dai_link *sof_card_dai_links_create(struct device *dev, goto devm_err; } break; + case LINK_BT: + ret = create_bt_offload_dai_links(dev, links, cpus, &id, ssp_bt); + if (ret < 0) { + dev_err(dev, "fail to create bt offload dai links, ret %d\n", + ret); + goto devm_err; + } + break; case LINK_NONE: /* caught here if it's not used as terminator in macro */ default: @@ -543,7 +605,7 @@ static int sof_audio_probe(struct platform_device *pdev) struct snd_soc_acpi_mach *mach; struct sof_card_private *ctx; int dmic_be_num, hdmi_num; - int ret, ssp_amp, ssp_codec; + int ret, ssp_bt, ssp_amp, ssp_codec; ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL); if (!ctx) @@ -568,6 +630,9 @@ static int sof_audio_probe(struct platform_device *pdev) dev_dbg(&pdev->dev, "sof_cs42l42_quirk = %lx\n", sof_cs42l42_quirk); + ssp_bt = (sof_cs42l42_quirk & SOF_CS42L42_SSP_BT_MASK) >> + SOF_CS42L42_SSP_BT_SHIFT; + ssp_amp = (sof_cs42l42_quirk & SOF_CS42L42_SSP_AMP_MASK) >> SOF_CS42L42_SSP_AMP_SHIFT; @@ -578,9 +643,11 @@ static int sof_audio_probe(struct platform_device *pdev) if (sof_cs42l42_quirk & SOF_SPEAKER_AMP_PRESENT) sof_audio_card_cs42l42.num_links++; + if (sof_cs42l42_quirk & SOF_BT_OFFLOAD_PRESENT) + sof_audio_card_cs42l42.num_links++; dai_links = sof_card_dai_links_create(&pdev->dev, ssp_codec, ssp_amp, - dmic_be_num, hdmi_num); + ssp_bt, dmic_be_num, hdmi_num); if (!dai_links) return -ENOMEM; @@ -621,6 +688,17 @@ static const struct platform_device_id board_ids[] = { SOF_CS42L42_SSP_AMP(1)) | SOF_CS42L42_DAILINK(LINK_HP, LINK_DMIC, LINK_HDMI, LINK_SPK, LINK_NONE), }, + { + .name = "adl_mx98360a_cs4242", + .driver_data = (kernel_ulong_t)(SOF_CS42L42_SSP_CODEC(0) | + SOF_SPEAKER_AMP_PRESENT | + SOF_MAX98360A_SPEAKER_AMP_PRESENT | + SOF_CS42L42_SSP_AMP(1) | + SOF_CS42L42_NUM_HDMIDEV(4) | + SOF_BT_OFFLOAD_PRESENT | + SOF_CS42L42_SSP_BT(2)) | + SOF_CS42L42_DAILINK(LINK_HP, LINK_DMIC, LINK_HDMI, LINK_SPK, LINK_BT), + }, { } }; MODULE_DEVICE_TABLE(platform, board_ids); 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 7c8cd00457f81..311adeb3afbc4 100644 --- a/sound/soc/intel/common/soc-acpi-intel-adl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-adl-match.c @@ -384,6 +384,13 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_machines[] = { .sof_fw_filename = "sof-adl.ri", .sof_tplg_filename = "sof-adl-cs35l41.tplg", }, + { + .id = "10134242", + .drv_name = "adl_mx98360a_cs4242", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &adl_max98360a_amp, + .sof_tplg_filename = "sof-adl-max98360a-rt5682.tplg", + }, {}, }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_adl_machines); -- 2.25.1

2 1

[PATCH] ALSA: usb-audio: Fix wrong kfree issue in snd_usb_endpoint_free_all
by Wan Jiabing 18 May '22

18 May '22

Fix following coccicheck error: ./sound/usb/endpoint.c:1671:8-10: ERROR: reference preceded by free on line 1671. Here should be 'cp' rather than 'ip'. Fixes: c11117b634f4 ("ALSA: usb-audio: Refcount multiple accesses on the single clock") Signed-off-by: Wan Jiabing <wanjiabing(a)vivo.com> --- sound/usb/endpoint.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index df5a70013a85..f9c921683948 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -1668,7 +1668,7 @@ void snd_usb_endpoint_free_all(struct snd_usb_audio *chip) kfree(ip); list_for_each_entry_safe(cp, cn, &chip->clock_ref_list, list) - kfree(ip); + kfree(cp); } /* -- 2.36.1

2 1

[PATCH v2 00/15] ASoC: Intel: avs: Driver core and PCM operations
by Cezary Rojewski 17 May '22

17 May '22

Part three of main AVS driver series. This series was originally part of the initial series which was later divided [1] into smaller, easier-to-review chunks. Thus, many patches found here were already present on the list. This series consists of code typical to many drivers - PCI driver operations, trace ability, PM operations - as well as PCM handlers for all standard audio interfaces, that is, HDA, I2S and DMIC are found here. Series starts with updating firmware boot flow - libraries are no longer ignored. This change is dependent on already merged topology code [2] and because of that could not be part of the initial series [1]. PCM operations are split into four changes. First component operations alone i.e. operations which are usually agnostic towards path position (FE/BE). Then it continues with "generic" FE operations - there is no interface split here as from Intel ADSP point of view, FE, or HOST side as it's called in the specs, involves HD-Audio operations only. BE (also known as LINK) side on the other hand is divided into "non-HD-Audio" and HD-Audio part. The former represents transfer over DMIC and I2S interfaces both. While patches implementing standard PCI driver operations along (again standard) HD-Audio initialization routines followed up by power management handlers are two major ones, series covers also other important subjects such as: While patches implementing standard PCI driver operations along (again standard) HD-Audio initialization routines followed up by power management handlers are two major ones, series covers also other important subjects such as: - event tracing - preparation for firmware tracing (debugability) - coredump (debugability) - recovery flow (attempt recovery after IPC timeout or exception) - D0ix (D0 device substate, complements standard power management) Series is finalized by actual addition of supported platforms: SKL and APL-based. Platform-specific files are limited to firmware-specific bits, that is, bits that are specific to given firmware generation. Everything else is shared and is part of already upstream messaging code found in ipc.c, messages.c and messages.h files. Changes in v2: - usage of avs_releast_last_firmware() dropped in error path for library loading procedure as suggested by Pierre - 'link_mask' usage replaced with 'i2s_link_mask' as requested by Pierre. Existing code addressed with new patch: "ASoC: Intel: avs: Replace link_mask usage with i2s_link_mask" - fixed possible race during recovery flow (->recovering flag is now atomic and tested in single location only). - dropped platform prefixes for basically all i2s board descriptors - 'ssp_test' renamed to 'i2s_test' to match naming convention of other boards - simplified PM implementation for current series, 'low_power' bits moved to future series - replaced SND_INTEL_DSP_DRIVER_SST with _AVS as suggested by Mark. Required changes to intel-dspcfg will be added in future series - number of typos across commit messages addressed [1]: https://lore.kernel.org/all/20220311153544.136854-1-cezary.rojewski@intel.c… [2]: https://lore.kernel.org/all/20220331135246.993089-1-cezary.rojewski@intel.c… Cezary Rojewski (14): ASoC: Intel: avs: Account for libraries when booting basefw ASoC: Intel: avs: Generic soc component driver ASoC: Intel: avs: Generic PCM FE operations ASoC: Intel: avs: non-HDA PCM BE operations ASoC: Intel: avs: HDA PCM BE operations ASoC: Intel: avs: Coredump and recovery flow ASoC: Intel: avs: Prepare for firmware tracing ASoC: Intel: avs: D0ix power state support ASoC: Intel: avs: Event tracing ASoC: Intel: avs: Machine board registration ASoC: Intel: avs: PCI driver implementation ASoC: Intel: avs: Power management ASoC: Intel: avs: SKL-based platforms support ASoC: Intel: avs: APL-based platforms support Piotr Maziarz (1): ASoC: Intel: avs: Replace link_mask usage with i2s_link_mask include/sound/intel-dsp-config.h | 3 +- include/sound/soc-acpi.h | 2 + sound/soc/intel/Kconfig | 2 + sound/soc/intel/avs/Makefile | 7 +- sound/soc/intel/avs/apl.c | 250 ++++++ sound/soc/intel/avs/avs.h | 79 ++ sound/soc/intel/avs/board_selection.c | 501 +++++++++++ sound/soc/intel/avs/core.c | 631 +++++++++++++ sound/soc/intel/avs/dsp.c | 27 +- sound/soc/intel/avs/ipc.c | 253 +++++- sound/soc/intel/avs/loader.c | 84 ++ sound/soc/intel/avs/messages.c | 35 +- sound/soc/intel/avs/messages.h | 51 ++ sound/soc/intel/avs/pcm.c | 1182 +++++++++++++++++++++++++ sound/soc/intel/avs/registers.h | 8 + sound/soc/intel/avs/skl.c | 125 +++ sound/soc/intel/avs/topology.c | 14 +- sound/soc/intel/avs/trace.c | 33 + sound/soc/intel/avs/trace.h | 158 ++++ sound/soc/intel/avs/utils.c | 23 + 20 files changed, 3447 insertions(+), 21 deletions(-) create mode 100644 sound/soc/intel/avs/apl.c create mode 100644 sound/soc/intel/avs/board_selection.c create mode 100644 sound/soc/intel/avs/pcm.c create mode 100644 sound/soc/intel/avs/skl.c create mode 100644 sound/soc/intel/avs/trace.c create mode 100644 sound/soc/intel/avs/trace.h -- 2.25.1

5 21

Re: Re: [PATCH] conf: vc4-hdmi: use a proper hdmi pcm, fix broken default pcm
by popcorn mix 17 May '22

17 May '22

Matthias Reichl wrote: > Userspace expects to see a HDMI pcm, so remove the somewhat broken > front and iec958 pcms and add a proper hdmi pcm instead. > > The hdmi pcm supports HD/HBR audio passthrough and relays the IEC958 > status bits on to the kernel driver so it can switch to HBR audio > packets if needed. The control hook is marked optional because kernels > before 5.14 didn't expose the IEC958 controls. > > The default pcm never worked as dmix doesn't support the iec958 format. > So drop dmix, only use plug and softvol, and use the hdmi pcm for > iec958 formatting. I can confirm the upstream cards/vc4-hdmi.conf is not usable, and RPiOS ships a patched version (which was an earlier version of this patch). I've tested this version with RPiOS bullseye and all seems good. I could play stereo and multi-channel PCM from VLC. I could also play DTS, DTS-HD and TrueHD formats as passthrough using kodi. It would be good if we can switch from a custom version of vc4-hdmi.conf to an upstream version in RPiOS when this is merged. Tested-by: Dom Cobley <popcornmix(a)gmail.com>

1 0

[PATCH] ASoC: ux500: mop500: Check before clk_put() not needed
by Yihao Han 17 May '22

17 May '22

clk_put() already checks the clk ptr using !clk and IS_ERR() so there is no need to check it again before calling it. Signed-off-by: Yihao Han <hanyihao(a)vivo.com> --- sound/soc/ux500/mop500_ab8500.c | 9 +++------ 1 file changed, 3 insertions(+), 6 deletions(-) diff --git a/sound/soc/ux500/mop500_ab8500.c b/sound/soc/ux500/mop500_ab8500.c index 3e654e708f78..1ea1729984a9 100644 --- a/sound/soc/ux500/mop500_ab8500.c +++ b/sound/soc/ux500/mop500_ab8500.c @@ -433,12 +433,9 @@ void mop500_ab8500_remove(struct snd_soc_card *card) { struct mop500_ab8500_drvdata *drvdata = snd_soc_card_get_drvdata(card); - if (drvdata->clk_ptr_sysclk != NULL) - clk_put(drvdata->clk_ptr_sysclk); - if (drvdata->clk_ptr_ulpclk != NULL) - clk_put(drvdata->clk_ptr_ulpclk); - if (drvdata->clk_ptr_intclk != NULL) - clk_put(drvdata->clk_ptr_intclk); + clk_put(drvdata->clk_ptr_sysclk); + clk_put(drvdata->clk_ptr_ulpclk); + clk_put(drvdata->clk_ptr_intclk); snd_soc_card_set_drvdata(card, drvdata); } -- 2.17.1

2 1

[PATCH] ASoC: SOF: ipc-msg-injector: Cap the rmaining to count in IPC4 mode
by Peter Ujfalusi 17 May '22

17 May '22

If user space provides smaller buffer than the IPC4 reply then it is possible that we corrupt user space memory since the IPC4 dfs_read function is not using the count directly in copy_to_user() due to the nature of an IPC4 message. Cap the remaining counter to make sure that we are not writing too much to the user space provided buffer. Add a check also to make sure that the buffer is at least the size of the IPC4 header. Fixes: 066c67624d8c: "ASoC: SOF: ipc-msg-injector: Add support for IPC4 messages" Reported-by: Dan Carpenter <dan.carpenter(a)oracle.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> --- sound/soc/sof/sof-client-ipc-msg-injector.c | 15 +++++++++++---- 1 file changed, 11 insertions(+), 4 deletions(-) diff --git a/sound/soc/sof/sof-client-ipc-msg-injector.c b/sound/soc/sof/sof-client-ipc-msg-injector.c index c2480317730c..03490a4d4ae7 100644 --- a/sound/soc/sof/sof-client-ipc-msg-injector.c +++ b/sound/soc/sof/sof-client-ipc-msg-injector.c @@ -76,12 +76,17 @@ static ssize_t sof_msg_inject_ipc4_dfs_read(struct file *file, struct sof_client_dev *cdev = file->private_data; struct sof_msg_inject_priv *priv = cdev->data; struct sof_ipc4_msg *ipc4_msg = priv->rx_buffer; + size_t header_size = sizeof(ipc4_msg->header_u64); size_t remaining; if (!ipc4_msg->header_u64 || !count || *ppos) return 0; - remaining = sizeof(ipc4_msg->header_u64); + /* we need space for the header at minimum (u64) */ + if (count < header_size) + return -ENOSPC; + + remaining = header_size; /* Only get large config have payload */ if (SOF_IPC4_MSG_IS_MODULE_MSG(ipc4_msg->primary) && @@ -90,13 +95,15 @@ static ssize_t sof_msg_inject_ipc4_dfs_read(struct file *file, if (count > remaining) count = remaining; + else if (count < remaining) + remaining = count; /* copy the header first */ - if (copy_to_user(buffer, &ipc4_msg->header_u64, sizeof(ipc4_msg->header_u64))) + if (copy_to_user(buffer, &ipc4_msg->header_u64, header_size)) return -EFAULT; - *ppos += sizeof(ipc4_msg->header_u64); - remaining -= sizeof(ipc4_msg->header_u64); + *ppos += header_size; + remaining -= header_size; if (!remaining) return count; -- 2.36.1

2 1

[PATCH -next] ASoC: amd: vangogh: Remove duplicate include files
by YueHaibing 17 May '22

17 May '22

Remove duplicated includes. Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/soc/amd/vangogh/acp5x-mach.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/amd/vangogh/acp5x-mach.c b/sound/soc/amd/vangogh/acp5x-mach.c index d8b25622f911..727de46860b1 100644 --- a/sound/soc/amd/vangogh/acp5x-mach.c +++ b/sound/soc/amd/vangogh/acp5x-mach.c @@ -17,10 +17,8 @@ #include <linux/clk.h> #include <linux/gpio.h> #include <linux/gpio/consumer.h> -#include <linux/module.h> #include <linux/i2c.h> #include <linux/input.h> -#include <linux/io.h> #include <linux/acpi.h> #include <linux/dmi.h> -- 2.17.1

2 1

[PATCH 0/3] ASoC: fsl_sai: Add support for i.MX8MM, MP, ULP
by Shengjiu Wang 17 May '22

17 May '22

ASoC: fsl_sai: Add support for i.MX8MM, MP, ULP platforms Shengjiu Wang (3): ASoC: fsl_sai: Add support for i.MX8MM ASoC: fsl_sai: Add support for i.MX8M Plus ASoC: fsl_sai: Add support for i.MX8ULP sound/soc/fsl/fsl_sai.c | 43 ++++++++++++++++++++++++++++++++++++++++- sound/soc/fsl/fsl_sai.h | 1 + 2 files changed, 43 insertions(+), 1 deletion(-) -- 2.17.1

3 5

[PATCH v0] ASoC: rt5645: Fix errorenous cleanup order
by Lin Ma 17 May '22

17 May '22

There is a logic error when removing rt5645 device as the function rt5645_i2c_remove() first cancel the &rt5645->jack_detect_work and delete the &rt5645->btn_check_timer latter. However, since the timer handler rt5645_btn_check_callback() will re-queue the jack_detect_work, this cleanup order is buggy. That is, once the del_timer_sync in rt5645_i2c_remove is concurrently run with the rt5645_btn_check_callback, the canceled jack_detect_work will be rescheduled again, leading to possible use-after-free. This patch fix the issue by placing the del_timer_sync function before the cancel_delayed_work_sync. Signed-off-by: Lin Ma <linma(a)zju.edu.cn> --- sound/soc/codecs/rt5645.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 197c56047947..71a8990ba548 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -4154,9 +4154,14 @@ static int rt5645_i2c_remove(struct i2c_client *i2c) if (i2c->irq) free_irq(i2c->irq, rt5645); + /* + * Since the rt5645_btn_check_callback() can queue jack_detect_work, + * the timer need to be delted first + */ + del_timer_sync(&rt5645->btn_check_timer); + cancel_delayed_work_sync(&rt5645->jack_detect_work); cancel_delayed_work_sync(&rt5645->rcclock_work); - del_timer_sync(&rt5645->btn_check_timer); regulator_bulk_disable(ARRAY_SIZE(rt5645->supplies), rt5645->supplies); -- 2.35.1

2 1

[PATCH 1/3] ASoC: core: Correct spelling fliped -> flipped
by Charles Keepax 17 May '22

17 May '22

Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- include/sound/soc.h | 2 +- sound/soc/generic/audio-graph-card2.c | 2 +- sound/soc/soc-core.c | 6 +++--- 3 files changed, 5 insertions(+), 5 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index f906e5a708308..f20f5f890794a 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1259,7 +1259,7 @@ int snd_soc_of_parse_audio_routing(struct snd_soc_card *card, const char *propname); int snd_soc_of_parse_aux_devs(struct snd_soc_card *card, const char *propname); -unsigned int snd_soc_daifmt_clock_provider_fliped(unsigned int dai_fmt); +unsigned int snd_soc_daifmt_clock_provider_flipped(unsigned int dai_fmt); unsigned int snd_soc_daifmt_clock_provider_from_bitmap(unsigned int bit_frame); unsigned int snd_soc_daifmt_parse_format(struct device_node *np, const char *prefix); diff --git a/sound/soc/generic/audio-graph-card2.c b/sound/soc/generic/audio-graph-card2.c index 2b5d20f02f8f8..77ac4051b8276 100644 --- a/sound/soc/generic/audio-graph-card2.c +++ b/sound/soc/generic/audio-graph-card2.c @@ -711,7 +711,7 @@ static void graph_link_init(struct asoc_simple_priv *priv, */ daiclk = snd_soc_daifmt_clock_provider_from_bitmap(bit_frame); if (is_cpu_node) - daiclk = snd_soc_daifmt_clock_provider_fliped(daiclk); + daiclk = snd_soc_daifmt_clock_provider_flipped(daiclk); dai_link->dai_fmt = daifmt | daiclk; dai_link->init = asoc_simple_dai_init; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index d68e64b73eea7..32267a38130b3 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1230,7 +1230,7 @@ int snd_soc_runtime_set_dai_fmt(struct snd_soc_pcm_runtime *rtd, /* * Flip the polarity for the "CPU" end of a CODEC<->CODEC link */ - inv_dai_fmt = snd_soc_daifmt_clock_provider_fliped(dai_fmt); + inv_dai_fmt = snd_soc_daifmt_clock_provider_flipped(dai_fmt); for_each_rtd_cpu_dais(rtd, i, cpu_dai) { unsigned int fmt = dai_fmt; @@ -3035,7 +3035,7 @@ int snd_soc_of_parse_aux_devs(struct snd_soc_card *card, const char *propname) } EXPORT_SYMBOL_GPL(snd_soc_of_parse_aux_devs); -unsigned int snd_soc_daifmt_clock_provider_fliped(unsigned int dai_fmt) +unsigned int snd_soc_daifmt_clock_provider_flipped(unsigned int dai_fmt) { unsigned int inv_dai_fmt = dai_fmt & ~SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK; @@ -3056,7 +3056,7 @@ unsigned int snd_soc_daifmt_clock_provider_fliped(unsigned int dai_fmt) return inv_dai_fmt; } -EXPORT_SYMBOL_GPL(snd_soc_daifmt_clock_provider_fliped); +EXPORT_SYMBOL_GPL(snd_soc_daifmt_clock_provider_flipped); unsigned int snd_soc_daifmt_clock_provider_from_bitmap(unsigned int bit_frame) { -- 2.30.2

3 4

[PATCH] ASoC: ti: j721e-evm: Fix refcount leak in j721e_soc_probe_*
by Miaoqian Lin 17 May '22

17 May '22

of_parse_phandle() returns a node pointer with refcount incremented, we should use of_node_put() on it when not needed anymore. Add missing of_node_put() to avoid refcount leak. Fixes: 6748d0559059 ("ASoC: ti: Add custom machine driver for j721e EVM (CPB and IVI)") Signed-off-by: Miaoqian Lin <linmq006(a)gmail.com> --- sound/soc/ti/j721e-evm.c | 44 ++++++++++++++++++++++++++++++---------- 1 file changed, 33 insertions(+), 11 deletions(-) diff --git a/sound/soc/ti/j721e-evm.c b/sound/soc/ti/j721e-evm.c index 4077e15ec48b..6a969874c927 100644 --- a/sound/soc/ti/j721e-evm.c +++ b/sound/soc/ti/j721e-evm.c @@ -630,17 +630,18 @@ static int j721e_soc_probe_cpb(struct j721e_priv *priv, int *link_idx, codec_node = of_parse_phandle(node, "ti,cpb-codec", 0); if (!codec_node) { dev_err(priv->dev, "CPB codec node is not provided\n"); - return -EINVAL; + ret = -EINVAL; + goto put_dai_node; } domain = &priv->audio_domains[J721E_AUDIO_DOMAIN_CPB]; ret = j721e_get_clocks(priv->dev, &domain->codec, "cpb-codec-scki"); if (ret) - return ret; + goto put_codec_node; ret = j721e_get_clocks(priv->dev, &domain->mcasp, "cpb-mcasp-auxclk"); if (ret) - return ret; + goto put_codec_node; /* * Common Processor Board, two links @@ -650,8 +651,10 @@ static int j721e_soc_probe_cpb(struct j721e_priv *priv, int *link_idx, comp_count = 6; compnent = devm_kzalloc(priv->dev, comp_count * sizeof(*compnent), GFP_KERNEL); - if (!compnent) - return -ENOMEM; + if (!compnent) { + ret = -ENOMEM; + goto put_codec_node; + } comp_idx = 0; priv->dai_links[*link_idx].cpus = &compnent[comp_idx++]; @@ -702,6 +705,12 @@ static int j721e_soc_probe_cpb(struct j721e_priv *priv, int *link_idx, (*conf_idx)++; return 0; + +put_codec_node: + of_node_put(codec_node); +put_dai_node: + of_node_put(dai_node); + return ret; } static int j721e_soc_probe_ivi(struct j721e_priv *priv, int *link_idx, @@ -726,23 +735,25 @@ static int j721e_soc_probe_ivi(struct j721e_priv *priv, int *link_idx, codeca_node = of_parse_phandle(node, "ti,ivi-codec-a", 0); if (!codeca_node) { dev_err(priv->dev, "IVI codec-a node is not provided\n"); - return -EINVAL; + ret = -EINVAL; + goto put_dai_node; } codecb_node = of_parse_phandle(node, "ti,ivi-codec-b", 0); if (!codecb_node) { dev_warn(priv->dev, "IVI codec-b node is not provided\n"); - return 0; + ret = 0; + goto put_codeca_node; } domain = &priv->audio_domains[J721E_AUDIO_DOMAIN_IVI]; ret = j721e_get_clocks(priv->dev, &domain->codec, "ivi-codec-scki"); if (ret) - return ret; + goto put_codecb_node; ret = j721e_get_clocks(priv->dev, &domain->mcasp, "ivi-mcasp-auxclk"); if (ret) - return ret; + goto put_codecb_node; /* * IVI extension, two links @@ -754,8 +765,10 @@ static int j721e_soc_probe_ivi(struct j721e_priv *priv, int *link_idx, comp_count = 8; compnent = devm_kzalloc(priv->dev, comp_count * sizeof(*compnent), GFP_KERNEL); - if (!compnent) - return -ENOMEM; + if (!compnent) { + ret = -ENOMEM; + goto put_codecb_node; + } comp_idx = 0; priv->dai_links[*link_idx].cpus = &compnent[comp_idx++]; @@ -816,6 +829,15 @@ static int j721e_soc_probe_ivi(struct j721e_priv *priv, int *link_idx, (*conf_idx)++; return 0; + + +put_codecb_node: + of_node_put(codecb_node); +put_codeca_node: + of_node_put(codeca_node); +put_dai_node: + of_node_put(dai_node); + return ret; } static int j721e_soc_probe(struct platform_device *pdev) -- 2.25.1

2 1

[PATCH v9 0/3] firmware: mtk: add adsp ipc protocol for SOF
by Tinghan Shen 17 May '22

17 May '22

This patch provides mediatek adsp ipc support for SOF. ADSP IPC protocol offers (send/recv) interfaces using mediatek-mailbox APIs. This patch was tested and confirmed to work with SOF fw on MT8195 cherry board and MT8186 krabby board. changes since v8: - fix patchset 2 and 3. move "depends on MTK_ADSP_IPC" from SND_SOC_SOF_MTK_COMMON to SND_SOC_SOF_MT8195/MT8186 to prevent generating wrong config. changes since v7: - rebase to linux-next/next-22020504 - use EXPORT_SYMBOL_GPL instead of EXPORT_SYMBOL in mtk-adsp-ipc.c - move mtk-adsp-ipc.c out from driver/firmware/mediatek - add user of mtk-adsp-ipc.h in patchset 2 and 3. changes since v6: - rebase to matthias.bgg/linux.git, v5.18-next/soc - Prefer "GPL" over "GPL v2" for MODULE_LICENSE changes since v5: - fix WARNING: modpost: missing MODULE_LICENSE() in drivers/mailbox /mtk-adsp-mailbox.o. Add MODULE_LICENSE in the last line. - Due to WARNING: Missing or malformed SPDX-License-Identifier tag in line 1 in checkpatch, we don't remove SPDX-License in line 1. changes since v4: - add error message for wrong mbox chan changes since v3: - rebase on v5.16-rc8 - update reviewers changes since v2: - add out tag for two memory free phases changes since v1: - add comments for mtk_adsp_ipc_send and mtk_adsp_ipc_recv - remove useless MODULE_LICENSE - change label name to out_free Allen-KH Cheng (1): ASoC: SOF: mediatek: Add ipc support for mt8195 TingHan Shen (1): firmware: mediatek: add adsp ipc protocol interface Tinghan Shen (1): ASoC: SOF: mediatek: Add mt8186 ipc support drivers/firmware/Kconfig | 9 + drivers/firmware/Makefile | 1 + drivers/firmware/mtk-adsp-ipc.c | 157 ++++++++++++++++++ .../linux/firmware/mediatek/mtk-adsp-ipc.h | 65 ++++++++ sound/soc/sof/mediatek/Kconfig | 2 + sound/soc/sof/mediatek/adsp_helper.h | 12 +- sound/soc/sof/mediatek/mt8186/mt8186-loader.c | 5 + sound/soc/sof/mediatek/mt8186/mt8186.c | 141 ++++++++++++++++ sound/soc/sof/mediatek/mt8195/mt8195.c | 138 ++++++++++++++- 9 files changed, 519 insertions(+), 11 deletions(-) create mode 100644 drivers/firmware/mtk-adsp-ipc.c create mode 100644 include/linux/firmware/mediatek/mtk-adsp-ipc.h -- 2.18.0

2 4

vc4-hdmi.conf: use a proper hdmi pcm, fix broken default pcm
by GitHub issues - opened 17 May '22

17 May '22

alsa-project/alsa-lib issue #229 was opened from macmpi: referring https://github.com/raspberrypi/linux/issues/4662#issuecomment-1114000673 and assorted [patch](https://mailman.alsa-project.org/pipermail/alsa-devel/2022-April/200… from @HiassofT Thanks for consideration. Issue URL : https://github.com/alsa-project/alsa-lib/issues/229 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH 1/2 v3] ASoC: ADAU7118: add bindings for managing pins drive strength
by Dylan Laduranty 17 May '22

17 May '22

This allows users to change SDATA and both PDM clocks pins drive strength during device probing according to their need. Signed-off-by: Dylan Laduranty <dylan.laduranty(a)mesotic.com> --- Changes since v2: Fixes commit description length sound/soc/codecs/adau7118.c | 62 ++++++++++++++++++++++++++++++++++--- 1 file changed, 58 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/adau7118.c b/sound/soc/codecs/adau7118.c index 841229dcbca1..18c1f246f911 100644 --- a/sound/soc/codecs/adau7118.c +++ b/sound/soc/codecs/adau7118.c @@ -29,6 +29,12 @@ FIELD_PREP(ADAU7118_LRCLK_BCLK_POL_MASK, x) #define ADAU7118_SPT_SLOT_MASK GENMASK(7, 4) #define ADAU7118_SPT_SLOT(x) FIELD_PREP(ADAU7118_SPT_SLOT_MASK, x) +#define ADAU7118_DS_PDM_CLK0_MASK GENMASK(1, 0) +#define ADAU7118_DS_PDM_CLK0(x) FIELD_PREP(ADAU7118_DS_PDM_CLK0_MASK, x) +#define ADAU7118_DS_PDM_CLK1_MASK GENMASK(3, 2) +#define ADAU7118_DS_PDM_CLK1(x) FIELD_PREP(ADAU7118_DS_PDM_CLK1_MASK, x) +#define ADAU7118_DS_SDATA_MASK GENMASK(5, 4) +#define ADAU7118_DS_SDATA(x) FIELD_PREP(ADAU7118_DS_SDATA_MASK, x) #define ADAU7118_FULL_SOFT_R_MASK BIT(1) #define ADAU7118_FULL_SOFT_R(x) FIELD_PREP(ADAU7118_FULL_SOFT_R_MASK, x) @@ -489,7 +495,7 @@ static int adau7118_regulator_setup(struct adau7118_data *st) static int adau7118_parset_dt(const struct adau7118_data *st) { int ret; - u32 dec_ratio = 0; + u32 val32 = 0; /* 4 inputs */ u32 clk_map[4], regval; @@ -497,9 +503,9 @@ static int adau7118_parset_dt(const struct adau7118_data *st) return 0; ret = device_property_read_u32(st->dev, "adi,decimation-ratio", - &dec_ratio); + &val32); if (!ret) { - switch (dec_ratio) { + switch (val32) { case 64: regval = ADAU7118_DEC_RATIO(0); break; @@ -510,7 +516,7 @@ static int adau7118_parset_dt(const struct adau7118_data *st) regval = ADAU7118_DEC_RATIO(2); break; default: - dev_err(st->dev, "Invalid dec ratio: %u", dec_ratio); + dev_err(st->dev, "Invalid dec ratio: %u", val32); return -EINVAL; } @@ -537,6 +543,54 @@ static int adau7118_parset_dt(const struct adau7118_data *st) return ret; } + ret = device_property_read_u32(st->dev, "adi,pdm-clk0-ds", + &val32); + if (!ret) { + if (val32 > 3) { + dev_err(st->dev, "Invalid pdm-clk0-ds: %u", val32); + return -EINVAL; + } + + ret = regmap_update_bits(st->map, + ADAU7118_REG_DRIVE_STRENGTH, + ADAU7118_DS_PDM_CLK0_MASK, + ADAU7118_DS_PDM_CLK0(val32)); + if (ret) + return ret; + } + + ret = device_property_read_u32(st->dev, "adi,pdm-clk1-ds", + &val32); + if (!ret) { + if (val32 > 3) { + dev_err(st->dev, "Invalid pdm-clk1-ds: %u", val32); + return -EINVAL; + } + + ret = regmap_update_bits(st->map, + ADAU7118_REG_DRIVE_STRENGTH, + ADAU7118_DS_PDM_CLK1_MASK, + ADAU7118_DS_PDM_CLK1(val32)); + if (ret) + return ret; + } + + ret = device_property_read_u32(st->dev, "adi,sdata-ds", + &val32); + if (!ret) { + if (val32 > 3) { + dev_err(st->dev, "Invalid sdata-ds: %u", val32); + return -EINVAL; + } + + ret = regmap_update_bits(st->map, + ADAU7118_REG_DRIVE_STRENGTH, + ADAU7118_DS_SDATA_MASK, + ADAU7118_DS_SDATA(val32)); + if (ret) + return ret; + } + return 0; } -- 2.17.1

2 4

UCM2 configuration guide
by Peter Wedder 17 May '22

17 May '22

Hello, I have been working on an article to describe the process of writing UCM2 configuration to split multi-channel devices into smaller devices, particularly in the context of USB audio interfaces. This is a common problem with enthusiast/studio USB audio interfaces and it happens to be a pet peeve of mine. I recently contributed a profile for the Behringer UMC204HD (https://github.com/alsa-project/alsa-ucm-conf/pull/128) and found the process to be a little intimidating, but with a lot of help from Perexg I managed to get it working. I would like to help other potential contributors out by explaining the process step by step. I was considering posting the article to the Fedora Magazine (https://fedoramagazine.org/), since that's the distro I use. However, I wanted to check with you, the ALSA maintainers, first, because the intent of the article is to encourage readers to contribute UCM configurations for their own cards, and I thought perhaps a potential surge of contributions is not something you want at the moment. Let me know what you think -- is this a welcome idea or should I forget it? I have a draft of the article available here: https://gist.github.com/brndd/ccec98a575f7c0422d50402937439227 Thanks, Peter Wedder

2 3

[PATCH] ALSA: seq: replace ternary operator with max()
by Guo Zhengkui 17 May '22

17 May '22

Fix the following coccicheck warning: sound/core/seq/seq_ports.c:142:12-14: WARNING opportunity for max() Signed-off-by: Guo Zhengkui <guozhengkui(a)vivo.com> --- sound/core/seq/seq_ports.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/seq/seq_ports.c b/sound/core/seq/seq_ports.c index 84d78630463e..25fcf5a2c71c 100644 --- a/sound/core/seq/seq_ports.c +++ b/sound/core/seq/seq_ports.c @@ -139,7 +139,7 @@ struct snd_seq_client_port *snd_seq_create_port(struct snd_seq_client *client, port_subs_info_init(&new_port->c_dest); snd_use_lock_use(&new_port->use_lock); - num = port >= 0 ? port : 0; + num = max(port, 0); mutex_lock(&client->ports_mutex); write_lock_irq(&client->ports_lock); list_for_each_entry(p, &client->ports_list_head, list) { -- 2.20.1

2 1

[PATCH] selftests: alsa: Better error messages
by Siddh Raman Pant 16 May '22

16 May '22

This allows for potentially better machine-parsing due to an expected / fixed format. Also because of eyecandy reasons. Whenever possible, errors will start with the name of the offending card/control, separated by the error with colons. Whenever there is a description of an error given, the generated error string from snd_strerror will be (mostly) enclosed in parentheses at the end. Clarity of error messages have been (hopefully) improved. Signed-off-by: Siddh Raman Pant <siddhpant.gh(a)gmail.com> --- tools/testing/selftests/alsa/mixer-test.c | 94 +++++++++++------------ 1 file changed, 47 insertions(+), 47 deletions(-) diff --git a/tools/testing/selftests/alsa/mixer-test.c b/tools/testing/selftests/alsa/mixer-test.c index a38b89c28..4719b12a5 100644 --- a/tools/testing/selftests/alsa/mixer-test.c +++ b/tools/testing/selftests/alsa/mixer-test.c @@ -114,7 +114,7 @@ static void find_controls(void) err = snd_config_load_string(&config, alsa_config, strlen(alsa_config)); if (err < 0) { - ksft_print_msg("Unable to parse custom alsa-lib configuration: %s\n", + ksft_print_msg("Unable to parse custom alsa-lib configuration (%s)\n", snd_strerror(err)); ksft_exit_fail(); } @@ -128,7 +128,7 @@ static void find_controls(void) err = snd_ctl_open_lconf(&card_data->handle, name, 0, config); if (err < 0) { - ksft_print_msg("Failed to get hctl for card %d: %s\n", + ksft_print_msg("Card %d: Failed to get hctl (%s)\n", card, snd_strerror(err)); goto next_card; } @@ -177,9 +177,8 @@ static void find_controls(void) err = snd_ctl_elem_info(card_data->handle, ctl_data->info); if (err < 0) { - ksft_print_msg("%s getting info for %d\n", - snd_strerror(err), - ctl_data->name); + ksft_print_msg("%s : %s while getting info\n", + ctl_data->name, snd_strerror(err)); } snd_ctl_elem_value_set_id(ctl_data->def_val, @@ -192,20 +191,20 @@ static void find_controls(void) /* Set up for events */ err = snd_ctl_subscribe_events(card_data->handle, true); if (err < 0) { - ksft_exit_fail_msg("snd_ctl_subscribe_events() failed for card %d: %d\n", + ksft_exit_fail_msg("Card %d : snd_ctl_subscribe_events() failed (%d)\n", card, err); } err = snd_ctl_poll_descriptors_count(card_data->handle); if (err != 1) { - ksft_exit_fail_msg("Unexpected descriptor count %d for card %d\n", - err, card); + ksft_exit_fail_msg("Card %d : Unexpected descriptor count %d\n", + card, err); } err = snd_ctl_poll_descriptors(card_data->handle, &card_data->pollfd, 1); if (err != 1) { - ksft_exit_fail_msg("snd_ctl_poll_descriptors() failed for %d\n", + ksft_exit_fail_msg("Card %d : snd_ctl_poll_descriptors() failed (%d)\n", card, err); } @@ -236,8 +235,8 @@ static int wait_for_event(struct ctl_data *ctl, int timeout) do { err = poll(&(ctl->card->pollfd), 1, timeout); if (err < 0) { - ksft_print_msg("poll() failed for %s: %s (%d)\n", - ctl->name, strerror(errno), errno); + ksft_print_msg("%s : poll() failed (%d - %s)\n", + ctl->name, errno, strerror(errno)); return -1; } /* Timeout */ @@ -248,12 +247,12 @@ static int wait_for_event(struct ctl_data *ctl, int timeout) &(ctl->card->pollfd), 1, &revents); if (err < 0) { - ksft_print_msg("snd_ctl_poll_descriptors_revents() failed for %s: %d\n", + ksft_print_msg("%s : snd_ctl_poll_descriptors_revents() failed (%d)\n", ctl->name, err); return err; } if (revents & POLLERR) { - ksft_print_msg("snd_ctl_poll_descriptors_revents() reported POLLERR for %s\n", + ksft_print_msg("%s : snd_ctl_poll_descriptors_revents() reported POLLERR\n", ctl->name); return -1; } @@ -265,7 +264,7 @@ static int wait_for_event(struct ctl_data *ctl, int timeout) err = snd_ctl_read(ctl->card->handle, event); if (err < 0) { - ksft_print_msg("snd_ctl_read() failed for %s: %d\n", + ksft_print_msg("%s : snd_ctl_read() failed (%d)\n", ctl->name, err); return err; } @@ -283,7 +282,7 @@ static int wait_for_event(struct ctl_data *ctl, int timeout) } if ((mask & SND_CTL_EVENT_MASK_REMOVE) == SND_CTL_EVENT_MASK_REMOVE) { - ksft_print_msg("Removal event for %s\n", + ksft_print_msg("%s : Removal event\n", ctl->name); return -1; } @@ -301,7 +300,7 @@ static bool ctl_value_index_valid(struct ctl_data *ctl, switch (snd_ctl_elem_info_get_type(ctl->info)) { case SND_CTL_ELEM_TYPE_NONE: - ksft_print_msg("%s.%d Invalid control type NONE\n", + ksft_print_msg("%s.%d : Invalid control type NONE\n", ctl->name, index); return false; @@ -312,7 +311,7 @@ static bool ctl_value_index_valid(struct ctl_data *ctl, case 1: break; default: - ksft_print_msg("%s.%d Invalid boolean value %ld\n", + ksft_print_msg("%s.%d : Invalid boolean value %ld\n", ctl->name, index, int_val); return false; } @@ -322,14 +321,14 @@ static bool ctl_value_index_valid(struct ctl_data *ctl, int_val = snd_ctl_elem_value_get_integer(val, index); if (int_val < snd_ctl_elem_info_get_min(ctl->info)) { - ksft_print_msg("%s.%d value %ld less than minimum %ld\n", + ksft_print_msg("%s.%d : Value %ld is less than the minimum (%ld)\n", ctl->name, index, int_val, snd_ctl_elem_info_get_min(ctl->info)); return false; } if (int_val > snd_ctl_elem_info_get_max(ctl->info)) { - ksft_print_msg("%s.%d value %ld more than maximum %ld\n", + ksft_print_msg("%s.%d : Value %ld is more than the maximum (%ld)\n", ctl->name, index, int_val, snd_ctl_elem_info_get_max(ctl->info)); return false; @@ -339,7 +338,7 @@ static bool ctl_value_index_valid(struct ctl_data *ctl, if (snd_ctl_elem_info_get_step(ctl->info) && (int_val - snd_ctl_elem_info_get_min(ctl->info) % snd_ctl_elem_info_get_step(ctl->info))) { - ksft_print_msg("%s.%d value %ld invalid for step %ld minimum %ld\n", + ksft_print_msg("%s.%d : Value %ld is invalid for step %ld, minimum %ld\n", ctl->name, index, int_val, snd_ctl_elem_info_get_step(ctl->info), snd_ctl_elem_info_get_min(ctl->info)); @@ -351,14 +350,14 @@ static bool ctl_value_index_valid(struct ctl_data *ctl, int64_val = snd_ctl_elem_value_get_integer64(val, index); if (int64_val < snd_ctl_elem_info_get_min64(ctl->info)) { - ksft_print_msg("%s.%d value %lld less than minimum %lld\n", + ksft_print_msg("%s.%d : Value %lld is less than the minimum (%lld)\n", ctl->name, index, int64_val, snd_ctl_elem_info_get_min64(ctl->info)); return false; } if (int64_val > snd_ctl_elem_info_get_max64(ctl->info)) { - ksft_print_msg("%s.%d value %lld more than maximum %lld\n", + ksft_print_msg("%s.%d : Value %lld is more than the maximum (%lld)\n", ctl->name, index, int64_val, snd_ctl_elem_info_get_max(ctl->info)); return false; @@ -368,7 +367,7 @@ static bool ctl_value_index_valid(struct ctl_data *ctl, if (snd_ctl_elem_info_get_step64(ctl->info) && (int64_val - snd_ctl_elem_info_get_min64(ctl->info)) % snd_ctl_elem_info_get_step64(ctl->info)) { - ksft_print_msg("%s.%d value %lld invalid for step %lld minimum %lld\n", + ksft_print_msg("%s.%d : Value %lld is invalid for step %lld, minimum %lld\n", ctl->name, index, int64_val, snd_ctl_elem_info_get_step64(ctl->info), snd_ctl_elem_info_get_min64(ctl->info)); @@ -380,13 +379,13 @@ static bool ctl_value_index_valid(struct ctl_data *ctl, int_val = snd_ctl_elem_value_get_enumerated(val, index); if (int_val < 0) { - ksft_print_msg("%s.%d negative value %ld for enumeration\n", + ksft_print_msg("%s.%d : Negative value %ld for enumeration\n", ctl->name, index, int_val); return false; } if (int_val >= snd_ctl_elem_info_get_items(ctl->info)) { - ksft_print_msg("%s.%d value %ld more than item count %ld\n", + ksft_print_msg("%s.%d : Value %ld is more than item count (%ld)\n", ctl->name, index, int_val, snd_ctl_elem_info_get_items(ctl->info)); return false; @@ -427,7 +426,7 @@ static void test_ctl_get_value(struct ctl_data *ctl) /* If the control is turned off let's be polite */ if (snd_ctl_elem_info_is_inactive(ctl->info)) { - ksft_print_msg("%s is inactive\n", ctl->name); + ksft_print_msg("%s : Inactive\n", ctl->name); ksft_test_result_skip("get_value.%d.%d\n", ctl->card->card, ctl->elem); return; @@ -435,7 +434,7 @@ static void test_ctl_get_value(struct ctl_data *ctl) /* Can't test reading on an unreadable control */ if (!snd_ctl_elem_info_is_readable(ctl->info)) { - ksft_print_msg("%s is not readable\n", ctl->name); + ksft_print_msg("%s : Not readable\n", ctl->name); ksft_test_result_skip("get_value.%d.%d\n", ctl->card->card, ctl->elem); return; @@ -443,7 +442,7 @@ static void test_ctl_get_value(struct ctl_data *ctl) err = snd_ctl_elem_read(ctl->card->handle, ctl->def_val); if (err < 0) { - ksft_print_msg("snd_ctl_elem_read() failed: %s\n", + ksft_print_msg("snd_ctl_elem_read() failed (%s)\n", snd_strerror(err)); goto out; } @@ -474,7 +473,7 @@ static void test_ctl_name(struct ctl_data *ctl) /* Only boolean controls should end in Switch */ if (strend(ctl->name, " Switch")) { if (snd_ctl_elem_info_get_type(ctl->info) != SND_CTL_ELEM_TYPE_BOOLEAN) { - ksft_print_msg("%d.%d %s ends in Switch but is not boolean\n", + ksft_print_msg("%d.%d %s : Not a boolean, but name ends in Switch\n", ctl->card->card, ctl->elem, ctl->name); name_ok = false; } @@ -484,7 +483,7 @@ static void test_ctl_name(struct ctl_data *ctl) if (snd_ctl_elem_info_get_type(ctl->info) == SND_CTL_ELEM_TYPE_BOOLEAN && snd_ctl_elem_info_is_writable(ctl->info)) { if (!strend(ctl->name, " Switch")) { - ksft_print_msg("%d.%d %s is a writeable boolean but not a Switch\n", + ksft_print_msg("%d.%d %s : Not a Switch despite being a writeable boolean\n", ctl->card->card, ctl->elem, ctl->name); name_ok = false; } @@ -542,11 +541,12 @@ static bool show_mismatch(struct ctl_data *ctl, int index, /* * NOTE: The volatile attribute means that the hardware * can voluntarily change the state of control element - * independent of any operation by software. + * independent of any operation by software. */ bool is_volatile = snd_ctl_elem_info_is_volatile(ctl->info); - ksft_print_msg("%s.%d expected %lld but read %lld, is_volatile %d\n", - ctl->name, index, expected_int, read_int, is_volatile); + ksft_print_msg("%s.%d : Expected %lld, but read %lld (%s)\n", + ctl->name, index, expected_int, read_int, + (is_volatile ? "Volatile" : "Non-volatile")); return !is_volatile; } else { return false; @@ -590,7 +590,7 @@ static int write_and_verify(struct ctl_data *ctl, err = snd_ctl_elem_read(ctl->card->handle, initial_val); if (err < 0) { - ksft_print_msg("snd_ctl_elem_read() failed: %s\n", + ksft_print_msg("snd_ctl_elem_read() failed (%s)\n", snd_strerror(err)); return err; } @@ -602,7 +602,7 @@ static int write_and_verify(struct ctl_data *ctl, */ err = snd_ctl_elem_write(ctl->card->handle, w_val); if (err < 0 && !error_expected) { - ksft_print_msg("snd_ctl_elem_write() failed: %s\n", + ksft_print_msg("snd_ctl_elem_write() failed (%s)\n", snd_strerror(err)); return err; } @@ -615,7 +615,7 @@ static int write_and_verify(struct ctl_data *ctl, err = snd_ctl_elem_read(ctl->card->handle, read_val); if (err < 0) { - ksft_print_msg("snd_ctl_elem_read() failed: %s\n", + ksft_print_msg("snd_ctl_elem_read() failed (%s)\n", snd_strerror(err)); return err; } @@ -631,13 +631,13 @@ static int write_and_verify(struct ctl_data *ctl, err = wait_for_event(ctl, 0); if (snd_ctl_elem_value_compare(initial_val, read_val)) { if (err < 1) { - ksft_print_msg("No event generated for %s\n", + ksft_print_msg("%s : No event generated\n", ctl->name); ctl->event_missing++; } } else { if (err != 0) { - ksft_print_msg("Spurious event generated for %s\n", + ksft_print_msg("%s : Spurious event generated\n", ctl->name); ctl->event_spurious++; } @@ -658,7 +658,7 @@ static int write_and_verify(struct ctl_data *ctl, mismatch_shown = true; if (!mismatch_shown) - ksft_print_msg("%s read and written values differ\n", + ksft_print_msg("%s : Read and written values differ\n", ctl->name); return -1; @@ -674,14 +674,14 @@ static void test_ctl_write_default(struct ctl_data *ctl) /* If the control is turned off let's be polite */ if (snd_ctl_elem_info_is_inactive(ctl->info)) { - ksft_print_msg("%s is inactive\n", ctl->name); + ksft_print_msg("%s : Inactive\n", ctl->name); ksft_test_result_skip("write_default.%d.%d\n", ctl->card->card, ctl->elem); return; } if (!snd_ctl_elem_info_is_writable(ctl->info)) { - ksft_print_msg("%s is not writeable\n", ctl->name); + ksft_print_msg("%s : Not writeable\n", ctl->name); ksft_test_result_skip("write_default.%d.%d\n", ctl->card->card, ctl->elem); return; @@ -689,7 +689,7 @@ static void test_ctl_write_default(struct ctl_data *ctl) /* No idea what the default was for unreadable controls */ if (!snd_ctl_elem_info_is_readable(ctl->info)) { - ksft_print_msg("%s couldn't read default\n", ctl->name); + ksft_print_msg("%s : Couldn't read default\n", ctl->name); ksft_test_result_skip("write_default.%d.%d\n", ctl->card->card, ctl->elem); return; @@ -808,14 +808,14 @@ static void test_ctl_write_valid(struct ctl_data *ctl) /* If the control is turned off let's be polite */ if (snd_ctl_elem_info_is_inactive(ctl->info)) { - ksft_print_msg("%s is inactive\n", ctl->name); + ksft_print_msg("%s : Inactive\n", ctl->name); ksft_test_result_skip("write_valid.%d.%d\n", ctl->card->card, ctl->elem); return; } if (!snd_ctl_elem_info_is_writable(ctl->info)) { - ksft_print_msg("%s is not writeable\n", ctl->name); + ksft_print_msg("%s : Not writeable\n", ctl->name); ksft_test_result_skip("write_valid.%d.%d\n", ctl->card->card, ctl->elem); return; @@ -868,7 +868,7 @@ static bool test_ctl_write_invalid_value(struct ctl_data *ctl, /* ...but some devices will clamp to an in range value */ err = snd_ctl_elem_read(ctl->card->handle, val); if (err < 0) { - ksft_print_msg("%s failed to read: %s\n", + ksft_print_msg("%s : Failed to read (%s)\n", ctl->name, snd_strerror(err)); return true; } @@ -1026,14 +1026,14 @@ static void test_ctl_write_invalid(struct ctl_data *ctl) /* If the control is turned off let's be polite */ if (snd_ctl_elem_info_is_inactive(ctl->info)) { - ksft_print_msg("%s is inactive\n", ctl->name); + ksft_print_msg("%s : Inactive\n", ctl->name); ksft_test_result_skip("write_invalid.%d.%d\n", ctl->card->card, ctl->elem); return; } if (!snd_ctl_elem_info_is_writable(ctl->info)) { - ksft_print_msg("%s is not writeable\n", ctl->name); + ksft_print_msg("%s : Not writeable\n", ctl->name); ksft_test_result_skip("write_invalid.%d.%d\n", ctl->card->card, ctl->elem); return; -- 2.35.1

3 4

[PATCH] kselftest: alsa: handle cross compilation
by Antonio Terceiro 16 May '22

16 May '22

Calling just `pkg-config` is adequate for native builds, but finding the foreign libraries with pkg-config needs pkg-config to be called via its architecture-specific wrapper. This works in Debian, where there is a corresponding *-pkg-config wrapper script for each enabled foreign architecture, just like there are *-gcc, *-ld. Signed-off-by: Antonio Terceiro <antonio.terceiro(a)linaro.org> --- tools/testing/selftests/alsa/Makefile | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/tools/testing/selftests/alsa/Makefile b/tools/testing/selftests/alsa/Makefile index f64d9090426d..f5675b3c929d 100644 --- a/tools/testing/selftests/alsa/Makefile +++ b/tools/testing/selftests/alsa/Makefile @@ -1,8 +1,10 @@ # SPDX-License-Identifier: GPL-2.0 # -CFLAGS += $(shell pkg-config --cflags alsa) -LDLIBS += $(shell pkg-config --libs alsa) +PKG_CONFIG ?= $(CROSS_COMPILE)pkg-config + +CFLAGS += $(shell $(PKG_CONFIG) --cflags alsa) +LDLIBS += $(shell $(PKG_CONFIG) --libs alsa) TEST_GEN_PROGS := mixer-test -- 2.35.1

2 1

Complete support for "Pro" USB multichannel cards
by GitHub issues - opened 16 May '22

16 May '22

alsa-project/alsa-ucm-conf issue #154 was opened from perexg: - [x] channel split using alsa-lib - [ ] channel split inside app - [ ] channel split hide the specific configs (alsa-lib, app side support) - [x] mixer control remap using alsa-lib - [ ] simplify the mixer control remap Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/154 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH 1/2] Revert "ASoC: amd: acp: Set gpio_spkr_en to None for max speaker amplifer in machine driver"
by V sujith kumar Reddy 16 May '22

16 May '22

RT1019 codec has two ways of controlling the en_spkr. one way is controlling through gpio pin method the another way is through codec register update. Now Speaker enable/disable is controlled through register update in BIOS. So this patch reverse gpio logic, which is no longer in use. This reverts commit 7fa5c33d043160eba3be9fb8e21588dff2a467c7. Signed-off-by: V sujith kumar Reddy <Vsujithkumar.Reddy(a)amd.com> --- sound/soc/amd/acp/acp-mach.h | 1 + sound/soc/amd/acp/acp-sof-mach.c | 4 ++-- 2 files changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/amd/acp/acp-mach.h b/sound/soc/amd/acp/acp-mach.h index c855f50d6b34..fd6299844ebe 100644 --- a/sound/soc/amd/acp/acp-mach.h +++ b/sound/soc/amd/acp/acp-mach.h @@ -21,6 +21,7 @@ #include <linux/gpio/consumer.h> #define EN_SPKR_GPIO_GB 0x11F +#define EN_SPKR_GPIO_NK 0x146 #define EN_SPKR_GPIO_NONE -EINVAL enum be_id { diff --git a/sound/soc/amd/acp/acp-sof-mach.c b/sound/soc/amd/acp/acp-sof-mach.c index bf61a1726f0e..8243765d490f 100644 --- a/sound/soc/amd/acp/acp-sof-mach.c +++ b/sound/soc/amd/acp/acp-sof-mach.c @@ -37,7 +37,7 @@ static struct acp_card_drvdata sof_rt5682_max_data = { .hs_codec_id = RT5682, .amp_codec_id = MAX98360A, .dmic_codec_id = DMIC, - .gpio_spkr_en = EN_SPKR_GPIO_NONE, + .gpio_spkr_en = EN_SPKR_GPIO_NK, }; static struct acp_card_drvdata sof_rt5682s_rt1019_data = { @@ -56,7 +56,7 @@ static struct acp_card_drvdata sof_rt5682s_max_data = { .hs_codec_id = RT5682S, .amp_codec_id = MAX98360A, .dmic_codec_id = DMIC, - .gpio_spkr_en = EN_SPKR_GPIO_NONE, + .gpio_spkr_en = EN_SPKR_GPIO_NK, }; static const struct snd_kcontrol_new acp_controls[] = { -- 2.25.1

3 3

[PATCH] ASoC: max98088: add support for reg_4a_cfg_bypass reg
by Tommaso Merciai 16 May '22

16 May '22

Add mixer controls support for M98088_REG_4A_CFG_BYPASS register References: - https://datasheets.maximintegrated.com/en/ds/MAX98089.pdf p71, p113 Signed-off-by: Tommaso Merciai <tommaso.merciai(a)amarulasolutions.com> --- sound/soc/codecs/max98088.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c index 429717d4ac5a..f8ec2f164e08 100644 --- a/sound/soc/codecs/max98088.c +++ b/sound/soc/codecs/max98088.c @@ -486,6 +486,11 @@ static const struct snd_kcontrol_new max98088_snd_controls[] = { SOC_SINGLE("EQ1 Switch", M98088_REG_49_CFG_LEVEL, 0, 1, 0), SOC_SINGLE("EQ2 Switch", M98088_REG_49_CFG_LEVEL, 1, 1, 0), + SOC_SINGLE("SPK Bypass Switch", M98088_REG_4A_CFG_BYPASS, 0, 1, 0), + SOC_SINGLE("REC Bypass Switch", M98088_REG_4A_CFG_BYPASS, 1, 1, 0), + SOC_SINGLE("MIC2 Bypass Switch", M98088_REG_4A_CFG_BYPASS, 4, 1, 0), + SOC_SINGLE("INA Bypass Switch", M98088_REG_4A_CFG_BYPASS, 7, 1, 0), + SOC_ENUM("EX Limiter Mode", max98088_exmode_enum), SOC_ENUM("EX Limiter Threshold", max98088_ex_thresh_enum), -- 2.25.1

2 9

[PATCH] ALSA: usb-audio: Restore Rane SL-1 quirk
by Takashi Iwai 16 May '22

16 May '22

At cleaning up and moving the device rename from the quirk table to its own table, we removed the entry for Rane SL-1 as we thought it's only for renaming. It turned out, however, that the quirk is required for matching with the device that declares itself as no standard audio but only as vendor-specific. Restore the quirk entry for Rane SL-1 to fix the regression. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=215887 Fixes: 5436f59bc5bc ("ALSA: usb-audio: Move device rename and profile quirks to an internal table") Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/quirks-table.h | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 0ea39565e623..40a5e3eb4ef2 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -3235,6 +3235,15 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, +/* Rane SL-1 */ +{ + USB_DEVICE(0x13e5, 0x0001), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + } +}, + /* disabled due to regression for other devices; * see https://bugzilla.kernel.org/show_bug.cgi?id=199905 */ -- 2.35.3

1 0

[bug report] ASoC: SOF: ipc-msg-injector: Add support for IPC4 messages
by Dan Carpenter 16 May '22

16 May '22

Hello Peter Ujfalusi, The patch 066c67624d8c: "ASoC: SOF: ipc-msg-injector: Add support for IPC4 messages" from May 6, 2022, leads to the following Smatch static checker warning: sound/soc/sof/sof-client-ipc-msg-injector.c:95 sof_msg_inject_ipc4_dfs_read() warn: userbuf overflow? is '8' <= 'count' sound/soc/sof/sof-client-ipc-msg-injector.c 72 static ssize_t sof_msg_inject_ipc4_dfs_read(struct file *file, 73 char __user *buffer, 74 size_t count, loff_t *ppos) 75 { 76 struct sof_client_dev *cdev = file->private_data; 77 struct sof_msg_inject_priv *priv = cdev->data; 78 struct sof_ipc4_msg *ipc4_msg = priv->rx_buffer; 79 size_t remaining; 80 81 if (!ipc4_msg->header_u64 || !count || *ppos) 82 return 0; 83 84 remaining = sizeof(ipc4_msg->header_u64); 85 86 /* Only get large config have payload */ 87 if (SOF_IPC4_MSG_IS_MODULE_MSG(ipc4_msg->primary) && 88 (SOF_IPC4_MSG_TYPE_GET(ipc4_msg->primary) == SOF_IPC4_MOD_LARGE_CONFIG_GET)) 89 remaining += ipc4_msg->data_size; 90 91 if (count > remaining) 92 count = remaining; 93 94 /* copy the header first */ --> 95 if (copy_to_user(buffer, &ipc4_msg->header_u64, sizeof(ipc4_msg->header_u64))) 96 return -EFAULT; 97 98 *ppos += sizeof(ipc4_msg->header_u64); 99 remaining -= sizeof(ipc4_msg->header_u64); 100 101 if (!remaining) 102 return count; 103 104 if (remaining > ipc4_msg->data_size) 105 remaining = ipc4_msg->data_size; 106 107 /* Copy the payload */ 108 if (copy_to_user(buffer + *ppos, ipc4_msg->data_ptr, remaining)) ^^^^^^^^^^^^^^^ Potentially writing more than count bytes resulting in corrupting the user space memory. 109 return -EFAULT; 110 111 *ppos += remaining; 112 return count; 113 } regards, dan carpenter

2 1

snd_pcm_wait_nocheck - use more finer timeout for the the poll wait
by GitHub issues - edited 16 May '22

16 May '22

alsa-project/alsa-lib issue #228 was edited from takaswie: Axfer calls polling system call(i.e. poll(2), epoll(7), select(7)) without explicit timeout when running for IRQ-based scheduling model. In a case that driver achieves to control sound hardware for periodical hardware IRQ, the call can return due to any event queued by the driver in hardware IRQ context. However, in a case that driver fails to control it, the call cannot return. Especially, this situation easily occurs when developers work to support hardware newly. It's better to care for the situation. How to reproduce: 1. prepare sound hardware which driver fail to control for periodical hardware IRQ. 2. execute axfer for the driver/hardware with options to: * use IRQ-based scheduling model: `--sched-model=irq` * use polling system call: without `--test-nowait` 3. the running process blocks as a call of polling system call (defaultly poll(2) by snd_pcm_wait()) Issue URL : https://github.com/alsa-project/alsa-lib/issues/228 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

axfer: call of polling system call never return when driver fails to queue no period-wakeup event due to the lack of hw IRQ context
by GitHub issues - edited 16 May '22

16 May '22

alsa-project/alsa-lib issue #228 was edited from takaswie: Axfer calls polling system call(i.e. poll(2), epoll(7), select(7)) without explicit timeout when running for IRQ-based scheduling model. In a case that driver achieves to control sound hardware for periodical hardware IRQ, the call can return due to any event queued by the driver in hardware IRQ context. However, in a case that driver fails to control it, the call cannot return. Especially, this situation easily occurs when developers work to support hardware newly. It's better to care for the situation. How to reproduce: 1. prepare sound hardware which driver fail to control for periodical hardware IRQ. 2. execute axfer for the driver/hardware with options to: * use IRQ-based scheduling model: `--sched-model=irq` * use polling system call: without `--test-nowait` 3. the running process blocks as a call of polling system call (defaultly poll(2) by snd_pcm_wait()) Issue URL : https://github.com/alsa-project/alsa-lib/issues/228 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH] ALSA: hda/realtek: fix right sounds and mute/micmute LEDs for HP machine
by Andy Chi 16 May '22

16 May '22

* The HP EliteBook 630 is using ALC236 codec which used 0x02 to control mute LED and 0x01 to control micmute LED. Therefore, add a quirk to make it works. Signed-off-by: Andy Chi <andy.chi(a)canonical.com> --- 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 33c439578a61..3a1500e487eb 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9091,6 +9091,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x8995, "HP EliteBook 855 G9", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x103c, 0x89a4, "HP ProBook 440 G9", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x89a6, "HP ProBook 450 G9", ALC236_FIXUP_HP_GPIO_LED), + 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, 0x89c3, "Zbook Studio G9", ALC245_FIXUP_CS35L41_SPI_4_HP_GPIO_LED), -- 2.25.1

2 1

[PATCH net-next] ALSA: ice1724: return void from snd_vt1724_chip_init()
by Lu Wei 16 May '22

16 May '22

Return int in snd_vt1724_chip_init() is useless since it always returns 0. Signed-off-by: Lu Wei <luwei32(a)huawei.com> --- sound/pci/ice1712/ice1724.c | 14 ++++---------- 1 file changed, 4 insertions(+), 10 deletions(-) diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c index 6fab2ad85bbe..dfef468ad6dc 100644 --- a/sound/pci/ice1712/ice1724.c +++ b/sound/pci/ice1712/ice1724.c @@ -2347,7 +2347,7 @@ static void snd_vt1724_chip_reset(struct snd_ice1712 *ice) msleep(10); } -static int snd_vt1724_chip_init(struct snd_ice1712 *ice) +static void snd_vt1724_chip_init(struct snd_ice1712 *ice) { outb(ice->eeprom.data[ICE_EEP2_SYSCONF], ICEREG1724(ice, SYS_CFG)); outb(ice->eeprom.data[ICE_EEP2_ACLINK], ICEREG1724(ice, AC97_CFG)); @@ -2369,8 +2369,6 @@ static int snd_vt1724_chip_init(struct snd_ice1712 *ice) * since they cause machine lockups */ outb(VT1724_MULTI_FIFO_ERR, ICEMT1724(ice, DMA_INT_MASK)); - - return 0; } static int snd_vt1724_spdif_build_controls(struct snd_ice1712 *ice) @@ -2506,8 +2504,8 @@ static int snd_vt1724_create(struct snd_card *card, snd_vt1724_chip_reset(ice); if (snd_vt1724_read_eeprom(ice, modelname) < 0) return -EIO; - if (snd_vt1724_chip_init(ice) < 0) - return -EIO; + + snd_vt1724_chip_init(ice); return 0; } @@ -2702,11 +2700,7 @@ static int snd_vt1724_resume(struct device *dev) return 0; snd_vt1724_chip_reset(ice); - - if (snd_vt1724_chip_init(ice) < 0) { - snd_card_disconnect(card); - return -EIO; - } + snd_vt1724_chip_init(ice); if (ice->pm_resume) ice->pm_resume(ice); -- 2.17.1

1 0

Weaknesses in alsa_info on new Ubuntu systems
by GitHub issues - opened 15 May '22

15 May '22

alsa-project/alsa-utils issue #146 was opened from mhampl: There are some weaknesses on current Ubuntu systems (e.g. release 22.04) where the alsa_info scripts does not deliver the desired information: The command "tree" is no more provided by default (but can be installed in addition) This results in the error message "./alsa-info.sh: line 661: tree: command not found" and causes missing output in "Sysfs card info" The command "dmesg" requires administrative authorizations (e.g. "sudo dmesg | grep ...") This results in the error message "dmesg: read kernel buffer failed: Operation not permitted" and causes missing output in "ALSA/HDA dmesg" Issue URL : https://github.com/alsa-project/alsa-utils/issues/146 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

Missing documentation
by GitHub issues - opened 15 May '22

15 May '22

alsa-project/alsa-lib issue #227 was opened from fedeb95: Hi, this probably isn't the right place, but I couldn't find anything specific for documentation bugs. The links under https://www.alsa-project.org/wiki/ALSA_Driver_Documentation currently give a 404. Issue URL : https://github.com/alsa-project/alsa-lib/issues/227 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

chore: Unify Travis CI process to GitHub Actions
by GitHub pull_request - opened 15 May '22

15 May '22

alsa-project/alsa-lib pull request #226 was opened from Korilakkuma: [Travis are now recommending removing the sudo tag](https://blog.travis-ci.com/2018-11-19-required-linux-infrastructure-mi…. Request URL : https://github.com/alsa-project/alsa-lib/pull/226 Patch URL : https://github.com/alsa-project/alsa-lib/pull/226.patch Repository URL: https://github.com/alsa-project/alsa-lib

1 0

chore: Add link to GitHub Actions
by GitHub pull_request - opened 15 May '22

15 May '22

alsa-project/alsa-lib pull request #225 was opened from Korilakkuma: This PR enables to transition GitHub Actions by clicking badge. Request URL : https://github.com/alsa-project/alsa-lib/pull/225 Patch URL : https://github.com/alsa-project/alsa-lib/pull/225.patch Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH v2] ASoC: hdmi-codec: Add option for ELD bypass
by Sugar Zhang 14 May '22

14 May '22

This patch allow users to enable "ELD Bypass" who don't care anything from EDID Link Data. Currently, this driver gets ELD(from EDID) to constraint channels and rates. Unfortunately, EDID is not always valid, maybe caused by the fragile HDMI port or cable, in this situation, the max features are limited to 48kHz stereo. So, add this option to allow user to select the manual way to output audio as expected. such as multi-channels LPCM(7.1), or HBR bitstream for these sink devices. Signed-off-by: Sugar Zhang <sugar.zhang(a)rock-chips.com> --- Changes in v2: - Use MACRO SOC_SINGLE_BOOL_EXT to simplify code. Fix event_missing checked by mixer-test. Add suffix "Switch" for "ELD Bypass". sound/soc/codecs/hdmi-codec.c | 32 ++++++++++++++++++++++++++++++-- 1 file changed, 30 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c index b07607a..be46fbd 100644 --- a/sound/soc/codecs/hdmi-codec.c +++ b/sound/soc/codecs/hdmi-codec.c @@ -275,6 +275,7 @@ struct hdmi_codec_priv { unsigned int chmap_idx; struct mutex lock; bool busy; + bool eld_bypass; struct snd_soc_jack *jack; unsigned int jack_status; u8 iec_status[AES_IEC958_STATUS_SIZE]; @@ -427,6 +428,31 @@ static int hdmi_codec_iec958_mask_get(struct snd_kcontrol *kcontrol, return 0; } +static int hdmi_codec_eld_bypass_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct hdmi_codec_priv *hcp = snd_soc_component_get_drvdata(component); + + ucontrol->value.integer.value[0] = hcp->eld_bypass; + + return 0; +} + +static int hdmi_codec_eld_bypass_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct hdmi_codec_priv *hcp = snd_soc_component_get_drvdata(component); + + if (hcp->eld_bypass == ucontrol->value.integer.value[0]) + return 0; + + hcp->eld_bypass = ucontrol->value.integer.value[0]; + + return 1; +} + static int hdmi_codec_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { @@ -447,7 +473,7 @@ static int hdmi_codec_startup(struct snd_pcm_substream *substream, goto err; } - if (tx && hcp->hcd.ops->get_eld) { + if (tx && !hcp->eld_bypass && hcp->hcd.ops->get_eld) { ret = hcp->hcd.ops->get_eld(dai->dev->parent, hcp->hcd.data, hcp->eld, sizeof(hcp->eld)); if (ret) @@ -770,6 +796,8 @@ static struct snd_kcontrol_new hdmi_codec_controls[] = { .info = hdmi_eld_ctl_info, .get = hdmi_eld_ctl_get, }, + SOC_SINGLE_BOOL_EXT("ELD Bypass Switch", 0, + hdmi_codec_eld_bypass_get, hdmi_codec_eld_bypass_put), }; static int hdmi_codec_pcm_new(struct snd_soc_pcm_runtime *rtd, @@ -854,7 +882,7 @@ static void plugged_cb(struct device *dev, bool plugged) struct hdmi_codec_priv *hcp = dev_get_drvdata(dev); if (plugged) { - if (hcp->hcd.ops->get_eld) { + if (!hcp->eld_bypass && hcp->hcd.ops->get_eld) { hcp->hcd.ops->get_eld(dev->parent, hcp->hcd.data, hcp->eld, sizeof(hcp->eld)); } -- 2.7.4

3 3

Add "Pro Audio" profile for UMC204HD
by GitHub pull_request - opened 14 May '22

14 May '22

alsa-project/alsa-ucm-conf pull request #153 was opened from brndd: Follow-up to #128 I figured that some users might prefer to keep the channel mapping untouched from what the card defaults to when unconfigured. Providing an alternative profile like this (implementation adapted from #145) for these users seems like a good idea. I believe PipeWire autogenerates profiles like these for unconfigured devices, so that's where the "Pro Audio" nomenclature comes from. Likely a good idea to match that since PipeWire is increasingly the default on most distributions. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/153 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/153.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH] sound: sound_core: changed spaces to tabs
by Paul Lemmermann 14 May '22

14 May '22

changed spaces to tabs in sound_core.c according to kernel styling guidelines. found using checkpatch.pl. Signed-off-by: Paul Lemmermann <thepaulodoom(a)thepaulodoom.com> --- sound/sound_core.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/sound_core.c b/sound/sound_core.c index 3332fe321..28bc038dc 100644 --- a/sound/sound_core.c +++ b/sound/sound_core.c @@ -290,7 +290,7 @@ static int sound_insert_unit(struct sound_unit **list, const struct file_operati * completed the removal before their file operations become * invalid. */ - + static void sound_remove_unit(struct sound_unit **list, int unit) { struct sound_unit *p; @@ -395,12 +395,12 @@ int register_sound_special_device(const struct file_operations *fops, int unit, name = "admmidi"; break; default: - { + { __unknown: sprintf(_name, "unknown%d", chain); - if (unit >= SOUND_STEP) - strcat(_name, "-"); - name = _name; + if (unit >= SOUND_STEP) + strcat(_name, "-"); + name = _name; } break; } -- 2.35.1

1 0

Re: [PATCH 19/32] afs: Use mem_to_flex_dup() with struct afs_acl
by David Howells 13 May '22

13 May '22

Kees Cook <keescook(a)chromium.org> wrote: > struct afs_acl { > - u32 size; > - u8 data[]; > + DECLARE_FLEX_ARRAY_ELEMENTS_COUNT(u32, size); > + DECLARE_FLEX_ARRAY_ELEMENTS(u8, data); > }; Oof... That's really quite unpleasant syntax. Is it not possible to have mem_to_flex_dup() and friends work without that? You are telling them the fields they have to fill in. > + struct afs_acl *acl = NULL; > > - acl = kmalloc(sizeof(*acl) + size, GFP_KERNEL); > - if (!acl) { > + if (mem_to_flex_dup(&acl, buffer, size, GFP_KERNEL)) { Please don't do that. Either do: acl = mem_to_flex_dup(buffer, size, GFP_KERNEL); if (!acl) or: acl = mem_to_flex_dup(buffer, size, GFP_KERNEL); if (IS_ERR(acl)) Please especially don't make it that an apparent 'true' return indicates an error. If you absolutely must return the acl pointer through the argument list (presumably because it's actually a macro), make it return false on failure: if (!mem_to_flex_dup(&acl, buffer, size, GFP_KERNEL)) or return and explicitly check for an error code: if (mem_to_flex_dup(&acl, buffer, size, GFP_KERNEL) < 0) or: ret = mem_to_flex_dup(&acl, buffer, size, GFP_KERNEL); if (ret < 0) (or use != 0 rather than < 0) David

2 1

[PATCH 1/2 v2] ASoC: ADAU7118: add bindings for managing pins drive strength
by Dylan Laduranty 13 May '22

13 May '22

This allows users to change SDATA and both PDM clocks pins drive strength during device probing according to their need. Signed-off-by: Dylan Laduranty <dylan.laduranty(a)mesotic.com> --- Changes since v1: split new bindings and documentation into separate patches sound/soc/codecs/adau7118.c | 62 ++++++++++++++++++++++++++++++++++--- 1 file changed, 58 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/adau7118.c b/sound/soc/codecs/adau7118.c index 841229dcbca1..18c1f246f911 100644 --- a/sound/soc/codecs/adau7118.c +++ b/sound/soc/codecs/adau7118.c @@ -29,6 +29,12 @@ FIELD_PREP(ADAU7118_LRCLK_BCLK_POL_MASK, x) #define ADAU7118_SPT_SLOT_MASK GENMASK(7, 4) #define ADAU7118_SPT_SLOT(x) FIELD_PREP(ADAU7118_SPT_SLOT_MASK, x) +#define ADAU7118_DS_PDM_CLK0_MASK GENMASK(1, 0) +#define ADAU7118_DS_PDM_CLK0(x) FIELD_PREP(ADAU7118_DS_PDM_CLK0_MASK, x) +#define ADAU7118_DS_PDM_CLK1_MASK GENMASK(3, 2) +#define ADAU7118_DS_PDM_CLK1(x) FIELD_PREP(ADAU7118_DS_PDM_CLK1_MASK, x) +#define ADAU7118_DS_SDATA_MASK GENMASK(5, 4) +#define ADAU7118_DS_SDATA(x) FIELD_PREP(ADAU7118_DS_SDATA_MASK, x) #define ADAU7118_FULL_SOFT_R_MASK BIT(1) #define ADAU7118_FULL_SOFT_R(x) FIELD_PREP(ADAU7118_FULL_SOFT_R_MASK, x) @@ -489,7 +495,7 @@ static int adau7118_regulator_setup(struct adau7118_data *st) static int adau7118_parset_dt(const struct adau7118_data *st) { int ret; - u32 dec_ratio = 0; + u32 val32 = 0; /* 4 inputs */ u32 clk_map[4], regval; @@ -497,9 +503,9 @@ static int adau7118_parset_dt(const struct adau7118_data *st) return 0; ret = device_property_read_u32(st->dev, "adi,decimation-ratio", - &dec_ratio); + &val32); if (!ret) { - switch (dec_ratio) { + switch (val32) { case 64: regval = ADAU7118_DEC_RATIO(0); break; @@ -510,7 +516,7 @@ static int adau7118_parset_dt(const struct adau7118_data *st) regval = ADAU7118_DEC_RATIO(2); break; default: - dev_err(st->dev, "Invalid dec ratio: %u", dec_ratio); + dev_err(st->dev, "Invalid dec ratio: %u", val32); return -EINVAL; } @@ -537,6 +543,54 @@ static int adau7118_parset_dt(const struct adau7118_data *st) return ret; } + ret = device_property_read_u32(st->dev, "adi,pdm-clk0-ds", + &val32); + if (!ret) { + if (val32 > 3) { + dev_err(st->dev, "Invalid pdm-clk0-ds: %u", val32); + return -EINVAL; + } + + ret = regmap_update_bits(st->map, + ADAU7118_REG_DRIVE_STRENGTH, + ADAU7118_DS_PDM_CLK0_MASK, + ADAU7118_DS_PDM_CLK0(val32)); + if (ret) + return ret; + } + + ret = device_property_read_u32(st->dev, "adi,pdm-clk1-ds", + &val32); + if (!ret) { + if (val32 > 3) { + dev_err(st->dev, "Invalid pdm-clk1-ds: %u", val32); + return -EINVAL; + } + + ret = regmap_update_bits(st->map, + ADAU7118_REG_DRIVE_STRENGTH, + ADAU7118_DS_PDM_CLK1_MASK, + ADAU7118_DS_PDM_CLK1(val32)); + if (ret) + return ret; + } + + ret = device_property_read_u32(st->dev, "adi,sdata-ds", + &val32); + if (!ret) { + if (val32 > 3) { + dev_err(st->dev, "Invalid sdata-ds: %u", val32); + return -EINVAL; + } + + ret = regmap_update_bits(st->map, + ADAU7118_REG_DRIVE_STRENGTH, + ADAU7118_DS_SDATA_MASK, + ADAU7118_DS_SDATA(val32)); + if (ret) + return ret; + } + return 0; } -- 2.17.1

2 5

[PATCH v2] ALSA: hda/realtek: Add quirk for TongFang devices with pop noise
by Werner Sembach 13 May '22

13 May '22

When audio stops playing there is an audible "pop"-noise when using headphones on the TongFang GMxMRxx, GKxNRxx, GMxZGxx, GMxTGxx and GMxAGxx. This quirk fixes this mostly. Signed-off-by: Werner Sembach <wse(a)tuxedocomputers.com> Cc: stable(a)vger.kernel.org --- sound/pci/hda/patch_realtek.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index cf531c1efa132..8003bf7441d6d 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9290,6 +9290,14 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1c06, 0x2013, "Lemote A1802", ALC269_FIXUP_LEMOTE_A1802), SND_PCI_QUIRK(0x1c06, 0x2015, "Lemote A190X", ALC269_FIXUP_LEMOTE_A190X), SND_PCI_QUIRK(0x1d05, 0x1132, "TongFang PHxTxX1", ALC256_FIXUP_SET_COEF_DEFAULTS), + SND_PCI_QUIRK(0x1d05, 0x1096, "TongFang GMxMRxx", ALC269_FIXUP_NO_SHUTUP), + SND_PCI_QUIRK(0x1d05, 0x1100, "TongFang GKxNRxx", ALC269_FIXUP_NO_SHUTUP), + SND_PCI_QUIRK(0x1d05, 0x1111, "TongFang GMxZGxx", ALC269_FIXUP_NO_SHUTUP), + SND_PCI_QUIRK(0x1d05, 0x1119, "TongFang GMxZGxx", ALC269_FIXUP_NO_SHUTUP), + SND_PCI_QUIRK(0x1d05, 0x1129, "TongFang GMxZGxx", ALC269_FIXUP_NO_SHUTUP), + SND_PCI_QUIRK(0x1d05, 0x1147, "TongFang GMxTGxx", ALC269_FIXUP_NO_SHUTUP), + SND_PCI_QUIRK(0x1d05, 0x115c, "TongFang GMxTGxx", ALC269_FIXUP_NO_SHUTUP), + SND_PCI_QUIRK(0x1d05, 0x121b, "TongFang GMxAGxx", ALC269_FIXUP_NO_SHUTUP), SND_PCI_QUIRK(0x1d72, 0x1602, "RedmiBook", ALC255_FIXUP_XIAOMI_HEADSET_MIC), SND_PCI_QUIRK(0x1d72, 0x1701, "XiaomiNotebook Pro", ALC298_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1d72, 0x1901, "RedmiBook 14", ALC256_FIXUP_ASUS_HEADSET_MIC), -- 2.25.1

2 1

Re: [PATCH 00/32] Introduce flexible array struct memcpy() helpers
by David Howells 12 May '22

12 May '22

Kees Cook <keescook(a)chromium.org> wrote: > I'm happy to also point out that the conversions (patches 5+) are actually > a net reduction in lines of code: > 49 files changed, 154 insertions(+), 244 deletions(-) That doesn't mean that it's actually code that's clearer to read. I would say that it's actually less clear. In a bunch of places, you've done something like: - e = kmalloc(...); - if (!e) + if (__mem_to_flex_dup(&e, ...)) The problem is that, to me at least, it looks like: - e = kmalloc(...); - if (kmalloc failed) + if (__mem_to_flex_dup(&e, ...) succeeded) David

1 0

[PATCH] ALSA: hda/intel: Add quirk for TongFang devices with pop noise
by Werner Sembach 12 May '22

12 May '22

When audio stops playing and sometimes when it starts playing, there is an audible "pop" noise when using headphones on most Tongfang GMxMxxx, GKxNxxx, GMxZxxx, GMxTxxx, and GMxAxxx devices. Disabling power saving for the Realtek codec fixes this noise. Presumably it is triggered on some power event in the audio circuit. Signed-off-by: Werner Sembach <wse(a)tuxedocomputers.com> Cc: stable(a)vger.kernel.org --- sound/pci/hda/hda_intel.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 0a83eb6b88b1f..8a1088e057ec0 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2201,6 +2201,13 @@ static const struct snd_pci_quirk power_save_denylist[] = { SND_PCI_QUIRK(0x17aa, 0x36a7, "Lenovo C50 All in one", 0), /* https://bugs.launchpad.net/bugs/1821663 */ SND_PCI_QUIRK(0x1631, 0xe017, "Packard Bell NEC IMEDIA 5204", 0), + /* Several TongFang barebones making popping sounds */ + SND_PCI_QUIRK(0x1d05, 0x1096, "TongFang GMxMxxx", 0), + SND_PCI_QUIRK(0x1d05, 0x1100, "TongFang GKxNxxx", 0), + SND_PCI_QUIRK(0x1d05, 0x1111, "TongFang GMxZxxx", 0), + SND_PCI_QUIRK(0x1d05, 0x1129, "TongFang GMxZxxx", 0), + SND_PCI_QUIRK(0x1d05, 0x1147, "TongFang GMxTxxx", 0), + SND_PCI_QUIRK(0x1d05, 0x121b, "TongFang GMxAxxx", 0), {} }; #endif /* CONFIG_PM */ -- 2.25.1

2 4

[PATCH -next] ASoC: SOF: amd: add missing platform_device_unregister in acp_pci_rn_probe
by Zheng Bin 12 May '22

12 May '22

acp_pci_rn_probe misses a call platform_device_unregister in error path, this patch fixes that. Signed-off-by: Zheng Bin <zhengbin13(a)huawei.com> --- sound/soc/sof/amd/pci-rn.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/sof/amd/pci-rn.c b/sound/soc/sof/amd/pci-rn.c index b8910bb7f27c..d5d9bcc2c997 100644 --- a/sound/soc/sof/amd/pci-rn.c +++ b/sound/soc/sof/amd/pci-rn.c @@ -101,6 +101,7 @@ static int acp_pci_rn_probe(struct pci_dev *pci, const struct pci_device_id *pci res = devm_kzalloc(&pci->dev, sizeof(struct resource) * ARRAY_SIZE(renoir_res), GFP_KERNEL); if (!res) { sof_pci_remove(pci); + platform_device_unregister(dmic_dev); return -ENOMEM; } -- 2.31.1

3 2

[PATCH] ASoC: samsung: Fix refcount leak in aries_audio_probe
by Miaoqian Lin 12 May '22

12 May '22

of_parse_phandle() returns a node pointer with refcount incremented, we should use of_node_put() on it when done. If extcon_find_edev_by_node() fails, it doesn't call of_node_put() Calling of_node_put() after extcon_find_edev_by_node() to fix this. Fixes: 7a3a7671fa6c ("ASoC: samsung: Add driver for Aries boards") Signed-off-by: Miaoqian Lin <linmq006(a)gmail.com> --- sound/soc/samsung/aries_wm8994.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/samsung/aries_wm8994.c b/sound/soc/samsung/aries_wm8994.c index 5265e546b124..83acbe57b248 100644 --- a/sound/soc/samsung/aries_wm8994.c +++ b/sound/soc/samsung/aries_wm8994.c @@ -585,10 +585,10 @@ static int aries_audio_probe(struct platform_device *pdev) extcon_np = of_parse_phandle(np, "extcon", 0); priv->usb_extcon = extcon_find_edev_by_node(extcon_np); + of_node_put(extcon_np); if (IS_ERR(priv->usb_extcon)) return dev_err_probe(dev, PTR_ERR(priv->usb_extcon), "Failed to get extcon device"); - of_node_put(extcon_np); priv->adc = devm_iio_channel_get(dev, "headset-detect"); if (IS_ERR(priv->adc)) -- 2.25.1

4 3

[PATCH 0/5] ASoC: SOF: Add IPC4 FW loader support
by Ranjani Sridharan 12 May '22

12 May '22

The patches in this series add support for FW loading for IPC4 in the SOF driver. Ranjani Sridharan (5): ASoC: SOF: Intel: HDA: Set IPC4-specific DSP ops for CaVS platforms ASoC: SOF: Add IPC4 private header ASoC: SOF: Add header for IPC4 manifest ASoC: SOF: IPC4: Add FW loader ops ASoC: SOF: Intel: HDA: Set up sof_ipc4_fw_data for IPC4 include/sound/sof/ext_manifest4.h | 119 +++++++++++++++++ sound/soc/sof/Makefile | 2 +- sound/soc/sof/intel/apl.c | 29 ++++- sound/soc/sof/intel/cnl.c | 109 +++++++++++++++- sound/soc/sof/intel/hda-ipc.c | 82 ++++++++++++ sound/soc/sof/intel/hda.h | 6 + sound/soc/sof/intel/tgl.c | 29 ++++- sound/soc/sof/ipc4-loader.c | 210 ++++++++++++++++++++++++++++++ sound/soc/sof/ipc4-priv.h | 44 +++++++ sound/soc/sof/ipc4.c | 2 + 10 files changed, 619 insertions(+), 13 deletions(-) create mode 100644 include/sound/sof/ext_manifest4.h create mode 100644 sound/soc/sof/ipc4-loader.c create mode 100644 sound/soc/sof/ipc4-priv.h -- 2.25.1

2 6