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[PATCH AUTOSEL 5.4 40/43] ALSA: hda: fix a runtime pm issue in SOF when integrated GPU is disabled
by Sasha Levin 07 Sep '20

07 Sep '20

From: Rander Wang <rander.wang(a)intel.com> [ Upstream commit 13774d81f38538c5fa2924bdcdfa509155480fa6 ] In snd_hdac_device_init pm_runtime_set_active is called to increase child_count in parent device. But when it is failed to build connection with GPU for one case that integrated graphic gpu is disabled, snd_hdac_ext_bus_device_exit will be invoked to clean up a HD-audio extended codec base device. At this time the child_count of parent is not decreased, which makes parent device can't get suspended. This patch calls pm_runtime_set_suspended to decrease child_count in parent device in snd_hdac_device_exit to match with snd_hdac_device_init. pm_runtime_set_suspended can make sure that it will not decrease child_count if the device is already suspended. Signed-off-by: Rander Wang <rander.wang(a)intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Link: https://lore.kernel.org/r/20200902154218.1440441-1-kai.vehmanen@linux.intel… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/hda/hdac_device.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/hda/hdac_device.c b/sound/hda/hdac_device.c index c946fd8beebca..b84e12f4f8046 100644 --- a/sound/hda/hdac_device.c +++ b/sound/hda/hdac_device.c @@ -127,6 +127,8 @@ EXPORT_SYMBOL_GPL(snd_hdac_device_init); void snd_hdac_device_exit(struct hdac_device *codec) { pm_runtime_put_noidle(&codec->dev); + /* keep balance of runtime PM child_count in parent device */ + pm_runtime_set_suspended(&codec->dev); snd_hdac_bus_remove_device(codec->bus, codec); kfree(codec->vendor_name); kfree(codec->chip_name); -- 2.25.1

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[PATCH AUTOSEL 5.4 39/43] ALSA: hda: hdmi - add Rocketlake support
by Sasha Levin 07 Sep '20

07 Sep '20

From: Rander Wang <rander.wang(a)intel.com> [ Upstream commit f804a324a41a880c1ab43cc5145d8b3e5790430d ] Add Rocketlake HDMI codec support. Rocketlake shares the pin-to-port mapping table with Tigerlake. Signed-off-by: Rander Wang <rander.wang(a)intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Link: https://lore.kernel.org/r/20200902154207.1440393-1-kai.vehmanen@linux.intel… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/patch_hdmi.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 330fb7656c636..3e74a9181bc74 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -4204,6 +4204,7 @@ HDA_CODEC_ENTRY(0x8086280c, "Cannonlake HDMI", patch_i915_glk_hdmi), HDA_CODEC_ENTRY(0x8086280d, "Geminilake HDMI", patch_i915_glk_hdmi), HDA_CODEC_ENTRY(0x8086280f, "Icelake HDMI", patch_i915_icl_hdmi), HDA_CODEC_ENTRY(0x80862812, "Tigerlake HDMI", patch_i915_tgl_hdmi), +HDA_CODEC_ENTRY(0x80862816, "Rocketlake HDMI", patch_i915_tgl_hdmi), HDA_CODEC_ENTRY(0x8086281a, "Jasperlake HDMI", patch_i915_icl_hdmi), HDA_CODEC_ENTRY(0x80862880, "CedarTrail HDMI", patch_generic_hdmi), HDA_CODEC_ENTRY(0x80862882, "Valleyview2 HDMI", patch_i915_byt_hdmi), -- 2.25.1

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[PATCH AUTOSEL 5.4 09/43] ALSA: hda/tegra: Program WAKEEN register for Tegra
by Sasha Levin 07 Sep '20

07 Sep '20

From: Mohan Kumar <mkumard(a)nvidia.com> [ Upstream commit 23d63a31d9f44d7daeac0d1fb65c6a73c70e5216 ] The WAKEEN bits are used to indicate which bits in the STATESTS register may cause wake event during the codec state change request. Configure the WAKEEN register for the Tegra to detect the wake events. Signed-off-by: Mohan Kumar <mkumard(a)nvidia.com> Acked-by: Sameer Pujar <spujar(a)nvidia.com> Link: https://lore.kernel.org/r/20200825052415.20626-3-mkumard@nvidia.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/hda_tegra.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/pci/hda/hda_tegra.c b/sound/pci/hda/hda_tegra.c index e5191584638ab..e378cb33c69df 100644 --- a/sound/pci/hda/hda_tegra.c +++ b/sound/pci/hda/hda_tegra.c @@ -169,6 +169,10 @@ static int __maybe_unused hda_tegra_runtime_suspend(struct device *dev) struct hdac_bus *bus = azx_bus(chip); if (chip && chip->running) { + /* enable controller wake up event */ + azx_writew(chip, WAKEEN, azx_readw(chip, WAKEEN) | + STATESTS_INT_MASK); + azx_stop_chip(chip); synchronize_irq(bus->irq); azx_enter_link_reset(chip); @@ -191,6 +195,9 @@ static int __maybe_unused hda_tegra_runtime_resume(struct device *dev) if (chip && chip->running) { hda_tegra_init(hda); azx_init_chip(chip, 1); + /* disable controller wake up event*/ + azx_writew(chip, WAKEEN, azx_readw(chip, WAKEEN) & + ~STATESTS_INT_MASK); } return 0; -- 2.25.1

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[PATCH AUTOSEL 5.4 08/43] ALSA: hda: Fix 2 channel swapping for Tegra
by Sasha Levin 07 Sep '20

07 Sep '20

From: Mohan Kumar <mkumard(a)nvidia.com> [ Upstream commit 216116eae43963c662eb84729507bad95214ca6b ] The Tegra HDA codec HW implementation has an issue related to not swapping the 2 channel Audio Sample Packet(ASP) channel mapping. Whatever the FL and FR mapping specified the left channel always comes out of left speaker and right channel on right speaker. So add condition to disallow the swapping of FL,FR during the playback. Signed-off-by: Mohan Kumar <mkumard(a)nvidia.com> Acked-by: Sameer Pujar <spujar(a)nvidia.com> Link: https://lore.kernel.org/r/20200825052415.20626-2-mkumard@nvidia.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/patch_hdmi.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index ec9460f3a288e..330fb7656c636 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -3677,6 +3677,7 @@ static int tegra_hdmi_build_pcms(struct hda_codec *codec) static int patch_tegra_hdmi(struct hda_codec *codec) { + struct hdmi_spec *spec; int err; err = patch_generic_hdmi(codec); @@ -3684,6 +3685,10 @@ static int patch_tegra_hdmi(struct hda_codec *codec) return err; codec->patch_ops.build_pcms = tegra_hdmi_build_pcms; + spec = codec->spec; + spec->chmap.ops.chmap_cea_alloc_validate_get_type = + nvhdmi_chmap_cea_alloc_validate_get_type; + spec->chmap.ops.chmap_validate = nvhdmi_chmap_validate; return 0; } -- 2.25.1

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[PATCH AUTOSEL 5.8 48/53] ALSA: hda: use consistent HDAudio spelling in comments/docs
by Sasha Levin 07 Sep '20

07 Sep '20

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit b79de57b4378a93115307be6962d05b099eb0f37 ] We use HDaudio and HDAudio, pick one to make searches easier. No functionality change Also fix timestamping typo in documentation. Reported-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> 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: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Link: https://lore.kernel.org/r/20200902154250.1440585-1-kai.vehmanen@linux.intel… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- Documentation/sound/designs/timestamping.rst | 2 +- sound/hda/intel-dsp-config.c | 10 +++++----- sound/x86/Kconfig | 2 +- 3 files changed, 7 insertions(+), 7 deletions(-) diff --git a/Documentation/sound/designs/timestamping.rst b/Documentation/sound/designs/timestamping.rst index 2b0fff5034151..7c7ecf5dbc4bd 100644 --- a/Documentation/sound/designs/timestamping.rst +++ b/Documentation/sound/designs/timestamping.rst @@ -143,7 +143,7 @@ timestamp shows when the information is put together by the driver before returning from the ``STATUS`` and ``STATUS_EXT`` ioctl. in most cases this driver_timestamp will be identical to the regular system tstamp. -Examples of typestamping with HDaudio: +Examples of timestamping with HDAudio: 1. DMA timestamp, no compensation for DMA+analog delay :: diff --git a/sound/hda/intel-dsp-config.c b/sound/hda/intel-dsp-config.c index 99aec73491676..1c5114dedda92 100644 --- a/sound/hda/intel-dsp-config.c +++ b/sound/hda/intel-dsp-config.c @@ -54,7 +54,7 @@ static const struct config_entry config_table[] = { #endif /* * Apollolake (Broxton-P) - * the legacy HDaudio driver is used except on Up Squared (SOF) and + * the legacy HDAudio driver is used except on Up Squared (SOF) and * Chromebooks (SST) */ #if IS_ENABLED(CONFIG_SND_SOC_SOF_APOLLOLAKE) @@ -89,7 +89,7 @@ static const struct config_entry config_table[] = { }, #endif /* - * Skylake and Kabylake use legacy HDaudio driver except for Google + * Skylake and Kabylake use legacy HDAudio driver except for Google * Chromebooks (SST) */ @@ -135,7 +135,7 @@ static const struct config_entry config_table[] = { #endif /* - * Geminilake uses legacy HDaudio driver except for Google + * Geminilake uses legacy HDAudio driver except for Google * Chromebooks */ /* Geminilake */ @@ -157,7 +157,7 @@ static const struct config_entry config_table[] = { /* * CoffeeLake, CannonLake, CometLake, IceLake, TigerLake use legacy - * HDaudio driver except for Google Chromebooks and when DMICs are + * HDAudio driver except for Google Chromebooks and when DMICs are * present. Two cases are required since Coreboot does not expose NHLT * tables. * @@ -391,7 +391,7 @@ int snd_intel_dsp_driver_probe(struct pci_dev *pci) if (pci->class == 0x040300) return SND_INTEL_DSP_DRIVER_LEGACY; if (pci->class != 0x040100 && pci->class != 0x040380) { - dev_err(&pci->dev, "Unknown PCI class/subclass/prog-if information (0x%06x) found, selecting HDA legacy driver\n", pci->class); + dev_err(&pci->dev, "Unknown PCI class/subclass/prog-if information (0x%06x) found, selecting HDAudio legacy driver\n", pci->class); return SND_INTEL_DSP_DRIVER_LEGACY; } diff --git a/sound/x86/Kconfig b/sound/x86/Kconfig index 77777192f6508..4ffcc5e623c22 100644 --- a/sound/x86/Kconfig +++ b/sound/x86/Kconfig @@ -9,7 +9,7 @@ menuconfig SND_X86 if SND_X86 config HDMI_LPE_AUDIO - tristate "HDMI audio without HDaudio on Intel Atom platforms" + tristate "HDMI audio without HDAudio on Intel Atom platforms" depends on DRM_I915 select SND_PCM help -- 2.25.1

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[PATCH AUTOSEL 5.8 47/53] ALSA: hda: fix a runtime pm issue in SOF when integrated GPU is disabled
by Sasha Levin 07 Sep '20

07 Sep '20

From: Rander Wang <rander.wang(a)intel.com> [ Upstream commit 13774d81f38538c5fa2924bdcdfa509155480fa6 ] In snd_hdac_device_init pm_runtime_set_active is called to increase child_count in parent device. But when it is failed to build connection with GPU for one case that integrated graphic gpu is disabled, snd_hdac_ext_bus_device_exit will be invoked to clean up a HD-audio extended codec base device. At this time the child_count of parent is not decreased, which makes parent device can't get suspended. This patch calls pm_runtime_set_suspended to decrease child_count in parent device in snd_hdac_device_exit to match with snd_hdac_device_init. pm_runtime_set_suspended can make sure that it will not decrease child_count if the device is already suspended. Signed-off-by: Rander Wang <rander.wang(a)intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Link: https://lore.kernel.org/r/20200902154218.1440441-1-kai.vehmanen@linux.intel… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/hda/hdac_device.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/hda/hdac_device.c b/sound/hda/hdac_device.c index 333220f0f8afc..3e9e9ac804f62 100644 --- a/sound/hda/hdac_device.c +++ b/sound/hda/hdac_device.c @@ -127,6 +127,8 @@ EXPORT_SYMBOL_GPL(snd_hdac_device_init); void snd_hdac_device_exit(struct hdac_device *codec) { pm_runtime_put_noidle(&codec->dev); + /* keep balance of runtime PM child_count in parent device */ + pm_runtime_set_suspended(&codec->dev); snd_hdac_bus_remove_device(codec->bus, codec); kfree(codec->vendor_name); kfree(codec->chip_name); -- 2.25.1

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[PATCH AUTOSEL 5.8 46/53] ALSA: hda: hdmi - add Rocketlake support
by Sasha Levin 07 Sep '20

07 Sep '20

From: Rander Wang <rander.wang(a)intel.com> [ Upstream commit f804a324a41a880c1ab43cc5145d8b3e5790430d ] Add Rocketlake HDMI codec support. Rocketlake shares the pin-to-port mapping table with Tigerlake. Signed-off-by: Rander Wang <rander.wang(a)intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Link: https://lore.kernel.org/r/20200902154207.1440393-1-kai.vehmanen@linux.intel… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/patch_hdmi.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 9442127d9b15d..43ae35338a383 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -4204,6 +4204,7 @@ HDA_CODEC_ENTRY(0x8086280c, "Cannonlake HDMI", patch_i915_glk_hdmi), HDA_CODEC_ENTRY(0x8086280d, "Geminilake HDMI", patch_i915_glk_hdmi), HDA_CODEC_ENTRY(0x8086280f, "Icelake HDMI", patch_i915_icl_hdmi), HDA_CODEC_ENTRY(0x80862812, "Tigerlake HDMI", patch_i915_tgl_hdmi), +HDA_CODEC_ENTRY(0x80862816, "Rocketlake HDMI", patch_i915_tgl_hdmi), HDA_CODEC_ENTRY(0x8086281a, "Jasperlake HDMI", patch_i915_icl_hdmi), HDA_CODEC_ENTRY(0x8086281b, "Elkhartlake HDMI", patch_i915_icl_hdmi), HDA_CODEC_ENTRY(0x80862880, "CedarTrail HDMI", patch_generic_hdmi), -- 2.25.1

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[PATCH AUTOSEL 5.8 12/53] ALSA: hda/tegra: Program WAKEEN register for Tegra
by Sasha Levin 07 Sep '20

07 Sep '20

From: Mohan Kumar <mkumard(a)nvidia.com> [ Upstream commit 23d63a31d9f44d7daeac0d1fb65c6a73c70e5216 ] The WAKEEN bits are used to indicate which bits in the STATESTS register may cause wake event during the codec state change request. Configure the WAKEEN register for the Tegra to detect the wake events. Signed-off-by: Mohan Kumar <mkumard(a)nvidia.com> Acked-by: Sameer Pujar <spujar(a)nvidia.com> Link: https://lore.kernel.org/r/20200825052415.20626-3-mkumard@nvidia.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/hda_tegra.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/pci/hda/hda_tegra.c b/sound/pci/hda/hda_tegra.c index 0cc5fad1af8a9..ae40ca3f29837 100644 --- a/sound/pci/hda/hda_tegra.c +++ b/sound/pci/hda/hda_tegra.c @@ -179,6 +179,10 @@ static int __maybe_unused hda_tegra_runtime_suspend(struct device *dev) struct hda_tegra *hda = container_of(chip, struct hda_tegra, chip); if (chip && chip->running) { + /* enable controller wake up event */ + azx_writew(chip, WAKEEN, azx_readw(chip, WAKEEN) | + STATESTS_INT_MASK); + azx_stop_chip(chip); azx_enter_link_reset(chip); } @@ -200,6 +204,9 @@ static int __maybe_unused hda_tegra_runtime_resume(struct device *dev) if (chip && chip->running) { hda_tegra_init(hda); azx_init_chip(chip, 1); + /* disable controller wake up event*/ + azx_writew(chip, WAKEEN, azx_readw(chip, WAKEEN) & + ~STATESTS_INT_MASK); } return 0; -- 2.25.1

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[PATCH AUTOSEL 5.8 11/53] ALSA: hda: Fix 2 channel swapping for Tegra
by Sasha Levin 07 Sep '20

07 Sep '20

From: Mohan Kumar <mkumard(a)nvidia.com> [ Upstream commit 216116eae43963c662eb84729507bad95214ca6b ] The Tegra HDA codec HW implementation has an issue related to not swapping the 2 channel Audio Sample Packet(ASP) channel mapping. Whatever the FL and FR mapping specified the left channel always comes out of left speaker and right channel on right speaker. So add condition to disallow the swapping of FL,FR during the playback. Signed-off-by: Mohan Kumar <mkumard(a)nvidia.com> Acked-by: Sameer Pujar <spujar(a)nvidia.com> Link: https://lore.kernel.org/r/20200825052415.20626-2-mkumard@nvidia.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/patch_hdmi.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index f0c6d2907e396..9442127d9b15d 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -3670,6 +3670,7 @@ static int tegra_hdmi_build_pcms(struct hda_codec *codec) static int patch_tegra_hdmi(struct hda_codec *codec) { + struct hdmi_spec *spec; int err; err = patch_generic_hdmi(codec); @@ -3677,6 +3678,10 @@ static int patch_tegra_hdmi(struct hda_codec *codec) return err; codec->patch_ops.build_pcms = tegra_hdmi_build_pcms; + spec = codec->spec; + spec->chmap.ops.chmap_cea_alloc_validate_get_type = + nvhdmi_chmap_cea_alloc_validate_get_type; + spec->chmap.ops.chmap_validate = nvhdmi_chmap_validate; return 0; } -- 2.25.1

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[PATCH 00/10] USB: new USB control message helper functions
by Greg Kroah-Hartman 07 Sep '20

07 Sep '20

In a recent discussion about a USB networking bug found by syzbot, and fixed by Himadri Pandya, the design of the existing usb_control_msg() call was brought up as not being the "nicest" thing to use by Dmitry Vyukov: https://lore.kernel.org/r/CACT4Y+YbDODLRFn8M5QcY4CazhpeCaunJnP_udXtAs0rYoAS… The function makes it hard to get right, in that it will return the number of bytes sent/received, but almost no one checks to see if a short read/write happens. With a malicious, or broken, USB device, this can cause drivers to act on data that they did not anticipate, and sometimes copy internal kernel data out to userspace. So let's fix this up by creating two new functions, usb_control_msg_send() and usb_control_msg_recv(). These functions either complete the full transation, or they return an error, a short send/recv is now an error. They also accept data off of the stack, saving individual drivers the pain of having to constantly allocate memory on their own for tiny messages, thereby saving overall kernel code space. The api also does not require a raw USB "pipe" to be sent to the function, as we know the direction, so just pass in the endpoint number instead, again making it easier on the USB driver author to use. This series first takes a helper function out of the sound core for verifying USB endpoints to be able to use internally, and then adds the new functions, converts over some internal USB code to use them, and then starts to clean up some drivers using these new functions, as an example of the savings that can happen by using these functions. Thanks to Dmitry and Himadri for the idea on how to do all of this. greg k-h Greg Kroah-Hartman (10): USB: move snd_usb_pipe_sanity_check into the USB core USB: add usb_control_msg_send() and usb_control_msg_recv() USB: core: message.c: use usb_control_msg_send() in a few places USB: core: hub.c: use usb_control_msg_send() in a few places USB: legousbtower: use usb_control_msg_recv() sound: usx2y: move to use usb_control_msg_send() sound: 6fire: move to use usb_control_msg_send() and usb_control_msg_recv() sound: line6: move to use usb_control_msg_send() and usb_control_msg_recv() sound: hiface: move to use usb_control_msg_send() Bluetooth: ath3k: use usb_control_msg_send() and usb_control_msg_recv() drivers/bluetooth/ath3k.c | 90 +++++------------ drivers/usb/core/hub.c | 128 +++++++++--------------- drivers/usb/core/message.c | 171 ++++++++++++++++++++++++++++---- drivers/usb/core/urb.c | 29 ++++-- drivers/usb/misc/legousbtower.c | 60 ++++------- include/linux/usb.h | 7 ++ sound/usb/6fire/firmware.c | 38 +++---- sound/usb/helper.c | 16 +-- sound/usb/helper.h | 1 - sound/usb/hiface/pcm.c | 14 ++- sound/usb/line6/driver.c | 69 +++++-------- sound/usb/line6/podhd.c | 17 ++-- sound/usb/line6/toneport.c | 8 +- sound/usb/mixer_scarlett_gen2.c | 2 +- sound/usb/quirks.c | 12 +-- sound/usb/usx2y/us122l.c | 42 ++------ 16 files changed, 345 insertions(+), 359 deletions(-) -- 2.28.0

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[PATCH] ASoC: rt5682: Have global name clock option for parent clk
by Akshu Agrawal 07 Sep '20

07 Sep '20

When adding parent clk(mclk) to wclk, this adds fallback option for the case where global clk name is used. Signed-off-by: Akshu Agrawal <akshu.agrawal(a)amd.com> --- sound/soc/codecs/rt5682.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index 93ebf0279b62..26928bc49bcb 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -2780,6 +2780,7 @@ static int rt5682_register_dai_clks(struct snd_soc_component *component) if (rt5682->mclk) { init.parent_data = &(struct clk_parent_data){ .fw_name = "mclk", + .name = __clk_get_name(rt5682->mclk), }; init.num_parents = 1; } -- 2.20.1

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[PATCH] [RFC] ASoC: core: Do not cleanup uninitialized dais on soc_pcm_open failure
by Cezary Rojewski 07 Sep '20

07 Sep '20

Introduce for_each_rtd_dais_rollback macro which behaves exactly like for_each_codec_dais_rollback and its cpu_dais equivalent but for all dais instead. Use newly added macro to fix soc_pcm_open error path and prevent uninitialized dais from being cleaned-up. Signed-off-by: Cezary Rojewski <cezary.rojewski(a)intel.com> --- include/sound/soc.h | 2 ++ sound/soc/soc-pcm.c | 2 +- 2 files changed, 3 insertions(+), 1 deletion(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index b94ebff12350..09a7d8409ade 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1193,6 +1193,8 @@ struct snd_soc_pcm_runtime { ((i) < (rtd)->num_cpus + (rtd)->num_codecs) && \ ((dai) = (rtd)->dais[i]); \ (i)++) +#define for_each_rtd_dais_rollback(rtd, i, dai) \ + for (; (--(i) >= 0) && ((dai) = (rtd)->dais[i]);) void snd_soc_close_delayed_work(struct snd_soc_pcm_runtime *rtd); diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 00ac1cbf6f88..4c9d4cd8cf0b 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -812,7 +812,7 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) return 0; config_err: - for_each_rtd_dais(rtd, i, dai) + for_each_rtd_dais_rollback(rtd, i, dai) snd_soc_dai_shutdown(dai, substream); snd_soc_link_shutdown(substream); -- 2.17.1

4 4

[PATCH 00/11] ALSA: Kill tasklet usage
by Takashi Iwai 07 Sep '20

07 Sep '20

Hi, here is a patch set to convert the tasklet usage in sound tree with either the threaded irq or the dedicated work. It's applied after the tasket API conversion series, found in topic/tasklet-convert branch of sound git tree (which will be included in the next pull request for 5.9-rc4). This contains only non-ASoC changes, the changes for ASoC will follow at next. Takashi === Takashi Iwai (11): ALSA: pcsp: Replace tasklet with work ALSA: timer: Replace tasklet with work ALSA: usb-audio: Replace tasklet with work ALSA: ua101: Replace tasklet with work ALSA: aloop: Replace tasklet with work ALSA: hdsp: Replace tasklet with work ALSA: hdspm: Replace tasklet with work ALSA: riptide: Replace tasklet with threaded irq ALSA: asihpi: Replace tasklet with threaded irq ALSA: firewire: Replace tasklet with work ALSA: mixart: Correct comment wrt obsoleted tasklet usage include/sound/timer.h | 8 +++--- sound/core/hrtimer.c | 2 +- sound/core/timer.c | 20 +++++++------- sound/drivers/aloop.c | 23 ++++++++-------- sound/drivers/pcsp/pcsp_lib.c | 12 ++++---- sound/firewire/amdtp-stream-trace.h | 2 +- sound/firewire/amdtp-stream.c | 25 +++++++++-------- sound/firewire/amdtp-stream.h | 2 +- sound/pci/asihpi/asihpi.c | 28 ++----------------- sound/pci/asihpi/hpioctl.c | 16 +++++++++-- sound/pci/mixart/mixart.h | 2 +- sound/pci/riptide/riptide.c | 20 ++++++++------ sound/pci/rme9652/hdsp.c | 55 ++++++++++++++++++------------------- sound/pci/rme9652/hdspm.c | 13 ++++----- sound/usb/midi.c | 13 +++++---- sound/usb/misc/ua101.c | 16 +++++------ 16 files changed, 122 insertions(+), 135 deletions(-) -- 2.16.4

2 13

[PATCH V2] ALSA: hda: fixup headset for ASUS GX502 laptop
by Luke D Jones 07 Sep '20

07 Sep '20

The GX502 requires a few steps to enable the headset i/o: pincfg, verbs to enable and unmute the amp used for headpone out, and a jacksense callback to toggle output via internal or jack using a verb. Signed-off-by: Luke D Jones <luke(a)ljones.dev> --- sound/pci/hda/patch_realtek.c | 65 +++++++++++++++++++++++++++++++++++ 1 file changed, 65 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 29f5878f0c50..16009297f771 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5975,6 +5975,40 @@ static void alc_fixup_disable_mic_vref(struct hda_codec *codec, snd_hda_codec_set_pin_target(codec, 0x19, PIN_VREFHIZ); } + +static void alc294_gx502_toggle_output(struct hda_codec *codec, + struct hda_jack_callback *cb) +{ + /* The Windows driver sets the codec up in a very different way where + * it appears to leave 0x10 = 0x8a20 set. For Linux we need to toggle it + */ + if (snd_hda_jack_detect_state(codec, 0x21) == HDA_JACK_PRESENT) + alc_write_coef_idx(codec, 0x10, 0x8a20); + else + alc_write_coef_idx(codec, 0x10, 0x0a20); +} + +static void alc294_fixup_gx502_hp(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + /* Pin 0x21: headphones/headset mic */ + if (!is_jack_detectable(codec, 0x21)) + return; + + switch (action) { + case HDA_FIXUP_ACT_PRE_PROBE: + snd_hda_jack_detect_enable_callback(codec, 0x21, + alc294_gx502_toggle_output); + break; + case HDA_FIXUP_ACT_INIT: + /* Make sure to start in a correct state, i.e. if + * headphones have been plugged in before powering up the system + */ + alc294_gx502_toggle_output(codec, NULL); + break; + } +} + static void alc285_fixup_hp_gpio_amp_init(struct hda_codec *codec, const struct hda_fixup *fix, int action) { @@ -6154,6 +6188,9 @@ enum { ALC285_FIXUP_THINKPAD_HEADSET_JACK, ALC294_FIXUP_ASUS_HPE, ALC294_FIXUP_ASUS_COEF_1B, + ALC294_FIXUP_ASUS_GX502_HP, + ALC294_FIXUP_ASUS_GX502_PINS, + ALC294_FIXUP_ASUS_GX502_VERBS, ALC285_FIXUP_HP_GPIO_LED, ALC285_FIXUP_HP_MUTE_LED, ALC236_FIXUP_HP_MUTE_LED, @@ -7307,6 +7344,33 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC294_FIXUP_ASUS_HEADSET_MIC }, + [ALC294_FIXUP_ASUS_GX502_PINS] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x19, 0x03a11050 }, /* front HP mic */ + { 0x1a, 0x01a11830 }, /* rear external mic */ + { 0x21, 0x03211020 }, /* front HP out */ + { } + }, + .chained = true, + .chain_id = ALC294_FIXUP_ASUS_GX502_VERBS + }, + [ALC294_FIXUP_ASUS_GX502_VERBS] = { + .type = HDA_FIXUP_VERBS, + .v.verbs = (const struct hda_verb[]) { + /* set 0x15 to HP-OUT ctrl */ + { 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0 }, + /* unmute the 0x15 amp */ + { 0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000 }, + { } + }, + .chained = true, + .chain_id = ALC294_FIXUP_ASUS_GX502_HP + }, + [ALC294_FIXUP_ASUS_GX502_HP] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc294_fixup_gx502_hp, + }, [ALC294_FIXUP_ASUS_COEF_1B] = { .type = HDA_FIXUP_VERBS, .v.verbs = (const struct hda_verb[]) { @@ -7588,6 +7652,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1ccd, "ASUS X555UB", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x1e11, "ASUS Zephyrus G15", ALC289_FIXUP_ASUS_GA502), SND_PCI_QUIRK(0x1043, 0x1f11, "ASUS Zephyrus G14", ALC289_FIXUP_ASUS_GA401), + SND_PCI_QUIRK(0x1043, 0x1881, "ASUS Zephyrus S/M", ALC294_FIXUP_ASUS_GX502_PINS), SND_PCI_QUIRK(0x1043, 0x3030, "ASUS ZN270IE", ALC256_FIXUP_ASUS_AIO_GPIO2), SND_PCI_QUIRK(0x1043, 0x831a, "ASUS P901", ALC269_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x1043, 0x834a, "ASUS S101", ALC269_FIXUP_STEREO_DMIC), -- 2.26.2

2 1

[PATCH] ALSA: hda: fixup headset for ASUS GX502 laptop
by Luke Jones 07 Sep '20

07 Sep '20

From: Luke D Jones <luke(a)ljones.dev> The GX502 requires a few steps to enable the headset i/o: pincfg, verbs to enable and unmute the amp used for headpone out, and a jacksense callback to toggle output via internal or jack using a verb. Signed-off-by: Luke Jones <luke(a)ljones.dev> --- sound/pci/hda/patch_realtek.c | 70 ++++++++++++++++++++++++++++++++++- 1 file changed, 69 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index c521a1f17096..d2052a6e3b13 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5424,7 +5424,7 @@ static void alc_fixup_headset_mode_alc255_no_hp_mic(struct hda_codec *codec, struct alc_spec *spec = codec->spec; spec->parse_flags |= HDA_PINCFG_HEADSET_MIC; alc255_set_default_jack_type(codec); - } + } else alc_fixup_headset_mode(codec, fix, action); } @@ -5993,6 +5993,41 @@ static void alc_fixup_disable_mic_vref(struct hda_codec *codec, snd_hda_codec_set_pin_target(codec, 0x19, PIN_VREFHIZ); } + +static void alc294_gx502_toggle_output(struct hda_codec *codec, + struct hda_jack_callback *cb) +{ + /* The Windows driver sets the codec up in a very different way where + * it appears to leave 0x10 = 0x8a20 set. For Linux we need to toggle it + */ + if (snd_hda_jack_detect_state(codec, 0x21) == HDA_JACK_PRESENT) { + alc_write_coef_idx(codec, 0x10, 0x8a20); + } else { + alc_write_coef_idx(codec, 0x10, 0x0a20); + } +} + +static void alc294_fixup_gx502_hp(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + /* Pin 0x21: headphones/headset mic */ + if (!is_jack_detectable(codec, 0x21)) + return; + + switch (action) { + case HDA_FIXUP_ACT_PRE_PROBE: + snd_hda_jack_detect_enable_callback(codec, 0x21, + alc294_gx502_toggle_output); + break; + case HDA_FIXUP_ACT_INIT: + /* Make sure to start in a correct state, i.e. if + * headphones have been plugged in before powering up the system + */ + alc294_gx502_toggle_output(codec, NULL); + break; + } +} + static void alc285_fixup_hp_gpio_amp_init(struct hda_codec *codec, const struct hda_fixup *fix, int action) { @@ -6172,6 +6207,9 @@ enum { ALC285_FIXUP_THINKPAD_X1_GEN7, ALC285_FIXUP_THINKPAD_HEADSET_JACK, ALC294_FIXUP_ASUS_HPE, + ALC294_FIXUP_ASUS_GX502_HP, + ALC294_FIXUP_ASUS_GX502_PINS, + ALC294_FIXUP_ASUS_GX502_VERBS, ALC294_FIXUP_ASUS_COEF_1B, ALC285_FIXUP_HP_GPIO_LED, ALC285_FIXUP_HP_MUTE_LED, @@ -7338,6 +7376,35 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC294_FIXUP_ASUS_HEADSET_MIC }, + [ALC294_FIXUP_ASUS_GX502_PINS] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x19, 0x03a11050 }, /* front HP mic */ + { 0x1a, 0x01a11830 }, //0x00a11030 }, /* rear external mic */ + { 0x21, 0x03211020 }, /* HP out */ + { } + }, + .chained = true, + .chain_id = ALC294_FIXUP_ASUS_GX502_VERBS + }, + [ALC294_FIXUP_ASUS_GX502_VERBS] = { + .type = HDA_FIXUP_VERBS, + .v.verbs = (const struct hda_verb[]) { + /* set 0x15 to HP-OUT ctrl */ + { 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0 }, + /* unmute the 0x15 amp */ + { 0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000 }, + /* set 0x0a input converter to digital */ + { 0x0a, 0x70d, 0x01 }, + { } + }, + .chained = true, + .chain_id = ALC294_FIXUP_ASUS_GX502_HP + }, + [ALC294_FIXUP_ASUS_GX502_HP] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc294_fixup_gx502_hp, + }, [ALC294_FIXUP_ASUS_COEF_1B] = { .type = HDA_FIXUP_VERBS, .v.verbs = (const struct hda_verb[]) { @@ -7711,6 +7778,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1ccd, "ASUS X555UB", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x1e11, "ASUS Zephyrus G15", ALC289_FIXUP_ASUS_GA502), SND_PCI_QUIRK(0x1043, 0x1f11, "ASUS Zephyrus G14", ALC289_FIXUP_ASUS_GA401), + SND_PCI_QUIRK(0x1043, 0x1881, "ASUS Zephyrus S/M", ALC294_FIXUP_ASUS_GX502_PINS), SND_PCI_QUIRK(0x1043, 0x3030, "ASUS ZN270IE", ALC256_FIXUP_ASUS_AIO_GPIO2), SND_PCI_QUIRK(0x1043, 0x831a, "ASUS P901", ALC269_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x1043, 0x834a, "ASUS S101", ALC269_FIXUP_STEREO_DMIC), -- 2.26.2

2 2

[PATCH] ALSA: hda: release resource when snd_hdac_device_init is failed
by Kai Vehmanen 07 Sep '20

07 Sep '20

From: Rander Wang <rander.wang(a)intel.com> When snd_hdac_device_init is failed, the codec is released by kfree immediately without releasing some resources. The vendor_name should be free if the memory is allocated and the runtime pm status should be restored, especially the runtime pm status of parent device. Signed-off-by: Rander Wang <rander.wang(a)intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> --- sound/hda/hdac_device.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/hda/hdac_device.c b/sound/hda/hdac_device.c index 3e9e9ac804f6..8d9d37225701 100644 --- a/sound/hda/hdac_device.c +++ b/sound/hda/hdac_device.c @@ -109,12 +109,16 @@ int snd_hdac_device_init(struct hdac_device *codec, struct hdac_bus *bus, codec->vendor_id & 0xffff); if (!codec->chip_name) { err = -ENOMEM; - goto error; + goto error_chip; } return 0; + error_chip: + kfree(codec->vendor_name); error: + pm_runtime_put_noidle(&codec->dev); + pm_runtime_set_suspended(&codec->dev); put_device(&codec->dev); return err; } -- 2.27.0

3 3

[RFC] soc_pcm_open: error path behavior change since v5.6
by Cezary Rojewski 07 Sep '20

07 Sep '20

Hello, Some time ago negative-tests found out that behavior of soc_pcm_open has changed, quite sure this might be a regression hence my email. Till v5.6 soc_pcm_open was invoking ::shutdown() for cpu_dai in error path only if ::startup() succeeded first (label: 'out'). After addition of commit: 5d9fa03e6c3514266fa94851ab1b6dd6e0595a13 ASoC: soc-pcm: tidyup soc_pcm_open() order this is no longer true. ::shutdown() can and will be invoked for given cpu_dai despite ::startup()'s failure. This complicated further since the merging of cpu & codec dais. The same applies to codec_dais: notice the usage of for_each_rtd_codec_dai_rollback macro instead of for_each_rtd_dais in error path in v5.6. Should dai's ::shutdown() be introducing some kind of state-check from now on? - similarly to how developers deal with some of the core pcm operations e.g.: ::prepare() (as it may get invoked multiple times in a row so check is there to prevent redundancy). Or, perhaps behavior change should be reverted with ::shutdown() routine again being called only after successful ::startup()? Czarek

3 5

[PATCH v2 00/20] Add Allwinner H3/H5/H6/A64 HDMI audio
by Clément Péron 05 Sep '20

05 Sep '20

Hi, Regarding the I2S LRCK polarity, Maxime Ripard test it and found that the LRCK is fine: https://lkml.org/lkml/2020/7/29/581. So the patch introduce this modification has been reverted. I have tested this on Allwinner H6 but this should be tested on older platform like sun4i family. Regards, Clement Change since v1: - rebase on next-20200828 - add revert LRCK polarity - remove all simple-audio-card,frame-inversion in dts - add Ondrej patches for Orange Pi board - Add arm64 defconfig patch Clément Péron (4): Revert "ASoC: sun4i-i2s: Fix the LRCK polarity" ASoC: sun4i-i2s: Fix sun8i volatile regs arm64: dts: allwinner: h6: Enable HDMI sound for Beelink GS1 arm64: defconfig: Enable Allwinner i2s driver Jernej Skrabec (3): ASoC: sun4i-i2s: Add support for H6 I2S dt-bindings: ASoC: sun4i-i2s: Add H6 compatible arm64: dts: allwinner: h6: Add HDMI audio node Marcus Cooper (10): ASoC: sun4i-i2s: Adjust LRCLK width ASoC: sun4i-i2s: Set sign extend sample ASoc: sun4i-i2s: Add 20 and 24 bit support arm: dts: sunxi: h3/h5: Add DAI node for HDMI arm: dts: sunxi: h3/h5: Add HDMI audio arm64: dts: allwinner: a64: Add DAI node for HDMI arm64: dts: allwinner: a64: Add HDMI audio arm: sun8i: h3: Add HDMI audio to Orange Pi 2 arm: sun8i: h3: Add HDMI audio to Beelink X2 arm64: dts: allwinner: a64: Add HDMI audio to Pine64 Ondrej Jirman (3): arm64: dts: allwinner: Enable HDMI audio on Orange Pi PC 2 ARM: dts: sun8i-h3: Enable HDMI audio on Orange Pi PC/One arm64: dts: sun50i-h6-orangepi-3: Enable HDMI audio .../sound/allwinner,sun4i-a10-i2s.yaml | 2 + arch/arm/boot/dts/sun8i-h3-beelink-x2.dts | 8 + arch/arm/boot/dts/sun8i-h3-orangepi-2.dts | 8 + arch/arm/boot/dts/sun8i-h3-orangepi-one.dts | 8 + arch/arm/boot/dts/sun8i-h3-orangepi-pc.dts | 8 + arch/arm/boot/dts/sunxi-h3-h5.dtsi | 32 ++ .../boot/dts/allwinner/sun50i-a64-pine64.dts | 8 + arch/arm64/boot/dts/allwinner/sun50i-a64.dtsi | 33 ++ .../dts/allwinner/sun50i-h5-orangepi-pc2.dts | 8 + .../dts/allwinner/sun50i-h6-beelink-gs1.dts | 8 + .../dts/allwinner/sun50i-h6-orangepi-3.dts | 8 + arch/arm64/boot/dts/allwinner/sun50i-h6.dtsi | 32 ++ arch/arm64/configs/defconfig | 1 + sound/soc/sunxi/sun4i-i2s.c | 285 ++++++++++++++++-- 14 files changed, 431 insertions(+), 18 deletions(-) -- 2.25.1

3 24

[PATCH 00/16] Add Allwinner H3/H5/H6/A64 HDMI audio
by Clément Péron 04 Sep '20

04 Sep '20

Hi, This a merge of serie: - Add Add H6 I2S support https://patchwork.kernel.org/cover/11497007/ - Add Allwinner H3/H5/A64 HDMI audio https://patchwork.kernel.org/cover/11510511/ I merge both serie because there is a similar issue regarding the I2S polarity. This need to be investigated under a scope and see if the comment is still true for Allwinner H6 and previous SoC. LibreElec team found that we have to introduce the "simple-audio-card,frame-inversion" property with recent mainline change. In order to make HDMI audio work. Maybe the I2S polarity is good or maybe the silicium has an issue but not present for HDMI I2S or not present in TDM mode... I cannot do it myself, so if someone want to to do it please feel free. Regarding the discussion we had here: https://patchwork.kernel.org/patch/11510521/ I didn't switch to generic hdmi card-name and used name like: sun8i-h3-hdmi, sun50i-h6-hdmi, etc.... Despite this wrong comment the rest of the serie introduce some fix that should be merged even if it's without the H6 support. Regards Clement Change since v1: - drop allwinner,playback-channels property - use coherent hdmi,card-name - indentation fix - collect tags Clément Péron (2): ASoC: sun4i-i2s: Fix sun8i volatile regs arm64: dts: allwinner: h6: Enable HDMI sound for Beelink GS1 Jernej Skrabec (3): ASoC: sun4i-i2s: Add support for H6 I2S dt-bindings: ASoC: sun4i-i2s: Add H6 compatible arm64: dts: allwinner: h6: Add HDMI audio node Marcus Cooper (11): ASoC: sun4i-i2s: Adjust LRCLK width ASoC: sun4i-i2s: Set sign extend sample ASoc: sun4i-i2s: Add 20 and 24 bit support ASoC: sun4i-i2s: Adjust regmap settings arm: dts: sunxi: h3/h5: Add DAI node for HDMI arm: dts: sunxi: h3/h5: Add HDMI audio arm64: dts: allwinner: a64: Add DAI node for HDMI arm64: dts: allwinner: a64: Add HDMI audio arm: sun8i: h3: Add HDMI audio to Orange Pi 2 arm: sun8i: h3: Add HDMI audio to Beelink X2 arm64: dts: allwinner: a64: Add HDMI audio to Pine64 .../sound/allwinner,sun4i-a10-i2s.yaml | 2 + arch/arm/boot/dts/sun8i-h3-beelink-x2.dts | 8 + arch/arm/boot/dts/sun8i-h3-orangepi-2.dts | 8 + arch/arm/boot/dts/sunxi-h3-h5.dtsi | 33 ++ .../boot/dts/allwinner/sun50i-a64-pine64.dts | 8 + arch/arm64/boot/dts/allwinner/sun50i-a64.dtsi | 35 +++ .../dts/allwinner/sun50i-h6-beelink-gs1.dts | 8 + arch/arm64/boot/dts/allwinner/sun50i-h6.dtsi | 33 ++ sound/soc/sunxi/sun4i-i2s.c | 281 +++++++++++++++++- 9 files changed, 410 insertions(+), 6 deletions(-) -- 2.25.1

6 39

[PATCH v3 0/5] Qualcomm's lpass-hdmi ASoC driver to support audio over dp port
by Srinivasa Rao Mandadapu 04 Sep '20

04 Sep '20

These patches are to support audio over DP port on Qualcomm's SC7180 LPASS Asoc. It includes machine driver, cpu driver, platform driver updates for HDMI path support, device tree documention, lpass variant structure optimization and configuration changes. These patches depends on the DP patch series https://patchwork.kernel.org/project/dri-devel/list/?series=332029 Changes Since v2: -- Audio buffer size(i.e. LPASS_PLATFORM_BUFFER_SIZE) in lpass-platform.c increased. V Sujith Kumar Reddy (5): ASoC: Add sc7180-lpass binding header hdmi define ASoC: dt-bindings: Add dt binding for lpass hdmi ASoC: qcom: Add support for lpass hdmi driver ASoC: qcom: Add support for audio over DP ASoC: qcom: Optimise lpass variant structure .../devicetree/bindings/sound/qcom,lpass-cpu.yaml | 51 +- include/dt-bindings/sound/sc7180-lpass.h | 1 + sound/soc/qcom/Kconfig | 5 + sound/soc/qcom/Makefile | 2 + sound/soc/qcom/lpass-apq8016.c | 25 +- sound/soc/qcom/lpass-cpu.c | 92 ++- sound/soc/qcom/lpass-hdmi.c | 685 +++++++++++++++++++++ sound/soc/qcom/lpass-hdmi.h | 129 ++++ sound/soc/qcom/lpass-ipq806x.c | 25 +- sound/soc/qcom/lpass-lpaif-reg.h | 51 +- sound/soc/qcom/lpass-platform.c | 287 +++++++-- sound/soc/qcom/lpass-sc7180.c | 147 ++++- sound/soc/qcom/lpass.h | 123 +++- 13 files changed, 1472 insertions(+), 151 deletions(-) create mode 100644 sound/soc/qcom/lpass-hdmi.c create mode 100644 sound/soc/qcom/lpass-hdmi.h -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

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[PATCH 0/3] ASoC: SOF: multi core support for 5.10
by Kai Vehmanen 04 Sep '20

04 Sep '20

This series extends the multi-core support in SOF. Capability to specify which core to use, on a per component basis, is added to topology. The topology load functionality in SOF is modified to power up/down host controlled cores based on the topology description. Guennadi Liakhovetski (2): ASoC: SOF: add a "core" parameter to widget loading functions ASoC: SOF: support topology components on secondary cores Ranjani Sridharan (1): ASoC: SOF: topology: fix core enable sequence include/uapi/sound/sof/tokens.h | 1 + sound/soc/sof/pm.c | 1 + sound/soc/sof/sof-audio.c | 25 ++++ sound/soc/sof/sof-audio.h | 5 + sound/soc/sof/sof-priv.h | 3 + sound/soc/sof/topology.c | 210 ++++++++++++++++++++++---------- 6 files changed, 184 insertions(+), 61 deletions(-) -- 2.27.0

2 4

[PATCH v2 0/3] ASoC: Add sdw stream operations to dailink ops.
by Bard Liao 04 Sep '20

04 Sep '20

Sdw stream operation APIs can be called once per stream. Move these operations to dailink ops. The linked series is "soundwire: Remove sdw stream operations from Intel soundwire dai". Pierre-Louis Bossart (3): ASoC: soc-dai: clarify return value for get_sdw_stream() ASOC: Intel: sof_sdw: add dailink .trigger callback ASOC: Intel: sof_sdw: add dailink .prepare and .hw_free callback include/sound/soc-dai.h | 3 +- sound/soc/intel/boards/sof_sdw.c | 81 ++++++++++++++++++++++++++++++++ 2 files changed, 83 insertions(+), 1 deletion(-) -- 2.17.1

2 4

[PATCH v2 0/9] soundwire: use FIELD_{GET|PREP} in subsystem
by Vinod Koul 04 Sep '20

04 Sep '20

Use the FIELD_{GET|PREP} in soundwire subsytem and remove the local SDW_REG_SHIFT(). This makes code IMO look much neater Tested this on db845c board Changes in v2: - fold in fixes provided by Bard and Pierre - Use GENMAS_ULL for addr - rebase to next and fold in new intel change Vinod Koul (9): soundwire: define and use addr bit masks soundwire: bus: use FIELD_GET() soundwire: slave: use SDW_DISCO_LINK_ID() soundwire: stream: use FIELD_{GET|PREP} soundwire: qcom : use FIELD_{GET|PREP} soundwire: cadence: use FIELD_{GET|PREP} soundwire: intel: use FIELD_{GET|PREP} soundwire: intel_init: use FIELD_{GET|PREP} soundwire: remove SDW_REG_SHIFT() drivers/soundwire/bus.c | 6 +-- drivers/soundwire/cadence_master.c | 61 +++++++++++-------------- drivers/soundwire/intel.c | 52 ++++++++------------- drivers/soundwire/intel_init.c | 2 +- drivers/soundwire/qcom.c | 22 +++------ drivers/soundwire/slave.c | 2 +- drivers/soundwire/stream.c | 13 ++---- include/linux/soundwire/sdw.h | 21 ++++++--- include/linux/soundwire/sdw_registers.h | 7 --- 9 files changed, 75 insertions(+), 111 deletions(-) -- 2.26.2

2 11

[PATCH 0/7] ASoC: soundwire: Move sdw stream operations to
by Bard Liao 04 Sep '20

04 Sep '20

sdw stream operation APIs can be called once per stream. dailink callbacks are good places to call these APIs. Pierre-Louis Bossart (7): ASoC: soc-dai: clarify return value for get_sdw_stream() soundwire: stream: fix NULL/IS_ERR confusion soundwire: intel: fix NULL/ERR_PTR confusion ASOC: Intel: sof_sdw: add dailink .trigger callback ASOC: Intel: sof_sdw: add dailink .prepare and .hw_free callback soundwire: intel: remove .trigger operation soundwire: intel: remove stream handling from .prepare and .hw_free drivers/soundwire/intel.c | 60 ++++------------------- drivers/soundwire/stream.c | 2 +- include/sound/soc-dai.h | 3 +- sound/soc/intel/boards/sof_sdw.c | 81 ++++++++++++++++++++++++++++++++ 4 files changed, 92 insertions(+), 54 deletions(-) -- 2.17.1

4 11

[PATCH v3 0/7] Add H6 I2S support
by Clément Péron 04 Sep '20

04 Sep '20

Hi, This is a sequel of Marcus Cooper serie[0], where remarks made by Maxime have been fixed. I have tested it on my Beelink GS1 board. Thanks, Clement 0: https://lore.kernel.org/patchwork/cover/1139949/ Changes since v2 (thanks Maxime): - Add details in commit log about sign extend sample - Only set FIFO regs as volatile in regmap - Missing a space (detected by checkpatch) Changes since v1: - Fix missing header in set sign extend sample Jernej Skrabec (3): dt-bindings: ASoC: sun4i-i2s: Add H6 compatible ASoC: sun4i-i2s: Add support for H6 I2S arm64: dts: sun50i-h6: Add HDMI audio to H6 DTSI Marcus Cooper (4): ASoC: sun4i-i2s: Adjust LRCLK width ASoC: sun4i-i2s: Set sign extend sample ASoc: sun4i-i2s: Add 20 and 24 bit support ASoC: sun4i-i2s: Adjust regmap settings .../sound/allwinner,sun4i-a10-i2s.yaml | 2 + arch/arm64/boot/dts/allwinner/sun50i-h6.dtsi | 31 ++ sound/soc/sunxi/sun4i-i2s.c | 265 +++++++++++++++++- 3 files changed, 296 insertions(+), 2 deletions(-) -- 2.20.1

6 28

[PATCH 1/3] ALSA: hda/realtek: Add quirk for System76 Lemur Pro (lemp9)
by Jeremy Soller 03 Sep '20

03 Sep '20

The System76 Lemur Pro (lemp9) requires the ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC quirk for the headset microphone to function. Signed-off-by: Jeremy Soller <jeremy(a)system76.com> Cc: alsa-devel(a)alsa-project.org --- diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index a1fa983d2a94..58d5ebb6e196 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7702,6 +7702,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1462, 0xb120, "MSI Cubi MS-B120", ALC283_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x1462, 0xb171, "Cubi N 8GL (MS-B171)", ALC283_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x1558, 0x1325, "System76 Darter Pro (darp5)", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1558, 0x1401, "System76 Lemur Pro (lemp9)", ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC), SND_PCI_QUIRK(0x1558, 0x8550, "System76 Gazelle (gaze14)", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x8551, "System76 Gazelle (gaze14)", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x8560, "System76 Gazelle (gaze14)", ALC269_FIXUP_HEADSET_MIC),

1 2

[PATCH 1/2] dt-bindings: sound: midas-audio: Correct parsing sound-dai phandles
by Krzysztof Kozlowski 03 Sep '20

03 Sep '20

The "sound-dai" property has cells therefore phandle-array should be used, even if it is just one phandle. This fixes dtbs_check warnings like: arch/arm/boot/dts/exynos4412-trats2.dt.yaml: sound: cpu:sound-dai:0:1: missing phandle tag in 0 arch/arm/boot/dts/exynos4412-trats2.dt.yaml: sound: cpu:sound-dai:0: [158, 0] is too long Signed-off-by: Krzysztof Kozlowski <krzk(a)kernel.org> --- .../devicetree/bindings/sound/samsung,midas-audio.yaml | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/samsung,midas-audio.yaml b/Documentation/devicetree/bindings/sound/samsung,midas-audio.yaml index 1c755de686f7..578928e67e5c 100644 --- a/Documentation/devicetree/bindings/sound/samsung,midas-audio.yaml +++ b/Documentation/devicetree/bindings/sound/samsung,midas-audio.yaml @@ -21,7 +21,8 @@ properties: type: object properties: sound-dai: - $ref: /schemas/types.yaml#/definitions/phandle + $ref: /schemas/types.yaml#/definitions/phandle-array + maxItems: 1 description: phandle to the I2S controller required: - sound-dai @@ -30,7 +31,8 @@ properties: type: object properties: sound-dai: - $ref: /schemas/types.yaml#/definitions/phandle + $ref: /schemas/types.yaml#/definitions/phandle-array + maxItems: 1 description: phandle to the WM1811 CODEC required: - sound-dai -- 2.17.1

4 13

[PATCH 1/2] control: Improve general control interface documentation.
by Tanjeff-N. Moos 03 Sep '20

03 Sep '20

Signed-off-by: Tanjeff-N. Moos <tanjeff(a)cccmz.de> --- src/control/control.c | 72 +++++++++++++++++++++++++++++++++++++------ 1 file changed, 63 insertions(+), 9 deletions(-) diff --git a/src/control/control.c b/src/control/control.c index 1bcf1ab2..497a5399 100644 --- a/src/control/control.c +++ b/src/control/control.c @@ -31,7 +31,7 @@ /*! \page control Control interface <P>Control interface is designed to access primitive controls. There is -also interface notifying about control and structure changes. +also an interface for notifying about control and structure changes. \section control_general_overview General overview @@ -43,20 +43,74 @@ are managed according to below model. Some element sets can be added to a sound card by drivers in kernel and userspace applications. - element - - An element can be identified by userspace applications. Each element has - own identical information. + - A control element might be a master volume control, for example, or a + read-only indicator, such as a sync status. An element has a type (e.g. + INTEGER or BOOLEAN) and - depending on the type - min/max values, a step + size, a set of possible values (for enums), etc. - member - - An element includes some members to have a value. The value of each member - can be changed by both of userspace applications and drivers in kernel. -Each element can be identified by two ways; the numerical number (numid), or the -combination of interface, device, subdevice, name, and index. + - An element includes one or more member(s) to have a value. For + example, a stereo volume control element has two members (for + left/right). The members share the same properties (e.g. both + volume controls have the same min/max values). The value of each + member can be changed by both of userspace applications and + drivers in kernel. -The type of element set is one of integer, integerr64, boolean, enumerators, + +\section identifying_elements Identifying the Elements + +Each element has the following identifying properties: + + - The numid (a numeric identifier, assigned when the sound card is + detected, constant while the sound card is kept connected) + + - The interface type (e.g. MIXER, CARD or PCM) + - The device + - The subdevice + - Its name + - Its index + +An element can be identified either by its numid or by the tuple +(interface type, device, subdevice, name, index). This tuple is always +the same (driver updates can change it, but in practice this is +rare). The numid can change on each boot. In case of an USB sound +card, the numid can also change when it is reconnected. + + +\section element_lists Element Lists + +An element list can be used to obtain a list of all elements of the +sound card. The list contains generic information (e.g. how many +elements the card has), and the identifying properties of the elements +(numid, card, name, ...). See #snd_ctl_elem_list_t to learn more about +element lists. + + +\section working_with_elements Working with Elements + +It is possible to obtain information about an element using the +snd_ctl_elem_info_*() functions. For enums, the allowed values can be +obtained, for integers, the min/max values can be obtained, and so +on. In addition, these functions can report the identifying +properties. E.g. when the element is addressed using its numid, the +functions complements the name, index, etc. + +To access the values of a control, use the snd_ctl_elem_value*() +functions. These allow to get and set the actual values or +settings. It is also possible to get and set the ID values (such as +the numid or the name). + + +\section element_sets Element Sets + +The type of element set is one of integer, integer64, boolean, enumerators, bytes and IEC958 structure. This indicates the type of value for each member in elements included in the element set. -When the value of member is changed, corresponding events are transferred to + +\section events Events + +When the value of a member is changed, corresponding events are transferred to userspace applications. The applications should subscribe any events in advance. \section tlv_blob Supplemental data for elements in an element set -- 2.17.1

3 4

[PATCH v2] ASoC: fsl_sai: Set SAI Channel Mode to Output Mode
by Shengjiu Wang 03 Sep '20

03 Sep '20

Transmit data pins will output zero when slots are masked or channels are disabled. In CHMOD TDM mode, transmit data pins are tri-stated when slots are masked or channels are disabled. When data pins are tri-stated, there is noise on some channels when FS clock value is high and data is read while fsclk is transitioning from high to low. Signed-off-by: Cosmin-Gabriel Samoila <cosmin.samoila(a)nxp.com> Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> Acked-by: Nicolin Chen <nicoleotsuka(a)gmail.com> --- changes in v2 - update note - add acked-by Nicolin sound/soc/fsl/fsl_sai.c | 10 ++++++++-- sound/soc/fsl/fsl_sai.h | 2 ++ 2 files changed, 10 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 38c7bcbb361d..b2d65e53dbc4 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -489,6 +489,10 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, val_cr4 |= FSL_SAI_CR4_FRSZ(slots); + /* Set to output mode to avoid tri-stated data pins */ + if (tx) + val_cr4 |= FSL_SAI_CR4_CHMOD; + /* * For SAI master mode, when Tx(Rx) sync with Rx(Tx) clock, Rx(Tx) will * generate bclk and frame clock for Tx(Rx), we should set RCR4(TCR4), @@ -497,7 +501,8 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, if (!sai->is_slave_mode && fsl_sai_dir_is_synced(sai, adir)) { regmap_update_bits(sai->regmap, FSL_SAI_xCR4(!tx, ofs), - FSL_SAI_CR4_SYWD_MASK | FSL_SAI_CR4_FRSZ_MASK, + FSL_SAI_CR4_SYWD_MASK | FSL_SAI_CR4_FRSZ_MASK | + FSL_SAI_CR4_CHMOD_MASK, val_cr4); regmap_update_bits(sai->regmap, FSL_SAI_xCR5(!tx, ofs), FSL_SAI_CR5_WNW_MASK | FSL_SAI_CR5_W0W_MASK | @@ -508,7 +513,8 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, FSL_SAI_CR3_TRCE_MASK, FSL_SAI_CR3_TRCE((1 << pins) - 1)); regmap_update_bits(sai->regmap, FSL_SAI_xCR4(tx, ofs), - FSL_SAI_CR4_SYWD_MASK | FSL_SAI_CR4_FRSZ_MASK, + FSL_SAI_CR4_SYWD_MASK | FSL_SAI_CR4_FRSZ_MASK | + FSL_SAI_CR4_CHMOD_MASK, val_cr4); regmap_update_bits(sai->regmap, FSL_SAI_xCR5(tx, ofs), FSL_SAI_CR5_WNW_MASK | FSL_SAI_CR5_W0W_MASK | diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h index 5f630be74853..736a437450c8 100644 --- a/sound/soc/fsl/fsl_sai.h +++ b/sound/soc/fsl/fsl_sai.h @@ -119,6 +119,8 @@ #define FSL_SAI_CR4_FRSZ_MASK (0x1f << 16) #define FSL_SAI_CR4_SYWD(x) (((x) - 1) << 8) #define FSL_SAI_CR4_SYWD_MASK (0x1f << 8) +#define FSL_SAI_CR4_CHMOD BIT(5) +#define FSL_SAI_CR4_CHMOD_MASK BIT(5) #define FSL_SAI_CR4_MF BIT(4) #define FSL_SAI_CR4_FSE BIT(3) #define FSL_SAI_CR4_FSP BIT(1) -- 2.27.0

2 1

[PATCH] ASoC: fsl_sai: Support multiple data channel enable bits
by Shengjiu Wang 03 Sep '20

03 Sep '20

One data channel is one data line. From imx7ulp, the SAI IP is enhanced to support multiple data channels. If there is only two channels input and slots is 2, then enable one data channel is enough for data transfer. So enable the TCE/RCE and transmit/receive mask register according to the input channels and slots configuration. Move the data channel enablement from startup() to hw_params(). Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/fsl_sai.c | 30 ++++++++++++------------------ sound/soc/fsl/fsl_sai.h | 2 +- 2 files changed, 13 insertions(+), 19 deletions(-) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 62c5fdb678fc..38c7bcbb361d 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -443,6 +443,7 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, u32 slots = (channels == 1) ? 2 : channels; u32 slot_width = word_width; int adir = tx ? RX : TX; + u32 pins; int ret; if (sai->slots) @@ -451,6 +452,8 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, if (sai->slot_width) slot_width = sai->slot_width; + pins = DIV_ROUND_UP(channels, slots); + if (!sai->is_slave_mode) { if (sai->bclk_ratio) ret = fsl_sai_set_bclk(cpu_dai, tx, @@ -501,13 +504,17 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, FSL_SAI_CR5_FBT_MASK, val_cr5); } + regmap_update_bits(sai->regmap, FSL_SAI_xCR3(tx, ofs), + FSL_SAI_CR3_TRCE_MASK, + FSL_SAI_CR3_TRCE((1 << pins) - 1)); regmap_update_bits(sai->regmap, FSL_SAI_xCR4(tx, ofs), FSL_SAI_CR4_SYWD_MASK | FSL_SAI_CR4_FRSZ_MASK, val_cr4); regmap_update_bits(sai->regmap, FSL_SAI_xCR5(tx, ofs), FSL_SAI_CR5_WNW_MASK | FSL_SAI_CR5_W0W_MASK | FSL_SAI_CR5_FBT_MASK, val_cr5); - regmap_write(sai->regmap, FSL_SAI_xMR(tx), ~0UL - ((1 << channels) - 1)); + regmap_write(sai->regmap, FSL_SAI_xMR(tx), + ~0UL - ((1 << min(channels, slots)) - 1)); return 0; } @@ -517,6 +524,10 @@ static int fsl_sai_hw_free(struct snd_pcm_substream *substream, { struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai); bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + unsigned int ofs = sai->soc_data->reg_offset; + + regmap_update_bits(sai->regmap, FSL_SAI_xCR3(tx, ofs), + FSL_SAI_CR3_TRCE_MASK, 0); if (!sai->is_slave_mode && sai->mclk_streams & BIT(substream->stream)) { @@ -651,14 +662,9 @@ static int fsl_sai_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai) { struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai); - unsigned int ofs = sai->soc_data->reg_offset; bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; int ret; - regmap_update_bits(sai->regmap, FSL_SAI_xCR3(tx, ofs), - FSL_SAI_CR3_TRCE_MASK, - FSL_SAI_CR3_TRCE); - /* * EDMA controller needs period size to be a multiple of * tx/rx maxburst @@ -675,17 +681,6 @@ static int fsl_sai_startup(struct snd_pcm_substream *substream, return ret; } -static void fsl_sai_shutdown(struct snd_pcm_substream *substream, - struct snd_soc_dai *cpu_dai) -{ - struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai); - unsigned int ofs = sai->soc_data->reg_offset; - bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; - - regmap_update_bits(sai->regmap, FSL_SAI_xCR3(tx, ofs), - FSL_SAI_CR3_TRCE_MASK, 0); -} - static const struct snd_soc_dai_ops fsl_sai_pcm_dai_ops = { .set_bclk_ratio = fsl_sai_set_dai_bclk_ratio, .set_sysclk = fsl_sai_set_dai_sysclk, @@ -695,7 +690,6 @@ static const struct snd_soc_dai_ops fsl_sai_pcm_dai_ops = { .hw_free = fsl_sai_hw_free, .trigger = fsl_sai_trigger, .startup = fsl_sai_startup, - .shutdown = fsl_sai_shutdown, }; static int fsl_sai_dai_probe(struct snd_soc_dai *cpu_dai) diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h index 6aba7d28f5f3..5f630be74853 100644 --- a/sound/soc/fsl/fsl_sai.h +++ b/sound/soc/fsl/fsl_sai.h @@ -109,7 +109,7 @@ #define FSL_SAI_CR2_DIV_MASK 0xff /* SAI Transmit and Receive Configuration 3 Register */ -#define FSL_SAI_CR3_TRCE BIT(16) +#define FSL_SAI_CR3_TRCE(x) ((x) << 16) #define FSL_SAI_CR3_TRCE_MASK GENMASK(23, 16) #define FSL_SAI_CR3_WDFL(x) (x) #define FSL_SAI_CR3_WDFL_MASK 0x1f -- 2.27.0

3 2

[PATCH v2 00/10] soundwire: intel: add multi-link support
by Bard Liao 03 Sep '20

03 Sep '20

This series enables multi-link support for Intel platforms. Changes in v2: - Remove the "soundwire: intel: Only call sdw stream APIs for the first cpu_dai" patch form this series. We will move the sounwdire stream operations to machine driver in another series. - Update the commit message and title of "soundwire: intel: add error log for clock-stop invalid configs" Pierre-Louis Bossart (10): soundwire: intel: disable shim wake on suspend soundwire: intel: ignore software command retries soundwire: intel: add multi-link support soundwire: intel: add missing support for all clock stop modes soundwire: bus: update multi-link definition with hw sync details soundwire: intel: add multi-link hw_synchronization information soundwire: stream: enable hw_sync as needed by hardware soundwire: intel: add error log for clock-stop invalid configs soundwire: intel: pass link_mask information to each master soundwire: intel: don't manage link power individually drivers/soundwire/intel.c | 264 ++++++++++++++++++++++++++++----- drivers/soundwire/intel.h | 2 + drivers/soundwire/intel_init.c | 1 + drivers/soundwire/stream.c | 15 +- include/linux/soundwire/sdw.h | 6 + 5 files changed, 243 insertions(+), 45 deletions(-) -- 2.17.1

2 11

[PATCH v4 0/3] ASoC: soundwire: fix port_ready[] dynamic allocation
by Bard Liao 03 Sep '20

03 Sep '20

The existing code allocates memory for the total number of ports. This only works if the ports are contiguous, but will break if e.g. a Devices uses port0, 1, and 14. The port_ready[] array would contain 3 elements, which would lead to an out-of-bounds access. Conversely in other cases, the wrong port index would be used leading to timeouts on prepare. This can be fixed by allocating for the worst-case of 15 ports (DP0..DP14). In addition since the number is now fixed, we can use an array instead of a dynamic allocation. Changes in v4: - Reorder the patches order to allow bisect. Changes in v3: - Add ASoC tag in the cover letter title. - Edit the title and commit message of the third patch for better understanding. Changes in v2: - Split patches into sdw and asoc patches. Please note that "soundwire: fix port_ready[] dynamic allocation in mipi_disco" and "ASoC: codecs: fix port_ready[] dynamic allocation in ASoC codecs" should be merged at the same time. Pierre-Louis Bossart (3): ASoC: codecs: soundwire: remove port_ready[] usage from codecs. soundwire: add definition for maximum number of ports soundwire: fix port_ready[] dynamic allocation in mipi_disco drivers/soundwire/mipi_disco.c | 18 +----------------- drivers/soundwire/slave.c | 4 ++++ include/linux/soundwire/sdw.h | 5 +++-- sound/soc/codecs/max98373-sdw.c | 15 +-------------- sound/soc/codecs/rt1308-sdw.c | 14 +------------- sound/soc/codecs/rt5682-sdw.c | 15 +-------------- sound/soc/codecs/rt700-sdw.c | 15 +-------------- sound/soc/codecs/rt711-sdw.c | 15 +-------------- sound/soc/codecs/rt715-sdw.c | 33 +-------------------------------- 9 files changed, 14 insertions(+), 120 deletions(-) -- 2.17.1

3 5

a saner API for allocating DMA addressable pages
by Christoph Hellwig 03 Sep '20

03 Sep '20

Hi all, this series replaced the DMA_ATTR_NON_CONSISTENT flag to dma_alloc_attrs with a separate new dma_alloc_pages API, which is available on all platforms. In addition to cleaning up the convoluted code path, this ensures that other drivers that have asked for better support for non-coherent DMA to pages with incurring bounce buffering over can finally be properly supported. I'm still a little unsure about the API naming, as alloc_pages sort of implies a struct page return value, but we return a kernel virtual address. The other alternative would be to name the API dma_alloc_noncoherent, but the whole non-coherent naming seems to put people off. As a follow up I plan to move the implementation of the DMA_ATTR_NO_KERNEL_MAPPING flag over to this framework as well, given that is also is a fundamentally non coherent allocation. The replacement for that flag would then return a struct page, as it is allowed to actually return pages without a kernel mapping as the name suggested (although most of the time they will actually have a kernel mapping..) In addition to the conversions of the existing non-coherent DMA users the last three patches also convert the DMA coherent allocations in the NVMe driver to use this new framework through a dmapool addition. This was both to give me a good testing vehicle, but also because it should speed up the NVMe driver on platforms with non-coherent DMA nicely, without a downside on platforms with cache coherent DMA. A git tree is available here: git://git.infradead.org/users/hch/misc.git dma_alloc_pages Gitweb: http://git.infradead.org/users/hch/misc.git/shortlog/refs/heads/dma_alloc_p… Diffstat: Documentation/core-api/dma-api.rst | 92 ++---- Documentation/core-api/dma-attributes.rst | 8 Documentation/userspace-api/media/v4l/buffer.rst | 17 - Documentation/userspace-api/media/v4l/vidioc-reqbufs.rst | 1 arch/alpha/kernel/pci_iommu.c | 2 arch/arm/include/asm/dma-direct.h | 4 arch/arm/mm/dma-mapping-nommu.c | 2 arch/arm/mm/dma-mapping.c | 4 arch/ia64/Kconfig | 3 arch/ia64/hp/common/sba_iommu.c | 2 arch/ia64/kernel/dma-mapping.c | 14 arch/ia64/mm/init.c | 3 arch/mips/Kconfig | 1 arch/mips/bmips/dma.c | 4 arch/mips/cavium-octeon/dma-octeon.c | 4 arch/mips/include/asm/dma-direct.h | 4 arch/mips/include/asm/jazzdma.h | 2 arch/mips/jazz/jazzdma.c | 102 +------ arch/mips/loongson2ef/fuloong-2e/dma.c | 4 arch/mips/loongson2ef/lemote-2f/dma.c | 4 arch/mips/loongson64/dma.c | 4 arch/mips/mm/dma-noncoherent.c | 48 +-- arch/mips/pci/pci-ar2315.c | 4 arch/mips/pci/pci-xtalk-bridge.c | 4 arch/mips/sgi-ip32/ip32-dma.c | 4 arch/parisc/Kconfig | 1 arch/parisc/kernel/pci-dma.c | 6 arch/powerpc/include/asm/dma-direct.h | 4 arch/powerpc/kernel/dma-iommu.c | 2 arch/powerpc/platforms/ps3/system-bus.c | 4 arch/powerpc/platforms/pseries/vio.c | 2 arch/s390/pci/pci_dma.c | 2 arch/x86/kernel/amd_gart_64.c | 8 drivers/gpu/drm/exynos/exynos_drm_gem.c | 2 drivers/gpu/drm/nouveau/nvkm/subdev/instmem/gk20a.c | 3 drivers/iommu/dma-iommu.c | 2 drivers/iommu/intel/iommu.c | 6 drivers/media/common/videobuf2/videobuf2-core.c | 36 -- drivers/media/common/videobuf2/videobuf2-dma-contig.c | 19 - drivers/media/common/videobuf2/videobuf2-dma-sg.c | 3 drivers/media/common/videobuf2/videobuf2-v4l2.c | 12 drivers/net/ethernet/amd/au1000_eth.c | 15 - drivers/net/ethernet/i825xx/lasi_82596.c | 36 +- drivers/net/ethernet/i825xx/lib82596.c | 148 +++++----- drivers/net/ethernet/i825xx/sni_82596.c | 23 - drivers/net/ethernet/seeq/sgiseeq.c | 24 - drivers/nvme/host/pci.c | 79 ++--- drivers/parisc/ccio-dma.c | 2 drivers/parisc/sba_iommu.c | 2 drivers/scsi/53c700.c | 120 ++++---- drivers/scsi/53c700.h | 9 drivers/scsi/sgiwd93.c | 14 drivers/xen/swiotlb-xen.c | 2 include/linux/dma-direct.h | 55 ++- include/linux/dma-mapping.h | 32 +- include/linux/dma-noncoherent.h | 21 - include/linux/dmapool.h | 23 + include/linux/gfp.h | 6 include/media/videobuf2-core.h | 3 include/uapi/linux/videodev2.h | 2 kernel/dma/Kconfig | 9 kernel/dma/Makefile | 1 kernel/dma/coherent.c | 17 + kernel/dma/direct.c | 112 +++++-- kernel/dma/mapping.c | 104 ++----- kernel/dma/ops_helpers.c | 86 ++++++ kernel/dma/pool.c | 2 kernel/dma/swiotlb.c | 4 kernel/dma/virt.c | 2 mm/dmapool.c | 211 +++++++++------ sound/mips/hal2.c | 58 +--- 71 files changed, 872 insertions(+), 803 deletions(-)

8 76

[PATCH v2] soundwire: fix double free of dangling pointer
by trix＠redhat.com 03 Sep '20

03 Sep '20

From: Tom Rix <trix(a)redhat.com> clang static analysis flags this problem stream.c:844:9: warning: Use of memory after it is freed kfree(bus->defer_msg.msg->buf); ^~~~~~~~~~~~~~~~~~~~~~~ This happens in an error handler cleaning up memory allocated for elements in a list. list_for_each_entry(m_rt, &stream->master_list, stream_node) { bus = m_rt->bus; kfree(bus->defer_msg.msg->buf); kfree(bus->defer_msg.msg); } And is triggered when the call to sdw_bank_switch() fails. There are a two problems. First, when sdw_bank_switch() fails, though it frees memory it does not clear bus's reference 'defer_msg.msg' to that memory. The second problem is the freeing msg->buf. In some cases msg will be NULL so this will dereference a null pointer. Need to check before freeing. Fixes: 99b8a5d608a6 ("soundwire: Add bank switch routine") Signed-off-by: Tom Rix <trix(a)redhat.com> Reviewed-by: Nick Desaulniers <ndesaulniers(a)google.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- v2 : change title, was 'soundwire: fix error handling' --- drivers/soundwire/stream.c | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) diff --git a/drivers/soundwire/stream.c b/drivers/soundwire/stream.c index 37290a799023..6e36deb505b1 100644 --- a/drivers/soundwire/stream.c +++ b/drivers/soundwire/stream.c @@ -717,6 +717,7 @@ static int sdw_bank_switch(struct sdw_bus *bus, int m_rt_count) kfree(wbuf); error_1: kfree(wr_msg); + bus->defer_msg.msg = NULL; return ret; } @@ -840,9 +841,10 @@ static int do_bank_switch(struct sdw_stream_runtime *stream) error: list_for_each_entry(m_rt, &stream->master_list, stream_node) { bus = m_rt->bus; - - kfree(bus->defer_msg.msg->buf); - kfree(bus->defer_msg.msg); + if (bus->defer_msg.msg) { + kfree(bus->defer_msg.msg->buf); + kfree(bus->defer_msg.msg); + } } msg_unlock: -- 2.18.1

3 2

[PATCH] ALSA: hda/realtek - Improved routing for Thinkpad X1 7th/8th Gen
by Takashi Iwai 03 Sep '20

03 Sep '20

There've been quite a few regression reports about the lowered volume (reduced to ca 65% from the previous level) on Lenovo Thinkpad X1 after the commit d2cd795c4ece ("ALSA: hda - fixup for the bass speaker on Lenovo Carbon X1 7th gen"). Although the commit itself does the right thing from HD-audio POV in order to have a volume control for bass speakers, it seems that the machine has some secret recipe under the hood. Through experiments, Benjamin Poirier found out that the following routing gives the best result: * DAC1 (NID 0x02) -> Speaker pin (NID 0x14) * DAC2 (NID 0x03) -> Shared by both Bass Speaker pin (NID 0x17) & Headphone pin (0x21) * DAC3 (NID 0x06) -> Unused DAC1 seems to have some equalizer internally applied, and you'd get again the output in a bad quality if you connect this to the headphone pin. Hence the headphone is connected to DAC2, which is now shared with the bass speaker pin. DAC3 has no volume amp, hence it's not connected at all. For achieving the routing above, this patch introduced a couple of workarounds: * The connection list of bass speaker pin (NID 0x17) is reduced not to include DAC3 (NID 0x06) * Pass preferred_pairs array to specify the fixed connection Here, both workarounds are needed because the generic parser prefers the individual DAC assignment over others. When the routing above is applied, the generic parser creates the two volume controls "Front" and "Bass Speaker". Since we have only two DACs for three output pins, those are not fully controlling each output individually, and it would confuse PulseAudio. For avoiding the pitfall, in this patch, we rename those volume controls to some unique ones ("DAC1" and "DAC2"). Then PulseAudio ignore them and concentrate only on the still good-working "Master" volume control. If a user still wants to control each DAC volume, they can still change manually via "DAC1" and "DAC2" volume controls. Fixes: d2cd795c4ece ("ALSA: hda - fixup for the bass speaker on Lenovo Carbon X1 7th gen") Reported-by: Benjamin Poirier <benjamin.poirier(a)gmail.com> Reviewed-by: Jaroslav Kysela <perex(a)perex.cz> Tested-by: Benjamin Poirier <benjamin.poirier(a)gmail.com> Cc: <stable(a)vger.kernel.org> BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207407#c10 BugLink: https://gist.github.com/hamidzr/dd81e429dc86f4327ded7a2030e7d7d9#gistcommen… BugLink: https://gist.github.com/hamidzr/dd81e429dc86f4327ded7a2030e7d7d9#gistcommen… Link: https://lore/kernel.org/r/20200829112746.3118-1-benjamin.poirier@gmail.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 42 +++++++++++++++++++++++++++++++++++++++++- 1 file changed, 41 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 2ef8b080d84b..c521a1f17096 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5868,6 +5868,39 @@ static void alc275_fixup_gpio4_off(struct hda_codec *codec, } } +/* Quirk for Thinkpad X1 7th and 8th Gen + * The following fixed routing needed + * DAC1 (NID 0x02) -> Speaker (NID 0x14); some eq applied secretly + * DAC2 (NID 0x03) -> Bass (NID 0x17) & Headphone (NID 0x21); sharing a DAC + * DAC3 (NID 0x06) -> Unused, due to the lack of volume amp + */ +static void alc285_fixup_thinkpad_x1_gen7(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + static const hda_nid_t conn[] = { 0x02, 0x03 }; /* exclude 0x06 */ + static const hda_nid_t preferred_pairs[] = { + 0x14, 0x02, 0x17, 0x03, 0x21, 0x03, 0 + }; + struct alc_spec *spec = codec->spec; + + switch (action) { + case HDA_FIXUP_ACT_PRE_PROBE: + snd_hda_override_conn_list(codec, 0x17, ARRAY_SIZE(conn), conn); + spec->gen.preferred_dacs = preferred_pairs; + break; + case HDA_FIXUP_ACT_BUILD: + /* The generic parser creates somewhat unintuitive volume ctls + * with the fixed routing above, and the shared DAC2 may be + * confusing for PA. + * Rename those to unique names so that PA doesn't touch them + * and use only Master volume. + */ + rename_ctl(codec, "Front Playback Volume", "DAC1 Playback Volume"); + rename_ctl(codec, "Bass Speaker Playback Volume", "DAC2 Playback Volume"); + break; + } +} + static void alc233_alc662_fixup_lenovo_dual_codecs(struct hda_codec *codec, const struct hda_fixup *fix, int action) @@ -6136,6 +6169,7 @@ enum { ALC289_FIXUP_DUAL_SPK, ALC294_FIXUP_SPK2_TO_DAC1, ALC294_FIXUP_ASUS_DUAL_SPK, + ALC285_FIXUP_THINKPAD_X1_GEN7, ALC285_FIXUP_THINKPAD_HEADSET_JACK, ALC294_FIXUP_ASUS_HPE, ALC294_FIXUP_ASUS_COEF_1B, @@ -7281,11 +7315,17 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC294_FIXUP_SPK2_TO_DAC1 }, + [ALC285_FIXUP_THINKPAD_X1_GEN7] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc285_fixup_thinkpad_x1_gen7, + .chained = true, + .chain_id = ALC269_FIXUP_THINKPAD_ACPI + }, [ALC285_FIXUP_THINKPAD_HEADSET_JACK] = { .type = HDA_FIXUP_FUNC, .v.func = alc_fixup_headset_jack, .chained = true, - .chain_id = ALC285_FIXUP_SPEAKER2_TO_DAC1 + .chain_id = ALC285_FIXUP_THINKPAD_X1_GEN7 }, [ALC294_FIXUP_ASUS_HPE] = { .type = HDA_FIXUP_VERBS, -- 2.16.4

1 0

[PATCH v4 00/10] topology: decode: Various fixes
by Piotr Maziarz 03 Sep '20

03 Sep '20

This series fixes various problems with topology decoding mechanism. Some of the problems were critical like improper memory management or infinite loops that were causing undefined behaviour or program crashes, while other resulted in losing some data during conversion. Bugs found while testing with Intel SST topologies. Signed-off-by: Piotr Maziarz <piotrx.maziarz(a)linux.intel.com> Reviewed-by: Cezary Rojewski <cezary.rojewski(a)intel.com> Reviewed-by: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Changelog: v2: -Divide into more patches, critical fixes are in separate patches now -More specific descriptions -Fix a typo UML to UCM -Add error reporting in topology: decode: fix channel map memory allocation -Remove goto again in topology: Make buffer for saving dynamic size for better readability v3: -No functional changes -Changed UCM to more descriptive standard ALSA configuration files -removed Gerrit's Change-Id -Added missing signed-offs v4: -Resend, added Reviewed-by tags Piotr Maziarz (10): topology: decode: Fix channel map memory allocation topology: decode: Fix infinite loop in decoding enum control topology: decode: Remove decoding values for enum control topology: decode: Add enum control texts as separate element topology: decode: Fix printing texts section topology: decode: Change declaration of enum decoding function topology: decode: Fix decoding PCM formats and rates topology: decode: Print sig_bits field in PCM capabilities section topology: decode: Add DAI name printing topology: Make buffer for saving dynamic size src/topology/ctl.c | 51 ++++++++++++++++++++++------------------------- src/topology/dapm.c | 3 +-- src/topology/pcm.c | 11 +++++++--- src/topology/save.c | 34 ++++++++++++++++++++++++++----- src/topology/text.c | 2 +- src/topology/tplg_local.h | 2 +- 6 files changed, 64 insertions(+), 39 deletions(-) -- 2.7.4

2 13

[PATCH 00/18] ALSA: cleanups for sparse/cppcheck warnings
by Pierre-Louis Bossart 03 Sep '20

03 Sep '20

Various cleanups for ALSA core, compressed, hda, usb, ac97 and Digigram cards. Pierre-Louis Bossart (18): ALSA: core: pcm: simplify locking for timers ALSA: core: memalloc: fix fallthrough position ALSA: core: pcm_memory: dereference pointer after NULL checks ALSA: core: timer: remove redundant assignment ALSA: core: timer: clarify operator precedence ALSA: compress_offload: dereference after checking for NULL pointer ALSA: compress_offload: remove redundant initialization ALSA: core: init: use DECLARE_COMPLETION_ONSTACK() macro ALSA: aoa: i2sbus: use DECLARE_COMPLETION_ONSTACK() macro ALSA: hda: auto_parser: remove shadowed variable declaration ALSA: hda: (cosmetic) align function parameters ALSA: usb: scarless_gen2: fix endianness issue ALSA: ac97: (cosmetic) align argument names ALSA: atmel: ac97: clarify operator precedence ALSA: rawmidi: (cosmetic) align function parameters ALSA: vx: vx_core: clarify operator precedence ALSA: vx: vx_pcm: remove redundant assignment ALSA: vx: vx_pcm: remove redundant assignment sound/ac97/ac97_core.h | 2 +- sound/aoa/soundbus/i2sbus/pcm.c | 3 +-- sound/atmel/ac97c.c | 22 +++++++++++----------- sound/core/compress_offload.c | 5 +++-- sound/core/init.c | 3 +-- sound/core/memalloc.c | 2 +- sound/core/pcm.c | 8 +++++--- sound/core/pcm_memory.c | 3 ++- sound/core/rawmidi.c | 2 +- sound/core/timer.c | 6 +++--- sound/drivers/vx/vx_core.c | 4 ++-- sound/drivers/vx/vx_pcm.c | 2 -- sound/pci/hda/hda_auto_parser.c | 2 +- sound/pci/hda/hda_jack.h | 2 +- sound/pci/hda/hda_local.h | 8 ++++---- sound/usb/mixer_scarlett_gen2.c | 2 +- 16 files changed, 38 insertions(+), 38 deletions(-) -- 2.25.1

3 21

[PATCH] ALSA: hda/realtek - Add control fixup for Lenovo Thinkpad X1 Carbon 7th
by Benjamin Poirier 03 Sep '20

03 Sep '20

As a result of commit d2cd795c4ece ("ALSA: hda - fixup for the bass speaker on Lenovo Carbon X1 7th gen"), the sound output level on my machine, an X1 Carbon 7th gen, was reduced to ~65% of its previous level when playing certain sounds. [1] Internally, this laptop model has three outputs (PCM-OUT1, connection 0x02; PCM-OUT2, connection 0x03; SP-OUT PCM, connection 0x06) which can be routed to two sets of stereo speakers: Front (tweeters, node 0x14) and Bass (woofers, node 0x17, aka Rear in some contexts) and one headphone output (node 0x21). The tweeters are noticeably less powerful than the woofers. [2] Before commit d2cd795c4ece, the bass speakers were connected to SP-OUT PCM. SP-OUT PCM is meant for s/pdif output and does not have volume control. This connection made volume control commonly ineffective (using the Master slider in alsa or pulseaudio apparently had little effect or alternated between mute or max with nothing in between). commit d2cd795c4ece added quirk ALC285_FIXUP_SPEAKER2_TO_DAC1 which resulted in assigning both sets of speakers to PCM-OUT1, bringing the two sets of speakers under one effective volume control but also lowering the output volume noticeably. Fix this by connecting PCM-OUT1 to Front speakers and PCM-OUT2 to Rear speakers. The max output volume is restored to what it was before commit d2cd795c4ece. This is done by setting the connection of node 0x17 to 0x03. However, when we do this, the HDA auto config automatically changes the connection of node 0x21 to 0x02. This output, meant for the front speakers, has some "secret" equalizer which changes the output volume according to the level of what's being played, after some delay[3]. This is undesirable with headphones. Therefore, this patch manually limits the connection of node 0x21 to 0x03. Even though there are three speaker/line outputs (nodes 0x14, 0x17, 0x21), there are only two amps for volume control (nodes 0x02, 0x03). When setting node 0x17 to connection 0x03 and node 0x21 to connection 0x03, amp 0x03 is shared by the Woofer and Headphone outputs. Therefore, they cannot have independent volume control. This patch uses the Speaker control element to regulate nodes 0x02 (acts on tweeters) and 0x03 (acts on woofers and headphones). A virtual control element is created for the Headphone mixer which mirrors the Speaker mixer. These changes are only for volume controls; there are 3 hardware mute controls. The current fixup chain applied to this machine is ALC285_FIXUP_THINKPAD_HEADSET_JACK ALC285_FIXUP_SPEAKER2_TO_DAC1 ALC269_FIXUP_THINKPAD_ACPI ALC269_FIXUP_SKU_IGNORE The fixup added by this patch is an alternative to ALC285_FIXUP_SPEAKER2_TO_DAC1 which must be selected explicitly using the module parameter snd_sof_intel_hda_common.hda_model=alc285-tpx1-dual-speakers when using the new SOF driver architecture, or snd_hda_intel.model=alc285-tpx1-dual-speakers when using the snd_hda_intel driver in legacy mode (see the snd_intel_dspcfg.dsp_driver module parameter). In order to be able to apply ALC285_FIXUP_THINKPAD_HEADSET_JACK without changing current behavior, a new variant of that fixup is introduced. When using the model added by this patch, the fixup chain is: ALC285_FIXUP_TPX1_DUAL_SPEAKERS ALC285_FIXUP_THINKPAD_HEADSET_JACK_ACPI ALC269_FIXUP_THINKPAD_ACPI ALC269_FIXUP_SKU_IGNORE [1] https://gist.github.com/hamidzr/dd81e429dc86f4327ded7a2030e7d7d9#gistcommen… [2] https://bugzilla.kernel.org/show_bug.cgi?id=207407#c10 [3] https://gist.github.com/hamidzr/dd81e429dc86f4327ded7a2030e7d7d9#gistcommen… Fixes: d2cd795c4ece ("ALSA: hda - fixup for the bass speaker on Lenovo Carbon X1 7th gen") Link: https://lore.kernel.org/alsa-devel/20200210025249.GA2700@f3/ Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Kailang Yang <kailang(a)realtek.com> Tested-by: Even Brenden <evenbrenden(a)gmail.com> Tested-by: Vincent Bernat <vincent(a)bernat.ch> Signed-off-by: Benjamin Poirier <benjamin.poirier(a)gmail.com> --- This is an updated version of the patchset (now a single patch) to fix the low output volume on the Lenovo ThinkPad X1 Carbon 7th gen. With this patch, the volume output with the Linux driver matches the one with the Windows driver. Volume control is effective and headphone output is unaffected. Multiple users of this machine have reported a low output volume [1] when using a kernel with commit d2cd795c4ece ("ALSA: hda - fixup for the bass speaker on Lenovo Carbon X1 7th gen"). Many have tried these patches or a similar workaround using `hda-verb` and prefer this configuration[2]. [1] https://gist.github.com/hamidzr/dd81e429dc86f4327ded7a2030e7d7d9#gistcommen… https://gist.github.com/hamidzr/dd81e429dc86f4327ded7a2030e7d7d9#gistcommen… https://gist.github.com/hamidzr/dd81e429dc86f4327ded7a2030e7d7d9#gistcommen… https://gist.github.com/hamidzr/dd81e429dc86f4327ded7a2030e7d7d9#gistcommen… https://gist.github.com/hamidzr/dd81e429dc86f4327ded7a2030e7d7d9#gistcommen… https://gist.github.com/hamidzr/dd81e429dc86f4327ded7a2030e7d7d9#gistcommen… https://gist.github.com/hamidzr/dd81e429dc86f4327ded7a2030e7d7d9#gistcommen… ... [2] https://gist.github.com/hamidzr/dd81e429dc86f4327ded7a2030e7d7d9#gistcommen… https://gist.github.com/hamidzr/dd81e429dc86f4327ded7a2030e7d7d9#gistcommen… https://gist.github.com/hamidzr/dd81e429dc86f4327ded7a2030e7d7d9#gistcommen… https://gist.github.com/hamidzr/dd81e429dc86f4327ded7a2030e7d7d9#gistcommen… ... Changes: v1 == https://lore.kernel.org/alsa-devel/20200211055651.4405-1-benjamin.poirier@g… v2 == https://lore.kernel.org/alsa-devel/20200703080005.8942-1-benjamin.poirier@g… * patch 1 Shortened log * patch 2 Change log to replace experimental observations by information from diagram posted on kernel.org bugzilla. Select exactly the desired connection for node 0x17 (bass speakers) instead of reusing alc295_fixup_disable_dac3(). Control node 0x21 (headphones) connection to avoid the volume "wobble" effect described in the log. Include SET_CONNECT_SEL command in fixup since it is not done by the hda core. Rename one control to reflect its new function and to use as a condition in the ucm config. v3 == * Remove patch 1 which was merged separately as commit 9774dc218bb6 ("ALSA: hda/realtek - Fix Lenovo Thinkpad X1 Carbon 7th quirk subdevice id") * Make fixup activation manual using "alc285-tpx1-dual-speakers" model instead of automatic based on codec id. Suggested by Jaroslav. * Control both tweeter and woofer volume from "Speaker Playback Volume" control. This way a separate ucm change is not needed, unlike in v2 of the patchset. Suggested by Jaroslav. sound/pci/hda/patch_realtek.c | 355 +++++++++++++++++++++++++++++++++- 1 file changed, 351 insertions(+), 4 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index a1fa983d2a94..8012c4aaf282 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5821,6 +5821,338 @@ static void alc285_fixup_speaker2_to_dac1(struct hda_codec *codec, } } +static struct snd_kcontrol *find_ctl_verbose(struct hda_codec *codec, + const char *name) +{ + struct snd_kcontrol *kctl; + + kctl = snd_hda_find_mixer_ctl(codec, name); + if (!kctl) + codec_warn(codec, "Did not find control \"%s\"\n", name); + return kctl; +} + +struct tpx1_dual_speaker { + struct hda_codec *codec; + struct snd_kcontrol underlying, hp_vol; +}; + +static int tpx1_dual_speaker_vol_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct tpx1_dual_speaker *speaker_priv = snd_kcontrol_chip(kcontrol); + + return speaker_priv->underlying.info(&speaker_priv->underlying, uinfo); +} + +static int tpx1_dual_speaker_vol_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tpx1_dual_speaker *speaker_priv = snd_kcontrol_chip(kcontrol); + + return speaker_priv->underlying.get(&speaker_priv->underlying, + ucontrol); +} + +static int tpx1_dual_speaker_vol_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tpx1_dual_speaker *speaker_priv = snd_kcontrol_chip(kcontrol); + int err; + + /* Control tweeter volume */ + err = speaker_priv->underlying.put(&speaker_priv->underlying, + ucontrol); + if (err < 0) + return err; + + /* Control woofer volume (shared with headphone) */ + err = speaker_priv->hp_vol.put(&speaker_priv->hp_vol, ucontrol); + if (err < 0) + return err; + + snd_ctl_notify(speaker_priv->codec->card, SNDRV_CTL_EVENT_MASK_VALUE, + &speaker_priv->hp_vol.id); + return err; +} + +static int tpx1_dual_speaker_vol_tlv(struct snd_kcontrol *kcontrol, + int op_flag, unsigned int size, + unsigned int __user *tlv) +{ + struct tpx1_dual_speaker *speaker_priv = snd_kcontrol_chip(kcontrol); + + return speaker_priv->underlying.tlv.c(&speaker_priv->underlying, + op_flag, size, tlv); +} + +static void tpx1_dual_speaker_vol_free(struct snd_kcontrol *kcontrol) +{ + struct tpx1_dual_speaker *speaker_priv = snd_kcontrol_chip(kcontrol); + + if (speaker_priv->underlying.private_free) + speaker_priv->underlying.private_free( + &speaker_priv->underlying); + kfree(speaker_priv); +} + +static int tpx1_dual_override_speaker_vol(struct hda_codec *codec, + struct snd_kcontrol *speaker_vol, + struct snd_kcontrol *hp_vol) +{ + struct tpx1_dual_speaker *speaker_priv; + + speaker_priv = kmalloc(sizeof(struct tpx1_dual_speaker), GFP_KERNEL); + if (!speaker_priv) + return -ENOMEM; + speaker_priv->codec = codec; + memcpy(&speaker_priv->underlying, speaker_vol, + sizeof(struct snd_kcontrol)); + memcpy(&speaker_priv->hp_vol, hp_vol, sizeof(struct snd_kcontrol)); + + speaker_vol->info = &tpx1_dual_speaker_vol_info; + speaker_vol->get = &tpx1_dual_speaker_vol_get; + speaker_vol->put = &tpx1_dual_speaker_vol_put; + speaker_vol->tlv.c = &tpx1_dual_speaker_vol_tlv; + speaker_vol->private_data = speaker_priv; + speaker_vol->private_free = &tpx1_dual_speaker_vol_free; + + return 0; +} + +struct tpx1_dual_hp { + struct hda_codec *codec; + struct snd_kcontrol underlying, *speaker_vol; +}; + +static int tpx1_dual_hp_vol_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct tpx1_dual_hp *hp_priv = snd_kcontrol_chip(kcontrol); + + return hp_priv->underlying.info(&hp_priv->underlying, uinfo); +} + +static int tpx1_dual_hp_vol_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tpx1_dual_hp *hp_priv = snd_kcontrol_chip(kcontrol); + + return hp_priv->speaker_vol->get(hp_priv->speaker_vol, ucontrol); +} + +static int tpx1_dual_hp_vol_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tpx1_dual_hp *hp_priv = snd_kcontrol_chip(kcontrol); + int err; + + err = hp_priv->speaker_vol->put(hp_priv->speaker_vol, ucontrol); + if (err < 0) + return err; + snd_ctl_notify(hp_priv->codec->card, SNDRV_CTL_EVENT_MASK_VALUE, + &hp_priv->speaker_vol->id); + + return 0; +} + +static int tpx1_dual_hp_vol_tlv(struct snd_kcontrol *kcontrol, + int op_flag, unsigned int size, + unsigned int __user *tlv) +{ + struct tpx1_dual_hp *hp_priv = snd_kcontrol_chip(kcontrol); + + return hp_priv->underlying.tlv.c(&hp_priv->underlying, op_flag, size, + tlv); +} + +static void tpx1_dual_hp_vol_free(struct snd_kcontrol *kcontrol) +{ + struct tpx1_dual_hp *hp_priv = snd_kcontrol_chip(kcontrol); + + if (hp_priv->underlying.private_free) + hp_priv->underlying.private_free(&hp_priv->underlying); + kfree(hp_priv); +} + +static int tpx1_dual_override_hp_vol(struct hda_codec *codec, + struct snd_kcontrol *speaker_vol, + struct snd_kcontrol *hp_vol) +{ + struct tpx1_dual_hp *hp_priv; + + hp_priv = kmalloc(sizeof(struct tpx1_dual_hp), GFP_KERNEL); + if (!hp_priv) + return -ENOMEM; + hp_priv->codec = codec; + memcpy(&hp_priv->underlying, hp_vol, sizeof(struct snd_kcontrol)); + hp_priv->speaker_vol = speaker_vol; + + hp_vol->info = &tpx1_dual_hp_vol_info; + hp_vol->get = &tpx1_dual_hp_vol_get; + hp_vol->put = &tpx1_dual_hp_vol_put; + hp_vol->tlv.c = &tpx1_dual_hp_vol_tlv; + hp_vol->private_data = hp_priv; + hp_vol->private_free = &tpx1_dual_hp_vol_free; + + return 0; +} + +/* We cannot use snd_hda_add_vmaster() because it expects the controls to be + * HDA (see init_follower_0dB()). This is not the case here. + */ +static int tpx1_dual_create_main_volume_ctl(struct hda_codec *codec) +{ + struct snd_kcontrol *main_vol, *speaker_vol; + struct snd_ctl_elem_value *ucontrol = NULL; + struct hda_gen_spec *spec = codec->spec; + struct snd_ctl_elem_info *uinfo; + int ch, err; + + speaker_vol = find_ctl_verbose(codec, "Speaker Playback Volume"); + if (!speaker_vol) + return -ENOENT; + + main_vol = snd_ctl_make_virtual_master("Master Playback Volume", + spec->vmaster_tlv); + if (!main_vol) + return -ENOMEM; + err = snd_hda_ctl_add(codec, 0, main_vol); + if (err < 0) + return err; + err = snd_ctl_add_follower(main_vol, speaker_vol); + if (err < 0) + return err; + + /* Initialize values */ + uinfo = kmalloc(sizeof(struct snd_ctl_elem_info), GFP_KERNEL); + if (!uinfo) + return -ENOMEM; + + uinfo->id = speaker_vol->id; + err = speaker_vol->info(speaker_vol, uinfo); + if (err) + goto out; + + ucontrol = kmalloc(sizeof(struct snd_ctl_elem_value), GFP_KERNEL); + if (!ucontrol) { + err = -ENOMEM; + goto out; + } + + ucontrol->id = speaker_vol->id; + for (ch = 0; ch < uinfo->count; ch++) { + ucontrol->value.integer.value[ch] = + uinfo->value.integer.max; + } + err = speaker_vol->put(speaker_vol, ucontrol); + if (err < 0) + goto out; + else + err = 0; + +out: + kfree(ucontrol); + kfree(uinfo); + return err; +} + +static void alc285_fixup_tpx1_dual_speakers(struct hda_codec *codec, + const struct hda_fixup *fix, + int action) +{ + struct alc_spec *spec = codec->spec; + + if (action == HDA_FIXUP_ACT_PRE_PROBE) { + static const hda_nid_t conn[] = { 0x03 }; + + /* For NID 0x17 (bass speakers), the connection list is {0x02, + * 0x03, 0x06}. Disable SP-OUT PCM (0x06) selection since it + * has no volume control, disable PCM1 (0x02) selection since + * it is for front speakers. This leaves PCM2 (0x03). + */ + snd_hda_override_conn_list(codec, 0x17, ARRAY_SIZE(conn), + conn); + /* For NID 0x21 (headphone out), the connection list is {0x02, + * 0x03}. Disable LOUT1 (0x02) selection since its volume + * fluctuates according to input level. This leaves LOUT2 + * (0x03). + */ + snd_hda_override_conn_list(codec, 0x21, ARRAY_SIZE(conn), + conn); + + /* "Master" controls will be created manually after + * subordinate controls have been changed in the + * HDA_FIXUP_ACT_BUILD phase. + */ + spec->gen.suppress_vmaster = 1; + } else if (action == HDA_FIXUP_ACT_INIT) { + /* Because the overridden connection lists contain only a + * single node, __parse_nid_path() does not label the output + * as "multi". This leads snd_hda_activate_path() to skip the + * AC_VERB_SET_CONNECT_SEL even though it might be needed. Do + * it here instead. + * Note that when doing AC_VERB_SET_CONNECT_SEL, the + * connection is specified by index instead of nid. + */ + snd_hda_codec_write(codec, 0x17, 0, AC_VERB_SET_CONNECT_SEL, + 0x1); + snd_hda_codec_write(codec, 0x21, 0, AC_VERB_SET_CONNECT_SEL, + 0x1); + } else if (action == HDA_FIXUP_ACT_BUILD) { + struct snd_kcontrol *speaker_vol, *hp_vol; + int err; + + speaker_vol = find_ctl_verbose(codec, "Speaker Playback Volume"); + if (!speaker_vol) + return; + + hp_vol = find_ctl_verbose(codec, "Headphone Playback Volume"); + if (!hp_vol) + return; + + err = tpx1_dual_override_speaker_vol(codec, speaker_vol, hp_vol); + if (err) { + codec_warn(codec, "Failed to override speaker volume control, err %d\n", + err); + return; + } + + err = tpx1_dual_override_hp_vol(codec, speaker_vol, hp_vol); + if (err) { + codec_warn(codec, "Failed to override headphone volume control, err %d\n", + err); + return; + } + + err = tpx1_dual_create_main_volume_ctl(codec); + if (err) { + codec_warn(codec, "Failed to create main volume control, err %d\n", + err); + return; + } + + err = __snd_hda_add_vmaster( + codec, "Master Playback Switch", NULL, + ((const char * const[]){ + "Speaker", "Bass Speaker", "Headphone", NULL + }), "Playback Switch", true, + &spec->gen.vmaster_mute.sw_kctl); + if (err) { + codec_warn(codec, "Failed to create main mute control, err %d\n", + err); + return; + } + + if (spec->gen.vmaster_mute.hook) { + snd_hda_add_vmaster_hook(codec, + &spec->gen.vmaster_mute, + spec->gen.vmaster_mute_enum); + snd_hda_sync_vmaster_hook(&spec->gen.vmaster_mute); + } + } +} + /* Hook to update amp GPIO4 for automute */ static void alc280_hp_gpio4_automute_hook(struct hda_codec *codec, struct hda_jack_callback *jack) @@ -6083,6 +6415,7 @@ enum { ALC225_FIXUP_DISABLE_MIC_VREF, ALC225_FIXUP_DELL1_MIC_NO_PRESENCE, ALC295_FIXUP_DISABLE_DAC3, + ALC285_FIXUP_TPX1_DUAL_SPEAKERS, ALC285_FIXUP_SPEAKER2_TO_DAC1, ALC280_FIXUP_HP_HEADSET_MIC, ALC221_FIXUP_HP_FRONT_MIC, @@ -6135,7 +6468,8 @@ enum { ALC289_FIXUP_DUAL_SPK, ALC294_FIXUP_SPK2_TO_DAC1, ALC294_FIXUP_ASUS_DUAL_SPK, - ALC285_FIXUP_THINKPAD_HEADSET_JACK, + ALC285_FIXUP_THINKPAD_HEADSET_JACK_SPK, + ALC285_FIXUP_THINKPAD_HEADSET_JACK_ACPI, ALC294_FIXUP_ASUS_HPE, ALC294_FIXUP_ASUS_COEF_1B, ALC285_FIXUP_HP_GPIO_LED, @@ -6905,6 +7239,12 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc295_fixup_disable_dac3, }, + [ALC285_FIXUP_TPX1_DUAL_SPEAKERS] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc285_fixup_tpx1_dual_speakers, + .chained = true, + .chain_id = ALC285_FIXUP_THINKPAD_HEADSET_JACK_ACPI + }, [ALC285_FIXUP_SPEAKER2_TO_DAC1] = { .type = HDA_FIXUP_FUNC, .v.func = alc285_fixup_speaker2_to_dac1, @@ -7280,12 +7620,18 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC294_FIXUP_SPK2_TO_DAC1 }, - [ALC285_FIXUP_THINKPAD_HEADSET_JACK] = { + [ALC285_FIXUP_THINKPAD_HEADSET_JACK_SPK] = { .type = HDA_FIXUP_FUNC, .v.func = alc_fixup_headset_jack, .chained = true, .chain_id = ALC285_FIXUP_SPEAKER2_TO_DAC1 }, + [ALC285_FIXUP_THINKPAD_HEADSET_JACK_ACPI] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc_fixup_headset_jack, + .chained = true, + .chain_id = ALC269_FIXUP_THINKPAD_ACPI + }, [ALC294_FIXUP_ASUS_HPE] = { .type = HDA_FIXUP_VERBS, .v.verbs = (const struct hda_verb[]) { @@ -7741,8 +8087,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x224c, "Thinkpad", ALC298_FIXUP_TPT470_DOCK), SND_PCI_QUIRK(0x17aa, 0x224d, "Thinkpad", ALC298_FIXUP_TPT470_DOCK), SND_PCI_QUIRK(0x17aa, 0x225d, "Thinkpad T480", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), - SND_PCI_QUIRK(0x17aa, 0x2292, "Thinkpad X1 Carbon 7th", ALC285_FIXUP_THINKPAD_HEADSET_JACK), - SND_PCI_QUIRK(0x17aa, 0x22be, "Thinkpad X1 Carbon 8th", ALC285_FIXUP_THINKPAD_HEADSET_JACK), + SND_PCI_QUIRK(0x17aa, 0x2292, "Thinkpad X1 Carbon 7th", ALC285_FIXUP_THINKPAD_HEADSET_JACK_SPK), + SND_PCI_QUIRK(0x17aa, 0x22be, "Thinkpad X1 Carbon 8th", ALC285_FIXUP_THINKPAD_HEADSET_JACK_SPK), SND_PCI_QUIRK(0x17aa, 0x30bb, "ThinkCentre AIO", ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY), SND_PCI_QUIRK(0x17aa, 0x30e2, "ThinkCentre AIO", ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY), SND_PCI_QUIRK(0x17aa, 0x310c, "ThinkCentre Station", ALC294_FIXUP_LENOVO_MIC_LOCATION), @@ -7934,6 +8280,7 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {.id = ALC255_FIXUP_DELL_SPK_NOISE, .name = "dell-spk-noise"}, {.id = ALC225_FIXUP_DELL1_MIC_NO_PRESENCE, .name = "alc225-dell1"}, {.id = ALC295_FIXUP_DISABLE_DAC3, .name = "alc295-disable-dac3"}, + {.id = ALC285_FIXUP_TPX1_DUAL_SPEAKERS, .name = "alc285-tpx1-dual-speakers"}, {.id = ALC285_FIXUP_SPEAKER2_TO_DAC1, .name = "alc285-speaker2-to-dac1"}, {.id = ALC280_FIXUP_HP_HEADSET_MIC, .name = "alc280-hp-headset"}, {.id = ALC221_FIXUP_HP_FRONT_MIC, .name = "alc221-hp-mic"}, -- 2.28.0

3 6

[PATCH] ASoC: fsl_sai: Set SAI Channel Mode to Output Mode
by Shengjiu Wang 03 Sep '20

03 Sep '20

Transmit data pins will output zero when slots are masked or channels are disabled. In CHMOD TDM mode, transmit data pins are tri-stated when slots are masked or channels are disabled. When data pins are tri-stated, there is noise on some channels when FS clock value is high and data is read while fsclk is transitioning from high to low. Signed-off-by: Cosmin-Gabriel Samoila <cosmin.samoila(a)nxp.com> Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/fsl_sai.c | 12 ++++++++++-- sound/soc/fsl/fsl_sai.h | 2 ++ 2 files changed, 12 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 62c5fdb678fc..33b194a5c1dc 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -486,6 +486,12 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, val_cr4 |= FSL_SAI_CR4_FRSZ(slots); + /* Output Mode - data pins transmit 0 when slots are masked + * or channels are disabled + */ + if (tx) + val_cr4 |= FSL_SAI_CR4_CHMOD; + /* * For SAI master mode, when Tx(Rx) sync with Rx(Tx) clock, Rx(Tx) will * generate bclk and frame clock for Tx(Rx), we should set RCR4(TCR4), @@ -494,7 +500,8 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, if (!sai->is_slave_mode && fsl_sai_dir_is_synced(sai, adir)) { regmap_update_bits(sai->regmap, FSL_SAI_xCR4(!tx, ofs), - FSL_SAI_CR4_SYWD_MASK | FSL_SAI_CR4_FRSZ_MASK, + FSL_SAI_CR4_SYWD_MASK | FSL_SAI_CR4_FRSZ_MASK | + FSL_SAI_CR4_CHMOD_MASK, val_cr4); regmap_update_bits(sai->regmap, FSL_SAI_xCR5(!tx, ofs), FSL_SAI_CR5_WNW_MASK | FSL_SAI_CR5_W0W_MASK | @@ -502,7 +509,8 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, } regmap_update_bits(sai->regmap, FSL_SAI_xCR4(tx, ofs), - FSL_SAI_CR4_SYWD_MASK | FSL_SAI_CR4_FRSZ_MASK, + FSL_SAI_CR4_SYWD_MASK | FSL_SAI_CR4_FRSZ_MASK | + FSL_SAI_CR4_CHMOD_MASK, val_cr4); regmap_update_bits(sai->regmap, FSL_SAI_xCR5(tx, ofs), FSL_SAI_CR5_WNW_MASK | FSL_SAI_CR5_W0W_MASK | diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h index 6aba7d28f5f3..19cd4e1bbff9 100644 --- a/sound/soc/fsl/fsl_sai.h +++ b/sound/soc/fsl/fsl_sai.h @@ -119,6 +119,8 @@ #define FSL_SAI_CR4_FRSZ_MASK (0x1f << 16) #define FSL_SAI_CR4_SYWD(x) (((x) - 1) << 8) #define FSL_SAI_CR4_SYWD_MASK (0x1f << 8) +#define FSL_SAI_CR4_CHMOD BIT(5) +#define FSL_SAI_CR4_CHMOD_MASK BIT(5) #define FSL_SAI_CR4_MF BIT(4) #define FSL_SAI_CR4_FSE BIT(3) #define FSL_SAI_CR4_FSP BIT(1) -- 2.27.0

3 2

[PATCH] ASoC: Intel: bytcr_rt5640: Add quirk for MPMAN Converter9 2-in-1
by Hans de Goede 02 Sep '20

02 Sep '20

The MPMAN Converter9 2-in-1 almost fully works with out default settings. The only problem is that it has only 1 speaker so any sounds only playing on the right channel get lost. Add a quirk for this model using the default settings + MONO_SPEAKER. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- sound/soc/intel/boards/bytcr_rt5640.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 479992f4e97a..fc202747ba83 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -591,6 +591,16 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { BYT_RT5640_SSP0_AIF1 | BYT_RT5640_MCLK_EN), }, + { /* MPMAN Converter 9, similar hw as the I.T.Works TW891 2-in-1 */ + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "MPMAN"), + DMI_MATCH(DMI_PRODUCT_NAME, "Converter9"), + }, + .driver_data = (void *)(BYTCR_INPUT_DEFAULTS | + BYT_RT5640_MONO_SPEAKER | + BYT_RT5640_SSP0_AIF1 | + BYT_RT5640_MCLK_EN), + }, { /* MPMAN MPWIN895CL */ .matches = { -- 2.28.0

3 2

[PATCH] ALSA: hda: use consistent HDAudio spelling in comments/docs
by Kai Vehmanen 02 Sep '20

02 Sep '20

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> We use HDaudio and HDAudio, pick one to make searches easier. No functionality change Also fix timestamping typo in documentation. Reported-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> 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: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> --- Documentation/sound/designs/timestamping.rst | 2 +- sound/hda/intel-dsp-config.c | 10 +++++----- sound/x86/Kconfig | 2 +- 3 files changed, 7 insertions(+), 7 deletions(-) diff --git a/Documentation/sound/designs/timestamping.rst b/Documentation/sound/designs/timestamping.rst index 2b0fff503415..7c7ecf5dbc4b 100644 --- a/Documentation/sound/designs/timestamping.rst +++ b/Documentation/sound/designs/timestamping.rst @@ -143,7 +143,7 @@ timestamp shows when the information is put together by the driver before returning from the ``STATUS`` and ``STATUS_EXT`` ioctl. in most cases this driver_timestamp will be identical to the regular system tstamp. -Examples of typestamping with HDaudio: +Examples of timestamping with HDAudio: 1. DMA timestamp, no compensation for DMA+analog delay :: diff --git a/sound/hda/intel-dsp-config.c b/sound/hda/intel-dsp-config.c index 99aec7349167..1c5114dedda9 100644 --- a/sound/hda/intel-dsp-config.c +++ b/sound/hda/intel-dsp-config.c @@ -54,7 +54,7 @@ static const struct config_entry config_table[] = { #endif /* * Apollolake (Broxton-P) - * the legacy HDaudio driver is used except on Up Squared (SOF) and + * the legacy HDAudio driver is used except on Up Squared (SOF) and * Chromebooks (SST) */ #if IS_ENABLED(CONFIG_SND_SOC_SOF_APOLLOLAKE) @@ -89,7 +89,7 @@ static const struct config_entry config_table[] = { }, #endif /* - * Skylake and Kabylake use legacy HDaudio driver except for Google + * Skylake and Kabylake use legacy HDAudio driver except for Google * Chromebooks (SST) */ @@ -135,7 +135,7 @@ static const struct config_entry config_table[] = { #endif /* - * Geminilake uses legacy HDaudio driver except for Google + * Geminilake uses legacy HDAudio driver except for Google * Chromebooks */ /* Geminilake */ @@ -157,7 +157,7 @@ static const struct config_entry config_table[] = { /* * CoffeeLake, CannonLake, CometLake, IceLake, TigerLake use legacy - * HDaudio driver except for Google Chromebooks and when DMICs are + * HDAudio driver except for Google Chromebooks and when DMICs are * present. Two cases are required since Coreboot does not expose NHLT * tables. * @@ -391,7 +391,7 @@ int snd_intel_dsp_driver_probe(struct pci_dev *pci) if (pci->class == 0x040300) return SND_INTEL_DSP_DRIVER_LEGACY; if (pci->class != 0x040100 && pci->class != 0x040380) { - dev_err(&pci->dev, "Unknown PCI class/subclass/prog-if information (0x%06x) found, selecting HDA legacy driver\n", pci->class); + dev_err(&pci->dev, "Unknown PCI class/subclass/prog-if information (0x%06x) found, selecting HDAudio legacy driver\n", pci->class); return SND_INTEL_DSP_DRIVER_LEGACY; } diff --git a/sound/x86/Kconfig b/sound/x86/Kconfig index 77777192f650..4ffcc5e623c2 100644 --- a/sound/x86/Kconfig +++ b/sound/x86/Kconfig @@ -9,7 +9,7 @@ menuconfig SND_X86 if SND_X86 config HDMI_LPE_AUDIO - tristate "HDMI audio without HDaudio on Intel Atom platforms" + tristate "HDMI audio without HDAudio on Intel Atom platforms" depends on DRM_I915 select SND_PCM help -- 2.27.0

2 1

[PATCH] ALSA: hda: add dev_dbg log when driver is not selected
by Kai Vehmanen 02 Sep '20

02 Sep '20

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> On SKL+ Intel platforms, the driver selection is handled by the snd_intel_dspcfg, and when the HDaudio legacy driver is not selected, be it with the auto-selection or user preferences with a kernel parameter, the probe aborts with no logs, only a -ENODEV return value. Having no dmesg trace, even with dynamic debug enabled, makes support more complicated than it needs to be, and even experienced users can be fooled. A simple dev_dbg() trace solves this problem. BugLink: https://github.com/thesofproject/linux/issues/2330 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: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> --- sound/pci/hda/hda_intel.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 0f86e3765bb3..36a9dbc33aa0 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2127,9 +2127,10 @@ static int azx_probe(struct pci_dev *pci, */ if (dmic_detect) { err = snd_intel_dsp_driver_probe(pci); - if (err != SND_INTEL_DSP_DRIVER_ANY && - err != SND_INTEL_DSP_DRIVER_LEGACY) + if (err != SND_INTEL_DSP_DRIVER_ANY && err != SND_INTEL_DSP_DRIVER_LEGACY) { + dev_dbg(&pci->dev, "HDAudio driver not selected, aborting probe\n"); return -ENODEV; + } } else { dev_warn(&pci->dev, "dmic_detect option is deprecated, pass snd-intel-dspcfg.dsp_driver=1 option instead\n"); } -- 2.27.0

2 1

[PATCH] ALSA: hda: fix a runtime pm issue in SOF when integrated GPU is disabled
by Kai Vehmanen 02 Sep '20

02 Sep '20

From: Rander Wang <rander.wang(a)intel.com> In snd_hdac_device_init pm_runtime_set_active is called to increase child_count in parent device. But when it is failed to build connection with GPU for one case that integrated graphic gpu is disabled, snd_hdac_ext_bus_device_exit will be invoked to clean up a HD-audio extended codec base device. At this time the child_count of parent is not decreased, which makes parent device can't get suspended. This patch calls pm_runtime_set_suspended to decrease child_count in parent device in snd_hdac_device_exit to match with snd_hdac_device_init. pm_runtime_set_suspended can make sure that it will not decrease child_count if the device is already suspended. Signed-off-by: Rander Wang <rander.wang(a)intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> --- sound/hda/hdac_device.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/hda/hdac_device.c b/sound/hda/hdac_device.c index 333220f0f8af..3e9e9ac804f6 100644 --- a/sound/hda/hdac_device.c +++ b/sound/hda/hdac_device.c @@ -127,6 +127,8 @@ EXPORT_SYMBOL_GPL(snd_hdac_device_init); void snd_hdac_device_exit(struct hdac_device *codec) { pm_runtime_put_noidle(&codec->dev); + /* keep balance of runtime PM child_count in parent device */ + pm_runtime_set_suspended(&codec->dev); snd_hdac_bus_remove_device(codec->bus, codec); kfree(codec->vendor_name); kfree(codec->chip_name); -- 2.27.0

2 1

[PATCH] ALSA: hda: hdmi - add Rocketlake support
by Kai Vehmanen 02 Sep '20

02 Sep '20

From: Rander Wang <rander.wang(a)intel.com> Add Rocketlake HDMI codec support. Rocketlake shares the pin-to-port mapping table with Tigerlake. Signed-off-by: Rander Wang <rander.wang(a)intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> --- sound/pci/hda/patch_hdmi.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 1e1b13eb7829..402050088090 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -4269,6 +4269,7 @@ HDA_CODEC_ENTRY(0x8086280c, "Cannonlake HDMI", patch_i915_glk_hdmi), HDA_CODEC_ENTRY(0x8086280d, "Geminilake HDMI", patch_i915_glk_hdmi), HDA_CODEC_ENTRY(0x8086280f, "Icelake HDMI", patch_i915_icl_hdmi), HDA_CODEC_ENTRY(0x80862812, "Tigerlake HDMI", patch_i915_tgl_hdmi), +HDA_CODEC_ENTRY(0x80862816, "Rocketlake HDMI", patch_i915_tgl_hdmi), HDA_CODEC_ENTRY(0x8086281a, "Jasperlake HDMI", patch_i915_icl_hdmi), HDA_CODEC_ENTRY(0x8086281b, "Elkhartlake HDMI", patch_i915_icl_hdmi), HDA_CODEC_ENTRY(0x80862880, "CedarTrail HDMI", patch_generic_hdmi), -- 2.27.0

2 1

general protection fault in snd_ctl_release
by syzbot 02 Sep '20

02 Sep '20

Hello, syzbot found the following issue on: HEAD commit: b51594df Merge tag 'docs-5.9-3' of git://git.lwn.net/linux git tree: upstream console output: https://syzkaller.appspot.com/x/log.txt?x=172fea15900000 kernel config: https://syzkaller.appspot.com/x/.config?x=3c5f6ce8d5b68299 dashboard link: https://syzkaller.appspot.com/bug?extid=dd94e1d44f61c258d538 compiler: gcc (GCC) 10.1.0-syz 20200507 syz repro: https://syzkaller.appspot.com/x/repro.syz?x=179811c1900000 The issue was bisected to: commit a9ed4a6560b8562b7e2e2bed9527e88001f7b682 Author: Marc Zyngier <maz(a)kernel.org> Date: Wed Aug 19 16:12:17 2020 +0000 epoll: Keep a reference on files added to the check list bisection log: https://syzkaller.appspot.com/x/bisect.txt?x=14406199900000 final oops: https://syzkaller.appspot.com/x/report.txt?x=16406199900000 console output: https://syzkaller.appspot.com/x/log.txt?x=12406199900000 IMPORTANT: if you fix the issue, please add the following tag to the commit: Reported-by: syzbot+dd94e1d44f61c258d538(a)syzkaller.appspotmail.com Fixes: a9ed4a6560b8 ("epoll: Keep a reference on files added to the check list") general protection fault, probably for non-canonical address 0xdffffc0000000002: 0000 [#1] PREEMPT SMP KASAN KASAN: null-ptr-deref in range [0x0000000000000010-0x0000000000000017] CPU: 0 PID: 7429 Comm: syz-executor.0 Not tainted 5.9.0-rc3-syzkaller #0 Hardware name: Google Google Compute Engine/Google Compute Engine, BIOS Google 01/01/2011 RIP: 0010:snd_ctl_release+0x69/0x3e0 sound/core/control.c:114 Code: 44 24 10 4c 8b b0 b0 01 00 00 48 c7 80 b0 01 00 00 00 00 00 00 48 b8 00 00 00 00 00 fc ff df 49 8d 7e 10 48 89 fa 48 c1 ea 03 <80> 3c 02 00 0f 85 f7 02 00 00 49 8b 46 10 4d 8d 6e 08 4c 8d a0 48 RSP: 0018:ffffc900073bfe10 EFLAGS: 00010202 RAX: dffffc0000000000 RBX: ffff8880a2a9fcc0 RCX: 1ffffffff11eb3f0 RDX: 0000000000000002 RSI: ffffffff860cbd2b RDI: 0000000000000010 RBP: ffff8880a2a9fcc0 R08: 0000000000000001 R09: ffffffff89cc5567 R10: 0000000000000000 R11: 0000000000000000 R12: ffff8880a2a9fd44 R13: ffff8880a2a9fce8 R14: 0000000000000000 R15: ffff8880a4bc1468 FS: 00007f30f5cd0700(0000) GS:ffff8880ae600000(0000) knlGS:0000000000000000 CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 CR2: 0000000000000000 CR3: 0000000096ced000 CR4: 00000000001506f0 DR0: 0000000000000000 DR1: 0000000000000000 DR2: 0000000000000000 DR3: 0000000000000000 DR6: 00000000fffe0ff0 DR7: 0000000000000400 Call Trace: __fput+0x285/0x920 fs/file_table.c:281 task_work_run+0xdd/0x190 kernel/task_work.c:141 tracehook_notify_resume include/linux/tracehook.h:188 [inline] exit_to_user_mode_loop kernel/entry/common.c:140 [inline] exit_to_user_mode_prepare+0x1e1/0x200 kernel/entry/common.c:167 syscall_exit_to_user_mode+0x7e/0x2e0 kernel/entry/common.c:242 entry_SYSCALL_64_after_hwframe+0x44/0xa9 RIP: 0033:0x45d5b9 Code: 5d b4 fb ff c3 66 2e 0f 1f 84 00 00 00 00 00 66 90 48 89 f8 48 89 f7 48 89 d6 48 89 ca 4d 89 c2 4d 89 c8 4c 8b 4c 24 08 0f 05 <48> 3d 01 f0 ff ff 0f 83 2b b4 fb ff c3 66 2e 0f 1f 84 00 00 00 00 RSP: 002b:00007f30f5ccfc78 EFLAGS: 00000246 ORIG_RAX: 00000000000000e9 RAX: 0000000000000000 RBX: 0000000000002ac0 RCX: 000000000045d5b9 RDX: 0000000000000004 RSI: 0000000000000001 RDI: 0000000000000006 RBP: 000000000118cf88 R08: 0000000000000000 R09: 0000000000000000 R10: 0000000020000180 R11: 0000000000000246 R12: 000000000118cf4c R13: 000000000169fb6f R14: 00007f30f5cd09c0 R15: 000000000118cf4c Modules linked in: ---[ end trace d1d5678dff9246de ]--- RIP: 0010:snd_ctl_release+0x69/0x3e0 sound/core/control.c:114 Code: 44 24 10 4c 8b b0 b0 01 00 00 48 c7 80 b0 01 00 00 00 00 00 00 48 b8 00 00 00 00 00 fc ff df 49 8d 7e 10 48 89 fa 48 c1 ea 03 <80> 3c 02 00 0f 85 f7 02 00 00 49 8b 46 10 4d 8d 6e 08 4c 8d a0 48 RSP: 0018:ffffc900073bfe10 EFLAGS: 00010202 RAX: dffffc0000000000 RBX: ffff8880a2a9fcc0 RCX: 1ffffffff11eb3f0 RDX: 0000000000000002 RSI: ffffffff860cbd2b RDI: 0000000000000010 RBP: ffff8880a2a9fcc0 R08: 0000000000000001 R09: ffffffff89cc5567 R10: 0000000000000000 R11: 0000000000000000 R12: ffff8880a2a9fd44 R13: ffff8880a2a9fce8 R14: 0000000000000000 R15: ffff8880a4bc1468 FS: 00007f30f5cd0700(0000) GS:ffff8880ae600000(0000) knlGS:0000000000000000 CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 CR2: 0000000000000000 CR3: 0000000096ced000 CR4: 00000000001506f0 DR0: 0000000000000000 DR1: 0000000000000000 DR2: 0000000000000000 DR3: 0000000000000000 DR6: 00000000fffe0ff0 DR7: 0000000000000400 --- 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. For information about bisection process see: https://goo.gl/tpsmEJ#bisection syzbot can test patches for this issue, for details see: https://goo.gl/tpsmEJ#testing-patches

3 4

Miss mapping to unmute the middle speakers for Creative X-Fi with 20K1 chipset cards in analog-output.conf
by GitHub issues - edited 02 Sep '20

02 Sep '20

alsa-project/alsa-utils issue #54 was edited from deathxxx123: Hi there. I want to report some bug, that is present from long time and i think that can be fixed now. With any configuration that uses center/subwoofer, that center gets muted by default for Creative X-Fi with 20K1 chipset cards, but if you have other card, that will not happen, because of missing configuration for Center/Subwoofer in `analog-output.conf`. The missing thing in `analog-output.conf` is: ``` [Element Center/LFE] switch = mute volume = merge override-map.1 = center,lfe ``` You can see my issue and fix there: https://github.com/gamer-os/gamer-os/issues/131 Issue URL : https://github.com/alsa-project/alsa-utils/issues/54 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

Miss mapping to unmute the middle speakers for Creative X-Fi with 20K1 chipset cards in analog-output.conf
by GitHub issues - edited 02 Sep '20

02 Sep '20

alsa-project/alsa-utils issue #54 was edited from deathxxx123: Hi there. I want to report some bug, that is present from long time and i think that can be fixed now. With any configuration that uses center/subwoofer, that center gets muted by default for Creative X-Fi with 20K1 chipset cards, but if you have other card, that will not happen, because of missing configuration in `analog-output.conf`. The missing thing in `analog-output.conf` is: ``` [Element Center/LFE] switch = mute volume = merge override-map.1 = center,lfe ``` You can see my issue and fix there: https://github.com/gamer-os/gamer-os/issues/131 Issue URL : https://github.com/alsa-project/alsa-utils/issues/54 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

Miss mapping to unmute the middle speakers for Creative X-Fi with 20K1 chipset cards in analog-output.conf
by GitHub issues - edited 02 Sep '20

02 Sep '20

alsa-project/alsa-utils issue #54 was edited from deathxxx123: Hi there. I want to report some bug, that is present from long time and i think that can be fixed now. With any configuration that uses center/subwoofer, that center gets muted by default for Creative X-Fi with 20K1 chipset cards, but if you have other card, that will not happen, because of missing mapping in `analog-output.conf`. The missing thing in `analog-output.conf` is: ``` [Element Center/LFE] switch = mute volume = merge override-map.1 = center,lfe ``` You can see my issue and fix there: https://github.com/gamer-os/gamer-os/issues/131 Issue URL : https://github.com/alsa-project/alsa-utils/issues/54 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

Miss mapping to unmute the middle speakers for Creative X-Fi with 20K1 chipset cards in analog-output.conf
by GitHub issues - opened 02 Sep '20

02 Sep '20

alsa-project/alsa-utils issue #54 was opened from deathxxx123: Hi there. I want to report some bug, that is present from long time and i think that can be fixed now. With any configuration that uses center/subwoofer, that center gets muted by default with Creative X-Fi with 20K1 cards, but if you have other card, that will not happen, because of missing mapping in `analog-output.conf`. The missing thing in `analog-output.conf` is: ``` [Element Center/LFE] switch = mute volume = merge override-map.1 = center,lfe ``` You can see my issue and fix there: https://github.com/gamer-os/gamer-os/issues/131 Issue URL : https://github.com/alsa-project/alsa-utils/issues/54 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[PATCH] ALSA: hda: Refactor calculating SDnFMT according to specification
by Pawel Harlozinski 02 Sep '20

02 Sep '20

Fix setting SDnFMT based on High Definition Audio Specification Rev. 1.0a page 48. Bits per Sample (BITS): 000 = 8 bits. The data will be packed in memory in 8-bit containers on 16-bit boundaries. 001 = 16 bits. The data will be packed in memory in 16-bit containers on 16-bit boundaries. 010 = 20 bits. The data will be packed in memory in 32-bit containers on 32-bit boundaries. 011 = 24 bits. The data will be packed in memory in 32-bit containers on 32-bit boundaries. 100 = 32 bits. The data will be packed in memory in 32-bit containers on 32-bit boundaries. 101-111 = Reserved Set SDnFMT depending on which format was given, as maxbps only describes container size. Henceforth split cases for formats 20, 24, 32 bits. For format SNDRV_PCM_FORMAT_FLOAT_LE width is equal 32 so as previously it will end up with mask for 32 bits. Signed-off-by: Pawel Harlozinski <pawel.harlozinski(a)linux.intel.com> --- sound/hda/hdac_device.c | 11 +++++------ 1 file changed, 5 insertions(+), 6 deletions(-) diff --git a/sound/hda/hdac_device.c b/sound/hda/hdac_device.c index 333220f0f8af..2ccf79866f99 100644 --- a/sound/hda/hdac_device.c +++ b/sound/hda/hdac_device.c @@ -762,14 +762,13 @@ unsigned int snd_hdac_calc_stream_format(unsigned int rate, val |= AC_FMT_BITS_16; break; case 20: + val |= AC_FMT_BITS_20; + break; case 24: + val |= AC_FMT_BITS_24; + break; case 32: - if (maxbps >= 32 || format == SNDRV_PCM_FORMAT_FLOAT_LE) - val |= AC_FMT_BITS_32; - else if (maxbps >= 24) - val |= AC_FMT_BITS_24; - else - val |= AC_FMT_BITS_20; + val |= AC_FMT_BITS_32; break; default: return 0; -- 2.17.1

4 4

[PATCH] Revert "ASoC: Intel: haswell: Power transition refactor"
by Cezary Rojewski 02 Sep '20

02 Sep '20

This reverts commit 8ec7d6043263ecf250b9b7c0dd8ade899487538a. While addressing existing power-cycle limitations for sound/soc/intel/haswell solution, change brings regression for standard audio userspace flows e.g.: when using PulseAudio. Occasional sound-card initialization fail is still better than pernament audio distortions, so revert the change. Fixes: 8ec7d6043263 ("ASoC: Intel: haswell: Power transition refactor") Reported-by: Christian Bundy <christianbundy(a)fraction.io> Signed-off-by: Cezary Rojewski <cezary.rojewski(a)intel.com> --- sound/soc/intel/haswell/sst-haswell-dsp.c | 185 ++++++++++------------ 1 file changed, 81 insertions(+), 104 deletions(-) diff --git a/sound/soc/intel/haswell/sst-haswell-dsp.c b/sound/soc/intel/haswell/sst-haswell-dsp.c index de80e19454c1..88c3f63bded9 100644 --- a/sound/soc/intel/haswell/sst-haswell-dsp.c +++ b/sound/soc/intel/haswell/sst-haswell-dsp.c @@ -243,92 +243,45 @@ static irqreturn_t hsw_irq(int irq, void *context) return ret; } -#define CSR_DEFAULT_VALUE 0x8480040E -#define ISC_DEFAULT_VALUE 0x0 -#define ISD_DEFAULT_VALUE 0x0 -#define IMC_DEFAULT_VALUE 0x7FFF0003 -#define IMD_DEFAULT_VALUE 0x7FFF0003 -#define IPCC_DEFAULT_VALUE 0x0 -#define IPCD_DEFAULT_VALUE 0x0 -#define CLKCTL_DEFAULT_VALUE 0x7FF -#define CSR2_DEFAULT_VALUE 0x0 -#define LTR_CTRL_DEFAULT_VALUE 0x0 -#define HMD_CTRL_DEFAULT_VALUE 0x0 - -static void hsw_set_shim_defaults(struct sst_dsp *sst) -{ - sst_dsp_shim_write_unlocked(sst, SST_CSR, CSR_DEFAULT_VALUE); - sst_dsp_shim_write_unlocked(sst, SST_ISRX, ISC_DEFAULT_VALUE); - sst_dsp_shim_write_unlocked(sst, SST_ISRD, ISD_DEFAULT_VALUE); - sst_dsp_shim_write_unlocked(sst, SST_IMRX, IMC_DEFAULT_VALUE); - sst_dsp_shim_write_unlocked(sst, SST_IMRD, IMD_DEFAULT_VALUE); - sst_dsp_shim_write_unlocked(sst, SST_IPCX, IPCC_DEFAULT_VALUE); - sst_dsp_shim_write_unlocked(sst, SST_IPCD, IPCD_DEFAULT_VALUE); - sst_dsp_shim_write_unlocked(sst, SST_CLKCTL, CLKCTL_DEFAULT_VALUE); - sst_dsp_shim_write_unlocked(sst, SST_CSR2, CSR2_DEFAULT_VALUE); - sst_dsp_shim_write_unlocked(sst, SST_LTRC, LTR_CTRL_DEFAULT_VALUE); - sst_dsp_shim_write_unlocked(sst, SST_HMDC, HMD_CTRL_DEFAULT_VALUE); -} - -/* all clock-gating minus DCLCGE and DTCGE */ -#define SST_VDRTCL2_CG_OTHER 0xB7D - static void hsw_set_dsp_D3(struct sst_dsp *sst) { + u32 val; u32 reg; - /* disable clock core gating */ + /* Disable core clock gating (VDRTCTL2.DCLCGE = 0) */ reg = readl(sst->addr.pci_cfg + SST_VDRTCTL2); - reg &= ~(SST_VDRTCL2_DCLCGE); + reg &= ~(SST_VDRTCL2_DCLCGE | SST_VDRTCL2_DTCGE); writel(reg, sst->addr.pci_cfg + SST_VDRTCTL2); - /* stall, reset and set 24MHz XOSC */ - sst_dsp_shim_update_bits_unlocked(sst, SST_CSR, - SST_CSR_24MHZ_LPCS | SST_CSR_STALL | SST_CSR_RST, - SST_CSR_24MHZ_LPCS | SST_CSR_STALL | SST_CSR_RST); - - /* DRAM power gating all */ - reg = readl(sst->addr.pci_cfg + SST_VDRTCTL0); - reg |= SST_VDRTCL0_ISRAMPGE_MASK | - SST_VDRTCL0_DSRAMPGE_MASK; - reg &= ~(SST_VDRTCL0_D3SRAMPGD); - reg |= SST_VDRTCL0_D3PGD; - writel(reg, sst->addr.pci_cfg + SST_VDRTCTL0); - udelay(50); + /* enable power gating and switch off DRAM & IRAM blocks */ + val = readl(sst->addr.pci_cfg + SST_VDRTCTL0); + val |= SST_VDRTCL0_DSRAMPGE_MASK | + SST_VDRTCL0_ISRAMPGE_MASK; + val &= ~(SST_VDRTCL0_D3PGD | SST_VDRTCL0_D3SRAMPGD); + writel(val, sst->addr.pci_cfg + SST_VDRTCTL0); - /* PLL shutdown enable */ - reg = readl(sst->addr.pci_cfg + SST_VDRTCTL2); - reg |= SST_VDRTCL2_APLLSE_MASK; - writel(reg, sst->addr.pci_cfg + SST_VDRTCTL2); + /* switch off audio PLL */ + val = readl(sst->addr.pci_cfg + SST_VDRTCTL2); + val |= SST_VDRTCL2_APLLSE_MASK; + writel(val, sst->addr.pci_cfg + SST_VDRTCTL2); - /* disable MCLK */ + /* disable MCLK(clkctl.smos = 0) */ sst_dsp_shim_update_bits_unlocked(sst, SST_CLKCTL, - SST_CLKCTL_MASK, 0); - - /* switch clock gating */ - reg = readl(sst->addr.pci_cfg + SST_VDRTCTL2); - reg |= SST_VDRTCL2_CG_OTHER; - reg &= ~(SST_VDRTCL2_DTCGE); - writel(reg, sst->addr.pci_cfg + SST_VDRTCTL2); - /* enable DTCGE separatelly */ - reg = readl(sst->addr.pci_cfg + SST_VDRTCTL2); - reg |= SST_VDRTCL2_DTCGE; - writel(reg, sst->addr.pci_cfg + SST_VDRTCTL2); + SST_CLKCTL_MASK, 0); - /* set shim defaults */ - hsw_set_shim_defaults(sst); - - /* set D3 */ - reg = readl(sst->addr.pci_cfg + SST_PMCS); - reg |= SST_PMCS_PS_MASK; - writel(reg, sst->addr.pci_cfg + SST_PMCS); + /* Set D3 state, delay 50 us */ + val = readl(sst->addr.pci_cfg + SST_PMCS); + val |= SST_PMCS_PS_MASK; + writel(val, sst->addr.pci_cfg + SST_PMCS); udelay(50); - /* enable clock core gating */ + /* Enable core clock gating (VDRTCTL2.DCLCGE = 1), delay 50 us */ reg = readl(sst->addr.pci_cfg + SST_VDRTCTL2); - reg |= SST_VDRTCL2_DCLCGE; + reg |= SST_VDRTCL2_DCLCGE | SST_VDRTCL2_DTCGE; writel(reg, sst->addr.pci_cfg + SST_VDRTCTL2); + udelay(50); + } static void hsw_reset(struct sst_dsp *sst) @@ -346,62 +299,75 @@ static void hsw_reset(struct sst_dsp *sst) SST_CSR_RST | SST_CSR_STALL, SST_CSR_STALL); } -/* recommended CSR state for power-up */ -#define SST_CSR_D0_MASK (0x18A09C0C | SST_CSR_DCS_MASK) - static int hsw_set_dsp_D0(struct sst_dsp *sst) { - u32 reg; + int tries = 10; + u32 reg, fw_dump_bit; - /* disable clock core gating */ + /* Disable core clock gating (VDRTCTL2.DCLCGE = 0) */ reg = readl(sst->addr.pci_cfg + SST_VDRTCTL2); - reg &= ~(SST_VDRTCL2_DCLCGE); + reg &= ~(SST_VDRTCL2_DCLCGE | SST_VDRTCL2_DTCGE); writel(reg, sst->addr.pci_cfg + SST_VDRTCTL2); - /* switch clock gating */ - reg = readl(sst->addr.pci_cfg + SST_VDRTCTL2); - reg |= SST_VDRTCL2_CG_OTHER; - reg &= ~(SST_VDRTCL2_DTCGE); - writel(reg, sst->addr.pci_cfg + SST_VDRTCTL2); + /* Disable D3PG (VDRTCTL0.D3PGD = 1) */ + reg = readl(sst->addr.pci_cfg + SST_VDRTCTL0); + reg |= SST_VDRTCL0_D3PGD; + writel(reg, sst->addr.pci_cfg + SST_VDRTCTL0); - /* set D0 */ + /* Set D0 state */ reg = readl(sst->addr.pci_cfg + SST_PMCS); - reg &= ~(SST_PMCS_PS_MASK); + reg &= ~SST_PMCS_PS_MASK; writel(reg, sst->addr.pci_cfg + SST_PMCS); - /* DRAM power gating none */ - reg = readl(sst->addr.pci_cfg + SST_VDRTCTL0); - reg &= ~(SST_VDRTCL0_ISRAMPGE_MASK | - SST_VDRTCL0_DSRAMPGE_MASK); - reg |= SST_VDRTCL0_D3SRAMPGD; - reg |= SST_VDRTCL0_D3PGD; - writel(reg, sst->addr.pci_cfg + SST_VDRTCTL0); - mdelay(10); + /* check that ADSP shim is enabled */ + while (tries--) { + reg = readl(sst->addr.pci_cfg + SST_PMCS) & SST_PMCS_PS_MASK; + if (reg == 0) + goto finish; + + msleep(1); + } + + return -ENODEV; - /* set shim defaults */ - hsw_set_shim_defaults(sst); +finish: + /* select SSP1 19.2MHz base clock, SSP clock 0, turn off Low Power Clock */ + sst_dsp_shim_update_bits_unlocked(sst, SST_CSR, + SST_CSR_S1IOCS | SST_CSR_SBCS1 | SST_CSR_LPCS, 0x0); + + /* stall DSP core, set clk to 192/96Mhz */ + sst_dsp_shim_update_bits_unlocked(sst, + SST_CSR, SST_CSR_STALL | SST_CSR_DCS_MASK, + SST_CSR_STALL | SST_CSR_DCS(4)); - /* restore MCLK */ + /* Set 24MHz MCLK, prevent local clock gating, enable SSP0 clock */ sst_dsp_shim_update_bits_unlocked(sst, SST_CLKCTL, - SST_CLKCTL_MASK, SST_CLKCTL_MASK); + SST_CLKCTL_MASK | SST_CLKCTL_DCPLCG | SST_CLKCTL_SCOE0, + SST_CLKCTL_MASK | SST_CLKCTL_DCPLCG | SST_CLKCTL_SCOE0); - /* PLL shutdown disable */ + /* Stall and reset core, set CSR */ + hsw_reset(sst); + + /* Enable core clock gating (VDRTCTL2.DCLCGE = 1), delay 50 us */ reg = readl(sst->addr.pci_cfg + SST_VDRTCTL2); - reg &= ~(SST_VDRTCL2_APLLSE_MASK); + reg |= SST_VDRTCL2_DCLCGE | SST_VDRTCL2_DTCGE; writel(reg, sst->addr.pci_cfg + SST_VDRTCTL2); - sst_dsp_shim_update_bits_unlocked(sst, SST_CSR, - SST_CSR_D0_MASK, SST_CSR_SBCS0 | SST_CSR_SBCS1 | - SST_CSR_STALL | SST_CSR_DCS(4)); udelay(50); - /* enable clock core gating */ + /* switch on audio PLL */ reg = readl(sst->addr.pci_cfg + SST_VDRTCTL2); - reg |= SST_VDRTCL2_DCLCGE; + reg &= ~SST_VDRTCL2_APLLSE_MASK; writel(reg, sst->addr.pci_cfg + SST_VDRTCTL2); - /* clear reset */ - sst_dsp_shim_update_bits_unlocked(sst, SST_CSR, SST_CSR_RST, 0); + /* set default power gating control, enable power gating control for all blocks. that is, + can't be accessed, please enable each block before accessing. */ + reg = readl(sst->addr.pci_cfg + SST_VDRTCTL0); + reg |= SST_VDRTCL0_DSRAMPGE_MASK | SST_VDRTCL0_ISRAMPGE_MASK; + /* for D0, always enable the block(DSRAM[0]) used for FW dump */ + fw_dump_bit = 1 << SST_VDRTCL0_DSRAMPGE_SHIFT; + writel(reg & ~fw_dump_bit, sst->addr.pci_cfg + SST_VDRTCTL0); + /* disable DMA finish function for SSP0 & SSP1 */ sst_dsp_shim_update_bits_unlocked(sst, SST_CSR2, SST_CSR2_SDFD_SSP1, @@ -418,6 +384,12 @@ static int hsw_set_dsp_D0(struct sst_dsp *sst) sst_dsp_shim_update_bits(sst, SST_IMRD, (SST_IMRD_DONE | SST_IMRD_BUSY | SST_IMRD_SSP0 | SST_IMRD_DMAC), 0x0); + /* clear IPC registers */ + sst_dsp_shim_write(sst, SST_IPCX, 0x0); + sst_dsp_shim_write(sst, SST_IPCD, 0x0); + sst_dsp_shim_write(sst, 0x80, 0x6); + sst_dsp_shim_write(sst, 0xe0, 0x300a); + return 0; } @@ -443,6 +415,11 @@ static void hsw_sleep(struct sst_dsp *sst) { dev_dbg(sst->dev, "HSW_PM dsp runtime suspend\n"); + /* put DSP into reset and stall */ + sst_dsp_shim_update_bits(sst, SST_CSR, + SST_CSR_24MHZ_LPCS | SST_CSR_RST | SST_CSR_STALL, + SST_CSR_RST | SST_CSR_STALL | SST_CSR_24MHZ_LPCS); + hsw_set_dsp_D3(sst); dev_dbg(sst->dev, "HSW_PM dsp runtime suspend exit\n"); } -- 2.17.1

4 4

[PATCH v2 00/10] ALSA: convert tasklets to use new
by Allen Pais 02 Sep '20

02 Sep '20

Commit 12cc923f1ccc ("tasklet: Introduce new initialization API")' introduced a new tasklet initialization API. This series converts all the sound drivers to use the new tasklet_setup() API The series is based on 5.9-rc3 (f75aef392f86) v2: Fixed subject lines. Allen Pais (10): ALSA: core: convert tasklets to use new tasklet_setup() API ALSA: firewire: convert tasklets to use new tasklet_setup() API ALSA: pci/asihpi: convert tasklets to use new tasklet_setup() API ALSA: riptide: convert tasklets to use new tasklet_setup() API ALSA: hdsp: convert tasklets to use new tasklet_setup() API ASoc: fsl_esai: convert tasklets to use new tasklet_setup() API ASoC: siu: convert tasklets to use new tasklet_setup() API ASoC: txx9: convert tasklets to use new tasklet_setup() API ALSA: usb-audio: convert tasklets to use new tasklet_setup() API ALSA: ua101: convert tasklets to use new tasklet_setup() API sound/core/timer.c | 7 +++---- sound/firewire/amdtp-stream.c | 8 ++++---- sound/pci/asihpi/asihpi.c | 9 ++++----- sound/pci/riptide/riptide.c | 6 +++--- sound/pci/rme9652/hdsp.c | 6 +++--- sound/pci/rme9652/hdspm.c | 7 +++---- sound/soc/fsl/fsl_esai.c | 7 +++---- sound/soc/sh/siu_pcm.c | 10 ++++------ sound/soc/txx9/txx9aclc.c | 7 +++---- sound/usb/midi.c | 7 +++---- sound/usb/misc/ua101.c | 7 +++---- 11 files changed, 36 insertions(+), 45 deletions(-) -- 2.25.1

4 16

[bug report] 'ASoC: Intel: haswell: Power transition refactor' and PulseAudio
by Christian Bundy 01 Sep '20

01 Sep '20

After upgrading to Linux 5.8 I discovered an audio issue on my device that was introduced in 8ec7d6043263ecf250b9b7c0dd8ade899487538a [0]. I used 'git bisect' to identify the commit that introduced the bug and have confirmed that reverting the commit resolves the problem Reproduction: 1. Play any audio via PulseAudio. 2. Observe that the audio output is fuzzy and choppy. I can use programs like mpv to play audio without PulseAudio, and the audio is fine, but as soon as I open a process that uses PulseAudio it will ruin the audio output for all processes (including mpv) until I reboot. I'm using a 2015 Chromebook Pixel ("Samus") and have confirmed this problem with a friend who has the same device. Is there anything I can do to help debug this instead of sending a patch to revert the commit? Relevant lspci output: 00:03.0 Audio device: Intel Corporation Broadwell-U Audio Controller (rev 09) Subsystem: Intel Corporation Broadwell-U Audio Controller Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx+ Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0, Cache Line Size: 64 bytes Interrupt: pin A routed to IRQ 50 Region 0: Memory at e1218000 (64-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 2 Flags: PMEClk- DSI- D1- D2- AuxCurrent=0mA PME(D0-,D1-,D2-,D3hot-,D3cold-) Status: D0 NoSoftRst- PME-Enable- DSel=0 DScale=0 PME- Capabilities: [60] MSI: Enable+ Count=1/1 Maskable- 64bit- Address: fee00378 Data: 0000 Capabilities: [70] Express (v1) Root Complex Integrated Endpoint, MSI 00 DevCap: MaxPayload 128 bytes, PhantFunc 0 ExtTag- RBE- FLReset+ DevCtl: CorrErr- NonFatalErr- FatalErr- UnsupReq- RlxdOrd- ExtTag- PhantFunc- AuxPwr- NoSnoop- FLReset- MaxPayload 128 bytes, MaxReadReq 128 bytes DevSta: CorrErr- NonFatalErr- FatalErr- UnsupReq- AuxPwr- TransPend- Kernel driver in use: snd_hda_intel Kernel modules: snd_hda_intel [0]: https://git.kernel.org/pub/scm/linux/kernel/git/stable/linux.git/commit/?h=…

4 7

[PATCH] soundwire: fix error handling
by trix＠redhat.com 01 Sep '20

01 Sep '20

From: Tom Rix <trix(a)redhat.com> clang static analysis flags this problem stream.c:844:9: warning: Use of memory after it is freed kfree(bus->defer_msg.msg->buf); ^~~~~~~~~~~~~~~~~~~~~~~ This happens in an error handler cleaning up memory allocated for elements in a list. list_for_each_entry(m_rt, &stream->master_list, stream_node) { bus = m_rt->bus; kfree(bus->defer_msg.msg->buf); kfree(bus->defer_msg.msg); } And is triggered when the call to sdw_bank_switch() fails. There are a two problems. First, when sdw_bank_switch() fails, though it frees memory it does not clear bus's reference 'defer_msg.msg' to that memory. The second problem is the freeing msg->buf. In some cases msg will be NULL so this will dereference a null pointer. Need to check before freeing. Fixes: 99b8a5d608a6 ("soundwire: Add bank switch routine") Signed-off-by: Tom Rix <trix(a)redhat.com> --- drivers/soundwire/stream.c | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) diff --git a/drivers/soundwire/stream.c b/drivers/soundwire/stream.c index 37290a799023..6e36deb505b1 100644 --- a/drivers/soundwire/stream.c +++ b/drivers/soundwire/stream.c @@ -717,6 +717,7 @@ static int sdw_bank_switch(struct sdw_bus *bus, int m_rt_count) kfree(wbuf); error_1: kfree(wr_msg); + bus->defer_msg.msg = NULL; return ret; } @@ -840,9 +841,10 @@ static int do_bank_switch(struct sdw_stream_runtime *stream) error: list_for_each_entry(m_rt, &stream->master_list, stream_node) { bus = m_rt->bus; - - kfree(bus->defer_msg.msg->buf); - kfree(bus->defer_msg.msg); + if (bus->defer_msg.msg) { + kfree(bus->defer_msg.msg->buf); + kfree(bus->defer_msg.msg); + } } msg_unlock: -- 2.18.1

5 8

[PATCH v4] ASoC: tlv320adcx140: Fix accessing uninitialized adcx140->dev
by Camel Guo 01 Sep '20

01 Sep '20

From: Camel Guo <camelg(a)axis.com> In adcx140_i2c_probe, adcx140->dev is accessed before its initialization. This commit fixes this bug. Fixes: 689c7655b50c ("ASoC: tlv320adcx140: Add the tlv320adcx140 codec driver family") Acked-by: Dan Murphy <dmurphy(a)ti.com> Signed-off-by: Camel Guo <camel.guo(a)axis.com> --- v4: - Add Acked-By from Dan v3: - Add Fixes tag v2: - Fix typo sound/soc/codecs/tlv320adcx140.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/tlv320adcx140.c b/sound/soc/codecs/tlv320adcx140.c index 5cd50d841177..7ae6ec374be3 100644 --- a/sound/soc/codecs/tlv320adcx140.c +++ b/sound/soc/codecs/tlv320adcx140.c @@ -980,6 +980,8 @@ static int adcx140_i2c_probe(struct i2c_client *i2c, if (!adcx140) return -ENOMEM; + adcx140->dev = &i2c->dev; + adcx140->gpio_reset = devm_gpiod_get_optional(adcx140->dev, "reset", GPIOD_OUT_LOW); if (IS_ERR(adcx140->gpio_reset)) @@ -1007,7 +1009,7 @@ static int adcx140_i2c_probe(struct i2c_client *i2c, ret); return ret; } - adcx140->dev = &i2c->dev; + i2c_set_clientdata(i2c, adcx140); return devm_snd_soc_register_component(&i2c->dev, -- 2.20.1

2 1

[PATCH] ASoC: rt5682: Prefer async probe
by Douglas Anderson 01 Sep '20

01 Sep '20

The probe of rt5682 is pretty slow. A quick measurement shows that it takes ~650 ms on at least one board. There's no reason to block all other drivers waiting for this probe to finish. Set the flag to allow other drivers to probe while we're probing. Signed-off-by: Douglas Anderson <dianders(a)chromium.org> --- NOTE: I haven't done any analysis of the driver to see _why_ it's so slow, only that I have measured it to be slow. Someone could certainly take the time to profile / optimize it, but in any case it still won't hurt to be async. This is a very safe flag to turn on since: 1. It's not like our probe order was defined by anything anyway. When we probe is at the whim of when our i2c controller probes and that can be any time. 2. If some other driver needs us then they have to handle the fact that we might not have probed yet anyway. 3. There may be other drivers probing at the same time as us anyway because _they_ used async probe. While I won't say that it's impossible to tickle a bug by turning on async probe, I would assert that in almost all cases the bug was already there and needed to be fixed anyway. sound/soc/codecs/rt5682-i2c.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/rt5682-i2c.c b/sound/soc/codecs/rt5682-i2c.c index 85aba311bdc8..6b4e0eb30c89 100644 --- a/sound/soc/codecs/rt5682-i2c.c +++ b/sound/soc/codecs/rt5682-i2c.c @@ -294,6 +294,7 @@ static struct i2c_driver rt5682_i2c_driver = { .name = "rt5682", .of_match_table = rt5682_of_match, .acpi_match_table = rt5682_acpi_match, + .probe_type = PROBE_PREFER_ASYNCHRONOUS, }, .probe = rt5682_i2c_probe, .shutdown = rt5682_i2c_shutdown, -- 2.28.0.402.g5ffc5be6b7-goog

4 3

[PATCH 1/2] ASoC: wm8994: Skip setting of the WM8994_MICBIAS register for WM1811
by Sylwester Nawrocki 01 Sep '20

01 Sep '20

The WM8994_MICBIAS register is not available in the WM1811 CODEC so skip initialization of that register for that device. This suppresses an error during boot: "wm8994-codec: ASoC: error at snd_soc_component_update_bits on wm8994-codec" Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> --- sound/soc/codecs/wm8994.c | 2 ++ sound/soc/codecs/wm_hubs.c | 3 +++ sound/soc/codecs/wm_hubs.h | 1 + 3 files changed, 6 insertions(+) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 038be66..b3ba053 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -4193,11 +4193,13 @@ static int wm8994_component_probe(struct snd_soc_component *component) wm8994->hubs.dcs_readback_mode = 2; break; } + wm8994->hubs.micd_scthr = true; break; case WM8958: wm8994->hubs.dcs_readback_mode = 1; wm8994->hubs.hp_startup_mode = 1; + wm8994->hubs.micd_scthr = true; switch (control->revision) { case 0: diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index 891effe..0c88184 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -1223,6 +1223,9 @@ int wm_hubs_handle_analogue_pdata(struct snd_soc_component *component, snd_soc_component_update_bits(component, WM8993_ADDITIONAL_CONTROL, WM8993_LINEOUT2_FB, WM8993_LINEOUT2_FB); + if (!hubs->micd_scthr) + return 0; + snd_soc_component_update_bits(component, WM8993_MICBIAS, WM8993_JD_SCTHR_MASK | WM8993_JD_THR_MASK | WM8993_MICB1_LVL | WM8993_MICB2_LVL, diff --git a/sound/soc/codecs/wm_hubs.h b/sound/soc/codecs/wm_hubs.h index 4b8e5f0..988b29e 100644 --- a/sound/soc/codecs/wm_hubs.h +++ b/sound/soc/codecs/wm_hubs.h @@ -27,6 +27,7 @@ struct wm_hubs_data { int hp_startup_mode; int series_startup; int no_series_update; + bool micd_scthr; bool no_cache_dac_hp_direct; struct list_head dcs_cache; -- 2.7.4

4 8

[PATCH 00/11] soundwire: intel: add multi-link support
by Bard Liao 01 Sep '20

01 Sep '20

This series enables multi-link support for Intel platforms. Bard Liao (1): soundwire: intel: Only call sdw stream APIs for the first cpu_dai Pierre-Louis Bossart (10): soundwire: intel: disable shim wake on suspend soundwire: intel: ignore software command retries soundwire: intel: add multi-link support soundwire: intel: add missing support for all clock stop modes soundwire: bus: update multi-link definition with hw sync details soundwire: intel: add multi-link hw_synchronization information soundwire: stream: enable hw_sync as needed by hardware soundwire: intel: add dynamic debug trace for clock-stop invalid configs soundwire: intel: pass link_mask information to each master soundwire: intel: don't manage link power individually drivers/soundwire/intel.c | 309 ++++++++++++++++++++++++++++----- drivers/soundwire/intel.h | 2 + drivers/soundwire/intel_init.c | 1 + drivers/soundwire/stream.c | 15 +- include/linux/soundwire/sdw.h | 6 + 5 files changed, 282 insertions(+), 51 deletions(-) -- 2.17.1

4 28

[PATCH] ALSA: pcm: oss: Remove superfluous WARN_ON() for mulaw sanity check
by Takashi Iwai 01 Sep '20

01 Sep '20

The PCM OSS mulaw plugin has a check of the format of the counter part whether it's a linear format. The check is with snd_BUG_ON() that emits WARN_ON() when the debug config is set, and it confuses syzkaller as if it were a serious issue. Let's drop snd_BUG_ON() for avoiding that. While we're at it, correct the error code to a more suitable, EINVAL. Reported-by: syzbot+23b22dc2e0b81cbfcc95(a)syzkaller.appspotmail.com Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/oss/mulaw.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/core/oss/mulaw.c b/sound/core/oss/mulaw.c index 3788906421a7..fe27034f2846 100644 --- a/sound/core/oss/mulaw.c +++ b/sound/core/oss/mulaw.c @@ -329,8 +329,8 @@ int snd_pcm_plugin_build_mulaw(struct snd_pcm_substream *plug, snd_BUG(); return -EINVAL; } - if (snd_BUG_ON(!snd_pcm_format_linear(format->format))) - return -ENXIO; + if (!snd_pcm_format_linear(format->format)) + return -EINVAL; err = snd_pcm_plugin_build(plug, "Mu-Law<->linear conversion", src_format, dst_format, -- 2.16.4

1 0

[PATCH v3] ASoC: tlv320adcx140: Fix accessing uninitialized adcx140->dev
by Camel Guo 01 Sep '20

01 Sep '20

From: Camel Guo <camelg(a)axis.com> In adcx140_i2c_probe, adcx140->dev is accessed before its initialization. This commit fixes this bug. Fixes: 689c7655b50c ("ASoC: tlv320adcx140: Add the tlv320adcx140 codec driver family") Signed-off-by: Camel Guo <camel.guo(a)axis.com> --- sound/soc/codecs/tlv320adcx140.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/tlv320adcx140.c b/sound/soc/codecs/tlv320adcx140.c index 5cd50d841177..7ae6ec374be3 100644 --- a/sound/soc/codecs/tlv320adcx140.c +++ b/sound/soc/codecs/tlv320adcx140.c @@ -980,6 +980,8 @@ static int adcx140_i2c_probe(struct i2c_client *i2c, if (!adcx140) return -ENOMEM; + adcx140->dev = &i2c->dev; + adcx140->gpio_reset = devm_gpiod_get_optional(adcx140->dev, "reset", GPIOD_OUT_LOW); if (IS_ERR(adcx140->gpio_reset)) @@ -1007,7 +1009,7 @@ static int adcx140_i2c_probe(struct i2c_client *i2c, ret); return ret; } - adcx140->dev = &i2c->dev; + i2c_set_clientdata(i2c, adcx140); return devm_snd_soc_register_component(&i2c->dev, -- 2.20.1

1 1

[PATCH v2] ASoC: tlv320adcx140: Fix accessing uninitialized adcx140->dev
by Camel Guo 01 Sep '20

01 Sep '20

From: Camel Guo <camelg(a)axis.com> In adcx140_i2c_probe, adcx140->dev is accessed before its initialization. This commit fixes this bug. Signed-off-by: Camel Guo <camel.guo(a)axis.com> --- sound/soc/codecs/tlv320adcx140.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/tlv320adcx140.c b/sound/soc/codecs/tlv320adcx140.c index 5cd50d841177..7ae6ec374be3 100644 --- a/sound/soc/codecs/tlv320adcx140.c +++ b/sound/soc/codecs/tlv320adcx140.c @@ -980,6 +980,8 @@ static int adcx140_i2c_probe(struct i2c_client *i2c, if (!adcx140) return -ENOMEM; + adcx140->dev = &i2c->dev; + adcx140->gpio_reset = devm_gpiod_get_optional(adcx140->dev, "reset", GPIOD_OUT_LOW); if (IS_ERR(adcx140->gpio_reset)) @@ -1007,7 +1009,7 @@ static int adcx140_i2c_probe(struct i2c_client *i2c, ret); return ret; } - adcx140->dev = &i2c->dev; + i2c_set_clientdata(i2c, adcx140); return devm_snd_soc_register_component(&i2c->dev, -- 2.20.1

1 0

WARNING in snd_pcm_plugin_build_mulaw
by syzbot 01 Sep '20

01 Sep '20

Hello, syzbot found the following issue on: HEAD commit: 15bc20c6 Merge tag 'tty-5.9-rc3' of git://git.kernel.org/p.. git tree: upstream console output: https://syzkaller.appspot.com/x/log.txt?x=1268efb1900000 kernel config: https://syzkaller.appspot.com/x/.config?x=891ca5711a9f1650 dashboard link: https://syzkaller.appspot.com/bug?extid=23b22dc2e0b81cbfcc95 compiler: clang version 10.0.0 (https://github.com/llvm/llvm-project/ c2443155a0fb245c8f17f2c1c72b6ea391e86e81) syz repro: https://syzkaller.appspot.com/x/repro.syz?x=11a95f41900000 C reproducer: https://syzkaller.appspot.com/x/repro.c?x=11a452f2900000 The issue was bisected to: commit f3ca3f5b0966cfc2737178cca8a7a65b09991898 Author: Fabio Estevam <festevam(a)gmail.com> Date: Fri Mar 27 15:57:21 2020 +0000 dt-bindings: sound: cs42l51: Remove unneeded I2C unit name bisection log: https://syzkaller.appspot.com/x/bisect.txt?x=15395f41900000 final oops: https://syzkaller.appspot.com/x/report.txt?x=17395f41900000 console output: https://syzkaller.appspot.com/x/log.txt?x=13395f41900000 IMPORTANT: if you fix the issue, please add the following tag to the commit: Reported-by: syzbot+23b22dc2e0b81cbfcc95(a)syzkaller.appspotmail.com Fixes: f3ca3f5b0966 ("dt-bindings: sound: cs42l51: Remove unneeded I2C unit name") ------------[ cut here ]------------ WARNING: CPU: 0 PID: 6835 at sound/core/oss/mulaw.c:332 snd_pcm_plugin_build_mulaw+0x4ef/0x6e0 sound/core/oss/mulaw.c:332 Kernel panic - not syncing: panic_on_warn set ... CPU: 0 PID: 6835 Comm: syz-executor088 Not tainted 5.9.0-rc2-syzkaller #0 Hardware name: Google Google Compute Engine/Google Compute Engine, BIOS Google 01/01/2011 Call Trace: __dump_stack lib/dump_stack.c:77 [inline] dump_stack+0x1f0/0x31e lib/dump_stack.c:118 panic+0x264/0x7a0 kernel/panic.c:231 __warn+0x227/0x250 kernel/panic.c:600 report_bug+0x1b1/0x2e0 lib/bug.c:198 handle_bug+0x42/0x80 arch/x86/kernel/traps.c:234 exc_invalid_op+0x16/0x40 arch/x86/kernel/traps.c:254 asm_exc_invalid_op+0x12/0x20 arch/x86/include/asm/idtentry.h:536 RIP: 0010:snd_pcm_plugin_build_mulaw+0x4ef/0x6e0 sound/core/oss/mulaw.c:332 Code: 0f 0b bb fa ff ff ff eb cd e8 fd e3 2d fb 0f 0b bb fa ff ff ff eb bf e8 ef e3 2d fb 0f 0b bb fa ff ff ff eb b1 e8 e1 e3 2d fb <0f> 0b bb fa ff ff ff eb a3 89 e9 80 e1 07 80 c1 03 38 c1 0f 8c 77 RSP: 0018:ffffc90001ed7950 EFLAGS: 00010293 RAX: ffffffff8647143f RBX: 0000000000000000 RCX: ffff88808dcbc540 RDX: 0000000000000000 RSI: 0000000000000000 RDI: 0000000000000000 RBP: ffffffff86471e70 R08: ffffffff864710ee R09: fffff520003daf47 R10: fffff520003daf47 R11: 0000000000000000 R12: 1ffff920003daf45 R13: ffffc90001ed7a18 R14: ffffc90001ed7a28 R15: ffffc90001ed7a28 snd_pcm_plug_format_plugins+0xbbd/0x1640 sound/core/oss/pcm_plugin.c:494 snd_pcm_oss_change_params_locked+0x2c29/0x4410 sound/core/oss/pcm_oss.c:952 snd_pcm_oss_change_params sound/core/oss/pcm_oss.c:1084 [inline] snd_pcm_oss_make_ready+0x115/0x2b0 sound/core/oss/pcm_oss.c:1143 snd_pcm_oss_sync+0x265/0x900 sound/core/oss/pcm_oss.c:1642 snd_pcm_oss_release+0x119/0x270 sound/core/oss/pcm_oss.c:2546 __fput+0x34f/0x7b0 fs/file_table.c:281 task_work_run+0x137/0x1c0 kernel/task_work.c:141 exit_task_work include/linux/task_work.h:25 [inline] do_exit+0x5f3/0x1f20 kernel/exit.c:806 do_group_exit+0x161/0x2d0 kernel/exit.c:903 __do_sys_exit_group+0x13/0x20 kernel/exit.c:914 __se_sys_exit_group+0x10/0x10 kernel/exit.c:912 __x64_sys_exit_group+0x37/0x40 kernel/exit.c:912 do_syscall_64+0x31/0x70 arch/x86/entry/common.c:46 entry_SYSCALL_64_after_hwframe+0x44/0xa9 RIP: 0033:0x440038 Code: Bad RIP value. RSP: 002b:00007ffe721a2058 EFLAGS: 00000246 ORIG_RAX: 00000000000000e7 RAX: ffffffffffffffda RBX: 0000000000000000 RCX: 0000000000440038 RDX: 0000000000000000 RSI: 000000000000003c RDI: 0000000000000000 RBP: 00000000004bf870 R08: 00000000000000e7 R09: ffffffffffffffd0 R10: 0000000000000000 R11: 0000000000000246 R12: 0000000000000001 R13: 00000000006d1180 R14: 0000000000000000 R15: 0000000000000000 Kernel Offset: disabled Rebooting in 86400 seconds.. --- 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. For information about bisection process see: https://goo.gl/tpsmEJ#bisection syzbot can test patches for this issue, for details see: https://goo.gl/tpsmEJ#testing-patches

3 2

[PATCH] ASoC: tlv320adcx140: Fix accessing uninitialized abcx140->dev
by Camel Guo 01 Sep '20

01 Sep '20

From: Camel Guo <camelg(a)axis.com> in adcx140_i2c_probe, adcx140->dev is accessed before its initialization. This commit fixes this bug. Signed-off-by: Camel Guo <camel.guo(a)axis.com> --- sound/soc/codecs/tlv320adcx140.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/tlv320adcx140.c b/sound/soc/codecs/tlv320adcx140.c index 5cd50d841177..7ae6ec374be3 100644 --- a/sound/soc/codecs/tlv320adcx140.c +++ b/sound/soc/codecs/tlv320adcx140.c @@ -980,6 +980,8 @@ static int adcx140_i2c_probe(struct i2c_client *i2c, if (!adcx140) return -ENOMEM; + adcx140->dev = &i2c->dev; + adcx140->gpio_reset = devm_gpiod_get_optional(adcx140->dev, "reset", GPIOD_OUT_LOW); if (IS_ERR(adcx140->gpio_reset)) @@ -1007,7 +1009,7 @@ static int adcx140_i2c_probe(struct i2c_client *i2c, ret); return ret; } - adcx140->dev = &i2c->dev; + i2c_set_clientdata(i2c, adcx140); return devm_snd_soc_register_component(&i2c->dev, -- 2.20.1

2 1

[PATCH 00/10] sound: convert tasklets to use new tasklet_setup()
by Allen Pais 01 Sep '20

01 Sep '20

From: Allen Pais <allen.lkml(a)gmail.com> Commit 12cc923f1ccc ("tasklet: Introduce new initialization API")' introduced a new tasklet initialization API. This series converts all the sound drivers to use the new tasklet_setup() API Allen Pais (10): sound: core: convert tasklets to use new tasklet_setup() API sound: firewire: convert tasklets to use new tasklet_setup() API sound: asihpi: convert tasklets to use new tasklet_setup() API sound: riptide: convert tasklets to use new tasklet_setup() API sound: rm9652: convert tasklets to use new tasklet_setup() API sound/soc: fsl_esai: convert tasklets to use new tasklet_setup() API sound/soc: sh: convert tasklets to use new tasklet_setup() API sound/soc: txx9: convert tasklets to use new tasklet_setup() API sound: midi: convert tasklets to use new tasklet_setup() API sound: ua101: convert tasklets to use new tasklet_setup() API sound/core/timer.c | 7 +++---- sound/firewire/amdtp-stream.c | 8 ++++---- sound/pci/asihpi/asihpi.c | 9 ++++----- sound/pci/riptide/riptide.c | 6 +++--- sound/pci/rme9652/hdsp.c | 6 +++--- sound/pci/rme9652/hdspm.c | 7 +++---- sound/soc/fsl/fsl_esai.c | 7 +++---- sound/soc/sh/siu_pcm.c | 10 ++++------ sound/soc/txx9/txx9aclc.c | 7 +++---- sound/usb/midi.c | 7 +++---- sound/usb/misc/ua101.c | 7 +++---- 11 files changed, 36 insertions(+), 45 deletions(-) -- 2.17.1

4 22

[PATCH v3 0/3] ASoC: soundwire: fix port_ready[] dynamic allocation
by Bard Liao 01 Sep '20

01 Sep '20

The existing code allocates memory for the total number of ports. This only works if the ports are contiguous, but will break if e.g. a Devices uses port0, 1, and 14. The port_ready[] array would contain 3 elements, which would lead to an out-of-bounds access. Conversely in other cases, the wrong port index would be used leading to timeouts on prepare. This can be fixed by allocating for the worst-case of 15 ports (DP0..DP14). In addition since the number is now fixed, we can use an array instead of a dynamic allocation. Changes in v3: - Add ASoC tag in the cover letter title. - Edit the title and commit message of the third patch for better understanding. Changes in v2: - Split patches into sdw and asoc patches. Please note that "soundwire: fix port_ready[] dynamic allocation in mipi_disco" and "ASoC: codecs: fix port_ready[] dynamic allocation in ASoC codecs" should be merged at the same time. Pierre-Louis Bossart (3): soundwire: add definition for maximum number of ports soundwire: fix port_ready[] dynamic allocation in mipi_disco ASoC: codecs: soundwire: remove port_ready[] usage from codecs. drivers/soundwire/mipi_disco.c | 18 +----------------- drivers/soundwire/slave.c | 4 ++++ include/linux/soundwire/sdw.h | 5 +++-- sound/soc/codecs/max98373-sdw.c | 15 +-------------- sound/soc/codecs/rt1308-sdw.c | 14 +------------- sound/soc/codecs/rt5682-sdw.c | 15 +-------------- sound/soc/codecs/rt700-sdw.c | 15 +-------------- sound/soc/codecs/rt711-sdw.c | 15 +-------------- sound/soc/codecs/rt715-sdw.c | 33 +-------------------------------- 9 files changed, 14 insertions(+), 120 deletions(-) -- 2.17.1

2 5

[PATCH 0/9] soundwire: use FIELD_{GET|PREP} in subsystem
by Vinod Koul 31 Aug '20

31 Aug '20

Use the FIELD_{GET|PREP} in soundwire subsytem and remove the local SDW_REG_SHIFT(). This makes code IMO look much neater Tested this on db845c board Bard, can you please verify this on intel boards. Vinod Koul (9): soundwire: define and use addr bit masks soundwire: bus: use FIELD_GET() soundwire: slave: use SDW_DISCO_LINK_ID() soundwire: stream: use FIELD_{GET|PREP} soundwire: qcom : use FIELD_{GET|PREP} soundwire: cadence: use FIELD_{GET|PREP} soundwire: intel: use FIELD_{GET|PREP} soundwire: intel_init: use FIELD_{GET|PREP} soundwire: remove SDW_REG_SHIFT() drivers/soundwire/bus.c | 6 +-- drivers/soundwire/cadence_master.c | 53 ++++++++++--------------- drivers/soundwire/intel.c | 40 +++++++------------ drivers/soundwire/intel_init.c | 2 +- drivers/soundwire/qcom.c | 22 ++++------ drivers/soundwire/slave.c | 2 +- drivers/soundwire/stream.c | 13 +++--- include/linux/soundwire/sdw.h | 21 ++++++---- include/linux/soundwire/sdw_registers.h | 7 ---- 9 files changed, 67 insertions(+), 99 deletions(-) -- 2.26.2

3 15

general protection fault in line6_midibuf_write
by syzbot 31 Aug '20

31 Aug '20

Hello, syzbot found the following issue on: HEAD commit: 3ed8e1c2 usb: typec: tcpm: Migrate workqueue to RT priorit.. git tree: https://git.kernel.org/pub/scm/linux/kernel/git/gregkh/usb.git usb-testing console output: https://syzkaller.appspot.com/x/log.txt?x=137be056900000 kernel config: https://syzkaller.appspot.com/x/.config?x=ccafc70ac3d5f49c dashboard link: https://syzkaller.appspot.com/bug?extid=b333bd4e812c1d42827a compiler: gcc (GCC) 10.1.0-syz 20200507 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+b333bd4e812c1d42827a(a)syzkaller.appspotmail.com general protection fault, probably for non-canonical address 0xdffffc0000000019: 0000 [#1] SMP KASAN KASAN: null-ptr-deref in range [0x00000000000000c8-0x00000000000000cf] CPU: 1 PID: 21 Comm: kworker/1:1 Not tainted 5.9.0-rc1-syzkaller #0 Hardware name: Google Google Compute Engine/Google Compute Engine, BIOS Google 01/01/2011 Workqueue: usb_hub_wq hub_event RIP: 0010:midibuf_is_full sound/usb/line6/midibuf.c:44 [inline] RIP: 0010:line6_midibuf_write+0x30/0x3f0 sound/usb/line6/midibuf.c:91 Code: 49 89 f5 41 54 55 89 d5 53 48 89 fb 4c 8d 7b 18 48 83 ec 28 e8 11 68 5f fc 4c 89 fa 48 b8 00 00 00 00 00 fc ff df 48 c1 ea 03 <0f> b6 04 02 84 c0 74 08 3c 03 0f 8e 0a 03 00 00 44 8b 63 18 31 ff RSP: 0018:ffff8881db3099d8 EFLAGS: 00010002 RAX: dffffc0000000000 RBX: 00000000000000b0 RCX: ffffc900006a8000 RDX: 0000000000000019 RSI: ffffffff84e098af RDI: 00000000000000b0 RBP: 0000000000000000 R08: 0000000000000000 R09: ffffffff8734d643 R10: 0000000000000000 R11: 000000000004e017 R12: 00000000000000b0 R13: ffff8881d40aec80 R14: ffff8881d4464984 R15: 00000000000000c8 FS: 0000000000000000(0000) GS:ffff8881db300000(0000) knlGS:0000000000000000 CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 CR2: 00005629deed5130 CR3: 00000001d2254000 CR4: 00000000001506e0 DR0: 0000000000000000 DR1: 0000000000000000 DR2: 0000000000000000 DR3: 0000000000000000 DR6: 00000000fffe0ff0 DR7: 0000000000000400 Call Trace: <IRQ> line6_data_received+0x281/0x520 sound/usb/line6/driver.c:296 __usb_hcd_giveback_urb+0x32d/0x560 drivers/usb/core/hcd.c:1650 usb_hcd_giveback_urb+0x367/0x410 drivers/usb/core/hcd.c:1716 dummy_timer+0x11f2/0x3240 drivers/usb/gadget/udc/dummy_hcd.c:1967 call_timer_fn+0x1ac/0x6e0 kernel/time/timer.c:1413 expire_timers kernel/time/timer.c:1458 [inline] __run_timers.part.0+0x67c/0xa60 kernel/time/timer.c:1755 __run_timers kernel/time/timer.c:1736 [inline] run_timer_softirq+0x80/0x120 kernel/time/timer.c:1768 __do_softirq+0x1af/0x91c kernel/softirq.c:298 asm_call_on_stack+0xf/0x20 arch/x86/entry/entry_64.S:706 </IRQ> __run_on_irqstack arch/x86/include/asm/irq_stack.h:22 [inline] run_on_irqstack_cond arch/x86/include/asm/irq_stack.h:48 [inline] do_softirq_own_stack+0x73/0x90 arch/x86/kernel/irq_64.c:77 invoke_softirq kernel/softirq.c:393 [inline] __irq_exit_rcu kernel/softirq.c:423 [inline] irq_exit_rcu+0x107/0x1a0 kernel/softirq.c:435 sysvec_apic_timer_interrupt+0x43/0x90 arch/x86/kernel/apic/apic.c:1091 asm_sysvec_apic_timer_interrupt+0x12/0x20 arch/x86/include/asm/idtentry.h:581 RIP: 0010:arch_local_irq_restore arch/x86/include/asm/irqflags.h:85 [inline] RIP: 0010:console_unlock+0xa99/0xcd0 kernel/printk/printk.c:2509 Code: 00 89 ee 48 c7 c7 a0 0a 35 87 e8 12 b9 03 00 65 ff 0d 3b 48 d8 7e e9 87 f9 ff ff e8 f1 59 16 00 e8 dc f2 1b 00 ff 74 24 30 9d <e9> 20 fe ff ff e8 dd 59 16 00 48 8d 7d 08 48 89 f8 48 c1 e8 03 42 RSP: 0018:ffff8881da34f5e0 EFLAGS: 00000212 RAX: 00000000000d56ed RBX: 0000000000000200 RCX: 0000000000000006 RDX: 0000000000000000 RSI: 0000000000000000 RDI: ffffffff8129a6d4 RBP: 0000000000000000 R08: 0000000000000001 R09: ffffffff895c65e7 R10: fffffbfff12b8cbc R11: 0000000000003254 R12: ffffffff82b3d6f0 R13: ffffffff876fa450 R14: 0000000000000054 R15: dffffc0000000000 vprintk_emit+0x1b2/0x460 kernel/printk/printk.c:2029 dev_vprintk_emit+0x3eb/0x436 drivers/base/core.c:4133 dev_printk_emit+0xba/0xf1 drivers/base/core.c:4144 __dev_printk+0xcf/0xf5 drivers/base/core.c:4156 _dev_info+0xd7/0x109 drivers/base/core.c:4202 hub_port_init.cold+0x264/0x32f drivers/usb/core/hub.c:4628 hub_port_connect drivers/usb/core/hub.c:5140 [inline] hub_port_connect_change drivers/usb/core/hub.c:5348 [inline] port_event drivers/usb/core/hub.c:5494 [inline] hub_event+0x2191/0x4390 drivers/usb/core/hub.c:5576 process_one_work+0x94c/0x15f0 kernel/workqueue.c:2269 process_scheduled_works kernel/workqueue.c:2331 [inline] worker_thread+0x82b/0x1120 kernel/workqueue.c:2417 kthread+0x392/0x470 kernel/kthread.c:292 ret_from_fork+0x1f/0x30 arch/x86/entry/entry_64.S:294 Modules linked in: ---[ end trace 3dff073ad8b565c9 ]--- RIP: 0010:midibuf_is_full sound/usb/line6/midibuf.c:44 [inline] RIP: 0010:line6_midibuf_write+0x30/0x3f0 sound/usb/line6/midibuf.c:91 Code: 49 89 f5 41 54 55 89 d5 53 48 89 fb 4c 8d 7b 18 48 83 ec 28 e8 11 68 5f fc 4c 89 fa 48 b8 00 00 00 00 00 fc ff df 48 c1 ea 03 <0f> b6 04 02 84 c0 74 08 3c 03 0f 8e 0a 03 00 00 44 8b 63 18 31 ff RSP: 0018:ffff8881db3099d8 EFLAGS: 00010002 RAX: dffffc0000000000 RBX: 00000000000000b0 RCX: ffffc900006a8000 RDX: 0000000000000019 RSI: ffffffff84e098af RDI: 00000000000000b0 RBP: 0000000000000000 R08: 0000000000000000 R09: ffffffff8734d643 R10: 0000000000000000 R11: 000000000004e017 R12: 00000000000000b0 R13: ffff8881d40aec80 R14: ffff8881d4464984 R15: 00000000000000c8 FS: 0000000000000000(0000) GS:ffff8881db300000(0000) knlGS:0000000000000000 CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 CR2: 00005629deed5130 CR3: 00000001d2254000 CR4: 00000000001506e0 DR0: 0000000000000000 DR1: 0000000000000000 DR2: 0000000000000000 DR3: 0000000000000000 DR6: 00000000fffe0ff0 DR7: 0000000000000400 --- 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] ALSA: opti9xx: Fix memleak in snd_opti9xx_read_check
by Dinghao Liu 31 Aug '20

31 Aug '20

When request_region() on chip->res_mc_indir fails, chip->res_mc_base should be freed to prevent memleak. Signed-off-by: Dinghao Liu <dinghao.liu(a)zju.edu.cn> --- sound/isa/opti9xx/opti92x-ad1848.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/isa/opti9xx/opti92x-ad1848.c b/sound/isa/opti9xx/opti92x-ad1848.c index 881d3b5711d2..06ba18d1737c 100644 --- a/sound/isa/opti9xx/opti92x-ad1848.c +++ b/sound/isa/opti9xx/opti92x-ad1848.c @@ -679,8 +679,11 @@ static int snd_opti9xx_read_check(struct snd_opti9xx *chip) #else /* OPTi93X */ chip->res_mc_indir = request_region(chip->mc_indir_index, 2, "OPTi93x MC"); - if (chip->res_mc_indir == NULL) + if (chip->res_mc_indir == NULL) { + release_and_free_resource(chip->res_mc_base); + chip->res_mc_base = NULL; return -EBUSY; + } spin_lock_irqsave(&chip->lock, flags); outb(chip->password, chip->mc_base + chip->pwd_reg); -- 2.17.1

2 1

KASAN: use-after-free Read in snd_pcm_oss_release
by syzbot 31 Aug '20

31 Aug '20

Hello, syzbot found the following issue on: HEAD commit: f37be724 Add linux-next specific files for 20200826 git tree: linux-next console output: https://syzkaller.appspot.com/x/log.txt?x=137c7cb9900000 kernel config: https://syzkaller.appspot.com/x/.config?x=1e5a1cf036089d95 dashboard link: https://syzkaller.appspot.com/bug?extid=12a3fcd1493ec6585007 compiler: gcc (GCC) 10.1.0-syz 20200507 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+12a3fcd1493ec6585007(a)syzkaller.appspotmail.com ================================================================== BUG: KASAN: use-after-free in snd_pcm_oss_release_file sound/core/oss/pcm_oss.c:2367 [inline] BUG: KASAN: use-after-free in snd_pcm_oss_release_file sound/core/oss/pcm_oss.c:2361 [inline] BUG: KASAN: use-after-free in snd_pcm_oss_release+0x28d/0x300 sound/core/oss/pcm_oss.c:2548 Read of size 8 at addr ffff888026b66480 by task syz-executor.1/27751 CPU: 1 PID: 27751 Comm: syz-executor.1 Not tainted 5.9.0-rc2-next-20200826-syzkaller #0 Hardware name: Google Google Compute Engine/Google Compute Engine, BIOS Google 01/01/2011 Call Trace: __dump_stack lib/dump_stack.c:77 [inline] dump_stack+0x18f/0x20d lib/dump_stack.c:118 print_address_description.constprop.0.cold+0xae/0x497 mm/kasan/report.c:383 __kasan_report mm/kasan/report.c:513 [inline] kasan_report.cold+0x1f/0x37 mm/kasan/report.c:530 snd_pcm_oss_release_file sound/core/oss/pcm_oss.c:2367 [inline] snd_pcm_oss_release_file sound/core/oss/pcm_oss.c:2361 [inline] snd_pcm_oss_release+0x28d/0x300 sound/core/oss/pcm_oss.c:2548 __fput+0x285/0x920 fs/file_table.c:281 task_work_run+0xdd/0x190 kernel/task_work.c:141 tracehook_notify_resume include/linux/tracehook.h:188 [inline] exit_to_user_mode_loop kernel/entry/common.c:140 [inline] exit_to_user_mode_prepare+0x195/0x1c0 kernel/entry/common.c:167 syscall_exit_to_user_mode+0x59/0x2b0 kernel/entry/common.c:242 entry_SYSCALL_64_after_hwframe+0x44/0xa9 RIP: 0033:0x416f01 Code: 75 14 b8 03 00 00 00 0f 05 48 3d 01 f0 ff ff 0f 83 04 1b 00 00 c3 48 83 ec 08 e8 0a fc ff ff 48 89 04 24 b8 03 00 00 00 0f 05 <48> 8b 3c 24 48 89 c2 e8 53 fc ff ff 48 89 d0 48 83 c4 08 48 3d 01 RSP: 002b:00007ffdd39a5e70 EFLAGS: 00000293 ORIG_RAX: 0000000000000003 RAX: 0000000000000000 RBX: 0000000000000004 RCX: 0000000000416f01 RDX: 0000000000000001 RSI: 0000000001190428 RDI: 0000000000000003 RBP: 0000000000000000 R08: 0000000000000000 R09: 0000000000000000 R10: 00007ffdd39a5f60 R11: 0000000000000293 R12: 0000000001190428 R13: 0000000000268c54 R14: ffffffffffffffff R15: 000000000118cfec Allocated by task 27754: kasan_save_stack+0x1b/0x40 mm/kasan/common.c:48 kasan_set_track mm/kasan/common.c:56 [inline] __kasan_kmalloc.constprop.0+0xbf/0xd0 mm/kasan/common.c:461 kmem_cache_alloc_trace+0x16e/0x2c0 mm/slab.c:3550 kmalloc include/linux/slab.h:554 [inline] kzalloc include/linux/slab.h:666 [inline] snd_pcm_oss_open_file sound/core/oss/pcm_oss.c:2389 [inline] snd_pcm_oss_open.part.0+0x55e/0x1290 sound/core/oss/pcm_oss.c:2491 snd_pcm_oss_open+0x3c/0x50 sound/core/oss/pcm_oss.c:2455 soundcore_open+0x445/0x600 sound/sound_core.c:593 chrdev_open+0x266/0x770 fs/char_dev.c:414 do_dentry_open+0x4b9/0x11b0 fs/open.c:817 do_open fs/namei.c:3251 [inline] path_openat+0x1b9a/0x2730 fs/namei.c:3368 do_filp_open+0x17e/0x3c0 fs/namei.c:3395 do_sys_openat2+0x16d/0x420 fs/open.c:1168 do_sys_open fs/open.c:1184 [inline] __do_sys_openat fs/open.c:1200 [inline] __se_sys_openat fs/open.c:1195 [inline] __x64_sys_openat+0x13f/0x1f0 fs/open.c:1195 do_syscall_64+0x2d/0x70 arch/x86/entry/common.c:46 entry_SYSCALL_64_after_hwframe+0x44/0xa9 Freed by task 27754: kasan_save_stack+0x1b/0x40 mm/kasan/common.c:48 kasan_set_track+0x1c/0x30 mm/kasan/common.c:56 kasan_set_free_info+0x1b/0x30 mm/kasan/generic.c:355 __kasan_slab_free+0xd8/0x120 mm/kasan/common.c:422 __cache_free mm/slab.c:3418 [inline] kfree+0x103/0x2c0 mm/slab.c:3756 snd_pcm_oss_release_file sound/core/oss/pcm_oss.c:2371 [inline] snd_pcm_oss_release_file sound/core/oss/pcm_oss.c:2361 [inline] snd_pcm_oss_release+0x17e/0x300 sound/core/oss/pcm_oss.c:2548 __fput+0x285/0x920 fs/file_table.c:281 task_work_run+0xdd/0x190 kernel/task_work.c:141 tracehook_notify_resume include/linux/tracehook.h:188 [inline] exit_to_user_mode_loop kernel/entry/common.c:140 [inline] exit_to_user_mode_prepare+0x195/0x1c0 kernel/entry/common.c:167 syscall_exit_to_user_mode+0x59/0x2b0 kernel/entry/common.c:242 entry_SYSCALL_64_after_hwframe+0x44/0xa9 The buggy address belongs to the object at ffff888026b66480 which belongs to the cache kmalloc-32 of size 32 The buggy address is located 0 bytes inside of 32-byte region [ffff888026b66480, ffff888026b664a0) The buggy address belongs to the page: page:0000000076bbbb49 refcount:1 mapcount:0 mapping:0000000000000000 index:0xffff888026b66fc1 pfn:0x26b66 flags: 0xfffe0000000200(slab) raw: 00fffe0000000200 ffffea0002385208 ffffea0000782448 ffff8880aa040100 raw: ffff888026b66fc1 ffff888026b66000 000000010000002e 0000000000000000 page dumped because: kasan: bad access detected Memory state around the buggy address: ffff888026b66380: fa fb fb fb fc fc fc fc fb fb fb fb fc fc fc fc ffff888026b66400: 00 fc fc fc fc fc fc fc 00 00 00 00 fc fc fc fc >ffff888026b66480: fa fb fb fb fc fc fc fc fb fb fb fb fc fc fc fc ^ ffff888026b66500: 00 fc fc fc fc fc fc fc 00 fc fc fc fc fc fc fc ffff888026b66580: fa fb fb fb fc fc fc fc fb fb fb fb fc fc fc fc ================================================================== --- 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.

3 2

Re: [PATCH v2] ALSA: HDA: Early Forbid of runtime PM
by Takashi Iwai 31 Aug '20

31 Aug '20

On Fri, 28 Aug 2020 19:06:51 +0200, Jillela, Emmanuel wrote: > > > > Subject: Re: [PATCH v2] ALSA: HDA: Early Forbid of runtime PM > > > > Hey, > > > > On Fri, 28 Aug 2020, Takashi Iwai wrote: > > > > > On Fri, 28 Aug 2020 01:05:36 +0200, Harsha Priya wrote: > > > > This issue being common across all codecs, pm_runtime_forbid() is > > > > called when the codec object is created to fix this issue. > > > > A codec enables or disables runtime pm in its own probe function. > > > > > > Thanks. The only concern is about the influence on the relevant ASoC > > > code, especially hdac_hda.c. > > > > > > Kai, could you check whether this still works? > > > > I believe Harsha is testing mostly with ASoC, so hdac_hda.c should be covered. > > > > I did queue a SOF CI job for this v2 patch and I'm seeing some failures in module > > load/unload test that might be related and need checking before we merge: > > > > https://sof-ci.01.org/linuxpr/PR2403/build4400/devicetest/ > > > > The actual runtime-PM tests in our test set pass, but module load/unload has failures >on > > some platforms. I'll follow-up when I have a better understanding what goes wrong. > > > > Br, Kai > I am just trying to think ahead to get something that may work. We can add an if check around the pm_runtime_forbid like below > If (!codec->auto_runtime_pm) > Pm_runtime_forbid(&codec->core.dev) > Do you think that will work?? snd_hda_codec_device_new() is called before setting the flag, so it's no use for adding the conditional. thanks, Takashi

1 0

Crash when trying to capture vdownmix
by GitHub issues - opened 31 Aug '20

31 Aug '20

alsa-project/alsa-plugins issue #13 was opened from s09bQ5: Our game captures audio from microphones and tests the available devices on startup. For some reason PortAudio tells us that vdownmix is a 6 channel capture device. When the game tries to test that device, the plugin crashes inside vdownmix_transfer because the src_areas array contained only two valid pointers while mix->channels is 6. I assume the plugin is not prepared to handle the swapped roles of input and output during capture. A simple way to cause the crash is to execute `arecord -D vdownmix -f dat -c 6`. The expected behavior is that the plugin either enforces 2 channel capture with a 4/5/6 channel slave pcm or completely rejects capture. In any case, it should not crash with a SIGSEGV. Issue URL : https://github.com/alsa-project/alsa-plugins/issues/13 Repository URL: https://github.com/alsa-project/alsa-plugins

1 0

[PATCH v2 1/2] ASoC: rockchip-spdif: add description for rk3308
by Johan Jonker 30 Aug '20

30 Aug '20

A test with the command below shows that the compatible string "rockchip,rk3308-spdif", "rockchip,rk3328-spdif" is already in use, but is not added to a document. The current fallback string "rockchip,rk3328-spdif" points to a data set enum RK_SPDIF_RK3366 in rockchip_spdif.c that is not used both in the mainline as in the manufacturer kernel. (Of the enum only RK_SPDIF_RK3288 is used.) So if the properties don't change we might as well use the first SoC in line as fallback string and add the description for rk3308 as: "rockchip,rk3308-spdif", "rockchip,rk3066-spdif" make ARCH=arm64 dtbs_check DT_SCHEMA_FILES=Documentation/devicetree/bindings/sound/rockchip-spdif.yaml Signed-off-by: Johan Jonker <jbx6244(a)gmail.com> --- Changed V2: rebase --- Documentation/devicetree/bindings/sound/rockchip-spdif.yaml | 1 + 1 file changed, 1 insertion(+) diff --git a/Documentation/devicetree/bindings/sound/rockchip-spdif.yaml b/Documentation/devicetree/bindings/sound/rockchip-spdif.yaml index 7bad6f16f..62a61b68d 100644 --- a/Documentation/devicetree/bindings/sound/rockchip-spdif.yaml +++ b/Documentation/devicetree/bindings/sound/rockchip-spdif.yaml @@ -27,6 +27,7 @@ properties: - enum: - rockchip,rk3188-spdif - rockchip,rk3288-spdif + - rockchip,rk3308-spdif - const: rockchip,rk3066-spdif reg: -- 2.11.0

3 3

[PATCH v2 0/9] Audio graph card updates and usage with Tegra210 audio
by Sameer Pujar 29 Aug '20

29 Aug '20

This series proposes following enhancements to audio-graph card driver. * Support multiple instances of a component. * Support open platforms with empty Codec endpoint. * Identify no-pcm DPCM DAI links which can be used in BE<->BE connections. * Add new compatible to support DPCM based DAI chaining. This pushes DT support for Tegra210 based platforms which uses audio-graph card and above enhancements. The series is based on following references where DPCM usgae for Tegra Audio and simple-card driver proposal were discussed. * https://lkml.org/lkml/2020/4/30/519 (DPCM for Tegra) * https://lkml.org/lkml/2020/6/27/4 (simple-card driver) Changelog ========= v1 -> v2 -------- * Re-organized ports/endpoints description for ADMAIF and XBAR. Updated DT patches accordingly. * After above change, multiple Codec endpoint support is not required and hence dropped for now. This will be considered separately if at all required in future. * Re-ordered patches in the series. Sameer Pujar (9): ASoC: soc-core: Fix component name_prefix parsing ASoC: audio-graph: Use of_node and DAI for DPCM DAI link names ASoC: audio-graph: Identify 'no_pcm' DAI links for DPCM ASoC: soc-pcm: Get all BEs along DAPM path ASoC: dt-bindings: audio-graph-card: Support for component chaining ASoC: audio-graph: Add support for component chaining ASoC: audio-graph: Support empty Codec endpoint arm64: tegra: Audio graph header for Tegra210 arm64: tegra: Audio graph sound card for Jetson Nano and TX1 .../devicetree/bindings/sound/audio-graph-card.txt | 1 + .../boot/dts/nvidia/tegra210-audio-graph.dtsi | 141 +++++++++++++ arch/arm64/boot/dts/nvidia/tegra210-p2371-2180.dts | 217 +++++++++++++++++++++ arch/arm64/boot/dts/nvidia/tegra210-p3450-0000.dts | 122 ++++++++++++ include/sound/soc.h | 1 + sound/soc/generic/audio-graph-card.c | 69 ++++++- sound/soc/soc-core.c | 3 +- sound/soc/soc-pcm.c | 3 +- 8 files changed, 545 insertions(+), 12 deletions(-) create mode 100644 arch/arm64/boot/dts/nvidia/tegra210-audio-graph.dtsi -- 2.7.4

2 26

[PATCH 1/1] Fix silent audio output and corrupted input on MSI X570-A PRO
by Dan Crawford 29 Aug '20

29 Aug '20

From: Dan Crawford <dnlcrwfrd(a)gmail.com> Following Christian Lachner's patch for Gigabyte X570-based motherboards, also patch the MSI X570-A PRO motherboard; the ALC1220 codec requires the same workaround for Clevo laptops to enforce the DAC/mixer connection path. Set up a quirk entry for that. I suspect most if all X570 motherboards will require similar patches. Related buglink: https://bugzilla.kernel.org/show_bug.cgi?id=205275 Signed-off-by: Dan Crawford <dnlcrwfrd(a)gmail.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 a1fa983d2..a52d8640e 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2474,6 +2474,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x1462, 0x1275, "MSI-GL63", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK(0x1462, 0x1276, "MSI-GL73", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK(0x1462, 0x1293, "MSI-GP65", ALC1220_FIXUP_CLEVO_P950), + SND_PCI_QUIRK(0x1462, 0x9c37, "MSI X570-A PRO", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK(0x1462, 0x7350, "MSI-7350", ALC889_FIXUP_CD), SND_PCI_QUIRK(0x1462, 0xda57, "MSI Z270-Gaming", ALC1220_FIXUP_GB_DUAL_CODECS), SND_PCI_QUIRK_VENDOR(0x1462, "MSI", ALC882_FIXUP_GPIO3), -- 2.28.0

2 1

[PATCH v6 0/2] Add documentation and machine driver for SC7180 sound card
by Cheng-Yi Chiang 29 Aug '20

29 Aug '20

Note: - The machine driver patch is made by the collaboration of Cheng-Yi Chiang <cychiang(a)chromium.org> Rohit kumar <rohitkr(a)codeaurora.org> Ajit Pandey <ajitp(a)codeaurora.org> But Ajit has left codeaurora. Changes from v1 to v2: - Ducumentation: Addressed all suggestions from Doug. - Machine driver: - Fix comment style for license. - Sort includes. - Remove sc7180_snd_hw_params. - Remove sc7180_dai_init and use aux device instead for headset jack registration. - Statically define format for Primary MI2S. - Atomic is not a concern because there is mutex in card to make sure startup and shutdown happen sequentially. - Fix missing return -EINVAL in startup. - Use static sound card. - Use devm_kzalloc to avoid kfree. Changes from v2 to v3: - Ducumentation: Addressed suggestions from Srini. - Machine driver: - Reuse qcom_snd_parse_of to parse properties. - Remove playback-only and capture-only. - Misc fixes to address comments. Changes from v3 to v4: - Ducumentation: Addressed suggestions from Rob. - Remove definition of dai. - Use 'sound-dai: true' for sound-dai schema. - Add reg property to pass 'make dt_binding_check' check although reg is not used in the driver. - Machine driver: - Add Reviewed-by: Tzung-Bi Shih <tzungbi(a)google.com> Changes from v4 to v5: - Documentation: Addressed suggestions from Rob. - Add definition for "#address-cells" and "#size-cells". - Add additionalProperties: false - Add required properties. Changes from v5 to v6: - Documentation: Addressed suggestions from Rob. - Drop contains in compatible strings. - Only allow dai-link@[0-9] - Remove reg ref since it has a type definition already. Ajit Pandey (1): ASoC: qcom: sc7180: Add machine driver for sound card registration Cheng-Yi Chiang (1): ASoC: qcom: dt-bindings: Add sc7180 machine bindings .../bindings/sound/qcom,sc7180.yaml | 124 +++++++++ sound/soc/qcom/Kconfig | 12 + sound/soc/qcom/Makefile | 2 + sound/soc/qcom/sc7180.c | 244 ++++++++++++++++++ 4 files changed, 382 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/qcom,sc7180.yaml create mode 100644 sound/soc/qcom/sc7180.c -- 2.28.0.297.g1956fa8f8d-goog

5 8

[PATCH] ASoC: meson: axg-toddr: fix channel order on g12 platforms
by Jerome Brunet 28 Aug '20

28 Aug '20

On g12 and following platforms, The first channel of record with more than 2 channels ends being placed randomly on an even channel of the output. On these SoCs, a bit was added to force the first channel to be placed at the beginning of the output. Apparently the behavior if the bit is not set is not easily predictable. According to the documentation, this bit is not present on the axg series. Set the bit on g12 and fix the problem. Fixes: a3c23a8ad4dc ("ASoC: meson: axg-toddr: add g12a support") Reported-by: Nicolas Belin <nbelin(a)baylibre.com> Signed-off-by: Jerome Brunet <jbrunet(a)baylibre.com> --- sound/soc/meson/axg-toddr.c | 24 +++++++++++++++++++++++- 1 file changed, 23 insertions(+), 1 deletion(-) diff --git a/sound/soc/meson/axg-toddr.c b/sound/soc/meson/axg-toddr.c index e711abcf8c12..d6adf7edea41 100644 --- a/sound/soc/meson/axg-toddr.c +++ b/sound/soc/meson/axg-toddr.c @@ -18,6 +18,7 @@ #define CTRL0_TODDR_SEL_RESAMPLE BIT(30) #define CTRL0_TODDR_EXT_SIGNED BIT(29) #define CTRL0_TODDR_PP_MODE BIT(28) +#define CTRL0_TODDR_SYNC_CH BIT(27) #define CTRL0_TODDR_TYPE_MASK GENMASK(15, 13) #define CTRL0_TODDR_TYPE(x) ((x) << 13) #define CTRL0_TODDR_MSB_POS_MASK GENMASK(12, 8) @@ -189,10 +190,31 @@ static const struct axg_fifo_match_data axg_toddr_match_data = { .dai_drv = &axg_toddr_dai_drv }; +static int g12a_toddr_dai_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct axg_fifo *fifo = snd_soc_dai_get_drvdata(dai); + int ret; + + ret = axg_toddr_dai_startup(substream, dai); + if (ret) + return ret; + + /* + * Make sure the first channel ends up in the at beginning of the output + * As weird as it looks, without this the first channel may be misplaced + * in memory, with a random shift of 2 channels. + */ + regmap_update_bits(fifo->map, FIFO_CTRL0, CTRL0_TODDR_SYNC_CH, + CTRL0_TODDR_SYNC_CH); + + return 0; +} + static const struct snd_soc_dai_ops g12a_toddr_ops = { .prepare = g12a_toddr_dai_prepare, .hw_params = axg_toddr_dai_hw_params, - .startup = axg_toddr_dai_startup, + .startup = g12a_toddr_dai_startup, .shutdown = axg_toddr_dai_shutdown, }; -- 2.25.4

2 1

[PATCH] ASoC: tlv320adcx140: Add digital mic channel enable routing
by Dan Murphy 28 Aug '20

28 Aug '20

Add the audio routing map to enable the digital mic paths when the analog mic paths are not enabled. Signed-off-by: Dan Murphy <dmurphy(a)ti.com> --- sound/soc/codecs/tlv320adcx140.c | 37 ++++++++++++++++++++++++++++++++ 1 file changed, 37 insertions(+) diff --git a/sound/soc/codecs/tlv320adcx140.c b/sound/soc/codecs/tlv320adcx140.c index 5cd50d841177..5d1b2a03f0ac 100644 --- a/sound/soc/codecs/tlv320adcx140.c +++ b/sound/soc/codecs/tlv320adcx140.c @@ -412,6 +412,16 @@ static const struct snd_soc_dapm_widget adcx140_dapm_widgets[] = { SND_SOC_DAPM_ADC("CH3_ADC", "CH3 Capture", ADCX140_IN_CH_EN, 5, 0), SND_SOC_DAPM_ADC("CH4_ADC", "CH4 Capture", ADCX140_IN_CH_EN, 4, 0), + SND_SOC_DAPM_ADC("CH1_DIG", "CH1 Capture", ADCX140_IN_CH_EN, 7, 0), + SND_SOC_DAPM_ADC("CH2_DIG", "CH2 Capture", ADCX140_IN_CH_EN, 6, 0), + SND_SOC_DAPM_ADC("CH3_DIG", "CH3 Capture", ADCX140_IN_CH_EN, 5, 0), + SND_SOC_DAPM_ADC("CH4_DIG", "CH4 Capture", ADCX140_IN_CH_EN, 4, 0), + SND_SOC_DAPM_ADC("CH5_DIG", "CH5 Capture", ADCX140_IN_CH_EN, 3, 0), + SND_SOC_DAPM_ADC("CH6_DIG", "CH6 Capture", ADCX140_IN_CH_EN, 2, 0), + SND_SOC_DAPM_ADC("CH7_DIG", "CH7 Capture", ADCX140_IN_CH_EN, 1, 0), + SND_SOC_DAPM_ADC("CH8_DIG", "CH8 Capture", ADCX140_IN_CH_EN, 0, 0), + + SND_SOC_DAPM_SWITCH("CH1_ASI_EN", SND_SOC_NOPM, 0, 0, &adcx140_dapm_ch1_en_switch), SND_SOC_DAPM_SWITCH("CH2_ASI_EN", SND_SOC_NOPM, 0, 0, @@ -470,6 +480,15 @@ static const struct snd_soc_dapm_route adcx140_audio_map[] = { {"CH3_ASI_EN", "Switch", "CH3_ADC"}, {"CH4_ASI_EN", "Switch", "CH4_ADC"}, + {"CH1_ASI_EN", "Switch", "CH1_DIG"}, + {"CH2_ASI_EN", "Switch", "CH2_DIG"}, + {"CH3_ASI_EN", "Switch", "CH3_DIG"}, + {"CH4_ASI_EN", "Switch", "CH4_DIG"}, + {"CH5_ASI_EN", "Switch", "CH5_DIG"}, + {"CH6_ASI_EN", "Switch", "CH6_DIG"}, + {"CH7_ASI_EN", "Switch", "CH7_DIG"}, + {"CH8_ASI_EN", "Switch", "CH8_DIG"}, + {"CH5_ASI_EN", "Switch", "CH5_OUT"}, {"CH6_ASI_EN", "Switch", "CH6_OUT"}, {"CH7_ASI_EN", "Switch", "CH7_OUT"}, @@ -541,6 +560,15 @@ static const struct snd_soc_dapm_route adcx140_audio_map[] = { {"PDM Clk Div Select", "705.6 kHz", "MIC1P Input Mux"}, {"PDM Clk Div Select", "5.6448 MHz", "MIC1P Input Mux"}, + {"MIC1P Input Mux", NULL, "CH1_DIG"}, + {"MIC1M Input Mux", NULL, "CH2_DIG"}, + {"MIC2P Input Mux", NULL, "CH3_DIG"}, + {"MIC2M Input Mux", NULL, "CH4_DIG"}, + {"MIC3P Input Mux", NULL, "CH5_DIG"}, + {"MIC3M Input Mux", NULL, "CH6_DIG"}, + {"MIC4P Input Mux", NULL, "CH7_DIG"}, + {"MIC4M Input Mux", NULL, "CH8_DIG"}, + {"MIC1 Analog Mux", "Line In", "MIC1P"}, {"MIC2 Analog Mux", "Line In", "MIC2P"}, {"MIC3 Analog Mux", "Line In", "MIC3P"}, @@ -554,6 +582,15 @@ static const struct snd_soc_dapm_route adcx140_audio_map[] = { {"MIC3M Input Mux", "Analog", "MIC3M"}, {"MIC4P Input Mux", "Analog", "MIC4P"}, {"MIC4M Input Mux", "Analog", "MIC4M"}, + + {"MIC1P Input Mux", "Digital", "MIC1P"}, + {"MIC1M Input Mux", "Digital", "MIC1M"}, + {"MIC2P Input Mux", "Digital", "MIC2P"}, + {"MIC2M Input Mux", "Digital", "MIC2M"}, + {"MIC3P Input Mux", "Digital", "MIC3P"}, + {"MIC3M Input Mux", "Digital", "MIC3M"}, + {"MIC4P Input Mux", "Digital", "MIC4P"}, + {"MIC4M Input Mux", "Digital", "MIC4M"}, }; static const struct snd_kcontrol_new adcx140_snd_controls[] = { -- 2.28.0

2 1

[PATCH] ASoC: core: remove artificial component and DAI name constraint
by Dmitry Baryshkov 28 Aug '20

28 Aug '20

Current fmt_single_name code limits maximum name of a DAI or component to 32 bytes. On some systems corresponding device names might be longer than that (e.g. 17300000.remoteproc:glink-edge:apr:apr-service@8:routing). This will result in duplicate DAI/component names. Rewrite fmt_single_name() to remove such length limitations. Signed-off-by: Dmitry Baryshkov <dmitry.baryshkov(a)linaro.org> --- sound/soc/soc-core.c | 33 +++++++++++++++------------------ 1 file changed, 15 insertions(+), 18 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 2fe1b2ec7c8f..4f2e2270a0d3 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -44,8 +44,6 @@ #define CREATE_TRACE_POINTS #include <trace/events/asoc.h> -#define NAME_SIZE 32 - static DEFINE_MUTEX(client_mutex); static LIST_HEAD(component_list); static LIST_HEAD(unbind_card_list); @@ -2218,13 +2216,14 @@ EXPORT_SYMBOL_GPL(snd_soc_unregister_card); */ static char *fmt_single_name(struct device *dev, int *id) { - char *found, name[NAME_SIZE]; + const char *devname = dev_name(dev); + char *found, *name; int id1, id2; - if (dev_name(dev) == NULL) + if (devname == NULL) return NULL; - strlcpy(name, dev_name(dev), NAME_SIZE); + name = devm_kstrdup(dev, devname, GFP_KERNEL); /* are we a "%s.%d" name (platform and SPI components) */ found = strstr(name, dev->driver->name); @@ -2237,23 +2236,21 @@ static char *fmt_single_name(struct device *dev, int *id) found[strlen(dev->driver->name)] = '\0'; } - } else { - /* I2C component devices are named "bus-addr" */ - if (sscanf(name, "%x-%x", &id1, &id2) == 2) { - char tmp[NAME_SIZE]; + /* I2C component devices are named "bus-addr" */ + } else if (sscanf(name, "%x-%x", &id1, &id2) == 2) { - /* create unique ID number from I2C addr and bus */ - *id = ((id1 & 0xffff) << 16) + id2; + /* create unique ID number from I2C addr and bus */ + *id = ((id1 & 0xffff) << 16) + id2; - /* sanitize component name for DAI link creation */ - snprintf(tmp, NAME_SIZE, "%s.%s", dev->driver->name, - name); - strlcpy(name, tmp, NAME_SIZE); - } else - *id = 0; + devm_kfree(dev, name); + + /* sanitize component name for DAI link creation */ + name = devm_kasprintf(dev, GFP_KERNEL, "%s.%s", dev->driver->name, devname); + } else { + *id = 0; } - return devm_kstrdup(dev, name, GFP_KERNEL); + return name; } /* -- 2.28.0

2 4

[PATCH 0/4] regmap: add SoundWire 1.2 MBQ support
by Pierre-Louis Bossart 28 Aug '20

28 Aug '20

In preparation of the upstream contribution of SDCA (SoundWire Device Class for Audio) ASoC codec drivers [1] [2], add regmap support SoundWire 1.2 MBQ support. The MBQ (Multi-Byte Quantity) registers need to be handled in a different way from regular 8-bit SoundWire registers, their main application is going to be for volume/gain controls. The 3rd patch was initially suggested for inclusion in the SoundWire tree, and was modified to add more background information in the commit message as requested by Vinod Koul. Pierre-Louis Bossart (4): regmap: sdw: move to -EOPNOTSUPP regmap: sdw: add required header files soundwire: SDCA: add helper macro to access controls regmap: sdw: add support for SoundWire 1.2 MBQ drivers/base/regmap/Kconfig | 6 +- drivers/base/regmap/Makefile | 1 + drivers/base/regmap/regmap-sdw-mbq.c | 102 ++++++++++++++++++++++++ drivers/base/regmap/regmap-sdw.c | 8 +- include/linux/regmap.h | 20 +++++ include/linux/soundwire/sdw_registers.h | 13 +++ 6 files changed, 146 insertions(+), 4 deletions(-) create mode 100644 drivers/base/regmap/regmap-sdw-mbq.c base-commit: 9123e3a74ec7b934a4a099e98af6a61c2f80bbf5 -- 2.25.1

5 25

[PATCH v1] ASoC: core: Replace dma_request_slave_channel() with dma_request_chan()
by Andy Shevchenko 28 Aug '20

28 Aug '20

Drivers should use dma_request_chan() instead of dma_request_slave_channel(). dma_request_slave_channel() is a simple wrapper for dma_request_chan() eating up the error code for channel request failure and makes deferred probing impossible. Signed-off-by: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> --- sound/soc/soc-generic-dmaengine-pcm.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index fb95c1464e66..ab13e08cd636 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -249,8 +249,7 @@ static int dmaengine_pcm_new(struct snd_soc_component *component, continue; if (!pcm->chan[i] && config && config->chan_names[i]) - pcm->chan[i] = dma_request_slave_channel(dev, - config->chan_names[i]); + pcm->chan[i] = dma_request_chan(dev, config->chan_names[i]); if (!pcm->chan[i] && (pcm->flags & SND_DMAENGINE_PCM_FLAG_COMPAT)) { pcm->chan[i] = dmaengine_pcm_compat_request_channel( -- 2.28.0

1 1

[PATCH 0/7] Allwinner A64 digital audio codec fixes
by Samuel Holland 28 Aug '20

28 Aug '20

This series fixes a couple of issues with the digital audio codec in the Allwinner A64 SoC: 1) Left/right channels were swapped when playing/recording audio 2) DAPM topology was wrong, breaking some kcontrols This is the minimum set of changes necessary to fix these issues in a backward-compatible way. For that reason, some DAPM widgets still have incorrect or confusing names; those and other issues will be fixed in later patch sets. Samuel Holland (7): ASoC: dt-bindings: Add a new compatible for the A64 codec ASoC: sun8i-codec: Fix DAPM to match the hardware topology ASoC: sun8i-codec: Add missing mixer routes ASoC: sun8i-codec: Add a quirk for LRCK inversion ARM: dts: sun8i: a33: Update codec widget names arm64: dts: allwinner: a64: Update codec widget names arm64: dts: allwinner: a64: Update the audio codec compatible .../sound/allwinner,sun8i-a33-codec.yaml | 6 +- arch/arm/boot/dts/sun8i-a33-olinuxino.dts | 4 +- arch/arm/boot/dts/sun8i-a33.dtsi | 4 +- .../dts/allwinner/sun50i-a64-bananapi-m64.dts | 8 +- .../dts/allwinner/sun50i-a64-orangepi-win.dts | 8 +- .../boot/dts/allwinner/sun50i-a64-pine64.dts | 8 +- .../dts/allwinner/sun50i-a64-pinebook.dts | 8 +- .../dts/allwinner/sun50i-a64-pinephone.dtsi | 8 +- .../boot/dts/allwinner/sun50i-a64-pinetab.dts | 8 +- .../allwinner/sun50i-a64-sopine-baseboard.dts | 8 +- .../boot/dts/allwinner/sun50i-a64-teres-i.dts | 8 +- arch/arm64/boot/dts/allwinner/sun50i-a64.dtsi | 11 +- sound/soc/sunxi/sun8i-codec.c | 137 ++++++++++++++---- 13 files changed, 155 insertions(+), 71 deletions(-) -- 2.26.2

5 14

[PATCH v2 0/3] soundwire: fix port_ready[] dynamic allocation
by Bard Liao 28 Aug '20

28 Aug '20

The existing code allocates memory for the total number of ports. This only works if the ports are contiguous, but will break if e.g. a Devices uses port0, 1, and 14. The port_ready[] array would contain 3 elements, which would lead to an out-of-bounds access. Conversely in other cases, the wrong port index would be used leading to timeouts on prepare. This can be fixed by allocating for the worst-case of 15 ports (DP0..DP14). In addition since the number is now fixed, we can use an array instead of a dynamic allocation. Changes in v2: - Split patches into sdw and asoc patches. Please note that "soundwire: fix port_ready[] dynamic allocation in mipi_disco" and "ASoC: codecs: fix port_ready[] dynamic allocation in ASoC codecs" should be merged at the same time. Pierre-Louis Bossart (3): soundwire: add definition for maximum number of ports soundwire: fix port_ready[] dynamic allocation in mipi_disco ASoC: codecs: fix port_ready[] dynamic allocation in ASoC codecs drivers/soundwire/mipi_disco.c | 18 +----------------- drivers/soundwire/slave.c | 4 ++++ include/linux/soundwire/sdw.h | 5 +++-- sound/soc/codecs/max98373-sdw.c | 15 +-------------- sound/soc/codecs/rt1308-sdw.c | 14 +------------- sound/soc/codecs/rt5682-sdw.c | 15 +-------------- sound/soc/codecs/rt700-sdw.c | 15 +-------------- sound/soc/codecs/rt711-sdw.c | 15 +-------------- sound/soc/codecs/rt715-sdw.c | 33 +-------------------------------- 9 files changed, 14 insertions(+), 120 deletions(-) -- 2.17.1

3 5

[PATCH] ASoC: soc-core: add snd_soc_find_dai_with_mutex()
by Kuninori Morimoto 28 Aug '20

28 Aug '20

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> commit 25612477d20b52 ("ASoC: soc-dai: set dai_link dpcm_ flags with a helper") added snd_soc_dai_link_set_capabilities(). But it is using snd_soc_find_dai() (A) which is required client_mutex (B). And client_mutex is soc-core.c local. struct snd_soc_dai *snd_soc_find_dai(xxx) { ... (B) lockdep_assert_held(&client_mutex); ... } void snd_soc_dai_link_set_capabilities(xxx) { ... for_each_pcm_streams(direction) { ... for_each_link_cpus(dai_link, i, cpu) { (A) dai = snd_soc_find_dai(cpu); ... } ... for_each_link_codecs(dai_link, i, codec) { (A) dai = snd_soc_find_dai(codec); ... } } ... } Because of these background, we will get WARNING if .config has CONFIG_LOCKDEP. WARNING: CPU: 2 PID: 53 at sound/soc/soc-core.c:814 snd_soc_find_dai+0xf8/0x100 CPU: 2 PID: 53 Comm: kworker/2:1 Not tainted 5.7.0-rc1+ #328 Hardware name: Renesas H3ULCB Kingfisher board based on r8a77951 (DT) Workqueue: events deferred_probe_work_func pstate: 60000005 (nZCv daif -PAN -UAO) pc : snd_soc_find_dai+0xf8/0x100 lr : snd_soc_find_dai+0xf4/0x100 ... Call trace: snd_soc_find_dai+0xf8/0x100 snd_soc_dai_link_set_capabilities+0xa0/0x16c graph_dai_link_of_dpcm+0x390/0x3c0 graph_for_each_link+0x134/0x200 graph_probe+0x144/0x230 platform_drv_probe+0x5c/0xb0 really_probe+0xe4/0x430 driver_probe_device+0x60/0xf4 snd_soc_find_dai() will be used from (X) CPU/Codec/Platform driver with mutex lock, and (Y) Card driver without mutex lock. This snd_soc_dai_link_set_capabilities() is for Card driver, this means called without mutex. This patch adds snd_soc_find_dai_with_mutex() to solve it. Fixes: 25612477d20b52 ("ASoC: soc-dai: set dai_link dpcm_ flags with a helper") Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- This patch was originally "ASoC: soc-core: move snd_soc_dai_link_set_capabilities() to soc-core.c". This is for asoc/for-5.9 branch include/sound/soc.h | 2 ++ sound/soc/soc-core.c | 13 +++++++++++++ sound/soc/soc-dai.c | 4 ++-- 3 files changed, 17 insertions(+), 2 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 5e3919ffb00c..4176071f61bf 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1361,6 +1361,8 @@ void snd_soc_unregister_dai(struct snd_soc_dai *dai); struct snd_soc_dai *snd_soc_find_dai( const struct snd_soc_dai_link_component *dlc); +struct snd_soc_dai *snd_soc_find_dai_with_mutex( + const struct snd_soc_dai_link_component *dlc); #include <sound/soc-dai.h> diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 2fe1b2ec7c8f..fe11856d7a63 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -834,6 +834,19 @@ struct snd_soc_dai *snd_soc_find_dai( } EXPORT_SYMBOL_GPL(snd_soc_find_dai); +struct snd_soc_dai *snd_soc_find_dai_with_mutex( + const struct snd_soc_dai_link_component *dlc) +{ + struct snd_soc_dai *dai; + + mutex_lock(&client_mutex); + dai = snd_soc_find_dai(dlc); + mutex_unlock(&client_mutex); + + return dai; +} +EXPORT_SYMBOL_GPL(snd_soc_find_dai_with_mutex); + static int soc_dai_link_sanity_check(struct snd_soc_card *card, struct snd_soc_dai_link *link) { diff --git a/sound/soc/soc-dai.c b/sound/soc/soc-dai.c index 91a2551e4cef..0dbd312aad08 100644 --- a/sound/soc/soc-dai.c +++ b/sound/soc/soc-dai.c @@ -412,14 +412,14 @@ void snd_soc_dai_link_set_capabilities(struct snd_soc_dai_link *dai_link) supported_codec = false; for_each_link_cpus(dai_link, i, cpu) { - dai = snd_soc_find_dai(cpu); + dai = snd_soc_find_dai_with_mutex(cpu); if (dai && snd_soc_dai_stream_valid(dai, direction)) { supported_cpu = true; break; } } for_each_link_codecs(dai_link, i, codec) { - dai = snd_soc_find_dai(codec); + dai = snd_soc_find_dai_with_mutex(codec); if (dai && snd_soc_dai_stream_valid(dai, direction)) { supported_codec = true; break; -- 2.25.1

2 3

[PATCH 1/3] mfd: madera: Simplify with dev_err_probe()
by Krzysztof Kozlowski 28 Aug '20

28 Aug '20

Common pattern of handling deferred probe can be simplified with dev_err_probe(). Less code and also it prints the error value. Signed-off-by: Krzysztof Kozlowski <krzk(a)kernel.org> --- drivers/mfd/madera-core.c | 11 +++-------- 1 file changed, 3 insertions(+), 8 deletions(-) diff --git a/drivers/mfd/madera-core.c b/drivers/mfd/madera-core.c index 8a8d733fdce5..4ed6ad8ce002 100644 --- a/drivers/mfd/madera-core.c +++ b/drivers/mfd/madera-core.c @@ -369,19 +369,14 @@ EXPORT_SYMBOL_GPL(madera_of_match); static int madera_get_reset_gpio(struct madera *madera) { struct gpio_desc *reset; - int ret; if (madera->pdata.reset) return 0; reset = devm_gpiod_get_optional(madera->dev, "reset", GPIOD_OUT_LOW); - if (IS_ERR(reset)) { - ret = PTR_ERR(reset); - if (ret != -EPROBE_DEFER) - dev_err(madera->dev, "Failed to request /RESET: %d\n", - ret); - return ret; - } + if (IS_ERR(reset)) + return dev_err_probe(madera->dev, PTR_ERR(reset), + "Failed to request /RESET"); /* * A hard reset is needed for full reset of the chip. We allow running -- 2.17.1

3 6

[PATCH] ALSA: Realtek: Early Forbid of runtime PM
by Harsha Priya 28 Aug '20

28 Aug '20

For Realtek codecs, pm_runtime_forbid() is called in the probe function after the build_controls(). In a stress test, occasionally the runtime PM calls are invoked before controls are built. This causes the codec to be runtime suspended before probe completes. Because of this, not all controls are enumerated correctly and audio does not work until system is rebooted. This patch calls pm_runtime_forbid() early to fix the issue. Multiple stress tests of 2000+ cycles has been done to test the fix. Signed-off-by: Harsha Priya <harshapriya.n(a)intel.com> Signed-off-by: Emmanuel Jillela <emmanuel.jillela(a)intel.com> Reviewed-by: Kailang Yang <kailang(a)realtek.com> --- sound/pci/hda/patch_realtek.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 98789691a479..30ea07fd6735 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8418,6 +8418,7 @@ static int patch_alc269(struct hda_codec *codec) { struct alc_spec *spec; int err; + struct device *dev = hda_codec_dev(codec); err = alc_alloc_spec(codec, 0x0b); if (err < 0) @@ -8430,6 +8431,8 @@ static int patch_alc269(struct hda_codec *codec) #ifdef CONFIG_PM codec->patch_ops.suspend = alc269_suspend; codec->patch_ops.resume = alc269_resume; + pm_runtime_dont_use_autosuspend(dev); + pm_runtime_forbid(dev); #endif spec->shutup = alc_default_shutup; spec->init_hook = alc_default_init; -- 2.17.1

3 2

Re: [PATCH 4/5] soundwire: qcom: avoid dependency on CONFIG_SLIMBUS
by Dmitry Baryshkov 27 Aug '20

27 Aug '20

On 08/06/2020 23:43, Jonathan Marek wrote: > The driver may be used without slimbus, so don't depend on slimbus. > > Signed-off-by: Jonathan Marek <jonathan(a)marek.ca> > --- > drivers/soundwire/Kconfig | 1 - > drivers/soundwire/qcom.c | 5 +++++ > 2 files changed, 5 insertions(+), 1 deletion(-) > > diff --git a/drivers/soundwire/Kconfig b/drivers/soundwire/Kconfig > index fa2b4ab92ed9..d121cf739090 100644 > --- a/drivers/soundwire/Kconfig > +++ b/drivers/soundwire/Kconfig > @@ -33,7 +33,6 @@ config SOUNDWIRE_INTEL > > config SOUNDWIRE_QCOM > tristate "Qualcomm SoundWire Master driver" > - depends on SLIMBUS I'd suggest: depends on SLIMBUS || !SLIMBUS #if SLIMBUS=m, this can not be builtin This would allow building both SLIMBUS and SOUNDWIRE_QCOM as modules > depends on SND_SOC > help > SoundWire Qualcomm Master driver. > diff --git a/drivers/soundwire/qcom.c b/drivers/soundwire/qcom.c > index 14334442615f..ac81c64768ea 100644 > --- a/drivers/soundwire/qcom.c > +++ b/drivers/soundwire/qcom.c > @@ -769,13 +769,18 @@ static int qcom_swrm_probe(struct platform_device *pdev) > if (!ctrl) > return -ENOMEM; > > +#ifdef CONFIG_SLIMBUS and then #if IS_ENABLED(CONFIG_SLIBMUS) here > if (dev->parent->bus == &slimbus_bus) { > +#else > + if (false) { > +#endif > ctrl->reg_read = qcom_swrm_ahb_reg_read; > ctrl->reg_write = qcom_swrm_ahb_reg_write; > ctrl->regmap = dev_get_regmap(dev->parent, NULL); > if (!ctrl->regmap) > return -EINVAL; > } else { > + > res = platform_get_resource(pdev, IORESOURCE_MEM, 0); > > ctrl->reg_read = qcom_swrm_cpu_reg_read; > -- With best wishes Dmitry

1 0

INFO: task can't die in p9_fd_close
by syzbot 27 Aug '20

27 Aug '20

Hello, syzbot found the following issue on: HEAD commit: 494d311a Add linux-next specific files for 20200821 git tree: linux-next console output: https://syzkaller.appspot.com/x/log.txt?x=10615b36900000 kernel config: https://syzkaller.appspot.com/x/.config?x=a61d44f28687f508 dashboard link: https://syzkaller.appspot.com/bug?extid=fbe34b643e462f65e542 compiler: gcc (GCC) 10.1.0-syz 20200507 syz repro: https://syzkaller.appspot.com/x/repro.syz?x=15920a05900000 C reproducer: https://syzkaller.appspot.com/x/repro.c?x=13a78539900000 The issue was bisected to: commit af3acca3e35c01920fe476f730dca7345d0a48df Author: Daniel Baluta <daniel.baluta(a)nxp.com> Date: Tue Feb 20 12:53:10 2018 +0000 ASoC: ak5558: Fix style for SPDX identifier bisection log: https://syzkaller.appspot.com/x/bisect.txt?x=12ea5d39900000 final oops: https://syzkaller.appspot.com/x/report.txt?x=11ea5d39900000 console output: https://syzkaller.appspot.com/x/log.txt?x=16ea5d39900000 IMPORTANT: if you fix the issue, please add the following tag to the commit: Reported-by: syzbot+fbe34b643e462f65e542(a)syzkaller.appspotmail.com Fixes: af3acca3e35c ("ASoC: ak5558: Fix style for SPDX identifier") INFO: task syz-executor475:7005 can't die for more than 143 seconds. task:syz-executor475 state:D stack:27208 pid: 7005 ppid: 6875 flags:0x00004004 Call Trace: context_switch kernel/sched/core.c:3778 [inline] __schedule+0x8e5/0x21e0 kernel/sched/core.c:4527 schedule+0xd0/0x2a0 kernel/sched/core.c:4602 schedule_timeout+0x1d8/0x250 kernel/time/timer.c:1855 do_wait_for_common kernel/sched/completion.c:85 [inline] __wait_for_common kernel/sched/completion.c:106 [inline] wait_for_common kernel/sched/completion.c:117 [inline] wait_for_completion+0x163/0x260 kernel/sched/completion.c:138 __flush_work+0x51f/0xab0 kernel/workqueue.c:3046 __cancel_work_timer+0x5de/0x700 kernel/workqueue.c:3133 p9_conn_destroy net/9p/trans_fd.c:889 [inline] p9_fd_close+0x305/0x520 net/9p/trans_fd.c:919 p9_client_destroy+0xbe/0x360 net/9p/client.c:1086 v9fs_session_close+0x45/0x2c0 fs/9p/v9fs.c:498 v9fs_kill_super+0x49/0x90 fs/9p/vfs_super.c:222 deactivate_locked_super+0x94/0x160 fs/super.c:335 v9fs_mount+0x77c/0x970 fs/9p/vfs_super.c:203 legacy_get_tree+0x105/0x220 fs/fs_context.c:592 vfs_get_tree+0x89/0x2f0 fs/super.c:1547 do_new_mount fs/namespace.c:2896 [inline] path_mount+0x1287/0x1e20 fs/namespace.c:3214 do_mount fs/namespace.c:3227 [inline] __do_sys_mount fs/namespace.c:3435 [inline] __se_sys_mount fs/namespace.c:3412 [inline] __x64_sys_mount+0x27f/0x300 fs/namespace.c:3412 do_syscall_64+0x2d/0x70 arch/x86/entry/common.c:46 entry_SYSCALL_64_after_hwframe+0x44/0xa9 RIP: 0033:0x446eb9 Code: Bad RIP value. RSP: 002b:00007f6a133add98 EFLAGS: 00000246 ORIG_RAX: 00000000000000a5 RAX: ffffffffffffffda RBX: 00000000006dcc48 RCX: 0000000000446eb9 RDX: 0000000020000200 RSI: 0000000020000000 RDI: 0000000000000000 RBP: 00000000006dcc40 R08: 00000000200028c0 R09: 0000000000000000 R10: 0000000000000000 R11: 0000000000000246 R12: 00000000006dcc4c R13: 00000000200003c0 R14: 00000000004af538 R15: 0000000000000000 INFO: task syz-executor475:7005 blocked for more than 143 seconds. Not tainted 5.9.0-rc1-next-20200821-syzkaller #0 "echo 0 > /proc/sys/kernel/hung_task_timeout_secs" disables this message. task:syz-executor475 state:D stack:27208 pid: 7005 ppid: 6875 flags:0x00004004 Call Trace: context_switch kernel/sched/core.c:3778 [inline] __schedule+0x8e5/0x21e0 kernel/sched/core.c:4527 schedule+0xd0/0x2a0 kernel/sched/core.c:4602 schedule_timeout+0x1d8/0x250 kernel/time/timer.c:1855 do_wait_for_common kernel/sched/completion.c:85 [inline] __wait_for_common kernel/sched/completion.c:106 [inline] wait_for_common kernel/sched/completion.c:117 [inline] wait_for_completion+0x163/0x260 kernel/sched/completion.c:138 __flush_work+0x51f/0xab0 kernel/workqueue.c:3046 __cancel_work_timer+0x5de/0x700 kernel/workqueue.c:3133 p9_conn_destroy net/9p/trans_fd.c:889 [inline] p9_fd_close+0x305/0x520 net/9p/trans_fd.c:919 p9_client_destroy+0xbe/0x360 net/9p/client.c:1086 v9fs_session_close+0x45/0x2c0 fs/9p/v9fs.c:498 v9fs_kill_super+0x49/0x90 fs/9p/vfs_super.c:222 deactivate_locked_super+0x94/0x160 fs/super.c:335 v9fs_mount+0x77c/0x970 fs/9p/vfs_super.c:203 legacy_get_tree+0x105/0x220 fs/fs_context.c:592 vfs_get_tree+0x89/0x2f0 fs/super.c:1547 do_new_mount fs/namespace.c:2896 [inline] path_mount+0x1287/0x1e20 fs/namespace.c:3214 do_mount fs/namespace.c:3227 [inline] __do_sys_mount fs/namespace.c:3435 [inline] __se_sys_mount fs/namespace.c:3412 [inline] __x64_sys_mount+0x27f/0x300 fs/namespace.c:3412 do_syscall_64+0x2d/0x70 arch/x86/entry/common.c:46 entry_SYSCALL_64_after_hwframe+0x44/0xa9 RIP: 0033:0x446eb9 Code: Bad RIP value. RSP: 002b:00007f6a133add98 EFLAGS: 00000246 ORIG_RAX: 00000000000000a5 RAX: ffffffffffffffda RBX: 00000000006dcc48 RCX: 0000000000446eb9 RDX: 0000000020000200 RSI: 0000000020000000 RDI: 0000000000000000 RBP: 00000000006dcc40 R08: 00000000200028c0 R09: 0000000000000000 R10: 0000000000000000 R11: 0000000000000246 R12: 00000000006dcc4c R13: 00000000200003c0 R14: 00000000004af538 R15: 0000000000000000 Showing all locks held in the system: 1 lock held by khungtaskd/1173: #0: ffffffff89c675c0 (rcu_read_lock){....}-{1:2}, at: debug_show_all_locks+0x53/0x260 kernel/locking/lockdep.c:5825 1 lock held by in:imklog/6540: 2 locks held by kworker/0:1/6858: #0: ffff8880ae635dd8 (&rq->lock){-.-.}-{2:2}, at: rq_lock kernel/sched/sched.h:1292 [inline] #0: ffff8880ae635dd8 (&rq->lock){-.-.}-{2:2}, at: __schedule+0x232/0x21e0 kernel/sched/core.c:4445 #1: ffff8880ae620ec8 (&per_cpu_ptr(group->pcpu, cpu)->seq){-.-.}-{0:0}, at: psi_task_switch+0x2fb/0x400 kernel/sched/psi.c:833 2 locks held by kworker/0:2/6898: #0: ffff8880aa063d38 ((wq_completion)events){+.+.}-{0:0}, at: arch_atomic64_set arch/x86/include/asm/atomic64_64.h:34 [inline] #0: ffff8880aa063d38 ((wq_completion)events){+.+.}-{0:0}, at: atomic64_set include/asm-generic/atomic-instrumented.h:856 [inline] #0: ffff8880aa063d38 ((wq_completion)events){+.+.}-{0:0}, at: atomic_long_set include/asm-generic/atomic-long.h:41 [inline] #0: ffff8880aa063d38 ((wq_completion)events){+.+.}-{0:0}, at: set_work_data kernel/workqueue.c:616 [inline] #0: ffff8880aa063d38 ((wq_completion)events){+.+.}-{0:0}, at: set_work_pool_and_clear_pending kernel/workqueue.c:643 [inline] #0: ffff8880aa063d38 ((wq_completion)events){+.+.}-{0:0}, at: process_one_work+0x82b/0x1670 kernel/workqueue.c:2240 #1: ffffc90005577da8 ((work_completion)(&m->wq)){+.+.}-{0:0}, at: process_one_work+0x85f/0x1670 kernel/workqueue.c:2244 1 lock held by syz-executor475/6960: #0: ffff8880a7c7a0e0 (&type->s_umount_key#45/1){+.+.}-{3:3}, at: alloc_super+0x201/0xa90 fs/super.c:229 1 lock held by syz-executor475/7064: #0: ffff8880a6c9c0e0 (&type->s_umount_key#45/1){+.+.}-{3:3}, at: alloc_super+0x201/0xa90 fs/super.c:229 2 locks held by kworker/0:3/7070: #0: ffff8880aa073538 ((wq_completion)rcu_gp){+.+.}-{0:0}, at: arch_atomic64_set arch/x86/include/asm/atomic64_64.h:34 [inline] #0: ffff8880aa073538 ((wq_completion)rcu_gp){+.+.}-{0:0}, at: atomic64_set include/asm-generic/atomic-instrumented.h:856 [inline] #0: ffff8880aa073538 ((wq_completion)rcu_gp){+.+.}-{0:0}, at: atomic_long_set include/asm-generic/atomic-long.h:41 [inline] #0: ffff8880aa073538 ((wq_completion)rcu_gp){+.+.}-{0:0}, at: set_work_data kernel/workqueue.c:616 [inline] #0: ffff8880aa073538 ((wq_completion)rcu_gp){+.+.}-{0:0}, at: set_work_pool_and_clear_pending kernel/workqueue.c:643 [inline] #0: ffff8880aa073538 ((wq_completion)rcu_gp){+.+.}-{0:0}, at: process_one_work+0x82b/0x1670 kernel/workqueue.c:2240 #1: ffffc90006157da8 ((work_completion)(&rew.rew_work)){+.+.}-{0:0}, at: process_one_work+0x85f/0x1670 kernel/workqueue.c:2244 1 lock held by syz-executor475/7279: #0: ffff88809476e0e0 (&type->s_umount_key#45/1){+.+.}-{3:3}, at: alloc_super+0x201/0xa90 fs/super.c:229 1 lock held by syz-executor475/32328: #0: ffffffff89c6bc28 (rcu_state.exp_mutex){+.+.}-{3:3}, at: exp_funnel_lock kernel/rcu/tree_exp.h:322 [inline] #0: ffffffff89c6bc28 (rcu_state.exp_mutex){+.+.}-{3:3}, at: synchronize_rcu_expedited+0x274/0x5f0 kernel/rcu/tree_exp.h:836 1 lock held by syz-executor475/32335: #0: ffffffff89c6bc28 (rcu_state.exp_mutex){+.+.}-{3:3}, at: exp_funnel_lock kernel/rcu/tree_exp.h:322 [inline] #0: ffffffff89c6bc28 (rcu_state.exp_mutex){+.+.}-{3:3}, at: synchronize_rcu_expedited+0x274/0x5f0 kernel/rcu/tree_exp.h:836 ============================================= NMI backtrace for cpu 1 CPU: 1 PID: 1173 Comm: khungtaskd Not tainted 5.9.0-rc1-next-20200821-syzkaller #0 Hardware name: Google Google Compute Engine/Google Compute Engine, BIOS Google 01/01/2011 Call Trace: __dump_stack lib/dump_stack.c:77 [inline] dump_stack+0x18f/0x20d lib/dump_stack.c:118 nmi_cpu_backtrace.cold+0x44/0xd7 lib/nmi_backtrace.c:105 nmi_trigger_cpumask_backtrace+0x1b3/0x223 lib/nmi_backtrace.c:62 trigger_all_cpu_backtrace include/linux/nmi.h:147 [inline] check_hung_uninterruptible_tasks kernel/hung_task.c:253 [inline] watchdog+0xd89/0xf30 kernel/hung_task.c:339 kthread+0x3b5/0x4a0 kernel/kthread.c:292 ret_from_fork+0x1f/0x30 arch/x86/entry/entry_64.S:294 Sending NMI from CPU 1 to CPUs 0: NMI backtrace for cpu 0 CPU: 0 PID: 32335 Comm: syz-executor475 Not tainted 5.9.0-rc1-next-20200821-syzkaller #0 Hardware name: Google Google Compute Engine/Google Compute Engine, BIOS Google 01/01/2011 RIP: 0010:write_comp_data+0x0/0x80 kernel/kcov.c:212 Code: e8 ff ff cc cc cc cc cc cc cc 65 48 8b 04 25 c0 fe 01 00 48 8b 80 20 14 00 00 c3 0f 1f 44 00 00 66 2e 0f 1f 84 00 00 00 00 00 <49> 89 f2 65 8b 05 76 90 8d 7e 65 48 8b 34 25 c0 fe 01 00 a9 00 01 RSP: 0018:ffffc90009c47ab8 EFLAGS: 00000086 RAX: 0000000000000000 RBX: ffff888096e0e000 RCX: ffffffff8134b5d6 RDX: ffff888116e0e000 RSI: ffff888096e0e000 RDI: 0000000000000006 RBP: ffff888116e0e000 R08: 0000000000000001 R09: ffff8880ae736dc7 R10: 0000000000000000 R11: 0000000000000001 R12: 0000000000000000 R13: ffff888096e0e000 R14: ffff8880ae632c80 R15: ffff8880a43f9f00 FS: 00007f6a133ae700(0000) GS:ffff8880ae600000(0000) knlGS:0000000000000000 CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 CR2: 00007f19a2450710 CR3: 0000000097214000 CR4: 00000000001506f0 DR0: 0000000000000000 DR1: 0000000000000000 DR2: 0000000000000000 DR3: 0000000000000000 DR6: 00000000fffe0ff0 DR7: 0000000000000400 Call Trace: __phys_addr+0x26/0x110 arch/x86/mm/physaddr.c:20 virt_to_head_page include/linux/mm.h:846 [inline] qlink_to_cache mm/kasan/quarantine.c:128 [inline] qlist_move_cache+0x72/0xd0 mm/kasan/quarantine.c:278 per_cpu_remove_cache+0x47/0x60 mm/kasan/quarantine.c:296 on_each_cpu+0xf0/0x240 kernel/smp.c:834 quarantine_remove_cache+0x40/0xd0 mm/kasan/quarantine.c:313 shutdown_cache mm/slab_common.c:449 [inline] kmem_cache_destroy+0x50/0x120 mm/slab_common.c:497 p9_client_destroy+0x24b/0x360 net/9p/client.c:1097 v9fs_session_close+0x45/0x2c0 fs/9p/v9fs.c:498 v9fs_kill_super+0x49/0x90 fs/9p/vfs_super.c:222 deactivate_locked_super+0x94/0x160 fs/super.c:335 v9fs_mount+0x77c/0x970 fs/9p/vfs_super.c:203 legacy_get_tree+0x105/0x220 fs/fs_context.c:592 vfs_get_tree+0x89/0x2f0 fs/super.c:1547 do_new_mount fs/namespace.c:2896 [inline] path_mount+0x1287/0x1e20 fs/namespace.c:3214 do_mount fs/namespace.c:3227 [inline] __do_sys_mount fs/namespace.c:3435 [inline] __se_sys_mount fs/namespace.c:3412 [inline] __x64_sys_mount+0x27f/0x300 fs/namespace.c:3412 do_syscall_64+0x2d/0x70 arch/x86/entry/common.c:46 entry_SYSCALL_64_after_hwframe+0x44/0xa9 RIP: 0033:0x446eb9 Code: e8 dc e6 ff ff 48 83 c4 18 c3 0f 1f 80 00 00 00 00 48 89 f8 48 89 f7 48 89 d6 48 89 ca 4d 89 c2 4d 89 c8 4c 8b 4c 24 08 0f 05 <48> 3d 01 f0 ff ff 0f 83 ab 05 fc ff c3 66 2e 0f 1f 84 00 00 00 00 RSP: 002b:00007f6a133add98 EFLAGS: 00000246 ORIG_RAX: 00000000000000a5 RAX: ffffffffffffffda RBX: 00000000006dcc48 RCX: 0000000000446eb9 RDX: 0000000020000200 RSI: 0000000020000000 RDI: 0000000000000000 RBP: 00000000006dcc40 R08: 00000000200028c0 R09: 0000000000000000 R10: 0000000000000000 R11: 0000000000000246 R12: 00000000006dcc4c R13: 00000000200003c0 R14: 00000000004af538 R15: 0000000000000000 --- 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. For information about bisection process see: https://goo.gl/tpsmEJ#bisection syzbot can test patches for this issue, for details see: https://goo.gl/tpsmEJ#testing-patches

3 2

[PATCH 00/25] alsa-tools: efw-downloader: add initial version of firmwre downloader for Echo Audio Fireworks devices
by Takashi Sakamoto 27 Aug '20

27 Aug '20

Hi, This patchset is for alsa-tool repository[1] to add a new command-line tool, efw-downloader. The tools is designed to operate on-board flash memory for devices based on Fireworks board module. The patches are also available in my personal repository in github.com[2]. Fireworks board module was designed by Echo Digital Audio corporation. The board module has on-board flash memory to store firmware blob and session data. The contents of flash memory can be operated by software by a pair of asynchronous transactions defined by Echo Digital Audio corporation. Echo Digital Audio corporation also designed file format of firmware. Hardware Vendors including Echo Digital Audio corporation shipped several versions of firmware by the format in driver packages for Windows and macOS. The goal of this tool is to operate the flash memory to download any version of firmware. In this patchset, limited functionalities are added; read from the flash memory, and parse the content of file. The other functionalities are planned to added in future work. I thinks it possible to put actual firmware blobs into somewhere suitable for them (alsa-firmware or linux-firmware repositories). I think it better to prepare the files by reading on-board flash memory, with enough care of copyright of original firmware files shipped by vendor. In the case, it's preferable to use file format different from the original one. But it's my first time for this kind of work. I'd like to ask some advices to alsa developers. [1] https://github.com/alsa-project/alsa-tools/ [2] https://github.com/takaswie/alsa-tools/tree/topic/efw-downloader Takashi Sakamoto (25): efw-downloader: start a new project to operate on-board flash memory for Fireworks board module efw-downloader: efw-proto: define EfwProto as derived object of HinawaFwResp efw-downloader: efw-proto: add constructor, destructor, bind, unbind functions efw-downloader: efw-proto: add responded signal efw-downloader: efw-proto: add class virtual method to handle responded signal efw-downloader: efw-proto: add instance private structure efw-downloader: efw-proto: emit responded signal at receiving response efw-downloader: efw-proto: add function to finish transaction for command frame efw-downloader: efw-proto: add function to finish a pair of transactions efw-downloader: add parser for sub commands efw-downloader: subcmd-device: implement 'device' sub command to operate actual device efw-downloader: subcmd-device: open firewire character device by HinawaFwNode efw-downloader: config-rom: parse config rom to detect supported device efw-downloader: subcmd-device: check supported models or not efw-downloader: subcmd-device: bind Fireworks protocol efw-downloader: node_dispatcher: add event dispatcher utilizing GLib MainContext/MainLoop efw-downloader: subcmd-device: support debug output for response of Fireworks protocol efw-downloader: efw-commands: add support for a part of hardware command efw-downloader: efw-commands: add support for commands in flash category efw-downloader: subcmd-device: add read operation efw-downloader: file-cntr: add parser for data binary shipped by Echo Digital Audio corporation efw-downloader: subcmd-file: add 'file' sub command efw-downloader: subcmd-file: add parse operation efw-downloader: man: add online manual efw-downloader: add README formatted by reStructuredText efw-downloader/COPYING | 674 ++++++++++++++++++ efw-downloader/README.rst | 167 +++++ efw-downloader/man/efw-downloader.1 | 162 +++++ efw-downloader/meson.build | 14 + efw-downloader/meson_options.txt | 5 + efw-downloader/src/config-rom.c | 60 ++ efw-downloader/src/config-rom.h | 26 + efw-downloader/src/efw-commands.c | 243 +++++++ efw-downloader/src/efw-commands.h | 57 ++ .../src/efw-proto-sigs-marshal.list | 1 + efw-downloader/src/efw-proto.c | 401 +++++++++++ efw-downloader/src/efw-proto.h | 80 +++ efw-downloader/src/file-cntr.c | 183 +++++ efw-downloader/src/file-cntr.h | 39 + efw-downloader/src/main.c | 54 ++ efw-downloader/src/meson.build | 54 ++ efw-downloader/src/node-dispatcher.c | 86 +++ efw-downloader/src/node-dispatcher.h | 21 + efw-downloader/src/op-device-read.c | 104 +++ efw-downloader/src/op-file-parse.c | 106 +++ efw-downloader/src/subcmd-device.c | 182 +++++ efw-downloader/src/subcmd-file.c | 87 +++ efw-downloader/src/subcmds.h | 16 + 23 files changed, 2822 insertions(+) create mode 100644 efw-downloader/COPYING create mode 100644 efw-downloader/README.rst create mode 100644 efw-downloader/man/efw-downloader.1 create mode 100644 efw-downloader/meson.build create mode 100644 efw-downloader/meson_options.txt create mode 100644 efw-downloader/src/config-rom.c create mode 100644 efw-downloader/src/config-rom.h create mode 100644 efw-downloader/src/efw-commands.c create mode 100644 efw-downloader/src/efw-commands.h create mode 100644 efw-downloader/src/efw-proto-sigs-marshal.list create mode 100644 efw-downloader/src/efw-proto.c create mode 100644 efw-downloader/src/efw-proto.h create mode 100644 efw-downloader/src/file-cntr.c create mode 100644 efw-downloader/src/file-cntr.h create mode 100644 efw-downloader/src/main.c create mode 100644 efw-downloader/src/meson.build create mode 100644 efw-downloader/src/node-dispatcher.c create mode 100644 efw-downloader/src/node-dispatcher.h create mode 100644 efw-downloader/src/op-device-read.c create mode 100644 efw-downloader/src/op-file-parse.c create mode 100644 efw-downloader/src/subcmd-device.c create mode 100644 efw-downloader/src/subcmd-file.c create mode 100644 efw-downloader/src/subcmds.h -- 2.25.1

2 30

[PATCH] ASoC: intel/skl/hda - fix probe regression on systems without i915
by Kai Vehmanen 27 Aug '20

27 Aug '20

Starting in commit cbc7a6b5a87a ("ASoC: soc-card: add snd_soc_card_add_dai_link()"), error value from ASoc add_dai_link() is no longer ignored. The generic HDA machine driver relied on the old semantics to disable i915 HDMI/DP audio codec at runtime. If no display codec was present, add_dai_link() returned an error, but this was ignored and rest of the card was successfully probed. Fix the problem by changing the machine driver add_dai_link() to not return an error in this case. Fixes: cbc7a6b5a87a ("ASoC: soc-card: add snd_soc_card_add_dai_link()") BugLink: https://github.com/thesofproject/linux/issues/2261 Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Rander Wang <rander.wang(a)linux.intel.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> --- Notes: base broonie/for-5.8 at commit 12eb3ad0638c ("ASoC: rt286: fix unexpected interrupt happens") sound/soc/intel/boards/skl_hda_dsp_common.h | 1 + sound/soc/intel/boards/skl_hda_dsp_generic.c | 17 +++++++++++------ 2 files changed, 12 insertions(+), 6 deletions(-) diff --git a/sound/soc/intel/boards/skl_hda_dsp_common.h b/sound/soc/intel/boards/skl_hda_dsp_common.h index 507750ef67f3..4b0b3959182e 100644 --- a/sound/soc/intel/boards/skl_hda_dsp_common.h +++ b/sound/soc/intel/boards/skl_hda_dsp_common.h @@ -33,6 +33,7 @@ struct skl_hda_private { int dai_index; const char *platform_name; bool common_hdmi_codec_drv; + bool idisp_codec; }; extern struct snd_soc_dai_link skl_hda_be_dai_links[HDA_DSP_MAX_BE_DAI_LINKS]; diff --git a/sound/soc/intel/boards/skl_hda_dsp_generic.c b/sound/soc/intel/boards/skl_hda_dsp_generic.c index 79c8947f840b..ca4900036ead 100644 --- a/sound/soc/intel/boards/skl_hda_dsp_generic.c +++ b/sound/soc/intel/boards/skl_hda_dsp_generic.c @@ -79,6 +79,9 @@ skl_hda_add_dai_link(struct snd_soc_card *card, struct snd_soc_dai_link *link) link->platforms->name = ctx->platform_name; link->nonatomic = 1; + if (!ctx->idisp_codec) + return 0; + if (strstr(link->name, "HDMI")) { ret = skl_hda_hdmi_add_pcm(card, ctx->pcm_count); @@ -118,19 +121,20 @@ static char hda_soc_components[30]; static int skl_hda_fill_card_info(struct snd_soc_acpi_mach_params *mach_params) { struct snd_soc_card *card = &hda_soc_card; + struct skl_hda_private *ctx = snd_soc_card_get_drvdata(card); struct snd_soc_dai_link *dai_link; - u32 codec_count, codec_mask, idisp_mask; + u32 codec_count, codec_mask; int i, num_links, num_route; codec_mask = mach_params->codec_mask; codec_count = hweight_long(codec_mask); - idisp_mask = codec_mask & IDISP_CODEC_MASK; + ctx->idisp_codec = !!(codec_mask & IDISP_CODEC_MASK); if (!codec_count || codec_count > 2 || - (codec_count == 2 && !idisp_mask)) + (codec_count == 2 && !ctx->idisp_codec)) return -EINVAL; - if (codec_mask == idisp_mask) { + if (codec_mask == IDISP_CODEC_MASK) { /* topology with iDisp as the only HDA codec */ num_links = IDISP_DAI_COUNT + DMIC_DAI_COUNT; num_route = IDISP_ROUTE_COUNT; @@ -152,7 +156,7 @@ static int skl_hda_fill_card_info(struct snd_soc_acpi_mach_params *mach_params) num_route = ARRAY_SIZE(skl_hda_map); card->dapm_widgets = skl_hda_widgets; card->num_dapm_widgets = ARRAY_SIZE(skl_hda_widgets); - if (!idisp_mask) { + if (!ctx->idisp_codec) { for (i = 0; i < IDISP_DAI_COUNT; i++) { skl_hda_be_dai_links[i].codecs = dummy_codec; skl_hda_be_dai_links[i].num_codecs = @@ -211,6 +215,8 @@ static int skl_hda_audio_probe(struct platform_device *pdev) if (!mach) return -EINVAL; + snd_soc_card_set_drvdata(&hda_soc_card, ctx); + ret = skl_hda_fill_card_info(&mach->mach_params); if (ret < 0) { dev_err(&pdev->dev, "Unsupported HDAudio/iDisp configuration found\n"); @@ -223,7 +229,6 @@ static int skl_hda_audio_probe(struct platform_device *pdev) ctx->common_hdmi_codec_drv = mach->mach_params.common_hdmi_codec_drv; hda_soc_card.dev = &pdev->dev; - snd_soc_card_set_drvdata(&hda_soc_card, ctx); if (mach->mach_params.dmic_num > 0) { snprintf(hda_soc_components, sizeof(hda_soc_components), -- 2.27.0

3 3

[PATCH] ASoC: meson: cards: deal dpcm flag change
by Jerome Brunet 27 Aug '20

27 Aug '20

Commit b73287f0b074 ("ASoC: soc-pcm: dpcm: fix playback/capture checks") changed the meaning of dpcm_playback/dpcm_capture and now requires the CPU DAI BE to aligned with those flags. This broke all Amlogic cards with uni-directional backends (All gx and most axg cards). While I'm still confused as to how this change is an improvement, those cards can't remain broken forever. Hopefully, next time an API change is done like that, all the users will be updated as part of the change, and not left to fend for themselves. Fixes: b73287f0b074 ("ASoC: soc-pcm: dpcm: fix playback/capture checks") Signed-off-by: Jerome Brunet <jbrunet(a)baylibre.com> --- Hi, For the cards be actually fixed, change 'ASoC: core: use less strict tests for dailink capabilities' will need to merged as well. snd_soc_dai_link_set_capabilities() as implemented in 25612477d20b5 ('ASoC: soc-dai: set dai_link dpcm_ flags with a helper') does not support link with multiple codec with different direction capabilities. This quite a common case on these cards. sound/soc/meson/axg-card.c | 18 ++++++++++-------- sound/soc/meson/gx-card.c | 18 +++++++++--------- sound/soc/meson/meson-card-utils.c | 4 ---- 3 files changed, 19 insertions(+), 21 deletions(-) diff --git a/sound/soc/meson/axg-card.c b/sound/soc/meson/axg-card.c index 5176be165210..2b77010c2c5c 100644 --- a/sound/soc/meson/axg-card.c +++ b/sound/soc/meson/axg-card.c @@ -327,20 +327,22 @@ static int axg_card_add_link(struct snd_soc_card *card, struct device_node *np, return ret; if (axg_card_cpu_is_playback_fe(dai_link->cpus->of_node)) - ret = meson_card_set_fe_link(card, dai_link, np, true); + return meson_card_set_fe_link(card, dai_link, np, true); else if (axg_card_cpu_is_capture_fe(dai_link->cpus->of_node)) - ret = meson_card_set_fe_link(card, dai_link, np, false); - else - ret = meson_card_set_be_link(card, dai_link, np); + return meson_card_set_fe_link(card, dai_link, np, false); + + ret = meson_card_set_be_link(card, dai_link, np); if (ret) return ret; - if (axg_card_cpu_is_tdm_iface(dai_link->cpus->of_node)) - ret = axg_card_parse_tdm(card, np, index); - else if (axg_card_cpu_is_codec(dai_link->cpus->of_node)) { + if (axg_card_cpu_is_codec(dai_link->cpus->of_node)) { dai_link->params = &codec_params; - dai_link->no_pcm = 0; /* link is not a DPCM BE */ + } else { + dai_link->no_pcm = 1; + snd_soc_dai_link_set_capabilities(dai_link); + if (axg_card_cpu_is_tdm_iface(dai_link->cpus->of_node)) + ret = axg_card_parse_tdm(card, np, index); } return ret; diff --git a/sound/soc/meson/gx-card.c b/sound/soc/meson/gx-card.c index 6da8535f4dd2..5119434a81c4 100644 --- a/sound/soc/meson/gx-card.c +++ b/sound/soc/meson/gx-card.c @@ -96,21 +96,21 @@ static int gx_card_add_link(struct snd_soc_card *card, struct device_node *np, return ret; if (gx_card_cpu_identify(dai_link->cpus, "FIFO")) - ret = meson_card_set_fe_link(card, dai_link, np, true); - else - ret = meson_card_set_be_link(card, dai_link, np); + return meson_card_set_fe_link(card, dai_link, np, true); + ret = meson_card_set_be_link(card, dai_link, np); if (ret) return ret; - /* Check if the cpu is the i2s encoder and parse i2s data */ - if (gx_card_cpu_identify(dai_link->cpus, "I2S Encoder")) - ret = gx_card_parse_i2s(card, np, index); - /* Or apply codec to codec params if necessary */ - else if (gx_card_cpu_identify(dai_link->cpus, "CODEC CTRL")) { + if (gx_card_cpu_identify(dai_link->cpus, "CODEC CTRL")) { dai_link->params = &codec_params; - dai_link->no_pcm = 0; /* link is not a DPCM BE */ + } else { + dai_link->no_pcm = 1; + snd_soc_dai_link_set_capabilities(dai_link); + /* Check if the cpu is the i2s encoder and parse i2s data */ + if (gx_card_cpu_identify(dai_link->cpus, "I2S Encoder")) + ret = gx_card_parse_i2s(card, np, index); } return ret; diff --git a/sound/soc/meson/meson-card-utils.c b/sound/soc/meson/meson-card-utils.c index f9ce03f3921f..6a64ac01b5ca 100644 --- a/sound/soc/meson/meson-card-utils.c +++ b/sound/soc/meson/meson-card-utils.c @@ -147,10 +147,6 @@ int meson_card_set_be_link(struct snd_soc_card *card, struct device_node *np; int ret, num_codecs; - link->no_pcm = 1; - link->dpcm_playback = 1; - link->dpcm_capture = 1; - num_codecs = of_get_child_count(node); if (!num_codecs) { dev_err(card->dev, "be link %s has no codec\n", -- 2.25.4

3 4

[PATCH 0/8] SOF fixes and updates for FW boot
by Ranjani Sridharan 27 Aug '20

27 Aug '20

This series includes fixes and updates for the FW boot sequence for Intel platforms. Ranjani Sridharan (7): ALSA: hda: fix VS_LTRC register name ASoC: SOF: Intel: hda: Add helper function to program ICCMAX stream ASoC: SOF: Intel: hda: modify the signature of get_stream_with_tag() ASoC: SOF: Intel: hda: define macro for code loader stream format ASoC: SOF: Intel: hda: Define FW boot sequence with ICCMAX ASoC: SOF: Intel: hda: Add sof_tgl_ops for TGL platforms ASoC: SOF: Intel: hda: Simplify error handling during FW boot Yong Zhi (1): ASoC: SOF: Intel: hda: Remove unused parameters in cl_dsp_init() include/sound/hda_register.h | 2 +- sound/soc/sof/intel/Makefile | 2 +- sound/soc/sof/intel/cnl.c | 23 +---- sound/soc/sof/intel/hda-ipc.h | 4 + sound/soc/sof/intel/hda-loader.c | 145 ++++++++++++++++++------------- sound/soc/sof/intel/hda-stream.c | 69 +++++++++++++++ sound/soc/sof/intel/hda.h | 6 ++ sound/soc/sof/intel/tgl.c | 137 +++++++++++++++++++++++++++++ sound/soc/sof/sof-pci-dev.c | 2 +- 9 files changed, 306 insertions(+), 84 deletions(-) create mode 100644 sound/soc/sof/intel/tgl.c -- 2.25.1

3 11

[v2] ASoC: AMD: Clean kernel log from deferred probe error messages
by Akshu Agrawal 27 Aug '20

27 Aug '20

While the driver waits for DAIs to be probed and retries probing, have the error messages at debug level instead of error. Signed-off-by: Akshu Agrawal <akshu.agrawal(a)amd.com> --- v2: Add a debug level message for -EPROBE_DEFER sound/soc/amd/acp3x-rt5682-max9836.c | 11 ++++++++--- 1 file changed, 8 insertions(+), 3 deletions(-) diff --git a/sound/soc/amd/acp3x-rt5682-max9836.c b/sound/soc/amd/acp3x-rt5682-max9836.c index 406526e79af3..1a4e8ca0f99c 100644 --- a/sound/soc/amd/acp3x-rt5682-max9836.c +++ b/sound/soc/amd/acp3x-rt5682-max9836.c @@ -472,12 +472,17 @@ static int acp3x_probe(struct platform_device *pdev) ret = devm_snd_soc_register_card(&pdev->dev, card); if (ret) { - dev_err(&pdev->dev, + if (ret != -EPROBE_DEFER) + dev_err(&pdev->dev, "devm_snd_soc_register_card(%s) failed: %d\n", card->name, ret); - return ret; + else + dev_dbg(&pdev->dev, + "devm_snd_soc_register_card(%s) probe deferred: %d\n", + card->name, ret); } - return 0; + + return ret; } static const struct acpi_device_id acp3x_audio_acpi_match[] = { -- 2.20.1

2 1

[PATCH v3] ASoC: max9867: shutdown codec when changing filter type
by Pavel Dobias 27 Aug '20

27 Aug '20

Changing filter type without disabling codec results in filter malfunction. Disable codec when changing filter type. Signed-off-by: Pavel Dobias <dobias(a)2n.cz> --- Changes in v2: - don't allow change if component active - avoid codec shutdown in max9867_filter_set if value not changed Changes in v3: - don't allow change if ADC/DAC active, not whole component - ternary operator rewritten to normal condition sound/soc/codecs/max9867.c | 131 +++++++++++++++++++++++++++++++++---- sound/soc/codecs/max9867.h | 4 +- 2 files changed, 120 insertions(+), 15 deletions(-) diff --git a/sound/soc/codecs/max9867.c b/sound/soc/codecs/max9867.c index fcb31144d69c..aef2746bfb94 100644 --- a/sound/soc/codecs/max9867.c +++ b/sound/soc/codecs/max9867.c @@ -15,6 +15,14 @@ #include <sound/tlv.h> #include "max9867.h" +struct max9867_priv { + struct regmap *regmap; + const struct snd_pcm_hw_constraint_list *constraints; + unsigned int sysclk, pclk; + bool master, dsp_a; + unsigned int adc_dac_active; +}; + static const char *const max9867_spmode[] = { "Stereo Diff", "Mono Diff", "Stereo Cap", "Mono Cap", @@ -32,8 +40,102 @@ static const char *const max9867_adc_dac_filter_text[] = { "Butterworth/8-24" }; -static SOC_ENUM_SINGLE_DECL(max9867_filter, MAX9867_CODECFLTR, 7, - max9867_filter_text); +enum max9867_adc_dac { + MAX9867_ADC_LEFT, + MAX9867_ADC_RIGHT, + MAX9867_DAC_LEFT, + MAX9867_DAC_RIGHT, +}; + +static int max9867_adc_dac_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); + struct max9867_priv *max9867 = snd_soc_component_get_drvdata(component); + enum max9867_adc_dac adc_dac; + + if (!strcmp(w->name, "ADCL")) + adc_dac = MAX9867_ADC_LEFT; + else if (!strcmp(w->name, "ADCR")) + adc_dac = MAX9867_ADC_RIGHT; + else if (!strcmp(w->name, "DACL")) + adc_dac = MAX9867_DAC_LEFT; + else if (!strcmp(w->name, "DACR")) + adc_dac = MAX9867_DAC_RIGHT; + else + return 0; + + if (SND_SOC_DAPM_EVENT_ON(event)) + max9867->adc_dac_active |= BIT(adc_dac); + else if (SND_SOC_DAPM_EVENT_OFF(event)) + max9867->adc_dac_active &= ~BIT(adc_dac); + + return 0; +} + +static int max9867_filter_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); + struct max9867_priv *max9867 = snd_soc_component_get_drvdata(component); + unsigned int reg; + int ret; + + ret = regmap_read(max9867->regmap, MAX9867_CODECFLTR, &reg); + if (ret) + return -EINVAL; + + if (reg & MAX9867_CODECFLTR_MODE) + ucontrol->value.enumerated.item[0] = 1; + else + ucontrol->value.enumerated.item[0] = 0; + + return 0; +} + +static int max9867_filter_set(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); + struct max9867_priv *max9867 = snd_soc_component_get_drvdata(component); + unsigned int reg, mode = ucontrol->value.enumerated.item[0]; + int ret; + + if (mode > 1) + return -EINVAL; + + /* don't allow change if ADC/DAC active */ + if (max9867->adc_dac_active) + return -EBUSY; + + /* read current filter mode */ + ret = regmap_read(max9867->regmap, MAX9867_CODECFLTR, &reg); + if (ret) + return -EINVAL; + + if (mode) + mode = MAX9867_CODECFLTR_MODE; + + /* check if change is needed */ + if ((reg & MAX9867_CODECFLTR_MODE) == mode) + return 0; + + /* shutdown codec before switching filter mode */ + regmap_update_bits(max9867->regmap, MAX9867_PWRMAN, + MAX9867_PWRMAN_SHDN, 0); + + /* switch filter mode */ + regmap_update_bits(max9867->regmap, MAX9867_CODECFLTR, + MAX9867_CODECFLTR_MODE, mode); + + /* out of shutdown now */ + regmap_update_bits(max9867->regmap, MAX9867_PWRMAN, + MAX9867_PWRMAN_SHDN, MAX9867_PWRMAN_SHDN); + + return 0; +} + +static SOC_ENUM_SINGLE_EXT_DECL(max9867_filter, max9867_filter_text); static SOC_ENUM_SINGLE_DECL(max9867_dac_filter, MAX9867_CODECFLTR, 0, max9867_adc_dac_filter_text); static SOC_ENUM_SINGLE_DECL(max9867_adc_filter, MAX9867_CODECFLTR, 4, @@ -76,7 +178,7 @@ static const struct snd_kcontrol_new max9867_snd_controls[] = { SOC_ENUM("Speaker Mode", max9867_spkmode), SOC_SINGLE("Volume Smoothing Switch", MAX9867_MODECONFIG, 6, 1, 0), SOC_SINGLE("Line ZC Switch", MAX9867_MODECONFIG, 5, 1, 0), - SOC_ENUM("DSP Filter", max9867_filter), + SOC_ENUM_EXT("DSP Filter", max9867_filter, max9867_filter_get, max9867_filter_set), SOC_ENUM("ADC Filter", max9867_adc_filter), SOC_ENUM("DAC Filter", max9867_dac_filter), SOC_SINGLE("Mono Playback Switch", MAX9867_IFC1B, 3, 1, 0), @@ -134,8 +236,12 @@ static const struct snd_soc_dapm_widget max9867_dapm_widgets[] = { &max9867_left_dmic_mux), SND_SOC_DAPM_MUX("DMICR Mux", SND_SOC_NOPM, 0, 0, &max9867_right_dmic_mux), - SND_SOC_DAPM_ADC("ADCL", "HiFi Capture", SND_SOC_NOPM, 0, 0), - SND_SOC_DAPM_ADC("ADCR", "HiFi Capture", SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_ADC_E("ADCL", "HiFi Capture", SND_SOC_NOPM, 0, 0, + max9867_adc_dac_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_ADC_E("ADCR", "HiFi Capture", SND_SOC_NOPM, 0, 0, + max9867_adc_dac_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_MIXER("Digital", SND_SOC_NOPM, 0, 0, max9867_sidetone_mixer_controls, @@ -143,8 +249,12 @@ static const struct snd_soc_dapm_widget max9867_dapm_widgets[] = { SND_SOC_DAPM_MIXER_NAMED_CTL("Output Mixer", SND_SOC_NOPM, 0, 0, max9867_output_mixer_controls, ARRAY_SIZE(max9867_output_mixer_controls)), - SND_SOC_DAPM_DAC("DACL", "HiFi Playback", SND_SOC_NOPM, 0, 0), - SND_SOC_DAPM_DAC("DACR", "HiFi Playback", SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC_E("DACL", "HiFi Playback", SND_SOC_NOPM, 0, 0, + max9867_adc_dac_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_DAC_E("DACR", "HiFi Playback", SND_SOC_NOPM, 0, 0, + max9867_adc_dac_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_SWITCH("Master Playback", SND_SOC_NOPM, 0, 0, &max9867_line_out_control), SND_SOC_DAPM_OUTPUT("LOUT"), @@ -197,13 +307,6 @@ static const struct snd_pcm_hw_constraint_list max9867_constraints_48k = { .count = ARRAY_SIZE(max9867_rates_48k), }; -struct max9867_priv { - struct regmap *regmap; - const struct snd_pcm_hw_constraint_list *constraints; - unsigned int sysclk, pclk; - bool master, dsp_a; -}; - static int max9867_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { diff --git a/sound/soc/codecs/max9867.h b/sound/soc/codecs/max9867.h index 3092c3b99075..b6b880631b13 100644 --- a/sound/soc/codecs/max9867.h +++ b/sound/soc/codecs/max9867.h @@ -44,7 +44,8 @@ #define MAX9867_IFC1B_PCLK_4 0x05 #define MAX9867_IFC1B_PCLK_8 0x06 #define MAX9867_IFC1B_PCLK_16 0x07 -#define MAX9867_CODECFLTR 0x0a +#define MAX9867_CODECFLTR 0x0a +#define MAX9867_CODECFLTR_MODE (1<<7) #define MAX9867_SIDETONE 0x0b #define MAX9867_DACLEVEL 0x0c #define MAX9867_ADCLEVEL 0x0d @@ -58,6 +59,7 @@ #define MAX9867_MICCONFIG 0x15 #define MAX9867_MODECONFIG 0x16 #define MAX9867_PWRMAN 0x17 +#define MAX9867_PWRMAN_SHDN (1<<7) #define MAX9867_REVISION 0xff #define MAX9867_CACHEREGNUM 10 -- 2.20.1

2 1

Re: WARNING: ODEBUG bug in get_signal
by syzbot 27 Aug '20

27 Aug '20

syzbot has bisected this issue to: commit a9ed4a6560b8562b7e2e2bed9527e88001f7b682 Author: Marc Zyngier <maz(a)kernel.org> Date: Wed Aug 19 16:12:17 2020 +0000 epoll: Keep a reference on files added to the check list bisection log: https://syzkaller.appspot.com/x/bisect.txt?x=15e57751900000 start commit: 15bc20c6 Merge tag 'tty-5.9-rc3' of git://git.kernel.org/p.. git tree: upstream final oops: https://syzkaller.appspot.com/x/report.txt?x=17e57751900000 console output: https://syzkaller.appspot.com/x/log.txt?x=13e57751900000 kernel config: https://syzkaller.appspot.com/x/.config?x=978db74cb30aa994 dashboard link: https://syzkaller.appspot.com/bug?extid=e3cf8f93cf86936710db userspace arch: i386 syz repro: https://syzkaller.appspot.com/x/repro.syz?x=13236eb6900000 Reported-by: syzbot+e3cf8f93cf86936710db(a)syzkaller.appspotmail.com Fixes: a9ed4a6560b8 ("epoll: Keep a reference on files added to the check list") For information about bisection process see: https://goo.gl/tpsmEJ#bisection

1 0

[PATCH v4 00/13] ASoC: Intel: Catpt - Lynx and Wildcat point
by Cezary Rojewski 27 Aug '20

27 Aug '20

Implement support for Lynxpoint and Wildcat Point AudioDSP. Catpt solution deprecates existing sound/soc/intel/haswell which is removed in the following series. This cover-letter is followed by 'Developer's deep dive' message schedding light on catpt's key concepts and areas addressed. Due to high range of errors and desynchronization from recommendations set by Windows solution, re-write came as a lower-cost solution compared to refactoring /haswell/ with several series of patches. Special thanks go to Marcin Barlik and Piotr Papierkowski for sharing their LPT/WPT AudioDSP architecture expertise as well as helping backtrack its historical background. My thanks go to Amadeusz Slawinski for reviews and improvements proposed on and off the internal list. Most of internal diff below is his contribution. Krzysztof Hejmowski helped me setup my own Xtensa environment and recompile LPT/WPT FW binary sources what sped up the development greatly. This would not have been possible without help from these champions, especially considering how quickly the catpt was written: 2 weeks features, 3 weeks optimizations. Thank you. Userspace-exposed members are compatible with what is exposed by deprecated solution as well as FW binary being re-used thus no harm is done. The only visible differences are: the newly added 'Loopback Mute' kcontrol and volume support extending to quad from stereo. On top of fixing erros and design flows, catpt also adds module reload, dynamic SRAM memory allocation during PCM runtime and exposes missing userspace API: 'Loopback Mute' kcontrol, quad volume controls and sysfs fw-version entries. Event tracing is provided to easy solution debugging. Following are not included in this update and are scheduled as later addition: - fw logging - module (library) support Note: LPT power up/down sequences might get aligned with WPT once enough testing is done as capabilities are shared for both DSPs. Note #2: Both LPT and WPT power up/down sequences may get optimized in future updates as thanks to help from the Windows team, most of nuances behind why/what/when in regard to hw registers have been backtracked and reviewed again. Link to developer's deep dive message: https://www.spinics.net/lists/alsa-devel/msg113563.html Changes in v4: - fixed compilation with i386 kconfig (conflicting names) - streamlined naming for SHIM and PCI registers to match SSP ones (SHIM_REG -> SHIM) - catpt_component_probe removed and kcontrols again initializzed statically via snd_kcontrol_new array: this is to remove kctl->id.device shenanigans - renamed catpt_set_ctlvol to catpt_set_dspvol - function name wasn't matching its purpose Changes in v3: - fixed IRAM mask usage in lpt_dsp_power_up (dsp.c) - updated dbg message formatting in catpt_restore_fwimage as suggested by Andy - fixed alignment for struct catpt_ssp_device_format - catpt_set_ctlvol now verifies all-equal scenario based on all channels rather than just first two as requested by Amadeo - fixed SPDX for registers.h Changes in v2: - fixed SPDX formatting for all header files as well as pcm.c - fixed size used when memcping fs.c::fw_build_read as reported by Mark - renamed struct catpt_pdata to struct catpt_spec (cosmetic) - fixed erroneous path in catpt_load_block: region is properly released - trace.h events for updating registers have been removed and usages replaced by dev_dbg (SRAMPGE/ LPCS) - as requested by Andy, struct resource has replaced struct catpt_mbank and struct catpt_mregion. This change cascaded into: - catpt_mbank_size and catpt_mregion_size replaced by resource_size - catpt_mregion_intersects replaced by resource_overlaps - all catpt_mbank_ and catpt_mregion_ handlers found in loader.c (_request, _reserve, _release, _extract, _split, _join) have been removed - __request_region and __release_region have been enlisted in their place - catpt_mregion_intersecting renamed to catpt_resource_overlapping - catpt_request_region helper has been provided to deal with -size based requests o haven't found direct replacements in resource.c/ ioport.h for both functions - catpt_mbank_create and catpt_mbank_remove renamed to catpt_sram_init and catpt_sram_free respectively - catpt_sram_init now returns void instead of int and has been converted to simple initialized. This change ultimately cascaded into: o both SRAM banks initialization being moved to catpt_dev_init from catpt_acpi_probe (device.c) o catpt_dev::spec is now initialized first, with catpt_dev_init following it soon after o catpt_acpi_probe erroneous path has been simplified as SRAM banks no longer need to be freed - catpt_sram_free now frees all resources via child -> sibling enumeration rather than region_list iteration - catpt_dsp_update_srampge and catpt_dsp_set_srampge now accept new argument: unsigned long mask. Caused by removal of catpt_mbank - mask is taken directly from catpt_dev::spec::d/iram_mask - trace.h events for catpt_mbank and catpt_mregion have been removed Diff against last drop on internal list: - replaced spinlock with mutex for mregion allocation and release to address sleeping in atomic context warnings - fixed coredump fw_hash dumping - kcontrol values are now always stored regardless of stream of interest is running or not - kcontrol values are now applied after stream is prepared instead of ignoring what has been set by user initially - catpt_pdata instances have been renamed from hsw_ and bdw_ to lpt_ and wpt_ respectively - reordered Makefile .o(s) (cosmetic) Cezary Rojewski (13): ASoC: Intel: Add catpt device ASoC: Intel: catpt: Define DSP operations ASoC: Intel: catpt: Firmware loading and context restore ASoC: Intel: catpt: Implement IPC protocol ASoC: Intel: catpt: Add IPC messages ASoC: Intel: catpt: PCM operations ASoC: Intel: catpt: Event tracing ASoC: Intel: catpt: Simple sysfs attributes ASoC: Intel: Select catpt and deprecate haswell ASoC: Intel: haswell: Remove haswell-solution specific code ASoC: Intel: broadwell: Remove haswell-solution specific code ASoC: Intel: bdw-5650: Remove haswell-solution specific code ASoC: Intel: bdw-5677: Remove haswell-solution specific code sound/soc/intel/Kconfig | 22 +- sound/soc/intel/Makefile | 2 +- sound/soc/intel/boards/Kconfig | 8 +- sound/soc/intel/boards/bdw-rt5650.c | 36 - sound/soc/intel/boards/bdw-rt5677.c | 33 - sound/soc/intel/boards/broadwell.c | 33 - sound/soc/intel/boards/haswell.c | 28 +- sound/soc/intel/catpt/Makefile | 6 + sound/soc/intel/catpt/core.h | 187 +++++ sound/soc/intel/catpt/device.c | 376 +++++++++ sound/soc/intel/catpt/dsp.c | 596 +++++++++++++ sound/soc/intel/catpt/fs.c | 79 ++ sound/soc/intel/catpt/ipc.c | 298 +++++++ sound/soc/intel/catpt/loader.c | 673 +++++++++++++++ sound/soc/intel/catpt/messages.c | 312 +++++++ sound/soc/intel/catpt/messages.h | 401 +++++++++ sound/soc/intel/catpt/pcm.c | 1212 +++++++++++++++++++++++++++ sound/soc/intel/catpt/registers.h | 191 +++++ sound/soc/intel/catpt/trace.h | 83 ++ 19 files changed, 4434 insertions(+), 142 deletions(-) create mode 100644 sound/soc/intel/catpt/Makefile create mode 100644 sound/soc/intel/catpt/core.h create mode 100644 sound/soc/intel/catpt/device.c create mode 100644 sound/soc/intel/catpt/dsp.c create mode 100644 sound/soc/intel/catpt/fs.c create mode 100644 sound/soc/intel/catpt/ipc.c create mode 100644 sound/soc/intel/catpt/loader.c create mode 100644 sound/soc/intel/catpt/messages.c create mode 100644 sound/soc/intel/catpt/messages.h create mode 100644 sound/soc/intel/catpt/pcm.c create mode 100644 sound/soc/intel/catpt/registers.h create mode 100644 sound/soc/intel/catpt/trace.h -- 2.17.1

5 42

[PATCH 0/5] Qualcomm's lpass-hdmi ASoC driver to support audio over dp port
by Srinivasa Rao Mandadapu 27 Aug '20

27 Aug '20

These patches are to support audio over DP port on Qualcomm's SC7180 LPASS Asoc. It includes machine driver, cpu driver, platform driver updates for HDMI path support, device tree documention, lpass variant structure optimization and configuration changes. These patches depends on the DP patch series https://patchwork.kernel.org/project/dri-devel/list/?series=332029 V Sujith Kumar Reddy (5): ASoC: Add sc7180-lpass binding header hdmi define ASoC: dt-bindings: Add dt binding for lpass hdmi ASoC: qcom: Add support for lpass hdmi driver ASoC: qcom: Add support for audio over DP ASoC: qcom: Optimise lpass variant structure .../devicetree/bindings/sound/qcom,lpass-cpu.yaml | 51 +- include/dt-bindings/sound/sc7180-lpass.h | 1 + sound/soc/qcom/Kconfig | 5 + sound/soc/qcom/Makefile | 2 + sound/soc/qcom/lpass-apq8016.c | 25 +- sound/soc/qcom/lpass-cpu.c | 92 ++- sound/soc/qcom/lpass-hdmi.c | 685 +++++++++++++++++++++ sound/soc/qcom/lpass-hdmi.h | 129 ++++ sound/soc/qcom/lpass-ipq806x.c | 25 +- sound/soc/qcom/lpass-lpaif-reg.h | 51 +- sound/soc/qcom/lpass-platform.c | 287 +++++++-- sound/soc/qcom/lpass-sc7180.c | 147 ++++- sound/soc/qcom/lpass.h | 123 +++- 13 files changed, 1472 insertions(+), 151 deletions(-) create mode 100644 sound/soc/qcom/lpass-hdmi.c create mode 100644 sound/soc/qcom/lpass-hdmi.h -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

1 0

[PATCH v2 1/5] ASoC: Add sc7180-lpass binding header hdmi define
by Srinivasa Rao Mandadapu 27 Aug '20

27 Aug '20

From: V Sujith Kumar Reddy <vsujithk(a)codeaurora.org> Add header defining hdmi dai-id for SC7180 lpass soc in dt bindings. Signed-off-by: Srinivasa Rao <srivasam(a)codeaurora.org> Signed-off-by: V Sujith Kumar Reddy <vsujithk(a)codeaurora.org> --- include/dt-bindings/sound/sc7180-lpass.h | 1 + 1 file changed, 1 insertion(+) diff --git a/include/dt-bindings/sound/sc7180-lpass.h b/include/dt-bindings/sound/sc7180-lpass.h index 7d988f6..0b4991e 100644 --- a/include/dt-bindings/sound/sc7180-lpass.h +++ b/include/dt-bindings/sound/sc7180-lpass.h @@ -4,6 +4,7 @@ #define MI2S_PRIMARY 0 #define MI2S_SECONDARY 1 +#define HDMI 2 #define LPASS_MCLK0 0 -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

1 4

[PATCH 0/2] do exception handling appropriately in imx_es8328_probe()
by Yu Kuai 27 Aug '20

27 Aug '20

Yu Kuai (2): ASoC: fsl: imx-es8328: add missing kfree() call in imx_es8328_probe() ASoC: fsl: imx-es8328: add missing put_device() call in imx_es8328_probe() sound/soc/fsl/imx-es8328.c | 16 +++++++++++----- 1 file changed, 11 insertions(+), 5 deletions(-) -- 2.25.4

4 5

[PATCH v2 0/7] ASoC: merge soc_pcm_open() rollback and soc_pcm_close()
by Kuninori Morimoto 27 Aug '20

27 Aug '20

Hi Mark This is v2 patch-set. soc_pcm_open() does rollback when failed (A), but, it is almost same as soc_pcm_close(). static int soc_pcm_open(xxx) { ... if (ret < 0) goto xxx_err; ... return 0; ^ config_err: | ... | rtd_startup_err: (A) ... | component_err: | ... v return ret; } This kind of duplicated code can be a hotbed of bugs, thus, this patch-set share soc_pcm_close() and rollback. v1 -> v2 - indicate more detail background/logic on git-log Link: https://lore.kernel.org/r/87wo2oku0m.wl-kuninori.morimoto.gx@renesas.com Kuninori Morimoto (7): ASoC: soc-dai: add mark for snd_soc_dai_startup/shutdown() ASoC: soc-link: add mark for snd_soc_link_startup/shutdown() ASoC: soc-component: add mark for soc_pcm_components_open/close() ASoC: soc-component: add mark for snd_soc_pcm_component_pm_runtime_get/put() ASoC: soc-pcm: add soc_pcm_clean() and call it from soc_pcm_open/close() ASoC: soc-pcm: remove unneeded dev_err() for snd_soc_dai_startup() ASoC: soc-pcm: remove unneeded dev_err() for snd_soc_component_module/open() include/sound/soc-component.h | 28 +++++--- include/sound/soc-dai.h | 5 +- include/sound/soc-link.h | 3 +- include/sound/soc.h | 3 + sound/soc/soc-component.c | 73 ++++++++++++++++++++- sound/soc/soc-compress.c | 30 +++------ sound/soc/soc-dai.c | 21 +++++- sound/soc/soc-dapm.c | 4 +- sound/soc/soc-link.c | 21 +++++- sound/soc/soc-pcm.c | 120 ++++++++++++---------------------- 10 files changed, 190 insertions(+), 118 deletions(-) -- 2.25.1

1 7

[PATCH 0/2] ASoC: dt-bindings: audio-graph-card: switch to yaml base Documentation
by Kuninori Morimoto 27 Aug '20

27 Aug '20

Hi Rob, Mark These patches switch to yaml base Documentation for audio-graph-card, and renesas,rsnd which uses audio-graph-card. Kuninori Morimoto (2): ASoC: dt-bindings: audio-graph-card: switch to yaml base Documentation ASoC: dt-bindings: renesas,rsnd: switch to yaml base Documentation .../bindings/sound/audio-graph-card.txt | 337 ------------ .../bindings/sound/audio-graph-card.yaml | 135 +++++ .../bindings/sound/renesas,rsnd.txt | 520 ------------------ .../bindings/sound/renesas,rsnd.yaml | 454 +++++++++++++++ .../devicetree/bindings/vendor-prefixes.yaml | 2 +- 5 files changed, 590 insertions(+), 858 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/audio-graph-card.txt create mode 100644 Documentation/devicetree/bindings/sound/audio-graph-card.yaml create mode 100644 Documentation/devicetree/bindings/sound/renesas,rsnd.yaml -- 2.25.1

2 4

[PATCH] ASoC: soc-core: move snd_soc_dai_link_set_capabilities() to soc-core.c
by Kuninori Morimoto 27 Aug '20

27 Aug '20

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> commit 25612477d20b52 ("ASoC: soc-dai: set dai_link dpcm_ flags with a helper") added snd_soc_dai_link_set_capabilities(). But it is using snd_soc_find_dai() (A) which is required client_mutex (B). And client_mutex is soc-core.c local. struct snd_soc_dai *snd_soc_find_dai(xxx) { ... (B) lockdep_assert_held(&client_mutex); ... } void snd_soc_dai_link_set_capabilities(xxx) { ... for_each_pcm_streams(direction) { ... for_each_link_cpus(dai_link, i, cpu) { (A) dai = snd_soc_find_dai(cpu); ... } ... for_each_link_codecs(dai_link, i, codec) { (A) dai = snd_soc_find_dai(codec); ... } } ... } Because of this, we will get WARNING. WARNING: CPU: 2 PID: 53 at sound/soc/soc-core.c:814 snd_soc_find_dai+0xf8/0x100 CPU: 2 PID: 53 Comm: kworker/2:1 Not tainted 5.7.0-rc1+ #328 Hardware name: Renesas H3ULCB Kingfisher board based on r8a77951 (DT) Workqueue: events deferred_probe_work_func pstate: 60000005 (nZCv daif -PAN -UAO) pc : snd_soc_find_dai+0xf8/0x100 lr : snd_soc_find_dai+0xf4/0x100 ... Call trace: snd_soc_find_dai+0xf8/0x100 snd_soc_dai_link_set_capabilities+0xa0/0x16c graph_dai_link_of_dpcm+0x390/0x3c0 graph_for_each_link+0x134/0x200 graph_probe+0x144/0x230 platform_drv_probe+0x5c/0xb0 really_probe+0xe4/0x430 driver_probe_device+0x60/0xf4 Because not only snd_soc_dai_link_set_capabilities() but many drivers are already using snd_soc_find_dai(), 1st idea for this issue was creating snd_soc_find_dai() of _with_mutex / _without_mutex, but it was not enough and/or confusable. soc-dai.c which is implemented snd_soc_dai_link_set_capabilities() currently is mainly for snd_soc_dai related functions, not for snd_soc_dai_link. Thus as 2nd idea, this patch moves snd_soc_dai_link_set_capabilities() from soc-dai.c to soc-core.c, and use mutex_lock(). Fixes: 25612477d20b52 ("ASoC: soc-dai: set dai_link dpcm_ flags with a helper") Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/soc-core.c | 41 +++++++++++++++++++++++++++++++++++++++++ sound/soc/soc-dai.c | 38 -------------------------------------- 2 files changed, 41 insertions(+), 38 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index fe23e936e2d1..f46979f8d965 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -344,6 +344,47 @@ struct snd_soc_component *snd_soc_lookup_component(struct device *dev, } EXPORT_SYMBOL_GPL(snd_soc_lookup_component); +/* + * snd_soc_dai_link_set_capabilities() - set dai_link properties based on its DAIs + */ +void snd_soc_dai_link_set_capabilities(struct snd_soc_dai_link *dai_link) +{ + struct snd_soc_dai_link_component *cpu; + struct snd_soc_dai_link_component *codec; + struct snd_soc_dai *dai; + bool supported[SNDRV_PCM_STREAM_LAST + 1]; + int direction; + int i; + + mutex_lock(&client_mutex); + + for_each_pcm_streams(direction) { + supported[direction] = true; + + for_each_link_cpus(dai_link, i, cpu) { + dai = snd_soc_find_dai(cpu); + if (!dai || !snd_soc_dai_stream_valid(dai, direction)) { + supported[direction] = false; + break; + } + } + if (!supported[direction]) + continue; + for_each_link_codecs(dai_link, i, codec) { + dai = snd_soc_find_dai(codec); + if (!dai || !snd_soc_dai_stream_valid(dai, direction)) { + supported[direction] = false; + break; + } + } + } + mutex_unlock(&client_mutex); + + dai_link->dpcm_playback = supported[SNDRV_PCM_STREAM_PLAYBACK]; + dai_link->dpcm_capture = supported[SNDRV_PCM_STREAM_CAPTURE]; +} +EXPORT_SYMBOL_GPL(snd_soc_dai_link_set_capabilities); + struct snd_soc_pcm_runtime *snd_soc_get_pcm_runtime(struct snd_soc_card *card, struct snd_soc_dai_link *dai_link) diff --git a/sound/soc/soc-dai.c b/sound/soc/soc-dai.c index 693893420bf0..783b7928205e 100644 --- a/sound/soc/soc-dai.c +++ b/sound/soc/soc-dai.c @@ -393,44 +393,6 @@ bool snd_soc_dai_stream_valid(struct snd_soc_dai *dai, int dir) return stream->channels_min; } -/* - * snd_soc_dai_link_set_capabilities() - set dai_link properties based on its DAIs - */ -void snd_soc_dai_link_set_capabilities(struct snd_soc_dai_link *dai_link) -{ - struct snd_soc_dai_link_component *cpu; - struct snd_soc_dai_link_component *codec; - struct snd_soc_dai *dai; - bool supported[SNDRV_PCM_STREAM_LAST + 1]; - int direction; - int i; - - for_each_pcm_streams(direction) { - supported[direction] = true; - - for_each_link_cpus(dai_link, i, cpu) { - dai = snd_soc_find_dai(cpu); - if (!dai || !snd_soc_dai_stream_valid(dai, direction)) { - supported[direction] = false; - break; - } - } - if (!supported[direction]) - continue; - for_each_link_codecs(dai_link, i, codec) { - dai = snd_soc_find_dai(codec); - if (!dai || !snd_soc_dai_stream_valid(dai, direction)) { - supported[direction] = false; - break; - } - } - } - - dai_link->dpcm_playback = supported[SNDRV_PCM_STREAM_PLAYBACK]; - dai_link->dpcm_capture = supported[SNDRV_PCM_STREAM_CAPTURE]; -} -EXPORT_SYMBOL_GPL(snd_soc_dai_link_set_capabilities); - void snd_soc_dai_action(struct snd_soc_dai *dai, int stream, int action) { -- 2.25.1

2 3

[PATCH] ASoC: qcom: common: Fix refcount imbalance on error
by Dinghao Liu 26 Aug '20

26 Aug '20

for_each_child_of_node returns a node pointer np with refcount incremented. So when devm_kzalloc fails, a pairing refcount decrement is needed to keep np's refcount balanced. Fixes: 16395ceee11f8 ("ASoC: qcom: common: Fix NULL pointer in of parser") Signed-off-by: Dinghao Liu <dinghao.liu(a)zju.edu.cn> --- sound/soc/qcom/common.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/qcom/common.c b/sound/soc/qcom/common.c index 5194d90ddb96..fd69cf8b1f23 100644 --- a/sound/soc/qcom/common.c +++ b/sound/soc/qcom/common.c @@ -52,8 +52,10 @@ int qcom_snd_parse_of(struct snd_soc_card *card) for_each_child_of_node(dev->of_node, np) { dlc = devm_kzalloc(dev, 2 * sizeof(*dlc), GFP_KERNEL); - if (!dlc) - return -ENOMEM; + if (!dlc) { + ret = -ENOMEM; + goto err; + } link->cpus = &dlc[0]; link->platforms = &dlc[1]; -- 2.17.1

2 1

[PATCH v2 0/5] ASoC: Fix return check for devm_regmap_init_sdw()
by Vinod Koul 26 Aug '20

26 Aug '20

This series attempts to fix return check for devm_regmap_init_sdw() Changes in v2: - Add missing patch for rt711 - Add patch for rt700 Vinod Koul (5): ASoC: max98373: Fix return check for devm_regmap_init_sdw() ASoC: rt1308-sdw: Fix return check for devm_regmap_init_sdw() ASoC: rt711: Fix return check for devm_regmap_init_sdw() ASoC: rt715: Fix return check for devm_regmap_init_sdw() ASoC: rt700: Fix return check for devm_regmap_init_sdw() sound/soc/codecs/max98373-sdw.c | 4 ++-- sound/soc/codecs/rt1308-sdw.c | 4 ++-- sound/soc/codecs/rt700-sdw.c | 4 ++-- sound/soc/codecs/rt711-sdw.c | 4 ++-- sound/soc/codecs/rt715-sdw.c | 4 ++-- 5 files changed, 10 insertions(+), 10 deletions(-) -- 2.26.2

3 7

[PATCH] ALSA: hda/hdmi: always check pin power status in i915 pin fixup
by Kai Vehmanen 26 Aug '20

26 Aug '20

When system is suspended with active audio playback to HDMI/DP, two alternative sequences can happen at resume: a) monitor is detected first and ALSA prepare follows normal stream setup sequence, or b) ALSA prepare is called first, but monitor is not yet detected, so PCM is restarted without a pin, In case of (b), on i915 systems, haswell_verify_D0() is not called at resume and the pin power state may be incorrect. Result is lack of audio after resume with no error reported back to user-space. Fix the problem by always verifying converter and pin state in the i915_pin_cvt_fixup(). BugLink: https://github.com/thesofproject/linux/issues/2388 Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> --- sound/pci/hda/patch_hdmi.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 3259d713ace9..1e1b13eb7829 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -2794,6 +2794,7 @@ static void i915_pin_cvt_fixup(struct hda_codec *codec, hda_nid_t cvt_nid) { if (per_pin) { + haswell_verify_D0(codec, per_pin->cvt_nid, per_pin->pin_nid); snd_hda_set_dev_select(codec, per_pin->pin_nid, per_pin->dev_id); intel_verify_pin_cvt_connect(codec, per_pin); -- 2.27.0

2 1

[PATCH] ASoC: AMD: Clean kernel log from deferred probe error messages
by Akshu Agrawal 26 Aug '20

26 Aug '20

While the driver waits for DAIs to be probed and retries probing, avoid printing error messages. Signed-off-by: Akshu Agrawal <akshu.agrawal(a)amd.com> --- sound/soc/amd/acp3x-rt5682-max9836.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) diff --git a/sound/soc/amd/acp3x-rt5682-max9836.c b/sound/soc/amd/acp3x-rt5682-max9836.c index 406526e79af3..67f80ba51de0 100644 --- a/sound/soc/amd/acp3x-rt5682-max9836.c +++ b/sound/soc/amd/acp3x-rt5682-max9836.c @@ -471,13 +471,11 @@ static int acp3x_probe(struct platform_device *pdev) } ret = devm_snd_soc_register_card(&pdev->dev, card); - if (ret) { + if (ret && ret != -EPROBE_DEFER) dev_err(&pdev->dev, "devm_snd_soc_register_card(%s) failed: %d\n", card->name, ret); - return ret; - } - return 0; + return ret; } static const struct acpi_device_id acp3x_audio_acpi_match[] = { -- 2.20.1

3 2

[PATCH v2] ASoC: max9867: shutdown codec when changing filter type
by Pavel Dobias 26 Aug '20

26 Aug '20

Changing filter type without disabling codec results in filter malfunction. Disable codec when changing filter type. Signed-off-by: Pavel Dobias <dobias(a)2n.cz> --- Changes in v2: - don't allow change if component active - avoid codec shutdown in max9867_filter_set if value not changed sound/soc/codecs/max9867.c | 80 +++++++++++++++++++++++++++++++++----- sound/soc/codecs/max9867.h | 4 +- 2 files changed, 73 insertions(+), 11 deletions(-) diff --git a/sound/soc/codecs/max9867.c b/sound/soc/codecs/max9867.c index fcb31144d69c..749de1ab1093 100644 --- a/sound/soc/codecs/max9867.c +++ b/sound/soc/codecs/max9867.c @@ -15,6 +15,13 @@ #include <sound/tlv.h> #include "max9867.h" +struct max9867_priv { + struct regmap *regmap; + const struct snd_pcm_hw_constraint_list *constraints; + unsigned int sysclk, pclk; + bool master, dsp_a; +}; + static const char *const max9867_spmode[] = { "Stereo Diff", "Mono Diff", "Stereo Cap", "Mono Cap", @@ -32,8 +39,68 @@ static const char *const max9867_adc_dac_filter_text[] = { "Butterworth/8-24" }; -static SOC_ENUM_SINGLE_DECL(max9867_filter, MAX9867_CODECFLTR, 7, - max9867_filter_text); +static int max9867_filter_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); + struct max9867_priv *max9867 = snd_soc_component_get_drvdata(component); + unsigned int reg; + int ret; + + ret = regmap_read(max9867->regmap, MAX9867_CODECFLTR, &reg); + if (ret) + return -EINVAL; + + if (reg & MAX9867_CODECFLTR_MODE) + ucontrol->value.enumerated.item[0] = 1; + else + ucontrol->value.enumerated.item[0] = 0; + + return 0; +} + +static int max9867_filter_set(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); + struct max9867_priv *max9867 = snd_soc_component_get_drvdata(component); + unsigned int reg, mode = ucontrol->value.enumerated.item[0]; + int ret; + + /* don't allow change if component active */ + if (snd_soc_component_active(component)) { + return -EBUSY; + } + + if (mode > 1) + return -EINVAL; + mode = mode ? MAX9867_CODECFLTR_MODE : 0; + + /* read current filter mode */ + ret = regmap_read(max9867->regmap, MAX9867_CODECFLTR, &reg); + if (ret) + return -EINVAL; + + /* check if change is needed */ + if ((reg & MAX9867_CODECFLTR_MODE) == mode) + return 0; + + /* shutdown codec before switching filter mode */ + regmap_update_bits(max9867->regmap, MAX9867_PWRMAN, + MAX9867_PWRMAN_SHDN, 0); + + /* switch filter mode */ + regmap_update_bits(max9867->regmap, MAX9867_CODECFLTR, + MAX9867_CODECFLTR_MODE, mode); + + /* out of shutdown now */ + regmap_update_bits(max9867->regmap, MAX9867_PWRMAN, + MAX9867_PWRMAN_SHDN, MAX9867_PWRMAN_SHDN); + + return 0; +} + +static SOC_ENUM_SINGLE_EXT_DECL(max9867_filter, max9867_filter_text); static SOC_ENUM_SINGLE_DECL(max9867_dac_filter, MAX9867_CODECFLTR, 0, max9867_adc_dac_filter_text); static SOC_ENUM_SINGLE_DECL(max9867_adc_filter, MAX9867_CODECFLTR, 4, @@ -76,7 +143,7 @@ static const struct snd_kcontrol_new max9867_snd_controls[] = { SOC_ENUM("Speaker Mode", max9867_spkmode), SOC_SINGLE("Volume Smoothing Switch", MAX9867_MODECONFIG, 6, 1, 0), SOC_SINGLE("Line ZC Switch", MAX9867_MODECONFIG, 5, 1, 0), - SOC_ENUM("DSP Filter", max9867_filter), + SOC_ENUM_EXT("DSP Filter", max9867_filter, max9867_filter_get, max9867_filter_set), SOC_ENUM("ADC Filter", max9867_adc_filter), SOC_ENUM("DAC Filter", max9867_dac_filter), SOC_SINGLE("Mono Playback Switch", MAX9867_IFC1B, 3, 1, 0), @@ -197,13 +264,6 @@ static const struct snd_pcm_hw_constraint_list max9867_constraints_48k = { .count = ARRAY_SIZE(max9867_rates_48k), }; -struct max9867_priv { - struct regmap *regmap; - const struct snd_pcm_hw_constraint_list *constraints; - unsigned int sysclk, pclk; - bool master, dsp_a; -}; - static int max9867_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { diff --git a/sound/soc/codecs/max9867.h b/sound/soc/codecs/max9867.h index 3092c3b99075..b6b880631b13 100644 --- a/sound/soc/codecs/max9867.h +++ b/sound/soc/codecs/max9867.h @@ -44,7 +44,8 @@ #define MAX9867_IFC1B_PCLK_4 0x05 #define MAX9867_IFC1B_PCLK_8 0x06 #define MAX9867_IFC1B_PCLK_16 0x07 -#define MAX9867_CODECFLTR 0x0a +#define MAX9867_CODECFLTR 0x0a +#define MAX9867_CODECFLTR_MODE (1<<7) #define MAX9867_SIDETONE 0x0b #define MAX9867_DACLEVEL 0x0c #define MAX9867_ADCLEVEL 0x0d @@ -58,6 +59,7 @@ #define MAX9867_MICCONFIG 0x15 #define MAX9867_MODECONFIG 0x16 #define MAX9867_PWRMAN 0x17 +#define MAX9867_PWRMAN_SHDN (1<<7) #define MAX9867_REVISION 0xff #define MAX9867_CACHEREGNUM 10 -- 2.20.1

2 1

[PATCH 1/4] ASoC: max98373: Fix return check for devm_regmap_init_sdw()
by Vinod Koul 26 Aug '20

26 Aug '20

devm_regmap_init_sdw() returns a valid pointer on success or ERR_PTR on failure which should be checked with IS_ERR. Also use PTR_ERR for returning error codes. Reported-by: Takashi Iwai <tiwai(a)suse.de> Fixes: 56a5b7910e96 ("ASoC: codecs: max98373: add SoundWire support") Signed-off-by: Vinod Koul <vkoul(a)kernel.org> --- sound/soc/codecs/max98373-sdw.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/max98373-sdw.c b/sound/soc/codecs/max98373-sdw.c index 5fe724728e84..e4675cfff7b2 100644 --- a/sound/soc/codecs/max98373-sdw.c +++ b/sound/soc/codecs/max98373-sdw.c @@ -838,8 +838,8 @@ static int max98373_sdw_probe(struct sdw_slave *slave, /* Regmap Initialization */ regmap = devm_regmap_init_sdw(slave, &max98373_sdw_regmap); - if (!regmap) - return -EINVAL; + if (IS_ERR(regmap)) + return PTR_ERR(regmap); return max98373_init(slave, regmap); } -- 2.26.2

2 5

Question about DPCM
by Shengjiu Wang 26 Aug '20

26 Aug '20

Hi ALSA-experts Can DPCM support such a pipeline? FE -> (FE/BE) ->BE which means I want to connect several cpu-dai together: CPU-DAI1 -> CPU-DAI2 -> CPU-DAI3 -> CPU-DAI4 -> Codec Could you please recommend what I can do? Thanks for you help! Best regards Wang Shengjiu

2 1

[PATCH] ASoC: max9867: shutdown codec when changing filter type
by Pavel Dobias 26 Aug '20

26 Aug '20

Changing filter type without disabling codec results in filter malfunction. Disable codec when changing filter type. Signed-off-by: Pavel Dobias <dobias(a)2n.cz> --- sound/soc/codecs/max9867.c | 64 ++++++++++++++++++++++++++++++++------ sound/soc/codecs/max9867.h | 4 ++- 2 files changed, 57 insertions(+), 11 deletions(-) diff --git a/sound/soc/codecs/max9867.c b/sound/soc/codecs/max9867.c index fcb31144d69c..c5bceb93b171 100644 --- a/sound/soc/codecs/max9867.c +++ b/sound/soc/codecs/max9867.c @@ -15,6 +15,13 @@ #include <sound/tlv.h> #include "max9867.h" +struct max9867_priv { + struct regmap *regmap; + const struct snd_pcm_hw_constraint_list *constraints; + unsigned int sysclk, pclk; + bool master, dsp_a; +}; + static const char *const max9867_spmode[] = { "Stereo Diff", "Mono Diff", "Stereo Cap", "Mono Cap", @@ -32,8 +39,52 @@ static const char *const max9867_adc_dac_filter_text[] = { "Butterworth/8-24" }; -static SOC_ENUM_SINGLE_DECL(max9867_filter, MAX9867_CODECFLTR, 7, - max9867_filter_text); +static int max9867_filter_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); + struct max9867_priv *max9867 = snd_soc_component_get_drvdata(component); + unsigned int reg; + int ret; + + ret = regmap_read(max9867->regmap, MAX9867_CODECFLTR, &reg); + if (ret) + return -EINVAL; + + if (reg & MAX9867_CODECFLTR_MODE) + ucontrol->value.enumerated.item[0] = 1; + else + ucontrol->value.enumerated.item[0] = 0; + + return 0; +} + +static int max9867_filter_set(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); + struct max9867_priv *max9867 = snd_soc_component_get_drvdata(component); + unsigned int mode = ucontrol->value.enumerated.item[0]; + + if (mode > 1) + return -EINVAL; + + /* shutdown codec before switching filter mode */ + regmap_update_bits(max9867->regmap, MAX9867_PWRMAN, + MAX9867_PWRMAN_SHDN, 0); + + /* switch filter mode */ + regmap_update_bits(max9867->regmap, MAX9867_CODECFLTR, + MAX9867_CODECFLTR_MODE, mode ? MAX9867_CODECFLTR_MODE : 0); + + /* out of shutdown now */ + regmap_update_bits(max9867->regmap, MAX9867_PWRMAN, + MAX9867_PWRMAN_SHDN, MAX9867_PWRMAN_SHDN); + + return 0; +} + +static SOC_ENUM_SINGLE_EXT_DECL(max9867_filter, max9867_filter_text); static SOC_ENUM_SINGLE_DECL(max9867_dac_filter, MAX9867_CODECFLTR, 0, max9867_adc_dac_filter_text); static SOC_ENUM_SINGLE_DECL(max9867_adc_filter, MAX9867_CODECFLTR, 4, @@ -76,7 +127,7 @@ static const struct snd_kcontrol_new max9867_snd_controls[] = { SOC_ENUM("Speaker Mode", max9867_spkmode), SOC_SINGLE("Volume Smoothing Switch", MAX9867_MODECONFIG, 6, 1, 0), SOC_SINGLE("Line ZC Switch", MAX9867_MODECONFIG, 5, 1, 0), - SOC_ENUM("DSP Filter", max9867_filter), + SOC_ENUM_EXT("DSP Filter", max9867_filter, max9867_filter_get, max9867_filter_set), SOC_ENUM("ADC Filter", max9867_adc_filter), SOC_ENUM("DAC Filter", max9867_dac_filter), SOC_SINGLE("Mono Playback Switch", MAX9867_IFC1B, 3, 1, 0), @@ -197,13 +248,6 @@ static const struct snd_pcm_hw_constraint_list max9867_constraints_48k = { .count = ARRAY_SIZE(max9867_rates_48k), }; -struct max9867_priv { - struct regmap *regmap; - const struct snd_pcm_hw_constraint_list *constraints; - unsigned int sysclk, pclk; - bool master, dsp_a; -}; - static int max9867_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { diff --git a/sound/soc/codecs/max9867.h b/sound/soc/codecs/max9867.h index 3092c3b99075..b6b880631b13 100644 --- a/sound/soc/codecs/max9867.h +++ b/sound/soc/codecs/max9867.h @@ -44,7 +44,8 @@ #define MAX9867_IFC1B_PCLK_4 0x05 #define MAX9867_IFC1B_PCLK_8 0x06 #define MAX9867_IFC1B_PCLK_16 0x07 -#define MAX9867_CODECFLTR 0x0a +#define MAX9867_CODECFLTR 0x0a +#define MAX9867_CODECFLTR_MODE (1<<7) #define MAX9867_SIDETONE 0x0b #define MAX9867_DACLEVEL 0x0c #define MAX9867_ADCLEVEL 0x0d @@ -58,6 +59,7 @@ #define MAX9867_MICCONFIG 0x15 #define MAX9867_MODECONFIG 0x16 #define MAX9867_PWRMAN 0x17 +#define MAX9867_PWRMAN_SHDN (1<<7) #define MAX9867_REVISION 0xff #define MAX9867_CACHEREGNUM 10 -- 2.20.1

2 1

[PATCH 0/4] SOF FW loader fixes and updates
by Ranjani Sridharan 26 Aug '20

26 Aug '20

This series includes fixes and updates to the SOF firmware loader. Iulian Olaru (1): ASoC: SOF: loader: Add debug box region Karol Trzcinski (2): ASoC: SOF: IPC: make sof_ipc_window monosized ASoC: SOF: ext_manifest: Parse debug ABI version Pierre-Louis Bossart (1): ASoC: SOF: loader: fix memory leak in get_ext_windows include/sound/sof/ext_manifest.h | 7 +++++++ include/sound/sof/info.h | 2 +- include/uapi/sound/sof/abi.h | 2 +- sound/soc/sof/loader.c | 34 +++++++++++++++++++++++++++++--- sound/soc/sof/sof-priv.h | 1 + 5 files changed, 41 insertions(+), 5 deletions(-) -- 2.25.1

2 5

Plugin support in tinycompress
by GitHub pull_request - opened 26 Aug '20

26 Aug '20

alsa-project/tinycompress pull request #6 was opened from rohkkumar: These are initial set of patches to add the plugins support to tinycompress. This pull request corresponds to https://github.com/alsa-project/tinycompress/issues/5 Request URL : https://github.com/alsa-project/tinycompress/pull/6 Patch URL : https://github.com/alsa-project/tinycompress/pull/6.patch Repository URL: https://github.com/alsa-project/tinycompress

1 0

[PATCH] ALSA: hda/realtek: Add quirk for Samsung Galaxy Book Ion NT950XCJ-X716A
by Adrien Crivelli 26 Aug '20

26 Aug '20

The Galaxy Book Ion NT950XCJ-X716A (15 inches) uses the same ALC298 codec as other Samsung laptops which have the no headphone sound bug. I confirmed on my own hardware that this fixes the bug. This also correct the model name for the 13 inches version. It was incorrectly referenced as NT950XCJ-X716A in commit e17f02d05. But it should have been NP930XCJ-K01US. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207423 Signed-off-by: Adrien Crivelli <adrien.crivelli(a)gmail.com> --- sound/pci/hda/patch_realtek.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index a1fa983d2..98789691a 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7695,7 +7695,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x144d, 0xc169, "Samsung Notebook 9 Pen (NP930SBE-K01US)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET), SND_PCI_QUIRK(0x144d, 0xc176, "Samsung Notebook 9 Pro (NP930MBE-K04US)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET), SND_PCI_QUIRK(0x144d, 0xc189, "Samsung Galaxy Flex Book (NT950QCG-X716)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET), - SND_PCI_QUIRK(0x144d, 0xc18a, "Samsung Galaxy Book Ion (NT950XCJ-X716A)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET), + SND_PCI_QUIRK(0x144d, 0xc18a, "Samsung Galaxy Book Ion (NP930XCJ-K01US)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET), + SND_PCI_QUIRK(0x144d, 0xc830, "Samsung Galaxy Book Ion (NT950XCJ-X716A)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET), SND_PCI_QUIRK(0x144d, 0xc740, "Samsung Ativ book 8 (NP870Z5G)", ALC269_FIXUP_ATIV_BOOK_8), SND_PCI_QUIRK(0x144d, 0xc812, "Samsung Notebook Pen S (NT950SBE-X58)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET), SND_PCI_QUIRK(0x1458, 0xfa53, "Gigabyte BXBT-2807", ALC283_FIXUP_HEADSET_MIC), -- 2.25.1

2 2

[PATCH v2] soundwire: intel: fix intel_suspend/resume defined but not used warning
by Bard Liao 26 Aug '20

26 Aug '20

When CONFIG_PM_SLEEP is not defined, GCC throws compilation warnings: drivers/soundwire/intel.c:1799:12: warning: ‘intel_resume’ defined but not used [-Wunused-function] static int intel_resume(struct device *dev) ^~~~~~~~~~~~ drivers/soundwire/intel.c:1683:12: warning: ‘intel_suspend’ defined but not used [-Wunused-function] static int intel_suspend(struct device *dev) ^~~~~~~~~~~~~ Fix by using __maybe_unused macro. Suggested-by: Vinod Koul <vkoul(a)kernel.org> Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- Changelog: v2: - use __maybe_unused macro instead of wrapping in another ifdef. --- drivers/soundwire/intel.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/drivers/soundwire/intel.c b/drivers/soundwire/intel.c index dbcbe2708563..ebca8ced59ec 100644 --- a/drivers/soundwire/intel.c +++ b/drivers/soundwire/intel.c @@ -1527,7 +1527,7 @@ int intel_master_process_wakeen_event(struct platform_device *pdev) #ifdef CONFIG_PM -static int intel_suspend(struct device *dev) +static int __maybe_unused intel_suspend(struct device *dev) { struct sdw_cdns *cdns = dev_get_drvdata(dev); struct sdw_intel *sdw = cdns_to_intel(cdns); @@ -1625,7 +1625,7 @@ static int intel_suspend_runtime(struct device *dev) return ret; } -static int intel_resume(struct device *dev) +static int __maybe_unused intel_resume(struct device *dev) { struct sdw_cdns *cdns = dev_get_drvdata(dev); struct sdw_intel *sdw = cdns_to_intel(cdns); -- 2.17.1

2 1

Re: [PATCH] ALSA: hda/realtek: Add quirk for Samsung Galaxy Book Ion 15 inches
by Alex Dewar 26 Aug '20

26 Aug '20

There's a mistake in this. The ID numbers are the wrong way round. Alex

3 5

[PATCH 0/2] ASoC: qcom: common: Parse auxiliary devices from device tree
by Stephan Gerhold 26 Aug '20

26 Aug '20

In some cases we need to probe additional audio components that do not appear as part of the DAI links specified in the device tree. Examples for this are auxiliary devices such as analog amplifiers or codecs. The ASoC core provides a way to probe these components by adding them to snd_soc_card->aux_dev. This patch set allows specifying them in the device tree through a new "aux-devs" property. Stephan Gerhold (2): ASoC: dt-bindings: qcom: Document "aux-devs" property ASoC: qcom: common: Parse auxiliary devices from device tree .../devicetree/bindings/sound/qcom,apq8016-sbc.txt | 7 +++++++ Documentation/devicetree/bindings/sound/qcom,apq8096.txt | 8 ++++++++ Documentation/devicetree/bindings/sound/qcom,sdm845.txt | 8 ++++++++ sound/soc/qcom/common.c | 4 ++++ 4 files changed, 27 insertions(+) -- 2.28.0

2 4

[PATCH v2 00/20] ALSA: hda/ca0132 - Add AE-7 support/Control cleanup.
by Connor McAdams 26 Aug '20

26 Aug '20

This patch series adds support for the Sound Blaster AE-7 along with adding new controls and cleaning up the old ones. I have personally tested the Recon3D, Recon3Di, Sound Blaster Z/ZxR, and AE-5 cards to confirm they work with these patches. Another user has confirmed that they work on the AE-7. --- v2: Change control names to omit unnecessary information. Connor McAdams (20): ALSA: hda/ca0132 - Cleanup ca0132_mmio_init function. ALSA: hda/ca0132 - Add speaker tuning initialization commands. ALSA: hda/ca0132 - Add surround channel config control. ALSA: hda/ca0132 - Add full-range speaker selection controls. ALSA: hda/ca0132 - Add bass redirection controls. ALSA: hda/ca0132 - Remove surround output selection. ALSA: hda/ca0132 - Clean up ca0132_alt_out_select. ALSA: hda/ca0132 - Add quirk output selection structures. ALSA: hda/ca0132 - Fix Recon3D Center/LFE output. ALSA: hda/ca0132 - Add new quirk ID for SoundBlaster AE-7. ALSA: hda/ca0132 - Add SoundBlaster AE-7 pincfg. ALSA: hda/ca0132 - Set AE-7 bools and select mixer. ALSA: hda/ca0132 - Add ca0132_mmio_init data for SoundBlaster AE-7. ALSA: hda/ca0132 - Add pre-init function for SoundBlaster AE-7. ALSA: hda/ca0132 - Add init data for SoundBlaster AE-7. ALSA: hda/ca0132 - Add DSP setup functions for AE-7. ALSA: hda/ca0132 - Add output selection for SoundBlaster AE-7. ALSA: hda/ca0132 - Add AE-7 microphone selection commands. ALSA: hda/ca0132 - Add AE-7 custom controls. ALSA: hda/ca0132 - Add AE-7 exit commands. sound/pci/hda/patch_ca0132.c | 1794 +++++++++++++++++++++++++++------- 1 file changed, 1430 insertions(+), 364 deletions(-) -- 2.20.1

2 21

[PATCH] ALSA: usb-audio: Add implicit feedback quirk for UR22C
by Joshua Sivec 26 Aug '20

26 Aug '20

This uses the same quirk as the Motu and SSL2 devices. Tested on the UR22C. Fixes bug 208851. Signed-off-by: Joshua Sivec <sivec(a)posteo.net> BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=208851 --- sound/usb/pcm.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 5600751803cf..10d342270dd9 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -369,6 +369,7 @@ static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs, case USB_ID(0x07fd, 0x0008): /* MOTU M Series */ case USB_ID(0x31e9, 0x0001): /* Solid State Logic SSL2 */ case USB_ID(0x31e9, 0x0002): /* Solid State Logic SSL2+ */ + case USB_ID(0x0499, 0x172f): /* Steinberg UR22C */ case USB_ID(0x0d9a, 0x00df): /* RTX6001 */ ep = 0x81; ifnum = 2; -- 2.26.2

2 1

[bug report] ASoC: Intel: KMB: Enable TDM audio capture
by Dan Carpenter 26 Aug '20

26 Aug '20

Hello Michael Sit Wei Hong, The patch 9c3bab3c4f15: "ASoC: Intel: KMB: Enable TDM audio capture" from Aug 11, 2020, leads to the following static checker warning: sound/soc/intel/keembay/kmb_platform.c:518 kmb_dai_hw_params() warn: potential ! vs ~ typo sound/soc/intel/keembay/kmb_platform.c 502 } 503 504 config->chan_nr = params_channels(hw_params); 505 506 switch (config->chan_nr) { 507 case 8: 508 case 4: 509 /* 510 * Platform is not capable of providing clocks for 511 * multi channel audio 512 */ 513 if (kmb_i2s->master) 514 return -EINVAL; 515 516 write_val = ((config->chan_nr / 2) << TDM_CHANNEL_CONFIG_BIT) | 517 (config->data_width << DATA_WIDTH_CONFIG_BIT) | 518 !MASTER_MODE | TDM_OPERATION; ^^^^^^^^^^^^ MASTER_MODE is BIT(13). It's unclear what this is supposed to be. My best guess is that the ! should just be deleted. 519 520 writel(write_val, kmb_i2s->pss_base + I2S_GEN_CFG_0); 521 break; 522 case 2: 523 /* 524 * Platform is only capable of providing clocks need for 525 * 2 channel master mode 526 */ 527 if (!(kmb_i2s->master)) 528 return -EINVAL; 529 530 write_val = ((config->chan_nr / 2) << TDM_CHANNEL_CONFIG_BIT) | 531 (config->data_width << DATA_WIDTH_CONFIG_BIT) | 532 MASTER_MODE | I2S_OPERATION; 533 534 writel(write_val, kmb_i2s->pss_base + I2S_GEN_CFG_0); 535 break; 536 default: 537 dev_dbg(kmb_i2s->dev, "channel not supported\n"); 538 return -EINVAL; 539 } regards, dan carpenter

3 7

[PATCH] ASoC: ti: fixup ams_delta_mute() function name
by Kuninori Morimoto 25 Aug '20

25 Aug '20

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> commit 059374fe9ea5d ("ASoC: ti: merge .digital_mute() into .mute_stream()") merged .digital_mute() into .mute_stream(). But it didn't rename ams_delta_digital_mute() to ams_delta_mute(). This patch fixup it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Reported-by: kernel test robot <lkp(a)intel.com> --- sound/soc/ti/ams-delta.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/ti/ams-delta.c b/sound/soc/ti/ams-delta.c index 5c47de96c529..57feb473a579 100644 --- a/sound/soc/ti/ams-delta.c +++ b/sound/soc/ti/ams-delta.c @@ -446,12 +446,12 @@ static const struct snd_soc_dai_ops ams_delta_dai_ops = { /* Will be used if the codec ever has its own digital_mute function */ static int ams_delta_startup(struct snd_pcm_substream *substream) { - return ams_delta_digital_mute(NULL, 0, substream->stream); + return ams_delta_mute(NULL, 0, substream->stream); } static void ams_delta_shutdown(struct snd_pcm_substream *substream) { - ams_delta_digital_mute(NULL, 1, substream->stream); + ams_delta_mute(NULL, 1, substream->stream); } -- 2.25.1

2 2

[PATCH] ASoC: pcm3168a: ignore 0 Hz settings
by Kuninori Morimoto 25 Aug '20

25 Aug '20

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Some sound card try to set 0 Hz as reset, but it is impossible. This patch ignores it to avoid error return. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/codecs/pcm3168a.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/codecs/pcm3168a.c b/sound/soc/codecs/pcm3168a.c index 9711fab296eb..045c6f8b26be 100644 --- a/sound/soc/codecs/pcm3168a.c +++ b/sound/soc/codecs/pcm3168a.c @@ -306,6 +306,13 @@ static int pcm3168a_set_dai_sysclk(struct snd_soc_dai *dai, struct pcm3168a_priv *pcm3168a = snd_soc_component_get_drvdata(dai->component); int ret; + /* + * Some sound card sets 0 Hz as reset, + * but it is impossible to set. Ignore it here + */ + if (freq == 0) + return 0; + if (freq > PCM3168A_MAX_SYSCLK) return -EINVAL; -- 2.25.1

2 2

[PATCH 0/8] ASoC: SOF: trivial code/log/comment improvements
by Pierre-Louis Bossart 25 Aug '20

25 Aug '20

Remove useless variable initialization and allocation, adjust log levels to make support easier, and fix comments. No functional changes. Guennadi Liakhovetski (2): ASoC: SOF: topology: (cosmetic) remove redundant variable initialisations ASoC: SOF: (cosmetic) use the "bool" type where it makes sense Pierre-Louis Bossart (4): ASoC: SOF: IPC: reduce verbosity of IPC pointer updates ASoC: SOF: acpi: add dev_dbg() log for probe completion ASoC: SOF: Intel: add dev_dbg log when driver is not selected ASoC: Intel: use consistent HDAudio spelling in comments/docs Ranjani Sridharan (2): ASoC: SOF: topology: remove unnecessary memory alloc for sdev->private ASoC: SOF: topology: reduce the log level for unhandled widgets include/sound/soc-acpi.h | 2 +- sound/soc/intel/Kconfig | 2 +- sound/soc/intel/skylake/skl.c | 6 +++--- sound/soc/sof/Kconfig | 2 +- sound/soc/sof/intel/Kconfig | 2 +- sound/soc/sof/ipc.c | 16 +++++++++++----- sound/soc/sof/pcm.c | 8 ++++---- sound/soc/sof/sof-acpi-dev.c | 2 ++ sound/soc/sof/sof-pci-dev.c | 6 +++--- sound/soc/sof/sof-priv.h | 10 +++++----- sound/soc/sof/topology.c | 20 ++++---------------- 11 files changed, 36 insertions(+), 40 deletions(-) base-commit: aafdeba5cbc14cecee3797e669473b70a2b3e81e -- 2.25.1

2 9

[PATCH] ASoC: hdac_hdmi: tidy up a memset()
by Dan Carpenter 25 Aug '20

25 Aug '20

The ARRAY_SIZE() is the number of the elements but we want to use the number of bytes. Fortunately, in this case the value is the same so it doesn't affect runtime. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- sound/soc/codecs/hdac_hdmi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c index 869d1547ae5d..c61cce53f513 100644 --- a/sound/soc/codecs/hdac_hdmi.c +++ b/sound/soc/codecs/hdac_hdmi.c @@ -1474,7 +1474,7 @@ static int hdac_hdmi_eld_ctl_get(struct snd_kcontrol *kcontrol, struct hdac_hdmi_port *port; struct hdac_hdmi_eld *eld; - memset(ucontrol->value.bytes.data, 0, ARRAY_SIZE(ucontrol->value.bytes.data)); + memset(ucontrol->value.bytes.data, 0, sizeof(ucontrol->value.bytes.data)); pcm = get_hdmi_pcm_from_id(hdmi, kcontrol->id.device); if (!pcm) { -- 2.28.0

2 1

[PATCH 0/5]: Qualcomm's lpass-hdmi ASoC driver to support audio over dp port
by Srinivasa Rao 25 Aug '20

25 Aug '20

These patches are to support audio over DP port on Qualcomm's SC7180 LPASS Asoc. It includes machine driver, cpu driver, platform driver updates for HDMI path support, device tree documention, lpass variant structure optimization and configuration changes. These patches depends on the DP patch series https://patchwork.kernel.org/project/dri-devel/list/?series=332029 vsujithk (5): dt-bindings: Add sc7180-lpass HDMI header define dt-bindings: Add dts entry for lpass-hdmi dp audio ASoC: qcom: add support for lpass hdmi driver ASoC: qcom: Add support for audio over DP Optimise lpass variant structure .../devicetree/bindings/sound/qcom,lpass-cpu.yaml | 42 ++ include/dt-bindings/sound/sc7180-lpass.h | 1 + sound/soc/qcom/Kconfig | 5 + sound/soc/qcom/Makefile | 2 + sound/soc/qcom/lpass-apq8016.c | 25 +- sound/soc/qcom/lpass-cpu.c | 92 ++- sound/soc/qcom/lpass-hdmi.c | 685 +++++++++++++++++++++ sound/soc/qcom/lpass-hdmi.h | 129 ++++ sound/soc/qcom/lpass-ipq806x.c | 25 +- sound/soc/qcom/lpass-lpaif-reg.h | 51 +- sound/soc/qcom/lpass-platform.c | 287 +++++++-- sound/soc/qcom/lpass-sc7180.c | 147 ++++- sound/soc/qcom/lpass.h | 123 +++- 13 files changed, 1467 insertions(+), 147 deletions(-) create mode 100644 sound/soc/qcom/lpass-hdmi.c create mode 100644 sound/soc/qcom/lpass-hdmi.h -- 2.7.4

2 7

[PATCH v5 0/2] Add documentation and machine driver for SC7180 sound card
by Cheng-Yi Chiang 25 Aug '20

25 Aug '20

Note: - The machine driver patch depends on LPASS patch series so it is not ready to be merged now. ASoC: qcom: Add support for SC7180 lpass variant https://patchwork.kernel.org/cover/11714317/ - The machine driver patch is made by the collaboration of Cheng-Yi Chiang <cychiang(a)chromium.org> Rohit kumar <rohitkr(a)codeaurora.org> Ajit Pandey <ajitp(a)codeaurora.org> But Ajit has left codeaurora. Changes from v1 to v2: - Ducumentation: Addressed all suggestions from Doug. - Machine driver: - Fix comment style for license. - Sort includes. - Remove sc7180_snd_hw_params. - Remove sc7180_dai_init and use aux device instead for headset jack registration. - Statically define format for Primary MI2S. - Atomic is not a concern because there is mutex in card to make sure startup and shutdown happen sequentially. - Fix missing return -EINVAL in startup. - Use static sound card. - Use devm_kzalloc to avoid kfree. Changes from v2 to v3: - Ducumentation: Addressed suggestions from Srini. - Machine driver: - Reuse qcom_snd_parse_of to parse properties. - Remove playback-only and capture-only. - Misc fixes to address comments. Changes from v3 to v4: - Ducumentation: Addressed suggestions from Rob. - Remove definition of dai. - Use 'sound-dai: true' for sound-dai schema. - Add reg property to pass 'make dt_binding_check' check although reg is not used in the driver. - Machine driver: - Add Reviewed-by: Tzung-Bi Shih <tzungbi(a)google.com> Changes from v4 to v5: - Documentation: Addressed suggestions from Rob. - Add definition for "#address-cells" and "#size-cells". - Add additionalProperties: false - Add required properties. Ajit Pandey (1): ASoC: qcom: sc7180: Add machine driver for sound card registration Cheng-Yi Chiang (1): ASoC: qcom: dt-bindings: Add sc7180 machine bindings .../bindings/sound/qcom,sc7180.yaml | 127 +++++++++ sound/soc/qcom/Kconfig | 12 + sound/soc/qcom/Makefile | 2 + sound/soc/qcom/sc7180.c | 244 ++++++++++++++++++ 4 files changed, 385 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/qcom,sc7180.yaml create mode 100644 sound/soc/qcom/sc7180.c -- 2.28.0.220.ged08abb693-goog

2 3

[PATCH] Add implicit feedback quirk for UR22C
by Joshua Sivec 25 Aug '20

25 Aug '20

This uses the same quirk as the Motu and SSL2 devices. Tested on the UR22C. Fixes bug 208851. Signed-off-by: Joshua Sivec <sivec(a)posteo.net> --- sound/usb/pcm.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 5600751803cf..10d342270dd9 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -369,6 +369,7 @@ static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs, case USB_ID(0x07fd, 0x0008): /* MOTU M Series */ case USB_ID(0x31e9, 0x0001): /* Solid State Logic SSL2 */ case USB_ID(0x31e9, 0x0002): /* Solid State Logic SSL2+ */ + case USB_ID(0x0499, 0x172f): /* Steinberg UR22C */ case USB_ID(0x0d9a, 0x00df): /* RTX6001 */ ep = 0x81; ifnum = 2; -- 2.26.2

2 1

[PATCH] Revert "ALSA: hda: Add support for Loongson 7A1000 controller"
by Tiezhu Yang 25 Aug '20

25 Aug '20

This reverts commit 61eee4a7fc40 ("ALSA: hda: Add support for Loongson 7A1000 controller"). With this patch, there exists the following error on the Loongson LS7A platform: [ 216.639938] rcu: INFO: rcu_preempt self-detected stall on CPU [ 216.645685] rcu: 0-....: (1 GPs behind) idle=d5a/1/0x4000000000000004 softirq=562/563 fqs=16476 [ 216.654565] (t=53772 jiffies g=-463 q=11976) [ 216.658923] NMI backtrace for cpu 0 [ 216.662417] CPU: 0 PID: 68 Comm: kworker/0:2 Not tainted 5.8.0+ #3 [ 216.668587] Hardware name: , BIOS [ 216.672174] Workqueue: events azx_probe_work [snd_hda_intel] [ 216.677829] Stack : 0000000000000000 0000000000000000 ffffffff95004ce0 d786f9efa2288403 [ 216.685848] d786f9efa2288403 0000000000000000 98000001102638c8 ffffffff80cee270 [ 216.693866] 0000000000000000 0000000000000000 0000000000000000 00000000000002b4 [ 216.701883] 206b726f775f6562 0000000000000001 0000000000000000 ffffffff80f30000 [ 216.709902] ffffffff80f30000 ffffffff80d90000 0000000000000000 0000000000000000 [ 216.717919] 0000000000000000 0000000000000000 0000000000000000 ffffffff80d90000 [ 216.725937] ffffffff80d90000 0000000000000007 ffffffff806aff18 0000000000000000 [ 216.733955] ffffffff80f00000 9800000110cc4000 98000001102638c0 ffffffff80d9db80 [ 216.741974] ffffffff8065a740 0000000000000000 0000000000000000 0000000000000000 [ 216.749991] 000073746e657665 0000000000000000 ffffffff80211a64 d786f9efa2288403 [ 216.758009] ... [ 216.760464] Call Trace: [ 216.762920] [<ffffffff80211a64>] show_stack+0x9c/0x130 [ 216.768058] [<ffffffff8065a740>] dump_stack+0xb0/0xf0 [ 216.773110] [<ffffffff80665774>] nmi_cpu_backtrace+0x134/0x140 [ 216.778939] [<ffffffff80665910>] nmi_trigger_cpumask_backtrace+0x190/0x200 [ 216.785805] [<ffffffff802b1abc>] rcu_dump_cpu_stacks+0x12c/0x190 [ 216.791806] [<ffffffff802b08cc>] rcu_sched_clock_irq+0xa2c/0xfc8 [ 216.797808] [<ffffffff802b91d4>] update_process_times+0x2c/0xb8 [ 216.803724] [<ffffffff802cad80>] tick_sched_timer+0x40/0xb8 [ 216.809293] [<ffffffff802ba5f0>] __hrtimer_run_queues+0x118/0x1d0 [ 216.815380] [<ffffffff802bab74>] hrtimer_interrupt+0x12c/0x2d8 [ 216.821208] [<ffffffff8021547c>] c0_compare_interrupt+0x74/0xa0 [ 216.827124] [<ffffffff80296bd0>] __handle_irq_event_percpu+0xa8/0x198 [ 216.833558] [<ffffffff80296cf0>] handle_irq_event_percpu+0x30/0x90 [ 216.839732] [<ffffffff8029d958>] handle_percpu_irq+0x88/0xb8 [ 216.845388] [<ffffffff80296124>] generic_handle_irq+0x44/0x60 [ 216.851131] [<ffffffff80b3cfd0>] do_IRQ+0x18/0x28 [ 216.855838] [<ffffffff8067ace4>] plat_irq_dispatch+0x64/0x100 [ 216.861579] [<ffffffff80209a20>] handle_int+0x140/0x14c [ 216.866802] [<ffffffff802402e8>] irq_exit+0xf8/0x100 This is because AZX_DRIVER_GENERIC can not work well for Loongson LS7A HDA controller, it needs some workarounds which are not merged into the upstream kernel, so it should revert this patch now. There will be a better solution that has been tested carefully to support Loongson LS7A HDA controller. Fixes: 61eee4a7fc40 ("ALSA: hda: Add support for Loongson 7A1000 controller") Cc: <stable(a)vger.kernel.org> # 5.9-rc1+ Signed-off-by: Tiezhu Yang <yangtiezhu(a)loongson.cn> --- sound/pci/hda/hda_intel.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index e34a4d5..0f86e37 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2745,8 +2745,6 @@ static const struct pci_device_id azx_ids[] = { .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_ATI_HDMI }, /* Zhaoxin */ { PCI_DEVICE(0x1d17, 0x3288), .driver_data = AZX_DRIVER_ZHAOXIN }, - /* Loongson */ - { PCI_DEVICE(0x0014, 0x7a07), .driver_data = AZX_DRIVER_GENERIC }, { 0, } }; MODULE_DEVICE_TABLE(pci, azx_ids); -- 2.1.0

2 2

[PATCH 0/2] Tegra HDA specific fixes
by Mohan Kumar 25 Aug '20

25 Aug '20

This series has the fix for - 2 Channel swapping issue for Tegra - WAKEEN programming to detect wake events. Mohan Kumar (2): ALSA: hda: Fix 2 channel swapping for Tegra ALSA: hda/tegra: Program WAKEEN register for Tegra sound/pci/hda/hda_tegra.c | 7 +++++++ sound/pci/hda/patch_hdmi.c | 5 +++++ 2 files changed, 12 insertions(+) -- 2.17.1

3 5

[PATCH] ALSA: ca0106: fix error code handling
by Tong Zhang 25 Aug '20

25 Aug '20

snd_ca0106_spi_write() returns 1 on error, snd_ca0106_pcm_power_dac() is returning the error code directly, and the caller is expecting an negative error code Signed-off-by: Tong Zhang <ztong0001(a)gmail.com> --- sound/pci/ca0106/ca0106_main.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/pci/ca0106/ca0106_main.c b/sound/pci/ca0106/ca0106_main.c index 70d775ff967e..c189f70c82cb 100644 --- a/sound/pci/ca0106/ca0106_main.c +++ b/sound/pci/ca0106/ca0106_main.c @@ -537,7 +537,8 @@ static int snd_ca0106_pcm_power_dac(struct snd_ca0106 *chip, int channel_id, else /* Power down */ chip->spi_dac_reg[reg] |= bit; - return snd_ca0106_spi_write(chip, chip->spi_dac_reg[reg]); + if (snd_ca0106_spi_write(chip, chip->spi_dac_reg[reg]) != 0) + return -ENXIO; } return 0; } -- 2.25.1

2 1

Re: [PATCH] dt-bindings: Whitespace clean-ups in schema files
by Wolfram Sang 25 Aug '20

25 Aug '20

On Wed, Aug 12, 2020 at 02:36:18PM -0600, Rob Herring wrote: > Clean-up incorrect indentation, extra spaces, long lines, and missing > EOF newline in schema files. Most of the clean-ups are for list > indentation which should always be 2 spaces more than the preceding > keyword. > > Found with yamllint (which I plan to integrate into the checks). > > Cc: linux-arm-kernel(a)lists.infradead.org > Cc: linux-clk(a)vger.kernel.org > Cc: dri-devel(a)lists.freedesktop.org > Cc: linux-spi(a)vger.kernel.org > Cc: linux-gpio(a)vger.kernel.org > Cc: linux-remoteproc(a)vger.kernel.org > Cc: linux-hwmon(a)vger.kernel.org > Cc: linux-i2c(a)vger.kernel.org > Cc: linux-fbdev(a)vger.kernel.org > Cc: linux-iio(a)vger.kernel.org > Cc: linux-input(a)vger.kernel.org > Cc: linux-pm(a)vger.kernel.org > Cc: linux-media(a)vger.kernel.org > Cc: alsa-devel(a)alsa-project.org > Cc: linux-mmc(a)vger.kernel.org > Cc: linux-mtd(a)lists.infradead.org > Cc: netdev(a)vger.kernel.org > Cc: linux-rtc(a)vger.kernel.org > Cc: linux-serial(a)vger.kernel.org > Cc: linux-usb(a)vger.kernel.org > Signed-off-by: Rob Herring <robh(a)kernel.org> I trust you guys in figuring out the details, so for touching I2C: Acked-by: Wolfram Sang <wsa(a)kernel.org>

1 0

pcm|dmaengine|imx-sdma race condition on i.MX6
by Richard Leitner 25 Aug '20

25 Aug '20

Hi, we've found a race condition with the PCM on the i.MX6 which results in an -EIO for the SNDRV_PCM_IOCTL_READI_FRAMES ioctl after an -EPIPE (XRUN). A possible reproduction may look like the following reduced call graph during a PCM capture: us -> ioctl(SNDRV_PCM_IOCTL_READI_FRAMES) - wait_for_avail() - schedule_timeout() -> snd_pcm_update_hw_ptr0() - snd_pcm_update_state: EPIPE (XRUN) - sdma_disable_channel_async() # get's scheduled away due to sleep us <- ioctl(SNDRV_PCM_IOCTL_READI_FRAMES) returns -EPIPE us -> ioctl(SNDRV_PCM_IOCTL_PREPARE) # as reaction to the EPIPE (XRUN) us -> ioctl(SNDRV_PCM_IOCTL_READI_FRAMES) # next try to capture frames - sdma_prep_dma_cyclic() - sdma_load_context() # not loaded as context_loaded is 1 - wait_for_avail() - schedule_timeout() # now the sdma_channel_terminate_work() comes back and sets # context_loaded = false and frees in vchan_dma_desc_free_list(). us <- ioctl returns -EIO (capture write error (DMA or IRQ trouble?)) What we have found out, based on our understanding: The dmaengine docu states that a dmaengine_terminate_async() must be followed by a dmaengine_synchronize(). However, in the pcm_dmaengine.c, only dmaengine_terminate_async() is called (for performance reasons and because it might be called from an interrupt handler). In our tests, we saw that the user-space immediately calls ioctl(SNDRV_PCM_IOCTL_PREPARE) as a handler for the happened xrun (previous ioctl(SNDRV_PCM_IOCTL_READI_FRAMES) returns with -EPIPE). In our case (imx-sdma.c), the terminate really happens asynchronously with a worker thread which is not awaited/synchronized by the ioctl(SNDRV_PCM_IOCTL_PREPARE) call. Since the syscall immediately enters an atomic context (snd_pcm_stream_lock_irq()), we are not able to flush the work of the termination worker from within the DMA context. This leads to an unterminated DMA getting re-initialized and then terminated. On the i.MX6 platform the problem is (if I got it correctly) that the sdma_channel_terminate_work() called after the -EPIPE gets scheduled away (for the 1-2ms sleep [1]). During that time the userspace already sends in the ioctl(SNDRV_PCM_IOCTL_PREPARE) and ioctl(SNDRV_PCM_IOCTL_READI_FRAMES). As none of them are anyhow synchronized to the terminate_worker the vchan_dma_desc_free_list() [2] and "sdmac->context_loaded = false;" [3] are executed during the wait_for_avail() [4] of the ioctl(SNDRV_PCM_IOCTL_READI_FRAMES). To make sure we identified the problem correctly we've tested to add a "dmaengine_synchronize()" before the snd_pcm_prepare() in [5]. This fixed the race condition in all our tests. (Before we were able to reproduce it in 100% of the test runs). Based on our understanding, there are two different points to ensure the termination: Either ensure that the termination is finished within the previous SNDRV_PCM_IOCTL_READI_FRAMES call (inside the DMA context) or finishing it in the SNDRV_PCM_IOCTL_PREPARE call (and all other applicable ioclts) before entering the atomic context (from the PCM context). We initially thought about implementing the first approach, basically splitting up the dma_device terminate_all operation into a sync (busy-wait) and a async one. This would align the operations with the DMAengine interface and would enable a sync termination variant from atomic contexts. However, we saw that the dma_free_attrs() function has a WARN_ON on irqs disabled, which would be the case for the sync variant. Side note: We found this issue on the current v5.4.y LTS branch, but it also affects v5.8.y. Any feedback or pointers how we may fix the problem are warmly welcome! If anything is unclear please just ask :-) regards; Richard Leitner Benjamin Bara [1]https://elixir.bootlin.com/linux/v5.8/source/drivers/dma/imx-sdma.c#L1066 [2]https://elixir.bootlin.com/linux/v5.8/source/drivers/dma/imx-sdma.c#L1071 [3]https://elixir.bootlin.com/linux/v5.8/source/drivers/dma/imx-sdma.c#L1072 [4]https://elixir.bootlin.com/linux/v5.8/source/sound/core/pcm_lib.c#L1825 [5]https://elixir.bootlin.com/linux/v5.8/source/sound/core/pcm_native.c#L3226

5 16

[PATCH 00/14] ASoC: Intel: machine driver updates for 5.10
by Pierre-Louis Bossart 24 Aug '20

24 Aug '20

This series updates the tables used to select SoundWire configurations for CometLake and TigerLake, and adds support for SDCA (SoundWire Device Class for Audio) codecs in the common machine driver. These codec drivers are still being tested on early silicon/boards and will be contributed at a later time. For TigerLake Chromebooks a new DMI quirk is added, as well as a means to override the topology names. A pm_runtime fix is also provided to deal with playback/capture dependencies with an amplifier w/ feedback. I also included a minor codec correction for the TGL amplifier. Bard Liao (5): ASoC: Intel: modify SoundWire version id in acpi match table ASoC: Intel: sof_sdw: check SoundWire version when matching codec ASoC: Intel: sof_sdw: rename id as part_id ASoC: Intel: sof_sdw: add rt711 rt1316 rt714 SDCA codec support. ASoC: Intel: sof_sdw: clean-up inclusion of header files Pierre-Louis Bossart (5): ASoC: Intel: soc-acpi: cnl: add support for rt5682 on SoundWire link2 ASoC: Intel: sof-soundwire: add support for rt5682 on link2 ASoC: Intel: soc-acpi: mirror CML and TGL configurations ASoC: Intel: soc-acpi: add support for SDCA boards ASoC: codecs: max98373-sdw: add missing test on resume Rander Wang (2): ASoC: Intel: tgl_max98373: fix a runtime pm issue in multi-thread case ASoC: Intel: sof_sdw: Add support for product Ripto Sathyanarayana Nujella (2): ASoC: Intel: sof_rt5682: override quirk data for tgl_max98373_rt5682 ASoC: SOF: Add topology filename override based on dmi data match sound/soc/codecs/max98373-sdw.c | 3 + sound/soc/intel/boards/Kconfig | 3 + sound/soc/intel/boards/Makefile | 7 +- sound/soc/intel/boards/sof_maxim_common.c | 7 +- sound/soc/intel/boards/sof_rt5682.c | 13 ++ sound/soc/intel/boards/sof_sdw.c | 98 +++++++--- sound/soc/intel/boards/sof_sdw_common.h | 22 ++- sound/soc/intel/boards/sof_sdw_dmic.c | 1 + sound/soc/intel/boards/sof_sdw_max98373.c | 2 + sound/soc/intel/boards/sof_sdw_rt1308.c | 2 + sound/soc/intel/boards/sof_sdw_rt1316.c | 113 ++++++++++++ sound/soc/intel/boards/sof_sdw_rt5682.c | 2 + sound/soc/intel/boards/sof_sdw_rt700.c | 2 + sound/soc/intel/boards/sof_sdw_rt711.c | 2 + sound/soc/intel/boards/sof_sdw_rt711_sdca.c | 174 ++++++++++++++++++ sound/soc/intel/boards/sof_sdw_rt715_sdca.c | 42 +++++ .../intel/common/soc-acpi-intel-cml-match.c | 79 +++++++- .../intel/common/soc-acpi-intel-cnl-match.c | 33 +++- .../intel/common/soc-acpi-intel-icl-match.c | 10 +- .../intel/common/soc-acpi-intel-tgl-match.c | 165 ++++++++++++++++- sound/soc/sof/intel/hda.c | 8 +- sound/soc/sof/sof-pci-dev.c | 24 +++ 22 files changed, 764 insertions(+), 48 deletions(-) create mode 100644 sound/soc/intel/boards/sof_sdw_rt1316.c create mode 100644 sound/soc/intel/boards/sof_sdw_rt711_sdca.c create mode 100644 sound/soc/intel/boards/sof_sdw_rt715_sdca.c base-commit: fcea8b023a5f06ea0180ae65b01520b0414ee325 -- 2.25.1

2 21

[PATCH AUTOSEL 4.19 11/21] ASoC: wm8994: Avoid attempts to read unreadable registers
by Sasha Levin 24 Aug '20

24 Aug '20

From: Sylwester Nawrocki <s.nawrocki(a)samsung.com> [ Upstream commit f082bb59b72039a2326ec1a44496899fb8aa6d0e ] The driver supports WM1811, WM8994, WM8958 devices but according to documentation and the regmap definitions the WM8958_DSP2_* registers are only available on WM8958. In current code these registers are being accessed as if they were available on all the three chips. When starting playback on WM1811 CODEC multiple errors like: "wm8994-codec wm8994-codec: ASoC: error at soc_component_read_no_lock on wm8994-codec: -5" can be seen, which is caused by attempts to read an unavailable WM8958_DSP2_PROGRAM register. The issue has been uncovered by recent commit "e2329ee ASoC: soc-component: add soc_component_err()". This patch adds a check in wm8958_aif_ev() callback so the DSP2 handling is only done for WM8958. Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> Acked-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20200731173834.23832-1-s.nawrocki@samsung.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/wm8958-dsp2.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/codecs/wm8958-dsp2.c b/sound/soc/codecs/wm8958-dsp2.c index 108e8bf42a346..f0a409504a13b 100644 --- a/sound/soc/codecs/wm8958-dsp2.c +++ b/sound/soc/codecs/wm8958-dsp2.c @@ -419,8 +419,12 @@ int wm8958_aif_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); + struct wm8994 *control = dev_get_drvdata(component->dev->parent); int i; + if (control->type != WM8958) + return 0; + switch (event) { case SND_SOC_DAPM_POST_PMU: case SND_SOC_DAPM_PRE_PMU: -- 2.25.1

1 0

[PATCH AUTOSEL 5.4 22/38] ASoC: wm8994: Avoid attempts to read unreadable registers
by Sasha Levin 24 Aug '20

24 Aug '20

From: Sylwester Nawrocki <s.nawrocki(a)samsung.com> [ Upstream commit f082bb59b72039a2326ec1a44496899fb8aa6d0e ] The driver supports WM1811, WM8994, WM8958 devices but according to documentation and the regmap definitions the WM8958_DSP2_* registers are only available on WM8958. In current code these registers are being accessed as if they were available on all the three chips. When starting playback on WM1811 CODEC multiple errors like: "wm8994-codec wm8994-codec: ASoC: error at soc_component_read_no_lock on wm8994-codec: -5" can be seen, which is caused by attempts to read an unavailable WM8958_DSP2_PROGRAM register. The issue has been uncovered by recent commit "e2329ee ASoC: soc-component: add soc_component_err()". This patch adds a check in wm8958_aif_ev() callback so the DSP2 handling is only done for WM8958. Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> Acked-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20200731173834.23832-1-s.nawrocki@samsung.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/wm8958-dsp2.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/codecs/wm8958-dsp2.c b/sound/soc/codecs/wm8958-dsp2.c index 18535b326680a..04f23477039a5 100644 --- a/sound/soc/codecs/wm8958-dsp2.c +++ b/sound/soc/codecs/wm8958-dsp2.c @@ -416,8 +416,12 @@ int wm8958_aif_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); + struct wm8994 *control = dev_get_drvdata(component->dev->parent); int i; + if (control->type != WM8958) + return 0; + switch (event) { case SND_SOC_DAPM_POST_PMU: case SND_SOC_DAPM_PRE_PMU: -- 2.25.1

1 0

[PATCH AUTOSEL 5.4 20/38] ALSA: hda/realtek: Add model alc298-samsung-headphone
by Sasha Levin 24 Aug '20

24 Aug '20

From: Mike Pozulp <pozulp.kernel(a)gmail.com> [ Upstream commit 23dc958689449be85e39351a8c809c3d344b155b ] The very quiet and distorted headphone output bug that afflicted my Samsung Notebook 9 is appearing in many other Samsung laptops. Expose the quirk which fixed my laptop as a model so other users can try it. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207423 Signed-off-by: Mike Pozulp <pozulp.kernel(a)gmail.com> Link: https://lore.kernel.org/r/20200817043219.458889-1-pozulp.kernel@gmail.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- 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 88629906f314c..e230909af980c 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7926,6 +7926,7 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {.id = ALC299_FIXUP_PREDATOR_SPK, .name = "predator-spk"}, {.id = ALC298_FIXUP_HUAWEI_MBX_STEREO, .name = "huawei-mbx-stereo"}, {.id = ALC256_FIXUP_MEDION_HEADSET_NO_PRESENCE, .name = "alc256-medion-headset"}, + {.id = ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET, .name = "alc298-samsung-headphone"}, {} }; #define ALC225_STANDARD_PINS \ -- 2.25.1

1 0

[PATCH AUTOSEL 5.7 31/54] ALSA: usb-audio: ignore broken processing/extension unit
by Sasha Levin 24 Aug '20

24 Aug '20

From: Tom Yan <tom.ty89(a)gmail.com> [ Upstream commit d8d0db7bb358ef65d60726a61bfcd08eccff0bc0 ] Some devices have broken extension unit where getting current value doesn't work. Attempt that once when creating mixer control for it. If it fails, just ignore it, so that it won't cripple the device entirely (and/or make the error floods). Signed-off-by: Tom Yan <tom.ty89(a)gmail.com> Link: https://lore.kernel.org/r/5f3abc52.1c69fb81.9cf2.fe91@mx.google.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/usb/mixer.c | 8 +++++++- 1 file changed, 7 insertions(+), 1 deletion(-) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index eab0fd4fd7c33..e0b7174c10430 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -2367,7 +2367,7 @@ static int build_audio_procunit(struct mixer_build *state, int unitid, int num_ins; struct usb_mixer_elem_info *cval; struct snd_kcontrol *kctl; - int i, err, nameid, type, len; + int i, err, nameid, type, len, val; const struct procunit_info *info; const struct procunit_value_info *valinfo; const struct usbmix_name_map *map; @@ -2470,6 +2470,12 @@ static int build_audio_procunit(struct mixer_build *state, int unitid, break; } + err = get_cur_ctl_value(cval, cval->control << 8, &val); + if (err < 0) { + usb_mixer_elem_info_free(cval); + return -EINVAL; + } + kctl = snd_ctl_new1(&mixer_procunit_ctl, cval); if (!kctl) { usb_mixer_elem_info_free(cval); -- 2.25.1

1 0

[PATCH AUTOSEL 5.7 30/54] ASoC: wm8994: Avoid attempts to read unreadable registers
by Sasha Levin 24 Aug '20

24 Aug '20

From: Sylwester Nawrocki <s.nawrocki(a)samsung.com> [ Upstream commit f082bb59b72039a2326ec1a44496899fb8aa6d0e ] The driver supports WM1811, WM8994, WM8958 devices but according to documentation and the regmap definitions the WM8958_DSP2_* registers are only available on WM8958. In current code these registers are being accessed as if they were available on all the three chips. When starting playback on WM1811 CODEC multiple errors like: "wm8994-codec wm8994-codec: ASoC: error at soc_component_read_no_lock on wm8994-codec: -5" can be seen, which is caused by attempts to read an unavailable WM8958_DSP2_PROGRAM register. The issue has been uncovered by recent commit "e2329ee ASoC: soc-component: add soc_component_err()". This patch adds a check in wm8958_aif_ev() callback so the DSP2 handling is only done for WM8958. Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> Acked-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20200731173834.23832-1-s.nawrocki@samsung.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/wm8958-dsp2.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/codecs/wm8958-dsp2.c b/sound/soc/codecs/wm8958-dsp2.c index ca42445b649d4..b471892d84778 100644 --- a/sound/soc/codecs/wm8958-dsp2.c +++ b/sound/soc/codecs/wm8958-dsp2.c @@ -412,8 +412,12 @@ int wm8958_aif_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); + struct wm8994 *control = dev_get_drvdata(component->dev->parent); int i; + if (control->type != WM8958) + return 0; + switch (event) { case SND_SOC_DAPM_POST_PMU: case SND_SOC_DAPM_PRE_PMU: -- 2.25.1

1 0

[PATCH AUTOSEL 5.7 28/54] ALSA: hda/realtek: Add model alc298-samsung-headphone
by Sasha Levin 24 Aug '20

24 Aug '20

From: Mike Pozulp <pozulp.kernel(a)gmail.com> [ Upstream commit 23dc958689449be85e39351a8c809c3d344b155b ] The very quiet and distorted headphone output bug that afflicted my Samsung Notebook 9 is appearing in many other Samsung laptops. Expose the quirk which fixed my laptop as a model so other users can try it. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207423 Signed-off-by: Mike Pozulp <pozulp.kernel(a)gmail.com> Link: https://lore.kernel.org/r/20200817043219.458889-1-pozulp.kernel@gmail.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- 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 313eecfb91b44..e2bb0b6986182 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7926,6 +7926,7 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {.id = ALC299_FIXUP_PREDATOR_SPK, .name = "predator-spk"}, {.id = ALC298_FIXUP_HUAWEI_MBX_STEREO, .name = "huawei-mbx-stereo"}, {.id = ALC256_FIXUP_MEDION_HEADSET_NO_PRESENCE, .name = "alc256-medion-headset"}, + {.id = ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET, .name = "alc298-samsung-headphone"}, {} }; #define ALC225_STANDARD_PINS \ -- 2.25.1

1 0

[PATCH AUTOSEL 5.8 36/63] ALSA: usb-audio: ignore broken processing/extension unit
by Sasha Levin 24 Aug '20

24 Aug '20

From: Tom Yan <tom.ty89(a)gmail.com> [ Upstream commit d8d0db7bb358ef65d60726a61bfcd08eccff0bc0 ] Some devices have broken extension unit where getting current value doesn't work. Attempt that once when creating mixer control for it. If it fails, just ignore it, so that it won't cripple the device entirely (and/or make the error floods). Signed-off-by: Tom Yan <tom.ty89(a)gmail.com> Link: https://lore.kernel.org/r/5f3abc52.1c69fb81.9cf2.fe91@mx.google.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/usb/mixer.c | 8 +++++++- 1 file changed, 7 insertions(+), 1 deletion(-) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index eab0fd4fd7c33..e0b7174c10430 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -2367,7 +2367,7 @@ static int build_audio_procunit(struct mixer_build *state, int unitid, int num_ins; struct usb_mixer_elem_info *cval; struct snd_kcontrol *kctl; - int i, err, nameid, type, len; + int i, err, nameid, type, len, val; const struct procunit_info *info; const struct procunit_value_info *valinfo; const struct usbmix_name_map *map; @@ -2470,6 +2470,12 @@ static int build_audio_procunit(struct mixer_build *state, int unitid, break; } + err = get_cur_ctl_value(cval, cval->control << 8, &val); + if (err < 0) { + usb_mixer_elem_info_free(cval); + return -EINVAL; + } + kctl = snd_ctl_new1(&mixer_procunit_ctl, cval); if (!kctl) { usb_mixer_elem_info_free(cval); -- 2.25.1

1 0

[PATCH AUTOSEL 5.8 35/63] ASoC: wm8994: Avoid attempts to read unreadable registers
by Sasha Levin 24 Aug '20

24 Aug '20

From: Sylwester Nawrocki <s.nawrocki(a)samsung.com> [ Upstream commit f082bb59b72039a2326ec1a44496899fb8aa6d0e ] The driver supports WM1811, WM8994, WM8958 devices but according to documentation and the regmap definitions the WM8958_DSP2_* registers are only available on WM8958. In current code these registers are being accessed as if they were available on all the three chips. When starting playback on WM1811 CODEC multiple errors like: "wm8994-codec wm8994-codec: ASoC: error at soc_component_read_no_lock on wm8994-codec: -5" can be seen, which is caused by attempts to read an unavailable WM8958_DSP2_PROGRAM register. The issue has been uncovered by recent commit "e2329ee ASoC: soc-component: add soc_component_err()". This patch adds a check in wm8958_aif_ev() callback so the DSP2 handling is only done for WM8958. Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> Acked-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20200731173834.23832-1-s.nawrocki@samsung.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/wm8958-dsp2.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/codecs/wm8958-dsp2.c b/sound/soc/codecs/wm8958-dsp2.c index ca42445b649d4..b471892d84778 100644 --- a/sound/soc/codecs/wm8958-dsp2.c +++ b/sound/soc/codecs/wm8958-dsp2.c @@ -412,8 +412,12 @@ int wm8958_aif_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); + struct wm8994 *control = dev_get_drvdata(component->dev->parent); int i; + if (control->type != WM8958) + return 0; + switch (event) { case SND_SOC_DAPM_POST_PMU: case SND_SOC_DAPM_PRE_PMU: -- 2.25.1

1 0

[PATCH AUTOSEL 5.8 33/63] ALSA: hda/realtek: Add model alc298-samsung-headphone
by Sasha Levin 24 Aug '20

24 Aug '20

From: Mike Pozulp <pozulp.kernel(a)gmail.com> [ Upstream commit 23dc958689449be85e39351a8c809c3d344b155b ] The very quiet and distorted headphone output bug that afflicted my Samsung Notebook 9 is appearing in many other Samsung laptops. Expose the quirk which fixed my laptop as a model so other users can try it. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207423 Signed-off-by: Mike Pozulp <pozulp.kernel(a)gmail.com> Link: https://lore.kernel.org/r/20200817043219.458889-1-pozulp.kernel@gmail.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- 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 8626e59f1e6a9..ce9cf6013e9d3 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7956,6 +7956,7 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {.id = ALC299_FIXUP_PREDATOR_SPK, .name = "predator-spk"}, {.id = ALC298_FIXUP_HUAWEI_MBX_STEREO, .name = "huawei-mbx-stereo"}, {.id = ALC256_FIXUP_MEDION_HEADSET_NO_PRESENCE, .name = "alc256-medion-headset"}, + {.id = ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET, .name = "alc298-samsung-headphone"}, {} }; #define ALC225_STANDARD_PINS \ -- 2.25.1

1 0

[PATCH 0/5] ASoC: SOF/Intel: fix cppcheck warnings
by Pierre-Louis Bossart 24 Aug '20

24 Aug '20

A small set of fixes to reduce the number of warnings. Pierre-Louis Bossart (5): ASOC: SOF: Intel: hda-codec: move unused label to correct position ASoC: SOF: Intel: hda-codec: move variable used conditionally ASoC: Intel: rename shadowed variable for all broadwell boards ASoC: Intel: bytcht_cx2072x: simplify return handling ASoC: Intel: sof_sdw: clarify operator precedence sound/soc/intel/boards/bdw-rt5650.c | 10 +++++----- sound/soc/intel/boards/bdw-rt5677.c | 8 ++++---- sound/soc/intel/boards/broadwell.c | 8 ++++---- sound/soc/intel/boards/bytcht_cx2072x.c | 2 +- sound/soc/intel/boards/sof_sdw.c | 2 +- sound/soc/sof/intel/hda-codec.c | 7 ++++++- 6 files changed, 21 insertions(+), 16 deletions(-) -- 2.25.1

3 12

[PATCH RESEND] ASoC: fsl_sai: Add -EPROBE_DEFER check for regmap init
by Shengjiu Wang 24 Aug '20

24 Aug '20

Regmap initialization may return -EPROBE_DEFER for clock may not be ready, so check -EPROBE_DEFER error type before start another Regmap initialization. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> Acked-by: Nicolin Chen <nicoleotsuka(a)gmail.com> --- sound/soc/fsl/fsl_sai.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index f6969a5d49e3..62c5fdb678fc 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -959,7 +959,7 @@ static int fsl_sai_probe(struct platform_device *pdev) "bus", base, &fsl_sai_regmap_config); /* Compatible with old DTB cases */ - if (IS_ERR(sai->regmap)) + if (IS_ERR(sai->regmap) && PTR_ERR(sai->regmap) != -EPROBE_DEFER) sai->regmap = devm_regmap_init_mmio_clk(&pdev->dev, "sai", base, &fsl_sai_regmap_config); if (IS_ERR(sai->regmap)) { -- 2.27.0

2 1

[PATCH 1/2] ASoC: dt-bindings: ak5558: Add power supply property
by Shengjiu Wang 24 Aug '20

24 Aug '20

AVDD-supply is for Analog power supply DVDD-supply is for Digital power supply Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- Documentation/devicetree/bindings/sound/ak5558.txt | 2 ++ 1 file changed, 2 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/ak5558.txt b/Documentation/devicetree/bindings/sound/ak5558.txt index 7d67ca6ced80..36934098170c 100644 --- a/Documentation/devicetree/bindings/sound/ak5558.txt +++ b/Documentation/devicetree/bindings/sound/ak5558.txt @@ -10,6 +10,8 @@ Required properties: Optional properties: - reset-gpios: A GPIO specifier for the power down & reset pin. +- AVDD-supply: Analog power supply +- DVDD-supply: Digital power supply Example: -- 2.27.0

2 2

[PATCH] ALSA: hda/realtek: Add quirk for Samsung Galaxy Book Ion 15 inches
by Adrien Crivelli 24 Aug '20

24 Aug '20

The Galaxy Book Ion 15 inches uses the same ALC298 codec as other Samsung laptops which have the no headphone sound bug. I confirmed on my own hardware that this fixes the bug. Commit e17f02d05 mixed up 13 inches and 15 inches models. This clarifies the situation with the correct model names. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207423 Signed-off-by: Adrien Crivelli <adrien.crivelli(a)gmail.com> --- sound/pci/hda/patch_realtek.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index a1fa983d2..98789691a 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7695,7 +7695,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x144d, 0xc169, "Samsung Notebook 9 Pen (NP930SBE-K01US)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET), SND_PCI_QUIRK(0x144d, 0xc176, "Samsung Notebook 9 Pro (NP930MBE-K04US)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET), SND_PCI_QUIRK(0x144d, 0xc189, "Samsung Galaxy Flex Book (NT950QCG-X716)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET), - SND_PCI_QUIRK(0x144d, 0xc18a, "Samsung Galaxy Book Ion (NT950XCJ-X716A)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET), + SND_PCI_QUIRK(0x144d, 0xc18a, "Samsung Galaxy Book Ion (NP930XCJ-K01US)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET), + SND_PCI_QUIRK(0x144d, 0xc830, "Samsung Galaxy Book Ion (NT950XCJ-X716A)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET), SND_PCI_QUIRK(0x144d, 0xc740, "Samsung Ativ book 8 (NP870Z5G)", ALC269_FIXUP_ATIV_BOOK_8), SND_PCI_QUIRK(0x144d, 0xc812, "Samsung Notebook Pen S (NT950SBE-X58)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET), SND_PCI_QUIRK(0x1458, 0xfa53, "Gigabyte BXBT-2807", ALC283_FIXUP_HEADSET_MIC), -- 2.25.1

1 0

Help with EDIROL UA-101 midi not recognized on kernel / UA101 improvements
by nick83ola 24 Aug '20

24 Aug '20

Hi alsa-devel, I have an old edirol interface (ua-101) that is still kicking (8 channel I/O in a small package) The problem is that recently (I don't know from what kernel) the midi is not recognized anymore. Also I would like to implement an application to control the internal mixer. Can someone help me out/point me in the right direction where to start?Or help me with this? I never debugged an USB driver before. Thank you Nicola Lunghi

1 0

[PATCH] Documentation: sound/cards: fix heading underline lengths for https: changes
by Randy Dunlap 24 Aug '20

24 Aug '20

From: Randy Dunlap <rdunlap(a)infradead.org> Fix documentation build warnings for underline length too short, caused by s/http/https/ and not changing the accompanying underlines. Documentation/sound/cards/audigy-mixer.rst:335: WARNING: Title underline too short. US Patents (https://www.uspto.gov/) ---------------------------------- Documentation/sound/cards/sb-live-mixer.rst:340: WARNING: Title underline too short. US Patents (https://www.uspto.gov/) ---------------------------------- Fixes: 7ed33ea6b4fa ("ALSA: Replace HTTP links with HTTPS ones") Signed-off-by: Randy Dunlap <rdunlap(a)infradead.org> Cc: Alexander A. Klimov <grandmaster(a)al2klimov.de> Cc: Takashi Iwai <tiwai(a)suse.de> --- Documentation/sound/cards/audigy-mixer.rst | 2 +- Documentation/sound/cards/sb-live-mixer.rst | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) --- lnx-59-rc2.orig/Documentation/sound/cards/audigy-mixer.rst +++ lnx-59-rc2/Documentation/sound/cards/audigy-mixer.rst @@ -332,7 +332,7 @@ WO 9901953 (A1) US Patents (https://www.uspto.gov/) ----------------------------------- +----------------------------------- US 5925841 Digital Sampling Instrument employing cache memory (Jul. 20, 1999) --- lnx-59-rc2.orig/Documentation/sound/cards/sb-live-mixer.rst +++ lnx-59-rc2/Documentation/sound/cards/sb-live-mixer.rst @@ -337,7 +337,7 @@ WO 9901953 (A1) US Patents (https://www.uspto.gov/) ----------------------------------- +----------------------------------- US 5925841 Digital Sampling Instrument employing cache memory (Jul. 20, 1999)

2 1

[PATCH] ALSA: usb-audio: Disable autosuspend for Lenovo ThinkStation P620
by Kai-Heng Feng 24 Aug '20

24 Aug '20

If USB autosuspend is enabled, both front and rear panel can no longer detect jack insertion. Enable USB remote wakeup, i.e. needs_remote_wakeup = 1, doesn't help either. So disable USB autosuspend to prevent missing jack detection event. Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> --- sound/usb/quirks-table.h | 18 ++++++++++++++---- sound/usb/quirks.c | 10 ++++++++++ sound/usb/usbaudio.h | 1 + 3 files changed, 25 insertions(+), 4 deletions(-) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index f4fb002e3ef4..416de71c6895 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -2827,14 +2827,24 @@ YAMAHA_DEVICE(0x7010, "UB99"), /* Lenovo ThinkStation P620 Rear Line-in, Line-out and Microphone */ { USB_DEVICE(0x17aa, 0x1046), - QUIRK_DEVICE_PROFILE("Lenovo", "ThinkStation P620 Rear", - "Lenovo-ThinkStation-P620-Rear"), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + .vendor_name = "Lenovo", + .product_name = "ThinkStation P620 Rear", + .profile_name = "Lenovo-ThinkStation-P620-Rear", + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_SETUP_DISABLE_AUTOSUSPEND + } }, /* Lenovo ThinkStation P620 Internal Speaker + Front Headset */ { USB_DEVICE(0x17aa, 0x104d), - QUIRK_DEVICE_PROFILE("Lenovo", "ThinkStation P620 Main", - "Lenovo-ThinkStation-P620-Main"), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + .vendor_name = "Lenovo", + .product_name = "ThinkStation P620 Main", + .profile_name = "Lenovo-ThinkStation-P620-Main", + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_SETUP_DISABLE_AUTOSUSPEND + } }, /* Native Instruments MK2 series */ diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index abf99b814a0f..b800fd92106c 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -518,6 +518,15 @@ static int setup_fmt_after_resume_quirk(struct snd_usb_audio *chip, return 1; /* Continue with creating streams and mixer */ } +static int setup_disable_autosuspend(struct snd_usb_audio *chip, + struct usb_interface *iface, + struct usb_driver *driver, + const struct snd_usb_audio_quirk *quirk) +{ + driver->supports_autosuspend = 0; + return 1; /* Continue with creating streams and mixer */ +} + /* * audio-interface quirks * @@ -557,6 +566,7 @@ int snd_usb_create_quirk(struct snd_usb_audio *chip, [QUIRK_AUDIO_ALIGN_TRANSFER] = create_align_transfer_quirk, [QUIRK_AUDIO_STANDARD_MIXER] = create_standard_mixer_quirk, [QUIRK_SETUP_FMT_AFTER_RESUME] = setup_fmt_after_resume_quirk, + [QUIRK_SETUP_DISABLE_AUTOSUSPEND] = setup_disable_autosuspend, }; if (quirk->type < QUIRK_TYPE_COUNT) { diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index b91c4c0807ec..6839915a0128 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -102,6 +102,7 @@ enum quirk_type { QUIRK_AUDIO_ALIGN_TRANSFER, QUIRK_AUDIO_STANDARD_MIXER, QUIRK_SETUP_FMT_AFTER_RESUME, + QUIRK_SETUP_DISABLE_AUTOSUSPEND, QUIRK_TYPE_COUNT }; -- 2.17.1

2 1

[PATCH 0/3] ALSA: usb-audio: Reduce device table
by Takashi Iwai 23 Aug '20

23 Aug '20

Hi, this is a patch set for USB-audio driver to reduce the unnecessary entries from the exported device table. Also it contains the corrections of the device matching for a few entries to be more strict. Takashi === Takashi Iwai (3): ALSA: usb-audio: Move device rename and profile quirks to an internal table ALSA: usb-audio: Simplify quirk entries with a macro ALSA: usb-audio: Properly match with audio interface class sound/usb/card.c | 133 +++++++++++++++++++++-- sound/usb/quirks-table.h | 276 +++++++---------------------------------------- sound/usb/usbaudio.h | 1 - 3 files changed, 162 insertions(+), 248 deletions(-) -- 2.16.4

2 7

[PATCH] ALSA: usb-audio: Correct wrongly matching entries with audio class
by Takashi Iwai 23 Aug '20

23 Aug '20

The commit 51ab5d77dcb8 ("ALSA: usb-audio: Properly match with audio interface class") converted the quirk entries that have both vid/pid pair and bInterface fields to match with all those with a new macro USB_AUDIO_CLASS(). However, it turned out that those are false conversions; all those (but the unknown KeithMcMillen device) are actually with vendor-specific interface class, hence the conversions broke the matching. This patch corrects those entries to the right one, USB_DEVICE_VENDOR_SPEC() (and USB_DEVICE() for KeithMcMillen to be sure), and drop the unused USB_AUDIO_CLASS macro again. Fixes: 51ab5d77dcb8 ("ALSA: usb-audio: Properly match with audio interface class") Reported-by: Alexander Tsoy <alexander(a)tsoy.me> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/quirks-table.h | 21 ++++++--------------- 1 file changed, 6 insertions(+), 15 deletions(-) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 969c79d0c688..7fe9d3e75d59 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -35,14 +35,6 @@ .bInterfaceClass = USB_CLASS_AUDIO, \ .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL -/* Another standard entry matching with vid/pid and the audio class */ -#define USB_AUDIO_CLASS(vend, prod) \ - .match_flags = USB_DEVICE_ID_MATCH_DEVICE | \ - USB_DEVICE_ID_MATCH_INT_CLASS, \ - .idVendor = vend, \ - .idProduct = prod, \ - .bInterfaceClass = USB_CLASS_AUDIO - /* FTDI devices */ { USB_DEVICE(0x0403, 0xb8d8), @@ -77,13 +69,13 @@ }, /* E-Mu 0202 USB */ -{ USB_AUDIO_CLASS(0x041e, 0x3f02) }, +{ USB_DEVICE_VENDOR_SPEC(0x041e, 0x3f02) }, /* E-Mu 0404 USB */ -{ USB_AUDIO_CLASS(0x041e, 0x3f04) }, +{ USB_DEVICE_VENDOR_SPEC(0x041e, 0x3f04) }, /* E-Mu Tracker Pre */ -{ USB_AUDIO_CLASS(0x041e, 0x3f0a) }, +{ USB_DEVICE_VENDOR_SPEC(0x041e, 0x3f0a) }, /* E-Mu 0204 USB */ -{ USB_AUDIO_CLASS(0x041e, 0x3f19) }, +{ USB_DEVICE_VENDOR_SPEC(0x041e, 0x3f19) }, /* * HP Wireless Audio @@ -2756,7 +2748,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), }, /* KeithMcMillen Stringport */ -{ USB_AUDIO_CLASS(0x1f38, 0x0001) }, +{ USB_DEVICE(0x1f38, 0x0001) }, /* FIXME: should be more restrictive matching */ /* Miditech devices */ { @@ -2979,7 +2971,7 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"), }, { /* Tascam US122 MKII - playback-only support */ - USB_AUDIO_CLASS(0x0644, 0x8021), + USB_DEVICE_VENDOR_SPEC(0x0644, 0x8021), .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { .vendor_name = "TASCAM", .product_name = "US122 MKII", @@ -3611,4 +3603,3 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"), #undef USB_DEVICE_VENDOR_SPEC #undef USB_AUDIO_DEVICE -#undef USB_AUDIO_CLASS -- 2.16.4

1 0

[PATCH] topology: hda-dsp: Remove srcintc module from HDA DSP topology
by Mateusz Gorski 23 Aug '20

23 Aug '20

The main goal of this topology is to handle every device with hda-dsp configuration on Skylake SST driver. The module_id value for srcintc module varies by FW branch, which makes it impossible to have one universal topology with srcintc module for all of them. Remove srcintc from this topology to make it truly universal. Signed-off-by: Mateusz Gorski <mateusz.gorski(a)linux.intel.com> --- .../hda-dsp/skl_hda_dsp_generic-tplg.conf | 218 +----------------- 1 file changed, 7 insertions(+), 211 deletions(-) diff --git a/topology/hda-dsp/skl_hda_dsp_generic-tplg.conf b/topology/hda-dsp/skl_hda_dsp_generic-tplg.conf index 32ba9a7..17b7af0 100644 --- a/topology/hda-dsp/skl_hda_dsp_generic-tplg.conf +++ b/topology/hda-dsp/skl_hda_dsp_generic-tplg.conf @@ -102,7 +102,7 @@ SectionVendorTuples."mod_type_data_size_desc" { } tuples."short.u16_size_desc" { - SKL_TKN_U16_BLOCK_SIZE "6436" + SKL_TKN_U16_BLOCK_SIZE "6072" } } @@ -114,7 +114,7 @@ SectionVendorTuples."mod_type_data" { tokens "skl_tokens" tuples."byte.u8_num_mod" { - SKL_TKN_U8_NUM_MOD "5" + SKL_TKN_U8_NUM_MOD "4" } tuples."uuid.mod_0" { @@ -1252,74 +1252,6 @@ SectionVendorTuples."mod_type_data" { SKL_TKN_U32_FMT_SAMPLE_TYPE "0" SKL_TKN_U32_FMT_CH_MAP "4294967056" } - - tuples."uuid.mod_4" { - SKL_TKN_UUID "141, 178, 27, 230, 154, 20, 31, 76, 183, 9, 70, 130, 62, 245, 245, 174" - } - - tuples."byte.u8_mod_type_4" { - SKL_TKN_MM_U8_MOD_IDX "4" - SKL_TKN_U8_IN_PIN_TYPE "0" - SKL_TKN_U8_OUT_PIN_TYPE "0" - SKL_TKN_U8_IN_QUEUE_COUNT "1" - SKL_TKN_U8_OUT_QUEUE_COUNT "1" - SKL_TKN_MM_U8_NUM_RES "1" - SKL_TKN_MM_U8_NUM_INTF "1" - } - - tuples."word.u32_mod_type_4_res_0" { - SKL_TKN_MM_U32_RES_ID "0" - SKL_TKN_U32_MEM_PAGES "4" - SKL_TKN_MM_U32_CPS "100000" - SKL_TKN_U32_IBS "384" - SKL_TKN_U32_OBS "384" - SKL_TKN_MM_U32_DMA_SIZE "0" - SKL_TKN_MM_U32_CPC "10157" - } - - tuples."word.u32_mod_type_4_res_0_input_0" { - SKL_TKN_U32_DIR_PIN_COUNT "0" - SKL_TKN_MM_U32_RES_PIN_ID "0" - SKL_TKN_MM_U32_PIN_BUF "384" - } - - tuples."word.u32_mod_type_4_res_0_output_0" { - SKL_TKN_U32_DIR_PIN_COUNT "1" - SKL_TKN_MM_U32_RES_PIN_ID "0" - SKL_TKN_MM_U32_PIN_BUF "384" - } - - tuples."word.u32_mod_type_4_intf_0" { - SKL_TKN_MM_U32_FMT_ID "0" - SKL_TKN_MM_U32_NUM_IN_FMT "1" - SKL_TKN_MM_U32_NUM_OUT_FMT "1" - } - - tuples."word.u32_mod_type_4_intf_0_input_0" { - SKL_TKN_U32_DIR_PIN_COUNT "0" - SKL_TKN_MM_U32_INTF_PIN_ID "0" - SKL_TKN_U32_FMT_CH "2" - SKL_TKN_U32_FMT_FREQ "48000" - SKL_TKN_U32_FMT_BIT_DEPTH "32" - SKL_TKN_U32_FMT_CH_CONFIG "1" - SKL_TKN_U32_FMT_INTERLEAVE "0" - SKL_TKN_U32_FMT_SAMPLE_SIZE "32" - SKL_TKN_U32_FMT_SAMPLE_TYPE "0" - SKL_TKN_U32_FMT_CH_MAP "4294967056" - } - - tuples."word.u32_mod_type_4_intf_0_output_0" { - SKL_TKN_U32_DIR_PIN_COUNT "1" - SKL_TKN_MM_U32_INTF_PIN_ID "0" - SKL_TKN_U32_FMT_CH "2" - SKL_TKN_U32_FMT_FREQ "48000" - SKL_TKN_U32_FMT_BIT_DEPTH "32" - SKL_TKN_U32_FMT_CH_CONFIG "1" - SKL_TKN_U32_FMT_INTERLEAVE "0" - SKL_TKN_U32_FMT_SAMPLE_SIZE "32" - SKL_TKN_U32_FMT_SAMPLE_TYPE "0" - SKL_TKN_U32_FMT_CH_MAP "4294967056" - } } SectionData."mod_type_data" { @@ -1597,12 +1529,12 @@ SectionVendorTuples."media0_in updwmix 0" { tuples."word.out_pin_0" { SKL_TKN_U32_DIR_PIN_COUNT "1" - SKL_TKN_U32_PIN_MOD_ID "8" + SKL_TKN_U32_PIN_MOD_ID "0" SKL_TKN_U32_PIN_INST_ID "0" } tuples."uuid.out_pin_0" { - SKL_TKN_UUID "141, 178, 27, 230, 154, 20, 31, 76, 183, 9, 70, 130, 62, 245, 245, 174" + SKL_TKN_UUID "178, 110, 101, 57, 113, 59, 73, 64, 141, 63, 249, 44, 213, 196, 60, 9" } } @@ -1623,141 +1555,6 @@ SectionWidget."media0_in updwmix 0" { ] } -SectionVendorTuples."media0_in srcintc 0 num_desc" { - tokens "skl_tokens" - - tuples."byte.u8_num_blocks" { - SKL_TKN_U8_NUM_BLOCKS "1" - } -} - -SectionData."media0_in srcintc 0 num_desc" { - tuples "media0_in srcintc 0 num_desc" -} - -SectionVendorTuples."media0_in srcintc 0_size_desc" { - tokens "skl_tokens" - - tuples."byte.u8_block_type" { - SKL_TKN_U8_BLOCK_TYPE "0" - } - - tuples."short.u16_size_desc" { - SKL_TKN_U16_BLOCK_SIZE "388" - } -} - -SectionData."media0_in srcintc 0_size_desc" { - tuples "media0_in srcintc 0_size_desc" -} - -SectionVendorTuples."media0_in srcintc 0" { - tokens "skl_tokens" - - tuples."uuid" { - SKL_TKN_UUID "141, 178, 27, 230, 154, 20, 31, 76, 183, 9, 70, 130, 62, 245, 245, 174" - } - - tuples."byte.u8_data" { - SKL_TKN_U8_IN_PIN_TYPE "0" - SKL_TKN_U8_OUT_PIN_TYPE "0" - SKL_TKN_U8_DYN_IN_PIN "0" - SKL_TKN_U8_DYN_OUT_PIN "0" - SKL_TKN_U8_TIME_SLOT "0" - SKL_TKN_U8_CORE_ID "0" - SKL_TKN_U8_MODULE_TYPE "3" - SKL_TKN_U8_CONN_TYPE "0" - SKL_TKN_U8_HW_CONN_TYPE "1" - SKL_TKN_U8_DEV_TYPE "6" - } - - tuples."short.u16_data" { - SKL_TKN_U16_MOD_INST_ID "0" - } - - tuples."word.u32_data" { - SKL_TKN_U32_VBUS_ID "4294967295" - SKL_TKN_U32_PARAMS_FIXUP "0" - SKL_TKN_U32_CONVERTER "0" - SKL_TKN_U32_PIPE_ID "0" - SKL_TKN_U32_PIPE_CONN_TYPE "1" - SKL_TKN_U32_PIPE_PRIORITY "0" - SKL_TKN_U32_PMODE "0" - SKL_TKN_U32_D0I3_CAPS "1" - SKL_TKN_U32_PROC_DOMAIN "1" - SKL_TKN_U32_PIPE_DIRECTION "0" - SKL_TKN_U32_NUM_CONFIGS "1" - SKL_TKN_U32_DMA_BUF_SIZE "2" - } - - tuples."word._pipe_0" { - SKL_TKN_U32_PIPE_CONFIG_ID "0" - SKL_TKN_U32_PATH_MEM_PGS "8" - } - - tuples."word._pipe_u32_cfg_in_fmt_0" { - SKL_TKN_U32_DIR_PIN_COUNT "0" - SKL_TKN_U32_CFG_FREQ "48000" - } - - tuples."word._pipe_u8_cfg_in_fmt_0" { - SKL_TKN_U8_CFG_BPS "16" - SKL_TKN_U8_CFG_CHAN "2" - } - - tuples."word._pipe_u32_cfg_out_fmt_0" { - SKL_TKN_U32_DIR_PIN_COUNT "1" - SKL_TKN_U32_CFG_FREQ "48000" - } - - tuples."word._pipe_u8_cfg_out_fmt_0" { - SKL_TKN_U8_CFG_BPS "32" - SKL_TKN_U8_CFG_CHAN "2" - } - - tuples."short.u16_pipe_mod_cfg_0" { - SKL_TKN_CFG_MOD_RES_ID "0" - SKL_TKN_CFG_MOD_FMT_ID "0" - } - - tuples."word.in_pin_0" { - SKL_TKN_U32_DIR_PIN_COUNT "0" - SKL_TKN_U32_PIN_MOD_ID "7" - SKL_TKN_U32_PIN_INST_ID "0" - } - - tuples."uuid.in_pin_0" { - SKL_TKN_UUID "12, 6, 248, 66, 47, 131, 191, 77, 178, 71, 81, 233, 97, 153, 123, 52" - } - - tuples."word.out_pin_0" { - SKL_TKN_U32_DIR_PIN_COUNT "1" - SKL_TKN_U32_PIN_MOD_ID "0" - SKL_TKN_U32_PIN_INST_ID "0" - } - - tuples."uuid.out_pin_0" { - SKL_TKN_UUID "178, 110, 101, 57, 113, 59, 73, 64, 141, 63, 249, 44, 213, 196, 60, 9" - } -} - -SectionData."media0_in srcintc 0" { - tuples "media0_in srcintc 0" -} - -SectionWidget."media0_in srcintc 0" { - index "0" - type "pga" - no_pm "true" - event_type "4" - - data [ - "media0_in srcintc 0 num_desc" - "media0_in srcintc 0_size_desc" - "media0_in srcintc 0" - ] -} - SectionVendorTuples."media0_in mi num_desc" { tokens "skl_tokens" @@ -1857,12 +1654,12 @@ SectionVendorTuples."media0_in mi" { tuples."word.in_pin_0" { SKL_TKN_U32_DIR_PIN_COUNT "0" - SKL_TKN_U32_PIN_MOD_ID "8" + SKL_TKN_U32_PIN_MOD_ID "7" SKL_TKN_U32_PIN_INST_ID "0" } tuples."uuid.in_pin_0" { - SKL_TKN_UUID "141, 178, 27, 230, 154, 20, 31, 76, 183, 9, 70, 130, 62, 245, 245, 174" + SKL_TKN_UUID "12, 6, 248, 66, 47, 131, 191, 77, 178, 71, 81, 233, 97, 153, 123, 52" } tuples."word.out_pin_0" { @@ -8013,8 +7810,7 @@ SectionGraph."Pipeline 1 Graph" { lines [ "media0_in cpr 0, , hda-dsp-analog-playback" "media0_in updwmix 0, , media0_in cpr 0" - "media0_in srcintc 0, , media0_in updwmix 0" - "media0_in mi, , media0_in srcintc 0" + "media0_in mi, , media0_in updwmix 0" "codec0_out cpr 1, , codec0_out mo" "codec0_out, , codec0_out cpr 1" "media1_in cpr 2, , hda-dsp-digital-playback" -- 2.17.1

2 1

[PATCH 2/2] ALSA: firewire-digi00x: exclude Avid Adrenaline from detection
by Takashi Sakamoto 23 Aug '20

23 Aug '20

Avid Adrenaline is reported that ALSA firewire-digi00x driver is bound to. However, as long as he investigated, the design of this model is hardly similar to the one of Digi 00x family. It's better to exclude the model from modalias of ALSA firewire-digi00x driver. This commit changes device entries so that the model is excluded. $ python3 crpp < ~/git/am-config-rom/misc/avid-adrenaline.img ROM header and bus information block ----------------------------------------------------------------- 400 04203a9c bus_info_length 4, crc_length 32, crc 15004 404 31333934 bus_name "1394" 408 e064a002 irmc 1, cmc 1, isc 1, bmc 0, cyc_clk_acc 100, max_rec 10 (2048) 40c 00a07e01 company_id 00a07e | 410 00085257 device_id 0100085257 | EUI-64 00a07e0100085257 root directory ----------------------------------------------------------------- 414 0005d08c directory_length 5, crc 53388 418 0300a07e vendor 41c 8100000c --> descriptor leaf at 44c 420 0c008380 node capabilities 424 8d000002 --> eui-64 leaf at 42c 428 d1000004 --> unit directory at 438 eui-64 leaf at 42c ----------------------------------------------------------------- 42c 0002410f leaf_length 2, crc 16655 430 00a07e01 company_id 00a07e | 434 00085257 device_id 0100085257 | EUI-64 00a07e0100085257 unit directory at 438 ----------------------------------------------------------------- 438 0004d6c9 directory_length 4, crc 54985 43c 1200a02d specifier id: 1394 TA 440 13014001 version: Vender Unique and AV/C 444 17000001 model 448 81000009 --> descriptor leaf at 46c descriptor leaf at 44c ----------------------------------------------------------------- 44c 00077205 leaf_length 7, crc 29189 450 00000000 textual descriptor 454 00000000 minimal ASCII 458 41766964 "Avid" 45c 20546563 " Tec" 460 686e6f6c "hnol" 464 6f677900 "ogy" 468 00000000 descriptor leaf at 46c ----------------------------------------------------------------- 46c 000599a5 leaf_length 5, crc 39333 470 00000000 textual descriptor 474 00000000 minimal ASCII 478 41647265 "Adre" 47c 6e616c69 "nali" 480 6e650000 "ne" Reported-by: Simon Wood <simon(a)mungewell.org> Fixes: 9edf723fd858 ("ALSA: firewire-digi00x: add skeleton for Digi 002/003 family") Cc: <stable(a)vger.kernel.org> # 4.4+ Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/digi00x/digi00x.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/firewire/digi00x/digi00x.c b/sound/firewire/digi00x/digi00x.c index c84b913a9fe0..ab8408966ec3 100644 --- a/sound/firewire/digi00x/digi00x.c +++ b/sound/firewire/digi00x/digi00x.c @@ -14,6 +14,7 @@ MODULE_LICENSE("GPL v2"); #define VENDOR_DIGIDESIGN 0x00a07e #define MODEL_CONSOLE 0x000001 #define MODEL_RACK 0x000002 +#define SPEC_VERSION 0x000001 static int name_card(struct snd_dg00x *dg00x) { @@ -175,14 +176,18 @@ static const struct ieee1394_device_id snd_dg00x_id_table[] = { /* Both of 002/003 use the same ID. */ { .match_flags = IEEE1394_MATCH_VENDOR_ID | + IEEE1394_MATCH_VERSION | IEEE1394_MATCH_MODEL_ID, .vendor_id = VENDOR_DIGIDESIGN, + .version = SPEC_VERSION, .model_id = MODEL_CONSOLE, }, { .match_flags = IEEE1394_MATCH_VENDOR_ID | + IEEE1394_MATCH_VERSION | IEEE1394_MATCH_MODEL_ID, .vendor_id = VENDOR_DIGIDESIGN, + .version = SPEC_VERSION, .model_id = MODEL_RACK, }, {} -- 2.25.1

2 1

[PATCH 1/2] ALSA; firewire-tascam: exclude Tascam FE-8 from detection
by Takashi Sakamoto 23 Aug '20

23 Aug '20

Tascam FE-8 is known to support communication by asynchronous transaction only. The support can be implemented in userspace application and snd-firewire-ctl-services project has the support. However, ALSA firewire-tascam driver is bound to the model. This commit changes device entries so that the model is excluded. In a commit 53b3ffee7885 ("ALSA: firewire-tascam: change device probing processing"), I addressed to the concern that version field in configuration differs depending on installed firmware. However, as long as I checked, the version number is fixed. It's safe to return version number back to modalias. Fixes: 53b3ffee7885 ("ALSA: firewire-tascam: change device probing processing") Cc: <stable(a)vger.kernel.org> # 4.4+ Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/tascam/tascam.c | 33 +++++++++++++++++++++++++++++---- 1 file changed, 29 insertions(+), 4 deletions(-) diff --git a/sound/firewire/tascam/tascam.c b/sound/firewire/tascam/tascam.c index 5dac0d9fc58e..75f2edd8e78f 100644 --- a/sound/firewire/tascam/tascam.c +++ b/sound/firewire/tascam/tascam.c @@ -39,9 +39,6 @@ static const struct snd_tscm_spec model_specs[] = { .midi_capture_ports = 2, .midi_playback_ports = 4, }, - // This kernel module doesn't support FE-8 because the most of features - // can be implemented in userspace without any specific support of this - // module. }; static int identify_model(struct snd_tscm *tscm) @@ -211,11 +208,39 @@ static void snd_tscm_remove(struct fw_unit *unit) } static const struct ieee1394_device_id snd_tscm_id_table[] = { + // Tascam, FW-1884. + { + .match_flags = IEEE1394_MATCH_VENDOR_ID | + IEEE1394_MATCH_SPECIFIER_ID | + IEEE1394_MATCH_VERSION, + .vendor_id = 0x00022e, + .specifier_id = 0x00022e, + .version = 0x800000, + }, + // Tascam, FE-8 (.version = 0x800001) + // This kernel module doesn't support FE-8 because the most of features + // can be implemented in userspace without any specific support of this + // module. + // + // .version = 0x800002 is unknown. + // + // Tascam, FW-1082. + { + .match_flags = IEEE1394_MATCH_VENDOR_ID | + IEEE1394_MATCH_SPECIFIER_ID | + IEEE1394_MATCH_VERSION, + .vendor_id = 0x00022e, + .specifier_id = 0x00022e, + .version = 0x800003, + }, + // Tascam, FW-1804. { .match_flags = IEEE1394_MATCH_VENDOR_ID | - IEEE1394_MATCH_SPECIFIER_ID, + IEEE1394_MATCH_SPECIFIER_ID | + IEEE1394_MATCH_VERSION, .vendor_id = 0x00022e, .specifier_id = 0x00022e, + .version = 0x800004, }, {} }; -- 2.25.1

2 1

Is raw MIDI API really raw ? (Running status)
by GitHub issues - opened 23 Aug '20

23 Aug '20

alsa-project/alsa-lib issue #80 was opened from MusicMaker: A device is sending always 3 bytes for note off/on event. The host application uses a virtual port to receive the data The ALSA RAW MIDI API seems to remove the status byte if the channel of the status byte is the same. Once the channel changes, the RAW MIDI API provide the status byte to the higher level layer. Is this the bahaviour of the RAW MIDI API ? if yes then raw isn't actually raw. Can this be disabled ? Could not find a parameter at port open to do this. (not using the sequencer) Issue URL : https://github.com/alsa-project/alsa-lib/issues/80 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH 1/2] ucm: Add JackSwitch definition
by Curtis Malainey 22 Aug '20

22 Aug '20

Some codecs can differentiate headsets based on the number of rings. Should we choose to differentiate the config we will need to be able to select based on the jack event code. Signed-off-by: Curtis Malainey <cujomalainey(a)chromium.org> --- include/use-case.h | 3 +++ 1 file changed, 3 insertions(+) diff --git a/include/use-case.h b/include/use-case.h index 2363c3ed..80148a7b 100644 --- a/include/use-case.h +++ b/include/use-case.h @@ -386,6 +386,9 @@ int snd_use_case_get_list(snd_use_case_mgr_t *uc_mgr, * to e.g. automatically mute speakers when headphones are plugged in, * but that's application policy configuration that doesn't belong * to UCM configuration files. + * - JackSwitch + * - Represents the jack switch value for this device. Value is a base 10 + * value of a switch code in input_event_codes.h * - MinBufferLevel * - This is used on platform where reported buffer level is not accurate. * E.g. "512", which holds 512 samples in device buffer. Note: this will -- 2.28.0.297.g1956fa8f8d-goog

3 5

[PATCH 00/20] ALSA: hda/ca0132 - Add AE-7 support/Control cleanup.
by Connor McAdams 22 Aug '20

22 Aug '20

This patch series adds support for the Sound Blaster AE-7 along with adding new controls and cleaning up the old ones. I have personally tested the Recon3D, Recon3Di, Sound Blaster Z/ZxR, and AE-5 cards to confirm they work with these patches. Another user has confirmed that they work on the AE-7. Connor McAdams (20): ALSA: hda/ca0132 - Cleanup ca0132_mmio_init function. ALSA: hda/ca0132 - Add speaker tuning initialization commands. ALSA: hda/ca0132 - Add surround channel config control. ALSA: hda/ca0132 - Add full-range speaker selection controls. ALSA: hda/ca0132 - Add bass redirection controls. ALSA: hda/ca0132 - Remove surround output selection. ALSA: hda/ca0132 - Clean up ca0132_alt_out_select. ALSA: hda/ca0132 - Add quirk output selection structures. ALSA: hda/ca0132 - Fix Recon3D Center/LFE output. ALSA: hda/ca0132 - Add new quirk ID for SoundBlaster AE-7. ALSA: hda/ca0132 - Add SoundBlaster AE-7 pincfg. ALSA: hda/ca0132 - Set AE-7 bools and select mixer. ALSA: hda/ca0132 - Add ca0132_mmio_init data for SoundBlaster AE-7. ALSA: hda/ca0132 - Add pre-init function for SoundBlaster AE-7. ALSA: hda/ca0132 - Add init data for SoundBlaster AE-7. ALSA: hda/ca0132 - Add DSP setup functions for AE-7. ALSA: hda/ca0132 - Add output selection for SoundBlaster AE-7. ALSA: hda/ca0132 - Add AE-7 microphone selection commands. ALSA: hda/ca0132 - Add AE-7 custom controls. ALSA: hda/ca0132 - Add AE-7 exit commands. sound/pci/hda/patch_ca0132.c | 1794 +++++++++++++++++++++++++++------- 1 file changed, 1430 insertions(+), 364 deletions(-) -- 2.20.1

2 22

aserver + shm plug-in information?
by Stuart Longland 22 Aug '20

22 Aug '20

[Hi all, I sent this to the alsa-user list about 2 months ago, and haven't seen any replies. Hoping someone here might know? Thanks in advance.] Hi all, I'm looking at ways to try and share a sound card over a network in a lossless and low-latency manner. The application here will be to be able to use the sound card on a Raspberry Pi to interface to an amateur radio transceiver, and use that interface to send things like slow-scan television from a more powerful workstation. For this to work, it is utterly *essential* that no audio compression is used. So PulseAudio, JACK, etc… unless I'm very much mistaken, are not options as they use lossy audio CODECs for compression. Audio itself might only be 8kHz at 16-bit linear PCM, mono… so not terribly high bit-rate, and even latency is negotiable, but lossy compression is likely to mess things up. PulseAudio I'd imagine is also asking for trouble where digital modes are concerned, as it'll be trying to re-sample everything to keep a consistent sample rate. I want as little extra DSP on streams as possible. I'd prefer to let PulseAudio just handle "audio" only. I'm considering writing an ALSA PCM plug-in to accomplish this, that would expose a "virtual" sound card on my workstation, and allow me to stream it (by custom protocol, RTP, whatever) to another machine. In researching this, I've stumbled on the `shm` plug-in. https://www.alsa-project.org/alsa-doc/alsa-lib/pcm_plugins.html#pcm_plugins… It apparently talks to something called `aserver`. `man aserver` gives me nothing. Google search just takes me to its source code. There seems to be scant information about what this does, how to configure it, etc. Alternatively, it'd be nice to know how `aserver` and the `shm` plug-in communicate, maybe I can write my own client daemon that acts like `aserver` and looks after streaming to the desired end-point. Alternatively, if people feel I'm barking up the wrong tree, I'm willing to consider alternatives. Regards, -- Stuart Longland (aka Redhatter, VK4MSL) I haven't lost my mind... ...it's backed up on a tape somewhere.

1 0

[PATCH 0/2] Add period size constraint for Atom Chromebook
by Brent Lu 21 Aug '20

21 Aug '20

Two different constraints are implemented: one is in platform's CPU DAI to enforce period sizes which are already used in Android BSP. The other is in Atom Chromebook's machine driver to use 240 as period size. Brent Lu (1): ASoC: intel: atom: Add period size constraint Yu-Hsuan Hsu (1): ASoC: Intel: Add period size constraint on strago board sound/soc/intel/atom/sst-mfld-platform-pcm.c | 15 +++++++++++++++ sound/soc/intel/boards/cht_bsw_max98090_ti.c | 14 +++++++++++++- sound/soc/intel/boards/cht_bsw_rt5645.c | 14 +++++++++++++- 3 files changed, 41 insertions(+), 2 deletions(-) -- 2.7.4

7 53

[PATCH] ASoC: da7219: Fix I/O voltage range configuration during probe
by Adam Thomson 21 Aug '20

21 Aug '20

Previous improvements around handling device and codec level probe functionality added the possibility of the voltage level being undefined for the scenario where the IO voltage retrieved from the regulator supply was below 1.2V, whereas previously the code defaulted to the 2.5V to 3.6V range in that case. This commit restores the default value to avoid this happening. Fixes: aa5b18d1c290 ("ASoC: da7219: Move soft reset handling to codec level probe") Reported-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Adam Thomson <Adam.Thomson.Opensource(a)diasemi.com> --- sound/soc/codecs/da7219.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/da7219.c b/sound/soc/codecs/da7219.c index ddd422c..045aeb6 100644 --- a/sound/soc/codecs/da7219.c +++ b/sound/soc/codecs/da7219.c @@ -1862,6 +1862,9 @@ static int da7219_handle_supplies(struct snd_soc_component *component, return ret; } + /* Default to upper range */ + *io_voltage_lvl = DA7219_IO_VOLTAGE_LEVEL_2_5V_3_6V; + /* Determine VDDIO voltage provided */ vddio = da7219->supplies[DA7219_SUPPLY_VDDIO].consumer; ret = regulator_get_voltage(vddio); @@ -1869,8 +1872,6 @@ static int da7219_handle_supplies(struct snd_soc_component *component, dev_warn(component->dev, "Invalid VDDIO voltage\n"); else if (ret < 2800000) *io_voltage_lvl = DA7219_IO_VOLTAGE_LEVEL_1_2V_2_8V; - else - *io_voltage_lvl = DA7219_IO_VOLTAGE_LEVEL_2_5V_3_6V; /* Enable main supplies */ ret = regulator_bulk_enable(DA7219_NUM_SUPPLIES, da7219->supplies); -- 1.9.1

2 1

[PATCH] ASoC: qcom: Set card->owner to avoid warnings
by Stephan Gerhold 21 Aug '20

21 Aug '20

On Linux 5.9-rc1 I get the following warning with apq8016-sbc: WARNING: CPU: 2 PID: 69 at sound/core/init.c:207 snd_card_new+0x36c/0x3b0 [snd] CPU: 2 PID: 69 Comm: kworker/2:1 Not tainted 5.9.0-rc1 #1 Workqueue: events deferred_probe_work_func pc : snd_card_new+0x36c/0x3b0 [snd] lr : snd_card_new+0xf4/0x3b0 [snd] Call trace: snd_card_new+0x36c/0x3b0 [snd] snd_soc_bind_card+0x340/0x9a0 [snd_soc_core] snd_soc_register_card+0xf4/0x110 [snd_soc_core] devm_snd_soc_register_card+0x44/0xa0 [snd_soc_core] apq8016_sbc_platform_probe+0x11c/0x140 [snd_soc_apq8016_sbc] This warning was introduced in commit 81033c6b584b ("ALSA: core: Warn on empty module"). It looks like we are supposed to set card->owner to THIS_MODULE. Fix this for all the qcom ASoC drivers. Cc: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Fixes: 79119c798649 ("ASoC: qcom: Add Storm machine driver") Fixes: bdb052e81f62 ("ASoC: qcom: add apq8016 sound card support") Fixes: a6f933f63f2f ("ASoC: qcom: apq8096: Add db820c machine driver") Fixes: 6b1687bf76ef ("ASoC: qcom: add sdm845 sound card support") Signed-off-by: Stephan Gerhold <stephan(a)gerhold.net> --- sound/soc/qcom/apq8016_sbc.c | 1 + sound/soc/qcom/apq8096.c | 1 + sound/soc/qcom/sdm845.c | 1 + sound/soc/qcom/storm.c | 1 + 4 files changed, 4 insertions(+) diff --git a/sound/soc/qcom/apq8016_sbc.c b/sound/soc/qcom/apq8016_sbc.c index 083413abc2f6..575e2aefefe3 100644 --- a/sound/soc/qcom/apq8016_sbc.c +++ b/sound/soc/qcom/apq8016_sbc.c @@ -143,6 +143,7 @@ static int apq8016_sbc_platform_probe(struct platform_device *pdev) card = &data->card; card->dev = dev; + card->owner = THIS_MODULE; card->dapm_widgets = apq8016_sbc_dapm_widgets; card->num_dapm_widgets = ARRAY_SIZE(apq8016_sbc_dapm_widgets); diff --git a/sound/soc/qcom/apq8096.c b/sound/soc/qcom/apq8096.c index 253549600c5a..1a69baefc5ce 100644 --- a/sound/soc/qcom/apq8096.c +++ b/sound/soc/qcom/apq8096.c @@ -114,6 +114,7 @@ static int apq8096_platform_probe(struct platform_device *pdev) return -ENOMEM; card->dev = dev; + card->owner = THIS_MODULE; dev_set_drvdata(dev, card); ret = qcom_snd_parse_of(card); if (ret) diff --git a/sound/soc/qcom/sdm845.c b/sound/soc/qcom/sdm845.c index 0d10fba53945..ab1bf23c21a6 100644 --- a/sound/soc/qcom/sdm845.c +++ b/sound/soc/qcom/sdm845.c @@ -555,6 +555,7 @@ static int sdm845_snd_platform_probe(struct platform_device *pdev) card->dapm_widgets = sdm845_snd_widgets; card->num_dapm_widgets = ARRAY_SIZE(sdm845_snd_widgets); card->dev = dev; + card->owner = THIS_MODULE; dev_set_drvdata(dev, card); ret = qcom_snd_parse_of(card); if (ret) diff --git a/sound/soc/qcom/storm.c b/sound/soc/qcom/storm.c index c0c388d4db82..80c9cf2f254a 100644 --- a/sound/soc/qcom/storm.c +++ b/sound/soc/qcom/storm.c @@ -96,6 +96,7 @@ static int storm_platform_probe(struct platform_device *pdev) return -ENOMEM; card->dev = &pdev->dev; + card->owner = THIS_MODULE; ret = snd_soc_of_parse_card_name(card, "qcom,model"); if (ret) { -- 2.28.0

2 1

[PATCH 1/2] ASoC: bindings: fsl-asoc-card: add compatible string for TLV320AIC32x4 codec
by Matthias Schiffer 21 Aug '20

21 Aug '20

The TLV320AIC32x4 is commonly used on TQ-Systems starterkit mainboards for i.MX-based SoMs (i.MX6Q/DL, i.MX6UL, i.MX7) and LS1021A. Signed-off-by: Matthias Schiffer <matthias.schiffer(a)ew.tq-group.com> --- Documentation/devicetree/bindings/sound/fsl-asoc-card.txt | 2 ++ 1 file changed, 2 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt b/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt index 63ebf52b43e8..f339be62e7e4 100644 --- a/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt +++ b/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt @@ -38,6 +38,8 @@ The compatible list for this generic sound card currently: "fsl,imx-audio-wm8524" + "fsl,imx-audio-tlv320aic32x4" + Required properties: - compatible : Contains one of entries in the compatible list. -- 2.17.1

3 4

[PATCH] ASoC: intel: sof_sdw: add .exit callback function
by Bard Liao 21 Aug '20

21 Aug '20

We may allocate some resources in sof_sdw_codec_info .init function. Adding a corresponding .exit function can help to release these resources. Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> --- sound/soc/intel/boards/sof_sdw.c | 31 +++++++++++++++++++++++++ sound/soc/intel/boards/sof_sdw_common.h | 1 + 2 files changed, 32 insertions(+) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index 2463d432bf4d..4bc1ed757009 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -1032,12 +1032,43 @@ static int mc_probe(struct platform_device *pdev) return ret; } +static int mc_remove(struct platform_device *pdev) +{ + struct snd_soc_card *card = platform_get_drvdata(pdev); + struct snd_soc_dai_link *link; + int ret; + int i, j; + + for (i = 0; i < ARRAY_SIZE(codec_info_list); i++) { + if (!codec_info_list[i].exit) + continue; + /* + * We don't need to call .exit function if there is no matched + * dai link found. + */ + for_each_card_prelinks(card, j, link) { + if (!strcmp(link->codecs[0].dai_name, + codec_info_list[i].dai_name)) { + ret = codec_info_list[i].exit(&pdev->dev, link); + if (ret) + dev_warn(&pdev->dev, + "codec exit failed %d\n", + ret); + break; + } + } + } + + return 0; +} + static struct platform_driver sof_sdw_driver = { .driver = { .name = "sof_sdw", .pm = &snd_soc_pm_ops, }, .probe = mc_probe, + .remove = mc_remove, }; module_platform_driver(sof_sdw_driver); diff --git a/sound/soc/intel/boards/sof_sdw_common.h b/sound/soc/intel/boards/sof_sdw_common.h index 12e32439ba46..76d6c0c3839d 100644 --- a/sound/soc/intel/boards/sof_sdw_common.h +++ b/sound/soc/intel/boards/sof_sdw_common.h @@ -63,6 +63,7 @@ struct sof_sdw_codec_info { struct sof_sdw_codec_info *info, bool playback); + int (*exit)(struct device *dev, struct snd_soc_dai_link *dai_link); bool late_probe; int (*codec_card_late_probe)(struct snd_soc_card *card); }; -- 2.17.1

2 1

[PATCH AUTOSEL 4.4 01/22] ALSA: pci: delete repeated words in comments
by Sasha Levin 21 Aug '20

21 Aug '20

From: Randy Dunlap <rdunlap(a)infradead.org> [ Upstream commit c7fabbc51352f50cc58242a6dc3b9c1a3599849b ] Drop duplicated words in sound/pci/. {and, the, at} Signed-off-by: Randy Dunlap <rdunlap(a)infradead.org> Link: https://lore.kernel.org/r/20200806021926.32418-1-rdunlap@infradead.org Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/cs46xx/cs46xx_lib.c | 2 +- sound/pci/cs46xx/dsp_spos_scb_lib.c | 2 +- sound/pci/hda/hda_codec.c | 2 +- sound/pci/hda/hda_generic.c | 2 +- sound/pci/hda/patch_sigmatel.c | 2 +- sound/pci/ice1712/prodigy192.c | 2 +- sound/pci/oxygen/xonar_dg.c | 2 +- 7 files changed, 7 insertions(+), 7 deletions(-) diff --git a/sound/pci/cs46xx/cs46xx_lib.c b/sound/pci/cs46xx/cs46xx_lib.c index 2706f271a83b0..8a174c170e0aa 100644 --- a/sound/pci/cs46xx/cs46xx_lib.c +++ b/sound/pci/cs46xx/cs46xx_lib.c @@ -780,7 +780,7 @@ static void snd_cs46xx_set_capture_sample_rate(struct snd_cs46xx *chip, unsigned rate = 48000 / 9; /* - * We can not capture at at rate greater than the Input Rate (48000). + * We can not capture at a rate greater than the Input Rate (48000). * Return an error if an attempt is made to stray outside that limit. */ if (rate > 48000) diff --git a/sound/pci/cs46xx/dsp_spos_scb_lib.c b/sound/pci/cs46xx/dsp_spos_scb_lib.c index 7488e1b7a7707..4e726d39b05d1 100644 --- a/sound/pci/cs46xx/dsp_spos_scb_lib.c +++ b/sound/pci/cs46xx/dsp_spos_scb_lib.c @@ -1742,7 +1742,7 @@ int cs46xx_iec958_pre_open (struct snd_cs46xx *chip) struct dsp_spos_instance * ins = chip->dsp_spos_instance; if ( ins->spdif_status_out & DSP_SPDIF_STATUS_OUTPUT_ENABLED ) { - /* remove AsynchFGTxSCB and and PCMSerialInput_II */ + /* remove AsynchFGTxSCB and PCMSerialInput_II */ cs46xx_dsp_disable_spdif_out (chip); /* save state */ diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 825d9b27dbe12..4962a9d8a572b 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -3496,7 +3496,7 @@ EXPORT_SYMBOL_GPL(snd_hda_set_power_save); * @nid: NID to check / update * * Check whether the given NID is in the amp list. If it's in the list, - * check the current AMP status, and update the the power-status according + * check the current AMP status, and update the power-status according * to the mute status. * * This function is supposed to be set or called from the check_power_status diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 869c322ddae31..7cd1047a4edf3 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -837,7 +837,7 @@ static void activate_amp_in(struct hda_codec *codec, struct nid_path *path, } } -/* sync power of each widget in the the given path */ +/* sync power of each widget in the given path */ static hda_nid_t path_power_update(struct hda_codec *codec, struct nid_path *path, bool allow_powerdown) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index d1a6d20ace0da..80b72d0702c5e 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -862,7 +862,7 @@ static int stac_auto_create_beep_ctls(struct hda_codec *codec, static struct snd_kcontrol_new beep_vol_ctl = HDA_CODEC_VOLUME(NULL, 0, 0, 0); - /* check for mute support for the the amp */ + /* check for mute support for the amp */ if ((caps & AC_AMPCAP_MUTE) >> AC_AMPCAP_MUTE_SHIFT) { const struct snd_kcontrol_new *temp; if (spec->anabeep_nid == nid) diff --git a/sound/pci/ice1712/prodigy192.c b/sound/pci/ice1712/prodigy192.c index 3919aed39ca03..5e52086d7b986 100644 --- a/sound/pci/ice1712/prodigy192.c +++ b/sound/pci/ice1712/prodigy192.c @@ -31,7 +31,7 @@ * Experimentally I found out that only a combination of * OCKS0=1, OCKS1=1 (128fs, 64fs output) and ice1724 - * VT1724_MT_I2S_MCLK_128X=0 (256fs input) yields correct - * sampling rate. That means the the FPGA doubles the + * sampling rate. That means that the FPGA doubles the * MCK01 rate. * * Copyright (c) 2003 Takashi Iwai <tiwai(a)suse.de> diff --git a/sound/pci/oxygen/xonar_dg.c b/sound/pci/oxygen/xonar_dg.c index 4cf3200e988b0..df44135e1b0c9 100644 --- a/sound/pci/oxygen/xonar_dg.c +++ b/sound/pci/oxygen/xonar_dg.c @@ -39,7 +39,7 @@ * GPIO 4 <- headphone detect * GPIO 5 -> enable ADC analog circuit for the left channel * GPIO 6 -> enable ADC analog circuit for the right channel - * GPIO 7 -> switch green rear output jack between CS4245 and and the first + * GPIO 7 -> switch green rear output jack between CS4245 and the first * channel of CS4361 (mechanical relay) * GPIO 8 -> enable output to speakers * -- 2.25.1

1 1

[PATCH AUTOSEL 4.9 01/26] ALSA: pci: delete repeated words in comments
by Sasha Levin 21 Aug '20

21 Aug '20

From: Randy Dunlap <rdunlap(a)infradead.org> [ Upstream commit c7fabbc51352f50cc58242a6dc3b9c1a3599849b ] Drop duplicated words in sound/pci/. {and, the, at} Signed-off-by: Randy Dunlap <rdunlap(a)infradead.org> Link: https://lore.kernel.org/r/20200806021926.32418-1-rdunlap@infradead.org Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/cs46xx/cs46xx_lib.c | 2 +- sound/pci/cs46xx/dsp_spos_scb_lib.c | 2 +- sound/pci/hda/hda_codec.c | 2 +- sound/pci/hda/hda_generic.c | 2 +- sound/pci/hda/patch_sigmatel.c | 2 +- sound/pci/ice1712/prodigy192.c | 2 +- sound/pci/oxygen/xonar_dg.c | 2 +- 7 files changed, 7 insertions(+), 7 deletions(-) diff --git a/sound/pci/cs46xx/cs46xx_lib.c b/sound/pci/cs46xx/cs46xx_lib.c index 528102cc2d5d0..d824ff4ae3e3b 100644 --- a/sound/pci/cs46xx/cs46xx_lib.c +++ b/sound/pci/cs46xx/cs46xx_lib.c @@ -780,7 +780,7 @@ static void snd_cs46xx_set_capture_sample_rate(struct snd_cs46xx *chip, unsigned rate = 48000 / 9; /* - * We can not capture at at rate greater than the Input Rate (48000). + * We can not capture at a rate greater than the Input Rate (48000). * Return an error if an attempt is made to stray outside that limit. */ if (rate > 48000) diff --git a/sound/pci/cs46xx/dsp_spos_scb_lib.c b/sound/pci/cs46xx/dsp_spos_scb_lib.c index 7488e1b7a7707..4e726d39b05d1 100644 --- a/sound/pci/cs46xx/dsp_spos_scb_lib.c +++ b/sound/pci/cs46xx/dsp_spos_scb_lib.c @@ -1742,7 +1742,7 @@ int cs46xx_iec958_pre_open (struct snd_cs46xx *chip) struct dsp_spos_instance * ins = chip->dsp_spos_instance; if ( ins->spdif_status_out & DSP_SPDIF_STATUS_OUTPUT_ENABLED ) { - /* remove AsynchFGTxSCB and and PCMSerialInput_II */ + /* remove AsynchFGTxSCB and PCMSerialInput_II */ cs46xx_dsp_disable_spdif_out (chip); /* save state */ diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index cbe0248225c1c..4e67614f15f8e 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -3496,7 +3496,7 @@ EXPORT_SYMBOL_GPL(snd_hda_set_power_save); * @nid: NID to check / update * * Check whether the given NID is in the amp list. If it's in the list, - * check the current AMP status, and update the the power-status according + * check the current AMP status, and update the power-status according * to the mute status. * * This function is supposed to be set or called from the check_power_status diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 949c90a859fab..184089c5e8cbc 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -820,7 +820,7 @@ static void activate_amp_in(struct hda_codec *codec, struct nid_path *path, } } -/* sync power of each widget in the the given path */ +/* sync power of each widget in the given path */ static hda_nid_t path_power_update(struct hda_codec *codec, struct nid_path *path, bool allow_powerdown) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index d1a6d20ace0da..80b72d0702c5e 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -862,7 +862,7 @@ static int stac_auto_create_beep_ctls(struct hda_codec *codec, static struct snd_kcontrol_new beep_vol_ctl = HDA_CODEC_VOLUME(NULL, 0, 0, 0); - /* check for mute support for the the amp */ + /* check for mute support for the amp */ if ((caps & AC_AMPCAP_MUTE) >> AC_AMPCAP_MUTE_SHIFT) { const struct snd_kcontrol_new *temp; if (spec->anabeep_nid == nid) diff --git a/sound/pci/ice1712/prodigy192.c b/sound/pci/ice1712/prodigy192.c index 3919aed39ca03..5e52086d7b986 100644 --- a/sound/pci/ice1712/prodigy192.c +++ b/sound/pci/ice1712/prodigy192.c @@ -31,7 +31,7 @@ * Experimentally I found out that only a combination of * OCKS0=1, OCKS1=1 (128fs, 64fs output) and ice1724 - * VT1724_MT_I2S_MCLK_128X=0 (256fs input) yields correct - * sampling rate. That means the the FPGA doubles the + * sampling rate. That means that the FPGA doubles the * MCK01 rate. * * Copyright (c) 2003 Takashi Iwai <tiwai(a)suse.de> diff --git a/sound/pci/oxygen/xonar_dg.c b/sound/pci/oxygen/xonar_dg.c index 4cf3200e988b0..df44135e1b0c9 100644 --- a/sound/pci/oxygen/xonar_dg.c +++ b/sound/pci/oxygen/xonar_dg.c @@ -39,7 +39,7 @@ * GPIO 4 <- headphone detect * GPIO 5 -> enable ADC analog circuit for the left channel * GPIO 6 -> enable ADC analog circuit for the right channel - * GPIO 7 -> switch green rear output jack between CS4245 and and the first + * GPIO 7 -> switch green rear output jack between CS4245 and the first * channel of CS4361 (mechanical relay) * GPIO 8 -> enable output to speakers * -- 2.25.1

1 1

[PATCH AUTOSEL 4.14 01/30] ALSA: pci: delete repeated words in comments
by Sasha Levin 21 Aug '20

21 Aug '20

From: Randy Dunlap <rdunlap(a)infradead.org> [ Upstream commit c7fabbc51352f50cc58242a6dc3b9c1a3599849b ] Drop duplicated words in sound/pci/. {and, the, at} Signed-off-by: Randy Dunlap <rdunlap(a)infradead.org> Link: https://lore.kernel.org/r/20200806021926.32418-1-rdunlap@infradead.org Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/cs46xx/cs46xx_lib.c | 2 +- sound/pci/cs46xx/dsp_spos_scb_lib.c | 2 +- sound/pci/hda/hda_codec.c | 2 +- sound/pci/hda/hda_generic.c | 2 +- sound/pci/hda/patch_sigmatel.c | 2 +- sound/pci/ice1712/prodigy192.c | 2 +- sound/pci/oxygen/xonar_dg.c | 2 +- 7 files changed, 7 insertions(+), 7 deletions(-) diff --git a/sound/pci/cs46xx/cs46xx_lib.c b/sound/pci/cs46xx/cs46xx_lib.c index 0020fd0efc466..09c547f4cc186 100644 --- a/sound/pci/cs46xx/cs46xx_lib.c +++ b/sound/pci/cs46xx/cs46xx_lib.c @@ -780,7 +780,7 @@ static void snd_cs46xx_set_capture_sample_rate(struct snd_cs46xx *chip, unsigned rate = 48000 / 9; /* - * We can not capture at at rate greater than the Input Rate (48000). + * We can not capture at a rate greater than the Input Rate (48000). * Return an error if an attempt is made to stray outside that limit. */ if (rate > 48000) diff --git a/sound/pci/cs46xx/dsp_spos_scb_lib.c b/sound/pci/cs46xx/dsp_spos_scb_lib.c index 7488e1b7a7707..4e726d39b05d1 100644 --- a/sound/pci/cs46xx/dsp_spos_scb_lib.c +++ b/sound/pci/cs46xx/dsp_spos_scb_lib.c @@ -1742,7 +1742,7 @@ int cs46xx_iec958_pre_open (struct snd_cs46xx *chip) struct dsp_spos_instance * ins = chip->dsp_spos_instance; if ( ins->spdif_status_out & DSP_SPDIF_STATUS_OUTPUT_ENABLED ) { - /* remove AsynchFGTxSCB and and PCMSerialInput_II */ + /* remove AsynchFGTxSCB and PCMSerialInput_II */ cs46xx_dsp_disable_spdif_out (chip); /* save state */ diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 7d65fe31c8257..a56f018d586f5 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -3394,7 +3394,7 @@ EXPORT_SYMBOL_GPL(snd_hda_set_power_save); * @nid: NID to check / update * * Check whether the given NID is in the amp list. If it's in the list, - * check the current AMP status, and update the the power-status according + * check the current AMP status, and update the power-status according * to the mute status. * * This function is supposed to be set or called from the check_power_status diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 28ef409a9e6ae..9dee657ce9e27 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -823,7 +823,7 @@ static void activate_amp_in(struct hda_codec *codec, struct nid_path *path, } } -/* sync power of each widget in the the given path */ +/* sync power of each widget in the given path */ static hda_nid_t path_power_update(struct hda_codec *codec, struct nid_path *path, bool allow_powerdown) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 7cd147411b22d..f7896a9ae3d65 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -863,7 +863,7 @@ static int stac_auto_create_beep_ctls(struct hda_codec *codec, static struct snd_kcontrol_new beep_vol_ctl = HDA_CODEC_VOLUME(NULL, 0, 0, 0); - /* check for mute support for the the amp */ + /* check for mute support for the amp */ if ((caps & AC_AMPCAP_MUTE) >> AC_AMPCAP_MUTE_SHIFT) { const struct snd_kcontrol_new *temp; if (spec->anabeep_nid == nid) diff --git a/sound/pci/ice1712/prodigy192.c b/sound/pci/ice1712/prodigy192.c index 3919aed39ca03..5e52086d7b986 100644 --- a/sound/pci/ice1712/prodigy192.c +++ b/sound/pci/ice1712/prodigy192.c @@ -31,7 +31,7 @@ * Experimentally I found out that only a combination of * OCKS0=1, OCKS1=1 (128fs, 64fs output) and ice1724 - * VT1724_MT_I2S_MCLK_128X=0 (256fs input) yields correct - * sampling rate. That means the the FPGA doubles the + * sampling rate. That means that the FPGA doubles the * MCK01 rate. * * Copyright (c) 2003 Takashi Iwai <tiwai(a)suse.de> diff --git a/sound/pci/oxygen/xonar_dg.c b/sound/pci/oxygen/xonar_dg.c index 4cf3200e988b0..df44135e1b0c9 100644 --- a/sound/pci/oxygen/xonar_dg.c +++ b/sound/pci/oxygen/xonar_dg.c @@ -39,7 +39,7 @@ * GPIO 4 <- headphone detect * GPIO 5 -> enable ADC analog circuit for the left channel * GPIO 6 -> enable ADC analog circuit for the right channel - * GPIO 7 -> switch green rear output jack between CS4245 and and the first + * GPIO 7 -> switch green rear output jack between CS4245 and the first * channel of CS4361 (mechanical relay) * GPIO 8 -> enable output to speakers * -- 2.25.1

1 1

[PATCH AUTOSEL 4.19 01/38] ALSA: pci: delete repeated words in comments
by Sasha Levin 21 Aug '20

21 Aug '20

From: Randy Dunlap <rdunlap(a)infradead.org> [ Upstream commit c7fabbc51352f50cc58242a6dc3b9c1a3599849b ] Drop duplicated words in sound/pci/. {and, the, at} Signed-off-by: Randy Dunlap <rdunlap(a)infradead.org> Link: https://lore.kernel.org/r/20200806021926.32418-1-rdunlap@infradead.org Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/cs46xx/cs46xx_lib.c | 2 +- sound/pci/cs46xx/dsp_spos_scb_lib.c | 2 +- sound/pci/hda/hda_codec.c | 2 +- sound/pci/hda/hda_generic.c | 2 +- sound/pci/hda/patch_sigmatel.c | 2 +- sound/pci/ice1712/prodigy192.c | 2 +- sound/pci/oxygen/xonar_dg.c | 2 +- 7 files changed, 7 insertions(+), 7 deletions(-) diff --git a/sound/pci/cs46xx/cs46xx_lib.c b/sound/pci/cs46xx/cs46xx_lib.c index 146e1a3498c73..419da70cd942a 100644 --- a/sound/pci/cs46xx/cs46xx_lib.c +++ b/sound/pci/cs46xx/cs46xx_lib.c @@ -780,7 +780,7 @@ static void snd_cs46xx_set_capture_sample_rate(struct snd_cs46xx *chip, unsigned rate = 48000 / 9; /* - * We can not capture at at rate greater than the Input Rate (48000). + * We can not capture at a rate greater than the Input Rate (48000). * Return an error if an attempt is made to stray outside that limit. */ if (rate > 48000) diff --git a/sound/pci/cs46xx/dsp_spos_scb_lib.c b/sound/pci/cs46xx/dsp_spos_scb_lib.c index 8d0a3d3573457..8ef51a29380af 100644 --- a/sound/pci/cs46xx/dsp_spos_scb_lib.c +++ b/sound/pci/cs46xx/dsp_spos_scb_lib.c @@ -1739,7 +1739,7 @@ int cs46xx_iec958_pre_open (struct snd_cs46xx *chip) struct dsp_spos_instance * ins = chip->dsp_spos_instance; if ( ins->spdif_status_out & DSP_SPDIF_STATUS_OUTPUT_ENABLED ) { - /* remove AsynchFGTxSCB and and PCMSerialInput_II */ + /* remove AsynchFGTxSCB and PCMSerialInput_II */ cs46xx_dsp_disable_spdif_out (chip); /* save state */ diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index f3a6b1d869d8a..dbeb62362f1c3 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -3410,7 +3410,7 @@ EXPORT_SYMBOL_GPL(snd_hda_set_power_save); * @nid: NID to check / update * * Check whether the given NID is in the amp list. If it's in the list, - * check the current AMP status, and update the the power-status according + * check the current AMP status, and update the power-status according * to the mute status. * * This function is supposed to be set or called from the check_power_status diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 2609161707a41..97adb7e340f99 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -825,7 +825,7 @@ static void activate_amp_in(struct hda_codec *codec, struct nid_path *path, } } -/* sync power of each widget in the the given path */ +/* sync power of each widget in the given path */ static hda_nid_t path_power_update(struct hda_codec *codec, struct nid_path *path, bool allow_powerdown) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index d8168aa2cef38..85c33f528d7b3 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -845,7 +845,7 @@ static int stac_auto_create_beep_ctls(struct hda_codec *codec, static struct snd_kcontrol_new beep_vol_ctl = HDA_CODEC_VOLUME(NULL, 0, 0, 0); - /* check for mute support for the the amp */ + /* check for mute support for the amp */ if ((caps & AC_AMPCAP_MUTE) >> AC_AMPCAP_MUTE_SHIFT) { const struct snd_kcontrol_new *temp; if (spec->anabeep_nid == nid) diff --git a/sound/pci/ice1712/prodigy192.c b/sound/pci/ice1712/prodigy192.c index 3919aed39ca03..5e52086d7b986 100644 --- a/sound/pci/ice1712/prodigy192.c +++ b/sound/pci/ice1712/prodigy192.c @@ -31,7 +31,7 @@ * Experimentally I found out that only a combination of * OCKS0=1, OCKS1=1 (128fs, 64fs output) and ice1724 - * VT1724_MT_I2S_MCLK_128X=0 (256fs input) yields correct - * sampling rate. That means the the FPGA doubles the + * sampling rate. That means that the FPGA doubles the * MCK01 rate. * * Copyright (c) 2003 Takashi Iwai <tiwai(a)suse.de> diff --git a/sound/pci/oxygen/xonar_dg.c b/sound/pci/oxygen/xonar_dg.c index 4cf3200e988b0..df44135e1b0c9 100644 --- a/sound/pci/oxygen/xonar_dg.c +++ b/sound/pci/oxygen/xonar_dg.c @@ -39,7 +39,7 @@ * GPIO 4 <- headphone detect * GPIO 5 -> enable ADC analog circuit for the left channel * GPIO 6 -> enable ADC analog circuit for the right channel - * GPIO 7 -> switch green rear output jack between CS4245 and and the first + * GPIO 7 -> switch green rear output jack between CS4245 and the first * channel of CS4361 (mechanical relay) * GPIO 8 -> enable output to speakers * -- 2.25.1

1 3

[PATCH AUTOSEL 5.4 01/48] ALSA: hda/hdmi: Add quirk to force connectivity
by Sasha Levin 21 Aug '20

21 Aug '20

From: Kai-Heng Feng <kai.heng.feng(a)canonical.com> [ Upstream commit cd72c317a0a11f64225b9a3f1fe503bb8c7327b5 ] HDMI on some platforms doesn't enable audio support because its Port Connectivity [31:30] is set to AC_JACK_PORT_NONE: Node 0x05 [Pin Complex] wcaps 0x40778d: 8-Channels Digital Amp-Out CP Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0b000094: OUT Detect HBR HDMI DP Pin Default 0x58560010: [N/A] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0x1, Sequence = 0x0 Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Power states: D0 D3 EPSS Power: setting=D0, actual=D0 Devices: 0 Connection: 3 0x02 0x03* 0x04 For now, use a quirk to force connectivity based on SSID. If there are more platforms affected by the same issue, we can eye for a more generic solution. Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> Link: https://lore.kernel.org/r/20200804155836.16252-1-kai.heng.feng@canonical.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/patch_hdmi.c | 16 +++++++++++++++- 1 file changed, 15 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 908b68fda24c9..a9559fb29e209 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -176,6 +176,7 @@ struct hdmi_spec { bool use_jack_detect; /* jack detection enabled */ bool use_acomp_notifier; /* use eld_notify callback for hotplug */ bool acomp_registered; /* audio component registered in this driver */ + bool force_connect; /* force connectivity */ struct drm_audio_component_audio_ops drm_audio_ops; int (*port2pin)(struct hda_codec *, int); /* reverse port/pin mapping */ @@ -1711,7 +1712,8 @@ static int hdmi_add_pin(struct hda_codec *codec, hda_nid_t pin_nid) * all device entries on the same pin */ config = snd_hda_codec_get_pincfg(codec, pin_nid); - if (get_defcfg_connect(config) == AC_JACK_PORT_NONE) + if (get_defcfg_connect(config) == AC_JACK_PORT_NONE && + !spec->force_connect) return 0; /* @@ -1815,11 +1817,18 @@ static int hdmi_add_cvt(struct hda_codec *codec, hda_nid_t cvt_nid) return 0; } +static const struct snd_pci_quirk force_connect_list[] = { + SND_PCI_QUIRK(0x103c, 0x871a, "HP", 1), + {} +}; + static int hdmi_parse_codec(struct hda_codec *codec) { + struct hdmi_spec *spec = codec->spec; hda_nid_t start_nid; unsigned int caps; int i, nodes; + const struct snd_pci_quirk *q; nodes = snd_hda_get_sub_nodes(codec, codec->core.afg, &start_nid); if (!start_nid || nodes < 0) { @@ -1827,6 +1836,11 @@ static int hdmi_parse_codec(struct hda_codec *codec) return -EINVAL; } + q = snd_pci_quirk_lookup(codec->bus->pci, force_connect_list); + + if (q && q->value) + spec->force_connect = true; + /* * hdmi_add_pin() assumes total amount of converters to * be known, so first discover all converters -- 2.25.1

1 7

[PATCH AUTOSEL 5.7 01/61] ALSA: hda/hdmi: Add quirk to force connectivity
by Sasha Levin 21 Aug '20

21 Aug '20

From: Kai-Heng Feng <kai.heng.feng(a)canonical.com> [ Upstream commit cd72c317a0a11f64225b9a3f1fe503bb8c7327b5 ] HDMI on some platforms doesn't enable audio support because its Port Connectivity [31:30] is set to AC_JACK_PORT_NONE: Node 0x05 [Pin Complex] wcaps 0x40778d: 8-Channels Digital Amp-Out CP Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0b000094: OUT Detect HBR HDMI DP Pin Default 0x58560010: [N/A] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0x1, Sequence = 0x0 Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Power states: D0 D3 EPSS Power: setting=D0, actual=D0 Devices: 0 Connection: 3 0x02 0x03* 0x04 For now, use a quirk to force connectivity based on SSID. If there are more platforms affected by the same issue, we can eye for a more generic solution. Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> Link: https://lore.kernel.org/r/20200804155836.16252-1-kai.heng.feng@canonical.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/patch_hdmi.c | 16 +++++++++++++++- 1 file changed, 15 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 37391c3d2f47c..de9570032e07c 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -160,6 +160,7 @@ struct hdmi_spec { bool use_acomp_notifier; /* use eld_notify callback for hotplug */ bool acomp_registered; /* audio component registered in this driver */ + bool force_connect; /* force connectivity */ struct drm_audio_component_audio_ops drm_audio_ops; int (*port2pin)(struct hda_codec *, int); /* reverse port/pin mapping */ @@ -1700,7 +1701,8 @@ static int hdmi_add_pin(struct hda_codec *codec, hda_nid_t pin_nid) * all device entries on the same pin */ config = snd_hda_codec_get_pincfg(codec, pin_nid); - if (get_defcfg_connect(config) == AC_JACK_PORT_NONE) + if (get_defcfg_connect(config) == AC_JACK_PORT_NONE && + !spec->force_connect) return 0; /* @@ -1802,11 +1804,18 @@ static int hdmi_add_cvt(struct hda_codec *codec, hda_nid_t cvt_nid) return 0; } +static const struct snd_pci_quirk force_connect_list[] = { + SND_PCI_QUIRK(0x103c, 0x871a, "HP", 1), + {} +}; + static int hdmi_parse_codec(struct hda_codec *codec) { + struct hdmi_spec *spec = codec->spec; hda_nid_t start_nid; unsigned int caps; int i, nodes; + const struct snd_pci_quirk *q; nodes = snd_hda_get_sub_nodes(codec, codec->core.afg, &start_nid); if (!start_nid || nodes < 0) { @@ -1814,6 +1823,11 @@ static int hdmi_parse_codec(struct hda_codec *codec) return -EINVAL; } + q = snd_pci_quirk_lookup(codec->bus->pci, force_connect_list); + + if (q && q->value) + spec->force_connect = true; + /* * hdmi_add_pin() assumes total amount of converters to * be known, so first discover all converters -- 2.25.1

1 8

[PATCH AUTOSEL 5.8 01/62] ALSA: hda/hdmi: Add quirk to force connectivity
by Sasha Levin 21 Aug '20

21 Aug '20

From: Kai-Heng Feng <kai.heng.feng(a)canonical.com> [ Upstream commit cd72c317a0a11f64225b9a3f1fe503bb8c7327b5 ] HDMI on some platforms doesn't enable audio support because its Port Connectivity [31:30] is set to AC_JACK_PORT_NONE: Node 0x05 [Pin Complex] wcaps 0x40778d: 8-Channels Digital Amp-Out CP Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0b000094: OUT Detect HBR HDMI DP Pin Default 0x58560010: [N/A] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0x1, Sequence = 0x0 Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Power states: D0 D3 EPSS Power: setting=D0, actual=D0 Devices: 0 Connection: 3 0x02 0x03* 0x04 For now, use a quirk to force connectivity based on SSID. If there are more platforms affected by the same issue, we can eye for a more generic solution. Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> Link: https://lore.kernel.org/r/20200804155836.16252-1-kai.heng.feng@canonical.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/patch_hdmi.c | 16 +++++++++++++++- 1 file changed, 15 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index cd46247988e4d..b62cd3abb8273 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -160,6 +160,7 @@ struct hdmi_spec { bool use_acomp_notifier; /* use eld_notify callback for hotplug */ bool acomp_registered; /* audio component registered in this driver */ + bool force_connect; /* force connectivity */ struct drm_audio_component_audio_ops drm_audio_ops; int (*port2pin)(struct hda_codec *, int); /* reverse port/pin mapping */ @@ -1701,7 +1702,8 @@ static int hdmi_add_pin(struct hda_codec *codec, hda_nid_t pin_nid) * all device entries on the same pin */ config = snd_hda_codec_get_pincfg(codec, pin_nid); - if (get_defcfg_connect(config) == AC_JACK_PORT_NONE) + if (get_defcfg_connect(config) == AC_JACK_PORT_NONE && + !spec->force_connect) return 0; /* @@ -1803,11 +1805,18 @@ static int hdmi_add_cvt(struct hda_codec *codec, hda_nid_t cvt_nid) return 0; } +static const struct snd_pci_quirk force_connect_list[] = { + SND_PCI_QUIRK(0x103c, 0x871a, "HP", 1), + {} +}; + static int hdmi_parse_codec(struct hda_codec *codec) { + struct hdmi_spec *spec = codec->spec; hda_nid_t start_nid; unsigned int caps; int i, nodes; + const struct snd_pci_quirk *q; nodes = snd_hda_get_sub_nodes(codec, codec->core.afg, &start_nid); if (!start_nid || nodes < 0) { @@ -1815,6 +1824,11 @@ static int hdmi_parse_codec(struct hda_codec *codec) return -EINVAL; } + q = snd_pci_quirk_lookup(codec->bus->pci, force_connect_list); + + if (q && q->value) + spec->force_connect = true; + /* * hdmi_add_pin() assumes total amount of converters to * be known, so first discover all converters -- 2.25.1

1 8

[PATCH] soundwire: intel: fix CONFIG_PM and CONFIG_PM_SLEEP confusion
by Bard Liao 21 Aug '20

21 Aug '20

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> When CONFIG_PM_SLEEP is not defined, GCC throws compilation warnings: drivers/soundwire/intel.c:1816:12: warning: ‘intel_resume’ defined but not used [-Wunused-function] 1816 | static int intel_resume(struct device *dev) | ^~~~~~~~~~~~ drivers/soundwire/intel.c:1697:12: warning: ‘intel_suspend’ defined but not used [-Wunused-function] 1697 | static int intel_suspend(struct device *dev) Fix by adding the missing CONFIG_PM_SLEEP. Note that we could move code around and use only 2 ifdefs, but this will generate conflicts so let's do this when all the pm handling is merged. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> --- drivers/soundwire/intel.c | 12 ++++++++++-- 1 file changed, 10 insertions(+), 2 deletions(-) diff --git a/drivers/soundwire/intel.c b/drivers/soundwire/intel.c index dbcbe2708563..a2f0026cb2c1 100644 --- a/drivers/soundwire/intel.c +++ b/drivers/soundwire/intel.c @@ -1525,7 +1525,7 @@ int intel_master_process_wakeen_event(struct platform_device *pdev) * PM calls */ -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP static int intel_suspend(struct device *dev) { @@ -1562,6 +1562,9 @@ static int intel_suspend(struct device *dev) return 0; } +#endif + +#ifdef CONFIG_PM static int intel_suspend_runtime(struct device *dev) { @@ -1624,6 +1627,9 @@ static int intel_suspend_runtime(struct device *dev) return ret; } +#endif + +#ifdef CONFIG_PM_SLEEP static int intel_resume(struct device *dev) { @@ -1691,6 +1697,9 @@ static int intel_resume(struct device *dev) return ret; } +#endif + +#ifdef CONFIG_PM static int intel_resume_runtime(struct device *dev) { @@ -1797,7 +1806,6 @@ static int intel_resume_runtime(struct device *dev) return ret; } - #endif static const struct dev_pm_ops intel_pm = { -- 2.17.1

3 2

[alsa-devel] Attempting to understand odd snd-usb-audio code and behavior
by Keith A. Milner 21 Aug '20

21 Aug '20

Over the last few weeks I've been trying to dig through the snd-usb-audio code to try to get an understanding of it. I've used ftrace and also a local version which I have heavily polluted with kprintf(KERN_INFO statements so I can track what it's doing in practice. I've been playing with this with some USB audio interfaces I have some which work, and some which don't, and trying to understand the differences and, ultimately, what I can do to make the non-functioning ones work. I've got to the point where I have a fairly good understanding of some parts, but I'm struggling to understand why the code is behaving as it does in practice. My observation when plugging the USB device in is that, as expected, it tries to build the audio device, and add components to it including attaching capture and playback streams. It seems to go through this phase without too many surprises, and we get to the standard dmesg output: usbcore: registered new interface driver snd-usb-audio After this, it gets confusing. It seems to open and close the PCMs (e.g, snd_usb_capture_open) several times in a row. Each time it sets up the hardware info, checks and sets the format and then closes the PCM. I don't see anything obviously different between most of these cycles. Here's an example my logging output (hopefully this will make sense): Aug 22 20:04:16 kerneldev kernel: [ 923.140789] Called: pcm.c - snd_usb_capture_open for pcm substream 0 Aug 22 20:04:16 kerneldev kernel: [ 923.140789] Called: pcm.c - snd_usb_pcm_open for pcm substream 0 Aug 22 20:04:16 kerneldev kernel: [ 923.140789] Setting up hw info Aug 22 20:04:16 kerneldev kernel: [ 923.140789] Checking format: Aug 22 20:04:16 kerneldev kernel: [ 923.140790] Interface number = 2 Aug 22 20:04:16 kerneldev kernel: [ 923.140790] alt setting = 1 Aug 22 20:04:16 kerneldev kernel: [ 923.140790] alt setting index = 1 Aug 22 20:04:16 kerneldev kernel: [ 923.140790] format type = 1 Aug 22 20:04:16 kerneldev kernel: [ 923.140790] frame size = 0 Aug 22 20:04:16 kerneldev kernel: [ 923.140791] Setting pt=125 ptmin=125 Aug 22 20:04:16 kerneldev kernel: [ 923.140792] Checking for unusual bit rates Aug 22 20:04:16 kerneldev kernel: [ 923.140794] Called: init.c - snd_card_file_remove Aug 22 20:04:16 kerneldev kernel: [ 923.140921] Called: pcm.c - snd_usb_hw_params for pcm substream 0 Aug 22 20:04:16 kerneldev kernel: [ 923.140945] Setting format: Aug 22 20:04:16 kerneldev kernel: [ 923.140945] Bytes per period = 8816 Aug 22 20:04:16 kerneldev kernel: [ 923.140945] Frames per period = 1102 Aug 22 20:04:16 kerneldev kernel: [ 923.140946] Periods per buffer = 4 Aug 22 20:04:16 kerneldev kernel: [ 923.140946] Channels = 2 Aug 22 20:04:16 kerneldev kernel: [ 923.140946] Rate = 44100 Aug 22 20:04:16 kerneldev kernel: [ 923.141279] Called: endpoint.c - snd_usb_add_endpoint of type 0 with direction 1 Aug 22 20:04:16 kerneldev kernel: [ 923.141292] Called: pcm.c - snd_usb_pcm_prepare for pcm substream 0 Aug 22 20:04:16 kerneldev kernel: [ 923.141293] Called: pcm.c - configure_endpoint Aug 22 20:04:16 kerneldev kernel: [ 923.141294] Called: endpoint.c - snd_usb_endpoint_set_params for endpoint 142, type 0 Aug 22 20:04:16 kerneldev kernel: [ 923.141294] Setting endpoint params for endpoint num 142 Aug 22 20:04:16 kerneldev kernel: [ 923.141294] channels=2 Aug 22 20:04:16 kerneldev kernel: [ 923.141294] period bytes=8816 Aug 22 20:04:16 kerneldev kernel: [ 923.141295] frames per period=1102 Aug 22 20:04:16 kerneldev kernel: [ 923.141295] periods per buffer=4 Aug 22 20:04:16 kerneldev kernel: [ 923.141296] Called: endpoint.c - data_ep_set_params for endpoint 142, type 0 Aug 22 20:04:16 kerneldev kernel: [ 923.141301] Called: pcm.c - snd_usb_pcm_prepare for pcm substream 0 Aug 22 20:04:16 kerneldev kernel: [ 923.142768] Called: init.c - snd_card_file_add Aug 22 20:04:16 kerneldev kernel: [ 923.142772] Called: init.c - snd_card_file_remove Aug 22 20:04:16 kerneldev kernel: [ 923.142774] Called: init.c - snd_card_file_add Aug 22 20:04:16 kerneldev kernel: [ 923.142830] Called: init.c - snd_card_file_remove Aug 22 20:04:16 kerneldev kernel: [ 923.142836] Called: pcm.c - snd_usb_hw_free for pcm substream 0 Aug 22 20:04:16 kerneldev kernel: [ 923.142837] Called: endpoint.c - snd_usb_endpoint_deactivate for endpoint 142, type 0 Aug 22 20:04:16 kerneldev kernel: [ 923.142846] Called: pcm.c - snd_usb_capture_close for pcm substream 0 Aug 22 20:04:16 kerneldev kernel: [ 923.146760] Called: init.c - snd_card_file_remove Aug 22 20:04:16 kerneldev kernel: [ 923.146834] Called: init.c - snd_card_file_add Aug 22 20:04:16 kerneldev kernel: [ 923.146838] Called: init.c - snd_card_file_remove Aug 22 20:04:16 kerneldev kernel: [ 923.146841] Called: init.c - snd_card_file_add Aug 22 20:04:16 kerneldev kernel: [ 923.146845] Called: init.c - snd_card_file_add Aug 22 20:04:16 kerneldev kernel: [ 923.146848] Called: pcm.c - snd_usb_capture_open for pcm substream 0 Aug 22 20:04:16 kerneldev kernel: [ 923.146849] Called: pcm.c - snd_usb_pcm_open for pcm substream 0 Aug 22 20:04:16 kerneldev kernel: [ 923.146849] Setting up hw info It seems to do this several times, before the behavior changing slightly, as it now starts to try to set the endpoint parameters and seems to leave the stream open. However, it then tries to open the stream and set the endpoint parameters again resulting in errors: Aug 22 20:22:00 kerneldev kernel: [ 1987.452945] Called: endpoint.c - snd_usb_endpoint_set_params for endpoint 142, type 0 Aug 22 20:22:00 kerneldev kernel: [ 1987.452945] Setting endpoint params for endpoint num 142 Aug 22 20:22:00 kerneldev kernel: [ 1987.452945] channels=2 Aug 22 20:22:00 kerneldev kernel: [ 1987.452945] period bytes=1768 Aug 22 20:22:00 kerneldev kernel: [ 1987.452946] frames per period=221 Aug 22 20:22:00 kerneldev kernel: [ 1987.452946] periods per buffer=593 Aug 22 20:22:00 kerneldev kernel: [ 1987.452947] usb 1-1: Unable to change format on ep #8e: already in use Maybe this is driven by hotplug or something, but it seems to be performing a lot of pointless activities. Eventually it seems to try to open the PCM playback or capture stream (with some devices it opens playback, with others capture) and then checks the pointer: Aug 22 20:22:00 kerneldev kernel: [ 1987.470194] Called: pcm.c - snd_usb_substream_playback_trigger for pcm substream 0 Aug 22 20:22:00 kerneldev kernel: [ 1987.470198] Called: pcm.c - snd_usb_pcm_pointer for pcm substream 0 Aug 22 20:22:00 kerneldev kernel: [ 1987.470198] Pointer is 0 Aug 22 20:22:00 kerneldev kernel: [ 1987.470656] Called: pcm.c - snd_usb_pcm_pointer for pcm substream 0 Aug 22 20:22:00 kerneldev kernel: [ 1987.470657] Pointer is 0 Aug 22 20:22:00 kerneldev kernel: [ 1987.470660] Called: pcm.c - snd_usb_pcm_pointer for pcm substream 0 Aug 22 20:22:00 kerneldev kernel: [ 1987.470660] Pointer is 0 Aug 22 20:22:05 kerneldev kernel: [ 1992.473567] Called: pcm.c - snd_usb_pcm_pointer for pcm substream 0 Aug 22 20:22:05 kerneldev kernel: [ 1992.473569] Pointer is 0 The above is from a non-working device. For a working device the pointer increments. Eventually this activity ends: Aug 22 20:22:05 kerneldev kernel: [ 1992.473575] Called: pcm.c - snd_usb_substream_playback_trigger for pcm substream 0 Aug 22 20:22:05 kerneldev kernel: [ 1992.473576] Called: endpoint.c - snd_usb_endpoint_stop for endpoint 142, type 0 Aug 22 20:22:05 kerneldev kernel: [ 1992.473578] Called: endpoint.c - snd_usb_endpoint_stop for endpoint 13, type 0 Aug 22 20:22:05 kerneldev kernel: [ 1992.473602] Called: pcm.c - snd_usb_hw_free for pcm substream 0 Aug 22 20:22:05 kerneldev kernel: [ 1992.503287] Called: endpoint.c - snd_usb_endpoint_deactivate for endpoint 142, type 0 Aug 22 20:22:05 kerneldev kernel: [ 1992.503288] Called: endpoint.c - snd_usb_endpoint_deactivate for endpoint 13, type 0 Aug 22 20:22:05 kerneldev kernel: [ 1992.503323] Called: pcm.c - snd_usb_playback_close for pcm substream 0 Aug 22 20:22:05 kerneldev kernel: [ 1992.503656] Called: init.c - snd_card_file_remove This seems as if it might be some sort of test of the PCMs, to make sure they are working. Certainly in non-working devices the pcm_pointer stays at zero and in working devices it increments. It also seems to try harder with some devices than others, even though the device descriptors seem very similar. Can anyone explain any of this to me, particularly why it's exhibiting this behaviour. I've uploaded one of the logs from a non-working device to Pastebin if anyone wants to look at a full example: https://pastebin.com/uBZEuQ57 Also, is this the best way to approach this, or are there other tools/techniques that can help? I've experimented with ftrace and that was useful, but didn't give the level of detail I wanted. Cheers, Keith

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[PATCH 0/2] soundwire: fix port_ready[] dynamic allocation in
by Bard Liao 21 Aug '20

21 Aug '20

The existing code allocates memory for the total number of ports. This only works if the ports are contiguous, but will break if e.g. a Devices uses port0, 1, and 14. The port_ready[] array would contain 3 elements, which would lead to an out-of-bounds access. Conversely in other cases, the wrong port index would be used leading to timeouts on prepare. This can be fixed by allocating for the worst-case of 15 ports (DP0..DP14). In addition since the number is now fixed, we can use an array instead of a dynamic allocation. Pierre-Louis Bossart (2): soundwire: add definition for maximum number of ports soundwire: fix port_ready[] dynamic allocation in mipi_disco and ASoC codecs drivers/soundwire/mipi_disco.c | 18 +----------------- drivers/soundwire/slave.c | 4 ++++ include/linux/soundwire/sdw.h | 5 +++-- sound/soc/codecs/max98373-sdw.c | 15 +-------------- sound/soc/codecs/rt1308-sdw.c | 14 +------------- sound/soc/codecs/rt5682-sdw.c | 15 +-------------- sound/soc/codecs/rt700-sdw.c | 15 +-------------- sound/soc/codecs/rt711-sdw.c | 15 +-------------- sound/soc/codecs/rt715-sdw.c | 33 +-------------------------------- 9 files changed, 14 insertions(+), 120 deletions(-) -- 2.17.1

4 9

[PATCH v2] soundwire: SDCA: add helper macro to access controls
by Bard Liao 21 Aug '20

21 Aug '20

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> The upcoming SDCA (SoundWire Device Class Audio) specification defines a hiearchical encoding to interface with Class-defined capabilities, based on which audio function, entity, control and channel being used. The specification is not yet accessible to the general public but this information is released with explicit permission from the MIPI Board to avoid delays with SDCA support on Linux platforms. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Rander Wang <rander.wang(a)linux.intel.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> --- Changelog: v2: - add SDW_SDCA_MBQ_CTL --- include/linux/soundwire/sdw_registers.h | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/include/linux/soundwire/sdw_registers.h b/include/linux/soundwire/sdw_registers.h index 5d3c271af7d1..4517c0f65b4f 100644 --- a/include/linux/soundwire/sdw_registers.h +++ b/include/linux/soundwire/sdw_registers.h @@ -305,4 +305,16 @@ #define SDW_CASC_PORT_MASK_INTSTAT3 1 #define SDW_CASC_PORT_REG_OFFSET_INTSTAT3 2 +/* v1.2 device - SDCA address mapping */ +#define SDW_SDCA_CTL(fun, ent, ctl, ch) (BIT(30) | \ + (((fun) & 0x7) << 22) | \ + (((ent) & 0x40) << 15) | \ + (((ent) & 0x3f) << 7) | \ + (((ctl) & 0x30) << 15) | \ + (((ctl) & 0x0f) << 3) | \ + (((ch) & 0x38) << 12) | \ + ((ch) & 0x07)) + +#define SDW_SDCA_MBQ_CTL(reg) ((reg) | BIT(13)) + #endif /* __SDW_REGISTERS_H */ -- 2.17.1

3 3

[RESEND] ASoC: SOF: Intel: add build support for SoundWire
by Bard Liao 21 Aug '20

21 Aug '20

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Select SoundWire capabilities on newer Intel platforms, starting with CannonLake/CoffeeLake/CometLake. As done for HDaudio, the SoundWire link is an opt-in capability. We explicitly test for ACPI to avoid warnings on unmet dependencies on the SoundWire side. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> --- sound/soc/sof/intel/Kconfig | 29 +++++++++++++++++++++++++++++ 1 file changed, 29 insertions(+) diff --git a/sound/soc/sof/intel/Kconfig b/sound/soc/sof/intel/Kconfig index 3aaf25e4f766..f644010cc568 100644 --- a/sound/soc/sof/intel/Kconfig +++ b/sound/soc/sof/intel/Kconfig @@ -166,6 +166,7 @@ config SND_SOC_SOF_CANNONLAKE_SUPPORT config SND_SOC_SOF_CANNONLAKE tristate select SND_SOC_SOF_HDA_COMMON + select SND_SOC_SOF_INTEL_SOUNDWIRE_LINK_BASELINE help This option is not user-selectable but automagically handled by 'select' statements at a higher level @@ -181,6 +182,7 @@ config SND_SOC_SOF_COFFEELAKE_SUPPORT config SND_SOC_SOF_COFFEELAKE tristate select SND_SOC_SOF_HDA_COMMON + select SND_SOC_SOF_INTEL_SOUNDWIRE_LINK_BASELINE help This option is not user-selectable but automagically handled by 'select' statements at a higher level @@ -196,6 +198,7 @@ config SND_SOC_SOF_ICELAKE_SUPPORT config SND_SOC_SOF_ICELAKE tristate select SND_SOC_SOF_HDA_COMMON + select SND_SOC_SOF_INTEL_SOUNDWIRE_LINK_BASELINE help This option is not user-selectable but automagically handled by 'select' statements at a higher level @@ -203,6 +206,7 @@ config SND_SOC_SOF_ICELAKE config SND_SOC_SOF_COMETLAKE tristate select SND_SOC_SOF_HDA_COMMON + select SND_SOC_SOF_INTEL_SOUNDWIRE_LINK_BASELINE help This option is not user-selectable but automagically handled by 'select' statements at a higher level @@ -229,6 +233,7 @@ config SND_SOC_SOF_TIGERLAKE_SUPPORT config SND_SOC_SOF_TIGERLAKE tristate select SND_SOC_SOF_HDA_COMMON + select SND_SOC_SOF_INTEL_SOUNDWIRE_LINK_BASELINE help This option is not user-selectable but automagically handled by 'select' statements at a higher level @@ -244,6 +249,7 @@ config SND_SOC_SOF_ELKHARTLAKE_SUPPORT config SND_SOC_SOF_ELKHARTLAKE tristate select SND_SOC_SOF_HDA_COMMON + select SND_SOC_SOF_INTEL_SOUNDWIRE_LINK_BASELINE help This option is not user-selectable but automagically handled by 'select' statements at a higher level @@ -329,6 +335,29 @@ config SND_SOC_SOF_HDA This option is not user-selectable but automagically handled by 'select' statements at a higher level +config SND_SOC_SOF_INTEL_SOUNDWIRE_LINK + bool "SOF support for SoundWire" + depends on SOUNDWIRE && ACPI + help + This adds support for SoundWire with Sound Open Firmware + for Intel(R) platforms. + Say Y if you want to enable SoundWire links with SOF. + If unsure select "N". + +config SND_SOC_SOF_INTEL_SOUNDWIRE_LINK_BASELINE + tristate + select SND_SOC_SOF_INTEL_SOUNDWIRE if SND_SOC_SOF_INTEL_SOUNDWIRE_LINK + help + This option is not user-selectable but automagically handled by + 'select' statements at a higher level + +config SND_SOC_SOF_INTEL_SOUNDWIRE + tristate + select SOUNDWIRE_INTEL + help + This option is not user-selectable but automagically handled by + 'select' statements at a higher level + endif ## SND_SOC_SOF_INTEL_PCI endif ## SND_SOC_SOF_INTEL_TOPLEVEL -- 2.17.1

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[RESEND] ASoC: SOF: Intel: hda: import SOUNDWIRE_INIT namespace
by Bard Liao 21 Aug '20

21 Aug '20

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Make sure the SoundWire driver can be loaded Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> --- sound/soc/sof/intel/hda.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c index 63ca920c8e6e..c0b75d682750 100644 --- a/sound/soc/sof/intel/hda.c +++ b/sound/soc/sof/intel/hda.c @@ -1207,3 +1207,4 @@ MODULE_LICENSE("Dual BSD/GPL"); MODULE_IMPORT_NS(SND_SOC_SOF_HDA_AUDIO_CODEC); MODULE_IMPORT_NS(SND_SOC_SOF_HDA_AUDIO_CODEC_I915); MODULE_IMPORT_NS(SND_SOC_SOF_XTENSA); +MODULE_IMPORT_NS(SOUNDWIRE_INTEL_INIT); -- 2.17.1

2 1

[PATCH][next] ASoC: qcom: add missing out of memory check on drvdata->clks allocation
by Colin King 21 Aug '20

21 Aug '20

From: Colin Ian King <colin.king(a)canonical.com> Currently drvdata->clks is not being checked for an allocation failure, leading to potential null pointer dereferencing. Fix this by adding a check and returning -ENOMEM if an error occurred. Addresses-Coverity: ("Dereference null return value") Fixes: 1220f6a76e77 ("ASoC: qcom: Add common array to initialize soc based core clocks") Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/qcom/lpass-apq8016.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/qcom/lpass-apq8016.c b/sound/soc/qcom/lpass-apq8016.c index dd9e3dd014f6..5c8ae225cd5d 100644 --- a/sound/soc/qcom/lpass-apq8016.c +++ b/sound/soc/qcom/lpass-apq8016.c @@ -168,6 +168,8 @@ static int apq8016_lpass_init(struct platform_device *pdev) drvdata->clks = devm_kcalloc(dev, variant->num_clks, sizeof(*drvdata->clks), GFP_KERNEL); + if (!drvdata->clks) + return -ENOMEM; drvdata->num_clks = variant->num_clks; for (i = 0; i < drvdata->num_clks; i++) -- 2.27.0

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Re: [PATCH v2 00/41] spi / fbdev / cpufreq / usb / mmc / hwmon / ARM: Prepare for multiplatform S3C
by Krzysztof Kozlowski 20 Aug '20

20 Aug '20

On Thu, Aug 06, 2020 at 08:19:32PM +0200, Krzysztof Kozlowski wrote: > Hi All, > > Shortly > ======= > This is a continuation of Arnd's work from 2019 [1]. The goal is to > cleanup, merge and finally make the Samsung S3C24xx and S3C64xx > architectures multiplatform. The multiplatform did not happen yet > here... just cleaning up and merging into one mach-s3c. > > I intend to take it through Samsung SoC tree so all Acks are welcomed. > > Changes since v1 > ================ > 1. Rebased, > 2. Addressed comments (including mine), > 3. Few new patches. > > Please see individual changelogs (per patch). > > [1] https://patchwork.kernel.org/project/linux-samsung-soc/list/?series=185855&… > Hi All, I applied second part of the set which finishes this step of S3C cleanup. Thanks to Arnd for the work! Best regards, Krzysztof

1 0

JamVOX usb soundcard works only after windows reboot
by Krabador 20 Aug '20

20 Aug '20

Hi alsa-devel team. This device from linux from 5.2 (my first plug) to now with 5.8.1, the only way to use it properly is to run linux after windows reboot. If linux is run at poweron, the device is correctly seen by the kernel but muted. Here the results of alsa-info.sh when card is working , not working, a diff, dmesg and lsusb -v https://paste.ee/p/YrJSP#pRmvQ58T7sCeulHr1ZSluw2cTQt7exsh Waiting to give more information, Best Regards.

1 1

[PATCH v6 0/2] Add mediatek codec mt6359 driver
by Jiaxin Yu 20 Aug '20

20 Aug '20

Add mediatek codec (MT6359) driver MT6359 support playback and capture feature. On downlink path, it includes three DACs for handset, headset, and lineout path. On unlink path, it includeds three ADCs for main mic, second mic, 3rd mic, and headset mic. By scenario, select *_MUX widget to create damp path. And by select mic_type_mux to choose DMIC/AMIC/.... For example, select these MUX widget to create headset path (1) DAC In Mux --> "Normal Path" (2) HP Mux --> "Audio Playback" v6 changes: 1. Remove the compatible string in mt6359codec because MFD should be registering the platform device. v5 changes: 1. Don't need to unregister the component whic is already relegated to devm. 2. patchwork link: https://patchwork.kernel.org/cover/11716387/ https://patchwork.kernel.org/patch/11717757/ https://patchwork.kernel.org/patch/11716491/ v4 changes: 1. Add a remove() function to undo regulator_enable(). 2. Removed unnecessary logs. 3. patchwork link: https://patchwork.kernel.org/cover/11715553/ https://patchwork.kernel.org/patch/11716015/ https://patchwork.kernel.org/patch/11715557/ v3 changes: 1. patchwork link: https://patchwork.kernel.org/cover/11714291/ https://patchwork.kernel.org/patch/11714295/ https://patchwork.kernel.org/patch/11714339/ v2 changes: 1. patchwork link: https://patchwork.kernel.org/cover/11706935/ https://patchwork.kernel.org/patch/11708865/ https://patchwork.kernel.org/patch/11706937/ v1 changes: 1.lkml link: https://lkml.org/lkml/2020/3/5/1257 Jiaxin Yu (2): ASoC: mediatek: mt6359: add codec driver dt-bindings: mediatek: mt6359: add codec document .../devicetree/bindings/sound/mt6359.yaml | 61 + sound/soc/codecs/Kconfig | 8 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/mt6359.c | 2753 +++++++++++++++++ sound/soc/codecs/mt6359.h | 2640 ++++++++++++++++ 5 files changed, 5464 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/mt6359.yaml create mode 100644 sound/soc/codecs/mt6359.c create mode 100644 sound/soc/codecs/mt6359.h -- 2.18.0

2 3

Request to pick up couple NVIDIA Tegra ASoC patches into 5.7 kernel
by Dmitry Osipenko 20 Aug '20

20 Aug '20

Hello, stable-kernel maintainers! Could you please cherry-pick these commits into the v5.7.x kernel? commit 0de6db30ef79b391cedd749801a49c485d2daf4b Author: Sowjanya Komatineni <skomatineni(a)nvidia.com> Date: Mon Jan 13 23:24:17 2020 -0800 ASoC: tegra: Use device managed resource APIs to get the clock commit 1e4e0bf136aa4b4aa59c1e6af19844bd6d807794 Author: Sowjanya Komatineni <skomatineni(a)nvidia.com> Date: Mon Jan 13 23:24:23 2020 -0800 ASoC: tegra: Add audio mclk parent configuration commit ff5d18cb04f4ecccbcf05b7f83ab6df2a0d95c16 Author: Sowjanya Komatineni <skomatineni(a)nvidia.com> Date: Mon Jan 13 23:24:24 2020 -0800 ASoC: tegra: Enable audio mclk during tegra_asoc_utils_init() It will fix a huge warnings splat during of kernel boot on NVIDIA Tegra SoCs. For some reason these patches haven't made into 5.7 when it was released and several people complained about the warnings. Thanks in advance!

3 3

[PATCH] ALSA: usb-audio: Add prevent wakeup from s3 state trig by Lenovo ThinkCentre TI024Gen3 USB-audio
by penghao 20 Aug '20

20 Aug '20

TI024Gen3 USB-audio is controlled by TI024Gen3,when TI024Gens enter sleep mode, USB-audio will disconnect from USB bus port, wakup form s3 state Signed-off-by: penghao <penghao(a)uniontech.com> --- sound/usb/card.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/usb/card.c b/sound/usb/card.c index 696e788c5d31..6bdbb34009b3 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -658,6 +658,12 @@ static int usb_audio_probe(struct usb_interface *intf, } dev_set_drvdata(&dev->dev, chip); + /* + *ALSA: usb-audio: Add prevent wakeup from s3 state trig by Lenovo + *ThinkCentre TI024Gen3 usb-audio + */ + if ((usb_id->idVendor == 0x17ef) && (usb_id->idProduct == 0xa012)) + device_set_wakeup_enable(＆dev->dev, 0); /* * For devices with more than one control interface, we assume the -- 2.11.0

6 7

[PATCH v2] ALSA: hda: avoid reset of sdo_limit
by Sameer Pujar 19 Aug '20

19 Aug '20

By default 'sdo_limit' is initialized with a default value of '8' as per spec. This is overridden in cases where a different value is required. However this is getting reset when snd_hdac_bus_init_chip() is called again, which happens during runtime PM cycle. Avoid this reset by moving 'sdo_limit' setup to 'snd_hdac_bus_init()' function which would be called only once. Fixes: 67ae482a59e9 ("ALSA: hda: add member to store ratio for stripe control") Cc: <stable(a)vger.kernel.org> Signed-off-by: Sameer Pujar <spujar(a)nvidia.com> --- sound/hda/hdac_bus.c | 12 ++++++++++++ sound/hda/hdac_controller.c | 11 ----------- 2 files changed, 12 insertions(+), 11 deletions(-) diff --git a/sound/hda/hdac_bus.c b/sound/hda/hdac_bus.c index 09ddab5..9766f6a 100644 --- a/sound/hda/hdac_bus.c +++ b/sound/hda/hdac_bus.c @@ -46,6 +46,18 @@ int snd_hdac_bus_init(struct hdac_bus *bus, struct device *dev, INIT_LIST_HEAD(&bus->hlink_list); init_waitqueue_head(&bus->rirb_wq); bus->irq = -1; + + /* + * Default value of '8' is as per the HD audio specification (Rev 1.0a). + * Following relation is used to derive STRIPE control value. + * For sample rate <= 48K: + * { ((num_channels * bits_per_sample) / number of SDOs) >= 8 } + * For sample rate > 48K: + * { ((num_channels * bits_per_sample * rate/48000) / + * number of SDOs) >= 8 } + */ + bus->sdo_limit = 8; + return 0; } EXPORT_SYMBOL_GPL(snd_hdac_bus_init); diff --git a/sound/hda/hdac_controller.c b/sound/hda/hdac_controller.c index 011b17c..b98449f 100644 --- a/sound/hda/hdac_controller.c +++ b/sound/hda/hdac_controller.c @@ -529,17 +529,6 @@ bool snd_hdac_bus_init_chip(struct hdac_bus *bus, bool full_reset) bus->chip_init = true; - /* - * Default value of '8' is as per the HD audio specification (Rev 1.0a). - * Following relation is used to derive STRIPE control value. - * For sample rate <= 48K: - * { ((num_channels * bits_per_sample) / number of SDOs) >= 8 } - * For sample rate > 48K: - * { ((num_channels * bits_per_sample * rate/48000) / - * number of SDOs) >= 8 } - */ - bus->sdo_limit = 8; - return true; } EXPORT_SYMBOL_GPL(snd_hdac_bus_init_chip); -- 2.7.4

2 1

[PATCH] ALSA: hda: avoid reset of sdo_limit
by Sameer Pujar 19 Aug '20

19 Aug '20

By default 'sdo_limit' is initialized with a default value of '8' as per spec. This is overridden in cases where a different value is required. However this is getting reset when snd_hdac_bus_init_chip() is called again, which happens during runtime PM cycle. Avoid reset by not initializing to default value everytime. Fixes: 67ae482a59e9 ("ALSA: hda: add member to store ratio for stripe control") Cc: <stable(a)vger.kernel.org> Signed-off-by: Sameer Pujar <spujar(a)nvidia.com> --- sound/hda/hdac_controller.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/hda/hdac_controller.c b/sound/hda/hdac_controller.c index 011b17c..0e26e96 100644 --- a/sound/hda/hdac_controller.c +++ b/sound/hda/hdac_controller.c @@ -538,7 +538,8 @@ bool snd_hdac_bus_init_chip(struct hdac_bus *bus, bool full_reset) * { ((num_channels * bits_per_sample * rate/48000) / * number of SDOs) >= 8 } */ - bus->sdo_limit = 8; + if (!bus->sdo_limit) + bus->sdo_limit = 8; return true; } -- 2.7.4

2 2

[PATCH] ASoC: codecs: rt*-sdw: use SDW_SLAVE_ENTRY_EXT
by Bard Liao 19 Aug '20

19 Aug '20

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Add version and class information explicitly to prepare for support for new devices. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Rander Wang <rander.wang(a)linux.intel.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Reviewed-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> --- sound/soc/codecs/rt1308-sdw.c | 2 +- sound/soc/codecs/rt5682-sdw.c | 2 +- sound/soc/codecs/rt700-sdw.c | 2 +- sound/soc/codecs/rt711-sdw.c | 2 +- sound/soc/codecs/rt715-sdw.c | 2 +- 5 files changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/rt1308-sdw.c b/sound/soc/codecs/rt1308-sdw.c index 81e3e2caeb2f..ecb03ad8102e 100644 --- a/sound/soc/codecs/rt1308-sdw.c +++ b/sound/soc/codecs/rt1308-sdw.c @@ -696,7 +696,7 @@ static int rt1308_sdw_probe(struct sdw_slave *slave, } static const struct sdw_device_id rt1308_id[] = { - SDW_SLAVE_ENTRY(0x025d, 0x1308, 0), + SDW_SLAVE_ENTRY_EXT(0x025d, 0x1308, 0x2, 0, 0), {}, }; MODULE_DEVICE_TABLE(sdw, rt1308_id); diff --git a/sound/soc/codecs/rt5682-sdw.c b/sound/soc/codecs/rt5682-sdw.c index 148d4cff267b..4d5bc8ac38df 100644 --- a/sound/soc/codecs/rt5682-sdw.c +++ b/sound/soc/codecs/rt5682-sdw.c @@ -722,7 +722,7 @@ static int rt5682_sdw_remove(struct sdw_slave *slave) } static const struct sdw_device_id rt5682_id[] = { - SDW_SLAVE_ENTRY(0x025d, 0x5682, 0), + SDW_SLAVE_ENTRY_EXT(0x025d, 0x5682, 0x2, 0, 0), {}, }; MODULE_DEVICE_TABLE(sdw, rt5682_id); diff --git a/sound/soc/codecs/rt700-sdw.c b/sound/soc/codecs/rt700-sdw.c index 2d475405b20d..b08ccc3da1dd 100644 --- a/sound/soc/codecs/rt700-sdw.c +++ b/sound/soc/codecs/rt700-sdw.c @@ -483,7 +483,7 @@ static int rt700_sdw_remove(struct sdw_slave *slave) } static const struct sdw_device_id rt700_id[] = { - SDW_SLAVE_ENTRY(0x025d, 0x700, 0), + SDW_SLAVE_ENTRY_EXT(0x025d, 0x700, 0x1, 0, 0), {}, }; MODULE_DEVICE_TABLE(sdw, rt700_id); diff --git a/sound/soc/codecs/rt711-sdw.c b/sound/soc/codecs/rt711-sdw.c index 7a1ae7442e75..c72cd4fb64fd 100644 --- a/sound/soc/codecs/rt711-sdw.c +++ b/sound/soc/codecs/rt711-sdw.c @@ -484,7 +484,7 @@ static int rt711_sdw_remove(struct sdw_slave *slave) } static const struct sdw_device_id rt711_id[] = { - SDW_SLAVE_ENTRY(0x025d, 0x711, 0), + SDW_SLAVE_ENTRY_EXT(0x025d, 0x711, 0x2, 0, 0), {}, }; MODULE_DEVICE_TABLE(sdw, rt711_id); diff --git a/sound/soc/codecs/rt715-sdw.c b/sound/soc/codecs/rt715-sdw.c index 761d4663e813..c87a3fc0678b 100644 --- a/sound/soc/codecs/rt715-sdw.c +++ b/sound/soc/codecs/rt715-sdw.c @@ -546,7 +546,7 @@ static int rt715_sdw_probe(struct sdw_slave *slave, } static const struct sdw_device_id rt715_id[] = { - SDW_SLAVE_ENTRY(0x025d, 0x715, 0), + SDW_SLAVE_ENTRY_EXT(0x025d, 0x715, 0x2, 0, 0), {}, }; MODULE_DEVICE_TABLE(sdw, rt715_id); -- 2.17.1

2 1

[PATCH] ASoC: Intel: skl_hda_dsp_generic: Fix NULLptr dereference in autosuspend delay
by Mateusz Gorski 19 Aug '20

19 Aug '20

Different modules for HDMI codec are used depending on the "hda_codec_use_common_hdmi" option being enabled or not. Driver private context for both of them is different. This leads to null-pointer dereference error when driver tries to set autosuspend delay for HDMI codec while the option is off (hdac_hdmi module is used for HDMI). Change the string in conditional statement to "ehdaudio0D0" to ensure that only the HDAudio codec is handled by this function. Fixes: 5bf73b1b1dec ("ASoC: intel/skl/hda - fix oops on systems without i915 audio codec") Signed-off-by: Mateusz Gorski <mateusz.gorski(a)linux.intel.com> --- sound/soc/intel/boards/skl_hda_dsp_generic.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/boards/skl_hda_dsp_generic.c b/sound/soc/intel/boards/skl_hda_dsp_generic.c index ca4900036ead..bc50eda297ab 100644 --- a/sound/soc/intel/boards/skl_hda_dsp_generic.c +++ b/sound/soc/intel/boards/skl_hda_dsp_generic.c @@ -181,7 +181,7 @@ static void skl_set_hda_codec_autosuspend_delay(struct snd_soc_card *card) struct snd_soc_dai *dai; for_each_card_rtds(card, rtd) { - if (!strstr(rtd->dai_link->codecs->name, "ehdaudio")) + if (!strstr(rtd->dai_link->codecs->name, "ehdaudio0D0")) continue; dai = asoc_rtd_to_codec(rtd, 0); hda_pvt = snd_soc_component_get_drvdata(dai->component); -- 2.17.1

6 7

[PATCH V2] dt-bindings: sound: Convert NXP spdif to json-schema
by Anson Huang 19 Aug '20

19 Aug '20

Convert the NXP SPDIF binding to DT schema format using json-schema. Signed-off-by: Anson Huang <Anson.Huang(a)nxp.com> Reviewed-by: Rob Herring <robh(a)kernel.org> --- changes since V1: - add "additionalProperties: false". --- .../devicetree/bindings/sound/fsl,spdif.txt | 68 ------------- .../devicetree/bindings/sound/fsl,spdif.yaml | 110 +++++++++++++++++++++ 2 files changed, 110 insertions(+), 68 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/fsl,spdif.txt create mode 100644 Documentation/devicetree/bindings/sound/fsl,spdif.yaml diff --git a/Documentation/devicetree/bindings/sound/fsl,spdif.txt b/Documentation/devicetree/bindings/sound/fsl,spdif.txt deleted file mode 100644 index e1365b0..0000000 --- a/Documentation/devicetree/bindings/sound/fsl,spdif.txt +++ /dev/null @@ -1,68 +0,0 @@ -Freescale Sony/Philips Digital Interface Format (S/PDIF) Controller - -The Freescale S/PDIF audio block is a stereo transceiver that allows the -processor to receive and transmit digital audio via an coaxial cable or -a fibre cable. - -Required properties: - - - compatible : Compatible list, should contain one of the following - compatibles: - "fsl,imx35-spdif", - "fsl,vf610-spdif", - "fsl,imx6sx-spdif", - - - reg : Offset and length of the register set for the device. - - - interrupts : Contains the spdif interrupt. - - - dmas : Generic dma devicetree binding as described in - Documentation/devicetree/bindings/dma/dma.txt. - - - dma-names : Two dmas have to be defined, "tx" and "rx". - - - clocks : Contains an entry for each entry in clock-names. - - - clock-names : Includes the following entries: - "core" The core clock of spdif controller. - "rxtx<0-7>" Clock source list for tx and rx clock. - This clock list should be identical to the source - list connecting to the spdif clock mux in "SPDIF - Transceiver Clock Diagram" of SoC reference manual. - It can also be referred to TxClk_Source bit of - register SPDIF_STC. - "spba" The spba clock is required when SPDIF is placed as a - bus slave of the Shared Peripheral Bus and when two - or more bus masters (CPU, DMA or DSP) try to access - it. This property is optional depending on the SoC - design. - -Optional properties: - - - big-endian : If this property is absent, the native endian mode - will be in use as default, or the big endian mode - will be in use for all the device registers. - -Example: - -spdif: spdif@2004000 { - compatible = "fsl,imx35-spdif"; - reg = <0x02004000 0x4000>; - interrupts = <0 52 0x04>; - dmas = <&sdma 14 18 0>, - <&sdma 15 18 0>; - dma-names = "rx", "tx"; - - clocks = <&clks 197>, <&clks 3>, - <&clks 197>, <&clks 107>, - <&clks 0>, <&clks 118>, - <&clks 62>, <&clks 139>, - <&clks 0>; - clock-names = "core", "rxtx0", - "rxtx1", "rxtx2", - "rxtx3", "rxtx4", - "rxtx5", "rxtx6", - "rxtx7"; - - big-endian; -}; diff --git a/Documentation/devicetree/bindings/sound/fsl,spdif.yaml b/Documentation/devicetree/bindings/sound/fsl,spdif.yaml new file mode 100644 index 0000000..2ac671f --- /dev/null +++ b/Documentation/devicetree/bindings/sound/fsl,spdif.yaml @@ -0,0 +1,110 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/fsl,spdif.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Freescale Sony/Philips Digital Interface Format (S/PDIF) Controller + +maintainers: + - Shengjiu Wang <shengjiu.wang(a)nxp.com> + +description: | + The Freescale S/PDIF audio block is a stereo transceiver that allows the + processor to receive and transmit digital audio via an coaxial cable or + a fibre cable. + +properties: + compatible: + enum: + - fsl,imx35-spdif + - fsl,vf610-spdif + - fsl,imx6sx-spdif + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + dmas: + items: + - description: DMA controller phandle and request line for RX + - description: DMA controller phandle and request line for TX + + dma-names: + items: + - const: rx + - const: tx + + clocks: + items: + - description: The core clock of spdif controller. + - description: Clock for tx0 and rx0. + - description: Clock for tx1 and rx1. + - description: Clock for tx2 and rx2. + - description: Clock for tx3 and rx3. + - description: Clock for tx4 and rx4. + - description: Clock for tx5 and rx5. + - description: Clock for tx6 and rx6. + - description: Clock for tx7 and rx7. + - description: The spba clock is required when SPDIF is placed as a bus + slave of the Shared Peripheral Bus and when two or more bus masters + (CPU, DMA or DSP) try to access it. This property is optional depending + on the SoC design. + minItems: 9 + + clock-names: + items: + - const: core + - const: rxtx0 + - const: rxtx1 + - const: rxtx2 + - const: rxtx3 + - const: rxtx4 + - const: rxtx5 + - const: rxtx6 + - const: rxtx7 + - const: spba + minItems: 9 + + big-endian: + $ref: /schemas/types.yaml#/definitions/flag + description: | + If this property is absent, the native endian mode will be in use + as default, or the big endian mode will be in use for all the device + registers. Set this flag for HCDs with big endian descriptors and big + endian registers. + +required: + - compatible + - reg + - interrupts + - dmas + - dma-names + - clocks + - clock-names + +additionalProperties: false + +examples: + - | + spdif@2004000 { + compatible = "fsl,imx35-spdif"; + reg = <0x02004000 0x4000>; + interrupts = <0 52 0x04>; + dmas = <&sdma 14 18 0>, + <&sdma 15 18 0>; + dma-names = "rx", "tx"; + clocks = <&clks 197>, <&clks 3>, + <&clks 197>, <&clks 107>, + <&clks 0>, <&clks 118>, + <&clks 62>, <&clks 139>, + <&clks 0>; + clock-names = "core", "rxtx0", + "rxtx1", "rxtx2", + "rxtx3", "rxtx4", + "rxtx5", "rxtx6", + "rxtx7"; + big-endian; + }; -- 2.7.4

2 1

[PATCH] soundwire: bus: fix typo in comment on INTSTAT registers
by Bard Liao 19 Aug '20

19 Aug '20

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> s/Instat/Intstat/ Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> --- drivers/soundwire/bus.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/drivers/soundwire/bus.c b/drivers/soundwire/bus.c index e6e0fb9a81b4..da0201693c24 100644 --- a/drivers/soundwire/bus.c +++ b/drivers/soundwire/bus.c @@ -1372,7 +1372,7 @@ static int sdw_handle_slave_alerts(struct sdw_slave *slave) return ret; } - /* Read Instat 1, Instat 2 and Instat 3 registers */ + /* Read Intstat 1, Intstat 2 and Intstat 3 registers */ ret = sdw_read(slave, SDW_SCP_INT1); if (ret < 0) { dev_err(slave->bus->dev, -- 2.17.1

2 1

[PATCH] ALSA: hda/realtek: Add quirk for Samsung Galaxy Book Ion
by Mike Pozulp 19 Aug '20

19 Aug '20

The Galaxy Book Ion uses the same ALC298 codec as other Samsung laptops which have the no headphone sound bug, like my Samsung Notebook. The Galaxy Book owner confirmed that this patch fixes the bug. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207423 Signed-off-by: Mike Pozulp <pozulp.kernel(a)gmail.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 29f5878f0c50..72aa3da479c9 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7612,6 +7612,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x144d, 0xc109, "Samsung Ativ book 9 (NP900X3G)", ALC269_FIXUP_INV_DMIC), SND_PCI_QUIRK(0x144d, 0xc169, "Samsung Notebook 9 Pen (NP930SBE-K01US)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET), SND_PCI_QUIRK(0x144d, 0xc176, "Samsung Notebook 9 Pro (NP930MBE-K04US)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET), + SND_PCI_QUIRK(0x144d, 0xc18a, "Samsung Galaxy Book Ion (NT950XCJ-X716A)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET), SND_PCI_QUIRK(0x144d, 0xc740, "Samsung Ativ book 8 (NP870Z5G)", ALC269_FIXUP_ATIV_BOOK_8), SND_PCI_QUIRK(0x144d, 0xc812, "Samsung Notebook Pen S (NT950SBE-X58)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET), SND_PCI_QUIRK(0x1458, 0xfa53, "Gigabyte BXBT-2807", ALC283_FIXUP_HEADSET_MIC), -- 2.28.0

2 1

[GIT PULL] ASoC fixes for v5.9-rc1
by Mark Brown 19 Aug '20

19 Aug '20

The following changes since commit 8e34f1e867b572f1e20b5250c2897fe5f041c99f: Merge series "ASoC: core: Two step component registration" from Cezary Rojewski <cezary.rojewski(a)intel.com>: (2020-07-31 19:36:00 +0100) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v5.9-rc1 for you to fetch changes up to 062fa09f44f4fb3776a23184d5d296b0c8872eb9: ASoC: intel: Fix memleak in sst_media_open (2020-08-17 12:28:57 +0100) ---------------------------------------------------------------- ASoC: Fixes for v5.9 A bunch of fixes that came in during the merge window, mostly for issues that were uncovered by the changes to report errors on invalid register access plus one important fix in that code itself. ---------------------------------------------------------------- Dinghao Liu (1): ASoC: intel: Fix memleak in sst_media_open Fabio Estevam (1): ASoC: wm8962: Do not remove ADDITIONAL_CONTROL_4 from readable register list Hui Wang (1): ASoC: amd: renoir: restore two more registers during resume Mark Brown (1): Merge series "ASoC: tegra: Fix compile warning with CONFIG_PM=n" from Takashi Iwai <tiwai(a)suse.de>: Ravulapati Vishnu vardhan rao (1): ASoC: amd: Replacing component->name with codec_dai->name. Shengjiu Wang (1): ASoC: fsl-asoc-card: Remove fsl_asoc_card_set_bias_level function Srinivas Kandagatla (3): ASoC: q6afe-dai: mark all widgets registers as SND_SOC_NOPM ASoC: q6routing: add dummy register read/write function ASoC: msm8916-wcd-analog: fix register Interrupt offset Sylwester Nawrocki (2): ASoC: wm8994: Prevent access to invalid VU register bits on WM1811 ASoC: wm8994: Avoid attempts to read unreadable registers Takashi Iwai (7): ASoC: tegra: tegra186_dspk: Fix compile warning with CONFIG_PM=n ASoC: tegra: tegra210_admaif: Fix compile warning with CONFIG_PM=n ASoC: tegra: tegra210_ahub: Fix compile warning with CONFIG_PM=n ASoC: tegra: tegra210_dmic: Fix compile warning with CONFIG_PM=n ASoC: tegra: tegra210_i2s: Fix compile warning with CONFIG_PM=n ASoC: fsl: Fix unused variable warning ASoC: Make soc_component_read() returning an error code again sound/soc/amd/acp3x-rt5682-max9836.c | 2 +- sound/soc/amd/renoir/acp3x-pdm-dma.c | 29 ++-- sound/soc/codecs/msm8916-wcd-analog.c | 4 +- sound/soc/codecs/wm8958-dsp2.c | 4 + sound/soc/codecs/wm8962.c | 2 +- sound/soc/codecs/wm8994.c | 60 ++++++-- sound/soc/fsl/fsl-asoc-card.c | 154 +++++++++----------- sound/soc/fsl/mpc5200_dma.c | 1 - sound/soc/intel/atom/sst-mfld-platform-pcm.c | 5 +- sound/soc/qcom/qdsp6/q6afe-dai.c | 210 +++++++++++++-------------- sound/soc/qcom/qdsp6/q6routing.c | 16 ++ sound/soc/soc-component.c | 2 +- sound/soc/tegra/tegra186_dspk.c | 4 +- sound/soc/tegra/tegra210_admaif.c | 4 +- sound/soc/tegra/tegra210_ahub.c | 4 +- sound/soc/tegra/tegra210_dmic.c | 4 +- sound/soc/tegra/tegra210_i2s.c | 4 +- 17 files changed, 266 insertions(+), 243 deletions(-)

2 1

ucm: let rval_sysfs check the device/driver/module first
by GitHub pull_request - opened 19 Aug '20

19 Aug '20

alsa-project/alsa-lib pull request #79 was opened from jason77-wang: We plan to use the sound driver module's name for the top ucm's name, I checked it on 3 different ASoC machines, the device/driver links to machine driver while deice/driver/module links to the sound driver module, so change the code to readlink the device/driver/module first, if it fails then readlink the device/driver. This is the output from those 3 machines: $readlink /sys/class/sound/card0/device/driver ../../../../bus/platform/drivers/skl_hda_dsp_generic $readlink /sys/class/sound/card0/device/driver/module ../../../../module/snd_soc_skl_hda_dsp $readlink /sys/class/sound/card2/device/driver ../../../../../bus/platform/drivers/acp_pdm_mach $readlink /sys/class/sound/card2/device/driver/module ../../../../module/snd_acp3x_rn $readlink /sys/class/sound/card0/device/driver ../../../../bus/platform/drivers/sof_sdw $readlink /sys/class/sound/card0/device/driver/module ../../../../module/snd_soc_sof_sdw Signed-off-by: Hui Wang <hui.wang(a)canonical.com> Request URL : https://github.com/alsa-project/alsa-lib/pull/79 Patch URL : https://github.com/alsa-project/alsa-lib/pull/79.patch Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH] ALSA: usb-audio: Add prevent wakeup from s3 state trig by Lenovo ThinkCentre TI024Gen3 USB-audio
by penghao 19 Aug '20

19 Aug '20

TI024Gen3 USB-audio is controlled by TI024Gen3,when TI024Gens enter sleep mode, USB-audio will disconnect from USB bus port, wakup form s3 state Signed-off-by: penghao <penghao(a)uniontech.com> --- sound/usb/card.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/usb/card.c b/sound/usb/card.c index 696e788c5d31..6bdbb34009b3 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -658,6 +658,12 @@ static int usb_audio_probe(struct usb_interface *intf, } dev_set_drvdata(&dev->dev, chip); + /* + *ALSA: usb-audio: Add prevent wakeup from s3 state trig by Lenovo + *ThinkCentre TI024Gen3 usb-audio + */ + if ((usb_id->idVendor == 0x17ef) && (usb_id->idProduct == 0xa012)) + device_set_wakeup_enable(＆dev->dev, 0); /* * For devices with more than one control interface, we assume the -- 2.11.0

1 0

[PATCH v2 0/3] Fix Kconfig dependency issue with DMAENGINES selection
by Laurent Pinchart 19 Aug '20

19 Aug '20

Hello, This small series fixes a Kconfig dependency issue with the recently merged Xilixn DPSUB DRM/KMS driver. The fix is in patch 3/3, but requires a separate fixes in patches 1/3 and 2/3 to avoid circular dependencies: drivers/i2c/Kconfig:8:error: recursive dependency detected! drivers/i2c/Kconfig:8: symbol I2C is selected by FB_DDC drivers/video/fbdev/Kconfig:63: symbol FB_DDC depends on FB drivers/video/fbdev/Kconfig:12: symbol FB is selected by DRM_KMS_FB_HELPER drivers/gpu/drm/Kconfig:80: symbol DRM_KMS_FB_HELPER depends on DRM_KMS_HELPER drivers/gpu/drm/Kconfig:74: symbol DRM_KMS_HELPER is selected by DRM_ZYNQMP_DPSUB drivers/gpu/drm/xlnx/Kconfig:1: symbol DRM_ZYNQMP_DPSUB depends on DMA_ENGINE drivers/dma/Kconfig:44: symbol DMA_ENGINE depends on DMADEVICES drivers/dma/Kconfig:6: symbol DMADEVICES is selected by SND_SOC_SH4_SIU sound/soc/sh/Kconfig:30: symbol SND_SOC_SH4_SIU is selected by SND_SIU_MIGOR sound/soc/sh/Kconfig:60: symbol SND_SIU_MIGOR depends on I2C For a resolution refer to Documentation/kbuild/kconfig-language.rst subsection "Kconfig recursive dependency limitations" Due to the DPSUB driver being merged in v5.9, this is a candidate fix for v5.9 as well. 1/3 and 2/3 can be merged independently, 3/3 depends on the first two. What's the best course of action, can I merge this all in a single tree, or should the rapidio and ASoC patches be merged independently early in the -rc cycle, and the DRM patch later on top ? I don't expect conflicts (especially in 2/3 and 3/3), so merging the whole series in one go would be simpler in my opinion. Compared to v1, commit messages have been fixed to mention DMADEVICES instead of DMAENGINES. Laurent Pinchart (3): rapidio: Replace 'select' DMADEVICES 'with depends on' ASoC: sh: Replace 'select' DMADEVICES 'with depends on' drm: xlnx: dpsub: Fix DMADEVICES Kconfig dependency drivers/gpu/drm/xlnx/Kconfig | 1 + drivers/rapidio/Kconfig | 2 +- sound/soc/sh/Kconfig | 2 +- 3 files changed, 3 insertions(+), 2 deletions(-) -- Regards, Laurent Pinchart

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[RFC PATCH v1 1/2] ASoC: rockchip-spdif: add description for rk3308
by Johan Jonker 18 Aug '20

18 Aug '20

A test with the command below shows that the compatible string "rockchip,rk3308-spdif", "rockchip,rk3328-spdif" is already in use, but is not added to a document. The current fallback string "rockchip,rk3328-spdif" points to a data set enum RK_SPDIF_RK3366 in rockchip_spdif.c that is not used both in the mainline as in the manufacturer kernel. (Of the enum only RK_SPDIF_RK3288 is used.) So if the properties don't change we might as well use the first SoC in line as fallback string and add the description for rk3308 as: "rockchip,rk3308-spdif", "rockchip,rk3066-spdif" make ARCH=arm64 dtbs_check DT_SCHEMA_FILES=Documentation/devicetree/bindings/sound/rockchip-spdif.yaml Signed-off-by: Johan Jonker <jbx6244(a)gmail.com> --- enum rk_spdif_type { RK_SPDIF_RK3066, RK_SPDIF_RK3188, RK_SPDIF_RK3288, RK_SPDIF_RK3366, }; static const struct of_device_id rk_spdif_match[] = { { .compatible = "rockchip,rk3066-spdif", .data = (void *)RK_SPDIF_RK3066 }, { .compatible = "rockchip,rk3188-spdif", .data = (void *)RK_SPDIF_RK3188 }, { .compatible = "rockchip,rk3228-spdif", .data = (void *)RK_SPDIF_RK3366 }, { .compatible = "rockchip,rk3288-spdif", .data = (void *)RK_SPDIF_RK3288 }, { .compatible = "rockchip,rk3328-spdif", .data = (void *)RK_SPDIF_RK3366 }, { .compatible = "rockchip,rk3366-spdif", .data = (void *)RK_SPDIF_RK3366 }, { .compatible = "rockchip,rk3368-spdif", .data = (void *)RK_SPDIF_RK3366 }, { .compatible = "rockchip,rk3399-spdif", .data = (void *)RK_SPDIF_RK3366 }, {}, }; --- Documentation/devicetree/bindings/sound/rockchip-spdif.yaml | 1 + 1 file changed, 1 insertion(+) diff --git a/Documentation/devicetree/bindings/sound/rockchip-spdif.yaml b/Documentation/devicetree/bindings/sound/rockchip-spdif.yaml index c46715265..26e986097 100644 --- a/Documentation/devicetree/bindings/sound/rockchip-spdif.yaml +++ b/Documentation/devicetree/bindings/sound/rockchip-spdif.yaml @@ -27,6 +27,7 @@ properties: - enum: - rockchip,rk3188-spdif - rockchip,rk3288-spdif + - rockchip,rk3308-spdif - const: rockchip,rk3066-spdif reg: -- 2.11.0

2 3

[PATCH] ASoC: hdac_hdmi: support 'ELD' mixer
by Brent Lu 18 Aug '20

18 Aug '20

Add an binary mixer 'ELD' to each HDMI PCM device so user space could read the ELD data of external HDMI display. Signed-off-by: Brent Lu <brent.lu(a)intel.com> --- sound/soc/codecs/hdac_hdmi.c | 139 +++++++++++++++++++++++++++++++++++ 1 file changed, 139 insertions(+) diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c index f26b77faed59..83656bbf22a7 100644 --- a/sound/soc/codecs/hdac_hdmi.c +++ b/sound/soc/codecs/hdac_hdmi.c @@ -9,6 +9,7 @@ * * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ */ + #include <linux/init.h> #include <linux/delay.h> #include <linux/module.h> @@ -107,6 +108,7 @@ struct hdac_hdmi_pcm { unsigned char chmap[8]; /* ALSA API channel-map */ struct mutex lock; int jack_event; + struct snd_kcontrol *eld_ctl; }; struct hdac_hdmi_dai_port_map { @@ -1248,6 +1250,7 @@ static void hdac_hdmi_present_sense(struct hdac_hdmi_pin *pin, struct hdac_hdmi_pcm *pcm; int size = 0; int port_id = -1; + bool eld_valid, eld_changed; if (!hdmi) return; @@ -1273,6 +1276,8 @@ static void hdac_hdmi_present_sense(struct hdac_hdmi_pin *pin, size = -EINVAL; } + eld_valid = port->eld.eld_valid; + if (size > 0) { port->eld.eld_valid = true; port->eld.eld_size = size; @@ -1281,6 +1286,8 @@ static void hdac_hdmi_present_sense(struct hdac_hdmi_pin *pin, port->eld.eld_size = 0; } + eld_changed = (eld_valid != port->eld.eld_valid); + pcm = hdac_hdmi_get_pcm(hdev, port); if (!port->eld.monitor_present || !port->eld.eld_valid) { @@ -1313,6 +1320,12 @@ static void hdac_hdmi_present_sense(struct hdac_hdmi_pin *pin, } mutex_unlock(&hdmi->pin_mutex); + + if (eld_changed && pcm) + snd_ctl_notify(hdmi->card, + SNDRV_CTL_EVENT_MASK_VALUE | + SNDRV_CTL_EVENT_MASK_INFO, + &pcm->eld_ctl->id); } static int hdac_hdmi_add_ports(struct hdac_device *hdev, @@ -1411,6 +1424,122 @@ static void hdac_hdmi_skl_enable_dp12(struct hdac_device *hdev) } +static int hdac_hdmi_eld_ctl_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct hdac_hdmi_priv *hdmi = snd_soc_component_get_drvdata(component); + struct hdac_hdmi_pcm *pcm; + struct hdac_hdmi_port *port; + struct hdac_hdmi_eld *eld; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_BYTES; + uinfo->count = 0; + + pcm = get_hdmi_pcm_from_id(hdmi, kcontrol->id.device); + if (!pcm) { + dev_dbg(component->dev, "%s: no pcm, device %d\n", __func__, + kcontrol->id.device); + return 0; + } + + if (list_empty(&pcm->port_list)) { + dev_dbg(component->dev, "%s: empty port list, device %d\n", + __func__, kcontrol->id.device); + return 0; + } + + mutex_lock(&hdmi->pin_mutex); + + list_for_each_entry(port, &pcm->port_list, head) { + eld = &port->eld; + + if (eld->eld_valid) { + uinfo->count = eld->eld_size; + break; + } + } + + mutex_unlock(&hdmi->pin_mutex); + + return 0; +} + +static int hdac_hdmi_eld_ctl_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct hdac_hdmi_priv *hdmi = snd_soc_component_get_drvdata(component); + struct hdac_hdmi_pcm *pcm; + struct hdac_hdmi_port *port; + struct hdac_hdmi_eld *eld; + + memset(ucontrol->value.bytes.data, 0, ARRAY_SIZE(ucontrol->value.bytes.data)); + + pcm = get_hdmi_pcm_from_id(hdmi, kcontrol->id.device); + if (!pcm) { + dev_dbg(component->dev, "%s: no pcm, device %d\n", __func__, + kcontrol->id.device); + return 0; + } + + if (list_empty(&pcm->port_list)) { + dev_dbg(component->dev, "%s: empty port list, device %d\n", + __func__, kcontrol->id.device); + return 0; + } + + mutex_lock(&hdmi->pin_mutex); + + list_for_each_entry(port, &pcm->port_list, head) { + eld = &port->eld; + + if (!eld->eld_valid) + continue; + + if (eld->eld_size > ARRAY_SIZE(ucontrol->value.bytes.data) || + eld->eld_size > ELD_MAX_SIZE) { + mutex_unlock(&hdmi->pin_mutex); + + dev_err(component->dev, "%s: buffer too small, device %d eld_size %d\n", + __func__, kcontrol->id.device, eld->eld_size); + snd_BUG(); + return -EINVAL; + } + + memcpy(ucontrol->value.bytes.data, eld->eld_buffer, + eld->eld_size); + break; + } + + mutex_unlock(&hdmi->pin_mutex); + + return 0; +} + +static int hdac_hdmi_create_eld_ctl(struct snd_soc_component *component, struct hdac_hdmi_pcm *pcm) +{ + struct snd_kcontrol *kctl; + struct snd_kcontrol_new hdmi_eld_ctl = { + .access = SNDRV_CTL_ELEM_ACCESS_READ | + SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .name = "ELD", + .info = hdac_hdmi_eld_ctl_info, + .get = hdac_hdmi_eld_ctl_get, + .device = pcm->pcm_id, + }; + + /* add ELD ctl with the device number corresponding to the PCM stream */ + kctl = snd_ctl_new1(&hdmi_eld_ctl, component); + if (!kctl) + return -ENOMEM; + + pcm->eld_ctl = kctl; + + return snd_ctl_add(component->card->snd_card, kctl); +} + static const struct snd_soc_dai_ops hdmi_dai_ops = { .startup = hdac_hdmi_pcm_open, .shutdown = hdac_hdmi_pcm_close, @@ -1784,6 +1913,16 @@ int hdac_hdmi_jack_init(struct snd_soc_dai *dai, int device, } } + /* add control for ELD Bytes */ + err = hdac_hdmi_create_eld_ctl(component, pcm); + if (err < 0) { + dev_err(&hdev->dev, + "eld control add failed with err: %d for pcm: %d\n", + err, device); + kfree(pcm); + return err; + } + list_add_tail(&pcm->head, &hdmi->pcm_list); return 0; -- 2.17.1

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[PATCH] ASoC: wm_adsp: Update naming in error handling
by Charles Keepax 18 Aug '20

18 Aug '20

It seems the datasheet has never used the word slave for this error status bit and has always used the term address error. So update the driver to match the datasheets and also in the process align a bit better with avoiding the use of such words where possible. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- sound/soc/codecs/wm_adsp.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 410cca57da52d..4ff0c6cfa32de 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -199,7 +199,7 @@ #define ADSP2_PMEM_ERR_ADDR_XMEM_ERR_ADDR 0x7C #define ADSP2_REGION_LOCK_ERR_MASK 0x8000 -#define ADSP2_SLAVE_ERR_MASK 0x4000 +#define ADSP2_ADDR_ERR_MASK 0x4000 #define ADSP2_WDT_TIMEOUT_STS_MASK 0x2000 #define ADSP2_CTRL_ERR_PAUSE_ENA 0x0002 #define ADSP2_CTRL_ERR_EINT 0x0001 @@ -4364,9 +4364,9 @@ irqreturn_t wm_adsp2_bus_error(int irq, void *data) wm_adsp_fatal_error(dsp); } - if (val & (ADSP2_SLAVE_ERR_MASK | ADSP2_REGION_LOCK_ERR_MASK)) { - if (val & ADSP2_SLAVE_ERR_MASK) - adsp_err(dsp, "bus error: slave error\n"); + if (val & (ADSP2_ADDR_ERR_MASK | ADSP2_REGION_LOCK_ERR_MASK)) { + if (val & ADSP2_ADDR_ERR_MASK) + adsp_err(dsp, "bus error: address error\n"); else adsp_err(dsp, "bus error: region lock error\n"); -- 2.11.0

2 1

[PATCH 0/3] ASoC: da7219: Reorganise device/codec level probe/remove
by Adam Thomson 18 Aug '20

18 Aug '20

This patch set reorganises and fixes device and codec level probe/remove handling within the driver, to allow clean probe and remove at the codec level. This set relates to an issue raised by Yong Zhi where a codec level re-probe would fail due to clks still being registered from the previous instantiation. In addition some improvements around regulator handling and soft reset have also been included. Adam Thomson (3): ASoC: da7219: Move required devm_* allocations to device level code ASoC: da7219: Move soft reset handling to codec level probe ASoC: da7219: Fix clock handling around codec level probe sound/soc/codecs/da7219-aad.c | 85 +++++--- sound/soc/codecs/da7219-aad.h | 3 + sound/soc/codecs/da7219.c | 493 +++++++++++++++++++++++------------------- sound/soc/codecs/da7219.h | 1 + 4 files changed, 328 insertions(+), 254 deletions(-) -- 1.9.1

3 6

[PATCH 0/3] ASoC: rt5682: Use clk APIs better
by Stephen Boyd 18 Aug '20

18 Aug '20

This patch series drops a printk message down to dev_dbg() because it was noisy and then migrates this driver to use clk_hw based APIs instead of clk based APIs because this device is a clk provider, not a clk consumer. I've only lightly tested the last two patches but I don't have all combinations of clks for this device. Cc: Cheng-Yi Chiang <cychiang(a)chromium.org> Cc: Shuming Fan <shumingf(a)realtek.com> Stephen Boyd (3): ASoC: rt5682: Use dev_dbg() in rt5682_clk_check() ASoC: rt5682: Drop usage of __clk_get_name() ASoC: rt5682: Use clk_hw based APIs for registration sound/soc/codecs/rt5682.c | 73 ++++++++++++--------------------------- sound/soc/codecs/rt5682.h | 2 -- 2 files changed, 23 insertions(+), 52 deletions(-) Based on the last patch to this driver in linux-next. base-commit: 6301adf942a31bed65e026a554e5bd55d9e731e1 -- Sent by a computer, using git, on the internet

2 4

[PATCH RFC] ASoC: wm8994: Avoid attempts to read unreadable registers
by Sylwester Nawrocki 18 Aug '20

18 Aug '20

The driver supports WM1811, WM8994, WM8958 devices but according to documentation and the regmap definitions the WM8958_DSP2_* registers are only available on WM8958. In current code these registers are being accessed as if they were available on all the three chips. When starting playback on WM1811 CODEC multiple errors like: "wm8994-codec wm8994-codec: ASoC: error at soc_component_read_no_lock on wm8994-codec: -5" can be seen, which is caused by attempts to read an unavailable WM8958_DSP2_PROGRAM register. The issue has been uncovered by recent commit "e2329ee ASoC: soc-component: add soc_component_err()". This patch adds a check in wm8958_aif_ev() callback so the DSP2 handling is only done for WM8958. Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> --- sound/soc/codecs/wm8958-dsp2.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/codecs/wm8958-dsp2.c b/sound/soc/codecs/wm8958-dsp2.c index 68a3b48..3bce9a1 100644 --- a/sound/soc/codecs/wm8958-dsp2.c +++ b/sound/soc/codecs/wm8958-dsp2.c @@ -412,8 +412,12 @@ int wm8958_aif_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); + struct wm8994 *control = dev_get_drvdata(component->dev->parent); int i; + if (control->type != WM8958) + return 0; + switch (event) { case SND_SOC_DAPM_POST_PMU: case SND_SOC_DAPM_PRE_PMU: -- 2.7.4

3 2

[PATCH] sound: remove duplicate "the the" phrase in Kconfig text
by Colin King 18 Aug '20

18 Aug '20

From: Colin Ian King <colin.king(a)canonical.com> There are a couple of occurrences of "the the" in the Kconfig text. Fix these. Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/xilinx/Kconfig | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/xilinx/Kconfig b/sound/soc/xilinx/Kconfig index 1d3586b68db7..5bd2730aab76 100644 --- a/sound/soc/xilinx/Kconfig +++ b/sound/soc/xilinx/Kconfig @@ -9,14 +9,14 @@ config SND_SOC_XILINX_I2S encapsulates PCM in AES format and sends AES data. config SND_SOC_XILINX_AUDIO_FORMATTER - tristate "Audio support for the the Xilinx audio formatter" + tristate "Audio support for the Xilinx audio formatter" help Select this option to enable Xilinx audio formatter support. This provides DMA platform device support for audio functionality. config SND_SOC_XILINX_SPDIF - tristate "Audio support for the the Xilinx SPDIF" + tristate "Audio support for the Xilinx SPDIF" help Select this option to enable Xilinx SPDIF Audio. This provides playback and capture of SPDIF audio in -- 2.27.0

3 2

[PATCH v3 00/10] ASoC: qdsp6: add gapless compressed audio support
by Srinivas Kandagatla 18 Aug '20

18 Aug '20

This patchset adds gapless compressed audio support on q6asm. Gapless on q6asm is implemented using 2 streams in a single q6asm session. First few patches such as stream id per each command, gapless flags and silence meta data are for preparedness for adding gapless support. Last patch implements copy callback to allow finer control over buffer offsets, specially in partial drain cases. This patchset is tested on RB3 aka DB845c platform. This patchset as it is will support gapless however QDSP can also support switching decoders on a single stream. Patches to support such feature are send in different patchset which involves adding generic interfaces. Thanks, srini Changes since v2:(mostly suggested by Pierre) - removed unnessary kernel style comments, - moved TIMESTAMP flag to respective patch. - move preparatory code from gapless support patch to new one. - fix subject prefix of one patch. - add comments to clarify valid stream_ids Srinivas Kandagatla (10): ASoC: q6asm: rename misleading session id variable ASoC: q6asm: make commands specific to streams ASoC: q6asm: use flags directly from q6asm-dai ASoC: q6asm: add length to write command token ASoC: q6asm: add support to remove intial and trailing silence ASoC: q6asm: add support to gapless flag in q6asm open ASoC: q6asm-dai: add next track metadata support ASoC: q6asm-dai: prepare set params to accept profile change ASoC: q6asm-dai: add gapless support ASoC: q6asm-dai: add support to copy callback sound/soc/qcom/qdsp6/q6asm-dai.c | 414 +++++++++++++++++++++++-------- sound/soc/qcom/qdsp6/q6asm.c | 169 +++++++++---- sound/soc/qcom/qdsp6/q6asm.h | 49 ++-- 3 files changed, 469 insertions(+), 163 deletions(-) -- 2.21.0

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[PATCH] sound/soc/intel: Fix spelling mistake "cant" --> "can't"
by Youling Tang 18 Aug '20

18 Aug '20

There is some spelling mistakes in a dev_err message. Fix it. Signed-off-by: Youling Tang <tangyouling(a)loongson.cn> --- sound/soc/intel/common/sst-firmware.c | 4 ++-- sound/soc/intel/haswell/sst-haswell-ipc.c | 6 +++--- 2 files changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/intel/common/sst-firmware.c b/sound/soc/intel/common/sst-firmware.c index 0594f89..1189ec3 100644 --- a/sound/soc/intel/common/sst-firmware.c +++ b/sound/soc/intel/common/sst-firmware.c @@ -130,7 +130,7 @@ static void block_list_remove(struct sst_dsp *dsp, err = block->ops->disable(block); if (err < 0) dev_err(dsp->dev, - "error: cant disable block %d:%d\n", + "error: can't disable block %d:%d\n", block->type, block->index); } } @@ -158,7 +158,7 @@ static int block_list_prepare(struct sst_dsp *dsp, ret = block->ops->enable(block); if (ret < 0) { dev_err(dsp->dev, - "error: cant disable block %d:%d\n", + "error: can't disable block %d:%d\n", block->type, block->index); goto err; } diff --git a/sound/soc/intel/haswell/sst-haswell-ipc.c b/sound/soc/intel/haswell/sst-haswell-ipc.c index 0ff89ea..773688b 100644 --- a/sound/soc/intel/haswell/sst-haswell-ipc.c +++ b/sound/soc/intel/haswell/sst-haswell-ipc.c @@ -1507,7 +1507,7 @@ static int sst_hsw_dx_state_dump(struct sst_hsw *hsw) ret = sst_dsp_dma_get_channel(sst, 0); if (ret < 0) { - dev_err(hsw->dev, "error: cant allocate dma channel %d\n", ret); + dev_err(hsw->dev, "error: can't allocate dma channel %d\n", ret); return ret; } @@ -1587,7 +1587,7 @@ int sst_hsw_dsp_load(struct sst_hsw *hsw) ret = sst_dsp_dma_get_channel(dsp, 0); if (ret < 0) { - dev_err(hsw->dev, "error: cant allocate dma channel %d\n", ret); + dev_err(hsw->dev, "error: can't allocate dma channel %d\n", ret); return ret; } @@ -1616,7 +1616,7 @@ static int sst_hsw_dsp_restore(struct sst_hsw *hsw) ret = sst_dsp_dma_get_channel(dsp, 0); if (ret < 0) { - dev_err(hsw->dev, "error: cant allocate dma channel %d\n", ret); + dev_err(hsw->dev, "error: can't allocate dma channel %d\n", ret); return ret; } -- 2.1.0

2 1