September 2020 - Alsa-devel - mailman.alsa-project.org

[PATCH] ALSA: ctl: Workaround for lockdep warning wrt card->ctl_files_rwlock
by Takashi Iwai 22 Sep '20

22 Sep '20

The recent change in lockdep for read lock caused the deadlock warnings in ALSA control code which uses the read_lock() for notification and else while write_lock_irqsave() is used for adding and removing the list entry. Although a deadlock would practically never hit in a real usage (the addition and the deletion can't happen with the notification), it's better to fix the read_lock() usage in a semantically correct way. This patch replaces the read_lock() calls with read_lock_irqsave() version for avoiding a reported deadlock. The notification code path takes the irq disablement in anyway, and other code paths are very short execution, hence there shouldn't be any big performance hit by this change. Fixes: e918188611f0 ("locking: More accurate annotations for read_lock()") Reported-by: syzbot+561a74f84100162990b2(a)syzkaller.appspotmail.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/control.c | 18 ++++++++++-------- 1 file changed, 10 insertions(+), 8 deletions(-) diff --git a/sound/core/control.c b/sound/core/control.c index e014598142df..421ddc76f264 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -150,14 +150,14 @@ void snd_ctl_notify(struct snd_card *card, unsigned int mask, return; if (card->shutdown) return; - read_lock(&card->ctl_files_rwlock); + read_lock_irqsave(&card->ctl_files_rwlock, flags); #if IS_ENABLED(CONFIG_SND_MIXER_OSS) card->mixer_oss_change_count++; #endif list_for_each_entry(ctl, &card->ctl_files, list) { if (!ctl->subscribed) continue; - spin_lock_irqsave(&ctl->read_lock, flags); + spin_lock(&ctl->read_lock); list_for_each_entry(ev, &ctl->events, list) { if (ev->id.numid == id->numid) { ev->mask |= mask; @@ -174,10 +174,10 @@ void snd_ctl_notify(struct snd_card *card, unsigned int mask, } _found: wake_up(&ctl->change_sleep); - spin_unlock_irqrestore(&ctl->read_lock, flags); + spin_unlock(&ctl->read_lock); kill_fasync(&ctl->fasync, SIGIO, POLL_IN); } - read_unlock(&card->ctl_files_rwlock); + read_unlock_irqrestore(&card->ctl_files_rwlock, flags); } EXPORT_SYMBOL(snd_ctl_notify); @@ -1951,8 +1951,9 @@ int snd_ctl_get_preferred_subdevice(struct snd_card *card, int type) { struct snd_ctl_file *kctl; int subdevice = -1; + unsigned long flags; - read_lock(&card->ctl_files_rwlock); + read_lock_irqsave(&card->ctl_files_rwlock, flags); list_for_each_entry(kctl, &card->ctl_files, list) { if (kctl->pid == task_pid(current)) { subdevice = kctl->preferred_subdevice[type]; @@ -1960,7 +1961,7 @@ int snd_ctl_get_preferred_subdevice(struct snd_card *card, int type) break; } } - read_unlock(&card->ctl_files_rwlock); + read_unlock_irqrestore(&card->ctl_files_rwlock, flags); return subdevice; } EXPORT_SYMBOL_GPL(snd_ctl_get_preferred_subdevice); @@ -2009,13 +2010,14 @@ static int snd_ctl_dev_disconnect(struct snd_device *device) { struct snd_card *card = device->device_data; struct snd_ctl_file *ctl; + unsigned long flags; - read_lock(&card->ctl_files_rwlock); + read_lock_irqsave(&card->ctl_files_rwlock, flags); list_for_each_entry(ctl, &card->ctl_files, list) { wake_up(&ctl->change_sleep); kill_fasync(&ctl->fasync, SIGIO, POLL_ERR); } - read_unlock(&card->ctl_files_rwlock); + read_unlock_irqrestore(&card->ctl_files_rwlock, flags); return snd_unregister_device(&card->ctl_dev); } -- 2.16.4

1 0

[PATCH 00/20] Add support for SATA/PCIe/USB2[3]/VIN/CSI on R8A774E1
by Lad Prabhakar 22 Sep '20

22 Sep '20

Hi All, This patch series adds support for the following peripherals on RZ/G2H SoC * PCIe * SATA * USB2 * USB3 * Audio * VIN * CSI Cheers, Prabhakar Lad Prabhakar (20): dt-bindings: pci: rcar-pci: Add device tree support for r8a774e1 arm64: dts: renesas: r8a774e1: Add PCIe device nodes dt-bindings: ata: renesas,rcar-sata: Add r8a774e1 support arm64: dts: renesas: r8a774e1: Add SATA controller node dt-bindings: phy: renesas,usb2-phy: Add r8a774e1 support arm64: dts: renesas: r8a774e1: Add USB2.0 phy and host (EHCI/OHCI) device nodes dt-bindings: usb: renesas,usb3-peri: Document r8a774e1 support dt-bindings: usb: usb-xhci: Document r8a774e1 support dt-bindings: phy: renesas,usb3-phy: Add r8a774e1 support arm64: dts: renesas: r8a774e1: Add USB3.0 device nodes dt-bindings: usb: renesas,usbhs: Add r8a774e1 support dt-bindings: dma: renesas,usb-dmac: Add binding for r8a774e1 arm64: dts: renesas: r8a774e1: Add USB-DMAC and HSUSB device nodes dt-bindings: sound: renesas,rsnd: Document r8a774e1 bindings arm64: dts: renesas: r8a774e1: Add audio support dt-bindings: media: renesas,csi2: Add R8A774E1 support dt-bindings: media: renesas,vin: Add R8A774E1 support media: rcar-csi2: Enable support for R8A774E1 media: rcar-vin: Enable support for R8A774E1 arm64: dts: renesas: r8a774e1: Add VIN and CSI-2 nodes .../bindings/ata/renesas,rcar-sata.yaml | 1 + .../bindings/dma/renesas,usb-dmac.yaml | 1 + .../bindings/media/renesas,csi2.yaml | 1 + .../bindings/media/renesas,vin.yaml | 1 + .../devicetree/bindings/pci/rcar-pci.txt | 1 + .../bindings/phy/renesas,usb2-phy.yaml | 1 + .../bindings/phy/renesas,usb3-phy.yaml | 1 + .../bindings/sound/renesas,rsnd.txt | 1 + .../bindings/usb/renesas,usb3-peri.yaml | 1 + .../bindings/usb/renesas,usbhs.yaml | 1 + .../devicetree/bindings/usb/usb-xhci.txt | 1 + arch/arm64/boot/dts/renesas/r8a774e1.dtsi | 989 +++++++++++++++++- drivers/media/platform/rcar-vin/rcar-core.c | 40 + drivers/media/platform/rcar-vin/rcar-csi2.c | 4 + 14 files changed, 1022 insertions(+), 22 deletions(-) -- 2.17.1

11 77

[PATCH] ALSA: seq: oss: Avoid mutex lock for a long-time ioctl
by Takashi Iwai 22 Sep '20

22 Sep '20

Recently we applied a fix to cover the whole OSS sequencer ioctls with the mutex for dealing with the possible races. This works fine in general, but in theory, this may lead to unexpectedly long stall if an ioctl like SNDCTL_SEQ_SYNC is issued and an event with the far future timestamp was queued. For fixing such a potential stall, this patch changes the mutex lock applied conditionally excluding such an ioctl command. Also, change the mutex_lock() with the interruptible version for user to allow escaping from the big-hammer mutex. Fixes: 80982c7e834e ("ALSA: seq: oss: Serialize ioctls") Suggested-by: Pavel Machek <pavel(a)ucw.cz> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/seq/oss/seq_oss.c | 7 +++++-- 1 file changed, 5 insertions(+), 2 deletions(-) diff --git a/sound/core/seq/oss/seq_oss.c b/sound/core/seq/oss/seq_oss.c index c8b9c0b315d8..250a92b18726 100644 --- a/sound/core/seq/oss/seq_oss.c +++ b/sound/core/seq/oss/seq_oss.c @@ -174,9 +174,12 @@ odev_ioctl(struct file *file, unsigned int cmd, unsigned long arg) if (snd_BUG_ON(!dp)) return -ENXIO; - mutex_lock(&register_mutex); + if (cmd != SNDCTL_SEQ_SYNC && + mutex_lock_interruptible(&register_mutex)) + return -ERESTARTSYS; rc = snd_seq_oss_ioctl(dp, cmd, arg); - mutex_unlock(&register_mutex); + if (cmd != SNDCTL_SEQ_SYNC) + mutex_unlock(&register_mutex); return rc; } -- 2.16.4

1 0

[PATCH] ASoC: hdac: make SOF HDA codec driver probe deterministic
by Kai Vehmanen 22 Sep '20

22 Sep '20

To provide backward compatibility to older systems, the SOF HDA driver allows user to specify which HDMI codec driver to use at runtime via kernel parameter. This mechanism has a subtle flaw in that it assumes the codec drivers not to be loaded when the SOF PCI driver is loaded. The problem is rooted in use of the hdev->type field. snd_hdac_ext_bus_device_init() initializes this field to HDA_DEV_ASOC. This signals the HDA core that ASoC drivers should be considered in driver matching (hda_bus_match()). The SOF and SST drivers continue by overriding this field to HDA_DEV_LEGACY and proceeding to load driver modules with request_module(). Correct drivers will get loaded and attached. If however the codec drivers are already loaded when snd_hdac_ext_bus_device_init() is called, the matching will not work as expected as device type is still set to HDA_DEV_ASOC. Specifically if hdac-hdmi is attached when machine driver is configured to use hdac-hda, this leads to out-of-bounds memory access in hda_dsp_hdmi_build_controls(). Fix the issue by adding codec type as a parameter to snd_hdac_ext_bus_device_init() and ensuring type is set correctly from the start. BugLink: https://github.com/thesofproject/linux/issues/2311 Fixes: 139c7febad1a ("ASoC: SOF: Intel: add support for snd-hda-codec-hdmi") Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- include/sound/hdaudio_ext.h | 2 +- sound/hda/ext/hdac_ext_bus.c | 5 +++-- sound/soc/intel/skylake/skl.c | 4 ++-- sound/soc/sof/intel/hda-codec.c | 17 ++++++++--------- 4 files changed, 14 insertions(+), 14 deletions(-) diff --git a/include/sound/hdaudio_ext.h b/include/sound/hdaudio_ext.h index ef88b20c7b0a..7abf74c1c474 100644 --- a/include/sound/hdaudio_ext.h +++ b/include/sound/hdaudio_ext.h @@ -10,7 +10,7 @@ int snd_hdac_ext_bus_init(struct hdac_bus *bus, struct device *dev, void snd_hdac_ext_bus_exit(struct hdac_bus *bus); int snd_hdac_ext_bus_device_init(struct hdac_bus *bus, int addr, - struct hdac_device *hdev); + struct hdac_device *hdev, int type); void snd_hdac_ext_bus_device_exit(struct hdac_device *hdev); void snd_hdac_ext_bus_device_remove(struct hdac_bus *bus); diff --git a/sound/hda/ext/hdac_ext_bus.c b/sound/hda/ext/hdac_ext_bus.c index d0a604c939df..765c40a6ccba 100644 --- a/sound/hda/ext/hdac_ext_bus.c +++ b/sound/hda/ext/hdac_ext_bus.c @@ -70,11 +70,12 @@ static void default_release(struct device *dev) * @bus: hdac bus to attach to * @addr: codec address * @hdev: hdac device to init + * @type: codec type (HDAC_DEV_*) to use for this device * * Returns zero for success or a negative error code. */ int snd_hdac_ext_bus_device_init(struct hdac_bus *bus, int addr, - struct hdac_device *hdev) + struct hdac_device *hdev, int type) { char name[15]; int ret; @@ -88,7 +89,7 @@ int snd_hdac_ext_bus_device_init(struct hdac_bus *bus, int addr, dev_err(bus->dev, "device init failed for hdac device\n"); return ret; } - hdev->type = HDA_DEV_ASOC; + hdev->type = type; hdev->dev.release = default_release; ret = snd_hdac_device_register(hdev); diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index ea00cf61d1a8..8b993722f74e 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -721,7 +721,7 @@ static int probe_codec(struct hdac_bus *bus, int addr) hda_codec->codec.bus = skl_to_hbus(skl); hdev = &hda_codec->codec.core; - err = snd_hdac_ext_bus_device_init(bus, addr, hdev); + err = snd_hdac_ext_bus_device_init(bus, addr, hdev, HDA_DEV_ASOC); if (err < 0) return err; @@ -736,7 +736,7 @@ static int probe_codec(struct hdac_bus *bus, int addr) if (!hdev) return -ENOMEM; - return snd_hdac_ext_bus_device_init(bus, addr, hdev); + return snd_hdac_ext_bus_device_init(bus, addr, hdev, HDA_DEV_ASOC); #endif /* CONFIG_SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC */ } diff --git a/sound/soc/sof/intel/hda-codec.c b/sound/soc/sof/intel/hda-codec.c index f6ba3b593e1f..6875fa570c2c 100644 --- a/sound/soc/sof/intel/hda-codec.c +++ b/sound/soc/sof/intel/hda-codec.c @@ -117,6 +117,7 @@ static int hda_codec_probe(struct snd_sof_dev *sdev, int address, #if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA_AUDIO_CODEC) struct hdac_hda_priv *hda_priv; struct hda_codec *codec; + int type = HDA_DEV_LEGACY; #endif struct hda_bus *hbus = sof_to_hbus(sdev); struct hdac_device *hdev; @@ -143,7 +144,11 @@ static int hda_codec_probe(struct snd_sof_dev *sdev, int address, hdev = &hda_priv->codec.core; codec = &hda_priv->codec; - ret = snd_hdac_ext_bus_device_init(&hbus->core, address, hdev); + /* only probe ASoC codec drivers for HDAC-HDMI */ + if (!hda_codec_use_common_hdmi && (resp & 0xFFFF0000) == IDISP_VID_INTEL) + type = HDA_DEV_ASOC; + + ret = snd_hdac_ext_bus_device_init(&hbus->core, address, hdev, type); if (ret < 0) return ret; @@ -161,13 +166,7 @@ static int hda_codec_probe(struct snd_sof_dev *sdev, int address, else codec->probe_id = 0; - /* - * if common HDMI codec driver is not used, codec load - * is skipped here and hdac_hdmi is used instead - */ - if (hda_codec_use_common_hdmi || - (resp & 0xFFFF0000) != IDISP_VID_INTEL) { - hdev->type = HDA_DEV_LEGACY; + if (type == HDA_DEV_LEGACY) { ret = hda_codec_load_module(codec); /* * handle ret==0 (no driver bound) as an error, but pass @@ -188,7 +187,7 @@ static int hda_codec_probe(struct snd_sof_dev *sdev, int address, if (!hdev) return -ENOMEM; - ret = snd_hdac_ext_bus_device_init(&hbus->core, address, hdev); + ret = snd_hdac_ext_bus_device_init(&hbus->core, address, hdev, HDA_DEV_ASOC); return ret; #endif -- 2.27.0

2 1

[PATCH] dt-bindings: tas2770: Fix I2C addresses for the TAS2770
by Dan Murphy 22 Sep '20

22 Sep '20

The I2C addresses listed in the yaml are not correct. The addresses can range from 0x41 through 0x48 based on register configurations. Fix the example and the description. Fixes: 4b7151dadfd4 ("dt-bindings: ASoC: Add tas2770 smart PA dt bindings") Signed-off-by: Dan Murphy <dmurphy(a)ti.com> --- Documentation/devicetree/bindings/sound/tas2770.yaml | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/tas2770.yaml b/Documentation/devicetree/bindings/sound/tas2770.yaml index 33a90f829c80..bb26d081c9fa 100644 --- a/Documentation/devicetree/bindings/sound/tas2770.yaml +++ b/Documentation/devicetree/bindings/sound/tas2770.yaml @@ -24,7 +24,7 @@ properties: reg: maxItems: 1 description: | - I2C address of the device can be one of these 0x4c, 0x4d, 0x4e or 0x4f + I2C address of the device can be between 0x41 to 0x48. reset-gpio: description: GPIO used to reset the device. @@ -62,9 +62,9 @@ examples: i2c0 { #address-cells = <1>; #size-cells = <0>; - codec: codec@4c { + codec: codec@41 { compatible = "ti,tas2770"; - reg = <0x4c>; + reg = <0x41>; #sound-dai-cells = <1>; interrupt-parent = <&gpio1>; interrupts = <14>; -- 2.28.0

2 2

[PATCH 1/2] dt-bindings: tas2562: Add the TAS2110 amplifier
by Dan Murphy 22 Sep '20

22 Sep '20

Add the TAS2110 amplifier compatible. Signed-off-by: Dan Murphy <dmurphy(a)ti.com> --- This patchset is based on top of patchset https://lore.kernel.org/patchwork/project/lkml/list/?series=463709 Documentation/devicetree/bindings/sound/tas2562.yaml | 2 ++ 1 file changed, 2 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/tas2562.yaml b/Documentation/devicetree/bindings/sound/tas2562.yaml index d3ceda356aaa..27f7132ba2ef 100644 --- a/Documentation/devicetree/bindings/sound/tas2562.yaml +++ b/Documentation/devicetree/bindings/sound/tas2562.yaml @@ -20,6 +20,7 @@ description: | https://www.ti.com/lit/gpn/tas2562 https://www.ti.com/lit/gpn/tas2563 https://www.ti.com/lit/gpn/tas2564 + https://www.ti.com/lit/gpn/tas2110 properties: compatible: @@ -27,6 +28,7 @@ properties: - ti,tas2562 - ti,tas2563 - ti,tas2564 + - ti,tas2110 reg: maxItems: 1 -- 2.28.0

2 2

[PATCH 0/3] tlv320aic3xx4 updates
by Miquel Raynal 22 Sep '20

22 Sep '20

Hello, While doing a kernel update on a sama5-based board I figured out the sound system was not working anymore, the debug session led me to the following commits. As I am not an audio expert at all, I am fully open to comments and suggestions. Thanks, Miquèl Miquel Raynal (3): ASoC: tlv320aic32x4: Ensure a minimum delay before clock stabilization ASoC: tlv320aic32x4: Fix bdiv clock rate derivation ASoC: tlv320aic32x4: Enable fast charge sound/soc/codecs/tlv320aic32x4-clk.c | 9 ++++++++- sound/soc/codecs/tlv320aic32x4.c | 17 ++++++++++++++--- sound/soc/codecs/tlv320aic32x4.h | 7 +++++++ 3 files changed, 29 insertions(+), 4 deletions(-) -- 2.20.1

3 11

[PATCH v6 00/14] ASoC: Intel: Catpt - Lynx and Wildcat point
by Cezary Rojewski 22 Sep '20

22 Sep '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. Series is dependent on linux-spi change: spi: pxa2xx: Add SSC2 and SSPSP2 SSP registers https://www.spinics.net/lists/linux-spi/msg23885.html which has been already merged and is now part of linux-spi tree. Bulk of series content is device driver core code - everything up to patch 7/14 - with fs entries and trace macros introduced right after. While each core patch is shaped in such a way that no unavailable members are ever called, until patch 10/14 is applied, no code compilation can occur as no Makefile is present. Once said patch is added, Makefile and Kconfig are implemented and driver module compiles as expected. 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 v6: - reordered and reorganized code for patches 1/13 - 8/13 of v5, so each patches makes use of no member or function which is unavailable to it. Series size increased from 13 to 14 patches: addition of base members e.g.: registers has been split from addition of device.c file which describes acpi device behavior Changes in v5: https://www.spinics.net/lists/alsa-devel/msg115621.html Basically everything below is result of Andy's review. Thank you Andy for taking time into this detailed review - catpt now makes use of common linux/pxa2xx_ssp.h header file, removing redundant SSP register declarations in the process. As stated in the opening, this is dependent upon linux-spi change: spi: pxa2xx: Add SSC2 and SSPSP2 SSP registers - updated Kconfig by removing DMADEVICES and adding COMPILE_TEST as optional depends-on - updated all register macros definitions to be more safe against common arithmetics when specifying macro's parameters - removed CONFIG_PM and CONFIG_PM_SLEEP usage in favor of __maybe_unused - all 'if (ret < 0)' converted to simple 'if (ret)' whenever possible - fixed erroneous check for platform_device_register_data within catpt_register_board() - _SLAVE/_MASTER replaced with more inclusive _CONSUMER/_PROVIDER for enum catpt_ssp_mode - catpt_acpi_probe() is now making use of high-level wrappers for ioremapping and resource assignment, reducing function's code size - due to improved catpt_acpi_probe() behavior, catpt_acpi_remove() needs not to cast dma_free_coherent() any longer - DMA source and destrination maxburst now of value 16, see: https://www.spinics.net/lists/alsa-devel/msg114394.html - simplified catpt_dsp_update_lpclock() as list_for_each_entry() is empty-safe by default - dropped '_SSP_' from all names of all CATPT_SSP_SSXXX_DEFAULT macros - catpt_updatel_pci now makes use of linux/pci.h and uapi/linux/pci.h constants such as: PCI_PM_CTRL_STATE_MASK and PCI_D3hot Changes in v4: https://www.spinics.net/lists/alsa-devel/msg113762.html - 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: https://www.spinics.net/lists/alsa-devel/msg113660.html - fixed SPDX formatting for all header files as well as pcm.c - fixed size provided to memcpy() in 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: https://www.spinics.net/lists/alsa-devel/msg113549.html - 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 (14): ASoC: Intel: Add catpt base members ASoC: Intel: catpt: Implement IPC protocol ASoC: Intel: catpt: Add IPC message handlers ASoC: Intel: catpt: Define DSP operations ASoC: Intel: catpt: Firmware loading and context restore ASoC: Intel: catpt: PCM operations ASoC: Intel: catpt: Device driver lifecycle 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 | 24 +- 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 | 189 +++++ sound/soc/intel/catpt/device.c | 354 ++++++++ sound/soc/intel/catpt/dsp.c | 572 +++++++++++++ sound/soc/intel/catpt/fs.c | 79 ++ sound/soc/intel/catpt/ipc.c | 298 +++++++ sound/soc/intel/catpt/loader.c | 671 +++++++++++++++ sound/soc/intel/catpt/messages.c | 313 +++++++ sound/soc/intel/catpt/messages.h | 401 +++++++++ sound/soc/intel/catpt/pcm.c | 1212 +++++++++++++++++++++++++++ sound/soc/intel/catpt/registers.h | 178 ++++ sound/soc/intel/catpt/trace.h | 83 ++ 19 files changed, 4377 insertions(+), 143 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 23

[PATCH v2 0/2] DAI driver for new XCVR IP
by Viorel Suman (OSS) 21 Sep '20

21 Sep '20

From: Viorel Suman <viorel.suman(a)nxp.com> DAI driver for new XCVR IP found in i.MX8MP. Viorel Suman (2): ASoC: fsl_xcvr: Add XCVR ASoC CPU DAI driver ASoC: dt-bindings: fsl_xcvr: Add document for XCVR Changes since v1: - improved 6- and 12-ch layout comment - used regmap polling function, improved clocks handling in runtime_resume - added FW size check in FW load function, improved IRQ handler, removed dummy IRQ handlers - fixed yaml file .../devicetree/bindings/sound/fsl,xcvr.yaml | 103 ++ sound/soc/fsl/Kconfig | 10 + sound/soc/fsl/Makefile | 2 + sound/soc/fsl/fsl_xcvr.c | 1343 ++++++++++++++++++++ sound/soc/fsl/fsl_xcvr.h | 266 ++++ 5 files changed, 1724 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/fsl,xcvr.yaml create mode 100644 sound/soc/fsl/fsl_xcvr.c create mode 100644 sound/soc/fsl/fsl_xcvr.h -- 2.7.4

1 2

[PATCH] ALSA: usb-audio: Add mixer support for Pioneer DJ DJM-250MK2
by František Kučera 21 Sep '20

21 Sep '20

From: František Kučera <franta-linux(a)frantovo.cz> This patch extends support for DJM-250MK2 and allows mapping playback and capture channels to available sources. Configures the card through USB commands. --- sound/usb/mixer_quirks.c | 216 +++++++++++++++++++++++++++++++++++++++ 1 file changed, 216 insertions(+) diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index 199cdbfdc761..b7816a71c42d 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -2602,6 +2602,219 @@ static int snd_bbfpro_controls_create(struct usb_mixer_interface *mixer) return 0; } +/* + * Pioneer DJ DJM-250MK2 and maybe other DJM models + * + * For playback, no duplicate mapping should be set. + * There are three mixer stereo channels (CH1, CH2, AUX) + * and three stereo sources (Playback 1-2, Playback 3-4, Playback 5-6). + * Each channel should be mapped just once to one source. + * If mapped multiple times, only one source will play on given channel + * (sources are not mixed together). + * + * For recording, duplicate mapping is OK. We will get the same signal multiple times. + * + * Channels 7-8 are in both directions fixed to FX SEND / FX RETURN. + * + * See also notes in the quirks-table.h file. + */ + +typedef struct { + const u16 wIndex; + const u16 wValue; + const char *name; +} snd_pioneer_djm_option; + +static const snd_pioneer_djm_option SND_PIONEER_DJM_OPTIONS_CAPTURE_LEVEL[] = { + { .name = "-5 dB", .wValue = 0x0300, .wIndex = 0x8003 }, + { .name = "-10 dB", .wValue = 0x0200, .wIndex = 0x8003 }, + { .name = "-15 dB", .wValue = 0x0100, .wIndex = 0x8003 }, + { .name = "-19 dB", .wValue = 0x0000, .wIndex = 0x8003 } +}; + +static const snd_pioneer_djm_option SND_PIONEER_DJM_OPTIONS_CAPTURE_CH12[] = { + { .name = "CH1 Control Tone PHONO", .wValue = 0x0103, .wIndex = 0x8002 }, + { .name = "CH1 Control Tone LINE", .wValue = 0x0100, .wIndex = 0x8002 }, + { .name = "Post CH1 Fader", .wValue = 0x0106, .wIndex = 0x8002 }, + { .name = "Cross Fader A", .wValue = 0x0107, .wIndex = 0x8002 }, + { .name = "Cross Fader B", .wValue = 0x0108, .wIndex = 0x8002 }, + { .name = "MIC", .wValue = 0x0109, .wIndex = 0x8002 }, + { .name = "AUX", .wValue = 0x010d, .wIndex = 0x8002 }, + { .name = "REC OUT", .wValue = 0x010a, .wIndex = 0x8002 } +}; + +static const snd_pioneer_djm_option SND_PIONEER_DJM_OPTIONS_CAPTURE_CH34[] = { + { .name = "CH2 Control Tone PHONO", .wValue = 0x0203, .wIndex = 0x8002 }, + { .name = "CH2 Control Tone LINE", .wValue = 0x0200, .wIndex = 0x8002 }, + { .name = "Post CH2 Fader", .wValue = 0x0206, .wIndex = 0x8002 }, + { .name = "Cross Fader A", .wValue = 0x0207, .wIndex = 0x8002 }, + { .name = "Cross Fader B", .wValue = 0x0208, .wIndex = 0x8002 }, + { .name = "MIC", .wValue = 0x0209, .wIndex = 0x8002 }, + { .name = "AUX", .wValue = 0x020d, .wIndex = 0x8002 }, + { .name = "REC OUT", .wValue = 0x020a, .wIndex = 0x8002 } +}; + +static const snd_pioneer_djm_option SND_PIONEER_DJM_OPTIONS_CAPTURE_CH56[] = { + { .name = "REC OUT", .wValue = 0x030a, .wIndex = 0x8002 }, + { .name = "Post CH1 Fader", .wValue = 0x0311, .wIndex = 0x8002 }, + { .name = "Post CH2 Fader", .wValue = 0x0312, .wIndex = 0x8002 }, + { .name = "Cross Fader A", .wValue = 0x0307, .wIndex = 0x8002 }, + { .name = "Cross Fader B", .wValue = 0x0308, .wIndex = 0x8002 }, + { .name = "MIC", .wValue = 0x0309, .wIndex = 0x8002 }, + { .name = "AUX", .wValue = 0x030d, .wIndex = 0x8002 } +}; + +static const snd_pioneer_djm_option SND_PIONEER_DJM_OPTIONS_PLAYBACK_12[] = { + { .name = "CH1", .wValue = 0x0100, .wIndex = 0x8016 }, + { .name = "CH2", .wValue = 0x0101, .wIndex = 0x8016 }, + { .name = "AUX", .wValue = 0x0104, .wIndex = 0x8016 } +}; + +static const snd_pioneer_djm_option SND_PIONEER_DJM_OPTIONS_PLAYBACK_34[] = { + { .name = "CH1", .wValue = 0x0200, .wIndex = 0x8016 }, + { .name = "CH2", .wValue = 0x0201, .wIndex = 0x8016 }, + { .name = "AUX", .wValue = 0x0204, .wIndex = 0x8016 } +}; + +static const snd_pioneer_djm_option SND_PIONEER_DJM_OPTIONS_PLAYBACK_56[] = { + { .name = "CH1", .wValue = 0x0300, .wIndex = 0x8016 }, + { .name = "CH2", .wValue = 0x0301, .wIndex = 0x8016 }, + { .name = "AUX", .wValue = 0x0304, .wIndex = 0x8016 } +}; + +typedef struct { + const char *name; + const snd_pioneer_djm_option *options; + const size_t count; + const u16 default_value; +} snd_pioneer_djm_option_group; + +#define SND_PIONEER_DJM_OPTION_GROUP_ITEM(_name, suffix, _default_value) { \ + .name = _name, \ + .options = SND_PIONEER_DJM_OPTIONS_##suffix, \ + .count = ARRAY_SIZE(SND_PIONEER_DJM_OPTIONS_##suffix), \ + .default_value = _default_value } + +static const snd_pioneer_djm_option_group SND_PIONEER_DJM_OPTION_GROUPS[] = { + SND_PIONEER_DJM_OPTION_GROUP_ITEM("Master Capture Level Capture Switch", CAPTURE_LEVEL, 0), + SND_PIONEER_DJM_OPTION_GROUP_ITEM("Capture 1-2 Capture Switch", CAPTURE_CH12, 2), + SND_PIONEER_DJM_OPTION_GROUP_ITEM("Capture 3-4 Capture Switch", CAPTURE_CH34, 2), + SND_PIONEER_DJM_OPTION_GROUP_ITEM("Capture 5-6 Capture Switch", CAPTURE_CH56, 0), + SND_PIONEER_DJM_OPTION_GROUP_ITEM("Playback 1-2 Playback Switch", PLAYBACK_12, 0), + SND_PIONEER_DJM_OPTION_GROUP_ITEM("Playback 3-4 Playback Switch", PLAYBACK_34, 1), + SND_PIONEER_DJM_OPTION_GROUP_ITEM("Playback 5-6 Playback Switch", PLAYBACK_56, 2) +}; + +// layout of the kcontrol->private_value: +#define SND_PIONEER_DJM_VALUE_MASK 0x0000ffff +#define SND_PIONEER_DJM_GROUP_MASK 0xffff0000 +#define SND_PIONEER_DJM_GROUP_SHIFT 16 + +static int snd_pioneer_djm_controls_info(struct snd_kcontrol *kctl, struct snd_ctl_elem_info *info) +{ + u16 group_index = kctl->private_value >> SND_PIONEER_DJM_GROUP_SHIFT; + size_t count; + const char *name; + const snd_pioneer_djm_option_group *group; + + if (group_index < ARRAY_SIZE(SND_PIONEER_DJM_OPTION_GROUPS)) { + group = &SND_PIONEER_DJM_OPTION_GROUPS[group_index]; + count = group->count; + if (info->value.enumerated.item >= count) + info->value.enumerated.item = count - 1; + name = group->options[info->value.enumerated.item].name; + strlcpy(info->value.enumerated.name, name, sizeof(info->value.enumerated.name)); + info->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + info->count = 1; + info->value.enumerated.items = count; + return 0; + } else { + return -EINVAL; + } +} + +static int snd_pioneer_djm_controls_update(struct usb_mixer_interface *mixer, u16 group, u16 value) +{ + int err; + + if (group < ARRAY_SIZE(SND_PIONEER_DJM_OPTION_GROUPS) + && value < SND_PIONEER_DJM_OPTION_GROUPS[group].count) { + err = snd_usb_lock_shutdown(mixer->chip); + if (err) + return err; + + err = snd_usb_ctl_msg( + mixer->chip->dev, usb_sndctrlpipe(mixer->chip->dev, 0), + USB_REQ_SET_FEATURE, + USB_DIR_OUT | USB_TYPE_VENDOR | USB_RECIP_DEVICE, + SND_PIONEER_DJM_OPTION_GROUPS[group].options[value].wValue, + SND_PIONEER_DJM_OPTION_GROUPS[group].options[value].wIndex, + NULL, 0); + + snd_usb_unlock_shutdown(mixer->chip); + return err; + } else { + return -EINVAL; + } +} + +static int snd_pioneer_djm_controls_get(struct snd_kcontrol *kctl, struct snd_ctl_elem_value *elem) +{ + elem->value.enumerated.item[0] = kctl->private_value & SND_PIONEER_DJM_VALUE_MASK; + return 0; +} + +static int snd_pioneer_djm_controls_put(struct snd_kcontrol *kctl, struct snd_ctl_elem_value *elem) +{ + struct usb_mixer_elem_list *list = snd_kcontrol_chip(kctl); + struct usb_mixer_interface *mixer = list->mixer; + unsigned long private_value = kctl->private_value; + + u16 group = (private_value & SND_PIONEER_DJM_GROUP_MASK) >> SND_PIONEER_DJM_GROUP_SHIFT; + u16 value = elem->value.enumerated.item[0]; + + kctl->private_value = group << SND_PIONEER_DJM_GROUP_SHIFT | value; + + return snd_pioneer_djm_controls_update(mixer, group, value); +} + +static int snd_pioneer_djm_controls_resume(struct usb_mixer_elem_list *list) +{ + unsigned long private_value = list->kctl->private_value; + u16 group = (private_value & SND_PIONEER_DJM_GROUP_MASK) >> SND_PIONEER_DJM_GROUP_SHIFT; + u16 value = (private_value & SND_PIONEER_DJM_VALUE_MASK); + + return snd_pioneer_djm_controls_update(list->mixer, group, value); +} + +static int snd_pioneer_djm_controls_create(struct usb_mixer_interface *mixer) +{ + int err, i; + const snd_pioneer_djm_option_group *group; + struct snd_kcontrol_new knew = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .index = 0, + .info = snd_pioneer_djm_controls_info, + .get = snd_pioneer_djm_controls_get, + .put = snd_pioneer_djm_controls_put + }; + + for (i = 0; i < ARRAY_SIZE(SND_PIONEER_DJM_OPTION_GROUPS); i++) { + group = &SND_PIONEER_DJM_OPTION_GROUPS[i]; + knew.name = group->name; + knew.private_value = i << SND_PIONEER_DJM_GROUP_SHIFT | group->default_value; + err = snd_pioneer_djm_controls_update(mixer, i, group->default_value); + if (err) + return err; + err = add_single_ctl_with_resume(mixer, 0, snd_pioneer_djm_controls_resume, + &knew, NULL); + if (err) + return err; + } + return 0; +} + int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) { int err = 0; @@ -2706,6 +2919,9 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) case USB_ID(0x2a39, 0x3fb0): /* RME Babyface Pro FS */ err = snd_bbfpro_controls_create(mixer); break; + case USB_ID(0x2b73, 0x0017): /* Pioneer DJ DJM-250MK2 */ + err = snd_pioneer_djm_controls_create(mixer); + break; } return err; -- 2.20.1

1 0

[PATCH] ASoC: SOF: pm: Fix prepare callback behavior for OF usecase
by Kai Vehmanen 21 Sep '20

21 Sep '20

From: Daniel Baluta <daniel.baluta(a)nxp.com> On i.MX platforms PM is not managed via ACPI although CONFIG_ACPI can be set. So, in order to correctly set the system target state we introduce a flag for platforms that require to use acpi target states. Signed-off-by: Daniel Baluta <daniel.baluta(a)nxp.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> --- include/sound/sof.h | 2 ++ sound/soc/sof/pm.c | 12 +++++++----- sound/soc/sof/sof-pci-dev.c | 9 +++++++++ 3 files changed, 18 insertions(+), 5 deletions(-) diff --git a/include/sound/sof.h b/include/sound/sof.h index f3e716c8ce1c..9aa055289dcc 100644 --- a/include/sound/sof.h +++ b/include/sound/sof.h @@ -66,6 +66,8 @@ struct sof_dev_desc { /* alternate list of machines using this configuration */ struct snd_soc_acpi_mach *alt_machines; + bool use_acpi_target_states; + /* Platform resource indexes in BAR / ACPI resources. */ /* Must set to -1 if not used - add new items to end */ int resindex_lpe_base; diff --git a/sound/soc/sof/pm.c b/sound/soc/sof/pm.c index a5f7c7f024a1..c83fb6255961 100644 --- a/sound/soc/sof/pm.c +++ b/sound/soc/sof/pm.c @@ -305,15 +305,17 @@ EXPORT_SYMBOL(snd_sof_suspend); int snd_sof_prepare(struct device *dev) { struct snd_sof_dev *sdev = dev_get_drvdata(dev); + const struct sof_dev_desc *desc = sdev->pdata->desc; + + /* will suspend to S3 by default */ + sdev->system_suspend_target = SOF_SUSPEND_S3; + + if (!desc->use_acpi_target_states) + return 0; #if defined(CONFIG_ACPI) if (acpi_target_system_state() == ACPI_STATE_S0) sdev->system_suspend_target = SOF_SUSPEND_S0IX; - else - sdev->system_suspend_target = SOF_SUSPEND_S3; -#else - /* will suspend to S3 by default */ - sdev->system_suspend_target = SOF_SUSPEND_S3; #endif return 0; diff --git a/sound/soc/sof/sof-pci-dev.c b/sound/soc/sof/sof-pci-dev.c index 8c53f6935417..8f62e3487dc1 100644 --- a/sound/soc/sof/sof-pci-dev.c +++ b/sound/soc/sof/sof-pci-dev.c @@ -77,6 +77,7 @@ static const struct dmi_system_id community_key_platforms[] = { #if IS_ENABLED(CONFIG_SND_SOC_SOF_APOLLOLAKE) static const struct sof_dev_desc bxt_desc = { .machines = snd_soc_acpi_intel_bxt_machines, + .use_acpi_target_states = true, .resindex_lpe_base = 0, .resindex_pcicfg_base = -1, .resindex_imr_base = -1, @@ -94,6 +95,7 @@ static const struct sof_dev_desc bxt_desc = { #if IS_ENABLED(CONFIG_SND_SOC_SOF_GEMINILAKE) static const struct sof_dev_desc glk_desc = { .machines = snd_soc_acpi_intel_glk_machines, + .use_acpi_target_states = true, .resindex_lpe_base = 0, .resindex_pcicfg_base = -1, .resindex_imr_base = -1, @@ -139,6 +141,7 @@ static const struct sof_dev_desc tng_desc = { static const struct sof_dev_desc cnl_desc = { .machines = snd_soc_acpi_intel_cnl_machines, .alt_machines = snd_soc_acpi_intel_cnl_sdw_machines, + .use_acpi_target_states = true, .resindex_lpe_base = 0, .resindex_pcicfg_base = -1, .resindex_imr_base = -1, @@ -157,6 +160,7 @@ static const struct sof_dev_desc cnl_desc = { static const struct sof_dev_desc cfl_desc = { .machines = snd_soc_acpi_intel_cfl_machines, .alt_machines = snd_soc_acpi_intel_cfl_sdw_machines, + .use_acpi_target_states = true, .resindex_lpe_base = 0, .resindex_pcicfg_base = -1, .resindex_imr_base = -1, @@ -175,6 +179,7 @@ static const struct sof_dev_desc cfl_desc = { static const struct sof_dev_desc cml_desc = { .machines = snd_soc_acpi_intel_cml_machines, .alt_machines = snd_soc_acpi_intel_cml_sdw_machines, + .use_acpi_target_states = true, .resindex_lpe_base = 0, .resindex_pcicfg_base = -1, .resindex_imr_base = -1, @@ -193,6 +198,7 @@ static const struct sof_dev_desc cml_desc = { static const struct sof_dev_desc icl_desc = { .machines = snd_soc_acpi_intel_icl_machines, .alt_machines = snd_soc_acpi_intel_icl_sdw_machines, + .use_acpi_target_states = true, .resindex_lpe_base = 0, .resindex_pcicfg_base = -1, .resindex_imr_base = -1, @@ -211,6 +217,7 @@ static const struct sof_dev_desc icl_desc = { static const struct sof_dev_desc tgl_desc = { .machines = snd_soc_acpi_intel_tgl_machines, .alt_machines = snd_soc_acpi_intel_tgl_sdw_machines, + .use_acpi_target_states = true, .resindex_lpe_base = 0, .resindex_pcicfg_base = -1, .resindex_imr_base = -1, @@ -244,6 +251,7 @@ static const struct sof_dev_desc tglh_desc = { #if IS_ENABLED(CONFIG_SND_SOC_SOF_ELKHARTLAKE) static const struct sof_dev_desc ehl_desc = { .machines = snd_soc_acpi_intel_ehl_machines, + .use_acpi_target_states = true, .resindex_lpe_base = 0, .resindex_pcicfg_base = -1, .resindex_imr_base = -1, @@ -261,6 +269,7 @@ static const struct sof_dev_desc ehl_desc = { #if IS_ENABLED(CONFIG_SND_SOC_SOF_JASPERLAKE) static const struct sof_dev_desc jsl_desc = { .machines = snd_soc_acpi_intel_jsl_machines, + .use_acpi_target_states = true, .resindex_lpe_base = 0, .resindex_pcicfg_base = -1, .resindex_imr_base = -1, -- 2.27.0

2 1

[PATCH v3 1/2] dt-bindings: tlv320adcx140: Add GPIO config and drive config
by Camel Guo 21 Sep '20

21 Sep '20

From: Camel Guo <camelg(a)axis.com> Add properties for configuring the General Purpose Input Output (GPIO). There are 2 settings for GPIO, configuration and the output drive type. Signed-off-by: Camel Guo <camelg(a)axis.com> Acked-by: Dan Murphy <dmurphy(a)ti.com> --- v3: - Fix typo - Add Acked-By from Dan .../bindings/sound/tlv320adcx140.yaml | 44 +++++++++++++++++++ 1 file changed, 44 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/tlv320adcx140.yaml b/Documentation/devicetree/bindings/sound/tlv320adcx140.yaml index f578f17f3e04..2f95ccde4dc3 100644 --- a/Documentation/devicetree/bindings/sound/tlv320adcx140.yaml +++ b/Documentation/devicetree/bindings/sound/tlv320adcx140.yaml @@ -134,6 +134,49 @@ patternProperties: 4d - Drive weak low and active high 5d - Drive Hi-Z and active high + ti,gpio-config: + description: | + Defines the configuration and output drive for the General Purpose + Input and Output pin (GPIO1). Its value is a pair, the first value is for + the configuration type and the second value is for the output drive + type. The array is defined as <GPIO1_CFG GPIO1_DRV> + + configuration for the GPIO pin can be one of the following: + 0 - disabled + 1 - GPIO1 is configured as a general-purpose output (GPO) + 2 - (default) GPIO1 is configured as a device interrupt output (IRQ) + 3 - GPIO1 is configured as a secondary ASI output (SDOUT2) + 4 - GPIO1 is configured as a PDM clock output (PDMCLK) + 8 - GPIO1 is configured as an input to control when MICBIAS turns on or + off (MICBIAS_EN) + 9 - GPIO1 is configured as a general-purpose input (GPI) + 10 - GPIO1 is configured as a master clock input (MCLK) + 11 - GPIO1 is configured as an ASI input for daisy-chain (SDIN) + 12 - GPIO1 is configured as a PDM data input for channel 1 and channel 2 + (PDMDIN1) + 13 - GPIO1 is configured as a PDM data input for channel 3 and channel 4 + (PDMDIN2) + 14 - GPIO1 is configured as a PDM data input for channel 5 and channel 6 + (PDMDIN3) + 15 - GPIO1 is configured as a PDM data input for channel 7 and channel 8 + (PDMDIN4) + + output drive type for the GPIO pin can be one of the following: + 0 - Hi-Z output + 1 - Drive active low and active high + 2 - (default) Drive active low and weak high + 3 - Drive active low and Hi-Z + 4 - Drive weak low and active high + 5 - Drive Hi-Z and active high + + allOf: + - $ref: /schemas/types.yaml#/definitions/uint32-array + - minItems: 2 + maxItems: 2 + items: + maximum: 15 + default: [2, 2] + required: - compatible - reg @@ -150,6 +193,7 @@ examples: ti,mic-bias-source = <6>; ti,pdm-edge-select = <0 1 0 1>; ti,gpi-config = <4 5 6 7>; + ti,gpio-config = <10 2>; ti,gpo-config-1 = <0 0>; ti,gpo-config-2 = <0 0>; reset-gpios = <&gpio0 14 GPIO_ACTIVE_HIGH>; -- 2.20.1

3 4

[PATCH v5 1/2] dt-bindings: tlv320adcx140: Add GPIO config and drive config
by Camel Guo 21 Sep '20

21 Sep '20

From: Camel Guo <camelg(a)axis.com> Add properties for configuring the General Purpose Input Output (GPIO). There are 2 settings for GPIO, configuration and the output drive type. Signed-off-by: Camel Guo <camelg(a)axis.com> Acked-by: Dan Murphy <dmurphy(a)ti.com> --- v4: - Rebase v3: - Fix typo - Add Acked-By from Dan .../bindings/sound/tlv320adcx140.yaml | 44 +++++++++++++++++++ 1 file changed, 44 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/tlv320adcx140.yaml b/Documentation/devicetree/bindings/sound/tlv320adcx140.yaml index e79f8d1891e4..798b366fe6d1 100644 --- a/Documentation/devicetree/bindings/sound/tlv320adcx140.yaml +++ b/Documentation/devicetree/bindings/sound/tlv320adcx140.yaml @@ -140,6 +140,49 @@ patternProperties: 4d - Drive weak low and active high 5d - Drive Hi-Z and active high + ti,gpio-config: + description: | + Defines the configuration and output drive for the General Purpose + Input and Output pin (GPIO1). Its value is a pair, the first value is for + the configuration type and the second value is for the output drive + type. The array is defined as <GPIO1_CFG GPIO1_DRV> + + configuration for the GPIO pin can be one of the following: + 0 - disabled + 1 - GPIO1 is configured as a general-purpose output (GPO) + 2 - (default) GPIO1 is configured as a device interrupt output (IRQ) + 3 - GPIO1 is configured as a secondary ASI output (SDOUT2) + 4 - GPIO1 is configured as a PDM clock output (PDMCLK) + 8 - GPIO1 is configured as an input to control when MICBIAS turns on or + off (MICBIAS_EN) + 9 - GPIO1 is configured as a general-purpose input (GPI) + 10 - GPIO1 is configured as a master clock input (MCLK) + 11 - GPIO1 is configured as an ASI input for daisy-chain (SDIN) + 12 - GPIO1 is configured as a PDM data input for channel 1 and channel 2 + (PDMDIN1) + 13 - GPIO1 is configured as a PDM data input for channel 3 and channel 4 + (PDMDIN2) + 14 - GPIO1 is configured as a PDM data input for channel 5 and channel 6 + (PDMDIN3) + 15 - GPIO1 is configured as a PDM data input for channel 7 and channel 8 + (PDMDIN4) + + output drive type for the GPIO pin can be one of the following: + 0 - Hi-Z output + 1 - Drive active low and active high + 2 - (default) Drive active low and weak high + 3 - Drive active low and Hi-Z + 4 - Drive weak low and active high + 5 - Drive Hi-Z and active high + + allOf: + - $ref: /schemas/types.yaml#/definitions/uint32-array + - minItems: 2 + maxItems: 2 + items: + maximum: 15 + default: [2, 2] + required: - compatible - reg @@ -156,6 +199,7 @@ examples: ti,mic-bias-source = <6>; ti,pdm-edge-select = <0 1 0 1>; ti,gpi-config = <4 5 6 7>; + ti,gpio-config = <10 2>; ti,gpo-config-1 = <0 0>; ti,gpo-config-2 = <0 0>; reset-gpios = <&gpio0 14 GPIO_ACTIVE_HIGH>; -- 2.20.1

2 2

[PATCH] ASoC: SOF: imx: add missing MODULE_LICENSE() for imx-common
by Kai Vehmanen 21 Sep '20

21 Sep '20

Fix build warning: WARNING: modpost: missing MODULE_LICENSE() in sound/soc/sof/imx/imx-common.o Fixes: 18ebffe4d043 ("ASoC: SOF: imx: Add debug support for imx platforms") Reported-by: Stephen Rothwell <sfr(a)canb.auug.org.au> Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Daniel Baluta <daniel.baluta(a)gmail.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/sof/imx/imx-common.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/sof/imx/imx-common.c b/sound/soc/sof/imx/imx-common.c index 63d8a5d7bc44..5fee637834c2 100644 --- a/sound/soc/sof/imx/imx-common.c +++ b/sound/soc/sof/imx/imx-common.c @@ -4,6 +4,7 @@ // // Common helpers for the audio DSP on i.MX8 +#include <linux/module.h> #include <sound/sof/xtensa.h> #include "../ops.h" @@ -70,3 +71,5 @@ void imx8_dump(struct snd_sof_dev *sdev, u32 flags) IMX8_STACK_DUMP_SIZE); } EXPORT_SYMBOL(imx8_dump); + +MODULE_LICENSE("Dual BSD/GPL"); -- 2.27.0

2 1

[PATCH] ASoC: cs42l51: add additional ADC volume controls
by Guillermo Rodríguez 21 Sep '20

21 Sep '20

Add volume controls for: - Analog programmable gain amplifier (PGA) (-3 .. +12 dB) - ADC attenuator (0 .. -96 dB) Signed-off-by: Guillermo Rodríguez <guille.rodriguez(a)gmail.com> --- sound/soc/codecs/cs42l51.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c index 764f2ef8f59d..097c4e8d9950 100644 --- a/sound/soc/codecs/cs42l51.c +++ b/sound/soc/codecs/cs42l51.c @@ -122,6 +122,9 @@ static const char *chan_mix[] = { "R L", }; +static const DECLARE_TLV_DB_SCALE(pga_tlv, -300, 50, 0); +static const DECLARE_TLV_DB_SCALE(adc_att_tlv, -9600, 100, 0); + static SOC_ENUM_SINGLE_EXT_DECL(cs42l51_chan_mix, chan_mix); static const struct snd_kcontrol_new cs42l51_snd_controls[] = { @@ -138,6 +141,12 @@ static const struct snd_kcontrol_new cs42l51_snd_controls[] = { 0, 0x19, 0x7F, adc_pcm_tlv), SOC_DOUBLE_R("ADC Mixer Switch", CS42L51_ADCA_VOL, CS42L51_ADCB_VOL, 7, 1, 1), + SOC_DOUBLE_R_SX_TLV("ADC Attenuator Volume", + CS42L51_ADCA_ATT, CS42L51_ADCB_ATT, + 0, 0xA0, 96, adc_att_tlv), + SOC_DOUBLE_R_SX_TLV("PGA Volume", + CS42L51_ALC_PGA_CTL, CS42L51_ALC_PGB_CTL, + 0, 0x1A, 30, pga_tlv), SOC_SINGLE("Playback Deemphasis Switch", CS42L51_DAC_CTL, 3, 1, 0), SOC_SINGLE("Auto-Mute Switch", CS42L51_DAC_CTL, 2, 1, 0), SOC_SINGLE("Soft Ramp Switch", CS42L51_DAC_CTL, 1, 1, 0), -- 2.26.0

3 2

[PATCH 1/2] dt-bindings: tas2562: Add TAS2564 to binding
by Dan Murphy 21 Sep '20

21 Sep '20

Add the TAS2564 compatible and data sheet link to the binding. Signed-off-by: Dan Murphy <dmurphy(a)ti.com> --- Documentation/devicetree/bindings/sound/tas2562.yaml | 2 ++ 1 file changed, 2 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/tas2562.yaml b/Documentation/devicetree/bindings/sound/tas2562.yaml index c3b7e19a0d44..d3ceda356aaa 100644 --- a/Documentation/devicetree/bindings/sound/tas2562.yaml +++ b/Documentation/devicetree/bindings/sound/tas2562.yaml @@ -19,12 +19,14 @@ description: | Specifications about the audio amplifier can be found at: https://www.ti.com/lit/gpn/tas2562 https://www.ti.com/lit/gpn/tas2563 + https://www.ti.com/lit/gpn/tas2564 properties: compatible: enum: - ti,tas2562 - ti,tas2563 + - ti,tas2564 reg: maxItems: 1 -- 2.28.0

2 2

[PATCH] ASoC: SOF: topology: fix the process being scheduled on core0 always
by Kai Vehmanen 21 Sep '20

21 Sep '20

From: Keyon Jie <yang.jie(a)linux.intel.com> In commit 783898ce68de ("ASoC: SOF: append extended data to sof_ipc_comp_process") the process components are set to run on the fixed core 0, this break us from scheduling components on any other DSP core. Since we can get the DSP core index from swidget->core, it is duplicated to pass the extra 'core' argument for those sof_widget_load_xx() functions. Here removes the duplicate 'core' argument and get component core from swidget->core directly to fix the issue mentioned above. Fixes: 783898ce68de ("ASoC: SOF: append extended data to sof_ipc_comp_process") Signed-off-by: Keyon Jie <yang.jie(a)linux.intel.com> Reviewed-by: Jaska Uimonen <jaska.uimonen(a)intel.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> --- sound/soc/sof/topology.c | 82 +++++++++++++++++----------------------- 1 file changed, 35 insertions(+), 47 deletions(-) diff --git a/sound/soc/sof/topology.c b/sound/soc/sof/topology.c index eaa1122d5a68..69313fbdb636 100644 --- a/sound/soc/sof/topology.c +++ b/sound/soc/sof/topology.c @@ -1465,13 +1465,11 @@ static int sof_connect_dai_widget(struct snd_soc_component *scomp, * @ipc_size: IPC payload size that will be updated depending on valid * extended data. * @index: ID of the pipeline the component belongs to - * @core: index of the DSP core that the component should run on * * Return: The pointer to the new allocated component, NULL if failed. */ static struct sof_ipc_comp *sof_comp_alloc(struct snd_sof_widget *swidget, - size_t *ipc_size, int index, - int core) + size_t *ipc_size, int index) { u8 nil_uuid[SOF_UUID_SIZE] = {0}; struct sof_ipc_comp *comp; @@ -1490,7 +1488,7 @@ static struct sof_ipc_comp *sof_comp_alloc(struct snd_sof_widget *swidget, comp->hdr.cmd = SOF_IPC_GLB_TPLG_MSG | SOF_IPC_TPLG_COMP_NEW; comp->id = swidget->comp_id; comp->pipeline_id = index; - comp->core = core; + comp->core = swidget->core; /* handle the extended data if needed */ if (total_size > *ipc_size) { @@ -1505,7 +1503,7 @@ static struct sof_ipc_comp *sof_comp_alloc(struct snd_sof_widget *swidget, } static int sof_widget_load_dai(struct snd_soc_component *scomp, int index, - struct snd_sof_widget *swidget, int core, + struct snd_sof_widget *swidget, struct snd_soc_tplg_dapm_widget *tw, struct sof_ipc_comp_reply *r, struct snd_sof_dai *dai) @@ -1517,7 +1515,7 @@ static int sof_widget_load_dai(struct snd_soc_component *scomp, int index, int ret; comp_dai = (struct sof_ipc_comp_dai *) - sof_comp_alloc(swidget, &ipc_size, index, core); + sof_comp_alloc(swidget, &ipc_size, index); if (!comp_dai) return -ENOMEM; @@ -1571,7 +1569,7 @@ static int sof_widget_load_dai(struct snd_soc_component *scomp, int index, */ static int sof_widget_load_buffer(struct snd_soc_component *scomp, int index, - struct snd_sof_widget *swidget, int core, + struct snd_sof_widget *swidget, struct snd_soc_tplg_dapm_widget *tw, struct sof_ipc_comp_reply *r) { @@ -1590,7 +1588,7 @@ static int sof_widget_load_buffer(struct snd_soc_component *scomp, int index, buffer->comp.id = swidget->comp_id; buffer->comp.type = SOF_COMP_BUFFER; buffer->comp.pipeline_id = index; - buffer->comp.core = core; + buffer->comp.core = swidget->core; ret = sof_parse_tokens(scomp, buffer, buffer_tokens, ARRAY_SIZE(buffer_tokens), private->array, @@ -1642,7 +1640,7 @@ static int spcm_bind(struct snd_soc_component *scomp, struct snd_sof_pcm *spcm, */ static int sof_widget_load_pcm(struct snd_soc_component *scomp, int index, - struct snd_sof_widget *swidget, int core, + struct snd_sof_widget *swidget, enum sof_ipc_stream_direction dir, struct snd_soc_tplg_dapm_widget *tw, struct sof_ipc_comp_reply *r) @@ -1654,7 +1652,7 @@ static int sof_widget_load_pcm(struct snd_soc_component *scomp, int index, int ret; host = (struct sof_ipc_comp_host *) - sof_comp_alloc(swidget, &ipc_size, index, core); + sof_comp_alloc(swidget, &ipc_size, index); if (!host) return -ENOMEM; @@ -1779,7 +1777,7 @@ static int sof_widget_load_pipeline(struct snd_soc_component *scomp, int index, */ static int sof_widget_load_mixer(struct snd_soc_component *scomp, int index, - struct snd_sof_widget *swidget, int core, + struct snd_sof_widget *swidget, struct snd_soc_tplg_dapm_widget *tw, struct sof_ipc_comp_reply *r) { @@ -1790,7 +1788,7 @@ static int sof_widget_load_mixer(struct snd_soc_component *scomp, int index, int ret; mixer = (struct sof_ipc_comp_mixer *) - sof_comp_alloc(swidget, &ipc_size, index, core); + sof_comp_alloc(swidget, &ipc_size, index); if (!mixer) return -ENOMEM; @@ -1824,7 +1822,7 @@ static int sof_widget_load_mixer(struct snd_soc_component *scomp, int index, * Mux topology */ static int sof_widget_load_mux(struct snd_soc_component *scomp, int index, - struct snd_sof_widget *swidget, int core, + struct snd_sof_widget *swidget, struct snd_soc_tplg_dapm_widget *tw, struct sof_ipc_comp_reply *r) { @@ -1835,7 +1833,7 @@ static int sof_widget_load_mux(struct snd_soc_component *scomp, int index, int ret; mux = (struct sof_ipc_comp_mux *) - sof_comp_alloc(swidget, &ipc_size, index, core); + sof_comp_alloc(swidget, &ipc_size, index); if (!mux) return -ENOMEM; @@ -1870,7 +1868,7 @@ static int sof_widget_load_mux(struct snd_soc_component *scomp, int index, */ static int sof_widget_load_pga(struct snd_soc_component *scomp, int index, - struct snd_sof_widget *swidget, int core, + struct snd_sof_widget *swidget, struct snd_soc_tplg_dapm_widget *tw, struct sof_ipc_comp_reply *r) { @@ -1884,7 +1882,7 @@ static int sof_widget_load_pga(struct snd_soc_component *scomp, int index, int ret; volume = (struct sof_ipc_comp_volume *) - sof_comp_alloc(swidget, &ipc_size, index, core); + sof_comp_alloc(swidget, &ipc_size, index); if (!volume) return -ENOMEM; @@ -1946,7 +1944,7 @@ static int sof_widget_load_pga(struct snd_soc_component *scomp, int index, */ static int sof_widget_load_src(struct snd_soc_component *scomp, int index, - struct snd_sof_widget *swidget, int core, + struct snd_sof_widget *swidget, struct snd_soc_tplg_dapm_widget *tw, struct sof_ipc_comp_reply *r) { @@ -1957,7 +1955,7 @@ static int sof_widget_load_src(struct snd_soc_component *scomp, int index, int ret; src = (struct sof_ipc_comp_src *) - sof_comp_alloc(swidget, &ipc_size, index, core); + sof_comp_alloc(swidget, &ipc_size, index); if (!src) return -ENOMEM; @@ -2003,7 +2001,7 @@ static int sof_widget_load_src(struct snd_soc_component *scomp, int index, */ static int sof_widget_load_asrc(struct snd_soc_component *scomp, int index, - struct snd_sof_widget *swidget, int core, + struct snd_sof_widget *swidget, struct snd_soc_tplg_dapm_widget *tw, struct sof_ipc_comp_reply *r) { @@ -2014,7 +2012,7 @@ static int sof_widget_load_asrc(struct snd_soc_component *scomp, int index, int ret; asrc = (struct sof_ipc_comp_asrc *) - sof_comp_alloc(swidget, &ipc_size, index, core); + sof_comp_alloc(swidget, &ipc_size, index); if (!asrc) return -ENOMEM; @@ -2062,7 +2060,7 @@ static int sof_widget_load_asrc(struct snd_soc_component *scomp, int index, */ static int sof_widget_load_siggen(struct snd_soc_component *scomp, int index, - struct snd_sof_widget *swidget, int core, + struct snd_sof_widget *swidget, struct snd_soc_tplg_dapm_widget *tw, struct sof_ipc_comp_reply *r) { @@ -2073,7 +2071,7 @@ static int sof_widget_load_siggen(struct snd_soc_component *scomp, int index, int ret; tone = (struct sof_ipc_comp_tone *) - sof_comp_alloc(swidget, &ipc_size, index, core); + sof_comp_alloc(swidget, &ipc_size, index); if (!tone) return -ENOMEM; @@ -2229,7 +2227,7 @@ static int sof_process_load(struct snd_soc_component *scomp, int index, } process = (struct sof_ipc_comp_process *) - sof_comp_alloc(swidget, &ipc_size, index, 0); + sof_comp_alloc(swidget, &ipc_size, index); if (!process) { ret = -ENOMEM; goto out; @@ -2307,7 +2305,7 @@ static int sof_process_load(struct snd_soc_component *scomp, int index, */ static int sof_widget_load_process(struct snd_soc_component *scomp, int index, - struct snd_sof_widget *swidget, int core, + struct snd_sof_widget *swidget, struct snd_soc_tplg_dapm_widget *tw, struct sof_ipc_comp_reply *r) { @@ -2322,7 +2320,7 @@ static int sof_widget_load_process(struct snd_soc_component *scomp, int index, } memset(&config, 0, sizeof(config)); - config.comp.core = core; + config.comp.core = swidget->core; /* get the process token */ ret = sof_parse_tokens(scomp, &config, process_tokens, @@ -2450,8 +2448,7 @@ static int sof_widget_ready(struct snd_soc_component *scomp, int index, return -ENOMEM; } - ret = sof_widget_load_dai(scomp, index, swidget, comp.core, - tw, &reply, dai); + ret = sof_widget_load_dai(scomp, index, swidget, tw, &reply, dai); if (ret == 0) { sof_connect_dai_widget(scomp, w, tw, dai); list_add(&dai->list, &sdev->dai_list); @@ -2461,12 +2458,10 @@ static int sof_widget_ready(struct snd_soc_component *scomp, int index, } break; case snd_soc_dapm_mixer: - ret = sof_widget_load_mixer(scomp, index, swidget, comp.core, - tw, &reply); + ret = sof_widget_load_mixer(scomp, index, swidget, tw, &reply); break; case snd_soc_dapm_pga: - ret = sof_widget_load_pga(scomp, index, swidget, comp.core, - tw, &reply); + ret = sof_widget_load_pga(scomp, index, swidget, tw, &reply); /* Find scontrol for this pga and set readback offset*/ list_for_each_entry(scontrol, &sdev->kcontrol_list, list) { if (scontrol->comp_id == swidget->comp_id) { @@ -2476,41 +2471,34 @@ static int sof_widget_ready(struct snd_soc_component *scomp, int index, } break; case snd_soc_dapm_buffer: - ret = sof_widget_load_buffer(scomp, index, swidget, comp.core, - tw, &reply); + ret = sof_widget_load_buffer(scomp, index, swidget, tw, &reply); break; case snd_soc_dapm_scheduler: - ret = sof_widget_load_pipeline(scomp, index, swidget, - tw, &reply); + ret = sof_widget_load_pipeline(scomp, index, swidget, tw, &reply); break; case snd_soc_dapm_aif_out: - ret = sof_widget_load_pcm(scomp, index, swidget, comp.core, + ret = sof_widget_load_pcm(scomp, index, swidget, SOF_IPC_STREAM_CAPTURE, tw, &reply); break; case snd_soc_dapm_aif_in: - ret = sof_widget_load_pcm(scomp, index, swidget, comp.core, + ret = sof_widget_load_pcm(scomp, index, swidget, SOF_IPC_STREAM_PLAYBACK, tw, &reply); break; case snd_soc_dapm_src: - ret = sof_widget_load_src(scomp, index, swidget, comp.core, - tw, &reply); + ret = sof_widget_load_src(scomp, index, swidget, tw, &reply); break; case snd_soc_dapm_asrc: - ret = sof_widget_load_asrc(scomp, index, swidget, comp.core, - tw, &reply); + ret = sof_widget_load_asrc(scomp, index, swidget, tw, &reply); break; case snd_soc_dapm_siggen: - ret = sof_widget_load_siggen(scomp, index, swidget, comp.core, - tw, &reply); + ret = sof_widget_load_siggen(scomp, index, swidget, tw, &reply); break; case snd_soc_dapm_effect: - ret = sof_widget_load_process(scomp, index, swidget, comp.core, - tw, &reply); + ret = sof_widget_load_process(scomp, index, swidget, tw, &reply); break; case snd_soc_dapm_mux: case snd_soc_dapm_demux: - ret = sof_widget_load_mux(scomp, index, swidget, comp.core, - tw, &reply); + ret = sof_widget_load_mux(scomp, index, swidget, tw, &reply); break; case snd_soc_dapm_switch: case snd_soc_dapm_dai_link: -- 2.27.0

2 1

[PATCH] ALSA: usb-audio: Add mixer support for Pioneer DJ DJM-250MK2
by František Kučera 21 Sep '20

21 Sep '20

From: František Kučera <franta-linux(a)frantovo.cz> This patch extends support for DJM-250MK2 and allows mapping playback and capture channels to available sources. Configures the card through USB commands. --- sound/usb/mixer_quirks.c | 194 +++++++++++++++++++++++++++++++++++++++ 1 file changed, 194 insertions(+) diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index 199cdbfdc761..f2aeb5ac3611 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -2602,6 +2602,197 @@ static int snd_bbfpro_controls_create(struct usb_mixer_interface *mixer) return 0; } +/* + * Pioneer DJ DJM-250MK2 and maybe other DJM models + * + * For playback, no duplicate mapping should be set. + * There are three mixer stereo channels (CH1, CH2, AUX) and three stereo sources (PLAYBACK 1-2, PLAYBACK 3-4, PLAYBACK 5-6). + * Each channel should be mapped just once to one source. + * If mapped multiple times, only one source will play on given channel (sources are not mixed together). + * + * For recording, duplicate mapping is OK. We will get the same signal multiple times. + * + * Channels 7-8 are in both directions fixed to FX SEND / FX RETURN. + * + * See also notes in the quirks-table.h file. + */ + +typedef struct { + const u16 wIndex; + const u16 wValue; + const char *name; +} snd_pioneer_djm_option; + +static const snd_pioneer_djm_option SND_PIONEER_DJM_OPTIONS_CAPTURE_LEVEL[] = { + { .name = "-5 dB", .wValue = 0x0300, .wIndex = 0x8003 }, + { .name = "-10 dB", .wValue = 0x0200, .wIndex = 0x8003 }, + { .name = "-15 dB", .wValue = 0x0100, .wIndex = 0x8003 }, + { .name = "-19 dB", .wValue = 0x0000, .wIndex = 0x8003 } +}; + +static const snd_pioneer_djm_option SND_PIONEER_DJM_OPTIONS_CAPTURE_CH12[] = { + { .name = "CH1 Control Tone PHONO", .wValue = 0x0103, .wIndex = 0x8002 }, + { .name = "CH1 Control Tone LINE", .wValue = 0x0100, .wIndex = 0x8002 }, + { .name = "Post CH1 Fader", .wValue = 0x0106, .wIndex = 0x8002 }, + { .name = "Cross Fader A", .wValue = 0x0107, .wIndex = 0x8002 }, + { .name = "Cross Fader B", .wValue = 0x0108, .wIndex = 0x8002 }, + { .name = "MIC", .wValue = 0x0109, .wIndex = 0x8002 }, + { .name = "AUX", .wValue = 0x010d, .wIndex = 0x8002 }, + { .name = "REC OUT", .wValue = 0x010a, .wIndex = 0x8002 } +}; + +static const snd_pioneer_djm_option SND_PIONEER_DJM_OPTIONS_CAPTURE_CH34[] = { + { .name = "CH2 Control Tone PHONO", .wValue = 0x0203, .wIndex = 0x8002 }, + { .name = "CH2 Control Tone LINE", .wValue = 0x0200, .wIndex = 0x8002 }, + { .name = "Post CH2 Fader", .wValue = 0x0206, .wIndex = 0x8002 }, + { .name = "Cross Fader A", .wValue = 0x0207, .wIndex = 0x8002 }, + { .name = "Cross Fader B", .wValue = 0x0208, .wIndex = 0x8002 }, + { .name = "MIC", .wValue = 0x0209, .wIndex = 0x8002 }, + { .name = "AUX", .wValue = 0x020d, .wIndex = 0x8002 }, + { .name = "REC OUT", .wValue = 0x020a, .wIndex = 0x8002 } +}; + +static const snd_pioneer_djm_option SND_PIONEER_DJM_OPTIONS_CAPTURE_CH56[] = { + { .name = "REC OUT", .wValue = 0x030a, .wIndex = 0x8002 }, + { .name = "Post CH1 Fader", .wValue = 0x0311, .wIndex = 0x8002 }, + { .name = "Post CH2 Fader", .wValue = 0x0312, .wIndex = 0x8002 }, + { .name = "Cross Fader A", .wValue = 0x0307, .wIndex = 0x8002 }, + { .name = "Cross Fader B", .wValue = 0x0308, .wIndex = 0x8002 }, + { .name = "MIC", .wValue = 0x0309, .wIndex = 0x8002 }, + { .name = "AUX", .wValue = 0x030d, .wIndex = 0x8002 } +}; + +static const snd_pioneer_djm_option SND_PIONEER_DJM_OPTIONS_PLAYBACK_12[] = { + { .name = "CH1", .wValue = 0x0100, .wIndex = 0x8016 }, + { .name = "CH2", .wValue = 0x0101, .wIndex = 0x8016 }, + { .name = "AUX", .wValue = 0x0104, .wIndex = 0x8016 } +}; + +static const snd_pioneer_djm_option SND_PIONEER_DJM_OPTIONS_PLAYBACK_34[] = { + { .name = "CH1", .wValue = 0x0200, .wIndex = 0x8016 }, + { .name = "CH2", .wValue = 0x0201, .wIndex = 0x8016 }, + { .name = "AUX", .wValue = 0x0204, .wIndex = 0x8016 } +}; + +static const snd_pioneer_djm_option SND_PIONEER_DJM_OPTIONS_PLAYBACK_56[] = { + { .name = "CH1", .wValue = 0x0300, .wIndex = 0x8016 }, + { .name = "CH2", .wValue = 0x0301, .wIndex = 0x8016 }, + { .name = "AUX", .wValue = 0x0304, .wIndex = 0x8016 } +}; + +typedef struct { + const char *name; + const snd_pioneer_djm_option *options; + const size_t count; + const u16 default_value; +} snd_pioneer_djm_option_group; + +#define SND_PIONEER_DJM_OPTION_GROUP_ITEM(_name, suffix, _default_value) \ + { .name = _name, .options = SND_PIONEER_DJM_OPTIONS_##suffix, .count = ARRAY_SIZE(SND_PIONEER_DJM_OPTIONS_##suffix), .default_value = _default_value } + +static const snd_pioneer_djm_option_group SND_PIONEER_DJM_OPTION_GROUPS[] = { + SND_PIONEER_DJM_OPTION_GROUP_ITEM("Master Capture Level Capture Switch", CAPTURE_LEVEL, 0), + SND_PIONEER_DJM_OPTION_GROUP_ITEM("Capture 1-2 Capture Switch", CAPTURE_CH12, 2), + SND_PIONEER_DJM_OPTION_GROUP_ITEM("Capture 3-4 Capture Switch", CAPTURE_CH34, 2), + SND_PIONEER_DJM_OPTION_GROUP_ITEM("Capture 5-6 Capture Switch", CAPTURE_CH56, 0), + SND_PIONEER_DJM_OPTION_GROUP_ITEM("Playback 1-2 Playback Switch", PLAYBACK_12, 0), + SND_PIONEER_DJM_OPTION_GROUP_ITEM("Playback 3-4 Playback Switch", PLAYBACK_34, 1), + SND_PIONEER_DJM_OPTION_GROUP_ITEM("Playback 5-6 Playback Switch", PLAYBACK_56, 2) +}; + +// layout of the kcontrol->private_value: +#define SND_PIONEER_DJM_VALUE_MASK 0x0000ffff +#define SND_PIONEER_DJM_GROUP_MASK 0xffff0000 +#define SND_PIONEER_DJM_GROUP_SHIFT 16 + +static int snd_pioneer_djm_controls_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *info) +{ + u16 group = kcontrol->private_value >> SND_PIONEER_DJM_GROUP_SHIFT; + size_t count; + const char *name; + if (group < ARRAY_SIZE(SND_PIONEER_DJM_OPTION_GROUPS)) { + count = SND_PIONEER_DJM_OPTION_GROUPS[group].count; + if (info->value.enumerated.item >= count) info->value.enumerated.item = count - 1; + name = SND_PIONEER_DJM_OPTION_GROUPS[group].options[info->value.enumerated.item].name; + strlcpy(info->value.enumerated.name, name, sizeof (info->value.enumerated.name)); + info->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + info->count = 1; + info->value.enumerated.items = count; + return 0; + } else { + return -EINVAL; + } +} + +static int snd_pioneer_djm_controls_update(struct usb_mixer_interface *mixer, u16 group, u16 value) +{ + int err; + if (group < ARRAY_SIZE(SND_PIONEER_DJM_OPTION_GROUPS) && value < SND_PIONEER_DJM_OPTION_GROUPS[group].count) { + err = snd_usb_lock_shutdown(mixer->chip); + if (err) return err; + + err = snd_usb_ctl_msg( + mixer->chip->dev, usb_sndctrlpipe(mixer->chip->dev, 0), + USB_REQ_SET_FEATURE, + USB_DIR_OUT | USB_TYPE_VENDOR | USB_RECIP_DEVICE, + SND_PIONEER_DJM_OPTION_GROUPS[group].options[value].wValue, + SND_PIONEER_DJM_OPTION_GROUPS[group].options[value].wIndex, + NULL, 0); + + snd_usb_unlock_shutdown(mixer->chip); + return err; + } else { + return -EINVAL; + } +} + +static int snd_pioneer_djm_controls_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.enumerated.item[0] = kcontrol->private_value & SND_PIONEER_DJM_VALUE_MASK; + return 0; +} + +static int snd_pioneer_djm_controls_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_list *list = snd_kcontrol_chip(kcontrol); + struct usb_mixer_interface *mixer = list->mixer; + + u16 group = (kcontrol->private_value & SND_PIONEER_DJM_GROUP_MASK) >> SND_PIONEER_DJM_GROUP_SHIFT; + u16 value = ucontrol->value.enumerated.item[0]; + kcontrol->private_value = group << SND_PIONEER_DJM_GROUP_SHIFT | value; + + return snd_pioneer_djm_controls_update(mixer, group, value); +} + +static int snd_pioneer_djm_controls_resume(struct usb_mixer_elem_list *list) +{ + u16 group = (list->kctl->private_value & SND_PIONEER_DJM_GROUP_MASK) >> SND_PIONEER_DJM_GROUP_SHIFT; + u16 value = (list->kctl->private_value & SND_PIONEER_DJM_VALUE_MASK); + return snd_pioneer_djm_controls_update(list->mixer, group, value); +} + +static int snd_pioneer_djm_controls_create(struct usb_mixer_interface *mixer) +{ + int err, group; + for (group = 0; group < ARRAY_SIZE(SND_PIONEER_DJM_OPTION_GROUPS); group++ ) { + struct snd_kcontrol_new knew = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .index = 0, + .info = snd_pioneer_djm_controls_info, + .get = snd_pioneer_djm_controls_get, + .put = snd_pioneer_djm_controls_put, + .name = SND_PIONEER_DJM_OPTION_GROUPS[group].name, + .private_value = group << SND_PIONEER_DJM_GROUP_SHIFT | SND_PIONEER_DJM_OPTION_GROUPS[group].default_value + }; + err = snd_pioneer_djm_controls_update(mixer, group, SND_PIONEER_DJM_OPTION_GROUPS[group].default_value); + if (err) return err; + err = add_single_ctl_with_resume(mixer, 0, snd_pioneer_djm_controls_resume, &knew, NULL); + if (err) return err; + } + return 0; +} + int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) { int err = 0; @@ -2706,6 +2897,9 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) case USB_ID(0x2a39, 0x3fb0): /* RME Babyface Pro FS */ err = snd_bbfpro_controls_create(mixer); break; + case USB_ID(0x2b73, 0x0017): /* Pioneer DJ DJM-250MK2 */ + err = snd_pioneer_djm_controls_create(mixer); + break; } return err; -- 2.20.1

2 1

[PATCH 0/4] ALSA: hda - add Intel DG1 support
by Kai Vehmanen 21 Sep '20

21 Sep '20

Series adding support for HDMI/DP audio for Intel DG1. Kai Vehmanen (4): ALSA: hda - add Intel DG1 PCI and HDMI ids ALSA: hda - controller is in GPU on the DG1 ALSA: hda - handle multiple i915 device instances ALSA: hda - fix CONTROLLER_IN_GPU macro name sound/hda/hdac_i915.c | 48 ++++++++++++++++++++++++++++++++++---- sound/pci/hda/hda_intel.c | 6 ++++- sound/pci/hda/patch_hdmi.c | 1 + 3 files changed, 50 insertions(+), 5 deletions(-) -- 2.27.0

2 5

[PATCH AUTOSEL 4.19 1/9] ASoC: wm8994: Skip setting of the WM8994_MICBIAS register for WM1811
by Sasha Levin 21 Sep '20

21 Sep '20

From: Sylwester Nawrocki <s.nawrocki(a)samsung.com> [ Upstream commit 811c5494436789e7149487c06e0602b507ce274b ] 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> Acked-by: Krzysztof Kozlowski <krzk(a)kernel.org> Acked-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20200827173357.31891-1-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/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 01acb8da2f48e..cd089b4143029 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -4051,11 +4051,13 @@ 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 fed6ea9b019f7..da7fa6f5459e6 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -1227,6 +1227,9 @@ 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 ee339ad8514d1..1433d73e09bf8 100644 --- a/sound/soc/codecs/wm_hubs.h +++ b/sound/soc/codecs/wm_hubs.h @@ -31,6 +31,7 @@ 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; -- 2.25.1

1 2

[PATCH AUTOSEL 5.4 05/15] ASoC: Intel: bytcr_rt5640: Add quirk for MPMAN Converter9 2-in-1
by Sasha Levin 21 Sep '20

21 Sep '20

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit 6a0137101f47301fff2da6ba4b9048383d569909 ] 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> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20200901080623.4987-1-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/bytcr_rt5640.c | 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 f7964d1ec486f..6012367f6fe48 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 = { -- 2.25.1

1 0

[PATCH AUTOSEL 5.4 04/15] ASoC: wm8994: Ensure the device is resumed in wm89xx_mic_detect functions
by Sasha Levin 21 Sep '20

21 Sep '20

From: Sylwester Nawrocki <s.nawrocki(a)samsung.com> [ Upstream commit f5a2cda4f1db89776b64c4f0f2c2ac609527ac70 ] When the wm8958_mic_detect, wm8994_mic_detect functions get called from the machine driver, e.g. from the card's late_probe() callback, the CODEC device may be PM runtime suspended and any regmap writes have no effect. Add PM runtime calls to these functions to ensure the device registers are updated as expected. This suppresses an error during boot "wm8994-codec: ASoC: error at snd_soc_component_update_bits on wm8994-codec" caused by the regmap access error due to the cache_only flag being set. Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> Acked-by: Krzysztof Kozlowski <krzk(a)kernel.org> Acked-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20200827173357.31891-2-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/wm8994.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 64635f9cdae65..6dbab3fc6537e 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3372,6 +3372,8 @@ int wm8994_mic_detect(struct snd_soc_component *component, struct snd_soc_jack * return -EINVAL; } + pm_runtime_get_sync(component->dev); + switch (micbias) { case 1: micdet = &wm8994->micdet[0]; @@ -3419,6 +3421,8 @@ int wm8994_mic_detect(struct snd_soc_component *component, struct snd_soc_jack * snd_soc_dapm_sync(dapm); + pm_runtime_put(component->dev); + return 0; } EXPORT_SYMBOL_GPL(wm8994_mic_detect); @@ -3786,6 +3790,8 @@ int wm8958_mic_detect(struct snd_soc_component *component, struct snd_soc_jack * return -EINVAL; } + pm_runtime_get_sync(component->dev); + if (jack) { snd_soc_dapm_force_enable_pin(dapm, "CLK_SYS"); snd_soc_dapm_sync(dapm); @@ -3854,6 +3860,8 @@ int wm8958_mic_detect(struct snd_soc_component *component, struct snd_soc_jack * snd_soc_dapm_sync(dapm); } + pm_runtime_put(component->dev); + return 0; } EXPORT_SYMBOL_GPL(wm8958_mic_detect); -- 2.25.1

1 0

[PATCH AUTOSEL 5.4 03/15] ASoC: wm8994: Skip setting of the WM8994_MICBIAS register for WM1811
by Sasha Levin 21 Sep '20

21 Sep '20

From: Sylwester Nawrocki <s.nawrocki(a)samsung.com> [ Upstream commit 811c5494436789e7149487c06e0602b507ce274b ] 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> Acked-by: Krzysztof Kozlowski <krzk(a)kernel.org> Acked-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20200827173357.31891-1-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/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 d5fb7f5dd551c..64635f9cdae65 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -4047,11 +4047,13 @@ 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 e93af7edd8f75..dd421e2fe7b21 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 4b8e5f0d6e32d..988b29e630607 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; -- 2.25.1

1 0

[PATCH AUTOSEL 5.4 02/15] ASoC: pcm3168a: ignore 0 Hz settings
by Sasha Levin 21 Sep '20

21 Sep '20

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> [ Upstream commit 7ad26d6671db758c959d7e1d100b138a38483612 ] 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> Link: https://lore.kernel.org/r/87a6yjy5sy.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/pcm3168a.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/codecs/pcm3168a.c b/sound/soc/codecs/pcm3168a.c index 88b75695fbf7f..b37e5fbbd301a 100644 --- a/sound/soc/codecs/pcm3168a.c +++ b/sound/soc/codecs/pcm3168a.c @@ -302,6 +302,13 @@ 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

1 0

[PATCH AUTOSEL 5.8 05/20] ASoC: Intel: bytcr_rt5640: Add quirk for MPMAN Converter9 2-in-1
by Sasha Levin 21 Sep '20

21 Sep '20

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit 6a0137101f47301fff2da6ba4b9048383d569909 ] 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> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20200901080623.4987-1-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/bytcr_rt5640.c | 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 1fdb70b9e4788..5f885062145fe 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 = { -- 2.25.1

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