June 2022 - Alsa-devel - mailman.alsa-project.org

Regression playing 24bit/96kHz audio on USB audio interface between 5.10.94 and 5.15.21
by Ruud van Asseldonk 02 Jan '23

02 Jan '23

Hi, I have a program that configures a Behringer UMC404HD 192k USB audio interface for playback as follows (pseudocode): snd_pcm_hw_params_set_channels(pcm, hwp, 2); snd_pcm_hw_params_set_format(pcm, hwp, SND_PCM_FORMAT_S24_3LE); snd_pcm_hw_params_set_access(pcm, hwp, SND_PCM_ACCESS_MMAP_INTERLEAVED); snd_pcm_hw_params_set_rate(pcm, hwp, val=96000, dir=0); snd_pcm_hw_params_set_period_size_near(pcm, hwp, val=513, dir=0); // If I confirm at this point with snd_pcm_hw_params_get_period_size, // I get back 513. snd_pcm_hw_params_set_buffer_size_near(pcm, hwp, val=2052); snd_pcm_hw_params(pcm, hwp); // If I confirm with snd_pcm_hw_params_get_buffer_size, // I get back 2052. On Linux 5.10.94 (specifically Arch Linux ARM package linux-rpi-5.10.94-1), this succeeds, and I can play back audio just fine. On 5.15.21 (specifically linux-rpi-5.15.21-3), all the calls succeed apart from the last one, which returns EINVAL. (My actual program is written in Rust and uses a Rust wrapper library for libasound, https://docs.rs/alsa/0.6.0/alsa/, but the calls should be the ones listed.) I confirmed with snd_pcm_hw_params_test_format that the S24_3LE format is supported. On both kernels it is successful. I confirmed with snd_pcm_hw_params_test_rate that 96 kHz is supported. On both kernels it is successful. There are differences though. When I query these: snd_pcm_hw_params_get_period_size_min snd_pcm_hw_params_get_period_size_max snd_pcm_hw_params_get_buffer_size_min snd_pcm_hw_params_get_buffer_size_max Then on 5.10.94 (the good version), I get: period size min/max: 12/32768, buffer size min/max: 24/65536 But on 5.15.21 (the bad version), I get: period size min/max: 13/71332, buffer size min/max: 26/142663 Also, after I set the period size to 513, if I then query the buffer size min/max, on 5.10.94 I get: buffer size min/max: 1026/65536 But on 5.15.21 I get: buffer size min/max: 1027/142663 In lsmod, snd_usb_audio shows up, and from lsusb -v, the UMC404HD appears to be a standard USB audio interface. The lsusb output is identical on both kernels. Please let me know if there is anything I can do to help further diagnose the issue, or if I should report this elsewhere. Kind regards, Ruud van Asseldonk

5 10

[alsa-devel] [PATCH 0/6] ASoC: mediatek: mt8183: support DP audio
by Tzung-Bi Shih 24 Nov '22

24 Nov '22

(This series depends on https://lore.kernel.org/patchwork/patch/1126819/ which has not applied to the maintainer's tree yet.) This series makes mt8183-mt6358-ts3a227-max98357 support DP (display port) audio output. ASoC: hdmi-codec: add PCM trigger operator - add a PCM trigger callback if codec driver is interested in PCM events. drm: bridge: it6505: bridge to hdmi-codec - add audio feature support of it6505. - bridge to hdmi-codec so that ASoC can use the audio feature. - defer the "enabling" after receiving PCM START or RESUME event. ASoC: dt-bindings: mt8183: add a property "mediatek,hdmi-codec" ASoC: mediatek: mt8183: use hdmi-codec - bind to hdmi-codec in mt8183-mt6358-ts3a227-max98357. drm: bridge: it6505: report connector status ASoC: mediatek: mt8183: support HDMI jack reporting - use the mechanism from the following patch https://mailman.alsa-project.org/pipermail/alsa-devel/2019-September/155302… to report jack status (from DRM to ASoC). Allen Chen (1): drm: bridge: it6505: bridge to hdmi-codec Tzung-Bi Shih (5): ASoC: hdmi-codec: add PCM trigger operator ASoC: dt-bindings: mt8183: add a property "mediatek,hdmi-codec" ASoC: mediatek: mt8183: use hdmi-codec drm: bridge: it6505: report connector status ASoC: mediatek: mt8183: support HDMI jack reporting .../sound/mt8183-mt6358-ts3a227-max98357.txt | 2 + drivers/gpu/drm/bridge/ite-it6505.c | 191 +++++++++++++++++- include/sound/hdmi-codec.h | 13 ++ sound/soc/codecs/hdmi-codec.c | 31 +++ sound/soc/mediatek/Kconfig | 3 +- .../mt8183/mt8183-mt6358-ts3a227-max98357.c | 36 +++- 6 files changed, 266 insertions(+), 10 deletions(-) -- 2.23.0.866.gb869b98d4c-goog

3 9

[PATCH v6 00/10] Add support for CS35L41 in HDA systems
by Lucas Tanure 31 Oct '22

31 Oct '22

Add support for laptops that have CS35L41 connected to an HDA codec by I2S and direct I2C connection to the CPU. Laptops that use CS35L41 and are SPI will be added in the future, after the support for it is resolved at i2c-multi-instantiate driver. i2c-multi-instantiate thread: https://lkml.org/lkml/2021/12/10/557 Hardware: - Some laptops have two CS35L41 amplifiers, connected to Realtek ALC287 by an I2S bus and by and direct I2C to the CPU. - The ALC287 codec is connected to the CPU by an HDA bus. - The CS35L41 has a DSP which will require firmware to be loaded. Architecture: - Creation of a library of shared functions for CS35L41 ASoC and HDA - HDA codec driver (RealTek) and CS35L41 HDA driver are combined by using component binding, where it uses device names to find the components and bind to the master - The HDA CS35L41 driver applies pre-defined registers sequences for each action in playback for HDA Sound card Changes from V5: - Fix build issues reported by Intel Test Bot - Check devm_kasprintf return - Add ACPI dependency to HDA drivers V5: https://lkml.org/lkml/2021/12/16/430 Changes from V4: - Save index received from ACPI for latter use in DSP Support - Move CS35L41_DSP1_CCM_CORE_CTRL to regmap_write so doesn't affect DSP hibernation feature V4: https://lkml.org/lkml/2021/12/14/487 Changes from V3: - SPI bus driver starter added - Use separate modules approach instead of liking library - Add CSC3551 ACPI _HID for more I2C laptops - Removed regulators from HDA driver - Add note about Non-conforming _HID V3: https://lkml.org/lkml/2021/11/23/723 Changes from V2: - Not an RFC - Create a new HDA driver for CS35L41 instead of using the ASoC one - Use component binding and device names to find the CS35L41 driver - Create a shared library for ASoC and HDA CS35L41 drivers v2: https://lkml.org/lkml/2021/10/8/344 Lucas Tanure (9): ASoC: cs35l41: Convert tables to shared source code ASoC: cs35l41: Move cs35l41_otp_unpack to shared code ASoC: cs35l41: Move power initializations to reg_sequence ASoC: cs35l41: Create shared function for errata patches ASoC: cs35l41: Create shared function for setting channels ASoC: cs35l41: Create shared function for boost configuration hda: cs35l41: Add support for CS35L41 in HDA systems ACPI / scan: Create platform device for CLSA0100 and CSC3551 ACPI nodes ALSA: hda/realtek: Add support for Legion 7 16ACHg6 laptop Stefan Binding (1): ALSA: hda/realtek: Add CS35L41 support for Thinkpad laptops MAINTAINERS | 2 + drivers/acpi/scan.c | 3 + drivers/platform/x86/i2c-multi-instantiate.c | 11 + include/sound/cs35l41.h | 739 ++++++++++++++++++ sound/pci/hda/Kconfig | 29 + sound/pci/hda/Makefile | 10 + sound/pci/hda/cs35l41_hda.c | 527 +++++++++++++ sound/pci/hda/cs35l41_hda.h | 69 ++ sound/pci/hda/cs35l41_hda_i2c.c | 66 ++ sound/pci/hda/cs35l41_hda_spi.c | 63 ++ sound/pci/hda/hda_component.h | 20 + sound/pci/hda/patch_realtek.c | 146 ++++ sound/soc/codecs/Kconfig | 11 +- sound/soc/codecs/Makefile | 4 +- sound/soc/codecs/cs35l41-i2c.c | 1 - .../{cs35l41-tables.c => cs35l41-lib.c} | 355 ++++++++- sound/soc/codecs/cs35l41-spi.c | 1 - sound/soc/codecs/cs35l41.c | 360 +-------- sound/soc/codecs/cs35l41.h | 734 ----------------- 19 files changed, 2055 insertions(+), 1096 deletions(-) create mode 100644 sound/pci/hda/cs35l41_hda.c create mode 100644 sound/pci/hda/cs35l41_hda.h create mode 100644 sound/pci/hda/cs35l41_hda_i2c.c create mode 100644 sound/pci/hda/cs35l41_hda_spi.c create mode 100644 sound/pci/hda/hda_component.h rename sound/soc/codecs/{cs35l41-tables.c => cs35l41-lib.c} (71%) -- 2.34.1

11 28

[PATCH 1/2] ALSA: hda: Add Intel DG2 PCI ID and HDMI codec vid
by Kai Vehmanen 14 Oct '22

14 Oct '22

Add HD Audio PCI ID and HDMI codec vendor ID for Intel DG2. Reviewed-by: Uma Shankar <uma.shankar(a)intel.com> Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> --- sound/pci/hda/hda_intel.c | 12 +++++++++++- sound/pci/hda/patch_hdmi.c | 1 + 2 files changed, 12 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 221afacbc7fd..b98d7975adea 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -335,7 +335,10 @@ enum { ((pci)->device == 0x0c0c) || \ ((pci)->device == 0x0d0c) || \ ((pci)->device == 0x160c) || \ - ((pci)->device == 0x490d)) + ((pci)->device == 0x490d) || \ + ((pci)->device == 0x4f90) || \ + ((pci)->device == 0x4f91) || \ + ((pci)->device == 0x4f92)) #define IS_BXT(pci) ((pci)->vendor == 0x8086 && (pci)->device == 0x5a98) @@ -2477,6 +2480,13 @@ static const struct pci_device_id azx_ids[] = { /* DG1 */ { PCI_DEVICE(0x8086, 0x490d), .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, + /* DG2 */ + { PCI_DEVICE(0x8086, 0x4f90), + .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, + { PCI_DEVICE(0x8086, 0x4f91), + .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, + { PCI_DEVICE(0x8086, 0x4f92), + .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, /* Alderlake-S */ { PCI_DEVICE(0x8086, 0x7ad0), .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 65d2c5539919..98633d2684de 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -4382,6 +4382,7 @@ HDA_CODEC_ENTRY(0x80862814, "DG1 HDMI", patch_i915_tgl_hdmi), HDA_CODEC_ENTRY(0x80862815, "Alderlake HDMI", patch_i915_tgl_hdmi), HDA_CODEC_ENTRY(0x8086281c, "Alderlake-P HDMI", patch_i915_tgl_hdmi), HDA_CODEC_ENTRY(0x80862816, "Rocketlake HDMI", patch_i915_tgl_hdmi), +HDA_CODEC_ENTRY(0x80862819, "DG2 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), base-commit: d7aca8d8f548570f650d9b1dfedf6902f56f2bce -- 2.34.1

3 3

[PATCH v6 0/4] Let userspace know when snd-hda-intel needs i915
by Mauro Carvalho Chehab 28 Sep '22

28 Sep '22

Currently, kernel/module annotates module dependencies when request_symbol is used, but it doesn't cover more complex inter-driver dependencies that are subsystem and/or driver-specific. That's because module_try_get() and symbol_get() doesn't try to setup the module owner. In the case of hdmi sound, depending on the CPU/GPU, sometimes the snd_hda_driver can talk directly with the hardware, but sometimes, it uses the i915 driver. When the snd_hda_driver uses i915, it should first be unbind/rmmod, as otherwise trying to unbind/rmmod the i915 driver cause driver issues, as as reported by CI tools with different GPU models: https://intel-gfx-ci.01.org/tree/drm-tip/IGT_6415/fi-tgl-1115g4/igt@core_ho… https://intel-gfx-ci.01.org/tree/drm-tip/CI_DRM_11495/bat-adlm-1/igt@i915_m… In the past, just a few CPUs were doing such bindings, but this issue now applies to all "modern" Intel CPUs that have onboard graphics, as well as to the newer discrete GPUs. With the discrete GPU case, the HDA controller is physically separate and requires i915 to power on the hardware for all hardware access. In this case, the issue is hit basicly 100% of the time. With on-board graphics, i915 driver is needed only when the display codec is accessed. If i915 is unbind during runtime suspend, while snd-hda-intel is still bound, nothing bad happens, but unbinding i915 on other situations may also cause issues. So, add support at kernel/modules to properly set the holders when try_module_get() and symbol_get() are used. This allow allow audio drivers to properly annotate when a dependency on a DRM driver dependencies exists, and add a call to such new function at the snd-hda driver when it successfully binds into the DRM driver. With that, userspace tools can now check and properly remove the audio driver before trying to remove or unbind the GPU driver. It should be noticed that this series conveys the hidden module dependencies. Other changes are needed in order to allow removing or unbinding the i915 driver while keeping the snd-hda-intel driver loaded/bound. With that regards, there are some discussions on how to improve this at alsa-devel a while back: https://mailman.alsa-project.org/pipermail/alsa-devel/2021-September/190099… So, future improvements on both in i915 and the audio drivers could be made. E.g. with discrete GPUs, it's the only codec of the card, so it seems feasible to detach the ALSA card if i915 is bound (using infra made for VGA switcheroo), but, until these improvements are done and land in upstream, audio drivers needs to be unbound if i915 driver goes unbind. Yet, even if such fixes got merged, this series is still needed, as it makes such dependencies more explicit and easier to debug. PS.: This series was generated against next-20220506. --- v6: - dropped an unused function prototype for __symbol_get_gpl(); - addressed several issues that were noticed while testing the series on an slow atom machine; - also add holders when symbol_get() is used. v5: - while v4 works fine, it ends calling try_module_format() recursively, which is not what it it was supposed to do. So, change the logic to avoid such recursion, by adding a static __try_module_format() and renaming the new version that takes two arguments as try_module_format_owner(). v4: - fix a compilation warning reported by Intel's Kernel robot when !CONFIG_MODULE_UNLOAD or !CONFIG_MODULE. v3: minor fixes: - fixed a checkpatch warning; - use a single line for the new function prototype. v2: - the dependencies are now handled directly at try_module_get(). Mauro Carvalho Chehab (4): module: drop prototype for non-existing __symbol_get_gpl() module: update dependencies at try_module_get() module: set holders when symbol_get() is used ALSA: hda - identify when audio is provided by a video driver drivers/mtd/chips/gen_probe.c | 4 +- include/linux/module.h | 13 +++-- kernel/module/main.c | 76 ++++++++++++++++++++----- samples/hw_breakpoint/data_breakpoint.c | 2 +- sound/hda/hdac_component.c | 2 +- 5 files changed, 72 insertions(+), 25 deletions(-) -- 2.35.3

5 9

[PATCH v2 0/3] ASoC: dt-bindings: Add bindings for WCD9335 DAIs
by Yassine Oudjana 15 Sep '22

15 Sep '22

Add DT bindings for WCD9335 DAIs and use them in the driver as well as all device trees currently using WCD9335. Changes since v1: - Make header guard match path - Maintain the alphabetical order in msm8996-xiaomi-gemini includes Yassine Oudjana (3): ASoC: dt-bindings: Add bindings for WCD9335 DAIs ASoC: wcd9335: Use DT bindings instead of local DAI definitions arm64: dts: qcom: Use WCD9335 DT bindings MAINTAINERS | 1 + arch/arm64/boot/dts/qcom/apq8096-db820c.dts | 5 +++-- .../arm64/boot/dts/qcom/msm8996-xiaomi-gemini.dts | 5 +++-- .../boot/dts/qcom/msm8996-xiaomi-scorpio.dts | 5 +++-- include/dt-bindings/sound/qcom,wcd9335.h | 15 +++++++++++++++ sound/soc/codecs/wcd9335.c | 13 ++----------- 6 files changed, 27 insertions(+), 17 deletions(-) create mode 100644 include/dt-bindings/sound/qcom,wcd9335.h -- 2.36.1

4 6

[PATCH] ALSA: usb-audio: Refcount multiple accesses on the single clock
by Takashi Iwai 05 Sep '22

05 Sep '22

When a clock source is connected to multiple nodes / endpoints, the current USB-audio driver tries to set up at each time one of them is configured. Although it reads the current rate and updates only if it differs, some devices seem unhappy with this behavior and spew the errors when reading/updating the rate unnecessarily. This patch tries to reduce the redundant clock setup by introducing a refcount for each clock source. When the stream is actually running, a clock rate is "locked", and it bypasses the clock and/or refuse to change any longer. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=215934 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/card.c | 1 + sound/usb/card.h | 3 +- sound/usb/endpoint.c | 90 +++++++++++++++++++++++++++++++++++++++----- sound/usb/usbaudio.h | 1 + 4 files changed, 85 insertions(+), 10 deletions(-) diff --git a/sound/usb/card.c b/sound/usb/card.c index 376962291c4d..0fff96a5d3ab 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -635,6 +635,7 @@ static int snd_usb_audio_create(struct usb_interface *intf, INIT_LIST_HEAD(&chip->pcm_list); INIT_LIST_HEAD(&chip->ep_list); INIT_LIST_HEAD(&chip->iface_ref_list); + INIT_LIST_HEAD(&chip->clock_ref_list); INIT_LIST_HEAD(&chip->midi_list); INIT_LIST_HEAD(&chip->mixer_list); diff --git a/sound/usb/card.h b/sound/usb/card.h index 87f042d06ce0..ca75f2206170 100644 --- a/sound/usb/card.h +++ b/sound/usb/card.h @@ -44,6 +44,7 @@ struct audioformat { struct snd_usb_substream; struct snd_usb_iface_ref; +struct snd_usb_clock_ref; struct snd_usb_endpoint; struct snd_usb_power_domain; @@ -62,6 +63,7 @@ struct snd_urb_ctx { struct snd_usb_endpoint { struct snd_usb_audio *chip; struct snd_usb_iface_ref *iface_ref; + struct snd_usb_clock_ref *clock_ref; int opened; /* open refcount; protect with chip->mutex */ atomic_t running; /* running status */ @@ -138,7 +140,6 @@ struct snd_usb_endpoint { unsigned int cur_period_frames; unsigned int cur_period_bytes; unsigned int cur_buffer_periods; - unsigned char cur_clock; spinlock_t lock; struct list_head list; diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index 743b8287cfcd..df5a70013a85 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -35,6 +35,14 @@ struct snd_usb_iface_ref { struct list_head list; }; +/* clock refcounting */ +struct snd_usb_clock_ref { + unsigned char clock; + atomic_t locked; + int rate; + struct list_head list; +}; + /* * snd_usb_endpoint is a model that abstracts everything related to an * USB endpoint and its streaming. @@ -591,6 +599,25 @@ iface_ref_find(struct snd_usb_audio *chip, int iface) return ip; } +/* Similarly, a refcount object for clock */ +static struct snd_usb_clock_ref * +clock_ref_find(struct snd_usb_audio *chip, int clock) +{ + struct snd_usb_clock_ref *ref; + + list_for_each_entry(ref, &chip->clock_ref_list, list) + if (ref->clock == clock) + return ref; + + ref = kzalloc(sizeof(*ref), GFP_KERNEL); + if (!ref) + return NULL; + ref->clock = clock; + atomic_set(&ref->locked, 0); + list_add_tail(&ref->list, &chip->clock_ref_list); + return ref; +} + /* * Get the existing endpoint object corresponding EP * Returns NULL if not present. @@ -768,6 +795,14 @@ snd_usb_endpoint_open(struct snd_usb_audio *chip, goto unlock; } + if (fp->protocol != UAC_VERSION_1) { + ep->clock_ref = clock_ref_find(chip, fp->clock); + if (!ep->clock_ref) { + ep = NULL; + goto unlock; + } + } + ep->cur_audiofmt = fp; ep->cur_channels = fp->channels; ep->cur_rate = params_rate(params); @@ -777,7 +812,6 @@ snd_usb_endpoint_open(struct snd_usb_audio *chip, ep->cur_period_frames = params_period_size(params); ep->cur_period_bytes = ep->cur_period_frames * ep->cur_frame_bytes; ep->cur_buffer_periods = params_periods(params); - ep->cur_clock = fp->clock; if (ep->type == SND_USB_ENDPOINT_TYPE_SYNC) endpoint_set_syncinterval(chip, ep); @@ -894,8 +928,8 @@ void snd_usb_endpoint_close(struct snd_usb_audio *chip, ep->altsetting = 0; ep->cur_audiofmt = NULL; ep->cur_rate = 0; - ep->cur_clock = 0; ep->iface_ref = NULL; + ep->clock_ref = NULL; usb_audio_dbg(chip, "EP 0x%x closed\n", ep->ep_num); } mutex_unlock(&chip->mutex); @@ -907,6 +941,8 @@ void snd_usb_endpoint_suspend(struct snd_usb_endpoint *ep) ep->need_setup = true; if (ep->iface_ref) ep->iface_ref->need_setup = true; + if (ep->clock_ref) + ep->clock_ref->rate = 0; } /* @@ -1314,6 +1350,33 @@ static int snd_usb_endpoint_set_params(struct snd_usb_audio *chip, return 0; } +static int init_sample_rate(struct snd_usb_audio *chip, + struct snd_usb_endpoint *ep) +{ + struct snd_usb_clock_ref *clock = ep->clock_ref; + int err; + + if (clock) { + if (atomic_read(&clock->locked)) + return 0; + if (clock->rate == ep->cur_rate) + return 0; + if (clock->rate && clock->rate != ep->cur_rate) { + usb_audio_dbg(chip, "Mismatched sample rate %d vs %d for EP 0x%x\n", + clock->rate, ep->cur_rate, ep->ep_num); + return -EINVAL; + } + } + + err = snd_usb_init_sample_rate(chip, ep->cur_audiofmt, ep->cur_rate); + if (err < 0) + return err; + + if (clock) + clock->rate = ep->cur_rate; + return 0; +} + /* * snd_usb_endpoint_configure: Configure the endpoint * @@ -1343,8 +1406,7 @@ int snd_usb_endpoint_configure(struct snd_usb_audio *chip, * to update at each EP configuration */ if (ep->cur_audiofmt->protocol == UAC_VERSION_1) { - err = snd_usb_init_sample_rate(chip, ep->cur_audiofmt, - ep->cur_rate); + err = init_sample_rate(chip, ep); if (err < 0) goto unlock; } @@ -1374,7 +1436,7 @@ int snd_usb_endpoint_configure(struct snd_usb_audio *chip, if (err < 0) goto unlock; - err = snd_usb_init_sample_rate(chip, ep->cur_audiofmt, ep->cur_rate); + err = init_sample_rate(chip, ep); if (err < 0) goto unlock; @@ -1407,15 +1469,15 @@ int snd_usb_endpoint_configure(struct snd_usb_audio *chip, /* get the current rate set to the given clock by any endpoint */ int snd_usb_endpoint_get_clock_rate(struct snd_usb_audio *chip, int clock) { - struct snd_usb_endpoint *ep; + struct snd_usb_clock_ref *ref; int rate = 0; if (!clock) return 0; mutex_lock(&chip->mutex); - list_for_each_entry(ep, &chip->ep_list, list) { - if (ep->cur_clock == clock && ep->cur_rate) { - rate = ep->cur_rate; + list_for_each_entry(ref, &chip->clock_ref_list, list) { + if (ref->clock == clock) { + rate = ref->rate; break; } } @@ -1456,6 +1518,9 @@ int snd_usb_endpoint_start(struct snd_usb_endpoint *ep) if (atomic_inc_return(&ep->running) != 1) return 0; + if (ep->clock_ref) + atomic_inc(&ep->clock_ref->locked); + ep->active_mask = 0; ep->unlink_mask = 0; ep->phase = 0; @@ -1565,6 +1630,9 @@ void snd_usb_endpoint_stop(struct snd_usb_endpoint *ep, bool keep_pending) if (ep->sync_source) WRITE_ONCE(ep->sync_source->sync_sink, NULL); stop_urbs(ep, false, keep_pending); + if (ep->clock_ref) + if (!atomic_dec_return(&ep->clock_ref->locked)) + ep->clock_ref->rate = 0; } } @@ -1591,12 +1659,16 @@ void snd_usb_endpoint_free_all(struct snd_usb_audio *chip) { struct snd_usb_endpoint *ep, *en; struct snd_usb_iface_ref *ip, *in; + struct snd_usb_clock_ref *cp, *cn; list_for_each_entry_safe(ep, en, &chip->ep_list, list) kfree(ep); list_for_each_entry_safe(ip, in, &chip->iface_ref_list, list) kfree(ip); + + list_for_each_entry_safe(cp, cn, &chip->clock_ref_list, list) + kfree(ip); } /* diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index 044cd7ab27cb..ffbb4b0d09a0 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -45,6 +45,7 @@ struct snd_usb_audio { struct list_head pcm_list; /* list of pcm streams */ struct list_head ep_list; /* list of audio-related endpoints */ struct list_head iface_ref_list; /* list of interface refcounts */ + struct list_head clock_ref_list; /* list of clock refcounts */ int pcm_devs; struct list_head midi_list; /* list of midi interfaces */ -- 2.35.3

2 2

[PATCH] sound: soc/codecs: tlv320adc3xxx: needs GPIOLIB
by Randy Dunlap 31 Aug '22

31 Aug '22

Fix build errors when CONFIG_GPIOLIB is not enabled: ../sound/soc/codecs/tlv320adc3xxx.c: In function 'adc3xxx_i2c_probe': ../sound/soc/codecs/tlv320adc3xxx.c:1352:28: error: implicit declaration of function 'devm_gpiod_get'; did you mean 'devm_gpio_free'? [-Werror=implicit-function-declaration] 1352 | adc3xxx->rst_pin = devm_gpiod_get(dev, "reset", GPIOD_OUT_LOW); ../sound/soc/codecs/tlv320adc3xxx.c:1352:57: error: 'GPIOD_OUT_LOW' undeclared (first use in this function); did you mean 'GPIOF_INIT_LOW'? 1352 | adc3xxx->rst_pin = devm_gpiod_get(dev, "reset", GPIOD_OUT_LOW); CC lib/dynamic_debug.o ../sound/soc/codecs/tlv320adc3xxx.c:1400:9: error: implicit declaration of function 'gpiod_set_value_cansleep'; did you mean 'gpio_set_value_cansleep'? [-Werror=implicit-function-declaration] 1400 | gpiod_set_value_cansleep(adc3xxx->rst_pin, 1); Fixes: e9a3b57efd28 ("ASoC: codec: tlv320adc3xxx: New codec driver") Signed-off-by: Randy Dunlap <rdunlap(a)infradead.org> Reported-by: kernel test robot <lkp(a)intel.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Ricard Wanderlof <ricardw(a)axis.com> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: alsa-devel(a)alsa-project.org --- sound/soc/codecs/Kconfig | 1 + 1 file changed, 1 insertion(+) --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -1567,6 +1567,7 @@ config SND_SOC_TFA989X config SND_SOC_TLV320ADC3XXX tristate "Texas Instruments TLV320ADC3001/3101 audio ADC" depends on I2C + depends on GPIOLIB help Enable support for Texas Instruments TLV320ADC3001 and TLV320ADC3101 ADCs.

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Request for setup of new repositories
by Takashi Sakamoto 27 Aug '22

27 Aug '22

Hi Jaroslav, Iwai-san, Thanks for your maintenance for alsa-project organization in github.com. Currently I'd like to add new three repositories under the organization as a part of my work for ALSA firewire stack. I've been maintaining libhinawa since 2014 and recently realized that current design is not necessarily convenient since it includes two functions; operation to Linux FireWire cdev, and operation of ALSA HwDep cdev. Currently I'm working for new library to split the latter operation. Then I'd like you to setup below repositories: * 'libhitaki' * 'libhitaki-doc' * 'hitaki-rs' The library itself and its Rust binding are utilized by 'snd-firewire-ctl-services'[2], thus it's preferable to register them under 'GObject Introspection' Team. Thanks for your assist for my work. [1] https://github.com/alsa-project/libhinawa [2] https://github.com/alsa-project/snd-firewire-ctl-services Regards Takashi Sakamoto

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[PATCH] ASoC: ops: Don't modify the driver's plaform_max when reading state
by Mark Brown 22 Aug '22

22 Aug '22

Currently snd_soc_info_volsw() will set a platform_max based on the limit the control has if one is not already set. This isn't really great, we shouldn't be modifying the passed in driver data especially in a path like this which may not ever be executed or where we may execute other callbacks before this one. Instead make this function leave the data unchanged, and clarify things a bit by referring to max rather than platform_max within the function. platform_max is now applied as a limit after working out the natural maximum value for the control. This means that platform_max is no longer treated as a direct register value for controls were min is non-zero. The put() callbacks already validate on this basis, and there do not appear to be any in tree users that would be affected. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/soc-ops.c | 13 +++++++------ 1 file changed, 7 insertions(+), 6 deletions(-) diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c index 2d39370ddeca..93e72a016b4d 100644 --- a/sound/soc/soc-ops.c +++ b/sound/soc/soc-ops.c @@ -176,20 +176,21 @@ int snd_soc_info_volsw(struct snd_kcontrol *kcontrol, { struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; - int platform_max; + int max; - if (!mc->platform_max) - mc->platform_max = mc->max; - platform_max = mc->platform_max; + max = uinfo->value.integer.max = mc->max - mc->min; + if (mc->platform_max && mc->platform_max < max) + max = mc->platform_max; - if (platform_max == 1 && !strstr(kcontrol->id.name, " Volume")) + if (max == 1 && !strstr(kcontrol->id.name, " Volume")) uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; else uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; uinfo->count = snd_soc_volsw_is_stereo(mc) ? 2 : 1; uinfo->value.integer.min = 0; - uinfo->value.integer.max = platform_max - mc->min; + uinfo->value.integer.max = max; + return 0; } EXPORT_SYMBOL_GPL(snd_soc_info_volsw); -- 2.30.2

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