December 2023 - Alsa-devel - mailman.alsa-project.org

[PATCHv3 RESEND 00/10] ASoC: fsl-asoc-card: compatibility integration of a generic codec use case for use with S/PDIF controller
by Elinor Montmasson 16 Jan '24

16 Jan '24

Hello, This is the v3 of the series of patch aiming to make the machine driver "fsl-asoc-card" compatible with use cases where there is no real codec driver. It proposes to use the "spdif_receiver" and "spdif_transmitter" drivers instead of the dummy codec. This is a first step in using the S/PDIF controller with the ASRC. The five first patches add compatibility with the pair of codecs "spdif_receiver" and "spdif_transmitter" with a new compatible, "fsl,imx-audio-generic". Codec parameters are set with default values. Consequently, the driver is modified to work with multi-codec use cases. It can get 2 codecs phandles from the device tree, and the "fsl_asoc_card_priv" struct now has 2 "codec_priv" to store properties of both codecs. It is fixed to 2 codecs as only "fsl,imx-audio-generic" uses multiple codecs at the moment. However, the driver now uses for_each_codecs macros when possible to ease future implementations of multi-codec configurations. The remaining patches add configuration options for the device tree. They configure the CPU DAI when using "fsl,imx-audio-generic". These options are usually hard-coded in "fsl-asoc-card.c" for each audio codec. Because the generic codec could be used with other CPU DAIs than the S/PDIF controller, setting these parameters could be required. This series of patch was successfully built for arm64 and x86 on top of the latest for-next branch of the ASoC git tree on the 14th of December. These modifications have also been tested on an i.MX8MN evaluation board, with a linux kernel RT v6.1.26-rt8. We also have a few questions, following remarks made by Krzysztof Kozlowski in a previous email for patch 10/10: > >>> The compatible list for this generic sound card currently: > >>> @@ -48,6 +51,8 @@ The compatible list for this generic sound card > >>> currently: > >>> > >>> "fsl,imx-audio-nau8822" > >>> > >>> + "fsl,imx-audio-generic" > >> > >> Generic does not look like hardware specific. > > > > Even if our end goal is to use it with the S/PDIF controller, this new > > support can be used with different hardware that doesn't > > require a codec. Thus, we don't really want to specify "spdif" in it. > > > > Is this compatible string not suitable ? > > Should we rename it to something else, like "fsl,imx-audio-no-codec" ? > > Maybe Mark or Rob will help here, but for me "imx-audio" is just way too > generic. * Which generic name should we use ? Or how should we change it? > Also, you add several new properties, so I really expect either > converting old binding to DT schema first or adding new device in DT > schema format. * fsl-asoc-card.txt currently follows the old dt-bindings format. Should we update it to DT schema format in this patch series before adding my new properties? Best regards, Elinor Montmasson Changelog: v2 -> v3: * when the bitmaster or framemaster are retrieved from the device tree, the driver will now compare them with the two codecs possibly given in the device tree, and not just the first codec. * improve driver modifications to use multiple codecs for better integration of future multi-codec use cases: * use for_each_codec macros when possible. * "fsl_asoc_card_priv" struct now has 2 "codec_priv" as the driver can currently retrieve 2 codec phandles from the device tree. * fix subject of patch 10/10 to follow the style of the subsystem and previous commits of the file. * v2 patch series at: https://lore.kernel.org/alsa-devel/20231027144734.3654829-1-elinor.montmass… v1 -> v2: * replace use of the dummy codec by the pair of codecs "spdif_receiver" / " spdif_transmitter". * adapt how dai links codecs are used to take into account the possibility for multiple codecs per link. * change compatible name. * adapt driver to be able to register two codecs given in the device tree. * v1 patch series at: https://lore.kernel.org/alsa-devel/20230901144550.520072-1-elinor.montmasso… Elinor Montmasson (10): ASoC: fsl-asoc-card: add support for dai links with multiple codecs ASoC: fsl-asoc-card: add second dai link component for codecs ASoC: fsl-asoc-card: add compatibility to use 2 codecs in dai-links ASoC: fsl-asoc-card: add new compatible for a generic codec use case ASoC: fsl-asoc-card: set generic codec as clock provider ASoC: fsl-asoc-card: add dts property "cpu-slot-width" ASoC: fsl-asoc-card: add dts property "cpu-slot-num" ASoC: fsl-asoc-card: add dts properties "cpu-sysclk-freq" ASoC: fsl-asoc-card: add dts properties "cpu-sysclk-dir-out" ASoC: bindings: fsl-asoc-card: add compatible for generic codec .../bindings/sound/fsl-asoc-card.txt | 28 +- sound/soc/fsl/fsl-asoc-card.c | 299 +++++++++++------- 2 files changed, 218 insertions(+), 109 deletions(-) -- 2.25.1

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[RESEND PATCH 0/1] ALSA: virtio: add support for audio controls
by Aiswarya Cyriac 15 Jan '24

15 Jan '24

From: Anton Yakovlev <anton.yakovlev(a)opensynergy.com> Implementation of support for audio controls in accordance with the extension of the virtio sound device specification[1] planned for virtio-v1.3-cs01. Design　of virtual audio controls is based on and derived from ALSA audio controls. It allows the driver to perform all standard operations, such as reading and writing audio control value, as well as working with metadata (represented in the TLV form). The driver part was tested on top of the Linux 5.10 kernel. As a device part was used OpenSynergy proprietary implementation. [1] https://lists.oasis-open.org/archives/virtio-comment/202104/msg00013.html Anton Yakovlev (1): ALSA: virtio: add support for audio controls include/uapi/linux/virtio_snd.h | 154 +++++++++++ sound/virtio/Makefile | 1 + sound/virtio/virtio_card.c | 21 ++ sound/virtio/virtio_card.h | 22 ++ sound/virtio/virtio_kctl.c | 464 ++++++++++++++++++++++++++++++++ 5 files changed, 662 insertions(+) create mode 100644 sound/virtio/virtio_kctl.c -- 2.25.1

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[PATCH 00/12] soundwire/SOF: add SoundWire Interface support for AMD SOF stack
by Vijendar Mukunda 09 Jan '24

09 Jan '24

This patch series is to redesign existing platform device creation logic for SoundWire managers and Implement generic functions for SoundWire manager probe, start and exit sequence which are common for both Legacy(NO DSP enabled) and SOF stack, and add SoundWire Interface support for AMD SOF stack (ACP 6.3 based platforms). Vijendar Mukunda (12): ASoC/soundwire: implement generic api for scanning amd soundwire controller drivers: soundwire: refactor amd soundwire manager device node creation drivers: soundwire: implement function to extract slave information drivers: soundwire: refactor soundwire pads enable drivers: soundwire: refactor register mask structure ASoC: SOF: amd: add code for invoking soundwire manager helper functions ASoC: SOF: amd: add interrupt handling for SoundWire manager devices ASoC: SOF: amd: Add Soundwire DAI configuration support for AMD platforms ASoC: SOF: amd: add machine select logic for soundwire based platforms ASoC: SOF: amd: update descriptor fields for acp6.3 based platform ASoC: SOF: amd: select soundwire dependency flag for acp6.3 based platform ASoC: SOF: amd: refactor acp driver pm ops drivers/soundwire/Makefile | 2 +- drivers/soundwire/amd_init.c | 235 +++++++++++++++++++++++++++++ drivers/soundwire/amd_init.h | 13 ++ drivers/soundwire/amd_manager.c | 41 +---- drivers/soundwire/amd_manager.h | 12 +- include/linux/soundwire/sdw_amd.h | 79 ++++++++-- include/sound/sof/dai-amd.h | 7 + include/sound/sof/dai.h | 2 + include/uapi/sound/sof/tokens.h | 4 + sound/soc/amd/acp/Kconfig | 7 + sound/soc/amd/acp/Makefile | 2 + sound/soc/amd/acp/amd-sdw-acpi.c | 62 ++++++++ sound/soc/sof/amd/Kconfig | 18 +++ sound/soc/sof/amd/acp-common.c | 65 +++++++- sound/soc/sof/amd/acp-dsp-offset.h | 10 ++ sound/soc/sof/amd/acp.c | 202 ++++++++++++++++++++++++- sound/soc/sof/amd/acp.h | 26 +++- sound/soc/sof/amd/pci-acp63.c | 7 + sound/soc/sof/ipc3-pcm.c | 25 +++ sound/soc/sof/ipc3-topology.c | 40 +++++ sound/soc/sof/sof-audio.h | 1 + sound/soc/sof/topology.c | 5 + 22 files changed, 798 insertions(+), 67 deletions(-) create mode 100644 drivers/soundwire/amd_init.c create mode 100644 drivers/soundwire/amd_init.h create mode 100644 sound/soc/amd/acp/amd-sdw-acpi.c -- 2.34.1

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sof-audio-pci-intel-tgl/soundwire 6.6.8 kernel outputs no sound on speakers but works on headphones
by Marc MERLIN 02 Jan '24

02 Jan '24

Howdy, First, apologies if I'm not sending this to the right place, I could not find an approrpriate soundwire/sof mailing list, please redirect me as needed (I had to find an old soundwire message for reference) laptop; Dell XPS 17 9730 merlin:~# lspci | grep -i audio 0000:00:1f.3 Multimedia audio controller: Intel Corporation Device 51ca (rev 01) stock mainline 6.6.8 shows the device, but no sound comes out, as if it were muted, but I do not see anyting muted in alsamixer. If I plug in headphones, sound works, if I unplug them, no sound on speakers. Nothing is shown as muted alsamixer and the sound levels look ok. I ran the test as root from console so that pulseaudio/pipewire doesn't get in the way. Under xorg pavucontrol detects when I plug in headphones, and says speaker unvailable, when I switch to headphones output it works, when I unplug the headphones, speaker unavailable goes back to speaker. So I kill X and pulseaudio, play audio on the console directly to alsa instead of pulse and same thing, it plays but no sound on the speakers, but sound on headphones when I plug them in Alsamixer is using default:0 sof-soundwire Any idea what could be muting the speakers if alsamixer shows them unmuted? It is a kernel bug or userspace bug? merlin:~# cat /proc/asound/* cat: /proc/asound/card0: Is a directory 0 [sofsoundwire ]: sof-soundwire - sof-soundwire Intel Soundwire SOF 1: : sequencer 2: [ 0- 0]: digital audio playback 3: [ 0- 1]: digital audio capture 4: [ 0- 2]: digital audio playback 5: [ 0- 4]: digital audio capture 6: [ 0- 5]: digital audio playback 7: [ 0- 6]: digital audio playback 8: [ 0- 7]: digital audio playback 9: [ 0- 2]: hardware dependent 10: [ 0] : control 33: : timer 00-02: HDA Codec 2 0 snd_soc_sof_sdw cat: /proc/asound/oss: Is a directory 00-00: Jack Out (*) : : playback 1 00-01: Jack In (*) : : capture 1 00-02: Speaker (*) : : playback 1 00-04: Microphone (*) : : capture 1 00-05: HDMI 1 (*) : : playback 1 00-06: HDMI 2 (*) : : playback 1 00-07: HDMI 3 (*) : : playback 1 cat: /proc/asound/seq: Is a directory cat: /proc/asound/sofsoundwire: Is a directory G0: system timer : 4000.000us (10000000 ticks) G3: HR timer : 0.001us (1000000000 ticks) Client sequencer queue 1 : stopped P0-0-0: PCM playback 0-0-0 : 21333.333us (1 ticks) SLAVE P0-1-1: PCM capture 0-1-1 : SLAVE P0-2-0: PCM playback 0-2-0 : SLAVE P0-4-1: PCM capture 0-4-1 : SLAVE P0-5-0: PCM playback 0-5-0 : SLAVE P0-6-0: PCM playback 0-6-0 : SLAVE P0-7-0: PCM playback 0-7-0 : SLAVE Advanced Linux Sound Architecture Driver Version k6.6.8-amd64-volpre-sysrq-20231218. merlin:~# dmesg | grep -E '(snd|sof|soundcore)' [ 0.090196] software IO TLB: area num 32. [ 1.567086] PCI-DMA: Using software bounce buffering for IO (SWIOTLB) [ 1.567087] software IO TLB: mapped [mem 0x0000000053683000-0x0000000057683000] (64MB) [ 15.941644] snd_hda_intel 0000:00:1f.3: vgaarb: pci_notify [ 15.941656] snd_hda_intel 0000:00:1f.3: runtime IRQ mapping not provided by arch [ 15.941662] snd_hda_intel 0000:00:1f.3: DSP detected with PCI class/subclass/prog-if info 0x040100 [ 15.942126] snd_hda_intel 0000:00:1f.3: SoundWire enabled on CannonLake+ platform, using SOF driver [ 15.942151] snd_hda_intel 0000:00:1f.3: vgaarb: pci_notify [ 16.089302] sof-audio-pci-intel-tgl 0000:00:1f.3: vgaarb: pci_notify [ 16.089311] sof-audio-pci-intel-tgl 0000:00:1f.3: runtime IRQ mapping not provided by arch [ 16.089317] sof-audio-pci-intel-tgl 0000:00:1f.3: power state changed by ACPI to D0 [ 16.089320] sof-audio-pci-intel-tgl 0000:00:1f.3: ACPI _REG connect evaluation failed (5) [ 16.089417] sof-audio-pci-intel-tgl 0000:00:1f.3: DSP detected with PCI class/subclass/prog-if info 0x040100 [ 16.089503] sof-audio-pci-intel-tgl 0000:00:1f.3: SoundWire enabled on CannonLake+ platform, using SOF driver [ 16.089542] sof-audio-pci-intel-tgl 0000:00:1f.3: enabling device (0000 -> 0002) [ 16.089937] sof-audio-pci-intel-tgl 0000:00:1f.3: vgaarb: pci_notify [ 16.090083] sof-audio-pci-intel-tgl 0000:00:1f.3: DSP detected with PCI class/subclass/prog-if 0x040100 [ 16.090473] sof-audio-pci-intel-tgl 0000:00:1f.3: bound 0000:00:02.0 (ops i915_audio_component_bind_ops [i915]) [ 16.102000] sof-audio-pci-intel-tgl 0000:00:1f.3: use msi interrupt mode [ 16.102152] sof-audio-pci-intel-tgl 0000:00:1f.3: enabling bus mastering [ 16.120035] sof-audio-pci-intel-tgl 0000:00:1f.3: hda codecs found, mask 4 [ 16.125472] sof-audio-pci-intel-tgl 0000:00:1f.3: Firmware info: version 2:2:0-57864 [ 16.125499] sof-audio-pci-intel-tgl 0000:00:1f.3: Firmware: ABI 3:22:1 Kernel ABI 3:23:0 [ 16.125522] sof-audio-pci-intel-tgl 0000:00:1f.3: unknown sof_ext_man header type 3 size 0x30 [ 16.218672] sof-audio-pci-intel-tgl 0000:00:1f.3: Firmware info: version 2:2:0-57864 [ 16.218704] sof-audio-pci-intel-tgl 0000:00:1f.3: Firmware: ABI 3:22:1 Kernel ABI 3:23:0 [ 16.235176] sof-audio-pci-intel-tgl 0000:00:1f.3: Topology: ABI 3:22:1 Kernel ABI 3:23:0 [ 16.235457] sof_sdw sof_sdw: ASoC: Parent card not yet available, widget card binding deferred [ 16.285146] sof_sdw sof_sdw: hda_dsp_hdmi_build_controls: no PCM in topology for HDMI converter 3 [ 16.304414] input: sof-soundwire Headset Jack as /devices/pci0000:00/0000:00:1f.3/sof_sdw/sound/card0/input12 [ 16.307681] input: sof-soundwire HDMI/DP,pcm=5 as /devices/pci0000:00/0000:00:1f.3/sof_sdw/sound/card0/input13 [ 16.314904] input: sof-soundwire HDMI/DP,pcm=6 as /devices/pci0000:00/0000:00:1f.3/sof_sdw/sound/card0/input14 [ 16.321892] input: sof-soundwire HDMI/DP,pcm=7 as /devices/pci0000:00/0000:00:1f.3/sof_sdw/sound/card0/input15 merlin:~# lsmod | grep -E "(^snd|sof|soundwire)" snd_seq_dummy 12288 0 snd_hrtimer 12288 1 snd_ctl_led 24576 0 snd_soc_sof_sdw 86016 3 snd_soc_intel_hda_dsp_common 16384 1 snd_soc_sof_sdw snd_soc_intel_sof_maxim_common 20480 1 snd_soc_sof_sdw snd_sof_probes 24576 0 snd_soc_rt711_sdca 53248 1 snd_soc_rt715_sdca 45056 1 snd_hda_codec_hdmi 90112 1 snd_soc_rt1316_sdw 28672 2 snd_soc_dmic 12288 0 snd_sof_pci_intel_tgl 12288 2 snd_sof_intel_hda_common 208896 1 snd_sof_pci_intel_tgl snd_sof_intel_hda 24576 1 snd_sof_intel_hda_common snd_sof_pci 24576 2 snd_sof_intel_hda_common,snd_sof_pci_intel_tgl snd_sof_xtensa_dsp 12288 1 snd_sof_intel_hda_common snd_sof 352256 4 snd_sof_pci,snd_sof_intel_hda_common,snd_sof_probes,snd_sof_intel_hda snd_sof_utils 16384 1 snd_sof snd_soc_hdac_hda 24576 1 snd_sof_intel_hda_common snd_soc_acpi_intel_match 98304 2 snd_sof_intel_hda_common,snd_sof_pci_intel_tgl snd_soc_acpi 16384 2 snd_soc_acpi_intel_match,snd_sof_intel_hda_common snd_hda_intel 57344 0 snd_intel_dspcfg 36864 3 snd_hda_intel,snd_sof,snd_sof_intel_hda_common snd_intel_sdw_acpi 16384 2 snd_sof_intel_hda_common,snd_intel_dspcfg snd_hda_codec 217088 5 snd_hda_codec_hdmi,snd_hda_intel,snd_soc_intel_hda_dsp_common,snd_soc_hdac_hda,snd_sof_intel_hda snd_hwdep 20480 1 snd_hda_codec soundwire_intel 69632 5 snd_sof_intel_hda_common snd_sof_intel_hda_mlink 40960 2 soundwire_intel,snd_sof_intel_hda_common soundwire_cadence 40960 1 soundwire_intel snd_hda_ext_core 40960 4 snd_sof_intel_hda_common,snd_soc_hdac_hda,snd_sof_intel_hda_mlink,snd_sof_intel_hda snd_hda_core 147456 8 snd_hda_codec_hdmi,snd_hda_intel,snd_hda_ext_core,snd_hda_codec,snd_soc_intel_hda_dsp_common,snd_sof_intel_hda_common,snd_soc_hdac_hda,snd_sof_intel_hda soundwire_generic_allocation 12288 1 soundwire_intel snd_soc_core 421888 11 snd_soc_sof_sdw,snd_soc_rt715_sdca,snd_soc_rt1316_sdw,soundwire_intel,snd_sof,snd_sof_intel_hda_common,snd_soc_hdac_hda,snd_soc_rt711_sdca,snd_soc_intel_sof_maxim_common,snd_sof_probes,snd_soc_dmic snd_compress 28672 2 snd_soc_core,snd_sof_probes snd_pcm_dmaengine 16384 1 snd_soc_core snd_pcm 192512 16 snd_soc_rt715_sdca,snd_hda_codec_hdmi,snd_soc_rt1316_sdw,snd_hda_intel,snd_hda_codec,soundwire_intel,snd_sof,snd_sof_intel_hda_common,snd_compress,snd_soc_rt711_sdca,snd_soc_core,snd_sof_utils,snd_soc_intel_sof_maxim_common,snd_hda_core,snd_pcm_dmaengine snd_seq_midi 20480 0 snd_seq_midi_event 16384 1 snd_seq_midi snd_rawmidi 53248 1 snd_seq_midi snd_seq 106496 9 snd_seq_midi,snd_seq_midi_event,snd_seq_dummy snd_seq_device 16384 3 snd_seq,snd_seq_midi,snd_rawmidi snd_timer 49152 3 snd_seq,snd_hrtimer,snd_pcm snd 155648 21 snd_ctl_led,snd_soc_sof_sdw,snd_seq,snd_seq_device,snd_hda_codec_hdmi,snd_hwdep,snd_hda_intel,snd_hda_codec,snd_sof,snd_timer,snd_compress,snd_soc_core,snd_pcm,snd_rawmidi Thanks, Marc -- "A mouse is a device used to point at the xterm you want to type in" - A.S.R. Home page: http://marc.merlins.org/

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[PATCH v2 0/3] Support Dell models without _DSD
by Stefan Binding 02 Jan '24

02 Jan '24

Add Quirks and driver properties for Dell models. Driver properties are required since these models do not have _DSD. Additionally, some laptops, including some of these, have an issue with their BIOS which causes the SPI speed to be set too slow. To ensure a decent performance for laptops with this slow speed, disable firmware loading. Running without firmware results in lower volume. Changes since v1: - Rebased onto for-linus Stefan Binding (3): ALSA: hda: cs35l41: Support additional Dell models without _DSD ALSA: hda: cs35l41: Prevent firmware load if SPI speed too low ALSA: hda/realtek: Add quirks for Dell models sound/pci/hda/cs35l41_hda.c | 25 ++++++++-- sound/pci/hda/cs35l41_hda.h | 12 ++++- sound/pci/hda/cs35l41_hda_i2c.c | 2 +- sound/pci/hda/cs35l41_hda_property.c | 74 +++++++++++++++------------- sound/pci/hda/cs35l41_hda_spi.c | 2 +- sound/pci/hda/patch_realtek.c | 13 +++++ 6 files changed, 87 insertions(+), 41 deletions(-) -- 2.34.1

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[PATCH v2] ALSA: hda: cs35l41: Add HP override
by Lorenz Brun 02 Jan '24

02 Jan '24

This adds an override for a series of notebooks using a common config taken from HP's proprietary Windows driver. This has been tested on a HP 15-ey0xxxx device (subsystem 103C8A31) together with another Realtek quirk and the calibration files from the proprietary driver. Signed-off-by: Lorenz Brun <lorenz(a)brun.one> --- Changes in v2: - Use newly-introduced config table instead of function - Change boost cap value from 10uF to 24uF after cross-referencing my sources with the preexisting 103C89C6 entry. Just noticed that yesterday I sent out another copy of v1 instead of v2, sorry about that. sound/pci/hda/cs35l41_hda_property.c | 46 ++++++++++++++++++++++++++++ 1 file changed, 46 insertions(+) diff --git a/sound/pci/hda/cs35l41_hda_property.c b/sound/pci/hda/cs35l41_hda_property.c index c9eb70290973..c65b106ff1ab 100644 --- a/sound/pci/hda/cs35l41_hda_property.c +++ b/sound/pci/hda/cs35l41_hda_property.c @@ -42,6 +42,30 @@ static const struct cs35l41_config cs35l41_config_table[] = { * in the ACPI. The Reset GPIO is also valid, so we can use the Reset defined in _DSD. */ { "103C89C6", SPI, 2, INTERNAL, { CS35L41_RIGHT, CS35L41_LEFT, 0, 0 }, -1, -1, -1, 1000, 4500, 24 }, + + { "103C8A28", I2C, 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4100, 24 }, + { "103C8A29", I2C, 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4100, 24 }, + { "103C8A2A", I2C, 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4100, 24 }, + { "103C8A2B", I2C, 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4100, 24 }, + { "103C8A2C", I2C, 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4100, 24 }, + { "103C8A2D", I2C, 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4100, 24 }, + { "103C8A2E", I2C, 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4100, 24 }, + { "103C8A30", I2C, 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4100, 24 }, + { "103C8A31", I2C, 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4100, 24 }, + { "103C8BB3", I2C, 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4100, 24 }, + { "103C8BB4", I2C, 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4100, 24 }, + { "103C8BDF", I2C, 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4100, 24 }, + { "103C8BE0", I2C, 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4100, 24 }, + { "103C8BE1", I2C, 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4100, 24 }, + { "103C8BE2", I2C, 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4100, 24 }, + { "103C8BE9", I2C, 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4100, 24 }, + { "103C8BDD", I2C, 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4100, 24 }, + { "103C8BDE", I2C, 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4100, 24 }, + { "103C8BE3", I2C, 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4100, 24 }, + { "103C8BE5", I2C, 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4100, 24 }, + { "103C8BE6", I2C, 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4100, 24 }, + { "103C8B3A", I2C, 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4100, 24 }, + { "104312AF", SPI, 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 1000, 4500, 24 }, { "10431433", I2C, 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4500, 24 }, { "10431463", I2C, 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4500, 24 }, @@ -374,6 +398,28 @@ static const struct cs35l41_prop_model cs35l41_prop_model_table[] = { { "CSC3551", "10431F12", generic_dsd_config }, { "CSC3551", "10431F1F", generic_dsd_config }, { "CSC3551", "10431F62", generic_dsd_config }, + { "CSC3551", "103C8A28", generic_dsd_config }, + { "CSC3551", "103C8A29", generic_dsd_config }, + { "CSC3551", "103C8A2A", generic_dsd_config }, + { "CSC3551", "103C8A2B", generic_dsd_config }, + { "CSC3551", "103C8A2C", generic_dsd_config }, + { "CSC3551", "103C8A2D", generic_dsd_config }, + { "CSC3551", "103C8A2E", generic_dsd_config }, + { "CSC3551", "103C8A30", generic_dsd_config }, + { "CSC3551", "103C8A31", generic_dsd_config }, + { "CSC3551", "103C8BB3", generic_dsd_config }, + { "CSC3551", "103C8BB4", generic_dsd_config }, + { "CSC3551", "103C8BDF", generic_dsd_config }, + { "CSC3551", "103C8BE0", generic_dsd_config }, + { "CSC3551", "103C8BE1", generic_dsd_config }, + { "CSC3551", "103C8BE2", generic_dsd_config }, + { "CSC3551", "103C8BE9", generic_dsd_config }, + { "CSC3551", "103C8BDD", generic_dsd_config }, + { "CSC3551", "103C8BDE", generic_dsd_config }, + { "CSC3551", "103C8BE3", generic_dsd_config }, + { "CSC3551", "103C8BE5", generic_dsd_config }, + { "CSC3551", "103C8BE6", generic_dsd_config }, + { "CSC3551", "103C8B3A", generic_dsd_config }, {} }; -- 2.42.0

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[PATCH 00/16] ALSA: hda/tas2781: Add tas2563 support
by Gergo Koteles 02 Jan '24

02 Jan '24

The tas2781-hda driver can be modified to support tas2563 as well. Before knowing this information, I created another series for a new driver. https://lore.kernel.org/lkml/cover.1701733441.git.soyer@irl.hu/ This series now extends tas2781-hda, addresses differences and fixes various bugs. The tas2563 is a smart amplifier. Similar to tas2562 but with DSP. Some Lenovo laptops have it to drive the bass speakers. By default, it is in software shutdown state. To make the DSP work it needs a firmware and some calibration data. The latter can be read from the EFI in Lenovo laptops. For the correct configuration it needs additional register data. It captured after running the Windows driver. The firmware can be extracted as TAS2563Firmware.bin from the Windows driver with innoextract. https://download.lenovo.com/consumer/mobiles/h5yd037fbfyy7kd0.exe The driver will search for it as TAS2XXX3870.bin with the 14ARB7. The captured registers extracted with TI's regtool: https://github.com/soyersoyer/tas2563rca/raw/main/INT8866RCA2.bin Gergo Koteles (16): ASoC: tas2781: add support for fw version 0x0503 ALSA: hda/tas2781: leave hda_component in usable state ASoC: tas2781: disable regmap regcache ALSA: hda/tas2781: handle missing calibration data ALSA: hda/tas2781: fix typos in comment ASoC: tas2781: add ptrs to calibration functions ALSA: hda/tas2781: load_calibration just load ASoC: tas2781: add configurable global_addr ALSA: hda/tas2781: add TAS2563 support for 14ARB7 ASoC: tas2781: check negative indexes ASoC: tas2781: use 0 as default prog/conf index ASoC: tas2781: move set_drv_data outside tasdevice_init ALSA: hda/tas2781: remove sound controls in unbind ALSA: hda/tas2781: call cleaner functions only once ALSA: hda/tas2781: reset the amp before component_add ALSA: hda/tas2781: configure the amp after firmware load include/sound/tas2781.h | 8 + sound/pci/hda/tas2781_hda_i2c.c | 364 +++++++++++++++++++----------- sound/soc/codecs/tas2781-comlib.c | 23 +- sound/soc/codecs/tas2781-fmwlib.c | 11 +- sound/soc/codecs/tas2781-i2c.c | 2 + 5 files changed, 270 insertions(+), 138 deletions(-) base-commit: ffc253263a1375a65fa6c9f62a893e9767fbebfa -- 2.43.0

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[PATCH 00/23] ALSA: scarlett2: Refactor in preparation for gen4
by Geoffrey D. Bennett 02 Jan '24

02 Jan '24

Hi Takashi, This series is preparation for adding support for the Focusrite Scarlett 4th Gen devices. It applies on top of the previous series https://lore.kernel.org/linux-sound/cover.1703001053.git.g@b4.vu/ "ALSA: scarlett2: Firmware Upgrade and Error Handling Improvements". There should be no notable functional changes in this series, just refactoring/restructuring/renaming/reformatting. Regards, Geoffrey. Geoffrey D. Bennett (23): ALSA: scarlett2: Simplify enums by removing explicit values ALSA: scarlett2: Infer has_msd_mode from config items ALSA: scarlett2: Infer standalone switch from config items ALSA: scarlett2: Check for phantom persistence config item ALSA: scarlett2: Check presence of mixer using mux_assignment ALSA: scarlett2: Add config set struct ALSA: scarlett2: Remove scarlett2_config_sets array ALSA: scarlett2: Add check for config_item presence ALSA: scarlett2: Refactor scarlett2_usb_set_config() ALSA: scarlett2: Refactor scarlett2_config_save() ALSA: scarlett2: Formatting fixes ALSA: scarlett2: Parameterise notifications ALSA: scarlett2: Change num_mux_* from int to u8 ALSA: scarlett2: Refactor common port_count lookups ALSA: scarlett2: Remove struct scarlett2_usb_volume_status ALSA: scarlett2: Split dim_mute_update from vol_updated ALSA: scarlett2: Remove line_out_hw_vol device info entry ALSA: scarlett2: Allow for interfaces without per-channel volume ALSA: scarlett2: Add scarlett2_mixer_value_to_db() ALSA: scarlett2: Add #define for SCARLETT2_MIX_MAX ALSA: scarlett2: Rename db_scale_scarlett2_gain to volume ALSA: scarlett2: Split input_other into level/pad/air/phantom ALSA: scarlett2: Split direct_monitor out from monitor_other sound/usb/mixer_scarlett2.c | 1536 ++++++++++++++++++++--------------- 1 file changed, 879 insertions(+), 657 deletions(-) -- 2.43.0

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[PATCH] ALSA: hda/tas2781: fix typos in comment
by Gergo Koteles 02 Jan '24

02 Jan '24

Correct typos. Signed-off-by: Gergo Koteles <soyer(a)irl.hu> --- sound/pci/hda/tas2781_hda_i2c.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/tas2781_hda_i2c.c b/sound/pci/hda/tas2781_hda_i2c.c index fb802802939e..8f63e3099d3e 100644 --- a/sound/pci/hda/tas2781_hda_i2c.c +++ b/sound/pci/hda/tas2781_hda_i2c.c @@ -421,9 +421,9 @@ static void tas2781_apply_calib(struct tasdevice_priv *tas_priv) } } -/* Update the calibrate data, including speaker impedance, f0, etc, into algo. +/* Update the calibration data, including speaker impedance, f0, etc, into algo. * Calibrate data is done by manufacturer in the factory. These data are used - * by Algo for calucating the speaker temperature, speaker membrance excursion + * by Algo for calculating the speaker temperature, speaker membrane excursion * and f0 in real time during playback. */ static int tas2781_save_calibration(struct tasdevice_priv *tas_priv) base-commit: c7e37b07cc7564a07125ae48c11fd1ca2bcbeae2 -- 2.43.0

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[PATCH 00/11] ALSA: scarlett2: Firmware Upgrade and Error Handling Improvements
by Geoffrey D. Bennett 02 Jan '24

02 Jan '24

Hi Takashi, 11 patches to improve error handling and add firmware upgrade support to the scarlett2 driver. The notes below incorporate your feedback & include answers to your questions in response to my previous RFC v2 email. Patch 1 adds GitHub links for the repo where I share this code before submission here, and for issue reporting. Patches 2-5 add missing error/bounds checks. Patch 6 adds a missing lock. Patches 7-11 add support for firmware upgrades. The usage of the hwdep interface is as per my previous proposal: - ioctl pversion to check the protocol version - ioctl select_flash_segment SCARLETT2_SEGMENT_ID_SETTINGS - ioctl erase_flash_segment - ioctl get_erase_progress (optional) - ioctl select_flash_segment SCARLETT2_SEGMENT_ID_FIRMWARE - ioctl erase_flash_segment - ioctl get_erase_progress (optional) - write() the firmware - ioctl reboot Notes on the hwdep interface: - Conflicting mixer operations are denied with EBUSY. - The hwdep interface is marked as exclusive so there can't be conflicting hwdep operations. - Invalid sequences of operations (e.g. erase before select, write before erase) return an error. - The erase operation is asynchronous, and the progress can optionally be monitored by calling get_erase_progress. - If the erase progress is not monitored then subsequent hwdep operations will wait until the erase is complete. - The write operation is synchronous, but <1KB can be written per call, and it returns very quickly. On the user-side it looks like this with error checking omitted: while (offset < len) offset += snd_hwdep_write(hwdep, buf + offset, len - offset); - By using a subset of the firmware-upgrade steps, other useful high-level operations can be performed: reset-to-factory-defaults, reset-to-factory-firmware, or just reboot. - I considered combining the select and erase operations, but that would prevent a future read-flash operation if that ever become possible. - I considered separate one-shot ioctls for reset-to-factory, etc., but that prevents providing progress feedback. - In case the purpose of the other firmware segments is discovered in the future, this implementation would be able to support erase/write/read of them with minimal changes. Here is a user-space implementation of all of the above: https://github.com/geoffreybennett/scarlett2 This has been tested by me on every supported Scarlett 2nd, 3rd, 4th Gen and Clarett+ device, and tested by a handful of others on their Scarlett 4th Gen devices. Thanks, Geoffrey. Geoffrey D. Bennett (11): ALSA: scarlett2: Update maintainer info ALSA: scarlett2: Add missing error check to scarlett2_config_save() ALSA: scarlett2: Add missing error check to scarlett2_usb_set_config() ALSA: scarlett2: Add missing error checks to *_ctl_get() ALSA: scarlett2: Add clamp() in scarlett2_mixer_ctl_put() ALSA: scarlett2: Add missing mutex lock around get meter levels ALSA: scarlett2: Add #defines for firmware upgrade ALSA: scarlett2: Retrieve useful flash segment numbers ALSA: scarlett2: Add skeleton hwdep/ioctl interface ALSA: scarlett2: Add ioctl commands to erase flash segments ALSA: scarlett2: Add support for uploading new firmware MAINTAINERS | 7 +- include/uapi/sound/scarlett2.h | 54 ++ sound/usb/mixer_scarlett2.c | 914 ++++++++++++++++++++++++++++++--- 3 files changed, 901 insertions(+), 74 deletions(-) create mode 100644 include/uapi/sound/scarlett2.h -- 2.43.0

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