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Re: [PATCH] Add support for 2024 ROG Strix G634JZR
by Takashi Iwai 02 Apr '24

02 Apr '24

On Tue, 02 Apr 2024 08:04:39 +0200, Takashi Iwai wrote: > > On Mon, 01 Apr 2024 16:31:30 +0200, > Daniel Berlin via B4 Relay wrote: > > > > From: Daniel Berlin <dberlin(a)dberlin.org> > > > > The windows driver inf says it uses the exact same tunings and setup as the > > older G634, and then reuses the data for ID 1043:1CAF. > > > > We do the same here. > > Tested on an actual laptop, sound works fine with this patch plus > > default fallback firmware. > > > > Signed-off-by: Daniel Berlin <dberlin(a)dberlin.org> > > Applied now. Thanks. ... and now I see that Luke has another patch that modified the entry in patch_realtek.c, and I took his version instead. https://lore.kernel.org/r/20240402015126.21115-1-luke@ljones.dev Let me know if this causes problem for you. thanks, Takashi

1 0

[PATCH] ALSA: hda/realtek: Update Panasonic CF-SZ6 quirk to support headset with microphone
by I Gede Agastya Darma Laksana 02 Apr '24

02 Apr '24

This patch addresses an issue with the Panasonic CF-SZ6's existing quirk, specifically its headset microphone functionality. Previously, the quirk used ALC269_FIXUP_HEADSET_MODE, which does not support the CF-SZ6's design of a single 3.5mm jack for both mic and audio output effectively. The device uses pin 0x19 for the headset mic without jack detection. Following verification on the CF-SZ6 and discussions with the original patch author, i determined that the update to ALC269_FIXUP_ASPIRE_HEADSET_MIC is the appropriate solution. This change is custom-designed for the CF-SZ6's unique hardware setup, which includes a single 3.5mm jack for both mic and audio output, connecting the headset microphone to pin 0x19 without the use of jack detection. Signed-off-by: I Gede Agastya Darma Laksana <gedeagas22(a)gmail.com> --- sound/pci/hda/patch_realtek.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index a17c36a36..9dedb6504 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -10159,7 +10159,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x10ec, 0x1254, "Intel Reference board", ALC295_FIXUP_CHROME_BOOK), SND_PCI_QUIRK(0x10ec, 0x12cc, "Intel Reference board", ALC295_FIXUP_CHROME_BOOK), SND_PCI_QUIRK(0x10ec, 0x12f6, "Intel Reference board", ALC295_FIXUP_CHROME_BOOK), - SND_PCI_QUIRK(0x10f7, 0x8338, "Panasonic CF-SZ6", ALC269_FIXUP_HEADSET_MODE), + SND_PCI_QUIRK(0x10f7, 0x8338, "Panasonic CF-SZ6", ALC269_FIXUP_ASPIRE_HEADSET_MIC), SND_PCI_QUIRK(0x144d, 0xc109, "Samsung Ativ book 9 (NP900X3G)", ALC269_FIXUP_INV_DMIC), SND_PCI_QUIRK(0x144d, 0xc169, "Samsung Notebook 9 Pen (NP930SBE-K01US)", ALC298_FIXUP_SAMSUNG_AMP), SND_PCI_QUIRK(0x144d, 0xc176, "Samsung Notebook 9 Pro (NP930MBE-K04US)", ALC298_FIXUP_SAMSUNG_AMP), -- 2.34.1

2 1

[PATCH] Add sound quirks for Lenovo Legion slim 7 16ARHA7 models.
by Christian＠bendiksen.me 02 Apr '24

02 Apr '24

From: Christian Bendiksen <christian(a)bendiksen.me> This fixes the sound not working from internal speakers on Lenovo Legion Slim 7 16ARHA7 models. The correct subsystem ID have been added to cs35l41_hda_property.c and patch_realtek.c. Signed-off-by: Christian Bendiksen <christian(a)bendiksen.me> --- sound/pci/hda/cs35l41_hda_property.c | 4 ++++ sound/pci/hda/patch_realtek.c | 2 ++ 2 files changed, 6 insertions(+) diff --git a/sound/pci/hda/cs35l41_hda_property.c b/sound/pci/hda/cs35l41_hda_property.c index 72ec872afb8d..959864994eca 100644 --- a/sound/pci/hda/cs35l41_hda_property.c +++ b/sound/pci/hda/cs35l41_hda_property.c @@ -115,6 +115,8 @@ static const struct cs35l41_config cs35l41_config_table[] = { { "17AA38B5", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 0, 0, 0 }, { "17AA38B6", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 0, 0, 0 }, { "17AA38B7", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 0, 0, 0 }, + { "17AA3877", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 0, 0, 0 }, + { "17AA3878", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 0, 0, 0 }, {} }; @@ -503,6 +505,8 @@ static const struct cs35l41_prop_model cs35l41_prop_model_table[] = { { "CSC3551", "17AA38B5", generic_dsd_config }, { "CSC3551", "17AA38B6", generic_dsd_config }, { "CSC3551", "17AA38B7", generic_dsd_config }, + { "CSC3551", "17AA3877", generic_dsd_config }, + { "CSC3551", "17AA3878", generic_dsd_config }, {} }; diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index a17c36a36aa5..540868d6c602 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -10347,6 +10347,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x38b5, "Legion Slim 7 16IRH8", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x17aa, 0x38b6, "Legion Slim 7 16APH8", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x17aa, 0x38b7, "Legion Slim 7 16APH8", ALC287_FIXUP_CS35L41_I2C_2), + SND_PCI_QUIRK(0x17aa, 0x3877, "Legion Slim 7 16ARHA7", ALC287_FIXUP_CS35L41_I2C_2), + SND_PCI_QUIRK(0x17aa, 0x3878, "Legion Slim 7 16ARHA7", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x17aa, 0x38ba, "Yoga S780-14.5 Air AMD quad YC", ALC287_FIXUP_TAS2781_I2C), SND_PCI_QUIRK(0x17aa, 0x38bb, "Yoga S780-14.5 Air AMD quad AAC", ALC287_FIXUP_TAS2781_I2C), SND_PCI_QUIRK(0x17aa, 0x38be, "Yoga S980-14.5 proX YC Dual", ALC287_FIXUP_TAS2781_I2C), -- 2.44.0

3 5

[PATCH 0/9] enabled -Wformat-truncation for clang
by Arnd Bergmann 02 Apr '24

02 Apr '24

From: Arnd Bergmann <arnd(a)arndb.de> With randconfig build testing, I found only eight files that produce warnings with clang when -Wformat-truncation is enabled. This means we can just turn it on by default rather than only enabling it for "make W=1". Unfortunately, gcc produces a lot more warnings when the option is enabled, so it's not yet possible to turn it on both both compilers. I hope that the patches can get picked up by platform maintainers directly, so the final patch can go in later on. Arnd Arnd Bergmann (9): fbdev: shmobile: fix snprintf truncation enetc: avoid truncating error message qed: avoid truncating work queue length mlx5: avoid truncating error message surface3_power: avoid format string truncation warning Input: IMS: fix printf string overflow scsi: mylex: fix sysfs buffer lengths ALSA: aoa: avoid false-positive format truncation warning kbuild: enable -Wformat-truncation on clang drivers/input/misc/ims-pcu.c | 4 ++-- drivers/net/ethernet/freescale/enetc/enetc.c | 2 +- .../ethernet/mellanox/mlx5/core/esw/bridge.c | 2 +- drivers/net/ethernet/qlogic/qed/qed_main.c | 9 ++++--- drivers/platform/surface/surface3_power.c | 2 +- drivers/scsi/myrb.c | 20 ++++++++-------- drivers/scsi/myrs.c | 24 +++++++++---------- drivers/video/fbdev/sh_mobile_lcdcfb.c | 2 +- scripts/Makefile.extrawarn | 2 ++ sound/aoa/soundbus/i2sbus/core.c | 2 +- 10 files changed, 35 insertions(+), 34 deletions(-) -- 2.39.2 Cc: Dmitry Torokhov <dmitry.torokhov(a)gmail.com> Cc: Claudiu Manoil <claudiu.manoil(a)nxp.com> Cc: Vladimir Oltean <vladimir.oltean(a)nxp.com> Cc: Jakub Kicinski <kuba(a)kernel.org> Cc: Saeed Mahameed <saeedm(a)nvidia.com> Cc: Leon Romanovsky <leon(a)kernel.org> Cc: Ariel Elior <aelior(a)marvell.com> Cc: Manish Chopra <manishc(a)marvell.com> Cc: Hans de Goede <hdegoede(a)redhat.com> Cc: "Ilpo Järvinen" <ilpo.jarvinen(a)linux.intel.com> Cc: Maximilian Luz <luzmaximilian(a)gmail.com> Cc: Hannes Reinecke <hare(a)kernel.org> Cc: "Martin K. Petersen" <martin.petersen(a)oracle.com> Cc: Helge Deller <deller(a)gmx.de> Cc: Masahiro Yamada <masahiroy(a)kernel.org> Cc: Nathan Chancellor <nathan(a)kernel.org> Cc: Nicolas Schier <nicolas(a)fjasle.eu> Cc: Johannes Berg <johannes(a)sipsolutions.net> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: Nick Desaulniers <ndesaulniers(a)google.com> Cc: Bill Wendling <morbo(a)google.com> Cc: Justin Stitt <justinstitt(a)google.com> Cc: linux-input(a)vger.kernel.org Cc: linux-kernel(a)vger.kernel.org Cc: netdev(a)vger.kernel.org Cc: linux-rdma(a)vger.kernel.org Cc: platform-driver-x86(a)vger.kernel.org Cc: linux-scsi(a)vger.kernel.org Cc: linux-fbdev(a)vger.kernel.org Cc: dri-devel(a)lists.freedesktop.org Cc: linux-kbuild(a)vger.kernel.org Cc: linuxppc-dev(a)lists.ozlabs.org Cc: alsa-devel(a)alsa-project.org Cc: linux-sound(a)vger.kernel.org Cc: llvm(a)lists.linux.dev

5 5

[PATCH] Revert "ALSA: emu10k1: fix synthesizer sample playback position and caching"
by Oswald Buddenhagen 01 Apr '24

01 Apr '24

As already anticipated in the original commit, playback was broken for very short samples. I just didn't expect it to be an actual problem, because we're talking about less than 1.5 milliseconds here. But clearly such wavetable samples do actually exist. The problem was that for such short samples we'd set the current position beyond the end of the loop, so we'd run off the end of the sample and play garbage. This is a bigger (more audible) problem than the original one, which was that we'd start playback with garbage (whatever was still in the cache), which would be mostly masked by the note's attack phase. So revert to the old behavior for now. We'll subsequently fix it properly with a bigger patch series. Note that this isn't a full revert - the dead code is not re-introduced, because that would be silly. Fixes: df335e9a8b (ALSA: emu10k1: fix synthesizer sample playback position and caching, 2023-05-18) Link: https://bugzilla.kernel.org/show_bug.cgi?id=218625 Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen(a)gmx.de> --- sound/pci/emu10k1/emu10k1_callback.c | 7 ++----- 1 file changed, 2 insertions(+), 5 deletions(-) diff --git a/sound/pci/emu10k1/emu10k1_callback.c b/sound/pci/emu10k1/emu10k1_callback.c index d36234b88fb4..941bfbf812ed 100644 --- a/sound/pci/emu10k1/emu10k1_callback.c +++ b/sound/pci/emu10k1/emu10k1_callback.c @@ -255,7 +255,7 @@ lookup_voices(struct snd_emux *emu, struct snd_emu10k1 *hw, /* check if sample is finished playing (non-looping only) */ if (bp != best + V_OFF && bp != best + V_FREE && (vp->reg.sample_mode & SNDRV_SFNT_SAMPLE_SINGLESHOT)) { - val = snd_emu10k1_ptr_read(hw, CCCA_CURRADDR, vp->ch) - 64; + val = snd_emu10k1_ptr_read(hw, CCCA_CURRADDR, vp->ch); if (val >= vp->reg.loopstart) bp = best + V_OFF; } @@ -362,7 +362,7 @@ start_voice(struct snd_emux_voice *vp) map = (hw->silent_page.addr << hw->address_mode) | (hw->address_mode ? MAP_PTI_MASK1 : MAP_PTI_MASK0); - addr = vp->reg.start + 64; + addr = vp->reg.start; temp = vp->reg.parm.filterQ; ccca = (temp << 28) | addr; if (vp->apitch < 0xe400) @@ -430,9 +430,6 @@ start_voice(struct snd_emux_voice *vp) /* Q & current address (Q 4bit value, MSB) */ CCCA, ccca, - /* cache */ - CCR, REG_VAL_PUT(CCR_CACHEINVALIDSIZE, 64), - /* reset volume */ VTFT, vtarget | vp->ftarget, CVCF, vtarget | CVCF_CURRENTFILTER_MASK, -- 2.42.0.419.g70bf8a5751

2 1

[PATCH] Add support for 2024 ROG Strix G634JZR
by Daniel Berlin via B4 Relay 01 Apr '24

01 Apr '24

From: Daniel Berlin <dberlin(a)dberlin.org> The windows driver inf says it uses the exact same tunings and setup as the older G634, and then reuses the data for ID 1043:1CAF. We do the same here. Tested on an actual laptop, sound works fine with this patch plus default fallback firmware. Signed-off-by: Daniel Berlin <dberlin(a)dberlin.org> --- sound/pci/hda/cs35l41_hda_property.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/cs35l41_hda_property.c b/sound/pci/hda/cs35l41_hda_property.c index 72ec872afb8d..25c117db3317 100644 --- a/sound/pci/hda/cs35l41_hda_property.c +++ b/sound/pci/hda/cs35l41_hda_property.c @@ -108,6 +108,7 @@ static const struct cs35l41_config cs35l41_config_table[] = { { "10431F12", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4500, 24 }, { "10431F1F", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, -1, 0, 0, 0, 0 }, { "10431F62", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 0, 0, 0 }, + { "10433A60", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 1000, 4500, 24 }, { "17AA386F", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, -1, -1, 0, 0, 0 }, { "17AA38A9", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 2, -1, 0, 0, 0 }, { "17AA38AB", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 2, -1, 0, 0, 0 }, @@ -496,6 +497,7 @@ 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", "10433A60", generic_dsd_config }, { "CSC3551", "17AA386F", generic_dsd_config }, { "CSC3551", "17AA38A9", generic_dsd_config }, { "CSC3551", "17AA38AB", generic_dsd_config }, --- base-commit: 39cd87c4eb2b893354f3b850f916353f2658ae6f change-id: 20240401-add-rog-strix-16-g634jzr-9e5490cadc2b Best regards, -- Daniel Berlin <dberlin(a)dberlin.org>

1 0

[PATCH] ALSA: soc/au1x: declare that FIFO is reported in frames
by Oswald Buddenhagen 01 Apr '24

01 Apr '24

Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen(a)gmx.de> --- Is this right? The comment on the field says "fifo entries of AC97/I2S PSC", which suggests frames, but that may mean whatever. Cc: Manuel Lauss <mano(a)roarinelk.homelinux.net> --- sound/soc/au1x/dbdma2.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/au1x/dbdma2.c b/sound/soc/au1x/dbdma2.c index ea01d6490cec..2c8c01a5fb8f 100644 --- a/sound/soc/au1x/dbdma2.c +++ b/sound/soc/au1x/dbdma2.c @@ -66,7 +66,8 @@ struct au1xpsc_audio_dmadata { /* PCM hardware DMA capabilities - platform specific */ static const struct snd_pcm_hardware au1xpsc_pcm_hardware = { .info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BATCH, + SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BATCH | + SNDRV_PCM_INFO_FIFO_IN_FRAMES, .period_bytes_min = AU1XPSC_PERIOD_MIN_BYTES, .period_bytes_max = 4096 * 1024 - 1, .periods_min = 2, -- 2.42.0.419.g70bf8a5751

2 1

[PATCH 00/18] ALSA: emu10k1 & emux: fixes related to wavetable playback
by Oswald Buddenhagen 01 Apr '24

01 Apr '24

this fixes the regression i introduced (though arguably, i just made it broken in a different way), and then some more. Oswald Buddenhagen (18): ALSA: emux: fix /proc teardown at module unload ALSA: emux: prune unused parameter from snd_soundfont_load_guspatch() ALSA: emux: fix validation of snd_emux.num_ports ALSA: emux: fix init of patch_info.truesize in load_data() ALSA: emu10k1: prune vestiges of SNDRV_SFNT_SAMPLE_{BIDIR,REVERSE}_LOOP support ALSA: emux: centralize & improve patch info validation ALSA: emux: improve patch ioctl data validation ALSA: emu10k1: move patch loader assertions into low-level functions ALSA: emu10k1: fix sample signedness issues in wavetable loader ALSA: emu10k1: fix playback of 8-bit wavetable samples ALSA: emu10k1: make wavetable sample playback start position exact ALSA: emu10k1: shrink blank space in front of wavetable samples ALSA: emu10k1: merge conditions in patch loader ALSA: emu10k1: fix wavetable offset recalculation ALSA: emu10k1: de-duplicate size calculations for 16-bit samples ALSA: emu10k1: improve cache behavior documentation ALSA: emu10k1: fix playback of short wavetable samples ALSA: emux: simplify snd_sf_list.callback handling include/sound/emu10k1.h | 32 +++-- include/sound/soundfont.h | 2 +- sound/isa/sb/emu8000_patch.c | 13 -- sound/pci/emu10k1/emu10k1_callback.c | 10 +- sound/pci/emu10k1/emu10k1_patch.c | 207 +++++++++++---------------- sound/pci/emu10k1/memory.c | 55 +++++-- sound/synth/emux/emux.c | 6 +- sound/synth/emux/emux_hwdep.c | 3 +- sound/synth/emux/emux_oss.c | 3 +- sound/synth/emux/emux_proc.c | 1 + sound/synth/emux/emux_seq.c | 6 +- sound/synth/emux/soundfont.c | 73 +++++++--- 12 files changed, 216 insertions(+), 195 deletions(-) -- 2.42.0.419.g70bf8a5751

2 21

[PATCH v6 0/2] ASoC: nau8325: Modify driver code and dtschema.
by Seven Lee 01 Apr '24

01 Apr '24

Revise properties description and use standard units in dtschema. The unit conversion driver based on the attribute must also be changed accordingly. Change: V1 -> V2: - Revise the driver description part for the C++ comment. - In the nau8325_clkdet_put function, modify the max variable to hard code. - Removed "Clock Detection" switch control. - modify the "ALC Enable" switch name. - Revise the dtschema for "nuvoton,dac-vref". V2 -> V3: - Properties use standard unit suffixes. - Modify the enum definition. - Driver code should be used dev_dbg(). V3 -> V4: - Properties use standard unit suffixes. - Modify the enum definition. V4 -> V5: - Properties use standard unit suffixes for vref-impedance and dac-vref. - Revise the enum definition. - Modify the code related to properties usage in the NAU8325 codec driver. - Re-arrange header files in alphabetical order. V5 -> V6: - Revise the nau8325_dac_oversampl_enum structure definition. Seven Lee (2): ASoC: dt-bindings: Added schema for "nuvoton,nau8325" ASoC: nau8325: new driver .../bindings/sound/nuvoton,nau8325.yaml | 80 ++ sound/soc/codecs/nau8325.c | 900 ++++++++++++++++++ sound/soc/codecs/nau8325.h | 391 ++++++++ 3 files changed, 1371 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/nuvoton,nau8325.yaml create mode 100644 sound/soc/codecs/nau8325.c create mode 100644 sound/soc/codecs/nau8325.h -- 2.25.1

3 4

Segmentation fault on Python 3.12
by GitHub issues - opened 30 Mar '24

30 Mar '24

alsa-project/alsa-python issue #9 was opened from FrancescoCeruti: When using Python 3.12, I'm getting a segmentation fault, by simply importing the `alsasq` module (`from pyalsa import alsaseq`) Not sure if this it's the same as #8, here's the backtrace: ``` (gdb) bt #0 __memset_avx512_unaligned_erms () at ../sysdeps/x86_64/multiarch/memset-vec-unaligned-erms.S:216 #1 0x00007ffff79e6122 in _PyType_AllocNoTrack (nitems=1, type=0x7ffff7f571c0 <ConstantType>) at Objects/typeobject.c:1708 #2 PyType_GenericAlloc (type=0x7ffff7f571c0 <ConstantType>, nitems=1) at Objects/typeobject.c:1722 #3 0x00007ffff79b0a0e in long_subtype_new (obase=<optimized out>, x=<optimized out>, type=0x7ffff7f571c0 <ConstantType>) at Objects/longobject.c:5616 #4 long_new_impl (obase=<optimized out>, x=<optimized out>, type=0x7ffff7f571c0 <ConstantType>) at Objects/longobject.c:5555 #5 long_new (type=0x7ffff7f571c0 <ConstantType>, args=<optimized out>, kwargs=<optimized out>) at Objects/clinic/longobject.c.h:65 #6 0x00007ffff7f49963 in Constant_create (name=name@entry=0x7ffff7f4e47c "SEQ_OPEN_INPUT", value=value@entry=2) at pyalsa/alsaseq.c:387 #7 0x00007ffff7f4abce in PyInit_alsaseq () at pyalsa/alsaseq.c:3337 #8 0x00007ffff7ac1ed1 in _PyImport_LoadDynamicModuleWithSpec (spec=spec@entry=0x7ffff71fe840, fp=fp@entry=0x0) at ./Python/importdl.c:169 #9 0x00007ffff7abee8a in _imp_create_dynamic_impl (module=<optimized out>, file=0x0, spec=0x7ffff71fe840) at Python/import.c:3775 #10 _imp_create_dynamic (module=<optimized out>, args=<optimized out>, nargs=<optimized out>) at Python/clinic/import.c.h:506 #11 0x00007ffff79cac88 in cfunction_vectorcall_FASTCALL (func=0x7ffff71972e0, args=0x7ffff71fe6c8, nargsf=<optimized out>, kwnames=0x0) at Objects/methodobject.c:422 #12 0x00007ffff790c8bf in _PyEval_EvalFrameDefault (tstate=<optimized out>, frame=<optimized out>, throwflag=<optimized out>) at Python/bytecodes.c:3254 #13 0x00007ffff7972487 in _PyObject_VectorcallTstate (kwnames=0x0, nargsf=2, args=0x7fffffffbae0, callable=0x7ffff71a4040, tstate=0x7ffff7ec3988 <_PyRuntime+459656>) at ./Include/internal/pycore_call.h:92 #14 object_vacall (tstate=tstate@entry=0x7ffff7ec3988 <_PyRuntime+459656>, base=base@entry=0x0, callable=0x7ffff71a4040, vargs=vargs@entry=0x7fffffffbb70) at Objects/call.c:850 #15 0x00007ffff79726e7 in PyObject_CallMethodObjArgs (obj=0x0, name=0x7ffff7e5c0b8 <_PyRuntime+35512>) at Objects/call.c:911 #16 0x00007ffff7ac076b in import_find_and_load (abs_name=0x7ffff7028eb0, tstate=0x7ffff7ec3988 <_PyRuntime+459656>) at Python/import.c:2779 #17 PyImport_ImportModuleLevelObject (name=name@entry=0x7ffff7028eb0, globals=globals@entry=0x0, locals=locals@entry=0x0, fromlist=fromlist@entry=0x0, level=0) at Python/import.c:2862 #18 0x00007ffff7a82496 in builtin___import___impl (level=<optimized out>, fromlist=0x0, locals=0x0, globals=0x0, name=0x7ffff7028eb0, module=<optimized out>) at Python/bltinmodule.c:275 #19 builtin___import__ (module=<optimized out>, args=<optimized out>, nargs=<optimized out>, kwnames=<optimized out>) at Python/clinic/bltinmodule.c.h:107 #20 0x00007ffff79ca9fd in cfunction_vectorcall_FASTCALL_KEYWORDS (func=0x7ffff71959e0, args=0x7ffff719ad28, nargsf=<optimized out>, kwnames=0x0) at Objects/methodobject.c:438 #21 0x00007ffff790c8bf in _PyEval_EvalFrameDefault (tstate=<optimized out>, frame=<optimized out>, throwflag=<optimized out>) at Python/bytecodes.c:3254 #22 0x00007ffff7972487 in _PyObject_VectorcallTstate (kwnames=0x0, nargsf=3, args=0x7fffffffbf60, callable=0x7ffff71a4180, tstate=0x7ffff7ec3988 <_PyRuntime+459656>) at ./Include/internal/pycore_call.h:92 #23 object_vacall (tstate=tstate@entry=0x7ffff7ec3988 <_PyRuntime+459656>, base=base@entry=0x0, callable=0x7ffff71a4180, vargs=vargs@entry=0x7fffffffbff0) at Objects/call.c:850 #24 0x00007ffff79726e7 in PyObject_CallMethodObjArgs (obj=0x0, name=name@entry=0x7ffff7e5c190 <_PyRuntime+35728>) at Objects/call.c:911 #25 0x00007ffff7ac0326 in PyImport_ImportModuleLevelObject (name=name@entry=0x7ffff71fd7d0, globals=<optimized out>, locals=locals@entry=0x7ffff71f9880, fromlist=fromlist@entry=0x7ffff71fc2e0, level=0) at Python/import.c:2931 #26 0x00007ffff791562e in import_name (level=0x7ffff7e542c8 <_PyRuntime+3272>, fromlist=0x7ffff71fc2e0, name=0x7ffff71fd7d0, frame=<optimized out>, tstate=0x7ffff71959e0) at Python/ceval.c:2482 #27 _PyEval_EvalFrameDefault (tstate=tstate@entry=0x7ffff7ec3988 <_PyRuntime+459656>, frame=<optimized out>, frame@entry=0x7ffff7f9b020, throwflag=throwflag@entry=0) at Python/bytecodes.c:2135 #28 0x00007ffff7a89247 in _PyEval_EvalFrame (throwflag=0, frame=0x7ffff7f9b020, tstate=0x7ffff7ec3988 <_PyRuntime+459656>) at ./Include/internal/pycore_ceval.h:89 #29 _PyEval_Vector (args=0x0, argcount=0, kwnames=0x0, locals=0x7ffff71f9880, func=0x7ffff71da160, tstate=0x7ffff7ec3988 <_PyRuntime+459656>) at Python/ceval.c:1683 #30 PyEval_EvalCode (co=co@entry=0x7ffff70183f0, globals=globals@entry=0x7ffff71f9880, locals=locals@entry=0x7ffff71f9880) at Python/ceval.c:578 #31 0x00007ffff7ae3506 in run_eval_code_obj (tstate=tstate@entry=0x7ffff7ec3988 <_PyRuntime+459656>, co=co@entry=0x7ffff70183f0, globals=globals@entry=0x7ffff71f9880, locals=locals@entry=0x7ffff71f9880) at Python/pythonrun.c:1722 #32 0x00007ffff7ae3619 in run_mod (mod=<optimized out>, filename=filename@entry=0x7ffff712f450, globals=globals@entry=0x7ffff71f9880, locals=locals@entry=0x7ffff71f9880, flags=flags@entry=0x7fffffffc520, arena=arena@entry=0x7ffff711be30) at Python/pythonrun.c:1743 #33 0x00007ffff7ae650c in pyrun_file (flags=0x7fffffffc520, closeit=1, locals=0x7ffff71f9880, globals=0x7ffff71f9880, start=257, filename=0x7ffff712f450, fp=0x55555555a590) at Python/pythonrun.c:1643 #34 _PyRun_SimpleFileObject (fp=fp@entry=0x55555555a590, filename=filename@entry=0x7ffff712f450, closeit=closeit@entry=1, flags=flags@entry=0x7fffffffc520) at Python/pythonrun.c:433 #35 0x00007ffff7ae6afc in _PyRun_AnyFileObject (fp=0x55555555a590, filename=filename@entry=0x7ffff712f450, closeit=closeit@entry=1, flags=flags@entry=0x7fffffffc520) at Python/pythonrun.c:78 #36 0x00007ffff7b0d72c in pymain_run_file_obj (skip_source_first_line=0, filename=0x7ffff712f450, program_name=0x7ffff7148880) at Modules/main.c:360 #37 pymain_run_file (config=0x7ffff7e66568 <_PyRuntime+77672>) at Modules/main.c:379 #38 pymain_run_python (exitcode=exitcode@entry=0x7fffffffc634) at Modules/main.c:629 #39 0x00007ffff7b0db66 in Py_RunMain () at Modules/main.c:709 #40 0x00007ffff7b0dd1a in pymain_main (args=0x7fffffffc660) at Modules/main.c:739 #41 Py_BytesMain (argc=<optimized out>, argv=<optimized out>) at Modules/main.c:763 #42 0x00007ffff7557cd0 in __libc_start_call_main (main=main@entry=0x555555555040 <main>, argc=argc@entry=2, argv=argv@entry=0x7fffffffc7e8) at ../sysdeps/nptl/libc_start_call_main.h:58 #43 0x00007ffff7557d8a in __libc_start_main_impl (main=0x555555555040 <main>, argc=2, argv=0x7fffffffc7e8, init=<optimized out>, fini=<optimized out>, rtld_fini=<optimized out>, stack_end=0x7fffffffc7d8) at ../csu/libc-start.c:360 #44 0x0000555555555075 in _start () ``` Issue URL : https://github.com/alsa-project/alsa-python/issues/9 Repository URL: https://github.com/alsa-project/alsa-python

1 0

[PATCH] ALSA: hda/realtek: Add quirks for ASUS Laptops using CS35L56
by Simon Trimmer 30 Mar '24

30 Mar '24

These ASUS laptops use the Realtek HDA codec combined with a number of CS35L56 amplifiers. The SSID of the GA403U matches a previous ASUS laptop - we can tell them apart because they use different codecs. Signed-off-by: Simon Trimmer <simont(a)opensource.cirrus.com> --- sound/pci/hda/patch_realtek.c | 53 ++++++++++++++++++++++++++++++++++- 1 file changed, 52 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index a17c36a36aa5..7d229f4a8aa6 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6875,11 +6875,38 @@ static void alc287_fixup_legion_16ithg6_speakers(struct hda_codec *cdc, const st comp_generic_fixup(cdc, action, "i2c", "CLSA0101", "-%s:00-cs35l41-hda.%d", 2); } +static void cs35l56_fixup_i2c_two(struct hda_codec *cdc, const struct hda_fixup *fix, int action) +{ + comp_generic_fixup(cdc, action, "i2c", "CSC3556", "-%s:00-cs35l56-hda.%d", 2); +} + +static void cs35l56_fixup_i2c_four(struct hda_codec *cdc, const struct hda_fixup *fix, int action) +{ + comp_generic_fixup(cdc, action, "i2c", "CSC3556", "-%s:00-cs35l56-hda.%d", 4); +} + +static void cs35l56_fixup_spi_two(struct hda_codec *cdc, const struct hda_fixup *fix, int action) +{ + comp_generic_fixup(cdc, action, "spi", "CSC3556", "-%s:00-cs35l56-hda.%d", 2); +} + static void cs35l56_fixup_spi_four(struct hda_codec *cdc, const struct hda_fixup *fix, int action) { comp_generic_fixup(cdc, action, "spi", "CSC3556", "-%s:00-cs35l56-hda.%d", 4); } +static void alc285_fixup_asus_ga403u(struct hda_codec *cdc, const struct hda_fixup *fix, int action) +{ + /* + * The same SSID has been re-used in different hardware, they have + * different codecs and the newer GA403U has a ALC285. + */ + if (cdc->core.vendor_id == 0x10ec0285) + cs35l56_fixup_i2c_two(cdc, fix, action); + else + alc_fixup_inv_dmic(cdc, fix, action); +} + static void tas2781_fixup_i2c(struct hda_codec *cdc, const struct hda_fixup *fix, int action) { @@ -7436,6 +7463,10 @@ enum { ALC256_FIXUP_ACER_SFG16_MICMUTE_LED, ALC256_FIXUP_HEADPHONE_AMP_VOL, ALC245_FIXUP_HP_SPECTRE_X360_EU0XXX, + ALC285_FIXUP_CS35L56_SPI_2, + ALC285_FIXUP_CS35L56_I2C_2, + ALC285_FIXUP_CS35L56_I2C_4, + ALC285_FIXUP_ASUS_GA403U, }; /* A special fixup for Lenovo C940 and Yoga Duet 7; @@ -9643,6 +9674,22 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc245_fixup_hp_spectre_x360_eu0xxx, }, + [ALC285_FIXUP_CS35L56_SPI_2] = { + .type = HDA_FIXUP_FUNC, + .v.func = cs35l56_fixup_spi_two, + }, + [ALC285_FIXUP_CS35L56_I2C_2] = { + .type = HDA_FIXUP_FUNC, + .v.func = cs35l56_fixup_i2c_two, + }, + [ALC285_FIXUP_CS35L56_I2C_4] = { + .type = HDA_FIXUP_FUNC, + .v.func = cs35l56_fixup_i2c_four, + }, + [ALC285_FIXUP_ASUS_GA403U] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc285_fixup_asus_ga403u, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -10096,7 +10143,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1a83, "ASUS UM5302LA", ALC294_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x1043, 0x1a8f, "ASUS UX582ZS", ALC245_FIXUP_CS35L41_SPI_2), SND_PCI_QUIRK(0x1043, 0x1b11, "ASUS UX431DA", ALC294_FIXUP_ASUS_COEF_1B), - SND_PCI_QUIRK(0x1043, 0x1b13, "Asus U41SV", ALC269_FIXUP_INV_DMIC), + SND_PCI_QUIRK(0x1043, 0x1b13, "ASUS U41SV/GA403U", ALC285_FIXUP_ASUS_GA403U), SND_PCI_QUIRK(0x1043, 0x1b93, "ASUS G614JVR/JIR", ALC245_FIXUP_CS35L41_SPI_2), SND_PCI_QUIRK(0x1043, 0x1bbd, "ASUS Z550MA", ALC255_FIXUP_ASUS_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1043, 0x1c03, "ASUS UM3406HA", ALC287_FIXUP_CS35L41_I2C_2), @@ -10104,6 +10151,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1c33, "ASUS UX5304MA", ALC245_FIXUP_CS35L41_SPI_2), SND_PCI_QUIRK(0x1043, 0x1c43, "ASUS UX8406MA", ALC245_FIXUP_CS35L41_SPI_2), SND_PCI_QUIRK(0x1043, 0x1c62, "ASUS GU603", ALC289_FIXUP_ASUS_GA401), + SND_PCI_QUIRK(0x1043, 0x1c63, "ASUS GU605M", ALC285_FIXUP_CS35L56_SPI_2), SND_PCI_QUIRK(0x1043, 0x1c92, "ASUS ROG Strix G15", ALC285_FIXUP_ASUS_G533Z_PINS), SND_PCI_QUIRK(0x1043, 0x1c9f, "ASUS G614JU/JV/JI", ALC285_FIXUP_ASUS_HEADSET_MIC), SND_PCI_QUIRK(0x1043, 0x1caf, "ASUS G634JY/JZ/JI/JG", ALC285_FIXUP_ASUS_SPI_REAR_SPEAKERS), @@ -10115,11 +10163,14 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1d42, "ASUS Zephyrus G14 2022", ALC289_FIXUP_ASUS_GA401), SND_PCI_QUIRK(0x1043, 0x1d4e, "ASUS TM420", ALC256_FIXUP_ASUS_HPE), SND_PCI_QUIRK(0x1043, 0x1da2, "ASUS UP6502ZA/ZD", ALC245_FIXUP_CS35L41_SPI_2), + SND_PCI_QUIRK(0x1043, 0x1df3, "ASUS UM5606", ALC285_FIXUP_CS35L56_I2C_4), SND_PCI_QUIRK(0x1043, 0x1e02, "ASUS UX3402ZA", ALC245_FIXUP_CS35L41_SPI_2), SND_PCI_QUIRK(0x1043, 0x1e11, "ASUS Zephyrus G15", ALC289_FIXUP_ASUS_GA502), SND_PCI_QUIRK(0x1043, 0x1e12, "ASUS UM3402", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x1043, 0x1e51, "ASUS Zephyrus M15", ALC294_FIXUP_ASUS_GU502_PINS), SND_PCI_QUIRK(0x1043, 0x1e5e, "ASUS ROG Strix G513", ALC294_FIXUP_ASUS_G513_PINS), + SND_PCI_QUIRK(0x1043, 0x1e63, "ASUS H7606W", ALC285_FIXUP_CS35L56_I2C_2), + SND_PCI_QUIRK(0x1043, 0x1e83, "ASUS GA605W", ALC285_FIXUP_CS35L56_I2C_2), SND_PCI_QUIRK(0x1043, 0x1e8e, "ASUS Zephyrus G15", ALC289_FIXUP_ASUS_GA401), SND_PCI_QUIRK(0x1043, 0x1ee2, "ASUS UM6702RA/RC", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x1043, 0x1c52, "ASUS Zephyrus G15 2022", ALC289_FIXUP_ASUS_GA401), -- 2.34.1

2 1

[PATCH v18 00/41] Introduce QC USB SND audio offloading support
by Wesley Cheng 30 Mar '24

30 Mar '24

Several Qualcomm based chipsets can support USB audio offloading to a dedicated audio DSP, which can take over issuing transfers to the USB host controller. The intention is to reduce the load on the main processors in the SoC, and allow them to be placed into lower power modes. There are several parts to this design: 1. Adding ASoC binding layer 2. Create a USB backend for Q6DSP 3. Introduce XHCI interrupter support 4. Create vendor ops for the USB SND driver USB | ASoC -------------------------------------------------------------------- | _________________________ | |sm8250 platform card | | |_________________________| | | | | ___V____ ____V____ | |Q6USB | |Q6AFE | | |"codec" | |"cpu" | | |________| |_________| | ^ ^ ^ | | |________| | ___V____ | | |SOC-USB | | ________ ________ | | | |USB SND |<--->|QC offld|<------------>|________| | |(card.c)| | |<---------- | |________| |________|___ | | | ^ ^ | | | ____________V_________ | | | | | |APR/GLINK | __ V_______________V_____ | | | |______________________| |USB SND (endpoint.c) | | | | ^ |_________________________| | | | | ^ | | | ___________V___________ | | | |->|audio DSP | ___________V_____________ | | |_______________________| |XHCI HCD |<- | |_________________________| | Adding ASoC binding layer: soc-usb: Intention is to treat a USB port similar to a headphone jack. The port is always present on the device, but cable/pin status can be enabled/disabled. Expose mechanisms for USB backend ASoC drivers to communicate with USB SND. Create a USB backend for Q6DSP: q6usb: Basic backend driver that will be responsible for maintaining the resources needed to initiate a playback stream using the Q6DSP. Will be the entity that checks to make sure the connected USB audio device supports the requested PCM format. If it does not, the PCM open call will fail, and userpsace ALSA can take action accordingly. Introduce XHCI interrupter support: XHCI HCD supports multiple interrupters, which allows for events to be routed to different event rings. This is determined by "Interrupter Target" field specified in Section "6.4.1.1 Normal TRB" of the XHCI specification. Events in the offloading case will be routed to an event ring that is assigned to the audio DSP. Create vendor ops for the USB SND driver: qc_audio_offload: This particular driver has several components associated with it: - QMI stream request handler - XHCI interrupter and resource management - audio DSP memory management When the audio DSP wants to enable a playback stream, the request is first received by the ASoC platform sound card. Depending on the selected route, ASoC will bring up the individual DAIs in the path. The Q6USB backend DAI will send an AFE port start command (with enabling the USB playback path), and the audio DSP will handle the request accordingly. Part of the AFE USB port start handling will have an exchange of control messages using the QMI protocol. The qc_audio_offload driver will populate the buffer information: - Event ring base address - EP transfer ring base address and pass it along to the audio DSP. All endpoint management will now be handed over to the DSP, and the main processor is not involved in transfers. Overall, implementing this feature will still expose separate sound card and PCM devices for both the platorm card and USB audio device: 0 [SM8250MTPWCD938]: sm8250 - SM8250-MTP-WCD9380-WSA8810-VA-D SM8250-MTP-WCD9380-WSA8810-VA-DMIC 1 [Audio ]: USB-Audio - USB Audio Generic USB Audio at usb-xhci-hcd.1.auto-1.4, high speed This is to ensure that userspace ALSA entities can decide which route to take when executing the audio playback. In the above, if card#1 is selected, then USB audio data will take the legacy path over the USB PCM drivers, etc... This feature was validated using: - tinymix: set/enable the multimedia path to route to USB backend - tinyplay: issue playback on platform card Changelog -------------------------------------------- Changes in v18: - Rebased to usb-next, which merged in part of the series. Removed these patches. - Reworked Kconfigs for the ASoC USB related components from QCOM Q6DSP drivers to keep dependencies in place for SoC USB and USB SND. - Removed the repurposing of the stop ep sync API into existing XHCI operations. This will be solely used by the XHCI sideband for now. Changes in v17: - Fixed an issue where one patch was squashed into another. - Re-added some kconfig checks for helpers exposed in USB SND for the soc usb driver, after running different kconfigs. Changes in v16: - Modified some code layer dependencies so that soc usb can be split as a separate module. - Split the kcontrols from ASoC QCOM common layer into a separate driver - Reworked SOC USB kcontrols for controlling card + pcm offload routing and status so that there are individual controls for card and pcm devices. - Added a kcontrol remove API in SOC USB to remove the controls on the fly. This required to add some kcontrol management to SOC USB. - Removed the disconnect work and workqueue for the QC USB offload as it is not required, since QMI interface driver ensures events are handled in its own WQ. Changes in v15: - Removed some already merged XHCI changes - Separated SOC USB driver from being always compiled into SOC core. Now configurable from kconfig. - Fixed up ASoC kcontrol naming to fit guidelines. - Removed some unnecessary dummy ifdefs. - Moved usb snd offload capable kcontrol to be initialized by the platform offloading driver. Changes in v14: - Cleaned up some USB SND related feedback: - Renamed SNDUSB OFFLD playback available --> USB offload capable card - Fixed locking while checking if stream is in use - Replaced some mutex pairs with guard(mutex) Changes in v13: - Pulled in secondary/primary interrupter rework from Mathias from: https://git.kernel.org/pub/scm/linux/kernel/git/mnyman/xhci.git/log/drivers… - Did some cleanup and commit message updates, and tested on current code base. - Added mutex locking to xhci sideband to help prevent any race conditions, esp. for when accessing shared references. - Addresed concerns from Hillf about gfp_flags and locking used in qc_usb_audio_offload. - Rebased onto usb-next Changes in v12: - Updated copyright year to 2024. Happy new years! - Fixed newline format on mixer offload driver. Changes in v11: - Modified QMI format structures to be const Changes in v10: - Added new mixer for exposing kcontrol for sound card created by USB SND. This allows for applications to know which platform sound card has offload support. Will return the card number. - Broke down and cleaned up some functions/APIs within qc_audio_offload driver. - Exported xhci_initialize_ring_info(), and modified XHCI makefile to allow for the XHCI sideband to exist as a module. - Reworked the jack registration and moved it to the QCOM platform card driver, ie sm8250. - Added an SOC USB API to fetch a standard component tag that can be appended to the platform sound card. Added this tag to sm8250 if any USB path exists within the DT node. - Moved kcontrols that existed in the Q6USB driver, and made it a bit more generic, so that naming can be standardized across solutions. SOC USB is now responsible for creation of these kcontrols. - Added a SOC USB RST document explaining some code flows and implementation details so that other vendors can utilize the framework. - Addressed a case where USB device connection events are lost if usb offload driver (qc_audio_offload) is not probed when everything else has been initialized, ie USB SND, SOC USB and ASoC sound card. Add a rediscover device call during module init, to ensure that connection events will be propagated. - Rebased to usb-next. Changes in v9: - Fixed the dt binding check issue with regards to num-hc-interrupters. Changes in v8: - Cleaned up snd_soc_usb_find_priv_data() based on Mark's feedback. Removed some of the duplicate looping code that was present on previous patches. Also renamed the API. - Integrated Mathias' suggestions on his new sideband changes: https://git.kernel.org/pub/scm/linux/kernel/git/mnyman/xhci.git/log/?h=feat… - Addressed some of Mathias' fixme tags, such as: - Resetting transfer ring dequeue/enqueue pointers - Issuing stop endpoint command during ep removal - Reset ERDP properly to first segment ring during interrupter removal. (this is currently just being cleared to 0, but should be pointing to a valid segment if controller is still running. Changes in v7: - Fixed dt check error for q6usb bindings - Updated q6usb property from qcom,usb-audio-intr-num --> qcom,usb-audio-intr-idx - Removed separate DWC3 HC interrupters num property, and place limits to XHCI one. - Modified xhci_ring_to_sgtable() to use assigned IOVA/DMA address to fetch pages, as it is not ensured event ring allocated is always done in the vmalloc range. Changes in v6: - Fixed limits and description on several DT bindings (XHCI and Q6USB) - Fixed patch subjects to follow other ALSA/ASoC notations. USB SND - Addressed devices which expose multiple audio (UAC) interfaces. These devices will create a single USB sound card with multiple audio streams, and receive multiple interface probe routines. QC offload was not properly considering cases with multiple probe calls. - Renamed offload module name and kconfig to fit within the SND domain. - Renamed attach/detach endpoint API to keep the hw_params notation. Changes in v5: - Removed some unnescessary files that were included - Fixed some typos mentioned - Addressed dt-binding issues and added hc-interrupters definition to usb-xhci.yaml XHCI: - Moved secondary skip events API to xhci-ring and updated implementation - Utilized existing XHCI APIs, such as inc_deq and xhci_update_erst_dequeue() USB SND - Renamed and reworked the APIs in "sound: usb: Export USB SND APIs for modules" patch to include suggestions to utilize snd_usb_hw_params/free and to avoid generic naming. - Added a resume_cb() op for completion sake. - Addressed some locking concerns with regards to when registering for platform hooks. - Added routine to disconnect all offloaded devices during module unbind. ASoC - Replaced individual PCM parameter arguments in snd_soc_usb_connect() with new snd_soc_usb_device structure to pass along PCM info. - Modified snd_jack set report to notify HEADPHONE event, as we do not support record path. Changes in v4: - Rebased to xhci/for-usb-next - Addressed some dt-bindings comments XHCI: - Pulled in latest changes from Mathias' feature_interrupters branch: https://git.kernel.org/pub/scm/linux/kernel/git/mnyman/xhci.git/log/?h=feat… - Fixed commit text and signage for the XHCI sideband/interrupter related changes - Added some logic to address the FIXME tags mentioned throughout the commits, such as handling multi segment rings and building the SGT, locking concerns, and ep cleanup operations. - Removed some fixme tags for conditions that may not be needed/addressed. - Repurposed the new endpoint stop sync API to be utilized in other places. - Fixed potential compile issue if XHCI sideband config is not defined. ASoC: - Added sound jack control into the Q6USB driver. Allows for userpsace to know when an offload capable device is connected. USB SND: - Avoided exporting _snd_pcm_hw_param_set based on Takashi's recommendation. - Split USB QMI packet header definitions into a separate commit. This is used to properly allow the QMI interface driver to parse and route QMI packets accordingly - Added a "depends on" entry when enabling QC audio offload to avoid compile time issues. Changes in v3: - Changed prefix from RFC to PATCH - Rebased entire series to usb-next - Updated copyright years XHCI: - Rebased changes on top of XHCI changes merged into usb-next, and only added changes that were still under discussion. - Added change to read in the "num-hc-interrupters" device property. ASoC: - qusb6 USB backend - Incorporated suggestions to fetch iommu information with existing APIs - Added two new sound kcontrols to fetch offload status and offload device selection. - offload status - will return the card and pcm device in use tinymix -D 0 get 1 --> 1, 0 (offload in progress on card#1 pcm#0) - device selection - set the card and pcm device to enable offload on. Ex.: tinymix -D 0 set 1 2 0 --> sets offload on card#2 pcm#0 (this should be the USB card) USB SND: - Fixed up some locking related concerns for registering platform ops. - Moved callbacks under the register_mutex, so that - Modified APIs to properly pass more information about the USB SND device, so that the Q6USB backend can build a device list/map, in order to monitor offload status and device selection. Changes in v2: XHCI: - Replaced XHCI and HCD changes with Mathias' XHCI interrupter changes in his tree: https://git.kernel.org/pub/scm/linux/kernel/git/mnyman/xhci.git/log/?h=feat… Adjustments made to Mathias' changes: - Created xhci-intr.h to export/expose interrupter APIs versus exposing xhci.h. Moved dependent structures to this file as well. (so clients can parse out information from "struct xhci_interrupter") - Added some basic locking when requesting interrupters. - Fixed up some sanity checks. - Removed clearing of the ERSTBA during freeing of the interrupter. (pending issue where SMMU fault occurs if DMA addr returned is 64b - TODO) - Clean up pending events in the XHCI secondary interrupter. While testing USB bus suspend, it was seen that on bus resume, the xHCI HC would run into a command timeout. - Added offloading APIs to xHCI to fetch transfer and event ring information. ASoC: - Modified soc-usb to allow for multiple USB port additions. For this to work, the USB offload driver has to have a reference to the USB backend by adding a "usb-soc-be" DT entry to the device saved into XHCI sysdev. - Created separate dt-bindings for defining USB_RX port. - Increased APR timeout to accommodate the situation where the AFE port start command could be delayed due to having to issue a USB bus resume while handling the QMI stream start command. USB SND: - Added a platform ops during usb_audio_suspend(). This allows for the USB offload driver to halt the audio stream when system enters PM suspend. This ensures the audio DSP is not issuing transfers on the USB bus. - Do not override platform ops if they are already populated. - Introduce a shared status variable between the USB offload and USB SND layers, to ensure that only one path is active at a time. If the USB bus is occupied, then userspace is notified that the path is busy. Mathias Nyman (2): xhci: add helper to stop endpoint and wait for completion xhci: sideband: add initial api to register a sideband entity Wesley Cheng (39): usb: host: xhci: Export enable and disable interrupter APIs usb: host: xhci: Repurpose event handler for skipping interrupter events xhci: export XHCI IMOD setting helper for interrupters usb: host: xhci-sideband: Expose a sideband interrupter enable API usb: host: xhci-mem: Cleanup pending secondary event ring events usb: host: xhci-mem: Allow for interrupter clients to choose specific index ASoC: Add SOC USB APIs for adding an USB backend ASoC: dt-bindings: qcom,q6dsp-lpass-ports: Add USB_RX port ASoC: qcom: qdsp6: Introduce USB AFE port to q6dsp ASoC: qdsp6: q6afe: Increase APR timeout ASoC: qcom: qdsp6: Add USB backend ASoC driver for Q6 ALSA: usb-audio: Introduce USB SND platform op callbacks ALSA: usb-audio: Export USB SND APIs for modules ALSA: usb-audio: Save UAC sample size information usb: dwc3: Specify maximum number of XHCI interrupters usb: host: xhci-plat: Set XHCI max interrupters if property is present ALSA: usb-audio: qcom: Add USB QMI definitions ALSA: usb-audio: qcom: Introduce QC USB SND offloading support ALSA: usb-audio: Check for support for requested audio format ASoC: usb: Add PCM format check API for USB backend ASoC: qcom: qdsp6: Ensure PCM format is supported by USB audio device ALSA: usb-audio: Prevent starting of audio stream if in use ALSA: usb-audio: Do not allow USB offload path if PCM device is in use ASoC: dt-bindings: Update example for enabling USB offload on SM8250 ALSA: usb-audio: qcom: Populate PCM and USB chip information ASoC: qcom: qdsp6: Add support to track available USB PCM devices ASoC: Introduce SND kcontrols to select sound card and PCM device ASoC: qcom: qdsp6: Add SOC USB offload select get/put callbacks ASoC: Introduce SND kcontrols to track USB offloading state ASoC: qcom: qdsp6: Add PCM ops to track current state ASoC: usb: Create SOC USB SND jack kcontrol ASoC: qcom: qdsp6: Add headphone jack for offload connection status ASoC: usb: Fetch ASoC sound card information ALSA: usb-audio: Add USB offloading capable kcontrol ALSA: usb-audio: Allow for rediscovery of connected USB SND devices ALSA: usb-audio: qcom: Use card and PCM index from QMI request ASoC: usb: Rediscover USB SND devices on USB port add ASoC: qcom: Populate SoC components string ASoC: doc: Add documentation for SOC USB .../bindings/sound/qcom,sm8250.yaml | 15 + Documentation/sound/soc/index.rst | 1 + Documentation/sound/soc/usb.rst | 611 ++++++ drivers/usb/dwc3/core.c | 12 + drivers/usb/dwc3/core.h | 2 + drivers/usb/dwc3/host.c | 3 + drivers/usb/host/Kconfig | 9 + drivers/usb/host/Makefile | 2 + drivers/usb/host/xhci-mem.c | 37 +- drivers/usb/host/xhci-plat.c | 2 + drivers/usb/host/xhci-ring.c | 50 +- drivers/usb/host/xhci-sideband.c | 439 ++++ drivers/usb/host/xhci.c | 48 +- drivers/usb/host/xhci.h | 18 +- .../sound/qcom,q6dsp-lpass-ports.h | 1 + include/linux/usb/xhci-sideband.h | 70 + include/sound/q6usboffload.h | 20 + include/sound/soc-usb.h | 182 ++ sound/soc/Kconfig | 10 + sound/soc/Makefile | 2 + sound/soc/qcom/Kconfig | 15 + sound/soc/qcom/Makefile | 2 + sound/soc/qcom/qdsp6/Makefile | 1 + sound/soc/qcom/qdsp6/q6afe-dai.c | 60 + sound/soc/qcom/qdsp6/q6afe.c | 193 +- sound/soc/qcom/qdsp6/q6afe.h | 36 +- sound/soc/qcom/qdsp6/q6dsp-lpass-ports.c | 23 + sound/soc/qcom/qdsp6/q6dsp-lpass-ports.h | 1 + sound/soc/qcom/qdsp6/q6routing.c | 9 + sound/soc/qcom/qdsp6/q6usb.c | 408 ++++ sound/soc/qcom/sm8250.c | 15 +- sound/soc/qcom/usb_offload_utils.c | 51 + sound/soc/qcom/usb_offload_utils.h | 26 + sound/soc/soc-usb.c | 660 ++++++ sound/usb/Kconfig | 25 + sound/usb/Makefile | 2 +- sound/usb/card.c | 109 + sound/usb/card.h | 15 + sound/usb/endpoint.c | 1 + sound/usb/format.c | 1 + sound/usb/helper.c | 1 + sound/usb/pcm.c | 104 +- sound/usb/pcm.h | 11 + sound/usb/qcom/Makefile | 6 + sound/usb/qcom/mixer_usb_offload.c | 65 + sound/usb/qcom/mixer_usb_offload.h | 17 + sound/usb/qcom/qc_audio_offload.c | 1914 +++++++++++++++++ sound/usb/qcom/usb_audio_qmi_v01.c | 892 ++++++++ sound/usb/qcom/usb_audio_qmi_v01.h | 162 ++ 49 files changed, 6306 insertions(+), 53 deletions(-) create mode 100644 Documentation/sound/soc/usb.rst create mode 100644 drivers/usb/host/xhci-sideband.c create mode 100644 include/linux/usb/xhci-sideband.h create mode 100644 include/sound/q6usboffload.h create mode 100644 include/sound/soc-usb.h create mode 100644 sound/soc/qcom/qdsp6/q6usb.c create mode 100644 sound/soc/qcom/usb_offload_utils.c create mode 100644 sound/soc/qcom/usb_offload_utils.h create mode 100644 sound/soc/soc-usb.c create mode 100644 sound/usb/qcom/Makefile create mode 100644 sound/usb/qcom/mixer_usb_offload.c create mode 100644 sound/usb/qcom/mixer_usb_offload.h create mode 100644 sound/usb/qcom/qc_audio_offload.c create mode 100644 sound/usb/qcom/usb_audio_qmi_v01.c create mode 100644 sound/usb/qcom/usb_audio_qmi_v01.h

2 47

Sending 0xFx events causes libasound to segfault
by GitHub issues - edited 29 Mar '24

29 Mar '24

alsa-project/alsa-lib issue #392 was edited from human222: ALSA 1.1.9. This issue breaks DOSBox-X MIDI playback. Discovered through this thread: https://github.com/joncampbell123/dosbox-x/issues/4873 In particular, this comment: https://github.com/joncampbell123/dosbox-x/issues/4873#issuecomment-2026694… Issue URL : https://github.com/alsa-project/alsa-lib/issues/392 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

Sending 0xFx events causes libasound to segfault
by GitHub issues - opened 29 Mar '24

29 Mar '24

alsa-project/alsa-lib issue #392 was opened from human222: ALSA 1.1.9. This issue breaks DOSBox-X MIDI playback. Discovered through this thread: https://github.com/joncampbell123/dosbox-x/issues/4873 Issue URL : https://github.com/alsa-project/alsa-lib/issues/392 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH V2] ASoC: amd: acp: fix for acp_init function error handling
by Vijendar Mukunda 29 Mar '24

29 Mar '24

If acp_init() fails, acp pci driver probe should return error. Add acp_init() function return value check logic. Fixes: e61b415515d3 ("ASoC: amd: acp: refactor the acp init and de-init sequence") Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- changes from v1->v2: - handle clean up sequence instead of returning error directly sound/soc/amd/acp/acp-pci.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/amd/acp/acp-pci.c b/sound/soc/amd/acp/acp-pci.c index 440b91d4f261..5f35b90eab8d 100644 --- a/sound/soc/amd/acp/acp-pci.c +++ b/sound/soc/amd/acp/acp-pci.c @@ -115,7 +115,10 @@ static int acp_pci_probe(struct pci_dev *pci, const struct pci_device_id *pci_id goto unregister_dmic_dev; } - acp_init(chip); + ret = acp_init(chip); + if (ret) + goto unregister_dmic_dev; + res = devm_kcalloc(&pci->dev, num_res, sizeof(struct resource), GFP_KERNEL); if (!res) { ret = -ENOMEM; -- 2.34.1

2 1

alsa-info uploads re periodic sound failures on P16 w/ Mint
by Robert Hoskin 29 Mar '24

29 Mar '24

Greetings! The alsa-info output links below is from a Lenovo P16 Gen2, running Linux Mint 21.3, with an intermittent sound problem. The problem is that: Sound will be available from either built in speakers or headphones, perhaps for several days after a startup, but at some point will fail, with both Headphones (via jack) and internal Speakers disappearing from the "sound" app, and "Dummy Output" appearing in their place. If it's possible to reproduce this failure, I haven't figured out how. It happens when it happens, usually after a few days of uptime. I've found that sound can be restored with "sudo alsa force-reload", so I do have a workaround. *** For before-and-after comparison purposes... *** This report was taken when "Dummy Output" was shown in the Sound app: http://alsa-project.org/db/?f=4ec46710d176deb77fcdd950520eaafc4c9db1c3 *** This report was taken after a "sudo alsa force-reload", which restored both headphones and speakers to the Sound app" http://alsa-project.org/db/?f=3832b192eaa22ea4d51b0e6073f389318b983356 *** In the "System Reports" app, there are no crash reports shown. BUT... In the system logs app, I did find some lines in the Hardware section that may be relevant: 09:41:50 kernel: sof-audio-pci-intel-tgl 0000:00:1f.3: ASoC: error at snd_soc_pcm_component_prepare on 0000:00:1f.3: -22 09:41:50 kernel: sof-audio-pci-intel-tgl 0000:00:1f.3: ASoC: error at snd_soc_pcm_component_prepare on 0000:00:1f.3: -22 09:41:50 kernel: sof-audio-pci-intel-tgl 0000:00:1f.3: error: set pcm hw_params after resume 09:41:50 kernel: sof-audio-pci-intel-tgl 0000:00:1f.3: HW params ipc failed for stream 1 09:41:50 kernel: sof-audio-pci-intel-tgl 0000:00:1f.3: ipc tx error for 0x60010000 (msg/reply size: 108/20): -22 I really have no idea whether I'm submitting this in the right place or not, whether this is a kernel problem, a driver problem, a config problem, or whatever. I can tell you that this Mint install is new on a new machine, otherwise well-behaved, with subsequent updates that have not changed this sound behavior. Hope this is useful to you, love to have it fixed. Thanks! - Bob -- Robert Hoskin rhoskin(a)fastmail.org

1 1

[PATCH 00/22] virtio: store owner from modules with register_virtio_driver()
by Krzysztof Kozlowski 29 Mar '24

29 Mar '24

Merging ======= All further patches depend on the first virtio patch, therefore please ack and this should go via one tree: virtio? Description =========== Modules registering driver with register_virtio_driver() often forget to set .owner field. Solve the problem by moving this task away from the drivers to the core amba bus code, just like we did for platform_driver in commit 9447057eaff8 ("platform_device: use a macro instead of platform_driver_register"). Best regards, Krzysztof Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- Krzysztof Kozlowski (22): virtio: store owner from modules with register_virtio_driver() um: virt-pci: drop owner assignment virtio_blk: drop owner assignment bluetooth: virtio: drop owner assignment hwrng: virtio: drop owner assignment virtio_console: drop owner assignment crypto: virtio - drop owner assignment firmware: arm_scmi: virtio: drop owner assignment gpio: virtio: drop owner assignment drm/virtio: drop owner assignment iommu: virtio: drop owner assignment misc: nsm: drop owner assignment net: caif: virtio: drop owner assignment net: virtio: drop owner assignment net: 9p: virtio: drop owner assignment net: vmw_vsock: virtio: drop owner assignment wireless: mac80211_hwsim: drop owner assignment nvdimm: virtio_pmem: drop owner assignment rpmsg: virtio: drop owner assignment scsi: virtio: drop owner assignment fuse: virtio: drop owner assignment sound: virtio: drop owner assignment Documentation/driver-api/virtio/writing_virtio_drivers.rst | 1 - arch/um/drivers/virt-pci.c | 1 - drivers/block/virtio_blk.c | 1 - drivers/bluetooth/virtio_bt.c | 1 - drivers/char/hw_random/virtio-rng.c | 1 - drivers/char/virtio_console.c | 2 -- drivers/crypto/virtio/virtio_crypto_core.c | 1 - drivers/firmware/arm_scmi/virtio.c | 1 - drivers/gpio/gpio-virtio.c | 1 - drivers/gpu/drm/virtio/virtgpu_drv.c | 1 - drivers/iommu/virtio-iommu.c | 1 - drivers/misc/nsm.c | 1 - drivers/net/caif/caif_virtio.c | 1 - drivers/net/virtio_net.c | 1 - drivers/net/wireless/virtual/mac80211_hwsim.c | 1 - drivers/nvdimm/virtio_pmem.c | 1 - drivers/rpmsg/virtio_rpmsg_bus.c | 1 - drivers/scsi/virtio_scsi.c | 1 - drivers/virtio/virtio.c | 6 ++++-- fs/fuse/virtio_fs.c | 1 - include/linux/virtio.h | 7 +++++-- net/9p/trans_virtio.c | 1 - net/vmw_vsock/virtio_transport.c | 1 - sound/virtio/virtio_card.c | 1 - 24 files changed, 9 insertions(+), 27 deletions(-) --- base-commit: 7fdcff3312e16ba8d1419f8a18f465c5cc235ecf change-id: 20240327-module-owner-virtio-546763b3ca22 Best regards, -- Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org>

10 38

[PATCH 00/13] ASoC: ti: davinci-i2s: Add features to McBSP driver
by Bastien Curutchet 29 Mar '24

29 Mar '24

This series aims to add some features to McBSP driver. Convert bindings from .txt to .yaml. Add possibility to use an external clock as sample rate generator's input. Add handling of new formats (TDM, S24_LE, BP_FC). Add optional properties in DT: - ti,enable-sync-err : Enable the detection of unexpected frame pulses - ti,disable-free-run : Disable the free-running mode where McBSP drives serial clocks during emulation halt - ti,drive-dx : Outputs a chosen pattern on DX pin during capture streams. This has been tested on a platform designed off of the DAVINCI/OMAP-L138 connected to 3 daisy-chained AD7767. An external clock drives the sample rate generator through the CLKS pin. The hardware I have only allowed me to test acquisition side of McBSP. It is connected to a 6 channels TDM and acts as Bit clock provider and Frame clock consumer. Bastien Curutchet (13): ASoC: dt-bindings: davinci-mcbsp: convert McBSP bindings to yaml schema ASoC: dt-bindings: davinci-mcbsp: Add new properties ASoC: ti: davinci-i2s: Remove the unused clk_input_pin attribute ASoC: ti: davinci-i2s: Replace dev_err with dev_err_probe ASoC: ti: davinci-i2s: Use external clock to drive sample rate generator ASoC: ti: davinci-i2s: Delete unnecessary assignment ASoC: ti: davinci-i2s: Add TDM support ASoC: ti: davinci-i2s: Add handling of BP_FC format ASoC: ti: davinci-i2s: Enable unexpected frame pulses detection ASoC: ti: davinci-i2s: Make free-running mode optional ASoC: ti: davinci-i2s: Add S24_LE to supported formats ASoC: dt-bindings: davinic-mcbsp: Add the 'ti,drive-dx' property ASoC: ti: davinci-i2s: Opitonally drive DX pin during capture streams .../bindings/sound/davinci-mcbsp.txt | 50 --- .../bindings/sound/davinci-mcbsp.yaml | 119 +++++++ include/linux/platform_data/davinci_asp.h | 15 - sound/soc/ti/davinci-i2s.c | 333 ++++++++++++++---- 4 files changed, 376 insertions(+), 141 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/davinci-mcbsp.txt create mode 100644 Documentation/devicetree/bindings/sound/davinci-mcbsp.yaml -- 2.43.2

5 38

[PATCH AUTOSEL 5.15 13/34] ALSA: firewire-lib: handle quirk to calculate payload quadlets as data block counter
by Sasha Levin 29 Mar '24

29 Mar '24

From: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> [ Upstream commit 4a486439d2ca85752c46711f373b6ddc107bb35d ] Miglia Harmony Audio (OXFW970) has a quirk to put the number of accumulated quadlets in CIP payload into the dbc field of CIP header. This commit handles the quirk in the packet processing layer. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> Link: https://lore.kernel.org/r/20240218074128.95210-4-o-takashi@sakamocchi.jp Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/firewire/amdtp-stream.c | 12 ++++++++---- sound/firewire/amdtp-stream.h | 4 ++++ 2 files changed, 12 insertions(+), 4 deletions(-) diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c index f8b644cb9157a..8753125683692 100644 --- a/sound/firewire/amdtp-stream.c +++ b/sound/firewire/amdtp-stream.c @@ -771,10 +771,14 @@ static int check_cip_header(struct amdtp_stream *s, const __be32 *buf, } else { unsigned int dbc_interval; - if (*data_blocks > 0 && s->ctx_data.tx.dbc_interval > 0) - dbc_interval = s->ctx_data.tx.dbc_interval; - else - dbc_interval = *data_blocks; + if (!(s->flags & CIP_DBC_IS_PAYLOAD_QUADLETS)) { + if (*data_blocks > 0 && s->ctx_data.tx.dbc_interval > 0) + dbc_interval = s->ctx_data.tx.dbc_interval; + else + dbc_interval = *data_blocks; + } else { + dbc_interval = payload_length / sizeof(__be32); + } lost = dbc != ((*data_block_counter + dbc_interval) & 0xff); } diff --git a/sound/firewire/amdtp-stream.h b/sound/firewire/amdtp-stream.h index 1f957c946c956..cf9ab347277f2 100644 --- a/sound/firewire/amdtp-stream.h +++ b/sound/firewire/amdtp-stream.h @@ -37,6 +37,9 @@ * the value of current SYT_INTERVAL; e.g. initial value is not zero. * @CIP_UNAWARE_SYT: For outgoing packet, the value in SYT field of CIP is 0xffff. * For incoming packet, the value in SYT field of CIP is not handled. + * @CIP_DBC_IS_PAYLOAD_QUADLETS: Available for incoming packet, and only effective with + * CIP_DBC_IS_END_EVENT flag. The value of dbc field is the number of accumulated quadlets + * in CIP payload, instead of the number of accumulated data blocks. */ enum cip_flags { CIP_NONBLOCKING = 0x00, @@ -51,6 +54,7 @@ enum cip_flags { CIP_NO_HEADER = 0x100, CIP_UNALIGHED_DBC = 0x200, CIP_UNAWARE_SYT = 0x400, + CIP_DBC_IS_PAYLOAD_QUADLETS = 0x800, }; /** -- 2.43.0

1 0

[PATCH AUTOSEL 6.1 20/52] ALSA: firewire-lib: handle quirk to calculate payload quadlets as data block counter
by Sasha Levin 29 Mar '24

29 Mar '24

From: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> [ Upstream commit 4a486439d2ca85752c46711f373b6ddc107bb35d ] Miglia Harmony Audio (OXFW970) has a quirk to put the number of accumulated quadlets in CIP payload into the dbc field of CIP header. This commit handles the quirk in the packet processing layer. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> Link: https://lore.kernel.org/r/20240218074128.95210-4-o-takashi@sakamocchi.jp Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/firewire/amdtp-stream.c | 12 ++++++++---- sound/firewire/amdtp-stream.h | 4 ++++ 2 files changed, 12 insertions(+), 4 deletions(-) diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c index f8b644cb9157a..8753125683692 100644 --- a/sound/firewire/amdtp-stream.c +++ b/sound/firewire/amdtp-stream.c @@ -771,10 +771,14 @@ static int check_cip_header(struct amdtp_stream *s, const __be32 *buf, } else { unsigned int dbc_interval; - if (*data_blocks > 0 && s->ctx_data.tx.dbc_interval > 0) - dbc_interval = s->ctx_data.tx.dbc_interval; - else - dbc_interval = *data_blocks; + if (!(s->flags & CIP_DBC_IS_PAYLOAD_QUADLETS)) { + if (*data_blocks > 0 && s->ctx_data.tx.dbc_interval > 0) + dbc_interval = s->ctx_data.tx.dbc_interval; + else + dbc_interval = *data_blocks; + } else { + dbc_interval = payload_length / sizeof(__be32); + } lost = dbc != ((*data_block_counter + dbc_interval) & 0xff); } diff --git a/sound/firewire/amdtp-stream.h b/sound/firewire/amdtp-stream.h index 1f957c946c956..cf9ab347277f2 100644 --- a/sound/firewire/amdtp-stream.h +++ b/sound/firewire/amdtp-stream.h @@ -37,6 +37,9 @@ * the value of current SYT_INTERVAL; e.g. initial value is not zero. * @CIP_UNAWARE_SYT: For outgoing packet, the value in SYT field of CIP is 0xffff. * For incoming packet, the value in SYT field of CIP is not handled. + * @CIP_DBC_IS_PAYLOAD_QUADLETS: Available for incoming packet, and only effective with + * CIP_DBC_IS_END_EVENT flag. The value of dbc field is the number of accumulated quadlets + * in CIP payload, instead of the number of accumulated data blocks. */ enum cip_flags { CIP_NONBLOCKING = 0x00, @@ -51,6 +54,7 @@ enum cip_flags { CIP_NO_HEADER = 0x100, CIP_UNALIGHED_DBC = 0x200, CIP_UNAWARE_SYT = 0x400, + CIP_DBC_IS_PAYLOAD_QUADLETS = 0x800, }; /** -- 2.43.0

1 0

[PATCH AUTOSEL 6.1 13/52] ASoC: Intel: common: DMI remap for rebranded Intel NUC M15 (LAPRC710) laptops
by Sasha Levin 29 Mar '24

29 Mar '24

From: mosomate <mosomate(a)gmail.com> [ Upstream commit c13e03126a5be90781084437689724254c8226e1 ] Added DMI quirk to handle the rebranded variants of Intel NUC M15 (LAPRC710) laptops. The DMI matching is based on motherboard attributes. Link: https://github.com/thesofproject/linux/issues/4218 Signed-off-by: Máté Mosonyi <mosomate(a)gmail.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20240208165545.93811-20-pierre-louis.bossart@linu… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- drivers/soundwire/dmi-quirks.c | 8 ++++++++ sound/soc/intel/boards/sof_sdw.c | 11 +++++++++++ 2 files changed, 19 insertions(+) diff --git a/drivers/soundwire/dmi-quirks.c b/drivers/soundwire/dmi-quirks.c index 9ebdd0cd0b1cf..91ab97a456fa9 100644 --- a/drivers/soundwire/dmi-quirks.c +++ b/drivers/soundwire/dmi-quirks.c @@ -130,6 +130,14 @@ static const struct dmi_system_id adr_remap_quirk_table[] = { }, .driver_data = (void *)intel_rooks_county, }, + { + /* quirk used for NUC15 LAPRC710 skew */ + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "Intel Corporation"), + DMI_MATCH(DMI_BOARD_NAME, "LAPRC710"), + }, + .driver_data = (void *)intel_rooks_county, + }, { .matches = { DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"), diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index 985012f2003e2..d1e6e4208c376 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -224,6 +224,17 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { SOF_SDW_PCH_DMIC | RT711_JD2_100K), }, + { + /* NUC15 LAPRC710 skews */ + .callback = sof_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "Intel Corporation"), + DMI_MATCH(DMI_BOARD_NAME, "LAPRC710"), + }, + .driver_data = (void *)(SOF_SDW_TGL_HDMI | + SOF_SDW_PCH_DMIC | + RT711_JD2_100K), + }, /* TigerLake-SDCA devices */ { .callback = sof_sdw_quirk_cb, -- 2.43.0

1 0

[PATCH AUTOSEL 6.6 33/75] ASoC: Intel: avs: Populate board selection with new I2S entries
by Sasha Levin 29 Mar '24

29 Mar '24

From: Cezary Rojewski <cezary.rojewski(a)intel.com> [ Upstream commit 5b417fe0cded0b5917683398e6519aae8045cd40 ] Update board selection with tables specifying supported I2S configurations. DMIC/HDAudio board selection require no update as dmic/hdaudio machine boards are generic and not tied to any specific codec. Reviewed-by: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> Signed-off-by: Cezary Rojewski <cezary.rojewski(a)intel.com> Link: https://msgid.link/r/20240220115035.770402-11-cezary.rojewski@intel.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/avs/board_selection.c | 85 +++++++++++++++++++++++++++ 1 file changed, 85 insertions(+) diff --git a/sound/soc/intel/avs/board_selection.c b/sound/soc/intel/avs/board_selection.c index 59a13feec57b2..0db5e530a8de2 100644 --- a/sound/soc/intel/avs/board_selection.c +++ b/sound/soc/intel/avs/board_selection.c @@ -227,6 +227,82 @@ static struct snd_soc_acpi_mach avs_gml_i2s_machines[] = { {}, }; +static struct snd_soc_acpi_mach avs_cnl_i2s_machines[] = { + { + .id = "INT34C2", + .drv_name = "avs_rt274", + .mach_params = { + .i2s_link_mask = AVS_SSP(0), + }, + .tplg_filename = "rt274-tplg.bin", + }, + { + .id = "10EC5682", + .drv_name = "avs_rt5682", + .mach_params = { + .i2s_link_mask = AVS_SSP(1), + }, + .tplg_filename = "rt5682-tplg.bin", + }, + {}, +}; + +static struct snd_soc_acpi_mach avs_icl_i2s_machines[] = { + { + .id = "INT343A", + .drv_name = "avs_rt298", + .mach_params = { + .i2s_link_mask = AVS_SSP(0), + }, + .tplg_filename = "rt298-tplg.bin", + }, + { + .id = "INT34C2", + .drv_name = "avs_rt274", + .mach_params = { + .i2s_link_mask = AVS_SSP(0), + }, + .tplg_filename = "rt274-tplg.bin", + }, + {}, +}; + +static struct snd_soc_acpi_mach avs_tgl_i2s_machines[] = { + { + .id = "INT34C2", + .drv_name = "avs_rt274", + .mach_params = { + .i2s_link_mask = AVS_SSP(0), + }, + .tplg_filename = "rt274-tplg.bin", + }, + { + .id = "10EC0298", + .drv_name = "avs_rt298", + .mach_params = { + .i2s_link_mask = AVS_SSP(0), + }, + .tplg_filename = "rt298-tplg.bin", + }, + { + .id = "10EC1308", + .drv_name = "avs_rt1308", + .mach_params = { + .i2s_link_mask = AVS_SSP(1), + }, + .tplg_filename = "rt1308-tplg.bin", + }, + { + .id = "ESSX8336", + .drv_name = "avs_es8336", + .mach_params = { + .i2s_link_mask = AVS_SSP(0), + }, + .tplg_filename = "es8336-tplg.bin", + }, + {}, +}; + static struct snd_soc_acpi_mach avs_test_i2s_machines[] = { { .drv_name = "avs_i2s_test", @@ -287,6 +363,15 @@ static const struct avs_acpi_boards i2s_boards[] = { AVS_MACH_ENTRY(HDA_KBL_LP, avs_kbl_i2s_machines), AVS_MACH_ENTRY(HDA_APL, avs_apl_i2s_machines), AVS_MACH_ENTRY(HDA_GML, avs_gml_i2s_machines), + AVS_MACH_ENTRY(HDA_CNL_LP, avs_cnl_i2s_machines), + AVS_MACH_ENTRY(HDA_CNL_H, avs_cnl_i2s_machines), + AVS_MACH_ENTRY(HDA_CML_LP, avs_cnl_i2s_machines), + AVS_MACH_ENTRY(HDA_ICL_LP, avs_icl_i2s_machines), + AVS_MACH_ENTRY(HDA_TGL_LP, avs_tgl_i2s_machines), + AVS_MACH_ENTRY(HDA_EHL_0, avs_tgl_i2s_machines), + AVS_MACH_ENTRY(HDA_ADL_P, avs_tgl_i2s_machines), + AVS_MACH_ENTRY(HDA_RPL_P_0, avs_tgl_i2s_machines), + AVS_MACH_ENTRY(HDA_RPL_M, avs_tgl_i2s_machines), {}, }; -- 2.43.0

1 0

[PATCH AUTOSEL 6.6 32/75] ALSA: firewire-lib: handle quirk to calculate payload quadlets as data block counter
by Sasha Levin 29 Mar '24

29 Mar '24

From: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> [ Upstream commit 4a486439d2ca85752c46711f373b6ddc107bb35d ] Miglia Harmony Audio (OXFW970) has a quirk to put the number of accumulated quadlets in CIP payload into the dbc field of CIP header. This commit handles the quirk in the packet processing layer. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> Link: https://lore.kernel.org/r/20240218074128.95210-4-o-takashi@sakamocchi.jp Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/firewire/amdtp-stream.c | 12 ++++++++---- sound/firewire/amdtp-stream.h | 4 ++++ 2 files changed, 12 insertions(+), 4 deletions(-) diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c index 7be17bca257f0..c9f153f85ae6b 100644 --- a/sound/firewire/amdtp-stream.c +++ b/sound/firewire/amdtp-stream.c @@ -773,10 +773,14 @@ static int check_cip_header(struct amdtp_stream *s, const __be32 *buf, } else { unsigned int dbc_interval; - if (*data_blocks > 0 && s->ctx_data.tx.dbc_interval > 0) - dbc_interval = s->ctx_data.tx.dbc_interval; - else - dbc_interval = *data_blocks; + if (!(s->flags & CIP_DBC_IS_PAYLOAD_QUADLETS)) { + if (*data_blocks > 0 && s->ctx_data.tx.dbc_interval > 0) + dbc_interval = s->ctx_data.tx.dbc_interval; + else + dbc_interval = *data_blocks; + } else { + dbc_interval = payload_length / sizeof(__be32); + } lost = dbc != ((*data_block_counter + dbc_interval) & 0xff); } diff --git a/sound/firewire/amdtp-stream.h b/sound/firewire/amdtp-stream.h index b7ff44751ab93..a1ed2e80f91a7 100644 --- a/sound/firewire/amdtp-stream.h +++ b/sound/firewire/amdtp-stream.h @@ -37,6 +37,9 @@ * the value of current SYT_INTERVAL; e.g. initial value is not zero. * @CIP_UNAWARE_SYT: For outgoing packet, the value in SYT field of CIP is 0xffff. * For incoming packet, the value in SYT field of CIP is not handled. + * @CIP_DBC_IS_PAYLOAD_QUADLETS: Available for incoming packet, and only effective with + * CIP_DBC_IS_END_EVENT flag. The value of dbc field is the number of accumulated quadlets + * in CIP payload, instead of the number of accumulated data blocks. */ enum cip_flags { CIP_NONBLOCKING = 0x00, @@ -51,6 +54,7 @@ enum cip_flags { CIP_NO_HEADER = 0x100, CIP_UNALIGHED_DBC = 0x200, CIP_UNAWARE_SYT = 0x400, + CIP_DBC_IS_PAYLOAD_QUADLETS = 0x800, }; /** -- 2.43.0

1 0

[PATCH AUTOSEL 6.6 17/75] ASoC: Intel: common: DMI remap for rebranded Intel NUC M15 (LAPRC710) laptops
by Sasha Levin 29 Mar '24

29 Mar '24

From: mosomate <mosomate(a)gmail.com> [ Upstream commit c13e03126a5be90781084437689724254c8226e1 ] Added DMI quirk to handle the rebranded variants of Intel NUC M15 (LAPRC710) laptops. The DMI matching is based on motherboard attributes. Link: https://github.com/thesofproject/linux/issues/4218 Signed-off-by: Máté Mosonyi <mosomate(a)gmail.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20240208165545.93811-20-pierre-louis.bossart@linu… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- drivers/soundwire/dmi-quirks.c | 8 ++++++++ sound/soc/intel/boards/sof_sdw.c | 11 +++++++++++ 2 files changed, 19 insertions(+) diff --git a/drivers/soundwire/dmi-quirks.c b/drivers/soundwire/dmi-quirks.c index 9ebdd0cd0b1cf..91ab97a456fa9 100644 --- a/drivers/soundwire/dmi-quirks.c +++ b/drivers/soundwire/dmi-quirks.c @@ -130,6 +130,14 @@ static const struct dmi_system_id adr_remap_quirk_table[] = { }, .driver_data = (void *)intel_rooks_county, }, + { + /* quirk used for NUC15 LAPRC710 skew */ + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "Intel Corporation"), + DMI_MATCH(DMI_BOARD_NAME, "LAPRC710"), + }, + .driver_data = (void *)intel_rooks_county, + }, { .matches = { DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"), diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index 9ed572141fe5b..0ea7812125fee 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -243,6 +243,17 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { SOF_SDW_PCH_DMIC | RT711_JD2_100K), }, + { + /* NUC15 LAPRC710 skews */ + .callback = sof_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "Intel Corporation"), + DMI_MATCH(DMI_BOARD_NAME, "LAPRC710"), + }, + .driver_data = (void *)(SOF_SDW_TGL_HDMI | + SOF_SDW_PCH_DMIC | + RT711_JD2_100K), + }, /* TigerLake-SDCA devices */ { .callback = sof_sdw_quirk_cb, -- 2.43.0

1 0

[PATCH AUTOSEL 6.8 48/98] ASoC: Intel: avs: Populate board selection with new I2S entries
by Sasha Levin 29 Mar '24

29 Mar '24

From: Cezary Rojewski <cezary.rojewski(a)intel.com> [ Upstream commit 5b417fe0cded0b5917683398e6519aae8045cd40 ] Update board selection with tables specifying supported I2S configurations. DMIC/HDAudio board selection require no update as dmic/hdaudio machine boards are generic and not tied to any specific codec. Reviewed-by: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> Signed-off-by: Cezary Rojewski <cezary.rojewski(a)intel.com> Link: https://msgid.link/r/20240220115035.770402-11-cezary.rojewski@intel.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/avs/board_selection.c | 85 +++++++++++++++++++++++++++ 1 file changed, 85 insertions(+) diff --git a/sound/soc/intel/avs/board_selection.c b/sound/soc/intel/avs/board_selection.c index 8e91eece992db..8360ce557401c 100644 --- a/sound/soc/intel/avs/board_selection.c +++ b/sound/soc/intel/avs/board_selection.c @@ -236,6 +236,82 @@ static struct snd_soc_acpi_mach avs_gml_i2s_machines[] = { {}, }; +static struct snd_soc_acpi_mach avs_cnl_i2s_machines[] = { + { + .id = "INT34C2", + .drv_name = "avs_rt274", + .mach_params = { + .i2s_link_mask = AVS_SSP(0), + }, + .tplg_filename = "rt274-tplg.bin", + }, + { + .id = "10EC5682", + .drv_name = "avs_rt5682", + .mach_params = { + .i2s_link_mask = AVS_SSP(1), + }, + .tplg_filename = "rt5682-tplg.bin", + }, + {}, +}; + +static struct snd_soc_acpi_mach avs_icl_i2s_machines[] = { + { + .id = "INT343A", + .drv_name = "avs_rt298", + .mach_params = { + .i2s_link_mask = AVS_SSP(0), + }, + .tplg_filename = "rt298-tplg.bin", + }, + { + .id = "INT34C2", + .drv_name = "avs_rt274", + .mach_params = { + .i2s_link_mask = AVS_SSP(0), + }, + .tplg_filename = "rt274-tplg.bin", + }, + {}, +}; + +static struct snd_soc_acpi_mach avs_tgl_i2s_machines[] = { + { + .id = "INT34C2", + .drv_name = "avs_rt274", + .mach_params = { + .i2s_link_mask = AVS_SSP(0), + }, + .tplg_filename = "rt274-tplg.bin", + }, + { + .id = "10EC0298", + .drv_name = "avs_rt298", + .mach_params = { + .i2s_link_mask = AVS_SSP(0), + }, + .tplg_filename = "rt298-tplg.bin", + }, + { + .id = "10EC1308", + .drv_name = "avs_rt1308", + .mach_params = { + .i2s_link_mask = AVS_SSP(1), + }, + .tplg_filename = "rt1308-tplg.bin", + }, + { + .id = "ESSX8336", + .drv_name = "avs_es8336", + .mach_params = { + .i2s_link_mask = AVS_SSP(0), + }, + .tplg_filename = "es8336-tplg.bin", + }, + {}, +}; + static struct snd_soc_acpi_mach avs_test_i2s_machines[] = { { .drv_name = "avs_i2s_test", @@ -296,6 +372,15 @@ static const struct avs_acpi_boards i2s_boards[] = { AVS_MACH_ENTRY(HDA_KBL_LP, avs_kbl_i2s_machines), AVS_MACH_ENTRY(HDA_APL, avs_apl_i2s_machines), AVS_MACH_ENTRY(HDA_GML, avs_gml_i2s_machines), + AVS_MACH_ENTRY(HDA_CNL_LP, avs_cnl_i2s_machines), + AVS_MACH_ENTRY(HDA_CNL_H, avs_cnl_i2s_machines), + AVS_MACH_ENTRY(HDA_CML_LP, avs_cnl_i2s_machines), + AVS_MACH_ENTRY(HDA_ICL_LP, avs_icl_i2s_machines), + AVS_MACH_ENTRY(HDA_TGL_LP, avs_tgl_i2s_machines), + AVS_MACH_ENTRY(HDA_EHL_0, avs_tgl_i2s_machines), + AVS_MACH_ENTRY(HDA_ADL_P, avs_tgl_i2s_machines), + AVS_MACH_ENTRY(HDA_RPL_P_0, avs_tgl_i2s_machines), + AVS_MACH_ENTRY(HDA_RPL_M, avs_tgl_i2s_machines), {}, }; -- 2.43.0

1 0

[PATCH AUTOSEL 6.8 45/98] ALSA: firewire-lib: handle quirk to calculate payload quadlets as data block counter
by Sasha Levin 29 Mar '24

29 Mar '24

From: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> [ Upstream commit 4a486439d2ca85752c46711f373b6ddc107bb35d ] Miglia Harmony Audio (OXFW970) has a quirk to put the number of accumulated quadlets in CIP payload into the dbc field of CIP header. This commit handles the quirk in the packet processing layer. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> Link: https://lore.kernel.org/r/20240218074128.95210-4-o-takashi@sakamocchi.jp Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/firewire/amdtp-stream.c | 12 ++++++++---- sound/firewire/amdtp-stream.h | 4 ++++ 2 files changed, 12 insertions(+), 4 deletions(-) diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c index 7be17bca257f0..c9f153f85ae6b 100644 --- a/sound/firewire/amdtp-stream.c +++ b/sound/firewire/amdtp-stream.c @@ -773,10 +773,14 @@ static int check_cip_header(struct amdtp_stream *s, const __be32 *buf, } else { unsigned int dbc_interval; - if (*data_blocks > 0 && s->ctx_data.tx.dbc_interval > 0) - dbc_interval = s->ctx_data.tx.dbc_interval; - else - dbc_interval = *data_blocks; + if (!(s->flags & CIP_DBC_IS_PAYLOAD_QUADLETS)) { + if (*data_blocks > 0 && s->ctx_data.tx.dbc_interval > 0) + dbc_interval = s->ctx_data.tx.dbc_interval; + else + dbc_interval = *data_blocks; + } else { + dbc_interval = payload_length / sizeof(__be32); + } lost = dbc != ((*data_block_counter + dbc_interval) & 0xff); } diff --git a/sound/firewire/amdtp-stream.h b/sound/firewire/amdtp-stream.h index b7ff44751ab93..a1ed2e80f91a7 100644 --- a/sound/firewire/amdtp-stream.h +++ b/sound/firewire/amdtp-stream.h @@ -37,6 +37,9 @@ * the value of current SYT_INTERVAL; e.g. initial value is not zero. * @CIP_UNAWARE_SYT: For outgoing packet, the value in SYT field of CIP is 0xffff. * For incoming packet, the value in SYT field of CIP is not handled. + * @CIP_DBC_IS_PAYLOAD_QUADLETS: Available for incoming packet, and only effective with + * CIP_DBC_IS_END_EVENT flag. The value of dbc field is the number of accumulated quadlets + * in CIP payload, instead of the number of accumulated data blocks. */ enum cip_flags { CIP_NONBLOCKING = 0x00, @@ -51,6 +54,7 @@ enum cip_flags { CIP_NO_HEADER = 0x100, CIP_UNALIGHED_DBC = 0x200, CIP_UNAWARE_SYT = 0x400, + CIP_DBC_IS_PAYLOAD_QUADLETS = 0x800, }; /** -- 2.43.0

1 0

[PATCH AUTOSEL 6.8 27/98] ASoC: Intel: common: DMI remap for rebranded Intel NUC M15 (LAPRC710) laptops
by Sasha Levin 29 Mar '24

29 Mar '24

From: mosomate <mosomate(a)gmail.com> [ Upstream commit c13e03126a5be90781084437689724254c8226e1 ] Added DMI quirk to handle the rebranded variants of Intel NUC M15 (LAPRC710) laptops. The DMI matching is based on motherboard attributes. Link: https://github.com/thesofproject/linux/issues/4218 Signed-off-by: Máté Mosonyi <mosomate(a)gmail.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20240208165545.93811-20-pierre-louis.bossart@linu… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- drivers/soundwire/dmi-quirks.c | 8 ++++++++ sound/soc/intel/boards/sof_sdw.c | 11 +++++++++++ 2 files changed, 19 insertions(+) diff --git a/drivers/soundwire/dmi-quirks.c b/drivers/soundwire/dmi-quirks.c index 9ebdd0cd0b1cf..91ab97a456fa9 100644 --- a/drivers/soundwire/dmi-quirks.c +++ b/drivers/soundwire/dmi-quirks.c @@ -130,6 +130,14 @@ static const struct dmi_system_id adr_remap_quirk_table[] = { }, .driver_data = (void *)intel_rooks_county, }, + { + /* quirk used for NUC15 LAPRC710 skew */ + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "Intel Corporation"), + DMI_MATCH(DMI_BOARD_NAME, "LAPRC710"), + }, + .driver_data = (void *)intel_rooks_county, + }, { .matches = { DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"), diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index 300391fbc2fc2..cf14bd3053fc9 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -236,6 +236,17 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { SOF_SDW_PCH_DMIC | RT711_JD2_100K), }, + { + /* NUC15 LAPRC710 skews */ + .callback = sof_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "Intel Corporation"), + DMI_MATCH(DMI_BOARD_NAME, "LAPRC710"), + }, + .driver_data = (void *)(SOF_SDW_TGL_HDMI | + SOF_SDW_PCH_DMIC | + RT711_JD2_100K), + }, /* TigerLake-SDCA devices */ { .callback = sof_sdw_quirk_cb, -- 2.43.0

1 0

[PATCH AUTOSEL 6.8 26/98] ASoC: Intel: sof_rt5682: dmi quirk cleanup for mtl boards
by Sasha Levin 29 Mar '24

29 Mar '24

From: Brent Lu <brent.lu(a)intel.com> [ Upstream commit 7a2a8730d51f95b263a1e8b888598dc6395220dc ] Some dmi quirks are duplicated since codec and amplifier type are removed from board quirk. Remove redundant quirks. Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Signed-off-by: Brent Lu <brent.lu(a)intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20240208165545.93811-7-pierre-louis.bossart@linux… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/sof_rt5682.c | 40 ----------------------------- 1 file changed, 40 deletions(-) diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index cd50f26d1edbe..02705034e5713 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -154,46 +154,6 @@ static const struct dmi_system_id sof_rt5682_quirk_table[] = { SOF_RT5682_SSP_AMP(2) | SOF_RT5682_NUM_HDMIDEV(4)), }, - { - .callback = sof_rt5682_quirk_cb, - .matches = { - DMI_MATCH(DMI_PRODUCT_FAMILY, "Google_Rex"), - DMI_MATCH(DMI_OEM_STRING, "AUDIO-MAX98360_ALC5682I_I2S"), - }, - .driver_data = (void *)(SOF_RT5682_MCLK_EN | - SOF_RT5682_SSP_CODEC(2) | - SOF_RT5682_SSP_AMP(0) | - SOF_RT5682_NUM_HDMIDEV(3) | - SOF_BT_OFFLOAD_SSP(1) | - SOF_SSP_BT_OFFLOAD_PRESENT - ), - }, - { - .callback = sof_rt5682_quirk_cb, - .matches = { - DMI_MATCH(DMI_PRODUCT_FAMILY, "Google_Rex"), - DMI_MATCH(DMI_OEM_STRING, "AUDIO-MAX98360_ALC5682I_DISCRETE_I2S_BT"), - }, - .driver_data = (void *)(SOF_RT5682_MCLK_EN | - SOF_RT5682_SSP_CODEC(2) | - SOF_RT5682_SSP_AMP(0) | - SOF_RT5682_NUM_HDMIDEV(3) | - SOF_BT_OFFLOAD_SSP(1) | - SOF_SSP_BT_OFFLOAD_PRESENT - ), - }, - { - .callback = sof_rt5682_quirk_cb, - .matches = { - DMI_MATCH(DMI_PRODUCT_FAMILY, "Google_Rex"), - DMI_MATCH(DMI_OEM_STRING, "AUDIO-ALC1019_ALC5682I_I2S"), - }, - .driver_data = (void *)(SOF_RT5682_MCLK_EN | - SOF_RT5682_SSP_CODEC(2) | - SOF_RT5682_SSP_AMP(0) | - SOF_RT5682_NUM_HDMIDEV(3) - ), - }, { .callback = sof_rt5682_quirk_cb, .matches = { -- 2.43.0

1 0

[PATCH 0/6] ASoC: rt-sdw: fix locking and improve error logs
by Pierre-Louis Bossart 29 Mar '24

29 Mar '24

Fix a set of problematic locking sequences and update error messages, tested on SOF/SoundWire platforms. Pierre-Louis Bossart (6): ASoC: rt5682-sdw: fix locking sequence ASoC: rt711-sdca: fix locking sequence ASoC: rt711-sdw: fix locking sequence ASoC: rt712-sdca-sdw: fix locking sequence ASoC: rt722-sdca-sdw: fix locking sequence ASoC: rt-sdw*: add __func__ to all error logs sound/soc/codecs/rt1316-sdw.c | 8 +++--- sound/soc/codecs/rt1318-sdw.c | 8 +++--- sound/soc/codecs/rt5682-sdw.c | 16 +++++------ sound/soc/codecs/rt700.c | 16 +++++------ sound/soc/codecs/rt711-sdca-sdw.c | 6 ++-- sound/soc/codecs/rt711-sdca.c | 18 ++++++------ sound/soc/codecs/rt711-sdw.c | 8 +++--- sound/soc/codecs/rt711.c | 16 +++++------ sound/soc/codecs/rt712-sdca-dmic.c | 24 +++++++++------- sound/soc/codecs/rt712-sdca-sdw.c | 7 +++-- sound/soc/codecs/rt712-sdca.c | 20 ++++++------- sound/soc/codecs/rt715-sdca-sdw.c | 2 +- sound/soc/codecs/rt715-sdca.c | 46 +++++++++++++++--------------- sound/soc/codecs/rt715-sdw.c | 4 +-- sound/soc/codecs/rt715.c | 24 ++++++++-------- sound/soc/codecs/rt722-sdca-sdw.c | 4 +-- sound/soc/codecs/rt722-sdca.c | 21 +++++++------- 17 files changed, 126 insertions(+), 122 deletions(-) -- 2.40.1

2 8

[PATCH v5 0/2] ASoC: nau8325: Modify driver code and dtschema.
by Seven Lee 29 Mar '24

29 Mar '24

Revise properties description and use standard units in dtschema. The unit conversion driver based on the attribute must also be changed accordingly. Change: V1 -> V2: - Revise the driver description part for the C++ comment. - In the nau8325_clkdet_put function, modify the max variable to hard code. - Removed "Clock Detection" switch control. - modify the "ALC Enable" switch name. - Revise the dtschema for "nuvoton,dac-vref". V2 -> V3: - Properties use standard unit suffixes. - Modify the enum definition. - Driver code should be used dev_dbg(). V3 -> V4: - Properties use standard unit suffixes. - Modify the enum definition. V4->V5: - Properties use standard unit suffixes for vref-impedance and dac-vref. - Revise the enum definition. - Modify the code related to properties usage in the NAU8325 codec driver. - Re-arrange header files in alphabetical order. Seven Lee (2): ASoC: dt-bindings: Added schema for "nuvoton,nau8325" ASoC: nau8325: new driver .../bindings/sound/nuvoton,nau8325.yaml | 80 ++ sound/soc/codecs/nau8325.c | 893 ++++++++++++++++++ sound/soc/codecs/nau8325.h | 391 ++++++++ 3 files changed, 1364 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/nuvoton,nau8325.yaml create mode 100644 sound/soc/codecs/nau8325.c create mode 100644 sound/soc/codecs/nau8325.h -- 2.25.1

4 6

[PATCH 00/18] ASoC: drop driver owner assignment
by Krzysztof Kozlowski 29 Mar '24

29 Mar '24

Core for several drivers already sets the driver.owner, so driver does not need to. Simplify the drivers. Patches are independent, no dependencies. Best regards, Krzysztof --- Krzysztof Kozlowski (18): ASoC: loongson: i2s/pci: drop driver owner assignment ASoC: tlv320aic32x4-spi: drop driver owner assignment ASoC: tlv320aic3x-spi: drop driver owner assignment ASoC: max98373-sdw: drop driver owner assignment ASoC: rt1017-sdca-sdw: drop driver owner assignment ASoC: rt1308-sdw: drop driver owner assignment ASoC: rt1316-sdw: drop driver owner assignment ASoC: rt1318-sdw: drop driver owner assignment ASoC: rt5682-sdw: drop driver owner assignment ASoC: rt700-sdw: drop driver owner assignment ASoC: rt711-sdca-sdw: drop driver owner assignment ASoC: rt711-sdw: drop driver owner assignment ASoC: rt712-sdca-dmic: drop driver owner assignment ASoC: rt712-sdca-sdw: drop driver owner assignment ASoC: rt715-sdca-sdw: drop driver owner assignment ASoC: rt715-sdw: drop driver owner assignment ASoC: rt722-sdca-sdw: drop driver owner assignment ASoC: sdw-mockup: drop driver owner assignment sound/soc/codecs/max98373-sdw.c | 1 - sound/soc/codecs/rt1017-sdca-sdw.c | 1 - sound/soc/codecs/rt1308-sdw.c | 1 - sound/soc/codecs/rt1316-sdw.c | 1 - sound/soc/codecs/rt1318-sdw.c | 1 - sound/soc/codecs/rt5682-sdw.c | 1 - sound/soc/codecs/rt700-sdw.c | 1 - sound/soc/codecs/rt711-sdca-sdw.c | 1 - sound/soc/codecs/rt711-sdw.c | 1 - sound/soc/codecs/rt712-sdca-dmic.c | 1 - sound/soc/codecs/rt712-sdca-sdw.c | 1 - sound/soc/codecs/rt715-sdca-sdw.c | 1 - sound/soc/codecs/rt715-sdw.c | 1 - sound/soc/codecs/rt722-sdca-sdw.c | 1 - sound/soc/codecs/sdw-mockup.c | 1 - sound/soc/codecs/tlv320aic32x4-spi.c | 1 - sound/soc/codecs/tlv320aic3x-spi.c | 1 - sound/soc/loongson/loongson_i2s_pci.c | 1 - 18 files changed, 18 deletions(-) --- base-commit: 1cadb9c1269d4e90ec23f3dc3e5373ee405462d1 change-id: 20240327-module-owner-var-2744d140fcdf Best regards, -- Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org>

3 20

[PATCH 00/18] ASoC: Intel: boards: updates for 6.10 - part3
by Pierre-Louis Bossart 29 Mar '24

29 Mar '24

This last part is the continuation of Brent Lu's cleanups. Multiple quirks have been removed to use "default" configurations and ACPI-based detection of codecs and selection of topology files. This cleanup has been done in multiple steps/phases since Fall 2023, thanks Brent for this contribution! Brent Lu (18): ASoC: Intel: ssp-common: relocate source file ASoC: Intel: ssp-common: relocate header file ASoC: Intel: ssp-common: naming convention change ASoC: Intel: ssp-common: module name change ASoC: Intel: ssp-common: delete module ASoC: Intel: ssp-common: get codec tplg suffix function ASoC: SOF: Intel: support tplg suffix detection ASoC: Intel: sof_nau8825: mach cleanup for adl boards ASoC: Intel: sof_nau8825: mach cleanup for rpl boards ASoC: Intel: sof_nau8825: add mtl_nau8825_def for mtl boards ASoC: Intel: sof_rt5682: mach cleanup for tgl boards ASoC: Intel: sof_rt5682: mach cleanup for adl boards ASoC: Intel: sof_rt5682: mach cleanup for rpl boards ASoC: Intel: sof_rt5682: mach cleanup for mtl boards ASoC: Intel: sof_cs42l42: board id cleanup for adl boards ASoC: Intel: sof_cs42l42: mach cleanup for adl boards ASoC: Intel: sof_cs42l42: add rpl_cs42l42_def for rpl boards ASoC: Intel: sof_cs42l42: add mtl_cs42l42_def for mtl boards .../sound/soc-acpi-intel-ssp-common.h | 22 ++- include/sound/soc-acpi.h | 12 ++ sound/soc/intel/boards/Kconfig | 13 +- sound/soc/intel/boards/Makefile | 3 - sound/soc/intel/boards/sof_board_helpers.c | 15 +- sound/soc/intel/boards/sof_board_helpers.h | 6 +- sound/soc/intel/boards/sof_cirrus_common.h | 2 +- sound/soc/intel/boards/sof_cs42l42.c | 17 +- sound/soc/intel/boards/sof_maxim_common.h | 2 +- sound/soc/intel/boards/sof_nau8825.c | 7 + sound/soc/intel/boards/sof_nuvoton_common.h | 2 +- sound/soc/intel/boards/sof_realtek_common.h | 2 +- sound/soc/intel/boards/sof_ssp_common.c | 122 -------------- sound/soc/intel/common/Makefile | 2 + .../intel/common/soc-acpi-intel-adl-match.c | 139 ++++----------- .../intel/common/soc-acpi-intel-mtl-match.c | 52 +++--- .../intel/common/soc-acpi-intel-rpl-match.c | 90 ++++------ .../intel/common/soc-acpi-intel-ssp-common.c | 159 ++++++++++++++++++ .../intel/common/soc-acpi-intel-tgl-match.c | 50 ++---- sound/soc/sof/intel/hda.c | 82 +++++++++ 20 files changed, 423 insertions(+), 376 deletions(-) rename sound/soc/intel/boards/sof_ssp_common.h => include/sound/soc-acpi-intel-ssp-common.h (66%) delete mode 100644 sound/soc/intel/boards/sof_ssp_common.c create mode 100644 sound/soc/intel/common/soc-acpi-intel-ssp-common.c -- 2.40.1

2 19

[PATCH] ASoC: tas2781: mark dvc_tlv with __maybe_unused
by Gergo Koteles 29 Mar '24

29 Mar '24

Since we put dvc_tlv static variable to a header file it's copied to each module that includes the header. But not all of them are actually used it. Fix this W=1 build warning: include/sound/tas2781-tlv.h:18:35: warning: 'dvc_tlv' defined but not used [-Wunused-const-variable=] Reported-by: kernel test robot <lkp(a)intel.com> Closes: https://lore.kernel.org/oe-kbuild-all/202403290354.v0StnRpc-lkp@intel.com/ Fixes: ae065d0ce9e3 ("ALSA: hda/tas2781: remove digital gain kcontrol") Signed-off-by: Gergo Koteles <soyer(a)irl.hu> --- include/sound/tas2781-tlv.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/include/sound/tas2781-tlv.h b/include/sound/tas2781-tlv.h index 4038dd421150..1dc59005d241 100644 --- a/include/sound/tas2781-tlv.h +++ b/include/sound/tas2781-tlv.h @@ -15,7 +15,7 @@ #ifndef __TAS2781_TLV_H__ #define __TAS2781_TLV_H__ -static const DECLARE_TLV_DB_SCALE(dvc_tlv, -10000, 100, 0); +static const __maybe_unused DECLARE_TLV_DB_SCALE(dvc_tlv, -10000, 100, 0); static const DECLARE_TLV_DB_SCALE(amp_vol_tlv, 1100, 50, 0); #endif base-commit: 2d0401ee38d43ab0e4cdd02dfc9d402befb2b5c8 -- 2.44.0

2 1

[PATCH] dt-bindings: ti,pcm1681: Convert to dtschema
by Animesh Agarwal 29 Mar '24

29 Mar '24

Convert the Texas Instruments PCM1681 bindings to DT schema. Signed-off-by: Animesh Agarwal <animeshagarwal28(a)gmail.com> --- .../devicetree/bindings/sound/ti,pcm1681.txt | 15 -------- .../devicetree/bindings/sound/ti,pcm1681.yaml | 35 +++++++++++++++++++ 2 files changed, 35 insertions(+), 15 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/ti,pcm1681.txt create mode 100644 Documentation/devicetree/bindings/sound/ti,pcm1681.yaml diff --git a/Documentation/devicetree/bindings/sound/ti,pcm1681.txt b/Documentation/devicetree/bindings/sound/ti,pcm1681.txt deleted file mode 100644 index 4df17185ab80..000000000000 --- a/Documentation/devicetree/bindings/sound/ti,pcm1681.txt +++ /dev/null @@ -1,15 +0,0 @@ -Texas Instruments PCM1681 8-channel PWM Processor - -Required properties: - - - compatible: Should contain "ti,pcm1681". - - reg: The i2c address. Should contain <0x4c>. - -Examples: - - i2c_bus { - pcm1681@4c { - compatible = "ti,pcm1681"; - reg = <0x4c>; - }; - }; diff --git a/Documentation/devicetree/bindings/sound/ti,pcm1681.yaml b/Documentation/devicetree/bindings/sound/ti,pcm1681.yaml new file mode 100644 index 000000000000..4093d0ff654d --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ti,pcm1681.yaml @@ -0,0 +1,35 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/ti,pcm1681.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Texas Instruments PCM1681 8-channel PWM Processor + +maintainers: + - Animesh Agarwal <animeshagarwal28(a)gmail.com> + +properties: + compatible: + const: ti,pcm1681 + + reg: + maxItems: 1 + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + + pcm1681@4c { + compatible = "ti,pcm1681"; + reg = <0x4c>; + }; + }; -- 2.44.0

2 2

[PATCH] ASoC: ops: Fix wraparound for mask in snd_soc_get_volsw
by Stephen Lee 29 Mar '24

29 Mar '24

In snd_soc_info_volsw(), mask is generated by figuring out the index of the most significant bit set in max and converting the index to a bitmask through bit shift 1. Unintended wraparound occurs when max is an integer value with msb bit set. Since the bit shift value 1 is treated as an integer type, the left shift operation will wraparound and set mask to 0 instead of all 1's. In order to fix this, we type cast 1 as `1ULL` to prevent the wraparound. Fixes: 7077148fb50a ("ASoC: core: Split ops out of soc-core.c") Signed-off-by: Stephen Lee <slee08177(a)gmail.com> --- This seems to show up in quite a few places in the alsa subsystem, should they be addressed altogether? --- sound/soc/soc-ops.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c index 2d25748ca706..b27e89ff6a16 100644 --- a/sound/soc/soc-ops.c +++ b/sound/soc/soc-ops.c @@ -263,7 +263,7 @@ int snd_soc_get_volsw(struct snd_kcontrol *kcontrol, int max = mc->max; int min = mc->min; int sign_bit = mc->sign_bit; - unsigned int mask = (1 << fls(max)) - 1; + unsigned int mask = (1ULL << fls(max)) - 1; unsigned int invert = mc->invert; int val; int ret; -- 2.43.2

2 2

[PATCH 1/2] ASoC: amd: acp: fix for acp_init function error handling
by Vijendar Mukunda 29 Mar '24

29 Mar '24

If acp_init() fails, acp pci driver probe should return error. Add acp_init() function return value check logic. Fixes: e61b415515d3 ("ASoC: amd: acp: refactor the acp init and de-init sequence") Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- sound/soc/amd/acp/acp-pci.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/amd/acp/acp-pci.c b/sound/soc/amd/acp/acp-pci.c index 8c8b1dcac628..c1d4140f0746 100644 --- a/sound/soc/amd/acp/acp-pci.c +++ b/sound/soc/amd/acp/acp-pci.c @@ -115,7 +115,10 @@ static int acp_pci_probe(struct pci_dev *pci, const struct pci_device_id *pci_id goto unregister_dmic_dev; } - acp_init(chip); + ret = acp_init(chip); + if (ret) + return ret; + res = devm_kcalloc(&pci->dev, num_res, sizeof(struct resource), GFP_KERNEL); if (!res) { ret = -ENOMEM; -- 2.34.1

3 4

[PATCH] ASoC: hdac_hda: improve error logs
by Pierre-Louis Bossart 28 Mar '24

28 Mar '24

We have a couple of duplicate logs and missing information, add __func__ consistently and make sure useful error codes are logged. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Reviewed-by: Péter Ujfalusi <peter.ujfalusi(a)linux.intel.com> --- sound/soc/codecs/hdac_hda.c | 44 ++++++++++++++++++++----------------- 1 file changed, 24 insertions(+), 20 deletions(-) diff --git a/sound/soc/codecs/hdac_hda.c b/sound/soc/codecs/hdac_hda.c index 6aa3223985be..29c88de5508b 100644 --- a/sound/soc/codecs/hdac_hda.c +++ b/sound/soc/codecs/hdac_hda.c @@ -230,7 +230,8 @@ static int hdac_hda_dai_hw_params(struct snd_pcm_substream *substream, format_val = snd_hdac_stream_format(params_channels(params), bits, params_rate(params)); if (!format_val) { dev_err(dai->dev, - "invalid format_val, rate=%d, ch=%d, format=%d, maxbps=%d\n", + "%s: invalid format_val, rate=%d, ch=%d, format=%d, maxbps=%d\n", + __func__, params_rate(params), params_channels(params), params_format(params), maxbps); @@ -266,14 +267,12 @@ static int hdac_hda_dai_prepare(struct snd_pcm_substream *substream, struct snd_soc_component *component = dai->component; struct hda_pcm_stream *hda_stream; struct hdac_hda_priv *hda_pvt; - struct hdac_device *hdev; unsigned int format_val; struct hda_pcm *pcm; unsigned int stream; int ret = 0; hda_pvt = snd_soc_component_get_drvdata(component); - hdev = &hda_pvt->codec->core; pcm = snd_soc_find_pcm_from_dai(hda_pvt, dai); if (!pcm) return -EINVAL; @@ -286,7 +285,7 @@ static int hdac_hda_dai_prepare(struct snd_pcm_substream *substream, ret = snd_hda_codec_prepare(hda_pvt->codec, hda_stream, stream, format_val, substream); if (ret < 0) - dev_err(&hdev->dev, "codec prepare failed %d\n", ret); + dev_err(dai->dev, "%s: failed %d\n", __func__, ret); return ret; } @@ -298,6 +297,7 @@ static int hdac_hda_dai_open(struct snd_pcm_substream *substream, struct hdac_hda_priv *hda_pvt; struct hda_pcm_stream *hda_stream; struct hda_pcm *pcm; + int ret; hda_pvt = snd_soc_component_get_drvdata(component); pcm = snd_soc_find_pcm_from_dai(hda_pvt, dai); @@ -308,7 +308,11 @@ static int hdac_hda_dai_open(struct snd_pcm_substream *substream, hda_stream = &pcm->stream[substream->stream]; - return hda_stream->ops.open(hda_stream, hda_pvt->codec, substream); + ret = hda_stream->ops.open(hda_stream, hda_pvt->codec, substream); + if (ret < 0) + dev_err(dai->dev, "%s: failed %d\n", __func__, ret); + + return ret; } static void hdac_hda_dai_close(struct snd_pcm_substream *substream, @@ -367,7 +371,7 @@ static struct hda_pcm *snd_soc_find_pcm_from_dai(struct hdac_hda_priv *hda_pvt, pcm_name = "HDMI 3"; break; default: - dev_err(&hcodec->core.dev, "invalid dai id %d\n", dai->id); + dev_err(dai->dev, "%s: invalid dai id %d\n", __func__, dai->id); return NULL; } @@ -381,7 +385,7 @@ static struct hda_pcm *snd_soc_find_pcm_from_dai(struct hdac_hda_priv *hda_pvt, } } - dev_err(&hcodec->core.dev, "didn't find PCM for DAI %s\n", dai->name); + dev_err(dai->dev, "%s: didn't find PCM for DAI %s\n", __func__, dai->name); return NULL; } @@ -411,7 +415,7 @@ static int hdac_hda_codec_probe(struct snd_soc_component *component) hlink = snd_hdac_ext_bus_get_hlink_by_name(hdev->bus, dev_name(&hdev->dev)); if (!hlink) { - dev_err(&hdev->dev, "hdac link not found\n"); + dev_err(&hdev->dev, "%s: hdac link not found\n", __func__); return -EIO; } @@ -429,7 +433,7 @@ static int hdac_hda_codec_probe(struct snd_soc_component *component) ret = snd_hda_codec_device_new(hcodec->bus, component->card->snd_card, hdev->addr, hcodec, true); if (ret < 0) { - dev_err(&hdev->dev, "failed to create hda codec %d\n", ret); + dev_err(&hdev->dev, "%s: failed to create hda codec %d\n", __func__, ret); goto error_no_pm; } @@ -446,7 +450,7 @@ static int hdac_hda_codec_probe(struct snd_soc_component *component) if (fw) { ret = snd_hda_load_patch(hcodec->bus, fw->size, fw->data); if (ret < 0) { - dev_err(&hdev->dev, "failed to load hda patch %d\n", ret); + dev_err(&hdev->dev, "%s: failed to load hda patch %d\n", __func__, ret); goto error_no_pm; } release_firmware(fw); @@ -470,13 +474,13 @@ static int hdac_hda_codec_probe(struct snd_soc_component *component) ret = snd_hda_codec_set_name(hcodec, hcodec->preset->name); if (ret < 0) { - dev_err(&hdev->dev, "name failed %s\n", hcodec->preset->name); + dev_err(&hdev->dev, "%s: name failed %s\n", __func__, hcodec->preset->name); goto error_pm; } ret = snd_hdac_regmap_init(&hcodec->core); if (ret < 0) { - dev_err(&hdev->dev, "regmap init failed\n"); + dev_err(&hdev->dev, "%s: regmap init failed\n", __func__); goto error_pm; } @@ -484,16 +488,16 @@ static int hdac_hda_codec_probe(struct snd_soc_component *component) if (patch) { ret = patch(hcodec); if (ret < 0) { - dev_err(&hdev->dev, "patch failed %d\n", ret); + dev_err(&hdev->dev, "%s: patch failed %d\n", __func__, ret); goto error_regmap; } } else { - dev_dbg(&hdev->dev, "no patch file found\n"); + dev_dbg(&hdev->dev, "%s: no patch file found\n", __func__); } ret = snd_hda_codec_parse_pcms(hcodec); if (ret < 0) { - dev_err(&hdev->dev, "unable to map pcms to dai %d\n", ret); + dev_err(&hdev->dev, "%s: unable to map pcms to dai %d\n", __func__, ret); goto error_patch; } @@ -501,8 +505,8 @@ static int hdac_hda_codec_probe(struct snd_soc_component *component) if (!is_hdmi_codec(hcodec)) { ret = snd_hda_codec_build_controls(hcodec); if (ret < 0) { - dev_err(&hdev->dev, "unable to create controls %d\n", - ret); + dev_err(&hdev->dev, "%s: unable to create controls %d\n", + __func__, ret); goto error_patch; } } @@ -548,7 +552,7 @@ static void hdac_hda_codec_remove(struct snd_soc_component *component) hlink = snd_hdac_ext_bus_get_hlink_by_name(hdev->bus, dev_name(&hdev->dev)); if (!hlink) { - dev_err(&hdev->dev, "hdac link not found\n"); + dev_err(&hdev->dev, "%s: hdac link not found\n", __func__); return; } @@ -624,7 +628,7 @@ static int hdac_hda_dev_probe(struct hdac_device *hdev) /* hold the ref while we probe */ hlink = snd_hdac_ext_bus_get_hlink_by_name(hdev->bus, dev_name(&hdev->dev)); if (!hlink) { - dev_err(&hdev->dev, "hdac link not found\n"); + dev_err(&hdev->dev, "%s: hdac link not found\n", __func__); return -EIO; } snd_hdac_ext_bus_link_get(hdev->bus, hlink); @@ -640,7 +644,7 @@ static int hdac_hda_dev_probe(struct hdac_device *hdev) ARRAY_SIZE(hdac_hda_dais)); if (ret < 0) { - dev_err(&hdev->dev, "failed to register HDA codec %d\n", ret); + dev_err(&hdev->dev, "%s: failed to register HDA codec %d\n", __func__, ret); return ret; } -- 2.40.1

2 2

[tiwai-sound:for-linus 12/17] include/sound/tas2781-tlv.h:18:35: warning: 'dvc_tlv' defined but not used
by kernel test robot 28 Mar '24

28 Mar '24

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git for-linus head: 2d0401ee38d43ab0e4cdd02dfc9d402befb2b5c8 commit: ae065d0ce9e36ca4efdfb9b96ce3395bd1c19372 [12/17] ALSA: hda/tas2781: remove digital gain kcontrol config: i386-allmodconfig (https://download.01.org/0day-ci/archive/20240329/202403290354.v0StnRpc-lkp@…) compiler: gcc-12 (Ubuntu 12.3.0-9ubuntu2) 12.3.0 reproduce (this is a W=1 build): (https://download.01.org/0day-ci/archive/20240329/202403290354.v0StnRpc-lkp@…) If you fix the issue in a separate patch/commit (i.e. not just a new version of the same patch/commit), kindly add following tags | Reported-by: kernel test robot <lkp(a)intel.com> | Closes: https://lore.kernel.org/oe-kbuild-all/202403290354.v0StnRpc-lkp@intel.com/ All warnings (new ones prefixed by >>): In file included from include/sound/tlv.h:10, from sound/pci/hda/tas2781_hda_i2c.c:22: >> include/sound/tas2781-tlv.h:18:35: warning: 'dvc_tlv' defined but not used [-Wunused-const-variable=] 18 | static const DECLARE_TLV_DB_SCALE(dvc_tlv, -10000, 100, 0); | ^~~~~~~ include/uapi/sound/tlv.h:53:22: note: in definition of macro 'SNDRV_CTL_TLVD_DECLARE_DB_SCALE' 53 | unsigned int name[] = { \ | ^~~~ include/sound/tas2781-tlv.h:18:14: note: in expansion of macro 'DECLARE_TLV_DB_SCALE' 18 | static const DECLARE_TLV_DB_SCALE(dvc_tlv, -10000, 100, 0); | ^~~~~~~~~~~~~~~~~~~~ vim +/dvc_tlv +18 include/sound/tas2781-tlv.h 678f38eba1f2fe Shenghao Ding 2023-06-18 17 678f38eba1f2fe Shenghao Ding 2023-06-18 @18 static const DECLARE_TLV_DB_SCALE(dvc_tlv, -10000, 100, 0); 678f38eba1f2fe Shenghao Ding 2023-06-18 19 static const DECLARE_TLV_DB_SCALE(amp_vol_tlv, 1100, 50, 0); 678f38eba1f2fe Shenghao Ding 2023-06-18 20 :::::: The code at line 18 was first introduced by commit :::::: 678f38eba1f2fe33ff700e85390ac98393e609ef ASoC: tas2781: Add Header file for tas2781 driver :::::: TO: Shenghao Ding <13916275206(a)139.com> :::::: CC: Mark Brown <broonie(a)kernel.org> -- 0-DAY CI Kernel Test Service https://github.com/intel/lkp-tests/wiki

1 0

[PATCH] soundwire: qcom: Convert to platform remove callback returning void
by Uwe Kleine-König 28 Mar '24

28 Mar '24

The .remove() callback for a platform driver returns an int which makes many driver authors wrongly assume it's possible to do error handling by returning an error code. However the value returned is ignored (apart from emitting a warning) and this typically results in resource leaks. To improve here there is a quest to make the remove callback return void. In the first step of this quest all drivers are converted to .remove_new(), which already returns void. Eventually after all drivers are converted, .remove_new() will be renamed to .remove(). Trivially convert this driver from always returning zero in the remove callback to the void returning variant. Signed-off-by: Uwe Kleine-König <u.kleine-koenig(a)pengutronix.de> --- drivers/soundwire/qcom.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) diff --git a/drivers/soundwire/qcom.c b/drivers/soundwire/qcom.c index 3c4d6debab1f..fb70afe64fcc 100644 --- a/drivers/soundwire/qcom.c +++ b/drivers/soundwire/qcom.c @@ -1636,14 +1636,12 @@ static int qcom_swrm_probe(struct platform_device *pdev) return ret; } -static int qcom_swrm_remove(struct platform_device *pdev) +static void qcom_swrm_remove(struct platform_device *pdev) { struct qcom_swrm_ctrl *ctrl = dev_get_drvdata(&pdev->dev); sdw_bus_master_delete(&ctrl->bus); clk_disable_unprepare(ctrl->hclk); - - return 0; } static int __maybe_unused swrm_runtime_resume(struct device *dev) @@ -1769,7 +1767,7 @@ MODULE_DEVICE_TABLE(of, qcom_swrm_of_match); static struct platform_driver qcom_swrm_driver = { .probe = &qcom_swrm_probe, - .remove = &qcom_swrm_remove, + .remove_new = qcom_swrm_remove, .driver = { .name = "qcom-soundwire", .of_match_table = qcom_swrm_of_match, base-commit: 11afac187274a6177a7ac82997f8691c0f469e41 -- 2.43.0

2 1

[PATCH 0/7] Soundwire: clean up sysfs group creation
by Greg Kroah-Hartman 28 Mar '24

28 Mar '24

Note, this is a redone version of a very old series I wrote back in 2022: https://lore.kernel.org/r/20220824135951.3604059-1-gregkh@linuxfoundation.o… but everyone has forgotten about it now, and I've reworked it, so I'm considering it a "new" version, and not v2. Here's a series that adds the functionality to the driver core to hide entire attribute groups, in a much saner way than we have attempted in the past (i.e. dynamically figuring it out.) Many thanks to Dan for this patch. I'll also be taking this into my driver-core branch and creating a stable tag for anyone else to pull from to get it into their trees, as I think it will want to be in many for this development cycle. After the driver core change, there's cleanups to the soundwire core for how the attribute groups are created, to remove the "manual" creation of them, and allow the driver core to create them correctly, as needed, when needed, which makes things much smaller for the soundwire code to manage. Comments appreciated! thanks, greg k-h Dan Williams (1): sysfs: Introduce a mechanism to hide static attribute_groups Greg Kroah-Hartman (5): soundwire: sysfs: move sdw_slave_dev_attr_group into the existing list of groups soundwire: sysfs: cleanup the logic for creating the dp0 sysfs attributes soundwire: sysfs: have the driver core handle the creation of the device groups soundwire: sysfs: remove sdw_slave_sysfs_init() soundwire: sysfs: remove unneeded ATTRIBUTE_GROUPS() comments drivers/soundwire/bus_type.c | 5 ++- drivers/soundwire/sysfs_local.h | 4 +- drivers/soundwire/sysfs_slave.c | 64 ++++++++++++++--------------- drivers/soundwire/sysfs_slave_dpn.c | 3 ++ fs/sysfs/group.c | 45 ++++++++++++++++---- include/linux/sysfs.h | 63 ++++++++++++++++++++++------ 6 files changed, 126 insertions(+), 58 deletions(-) -- 2.43.0

5 22

[PATCH 1/2] soundwire: amd: use inline function for register update
by Vijendar Mukunda 28 Mar '24

28 Mar '24

Define common inline function for register update. Use this inline function for updating SoundWire Pad registers and enable/disable SoundWire interrupt control registers. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- drivers/soundwire/amd_init.c | 36 +++++++++++++++------------------ drivers/soundwire/amd_init.h | 8 ++++++++ drivers/soundwire/amd_manager.c | 13 ++++++------ 3 files changed, 30 insertions(+), 27 deletions(-) diff --git a/drivers/soundwire/amd_init.c b/drivers/soundwire/amd_init.c index e45dc8261ab1..4cd26f3a21f5 100644 --- a/drivers/soundwire/amd_init.c +++ b/drivers/soundwire/amd_init.c @@ -17,42 +17,38 @@ #define ACP_PAD_PULLDOWN_CTRL 0x0001448 #define ACP_SW_PAD_KEEPER_EN 0x0001454 -#define AMD_SDW_PAD_PULLDOWN_CTRL_ENABLE_MASK 0x7f9a -#define AMD_SDW0_PAD_PULLDOWN_CTRL_ENABLE_MASK 0x7f9f -#define AMD_SDW1_PAD_PULLDOWN_CTRL_ENABLE_MASK 0x7ffa -#define AMD_SDW0_PAD_EN_MASK 1 -#define AMD_SDW1_PAD_EN_MASK 0x10 -#define AMD_SDW_PAD_EN_MASK (AMD_SDW0_PAD_EN_MASK | AMD_SDW1_PAD_EN_MASK) +#define AMD_SDW0_PAD_CTRL_MASK 0x60 +#define AMD_SDW1_PAD_CTRL_MASK 5 +#define AMD_SDW_PAD_CTRL_MASK (AMD_SDW0_PAD_CTRL_MASK | AMD_SDW1_PAD_CTRL_MASK) +#define AMD_SDW0_PAD_EN 1 +#define AMD_SDW1_PAD_EN 0x10 +#define AMD_SDW_PAD_EN (AMD_SDW0_PAD_EN | AMD_SDW1_PAD_EN) static int amd_enable_sdw_pads(void __iomem *mmio, u32 link_mask, struct device *dev) { - u32 val; - u32 pad_keeper_en_mask, pad_pulldown_ctrl_mask; + u32 pad_keeper_en, pad_pulldown_ctrl_mask; switch (link_mask) { case 1: - pad_keeper_en_mask = AMD_SDW0_PAD_EN_MASK; - pad_pulldown_ctrl_mask = AMD_SDW0_PAD_PULLDOWN_CTRL_ENABLE_MASK; + pad_keeper_en = AMD_SDW0_PAD_EN; + pad_pulldown_ctrl_mask = AMD_SDW0_PAD_CTRL_MASK; break; case 2: - pad_keeper_en_mask = AMD_SDW1_PAD_EN_MASK; - pad_pulldown_ctrl_mask = AMD_SDW1_PAD_PULLDOWN_CTRL_ENABLE_MASK; + pad_keeper_en = AMD_SDW1_PAD_EN; + pad_pulldown_ctrl_mask = AMD_SDW1_PAD_CTRL_MASK; break; case 3: - pad_keeper_en_mask = AMD_SDW_PAD_EN_MASK; - pad_pulldown_ctrl_mask = AMD_SDW_PAD_PULLDOWN_CTRL_ENABLE_MASK; + pad_keeper_en = AMD_SDW_PAD_EN; + pad_pulldown_ctrl_mask = AMD_SDW_PAD_CTRL_MASK; break; default: dev_err(dev, "No SDW Links are enabled\n"); return -ENODEV; } - val = readl(mmio + ACP_SW_PAD_KEEPER_EN); - val |= pad_keeper_en_mask; - writel(val, mmio + ACP_SW_PAD_KEEPER_EN); - val = readl(mmio + ACP_PAD_PULLDOWN_CTRL); - val &= pad_pulldown_ctrl_mask; - writel(val, mmio + ACP_PAD_PULLDOWN_CTRL); + amd_updatel(mmio, ACP_SW_PAD_KEEPER_EN, pad_keeper_en, pad_keeper_en); + amd_updatel(mmio, ACP_PAD_PULLDOWN_CTRL, pad_pulldown_ctrl_mask, 0); + return 0; } diff --git a/drivers/soundwire/amd_init.h b/drivers/soundwire/amd_init.h index 928b0c707162..5e7b43836a37 100644 --- a/drivers/soundwire/amd_init.h +++ b/drivers/soundwire/amd_init.h @@ -10,4 +10,12 @@ int amd_sdw_manager_start(struct amd_sdw_manager *amd_manager); +static inline void amd_updatel(void __iomem *mmio, int offset, u32 mask, u32 val) +{ + u32 tmp; + + tmp = readl(mmio + offset); + tmp = (tmp & ~mask) | val; + writel(tmp, mmio + offset); +} #endif diff --git a/drivers/soundwire/amd_manager.c b/drivers/soundwire/amd_manager.c index 7cd24bd8e224..1066d87aa011 100644 --- a/drivers/soundwire/amd_manager.c +++ b/drivers/soundwire/amd_manager.c @@ -89,9 +89,8 @@ static void amd_enable_sdw_interrupts(struct amd_sdw_manager *amd_manager) u32 val; mutex_lock(amd_manager->acp_sdw_lock); - val = readl(amd_manager->acp_mmio + ACP_EXTERNAL_INTR_CNTL(amd_manager->instance)); - val |= sdw_manager_reg_mask_array[amd_manager->instance]; - writel(val, amd_manager->acp_mmio + ACP_EXTERNAL_INTR_CNTL(amd_manager->instance)); + val = sdw_manager_reg_mask_array[amd_manager->instance]; + amd_updatel(amd_manager->acp_mmio, ACP_EXTERNAL_INTR_CNTL(amd_manager->instance), val, val); mutex_unlock(amd_manager->acp_sdw_lock); writel(AMD_SDW_IRQ_MASK_0TO7, amd_manager->mmio + @@ -103,12 +102,12 @@ static void amd_enable_sdw_interrupts(struct amd_sdw_manager *amd_manager) static void amd_disable_sdw_interrupts(struct amd_sdw_manager *amd_manager) { - u32 val; + u32 irq_mask; mutex_lock(amd_manager->acp_sdw_lock); - val = readl(amd_manager->acp_mmio + ACP_EXTERNAL_INTR_CNTL(amd_manager->instance)); - val &= ~sdw_manager_reg_mask_array[amd_manager->instance]; - writel(val, amd_manager->acp_mmio + ACP_EXTERNAL_INTR_CNTL(amd_manager->instance)); + irq_mask = sdw_manager_reg_mask_array[amd_manager->instance]; + amd_updatel(amd_manager->acp_mmio, ACP_EXTERNAL_INTR_CNTL(amd_manager->instance), + irq_mask, 0); mutex_unlock(amd_manager->acp_sdw_lock); writel(0x00, amd_manager->mmio + ACP_SW_STATE_CHANGE_STATUS_MASK_0TO7); -- 2.34.1

2 3

[PATCH] ALSA: hda: cs35l56: Add ACPI device match tables
by Simon Trimmer 28 Mar '24

28 Mar '24

Adding the ACPI HIDs to the match table triggers the cs35l56-hda modules to be loaded on boot so that Serial Multi Instantiate can add the devices to the bus and begin the driver init sequence. Signed-off-by: Simon Trimmer <simont(a)opensource.cirrus.com> Fixes: 73cfbfa9caea ("ALSA: hda/cs35l56: Add driver for Cirrus Logic CS35L56 amplifier") --- sound/pci/hda/cs35l56_hda_i2c.c | 13 +++++++++++-- sound/pci/hda/cs35l56_hda_spi.c | 13 +++++++++++-- 2 files changed, 22 insertions(+), 4 deletions(-) diff --git a/sound/pci/hda/cs35l56_hda_i2c.c b/sound/pci/hda/cs35l56_hda_i2c.c index 13beee807308..40f2f97944d5 100644 --- a/sound/pci/hda/cs35l56_hda_i2c.c +++ b/sound/pci/hda/cs35l56_hda_i2c.c @@ -56,10 +56,19 @@ static const struct i2c_device_id cs35l56_hda_i2c_id[] = { {} }; +static const struct acpi_device_id cs35l56_acpi_hda_match[] = { + { "CSC3554", 0 }, + { "CSC3556", 0 }, + { "CSC3557", 0 }, + {} +}; +MODULE_DEVICE_TABLE(acpi, cs35l56_acpi_hda_match); + static struct i2c_driver cs35l56_hda_i2c_driver = { .driver = { - .name = "cs35l56-hda", - .pm = &cs35l56_hda_pm_ops, + .name = "cs35l56-hda", + .acpi_match_table = cs35l56_acpi_hda_match, + .pm = &cs35l56_hda_pm_ops, }, .id_table = cs35l56_hda_i2c_id, .probe = cs35l56_hda_i2c_probe, diff --git a/sound/pci/hda/cs35l56_hda_spi.c b/sound/pci/hda/cs35l56_hda_spi.c index a3b2fa76663d..7f02155fe61e 100644 --- a/sound/pci/hda/cs35l56_hda_spi.c +++ b/sound/pci/hda/cs35l56_hda_spi.c @@ -56,10 +56,19 @@ static const struct spi_device_id cs35l56_hda_spi_id[] = { {} }; +static const struct acpi_device_id cs35l56_acpi_hda_match[] = { + { "CSC3554", 0 }, + { "CSC3556", 0 }, + { "CSC3557", 0 }, + {} +}; +MODULE_DEVICE_TABLE(acpi, cs35l56_acpi_hda_match); + static struct spi_driver cs35l56_hda_spi_driver = { .driver = { - .name = "cs35l56-hda", - .pm = &cs35l56_hda_pm_ops, + .name = "cs35l56-hda", + .acpi_match_table = cs35l56_acpi_hda_match, + .pm = &cs35l56_hda_pm_ops, }, .id_table = cs35l56_hda_spi_id, .probe = cs35l56_hda_spi_probe, -- 2.34.1

2 1

[PATCH 0/3] ASoC: Use snd_ctl_find_id_mixer() instead of open-coding
by Richard Fitzgerald 28 Mar '24

28 Mar '24

The first two patches change snd_soc_card_get_kcontrol() to use the core snd_ctl_find_id_mixer() functionality instead of open-coding its own list walk. The last patch adds a KUnit test for this, which was tested on the original and modified code. Richard Fitzgerald (3): ALSA: control: Introduce snd_ctl_find_id_mixer_locked() ASoC: soc-card: Use snd_ctl_find_id_mixer() instead of open-coding ASoC: soc-card: Add KUnit test case for snd_soc_card_get_kcontrol include/sound/control.h | 23 +++++ sound/soc/Kconfig | 8 ++ sound/soc/Makefile | 4 + sound/soc/soc-card-test.c | 191 ++++++++++++++++++++++++++++++++++++++ sound/soc/soc-card.c | 21 +---- 5 files changed, 230 insertions(+), 17 deletions(-) create mode 100644 sound/soc/soc-card-test.c -- 2.39.2

2 4

Dummy output with ES8326 codec
by GitHub issues - opened 28 Mar '24

28 Mar '24

alsa-project/alsa-ucm-conf issue #407 was opened from Henrythek: ## Description During the process of using the ES8326 codec, I encountered a dummy output issue. killall pulseaudio do not work, even after masking and then unmasking the pulseaudio service and attempting to restart it, the issue persists. The dummy output occurs intermittently when plugging in or unplugging headphones, and rebooting can temporarily resolve the issue. I am using the following configuration: - Operating System: Debian GNU/Linux 12 - Kernel Version: 6.7.0 ## Error Messages E: [pulseaudio] module-alsa-card.c: Failed to find a working profile. E: [pulseaudio] module.c: Failed to load module "module-alsa-card" (argument: "device_id="0" name="pci-0000_00_1f.3-platform-sof-essx8336" card_name="alsa_card.pci-0000_00_1f.3-platform-sof-essx8336" namereg_fail=false tsched=yes fixed_latency_range=no ignore_dB=no deferred_volume=yes use_ucm=yes avoid_resampling=no card_properties="module-udev-detect.discovered=1""): initialization failed. I: [pulseaudio] module-udev-detect.c: Card /devices/pci0000:00/0000:00:1f.3/sof-essx8336/sound/card0 (alsa_card.pci-0000_00_1f.3-platform-sof-essx8336) failed to load module. ## Additional Information The UCM (Use Case Manager) configuration used is from the following repository: [alsa-ucm-conf/ucm2/Intel/sof-essx8336](https://github.com/alsa-project/alsa… ## Steps to Reproduce 1. Use the ES8326 codec on a Debian GNU/Linux 12 system with Kernel 6.7.0. 2. Attempt to use the audio functionality. 3. Plug in or unplug headphones. 4. Observe the dummy output issue and the error messages mentioned above. ## Additional Context [alsa-info.txt](https://github.com/alsa-project/alsa-ucm-conf/files/14782974… [aplay.txt](https://github.com/alsa-project/alsa-ucm-conf/files/14782975/apl… [pulseaudio_v.txt](https://github.com/alsa-project/alsa-ucm-conf/files/14782… ![Screenshot from 2024-03-27 21-44-34](https://github.com/alsa-project/alsa-ucm-conf/assets/68213530/925… Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/407 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH 00/15] ASoC: replace dpcm_playback/capture to playback/capture_only
by Kuninori Morimoto 28 Mar '24

28 Mar '24

Hi Mark When we use DPCM, we need to set dpcm_playback/capture flag. If these flag are set, soc_get_playback_capture() will check its availability, but non DPCM case is checked without special flags. OTOH, it cares playback/capture_only flag. It is needed. This patch remove DPCM special flag, and replace it playback/capture_only flag if needed. Kuninori Morimoto (15): ASoC: soc-pcm.c: cleanup soc_get_playback_capture() ASoC: amd: replace dpcm_playback/capture to playback/capture_only ASoC: fsl: replace dpcm_playback/capture to playback/capture_only ASoC: sof: replace dpcm_playback/capture to playback/capture_only ASoC: meson: replace dpcm_playback/capture to playback/capture_only ASoC: Intel: replace dpcm_playback/capture to playback/capture_only ASoC: samsung: replace dpcm_playback/capture to playback/capture_only ASoC: mediatek: replace dpcm_playback/capture to playback/capture_only ASoC: Intel/avs: replace dpcm_playback/capture to playback/capture_only ASoC: soc-core.c: replace dpcm_playback/capture to playback/capture_only ASoC: soc-topology.c: replace dpcm_playback/capture to playback/capture_only ASoC: soc-compress.c: replace dpcm_playback/capture to playback/capture_only ASoC: remove snd_soc_dai_link_set_capabilities() ASoC: soc-pcm.c: remove dpcm_playback/capture ASoC: soc-pcm: indicate warning if DPCM BE Codec has no settings include/sound/soc-dai.h | 1 - include/sound/soc.h | 4 - sound/soc/amd/acp-da7219-max98357a.c | 20 ++--- sound/soc/amd/acp-es8336.c | 2 - sound/soc/amd/acp/acp-mach-common.c | 24 ++--- sound/soc/amd/acp3x-rt5682-max9836.c | 6 +- sound/soc/amd/vangogh/acp5x-mach.c | 6 -- sound/soc/fsl/fsl-asoc-card.c | 16 ++-- sound/soc/fsl/imx-audmix.c | 6 +- sound/soc/fsl/imx-card.c | 7 +- sound/soc/generic/audio-graph-card.c | 2 - sound/soc/generic/audio-graph-card2.c | 2 - sound/soc/generic/simple-card.c | 2 - sound/soc/intel/avs/boards/da7219.c | 2 - sound/soc/intel/avs/boards/dmic.c | 4 +- sound/soc/intel/avs/boards/es8336.c | 2 - sound/soc/intel/avs/boards/hdaudio.c | 4 - sound/soc/intel/avs/boards/i2s_test.c | 2 - sound/soc/intel/avs/boards/max98357a.c | 2 +- sound/soc/intel/avs/boards/max98373.c | 2 - sound/soc/intel/avs/boards/max98927.c | 2 - sound/soc/intel/avs/boards/nau8825.c | 2 - sound/soc/intel/avs/boards/rt274.c | 2 - sound/soc/intel/avs/boards/rt286.c | 2 - sound/soc/intel/avs/boards/rt298.c | 2 - sound/soc/intel/avs/boards/rt5514.c | 2 +- sound/soc/intel/avs/boards/rt5663.c | 2 - sound/soc/intel/avs/boards/rt5682.c | 2 - sound/soc/intel/avs/boards/ssm4567.c | 2 - sound/soc/intel/boards/bdw-rt5650.c | 4 - sound/soc/intel/boards/bdw-rt5677.c | 4 - sound/soc/intel/boards/bdw_rt286.c | 10 +-- sound/soc/intel/boards/bxt_da7219_max98357a.c | 32 ++++--- sound/soc/intel/boards/bxt_rt298.c | 26 +++--- sound/soc/intel/boards/bytcht_cx2072x.c | 6 +- sound/soc/intel/boards/bytcht_da7213.c | 6 +- sound/soc/intel/boards/bytcht_es8316.c | 6 +- sound/soc/intel/boards/bytcht_nocodec.c | 6 +- sound/soc/intel/boards/bytcr_rt5640.c | 6 +- sound/soc/intel/boards/bytcr_rt5651.c | 6 +- sound/soc/intel/boards/bytcr_wm5102.c | 6 +- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 6 +- sound/soc/intel/boards/cht_bsw_nau8824.c | 6 +- sound/soc/intel/boards/cht_bsw_rt5645.c | 6 +- sound/soc/intel/boards/cht_bsw_rt5672.c | 6 +- sound/soc/intel/boards/cml_rt1011_rt5682.c | 14 ++- sound/soc/intel/boards/ehl_rt5660.c | 14 ++- sound/soc/intel/boards/glk_rt5682_max98357a.c | 30 +++---- sound/soc/intel/boards/hsw_rt5640.c | 10 +-- sound/soc/intel/boards/kbl_da7219_max98357a.c | 26 +++--- sound/soc/intel/boards/kbl_da7219_max98927.c | 54 +++++------- sound/soc/intel/boards/kbl_rt5660.c | 18 ++-- sound/soc/intel/boards/kbl_rt5663_max98927.c | 44 +++++----- .../intel/boards/kbl_rt5663_rt5514_max98927.c | 22 ++--- sound/soc/intel/boards/skl_hda_dsp_common.c | 14 ++- .../soc/intel/boards/skl_nau88l25_max98357a.c | 26 +++--- sound/soc/intel/boards/skl_nau88l25_ssm4567.c | 26 +++--- sound/soc/intel/boards/skl_rt286.c | 26 +++--- sound/soc/intel/boards/sof_board_helpers.c | 12 +-- sound/soc/intel/boards/sof_da7219.c | 33 +++---- sound/soc/intel/boards/sof_es8336.c | 8 +- sound/soc/intel/boards/sof_pcm512x.c | 8 +- sound/soc/intel/boards/sof_sdw.c | 4 +- sound/soc/intel/boards/sof_wm8804.c | 2 - sound/soc/mediatek/mt2701/mt2701-cs42448.c | 20 ++--- sound/soc/mediatek/mt2701/mt2701-wm8960.c | 6 +- sound/soc/mediatek/mt6797/mt6797-mt6351.c | 24 ++--- sound/soc/mediatek/mt7986/mt7986-wm8960.c | 6 +- sound/soc/mediatek/mt8173/mt8173-max98090.c | 6 +- .../mediatek/mt8173/mt8173-rt5650-rt5514.c | 6 +- .../mediatek/mt8173/mt8173-rt5650-rt5676.c | 10 +-- sound/soc/mediatek/mt8173/mt8173-rt5650.c | 10 +-- .../mediatek/mt8183/mt8183-da7219-max98357.c | 34 +++---- .../mt8183/mt8183-mt6358-ts3a227-max98357.c | 34 +++---- .../mt8186/mt8186-mt6366-da7219-max98357.c | 86 +++++++----------- .../mt8186/mt8186-mt6366-rt1019-rt5682s.c | 86 +++++++----------- sound/soc/mediatek/mt8188/mt8188-mt6359.c | 58 ++++++------ .../mt8192/mt8192-mt6359-rt1015-rt5682.c | 78 ++++++++-------- sound/soc/mediatek/mt8195/mt8195-mt6359.c | 60 ++++++------- sound/soc/meson/axg-card.c | 9 +- sound/soc/meson/gx-card.c | 1 - sound/soc/meson/meson-card-utils.c | 4 +- sound/soc/qcom/common.c | 1 - sound/soc/samsung/odroid.c | 11 ++- sound/soc/soc-compress.c | 10 ++- sound/soc/soc-core.c | 20 +---- sound/soc/soc-dai.c | 38 -------- sound/soc/soc-pcm.c | 88 ++++++++----------- sound/soc/soc-topology-test.c | 2 - sound/soc/soc-topology.c | 4 +- sound/soc/sof/nocodec.c | 4 - 91 files changed, 507 insertions(+), 876 deletions(-) -- 2.25.1

4 25

[PATCH 00/34] ASoC: Intel: boards: updates for 6.10 - part2
by Pierre-Louis Bossart 27 Mar '24

27 Mar '24

This second part provides SoundWire-related cleanups and extensions required by Realtek RT722 and Cirrus Logic codecs. Also included is a cleanup of the RT715-sdca DAI naming and new tables for ACPI-based board detections. Balamurugan C (1): ASoC: Intel: soc-acpi-intel-arl-match: Add rt711 sdca codec support Bard Liao (7): ASoC: Intel: sof_sdw: use generic rtd_init function for Realtek SDW DMICs ASoC: Intel: sof_sdw: remove unused rt dmic rtd_init ASoC: Intel: sof_sdw_rt722_sdca: set rtd_init in codec_info_list[] ASoC: Intel: sof_sdw_rt722_sdca: use rt_dmic_rtd_init ASoC: rt715-sdca: rename dai name with rt715-sdca prefix ASoC: Intel: sof_sdw: change rt715-sdca dai name ASoC: Intel: change cs35l56 name_prefix Charles Keepax (26): ASoC: intel: sof_sdw: Make find_codec_info_part() return a pointer ASoC: intel: sof_sdw: Make find_codec_info_acpi() return a pointer ASoC: intel: sof_sdw: Make find_codec_info_dai() return a pointer ASoC: intel: sof_sdw: Only pass codec_conf pointer around ASoC: intel: sof_sdw: Set channel map directly from endpoints ASoC: Intel: sof_sdw: Move get_codec_dai_by_name() into sof_sdw itself ASoC: Intel: sof_sdw: Move flags to private struct ASoC: Intel: sof_sdw: Only pass dai_link pointer around ASoC: Intel: sof_sdw: Use for_each_set_bit ASoC: Intel: sof_sdw: Factor out SSP DAI creation ASoC: Intel: sof_sdw: Factor out DMIC DAI creation. ASoC: Intel: sof_sdw: Factor out HDMI DAI creation ASoC: Intel: sof_sdw: Factor out BlueTooth DAI creation ASoC: Intel: sof_sdw: Factor out codec name generation ASoC: Intel: sof_sdw: Remove no longer supported quirk ASoC: intel: soc-acpi: Add missing cs42l43 endpoints ASoC: Intel: sof-sdw: Add new code for parsing the snd_soc_acpi structs ASoC: Intel: sof_sdw: Move counting and codec_conf to new parsing ASoC: Intel: sof_sdw: Move ignore_pch_dmic to new parsing ASoC: Intel: sof_sdw: Move append_dai_type to new parsing ASoC: Intel: sof_sdw: Move generation of DAI links to new parsing ASoC: intel: sof_sdw: Factor out SoundWire DAI creation ASoC: Intel: sof_sdw: Don't pass acpi_link_adr to init functions ASoC: Intel: sof_sdw: Remove redundant initialisations ASoC: Intel: sof_sdw: Add quirk for optional codec speakers ASoC: Intel: sof_sdw: Add support for cs42l43 optional speaker output sound/soc/codecs/rt715-sdca.c | 4 +- sound/soc/intel/boards/Kconfig | 1 - sound/soc/intel/boards/Makefile | 4 +- sound/soc/intel/boards/sof_board_helpers.c | 18 - sound/soc/intel/boards/sof_board_helpers.h | 3 - sound/soc/intel/boards/sof_sdw.c | 1118 ++++++++--------- sound/soc/intel/boards/sof_sdw_common.h | 42 +- sound/soc/intel/boards/sof_sdw_cs42l42.c | 1 - sound/soc/intel/boards/sof_sdw_cs42l43.c | 50 + sound/soc/intel/boards/sof_sdw_cs_amp.c | 1 - sound/soc/intel/boards/sof_sdw_maxim.c | 1 - sound/soc/intel/boards/sof_sdw_rt5682.c | 1 - sound/soc/intel/boards/sof_sdw_rt700.c | 1 - sound/soc/intel/boards/sof_sdw_rt711.c | 2 - sound/soc/intel/boards/sof_sdw_rt712_sdca.c | 25 - sound/soc/intel/boards/sof_sdw_rt715.c | 26 - sound/soc/intel/boards/sof_sdw_rt715_sdca.c | 26 - sound/soc/intel/boards/sof_sdw_rt722_sdca.c | 38 +- sound/soc/intel/boards/sof_sdw_rt_amp.c | 1 - sound/soc/intel/boards/sof_sdw_rt_dmic.c | 54 + .../boards/sof_sdw_rt_sdca_jack_common.c | 4 +- sound/soc/intel/boards/sof_ssp_common.h | 7 - .../intel/common/soc-acpi-intel-arl-match.c | 24 + .../intel/common/soc-acpi-intel-mtl-match.c | 35 +- .../intel/common/soc-acpi-intel-tgl-match.c | 35 +- 25 files changed, 753 insertions(+), 769 deletions(-) delete mode 100644 sound/soc/intel/boards/sof_sdw_rt715.c delete mode 100644 sound/soc/intel/boards/sof_sdw_rt715_sdca.c create mode 100644 sound/soc/intel/boards/sof_sdw_rt_dmic.c -- 2.40.1

2 35

[PATCH 00/25] Introduce support of audio for Amlogic A1 SoC family
by Jan Dakinevich 27 Mar '24

27 Mar '24

This series includes the following: - new audio clock and reset controller data and adaptation for it of existing code (patches 0001..0004); - adaptation of existing audio components for A1 Soc (patches 0005..0021); - handy cosmetics for dai-link naming (patches 0022..0023); - integration of audio devices into common trees (patch 0024); - audio support bring up on Amlogic ad402 reference board (patch 0025). This patch is not actually checked on real hardware (because all ad402 that we had were burned out). This patch is based on ad402's schematics and on experience with our own hardware (which is very close to reference board); Dmitry Rokosov (2): ASoC: dt-bindings: meson: introduce link-name optional property ASoC: meson: implement link-name optional property in meson card utils Jan Dakinevich (23): clk: meson: a1: restrict an amount of 'hifi_pll' params clk: meson: axg: move reset controller's code to separate module dt-bindings: clock: meson: add A1 audio clock and reset controller bindings clk: meson: a1: add the audio clock controller driver ASoC: meson: codec-glue: add support for capture stream ASoC: meson: g12a-toacodec: fix "Lane Select" width ASoC: meson: g12a-toacodec: rework the definition of bits ASoC: dt-bindings: meson: g12a-toacodec: add support for A1 SoC family ASoC: meson: g12a-toacodec: add support for A1 SoC family ASoC: meson: t9015: prepare to adding new platforms ASoC: dt-bindings: meson: t9015: add support for A1 SoC family ASoC: meson: t9015: add support for A1 SoC family ASoC: dt-bindings: meson: axg-pdm: document 'sysrate' property ASoC: meson: axg-pdm: introduce 'sysrate' property pinctrl/meson: fix typo in PDM's pin name ASoC: dt-bindings: meson: meson-axg-audio-arb: claim support of A1 SoC family ASoC: dt-bindings: meson: axg-fifo: claim support of A1 SoC family ASoC: dt-bindings: meson: axg-pdm: claim support of A1 SoC family ASoC: dt-bindings: meson: axg-sound-card: claim support of A1 SoC family ASoC: dt-bindings: meson: axg-tdm-formatters: claim support of A1 SoC family ASoC: dt-bindings: meson: axg-tdm-iface: claim support of A1 SoC family arm64: dts: meson: a1: add audio devices arm64: dts: ad402: enable audio .../bindings/clock/amlogic,a1-audio-clkc.yaml | 83 +++ .../reset/amlogic,meson-axg-audio-arb.yaml | 10 +- .../bindings/sound/amlogic,axg-fifo.yaml | 8 + .../bindings/sound/amlogic,axg-pdm.yaml | 5 + .../sound/amlogic,axg-sound-card.yaml | 12 +- .../sound/amlogic,axg-tdm-formatters.yaml | 22 +- .../bindings/sound/amlogic,axg-tdm-iface.yaml | 6 +- .../bindings/sound/amlogic,g12a-toacodec.yaml | 1 + .../bindings/sound/amlogic,gx-sound-card.yaml | 6 + .../bindings/sound/amlogic,t9015.yaml | 4 +- .../arm64/boot/dts/amlogic/meson-a1-ad402.dts | 126 ++++ arch/arm64/boot/dts/amlogic/meson-a1.dtsi | 471 +++++++++++++++ drivers/clk/meson/Kconfig | 18 + drivers/clk/meson/Makefile | 2 + drivers/clk/meson/a1-audio.c | 556 ++++++++++++++++++ drivers/clk/meson/a1-audio.h | 58 ++ drivers/clk/meson/a1-pll.c | 8 +- drivers/clk/meson/axg-audio.c | 95 +-- drivers/clk/meson/meson-audio-rstc.c | 109 ++++ drivers/clk/meson/meson-audio-rstc.h | 12 + drivers/pinctrl/meson/pinctrl-meson-a1.c | 6 +- .../dt-bindings/clock/amlogic,a1-audio-clkc.h | 122 ++++ .../reset/amlogic,meson-a1-audio-reset.h | 29 + .../dt-bindings/sound/meson-g12a-toacodec.h | 5 + sound/soc/meson/axg-pdm.c | 10 +- sound/soc/meson/g12a-toacodec.c | 298 ++++++++-- sound/soc/meson/meson-card-utils.c | 12 +- sound/soc/meson/meson-codec-glue.c | 174 ++++-- sound/soc/meson/meson-codec-glue.h | 23 + sound/soc/meson/t9015.c | 326 +++++++++- 30 files changed, 2394 insertions(+), 223 deletions(-) create mode 100644 Documentation/devicetree/bindings/clock/amlogic,a1-audio-clkc.yaml create mode 100644 drivers/clk/meson/a1-audio.c create mode 100644 drivers/clk/meson/a1-audio.h create mode 100644 drivers/clk/meson/meson-audio-rstc.c create mode 100644 drivers/clk/meson/meson-audio-rstc.h create mode 100644 include/dt-bindings/clock/amlogic,a1-audio-clkc.h create mode 100644 include/dt-bindings/reset/amlogic,meson-a1-audio-reset.h -- 2.34.1

8 84

[PATCH v2 0/4] ALSA: hda/tas2781: fixes for 6.9-rc1
by Gergo Koteles 27 Mar '24

27 Mar '24

Hi, This series removes the no longer needed digital gain kcontrol, which caused problems in tas2563, because the register does not exists there. This series also adds locking and debug statements to the other kcontrols. They sometimes ran in parallel with tasdev_fw_ready, and caused weird sound problems. https://github.com/tomsom/yoga-linux/issues/58 Regards, Gergo Changes in v2: - Do not remove dvc_tlv scale from tas2781-tlv.h as it is also used by sound/soc/codecs/tas2781-i2c.c - Add kcontrol name to debug statements - Add a new patch to remove a useless debug statement. [1]: https://lore.kernel.org/all/cover.1711401621.git.soyer@irl.hu/ Gergo Koteles (4): ALSA: hda/tas2781: remove digital gain kcontrol ALSA: hda/tas2781: add locks to kcontrols ALSA: hda/tas2781: add debug statements to kcontrols ALSA: hda/tas2781: remove useless dev_dbg from playback_hook sound/pci/hda/tas2781_hda_i2c.c | 118 +++++++++++++++++++++----------- 1 file changed, 77 insertions(+), 41 deletions(-) base-commit: 4cece764965020c22cff7665b18a012006359095 -- 2.44.0

2 5

How to add jack GPIOs to audio-graph-cards?
by John Watts 27 Mar '24

27 Mar '24

Hi there, audio-graph-card's device tree schema indicates it supports the following: - pa-gpios - hp-det-gpio - mic-det-gpio But only implements pa-gpios. audio-graph-card2 doesn't implement any of these but will still attempts to grab pa-gpios but never use it (possibly a bug?) When using audio graph cards, how are you intended to add detection GPIOs? Are there future plans to add a jacks { } section of some sort for this? John.

2 1

Writing documentation for simple device tree ASoC cards
by John Watts 27 Mar '24

27 Mar '24

Hello! I've spent way too much time digging through emails, documentation and kernel source code to form a mental model that lets me understand and use simple-sound-card, audio-graph-card and audio-graph-card2. There's really no introductory documentation geared towards people who are writing device trees intending to glue together multiple audio components in to a single sound card. I'm deciding I'll try and take up this task to provide some basic documentation for this. Before starting I would like to quickly go over some notes in case I'm actually just incredibly wrong. I'd rather find that out now than later. As a general ASoC refresher: - ALSA provides PCM interfaces and controls to a sound card - ASoC introduces the concept of components - Components are what you instantiate with the device tree - Components can be asked to set a clock or create a PCM stream - Components have one or more DAIs - DAIs represent capabilities of a stream - Components have DAPM widgets for power management - These can be related to physical pins on the board or DAI streams - CPU components are ones that ALSA can use directly to process data - Codec components provide information about DAIs or drive streams Most of this is already explained in the documentation. The generic sound cards you can create using simple-sound-card and audio-graph-card are machine drivers. Their jobs are to ensure that when using a DAI all the related components, clocking and power management is set up and ready for use. This is done by: - Linking multiple DAIs together - Connecting DAPM widgets together to power on components - Calculate and setting the shared clock (usually mclk) - Setting DAI properties (like TDM slots and width) - Adding widgets for jack detection DAI linking isn't explained in the documentation but was a core part of my confusion when trying to use these cards: I didn't understand if it represented some relationship between physical pins, PCM channels, etc. My fuzzy model of a DAI link is that it exists to group DAIs (and thus components) necessary to acheive a desired data flow, where that flow is left up to the components themselves to implement. There's a lot of examples for actually using the simple family of cards but I imagine documenting the above would help people troubleshoot or describe their problems better. Other minor notes: - sound-name-prefix is very useful and fits in somewhere here - audio-graph-cards don't support GPIO jacks from what I can see I also have noticed that with audio-graph-card you can't use a DAI multiple times in different cards. This makes sense but is this due to the way graphs and endpoints work or due to the underlying sound system? This problem and jacks seem to point towards maybe allowing specification of DAPM widgets in the device tree, and maybe kcontrols for selecting which codec to use? I'm not sure. It would also be useful to understand the scope of these simple cards. Complex audio needs like mine seem to be outside the scope at the moment and require writing a custom card driver. Perhaps it's worth taking that as an opportunity to write a driver using the simple card framework as an example? John.

1 0

[PATCH 0/3] ALSA: hda/tas2781: fixes for 6.9-rc1
by Gergo Koteles 27 Mar '24

27 Mar '24

Hi, This series removes the no longer needed digital gain kcontrol, which caused problems in tas2563, because the register does not exists there. This series also adds locking and debug statements to the other kcontrols. They sometimes ran in parallel with tasdev_fw_ready, and caused weird sound problems. https://github.com/tomsom/yoga-linux/issues/58 Regards, Gergo Gergo Koteles (3): ALSA: hda/tas2781: remove digital gain kcontrol ALSA: hda/tas2781: add locks to kcontrols ALSA: hda/tas2781: add debug statements to kcontrols include/sound/tas2781-tlv.h | 1 - sound/pci/hda/tas2781_hda_i2c.c | 103 +++++++++++++++++++++----------- 2 files changed, 68 insertions(+), 36 deletions(-) base-commit: 4cece764965020c22cff7665b18a012006359095 -- 2.44.0

5 9

[PATCH] ASoC: sti: uniperif: fix the undefined bitwise shift behavior problem
by Su Hui 27 Mar '24

27 Mar '24

Clang static checker(scan-build): sound/soc/sti/uniperif_player.c:1115:12: warning: The result of the left shift is undefined because the right operand is negative [core.UndefinedBinaryOperatorResult] When UNIPERIF_CONFIG_BACK_STALL_REQ_SHIFT(ip) equals to -1, the result of SET_UNIPERIF_CONFIG_BACK_STALL_REQ_DISABLE(ip) is undefined. Here are some results of using different compilers. 1UL >> -1 1UL << -1 gcc 10.2.1 0x2 0 gcc 11.4.0 0 0x8000000000000000 clang 14.0.0 0x64b8a45d72a0 0x64b8a45d72a0 clang 17.0.0 0x556c43b0f2a0 0x556c43b0f2a0 Add some macros to ensure that when right opreand is negative or other invalid values, the results of bitwise shift is zero. Fixes: e1ecace6a685 ("ASoC: sti: Add uniperipheral header file") Signed-off-by: Su Hui <suhui(a)nfschina.com> --- sound/soc/sti/uniperif.h | 17 ++++++++++++++--- 1 file changed, 14 insertions(+), 3 deletions(-) diff --git a/sound/soc/sti/uniperif.h b/sound/soc/sti/uniperif.h index 2a5de328501c..1cbff01fbff0 100644 --- a/sound/soc/sti/uniperif.h +++ b/sound/soc/sti/uniperif.h @@ -12,17 +12,28 @@ #include <sound/dmaengine_pcm.h> +#define SR_SHIFT(a, b) ({unsigned long __a = (a); \ + unsigned int __b = (b); \ + __b < BITS_PER_LONG ? \ + __a >> __b : 0; }) + +#define SL_SHIFT(a, b) ({unsigned long __a = (a); \ + unsigned int __b = (b); \ + __b < BITS_PER_LONG ? \ + __a << __b : 0; }) + /* * Register access macros */ #define GET_UNIPERIF_REG(ip, offset, shift, mask) \ - ((readl_relaxed(ip->base + offset) >> shift) & mask) + (SR_SHIFT(readl_relaxed(ip->base + offset), shift) & mask) #define SET_UNIPERIF_REG(ip, offset, shift, mask, value) \ writel_relaxed(((readl_relaxed(ip->base + offset) & \ - ~(mask << shift)) | (((value) & mask) << shift)), ip->base + offset) + ~SL_SHIFT(mask, shift)) | SL_SHIFT(((value) & mask), shift)),\ + ip->base + offset) #define SET_UNIPERIF_BIT_REG(ip, offset, shift, mask, value) \ - writel_relaxed((((value) & mask) << shift), ip->base + offset) + writel_relaxed(SL_SHIFT(((value) & mask), shift), ip->base + offset) /* * UNIPERIF_SOFT_RST reg -- 2.30.2

2 3

[PATCH v4 0/7] Add support for the internal RK3308 audio codec
by Luca Ceresoli 26 Mar '24

26 Mar '24

This series adds a driver for the internal audio codec of the Rockchip RK3308 SoC, along with some related patches. This codec is internally connected to the I2S peripherals on the same chip, and it has some peculiarities arising from that interconnection. For proper bidirectional operation with the internal codec at any possible combination of sampling rates, the I2S peripheral needs two clock sources (tx and rx), while connection with an external codec commonly needs only one. Since v5.16 there is a driver for the I2S in sound/soc/rockchip/rockchip_i2s_tdm.c, but in some cases it does not configure correctly the clocks, resulting in an unnecessarily inaccurate rate. Patch 1 fixes this. Patches 2-4 add the codec driver along with the bindings and a new helper macro. Patches 5-7 add to the SoC DT file two I2S controllers (those which are internally connected to the internal codec) and the codec itself and enable the driver in the ARM64 defconfig. Luca Signed-off-by: Luca Ceresoli <luca.ceresoli(a)bootlin.com> --- Changes in v4: - several cleanups in the codec probe function - Link to v3: https://lore.kernel.org/r/20240221-rk3308-audio-codec-v3-0-dfa34abfcef6@boo… Changes in v3: - Add the I2S clock fix patch and remove a previous fix which is now superseded - Codec driver: fix silent playback until a given amplitude of sigital value, seen at >= 96 kHz rate - various other changes, listed per-patch - Link to v2: https://lore.kernel.org/r/20231219-rk3308-audio-codec-v2-0-c70d06021946@boo… Changes in v2: - largely rewrote the codec driver to use DAPM and lots of improvements and cleanups - removed the RK3308 audio card and related patches - various other changes, listed per-patch - Link to v1: https://lore.kernel.org/all/20220907142124.2532620-1-luca.ceresoli@bootlin.… --- Luca Ceresoli (7): ASoC: rockchip: i2s-tdm: Fix inaccurate sampling rates ASoC: dt-bindings: Add Rockchip RK3308 internal audio codec ASoC: core: add SOC_DOUBLE_RANGE_TLV() helper macro ASoC: codecs: Add RK3308 internal audio codec driver arm64: defconfig: enable Rockchip RK3308 internal audio codec driver arm64: dts: rockchip: add i2s_8ch_2 and i2s_8ch_3 arm64: dts: rockchip: add the internal audio codec .../bindings/sound/rockchip,rk3308-codec.yaml | 98 +++ MAINTAINERS | 7 + arch/arm64/boot/dts/rockchip/rk3308.dtsi | 56 ++ arch/arm64/configs/defconfig | 1 + include/sound/soc.h | 12 + sound/soc/codecs/Kconfig | 11 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/rk3308_codec.c | 974 +++++++++++++++++++++ sound/soc/codecs/rk3308_codec.h | 579 ++++++++++++ sound/soc/rockchip/rockchip_i2s_tdm.c | 352 +------- 10 files changed, 1746 insertions(+), 346 deletions(-) --- base-commit: dfda120c512b3edca1436f770924e91b14f93a98 change-id: 20231219-rk3308-audio-codec-a5558ba8949d Best regards, -- Luca Ceresoli <luca.ceresoli(a)bootlin.com>

4 12

[RFC PATCH v2] ASoC: meson: g12a-toacodec: rework the definition of bits
by Jan Dakinevich 26 Mar '24

26 Mar '24

There is a lot of defines, but almost all of them are not used. Lets rework them: - keep separate the definition for different platforms to make easier checking that they match documentation. - use LSB/MSB sufixes for uniformity. - don't use hard-coded values for already declared defines. Signed-off-by: Jan Dakinevich <jan.dakinevich(a)salutedevices.com> --- Links: [1] https://lore.kernel.org/lkml/20240314232201.2102178-1-jan.dakinevich@salute… Changes v1 -> v2: - Detached from v1's series (patch 7). - Fixed my wrong understanding of SOC_SINGLE's input parameters. sound/soc/meson/g12a-toacodec.c | 79 ++++++++++++++++++++------------- 1 file changed, 49 insertions(+), 30 deletions(-) diff --git a/sound/soc/meson/g12a-toacodec.c b/sound/soc/meson/g12a-toacodec.c index 531bb8707a3e..22181f4bab72 100644 --- a/sound/soc/meson/g12a-toacodec.c +++ b/sound/soc/meson/g12a-toacodec.c @@ -20,26 +20,37 @@ #define G12A_TOACODEC_DRV_NAME "g12a-toacodec" #define TOACODEC_CTRL0 0x0 -#define CTRL0_ENABLE_SHIFT 31 -#define CTRL0_DAT_SEL_SM1_MSB 19 -#define CTRL0_DAT_SEL_SM1_LSB 18 -#define CTRL0_DAT_SEL_MSB 15 -#define CTRL0_DAT_SEL_LSB 14 -#define CTRL0_LANE_SEL_SM1 16 -#define CTRL0_LANE_SEL 12 -#define CTRL0_LRCLK_SEL_SM1_MSB 14 -#define CTRL0_LRCLK_SEL_SM1_LSB 12 -#define CTRL0_LRCLK_SEL_MSB 9 -#define CTRL0_LRCLK_SEL_LSB 8 -#define CTRL0_LRCLK_INV_SM1 BIT(10) -#define CTRL0_BLK_CAP_INV_SM1 BIT(9) -#define CTRL0_BLK_CAP_INV BIT(7) -#define CTRL0_BCLK_O_INV_SM1 BIT(8) -#define CTRL0_BCLK_O_INV BIT(6) -#define CTRL0_BCLK_SEL_SM1_MSB 6 -#define CTRL0_BCLK_SEL_MSB 5 -#define CTRL0_BCLK_SEL_LSB 4 -#define CTRL0_MCLK_SEL GENMASK(2, 0) + +/* Common bits */ +#define CTRL0_ENABLE_SHIFT 31 +#define CTRL0_MCLK_SEL GENMASK(2, 0) + +/* G12A bits */ +#define CTRL0_DAT_SEL_G12A_MSB 15 +#define CTRL0_DAT_SEL_G12A_LSB 14 +#define CTRL0_LANE_SEL_G12A_MSB 13 +#define CTRL0_LANE_SEL_G12A_LSB 12 +#define CTRL0_LANE_SEL_G12A_MAX 3 +#define CTRL0_LRCLK_SEL_G12A_MSB 9 +#define CTRL0_LRCLK_SEL_G12A_LSB 8 +#define CTRL0_BLK_CAP_INV_G12A BIT(7) +#define CTRL0_BCLK_O_INV_G12A BIT(6) +#define CTRL0_BCLK_SEL_G12A_MSB 5 +#define CTRL0_BCLK_SEL_G12A_LSB 4 + +/* SM1 bits */ +#define CTRL0_DAT_SEL_SM1_MSB 19 +#define CTRL0_DAT_SEL_SM1_LSB 18 +#define CTRL0_LANE_SEL_SM1_MSB 17 +#define CTRL0_LANE_SEL_SM1_LSB 16 +#define CTRL0_LANE_SEL_SM1_MAX 3 +#define CTRL0_LRCLK_SEL_SM1_MSB 14 +#define CTRL0_LRCLK_SEL_SM1_LSB 12 +#define CTRL0_LRCLK_INV_SM1 BIT(10) +#define CTRL0_BLK_CAP_INV_SM1 BIT(9) +#define CTRL0_BCLK_O_INV_SM1 BIT(8) +#define CTRL0_BCLK_SEL_SM1_MSB 6 +#define CTRL0_BCLK_SEL_SM1_LSB 4 #define TOACODEC_OUT_CHMAX 2 @@ -108,7 +119,7 @@ static int g12a_toacodec_mux_put_enum(struct snd_kcontrol *kcontrol, } static SOC_ENUM_SINGLE_DECL(g12a_toacodec_mux_enum, TOACODEC_CTRL0, - CTRL0_DAT_SEL_LSB, + CTRL0_DAT_SEL_G12A_LSB, g12a_toacodec_mux_texts); static SOC_ENUM_SINGLE_DECL(sm1_toacodec_mux_enum, TOACODEC_CTRL0, @@ -210,7 +221,7 @@ static int g12a_toacodec_component_probe(struct snd_soc_component *c) { /* Initialize the static clock parameters */ return snd_soc_component_write(c, TOACODEC_CTRL0, - CTRL0_BLK_CAP_INV); + CTRL0_BLK_CAP_INV_G12A); } static int sm1_toacodec_component_probe(struct snd_soc_component *c) @@ -229,11 +240,13 @@ static const struct snd_soc_dapm_route g12a_toacodec_routes[] = { }; static const struct snd_kcontrol_new g12a_toacodec_controls[] = { - SOC_SINGLE("Lane Select", TOACODEC_CTRL0, CTRL0_LANE_SEL, 3, 0), + SOC_SINGLE("Lane Select", TOACODEC_CTRL0, CTRL0_LANE_SEL_G12A_LSB, + CTRL0_LANE_SEL_G12A_MAX, 0), }; static const struct snd_kcontrol_new sm1_toacodec_controls[] = { - SOC_SINGLE("Lane Select", TOACODEC_CTRL0, CTRL0_LANE_SEL_SM1, 3, 0), + SOC_SINGLE("Lane Select", TOACODEC_CTRL0, CTRL0_LANE_SEL_SM1_LSB, + CTRL0_LANE_SEL_SM1_MAX, 0), }; static const struct snd_soc_component_driver g12a_toacodec_component_drv = { @@ -266,16 +279,22 @@ static const struct regmap_config g12a_toacodec_regmap_cfg = { static const struct g12a_toacodec_match_data g12a_toacodec_match_data = { .component_drv = &g12a_toacodec_component_drv, - .field_dat_sel = REG_FIELD(TOACODEC_CTRL0, 14, 15), - .field_lrclk_sel = REG_FIELD(TOACODEC_CTRL0, 8, 9), - .field_bclk_sel = REG_FIELD(TOACODEC_CTRL0, 4, 5), + .field_dat_sel = REG_FIELD(TOACODEC_CTRL0, CTRL0_DAT_SEL_G12A_LSB, + CTRL0_DAT_SEL_G12A_MSB), + .field_lrclk_sel = REG_FIELD(TOACODEC_CTRL0, CTRL0_LRCLK_SEL_G12A_LSB, + CTRL0_LRCLK_SEL_G12A_MSB), + .field_bclk_sel = REG_FIELD(TOACODEC_CTRL0, CTRL0_BCLK_SEL_G12A_LSB, + CTRL0_BCLK_SEL_G12A_MSB), }; static const struct g12a_toacodec_match_data sm1_toacodec_match_data = { .component_drv = &sm1_toacodec_component_drv, - .field_dat_sel = REG_FIELD(TOACODEC_CTRL0, 18, 19), - .field_lrclk_sel = REG_FIELD(TOACODEC_CTRL0, 12, 14), - .field_bclk_sel = REG_FIELD(TOACODEC_CTRL0, 4, 6), + .field_dat_sel = REG_FIELD(TOACODEC_CTRL0, CTRL0_DAT_SEL_SM1_LSB, + CTRL0_DAT_SEL_SM1_MSB), + .field_lrclk_sel = REG_FIELD(TOACODEC_CTRL0, CTRL0_LRCLK_SEL_SM1_LSB, + CTRL0_LRCLK_SEL_SM1_MSB), + .field_bclk_sel = REG_FIELD(TOACODEC_CTRL0, CTRL0_BCLK_SEL_SM1_LSB, + CTRL0_BCLK_SEL_SM1_MSB), }; static const struct of_device_id g12a_toacodec_of_match[] = { -- 2.34.1

2 2

[PATCH 00/21] ASoC: Intel: boards: updates for 6.10 - part1
by Pierre-Louis Bossart 26 Mar '24

26 Mar '24

We have quite a few machine driver patches, spread in 3 series of about 20 patches each. This first part from Brent Lu adds common helpers and board configurations to reduce the number of quirks. Brent Lu (21): ASoC: Intel: board_helpers: support DAI link ID customization ASoC: Intel: sof_ssp_amp: use common module for DAI link generation ASoC: Intel: board_helpers: change dai link helpers to static function ASoC: Intel: sof_da7219: add rpl_mx98360_da7219 board config ASoC: Intel: sof_rt5682: support ALC5650 on RPL boards ASoC: Intel: sof_cs42l42: rename BT offload quirk ASoC: Intel: board_helpers: support sof_card_private initialization ASoC: Intel: sof_cs42l42: use common module for sof_card_private initialization ASoC: Intel: sof_nau8825: use common module for sof_card_private initialization ASoC: Intel: sof_rt5682: use common module for sof_card_private initialization ASoC: Intel: sof_ssp_amp: use common module for sof_card_private initialization ASoC: Intel: sof_da7219: use common module for DAI link generation ASoC: Intel: sof_da7219: add codec exit function ASoC: Intel: sof_da7219: add SOF_DA7219_MCLK_EN quirk ASoC: Intel: sof_da7219: board id cleanup for jsl boards ASoC: Intel: sof_da7219: board id cleanup for adl boards ASoC: Intel: sof_da7219: board id cleanup for rpl boards ASoC: Intel: sof_rt5682: remove unnecessary idisp HDMI quirk ASoC: Intel: sof_ssp_amp: remove unnecessary idisp HDMI quirk ASoC: Intel: sof_nau8825: remove sof_nau8825 board id ASoC: Intel: sof_rt5682: board id cleanup for cml boards sound/soc/intel/boards/Kconfig | 1 + sound/soc/intel/boards/sof_board_helpers.c | 146 ++++--- sound/soc/intel/boards/sof_board_helpers.h | 97 ++++- sound/soc/intel/boards/sof_cs42l42.c | 72 +-- sound/soc/intel/boards/sof_da7219.c | 410 ++++++------------ sound/soc/intel/boards/sof_nau8825.c | 92 +--- sound/soc/intel/boards/sof_rt5682.c | 222 ++++------ sound/soc/intel/boards/sof_ssp_amp.c | 264 ++++------- .../intel/common/soc-acpi-intel-adl-match.c | 4 +- .../intel/common/soc-acpi-intel-cml-match.c | 6 +- .../intel/common/soc-acpi-intel-jsl-match.c | 4 +- .../intel/common/soc-acpi-intel-rpl-match.c | 12 + 12 files changed, 518 insertions(+), 812 deletions(-) -- 2.40.1

2 22

[PATCH v1 0/2] ASoC: codecs: ES8326: Reducin powerconsumption and
by Zhang Yi 26 Mar '24

26 Mar '24

We changed the configuration related to hibernation. and delete the REG_SUPPLY to cover mute issue. Zhang Yi (2): ASoC: codecs: ES8326: Reducing power consumption ASoC: codecs: ES8326: Delete unused REG_SUPPLY sound/soc/codecs/es8326.c | 41 ++++++++++++++++++++++++++++++--------- 1 file changed, 32 insertions(+), 9 deletions(-) -- 2.17.1

2 3

[PATCH v3 00/22] SoC: Cleanup MediaTek soundcard machine drivers
by AngeloGioacchino Del Regno 26 Mar '24

26 Mar '24

Changes in v3: - Added audio-routing names in enum in all yaml files - Added mention of disallowing old and new properties together in commit message of bindings patches - Fixed validation errors with sound-card-common.yaml inclusion due to missing model property in examples - Removed `else` enforcing headset-codec/speaker-codecs requirement if xxx-dai-link not present to avoid future commit noise as the deprecated statement will disallow deprecated properties as required Changes in v2: - Bindings: Changed link-name/codec/clk-provider to remove `items` and leave just the enum - Moved .*-dai-link pattern additionalProperties after `type: object` - Added ref to sound-card-common.yaml - Fixed dai-link-xxx -> xxx-dai-link typo in example comment This series performs a cleanup of most of the MediaTek AFE drivers and soundcard machine drivers, reducing code duplication and setting a base to be consistent with their devicetree bindings, as many of those are using different properties and nodes for no good reason. Summarizing: - Commonizes functions and ops where possible - Adds a common probe mechanism, increasing maintainability of soundcard drivers for older MediaTek SoCs - Migrates all drivers to support the new bindings - Obviously retains compatibility with old device trees - Reduces machine-specific parameters hardcoding in drivers - Can now set machine-specific params in device tree - Uses the `audio-routing` and `dai-link` nodes like some other non-MediaTek SoC sound drivers - Imposes consistency between MediaTek ASoC machine soundcard drivers bindings - Reduces code size and greatly reduces the amount of code that will be required for newer drivers (retaining compatibility with the old bindings was costly in terms of code size, otherwise this series would've removed ~1000 more lines, or something along that line). This series was (manually) tested on MT8173, MT8192, MT8195 and MT8186 Chromebooks. AngeloGioacchino Del Regno (22): ASoC: mediatek: mt8192-afe-pcm: Convert to devm_pm_runtime_enable() ASoC: mediatek: mt8192-afe-pcm: Simplify with dev_err_probe() ASoC: mediatek: Commonize ADDA rate transform functions and enums ASoC: mediatek: Assign dummy when codec not specified for a DAI link ASoC: mediatek: Add common machine soundcard driver probe mechanism ASoC: mediatek: common: Constify struct mtk_sof_priv ASoC: mediatek: mt8188: Migrate to mtk_soundcard_common_probe ASoC: mediatek: mt8195: Migrate to mtk_soundcard_common_probe ASoC: mediatek: mt8192: Migrate to mtk_soundcard_common_probe ASoC: mediatek: mt8186: Migrate to mtk_soundcard_common_probe ASoC: mediatek: Add common snd_soc_ops .startup() callback ASoC: mediatek: mt8195: Migrate to the common mtk_soundcard_startup ASoC: mediatek: mt8192: Migrate to the common mtk_soundcard_startup ASoC: mediatek: mt8186-rt1019: Migrate to the common mtk_soundcard_startup ASoC: mediatek: Add common mtk_afe_component_probe callback ASoC: mediatek: Use common mtk_afe_pcm_platform with common probe cb ASoC: mediatek: mt8186: Unify mt8186-mt6366 machine drivers ASoC: dt-bindings: mt8195: Document audio-routing and dai-link subnode ASoC: dt-bindings: mt8192: Document audio-routing and dai-link subnode ASoC: dt-bindings: mt8186: Document audio-routing and dai-link subnode arm64: dts: mediatek: mt8195-cherry: Specify sound DAI links and routing arm64: dts: mediatek: mt8186-corsola: Specify sound DAI links and routing .../sound/mt8186-mt6366-da7219-max98357.yaml | 131 +- .../sound/mt8186-mt6366-rt1019-rt5682s.yaml | 120 +- .../sound/mt8192-mt6359-rt1015-rt5682.yaml | 139 +- .../bindings/sound/mt8195-mt6359.yaml | 134 ++ .../boot/dts/mediatek/mt8186-corsola.dtsi | 42 +- .../boot/dts/mediatek/mt8195-cherry.dtsi | 45 + sound/soc/mediatek/Kconfig | 24 +- sound/soc/mediatek/common/Makefile | 2 + .../mediatek/common/mtk-afe-platform-driver.c | 18 + .../soc/mediatek/common/mtk-dai-adda-common.c | 70 + .../soc/mediatek/common/mtk-dai-adda-common.h | 45 + .../soc/mediatek/common/mtk-dsp-sof-common.c | 15 +- .../soc/mediatek/common/mtk-dsp-sof-common.h | 1 - sound/soc/mediatek/common/mtk-soc-card.h | 7 +- .../mediatek/common/mtk-soundcard-driver.c | 205 ++- .../mediatek/common/mtk-soundcard-driver.h | 42 + sound/soc/mediatek/mt6797/mt6797-afe-pcm.c | 14 +- sound/soc/mediatek/mt6797/mt6797-dai-adda.c | 85 +- sound/soc/mediatek/mt7986/mt7986-afe-pcm.c | 14 +- sound/soc/mediatek/mt8183/mt8183-afe-pcm.c | 14 +- sound/soc/mediatek/mt8183/mt8183-dai-adda.c | 90 +- sound/soc/mediatek/mt8186/Makefile | 3 +- sound/soc/mediatek/mt8186/mt8186-afe-pcm.c | 19 +- sound/soc/mediatek/mt8186/mt8186-dai-adda.c | 94 +- .../mt8186/mt8186-mt6366-da7219-max98357.c | 1189 ----------------- ...t6366-rt1019-rt5682s.c => mt8186-mt6366.c} | 578 ++++---- sound/soc/mediatek/mt8188/mt8188-afe-pcm.c | 21 +- sound/soc/mediatek/mt8188/mt8188-dai-adda.c | 89 +- sound/soc/mediatek/mt8188/mt8188-mt6359.c | 203 +-- sound/soc/mediatek/mt8192/mt8192-afe-pcm.c | 114 +- sound/soc/mediatek/mt8192/mt8192-dai-adda.c | 90 +- .../mt8192/mt8192-mt6359-rt1015-rt5682.c | 301 ++--- sound/soc/mediatek/mt8195/mt8195-afe-pcm.c | 21 +- sound/soc/mediatek/mt8195/mt8195-dai-adda.c | 90 +- sound/soc/mediatek/mt8195/mt8195-mt6359.c | 487 +++---- 35 files changed, 1781 insertions(+), 2775 deletions(-) create mode 100644 sound/soc/mediatek/common/mtk-dai-adda-common.c create mode 100644 sound/soc/mediatek/common/mtk-dai-adda-common.h delete mode 100644 sound/soc/mediatek/mt8186/mt8186-mt6366-da7219-max98357.c rename sound/soc/mediatek/mt8186/{mt8186-mt6366-rt1019-rt5682s.c => mt8186-mt6366.c} (72%) -- 2.44.0

3 28

[PATCH v2 0/5] ASoC: Harden DAPM route checks and Intel fixes
by Cezary Rojewski 26 Mar '24

26 Mar '24

Set of loosely connected patches. Most impactful change is dropping any permisiveness when snd_soc_dapm_add_routes() fails in soc-topology.c To do it safely, disable route checks for all skylake-driver boards. Relevant background: Since commit daa480bde6b3 ("ASoC: soc-core: tidyup for snd_soc_dapm_add_routes()") route checks are no longer permissive. Probe failures for Intel boards have been partially addressed by commit a22ae72b86a4 ("ASoC: soc-core: isable route checks for legacy devices") and its follow up but only skl_nau88l25_ssm4567.c is patched. The rest of the boards still need fixing. After that, removal of copy-pastas found in ssm4567.c and redundant code in i2s_test.c for avs-boards. Changes in v2: - glk_rt5682_max98357a.c and skl_hda_dsp_generic.c now disable route checks only for the skylake-drvier - asoc now logs failures of snd_soc_dapm_add_routes() in soc-topology.c Amadeusz Sławiński (1): ASoC: Intel: avs: i2s_test: Remove redundant dapm routes Cezary Rojewski (4): ASoC: Intel: Disable route checks for Skylake boards ASoC: topology: Do not ignore route checks when parsing graphs ASoC: Intel: avs: ssm4567: Do not ignore route checks ASoC: Intel: avs: ssm4567: Board cleanup sound/soc/intel/avs/boards/i2s_test.c | 79 ------------------- sound/soc/intel/avs/boards/ssm4567.c | 5 +- sound/soc/intel/boards/bxt_da7219_max98357a.c | 1 + sound/soc/intel/boards/bxt_rt298.c | 1 + sound/soc/intel/boards/glk_rt5682_max98357a.c | 2 + sound/soc/intel/boards/kbl_da7219_max98357a.c | 1 + sound/soc/intel/boards/kbl_da7219_max98927.c | 4 + sound/soc/intel/boards/kbl_rt5660.c | 1 + sound/soc/intel/boards/kbl_rt5663_max98927.c | 2 + .../intel/boards/kbl_rt5663_rt5514_max98927.c | 1 + sound/soc/intel/boards/skl_hda_dsp_generic.c | 2 + .../soc/intel/boards/skl_nau88l25_max98357a.c | 1 + sound/soc/intel/boards/skl_rt286.c | 1 + sound/soc/soc-topology.c | 11 ++- 14 files changed, 27 insertions(+), 85 deletions(-) -- 2.25.1

3 8

[PATCH v3 0/5] ASoC: fsl: Support register and unregister rpmsg sound card through remoteproc
by Chancel Liu 26 Mar '24

26 Mar '24

echo /lib/firmware/fw.elf > /sys/class/remoteproc/remoteproc0/firmware (A) echo start > /sys/class/remoteproc/remoteproc0/state (B) echo stop > /sys/class/remoteproc/remoteproc0/state The rpmsg sound card is registered in (A) and unregistered in (B). After "start", imx-audio-rpmsg registers devices for ASoC platform driver and machine driver. Then sound card is registered. After "stop", imx-audio-rpmsg unregisters devices for ASoC platform driver and machine driver. Then sound card is unregistered. changes in v2 - Fix build errors reported by kernel test robot changes in v3 - Add a new patch for fsl_rpmsg to register CPU DAI with rpmsg channel name - Update imx-rpmsg.c to get DT node of ASoC CPU DAI device with rpmsg channel name instead of using undocumented bindings Chancel Liu (5): ASoC: fsl: imx-pcm-rpmsg: Register component with rpmsg channel name ASoC: fsl: imx-audio-rpmsg: Register device with rpmsg channel name ASoC: fsl: Let imx-audio-rpmsg register platform device for card ASoC: fsl: fsl_rpmsg: Register CPU DAI with name of rpmsg channel ASoC: fsl: imx-rpmsg: Update to correct DT node sound/soc/fsl/fsl_rpmsg.c | 43 ++++++++++++++++++++------------- sound/soc/fsl/imx-audio-rpmsg.c | 21 +++++++++++++--- sound/soc/fsl/imx-pcm-rpmsg.c | 11 ++++++--- sound/soc/fsl/imx-rpmsg.c | 28 ++++++++++++++------- 4 files changed, 71 insertions(+), 32 deletions(-) -- 2.43.0

3 7

[PATCH 0/7] ASoC: mediatek: AFE drivers cleanups
by AngeloGioacchino Del Regno 26 Mar '24

26 Mar '24

This series converts MediaTek AFE drivers' probe functions to use dev_err_probe() and devm functions where possible and, in some cases, dropping the .remove_new() callback, reducing the size. Cheers! AngeloGioacchino Del Regno (7): ASoC: mediatek: mt8173-afe-pcm: Convert to devm_pm_runtime_enable() ASoC: mediatek: mt8173-afe-pcm: Use devm_snd_soc_register_component() ASoC: mediatek: mt8183-afe-pcm: Convert to devm_pm_runtime_enable() ASoC: mediatek: mt8183-afe-pcm: Simplify with dev_err_probe() ASoC: mediatek: mt8192-afe-pcm: Convert to devm_pm_runtime_enable() ASoC: mediatek: mt8192-afe-pcm: Simplify with dev_err_probe() ASoC: mediatek: mt8195-afe-pcm: Drop .remove_new() callback sound/soc/mediatek/mt8173/mt8173-afe-pcm.c | 78 +++------------ sound/soc/mediatek/mt8183/mt8183-afe-pcm.c | 108 +++++++-------------- sound/soc/mediatek/mt8192/mt8192-afe-pcm.c | 95 ++++++------------ sound/soc/mediatek/mt8195/mt8195-afe-pcm.c | 12 +-- 4 files changed, 81 insertions(+), 212 deletions(-) -- 2.43.0

5 15

[PATCH v11] ASoc: tas2783: Add tas2783 codec driver
by Shenghao Ding 26 Mar '24

26 Mar '24

The tas2783 is a smart audio amplifier with integrated MIPI SoundWire interface (Version 1.2.1 compliant), I2C, and I2S/TDM interfaces designed for portable applications. An on-chip DSP supports Texas Instruments SmartAmp speaker protection algorithm. The integrated speaker voltage and current sense provides for real-time monitoring of loudspeakers. The ASoC component provides the majority of the functionality of the device, all the audio functions. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- Change in v11: - simplify tasdevice_set_sdw_stream - fixed some Linux coding style - fixed the spelling mistakes - Select left/right channel based on unique id - a longer description has been added - remove unused crc8 in KCONFIG - Invert of RE here means reciprocal of RE - correct indentation - correct the wrong pow calibrated register - correct comment on RESET register - fixed usually lines don't end with an open parenthesis - fixed tasdevice_sdw_shutdown - use defines for function, entity, etc. for readable - remove unnecessary codec_lock in component_probe() - remove struct sdw_stream_data - remove unused macro definitions - use BIT(2) and BIT(1) instead of 0x2 and 0x1 - implement pm_runtime - reg_start is u8, reg_start + i only adds i byte, it should be reg_start + i * 4, because the calibration values - fixed total_sz in tas2783_calibration - add a define for TASDEVICE_REG(0, 0, 0x0a) - rework macro style in header file - Add select CRC32 - rewrite the mute and the return value handling - Add comments on bin file - Add link_id into the fw name to distinguish the same sdw id on different sdw id. - remove unnecessary mutex_lock - use set_active() in tasdevice_io_init and move all other parts in the probe. - move pm_runtime_get_noresume to the probe - fix the return value of putvol function - add put_noidle in tasdevice_sdw_remove - remove unused header file(of.h) - correct the style of the multi-line comment - Set the readable registers in tas2783_readable_register for debugfs - Use new PM macros - Use default put and get inline SOC_SINGLE_RANGE_TLV and SOC_SINGLE_SX_TLV - Use respective %pt to log the calibrated time - Remove unnecessary struct tm - Add trailing comma tasdevice_dapm_widgets and tasdevice_audio_map - Rewrite the return value handling in tasdevice_sdw_probe - use dev_num instead of id.unique_id - rework the log - Add code to check the size of calibrated data in UEFI - Add description of Layout of calibrated Data in UEFI - switch back link_id and unique_id to identify the devcie - unique_id offset as the device index of the calibrated data - update calibrated data layout stored in UEFI, add link_id and unique_id into the calibrated data in case of tas2783 sitting in different SDQW buses. - correct the wrong unique_id - set the unique_id as hex style in dsp bin file name - use __func__ in the dev_err message in two identical logs intas2783_apply_calibv1/v2. - In tas2783_apply_calibv2, the very least move initializations to a different line. - No need for separate j variable, reuse i in tasdevice_read_prop. - remove unused "struct sdw_bus_params params", unsigned char dev_name[32] pstream, and chip_id - define dspfw_binaryname as a local variable in tasdevice_comp_probe. - reduce a lot of indentation in tasdevice_mute and spilte tasdevice_mute into tasdevice_enable_echo_ref and tasdevice_playback - remove tas2783_reg_defaults - use REGCACHE_MAPLE instead of REGCACHE_RBTREE - define fw_node_sz to store the node size --- sound/soc/codecs/Kconfig | 16 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tas2783-sdw.c | 893 +++++++++++++++++++++++++++++++++ sound/soc/codecs/tas2783.h | 107 ++++ 4 files changed, 1018 insertions(+) create mode 100644 sound/soc/codecs/tas2783-sdw.c create mode 100644 sound/soc/codecs/tas2783.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index f78ea2f86fa6..939384837d5e 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -248,6 +248,7 @@ config SND_SOC_ALL_CODECS imply SND_SOC_TAS2781_COMLIB imply SND_SOC_TAS2781_FMWLIB imply SND_SOC_TAS2781_I2C + imply SND_SOC_TAS2783 imply SND_SOC_TAS5086 imply SND_SOC_TAS571X imply SND_SOC_TAS5720 @@ -1880,6 +1881,21 @@ config SND_SOC_TAS2781_I2C algo coefficient setting, for one, two or even multiple TAS2781 chips. +config SND_SOC_TAS2783 + tristate "Texas Instruments TAS2783 speaker amplifier (sdw)" + depends on SOUNDWIRE + depends on EFI + select REGMAP + select REGMAP_SOUNDWIRE + select CRC32 + select CRC32_SARWATE + help + Enable support for Texas Instruments TAS2783 Smart Amplifier + Digital input mono Class-D and DSP-inside audio power amplifiers. + TAS2783 supports the MIPI SoundWire v1.2 compatible interface + for audio and control data. At the end of calibrated data appends + the crc data with crc32_sarwate for error detection. + config SND_SOC_TAS5086 tristate "Texas Instruments TAS5086 speaker amplifier" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 7c075539dc47..6f27f810affc 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -287,6 +287,7 @@ snd-soc-tas2770-objs := tas2770.o snd-soc-tas2781-comlib-objs := tas2781-comlib.o snd-soc-tas2781-fmwlib-objs := tas2781-fmwlib.o snd-soc-tas2781-i2c-objs := tas2781-i2c.o +snd-soc-tas2783-objs := tas2783-sdw.o snd-soc-tfa9879-objs := tfa9879.o snd-soc-tfa989x-objs := tfa989x.o snd-soc-tlv320adc3xxx-objs := tlv320adc3xxx.o @@ -672,6 +673,7 @@ obj-$(CONFIG_SND_SOC_TAS2780) += snd-soc-tas2780.o obj-$(CONFIG_SND_SOC_TAS2781_COMLIB) += snd-soc-tas2781-comlib.o obj-$(CONFIG_SND_SOC_TAS2781_FMWLIB) += snd-soc-tas2781-fmwlib.o obj-$(CONFIG_SND_SOC_TAS2781_I2C) += snd-soc-tas2781-i2c.o +obj-$(CONFIG_SND_SOC_TAS2783) += snd-soc-tas2783.o obj-$(CONFIG_SND_SOC_TAS5086) += snd-soc-tas5086.o obj-$(CONFIG_SND_SOC_TAS571X) += snd-soc-tas571x.o obj-$(CONFIG_SND_SOC_TAS5720) += snd-soc-tas5720.o diff --git a/sound/soc/codecs/tas2783-sdw.c b/sound/soc/codecs/tas2783-sdw.c new file mode 100644 index 000000000000..2844cc5189dc --- /dev/null +++ b/sound/soc/codecs/tas2783-sdw.c @@ -0,0 +1,893 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// ALSA SoC Texas Instruments TAS2783 Audio Smart Amplifier +// +// Copyright (C) 2023 - 2024 Texas Instruments Incorporated +// https://www.ti.com +// +// Author: Baojun Xu <baojun.xu(a)ti.com> +// Kevin Lu <kevin-lu(a)ti.com> +// Shenghao Ding <shenghao-ding(a)ti.com> +// + +#include <asm/unaligned.h> +#include <linux/acpi.h> +#include <linux/crc32.h> +#include <linux/efi.h> +#include <linux/err.h> +#include <linux/firmware.h> +#include <linux/init.h> +#include <linux/module.h> +#include <linux/pm.h> +#include <linux/pm_runtime.h> +#include <linux/regmap.h> +#include <linux/soundwire/sdw.h> +#include <linux/soundwire/sdw_registers.h> +#include <linux/soundwire/sdw_type.h> +#include <sound/pcm_params.h> +#include <sound/sdw.h> +#include <sound/soc.h> +#include <sound/tlv.h> + +#include <sound/tas2781-tlv.h> +#include "tas2783.h" + +struct calibdatav2_info { + unsigned int number_of_devices; + unsigned int crc32_indx; + unsigned int byt_sz; + unsigned int *cali_data; +}; + +static const unsigned int tas2783_cali_reg[] = { + TAS2783_CALIBRATION_RE, /* Resistance */ + TAS2783_CALIBRATION_RE_LOW, /* Low limitation of RE */ + TAS2783_CALIBRATION_INV_RE, /* Reciprocal of RE */ + TAS2783_CALIBRATION_POW, /* RMS Power */ + TAS2783_CALIBRATION_TLIMIT /* Temperature limitation */ +}; + +static bool tas2783_readable_register(struct device *dev, + unsigned int reg) +{ + switch (reg) { + /* Page 0 */ + case 0x8000 ... 0x807F: + return true; + default: + return false; + } +} + +static bool tas2783_volatile_register(struct device *dev, + unsigned int reg) +{ + switch (reg) { + /* + * Only software reset register was volatiled. + * Bit is self clearing. + * 0b = Don't reset + * 1b = reset + */ + case 0x8001: + return true; + default: + return false; + } +} + +static const struct regmap_config tasdevice_regmap = { + .reg_bits = 32, + .val_bits = 8, + .readable_reg = tas2783_readable_register, + .volatile_reg = tas2783_volatile_register, + .max_register = 0x44ffffff, + .cache_type = REGCACHE_MAPLE, + .use_single_read = true, + .use_single_write = true, +}; + +static const struct snd_kcontrol_new tas2783_snd_controls[] = { + SOC_SINGLE_RANGE_TLV("Digital Volume", TAS2783_DVC_LVL, + 0, 0, 200, 1, dvc_tlv), + SOC_SINGLE_SX_TLV("Amp Gain Volume", TAS2783_AMP_LEVEL, + 1, 0, 20, amp_vol_tlv), +}; + +/* + * Load the calibration data, including speaker impedance, f0, etc. + * Calibration is done by the manufacturer in the factory. The calibration + * data are used by the algorithm for calculating the speaker temperature, + * speaker membrane excursion and f0 in real time during playback. + * The DSP will work with default data values if calibrated data are + * missing or are invalid. + * Layout of calibrated Data V1 in UEFI(total 256 bytes): + * Calibrated Data of Dev 0(unique_id offset 0) (20 bytes) + * Calibrated Data of Dev 1(unique_id offset 1) (20 bytes) + * Calibrated Data of Dev 2(unique_id offset 2) (20 bytes) + * Calibrated Data of Dev 3(unique_id offset 3) (20 bytes) + * Calibrated Data of Dev 4(unique_id offset 4) (20 bytes) + * Calibrated Data of Dev 5(unique_id offset 5) (20 bytes) + * Calibrated Data of Dev 6(unique_id offset 6) (20 bytes) + * Calibrated Data of Dev 7(unique_id offset 7) (20 bytes) + * TimeStamp of Calibration (4 bytes) + * CRC (4 bytes) + * Reserved (88 bytes) + */ +static void tas2783_apply_calibv1(struct tasdevice_priv *tas_dev, + unsigned int *cali_data) +{ + struct regmap *map = tas_dev->regmap; + u8 *cali_start; + u16 offset; + int ret; + + if (!tas_dev->sdw_peripheral) { + dev_err(tas_dev->dev, "%s: peripheral doesn't exist.\n", + __func__); + return; + } + + offset = tas_dev->sdw_peripheral->id.unique_id - + TAS2783_UNIQUE_ID_BASE; + if (offset >= TAS2783_MAX_DEV_NUM) { + dev_err(tas_dev->dev, "%s: offset(%u) is out of range.\n", + __func__, offset); + return; + } + + /* + * The area saving tas2783 calibrated data is specified by its + * unique_id offset. cali_start is the first address of current + * tas2783's calibrated data. + */ + cali_start = (u8 *)(cali_data + offset * sizeof(tas2783_cali_reg)); + for (int i = 0; i < ARRAY_SIZE(tas2783_cali_reg); i++) { + ret = regmap_bulk_write(map, tas2783_cali_reg[i], + &cali_start[4 * i], 4); + if (ret) { + dev_err(tas_dev->dev, "%s: Cali failed %x:%d\n", + __func__, tas2783_cali_reg[i], ret); + break; + } + } +} + +/* + * New features were added in calibrated Data V2: + * 1. Added some the fields to store the link_id and uniqie_id for + * multi-link solutions + * 2. Support flexible number of devices instead of fixed one in V1. + * Layout of calibrated data V2 in UEFI(total 256 bytes): + * ChipID (2783, 4 bytes) + * Device-Sum (4 bytes) + * TimeStamp of Calibration (4 bytes) + * for (i = 0; i < Device-Sum; i++) { + * Device #i index_info () { + * SDW link id (2bytes) + * SDW unique_id (2bytes) + * } + * Calibrated Data of Device #i (20 bytes) + * } + * CRC (4 bytes) + * Reserved (the rest) + */ +static void tas2783_apply_calibv2(struct tasdevice_priv *tas_dev, + const struct calibdatav2_info *p) +{ + const unsigned int arr_size = ARRAY_SIZE(tas2783_cali_reg); + struct regmap *map = tas_dev->regmap; + unsigned int i, j, k; + u8 *cali_start; + u16 dev_indx; + int ret; + + if (!tas_dev->sdw_peripheral) { + dev_err(tas_dev->dev, "%s: peripheral doesn't exist.\n", + __func__); + return; + } + + dev_indx = tas_dev->sdw_peripheral->bus->link_id | + tas_dev->sdw_peripheral->id.unique_id << 16; + + for (i = 0; i < p->number_of_devices; i++) { + k = i * (arr_size + 1); + + if (dev_indx != p->cali_data[k]) + continue; + + /* skip 4 byte dev-indx */ + cali_start = (u8 *)&p->cali_data[k + 1]; + + for (j = 0; j < arr_size; j++) { + k = 4 * j; + ret = regmap_bulk_write(map, tas2783_cali_reg[j], + &cali_start[k], 4); + if (ret) { + dev_err(tas_dev->dev, + "%s: Cali failed %x:%d\n", __func__, + tas2783_cali_reg[j], ret); + break; + } + } + return; + } +} + +static void tas2783_calibration(struct tasdevice_priv *tas_dev) +{ + efi_guid_t efi_guid = EFI_GUID(0x1f52d2a1, 0xbb3a, 0x457d, 0xbc, + 0x09, 0x43, 0xa3, 0xf4, 0x31, 0x0a, 0x92); + static efi_char16_t efi_name[] = L"CALI_DATA"; + struct calibration_data cali_data; + unsigned int *tmp_val; + unsigned int crc; + efi_status_t status; + + cali_data.total_sz = 0; + + status = efi.get_variable(efi_name, &efi_guid, NULL, + &cali_data.total_sz, NULL); + if (status == EFI_BUFFER_TOO_SMALL + && cali_data.total_sz < TAS2783_MAX_CALIDATA_SIZE) { + status = efi.get_variable(efi_name, &efi_guid, NULL, + &cali_data.total_sz, + cali_data.data); + dev_dbg(tas_dev->dev, "%s: cali get %lx bytes result:%ld\n", + __func__, cali_data.total_sz, status); + } + if (status != 0) { + /* Failed got calibration data from EFI. */ + dev_dbg(tas_dev->dev, "No calibration data in UEFI."); + return; + } + + tmp_val = (unsigned int *)cali_data.data; + + /* Check Calibrated Data V1 */ + crc = crc32(~0, cali_data.data, TAS2783_CALIDATAV1_BYTE_SIZE) ^ ~0; + if (crc == tmp_val[TAS2783_CALIDATAV1_CRC32_INDX]) { + /* Date and time of when calibration was done. */ + tas2783_apply_calibv1(tas_dev, tmp_val); + dev_dbg(tas_dev->dev, "V1: %ptTs", + &tmp_val[TAS2783_CALIDATAV1_TIMESTAMP_INDX]); + return; + } + + /* Check Calibrated Data V2 */ + if (tmp_val[0] == 2783) { + const struct calibdatav2_info calib_info = { + .number_of_devices = tmp_val[1], + .crc32_indx = 3 + tmp_val[1] * 6, + .byt_sz = 12 + tmp_val[1] * 24, + .cali_data = &tmp_val[3] + }; + + if (calib_info.number_of_devices > TAS2783_MAX_DEV_NUM || + calib_info.number_of_devices == 0) { + dev_dbg(tas_dev->dev, "No dev in calibrated data V2."); + return; + } + crc = crc32(~0, cali_data.data, calib_info.byt_sz) + ^ ~0; + if (crc == tmp_val[calib_info.crc32_indx]) { + tas2783_apply_calibv2(tas_dev, &calib_info); + dev_dbg(tas_dev->dev, "V2: %ptTs", + &tmp_val[TAS2783_CALIDATAV2_TIMESTAMP_INDX]); + } else { + dev_dbg(tas_dev->dev, + "V2: CRC 0x%08x not match 0x%08x\n", + crc, tmp_val[calib_info.crc32_indx]); + } + } else { + dev_err(tas_dev->dev, "Non-2783 chip\n"); + } +} + +static void tasdevice_dspfw_ready(const struct firmware *fmw, + void *context) +{ + struct tasdevice_priv *tas_dev = (struct tasdevice_priv *) context; + struct regmap *map = tas_dev->regmap; + struct tas2783_firmware_node *p; + unsigned char *buf; + int fw_node_sz = sizeof(struct tas2783_firmware_node); + int img_sz, ret; + int offset = 0; + + if (!fmw || !fmw->data) { + dev_warn(tas_dev->dev, + "%s: failed to read fw, work in bypass-dsp mode.\n", + __func__); + return; + } + buf = (unsigned char *)fmw->data; + + img_sz = get_unaligned_le32(&buf[offset]); + offset += sizeof(img_sz); + if (img_sz != fmw->size) { + dev_warn(tas_dev->dev, "%s: size not matching, %d %u.", + __func__, (int)fmw->size, img_sz); + return; + } + + while (offset < img_sz) { + p = (struct tas2783_firmware_node *)(buf + offset); + if (p->length > 1) + ret = regmap_bulk_write(map, p->download_addr, + buf + offset + fw_node_sz, p->length); + else + ret = regmap_write(map, p->download_addr, + *(buf + offset + fw_node_sz)); + + if (ret != 0) { + dev_dbg(tas_dev->dev, + "%s: load FW fail: %d, work in bypass.\n", + __func__, ret); + return; + } + offset += fw_node_sz + p->length; + } + + tas2783_calibration(tas_dev); +} + +static const struct snd_soc_dapm_widget tasdevice_dapm_widgets[] = { + SND_SOC_DAPM_AIF_IN("ASI", "ASI Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("ASI OUT", "ASI Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_OUTPUT("OUT"), + SND_SOC_DAPM_INPUT("DMIC"), +}; + +static const struct snd_soc_dapm_route tasdevice_audio_map[] = { + {"OUT", NULL, "ASI"}, + {"ASI OUT", NULL, "DMIC"}, +}; + +static int tasdevice_set_sdw_stream(struct snd_soc_dai *dai, + void *sdw_stream, int direction) +{ + snd_soc_dai_dma_data_set(dai, direction, sdw_stream); + + return 0; +} + +static void tasdevice_sdw_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + snd_soc_dai_set_dma_data(dai, substream, NULL); +} + +static int tasdevice_sdw_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct tasdevice_priv *tas_priv = + snd_soc_component_get_drvdata(component); + struct sdw_stream_config stream_config = {0}; + struct sdw_port_config port_config = {0}; + struct sdw_stream_runtime *sdw_stream; + int ret; + + dev_dbg(dai->dev, "%s: dai_name %s", __func__, dai->name); + sdw_stream = snd_soc_dai_get_dma_data(dai, substream); + + if (!sdw_stream) + return -EINVAL; + + if (!tas_priv->sdw_peripheral) + return -EINVAL; + + /* SoundWire specific configuration */ + snd_sdw_params_to_config(substream, params, + &stream_config, &port_config); + + /* port 1 for playback */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + port_config.num = 1; + else + port_config.num = 2; + + ret = sdw_stream_add_slave(tas_priv->sdw_peripheral, + &stream_config, &port_config, 1, sdw_stream); + if (ret) { + dev_err(dai->dev, "%s: Unable to configure port\n", __func__); + return ret; + } + + dev_dbg(dai->dev, "%s: format: %i rate: %i\n", __func__, + params_format(params), params_rate(params)); + + return 0; +} + +static int tasdevice_sdw_pcm_hw_free(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct tasdevice_priv *tas_priv = + snd_soc_component_get_drvdata(component); + struct sdw_stream_runtime *sdw_stream = + snd_soc_dai_get_dma_data(dai, substream); + + if (!tas_priv->sdw_peripheral) + return -EINVAL; + + sdw_stream_remove_slave(tas_priv->sdw_peripheral, sdw_stream); + + return 0; +} + +/* Capture stream is the echo ref data for voice mode.*/ +static int tasdevice_enable_echo_ref(struct tasdevice_priv *tas_dev, + int mute) +{ + struct regmap *map = tas_dev->regmap; + unsigned int val; + int ret; + + if (mute) + val = TAS2783_REG_AEF_INACTIVE; + else + val = TAS2783_REG_AEF_ACTIVE; + + ret = regmap_update_bits(map, TAS2873_REG_PWR_CTRL, + TAS2783_REG_AEF_MASK, TAS2783_REG_AEF_INACTIVE); + if (ret) + dev_err(tas_dev->dev, "%s: Disable AEF failed.\n", __func__); + + return ret; +} + +static int tasdevice_playback(struct tasdevice_priv *tas_dev, int mute) +{ + struct regmap *map = tas_dev->regmap; + int ret; + + if (mute) { + /* FU23 mute (0x40400108) */ + ret = regmap_write(map, + SDW_SDCA_CTL(TAS2783_FUNC_TYPE_SMART_AMP, + TAS2783_SDCA_ENT_FU23, + TAS2783_SDCA_CTL_FU_MUTE, 0), 1); + if (ret) { + dev_err(tas_dev->dev, "%s: FU23 mute failed.\n", + __func__); + goto out; + } + + ret = regmap_update_bits(map, TAS2873_REG_PWR_CTRL, + TAS2783_REG_PWR_MODE_MASK | + TAS2783_REG_AEF_MASK, + TAS2783_REG_PWR_MODE_ACTIVE | + TAS2783_REG_PWR_MODE_SW_PWD); + if (ret) { + dev_err(tas_dev->dev, "%s: shutdown failed.\n", + __func__); + } + } else { + /* FU23 Unmute, 0x40400108. */ + ret = regmap_write(map, + SDW_SDCA_CTL(TAS2783_FUNC_TYPE_SMART_AMP, + TAS2783_SDCA_ENT_FU23, + TAS2783_SDCA_CTL_FU_MUTE, 0), 0); + if (ret) { + dev_err(tas_dev->dev, + "%s: FU23 Unmute failed.\n", __func__); + goto out; + } + ret = regmap_update_bits(map, TAS2873_REG_PWR_CTRL, + TAS2783_REG_PWR_MODE_MASK, + TAS2783_REG_PWR_MODE_ACTIVE); + if (ret) { + dev_err(tas_dev->dev, + "%s: PWR Unmute failed.\n", __func__); + } + } +out: + return ret; +} + +static int tasdevice_mute(struct snd_soc_dai *dai, int mute, + int direction) +{ + struct snd_soc_component *component = dai->component; + struct tasdevice_priv *tas_dev = + snd_soc_component_get_drvdata(component); + + int ret; + + dev_dbg(tas_dev->dev, "%s: %d.\n", __func__, mute); + + if (direction == SNDRV_PCM_STREAM_CAPTURE) + ret = tasdevice_enable_echo_ref(tas_dev, mute); + else + ret = tasdevice_playback(tas_dev, mute); + + if (ret) + dev_err(tas_dev->dev, "Mute or unmute %d failed %d.\n", + mute, ret); + + return ret; +} + +static const struct snd_soc_dai_ops tasdevice_dai_ops = { + .mute_stream = tasdevice_mute, + .hw_params = tasdevice_sdw_hw_params, + .hw_free = tasdevice_sdw_pcm_hw_free, + .set_stream = tasdevice_set_sdw_stream, + .shutdown = tasdevice_sdw_shutdown, +}; + +static struct snd_soc_dai_driver tasdevice_dai_driver[] = { + { + .name = "tas2783-codec", + .id = 0, + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 4, + .rates = TAS2783_DEVICE_RATES, + .formats = TAS2783_DEVICE_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 4, + .rates = TAS2783_DEVICE_RATES, + .formats = TAS2783_DEVICE_FORMATS, + }, + .ops = &tasdevice_dai_ops, + .symmetric_rate = 1, + }, +}; + +static int tasdevice_comp_probe(struct snd_soc_component *comp) +{ + struct tasdevice_priv *tas_dev = snd_soc_component_get_drvdata(comp); + unsigned char dspfw_binaryname[TAS2783_DSPFW_FILENAME_LEN]; + acpi_handle handle = ACPI_HANDLE(tas_dev->dev); + const struct firmware *fw_entry = NULL; + const char *sub = NULL; + int ret; + + if (handle) { + sub = acpi_get_subsystem_id(handle); + if (IS_ERR(sub)) + sub = NULL; + } + + tas_dev->component = comp; + + /* + * Each tas2783 in the system has its own dspfw. + */ + if (comp->name_prefix) { + /* + * name_prefix.bin stores the dsp firmware including speaker + * protection algorithm, audio acoustic algorithm, speaker + * characters and algorithm params, it must be copied into + * firmware folder. + */ + scnprintf(dspfw_binaryname, TAS2783_DSPFW_FILENAME_LEN, + "%s-tas2783.bin", comp->name_prefix); + } else { + /* Compatible with the previous naming rule */ + if (sub) { + /* + * subsys_id-link_id[0,1,...,N]-unique_id + * [0x8,...,0xf].bin stores the dsp firmware including + * speaker protection algorithm, audio acoustic + * algorithm, speaker characters and algorithm params, + * it must be copied into firmware folder. + */ + scnprintf(dspfw_binaryname, TAS2783_DSPFW_FILENAME_LEN, + "%s-%d-0x%x.bin", sub, + tas_dev->sdw_peripheral->bus->link_id, + tas_dev->sdw_peripheral->id.unique_id); + } else { + /* + * tas2783-link_id[0,1,...,N]-unique_id + * [0x8,...,0xf].bin stores the dsp firmware including + * speaker protection algorithm, audio acoustic + * algorithm, speaker characters and algorithm params, + * it must be copied into firmware folder. + */ + scnprintf(dspfw_binaryname, TAS2783_DSPFW_FILENAME_LEN, + "tas2783-%d-0x%x.bin", + tas_dev->sdw_peripheral->bus->link_id, + tas_dev->sdw_peripheral->id.unique_id); + } + } + + ret = request_firmware(&fw_entry, dspfw_binaryname, tas_dev->dev); + if (ret) { + dev_err(tas_dev->dev, + "%s: request_firmware %x open status: %d.\n", __func__, + tas_dev->sdw_peripheral->id.unique_id, ret); + goto out; + } + + tasdevice_dspfw_ready(fw_entry, tas_dev); + +out: + if (fw_entry) + release_firmware(fw_entry); + return 0; +} + +static const struct snd_soc_component_driver + soc_codec_driver_tasdevice = { + .probe = tasdevice_comp_probe, + .controls = tas2783_snd_controls, + .num_controls = ARRAY_SIZE(tas2783_snd_controls), + .dapm_widgets = tasdevice_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(tasdevice_dapm_widgets), + .dapm_routes = tasdevice_audio_map, + .num_dapm_routes = ARRAY_SIZE(tasdevice_audio_map), + .idle_bias_on = 1, + .endianness = 1, +}; + +static int tasdevice_init(struct tasdevice_priv *tas_dev) +{ + int ret; + + dev_set_drvdata(tas_dev->dev, tas_dev); + + ret = devm_snd_soc_register_component(tas_dev->dev, + &soc_codec_driver_tasdevice, tasdevice_dai_driver, + ARRAY_SIZE(tasdevice_dai_driver)); + if (ret) { + dev_err(tas_dev->dev, "%s: codec register error:%d.\n", + __func__, ret); + return ret; + } + + /* set autosuspend parameters */ + pm_runtime_set_autosuspend_delay(tas_dev->dev, 3000); + pm_runtime_use_autosuspend(tas_dev->dev); + + /* make sure the device does not suspend immediately */ + pm_runtime_mark_last_busy(tas_dev->dev); + pm_runtime_enable(tas_dev->dev); + + return 0; +} + +static int tasdevice_read_prop(struct sdw_slave *slave) +{ + struct sdw_slave_prop *prop = &slave->prop; + struct sdw_dpn_prop *dpn; + unsigned long addr; + int nval, i; + u32 bit; + + prop->scp_int1_mask = SDW_SCP_INT1_BUS_CLASH | SDW_SCP_INT1_PARITY; + prop->quirks = SDW_SLAVE_QUIRKS_INVALID_INITIAL_PARITY; + + prop->paging_support = true; + + /* first we need to allocate memory for set bits in port lists */ + prop->source_ports = BIT(2); /* BITMAP: 00000100 */ + prop->sink_ports = BIT(1); /* BITMAP: 00000010 */ + + nval = hweight32(prop->source_ports); + prop->src_dpn_prop = devm_kcalloc(&slave->dev, nval, + sizeof(*prop->src_dpn_prop), GFP_KERNEL); + if (!prop->src_dpn_prop) + return -ENOMEM; + + i = 0; + dpn = prop->src_dpn_prop; + addr = prop->source_ports; + for_each_set_bit(bit, &addr, 32) { + dpn[i].num = bit; + dpn[i].type = SDW_DPN_FULL; + dpn[i].simple_ch_prep_sm = true; + dpn[i].ch_prep_timeout = 10; + i++; + } + + /* do this again for sink now */ + nval = hweight32(prop->sink_ports); + prop->sink_dpn_prop = devm_kcalloc(&slave->dev, nval, + sizeof(*prop->sink_dpn_prop), GFP_KERNEL); + if (!prop->sink_dpn_prop) + return -ENOMEM; + + i = 0; + dpn = prop->sink_dpn_prop; + addr = prop->sink_ports; + for_each_set_bit(bit, &addr, 32) { + dpn[i].num = bit; + dpn[i].type = SDW_DPN_FULL; + dpn[i].simple_ch_prep_sm = true; + dpn[i].ch_prep_timeout = 10; + i++; + } + + /* set the timeout values */ + prop->clk_stop_timeout = 20; + + return 0; +} + +static int tasdevice_io_init(struct device *dev, + struct sdw_slave *slave) +{ + struct tasdevice_priv *tas_priv = dev_get_drvdata(dev); + int ret; + + regcache_cache_only(tas_priv->regmap, false); + + if (tas_priv->first_hw_init) { + regcache_cache_bypass(tas_priv->regmap, true); + } else { + /* + * PM runtime is only enabled when a Slave reports as Attached + * Update count of parent 'active' children. + */ + pm_runtime_set_active(&slave->dev); + } + + /* sw reset */ + ret = regmap_write(tas_priv->regmap, TAS2873_REG_SWRESET, + TAS2873_REG_SWRESET_RESET); + if (ret) { + dev_err(tas_priv->dev, "Reset failed.\n"); + goto out; + } + + if (tas_priv->first_hw_init) { + regcache_cache_bypass(tas_priv->regmap, false); + regcache_mark_dirty(tas_priv->regmap); + } + + tas_priv->first_hw_init = true; + tas_priv->hw_init = true; + +out: + return ret; +} + +static int tasdevice_update_status(struct sdw_slave *slave, + enum sdw_slave_status status) +{ + struct tasdevice_priv *tas_priv = dev_get_drvdata(&slave->dev); + + if (status == SDW_SLAVE_UNATTACHED) + tas_priv->hw_init = false; + /* + * Perform initialization only if slave status + * is present and hw_init flag is false + */ + if (tas_priv->hw_init || status != SDW_SLAVE_ATTACHED) + return 0; + + /* perform I/O transfers required for Slave initialization */ + return tasdevice_io_init(&slave->dev, slave); +} + +/* + * slave_ops: callbacks for get_clock_stop_mode, clock_stop and + * port_prep are not defined for now + */ +static const struct sdw_slave_ops tasdevice_sdw_ops = { + .read_prop = tasdevice_read_prop, + .update_status = tasdevice_update_status, +}; + +static int tasdevice_sdw_probe(struct sdw_slave *peripheral, + const struct sdw_device_id *id) +{ + struct device *dev = &peripheral->dev; + struct tasdevice_priv *tas_dev; + + tas_dev = devm_kzalloc(dev, sizeof(*tas_dev), GFP_KERNEL); + if (!tas_dev) + return -ENOMEM; + + tas_dev->dev = dev; + tas_dev->sdw_peripheral = peripheral; + /* + * Mark hw_init to false + * HW init will be performed when device reports present + */ + tas_dev->hw_init = false; + tas_dev->first_hw_init = false; + + dev_set_drvdata(dev, tas_dev); + + tas_dev->regmap = devm_regmap_init_sdw(peripheral, &tasdevice_regmap); + if (IS_ERR(tas_dev->regmap)) + return dev_err_probe(dev, PTR_ERR(tas_dev->regmap), + "Failed devm_regmap_init_sdw."); + + return tasdevice_init(tas_dev); +} + +static int tasdevice_sdw_remove(struct sdw_slave *peripheral) +{ + struct tasdevice_priv *tas_dev = dev_get_drvdata(&peripheral->dev); + + if (tas_dev->first_hw_init) + pm_runtime_disable(tas_dev->dev); + + return 0; +} + +static const struct sdw_device_id tasdevice_sdw_id[] = { + SDW_SLAVE_ENTRY(0x0102, 0x0000, 0), + {} +}; +MODULE_DEVICE_TABLE(sdw, tasdevice_sdw_id); + +static int __maybe_unused tas2783_sdca_dev_suspend(struct device *dev) +{ + struct tasdevice_priv *tas_priv = dev_get_drvdata(dev); + + if (!tas_priv->hw_init) + return 0; + + regcache_cache_only(tas_priv->regmap, true); + + return 0; +} + +#define TAS2783_PROBE_TIMEOUT_MS 5000 + +static int __maybe_unused tas2783_sdca_dev_resume(struct device *dev) +{ + struct sdw_slave *slave = dev_to_sdw_dev(dev); + struct tasdevice_priv *tas_priv = dev_get_drvdata(dev); + unsigned long time; + + if (!tas_priv->first_hw_init) + return 0; + + if (!slave->unattach_request) + goto regmap_sync; + + time = wait_for_completion_timeout(&slave->initialization_complete, + msecs_to_jiffies(TAS2783_PROBE_TIMEOUT_MS)); + if (!time) { + dev_err(&slave->dev, "Init not complete, timed out\n"); + sdw_show_ping_status(slave->bus, true); + + return -ETIMEDOUT; + } + +regmap_sync: + slave->unattach_request = 0; + regcache_cache_only(tas_priv->regmap, false); + regcache_sync(tas_priv->regmap); + + return 0; +} + +static const struct dev_pm_ops tas2783_sdca_pm = { + SYSTEM_SLEEP_PM_OPS(tas2783_sdca_dev_suspend, + tas2783_sdca_dev_resume) + RUNTIME_PM_OPS(tas2783_sdca_dev_suspend, + tas2783_sdca_dev_resume, NULL) +}; + +static struct sdw_driver tasdevice_sdw_driver = { + .driver = { + .name = "slave-tas2783", + .pm = &tas2783_sdca_pm, + }, + .probe = tasdevice_sdw_probe, + .remove = tasdevice_sdw_remove, + .ops = &tasdevice_sdw_ops, + .id_table = tasdevice_sdw_id, +}; + +module_sdw_driver(tasdevice_sdw_driver); + +MODULE_AUTHOR("Baojun Xu <baojun.xu(a)ti.com>"); +MODULE_AUTHOR("Shenghao Ding <shenghao-ding(a)ti.com>"); +MODULE_DESCRIPTION("ASoC TAS2783 SoundWire Driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/tas2783.h b/sound/soc/codecs/tas2783.h new file mode 100644 index 000000000000..de72e07602bc --- /dev/null +++ b/sound/soc/codecs/tas2783.h @@ -0,0 +1,107 @@ +/* SPDX-License-Identifier: GPL-2.0 + * + * ALSA SoC Texas Instruments TAS2783 Audio Smart Amplifier + * + * Copyright (C) 2023 - 2024 Texas Instruments Incorporated + * https://www.ti.com + * + * Author: Baojun Xu <baojun.xu(a)ti.com> + * Shenghao Ding <shenghao-ding(a)ti.com> + */ + +#ifndef __TAS2783_H__ +#define __TAS2783_H__ + +/* Unique id start */ +#define TAS2783_UNIQUE_ID_BASE 0x08 +#define TAS2783_MAX_DEV_NUM 8 + +#define TAS2783_DEVICE_RATES (SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000 | \ + SNDRV_PCM_RATE_96000 | \ + SNDRV_PCM_RATE_88200) + +#define TAS2783_DEVICE_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE) + +/* BOOK, PAGE Control Register */ +#define TASDEVICE_REG(book, page, reg) ((book * 256 * 256) + 0x8000 + \ + (page * 128) + reg) + +/*Software Reset */ +#define TAS2873_REG_SWRESET TASDEVICE_REG(0x0, 0x00, 0x01) +#define TAS2873_REG_SWRESET_MASK BIT(0) +#define TAS2873_REG_SWRESET_RESET BIT(0) + +#define TAS2873_REG_PWR_CTRL TASDEVICE_REG(0x0, 0x00, 0x02) +#define TAS2783_REG_AEF_MASK GENMASK(4, 3) +#define TAS2783_REG_AEF_ACTIVE 0 +#define TAS2783_REG_AEF_INACTIVE GENMASK(4, 3) +#define TAS2783_REG_PWR_MODE_MASK GENMASK(2, 0) +#define TAS2783_REG_PWR_MODE_ACTIVE 0 +#define TAS2783_REG_PWR_MODE_SW_PWD BIT(1) + +#define TAS2783_REG_TDM_RX_CFG TASDEVICE_REG(0x0, 0x00, 0x0a) + +/* Volume control */ +#define TAS2783_DVC_LVL TASDEVICE_REG(0x0, 0x00, 0x1a) +#define TAS2783_AMP_LEVEL TASDEVICE_REG(0x0, 0x00, 0x03) +#define TAS2783_AMP_LEVEL_MASK GENMASK(5, 1) + +/* Calibration data */ +#define TAS2783_CALIBRATION_RE TASDEVICE_REG(0x0, 0x17, 0x74) +#define TAS2783_CALIBRATION_RE_LOW TASDEVICE_REG(0x0, 0x18, 0x14) +#define TAS2783_CALIBRATION_INV_RE TASDEVICE_REG(0x0, 0x18, 0x0c) +#define TAS2783_CALIBRATION_POW TASDEVICE_REG(0x0, 0x13, 0x70) +#define TAS2783_CALIBRATION_TLIMIT TASDEVICE_REG(0x0, 0x18, 0x7c) + +/* TAS2783 SDCA Control - function number */ +#define TAS2783_FUNC_TYPE_SMART_AMP 0x01 + +/* TAS2783 SDCA entity */ +#define TAS2783_SDCA_ENT_FU21 0x01 +#define TAS2783_SDCA_ENT_FU23 0x02 + +/* TAS2783 SDCA control */ +#define TAS2783_SDCA_CTL_REQ_POWER_STATE 0x01 +#define TAS2783_SDCA_CTL_FU_MUTE 0x01 +#define TAS2783_SDCA_CTL_FU_VOLUME 0x02 +#define TAS2783_SDCA_CTL_UDMPU_CLUSTER 0x10 + +#define TAS2783_DEVICE_CHANNEL_LEFT 1 +#define TAS2783_DEVICE_CHANNEL_RIGHT 2 +#define TAS2783_DSPFW_FILENAME_LEN 64 + +#define TAS2783_MAX_CALIDATA_SIZE 256 + +#define TAS2783_CALIDATAV1_TIMESTAMP_INDX 40 +#define TAS2783_CALIDATAV1_CRC32_INDX 41 +/* 8 devs * 20 bytes calibrated data/dev + 4 bytes Timestamp */ +#define TAS2783_CALIDATAV1_BYTE_SIZE (8 * 20 + 4) + +#define TAS2783_CALIDATAV2_TIMESTAMP_INDX 2 + +struct tas2783_firmware_node { + unsigned int vendor_id; + unsigned int file_id; + unsigned int version_id; + unsigned int length; + unsigned int download_addr; +}; + +struct calibration_data { + unsigned long total_sz; + unsigned char data[TAS2783_MAX_CALIDATA_SIZE]; +}; + +struct tasdevice_priv { + struct snd_soc_component *component; + struct sdw_slave *sdw_peripheral; + struct regmap *regmap; + struct device *dev; + bool hw_init; + bool first_hw_init; +}; + +#endif /*__TAS2783_H__ */ -- 2.34.1

3 3

[PATCH] ASoC: cs42l43: Correct extraction of data pointer in suspend/resume
by Charles Keepax 26 Mar '24

26 Mar '24

The current code is pulling the wrong pointer causing it to disable the wrong IRQ. Correct the code to pull the correct cs42l43 core data pointer. Fixes: 64353af49fec ("ASoC: cs42l43: Add system suspend ops to disable IRQ") Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- sound/soc/codecs/cs42l43.c | 12 ++++++++---- 1 file changed, 8 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/cs42l43.c b/sound/soc/codecs/cs42l43.c index 860d5cda67bf..94685449f0f4 100644 --- a/sound/soc/codecs/cs42l43.c +++ b/sound/soc/codecs/cs42l43.c @@ -2364,7 +2364,8 @@ static int cs42l43_codec_runtime_resume(struct device *dev) static int cs42l43_codec_suspend(struct device *dev) { - struct cs42l43 *cs42l43 = dev_get_drvdata(dev); + struct cs42l43_codec *priv = dev_get_drvdata(dev); + struct cs42l43 *cs42l43 = priv->core; disable_irq(cs42l43->irq); @@ -2373,7 +2374,8 @@ static int cs42l43_codec_suspend(struct device *dev) static int cs42l43_codec_suspend_noirq(struct device *dev) { - struct cs42l43 *cs42l43 = dev_get_drvdata(dev); + struct cs42l43_codec *priv = dev_get_drvdata(dev); + struct cs42l43 *cs42l43 = priv->core; enable_irq(cs42l43->irq); @@ -2382,7 +2384,8 @@ static int cs42l43_codec_suspend_noirq(struct device *dev) static int cs42l43_codec_resume(struct device *dev) { - struct cs42l43 *cs42l43 = dev_get_drvdata(dev); + struct cs42l43_codec *priv = dev_get_drvdata(dev); + struct cs42l43 *cs42l43 = priv->core; enable_irq(cs42l43->irq); @@ -2391,7 +2394,8 @@ static int cs42l43_codec_resume(struct device *dev) static int cs42l43_codec_resume_noirq(struct device *dev) { - struct cs42l43 *cs42l43 = dev_get_drvdata(dev); + struct cs42l43_codec *priv = dev_get_drvdata(dev); + struct cs42l43 *cs42l43 = priv->core; disable_irq(cs42l43->irq); -- 2.39.2

2 1

[PATCH v1 1/1] ASoC: fsl: imx-es8328: Remove leftover gpio initialisation
by Andy Shevchenko 26 Mar '24

26 Mar '24

The gpio field is not used anymore, remove the leftover. This also fixes the compilation error after the ... Fixes: 9855f05e5536 ("ASoC: fsl: imx-es8328: Switch to using gpiod API") Signed-off-by: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> --- sound/soc/fsl/imx-es8328.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/fsl/imx-es8328.c b/sound/soc/fsl/imx-es8328.c index d67b8a149bff..5b9648f3b087 100644 --- a/sound/soc/fsl/imx-es8328.c +++ b/sound/soc/fsl/imx-es8328.c @@ -28,7 +28,6 @@ struct imx_es8328_data { static struct snd_soc_jack_gpio headset_jack_gpios[] = { { - .gpio = -1, .name = "headset-gpio", .report = SND_JACK_HEADSET, .invert = 0, -- 2.43.0.rc1.1.gbec44491f096

2 1

[PATCH v2][next] ALSA: firewire-lib: Avoid -Wflex-array-member-not-at-end warning
by Gustavo A. R. Silva 26 Mar '24

26 Mar '24

Use the `DEFINE_FLEX()` helper for an on-stack definition of a flexible structure where the size of the flexible-array member is known at compile-time, and refactor the rest of the code, accordingly. So, with these changes, fix the following warning: sound/firewire/amdtp-stream.c:1184:46: warning: structure containing a flexible array member is not at the end of another structure [-Wflex-array-member-not-at-end] Link: https://github.com/KSPP/linux/issues/202 Signed-off-by: Gustavo A. R. Silva <gustavoars(a)kernel.org> --- Changes in v2: - Use DEFINE_FLEX() helper instead of a new tagged struct. v1: - Link: https://lore.kernel.org/linux-hardening/d3a764eb76909b16b8a22d9ff530e5edf0e… sound/firewire/amdtp-stream.c | 10 ++++------ 1 file changed, 4 insertions(+), 6 deletions(-) diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c index c9f153f85ae6..d35d0a420ee0 100644 --- a/sound/firewire/amdtp-stream.c +++ b/sound/firewire/amdtp-stream.c @@ -1180,13 +1180,11 @@ static void process_rx_packets(struct fw_iso_context *context, u32 tstamp, size_ (void)fw_card_read_cycle_time(fw_parent_device(s->unit)->card, &curr_cycle_time); for (i = 0; i < packets; ++i) { - struct { - struct fw_iso_packet params; - __be32 header[CIP_HEADER_QUADLETS]; - } template = { {0}, {0} }; + DEFINE_FLEX(struct fw_iso_packet, template, header, + header_length, CIP_HEADER_QUADLETS); bool sched_irq = false; - build_it_pkt_header(s, desc->cycle, &template.params, pkt_header_length, + build_it_pkt_header(s, desc->cycle, template, pkt_header_length, desc->data_blocks, desc->data_block_counter, desc->syt, i, curr_cycle_time); @@ -1198,7 +1196,7 @@ static void process_rx_packets(struct fw_iso_context *context, u32 tstamp, size_ } } - if (queue_out_packet(s, &template.params, sched_irq) < 0) { + if (queue_out_packet(s, template, sched_irq) < 0) { cancel_stream(s); return; } -- 2.34.1

3 2

Scan hardware to find speakers not detected by OS
by Jacek Blocki 26 Mar '24

26 Mar '24

Hello, In Alsa-devel Digest, Vol 205, Issue 136 there is my message on internal spekaker and headphones not working on Huawei D14 laptop. SImilar issue was reported in Vol 205, Issue 142 (No sound "dummy output" - snd_hda_intel on Pop OS 22.04 LTS on Acer chromebook). Is there any tool/troubleshooting guide allowing to diagnose above problem? In my cases (Huawei laptop) i guess OS does not detect speakers. Is there any way to scan hardware to find components undetected by OS? Regards, Jacek Blocki

1 0

ucm2: MediaTek: mt8195-sof: Add support for Tomato RT5682s variant
by GitHub pull_request - opened 26 Mar '24

26 Mar '24

alsa-project/alsa-ucm-conf pull request #406 was opened from kholk: A later revision of the MT8195 Cherry Tomato Chromebook (Acer Chromebook Spin 513 CP513-2H, Revision 3 and 4) are using the RT5682s codec instead of RT5682i. The differences are only about a couple of missing switches, where the 'i' variant had a switch for the L/R Headphone output and a configurable DAC L/R Mux, while the 's' one misses the mux control and solely relies on the main Headphone Switch. This configuration has been tested on Arch Linux with Pipewire 10.03 + WirePlumber. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/406 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/406.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH 0/2] ALSA/ASoC/SOF: SSP dev_type support
by Peter Ujfalusi 26 Mar '24

26 Mar '24

Hi, SOF always passed 0 as dev_type to intel_nhlt_get_endpoint_blob() when looking up the blob from the NHLT table. This causes issues since alsa-utils commit 3a47ef2487ed ("topology: nhlt: intel: support more device types and directions") The dev_type is no longer always 0 in the topology embedded NHLT table resulting lookup failures for analog codecs since they will have dev_type=4. With SOF the dev_type is not used, we always use the SSP port index for looking up the configuration blob. The solution for the issue is to fetch the dev_type for the SSP port and feed it back to the intel_nhlt_get_endpoint_blob() to be able to find the configuration regardless of the assigned dev_type. Regards, Peter --- Brent Lu (2): ALSA: hda: intel-nhlt: add intel_nhlt_ssp_device_type() function ASoC: SOF: ipc4-topology: support NHLT device type include/sound/intel-nhlt.h | 10 ++++++++++ sound/hda/intel-nhlt.c | 26 ++++++++++++++++++++++++++ sound/soc/sof/ipc4-topology.c | 19 ++++++++++++++++--- 3 files changed, 52 insertions(+), 3 deletions(-) -- 2.43.0

3 6

[PATCH] ASoC: cs-amp-lib: Check for no firmware controls when writing calibration
by Richard Fitzgerald 26 Mar '24

26 Mar '24

From: Simon Trimmer <simont(a)opensource.cirrus.com> When a wmfw file has not been loaded the firmware control descriptions necessary to write a stored calibration are not present. In this case print a more descriptive error message. The message is logged at info level because it is not fatal, and does not necessarily imply that anything is broken. Signed-off-by: Simon Trimmer <simont(a)opensource.cirrus.com> Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> --- sound/soc/codecs/cs-amp-lib.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/soc/codecs/cs-amp-lib.c b/sound/soc/codecs/cs-amp-lib.c index 01ef4db5407d..287ac01a3873 100644 --- a/sound/soc/codecs/cs-amp-lib.c +++ b/sound/soc/codecs/cs-amp-lib.c @@ -56,6 +56,11 @@ static int _cs_amp_write_cal_coeffs(struct cs_dsp *dsp, dev_dbg(dsp->dev, "Calibration: Ambient=%#x, Status=%#x, CalR=%d\n", data->calAmbient, data->calStatus, data->calR); + if (list_empty(&dsp->ctl_list)) { + dev_info(dsp->dev, "Calibration disabled due to missing firmware controls\n"); + return -ENOENT; + } + ret = cs_amp_write_cal_coeff(dsp, controls, controls->ambient, data->calAmbient); if (ret) return ret; -- 2.39.2

2 1

[RESEND] ASoC: amd: simplify soundwire dependencies for legacy stack
by Vijendar Mukunda 25 Mar '24

25 Mar '24

The CONFIG_SOUNDWIRE_AMD is a user-visible option, it should be never selected by another driver. So replace the extra complexity with a normal Kconfig dependency in SND_SOC_AMD_SOUNDWIRE. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- sound/soc/amd/Kconfig | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/soc/amd/Kconfig b/sound/soc/amd/Kconfig index fa74635cee08..3508f5a96b75 100644 --- a/sound/soc/amd/Kconfig +++ b/sound/soc/amd/Kconfig @@ -134,15 +134,14 @@ config SND_SOC_AMD_RPL_ACP6x config SND_SOC_AMD_SOUNDWIRE_LINK_BASELINE tristate - select SOUNDWIRE_AMD if SND_SOC_AMD_SOUNDWIRE != n select SND_AMD_SOUNDWIRE_ACPI if ACPI config SND_SOC_AMD_SOUNDWIRE tristate "Support for SoundWire based AMD platforms" default SND_SOC_AMD_SOUNDWIRE_LINK_BASELINE depends on SND_SOC_AMD_SOUNDWIRE_LINK_BASELINE - depends on ACPI && SOUNDWIRE - depends on !(SOUNDWIRE=m && SND_SOC_AMD_SOUNDWIRE_LINK_BASELINE=y) + depends on ACPI + depends on SOUNDWIRE_AMD help This adds support for SoundWire for AMD platforms. Say Y if you want to enable SoundWire links with SOF. -- 2.34.1

2 1

[PATCH v2 1/1] ASoC: fsl: imx-es8328: Switch to using gpiod API
by Andy Shevchenko 25 Mar '24

25 Mar '24

This updates the driver to gpiod API, and removes yet another use of of_get_named_gpio(). Signed-off-by: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> --- v2: fixed compilation error sound/soc/fsl/imx-es8328.c | 16 ++++++++++------ 1 file changed, 10 insertions(+), 6 deletions(-) diff --git a/sound/soc/fsl/imx-es8328.c b/sound/soc/fsl/imx-es8328.c index 6f0d031c1d5f..d67b8a149bff 100644 --- a/sound/soc/fsl/imx-es8328.c +++ b/sound/soc/fsl/imx-es8328.c @@ -3,7 +3,7 @@ // Copyright 2012 Freescale Semiconductor, Inc. // Copyright 2012 Linaro Ltd. -#include <linux/gpio.h> +#include <linux/gpio/consumer.h> #include <linux/module.h> #include <linux/of.h> #include <linux/of_platform.h> @@ -23,7 +23,7 @@ struct imx_es8328_data { struct snd_soc_card card; char codec_dai_name[DAI_NAME_SIZE]; char platform_name[DAI_NAME_SIZE]; - int jack_gpio; + struct gpio_desc *jack_gpiod; }; static struct snd_soc_jack_gpio headset_jack_gpios[] = { @@ -54,8 +54,8 @@ static int imx_es8328_dai_init(struct snd_soc_pcm_runtime *rtd) struct imx_es8328_data, card); int ret = 0; - /* Headphone jack detection */ - if (gpio_is_valid(data->jack_gpio)) { + if (data->jack_gpiod) { + /* Headphone jack detection */ ret = snd_soc_card_jack_new_pins(rtd->card, "Headphone", SND_JACK_HEADSET | SND_JACK_BTN_0, &headset_jack, @@ -64,7 +64,7 @@ static int imx_es8328_dai_init(struct snd_soc_pcm_runtime *rtd) if (ret) return ret; - headset_jack_gpios[0].gpio = data->jack_gpio; + headset_jack_gpios[0].desc = data->jack_gpiod; ret = snd_soc_jack_add_gpios(&headset_jack, ARRAY_SIZE(headset_jack_gpios), headset_jack_gpios); @@ -174,7 +174,11 @@ static int imx_es8328_probe(struct platform_device *pdev) data->dev = dev; - data->jack_gpio = of_get_named_gpio(pdev->dev.of_node, "jack-gpio", 0); + data->jack_gpiod = devm_gpiod_get_optional(dev, "jack", GPIOD_IN); + if (IS_ERR(data->jack_gpiod)) { + ret = PTR_ERR(data->jack_gpiod); + goto put_device; + } /* * CPU == Platform -- 2.43.0.rc1.1.gbec44491f096

2 2

[PATCH 1/1] ASoC: fsl: fsl_ssi: Add dev_err_probe if PCM DMA init fails
by Alexander Stein 25 Mar '24

25 Mar '24

This happens especially if this driver is built-in, but SDMA driver is configured as module. Signed-off-by: Alexander Stein <alexander.stein(a)ew.tq-group.com> --- sound/soc/fsl/fsl_ssi.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index ab6ec1974807..4ca3a16f7ac0 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1401,8 +1401,10 @@ static int fsl_ssi_imx_probe(struct platform_device *pdev, goto error_pcm; } else { ret = imx_pcm_dma_init(pdev); - if (ret) + if (ret) { + dev_err_probe(dev, ret, "Failed to init PCM DMA\n"); goto error_pcm; + } } return 0; -- 2.34.1

3 2

[PATCH] ALSA: hda: cs35l56: Set the init_done flag before component_add()
by Richard Fitzgerald 25 Mar '24

25 Mar '24

From: Simon Trimmer <simont(a)opensource.cirrus.com> Initialization is completed before adding the component as that can start the process of the device binding and trigger actions that check init_done. Signed-off-by: Simon Trimmer <simont(a)opensource.cirrus.com> Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> Fixes: 73cfbfa9caea ("ALSA: hda/cs35l56: Add driver for Cirrus Logic CS35L56 amplifier") --- sound/pci/hda/cs35l56_hda.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/cs35l56_hda.c b/sound/pci/hda/cs35l56_hda.c index f3c5715f5e02..1a3f84599cb5 100644 --- a/sound/pci/hda/cs35l56_hda.c +++ b/sound/pci/hda/cs35l56_hda.c @@ -1045,14 +1045,14 @@ int cs35l56_hda_common_probe(struct cs35l56_hda *cs35l56, int hid, int id) pm_runtime_mark_last_busy(cs35l56->base.dev); pm_runtime_enable(cs35l56->base.dev); + cs35l56->base.init_done = true; + ret = component_add(cs35l56->base.dev, &cs35l56_hda_comp_ops); if (ret) { dev_err(cs35l56->base.dev, "Register component failed: %d\n", ret); goto pm_err; } - cs35l56->base.init_done = true; - return 0; pm_err: -- 2.39.2

2 1

[PATCH] ALSA: hda: cs35l56: Raise device name message log level
by Richard Fitzgerald 25 Mar '24

25 Mar '24

From: Simon Trimmer <simont(a)opensource.cirrus.com> The system and amplifier names influence which firmware and tuning files are downloaded to the device; log these values to aid end-user system support. Signed-off-by: Simon Trimmer <simont(a)opensource.cirrus.com> Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> --- sound/pci/hda/cs35l56_hda.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/cs35l56_hda.c b/sound/pci/hda/cs35l56_hda.c index 41974b3897a7..f3c5715f5e02 100644 --- a/sound/pci/hda/cs35l56_hda.c +++ b/sound/pci/hda/cs35l56_hda.c @@ -1024,8 +1024,8 @@ int cs35l56_hda_common_probe(struct cs35l56_hda *cs35l56, int hid, int id) goto err; } - dev_dbg(cs35l56->base.dev, "DSP system name: '%s', amp name: '%s'\n", - cs35l56->system_name, cs35l56->amp_name); + dev_info(cs35l56->base.dev, "DSP system name: '%s', amp name: '%s'\n", + cs35l56->system_name, cs35l56->amp_name); regmap_multi_reg_write(cs35l56->base.regmap, cs35l56_hda_dai_config, ARRAY_SIZE(cs35l56_hda_dai_config)); -- 2.39.2

2 1

aplay: fix S24_LE wav header
by GitHub pull_request - edited 25 Mar '24

25 Mar '24

alsa-project/alsa-utils pull request #260 was edited from psi-sue: S24_LE is 32 bits in width storing 24 bits of data and 8 bits of padding So wav header needs to be 32 bits not 24 Request URL : https://github.com/alsa-project/alsa-utils/pull/260 Patch URL : https://github.com/alsa-project/alsa-utils/pull/260.patch Repository URL: https://github.com/alsa-project/alsa-utils

1 0

aplay: fix S24_LE wav header
by GitHub pull_request - opened 25 Mar '24

25 Mar '24

alsa-project/alsa-utils pull request #260 was opened from psi-sue: S24_LE is 32 bits in width storing 24 bits of data and 8 bytes of padding So wav header needs to be 32 bits not 24 Request URL : https://github.com/alsa-project/alsa-utils/pull/260 Patch URL : https://github.com/alsa-project/alsa-utils/pull/260.patch Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[PATCH v4 0/2] ASoC: nau8325: Modify driver code and dtschema.
by Seven Lee 25 Mar '24

25 Mar '24

Modify properties description and use standard units in dtschema. Change: V1 -> V2: - Revise the driver description part for the C++ comment. - In the nau8325_clkdet_put function, modify the max variable to hard code. - Removed "Clock Detection" switch control. - modify the "ALC Enable" switch name. - Revise the dtschema for "nuvoton,dac-vref". V2 -> V3: - Properties use standard unit suffixes. - Modify the enum definition. - Driver code should be used dev_dbg(). V3 -> V4: - Properties use standard unit suffixes. - Modify the enum definition. Seven Lee (2): ASoC: dt-bindings: Added schema for "nuvoton,nau8325" ASoC: nau8325: new driver .../bindings/sound/nuvoton,nau8325.yaml | 90 ++ sound/soc/codecs/nau8325.c | 856 ++++++++++++++++++ sound/soc/codecs/nau8325.h | 391 ++++++++ 3 files changed, 1337 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/nuvoton,nau8325.yaml create mode 100644 sound/soc/codecs/nau8325.c create mode 100644 sound/soc/codecs/nau8325.h -- 2.25.1

4 6

[PATCH v3 00/32] spi: get rid of some legacy macros
by Uwe Kleine-König 25 Mar '24

25 Mar '24

Changes since v2 (https://lore.kernel.org/linux-spi/cover.1705944943.git.u.kleine-koenig@peng…): - Drop patch "mtd: rawnand: fsl_elbc: Let .probe retry if local bus is missing" which doesn't belong into this series. - Fix a build failure noticed by the kernel build bot in drivers/spi/spi-au1550.c. (I failed to catch this because this driver is mips only, but not enabled in a mips allmodconfig. That's a bit unfortunate, but not easily fixable.) - Add the Reviewed-by: and Acked-by: tags I received for v2. Mark already announced for v2 that he is willing to apply the whole series to his spi tree. Assuming no other show stoper are found in this v3, I assume that's the plan still for this series now. Thanks Uwe Uwe Kleine-König (32): fpga: ice40-spi: Follow renaming of SPI "master" to "controller" ieee802154: ca8210: Follow renaming of SPI "master" to "controller" iio: adc: ad_sigma_delta: Follow renaming of SPI "master" to "controller" Input: pxspad - follow renaming of SPI "master" to "controller" Input: synaptics-rmi4 - follow renaming of SPI "master" to "controller" media: mgb4: Follow renaming of SPI "master" to "controller" media: netup_unidvb: Follow renaming of SPI "master" to "controller" media: usb/msi2500: Follow renaming of SPI "master" to "controller" media: v4l2-subdev: Follow renaming of SPI "master" to "controller" misc: gehc-achc: Follow renaming of SPI "master" to "controller" mmc: mmc_spi: Follow renaming of SPI "master" to "controller" mtd: dataflash: Follow renaming of SPI "master" to "controller" net: ks8851: Follow renaming of SPI "master" to "controller" net: vertexcom: mse102x: Follow renaming of SPI "master" to "controller" platform/chrome: cros_ec_spi: Follow renaming of SPI "master" to "controller" spi: bitbang: Follow renaming of SPI "master" to "controller" spi: cadence-quadspi: Don't emit error message on allocation error spi: cadence-quadspi: Follow renaming of SPI "master" to "controller" spi: cavium: Follow renaming of SPI "master" to "controller" spi: geni-qcom: Follow renaming of SPI "master" to "controller" spi: loopback-test: Follow renaming of SPI "master" to "controller" spi: slave-mt27xx: Follow renaming of SPI "master" to "controller" spi: spidev: Follow renaming of SPI "master" to "controller" staging: fbtft: Follow renaming of SPI "master" to "controller" staging: greybus: spi: Follow renaming of SPI "master" to "controller" tpm_tis_spi: Follow renaming of SPI "master" to "controller" usb: gadget: max3420_udc: Follow renaming of SPI "master" to "controller" video: fbdev: mmp: Follow renaming of SPI "master" to "controller" wifi: libertas: Follow renaming of SPI "master" to "controller" spi: fsl-lib: Follow renaming of SPI "master" to "controller" spi: Drop compat layer from renaming "master" to "controller" Documentation: spi: Update documentation for renaming "master" to "controller" .../driver-api/driver-model/devres.rst | 2 +- Documentation/spi/spi-summary.rst | 74 +++++++++---------- drivers/char/tpm/tpm_tis_spi_main.c | 4 +- drivers/fpga/ice40-spi.c | 4 +- drivers/iio/adc/ad_sigma_delta.c | 14 ++-- drivers/input/joystick/psxpad-spi.c | 4 +- drivers/input/rmi4/rmi_spi.c | 2 +- drivers/media/pci/mgb4/mgb4_core.c | 14 ++-- .../media/pci/netup_unidvb/netup_unidvb_spi.c | 48 ++++++------ drivers/media/usb/msi2500/msi2500.c | 38 +++++----- drivers/media/v4l2-core/v4l2-spi.c | 4 +- drivers/misc/gehc-achc.c | 8 +- drivers/mmc/host/mmc_spi.c | 6 +- drivers/mtd/devices/mtd_dataflash.c | 2 +- drivers/net/ethernet/micrel/ks8851_spi.c | 4 +- drivers/net/ethernet/vertexcom/mse102x.c | 2 +- drivers/net/ieee802154/ca8210.c | 2 +- .../net/wireless/marvell/libertas/if_spi.c | 2 +- drivers/platform/chrome/cros_ec_spi.c | 8 +- drivers/spi/spi-ath79.c | 4 +- drivers/spi/spi-au1550.c | 2 +- drivers/spi/spi-bitbang.c | 64 ++++++++-------- drivers/spi/spi-butterfly.c | 6 +- drivers/spi/spi-cadence-quadspi.c | 7 +- drivers/spi/spi-cavium.c | 6 +- drivers/spi/spi-cavium.h | 2 +- drivers/spi/spi-davinci.c | 6 +- drivers/spi/spi-fsl-lib.c | 14 ++-- drivers/spi/spi-geni-qcom.c | 2 +- drivers/spi/spi-gpio.c | 2 +- drivers/spi/spi-lm70llp.c | 6 +- drivers/spi/spi-loopback-test.c | 4 +- drivers/spi/spi-oc-tiny.c | 6 +- drivers/spi/spi-omap-uwire.c | 4 +- drivers/spi/spi-sh-sci.c | 10 +-- drivers/spi/spi-slave-mt27xx.c | 2 +- drivers/spi/spi-xilinx.c | 4 +- drivers/spi/spi-xtensa-xtfpga.c | 2 +- drivers/spi/spi.c | 2 +- drivers/spi/spidev.c | 2 +- drivers/staging/fbtft/fbtft-core.c | 4 +- drivers/staging/greybus/spilib.c | 66 ++++++++--------- drivers/usb/gadget/udc/max3420_udc.c | 2 +- drivers/video/fbdev/mmp/hw/mmp_spi.c | 26 +++---- include/linux/spi/spi.h | 20 +---- include/linux/spi/spi_bitbang.h | 2 +- include/media/v4l2-common.h | 6 +- 47 files changed, 253 insertions(+), 272 deletions(-) base-commit: b9b98f594b6f4c0b0fb2da4493453aef183bca4b -- 2.43.0

3 4

No sound "dummy output" - snd_hda_intel on Pop OS 22.04 LTS on Acer chromebook plus 514 (Google Markath rev 3)
by GitHub issues - edited 24 Mar '24

24 Mar '24

alsa-project/alsa-lib issue #391 was edited from judicandus: Installed a fresh Pop OS installation on an Acer Chromebook Plus 514, but audio is not working with just one interface available "Dummy Output". Alsa-info attached: [alsa-info.txt](https://github.com/alsa-project/alsa-lib/files/14733718/alsa… **1) What I already tried to do** All the instructions in: https://support.system76.com/articles/audio/ , specifically: i. reinstalling pipewire and deleting configurations: systemctl --user restart wireplumber pipewire pipewire-pulse rm -r ~/.config/pulse/* rm -r ~/.local/state/wireplumber/* ii. reinstalling apps sudo apt install --reinstall alsa-base alsa-utils linux-sound-base libasound2 sudo apt reinstall libpipewire-0.3-0 libpipewire-0.3-common libpipewire-0.3-modules pipewire pipewire-audio-client-libraries pipewire-bin pipewire-pulse sudo alsa force-reload iii. snh_hda_intel specific bug fix echo "options snd-hda-intel dmic_detect=0" | sudo tee -a /etc/modprobe.d/alsa-base.conf echo "blacklist snd_soc_skl" | sudo tee -a /etc/modprobe.d/blacklist.conf (reboot) iv. installing SOF driver according to https://github.com/thesofproject/sof-bin from the tarball Nothing worked. **2) Output** ``` aplay -l **** Liste des périphériques matériels PLAYBACK **** carte 0 : Generic [HD-Audio Generic], périphérique 3 : HDMI 0 [HDMI 0] Sous-périphériques : 1/1 Sous-périphérique #0 : subdevice #0 carte 0 : Generic [HD-Audio Generic], périphérique 7 : HDMI 1 [HDMI 1] Sous-périphériques : 1/1 Sous-périphérique #0 : subdevice #0 carte 0 : Generic [HD-Audio Generic], périphérique 8 : HDMI 2 [HDMI 2] Sous-périphériques : 1/1 Sous-périphérique #0 : subdevice #0 carte 0 : Generic [HD-Audio Generic], périphérique 9 : HDMI 3 [HDMI 3] Sous-périphériques : 1/1 Sous-périphérique #0 : subdevice #0 ``` ``` arecord -l **** Liste des périphériques matériels CAPTURE **** ``` ``` lsmod | grep snd_hda_intel snd_hda_intel 61440 1 snd_intel_dspcfg 36864 2 snd_hda_intel,snd_sof snd_hda_codec 221184 2 snd_hda_codec_hdmi,snd_hda_intel snd_hda_core 151552 3 snd_hda_codec_hdmi,snd_hda_intel,snd_hda_codec snd_pcm 196608 14 snd_sof_amd_acp,snd_hda_codec_hdmi,snd_pci_acp6x,snd_hda_intel,snd_hda_codec,snd_sof,snd_compress,snd_soc_core,snd_sof_utils,snd_hda_core,snd_soc_rt1019,snd_pci_ps,snd_soc_rt5682s,snd_pcm_dmaengine snd 147456 16 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 ``` ``` lspci -nnk | grep -A2 Audio 04:00.1 Audio device [0403]: Advanced Micro Devices, Inc. [AMD/ATI] Device [1002:1640] Subsystem: Advanced Micro Devices, Inc. [AMD/ATI] Device [1002:1640] Kernel driver in use: snd_hda_intel -- 04:00.5 Multimedia controller [0480]: Advanced Micro Devices, Inc. [AMD] Raven/Raven2/FireFlight/Renoir Audio Processor [1022:15e2] (rev 6f) DeviceName: Multimedia controller Subsystem: Advanced Micro Devices, Inc. [AMD] Raven/Raven2/FireFlight/Renoir Audio Processor [1022:15e2] Kernel driver in use: snd_sof_amd_rembrandt Kernel modules: snd_pci_acp3x, snd_rn_pci_acp3x, snd_pci_acp5x, snd_pci_acp6x, snd_acp_pci, snd_rpl_pci_acp6x, snd_pci_ps, snd_sof_amd_renoir, snd_sof_amd_rembrandt, snd_sof_amd_vangogh ``` ``` cat /proc/asound/cards 0 [Generic ]: HDA-Intel - HD-Audio Generic HD-Audio Generic at 0xd06c0000 irq 77 ``` ``` sudo dmesg | grep -i -e audio -e error -e fail [ 0.282615] ACPI: _OSC evaluation for CPUs failed, trying _PDC [ 0.629902] ACPI Error: Could not enable SleepButton event (20230628/evxfevnt-182) [ 6.100083] RAS: Correctable Errors collector initialized. [ 6.327488] hub 6-0:1.0: config failed, hub doesn't have any ports! (err -19) [ 8.787783] system76_acpi: module verification failed: signature and/or required key missing - tainting kernel [ 10.353860] snd_hda_intel 0000:04:00.1: bound 0000:04:00.0 (ops amdgpu_dm_audio_component_bind_ops [amdgpu]) [ 10.454926] input: HD-Audio Generic HDMI/DP,pcm=3 as /devices/pci0000:00/0000:00:08.1/0000:04:00.1/sound/card0/input10 [ 10.461777] input: HD-Audio Generic HDMI/DP,pcm=7 as /devices/pci0000:00/0000:00:08.1/0000:04:00.1/sound/card0/input11 [ 10.491432] input: HD-Audio Generic HDMI/DP,pcm=8 as /devices/pci0000:00/0000:00:08.1/0000:04:00.1/sound/card0/input12 [ 10.518137] input: HD-Audio Generic HDMI/DP,pcm=9 as /devices/pci0000:00/0000:00:08.1/0000:04:00.1/sound/card0/input13 [ 10.659161] snd_sof_amd_rembrandt 0000:04:00.5: Direct firmware load for amd/sof/community/sof-rmb.ri failed with error -2 [ 10.659196] snd_sof_amd_rembrandt 0000:04:00.5: error: sof firmware file is missing, you might need to [ 10.659208] snd_sof_amd_rembrandt 0000:04:00.5: error: failed to load DSP firmware -2 [ 10.659441] snd_sof_amd_rembrandt 0000:04:00.5: error: sof_probe_work failed err: -2 ``` ``` systemctl --user status pipewire pipewire.service - PipeWire Multimedia Service Loaded: loaded (/usr/lib/systemd/user/pipewire.service; enabled; vendor preset: enabled) Active: active (running) since Sat 2024-03-23 22:15:36 CET; 58min ago TriggeredBy: ● pipewire.socket Main PID: 1320 (pipewire) Tasks: 3 (limit: 9239) Memory: 6.6M CPU: 335ms CGroup: /user.slice/user-1000.slice/user(a)1000.service/session.slice/pipewire.service └─1320 /usr/bin/pipewire mars 23 22:15:36 pop-os systemd[1310]: Started PipeWire Multimedia Service. mars 23 22:15:36 pop-os pipewire[1320]: mod.jackdbus-detect: Failed to receive jackdbus reply: org.freedesktop.DBus.Error.ServiceUnknown: The name is not activatable ``` I am not sure if this is a bug but since I tried all solutions I could find on the internet, I thought I might try my luck here. Thank you beforehand for any assistance or insight. Issue URL : https://github.com/alsa-project/alsa-lib/issues/391 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

No sound "dummy output" - snd_hda_intel on Pop OS 22.04 LTS on Acer chromebook plus 514 (Google Markath rev 3)
by GitHub issues - opened 24 Mar '24

24 Mar '24

alsa-project/alsa-lib issue #391 was opened from judicandus: Installed a fresh Pop OS installation on an Acer Chromebook Plus 514, but audio is not working with just one interface available "Dummy Output". Alsa-info attached: [alsa-info.txt](https://github.com/alsa-project/alsa-lib/files/14733718/alsa… **1) What I already tried to do** All the instructions in: https://support.system76.com/articles/audio/ , specifically: i. reinstalling pipewire and deleting configurations: systemctl --user restart wireplumber pipewire pipewire-pulse rm -r ~/.config/pulse/* rm -r ~/.local/state/wireplumber/* ii. reinstalling apps sudo apt install --reinstall alsa-base alsa-utils linux-sound-base libasound2 sudo apt reinstall libpipewire-0.3-0 libpipewire-0.3-common libpipewire-0.3-modules pipewire pipewire-audio-client-libraries pipewire-bin pipewire-pulse sudo alsa force-reload iii. snh_hda_intel specific bug fix echo "options snd-hda-intel dmic_detect=0" | sudo tee -a /etc/modprobe.d/alsa-base.conf echo "blacklist snd_soc_skl" | sudo tee -a /etc/modprobe.d/blacklist.conf (reboot) iv. installing SOF driver according to https://github.com/thesofproject/sof-bin from the tarball Nothing worked. **2) Output** aplay -l **** Liste des périphériques matériels PLAYBACK **** carte 0 : Generic [HD-Audio Generic], périphérique 3 : HDMI 0 [HDMI 0] Sous-périphériques : 1/1 Sous-périphérique #0 : subdevice #0 carte 0 : Generic [HD-Audio Generic], périphérique 7 : HDMI 1 [HDMI 1] Sous-périphériques : 1/1 Sous-périphérique #0 : subdevice #0 carte 0 : Generic [HD-Audio Generic], périphérique 8 : HDMI 2 [HDMI 2] Sous-périphériques : 1/1 Sous-périphérique #0 : subdevice #0 carte 0 : Generic [HD-Audio Generic], périphérique 9 : HDMI 3 [HDMI 3] Sous-périphériques : 1/1 Sous-périphérique #0 : subdevice #0 arecord -l **** Liste des périphériques matériels CAPTURE **** lsmod | grep snd_hda_intel snd_hda_intel 61440 1 snd_intel_dspcfg 36864 2 snd_hda_intel,snd_sof snd_hda_codec 221184 2 snd_hda_codec_hdmi,snd_hda_intel snd_hda_core 151552 3 snd_hda_codec_hdmi,snd_hda_intel,snd_hda_codec snd_pcm 196608 14 snd_sof_amd_acp,snd_hda_codec_hdmi,snd_pci_acp6x,snd_hda_intel,snd_hda_codec,snd_sof,snd_compress,snd_soc_core,snd_sof_utils,snd_hda_core,snd_soc_rt1019,snd_pci_ps,snd_soc_rt5682s,snd_pcm_dmaengine snd 147456 16 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 lspci -nnk | grep -A2 Audio 04:00.1 Audio device [0403]: Advanced Micro Devices, Inc. [AMD/ATI] Device [1002:1640] Subsystem: Advanced Micro Devices, Inc. [AMD/ATI] Device [1002:1640] Kernel driver in use: snd_hda_intel -- 04:00.5 Multimedia controller [0480]: Advanced Micro Devices, Inc. [AMD] Raven/Raven2/FireFlight/Renoir Audio Processor [1022:15e2] (rev 6f) DeviceName: Multimedia controller Subsystem: Advanced Micro Devices, Inc. [AMD] Raven/Raven2/FireFlight/Renoir Audio Processor [1022:15e2] Kernel driver in use: snd_sof_amd_rembrandt Kernel modules: snd_pci_acp3x, snd_rn_pci_acp3x, snd_pci_acp5x, snd_pci_acp6x, snd_acp_pci, snd_rpl_pci_acp6x, snd_pci_ps, snd_sof_amd_renoir, snd_sof_amd_rembrandt, snd_sof_amd_vangogh cat /proc/asound/cards 0 [Generic ]: HDA-Intel - HD-Audio Generic HD-Audio Generic at 0xd06c0000 irq 77 sudo dmesg | grep -i -e audio -e error -e fail [ 0.282615] ACPI: _OSC evaluation for CPUs failed, trying _PDC [ 0.629902] ACPI Error: Could not enable SleepButton event (20230628/evxfevnt-182) [ 6.100083] RAS: Correctable Errors collector initialized. [ 6.327488] hub 6-0:1.0: config failed, hub doesn't have any ports! (err -19) [ 8.787783] system76_acpi: module verification failed: signature and/or required key missing - tainting kernel [ 10.353860] snd_hda_intel 0000:04:00.1: bound 0000:04:00.0 (ops amdgpu_dm_audio_component_bind_ops [amdgpu]) [ 10.454926] input: HD-Audio Generic HDMI/DP,pcm=3 as /devices/pci0000:00/0000:00:08.1/0000:04:00.1/sound/card0/input10 [ 10.461777] input: HD-Audio Generic HDMI/DP,pcm=7 as /devices/pci0000:00/0000:00:08.1/0000:04:00.1/sound/card0/input11 [ 10.491432] input: HD-Audio Generic HDMI/DP,pcm=8 as /devices/pci0000:00/0000:00:08.1/0000:04:00.1/sound/card0/input12 [ 10.518137] input: HD-Audio Generic HDMI/DP,pcm=9 as /devices/pci0000:00/0000:00:08.1/0000:04:00.1/sound/card0/input13 [ 10.659161] snd_sof_amd_rembrandt 0000:04:00.5: Direct firmware load for amd/sof/community/sof-rmb.ri failed with error -2 [ 10.659196] snd_sof_amd_rembrandt 0000:04:00.5: error: sof firmware file is missing, you might need to [ 10.659208] snd_sof_amd_rembrandt 0000:04:00.5: error: failed to load DSP firmware -2 [ 10.659441] snd_sof_amd_rembrandt 0000:04:00.5: error: sof_probe_work failed err: -2 systemctl --user status pipewire pipewire.service - PipeWire Multimedia Service Loaded: loaded (/usr/lib/systemd/user/pipewire.service; enabled; vendor preset: enabled) Active: active (running) since Sat 2024-03-23 22:15:36 CET; 58min ago TriggeredBy: ● pipewire.socket Main PID: 1320 (pipewire) Tasks: 3 (limit: 9239) Memory: 6.6M CPU: 335ms CGroup: /user.slice/user-1000.slice/user(a)1000.service/session.slice/pipewire.service └─1320 /usr/bin/pipewire mars 23 22:15:36 pop-os systemd[1310]: Started PipeWire Multimedia Service. mars 23 22:15:36 pop-os pipewire[1320]: mod.jackdbus-detect: Failed to receive jackdbus reply: org.freedesktop.DBus.Error.ServiceUnknown: The name is not activatable I am not sure if this is a bug but since I tried all solutions I could find on the internet, I thought I might try my luck here. Thank you beforehand for any assistance or insight. Issue URL : https://github.com/alsa-project/alsa-lib/issues/391 Repository URL: https://github.com/alsa-project/alsa-lib

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[PATCH] amdtp-stream: Checking a variable for zero before dividing
by Andrey Shumilin 23 Mar '24

23 Mar '24

The step variable is initialized to zero. It changes in the loop, but if it doesn’t change it will remain zero. The patch added a variable check. Found by Linux Verification Center (linuxtesting.org) with SVACE. Signed-off-by: Andrey Shumilin <shum.sdl(a)nppct.ru> --- sound/firewire/amdtp-stream.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c index c9f153f85ae6..b61aa30f43a9 100644 --- a/sound/firewire/amdtp-stream.c +++ b/sound/firewire/amdtp-stream.c @@ -168,7 +168,8 @@ static int apply_constraint_to_size(struct snd_pcm_hw_params *params, if (snd_interval_test(r, amdtp_rate_table[i])) step = max(step, amdtp_syt_intervals[i]); } - + if (step == 0) + return -EINVAL; t.min = roundup(s->min, step); t.max = rounddown(s->max, step); t.integer = 1; -- 2.30.2

1 0

Variable mclk-fs in ASoC audio cards?
by John Watts 22 Mar '24

22 Mar '24

Greetings fellow ALSAers :) Over the past month I have been working on a driver for the 8-channel TDM CS5368 audio codec. However I've run in to an interesting snag that seems to be an open problem. Let me explain. The audio codec generally runs at 8 channels of 32-bit data, so: MCLK = BCLK = LRCK * 256 However it is possible to read 4 or 2 channels instead: MCLK = BCLK = LRCK * 128 MCLK = BCLK = LRCK * 64 This trade-off allows avoiding high frequency clocking at the expense of a reduced channel count. From what I current see the way to indicate the multiplier here is using mclk-fs in the device tree, but this would mean effectively fixing the channel count in the device tree. I have searched around found someone has suggested mapping mclk-fs to rates here: https://lore.kernel.org/all/CAEnQRZDpJu27_pB=-ZYQjsNPC8eA4-Ur-36niT2hCosMWt… This functionality is somewhat adjacent to it, but I was wondering if it would be better to somehow factor dai-tdm-slot-* in to mclk-fs calculation. Perhaps mclk-fs could be divided by dai-tdm-slot-num then multiplied by slots? Has anyone hit this problem before? How did you solve it? John.

1 0

[PATCH] [RESEND] ASoC: amd: simplify soundwire dependencies for
by Vijendar Mukunda 22 Mar '24

22 Mar '24

The CONFIG_SOUNDWIRE_AMD is a user-visible option, it should be never selected by another driver. So replace the extra complexity with a normal Kconfig dependency in SND_SOC_AMD_SOUNDWIRE. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- sound/soc/amd/Kconfig | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/soc/amd/Kconfig b/sound/soc/amd/Kconfig index fa74635cee08..3508f5a96b75 100644 --- a/sound/soc/amd/Kconfig +++ b/sound/soc/amd/Kconfig @@ -134,15 +134,14 @@ config SND_SOC_AMD_RPL_ACP6x config SND_SOC_AMD_SOUNDWIRE_LINK_BASELINE tristate - select SOUNDWIRE_AMD if SND_SOC_AMD_SOUNDWIRE != n select SND_AMD_SOUNDWIRE_ACPI if ACPI config SND_SOC_AMD_SOUNDWIRE tristate "Support for SoundWire based AMD platforms" default SND_SOC_AMD_SOUNDWIRE_LINK_BASELINE depends on SND_SOC_AMD_SOUNDWIRE_LINK_BASELINE - depends on ACPI && SOUNDWIRE - depends on !(SOUNDWIRE=m && SND_SOC_AMD_SOUNDWIRE_LINK_BASELINE=y) + depends on ACPI + depends on SOUNDWIRE_AMD help This adds support for SoundWire for AMD platforms. Say Y if you want to enable SoundWire links with SOF. -- 2.34.1

2 1

[GIT PULL] ASoC fixes for v6.9-merge-window
by Mark Brown 21 Mar '24

21 Mar '24

The following changes since commit e8f897f4afef0031fe618a8e94127a0934896aba: Linux 6.8 (2024-03-10 13:38:09 -0700) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.9-merge-window for you to fetch changes up to 9a8b202f8cb7ebebc71f1f2a353a21c76d3063a8: ASoC: soc-compress: Fix and add DPCM locking (2024-03-18 14:41:51 +0000) ---------------------------------------------------------------- ASoC: Fixes for v6.9 A bunch of fixes that came in during the merge window, probably the most substantial thing is the DPCM locking fix for compressed audio which has been lurking for a while. ---------------------------------------------------------------- Chancel Liu (1): ASoC: soc-core.c: Skip dummy codec when adding platforms Cristian Ciocaltea (2): ASoC: SOF: amd: Move signed_fw_image to struct acp_quirk_entry ASoC: SOF: amd: Skip IRAM/DRAM size modification for Steam Deck OLED Jiawei Wang (2): ASoC: amd: yc: Revert "Fix non-functional mic on Lenovo 21J2" ASoC: amd: yc: Revert "add new YC platform variant (0x63) support" Luca Ceresoli (1): ASoC: rockchip: i2s-tdm: Fix inaccurate sampling rates M Cooley (1): ASoC: amd: yc: Fix non-functional mic on ASUS M7600RE Mark Brown (3): ASoC: Merge up release Add support for the internal RK3308 audio codec ASoC: SOF: amd: Skip IRAM/DRAM size modification Rob Herring (1): ASoC: dt-bindings: cirrus,cs42l43: Fix 'gpio-ranges' schema Shalini Manjunatha (1): ASoC: soc-compress: Fix and add DPCM locking Uwe Kleine-König (1): ASoC: tlv320adc3xxx: Don't strip remove function when driver is builtin .../ABI/testing/sysfs-bus-pci-devices-avs | 8 + .../bindings/sound/atmel,asoc-wm8904.yaml | 84 + .../bindings/sound/atmel,sam9x5-wm8731-audio.yaml | 76 + .../bindings/sound/atmel,sama5d2-classd.yaml | 7 +- .../bindings/sound/atmel-sam9x5-wm8731-audio.txt | 35 - .../devicetree/bindings/sound/atmel-wm8904.txt | 55 - .../bindings/sound/audio-graph-port.yaml | 2 +- .../devicetree/bindings/sound/cirrus,cs35l45.yaml | 3 + .../devicetree/bindings/sound/cirrus,cs42l43.yaml | 11 +- .../devicetree/bindings/sound/everest,es8326.yaml | 8 +- .../devicetree/bindings/sound/fsl,asrc.txt | 80 - .../devicetree/bindings/sound/fsl,easrc.yaml | 4 +- .../devicetree/bindings/sound/fsl,imx-asrc.yaml | 162 + .../devicetree/bindings/sound/fsl,micfil.yaml | 14 +- .../devicetree/bindings/sound/fsl,sai.yaml | 6 + .../bindings/sound/infineon,peb2466.yaml | 2 +- .../bindings/sound/microchip,sama7g5-i2smcc.yaml | 11 +- .../devicetree/bindings/sound/qcom,sm8250.yaml | 2 +- .../devicetree/bindings/sound/qcom,wcd938x.yaml | 81 +- .../bindings/sound/qcom,wcd939x-sdw.yaml | 69 + .../devicetree/bindings/sound/qcom,wcd939x.yaml | 96 + .../bindings/sound/qcom,wcd93xx-common.yaml | 95 + .../devicetree/bindings/sound/qcom,wsa8840.yaml | 11 +- .../devicetree/bindings/sound/realtek,rt1015.yaml | 41 + Documentation/devicetree/bindings/sound/rt1015.txt | 23 - .../devicetree/bindings/sound/samsung,tm2.yaml | 7 +- drivers/firmware/cirrus/cs_dsp.c | 7 +- drivers/soundwire/Makefile | 2 +- drivers/soundwire/amd_init.c | 235 ++ drivers/soundwire/amd_init.h | 13 + drivers/soundwire/amd_manager.c | 47 +- drivers/soundwire/amd_manager.h | 16 +- drivers/soundwire/dmi-quirks.c | 8 + include/linux/firmware/cirrus/cs_dsp.h | 1 - include/linux/soundwire/sdw_amd.h | 83 +- include/linux/spi/spi.h | 2 - include/sound/cs-amp-lib.h | 66 + include/sound/cs35l56.h | 10 + include/sound/cs42l42.h | 5 +- include/sound/hda-mlink.h | 2 +- include/sound/hda_register.h | 2 + include/sound/soc.h | 4 +- include/sound/sof/dai-amd.h | 7 + include/sound/sof/dai.h | 2 + include/sound/tas2781.h | 1 - include/trace/events/asoc.h | 45 +- include/uapi/sound/intel/avs/tokens.h | 9 + include/uapi/sound/sof/tokens.h | 4 + sound/pci/hda/Kconfig | 2 + sound/pci/hda/cs35l56_hda.c | 39 +- sound/pci/hda/tas2781_hda_i2c.c | 12 - sound/soc/amd/Kconfig | 17 + sound/soc/amd/Makefile | 2 +- sound/soc/amd/acp/Kconfig | 7 + sound/soc/amd/acp/Makefile | 2 + sound/soc/amd/acp/acp-mach-common.c | 6 +- sound/soc/amd/acp/acp-sof-mach.c | 26 +- sound/soc/amd/acp/amd-sdw-acpi.c | 62 + sound/soc/amd/ps/Makefile | 2 +- sound/soc/amd/ps/acp63.h | 91 +- sound/soc/amd/ps/pci-ps.c | 565 +-- sound/soc/amd/ps/ps-mach.c | 2 +- sound/soc/amd/ps/ps-pdm-dma.c | 2 +- sound/soc/amd/ps/ps-sdw-dma.c | 2 +- sound/soc/amd/yc/acp6x-mach.c | 14 +- sound/soc/amd/yc/pci-acp6x.c | 1 - sound/soc/atmel/mikroe-proto.c | 8 +- sound/soc/codecs/Kconfig | 55 +- sound/soc/codecs/Makefile | 13 + sound/soc/codecs/cs-amp-lib-test.c | 709 ++++ sound/soc/codecs/cs-amp-lib.c | 277 ++ sound/soc/codecs/cs35l56-sdw.c | 20 + sound/soc/codecs/cs35l56-shared.c | 84 + sound/soc/codecs/cs35l56.c | 44 +- sound/soc/codecs/cs42l42.c | 1 - sound/soc/codecs/cs42l43-jack.c | 27 +- sound/soc/codecs/cs42l43-sdw.c | 1 + sound/soc/codecs/cs42l43.c | 82 +- sound/soc/codecs/cs42l43.h | 21 +- sound/soc/codecs/da7213.c | 12 +- sound/soc/codecs/es8326.c | 110 +- sound/soc/codecs/es8326.h | 5 +- sound/soc/codecs/framer-codec.c | 413 +++ sound/soc/codecs/lpass-macro-common.h | 7 + sound/soc/codecs/lpass-tx-macro.c | 693 +++- sound/soc/codecs/lpass-va-macro.c | 57 + sound/soc/codecs/max98363.c | 2 +- sound/soc/codecs/max98373-sdw.c | 2 +- sound/soc/codecs/nau8540.c | 116 +- sound/soc/codecs/nau8540.h | 13 +- sound/soc/codecs/nau8825.c | 12 +- sound/soc/codecs/rt1017-sdca-sdw.c | 2 +- sound/soc/codecs/rt274.c | 2 +- sound/soc/codecs/rt286.c | 2 +- sound/soc/codecs/rt298.c | 2 +- sound/soc/codecs/rt5514-spi.c | 2 +- sound/soc/codecs/rt5514.c | 9 +- sound/soc/codecs/rt5616.c | 9 +- sound/soc/codecs/rt5640.c | 9 +- sound/soc/codecs/rt5645.c | 2 +- sound/soc/codecs/rt5651.c | 2 +- sound/soc/codecs/rt5659.c | 2 +- sound/soc/codecs/rt5660.c | 9 +- sound/soc/codecs/rt5663.c | 2 +- sound/soc/codecs/rt5665.c | 2 +- sound/soc/codecs/rt5668.c | 2 +- sound/soc/codecs/rt5682-i2c.c | 2 +- sound/soc/codecs/rt5682s.c | 2 +- sound/soc/codecs/rt712-sdca-dmic.c | 2 +- sound/soc/codecs/rt712-sdca-sdw.c | 2 +- sound/soc/codecs/rt722-sdca-sdw.c | 2 +- sound/soc/codecs/tas2781-comlib.c | 1 - sound/soc/codecs/tas2781-fmwlib.c | 2 +- sound/soc/codecs/tlv320adc3xxx.c | 4 +- sound/soc/codecs/wcd-clsh-v2.h | 1 + sound/soc/codecs/wcd-mbhc-v2.c | 95 +- sound/soc/codecs/wcd-mbhc-v2.h | 3 + sound/soc/codecs/wcd939x-sdw.c | 1551 ++++++++ sound/soc/codecs/wcd939x.c | 3686 ++++++++++++++++++++ sound/soc/codecs/wcd939x.h | 989 ++++++ sound/soc/codecs/wm_adsp.c | 27 +- sound/soc/codecs/wm_adsp.h | 2 + sound/soc/codecs/wsa884x.c | 53 +- sound/soc/fsl/eukrea-tlv320.c | 8 +- sound/soc/fsl/fsl_sai.c | 13 + sound/soc/fsl/p1022_rdk.c | 33 +- sound/soc/intel/atom/sst/sst_ipc.c | 3 +- sound/soc/intel/avs/Makefile | 5 +- sound/soc/intel/avs/apl.c | 58 +- sound/soc/intel/avs/avs.h | 70 +- sound/soc/intel/avs/board_selection.c | 85 + sound/soc/intel/avs/cnl.c | 61 + sound/soc/intel/avs/core.c | 161 +- sound/soc/intel/avs/icl.c | 197 ++ sound/soc/intel/avs/ipc.c | 66 +- sound/soc/intel/avs/loader.c | 2 +- sound/soc/intel/avs/messages.c | 1 + sound/soc/intel/avs/messages.h | 38 +- sound/soc/intel/avs/path.c | 33 + sound/soc/intel/avs/pcm.c | 77 +- sound/soc/intel/avs/registers.h | 21 +- sound/soc/intel/avs/skl.c | 59 +- sound/soc/intel/avs/sysfs.c | 35 + sound/soc/intel/avs/tgl.c | 54 + sound/soc/intel/avs/topology.c | 164 +- sound/soc/intel/avs/topology.h | 13 + sound/soc/intel/boards/Kconfig | 1 + sound/soc/intel/boards/sof_board_helpers.c | 249 +- sound/soc/intel/boards/sof_board_helpers.h | 29 + sound/soc/intel/boards/sof_cs42l42.c | 236 +- sound/soc/intel/boards/sof_rt5682.c | 151 +- sound/soc/intel/boards/sof_sdw.c | 106 +- sound/soc/intel/boards/sof_sdw_common.h | 82 +- sound/soc/intel/boards/sof_sdw_cs42l42.c | 35 +- sound/soc/intel/boards/sof_sdw_cs42l43.c | 25 +- sound/soc/intel/boards/sof_sdw_cs_amp.c | 3 +- sound/soc/intel/boards/sof_sdw_maxim.c | 4 +- sound/soc/intel/boards/sof_sdw_rt5682.c | 35 +- sound/soc/intel/boards/sof_sdw_rt700.c | 35 +- sound/soc/intel/boards/sof_sdw_rt711.c | 19 +- sound/soc/intel/boards/sof_sdw_rt712_sdca.c | 39 +- sound/soc/intel/boards/sof_sdw_rt715.c | 12 +- sound/soc/intel/boards/sof_sdw_rt715_sdca.c | 12 +- sound/soc/intel/boards/sof_sdw_rt_amp.c | 53 +- .../soc/intel/boards/sof_sdw_rt_sdca_jack_common.c | 19 +- sound/soc/intel/boards/sof_ssp_common.h | 8 + sound/soc/intel/catpt/dsp.c | 4 +- sound/soc/intel/common/soc-acpi-intel-adl-match.c | 12 +- sound/soc/intel/common/soc-acpi-intel-jsl-match.c | 10 +- sound/soc/intel/common/soc-acpi-intel-lnl-match.c | 102 + sound/soc/intel/common/soc-acpi-intel-mtl-match.c | 61 +- sound/soc/intel/common/soc-acpi-intel-rpl-match.c | 4 +- sound/soc/intel/common/soc-acpi-intel-tgl-match.c | 6 +- sound/soc/meson/aiu.c | 19 +- sound/soc/meson/aiu.h | 1 - sound/soc/meson/axg-fifo.c | 25 +- sound/soc/meson/axg-fifo.h | 14 +- sound/soc/meson/axg-frddr.c | 13 +- sound/soc/meson/axg-spdifin.c | 6 +- sound/soc/meson/axg-tdm-interface.c | 31 +- sound/soc/meson/axg-toddr.c | 30 +- sound/soc/meson/t9015.c | 20 +- sound/soc/pxa/Kconfig | 3 - sound/soc/qcom/common.c | 2 +- sound/soc/rockchip/rockchip_i2s_tdm.c | 352 +- sound/soc/sh/rz-ssi.c | 2 +- sound/soc/soc-compress.c | 4 + sound/soc/soc-core.c | 24 +- sound/soc/soc-dapm.c | 8 +- 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-loader.c | 34 +- sound/soc/sof/amd/acp.c | 252 +- sound/soc/sof/amd/acp.h | 33 +- sound/soc/sof/amd/pci-acp63.c | 7 + sound/soc/sof/amd/vangogh.c | 9 +- sound/soc/sof/core.c | 10 + sound/soc/sof/debug.c | 8 +- sound/soc/sof/fw-file-profile.c | 18 +- sound/soc/sof/imx/imx8.c | 16 + sound/soc/sof/imx/imx8m.c | 10 + sound/soc/sof/imx/imx8ulp.c | 10 + sound/soc/sof/intel/hda-common-ops.c | 1 + sound/soc/sof/intel/hda-dai-ops.c | 51 +- sound/soc/sof/intel/hda-dai.c | 17 +- sound/soc/sof/intel/hda-dsp.c | 5 + sound/soc/sof/intel/hda-stream.c | 9 + sound/soc/sof/intel/hda.c | 80 +- sound/soc/sof/intel/hda.h | 5 + sound/soc/sof/intel/lnl.c | 24 +- sound/soc/sof/ipc3-loader.c | 2 + sound/soc/sof/ipc3-pcm.c | 25 + sound/soc/sof/ipc3-topology.c | 40 + sound/soc/sof/ipc4-pcm.c | 19 +- sound/soc/sof/ipc4-priv.h | 4 + sound/soc/sof/ipc4-topology.c | 28 +- sound/soc/sof/ops.h | 9 + sound/soc/sof/sof-audio.c | 8 +- sound/soc/sof/sof-audio.h | 2 + sound/soc/sof/sof-priv.h | 10 + sound/soc/sof/topology.c | 30 +- sound/soc/ti/j721e-evm.c | 4 +- sound/soc/ti/omap-hdmi.c | 10 +- 224 files changed, 13139 insertions(+), 2591 deletions(-) create mode 100644 Documentation/ABI/testing/sysfs-bus-pci-devices-avs create mode 100644 Documentation/devicetree/bindings/sound/atmel,asoc-wm8904.yaml create mode 100644 Documentation/devicetree/bindings/sound/atmel,sam9x5-wm8731-audio.yaml delete mode 100644 Documentation/devicetree/bindings/sound/atmel-sam9x5-wm8731-audio.txt delete mode 100644 Documentation/devicetree/bindings/sound/atmel-wm8904.txt delete mode 100644 Documentation/devicetree/bindings/sound/fsl,asrc.txt create mode 100644 Documentation/devicetree/bindings/sound/fsl,imx-asrc.yaml create mode 100644 Documentation/devicetree/bindings/sound/qcom,wcd939x-sdw.yaml create mode 100644 Documentation/devicetree/bindings/sound/qcom,wcd939x.yaml create mode 100644 Documentation/devicetree/bindings/sound/qcom,wcd93xx-common.yaml create mode 100644 Documentation/devicetree/bindings/sound/realtek,rt1015.yaml delete mode 100644 Documentation/devicetree/bindings/sound/rt1015.txt create mode 100644 drivers/soundwire/amd_init.c create mode 100644 drivers/soundwire/amd_init.h create mode 100644 include/sound/cs-amp-lib.h create mode 100644 sound/soc/amd/acp/amd-sdw-acpi.c create mode 100644 sound/soc/codecs/cs-amp-lib-test.c create mode 100644 sound/soc/codecs/cs-amp-lib.c create mode 100644 sound/soc/codecs/framer-codec.c create mode 100644 sound/soc/codecs/wcd939x-sdw.c create mode 100644 sound/soc/codecs/wcd939x.c create mode 100644 sound/soc/codecs/wcd939x.h create mode 100644 sound/soc/intel/avs/cnl.c create mode 100644 sound/soc/intel/avs/icl.c create mode 100644 sound/soc/intel/avs/sysfs.c create mode 100644 sound/soc/intel/avs/tgl.c

2 1

synaptics hi-res audio usb device duplex, usb bandwidth issues
by Ian Malone 21 Mar '24

21 Mar '24

Hi, In short I have a USB to 3.5mm adapter which seems not to work in duplex mode on USB2.0 systems, possibly due to a bandwidth calculation bug. This is an evolution of an issue I previously posted on the users list with no luck. I have an Anker USB-C to 3.5mm audio dongle (lsusb: Conexant Systems (Rockwell), Inc. Hi-Res Audio) which I've used for some time on my phone (Android with USB-3.2). On trying to use it with an older T420 laptop recently with only USB-2.0 ports I discovered it will not work in duplex mode. Input-only and output-only profiles work (tested recording and playback with audacity), but with duplex no sound is recorded (Fedora 39, pipewire). This is easily reproduced by looking at the pavucontrol volume monitor for Output Devices, if I switch the device to Analog or Digital Input in configuration then the Input Devices level monitor for it shows activity if I speak into or tap the microphone. With duplex selected there is no activity, the level monitor bar may or may not display. I can switch between the two behaviours by changing the profile. Various applications such as Audacity and Zoom appear to hang when accessing this microphone with the duplex profile set. I've used pipewire configuration to force the format to 16LE only (playback and recording), but this has not helped. In dmesg this error appears when this happens (microphone opened, for example by pavucontrol): [ 294.825544] usb 1-1.1: cannot submit urb 0, error -28: not enough bandwidth (T420, Fedora 39, kernel 6.7.5) The T420 has USB 2 Type A ports, so a Type C to Type A adapter is needed, but the problem is not limited to the T420 laptop. On an older Samsung Tablet (Tab-A 2019, USB-2.0 Type C, kernel 4.4.177) I've been able to test using Google Meet. Although there are no errors or freezing observed, in headphone mode over USB the headphone microphone is not being used and the tablet uses its onboard microphone instead (although headphones are selected and playback is over the headphones). Again, I can record (via voice recorder) from the headset microphone, just no duplex. Trying google meet on my phone (USB3.2) I do get duplex over the USB-3.5 adapter. On a newer laptop with USB-3.2 and a mix of type A and C ports I also get duplex (F37, kernel 6.5.12), however dmesg shows even here not all is well, using the type A and C ports on the side: [ 9172.326602] usb 3-1: new full-speed USB device number 6 using xhci_hcd [ 9172.490169] usb 3-1: New USB device found, idVendor=0572, idProduct=1b08, bcdDevice= 0.10 [ 9172.490174] usb 3-1: New USB device strings: Mfr=1, Product=2, SerialNumber=3 [ 9172.490177] usb 3-1: Product: Hi-Res Audio [ 9172.490178] usb 3-1: Manufacturer: Synaptics [ 9172.490180] usb 3-1: SerialNumber: 000000000000000000000000 [ 9172.683435] input: Synaptics Hi-Res Audio as /devices/pci0000:00/0000:00:08.1/0000:05:00.4/usb3/3-1/3-1:1.3/0003:0572:1B08.000A/input/input36 [ 9172.736025] hid-generic 0003:0572:1B08.000A: input,hidraw5: USB HID v1.11 Device [Synaptics Hi-Res Audio] on usb-0000:05:00.4-1/input3 [ 9173.386988] usb 3-1: Not enough bandwidth for new device state. [ 9173.386994] usb 3-1: Not enough bandwidth for altsetting 1 [ 9173.386996] endpoint_set_interface: 70 callbacks suppressed [ 9173.386998] usb 3-1: 1:1: usb_set_interface failed (-28) [ 9173.387110] usb 3-1: Not enough bandwidth for new device state. [ 9173.387113] usb 3-1: Not enough bandwidth for altsetting 1 (...these 3 lines repeat 9 more times, followed by...) [ 9173.388785] usb 3-1: Not enough bandwidth for new device state. [ 9173.388789] usb 3-1: Not enough bandwidth for altsetting 1 (...these 2 lines repeat 5 more times, then stop) This is a USB3.2 system, the onboard camera has a hardware switch, so I can disable it, and the only other peripheral that appears is the AX210 bluetooth (I'd thought that would have its own bus as it's part of a separate wifi chip, but apparently not, lsusb -t below), bandwidth for even a 96kHz 24bit duplex audio device shouldn't really be an issue. $ lsusb Bus 004 Device 001: ID 1d6b:0003 Linux Foundation 3.0 root hub Bus 003 Device 003: ID 8087:0032 Intel Corp. AX210 Bluetooth Bus 003 Device 002: ID 048d:c965 Integrated Technology Express, Inc. ITE Device(8295) Bus 003 Device 007: ID 0572:1b08 Conexant Systems (Rockwell), Inc. Hi-Res Audio Bus 003 Device 001: ID 1d6b:0002 Linux Foundation 2.0 root hub Bus 002 Device 002: ID 0bda:0411 Realtek Semiconductor Corp. Hub Bus 002 Device 001: ID 1d6b:0003 Linux Foundation 3.0 root hub Bus 001 Device 004: ID 048d:c101 Integrated Technology Express, Inc. ITE Device(8910) Bus 001 Device 002: ID 0bda:5411 Realtek Semiconductor Corp. RTS5411 Hub Bus 001 Device 001: ID 1d6b:0002 Linux Foundation 2.0 root hub $ lsusb -t /: Bus 04.Port 1: Dev 1, Class=root_hub, Driver=xhci_hcd/2p, 10000M /: Bus 03.Port 1: Dev 1, Class=root_hub, Driver=xhci_hcd/4p, 480M |__ Port 1: Dev 7, If 0, Class=Audio, Driver=snd-usb-audio, 12M |__ Port 1: Dev 7, If 1, Class=Audio, Driver=snd-usb-audio, 12M |__ Port 1: Dev 7, If 2, Class=Audio, Driver=snd-usb-audio, 12M |__ Port 1: Dev 7, If 3, Class=Human Interface Device, Driver=usbhid, 12M |__ Port 3: Dev 2, If 0, Class=Human Interface Device, Driver=usbhid, 12M |__ Port 3: Dev 2, If 1, Class=Human Interface Device, Driver=usbhid, 12M |__ Port 4: Dev 3, If 0, Class=Wireless, Driver=btusb, 12M |__ Port 4: Dev 3, If 1, Class=Wireless, Driver=btusb, 12M /: Bus 02.Port 1: Dev 1, Class=root_hub, Driver=xhci_hcd/2p, 10000M |__ Port 2: Dev 2, If 0, Class=Hub, Driver=hub/4p, 5000M /: Bus 01.Port 1: Dev 1, Class=root_hub, Driver=xhci_hcd/4p, 480M |__ Port 2: Dev 2, If 0, Class=Hub, Driver=hub/4p, 480M |__ Port 4: Dev 4, If 0, Class=Human Interface Device, Driver=usbhid, 12M The only situation in which I *don't* get these dmesg errors is if I use one of the rear USB 3.2 ports that seem to lead to it being on its own bus: $ lsusb -t /: Bus 04.Port 1: Dev 1, Class=root_hub, Driver=xhci_hcd/2p, 10000M /: Bus 03.Port 1: Dev 1, Class=root_hub, Driver=xhci_hcd/4p, 480M |__ Port 3: Dev 2, If 0, Class=Human Interface Device, Driver=usbhid, 12M |__ Port 3: Dev 2, If 1, Class=Human Interface Device, Driver=usbhid, 12M |__ Port 4: Dev 3, If 0, Class=Wireless, Driver=btusb, 12M |__ Port 4: Dev 3, If 1, Class=Wireless, Driver=btusb, 12M /: Bus 02.Port 1: Dev 1, Class=root_hub, Driver=xhci_hcd/2p, 10000M |__ Port 2: Dev 2, If 0, Class=Hub, Driver=hub/4p, 5000M /: Bus 01.Port 1: Dev 1, Class=root_hub, Driver=xhci_hcd/4p, 480M |__ Port 2: Dev 2, If 0, Class=Hub, Driver=hub/4p, 480M |__ Port 2: Dev 7, If 0, Class=Audio, Driver=snd-usb-audio, 12M |__ Port 2: Dev 7, If 1, Class=Audio, Driver=snd-usb-audio, 12M |__ Port 2: Dev 7, If 2, Class=Audio, Driver=snd-usb-audio, 12M |__ Port 2: Dev 7, If 3, Class=Human Interface Device, Driver=usbhid, 12M |__ Port 4: Dev 4, If 0, Class=Human Interface Device, Driver=usbhid, 12M In this case the dmesg output is clear after device connection. This seems like some kind of bandwidth calculation problem in snd_usb_audio. I've tried building a kernel and modifying various of the defines in sound/usb/card.h (currently MAX_PACKS 4 and MAX_PACKS_HS (MAX_PACKS * 4), compared to 6 and *8) but not hit on a winning formula yet. Is there any information I could collect to try to fix this? -- imalone

2 6

Internal speakers and headphone not working on Huawei notebook
by Jacek Blocki 20 Mar '24

20 Mar '24

Hello, I have Huawei notebook (KLVL-WXXW-PCB) with Ubuntu 22.04.3 LTS installed alongside Windows. In Ubuntu, neither internal speakers nor headphone jack work. I only see Dummy Output while trying to adjust volume, but sound works with external HDMI monitor. Internal speakers work with Windows, so I believe hardware is fine and kernel driver (snd_hda_intel?) needs to be modified to fix the problem. I have submitted Ubuntu bug report back in January, but nobody touched it. So I try this list hoping for better result. $ cat /proc/asound/cards 0 [Generic ]: HDA-Intel - HD-Audio Generic HD-Audio Generic at 0xd03c0000 irq 64 1 [acp ]: acp - acp HUAWEI-KLVL_WXXW-M1010-KLVL_WXXW_PCB I run 6.5.0-26-generic kernel. Regards, Jacek Błocki

1 0

[sound/soc/qcom] Question about memory allocation in asoc_qcom_lpass_cpu_platform_probe
by Zijie Zhao 20 Mar '24

20 Mar '24

Dear QCOM AUDIO maintainers, We encountered an usual usage of devm_kzalloc while performing a static analysis for kernel code. https://elixir.bootlin.com/linux/v6.8/source/sound/soc/qcom/lpass-cpu.c#L12… ``` int asoc_qcom_lpass_cpu_platform_probe(struct platform_device *pdev) { ... drvdata = devm_kzalloc(dev, sizeof(struct lpass_data), GFP_KERNEL); if (!drvdata) return -ENOMEM; ... /* Allocation for i2sctl regmap fields */ drvdata->i2sctl = devm_kzalloc(&pdev->dev, sizeof(struct lpaif_i2sctl), GFP_KERNEL); /* Initialize bitfields for dai I2SCTL register */ ret = lpass_cpu_init_i2sctl_bitfields(dev, drvdata->i2sctl, drvdata->lpaif_map); if (ret) { dev_err(dev, "error init i2sctl field: %d\n", ret); return ret; } ... } ``` ``` static int lpass_cpu_init_i2sctl_bitfields(struct device *dev, struct lpaif_i2sctl *i2sctl, struct regmap *map) { struct lpass_data *drvdata = dev_get_drvdata(dev); const struct lpass_variant *v = drvdata->variant; i2sctl->loopback = devm_regmap_field_alloc(dev, map, v->loopback); ... } ``` Here devm_kzalloc might return NULL for drvdata->i2sctl if out-of-memory. Then i2sctl->loopback will cause segfault since i2sctl is never checked against NULL. Would a NULL check needed here? Like the drvdata above. Please let us know if we missed any key information or assumption! We appreciate your time! Best, Zijie

1 0

[PATCH] ASoC: SOF: Remove libraries from topology lookups
by cujomalainey＠chromium.org 20 Mar '24

20 Mar '24

From: Curtis Malainey <cujomalainey(a)chromium.org> Default firmware shipped in open source are not licensed for 3P libraries, therefore topologies should not reference them. If a OS wants to use 3P (that they have licensed) then they should use the appropriate topology override mechanisms. Fixes: 8a7d5d85ed2161 ("ASoC: SOF: mediatek: mt8195: Add devicetree support to select topologies") Signed-off-by: Curtis Malainey <cujomalainey(a)chromium.org> Cc: Wojciech Macek <wmacek(a)google.com> --- sound/soc/sof/mediatek/mt8195/mt8195.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sof/mediatek/mt8195/mt8195.c b/sound/soc/sof/mediatek/mt8195/mt8195.c index 8ee7ee246344c..f143baf75af60 100644 --- a/sound/soc/sof/mediatek/mt8195/mt8195.c +++ b/sound/soc/sof/mediatek/mt8195/mt8195.c @@ -573,7 +573,7 @@ static struct snd_sof_dsp_ops sof_mt8195_ops = { static struct snd_sof_of_mach sof_mt8195_machs[] = { { .compatible = "google,tomato", - .sof_tplg_filename = "sof-mt8195-mt6359-rt1019-rt5682-dts.tplg" + .sof_tplg_filename = "sof-mt8195-mt6359-rt1019-rt5682.tplg" }, { .compatible = "mediatek,mt8195", .sof_tplg_filename = "sof-mt8195.tplg" -- 2.44.0.291.gc1ea87d7ee-goog

4 6

[PATCH 1/2] ASoC: codecs: ES8326: Reducing power consumption
by Zhang Yi 20 Mar '24

20 Mar '24

> > For lower power consumption during hibernation, the configuration of > > es8326_suspend and es8326_remove will be adjusted. > > Adding es8326_i2c_shutdown and es8326_i2c_remove to cover different > > situations > > > > Signed-off-by: Zhang Yi <zhangyi(a)everest-semi.com> > > --- > > sound/soc/codecs/es8326.c | 34 +++++++++++++++++++++++++++++++++- > > 1 file changed, 33 insertions(+), 1 deletion(-) > > > > diff --git a/sound/soc/codecs/es8326.c b/sound/soc/codecs/es8326.c > > index 15289dadafea..5c89323035a2 100755 > > --- a/sound/soc/codecs/es8326.c > > +++ b/sound/soc/codecs/es8326.c > > @@ -1072,12 +1072,13 @@ static int es8326_suspend(struct snd_soc_component *component) > > es8326->calibrated = false; > > regmap_write(es8326->regmap, ES8326_CLK_CTL, ES8326_CLK_OFF); > > regcache_cache_only(es8326->regmap, true); > > - regcache_mark_dirty(es8326->regmap); > > > > /* reset register value to default */ > > regmap_write(es8326->regmap, ES8326_CSM_I2C_STA, 0x01); > > usleep_range(1000, 3000); > > regmap_write(es8326->regmap, ES8326_CSM_I2C_STA, 0x00); > > + > > + regcache_mark_dirty(es8326->regmap); > > return 0; > > } > > > > @@ -1163,8 +1164,13 @@ static int es8326_set_jack(struct snd_soc_component *component, > > > > static void es8326_remove(struct snd_soc_component *component) > > { > > + struct es8326_priv *es8326 = snd_soc_component_get_drvdata(component); > > + > > es8326_disable_jack_detect(component); > > es8326_set_bias_level(component, SND_SOC_BIAS_OFF); > > + regmap_write(es8326->regmap, ES8326_CSM_I2C_STA, 0x01); > > + usleep_range(1000, 3000); > > + regmap_write(es8326->regmap, ES8326_CSM_I2C_STA, 0x00); > > } > > > > static const struct snd_soc_component_driver soc_component_dev_es8326 = { > > @@ -1236,6 +1242,30 @@ static int es8326_i2c_probe(struct i2c_client *i2c) > > &es8326_dai, 1); > > } > > > > + > > +static void es8326_i2c_shutdown(struct i2c_client *i2c) > > +{ > > + struct snd_soc_component *component; > > + struct es8326_priv *es8326; > > + > > + dev_dbg(component->dev, "Enter into %s\n", __func__); > Above line works on unset pointer, you need to set component before > dereferencing it. > > + es8326 = i2c_get_clientdata(i2c); > > + component = es8326->component; > > + cancel_delayed_work_sync(&es8326->jack_detect_work); > > + cancel_delayed_work_sync(&es8326->button_press_work); > > + > > + regmap_write(es8326->regmap, ES8326_CSM_I2C_STA, 0x01); > > + usleep_range(1000, 3000); > > + regmap_write(es8326->regmap, ES8326_CSM_I2C_STA, 0x00); > > + > > + return; > > +} > > + > > +static void es8326_i2c_remove(struct i2c_client *i2c) > > +{ > > + es8326_i2c_shutdown(i2c); > > +} > > + > > static const struct i2c_device_id es8326_i2c_id[] = { > > {"es8326", 0 }, > > {} > > @@ -1265,6 +1295,8 @@ static struct i2c_driver es8326_i2c_driver = { > > .of_match_table = of_match_ptr(es8326_of_match), > > }, > > .probe = es8326_i2c_probe, > > + .shutdown = es8326_i2c_shutdown, > > + .remove = es8326_i2c_remove, > > .id_table = es8326_i2c_id, > > }; > > module_i2c_driver(es8326_i2c_driver); I'm sorry. I'll fix my mistake.

1 0

[PATCH 0/2] ASoC: codecs: ES8326: Reducin powerconsumption and
by Zhang Yi 20 Mar '24

20 Mar '24

We changed the configuration related to hibernation. and delete the REG_SUPPLY to cover mute issue. Zhang Yi (2): ASoC: codecs: ES8326: Reducing power consumption ASoC: codecs: ES8326: Delete unused REG_SUPPLY sound/soc/codecs/es8326.c | 42 ++++++++++++++++++++++++++++++--------- 1 file changed, 33 insertions(+), 9 deletions(-) -- 2.17.1

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[PATCH] ALSA: hda/realtek: Add quirks for some Clevo laptops
by Tim Crawford 20 Mar '24

20 Mar '24

Add audio quirks to fix speaker output and headset detection on some new Clevo models: - L240TU (ALC245) - PE60SNE-G (ALC1220) - V350SNEQ (ALC245) Co-authored-by: Jeremy Soller <jeremy(a)system76.com> Signed-off-by: Tim Crawford <tcrawford(a)system76.com> --- sound/pci/hda/patch_realtek.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index d463d416fc23..25183996756c 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2645,6 +2645,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x1558, 0x65f1, "Clevo PC50HS", ALC1220_FIXUP_CLEVO_PB51ED_PINS), SND_PCI_QUIRK(0x1558, 0x65f5, "Clevo PD50PN[NRT]", ALC1220_FIXUP_CLEVO_PB51ED_PINS), SND_PCI_QUIRK(0x1558, 0x66a2, "Clevo PE60RNE", ALC1220_FIXUP_CLEVO_PB51ED_PINS), + SND_PCI_QUIRK(0x1558, 0x66a6, "Clevo PE60SN[CDE]-[GS]", ALC1220_FIXUP_CLEVO_PB51ED_PINS), SND_PCI_QUIRK(0x1558, 0x67d1, "Clevo PB71[ER][CDF]", ALC1220_FIXUP_CLEVO_PB51ED_PINS), SND_PCI_QUIRK(0x1558, 0x67e1, "Clevo PB71[DE][CDF]", ALC1220_FIXUP_CLEVO_PB51ED_PINS), SND_PCI_QUIRK(0x1558, 0x67e5, "Clevo PC70D[PRS](?:-D|-G)?", ALC1220_FIXUP_CLEVO_PB51ED_PINS), @@ -10139,12 +10140,14 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1462, 0xb120, "MSI Cubi MS-B120", ALC283_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x1462, 0xb171, "Cubi N 8GL (MS-B171)", ALC283_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x152d, 0x1082, "Quanta NL3", ALC269_FIXUP_LIFEBOOK), + SND_PCI_QUIRK(0x1558, 0x0353, "Clevo V35[05]SN[CDE]Q", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x1323, "Clevo N130ZU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x1325, "Clevo N15[01][CW]U", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x1401, "Clevo L140[CZ]U", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x1403, "Clevo N140CU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x1404, "Clevo N150CU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x14a1, "Clevo L141MU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1558, 0x2624, "Clevo L240TU", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x4018, "Clevo NV40M[BE]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x4019, "Clevo NV40MZ", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x4020, "Clevo NV40MB", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), -- 2.44.0

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Re: [PATCH] Add Asus VivoBook S 16X to quirks table for acp6x
by Geraldo Nascimento 20 Mar '24

20 Mar '24

On Tue, Mar 19, 2024 at 05:16:10PM +0800, marcel wrote: > Hi again, > > it's been a while since i sent my patch and I still can't see it in > kernel on github, so there's my question because I'm not sure if I > understand correctly. Should I just wait a little bit more and it just > takes time or send patch again and change things you pointed? Also which > version should I type after [PATCH v...], "1" ? Because it's my first > attempt? Sorry for taking your time and if the questions are obvious, > but it's my first time doing something like this. Hi Marcel, please address my past concerns regarding your patch and resubmit as v2. Remember to Cc: Takashi Iwai and Jaroslav Kysela besides linux-sound mailing list or your patch will fall into limbo again, overlooked by the maintainers. There's also a get_maintainers.pl script in the linux tree. You might wanna use that as it adds some relevant people who might take a interest in your patch. Thanks, Geraldo Nascimento > > Greetings

1 0

[PATCH v14 00/16] Add audio support in v4l2 framework
by Shengjiu Wang 19 Mar '24

19 Mar '24

Audio signal processing also has the requirement for memory to memory similar as Video. This asrc memory to memory (memory ->asrc->memory) case is a non real time use case. User fills the input buffer to the asrc module, after conversion, then asrc sends back the output buffer to user. So it is not a traditional ALSA playback and capture case. It is a specific use case, there is no reference in current kernel. v4l2 memory to memory is the closed implementation, v4l2 current support video, image, radio, tuner, touch devices, so it is not complicated to add support for this specific audio case. Because we had implemented the "memory -> asrc ->i2s device-> codec" use case in ALSA. Now the "memory->asrc->memory" needs to reuse the code in asrc driver, so the first 3 patches is for refining the code to make it can be shared by the "memory->asrc->memory" driver. The main change is in the v4l2 side, A /dev/vl4-audioX will be created, user applications only use the ioctl of v4l2 framework. Other change is to add memory to memory support for two kinds of i.MX ASRC module. changes in v14: - document the reservation of 'AUXX' fourcc format. - add v4l2_audfmt_to_fourcc() definition. changes in v13 - change 'pixelformat' to 'audioformat' in dev-audio-mem2mem.rst - add more description for clock drift in ext-ctrls-audio-m2m.rst - Add "media: v4l2-ctrls: add support for fraction_bits" from Hans to avoid build issue for kernel test robot changes in v12 - minor changes according to comments - drop min_buffers_needed = 1 and V4L2_CTRL_FLAG_UPDATE flag - drop bus_info changes in v11 - add add-fixed-point-test-controls in vivid. - add v4l2_ctrl_fp_compose() helper function for min and max changes in v10 - remove FIXED_POINT type - change code base on media: v4l2-ctrls: add support for fraction_bits - fix issue reported by kernel test robot - remove module_alias changes in v9: - add MEDIA_ENT_F_PROC_AUDIO_RESAMPLER. - add MEDIA_INTF_T_V4L_AUDIO - add media controller support - refine the vim2m-audio to support 8k<->16k conversion. changes in v8: - refine V4L2_CAP_AUDIO_M2M to be 0x00000008 - update doc for FIXED_POINT - address comments for imx-asrc changes in v7: - add acked-by from Mark - separate commit for fixed point, m2m audio class, audio rate controls - use INTEGER_MENU for rate, FIXED_POINT for rate offset - remove used fmts - address other comments for Hans changes in v6: - use m2m_prepare/m2m_unprepare/m2m_start/m2m_stop to replace m2m_start_part_one/m2m_stop_part_one, m2m_start_part_two/m2m_stop_part_two. - change V4L2_CTRL_TYPE_ASRC_RATE to V4L2_CTRL_TYPE_FIXED_POINT - fix warning by kernel test rebot - remove some unused format V4L2_AUDIO_FMT_XX - Get SNDRV_PCM_FORMAT from V4L2_AUDIO_FMT in driver. - rename audm2m to viaudm2m. changes in v5: - remove V4L2_AUDIO_FMT_LPCM - define audio pixel format like V4L2_AUDIO_FMT_S8... - remove rate and format in struct v4l2_audio_format. - Add V4L2_CID_ASRC_SOURCE_RATE and V4L2_CID_ASRC_DEST_RATE controls - updata document accordingly. changes in v4: - update document style - separate V4L2_AUDIO_FMT_LPCM and V4L2_CAP_AUDIO_M2M in separate commit changes in v3: - Modify documents for adding audio m2m support - Add audio virtual m2m driver - Defined V4L2_AUDIO_FMT_LPCM format type for audio. - Defined V4L2_CAP_AUDIO_M2M capability type for audio m2m case. - with modification in v4l-utils, pass v4l2-compliance test. changes in v2: - decouple the implementation in v4l2 and ALSA - implement the memory to memory driver as a platfrom driver and move it to driver/media - move fsl_asrc_common.h to include/sound folder Hans Verkuil (1): media: v4l2-ctrls: add support for fraction_bits Shengjiu Wang (15): ASoC: fsl_asrc: define functions for memory to memory usage ASoC: fsl_easrc: define functions for memory to memory usage ASoC: fsl_asrc: move fsl_asrc_common.h to include/sound ASoC: fsl_asrc: register m2m platform device ASoC: fsl_easrc: register m2m platform device media: uapi: Add V4L2_CAP_AUDIO_M2M capability flag media: v4l2: Add audio capture and output support media: uapi: Define audio sample format fourcc type media: uapi: Add V4L2_CTRL_CLASS_M2M_AUDIO media: uapi: Add audio rate controls support media: uapi: Declare interface types for Audio media: uapi: Add an entity type for audio resampler media: vivid: add fixed point test controls media: imx-asrc: Add memory to memory driver media: vim2m-audio: add virtual driver for audio memory to memory .../media/mediactl/media-types.rst | 11 + .../userspace-api/media/v4l/buffer.rst | 6 + .../userspace-api/media/v4l/common.rst | 1 + .../media/v4l/dev-audio-mem2mem.rst | 71 + .../userspace-api/media/v4l/devices.rst | 1 + .../media/v4l/ext-ctrls-audio-m2m.rst | 59 + .../userspace-api/media/v4l/pixfmt-audio.rst | 100 ++ .../userspace-api/media/v4l/pixfmt.rst | 1 + .../media/v4l/vidioc-enum-fmt.rst | 2 + .../media/v4l/vidioc-g-ext-ctrls.rst | 4 + .../userspace-api/media/v4l/vidioc-g-fmt.rst | 4 + .../media/v4l/vidioc-querycap.rst | 3 + .../media/v4l/vidioc-queryctrl.rst | 11 +- .../media/videodev2.h.rst.exceptions | 3 + .../media/common/videobuf2/videobuf2-v4l2.c | 4 + drivers/media/platform/nxp/Kconfig | 13 + drivers/media/platform/nxp/Makefile | 1 + drivers/media/platform/nxp/imx-asrc.c | 1256 +++++++++++++++++ drivers/media/test-drivers/Kconfig | 10 + drivers/media/test-drivers/Makefile | 1 + drivers/media/test-drivers/vim2m-audio.c | 793 +++++++++++ drivers/media/test-drivers/vivid/vivid-core.h | 2 + .../media/test-drivers/vivid/vivid-ctrls.c | 26 + drivers/media/v4l2-core/v4l2-compat-ioctl32.c | 9 + drivers/media/v4l2-core/v4l2-ctrls-api.c | 1 + drivers/media/v4l2-core/v4l2-ctrls-core.c | 93 +- drivers/media/v4l2-core/v4l2-ctrls-defs.c | 10 + drivers/media/v4l2-core/v4l2-dev.c | 21 + drivers/media/v4l2-core/v4l2-ioctl.c | 66 + drivers/media/v4l2-core/v4l2-mem2mem.c | 13 +- include/media/v4l2-ctrls.h | 13 +- include/media/v4l2-dev.h | 2 + include/media/v4l2-ioctl.h | 34 + .../fsl => include/sound}/fsl_asrc_common.h | 60 + include/uapi/linux/media.h | 2 + include/uapi/linux/v4l2-controls.h | 9 + include/uapi/linux/videodev2.h | 50 +- sound/soc/fsl/fsl_asrc.c | 144 ++ sound/soc/fsl/fsl_asrc.h | 4 +- sound/soc/fsl/fsl_asrc_dma.c | 2 +- sound/soc/fsl/fsl_easrc.c | 233 +++ sound/soc/fsl/fsl_easrc.h | 6 +- 42 files changed, 3128 insertions(+), 27 deletions(-) create mode 100644 Documentation/userspace-api/media/v4l/dev-audio-mem2mem.rst create mode 100644 Documentation/userspace-api/media/v4l/ext-ctrls-audio-m2m.rst create mode 100644 Documentation/userspace-api/media/v4l/pixfmt-audio.rst create mode 100644 drivers/media/platform/nxp/imx-asrc.c create mode 100644 drivers/media/test-drivers/vim2m-audio.c rename {sound/soc/fsl => include/sound}/fsl_asrc_common.h (60%) -- 2.34.1

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[PATCH v1 1/1] ASoC: fsl: imx-es8328: Switch to using gpiod API
by Andy Shevchenko 18 Mar '24

18 Mar '24

This updates the driver to gpiod API, and removes yet another use of of_get_named_gpio(). Signed-off-by: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> --- sound/soc/fsl/imx-es8328.c | 16 ++++++++++------ 1 file changed, 10 insertions(+), 6 deletions(-) diff --git a/sound/soc/fsl/imx-es8328.c b/sound/soc/fsl/imx-es8328.c index 6f0d031c1d5f..ad915c55cc35 100644 --- a/sound/soc/fsl/imx-es8328.c +++ b/sound/soc/fsl/imx-es8328.c @@ -3,7 +3,7 @@ // Copyright 2012 Freescale Semiconductor, Inc. // Copyright 2012 Linaro Ltd. -#include <linux/gpio.h> +#include <linux/gpio/consumer.h> #include <linux/module.h> #include <linux/of.h> #include <linux/of_platform.h> @@ -23,7 +23,7 @@ struct imx_es8328_data { struct snd_soc_card card; char codec_dai_name[DAI_NAME_SIZE]; char platform_name[DAI_NAME_SIZE]; - int jack_gpio; + struct gpio_desc *jack_gpiod; }; static struct snd_soc_jack_gpio headset_jack_gpios[] = { @@ -54,8 +54,8 @@ static int imx_es8328_dai_init(struct snd_soc_pcm_runtime *rtd) struct imx_es8328_data, card); int ret = 0; - /* Headphone jack detection */ - if (gpio_is_valid(data->jack_gpio)) { + if (data->jack_gpiod) { + /* Headphone jack detection */ ret = snd_soc_card_jack_new_pins(rtd->card, "Headphone", SND_JACK_HEADSET | SND_JACK_BTN_0, &headset_jack, @@ -64,7 +64,7 @@ static int imx_es8328_dai_init(struct snd_soc_pcm_runtime *rtd) if (ret) return ret; - headset_jack_gpios[0].gpio = data->jack_gpio; + headset_jack_gpios[0].desc = data->jack_gpiod; ret = snd_soc_jack_add_gpios(&headset_jack, ARRAY_SIZE(headset_jack_gpios), headset_jack_gpios); @@ -174,7 +174,11 @@ static int imx_es8328_probe(struct platform_device *pdev) data->dev = dev; - data->jack_gpio = of_get_named_gpio(pdev->dev.of_node, "jack-gpio", 0); + data->jack_gpiod = devm_gpiod_get_optional(dev, "jack"); + if (IS_ERR(data->jack_gpiod)) { + ret = PTR_ERR(data->jack_gpiod); + goto put_device; + } /* * CPU == Platform -- 2.43.0.rc1.1.gbec44491f096

1 1

[PATCH V0 0/1] ASoC: soc-compress: Fix and add missing DPCM locking
by Shalini Manjunatha 18 Mar '24

18 Mar '24

We find mising DPCM locking inside soc_compr_set_params_fe() before calling dpcm_be_dai_hw_params() and dpcm_be_dai_prepare() which cause lockdep assert for DPCM lock not held in __soc_pcm_hw_params() and __soc_pcm_prepare() in above flow when ever we play compress offload audio playback use case. To fix this issue added DPCM lock and unlock in both places of above code flow mentioned. Shalini Manjunatha (1): ASoC: soc-compress: Fix and add DPCM locking sound/soc/soc-compress.c | 4 ++++ 1 file changed, 4 insertions(+) -- 2.7.4

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[PATCH] ALSA: hda/realtek: fix the hp playback volume issue for LG machines
by Hui Wang 18 Mar '24

18 Mar '24

Recently we tested the headphone playback on 2 LG machines, if we set the volume to the max value or near to the max value, the sound is too loud, it could even bring harm to listeners. A workaround is to decrease the max volume to a reasonable value for the headphone's amplifier, then the users couldn't set the volume bigger than that value from the userspace. Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 26 ++++++++++++++++++++++++++ 1 file changed, 26 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index e904f62e1952..d463d416fc23 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6964,6 +6964,25 @@ static void alc256_fixup_mic_no_presence_and_resume(struct hda_codec *codec, } } +static void alc256_decrease_headphone_amp_val(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + u32 caps; + u8 nsteps, offs; + + if (action != HDA_FIXUP_ACT_PRE_PROBE) + return; + + caps = query_amp_caps(codec, 0x3, HDA_OUTPUT); + nsteps = ((caps & AC_AMPCAP_NUM_STEPS) >> AC_AMPCAP_NUM_STEPS_SHIFT) - 10; + offs = ((caps & AC_AMPCAP_OFFSET) >> AC_AMPCAP_OFFSET_SHIFT) - 10; + caps &= ~AC_AMPCAP_NUM_STEPS & ~AC_AMPCAP_OFFSET; + caps |= (nsteps << AC_AMPCAP_NUM_STEPS_SHIFT) | (offs << AC_AMPCAP_OFFSET_SHIFT); + + if (snd_hda_override_amp_caps(codec, 0x3, HDA_OUTPUT, caps)) + codec_warn(codec, "failed to override amp caps for NID 0x3\n"); +} + static void alc_fixup_dell4_mic_no_presence_quiet(struct hda_codec *codec, const struct hda_fixup *fix, int action) @@ -7382,6 +7401,7 @@ enum { ALC294_FIXUP_CS35L41_I2C_2, ALC245_FIXUP_CS35L56_SPI_4_HP_GPIO_LED, ALC256_FIXUP_ACER_SFG16_MICMUTE_LED, + ALC256_FIXUP_HEADPHONE_AMP_VOL, }; /* A special fixup for Lenovo C940 and Yoga Duet 7; @@ -9581,6 +9601,10 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc256_fixup_acer_sfg16_micmute_led, }, + [ALC256_FIXUP_HEADPHONE_AMP_VOL] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc256_decrease_headphone_amp_val, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -10319,6 +10343,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x9e56, "Lenovo ZhaoYang CF4620Z", ALC286_FIXUP_SONY_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1849, 0x1233, "ASRock NUC Box 1100", ALC233_FIXUP_NO_AUDIO_JACK), SND_PCI_QUIRK(0x1849, 0xa233, "Positivo Master C6300", ALC269_FIXUP_HEADSET_MIC), + SND_PCI_QUIRK(0x1854, 0x0440, "LG CQ6", ALC256_FIXUP_HEADPHONE_AMP_VOL), + SND_PCI_QUIRK(0x1854, 0x0441, "LG CQ6 AIO", ALC256_FIXUP_HEADPHONE_AMP_VOL), SND_PCI_QUIRK(0x19e5, 0x3204, "Huawei MACH-WX9", ALC256_FIXUP_HUAWEI_MACH_WX9_PINS), SND_PCI_QUIRK(0x19e5, 0x320f, "Huawei WRT-WX9 ", ALC256_FIXUP_ASUS_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1b35, 0x1235, "CZC B20", ALC269_FIXUP_CZC_B20), -- 2.34.1

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[PATCH] ALSA: hda: cs35l41: Support HP Spectre x360 14 eu0000
by Anthony I Gilea 18 Mar '24

18 Mar '24

The new HP Spectre x360 has _DSD for CS35L41 amps in ACPI but reset-gpios and spk-id-gpios are merged into single Package of size 4 so _DSD parser fails to parse it. Overwrite broken _DSD with the correct configuration. Signed-off-by: Anthony I Gilea <i(a)cpp.in> --- sound/pci/hda/cs35l41_hda_property.c | 2 ++ 1 file changed, 2 insertions(+) diff --ruNp a/sound/pci/hda/cs35l41_hda_property.c b/sound/pci/hda/cs35l41_hda_property.c --- a/sound/pci/hda/cs35l41_hda_property.c 2024-03-10 23:38:09.000000000 +0300 +++ b/sound/pci/hda/cs35l41_hda_property.c 2024-03-14 18:22:01.887566917 +0300 @@ -64,6 +64,7 @@ static const struct cs35l41_config cs35l { "103C8BE5", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4100, 24 }, { "103C8BE6", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4100, 24 }, { "103C8B3A", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4100, 24 }, + { "103C8C15", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4100, 24 }, { "104312AF", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 1000, 4500, 24 }, { "10431433", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4500, 24 }, { "10431463", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4500, 24 }, @@ -405,6 +406,7 @@ static const struct cs35l41_prop_model c { "CSC3551", "103C8BE5", generic_dsd_config }, { "CSC3551", "103C8BE6", generic_dsd_config }, { "CSC3551", "103C8B3A", generic_dsd_config }, + { "CSC3551", "103C8C15", generic_dsd_config }, { "CSC3551", "104312AF", generic_dsd_config }, { "CSC3551", "10431433", generic_dsd_config }, { "CSC3551", "10431463", generic_dsd_config },

3 2

[PATCH v11] ASoc: tas2783: Add tas2783 codec driver
by Shenghao Ding 18 Mar '24

18 Mar '24

The tas2783 is a smart audio amplifier with integrated MIPI SoundWire interface (Version 1.2.1 compliant), I2C, and I2S/TDM interfaces designed for portable applications. An on-chip DSP supports Texas Instruments SmartAmp speaker protection algorithm. The integrated speaker voltage and current sense provides for real-time monitoring of loudspeakers. The ASoC component provides the majority of the functionality of the device, all the audio functions. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- Change in v11: - simplify tasdevice_set_sdw_stream - fixed some Linux coding style - fixed the spelling mistakes - Select left/right channel based on unique id - a longer description has been added - remove unused crc8 in KCONFIG - Invert of RE here means reciprocal of RE - correct indentation - correct the wrong pow calibrated register - correct comment on RESET register - fixed usually lines don't end with an open parenthesis - fixed tasdevice_sdw_shutdown - use defines for function, entity, etc. for readable - remove unnecessary codec_lock in component_probe() - remove struct sdw_stream_data - remove unused macro definitions - use BIT(2) and BIT(1) instead of 0x2 and 0x1 - implement pm_runtime - reg_start is u8, reg_start + i only adds i byte, it should be reg_start + i * 4, because the calibration values - fixed total_sz in tas2783_calibration - add a define for TASDEVICE_REG(0, 0, 0x0a) - rework macro style in header file - Add select CRC32 - rewrite the mute and the return value handling - Add comments on bin file - Add link_id into the fw name to distinguish the same sdw id on different sdw id. - remove unnecessary mutex_lock - use set_active() in tasdevice_io_init and move all other parts in the probe. - move pm_runtime_get_noresume to the probe - fix the return value of putvol function - add put_noidle in tasdevice_sdw_remove - remove unused header file(of.h) - correct the style of the multi-line comment - Set the readable registers in tas2783_readable_register for debugfs - Use new PM macros - Use default put and get inline SOC_SINGLE_RANGE_TLV and SOC_SINGLE_SX_TLV - Use respective %pt to log the calibrated time - Remove unnecessary struct tm - Add trailing comma tasdevice_dapm_widgets and tasdevice_audio_map - Rewrite the return value handling in tasdevice_sdw_probe - use dev_num instead of id.unique_id - rework the log - Add code to check the size of calibrated data in UEFI - Add description of Layout of calibrated Data in UEFI - switch back link_id and unique_id to identify the devcie - unique_id offset as the device index of the calibrated data - update calibrated data layout stored in UEFI, add link_id and unique_id into the calibrated data in case of tas2783 sitting in different SDQW buses. - correct the wrong unique_id - set the unique_id as hex style in dsp bin file name --- sound/soc/codecs/Kconfig | 16 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tas2783-sdw.c | 932 +++++++++++++++++++++++++++++++++ sound/soc/codecs/tas2783.h | 106 ++++ 4 files changed, 1056 insertions(+) create mode 100644 sound/soc/codecs/tas2783-sdw.c create mode 100644 sound/soc/codecs/tas2783.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 59f9742e9ff4..6ba3fd0c0811 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -247,6 +247,7 @@ config SND_SOC_ALL_CODECS imply SND_SOC_TAS2781_COMLIB imply SND_SOC_TAS2781_FMWLIB imply SND_SOC_TAS2781_I2C + imply SND_SOC_TAS2783 imply SND_SOC_TAS5086 imply SND_SOC_TAS571X imply SND_SOC_TAS5720 @@ -1847,6 +1848,21 @@ config SND_SOC_TAS2781_I2C algo coefficient setting, for one, two or even multiple TAS2781 chips. +config SND_SOC_TAS2783 + tristate "Texas Instruments TAS2783 speaker amplifier (sdw)" + depends on SOUNDWIRE + depends on EFI + select REGMAP + select REGMAP_SOUNDWIRE + select CRC32 + select CRC32_SARWATE + help + Enable support for Texas Instruments TAS2783 Smart Amplifier + Digital input mono Class-D and DSP-inside audio power amplifiers. + TAS2783 supports the MIPI SoundWire v1.2 compatible interface + for audio and control data. At the end of calibrated data appends + the crc data with crc32_sarwate for error detection. + config SND_SOC_TAS5086 tristate "Texas Instruments TAS5086 speaker amplifier" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index f53baa2b9565..6afceb126ae3 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -284,6 +284,7 @@ snd-soc-tas2770-objs := tas2770.o snd-soc-tas2781-comlib-objs := tas2781-comlib.o snd-soc-tas2781-fmwlib-objs := tas2781-fmwlib.o snd-soc-tas2781-i2c-objs := tas2781-i2c.o +snd-soc-tas2783-objs := tas2783-sdw.o snd-soc-tfa9879-objs := tfa9879.o snd-soc-tfa989x-objs := tfa989x.o snd-soc-tlv320adc3xxx-objs := tlv320adc3xxx.o @@ -664,6 +665,7 @@ obj-$(CONFIG_SND_SOC_TAS2780) += snd-soc-tas2780.o obj-$(CONFIG_SND_SOC_TAS2781_COMLIB) += snd-soc-tas2781-comlib.o obj-$(CONFIG_SND_SOC_TAS2781_FMWLIB) += snd-soc-tas2781-fmwlib.o obj-$(CONFIG_SND_SOC_TAS2781_I2C) += snd-soc-tas2781-i2c.o +obj-$(CONFIG_SND_SOC_TAS2783) += snd-soc-tas2783.o obj-$(CONFIG_SND_SOC_TAS5086) += snd-soc-tas5086.o obj-$(CONFIG_SND_SOC_TAS571X) += snd-soc-tas571x.o obj-$(CONFIG_SND_SOC_TAS5720) += snd-soc-tas5720.o diff --git a/sound/soc/codecs/tas2783-sdw.c b/sound/soc/codecs/tas2783-sdw.c new file mode 100644 index 000000000000..71c218792bc9 --- /dev/null +++ b/sound/soc/codecs/tas2783-sdw.c @@ -0,0 +1,932 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// ALSA SoC Texas Instruments TAS2783 Audio Smart Amplifier +// +// Copyright (C) 2023 - 2024 Texas Instruments Incorporated +// https://www.ti.com +// +// Author: Baojun Xu <baojun.xu(a)ti.com> +// Kevin Lu <kevin-lu(a)ti.com> +// Shenghao Ding <shenghao-ding(a)ti.com> +// + +#include <asm/unaligned.h> +#include <linux/acpi.h> +#include <linux/crc32.h> +#include <linux/efi.h> +#include <linux/err.h> +#include <linux/firmware.h> +#include <linux/init.h> +#include <linux/module.h> +#include <linux/pm.h> +#include <linux/pm_runtime.h> +#include <linux/regmap.h> +#include <linux/soundwire/sdw.h> +#include <linux/soundwire/sdw_registers.h> +#include <linux/soundwire/sdw_type.h> +#include <sound/pcm_params.h> +#include <sound/sdw.h> +#include <sound/soc.h> +#include <sound/tlv.h> + +#include <sound/tas2781-tlv.h> +#include "tas2783.h" + +static const unsigned int tas2783_cali_reg[] = { + TAS2783_CALIBRATION_RE, /* Resistance */ + TAS2783_CALIBRATION_RE_LOW, /* Low limitation of RE */ + TAS2783_CALIBRATION_INV_RE, /* Reciprocal of RE */ + TAS2783_CALIBRATION_POW, /* RMS Power */ + TAS2783_CALIBRATION_TLIMIT /* Temperature limitation */ +}; + +static const struct reg_default tas2783_reg_defaults[] = { + /* Default values for ROM mode. Activated. */ + { 0x8002, 0x1a }, /* AMP inactive. */ + { 0x8097, 0xc8 }, + { 0x80b5, 0x74 }, + { 0x8099, 0x20 }, + { 0xfe8d, 0x0d }, + { 0xfebe, 0x4a }, + { 0x8230, 0x00 }, + { 0x8231, 0x00 }, + { 0x8232, 0x00 }, + { 0x8233, 0x01 }, + { 0x8418, 0x00 }, /* Set volume to 0 dB. */ + { 0x8419, 0x00 }, + { 0x841a, 0x00 }, + { 0x841b, 0x00 }, + { 0x8428, 0x40 }, /* Unmute channel */ + { 0x8429, 0x00 }, + { 0x842a, 0x00 }, + { 0x842b, 0x00 }, + { 0x8548, 0x00 }, /* Set volume to 0 dB. */ + { 0x8549, 0x00 }, + { 0x854a, 0x00 }, + { 0x854b, 0x00 }, + { 0x8558, 0x40 }, /* Unmute channel */ + { 0x8559, 0x00 }, + { 0x855a, 0x00 }, + { 0x855b, 0x00 }, + { 0x800a, 0x3a }, /* Enable both channel */ + { 0x800e, 0x44 }, + { 0x800f, 0x40 }, + { 0x805c, 0x99 }, + { SDW_SDCA_CTL(TAS2783_FUNC_TYPE_SMART_AMP, TAS2783_SDCA_ENT_FU21, + TAS2783_SDCA_CTL_FU_MUTE, 0), 0 }, + { SDW_SDCA_CTL(TAS2783_FUNC_TYPE_SMART_AMP, TAS2783_SDCA_ENT_FU21, + TAS2783_SDCA_CTL_FU_VOLUME, 0), 0 }, + { SDW_SDCA_CTL(TAS2783_FUNC_TYPE_SMART_AMP, TAS2783_SDCA_ENT_FU23, + TAS2783_SDCA_CTL_FU_MUTE, 0), 0 }, +}; + +static bool tas2783_readable_register(struct device *dev, + unsigned int reg) +{ + switch (reg) { + case 0x8000 ... 0x807F: /* Page 0 */ + return true; + default: + return false; + } +} + +static bool tas2783_volatile_register(struct device *dev, + unsigned int reg) +{ + switch (reg) { + /* + * Only software reset register was volatiled. + * Bit is self clearing. + * 0b = Don't reset + * 1b = reset + */ + case 0x8001: + return true; + default: + return false; + } +} + +static const struct regmap_config tasdevice_regmap = { + .reg_bits = 32, + .val_bits = 8, + .readable_reg = tas2783_readable_register, + .volatile_reg = tas2783_volatile_register, + .max_register = 0x44ffffff, + .reg_defaults = tas2783_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(tas2783_reg_defaults), + .cache_type = REGCACHE_RBTREE, + .use_single_read = true, + .use_single_write = true, +}; + +static const struct snd_kcontrol_new tas2783_snd_controls[] = { + SOC_SINGLE_RANGE_TLV("Digital Volume", TAS2783_DVC_LVL, + 0, 0, 200, 1, dvc_tlv), + SOC_SINGLE_SX_TLV("Amp Gain Volume", TAS2783_AMP_LEVEL, + 1, 0, 20, amp_vol_tlv), +}; + +static void tas2783_apply_calibv1(struct tasdevice_priv *tas_dev, + unsigned int *cali_data) +{ + struct regmap *map = tas_dev->regmap; + u8 *cali_start; + u16 offset; + int ret; + + if (!tas_dev->sdw_peripheral) { + dev_err(tas_dev->dev, "%s: peripheral doesn't exist.\n", + __func__); + return; + } + + offset = tas_dev->sdw_peripheral->id.unique_id - + TAS2783_UNIQUE_ID_BASE; + if (offset >= TAS2783_MAX_DEV_NUM) { + dev_err(tas_dev->dev, "%s: offset(%u) is out of range.\n", + __func__, offset); + return; + } + + /* + * The area saving tas2783 calibrated data is specified by its + * unique_id offset. cali_start is the first address of current + * tas2783's calibrated data. + */ + cali_start = (u8 *)(cali_data + offset * sizeof(tas2783_cali_reg)); + for (int i = 0; i < ARRAY_SIZE(tas2783_cali_reg); i++) { + ret = regmap_bulk_write(map, tas2783_cali_reg[i], + &cali_start[4 * i], 4); + if (ret) { + dev_err(tas_dev->dev, "Cali failed %x:%d\n", + tas2783_cali_reg[i], ret); + break; + } + } +} + +static void tas2783_apply_calibv2(struct tasdevice_priv *tas_dev, + unsigned int *cali_data) +{ + const unsigned int arr_size = ARRAY_SIZE(tas2783_cali_reg); + struct regmap *map = tas_dev->regmap; + unsigned int dev_sum = cali_data[1], i, j, k; + u8 *cali_start; + u16 dev_info; + int ret; + + if (!tas_dev->sdw_peripheral) { + dev_err(tas_dev->dev, "%s: peripheral doesn't exist.\n", + __func__); + return; + } + + dev_info = tas_dev->sdw_peripheral->bus->link_id | + tas_dev->sdw_peripheral->id.unique_id << 16; + + /* + * The area saving tas2783 calibrated data is specified by its + * unique_id offset. cali_start is the first address of current + * tas2783's calibrated data. + */ + cali_start = (u8 *)&cali_data[3]; + for (i = 0; i < dev_sum; i++) { + k = i * (arr_size + 1) + 3; + if (dev_info != cali_data[k]) { + for (j = 0; j < arr_size; j++) { + k = 4 * (k + 1 + j); + ret = regmap_bulk_write(map, + tas2783_cali_reg[j], + &cali_start[k], 4); + if (ret) { + dev_err(tas_dev->dev, + "Cali failed %x:%d\n", + tas2783_cali_reg[j], ret); + break; + } + } + break; + } + } +} + +/* + * Load the calibration data, including speaker impedance, f0, etc. + * Calibration is done by the manufacturer in the factory. The calibration + * data are used by the algorithm for calculating the speaker temperature, + * speaker membrane excursion and f0 in real time during playback. + * The DSP will work with default data values if calibrated data are + * missing or are invalid. + * Layout of calibrated Data in UEFI(total 256 bytes): + * V2: + * ChipID (2783, 4 bytes) + * Device-Sum (4 bytes) + * TimeStamp of Calibration (4 bytes) + * for (i = 0; i < Device-Sum; i++) { + * Device #i info () { + * SDW link id (2bytes) + * SDW unique_id (2bytes) + * } + * Calibrated Data of Device #i (20 bytes) + * } + * CRC (4 bytes) + * Reserved (the rest) + * Or + * V1: + * Calibrated Data of Dev 0(unique_id offset 0) (20 bytes) + * Calibrated Data of Dev 1(unique_id offset 1) (20 bytes) + * Calibrated Data of Dev 2(unique_id offset 2) (20 bytes) + * Calibrated Data of Dev 3(unique_id offset 3) (20 bytes) + * Calibrated Data of Dev 4(unique_id offset 4) (20 bytes) + * Calibrated Data of Dev 5(unique_id offset 5) (20 bytes) + * Calibrated Data of Dev 6(unique_id offset 6) (20 bytes) + * Calibrated Data of Dev 7(unique_id offset 7) (20 bytes) + * TimeStamp of Calibration (4 bytes) + * CRC (4 bytes) + * Reserved (88 bytes) + */ +static void tas2783_calibration(struct tasdevice_priv *tas_dev) +{ + efi_guid_t efi_guid = EFI_GUID(0x1f52d2a1, 0xbb3a, 0x457d, 0xbc, + 0x09, 0x43, 0xa3, 0xf4, 0x31, 0x0a, 0x92); + static efi_char16_t efi_name[] = L"CALI_DATA"; + struct calibration_data cali_data; + unsigned int *tmp_val; + unsigned int crc; + efi_status_t status; + + cali_data.total_sz = 0; + + status = efi.get_variable(efi_name, &efi_guid, NULL, + &cali_data.total_sz, NULL); + if (status == EFI_BUFFER_TOO_SMALL + && cali_data.total_sz < TAS2783_MAX_CALIDATA_SIZE) { + status = efi.get_variable(efi_name, &efi_guid, NULL, + &cali_data.total_sz, + cali_data.data); + dev_dbg(tas_dev->dev, "%s: cali get %lx bytes result:%ld\n", + __func__, cali_data.total_sz, status); + } + if (status != 0) { + /* Failed got calibration data from EFI. */ + dev_dbg(tas_dev->dev, "No calibration data in UEFI."); + return; + } + + tmp_val = (unsigned int *)cali_data.data; + + if (tmp_val[0] == 2783) { + /* Calibrated Data V2 */ + unsigned int dev_sum = tmp_val[1]; + + if (dev_sum > TAS2783_MAX_DEV_NUM || + dev_sum == 0) { + dev_dbg(tas_dev->dev, "No dev in calibrated data V2."); + return; + } + crc = crc32(~0, cali_data.data, 12 + dev_sum * 24) ^ ~0; + if (crc == tmp_val[3 + dev_sum * 6]) { + tas2783_apply_calibv2(tas_dev, tmp_val); + dev_dbg(tas_dev->dev, "V2: %ptTs", &tmp_val[40]); + } else { + dev_dbg(tas_dev->dev, + "V2: CRC 0x%08x not match 0x%08x\n", + crc, tmp_val[41]); + } + } else { + /* Calibrated Data V1 */ + /* 8 devs * 20 bytes calibrated data/dev + 4 bytes Timestamp */ + crc = crc32(~0, cali_data.data, 164) ^ ~0; + if (crc == tmp_val[41]) { + /* Date and time of when calibration was done. */ + tas2783_apply_calibv1(tas_dev, tmp_val); + dev_dbg(tas_dev->dev, "V1: %ptTs", &tmp_val[40]); + } else { + dev_dbg(tas_dev->dev, + "V1: CRC 0x%08x not match 0x%08x\n", + crc, tmp_val[41]); + } + } +} + +static void tasdevice_dspfw_ready(const struct firmware *fmw, + void *context) +{ + struct tasdevice_priv *tas_dev = + (struct tasdevice_priv *) context; + struct tas2783_firmware_node *p; + struct regmap *map = tas_dev->regmap; + unsigned char *buf = NULL; + int offset = 0, img_sz; + int ret; + + if (!fmw || !fmw->data) { + dev_warn(tas_dev->dev, + "%s: failed to read %s: work in bypass-dsp mode.\n", + __func__, tas_dev->dspfw_binaryname); + return; + } + buf = (unsigned char *)fmw->data; + + img_sz = get_unaligned_le32(&buf[offset]); + offset += sizeof(img_sz); + if (img_sz != fmw->size) { + dev_warn(tas_dev->dev, "%s: size not matching, %d %u.", + __func__, (int)fmw->size, img_sz); + return; + } + + while (offset < img_sz) { + p = (struct tas2783_firmware_node *)(buf + offset); + if (p->length > 1) { + ret = regmap_bulk_write(map, p->download_addr, + buf + offset + sizeof(unsigned int) * 5, p->length); + } else { + ret = regmap_write(map, p->download_addr, + *(buf + offset + sizeof(unsigned int) * 5)); + } + + if (ret != 0) { + dev_dbg(tas_dev->dev, + "%s: load FW fail: %d, work in bypass.\n", + __func__, ret); + return; + } + offset += sizeof(unsigned int) * 5 + p->length; + } + + tas2783_calibration(tas_dev); +} + +static const struct snd_soc_dapm_widget tasdevice_dapm_widgets[] = { + SND_SOC_DAPM_AIF_IN("ASI", "ASI Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("ASI OUT", "ASI Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_OUTPUT("OUT"), + SND_SOC_DAPM_INPUT("DMIC"), +}; + +static const struct snd_soc_dapm_route tasdevice_audio_map[] = { + {"OUT", NULL, "ASI"}, + {"ASI OUT", NULL, "DMIC"}, +}; + +static int tasdevice_set_sdw_stream(struct snd_soc_dai *dai, + void *sdw_stream, int direction) +{ + snd_soc_dai_dma_data_set(dai, direction, sdw_stream); + + return 0; +} + +static void tasdevice_sdw_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + snd_soc_dai_set_dma_data(dai, substream, NULL); +} + +static int tasdevice_sdw_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct tasdevice_priv *tas_priv = + snd_soc_component_get_drvdata(component); + struct sdw_stream_config stream_config = {0}; + struct sdw_port_config port_config = {0}; + struct sdw_stream_runtime *sdw_stream; + int ret; + + dev_dbg(dai->dev, "%s: dai_name %s", __func__, dai->name); + sdw_stream = snd_soc_dai_get_dma_data(dai, substream); + + if (!sdw_stream) + return -EINVAL; + + if (!tas_priv->sdw_peripheral) + return -EINVAL; + + /* SoundWire specific configuration */ + snd_sdw_params_to_config(substream, params, + &stream_config, &port_config); + + /* port 1 for playback */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + port_config.num = 1; + else + port_config.num = 2; + + ret = sdw_stream_add_slave(tas_priv->sdw_peripheral, + &stream_config, &port_config, 1, sdw_stream); + if (ret) { + dev_err(dai->dev, "%s: Unable to configure port\n", __func__); + return ret; + } + + dev_dbg(dai->dev, "%s: format: %i rate: %i\n", __func__, + params_format(params), params_rate(params)); + + return 0; +} + +static int tasdevice_sdw_pcm_hw_free(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct tasdevice_priv *tas_priv = + snd_soc_component_get_drvdata(component); + struct sdw_stream_runtime *sdw_stream = + snd_soc_dai_get_dma_data(dai, substream); + + if (!tas_priv->sdw_peripheral) + return -EINVAL; + + sdw_stream_remove_slave(tas_priv->sdw_peripheral, sdw_stream); + + return 0; +} + +static int tasdevice_mute(struct snd_soc_dai *dai, int mute, + int direction) +{ + struct snd_soc_component *component = dai->component; + struct tasdevice_priv *tas_dev = + snd_soc_component_get_drvdata(component); + struct regmap *map = tas_dev->regmap; + int ret; + + dev_dbg(tas_dev->dev, "%s: %d.\n", __func__, mute); + + if (mute) { + if (direction == SNDRV_PCM_STREAM_CAPTURE) { + ret = regmap_update_bits(map, TAS2873_REG_PWR_CTRL, + TAS2783_REG_AEF_MASK, + TAS2783_REG_AEF_INACTIVE); + if (ret) + dev_err(tas_dev->dev, + "%s: Disable AEF failed.\n", __func__); + } else { + /* FU23 mute (0x40400108) */ + ret = regmap_write(map, + SDW_SDCA_CTL(TAS2783_FUNC_TYPE_SMART_AMP, + TAS2783_SDCA_ENT_FU23, + TAS2783_SDCA_CTL_FU_MUTE, 0), 1); + if (ret) { + dev_err(tas_dev->dev, + "%s: FU23 mute failed.\n", __func__); + goto out; + } + /* + * Both playback and echo data will be shutdown in + * playback stream. + */ + ret = regmap_update_bits(map, TAS2873_REG_PWR_CTRL, + TAS2783_REG_PWR_MODE_MASK | + TAS2783_REG_AEF_MASK, + TAS2783_REG_PWR_MODE_ACTIVE | + TAS2783_REG_PWR_MODE_SW_PWD); + if (ret) { + dev_err(tas_dev->dev, + "%s: PWR&AEF shutdown failed.\n", + __func__); + goto out; + } + tas_dev->pstream = false; + } + } else { + /* FU23 Unmute, 0x40400108. */ + if (direction == SNDRV_PCM_STREAM_PLAYBACK) { + ret = regmap_write(map, + SDW_SDCA_CTL(TAS2783_FUNC_TYPE_SMART_AMP, + TAS2783_SDCA_ENT_FU23, + TAS2783_SDCA_CTL_FU_MUTE, 0), 0); + if (ret) { + dev_err(tas_dev->dev, + "%s: FU23 Unmute failed.\n", __func__); + goto out; + } + ret = regmap_update_bits(map, TAS2873_REG_PWR_CTRL, + TAS2783_REG_PWR_MODE_MASK, + TAS2783_REG_PWR_MODE_ACTIVE); + if (ret) { + dev_err(tas_dev->dev, + "%s: PWR Unmute failed.\n", __func__); + goto out; + } + tas_dev->pstream = true; + } else { + /* Capture stream is the echo ref data for voice. + * Without playback, it can't be active. + */ + if (tas_dev->pstream == true) { + ret = regmap_update_bits(map, + TAS2873_REG_PWR_CTRL, + TAS2783_REG_AEF_MASK, + TAS2783_REG_AEF_ACTIVE); + if (ret) { + dev_err(tas_dev->dev, + "%s: AEF enable failed.\n", + __func__); + goto out; + } + } else { + dev_err(tas_dev->dev, + "%s: No playback, no AEF!", __func__); + ret = -EINVAL; + } + } + } +out: + if (ret) + dev_err(tas_dev->dev, "Mute or unmute %d failed %d.\n", + mute, ret); + + return ret; +} + +static const struct snd_soc_dai_ops tasdevice_dai_ops = { + .mute_stream = tasdevice_mute, + .hw_params = tasdevice_sdw_hw_params, + .hw_free = tasdevice_sdw_pcm_hw_free, + .set_stream = tasdevice_set_sdw_stream, + .shutdown = tasdevice_sdw_shutdown, +}; + +static struct snd_soc_dai_driver tasdevice_dai_driver[] = { + { + .name = "tas2783-codec", + .id = 0, + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 4, + .rates = TAS2783_DEVICE_RATES, + .formats = TAS2783_DEVICE_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 4, + .rates = TAS2783_DEVICE_RATES, + .formats = TAS2783_DEVICE_FORMATS, + }, + .ops = &tasdevice_dai_ops, + .symmetric_rate = 1, + }, +}; + +static int tasdevice_comp_probe(struct snd_soc_component *comp) +{ + struct tasdevice_priv *tas_dev = snd_soc_component_get_drvdata(comp); + acpi_handle handle = ACPI_HANDLE(tas_dev->dev); + const struct firmware *fw_entry = NULL; + const char *sub = NULL; + int ret; + + if (handle) { + sub = acpi_get_subsystem_id(handle); + if (IS_ERR(sub)) + sub = NULL; + } + + tas_dev->component = comp; + + /* + * Each tas2783 in the system has its own dspfw. + */ + if (comp->name_prefix) { + /* + * name_prefix.bin stores the dsp firmware including speaker + * protection algorithm, audio acoustic algorithm, speaker + * characters and algorithm params, it must be copied into + * firmware folder. + */ + scnprintf(tas_dev->dspfw_binaryname, + TAS2783_DSPFW_FILENAME_LEN, "%s-tas2783.bin", + comp->name_prefix); + } else { + /* Compatible with the previous naming rule */ + if (sub) { + /* + * subsys_id-link_id[0,1,...,N]-unique_id + * [0x8,...,0xf].bin stores the dsp firmware including + * speaker protection algorithm, audio acoustic + * algorithm, speaker characters and algorithm params, + * it must be copied into firmware folder. + */ + scnprintf(tas_dev->dspfw_binaryname, + TAS2783_DSPFW_FILENAME_LEN, + "%s-%d-0x%x.bin", sub, + tas_dev->sdw_peripheral->bus->link_id, + tas_dev->sdw_peripheral->id.unique_id); + } else { + /* + * tas2783-link_id[0,1,...,N]-unique_id + * [0x8,...,0xf].bin stores the dsp firmware including + * speaker protection algorithm, audio acoustic + * algorithm, speaker characters and algorithm params, + * it must be copied into firmware folder. + */ + scnprintf(tas_dev->dspfw_binaryname, + TAS2783_DSPFW_FILENAME_LEN, + "tas2783-%d-0x%x.bin", + tas_dev->sdw_peripheral->bus->link_id, + tas_dev->sdw_peripheral->id.unique_id); + } + } + + ret = request_firmware(&fw_entry, tas_dev->dspfw_binaryname, + tas_dev->dev); + if (ret) { + dev_err(tas_dev->dev, + "%s: request_firmware %x open status: %d.\n", __func__, + tas_dev->sdw_peripheral->id.unique_id, ret); + goto out; + } + + tasdevice_dspfw_ready(fw_entry, tas_dev); + +out: + if (fw_entry) + release_firmware(fw_entry); + return 0; +} + +static const struct snd_soc_component_driver + soc_codec_driver_tasdevice = { + .probe = tasdevice_comp_probe, + .controls = tas2783_snd_controls, + .num_controls = ARRAY_SIZE(tas2783_snd_controls), + .dapm_widgets = tasdevice_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(tasdevice_dapm_widgets), + .dapm_routes = tasdevice_audio_map, + .num_dapm_routes = ARRAY_SIZE(tasdevice_audio_map), + .idle_bias_on = 1, + .endianness = 1, +}; + +static int tasdevice_init(struct tasdevice_priv *tas_dev) +{ + int ret; + + dev_set_drvdata(tas_dev->dev, tas_dev); + + ret = devm_snd_soc_register_component(tas_dev->dev, + &soc_codec_driver_tasdevice, + tasdevice_dai_driver, ARRAY_SIZE(tasdevice_dai_driver)); + if (ret) { + dev_err(tas_dev->dev, "%s: codec register error:%d.\n", + __func__, ret); + return ret; + } + + /* set autosuspend parameters */ + pm_runtime_set_autosuspend_delay(tas_dev->dev, 3000); + pm_runtime_use_autosuspend(tas_dev->dev); + + /* make sure the device does not suspend immediately */ + pm_runtime_mark_last_busy(tas_dev->dev); + pm_runtime_enable(tas_dev->dev); + + return 0; +} + +static int tasdevice_read_prop(struct sdw_slave *slave) +{ + struct sdw_slave_prop *prop = &slave->prop; + struct sdw_dpn_prop *dpn; + unsigned long addr; + int nval, i, j; + u32 bit; + + prop->scp_int1_mask = SDW_SCP_INT1_BUS_CLASH | SDW_SCP_INT1_PARITY; + prop->quirks = SDW_SLAVE_QUIRKS_INVALID_INITIAL_PARITY; + + prop->paging_support = true; + + /* first we need to allocate memory for set bits in port lists */ + prop->source_ports = BIT(2); /* BITMAP: 00000100 */ + prop->sink_ports = BIT(1); /* BITMAP: 00000010 */ + + nval = hweight32(prop->source_ports); + prop->src_dpn_prop = devm_kcalloc(&slave->dev, nval, + sizeof(*prop->src_dpn_prop), GFP_KERNEL); + if (!prop->src_dpn_prop) + return -ENOMEM; + + i = 0; + dpn = prop->src_dpn_prop; + addr = prop->source_ports; + for_each_set_bit(bit, &addr, 32) { + dpn[i].num = bit; + dpn[i].type = SDW_DPN_FULL; + dpn[i].simple_ch_prep_sm = true; + dpn[i].ch_prep_timeout = 10; + i++; + } + + /* do this again for sink now */ + nval = hweight32(prop->sink_ports); + prop->sink_dpn_prop = devm_kcalloc(&slave->dev, nval, + sizeof(*prop->sink_dpn_prop), GFP_KERNEL); + if (!prop->sink_dpn_prop) + return -ENOMEM; + + j = 0; + dpn = prop->sink_dpn_prop; + addr = prop->sink_ports; + for_each_set_bit(bit, &addr, 32) { + dpn[j].num = bit; + dpn[j].type = SDW_DPN_FULL; + dpn[j].simple_ch_prep_sm = true; + dpn[j].ch_prep_timeout = 10; + j++; + } + + /* set the timeout values */ + prop->clk_stop_timeout = 20; + + return 0; +} + +static int tasdevice_io_init(struct device *dev, + struct sdw_slave *slave) +{ + struct tasdevice_priv *tas_priv = dev_get_drvdata(dev); + int ret; + + regcache_cache_only(tas_priv->regmap, false); + + if (tas_priv->first_hw_init) { + regcache_cache_bypass(tas_priv->regmap, true); + } else { + /* + * PM runtime is only enabled when a Slave reports as Attached + * Update count of parent 'active' children. + */ + pm_runtime_set_active(&slave->dev); + } + + /* sw reset */ + ret = regmap_write(tas_priv->regmap, TAS2873_REG_SWRESET, + TAS2873_REG_SWRESET_RESET); + if (ret) { + dev_err(tas_priv->dev, "Reset failed.\n"); + goto out; + } + + if (tas_priv->first_hw_init) { + regcache_cache_bypass(tas_priv->regmap, false); + regcache_mark_dirty(tas_priv->regmap); + } + + tas_priv->first_hw_init = true; + tas_priv->hw_init = true; + +out: + return ret; +} + +static int tasdevice_update_status(struct sdw_slave *slave, + enum sdw_slave_status status) +{ + struct tasdevice_priv *tas_priv = dev_get_drvdata(&slave->dev); + + /* Update the status */ + tas_priv->status = status; + + if (status == SDW_SLAVE_UNATTACHED) + tas_priv->hw_init = false; + /* + * Perform initialization only if slave status + * is present and hw_init flag is false + */ + if (tas_priv->hw_init || tas_priv->status != SDW_SLAVE_ATTACHED) + return 0; + + /* perform I/O transfers required for Slave initialization */ + return tasdevice_io_init(&slave->dev, slave); +} + +/* + * slave_ops: callbacks for get_clock_stop_mode, clock_stop and + * port_prep are not defined for now + */ +static const struct sdw_slave_ops tasdevice_sdw_ops = { + .read_prop = tasdevice_read_prop, + .update_status = tasdevice_update_status, +}; + +static int tasdevice_sdw_probe(struct sdw_slave *peripheral, + const struct sdw_device_id *id) +{ + struct device *dev = &peripheral->dev; + struct tasdevice_priv *tas_dev; + + tas_dev = devm_kzalloc(dev, sizeof(*tas_dev), GFP_KERNEL); + if (!tas_dev) + return -ENOMEM; + + tas_dev->dev = dev; + tas_dev->chip_id = id->driver_data; + tas_dev->sdw_peripheral = peripheral; + /* + * Mark hw_init to false + * HW init will be performed when device reports present + */ + tas_dev->hw_init = false; + tas_dev->first_hw_init = false; + + dev_set_drvdata(dev, tas_dev); + + tas_dev->regmap = devm_regmap_init_sdw(peripheral, &tasdevice_regmap); + if (IS_ERR(tas_dev->regmap)) + return dev_err_probe(dev, PTR_ERR(tas_dev->regmap), + "Failed devm_regmap_init_sdw."); + + return tasdevice_init(tas_dev); +} + +static int tasdevice_sdw_remove(struct sdw_slave *peripheral) +{ + struct tasdevice_priv *tas_dev = dev_get_drvdata(&peripheral->dev); + + if (tas_dev->first_hw_init) + pm_runtime_disable(tas_dev->dev); + + return 0; +} + +static const struct sdw_device_id tasdevice_sdw_id[] = { + SDW_SLAVE_ENTRY(0x0102, 0x0000, 0), + {} +}; +MODULE_DEVICE_TABLE(sdw, tasdevice_sdw_id); + +static int __maybe_unused tas2783_sdca_dev_suspend(struct device *dev) +{ + struct tasdevice_priv *tas_priv = dev_get_drvdata(dev); + + if (!tas_priv->hw_init) + return 0; + + regcache_cache_only(tas_priv->regmap, true); + + return 0; +} + +#define TAS2783_PROBE_TIMEOUT_MS 5000 + +static int __maybe_unused tas2783_sdca_dev_resume(struct device *dev) +{ + struct sdw_slave *slave = dev_to_sdw_dev(dev); + struct tasdevice_priv *tas_priv = dev_get_drvdata(dev); + unsigned long time; + + if (!tas_priv->first_hw_init) + return 0; + + if (!slave->unattach_request) + goto regmap_sync; + + time = wait_for_completion_timeout(&slave->initialization_complete, + msecs_to_jiffies(TAS2783_PROBE_TIMEOUT_MS)); + if (!time) { + dev_err(&slave->dev, "Init not complete, timed out\n"); + sdw_show_ping_status(slave->bus, true); + + return -ETIMEDOUT; + } + +regmap_sync: + slave->unattach_request = 0; + regcache_cache_only(tas_priv->regmap, false); + regcache_sync(tas_priv->regmap); + + return 0; +} + +static const struct dev_pm_ops tas2783_sdca_pm = { + SYSTEM_SLEEP_PM_OPS(tas2783_sdca_dev_suspend, + tas2783_sdca_dev_resume) + RUNTIME_PM_OPS(tas2783_sdca_dev_suspend, + tas2783_sdca_dev_resume, NULL) +}; + +static struct sdw_driver tasdevice_sdw_driver = { + .driver = { + .name = "slave-tas2783", + .pm = &tas2783_sdca_pm, + }, + .probe = tasdevice_sdw_probe, + .remove = tasdevice_sdw_remove, + .ops = &tasdevice_sdw_ops, + .id_table = tasdevice_sdw_id, +}; + +module_sdw_driver(tasdevice_sdw_driver); + +MODULE_AUTHOR("Baojun Xu <baojun.xu(a)ti.com>"); +MODULE_AUTHOR("Shenghao Ding <shenghao-ding(a)ti.com>"); +MODULE_DESCRIPTION("ASoC TAS2783 SoundWire Driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/tas2783.h b/sound/soc/codecs/tas2783.h new file mode 100644 index 000000000000..631776dcec4c --- /dev/null +++ b/sound/soc/codecs/tas2783.h @@ -0,0 +1,106 @@ +/* SPDX-License-Identifier: GPL-2.0 + * + * ALSA SoC Texas Instruments TAS2783 Audio Smart Amplifier + * + * Copyright (C) 2023 - 2024 Texas Instruments Incorporated + * https://www.ti.com + * + * Author: Baojun Xu <baojun.xu(a)ti.com> + * Shenghao Ding <shenghao-ding(a)ti.com> + */ + +#ifndef __TAS2783_H__ +#define __TAS2783_H__ + +/* Unique id start */ +#define TAS2783_UNIQUE_ID_BASE 0x08 +#define TAS2783_MAX_DEV_NUM 8 + +#define TAS2783_DEVICE_RATES (SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000 | \ + SNDRV_PCM_RATE_96000 | \ + SNDRV_PCM_RATE_88200) + +#define TAS2783_DEVICE_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE) + +/* BOOK, PAGE Control Register */ +#define TASDEVICE_REG(book, page, reg) ((book * 256 * 256) + 0x8000 + \ + (page * 128) + reg) + +/*Software Reset */ +#define TAS2873_REG_SWRESET TASDEVICE_REG(0x0, 0x00, 0x01) +#define TAS2873_REG_SWRESET_MASK BIT(0) +#define TAS2873_REG_SWRESET_RESET BIT(0) + +#define TAS2873_REG_PWR_CTRL TASDEVICE_REG(0x0, 0x00, 0x02) +#define TAS2783_REG_AEF_MASK GENMASK(4, 3) +#define TAS2783_REG_AEF_ACTIVE 0 +#define TAS2783_REG_AEF_INACTIVE GENMASK(4, 3) +#define TAS2783_REG_PWR_MODE_MASK GENMASK(2, 0) +#define TAS2783_REG_PWR_MODE_ACTIVE 0 +#define TAS2783_REG_PWR_MODE_SW_PWD BIT(1) + +#define TAS2783_REG_TDM_RX_CFG TASDEVICE_REG(0x0, 0x00, 0x0a) + +/* Volume control */ +#define TAS2783_DVC_LVL TASDEVICE_REG(0x0, 0x00, 0x1a) +#define TAS2783_AMP_LEVEL TASDEVICE_REG(0x0, 0x00, 0x03) +#define TAS2783_AMP_LEVEL_MASK GENMASK(5, 1) + +/* Calibration data */ +#define TAS2783_CALIBRATION_RE TASDEVICE_REG(0x0, 0x17, 0x74) +#define TAS2783_CALIBRATION_RE_LOW TASDEVICE_REG(0x0, 0x18, 0x14) +#define TAS2783_CALIBRATION_INV_RE TASDEVICE_REG(0x0, 0x18, 0x0c) +#define TAS2783_CALIBRATION_POW TASDEVICE_REG(0x0, 0x13, 0x70) +#define TAS2783_CALIBRATION_TLIMIT TASDEVICE_REG(0x0, 0x18, 0x7c) + +/* TAS2783 SDCA Control - function number */ +#define TAS2783_FUNC_TYPE_SMART_AMP 0x01 + +/* TAS2783 SDCA entity */ +#define TAS2783_SDCA_ENT_FU21 0x01 +#define TAS2783_SDCA_ENT_FU23 0x02 + +/* TAS2783 SDCA control */ +#define TAS2783_SDCA_CTL_REQ_POWER_STATE 0x01 +#define TAS2783_SDCA_CTL_FU_MUTE 0x01 +#define TAS2783_SDCA_CTL_FU_VOLUME 0x02 +#define TAS2783_SDCA_CTL_UDMPU_CLUSTER 0x10 + +#define TAS2783_DEVICE_CHANNEL_LEFT 1 +#define TAS2783_DEVICE_CHANNEL_RIGHT 2 +#define TAS2783_DSPFW_FILENAME_LEN 64 + +#define TAS2783_MAX_CALIDATA_SIZE 256 + +struct tas2783_firmware_node { + unsigned int vendor_id; + unsigned int file_id; + unsigned int version_id; + unsigned int length; + unsigned int download_addr; +}; + +struct calibration_data { + unsigned long total_sz; + unsigned char data[TAS2783_MAX_CALIDATA_SIZE]; +}; + +struct tasdevice_priv { + struct snd_soc_component *component; + struct sdw_slave *sdw_peripheral; + enum sdw_slave_status status; + struct sdw_bus_params params; + struct regmap *regmap; + struct device *dev; + unsigned char dspfw_binaryname[TAS2783_DSPFW_FILENAME_LEN]; + unsigned char dev_name[32]; + unsigned int chip_id; + bool pstream; + bool hw_init; + bool first_hw_init; +}; + +#endif /*__TAS2783_H__ */ -- 2.34.1

4 6

[PATCH] ALSA: core: add kunitconfig
by Takashi Sakamoto 17 Mar '24

17 Mar '24

It is helpful to add .kunitconfig if we work with the tools provided by KUnit project. The file describes the series of kernel configurations to satisfy the dependency to build the target test. For example: $ ./tools/testing/kunit/kunit.py run --arch=arm64 --cross_compile=aarch64-linux-gnu- --kunitconfig=sound/core/ [11:35:13] Configuring KUnit Kernel ... Regenerating .config ... Populating config with: $ make ARCH=arm64 O=.kunit olddefconfig CROSS_COMPILE=aarch64-linux-gnu- [11:35:19] Building KUnit Kernel ... Populating config with: $ make ARCH=arm64 O=.kunit olddefconfig CROSS_COMPILE=aarch64-linux-gnu- Building with: $ make ARCH=arm64 O=.kunit --jobs=8 CROSS_COMPILE=aarch64-linux-gnu- [11:37:35] Starting KUnit Kernel (1/1)... [11:37:35] ============================================================ Running tests with: $ qemu-system-aarch64 -nodefaults -m 1024 -kernel .kunit/arch/arm64/boot/Image.gz -append 'kunit.enable=1 console=ttyAMA0 kunit_shutdown=reboot' -no-reboot -nographic -serial stdio -machine virt -cpu max,pauth-impdef=on [11:37:35] ============== sound-core-test (10 subtests) =============== [11:37:35] [PASSED] test_phys_format_size [11:37:35] [PASSED] test_format_width [11:37:35] [PASSED] test_format_endianness [11:37:35] [PASSED] test_format_signed [11:37:35] [PASSED] test_format_fill_silence [11:37:35] [PASSED] test_playback_avail [11:37:35] [PASSED] test_capture_avail [11:37:35] [PASSED] test_card_set_id [11:37:35] [PASSED] test_pcm_format_name [11:37:35] [PASSED] test_card_add_component [11:37:35] ================= [PASSED] sound-core-test ================= [11:37:35] ============================================================ [11:37:35] Testing complete. Ran 10 tests: passed: 10 [11:37:35] Elapsed time: 142.333s total, 5.617s configuring, 136.047s building, 0.630s running Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/core/.kunitconfig | 5 +++++ 1 file changed, 5 insertions(+) create mode 100644 sound/core/.kunitconfig diff --git a/sound/core/.kunitconfig b/sound/core/.kunitconfig new file mode 100644 index 000000000000..440f974ba0b7 --- /dev/null +++ b/sound/core/.kunitconfig @@ -0,0 +1,5 @@ +CONFIG_KUNIT=y +CONFIG_SOUND=y +CONFIG_SND=y +CONFIG_SND_PCM=y +CONFIG_SND_CORE_TEST=y -- 2.43.0

2 1

[PATCH] ALSA: hda/realtek: add in quirk for Acer Swift Go 16 - SFG16-71
by Ian Murphy 17 Mar '24

17 Mar '24

Keyboard has an LED that is ON/OFF when mic is muted/active - LED is controlled by GPIO pin - Patch enables led to appear in /sys/class/leds/ as hda::micmute - Enables LED when mic is MUTED - Disables LED when mic is active Signed-off-by: Ian Murphy <iano200(a)gmail.com> --- sound/pci/hda/patch_realtek.c | 13 +++++++++++++ 1 file changed, 13 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index a91d5ec29bfb..05c685485606 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6891,6 +6891,13 @@ static void yoga7_14arb7_fixup_i2c(struct hda_codec *cdc, comp_generic_fixup(cdc, action, "i2c", "INT8866", "-%s:00", 1); } +static void alc256_fixup_acer_sfg16_micmute_led(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + alc_fixup_hp_gpio_led(codec, action, 0, 0x04); +} + + /* for alc295_fixup_hp_top_speakers */ #include "hp_x360_helper.c" @@ -7374,6 +7381,7 @@ enum { ALC289_FIXUP_DELL_CS35L41_SPI_2, ALC294_FIXUP_CS35L41_I2C_2, ALC245_FIXUP_CS35L56_SPI_4_HP_GPIO_LED, + ALC256_FIXUP_ACER_SFG16_MICMUTE_LED, }; /* A special fixup for Lenovo C940 and Yoga Duet 7; @@ -9569,6 +9577,10 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC285_FIXUP_HP_GPIO_LED, }, + [ALC256_FIXUP_ACER_SFG16_MICMUTE_LED] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc256_fixup_acer_sfg16_micmute_led, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -9612,6 +9624,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x1430, "Acer TravelMate B311R-31", ALC256_FIXUP_ACER_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1025, 0x1466, "Acer Aspire A515-56", ALC255_FIXUP_ACER_HEADPHONE_AND_MIC), SND_PCI_QUIRK(0x1025, 0x1534, "Acer Predator PH315-54", ALC255_FIXUP_ACER_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1025, 0x169a, "Acer Swift SFG16", ALC256_FIXUP_ACER_SFG16_MICMUTE_LED), SND_PCI_QUIRK(0x1028, 0x0470, "Dell M101z", ALC269_FIXUP_DELL_M101Z), SND_PCI_QUIRK(0x1028, 0x053c, "Dell Latitude E5430", ALC292_FIXUP_DELL_E7X), SND_PCI_QUIRK(0x1028, 0x054b, "Dell XPS one 2710", ALC275_FIXUP_DELL_XPS), -- 2.44.0

2 1

chore: Use actions/checkout@v4
by GitHub pull_request - opened 16 Mar '24

16 Mar '24

alsa-project/alsa-utils pull request #259 was opened from Korilakkuma: Refer to https://github.blog/changelog/2023-09-22-github-actions-transitioning-from-…. > Node.js 16 actions are deprecated. Please update the following actions to use Node.js 20: actions/checkout@v3. Request URL : https://github.com/alsa-project/alsa-utils/pull/259 Patch URL : https://github.com/alsa-project/alsa-utils/pull/259.patch Repository URL: https://github.com/alsa-project/alsa-utils

1 0

GitHub Actions: Use actions/checkout@v4
by GitHub pull_request - opened 16 Mar '24

16 Mar '24

alsa-project/alsa-lib pull request #390 was opened from ryandesign: Fixes: #389 Request URL : https://github.com/alsa-project/alsa-lib/pull/390 Patch URL : https://github.com/alsa-project/alsa-lib/pull/390.patch Repository URL: https://github.com/alsa-project/alsa-lib

1 0

Linux 6.9 pre-rc1 might have broken UCM matching for USB devices
by GitHub issues - edited 15 Mar '24

15 Mar '24

alsa-project/alsa-ucm-conf issue #404 was edited from pinkflames: While testing Torvalds' master branch of Linux which will become 6.9-rc1 eventually, I have stumbled upon PipeWire/WirePlumber apparently no longer applying the UCM profile for my USB connected on-board audio controller. Git bisection leads to the following commit: torvalds/linux@1601cd53c7e3197181277326dbfc131d20a74e46 . It's possible that I'm wrong in concluding that the issue is UCM no longer applying/matching but that's my best guess looking at seemingly ACP named devices and the lack of any `ucm` strings in the output of pw-dump, when the regression is present. Since this is likely a kernel regression, I'll be reporting that to the [kernel bugzilla](https://bugzilla.kernel.org/show_bug.cgi?id=218605), too, but I thought alsa-ucm-conf might be interested in this change as well. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/404 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Linux 6.9 pre-rc1 might have broken UCM matching
by GitHub issues - edited 15 Mar '24

15 Mar '24

alsa-project/alsa-ucm-conf issue #404 was edited from pinkflames: While testing Torvalds' master branch of Linux which will become 6.9-rc1 eventually, I have stumbled upon PipeWire/WirePlumber apparently no longer applying the UCM profile for my USB connected on-board audio controller. Git bisection leads to the following commit: torvalds/linux@1601cd53c7e3197181277326dbfc131d20a74e46 . It's possible that I'm wrong in concluding that the issue is UCM no longer applying/matching but that's my best guess looking at seemingly ACP named devices and the lack of any `ucm` strings in the output of pw-dump, when the regression is present. Since this is likely a kernel regression, I'll be reporting that to the [kernel bugzilla](https://bugzilla.kernel.org/show_bug.cgi?id=218605), too, but I thought alsa-ucm-conf might be interested in this change as well. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/404 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Linux 6.9 pre-rc1 might have broken UCM matching
by GitHub issues - opened 15 Mar '24

15 Mar '24

alsa-project/alsa-ucm-conf issue #404 was opened from pinkflames: While testing Torvalds' master branch of Linux which will become 6.9-rc1 eventually, I have stumbled upon PipeWire/WirePlumber apparently no longer applying the UCM profile for my USB connected on-board audio controller. Git bisection leads to the following commit: torvalds/linux@1601cd53c7e3197181277326dbfc131d20a74e46 . It's possible that I'm wrong in concluding that the issue is UCM no longer applying/matching but that's my best guess looking at seemingly ACP named devices and the lack of any `ucm` strings in the output of pw-dump, when the regression is present. Since this is likely a kernel regression, I'll be reporting that to the kernel bugzilla, too, but I thought alsa-ucm-conf might be interested in this change as well. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/404 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH] ALSA: hda/realtek: Add quirk for Lenovo Yoga 9 14IMH9
by i＠jichi.ca 15 Mar '24

15 Mar '24

From: Jichi Zhang <i(a)jichi.ca> The speakers on the Lenovo Yoga 9 14IMH9 are similar to previous generations such as the 14IAP7, and the bass speakers can be fixed using similar methods with one caveat: 14IMH9 uses CS35L41 amplifiers which need to be activated seperately. Signed-off-by: Jichi Zhang <i(a)jichi.ca> --- sound/pci/hda/patch_realtek.c | 33 +++++++++++++++++++++++++++++++++ 1 file changed, 33 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index b6cd13b17..6288d837a 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7360,6 +7384,7 @@ enum { ALC287_FIXUP_LEGION_16ITHG6, ALC287_FIXUP_YOGA9_14IAP7_BASS_SPK, ALC287_FIXUP_YOGA9_14IAP7_BASS_SPK_PIN, + ALC287_FIXUP_YOGA9_14IMH9_BASS_SPK_PIN, ALC295_FIXUP_DELL_INSPIRON_TOP_SPEAKERS, ALC236_FIXUP_DELL_DUAL_CODECS, ALC287_FIXUP_CS35L41_I2C_2_THINKPAD_ACPI, @@ -9490,6 +9515,12 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC287_FIXUP_YOGA9_14IAP7_BASS_SPK, }, + [ALC287_FIXUP_YOGA9_14IMH9_BASS_SPK_PIN] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc287_fixup_yoga9_14iap7_bass_spk_pin, + .chained = true, + .chain_id = ALC287_FIXUP_CS35L41_I2C_2, + }, [ALC295_FIXUP_DELL_INSPIRON_TOP_SPEAKERS] = { .type = HDA_FIXUP_FUNC, .v.func = alc295_fixup_dell_inspiron_top_speakers, @@ -10270,6 +10301,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x38c3, "Y980 DUAL", ALC287_FIXUP_TAS2781_I2C), SND_PCI_QUIRK(0x17aa, 0x38cb, "Y790 YG DUAL", ALC287_FIXUP_TAS2781_I2C), SND_PCI_QUIRK(0x17aa, 0x38cd, "Y790 VECO DUAL", ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x17aa, 0x38d2, "Lenovo Yoga 9 14IMH9", ALC287_FIXUP_YOGA9_14IMH9_BASS_SPK_PIN), + SND_PCI_QUIRK(0x17aa, 0x38d7, "Lenovo Yoga 9 14IMH9", ALC287_FIXUP_YOGA9_14IMH9_BASS_SPK_PIN), SND_PCI_QUIRK(0x17aa, 0x3902, "Lenovo E50-80", ALC269_FIXUP_DMIC_THINKPAD_ACPI), SND_PCI_QUIRK(0x17aa, 0x3977, "IdeaPad S210", ALC283_FIXUP_INT_MIC), SND_PCI_QUIRK(0x17aa, 0x3978, "Lenovo B50-70", ALC269_FIXUP_DMIC_THINKPAD_ACPI), -- 2.44.0

2 1

ctl/hwdep/seq: optimize to gi-docgen 2023.3 or later
by GitHub pull_request - opened 15 Mar '24

15 Mar '24

alsa-project/alsa-gobject pull request #84 was opened from takaswie: The version of gi-docgen applies the restrict rule for the version expression. In the rule, the dot following to the number causes failure. The series of changes in this MR solves the failure. Request URL : https://github.com/alsa-project/alsa-gobject/pull/84 Patch URL : https://github.com/alsa-project/alsa-gobject/pull/84.patch Repository URL: https://github.com/alsa-project/alsa-gobject

1 0

Update CI to actions/checkout@v4
by GitHub issues - opened 15 Mar '24

15 Mar '24

alsa-project/alsa-lib issue #389 was opened from ryandesign: As it says in the annotations of recent CI builds: > Node.js 16 actions are deprecated. Please update the following actions to use Node.js 20: actions/checkout@v3. For more information see: https://github.blog/changelog/2023-09-22-github-actions-transitioning-from-…. In other words, switch to actions/checkout@v4. Issue URL : https://github.com/alsa-project/alsa-lib/issues/389 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH v9 00/27] Add support for QMC HDLC, framer infrastructure and PEF2256 framer
by Herve Codina 14 Mar '24

14 Mar '24

Hi, I have a system where I need to handle an HDLC interface and some audio data. The HDLC data are transferred using a TDM bus on which a PEF2256 (E1/T1 framer) is present. The PEF2256 transfers data from/to the TDM bus to/from the E1 line. This PEF2256 is connected to a PowerQUICC SoC for the control path and the TDM is connected to the SoC (QMC component) for the data path. From the QMC HDLC driver, I need to handle HDLC data using the QMC, carrier detection using the PEF2256 (E1 line carrier) and set/get some PEF2256 configuration. The QMC HDLC driver considers the PEF2256 as a generic framer. It performs operations that involve the PEF2256 through the generic framer API. The audio data are exchanged with the PEF2256 using a CPU DAI connected to the TDM bus through the QMC and the PEF2256 needs to be seen as a codec in order to be linked to the CPU DAI. The codec handles the carrier detection using the PEF2256 and reports the carrier state to the ALSA subsystem using the ASoC jack detection. The codec, even if instantiated by the PEF2256 driver, considers the PEF2256 as a generic framer. The generic framer has: - 2 consumers (QMC HDLC drv and codec) - 1 provider (PEF2256) So, the design is the following: +------------------+ +---------+ | QMC | <- TDM -> | PEF2256 | <-> E1 +---------+ | +-------------+ | | | | CPU DAI | <-data--> | QMC channel | | | | +---------+ | +-------------+ | | | +--------------+ | +-------------+ | | | | QMC HDLC drv | <-data--> | QMC channel | | | | +--------------+ | +-------------+ | | | ^ +------------------+ | | | +--------+ +-------------+ | | +-> | framer | <-> | PEF2256 drv | <- local bus ->| | | | | | +---------+ +-> | | | | | +--------+ | +-------+ | +-------------------> | codec | | | +-------+ | +-------------+ Further more, the TDM timeslots used by the QMC HDLC driver need to be configured at runtime (QMC dynamic timeslots). Several weeks ago, I sent two series related to this topic: - Add the Lantiq PEF2256 audio support [1] - RFC Add support for QMC HDLC and PHY [2] This current series is a rework of these two series taking into account feedbacks previously received. In order to implement all of this, I do the following: 1) Perform some fixes (patches 1, 2, 3, 4, 5 ) 2) Introduce the QMC HDLC driver (patches 6, 7, 8) 3) Add QMC dynamic timeslot support (patches 9 - 19) 4) Add timeslots change support in QMC HDLC (patch 20) 5) Introduce framer infrastructure (patch 21) 6) Add PEF2256 framer provider (patches 22, 23, 24, 25) 7) Add framer codec as a framer consumer (patch 26) 8) Add framer support as a framer consumer in QMC HDLC (patch 27) The series contains the full story and detailed modifications. If needed, the series can be split and/or commits can be squashed. Let me know. Compare to the previous iteration https://lore.kernel.org/linux-kernel/20231011061437.64213-1-herve.codina@bo… This v9 series: - Removes some patches already applied - Adds some 'Acked-by' tags Best regards, Hervé [1]: https://lore.kernel.org/all/20230417171601.74656-1-herve.codina@bootlin.com/ [2]: https://lore.kernel.org/all/20230323103154.264546-1-herve.codina@bootlin.co… Changes v8 -> v9 - Removed Patches 6, 7 and 8 (patches applied) - Patches 7, 20, 21, 23 (patches 10, 23, 24, 26 in v8) Add 'Acked-by: Jakub Kicinski <kuba(a)kernel.org>' Changes v7 -> v8 - Patch 10 Fix a race condition when stopping the queue in qmc_hdlc_close() - Patch 24 Move to menuconfig and hide CONFIG_GENERIC_FRAMER Remove unneeded check (defensive programming) Remove unneeded variable assignment - Patch 25 Add 'Reviewed-by: Rob Herring <robh(a)kernel.org>' - Patch 26 Use array notation for 'audio_devs' Changes v6 -> v7 - Patch 8 Move the 'compatible' property to the first property. Add device/SoC specific compatible strings. Add 'Reviewed-by: Rob Herring <robh(a)kernel.org>' - Patch 25 Remove '$ref' from the lantiq,data-rate-bps property. '-bps' is now a standard suffix. Changes v5 -> v6 - All patches Remove 'Signed-off-by: Christophe Leroy <christophe.leroy(a)csgroup.eu>' These Sob were added by Christophe when he sent the v4 while I was on vacation. https://lore.kernel.org/linux-kernel/992a2b31-e21f-eee3-8bfc-a65b69fe5bd7@c… - Patch 6, 7 Add 'Acked-by: Conor Dooley <conor.dooley(a)microchip.com>' Add 'Acked-by: Rob Herring <robh(a)kernel.org>' - Patch 8 Add a constraint on fsl,framer. - Patch 25 Fix clocks description (inconsistent capitalisation) Use '8-bit' instead of '8bit' Add 'Reviewed-by: Christophe Leroy <christophe.leroy(a)csgroup.eu>' - Removed patch 26 (patch applied) - Patch 26 (patch 27 in v5) Add 'Reviewed-by: Linus Walleij <linus.walleij(a)linaro.org>' - Patch 27 (patch 28 in v5) Move registers definition to .c file and remove .h file Add 'select PINCONF' in Kconfig Use dev_err_probe() Add 'Reviewed-by: Linus Walleij <linus.walleij(a)linaro.org>' - Patch 28 (patch 29 in v5) Remove .h file Changes v4 -> v5 - Patches 1 to 5 No changes - Patch 6 (new in v5) Fix QMC binding example - Patch 7 (new in v5) Add missing 'additionalProperties: false' - Patch 8 (new in v5, replace v4 patch 6) Add QMC HDLC properties in the QMC channel node Renamed the 'framer' property to 'fsl,framer' - Patch 9 (new in v5) Add support for QMC child devices - Patch 10 (patch 7 in v4) No changes - Patch 11 (patch 8 in v4) Remove fsl,qmc-hdlc.yaml (no more existing file) - Patches 12 to 22 (patches 9 to 19 in v4) No changes - Patch 23 (patch 20 in v4) Remove unused variable initializations Remove extra space - Patch 24 (patch 21 in v4) Improve Kconfig help text Fix variable declaration (reverse xmas tree) Fix typos and extra spaces Fix documentation issues raised by 'kernel-doc -none' Move of_node_put() and kfree() out of the mutex Replace ida_simple_{get,remove}() by ida_{alloc,free}() Support framer device-tree nodes without '#framer-cells' property - Patch 25 (patch 22 in v4) Fix $ref in the pinctrl subnode Remove '#framer-cells' property Add needed '|' - Patch 26 (patch 23 in v4) Fix a typo in the commit subject - Patches 27, 28, 29 (patch 24, 25, 26 in v4) No changes - Patch 30 (patch 27 in v4) Fix a typo in the commit log - Patch 31 (patch 28 in v4) Used 'fsl,framer' property name instead of 'framer' Changes v3 -> v4 - Patch 21 Fixes build failure with CONFIG_MODULES Changes v2 -> v3 - Patches 1, 2, 3, 4 Add 'Reviewed-by: Christophe Leroy <christophe.leroy(a)csgroup.eu>' - New patch Remove inline keyword from the existing registers accessors helpers - Patch 6 (patches 5, 27 in v2) Update the binding title Squash patch 27 - Patch 7 (patch 6 in v2) Remove the cast in netdev_to_qmc_hdlc() Add 'Reviewed-by: Christophe Leroy <christophe.leroy(a)csgroup.eu>' - Patch 8 (patch 7 in v2): No change - Patches 9, 10 (patches 8, 9 in v2) Add 'Reviewed-by: Christophe Leroy <christophe.leroy(a)csgroup.eu>' - Patch 11 (patch 10 in v2) Remove inline keyword from the introduced qmc_clrsetbits16() helper Add 'Reviewed-by: Christophe Leroy <christophe.leroy(a)csgroup.eu>' - Patches 12, 13, 14, 15, 16, 17, 18, 19, 20 Add 'Reviewed-by: Christophe Leroy <christophe.leroy(a)csgroup.eu>' - Patch 21 (patch 20 in v2) Remove unneeded framer NULL pointer check Add 'Reviewed-by: Christophe Leroy <christophe.leroy(a)csgroup.eu>' - Patch 22 (patch 21 in v2) Change sclkr and sclkx clocks description Remove the framer phandle property from the framer subnodes (ie. from framer-codec nodes) - Patch 23 (patch 22 in v2) Initialize 'disabled' variable at declaration Fix commit log Add 'Reviewed-by: Christophe Leroy <christophe.leroy(a)csgroup.eu>' - Patch 24 (patch 23 in v2) Remove inline keyword from the existing registers accessors helpers Use dev_warn_ratelimited() in default interrupt handler Add 'Reviewed-by: Christophe Leroy <christophe.leroy(a)csgroup.eu>' - Patch 25 (patch 24 in v2) Replace #include "linux/bitfield.h" by #include <linux/bitfield.h> Fold the pinctrl anonymous struct into the struct pef2256_pinctrl Update commit log Add 'Reviewed-by: Christophe Leroy <christophe.leroy(a)csgroup.eu>' - Patch 26 (patch 25 in v2) Add 'Reviewed-by: Christophe Leroy <christophe.leroy(a)csgroup.eu>' - Patch 27 (patch 26 in v2) Fix error message Changed the ch.max computation in framer_dai_hw_rule_channels_by_format() Add 'Reviewed-by: Christophe Leroy <christophe.leroy(a)csgroup.eu>' - Patch 28 Add 'Reviewed-by: Christophe Leroy <christophe.leroy(a)csgroup.eu>' Changes v1 -> v2 - Patches 1, 2 (New in v2) Fix __iomem addresses declaration - Patch 19 (17 in v1) Fix a compilation warning - Patch 26 (24 in v1) Fix a typo in Kconfig file Fix issues raised by sparse (make C=1) Herve Codina (27): soc: fsl: cpm1: tsa: Fix __iomem addresses declaration soc: fsl: cpm1: qmc: Fix __iomem addresses declaration soc: fsl: cpm1: qmc: Fix rx channel reset soc: fsl: cpm1: qmc: Extend the API to provide Rx status soc: fsl: cpm1: qmc: Remove inline function specifiers soc: fsl: cpm1: qmc: Add support for child devices net: wan: Add support for QMC HDLC MAINTAINERS: Add the Freescale QMC HDLC driver entry soc: fsl: cpm1: qmc: Introduce available timeslots masks soc: fsl: cpm1: qmc: Rename qmc_setup_tsa* to qmc_init_tsa* soc: fsl: cpm1: qmc: Introduce qmc_chan_setup_tsa* soc: fsl: cpm1: qmc: Remove no more needed checks from qmc_check_chans() soc: fsl: cpm1: qmc: Check available timeslots in qmc_check_chans() soc: fsl: cpm1: qmc: Add support for disabling channel TSA entries soc: fsl: cpm1: qmc: Split Tx and Rx TSA entries setup soc: fsl: cpm1: qmc: Introduce is_tsa_64rxtx flag soc: fsl: cpm1: qmc: Handle timeslot entries at channel start() and stop() soc: fsl: cpm1: qmc: Remove timeslots handling from setup_chan() soc: fsl: cpm1: qmc: Introduce functions to change timeslots at runtime wan: qmc_hdlc: Add runtime timeslots changes support net: wan: Add framer framework support dt-bindings: net: Add the Lantiq PEF2256 E1/T1/J1 framer net: wan: framer: Add support for the Lantiq PEF2256 framer pinctrl: Add support for the Lantic PEF2256 pinmux MAINTAINERS: Add the Lantiq PEF2256 driver entry ASoC: codecs: Add support for the framer codec net: wan: fsl_qmc_hdlc: Add framer support .../bindings/net/lantiq,pef2256.yaml | 213 +++++ MAINTAINERS | 15 + drivers/net/wan/Kconfig | 14 + drivers/net/wan/Makefile | 3 + drivers/net/wan/framer/Kconfig | 41 + drivers/net/wan/framer/Makefile | 7 + drivers/net/wan/framer/framer-core.c | 882 ++++++++++++++++++ drivers/net/wan/framer/pef2256/Makefile | 8 + drivers/net/wan/framer/pef2256/pef2256-regs.h | 250 +++++ drivers/net/wan/framer/pef2256/pef2256.c | 880 +++++++++++++++++ drivers/net/wan/fsl_qmc_hdlc.c | 820 ++++++++++++++++ drivers/pinctrl/Kconfig | 15 + drivers/pinctrl/Makefile | 1 + drivers/pinctrl/pinctrl-pef2256.c | 358 +++++++ drivers/soc/fsl/qe/qmc.c | 592 +++++++++--- drivers/soc/fsl/qe/tsa.c | 22 +- include/linux/framer/framer-provider.h | 194 ++++ include/linux/framer/framer.h | 205 ++++ include/linux/framer/pef2256.h | 31 + include/soc/fsl/qe/qmc.h | 27 +- sound/soc/codecs/Kconfig | 15 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/framer-codec.c | 413 ++++++++ sound/soc/fsl/fsl_qmc_audio.c | 2 +- 24 files changed, 4873 insertions(+), 137 deletions(-) create mode 100644 Documentation/devicetree/bindings/net/lantiq,pef2256.yaml create mode 100644 drivers/net/wan/framer/Kconfig create mode 100644 drivers/net/wan/framer/Makefile create mode 100644 drivers/net/wan/framer/framer-core.c create mode 100644 drivers/net/wan/framer/pef2256/Makefile create mode 100644 drivers/net/wan/framer/pef2256/pef2256-regs.h create mode 100644 drivers/net/wan/framer/pef2256/pef2256.c create mode 100644 drivers/net/wan/fsl_qmc_hdlc.c create mode 100644 drivers/pinctrl/pinctrl-pef2256.c create mode 100644 include/linux/framer/framer-provider.h create mode 100644 include/linux/framer/framer.h create mode 100644 include/linux/framer/pef2256.h create mode 100644 sound/soc/codecs/framer-codec.c -- 2.41.0

6 36

[PATCH v7 0/5] ASoC: makes CPU/Codec channel connection map more generic
by Kuninori Morimoto 14 Mar '24

14 Mar '24

Hi Mark Cc Bard, Pierre-Louis, Jerome, DT-ML This is v7 patch-set. Current ASoC is supporting CPU/Codec = N:M (N < M) connection by using ch_map idea. This patch-set expands it that all connection uses this idea, and no longer N < M limit [1][2]. Link: https://lore.kernel.org/r/87fs6wuszr.wl-kuninori.morimoto.gx@renesas.com [1] Link: https://lore.kernel.org/r/878r7yqeo4.wl-kuninori.morimoto.gx@renesas.com [2] ASoC core code ([PATCH 1/5]) is same as v6 and it was tested by Pierre-Louis, and Jerome. Big change on v7 is basically for Audio-Graph-Card2. v6 -> v7 - use "endpoint" for N:M connection instead of ch-map-idx on Audio-Graph-Card2 v5 -> v6 - tidyup some warnings - rename "ch-map-idx" -> "channel-map-index" - Update "channel-map-index" description - add Tested-by from Pierre-Louis / Jerome v4 -> v5 - use cpu/codec index on ch_maps - separate card2 sample DT patch into prepare and new feature - ch-maps -> ch-map-idx on DT v3 -> v4 - add Jerome on To - add "description" on "ch-maps" v2 -> v3 - tidyup comment - use more clear connection image on DT - "ch_maps" -> "ch-maps" on DT - Add DT maintainer on "To:" for all patches v1 -> v2 - makes CPU/Codec connection relation clear on comment/sample - fixup type "connction" -> "connection" - makes error message clear Kuninori Morimoto (5): ASoC: makes CPU/Codec channel connection map more generic ASoC: audio-graph-card2: use better image for Multi connection ASoC: audio-graph-card2: add CPU:Codec = N:M support ASoC: audio-graph-card2-custom-sample: Add connection image ASoC: audio-graph-card2-custom-sample: add CPU/Codec = N:M sample include/sound/soc.h | 56 ++- .../audio-graph-card2-custom-sample.dtsi | 380 ++++++++++++++++-- sound/soc/generic/audio-graph-card2.c | 277 ++++++++++--- sound/soc/intel/boards/sof_sdw.c | 28 +- sound/soc/soc-core.c | 95 ++++- sound/soc/soc-dapm.c | 45 +-- sound/soc/soc-pcm.c | 44 +- 7 files changed, 754 insertions(+), 171 deletions(-) -- 2.25.1

3 15

[PATCH v3] powerpc: macio: Make remove callback of macio driver void returned
by Dawei Li 13 Mar '24

13 Mar '24

Commit fc7a6209d571 ("bus: Make remove callback return void") forces bus_type::remove be void-returned, it doesn't make much sense for any bus based driver implementing remove callbalk to return non-void to its caller. This change is for macio bus based drivers. Signed-off-by: Dawei Li <set_pte_at(a)outlook.com> --- v2 -> v3 - Rebased on latest powerpc/next. - cc' to relevant subsysem lists. v1 -> v2 - Revert unneeded changes. - Rebased on latest powerpc/next. v1 - https://lore.kernel.org/all/TYCP286MB2323FCDC7ECD87F8D97CB74BCA189@TYCP286M… --- arch/powerpc/include/asm/macio.h | 2 +- drivers/ata/pata_macio.c | 4 +--- drivers/macintosh/rack-meter.c | 4 +--- drivers/net/ethernet/apple/bmac.c | 4 +--- drivers/net/ethernet/apple/mace.c | 4 +--- drivers/net/wireless/intersil/orinoco/airport.c | 4 +--- drivers/scsi/mac53c94.c | 5 +---- drivers/scsi/mesh.c | 5 +---- drivers/tty/serial/pmac_zilog.c | 7 ++----- sound/aoa/soundbus/i2sbus/core.c | 4 +--- 10 files changed, 11 insertions(+), 32 deletions(-) diff --git a/arch/powerpc/include/asm/macio.h b/arch/powerpc/include/asm/macio.h index ff5fd82d9ff0..cb9c386dacf8 100644 --- a/arch/powerpc/include/asm/macio.h +++ b/arch/powerpc/include/asm/macio.h @@ -125,7 +125,7 @@ static inline struct pci_dev *macio_get_pci_dev(struct macio_dev *mdev) struct macio_driver { int (*probe)(struct macio_dev* dev, const struct of_device_id *match); - int (*remove)(struct macio_dev* dev); + void (*remove)(struct macio_dev *dev); int (*suspend)(struct macio_dev* dev, pm_message_t state); int (*resume)(struct macio_dev* dev); diff --git a/drivers/ata/pata_macio.c b/drivers/ata/pata_macio.c index 9ccaac9e2bc3..653106716a4b 100644 --- a/drivers/ata/pata_macio.c +++ b/drivers/ata/pata_macio.c @@ -1187,7 +1187,7 @@ static int pata_macio_attach(struct macio_dev *mdev, return rc; } -static int pata_macio_detach(struct macio_dev *mdev) +static void pata_macio_detach(struct macio_dev *mdev) { struct ata_host *host = macio_get_drvdata(mdev); struct pata_macio_priv *priv = host->private_data; @@ -1202,8 +1202,6 @@ static int pata_macio_detach(struct macio_dev *mdev) ata_host_detach(host); unlock_media_bay(priv->mdev->media_bay); - - return 0; } #ifdef CONFIG_PM_SLEEP diff --git a/drivers/macintosh/rack-meter.c b/drivers/macintosh/rack-meter.c index c28893e41a8b..f2f83c4f3af5 100644 --- a/drivers/macintosh/rack-meter.c +++ b/drivers/macintosh/rack-meter.c @@ -523,7 +523,7 @@ static int rackmeter_probe(struct macio_dev* mdev, return rc; } -static int rackmeter_remove(struct macio_dev* mdev) +static void rackmeter_remove(struct macio_dev *mdev) { struct rackmeter *rm = dev_get_drvdata(&mdev->ofdev.dev); @@ -558,8 +558,6 @@ static int rackmeter_remove(struct macio_dev* mdev) /* Get rid of me */ kfree(rm); - - return 0; } static int rackmeter_shutdown(struct macio_dev* mdev) diff --git a/drivers/net/ethernet/apple/bmac.c b/drivers/net/ethernet/apple/bmac.c index 9e653e2925f7..292b1f9cd9e7 100644 --- a/drivers/net/ethernet/apple/bmac.c +++ b/drivers/net/ethernet/apple/bmac.c @@ -1591,7 +1591,7 @@ bmac_proc_info(char *buffer, char **start, off_t offset, int length) } #endif -static int bmac_remove(struct macio_dev *mdev) +static void bmac_remove(struct macio_dev *mdev) { struct net_device *dev = macio_get_drvdata(mdev); struct bmac_data *bp = netdev_priv(dev); @@ -1609,8 +1609,6 @@ static int bmac_remove(struct macio_dev *mdev) macio_release_resources(mdev); free_netdev(dev); - - return 0; } static const struct of_device_id bmac_match[] = diff --git a/drivers/net/ethernet/apple/mace.c b/drivers/net/ethernet/apple/mace.c index fd1b008b7208..e6350971c707 100644 --- a/drivers/net/ethernet/apple/mace.c +++ b/drivers/net/ethernet/apple/mace.c @@ -272,7 +272,7 @@ static int mace_probe(struct macio_dev *mdev, const struct of_device_id *match) return rc; } -static int mace_remove(struct macio_dev *mdev) +static void mace_remove(struct macio_dev *mdev) { struct net_device *dev = macio_get_drvdata(mdev); struct mace_data *mp; @@ -296,8 +296,6 @@ static int mace_remove(struct macio_dev *mdev) free_netdev(dev); macio_release_resources(mdev); - - return 0; } static void dbdma_reset(volatile struct dbdma_regs __iomem *dma) diff --git a/drivers/net/wireless/intersil/orinoco/airport.c b/drivers/net/wireless/intersil/orinoco/airport.c index a890bfa0d5cc..276a06cdd1f5 100644 --- a/drivers/net/wireless/intersil/orinoco/airport.c +++ b/drivers/net/wireless/intersil/orinoco/airport.c @@ -85,7 +85,7 @@ airport_resume(struct macio_dev *mdev) return err; } -static int +static void airport_detach(struct macio_dev *mdev) { struct orinoco_private *priv = dev_get_drvdata(&mdev->ofdev.dev); @@ -111,8 +111,6 @@ airport_detach(struct macio_dev *mdev) macio_set_drvdata(mdev, NULL); free_orinocodev(priv); - - return 0; } static int airport_hard_reset(struct orinoco_private *priv) diff --git a/drivers/scsi/mac53c94.c b/drivers/scsi/mac53c94.c index f75928f7773e..42648ca9b8ed 100644 --- a/drivers/scsi/mac53c94.c +++ b/drivers/scsi/mac53c94.c @@ -508,7 +508,7 @@ static int mac53c94_probe(struct macio_dev *mdev, const struct of_device_id *mat return rc; } -static int mac53c94_remove(struct macio_dev *mdev) +static void mac53c94_remove(struct macio_dev *mdev) { struct fsc_state *fp = (struct fsc_state *)macio_get_drvdata(mdev); struct Scsi_Host *host = fp->host; @@ -526,11 +526,8 @@ static int mac53c94_remove(struct macio_dev *mdev) scsi_host_put(host); macio_release_resources(mdev); - - return 0; } - static struct of_device_id mac53c94_match[] = { { diff --git a/drivers/scsi/mesh.c b/drivers/scsi/mesh.c index 84b541a57b7b..cd2575b88c85 100644 --- a/drivers/scsi/mesh.c +++ b/drivers/scsi/mesh.c @@ -1986,7 +1986,7 @@ static int mesh_probe(struct macio_dev *mdev, const struct of_device_id *match) return -ENODEV; } -static int mesh_remove(struct macio_dev *mdev) +static void mesh_remove(struct macio_dev *mdev) { struct mesh_state *ms = (struct mesh_state *)macio_get_drvdata(mdev); struct Scsi_Host *mesh_host = ms->host; @@ -2013,11 +2013,8 @@ static int mesh_remove(struct macio_dev *mdev) macio_release_resources(mdev); scsi_host_put(mesh_host); - - return 0; } - static struct of_device_id mesh_match[] = { { diff --git a/drivers/tty/serial/pmac_zilog.c b/drivers/tty/serial/pmac_zilog.c index 13668ffdb1e7..d4640479c338 100644 --- a/drivers/tty/serial/pmac_zilog.c +++ b/drivers/tty/serial/pmac_zilog.c @@ -1507,12 +1507,12 @@ static int pmz_attach(struct macio_dev *mdev, const struct of_device_id *match) * That one should not be called, macio isn't really a hotswap device, * we don't expect one of those serial ports to go away... */ -static int pmz_detach(struct macio_dev *mdev) +static void pmz_detach(struct macio_dev *mdev) { struct uart_pmac_port *uap = dev_get_drvdata(&mdev->ofdev.dev); if (!uap) - return -ENODEV; + return; uart_remove_one_port(&pmz_uart_reg, &uap->port); @@ -1523,11 +1523,8 @@ static int pmz_detach(struct macio_dev *mdev) dev_set_drvdata(&mdev->ofdev.dev, NULL); uap->dev = NULL; uap->port.dev = NULL; - - return 0; } - static int pmz_suspend(struct macio_dev *mdev, pm_message_t pm_state) { struct uart_pmac_port *uap = dev_get_drvdata(&mdev->ofdev.dev); diff --git a/sound/aoa/soundbus/i2sbus/core.c b/sound/aoa/soundbus/i2sbus/core.c index 51ed2f34b276..35f39727994d 100644 --- a/sound/aoa/soundbus/i2sbus/core.c +++ b/sound/aoa/soundbus/i2sbus/core.c @@ -364,15 +364,13 @@ static int i2sbus_probe(struct macio_dev* dev, const struct of_device_id *match) return 0; } -static int i2sbus_remove(struct macio_dev* dev) +static void i2sbus_remove(struct macio_dev *dev) { struct i2sbus_control *control = dev_get_drvdata(&dev->ofdev.dev); struct i2sbus_dev *i2sdev, *tmp; list_for_each_entry_safe(i2sdev, tmp, &control->list, item) soundbus_remove_one(&i2sdev->sound); - - return 0; } #ifdef CONFIG_PM -- 2.25.1

6 6

[PATCH V4 00/13] soundwire/SOF: add SoundWire Interface support for AMD SOF stack
by Vijendar Mukunda 13 Mar '24

13 Mar '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 platform). Below patch series is reviewed in https://github.com/thesofproject/linux/pull/4699 Changes since V1: - add new commit to update license for amd soundwire manager driver files. - wrap the commit message with in 75 characters. - add amd tag in commit title for soundwire patches Changes since V2: - drop "drivers" tag from commit title for SoundWire patches. changes since v3: - update copyright year - add missing blank line in sdw_amd.h file This patch series has dependencies. "soundwire: amd: refactor amd soundwire manager device node creation" patch has dependency on below patch. "ASoC/soundwire: implement generic api for scanning amd soundwire controller" Similarly SOF patches has dependency on Soundwire tree patches. This patch series has to be reviewed and merged in one go. Vijendar Mukunda (13): ASoC/soundwire: implement generic api for scanning amd soundwire controller soundwire: amd: update license soundwire: amd: refactor amd soundwire manager device node creation soundwire: amd: implement function to extract slave information soundwire: amd: refactor soundwire pads enable soundwire: amd: 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 | 47 ++---- drivers/soundwire/amd_manager.h | 16 +- include/linux/soundwire/sdw_amd.h | 83 ++++++++-- 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, 805 insertions(+), 74 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

4 25

alsa-utils-1.2.11 does validate against signature
by GitHub issues - opened 13 Mar '24

13 Mar '24

alsa-project/alsa-utils issue #258 was opened from egtvedt: Hello, downloading alsa-utils-1.2.11.tar.bz2 from https://www.alsa-project.org/wiki/Download gives me a file with sha512sum 5ce76807b53357584bfb4ace5acfdac4db9168ffaf5cdd1e499738eec046c36112bf84a99970f66368063a9baf73bad93af2d439630572f3eba5c9321071172d However, validating this archive against the GPG signing key with fingerprint F04D F507 37AC 1A88 4C4B 3D71 8380 596D A6E5 9C91 does not result in success. Signature file also downloaded from https://www.alsa-project.org/wiki/Download Doing the above steps for alsa-lib-1.2.11.tar.bz2 archive is successful. I looked at the diff between alsa-utils-1.2.11.tar.bz2 and git tag, and couldn't spot anything suspicious. Could a maintainer please verify alsa-utils is properly signed, if not, refresh the signature file to match released binary? Issue URL : https://github.com/alsa-project/alsa-utils/issues/258 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[PATCH] ALSA: hda/tas2781: remove unnecessary runtime_pm calls
by Pierre-Louis Bossart 12 Mar '24

12 Mar '24

The runtime_pm handling seems to have been loosely inspired by the cs32l41 driver, but in this case the get_noresume/put sequence is not required. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/pci/hda/tas2781_hda_i2c.c | 3 --- 1 file changed, 3 deletions(-) diff --git a/sound/pci/hda/tas2781_hda_i2c.c b/sound/pci/hda/tas2781_hda_i2c.c index 2eb1f9e443c0..4475cea8e9f7 100644 --- a/sound/pci/hda/tas2781_hda_i2c.c +++ b/sound/pci/hda/tas2781_hda_i2c.c @@ -796,11 +796,8 @@ static int tas2781_hda_i2c_probe(struct i2c_client *clt) pm_runtime_use_autosuspend(tas_hda->dev); pm_runtime_mark_last_busy(tas_hda->dev); pm_runtime_set_active(tas_hda->dev); - pm_runtime_get_noresume(tas_hda->dev); pm_runtime_enable(tas_hda->dev); - pm_runtime_put_autosuspend(tas_hda->dev); - tas2781_reset(tas_hda->priv); ret = component_add(tas_hda->dev, &tas2781_hda_comp_ops); -- 2.40.1

2 1

[PATCH] ASoC: dt-bindings: cirrus,cs42l43: Fix 'gpio-ranges' schema
by Rob Herring 12 Mar '24

12 Mar '24

'gpio-ranges' is a phandle-array which is really a matrix. The schema in cirrus,cs42l43 is incomplete as it doesn't define there's only a single entry. Add the outer array constraints that there is a single entry. Signed-off-by: Rob Herring <robh(a)kernel.org> --- .../devicetree/bindings/sound/cirrus,cs42l43.yaml | 11 ++++++----- 1 file changed, 6 insertions(+), 5 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/cirrus,cs42l43.yaml b/Documentation/devicetree/bindings/sound/cirrus,cs42l43.yaml index 7f9d8c7a635a..99a536601cc7 100644 --- a/Documentation/devicetree/bindings/sound/cirrus,cs42l43.yaml +++ b/Documentation/devicetree/bindings/sound/cirrus,cs42l43.yaml @@ -185,11 +185,12 @@ properties: gpio-ranges: items: - - description: A phandle to the CODEC pinctrl node - minimum: 0 - - const: 0 - - const: 0 - - const: 3 + - items: + - description: A phandle to the CODEC pinctrl node + minimum: 0 + - const: 0 + - const: 0 + - const: 3 patternProperties: "-state$": -- 2.43.0

2 1

Added new utility aseqsend
by GitHub pull_request - opened 12 Mar '24

12 Mar '24

alsa-project/alsa-utils pull request #257 was opened from mj15003: New utility **aseqsend** enables to send MIDI messages encoded as a string of hexadecimal characters to specified ALSA seqencer port. Similar functionality is already available with amidi tool, however only for raw midi ports. **Aseqsend** command has been developed mainly to conveniently test and debug another MIDI software. I think it might be useful for others to be included with general alsa-utils package. Request URL : https://github.com/alsa-project/alsa-utils/pull/257 Patch URL : https://github.com/alsa-project/alsa-utils/pull/257.patch Repository URL: https://github.com/alsa-project/alsa-utils

1 0

Focusrite USB UCM2 profile forcing 48kHz
by GitHub issues - opened 12 Mar '24

12 Mar '24

alsa-project/alsa-lib issue #388 was opened from geoffreybennett: I'm creating UCM2 profiles for the Focusrite USB interfaces. They support: ``` $ cat /proc/asound/Gen/stream0 Focusrite Scarlett 4i4 4th Gen at usb-0000:00:14.0-6.4.1, high speed : USB Audio Playback: Status: Running Interface = 1 Altset = 1 Packet Size = 216 Momentary freq = 48000 Hz (0x6.0000) Interface 1 Altset 1 Format: S32_LE Channels: 6 Endpoint: 0x01 (1 OUT) (SYNC) Rates: 44100, 48000, 88200, 96000, 176400, 192000 Data packet interval: 125 us Bits: 24 Channel map: FL FR FC LFE RL RR Capture: Status: Running Interface = 2 Altset = 1 Packet Size = 216 Momentary freq = 48000 Hz (0x6.0000) Interface 2 Altset 1 Format: S32_LE Channels: 6 Endpoint: 0x81 (1 IN) (SYNC) Rates: 44100, 48000, 88200, 96000, 176400, 192000 Data packet interval: 125 us Bits: 24 Channel map: FL FR FC LFE RL RR ``` When I use my UCM2 profile that splits the inputs and outputs up, PipeWire will only select 48kHz. The Pro Audio and Direct profiles can use all the sample rates. To me it sounded like https://github.com/alsa-project/alsa-ucm-conf/issues/313 but @perexg advised https://github.com/alsa-project/alsa-ucm-conf/issues/313#issuecomment-19893…: > For configurations using split macros, the default rate is hardcoded to 48kHz in the alsa-lib direct plugins. It would be better to create a bug in alsa-lib issue tracker. Issue URL : https://github.com/alsa-project/alsa-lib/issues/388 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH v3 0/2] ASoC: nau8325: Modify driver code and dtschema.
by Seven Lee 12 Mar '24

12 Mar '24

Modify properties of "nuvoton,dac-vref" in dtschema. For driver code, changed dev_err() to dev_dbg(). Change: V1 -> V2: - Revise the driver description part for the C++ comment. - In the nau8325_clkdet_put function, modify the max variable to hard code. - Removed "Clock Detection" switch control. - modify the "ALC Enable" switch name. - Revise the dtschema for "nuvoton,dac-vref". V2 -> V3: - Properties use standard unit suffixes. - Modify the enum definition. - Driver code should be used dev_dbg(). Seven Lee (2): ASoC: dt-bindings: Added schema for "nuvoton,nau8325" ASoC: nau8325: new driver .../bindings/sound/nuvoton,nau8325.yaml | 74 ++ sound/soc/codecs/nau8325.c | 856 ++++++++++++++++++ sound/soc/codecs/nau8325.h | 391 ++++++++ 3 files changed, 1321 insertions(+) create mode 100755 Documentation/devicetree/bindings/sound/nuvoton,nau8325.yaml create mode 100755 sound/soc/codecs/nau8325.c create mode 100644 sound/soc/codecs/nau8325.h -- 2.25.1

3 6

UCM2 for HP Chromebook X2 11" (sc7180-adau7002-max98357a)
by GitHub pull_request - opened 11 Mar '24

11 Mar '24

alsa-project/alsa-ucm-conf pull request #403 was opened from pulak1901: Hi, I've been trying to get sound working on this tablet. Since I'm quite new to alsa UCM & PA/PW, I took a look at the existing sc7180-rt5682-max98357a config as well as the [ChromeOS UCM for this system](https://chromium.googlesource.com/chromiumos/overlays/board-overlay…. It looks like the hardware does not have any hw controls (`alsamixer -c 0` shows no bars), so it was easy to get internal speakers & microphone working (albeit without any gain for the mic). However, I could not get the USB HDMI/DP output to work seamlessly. Even though I've mentioned the JackControl, PA tries to open the device, errors out, and fails to find a working profile. It only works if I reboot the system with HDMI plugged in. Jack detection itself seems to be fine(from the output of `amixer -c 0 contents`), so is there something else I should be doing here? <details><summary>amixer -c 0 contents</summary> <p> ``` numid=1,iface=CARD,name='HDMI Jack' ; type=BOOLEAN,access=r-------,values=1 : values=off numid=5,iface=PCM,name='ELD',device=2 ; type=BYTES,access=r--v----,values=128 : values=0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00 numid=4,iface=PCM,name='IEC958 Playback Default',device=2 ; type=IEC958,access=rw------,values=1 : values=[AES0=0x04 AES1=0x00 AES2=0x00 AES3=0x01] numid=3,iface=PCM,name='IEC958 Playback Mask',device=2 ; type=IEC958,access=r-------,values=1 : values=[AES0=0xff AES1=0xff AES2=0xff AES3=0xff] numid=2,iface=PCM,name='Playback Channel Map',device=2 ; type=INTEGER,access=r----R--,values=8,min=0,max=36,step=0 : values=0,0,0,0,0,0,0,0 | container | chmap-fixed=FL,FR ``` </p> </details> <details><summary>pulseaudio -vvv</summary> <p> ``` ... D: [pulseaudio] module-udev-detect.c: /dev/snd/controlC0 is accessible: yes D: [pulseaudio] module-udev-detect.c: /devices/platform/sound/sound/card0 is busy: no D: [pulseaudio] module-udev-detect.c: Loading module-alsa-card with arguments 'device_id="0" name="platform-sound" card_name="alsa_card.platform-sound" namereg_fail=false tsched=yes fixed_latency_range=no ignore_dB=no deferred_volume=yes use_ucm=yes avoid_resampling=no card_properties="module-udev-detect.discovered=1"' D: [pulseaudio] reserve-wrap.c: Unable to contact D-Bus session bus: org.freedesktop.DBus.Error.Spawn.ExecFailed: /usr/bin/dbus-launch terminated abnormally without any error message I: [pulseaudio] alsa-ucm.c: UCM available for card hw:0 D: [pulseaudio] alsa-ucm.c: UCM _alibpref=_ucm0006. I: [pulseaudio] alsa-ucm.c: Set UCM verb to HiFi D: [pulseaudio] alsa-ucm.c: Got PlaybackPCM for device HDMI: _ucm0006.hw:sc7180adau7002m,2 D: [pulseaudio] alsa-ucm.c: Got PlaybackCTL for device HDMI: _ucm0006.hw:sc7180adau7002m D: [pulseaudio] alsa-ucm.c: Got CaptureCTL for device HDMI: _ucm0006.hw:sc7180adau7002m D: [pulseaudio] alsa-ucm.c: Got JackControl for device HDMI: HDMI Jack I: [pulseaudio] alsa-ucm.c: UCM file does not specify 'PlaybackChannels' for device HDMI, assuming stereo. D: [pulseaudio] alsa-ucm.c: No _supporteddevs for device HDMI D: [pulseaudio] alsa-ucm.c: Got CapturePCM for device Mic: _ucm0006.hw:sc7180adau7002m,0 D: [pulseaudio] alsa-ucm.c: Got PlaybackCTL for device Mic: _ucm0006.hw:sc7180adau7002m D: [pulseaudio] alsa-ucm.c: Got CaptureCTL for device Mic: _ucm0006.hw:sc7180adau7002m D: [pulseaudio] alsa-ucm.c: Got CapturePriority for device Mic: 100 I: [pulseaudio] alsa-ucm.c: UCM file does not specify 'CaptureChannels' for device Mic, assuming stereo. D: [pulseaudio] alsa-ucm.c: No _conflictingdevs for device Mic D: [pulseaudio] alsa-ucm.c: No _supporteddevs for device Mic D: [pulseaudio] alsa-ucm.c: Got PlaybackPCM for device Speaker: _ucm0006.hw:sc7180adau7002m,1 D: [pulseaudio] alsa-ucm.c: Got PlaybackCTL for device Speaker: _ucm0006.hw:sc7180adau7002m D: [pulseaudio] alsa-ucm.c: Got PlaybackPriority for device Speaker: 100 D: [pulseaudio] alsa-ucm.c: Got PlaybackRate for device Speaker: 48000 D: [pulseaudio] alsa-ucm.c: Got PlaybackChannels for device Speaker: 2 D: [pulseaudio] alsa-ucm.c: Got CaptureCTL for device Speaker: _ucm0006.hw:sc7180adau7002m D: [pulseaudio] alsa-ucm.c: UCM playback device Speaker rate 48000 D: [pulseaudio] alsa-ucm.c: No _supporteddevs for device Speaker I: [pulseaudio] module-alsa-card.c: Found UCM profiles D: [pulseaudio] alsa-ucm.c: UCM mapping: HiFi: hw:sc7180adau7002m,2: sink dev HDMI D: [pulseaudio] alsa-ucm.c: UCM mapping: HiFi: hw:sc7180adau7002m,0: source dev Mic D: [pulseaudio] alsa-ucm.c: UCM mapping: HiFi: hw:sc7180adau7002m,1: sink dev Speaker D: [pulseaudio] alsa-mixer.c: Profile HiFi (Default), input=(null), output=(null) priority=8000, supported=yes n_input_mappings=1, n_output_mappings=2 D: [pulseaudio] alsa-mixer.c: Input HiFi: hw:sc7180adau7002m,0: source D: [pulseaudio] alsa-mixer.c: Output HiFi: hw:sc7180adau7002m,2: sink D: [pulseaudio] alsa-mixer.c: Output HiFi: hw:sc7180adau7002m,1: sink I: [pulseaudio] alsa-ucm.c: Set ucm verb to HiFi D: [pulseaudio] alsa-util.c: Trying _ucm0006.hw:sc7180adau7002m,2 with SND_PCM_NO_AUTO_FORMAT ... D: [pulseaudio] alsa-util.c: Managed to open _ucm0006.hw:sc7180adau7002m,2 D: [pulseaudio] alsa-util.c: snd_pcm_hw_params_set_format(Signed 16 bit Little Endian) failed: Invalid argument D: [pulseaudio] alsa-util.c: snd_pcm_hw_params_set_format(Signed 16 bit Big Endian) failed: Invalid argument D: [pulseaudio] alsa-util.c: snd_pcm_hw_params_set_format(Float 32 bit Little Endian) failed: Invalid argument D: [pulseaudio] alsa-util.c: snd_pcm_hw_params_set_format(Float 32 bit Big Endian) failed: Invalid argument D: [pulseaudio] alsa-util.c: snd_pcm_hw_params_set_format(Signed 32 bit Little Endian) failed: Invalid argument D: [pulseaudio] alsa-util.c: snd_pcm_hw_params_set_format(Signed 32 bit Big Endian) failed: Invalid argument D: [pulseaudio] alsa-util.c: Maximum hw buffer size is 128 ms I: [pulseaudio] (alsa-lib)pcm_hw.c: SNDRV_PCM_IOCTL_HW_PARAMS failed (-22) I: [pulseaudio] (alsa-lib)pcm_hw.c: SNDRV_PCM_IOCTL_HW_PARAMS failed (-22) I: [pulseaudio] (alsa-lib)pcm_hw.c: SNDRV_PCM_IOCTL_HW_PARAMS failed (-22) I: [pulseaudio] (alsa-lib)pcm_hw.c: SNDRV_PCM_IOCTL_HW_PARAMS failed (-22) D: [pulseaudio] alsa-util.c: Set neither period nor buffer size. I: [pulseaudio] (alsa-lib)pcm_hw.c: SNDRV_PCM_IOCTL_HW_PARAMS failed (-22) I: [pulseaudio] alsa-util.c: snd_pcm_hw_params failed: Invalid argument D: [pulseaudio] alsa-util.c: Trying _ucm0006.hw:sc7180adau7002m,2 without SND_PCM_NO_AUTO_FORMAT ... D: [pulseaudio] alsa-util.c: Managed to open _ucm0006.hw:sc7180adau7002m,2 D: [pulseaudio] alsa-util.c: snd_pcm_hw_params_set_format(Signed 16 bit Little Endian) failed: Invalid argument D: [pulseaudio] alsa-util.c: snd_pcm_hw_params_set_format(Signed 16 bit Big Endian) failed: Invalid argument D: [pulseaudio] alsa-util.c: snd_pcm_hw_params_set_format(Float 32 bit Little Endian) failed: Invalid argument D: [pulseaudio] alsa-util.c: snd_pcm_hw_params_set_format(Float 32 bit Big Endian) failed: Invalid argument D: [pulseaudio] alsa-util.c: snd_pcm_hw_params_set_format(Signed 32 bit Little Endian) failed: Invalid argument D: [pulseaudio] alsa-util.c: snd_pcm_hw_params_set_format(Signed 32 bit Big Endian) failed: Invalid argument D: [pulseaudio] alsa-util.c: Maximum hw buffer size is 128 ms I: [pulseaudio] (alsa-lib)pcm_hw.c: SNDRV_PCM_IOCTL_HW_PARAMS failed (-22) I: [pulseaudio] (alsa-lib)pcm_hw.c: SNDRV_PCM_IOCTL_HW_PARAMS failed (-22) I: [pulseaudio] (alsa-lib)pcm_hw.c: SNDRV_PCM_IOCTL_HW_PARAMS failed (-22) I: [pulseaudio] (alsa-lib)pcm_hw.c: SNDRV_PCM_IOCTL_HW_PARAMS failed (-22) D: [pulseaudio] alsa-util.c: Set neither period nor buffer size. I: [pulseaudio] (alsa-lib)pcm_hw.c: SNDRV_PCM_IOCTL_HW_PARAMS failed (-22) I: [pulseaudio] alsa-util.c: snd_pcm_hw_params failed: Invalid argument D: [pulseaudio] alsa-util.c: Trying plug:SLAVE='_ucm0006.hw:sc7180adau7002m,2' with SND_PCM_NO_AUTO_FORMAT ... I: [pulseaudio] (alsa-lib)pcm.c: Unknown PCM _ucm0006.hw:sc7180adau7002m,2 I: [pulseaudio] alsa-util.c: Error opening PCM device plug:SLAVE='_ucm0006.hw:sc7180adau7002m,2': No such file or directory D: [pulseaudio] alsa-mixer.c: Profile set 0xaaab034ad7e0, auto_profiles=no, probed=yes, n_mappings=0, n_profiles=0, n_decibel_fixes=0 E: [pulseaudio] module-alsa-card.c: Failed to find a working profile. E: [pulseaudio] module.c: Failed to load module "module-alsa-card" (argument: "device_id="0" name="platform-sound" card_name="alsa_card.platform-sound" namereg_fail=false tsched=yes fixed_latency_range=no ignore_dB=no deferred_volume=yes use_ucm=yes avoid_resampling=no card_properties="module-udev-detect.discovered=1""): initialization failed. I: [pulseaudio] module-udev-detect.c: Card /devices/platform/sound/sound/card0 (alsa_card.platform-sound) failed to load module. D: [pulseaudio] bluez5-util.c: Media application for adapter /org/bluez/hci0 was successfully registered D: [pulseaudio] module-udev-detect.c: /dev/snd/controlC0 is accessible: yes D: [pulseaudio] module-udev-detect.c: /devices/platform/sound/sound/card0 is busy: no W: [pulseaudio] module-udev-detect.c: Tried to configure /devices/platform/sound/sound/card0 (alsa_card.platform-sound) more often than 5 times in 10s ... ``` </p> </details> Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/403 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/403.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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[PATCH AUTOSEL 4.19 1/5] ASoC: Intel: bytcr_rt5640: Add an extra entry for the Chuwi Vi8 tablet
by Sasha Levin 11 Mar '24

11 Mar '24

From: Alban Boyé <alban.boye(a)protonmail.com> [ Upstream commit f8b0127aca8c60826e7354e504a12d4a46b1c3bb ] The bios version can differ depending if it is a dual-boot variant of the tablet. Therefore another DMI match is required. Signed-off-by: Alban Boyé <alban.boye(a)protonmail.com> Reviewed-by: Cezary Rojewski <cezary.rojewski(a)intel.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://msgid.link/r/20240228192807.15130-1-alban.boye@protonmail.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/bytcr_rt5640.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index d27dd170bedaf..19f425eb4a40f 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -523,6 +523,18 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { BYT_RT5640_SSP0_AIF1 | BYT_RT5640_MCLK_EN), }, + { /* Chuwi Vi8 dual-boot (CWI506) */ + .matches = { + DMI_EXACT_MATCH(DMI_SYS_VENDOR, "Insyde"), + DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "i86"), + /* The above are too generic, also match BIOS info */ + DMI_MATCH(DMI_BIOS_VERSION, "CHUWI2.D86JHBNR02"), + }, + .driver_data = (void *)(BYTCR_INPUT_DEFAULTS | + BYT_RT5640_MONO_SPEAKER | + BYT_RT5640_SSP0_AIF1 | + BYT_RT5640_MCLK_EN), + }, { /* Chuwi Vi10 (CWI505) */ .matches = { -- 2.43.0

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[PATCH AUTOSEL 5.4 1/5] ASoC: Intel: bytcr_rt5640: Add an extra entry for the Chuwi Vi8 tablet
by Sasha Levin 11 Mar '24

11 Mar '24

From: Alban Boyé <alban.boye(a)protonmail.com> [ Upstream commit f8b0127aca8c60826e7354e504a12d4a46b1c3bb ] The bios version can differ depending if it is a dual-boot variant of the tablet. Therefore another DMI match is required. Signed-off-by: Alban Boyé <alban.boye(a)protonmail.com> Reviewed-by: Cezary Rojewski <cezary.rojewski(a)intel.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://msgid.link/r/20240228192807.15130-1-alban.boye@protonmail.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/bytcr_rt5640.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index df3b370fe7292..c740dec00f83b 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -526,6 +526,18 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { BYT_RT5640_SSP0_AIF1 | BYT_RT5640_MCLK_EN), }, + { /* Chuwi Vi8 dual-boot (CWI506) */ + .matches = { + DMI_EXACT_MATCH(DMI_SYS_VENDOR, "Insyde"), + DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "i86"), + /* The above are too generic, also match BIOS info */ + DMI_MATCH(DMI_BIOS_VERSION, "CHUWI2.D86JHBNR02"), + }, + .driver_data = (void *)(BYTCR_INPUT_DEFAULTS | + BYT_RT5640_MONO_SPEAKER | + BYT_RT5640_SSP0_AIF1 | + BYT_RT5640_MCLK_EN), + }, { /* Chuwi Vi10 (CWI505) */ .matches = { -- 2.43.0

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[PATCH AUTOSEL 5.10 1/5] ASoC: Intel: bytcr_rt5640: Add an extra entry for the Chuwi Vi8 tablet
by Sasha Levin 11 Mar '24

11 Mar '24

From: Alban Boyé <alban.boye(a)protonmail.com> [ Upstream commit f8b0127aca8c60826e7354e504a12d4a46b1c3bb ] The bios version can differ depending if it is a dual-boot variant of the tablet. Therefore another DMI match is required. Signed-off-by: Alban Boyé <alban.boye(a)protonmail.com> Reviewed-by: Cezary Rojewski <cezary.rojewski(a)intel.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://msgid.link/r/20240228192807.15130-1-alban.boye@protonmail.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/bytcr_rt5640.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index f5b1b3b876980..1d049685e7075 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -529,6 +529,18 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { BYT_RT5640_SSP0_AIF1 | BYT_RT5640_MCLK_EN), }, + { /* Chuwi Vi8 dual-boot (CWI506) */ + .matches = { + DMI_EXACT_MATCH(DMI_SYS_VENDOR, "Insyde"), + DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "i86"), + /* The above are too generic, also match BIOS info */ + DMI_MATCH(DMI_BIOS_VERSION, "CHUWI2.D86JHBNR02"), + }, + .driver_data = (void *)(BYTCR_INPUT_DEFAULTS | + BYT_RT5640_MONO_SPEAKER | + BYT_RT5640_SSP0_AIF1 | + BYT_RT5640_MCLK_EN), + }, { /* Chuwi Vi10 (CWI505) */ .matches = { -- 2.43.0

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[PATCH AUTOSEL 5.15 1/5] ASoC: Intel: bytcr_rt5640: Add an extra entry for the Chuwi Vi8 tablet
by Sasha Levin 11 Mar '24

11 Mar '24

From: Alban Boyé <alban.boye(a)protonmail.com> [ Upstream commit f8b0127aca8c60826e7354e504a12d4a46b1c3bb ] The bios version can differ depending if it is a dual-boot variant of the tablet. Therefore another DMI match is required. Signed-off-by: Alban Boyé <alban.boye(a)protonmail.com> Reviewed-by: Cezary Rojewski <cezary.rojewski(a)intel.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://msgid.link/r/20240228192807.15130-1-alban.boye@protonmail.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/bytcr_rt5640.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 49dfbd29c5451..434679afa7e1a 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -668,6 +668,18 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { BYT_RT5640_SSP0_AIF1 | BYT_RT5640_MCLK_EN), }, + { /* Chuwi Vi8 dual-boot (CWI506) */ + .matches = { + DMI_EXACT_MATCH(DMI_SYS_VENDOR, "Insyde"), + DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "i86"), + /* The above are too generic, also match BIOS info */ + DMI_MATCH(DMI_BIOS_VERSION, "CHUWI2.D86JHBNR02"), + }, + .driver_data = (void *)(BYTCR_INPUT_DEFAULTS | + BYT_RT5640_MONO_SPEAKER | + BYT_RT5640_SSP0_AIF1 | + BYT_RT5640_MCLK_EN), + }, { /* Chuwi Vi10 (CWI505) */ .matches = { -- 2.43.0

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[PATCH AUTOSEL 6.1 7/7] ASoC: wm8962: Fix up incorrect error message in wm8962_set_fll
by Sasha Levin 11 Mar '24

11 Mar '24

From: Stuart Henderson <stuarth(a)opensource.cirrus.com> [ Upstream commit 96e202f8c52ac49452f83317cf3b34cd1ad81e18 ] Use source instead of ret, which seems to be unrelated and will always be zero. Signed-off-by: Stuart Henderson <stuarth(a)opensource.cirrus.com> Link: https://msgid.link/r/20240306161439.1385643-5-stuarth@opensource.cirrus.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/wm8962.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 806b69c9b2e36..d215e58c4a7b3 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -2933,7 +2933,7 @@ static int wm8962_set_fll(struct snd_soc_component *component, int fll_id, int s WM8962_FLL_FRC_NCO, WM8962_FLL_FRC_NCO); break; default: - dev_err(component->dev, "Unknown FLL source %d\n", ret); + dev_err(component->dev, "Unknown FLL source %d\n", source); return -EINVAL; } -- 2.43.0

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[PATCH AUTOSEL 6.1 6/7] ASoC: wm8962: Enable both SPKOUTR_ENA and SPKOUTL_ENA in mono mode
by Sasha Levin 11 Mar '24

11 Mar '24

From: Stuart Henderson <stuarth(a)opensource.cirrus.com> [ Upstream commit 6fa849e4d78b880e878138bf238e4fd2bac3c4fa ] Signed-off-by: Stuart Henderson <stuarth(a)opensource.cirrus.com> Link: https://msgid.link/r/20240306161439.1385643-2-stuarth@opensource.cirrus.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/wm8962.c | 23 ++++++++++++++--------- 1 file changed, 14 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 35d8cd4fc71a7..806b69c9b2e36 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -2229,6 +2229,9 @@ SND_SOC_DAPM_PGA_E("HPOUT", SND_SOC_NOPM, 0, 0, NULL, 0, hp_event, SND_SOC_DAPM_OUTPUT("HPOUTL"), SND_SOC_DAPM_OUTPUT("HPOUTR"), + +SND_SOC_DAPM_PGA("SPKOUTL Output", WM8962_CLASS_D_CONTROL_1, 6, 0, NULL, 0), +SND_SOC_DAPM_PGA("SPKOUTR Output", WM8962_CLASS_D_CONTROL_1, 7, 0, NULL, 0), }; static const struct snd_soc_dapm_widget wm8962_dapm_spk_mono_widgets[] = { @@ -2236,7 +2239,6 @@ SND_SOC_DAPM_MIXER("Speaker Mixer", WM8962_MIXER_ENABLES, 1, 0, spkmixl, ARRAY_SIZE(spkmixl)), SND_SOC_DAPM_MUX_E("Speaker PGA", WM8962_PWR_MGMT_2, 4, 0, &spkoutl_mux, out_pga_event, SND_SOC_DAPM_POST_PMU), -SND_SOC_DAPM_PGA("Speaker Output", WM8962_CLASS_D_CONTROL_1, 7, 0, NULL, 0), SND_SOC_DAPM_OUTPUT("SPKOUT"), }; @@ -2251,9 +2253,6 @@ SND_SOC_DAPM_MUX_E("SPKOUTL PGA", WM8962_PWR_MGMT_2, 4, 0, &spkoutl_mux, SND_SOC_DAPM_MUX_E("SPKOUTR PGA", WM8962_PWR_MGMT_2, 3, 0, &spkoutr_mux, out_pga_event, SND_SOC_DAPM_POST_PMU), -SND_SOC_DAPM_PGA("SPKOUTR Output", WM8962_CLASS_D_CONTROL_1, 7, 0, NULL, 0), -SND_SOC_DAPM_PGA("SPKOUTL Output", WM8962_CLASS_D_CONTROL_1, 6, 0, NULL, 0), - SND_SOC_DAPM_OUTPUT("SPKOUTL"), SND_SOC_DAPM_OUTPUT("SPKOUTR"), }; @@ -2366,12 +2365,18 @@ static const struct snd_soc_dapm_route wm8962_spk_mono_intercon[] = { { "Speaker PGA", "Mixer", "Speaker Mixer" }, { "Speaker PGA", "DAC", "DACL" }, - { "Speaker Output", NULL, "Speaker PGA" }, - { "Speaker Output", NULL, "SYSCLK" }, - { "Speaker Output", NULL, "TOCLK" }, - { "Speaker Output", NULL, "TEMP_SPK" }, + { "SPKOUTL Output", NULL, "Speaker PGA" }, + { "SPKOUTL Output", NULL, "SYSCLK" }, + { "SPKOUTL Output", NULL, "TOCLK" }, + { "SPKOUTL Output", NULL, "TEMP_SPK" }, + + { "SPKOUTR Output", NULL, "Speaker PGA" }, + { "SPKOUTR Output", NULL, "SYSCLK" }, + { "SPKOUTR Output", NULL, "TOCLK" }, + { "SPKOUTR Output", NULL, "TEMP_SPK" }, - { "SPKOUT", NULL, "Speaker Output" }, + { "SPKOUT", NULL, "SPKOUTL Output" }, + { "SPKOUT", NULL, "SPKOUTR Output" }, }; static const struct snd_soc_dapm_route wm8962_spk_stereo_intercon[] = { -- 2.43.0

1 0

[PATCH AUTOSEL 6.1 5/7] ASoC: wm8962: Enable oscillator if selecting WM8962_FLL_OSC
by Sasha Levin 11 Mar '24

11 Mar '24

From: Stuart Henderson <stuarth(a)opensource.cirrus.com> [ Upstream commit 03c7874106ca5032a312626b927b1c35f07b1f35 ] Signed-off-by: Stuart Henderson <stuarth(a)opensource.cirrus.com> Link: https://msgid.link/r/20240306161439.1385643-1-stuarth@opensource.cirrus.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/wm8962.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index b901e4c65e8a5..35d8cd4fc71a7 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -2914,8 +2914,12 @@ static int wm8962_set_fll(struct snd_soc_component *component, int fll_id, int s switch (fll_id) { case WM8962_FLL_MCLK: case WM8962_FLL_BCLK: + fll1 |= (fll_id - 1) << WM8962_FLL_REFCLK_SRC_SHIFT; + break; case WM8962_FLL_OSC: fll1 |= (fll_id - 1) << WM8962_FLL_REFCLK_SRC_SHIFT; + snd_soc_component_update_bits(component, WM8962_PLL2, + WM8962_OSC_ENA, WM8962_OSC_ENA); break; case WM8962_FLL_INT: snd_soc_component_update_bits(component, WM8962_FLL_CONTROL_1, -- 2.43.0

1 0

[PATCH AUTOSEL 6.1 2/7] ASoC: Intel: bytcr_rt5640: Add an extra entry for the Chuwi Vi8 tablet
by Sasha Levin 11 Mar '24

11 Mar '24

From: Alban Boyé <alban.boye(a)protonmail.com> [ Upstream commit f8b0127aca8c60826e7354e504a12d4a46b1c3bb ] The bios version can differ depending if it is a dual-boot variant of the tablet. Therefore another DMI match is required. Signed-off-by: Alban Boyé <alban.boye(a)protonmail.com> Reviewed-by: Cezary Rojewski <cezary.rojewski(a)intel.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://msgid.link/r/20240228192807.15130-1-alban.boye@protonmail.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/bytcr_rt5640.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 797d0a48d6066..094445036c20f 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -685,6 +685,18 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { BYT_RT5640_SSP0_AIF1 | BYT_RT5640_MCLK_EN), }, + { /* Chuwi Vi8 dual-boot (CWI506) */ + .matches = { + DMI_EXACT_MATCH(DMI_SYS_VENDOR, "Insyde"), + DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "i86"), + /* The above are too generic, also match BIOS info */ + DMI_MATCH(DMI_BIOS_VERSION, "CHUWI2.D86JHBNR02"), + }, + .driver_data = (void *)(BYTCR_INPUT_DEFAULTS | + BYT_RT5640_MONO_SPEAKER | + BYT_RT5640_SSP0_AIF1 | + BYT_RT5640_MCLK_EN), + }, { /* Chuwi Vi10 (CWI505) */ .matches = { -- 2.43.0

1 0

[PATCH AUTOSEL 6.6 12/12] ASoC: wm8962: Fix up incorrect error message in wm8962_set_fll
by Sasha Levin 11 Mar '24

11 Mar '24

From: Stuart Henderson <stuarth(a)opensource.cirrus.com> [ Upstream commit 96e202f8c52ac49452f83317cf3b34cd1ad81e18 ] Use source instead of ret, which seems to be unrelated and will always be zero. Signed-off-by: Stuart Henderson <stuarth(a)opensource.cirrus.com> Link: https://msgid.link/r/20240306161439.1385643-5-stuarth@opensource.cirrus.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/wm8962.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 914f8eb418357..4f50b07848fd8 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -2933,7 +2933,7 @@ static int wm8962_set_fll(struct snd_soc_component *component, int fll_id, int s WM8962_FLL_FRC_NCO, WM8962_FLL_FRC_NCO); break; default: - dev_err(component->dev, "Unknown FLL source %d\n", ret); + dev_err(component->dev, "Unknown FLL source %d\n", source); return -EINVAL; } -- 2.43.0

1 0

[PATCH AUTOSEL 6.6 11/12] ASoC: wm8962: Enable both SPKOUTR_ENA and SPKOUTL_ENA in mono mode
by Sasha Levin 11 Mar '24

11 Mar '24

From: Stuart Henderson <stuarth(a)opensource.cirrus.com> [ Upstream commit 6fa849e4d78b880e878138bf238e4fd2bac3c4fa ] Signed-off-by: Stuart Henderson <stuarth(a)opensource.cirrus.com> Link: https://msgid.link/r/20240306161439.1385643-2-stuarth@opensource.cirrus.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/wm8962.c | 23 ++++++++++++++--------- 1 file changed, 14 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index d444b7776ece1..914f8eb418357 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -2229,6 +2229,9 @@ SND_SOC_DAPM_PGA_E("HPOUT", SND_SOC_NOPM, 0, 0, NULL, 0, hp_event, SND_SOC_DAPM_OUTPUT("HPOUTL"), SND_SOC_DAPM_OUTPUT("HPOUTR"), + +SND_SOC_DAPM_PGA("SPKOUTL Output", WM8962_CLASS_D_CONTROL_1, 6, 0, NULL, 0), +SND_SOC_DAPM_PGA("SPKOUTR Output", WM8962_CLASS_D_CONTROL_1, 7, 0, NULL, 0), }; static const struct snd_soc_dapm_widget wm8962_dapm_spk_mono_widgets[] = { @@ -2236,7 +2239,6 @@ SND_SOC_DAPM_MIXER("Speaker Mixer", WM8962_MIXER_ENABLES, 1, 0, spkmixl, ARRAY_SIZE(spkmixl)), SND_SOC_DAPM_MUX_E("Speaker PGA", WM8962_PWR_MGMT_2, 4, 0, &spkoutl_mux, out_pga_event, SND_SOC_DAPM_POST_PMU), -SND_SOC_DAPM_PGA("Speaker Output", WM8962_CLASS_D_CONTROL_1, 7, 0, NULL, 0), SND_SOC_DAPM_OUTPUT("SPKOUT"), }; @@ -2251,9 +2253,6 @@ SND_SOC_DAPM_MUX_E("SPKOUTL PGA", WM8962_PWR_MGMT_2, 4, 0, &spkoutl_mux, SND_SOC_DAPM_MUX_E("SPKOUTR PGA", WM8962_PWR_MGMT_2, 3, 0, &spkoutr_mux, out_pga_event, SND_SOC_DAPM_POST_PMU), -SND_SOC_DAPM_PGA("SPKOUTR Output", WM8962_CLASS_D_CONTROL_1, 7, 0, NULL, 0), -SND_SOC_DAPM_PGA("SPKOUTL Output", WM8962_CLASS_D_CONTROL_1, 6, 0, NULL, 0), - SND_SOC_DAPM_OUTPUT("SPKOUTL"), SND_SOC_DAPM_OUTPUT("SPKOUTR"), }; @@ -2366,12 +2365,18 @@ static const struct snd_soc_dapm_route wm8962_spk_mono_intercon[] = { { "Speaker PGA", "Mixer", "Speaker Mixer" }, { "Speaker PGA", "DAC", "DACL" }, - { "Speaker Output", NULL, "Speaker PGA" }, - { "Speaker Output", NULL, "SYSCLK" }, - { "Speaker Output", NULL, "TOCLK" }, - { "Speaker Output", NULL, "TEMP_SPK" }, + { "SPKOUTL Output", NULL, "Speaker PGA" }, + { "SPKOUTL Output", NULL, "SYSCLK" }, + { "SPKOUTL Output", NULL, "TOCLK" }, + { "SPKOUTL Output", NULL, "TEMP_SPK" }, + + { "SPKOUTR Output", NULL, "Speaker PGA" }, + { "SPKOUTR Output", NULL, "SYSCLK" }, + { "SPKOUTR Output", NULL, "TOCLK" }, + { "SPKOUTR Output", NULL, "TEMP_SPK" }, - { "SPKOUT", NULL, "Speaker Output" }, + { "SPKOUT", NULL, "SPKOUTL Output" }, + { "SPKOUT", NULL, "SPKOUTR Output" }, }; static const struct snd_soc_dapm_route wm8962_spk_stereo_intercon[] = { -- 2.43.0

1 0

[PATCH AUTOSEL 6.6 10/12] ASoC: wm8962: Enable oscillator if selecting WM8962_FLL_OSC
by Sasha Levin 11 Mar '24

11 Mar '24

From: Stuart Henderson <stuarth(a)opensource.cirrus.com> [ Upstream commit 03c7874106ca5032a312626b927b1c35f07b1f35 ] Signed-off-by: Stuart Henderson <stuarth(a)opensource.cirrus.com> Link: https://msgid.link/r/20240306161439.1385643-1-stuarth@opensource.cirrus.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/wm8962.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 83ce5dbecc45d..d444b7776ece1 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -2914,8 +2914,12 @@ static int wm8962_set_fll(struct snd_soc_component *component, int fll_id, int s switch (fll_id) { case WM8962_FLL_MCLK: case WM8962_FLL_BCLK: + fll1 |= (fll_id - 1) << WM8962_FLL_REFCLK_SRC_SHIFT; + break; case WM8962_FLL_OSC: fll1 |= (fll_id - 1) << WM8962_FLL_REFCLK_SRC_SHIFT; + snd_soc_component_update_bits(component, WM8962_PLL2, + WM8962_OSC_ENA, WM8962_OSC_ENA); break; case WM8962_FLL_INT: snd_soc_component_update_bits(component, WM8962_FLL_CONTROL_1, -- 2.43.0

1 0

[PATCH AUTOSEL 6.6 06/12] ASoC: amd: yc: Add HP Pavilion Aero Laptop 13-be2xxx(8BD6) into DMI quirk table
by Sasha Levin 11 Mar '24

11 Mar '24

From: Al Raj Hassain <alrajhassain(a)gmail.com> [ Upstream commit b3a51137607cee7c814cd3a75d96f78b9ee1dc1f ] The HP Pavilion Aero Laptop 13-be2xxx(8BD6) requires a quirk entry for its internal microphone to function. Signed-off-by: Al Raj Hassain <alrajhassain(a)gmail.com> Reviewed-by: Mario Limonciello <mario.limonciello(a)amd.com> Link: https://msgid.link/r/20240304103924.13673-1-alrajhassain@gmail.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/amd/yc/acp6x-mach.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/amd/yc/acp6x-mach.c b/sound/soc/amd/yc/acp6x-mach.c index 80ad60d485ea0..ea9512efa6fa5 100644 --- a/sound/soc/amd/yc/acp6x-mach.c +++ b/sound/soc/amd/yc/acp6x-mach.c @@ -395,6 +395,13 @@ static const struct dmi_system_id yc_acp_quirk_table[] = { DMI_MATCH(DMI_BOARD_NAME, "8B2F"), } }, + { + .driver_data = &acp6x_card, + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "HP"), + DMI_MATCH(DMI_BOARD_NAME, "8BD6"), + } + }, { .driver_data = &acp6x_card, .matches = { -- 2.43.0

1 0

[PATCH AUTOSEL 6.6 02/12] ASoC: Intel: bytcr_rt5640: Add an extra entry for the Chuwi Vi8 tablet
by Sasha Levin 11 Mar '24

11 Mar '24

From: Alban Boyé <alban.boye(a)protonmail.com> [ Upstream commit f8b0127aca8c60826e7354e504a12d4a46b1c3bb ] The bios version can differ depending if it is a dual-boot variant of the tablet. Therefore another DMI match is required. Signed-off-by: Alban Boyé <alban.boye(a)protonmail.com> Reviewed-by: Cezary Rojewski <cezary.rojewski(a)intel.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://msgid.link/r/20240228192807.15130-1-alban.boye@protonmail.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/bytcr_rt5640.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 6b631a608ae8f..e609249cc38d5 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -685,6 +685,18 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { BYT_RT5640_SSP0_AIF1 | BYT_RT5640_MCLK_EN), }, + { /* Chuwi Vi8 dual-boot (CWI506) */ + .matches = { + DMI_EXACT_MATCH(DMI_SYS_VENDOR, "Insyde"), + DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "i86"), + /* The above are too generic, also match BIOS info */ + DMI_MATCH(DMI_BIOS_VERSION, "CHUWI2.D86JHBNR02"), + }, + .driver_data = (void *)(BYTCR_INPUT_DEFAULTS | + BYT_RT5640_MONO_SPEAKER | + BYT_RT5640_SSP0_AIF1 | + BYT_RT5640_MCLK_EN), + }, { /* Chuwi Vi10 (CWI505) */ .matches = { -- 2.43.0

1 0

[PATCH AUTOSEL 6.7 14/14] ASoC: wm8962: Fix up incorrect error message in wm8962_set_fll
by Sasha Levin 11 Mar '24

11 Mar '24

From: Stuart Henderson <stuarth(a)opensource.cirrus.com> [ Upstream commit 96e202f8c52ac49452f83317cf3b34cd1ad81e18 ] Use source instead of ret, which seems to be unrelated and will always be zero. Signed-off-by: Stuart Henderson <stuarth(a)opensource.cirrus.com> Link: https://msgid.link/r/20240306161439.1385643-5-stuarth@opensource.cirrus.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/wm8962.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 2256cc0a37ebd..7c6ed29831285 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -2933,7 +2933,7 @@ static int wm8962_set_fll(struct snd_soc_component *component, int fll_id, int s WM8962_FLL_FRC_NCO, WM8962_FLL_FRC_NCO); break; default: - dev_err(component->dev, "Unknown FLL source %d\n", ret); + dev_err(component->dev, "Unknown FLL source %d\n", source); return -EINVAL; } -- 2.43.0

1 0

[PATCH AUTOSEL 6.7 13/14] ASoC: wm8962: Enable both SPKOUTR_ENA and SPKOUTL_ENA in mono mode
by Sasha Levin 11 Mar '24

11 Mar '24

From: Stuart Henderson <stuarth(a)opensource.cirrus.com> [ Upstream commit 6fa849e4d78b880e878138bf238e4fd2bac3c4fa ] Signed-off-by: Stuart Henderson <stuarth(a)opensource.cirrus.com> Link: https://msgid.link/r/20240306161439.1385643-2-stuarth@opensource.cirrus.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/wm8962.c | 23 ++++++++++++++--------- 1 file changed, 14 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 6d7bb696b1356..2256cc0a37ebd 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -2229,6 +2229,9 @@ SND_SOC_DAPM_PGA_E("HPOUT", SND_SOC_NOPM, 0, 0, NULL, 0, hp_event, SND_SOC_DAPM_OUTPUT("HPOUTL"), SND_SOC_DAPM_OUTPUT("HPOUTR"), + +SND_SOC_DAPM_PGA("SPKOUTL Output", WM8962_CLASS_D_CONTROL_1, 6, 0, NULL, 0), +SND_SOC_DAPM_PGA("SPKOUTR Output", WM8962_CLASS_D_CONTROL_1, 7, 0, NULL, 0), }; static const struct snd_soc_dapm_widget wm8962_dapm_spk_mono_widgets[] = { @@ -2236,7 +2239,6 @@ SND_SOC_DAPM_MIXER("Speaker Mixer", WM8962_MIXER_ENABLES, 1, 0, spkmixl, ARRAY_SIZE(spkmixl)), SND_SOC_DAPM_MUX_E("Speaker PGA", WM8962_PWR_MGMT_2, 4, 0, &spkoutl_mux, out_pga_event, SND_SOC_DAPM_POST_PMU), -SND_SOC_DAPM_PGA("Speaker Output", WM8962_CLASS_D_CONTROL_1, 7, 0, NULL, 0), SND_SOC_DAPM_OUTPUT("SPKOUT"), }; @@ -2251,9 +2253,6 @@ SND_SOC_DAPM_MUX_E("SPKOUTL PGA", WM8962_PWR_MGMT_2, 4, 0, &spkoutl_mux, SND_SOC_DAPM_MUX_E("SPKOUTR PGA", WM8962_PWR_MGMT_2, 3, 0, &spkoutr_mux, out_pga_event, SND_SOC_DAPM_POST_PMU), -SND_SOC_DAPM_PGA("SPKOUTR Output", WM8962_CLASS_D_CONTROL_1, 7, 0, NULL, 0), -SND_SOC_DAPM_PGA("SPKOUTL Output", WM8962_CLASS_D_CONTROL_1, 6, 0, NULL, 0), - SND_SOC_DAPM_OUTPUT("SPKOUTL"), SND_SOC_DAPM_OUTPUT("SPKOUTR"), }; @@ -2366,12 +2365,18 @@ static const struct snd_soc_dapm_route wm8962_spk_mono_intercon[] = { { "Speaker PGA", "Mixer", "Speaker Mixer" }, { "Speaker PGA", "DAC", "DACL" }, - { "Speaker Output", NULL, "Speaker PGA" }, - { "Speaker Output", NULL, "SYSCLK" }, - { "Speaker Output", NULL, "TOCLK" }, - { "Speaker Output", NULL, "TEMP_SPK" }, + { "SPKOUTL Output", NULL, "Speaker PGA" }, + { "SPKOUTL Output", NULL, "SYSCLK" }, + { "SPKOUTL Output", NULL, "TOCLK" }, + { "SPKOUTL Output", NULL, "TEMP_SPK" }, + + { "SPKOUTR Output", NULL, "Speaker PGA" }, + { "SPKOUTR Output", NULL, "SYSCLK" }, + { "SPKOUTR Output", NULL, "TOCLK" }, + { "SPKOUTR Output", NULL, "TEMP_SPK" }, - { "SPKOUT", NULL, "Speaker Output" }, + { "SPKOUT", NULL, "SPKOUTL Output" }, + { "SPKOUT", NULL, "SPKOUTR Output" }, }; static const struct snd_soc_dapm_route wm8962_spk_stereo_intercon[] = { -- 2.43.0

1 0

[PATCH AUTOSEL 6.7 12/14] ASoC: wm8962: Enable oscillator if selecting WM8962_FLL_OSC
by Sasha Levin 11 Mar '24

11 Mar '24

From: Stuart Henderson <stuarth(a)opensource.cirrus.com> [ Upstream commit 03c7874106ca5032a312626b927b1c35f07b1f35 ] Signed-off-by: Stuart Henderson <stuarth(a)opensource.cirrus.com> Link: https://msgid.link/r/20240306161439.1385643-1-stuarth@opensource.cirrus.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/wm8962.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index fb90ae6a8a344..6d7bb696b1356 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -2914,8 +2914,12 @@ static int wm8962_set_fll(struct snd_soc_component *component, int fll_id, int s switch (fll_id) { case WM8962_FLL_MCLK: case WM8962_FLL_BCLK: + fll1 |= (fll_id - 1) << WM8962_FLL_REFCLK_SRC_SHIFT; + break; case WM8962_FLL_OSC: fll1 |= (fll_id - 1) << WM8962_FLL_REFCLK_SRC_SHIFT; + snd_soc_component_update_bits(component, WM8962_PLL2, + WM8962_OSC_ENA, WM8962_OSC_ENA); break; case WM8962_FLL_INT: snd_soc_component_update_bits(component, WM8962_FLL_CONTROL_1, -- 2.43.0

1 0

[PATCH AUTOSEL 6.7 08/14] ASoC: amd: yc: Add HP Pavilion Aero Laptop 13-be2xxx(8BD6) into DMI quirk table
by Sasha Levin 11 Mar '24

11 Mar '24

From: Al Raj Hassain <alrajhassain(a)gmail.com> [ Upstream commit b3a51137607cee7c814cd3a75d96f78b9ee1dc1f ] The HP Pavilion Aero Laptop 13-be2xxx(8BD6) requires a quirk entry for its internal microphone to function. Signed-off-by: Al Raj Hassain <alrajhassain(a)gmail.com> Reviewed-by: Mario Limonciello <mario.limonciello(a)amd.com> Link: https://msgid.link/r/20240304103924.13673-1-alrajhassain@gmail.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/amd/yc/acp6x-mach.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/amd/yc/acp6x-mach.c b/sound/soc/amd/yc/acp6x-mach.c index 80ad60d485ea0..ea9512efa6fa5 100644 --- a/sound/soc/amd/yc/acp6x-mach.c +++ b/sound/soc/amd/yc/acp6x-mach.c @@ -395,6 +395,13 @@ static const struct dmi_system_id yc_acp_quirk_table[] = { DMI_MATCH(DMI_BOARD_NAME, "8B2F"), } }, + { + .driver_data = &acp6x_card, + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "HP"), + DMI_MATCH(DMI_BOARD_NAME, "8BD6"), + } + }, { .driver_data = &acp6x_card, .matches = { -- 2.43.0

1 0

[PATCH AUTOSEL 6.7 07/14] ALSA: hda: cs35l41: Overwrite CS35L41 configuration for ASUS UM5302LA
by Sasha Levin 11 Mar '24

11 Mar '24

From: Stefan Binding <sbinding(a)opensource.cirrus.com> [ Upstream commit b603d95692e47dc6f5f733e93c3841dc0c01e624 ] Whilst this laptop contains _DSD inside the BIOS, there is an error in this configuration. Override the _DSD in the BIOS with the correct configuration for this laptop. Signed-off-by: Stefan Binding <sbinding(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20240301160154.158398-4-sbinding@opensource.cirru… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/cs35l41_hda_property.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/cs35l41_hda_property.c b/sound/pci/hda/cs35l41_hda_property.c index d74cf11eef1ea..2de06e3db1a9e 100644 --- a/sound/pci/hda/cs35l41_hda_property.c +++ b/sound/pci/hda/cs35l41_hda_property.c @@ -83,6 +83,7 @@ static const struct cs35l41_config cs35l41_config_table[] = { { "104317F3", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4500, 24 }, { "10431863", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 1000, 4500, 24 }, { "104318D3", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 0, 0, 0 }, + { "10431A83", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4500, 24 }, { "10431C9F", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 1000, 4500, 24 }, { "10431CAF", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 1000, 4500, 24 }, { "10431CCF", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 1000, 4500, 24 }, @@ -419,6 +420,7 @@ static const struct cs35l41_prop_model cs35l41_prop_model_table[] = { { "CSC3551", "104317F3", generic_dsd_config }, { "CSC3551", "10431863", generic_dsd_config }, { "CSC3551", "104318D3", generic_dsd_config }, + { "CSC3551", "10431A83", generic_dsd_config }, { "CSC3551", "10431C9F", generic_dsd_config }, { "CSC3551", "10431CAF", generic_dsd_config }, { "CSC3551", "10431CCF", generic_dsd_config }, -- 2.43.0

1 0

[PATCH AUTOSEL 6.7 06/14] ALSA: hda/realtek: Add quirks for Lenovo Thinkbook 16P laptops
by Sasha Levin 11 Mar '24

11 Mar '24

From: Stefan Binding <sbinding(a)opensource.cirrus.com> [ Upstream commit 6214e24cae9b10a7c1572f99552610a24614fffe ] These models use 2 CS35L41 amps with HDA using I2C. Both models have _DSD support inside cs35l41_hda_property.c. Closes: https://bugzilla.kernel.org/show_bug.cgi?id=218437 Signed-off-by: Stefan Binding <sbinding(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20240301160154.158398-3-sbinding@opensource.cirru… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/patch_realtek.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index eb45e5c3db8c6..9590fd31441bc 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -10288,6 +10288,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x3886, "Y780 VECO DUAL", ALC287_FIXUP_TAS2781_I2C), SND_PCI_QUIRK(0x17aa, 0x38a7, "Y780P AMD YG dual", ALC287_FIXUP_TAS2781_I2C), SND_PCI_QUIRK(0x17aa, 0x38a8, "Y780P AMD VECO dual", ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x17aa, 0x38a9, "Thinkbook 16P", ALC287_FIXUP_CS35L41_I2C_2), + SND_PCI_QUIRK(0x17aa, 0x38ab, "Thinkbook 16P", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x17aa, 0x38b4, "Legion Slim 7 16IRH8", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x17aa, 0x38b5, "Legion Slim 7 16IRH8", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x17aa, 0x38b6, "Legion Slim 7 16APH8", ALC287_FIXUP_CS35L41_I2C_2), -- 2.43.0

1 0

[PATCH AUTOSEL 6.7 02/14] ASoC: Intel: bytcr_rt5640: Add an extra entry for the Chuwi Vi8 tablet
by Sasha Levin 11 Mar '24

11 Mar '24

From: Alban Boyé <alban.boye(a)protonmail.com> [ Upstream commit f8b0127aca8c60826e7354e504a12d4a46b1c3bb ] The bios version can differ depending if it is a dual-boot variant of the tablet. Therefore another DMI match is required. Signed-off-by: Alban Boyé <alban.boye(a)protonmail.com> Reviewed-by: Cezary Rojewski <cezary.rojewski(a)intel.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://msgid.link/r/20240228192807.15130-1-alban.boye@protonmail.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/bytcr_rt5640.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 42466b4b1ca45..a290f498ba823 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -685,6 +685,18 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { BYT_RT5640_SSP0_AIF1 | BYT_RT5640_MCLK_EN), }, + { /* Chuwi Vi8 dual-boot (CWI506) */ + .matches = { + DMI_EXACT_MATCH(DMI_SYS_VENDOR, "Insyde"), + DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "i86"), + /* The above are too generic, also match BIOS info */ + DMI_MATCH(DMI_BIOS_VERSION, "CHUWI2.D86JHBNR02"), + }, + .driver_data = (void *)(BYTCR_INPUT_DEFAULTS | + BYT_RT5640_MONO_SPEAKER | + BYT_RT5640_SSP0_AIF1 | + BYT_RT5640_MCLK_EN), + }, { /* Chuwi Vi10 (CWI505) */ .matches = { -- 2.43.0

1 0

[PATCH] ASoC: tlv320adc3xxx: Don't strip remove function when driver is builtin
by Uwe Kleine-König 11 Mar '24

11 Mar '24

Using __exit for the remove function results in the remove callback being discarded with SND_SOC_TLV320ADC3XXX=y. When such a device gets unbound (e.g. using sysfs or hotplug), the driver is just removed without the cleanup being performed. This results in resource leaks. Fix it by compiling in the remove callback unconditionally. This also fixes a W=1 modpost warning: WARNING: modpost: sound/soc/codecs/snd-soc-tlv320adc3xxx: section mismatch in reference: adc3xxx_i2c_driver+0x10 (section: .data) -> adc3xxx_i2c_remove (section: .exit.text) (which only happens with SND_SOC_TLV320ADC3XXX=m). Fixes: e9a3b57efd28 ("ASoC: codec: tlv320adc3xxx: New codec driver") Signed-off-by: Uwe Kleine-König <u.kleine-koenig(a)pengutronix.de> --- Hello, note that this patch only applies as is after commit 4e8ff3587868 ("ASoC: codecs: tlv320adc3xxx: Wrap adc3xxx_i2c_remove() in __exit_p()") which was the wrong approach to fix the build error. Best regards Uwe sound/soc/codecs/tlv320adc3xxx.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/tlv320adc3xxx.c b/sound/soc/codecs/tlv320adc3xxx.c index 420bbf588efe..e100cc9f5c19 100644 --- a/sound/soc/codecs/tlv320adc3xxx.c +++ b/sound/soc/codecs/tlv320adc3xxx.c @@ -1429,7 +1429,7 @@ static int adc3xxx_i2c_probe(struct i2c_client *i2c) return ret; } -static void __exit adc3xxx_i2c_remove(struct i2c_client *client) +static void adc3xxx_i2c_remove(struct i2c_client *client) { struct adc3xxx *adc3xxx = i2c_get_clientdata(client); @@ -1452,7 +1452,7 @@ static struct i2c_driver adc3xxx_i2c_driver = { .of_match_table = tlv320adc3xxx_of_match, }, .probe = adc3xxx_i2c_probe, - .remove = __exit_p(adc3xxx_i2c_remove), + .remove = adc3xxx_i2c_remove, .id_table = adc3xxx_i2c_id, }; base-commit: 8ffc8b1bbd505e27e2c8439d326b6059c906c9dd -- 2.43.0

3 2

[GIT PULL] ASoC updates for v6.9
by Mark Brown 11 Mar '24

11 Mar '24

The following changes since commit b401b621758e46812da61fa58a67c3fd8d91de0d: Linux 6.8-rc5 (2024-02-18 12:56:25 -0800) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v6.9 for you to fetch changes up to 6c023ad32b192dea51a4f842cc6ecf89bb6238c9: ASoC: Intel: catpt: Carefully use PCI bitwise constants (2024-03-07 16:47:01 +0000) ---------------------------------------------------------------- ASoC: Updates for v6.9 This has been quite a small release, there's a lot of driver specific cleanups and minor enhancements but hardly anything on the core and only one new driver. Highlights include: - SoundWire support for AMD ACP 6.3 systems. - Support for reporting version information for AVS firmware. - Support DSPless mode for Intel Soundwire systems. - Support for configuring CS35L56 amplifiers using EFI calibration data. - Log which component is being operated on as part of power management trace events. - Support for Microchip SAM9x7, NXP i.MX95 and Qualcomm WCD939x ---------------------------------------------------------------- Alexander Stein (2): ASoC: dt-bindings: fsl-sai: Add power-domains ASoC: dt-bindings: fsl-sai: Support Rx-only SAI Amadeusz Sławiński (3): ASoC: Intel: avs: UAPI: Add tokens for initial config feature ASoC: Intel: avs: Add topology parsing support for initial config ASoC: Intel: avs: Send initial config to module if present Andy Shevchenko (2): ASoC: Intel: atom: sst_ipc: Remove unused intel-mid.h ASoC: Intel: catpt: Carefully use PCI bitwise constants Arnd Bergmann (3): ASoC: SOF: amd: fix SND_AMD_SOUNDWIRE_ACPI dependencies ASoC: fix SND_SOC_WCD939X dependencies ASoC: SOF: amd: fix soundwire dependencies Bard Liao (12): ASoC: Intel: sof_sdw: use single rtd_init for rt_amps ASoC: Intel: add get_codec_dai_by_name helper function ASoC: Intel: sof_sdw_rt_sdca_jack_common: use helper to get codec dai by name ASoC: Intel: sof_sdw_rt711: use helper to get codec dai by name ASoC: Intel: sof_sdw_rt712_sdca: use helper to get codec dai by name ASoC: Intel: sof_sdw_rt700: use helper to get codec dai by name ASoC: Intel: sof_sdw_cs42l42: use helper to get codec dai by name ASoC: Intel: sof_sdw_rt5682: use helper to get codec dai by name ASoC: Intel: sof_sdw: add common sdw dai link init ASoC: Intel: sof_sdw: remove .init callbacks ASoC: Intel: sof_sdw: starts non sdw BE id with the highest sdw BE id ASoC: SOF: ipc4-topology: set config_length based on device_count Brent Lu (8): ASoC: Intel: sof_rt5682: board id cleanup for jsl boards ASoC: Intel: sof_rt5682: board id cleanup for tgl boards ASoC: Intel: sof_rt5682: board id cleanup for adl boards ASoC: Intel: sof_rt5682: board id cleanup for rpl boards ASoC: Intel: sof_rt5682: board id cleanup for mtl boards ASoC: Intel: sof_rt5682: dmi quirk cleanup for mtl boards ASoC: Intel: board_helpers: support DAI link order customization ASoC: Intel: sof_cs42l42: use common module for DAI link generation Cezary Rojewski (17): ASoC: Intel: avs: Expose FW version with sysfs ASoC: Intel: avs: L1SEN reference counted ASoC: Intel: avs: Fix sound clipping in single capture scenario ASoC: Intel: avs: Prefix SKL/APL-specific members ASoC: Intel: avs: Abstract IPC handling ASoC: Intel: avs: Abstract IRQ handling ASoC: Intel: avs: CNL-based platforms support ASoC: Intel: avs: ICL-based platforms support ASoC: Intel: avs: TGL-based platforms support ASoC: Intel: avs: ICCMAX recommendations for ICL+ platforms ASoC: Intel: avs: Populate board selection with new I2S entries ASoC: codecs: da7213: Simplify mclk initialization ASoC: codecs: nau8825: Simplify mclk initialization ASoC: codecs: rt5514: Simplify mclk initialization ASoC: codecs: rt5616: Simplify mclk initialization ASoC: codecs: rt5640: Simplify mclk initialization ASoC: codecs: rt5660: Simplify mclk initialization Chancel Liu (4): ASoC: dt-bindings: fsl,sai: Add compatible string for i.MX95 platform ASoC: fsl_sai: Add support for i.MX95 platform ASoC: dt-bindings: fsl,micfil: Add compatible string for i.MX95 platform ASoC: soc-core.c: Prefer to return dai->driver->name in snd_soc_dai_name_get() Chao Song (2): ASoC: Intel: soc-acpi: add RT712 support for LNL ASoC: Intel: soc-acpi-intel-lnl-match: Add rt722 support Charles Keepax (12): ASoC: cs42l43: Tidy up header includes ASoC: cs42l43: Minor code tidy ups ASoC: cs42l43: Check error from device_property_read_u32_array() ASoC: cs42l43: Add pm_ptr around the power ops ASoC: cs42l43: Use USEC_PER_MSEC rather than hard coding ASoC: cs42l43: Refactor to use for_each_set_bit() ASoC: cs42l43: Use fls to calculate the pre-divider for the PLL ASoC: cs42l43: Add clear of stashed pointer on component remove ASoC: cs42l43: Sync the hp ilimit works when removing the component ASoC: cs42l43: Shut down jack detection on component remove ASoC: Intel: ssp-common: Add stub for sof_ssp_get_codec_name ASoC: Intel: sof_sdw: Remove unused function prototypes Colin Ian King (1): ASoC: tas2781: Remove redundant initialization of pointer 'data' Cristian Ciocaltea (8): ASoC: amd: acp: Drop redundant initialization of machine driver data ASoC: amd: acp: Make use of existing *_CODEC_DAI macros ASoC: amd: acp: Add missing error handling in sof-mach ASoC: amd: acp: Update MODULE_DESCRIPTION for sof-mach ASoC: SOF: amd: Fix memory leak in amd_sof_acp_probe() ASoC: SOF: amd: Optimize quirk for Valve Galileo ASoC: SOF: core: Skip firmware test for custom loaders ASoC: SOF: amd: Compute file paths on firmware load Dan Carpenter (1): ASoC: SOF: Add some bounds checking to firmware data Daniel Baluta (1): ASoC: SOF: imx: Add devicetree support to select topologies David Lin (1): ASoC: nau8540: Add pre-charge actions for input Dharma Balasubiramani (2): ASoC: dt-bindings: atmel,sam9x5-wm8731: Convert to json-schema ASoC: dt-bindings: atmel,asoc-wm8904: Convert to json-schema Erick Archer (2): ASoC: qcom: Use devm_kcalloc() instead of devm_kzalloc() ASoC: ti: j721e-evm: Use devm_kcalloc() instead of devm_kzalloc() Gergo Koteles (1): ASoC: tas2781: remove unused acpi_subysystem_id Herve Codina (1): ASoC: codecs: Add support for the framer codec Javier García (1): ASoC: dt-bindings: rt1015: Convert to dtschema Jerome Brunet (8): ASoC: meson: aiu: fix function pointer type mismatch ASoC: meson: t9015: fix function pointer type mismatch ASoC: meson: axg-tdm-interface: fix mclk setup without mclk-fs ASoC: meson: axg-tdm-interface: add frame rate constraint ASoC: meson: axg-tdm-interface: update error format error traces ASoC: meson: axg-spdifin: use max width for rate detection ASoC: meson: axg-fifo: take continuous rates ASoC: meson: axg-fifo: use FIELD helpers Konrad Dybcio (1): ASoC: dt-bindings: cs35l45: Add interrupts Krzysztof Kozlowski (11): ASoC: dt-bindings: Do not override firmware-name $ref ASoC: codecs: constify static sdw_slave_ops struct ASoC: dt-bindings: qcom,sm8250: Allow up to 8 codec DAIs ASoC: codecs: tx-macro: Drop unimplemented DMIC clock divider ASoC: codecs: tx-macro: Mark AMIC control registers as volatile ASoC: codecs: tx-macro: Simplify setting AMIC control ASoC: Constify pointer to of_phandle_args ASoC: dt-bindings: qcom,wsa8840: Add reset-gpios for shared line ASoC: codecs: wsa884x: Allow sharing reset GPIO ASoC: codecs: tx-macro: split widgets per different LPASS versions ASoC: codecs: tx-macro: correct TX SMIC MUXn widgets on SM8350+ Kuninori Morimoto (4): ASoC: ti: use devm_snd_soc_register_card() ASoC: fsl: use devm_snd_soc_register_card() ASoC: atmel: use devm_snd_soc_register_card() ASoC: soc-core: tidyup strcmp() param on snd_soc_is_matching_dai() Lad Prabhakar (1): ASoC: sh: rz-ssi: Fix error message print Luca Ceresoli (2): ASoC: trace: add component to set_bias_level trace events ASoC: trace: add event to snd_soc_dapm trace events Maciej Strozek (1): ASoC: Intel: mtl-match: Add cs42l43_l0 cs35l56_l23 for MTL Mark Brown (15): Improve SOF support for Steam Deck OLED ASoC: codecs: add support for WCD939x Codec ASoC: Merge up fixes ASoC: Support SAI and MICFIL on i.MX95 platform ASoC: use devm_snd_soc_register_card() ASoC: Merge up fixes due to dependency with new changes soundwire/SOF: add SoundWire Interface support for ASoC: Intel: avs: Add support for sending initial ASoC: SOF: Extend ChainDMA and DSPless mode to LNL+ ASoC: meson: aiu: fix function pointer type ASoC: Intel: avs: Fixes and new platforms support ALSA: cs35l56: Apply calibration from EFI ASoC: codecs: Simplify mclk initialization ASoC: meson: axg fixes and clean-up ASoC: codecs: ES8326: change members of private Masahiro Yamada (2): ASoC: pxa: remove duplicated CONFIG_SND_PXA2XX_AC97 entry ASoC: codecs: remove redundant 'tristate' in sound/soc/codecs/Kconfig Neil Armstrong (5): ASoC: dt-bindings: qcom,wcd938x: move out common properties ASoC: dt-bindings: document WCD939x Audio Codec ASoC: codec: wcd-mbhc-v2: add support when connected behind an USB-C audio mux ASoC: codecs: Add WCD939x Soundwire devices driver ASoC: codecs: Add WCD939x Codec driver Peter Ujfalusi (2): ASoC: SOF: Intel: lnl: Do not use LNL specific wrappers in DSPless mode ASoC: SOF: Fix runtime pm usage counter balance after fw exception Pierre-Louis Bossart (14): ASoC: SOF: ipc4-topology: change chain_dma handling in dai_config ASoC: SOF: ops: add new 'is_chain_dma_supported' callback ASoC: SOF: Intel: hda: add 'is_chain_dma_supported' callback ASoC: SOF: Intel: hda-dai-ops: enable chain_dma for ALH ASoC: SOF: ipc4: store number of playback/capture streams ASoC: SOF: ipc4-pcm: fix dma_id for CHAIN_DMA capture ASoC: SOF: ipc4-topology: allow chain_dma for all supported DAIs ASoC: SOF: Intel: hda-dai: remove dspless special case ASoC: SOF: topology: dynamically allocate and store DAI widget->private ASoC: SOF: Intel: start SoundWire links earlier for LNL+ devices ASoC: SOF: Intel: hda-dai-ops: use dai_type ASoC: SOF: Intel: hda-dai-ops: add SoundWire dspless mode ASoC: SOF: Intel: hda-dai: add support for dspless mode beyond HDAudio ASoC: SOF: Intel: hda-mlink: update incorrect comment R SUNDAR (1): spi: Remove the @multi_cs_cap to prevent kernel-doc warnings Randy Dunlap (1): ASoC: p1022_rdk: fix all kernel-doc warnings Ranjani Sridharan (1): ASoC: SOF: topology: Parse DAI type token for dspless mode Richard Fitzgerald (9): ASoC: wm_adsp: Add wm_adsp_start() and wm_adsp_stop() ASoC: cs-amp-lib: Add helpers for factory calibration data ASoC: cs35l56: Add helper functions for amp calibration ASoC: cs35l56: Apply amp calibration from EFI data ALSA: hda: cs35l56: Apply amp calibration from EFI data ASoC: cs35l56: Prevent bad sign extension in cs35l56_read_silicon_uid() ASoC: cs-amp-lib: Add KUnit test for calibration helpers firmware: cirrus: cs_dsp: Initialize debugfs_root to invalid firmware: cirrus: cs_dsp: Remove non-existent member from kerneldoc Rob Herring (2): ASoC: dt-bindings: samsung,tm2: Correct "audio-codec" constraints ASoC: dt-bindings: audio-graph-port: Drop type from "clocks" Sebastian Reichel (1): ASoC: dt-bindings: fsl,imx-asrc: convert to YAML Srinivas Kandagatla (1): ASoC: codecs: va-macro: add npl clk Varshini Rajendran (2): ASoC: dt-bindings: atmel-classd: add sam9x7 compatible ASoC: dt-bindings: microchip: add sam9x7 Vijendar Mukunda (20): ASoC/soundwire: implement generic api for scanning amd soundwire controller soundwire: amd: update license soundwire: amd: refactor amd soundwire manager device node creation soundwire: amd: implement function to extract slave information soundwire: amd: refactor soundwire pads enable soundwire: amd: 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 ASoC: amd: ps: refactor acp device configuration read logic ASoC: amd: ps: refactor acp child platform device creation code ASoC: amd: ps: remove acp_reset flag ASoC: amd: ps: fix for acp pme wake for soundwire configuration ASoC: amd: ps: add machine select and register code ASoC: amd: ps: update license ASoC: amd: ps: modify ACP register end address macro Vitaly Rodionov (1): ASoC: cs42l42: Remove redundant delays in suspend(). Yang Li (1): ASoC: codecs: Remove unneeded semicolon Yinchuan Guo (1): ASoC: codecs: fix TYPO 'reguest' to 'request' in error log Zhang Yi (2): ASoC: codecs: ES8326: Changing members of private structure ASoC: codecs: ES8326: change support for ES8326 Zhu Ning (1): ASoC: codecs: ES8326: Adding new volume kcontrols mosomate (1): ASoC: Intel: common: DMI remap for rebranded Intel NUC M15 (LAPRC710) laptops .../ABI/testing/sysfs-bus-pci-devices-avs | 8 + .../bindings/sound/atmel,asoc-wm8904.yaml | 84 + .../bindings/sound/atmel,sam9x5-wm8731-audio.yaml | 76 + .../bindings/sound/atmel,sama5d2-classd.yaml | 7 +- .../bindings/sound/atmel-sam9x5-wm8731-audio.txt | 35 - .../devicetree/bindings/sound/atmel-wm8904.txt | 55 - .../bindings/sound/audio-graph-port.yaml | 2 +- .../devicetree/bindings/sound/cirrus,cs35l45.yaml | 3 + .../devicetree/bindings/sound/everest,es8326.yaml | 8 +- .../devicetree/bindings/sound/fsl,asrc.txt | 80 - .../devicetree/bindings/sound/fsl,easrc.yaml | 4 +- .../devicetree/bindings/sound/fsl,imx-asrc.yaml | 162 + .../devicetree/bindings/sound/fsl,micfil.yaml | 14 +- .../devicetree/bindings/sound/fsl,sai.yaml | 6 + .../bindings/sound/infineon,peb2466.yaml | 2 +- .../bindings/sound/microchip,sama7g5-i2smcc.yaml | 11 +- .../devicetree/bindings/sound/qcom,sm8250.yaml | 2 +- .../devicetree/bindings/sound/qcom,wcd938x.yaml | 81 +- .../bindings/sound/qcom,wcd939x-sdw.yaml | 69 + .../devicetree/bindings/sound/qcom,wcd939x.yaml | 96 + .../bindings/sound/qcom,wcd93xx-common.yaml | 95 + .../devicetree/bindings/sound/qcom,wsa8840.yaml | 11 +- .../devicetree/bindings/sound/realtek,rt1015.yaml | 41 + Documentation/devicetree/bindings/sound/rt1015.txt | 23 - .../devicetree/bindings/sound/samsung,tm2.yaml | 7 +- drivers/firmware/cirrus/cs_dsp.c | 7 +- drivers/soundwire/Makefile | 2 +- drivers/soundwire/amd_init.c | 235 ++ drivers/soundwire/amd_init.h | 13 + drivers/soundwire/amd_manager.c | 47 +- drivers/soundwire/amd_manager.h | 16 +- drivers/soundwire/dmi-quirks.c | 8 + include/linux/firmware/cirrus/cs_dsp.h | 1 - include/linux/soundwire/sdw_amd.h | 83 +- include/linux/spi/spi.h | 2 - include/sound/cs-amp-lib.h | 66 + include/sound/cs35l56.h | 10 + include/sound/cs42l42.h | 5 +- include/sound/hda-mlink.h | 2 +- include/sound/hda_register.h | 2 + include/sound/soc.h | 4 +- include/sound/sof/dai-amd.h | 7 + include/sound/sof/dai.h | 2 + include/sound/tas2781.h | 1 - include/trace/events/asoc.h | 45 +- include/uapi/sound/intel/avs/tokens.h | 9 + include/uapi/sound/sof/tokens.h | 4 + sound/pci/hda/Kconfig | 2 + sound/pci/hda/cs35l56_hda.c | 39 +- sound/pci/hda/tas2781_hda_i2c.c | 12 - sound/soc/amd/Kconfig | 17 + sound/soc/amd/Makefile | 2 +- sound/soc/amd/acp/Kconfig | 7 + sound/soc/amd/acp/Makefile | 2 + sound/soc/amd/acp/acp-mach-common.c | 6 +- sound/soc/amd/acp/acp-sof-mach.c | 26 +- sound/soc/amd/acp/amd-sdw-acpi.c | 62 + sound/soc/amd/ps/Makefile | 2 +- sound/soc/amd/ps/acp63.h | 91 +- sound/soc/amd/ps/pci-ps.c | 565 +-- sound/soc/amd/ps/ps-mach.c | 2 +- sound/soc/amd/ps/ps-pdm-dma.c | 2 +- sound/soc/amd/ps/ps-sdw-dma.c | 2 +- sound/soc/atmel/mikroe-proto.c | 8 +- sound/soc/codecs/Kconfig | 55 +- sound/soc/codecs/Makefile | 13 + sound/soc/codecs/cs-amp-lib-test.c | 709 ++++ sound/soc/codecs/cs-amp-lib.c | 277 ++ sound/soc/codecs/cs35l56-sdw.c | 20 + sound/soc/codecs/cs35l56-shared.c | 84 + sound/soc/codecs/cs35l56.c | 44 +- sound/soc/codecs/cs42l42.c | 1 - sound/soc/codecs/cs42l43-jack.c | 27 +- sound/soc/codecs/cs42l43-sdw.c | 1 + sound/soc/codecs/cs42l43.c | 82 +- sound/soc/codecs/cs42l43.h | 21 +- sound/soc/codecs/da7213.c | 12 +- sound/soc/codecs/es8326.c | 110 +- sound/soc/codecs/es8326.h | 5 +- sound/soc/codecs/framer-codec.c | 413 +++ sound/soc/codecs/lpass-macro-common.h | 7 + sound/soc/codecs/lpass-tx-macro.c | 693 +++- sound/soc/codecs/lpass-va-macro.c | 57 + sound/soc/codecs/max98363.c | 2 +- sound/soc/codecs/max98373-sdw.c | 2 +- sound/soc/codecs/nau8540.c | 116 +- sound/soc/codecs/nau8540.h | 13 +- sound/soc/codecs/nau8825.c | 12 +- sound/soc/codecs/rt1017-sdca-sdw.c | 2 +- sound/soc/codecs/rt274.c | 2 +- sound/soc/codecs/rt286.c | 2 +- sound/soc/codecs/rt298.c | 2 +- sound/soc/codecs/rt5514-spi.c | 2 +- sound/soc/codecs/rt5514.c | 9 +- sound/soc/codecs/rt5616.c | 9 +- sound/soc/codecs/rt5640.c | 9 +- sound/soc/codecs/rt5645.c | 2 +- sound/soc/codecs/rt5651.c | 2 +- sound/soc/codecs/rt5659.c | 2 +- sound/soc/codecs/rt5660.c | 9 +- sound/soc/codecs/rt5663.c | 2 +- sound/soc/codecs/rt5665.c | 2 +- sound/soc/codecs/rt5668.c | 2 +- sound/soc/codecs/rt5682-i2c.c | 2 +- sound/soc/codecs/rt5682s.c | 2 +- sound/soc/codecs/rt712-sdca-dmic.c | 2 +- sound/soc/codecs/rt712-sdca-sdw.c | 2 +- sound/soc/codecs/rt722-sdca-sdw.c | 2 +- sound/soc/codecs/tas2781-comlib.c | 1 - sound/soc/codecs/tas2781-fmwlib.c | 2 +- sound/soc/codecs/wcd-clsh-v2.h | 1 + sound/soc/codecs/wcd-mbhc-v2.c | 95 +- sound/soc/codecs/wcd-mbhc-v2.h | 3 + sound/soc/codecs/wcd939x-sdw.c | 1551 ++++++++ sound/soc/codecs/wcd939x.c | 3686 ++++++++++++++++++++ sound/soc/codecs/wcd939x.h | 989 ++++++ sound/soc/codecs/wm_adsp.c | 27 +- sound/soc/codecs/wm_adsp.h | 2 + sound/soc/codecs/wsa884x.c | 53 +- sound/soc/fsl/eukrea-tlv320.c | 8 +- sound/soc/fsl/fsl_sai.c | 13 + sound/soc/fsl/p1022_rdk.c | 33 +- sound/soc/intel/atom/sst/sst_ipc.c | 3 +- sound/soc/intel/avs/Makefile | 5 +- sound/soc/intel/avs/apl.c | 58 +- sound/soc/intel/avs/avs.h | 70 +- sound/soc/intel/avs/board_selection.c | 85 + sound/soc/intel/avs/cnl.c | 61 + sound/soc/intel/avs/core.c | 161 +- sound/soc/intel/avs/icl.c | 197 ++ sound/soc/intel/avs/ipc.c | 66 +- sound/soc/intel/avs/loader.c | 2 +- sound/soc/intel/avs/messages.c | 1 + sound/soc/intel/avs/messages.h | 38 +- sound/soc/intel/avs/path.c | 33 + sound/soc/intel/avs/pcm.c | 77 +- sound/soc/intel/avs/registers.h | 21 +- sound/soc/intel/avs/skl.c | 59 +- sound/soc/intel/avs/sysfs.c | 35 + sound/soc/intel/avs/tgl.c | 54 + sound/soc/intel/avs/topology.c | 164 +- sound/soc/intel/avs/topology.h | 13 + sound/soc/intel/boards/Kconfig | 1 + sound/soc/intel/boards/sof_board_helpers.c | 249 +- sound/soc/intel/boards/sof_board_helpers.h | 29 + sound/soc/intel/boards/sof_cs42l42.c | 236 +- sound/soc/intel/boards/sof_rt5682.c | 151 +- sound/soc/intel/boards/sof_sdw.c | 106 +- sound/soc/intel/boards/sof_sdw_common.h | 82 +- sound/soc/intel/boards/sof_sdw_cs42l42.c | 35 +- sound/soc/intel/boards/sof_sdw_cs42l43.c | 25 +- sound/soc/intel/boards/sof_sdw_cs_amp.c | 3 +- sound/soc/intel/boards/sof_sdw_maxim.c | 4 +- sound/soc/intel/boards/sof_sdw_rt5682.c | 35 +- sound/soc/intel/boards/sof_sdw_rt700.c | 35 +- sound/soc/intel/boards/sof_sdw_rt711.c | 19 +- sound/soc/intel/boards/sof_sdw_rt712_sdca.c | 39 +- sound/soc/intel/boards/sof_sdw_rt715.c | 12 +- sound/soc/intel/boards/sof_sdw_rt715_sdca.c | 12 +- sound/soc/intel/boards/sof_sdw_rt_amp.c | 53 +- .../soc/intel/boards/sof_sdw_rt_sdca_jack_common.c | 19 +- sound/soc/intel/boards/sof_ssp_common.h | 8 + sound/soc/intel/catpt/dsp.c | 4 +- sound/soc/intel/common/soc-acpi-intel-adl-match.c | 12 +- sound/soc/intel/common/soc-acpi-intel-jsl-match.c | 10 +- sound/soc/intel/common/soc-acpi-intel-lnl-match.c | 102 + sound/soc/intel/common/soc-acpi-intel-mtl-match.c | 61 +- sound/soc/intel/common/soc-acpi-intel-rpl-match.c | 4 +- sound/soc/intel/common/soc-acpi-intel-tgl-match.c | 6 +- sound/soc/meson/aiu.c | 19 +- sound/soc/meson/aiu.h | 1 - sound/soc/meson/axg-fifo.c | 25 +- sound/soc/meson/axg-fifo.h | 14 +- sound/soc/meson/axg-frddr.c | 13 +- sound/soc/meson/axg-spdifin.c | 6 +- sound/soc/meson/axg-tdm-interface.c | 31 +- sound/soc/meson/axg-toddr.c | 30 +- sound/soc/meson/t9015.c | 20 +- sound/soc/pxa/Kconfig | 3 - sound/soc/qcom/common.c | 2 +- sound/soc/sh/rz-ssi.c | 2 +- sound/soc/soc-core.c | 21 +- sound/soc/soc-dapm.c | 8 +- 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-loader.c | 32 +- sound/soc/sof/amd/acp.c | 232 +- sound/soc/sof/amd/acp.h | 26 +- sound/soc/sof/amd/pci-acp63.c | 7 + sound/soc/sof/core.c | 10 + sound/soc/sof/debug.c | 8 +- sound/soc/sof/fw-file-profile.c | 18 +- sound/soc/sof/imx/imx8.c | 16 + sound/soc/sof/imx/imx8m.c | 10 + sound/soc/sof/imx/imx8ulp.c | 10 + sound/soc/sof/intel/hda-common-ops.c | 1 + sound/soc/sof/intel/hda-dai-ops.c | 51 +- sound/soc/sof/intel/hda-dai.c | 17 +- sound/soc/sof/intel/hda-dsp.c | 5 + sound/soc/sof/intel/hda-stream.c | 9 + sound/soc/sof/intel/hda.c | 80 +- sound/soc/sof/intel/hda.h | 5 + sound/soc/sof/intel/lnl.c | 24 +- sound/soc/sof/ipc3-loader.c | 2 + sound/soc/sof/ipc3-pcm.c | 25 + sound/soc/sof/ipc3-topology.c | 40 + sound/soc/sof/ipc4-pcm.c | 19 +- sound/soc/sof/ipc4-priv.h | 4 + sound/soc/sof/ipc4-topology.c | 28 +- sound/soc/sof/ops.h | 9 + sound/soc/sof/sof-audio.c | 8 +- sound/soc/sof/sof-audio.h | 2 + sound/soc/sof/sof-priv.h | 10 + sound/soc/sof/topology.c | 30 +- sound/soc/ti/j721e-evm.c | 4 +- sound/soc/ti/omap-hdmi.c | 10 +- 217 files changed, 13085 insertions(+), 2218 deletions(-) create mode 100644 Documentation/ABI/testing/sysfs-bus-pci-devices-avs create mode 100644 Documentation/devicetree/bindings/sound/atmel,asoc-wm8904.yaml create mode 100644 Documentation/devicetree/bindings/sound/atmel,sam9x5-wm8731-audio.yaml delete mode 100644 Documentation/devicetree/bindings/sound/atmel-sam9x5-wm8731-audio.txt delete mode 100644 Documentation/devicetree/bindings/sound/atmel-wm8904.txt delete mode 100644 Documentation/devicetree/bindings/sound/fsl,asrc.txt create mode 100644 Documentation/devicetree/bindings/sound/fsl,imx-asrc.yaml create mode 100644 Documentation/devicetree/bindings/sound/qcom,wcd939x-sdw.yaml create mode 100644 Documentation/devicetree/bindings/sound/qcom,wcd939x.yaml create mode 100644 Documentation/devicetree/bindings/sound/qcom,wcd93xx-common.yaml create mode 100644 Documentation/devicetree/bindings/sound/realtek,rt1015.yaml delete mode 100644 Documentation/devicetree/bindings/sound/rt1015.txt create mode 100644 drivers/soundwire/amd_init.c create mode 100644 drivers/soundwire/amd_init.h create mode 100644 include/sound/cs-amp-lib.h create mode 100644 sound/soc/amd/acp/amd-sdw-acpi.c create mode 100644 sound/soc/codecs/cs-amp-lib-test.c create mode 100644 sound/soc/codecs/cs-amp-lib.c create mode 100644 sound/soc/codecs/framer-codec.c create mode 100644 sound/soc/codecs/wcd939x-sdw.c create mode 100644 sound/soc/codecs/wcd939x.c create mode 100644 sound/soc/codecs/wcd939x.h create mode 100644 sound/soc/intel/avs/cnl.c create mode 100644 sound/soc/intel/avs/icl.c create mode 100644 sound/soc/intel/avs/sysfs.c create mode 100644 sound/soc/intel/avs/tgl.c

2 1

Strange trigger_htstamp after snd_pcm_drop
by GitHub issues - reopened 11 Mar '24

11 Mar '24

alsa-project/alsa-lib issue #387 was reopened from z-s-e: I have conceptually the following situation (actual code can be found [here](https://gitlab.com/z-s-e/alpe/-/blob/main/src/alpe_alsa.cpp?ref_type=…): ``` // sound is currently playing snd_pcm_status(pcm, pcm_status); snd_pcm_status_get_htstamp(pcm_status, &ts1); snd_pcm_drop(pcm); snd_pcm_status(pcm, pcm_status); snd_pcm_status_get_trigger_htstamp(status, &ts2); ``` On my hardware I get a `ts2` timestamp that is earlier than `ts1`. From my understanding that should be impossible, as `ts2` should contain the timestamp of when the drop happened, which should obviously be after `ts1`. Is that a bug or do I misunderstand the meaning of those timestamps? Issue URL : https://github.com/alsa-project/alsa-lib/issues/387 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

sof-soundwire: rt1308: Fix single amp configuration
by GitHub pull_request - opened 11 Mar '24

11 Mar '24

alsa-project/alsa-ucm-conf pull request #402 was opened from ujfalusi: If the device have single rt1308 amp then the codec's Channel Select should be set to "LR" and not to "LL". "LL" is only valid if we have 2 amps, in that case one is the Left and the other is the Right channel. This fixes for example Dell Latitude 9520 which only played on the Left speaker, Right is silent. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/402 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/402.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH v2 0/4] ASoC: fsl: Support register and unregister rpmsg sound card through remoteproc
by Chancel Liu 11 Mar '24

11 Mar '24

echo /lib/firmware/fw.elf > /sys/class/remoteproc/remoteproc0/firmware (A) echo start > /sys/class/remoteproc/remoteproc0/state (B) echo stop > /sys/class/remoteproc/remoteproc0/state The rpmsg sound card is registered in (A) and unregistered in (B). After "start", imx-audio-rpmsg registers devices for ASoC platform driver and machine driver. Then sound card is registered. After "stop", imx-audio-rpmsg unregisters devices for ASoC platform driver and machine driver. Then sound card is unregistered. changes in v2 - Fix build errors reported by kernel test robot Chancel Liu (4): ASoC: fsl: imx_pcm_rpmsg: Register component with rpmsg channel name ASoC: fsl: imx-audio-rpmsg: Register device with rpmsg channel name ASoC: fsl: Let imx-audio-rpmsg register platform device for card ASoC: fsl: imx-rpmsg: Update to correct DT node sound/soc/fsl/fsl_rpmsg.c | 11 ----------- sound/soc/fsl/imx-audio-rpmsg.c | 21 ++++++++++++++++++--- sound/soc/fsl/imx-pcm-rpmsg.c | 11 ++++++++--- sound/soc/fsl/imx-rpmsg.c | 21 ++++++++++++++++++--- 4 files changed, 44 insertions(+), 20 deletions(-) -- 2.43.0

2 8

[PATCH 0/5] ALSA: hda/tas2781: restore power state after system_resume
by Gergo Koteles 11 Mar '24

11 Mar '24

This series makes some simplifications, speeding up the start of the playback and restoring the power state of the amplifiers after system_resume. Gergo Koteles (5): ALSA: hda/tas2781: use dev_dbg in system_resume ALSA: hda/tas2781: add lock to system_suspend ALSA: hda/tas2781: do not reset cur_* values in runtime_suspend ALSA: hda/tas2781: do not call pm_runtime_force_* in system_resume/suspend ALSA: hda/tas2781: restore power state after system_resume sound/pci/hda/tas2781_hda_i2c.c | 35 ++++++++++++++++----------------- 1 file changed, 17 insertions(+), 18 deletions(-) base-commit: 90d35da658da8cff0d4ecbb5113f5fac9d00eb72 -- 2.44.0

2 6

[PATCH v13 00/16] Add audio support in v4l2 framework
by Shengjiu Wang 11 Mar '24

11 Mar '24

Audio signal processing also has the requirement for memory to memory similar as Video. This asrc memory to memory (memory ->asrc->memory) case is a non real time use case. User fills the input buffer to the asrc module, after conversion, then asrc sends back the output buffer to user. So it is not a traditional ALSA playback and capture case. It is a specific use case, there is no reference in current kernel. v4l2 memory to memory is the closed implementation, v4l2 current support video, image, radio, tuner, touch devices, so it is not complicated to add support for this specific audio case. Because we had implemented the "memory -> asrc ->i2s device-> codec" use case in ALSA. Now the "memory->asrc->memory" needs to reuse the code in asrc driver, so the first 3 patches is for refining the code to make it can be shared by the "memory->asrc->memory" driver. The main change is in the v4l2 side, A /dev/vl4-audioX will be created, user applications only use the ioctl of v4l2 framework. Other change is to add memory to memory support for two kinds of i.MX ASRC module. changes in v13 - change 'pixelformat' to 'audioformat' in dev-audio-mem2mem.rst - add more description for clock drift in ext-ctrls-audio-m2m.rst - Add "media: v4l2-ctrls: add support for fraction_bits" from Hans to avoid build issue for kernel test robot changes in v12 - minor changes according to comments - drop min_buffers_needed = 1 and V4L2_CTRL_FLAG_UPDATE flag - drop bus_info changes in v11 - add add-fixed-point-test-controls in vivid. - add v4l2_ctrl_fp_compose() helper function for min and max changes in v10 - remove FIXED_POINT type - change code base on media: v4l2-ctrls: add support for fraction_bits - fix issue reported by kernel test robot - remove module_alias changes in v9: - add MEDIA_ENT_F_PROC_AUDIO_RESAMPLER. - add MEDIA_INTF_T_V4L_AUDIO - add media controller support - refine the vim2m-audio to support 8k<->16k conversion. changes in v8: - refine V4L2_CAP_AUDIO_M2M to be 0x00000008 - update doc for FIXED_POINT - address comments for imx-asrc changes in v7: - add acked-by from Mark - separate commit for fixed point, m2m audio class, audio rate controls - use INTEGER_MENU for rate, FIXED_POINT for rate offset - remove used fmts - address other comments for Hans changes in v6: - use m2m_prepare/m2m_unprepare/m2m_start/m2m_stop to replace m2m_start_part_one/m2m_stop_part_one, m2m_start_part_two/m2m_stop_part_two. - change V4L2_CTRL_TYPE_ASRC_RATE to V4L2_CTRL_TYPE_FIXED_POINT - fix warning by kernel test rebot - remove some unused format V4L2_AUDIO_FMT_XX - Get SNDRV_PCM_FORMAT from V4L2_AUDIO_FMT in driver. - rename audm2m to viaudm2m. changes in v5: - remove V4L2_AUDIO_FMT_LPCM - define audio pixel format like V4L2_AUDIO_FMT_S8... - remove rate and format in struct v4l2_audio_format. - Add V4L2_CID_ASRC_SOURCE_RATE and V4L2_CID_ASRC_DEST_RATE controls - updata document accordingly. changes in v4: - update document style - separate V4L2_AUDIO_FMT_LPCM and V4L2_CAP_AUDIO_M2M in separate commit changes in v3: - Modify documents for adding audio m2m support - Add audio virtual m2m driver - Defined V4L2_AUDIO_FMT_LPCM format type for audio. - Defined V4L2_CAP_AUDIO_M2M capability type for audio m2m case. - with modification in v4l-utils, pass v4l2-compliance test. changes in v2: - decouple the implementation in v4l2 and ALSA - implement the memory to memory driver as a platfrom driver and move it to driver/media - move fsl_asrc_common.h to include/sound folder Hans Verkuil (1): media: v4l2-ctrls: add support for fraction_bits Shengjiu Wang (15): ASoC: fsl_asrc: define functions for memory to memory usage ASoC: fsl_easrc: define functions for memory to memory usage ASoC: fsl_asrc: move fsl_asrc_common.h to include/sound ASoC: fsl_asrc: register m2m platform device ASoC: fsl_easrc: register m2m platform device media: uapi: Add V4L2_CAP_AUDIO_M2M capability flag media: v4l2: Add audio capture and output support media: uapi: Define audio sample format fourcc type media: uapi: Add V4L2_CTRL_CLASS_M2M_AUDIO media: uapi: Add audio rate controls support media: uapi: Declare interface types for Audio media: uapi: Add an entity type for audio resampler media: vivid: add fixed point test controls media: imx-asrc: Add memory to memory driver media: vim2m-audio: add virtual driver for audio memory to memory .../media/mediactl/media-types.rst | 11 + .../userspace-api/media/v4l/buffer.rst | 6 + .../userspace-api/media/v4l/common.rst | 1 + .../media/v4l/dev-audio-mem2mem.rst | 71 + .../userspace-api/media/v4l/devices.rst | 1 + .../media/v4l/ext-ctrls-audio-m2m.rst | 59 + .../userspace-api/media/v4l/pixfmt-audio.rst | 87 ++ .../userspace-api/media/v4l/pixfmt.rst | 1 + .../media/v4l/vidioc-enum-fmt.rst | 2 + .../media/v4l/vidioc-g-ext-ctrls.rst | 4 + .../userspace-api/media/v4l/vidioc-g-fmt.rst | 4 + .../media/v4l/vidioc-querycap.rst | 3 + .../media/v4l/vidioc-queryctrl.rst | 11 +- .../media/videodev2.h.rst.exceptions | 3 + .../media/common/videobuf2/videobuf2-v4l2.c | 4 + drivers/media/platform/nxp/Kconfig | 13 + drivers/media/platform/nxp/Makefile | 1 + drivers/media/platform/nxp/imx-asrc.c | 1256 +++++++++++++++++ drivers/media/test-drivers/Kconfig | 10 + drivers/media/test-drivers/Makefile | 1 + drivers/media/test-drivers/vim2m-audio.c | 793 +++++++++++ drivers/media/test-drivers/vivid/vivid-core.h | 2 + .../media/test-drivers/vivid/vivid-ctrls.c | 26 + drivers/media/v4l2-core/v4l2-compat-ioctl32.c | 9 + drivers/media/v4l2-core/v4l2-ctrls-api.c | 1 + drivers/media/v4l2-core/v4l2-ctrls-core.c | 93 +- drivers/media/v4l2-core/v4l2-ctrls-defs.c | 10 + drivers/media/v4l2-core/v4l2-dev.c | 21 + drivers/media/v4l2-core/v4l2-ioctl.c | 66 + drivers/media/v4l2-core/v4l2-mem2mem.c | 13 +- include/media/v4l2-ctrls.h | 13 +- include/media/v4l2-dev.h | 2 + include/media/v4l2-ioctl.h | 34 + .../fsl => include/sound}/fsl_asrc_common.h | 60 + include/uapi/linux/media.h | 2 + include/uapi/linux/v4l2-controls.h | 9 + include/uapi/linux/videodev2.h | 44 +- sound/soc/fsl/fsl_asrc.c | 144 ++ sound/soc/fsl/fsl_asrc.h | 4 +- sound/soc/fsl/fsl_asrc_dma.c | 2 +- sound/soc/fsl/fsl_easrc.c | 233 +++ sound/soc/fsl/fsl_easrc.h | 6 +- 42 files changed, 3109 insertions(+), 27 deletions(-) create mode 100644 Documentation/userspace-api/media/v4l/dev-audio-mem2mem.rst create mode 100644 Documentation/userspace-api/media/v4l/ext-ctrls-audio-m2m.rst create mode 100644 Documentation/userspace-api/media/v4l/pixfmt-audio.rst create mode 100644 drivers/media/platform/nxp/imx-asrc.c create mode 100644 drivers/media/test-drivers/vim2m-audio.c rename {sound/soc/fsl => include/sound}/fsl_asrc_common.h (60%) -- 2.34.1

4 26

Incompatible syntax 6 in Valve-Jupiter-1.conf
by GitHub issues - opened 11 Mar '24

11 Mar '24

alsa-project/alsa-ucm-conf issue #401 was opened from Richard-Haes-Ellis: When installing the missing ucm2 files for the Steam Deck I get this error on systemctl status alsa-restore: ``` systemctl status alsa-restore ● alsa-restore.service - Save/Restore Sound Card State Loaded: loaded (/lib/systemd/system/alsa-restore.service; static) Active: active (exited) since Mon 2024-03-11 01:06:18 CET; 15min ago Docs: man:alsactl(1) Main PID: 608 (code=exited, status=99) CPU: 13ms mar 11 01:06:18 steam-decado alsactl[608]: /usr/sbin/alsactl: load_state:1689: Cannot open /var/lib/alsa/asound.state for reading: > mar 11 01:06:18 steam-decado alsactl[608]: alsa-lib parser.c:2021:(parse_master_file) Incompatible syntax 6 in HDA-Intel.conf mar 11 01:06:18 steam-decado alsactl[608]: alsa-lib main.c:1412:(snd_use_case_mgr_open) error: failed to import hw:0 use case confi> mar 11 01:06:18 steam-decado alsactl[608]: Found hardware: "HDA-Intel" "ATI R6xx HDMI" "HDA:1002aa01,00aa0100,00100800" "0x1e44" "0> mar 11 01:06:18 steam-decado alsactl[608]: Hardware is initialized using a generic method mar 11 01:06:18 steam-decado alsactl[608]: alsa-lib parser.c:2021:(parse_master_file) Incompatible syntax 6 in Valve-Jupiter-1.conf mar 11 01:06:18 steam-decado alsactl[608]: alsa-lib main.c:1412:(snd_use_case_mgr_open) error: failed to import hw:1 use case confi> mar 11 01:06:18 steam-decado alsactl[608]: Found hardware: "acp5x" "" "" "" "" mar 11 01:06:18 steam-decado alsactl[608]: Hardware is initialized using a generic method mar 11 01:06:18 steam-decado systemd[1]: Finished Save/Restore Sound Card State. ``` looking at my alsa packages these are the versions installed: ``` dpkg -l | grep alsa ii alsa-base 1.0.25+dfsg-0ubuntu7 all ALSA driver configuration files ii alsa-topology-conf 1.2.5.1-2 all ALSA topology configuration files ii alsa-ucm-conf 1.2.6.3-1ubuntu1.11 all ALSA Use Case Manager configuration files ii alsa-utils 1.2.6-1ubuntu1 amd64 Utilities for configuring and using ALSA ii gstreamer1.0-alsa:amd64 1.20.1-1ubuntu0.1 amd64 GStreamer plugin for ALSA ii pipewire-alsa:amd64 1.0.3-2~ubuntu22.04 amd64 PipeWire ALSA plugin ii pipewire-audio-client-libraries 1.0.3-2~ubuntu22.04 all transitional package for pipewire-alsa and pipewire-jack ``` I'm running Ubuntu 22.04.4 LTS on Steam Deck hardware Valve Jupiter Any ideas why it would't work? Thanks in advanced. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/401 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

ASUS VivoBook Pro 14: platform::micmute led not working
by GitHub issues - opened 10 Mar '24

10 Mar '24

alsa-project/alsa-ucm-conf issue #400 was opened from Dinolek: There is `/sys/class/leds/platform::micmute` from `asus-nb-wmi`, but `snd_ctl_led` isn't loaded. Microphone isn't connected to acp, so `snd_rn_pci_acp3x` isn't getting loaded. I can get led switching working if I attach 'Capture Switch' ``` modprobe snd_ctl_led echo 'Capture Switch' > /sys/class/sound/ctl-led/mic/card1/attach ``` [alsa-info](http://alsa-project.org/db/?f=05aff573aedf6ddd6954ca739d8de8f827… Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/400 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

alsa-plugins-1.2.7.1 fails to compile against latest ffmpeg git
by GitHub issues - opened 10 Mar '24

10 Mar '24

alsa-project/alsa-plugins issue #57 was opened from jospezial: ``` libtool: compile: x86_64-pc-linux-gnu-gcc -m32 -mfpmath=sse -DHAVE_CONFIG_H -I. -I/var/tmp/portage/media-plugins/alsa-plugins-1.2.7.1-r1/work/alsa-plugins-1.2.7.1/a52 -I.. -DNDEBUG -Wall -g -march=native -mtune=native -O2 -pipe -c /var/tmp/portage/media-plugins/alsa-plugins-1.2.7.1-r1/work/alsa-plugins-1.2.7.1/a52/pcm_a52.c -fPIC -DPIC -o .libs/pcm_a52.o /var/tmp/portage/media-plugins/alsa-plugins-1.2.7.1-r1/work/alsa-plugins-1.2.7.1/a52/pcm_a52.c: In function 'a52_free': /var/tmp/portage/media-plugins/alsa-plugins-1.2.7.1-r1/work/alsa-plugins-1.2.7.1/a52/pcm_a52.c:631:17: warning: 'avcodec_close' is deprecated [-Wdeprecated-declarations] 631 | avcodec_close(rec->avctx); | ^~~~~~~~~~~~~ In file included from /var/tmp/portage/media-plugins/alsa-plugins-1.2.7.1-r1/work/alsa-plugins-1.2.7.1/a52/pcm_a52.c:28: /usr/include/libavcodec/avcodec.h:2374:5: note: declared here 2374 | int avcodec_close(AVCodecContext *avctx); | ^~~~~~~~~~~~~ /var/tmp/portage/media-plugins/alsa-plugins-1.2.7.1-r1/work/alsa-plugins-1.2.7.1/a52/pcm_a52.c: In function 'set_channel_layout': /var/tmp/portage/media-plugins/alsa-plugins-1.2.7.1-r1/work/alsa-plugins-1.2.7.1/a52/pcm_a52.c:672:29: error: 'AVCodecContext' has no member named 'channel_layout'; did you mean 'ch_layout'? 672 | rec->avctx->channel_layout = AV_CH_LAYOUT_STEREO; | ^~~~~~~~~~~~~~ | ch_layout /var/tmp/portage/media-plugins/alsa-plugins-1.2.7.1-r1/work/alsa-plugins-1.2.7.1/a52/pcm_a52.c:675:29: error: 'AVCodecContext' has no member named 'channel_layout'; did you mean 'ch_layout'? 675 | rec->avctx->channel_layout = AV_CH_LAYOUT_QUAD; | ^~~~~~~~~~~~~~ | ch_layout /var/tmp/portage/media-plugins/alsa-plugins-1.2.7.1-r1/work/alsa-plugins-1.2.7.1/a52/pcm_a52.c:678:29: error: 'AVCodecContext' has no member named 'channel_layout'; did you mean 'ch_layout'? 678 | rec->avctx->channel_layout = AV_CH_LAYOUT_5POINT1; | ^~~~~~~~~~~~~~ | ch_layout /var/tmp/portage/media-plugins/alsa-plugins-1.2.7.1-r1/work/alsa-plugins-1.2.7.1/a52/pcm_a52.c: In function 'alloc_input_buffer': /var/tmp/portage/media-plugins/alsa-plugins-1.2.7.1-r1/work/alsa-plugins-1.2.7.1/a52/pcm_a52.c:698:19: error: 'AVFrame' has no member named 'channels' 698 | rec->frame->channels = rec->avctx->channels; | ^~ /var/tmp/portage/media-plugins/alsa-plugins-1.2.7.1-r1/work/alsa-plugins-1.2.7.1/a52/pcm_a52.c:698:42: error: 'AVCodecContext' has no member named 'channels' 698 | rec->frame->channels = rec->avctx->channels; | ^~ /var/tmp/portage/media-plugins/alsa-plugins-1.2.7.1-r1/work/alsa-plugins-1.2.7.1/a52/pcm_a52.c:699:21: error: 'AVFrame' has no member named 'channel_layout'; did you mean 'ch_layout'? 699 | rec->frame->channel_layout = rec->avctx->channel_layout; | ^~~~~~~~~~~~~~ | ch_layout /var/tmp/portage/media-plugins/alsa-plugins-1.2.7.1-r1/work/alsa-plugins-1.2.7.1/a52/pcm_a52.c:699:50: error: 'AVCodecContext' has no member named 'channel_layout'; did you mean 'ch_layout'? 699 | rec->frame->channel_layout = rec->avctx->channel_layout; | ^~~~~~~~~~~~~~ | ch_layout /var/tmp/portage/media-plugins/alsa-plugins-1.2.7.1-r1/work/alsa-plugins-1.2.7.1/a52/pcm_a52.c: In function 'a52_prepare': /var/tmp/portage/media-plugins/alsa-plugins-1.2.7.1-r1/work/alsa-plugins-1.2.7.1/a52/pcm_a52.c:734:19: error: 'AVCodecContext' has no member named 'channels' 734 | rec->avctx->channels = io->channels; | ^~ /var/tmp/portage/media-plugins/alsa-plugins-1.2.7.1-r1/work/alsa-plugins-1.2.7.1/a52/pcm_a52.c: In function '_snd_pcm_a52_open': /var/tmp/portage/media-plugins/alsa-plugins-1.2.7.1-r1/work/alsa-plugins-1.2.7.1/a52/pcm_a52.c:1124:28: warning: assignment discards 'const' qualifier from pointer target type [-Wdiscarded-qualifiers] 1124 | rec->codec = avcodec_find_encoder_by_name(avcodec); | ^ /var/tmp/portage/media-plugins/alsa-plugins-1.2.7.1-r1/work/alsa-plugins-1.2.7.1/a52/pcm_a52.c:1126:28: warning: assignment discards 'const' qualifier from pointer target type [-Wdiscarded-qualifiers] 1126 | rec->codec = avcodec_find_encoder_by_name("ac3_fixed"); | ^ /var/tmp/portage/media-plugins/alsa-plugins-1.2.7.1-r1/work/alsa-plugins-1.2.7.1/a52/pcm_a52.c:1128:36: warning: assignment discards 'const' qualifier from pointer target type [-Wdiscarded-qualifiers] 1128 | rec->codec = avcodec_find_encoder_by_name("ac3"); | ^ /var/tmp/portage/media-plugins/alsa-plugins-1.2.7.1-r1/work/alsa-plugins-1.2.7.1/a52/pcm_a52.c:1131:28: warning: assignment discards 'const' qualifier from pointer target type [-Wdiscarded-qualifiers] 1131 | rec->codec = avcodec_find_encoder(AV_CODEC_ID_AC3); | ^ make[2]: *** [Makefile:469: pcm_a52.lo] Error 1 make[2]: Leaving directory '/var/tmp/portage/media-plugins/alsa-plugins-1.2.7.1-r1/work/alsa-plugins-1.2.7.1-abi_x86_32.x86/a52' make[1]: *** [Makefile:445: all-recursive] Error 1 make[1]: Leaving directory '/var/tmp/portage/media-plugins/alsa-plugins-1.2.7.1-r1/work/alsa-plugins-1.2.7.1-abi_x86_32.x86' make: *** [Makefile:377: all] Error 2 * ERROR: media-plugins/alsa-plugins-1.2.7.1-r1::gentoo failed (compile phase): * emake failed ``` Issue URL : https://github.com/alsa-project/alsa-plugins/issues/57 Repository URL: https://github.com/alsa-project/alsa-plugins

1 0

PinePhone HiFi.conf regresion
by GitHub issues - opened 10 Mar '24

10 Mar '24

alsa-project/alsa-ucm-conf issue #399 was opened from nesnera: No Audio Playback with current HiFi.conf. As [Philip Müller](https://gitlab.manjaro.org/philm) [comments](https://gitlab.manjaro.org/manjaro-arm/issues/pinephone/phosh/-/i…, it looks like a regression. [This version](https://gitlab.manjaro.org/manjaro-arm/issues/pinephone/phosh/-/is… is functional. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/399 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

`seqtest[23].py` segfault with Python 3.12
by GitHub issues - opened 09 Mar '24

09 Mar '24

alsa-project/alsa-python issue #8 was opened from mgorny: ``` $ PYTHONPATH=build/lib.linux-x86_64-cpython-312/ gdb --args python test/seqtest2.py […] Starting program: /tmp/alsa-python/.venv/bin/python test/seqtest2.py [Thread debugging using libthread_db enabled] Using host libthread_db library "/lib64/libthread_db.so.1". GC DEBUG: init. ALSA lib /tmp/portage/media-libs/alsa-lib-1.2.11/work/alsa-lib-1.2.11/src/seq/seq_hw.c:528:(snd_seq_hw_open) open /dev/snd/seq failed: No such file or directory Traceback (most recent call last): File "/tmp/alsa-python/test/seqtest2.py", line 19, in <module> seq = alsaseq.Sequencer() ^^^^^^^^^^^^^^^^^^^ alsaseq.SequencerError: Failed to create sequencer: No such file or directory gc: collecting generation 2... gc: objects in each generation: 321 4946 0 gc: objects in permanent generation: 375 gc: done, 0 unreachable, 0 uncollectable, 0.0004s elapsed gc: collecting generation 2... gc: objects in each generation: 74 0 5052 gc: objects in permanent generation: 375 gc: collectable <dict 0x7ffff7c951c0> gc: collectable <module 0x7ffff7c72a70> gc: collectable <dict 0x7ffff7c95580> gc: collectable <builtin_function_or_method 0x7ffff7c72b10> gc: collectable <builtin_function_or_method 0x7ffff7c72bb0> gc: collectable <builtin_function_or_method 0x7ffff7c72c00> gc: collectable <builtin_function_or_method 0x7ffff7c72c50> gc: collectable <builtin_function_or_method 0x7ffff7c72ca0> gc: collectable <builtin_function_or_method 0x7ffff7c72cf0> gc: collectable <builtin_function_or_method 0x7ffff7c72d40> gc: collectable <builtin_function_or_method 0x7ffff7c72d90> gc: collectable <builtin_function_or_method 0x7ffff7c72de0> gc: collectable <builtin_function_or_method 0x7ffff7c72e30> gc: collectable <builtin_function_or_method 0x7ffff7c72e80> gc: collectable <builtin_function_or_method 0x7ffff7c72ed0> gc: collectable <builtin_function_or_method 0x7ffff7c72f20> gc: collectable <builtin_function_or_method 0x7ffff7c72f70> gc: collectable <builtin_function_or_method 0x7ffff7c72fc0> gc: collectable <builtin_function_or_method 0x7ffff7c73010> gc: collectable <builtin_function_or_method 0x7ffff7c73060> gc: collectable <builtin_function_or_method 0x7ffff7c730b0> gc: collectable <builtin_function_or_method 0x7ffff7c73100> gc: collectable <builtin_function_or_method 0x7ffff7c73150> gc: collectable <builtin_function_or_method 0x7ffff7c731a0> gc: collectable <builtin_function_or_method 0x7ffff7c731f0> gc: collectable <builtin_function_or_method 0x7ffff7c73240> gc: collectable <builtin_function_or_method 0x7ffff7c73290> gc: collectable <builtin_function_or_method 0x7ffff7c732e0> gc: collectable <type 0x555555c83830> gc: collectable <tuple 0x7ffff7c93fc0> gc: collectable <dict 0x7ffff7c95d80> gc: collectable <wrapper_descriptor 0x7ffff7c733d0> gc: collectable <method_descriptor 0x7ffff7c73420> gc: collectable <member_descriptor 0x7ffff7c95dc0> gc: collectable <member_descriptor 0x7ffff7c95e00> gc: collectable <member_descriptor 0x7ffff7c95e40> gc: collectable <member_descriptor 0x7ffff7c95e80> gc: collectable <member_descriptor 0x7ffff7c95ec0> gc: collectable <member_descriptor 0x7ffff7c95f00> gc: collectable <member_descriptor 0x7ffff7c95f40> gc: collectable <member_descriptor 0x7ffff7c95f80> gc: collectable <member_descriptor 0x7ffff7c95fc0> gc: collectable <member_descriptor 0x7ffff7c96040> gc: collectable <member_descriptor 0x7ffff7c96080> gc: collectable <list 0x7ffff7c6ed80> gc: collectable <type 0x555555c83dc0> gc: collectable <tuple 0x7ffff7c72ac0> gc: collectable <dict 0x7ffff7c95480> gc: collectable <type 0x555555c848b0> gc: collectable <tuple 0x7ffff7c961c0> gc: collectable <function 0x7ffff7c99080> gc: collectable <function 0x7ffff7c998a0> gc: collectable <function 0x7ffff7c99940> gc: collectable <function 0x7ffff7c999e0> gc: collectable <function 0x7ffff7c99a80> gc: collectable <function 0x7ffff7c99b20> gc: collectable <function 0x7ffff7c99bc0> gc: collectable <function 0x7ffff7c99c60> gc: collectable <function 0x7ffff7c99d00> gc: collectable <function 0x7ffff7c99da0> gc: collectable <function 0x7ffff7c99e40> gc: collectable <function 0x7ffff7c99ee0> gc: collectable <function 0x7ffff7c99f80> gc: collectable <function 0x7ffff7c9a020> gc: collectable <ModuleSpec 0x7ffff7c7fb60> gc: collectable <ModuleSpec 0x7ffff7c7fc50> gc: collectable <builtin_function_or_method 0x7ffff7c72b60> gc: collectable <builtin_function_or_method 0x7ffff7c73380> gc: collectable <getset_descriptor 0x7ffff7c95640> gc: collectable <dict 0x7ffff7c96200> gc: collectable <tuple 0x7ffff7c7fbe0> gc: collectable <types.SimpleNamespace 0x7ffff7c7fb80> gc: collectable <list 0x7ffff7c6cac0> gc: collectable <list 0x7ffff7c95680> gc: collectable <function 0x7ffff7c991c0> gc: collectable <function 0x7ffff7c99260> gc: collectable <function 0x7ffff7c99300> gc: collectable <function 0x7ffff7c993a0> gc: collectable <function 0x7ffff7c99440> gc: collectable <function 0x7ffff7c994e0> gc: collectable <function 0x7ffff7c99580> gc: collectable <function 0x7ffff7c99620> gc: collectable <function 0x7ffff7c996c0> gc: collectable <function 0x7ffff7c99760> gc: collectable <function 0x7ffff7c99800> gc: collectable <module 0x7ffff7ca8f40> gc: collectable <dict 0x7ffff7c96400> gc: collectable <builtin_function_or_method 0x7ffff7ca8ef0> gc: collectable <builtin_function_or_method 0x7ffff7ca8fe0> gc: collectable <builtin_function_or_method 0x7ffff7ca9030> gc: collectable <builtin_function_or_method 0x7ffff7ca9080> gc: collectable <builtin_function_or_method 0x7ffff7ca90d0> gc: collectable <builtin_function_or_method 0x7ffff7ca9120> gc: collectable <builtin_function_or_method 0x7ffff7ca9170> gc: collectable <builtin_function_or_method 0x7ffff7ca91c0> gc: collectable <builtin_function_or_method 0x7ffff7ca9210> gc: collectable <builtin_function_or_method 0x7ffff7ca9260> gc: collectable <builtin_function_or_method 0x7ffff7ca92b0> gc: collectable <builtin_function_or_method 0x7ffff7ca9300> gc: collectable <builtin_function_or_method 0x7ffff7ca9350> gc: collectable <builtin_function_or_method 0x7ffff7ca93a0> gc: collectable <builtin_function_or_method 0x7ffff7ca93f0> gc: collectable <builtin_function_or_method 0x7ffff7ca9440> gc: collectable <builtin_function_or_method 0x7ffff7ca9490> gc: collectable <builtin_function_or_method 0x7ffff7ca94e0> gc: collectable <type 0x555555c95610> gc: collectable <tuple 0x7ffff7ca9530> gc: collectable <dict 0x7ffff7cb8900> gc: collectable <type 0x555555c95ff0> gc: collectable <tuple 0x7ffff7ca6880> gc: collectable <dict 0x7ffff7cb8880> gc: collectable <wrapper_descriptor 0x7ffff7ca9620> gc: collectable <method_descriptor 0x7ffff7ca9670> gc: collectable <member_descriptor 0x7ffff7cb8d00> gc: collectable <member_descriptor 0x7ffff7cb8d40> gc: collectable <member_descriptor 0x7ffff7cb8d80> gc: collectable <member_descriptor 0x7ffff7cb8e00> gc: collectable <member_descriptor 0x7ffff7cb8e40> gc: collectable <member_descriptor 0x7ffff7cb8e80> gc: collectable <member_descriptor 0x7ffff7cb8ec0> gc: collectable <module 0x7ffff7ca9760> gc: collectable <dict 0x7ffff7c953c0> gc: collectable <list 0x7ffff7cb90c0> gc: collectable <module 0x7ffff7ca98a0> gc: collectable <dict 0x7ffff7cb8980> gc: collectable <function 0x7ffff7cb2480> gc: collectable <type 0x555555c964e0> gc: collectable <tuple 0x7ffff7cb9000> gc: collectable <type 0x555555c77ca0> gc: collectable <tuple 0x7ffff7cb9380> gc: collectable <type 0x555555c78080> gc: collectable <tuple 0x7ffff7cb9480> gc: collectable <dict 0x7ffff7cb9440> gc: collectable <getset_descriptor 0x7ffff7cb9500> gc: collectable <module 0x7ffff7ca99e0> gc: collectable <builtin_function_or_method 0x7ffff7ca9a80> gc: collectable <dict 0x7ffff7cb95c0> gc: collectable <builtin_function_or_method 0x7ffff7ca9a30> gc: collectable <builtin_function_or_method 0x7ffff7ca9ad0> gc: collectable <builtin_function_or_method 0x7ffff7ca9b20> gc: collectable <builtin_function_or_method 0x7ffff7ca9b70> gc: collectable <builtin_function_or_method 0x7ffff7ca9bc0> gc: collectable <builtin_function_or_method 0x7ffff7ca9c10> gc: collectable <builtin_function_or_method 0x7ffff7ca9c60> gc: collectable <type 0x555555c78460> gc: collectable <tuple 0x7ffff7cb81c0> gc: collectable <dict 0x7ffff7cb97c0> gc: collectable <type 0x555555c78840> gc: collectable <tuple 0x7ffff7cb9800> gc: collectable <function 0x7ffff7cb2520> gc: collectable <ABCMeta 0x555555c78ec0> gc: collectable <tuple 0x7ffff7cb8200> gc: collectable <dict 0x7ffff7cb98c0> gc: collectable <_abc._abc_data 0x7ffff7cb9980> gc: collectable <ABCMeta 0x555555c798c0> gc: collectable <tuple 0x7ffff7cb9100> gc: collectable <dict 0x7ffff7cb9900> gc: collectable <_abc._abc_data 0x7ffff7cb9140> gc: collectable <ABCMeta 0x555555c7a180> gc: collectable <tuple 0x7ffff7ca9e40> gc: collectable <dict 0x7ffff7cb9840> gc: collectable <_abc._abc_data 0x7ffff7cb9a00> gc: collectable <ABCMeta 0x555555c7a560> gc: collectable <tuple 0x7ffff7ca9e90> gc: collectable <dict 0x7ffff7cb99c0> gc: collectable <_abc._abc_data 0x7ffff7cb9ac0> gc: collectable <ABCMeta 0x555555c7a940> gc: collectable <tuple 0x7ffff7ca9f30> gc: collectable <dict 0x7ffff7cb9a40> gc: collectable <_abc._abc_data 0x7ffff7cb9b80> gc: collectable <set 0x7ffff7bd5540> gc: collectable <weakref.ReferenceType 0x7ffff7caa070> gc: collectable <set 0x7ffff7bd5620> gc: collectable <weakref.ReferenceType 0x7ffff7caa160> gc: collectable <set 0x7ffff7bd5700> gc: collectable <set 0x7ffff7bd57e0> gc: collectable <weakref.ReferenceType 0x7ffff7caa390> gc: collectable <weakref.ReferenceType 0x7ffff7caa430> gc: collectable <weakref.ReferenceType 0x7ffff7caa4d0> gc: collectable <weakref.ReferenceType 0x7ffff7caa570> gc: collectable <weakref.ReferenceType 0x7ffff7caa2a0> gc: collectable <weakref.ReferenceType 0x7ffff7caa610> gc: collectable <set 0x7ffff7bd58c0> gc: collectable <set 0x7ffff7bd59a0> gc: collectable <weakref.ReferenceType 0x7ffff7caa7f0> gc: collectable <weakref.ReferenceType 0x7ffff7caa660> gc: collectable <weakref.ReferenceType 0x7ffff7caa890> gc: collectable <dict 0x7ffff7cba380> gc: collectable <module 0x7ffff7caaac0> gc: collectable <dict 0x7ffff7cba3c0> gc: collectable <module 0x7ffff7caabb0> gc: collectable <dict 0x7ffff7cba580> gc: collectable <tuple 0x7ffff7c761c0> gc: collectable <module 0x7ffff7caac00> gc: collectable <builtin_function_or_method 0x7ffff7caaca0> gc: collectable <dict 0x7ffff7cbacc0> gc: collectable <builtin_function_or_method 0x7ffff7caac50> gc: collectable <builtin_function_or_method 0x7ffff7caacf0> gc: collectable <builtin_function_or_method 0x7ffff7caad40> gc: collectable <builtin_function_or_method 0x7ffff7caad90> gc: collectable <builtin_function_or_method 0x7ffff7caade0> gc: collectable <builtin_function_or_method 0x7ffff7caae30> gc: collectable <builtin_function_or_method 0x7ffff7caae80> gc: collectable <builtin_function_or_method 0x7ffff7caaed0> gc: collectable <builtin_function_or_method 0x7ffff7caaf20> gc: collectable <builtin_function_or_method 0x7ffff7caaf70> gc: collectable <builtin_function_or_method 0x7ffff7caafc0> gc: collectable <builtin_function_or_method 0x7ffff7cab010> gc: collectable <dict 0x7ffff7cba740> gc: collectable <list 0x7ffff7cbadc0> gc: collectable <function 0x7ffff7cb3420> gc: collectable <ABCMeta 0x555555c7b0f0> gc: collectable <tuple 0x7ffff7cbae40> gc: collectable <ABCMeta 0x555555c7cf60> gc: collectable <tuple 0x7ffff7cbaf40> gc: collectable <ABCMeta 0x555555c7d340> gc: collectable <tuple 0x7ffff7cbb600> gc: collectable <ABCMeta 0x555555c7d720> gc: collectable <tuple 0x7ffff7cbb180> gc: collectable <ABCMeta 0x555555c7db00> gc: collectable <tuple 0x7ffff7cbba00> gc: collectable <ABCMeta 0x555555ca4630> gc: collectable <tuple 0x7ffff7c73920> gc: collectable <ModuleSpec 0x7ffff7ca3d40> gc: collectable <getset_descriptor 0x7ffff7cb8940> gc: collectable <builtin_function_or_method 0x7ffff7ca95d0> gc: collectable <ModuleSpec 0x7ffff6f00740> gc: collectable <ModuleSpec 0x7ffff6f00890> gc: collectable <dict 0x7ffff7cb9080> gc: collectable <dict 0x7ffff7cb9340> gc: collectable <getset_descriptor 0x7ffff7cb94c0> gc: collectable <ModuleSpec 0x7ffff6f00ad0> gc: collectable <builtin_function_or_method 0x7ffff7ca9cb0> gc: collectable <dict 0x7ffff7cb9580> gc: collectable <frozenset 0x7ffff7bd4ba0> gc: collectable <tuple 0x7ffff7ca3cd0> gc: collectable <frozenset 0x7ffff7bd4c80> gc: collectable <tuple 0x7ffff7cb8240> gc: collectable <frozenset 0x7ffff7bd4d60> gc: collectable <tuple 0x7ffff7cb8080> gc: collectable <frozenset 0x7ffff7bd5380> gc: collectable <tuple 0x7ffff7cb9940> gc: collectable <frozenset 0x7ffff7bd5460> gc: collectable <builtin_function_or_method 0x7ffff7caa020> gc: collectable <builtin_function_or_method 0x7ffff7caa110> gc: collectable <builtin_function_or_method 0x7ffff7caa340> gc: collectable <builtin_function_or_method 0x7ffff7caa3e0> gc: collectable <builtin_function_or_method 0x7ffff7caa480> gc: collectable <builtin_function_or_method 0x7ffff7caa520> gc: collectable <builtin_function_or_method 0x7ffff7caa250> gc: collectable <builtin_function_or_method 0x7ffff7caa5c0> gc: collectable <builtin_function_or_method 0x7ffff7caa7a0> gc: collectable <builtin_function_or_method 0x7ffff7caa6b0> gc: collectable <builtin_function_or_method 0x7ffff7caa840> gc: collectable <ModuleSpec 0x7ffff6f010a0> gc: collectable <ModuleSpec 0x7ffff6f012b0> gc: collectable <dict 0x7ffff7cbb100> gc: collectable <dict 0x7ffff7cbb400> gc: collectable <dict 0x7ffff7cbb5c0> gc: collectable <tuple 0x7ffff6f01360> gc: collectable <dict 0x7ffff7cbb8c0> gc: collectable <dict 0x7ffff7cbb9c0> gc: collectable <tuple 0x7ffff6f01450> gc: collectable <dict 0x7ffff7cbbbc0> gc: collectable <tuple 0x7ffff6f014b0> gc: collectable <list 0x7ffff7ca4ec0> gc: collectable <types.SimpleNamespace 0x7ffff6f00760> gc: collectable <list 0x7ffff7cb49c0> gc: collectable <types.SimpleNamespace 0x7ffff6f008b0> gc: collectable <list 0x7ffff7cb9240> gc: collectable <function 0x7ffff7cb25c0> gc: collectable <getset_descriptor 0x7ffff7cb9280> gc: collectable <function 0x7ffff7cb2700> gc: collectable <getset_descriptor 0x7ffff7cb93c0> gc: collectable <list 0x7ffff7cb9740> gc: collectable <staticmethod 0x7ffff6f009d0> gc: collectable <function 0x7ffff7cb2980> gc: collectable <function 0x7ffff7cb2a20> gc: collectable <function 0x7ffff7cb2ac0> gc: collectable <function 0x7ffff7cb2b60> gc: collectable <function 0x7ffff7cb2c00> gc: collectable <function 0x7ffff7cb2ca0> gc: collectable <weakref.ReferenceType 0x7ffff7ca9fd0> gc: collectable <weakref.ReferenceType 0x7ffff7caa0c0> gc: collectable <weakref.ReferenceType 0x7ffff7caa2f0> gc: collectable <weakref.ReferenceType 0x7ffff7caa200> gc: collectable <weakref.ReferenceType 0x7ffff7caa750> gc: collectable <weakref.ReferenceType 0x7ffff7caa700> gc: collectable <types.SimpleNamespace 0x7ffff6f010c0> gc: collectable <list 0x7ffff7cba700> gc: collectable <list 0x7ffff7cbae80> gc: collectable <function 0x7ffff7cb22a0> gc: collectable <classmethod 0x7ffff6f011b0> gc: collectable <frozenset 0x7ffff7bd5d20> gc: collectable <_abc._abc_data 0x7ffff7cbb280> gc: collectable <function 0x7ffff7cb32e0> gc: collectable <classmethod 0x7ffff6f01330> gc: collectable <classmethod 0x7ffff6f011e0> gc: collectable <frozenset 0x7ffff7bd5e00> gc: collectable <_abc._abc_data 0x7ffff7cbb500> gc: collectable <function 0x7ffff7cb3100> gc: collectable <function 0x7ffff7cb3060> gc: collectable <function 0x7ffff7cb2fc0> gc: collectable <classmethod 0x7ffff6f01390> gc: collectable <frozenset 0x7ffff7bd5ee0> gc: collectable <_abc._abc_data 0x7ffff7cbb6c0> gc: collectable <function 0x7ffff7cb2de0> gc: collectable <classmethod 0x7ffff6f013f0> gc: collectable <classmethod 0x7ffff6f01420> gc: collectable <frozenset 0x7ffff7bd60a0> gc: collectable <_abc._abc_data 0x7ffff7cbb940> gc: collectable <function 0x7ffff7cb3600> gc: collectable <function 0x7ffff7cb36a0> gc: collectable <classmethod 0x7ffff6f01480> gc: collectable <frozenset 0x7ffff7bd6180> gc: collectable <_abc._abc_data 0x7ffff7cbbac0> gc: collectable <function 0x7ffff7cb3880> gc: collectable <function 0x7ffff7cb3920> gc: collectable <function 0x7ffff7cb39c0> gc: collectable <function 0x7ffff7cb3a60> gc: collectable <classmethod 0x7ffff6f014e0> gc: collectable <frozenset 0x7ffff7bd6260> gc: collectable <_abc._abc_data 0x7ffff7cbbc80> gc: collectable <tuple 0x7ffff6f007f0> gc: collectable <tuple 0x7ffff6f00910> gc: collectable <function 0x7ffff7cb28e0> gc: collectable <function 0x7ffff7cb3380> gc: collectable <dict 0x7ffff7cbb000> gc: collectable <function 0x7ffff7cb3240> gc: collectable <dict 0x7ffff7cbb0c0> gc: collectable <function 0x7ffff7cb2f20> gc: collectable <dict 0x7ffff7cbb3c0> gc: collectable <set 0x7ffff7bd5fc0> gc: collectable <function 0x7ffff7cb34c0> gc: collectable <dict 0x7ffff7cbb680> gc: collectable <function 0x7ffff7cb3740> gc: collectable <dict 0x7ffff7cbb840> gc: collectable <function 0x7ffff7cb3b00> gc: collectable <dict 0x7ffff7cbba80> gc: collectable <set 0x7ffff7bd6340> gc: collectable <cell 0x7ffff6f00790> gc: collectable <cell 0x7ffff6f008e0> gc: collectable <tuple 0x7ffff6f00a30> gc: collectable <weakref.ReferenceType 0x7ffff7cab1f0> gc: collectable <weakref.ReferenceType 0x7ffff7cab3d0> gc: collectable <cell 0x7ffff6f009a0> gc: collectable <builtin_function_or_method 0x7ffff7cab1a0> gc: collectable <builtin_function_or_method 0x7ffff7cab380> gc: collectable <weakref.ReferenceType 0x7ffff7cab150> gc: collectable <weakref.ReferenceType 0x7ffff7cab330> gc: collectable <ABCMeta 0x555555ca5010> gc: collectable <tuple 0x7ffff7cbb1c0> gc: collectable <ABCMeta 0x555555ca53f0> gc: collectable <tuple 0x7ffff6f04040> gc: collectable <ABCMeta 0x555555ca57d0> gc: collectable <tuple 0x7ffff6f04200> gc: collectable <ABCMeta 0x555555ca5bb0> gc: collectable <tuple 0x7ffff7cabd80> gc: collectable <ABCMeta 0x555555ca5f90> gc: collectable <tuple 0x7ffff7cbb200> gc: collectable <ABCMeta 0x555555ca6370> gc: collectable <tuple 0x7ffff7cbb240> gc: collectable <ABCMeta 0x555555ca6750> gc: collectable <tuple 0x7ffff7caa1b0> gc: collectable <ABCMeta 0x555555ca6b30> gc: collectable <tuple 0x7ffff7cbb480> gc: collectable <type 0x555555ca6f10> gc: collectable <tuple 0x7ffff6f04840> gc: collectable <function 0x7ffff7cb2340> gc: collectable <function 0x7ffff6f08f40> gc: collectable <ABCMeta 0x555555ca72f0> gc: collectable <tuple 0x7ffff7cbb4c0> gc: collectable <ABCMeta 0x555555ca76d0> gc: collectable <tuple 0x7ffff7bed7e0> gc: collectable <ABCMeta 0x555555ca7ab0> gc: collectable <tuple 0x7ffff7bee140> gc: collectable <ABCMeta 0x555555ca7e90> gc: collectable <tuple 0x7ffff7beda20> gc: collectable <ABCMeta 0x555555c8f170> gc: collectable <tuple 0x7ffff6f05640> gc: collectable <ABCMeta 0x555555ca8270> gc: collectable <tuple 0x7ffff7c08b30> gc: collectable <ABCMeta 0x555555ca8650> gc: collectable <tuple 0x7ffff7c08ac0> gc: collectable <ABCMeta 0x555555ca8a30> gc: collectable <tuple 0x7ffff7bee0e0> gc: collectable <ABCMeta 0x555555ca8e10> gc: collectable <tuple 0x7ffff7bee1a0> gc: collectable <ABCMeta 0x555555ca91f0> gc: collectable <tuple 0x7ffff7bee200> gc: collectable <type 0x555555ca95d0> gc: collectable <tuple 0x7ffff6f0c040> gc: collectable <_DeprecateByteStringMeta 0x555555ca99b0> gc: collectable <tuple 0x7ffff7c08ba0> gc: collectable <dict 0x7ffff6f05f00> gc: collectable <_abc._abc_data 0x7ffff6f05fc0> gc: collectable <set 0x7ffff6f14580> gc: collectable <set 0x7ffff6f14660> gc: collectable <weakref.ReferenceType 0x7ffff6f0d800> gc: collectable <weakref.ReferenceType 0x7ffff6f0d670> gc: collectable <weakref.ReferenceType 0x7ffff6f0d8a0> gc: collectable <ABCMeta 0x555555ca9d90> gc: collectable <tuple 0x7ffff7c08c10> gc: collectable <list 0x7ffff7cba680> gc: collectable <function 0x7ffff7cb20c0> gc: collectable <function 0x7ffff7cb2160> gc: collectable <module 0x7ffff6f0dc10> gc: collectable <dict 0x7ffff7cbad40> gc: collectable <module 0x7ffff6f0dd00> gc: collectable <dict 0x7ffff6f05f80> gc: collectable <list 0x7ffff6f063c0> gc: collectable <function 0x7ffff7cb3560> gc: collectable <function 0x7ffff6f18220> gc: collectable <function 0x7ffff6f182c0> gc: collectable <function 0x7ffff6f18360> gc: collectable <function 0x7ffff6f18400> gc: collectable <function 0x7ffff6f184a0> gc: collectable <function 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<function 0x7ffff6f19260> gc: collectable <function 0x7ffff6f19300> gc: collectable <function 0x7ffff6f193a0> gc: collectable <function 0x7ffff6f19440> gc: collectable <function 0x7ffff6f194e0> gc: collectable <function 0x7ffff6f19580> gc: collectable <set 0x7ffff6f149e0> gc: collectable <set 0x7ffff6f14ac0> gc: collectable <set 0x7ffff6f14ba0> gc: collectable <set 0x7ffff6f14c80> gc: collectable <function 0x7ffff6f08fe0> gc: collectable <function 0x7ffff6f0bf60> gc: collectable <function 0x7ffff6f19620> gc: collectable <function 0x7ffff6f196c0> gc: collectable <function 0x7ffff6f19760> gc: collectable <function 0x7ffff6f19800> gc: collectable <function 0x7ffff6f198a0> gc: collectable <function 0x7ffff6f19940> gc: collectable <function 0x7ffff6f199e0> gc: collectable <function 0x7ffff6f19a80> gc: collectable <function 0x7ffff6f19b20> gc: collectable <function 0x7ffff6f19bc0> gc: collectable <function 0x7ffff6f19c60> gc: collectable <function 0x7ffff6f19d00> gc: collectable <ABCMeta 0x555555caad70> gc: collectable <tuple 0x7ffff7c08c80> gc: collectable <_Environ 0x7ffff6f01dc0> gc: collectable <function 0x7ffff6f19da0> gc: collectable <_Environ 0x7ffff6f00ec0> gc: collectable <function 0x7ffff6f1a840> gc: collectable <function 0x7ffff6f1a980> gc: collectable <function 0x7ffff6f1aa20> gc: collectable <function 0x7ffff6f1a8e0> gc: collectable <function 0x7ffff6f1aac0> gc: collectable <function 0x7ffff6f1ab60> gc: collectable <function 0x7ffff6f1ac00> gc: collectable <function 0x7ffff6f1aca0> gc: collectable <function 0x7ffff6f1ad40> gc: collectable <function 0x7ffff6f1ade0> gc: collectable <function 0x7ffff6f1ae80> gc: collectable <function 0x7ffff6f1af20> gc: collectable <function 0x7ffff6f1afc0> gc: collectable <type 0x555555cab800> gc: collectable <tuple 0x7ffff6f04f00> gc: collectable <function 0x7ffff6f1b060> gc: collectable <function 0x7ffff6f1b4c0> gc: collectable <ABCMeta 0x555555cabef0> gc: collectable <tuple 0x7ffff6f06740> gc: collectable <module 0x7ffff6f0de40> gc: collectable <dict 0x7ffff7cba200> gc: collectable <type 0x555555cac5e0> gc: collectable <tuple 0x7ffff6f04f40> gc: collectable <type 0x555555caccd0> gc: collectable <tuple 0x7ffff6f04f80> gc: collectable <type 0x555555cad3c0> gc: collectable <tuple 0x7ffff6f04fc0> gc: collectable <function 0x7ffff7c64a40> gc: collectable <function 0x7ffff7c64c20> gc: collectable <function 0x7ffff6f1b560> gc: collectable <function 0x7ffff6f1be20> gc: collectable <function 0x7ffff6f1bec0> gc: collectable <function 0x7ffff6f1bf60> gc: collectable <function 0x7ffff6f20040> gc: collectable <function 0x7ffff6f200e0> gc: collectable <function 0x7ffff6f20180> gc: collectable <function 0x7ffff6f20220> gc: collectable <function 0x7ffff6f202c0> gc: collectable <function 0x7ffff6f20360> gc: collectable <function 0x7ffff6f20400> gc: collectable <function 0x7ffff6f204a0> gc: collectable <function 0x7ffff6f20540> gc: collectable <function 0x7ffff6f205e0> gc: collectable <function 0x7ffff6f20680> gc: collectable <function 0x7ffff6f20720> gc: collectable <function 0x7ffff6f207c0> gc: collectable <function 0x7ffff6f20860> gc: collectable <function 0x7ffff6f20900> gc: collectable <function 0x7ffff6f209a0> gc: collectable <function 0x7ffff6f20a40> gc: collectable <dict 0x7ffff7cbbec0> gc: collectable <dict 0x7ffff7cbbfc0> gc: collectable <tuple 0x7ffff6f015a0> gc: collectable <dict 0x7ffff6f041c0> gc: collectable <tuple 0x7ffff6f01600> gc: collectable <dict 0x7ffff6f04380> gc: collectable <tuple 0x7ffff6f01660> gc: collectable <dict 0x7ffff6f04580> gc: collectable <dict 0x7ffff6f04780> gc: collectable <dict 0x7ffff6f04880> gc: collectable <tuple 0x7ffff7cbaf00> gc: collectable <dict 0x7ffff6f04a80> gc: collectable <dict 0x7ffff6f04740> gc: collectable <dict 0x7ffff6f04d00> gc: collectable <dict 0x7ffff6f04e40> gc: collectable <tuple 0x7ffff6f01a50> gc: collectable <dict 0x7ffff6f05340> gc: collectable <tuple 0x7ffff6f01a20> gc: collectable <dict 0x7ffff6f05440> gc: collectable <tuple 0x7ffff6f01a80> gc: collectable <dict 0x7ffff6f05600> gc: collectable <tuple 0x7ffff6f01ab0> gc: collectable <dict 0x7ffff6f05780> gc: collectable <tuple 0x7ffff7cbb700> gc: collectable <dict 0x7ffff6f05900> gc: collectable <tuple 0x7ffff7cbbcc0> gc: collectable <dict 0x7ffff6f05940> gc: collectable <tuple 0x7ffff7cbbd00> gc: collectable <dict 0x7ffff6f05b00> gc: collectable <tuple 0x7ffff6f01b40> gc: collectable <dict 0x7ffff6f05c00> gc: collectable <tuple 0x7ffff6f04480> gc: collectable <dict 0x7ffff6f05a80> gc: collectable <tuple 0x7ffff6f01c00> gc: collectable <tuple 0x7ffff6f01cf0> gc: collectable <frozenset 0x7ffff6f144a0> gc: collectable <builtin_function_or_method 0x7ffff6f0d7b0> gc: collectable <builtin_function_or_method 0x7ffff6f0d6c0> gc: collectable <builtin_function_or_method 0x7ffff6f0d850> gc: collectable <dict 0x7ffff6f06000> gc: collectable <tuple 0x7ffff6f01d20> gc: collectable <ModuleSpec 0x7ffff6f01df0> gc: collectable <ModuleSpec 0x7ffff6f01ee0> gc: collectable <dict 0x7ffff6f05480> gc: collectable <tuple 0x7ffff6f00ee0> gc: collectable <function 0x7ffff6f1a660> gc: collectable <function 0x7ffff6f1a700> gc: collectable <function 0x7ffff6f1a7a0> gc: collectable <tuple 0x7ffff6f04d80> gc: collectable <tuple 0x7ffff6f04dc0> gc: collectable <dict 0x7ffff7cbad00> gc: collectable <dict 0x7ffff6f06700> gc: collectable <tuple 0x7ffff6f01f90> gc: collectable <ModuleSpec 0x7ffff6f020f0> gc: collectable <dict 0x7ffff6f06840> gc: collectable <dict 0x7ffff6f06980> gc: collectable <dict 0x7ffff6f06a40> gc: collectable <classmethod 0x7ffff6f01570> gc: collectable <frozenset 0x7ffff7bd6420> gc: collectable <_abc._abc_data 0x7ffff7cbbf40> gc: collectable <function 0x7ffff7cb3e20> gc: collectable <function 0x7ffff7cb3ec0> gc: collectable <classmethod 0x7ffff6f015d0> gc: collectable <frozenset 0x7ffff7bd6500> gc: collectable <_abc._abc_data 0x7ffff6f04100> gc: collectable <function 0x7ffff6f080e0> gc: collectable <classmethod 0x7ffff6f01630> gc: collectable <frozenset 0x7ffff7bd66c0> gc: collectable <_abc._abc_data 0x7ffff6f04280> gc: collectable <function 0x7ffff6f08220> gc: collectable <function 0x7ffff6f082c0> gc: collectable <function 0x7ffff6f08360> gc: collectable <function 0x7ffff6f08400> gc: collectable <classmethod 0x7ffff6f01690> gc: collectable <frozenset 0x7ffff7bd67a0> gc: collectable <_abc._abc_data 0x7ffff6f04440> gc: collectable <function 0x7ffff6f085e0> gc: collectable <classmethod 0x7ffff6f016f0> gc: collectable <frozenset 0x7ffff7bd6960> gc: collectable <_abc._abc_data 0x7ffff6f04600> gc: collectable <function 0x7ffff6f08720> gc: collectable <classmethod 0x7ffff6f01750> gc: collectable <classmethod 0x7ffff6f01720> gc: collectable <frozenset 0x7ffff7bd6a40> gc: collectable <_abc._abc_data 0x7ffff6f04800> gc: collectable <classmethod 0x7ffff6f01810> gc: collectable <frozenset 0x7ffff7bd6b20> gc: collectable <_abc._abc_data 0x7ffff6f04980> gc: collectable <function 0x7ffff6f08ae0> gc: collectable <classmethod 0x7ffff6f017e0> gc: collectable <frozenset 0x7ffff7bd6c00> gc: collectable <_abc._abc_data 0x7ffff6f04b00> gc: collectable <staticmethod 0x7ffff6f01930> gc: collectable <function 0x7ffff6f08cc0> gc: collectable <function 0x7ffff6f08d60> gc: collectable <function 0x7ffff6f08e00> gc: collectable <function 0x7ffff6f09080> gc: collectable <classmethod 0x7ffff6f01990> gc: collectable <classmethod 0x7ffff6f019c0> gc: collectable <frozenset 0x7ffff7bd6ce0> gc: collectable <_abc._abc_data 0x7ffff6f04e00> gc: collectable <function 0x7ffff6f091c0> gc: collectable <function 0x7ffff6f09260> gc: collectable <function 0x7ffff6f09300> gc: collectable <function 0x7ffff6f093a0> gc: collectable <function 0x7ffff6f09440> gc: collectable <classmethod 0x7ffff6f019f0> gc: collectable <function 0x7ffff6f09580> gc: collectable <function 0x7ffff6f09620> gc: collectable <function 0x7ffff6f096c0> gc: collectable <function 0x7ffff6f09760> gc: collectable <function 0x7ffff6f09800> gc: collectable <function 0x7ffff6f098a0> gc: collectable <function 0x7ffff6f09940> gc: collectable <frozenset 0x7ffff7bd6dc0> gc: collectable <_abc._abc_data 0x7ffff6f05280> gc: collectable <function 0x7ffff6f09a80> gc: collectable <function 0x7ffff6f09b20> gc: collectable <function 0x7ffff6f09bc0> gc: collectable <function 0x7ffff6f09c60> gc: collectable <function 0x7ffff6f09d00> gc: collectable <function 0x7ffff6f09da0> gc: collectable <function 0x7ffff6f09e40> gc: collectable <function 0x7ffff6f09ee0> gc: collectable <function 0x7ffff6f09f80> gc: collectable <frozenset 0x7ffff7bd7060> gc: collectable <_abc._abc_data 0x7ffff6f05400> gc: collectable <function 0x7ffff6f0a0c0> gc: collectable <function 0x7ffff6f0a160> gc: collectable <function 0x7ffff6f0a200> gc: collectable <function 0x7ffff6f0a2a0> gc: collectable <function 0x7ffff6f0a340> gc: collectable <function 0x7ffff6f0a3e0> gc: collectable <function 0x7ffff6f0a480> gc: collectable <frozenset 0x7ffff7bd7300> gc: collectable <_abc._abc_data 0x7ffff6f05500> gc: collectable <function 0x7ffff6f0a520> gc: collectable <function 0x7ffff6f0a5c0> gc: collectable <function 0x7ffff6f0a660> gc: collectable <classmethod 0x7ffff6f01b10> gc: collectable <member_descriptor 0x7ffff6f05680> gc: collectable <frozenset 0x7ffff7bd75a0> gc: collectable <_abc._abc_data 0x7ffff6f05700> gc: collectable <classmethod 0x7ffff6f01b70> gc: collectable <function 0x7ffff6f0a840> gc: collectable <function 0x7ffff6f0a8e0> gc: collectable <frozenset 0x7ffff7bd7680> gc: collectable <_abc._abc_data 0x7ffff6f05840> gc: collectable <classmethod 0x7ffff6f01bd0> gc: collectable <function 0x7ffff6f0aac0> gc: collectable <function 0x7ffff6f0ab60> gc: collectable <frozenset 0x7ffff7bd7920> gc: collectable <_abc._abc_data 0x7ffff6f05980> gc: collectable <function 0x7ffff6f0ac00> gc: collectable <function 0x7ffff6f0aca0> gc: collectable <frozenset 0x7ffff7bd7bc0> gc: collectable <_abc._abc_data 0x7ffff6f05a40> gc: collectable <function 0x7ffff6f0ad40> gc: collectable <function 0x7ffff6f0ade0> gc: collectable <function 0x7ffff6f0ae80> gc: collectable <function 0x7ffff6f0af20> gc: collectable <function 0x7ffff6f0afc0> gc: collectable <function 0x7ffff6f0b060> gc: collectable <function 0x7ffff6f0b100> gc: collectable <frozenset 0x7ffff7bd7e60> gc: collectable <_abc._abc_data 0x7ffff6f05bc0> gc: collectable <function 0x7ffff6f0b240> gc: collectable <function 0x7ffff6f0b2e0> gc: collectable <function 0x7ffff6f0b380> gc: collectable <function 0x7ffff6f0b420> gc: collectable <function 0x7ffff6f0b4c0> gc: collectable <function 0x7ffff6f0b560> gc: collectable <frozenset 0x7ffff6f14200> gc: collectable <_abc._abc_data 0x7ffff6f05d00> gc: collectable <staticmethod 0x7ffff6f01cc0> gc: collectable <function 0x7ffff6f0b740> gc: collectable <weakref.ReferenceType 0x7ffff6f0d760> gc: collectable <weakref.ReferenceType 0x7ffff6f0d710> gc: collectable <function 0x7ffff6f0b920> gc: collectable <function 0x7ffff6f0b9c0> gc: collectable <function 0x7ffff6f0ba60> gc: collectable <function 0x7ffff6f0bb00> gc: collectable <function 0x7ffff6f0bba0> gc: collectable <function 0x7ffff6f0bc40> gc: collectable <function 0x7ffff6f0bce0> gc: collectable <function 0x7ffff6f0bd80> gc: collectable <function 0x7ffff6f0be20> gc: collectable <function 0x7ffff6f0bec0> gc: collectable <frozenset 0x7ffff6f14740> gc: collectable <_abc._abc_data 0x7ffff6f06100> gc: collectable <types.SimpleNamespace 0x7ffff6f01e10> gc: collectable <list 0x7ffff7cba600> gc: collectable <types.SimpleNamespace 0x7ffff6f01f00> gc: collectable <list 0x7ffff6f06300> gc: collectable <function 0x7ffff6f19e40> gc: collectable <function 0x7ffff6f19ee0> gc: collectable <function 0x7ffff6f19f80> gc: collectable <function 0x7ffff6f1a020> gc: collectable <function 0x7ffff6f1a0c0> gc: collectable <function 0x7ffff6f1a160> gc: collectable <function 0x7ffff6f1a200> gc: collectable <function 0x7ffff6f1a2a0> gc: collectable <function 0x7ffff6f1a340> gc: collectable <function 0x7ffff6f1a3e0> gc: collectable <function 0x7ffff6f1a480> gc: collectable <function 0x7ffff6f1a520> gc: collectable <getset_descriptor 0x7ffff6f06280> gc: collectable <getset_descriptor 0x7ffff6f06240> gc: collectable <frozenset 0x7ffff6f14d60> gc: collectable <_abc._abc_data 0x7ffff6f06480> gc: collectable <tuple 0x7ffff6f01d80> gc: collectable <tuple 0x7ffff6f01ea0> gc: collectable <cell 0x7ffff6f01e70> gc: collectable <cell 0x7ffff6f01e40> gc: collectable <function 0x7ffff6f1b100> gc: collectable <function 0x7ffff6f1b1a0> gc: collectable <function 0x7ffff6f1b240> gc: collectable <function 0x7ffff6f1b2e0> gc: collectable <function 0x7ffff6f1b380> gc: collectable <function 0x7ffff6f1b420> gc: collectable <getset_descriptor 0x7ffff6f06500> gc: collectable <getset_descriptor 0x7ffff6f06540> gc: collectable <function 0x7ffff6f1b600> gc: collectable <classmethod 0x7ffff6f01ff0> gc: collectable <classmethod 0x7ffff6f02020> gc: collectable <getset_descriptor 0x7ffff6f06780> gc: collectable <getset_descriptor 0x7ffff6f067c0> gc: collectable <frozenset 0x7ffff6f14e40> gc: collectable <_abc._abc_data 0x7ffff6f06880> gc: collectable <types.SimpleNamespace 0x7ffff6f02110> gc: collectable <list 0x7ffff7cbaf80> gc: collectable <function 0x7ffff6f1b880> gc: collectable <function 0x7ffff6f1b920> gc: collectable <function 0x7ffff6f1b9c0> gc: collectable <getset_descriptor 0x7ffff6f06940> gc: collectable <getset_descriptor 0x7ffff6f06900> gc: collectable <function 0x7ffff6f1ba60> gc: collectable <function 0x7ffff6f1bb00> gc: collectable <function 0x7ffff6f1bba0> gc: collectable <function 0x7ffff6f1bc40> gc: collectable <getset_descriptor 0x7ffff6f069c0> gc: collectable <getset_descriptor 0x7ffff6f06a00> gc: collectable <function 0x7ffff6f1bce0> gc: collectable <function 0x7ffff6f1bd80> gc: collectable <getset_descriptor 0x7ffff6f06a80> gc: collectable <getset_descriptor 0x7ffff6f06ac0> gc: collectable <function 0x7ffff7cb3f60> gc: collectable <dict 0x7ffff7cbbe40> gc: collectable <set 0x7ffff7bd65e0> gc: collectable <function 0x7ffff6f08180> gc: collectable <dict 0x7ffff7cbbf80> gc: collectable <function 0x7ffff6f084a0> gc: collectable <dict 0x7ffff6f04240> gc: collectable <set 0x7ffff7bd6880> gc: collectable <function 0x7ffff6f08680> gc: collectable <dict 0x7ffff6f04400> gc: collectable <function 0x7ffff6f087c0> gc: collectable <dict 0x7ffff6f04540> gc: collectable <function 0x7ffff6f08900> gc: collectable <dict 0x7ffff6f04300> gc: collectable <function 0x7ffff6f08b80> gc: collectable <dict 0x7ffff6f04940> gc: collectable <function 0x7ffff6f08c20> gc: collectable <tuple 0x7ffff6f018d0> gc: collectable <tuple 0x7ffff6f01900> gc: collectable <function 0x7ffff6f09120> gc: collectable <dict 0x7ffff6f04b40> gc: collectable <dict 0x7ffff6f04bc0> gc: collectable <function 0x7ffff6f094e0> gc: collectable <dict 0x7ffff6f04a40> gc: collectable <set 0x7ffff7bd6f80> gc: collectable <set 0x7ffff7bd6ea0> gc: collectable <set 0x7ffff7bd7220> gc: collectable <set 0x7ffff7bd7140> gc: collectable <set 0x7ffff7bd74c0> gc: collectable <set 0x7ffff7bd73e0> gc: collectable <function 0x7ffff6f0a7a0> gc: collectable <dict 0x7ffff6f053c0> gc: collectable <set 0x7ffff7bd7840> gc: collectable <set 0x7ffff7bd7760> gc: collectable <function 0x7ffff6f0aa20> gc: collectable <dict 0x7ffff6f056c0> gc: collectable <set 0x7ffff7bd7ae0> gc: collectable <set 0x7ffff7bd7a00> gc: collectable <set 0x7ffff7bd7d80> gc: collectable <set 0x7ffff7bd7ca0> gc: collectable <set 0x7ffff6f14120> gc: collectable <set 0x7ffff6f14040> gc: collectable <set 0x7ffff6f143c0> gc: collectable <set 0x7ffff6f142e0> gc: collectable <function 0x7ffff6f0b6a0> gc: collectable <tuple 0x7ffff6f01c90> gc: collectable <set 0x7ffff6f14900> gc: collectable <set 0x7ffff6f14820> gc: collectable <cell 0x7ffff6f00f10> gc: collectable <function 0x7ffff6f1b6a0> gc: collectable <dict 0x7ffff6f06440> gc: collectable <weakref.ReferenceType 0x7ffff7cab6a0> gc: collectable <weakref.ReferenceType 0x7ffff7cab740> gc: collectable <weakref.ReferenceType 0x7ffff7cab9c0> gc: collectable <weakref.ReferenceType 0x7ffff7cab7e0> gc: collectable <weakref.ReferenceType 0x7ffff7cab880> gc: collectable <weakref.ReferenceType 0x7ffff7cab560> gc: collectable <weakref.ReferenceType 0x7ffff7caba60> gc: collectable <weakref.ReferenceType 0x7ffff7cabb00> gc: collectable <weakref.ReferenceType 0x7ffff7cab920> gc: collectable <weakref.ReferenceType 0x7ffff7cabba0> gc: collectable <weakref.ReferenceType 0x7ffff7cabc40> gc: collectable <weakref.ReferenceType 0x7ffff7cab600> gc: collectable <weakref.ReferenceType 0x7ffff7cabce0> gc: collectable <weakref.ReferenceType 0x7ffff7cabec0> gc: collectable <tuple 0x7ffff6f01840> gc: collectable <cell 0x7ffff6f01870> gc: collectable <weakref.ReferenceType 0x7ffff6f0c360> gc: collectable <weakref.ReferenceType 0x7ffff6f0c270> gc: collectable <weakref.ReferenceType 0x7ffff6f0c590> gc: collectable <weakref.ReferenceType 0x7ffff6f0c4a0> gc: collectable <weakref.ReferenceType 0x7ffff6f0c7c0> gc: collectable <weakref.ReferenceType 0x7ffff6f0c6d0> gc: collectable <weakref.ReferenceType 0x7ffff6f0ca40> gc: collectable <weakref.ReferenceType 0x7ffff6f0c950> gc: collectable <weakref.ReferenceType 0x7ffff6f0cc70> gc: collectable <weakref.ReferenceType 0x7ffff6f0cb80> gc: collectable <weakref.ReferenceType 0x7ffff6f0cea0> gc: collectable <weakref.ReferenceType 0x7ffff6f0cdb0> gc: collectable <weakref.ReferenceType 0x7ffff6f0d0d0> gc: collectable <weakref.ReferenceType 0x7ffff6f0cfe0> gc: collectable <weakref.ReferenceType 0x7ffff6f0d440> gc: collectable <weakref.ReferenceType 0x7ffff6f0d3a0> gc: collectable <weakref.ReferenceType 0x7ffff6f0d4e0> gc: collectable <weakref.ReferenceType 0x7ffff6f0d300> gc: collectable <weakref.ReferenceType 0x7ffff6f0d1c0> gc: collectable <tuple 0x7ffff6f01c60> gc: collectable <cell 0x7ffff6f01c30> gc: collectable <weakref.ReferenceType 0x7ffff6f0dad0> gc: collectable <weakref.ReferenceType 0x7ffff6f0db70> gc: collectable <weakref.ReferenceType 0x7ffff6f0d940> gc: collectable <builtin_function_or_method 0x7ffff7cab650> gc: collectable <builtin_function_or_method 0x7ffff7cab6f0> gc: collectable <builtin_function_or_method 0x7ffff7cab970> gc: collectable <builtin_function_or_method 0x7ffff7cab790> gc: collectable <builtin_function_or_method 0x7ffff7cab830> gc: collectable <builtin_function_or_method 0x7ffff7cab510> gc: collectable <builtin_function_or_method 0x7ffff7caba10> gc: collectable <builtin_function_or_method 0x7ffff7cabab0> gc: collectable <builtin_function_or_method 0x7ffff7cab8d0> gc: collectable <builtin_function_or_method 0x7ffff7cabb50> gc: collectable <builtin_function_or_method 0x7ffff7cabbf0> gc: collectable <builtin_function_or_method 0x7ffff7cab5b0> gc: collectable <builtin_function_or_method 0x7ffff7cabc90> gc: collectable <builtin_function_or_method 0x7ffff7cabe70> gc: collectable <builtin_function_or_method 0x7ffff6f0c310> gc: collectable <builtin_function_or_method 0x7ffff6f0c220> gc: collectable <builtin_function_or_method 0x7ffff6f0c540> gc: collectable <builtin_function_or_method 0x7ffff6f0c450> gc: collectable <builtin_function_or_method 0x7ffff6f0c770> gc: collectable <builtin_function_or_method 0x7ffff6f0c680> gc: collectable <builtin_function_or_method 0x7ffff6f0c9f0> gc: collectable <builtin_function_or_method 0x7ffff6f0c900> gc: collectable <builtin_function_or_method 0x7ffff6f0cc20> gc: collectable <builtin_function_or_method 0x7ffff6f0cb30> gc: collectable <builtin_function_or_method 0x7ffff6f0ce50> gc: collectable <builtin_function_or_method 0x7ffff6f0cd60> gc: collectable <builtin_function_or_method 0x7ffff6f0d080> gc: collectable <builtin_function_or_method 0x7ffff6f0cf90> gc: collectable <builtin_function_or_method 0x7ffff6f0d3f0> gc: collectable <builtin_function_or_method 0x7ffff6f0d350> gc: collectable <builtin_function_or_method 0x7ffff6f0d490> gc: collectable <builtin_function_or_method 0x7ffff6f0d2b0> gc: collectable <builtin_function_or_method 0x7ffff6f0d170> gc: collectable <builtin_function_or_method 0x7ffff6f0da80> gc: collectable <builtin_function_or_method 0x7ffff6f0db20> gc: collectable <builtin_function_or_method 0x7ffff6f0d990> gc: collectable <weakref.ReferenceType 0x7ffff7cab4c0> gc: collectable <weakref.ReferenceType 0x7ffff7cabe20> gc: collectable <weakref.ReferenceType 0x7ffff6f0c2c0> gc: collectable <weakref.ReferenceType 0x7ffff6f0c1d0> gc: collectable <weakref.ReferenceType 0x7ffff6f0c4f0> gc: collectable <weakref.ReferenceType 0x7ffff6f0c400> gc: collectable <weakref.ReferenceType 0x7ffff6f0c720> gc: collectable <weakref.ReferenceType 0x7ffff6f0c630> gc: collectable <weakref.ReferenceType 0x7ffff6f0c9a0> gc: collectable <weakref.ReferenceType 0x7ffff6f0c8b0> gc: collectable <weakref.ReferenceType 0x7ffff6f0cbd0> gc: collectable <weakref.ReferenceType 0x7ffff6f0cae0> gc: collectable <weakref.ReferenceType 0x7ffff6f0ce00> gc: collectable <weakref.ReferenceType 0x7ffff6f0cd10> gc: collectable <weakref.ReferenceType 0x7ffff6f0d030> gc: collectable <weakref.ReferenceType 0x7ffff6f0cf40> gc: collectable <weakref.ReferenceType 0x7ffff6f0d260> gc: collectable <weakref.ReferenceType 0x7ffff6f0d210> gc: collectable <weakref.ReferenceType 0x7ffff6f0da30> gc: collectable <weakref.ReferenceType 0x7ffff6f0d9e0> gc: collectable <DistutilsMetaFinder 0x7ffff6f032c0> gc: collectable <module 0x7ffff6f3cc20> gc: collectable <builtin_function_or_method 0x7ffff7ca8f90> gc: collectable <function 0x7ffff6f22340> gc: collectable <function 0x7ffff6f223e0> gc: collectable <function 0x7ffff6f22480> gc: collectable <ModuleSpec 0x7ffff6f00e60> gc: collectable <list 0x7ffff7cba6c0> gc: collectable <function 0x7ffff6f211c0> gc: collectable <ModuleSpec 0x7ffff6f00f80> gc: collectable <ModuleSpec 0x7ffff6f012e0> gc: collectable <function 0x7ffff7cb3c40> gc: collectable <classmethod 0x7ffff6f01540> gc: collectable <type 0x555555cb5910> gc: collectable <dict 0x7ffff6f06e80> gc: collectable <dict 0x7ffff6f2dac0> gc: collectable <dict 0x7ffff6f06d40> gc: collectable <types.SimpleNamespace 0x7ffff6f00e80> gc: collectable <list 0x7ffff7cb9200> gc: collectable <types.SimpleNamespace 0x7ffff6f00fa0> gc: collectable <list 0x7ffff7cb96c0> gc: collectable <types.SimpleNamespace 0x7ffff6f01300> gc: collectable <list 0x7ffff7cbae00> gc: collectable <function 0x7ffff7cb3ce0> gc: collectable <dict 0x7ffff7cbbc40> gc: collectable <dict 0x7ffff6f2ca80> gc: collectable <tuple 0x7ffff6f2cac0> gc: collectable <SourceFileLoader 0x7ffff6f02990> gc: collectable <ModuleSpec 0x7ffff6f028a0> gc: collectable <list 0x7ffff6f07a80> gc: collectable <function 0x7ffff6f20f40> gc: collectable <function 0x7ffff6f21260> gc: collectable <function 0x7ffff6f21300> gc: collectable <function 0x7ffff6f213a0> gc: collectable <function 0x7ffff6f21440> gc: collectable <type 0x555555cb5220> gc: collectable <function 0x7ffff6f214e0> gc: collectable <type 0x555555cbb2d0> gc: collectable <function 0x7ffff6f21b20> gc: collectable <function 0x7ffff6f21d00> gc: collectable <ModuleSpec 0x7ffff6f34e90> gc: collectable <builtin_function_or_method 0x7ffff6f3ccc0> gc: collectable <builtin_function_or_method 0x7ffff6f3cc70> gc: collectable <builtin_function_or_method 0x7ffff6f3cd10> gc: collectable <builtin_function_or_method 0x7ffff6f3cd60> gc: collectable <builtin_function_or_method 0x7ffff6f3cdb0> gc: collectable <builtin_function_or_method 0x7ffff6f3ce00> gc: collectable <builtin_function_or_method 0x7ffff6f3ce50> gc: collectable <builtin_function_or_method 0x7ffff6f3cea0> gc: collectable <builtin_function_or_method 0x7ffff6f3cef0> gc: collectable <builtin_function_or_method 0x7ffff6f3cf40> gc: collectable <builtin_function_or_method 0x7ffff6f3cf90> gc: collectable <builtin_function_or_method 0x7ffff6f3cfe0> gc: collectable <builtin_function_or_method 0x7ffff6f3d030> gc: collectable <builtin_function_or_method 0x7ffff6f3d080> gc: collectable <builtin_function_or_method 0x7ffff6f3d0d0> gc: collectable <builtin_function_or_method 0x7ffff6f3d120> gc: collectable <builtin_function_or_method 0x7ffff6f3d170> gc: collectable <builtin_function_or_method 0x7ffff6f3d1c0> gc: collectable <SourceFileLoader 0x7ffff6f34da0> gc: collectable <ModuleSpec 0x7ffff6f34d70> gc: collectable <function 0x7ffff6f216c0> gc: collectable <function 0x7ffff6f21760> gc: collectable <staticmethod 0x7ffff6f02ec0> gc: collectable <function 0x7ffff6f218a0> gc: collectable <classmethod 0x7ffff6f030a0> gc: collectable <staticmethod 0x7ffff6f03280> gc: collectable <function 0x7ffff6f21a80> gc: collectable <list 0x7ffff6f07e40> gc: collectable <getset_descriptor 0x7ffff6f2cb00> gc: collectable <getset_descriptor 0x7ffff6f2cb40> gc: collectable <list 0x7ffff6f077c0> gc: collectable <dict 0x7ffff6f07dc0> gc: collectable <tuple 0x7ffff6f06f40> gc: collectable <dict 0x7ffff6f2cb80> gc: collectable <tuple 0x7ffff6f2cbc0> gc: collectable <list 0x7ffff6f2dbc0> gc: collectable <list 0x7ffff7cb9c80> gc: collectable <function 0x7ffff6f21800> gc: collectable <dict 0x7ffff6f07d40> gc: collectable <function 0x7ffff6f21940> gc: collectable <dict 0x7ffff6f07c80> gc: collectable <function 0x7ffff6f219e0> gc: collectable <dict 0x7ffff6f2c840> gc: collectable <function 0x7ffff6f21580> gc: collectable <function 0x7ffff6f21620> gc: collectable <getset_descriptor 0x7ffff6f06f00> gc: collectable <getset_descriptor 0x7ffff6f2c800> gc: collectable <function 0x7ffff6f21bc0> gc: collectable <function 0x7ffff6f21c60> gc: collectable <getset_descriptor 0x7ffff6f2cc00> gc: collectable <getset_descriptor 0x7ffff6f2cc40> gc: done, 1005 unreachable, 0 uncollectable, 0.0081s elapsed Program received signal SIGSEGV, Segmentation fault. _PyObject_GC_UNTRACK (filename=filename@entry=0x5555558ae97d "Modules/gcmodule.c", lineno=lineno@entry=2242, op=op@entry=0x7ffff6f317c0) at ./Include/internal/pycore_object.h:246 246 _PyGCHead_SET_NEXT(prev, next); (gdb) bt #0 _PyObject_GC_UNTRACK (filename=filename@entry=0x5555558ae97d "Modules/gcmodule.c", lineno=lineno@entry=2242, op=op@entry=0x7ffff6f317c0) at ./Include/internal/pycore_object.h:246 #1 0x000055555582ed86 in PyObject_GC_UnTrack (op_raw=op_raw@entry=0x7ffff6f317c0) at Modules/gcmodule.c:2242 #2 0x00005555556f29d2 in dict_dealloc (mp=0x7ffff6f317c0) at Objects/dictobject.c:2363 #3 0x00005555556ff0bc in _Py_Dealloc (op=<optimized out>) at Objects/object.c:2625 #4 0x00005555556e922a in Py_DECREF (op=<optimized out>) at ./Include/object.h:706 #5 Py_XDECREF (op=<optimized out>) at ./Include/object.h:799 #6 0x00005555556e92aa in free_keys_object (interp=0x555555bbff08 <_PyRuntime+76392>, keys=0x555555cd34f0) at Objects/dictobject.c:673 #7 0x00005555556e9388 in dictkeys_decref (interp=interp@entry=0x555555bbff08 <_PyRuntime+76392>, dk=dk@entry=0x555555cd34f0) at Objects/dictobject.c:333 #8 0x00005555556f2aee in dict_dealloc (mp=0x7ffff6f2e1c0) at Objects/dictobject.c:2374 #9 0x00005555556ff0bc in _Py_Dealloc (op=<optimized out>) at Objects/object.c:2625 #10 0x00005555557e09d4 in Py_DECREF (op=<optimized out>) at ./Include/object.h:706 #11 _PyImport_ClearModulesByIndex (interp=interp@entry=0x555555bbff08 <_PyRuntime+76392>) at Python/import.c:577 #12 0x00005555557fdbd4 in finalize_modules (tstate=tstate@entry=0x555555c1d828 <_PyRuntime+459656>) at Python/pylifecycle.c:1620 #13 0x00005555557ff692 in Py_FinalizeEx () at Python/pylifecycle.c:1889 #14 0x000055555582c1b6 in Py_RunMain () at Modules/main.c:711 #15 0x000055555582c22b in pymain_main (args=args@entry=0x7fffffffcff0) at Modules/main.c:739 #16 0x000055555582c2fa in Py_BytesMain (argc=<optimized out>, argv=<optimized out>) at Modules/main.c:763 #17 0x00005555556547ae in main (argc=<optimized out>, argv=<optimized out>) at ./Programs/python.c:15 ``` Issue URL : https://github.com/alsa-project/alsa-python/issues/8 Repository URL: https://github.com/alsa-project/alsa-python

1 0

Unable to find amixer simple control
by GitHub issues - opened 09 Mar '24

09 Mar '24

alsa-project/alsa-utils issue #256 was opened from Ovalelephant35: Following are the output of terminal when trying to fetch amixer set:- ```0 amixer: Unable to find simple control 'DC Mode Enable',0 1709919772.617362 WARNING measure-activity: Problem with amixer set "DC Mode Enable" mute amixer: Unable to find simple control 'V_REFOUT Enable',0 1709919772.664983 WARNING measure-activity: Problem with amixer set "V_REFOUT Enable" unmute amixer: Unable to find simple control 'Mic Boost (+20dB)',0 1709919772.766386 WARNING measure-activity: Problem with amixer set "Mic Boost (+20dB)" unmute amixer: Unable to find simple control 'DC Mode Enable',0 1709919772.866294 WARNING measure-activity: amixer: Could not get DC Mode amixer: Unable to find simple control 'V_REFOUT Enable',0 1709919772.916730 WARNING measure-activity: amixer: Could not get mic bias voltage. amixer: Unable to find simple control 'Mic Boost (+20dB)',0 1709919773.016803 WARNING measure-activity: amixer: Could not get mic boost Traceback (most recent call last): File "/usr/bin/sugar-activity3", line 5, in <module> activityinstance.main() File "/usr/lib/python3.7/dist-packages/sugar3/activity/activityinstance.py", line 230, in main instance = create_activity_instance(activity_constructor, activity_handle) O File "/usr/lib/python3.7/dist-packages/sugar3/activity/activityinstance.py", line 59, in create_activity_insta activity = constructor(handle) File "/home/user/Activities/Measure/measure.py", line 187, in self.audiograb ag(self.wave.new buffer, self) init File "/home/user/Activities/Measure/audiograb.py", line 125, in init self.pipeline.add(self.alsasrc) TypeError: Argument 1 does not allow None as a value Exited with status 1, pid 20470 activity id 53d702f16908eb0372466a8f3d74df190d6d00ca ``` Installed alsa-utils when amixer was missing using this command ```sudo apt-get alsa-utils``` Currently on using command ```amixer controls``` this is the output ``` numid=3, iface-MIXER, name='Master Playback Volume' numid=2,iface-MIXER, name='Capture Switch' numid=1, iface-MIXER, name='Capture Volume' ``` Issue URL : https://github.com/alsa-project/alsa-utils/issues/256 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

Alsamixer is exiting upon receiving some keys, such as volume control ones
by GitHub issues - edited 09 Mar '24

09 Mar '24

alsa-project/alsa-utils issue #255 was edited from Porkepix: When in an environment not intercepting those keys (usually managed by your compositor/window manager), upon receiving some keycodes such as volume up or down, alsamixer is exiting and writing the UTF-8 keycode inside the terminal while it shouldn't. Examples: Mute key > 440u Volume down key > 438u Volume up key > 439u Curiously enough, the microphone mute one doesn't do anything. These should at the very least not exit alsamixer. A bonus would be, when they're not intercepted, to actually use them to apply mutes and volumes up/down. EDIT: Terminal here is Alacritty as that behavior might change with others eg. if they're not sending any key codes over to what's running. Issue URL : https://github.com/alsa-project/alsa-utils/issues/255 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[PATCH 0/3] ALSA: Add support for Cirrus Logic CS35L54 and CS35L57
by Richard Fitzgerald 08 Mar '24

08 Mar '24

The CS35L54 and CS35L57 are Boosted Smart Amplifiers. The CS35L54 has I2C/SPI control and I2S/TDM audio. The CS35L57 also has SoundWire control and audio. The hardware differences between L54, L56 and L57 do not affect the driver control interface so they can all be handled by the same driver. The HDA patch has build dependencies on the ASoC patch. The final patch updates serial-multi-instantiate and scan.c to trap the ACPI HID for HDA systems that declare multiple amps all under one Device() node. This patch does not have any build dependency on the first two patches so can be taken separately. Simon Trimmer (3): ASoC: cs35l56: Add support for CS35L54 and CS35L57 ALSA: hda: cs35l56: Add support for CS35L54 and CS35L57 platform/x86: serial-multi-instantiate: Add support for CS35L54 and CS35L57 drivers/acpi/scan.c | 2 ++ .../platform/x86/serial-multi-instantiate.c | 28 +++++++++++++++++++ include/sound/cs35l56.h | 1 + sound/pci/hda/cs35l56_hda.c | 16 +++++++---- sound/pci/hda/cs35l56_hda.h | 2 +- sound/pci/hda/cs35l56_hda_i2c.c | 7 +++-- sound/pci/hda/cs35l56_hda_spi.c | 7 +++-- sound/soc/codecs/cs35l56-sdw.c | 3 +- sound/soc/codecs/cs35l56-shared.c | 8 ++++-- sound/soc/codecs/cs35l56.c | 14 +++++++++- 10 files changed, 73 insertions(+), 15 deletions(-) -- 2.30.2

4 6

[PATCH v1] slimbus: qcom-ngd-ctrl: Reduce auto suspend delay
by Viken Dadhaniya 08 Mar '24

08 Mar '24

Currently we have auto suspend delay of 1s which is very high and it takes long time to driver for runtime suspend after use case is done. Hence to optimize runtime PM ops, reduce auto suspend delay to 100ms. Signed-off-by: Viken Dadhaniya <quic_vdadhani(a)quicinc.com> --- drivers/slimbus/qcom-ngd-ctrl.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/drivers/slimbus/qcom-ngd-ctrl.c b/drivers/slimbus/qcom-ngd-ctrl.c index efeba8275a66..5de45a0e3da5 100644 --- a/drivers/slimbus/qcom-ngd-ctrl.c +++ b/drivers/slimbus/qcom-ngd-ctrl.c @@ -81,7 +81,7 @@ #define SLIM_USR_MC_DISCONNECT_PORT 0x2E #define SLIM_USR_MC_REPEAT_CHANGE_VALUE 0x0 -#define QCOM_SLIM_NGD_AUTOSUSPEND MSEC_PER_SEC +#define QCOM_SLIM_NGD_AUTOSUSPEND (MSEC_PER_SEC / 10) #define SLIM_RX_MSGQ_TIMEOUT_VAL 0x10000 #define SLIM_LA_MGR 0xFF -- QUALCOMM INDIA, on behalf of Qualcomm Innovation Center, Inc. is a member of Code Aurora Forum, hosted by The Linux Foundation

2 2

[PATCH v1 0/2] Add PDM controller for StarFive JH8100 SoC
by Xingyu Wu 08 Mar '24

08 Mar '24

The Pulse Density Modulation (PDM) controller is a digital PDM microphone interface controller and decoder that supports both mono/stereo PDM format, and a Inter-IC Sound (I2S) transmitter that outputs standard stereo audio data to another device. The PDM controller includes two PDM modules, each PDM module can drive one bitstream sampling clock and two bitstream coming data with sampling clock rising and falling edge. On the JH8100 SoC, PDM and I2S are fixedly connected as follow: PDM module 0 --> I2S channel 0 PDM module 1 --> I2S channel 1 The first patch adds documentation to describe PDM controller bindings. And the second patch adds PDM driver for the StarFive JH8100 SoC. The DTS patch of PDM controller will be submitted after the patchs of JH8100 DTS are accepted. Xingyu Wu (2): dt-bindings: ASoC: Add PDM controller for the StarFive JH8100 SoC ASoC: starfive: Add PDM controller support .../bindings/sound/starfive,jh8100-pdm.yaml | 84 ++++ MAINTAINERS | 7 + sound/soc/starfive/Kconfig | 7 + sound/soc/starfive/Makefile | 1 + sound/soc/starfive/jh8100_pdm.c | 395 ++++++++++++++++++ 5 files changed, 494 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/starfive,jh8100-pdm.yaml create mode 100644 sound/soc/starfive/jh8100_pdm.c -- 2.25.1

3 10

[PATCH v2 00/22] ASoC: Cleanup MediaTek soundcard machine drivers
by AngeloGioacchino Del Regno 08 Mar '24

08 Mar '24

Changes in v2: - Bindings: Changed link-name/codec/clk-provider to remove `items` and leave just the enum - Moved .*-dai-link pattern additionalProperties after `type: object` - Added ref to sound-card-common.yaml - Fixed dai-link-xxx -> xxx-dai-link typo in example comment This series performs a cleanup of most of the MediaTek AFE drivers and soundcard machine drivers, reducing code duplication and setting a base to be consistent with their devicetree bindings, as many of those are using different properties and nodes for no good reason. Summarizing: - Commonizes functions and ops where possible - Adds a common probe mechanism, increasing maintainability of soundcard drivers for older MediaTek SoCs - Migrates all drivers to support the new bindings - Obviously retains compatibility with old device trees - Reduces machine-specific parameters hardcoding in drivers - Can now set machine-specific params in device tree - Uses the `audio-routing` and `dai-link` nodes like some other non-MediaTek SoC sound drivers - Imposes consistency between MediaTek ASoC machine soundcard drivers bindings - Reduces code size and greatly reduces the amount of code that will be required for newer drivers (retaining compatibility with the old bindings was costly in terms of code size, otherwise this series would've removed ~1000 more lines, or something along that line). This series was (manually) tested on MT8173, MT8192, MT8195 and MT8186 Chromebooks. AngeloGioacchino Del Regno (22): ASoC: mediatek: mt8192-afe-pcm: Convert to devm_pm_runtime_enable() ASoC: mediatek: mt8192-afe-pcm: Simplify with dev_err_probe() ASoC: mediatek: Commonize ADDA rate transform functions and enums ASoC: mediatek: Assign dummy when codec not specified for a DAI link ASoC: mediatek: Add common machine soundcard driver probe mechanism ASoC: mediatek: common: Constify struct mtk_sof_priv ASoC: mediatek: mt8188: Migrate to mtk_soundcard_common_probe ASoC: mediatek: mt8195: Migrate to mtk_soundcard_common_probe ASoC: mediatek: mt8192: Migrate to mtk_soundcard_common_probe ASoC: mediatek: mt8186: Migrate to mtk_soundcard_common_probe ASoC: mediatek: Add common snd_soc_ops .startup() callback ASoC: mediatek: mt8195: Migrate to the common mtk_soundcard_startup ASoC: mediatek: mt8192: Migrate to the common mtk_soundcard_startup ASoC: mediatek: mt8186-rt1019: Migrate to the common mtk_soundcard_startup ASoC: mediatek: Add common mtk_afe_component_probe callback ASoC: mediatek: Use common mtk_afe_pcm_platform with common probe cb ASoC: mediatek: mt8186: Unify mt8186-mt6366 machine drivers ASoC: dt-bindings: mt8195: Document audio-routing and dai-link subnode ASoC: dt-bindings: mt8192: Document audio-routing and dai-link subnode ASoC: dt-bindings: mt8186: Document audio-routing and dai-link subnode arm64: dts: mediatek: mt8195-cherry: Specify sound DAI links and routing arm64: dts: mediatek: mt8186-corsola: Specify sound DAI links and routing .../sound/mt8186-mt6366-da7219-max98357.yaml | 112 +- .../sound/mt8186-mt6366-rt1019-rt5682s.yaml | 102 +- .../sound/mt8192-mt6359-rt1015-rt5682.yaml | 124 +- .../bindings/sound/mt8195-mt6359.yaml | 115 ++ .../boot/dts/mediatek/mt8186-corsola.dtsi | 42 +- .../boot/dts/mediatek/mt8195-cherry.dtsi | 45 + sound/soc/mediatek/Kconfig | 24 +- sound/soc/mediatek/common/Makefile | 2 + .../mediatek/common/mtk-afe-platform-driver.c | 18 + .../soc/mediatek/common/mtk-dai-adda-common.c | 70 + .../soc/mediatek/common/mtk-dai-adda-common.h | 45 + .../soc/mediatek/common/mtk-dsp-sof-common.c | 15 +- .../soc/mediatek/common/mtk-dsp-sof-common.h | 1 - sound/soc/mediatek/common/mtk-soc-card.h | 7 +- .../mediatek/common/mtk-soundcard-driver.c | 205 ++- .../mediatek/common/mtk-soundcard-driver.h | 42 + sound/soc/mediatek/mt6797/mt6797-afe-pcm.c | 14 +- sound/soc/mediatek/mt6797/mt6797-dai-adda.c | 85 +- sound/soc/mediatek/mt7986/mt7986-afe-pcm.c | 14 +- sound/soc/mediatek/mt8183/mt8183-afe-pcm.c | 14 +- sound/soc/mediatek/mt8183/mt8183-dai-adda.c | 90 +- sound/soc/mediatek/mt8186/Makefile | 3 +- sound/soc/mediatek/mt8186/mt8186-afe-pcm.c | 19 +- sound/soc/mediatek/mt8186/mt8186-dai-adda.c | 94 +- .../mt8186/mt8186-mt6366-da7219-max98357.c | 1189 ----------------- ...t6366-rt1019-rt5682s.c => mt8186-mt6366.c} | 578 ++++---- sound/soc/mediatek/mt8188/mt8188-afe-pcm.c | 21 +- sound/soc/mediatek/mt8188/mt8188-dai-adda.c | 89 +- sound/soc/mediatek/mt8188/mt8188-mt6359.c | 203 +-- sound/soc/mediatek/mt8192/mt8192-afe-pcm.c | 114 +- sound/soc/mediatek/mt8192/mt8192-dai-adda.c | 90 +- .../mt8192/mt8192-mt6359-rt1015-rt5682.c | 301 ++--- sound/soc/mediatek/mt8195/mt8195-afe-pcm.c | 21 +- sound/soc/mediatek/mt8195/mt8195-dai-adda.c | 90 +- sound/soc/mediatek/mt8195/mt8195-mt6359.c | 487 +++---- 35 files changed, 1714 insertions(+), 2771 deletions(-) create mode 100644 sound/soc/mediatek/common/mtk-dai-adda-common.c create mode 100644 sound/soc/mediatek/common/mtk-dai-adda-common.h delete mode 100644 sound/soc/mediatek/mt8186/mt8186-mt6366-da7219-max98357.c rename sound/soc/mediatek/mt8186/{mt8186-mt6366-rt1019-rt5682s.c => mt8186-mt6366.c} (72%) -- 2.44.0

2 26

[GIT PULL] ASoC fixes for v6.8-rc7
by Mark Brown 08 Mar '24

08 Mar '24

The following changes since commit ed00a6945dc32462c2d3744a3518d2316da66fcc: ASoC: amd: yc: Fix non-functional mic on Lenovo 21J2 (2024-02-28 12:24:59 +0000) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.8-rc7 for you to fetch changes up to 96e202f8c52ac49452f83317cf3b34cd1ad81e18: ASoC: wm8962: Fix up incorrect error message in wm8962_set_fll (2024-03-06 17:44:22 +0000) ---------------------------------------------------------------- ASoC: Fixes for v6.8 Some more driver specific fixes for v6.8, plus one new x86 platform quirk. All good fixes to have if you have systems that use the relevant hardware. ---------------------------------------------------------------- Al Raj Hassain (1): ASoC: amd: yc: Add HP Pavilion Aero Laptop 13-be2xxx(8BD6) into DMI quirk table Alban Boyé (1): ASoC: Intel: bytcr_rt5640: Add an extra entry for the Chuwi Vi8 tablet Andreas Pape (1): ASoC: rcar: adg: correct TIMSEL setting for SSI9 Rob Herring (1): ASoC: dt-bindings: nvidia: Fix 'lge' vendor prefix Stuart Henderson (4): ASoC: madera: Fix typo in madera_set_fll_clks shift value ASoC: wm8962: Enable oscillator if selecting WM8962_FLL_OSC ASoC: wm8962: Enable both SPKOUTR_ENA and SPKOUTL_ENA in mono mode ASoC: wm8962: Fix up incorrect error message in wm8962_set_fll .../sound/nvidia,tegra-audio-max9808x.yaml | 2 +- sound/soc/amd/yc/acp6x-mach.c | 7 ++++++ sound/soc/codecs/madera.c | 2 +- sound/soc/codecs/wm8962.c | 29 ++++++++++++++-------- sound/soc/intel/boards/bytcr_rt5640.c | 12 +++++++++ sound/soc/sh/rcar/adg.c | 7 ++++++ 6 files changed, 47 insertions(+), 12 deletions(-)

2 1

[PATCH v1 1/1] ASoC: Intel: catpt: Carefully use PCI bitwise constants
by Andy Shevchenko 07 Mar '24

07 Mar '24

PM constants for PCI devices are defined with bitwise annotation. When used as is, sparse complains about that: .../catpt/dsp.c:390:9: warning: restricted pci_power_t degrades to integer .../catpt/dsp.c:414:9: warning: restricted pci_power_t degrades to integer Force them to be u32 in the driver. Signed-off-by: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> --- sound/soc/intel/catpt/dsp.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/catpt/dsp.c b/sound/soc/intel/catpt/dsp.c index 346bec000306..5454c6d9ab5b 100644 --- a/sound/soc/intel/catpt/dsp.c +++ b/sound/soc/intel/catpt/dsp.c @@ -387,7 +387,7 @@ int catpt_dsp_power_down(struct catpt_dev *cdev) mask = cdev->spec->d3srampgd_bit | cdev->spec->d3pgd_bit; catpt_updatel_pci(cdev, VDRTCTL0, mask, cdev->spec->d3pgd_bit); - catpt_updatel_pci(cdev, PMCS, PCI_PM_CTRL_STATE_MASK, PCI_D3hot); + catpt_updatel_pci(cdev, PMCS, PCI_PM_CTRL_STATE_MASK, (__force u32)PCI_D3hot); /* give hw time to drop off */ udelay(50); @@ -411,7 +411,7 @@ int catpt_dsp_power_up(struct catpt_dev *cdev) val = mask & (~CATPT_VDRTCTL2_DTCGE); catpt_updatel_pci(cdev, VDRTCTL2, mask, val); - catpt_updatel_pci(cdev, PMCS, PCI_PM_CTRL_STATE_MASK, PCI_D0); + catpt_updatel_pci(cdev, PMCS, PCI_PM_CTRL_STATE_MASK, (__force u32)PCI_D0); /* SRAM power gating none */ mask = cdev->spec->d3srampgd_bit | cdev->spec->d3pgd_bit; -- 2.43.0.rc1.1.gbec44491f096

3 2

[PATCH] ASoC: wm_adsp: Fix missing mutex_lock in wm_adsp_write_ctl()
by Richard Fitzgerald 07 Mar '24

07 Mar '24

wm_adsp_write_ctl() must hold the pwr_lock mutex when calling cs_dsp_get_ctl(). This was previously partially fixed by commit 781118bc2fc1 ("ASoC: wm_adsp: Fix missing locking in wm_adsp_[read|write]_ctl()") but this only put locking around the call to cs_dsp_coeff_write_ctrl(), missing the call to cs_dsp_get_ctl(). Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> Fixes: 781118bc2fc1 ("ASoC: wm_adsp: Fix missing locking in wm_adsp_[read|write]_ctl()") --- sound/soc/codecs/wm_adsp.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index e451c009f2d9..7d5c096e06cd 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -683,11 +683,12 @@ static void wm_adsp_control_remove(struct cs_dsp_coeff_ctl *cs_ctl) int wm_adsp_write_ctl(struct wm_adsp *dsp, const char *name, int type, unsigned int alg, void *buf, size_t len) { - struct cs_dsp_coeff_ctl *cs_ctl = cs_dsp_get_ctl(&dsp->cs_dsp, name, type, alg); + struct cs_dsp_coeff_ctl *cs_ctl; struct wm_coeff_ctl *ctl; int ret; mutex_lock(&dsp->cs_dsp.pwr_lock); + cs_ctl = cs_dsp_get_ctl(&dsp->cs_dsp, name, type, alg); ret = cs_dsp_coeff_write_ctrl(cs_ctl, 0, buf, len); mutex_unlock(&dsp->cs_dsp.pwr_lock); -- 2.30.2

2 1

[PATCH v2 0/2] ASoC: codecs: ES8326: change members of private structure
by Zhang Yi 07 Mar '24

07 Mar '24

We found that using 0x45 as the default value for interrupt-clk would cause a headset detection error.So we took 0x00 as the default value for interrupt-clk and passed the test. We removed mic1-src and mic2-src, which were not used. Zhang Yi (2): ASoC: codecs: ES8326: Changing members of private structure ASoC: codecs: ES8326: change support for ES8326 .../bindings/sound/everest,es8326.yaml | 8 ++++---- sound/soc/codecs/es8326.c | 18 +----------------- 2 files changed, 5 insertions(+), 21 deletions(-) -- 2.17.1

2 3

[PATCH 1/2] ALSA: hda: hda_component: Add missing #include guards
by Richard Fitzgerald 07 Mar '24

07 Mar '24

Add the conventional include guards around the content of the hda_component.h header file. This prevents double-declaration of struct hda_component if the header gets included multiple times. This isn't causing any problems with current code, so no need to backport to older kernels. Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> --- sound/pci/hda/hda_component.h | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/pci/hda/hda_component.h b/sound/pci/hda/hda_component.h index deae9dea01b4..a0fcc723483d 100644 --- a/sound/pci/hda/hda_component.h +++ b/sound/pci/hda/hda_component.h @@ -6,6 +6,9 @@ * Cirrus Logic International Semiconductor Ltd. */ +#ifndef __HDA_COMPONENT_H__ +#define __HDA_COMPONENT_H__ + #include <linux/acpi.h> #include <linux/component.h> @@ -82,3 +85,5 @@ static inline void hda_component_manager_unbind(struct hda_codec *cdc, { component_unbind_all(hda_codec_dev(cdc), comps); } + +#endif /* ifndef __HDA_COMPONENT_H__ */ -- 2.30.2

2 2

[PATCH] firmware: cirrus: cs_dsp: Remove non-existent member from kerneldoc
by Richard Fitzgerald 07 Mar '24

07 Mar '24

The kerneldoc for struct cs_dsp refers to a fw_file_name member but there's no such member. Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> --- include/linux/firmware/cirrus/cs_dsp.h | 1 - 1 file changed, 1 deletion(-) diff --git a/include/linux/firmware/cirrus/cs_dsp.h b/include/linux/firmware/cirrus/cs_dsp.h index 29cd11d5a3cf..23384a54d575 100644 --- a/include/linux/firmware/cirrus/cs_dsp.h +++ b/include/linux/firmware/cirrus/cs_dsp.h @@ -123,7 +123,6 @@ struct cs_dsp_client_ops; * @sysclk_mask: Mask of frequency bits within sysclk register (ADSP1 only) * @sysclk_shift: Shift of frequency bits within sysclk register (ADSP1 only) * @alg_regions: List of currently loaded algorithm regions - * @fw_file_name: Filename of the current firmware * @fw_name: Name of the current firmware * @fw_id: ID of the current firmware, obtained from the wmfw * @fw_id_version: Version of the firmware, obtained from the wmfw -- 2.30.2

2 1

[PATCH] firmware: cirrus: cs_dsp: Initialize debugfs_root to invalid
by Richard Fitzgerald 07 Mar '24

07 Mar '24

Initialize debugfs_root to -ENODEV so that if the client never sets a valid debugfs root the debugfs files will not be created. A NULL pointer passed to any of the debugfs_create_*() functions means "create in the root of debugfs". It doesn't mean "ignore". Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> --- drivers/firmware/cirrus/cs_dsp.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) diff --git a/drivers/firmware/cirrus/cs_dsp.c b/drivers/firmware/cirrus/cs_dsp.c index 79d4254d1f9b..9f3d665cfdcf 100644 --- a/drivers/firmware/cirrus/cs_dsp.c +++ b/drivers/firmware/cirrus/cs_dsp.c @@ -522,7 +522,7 @@ void cs_dsp_cleanup_debugfs(struct cs_dsp *dsp) { cs_dsp_debugfs_clear(dsp); debugfs_remove_recursive(dsp->debugfs_root); - dsp->debugfs_root = NULL; + dsp->debugfs_root = ERR_PTR(-ENODEV); } EXPORT_SYMBOL_NS_GPL(cs_dsp_cleanup_debugfs, FW_CS_DSP); #else @@ -2246,6 +2246,11 @@ static int cs_dsp_common_init(struct cs_dsp *dsp) mutex_init(&dsp->pwr_lock); +#ifdef CONFIG_DEBUG_FS + /* Ensure this is invalid if client never provides a debugfs root */ + dsp->debugfs_root = ERR_PTR(-ENODEV); +#endif + return 0; } -- 2.30.2

2 1

[PATCH 00/22] ASoC: Cleanup MediaTek soundcard machine drivers
by AngeloGioacchino Del Regno 07 Mar '24

07 Mar '24

This series performs a cleanup of most of the MediaTek AFE drivers and soundcard machine drivers, reducing code duplication and setting a base to be consistent with their devicetree bindings, as many of those are using different properties and nodes for no good reason. Summarizing: - Commonizes functions and ops where possible - Adds a common probe mechanism, increasing maintainability of soundcard drivers for older MediaTek SoCs - Migrates all drivers to support the new bindings - Obviously retains compatibility with old device trees - Reduces machine-specific parameters hardcoding in drivers - Can now set machine-specific params in device tree - Uses the `audio-routing` and `dai-link` nodes like some other non-MediaTek SoC sound drivers - Imposes consistency between MediaTek ASoC machine soundcard drivers bindings - Reduces code size and greatly reduces the amount of code that will be required for newer drivers (retaining compatibility with the old bindings was costly in terms of code size, otherwise this series would've removed ~1000 more lines, or something along that line). This series was (manually) tested on MT8173, MT8192, MT8195 and MT8186 Chromebooks. AngeloGioacchino Del Regno (22): ASoC: mediatek: mt8192-afe-pcm: Convert to devm_pm_runtime_enable() ASoC: mediatek: mt8192-afe-pcm: Simplify with dev_err_probe() ASoC: mediatek: Commonize ADDA rate transform functions and enums ASoC: mediatek: Assign dummy when codec not specified for a DAI link ASoC: mediatek: Add common machine soundcard driver probe mechanism ASoC: mediatek: common: Constify struct mtk_sof_priv ASoC: mediatek: mt8188: Migrate to mtk_soundcard_common_probe ASoC: mediatek: mt8195: Migrate to mtk_soundcard_common_probe ASoC: mediatek: mt8192: Migrate to mtk_soundcard_common_probe ASoC: mediatek: mt8186: Migrate to mtk_soundcard_common_probe ASoC: mediatek: Add common snd_soc_ops .startup() callback ASoC: mediatek: mt8195: Migrate to the common mtk_soundcard_startup ASoC: mediatek: mt8192: Migrate to the common mtk_soundcard_startup ASoC: mediatek: mt8186-rt1019: Migrate to the common mtk_soundcard_startup ASoC: mediatek: Add common mtk_afe_component_probe callback ASoC: mediatek: Use common mtk_afe_pcm_platform with common probe cb ASoC: mediatek: mt8186: Unify mt8186-mt6366 machine drivers ASoC: dt-bindings: mt8195: Document audio-routing and dai-link subnode ASoC: dt-bindings: mt8192: Document audio-routing and dai-link subnode ASoC: dt-bindings: mt8186: Document audio-routing and dai-link subnode arm64: dts: mediatek: mt8195-cherry: Specify sound DAI links and routing arm64: dts: mediatek: mt8186-corsola: Specify sound DAI links and routing .../sound/mt8186-mt6366-da7219-max98357.yaml | 112 +- .../sound/mt8186-mt6366-rt1019-rt5682s.yaml | 112 +- .../sound/mt8192-mt6359-rt1015-rt5682.yaml | 129 +- .../bindings/sound/mt8195-mt6359.yaml | 123 ++ .../boot/dts/mediatek/mt8186-corsola.dtsi | 42 +- .../boot/dts/mediatek/mt8195-cherry.dtsi | 45 + sound/soc/mediatek/Kconfig | 24 +- sound/soc/mediatek/common/Makefile | 2 + .../mediatek/common/mtk-afe-platform-driver.c | 18 + .../soc/mediatek/common/mtk-dai-adda-common.c | 70 + .../soc/mediatek/common/mtk-dai-adda-common.h | 45 + .../soc/mediatek/common/mtk-dsp-sof-common.c | 15 +- .../soc/mediatek/common/mtk-dsp-sof-common.h | 1 - sound/soc/mediatek/common/mtk-soc-card.h | 7 +- .../mediatek/common/mtk-soundcard-driver.c | 205 ++- .../mediatek/common/mtk-soundcard-driver.h | 42 + sound/soc/mediatek/mt6797/mt6797-afe-pcm.c | 14 +- sound/soc/mediatek/mt6797/mt6797-dai-adda.c | 85 +- sound/soc/mediatek/mt7986/mt7986-afe-pcm.c | 14 +- sound/soc/mediatek/mt8183/mt8183-afe-pcm.c | 14 +- sound/soc/mediatek/mt8183/mt8183-dai-adda.c | 90 +- sound/soc/mediatek/mt8186/Makefile | 3 +- sound/soc/mediatek/mt8186/mt8186-afe-pcm.c | 19 +- sound/soc/mediatek/mt8186/mt8186-dai-adda.c | 94 +- .../mt8186/mt8186-mt6366-da7219-max98357.c | 1189 ----------------- ...t6366-rt1019-rt5682s.c => mt8186-mt6366.c} | 578 ++++---- sound/soc/mediatek/mt8188/mt8188-afe-pcm.c | 21 +- sound/soc/mediatek/mt8188/mt8188-dai-adda.c | 89 +- sound/soc/mediatek/mt8188/mt8188-mt6359.c | 203 +-- sound/soc/mediatek/mt8192/mt8192-afe-pcm.c | 114 +- sound/soc/mediatek/mt8192/mt8192-dai-adda.c | 90 +- .../mt8192/mt8192-mt6359-rt1015-rt5682.c | 301 ++--- sound/soc/mediatek/mt8195/mt8195-afe-pcm.c | 21 +- sound/soc/mediatek/mt8195/mt8195-dai-adda.c | 90 +- sound/soc/mediatek/mt8195/mt8195-mt6359.c | 487 +++---- 35 files changed, 1739 insertions(+), 2769 deletions(-) create mode 100644 sound/soc/mediatek/common/mtk-dai-adda-common.c create mode 100644 sound/soc/mediatek/common/mtk-dai-adda-common.h delete mode 100644 sound/soc/mediatek/mt8186/mt8186-mt6366-da7219-max98357.c rename sound/soc/mediatek/mt8186/{mt8186-mt6366-rt1019-rt5682s.c => mt8186-mt6366.c} (72%) -- 2.44.0

4 31

Strange trigger_htstamp after snd_pcm_drop
by GitHub issues - opened 07 Mar '24

07 Mar '24

alsa-project/alsa-lib issue #387 was opened from z-s-e: I have conceptually the following situation (actual code can be found [here](https://gitlab.com/z-s-e/alpe/-/blob/main/src/alpe_alsa.cpp?ref_type=…): ``` // sound is currently playing snd_pcm_status(pcm, pcm_status); snd_pcm_status_get_htstamp(pcm_status, &ts1); snd_pcm_drop(pcm); snd_pcm_status(pcm, pcm_status); snd_pcm_status_get_trigger_htstamp(status, &ts2); ``` On my hardware I get a `ts2` timestamp that is earlier than `ts1`. From my understanding that should be impossible, as `ts2` should contain the timestamp of when the drop happened, which should obviously be after `ts1`. Is that a bug or do I misunderstand the meaning of those timestamps? Issue URL : https://github.com/alsa-project/alsa-lib/issues/387 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

RE: [EXTERNAL] Re: [PATCH v10] The tas2783 is a smart audio amplifier with integrated MIPI SoundWire interface (Version 1.2.1 compliant), I2C, and I2S/TDM interfaces designed for portable applications. An on-chip DSP supports Texas Instruments SmartAmp sp
by Ding, Shenghao 07 Mar '24

07 Mar '24

> -----Original Message----- > From: Mark Brown <broonie(a)kernel.org> > Sent: Wednesday, March 6, 2024 3:19 AM > To: Ding, Shenghao <shenghao-ding(a)ti.com> > Cc: andriy.shevchenko(a)linux.intel.com; lgirdwood(a)gmail.com; > perex(a)perex.cz; pierre-louis.bossart(a)linux.intel.com; > 13916275206(a)139.com; alsa-devel(a)alsa-project.org; linux- > kernel(a)vger.kernel.org; liam.r.girdwood(a)intel.com; bard.liao(a)intel.com; > mengdong.lin(a)intel.com; yung-chuan.liao(a)linux.intel.com; Xu, Baojun > <baojun.xu(a)ti.com>; Lu, Kevin <kevin-lu(a)ti.com>; tiwai(a)suse.de; > soyer(a)irl.hu; Baojun.Xu(a)fpt.com; Navada Kanyana, Mukund > <navada(a)ti.com> > Subject: [EXTERNAL] Re: [PATCH v10] The tas2783 is a smart audio amplifier > with integrated MIPI SoundWire interface (Version 1.2.1 compliant), I2C, and > I2S/TDM interfaces designed for portable applications. An on-chip DSP > supports Texas Instruments SmartAmp spe... > > On Tue, Mar 05, 2024 at 04:43:35PM +0800, Shenghao Ding wrote: > > The ASoC component provides the majority of the functionality of the > > device, all the audio functions. > > > +static const struct reg_default tas2783_reg_defaults[] = { > > + /* Default values for ROM mode. Activated. */ > > + { 0x8002, 0x1a }, /* AMP inactive. */ > > + { 0x8097, 0xc8 }, > > + { 0x80b5, 0x74 }, > > + { 0x8099, 0x20 }, > > + { 0xfe8d, 0x0d }, > > + { 0xfebe, 0x4a }, > > + { 0x8230, 0x00 }, > > + { 0x8231, 0x00 }, > > + { 0x8232, 0x00 }, > > + { 0x8233, 0x01 }, > > + { 0x8418, 0x00 }, /* Set volume to 0 dB. */ > > + { 0x8419, 0x00 }, > > + { 0x841a, 0x00 }, > > + { 0x841b, 0x00 }, > > + { 0x8428, 0x40 }, /* Unmute channel */ > > + { 0x8429, 0x00 }, > > + { 0x842a, 0x00 }, > > + { 0x842b, 0x00 }, > > + { 0x8548, 0x00 }, /* Set volume to 0 dB. */ > > + { 0x8549, 0x00 }, > > + { 0x854a, 0x00 }, > > + { 0x854b, 0x00 }, > > + { 0x8558, 0x40 }, /* Unmute channel */ > > + { 0x8559, 0x00 }, > > + { 0x855a, 0x00 }, > > + { 0x855b, 0x00 }, > > + { 0x800a, 0x3a }, /* Enable both channel */ > > These comments sound like this register default table is not actually the > physical default register values that the chip has... > > > +static const struct regmap_config tasdevice_regmap = { > > + .reg_bits = 32, > > + .val_bits = 8, > > + .readable_reg = tas2783_readable_register, > > + .volatile_reg = tas2783_volatile_register, > > + .max_register = 0x44ffffff, > > + .reg_defaults = tas2783_reg_defaults, > > + .num_reg_defaults = ARRAY_SIZE(tas2783_reg_defaults), > > ...but this is set as the register defaults. This will cause problems with things > like cache sync where we don't write values out if they're not the defaults. > We also try to keep default settings from the silicon except in the most > obvious cases, it avoids issues with trying to work out what to set and > accomodate different use cases for different systems. > > If you do need to set non-default values then either just regular writes during > probe or a regmap patch would do it. So, can I remove ".reg_defaults = tas2783_reg_defaults," and tas2783_reg_defaults from the code? > > > + .cache_type = REGCACHE_RBTREE, > > + .use_single_read = true, > > + .use_single_write = true, > > REGCACHE_MAPLE is generally the most sensible choice for modern devices > - it's a newer and fancier data structure underlying it and there's only a few > cases with low end devices, mostly doing block I/O which this doesn't > support, where the RBTREE cache is still better. Accept > > > + u16 dev_info; > > + int ret; > > + > > + if (!tas_dev->sdw_peripheral) { > > + dev_err(tas_dev->dev, "%s: peripheral doesn't exist.\n", > > + __func__); > > + return; > > + } > > + > > + dev_info = tas_dev->sdw_peripheral->bus->link_id | > > + tas_dev->sdw_peripheral->id.unique_id << 16; > > I'm kind of surprised dev_info works as a variable name without something > getting upset that it aliases the function of the same name. Accept

2 1

[PATCH 0/4] ASoC: fsl: Support register and unregister rpmsg sound card through remoteproc
by Chancel Liu 07 Mar '24

07 Mar '24

echo /lib/firmware/fw.elf > /sys/class/remoteproc/remoteproc0/firmware (A) echo start > /sys/class/remoteproc/remoteproc0/state (B) echo stop > /sys/class/remoteproc/remoteproc0/state The rpmsg sound card is registered in (A) and unregistered in (B). After "start", imx-audio-rpmsg registers devices for ASoC platform driver and machine driver. Then sound card is registered. After "stop", imx-audio-rpmsg unregisters devices for ASoC platform driver and machine driver. Then sound card is unregistered. Chancel Liu (4): ASoC: fsl: imx_pcm_rpmsg: Register component with rpmsg channel name ASoC: fsl: imx-audio-rpmsg: Register device with rpmsg channel name ASoC: fsl: Let imx-audio-rpmsg register platform device for card ASoC: fsl: imx-rpmsg: Update to correct DT node sound/soc/fsl/fsl_rpmsg.c | 11 ----------- sound/soc/fsl/imx-audio-rpmsg.c | 22 +++++++++++++++++++--- sound/soc/fsl/imx-pcm-rpmsg.c | 11 ++++++++--- sound/soc/fsl/imx-rpmsg.c | 21 ++++++++++++++++++--- 4 files changed, 45 insertions(+), 20 deletions(-) -- 2.43.0

2 6

[PATCH v1 2/2] ASoC: codecs: ES8326: change support for ES8326
by Zhang Yi 07 Mar '24

07 Mar '24

> > On Wed, Mar 06, 2024 at 09:34:14AM +0800, Zhang Yi wrote: > > > Removed mic1-src and mic2-src. and changed default value > > > of interrupt-clk > > > > We could do with a better changelog here. > > > > > - everest,mic1-src: > > > - $ref: /schemas/types.yaml#/definitions/uint8 > > > - description: > > > - the value of reg 2A when headset plugged. > > > - minimum: 0x00 > > > - maximum: 0x77 > > > - default: 0x22 > > > - > > > - everest,mic2-src: > > > - $ref: /schemas/types.yaml#/definitions/uint8 > > > - description: > > > - the value of reg 2A when headset unplugged. > > > - minimum: 0x00 > > > - maximum: 0x77 > > > - default: 0x44 > > > > This will make any DTs using the properties instantly buggy. I believe > > there's a way of marking properties as deprecated, you should use that > > instead. Or we could just leave the properties there and ignore them at > > runtime. > 'deprecated: true' is the way. OK.I will modify the properties, thank you for your support

1 0

[PATCH 1/5] ASoC: wm8962: Enable oscillator if selecting WM8962_FLL_OSC
by Stuart Henderson 07 Mar '24

07 Mar '24

Signed-off-by: Stuart Henderson <stuarth(a)opensource.cirrus.com> --- sound/soc/codecs/wm8962.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index fb90ae6a8a34..6d7bb696b135 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -2914,8 +2914,12 @@ static int wm8962_set_fll(struct snd_soc_component *component, int fll_id, int s switch (fll_id) { case WM8962_FLL_MCLK: case WM8962_FLL_BCLK: + fll1 |= (fll_id - 1) << WM8962_FLL_REFCLK_SRC_SHIFT; + break; case WM8962_FLL_OSC: fll1 |= (fll_id - 1) << WM8962_FLL_REFCLK_SRC_SHIFT; + snd_soc_component_update_bits(component, WM8962_PLL2, + WM8962_OSC_ENA, WM8962_OSC_ENA); break; case WM8962_FLL_INT: snd_soc_component_update_bits(component, WM8962_FLL_CONTROL_1, -- 2.39.2

2 8

[PATCH v1 0/2] ASoC: codecs: ES8326: change members of private structure
by Zhang Yi 06 Mar '24

06 Mar '24

We found that using 0x45 as the default value for interrupt-clk would cause a headset detection error.So we took 0x00 as the default value for interrupt-clk and passed the test. We removed mic1-src and mic2-src, which were not used. *** BLURB HERE *** Zhang Yi (2): ASoC: codecs: ES8326: Changing members of private structure ASoC: codecs: ES8326: change support for ES8326 .../bindings/sound/everest,es8326.yaml | 22 ++----------------- sound/soc/codecs/es8326.c | 18 +-------------- 2 files changed, 3 insertions(+), 37 deletions(-) -- 2.17.1

3 4

[PATCH 0/5] ASoC: Harden DAPM route checks and Intel fixes
by Cezary Rojewski 06 Mar '24

06 Mar '24

Set of loosely connected patches. Most impactful change is dropping any permisiveness when snd_soc_dapm_add_routes() fails in soc-topology.c To do it safely, disable route checks for all skylake-driver boards. Relevant background: Since commit daa480bde6b3 ("ASoC: soc-core: tidyup for snd_soc_dapm_add_routes()") route checks are no longer permissive. Probe failures for Intel boards have been partially addressed by commit a22ae72b86a4 ("ASoC: soc-core: isable route checks for legacy devices") and its follow up but only skl_nau88l25_ssm4567.c is patched. The rest of the boards still need fixing. After that, removal of copy-pastas found in ssm4567.c and redundant code in i2s_test.c for avs-boards. Amadeusz Sławiński (1): ASoC: Intel: avs: i2s_test: Remove redundant dapm routes Cezary Rojewski (4): ASoC: Intel: Disable route checks for Skylake boards ASoC: topology: Do not ignore route checks when parsing graphs ASoC: Intel: avs: ssm4567: Do not ignore route checks ASoC: Intel: avs: ssm4567: Board cleanup sound/soc/intel/avs/boards/i2s_test.c | 79 ------------------- sound/soc/intel/avs/boards/ssm4567.c | 5 +- sound/soc/intel/boards/bxt_da7219_max98357a.c | 1 + sound/soc/intel/boards/bxt_rt298.c | 1 + sound/soc/intel/boards/glk_rt5682_max98357a.c | 1 + sound/soc/intel/boards/kbl_da7219_max98357a.c | 1 + sound/soc/intel/boards/kbl_da7219_max98927.c | 4 + sound/soc/intel/boards/kbl_rt5660.c | 1 + sound/soc/intel/boards/kbl_rt5663_max98927.c | 2 + .../intel/boards/kbl_rt5663_rt5514_max98927.c | 1 + sound/soc/intel/boards/skl_hda_dsp_generic.c | 1 + .../soc/intel/boards/skl_nau88l25_max98357a.c | 1 + sound/soc/intel/boards/skl_rt286.c | 1 + sound/soc/soc-topology.c | 5 +- 14 files changed, 19 insertions(+), 85 deletions(-) -- 2.25.1

2 13

[PATCH] ASoC: soc-card: Fix missing locking in snd_soc_card_get_kcontrol()
by Richard Fitzgerald 06 Mar '24

06 Mar '24

snd_soc_card_get_kcontrol() must be holding a read lock on card->controls_rwsem while walking the controls list. Compare with snd_ctl_find_numid(). The existing function is renamed snd_soc_card_get_kcontrol_locked() so that it can be called from contexts that are already holding card->controls_rwsem (for example, control get/put functions). There are few direct or indirect callers of snd_soc_card_get_kcontrol(), and most are safe. Three require changes, which have been included in this patch: codecs/cs35l45.c: cs35l45_activate_ctl() is called from a control put() function so is changed to call snd_soc_card_get_kcontrol_locked(). codecs/cs35l56.c: cs35l56_sync_asp1_mixer_widgets_with_firmware() is called from control get()/put() functions so is changed to call snd_soc_card_get_kcontrol_locked(). fsl/fsl_xcvr.c: fsl_xcvr_activate_ctl() is called from three places, one of which already holds card->controls_rwsem: 1. fsl_xcvr_mode_put(), a control put function, which will already be holding card->controls_rwsem. 2. fsl_xcvr_startup(), a DAI startup function. 3. fsl_xcvr_shutdown(), a DAI shutdown function. To fix this, fsl_xcvr_activate_ctl() has been changed to call snd_soc_card_get_kcontrol_locked() so that it is safe to call directly from fsl_xcvr_mode_put(). The fsl_xcvr_startup() and fsl_xcvr_shutdown() functions have been changed to take a read lock on card->controls_rsem() around calls to fsl_xcvr_activate_ctl(). While this is not very elegant, it keeps the change small, to avoid this patch creating a large collateral churn in fsl/fsl_xcvr.c. Analysis of other callers of snd_soc_card_get_kcontrol() is that they do not need any changes, they are not holding card->controls_rwsem when they call snd_soc_card_get_kcontrol(). Direct callers of snd_soc_card_get_kcontrol(): fsl/fsl_spdif.c: fsl_spdif_dai_probe() - DAI probe function fsl/fsl_micfil.c: voice_detected_fn() - IRQ handler Indirect callers via soc_component_notify_control(): codecs/cs42l43: cs42l43_mic_shutter() - IRQ handler codecs/cs42l43: cs42l43_spk_shutter() - IRQ handler codecs/ak4118.c: ak4118_irq_handler() - IRQ handler codecs/wm_adsp.c: wm_adsp_write_ctl() - not currently used Indirect callers via snd_soc_limit_volume(): qcom/sc8280xp.c: sc8280xp_snd_init() - DAIlink init function ti/rx51.c: rx51_aic34_init() - DAI init function I don't have hardware to test the fsl/*, qcom/sc828xp.c, ti/rx51.c and ak4118.c changes. Backport note: The fsl/, qcom/, cs35l45, cs35l56 and cs42l43 callers were added since the Fixes commit so won't all be present on older kernels. Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> Fixes: 209c6cdfd283 ("ASoC: soc-card: move snd_soc_card_get_kcontrol() to soc-card") --- It would be great if people could test the fsl/, qcom/, ti/rx51 and ak4418 drivers. --- include/sound/soc-card.h | 2 ++ sound/soc/codecs/cs35l45.c | 2 +- sound/soc/codecs/cs35l56.c | 2 +- sound/soc/fsl/fsl_xcvr.c | 12 +++++++++++- sound/soc/soc-card.c | 24 ++++++++++++++++++++++-- 5 files changed, 37 insertions(+), 5 deletions(-) diff --git a/include/sound/soc-card.h b/include/sound/soc-card.h index ecc02e955279..1f4c39922d82 100644 --- a/include/sound/soc-card.h +++ b/include/sound/soc-card.h @@ -30,6 +30,8 @@ static inline void snd_soc_card_mutex_unlock(struct snd_soc_card *card) struct snd_kcontrol *snd_soc_card_get_kcontrol(struct snd_soc_card *soc_card, const char *name); +struct snd_kcontrol *snd_soc_card_get_kcontrol_locked(struct snd_soc_card *soc_card, + const char *name); int snd_soc_card_jack_new(struct snd_soc_card *card, const char *id, int type, struct snd_soc_jack *jack); int snd_soc_card_jack_new_pins(struct snd_soc_card *card, const char *id, diff --git a/sound/soc/codecs/cs35l45.c b/sound/soc/codecs/cs35l45.c index 44c221745c3b..2392c6effed8 100644 --- a/sound/soc/codecs/cs35l45.c +++ b/sound/soc/codecs/cs35l45.c @@ -184,7 +184,7 @@ static int cs35l45_activate_ctl(struct snd_soc_component *component, else snprintf(name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, "%s", ctl_name); - kcontrol = snd_soc_card_get_kcontrol(component->card, name); + kcontrol = snd_soc_card_get_kcontrol_locked(component->card, name); if (!kcontrol) { dev_err(component->dev, "Can't find kcontrol %s\n", name); return -EINVAL; diff --git a/sound/soc/codecs/cs35l56.c b/sound/soc/codecs/cs35l56.c index 2c1313e34cce..6dd0319bc843 100644 --- a/sound/soc/codecs/cs35l56.c +++ b/sound/soc/codecs/cs35l56.c @@ -114,7 +114,7 @@ static int cs35l56_sync_asp1_mixer_widgets_with_firmware(struct cs35l56_private name = full_name; } - kcontrol = snd_soc_card_get_kcontrol(dapm->card, name); + kcontrol = snd_soc_card_get_kcontrol_locked(dapm->card, name); if (!kcontrol) { dev_warn(cs35l56->base.dev, "Could not find control %s\n", name); continue; diff --git a/sound/soc/fsl/fsl_xcvr.c b/sound/soc/fsl/fsl_xcvr.c index f0fb33d719c2..c46f64557a7f 100644 --- a/sound/soc/fsl/fsl_xcvr.c +++ b/sound/soc/fsl/fsl_xcvr.c @@ -174,7 +174,9 @@ static int fsl_xcvr_activate_ctl(struct snd_soc_dai *dai, const char *name, struct snd_kcontrol *kctl; bool enabled; - kctl = snd_soc_card_get_kcontrol(card, name); + lockdep_assert_held(&card->snd_card->controls_rwsem); + + kctl = snd_soc_card_get_kcontrol_locked(card, name); if (kctl == NULL) return -ENOENT; @@ -576,10 +578,14 @@ static int fsl_xcvr_startup(struct snd_pcm_substream *substream, xcvr->streams |= BIT(substream->stream); if (!xcvr->soc_data->spdif_only) { + struct snd_soc_card *card = dai->component->card; + /* Disable XCVR controls if there is stream started */ + down_read(&card->snd_card->controls_rwsem); fsl_xcvr_activate_ctl(dai, fsl_xcvr_mode_kctl.name, false); fsl_xcvr_activate_ctl(dai, fsl_xcvr_arc_mode_kctl.name, false); fsl_xcvr_activate_ctl(dai, fsl_xcvr_earc_capds_kctl.name, false); + up_read(&card->snd_card->controls_rwsem); } return 0; @@ -598,11 +604,15 @@ static void fsl_xcvr_shutdown(struct snd_pcm_substream *substream, /* Enable XCVR controls if there is no stream started */ if (!xcvr->streams) { if (!xcvr->soc_data->spdif_only) { + struct snd_soc_card *card = dai->component->card; + + down_read(&card->snd_card->controls_rwsem); fsl_xcvr_activate_ctl(dai, fsl_xcvr_mode_kctl.name, true); fsl_xcvr_activate_ctl(dai, fsl_xcvr_arc_mode_kctl.name, (xcvr->mode == FSL_XCVR_MODE_ARC)); fsl_xcvr_activate_ctl(dai, fsl_xcvr_earc_capds_kctl.name, (xcvr->mode == FSL_XCVR_MODE_EARC)); + up_read(&card->snd_card->controls_rwsem); } ret = regmap_update_bits(xcvr->regmap, FSL_XCVR_EXT_IER0, FSL_XCVR_IRQ_EARC_ALL, 0); diff --git a/sound/soc/soc-card.c b/sound/soc/soc-card.c index 285ab4c9c716..8a2f163da6bc 100644 --- a/sound/soc/soc-card.c +++ b/sound/soc/soc-card.c @@ -5,6 +5,9 @@ // Copyright (C) 2019 Renesas Electronics Corp. // Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> // + +#include <linux/lockdep.h> +#include <linux/rwsem.h> #include <sound/soc.h> #include <sound/jack.h> @@ -26,12 +29,15 @@ static inline int _soc_card_ret(struct snd_soc_card *card, return ret; } -struct snd_kcontrol *snd_soc_card_get_kcontrol(struct snd_soc_card *soc_card, - const char *name) +struct snd_kcontrol *snd_soc_card_get_kcontrol_locked(struct snd_soc_card *soc_card, + const char *name) { struct snd_card *card = soc_card->snd_card; struct snd_kcontrol *kctl; + /* must be held read or write */ + lockdep_assert_held(&card->controls_rwsem); + if (unlikely(!name)) return NULL; @@ -40,6 +46,20 @@ struct snd_kcontrol *snd_soc_card_get_kcontrol(struct snd_soc_card *soc_card, return kctl; return NULL; } +EXPORT_SYMBOL_GPL(snd_soc_card_get_kcontrol_locked); + +struct snd_kcontrol *snd_soc_card_get_kcontrol(struct snd_soc_card *soc_card, + const char *name) +{ + struct snd_card *card = soc_card->snd_card; + struct snd_kcontrol *kctl; + + down_read(&card->controls_rwsem); + kctl = snd_soc_card_get_kcontrol_locked(soc_card, name); + up_read(&card->controls_rwsem); + + return kctl; +} EXPORT_SYMBOL_GPL(snd_soc_card_get_kcontrol); static int jack_new(struct snd_soc_card *card, const char *id, int type, -- 2.30.2

3 3

[PATCH v1 1/1] ASoC: Intel: atom: sst_ipc: Remove unused intel-mid.h
by Andy Shevchenko 06 Mar '24

06 Mar '24

intel-mid.h is providing some core parts of the South Complex PM, which are usually are not used by individual drivers. In particular, this driver doesn't use it, so simply remove the unused header. Signed-off-by: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> --- sound/soc/intel/atom/sst/sst_ipc.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/intel/atom/sst/sst_ipc.c b/sound/soc/intel/atom/sst/sst_ipc.c index 3fc2c9a6c44d..0630e58b9d6b 100644 --- a/sound/soc/intel/atom/sst/sst_ipc.c +++ b/sound/soc/intel/atom/sst/sst_ipc.c @@ -19,8 +19,9 @@ #include <sound/pcm.h> #include <sound/soc.h> #include <sound/compress_driver.h> -#include <asm/intel-mid.h> + #include <asm/platform_sst_audio.h> + #include "../sst-mfld-platform.h" #include "sst.h" -- 2.43.0.rc1.1.gbec44491f096

2 1

[PATCH] ASoC: dt-bindings: rt1015: Convert to dtschema
by Javier García 06 Mar '24

06 Mar '24

Convert the RT1015 Mono Class D Audio Amplifier to DT schema. Signed-off-by: Javier García <javier.gar.tab(a)gmail.com> --- .../bindings/sound/realtek,rt1015.yaml | 41 +++++++++++++++++++ .../devicetree/bindings/sound/rt1015.txt | 23 ----------- 2 files changed, 41 insertions(+), 23 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/realtek,rt1015.yaml delete mode 100644 Documentation/devicetree/bindings/sound/rt1015.txt diff --git a/Documentation/devicetree/bindings/sound/realtek,rt1015.yaml b/Documentation/devicetree/bindings/sound/realtek,rt1015.yaml new file mode 100644 index 000000000000..880196081a60 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/realtek,rt1015.yaml @@ -0,0 +1,41 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/realtek,rt1015.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: RT1015 Mono Class D Audio Amplifier + +maintainers: + - Jack Yu <jack.yu(a)realtek.com> + +properties: + compatible: + enum: + - realtek,rt1015 + + reg: + maxItems: 1 + + realtek,power-up-delay-ms: + description: Set a delay time for flush work to be completed, + this vlaue is adjustable depending on platform. + maxItems: 1 + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + codec@28 { + compatible = "realtek,rt1015"; + reg = <0x28>; + realtek,power-up-delay-ms = <50>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/rt1015.txt b/Documentation/devicetree/bindings/sound/rt1015.txt deleted file mode 100644 index e498966d436f..000000000000 --- a/Documentation/devicetree/bindings/sound/rt1015.txt +++ /dev/null @@ -1,23 +0,0 @@ -RT1015 Mono Class D Audio Amplifier - -This device supports I2C only. - -Required properties: - -- compatible : "realtek,rt1015". - -- reg : The I2C address of the device. - -Optional properties: - -- realtek,power-up-delay-ms - Set a delay time for flush work to be completed, - this value is adjustable depending on platform. - -Example: - -rt1015: codec@28 { - compatible = "realtek,rt1015"; - reg = <0x28>; - realtek,power-up-delay-ms = <50>; -}; -- 2.41.0

3 2

Alsamixer is exiting upon receiving some keys, such as volume control ones
by GitHub issues - opened 06 Mar '24

06 Mar '24

alsa-project/alsa-utils issue #255 was opened from Porkepix: When in an environment not intercepting those keys (usually managed by your compositor/window manager), upon receiving some keycodes such as volume up or down, alsamixer is exiting and writing the UTF-8 keycode inside the terminal while it shouldn't. Examples: Mute key > 440u Volume down key > 438u Volume up key > 439u Curiously enough, the microphone mute one doesn't do anything. These should at the very least not exit alsamixer. A bonus would be, when they're not intercepted, to actually use them to apply mutes and volumes up/down. Issue URL : https://github.com/alsa-project/alsa-utils/issues/255 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[PATCH v1] slimbus: stream: Add null pointer check to prevent crash
by Viken Dadhaniya 06 Mar '24

06 Mar '24

We are facing crash due to null pointer dereference of stream in slim_stream_disable(). there is a possible scenario where client driver is calling slimbus stream APIs in incorrect sequence and it might lead to null pointer access of the stream in slimbus enable/disable/prepare/unprepare/free functions. Fix this issue by adding null pointer check of the stream before accessing in all stream API’s exposed to client. Signed-off-by: Viken Dadhaniya <quic_vdadhani(a)quicinc.com> --- drivers/slimbus/stream.c | 37 +++++++++++++++++++++++++++++++++---- 1 file changed, 33 insertions(+), 4 deletions(-) diff --git a/drivers/slimbus/stream.c b/drivers/slimbus/stream.c index 1d6b38657917..c5a436fd0952 100644 --- a/drivers/slimbus/stream.c +++ b/drivers/slimbus/stream.c @@ -202,10 +202,16 @@ static int slim_get_prate_code(int rate) int slim_stream_prepare(struct slim_stream_runtime *rt, struct slim_stream_config *cfg) { - struct slim_controller *ctrl = rt->dev->ctrl; + struct slim_controller *ctrl; struct slim_port *port; int num_ports, i, port_id, prrate; + if (!rt || !cfg) { + pr_err("%s: Stream or cfg is NULL, Check from client side\n", __func__); + return -EINVAL; + } + + ctrl = rt->dev->ctrl; if (rt->ports) { dev_err(&rt->dev->dev, "Stream already Prepared\n"); return -EINVAL; @@ -358,9 +364,15 @@ int slim_stream_enable(struct slim_stream_runtime *stream) { DEFINE_SLIM_BCAST_TXN(txn, SLIM_MSG_MC_BEGIN_RECONFIGURATION, 3, SLIM_LA_MANAGER, NULL); - struct slim_controller *ctrl = stream->dev->ctrl; + struct slim_controller *ctrl; int ret, i; + if (!stream) { + pr_err("%s: Stream is NULL, Check from client side\n", __func__); + return -EINVAL; + } + + ctrl = stream->dev->ctrl; if (ctrl->enable_stream) { ret = ctrl->enable_stream(stream); if (ret) @@ -411,12 +423,18 @@ int slim_stream_disable(struct slim_stream_runtime *stream) { DEFINE_SLIM_BCAST_TXN(txn, SLIM_MSG_MC_BEGIN_RECONFIGURATION, 3, SLIM_LA_MANAGER, NULL); - struct slim_controller *ctrl = stream->dev->ctrl; + struct slim_controller *ctrl; int ret, i; + if (!stream) { + pr_err("%s: Stream is NULL, Check from client side\n", __func__); + return -EINVAL; + } + if (!stream->ports || !stream->num_ports) return -EINVAL; + ctrl = stream->dev->ctrl; if (ctrl->disable_stream) ctrl->disable_stream(stream); @@ -448,6 +466,11 @@ int slim_stream_unprepare(struct slim_stream_runtime *stream) { int i; + if (!stream) { + pr_err("%s: Stream is NULL, Check from client side\n", __func__); + return -EINVAL; + } + if (!stream->ports || !stream->num_ports) return -EINVAL; @@ -476,8 +499,14 @@ EXPORT_SYMBOL_GPL(slim_stream_unprepare); */ int slim_stream_free(struct slim_stream_runtime *stream) { - struct slim_device *sdev = stream->dev; + struct slim_device *sdev; + + if (!stream) { + pr_err("%s: Stream is NULL, Check from client side\n", __func__); + return -EINVAL; + } + sdev = stream->dev; spin_lock(&sdev->stream_list_lock); list_del(&stream->node); spin_unlock(&sdev->stream_list_lock); -- QUALCOMM INDIA, on behalf of Qualcomm Innovation Center, Inc. is a member of Code Aurora Forum, hosted by The Linux Foundation

1 0

[PATCH v1] slimbus: qcom-ngd-ctrl: Add timeout for wait operation
by Viken Dadhaniya 06 Mar '24

06 Mar '24

In current driver qcom_slim_ngd_up_worker() indefinitely waiting for ctrl->qmi_up completion object. This is resulting in workqueue lockup on Kthread. Added wait_for_completion_interruptible_timeout to allow the thread to wait for specific timeout period and bail out instead waiting infinitely. Signed-off-by: Viken Dadhaniya <quic_vdadhani(a)quicinc.com> --- drivers/slimbus/qcom-ngd-ctrl.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) diff --git a/drivers/slimbus/qcom-ngd-ctrl.c b/drivers/slimbus/qcom-ngd-ctrl.c index efeba8275a66..c27076d8b7db 100644 --- a/drivers/slimbus/qcom-ngd-ctrl.c +++ b/drivers/slimbus/qcom-ngd-ctrl.c @@ -83,6 +83,7 @@ #define QCOM_SLIM_NGD_AUTOSUSPEND MSEC_PER_SEC #define SLIM_RX_MSGQ_TIMEOUT_VAL 0x10000 +#define SLIM_QMI_TIMEOUT_MS 1000 #define SLIM_LA_MGR 0xFF #define SLIM_ROOT_FREQ 24576000 @@ -1451,7 +1452,11 @@ static void qcom_slim_ngd_up_worker(struct work_struct *work) ctrl = container_of(work, struct qcom_slim_ngd_ctrl, ngd_up_work); /* Make sure qmi service is up before continuing */ - wait_for_completion_interruptible(&ctrl->qmi_up); + if (!wait_for_completion_interruptible_timeout(&ctrl->qmi_up, + msecs_to_jiffies(SLIM_QMI_TIMEOUT_MS))) { + dev_err(ctrl->dev, "QMI wait timeout\n"); + return; + } mutex_lock(&ctrl->ssr_lock); qcom_slim_ngd_enable(ctrl, true); -- QUALCOMM INDIA, on behalf of Qualcomm Innovation Center, Inc. is a member of Code Aurora Forum, hosted by The Linux Foundation

2 2

[PATCH] ASoC: da7219: read fmw property to get mclk for non-dts systems
by cujomalainey＠chromium.org 06 Mar '24

06 Mar '24

From: Akshu Agrawal <akshu.agrawal(a)amd.com> Non-dts based systems can use ACPI DSDT to pass on the mclk to da7219. This enables da7219 mclk to be linked to system clock. Enable/Disable of the mclk is already handled in the codec so platform drivers don't have to explicitly do handling of mclk. Signed-off-by: Akshu Agrawal <akshu.agrawal(a)amd.com> Signed-off-by: Curtis Malainey <cujomalainey(a)chromium.org> Cc: Karthik Ramasubramanian <kramasub(a)google.com> --- include/sound/da7219.h | 2 ++ sound/soc/codecs/da7219.c | 8 +++++++- 2 files changed, 9 insertions(+), 1 deletion(-) diff --git a/include/sound/da7219.h b/include/sound/da7219.h index dde4542b084f3..4cf4c52be3f7f 100644 --- a/include/sound/da7219.h +++ b/include/sound/da7219.h @@ -40,6 +40,8 @@ struct da7219_pdata { const char *dai_clk_names[DA7219_DAI_NUM_CLKS]; + const char *mclk_name; + /* Mic */ enum da7219_micbias_voltage micbias_lvl; enum da7219_mic_amp_in_sel mic_amp_in_sel; diff --git a/sound/soc/codecs/da7219.c b/sound/soc/codecs/da7219.c index 311ea7918b312..800f61d993043 100644 --- a/sound/soc/codecs/da7219.c +++ b/sound/soc/codecs/da7219.c @@ -1782,6 +1782,8 @@ static struct da7219_pdata *da7219_fw_to_pdata(struct device *dev) pdata->dai_clk_names[DA7219_DAI_WCLK_IDX], pdata->dai_clk_names[DA7219_DAI_BCLK_IDX]); + device_property_read_string(dev, "dlg,mclk-name", &pdata->mclk_name); + if (device_property_read_u32(dev, "dlg,micbias-lvl", &of_val32) >= 0) pdata->micbias_lvl = da7219_fw_micbias_lvl(dev, of_val32); else @@ -2521,7 +2523,11 @@ static int da7219_probe(struct snd_soc_component *component) da7219_handle_pdata(component); /* Check if MCLK provided */ - da7219->mclk = clk_get(component->dev, "mclk"); + if (da7219->pdata->mclk_name) + da7219->mclk = clk_get(NULL, da7219->pdata->mclk_name); + if (!da7219->mclk) + da7219->mclk = clk_get(component->dev, "mclk"); + if (IS_ERR(da7219->mclk)) { if (PTR_ERR(da7219->mclk) != -ENOENT) { ret = PTR_ERR(da7219->mclk); -- 2.44.0.rc1.240.g4c46232300-goog

3 4

Dmic nhlt update
by GitHub pull_request - opened 05 Mar '24

05 Mar '24

alsa-project/alsa-utils pull request #254 was opened from jsarha: Driver version 4 update from for DMIC NHLT generator, and some error print improvements. Request URL : https://github.com/alsa-project/alsa-utils/pull/254 Patch URL : https://github.com/alsa-project/alsa-utils/pull/254.patch Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[PATCH v3] ASoC: soc-core.c: Prefer to return dai->driver->name in snd_soc_dai_name_get()
by Chancel Liu 05 Mar '24

05 Mar '24

ASoC machine driver can use snd_soc_{of_}get_dlc() (A) to get DAI name for dlc (snd_soc_dai_link_component). In this function call dlc->dai_name is parsed via snd_soc_dai_name_get() (B). (A) int snd_soc_get_dlc(...) { ... (B) dlc->dai_name = snd_soc_dai_name_get(dai); ... } (B) has a priority to return dai->name as dlc->dai_name. In most cases card can probe successfully. However it has an issue that ASoC tries to rebind card. Here is a simplified flow for example: | a) Card probes successfully at first | b) One of the component bound to this card is removed for some | reason the component->dev is released | c) That component is re-registered v d) ASoC calls snd_soc_try_rebind_card() a) points dlc->dai_name to dai->name. b) releases all resource of the old DAI. c) creates new DAI structure. In result d) can not use dlc->dai_name to add new created DAI. So it's reasonable that prefer to return dai->driver->name in snd_soc_dai_name_get() because dai->driver is a pre-defined global variable. Also update snd_soc_is_matching_dai() for alignment. Signed-off-by: Chancel Liu <chancel.liu(a)nxp.com> --- changes in v2: - update snd_soc_is_matching_dai() for alignment changes in v3: - resolve conflicts when applying patch, no others changed sound/soc/soc-core.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 507cd3015ff4..1e94edba12eb 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -283,13 +283,13 @@ static int snd_soc_is_matching_dai(const struct snd_soc_dai_link_component *dlc, /* see snd_soc_dai_name_get() */ - if (strcmp(dlc->dai_name, dai->name) == 0) - return 1; - if (dai->driver->name && strcmp(dlc->dai_name, dai->driver->name) == 0) return 1; + if (strcmp(dlc->dai_name, dai->name) == 0) + return 1; + if (dai->component->name && strcmp(dlc->dai_name, dai->component->name) == 0) return 1; @@ -300,12 +300,12 @@ static int snd_soc_is_matching_dai(const struct snd_soc_dai_link_component *dlc, const char *snd_soc_dai_name_get(struct snd_soc_dai *dai) { /* see snd_soc_is_matching_dai() */ - if (dai->name) - return dai->name; - if (dai->driver->name) return dai->driver->name; + if (dai->name) + return dai->name; + if (dai->component->name) return dai->component->name; -- 2.43.0

2 1

[PATCH v3] ASoC: cs-amp-lib: Add KUnit test for calibration helpers
by Richard Fitzgerald 05 Mar '24

05 Mar '24

Add a KUnit test for the cs-amp-lib library. This has test cases for cs_amp_get_efi_calibration_data() and cs_amp_write_cal_coeffs(). A KUNIT_STATIC_STUB_REDIRECT() has been added to cs_amp_get_efi_variable() and cs_amp_write_cal_coeff() so that the KUnit test can redirect these to test harness functions. Much of the testing involves invoking the same function with different parameters, i.e. the number of amps and the amp index within the array. This uses parameterization rather than looping. The idea is to avoid looping over configurations within one test case as that has a higher chance of having a bug that doesn't actually test all the expected cases. Having the test run exactly one configuration, and then tear-down, is less prone to accidentally skipped configurations. Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> --- include/sound/cs-amp-lib.h | 14 + sound/soc/codecs/Kconfig | 13 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cs-amp-lib-test.c | 709 +++++++++++++++++++++++++++++ sound/soc/codecs/cs-amp-lib.c | 18 +- 5 files changed, 754 insertions(+), 2 deletions(-) create mode 100644 sound/soc/codecs/cs-amp-lib-test.c diff --git a/include/sound/cs-amp-lib.h b/include/sound/cs-amp-lib.h index 077fe36885b5..f481148735e1 100644 --- a/include/sound/cs-amp-lib.h +++ b/include/sound/cs-amp-lib.h @@ -49,4 +49,18 @@ int cs_amp_write_cal_coeffs(struct cs_dsp *dsp, const struct cirrus_amp_cal_data *data); int cs_amp_get_efi_calibration_data(struct device *dev, u64 target_uid, int amp_index, struct cirrus_amp_cal_data *out_data); + +struct cs_amp_test_hooks { + efi_status_t (*get_efi_variable)(efi_char16_t *name, + efi_guid_t *guid, + unsigned long *size, + void *buf); + + int (*write_cal_coeff)(struct cs_dsp *dsp, + const struct cirrus_amp_cal_controls *controls, + const char *ctl_name, u32 val); +}; + +extern const struct cs_amp_test_hooks * const cs_amp_test_hooks; + #endif /* CS_AMP_LIB_H */ diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 15f287784d8b..f78ea2f86fa6 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -732,6 +732,19 @@ config SND_SOC_CROS_EC_CODEC config SND_SOC_CS_AMP_LIB tristate +config SND_SOC_CS_AMP_LIB_TEST + tristate "KUnit test for Cirrus Logic cs-amp-lib" + depends on KUNIT + default KUNIT_ALL_TESTS + select SND_SOC_CS_AMP_LIB + help + This builds KUnit tests for the Cirrus Logic common + amplifier library. + For more information on KUnit and unit tests in general, + please refer to the KUnit documentation in + Documentation/dev-tools/kunit/. + If in doubt, say "N". + config SND_SOC_CS35L32 tristate "Cirrus Logic CS35L32 CODEC" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 0fc40640e5d0..7c075539dc47 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -60,6 +60,7 @@ snd-soc-cpcap-objs := cpcap.o snd-soc-cq93vc-objs := cq93vc.o snd-soc-cros-ec-codec-objs := cros_ec_codec.o snd-soc-cs-amp-lib-objs := cs-amp-lib.o +snd-soc-cs-amp-lib-test-objs := cs-amp-lib-test.o snd-soc-cs35l32-objs := cs35l32.o snd-soc-cs35l33-objs := cs35l33.o snd-soc-cs35l34-objs := cs35l34.o @@ -454,6 +455,7 @@ obj-$(CONFIG_SND_SOC_CQ0093VC) += snd-soc-cq93vc.o obj-$(CONFIG_SND_SOC_CPCAP) += snd-soc-cpcap.o obj-$(CONFIG_SND_SOC_CROS_EC_CODEC) += snd-soc-cros-ec-codec.o obj-$(CONFIG_SND_SOC_CS_AMP_LIB) += snd-soc-cs-amp-lib.o +obj-$(CONFIG_SND_SOC_CS_AMP_LIB_TEST) += snd-soc-cs-amp-lib-test.o obj-$(CONFIG_SND_SOC_CS35L32) += snd-soc-cs35l32.o obj-$(CONFIG_SND_SOC_CS35L33) += snd-soc-cs35l33.o obj-$(CONFIG_SND_SOC_CS35L34) += snd-soc-cs35l34.o diff --git a/sound/soc/codecs/cs-amp-lib-test.c b/sound/soc/codecs/cs-amp-lib-test.c new file mode 100644 index 000000000000..15f991b2e16e --- /dev/null +++ b/sound/soc/codecs/cs-amp-lib-test.c @@ -0,0 +1,709 @@ +// SPDX-License-Identifier: GPL-2.0-only +// +// KUnit test for the Cirrus common amplifier library. +// +// Copyright (C) 2024 Cirrus Logic, Inc. and +// Cirrus Logic International Semiconductor Ltd. + +#include <kunit/test.h> +#include <kunit/static_stub.h> +#include <linux/firmware/cirrus/cs_dsp.h> +#include <linux/firmware/cirrus/wmfw.h> +#include <linux/gpio/driver.h> +#include <linux/list.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/random.h> +#include <sound/cs-amp-lib.h> + +struct cs_amp_lib_test_priv { + struct platform_device amp_pdev; + + struct cirrus_amp_efi_data *cal_blob; + struct list_head ctl_write_list; +}; + +struct cs_amp_lib_test_ctl_write_entry { + struct list_head list; + unsigned int value; + char name[16]; +}; + +struct cs_amp_lib_test_param { + int num_amps; + int amp_index; +}; + +static void cs_amp_lib_test_init_dummy_cal_blob(struct kunit *test, int num_amps) +{ + struct cs_amp_lib_test_priv *priv = test->priv; + unsigned int blob_size; + + blob_size = offsetof(struct cirrus_amp_efi_data, data) + + sizeof(struct cirrus_amp_cal_data) * num_amps; + + priv->cal_blob = kunit_kzalloc(test, blob_size, GFP_KERNEL); + KUNIT_ASSERT_NOT_NULL(test, priv->cal_blob); + + priv->cal_blob->size = blob_size; + priv->cal_blob->count = num_amps; + + get_random_bytes(priv->cal_blob->data, sizeof(struct cirrus_amp_cal_data) * num_amps); +} + +static u64 cs_amp_lib_test_get_target_uid(struct kunit *test) +{ + struct cs_amp_lib_test_priv *priv = test->priv; + const struct cs_amp_lib_test_param *param = test->param_value; + u64 uid; + + uid = priv->cal_blob->data[param->amp_index].calTarget[1]; + uid <<= 32; + uid |= priv->cal_blob->data[param->amp_index].calTarget[0]; + + return uid; +} + +/* Redirected get_efi_variable to simulate that the file is too short */ +static efi_status_t cs_amp_lib_test_get_efi_variable_nohead(efi_char16_t *name, + efi_guid_t *guid, + unsigned long *size, + void *buf) +{ + if (!buf) { + *size = offsetof(struct cirrus_amp_efi_data, data) - 1; + return EFI_BUFFER_TOO_SMALL; + } + + return EFI_NOT_FOUND; +} + +/* Should return -EOVERFLOW if the header is larger than the EFI data */ +static void cs_amp_lib_test_cal_data_too_short_test(struct kunit *test) +{ + struct cs_amp_lib_test_priv *priv = test->priv; + struct cirrus_amp_cal_data result_data; + int ret; + + /* Redirect calls to get EFI data */ + kunit_activate_static_stub(test, + cs_amp_test_hooks->get_efi_variable, + cs_amp_lib_test_get_efi_variable_nohead); + + ret = cs_amp_get_efi_calibration_data(&priv->amp_pdev.dev, 0, 0, &result_data); + KUNIT_EXPECT_EQ(test, ret, -EOVERFLOW); + + kunit_deactivate_static_stub(test, cs_amp_test_hooks->get_efi_variable); +} + +/* Redirected get_efi_variable to simulate that the count is larger than the file */ +static efi_status_t cs_amp_lib_test_get_efi_variable_bad_count(efi_char16_t *name, + efi_guid_t *guid, + unsigned long *size, + void *buf) +{ + struct kunit *test = kunit_get_current_test(); + struct cs_amp_lib_test_priv *priv = test->priv; + + if (!buf) { + /* + * Return a size that is shorter than required for the + * declared number of entries. + */ + *size = priv->cal_blob->size - 1; + return EFI_BUFFER_TOO_SMALL; + } + + memcpy(buf, priv->cal_blob, priv->cal_blob->size - 1); + + return EFI_SUCCESS; +} + +/* Should return -EOVERFLOW if the entry count is larger than the EFI data */ +static void cs_amp_lib_test_cal_count_too_big_test(struct kunit *test) +{ + struct cs_amp_lib_test_priv *priv = test->priv; + struct cirrus_amp_cal_data result_data; + int ret; + + cs_amp_lib_test_init_dummy_cal_blob(test, 8); + + /* Redirect calls to get EFI data */ + kunit_activate_static_stub(test, + cs_amp_test_hooks->get_efi_variable, + cs_amp_lib_test_get_efi_variable_bad_count); + + ret = cs_amp_get_efi_calibration_data(&priv->amp_pdev.dev, 0, 0, &result_data); + KUNIT_EXPECT_EQ(test, ret, -EOVERFLOW); + + kunit_deactivate_static_stub(test, cs_amp_test_hooks->get_efi_variable); +} + +/* Redirected get_efi_variable to simulate that the variable not found */ +static efi_status_t cs_amp_lib_test_get_efi_variable_none(efi_char16_t *name, + efi_guid_t *guid, + unsigned long *size, + void *buf) +{ + return EFI_NOT_FOUND; +} + +/* If EFI doesn't contain a cal data variable the result should be -ENOENT */ +static void cs_amp_lib_test_no_cal_data_test(struct kunit *test) +{ + struct cs_amp_lib_test_priv *priv = test->priv; + struct cirrus_amp_cal_data result_data; + int ret; + + /* Redirect calls to get EFI data */ + kunit_activate_static_stub(test, + cs_amp_test_hooks->get_efi_variable, + cs_amp_lib_test_get_efi_variable_none); + + ret = cs_amp_get_efi_calibration_data(&priv->amp_pdev.dev, 0, 0, &result_data); + KUNIT_EXPECT_EQ(test, ret, -ENOENT); + + kunit_deactivate_static_stub(test, cs_amp_test_hooks->get_efi_variable); +} + +/* Redirected get_efi_variable to simulate reading a cal data blob */ +static efi_status_t cs_amp_lib_test_get_efi_variable(efi_char16_t *name, + efi_guid_t *guid, + unsigned long *size, + void *buf) +{ + static const efi_char16_t expected_name[] = L"CirrusSmartAmpCalibrationData"; + static const efi_guid_t expected_guid = + EFI_GUID(0x02f9af02, 0x7734, 0x4233, 0xb4, 0x3d, 0x93, 0xfe, 0x5a, 0xa3, 0x5d, 0xb3); + struct kunit *test = kunit_get_current_test(); + struct cs_amp_lib_test_priv *priv = test->priv; + + KUNIT_EXPECT_NOT_ERR_OR_NULL(test, name); + KUNIT_EXPECT_NOT_ERR_OR_NULL(test, guid); + KUNIT_EXPECT_NOT_ERR_OR_NULL(test, size); + + KUNIT_EXPECT_MEMEQ(test, name, expected_name, sizeof(expected_name)); + KUNIT_EXPECT_MEMEQ(test, guid, &expected_guid, sizeof(expected_guid)); + + if (!buf) { + *size = priv->cal_blob->size; + return EFI_BUFFER_TOO_SMALL; + } + + KUNIT_ASSERT_GE_MSG(test, ksize(buf), priv->cal_blob->size, "Buffer to small"); + + memcpy(buf, priv->cal_blob, priv->cal_blob->size); + + return EFI_SUCCESS; +} + +/* Get cal data block for a given amp, matched by target UID. */ +static void cs_amp_lib_test_get_efi_cal_by_uid_test(struct kunit *test) +{ + struct cs_amp_lib_test_priv *priv = test->priv; + const struct cs_amp_lib_test_param *param = test->param_value; + struct cirrus_amp_cal_data result_data; + u64 target_uid; + int ret; + + cs_amp_lib_test_init_dummy_cal_blob(test, param->num_amps); + + /* Redirect calls to get EFI data */ + kunit_activate_static_stub(test, + cs_amp_test_hooks->get_efi_variable, + cs_amp_lib_test_get_efi_variable); + + target_uid = cs_amp_lib_test_get_target_uid(test); + ret = cs_amp_get_efi_calibration_data(&priv->amp_pdev.dev, target_uid, -1, &result_data); + KUNIT_EXPECT_EQ(test, ret, 0); + + kunit_deactivate_static_stub(test, cs_amp_test_hooks->get_efi_variable); + + KUNIT_EXPECT_EQ(test, result_data.calTarget[0], target_uid & 0xFFFFFFFFULL); + KUNIT_EXPECT_EQ(test, result_data.calTarget[1], target_uid >> 32); + KUNIT_EXPECT_EQ(test, result_data.calTime[0], + priv->cal_blob->data[param->amp_index].calTime[0]); + KUNIT_EXPECT_EQ(test, result_data.calTime[1], + priv->cal_blob->data[param->amp_index].calTime[1]); + KUNIT_EXPECT_EQ(test, result_data.calAmbient, + priv->cal_blob->data[param->amp_index].calAmbient); + KUNIT_EXPECT_EQ(test, result_data.calStatus, + priv->cal_blob->data[param->amp_index].calStatus); + KUNIT_EXPECT_EQ(test, result_data.calR, + priv->cal_blob->data[param->amp_index].calR); +} + +/* Get cal data block for a given amp index without checking target UID. */ +static void cs_amp_lib_test_get_efi_cal_by_index_unchecked_test(struct kunit *test) +{ + struct cs_amp_lib_test_priv *priv = test->priv; + const struct cs_amp_lib_test_param *param = test->param_value; + struct cirrus_amp_cal_data result_data; + int ret; + + cs_amp_lib_test_init_dummy_cal_blob(test, param->num_amps); + + /* Redirect calls to get EFI data */ + kunit_activate_static_stub(test, + cs_amp_test_hooks->get_efi_variable, + cs_amp_lib_test_get_efi_variable); + + ret = cs_amp_get_efi_calibration_data(&priv->amp_pdev.dev, 0, + param->amp_index, &result_data); + KUNIT_EXPECT_EQ(test, ret, 0); + + kunit_deactivate_static_stub(test, cs_amp_test_hooks->get_efi_variable); + + KUNIT_EXPECT_EQ(test, result_data.calTime[0], + priv->cal_blob->data[param->amp_index].calTime[0]); + KUNIT_EXPECT_EQ(test, result_data.calTime[1], + priv->cal_blob->data[param->amp_index].calTime[1]); + KUNIT_EXPECT_EQ(test, result_data.calAmbient, + priv->cal_blob->data[param->amp_index].calAmbient); + KUNIT_EXPECT_EQ(test, result_data.calStatus, + priv->cal_blob->data[param->amp_index].calStatus); + KUNIT_EXPECT_EQ(test, result_data.calR, + priv->cal_blob->data[param->amp_index].calR); +} + +/* Get cal data block for a given amp index with checked target UID. */ +static void cs_amp_lib_test_get_efi_cal_by_index_checked_test(struct kunit *test) +{ + struct cs_amp_lib_test_priv *priv = test->priv; + const struct cs_amp_lib_test_param *param = test->param_value; + struct cirrus_amp_cal_data result_data; + u64 target_uid; + int ret; + + cs_amp_lib_test_init_dummy_cal_blob(test, param->num_amps); + + /* Redirect calls to get EFI data */ + kunit_activate_static_stub(test, + cs_amp_test_hooks->get_efi_variable, + cs_amp_lib_test_get_efi_variable); + + target_uid = cs_amp_lib_test_get_target_uid(test); + ret = cs_amp_get_efi_calibration_data(&priv->amp_pdev.dev, target_uid, + param->amp_index, &result_data); + KUNIT_EXPECT_EQ(test, ret, 0); + + kunit_deactivate_static_stub(test, cs_amp_test_hooks->get_efi_variable); + + KUNIT_EXPECT_EQ(test, result_data.calTime[0], + priv->cal_blob->data[param->amp_index].calTime[0]); + KUNIT_EXPECT_EQ(test, result_data.calTime[1], + priv->cal_blob->data[param->amp_index].calTime[1]); + KUNIT_EXPECT_EQ(test, result_data.calAmbient, + priv->cal_blob->data[param->amp_index].calAmbient); + KUNIT_EXPECT_EQ(test, result_data.calStatus, + priv->cal_blob->data[param->amp_index].calStatus); + KUNIT_EXPECT_EQ(test, result_data.calR, + priv->cal_blob->data[param->amp_index].calR); +} + +/* + * Get cal data block for a given amp index with checked target UID. + * The UID does not match so the result should be -ENOENT. + */ +static void cs_amp_lib_test_get_efi_cal_by_index_uid_mismatch_test(struct kunit *test) +{ + struct cs_amp_lib_test_priv *priv = test->priv; + const struct cs_amp_lib_test_param *param = test->param_value; + struct cirrus_amp_cal_data result_data; + u64 target_uid; + int ret; + + cs_amp_lib_test_init_dummy_cal_blob(test, param->num_amps); + + /* Redirect calls to get EFI data */ + kunit_activate_static_stub(test, + cs_amp_test_hooks->get_efi_variable, + cs_amp_lib_test_get_efi_variable); + + /* Get a target UID that won't match the entry */ + target_uid = ~cs_amp_lib_test_get_target_uid(test); + ret = cs_amp_get_efi_calibration_data(&priv->amp_pdev.dev, target_uid, + param->amp_index, &result_data); + KUNIT_EXPECT_EQ(test, ret, -ENOENT); + + kunit_deactivate_static_stub(test, cs_amp_test_hooks->get_efi_variable); +} + +/* + * Get cal data block for a given amp, where the cal data does not + * specify calTarget so the lookup falls back to using the index + */ +static void cs_amp_lib_test_get_efi_cal_by_index_fallback_test(struct kunit *test) +{ + struct cs_amp_lib_test_priv *priv = test->priv; + const struct cs_amp_lib_test_param *param = test->param_value; + struct cirrus_amp_cal_data result_data; + static const u64 bad_target_uid = 0xBADCA100BABABABAULL; + int i, ret; + + cs_amp_lib_test_init_dummy_cal_blob(test, param->num_amps); + + /* Make all the target values zero so they are ignored */ + for (i = 0; i < priv->cal_blob->count; ++i) { + priv->cal_blob->data[i].calTarget[0] = 0; + priv->cal_blob->data[i].calTarget[1] = 0; + } + + /* Redirect calls to get EFI data */ + kunit_activate_static_stub(test, + cs_amp_test_hooks->get_efi_variable, + cs_amp_lib_test_get_efi_variable); + + ret = cs_amp_get_efi_calibration_data(&priv->amp_pdev.dev, bad_target_uid, + param->amp_index, &result_data); + KUNIT_EXPECT_EQ(test, ret, 0); + + kunit_deactivate_static_stub(test, cs_amp_test_hooks->get_efi_variable); + + KUNIT_EXPECT_EQ(test, result_data.calTime[0], + priv->cal_blob->data[param->amp_index].calTime[0]); + KUNIT_EXPECT_EQ(test, result_data.calTime[1], + priv->cal_blob->data[param->amp_index].calTime[1]); + KUNIT_EXPECT_EQ(test, result_data.calAmbient, + priv->cal_blob->data[param->amp_index].calAmbient); + KUNIT_EXPECT_EQ(test, result_data.calStatus, + priv->cal_blob->data[param->amp_index].calStatus); + KUNIT_EXPECT_EQ(test, result_data.calR, + priv->cal_blob->data[param->amp_index].calR); +} + +/* + * If the target UID isn't present in the cal data, and there isn't an + * index to fall back do, the result should be -ENOENT. + */ +static void cs_amp_lib_test_get_efi_cal_uid_not_found_noindex_test(struct kunit *test) +{ + struct cs_amp_lib_test_priv *priv = test->priv; + struct cirrus_amp_cal_data result_data; + static const u64 bad_target_uid = 0xBADCA100BABABABAULL; + int i, ret; + + cs_amp_lib_test_init_dummy_cal_blob(test, 8); + + /* Make all the target values != bad_target_uid */ + for (i = 0; i < priv->cal_blob->count; ++i) { + priv->cal_blob->data[i].calTarget[0] &= ~(bad_target_uid & 0xFFFFFFFFULL); + priv->cal_blob->data[i].calTarget[1] &= ~(bad_target_uid >> 32); + } + + /* Redirect calls to get EFI data */ + kunit_activate_static_stub(test, + cs_amp_test_hooks->get_efi_variable, + cs_amp_lib_test_get_efi_variable); + + ret = cs_amp_get_efi_calibration_data(&priv->amp_pdev.dev, bad_target_uid, -1, + &result_data); + KUNIT_EXPECT_EQ(test, ret, -ENOENT); + + kunit_deactivate_static_stub(test, cs_amp_test_hooks->get_efi_variable); +} + +/* + * If the target UID isn't present in the cal data, and the index is + * out of range, the result should be -ENOENT. + */ +static void cs_amp_lib_test_get_efi_cal_uid_not_found_index_not_found_test(struct kunit *test) +{ + struct cs_amp_lib_test_priv *priv = test->priv; + struct cirrus_amp_cal_data result_data; + static const u64 bad_target_uid = 0xBADCA100BABABABAULL; + int i, ret; + + cs_amp_lib_test_init_dummy_cal_blob(test, 8); + + /* Make all the target values != bad_target_uid */ + for (i = 0; i < priv->cal_blob->count; ++i) { + priv->cal_blob->data[i].calTarget[0] &= ~(bad_target_uid & 0xFFFFFFFFULL); + priv->cal_blob->data[i].calTarget[1] &= ~(bad_target_uid >> 32); + } + + /* Redirect calls to get EFI data */ + kunit_activate_static_stub(test, + cs_amp_test_hooks->get_efi_variable, + cs_amp_lib_test_get_efi_variable); + + ret = cs_amp_get_efi_calibration_data(&priv->amp_pdev.dev, bad_target_uid, 99, + &result_data); + KUNIT_EXPECT_EQ(test, ret, -ENOENT); + + kunit_deactivate_static_stub(test, cs_amp_test_hooks->get_efi_variable); +} + +/* + * If the target UID isn't given, and the index is out of range, the + * result should be -ENOENT. + */ +static void cs_amp_lib_test_get_efi_cal_no_uid_index_not_found_test(struct kunit *test) +{ + struct cs_amp_lib_test_priv *priv = test->priv; + struct cirrus_amp_cal_data result_data; + int ret; + + cs_amp_lib_test_init_dummy_cal_blob(test, 8); + + /* Redirect calls to get EFI data */ + kunit_activate_static_stub(test, + cs_amp_test_hooks->get_efi_variable, + cs_amp_lib_test_get_efi_variable); + + ret = cs_amp_get_efi_calibration_data(&priv->amp_pdev.dev, 0, 99, &result_data); + KUNIT_EXPECT_EQ(test, ret, -ENOENT); + + kunit_deactivate_static_stub(test, cs_amp_test_hooks->get_efi_variable); +} + +/* If neither the target UID or the index is given the result should be -ENOENT. */ +static void cs_amp_lib_test_get_efi_cal_no_uid_no_index_test(struct kunit *test) +{ + struct cs_amp_lib_test_priv *priv = test->priv; + struct cirrus_amp_cal_data result_data; + int ret; + + cs_amp_lib_test_init_dummy_cal_blob(test, 8); + + /* Redirect calls to get EFI data */ + kunit_activate_static_stub(test, + cs_amp_test_hooks->get_efi_variable, + cs_amp_lib_test_get_efi_variable); + + ret = cs_amp_get_efi_calibration_data(&priv->amp_pdev.dev, 0, -1, &result_data); + KUNIT_EXPECT_EQ(test, ret, -ENOENT); + + kunit_deactivate_static_stub(test, cs_amp_test_hooks->get_efi_variable); +} + +/* + * If the UID is passed as 0 this must not match an entry with an + * unpopulated calTarget + */ +static void cs_amp_lib_test_get_efi_cal_zero_not_matched_test(struct kunit *test) +{ + struct cs_amp_lib_test_priv *priv = test->priv; + struct cirrus_amp_cal_data result_data; + int i, ret; + + cs_amp_lib_test_init_dummy_cal_blob(test, 8); + + /* Make all the target values zero so they are ignored */ + for (i = 0; i < priv->cal_blob->count; ++i) { + priv->cal_blob->data[i].calTarget[0] = 0; + priv->cal_blob->data[i].calTarget[1] = 0; + } + + /* Redirect calls to get EFI data */ + kunit_activate_static_stub(test, + cs_amp_test_hooks->get_efi_variable, + cs_amp_lib_test_get_efi_variable); + + ret = cs_amp_get_efi_calibration_data(&priv->amp_pdev.dev, 0, -1, &result_data); + KUNIT_EXPECT_EQ(test, ret, -ENOENT); + + kunit_deactivate_static_stub(test, cs_amp_test_hooks->get_efi_variable); +} + +static const struct cirrus_amp_cal_controls cs_amp_lib_test_calibration_controls = { + .alg_id = 0x9f210, + .mem_region = WMFW_ADSP2_YM, + .ambient = "CAL_AMBIENT", + .calr = "CAL_R", + .status = "CAL_STATUS", + .checksum = "CAL_CHECKSUM", +}; + +static int cs_amp_lib_test_write_cal_coeff(struct cs_dsp *dsp, + const struct cirrus_amp_cal_controls *controls, + const char *ctl_name, u32 val) +{ + struct kunit *test = kunit_get_current_test(); + struct cs_amp_lib_test_priv *priv = test->priv; + struct cs_amp_lib_test_ctl_write_entry *entry; + + KUNIT_ASSERT_NOT_ERR_OR_NULL(test, ctl_name); + KUNIT_EXPECT_PTR_EQ(test, controls, &cs_amp_lib_test_calibration_controls); + + entry = kunit_kzalloc(test, sizeof(*entry), GFP_KERNEL); + KUNIT_ASSERT_NOT_ERR_OR_NULL(test, entry); + + INIT_LIST_HEAD(&entry->list); + strscpy(entry->name, ctl_name, sizeof(entry->name)); + entry->value = val; + + list_add_tail(&entry->list, &priv->ctl_write_list); + + return 0; +} + +static void cs_amp_lib_test_write_cal_data_test(struct kunit *test) +{ + struct cs_amp_lib_test_priv *priv = test->priv; + struct cs_amp_lib_test_ctl_write_entry *entry; + struct cirrus_amp_cal_data data; + struct cs_dsp *dsp; + int ret; + + dsp = kunit_kzalloc(test, sizeof(*dsp), GFP_KERNEL); + KUNIT_ASSERT_NOT_ERR_OR_NULL(test, dsp); + dsp->dev = &priv->amp_pdev.dev; + + get_random_bytes(&data, sizeof(data)); + + /* Redirect calls to write firmware controls */ + kunit_activate_static_stub(test, + cs_amp_test_hooks->write_cal_coeff, + cs_amp_lib_test_write_cal_coeff); + + ret = cs_amp_write_cal_coeffs(dsp, &cs_amp_lib_test_calibration_controls, &data); + KUNIT_EXPECT_EQ(test, ret, 0); + + kunit_deactivate_static_stub(test, cs_amp_test_hooks->write_cal_coeff); + + KUNIT_EXPECT_EQ(test, list_count_nodes(&priv->ctl_write_list), 4); + + /* Checksum control must be written last */ + entry = list_last_entry(&priv->ctl_write_list, typeof(*entry), list); + KUNIT_EXPECT_STREQ(test, entry->name, cs_amp_lib_test_calibration_controls.checksum); + KUNIT_EXPECT_EQ(test, entry->value, data.calR + 1); + list_del(&entry->list); + + entry = list_first_entry(&priv->ctl_write_list, typeof(*entry), list); + KUNIT_EXPECT_STREQ(test, entry->name, cs_amp_lib_test_calibration_controls.ambient); + KUNIT_EXPECT_EQ(test, entry->value, data.calAmbient); + list_del(&entry->list); + + entry = list_first_entry(&priv->ctl_write_list, typeof(*entry), list); + KUNIT_EXPECT_STREQ(test, entry->name, cs_amp_lib_test_calibration_controls.calr); + KUNIT_EXPECT_EQ(test, entry->value, data.calR); + list_del(&entry->list); + + entry = list_first_entry(&priv->ctl_write_list, typeof(*entry), list); + KUNIT_EXPECT_STREQ(test, entry->name, cs_amp_lib_test_calibration_controls.status); + KUNIT_EXPECT_EQ(test, entry->value, data.calStatus); +} + +static void cs_amp_lib_test_dev_release(struct device *dev) +{ +} + +static int cs_amp_lib_test_case_init(struct kunit *test) +{ + struct cs_amp_lib_test_priv *priv; + int ret; + + KUNIT_ASSERT_NOT_NULL(test, cs_amp_test_hooks); + + priv = kunit_kzalloc(test, sizeof(*priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + + test->priv = priv; + INIT_LIST_HEAD(&priv->ctl_write_list); + + /* Create dummy amp driver dev */ + priv->amp_pdev.name = "cs_amp_lib_test_drv"; + priv->amp_pdev.id = -1; + priv->amp_pdev.dev.release = cs_amp_lib_test_dev_release; + ret = platform_device_register(&priv->amp_pdev); + KUNIT_ASSERT_GE_MSG(test, ret, 0, "Failed to register amp platform device\n"); + + return 0; +} + +static void cs_amp_lib_test_case_exit(struct kunit *test) +{ + struct cs_amp_lib_test_priv *priv = test->priv; + + if (priv->amp_pdev.name) + platform_device_unregister(&priv->amp_pdev); +} + +static const struct cs_amp_lib_test_param cs_amp_lib_test_get_cal_param_cases[] = { + { .num_amps = 2, .amp_index = 0 }, + { .num_amps = 2, .amp_index = 1 }, + + { .num_amps = 3, .amp_index = 0 }, + { .num_amps = 3, .amp_index = 1 }, + { .num_amps = 3, .amp_index = 2 }, + + { .num_amps = 4, .amp_index = 0 }, + { .num_amps = 4, .amp_index = 1 }, + { .num_amps = 4, .amp_index = 2 }, + { .num_amps = 4, .amp_index = 3 }, + + { .num_amps = 5, .amp_index = 0 }, + { .num_amps = 5, .amp_index = 1 }, + { .num_amps = 5, .amp_index = 2 }, + { .num_amps = 5, .amp_index = 3 }, + { .num_amps = 5, .amp_index = 4 }, + + { .num_amps = 6, .amp_index = 0 }, + { .num_amps = 6, .amp_index = 1 }, + { .num_amps = 6, .amp_index = 2 }, + { .num_amps = 6, .amp_index = 3 }, + { .num_amps = 6, .amp_index = 4 }, + { .num_amps = 6, .amp_index = 5 }, + + { .num_amps = 8, .amp_index = 0 }, + { .num_amps = 8, .amp_index = 1 }, + { .num_amps = 8, .amp_index = 2 }, + { .num_amps = 8, .amp_index = 3 }, + { .num_amps = 8, .amp_index = 4 }, + { .num_amps = 8, .amp_index = 5 }, + { .num_amps = 8, .amp_index = 6 }, + { .num_amps = 8, .amp_index = 7 }, +}; + +static void cs_amp_lib_test_get_cal_param_desc(const struct cs_amp_lib_test_param *param, + char *desc) +{ + snprintf(desc, KUNIT_PARAM_DESC_SIZE, "num_amps:%d amp_index:%d", + param->num_amps, param->amp_index); +} + +KUNIT_ARRAY_PARAM(cs_amp_lib_test_get_cal, cs_amp_lib_test_get_cal_param_cases, + cs_amp_lib_test_get_cal_param_desc); + +static struct kunit_case cs_amp_lib_test_cases[] = { + /* Tests for getting calibration data from EFI */ + KUNIT_CASE(cs_amp_lib_test_cal_data_too_short_test), + KUNIT_CASE(cs_amp_lib_test_cal_count_too_big_test), + KUNIT_CASE(cs_amp_lib_test_no_cal_data_test), + KUNIT_CASE(cs_amp_lib_test_get_efi_cal_uid_not_found_noindex_test), + KUNIT_CASE(cs_amp_lib_test_get_efi_cal_uid_not_found_index_not_found_test), + KUNIT_CASE(cs_amp_lib_test_get_efi_cal_no_uid_index_not_found_test), + KUNIT_CASE(cs_amp_lib_test_get_efi_cal_no_uid_no_index_test), + KUNIT_CASE(cs_amp_lib_test_get_efi_cal_zero_not_matched_test), + KUNIT_CASE_PARAM(cs_amp_lib_test_get_efi_cal_by_uid_test, + cs_amp_lib_test_get_cal_gen_params), + KUNIT_CASE_PARAM(cs_amp_lib_test_get_efi_cal_by_index_unchecked_test, + cs_amp_lib_test_get_cal_gen_params), + KUNIT_CASE_PARAM(cs_amp_lib_test_get_efi_cal_by_index_checked_test, + cs_amp_lib_test_get_cal_gen_params), + KUNIT_CASE_PARAM(cs_amp_lib_test_get_efi_cal_by_index_uid_mismatch_test, + cs_amp_lib_test_get_cal_gen_params), + KUNIT_CASE_PARAM(cs_amp_lib_test_get_efi_cal_by_index_fallback_test, + cs_amp_lib_test_get_cal_gen_params), + + /* Tests for writing calibration data */ + KUNIT_CASE(cs_amp_lib_test_write_cal_data_test), + + { } /* terminator */ +}; + +static struct kunit_suite cs_amp_lib_test_suite = { + .name = "snd-soc-cs-amp-lib-test", + .init = cs_amp_lib_test_case_init, + .exit = cs_amp_lib_test_case_exit, + .test_cases = cs_amp_lib_test_cases, +}; + +kunit_test_suite(cs_amp_lib_test_suite); + +MODULE_IMPORT_NS(SND_SOC_CS_AMP_LIB); +MODULE_DESCRIPTION("KUnit test for Cirrus Logic amplifier library"); +MODULE_AUTHOR("Richard Fitzgerald <rf(a)opensource.cirrus.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/cs-amp-lib.c b/sound/soc/codecs/cs-amp-lib.c index 4e2e5157a73f..01ef4db5407d 100644 --- a/sound/soc/codecs/cs-amp-lib.c +++ b/sound/soc/codecs/cs-amp-lib.c @@ -6,6 +6,7 @@ // Cirrus Logic International Semiconductor Ltd. #include <asm/byteorder.h> +#include <kunit/static_stub.h> #include <linux/dev_printk.h> #include <linux/efi.h> #include <linux/firmware/cirrus/cs_dsp.h> @@ -27,6 +28,8 @@ static int cs_amp_write_cal_coeff(struct cs_dsp *dsp, __be32 beval = cpu_to_be32(val); int ret; + KUNIT_STATIC_STUB_REDIRECT(cs_amp_write_cal_coeff, dsp, controls, ctl_name, val); + if (IS_REACHABLE(CONFIG_FW_CS_DSP)) { mutex_lock(&dsp->pwr_lock); cs_ctl = cs_dsp_get_ctl(dsp, ctl_name, controls->mem_region, controls->alg_id); @@ -84,7 +87,7 @@ int cs_amp_write_cal_coeffs(struct cs_dsp *dsp, const struct cirrus_amp_cal_controls *controls, const struct cirrus_amp_cal_data *data) { - if (IS_REACHABLE(CONFIG_FW_CS_DSP)) + if (IS_REACHABLE(CONFIG_FW_CS_DSP) || IS_ENABLED(CONFIG_SND_SOC_CS_AMP_LIB_TEST)) return _cs_amp_write_cal_coeffs(dsp, controls, data); else return -ENODEV; @@ -98,6 +101,8 @@ static efi_status_t cs_amp_get_efi_variable(efi_char16_t *name, { u32 attr; + KUNIT_STATIC_STUB_REDIRECT(cs_amp_get_efi_variable, name, guid, size, buf); + if (IS_ENABLED(CONFIG_EFI)) return efi.get_variable(name, guid, &attr, size, buf); @@ -250,13 +255,22 @@ static int _cs_amp_get_efi_calibration_data(struct device *dev, u64 target_uid, int cs_amp_get_efi_calibration_data(struct device *dev, u64 target_uid, int amp_index, struct cirrus_amp_cal_data *out_data) { - if (IS_ENABLED(CONFIG_EFI)) + if (IS_ENABLED(CONFIG_EFI) || IS_ENABLED(CONFIG_SND_SOC_CS_AMP_LIB_TEST)) return _cs_amp_get_efi_calibration_data(dev, target_uid, amp_index, out_data); else return -ENOENT; } EXPORT_SYMBOL_NS_GPL(cs_amp_get_efi_calibration_data, SND_SOC_CS_AMP_LIB); +static const struct cs_amp_test_hooks cs_amp_test_hook_ptrs = { + .get_efi_variable = cs_amp_get_efi_variable, + .write_cal_coeff = cs_amp_write_cal_coeff, +}; + +const struct cs_amp_test_hooks * const cs_amp_test_hooks = + PTR_IF(IS_ENABLED(CONFIG_SND_SOC_CS_AMP_LIB_TEST), &cs_amp_test_hook_ptrs); +EXPORT_SYMBOL_NS_GPL(cs_amp_test_hooks, SND_SOC_CS_AMP_LIB); + MODULE_DESCRIPTION("Cirrus Logic amplifier library"); MODULE_AUTHOR("Richard Fitzgerald <rf(a)opensource.cirrus.com>"); MODULE_LICENSE("GPL"); -- 2.39.2

2 1

[PATCH v10] The tas2783 is a smart audio amplifier with integrated MIPI SoundWire interface (Version 1.2.1 compliant), I2C, and I2S/TDM interfaces designed for portable applications. An on-chip DSP supports Texas Instruments SmartAmp speaker protection algorithm. The integrated speaker voltage and current sense provides for real-time monitoring of loudspeakers.
by Shenghao Ding 05 Mar '24

05 Mar '24

The ASoC component provides the majority of the functionality of the device, all the audio functions. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- Change in v10: - simplify tasdevice_set_sdw_stream - fixed some Linux coding style - fixed the spelling mistakes - Select left/right channel based on unique id - a longer description has been added - remove unused crc8 in KCONFIG - Invert of RE here means reciprocal of RE - correct indentation - correct the wrong pow calibrated register - correct comment on RESET register - fixed usually lines don't end with an open parenthesis - fixed tasdevice_sdw_shutdown - use defines for function, entity, etc. for readable - remove unnecessary codec_lock in component_probe() - remove struct sdw_stream_data - remove unused macro definitions - use BIT(2) and BIT(1) instead of 0x2 and 0x1 - implement pm_runtime - reg_start is u8, reg_start + i only adds i byte, it should be reg_start + i * 4, because the calibration values - fixed total_sz in tas2783_calibration - add a define for TASDEVICE_REG(0, 0, 0x0a) - rework macro style in header file - Add select CRC32 - rewrite the mute and the return value handling - Add comments on bin file - Add link_id into the fw name to distinguish the same sdw id on different sdw id. - remove unnecessary mutex_lock - use set_active() in tasdevice_io_init and move all other parts in the probe. - move pm_runtime_get_noresume to the probe - fix the return value of putvol function - add put_noidle in tasdevice_sdw_remove - remove unused header file(of.h) - correct the style of the multi-line comment - Set the readable registers in tas2783_readable_register for debugfs - Use new PM macros - Use default put and get inline SOC_SINGLE_RANGE_TLV and SOC_SINGLE_SX_TLV - Use respective %pt to log the calibrated time - Remove unnecessary struct tm - Add trailing comma tasdevice_dapm_widgets and tasdevice_audio_map - Rewrite the return value handling in tasdevice_sdw_probe - use dev_num instead of id.unique_id - rework the log - Add code to check the size of calibrated data in UEFI - Add description of Layout of calibrated Data in UEFI - switch back link_id and unique_id to identify the devcie - unique_id offset as the device index of the calibrated data - update calibrated data layout stored in UEFI, add link_id and unique_id into the calibrated data in case of tas2783 sitting in different SDQW buses. --- sound/soc/codecs/Kconfig | 16 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tas2783-sdw.c | 932 +++++++++++++++++++++++++++++++++ sound/soc/codecs/tas2783.h | 106 ++++ 4 files changed, 1056 insertions(+) create mode 100644 sound/soc/codecs/tas2783-sdw.c create mode 100644 sound/soc/codecs/tas2783.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 59f9742e9ff4..6ba3fd0c0811 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -247,6 +247,7 @@ config SND_SOC_ALL_CODECS imply SND_SOC_TAS2781_COMLIB imply SND_SOC_TAS2781_FMWLIB imply SND_SOC_TAS2781_I2C + imply SND_SOC_TAS2783 imply SND_SOC_TAS5086 imply SND_SOC_TAS571X imply SND_SOC_TAS5720 @@ -1847,6 +1848,21 @@ config SND_SOC_TAS2781_I2C algo coefficient setting, for one, two or even multiple TAS2781 chips. +config SND_SOC_TAS2783 + tristate "Texas Instruments TAS2783 speaker amplifier (sdw)" + depends on SOUNDWIRE + depends on EFI + select REGMAP + select REGMAP_SOUNDWIRE + select CRC32 + select CRC32_SARWATE + help + Enable support for Texas Instruments TAS2783 Smart Amplifier + Digital input mono Class-D and DSP-inside audio power amplifiers. + TAS2783 supports the MIPI SoundWire v1.2 compatible interface + for audio and control data. At the end of calibrated data appends + the crc data with crc32_sarwate for error detection. + config SND_SOC_TAS5086 tristate "Texas Instruments TAS5086 speaker amplifier" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index f53baa2b9565..6afceb126ae3 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -284,6 +284,7 @@ snd-soc-tas2770-objs := tas2770.o snd-soc-tas2781-comlib-objs := tas2781-comlib.o snd-soc-tas2781-fmwlib-objs := tas2781-fmwlib.o snd-soc-tas2781-i2c-objs := tas2781-i2c.o +snd-soc-tas2783-objs := tas2783-sdw.o snd-soc-tfa9879-objs := tfa9879.o snd-soc-tfa989x-objs := tfa989x.o snd-soc-tlv320adc3xxx-objs := tlv320adc3xxx.o @@ -664,6 +665,7 @@ obj-$(CONFIG_SND_SOC_TAS2780) += snd-soc-tas2780.o obj-$(CONFIG_SND_SOC_TAS2781_COMLIB) += snd-soc-tas2781-comlib.o obj-$(CONFIG_SND_SOC_TAS2781_FMWLIB) += snd-soc-tas2781-fmwlib.o obj-$(CONFIG_SND_SOC_TAS2781_I2C) += snd-soc-tas2781-i2c.o +obj-$(CONFIG_SND_SOC_TAS2783) += snd-soc-tas2783.o obj-$(CONFIG_SND_SOC_TAS5086) += snd-soc-tas5086.o obj-$(CONFIG_SND_SOC_TAS571X) += snd-soc-tas571x.o obj-$(CONFIG_SND_SOC_TAS5720) += snd-soc-tas5720.o diff --git a/sound/soc/codecs/tas2783-sdw.c b/sound/soc/codecs/tas2783-sdw.c new file mode 100644 index 000000000000..7bd151942a83 --- /dev/null +++ b/sound/soc/codecs/tas2783-sdw.c @@ -0,0 +1,932 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// ALSA SoC Texas Instruments TAS2783 Audio Smart Amplifier +// +// Copyright (C) 2023 - 2024 Texas Instruments Incorporated +// https://www.ti.com +// +// Author: Baojun Xu <baojun.xu(a)ti.com> +// Kevin Lu <kevin-lu(a)ti.com> +// Shenghao Ding <shenghao-ding(a)ti.com> +// + +#include <asm/unaligned.h> +#include <linux/acpi.h> +#include <linux/crc32.h> +#include <linux/efi.h> +#include <linux/err.h> +#include <linux/firmware.h> +#include <linux/init.h> +#include <linux/module.h> +#include <linux/pm.h> +#include <linux/pm_runtime.h> +#include <linux/regmap.h> +#include <linux/soundwire/sdw.h> +#include <linux/soundwire/sdw_registers.h> +#include <linux/soundwire/sdw_type.h> +#include <sound/pcm_params.h> +#include <sound/sdw.h> +#include <sound/soc.h> +#include <sound/tlv.h> + +#include <sound/tas2781-tlv.h> +#include "tas2783.h" + +static const unsigned int tas2783_cali_reg[] = { + TAS2783_CALIBRATION_RE, /* Resistance */ + TAS2783_CALIBRATION_RE_LOW, /* Low limitation of RE */ + TAS2783_CALIBRATION_INV_RE, /* Reciprocal of RE */ + TAS2783_CALIBRATION_POW, /* RMS Power */ + TAS2783_CALIBRATION_TLIMIT /* Temperature limitation */ +}; + +static const struct reg_default tas2783_reg_defaults[] = { + /* Default values for ROM mode. Activated. */ + { 0x8002, 0x1a }, /* AMP inactive. */ + { 0x8097, 0xc8 }, + { 0x80b5, 0x74 }, + { 0x8099, 0x20 }, + { 0xfe8d, 0x0d }, + { 0xfebe, 0x4a }, + { 0x8230, 0x00 }, + { 0x8231, 0x00 }, + { 0x8232, 0x00 }, + { 0x8233, 0x01 }, + { 0x8418, 0x00 }, /* Set volume to 0 dB. */ + { 0x8419, 0x00 }, + { 0x841a, 0x00 }, + { 0x841b, 0x00 }, + { 0x8428, 0x40 }, /* Unmute channel */ + { 0x8429, 0x00 }, + { 0x842a, 0x00 }, + { 0x842b, 0x00 }, + { 0x8548, 0x00 }, /* Set volume to 0 dB. */ + { 0x8549, 0x00 }, + { 0x854a, 0x00 }, + { 0x854b, 0x00 }, + { 0x8558, 0x40 }, /* Unmute channel */ + { 0x8559, 0x00 }, + { 0x855a, 0x00 }, + { 0x855b, 0x00 }, + { 0x800a, 0x3a }, /* Enable both channel */ + { 0x800e, 0x44 }, + { 0x800f, 0x40 }, + { 0x805c, 0x99 }, + { SDW_SDCA_CTL(TAS2783_FUNC_TYPE_SMART_AMP, TAS2783_SDCA_ENT_FU21, + TAS2783_SDCA_CTL_FU_MUTE, 0), 0 }, + { SDW_SDCA_CTL(TAS2783_FUNC_TYPE_SMART_AMP, TAS2783_SDCA_ENT_FU21, + TAS2783_SDCA_CTL_FU_VOLUME, 0), 0 }, + { SDW_SDCA_CTL(TAS2783_FUNC_TYPE_SMART_AMP, TAS2783_SDCA_ENT_FU23, + TAS2783_SDCA_CTL_FU_MUTE, 0), 0 }, +}; + +static bool tas2783_readable_register(struct device *dev, + unsigned int reg) +{ + switch (reg) { + case 0x8000 ... 0x807F: /* Page 0 */ + return true; + default: + return false; + } +} + +static bool tas2783_volatile_register(struct device *dev, + unsigned int reg) +{ + switch (reg) { + /* + * Only software reset register was volatiled. + * Bit is self clearing. + * 0b = Don't reset + * 1b = reset + */ + case 0x8001: + return true; + default: + return false; + } +} + +static const struct regmap_config tasdevice_regmap = { + .reg_bits = 32, + .val_bits = 8, + .readable_reg = tas2783_readable_register, + .volatile_reg = tas2783_volatile_register, + .max_register = 0x44ffffff, + .reg_defaults = tas2783_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(tas2783_reg_defaults), + .cache_type = REGCACHE_RBTREE, + .use_single_read = true, + .use_single_write = true, +}; + +static const struct snd_kcontrol_new tas2783_snd_controls[] = { + SOC_SINGLE_RANGE_TLV("Digital Volume", TAS2783_DVC_LVL, + 0, 0, 200, 1, dvc_tlv), + SOC_SINGLE_SX_TLV("Amp Gain Volume", TAS2783_AMP_LEVEL, + 1, 0, 20, amp_vol_tlv), +}; + +static void tas2783_apply_calibv1(struct tasdevice_priv *tas_dev, + unsigned int *cali_data) +{ + struct regmap *map = tas_dev->regmap; + u8 *cali_start; + u16 offset; + int ret; + + if (!tas_dev->sdw_peripheral) { + dev_err(tas_dev->dev, "%s: peripheral doesn't exist.\n", + __func__); + return; + } + + offset = tas_dev->sdw_peripheral->id.unique_id - + TAS2783_UNIQUE_ID_BASE; + if (offset >= TAS2783_MAX_DEV_NUM) { + dev_err(tas_dev->dev, "%s: offset(%u) is out of range.\n", + __func__, offset); + return; + } + + /* + * The area saving tas2783 calibrated data is specified by its + * unique_id offset. cali_start is the first address of current + * tas2783's calibrated data. + */ + cali_start = (u8 *)(cali_data + offset * sizeof(tas2783_cali_reg)); + for (int i = 0; i < ARRAY_SIZE(tas2783_cali_reg); i++) { + ret = regmap_bulk_write(map, tas2783_cali_reg[i], + &cali_start[4 * i], 4); + if (ret) { + dev_err(tas_dev->dev, "Cali failed %x:%d\n", + tas2783_cali_reg[i], ret); + break; + } + } +} + +static void tas2783_apply_calibv2(struct tasdevice_priv *tas_dev, + unsigned int *cali_data) +{ + const unsigned int arr_size = ARRAY_SIZE(tas2783_cali_reg); + struct regmap *map = tas_dev->regmap; + unsigned int dev_sum = cali_data[1], i, j, k; + u8 *cali_start; + u16 dev_info; + int ret; + + if (!tas_dev->sdw_peripheral) { + dev_err(tas_dev->dev, "%s: peripheral doesn't exist.\n", + __func__); + return; + } + + dev_info = tas_dev->sdw_peripheral->bus->link_id | + tas_dev->sdw_peripheral->id.unique_id << 16; + + /* + * The area saving tas2783 calibrated data is specified by its + * unique_id offset. cali_start is the first address of current + * tas2783's calibrated data. + */ + cali_start = (u8 *)&cali_data[3]; + for (i = 0; i < dev_sum; i++) { + k = i * (arr_size + 1) + 3; + if (dev_info != cali_data[k]) { + for (j = 0; j < arr_size; j++) { + k = 4 * (k + 1 + j); + ret = regmap_bulk_write(map, + tas2783_cali_reg[j], + &cali_start[k], 4); + if (ret) { + dev_err(tas_dev->dev, + "Cali failed %x:%d\n", + tas2783_cali_reg[j], ret); + break; + } + } + break; + } + } +} + +/* + * Load the calibration data, including speaker impedance, f0, etc. + * Calibration is done by the manufacturer in the factory. The calibration + * data are used by the algorithm for calculating the speaker temperature, + * speaker membrane excursion and f0 in real time during playback. + * The DSP will work with default data values if calibrated data are + * missing or are invalid. + * Layout of calibrated Data in UEFI(total 256 bytes): + * V2: + * ChipID (2783, 4 bytes) + * Device-Sum (4 bytes) + * TimeStamp of Calibration (4 bytes) + * for (i = 0; i < Device-Sum; i++) { + * Device #i info () { + * SDW link id (2bytes) + * SDW unique_id (2bytes) + * } + * Calibrated Data of Device #i (20 bytes) + * } + * CRC (4 bytes) + * Reserved (the rest) + * Or + * V1: + * Calibrated Data of Dev 0(unique_id offset 0) (20 bytes) + * Calibrated Data of Dev 1(unique_id offset 1) (20 bytes) + * Calibrated Data of Dev 2(unique_id offset 2) (20 bytes) + * Calibrated Data of Dev 3(unique_id offset 3) (20 bytes) + * Calibrated Data of Dev 4(unique_id offset 4) (20 bytes) + * Calibrated Data of Dev 5(unique_id offset 5) (20 bytes) + * Calibrated Data of Dev 6(unique_id offset 6) (20 bytes) + * Calibrated Data of Dev 7(unique_id offset 7) (20 bytes) + * TimeStamp of Calibration (4 bytes) + * CRC (4 bytes) + * Reserved (88 bytes) + */ +static void tas2783_calibration(struct tasdevice_priv *tas_dev) +{ + efi_guid_t efi_guid = EFI_GUID(0x1f52d2a1, 0xbb3a, 0x457d, 0xbc, + 0x09, 0x43, 0xa3, 0xf4, 0x31, 0x0a, 0x92); + static efi_char16_t efi_name[] = L"CALI_DATA"; + struct calibration_data cali_data; + unsigned int *tmp_val; + unsigned int crc; + efi_status_t status; + + cali_data.total_sz = 0; + + status = efi.get_variable(efi_name, &efi_guid, NULL, + &cali_data.total_sz, NULL); + if (status == EFI_BUFFER_TOO_SMALL + && cali_data.total_sz < TAS2783_MAX_CALIDATA_SIZE) { + status = efi.get_variable(efi_name, &efi_guid, NULL, + &cali_data.total_sz, + cali_data.data); + dev_dbg(tas_dev->dev, "%s: cali get %lx bytes result:%ld\n", + __func__, cali_data.total_sz, status); + } + if (status != 0) { + /* Failed got calibration data from EFI. */ + dev_dbg(tas_dev->dev, "No calibration data in UEFI."); + return; + } + + tmp_val = (unsigned int *)cali_data.data; + + if (tmp_val[0] == 2783) { + /* Calibrated Data V2 */ + unsigned int dev_sum = tmp_val[1]; + + if (dev_sum > TAS2783_MAX_DEV_NUM || + dev_sum == 0) { + dev_dbg(tas_dev->dev, "No dev in calibrated data V2."); + return; + } + crc = crc32(~0, cali_data.data, 12 + dev_sum * 24) ^ ~0; + if (crc == tmp_val[3 + dev_sum * 6]) { + tas2783_apply_calibv2(tas_dev, tmp_val); + dev_dbg(tas_dev->dev, "V2: %ptTs", &tmp_val[40]); + } else { + dev_dbg(tas_dev->dev, + "V2: CRC 0x%08x not match 0x%08x\n", + crc, tmp_val[41]); + } + } else { + /* Calibrated Data V1 */ + /* 8 devs * 20 bytes calibrated data/dev + 4 bytes Timestamp */ + crc = crc32(~0, cali_data.data, 164) ^ ~0; + if (crc == tmp_val[41]) { + /* Date and time of when calibration was done. */ + tas2783_apply_calibv1(tas_dev, tmp_val); + dev_dbg(tas_dev->dev, "V1: %ptTs", &tmp_val[40]); + } else { + dev_dbg(tas_dev->dev, + "V1: CRC 0x%08x not match 0x%08x\n", + crc, tmp_val[41]); + } + } +} + +static void tasdevice_dspfw_ready(const struct firmware *fmw, + void *context) +{ + struct tasdevice_priv *tas_dev = + (struct tasdevice_priv *) context; + struct tas2783_firmware_node *p; + struct regmap *map = tas_dev->regmap; + unsigned char *buf = NULL; + int offset = 0, img_sz; + int ret; + + if (!fmw || !fmw->data) { + dev_warn(tas_dev->dev, + "%s: failed to read %s: work in bypass-dsp mode.\n", + __func__, tas_dev->dspfw_binaryname); + return; + } + buf = (unsigned char *)fmw->data; + + img_sz = get_unaligned_le32(&buf[offset]); + offset += sizeof(img_sz); + if (img_sz != fmw->size) { + dev_warn(tas_dev->dev, "%s: size not matching, %d %u.", + __func__, (int)fmw->size, img_sz); + return; + } + + while (offset < img_sz) { + p = (struct tas2783_firmware_node *)(buf + offset); + if (p->length > 1) { + ret = regmap_bulk_write(map, p->download_addr, + buf + offset + sizeof(unsigned int) * 5, p->length); + } else { + ret = regmap_write(map, p->download_addr, + *(buf + offset + sizeof(unsigned int) * 5)); + } + + if (ret != 0) { + dev_dbg(tas_dev->dev, + "%s: load FW fail: %d, work in bypass.\n", + __func__, ret); + return; + } + offset += sizeof(unsigned int) * 5 + p->length; + } + + tas2783_calibration(tas_dev); +} + +static const struct snd_soc_dapm_widget tasdevice_dapm_widgets[] = { + SND_SOC_DAPM_AIF_IN("ASI", "ASI Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("ASI OUT", "ASI Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_OUTPUT("OUT"), + SND_SOC_DAPM_INPUT("DMIC"), +}; + +static const struct snd_soc_dapm_route tasdevice_audio_map[] = { + {"OUT", NULL, "ASI"}, + {"ASI OUT", NULL, "DMIC"}, +}; + +static int tasdevice_set_sdw_stream(struct snd_soc_dai *dai, + void *sdw_stream, int direction) +{ + snd_soc_dai_dma_data_set(dai, direction, sdw_stream); + + return 0; +} + +static void tasdevice_sdw_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + snd_soc_dai_set_dma_data(dai, substream, NULL); +} + +static int tasdevice_sdw_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct tasdevice_priv *tas_priv = + snd_soc_component_get_drvdata(component); + struct sdw_stream_config stream_config = {0}; + struct sdw_port_config port_config = {0}; + struct sdw_stream_runtime *sdw_stream; + int ret; + + dev_dbg(dai->dev, "%s: dai_name %s", __func__, dai->name); + sdw_stream = snd_soc_dai_get_dma_data(dai, substream); + + if (!sdw_stream) + return -EINVAL; + + if (!tas_priv->sdw_peripheral) + return -EINVAL; + + /* SoundWire specific configuration */ + snd_sdw_params_to_config(substream, params, + &stream_config, &port_config); + + /* port 1 for playback */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + port_config.num = 1; + else + port_config.num = 2; + + ret = sdw_stream_add_slave(tas_priv->sdw_peripheral, + &stream_config, &port_config, 1, sdw_stream); + if (ret) { + dev_err(dai->dev, "%s: Unable to configure port\n", __func__); + return ret; + } + + dev_dbg(dai->dev, "%s: format: %i rate: %i\n", __func__, + params_format(params), params_rate(params)); + + return 0; +} + +static int tasdevice_sdw_pcm_hw_free(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct tasdevice_priv *tas_priv = + snd_soc_component_get_drvdata(component); + struct sdw_stream_runtime *sdw_stream = + snd_soc_dai_get_dma_data(dai, substream); + + if (!tas_priv->sdw_peripheral) + return -EINVAL; + + sdw_stream_remove_slave(tas_priv->sdw_peripheral, sdw_stream); + + return 0; +} + +static int tasdevice_mute(struct snd_soc_dai *dai, int mute, + int direction) +{ + struct snd_soc_component *component = dai->component; + struct tasdevice_priv *tas_dev = + snd_soc_component_get_drvdata(component); + struct regmap *map = tas_dev->regmap; + int ret; + + dev_dbg(tas_dev->dev, "%s: %d.\n", __func__, mute); + + if (mute) { + if (direction == SNDRV_PCM_STREAM_CAPTURE) { + ret = regmap_update_bits(map, TAS2873_REG_PWR_CTRL, + TAS2783_REG_AEF_MASK, + TAS2783_REG_AEF_INACTIVE); + if (ret) + dev_err(tas_dev->dev, + "%s: Disable AEF failed.\n", __func__); + } else { + /* FU23 mute (0x40400108) */ + ret = regmap_write(map, + SDW_SDCA_CTL(TAS2783_FUNC_TYPE_SMART_AMP, + TAS2783_SDCA_ENT_FU23, + TAS2783_SDCA_CTL_FU_MUTE, 0), 1); + if (ret) { + dev_err(tas_dev->dev, + "%s: FU23 mute failed.\n", __func__); + goto out; + } + /* + * Both playback and echo data will be shutdown in + * playback stream. + */ + ret = regmap_update_bits(map, TAS2873_REG_PWR_CTRL, + TAS2783_REG_PWR_MODE_MASK | + TAS2783_REG_AEF_MASK, + TAS2783_REG_PWR_MODE_ACTIVE | + TAS2783_REG_PWR_MODE_SW_PWD); + if (ret) { + dev_err(tas_dev->dev, + "%s: PWR&AEF shutdown failed.\n", + __func__); + goto out; + } + tas_dev->pstream = false; + } + } else { + /* FU23 Unmute, 0x40400108. */ + if (direction == SNDRV_PCM_STREAM_PLAYBACK) { + ret = regmap_write(map, + SDW_SDCA_CTL(TAS2783_FUNC_TYPE_SMART_AMP, + TAS2783_SDCA_ENT_FU23, + TAS2783_SDCA_CTL_FU_MUTE, 0), 0); + if (ret) { + dev_err(tas_dev->dev, + "%s: FU23 Unmute failed.\n", __func__); + goto out; + } + ret = regmap_update_bits(map, TAS2873_REG_PWR_CTRL, + TAS2783_REG_PWR_MODE_MASK, + TAS2783_REG_PWR_MODE_ACTIVE); + if (ret) { + dev_err(tas_dev->dev, + "%s: PWR Unmute failed.\n", __func__); + goto out; + } + tas_dev->pstream = true; + } else { + /* Capture stream is the echo ref data for voice. + * Without playback, it can't be active. + */ + if (tas_dev->pstream == true) { + ret = regmap_update_bits(map, + TAS2873_REG_PWR_CTRL, + TAS2783_REG_AEF_MASK, + TAS2783_REG_AEF_ACTIVE); + if (ret) { + dev_err(tas_dev->dev, + "%s: AEF enable failed.\n", + __func__); + goto out; + } + } else { + dev_err(tas_dev->dev, + "%s: No playback, no AEF!", __func__); + ret = -EINVAL; + } + } + } +out: + if (ret) + dev_err(tas_dev->dev, "Mute or unmute %d failed %d.\n", + mute, ret); + + return ret; +} + +static const struct snd_soc_dai_ops tasdevice_dai_ops = { + .mute_stream = tasdevice_mute, + .hw_params = tasdevice_sdw_hw_params, + .hw_free = tasdevice_sdw_pcm_hw_free, + .set_stream = tasdevice_set_sdw_stream, + .shutdown = tasdevice_sdw_shutdown, +}; + +static struct snd_soc_dai_driver tasdevice_dai_driver[] = { + { + .name = "tas2783-codec", + .id = 0, + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 4, + .rates = TAS2783_DEVICE_RATES, + .formats = TAS2783_DEVICE_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 4, + .rates = TAS2783_DEVICE_RATES, + .formats = TAS2783_DEVICE_FORMATS, + }, + .ops = &tasdevice_dai_ops, + .symmetric_rate = 1, + }, +}; + +static int tasdevice_comp_probe(struct snd_soc_component *comp) +{ + struct tasdevice_priv *tas_dev = snd_soc_component_get_drvdata(comp); + acpi_handle handle = ACPI_HANDLE(tas_dev->dev); + const struct firmware *fw_entry = NULL; + const char *sub = NULL; + int ret; + + if (handle) { + sub = acpi_get_subsystem_id(handle); + if (IS_ERR(sub)) + sub = NULL; + } + + tas_dev->component = comp; + + /* + * Each tas2783 in the system has its own dspfw. + */ + if (comp->name_prefix) { + /* + * name_prefix.bin stores the dsp firmware including speaker + * protection algorithm, audio acoustic algorithm, speaker + * characters and algorithm params, it must be copied into + * firmware folder. + */ + scnprintf(tas_dev->dspfw_binaryname, + TAS2783_DSPFW_FILENAME_LEN, "%s-tas2783.bin", + comp->name_prefix); + } else { + /* Compatible with the previous naming rule */ + if (sub) { + /* + * subsys_id-link_id[0,1,...,N]-unique_id[1,...,4].bin + * stores the dsp firmware including speaker protection + * algorithm, audio acoustic algorithm, speaker + * characters and algorithm params, it must be copied + * into firmware folder. + */ + scnprintf(tas_dev->dspfw_binaryname, + TAS2783_DSPFW_FILENAME_LEN, + "%s-%d-%d.bin", sub, + tas_dev->sdw_peripheral->bus->link_id, + tas_dev->sdw_peripheral->id.unique_id); + } else { + /* + * tas2783-link_id[0,1,...,N]-unique_id[1,...,4].bin + * stores the dsp firmware including speaker protection + * algorithm, audio acoustic algorithm, speaker + * characters and algorithm params, it must be copied + * into firmware folder. + */ + scnprintf(tas_dev->dspfw_binaryname, + TAS2783_DSPFW_FILENAME_LEN, + "tas2783-%d-%d.bin", + tas_dev->sdw_peripheral->bus->link_id, + tas_dev->sdw_peripheral->id.unique_id); + } + } + + ret = request_firmware(&fw_entry, tas_dev->dspfw_binaryname, + tas_dev->dev); + if (ret) { + dev_err(tas_dev->dev, + "%s: request_firmware %x open status: %d.\n", __func__, + tas_dev->sdw_peripheral->id.unique_id, ret); + goto out; + } + + tasdevice_dspfw_ready(fw_entry, tas_dev); + +out: + if (fw_entry) + release_firmware(fw_entry); + return 0; +} + +static const struct snd_soc_component_driver + soc_codec_driver_tasdevice = { + .probe = tasdevice_comp_probe, + .controls = tas2783_snd_controls, + .num_controls = ARRAY_SIZE(tas2783_snd_controls), + .dapm_widgets = tasdevice_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(tasdevice_dapm_widgets), + .dapm_routes = tasdevice_audio_map, + .num_dapm_routes = ARRAY_SIZE(tasdevice_audio_map), + .idle_bias_on = 1, + .endianness = 1, +}; + +static int tasdevice_init(struct tasdevice_priv *tas_dev) +{ + int ret; + + dev_set_drvdata(tas_dev->dev, tas_dev); + + ret = devm_snd_soc_register_component(tas_dev->dev, + &soc_codec_driver_tasdevice, + tasdevice_dai_driver, ARRAY_SIZE(tasdevice_dai_driver)); + if (ret) { + dev_err(tas_dev->dev, "%s: codec register error:%d.\n", + __func__, ret); + return ret; + } + + /* set autosuspend parameters */ + pm_runtime_set_autosuspend_delay(tas_dev->dev, 3000); + pm_runtime_use_autosuspend(tas_dev->dev); + + /* make sure the device does not suspend immediately */ + pm_runtime_mark_last_busy(tas_dev->dev); + pm_runtime_enable(tas_dev->dev); + + return 0; +} + +static int tasdevice_read_prop(struct sdw_slave *slave) +{ + struct sdw_slave_prop *prop = &slave->prop; + struct sdw_dpn_prop *dpn; + unsigned long addr; + int nval, i, j; + u32 bit; + + prop->scp_int1_mask = SDW_SCP_INT1_BUS_CLASH | SDW_SCP_INT1_PARITY; + prop->quirks = SDW_SLAVE_QUIRKS_INVALID_INITIAL_PARITY; + + prop->paging_support = true; + + /* first we need to allocate memory for set bits in port lists */ + prop->source_ports = BIT(2); /* BITMAP: 00000100 */ + prop->sink_ports = BIT(1); /* BITMAP: 00000010 */ + + nval = hweight32(prop->source_ports); + prop->src_dpn_prop = devm_kcalloc(&slave->dev, nval, + sizeof(*prop->src_dpn_prop), GFP_KERNEL); + if (!prop->src_dpn_prop) + return -ENOMEM; + + i = 0; + dpn = prop->src_dpn_prop; + addr = prop->source_ports; + for_each_set_bit(bit, &addr, 32) { + dpn[i].num = bit; + dpn[i].type = SDW_DPN_FULL; + dpn[i].simple_ch_prep_sm = true; + dpn[i].ch_prep_timeout = 10; + i++; + } + + /* do this again for sink now */ + nval = hweight32(prop->sink_ports); + prop->sink_dpn_prop = devm_kcalloc(&slave->dev, nval, + sizeof(*prop->sink_dpn_prop), GFP_KERNEL); + if (!prop->sink_dpn_prop) + return -ENOMEM; + + j = 0; + dpn = prop->sink_dpn_prop; + addr = prop->sink_ports; + for_each_set_bit(bit, &addr, 32) { + dpn[j].num = bit; + dpn[j].type = SDW_DPN_FULL; + dpn[j].simple_ch_prep_sm = true; + dpn[j].ch_prep_timeout = 10; + j++; + } + + /* set the timeout values */ + prop->clk_stop_timeout = 20; + + return 0; +} + +static int tasdevice_io_init(struct device *dev, + struct sdw_slave *slave) +{ + struct tasdevice_priv *tas_priv = dev_get_drvdata(dev); + int ret; + + regcache_cache_only(tas_priv->regmap, false); + + if (tas_priv->first_hw_init) { + regcache_cache_bypass(tas_priv->regmap, true); + } else { + /* + * PM runtime is only enabled when a Slave reports as Attached + * Update count of parent 'active' children. + */ + pm_runtime_set_active(&slave->dev); + } + + /* sw reset */ + ret = regmap_write(tas_priv->regmap, TAS2873_REG_SWRESET, + TAS2873_REG_SWRESET_RESET); + if (ret) { + dev_err(tas_priv->dev, "Reset failed.\n"); + goto out; + } + + if (tas_priv->first_hw_init) { + regcache_cache_bypass(tas_priv->regmap, false); + regcache_mark_dirty(tas_priv->regmap); + } + + tas_priv->first_hw_init = true; + tas_priv->hw_init = true; + +out: + return ret; +} + +static int tasdevice_update_status(struct sdw_slave *slave, + enum sdw_slave_status status) +{ + struct tasdevice_priv *tas_priv = dev_get_drvdata(&slave->dev); + + /* Update the status */ + tas_priv->status = status; + + if (status == SDW_SLAVE_UNATTACHED) + tas_priv->hw_init = false; + /* + * Perform initialization only if slave status + * is present and hw_init flag is false + */ + if (tas_priv->hw_init || tas_priv->status != SDW_SLAVE_ATTACHED) + return 0; + + /* perform I/O transfers required for Slave initialization */ + return tasdevice_io_init(&slave->dev, slave); +} + +/* + * slave_ops: callbacks for get_clock_stop_mode, clock_stop and + * port_prep are not defined for now + */ +static const struct sdw_slave_ops tasdevice_sdw_ops = { + .read_prop = tasdevice_read_prop, + .update_status = tasdevice_update_status, +}; + +static int tasdevice_sdw_probe(struct sdw_slave *peripheral, + const struct sdw_device_id *id) +{ + struct device *dev = &peripheral->dev; + struct tasdevice_priv *tas_dev; + + tas_dev = devm_kzalloc(dev, sizeof(*tas_dev), GFP_KERNEL); + if (!tas_dev) + return -ENOMEM; + + tas_dev->dev = dev; + tas_dev->chip_id = id->driver_data; + tas_dev->sdw_peripheral = peripheral; + /* + * Mark hw_init to false + * HW init will be performed when device reports present + */ + tas_dev->hw_init = false; + tas_dev->first_hw_init = false; + + dev_set_drvdata(dev, tas_dev); + + tas_dev->regmap = devm_regmap_init_sdw(peripheral, &tasdevice_regmap); + if (IS_ERR(tas_dev->regmap)) + return dev_err_probe(dev, PTR_ERR(tas_dev->regmap), + "Failed devm_regmap_init_sdw."); + + return tasdevice_init(tas_dev); +} + +static int tasdevice_sdw_remove(struct sdw_slave *peripheral) +{ + struct tasdevice_priv *tas_dev = dev_get_drvdata(&peripheral->dev); + + if (tas_dev->first_hw_init) + pm_runtime_disable(tas_dev->dev); + + return 0; +} + +static const struct sdw_device_id tasdevice_sdw_id[] = { + SDW_SLAVE_ENTRY(0x0102, 0x0000, 0), + {} +}; +MODULE_DEVICE_TABLE(sdw, tasdevice_sdw_id); + +static int __maybe_unused tas2783_sdca_dev_suspend(struct device *dev) +{ + struct tasdevice_priv *tas_priv = dev_get_drvdata(dev); + + if (!tas_priv->hw_init) + return 0; + + regcache_cache_only(tas_priv->regmap, true); + + return 0; +} + +#define TAS2783_PROBE_TIMEOUT_MS 5000 + +static int __maybe_unused tas2783_sdca_dev_resume(struct device *dev) +{ + struct sdw_slave *slave = dev_to_sdw_dev(dev); + struct tasdevice_priv *tas_priv = dev_get_drvdata(dev); + unsigned long time; + + if (!tas_priv->first_hw_init) + return 0; + + if (!slave->unattach_request) + goto regmap_sync; + + time = wait_for_completion_timeout(&slave->initialization_complete, + msecs_to_jiffies(TAS2783_PROBE_TIMEOUT_MS)); + if (!time) { + dev_err(&slave->dev, "Init not complete, timed out\n"); + sdw_show_ping_status(slave->bus, true); + + return -ETIMEDOUT; + } + +regmap_sync: + slave->unattach_request = 0; + regcache_cache_only(tas_priv->regmap, false); + regcache_sync(tas_priv->regmap); + + return 0; +} + +static const struct dev_pm_ops tas2783_sdca_pm = { + SYSTEM_SLEEP_PM_OPS(tas2783_sdca_dev_suspend, + tas2783_sdca_dev_resume) + RUNTIME_PM_OPS(tas2783_sdca_dev_suspend, + tas2783_sdca_dev_resume, NULL) +}; + +static struct sdw_driver tasdevice_sdw_driver = { + .driver = { + .name = "slave-tas2783", + .pm = &tas2783_sdca_pm, + }, + .probe = tasdevice_sdw_probe, + .remove = tasdevice_sdw_remove, + .ops = &tasdevice_sdw_ops, + .id_table = tasdevice_sdw_id, +}; + +module_sdw_driver(tasdevice_sdw_driver); + +MODULE_AUTHOR("Baojun Xu <baojun.xu(a)ti.com>"); +MODULE_AUTHOR("Shenghao Ding <shenghao-ding(a)ti.com>"); +MODULE_DESCRIPTION("ASoC TAS2783 SoundWire Driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/tas2783.h b/sound/soc/codecs/tas2783.h new file mode 100644 index 000000000000..631776dcec4c --- /dev/null +++ b/sound/soc/codecs/tas2783.h @@ -0,0 +1,106 @@ +/* SPDX-License-Identifier: GPL-2.0 + * + * ALSA SoC Texas Instruments TAS2783 Audio Smart Amplifier + * + * Copyright (C) 2023 - 2024 Texas Instruments Incorporated + * https://www.ti.com + * + * Author: Baojun Xu <baojun.xu(a)ti.com> + * Shenghao Ding <shenghao-ding(a)ti.com> + */ + +#ifndef __TAS2783_H__ +#define __TAS2783_H__ + +/* Unique id start */ +#define TAS2783_UNIQUE_ID_BASE 0x08 +#define TAS2783_MAX_DEV_NUM 8 + +#define TAS2783_DEVICE_RATES (SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000 | \ + SNDRV_PCM_RATE_96000 | \ + SNDRV_PCM_RATE_88200) + +#define TAS2783_DEVICE_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE) + +/* BOOK, PAGE Control Register */ +#define TASDEVICE_REG(book, page, reg) ((book * 256 * 256) + 0x8000 + \ + (page * 128) + reg) + +/*Software Reset */ +#define TAS2873_REG_SWRESET TASDEVICE_REG(0x0, 0x00, 0x01) +#define TAS2873_REG_SWRESET_MASK BIT(0) +#define TAS2873_REG_SWRESET_RESET BIT(0) + +#define TAS2873_REG_PWR_CTRL TASDEVICE_REG(0x0, 0x00, 0x02) +#define TAS2783_REG_AEF_MASK GENMASK(4, 3) +#define TAS2783_REG_AEF_ACTIVE 0 +#define TAS2783_REG_AEF_INACTIVE GENMASK(4, 3) +#define TAS2783_REG_PWR_MODE_MASK GENMASK(2, 0) +#define TAS2783_REG_PWR_MODE_ACTIVE 0 +#define TAS2783_REG_PWR_MODE_SW_PWD BIT(1) + +#define TAS2783_REG_TDM_RX_CFG TASDEVICE_REG(0x0, 0x00, 0x0a) + +/* Volume control */ +#define TAS2783_DVC_LVL TASDEVICE_REG(0x0, 0x00, 0x1a) +#define TAS2783_AMP_LEVEL TASDEVICE_REG(0x0, 0x00, 0x03) +#define TAS2783_AMP_LEVEL_MASK GENMASK(5, 1) + +/* Calibration data */ +#define TAS2783_CALIBRATION_RE TASDEVICE_REG(0x0, 0x17, 0x74) +#define TAS2783_CALIBRATION_RE_LOW TASDEVICE_REG(0x0, 0x18, 0x14) +#define TAS2783_CALIBRATION_INV_RE TASDEVICE_REG(0x0, 0x18, 0x0c) +#define TAS2783_CALIBRATION_POW TASDEVICE_REG(0x0, 0x13, 0x70) +#define TAS2783_CALIBRATION_TLIMIT TASDEVICE_REG(0x0, 0x18, 0x7c) + +/* TAS2783 SDCA Control - function number */ +#define TAS2783_FUNC_TYPE_SMART_AMP 0x01 + +/* TAS2783 SDCA entity */ +#define TAS2783_SDCA_ENT_FU21 0x01 +#define TAS2783_SDCA_ENT_FU23 0x02 + +/* TAS2783 SDCA control */ +#define TAS2783_SDCA_CTL_REQ_POWER_STATE 0x01 +#define TAS2783_SDCA_CTL_FU_MUTE 0x01 +#define TAS2783_SDCA_CTL_FU_VOLUME 0x02 +#define TAS2783_SDCA_CTL_UDMPU_CLUSTER 0x10 + +#define TAS2783_DEVICE_CHANNEL_LEFT 1 +#define TAS2783_DEVICE_CHANNEL_RIGHT 2 +#define TAS2783_DSPFW_FILENAME_LEN 64 + +#define TAS2783_MAX_CALIDATA_SIZE 256 + +struct tas2783_firmware_node { + unsigned int vendor_id; + unsigned int file_id; + unsigned int version_id; + unsigned int length; + unsigned int download_addr; +}; + +struct calibration_data { + unsigned long total_sz; + unsigned char data[TAS2783_MAX_CALIDATA_SIZE]; +}; + +struct tasdevice_priv { + struct snd_soc_component *component; + struct sdw_slave *sdw_peripheral; + enum sdw_slave_status status; + struct sdw_bus_params params; + struct regmap *regmap; + struct device *dev; + unsigned char dspfw_binaryname[TAS2783_DSPFW_FILENAME_LEN]; + unsigned char dev_name[32]; + unsigned int chip_id; + bool pstream; + bool hw_init; + bool first_hw_init; +}; + +#endif /*__TAS2783_H__ */ -- 2.34.1

2 1

[PATCH] mfd: cs42l43: Fix wrong GPIO_FN_SEL and SPI_CLK_CONFIG1 defaults
by Maciej Strozek 05 Mar '24

05 Mar '24

Two regs have wrong values in existing fields, change them to match the datasheet. Fixes: ace6d1448138 ("mfd: cs42l43: Add support for cs42l43 core driver") Signed-off-by: Maciej Strozek <mstrozek(a)opensource.cirrus.com> --- drivers/mfd/cs42l43.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/drivers/mfd/cs42l43.c b/drivers/mfd/cs42l43.c index 73c88ee6a866..1cea3f8f467d 100644 --- a/drivers/mfd/cs42l43.c +++ b/drivers/mfd/cs42l43.c @@ -84,7 +84,7 @@ const struct reg_default cs42l43_reg_default[CS42L43_N_DEFAULTS] = { { CS42L43_DRV_CTRL_5, 0x136C00C0 }, { CS42L43_GPIO_CTRL1, 0x00000707 }, { CS42L43_GPIO_CTRL2, 0x00000000 }, - { CS42L43_GPIO_FN_SEL, 0x00000000 }, + { CS42L43_GPIO_FN_SEL, 0x00000004 }, { CS42L43_MCLK_SRC_SEL, 0x00000000 }, { CS42L43_SAMPLE_RATE1, 0x00000003 }, { CS42L43_SAMPLE_RATE2, 0x00000003 }, @@ -217,7 +217,7 @@ const struct reg_default cs42l43_reg_default[CS42L43_N_DEFAULTS] = { { CS42L43_CTRL_REG, 0x00000006 }, { CS42L43_FDIV_FRAC, 0x40000000 }, { CS42L43_CAL_RATIO, 0x00000080 }, - { CS42L43_SPI_CLK_CONFIG1, 0x00000000 }, + { CS42L43_SPI_CLK_CONFIG1, 0x00000001 }, { CS42L43_SPI_CONFIG1, 0x00000000 }, { CS42L43_SPI_CONFIG2, 0x00000000 }, { CS42L43_SPI_CONFIG3, 0x00000001 }, -- 2.30.2

3 2

[BUG] snd_fireface: frozen buffer with full system lockup
by edmund.raile 04 Mar '24

04 Mar '24

Hello, I want to report a bug with snd_fireface. issue audio suddenly freezes, repeating the last buffer to infinity other symptoms Most of the time, the GUI also freezes to a point that no interaction is possible. Not even a user console can be reached (ctrl + alt + F#). The only way to get the system back in this scenario is to use REISUB or power it down, I can not reach any console. Luckily, sometimes the kernel messages at the bottom are written to the logfile. occurrence This issue is triggered by more "stressful" circumstances, specifically while using the Windows VST Kontakt in linux-native reaper through yabridge (wine). The timeframe is 5-20 minutes. Reaper connects to the interface using ALSA, no other application is playing audio. My default sample rate is 96k, increasing buffer size to 512 samples decreases CPU load but does not prevent this crash. variables I tried to exclude OS Manjaro or fresh Artix install makes no difference host application the crash happens just as well with linux-native bitwig under similar circumstances wine variants wine, wine-staging, wine-tkg w esync and fsync patches The reason why I attribute both these issues to snd_firewire is that I get no freezes / instabilities at all under these circumstances: * in Linux, using an usb audio interface or the onboard audio instead. To me, this suggests Wine, Reaper and Kontakt are not likely the culprit. * in Windows, using RME's dirver. To me, this suggests FW643 and my platform have no issue working together and the FireFace works as it should. * in Linux, passing the FW643 card through to a windows VM with vfio-pci. Here I am just booting the same physical drive in QEMU, so using the same RME driver. To me, this suggests there is no strange interaction between linux and FW643. system Interface: RME FireFace 800 FireWire controller: LSI FW643 rev 8 Platform: MSI ms7d25 Z690 Kernel: 6.8.0-rc5-1-mainline #8 SMP PREEMPT_DYNAMIC OS: Artix Xfce video driver: mesa An related issue? During normal use, as of recent kernel versions, just playing back audio from any pulse audio application, I get a similar freeze but less severe: no other applications are blocked and it can be removed by just stoping playback, then after 5 seconds the audible freeze stops and I can start playback again. This happens less often, 5-60 minutes between freezes. I managed to capture this unicorn event yesterday in a relatively short file using 'sudo trace-cmd record -e snd_firewire_lib:amdtp_packet', maybe it can give you a hint as to what may be going on: https://pixeldrain.com/u/tgoSA1mk I tried to do the same for my real issue (the one with the full system freeze but I can not stop the recording and the output after rebooting are files split by cpu cores. I hope this is the right place to turn to for help. It has prevented me from using my interface in linux for audio production for a long time and I want to help fix it. If I can provide any form of further assistance or somehow collect any more data, please let me know. Kind regards, Edmund Raile. watchdog: Watchdog detected hard LOCKUP on cpu 13 Modules linked in: ntfs3 fuse 8021q garp mrp stp llc bnep bluetooth ecdh_generic rfkill vfat fat snd_sof_pci_intel_tgl snd_sof_intel_hda_common soundwire_intel snd_sof_intel_hda_mlink snd_hda_codec_hdmi soundwire_cadence snd_sof_intel_hda snd_sof_pci snd_sof_xtensa_dsp snd_sof intel_rapl_msr snd_sof_utils intel_rapl_common snd_soc_hdac_hda snd_hda_ext_core snd_soc_acpi_intel_match intel_uncore_frequency intel_uncore_frequency_common snd_soc_acpi intel_tcc_cooling soundwire_generic_allocation snd_hda_codec_realtek x86_pkg_temp_thermal soundwire_bus intel_powerclamp snd_hda_codec_generic snd_soc_core coretemp snd_compress kvm_intel ac97_bus snd_pcm_dmaengine snd_hda_intel snd_usb_audio snd_intel_dspcfg snd_fireface snd_intel_sdw_acpi kvm snd_usbmidi_lib snd_dice snd_hda_codec snd_firewire_lib snd_ump snd_hda_core snd_rawmidi irqbypass mc snd_hwdep snd_seq_device rapl iTCO_wdt snd_pcm intel_pmc_bxt intel_cstate mei_pxp mei_hdcp iTCO_vendor_support ee1004 snd_timer intel_uncore psmouse pcspkr crypto_user spi_nor snd i2c_i801 mei_me mousedev wacom joydev mtd igc soundcore i2c_smbus intel_pmc_core intel_vsec mei pmt_telemetry pmt_class mac_hid coreboot_table intel_scu_pltdrv ext4 crc32c_generic crc16 mbcache jbd2 hid_logitech ff_memless hid_logitech_hidpp hid_logitech_dj dm_crypt cbc uas usb_storage encrypted_keys trusted asn1_encoder tee dm_mod usbhid crct10dif_pclmul crc32_pclmul crc32c_intel polyval_clmulni polyval_generic gf128mul ghash_clmulni_intel sha512_ssse3 sha256_ssse3 sha1_ssse3 serio_raw aesni_intel atkbd nvme libps2 spi_intel_pci vivaldi_fmap firewire_ohci crypto_simd nvme_core firewire_core cryptd xhci_pci spi_intel nvme_auth xhci_pci_renesas i8042 crc_itu_t serio i915 i2c_algo_bit drm_buddy video wmi ttm intel_gtt drm_display_helper cec CPU: 13 PID: 1993 Comm: reaper/audio Not tainted 6.8.0-rc5-1-mainline #8 c2061649224b2c533bd915b824b700586979f2da Hardware name: Micro-Star International Co., Ltd. MS-7D25/PRO Z690-A DDR4(MS-7D25), BIOS Dasharo (coreboot+UEFI) v1.1.3 01/17/2024 RIP: 0010:tasklet_unlock_spin_wait+0xd/0x20 Code: c0 c3 cc cc cc cc 0f 1f 80 00 00 00 00 90 90 90 90 90 90 90 90 90 90 90 90 90 90 90 90 f3 0f 1e fa 0f 1f 44 00 00 eb 02 f3 90 <48> 8b 47 08 a8 02 75 f6 c3 cc cc cc cc 66 0f 1f 44 00 00 90 90 90 RSP: 0018:ffffb828c1cdbd08 EFLAGS: 00000002 RAX: 0000000000000002 RBX: ffff951841aba028 RCX: 0000000000000000 RDX: 0000000000000001 RSI: ffff951841a2aa70 RDI: ffff951841aba0a8 RBP: ffffb828c1cdbea0 R08: 0000000000000000 R09: 0000000000000000 R10: 0000000000000000 R11: 0000000000000000 R12: 0000000000000000 R13: 0000000000075800 R14: ffff951846a9f600 R15: 0000000000000000 FS: 00007f7f394006c0(0000) GS:ffff952b9fb40000(0000) knlGS:0000000000000000 CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 CR2: 00007fc65c2a0000 CR3: 000000018516a000 CR4: 0000000000f50ef0 PKRU: 55555554 Call Trace: <NMI> ? watchdog_hardlockup_check+0xfc/0x1d0 ? __perf_event_overflow+0xe5/0x2a0 ? handle_pmi_common+0x16f/0x380 ? intel_pmu_handle_irq+0x104/0x480 ? perf_event_nmi_handler+0x2a/0x50 ? nmi_handle+0x5e/0x150 ? default_do_nmi+0x40/0x100 ? exc_nmi+0x139/0x1c0 ? end_repeat_nmi+0xf/0x60 ? tasklet_unlock_spin_wait+0xd/0x20 ? tasklet_unlock_spin_wait+0xd/0x20 ? tasklet_unlock_spin_wait+0xd/0x20 </NMI> <TASK> ohci_flush_iso_completions+0x1d/0x100 [firewire_ohci 83160820eab58a73fec0d1e78730dd51dd0ac98c] amdtp_domain_stream_pcm_pointer+0x41/0x50 [snd_firewire_lib 95578abd4cf50f7c2e87b0d8233b959be2154fc3] snd_pcm_update_hw_ptr0+0x6c/0x430 [snd_pcm df3bd3d8840cfa728bc0f17b93055a5a1595ec6d] ? sched_clock_cpu+0xf/0x190 ? __smp_call_single_queue+0xad/0x120 snd_pcm_delay+0x33/0x140 [snd_pcm df3bd3d8840cfa728bc0f17b93055a5a1595ec6d] snd_pcm_common_ioctl+0x922/0x12b0 [snd_pcm df3bd3d8840cfa728bc0f17b93055a5a1595ec6d] ? wake_up_q+0x4e/0x90 ? futex_wake+0x159/0x190 snd_pcm_ioctl+0x2e/0x50 [snd_pcm df3bd3d8840cfa728bc0f17b93055a5a1595ec6d] __x64_sys_ioctl+0x94/0xd0 do_syscall_64+0x86/0x170 ? do_syscall_64+0x96/0x170 ? do_syscall_64+0x96/0x170 ? do_syscall_64+0x96/0x170 ? do_syscall_64+0x96/0x170 ? __irq_exit_rcu+0x4b/0xc0 entry_SYSCALL_64_after_hwframe+0x6e/0x76 RIP: 0033:0x7f7f47cfd4ff Code: 00 48 89 44 24 18 31 c0 48 8d 44 24 60 c7 04 24 10 00 00 00 48 89 44 24 08 48 8d 44 24 20 48 89 44 24 10 b8 10 00 00 00 0f 05 <89> c2 3d 00 f0 ff ff 77 18 48 8b 44 24 18 64 48 2b 04 25 28 00 00 RSP: 002b:00007f7f393ffd50 EFLAGS: 00000246 ORIG_RAX: 0000000000000010 RAX: ffffffffffffffda RBX: 0000000003cfc060 RCX: 00007f7f47cfd4ff RDX: 00007f7f393ffdf0 RSI: 0000000080084121 RDI: 000000000000000a RBP: 0000000000000001 R08: 0000000000000001 R09: 0000000000000004 R10: 00007f7f393ffb20 R11: 0000000000000246 R12: 0000000000005000 R13: 0000000003272a80 R14: 0000000000000050 R15: 00000000032706c0 </TASK> watchdog: Watchdog detected hard LOCKUP on cpu 6 CPU: 6 PID: 0 Comm: swapper/6 Not tainted 6.8.0-rc5-1-mainline #8 c2061649224b2c533bd915b824b700586979f2da Hardware name: Micro-Star International Co., Ltd. MS-7D25/PRO Z690-A DDR4(MS-7D25), BIOS Dasharo (coreboot+UEFI) v1.1.3 01/17/2024 RIP: 0010:native_queued_spin_lock_slowpath+0x6e/0x2e0 Code: 77 7f f0 0f ba 2b 08 0f 92 c2 8b 03 0f b6 d2 c1 e2 08 30 e4 09 d0 3d ff 00 00 00 77 5b 85 c0 74 10 0f b6 03 84 c0 74 09 f3 90 <0f> b6 03 84 c0 75 f7 b8 01 00 00 00 66 89 03 65 48 ff 05 83 a6 23 RSP: 0018:ffffb828c01ece08 EFLAGS: 00000002 RAX: 0000000000000001 RBX: ffff951846a9f6f8 RCX: 0000000000000300 RDX: 0000000000000000 RSI: 0000000000000001 RDI: ffff951846a9f6f8 RBP: ffff95184684c438 R08: ffff951879f3f800 R09: ffff951841a94028 R10: ffff951879f3f800 R11: 0000000000000000 R12: 0000000000000031 R13: 0000000000000000 R14: ffff9518a562e000 R15: ffff9518c49e5800 FS: 0000000000000000(0000) GS:ffff952b9f980000(0000) knlGS:0000000000000000 CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 CR2: 00000001010fdac8 CR3: 000000018516a000 CR4: 0000000000f50ef0 PKRU: 55555554 Call Trace: <NMI> ? watchdog_hardlockup_check+0xfc/0x1d0 ? __perf_event_overflow+0xe5/0x2a0 ? handle_pmi_common+0x16f/0x380 ? intel_pmu_handle_irq+0x104/0x480 ? perf_event_nmi_handler+0x2a/0x50 ? nmi_handle+0x5e/0x150 ? default_do_nmi+0x40/0x100 ? exc_nmi+0x139/0x1c0 ? end_repeat_nmi+0xf/0x60 ? native_queued_spin_lock_slowpath+0x6e/0x2e0 ? native_queued_spin_lock_slowpath+0x6e/0x2e0 ? native_queued_spin_lock_slowpath+0x6e/0x2e0 </NMI> <IRQ> _raw_spin_lock_irqsave+0x3d/0x50 snd_pcm_period_elapsed+0x18/0x40 [snd_pcm df3bd3d8840cfa728bc0f17b93055a5a1595ec6d] process_rx_packets+0x245/0x5a0 [snd_firewire_lib 95578abd4cf50f7c2e87b0d8233b959be2154fc3] ? irqtime_account_irq+0x40/0xc0 irq_target_callback+0x16/0x20 [snd_firewire_lib 95578abd4cf50f7c2e87b0d8233b959be2154fc3] handle_it_packet+0x135/0x150 [firewire_ohci 83160820eab58a73fec0d1e78730dd51dd0ac98c] context_tasklet+0x140/0x160 [firewire_ohci 83160820eab58a73fec0d1e78730dd51dd0ac98c] tasklet_action_common.isra.0+0x146/0x240 __do_softirq+0xc9/0x2c8 __irq_exit_rcu+0xa3/0xc0 common_interrupt+0x86/0xa0 </IRQ> <TASK> asm_common_interrupt+0x26/0x40 RIP: 0010:cpuidle_enter_state+0xcc/0x440 Code: 2a 39 36 ff e8 05 f1 ff ff 8b 53 04 49 89 c5 0f 1f 44 00 00 31 ff e8 83 38 35 ff 45 84 ff 0f 85 56 02 00 00 fb 0f 1f 44 00 00 <45> 85 f6 0f 88 85 01 00 00 49 63 d6 48 8d 04 52 48 8d 04 82 49 8d RSP: 0018:ffffb828c00ffe90 EFLAGS: 00000246 RAX: ffff952b9f9b4800 RBX: ffff952b9f9be208 RCX: 000000000000001f RDX: 0000000000000006 RSI: 0000000022b8e38e RDI: 0000000000000000 RBP: 0000000000000001 R08: 0000000000000000 R09: 00000000000001c8 R10: 0000000000000018 R11: ffff952b9f9b31e4 R12: ffffffff83f49780 R13: 0000006b6114102b R14: 0000000000000001 R15: 0000000000000000 cpuidle_enter+0x2d/0x40 do_idle+0x1d8/0x230 cpu_startup_entry+0x2a/0x30 start_secondary+0x11e/0x140 secondary_startup_64_no_verify+0x184/0x18b </TASK> rcu: INFO: rcu_preempt detected stalls on CPUs/tasks: rcu: \x096-...0: (1 GPs behind) idle=d7ec/1/0x4000000000000002 softirq=52862/52863 fqs=5201 rcu: \x0913-...0: (1 GPs behind) idle=c9d4/1/0x4000000000000000 softirq=53654/53655 fqs=5201 rcu: \x09(detected by 2, t=18002 jiffies, g=76629, q=5124 ncpus=16) Sending NMI from CPU 2 to CPUs 6: NMI backtrace for cpu 6 CPU: 6 PID: 0 Comm: swapper/6 Not tainted 6.8.0-rc5-1-mainline #8 c2061649224b2c533bd915b824b700586979f2da Hardware name: Micro-Star International Co., Ltd. MS-7D25/PRO Z690-A DDR4(MS-7D25), BIOS Dasharo (coreboot+UEFI) v1.1.3 01/17/2024 RIP: 0010:native_queued_spin_lock_slowpath+0x6e/0x2e0 Code: 77 7f f0 0f ba 2b 08 0f 92 c2 8b 03 0f b6 d2 c1 e2 08 30 e4 09 d0 3d ff 00 00 00 77 5b 85 c0 74 10 0f b6 03 84 c0 74 09 f3 90 <0f> b6 03 84 c0 75 f7 b8 01 00 00 00 66 89 03 65 48 ff 05 83 a6 23 RSP: 0018:ffffb828c01ece08 EFLAGS: 00000002 RAX: 0000000000000001 RBX: ffff951846a9f6f8 RCX: 0000000000000300 RDX: 0000000000000000 RSI: 0000000000000001 RDI: ffff951846a9f6f8 RBP: ffff95184684c438 R08: ffff951879f3f800 R09: ffff951841a94028 R10: ffff951879f3f800 R11: 0000000000000000 R12: 0000000000000031 R13: 0000000000000000 R14: ffff9518a562e000 R15: ffff9518c49e5800 FS: 0000000000000000(0000) GS:ffff952b9f980000(0000) knlGS:0000000000000000 CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 CR2: 00000001010fdac8 CR3: 000000018516a000 CR4: 0000000000f50ef0 PKRU: 55555554 Call Trace: <NMI> ? nmi_cpu_backtrace+0x99/0x110 ? nmi_cpu_backtrace_handler+0x11/0x20 ? nmi_handle+0x5e/0x150 ? default_do_nmi+0x40/0x100 ? exc_nmi+0x139/0x1c0 ? end_repeat_nmi+0xf/0x60 ? native_queued_spin_lock_slowpath+0x6e/0x2e0 ? native_queued_spin_lock_slowpath+0x6e/0x2e0 ? native_queued_spin_lock_slowpath+0x6e/0x2e0 </NMI> <IRQ> _raw_spin_lock_irqsave+0x3d/0x50 snd_pcm_period_elapsed+0x18/0x40 [snd_pcm df3bd3d8840cfa728bc0f17b93055a5a1595ec6d] process_rx_packets+0x245/0x5a0 [snd_firewire_lib 95578abd4cf50f7c2e87b0d8233b959be2154fc3] ? irqtime_account_irq+0x40/0xc0 irq_target_callback+0x16/0x20 [snd_firewire_lib 95578abd4cf50f7c2e87b0d8233b959be2154fc3] handle_it_packet+0x135/0x150 [firewire_ohci 83160820eab58a73fec0d1e78730dd51dd0ac98c] context_tasklet+0x140/0x160 [firewire_ohci 83160820eab58a73fec0d1e78730dd51dd0ac98c] tasklet_action_common.isra.0+0x146/0x240 __do_softirq+0xc9/0x2c8 __irq_exit_rcu+0xa3/0xc0 common_interrupt+0x86/0xa0 </IRQ> <TASK> asm_common_interrupt+0x26/0x40 RIP: 0010:cpuidle_enter_state+0xcc/0x440 Code: 2a 39 36 ff e8 05 f1 ff ff 8b 53 04 49 89 c5 0f 1f 44 00 00 31 ff e8 83 38 35 ff 45 84 ff 0f 85 56 02 00 00 fb 0f 1f 44 00 00 <45> 85 f6 0f 88 85 01 00 00 49 63 d6 48 8d 04 52 48 8d 04 82 49 8d RSP: 0018:ffffb828c00ffe90 EFLAGS: 00000246 RAX: ffff952b9f9b4800 RBX: ffff952b9f9be208 RCX: 000000000000001f RDX: 0000000000000006 RSI: 0000000022b8e38e RDI: 0000000000000000 RBP: 0000000000000001 R08: 0000000000000000 R09: 00000000000001c8 R10: 0000000000000018 R11: ffff952b9f9b31e4 R12: ffffffff83f49780 R13: 0000006b6114102b R14: 0000000000000001 R15: 0000000000000000 cpuidle_enter+0x2d/0x40 do_idle+0x1d8/0x230 cpu_startup_entry+0x2a/0x30 start_secondary+0x11e/0x140 secondary_startup_64_no_verify+0x184/0x18b </TASK> Sending NMI from CPU 2 to CPUs 13: NMI backtrace for cpu 13 CPU: 13 PID: 1993 Comm: reaper/audio Not tainted 6.8.0-rc5-1-mainline #8 c2061649224b2c533bd915b824b700586979f2da Hardware name: Micro-Star International Co., Ltd. MS-7D25/PRO Z690-A DDR4(MS-7D25), BIOS Dasharo (coreboot+UEFI) v1.1.3 01/17/2024 RIP: 0010:tasklet_unlock_spin_wait+0xd/0x20 Code: c0 c3 cc cc cc cc 0f 1f 80 00 00 00 00 90 90 90 90 90 90 90 90 90 90 90 90 90 90 90 90 f3 0f 1e fa 0f 1f 44 00 00 eb 02 f3 90 <48> 8b 47 08 a8 02 75 f6 c3 cc cc cc cc 66 0f 1f 44 00 00 90 90 90 RSP: 0018:ffffb828c1cdbd08 EFLAGS: 00000002 RAX: 0000000000000002 RBX: ffff951841aba028 RCX: 0000000000000000 RDX: 0000000000000001 RSI: ffff951841a2aa70 RDI: ffff951841aba0a8 RBP: ffffb828c1cdbea0 R08: 0000000000000000 R09: 0000000000000000 R10: 0000000000000000 R11: 0000000000000000 R12: 0000000000000000 R13: 0000000000075800 R14: ffff951846a9f600 R15: 0000000000000000 FS: 00007f7f394006c0(0000) GS:ffff952b9fb40000(0000) knlGS:0000000000000000 CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 CR2: 00007fc65c2a0000 CR3: 000000018516a000 CR4: 0000000000f50ef0 PKRU: 55555554 Call Trace: <NMI> ? nmi_cpu_backtrace+0x99/0x110 ? nmi_cpu_backtrace_handler+0x11/0x20 ? nmi_handle+0x5e/0x150 ? default_do_nmi+0x40/0x100 ? exc_nmi+0x139/0x1c0 ? end_repeat_nmi+0xf/0x60 ? tasklet_unlock_spin_wait+0xd/0x20 ? tasklet_unlock_spin_wait+0xd/0x20 ? tasklet_unlock_spin_wait+0xd/0x20 </NMI> <TASK> ohci_flush_iso_completions+0x1d/0x100 [firewire_ohci 83160820eab58a73fec0d1e78730dd51dd0ac98c] amdtp_domain_stream_pcm_pointer+0x41/0x50 [snd_firewire_lib 95578abd4cf50f7c2e87b0d8233b959be2154fc3] snd_pcm_update_hw_ptr0+0x6c/0x430 [snd_pcm df3bd3d8840cfa728bc0f17b93055a5a1595ec6d] ? sched_clock_cpu+0xf/0x190 ? __smp_call_single_queue+0xad/0x120 snd_pcm_delay+0x33/0x140 [snd_pcm df3bd3d8840cfa728bc0f17b93055a5a1595ec6d] snd_pcm_common_ioctl+0x922/0x12b0 [snd_pcm df3bd3d8840cfa728bc0f17b93055a5a1595ec6d] ? wake_up_q+0x4e/0x90 ? futex_wake+0x159/0x190 snd_pcm_ioctl+0x2e/0x50 [snd_pcm df3bd3d8840cfa728bc0f17b93055a5a1595ec6d] __x64_sys_ioctl+0x94/0xd0 do_syscall_64+0x86/0x170 ? do_syscall_64+0x96/0x170 ? do_syscall_64+0x96/0x170 ? do_syscall_64+0x96/0x170 ? do_syscall_64+0x96/0x170 ? __irq_exit_rcu+0x4b/0xc0 entry_SYSCALL_64_after_hwframe+0x6e/0x76 RIP: 0033:0x7f7f47cfd4ff Code: 00 48 89 44 24 18 31 c0 48 8d 44 24 60 c7 04 24 10 00 00 00 48 89 44 24 08 48 8d 44 24 20 48 89 44 24 10 b8 10 00 00 00 0f 05 <89> c2 3d 00 f0 ff ff 77 18 48 8b 44 24 18 64 48 2b 04 25 28 00 00 RSP: 002b:00007f7f393ffd50 EFLAGS: 00000246 ORIG_RAX: 0000000000000010 RAX: ffffffffffffffda RBX: 0000000003cfc060 RCX: 00007f7f47cfd4ff RDX: 00007f7f393ffdf0 RSI: 0000000080084121 RDI: 000000000000000a RBP: 0000000000000001 R08: 0000000000000001 R09: 0000000000000004 R10: 00007f7f393ffb20 R11: 0000000000000246 R12: 0000000000005000 R13: 0000000003272a80 R14: 0000000000000050 R15: 00000000032706c0 </TASK> INFO: task khugepaged:124 blocked for more than 122 seconds. Not tainted 6.8.0-rc5-1-mainline #8 "echo 0 > /proc/sys/kernel/hung_task_timeout_secs" disables this message. task:khugepaged state:D stack:0 pid:124 tgid:124 ppid:2 flags:0x00004000 Call Trace: <TASK> __schedule+0x3e6/0x1520 ? place_entity+0x1b/0xf0 schedule+0x32/0xd0 schedule_timeout+0x151/0x160 wait_for_completion+0x8a/0x160 __flush_work.isra.0+0x173/0x280 ? __pfx_wq_barrier_func+0x10/0x10 __lru_add_drain_all+0x14c/0x200 khugepaged+0x72/0x950 ? __pfx_autoremove_wake_function+0x10/0x10 ? __pfx_khugepaged+0x10/0x10 kthread+0xe5/0x120 ? __pfx_kthread+0x10/0x10 ret_from_fork+0x31/0x50 ? __pfx_kthread+0x10/0x10 ret_from_fork_asm+0x1b/0x30 </TASK>

1 0

ASoC: rockchip_i2s_tdm calibration clocking problem
by Pavel Hofman 04 Mar '24

04 Mar '24

Hi, I am hitting a clock issue with rockchip_i2s_tdm.c + simple-audio-card (+ RK3308). At boot the mclk clk_i2s0_8ch_tx is (somehow) initialized to some (unimportant?) value 50176000 Hz. Note that this frequency is not multiple of either 48kHz or 44.1kHz. Method asoc_simple_parse_clk() reads this value and sets it to simple_dai->sysclk. Subsequently at asoc_simple_dai_init this "random" initial value is stored to i2s_tdm->mclk_tx_freq: 17.839330] rockchip_i2s_tdm_set_sysclk+0x50/0xbc [snd_soc_rockchip_i2s_tdm] [ 17.839367] snd_soc_dai_set_sysclk+0x38/0xb8 [snd_soc_core] [ 17.839596] asoc_simple_init_dai+0x94/0xc0 [snd_soc_simple_card_utils] [ 17.839640] asoc_simple_dai_init+0x130/0x230 [snd_soc_simple_card_utils] [ 17.839672] snd_soc_link_init+0x28/0x90 [snd_soc_core] [ 17.839843] snd_soc_bind_card+0x60c/0xbb4 [snd_soc_core] When starting playback, called by rockchip_i2s_tdm_hw_params(), rockchip_i2s_tdm_calibrate_mclk() correctly switches parent of mclk_parent to correct root pll clock mclk_root0/1 for the given samplerate and correctly configures mclk_parent frequency. https://github.com/torvalds/linux/blob/master/sound/soc/rockchip/rockchip_i… But right after that, the next line of rockchip_i2s_tdm_hw_params() calls rockchip_i2s_tdm_set_mclk() https://github.com/torvalds/linux/blob/master/sound/soc/rockchip/rockchip_i… This method calls clk_set_rate(i2s_tdm->mclk_tx, i2s_tdm->mclk_tx_freq), which resets the clock and its parental chain to the original incorrect value stored in i2s_tdm->mclk_tx_freq from the dai initialization. https://github.com/torvalds/linux/blob/master/sound/soc/rockchip/rockchip_i… As a result, no matter what sample rate is being played, the i2s mclk clock always ends up configured incorrectly. DTS I2S sets all clocks, therefore the clk calibration in rockchip_i2s_tdm.c should be (and is) used: i2s_8ch_0: i2s@ff300000 { compatible = "rockchip,rk3308-i2s-tdm"; reg = <0x0 0xff300000 0x0 0x1000>; interrupts = <GIC_SPI 48 IRQ_TYPE_LEVEL_HIGH>; clocks = <&cru SCLK_I2S0_8CH_TX>, <&cru SCLK_I2S0_8CH_RX>, <&cru HCLK_I2S0_8CH>, <&cru SCLK_I2S0_8CH_TX_SRC>, <&cru SCLK_I2S0_8CH_RX_SRC>, <&cru PLL_VPLL0>, <&cru PLL_VPLL1>; clock-names = "mclk_tx", "mclk_rx", "hclk", "mclk_tx_src", "mclk_rx_src", "mclk_root0", "mclk_root1"; ......... It seems to me that the calibration code should also rewrite the initially incorrect i2s_tdm->mclk_tx_freq and i2s_tdm->mclk_rx_freq with correct values corresponding to the momentary hw_params rate, or maybe rockchip_i2s_tdm_set_mclk() should not be called if rockchip_i2s_tdm_calibrate_mclk() is called a line above (i.e. putting the call to rockchip_i2s_tdm_set_mclk() into "else" branch). Thank you very much for help. With regards, Pavel.

2 1

[PATCH 5.10/5.15/6.1 0/1] soundwire: stream: use consistent pattern for freeing buffers
by Daniil Dulov 04 Mar '24

04 Mar '24

Svacer reports NULL-pointer dereference and double free issues in do_bank_switch() in case sdw_ml_sync_bank_switch() returns an error not on the first iteration of the list_for_each_entry() loop. These problems are present in 5.10, 5.15 and 6.1 stable releases. These problems have been fixed by the following upstream patch that can be cleanly applied to 5.10, 5.15 and 6.1 branches.

1 1

[PATCH] [alsa-lib][PATCH] conf: USB-Audio: Add Corsair HS60 Pro to the IEC958 blacklist
by David Senoner 04 Mar '24

04 Mar '24

This device is just an external USB soundcard with a female 3.5mm jack for a headset, no S/PDIF | IEC958 connector. Add it to the blacklist to prevent it being opened. Signed-off-by: David Senoner <seda18(a)rolmail.net> --- src/conf/cards/USB-Audio.conf | 1 + 1 file changed, 1 insertion(+) diff --git a/src/conf/cards/USB-Audio.conf b/src/conf/cards/USB-Audio.conf index 80631b2e..08346a1b 100644 --- a/src/conf/cards/USB-Audio.conf +++ b/src/conf/cards/USB-Audio.conf @@ -53,6 +53,7 @@ USB-Audio.pcm.iec958_device { "Blue Snowball" 999 "C-Media USB Headphone Set" 999 "Cmedia Audio" 999 + "Corsair HS60 PRO Surround USB S" 999 "DELL PROFESSIONAL SOUND BAR AE5" 999 "HP Digital Stereo Headset" 999 "GN 9330" 999 -- 2.43.2

3 2

[PATCH RESEND] conf: USB-Audio: Add more Scarlett devices to the IEC958 blacklist
by Geoffrey D. Bennett 04 Mar '24

04 Mar '24

The Scarlett Solo and 2i2 don't have S/PDIF outputs. Signed-off-by: Geoffrey D. Bennett <g(a)b4.vu> --- src/conf/cards/USB-Audio.conf | 3 +++ 1 file changed, 3 insertions(+) diff --git a/src/conf/cards/USB-Audio.conf b/src/conf/cards/USB-Audio.conf index 80631b2e..05384a32 100644 --- a/src/conf/cards/USB-Audio.conf +++ b/src/conf/cards/USB-Audio.conf @@ -66,8 +66,11 @@ USB-Audio.pcm.iec958_device { "Plantronics USB Headset" 999 "Plantronics Wireless Audio" 999 "SB WoW Headset" 999 + "Scarlett 2i2 4th Gen" 999 "Scarlett 2i2 USB" 999 "Scarlett 2i4 USB" 999 + "Scarlett Solo 4th Gen" 999 + "Scarlett Solo USB" 999 "Sennheiser USB headset" 999 "SWTOR Gaming Headset by Razer" 999 "ThinkStation P620 Main" 999 -- 2.43.0

2 1

[PATCH v3 0/5] ALSA/ASoC: Conditionally skip i915 init and cleanups
by Cezary Rojewski 04 Mar '24

04 Mar '24

A small set of changes to improve initialization of the audio stack on HDAudio devices and pair of cleanups. As the first change is the most important one here, following is the technical background for it: Commit 78f613ba1efb ("drm/i915: finish removal of CNL") and its friends removed support for i915 for all CNL-based platforms. HDAudio library, however, still treats such platforms as valid candidates for i915 binding. Update query mechanism to reflect changes made in drm tree. At the same time, i915 support for LKF-based platforms has not been provided so remove them from valid binding candidates. The snd_soc_hda change is a follow up for the above and the cleanup patches do not bring any functional changes. Changes in v3: - snd_soc_hda_codec now returns -ENODEV on attach() if i915 is not present - denylist now const - added new patch for the avs-driver to address -ENODEV during probe_codec() - note: retained reviewed-by for patch 1/4 as changes are minimal Changes in v2: - list of problematic VGA devices is now declared locally, no more touching drm stuff Cezary Rojewski (5): ALSA: hda: Skip i915 initialization on CNL/LKF-based platforms ASoC: codecs: hda: Skip HDMI/DP registration if i915 is missing ASoC: Intel: avs: Ignore codecs with no suppoting driver ASoC: codecs: hda: Cleanup error messages ALSA: hda: Reuse for_each_pcm_streams() sound/hda/hdac_i915.c | 32 +++++++++++++++++++++++++++++--- sound/pci/hda/hda_codec.c | 2 +- sound/soc/codecs/hda.c | 15 ++++++++++----- sound/soc/intel/avs/core.c | 9 +++++---- 4 files changed, 45 insertions(+), 13 deletions(-) -- 2.25.1

3 9

[PATCH v1 0/3] ALSA: hda: cs35l41: Add support for a couple of Lenovo and Asus Laptops
by Stefan Binding 04 Mar '24

04 Mar '24

Add support for a couple of Lenovo Thinkbook 16P laptops using CS35L41 HDA. Also fix a _DSD BIOS error for the ASUS UM5302LA laptop. Stefan Binding (3): ALSA: hda: cs35l41: Support Lenovo Thinkbook 16P ALSA: hda/realtek: Add quirks for Lenovo Thinkbook 16P laptops ALSA: hda: cs35l41: Overwrite CS35L41 configuration for ASUS UM5302LA sound/pci/hda/cs35l41_hda_property.c | 6 ++++++ sound/pci/hda/patch_realtek.c | 2 ++ 2 files changed, 8 insertions(+) -- 2.34.1

2 4

Enable Headset Mic for Acer NB platform
by Kailang 04 Mar '24

04 Mar '24

Hi Takashi, This patch will enable headset Mic for Acer NB platform. Many Thanks. BR, Kailang

2 3

Scarlett Solo USB profiles missing or broken in 1.2.11
by GitHub issues - edited 03 Mar '24

03 Mar '24

alsa-project/alsa-ucm-conf issue #398 was edited from Slabity: Prior to version 1.2.10, my Scarlett Solo USB profiles looked like the following, where I could select profiles that explicitly enabled or disabled either the Inputs or Outputs: ![goodaudiooutput](https://github.com/alsa-project/alsa-ucm-conf/assets/3788421/b1bbc857-82b6-4fa5-b478-0c00c26b281c) As of 1.2.11, these profiles have disappeared and have left only a couple of profiles that do not provide the same functionality: ![badaudiooutputs](https://github.com/alsa-project/alsa-ucm-conf/assets/3788421/6c321638-3eb1-45ef-94a5-734c16477886) Other devices like my GPU's HDMI/DP Audio and motherboard's "Starship/Matisse HD Audio Controller" continue to expose the wider variety of profiles. It is only this device that seems to have the issue. I originally reported this on #346 as I thought it was related, but was told it might be a different issue. What makes me worried is the mention that this is "expected" now despite it breaking my audio setup. I can no longer select the "Digital Stereo (IEC958) Input" to exclusively enable the Input Device *without* enabling the Output Device. Here's some more information about the device: ``` $ lsusb | grep Focusrite Bus 007 Device 003: ID 1235:801c Focusrite-Novation Scarlett Solo USB $ alsaucm listcards 0: hw:2 Focusrite Scarlett Solo USB at usb-0000:10:00.3-3, high speed $ alsaucm -c hw:2 dump text Verb.HiFi { Comment Default Device.Line1 { Comment "Headphones / Line 1-2" Values { CaptureCTL "_ucm0001.hw:USB" PlaybackCTL "_ucm0001.hw:USB" PlaybackPCM "_ucm0001.hw:USB" PlaybackPriority 200 } } Device.Mic1 { Comment "Input 1 Mic" Values { CaptureCTL "_ucm0001.hw:USB" CaptureChannels 1 CapturePCM "_ucm0001.scarlett2i_mono_in:USB,0,0" CapturePriority 200 PlaybackCTL "_ucm0001.hw:USB" } } Device.Mic2 { Comment "Input 2 Inst/Line" Values { CaptureCTL "_ucm0001.hw:USB" CaptureChannels 1 CapturePCM "_ucm0001.scarlett2i_mono_in:USB,0,1" CapturePriority 100 PlaybackCTL "_ucm0001.hw:USB" } } } Verb.Direct { Comment "Direct Scarlett Solo USB" Device.Direct { Comment "Direct Scarlett Solo USB" Values { CaptureCTL "_ucm0001.hw:USB" CaptureChannels 2 CapturePCM "_ucm0001.hw:USB" CapturePriority 1000 PlaybackCTL "_ucm0001.hw:USB" PlaybackChannels 2 PlaybackPCM "_ucm0001.hw:USB" PlaybackPriority 1000 } } } ``` Some OS info it it helps: Linux distribution: NixOS on nixpkgs-unstable Kernel: 6.7.7 ALSA version: Advanced Linux Sound Architecture Driver Version k6.7.7 Pulseaudio version: 15.0.0 Pipewire version: 1.0.3 Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/398 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Scarlett Solo USB profiles missing or broken in 1.2.11
by GitHub issues - edited 03 Mar '24

03 Mar '24

alsa-project/alsa-ucm-conf issue #398 was edited from Slabity: Prior to version 1.2.10, my Scarlett Solo USB profiles looked like the following, where I could select profiles that explicitly enabled or disabled either the Inputs or Outputs: ![goodaudiooutput](https://github.com/alsa-project/alsa-ucm-conf/assets/3788421/b1bbc857-82b6-4fa5-b478-0c00c26b281c) As of 1.2.11, these profiles have disappeared and have left only a couple of profiles that do not provide the same functionality: ![badaudiooutputs](https://github.com/alsa-project/alsa-ucm-conf/assets/3788421/6c321638-3eb1-45ef-94a5-734c16477886) Other devices like my GPU's HDMI/DP Audio and motherboard's "Starship/Matisse HD Audio Controller" continue to expose the wider variety of profiles. It is only this device that seems to have the issue. I originally reported this on #346 as I thought it was related, but was told it might be a different issue. What makes me worried is the mention that this is "expected" now despite it breaking my audio setup. I can no longer select the "Digital Stereo (IEC958) Input" to exclusively enable the Input Device *without* enabling the Output Device. Here's some more information about the device: ``` $ lsusb | grep Focusrite Bus 007 Device 003: ID 1235:801c Focusrite-Novation Scarlett Solo USB $ alsaucm listcards 0: hw:2 Focusrite Scarlett Solo USB at usb-0000:10:00.3-3, high speed $ alsaucm -c hw:2 dump text Verb.HiFi { Comment Default Device.Line1 { Comment "Headphones / Line 1-2" Values { CaptureCTL "_ucm0001.hw:USB" PlaybackCTL "_ucm0001.hw:USB" PlaybackPCM "_ucm0001.hw:USB" PlaybackPriority 200 } } Device.Mic1 { Comment "Input 1 Mic" Values { CaptureCTL "_ucm0001.hw:USB" CaptureChannels 1 CapturePCM "_ucm0001.scarlett2i_mono_in:USB,0,0" CapturePriority 200 PlaybackCTL "_ucm0001.hw:USB" } } Device.Mic2 { Comment "Input 2 Inst/Line" Values { CaptureCTL "_ucm0001.hw:USB" CaptureChannels 1 CapturePCM "_ucm0001.scarlett2i_mono_in:USB,0,1" CapturePriority 100 PlaybackCTL "_ucm0001.hw:USB" } } } Verb.Direct { Comment "Direct Scarlett Solo USB" Device.Direct { Comment "Direct Scarlett Solo USB" Values { CaptureCTL "_ucm0001.hw:USB" CaptureChannels 2 CapturePCM "_ucm0001.hw:USB" CapturePriority 1000 PlaybackCTL "_ucm0001.hw:USB" PlaybackChannels 2 PlaybackPCM "_ucm0001.hw:USB" PlaybackPriority 1000 } } } ``` Some OS info it it helps: Linux distribution: NixOS Kernel: 6.7.7 ALSA version: Advanced Linux Sound Architecture Driver Version k6.7.7 Pulseaudio version: 15.0.0 Pipewire version: 1.0.3 Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/398 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Scarlett Solo USB profiles missing or broken in 1.2.11
by GitHub issues - edited 03 Mar '24

03 Mar '24

alsa-project/alsa-ucm-conf issue #398 was edited from Slabity: Prior to version 1.2.10, my Scarlett Solo USB profiles looked like the following, where I could select profiles that explicitly enabled or disabled either the Inputs or Outputs: ![goodaudiooutput](https://github.com/alsa-project/alsa-ucm-conf/assets/3788421/b1bbc857-82b6-4fa5-b478-0c00c26b281c) As of 1.2.11, these profiles have disappeared and have left only a couple of profiles that do not provide the same functionality: ![badaudiooutputs](https://github.com/alsa-project/alsa-ucm-conf/assets/3788421/6c321638-3eb1-45ef-94a5-734c16477886) Other devices like my GPU's HDMI/DP Audio and motherboard's "Starship/Matisse HD Audio Controller" continue to expose the wider variety of profiles. It is only this device that seems to have the issue. I originally reported this on #346 as I thought it was related, but was told it might be a different issue. What makes me worried is the mention that this is "expected" now despite it breaking my audio setup. I can no longer select the "Digital Stereo (IEC958) Input" to exclusively enable the Input Device *without* enabling the Output Device. Here's some more information about the device: ``` $ lsusb | grep Focusrite Bus 007 Device 003: ID 1235:801c Focusrite-Novation Scarlett Solo USB $ alsaucm listcards 0: hw:2 Focusrite Scarlett Solo USB at usb-0000:10:00.3-3, high speed $ alsaucm -c hw:2 dump text Verb.HiFi { Comment Default Device.Line1 { Comment "Headphones / Line 1-2" Values { CaptureCTL "_ucm0001.hw:USB" PlaybackCTL "_ucm0001.hw:USB" PlaybackPCM "_ucm0001.hw:USB" PlaybackPriority 200 } } Device.Mic1 { Comment "Input 1 Mic" Values { CaptureCTL "_ucm0001.hw:USB" CaptureChannels 1 CapturePCM "_ucm0001.scarlett2i_mono_in:USB,0,0" CapturePriority 200 PlaybackCTL "_ucm0001.hw:USB" } } Device.Mic2 { Comment "Input 2 Inst/Line" Values { CaptureCTL "_ucm0001.hw:USB" CaptureChannels 1 CapturePCM "_ucm0001.scarlett2i_mono_in:USB,0,1" CapturePriority 100 PlaybackCTL "_ucm0001.hw:USB" } } } Verb.Direct { Comment "Direct Scarlett Solo USB" Device.Direct { Comment "Direct Scarlett Solo USB" Values { CaptureCTL "_ucm0001.hw:USB" CaptureChannels 2 CapturePCM "_ucm0001.hw:USB" CapturePriority 1000 PlaybackCTL "_ucm0001.hw:USB" PlaybackChannels 2 PlaybackPCM "_ucm0001.hw:USB" PlaybackPriority 1000 } } } ``` Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/398 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Scarlett Solo USB profiles missing or broken in 1.2.11
by GitHub issues - opened 03 Mar '24

03 Mar '24

alsa-project/alsa-ucm-conf issue #398 was opened from Slabity: Prior to version 1.2.10, my Scarlett Solo USB profiles looked like the following, where I could select profiles that explicitly enabled or disabled either the Inputs or Outputs: ![goodaudiooutput](https://github.com/alsa-project/alsa-ucm-conf/assets/3788421/b1bbc857-82b6-4fa5-b478-0c00c26b281c) As of 1.2.11, these profiles have disappeared and have left only a couple of profiles that do not provide the same functionality: ![badaudiooutputs](https://github.com/alsa-project/alsa-ucm-conf/assets/3788421/6c321638-3eb1-45ef-94a5-734c16477886) Other devices like my GPU's HDMI/DP Audio and motherboard's "Starship/Matisse HD Audio Controller" continue to expose the wider variety of profiles. It is only this device that seems to have the issue. I originally reported this on #346 as I thought it was related, but was told it might be a different issue. What makes me worried is the mention that this is "expected" now despite it breaking my audio setup. I can no longer select the "Digital Stereo (IEC958) Input" to exclusively enable the Input Device *without* enabling the Output Device. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/398 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH] soundwire: Use snd_soc_substream_to_rtd() to obtain rtd
by Cezary Rojewski 03 Mar '24

03 Mar '24

Utilize the helper function instead of casting from ->private_data directly. Signed-off-by: Cezary Rojewski <cezary.rojewski(a)intel.com> --- drivers/soundwire/stream.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/drivers/soundwire/stream.c b/drivers/soundwire/stream.c index f9c0adc0738d..4e9e7d2a942d 100644 --- a/drivers/soundwire/stream.c +++ b/drivers/soundwire/stream.c @@ -1718,7 +1718,7 @@ EXPORT_SYMBOL(sdw_deprepare_stream); static int set_stream(struct snd_pcm_substream *substream, struct sdw_stream_runtime *sdw_stream) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct snd_soc_dai *dai; int ret = 0; int i; @@ -1771,7 +1771,7 @@ EXPORT_SYMBOL(sdw_alloc_stream); int sdw_startup_stream(void *sdw_substream) { struct snd_pcm_substream *substream = sdw_substream; - struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct sdw_stream_runtime *sdw_stream; char *name; int ret; @@ -1815,7 +1815,7 @@ EXPORT_SYMBOL(sdw_startup_stream); void sdw_shutdown_stream(void *sdw_substream) { struct snd_pcm_substream *substream = sdw_substream; - struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct sdw_stream_runtime *sdw_stream; struct snd_soc_dai *dai; -- 2.25.1

2 1

[PATCH] soundwire: constify the struct device_type usage
by Ricardo B. Marliere 03 Mar '24

03 Mar '24

Since commit aed65af1cc2f ("drivers: make device_type const"), the driver core can properly handle constant struct device_type. Move the sdw_master_type and sdw_slave_type variables to be constant structures as well, placing it into read-only memory which can not be modified at runtime. Cc: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> Signed-off-by: Ricardo B. Marliere <ricardo(a)marliere.net> --- drivers/soundwire/master.c | 2 +- drivers/soundwire/slave.c | 2 +- include/linux/soundwire/sdw_type.h | 4 ++-- 3 files changed, 4 insertions(+), 4 deletions(-) diff --git a/drivers/soundwire/master.c b/drivers/soundwire/master.c index 51abedbbaa66..b2c64512739d 100644 --- a/drivers/soundwire/master.c +++ b/drivers/soundwire/master.c @@ -112,7 +112,7 @@ static const struct dev_pm_ops master_dev_pm = { pm_generic_runtime_resume, NULL) }; -struct device_type sdw_master_type = { +const struct device_type sdw_master_type = { .name = "soundwire_master", .release = sdw_master_device_release, .pm = &master_dev_pm, diff --git a/drivers/soundwire/slave.c b/drivers/soundwire/slave.c index 060c2982e26b..9963b92eb505 100644 --- a/drivers/soundwire/slave.c +++ b/drivers/soundwire/slave.c @@ -16,7 +16,7 @@ static void sdw_slave_release(struct device *dev) kfree(slave); } -struct device_type sdw_slave_type = { +const struct device_type sdw_slave_type = { .name = "sdw_slave", .release = sdw_slave_release, .uevent = sdw_slave_uevent, diff --git a/include/linux/soundwire/sdw_type.h b/include/linux/soundwire/sdw_type.h index b445f7200f06..693320b4f5c2 100644 --- a/include/linux/soundwire/sdw_type.h +++ b/include/linux/soundwire/sdw_type.h @@ -5,8 +5,8 @@ #define __SOUNDWIRE_TYPES_H extern const struct bus_type sdw_bus_type; -extern struct device_type sdw_slave_type; -extern struct device_type sdw_master_type; +extern const struct device_type sdw_slave_type; +extern const struct device_type sdw_master_type; static inline int is_sdw_slave(const struct device *dev) { --- base-commit: 81a7d0c4d059cb5c122110acbeec7bedfb91a741 change-id: 20240219-device_cleanup-soundwire-c037fffa2746 Best regards, -- Ricardo B. Marliere <ricardo(a)marliere.net>

2 1

[PATCH] soundwire: Use snd_soc_substream_to_rtd() to obtain rtd
by Amadeusz Sławiński 03 Mar '24

03 Mar '24

From: Cezary Rojewski <cezary.rojewski(a)intel.com> Utilize the helper function instead of casting from ->private_data directly. Signed-off-by: Cezary Rojewski <cezary.rojewski(a)intel.com> Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> --- drivers/soundwire/stream.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/drivers/soundwire/stream.c b/drivers/soundwire/stream.c index 69719b335bcb..652aae4e221d 100644 --- a/drivers/soundwire/stream.c +++ b/drivers/soundwire/stream.c @@ -1717,7 +1717,7 @@ EXPORT_SYMBOL(sdw_deprepare_stream); static int set_stream(struct snd_pcm_substream *substream, struct sdw_stream_runtime *sdw_stream) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct snd_soc_dai *dai; int ret = 0; int i; @@ -1770,7 +1770,7 @@ EXPORT_SYMBOL(sdw_alloc_stream); int sdw_startup_stream(void *sdw_substream) { struct snd_pcm_substream *substream = sdw_substream; - struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct sdw_stream_runtime *sdw_stream; char *name; int ret; @@ -1814,7 +1814,7 @@ EXPORT_SYMBOL(sdw_startup_stream); void sdw_shutdown_stream(void *sdw_substream) { struct snd_pcm_substream *substream = sdw_substream; - struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct sdw_stream_runtime *sdw_stream; struct snd_soc_dai *dai; -- 2.34.1

2 2

Internal mic does not work on IdeaPad 3 Chromebook
by GitHub issues - opened 03 Mar '24

03 Mar '24

alsa-project/alsa-ucm-conf issue #397 was opened from abhi-deshpande: Device - IdeaPad 3 Chromebook (Octopus Lick) OS - Fedora 39 Card - sof-glkda7219max I recently converted my Chromebook to Chultrabook and installed latest Fedora on it. It seems that most of the audio stuff works, but the internal DMIC doesn't work. I read some previous issues and commits and found that this is due to the fact that the DMIC needs channel split as CRAS does. This was being implemented in #154 I guess. I am currently on lastest alsa-lib and alsa-ucm-conf. What could be missing ? Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/397 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH V3 00/13] soundwire/SOF: add SoundWire Interface support for AMD SOF stack
by Vijendar Mukunda 02 Mar '24

02 Mar '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 platform). Below patch series is reviewed in https://github.com/thesofproject/linux/pull/4699 Changes since V1: - add new commit to update license for amd soundwire manager driver files. - wrap the commit message with in 75 characters. - add amd tag in commit title for soundwire patches Changes since V2: - drop "drivers" tag from commit title for SoundWire patches. This patch series has dependencies. "drivers: soundwire: amd: refactor amd soundwire manager device node creation" patch has dependency on below patch. "ASoC/soundwire: implement generic api for scanning amd soundwire controller" Similarly SOF patches has dependency on Soundwire tree patches. This patch series has to be reviewed and merged in one go. Vijendar Mukunda (13): ASoC/soundwire: implement generic api for scanning amd soundwire controller soundwire: amd: update license soundwire: amd: refactor amd soundwire manager device node creation soundwire: amd: implement function to extract slave information soundwire: amd: refactor soundwire pads enable soundwire: amd: 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 | 45 ++---- drivers/soundwire/amd_manager.h | 14 +- include/linux/soundwire/sdw_amd.h | 81 ++++++++-- 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, 802 insertions(+), 71 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

3 15