June 2021 - Alsa-devel - mailman.alsa-project.org

[PATCH] ASoC: mediatek: mtk-btcvsd: Fix an error handling path in 'mtk_btcvsd_snd_probe()'
by Christophe JAILLET 08 Jun '21

08 Jun '21

If an error occurs after a successful 'of_iomap()' call, it must be undone by a corresponding 'iounmap()' call, as already done in the remove function. While at it, remove the useless initialization of 'ret' at the beginning of the function. Fixes: 4bd8597dc36c ("ASoC: mediatek: add btcvsd driver") Signed-off-by: Christophe JAILLET <christophe.jaillet(a)wanadoo.fr> --- This is NOT compile tested. Even if CONFIG_SND_SOC_MTK_BTCVSD is correctly set in my .config file, I've not been able to have this file compiled. --- sound/soc/mediatek/common/mtk-btcvsd.c | 24 ++++++++++++++++++------ 1 file changed, 18 insertions(+), 6 deletions(-) diff --git a/sound/soc/mediatek/common/mtk-btcvsd.c b/sound/soc/mediatek/common/mtk-btcvsd.c index f85b5ea180ec..d884bb7c0fc7 100644 --- a/sound/soc/mediatek/common/mtk-btcvsd.c +++ b/sound/soc/mediatek/common/mtk-btcvsd.c @@ -1281,7 +1281,7 @@ static const struct snd_soc_component_driver mtk_btcvsd_snd_platform = { static int mtk_btcvsd_snd_probe(struct platform_device *pdev) { - int ret = 0; + int ret; int irq_id; u32 offset[5] = {0, 0, 0, 0, 0}; struct mtk_btcvsd_snd *btcvsd; @@ -1337,7 +1337,8 @@ static int mtk_btcvsd_snd_probe(struct platform_device *pdev) btcvsd->bt_sram_bank2_base = of_iomap(dev->of_node, 1); if (!btcvsd->bt_sram_bank2_base) { dev_err(dev, "iomap bt_sram_bank2_base fail\n"); - return -EIO; + ret = -EIO; + goto unmap_pkv_err; } btcvsd->infra = syscon_regmap_lookup_by_phandle(dev->of_node, @@ -1345,7 +1346,8 @@ static int mtk_btcvsd_snd_probe(struct platform_device *pdev) if (IS_ERR(btcvsd->infra)) { dev_err(dev, "cannot find infra controller: %ld\n", PTR_ERR(btcvsd->infra)); - return PTR_ERR(btcvsd->infra); + ret = PTR_ERR(btcvsd->infra); + goto unmap_bank2_err; } /* get offset */ @@ -1354,7 +1356,7 @@ static int mtk_btcvsd_snd_probe(struct platform_device *pdev) ARRAY_SIZE(offset)); if (ret) { dev_warn(dev, "%s(), get offset fail, ret %d\n", __func__, ret); - return ret; + goto unmap_bank2_err; } btcvsd->infra_misc_offset = offset[0]; btcvsd->conn_bt_cvsd_mask = offset[1]; @@ -1373,8 +1375,18 @@ static int mtk_btcvsd_snd_probe(struct platform_device *pdev) mtk_btcvsd_snd_set_state(btcvsd, btcvsd->tx, BT_SCO_STATE_IDLE); mtk_btcvsd_snd_set_state(btcvsd, btcvsd->rx, BT_SCO_STATE_IDLE); - return devm_snd_soc_register_component(dev, &mtk_btcvsd_snd_platform, - NULL, 0); + ret = devm_snd_soc_register_component(dev, &mtk_btcvsd_snd_platform, + NULL, 0); + if (ret) + goto unmap_bank2_err; + + return 0; + +unmap_bank2_err: + iounmap(btcvsd->bt_sram_bank2_base); +unmap_pkv_err: + iounmap(btcvsd->bt_pkv_base); + return ret; } static int mtk_btcvsd_snd_remove(struct platform_device *pdev) -- 2.30.2

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[PATCH] ASoC: remove unneeded semicolons in wcd934x.c
by Wan Jiabing 08 Jun '21

08 Jun '21

Fix following coccicheck warning: ./sound/soc/codecs/wcd934x.c:5136:2-3: Unneeded semicolon ./sound/soc/codecs/wcd934x.c:2466:2-3: Unneeded semicolon ./sound/soc/codecs/wcd934x.c:2527:2-3: Unneeded semicolon Signed-off-by: Wan Jiabing <wanjiabing(a)vivo.com> --- sound/soc/codecs/wcd934x.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wcd934x.c b/sound/soc/codecs/wcd934x.c index 16fd1ab62609..c496b359f2f4 100644 --- a/sound/soc/codecs/wcd934x.c +++ b/sound/soc/codecs/wcd934x.c @@ -2463,7 +2463,7 @@ static int wcd934x_micbias_control(struct snd_soc_component *component, dev_err(component->dev, "%s: Invalid micbias number: %d\n", __func__, micb_num); return -EINVAL; - }; + } mutex_lock(&wcd934x->micb_lock); switch (req) { @@ -2524,7 +2524,7 @@ static int wcd934x_micbias_control(struct snd_soc_component *component, wcd_mbhc_event_notify(wcd934x->mbhc, WCD_EVENT_POST_DAPM_MICBIAS_2_OFF); break; - }; + } mutex_unlock(&wcd934x->micb_lock); @@ -5133,7 +5133,7 @@ static int wcd934x_codec_enable_micbias(struct snd_soc_dapm_widget *w, case SND_SOC_DAPM_POST_PMD: wcd934x_micbias_control(component, micb_num, MICB_DISABLE, true); break; - }; + } return 0; } -- 2.20.1

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[GIT PULL] ASoC fixes for v5.13-rc4
by Mark Brown 08 Jun '21

08 Jun '21

The following changes since commit af2702549d68519ac78228e915d9b2c199056787: ASoC: qcom: lpass-cpu: Use optional clk APIs (2021-05-21 13:12:29 +0100) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v5.13-rc4 for you to fetch changes up to c8a4556d98510ca05bad8d02265a4918b03a8c0b: ASoC: qcom: lpass-cpu: Fix pop noise during audio capture begin (2021-06-07 15:54:08 +0100) ---------------------------------------------------------------- ASoC: Fixes for v5.13 A collection of fixes and device ID updates that have come up in the past few -rcs, none of which stand out particularly. ---------------------------------------------------------------- Colin Ian King (1): ASoC: topology: Fix spelling mistake "vesion" -> "version" Jack Yu (1): ASoC: rt5659: Fix the lost powers for the HDA header Jerome Brunet (1): ASoC: meson: gx-card: fix sound-dai dt schema Kai Vehmanen (1): ASoC: SOF: reset enabled_cores state at suspend Kefeng Wang (1): ASoC: core: Fix Null-point-dereference in fmt_single_name() Mark Pearson (2): ASoC: AMD Renoir - add DMI entry for Lenovo 2020 AMD platforms ASoC: AMD Renoir: Remove fix for DMI entry on Lenovo 2020 platforms Nicolas Cavallari (1): ASoC: fsl-asoc-card: Set .owner attribute when registering card. Oder Chiou (1): ASoC: rt5682: Fix the fast discharge for headset unplugging in soundwire mode Richard Weinberger (1): ASoC: tas2562: Fix TDM_CFG0_SAMPRATE values Srinivasa Rao Mandadapu (1): ASoC: qcom: lpass-cpu: Fix pop noise during audio capture begin .../bindings/sound/amlogic,gx-sound-card.yaml | 4 +- sound/soc/codecs/rt5659.c | 26 +++++-- sound/soc/codecs/rt5682-sdw.c | 3 +- sound/soc/codecs/tas2562.h | 14 ++-- sound/soc/fsl/fsl-asoc-card.c | 1 + sound/soc/qcom/lpass-cpu.c | 79 ++++++++++++++++++++++ sound/soc/qcom/lpass.h | 4 ++ sound/soc/soc-core.c | 2 + sound/soc/soc-topology.c | 6 +- sound/soc/sof/pm.c | 1 + 10 files changed, 122 insertions(+), 18 deletions(-)

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[GIT PULL] HDMI codec improvements
by Maxime Ripard 08 Jun '21

08 Jun '21

Hi, Here's a PR for the changes to hdmi-codec that need to be shared between drm and ASoC. Thanks! Maxime The following changes since commit 6efb943b8616ec53a5e444193dccf1af9ad627b5: Linux 5.13-rc1 (2021-05-09 14:17:44 -0700) are available in the Git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/mripard/linux.git tags/asoc-hdmi-codec-improvements for you to fetch changes up to aee87e9c8fc0efbe933f0a8371990a0754ac65fd: ASoC: hdmi-codec: Add a prepare hook (2021-06-08 17:06:00 +0200) ---------------------------------------------------------------- Improvements to the hdmi-codec driver and ALSA infrastructure around it to support the HDMI Channel Mapping and IEC958 controls ---------------------------------------------------------------- Maxime Ripard (5): ALSA: doc: Clarify IEC958 controls iface ALSA: iec958: Split status creation and fill ASoC: hdmi-codec: Rework to support more controls ASoC: hdmi-codec: Add iec958 controls ASoC: hdmi-codec: Add a prepare hook .../sound/kernel-api/writing-an-alsa-driver.rst | 13 +- include/sound/hdmi-codec.h | 12 +- include/sound/pcm_iec958.h | 8 + sound/core/pcm_iec958.c | 176 +++++++++++++---- sound/soc/codecs/hdmi-codec.c | 219 +++++++++++++++++---- 5 files changed, 337 insertions(+), 91 deletions(-)

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[PATCH 00/66] ALSA: Fix assignment in if condition
by Takashi Iwai 08 Jun '21

08 Jun '21

Hi, here is a largish patch set mostly for cleanup / a slight refactoring of the coding styles with assignment in if condition, typically such as: if ((err = xxx()) < 0) return err; This was a popular style in the old ages, but nowadays it's seen rather fairly harmful as it makes harder to read and easily hides bugs. Since we've hit quite a few issues (although most of them are trivial), I decided to fix those lines. While conversions, I spotted a couple of real bugs. They are old driver codes and only in certain error paths, so most likely no one ever hits it. Takashi === Takashi Iwai (66): ALSA: sb: Fix assignment in if condition ALSA: sb: Minor coding style fixes ALSA: sb: Fix potential double-free of CSP mixer elements ALSA: gus: Fix assignment in if condition ALSA: ad1816a: Fix assignment in if condition ALSA: wavefront: Fix assignment in if condition ALSA: cs423x: Fix assignment in if condition ALSA: opti9xx: Fix assignment in if condition ALSA: opl3sa2: Fix assignment in if condition ALSA: es18xx: Fix assignment in if condition ALSA: cmi8330: Fix assignment in if condition ALSA: als100: Fix assignment in if condition ALSA: azt2320: Fix assignment in if condition ALSA: isa: Fix assignment in if condition ALSA: ad1889: Fix assignment in if condition ALSA: ak4531: Fix assignment in if condition ALSA: als300: Fix assignment in if condition ALSA: als4000: Fix assignment in if condition ALSA: atiixp: Fix assignment in if condition ALSA: azt3328: Fix assignment in if condition ALSA: bt87x: Fix assignment in if condition ALSA: cmipci: Fix assignment in if condition ALSA: cs4281: Fix assignment in if condition ALSA: ens137x: Fix assignment in if condition ALSA: es1938: Fix assignment in if condition ALSA: es1968: Fix assignment in if condition ALSA: fm801: Fix assignment in if condition ALSA: intel8x0: Fix assignment in if condition ALSA: maestro3: Fix assignment in if condition ALSA: rme32: Fix assignment in if condition ALSA: rme96: Fix assignment in if condition ALSA: sonicvibes: Fix assignment in if condition ALSA: via82xx: Fix assignment in if condition ALSA: ac97: Fix assignment in if condition ALSA: au88x0: Fix assignment in if condition ALSA: ca0106: Fix assignment in if condition ALSA: cs46xx: Fix assignment in if condition ALSA: cs5535audio: Fix assignment in if condition ALSA: echoaudio: Fix assignment in if condition ALSA: emu10k1: Fix assignment in if condition ALSA: emu10k1x: Fix assignment in if condition ALSA: ice1712: Fix assignment in if condition ALSA: korg1212: Fix assignment in if condition ALSA: mixart: Fix assignment in if condition ALSA: nm256: Fix assignment in if condition ALSA: pcxhr: Fix assignment in if condition ALSA: riptide: Fix assignment in if condition ALSA: hdsp: Fix assignment in if condition ALSA: rme9652: Fix assignment in if condition ALSA: trident: Fix assignment in if condition ALSA: vx222: Fix assignment in if condition ALSA: ymfpci: Fix assignment in if condition ALSA: core: Fix assignment in if condition ALSA: pcm: Fix assignment in if condition ALSA: oss: Fix assignment in if condition ALSA: seq: Fix assignment in if condition ALSA: pcmcia: Fix assignment in if condition ALSA: sparc: Fix assignment in if condition ALSA: mpu401: Fix assignment in if condition ALSA: vx: Fix assignment in if condition ALSA: opl3: Fix assignment in if condition ALSA: serial: Fix assignment in if condition ALSA: synth: Fix assignment in if condition ALSA: poewrmac: Fix assignment in if condition ALSA: i2c: Fix assignment in if condition ALSA: parisc: Fix assignment in if condition sound/core/hwdep.c | 6 +- sound/core/info_oss.c | 3 +- sound/core/init.c | 12 +- sound/core/oss/mixer_oss.c | 45 +++-- sound/core/oss/pcm_oss.c | 70 +++++--- sound/core/oss/pcm_plugin.c | 26 +-- sound/core/pcm_compat.c | 6 +- sound/core/pcm_misc.c | 12 +- sound/core/pcm_native.c | 6 +- sound/core/seq/oss/seq_oss.c | 26 +-- sound/core/seq/oss/seq_oss_init.c | 9 +- sound/core/seq/oss/seq_oss_midi.c | 33 ++-- sound/core/seq/oss/seq_oss_rw.c | 3 +- sound/core/seq/oss/seq_oss_synth.c | 9 +- sound/core/seq/oss/seq_oss_writeq.c | 3 +- sound/core/seq/seq_clientmgr.c | 51 ++++-- sound/core/seq/seq_dummy.c | 3 +- sound/core/seq/seq_fifo.c | 3 +- sound/core/seq/seq_memory.c | 6 +- sound/core/seq/seq_midi.c | 27 +-- sound/core/seq/seq_queue.c | 21 ++- sound/core/seq/seq_virmidi.c | 9 +- sound/core/sound.c | 3 +- sound/core/sound_oss.c | 3 +- sound/drivers/mpu401/mpu401.c | 9 +- sound/drivers/mpu401/mpu401_uart.c | 19 +- sound/drivers/mtpav.c | 15 +- sound/drivers/mts64.c | 15 +- sound/drivers/opl3/opl3_lib.c | 42 +++-- sound/drivers/opl3/opl3_oss.c | 6 +- sound/drivers/opl3/opl3_seq.c | 9 +- sound/drivers/portman2x4.c | 15 +- sound/drivers/serial-u16550.c | 27 +-- sound/drivers/vx/vx_core.c | 60 ++++--- sound/drivers/vx/vx_hwdep.c | 12 +- sound/drivers/vx/vx_mixer.c | 39 +++-- sound/drivers/vx/vx_pcm.c | 32 ++-- sound/i2c/cs8427.c | 24 ++- sound/i2c/other/ak4114.c | 3 +- sound/i2c/other/ak4117.c | 3 +- sound/i2c/tea6330t.c | 24 ++- sound/isa/ad1816a/ad1816a.c | 23 ++- sound/isa/ad1816a/ad1816a_lib.c | 27 ++- sound/isa/als100.c | 18 +- sound/isa/azt2320.c | 21 ++- sound/isa/cmi8330.c | 37 ++-- sound/isa/cs423x/cs4236.c | 21 ++- sound/isa/cs423x/cs4236_lib.c | 12 +- sound/isa/es1688/es1688_lib.c | 3 +- sound/isa/es18xx.c | 51 ++++-- sound/isa/gus/gus_main.c | 18 +- sound/isa/gus/gus_mem.c | 3 +- sound/isa/gus/gus_mixer.c | 6 +- sound/isa/gus/gus_pcm.c | 23 ++- sound/isa/gus/gus_uart.c | 3 +- sound/isa/gus/gusclassic.c | 6 +- sound/isa/gus/gusextreme.c | 6 +- sound/isa/gus/gusmax.c | 36 ++-- sound/isa/gus/interwave.c | 90 ++++++---- sound/isa/opl3sa2.c | 68 +++++--- sound/isa/opti9xx/miro.c | 27 ++- sound/isa/opti9xx/opti92x-ad1848.c | 27 ++- sound/isa/sb/emu8000.c | 21 ++- sound/isa/sb/emu8000_patch.c | 3 +- sound/isa/sb/emu8000_pcm.c | 6 +- sound/isa/sb/sb16.c | 65 ++++--- sound/isa/sb/sb16_csp.c | 29 ++-- sound/isa/sb/sb16_main.c | 6 +- sound/isa/sb/sb8.c | 38 ++-- sound/isa/sb/sb8_main.c | 3 +- sound/isa/sb/sb8_midi.c | 3 +- sound/isa/sb/sb_common.c | 9 +- sound/isa/sb/sb_mixer.c | 55 +++--- sound/isa/wavefront/wavefront.c | 6 +- sound/isa/wavefront/wavefront_midi.c | 20 ++- sound/isa/wavefront/wavefront_synth.c | 56 +++--- sound/isa/wss/wss_lib.c | 9 +- sound/parisc/harmony.c | 7 +- sound/pci/ac97/ac97_codec.c | 205 ++++++++++++++-------- sound/pci/ac97/ac97_patch.c | 137 ++++++++++----- sound/pci/ad1889.c | 15 +- sound/pci/ak4531_codec.c | 9 +- sound/pci/als300.c | 22 ++- sound/pci/als4000.c | 55 +++--- sound/pci/atiixp.c | 33 ++-- sound/pci/atiixp_modem.c | 40 +++-- sound/pci/au88x0/au88x0.c | 77 +++++---- sound/pci/au88x0/au88x0_a3d.c | 28 +-- sound/pci/au88x0/au88x0_core.c | 47 ++--- sound/pci/au88x0/au88x0_eq.c | 20 ++- sound/pci/au88x0/au88x0_mixer.c | 3 +- sound/pci/au88x0/au88x0_mpu401.c | 14 +- sound/pci/au88x0/au88x0_pcm.c | 15 +- sound/pci/azt3328.c | 3 +- sound/pci/bt87x.c | 3 +- sound/pci/ca0106/ca0106_main.c | 18 +- sound/pci/ca0106/ca_midi.c | 3 +- sound/pci/cmipci.c | 83 ++++++--- sound/pci/cs4281.c | 54 ++++-- sound/pci/cs46xx/cs46xx.c | 31 ++-- sound/pci/cs46xx/cs46xx_lib.c | 51 ++++-- sound/pci/cs46xx/dsp_spos.c | 3 +- sound/pci/cs5535audio/cs5535audio.c | 28 +-- sound/pci/cs5535audio/cs5535audio_pcm.c | 10 +- sound/pci/echoaudio/darla20_dsp.c | 6 +- sound/pci/echoaudio/darla24_dsp.c | 6 +- sound/pci/echoaudio/echo3g_dsp.c | 3 +- sound/pci/echoaudio/echoaudio.c | 220 ++++++++++++++---------- sound/pci/echoaudio/echoaudio_dsp.c | 12 +- sound/pci/echoaudio/echoaudio_gml.c | 3 +- sound/pci/echoaudio/gina20_dsp.c | 6 +- sound/pci/echoaudio/gina24_dsp.c | 6 +- sound/pci/echoaudio/indigo_dsp.c | 6 +- sound/pci/echoaudio/indigodj_dsp.c | 6 +- sound/pci/echoaudio/indigoio_dsp.c | 6 +- sound/pci/echoaudio/layla20_dsp.c | 6 +- sound/pci/echoaudio/layla24_dsp.c | 9 +- sound/pci/echoaudio/mia_dsp.c | 6 +- sound/pci/echoaudio/midi.c | 4 +- sound/pci/echoaudio/mona_dsp.c | 6 +- sound/pci/emu10k1/emu10k1.c | 42 +++-- sound/pci/emu10k1/emu10k1_callback.c | 3 +- sound/pci/emu10k1/emu10k1x.c | 85 +++++---- sound/pci/emu10k1/emufx.c | 13 +- sound/pci/emu10k1/emumixer.c | 78 ++++++--- sound/pci/emu10k1/emumpu401.c | 12 +- sound/pci/emu10k1/emupcm.c | 24 ++- sound/pci/emu10k1/memory.c | 21 ++- sound/pci/emu10k1/p16v.c | 13 +- sound/pci/emu10k1/timer.c | 3 +- sound/pci/ens1370.c | 45 +++-- sound/pci/es1938.c | 42 +++-- sound/pci/es1968.c | 75 ++++---- sound/pci/fm801.c | 63 ++++--- sound/pci/ice1712/delta.c | 6 +- sound/pci/ice1712/ews.c | 24 ++- sound/pci/intel8x0.c | 46 +++-- sound/pci/intel8x0m.c | 36 ++-- sound/pci/korg1212/korg1212.c | 24 ++- sound/pci/maestro3.c | 21 ++- sound/pci/mixart/mixart.c | 29 ++-- sound/pci/mixart/mixart_hwdep.c | 17 +- sound/pci/mixart/mixart_mixer.c | 33 ++-- sound/pci/nm256/nm256.c | 27 ++- sound/pci/pcxhr/pcxhr.c | 22 ++- sound/pci/pcxhr/pcxhr_hwdep.c | 9 +- sound/pci/riptide/riptide.c | 88 ++++++---- sound/pci/rme32.c | 76 +++++--- sound/pci/rme96.c | 148 +++++++++------- sound/pci/rme9652/hdsp.c | 121 ++++++++----- sound/pci/rme9652/rme9652.c | 98 ++++++----- sound/pci/sonicvibes.c | 67 +++++--- sound/pci/trident/trident.c | 41 +++-- sound/pci/trident/trident_main.c | 84 ++++++--- sound/pci/via82xx.c | 78 ++++++--- sound/pci/via82xx_modem.c | 52 ++++-- sound/pci/vx222/vx222.c | 18 +- sound/pci/vx222/vx222_ops.c | 12 +- sound/pci/ymfpci/ymfpci.c | 71 ++++---- sound/pci/ymfpci/ymfpci_main.c | 72 +++++--- sound/pcmcia/pdaudiocf/pdaudiocf.c | 3 +- sound/pcmcia/vx/vxp_mixer.c | 6 +- sound/pcmcia/vx/vxp_ops.c | 6 +- sound/pcmcia/vx/vxpocket.c | 3 +- sound/ppc/beep.c | 5 +- sound/ppc/daca.c | 24 ++- sound/ppc/keywest.c | 6 +- sound/ppc/pmac.c | 6 +- sound/ppc/powermac.c | 21 ++- sound/ppc/tumbler.c | 72 +++++--- sound/sparc/amd7930.c | 14 +- sound/sparc/cs4231.c | 10 +- sound/sparc/dbri.c | 11 +- sound/synth/emux/emux.c | 3 +- sound/synth/emux/emux_effect.c | 13 +- sound/synth/emux/emux_hwdep.c | 6 +- sound/synth/emux/soundfont.c | 40 +++-- 177 files changed, 3186 insertions(+), 1831 deletions(-) -- 2.26.2

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[PATCH 0/2] ASoC: Intel: boards: use software node API
by Pierre-Louis Bossart 08 Jun '21

08 Jun '21

This is an update on an earlier contribution from Heikki Krogerus, with mainly changes to avoid the use of devm_ routines for Baytrail machine drivers. Heikki Krogerus (2): ASoC: Intel: boards: use software node API in SoundWire machines ASoC: Intel: boards: use software node API in Atom boards sound/soc/intel/boards/bytcht_es8316.c | 25 +++++++++-- sound/soc/intel/boards/bytcr_rt5640.c | 42 +++++++++++++++--- sound/soc/intel/boards/bytcr_rt5651.c | 47 +++++++++++++++++---- sound/soc/intel/boards/sof_sdw_rt711.c | 20 ++++++--- sound/soc/intel/boards/sof_sdw_rt711_sdca.c | 20 ++++++--- 5 files changed, 127 insertions(+), 27 deletions(-) -- 2.25.1

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[PATCH v3] ASoC: qcom: Add four speaker support on MI2S secondary
by Srinivasa Rao Mandadapu 08 Jun '21

08 Jun '21

Add four speaker support on MI2S secondary block by using I2S SD1 line on gpio52 pin, and add channel map control support in the lpass-cpu audio driver. Signed-off-by: Srinivasa Rao Mandadapu <srivasam(a)codeaurora.org> --- sound/soc/qcom/lpass-cpu.c | 40 ++++++++++++++++++++++++++++++++++- sound/soc/qcom/lpass-sc7180.c | 1 + sound/soc/qcom/lpass.h | 8 ++++++- 3 files changed, 47 insertions(+), 2 deletions(-) diff --git a/sound/soc/qcom/lpass-cpu.c b/sound/soc/qcom/lpass-cpu.c index c62d2612e8f5..d03cc359983a 100644 --- a/sound/soc/qcom/lpass-cpu.c +++ b/sound/soc/qcom/lpass-cpu.c @@ -324,10 +324,43 @@ const struct snd_soc_dai_ops asoc_qcom_lpass_cpu_dai_ops = { }; EXPORT_SYMBOL_GPL(asoc_qcom_lpass_cpu_dai_ops); +int lpass_cpu_pcm_new(struct snd_soc_pcm_runtime *rtd, + struct snd_soc_dai *dai) +{ + int ret; + struct snd_soc_dai_driver *drv = dai->driver; + struct lpass_data *drvdata = snd_soc_dai_get_drvdata(dai); + + ret = snd_pcm_add_chmap_ctls(rtd->pcm, SNDRV_PCM_STREAM_PLAYBACK, + snd_pcm_alt_chmaps, drv->playback.channels_max, 0, + &drvdata->chmap_info); + if (ret < 0) + return ret; + + return 0; +} +EXPORT_SYMBOL_GPL(lpass_cpu_pcm_new); + int asoc_qcom_lpass_cpu_dai_probe(struct snd_soc_dai *dai) { struct lpass_data *drvdata = snd_soc_dai_get_drvdata(dai); - int ret; + struct snd_soc_dai_driver *drv = dai->driver; + int ret, i; + + drvdata->rx_chmap.channels = drv->playback.channels_max; + drvdata->tx_chmap.channels = drv->playback.channels_max; + + if (drv->playback.channels_max == 2) + drvdata->chmap_idx = 1; + else if (drv->playback.channels_max == 4) + drvdata->chmap_idx = 2; + + for (i = 0; i < drv->playback.channels_max; i++) { + drvdata->tx_chmap.map[i] = + snd_pcm_alt_chmaps[drvdata->chmap_idx].map[i]; + drvdata->rx_chmap.map[i] = + snd_pcm_alt_chmaps[drvdata->chmap_idx].map[i]; + } /* ensure audio hardware is disabled */ ret = regmap_write(drvdata->lpaif_map, @@ -856,6 +889,11 @@ int asoc_qcom_lpass_cpu_platform_probe(struct platform_device *pdev) PTR_ERR(drvdata->mi2s_bit_clk[dai_id])); return PTR_ERR(drvdata->mi2s_bit_clk[dai_id]); } + if (drvdata->mi2s_playback_sd_mode[dai_id] == + LPAIF_I2SCTL_MODE_QUAD01) { + variant->dai_driver[dai_id].playback.channels_min = 4; + variant->dai_driver[dai_id].playback.channels_max = 4; + } } /* Allocation for i2sctl regmap fields */ diff --git a/sound/soc/qcom/lpass-sc7180.c b/sound/soc/qcom/lpass-sc7180.c index 8c168d3c589e..77a556b27cf0 100644 --- a/sound/soc/qcom/lpass-sc7180.c +++ b/sound/soc/qcom/lpass-sc7180.c @@ -58,6 +58,7 @@ static struct snd_soc_dai_driver sc7180_lpass_cpu_dai_driver[] = { }, .probe = &asoc_qcom_lpass_cpu_dai_probe, .ops = &asoc_qcom_lpass_cpu_dai_ops, + .pcm_new = lpass_cpu_pcm_new, }, { .id = LPASS_DP_RX, .name = "Hdmi", diff --git a/sound/soc/qcom/lpass.h b/sound/soc/qcom/lpass.h index 83b2e08ade06..db239ea52946 100644 --- a/sound/soc/qcom/lpass.h +++ b/sound/soc/qcom/lpass.h @@ -110,6 +110,11 @@ struct lpass_data { struct lpass_hdmitx_dmactl *hdmi_tx_dmactl[LPASS_MAX_HDMI_DMA_CHANNELS]; struct lpass_dp_metadata_ctl *meta_ctl; struct lpass_sstream_ctl *sstream_ctl; + /* Channel map information */ + struct snd_pcm_chmap *chmap_info; + struct snd_pcm_chmap_elem rx_chmap; + struct snd_pcm_chmap_elem tx_chmap; + int chmap_idx; }; /* Vairant data per each SOC */ @@ -259,5 +264,6 @@ void asoc_qcom_lpass_cpu_platform_shutdown(struct platform_device *pdev); int asoc_qcom_lpass_cpu_platform_probe(struct platform_device *pdev); int asoc_qcom_lpass_cpu_dai_probe(struct snd_soc_dai *dai); extern const struct snd_soc_dai_ops asoc_qcom_lpass_cpu_dai_ops; - +int lpass_cpu_pcm_new(struct snd_soc_pcm_runtime *rtd, + struct snd_soc_dai *dai); #endif /* __LPASS_H__ */ -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

1 0

[PATCH V2 0/2] Read all configuration at init time
by Geoffrey D. Bennett 08 Jun '21

08 Jun '21

Version two of patches to add support for reading the mixer volumes and mux configuration from the hardware when the driver is initialising. Changes since v1: - fix missing endian conversion spotted by ktest bot Geoffrey D. Bennett (2): ALSA: usb-audio: scarlett2: Read mixer volumes at init time ALSA: usb-audio: scarlett2: Read mux at init time sound/usb/mixer_scarlett_gen2.c | 227 ++++++++++++++++++++++---------- 1 file changed, 161 insertions(+), 66 deletions(-) -- 2.31.1

2 3

[PATCH] ALSA: hda/realtek: fix mute/micmute LEDs for HP ZBook Power G8
by Jeremy Szu 08 Jun '21

08 Jun '21

The HP ZBook Power G8 using ALC236 codec which using 0x02 to control mute LED and 0x01 to control micmute LED. Therefore, add a quirk to make it works. Signed-off-by: Jeremy Szu <jeremy.szu(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index dcd9bceace68..f2fa6953b8b7 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8335,6 +8335,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x886d, "HP ZBook Fury 17.3 Inch G8 Mobile Workstation PC", ALC285_FIXUP_HP_GPIO_AMP_INIT), SND_PCI_QUIRK(0x103c, 0x8870, "HP ZBook Fury 15.6 Inch G8 Mobile Workstation PC", ALC285_FIXUP_HP_GPIO_AMP_INIT), SND_PCI_QUIRK(0x103c, 0x8873, "HP ZBook Studio 15.6 Inch G8 Mobile Workstation PC", ALC285_FIXUP_HP_GPIO_AMP_INIT), + SND_PCI_QUIRK(0x103c, 0x888d, "HP ZBook Power 15.6 inch G8 Mobile Workstation PC", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8896, "HP EliteBook 855 G8 Notebook PC", ALC285_FIXUP_HP_MUTE_LED), SND_PCI_QUIRK(0x103c, 0x8898, "HP EliteBook 845 G8 Notebook PC", ALC285_FIXUP_HP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x1043, 0x103e, "ASUS X540SA", ALC256_FIXUP_ASUS_MIC), -- 2.31.1

2 1

[PATCH] ALSA: hda/realtek: headphone and mic don't work on an Acer laptop
by Hui Wang 08 Jun '21

08 Jun '21

There are 2 issues on this machine, the 1st one is mic's plug/unplug can't be detected, that is because the mic is set to manual detecting mode, need to apply ALC255_FIXUP_XIAOMI_HEADSET_MIC to set it to auto detecting mode. The other one is headphone's plug/unplug can't be detected by pulseaudio, that is because the pulseaudio will use ucm2/sof-hda-dsp on this machine, and the ucm2 only handle 'Headphone Jack', but on this machine the headphone's pincfg sets the location to Front, then the alsa mixer name is "Front Headphone Jack" instead of "Headphone Jack", so override the pincfg to change location to Left. BugLink: http://bugs.launchpad.net/bugs/1930188 Cc: <stable(a)vger.kernel.org> Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index dcd9bceace68..be4a26f6bb96 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6568,6 +6568,7 @@ enum { ALC285_FIXUP_HP_SPECTRE_X360, ALC287_FIXUP_IDEAPAD_BASS_SPK_AMP, ALC623_FIXUP_LENOVO_THINKSTATION_P340, + ALC255_FIXUP_ACER_HEADPHONE_AND_MIC, }; static const struct hda_fixup alc269_fixups[] = { @@ -8146,6 +8147,15 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC283_FIXUP_HEADSET_MIC, }, + [ALC255_FIXUP_ACER_HEADPHONE_AND_MIC] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x21, 0x03211030 }, /* Change the Headphone location to Left */ + { } + }, + .chained = true, + .chain_id = ALC255_FIXUP_XIAOMI_HEADSET_MIC + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -8182,6 +8192,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x132a, "Acer TravelMate B114-21", ALC233_FIXUP_ACER_HEADSET_MIC), SND_PCI_QUIRK(0x1025, 0x1330, "Acer TravelMate X514-51T", ALC255_FIXUP_ACER_HEADSET_MIC), 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(0x1028, 0x0470, "Dell M101z", ALC269_FIXUP_DELL_M101Z), SND_PCI_QUIRK(0x1028, 0x054b, "Dell XPS one 2710", ALC275_FIXUP_DELL_XPS), SND_PCI_QUIRK(0x1028, 0x05bd, "Dell Latitude E6440", ALC292_FIXUP_DELL_E7X), @@ -8740,6 +8751,7 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {.id = ALC285_FIXUP_HP_SPECTRE_X360, .name = "alc285-hp-spectre-x360"}, {.id = ALC287_FIXUP_IDEAPAD_BASS_SPK_AMP, .name = "alc287-ideapad-bass-spk-amp"}, {.id = ALC623_FIXUP_LENOVO_THINKSTATION_P340, .name = "alc623-lenovo-thinkstation-p340"}, + {.id = ALC255_FIXUP_ACER_HEADPHONE_AND_MIC, .name = "alc255-acer-headphone-and-mic"}, {} }; #define ALC225_STANDARD_PINS \ -- 2.25.1

2 1

[PATCH][next][V2] ASoC: rsnd: check for zero node count
by Colin King 08 Jun '21

08 Jun '21

From: Colin Ian King <colin.king(a)canonical.com> Most callers of_get_child_count() check that "nr" is non-zero so it causes a static checker warning when we don't do that here. This does not cause a problem or a crash, but having zero SSUIes does not make sense either so let's add a check. Addresses-Coverity: ("Unchecked return value") Fixes: c413983eb66a ("ASoC: rsnd: adjust disabled module") Acked-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- V2: Rephrase commit message based on Dan Carpenter's suggestion. Thanks Dan! --- sound/soc/sh/rcar/ssiu.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/sh/rcar/ssiu.c b/sound/soc/sh/rcar/ssiu.c index 5682c74bb7ff..0d8f97633dd2 100644 --- a/sound/soc/sh/rcar/ssiu.c +++ b/sound/soc/sh/rcar/ssiu.c @@ -515,6 +515,9 @@ int rsnd_ssiu_probe(struct rsnd_priv *priv) else nr = priv->ssi_nr; + if (!nr) + return -EINVAL; + ssiu = devm_kcalloc(dev, nr, sizeof(*ssiu), GFP_KERNEL); if (!ssiu) return -ENOMEM; -- 2.31.1

3 2

added an option to aseqnet to optionally set the midi process name
by GitHub pull_request - opened 07 Jun '21

07 Jun '21

alsa-project/alsa-utils pull request #95 was opened from BrainDamage: This option allows to run multiple instances of aseqnet without having to double check the assigned port number, since each one can get spawned with a unique name. The patch is pretty minimal and clean, thopefully here shouldn't be troubles merging it. Request URL : https://github.com/alsa-project/alsa-utils/pull/95 Patch URL : https://github.com/alsa-project/alsa-utils/pull/95.patch Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[PATCH 0/4] Support CS42L42 on JSL platform
by Brent Lu 07 Jun '21

07 Jun '21

This series consists four patches. Patch 1 adds dai sequence support for cml/jsl/tgl platforms which is different from the sequence on glk platform. Patch 2 adds max98360a support to the maxim-common module. Patch 3 adds driver data for jsl_cs4242_mx98360a which supports cs42l42 and max98360a running on jsl boards. Patch 4 refactor the sof_rt5682 to use the max98360 code in the maxim-common module. Brent Lu (4): ASoC: Intel: sof_cs42l42: support JSL DAI link sequence ASoC: Intel: maxim-common: support max98360a ASoC: intel: sof_cs42l42: add support for jsl_cs4242_mx98360a ASoC: Intel: sof_rt5682: code refactor for max98360a sound/soc/intel/boards/sof_cs42l42.c | 340 ++++++++++++------ sound/soc/intel/boards/sof_maxim_common.c | 17 +- sound/soc/intel/boards/sof_maxim_common.h | 4 +- sound/soc/intel/boards/sof_rt5682.c | 52 +-- .../intel/common/soc-acpi-intel-jsl-match.c | 8 + 5 files changed, 254 insertions(+), 167 deletions(-) -- 2.17.1

3 12

[PATCH 0/2] ALSA: usb-audio: scarlett2: Read all configuration at init time
by Geoffrey D. Bennett 07 Jun '21

07 Jun '21

These two patches add support for reading the mixer volumes and mux configuration from the hardware when the driver is initialising. Previously the ALSA volume controls were initialised to zero and the mux configuration set to a fixed default instead of being initialised to match the hardware state. The ALSA controls for the Scarlett Gen 2 interfaces should now always be in sync with the hardware. Thanks to Vladimir Sadovnikov for figuring out how to do this. Takashi, if these pass your review, I believe that they are appropriate for: #Cc: stable(a)vger.kernel.org Geoffrey D. Bennett (2): ALSA: usb-audio: scarlett2: Read mixer volumes at init time ALSA: usb-audio: scarlett2: Read mux at init time sound/usb/mixer_scarlett_gen2.c | 227 ++++++++++++++++++++++---------- 1 file changed, 161 insertions(+), 66 deletions(-) -- 2.31.1

4 7

[PATCH][V2][next] ASoC: rk817: remove redundant assignment to pointer node, add missing of_node_put
by Colin King 07 Jun '21

07 Jun '21

From: Colin Ian King <colin.king(a)canonical.com> The pointer node is being initialized with a value that is never read and it is being updated later with a new value. The initialization is redundant and can be removed. The function is missing a of_node_put on node, fix this by adding the call before returning. Addresses-Coverity: ("Unused value") Fixes: 0d6a04da9b25 ("ASoC: Add Rockchip rk817 audio CODEC support") Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- V2: Add missing of_node_put call, kudos to Dan Carpenter for spotting this issue. --- sound/soc/codecs/rk817_codec.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rk817_codec.c b/sound/soc/codecs/rk817_codec.c index fd3a5ba034a9..a44d3cad1119 100644 --- a/sound/soc/codecs/rk817_codec.c +++ b/sound/soc/codecs/rk817_codec.c @@ -456,7 +456,7 @@ static const struct snd_soc_component_driver soc_codec_dev_rk817 = { static void rk817_codec_parse_dt_property(struct device *dev, struct rk817_codec_priv *rk817) { - struct device_node *node = dev->parent->of_node; + struct device_node *node; node = of_get_child_by_name(dev->parent->of_node, "codec"); if (!node) { @@ -466,6 +466,8 @@ static void rk817_codec_parse_dt_property(struct device *dev, rk817->mic_in_differential = of_property_read_bool(node, "rockchip,mic-in-differential"); + + of_node_put(node); } static int rk817_platform_probe(struct platform_device *pdev) -- 2.31.1

3 2

[v7] ASoC: qcom: lpass-cpu: Fix pop noise during audio capture begin
by Judy Hsiao 07 Jun '21

07 Jun '21

From: Srinivasa Rao Mandadapu <srivasam(a)codeaurora.org> This patch fixes PoP noise of around 15ms observed during audio capture begin. Enables BCLK and LRCLK in snd_soc_dai_ops prepare call for introducing some delay before capture start. Co-developed-by: Judy Hsiao <judyhsiao(a)chromium.org> Signed-off-by: Judy Hsiao <judyhsiao(a)chromium.org> Signed-off-by: Srinivasa Rao Mandadapu <srivasam(a)codeaurora.org> (am from https://patchwork.kernel.org/patch/12276369/) (also found at https://lore.kernel.org/r/20210524142114.18676-1-srivasam@codeaurora.org) --- Changes Since V6: -- Removed clk_disable OSR clock in lpass_cpu_daiops_prepare error case as failure of prepare will result in calling shutdown which should take care of this. Changes Since V5: -- Fixed nit. -- Updated `mi2s_was_prepared[dai->driver->id] = true;` after checking for errors. Changes Since V4: -- Replaced the __clk_is_enabled(BCLK) check by the self maintained. mi2s_was_prepared bool state. -- Removed unrelated changes. -- Refined comments. Changes Since V3: -- Checked BCLK is off before enabling it in lpass_cpu_daiops_prepare as lpass_cpu_daiops_prepare can be called multiple times -- Checked BCLK is on before disabling it in lpass_cpu_daiops_shutdown to fix the WARN. It is because BCLK may not be enabled if lpass_cpu_daiops_prepare is not called before lpass_cpu_daiops_shutdown -- Added more comments Changes Since V2: -- Updated comments as per linux style -- Removed unrelated changes Changes Since V1: -- Enabled BCLK and LRCLK in dai ops prepare API instead of startup API -- Added comments sound/soc/qcom/lpass-cpu.c | 79 ++++++++++++++++++++++++++++++++++++++ sound/soc/qcom/lpass.h | 4 ++ 2 files changed, 83 insertions(+) diff --git a/sound/soc/qcom/lpass-cpu.c b/sound/soc/qcom/lpass-cpu.c index af8cb64924a0..647423a6cb57 100644 --- a/sound/soc/qcom/lpass-cpu.c +++ b/sound/soc/qcom/lpass-cpu.c @@ -93,8 +93,30 @@ static void lpass_cpu_daiops_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct lpass_data *drvdata = snd_soc_dai_get_drvdata(dai); + struct lpaif_i2sctl *i2sctl = drvdata->i2sctl; + unsigned int id = dai->driver->id; clk_disable_unprepare(drvdata->mi2s_osr_clk[dai->driver->id]); + /* + * Ensure LRCLK is disabled even in device node validation. + * Will not impact if disabled in lpass_cpu_daiops_trigger() + * suspend. + */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + regmap_fields_write(i2sctl->spken, id, LPAIF_I2SCTL_SPKEN_DISABLE); + else + regmap_fields_write(i2sctl->micen, id, LPAIF_I2SCTL_MICEN_DISABLE); + + /* + * BCLK may not be enabled if lpass_cpu_daiops_prepare is called before + * lpass_cpu_daiops_shutdown. It's paired with the clk_enable in + * lpass_cpu_daiops_prepare. + */ + if (drvdata->mi2s_was_prepared[dai->driver->id]) { + drvdata->mi2s_was_prepared[dai->driver->id] = false; + clk_disable(drvdata->mi2s_bit_clk[dai->driver->id]); + } + clk_unprepare(drvdata->mi2s_bit_clk[dai->driver->id]); } @@ -275,6 +297,18 @@ static int lpass_cpu_daiops_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + /* + * Ensure lpass BCLK/LRCLK is enabled during + * device resume as lpass_cpu_daiops_prepare() is not called + * after the device resumes. We don't check mi2s_was_prepared before + * enable/disable BCLK in trigger events because: + * 1. These trigger events are paired, so the BCLK + * enable_count is balanced. + * 2. the BCLK can be shared (ex: headset and headset mic), + * we need to increase the enable_count so that we don't + * turn off the shared BCLK while other devices are using + * it. + */ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { ret = regmap_fields_write(i2sctl->spken, id, LPAIF_I2SCTL_SPKEN_ENABLE); @@ -296,6 +330,10 @@ static int lpass_cpu_daiops_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + /* + * To ensure lpass BCLK/LRCLK is disabled during + * device suspend. + */ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { ret = regmap_fields_write(i2sctl->spken, id, LPAIF_I2SCTL_SPKEN_DISABLE); @@ -315,12 +353,53 @@ static int lpass_cpu_daiops_trigger(struct snd_pcm_substream *substream, return ret; } +static int lpass_cpu_daiops_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct lpass_data *drvdata = snd_soc_dai_get_drvdata(dai); + struct lpaif_i2sctl *i2sctl = drvdata->i2sctl; + unsigned int id = dai->driver->id; + int ret; + + /* + * Ensure lpass BCLK/LRCLK is enabled bit before playback/capture + * data flow starts. This allows other codec to have some delay before + * the data flow. + * (ex: to drop start up pop noise before capture starts). + */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + ret = regmap_fields_write(i2sctl->spken, id, LPAIF_I2SCTL_SPKEN_ENABLE); + else + ret = regmap_fields_write(i2sctl->micen, id, LPAIF_I2SCTL_MICEN_ENABLE); + + if (ret) { + dev_err(dai->dev, "error writing to i2sctl reg: %d\n", ret); + return ret; + } + + /* + * Check mi2s_was_prepared before enabling BCLK as lpass_cpu_daiops_prepare can + * be called multiple times. It's paired with the clk_disable in + * lpass_cpu_daiops_shutdown. + */ + if (!drvdata->mi2s_was_prepared[dai->driver->id]) { + ret = clk_enable(drvdata->mi2s_bit_clk[id]); + if (ret) { + dev_err(dai->dev, "error in enabling mi2s bit clk: %d\n", ret); + return ret; + } + drvdata->mi2s_was_prepared[dai->driver->id] = true; + } + return 0; +} + const struct snd_soc_dai_ops asoc_qcom_lpass_cpu_dai_ops = { .set_sysclk = lpass_cpu_daiops_set_sysclk, .startup = lpass_cpu_daiops_startup, .shutdown = lpass_cpu_daiops_shutdown, .hw_params = lpass_cpu_daiops_hw_params, .trigger = lpass_cpu_daiops_trigger, + .prepare = lpass_cpu_daiops_prepare, }; EXPORT_SYMBOL_GPL(asoc_qcom_lpass_cpu_dai_ops); diff --git a/sound/soc/qcom/lpass.h b/sound/soc/qcom/lpass.h index 83b2e08ade06..7f72214404ba 100644 --- a/sound/soc/qcom/lpass.h +++ b/sound/soc/qcom/lpass.h @@ -67,6 +67,10 @@ struct lpass_data { /* MI2S SD lines to use for playback/capture */ unsigned int mi2s_playback_sd_mode[LPASS_MAX_MI2S_PORTS]; unsigned int mi2s_capture_sd_mode[LPASS_MAX_MI2S_PORTS]; + + /* The state of MI2S prepare dai_ops was called */ + bool mi2s_was_prepared[LPASS_MAX_MI2S_PORTS]; + int hdmi_port_enable; /* low-power audio interface (LPAIF) registers */ -- 2.32.0.rc1.229.g3e70b5a671-goog

3 2

[PATCH v3 0/7] ASoC: adds new .auto_selectable_formats support
by Kuninori Morimoto 07 Jun '21

07 Jun '21

Hi Mark These are v3 of "ASoC: adds new .get_fmt support", but renamed Subject. This is a little bit challenging patch-set. The idea/code is almost same as v1 / v2. v3 has "priority" support. We need to set dai_link->dai_fmt to select CPU/Codec settings, and it is selected by Sound Card Driver, today. Because of it, Sound Card user need to know both CPU / Codec available dai_fmt, and needs to select it. For example simple-card / audio-graph case, it is selected by "format" and "bitclock/frame-master/inversion" on DT. But, it can be automatically selected if both CPU and Codec drivers indicate it to ALSA SoC Framework, somehow. By this patch, dai_fmt can be automatically selected from each driver if both CPU / Codec driver had .auto_selectable_formats. Automatically selectable *field* is depends on each drivers. For example, some driver want to select format "automatically", but want to select other fields "manually", because of complex limitation. Or other example, in case of both CPU and Codec are possible to be clock provider, but the quality was different. In these case, user need/want to *manually* select each fields from Sound Card driver. It uses Sound Card specified fields preferentially, and try to select non-specific fields from CPU and Codec driver settings if driver had .auto_selectable_formats. In other words, we can select all dai_fmt via Sound Card driver same as before. Select dai_fmt 100% automatically is very difficult and will be very complex, but select automatically some fields only is very easy, I guess. This patch-set is based on such assumption. v1 -> v2 - Add more detail explanation on git-log, code, comment. - Possible to be Clock/Frame provider is depends on driver's situation. v2 -> v3 - has priority - tidyup function explanation for snd_soc_dai_get_fmt() - Each driver don't try to have SND_SOC_DAIFMT_CBx_CFx to avoid confusion Link: https://lore.kernel.org/r/871rb3hypy.wl-kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/871racbx0w.wl-kuninori.morimoto.gx@renesas.com Kuninori Morimoto (7): ASoC: soc-core: move snd_soc_runtime_set_dai_fmt() to upside ASoC: soc-core: add snd_soc_runtime_get_dai_fmt() ASoC: ak4613: add .auto_selectable_formats support ASoC: pcm3168a: add .auto_selectable_formats support ASoC: rsnd: add .auto_selectable_formats support ASoC: fsi: add .auto_selectable_formats support ASoC: hdmi-codec: add .auto_selectable_formats support include/sound/soc-dai.h | 55 +++++++ sound/soc/codecs/ak4613.c | 11 ++ sound/soc/codecs/hdmi-codec.c | 21 +++ sound/soc/codecs/pcm3168a.c | 26 +++ sound/soc/sh/fsi.c | 15 ++ sound/soc/sh/rcar/core.c | 31 +++- sound/soc/soc-core.c | 288 ++++++++++++++++++++++++++-------- sound/soc/soc-dai.c | 63 ++++++++ sound/soc/soc-utils.c | 29 ++++ 9 files changed, 475 insertions(+), 64 deletions(-) -- 2.25.1

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[PATCH] ASoC: rk817: Remove unneeded semicolon
by Jiapeng Chong 07 Jun '21

07 Jun '21

Fix the following coccicheck warnings: ./sound/soc/codecs/rk817_codec.c:49:2-3: Unneeded semicolon. Reported-by: Abaci Robot <abaci(a)linux.alibaba.com> Signed-off-by: Jiapeng Chong <jiapeng.chong(a)linux.alibaba.com> --- sound/soc/codecs/rk817_codec.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rk817_codec.c b/sound/soc/codecs/rk817_codec.c index 17e672b..e95fe75 100644 --- a/sound/soc/codecs/rk817_codec.c +++ b/sound/soc/codecs/rk817_codec.c @@ -46,7 +46,8 @@ static int rk817_init(struct snd_soc_component *component) if (rk817->mic_in_differential) { snd_soc_component_update_bits(component, RK817_CODEC_AMIC_CFG0, MIC_DIFF_MASK, MIC_DIFF_EN); - }; + } + return 0; } -- 1.8.3.1

2 2

Re: [PATCH 0/2] ALSA: usb-audio: scarlett2: Read all configuration at init time
by Takashi Iwai 07 Jun '21

07 Jun '21

On Mon, 07 Jun 2021 17:00:10 +0200, Vladimir Sadovnikov wrote: > > Hello! > > I would like to say some words from my side. > > The Scarlett device (especially 18i20) is pretty complicated device > and holds a lot of settings in it's internal > configuration area (hardware and software). > > So this is not the only patch which will configure the driver in proper way. > Since the device stores it's internal state (and that's good for power > safety and mobility), ideally, we should get > the almost fully compatible mixer settings with the original Focusrite > Control Software. > > The huge amount of job I've already done i my fork of Geoffrey's driver: > https://github.com/sadko4u/focusrite-scarlett-backports/blob/master/prod-dr… > > So we're planning to work on integrating our changes into the common > patch sets and will submit changes here. Sure, I don't mean against the patches, this looks like an acceptable approach. So don't worry, I'd take the patches once when the fixed version is submitted. However, from the system design POV, all those configurations should be a software issue, and ideally we shouldn't rely on the hardware preset state which has been done *somehow* -- it may allow malfunction easily. One thing I've learned over years is that you can never trust hardware :) Takashi > > Best, > Vladimir > > 07.06.2021 10:23, Takashi Iwai пишет: > > On Sun, 06 Jun 2021 16:16:44 +0200, > > Geoffrey D. Bennett wrote: > >> These two patches add support for reading the mixer volumes and mux > >> configuration from the hardware when the driver is initialising. > >> > >> Previously the ALSA volume controls were initialised to zero and the > >> mux configuration set to a fixed default instead of being initialised > >> to match the hardware state. > >> > >> The ALSA controls for the Scarlett Gen 2 interfaces should now always > >> be in sync with the hardware. Thanks to Vladimir Sadovnikov for > >> figuring out how to do this. > >> > >> Takashi, if these pass your review, I believe that they are > >> appropriate for: > >> #Cc: stable(a)vger.kernel.org > > Well, in general, having a proper fixed value for the initial mixer > > value is the right thing, which is a part of the driver's role. > > Though, in snd-usb-audio, we don't set up the initial values just > > because of laziness; since the topology in USB audio is variable per > > device and often hard to parse correctly, it's difficult to determine > > the suitable initial values, hence we leave untouched. So, in that > > sense, setting the zero isn't wrong, rather safer, per se. > > > > However, Scarlett 2 seems to want to be different; it has already some > > initialization code to read the existing configs. So this change > > sounds more or less acceptable. But it's questionable whether it's > > really for stable as a "fix". > > > > In anyway, please fix the bug ktest bot spotted, the missing endian > > conversions and resubmit. > > > > > > thanks, > > > > Takashi > >

2 1

alsa-lib 1.2.5(-2 on arch) only one of two connected hdmi/displayport outputs visible at once. My mic is completely MIA
by GitHub issues - opened 07 Jun '21

07 Jun '21

alsa-project/alsa-lib issue #148 was opened from fkroener: Since the recent update to 1.2.5-2 on archlinux, only one of my two usb-c/displayport/hdmi devices show up in pavucontrol and kde mixer. I am able to switch between hdmi0 and hdmi1, by changing the Profile in pavucontrol of HD-Audio Generic. The microphone doesn't show up at all now. ![KDE Mixer](https://user-images.githubusercontent.com/28690007/121057421-8c36938… This doesn't appear related to #142 or #143, since I have revision 2 from arch installed, which should include the fixes. This is a Lenovo T14 AMD. ``` pactl list cards Card #0 Name: alsa_card.pci-0000_07_00.1 Driver: module-alsa-card.c Owner Module: 6 Properties: alsa.card = "0" alsa.card_name = "HD-Audio Generic" alsa.long_card_name = "HD-Audio Generic at 0xfd3c8000 irq 91" alsa.driver_name = "snd_hda_intel" device.bus_path = "pci-0000:07:00.1" sysfs.path = "/devices/pci0000:00/0000:00:08.1/0000:07:00.1/sound/card0" device.bus = "pci" device.vendor.id = "1002" device.vendor.name = "Advanced Micro Devices, Inc. [AMD/ATI]" device.product.id = "1637" device.string = "0" device.description = "HD-Audio Generic" module-udev-detect.discovered = "1" device.icon_name = "audio-card-pci" Profiles: output:hdmi-stereo: Digital Stereo (HDMI) Output (sinks: 1, sources: 0, priority: 5900, available: yes) output:hdmi-stereo-extra1: Digital Stereo (HDMI 2) Output (sinks: 1, sources: 0, priority: 5700, available: yes) output:hdmi-stereo-extra2: Digital Stereo (HDMI 3) Output (sinks: 1, sources: 0, priority: 5700, available: no) output:hdmi-surround-extra2: Digital Surround 5.1 (HDMI 3) Output (sinks: 1, sources: 0, priority: 600, available: no) output:hdmi-surround71-extra2: Digital Surround 7.1 (HDMI 3) Output (sinks: 1, sources: 0, priority: 600, available: no) off: Off (sinks: 0, sources: 0, priority: 0, available: yes) Active Profile: output:hdmi-stereo Ports: hdmi-output-0: HDMI / DisplayPort (type: HDMI, priority: 5900, latency offset: 0 usec, available) Properties: device.icon_name = "video-display" device.product.name = "P27h-20" Part of profile(s): output:hdmi-stereo hdmi-output-1: HDMI / DisplayPort 2 (type: HDMI, priority: 5800, latency offset: 0 usec, available) Properties: device.icon_name = "video-display" device.product.name = "P27h-20" Part of profile(s): output:hdmi-stereo-extra1 hdmi-output-2: HDMI / DisplayPort 3 (type: HDMI, priority: 5700, latency offset: 0 usec, not available) Properties: device.icon_name = "video-display" Part of profile(s): output:hdmi-stereo-extra2, output:hdmi-surround-extra2, output:hdmi-surround71-extra2 Card #1 Name: alsa_card.pci-0000_07_00.6 Driver: module-alsa-card.c Owner Module: 8 Properties: alsa.card = "1" alsa.card_name = "HD-Audio Generic" alsa.long_card_name = "HD-Audio Generic at 0xfd3c0000 irq 92" alsa.driver_name = "snd_hda_intel" device.bus_path = "pci-0000:07:00.6" sysfs.path = "/devices/pci0000:00/0000:00:08.1/0000:07:00.6/sound/card1" device.bus = "pci" device.vendor.id = "1022" device.vendor.name = "Advanced Micro Devices, Inc. [AMD]" device.product.id = "15e3" device.product.name = "Family 17h (Models 10h-1fh) HD Audio Controller" device.string = "1" device.description = "Family 17h (Models 10h-1fh) HD Audio Controller" module-udev-detect.discovered = "1" device.icon_name = "audio-card-pci" Profiles: input:analog-stereo: Analog Stereo Input (sinks: 0, sources: 1, priority: 65, available: no) output:analog-stereo: Analog Stereo Output (sinks: 1, sources: 0, priority: 6500, available: yes) output:analog-stereo+input:analog-stereo: Analog Stereo Duplex (sinks: 1, sources: 1, priority: 6565, available: no) off: Off (sinks: 0, sources: 0, priority: 0, available: yes) Active Profile: output:analog-stereo Ports: analog-input-mic: Microphone (type: Mic, priority: 8700, latency offset: 0 usec, not available) Properties: device.icon_name = "audio-input-microphone" Part of profile(s): input:analog-stereo, output:analog-stereo+input:analog-stereo analog-output-speaker: Speakers (type: Speaker, priority: 10000, latency offset: 0 usec, availability unknown) Properties: device.icon_name = "audio-speakers" Part of profile(s): output:analog-stereo, output:analog-stereo+input:analog-stereo analog-output-headphones: Headphones (type: Headphones, priority: 9900, latency offset: 0 usec, not available) Properties: device.icon_name = "audio-headphones" Part of profile(s): output:analog-stereo, output:analog-stereo+input:analog-stereo ``` If you need any more information, I'm gladly at your disposal. Thanks! Issue URL : https://github.com/alsa-project/alsa-lib/issues/148 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH AUTOSEL 5.12 27/49] ALSA: hda: Add AlderLake-M PCI ID
by Sasha Levin 07 Jun '21

07 Jun '21

From: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> [ Upstream commit 4ad7935df6a566225c3d51900bde8f2f0f8b6de3 ] Add HD Audio PCI ID for Intel AlderLake-M. Add rules to snd_intel_dsp_find_config() to choose SOF driver for ADL-M systems with PCH-DMIC or Soundwire codecs, and legacy driver for the rest. Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Péter Ujfalusi <peter.ujfalusi(a)intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20210528185123.48332-1-kai.vehmanen@linux.intel.c… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/hda/intel-dsp-config.c | 4 ++++ sound/pci/hda/hda_intel.c | 3 +++ 2 files changed, 7 insertions(+) diff --git a/sound/hda/intel-dsp-config.c b/sound/hda/intel-dsp-config.c index ab5ff7867eb9..d8be146793ee 100644 --- a/sound/hda/intel-dsp-config.c +++ b/sound/hda/intel-dsp-config.c @@ -331,6 +331,10 @@ static const struct config_entry config_table[] = { .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC_OR_SOUNDWIRE, .device = 0x51c8, }, + { + .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC_OR_SOUNDWIRE, + .device = 0x51cc, + }, #endif }; diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 79ade335c8a0..470753b36c8a 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2485,6 +2485,9 @@ static const struct pci_device_id azx_ids[] = { /* Alderlake-P */ { PCI_DEVICE(0x8086, 0x51c8), .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, + /* Alderlake-M */ + { PCI_DEVICE(0x8086, 0x51cc), + .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, /* Elkhart Lake */ { PCI_DEVICE(0x8086, 0x4b55), .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, -- 2.30.2

1 0

[PATCH 0/9] ALSA: firewire: cease from delayed card registration
by Takashi Sakamoto 07 Jun '21

07 Jun '21

Hi, Since Linux kernel v4.7, delayed card registration was introduced to all drivers in ALSA firewire stack. Nowadays it brings less benefit than code complication. This patchset ceases from it. Takashi Sakamoto (9): ALSA: bebob: cease from delayed card registration ALSA: fireworks: cease from delayed card registration ALSA: oxfw: cease from delayed card registration ALSA: dice: cease from delayed card registration ALSA: firewire-digi00x: cease from delayed card registration ALSA: firewire-tascam: cease from delayed card registration ALSA: firewire-motu: cease from delayed card registration ALSA: fireface: cease from delayed card registration ALSA: firewire-lib: delete unused kernel API sound/firewire/bebob/bebob.c | 159 ++++++++++----------------- sound/firewire/bebob/bebob.h | 4 - sound/firewire/dice/dice.c | 138 ++++++++--------------- sound/firewire/dice/dice.h | 4 - sound/firewire/digi00x/digi00x.c | 101 +++++------------ sound/firewire/digi00x/digi00x.h | 3 - sound/firewire/fireface/ff.c | 90 +++++---------- sound/firewire/fireface/ff.h | 3 - sound/firewire/fireworks/fireworks.c | 107 ++++++------------ sound/firewire/fireworks/fireworks.h | 3 - sound/firewire/lib.c | 32 ------ sound/firewire/lib.h | 3 - sound/firewire/motu/motu.c | 84 +++++--------- sound/firewire/motu/motu.h | 3 - sound/firewire/oxfw/oxfw.c | 128 ++++++++------------- sound/firewire/oxfw/oxfw.h | 6 +- sound/firewire/tascam/tascam.c | 92 +++++----------- sound/firewire/tascam/tascam.h | 2 - 18 files changed, 296 insertions(+), 666 deletions(-) -- 2.27.0

2 10

[PATCH v2] ASoC: qcom: Fix for DMA interrupt clear reg overwriting
by Srinivasa Rao Mandadapu 07 Jun '21

07 Jun '21

The DMA interrupt clear register overwritten during simultaneous playback and capture in lpass platform interrupt handler. It's causing playback or capture stuck in similtaneous plaback on speaker and capture on dmic test. Update appropriate reg fields of corresponding channel instead of entire register write. Fixes: commit c5c8635a04711 ("ASoC: qcom: Add LPASS platform driver") Signed-off-by: Srinivasa Rao Mandadapu <srivasam(a)codeaurora.org> --- sound/soc/qcom/lpass-platform.c | 17 +++++++++++------ 1 file changed, 11 insertions(+), 6 deletions(-) diff --git a/sound/soc/qcom/lpass-platform.c b/sound/soc/qcom/lpass-platform.c index 0df9481ea4c6..f220a2739ac3 100644 --- a/sound/soc/qcom/lpass-platform.c +++ b/sound/soc/qcom/lpass-platform.c @@ -526,7 +526,7 @@ static int lpass_platform_pcmops_trigger(struct snd_soc_component *component, return -EINVAL; } - ret = regmap_write(map, reg_irqclr, val_irqclr); + ret = regmap_update_bits(map, reg_irqclr, val_irqclr, val_irqclr); if (ret) { dev_err(soc_runtime->dev, "error writing to irqclear reg: %d\n", ret); return ret; @@ -650,7 +650,7 @@ static irqreturn_t lpass_dma_interrupt_handler( struct lpass_variant *v = drvdata->variant; irqreturn_t ret = IRQ_NONE; int rv; - unsigned int reg = 0, val = 0; + unsigned int reg, val, val_clr, val_mask; struct regmap *map; unsigned int dai_id = cpu_dai->driver->id; @@ -676,8 +676,9 @@ static irqreturn_t lpass_dma_interrupt_handler( return -EINVAL; } if (interrupts & LPAIF_IRQ_PER(chan)) { - - rv = regmap_write(map, reg, LPAIF_IRQ_PER(chan) | val); + val_clr = LPAIF_IRQ_PER(chan) | val; + val_mask = LPAIF_IRQ_ALL(chan); + rv = regmap_update_bits(map, reg, val_mask, val_clr); if (rv) { dev_err(soc_runtime->dev, "error writing to irqclear reg: %d\n", rv); @@ -688,7 +689,9 @@ static irqreturn_t lpass_dma_interrupt_handler( } if (interrupts & LPAIF_IRQ_XRUN(chan)) { - rv = regmap_write(map, reg, LPAIF_IRQ_XRUN(chan) | val); + val_clr = (LPAIF_IRQ_XRUN(chan) | val); + val_mask = LPAIF_IRQ_ALL(chan); + rv = regmap_update_bits(map, reg, val_mask, val_clr); if (rv) { dev_err(soc_runtime->dev, "error writing to irqclear reg: %d\n", rv); @@ -700,7 +703,9 @@ static irqreturn_t lpass_dma_interrupt_handler( } if (interrupts & LPAIF_IRQ_ERR(chan)) { - rv = regmap_write(map, reg, LPAIF_IRQ_ERR(chan) | val); + val_clr = (LPAIF_IRQ_ERR(chan) | val); + val_mask = LPAIF_IRQ_ALL(chan); + rv = regmap_update_bits(map, reg, val_mask, val_clr); if (rv) { dev_err(soc_runtime->dev, "error writing to irqclear reg: %d\n", rv); -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

2 2

[PATCH] ASoC: Remove unneeded semicolon in rk817_codec.c
by Wan Jiabing 07 Jun '21

07 Jun '21

Fix following coccicheck warning: ./sound/soc/codecs/rk817_codec.c:49:2-3: Unneeded semicolon Signed-off-by: Wan Jiabing <wanjiabing(a)vivo.com> --- sound/soc/codecs/rk817_codec.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rk817_codec.c b/sound/soc/codecs/rk817_codec.c index fd3a5ba034a9..414d21f22a05 100644 --- a/sound/soc/codecs/rk817_codec.c +++ b/sound/soc/codecs/rk817_codec.c @@ -46,7 +46,7 @@ static int rk817_init(struct snd_soc_component *component) if (rk817->mic_in_differential) { snd_soc_component_update_bits(component, RK817_CODEC_AMIC_CFG0, MIC_DIFF_MASK, MIC_DIFF_EN); - }; + } return 0; } -- 2.20.1

3 3

Possible ALSA bug: Cannot record via mic while PA is running
by Meghdad Moradi 07 Jun '21

07 Jun '21

Hello, Please take a look at the following thread that I created some time ago: [1]https://ubuntuforums.org/showthread.php?t=2462882&p=14041574#post140 41574 I have included most relevant and useful info in the first and last post. pa-info output: [2]https://gist.github.com/no149/5c239626048a860a882f8d412013a4e6 I'd provide further info, if it's needed. References 1. https://ubuntuforums.org/showthread.php?t=2462882&p=14041574#post14041574 2. https://gist.github.com/no149/5c239626048a860a882f8d412013a4e6

1 0

[RFC][PATCH 0/3] ALSA: pcm/firewire: allow to queue period elapse event in process context
by Takashi Sakamoto 07 Jun '21

07 Jun '21

Hi, All of drivers in ALSA firewire stack processes two chances to process isochronous packets in any isochronous context; in software IRQ context for 1394 OHCI, and in process context of ALSA PCM application. In the process context, callbacks of .pointer and .ack are utilized. The callbacks are done by ALSA PCM core under acquiring lock of PCM substream, In design of ALSA PCM core, call of snd_pcm_period_elapsed() is used for drivers to awaken user processes from waiting for available frames. The function voluntarily acquires lock of PCM substream, therefore it is not called in the process context since it causes dead lock. As a workaround to avoid the dead lock, all of drivers in ALSA firewire stack uses workqueue to delegate the call. This patchset is my attempt for the issue. A variant of 'snd_pcm_period_elapsed()' without lock acquisition is going to be added, named 'snd_pcm_period_elapsed_without_lock()'. This is used in callbacks of .pointer and .ack of snd_pcm_ops structure. The patchset is still under my test, but it looks to work well in my easy and rough test. Before posting for merge, I'd like to get your comment to the idea. When evaluating, please merge below two histories: * 64584f329352 (for-next) * 9981b20a5e36 (for-linus) Takashi Sakamoto (3): ALSA: pcm: add snd_pcm_period_elapsed() variant without acquiring lock of PCM substream ALSA: firewire-lib: queue event of period elapse in process context ALSA: firewire-lib: obsolete workqueue for period update include/sound/pcm.h | 53 ++++++++++++++++++++++++++++++++++- sound/core/pcm_lib.c | 25 +++-------------- sound/firewire/amdtp-stream.c | 46 +++++++++--------------------- sound/firewire/amdtp-stream.h | 1 - 4 files changed, 70 insertions(+), 55 deletions(-) -- 2.27.0

2 5

New lavc api
by GitHub pull_request - opened 06 Jun '21

06 Jun '21

alsa-project/alsa-plugins pull request #23 was opened from jamrial: This is untested (I don't have access to a system with alsa, but the change was straightforward enough that i could write it). It is unlikely that it will fix the silence issue described in #22, but it nonetheless is a change required in order to support future versions of ffmpeg. The silence is, as far as i could tell, related to the fact that starting with ffmpeg 4.4, the ac3_fixed encoder takes S32 planar as input, versus the S16 planar it used to. This plugin should look at the AVCodec sample_fmts[0] value and feed the encoder the appropriate data. Request URL : https://github.com/alsa-project/alsa-plugins/pull/23 Patch URL : https://github.com/alsa-project/alsa-plugins/pull/23.patch Repository URL: https://github.com/alsa-project/alsa-plugins

1 0

[PATCH 0/8] ALSA: usb-audio: scarlett2: cleanup
by Geoffrey D. Bennett 06 Jun '21

06 Jun '21

Hi Takashi, Over the next few weeks I will have ready about 40 patches to clean up the Scarlett Gen 2 mixer driver a bit, add a few missing features, and add support for all Scarlett Gen 3 models. I was planning on submitting them in batches of 5-10 at a time; is that okay? Here are the first 8 patches which are ready to go; only small cleanups/renames/fixes. They are intended to be applied after these two patches that I submitted earlier: - ALSA: usb-audio: scarlett2: Read mixer volumes at init time - ALSA: usb-audio: scarlett2: Read mux at init time Thanks, Geoffrey. Geoffrey D. Bennett (8): ALSA: usb-audio: scarlett2: Remove incorrect S/PDIF comment ALSA: usb-audio: scarlett2: Coding style improvements ALSA: usb-audio: scarlett2: Remove unused/useless code ALSA: usb-audio: scarlett2: Remove interrupt debug message ALSA: usb-audio: scarlett2: Rename buttons/interrupts ALSA: usb-audio: scarlett2: Rename struct scarlett2_mixer_data ALSA: usb-audio: scarlett2: Add temp variable for consistency ALSA: usb-audio: scarlett2: Fix data_mutex lock sound/usb/mixer_scarlett_gen2.c | 229 ++++++++++++++------------------ 1 file changed, 102 insertions(+), 127 deletions(-) -- 2.31.1

1 8

a52: not compatible with recent avcodec
by GitHub issues - opened 06 Jun '21

06 Jun '21

alsa-project/alsa-plugins issue #22 was opened from manio: Hi! Current a52 plugin is not compatible with recent changes in the FFmpeg lib. The symptom is that using newest FFmpeg the a52 plugin produces silent in the speakers. During the weekend I was able to bisect the problem and shed some light on this, and here is what I found out: 1. The a52 plugin is working fine before commit: https://github.com/FFmpeg/FFmpeg/commit/93016f5d1d280f9cb7856883af287fa66af… 2. Above commit is wrapping avcodec_encode_audio2(), which a52 plugin is using, into new API internally inside the avcodec 3. All applications should move to new so called ["avcodec_send_frame() API"](https://ffmpeg.org/doxygen/3.4/group__lavc__encdec.html) 4. After the commit in point 1 the a52 output plugin is not working anymore I was asking the commit author @jamrial about the commit and he said that the main culprit could rather be the commit: https://github.com/FFmpeg/FFmpeg/commit/2d85e6e7230406e33e61d903b9c29ab562e… which is changing the S16 to S32 for ac3_fixed encoder while the a52 plugin _"is ignoring the kind of format the encoder takes as input (Starting from the aforementioned commit, ac3_fixed takes S32 planar audio), and unconditionally feeding it S16 planar audio."_ I was trying to change the a52 from S16 to S32 but I think some more work is needed on this... I was also try to change the do_encode() function from avcodec_encode_audio2 to avcodec_send_frame() with no luck... Issue URL : https://github.com/alsa-project/alsa-plugins/issues/22 Repository URL: https://github.com/alsa-project/alsa-plugins

1 0

[PATCH] ALSA: firewire-motu: add support for hybrid model of MOTU Ultralite mk3
by Takashi Sakamoto 06 Jun '21

06 Jun '21

This commit adds support for the hybrid model of MOTU Ultralite mk3 with alpha connector, which is already discontinued. The hardware specification of the model is the same as the one of FireWire-only model. $ cd linux-firewire-utils $ python3 src/crpp < /sys/bus/firewire/devices/fw1/config_rom ROM header and bus information block ----------------------------------------------------------------- 400 04101573 bus_info_length 4, crc_length 16, crc 5491 404 31333934 bus_name "1394" 408 20ff7000 irmc 0, cmc 0, isc 1, bmc 0, cyc_clk_acc 255, max_rec 7 (256) 40c 0001f200 company_id 0001f2 | 410 000a059c device_id 00000a059c | EUI-64 0001f200000a059c root directory ----------------------------------------------------------------- 414 0004ef04 directory_length 4, crc 61188 418 030001f2 vendor 41c 0c0083c0 node capabilities per IEEE 1394 420 d1000002 --> unit directory at 428 424 8d000005 --> eui-64 leaf at 438 unit directory at 428 ----------------------------------------------------------------- 428 0003f00b directory_length 3, crc 61451 42c 120001f2 specifier id 430 13000030 version 434 17103800 model eui-64 leaf at 438 ----------------------------------------------------------------- 438 0002d89c leaf_length 2, crc 55452 43c 0001f200 company_id 0001f2 | 440 000a059c device_id 00000a059c | EUI-64 0001f200000a059c Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/Kconfig | 2 ++ sound/firewire/motu/motu.c | 1 + 2 files changed, 3 insertions(+) diff --git a/sound/firewire/Kconfig b/sound/firewire/Kconfig index def1f3d5ecf5..fc6858131b59 100644 --- a/sound/firewire/Kconfig +++ b/sound/firewire/Kconfig @@ -154,6 +154,8 @@ config SND_FIREWIRE_MOTU * 8pre * 828mk3 (FireWire only) * 828mk3 (Hybrid) + * Ultralite mk3 (FireWire only) + * Ultralite mk3 (Hybrid) * Audio Express * 4pre diff --git a/sound/firewire/motu/motu.c b/sound/firewire/motu/motu.c index a4929c1302dc..2a8a6ea2d3f1 100644 --- a/sound/firewire/motu/motu.c +++ b/sound/firewire/motu/motu.c @@ -190,6 +190,7 @@ static const struct ieee1394_device_id motu_id_table[] = { SND_MOTU_DEV_ENTRY(0x00000f, &snd_motu_spec_8pre), SND_MOTU_DEV_ENTRY(0x000015, &snd_motu_spec_828mk3), // FireWire only. SND_MOTU_DEV_ENTRY(0x000019, &snd_motu_spec_ultralite_mk3), // FireWire only. + SND_MOTU_DEV_ENTRY(0x000030, &snd_motu_spec_ultralite_mk3), // Hybrid. SND_MOTU_DEV_ENTRY(0x000035, &snd_motu_spec_828mk3), // Hybrid. SND_MOTU_DEV_ENTRY(0x000033, &snd_motu_spec_audio_express), SND_MOTU_DEV_ENTRY(0x000045, &snd_motu_spec_4pre), -- 2.27.0

2 1

[PATCH] ALSA: firewire-lib: remove useless operations for kernel preemption
by Takashi Sakamoto 06 Jun '21

06 Jun '21

In all of drivers of ALSA firewire stack, the callback of .pointer and .ack in snd_pcm_ops structure is done in acquired spin_lock of PCM substream, therefore already under disabled kernel preemption. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/amdtp-stream.c | 14 ++------------ 1 file changed, 2 insertions(+), 12 deletions(-) diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c index 945597ffacc2..1c566ddf0776 100644 --- a/sound/firewire/amdtp-stream.c +++ b/sound/firewire/amdtp-stream.c @@ -1751,13 +1751,8 @@ unsigned long amdtp_domain_stream_pcm_pointer(struct amdtp_domain *d, // Later, the process context will sometimes schedules software // IRQ context of the period_work. Then, no need to flush the // queue by the same reason as described in the above - if (current_work() != &s->period_work) { - // Queued packet should be processed without any kernel - // preemption to keep latency against bus cycle. - preempt_disable(); + if (current_work() != &s->period_work) fw_iso_context_flush_completions(irq_target->context); - preempt_enable(); - } } return READ_ONCE(s->pcm_buffer_pointer); @@ -1777,13 +1772,8 @@ int amdtp_domain_stream_pcm_ack(struct amdtp_domain *d, struct amdtp_stream *s) // Process isochronous packets for recent isochronous cycle to handle // queued PCM frames. - if (irq_target && amdtp_stream_running(irq_target)) { - // Queued packet should be processed without any kernel - // preemption to keep latency against bus cycle. - preempt_disable(); + if (irq_target && amdtp_stream_running(irq_target)) fw_iso_context_flush_completions(irq_target->context); - preempt_enable(); - } return 0; } -- 2.27.0

2 1

[PATCH] ALSA: firewire-lib: fix error codes for allocation failure
by Dan Carpenter 06 Jun '21

06 Jun '21

Return -ENOMEM if kcalloc() fails. Currently the code returns success. Fixes: f9e5ecdfc2c2 ("ALSA: firewire-lib: add replay target to cache sequence of packet") Fixes: 6f24bb8a157c ("ALSA: firewire-lib: pool sequence of packet in IT context independently") Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- sound/firewire/amdtp-stream.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c index 945597ffacc2..b37cec3cc579 100644 --- a/sound/firewire/amdtp-stream.c +++ b/sound/firewire/amdtp-stream.c @@ -1625,8 +1625,10 @@ static int amdtp_stream_start(struct amdtp_stream *s, int channel, int speed, s->ctx_data.tx.cache.tail = 0; s->ctx_data.tx.cache.descs = kcalloc(s->ctx_data.tx.cache.size, sizeof(*s->ctx_data.tx.cache.descs), GFP_KERNEL); - if (!s->ctx_data.tx.cache.descs) + if (!s->ctx_data.tx.cache.descs) { + err = -ENOMEM; goto err_context; + } } } else { static const struct { @@ -1643,8 +1645,10 @@ static int amdtp_stream_start(struct amdtp_stream *s, int channel, int speed, }; s->ctx_data.rx.seq.descs = kcalloc(queue_size, sizeof(*s->ctx_data.rx.seq.descs), GFP_KERNEL); - if (!s->ctx_data.rx.seq.descs) + if (!s->ctx_data.rx.seq.descs) { + err = -ENOMEM; goto err_context; + } s->ctx_data.rx.seq.size = queue_size; s->ctx_data.rx.seq.tail = 0; s->ctx_data.rx.seq.head = 0; -- 2.30.2

3 2

amixer and alsamixer unaware of controls for ThinkPad T14's 'acp' card
by GitHub issues - opened 06 Jun '21

06 Jun '21

alsa-project/alsa-lib issue #146 was opened from jwatt: I have the AMD Ryzen 7 Pro 4750U version of the [ThinkPad T14 gen 1 type 20UD](https://pcsupport.lenovo.com/gb/en/products/laptops-and-netbooks/thin… ([tech specs](https://www.lenovo.com/gb/en/laptops/thinkpad/t-series/ThinkPad-T14-…) with Fedora 34 installed. I am unable to control the built-in microphone using `amixer`/`alsamixer`. ``` $ cat /proc/asound/modules 0 snd_hda_intel 1 snd_hda_intel 2 snd_acp3x_rn ``` As best I can tell (I could be mistaken!), 0 is the HDMI port, 1 is the build-in speakers and headphones jack, and 2 is the Ryzen "Audio Co-Processor" (i.e. "acp" / "AMD TrueAudio") which is (or includes) the built-in microphone. To be specific, I only get microphone input if I select 'acp' as the input. For the 'acp' card, AlsaMixer v1.2.4 reports "This sound device does not have any controls." It appears support for the Ryzen 4000 mobile series Audio Co-Processor was added in kernel 5.8 (see the section "Add Renoir ACP driver" on the [5.8 release notes](https://kernelnewbies.org/Linux_5.8#Audio)). If the kernel supports it, it's unclear to me why the ALSA tools wouldn't allow me to control it. Is this currently expected? I'm happy to provide any other information that would be helpful - just let me know what you need. Issue URL : https://github.com/alsa-project/alsa-lib/issues/146 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

alsa-ucm-conf 1.2.5 broke front headphone output on pipewire-pulse
by GitHub issues - opened 06 Jun '21

06 Jun '21

alsa-project/alsa-ucm-conf issue #98 was opened from amezin: After upgrade to 1.2.5, front headphone output isn't working. In GNOME Settings, when I select front headphone output and click "Test", it shows a dialog with text "Click a speaker to test", but there are no buttons to click. `systemctl --user status pipewire` shows this: Jun 06 04:40:30 trx40 systemd[2732]: Started Multimedia Service. Jun 06 04:40:41 trx40 pipewire[44766]: '_ucm0005.hw:ALC1220VBDT,2': playback open failed: No such device Jun 06 04:40:41 trx40 pipewire[44766]: adapter 0x559741478670: can't get format: No such device Jun 06 04:40:41 trx40 pipewire[44766]: usage: node.name=<string> Jun 06 04:40:41 trx40 pipewire[44766]: '_ucm0005.hw:ALC1220VBDT,1': playback open failed: No such device Jun 06 04:40:41 trx40 pipewire[44766]: adapter 0x559741530560: can't get format: No such device Jun 06 04:40:41 trx40 pipewire[44766]: usage: node.name=<string> Downgrading alsa-ucm-conf to 1.2.4 fixes the issue Tried pipewire 0.3.29 and pipewire git master Though with pulseaudio 14.2 the output works Motherboard: ASUS ROG Strix TRX40-E With alsa-ucm-conf 1.2.4 - output working: alsa-info.sh output: [alsa-info-good.txt](https://github.com/alsa-project/alsa-ucm-conf/files/660… pw-dump output: [pw-dump-good.txt](https://github.com/alsa-project/alsa-ucm-conf/files/66033… With alsa-ucm-conf 1.2.5 - output not working: alsa-info.sh output: [alsa-info-bad.txt](https://github.com/alsa-project/alsa-ucm-conf/files/6603… pw-dump output: [pw-dump-bad.txt](https://github.com/alsa-project/alsa-ucm-conf/files/660336… Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/98 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH] ALSA: firewire-lib: fix the context to call snd_pcm_stop_xrun()
by Takashi Sakamoto 05 Jun '21

05 Jun '21

In the workqueue to queue wake-up event, isochronous context is not processed, thus it's useless to check context for the workqueue to switch status of runtime for PCM substream to XRUN. On the other hand, in software IRQ context of 1394 OHCI, it's needed. This commit fixes the bug introduced when tasklet was replaced with workqueue. Cc: <stable(a)vger.kernel.org> Fixes: 2b3d2987d800 ("ALSA: firewire: Replace tasklet with work") Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/amdtp-stream.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c index 945597ffacc2..19c343c53585 100644 --- a/sound/firewire/amdtp-stream.c +++ b/sound/firewire/amdtp-stream.c @@ -1032,7 +1032,7 @@ static void generate_pkt_descs(struct amdtp_stream *s, const __be32 *ctx_header, static inline void cancel_stream(struct amdtp_stream *s) { s->packet_index = -1; - if (current_work() == &s->period_work) + if (in_interrupt()) amdtp_stream_pcm_abort(s); WRITE_ONCE(s->pcm_buffer_pointer, SNDRV_PCM_POS_XRUN); } -- 2.27.0

2 1

[PATCH 1/3] ALSA: hda/realtek: fix mute/micmute LEDs and speaker for HP Elite Dragonfly G2
by Jeremy Szu 05 Jun '21

05 Jun '21

The HP Elite Dragonfly G2 using ALC285 codec which using 0x04 to control mute LED and 0x01 to control micmute LED. In the other hand, there is no output from right channel of speaker. Therefore, add a quirk to make it works. Signed-off-by: Jeremy Szu <jeremy.szu(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 43e37145eb5d..9f65171a902d 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8310,6 +8310,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x8519, "HP Spectre x360 15-df0xxx", ALC285_FIXUP_HP_SPECTRE_X360), SND_PCI_QUIRK(0x103c, 0x869d, "HP", ALC236_FIXUP_HP_MUTE_LED), SND_PCI_QUIRK(0x103c, 0x86c7, "HP Envy AiO 32", ALC274_FIXUP_HP_ENVY_GPIO), + SND_PCI_QUIRK(0x103c, 0x8716, "HP Elite Dragonfly G2 Notebook PC", ALC285_FIXUP_HP_GPIO_AMP_INIT), SND_PCI_QUIRK(0x103c, 0x8724, "HP EliteBook 850 G7", ALC285_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8729, "HP", ALC285_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8730, "HP ProBook 445 G7", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), -- 2.31.1

2 5

[PATCH] ALSA: hda/realtek: Change device names for quirks to barebone names
by Werner Sembach 05 Jun '21

05 Jun '21

Change the name string of several devices needing quirks to the Clevo-barebone ones. Also make the names follow the same pattern for multiple Clevo names referring to the same mainboard. Signed-off-by: Werner Sembach <wse(a)tuxedocomputers.com> --- sound/pci/hda/patch_realtek.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 552e2cb73291..39cf3667118c 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8368,7 +8368,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { 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, 0x1323, "Clevo N130ZU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), - SND_PCI_QUIRK(0x1558, 0x1325, "System76 Darter Pro (darp5)", 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), @@ -8403,11 +8403,11 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1558, 0x8521, "Clevo NH77D[CD]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x8535, "Clevo NH50D[BE]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x8536, "Clevo NH79D[BE]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), - SND_PCI_QUIRK(0x1558, 0x8550, "System76 Gazelle (gaze14)", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), - SND_PCI_QUIRK(0x1558, 0x8551, "System76 Gazelle (gaze14)", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), - SND_PCI_QUIRK(0x1558, 0x8560, "System76 Gazelle (gaze14)", ALC269_FIXUP_HEADSET_MIC), - SND_PCI_QUIRK(0x1558, 0x8561, "System76 Gazelle (gaze14)", ALC269_FIXUP_HEADSET_MIC), - SND_PCI_QUIRK(0x1558, 0x8562, "Clevo NH[5|7][0-9]RZ[Q]", ALC269_FIXUP_DMIC), + SND_PCI_QUIRK(0x1558, 0x8550, "Clevo NH[57][0-9][ER][ACDH]Q", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1558, 0x8551, "Clevo NH[57][0-9][ER][ACDH]Q", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1558, 0x8560, "Clevo NH[57][0-9][ER][ACDH]Q", ALC269_FIXUP_HEADSET_MIC), + SND_PCI_QUIRK(0x1558, 0x8561, "Clevo NH[57][0-9][ER][ACDH]Q", ALC269_FIXUP_HEADSET_MIC), + SND_PCI_QUIRK(0x1558, 0x8562, "Clevo NH[57][0-9]RZ[Q]", ALC269_FIXUP_DMIC), SND_PCI_QUIRK(0x1558, 0x8668, "Clevo NP50B[BE]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x8680, "Clevo NJ50LU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x8686, "Clevo NH50[CZ]U", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), @@ -8418,8 +8418,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1558, 0x9600, "Clevo N960K[PR]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x961d, "Clevo N960S[CDF]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x971d, "Clevo N970T[CDF]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), - SND_PCI_QUIRK(0x1558, 0xa500, "Clevo NL53RU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), - SND_PCI_QUIRK(0x1558, 0xa600, "Clevo NL5XNU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1558, 0xa500, "Clevo NL5[03]RU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1558, 0xa600, "Clevo NL50NU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0xb018, "Clevo NP50D[BE]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0xb019, "Clevo NH77D[BE]Q", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0xb022, "Clevo NH77D[DC][QW]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), -- 2.25.1

2 1

[PATCH] ASoC: qcom: Fix for DMA interrupt clear reg overwriting
by Srinivasa Rao Mandadapu 05 Jun '21

05 Jun '21

This patch fixes the DMA interrupt registers overwriting issue in lpass platform interrupt handler. Signed-off-by: Srinivasa Rao Mandadapu <srivasam(a)codeaurora.org> --- sound/soc/qcom/lpass-platform.c | 17 +++++++++++------ 1 file changed, 11 insertions(+), 6 deletions(-) diff --git a/sound/soc/qcom/lpass-platform.c b/sound/soc/qcom/lpass-platform.c index 0df9481ea4c6..e02caa121fa4 100644 --- a/sound/soc/qcom/lpass-platform.c +++ b/sound/soc/qcom/lpass-platform.c @@ -526,7 +526,7 @@ static int lpass_platform_pcmops_trigger(struct snd_soc_component *component, return -EINVAL; } - ret = regmap_write(map, reg_irqclr, val_irqclr); + ret = regmap_update_bits(map, reg_irqclr, val_irqclr, val_irqclr); if (ret) { dev_err(soc_runtime->dev, "error writing to irqclear reg: %d\n", ret); return ret; @@ -650,7 +650,7 @@ static irqreturn_t lpass_dma_interrupt_handler( struct lpass_variant *v = drvdata->variant; irqreturn_t ret = IRQ_NONE; int rv; - unsigned int reg = 0, val = 0; + unsigned int reg = 0, val = 0, val_clr = 0, val_mask = 0; struct regmap *map; unsigned int dai_id = cpu_dai->driver->id; @@ -676,8 +676,9 @@ static irqreturn_t lpass_dma_interrupt_handler( return -EINVAL; } if (interrupts & LPAIF_IRQ_PER(chan)) { - - rv = regmap_write(map, reg, LPAIF_IRQ_PER(chan) | val); + val_clr = (LPAIF_IRQ_PER(chan) | val); + val_mask = LPAIF_IRQ_ALL(chan); + rv = regmap_update_bits(map, reg, val_mask, val_clr); if (rv) { dev_err(soc_runtime->dev, "error writing to irqclear reg: %d\n", rv); @@ -688,7 +689,9 @@ static irqreturn_t lpass_dma_interrupt_handler( } if (interrupts & LPAIF_IRQ_XRUN(chan)) { - rv = regmap_write(map, reg, LPAIF_IRQ_XRUN(chan) | val); + val_clr = (LPAIF_IRQ_XRUN(chan) | val); + val_mask = LPAIF_IRQ_ALL(chan); + rv = regmap_update_bits(map, reg, val_mask, val_clr); if (rv) { dev_err(soc_runtime->dev, "error writing to irqclear reg: %d\n", rv); @@ -700,7 +703,9 @@ static irqreturn_t lpass_dma_interrupt_handler( } if (interrupts & LPAIF_IRQ_ERR(chan)) { - rv = regmap_write(map, reg, LPAIF_IRQ_ERR(chan) | val); + val_clr = (LPAIF_IRQ_ERR(chan) | val); + val_mask = LPAIF_IRQ_ALL(chan); + rv = regmap_update_bits(map, reg, val_mask, val_clr); if (rv) { dev_err(soc_runtime->dev, "error writing to irqclear reg: %d\n", rv); -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

2 2

[PATCH] ASoC: rt5682: Fix the fast discharge for headset unplugging in soundwire mode
by Oder Chiou 04 Jun '21

04 Jun '21

Based on ("5a15cd7fce20b1fd4aece6a0240e2b58cd6a225d"), the setting also should be set in soundwire mode. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5682-sdw.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5682-sdw.c b/sound/soc/codecs/rt5682-sdw.c index fed80c8f994f..e78ba3b064c4 100644 --- a/sound/soc/codecs/rt5682-sdw.c +++ b/sound/soc/codecs/rt5682-sdw.c @@ -462,7 +462,8 @@ static int rt5682_io_init(struct device *dev, struct sdw_slave *slave) regmap_update_bits(rt5682->regmap, RT5682_CBJ_CTRL_2, RT5682_EXT_JD_SRC, RT5682_EXT_JD_SRC_MANUAL); - regmap_write(rt5682->regmap, RT5682_CBJ_CTRL_1, 0xd042); + regmap_write(rt5682->regmap, RT5682_CBJ_CTRL_1, 0xd142); + regmap_update_bits(rt5682->regmap, RT5682_CBJ_CTRL_5, 0x0700, 0x0600); regmap_update_bits(rt5682->regmap, RT5682_CBJ_CTRL_3, RT5682_CBJ_IN_BUF_EN, RT5682_CBJ_IN_BUF_EN); regmap_update_bits(rt5682->regmap, RT5682_SAR_IL_CMD_1, -- 2.31.1

2 2

[PATCH v2][next] ASoC: ti: davinci-mcasp: Fix fall-through warning for Clang
by Gustavo A. R. Silva 04 Jun '21

04 Jun '21

In preparation to enable -Wimplicit-fallthrough for Clang, fix a warning by explicitly adding a fallthrough; statement. Link: https://github.com/KSPP/linux/issues/115 Signed-off-by: Gustavo A. R. Silva <gustavoars(a)kernel.org> --- Changes in v2: - Use fallthrough; instead of break; - Update changelog text. Link: https://lore.kernel.org/linux-hardening/25941c37-6e38-88ae-3afe-4f5af44380d… sound/soc/ti/davinci-mcasp.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/ti/davinci-mcasp.c b/sound/soc/ti/davinci-mcasp.c index b94220306d1a..017a5a5e56cd 100644 --- a/sound/soc/ti/davinci-mcasp.c +++ b/sound/soc/ti/davinci-mcasp.c @@ -2317,6 +2317,7 @@ static int davinci_mcasp_probe(struct platform_device *pdev) break; default: dev_err(&pdev->dev, "No DMA controller found (%d)\n", ret); + fallthrough; case -EPROBE_DEFER: goto err; } -- 2.27.0

4 3

[PATCH] ASoC: snd-soc-lpass requires REGMAP_MMIO
by Bertrand Jacquin 04 Jun '21

04 Jun '21

With CONFIG_SND_SOC_LPASS_RX_MACRO=m and CONFIG_REGMAP_MMIO undefined, build fails with the following error make -f /var/tmp/portage/sys-kernel/stable-sources-5.12.8/work/linux-5.12.8-stable/scripts/Makefile.modpost sed 's/\.ko$/\.o/' modules.order | scripts/mod/modpost -E -o modules-only.symvers -i vmlinux.symvers -T - ERROR: modpost: "__devm_regmap_init_mmio_clk" [sound/soc/codecs/snd-soc-lpass-rx-macro.ko] undefined! This does also apply to other Qualcomm Macro LPASS all making call to devm_regmap_init_mmio() Signed-off-by: Bertrand Jacquin <bertrand(a)jacquin.bzh> --- sound/soc/codecs/Kconfig | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 1c87b42606c9..f5614f47f44d 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -1824,18 +1824,22 @@ config SND_SOC_TPA6130A2 config SND_SOC_LPASS_WSA_MACRO depends on COMMON_CLK + select REGMAP_MMIO tristate "Qualcomm WSA Macro in LPASS(Low Power Audio SubSystem)" config SND_SOC_LPASS_VA_MACRO depends on COMMON_CLK + select REGMAP_MMIO tristate "Qualcomm VA Macro in LPASS(Low Power Audio SubSystem)" config SND_SOC_LPASS_RX_MACRO depends on COMMON_CLK + select REGMAP_MMIO tristate "Qualcomm RX Macro in LPASS(Low Power Audio SubSystem)" config SND_SOC_LPASS_TX_MACRO depends on COMMON_CLK + select REGMAP_MMIO tristate "Qualcomm TX Macro in LPASS(Low Power Audio SubSystem)" endmenu

2 1

[PATCH -next] ASoC: rk817: fix compile warning in rk817_probe()
by Yang Yingliang 04 Jun '21

04 Jun '21

Fix the compile warning by return ret: sound/soc/codecs/rk817_codec.c:418:6: warning: variable ‘ret’ set but not used [-Wunused-but-set-variable] 418 | int ret; | ^~~ Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: Yang Yingliang <yangyingliang(a)huawei.com> --- sound/soc/codecs/rk817_codec.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rk817_codec.c b/sound/soc/codecs/rk817_codec.c index 17e672b85ee5..56de5ec5f03b 100644 --- a/sound/soc/codecs/rk817_codec.c +++ b/sound/soc/codecs/rk817_codec.c @@ -430,7 +430,7 @@ static int rk817_probe(struct snd_soc_component *component) snd_soc_component_set_pll(component, 0, 0, 0, 0); - return 0; + return ret; } static void rk817_remove(struct snd_soc_component *component) -- 2.25.1

2 1

[v6] ASoC: qcom: lpass-cpu: Fix pop noise during audio capture begin
by Judy Hsiao 04 Jun '21

04 Jun '21

From: Srinivasa Rao Mandadapu <srivasam(a)codeaurora.org> This patch fixes PoP noise of around 15ms observed during audio capture begin. Enables BCLK and LRCLK in snd_soc_dai_ops prepare call for introducing some delay before capture start. Co-developed-by: Judy Hsiao <judyhsiao(a)chromium.org> Signed-off-by: Judy Hsiao <judyhsiao(a)chromium.org> Signed-off-by: Srinivasa Rao Mandadapu <srivasam(a)codeaurora.org> (am from https://patchwork.kernel.org/patch/12276369/) (also found at https://lore.kernel.org/r/20210524142114.18676-1-srivasam@codeaurora.org) --- Changes Since V5: -- Fixed nit. -- Updated `mi2s_was_prepared[dai->driver->id] = true;` after checking for errors. Changes Since V4: -- Replaced the __clk_is_enabled(BCLK) check by the self maintained. mi2s_was_prepared bool state. -- Removed unrelated changes. -- Refined comments. Changes Since V3: -- Checked BCLK is off before enabling it in lpass_cpu_daiops_prepare as lpass_cpu_daiops_prepare can be called multiple times -- Checked BCLK is on before disabling it in lpass_cpu_daiops_shutdown to fix the WARN. It is because BCLK may not be enabled if lpass_cpu_daiops_prepare is not called before lpass_cpu_daiops_shutdown -- Added more comments Changes Since V2: -- Updated comments as per linux style -- Removed unrelated changes Changes Since V1: -- Enabled BCLK and LRCLK in dai ops prepare API instead of startup API -- Added comments sound/soc/qcom/lpass-cpu.c | 80 ++++++++++++++++++++++++++++++++++++++ sound/soc/qcom/lpass.h | 4 ++ 2 files changed, 84 insertions(+) diff --git a/sound/soc/qcom/lpass-cpu.c b/sound/soc/qcom/lpass-cpu.c index af8cb64924a0..0430cc797391 100644 --- a/sound/soc/qcom/lpass-cpu.c +++ b/sound/soc/qcom/lpass-cpu.c @@ -93,8 +93,30 @@ static void lpass_cpu_daiops_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct lpass_data *drvdata = snd_soc_dai_get_drvdata(dai); + struct lpaif_i2sctl *i2sctl = drvdata->i2sctl; + unsigned int id = dai->driver->id; clk_disable_unprepare(drvdata->mi2s_osr_clk[dai->driver->id]); + /* + * Ensure LRCLK is disabled even in device node validation. + * Will not impact if disabled in lpass_cpu_daiops_trigger() + * suspend. + */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + regmap_fields_write(i2sctl->spken, id, LPAIF_I2SCTL_SPKEN_DISABLE); + else + regmap_fields_write(i2sctl->micen, id, LPAIF_I2SCTL_MICEN_DISABLE); + + /* + * BCLK may not be enabled if lpass_cpu_daiops_prepare is called before + * lpass_cpu_daiops_shutdown. It's paired with the clk_enable in + * lpass_cpu_daiops_prepare. + */ + if (drvdata->mi2s_was_prepared[dai->driver->id]) { + drvdata->mi2s_was_prepared[dai->driver->id] = false; + clk_disable(drvdata->mi2s_bit_clk[dai->driver->id]); + } + clk_unprepare(drvdata->mi2s_bit_clk[dai->driver->id]); } @@ -275,6 +297,18 @@ static int lpass_cpu_daiops_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + /* + * Ensure lpass BCLK/LRCLK is enabled during + * device resume as lpass_cpu_daiops_prepare() is not called + * after the device resumes. We don't check mi2s_was_prepared before + * enable/disable BCLK in trigger events because: + * 1. These trigger events are paired, so the BCLK + * enable_count is balanced. + * 2. the BCLK can be shared (ex: headset and headset mic), + * we need to increase the enable_count so that we don't + * turn off the shared BCLK while other devices are using + * it. + */ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { ret = regmap_fields_write(i2sctl->spken, id, LPAIF_I2SCTL_SPKEN_ENABLE); @@ -296,6 +330,10 @@ static int lpass_cpu_daiops_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + /* + * To ensure lpass BCLK/LRCLK is disabled during + * device suspend. + */ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { ret = regmap_fields_write(i2sctl->spken, id, LPAIF_I2SCTL_SPKEN_DISABLE); @@ -315,12 +353,54 @@ static int lpass_cpu_daiops_trigger(struct snd_pcm_substream *substream, return ret; } +static int lpass_cpu_daiops_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct lpass_data *drvdata = snd_soc_dai_get_drvdata(dai); + struct lpaif_i2sctl *i2sctl = drvdata->i2sctl; + unsigned int id = dai->driver->id; + int ret; + + /* + * Ensure lpass BCLK/LRCLK is enabled bit before playback/capture + * data flow starts. This allows other codec to have some delay before + * the data flow. + * (ex: to drop start up pop noise before capture starts). + */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + ret = regmap_fields_write(i2sctl->spken, id, LPAIF_I2SCTL_SPKEN_ENABLE); + else + ret = regmap_fields_write(i2sctl->micen, id, LPAIF_I2SCTL_MICEN_ENABLE); + + if (ret) { + dev_err(dai->dev, "error writing to i2sctl reg: %d\n", ret); + return ret; + } + + /* + * Check mi2s_was_prepared before enabling BCLK as lpass_cpu_daiops_prepare can + * be called multiple times. It's paired with the clk_disable in + * lpass_cpu_daiops_shutdown. + */ + if (!drvdata->mi2s_was_prepared[dai->driver->id]) { + ret = clk_enable(drvdata->mi2s_bit_clk[id]); + if (ret) { + dev_err(dai->dev, "error in enabling mi2s bit clk: %d\n", ret); + clk_disable(drvdata->mi2s_osr_clk[id]); + return ret; + } + drvdata->mi2s_was_prepared[dai->driver->id] = true; + } + return 0; +} + const struct snd_soc_dai_ops asoc_qcom_lpass_cpu_dai_ops = { .set_sysclk = lpass_cpu_daiops_set_sysclk, .startup = lpass_cpu_daiops_startup, .shutdown = lpass_cpu_daiops_shutdown, .hw_params = lpass_cpu_daiops_hw_params, .trigger = lpass_cpu_daiops_trigger, + .prepare = lpass_cpu_daiops_prepare, }; EXPORT_SYMBOL_GPL(asoc_qcom_lpass_cpu_dai_ops); diff --git a/sound/soc/qcom/lpass.h b/sound/soc/qcom/lpass.h index 83b2e08ade06..7f72214404ba 100644 --- a/sound/soc/qcom/lpass.h +++ b/sound/soc/qcom/lpass.h @@ -67,6 +67,10 @@ struct lpass_data { /* MI2S SD lines to use for playback/capture */ unsigned int mi2s_playback_sd_mode[LPASS_MAX_MI2S_PORTS]; unsigned int mi2s_capture_sd_mode[LPASS_MAX_MI2S_PORTS]; + + /* The state of MI2S prepare dai_ops was called */ + bool mi2s_was_prepared[LPASS_MAX_MI2S_PORTS]; + int hdmi_port_enable; /* low-power audio interface (LPAIF) registers */ -- 2.32.0.rc1.229.g3e70b5a671-goog

2 1

hda: improve the microphone device allocation
by GitHub pull_request - opened 04 Jun '21

04 Jun '21

alsa-project/alsa-ucm-conf pull request #97 was opened from perexg: This change use the local variables for the UCM microphone device names. Also, handle the no-dmic case for sof-hda-dsp hardware. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/97 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/97.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH][next] ASoC: ti: davinci-mcasp: Fix fall-through warning for Clang
by Gustavo A. R. Silva 04 Jun '21

04 Jun '21

In preparation to enable -Wimplicit-fallthrough for Clang, fix a warning by explicitly adding a goto statement instead of letting the code fall through to the next case. Link: https://github.com/KSPP/linux/issues/115 Signed-off-by: Gustavo A. R. Silva <gustavoars(a)kernel.org> --- JFYI: We had thousands of these sorts of warnings and now we are down to just 25 in linux-next. This is one of those last remaining warnings. sound/soc/ti/davinci-mcasp.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/ti/davinci-mcasp.c b/sound/soc/ti/davinci-mcasp.c index b94220306d1a..587967720135 100644 --- a/sound/soc/ti/davinci-mcasp.c +++ b/sound/soc/ti/davinci-mcasp.c @@ -2317,6 +2317,7 @@ static int davinci_mcasp_probe(struct platform_device *pdev) break; default: dev_err(&pdev->dev, "No DMA controller found (%d)\n", ret); + goto err; case -EPROBE_DEFER: goto err; } -- 2.27.0

4 3

[PATCH -next] ALSA: firewire-motu: fix error return code in snd_motu_stream_reserve_duplex()
by Yang Yingliang 04 Jun '21

04 Jun '21

Fix to return a negative error code from the error handling case instead of 0, as done elsewhere in this function. Fixes: e50dfac81f73 ("ALSA: firewire-motu: cache event ticks in source packet header per data block") Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: Yang Yingliang <yangyingliang(a)huawei.com> --- sound/firewire/motu/motu-stream.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/firewire/motu/motu-stream.c b/sound/firewire/motu/motu-stream.c index 43ff5be32b15..9e6ca39ebd7f 100644 --- a/sound/firewire/motu/motu-stream.c +++ b/sound/firewire/motu/motu-stream.c @@ -191,7 +191,7 @@ int snd_motu_stream_reserve_duplex(struct snd_motu *motu, unsigned int rate, if (!motu->cache.event_offsets) { fw_iso_resources_free(&motu->tx_resources); fw_iso_resources_free(&motu->rx_resources); - return err; + return -ENOMEM; } } -- 2.25.1

3 2

[v5] ASoC: qcom: lpass-cpu: Fix pop noise during audio capture begin
by Judy Hsiao 04 Jun '21

04 Jun '21

From: Srinivasa Rao Mandadapu <srivasam(a)codeaurora.org> This patch fixes PoP noise of around 15ms observed during audio capture begin. Enables BCLK and LRCLK in snd_soc_dai_ops prepare call for introducing some delay before capture start. Co-developed-by: Judy Hsiao <judyhsiao(a)chromium.org> Signed-off-by: Judy Hsiao <judyhsiao(a)chromium.org> Signed-off-by: Srinivasa Rao Mandadapu <srivasam(a)codeaurora.org> (am from https://patchwork.kernel.org/patch/12276369/) (also found at https://lore.kernel.org/r/20210524142114.18676-1-srivasam@codeaurora.org) --- Changes Since V4: -- Replaced the __clk_is_enabled(BCLK) check by the self maintained mi2s_was_prepared bool state. -- Removed unrelated changes. -- Refined comments. Changes Since V3: -- Checked BCLK is off before enabling it in lpass_cpu_daiops_prepare as lpass_cpu_daiops_prepare can be called multiple times -- Checked BCLK is on before disabling it in lpass_cpu_daiops_shutdown to fix the WARN. It is because BCLK may not be enabled if lpass_cpu_daiops_prepare is not called before lpass_cpu_daiops_shutdown -- Added more comments Changes Since V2: -- Updated comments as per linux style -- Removed unrelated changes Changes Since V1: -- Enabled BCLK and LRCLK in dai ops prepare API instead of startup API -- Added comments sound/soc/qcom/lpass-cpu.c | 80 ++++++++++++++++++++++++++++++++++++++ sound/soc/qcom/lpass.h | 4 ++ 2 files changed, 84 insertions(+) diff --git a/sound/soc/qcom/lpass-cpu.c b/sound/soc/qcom/lpass-cpu.c index af8cb64924a0..3129e7cc25a4 100644 --- a/sound/soc/qcom/lpass-cpu.c +++ b/sound/soc/qcom/lpass-cpu.c @@ -93,8 +93,30 @@ static void lpass_cpu_daiops_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct lpass_data *drvdata = snd_soc_dai_get_drvdata(dai); + struct lpaif_i2sctl *i2sctl = drvdata->i2sctl; + unsigned int id = dai->driver->id; clk_disable_unprepare(drvdata->mi2s_osr_clk[dai->driver->id]); + /* + * Ensure LRCLK is disabled even in device node validation. + * Will not impact if disabled in lpass_cpu_daiops_trigger() + * suspend. + */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + regmap_fields_write(i2sctl->spken, id, LPAIF_I2SCTL_SPKEN_DISABLE); + else + regmap_fields_write(i2sctl->micen, id, LPAIF_I2SCTL_MICEN_DISABLE); + + /* + * BCLK may not be enabled if lpass_cpu_daiops_prepare is called before + * lpass_cpu_daiops_shutdown. It's paired with the clk_enable in + * lpass_cpu_daiops_prepare. + */ + if (drvdata->mi2s_was_prepared[dai->driver->id]) { + drvdata->mi2s_was_prepared[dai->driver->id] = false; + clk_disable(drvdata->mi2s_bit_clk[dai->driver->id]); + } + clk_unprepare(drvdata->mi2s_bit_clk[dai->driver->id]); } @@ -275,6 +297,18 @@ static int lpass_cpu_daiops_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + /* + * Ensure lpass BCLK/LRCLK is enabled during + * device resume as lpass_cpu_daiops_prepare() is not called + * after the device resumes. We don't check mi2s_was_prepared before + * enable/disable BCLK in trigger events because: + * 1. These trigger events are paired, so the BCLK + * enable_count is balanced. + * 2. the BCLK can be shared (ex: headset and headset mic), + * we need to increase the enable_count so that we don't + * turn off the shared BCLK while other devices are using + * it. + */ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { ret = regmap_fields_write(i2sctl->spken, id, LPAIF_I2SCTL_SPKEN_ENABLE); @@ -296,6 +330,10 @@ static int lpass_cpu_daiops_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + /* + * To ensure lpass BCLK/LRCLK is disabled during + * device suspend. + */ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { ret = regmap_fields_write(i2sctl->spken, id, LPAIF_I2SCTL_SPKEN_DISABLE); @@ -315,12 +353,54 @@ static int lpass_cpu_daiops_trigger(struct snd_pcm_substream *substream, return ret; } +static int lpass_cpu_daiops_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct lpass_data *drvdata = snd_soc_dai_get_drvdata(dai); + struct lpaif_i2sctl *i2sctl = drvdata->i2sctl; + unsigned int id = dai->driver->id; + int ret; + /* + * Ensure lpass BCLK/LRCLK is enabled bit before playback/capture + * data flow starts. This allows other codec to have some delay before + * the data flow. + * (ex: to drop start up pop noise before capture starts). + */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + ret = regmap_fields_write(i2sctl->spken, id, LPAIF_I2SCTL_SPKEN_ENABLE); + else + ret = regmap_fields_write(i2sctl->micen, id, LPAIF_I2SCTL_MICEN_ENABLE); + + if (ret) { + dev_err(dai->dev, "error writing to i2sctl reg: %d\n", ret); + return ret; + } + + /* + * Check mi2s_was_prepared before enabling BCLK as lpass_cpu_daiops_prepare can + * be called multiple times. It's paired with the clk_disable in + * lpass_cpu_daiops_shutdown. + */ + if (!drvdata->mi2s_was_prepared[dai->driver->id]) { + drvdata->mi2s_was_prepared[dai->driver->id] = true; + ret = clk_enable(drvdata->mi2s_bit_clk[id]); + if (ret) { + dev_err(dai->dev, "error in enabling mi2s bit clk: %d\n", ret); + clk_disable(drvdata->mi2s_osr_clk[id]); + drvdata->mi2s_was_prepared[dai->driver->id] = false; + return ret; + } + } + return 0; +} + const struct snd_soc_dai_ops asoc_qcom_lpass_cpu_dai_ops = { .set_sysclk = lpass_cpu_daiops_set_sysclk, .startup = lpass_cpu_daiops_startup, .shutdown = lpass_cpu_daiops_shutdown, .hw_params = lpass_cpu_daiops_hw_params, .trigger = lpass_cpu_daiops_trigger, + .prepare = lpass_cpu_daiops_prepare, }; EXPORT_SYMBOL_GPL(asoc_qcom_lpass_cpu_dai_ops); diff --git a/sound/soc/qcom/lpass.h b/sound/soc/qcom/lpass.h index 83b2e08ade06..7f72214404ba 100644 --- a/sound/soc/qcom/lpass.h +++ b/sound/soc/qcom/lpass.h @@ -67,6 +67,10 @@ struct lpass_data { /* MI2S SD lines to use for playback/capture */ unsigned int mi2s_playback_sd_mode[LPASS_MAX_MI2S_PORTS]; unsigned int mi2s_capture_sd_mode[LPASS_MAX_MI2S_PORTS]; + + /* The state of MI2S prepare dai_ops was called */ + bool mi2s_was_prepared[LPASS_MAX_MI2S_PORTS]; + int hdmi_port_enable; /* low-power audio interface (LPAIF) registers */ -- 2.31.0

3 2

[PATCH] ASoC: snd-soc-lpass requires REGMAP_MMIO
by Bertrand Jacquin 04 Jun '21

04 Jun '21

With CONFIG_SND_SOC_LPASS_RX_MACRO=m and CONFIG_REGMAP_MMIO undefined, build fails with the following error make -f /var/tmp/portage/sys-kernel/stable-sources-5.12.8/work/linux-5.12.8-stable/scripts/Makefile.modpost sed 's/\.ko$/\.o/' modules.order | scripts/mod/modpost -E -o modules-only.symvers -i vmlinux.symvers -T - ERROR: modpost: "__devm_regmap_init_mmio_clk" [sound/soc/codecs/snd-soc-lpass-rx-macro.ko] undefined! This does also apply to other Qualcomm Macro LPASS all making call to devm_regmap_init_mmio() --- sound/soc/codecs/Kconfig | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 1c87b42606c9..f5614f47f44d 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -1824,18 +1824,22 @@ config SND_SOC_TPA6130A2 config SND_SOC_LPASS_WSA_MACRO depends on COMMON_CLK + select REGMAP_MMIO tristate "Qualcomm WSA Macro in LPASS(Low Power Audio SubSystem)" config SND_SOC_LPASS_VA_MACRO depends on COMMON_CLK + select REGMAP_MMIO tristate "Qualcomm VA Macro in LPASS(Low Power Audio SubSystem)" config SND_SOC_LPASS_RX_MACRO depends on COMMON_CLK + select REGMAP_MMIO tristate "Qualcomm RX Macro in LPASS(Low Power Audio SubSystem)" config SND_SOC_LPASS_TX_MACRO depends on COMMON_CLK + select REGMAP_MMIO tristate "Qualcomm TX Macro in LPASS(Low Power Audio SubSystem)" endmenu

2 2

[PATCH] ASoC: img: Fix PM reference leak in img_i2s_in_probe()
by Yufen Yu 03 Jun '21

03 Jun '21

pm_runtime_get_sync will increment pm usage counter even it failed. Forgetting to putting operation will result in reference leak here. Fix it by replacing it with pm_runtime_resume_and_get to keep usage counter balanced. Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: Yufen Yu <yuyufen(a)huawei.com> --- sound/soc/img/img-i2s-in.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/img/img-i2s-in.c b/sound/soc/img/img-i2s-in.c index 0843235d73c9..fd3432a1d6ab 100644 --- a/sound/soc/img/img-i2s-in.c +++ b/sound/soc/img/img-i2s-in.c @@ -464,7 +464,7 @@ static int img_i2s_in_probe(struct platform_device *pdev) if (ret) goto err_pm_disable; } - ret = pm_runtime_get_sync(&pdev->dev); + ret = pm_runtime_resume_and_get(&pdev->dev); if (ret < 0) goto err_suspend; -- 2.25.4

2 1

[PATCH] ASoC: AMD Renoir: Remove fix for DMI entry on Lenovo 2020 platforms
by Mark Pearson 03 Jun '21

03 Jun '21

Unfortunately the previous patch to fix issues using the AMD ACP bridge has the side effect of breaking the dmic in other cases and needs to be reverted. Removing the changes while we revisit the fix and find something better. Apologies for the churn. Suggested-by: Gabriel Craciunescu <unix.or.die(a)gmail.com> Signed-off-by: Mark Pearson <markpearson(a)lenovo.com> --- sound/soc/amd/renoir/rn-pci-acp3x.c | 35 ----------------------------- 1 file changed, 35 deletions(-) diff --git a/sound/soc/amd/renoir/rn-pci-acp3x.c b/sound/soc/amd/renoir/rn-pci-acp3x.c index c9fb1c8fbf8c..19438da5dfa5 100644 --- a/sound/soc/amd/renoir/rn-pci-acp3x.c +++ b/sound/soc/amd/renoir/rn-pci-acp3x.c @@ -199,41 +199,6 @@ static const struct dmi_system_id rn_acp_quirk_table[] = { DMI_EXACT_MATCH(DMI_BOARD_NAME, "20NLCTO1WW"), } }, - { - /* Lenovo ThinkPad P14s Gen 1 (20Y1) */ - .matches = { - DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "LENOVO"), - DMI_MATCH(DMI_BOARD_NAME, "20Y1"), - } - }, - { - /* Lenovo ThinkPad T14s Gen1 */ - .matches = { - DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "LENOVO"), - DMI_MATCH(DMI_BOARD_NAME, "20UH"), - } - }, - { - /* Lenovo ThinkPad T14s Gen1 Campus */ - .matches = { - DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "LENOVO"), - DMI_MATCH(DMI_BOARD_NAME, "20UJ"), - } - }, - { - /* Lenovo ThinkPad T14 Gen 1*/ - .matches = { - DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "LENOVO"), - DMI_MATCH(DMI_BOARD_NAME, "20UD"), - } - }, - { - /* Lenovo ThinkPad X13 Gen 1*/ - .matches = { - DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "LENOVO"), - DMI_MATCH(DMI_BOARD_NAME, "20UF"), - } - }, {} }; -- 2.31.1

2 1

[PATCH v2] ASoC: dt-bindings: Convert imx-audmux binding to json schema
by Oleksij Rempel 03 Jun '21

03 Jun '21

Convert the imx-audmux binding to DT schema format using json-schema Signed-off-by: Oleksij Rempel <o.rempel(a)pengutronix.de> Reviewed-by: Rob Herring <robh(a)kernel.org> --- changes v2: - fix yamllint warnings .../devicetree/bindings/sound/imx-audmux.txt | 28 ----- .../devicetree/bindings/sound/imx-audmux.yaml | 119 ++++++++++++++++++ 2 files changed, 119 insertions(+), 28 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/imx-audmux.txt create mode 100644 Documentation/devicetree/bindings/sound/imx-audmux.yaml diff --git a/Documentation/devicetree/bindings/sound/imx-audmux.txt b/Documentation/devicetree/bindings/sound/imx-audmux.txt deleted file mode 100644 index 2db4dcbee1b9..000000000000 --- a/Documentation/devicetree/bindings/sound/imx-audmux.txt +++ /dev/null @@ -1,28 +0,0 @@ -Freescale Digital Audio Mux (AUDMUX) device - -Required properties: - - - compatible : "fsl,imx21-audmux" for AUDMUX version firstly used - on i.MX21, or "fsl,imx31-audmux" for the version - firstly used on i.MX31. - - - reg : Should contain AUDMUX registers location and length. - -An initial configuration can be setup using child nodes. - -Required properties of optional child nodes: - - - fsl,audmux-port : Integer of the audmux port that is configured by this - child node. - - - fsl,port-config : List of configuration options for the specific port. - For imx31-audmux and above, it is a list of tuples - <ptcr pdcr>. For imx21-audmux it is a list of pcr - values. - -Example: - -audmux@21d8000 { - compatible = "fsl,imx6q-audmux", "fsl,imx31-audmux"; - reg = <0x021d8000 0x4000>; -}; diff --git a/Documentation/devicetree/bindings/sound/imx-audmux.yaml b/Documentation/devicetree/bindings/sound/imx-audmux.yaml new file mode 100644 index 000000000000..dab45c310670 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/imx-audmux.yaml @@ -0,0 +1,119 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/imx-audmux.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Freescale Digital Audio Mux device + +maintainers: + - Oleksij Rempel <o.rempel(a)pengutronix.de> + +properties: + compatible: + oneOf: + - items: + - enum: + - fsl,imx27-audmux + - const: fsl,imx21-audmux + - items: + - enum: + - fsl,imx25-audmux + - fsl,imx35-audmux + - fsl,imx50-audmux + - fsl,imx51-audmux + - fsl,imx53-audmux + - fsl,imx6q-audmux + - fsl,imx6sl-audmux + - fsl,imx6sll-audmux + - fsl,imx6sx-audmux + - const: fsl,imx31-audmux + + reg: + maxItems: 1 + + clocks: + maxItems: 1 + + clock-names: + items: + - const: audmux + +patternProperties: + "^mux-[0-9a-z]*$": + type: object + properties: + fsl,audmux-port: + $ref: /schemas/types.yaml#/definitions/uint32 + description: | + Integer of the audmux port that is configured by this child node + + fsl,port-config: + $ref: /schemas/types.yaml#/definitions/uint32-array + description: | + List of configuration options for the specific port. + For imx31-audmux and above, it is a list of tuples ptcr pdcr. + For imx21-audmux it is a list of pcr values. + + required: + - fsl,audmux-port + - fsl,port-config + + additionalProperties: false + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + audmux@21d8000 { + compatible = "fsl,imx6q-audmux", "fsl,imx31-audmux"; + reg = <0x021d8000 0x4000>; + }; + - | + audmux@10016000 { + compatible = "fsl,imx27-audmux", "fsl,imx21-audmux"; + reg = <0x10016000 0x1000>; + clocks = <&clks 1>; + clock-names = "audmux"; + + mux-ssi0 { + fsl,audmux-port = <0>; + fsl,port-config = <0xcb205000>; + }; + + mux-pins4 { + fsl,audmux-port = <2>; + fsl,port-config = <0x00001000>; + }; + }; + - | + #include <dt-bindings/sound/fsl-imx-audmux.h> + audmux@21d8000 { + compatible = "fsl,imx6q-audmux", "fsl,imx31-audmux"; + reg = <0x021d8000 0x4000>; + pinctrl-names = "default"; + pinctrl-0 = <&pinctrl_audmux>; + + mux-ssi1 { + fsl,audmux-port = <0>; + fsl,port-config = < + IMX_AUDMUX_V2_PTCR_SYN 0 + IMX_AUDMUX_V2_PTCR_TFSEL(2) 0 + IMX_AUDMUX_V2_PTCR_TCSEL(2) 0 + IMX_AUDMUX_V2_PTCR_TFSDIR 0 + IMX_AUDMUX_V2_PTCR_TCLKDIR IMX_AUDMUX_V2_PDCR_RXDSEL(2) + >; + }; + + mux-pins3 { + fsl,audmux-port = <2>; + fsl,port-config = < + IMX_AUDMUX_V2_PTCR_SYN IMX_AUDMUX_V2_PDCR_RXDSEL(0) + 0 IMX_AUDMUX_V2_PDCR_TXRXEN + >; + }; + }; -- 2.29.2

2 1

[PATCH] ASoC: tas2562: Fix TDM_CFG0_SAMPRATE values
by Richard Weinberger 03 Jun '21

03 Jun '21

TAS2562_TDM_CFG0_SAMPRATE_MASK starts at bit 1, not 0. So all values need to be left shifted by 1. Signed-off-by: Richard Weinberger <richard(a)nod.at> --- sound/soc/codecs/tas2562.h | 14 +++++++------- 1 file changed, 7 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/tas2562.h b/sound/soc/codecs/tas2562.h index 81866aeb3fbf..55b2a1f52ca3 100644 --- a/sound/soc/codecs/tas2562.h +++ b/sound/soc/codecs/tas2562.h @@ -57,13 +57,13 @@ #define TAS2562_TDM_CFG0_RAMPRATE_MASK BIT(5) #define TAS2562_TDM_CFG0_RAMPRATE_44_1 BIT(5) #define TAS2562_TDM_CFG0_SAMPRATE_MASK GENMASK(3, 1) -#define TAS2562_TDM_CFG0_SAMPRATE_7305_8KHZ 0x0 -#define TAS2562_TDM_CFG0_SAMPRATE_14_7_16KHZ 0x1 -#define TAS2562_TDM_CFG0_SAMPRATE_22_05_24KHZ 0x2 -#define TAS2562_TDM_CFG0_SAMPRATE_29_4_32KHZ 0x3 -#define TAS2562_TDM_CFG0_SAMPRATE_44_1_48KHZ 0x4 -#define TAS2562_TDM_CFG0_SAMPRATE_88_2_96KHZ 0x5 -#define TAS2562_TDM_CFG0_SAMPRATE_176_4_192KHZ 0x6 +#define TAS2562_TDM_CFG0_SAMPRATE_7305_8KHZ (0x0 << 1) +#define TAS2562_TDM_CFG0_SAMPRATE_14_7_16KHZ (0x1 << 1) +#define TAS2562_TDM_CFG0_SAMPRATE_22_05_24KHZ (0x2 << 1) +#define TAS2562_TDM_CFG0_SAMPRATE_29_4_32KHZ (0x3 << 1) +#define TAS2562_TDM_CFG0_SAMPRATE_44_1_48KHZ (0x4 << 1) +#define TAS2562_TDM_CFG0_SAMPRATE_88_2_96KHZ (0x5 << 1) +#define TAS2562_TDM_CFG0_SAMPRATE_176_4_192KHZ (0x6 << 1) #define TAS2562_TDM_CFG2_RIGHT_JUSTIFY BIT(6) -- 2.26.2

2 1

[PATCH] ASoC: meson: gx-card: fix sound-dai dt schema
by Jerome Brunet 03 Jun '21

03 Jun '21

There is a fair amount of warnings when running 'make dtbs_check' with amlogic,gx-sound-card.yaml. Ex: arch/arm64/boot/dts/amlogic/meson-gxm-q200.dt.yaml: sound: dai-link-0:sound-dai:0:1: missing phandle tag in 0 arch/arm64/boot/dts/amlogic/meson-gxm-q200.dt.yaml: sound: dai-link-0:sound-dai:0:2: missing phandle tag in 0 arch/arm64/boot/dts/amlogic/meson-gxm-q200.dt.yaml: sound: dai-link-0:sound-dai:0: [66, 0, 0] is too long The reason is that the sound-dai phandle provided has cells, and in such case the schema should use 'phandle-array' instead of 'phandle'. Fixes: fd00366b8e41 ("ASoC: meson: gx: add sound card dt-binding documentation") Signed-off-by: Jerome Brunet <jbrunet(a)baylibre.com> --- .../devicetree/bindings/sound/amlogic,gx-sound-card.yaml | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/amlogic,gx-sound-card.yaml b/Documentation/devicetree/bindings/sound/amlogic,gx-sound-card.yaml index db61f0731a20..2e35aeaa8781 100644 --- a/Documentation/devicetree/bindings/sound/amlogic,gx-sound-card.yaml +++ b/Documentation/devicetree/bindings/sound/amlogic,gx-sound-card.yaml @@ -57,7 +57,7 @@ patternProperties: rate sound-dai: - $ref: /schemas/types.yaml#/definitions/phandle + $ref: /schemas/types.yaml#/definitions/phandle-array description: phandle of the CPU DAI patternProperties: @@ -71,7 +71,7 @@ patternProperties: properties: sound-dai: - $ref: /schemas/types.yaml#/definitions/phandle + $ref: /schemas/types.yaml#/definitions/phandle-array description: phandle of the codec DAI required: -- 2.32.0.rc0

2 1

[PATCH 0/5] ASoC: rsnd: tidyup adg and header
by Kuninori Morimoto 03 Jun '21

03 Jun '21

Hi Mark I noticed that adg can be more clean code. And rsnd.h header comment was not so good because patch has been randomly added. This patch tidyup these. Kuninori Morimoto (5): ASoC: rsnd: adg: supply __printf(x, y) formatting for dbg_msg() ASoC: rsnd: adg: tidyup rsnd_adg_get_clkin/out() parameter ASoC: rsnd: adg: use more simple method for null_clk ASoC: rsnd: adg: check return value for rsnd_adg_get_clkin/out() ASoC: rsnd: tidyup __rsnd_mod_xxx macro comments sound/soc/sh/rcar/adg.c | 139 +++++++++++++++++++++++++-------------- sound/soc/sh/rcar/rsnd.h | 21 ++---- 2 files changed, 97 insertions(+), 63 deletions(-) -- 2.25.1

2 6

[PATCH AUTOSEL 4.4 02/15] ASoC: sti-sas: add missing MODULE_DEVICE_TABLE
by Sasha Levin 03 Jun '21

03 Jun '21

From: Zou Wei <zou_wei(a)huawei.com> [ Upstream commit e072b2671606c77538d6a4dd5dda80b508cb4816 ] This patch adds missing MODULE_DEVICE_TABLE definition which generates correct modalias for automatic loading of this driver when it is built as an external module. Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: Zou Wei <zou_wei(a)huawei.com> Link: https://lore.kernel.org/r/1620789145-14936-1-git-send-email-zou_wei@huawei.… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/sti-sas.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/sti-sas.c b/sound/soc/codecs/sti-sas.c index 160d61a66204..71a1fde5a7ef 100644 --- a/sound/soc/codecs/sti-sas.c +++ b/sound/soc/codecs/sti-sas.c @@ -542,6 +542,7 @@ static const struct of_device_id sti_sas_dev_match[] = { }, {}, }; +MODULE_DEVICE_TABLE(of, sti_sas_dev_match); static int sti_sas_driver_probe(struct platform_device *pdev) { -- 2.30.2

1 0

[PATCH AUTOSEL 4.9 02/17] ASoC: sti-sas: add missing MODULE_DEVICE_TABLE
by Sasha Levin 03 Jun '21

03 Jun '21

From: Zou Wei <zou_wei(a)huawei.com> [ Upstream commit e072b2671606c77538d6a4dd5dda80b508cb4816 ] This patch adds missing MODULE_DEVICE_TABLE definition which generates correct modalias for automatic loading of this driver when it is built as an external module. Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: Zou Wei <zou_wei(a)huawei.com> Link: https://lore.kernel.org/r/1620789145-14936-1-git-send-email-zou_wei@huawei.… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/sti-sas.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/sti-sas.c b/sound/soc/codecs/sti-sas.c index d6e00c77edcd..7cf76661c3cc 100644 --- a/sound/soc/codecs/sti-sas.c +++ b/sound/soc/codecs/sti-sas.c @@ -542,6 +542,7 @@ static const struct of_device_id sti_sas_dev_match[] = { }, {}, }; +MODULE_DEVICE_TABLE(of, sti_sas_dev_match); static int sti_sas_driver_probe(struct platform_device *pdev) { -- 2.30.2

1 0

[PATCH AUTOSEL 4.14 03/18] ASoC: sti-sas: add missing MODULE_DEVICE_TABLE
by Sasha Levin 03 Jun '21

03 Jun '21

From: Zou Wei <zou_wei(a)huawei.com> [ Upstream commit e072b2671606c77538d6a4dd5dda80b508cb4816 ] This patch adds missing MODULE_DEVICE_TABLE definition which generates correct modalias for automatic loading of this driver when it is built as an external module. Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: Zou Wei <zou_wei(a)huawei.com> Link: https://lore.kernel.org/r/1620789145-14936-1-git-send-email-zou_wei@huawei.… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/sti-sas.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/sti-sas.c b/sound/soc/codecs/sti-sas.c index 62c618765224..730dd453a744 100644 --- a/sound/soc/codecs/sti-sas.c +++ b/sound/soc/codecs/sti-sas.c @@ -407,6 +407,7 @@ static const struct of_device_id sti_sas_dev_match[] = { }, {}, }; +MODULE_DEVICE_TABLE(of, sti_sas_dev_match); static int sti_sas_driver_probe(struct platform_device *pdev) { -- 2.30.2

1 0

[PATCH AUTOSEL 4.19 05/23] ASoC: sti-sas: add missing MODULE_DEVICE_TABLE
by Sasha Levin 03 Jun '21

03 Jun '21

From: Zou Wei <zou_wei(a)huawei.com> [ Upstream commit e072b2671606c77538d6a4dd5dda80b508cb4816 ] This patch adds missing MODULE_DEVICE_TABLE definition which generates correct modalias for automatic loading of this driver when it is built as an external module. Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: Zou Wei <zou_wei(a)huawei.com> Link: https://lore.kernel.org/r/1620789145-14936-1-git-send-email-zou_wei@huawei.… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/sti-sas.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/sti-sas.c b/sound/soc/codecs/sti-sas.c index 7316c80b8179..27196126f710 100644 --- a/sound/soc/codecs/sti-sas.c +++ b/sound/soc/codecs/sti-sas.c @@ -411,6 +411,7 @@ static const struct of_device_id sti_sas_dev_match[] = { }, {}, }; +MODULE_DEVICE_TABLE(of, sti_sas_dev_match); static int sti_sas_driver_probe(struct platform_device *pdev) { -- 2.30.2

1 0

[PATCH AUTOSEL 4.19 04/23] ASoC: Intel: bytcr_rt5640: Add quirk for the Lenovo Miix 3-830 tablet
by Sasha Levin 03 Jun '21

03 Jun '21

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit f0353e1f53f92f7b3da91e6669f5d58ee222ebe8 ] The Lenovo Miix 3-830 tablet has only 1 speaker, has an internal analog mic on IN1 and uses JD2 for jack-detect, add a quirk to automatically apply these settings on Lenovo Miix 3-830 tablets. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Link: https://lore.kernel.org/r/20210508150146.28403-2-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/bytcr_rt5640.c | 14 ++++++++++++++ 1 file changed, 14 insertions(+) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 077529113ee7..186c0ee059da 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -643,6 +643,20 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { BYT_RT5640_MONO_SPEAKER | BYT_RT5640_MCLK_EN), }, + { /* Lenovo Miix 3-830 */ + .matches = { + DMI_EXACT_MATCH(DMI_SYS_VENDOR, "LENOVO"), + DMI_EXACT_MATCH(DMI_PRODUCT_VERSION, "Lenovo MIIX 3-830"), + }, + .driver_data = (void *)(BYT_RT5640_IN1_MAP | + BYT_RT5640_JD_SRC_JD2_IN4N | + BYT_RT5640_OVCD_TH_2000UA | + BYT_RT5640_OVCD_SF_0P75 | + BYT_RT5640_MONO_SPEAKER | + BYT_RT5640_DIFF_MIC | + BYT_RT5640_SSP0_AIF1 | + BYT_RT5640_MCLK_EN), + }, { /* Linx Linx7 tablet */ .matches = { DMI_EXACT_MATCH(DMI_SYS_VENDOR, "LINX"), -- 2.30.2

1 0

[PATCH AUTOSEL 4.19 03/23] ASoC: Intel: bytcr_rt5640: Add quirk for the Glavey TM800A550L tablet
by Sasha Levin 03 Jun '21

03 Jun '21

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit 28c268d3acdd4cbcd2ac320b85609e77f84e74a7 ] Add a quirk for the Glavey TM800A550L tablet, this BYTCR tablet has no CHAN package in its ACPI tables and uses SSP0-AIF1 rather then SSP0-AIF2 which is the default for BYTCR devices. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20210508150146.28403-1-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/bytcr_rt5640.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 4ebc023f1507..077529113ee7 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -565,6 +565,17 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { BYT_RT5640_SSP0_AIF1 | BYT_RT5640_MCLK_EN), }, + { /* Glavey TM800A550L */ + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "AMI Corporation"), + DMI_MATCH(DMI_BOARD_NAME, "Aptio CRB"), + /* Above strings are too generic, also match on BIOS version */ + DMI_MATCH(DMI_BIOS_VERSION, "ZY-8-BI-PX4S70VTR400-X423B-005-D"), + }, + .driver_data = (void *)(BYTCR_INPUT_DEFAULTS | + BYT_RT5640_SSP0_AIF1 | + BYT_RT5640_MCLK_EN), + }, { .matches = { DMI_EXACT_MATCH(DMI_SYS_VENDOR, "Hewlett-Packard"), -- 2.30.2

1 0

[PATCH AUTOSEL 5.4 01/31] ASoC: max98088: fix ni clock divider calculation
by Sasha Levin 03 Jun '21

03 Jun '21

From: Marco Felsch <m.felsch(a)pengutronix.de> [ Upstream commit 6c9762a78c325107dc37d20ee21002b841679209 ] The ni1/ni2 ratio formula [1] uses the pclk which is the prescaled mclk. The max98088 datasheet [2] has no such formula but table-12 equals so we can assume that it is the same for both devices. While on it make use of DIV_ROUND_CLOSEST_ULL(). [1] https://datasheets.maximintegrated.com/en/ds/MAX98089.pdf; page 86 [2] https://datasheets.maximintegrated.com/en/ds/MAX98088.pdf; page 82 Signed-off-by: Marco Felsch <m.felsch(a)pengutronix.de> Link: https://lore.kernel.org/r/20210423135402.32105-1-m.felsch@pengutronix.de Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/max98088.c | 13 +++++++++++-- 1 file changed, 11 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c index f031d2caa8b7..fa4cdbfd0b80 100644 --- a/sound/soc/codecs/max98088.c +++ b/sound/soc/codecs/max98088.c @@ -41,6 +41,7 @@ struct max98088_priv { enum max98088_type devtype; struct max98088_pdata *pdata; struct clk *mclk; + unsigned char mclk_prescaler; unsigned int sysclk; struct max98088_cdata dai[2]; int eq_textcnt; @@ -998,13 +999,16 @@ static int max98088_dai1_hw_params(struct snd_pcm_substream *substream, /* Configure NI when operating as master */ if (snd_soc_component_read32(component, M98088_REG_14_DAI1_FORMAT) & M98088_DAI_MAS) { + unsigned long pclk; + if (max98088->sysclk == 0) { dev_err(component->dev, "Invalid system clock frequency\n"); return -EINVAL; } ni = 65536ULL * (rate < 50000 ? 96ULL : 48ULL) * (unsigned long long int)rate; - do_div(ni, (unsigned long long int)max98088->sysclk); + pclk = DIV_ROUND_CLOSEST(max98088->sysclk, max98088->mclk_prescaler); + ni = DIV_ROUND_CLOSEST_ULL(ni, pclk); snd_soc_component_write(component, M98088_REG_12_DAI1_CLKCFG_HI, (ni >> 8) & 0x7F); snd_soc_component_write(component, M98088_REG_13_DAI1_CLKCFG_LO, @@ -1065,13 +1069,16 @@ static int max98088_dai2_hw_params(struct snd_pcm_substream *substream, /* Configure NI when operating as master */ if (snd_soc_component_read32(component, M98088_REG_1C_DAI2_FORMAT) & M98088_DAI_MAS) { + unsigned long pclk; + if (max98088->sysclk == 0) { dev_err(component->dev, "Invalid system clock frequency\n"); return -EINVAL; } ni = 65536ULL * (rate < 50000 ? 96ULL : 48ULL) * (unsigned long long int)rate; - do_div(ni, (unsigned long long int)max98088->sysclk); + pclk = DIV_ROUND_CLOSEST(max98088->sysclk, max98088->mclk_prescaler); + ni = DIV_ROUND_CLOSEST_ULL(ni, pclk); snd_soc_component_write(component, M98088_REG_1A_DAI2_CLKCFG_HI, (ni >> 8) & 0x7F); snd_soc_component_write(component, M98088_REG_1B_DAI2_CLKCFG_LO, @@ -1113,8 +1120,10 @@ static int max98088_dai_set_sysclk(struct snd_soc_dai *dai, */ if ((freq >= 10000000) && (freq < 20000000)) { snd_soc_component_write(component, M98088_REG_10_SYS_CLK, 0x10); + max98088->mclk_prescaler = 1; } else if ((freq >= 20000000) && (freq < 30000000)) { snd_soc_component_write(component, M98088_REG_10_SYS_CLK, 0x20); + max98088->mclk_prescaler = 2; } else { dev_err(component->dev, "Invalid master clock frequency\n"); return -EINVAL; -- 2.30.2

1 3

[PATCH AUTOSEL 5.10 01/39] ASoC: max98088: fix ni clock divider calculation
by Sasha Levin 03 Jun '21

03 Jun '21

From: Marco Felsch <m.felsch(a)pengutronix.de> [ Upstream commit 6c9762a78c325107dc37d20ee21002b841679209 ] The ni1/ni2 ratio formula [1] uses the pclk which is the prescaled mclk. The max98088 datasheet [2] has no such formula but table-12 equals so we can assume that it is the same for both devices. While on it make use of DIV_ROUND_CLOSEST_ULL(). [1] https://datasheets.maximintegrated.com/en/ds/MAX98089.pdf; page 86 [2] https://datasheets.maximintegrated.com/en/ds/MAX98088.pdf; page 82 Signed-off-by: Marco Felsch <m.felsch(a)pengutronix.de> Link: https://lore.kernel.org/r/20210423135402.32105-1-m.felsch@pengutronix.de Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/max98088.c | 13 +++++++++++-- 1 file changed, 11 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c index 4be24e7f51c8..f8e49e45ce33 100644 --- a/sound/soc/codecs/max98088.c +++ b/sound/soc/codecs/max98088.c @@ -41,6 +41,7 @@ struct max98088_priv { enum max98088_type devtype; struct max98088_pdata *pdata; struct clk *mclk; + unsigned char mclk_prescaler; unsigned int sysclk; struct max98088_cdata dai[2]; int eq_textcnt; @@ -998,13 +999,16 @@ static int max98088_dai1_hw_params(struct snd_pcm_substream *substream, /* Configure NI when operating as master */ if (snd_soc_component_read(component, M98088_REG_14_DAI1_FORMAT) & M98088_DAI_MAS) { + unsigned long pclk; + if (max98088->sysclk == 0) { dev_err(component->dev, "Invalid system clock frequency\n"); return -EINVAL; } ni = 65536ULL * (rate < 50000 ? 96ULL : 48ULL) * (unsigned long long int)rate; - do_div(ni, (unsigned long long int)max98088->sysclk); + pclk = DIV_ROUND_CLOSEST(max98088->sysclk, max98088->mclk_prescaler); + ni = DIV_ROUND_CLOSEST_ULL(ni, pclk); snd_soc_component_write(component, M98088_REG_12_DAI1_CLKCFG_HI, (ni >> 8) & 0x7F); snd_soc_component_write(component, M98088_REG_13_DAI1_CLKCFG_LO, @@ -1065,13 +1069,16 @@ static int max98088_dai2_hw_params(struct snd_pcm_substream *substream, /* Configure NI when operating as master */ if (snd_soc_component_read(component, M98088_REG_1C_DAI2_FORMAT) & M98088_DAI_MAS) { + unsigned long pclk; + if (max98088->sysclk == 0) { dev_err(component->dev, "Invalid system clock frequency\n"); return -EINVAL; } ni = 65536ULL * (rate < 50000 ? 96ULL : 48ULL) * (unsigned long long int)rate; - do_div(ni, (unsigned long long int)max98088->sysclk); + pclk = DIV_ROUND_CLOSEST(max98088->sysclk, max98088->mclk_prescaler); + ni = DIV_ROUND_CLOSEST_ULL(ni, pclk); snd_soc_component_write(component, M98088_REG_1A_DAI2_CLKCFG_HI, (ni >> 8) & 0x7F); snd_soc_component_write(component, M98088_REG_1B_DAI2_CLKCFG_LO, @@ -1113,8 +1120,10 @@ static int max98088_dai_set_sysclk(struct snd_soc_dai *dai, */ if ((freq >= 10000000) && (freq < 20000000)) { snd_soc_component_write(component, M98088_REG_10_SYS_CLK, 0x10); + max98088->mclk_prescaler = 1; } else if ((freq >= 20000000) && (freq < 30000000)) { snd_soc_component_write(component, M98088_REG_10_SYS_CLK, 0x20); + max98088->mclk_prescaler = 2; } else { dev_err(component->dev, "Invalid master clock frequency\n"); return -EINVAL; -- 2.30.2

1 4

[PATCH AUTOSEL 5.12 01/43] ASoC: max98088: fix ni clock divider calculation
by Sasha Levin 03 Jun '21

03 Jun '21

From: Marco Felsch <m.felsch(a)pengutronix.de> [ Upstream commit 6c9762a78c325107dc37d20ee21002b841679209 ] The ni1/ni2 ratio formula [1] uses the pclk which is the prescaled mclk. The max98088 datasheet [2] has no such formula but table-12 equals so we can assume that it is the same for both devices. While on it make use of DIV_ROUND_CLOSEST_ULL(). [1] https://datasheets.maximintegrated.com/en/ds/MAX98089.pdf; page 86 [2] https://datasheets.maximintegrated.com/en/ds/MAX98088.pdf; page 82 Signed-off-by: Marco Felsch <m.felsch(a)pengutronix.de> Link: https://lore.kernel.org/r/20210423135402.32105-1-m.felsch@pengutronix.de Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/max98088.c | 13 +++++++++++-- 1 file changed, 11 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c index 4be24e7f51c8..f8e49e45ce33 100644 --- a/sound/soc/codecs/max98088.c +++ b/sound/soc/codecs/max98088.c @@ -41,6 +41,7 @@ struct max98088_priv { enum max98088_type devtype; struct max98088_pdata *pdata; struct clk *mclk; + unsigned char mclk_prescaler; unsigned int sysclk; struct max98088_cdata dai[2]; int eq_textcnt; @@ -998,13 +999,16 @@ static int max98088_dai1_hw_params(struct snd_pcm_substream *substream, /* Configure NI when operating as master */ if (snd_soc_component_read(component, M98088_REG_14_DAI1_FORMAT) & M98088_DAI_MAS) { + unsigned long pclk; + if (max98088->sysclk == 0) { dev_err(component->dev, "Invalid system clock frequency\n"); return -EINVAL; } ni = 65536ULL * (rate < 50000 ? 96ULL : 48ULL) * (unsigned long long int)rate; - do_div(ni, (unsigned long long int)max98088->sysclk); + pclk = DIV_ROUND_CLOSEST(max98088->sysclk, max98088->mclk_prescaler); + ni = DIV_ROUND_CLOSEST_ULL(ni, pclk); snd_soc_component_write(component, M98088_REG_12_DAI1_CLKCFG_HI, (ni >> 8) & 0x7F); snd_soc_component_write(component, M98088_REG_13_DAI1_CLKCFG_LO, @@ -1065,13 +1069,16 @@ static int max98088_dai2_hw_params(struct snd_pcm_substream *substream, /* Configure NI when operating as master */ if (snd_soc_component_read(component, M98088_REG_1C_DAI2_FORMAT) & M98088_DAI_MAS) { + unsigned long pclk; + if (max98088->sysclk == 0) { dev_err(component->dev, "Invalid system clock frequency\n"); return -EINVAL; } ni = 65536ULL * (rate < 50000 ? 96ULL : 48ULL) * (unsigned long long int)rate; - do_div(ni, (unsigned long long int)max98088->sysclk); + pclk = DIV_ROUND_CLOSEST(max98088->sysclk, max98088->mclk_prescaler); + ni = DIV_ROUND_CLOSEST_ULL(ni, pclk); snd_soc_component_write(component, M98088_REG_1A_DAI2_CLKCFG_HI, (ni >> 8) & 0x7F); snd_soc_component_write(component, M98088_REG_1B_DAI2_CLKCFG_LO, @@ -1113,8 +1120,10 @@ static int max98088_dai_set_sysclk(struct snd_soc_dai *dai, */ if ((freq >= 10000000) && (freq < 20000000)) { snd_soc_component_write(component, M98088_REG_10_SYS_CLK, 0x10); + max98088->mclk_prescaler = 1; } else if ((freq >= 20000000) && (freq < 30000000)) { snd_soc_component_write(component, M98088_REG_10_SYS_CLK, 0x20); + max98088->mclk_prescaler = 2; } else { dev_err(component->dev, "Invalid master clock frequency\n"); return -EINVAL; -- 2.30.2

1 6

Allen&Heath Xone43C: USB communication
by Roman 03 Jun '21

03 Jun '21

Hello all, hello list, I am not sure whether this is the correct list, since it's not strictly an ALSA affair (fw to kernel-devel). Recent progress on driver support for Pioneer devices encouraged me to dissect USB communication with an Allen&Heath Xone43C, a mixer with USB interface. It does feature 16 channels (8 in, 8 out), but only 4 (2 in, 2 out) show up, 2 of which remain automagically silent. There is no linux driver yet, and maybe I can help? Addresses are: 0x80 Endpoint 0 (in) 0x00 Endpoint 0 (out) 0x81 Endpoint 1 (in) Interface 2.1 and Interface 2.2 (vendor spec) 0x02 Endpoint 2 (out) Interface 1.1 and Interface 1.2 (vendor spec) 0x85 Endpoint 5 (in) Interface 1.1 0x83 Endpoint 3 (in) Interface 3 MIDI 0x04 Endpoint 4 (out) Interface 3 MIDI ---------------------------------- I have a sniff here: https://drive.google.com/file/d/1UPXCN-r7jc9hfm1d-Hip8vckSs2RQNtC/view? usp=sharing --------------------------------- USB info: Bus 001 Device 006: ID 22f0:000f Allen&Heath Xone:43C Device Descriptor: bLength 18 bDescriptorType 1 bcdUSB 2.00 bDeviceClass 239 Miscellaneous Device bDeviceSubClass 2 bDeviceProtocol 1 Interface Association bMaxPacketSize0 64 idVendor 0x22f0 idProduct 0x000f bcdDevice 2.00 iManufacturer 1 Allen&Heath iProduct 2 Xone:43C iSerial 3 no serial number bNumConfigurations 1 Configuration Descriptor: bLength 9 bDescriptorType 2 wTotalLength 0x0126 bNumInterfaces 4 bConfigurationValue 1 iConfiguration 0 bmAttributes 0xc0 Self Powered MaxPower 0mA Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 1 Control Device bInterfaceProtocol 0 iInterface 0 AudioControl Interface Descriptor: bLength 11 bDescriptorType 36 bDescriptorSubtype 1 (HEADER) bcdADC 1.00 wTotalLength 0x0035 bInCollection 3 baInterfaceNr(0) 1 baInterfaceNr(1) 2 baInterfaceNr(2) 3 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 1 wTerminalType 0x0101 USB Streaming bAssocTerminal 0 bNrChannels 2 wChannelConfig 0x0003 Left Front (L) Right Front (R) iChannelNames 0 iTerminal 0 AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 3 (OUTPUT_TERMINAL) bTerminalID 2 wTerminalType 0x0602 Digital Audio Interface bAssocTerminal 0 bSourceID 1 iTerminal 0 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 3 wTerminalType 0x0602 Digital Audio Interface bAssocTerminal 0 bNrChannels 2 wChannelConfig 0x0003 Left Front (L) Right Front (R) iChannelNames 0 iTerminal 0 AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 3 (OUTPUT_TERMINAL) bTerminalID 4 wTerminalType 0x0101 USB Streaming bAssocTerminal 0 bSourceID 3 iTerminal 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 1 bNumEndpoints 2 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 1 bDelay 1 frames wFormatTag 0x0001 PCM AudioStreaming Interface Descriptor: bLength 11 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 2 bSubframeSize 3 bBitResolution 24 bSamFreqType 1 Discrete tSamFreq[ 0] 48000 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x02 EP 2 OUT bmAttributes 5 Transfer Type Isochronous Synch Type Asynchronous Usage Type Data wMaxPacketSize 0x0126 1x 294 bytes bInterval 4 bRefresh 0 bSynchAddress 133 AudioStreaming Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x01 Sampling Frequency bLockDelayUnits 2 Decoded PCM samples wLockDelay 0x0000 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x85 EP 5 IN bmAttributes 1 Transfer Type Isochronous Synch Type None Usage Type Data wMaxPacketSize 0x0003 1x 3 bytes bInterval 4 bRefresh 5 bSynchAddress 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 2 bNumEndpoints 1 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 0 bInterfaceProtocol 0 iInterface 0 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x02 EP 2 OUT bmAttributes 5 Transfer Type Isochronous Synch Type Asynchronous Usage Type Data wMaxPacketSize 0x0200 1x 512 bytes bInterval 1 bRefresh 0 bSynchAddress 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 4 bDelay 1 frames wFormatTag 0x0001 PCM AudioStreaming Interface Descriptor: bLength 11 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 2 bSubframeSize 3 bBitResolution 24 bSamFreqType 1 Discrete tSamFreq[ 0] 48000 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x81 EP 1 IN bmAttributes 5 Transfer Type Isochronous Synch Type Asynchronous Usage Type Data wMaxPacketSize 0x0126 1x 294 bytes bInterval 4 bRefresh 0 bSynchAddress 0 AudioStreaming Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x01 Sampling Frequency bLockDelayUnits 2 Decoded PCM samples wLockDelay 0x0000 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 2 bNumEndpoints 1 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 0 bInterfaceProtocol 0 iInterface 0 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x81 EP 1 IN bmAttributes 5 Transfer Type Isochronous Synch Type Asynchronous Usage Type Data wMaxPacketSize 0x0200 1x 512 bytes bInterval 1 bRefresh 0 bSynchAddress 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 3 bAlternateSetting 0 bNumEndpoints 2 bInterfaceClass 1 Audio bInterfaceSubClass 3 MIDI Streaming bInterfaceProtocol 0 iInterface 0 MIDIStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (HEADER) bcdADC 1.00 wTotalLength 0x0041 MIDIStreaming Interface Descriptor: bLength 6 bDescriptorType 36 bDescriptorSubtype 2 (MIDI_IN_JACK) bJackType 1 Embedded bJackID 1 iJack 0 MIDIStreaming Interface Descriptor: bLength 6 bDescriptorType 36 bDescriptorSubtype 2 (MIDI_IN_JACK) bJackType 2 External bJackID 2 iJack 0 MIDIStreaming Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 3 (MIDI_OUT_JACK) bJackType 1 Embedded bJackID 3 bNrInputPins 1 baSourceID( 0) 2 BaSourcePin( 0) 1 iJack 0 MIDIStreaming Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 3 (MIDI_OUT_JACK) bJackType 2 External bJackID 4 bNrInputPins 1 baSourceID( 0) 1 BaSourcePin( 0) 1 iJack 0 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x83 EP 3 IN bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0200 1x 512 bytes bInterval 1 bRefresh 0 bSynchAddress 0 MIDIStreaming Endpoint Descriptor: bLength 5 bDescriptorType 37 bDescriptorSubtype 1 (GENERAL) bNumEmbMIDIJack 1 baAssocJackID( 0) 3 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x04 EP 4 OUT bmAttributes 3 Transfer Type Interrupt Synch Type None Usage Type Data wMaxPacketSize 0x0010 1x 16 bytes bInterval 4 bRefresh 0 bSynchAddress 0 MIDIStreaming Endpoint Descriptor: bLength 5 bDescriptorType 37 bDescriptorSubtype 1 (GENERAL) bNumEmbMIDIJack 1 baAssocJackID( 0) 1 Device Qualifier (for other device speed): bLength 10 bDescriptorType 6 bcdUSB 2.00 bDeviceClass 0 bDeviceSubClass 0 bDeviceProtocol 0 bMaxPacketSize0 64 bNumConfigurations 1 Device Status: 0x0001 Self Powered --- Best wishes, Roman

2 1

[PATCH] ASoC: SOF: Intel: Remove duplicate include of shim.h
by Wan Jiabing 03 Jun '21

03 Jun '21

shim.h is included twice. As shim.h is not related to later header file, keep one which has comment and remove the former one. Signed-off-by: Wan Jiabing <wanjiabing(a)vivo.com> --- sound/soc/sof/intel/pci-tng.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/sof/intel/pci-tng.c b/sound/soc/sof/intel/pci-tng.c index 4ee1da3..4bded66 100644 --- a/sound/soc/sof/intel/pci-tng.c +++ b/sound/soc/sof/intel/pci-tng.c @@ -15,7 +15,6 @@ #include <sound/sof.h> #include "../ops.h" #include "atom.h" -#include "shim.h" #include "../sof-pci-dev.h" #include "../sof-audio.h" -- 2.7.4

2 1

[PATCH v2 0/4] ASoC: codecs: wcd934x: add Headset and button detection support
by Srinivas Kandagatla 03 Jun '21

03 Jun '21

This patchset adds support to MBHC(Multi Button Headset Control) block found in Qualcomm WCD codecs. MBHC support headset type detection, both Mechanical and electrical insert/removal detection along with 8 buttons detection, Over current interrupts on HPHL/R, Impedance Measurements on HPHL/R. Eventhough MBHC block supports things like OverCurrent detection, Currently its reported as a kernel debug message. Should this be reported as an uevent to userspace? like the way USB reports? Any suggestions? First patch adds a common mbhc driver and the second one wcd934x specific driver changes along with sdm845 soundcard related changes. Common wcd-mbhc-v2 driver should be reusable across multiple codecs like WCD9335, WCD934x, WCD937x and WCD938x. Most of the work is derived from downstream Qualcomm kernels. Credits to various Qualcomm authors from Patrick Lai's team who have contributed to this code. This patchset thanks, srini Changes since v1: - updated buttons-vthreshold bindings from review comments for wcd938x Srinivas Kandagatla (4): ASoC: dt-bindings: wcd934x: add bindings for Headset Button detection ASoC: codecs: wcd: add multi button Headset detection support ASoC: codecs: wcd934x: add mbhc support ASoC: qcom: sdm845: add jack support for WCD934x .../bindings/sound/qcom,wcd934x.yaml | 30 + include/linux/mfd/wcd934x/registers.h | 57 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/wcd-mbhc-v2.c | 1468 +++++++++++++++++ sound/soc/codecs/wcd-mbhc-v2.h | 340 ++++ sound/soc/codecs/wcd934x.c | 884 +++++++++- sound/soc/qcom/sdm845.c | 8 + 8 files changed, 2778 insertions(+), 15 deletions(-) create mode 100644 sound/soc/codecs/wcd-mbhc-v2.c create mode 100644 sound/soc/codecs/wcd-mbhc-v2.h -- 2.21.0

3 7

[PATCH][next] ASoC: rsnd: check for zero node count
by Colin King 03 Jun '21

03 Jun '21

From: Colin Ian King <colin.king(a)canonical.com> The call to rsnd_node_count can potentially return a zero node count so add a check for this corner case. (Note that the two other calls to rsnd_node_count in the kernel perform this check, so I think it justifies adding this). This avoids using a zero nr in a devm_kcalloc call. Addresses-Coverity: ("Unchecked return value") Fixes: c413983eb66a ("ASoC: rsnd: adjust disabled module") Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/sh/rcar/ssiu.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/sh/rcar/ssiu.c b/sound/soc/sh/rcar/ssiu.c index 5682c74bb7ff..0d8f97633dd2 100644 --- a/sound/soc/sh/rcar/ssiu.c +++ b/sound/soc/sh/rcar/ssiu.c @@ -515,6 +515,9 @@ int rsnd_ssiu_probe(struct rsnd_priv *priv) else nr = priv->ssi_nr; + if (!nr) + return -EINVAL; + ssiu = devm_kcalloc(dev, nr, sizeof(*ssiu), GFP_KERNEL); if (!ssiu) return -ENOMEM; -- 2.31.1

4 3

[PATCH v12 0/4] ASoC: codecs: add rk817 support
by Chris Morgan 03 Jun '21

03 Jun '21

From: Chris Morgan <macromorgan(a)hotmail.com> This series is to add support for the Rockchip rk817 audio codec which is present in all Rockchip rk817 MFD chips. Changes in v12: - Updated codec cell in MFD driver to a single line. Changes in v11: - Stopped checking for presence of parent node in codec driver, as driver should never be called if parent device doesn't exist to call it. - Made codec subnode optional and stopped returning errors if it is missing. - Downgraded missing codec clock message from dev_err to dev_dbg. Users who choose to use the rk817 PMIC but do not choose to use the codec will not get an error message unless they have debugging enabled, whereas users who wish to use the codec but don't have the clock defined will not get an error message unless they have debug messages enabled. This seems like a fair compromise between mututally exclusive users. Note that errors with the clock or codec will still result in a dev_err. - Updated DT documentation to note that codec subnode is optional if no properties are to exist beneath it. Changes in v10: - Correct order of test/ack/signed-off to chronological order. - Removed ifdef from mfd driver keep cell from being added when the codec was not to be built. - Changed codec DT parsing messages from error to debug to prevent errors in dmesg log in the rare but valid case a user has the rk817 PMIC but does not use the codec. Changes in v9: - Add cover letter. - Remove documentation for interrupt parent per Rob Herring's request. - Remove unused MODULE_DEVICE_TABLE to fix a bug identified by kernel test robot. Changes in v8: - Added additional documentation for missing properties of #sound-dai-cells, interrupt-parent, and wakeup-source for mfd documentation. - Corrected order of elements descriptions in device tree documentation. - Changed name of "mic-in-differential" to "rockchip,mic-in-differential". - Changed name of sound card from "rockchip,rk817-codec" to "Analog". - Removed unused resets and reset-names from the i2s1_2ch node. Changes in v7: - Removed ifdef around register definitions for MFD. - Replaced codec documentation with updates to MFD documentation. - Reordered elements in example to comply with upstream rules. - Added binding update back for Odroid Go Advance as requested. - Submitting patches from gmail now. Changes in v6: - Included additional project maintainers for correct subsystems. - Removed unneeded compatible from DT documentation. - Removed binding update for Odroid Go Advance (will do in separate series). Changes in v5: - Move register definitions from rk817_codec.h to main rk808.h register definitions. - Add volatile register for codec bits. - Add default values for codec bits. - Removed of_compatible from mtd driver (not necessary). - Switched to using parent regmap instead of private regmap for codec. Changes in v4: - Created set_pll() call. - Created user visible gain control in mic. - Check for return value of clk_prepare_enable(). - Removed duplicate clk_prepare_enable(). - Split DT documentation to separate commit. Changes in v3: - Use DAPM macros to set audio path. - Updated devicetree binding (as every rk817 has this codec chip). - Changed documentation to yaml format. - Split MFD changes to separate commit. Changes in v2: - Fixed audio path registers to solve some bugs. Chris Morgan (4): mfd: Add Rockchip rk817 audio CODEC support ASoC: Add Rockchip rk817 audio CODEC support dt-bindings: Add Rockchip rk817 audio CODEC support arm64: dts: rockchip: add rk817 codec to Odroid Go .../devicetree/bindings/mfd/rk808.txt | 188 ++++++ .../boot/dts/rockchip/rk3326-odroid-go2.dts | 36 +- drivers/mfd/rk808.c | 81 +++ include/linux/mfd/rk808.h | 81 +++ sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/rk817_codec.c | 539 ++++++++++++++++++ 7 files changed, 931 insertions(+), 2 deletions(-) create mode 100644 sound/soc/codecs/rk817_codec.c -- 2.25.1

4 13

Speakers + Microphone Volume Reset
by GitHub issues - edited 03 Jun '21

03 Jun '21

alsa-project/alsa-lib issue #144 was edited from rosenbergd: Every time I start up, connect or disconnect headphones, my volume resets to 74%, I've also noticed the microphone is doing the same thing. I'm not sure if this is a pulseaudio bug, but here are the logs from `pulseaudio -vv` which seems to show alsa increasing the volume to 74% after initially setting it to 25% which is the correct volume that it should be at: ``` D: [pulseaudio] module-alsa-card.c: Jack 'Headphone Jack' is now plugged in D: [pulseaudio] device-port.c: Setting port analog-output-headphones to status yes D: [pulseaudio] module-switch-on-port-available.c: Trying to switch to port analog-output-headphones I: [pulseaudio] alsa-sink.c: Successfully enabled deferred volume. I: [pulseaudio] alsa-sink.c: Hardware volume ranges from -179.00 dB to 0.00 dB. I: [pulseaudio] alsa-sink.c: Fixing base volume to 0.00 dB I: [pulseaudio] alsa-sink.c: Using hardware volume control. Hardware dB scale supported. I: [pulseaudio] alsa-sink.c: Using hardware mute control. D: [alsa-sink-ALC892 Analog] alsa-mixer.c: Activating path analog-output-headphones D: [alsa-sink-ALC892 Analog] alsa-mixer.c: Path analog-output-headphones (Headphones), direction=1, priority=99, probed=yes, supported=yes, has_mute=yes, has_volume=yes, has_dB=yes, min_volume=0, max_volume=64, min_dB=-179, max_dB=0 D: [alsa-sink-ALC892 Analog] alsa-mixer.c: Element 'Master', direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x7ffffffffffff, n_channels=1, override_map=yes D: [alsa-sink-ALC892 Analog] alsa-mixer.c: Element 'Headphone', direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=4, required_absent=0, mask=0x3600000000f66, n_channels=2, override_map=yes D: [alsa-sink-ALC892 Analog] alsa-mixer.c: Element 'Speaker', direction=1, switch=2, volume=2, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x6, n_channels=2, override_map=no D: [alsa-sink-ALC892 Analog] alsa-mixer.c: Element 'Bass Speaker', direction=1, switch=2, volume=2, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x6, n_channels=2, override_map=no D: [alsa-sink-ALC892 Analog] alsa-mixer.c: Element 'PCM', direction=1, switch=0, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x3600000000f66, n_channels=2, override_map=yes D: [alsa-sink-ALC892 Analog] alsa-mixer.c: Element 'IEC958', direction=1, switch=2, volume=0, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x0, n_channels=0, override_map=no D: [alsa-sink-ALC892 Analog] alsa-mixer.c: Jack Dock Headphone, alsa_name='Dock Headphone Jack', index='0', detection unavailable D: [alsa-sink-ALC892 Analog] alsa-mixer.c: Jack Dock Headphone Phantom, alsa_name='Dock Headphone Phantom Jack', index='0', detection unavailable D: [alsa-sink-ALC892 Analog] alsa-mixer.c: Jack Front Headphone, alsa_name='Front Headphone Jack', index='0', detection unavailable D: [alsa-sink-ALC892 Analog] alsa-mixer.c: Jack Front Headphone, alsa_name='Front Headphone Jack', index='1', detection unavailable D: [alsa-sink-ALC892 Analog] alsa-mixer.c: Jack Front Headphone Front, alsa_name='Front Headphone Front Jack', index='0', detection unavailable D: [alsa-sink-ALC892 Analog] alsa-mixer.c: Jack Front Headphone Surround, alsa_name='Front Headphone Surround Jack', index='0', detection unavailable D: [alsa-sink-ALC892 Analog] alsa-mixer.c: Jack Front Headphone Phantom, alsa_name='Front Headphone Phantom Jack', index='0', detection unavailable D: [alsa-sink-ALC892 Analog] alsa-mixer.c: Jack Headphone, alsa_name='Headphone Jack', index='0', detection possible D: [alsa-sink-ALC892 Analog] alsa-mixer.c: Jack Headphone Phantom, alsa_name='Headphone Phantom Jack', index='0', detection unavailable D: [alsa-sink-ALC892 Analog] alsa-mixer.c: Jack Headphone Mic, alsa_name='Headphone Mic Jack', index='0', detection unavailable D: [alsa-sink-ALC892 Analog] alsa-mixer.c: Jack Headphone - Output, alsa_name='Headphone - Output Jack', index='0', detection unavailable I: [pulseaudio] sink.c: Changed port of sink 0 "alsa_output.pci-0000_00_1f.3.analog-stereo" to analog-output-headphones I: [pulseaudio] module-device-restore.c: Restoring volume for sink alsa_output.pci-0000_00_1f.3.analog-stereo. D: [pulseaudio] sink.c: The reference volume of sink alsa_output.pci-0000_00_1f.3.analog-stereo changed from front-left: 1369 / 2% / -100.80 dB, front-right: 1369 / 2% / -100.80 dB to front-left: 16380 / 25% / -36.13 dB, front-right: 16380 / 25% / -36.13 dB. D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event. D: [alsa-sink-ALC892 Analog] alsa-sink.c: Requested volume: front-left: 16380 / 25% / -36.13 dB, front-right: 16380 / 25% / -36.13 dB D: [alsa-sink-ALC892 Analog] alsa-sink.c: Got hardware volume: front-left: 16462 / 25% / -36.00 dB, front-right: 16462 / 25% / -36.00 dB D: [alsa-sink-ALC892 Analog] alsa-sink.c: Calculated software volume: front-left: 65210 / 100% / -0.13 dB, front-right: 65210 / 100% / -0.13 dB (accurate-enough=yes) D: [alsa-sink-ALC892 Analog] sink.c: Volume going up to 16380 at 263091695 D: [alsa-sink-ALC892 Analog] sink.c: Volume change to 16380 at 263091695 was written 5 usec late I: [pulseaudio] module-device-restore.c: Restoring mute state for sink alsa_output.pci-0000_00_1f.3.analog-stereo. D: [pulseaudio] sink.c: The mute of sink alsa_output.pci-0000_00_1f.3.analog-stereo changed from yes to no. D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event. D: [pulseaudio] device-port.c: Setting port analog-output-speaker to status no D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event. D: [pulseaudio] card.c: Setting card alsa_card.pci-0000_00_1f.3 profile output:analog-surround-21 to availability status no D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event. D: [pulseaudio] card.c: Setting card alsa_card.pci-0000_00_1f.3 profile output:analog-surround-21+input:analog-stereo to availability status no D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event. D: [pulseaudio] card.c: Setting card alsa_card.pci-0000_00_1f.3 profile output:analog-surround-40 to availability status no D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event. D: [pulseaudio] card.c: Setting card alsa_card.pci-0000_00_1f.3 profile output:analog-surround-40+input:analog-stereo to availability status no D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event. D: [alsa-sink-ALC892 Analog] alsa-sink.c: Read hardware volume: front-left: 5578 / 9% / -64.20 dB, front-right: 5578 / 9% / -64.20 dB D: [pulseaudio] sink.c: The reference volume of sink alsa_output.pci-0000_00_1f.3.analog-stereo changed from front-left: 16380 / 25% / -36.13 dB, front-right: 16380 / 25% / -36.13 dB to front-left: 5578 / 9% / -64.20 dB, front-right: 5578 / 9% / -64.20 dB. D: [pulseaudio] sink.c: The mute of sink alsa_output.pci-0000_00_1f.3.analog-stereo changed from no to yes. D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event. D: [alsa-sink-ALC892 Analog] alsa-sink.c: Read hardware volume: front-left: 49714 / 76% / -7.20 dB, front-right: 49714 / 76% / -7.20 dB D: [pulseaudio] sink.c: The reference volume of sink alsa_output.pci-0000_00_1f.3.analog-stereo changed from front-left: 5578 / 9% / -64.20 dB, front-right: 5578 / 9% / -64.20 dB to front-left: 49714 / 76% / -7.20 dB, front-right: 49714 / 76% / -7.20 dB. D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event. D: [pulseaudio] sink.c: The mute of sink alsa_output.pci-0000_00_1f.3.analog-stereo changed from yes to no. D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event. D: [alsa-sink-ALC892 Analog] alsa-sink.c: Read hardware volume: front-left: 48583 / 74% / -7.80 dB, front-right: 48583 / 74% / -7.80 dB D: [pulseaudio] sink.c: The reference volume of sink alsa_output.pci-0000_00_1f.3.analog-stereo changed from front-left: 49714 / 76% / -7.20 dB, front-right: 49714 / 76% / -7.20 dB to front-left: 48583 / 74% / -7.80 dB, front-right: 48583 / 74% / -7.80 dB. D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event. D: [alsa-sink-ALC892 Analog] alsa-sink.c: Read hardware volume: front-left: 48583 / 74% / -7.80 dB, front-right: 48583 / 74% / -7.80 dB D: [alsa-sink-ALC892 Analog] alsa-sink.c: Read hardware volume: front-left: 48583 / 74% / -7.80 dB, front-right: 48583 / 74% / -7.80 dB D: [alsa-sink-ALC892 Analog] alsa-sink.c: Read hardware volume: front-left: 48583 / 74% / -7.80 dB, front-right: 48583 / 74% / -7.80 dB D: [alsa-sink-ALC892 Analog] alsa-sink.c: Read hardware volume: front-left: 48583 / 74% / -7.80 dB, front-right: 48583 / 74% / -7.80 dB D: [alsa-sink-ALC892 Analog] alsa-sink.c: Read hardware volume: front-left: 48583 / 74% / -7.80 dB, front-right: 48583 / 74% / -7.80 dB D: [alsa-sink-ALC892 Analog] alsa-sink.c: Read hardware volume: front-left: 48583 / 74% / -7.80 dB, front-right: 48583 / 74% / -7.80 dB D: [alsa-sink-ALC892 Analog] alsa-sink.c: Read hardware volume: front-left: 48583 / 74% / -7.80 dB, front-right: 48583 / 74% / -7.80 dB I: [pulseaudio] module-device-restore.c: Storing volume/mute for device+port sink:alsa_output.pci-0000_00_1f.3.analog-stereo:analog-output-headphones. I: [pulseaudio] module-device-restore.c: Synced. ``` I'm using Linux Mint 20.1 Cinnamon v4.8.6 Kernel `v5.4.0-74` (but this was happening on `-73` as well) Issue URL : https://github.com/alsa-project/alsa-lib/issues/144 Repository URL: https://github.com/alsa-project/alsa-lib

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Speakers + Microphone Volume Reset
by GitHub issues - opened 03 Jun '21

03 Jun '21

alsa-project/alsa-lib issue #144 was opened from rosenbergd: Every time I start up, connect or disconnect headphones, my volume resets to 74%, I've also noticed the microphone is doing the same thing. I'm not sure if this is a pulseaudio bug, but here are the logs from `pulseaudio -vv` which seems to show alsa increasing the volume to 74% after initially setting it to 25% which is the correct volume that it should be at: ``` D: [pulseaudio] module-alsa-card.c: Jack 'Headphone Jack' is now plugged in D: [pulseaudio] device-port.c: Setting port analog-output-headphones to status yes D: [pulseaudio] module-switch-on-port-available.c: Trying to switch to port analog-output-headphones I: [pulseaudio] alsa-sink.c: Successfully enabled deferred volume. I: [pulseaudio] alsa-sink.c: Hardware volume ranges from -179.00 dB to 0.00 dB. I: [pulseaudio] alsa-sink.c: Fixing base volume to 0.00 dB I: [pulseaudio] alsa-sink.c: Using hardware volume control. Hardware dB scale supported. I: [pulseaudio] alsa-sink.c: Using hardware mute control. D: [alsa-sink-ALC892 Analog] alsa-mixer.c: Activating path analog-output-headphones D: [alsa-sink-ALC892 Analog] alsa-mixer.c: Path analog-output-headphones (Headphones), direction=1, priority=99, probed=yes, supported=yes, has_mute=yes, has_volume=yes, has_dB=yes, min_volume=0, max_volume=64, min_dB=-179, max_dB=0 D: [alsa-sink-ALC892 Analog] alsa-mixer.c: Element 'Master', direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x7ffffffffffff, n_channels=1, override_map=yes D: [alsa-sink-ALC892 Analog] alsa-mixer.c: Element 'Headphone', direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=4, required_absent=0, mask=0x3600000000f66, n_channels=2, override_map=yes D: [alsa-sink-ALC892 Analog] alsa-mixer.c: Element 'Speaker', direction=1, switch=2, volume=2, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x6, n_channels=2, override_map=no D: [alsa-sink-ALC892 Analog] alsa-mixer.c: Element 'Bass Speaker', direction=1, switch=2, volume=2, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x6, n_channels=2, override_map=no D: [alsa-sink-ALC892 Analog] alsa-mixer.c: Element 'PCM', direction=1, switch=0, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x3600000000f66, n_channels=2, override_map=yes D: [alsa-sink-ALC892 Analog] alsa-mixer.c: Element 'IEC958', direction=1, switch=2, volume=0, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x0, n_channels=0, override_map=no D: [alsa-sink-ALC892 Analog] alsa-mixer.c: Jack Dock Headphone, alsa_name='Dock Headphone Jack', index='0', detection unavailable D: [alsa-sink-ALC892 Analog] alsa-mixer.c: Jack Dock Headphone Phantom, alsa_name='Dock Headphone Phantom Jack', index='0', detection unavailable D: [alsa-sink-ALC892 Analog] alsa-mixer.c: Jack Front Headphone, alsa_name='Front Headphone Jack', index='0', detection unavailable D: [alsa-sink-ALC892 Analog] alsa-mixer.c: Jack Front Headphone, alsa_name='Front Headphone Jack', index='1', detection unavailable D: [alsa-sink-ALC892 Analog] alsa-mixer.c: Jack Front Headphone Front, alsa_name='Front Headphone Front Jack', index='0', detection unavailable D: [alsa-sink-ALC892 Analog] alsa-mixer.c: Jack Front Headphone Surround, alsa_name='Front Headphone Surround Jack', index='0', detection unavailable D: [alsa-sink-ALC892 Analog] alsa-mixer.c: Jack Front Headphone Phantom, alsa_name='Front Headphone Phantom Jack', index='0', detection unavailable D: [alsa-sink-ALC892 Analog] alsa-mixer.c: Jack Headphone, alsa_name='Headphone Jack', index='0', detection possible D: [alsa-sink-ALC892 Analog] alsa-mixer.c: Jack Headphone Phantom, alsa_name='Headphone Phantom Jack', index='0', detection unavailable D: [alsa-sink-ALC892 Analog] alsa-mixer.c: Jack Headphone Mic, alsa_name='Headphone Mic Jack', index='0', detection unavailable D: [alsa-sink-ALC892 Analog] alsa-mixer.c: Jack Headphone - Output, alsa_name='Headphone - Output Jack', index='0', detection unavailable I: [pulseaudio] sink.c: Changed port of sink 0 "alsa_output.pci-0000_00_1f.3.analog-stereo" to analog-output-headphones I: [pulseaudio] module-device-restore.c: Restoring volume for sink alsa_output.pci-0000_00_1f.3.analog-stereo. D: [pulseaudio] sink.c: The reference volume of sink alsa_output.pci-0000_00_1f.3.analog-stereo changed from front-left: 1369 / 2% / -100.80 dB, front-right: 1369 / 2% / -100.80 dB to front-left: 16380 / 25% / -36.13 dB, front-right: 16380 / 25% / -36.13 dB. D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event. D: [alsa-sink-ALC892 Analog] alsa-sink.c: Requested volume: front-left: 16380 / 25% / -36.13 dB, front-right: 16380 / 25% / -36.13 dB D: [alsa-sink-ALC892 Analog] alsa-sink.c: Got hardware volume: front-left: 16462 / 25% / -36.00 dB, front-right: 16462 / 25% / -36.00 dB D: [alsa-sink-ALC892 Analog] alsa-sink.c: Calculated software volume: front-left: 65210 / 100% / -0.13 dB, front-right: 65210 / 100% / -0.13 dB (accurate-enough=yes) D: [alsa-sink-ALC892 Analog] sink.c: Volume going up to 16380 at 263091695 D: [alsa-sink-ALC892 Analog] sink.c: Volume change to 16380 at 263091695 was written 5 usec late I: [pulseaudio] module-device-restore.c: Restoring mute state for sink alsa_output.pci-0000_00_1f.3.analog-stereo. D: [pulseaudio] sink.c: The mute of sink alsa_output.pci-0000_00_1f.3.analog-stereo changed from yes to no. D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event. D: [pulseaudio] device-port.c: Setting port analog-output-speaker to status no D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event. D: [pulseaudio] card.c: Setting card alsa_card.pci-0000_00_1f.3 profile output:analog-surround-21 to availability status no D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event. D: [pulseaudio] card.c: Setting card alsa_card.pci-0000_00_1f.3 profile output:analog-surround-21+input:analog-stereo to availability status no D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event. D: [pulseaudio] card.c: Setting card alsa_card.pci-0000_00_1f.3 profile output:analog-surround-40 to availability status no D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event. D: [pulseaudio] card.c: Setting card alsa_card.pci-0000_00_1f.3 profile output:analog-surround-40+input:analog-stereo to availability status no D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event. D: [alsa-sink-ALC892 Analog] alsa-sink.c: Read hardware volume: front-left: 5578 / 9% / -64.20 dB, front-right: 5578 / 9% / -64.20 dB D: [pulseaudio] sink.c: The reference volume of sink alsa_output.pci-0000_00_1f.3.analog-stereo changed from front-left: 16380 / 25% / -36.13 dB, front-right: 16380 / 25% / -36.13 dB to front-left: 5578 / 9% / -64.20 dB, front-right: 5578 / 9% / -64.20 dB. D: [pulseaudio] sink.c: The mute of sink alsa_output.pci-0000_00_1f.3.analog-stereo changed from no to yes. D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event. D: [alsa-sink-ALC892 Analog] alsa-sink.c: Read hardware volume: front-left: 49714 / 76% / -7.20 dB, front-right: 49714 / 76% / -7.20 dB D: [pulseaudio] sink.c: The reference volume of sink alsa_output.pci-0000_00_1f.3.analog-stereo changed from front-left: 5578 / 9% / -64.20 dB, front-right: 5578 / 9% / -64.20 dB to front-left: 49714 / 76% / -7.20 dB, front-right: 49714 / 76% / -7.20 dB. D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event. D: [pulseaudio] sink.c: The mute of sink alsa_output.pci-0000_00_1f.3.analog-stereo changed from yes to no. D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event. D: [alsa-sink-ALC892 Analog] alsa-sink.c: Read hardware volume: front-left: 48583 / 74% / -7.80 dB, front-right: 48583 / 74% / -7.80 dB D: [pulseaudio] sink.c: The reference volume of sink alsa_output.pci-0000_00_1f.3.analog-stereo changed from front-left: 49714 / 76% / -7.20 dB, front-right: 49714 / 76% / -7.20 dB to front-left: 48583 / 74% / -7.80 dB, front-right: 48583 / 74% / -7.80 dB. D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event. D: [alsa-sink-ALC892 Analog] alsa-sink.c: Read hardware volume: front-left: 48583 / 74% / -7.80 dB, front-right: 48583 / 74% / -7.80 dB D: [alsa-sink-ALC892 Analog] alsa-sink.c: Read hardware volume: front-left: 48583 / 74% / -7.80 dB, front-right: 48583 / 74% / -7.80 dB D: [alsa-sink-ALC892 Analog] alsa-sink.c: Read hardware volume: front-left: 48583 / 74% / -7.80 dB, front-right: 48583 / 74% / -7.80 dB D: [alsa-sink-ALC892 Analog] alsa-sink.c: Read hardware volume: front-left: 48583 / 74% / -7.80 dB, front-right: 48583 / 74% / -7.80 dB D: [alsa-sink-ALC892 Analog] alsa-sink.c: Read hardware volume: front-left: 48583 / 74% / -7.80 dB, front-right: 48583 / 74% / -7.80 dB D: [alsa-sink-ALC892 Analog] alsa-sink.c: Read hardware volume: front-left: 48583 / 74% / -7.80 dB, front-right: 48583 / 74% / -7.80 dB D: [alsa-sink-ALC892 Analog] alsa-sink.c: Read hardware volume: front-left: 48583 / 74% / -7.80 dB, front-right: 48583 / 74% / -7.80 dB I: [pulseaudio] module-device-restore.c: Storing volume/mute for device+port sink:alsa_output.pci-0000_00_1f.3.analog-stereo:analog-output-headphones. I: [pulseaudio] module-device-restore.c: Synced. ``` Issue URL : https://github.com/alsa-project/alsa-lib/issues/144 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH v2] ASoC: fsl-asoc-card: change dev_err to dev_err_probe for defer probe
by Shengjiu Wang 03 Jun '21

03 Jun '21

Don't need to print error message for defer probe Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- changes in v2: - use dev_err_probe instead of dev_dbg sound/soc/fsl/fsl-asoc-card.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index 4f55b316cf0f..2fa371ec6b1c 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -709,8 +709,8 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) of_node_put(framemaster); if (!fsl_asoc_card_is_ac97(priv) && !codec_dev) { - dev_err(&pdev->dev, "failed to find codec device\n"); ret = -EPROBE_DEFER; + dev_err_probe(&pdev->dev, ret, "failed to find codec device\n"); goto asrc_fail; } -- 2.27.0

1 0

trivial patch
by josef radinger 02 Jun '21

02 Jun '21

just fix some trivial typo in comments, sorry for the third try. now the direction between diffs is correct. diff --git a/sound/soc/qcom/qdsp6/q6adm.c b/sound/soc/qcom/qdsp6/q6adm.c index 1855b805eba2..3c68df82c95c 100644 --- a/sound/soc/qcom/qdsp6/q6adm.c +++ b/sound/soc/qcom/qdsp6/q6adm.c @@ -373,7 +373,7 @@ static int q6adm_device_open(struct q6adm *adm, struct q6copp *copp, * @rate: rate at which copp is required. * @channel_mode: channel mode * @topology: adm topology id - * @perf_mode: performace mode. + * @perf_mode: performance mode. * @bit_width: audio sample bit width * @app_type: Application type. * @acdb_id: ACDB id @@ -452,7 +452,7 @@ EXPORT_SYMBOL_GPL(q6adm_get_copp_id); * @dev: Pointer to adm child device. * @path: playback or capture path. * @payload_map: map between session id and afe ports. - * @perf_mode: Performace mode. + * @perf_mode: Performance mode. * * Return: Will be an negative on error or a zero on success. */ yours josef

1 0

[PATCH] ASoC: AMD Renoir - add DMI entry for Lenovo 2020 AMD platforms
by Mark Pearson 02 Jun '21

02 Jun '21

More laptops identified where the AMD ACP bridge needs to be blocked or the microphone will not work when connected to HDMI. Use DMI to block the microphone PCM device for these platforms. Suggested-by: Gabriel Craciunescu <nix.or.die(a)gmail.com> Signed-off-by: Mark Pearson <markpearson(a)lenovo.com> --- sound/soc/amd/renoir/rn-pci-acp3x.c | 35 +++++++++++++++++++++++++++++ 1 file changed, 35 insertions(+) diff --git a/sound/soc/amd/renoir/rn-pci-acp3x.c b/sound/soc/amd/renoir/rn-pci-acp3x.c index 19438da5dfa5..c9fb1c8fbf8c 100644 --- a/sound/soc/amd/renoir/rn-pci-acp3x.c +++ b/sound/soc/amd/renoir/rn-pci-acp3x.c @@ -199,6 +199,41 @@ static const struct dmi_system_id rn_acp_quirk_table[] = { DMI_EXACT_MATCH(DMI_BOARD_NAME, "20NLCTO1WW"), } }, + { + /* Lenovo ThinkPad P14s Gen 1 (20Y1) */ + .matches = { + DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "LENOVO"), + DMI_MATCH(DMI_BOARD_NAME, "20Y1"), + } + }, + { + /* Lenovo ThinkPad T14s Gen1 */ + .matches = { + DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "LENOVO"), + DMI_MATCH(DMI_BOARD_NAME, "20UH"), + } + }, + { + /* Lenovo ThinkPad T14s Gen1 Campus */ + .matches = { + DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "LENOVO"), + DMI_MATCH(DMI_BOARD_NAME, "20UJ"), + } + }, + { + /* Lenovo ThinkPad T14 Gen 1*/ + .matches = { + DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "LENOVO"), + DMI_MATCH(DMI_BOARD_NAME, "20UD"), + } + }, + { + /* Lenovo ThinkPad X13 Gen 1*/ + .matches = { + DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "LENOVO"), + DMI_MATCH(DMI_BOARD_NAME, "20UF"), + } + }, {} }; -- 2.31.1

1 1

[PATCH] ALSA: timer: Fix master timer notification
by Takashi Iwai 02 Jun '21

02 Jun '21

snd_timer_notify1() calls the notification to each slave for a master event, but it passes a wrong event number. It should be +10 offset, corresponding to SNDRV_TIMER_EVENT_MXXX, but it's incorrectly with +100 offset. Casually this was spotted by UBSAN check via syzkaller. Reported-by: syzbot+d102fa5b35335a7e544e(a)syzkaller.appspotmail.com Cc: <stable(a)vger.kernel.org> Link: https://lore.kernel.org/r/000000000000e5560e05c3bd1d63@google.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/timer.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/core/timer.c b/sound/core/timer.c index 6898b1ac0d7f..92b7008fcdb8 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -520,9 +520,10 @@ static void snd_timer_notify1(struct snd_timer_instance *ti, int event) return; if (timer->hw.flags & SNDRV_TIMER_HW_SLAVE) return; + event += 10; /* convert to SNDRV_TIMER_EVENT_MXXX */ list_for_each_entry(ts, &ti->slave_active_head, active_list) if (ts->ccallback) - ts->ccallback(ts, event + 100, &tstamp, resolution); + ts->ccallback(ts, event, &tstamp, resolution); } /* start/continue a master timer */ -- 2.26.2

2 1

[PATCH] ASoC: AMD Renoir - add DMI entry for Lenovo 2020 AMD platforms
by Mark Pearson 02 Jun '21

02 Jun '21

More laptops identified where the AMD ACP bridge needs to be blocked or the microphone will not work when connected to HDMI. Use DMI to block the microphone PCM device for these platforms. Suggested-by: Gabriel Craciunescu <nix.or.die(a)gmail.com> Signed-off-by: Mark Pearson <markpearson(a)lenovo.com> --- sound/soc/amd/renoir/rn-pci-acp3x.c | 35 +++++++++++++++++++++++++++++ 1 file changed, 35 insertions(+) diff --git a/sound/soc/amd/renoir/rn-pci-acp3x.c b/sound/soc/amd/renoir/rn-pci-acp3x.c index 19438da5dfa5..c9fb1c8fbf8c 100644 --- a/sound/soc/amd/renoir/rn-pci-acp3x.c +++ b/sound/soc/amd/renoir/rn-pci-acp3x.c @@ -199,6 +199,41 @@ static const struct dmi_system_id rn_acp_quirk_table[] = { DMI_EXACT_MATCH(DMI_BOARD_NAME, "20NLCTO1WW"), } }, + { + /* Lenovo ThinkPad P14s Gen 1 (20Y1) */ + .matches = { + DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "LENOVO"), + DMI_MATCH(DMI_BOARD_NAME, "20Y1"), + } + }, + { + /* Lenovo ThinkPad T14s Gen1 */ + .matches = { + DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "LENOVO"), + DMI_MATCH(DMI_BOARD_NAME, "20UH"), + } + }, + { + /* Lenovo ThinkPad T14s Gen1 Campus */ + .matches = { + DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "LENOVO"), + DMI_MATCH(DMI_BOARD_NAME, "20UJ"), + } + }, + { + /* Lenovo ThinkPad T14 Gen 1*/ + .matches = { + DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "LENOVO"), + DMI_MATCH(DMI_BOARD_NAME, "20UD"), + } + }, + { + /* Lenovo ThinkPad X13 Gen 1*/ + .matches = { + DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "LENOVO"), + DMI_MATCH(DMI_BOARD_NAME, "20UF"), + } + }, {} }; -- 2.31.1

2 3

[PATCH v1 1/4] ASoC: dt-bindings: nxp,tfa989x: Add tfa9897 support
by Vincent Knecht 02 Jun '21

02 Jun '21

Document TFA9897 bindings. Signed-off-by: Vincent Knecht <vincent.knecht(a)mailoo.org> --- Documentation/devicetree/bindings/sound/nxp,tfa989x.yaml | 1 + 1 file changed, 1 insertion(+) diff --git a/Documentation/devicetree/bindings/sound/nxp,tfa989x.yaml b/Documentation/devicetree/bindings/sound/nxp,tfa989x.yaml index 45db5776550c..46ddc1f3fc0c 100644 --- a/Documentation/devicetree/bindings/sound/nxp,tfa989x.yaml +++ b/Documentation/devicetree/bindings/sound/nxp,tfa989x.yaml @@ -13,6 +13,7 @@ properties: compatible: enum: - nxp,tfa9895 + - nxp,tfa9897 reg: maxItems: 1 -- 2.31.1

3 7

[PATCH -next] ASoC: Intel: Skylake: use DEVICE_ATTR_RO macro
by YueHaibing 02 Jun '21

02 Jun '21

Use DEVICE_ATTR_RO() helper instead of plain DEVICE_ATTR(), which makes the code a bit shorter and easier to read. Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/soc/intel/skylake/skl-nhlt.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/intel/skylake/skl-nhlt.c b/sound/soc/intel/skylake/skl-nhlt.c index 87c891c46291..64226072f0ee 100644 --- a/sound/soc/intel/skylake/skl-nhlt.c +++ b/sound/soc/intel/skylake/skl-nhlt.c @@ -149,8 +149,8 @@ int skl_nhlt_update_topology_bin(struct skl_dev *skl) return 0; } -static ssize_t skl_nhlt_platform_id_show(struct device *dev, - struct device_attribute *attr, char *buf) +static ssize_t platform_id_show(struct device *dev, + struct device_attribute *attr, char *buf) { struct pci_dev *pci = to_pci_dev(dev); struct hdac_bus *bus = pci_get_drvdata(pci); @@ -166,7 +166,7 @@ static ssize_t skl_nhlt_platform_id_show(struct device *dev, return sprintf(buf, "%s\n", platform_id); } -static DEVICE_ATTR(platform_id, 0444, skl_nhlt_platform_id_show, NULL); +static DEVICE_ATTR_RO(platform_id); int skl_nhlt_create_sysfs(struct skl_dev *skl) { -- 2.17.1

3 2

[PATCH] ASoC: sigmadsp: Remove the repeated declaration
by Shaokun Zhang 02 Jun '21

02 Jun '21

Function 'sigmadsp_reset' is declared twice, so remove the repeated declaration. Cc: Mark Brown <broonie(a)kernel.org> Signed-off-by: Shaokun Zhang <zhangshaokun(a)hisilicon.com> --- sound/soc/codecs/sigmadsp.h | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/sigmadsp.h b/sound/soc/codecs/sigmadsp.h index d63b8c366efb..2783eff633a1 100644 --- a/sound/soc/codecs/sigmadsp.h +++ b/sound/soc/codecs/sigmadsp.h @@ -44,7 +44,6 @@ struct sigmadsp { struct sigmadsp *devm_sigmadsp_init(struct device *dev, const struct sigmadsp_ops *ops, const char *firmware_name); -void sigmadsp_reset(struct sigmadsp *sigmadsp); int sigmadsp_restrict_params(struct sigmadsp *sigmadsp, struct snd_pcm_substream *substream); -- 2.7.4

2 1

[v4] ASoC: qcom: lpass-cpu: Fix pop noise during audio capture begin
by Judy Hsiao 02 Jun '21

02 Jun '21

From: Srinivasa Rao Mandadapu <srivasam(a)codeaurora.org> This patch fixes PoP noise of around 15ms observed during audio capture begin. Enables BCLK and LRCLK in snd_soc_dai_ops prepare call for introducing some delay before capture start and clock enable. Co-developed-by: Judy Hsiao <judyhsiao(a)chromium.org> Signed-off-by: Judy Hsiao <judyhsiao(a)chromium.org> Signed-off-by: Srinivasa Rao Mandadapu <srivasam(a)codeaurora.org> (am from https://patchwork.kernel.org/patch/12276369/) (also found at https://lore.kernel.org/r/20210524142114.18676-1-srivasam@codeaurora.org) --- Chages Since V3: -- Check BCLK is off before enabling it in lpass_cpu_daiops_prepare as lpass_cpu_daiops_prepare can be called multiple times. -- Check BCLK is on before disabling it in lpass_cpu_daiops_shutdown to fix the WARN. It is because BCLK may not be enabled if lpass_cpu_daiops_prepare is not called before lpass_cpu_daiops_shutdown. -- Adds more comments. Changes Since V2: -- Updated comments as per linux style -- Removed unrelated changes. Changes Since V1: -- Enableed BCLK and LRCLK in dai ops prepare API instead of startup API -- Added comments sound/soc/qcom/lpass-cpu.c | 83 +++++++++++++++++++++++++++++++++++++- 1 file changed, 82 insertions(+), 1 deletion(-) diff --git a/sound/soc/qcom/lpass-cpu.c b/sound/soc/qcom/lpass-cpu.c index af8cb64924a0..b572d7874554 100644 --- a/sound/soc/qcom/lpass-cpu.c +++ b/sound/soc/qcom/lpass-cpu.c @@ -6,6 +6,7 @@ */ #include <linux/clk.h> +#include <linux/clk-provider.h> #include <linux/kernel.h> #include <linux/module.h> #include <linux/of.h> @@ -93,8 +94,28 @@ static void lpass_cpu_daiops_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct lpass_data *drvdata = snd_soc_dai_get_drvdata(dai); + struct lpaif_i2sctl *i2sctl = drvdata->i2sctl; + unsigned int id = dai->driver->id; clk_disable_unprepare(drvdata->mi2s_osr_clk[dai->driver->id]); + /* + * To ensure LRCLK disabled even in device node validation + * Will not impact if disabled in lpass_cpu_daiops_trigger() + * suspend. + */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + regmap_fields_write(i2sctl->spken, id, LPAIF_I2SCTL_SPKEN_DISABLE); + else + regmap_fields_write(i2sctl->micen, id, LPAIF_I2SCTL_MICEN_DISABLE); + + /* + * BCLK may not be enabled if lpass_cpu_daiops_prepare is called before + * lpass_cpu_daiops_shutdown. It's paired with the clk_enable in + * lpass_cpu_daiops_prepare. + */ + if (__clk_is_enabled(drvdata->mi2s_bit_clk[dai->driver->id])) + clk_disable(drvdata->mi2s_bit_clk[dai->driver->id]); + clk_unprepare(drvdata->mi2s_bit_clk[dai->driver->id]); } @@ -275,6 +296,18 @@ static int lpass_cpu_daiops_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + /* + * To ensure lpass BCLK/LRCLK is enabled during + * device resume as lpass_cpu_daiops_prepare() is not called + * after the device resumes. We don't check BCLK status before + * enable/disable BCLK in trigger event because: + * 1. These trigger events are paired, so the BCLK + * enable_count is balanced. + * 2. the BCLK can be shared (ex: headset and headset mic), + * we need to increase the enable_count so that we don't + * turn off the shared BCLK while other devices are using + * it. + */ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { ret = regmap_fields_write(i2sctl->spken, id, LPAIF_I2SCTL_SPKEN_ENABLE); @@ -288,7 +321,8 @@ static int lpass_cpu_daiops_trigger(struct snd_pcm_substream *substream, ret = clk_enable(drvdata->mi2s_bit_clk[id]); if (ret) { - dev_err(dai->dev, "error in enabling mi2s bit clk: %d\n", ret); + dev_err(dai->dev, + "error in enabling mi2s bit clk: %d\n", ret); clk_disable(drvdata->mi2s_osr_clk[id]); return ret; } @@ -296,6 +330,10 @@ static int lpass_cpu_daiops_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + /* + * To ensure lpass BCLK/LRCLK is disabled during + * device suspend. + */ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { ret = regmap_fields_write(i2sctl->spken, id, LPAIF_I2SCTL_SPKEN_DISABLE); @@ -315,12 +353,55 @@ static int lpass_cpu_daiops_trigger(struct snd_pcm_substream *substream, return ret; } +static int lpass_cpu_daiops_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct lpass_data *drvdata = snd_soc_dai_get_drvdata(dai); + struct lpaif_i2sctl *i2sctl = drvdata->i2sctl; + unsigned int id = dai->driver->id; + int ret; + /* + * To ensure lpass BCLK/LRCLK is enabled bit before playback/capture + * data flow starts. This allows other codec to have some delay before + * the data flow. + * (ex: to drop start up pop noise before capture starts). + */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + ret = regmap_fields_write(i2sctl->spken, id, + LPAIF_I2SCTL_SPKEN_ENABLE); + else + ret = regmap_fields_write(i2sctl->micen, id, + LPAIF_I2SCTL_MICEN_ENABLE); + + if (ret) { + dev_err(dai->dev, "error writing to i2sctl reg: %d\n", ret); + return ret; + } + + /* + * Check BCLK is off before enabling it as lpass_cpu_daiops_prepare can + * be called multiple times. It's paired with the clk_disable in + * lpass_cpu_daiops_shutdown. + */ + if (!__clk_is_enabled(drvdata->mi2s_bit_clk[dai->driver->id])) { + ret = clk_enable(drvdata->mi2s_bit_clk[id]); + if (ret) { + dev_err(dai->dev, + "error in enabling mi2s bit clk: %d\n", ret); + clk_disable(drvdata->mi2s_osr_clk[id]); + return ret; + } + } + return 0; +} + const struct snd_soc_dai_ops asoc_qcom_lpass_cpu_dai_ops = { .set_sysclk = lpass_cpu_daiops_set_sysclk, .startup = lpass_cpu_daiops_startup, .shutdown = lpass_cpu_daiops_shutdown, .hw_params = lpass_cpu_daiops_hw_params, .trigger = lpass_cpu_daiops_trigger, + .prepare = lpass_cpu_daiops_prepare, }; EXPORT_SYMBOL_GPL(asoc_qcom_lpass_cpu_dai_ops); -- 2.31.0

2 1

[PATCH] ALSA: hda: update the power_state during the direct-complete
by Hui Wang 02 Jun '21

02 Jun '21

The patch_realtek.c needs to check if the power_state.event equals PM_EVENT_SUSPEND, after using the direct-complete, the suspend() and resume() will be skipped if the codec is already rt_suspended, in this case, the patch_realtek.c will always get PM_EVENT_ON even the system is really resumed from S3. We could set power_state to PMSG_SUSPEND in the prepare(), if other PM functions are called before complete(), those functions will override power_state; if no other PM functions are called before complete(), we could know the suspend() and resume() are skipped since only S3 pm functions could be skipped by direct-complete, in this case set power_state to PMSG_RESUME in the complete(). This could guarantee the first time of calling hda_codec_runtime_resume() after complete() has the correct power_state. Fixes: 215a22ed31a1 ("ALSA: hda: Refactor codec PM to use direct-complete optimization") Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/hda_codec.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index a31009afc025..5462f771c2f9 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -2917,6 +2917,7 @@ static int hda_codec_runtime_resume(struct device *dev) #ifdef CONFIG_PM_SLEEP static int hda_codec_pm_prepare(struct device *dev) { + dev->power.power_state = PMSG_SUSPEND; return pm_runtime_suspended(dev); } @@ -2924,6 +2925,10 @@ static void hda_codec_pm_complete(struct device *dev) { struct hda_codec *codec = dev_to_hda_codec(dev); + /* If no other pm-functions are called between prepare() and complete() */ + if (dev->power.power_state.event == PM_EVENT_SUSPEND) + dev->power.power_state = PMSG_RESUME; + if (pm_runtime_suspended(dev) && (codec->jackpoll_interval || hda_codec_need_resume(codec) || codec->forced_resume)) pm_request_resume(dev); -- 2.25.1

2 1

[syzbot] UBSAN: shift-out-of-bounds in snd_timer_user_ccallback
by Dongliang Mu 02 Jun '21

02 Jun '21

> Hello, > > syzbot found the following issue on: > > HEAD commit: 5ff2756a Merge tag 'nfs-for-5.13-2' of git://git.linux-nfs.. > git tree: upstream > console output: https://syzkaller.appspot.com/x/log.txt?x=17872d5bd00000 > kernel config: https://syzkaller.appspot.com/x/.config?x=770708ea7cfd4916 > dashboard link: https://syzkaller.appspot.com/bug?extid=d102fa5b35335a7e544e > > Unfortunately, I don't have any reproducer for this issue yet. > > IMPORTANT: if you fix the issue, please add the following tag to the commit: > Reported-by: syzbot+d102fa...(a)syzkaller.appspotmail.com > > ================================================================================ > UBSAN: shift-out-of-bounds in sound/core/timer.c:1376:23 > shift exponent 105 is too large for 32-bit type 'int' > CPU: 1 PID: 10368 Comm: syz-executor.1 Not tainted 5.13.0-rc3-syzkaller #0 > Hardware name: Google Google Compute Engine/Google Compute Engine, BIOS Google 01/01/2011 > Call Trace: > __dump_stack lib/dump_stack.c:79 [inline] > dump_stack+0x141/0x1d7 lib/dump_stack.c:120 > ubsan_epilogue+0xb/0x5a lib/ubsan.c:148 > __ubsan_handle_shift_out_of_bounds.cold+0xb1/0x181 lib/ubsan.c:327 > snd_timer_user_ccallback.cold+0x19/0x1e sound/core/timer.c:1376 > > snd_timer_notify1+0x243/0x3b0 sound/core/timer.c:525 The root cause of this bug is in the snd_timer_notify1 [1]. At the end of this function, it calls "ts->ccallback(ts, event + 100, &tstamp, resolution)". Here the variable event is 5. It adds 100 and is passed as 2nd argument of snd_timer_user_ccallback. >From the variable naming, the 2nd argument should an event, and in the range of event enumeration. In fact, 105 (event + 100) is out of this range. I don't quite understand the meaning of adding 100. Any thought here? [1] https://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux.git/tree/sou… -- My best regards to you. No System Is Safe! Dongliang Mu

4 4

[PATCH RESEND] ALSA: timer: Fix master timer notification
by Takashi Iwai 02 Jun '21

02 Jun '21

snd_timer_notify1() calls the notification to each slave for a master event, but it passes a wrong event number. It should be +10 offset, corresponding to SNDRV_TIMER_EVENT_MXXX, but it's incorrectly with +100 offset. Casually this was spotted by UBSAN check via syzkaller. Reported-by: syzbot+d102fa5b35335a7e544e(a)syzkaller.appspotmail.com Cc: <stable(a)vger.kernel.org> Link: https://lore.kernel.org/r/000000000000e5560e05c3bd1d63@google.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/timer.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/core/timer.c b/sound/core/timer.c index 6898b1ac0d7f..92b7008fcdb8 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -520,9 +520,10 @@ static void snd_timer_notify1(struct snd_timer_instance *ti, int event) return; if (timer->hw.flags & SNDRV_TIMER_HW_SLAVE) return; + event += 10; /* convert to SNDRV_TIMER_EVENT_MXXX */ list_for_each_entry(ts, &ti->slave_active_head, active_list) if (ts->ccallback) - ts->ccallback(ts, event + 100, &tstamp, resolution); + ts->ccallback(ts, event, &tstamp, resolution); } /* start/continue a master timer */ -- 2.26.2

1 0

[PATCH][next] ASoC: rk817: remove redundant assignment to pointer node
by Colin King 02 Jun '21

02 Jun '21

From: Colin Ian King <colin.king(a)canonical.com> The pointer node is being initialized with a value that is never read and it is being updated later with a new value. The initialization is redundant and can be removed. Addresses-Coverity: ("Unused value") Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/codecs/rk817_codec.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rk817_codec.c b/sound/soc/codecs/rk817_codec.c index 17e672b85ee5..0d7cc26ded57 100644 --- a/sound/soc/codecs/rk817_codec.c +++ b/sound/soc/codecs/rk817_codec.c @@ -457,7 +457,7 @@ static const struct snd_soc_component_driver soc_codec_dev_rk817 = { static void rk817_codec_parse_dt_property(struct device *dev, struct rk817_codec_priv *rk817) { - struct device_node *node = dev->parent->of_node; + struct device_node *node; node = of_get_child_by_name(dev->parent->of_node, "codec"); if (!node) { -- 2.31.1

3 2

no sound via HDMI
by reinerschrutz＠web.de 02 Jun '21

02 Jun '21

Hi, missing sound via monitor and HDMI cable following ubuntuusers "HDA" article I found: 0[generic ] - HD-Audio Generic HD-Audio Generic at 0xfcc 88000 irg 62 1[generic_1 ] - HD-Audio Generic HD-Audio Generic at 0xfcc 88000 irq 63 looking for a pci device I found: 05:00.1 Audio device Advanced Micro Devices, Inc. [AMD/ATI] Device 1637 05:00.6 Audio device Advanced Mirco Devices, Inc. [AMD] Family 17h (Models 10h 1fh) HD Audio Controller Codec found: ATI R6xx HDMI My system: Asus Prime A520M-K, AMD Ryzen 7 Pro 4750G, 8x 3,6 GHz, 16 GD DDR4 RAM, 500 GB Samsung 870 EVO Advanced Linux Sound Architecture - Driver Cofiguration Guide said to give notice. Thanks. Stefan Ocken

1 0

[PATCH] ASoC: rk817: fix a warning in rk817_probe()
by Chris Morgan 02 Jun '21

02 Jun '21

From: Chris Morgan <macromorgan(a)hotmail.com> The return value of snd_soc_component_write() is stored but not evaluated and this results in a warning when W=1 is set. Stop storing the return value to be consistent with all other calls of snd_soc_component_write() and to remove the warning. Fixes: 0d6a04da9b25 ("ASoC: Add Rockchip rk817 audio CODEC support") Signed-off-by: Chris Morgan <macromorgan(a)hotmail.com> --- sound/soc/codecs/rk817_codec.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/codecs/rk817_codec.c b/sound/soc/codecs/rk817_codec.c index 17e672b85ee5..fd3a5ba034a9 100644 --- a/sound/soc/codecs/rk817_codec.c +++ b/sound/soc/codecs/rk817_codec.c @@ -415,12 +415,11 @@ static int rk817_probe(struct snd_soc_component *component) { struct rk817_codec_priv *rk817 = snd_soc_component_get_drvdata(component); struct rk808 *rk808 = dev_get_drvdata(component->dev->parent); - int ret; snd_soc_component_init_regmap(component, rk808->regmap); rk817->component = component; - ret = snd_soc_component_write(component, RK817_CODEC_DTOP_LPT_SRST, 0x40); + snd_soc_component_write(component, RK817_CODEC_DTOP_LPT_SRST, 0x40); rk817_init(component); -- 2.25.1

2 1

[PATCH 0/3] ALSA: firewire-motu: media clock recovery for sph-aware devices
by Takashi Sakamoto 02 Jun '21

02 Jun '21

Hi, In a commit f9e5ecdfc2c2 ("ALSA: firewire-lib: add replay target to cache sequence of packet"), I categorize devices supported by drivers in ALSA firewire stack in terms of the way to deliver effective sampling transfer frequency. This patchset is for the devices in group 3. The devices are known to have problems when ALSA firewire-motu driver handles. Many of them generate sound with noise. In the worst case, it generates no sound. The devices interpret presentation time to decide playback timing. Unlike the syt-aware devices, the devices interpret the presentation time in source packet header (SPH) per data block, instead of the presentation time in syt field of CIP header. Current implementation of the driver processes the sequence of outgoing packet by computation according to nominal sampling transfer frequency. However, the ideal sequence is not adequate to the devices, actually. With this patchset, the drivers are going to replay the sequence of incoming packets for media clock recovery, instead of nominal sampling transfer frequency. For the detail of sequence replay, please refer to a commit 39c2649c71d8 ("ALSA: firewire-lib: replay sequence of incoming packets for outgoing packets"). The sequence replay is done by two levels; the sequence of the number of data blocks per packet, and the sequence of SPH per data blocks in the packet. Takashi Sakamoto (3): ALSA: firewire-motu: use macro for magic numbers relevant to IEC 61883-1 ALSA: firewire-motu: cache event ticks in source packet header per data block ALSA: firewire-motu: sequence replay for source packet header sound/firewire/motu/amdtp-motu.c | 136 +++++++++++++++--------------- sound/firewire/motu/motu-stream.c | 27 +++++- sound/firewire/motu/motu.h | 14 ++- 3 files changed, 104 insertions(+), 73 deletions(-) -- 2.27.0

2 4

[PATCH v2] ALSA: control led: fix memory leak in snd_ctl_led_register
by Dongliang Mu 02 Jun '21

02 Jun '21

The snd_ctl_led_sysfs_add and snd_ctl_led_sysfs_remove should contain the refcount operations in pair. However, snd_ctl_led_sysfs_remove fails to decrease the refcount to zero, which causes device_release never to be invoked. This leads to memory leak to some resources, like struct device_private. In addition, we also free some other similar memory leaks in snd_ctl_led_init/snd_ctl_led_exit. Fix this by replacing device_del to device_unregister in snd_ctl_led_sysfs_remove/snd_ctl_led_init/snd_ctl_led_exit. Note that, when CONFIG_DEBUG_KOBJECT_RELEASE is enabled, put_device will call kobject_release and delay the release of kobject, which will cause use-after-free when the memory backing the kobject is freed at once. Reported-by: syzbot+08a7d8b51ea048a74ffb(a)syzkaller.appspotmail.com Fixes: a135dfb5de1 ("ALSA: led control - add sysfs kcontrol LED marking layer") Signed-off-by: Dongliang Mu <mudongliangabcd(a)gmail.com> --- v1->v2: fix other similar memory leaks; move kfree(led_card) to a release function. sound/core/control_led.c | 33 ++++++++++++++++++++++++++------- 1 file changed, 26 insertions(+), 7 deletions(-) diff --git a/sound/core/control_led.c b/sound/core/control_led.c index 25f57c14f294..a90e31dbde61 100644 --- a/sound/core/control_led.c +++ b/sound/core/control_led.c @@ -17,6 +17,9 @@ MODULE_LICENSE("GPL"); #define MAX_LED (((SNDRV_CTL_ELEM_ACCESS_MIC_LED - SNDRV_CTL_ELEM_ACCESS_SPK_LED) \ >> SNDRV_CTL_ELEM_ACCESS_LED_SHIFT) + 1) +#define to_led_card_dev(_dev) \ + container_of(_dev, struct snd_ctl_led_card, dev) + enum snd_ctl_led_mode { MODE_FOLLOW_MUTE = 0, MODE_FOLLOW_ROUTE, @@ -371,6 +374,21 @@ static void snd_ctl_led_disconnect(struct snd_card *card) snd_ctl_led_refresh(); } +static void snd_ctl_led_card_release(struct device *dev) +{ + struct snd_ctl_led_card *led_card = to_led_card_dev(dev); + + kfree(led_card); +} + +static void snd_ctl_led_release(struct device *dev) +{ +} + +static void snd_ctl_led_dev_release(struct device *dev) +{ +} + /* * sysfs */ @@ -663,6 +681,7 @@ static void snd_ctl_led_sysfs_add(struct snd_card *card) led_card->number = card->number; led_card->led = led; device_initialize(&led_card->dev); + led_card->dev.release = snd_ctl_led_card_release; if (dev_set_name(&led_card->dev, "card%d", card->number) < 0) goto cerr; led_card->dev.parent = &led->dev; @@ -681,7 +700,6 @@ static void snd_ctl_led_sysfs_add(struct snd_card *card) put_device(&led_card->dev); cerr2: printk(KERN_ERR "snd_ctl_led: unable to add card%d", card->number); - kfree(led_card); } } @@ -700,8 +718,7 @@ static void snd_ctl_led_sysfs_remove(struct snd_card *card) snprintf(link_name, sizeof(link_name), "led-%s", led->name); sysfs_remove_link(&card->ctl_dev.kobj, link_name); sysfs_remove_link(&led_card->dev.kobj, "card"); - device_del(&led_card->dev); - kfree(led_card); + device_unregister(&led_card->dev); led->cards[card->number] = NULL; } } @@ -723,6 +740,7 @@ static int __init snd_ctl_led_init(void) device_initialize(&snd_ctl_led_dev); snd_ctl_led_dev.class = sound_class; + snd_ctl_led_dev.release = snd_ctl_led_dev_release; dev_set_name(&snd_ctl_led_dev, "ctl-led"); if (device_add(&snd_ctl_led_dev)) { put_device(&snd_ctl_led_dev); @@ -733,15 +751,16 @@ static int __init snd_ctl_led_init(void) INIT_LIST_HEAD(&led->controls); device_initialize(&led->dev); led->dev.parent = &snd_ctl_led_dev; + led->dev.release = snd_ctl_led_release; led->dev.groups = snd_ctl_led_dev_attr_groups; dev_set_name(&led->dev, led->name); if (device_add(&led->dev)) { put_device(&led->dev); for (; group > 0; group--) { led = &snd_ctl_leds[group - 1]; - device_del(&led->dev); + device_unregister(&led->dev); } - device_del(&snd_ctl_led_dev); + device_unregister(&snd_ctl_led_dev); return -ENOMEM; } } @@ -767,9 +786,9 @@ static void __exit snd_ctl_led_exit(void) } for (group = 0; group < MAX_LED; group++) { led = &snd_ctl_leds[group]; - device_del(&led->dev); + device_unregister(&led->dev); } - device_del(&snd_ctl_led_dev); + device_unregister(&snd_ctl_led_dev); snd_ctl_led_clean(NULL); } -- 2.25.1

4 3

bytcr-rt5640 initiializes but has wrong settings
by GitHub issues - opened 02 Jun '21

02 Jun '21

alsa-project/alsa-ucm-conf issue #96 was opened from AnterCreeper: My device is Elitepad 1000 G2. The headphone and the mic inside cannot work. However, The speaker works properly. The mic IN1 is the headphone mic insted of the mic inside. I think that it may be because the audio was ported to the wrong place. Sorry for my bad English :D Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/96 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[syzbot] UBSAN: shift-out-of-bounds in snd_timer_user_ccallback
by syzbot 02 Jun '21

02 Jun '21

Hello, syzbot found the following issue on: HEAD commit: 5ff2756a Merge tag 'nfs-for-5.13-2' of git://git.linux-nfs.. git tree: upstream console output: https://syzkaller.appspot.com/x/log.txt?x=17872d5bd00000 kernel config: https://syzkaller.appspot.com/x/.config?x=770708ea7cfd4916 dashboard link: https://syzkaller.appspot.com/bug?extid=d102fa5b35335a7e544e Unfortunately, I don't have any reproducer for this issue yet. IMPORTANT: if you fix the issue, please add the following tag to the commit: Reported-by: syzbot+d102fa5b35335a7e544e(a)syzkaller.appspotmail.com ================================================================================ UBSAN: shift-out-of-bounds in sound/core/timer.c:1376:23 shift exponent 105 is too large for 32-bit type 'int' CPU: 1 PID: 10368 Comm: syz-executor.1 Not tainted 5.13.0-rc3-syzkaller #0 Hardware name: Google Google Compute Engine/Google Compute Engine, BIOS Google 01/01/2011 Call Trace: __dump_stack lib/dump_stack.c:79 [inline] dump_stack+0x141/0x1d7 lib/dump_stack.c:120 ubsan_epilogue+0xb/0x5a lib/ubsan.c:148 __ubsan_handle_shift_out_of_bounds.cold+0xb1/0x181 lib/ubsan.c:327 snd_timer_user_ccallback.cold+0x19/0x1e sound/core/timer.c:1376 snd_timer_notify1+0x243/0x3b0 sound/core/timer.c:525 snd_timer_stop1+0x641/0x890 sound/core/timer.c:656 snd_timer_pause+0x6e/0x90 sound/core/timer.c:738 seq_timer_stop sound/core/seq/seq_timer.c:331 [inline] snd_seq_timer_stop+0x93/0xd0 sound/core/seq/seq_timer.c:341 queue_delete+0x42/0xa0 sound/core/seq/seq_queue.c:133 snd_seq_queue_client_leave+0x37/0x1a0 sound/core/seq/seq_queue.c:552 seq_free_client1.part.0+0x10a/0x260 sound/core/seq/seq_clientmgr.c:280 seq_free_client1 sound/core/seq/seq_clientmgr.c:273 [inline] seq_free_client+0x7b/0xf0 sound/core/seq/seq_clientmgr.c:301 snd_seq_release+0x4d/0xe0 sound/core/seq/seq_clientmgr.c:382 __fput+0x288/0x920 fs/file_table.c:280 task_work_run+0xdd/0x1a0 kernel/task_work.c:164 tracehook_notify_resume include/linux/tracehook.h:189 [inline] exit_to_user_mode_loop kernel/entry/common.c:174 [inline] exit_to_user_mode_prepare+0x272/0x280 kernel/entry/common.c:208 __syscall_exit_to_user_mode_work kernel/entry/common.c:290 [inline] syscall_exit_to_user_mode+0x19/0x60 kernel/entry/common.c:301 do_syscall_64+0x47/0xb0 arch/x86/entry/common.c:57 entry_SYSCALL_64_after_hwframe+0x44/0xae RIP: 0033:0x4193eb Code: 0f 05 48 3d 00 f0 ff ff 77 45 c3 0f 1f 40 00 48 83 ec 18 89 7c 24 0c e8 63 fc ff ff 8b 7c 24 0c 41 89 c0 b8 03 00 00 00 0f 05 <48> 3d 00 f0 ff ff 77 35 44 89 c7 89 44 24 0c e8 a1 fc ff ff 8b 44 RSP: 002b:0000000000a9fb80 EFLAGS: 00000293 ORIG_RAX: 0000000000000003 RAX: 0000000000000000 RBX: 0000000000000005 RCX: 00000000004193eb RDX: 0000000000571b58 RSI: ffffffff89166efe RDI: 0000000000000004 RBP: 0000000000000001 R08: 0000000000000000 R09: 0000001b2ea280a8 R10: 00000000000001d1 R11: 0000000000000293 R12: 000000000014d248 R13: 00000000000003e8 R14: 000000000056bf80 R15: 000000000014d1dd ================================================================================ --- This report is generated by a bot. It may contain errors. See https://goo.gl/tpsmEJ for more information about syzbot. syzbot engineers can be reached at syzkaller(a)googlegroups.com. syzbot will keep track of this issue. See: https://goo.gl/tpsmEJ#status for how to communicate with syzbot.

1 0

[PATCH 0/4] ASoC: rsnd: adjust disabled module for R-Car D3
by Kuninori Morimoto 01 Jun '21

01 Jun '21

Hi Mark, Geert Renesas Sound driver is assuming that all SSI/SRC are connected. But some SSI/SRC are not connected on Some Renesas SoC. ex) H2 E2 SRC0 <= SRC1 SRC1 SRC2 SRC2 ... ... We accepted it by using "status = disabled" on DT before. ex) rcar_sound,src { src-0 { => status = "disabled"; }; src1: src-1 { ... }; ... But R-Car D3 have many disabled modules (It has SSI3/SSI4, SRC5/SRC6), and Renesas SoC maintainer don't want above style on DT. ex) rcar_sound,src { => src0: src-0 { status = "disabled"; }; => src1: src-1 { status = "disabled"; }; => src2: src-2 { status = "disabled"; }; => src3: src-3 { status = "disabled"; }; => src4: src-4 { status = "disabled"; }; src5: src-5 { ... }; ... This patch-set adjust to this situation, and enables to intuitive DT settings. rcar_sound,src { src5: src-5 { ... }; src6: src-6 { ... }; }; Kuninori Morimoto (4): ASoC: rsnd: tidyup rsnd_parse_connect_common() ASoC: rsnd: tidyup rsnd_dma_request_channel() ASoC: rsnd: tidyup rsnd_parse_connect_xxx() ASoC: rsnd: adjust disabled module sound/soc/sh/rcar/core.c | 58 ++++++++++++++++++++++++++++++++++++++-- sound/soc/sh/rcar/dma.c | 8 +++--- sound/soc/sh/rcar/dvc.c | 2 +- sound/soc/sh/rcar/rsnd.h | 16 ++++++----- sound/soc/sh/rcar/src.c | 6 +++-- sound/soc/sh/rcar/ssi.c | 12 ++++++--- sound/soc/sh/rcar/ssiu.c | 10 ++++--- 7 files changed, 91 insertions(+), 21 deletions(-) -- 2.25.1

2 5

[PATCH 1/2] ASoC: SOF: Intel: hda: clean up hda_dsp_dump()
by Kai Vehmanen 01 Jun '21

01 Jun '21

From: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Clean up the hda_dsp_dump() function to avoid duplicating the ROM status and error. Signed-off-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Bard Liao <bard.liao(a)intel.com> Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> --- sound/soc/sof/intel/hda.c | 17 +++++------------ 1 file changed, 5 insertions(+), 12 deletions(-) diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c index 126232a76a10..e1e368ff2b12 100644 --- a/sound/soc/sof/intel/hda.c +++ b/sound/soc/sof/intel/hda.c @@ -394,28 +394,21 @@ void hda_dsp_dump(struct snd_sof_dev *sdev, u32 flags) struct sof_ipc_dsp_oops_xtensa xoops; struct sof_ipc_panic_info panic_info; u32 stack[HDA_DSP_STACK_DUMP_SIZE]; - u32 status, panic; - /* try APL specific status message types first */ + /* print ROM/FW status */ hda_dsp_get_status(sdev); - /* now try generic SOF status messages */ - status = snd_sof_dsp_read(sdev, HDA_DSP_BAR, - HDA_DSP_SRAM_REG_FW_STATUS); - panic = snd_sof_dsp_read(sdev, HDA_DSP_BAR, HDA_DSP_SRAM_REG_FW_TRACEP); - + /* print panic info if FW boot is complete. Otherwise, print the extended ROM status */ if (sdev->fw_state == SOF_FW_BOOT_COMPLETE) { + u32 status = snd_sof_dsp_read(sdev, HDA_DSP_BAR, HDA_DSP_SRAM_REG_FW_STATUS); + u32 panic = snd_sof_dsp_read(sdev, HDA_DSP_BAR, HDA_DSP_SRAM_REG_FW_TRACEP); + hda_dsp_get_registers(sdev, &xoops, &panic_info, stack, HDA_DSP_STACK_DUMP_SIZE); snd_sof_get_status(sdev, status, panic, &xoops, &panic_info, stack, HDA_DSP_STACK_DUMP_SIZE); } else { - sof_dev_dbg_or_err(sdev->dev, flags & SOF_DBG_DUMP_FORCE_ERR_LEVEL, - "status = 0x%8.8x panic = 0x%8.8x\n", - status, panic); - hda_dsp_dump_ext_rom_status(sdev, flags); - hda_dsp_get_status(sdev); } } base-commit: 058efb40641845432c52777443b3372dbc97c032 -- 2.31.1

2 2

[PATCH] ASoC: SOF: Intel: pci-tgl: add ADL-M support
by Kai Vehmanen 01 Jun '21

01 Jun '21

Add PCI DID for Intel AlderLake-M. Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Péter Ujfalusi <peter.ujfalusi(a)intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/sof/intel/pci-tgl.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/sof/intel/pci-tgl.c b/sound/soc/sof/intel/pci-tgl.c index 88c3bf404dd7..a00262184efa 100644 --- a/sound/soc/sof/intel/pci-tgl.c +++ b/sound/soc/sof/intel/pci-tgl.c @@ -116,6 +116,8 @@ static const struct pci_device_id sof_pci_ids[] = { .driver_data = (unsigned long)&adls_desc}, { PCI_DEVICE(0x8086, 0x51c8), /* ADL-P */ .driver_data = (unsigned long)&adl_desc}, + { PCI_DEVICE(0x8086, 0x51cc), /* ADL-M */ + .driver_data = (unsigned long)&adl_desc}, { 0, } }; MODULE_DEVICE_TABLE(pci, sof_pci_ids); base-commit: 058efb40641845432c52777443b3372dbc97c032 -- 2.31.1

3 2

[PATCH] ASoC: fsl-asoc-card: Set .owner attribute when registering card.
by Nicolas Cavallari 01 Jun '21

01 Jun '21

Otherwise, when compiled as module, a WARN_ON is triggered: WARNING: CPU: 0 PID: 5 at sound/core/init.c:208 snd_card_new+0x310/0x39c [snd] [...] CPU: 0 PID: 5 Comm: kworker/0:0 Not tainted 5.10.39 #1 Hardware name: Freescale i.MX6 Quad/DualLite (Device Tree) Workqueue: events deferred_probe_work_func [<c0111988>] (unwind_backtrace) from [<c010c8ac>] (show_stack+0x10/0x14) [<c010c8ac>] (show_stack) from [<c092784c>] (dump_stack+0xdc/0x104) [<c092784c>] (dump_stack) from [<c0129710>] (__warn+0xd8/0x114) [<c0129710>] (__warn) from [<c0922a48>] (warn_slowpath_fmt+0x5c/0xc4) [<c0922a48>] (warn_slowpath_fmt) from [<bf0496f8>] (snd_card_new+0x310/0x39c [snd]) [<bf0496f8>] (snd_card_new [snd]) from [<bf1d7df8>] (snd_soc_bind_card+0x334/0x9c4 [snd_soc_core]) [<bf1d7df8>] (snd_soc_bind_card [snd_soc_core]) from [<bf1e9cd8>] (devm_snd_soc_register_card+0x30/0x6c [snd_soc_core]) [<bf1e9cd8>] (devm_snd_soc_register_card [snd_soc_core]) from [<bf22d964>] (fsl_asoc_card_probe+0x550/0xcc8 [snd_soc_fsl_asoc_card]) [<bf22d964>] (fsl_asoc_card_probe [snd_soc_fsl_asoc_card]) from [<c060c930>] (platform_drv_probe+0x48/0x98) [...] Signed-off-by: Nicolas Cavallari <nicolas.cavallari(a)green-communications.fr> --- sound/soc/fsl/fsl-asoc-card.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index c62bfd1c3ac7..4f55b316cf0f 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -744,6 +744,7 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) /* Initialize sound card */ priv->pdev = pdev; priv->card.dev = &pdev->dev; + priv->card.owner = THIS_MODULE; ret = snd_soc_of_parse_card_name(&priv->card, "model"); if (ret) { snprintf(priv->name, sizeof(priv->name), "%s-audio", -- 2.32.0.rc0

3 2

[PATCH] ASoC: SOF: reset enabled_cores state at suspend
by Kai Vehmanen 01 Jun '21

01 Jun '21

The recent changes to use common code to power up/down DSP cores also removed the reset of the core state at suspend. It turns out this is still needed. When the firmware state is reset to SOF_FW_BOOT_NOT_STARTED, also enabled_cores should be reset, and existing DSP drivers depend on this. BugLink: https://github.com/thesofproject/linux/issues/2824 Fixes: 42077f08b3 ("ASoC: SOF: update dsp core power status in common APIs") Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> --- sound/soc/sof/pm.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/sof/pm.c b/sound/soc/sof/pm.c index fd265803f7bc..c83fb6255961 100644 --- a/sound/soc/sof/pm.c +++ b/sound/soc/sof/pm.c @@ -256,6 +256,7 @@ static int sof_suspend(struct device *dev, bool runtime_suspend) /* reset FW state */ sdev->fw_state = SOF_FW_BOOT_NOT_STARTED; + sdev->enabled_cores_mask = 0; return ret; } base-commit: 058efb40641845432c52777443b3372dbc97c032 -- 2.31.1

2 1

[PATCH] ASoC: topology: Fix spelling mistake "vesion" -> "version"
by Colin King 01 Jun '21

01 Jun '21

From: Colin Ian King <colin.king(a)canonical.com> There are spelling mistakes in comments. Fix them. Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/soc-topology.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 5e65b72910e9..0a24d0d409d2 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -1881,7 +1881,7 @@ static void stream_caps_new_ver(struct snd_soc_tplg_stream_caps *dest, * @src: older version of pcm as a source * @pcm: latest version of pcm created from the source * - * Support from vesion 4. User should free the returned pcm manually. + * Support from version 4. User should free the returned pcm manually. */ static int pcm_new_ver(struct soc_tplg *tplg, struct snd_soc_tplg_pcm *src, @@ -2069,7 +2069,7 @@ static void set_link_hw_format(struct snd_soc_dai_link *link, * @src: old version of phyical link config as a source * @link: latest version of physical link config created from the source * - * Support from vesion 4. User need free the returned link config manually. + * Support from version 4. User need free the returned link config manually. */ static int link_new_ver(struct soc_tplg *tplg, struct snd_soc_tplg_link_config *src, @@ -2380,7 +2380,7 @@ static int soc_tplg_dai_elems_load(struct soc_tplg *tplg, * @src: old version of manifest as a source * @manifest: latest version of manifest created from the source * - * Support from vesion 4. Users need free the returned manifest manually. + * Support from version 4. Users need free the returned manifest manually. */ static int manifest_new_ver(struct soc_tplg *tplg, struct snd_soc_tplg_manifest *src, -- 2.31.1

2 1

[PATCH -next] ASoC: ti: omap-mcbsp: use DEVICE_ATTR_RW macro
by YueHaibing 01 Jun '21

01 Jun '21

Use DEVICE_ATTR_RW() helper instead of plain DEVICE_ATTR(), which makes the code a bit shorter and easier to read. Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/soc/ti/omap-mcbsp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/ti/omap-mcbsp.c b/sound/soc/ti/omap-mcbsp.c index db47981768c5..4479d74f0a45 100644 --- a/sound/soc/ti/omap-mcbsp.c +++ b/sound/soc/ti/omap-mcbsp.c @@ -539,7 +539,7 @@ static ssize_t prop##_store(struct device *dev, \ return size; \ } \ \ -static DEVICE_ATTR(prop, 0644, prop##_show, prop##_store) +static DEVICE_ATTR_RW(prop) THRESHOLD_PROP_BUILDER(max_tx_thres); THRESHOLD_PROP_BUILDER(max_rx_thres); -- 2.17.1

4 3

[PATCH 0/5] ASoC: Constify snd_compress_ops
by Rikard Falkeborn 01 Jun '21

01 Jun '21

The only use of the static and global snd_compress_ops structs is to assign their address to the compress_ops field in the snd_soc_component_driver struct which is a pointer to const. Make them const to allow the compiler to put them in read-only memory. Rikard Falkeborn (5): ASoC: cs47125: Constify static struct snd_compress_ops ASoC: wm5102: Constify static struct snd_compress_ops ASoC: wm5110: Constify static struct snd_compress_ops ASoC: qcom: q6asm-dai: Constify static struct snd_compress_ops ASoC: SOF: Intel: Constify sof_probe_compressed_ops sound/soc/codecs/cs47l24.c | 2 +- sound/soc/codecs/wm5102.c | 2 +- sound/soc/codecs/wm5110.c | 2 +- sound/soc/qcom/qdsp6/q6asm-dai.c | 2 +- sound/soc/sof/compress.c | 2 +- sound/soc/sof/compress.h | 2 +- 6 files changed, 6 insertions(+), 6 deletions(-) -- 2.31.1

3 7

alsa-lib 1.2.5 no devices on Lenovo Carbon (gen 7) - pipewire
by GitHub issues - edited 01 Jun '21

01 Jun '21

alsa-project/alsa-lib issue #142 was edited from sambo57u: I use Fedora 34 and updated alsa-lib and all other alsa products to version 1.2.5. They work fine on my desktop machines but on Lenovo laptop I am getting the message alsactl cannot find hw:0 and daemon deactivates with no devices showing. Downgrading to 1.2.4 solves the problem. Issue URL : https://github.com/alsa-project/alsa-lib/issues/142 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

Incomplete sound device detection in PulseAudio with alsa-lib 1.2.5
by GitHub issues - opened 01 Jun '21

01 Jun '21

alsa-project/alsa-lib issue #143 was opened from foutrelis: (Copying my comments over from #142 since it appears to be a different issue to that one. Sorry for the noise there!) After upgrading to alsa-lib 1.2.5 Chromium stopped playing audio until pulseaudio was restarted. Upon closer inspection, `pactl list cards` is now missing the `HDA Intel HDMI` card and only shows the `HDA Intel PCH` one. Some error messages I've gathered: - After upgrading to alsa-lib and before restarting pulseaudio, attempting to play a video in Chromium would result in: `pulseaudio[625]: Error opening PCM device front:0: Invalid argument` - Restarting pulseaudio with alsa-lib 1.2.5 installed logs the following to the journal: ``` pulseaudio[66345]: Failed to find a working profile. pulseaudio[66345]: Failed to load module "module-alsa-card" (argument: "device_id="1" name="pci-0000_00_03.0" card_name="alsa_card.pci-0000_00_03.0" 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. ``` Git bisect points to commit 63f7745be504e447923f2cde421177a2fca99340 as the first commit that results in the `Failed to find a working profile` error. After restarting pulseaudio with alsa-lib 1.2.5 installed, it appears that `pactl list cards` shows one fewer cards on my laptop which has this audio hardware: ``` $ lspci | grep Audio 00:03.0 Audio device: Intel Corporation Haswell-ULT HD Audio Controller (rev 0b) 00:1b.0 Audio device: Intel Corporation 8 Series HD Audio Controller (rev 04) ``` [alsa-lib-1.2.4-pactl-list-cards.txt](https://github.com/alsa-project/alsa-l… [alsa-lib-1.2.5-pactl-list-cards.txt](https://github.com/alsa-project/alsa-l… `pulseaudio -v` output: [pulseaudio-verbose.txt](https://github.com/alsa-project/alsa-lib/files/6577… Issue URL : https://github.com/alsa-project/alsa-lib/issues/143 Repository URL: https://github.com/alsa-project/alsa-lib

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[PATCH 0/5] ALSA: usb-audio: Delay account cleanup and latency reduction
by Takashi Iwai 01 Jun '21

01 Jun '21

Hi, this is a patch set for USB-audio driver for the code refactoring of PCM delay account as well as the reduction of the playback latency. The last change is a bit intrusive (despite of the smallness), and hopefully it won't break things. My initial tests through various backends were OK, but if anyone finds a problem, please let me know. Takashi === Takashi Iwai (5): ALSA: usb-audio: Make snd_usb_pcm_delay() static ALSA: usb-audio: Pre-calculate buffer byte size ALSA: usb-audio: Refactoring delay account code ALSA: usb-audio: Factor out DSD bitrev copy function ALSA: usb-audio: Reduce latency at playback start sound/usb/card.h | 8 +- sound/usb/endpoint.c | 1 + sound/usb/pcm.c | 202 ++++++++++++++++++++----------------------- sound/usb/pcm.h | 3 - 4 files changed, 101 insertions(+), 113 deletions(-) -- 2.26.2

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