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

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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 > >

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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

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[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

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[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

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[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.

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[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

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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

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