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

[PATCH 0/3] ASoC: cs35l56: Bugfixes and efficiency improvement
by Richard Fitzgerald 19 May '23

19 May '23

First two patches are bugfixes. Third patch skips the overhead of rebooting the amp after applying firmware files when we know that it isn't necessary. Simon Trimmer (3): ASoC: cs35l56: Move DSP part string generation so that it is done only once ASoC: cs35l56: sdw_write_no_pm() should be performed under a pm_runtime request ASoC: cs35l56: In secure mode skip SHUTDOWN and RESET around fw download include/sound/cs35l56.h | 1 + sound/soc/codecs/cs35l56.c | 65 +++++++++++++++++++++++++++----------- 2 files changed, 47 insertions(+), 19 deletions(-) -- 2.30.2

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[PATCH AUTOSEL 6.3 01/24] ASoC: jack: allow multiple interrupt per gpio
by Sasha Levin 19 May '23

19 May '23

From: Svyatoslav Ryhel <clamor95(a)gmail.com> [ Upstream commit a2d4051b0bd6dffcd736888ae89a550d6f60b060 ] This feature is required for coupled hp-mic quirk used by some Nvidia Tegra 3 based devices work properly. Signed-off-by: Svyatoslav Ryhel <clamor95(a)gmail.com> Link: https://lore.kernel.org/r/20230221183211.21964-3-clamor95@gmail.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/soc-jack.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index fcece5ca38c69..f951acb2ce362 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -367,6 +367,7 @@ int snd_soc_jack_add_gpios(struct snd_soc_jack *jack, int count, ret = request_any_context_irq(gpiod_to_irq(gpios[i].desc), gpio_handler, + IRQF_SHARED | IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING, gpios[i].name, -- 2.39.2

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[PATCH] ASoC: soc-pcm: test if a BE can be prepared
by Pierre-Louis Bossart 19 May '23

19 May '23

From: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> In the BE hw_params configuration, the existing code checks if any of the existing FEs are prepared, running, paused or suspended - and skips the configuration in those cases. This allows multiple calls of hw_params which the ALSA state machine supports. This check is not handled for the prepare stage, which can lead to the same BE being prepared multiple times. This patch adds a check similar to that of the hw_params, with the main difference being that the suspended state is allowed: the ALSA state machine allows a transition from suspended to prepared with hw_params skipped. This problem was detected on Intel IPC4/SoundWire devices, where the BE dailink .prepare stage is used to configure the SoundWire stream with a bank switch. Multiple .prepare calls lead to conflicts with the .trigger operation with IPC4 configurations. This problem was not detected earlier on Intel devices, HDaudio BE dailinks detect that the link is already prepared and skip the configuration, and for IPC3 devices there is no BE trigger. Link: https://github.com/thesofproject/sof/issues/7596 Signed-off-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- include/sound/soc-dpcm.h | 4 ++++ sound/soc/soc-pcm.c | 20 ++++++++++++++++++++ 2 files changed, 24 insertions(+) diff --git a/include/sound/soc-dpcm.h b/include/sound/soc-dpcm.h index 4d6ac7699833..ebd24753dd00 100644 --- a/include/sound/soc-dpcm.h +++ b/include/sound/soc-dpcm.h @@ -122,6 +122,10 @@ int snd_soc_dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe, int snd_soc_dpcm_can_be_params(struct snd_soc_pcm_runtime *fe, struct snd_soc_pcm_runtime *be, int stream); +/* can this BE perform prepare */ +int snd_soc_dpcm_can_be_prepared(struct snd_soc_pcm_runtime *fe, + struct snd_soc_pcm_runtime *be, int stream); + /* is the current PCM operation for this FE ? */ int snd_soc_dpcm_fe_can_update(struct snd_soc_pcm_runtime *fe, int stream); diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index adb69d7820a8..4fb1ac8e1c4a 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2405,6 +2405,9 @@ int dpcm_be_dai_prepare(struct snd_soc_pcm_runtime *fe, int stream) if (!snd_soc_dpcm_be_can_update(fe, be, stream)) continue; + if (!snd_soc_dpcm_can_be_prepared(fe, be, stream)) + continue; + if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_HW_PARAMS) && (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP) && (be->dpcm[stream].state != SND_SOC_DPCM_STATE_SUSPEND) && @@ -3042,3 +3045,20 @@ int snd_soc_dpcm_can_be_params(struct snd_soc_pcm_runtime *fe, return snd_soc_dpcm_check_state(fe, be, stream, state, ARRAY_SIZE(state)); } EXPORT_SYMBOL_GPL(snd_soc_dpcm_can_be_params); + +/* + * We can only prepare a BE DAI if any of it's FE are not prepared, + * running or paused for the specified stream direction. + */ +int snd_soc_dpcm_can_be_prepared(struct snd_soc_pcm_runtime *fe, + struct snd_soc_pcm_runtime *be, int stream) +{ + const enum snd_soc_dpcm_state state[] = { + SND_SOC_DPCM_STATE_START, + SND_SOC_DPCM_STATE_PAUSED, + SND_SOC_DPCM_STATE_PREPARE, + }; + + return snd_soc_dpcm_check_state(fe, be, stream, state, ARRAY_SIZE(state)); +} +EXPORT_SYMBOL_GPL(snd_soc_dpcm_can_be_prepared); -- 2.37.2

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[PATCH v2 0/7] Add Chameleon v3 ASoC audio
by Paweł Anikiel 19 May '23

19 May '23

The Google Chameleon v3 is a device made for testing audio and video paths of other devices. This patchset adds support for ASoC audio on this device. It has two audio sources: HDMI audio from the it68051 chip (RX only), and analog audio from the ssm2603 chip (RX and TX). The patchset adds the ASoC platform and codec drivers, as well as some changes to the existing ssm2602 codec driver. v2 changes: * Replace chv3-audio machine driver with simple-audio-card * Use regmap_register_patch() in ssm2602 workaround * Remove ssm2602 mute gpio patch, add a gpio hog instead * Simplify hw_pointer logic in chv3-i2s * Commit splits & other misc changes Paweł Anikiel (7): ASoC: Add Google Chameleon v3 i2s driver ASoC: Add Google Chameleon v3 codec driver ASoC: dt-bindings: Add Google Chameleon v3 i2s device ASoC: dt-bindings: Add Google Chameleon v3 audio codec ARM: dts: chameleonv3: Add mute gpio hog ARM: dts: chameleonv3: Add Chameleon v3 audio ASoC: ssm2602: Add workaround for playback distortions .../bindings/sound/google,chv3-codec.yaml | 31 ++ .../bindings/sound/google,chv3-i2s.yaml | 44 +++ .../boot/dts/socfpga_arria10_chameleonv3.dts | 62 ++++ sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/codecs/Kconfig | 8 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/chv3-codec.c | 41 +++ sound/soc/codecs/ssm2602.c | 15 + sound/soc/google/Kconfig | 6 + sound/soc/google/Makefile | 2 + sound/soc/google/chv3-i2s.c | 338 ++++++++++++++++++ 12 files changed, 551 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/google,chv3-codec.yaml create mode 100644 Documentation/devicetree/bindings/sound/google,chv3-i2s.yaml create mode 100644 sound/soc/codecs/chv3-codec.c create mode 100644 sound/soc/google/Kconfig create mode 100644 sound/soc/google/Makefile create mode 100644 sound/soc/google/chv3-i2s.c -- 2.40.1.521.gf1e218fcd8-goog

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[PATCH] SoC: rt5682: Disable jack detection interrupt during suspend
by Matthias Kaehlcke 18 May '23

18 May '23

The rt5682 driver switches its regmap to cache-only when the device suspends and back to regular mode on resume. When the jack detect interrupt fires rt5682_irq() schedules the jack detect work. This can result in invalid reads from the regmap in cache-only mode if the work runs before the device has resumed: [ 56.245502] rt5682 9-001a: ASoC: error at soc_component_read_no_lock on rt5682.9-001a for register: [0x000000f0] -16 Disable the jack detection interrupt during suspend and re-enable it on resume. The driver already schedules the jack detection work on resume, so any state change during suspend is still handled. This is essentially the same as commit f7d00a9be147 ("SoC: rt5682s: Disable jack detection interrupt during suspend") for the rt5682s. Cc: stable(a)kernel.org Signed-off-by: Matthias Kaehlcke <mka(a)chromium.org> --- sound/soc/codecs/rt5682-i2c.c | 4 +++- sound/soc/codecs/rt5682.c | 6 ++++++ sound/soc/codecs/rt5682.h | 1 + 3 files changed, 10 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5682-i2c.c b/sound/soc/codecs/rt5682-i2c.c index 2935c1bb81f3..5bc46b041786 100644 --- a/sound/soc/codecs/rt5682-i2c.c +++ b/sound/soc/codecs/rt5682-i2c.c @@ -267,7 +267,9 @@ static int rt5682_i2c_probe(struct i2c_client *i2c) ret = devm_request_threaded_irq(&i2c->dev, i2c->irq, NULL, rt5682_irq, IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING | IRQF_ONESHOT, "rt5682", rt5682); - if (ret) + if (!ret) + rt5682->irq = i2c->irq; + else dev_err(&i2c->dev, "Failed to reguest IRQ: %d\n", ret); } diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index f6c798b65c08..5d992543b791 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -2959,6 +2959,9 @@ static int rt5682_suspend(struct snd_soc_component *component) if (rt5682->is_sdw) return 0; + if (rt5682->irq) + disable_irq(rt5682->irq); + cancel_delayed_work_sync(&rt5682->jack_detect_work); cancel_delayed_work_sync(&rt5682->jd_check_work); if (rt5682->hs_jack && (rt5682->jack_type & SND_JACK_HEADSET) == SND_JACK_HEADSET) { @@ -3027,6 +3030,9 @@ static int rt5682_resume(struct snd_soc_component *component) mod_delayed_work(system_power_efficient_wq, &rt5682->jack_detect_work, msecs_to_jiffies(0)); + if (rt5682->irq) + enable_irq(rt5682->irq); + return 0; } #else diff --git a/sound/soc/codecs/rt5682.h b/sound/soc/codecs/rt5682.h index d568c6993c33..e8efd8a84a6c 100644 --- a/sound/soc/codecs/rt5682.h +++ b/sound/soc/codecs/rt5682.h @@ -1462,6 +1462,7 @@ struct rt5682_priv { int pll_out[RT5682_PLLS]; int jack_type; + int irq; int irq_work_delay_time; }; -- 2.40.1.606.ga4b1b128d6-goog

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[PATCH v2] ASoC:codecs: lpass: Fix for KASAN use_after_free out of bounds
by Ravulapati Vishnu Vardhan Rao 18 May '23

18 May '23

When we run syzkaller we get below Out of Bounds error. "KASAN: slab-out-of-bounds Read in regcache_flat_read" Below is the backtrace of the issue: BUG: KASAN: slab-out-of-bounds in regcache_flat_read+0x10c/0x110 Read of size 4 at addr ffffff8088fbf714 by task syz-executor.4/14144 CPU: 6 PID: 14144 Comm: syz-executor.4 Tainted: G W Hardware name: Qualcomm Technologies, Inc. sc7280 CRD platform (rev5+) (DT) Call trace: dump_backtrace+0x0/0x4ec show_stack+0x34/0x50 dump_stack_lvl+0xdc/0x11c print_address_description+0x30/0x2d8 kasan_report+0x178/0x1e4 __asan_report_load4_noabort+0x44/0x50 regcache_flat_read+0x10c/0x110 regcache_read+0xf8/0x5a0 _regmap_read+0x45c/0x86c _regmap_update_bits+0x128/0x290 regmap_update_bits_base+0xc0/0x15c snd_soc_component_update_bits+0xa8/0x22c snd_soc_component_write_field+0x68/0xd4 tx_macro_put_dec_enum+0x1d0/0x268 snd_ctl_elem_write+0x288/0x474 By Error checking and checking valid values issue gets rectifies. Signed-off-by: Ravulapati Vishnu Vardhan Rao <quic_visr(a)quicinc.com> --- sound/soc/codecs/lpass-tx-macro.c | 5 +++++ 1 file changed, 5 insertions(+) Changes Since v1: - Check value with enum item value getting from control. - Removing of previously added code which is unwanted as already we get out with -EINVAL if value is out of bound. diff --git a/sound/soc/codecs/lpass-tx-macro.c b/sound/soc/codecs/lpass-tx-macro.c index da6fcf7f0991..de978c3d70b7 100644 --- a/sound/soc/codecs/lpass-tx-macro.c +++ b/sound/soc/codecs/lpass-tx-macro.c @@ -746,6 +746,8 @@ static int tx_macro_put_dec_enum(struct snd_kcontrol *kcontrol, struct tx_macro *tx = snd_soc_component_get_drvdata(component); val = ucontrol->value.enumerated.item[0]; + if (val >= e->items) + return -EINVAL; switch (e->reg) { case CDC_TX_INP_MUX_ADC_MUX0_CFG0: @@ -772,6 +774,9 @@ static int tx_macro_put_dec_enum(struct snd_kcontrol *kcontrol, case CDC_TX_INP_MUX_ADC_MUX7_CFG0: mic_sel_reg = CDC_TX7_TX_PATH_CFG0; break; + default: + dev_err(component->dev, "Error in configuration!!\n"); + return -EINVAL; } if (val != 0) { -- 2.17.1

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[PATCH] ASoC: mediatek: mt8192-mt6359: Remove " Jack" from Headphone pin name
by AngeloGioacchino Del Regno 18 May '23

18 May '23

Function jack_kctl_name_gen() will remove the redundant " Jack" from the name, if present, and then it will add it back, so that all of the controls are named "(pin-name) Jack". Remove " Jack" from the Headphone pin name to spare some CPU cycles. This commit brings no functional changes. Signed-off-by: AngeloGioacchino Del Regno <angelogioacchino.delregno(a)collabora.com> --- sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c b/sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c index 5e163e23a207..4e0d5bf12b47 100644 --- a/sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c +++ b/sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c @@ -46,7 +46,7 @@ struct mt8192_mt6359_priv { /* Headset jack detection DAPM pins */ static struct snd_soc_jack_pin mt8192_jack_pins[] = { { - .pin = "Headphone Jack", + .pin = "Headphone", .mask = SND_JACK_HEADPHONE, }, { -- 2.40.1

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[PATCH 01/12] ASoC: codecs: rt1308: do not store status in state container
by Krzysztof Kozlowski 18 May '23

18 May '23

Driver in its update status callback stores Soundwire device status in state container but it never uses it later. Simplify the code a bit. Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- sound/soc/codecs/rt1308-sdw.c | 5 +---- sound/soc/codecs/rt1308-sdw.h | 1 - 2 files changed, 1 insertion(+), 5 deletions(-) diff --git a/sound/soc/codecs/rt1308-sdw.c b/sound/soc/codecs/rt1308-sdw.c index 1797af824f60..313e97c94532 100644 --- a/sound/soc/codecs/rt1308-sdw.c +++ b/sound/soc/codecs/rt1308-sdw.c @@ -304,9 +304,6 @@ static int rt1308_update_status(struct sdw_slave *slave, { struct rt1308_sdw_priv *rt1308 = dev_get_drvdata(&slave->dev); - /* Update the status */ - rt1308->status = status; - if (status == SDW_SLAVE_UNATTACHED) rt1308->hw_init = false; @@ -314,7 +311,7 @@ static int rt1308_update_status(struct sdw_slave *slave, * Perform initialization only if slave status is present and * hw_init flag is false */ - if (rt1308->hw_init || rt1308->status != SDW_SLAVE_ATTACHED) + if (rt1308->hw_init || status != SDW_SLAVE_ATTACHED) return 0; /* perform I/O transfers required for Slave initialization */ diff --git a/sound/soc/codecs/rt1308-sdw.h b/sound/soc/codecs/rt1308-sdw.h index 04ff18fa18e2..f816c73e247e 100644 --- a/sound/soc/codecs/rt1308-sdw.h +++ b/sound/soc/codecs/rt1308-sdw.h @@ -159,7 +159,6 @@ struct rt1308_sdw_priv { struct snd_soc_component *component; struct regmap *regmap; struct sdw_slave *sdw_slave; - enum sdw_slave_status status; struct sdw_bus_params params; bool hw_init; bool first_hw_init; -- 2.34.1

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[PATCH 0/2] ALSA: emu10k1: various improvements to the DSP-based mixer code
by Oswald Buddenhagen 18 May '23

18 May '23

Oswald Buddenhagen (2): ALSA: emu10k1: enable bit-exact playback, part 3: pitch ALSA: emu10k1: enable bit-exact playback, part 4: send amounts include/sound/emu10k1.h | 2 ++ sound/pci/emu10k1/emupcm.c | 34 ++++++++++++++++++++++++++++++---- 2 files changed, 32 insertions(+), 4 deletions(-) -- 2.40.0.152.g15d061e6df

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[PATCH 1/2] ALSA: emu10k1: fix synthesizer sample playback position and caching
by Oswald Buddenhagen 18 May '23

18 May '23

Compensate for the cache delay, and actually populate the cache. Without these, the playback would start with garbage (which would be (mostly?) masqueraded by the attack phase). Unlike for the PCM voices, this doesn't try to compensate for the interpolator read-ahead, because it's pointless to be super-exact here. Note that this code is probably still broken for particularly short samples, because we ignore the loop-related parts of CCR. But I'm not going to reverse-engineer that now ... Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen(a)gmx.de> --- sound/pci/emu10k1/emu10k1_callback.c | 36 ++++------------------------ 1 file changed, 5 insertions(+), 31 deletions(-) diff --git a/sound/pci/emu10k1/emu10k1_callback.c b/sound/pci/emu10k1/emu10k1_callback.c index 06440b97b5d7..aab8d64fd708 100644 --- a/sound/pci/emu10k1/emu10k1_callback.c +++ b/sound/pci/emu10k1/emu10k1_callback.c @@ -253,7 +253,7 @@ lookup_voices(struct snd_emux *emu, struct snd_emu10k1 *hw, /* check if sample is finished playing (non-looping only) */ if (bp != best + V_OFF && bp != best + V_FREE && (vp->reg.sample_mode & SNDRV_SFNT_SAMPLE_SINGLESHOT)) { - val = snd_emu10k1_ptr_read(hw, CCCA_CURRADDR, vp->ch); + val = snd_emu10k1_ptr_read(hw, CCCA_CURRADDR, vp->ch) - 64; if (val >= vp->reg.loopstart) bp = best + V_OFF; } @@ -360,7 +360,7 @@ start_voice(struct snd_emux_voice *vp) map = (hw->silent_page.addr << hw->address_mode) | (hw->address_mode ? MAP_PTI_MASK1 : MAP_PTI_MASK0); - addr = vp->reg.start; + addr = vp->reg.start + 64; temp = vp->reg.parm.filterQ; ccca = (temp << 28) | addr; if (vp->apitch < 0xe400) @@ -428,40 +428,14 @@ start_voice(struct snd_emux_voice *vp) /* Q & current address (Q 4bit value, MSB) */ CCCA, ccca, + /* cache */ + CCR, REG_VAL_PUT(CCR_CACHEINVALIDSIZE, 64), + /* reset volume */ VTFT, vtarget | vp->ftarget, CVCF, vtarget | CVCF_CURRENTFILTER_MASK, REGLIST_END); -#if 0 - /* cache */ - { - unsigned int val, sample; - val = 32; - if (vp->reg.sample_mode & SNDRV_SFNT_SAMPLE_8BITS) - sample = 0x80808080; - else { - sample = 0; - val *= 2; - } - - /* cache */ - snd_emu10k1_ptr_write(hw, CCR, ch, 0x1c << 16); - snd_emu10k1_ptr_write(hw, CDE, ch, sample); - snd_emu10k1_ptr_write(hw, CDF, ch, sample); - - /* invalidate maps */ - temp = ((unsigned int)hw->silent_page.addr << hw_address_mode) | (hw->address_mode ? MAP_PTI_MASK1 : MAP_PTI_MASK0); - snd_emu10k1_ptr_write(hw, MAPA, ch, temp); - snd_emu10k1_ptr_write(hw, MAPB, ch, temp); - - /* fill cache */ - val -= 4; - val <<= 25; - val |= 0x1c << 16; - snd_emu10k1_ptr_write(hw, CCR, ch, val); - } -#endif return 0; } -- 2.40.0.152.g15d061e6df

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