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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

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