December 2020 - Alsa-devel - mailman.alsa-project.org

[PATCH] ASoC: imx-hdmi: Fix warning of the uninitialized variable ret
by Shengjiu Wang 12 Jan '21

12 Jan '21

From: shengjiu wang <shengjiu.wang(a)nxp.com> When condition ((hdmi_out && hdmi_in) || (!hdmi_out && !hdmi_in)) is true, then goto fail, the uninitialized variable ret will be returned. Signed-off-by: shengjiu wang <shengjiu.wang(a)nxp.com> Reported-by: kernel test robot <lkp(a)intel.com> --- sound/soc/fsl/imx-hdmi.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/fsl/imx-hdmi.c b/sound/soc/fsl/imx-hdmi.c index 2c2a76a71940..ede4a9ad1054 100644 --- a/sound/soc/fsl/imx-hdmi.c +++ b/sound/soc/fsl/imx-hdmi.c @@ -164,6 +164,7 @@ static int imx_hdmi_probe(struct platform_device *pdev) if ((hdmi_out && hdmi_in) || (!hdmi_out && !hdmi_in)) { dev_err(&pdev->dev, "Invalid HDMI DAI link\n"); + ret = -EINVAL; goto fail; } -- 2.17.1

5 7

SoF support for 8086:9d71?
by Kai-Heng Feng 12 Jan '21

12 Jan '21

Hi, On more recent kernels legacy HDA stops working for device 8086:9d71: [ 47.211381] snd_hda_intel 0000:00:1f.3: DSP detected with PCI class/subclass/prog-if info 0x040100 [ 47.211383] snd_hda_intel 0000:00:1f.3: Digital mics found on Skylake+ platform, using SST driver However, only legacy HDA supports it: 00:1f.3 Multimedia audio controller [0401]: Intel Corporation Sunrise Point-LP HD Audio [8086:9d71] (rev 21) .. Kernel modules: snd_hda_intel Ubuntu kernel doesn't select SND_SOC_INTEL_SKYLAKE, which builds snd-soc-skl, because based on the Kconfig description SOF should support it. I wonder if SND_SOC_INTEL_SKYLAKE should be select again? BugLink: <https://bugs.launchpad.net/bugs/1902457>

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[PATCH] ASoC: Intel: bytcr_rt5640: Add quirks for DeeQ X5-Z8300
by Chris Chiu 11 Jan '21

11 Jan '21

DeeQ X5-Z8300 is a laptop empowered by Intel Atom Cherry Trail. Add quirks to select the correct input map, jack-detect options to enable jack sensing and internal/headset microphones. Signed-off-by: Chris Chiu <chiu(a)endlessos.org> --- sound/soc/intel/boards/bytcr_rt5651.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/soc/intel/boards/bytcr_rt5651.c b/sound/soc/intel/boards/bytcr_rt5651.c index 688b5e0a49e3..9df09e26b05a 100644 --- a/sound/soc/intel/boards/bytcr_rt5651.c +++ b/sound/soc/intel/boards/bytcr_rt5651.c @@ -507,6 +507,14 @@ static const struct dmi_system_id byt_rt5651_quirk_table[] = { BYT_RT5651_SSP0_AIF1 | BYT_RT5651_MONO_SPEAKER), }, + { + .callback = byt_rt5651_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "DeeQ"), + DMI_MATCH(DMI_PRODUCT_NAME, "X5-Z8300"), + }, + .driver_data = (void *)(BYT_RT5651_IN1_IN2_MAP | BYT_RT5651_JD1_1), + }, {} }; -- 2.20.1

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[PATCH] ASoC: audio-graph-card: Drop remote-endpoint as required property
by Sameer Pujar 11 Jan '21

11 Jan '21

The remote-endpoint may not be available if it is part of some pluggable module. One such example would be an audio card, the Codec endpoint will not be available until it is plugged in. Hence drop 'remote-endpoint' as a required property. Cc: Rob Herring <robh(a)kernel.org> Cc: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Sameer Pujar <spujar(a)nvidia.com> --- Documentation/devicetree/bindings/sound/audio-graph-port.yaml | 3 --- 1 file changed, 3 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/audio-graph-port.yaml b/Documentation/devicetree/bindings/sound/audio-graph-port.yaml index 2005014..766e910 100644 --- a/Documentation/devicetree/bindings/sound/audio-graph-port.yaml +++ b/Documentation/devicetree/bindings/sound/audio-graph-port.yaml @@ -71,9 +71,6 @@ properties: description: CPU to Codec rate channels. $ref: /schemas/types.yaml#/definitions/uint32 - required: - - remote-endpoint - ports: description: multi OF-Graph subnode type: object -- 2.7.4

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[PATCH 0/2] fix tegra-hda on tegra30 devices
by Peter Geis 08 Jan '21

08 Jan '21

The following patches fix tegra-hda on legacy tegra devices. Two issues were discovered preventing tegra-hda from functioning: The hda clocks on tegra30 were assigned to clk_m and running at too low of a rate to function. The tegra-hda encounters an input/output error when opening a stream. Since the only mainline device that used tegra-hda previously was the t124, it is unknown exactly when this was broken. Fortunately a recent patch was submitted that fixed the issue only on t194 devices. We can apply it universally to the tegra-hda device to resolve the issues across the board. Note that downstream devices used the spdif device instead of hda for hdmi audio. The spdif device lacks a driver on mainline. Peter Geis (2): clk: tegra30: Add hda clock default rates to clock driver ALSA: hda/tegra: fix tegra-hda on tegra30 soc drivers/clk/tegra/clk-tegra30.c | 2 ++ sound/pci/hda/hda_tegra.c | 3 +-- 2 files changed, 3 insertions(+), 2 deletions(-) -- 2.25.1

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ASoC: soc-pcm: disconnect BEs if the FE is not ready
by 朱灿灿 07 Jan '21

07 Jan '21

FE is connected to two BEs, BE1 is active, BE2 is deactive. When closing BE1, FE/BE1 is in HW_FREE state, then BE2 is startup by mixer runtime update. For FE is in HW_FREE state, dpcm_run_update_startup() will skip BE2's startup because FE's state is HW_FREE, BE2 stays in FE's be_clients list. During FE's closed, the dpcm_fe_dai_close() will close all related BEs, BE2 will be closed. This will lead to BE2's dpcm[stream].users mismatch. We need disconnet all pending BEs in the corner case. Signed-off-by: zhucancan <zhucancan(a)vivo.com> --- sound/soc/soc-pcm.c | 19 ++++++++++++++----- 1 file changed, 14 insertions(+), 5 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index dcab9527ba3d..04ad53f0ae4f 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2451,8 +2451,11 @@ static int dpcm_run_update_startup(struct snd_soc_pcm_runtime *fe, int stream) /* Only start the BE if the FE is ready */ if (fe->dpcm[stream].state == SND_SOC_DPCM_STATE_HW_FREE || - fe->dpcm[stream].state == SND_SOC_DPCM_STATE_CLOSE) - return -EINVAL; + fe->dpcm[stream].state == SND_SOC_DPCM_STATE_CLOSE) { + dev_err(fe->dev, "ASoC: FE %s is not ready %d\n", + fe->dai_link->name, fe->dpcm[stream].state); + goto disconnect; + } /* startup must always be called for new BEs */ ret = dpcm_be_dai_startup(fe, stream); @@ -2513,12 +2516,18 @@ static int dpcm_run_update_startup(struct snd_soc_pcm_runtime *fe, int stream) close: dpcm_be_dai_shutdown(fe, stream); disconnect: - /* disconnect any closed BEs */ + /* disconnect any pending BEs */ spin_lock_irqsave(&fe->card->dpcm_lock, flags); for_each_dpcm_be(fe, stream, dpcm) { struct snd_soc_pcm_runtime *be = dpcm->be; - if (be->dpcm[stream].state == SND_SOC_DPCM_STATE_CLOSE) - dpcm->state = SND_SOC_DPCM_LINK_STATE_FREE; + + /* is this op for this BE ? */ + if (!snd_soc_dpcm_be_can_update(fe, be, stream)) + continue; + + if (be->dpcm[stream].state == SND_SOC_DPCM_STATE_CLOSE || + be->dpcm[stream].state == SND_SOC_DPCM_STATE_NEW) + dpcm->state = SND_SOC_DPCM_LINK_STATE_FREE; } spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); -- 2.21.0

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[PATCH 00/41] USB audio refactoring for better implicit feedback support
by Takashi Iwai 07 Jan '21

07 Jan '21

Hi, here is a largish patch set for USB-audio. Its main goal is to address the long-standing issues wrt the implicit feedback mode. With the current driver, when the implicit fb is used, the full duplex doesn't work properly unless you start both streams *very* carefully. This patch set tries to ease such the configuration mismatch and fix many other bugs. In short, with this patch set applied, USB-audio driver will try to restrict the hw_params to match with the already opened counter-part stream automatically. So individual aplay and arecord invocation should work now, as well as PulseAudio. Also, the patch set includes more fixes, e.g. the generic detection of the implicit feedback mode on UAC2/3, and the fix for the sample rate selection on UAC2/3, as well as a few more quirks to follow the new implicit fb model. The latest patches are available in topic/usb-audio-refactoring branch. I specially thank Keith Milner and Dylan Robinson for their time spent for lots of testing. Without their patient help, I couldn't finish those patches, as I myself don't own such devices. Takashi === Dylan Robinson (1): ALSA: usb-audio: Fix MOTU M-Series quirks Takashi Iwai (40): ALSA: usb-audio: Handle discrete rates properly in hw constraints ALSA: usb-audio: Don't call usb_set_interface() at trigger callback ALSA: usb-audio: Check valid altsetting at parsing rates for UAC2/3 ALSA: usb-audio: Check implicit feedback EP generically for UAC2 ALSA: usb-audio: Add snd_usb_get_endpoint() helper ALSA: usb-audio: Set and clear sync EP link properly ALSA: usb-audio: Improve some debug prints ALSA: usb-audio: Track implicit fb sync endpoint in audioformat list ALSA: usb-audio: Move snd_usb_autoresume() call out of setup_hw_info() ALSA: usb-audio: Add hw constraint for implicit fb sync ALSA: usb-audio: Simplify hw_params rules ALSA: usb-audio: Drop debug.h ALSA: usb-audio: Avoid doubly initialization for implicit fb ALSA: usb-audio: Create endpoint objects at parsing phase ALSA: usb-audio: Drop keep_interface flag again ALSA: usb-audio: Add snd_usb_get_host_interface() helper ALSA: usb-audio: Don't set altsetting before initializing sample rate ALSA: usb-audio: Pass snd_usb_audio object to quirk functions ALSA: usb-audio: Simplify snd_usb_init_sample_rate() arguments ALSA: usb-audio: Simplify snd_usb_init_pitch() arguments ALSA: usb-audio: Stop both endpoints properly at error ALSA: usb-audio: Set callbacks via snd_usb_endpoint_set_callback() ALSA: usb-audio: Always set up the parameters after resume ALSA: usb-audio: Fix EP matching for continuous rates ALSA: usb-audio: Refactor endpoint management ALSA: usb-audio: Fix possible stall of implicit fb packet ring-buffer ALSA: usb-audio: Constify audioformat pointer references ALSA: usb-audio: Use atomic_t for endpoint use_count ALSA: usb-audio: Refactoring endpoint URB deactivation ALSA: usb-audio: Drop unneeded snd_usb_substream fields ALSA: usb-audio: Unify the code for the next packet size calculation ALSA: usb-audio: Simplify rate_min/max and rates set up ALSA: usb-audio: Replace slave/master terms ALSA: usb-audio: Use unsigned char for iface and altsettings fields ALSA: usb-audio: Show sync endpoint information in proc outputs ALSA: usb-audio: Quirk for BOSS GT-001 ALSA: usb-audio: Factor out the implicit feedback quirk code ALSA: usb-audio: Add generic implicit fb parsing ALSA: usb-audio: Add implicit_fb module option ALSA: usb-audio: Fix quirks for other BOSS devices Documentation/sound/alsa-configuration.rst | 5 + sound/usb/Makefile | 1 + sound/usb/card.c | 14 +- sound/usb/card.h | 53 +- sound/usb/clock.c | 152 ++-- sound/usb/clock.h | 11 +- sound/usb/debug.h | 16 - sound/usb/endpoint.c | 943 +++++++++++++---------- sound/usb/endpoint.h | 57 +- sound/usb/format.c | 125 +++- sound/usb/helper.c | 10 + sound/usb/helper.h | 3 + sound/usb/implicit.c | 402 ++++++++++ sound/usb/implicit.h | 14 + sound/usb/mixer.c | 46 -- sound/usb/pcm.c | 1117 ++++++++++------------------ sound/usb/pcm.h | 7 +- sound/usb/proc.c | 35 +- sound/usb/quirks-table.h | 8 - sound/usb/quirks.c | 60 +- sound/usb/quirks.h | 10 +- sound/usb/stream.c | 24 +- sound/usb/usbaudio.h | 5 +- 23 files changed, 1696 insertions(+), 1422 deletions(-) delete mode 100644 sound/usb/debug.h create mode 100644 sound/usb/implicit.c create mode 100644 sound/usb/implicit.h -- 2.16.4

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[PATCH] ASoC: rt715: add main capture switch and main capture volume control
by jack.yu＠realtek.com 06 Jan '21

06 Jan '21

From: Jack Yu <jack.yu(a)realtek.com> Add main capture switch and main capture volume for callback to be in single operation. Signed-off-by: Jack Yu <jack.yu(a)realtek.com> --- sound/soc/codecs/rt715.c | 174 +++++++++++++++++++++++---------------- 1 file changed, 105 insertions(+), 69 deletions(-) diff --git a/sound/soc/codecs/rt715.c b/sound/soc/codecs/rt715.c index 9a7d393d424a..cdcba70146da 100644 --- a/sound/soc/codecs/rt715.c +++ b/sound/soc/codecs/rt715.c @@ -80,85 +80,106 @@ static int rt715_set_amp_gain_put(struct snd_kcontrol *kcontrol, struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; struct rt715_priv *rt715 = snd_soc_component_get_drvdata(component); + unsigned int capture_reg_H[] = {RT715_SET_GAIN_MIC_ADC_H, + RT715_SET_GAIN_LINE_ADC_H, RT715_SET_GAIN_MIX_ADC_H, + RT715_SET_GAIN_MIX_ADC2_H}; + unsigned int capture_reg_L[] = {RT715_SET_GAIN_MIC_ADC_L, + RT715_SET_GAIN_LINE_ADC_L, RT715_SET_GAIN_MIX_ADC_L, + RT715_SET_GAIN_MIX_ADC2_L}; unsigned int addr_h, addr_l, val_h, val_ll, val_lr; - unsigned int read_ll, read_rl; - int i; - - /* Can't use update bit function, so read the original value first */ - addr_h = mc->reg; - addr_l = mc->rreg; - if (mc->shift == RT715_DIR_OUT_SFT) /* output */ - val_h = 0x80; - else /* input */ - val_h = 0x0; - - rt715_get_gain(rt715, addr_h, addr_l, val_h, &read_rl, &read_ll); + unsigned int read_ll, read_rl, i, j, loop_cnt; - /* L Channel */ - if (mc->invert) { - /* for mute */ - val_ll = (mc->max - ucontrol->value.integer.value[0]) << 7; - /* keep gain */ - read_ll = read_ll & 0x7f; - val_ll |= read_ll; - } else { - /* for gain */ - val_ll = ((ucontrol->value.integer.value[0]) & 0x7f); - if (val_ll > mc->max) - val_ll = mc->max; - /* keep mute status */ - read_ll = read_ll & 0x80; - val_ll |= read_ll; - } - - /* R Channel */ - if (mc->invert) { - regmap_write(rt715->regmap, - RT715_SET_AUDIO_POWER_STATE, AC_PWRST_D0); - /* for mute */ - val_lr = (mc->max - ucontrol->value.integer.value[1]) << 7; - /* keep gain */ - read_rl = read_rl & 0x7f; - val_lr |= read_rl; - } else { - /* for gain */ - val_lr = ((ucontrol->value.integer.value[1]) & 0x7f); - if (val_lr > mc->max) - val_lr = mc->max; - /* keep mute status */ - read_rl = read_rl & 0x80; - val_lr |= read_rl; - } - - for (i = 0; i < 3; i++) { /* retry 3 times at most */ + if (strstr(ucontrol->id.name, "Main Capture Switch") || + strstr(ucontrol->id.name, "Main Capture Volume")) + loop_cnt = 4; + else + loop_cnt = 1; - if (val_ll == val_lr) { - /* Set both L/R channels at the same time */ - val_h = (1 << mc->shift) | (3 << 4); - regmap_write(rt715->regmap, addr_h, - (val_h << 8 | val_ll)); - regmap_write(rt715->regmap, addr_l, - (val_h << 8 | val_ll)); + for (j = 0; j < loop_cnt; j++) { + /* Can't use update bit function, so read the original value first */ + if (loop_cnt == 1) { + addr_h = mc->reg; + addr_l = mc->rreg; } else { - /* Lch*/ - val_h = (1 << mc->shift) | (1 << 5); - regmap_write(rt715->regmap, addr_h, - (val_h << 8 | val_ll)); - /* Rch */ - val_h = (1 << mc->shift) | (1 << 4); - regmap_write(rt715->regmap, addr_l, - (val_h << 8 | val_lr)); + addr_h = capture_reg_H[j]; + addr_l = capture_reg_L[j]; } - /* check result */ + if (mc->shift == RT715_DIR_OUT_SFT) /* output */ val_h = 0x80; else /* input */ val_h = 0x0; - rt715_get_gain(rt715, addr_h, addr_l, val_h, - &read_rl, &read_ll); - if (read_rl == val_lr && read_ll == val_ll) - break; + rt715_get_gain(rt715, addr_h, addr_l, val_h, &read_rl, &read_ll); + + if (dapm->bias_level <= SND_SOC_BIAS_STANDBY) + regmap_write(rt715->regmap, + RT715_SET_AUDIO_POWER_STATE, AC_PWRST_D0); + + /* L Channel */ + if (mc->invert) { + /* for mute */ + val_ll = (mc->max - ucontrol->value.integer.value[0]) << 7; + /* keep gain */ + read_ll = read_ll & 0x7f; + val_ll |= read_ll; + } else { + /* for gain */ + val_ll = ((ucontrol->value.integer.value[0]) & 0x7f); + if (val_ll > mc->max) + val_ll = mc->max; + /* keep mute status */ + read_ll = read_ll & 0x80; + val_ll |= read_ll; + } + + /* R Channel */ + if (mc->invert) { + /* for mute */ + val_lr = (mc->max - ucontrol->value.integer.value[1]) << 7; + /* keep gain */ + read_rl = read_rl & 0x7f; + val_lr |= read_rl; + } else { + /* for gain */ + val_lr = ((ucontrol->value.integer.value[1]) & 0x7f); + if (val_lr > mc->max) + val_lr = mc->max; + /* keep mute status */ + read_rl = read_rl & 0x80; + val_lr |= read_rl; + } + + for (i = 0; i < 3; i++) { /* retry 3 times at most */ + + if (val_ll == val_lr) { + /* Set both L/R channels at the same time */ + val_h = (1 << mc->shift) | (3 << 4); + regmap_write(rt715->regmap, addr_h, + (val_h << 8 | val_ll)); + regmap_write(rt715->regmap, addr_l, + (val_h << 8 | val_ll)); + } else { + /* Lch*/ + val_h = (1 << mc->shift) | (1 << 5); + regmap_write(rt715->regmap, addr_h, + (val_h << 8 | val_ll)); + /* Rch */ + val_h = (1 << mc->shift) | (1 << 4); + regmap_write(rt715->regmap, addr_l, + (val_h << 8 | val_lr)); + } + /* check result */ + if (mc->shift == RT715_DIR_OUT_SFT) /* output */ + val_h = 0x80; + else /* input */ + val_h = 0x0; + + rt715_get_gain(rt715, addr_h, addr_l, val_h, + &read_rl, &read_ll); + if (read_rl == val_lr && read_ll == val_ll) + break; + } } /* D0:power on state, D3: power saving mode */ if (dapm->bias_level <= SND_SOC_BIAS_STANDBY) @@ -226,6 +247,13 @@ static const struct snd_kcontrol_new rt715_snd_controls[] = { SOC_DOUBLE_R_EXT("ADC 27 Capture Switch", RT715_SET_GAIN_MIX_ADC2_H, RT715_SET_GAIN_MIX_ADC2_L, RT715_DIR_IN_SFT, 1, 1, rt715_set_amp_gain_get, rt715_set_amp_gain_put), + /* + * "Main Capture Switch" looks the same as "ADC 07 Capture Switch", + * but the callback has different action internally. + */ + SOC_DOUBLE_R_EXT("Main Capture Switch", RT715_SET_GAIN_MIC_ADC_H, + RT715_SET_GAIN_MIC_ADC_L, RT715_DIR_IN_SFT, 1, 1, + rt715_set_amp_gain_get, rt715_set_amp_gain_put), /* Volume Control */ SOC_DOUBLE_R_EXT_TLV("ADC 07 Capture Volume", RT715_SET_GAIN_MIC_ADC_H, RT715_SET_GAIN_MIC_ADC_L, RT715_DIR_IN_SFT, 0x3f, 0, @@ -243,6 +271,14 @@ static const struct snd_kcontrol_new rt715_snd_controls[] = { RT715_SET_GAIN_MIX_ADC2_L, RT715_DIR_IN_SFT, 0x3f, 0, rt715_set_amp_gain_get, rt715_set_amp_gain_put, in_vol_tlv), + /* + * "Main Capture Volume" looks the same as "ADC 07 Capture Volume", + * but the callback has different action internally. + */ + SOC_DOUBLE_R_EXT_TLV("Main Capture Volume", RT715_SET_GAIN_MIC_ADC_H, + RT715_SET_GAIN_MIC_ADC_L, RT715_DIR_IN_SFT, 0x3f, 0, + rt715_set_amp_gain_get, rt715_set_amp_gain_put, + in_vol_tlv), /* MIC Boost Control */ SOC_DOUBLE_R_EXT_TLV("DMIC1 Boost", RT715_SET_GAIN_DMIC1_H, RT715_SET_GAIN_DMIC1_L, RT715_DIR_IN_SFT, 3, 0, -- 2.29.0

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[PATCH v2 0/9] soundwire/regmap: use _no_pm routines
by Bard Liao 06 Jan '21

06 Jan '21

When a Slave device is resumed, it may resume the bus and restart the enumeration. And Slave drivers will wait for initialization_complete complete in their resume function, however initialization_complete will complete after sdw_update_slave_status function is finished and codec driver usually call some IO functions in the update_status callback function. It will become a deadlock if we use regular read/write routines during the resuming process. This series touches both soundwire and regmap trees. commit fb5103f9d6ce ("regmap/SoundWire: sdw: add support for SoundWire 1.2 MBQ") is needed for soundwire tree to complie. On the other hands, commit 6e06a85556f9 ("soundwire: bus: add comments to explain interrupt loop filter") to commit 47b8520997a8 ("soundwire: bus: only clear valid DPN interrupts") are needed for regmap tree. v2: - Separate commits according to maintainer's comments. Bard Liao (4): soundwire: export sdw_write/read_no_pm functions regmap: sdw: use _no_pm functions in regmap_read/write regmap: sdw: use no_pm routines for SoundWire 1.2 MBQ regmap: sdw-mbq: use MODULE_LICENSE("GPL") Pierre-Louis Bossart (5): soundwire: bus: use sdw_update_no_pm when initializing a device soundwire: bus: use sdw_write_no_pm when setting the bus scale registers soundwire: bus: use no_pm IO routines for all interrupt handling soundwire: bus: fix confusion on device used by pm_runtime soundwire: bus: clarify dev_err/dbg device references drivers/base/regmap/regmap-sdw-mbq.c | 10 +- drivers/base/regmap/regmap-sdw.c | 4 +- drivers/soundwire/bus.c | 136 +++++++++++++++------------ include/linux/soundwire/sdw.h | 2 + 4 files changed, 85 insertions(+), 67 deletions(-) -- 2.17.1

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[PATCH] ASoC: rt1015: re-calibrate again when resuming
by Tzung-Bi Shih 05 Jan '21

05 Jan '21

Assuming the calibration state gets lost after system suspend. Re-calibrates again when resuming. The rt1015_priv is alloced by kzalloc. No need to initialize cali_done to 0 in component probe callback. Signed-off-by: Tzung-Bi Shih <tzungbi(a)google.com> --- The idea is inspired from previous discussion: https://mailman.alsa-project.org/pipermail/alsa-devel/2020-December/178229.… The patch applies after the series: https://mailman.alsa-project.org/pipermail/alsa-devel/2020-December/178608.… v1 attempt: https://mailman.alsa-project.org/pipermail/alsa-devel/2020-December/178614.… sound/soc/codecs/rt1015.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt1015.c b/sound/soc/codecs/rt1015.c index 4a9e2eaebe30..3b06eaddccaf 100644 --- a/sound/soc/codecs/rt1015.c +++ b/sound/soc/codecs/rt1015.c @@ -1055,7 +1055,6 @@ static int rt1015_probe(struct snd_soc_component *component) rt1015->component = component; rt1015->bclk_ratio = 0; - rt1015->cali_done = 0; INIT_DELAYED_WORK(&rt1015->flush_work, rt1015_flush_work); @@ -1113,6 +1112,10 @@ static int rt1015_resume(struct snd_soc_component *component) regcache_cache_only(rt1015->regmap, false); regcache_sync(rt1015->regmap); + + if (rt1015->cali_done) + rt1015_calibrate(rt1015); + return 0; } #else -- 2.29.2.729.g45daf8777d-goog

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