September 2019 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH -next] ASoC: sirf-audio: use devm_platform_ioremap_resource() to simplify code
by YueHaibing 04 Sep '19

04 Sep '19

Use devm_platform_ioremap_resource() to simplify the code a bit. This is detected by coccinelle. Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/soc/codecs/sirf-audio-codec.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/codecs/sirf-audio-codec.c b/sound/soc/codecs/sirf-audio-codec.c index 9009a74..a061d78 100644 --- a/sound/soc/codecs/sirf-audio-codec.c +++ b/sound/soc/codecs/sirf-audio-codec.c @@ -459,7 +459,6 @@ static int sirf_audio_codec_driver_probe(struct platform_device *pdev) int ret; struct sirf_audio_codec *sirf_audio_codec; void __iomem *base; - struct resource *mem_res; sirf_audio_codec = devm_kzalloc(&pdev->dev, sizeof(struct sirf_audio_codec), GFP_KERNEL); @@ -468,8 +467,7 @@ static int sirf_audio_codec_driver_probe(struct platform_device *pdev) platform_set_drvdata(pdev, sirf_audio_codec); - mem_res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - base = devm_ioremap_resource(&pdev->dev, mem_res); + base = devm_platform_ioremap_resource(pdev, 0); if (IS_ERR(base)) return PTR_ERR(base); -- 2.7.4

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[alsa-devel] [PATCH -next] ASoC: ep93xx: use devm_platform_ioremap_resource() to simplify code
by YueHaibing 04 Sep '19

04 Sep '19

Use devm_platform_ioremap_resource() to simplify the code a bit. This is detected by coccinelle. Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/soc/cirrus/ep93xx-ac97.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/cirrus/ep93xx-ac97.c b/sound/soc/cirrus/ep93xx-ac97.c index 84c967f..e21eaa1 100644 --- a/sound/soc/cirrus/ep93xx-ac97.c +++ b/sound/soc/cirrus/ep93xx-ac97.c @@ -362,7 +362,6 @@ static const struct snd_soc_component_driver ep93xx_ac97_component = { static int ep93xx_ac97_probe(struct platform_device *pdev) { struct ep93xx_ac97_info *info; - struct resource *res; int irq; int ret; @@ -370,8 +369,7 @@ static int ep93xx_ac97_probe(struct platform_device *pdev) if (!info) return -ENOMEM; - res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - info->regs = devm_ioremap_resource(&pdev->dev, res); + info->regs = devm_platform_ioremap_resource(pdev, 0); if (IS_ERR(info->regs)) return PTR_ERR(info->regs); -- 2.7.4

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[alsa-devel] [RFC PATCH 0/5] ASoC: SOF: Intel: SoundWire initial integration
by Pierre-Louis Bossart 04 Sep '19

04 Sep '19

This RFC is the companion of the other RFC 'soundwire: intel: simplify DAI/PDI handling'. Our purpose at this point is to gather feedback on the interfaces between the Intel SOF parts and the SoundWire code. The suggested solution is a simple init/release inserted at probe/remove and resume/suspend, as well as two callbacks for the SOF driver to generate IPC configurations with the firmware. That level of separation completely hides the details of the SoundWire DAIs and will allow for 'transparent' multi-cpu DAI support, which will be handled in the machine driver and the soundwire DAIs. This solution was tested on IceLake and CometLake, and captures the feedback from SOF contributors on an initial integration that was deemed too complicated (and rightly so). Pierre-Louis Bossart (5): ASoC: SOF: IPC: dai-intel: move ALH declarations in header file ASoC: SOF: Intel: hda: add helper to initialize SoundWire IP ASoC: SOF: Intel: hda: add SoundWire IP support ASoC: SOF: Intel: hda: add SoundWire stream config/free callbacks ASoC: SOF: Intel: add support for SoundWire suspend/resume include/sound/sof/dai-intel.h | 18 ++-- sound/soc/sof/intel/hda-dsp.c | 11 +++ sound/soc/sof/intel/hda.c | 157 ++++++++++++++++++++++++++++++++++ sound/soc/sof/intel/hda.h | 11 +++ 4 files changed, 188 insertions(+), 9 deletions(-) base-commit: 3b3aaa017e8072b1bfddda92be296b3463d870be -- 2.20.1

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[alsa-devel] [PATCH v3 1/4] ASoC: es8316: judge PCM rate at later timing
by Katsuhiro Suzuki 04 Sep '19

04 Sep '19

This patch change the judge timing about playing/capturing PCM rate. Original code set constraints list of PCM rate limits at set_sysclk. This strategy works well if system is using fixed rate clock. But some boards and SoC (such as RockPro64 and RockChip I2S) has connected SoC MCLK out to ES8316 MCLK in. In this case, SoC side I2S will choose suitable frequency of MCLK such as fs * mclk-fs when user starts playing or capturing. Bad scenario as follows (mclk-fs = 256): - Initialize sysclk by correct value (Ex. 12.288MHz) - ES8316 set constraints of PCM rate by sysclk 48kHz (1/256), 32kHz (1/384), 30.720kHz (1/400), 24kHz (1/512), 16kHz (1/768), 12kHz (1/1024) - Play 48kHz sound, it's acceptable - Sysclk is not changed - Play 32kHz sound, it's acceptable - Set sysclk by 8.192MHz (= fs * mclk-fs = 32k * 256) - ES8316 set constraints of PCM rate by sysclk 32kHz (1/256), 21.33kHz (1/384), 20.48kHz (1/400), 16kHz (1/512), 10.66kHz (1/768), 8kHz (1/1024) - Play 48kHz again, but it's NOT acceptable because constraints list does not allow 48kHz Root cause of this strange behavior is changing constraints list at set_sysclk timing. It seems that is too early to determine. So this patch does not use constraints list and check PCM rate limit more later timing at hw_params. Signed-off-by: Katsuhiro Suzuki <katsuhiro(a)katsuster.net> --- sound/soc/codecs/es8316.c | 37 +++++++++++++++---------------------- 1 file changed, 15 insertions(+), 22 deletions(-) diff --git a/sound/soc/codecs/es8316.c b/sound/soc/codecs/es8316.c index ed2959dbe1fb..229808fa627c 100644 --- a/sound/soc/codecs/es8316.c +++ b/sound/soc/codecs/es8316.c @@ -363,27 +363,12 @@ static int es8316_set_dai_sysclk(struct snd_soc_dai *codec_dai, { struct snd_soc_component *component = codec_dai->component; struct es8316_priv *es8316 = snd_soc_component_get_drvdata(component); - int i; - int count = 0; es8316->sysclk = freq; if (freq == 0) return 0; - /* Limit supported sample rates to ones that can be autodetected - * by the codec running in slave mode. - */ - for (i = 0; i < NR_SUPPORTED_MCLK_LRCK_RATIOS; i++) { - const unsigned int ratio = supported_mclk_lrck_ratios[i]; - - if (freq % ratio == 0) - es8316->allowed_rates[count++] = freq / ratio; - } - - es8316->sysclk_constraints.list = es8316->allowed_rates; - es8316->sysclk_constraints.count = count; - return 0; } @@ -449,13 +434,6 @@ static int es8316_pcm_startup(struct snd_pcm_substream *substream, return -EINVAL; } - /* The set of sample rates that can be supported depends on the - * MCLK supplied to the CODEC. - */ - snd_pcm_hw_constraint_list(substream->runtime, 0, - SNDRV_PCM_HW_PARAM_RATE, - &es8316->sysclk_constraints); - return 0; } @@ -466,12 +444,27 @@ static int es8316_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_soc_component *component = dai->component; struct es8316_priv *es8316 = snd_soc_component_get_drvdata(component); u8 wordlen = 0; + int i; if (!es8316->sysclk) { dev_err(component->dev, "No MCLK configured\n"); return -EINVAL; } + /* Limit supported sample rates to ones that can be autodetected + * by the codec running in slave mode. + */ + for (i = 0; i < NR_SUPPORTED_MCLK_LRCK_RATIOS; i++) { + const unsigned int ratio = supported_mclk_lrck_ratios[i]; + + if (es8316->sysclk % ratio != 0) + continue; + if (es8316->sysclk / ratio == params_rate(params)) + break; + } + if (i == NR_SUPPORTED_MCLK_LRCK_RATIOS) + return -EINVAL; + switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: wordlen = ES8316_SERDATA2_LEN_16; -- 2.23.0.rc1

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[alsa-devel] [PATCH] ALSA: hda/realtek - Fix the problem of two front mics on a ThinkCentre
by Hui Wang 04 Sep '19

04 Sep '19

This ThinkCentre machine has a new realtek codec alc222, it is not in the support list, we add it in the realtek.c then this machine can apply FIXUPs for the realtek codec. And this machine has two front mics which can't be handled by PA so far, it uses the pin 0x18 and 0x19 as the front mics, as a result the existing FIXUP ALC294_FIXUP_LENOVO_MIC_LOCATION doesn't work on this machine. Fortunately another FIXUP ALC283_FIXUP_HEADSET_MIC also can change the location for one of the two mics on this machine. Cc: <stable(a)vger.kernel.org> Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 3b9583aa63c1..74697c1dd1d5 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7151,6 +7151,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x312a, "ThinkCentre Station", ALC294_FIXUP_LENOVO_MIC_LOCATION), SND_PCI_QUIRK(0x17aa, 0x312f, "ThinkCentre Station", ALC294_FIXUP_LENOVO_MIC_LOCATION), SND_PCI_QUIRK(0x17aa, 0x313c, "ThinkCentre Station", ALC294_FIXUP_LENOVO_MIC_LOCATION), + SND_PCI_QUIRK(0x17aa, 0x3151, "ThinkCentre Station", ALC283_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x17aa, 0x3902, "Lenovo E50-80", ALC269_FIXUP_DMIC_THINKPAD_ACPI), SND_PCI_QUIRK(0x17aa, 0x3977, "IdeaPad S210", ALC283_FIXUP_INT_MIC), SND_PCI_QUIRK(0x17aa, 0x3978, "Lenovo B50-70", ALC269_FIXUP_DMIC_THINKPAD_ACPI), @@ -9038,6 +9039,7 @@ static int patch_alc680(struct hda_codec *codec) static const struct hda_device_id snd_hda_id_realtek[] = { HDA_CODEC_ENTRY(0x10ec0215, "ALC215", patch_alc269), HDA_CODEC_ENTRY(0x10ec0221, "ALC221", patch_alc269), + HDA_CODEC_ENTRY(0x10ec0222, "ALC222", patch_alc269), HDA_CODEC_ENTRY(0x10ec0225, "ALC225", patch_alc269), HDA_CODEC_ENTRY(0x10ec0231, "ALC231", patch_alc269), HDA_CODEC_ENTRY(0x10ec0233, "ALC233", patch_alc269), -- 2.17.1

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[alsa-devel] [PATCH] drm: bridge/dw_hdmi: add audio sample channel status setting
by Cheng-Yi Chiang 04 Sep '19

04 Sep '19

From: Yakir Yang <ykk(a)rock-chips.com> When transmitting IEC60985 linear PCM audio, we configure the Audio Sample Channel Status information of all the channel status bits in the IEC60958 frame. Refer to 60958-3 page 10 for frequency, original frequency, and wordlength setting. This fix the issue that audio does not come out on some monitors (e.g. LG 22CV241) Signed-off-by: Yakir Yang <ykk(a)rock-chips.com> Signed-off-by: Cheng-Yi Chiang <cychiang(a)chromium.org> --- drivers/gpu/drm/bridge/synopsys/dw-hdmi.c | 59 +++++++++++++++++++++++ drivers/gpu/drm/bridge/synopsys/dw-hdmi.h | 20 ++++++++ 2 files changed, 79 insertions(+) diff --git a/drivers/gpu/drm/bridge/synopsys/dw-hdmi.c b/drivers/gpu/drm/bridge/synopsys/dw-hdmi.c index bd65d0479683..34d46e25d610 100644 --- a/drivers/gpu/drm/bridge/synopsys/dw-hdmi.c +++ b/drivers/gpu/drm/bridge/synopsys/dw-hdmi.c @@ -582,6 +582,63 @@ static unsigned int hdmi_compute_n(unsigned int freq, unsigned long pixel_clk) return n; } +static void hdmi_set_schnl(struct dw_hdmi *hdmi) +{ + u8 aud_schnl_samplerate; + u8 aud_schnl_8; + + /* These registers are on RK3288 using version 2.0a. */ + if (hdmi->version != 0x200a) + return; + + switch (hdmi->sample_rate) { + case 32000: + aud_schnl_samplerate = HDMI_FC_AUDSCHNLS7_SMPRATE_32K; + break; + case 44100: + aud_schnl_samplerate = HDMI_FC_AUDSCHNLS7_SMPRATE_44K1; + break; + case 48000: + aud_schnl_samplerate = HDMI_FC_AUDSCHNLS7_SMPRATE_48K; + break; + case 88200: + aud_schnl_samplerate = HDMI_FC_AUDSCHNLS7_SMPRATE_88K2; + break; + case 96000: + aud_schnl_samplerate = HDMI_FC_AUDSCHNLS7_SMPRATE_96K; + break; + case 176400: + aud_schnl_samplerate = HDMI_FC_AUDSCHNLS7_SMPRATE_176K4; + break; + case 192000: + aud_schnl_samplerate = HDMI_FC_AUDSCHNLS7_SMPRATE_192K; + break; + case 768000: + aud_schnl_samplerate = HDMI_FC_AUDSCHNLS7_SMPRATE_768K; + break; + default: + dev_warn(hdmi->dev, "Unsupported audio sample rate (%u)\n", + hdmi->sample_rate); + return; + } + + /* set channel status register */ + hdmi_modb(hdmi, aud_schnl_samplerate, HDMI_FC_AUDSCHNLS7_SMPRATE_MASK, + HDMI_FC_AUDSCHNLS7); + + /* + * Set original frequency to be the same as frequency. + * Use one-complement value as stated in IEC60958-3 page 13. + */ + aud_schnl_8 = (~aud_schnl_samplerate) << + HDMI_FC_AUDSCHNLS8_ORIGSAMPFREQ_OFFSET; + + /* This means word length is 16 bit. Refer to IEC60958-3 page 12. */ + aud_schnl_8 |= 2 << HDMI_FC_AUDSCHNLS8_WORDLEGNTH_OFFSET; + + hdmi_writeb(hdmi, aud_schnl_8, HDMI_FC_AUDSCHNLS8); +} + static void hdmi_set_clk_regenerator(struct dw_hdmi *hdmi, unsigned long pixel_clk, unsigned int sample_rate) { @@ -620,6 +677,8 @@ static void hdmi_set_clk_regenerator(struct dw_hdmi *hdmi, hdmi->audio_cts = cts; hdmi_set_cts_n(hdmi, cts, hdmi->audio_enable ? n : 0); spin_unlock_irq(&hdmi->audio_lock); + + hdmi_set_schnl(hdmi); } static void hdmi_init_clk_regenerator(struct dw_hdmi *hdmi) diff --git a/drivers/gpu/drm/bridge/synopsys/dw-hdmi.h b/drivers/gpu/drm/bridge/synopsys/dw-hdmi.h index 6988f12d89d9..619ebc1c8354 100644 --- a/drivers/gpu/drm/bridge/synopsys/dw-hdmi.h +++ b/drivers/gpu/drm/bridge/synopsys/dw-hdmi.h @@ -158,6 +158,17 @@ #define HDMI_FC_SPDDEVICEINF 0x1062 #define HDMI_FC_AUDSCONF 0x1063 #define HDMI_FC_AUDSSTAT 0x1064 +#define HDMI_FC_AUDSV 0x1065 +#define HDMI_FC_AUDSU 0x1066 +#define HDMI_FC_AUDSCHNLS0 0x1067 +#define HDMI_FC_AUDSCHNLS1 0x1068 +#define HDMI_FC_AUDSCHNLS2 0x1069 +#define HDMI_FC_AUDSCHNLS3 0x106a +#define HDMI_FC_AUDSCHNLS4 0x106b +#define HDMI_FC_AUDSCHNLS5 0x106c +#define HDMI_FC_AUDSCHNLS6 0x106d +#define HDMI_FC_AUDSCHNLS7 0x106e +#define HDMI_FC_AUDSCHNLS8 0x106f #define HDMI_FC_DATACH0FILL 0x1070 #define HDMI_FC_DATACH1FILL 0x1071 #define HDMI_FC_DATACH2FILL 0x1072 @@ -706,6 +717,15 @@ enum { /* HDMI_FC_AUDSCHNLS7 field values */ HDMI_FC_AUDSCHNLS7_ACCURACY_OFFSET = 4, HDMI_FC_AUDSCHNLS7_ACCURACY_MASK = 0x30, + HDMI_FC_AUDSCHNLS7_SMPRATE_MASK = 0x0f, + HDMI_FC_AUDSCHNLS7_SMPRATE_192K = 0xe, + HDMI_FC_AUDSCHNLS7_SMPRATE_176K4 = 0xc, + HDMI_FC_AUDSCHNLS7_SMPRATE_96K = 0xa, + HDMI_FC_AUDSCHNLS7_SMPRATE_768K = 0x9, + HDMI_FC_AUDSCHNLS7_SMPRATE_88K2 = 0x8, + HDMI_FC_AUDSCHNLS7_SMPRATE_32K = 0x3, + HDMI_FC_AUDSCHNLS7_SMPRATE_48K = 0x2, + HDMI_FC_AUDSCHNLS7_SMPRATE_44K1 = 0x0, /* HDMI_FC_AUDSCHNLS8 field values */ HDMI_FC_AUDSCHNLS8_ORIGSAMPFREQ_MASK = 0xf0, -- 2.23.0.187.g17f5b7556c-goog

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[alsa-devel] [PATCH] soundwire: bus: set initial value to port_status
by Bard liao 04 Sep '19

04 Sep '19

From: Bard Liao <yung-chuan.liao(a)linux.intel.com> port_status[port_num] are assigned for each port_num in some if conditions. So some of the port_status may not be initialized. Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> --- drivers/soundwire/bus.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/drivers/soundwire/bus.c b/drivers/soundwire/bus.c index f6a1e4b4813d..33f41b3e642e 100644 --- a/drivers/soundwire/bus.c +++ b/drivers/soundwire/bus.c @@ -835,7 +835,7 @@ static int sdw_handle_port_interrupt(struct sdw_slave *slave, static int sdw_handle_slave_alerts(struct sdw_slave *slave) { struct sdw_slave_intr_status slave_intr; - u8 clear = 0, bit, port_status[15]; + u8 clear = 0, bit, port_status[15] = {0}; int port_num, stat, ret, count = 0; unsigned long port; bool slave_notify = false; -- 2.17.1

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[alsa-devel] some lock needed for flush_delayed_work ?
by Kuninori Morimoto 04 Sep '19

04 Sep '19

Hi ALSA ML I got below error when I unbind/bind CPU DAI. It seems we need to have some kind of lock ? I think => message came from linux/kernel/locking/lockdep.c :: assign_lock_key ... [ 6.392640] asoc-audio-graph-card sound: ASoC: CPU DAI not registered => [ 6.399430] INFO: trying to register non-static key. => [ 6.404393] the code is fine but needs lockdep annotation. => [ 6.409875] turning off the locking correctness validator. [ 6.415362] CPU: 2 PID: 193 Comm: test-bind-unbin Not tainted 5.3.0-rc1+ #625 [ 6.422495] Hardware name: Renesas H3ULCB Kingfisher board based on r8a7795 ES2.0+ (DT) [ 6.430497] Call trace: [ 6.432949] dump_backtrace+0x0/0x140 [ 6.436609] show_stack+0x24/0x30 [ 6.439927] dump_stack+0xc4/0x10c [ 6.443328] register_lock_class+0x4e0/0x618 [ 6.447594] __lock_acquire+0x7c/0x1de8 [ 6.451426] lock_acquire+0xdc/0x1e0 [ 6.455000] del_timer_sync+0x60/0xe8 [ 6.458660] flush_delayed_work+0x2c/0x70 [ 6.462671] soc_cleanup_card_resources+0x50/0x348 [ 6.467459] snd_soc_bind_card+0x1e4/0x990 [ 6.471551] snd_soc_add_component+0x248/0x310 [ 6.475992] snd_soc_register_component+0x54/0x70 [ 6.480696] devm_snd_soc_register_component+0x60/0xb8 [ 6.485833] rsnd_probe+0x164/0x448 [ 6.489321] platform_drv_probe+0x58/0xa8 [ 6.493330] really_probe+0xec/0x2a0 [ 6.496902] driver_probe_device+0x5c/0xf0 [ 6.500996] device_driver_attach+0x74/0x80 [ 6.505177] bind_store+0xdc/0x120 [ 6.508575] drv_attr_store+0x40/0x58 [ 6.512237] sysfs_kf_write+0x50/0x78 [ 6.515896] kernfs_fop_write+0xf0/0x1e8 [ 6.519819] __vfs_write+0x48/0x90 [ 6.523218] vfs_write+0xac/0x1b8 [ 6.526529] ksys_write+0x74/0xf8 [ 6.529840] __arm64_sys_write+0x24/0x30 [ 6.533762] el0_svc_common.constprop.0+0x98/0x170 [ 6.538550] el0_svc_compat_handler+0x2c/0x38 [ 6.542904] el0_svc_compat+0x8/0x10 Thank you for your help !! Best regards --- Kuninori Morimoto

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[alsa-devel] snd_usb_audio - ALSA woke us up to write...was actually nothing to write
by Nathan Royce 03 Sep '19

03 Sep '19

***** ... Sep 03 14:32:58 username kernel: perf: interrupt took too long (4174 > 4156), lowering kernel.perf_event_max_sample_rate to 47700 Sep 03 14:33:03 username plasmashell[1057]: qml: temp unit: 0 Sep 03 14:33:14 username org_kde_powerdevil[1089]: powerdevil: Releasing inhibition with cookie 14 Sep 03 14:33:14 username org_kde_powerdevil[1089]: powerdevil: Releasing inhibition with cookie 15 Sep 03 14:33:14 username org_kde_powerdevil[1089]: powerdevil: Restoring DPMS features after inhibition release Sep 03 14:33:14 username org_kde_powerdevil[1089]: powerdevil: Can't contact ck Sep 03 14:33:14 username org_kde_powerdevil[1089]: powerdevil: Releasing inhibition with cookie 16 Sep 03 14:33:14 username org_kde_powerdevil[1089]: powerdevil: Can't contact ck Sep 03 14:33:15 username plasmashell[1057]: qml: temp unit: 0 ... Sep 03 14:33:23 username pulseaudio[1081]: E: [alsa-sink-USB Audio] alsa-sink.c: ALSA woke us up to write new data to the device, but there was actually nothing to write. Sep 03 14:33:23 username pulseaudio[1081]: E: [alsa-sink-USB Audio] alsa-sink.c: Most likely this is a bug in the ALSA driver 'snd_usb_audio'. Please report this issue to the ALSA developers. Sep 03 14:33:23 username pulseaudio[1081]: E: [alsa-sink-USB Audio] alsa-sink.c: We were woken up with POLLOUT set -- however a subsequent snd_pcm_avail() returned 0 or another value < min_avail. Sep 03 14:33:24 username plasmashell[1057]: qml: temp unit: 0 ... ***** During this time period, I do believe I was starting an android emulator and also watching a twitter video that had audio.

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[alsa-devel] [PATCH] ASoC: soc-core: add comment to jack at soc_remove_component()
by Kuninori Morimoto 03 Sep '19

03 Sep '19

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Basically, driver which setups snd_soc_component_set_jack() need to release it by themselves. But, as framework level robustness, soc_remove_component() also releases it. To avoid code reader confuse, this patch makes it clarify. This patch makes it clarify. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/soc-core.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 659b90a..60f7e8b 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1171,7 +1171,9 @@ static void soc_remove_component(struct snd_soc_component *component) if (component->probed) snd_soc_component_remove(component); + /* For framework level robustness */ snd_soc_component_set_jack(component, NULL, NULL); + list_del(&component->card_list); snd_soc_dapm_free(snd_soc_component_get_dapm(component)); soc_cleanup_component_debugfs(component); -- 2.7.4

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