September 2017 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH v2] ALSA: hda - program ICT bits to support HBR audio
by Pierre-Louis Bossart 02 Oct '17

02 Oct '17

From: Sriram Periyasamy <sriramx.periyasamy(a)intel.com> On recent Intel platforms (Haswell, Broadwell, Skylake, ApolloLake, KabyLake, ...), the IEC Coding Type (ICT) bitfield in the Digital Converter Control #3 needs to be set explicitly for HDMI/DisplayPort High Bit Rate (HBR) audio playback to work. This was not required in earlier platforms when HBR was first introduced. The ICT bits are defined in Section 7.3.3.9 of the HDaudio 1.0a specification. Since the ICT bitfield was not specified for HDAudio 1.0 devices (before 2009), we only program it on machines more recent than Haswell. We tested that this fix is not needed on Baytrail-I (MinnowBoard Turbot) and believe by extension it also does not apply to Braswell. Bugzilla: https://bugs.freedesktop.org/show_bug.cgi?id=98797 Signed-off-by: Sriram Periyasamy <sriramx.periyasamy(a)intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty(a)intel.com> --- v2: move digi3 verb in Set section, only set ICT for haswell and newer include/sound/hda_verbs.h | 1 + sound/pci/hda/patch_hdmi.c | 21 +++++++++++++++++++++ 2 files changed, 22 insertions(+) diff --git a/include/sound/hda_verbs.h b/include/sound/hda_verbs.h index d0509db6d0ec..d2004d7feb2b 100644 --- a/include/sound/hda_verbs.h +++ b/include/sound/hda_verbs.h @@ -54,6 +54,7 @@ enum { #define AC_VERB_GET_EAPD_BTLENABLE 0x0f0c #define AC_VERB_GET_DIGI_CONVERT_1 0x0f0d #define AC_VERB_GET_DIGI_CONVERT_2 0x0f0e /* unused */ +#define AC_VERB_SET_DIGI_CONVERT_3 0x73e #define AC_VERB_GET_VOLUME_KNOB_CONTROL 0x0f0f /* f10-f1a: GPIO */ #define AC_VERB_GET_GPIO_DATA 0x0f15 diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 53f9311370de..8566de298ea3 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -906,6 +906,7 @@ static int hdmi_setup_stream(struct hda_codec *codec, hda_nid_t cvt_nid, hda_nid_t pin_nid, u32 stream_tag, int format) { struct hdmi_spec *spec = codec->spec; + unsigned int param; int err; err = spec->ops.pin_hbr_setup(codec, pin_nid, is_hbr_format(format)); @@ -915,6 +916,26 @@ static int hdmi_setup_stream(struct hda_codec *codec, hda_nid_t cvt_nid, return err; } + if (is_haswell_plus(codec)) { + + /* + * on recent platforms IEC Coding Type is required for HBR + * support, read current Digital Converter settings and set + * ICT bitfield if needed. + */ + param = snd_hda_codec_read(codec, cvt_nid, 0, + AC_VERB_GET_DIGI_CONVERT_1, 0); + + param = (param >> 16) & ~(AC_DIG3_ICT); + + /* on recent platforms ICT mode is required for HBR support */ + if (is_hbr_format(format)) + param |= 0x1; + + snd_hda_codec_write(codec, cvt_nid, 0, + AC_VERB_SET_DIGI_CONVERT_3, param); + } + snd_hda_codec_setup_stream(codec, cvt_nid, stream_tag, 0, format); return 0; } -- 2.11.0

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[alsa-devel] [PATCH 0/2] ASoC: soc-core: add missing component functions
by Kuninori Morimoto 02 Oct '17

02 Oct '17

Hi Mark These are resend of one of "replace to component" patch set. These add missing functions which exist in "platforms" but not exist in "component". Kuninori Morimoto (2): ASoC: soc-core: add component lookup functions ASoC: soc-core: add snd_soc_add_component() include/sound/soc.h | 7 ++++++ sound/soc/soc-core.c | 60 ++++++++++++++++++++++++++++++++++++++++++---------- 2 files changed, 56 insertions(+), 11 deletions(-) -- 1.9.1

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[alsa-devel] 4.14-rc2: audio stopped working?
by Pavel Machek 29 Sep '17

29 Sep '17

Hi! I've had sound problems in chromium for long long time. It produces choppy auduo just after boot and then it works, probably after chromium restart. May have something to do with pulseaudio. Not sure. Anyway... sound in flightgear usually worked, and now it does not. mplayer works, chromium works (sometimes choppy, as usual). Any ideas? Pavel -- (english) http://www.livejournal.com/~pavelmachek (cesky, pictures) http://atrey.karlin.mff.cuni.cz/~pavel/picture/horses/blog.html

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[alsa-devel] [RFC] ASoC: codecs: msm8916-wcd-analog: use btn0 released detection
by Damien Riegel 28 Sep '17

28 Sep '17

msm8916-wcd-analog uses button0 to differentiate between headphone and headset. Under some circumstances, button pressed and released interrupts are not fired as the driver expects it. For instance, with some connectors, there are spurious button-pressed interrupts when unplugging a headphone, without the corresponding button-released interrupt. But the codec always alternates between button pressed and released interrupts, it cannot fire two interrupts of the same kind in a row. That means that when the headphone is plugged back, only a button-released interrupt will be fired instead of pressed then released. This causes the driver to report headphone as headset. By changing the logic and relying on button 0 release interrupt, the driver could be made more robust for connectors that differ from the one used on the Dragonboard's audio mezzanine. Signed-off-by: Damien Riegel <damien.riegel(a)savoirfairelinux.com> --- sound/soc/codecs/msm8916-wcd-analog.c | 18 ++++++++---------- 1 file changed, 8 insertions(+), 10 deletions(-) diff --git a/sound/soc/codecs/msm8916-wcd-analog.c b/sound/soc/codecs/msm8916-wcd-analog.c index 549c269acc7d..f562f2d86907 100644 --- a/sound/soc/codecs/msm8916-wcd-analog.c +++ b/sound/soc/codecs/msm8916-wcd-analog.c @@ -285,7 +285,7 @@ struct pm8916_wcd_analog_priv { u16 codec_version; bool mbhc_btn_enabled; /* special event to detect accessory type */ - bool mbhc_btn0_pressed; + int mbhc_btn0_released; bool detect_accessory_type; struct clk *mclk; struct snd_soc_codec *codec; @@ -483,7 +483,7 @@ static void pm8916_wcd_setup_mbhc(struct pm8916_wcd_analog_priv *wcd) snd_soc_update_bits(codec, CDC_D_INT_EN_CLR, int_en_mask, 0); snd_soc_update_bits(codec, CDC_D_INT_EN_SET, int_en_mask, int_en_mask); - wcd->mbhc_btn0_pressed = false; + wcd->mbhc_btn0_released = false; wcd->detect_accessory_type = true; } @@ -950,7 +950,7 @@ static irqreturn_t mbhc_btn_release_irq_handler(int irq, void *arg) /* check if its BTN0 thats released */ if ((val != -1) && !(val & CDC_A_MBHC_RESULT_1_BTN_RESULT_MASK)) - priv->mbhc_btn0_pressed = false; + priv->mbhc_btn0_released = true; } else { snd_soc_jack_report(priv->jack, 0, btn_mask); @@ -983,9 +983,7 @@ static irqreturn_t mbhc_btn_press_irq_handler(int irq, void *arg) break; case 0x0: /* handle BTN_0 specially for type detection */ - if (priv->detect_accessory_type) - priv->mbhc_btn0_pressed = true; - else + if (!priv->detect_accessory_type) snd_soc_jack_report(priv->jack, SND_JACK_BTN_0, btn_mask); break; @@ -1029,19 +1027,19 @@ static irqreturn_t pm8916_mbhc_switch_irq_handler(int irq, void *arg) * both press and release event received then its * a headset. */ - if (priv->mbhc_btn0_pressed) + if (priv->mbhc_btn0_released) snd_soc_jack_report(priv->jack, - SND_JACK_HEADPHONE, hs_jack_mask); + SND_JACK_HEADSET, hs_jack_mask); else snd_soc_jack_report(priv->jack, - SND_JACK_HEADSET, hs_jack_mask); + SND_JACK_HEADPHONE, hs_jack_mask); priv->detect_accessory_type = false; } else { /* removal */ snd_soc_jack_report(priv->jack, 0, hs_jack_mask); priv->detect_accessory_type = true; - priv->mbhc_btn0_pressed = false; + priv->mbhc_btn0_released = false; } return IRQ_HANDLED; -- 2.14.1

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[alsa-devel] [PATCH] ASoC: Intel: Skylake: Fix jack name format substitution
by Chintan Patel 28 Sep '17

28 Sep '17

Jack name is not getting formatted correctly hence resulting in invalid name for HDMI/DP input devices. This was recently exposed due changes brought by MST: commit 3a13347f05fd ("ASoC: Intel: kbl: Add jack port initialize in kbl machine drivers") Signed-off-by: Chintan Patel <chintan.m.patel(a)intel.com> --- sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c index 4689991bcc86..c127b7b0ee3f 100644 --- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c @@ -611,6 +611,8 @@ static int kabylake_card_late_probe(struct snd_soc_card *card) list_for_each_entry(pcm, &ctx->hdmi_pcm_list, head) { codec = pcm->codec_dai->codec; + snprintf(jack_name, sizeof(jack_name), + "HDMI/DP, pcm=%d Jack", pcm->device); err = snd_soc_card_jack_new(card, jack_name, SND_JACK_AVOUT, &ctx->kabylake_hdmi[i], NULL, 0); -- 1.9.1

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[alsa-devel] [PATCH v3] ASoC: rt5670: add set_bclk_ratio in dai ops
by Bard Liao 28 Sep '17

28 Sep '17

We need to set a specific bit for 50 bclk rate. So add set_bclk_ratio function to set the bit. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- v2: * add define for tdm data format select bit. v3: * change if (ratio == 50) to if ((ratio % 50) == 0) since the bit needs to be set if bclk ratio is divisible by 50. --- sound/soc/codecs/rt5670.c | 19 +++++++++++++++++++ sound/soc/codecs/rt5670.h | 4 ++++ 2 files changed, 23 insertions(+) diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c index 9545764..78517d8 100644 --- a/sound/soc/codecs/rt5670.c +++ b/sound/soc/codecs/rt5670.c @@ -2582,6 +2582,24 @@ static int rt5670_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, return 0; } +static int rt5670_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio) +{ + struct snd_soc_codec *codec = dai->codec; + + dev_dbg(codec->dev, "%s ratio=%d\n", __func__, ratio); + if (dai->id != RT5670_AIF1) + return 0; + + if ((ratio % 50) == 0) + snd_soc_update_bits(codec, RT5670_GEN_CTRL3, + RT5670_TDM_DATA_MODE_SEL, RT5670_TDM_DATA_MODE_50FS); + else + snd_soc_update_bits(codec, RT5670_GEN_CTRL3, + RT5670_TDM_DATA_MODE_SEL, RT5670_TDM_DATA_MODE_NOR); + + return 0; +} + static int rt5670_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { @@ -2712,6 +2730,7 @@ static const struct snd_soc_dai_ops rt5670_aif_dai_ops = { .set_fmt = rt5670_set_dai_fmt, .set_tdm_slot = rt5670_set_tdm_slot, .set_pll = rt5670_set_dai_pll, + .set_bclk_ratio = rt5670_set_bclk_ratio, }; static struct snd_soc_dai_driver rt5670_dai[] = { diff --git a/sound/soc/codecs/rt5670.h b/sound/soc/codecs/rt5670.h index 5ba485c..265df80 100644 --- a/sound/soc/codecs/rt5670.h +++ b/sound/soc/codecs/rt5670.h @@ -1816,6 +1816,10 @@ #define RT5670_ZCD_HP_DIS (0x0 << 15) #define RT5670_ZCD_HP_EN (0x1 << 15) +/* General Control 3 (0xfc) */ +#define RT5670_TDM_DATA_MODE_SEL (0x1 << 11) +#define RT5670_TDM_DATA_MODE_NOR (0x0 << 11) +#define RT5670_TDM_DATA_MODE_50FS (0x1 << 11) /* Codec Private Register definition */ /* 3D Speaker Control (0x63) */ -- 2.7.4

2 1

[alsa-devel] [PATCH v2] ASoC: dapm: Avoid creating kcontrol for params
by yesanishhere＠gmail.com 28 Sep '17

28 Sep '17

From: anish kumar <yesanishhere(a)gmail.com> Currently in codec to codec dai link if there are multiple params defined then dapm can use created kcontrol to decide which param to apply at runtime. However, in case there is only single param configuration then there is no point in creating the kcontrol and also there is no point in allocating memory for kcontrol. In the snd_soc_dapm_new_pcm function, there is memory allocation happening for kcontrol which is later used or not used based on num_param. It is better to not allocate memory when there is only a single configuration. This change is to remedy that anomaly. Signed-off-by: anish kumar <yesanishhere(a)gmail.com> --- Changes in this version: - Review comments from charles to move common function of error handling in seperate function. Changes in v1: - Included the review comments from charles regarding releasing memory in error path. sound/soc/soc-dapm.c | 145 +++++++++++++++++++++++++++++++-------------------- 1 file changed, 89 insertions(+), 56 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index d55cac6..5dae9d8 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3778,18 +3778,27 @@ static int snd_soc_dapm_dai_link_put(struct snd_kcontrol *kcontrol, return 0; } -int snd_soc_dapm_new_pcm(struct snd_soc_card *card, - const struct snd_soc_pcm_stream *params, - unsigned int num_params, - struct snd_soc_dapm_widget *source, - struct snd_soc_dapm_widget *sink) +void +snd_soc_dapm_free_kcontrol(struct snd_soc_card *card, + unsigned long *private_value, + int num_params, + const char **w_param_text) +{ + int count; + + devm_kfree(card->dev, (void *)*private_value); + for (count = 0 ; count < num_params; count++) + devm_kfree(card->dev, (void *)w_param_text[count]); + devm_kfree(card->dev, w_param_text); +} + +static struct snd_kcontrol_new * +snd_soc_dapm_alloc_kcontrol(struct snd_soc_card *card, + char *link_name, + const struct snd_soc_pcm_stream *params, + int num_params, const char **w_param_text, + unsigned long *private_value) { - struct snd_soc_dapm_widget template; - struct snd_soc_dapm_widget *w; - char *link_name; - int ret, count; - unsigned long private_value; - const char **w_param_text; struct soc_enum w_param_enum[] = { SOC_ENUM_SINGLE(0, 0, 0, NULL), }; @@ -3798,19 +3807,9 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card, snd_soc_dapm_dai_link_get, snd_soc_dapm_dai_link_put), }; + struct snd_kcontrol_new *kcontrol_news; const struct snd_soc_pcm_stream *config = params; - - w_param_text = devm_kcalloc(card->dev, num_params, - sizeof(char *), GFP_KERNEL); - if (!w_param_text) - return -ENOMEM; - - link_name = devm_kasprintf(card->dev, GFP_KERNEL, "%s-%s", - source->name, sink->name); - if (!link_name) { - ret = -ENOMEM; - goto outfree_w_param; - } + int count; for (count = 0 ; count < num_params; count++) { if (!config->stream_name) { @@ -3827,47 +3826,87 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card, strlen(config->stream_name) + 1, GFP_KERNEL); } - if (!w_param_text[count]) { - ret = -ENOMEM; - goto outfree_link_name; - } + if (!w_param_text[count]) + goto outfree_w_param; config++; } + w_param_enum[0].items = num_params; w_param_enum[0].texts = w_param_text; - memset(&template, 0, sizeof(template)); - template.reg = SND_SOC_NOPM; - template.id = snd_soc_dapm_dai_link; - template.name = link_name; - template.event = snd_soc_dai_link_event; - template.event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | - SND_SOC_DAPM_PRE_PMD; - template.num_kcontrols = 1; - /* duplicate w_param_enum on heap so that memory persists */ - private_value = + *private_value = (unsigned long) devm_kmemdup(card->dev, (void *)(kcontrol_dai_link[0].private_value), sizeof(struct soc_enum), GFP_KERNEL); - if (!private_value) { + if (!*private_value) { dev_err(card->dev, "ASoC: Failed to create control for %s widget\n", link_name); - ret = -ENOMEM; - goto outfree_link_name; + goto outfree_w_param; } - kcontrol_dai_link[0].private_value = private_value; + kcontrol_dai_link[0].private_value = *private_value; /* duplicate kcontrol_dai_link on heap so that memory persists */ - template.kcontrol_news = - devm_kmemdup(card->dev, &kcontrol_dai_link[0], - sizeof(struct snd_kcontrol_new), - GFP_KERNEL); - if (!template.kcontrol_news) { + kcontrol_news = + devm_kmemdup(card->dev, &kcontrol_dai_link[0], + sizeof(struct snd_kcontrol_new), + GFP_KERNEL); + if (!kcontrol_news) { dev_err(card->dev, "ASoC: Failed to create control for %s widget\n", link_name); - ret = -ENOMEM; - goto outfree_private_value; + goto outfree_w_param; } + return kcontrol_news; +outfree_w_param: + snd_soc_dapm_free_kcontrol(card, private_value, num_params, w_param_text); + return NULL; +} + +int snd_soc_dapm_new_pcm(struct snd_soc_card *card, + const struct snd_soc_pcm_stream *params, + unsigned int num_params, + struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink) +{ + struct snd_soc_dapm_widget template; + struct snd_soc_dapm_widget *w; + const char **w_param_text; + unsigned long private_value; + char *link_name; + int ret; + + link_name = devm_kasprintf(card->dev, GFP_KERNEL, "%s-%s", + source->name, sink->name); + if (!link_name) + return -ENOMEM; + + memset(&template, 0, sizeof(template)); + template.reg = SND_SOC_NOPM; + template.id = snd_soc_dapm_dai_link; + template.name = link_name; + template.event = snd_soc_dai_link_event; + template.event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_PRE_PMD; + template.kcontrol_news = NULL; + + /* allocate memory for control, only in case of multiple configs */ + if (num_params > 1) { + w_param_text = devm_kcalloc(card->dev, num_params, + sizeof(char *), GFP_KERNEL); + if (!w_param_text) { + ret = -ENOMEM; + goto param_fail; + } + + template.num_kcontrols = 1; + template.kcontrol_news = + snd_soc_dapm_alloc_kcontrol(card, + link_name, params, num_params, + w_param_text, &private_value); + if (!template.kcontrol_news) { + ret = -ENOMEM; + goto param_fail; + } + } dev_dbg(card->dev, "ASoC: adding %s widget\n", link_name); w = snd_soc_dapm_new_control_unlocked(&card->dapm, &template); @@ -3899,15 +3938,9 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card, devm_kfree(card->dev, w); outfree_kcontrol_news: devm_kfree(card->dev, (void *)template.kcontrol_news); -outfree_private_value: - devm_kfree(card->dev, (void *)private_value); -outfree_link_name: + snd_soc_dapm_free_kcontrol(card, &private_value, num_params, w_param_text); +param_fail: devm_kfree(card->dev, link_name); - for (count = 0 ; count < num_params; count++) - devm_kfree(card->dev, (void *)w_param_text[count]); -outfree_w_param: - devm_kfree(card->dev, w_param_text); - return ret; } -- 2.5.4 (Apple Git-61)

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[alsa-devel] [PATCH] ASoC: kirkwood: make kirkwood_soc_platform const
by Bhumika Goyal 27 Sep '17

27 Sep '17

Make kirkwood_soc_platform const as it only passed to a const argument of the function snd_soc_register_platform in the file referencing it. Make the declaration const too. Done using Coccinelle. Signed-off-by: Bhumika Goyal <bhumirks(a)gmail.com> --- sound/soc/kirkwood/kirkwood-dma.c | 2 +- sound/soc/kirkwood/kirkwood.h | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/kirkwood/kirkwood-dma.c b/sound/soc/kirkwood/kirkwood-dma.c index cf23af1..505b0ff 100644 --- a/sound/soc/kirkwood/kirkwood-dma.c +++ b/sound/soc/kirkwood/kirkwood-dma.c @@ -318,7 +318,7 @@ static void kirkwood_dma_free_dma_buffers(struct snd_pcm *pcm) } } -struct snd_soc_platform_driver kirkwood_soc_platform = { +const struct snd_soc_platform_driver kirkwood_soc_platform = { .ops = &kirkwood_dma_ops, .pcm_new = kirkwood_dma_new, .pcm_free = kirkwood_dma_free_dma_buffers, diff --git a/sound/soc/kirkwood/kirkwood.h b/sound/soc/kirkwood/kirkwood.h index 90e32a7..783cb1a4 100644 --- a/sound/soc/kirkwood/kirkwood.h +++ b/sound/soc/kirkwood/kirkwood.h @@ -143,6 +143,6 @@ struct kirkwood_dma_data { int burst; }; -extern struct snd_soc_platform_driver kirkwood_soc_platform; +extern const struct snd_soc_platform_driver kirkwood_soc_platform; #endif -- 1.9.1

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[alsa-devel] [PATCH] ASoC: Intel: Kabylake machine driver changes for a FE DAI
by Harsha Priya 27 Sep '17

27 Sep '17

'Kbl Audio Headset Playback' FE DAI also needs SSP hw parama fix. Adding this dai also to be handled in kabylake_ssp_fixup() call. Signed-off-by: Harsha Priya <harshapriya.n(a)intel.com> --- sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c index 88ff542..e767283 100644 --- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c @@ -302,6 +302,7 @@ static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd, * The ADSP will convert the FE rate to 48k, stereo, 24 bit */ if (!strcmp(fe_dai_link->name, "Kbl Audio Port") || + !strcmp(fe_dai_link->name, "Kbl Audio Headset Playback") || !strcmp(fe_dai_link->name, "Kbl Audio Capture Port")) { rate->min = rate->max = 48000; channels->min = channels->max = 2; -- 1.9.1

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[alsa-devel] [PATCH] ASoC: pcm: Sync delayed work before releasing resources
by Takashi Iwai 27 Sep '17

27 Sep '17

When ASoC driver is unbound dynamically during its operation (i.e. a kind of hot-unplug), we may hit Oops due to the resource access after the release by a delayed work, something like: Unable to handle kernel paging request at virtual address dead000000000220 .... PC is at soc_dapm_dai_stream_event.isra.14+0x20/0xd0 LR is at snd_soc_dapm_stream_event+0x74/0xa8 .... [<ffff000008715610>] soc_dapm_dai_stream_event.isra.14+0x20/0xd0 [<ffff00000871989c>] snd_soc_dapm_stream_event+0x74/0xa8 [<ffff00000871b23c>] close_delayed_work+0x3c/0x50 [<ffff0000080bbd6c>] process_one_work+0x1ac/0x318 [<ffff0000080bbf20>] worker_thread+0x48/0x420 [<ffff0000080c201c>] kthread+0xfc/0x128 [<ffff0000080842f0>] ret_from_fork+0x10/0x18 For fixing the race, this patch adds a sync-point in pcm private_free callback to finish the delayed work before actually releasing the resources. Reported-by: Hiep Cao Minh <cm-hiep(a)jinso.co.jp> Reported-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Tested-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/soc/soc-pcm.c | 13 ++++++++++++- 1 file changed, 12 insertions(+), 1 deletion(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 94b88b897c3b..c0f0b09cb433 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2632,6 +2632,17 @@ static int dpcm_fe_dai_close(struct snd_pcm_substream *fe_substream) return ret; } +static void soc_pcm_private_free(struct snd_pcm *pcm) +{ + struct snd_soc_pcm_runtime *rtd = pcm->private_data; + struct snd_soc_platform *platform = rtd->platform; + + /* need to sync the delayed work before releasing resources */ + flush_delayed_work(&rtd->delayed_work); + if (platform->driver->pcm_free) + platform->driver->pcm_free(pcm); +} + /* create a new pcm */ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) { @@ -2757,7 +2768,7 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) } } - pcm->private_free = platform->driver->pcm_free; + pcm->private_free = soc_pcm_private_free; out: dev_info(rtd->card->dev, "%s <-> %s mapping ok\n", (rtd->num_codecs > 1) ? "multicodec" : rtd->codec_dai->name, -- 2.14.0

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