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

[alsa-devel] [RFC PATCH v2] ASoC: dpcm: prevent snd_soc_dpcm use after free
by KaiChieh Chuang 06 Mar '19

06 Mar '19

the dpcm get from fe_clients/be_clients may be free before use possible race condition between void dpcm_be_disconnect( ... list_del(&dpcm->list_be); list_del(&dpcm->list_fe); kfree(dpcm); ... and for_each_dpcm_fe() for_each_dpcm_be*() race condition example Thread 1: snd_soc_dapm_mixer_update_power() -> soc_dpcm_runtime_update() -> dpcm_be_disconnect() -> kfree(dpcm); Thread 2: dpcm_fe_dai_trigger() -> dpcm_be_dai_trigger() -> snd_soc_dpcm_can_be_free_stop() -> if (dpcm->fe == fe) Excpetion Scenario: two FE link to same BE FE1 -> BE FE2 -> Thread 1: switch off mixer between FE2 -> BE Thread 2: pcm_stop FE1 Add a spin lock at snd_soc_card level, to protect the dpcm instance. The lock may be used in atomic context, so use spin lock. Exception: Unable to handle kernel paging request at virtual address dead0000000000e0 pc=<> [<ffffff8960e2cd10>] dpcm_be_dai_trigger+0x29c/0x47c sound/soc/soc-pcm.c:3226 if (dpcm->fe == fe) lr=<> [<ffffff8960e2f694>] dpcm_fe_dai_do_trigger+0x94/0x26c Backtrace: [<ffffff89602dba80>] notify_die+0x68/0xb8 [<ffffff896028c7dc>] die+0x118/0x2a8 [<ffffff89602a2f84>] __do_kernel_fault+0x13c/0x14c [<ffffff89602a27f4>] do_translation_fault+0x64/0xa0 [<ffffff8960280cf8>] do_mem_abort+0x4c/0xd0 [<ffffff8960282ad0>] el1_da+0x24/0x40 [<ffffff8960e2cd10>] dpcm_be_dai_trigger+0x29c/0x47c [<ffffff8960e2f694>] dpcm_fe_dai_do_trigger+0x94/0x26c [<ffffff8960e2edec>] dpcm_fe_dai_trigger+0x3c/0x44 [<ffffff8960de5588>] snd_pcm_do_stop+0x50/0x5c [<ffffff8960dded24>] snd_pcm_action+0xb4/0x13c [<ffffff8960ddfdb4>] snd_pcm_drop+0xa0/0x128 [<ffffff8960de69bc>] snd_pcm_common_ioctl+0x9d8/0x30f0 [<ffffff8960de1cac>] snd_pcm_ioctl_compat+0x29c/0x2f14 [<ffffff89604c9d60>] compat_SyS_ioctl+0x128/0x244 [<ffffff8960283740>] el0_svc_naked+0x34/0x38 [<ffffffffffffffff>] 0xffffffffffffffff Signed-off-by: KaiChieh Chuang <kaichieh.chuang(a)mediatek.com> --- include/sound/soc.h | 2 ++ sound/soc/soc-core.c | 1 + sound/soc/soc-pcm.c | 31 ++++++++++++++++++++++++------- 3 files changed, 27 insertions(+), 7 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index eb7db605955b..1e2be35ed36f 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1083,6 +1083,8 @@ struct snd_soc_card { struct mutex mutex; struct mutex dapm_mutex; + spinlock_t dpcm_lock; + bool instantiated; bool topology_shortname_created; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 5a5764dba147..d88757659729 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2820,6 +2820,7 @@ int snd_soc_register_card(struct snd_soc_card *card) card->instantiated = 0; mutex_init(&card->mutex); mutex_init(&card->dapm_mutex); + spin_lock_init(&card->dpcm_lock); return snd_soc_bind_card(card); } diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index a5b40e82dea4..f31eaf58e750 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1294,9 +1294,11 @@ void dpcm_be_disconnect(struct snd_soc_pcm_runtime *fe, int stream) #ifdef CONFIG_DEBUG_FS debugfs_remove(dpcm->debugfs_state); #endif + spin_lock(&fe->card->dpcm_lock); list_del(&dpcm->list_be); list_del(&dpcm->list_fe); kfree(dpcm); + spin_unlock(&fe->card->dpcm_lock); } } @@ -1548,9 +1550,11 @@ void dpcm_clear_pending_state(struct snd_soc_pcm_runtime *fe, int stream) { struct snd_soc_dpcm *dpcm; + spin_lock(&fe->card->dpcm_lock); for_each_dpcm_be(fe, stream, dpcm) dpcm->be->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_NO; + spin_unlock(&fe->card->dpcm_lock); } static void dpcm_be_dai_startup_unwind(struct snd_soc_pcm_runtime *fe, @@ -2640,11 +2644,13 @@ static int dpcm_run_update_startup(struct snd_soc_pcm_runtime *fe, int stream) dpcm_be_dai_shutdown(fe, stream); disconnect: /* disconnect any non started BEs */ + spin_lock(&fe->card->dpcm_lock); for_each_dpcm_be(fe, stream, dpcm) { struct snd_soc_pcm_runtime *be = dpcm->be; if (be->dpcm[stream].state != SND_SOC_DPCM_STATE_START) dpcm->state = SND_SOC_DPCM_LINK_STATE_FREE; } + spin_unlock(&fe->card->dpcm_lock); return ret; } @@ -3220,7 +3226,9 @@ int snd_soc_dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe, { struct snd_soc_dpcm *dpcm; int state; + int ret = 1; + spin_lock(&fe->card->dpcm_lock); for_each_dpcm_fe(be, stream, dpcm) { if (dpcm->fe == fe) @@ -3229,12 +3237,15 @@ int snd_soc_dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe, state = dpcm->fe->dpcm[stream].state; if (state == SND_SOC_DPCM_STATE_START || state == SND_SOC_DPCM_STATE_PAUSED || - state == SND_SOC_DPCM_STATE_SUSPEND) - return 0; + state == SND_SOC_DPCM_STATE_SUSPEND) { + ret = 0; + break; + } } + spin_unlock(&fe->card->dpcm_lock); /* it's safe to free/stop this BE DAI */ - return 1; + return ret; } EXPORT_SYMBOL_GPL(snd_soc_dpcm_can_be_free_stop); @@ -3247,7 +3258,9 @@ int snd_soc_dpcm_can_be_params(struct snd_soc_pcm_runtime *fe, { struct snd_soc_dpcm *dpcm; int state; + int ret = 1; + spin_lock(&fe->card->dpcm_lock); for_each_dpcm_fe(be, stream, dpcm) { if (dpcm->fe == fe) @@ -3257,12 +3270,15 @@ int snd_soc_dpcm_can_be_params(struct snd_soc_pcm_runtime *fe, if (state == SND_SOC_DPCM_STATE_START || state == SND_SOC_DPCM_STATE_PAUSED || state == SND_SOC_DPCM_STATE_SUSPEND || - state == SND_SOC_DPCM_STATE_PREPARE) - return 0; + state == SND_SOC_DPCM_STATE_PREPARE) { + ret = 0 + break; + } } + spin_unlock(&fe->card->dpcm_lock); /* it's safe to change hw_params */ - return 1; + return ret; } EXPORT_SYMBOL_GPL(snd_soc_dpcm_can_be_params); @@ -3328,6 +3344,7 @@ static ssize_t dpcm_show_state(struct snd_soc_pcm_runtime *fe, goto out; } + spin_lock(&fe->card->dpcm_lock); for_each_dpcm_be(fe, stream, dpcm) { struct snd_soc_pcm_runtime *be = dpcm->be; params = &dpcm->hw_params; @@ -3348,7 +3365,7 @@ static ssize_t dpcm_show_state(struct snd_soc_pcm_runtime *fe, params_channels(params), params_rate(params)); } - + spin_unlock(&fe->card->dpcm_lock); out: return offset; } -- 2.18.0

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[alsa-devel] [PATCH] ALSA: hda: set the 2nd headphone's name to be Headphone2
by Hui Wang 06 Mar '19

06 Mar '19

Recently I saw two Headphones on some HP machines, the 1st Headphone comes from the Front Headphone Jack, the 2nd one comes from the Front Headset Jack, and the current HDA driver will set the same name to them and with different index, just like below: 'Headphone Playback Volume' 'Headphone Playback Switch' 'Headphone Playback Volume' index 1 'Headphone Playback Switch' index 1 'Front Headphone Jack' 'Front Headphone Jack' index 1 This can't be handled by pulseaudio, and pulseaduio already has a path analog-output-headphones-2.conf to handle the 2nd headphone, now we set the 2nd headphone's name to "Headphone2", leave the other headphones unchanged. The change of hda_auto_parser.c will change the name of Jack. The change of hda_generic.c will change the name of Volume and Switch. Cc: <stable(a)vger.kernel.org> Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- The alsa-info.txt before applying this patch: https://pastebin.ubuntu.com/p/CrPrHRykgT/ The alsa-info.txt after applying this patch: https://pastebin.ubuntu.com/p/9VbkyvMBkW/ sound/pci/hda/hda_auto_parser.c | 11 +++++++++-- sound/pci/hda/hda_generic.c | 2 ++ 2 files changed, 11 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/hda_auto_parser.c b/sound/pci/hda/hda_auto_parser.c index df0d636145f8..748c62a9c5eb 100644 --- a/sound/pci/hda/hda_auto_parser.c +++ b/sound/pci/hda/hda_auto_parser.c @@ -714,6 +714,8 @@ int snd_hda_get_pin_label(struct hda_codec *codec, hda_nid_t nid, unsigned int def_conf = snd_hda_codec_get_pincfg(codec, nid); const char *name = NULL; int i; + int ret; + int index; bool hdmi; if (indexp) @@ -729,8 +731,13 @@ int snd_hda_get_pin_label(struct hda_codec *codec, hda_nid_t nid, return fill_audio_out_name(codec, nid, cfg, "Speaker", label, maxlen, indexp); case AC_JACK_HP_OUT: - return fill_audio_out_name(codec, nid, cfg, "Headphone", - label, maxlen, indexp); + ret = fill_audio_out_name(codec, nid, cfg, "Headphone", + label, maxlen, &index); + if (index == 1) + snprintf(label, maxlen, "%s", "Headphone2"); + if (indexp) + *indexp = index; + return ret; case AC_JACK_SPDIF_OUT: case AC_JACK_DIG_OTHER_OUT: hdmi = is_hdmi_cfg(def_conf); diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 4095cd7c56c6..2a34e8a6e2e1 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -2175,6 +2175,8 @@ static int create_extra_outs(struct hda_codec *codec, int num_pins, if (num_pins == 2 && i == 1 && !strcmp(pfx, "Speaker")) name = "Bass Speaker"; + else if (num_pins == 2 && i == 1 && !strcmp(pfx, "Headphone")) + name = "Headphone2"; else if (num_pins >= 3) { snprintf(tmp, sizeof(tmp), "%s %s", pfx, channel_name[i]); -- 2.17.1

2 4

[alsa-devel] [RFC PATCH] ASoC: dpcm: prevent snd_soc_dpcm use after free
by KaiChieh Chuang 06 Mar '19

06 Mar '19

the dpcm get from fe_clients/be_clients may be free before use possible race condition between void dpcm_be_disconnect( ... list_del(&dpcm->list_be); list_del(&dpcm->list_fe); kfree(dpcm); ... and for_each_dpcm_fe() for_each_dpcm_be*() race condition example Thread 1: snd_soc_dapm_mixer_update_power() -> soc_dpcm_runtime_update() -> dpcm_be_disconnect() -> kfree(dpcm); Thread 2: dpcm_fe_dai_trigger() -> dpcm_be_dai_trigger() -> snd_soc_dpcm_can_be_free_stop() -> if (dpcm->fe == fe) Excpetion Scenario: two FE link to same BE FE1 -> BE FE2 -> Thread 1: switch of mixer between FE2 -> BE Thread 2: pcm_stop FE1 Add a spin lock at snd_soc_card level, to protect the dpcm instance. The lock may be used in atomic context, so use spin lock. Exception: Unable to handle kernel paging request at virtual address dead0000000000e0 pc=<> [<ffffff8960e2cd10>] dpcm_be_dai_trigger+0x29c/0x47c sound/soc/soc-pcm.c:3226 if (dpcm->fe == fe) lr=<> [<ffffff8960e2f694>] dpcm_fe_dai_do_trigger+0x94/0x26c Backtrace: [<ffffff89602dba80>] notify_die+0x68/0xb8 [<ffffff896028c7dc>] die+0x118/0x2a8 [<ffffff89602a2f84>] __do_kernel_fault+0x13c/0x14c [<ffffff89602a27f4>] do_translation_fault+0x64/0xa0 [<ffffff8960280cf8>] do_mem_abort+0x4c/0xd0 [<ffffff8960282ad0>] el1_da+0x24/0x40 [<ffffff8960e2cd10>] dpcm_be_dai_trigger+0x29c/0x47c [<ffffff8960e2f694>] dpcm_fe_dai_do_trigger+0x94/0x26c [<ffffff8960e2edec>] dpcm_fe_dai_trigger+0x3c/0x44 [<ffffff8960de5588>] snd_pcm_do_stop+0x50/0x5c [<ffffff8960dded24>] snd_pcm_action+0xb4/0x13c [<ffffff8960ddfdb4>] snd_pcm_drop+0xa0/0x128 [<ffffff8960de69bc>] snd_pcm_common_ioctl+0x9d8/0x30f0 [<ffffff8960de1cac>] snd_pcm_ioctl_compat+0x29c/0x2f14 [<ffffff89604c9d60>] compat_SyS_ioctl+0x128/0x244 [<ffffff8960283740>] el0_svc_naked+0x34/0x38 [<ffffffffffffffff>] 0xffffffffffffffff Signed-off-by: KaiChieh Chuang <kaichieh.chuang(a)mediatek.com> --- include/sound/soc.h | 2 ++ sound/soc/soc-core.c | 1 + sound/soc/soc-pcm.c | 21 ++++++++++++++++++--- 3 files changed, 21 insertions(+), 3 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index eb7db605955b..1e2be35ed36f 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1083,6 +1083,8 @@ struct snd_soc_card { struct mutex mutex; struct mutex dapm_mutex; + spinlock_t dpcm_lock; + bool instantiated; bool topology_shortname_created; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 5a5764dba147..d88757659729 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2820,6 +2820,7 @@ int snd_soc_register_card(struct snd_soc_card *card) card->instantiated = 0; mutex_init(&card->mutex); mutex_init(&card->dapm_mutex); + spin_lock_init(&card->dpcm_lock); return snd_soc_bind_card(card); } diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index a5b40e82dea4..ac3f773ca768 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1294,9 +1294,11 @@ void dpcm_be_disconnect(struct snd_soc_pcm_runtime *fe, int stream) #ifdef CONFIG_DEBUG_FS debugfs_remove(dpcm->debugfs_state); #endif + spin_lock(&fe->card->dpcm_lock); list_del(&dpcm->list_be); list_del(&dpcm->list_fe); kfree(dpcm); + spin_unlock(&fe->card->dpcm_lock); } } @@ -1548,9 +1550,11 @@ void dpcm_clear_pending_state(struct snd_soc_pcm_runtime *fe, int stream) { struct snd_soc_dpcm *dpcm; + spin_lock(&fe->card->dpcm_lock); for_each_dpcm_be(fe, stream, dpcm) dpcm->be->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_NO; + spin_unlock(&fe->card->dpcm_lock); } static void dpcm_be_dai_startup_unwind(struct snd_soc_pcm_runtime *fe, @@ -2640,11 +2644,13 @@ static int dpcm_run_update_startup(struct snd_soc_pcm_runtime *fe, int stream) dpcm_be_dai_shutdown(fe, stream); disconnect: /* disconnect any non started BEs */ + spin_lock(&fe->card->dpcm_lock); for_each_dpcm_be(fe, stream, dpcm) { struct snd_soc_pcm_runtime *be = dpcm->be; if (be->dpcm[stream].state != SND_SOC_DPCM_STATE_START) dpcm->state = SND_SOC_DPCM_LINK_STATE_FREE; } + spin_unlock(&fe->card->dpcm_lock); return ret; } @@ -3221,6 +3227,7 @@ int snd_soc_dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe, struct snd_soc_dpcm *dpcm; int state; + spin_lock(&fe->card->dpcm_lock); for_each_dpcm_fe(be, stream, dpcm) { if (dpcm->fe == fe) @@ -3229,9 +3236,12 @@ int snd_soc_dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe, state = dpcm->fe->dpcm[stream].state; if (state == SND_SOC_DPCM_STATE_START || state == SND_SOC_DPCM_STATE_PAUSED || - state == SND_SOC_DPCM_STATE_SUSPEND) + state == SND_SOC_DPCM_STATE_SUSPEND) { + spin_unlock(&fe->card->dpcm_lock); return 0; + } } + spin_unlock(&fe->card->dpcm_lock); /* it's safe to free/stop this BE DAI */ return 1; @@ -3248,6 +3258,7 @@ int snd_soc_dpcm_can_be_params(struct snd_soc_pcm_runtime *fe, struct snd_soc_dpcm *dpcm; int state; + spin_lock(&fe->card->dpcm_lock); for_each_dpcm_fe(be, stream, dpcm) { if (dpcm->fe == fe) @@ -3257,9 +3268,12 @@ int snd_soc_dpcm_can_be_params(struct snd_soc_pcm_runtime *fe, if (state == SND_SOC_DPCM_STATE_START || state == SND_SOC_DPCM_STATE_PAUSED || state == SND_SOC_DPCM_STATE_SUSPEND || - state == SND_SOC_DPCM_STATE_PREPARE) + state == SND_SOC_DPCM_STATE_PREPARE) { + spin_unlock(&fe->card->dpcm_lock); return 0; + } } + spin_unlock(&fe->card->dpcm_lock); /* it's safe to change hw_params */ return 1; @@ -3328,6 +3342,7 @@ static ssize_t dpcm_show_state(struct snd_soc_pcm_runtime *fe, goto out; } + spin_lock(&fe->card->dpcm_lock); for_each_dpcm_be(fe, stream, dpcm) { struct snd_soc_pcm_runtime *be = dpcm->be; params = &dpcm->hw_params; @@ -3348,7 +3363,7 @@ static ssize_t dpcm_show_state(struct snd_soc_pcm_runtime *fe, params_channels(params), params_rate(params)); } - + spin_unlock(&fe->card->dpcm_lock); out: return offset; } -- 2.18.0

2 1

[alsa-devel] [PATCH RESEND v5 0/3] Add NXP AUDMIX device and machine drivers
by Viorel Suman 04 Mar '19

04 Mar '19

The patchset adds NXP Audio Mixer (AUDMIX) device and machine drivers and related DT bindings documentation. Changes since V4: 1. Removed "model" attribute from device driver DT bindings documentation as suggested by Nicolin. Changes since V3: 1. Removed machine driver DT bindings documentation. 2. Trigger machine driver probe from device driver as suggested by Nicolin. Changes since V2: 1. Moved "dais" node from machine driver DTS node to device driver DTS node as suggested by Rob. Changes since V1: 1. Original patch split into distinct patches for the device driver and DT binding documentation. 2. Replaced AMIX with AUDMIX in both code and file names as it looks more RM-compliant. 3. Removed polarity control from CPU DAI driver as suggested by Nicolin. 4. Added machine driver and related DT binding documentation. Viorel Suman (3): ASoC: fsl: Add Audio Mixer CPU DAI driver ASoC: add fsl_audmix DT binding documentation ASoC: fsl: Add Audio Mixer machine driver .../devicetree/bindings/sound/fsl,audmix.txt | 50 ++ sound/soc/fsl/Kconfig | 16 + sound/soc/fsl/Makefile | 5 + sound/soc/fsl/fsl_audmix.c | 578 +++++++++++++++++++++ sound/soc/fsl/fsl_audmix.h | 102 ++++ sound/soc/fsl/imx-audmix.c | 327 ++++++++++++ 6 files changed, 1078 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/fsl,audmix.txt create mode 100644 sound/soc/fsl/fsl_audmix.c create mode 100644 sound/soc/fsl/fsl_audmix.h create mode 100644 sound/soc/fsl/imx-audmix.c -- 2.7.4

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[alsa-devel] [PATCH] ASoC: AMD: Configure master codec on all playback/capture cases
by Agrawal, Akshu 04 Mar '19

04 Mar '19

In the system design da7219 is the master codec and clocks are generated by it. Bclk is to be generated at the required rate for other codecs used when da7219 is acting only as clock master. For this call hw_params of da7219 during playback/capture on non da7219 codecs. Being able to set bclk at lower rate also fixes noise issue observed on some dmics. Signed-off-by: Akshu Agrawal <akshu.agrawal(a)amd.com> --- sound/soc/amd/acp-da7219-max98357a.c | 20 ++++++++++++++++++-- 1 file changed, 18 insertions(+), 2 deletions(-) diff --git a/sound/soc/amd/acp-da7219-max98357a.c b/sound/soc/amd/acp-da7219-max98357a.c index a5daad973ce5..ad327415290a 100644 --- a/sound/soc/amd/acp-da7219-max98357a.c +++ b/sound/soc/amd/acp-da7219-max98357a.c @@ -45,6 +45,7 @@ #define CZ_PLAT_CLK 48000000 #define DUAL_CHANNEL 2 +static struct snd_soc_dai *codec_dai; static struct snd_soc_jack cz_jack; static struct clk *da7219_dai_clk; extern int bt_uart_enable; @@ -53,8 +54,10 @@ static int cz_da7219_init(struct snd_soc_pcm_runtime *rtd) { int ret; struct snd_soc_card *card = rtd->card; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_component *component = codec_dai->component; + struct snd_soc_component *component; + + codec_dai = rtd->codec_dai; + component = codec_dai->component; dev_info(rtd->dev, "codec dai name = %s\n", codec_dai->name); @@ -255,6 +258,16 @@ static void cz_dmic_shutdown(struct snd_pcm_substream *substream) da7219_clk_disable(); } +static static int cz_da7219_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + /* da7219 Codec is clock master so setup as per the needs */ + if (codec_dai->driver->ops->hw_params) + return codec_dai->driver->ops->hw_params(substream, params, + codec_dai); + return 0; +} + static const struct snd_soc_ops cz_da7219_play_ops = { .startup = cz_da7219_play_startup, .shutdown = cz_da7219_shutdown, @@ -268,16 +281,19 @@ static const struct snd_soc_ops cz_da7219_cap_ops = { static const struct snd_soc_ops cz_max_play_ops = { .startup = cz_max_startup, .shutdown = cz_max_shutdown, + .hw_params = cz_da7219_hw_params, }; static const struct snd_soc_ops cz_dmic0_cap_ops = { .startup = cz_dmic0_startup, .shutdown = cz_dmic_shutdown, + .hw_params = cz_da7219_hw_params, }; static const struct snd_soc_ops cz_dmic1_cap_ops = { .startup = cz_dmic1_startup, .shutdown = cz_dmic_shutdown, + .hw_params = cz_da7219_hw_params, }; static struct snd_soc_dai_link cz_dai_7219_98357[] = { -- 2.17.1

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[alsa-devel] [PATCH RFC 0/3] tda998x updates for DAI formats and bclk_ratio
by Russell King - ARM Linux admin 04 Mar '19

04 Mar '19

This series addresses two issues with TDA998x that have been identified: 1) Peter found that the I2S format was not being explicitly set, and retains its value from whatever was previously running on the platform. Work around this by implementing support for setting the I2S format from the DAI format, rather than merely defaulting the register back to its power-on value. 2) Sven found that TDA998x does not work on his Freescale platform, which always uses a 64·fs bitclock. The TDA998x driver was deriving this information from the sample width, which, while it works for Beagle Bone Black, does not allow the driver to be used with other I2S sources that may have different behaviours. To work around that, we implement support for snd_soc_dai_set_bclk_ratio() in hdmi-codec, and propagate its value to TDA998x and other HDMI codecs via a new member. However, since snd_soc_dai_set_bclk_ratio() is never called, we need to avoid breaking any existing users, so we detect the lack of call by an impossible zero value, and subsitute a value corresponding with the TDA998x's old behaviour. It is hoped that snd_soc_dai_set_bclk_ratio() will see more adoption in ASoC, and the TDA998x specific defaulting can be removed. drivers/gpu/drm/i2c/tda998x_drv.c | 75 ++++++++++++++++++++++++++------------- include/drm/i2c/tda998x.h | 12 +++++-- include/sound/hdmi-codec.h | 1 + sound/soc/codecs/hdmi-codec.c | 45 +++++++++++++++++++++-- 4 files changed, 104 insertions(+), 29 deletions(-) -- RMK's Patch system: https://www.armlinux.org.uk/developer/patches/ FTTC broadband for 0.8mile line in suburbia: sync at 12.1Mbps down 622kbps up According to speedtest.net: 11.9Mbps down 500kbps up

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[alsa-devel] [PATCH] ASoC: hdmi-codec: avoid limiting params->msbits in hw_params()
by Russell King 04 Mar '19

04 Mar '19

Limiting the value of the passed in params->msbits in the hw_params() callback is redundant on three counts: 1. We already specify in the DAI driver that we can only handle up to 24 bits. This means msbits will be limited to 24 via the ALSA constraints imposed by the ASoC core, unless we have multiple codecs that can handle more bits. 2. Nothing in our hw_params() implementation uses this value. 3. The copy of the params that we are passed by the ASoC core never reads back the msbits value. Consequently, this code is unnecessary and does nothing useful. Remove it. Signed-off-by: Russell King <rmk+kernel(a)armlinux.org.uk> --- sound/soc/codecs/hdmi-codec.c | 3 --- 1 file changed, 3 deletions(-) diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c index d5f73c837281..35df73e42cbc 100644 --- a/sound/soc/codecs/hdmi-codec.c +++ b/sound/soc/codecs/hdmi-codec.c @@ -484,9 +484,6 @@ static int hdmi_codec_hw_params(struct snd_pcm_substream *substream, params_width(params), params_rate(params), params_channels(params)); - if (params_width(params) > 24) - params->msbits = 24; - ret = snd_pcm_create_iec958_consumer_hw_params(params, hp.iec.status, sizeof(hp.iec.status)); if (ret < 0) { -- 2.7.4

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[alsa-devel] Regression: ASoC: soc-core: clear platform pointers on error
by Jon Hunter 04 Mar '19

04 Mar '19

Hi Mark, Curtis, I am seeing a regression on -next where the soundcard on one of our Tegra boards fails to initialise following a probe deferral. The bisect points to the commit 78a24e10cd94420f1b4e2dc5923ae7109e2aaba1 ('ASoC: soc-core: clear platform pointers on error') and reverting this on top of -next fixes the problem. Looking at the bootlog from the failure I see ... tegra-snd-wm8903 sound: ASoC: failed to init link WM8903 tegra-snd-wm8903 sound: snd_soc_register_card failed (-517) tegra30-i2s 70080400.i2s: DMA channels sourced from device 70080000.ahub tegra-snd-wm8903 sound: ASoC: Both platform name/of_node are set for WM8903 tegra-snd-wm8903 sound: ASoC: failed to init link WM8903 tegra-snd-wm8903 sound: snd_soc_register_card failed (-22) tegra-snd-wm8903: probe of sound failed with error -22 With the above change I see soc_cleanup_platform() is ever being called when the probe is deferred and hence leads to the failure. Note that the initial failure, "ASoC: failed to init link WM8903" occurs very early in snd_soc_register_card() when initialising the prelinks. The following fixes it, but I have not scrutinised the code to see if there are other exit points that we need to handle. diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 93d316d5bf8e..6f66beb0c3ae 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2797,6 +2797,7 @@ int snd_soc_register_card(struct snd_soc_card *card) ret = soc_init_dai_link(card, link); if (ret) { + soc_cleanup_platform(card); dev_err(card->dev, "ASoC: failed to init link %s\n", link->name); mutex_unlock(&client_mutex); Cheers Jon -- nvpublic

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[alsa-devel] [PATCH v2] ASoC: tlv320aic3x: fix reset gpio reference counting
by Philipp Puschmann 04 Mar '19

04 Mar '19

This patch fixes a bug that prevents freeing the reset gpio on unloading the module. aic3x_i2c_probe is called when loading the module and it calls list_add with a probably uninitialized list entry aic3x->list (next = prev = NULL)). So even if list_del is called it does nothing and in the end the gpio_reset is not freed. Then a repeated module probing fails silently because gpio_request fails. When moving INIT_LIST_HEAD to aic3x_i2c_probe we also have to move list_del to aic3x_i2c_remove because aic3x_remove may be called multiple times without aic3x_i2c_remove being called which leads to a NULL pointer dereference. Signed-off-by: Philipp Puschmann <philipp.puschmann(a)emlix.com> --- Changes in v2: Fix typo in title tlv320aix3x -> tlv320aic3x sound/soc/codecs/tlv320aic3x.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 6aa0edf8c5ef..cea3ebecdb12 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -1609,7 +1609,6 @@ static int aic3x_probe(struct snd_soc_component *component) struct aic3x_priv *aic3x = snd_soc_component_get_drvdata(component); int ret, i; - INIT_LIST_HEAD(&aic3x->list); aic3x->component = component; for (i = 0; i < ARRAY_SIZE(aic3x->supplies); i++) { @@ -1692,7 +1691,6 @@ static void aic3x_remove(struct snd_soc_component *component) struct aic3x_priv *aic3x = snd_soc_component_get_drvdata(component); int i; - list_del(&aic3x->list); for (i = 0; i < ARRAY_SIZE(aic3x->supplies); i++) regulator_unregister_notifier(aic3x->supplies[i].consumer, &aic3x->disable_nb[i].nb); @@ -1890,6 +1888,7 @@ static int aic3x_i2c_probe(struct i2c_client *i2c, if (ret != 0) goto err_gpio; + INIT_LIST_HEAD(&aic3x->list); list_add(&aic3x->list, &reset_list); return 0; @@ -1906,6 +1905,8 @@ static int aic3x_i2c_remove(struct i2c_client *client) { struct aic3x_priv *aic3x = i2c_get_clientdata(client); + list_del(&aic3x->list); + if (gpio_is_valid(aic3x->gpio_reset) && !aic3x_is_shared_reset(aic3x)) { gpio_set_value(aic3x->gpio_reset, 0); -- 2.20.1

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[alsa-devel] Applied "ASoC: tlv320aic3x: fix reset gpio reference counting" to the asoc tree
by Mark Brown 04 Mar '19

04 Mar '19

The patch ASoC: tlv320aic3x: fix reset gpio reference counting has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 82ad759143ed77673db0d93d53c1cde7b99917ee Mon Sep 17 00:00:00 2001 From: Philipp Puschmann <philipp.puschmann(a)emlix.com> Date: Wed, 27 Feb 2019 16:17:33 +0100 Subject: [PATCH] ASoC: tlv320aic3x: fix reset gpio reference counting This patch fixes a bug that prevents freeing the reset gpio on unloading the module. aic3x_i2c_probe is called when loading the module and it calls list_add with a probably uninitialized list entry aic3x->list (next = prev = NULL)). So even if list_del is called it does nothing and in the end the gpio_reset is not freed. Then a repeated module probing fails silently because gpio_request fails. When moving INIT_LIST_HEAD to aic3x_i2c_probe we also have to move list_del to aic3x_i2c_remove because aic3x_remove may be called multiple times without aic3x_i2c_remove being called which leads to a NULL pointer dereference. Signed-off-by: Philipp Puschmann <philipp.puschmann(a)emlix.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/tlv320aic3x.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 6aa0edf8c5ef..cea3ebecdb12 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -1609,7 +1609,6 @@ static int aic3x_probe(struct snd_soc_component *component) struct aic3x_priv *aic3x = snd_soc_component_get_drvdata(component); int ret, i; - INIT_LIST_HEAD(&aic3x->list); aic3x->component = component; for (i = 0; i < ARRAY_SIZE(aic3x->supplies); i++) { @@ -1692,7 +1691,6 @@ static void aic3x_remove(struct snd_soc_component *component) struct aic3x_priv *aic3x = snd_soc_component_get_drvdata(component); int i; - list_del(&aic3x->list); for (i = 0; i < ARRAY_SIZE(aic3x->supplies); i++) regulator_unregister_notifier(aic3x->supplies[i].consumer, &aic3x->disable_nb[i].nb); @@ -1890,6 +1888,7 @@ static int aic3x_i2c_probe(struct i2c_client *i2c, if (ret != 0) goto err_gpio; + INIT_LIST_HEAD(&aic3x->list); list_add(&aic3x->list, &reset_list); return 0; @@ -1906,6 +1905,8 @@ static int aic3x_i2c_remove(struct i2c_client *client) { struct aic3x_priv *aic3x = i2c_get_clientdata(client); + list_del(&aic3x->list); + if (gpio_is_valid(aic3x->gpio_reset) && !aic3x_is_shared_reset(aic3x)) { gpio_set_value(aic3x->gpio_reset, 0); -- 2.20.1

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