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

[PATCH 0/1] ASoC: dpcm: allow started or stop during pause for backend
by 이경택 30 Mar '20

30 Mar '20

Hi, When compress offload is used with DPCM, error can be occurred on start or stop after pause. soc_compr_trigger_fe() allows start or stop after pause_push. In dpcm_be_dai_trigger(), however, only pause_release is allowed command after pause_push. Any feedback will be greatly appreciated. Gyeongtaek Lee (1): ASoC: dpcm: allow start or stop during pause for backend sound/soc/soc-pcm.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) base-commit: 76ccd234269bd05debdbc12c96eafe62dd9a6180 -- 2.21.0

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[PATCH 0/4] ASoC: remove rtd->cpu/codec_dai{s} v2
by Kuninori Morimoto 30 Mar '20

30 Mar '20

Hi Mark These are rebased "remove rtd->cpu/codec_dai" patches. Kuninori Morimoto (4): ASoC: soc: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: soc-core: set rtd->num_cpu/codec at soc_new_pcm_runtime() ASoC: soc-core: tidyup soc_new_pcm_runtime() rtd setups ASoC: soc-core: remove cpu_dai/codec_dai/cpu_dais/codec_dais include/sound/soc.h | 26 +++++++++-------- sound/soc/soc-compress.c | 36 +++++++++++------------ sound/soc/soc-core.c | 42 ++++++++++----------------- sound/soc/soc-dapm.c | 4 +-- sound/soc/soc-generic-dmaengine-pcm.c | 6 ++-- sound/soc/soc-pcm.c | 30 +++++++++---------- 6 files changed, 68 insertions(+), 76 deletions(-) -- 2.17.1

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[PATCH v2] fix some output pcms being namehint-ed as input pcms
by sylvain.bertrand＠gmail.com 29 Mar '20

29 Mar '20

This is additional cleanup following: commit d434638a683c01bd6decf8f41863044055237a50 namehint: remember the direction from the upper levels signed-off-by: Sylvain BERTRAND <sylvain.bertrand(a)legeek.net> --- based on arecord -L output --- a/share/alsa/pcm/front.conf +++ b/share/alsa/pcm/front.conf @@ -53,6 +53,7 @@ name defaults.namehint.basic } description "Front speakers" - device $DEV + device_output $DEV + omit_noargs true } } --- a/share/alsa/pcm/iec958.conf +++ b/share/alsa/pcm/iec958.conf @@ -78,6 +78,6 @@ name defaults.namehint.basic } description "IEC958 (S/PDIF) Digital Audio Output" - device $DEV + device_output $DEV } }

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[PATCH] ASoC: bcm2835-i2s: substream alignment now independent in hwparams
by Matt Flax 29 Mar '20

29 Mar '20

Substream sample alignment was being set in hwparams for both substreams at the same time. This became a problem when the Audio Injector isolated sound card needed to offset sample alignment for high sample rates. The latency difference between playback and capture occurs because of the digital isolation chip propagation time, particularly when the codec is master and the DAC return is twice delayed. This patch sets sample alignment registers based on the substream direction in hwparams. This gives the machine driver more control over sample alignment in the bcm2835 i2s driver. Signed-off-by: Matt Flax <flatmax(a)flatmax.org> --- sound/soc/bcm/bcm2835-i2s.c | 36 +++++++++++++++++++----------------- 1 file changed, 19 insertions(+), 17 deletions(-) diff --git a/sound/soc/bcm/bcm2835-i2s.c b/sound/soc/bcm/bcm2835-i2s.c index e6a12e271b07..9db542699a13 100644 --- a/sound/soc/bcm/bcm2835-i2s.c +++ b/sound/soc/bcm/bcm2835-i2s.c @@ -493,11 +493,6 @@ static int bcm2835_i2s_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - bcm2835_i2s_calc_channel_pos(&rx_ch1_pos, &rx_ch2_pos, - rx_mask, slot_width, data_delay, odd_slot_offset); - bcm2835_i2s_calc_channel_pos(&tx_ch1_pos, &tx_ch2_pos, - tx_mask, slot_width, data_delay, odd_slot_offset); - /* * Transmitting data immediately after frame start, eg * in left-justified or DSP mode A, only works stable @@ -508,19 +503,26 @@ static int bcm2835_i2s_hw_params(struct snd_pcm_substream *substream, "Unstable slave config detected, L/R may be swapped"); /* - * Set format for both streams. - * We cannot set another frame length - * (and therefore word length) anyway, - * so the format will be the same. + * Set format on a per stream basis. + * The alignment format can be different depending on direction. */ - regmap_write(dev->i2s_regmap, BCM2835_I2S_RXC_A_REG, - format - | BCM2835_I2S_CH1_POS(rx_ch1_pos) - | BCM2835_I2S_CH2_POS(rx_ch2_pos)); - regmap_write(dev->i2s_regmap, BCM2835_I2S_TXC_A_REG, - format - | BCM2835_I2S_CH1_POS(tx_ch1_pos) - | BCM2835_I2S_CH2_POS(tx_ch2_pos)); + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { + bcm2835_i2s_calc_channel_pos(&rx_ch1_pos, &rx_ch2_pos, + rx_mask, slot_width, data_delay, odd_slot_offset); + regmap_write(dev->i2s_regmap, BCM2835_I2S_RXC_A_REG, + format + | BCM2835_I2S_CH1_POS(rx_ch1_pos) + | BCM2835_I2S_CH2_POS(rx_ch2_pos)); + } + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + bcm2835_i2s_calc_channel_pos(&tx_ch1_pos, &tx_ch2_pos, + tx_mask, slot_width, data_delay, odd_slot_offset); + regmap_write(dev->i2s_regmap, BCM2835_I2S_TXC_A_REG, + format + | BCM2835_I2S_CH1_POS(tx_ch1_pos) + | BCM2835_I2S_CH2_POS(tx_ch2_pos)); + } /* Setup the I2S mode */ -- 2.20.1

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[PATCH v2] ALSA: hda/realtek - a fake key event is triggered by running shutup
by Hui Wang 29 Mar '20

29 Mar '20

On the Lenovo X1C7 machines, after we plug the headset, the rt_resume() and rt_suspend() of the codec driver will be called periodically, the driver can't stay in the rt_suspend state even users doen't use the sound card. Through debugging, I found when running rt_suspend(), it will call alc225_shutup(), in this function, it will change 3k pull down control by alc_update_coef_idx(codec, 0x4a, 0, 3 << 10), this will trigger a fake key event and that event will resume the codec, when codec suspend agin, it will trigger the fake key event one more time, this process will repeat. If disable the key event before changing the pull down control, it will not trigger fake key event. It also needs to restore the pull down control and re-enable the key event, otherwise the system can't get key event when codec is in rt_suspend state. Also move some functions ahead of alc225_shutup(), this can save the function declaration. Fixes: 76f7dec08fd6 (ALSA: hda/realtek - Add Headset Button supported for ThinkPad X1) Cc: Kailang Yang <kailang(a)realtek.com> Cc: <stable(a)vger.kernel.org> Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 170 +++++++++++++++++++++------------- 1 file changed, 107 insertions(+), 63 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 1ad8c2e2d1af..0af33f00617a 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -107,6 +107,7 @@ struct alc_spec { unsigned int done_hp_init:1; unsigned int no_shutup_pins:1; unsigned int ultra_low_power:1; + unsigned int has_hs_key:1; /* for PLL fix */ hda_nid_t pll_nid; @@ -2982,6 +2983,107 @@ static int alc269_parse_auto_config(struct hda_codec *codec) return alc_parse_auto_config(codec, alc269_ignore, ssids); } +static const struct hda_jack_keymap alc_headset_btn_keymap[] = { + { SND_JACK_BTN_0, KEY_PLAYPAUSE }, + { SND_JACK_BTN_1, KEY_VOICECOMMAND }, + { SND_JACK_BTN_2, KEY_VOLUMEUP }, + { SND_JACK_BTN_3, KEY_VOLUMEDOWN }, + {} +}; + +static void alc_headset_btn_callback(struct hda_codec *codec, + struct hda_jack_callback *jack) +{ + int report = 0; + + if (jack->unsol_res & (7 << 13)) + report |= SND_JACK_BTN_0; + + if (jack->unsol_res & (1 << 16 | 3 << 8)) + report |= SND_JACK_BTN_1; + + /* Volume up key */ + if (jack->unsol_res & (7 << 23)) + report |= SND_JACK_BTN_2; + + /* Volume down key */ + if (jack->unsol_res & (7 << 10)) + report |= SND_JACK_BTN_3; + + jack->jack->button_state = report; +} + +static void alc_disable_headset_jack_key(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + + if (!spec->has_hs_key) + return; + + switch (codec->core.vendor_id) { + case 0x10ec0215: + case 0x10ec0225: + case 0x10ec0285: + case 0x10ec0295: + case 0x10ec0289: + case 0x10ec0299: + alc_write_coef_idx(codec, 0x48, 0x0); + alc_update_coef_idx(codec, 0x49, 0x0045, 0x0); + alc_update_coef_idx(codec, 0x44, 0x0045 << 8, 0x0); + break; + case 0x10ec0236: + case 0x10ec0256: + alc_write_coef_idx(codec, 0x48, 0x0); + alc_update_coef_idx(codec, 0x49, 0x0045, 0x0); + break; + } +} + +static void alc_enable_headset_jack_key(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + + if (!spec->has_hs_key) + return; + + switch (codec->core.vendor_id) { + case 0x10ec0215: + case 0x10ec0225: + case 0x10ec0285: + case 0x10ec0295: + case 0x10ec0289: + case 0x10ec0299: + alc_write_coef_idx(codec, 0x48, 0xd011); + alc_update_coef_idx(codec, 0x49, 0x007f, 0x0045); + alc_update_coef_idx(codec, 0x44, 0x007f << 8, 0x0045 << 8); + break; + case 0x10ec0236: + case 0x10ec0256: + alc_write_coef_idx(codec, 0x48, 0xd011); + alc_update_coef_idx(codec, 0x49, 0x007f, 0x0045); + break; + } +} + +static void alc_fixup_headset_jack(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + struct alc_spec *spec = codec->spec; + + switch (action) { + case HDA_FIXUP_ACT_PRE_PROBE: + spec->has_hs_key = 1; + snd_hda_jack_detect_enable_callback(codec, 0x55, + alc_headset_btn_callback); + snd_hda_jack_add_kctl(codec, 0x55, "Headset Jack", false, + SND_JACK_HEADSET, alc_headset_btn_keymap); + break; + case HDA_FIXUP_ACT_INIT: + alc_enable_headset_jack_key(codec); + break; + } +} + static void alc269vb_toggle_power_output(struct hda_codec *codec, int power_up) { alc_update_coef_idx(codec, 0x04, 1 << 11, power_up ? (1 << 11) : 0); @@ -3372,6 +3474,8 @@ static void alc225_shutup(struct hda_codec *codec) if (!hp_pin) hp_pin = 0x21; + + alc_disable_headset_jack_key(codec); /* 3k pull low control for Headset jack. */ alc_update_coef_idx(codec, 0x4a, 0, 3 << 10); @@ -3411,6 +3515,9 @@ static void alc225_shutup(struct hda_codec *codec) alc_update_coef_idx(codec, 0x4a, 3<<4, 2<<4); msleep(30); } + + alc_update_coef_idx(codec, 0x4a, 3 << 10, 0); + alc_enable_headset_jack_key(codec); } static void alc_default_init(struct hda_codec *codec) @@ -5668,69 +5775,6 @@ static void alc285_fixup_invalidate_dacs(struct hda_codec *codec, snd_hda_override_wcaps(codec, 0x03, 0); } -static const struct hda_jack_keymap alc_headset_btn_keymap[] = { - { SND_JACK_BTN_0, KEY_PLAYPAUSE }, - { SND_JACK_BTN_1, KEY_VOICECOMMAND }, - { SND_JACK_BTN_2, KEY_VOLUMEUP }, - { SND_JACK_BTN_3, KEY_VOLUMEDOWN }, - {} -}; - -static void alc_headset_btn_callback(struct hda_codec *codec, - struct hda_jack_callback *jack) -{ - int report = 0; - - if (jack->unsol_res & (7 << 13)) - report |= SND_JACK_BTN_0; - - if (jack->unsol_res & (1 << 16 | 3 << 8)) - report |= SND_JACK_BTN_1; - - /* Volume up key */ - if (jack->unsol_res & (7 << 23)) - report |= SND_JACK_BTN_2; - - /* Volume down key */ - if (jack->unsol_res & (7 << 10)) - report |= SND_JACK_BTN_3; - - jack->jack->button_state = report; -} - -static void alc_fixup_headset_jack(struct hda_codec *codec, - const struct hda_fixup *fix, int action) -{ - - switch (action) { - case HDA_FIXUP_ACT_PRE_PROBE: - snd_hda_jack_detect_enable_callback(codec, 0x55, - alc_headset_btn_callback); - snd_hda_jack_add_kctl(codec, 0x55, "Headset Jack", false, - SND_JACK_HEADSET, alc_headset_btn_keymap); - break; - case HDA_FIXUP_ACT_INIT: - switch (codec->core.vendor_id) { - case 0x10ec0215: - case 0x10ec0225: - case 0x10ec0285: - case 0x10ec0295: - case 0x10ec0289: - case 0x10ec0299: - alc_write_coef_idx(codec, 0x48, 0xd011); - alc_update_coef_idx(codec, 0x49, 0x007f, 0x0045); - alc_update_coef_idx(codec, 0x44, 0x007f << 8, 0x0045 << 8); - break; - case 0x10ec0236: - case 0x10ec0256: - alc_write_coef_idx(codec, 0x48, 0xd011); - alc_update_coef_idx(codec, 0x49, 0x007f, 0x0045); - break; - } - break; - } -} - static void alc295_fixup_chromebook(struct hda_codec *codec, const struct hda_fixup *fix, int action) { -- 2.17.1

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about alc880 no sound and eapd coef
by 尤晓杰 29 Mar '20

29 Mar '20

dear friend, I am using ecs g420 laptop, use options snd-hda-intel model=generic the speaker and mic correctly, 2 pop sounds,when the kernel load(left channel) and the x11 load(right channel), but with standard way only one pop sound when kernel load and silent always. the mic is ok. I have got datasheet of alc880 and see the eapd function, but in your code I see this need coef parameter. so what should I write using verb to open eapd? the attachment is the schematic of sound part.in summary: the 0x14 hp out, 0x18 mic in,0x1c cd in, 0x1e spif out, no jack sense. eapd shared with spdif in. all are correct with default of datasheet's describe. I use verb to dectt each pin widget's capability , take 0x14 as exaple, no eapd capability.To set 0x14's eapd with hda-verb /dev/snd/hwC0D0 0x70c 0x2, failed to read it again alwas 0. and bios also reported 0x16, I think it is bios's bug. I what to use "Early Patching" method to deal with it without changing source code, is it feasible? could you give me some idea? thanks!!!! dmesg: [ 8.009944] snd_hda_codec_realtek hdaudioC0D0: autoconfig for ALC880: line_outs=2 (0x14/0x16/0x0/0x0/0x0) type:hp [ 8.014023] snd_hda_codec_realtek hdaudioC0D0: speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) [ 8.017999] snd_hda_codec_realtek hdaudioC0D0: hp_outs=0 (0x0/0x0/0x0/0x0/0x0) [ 8.021625] snd_hda_codec_realtek hdaudioC0D0: mono: mono_out=0x0 [ 8.024255] snd_hda_codec_realtek hdaudioC0D0: dig-out=0x1e/0x0 [ 8.026741] snd_hda_codec_realtek hdaudioC0D0: inputs: [ 8.038468] mc: Linux media interface: v0.10 [ 8.067515] snd_hda_codec_realtek hdaudioC0D0: Mic=0x18 [ 8.078984] snd_hda_codec_realtek hdaudioC0D0: CD=0x1c [ 8.290306] input: HDA Digital PCBeep as /devices/pci0000:00/0000:00:1b.0/sound/card0/input13

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Port HDA Analyzer to python 3
by andythe_great 29 Mar '20

29 Mar '20

Hello, Is it possible to port HDA Analyzer to python 3? https://www.alsa-project.org/main/index.php/HDA_Analyzer Because openSUSE is retiring python 2. I tried using python-modernize https://python-modernize.readthedocs.io/en/latest/ but many could not get it to run. Thanks. Andy

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[PATCH] ALSA: hda: default enable CA0132 DSP support
by Rouven Czerwinski 29 Mar '20

29 Mar '20

From: Rouven Czerwinski <rouven(a)czerwinskis.de> If SND_HDA_CODEC_CA0132 is enabled, the DSP support should be enabled as well. Disabled DSP support leads to a hanging alsa system and no sound output on the card otherwise. Tested on: 06:00.0 Audio device: Creative Labs Sound Core3D [Sound Blaster Recon3D / Z-Series] (rev 01) Signed-off-by: Rouven Czerwinski <rouven(a)czerwinskis.de> --- sound/pci/hda/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig index bd48335d09d7..e1d3082a4fe9 100644 --- a/sound/pci/hda/Kconfig +++ b/sound/pci/hda/Kconfig @@ -184,6 +184,7 @@ comment "Set to Y if you want auto-loading the codec driver" config SND_HDA_CODEC_CA0132_DSP bool "Support new DSP code for CA0132 codec" depends on SND_HDA_CODEC_CA0132 + default y select SND_HDA_DSP_LOADER select FW_LOADER help base-commit: e595dd94515ed6bc5ba38fce0f9598db8c0ee9a9 -- 2.25.1

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[alsa-devel] [PATCH 0/3] ALSA: bebob: misc improvements for starting/stopping packet streaming
by Takashi Sakamoto 27 Mar '20

27 Mar '20

Hi, This patchset improves model-specific issues in a point of protocol version supported by the model. The changes are relevant to packet streaming, thus it can brings some regressions. For testers, I prepare for a remote branch to backport the latest codes to Linux v4.17 or later. I'm happy to receive any report. https://github.com/takaswie/snd-firewire-improve/tree/topic/bebob-connectio… Takashi Sakamoto (3): ALSA: bebob: expand sleep just after breaking connections for protocol version 1 ALSA: bebob: expand delay of start for IR context just for version 3 firmware ALSA: bebob: link the order of establishing connections and Syt-match clock mode sound/firewire/bebob/bebob_stream.c | 76 ++++++++++++++--------------- 1 file changed, 38 insertions(+), 38 deletions(-) -- 2.20.1

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[PATCH] ASoC: topology: Add missing memory checks
by Amadeusz Sławiński 27 Mar '20

27 Mar '20

kstrdup is an allocation function and it can fail, so its return value should be checked and handled appropriately. In order to check all cases, we need to modify set_stream_info to return a value, so check that everything went correctly when doing kstrdup(). Later add proper checks and error handlers. Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> --- sound/soc/soc-topology.c | 65 +++++++++++++++++++++++++++++++--------- 1 file changed, 51 insertions(+), 14 deletions(-) diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 575da6aba807..0bec3ff782c1 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -1766,10 +1766,13 @@ static int soc_tplg_dapm_complete(struct soc_tplg *tplg) return 0; } -static void set_stream_info(struct snd_soc_pcm_stream *stream, +static int set_stream_info(struct snd_soc_pcm_stream *stream, struct snd_soc_tplg_stream_caps *caps) { stream->stream_name = kstrdup(caps->name, GFP_KERNEL); + if (!stream->stream_name) + return -ENOMEM; + stream->channels_min = le32_to_cpu(caps->channels_min); stream->channels_max = le32_to_cpu(caps->channels_max); stream->rates = le32_to_cpu(caps->rates); @@ -1777,6 +1780,8 @@ static void set_stream_info(struct snd_soc_pcm_stream *stream, stream->rate_max = le32_to_cpu(caps->rate_max); stream->formats = le64_to_cpu(caps->formats); stream->sig_bits = le32_to_cpu(caps->sig_bits); + + return 0; } static void set_dai_flags(struct snd_soc_dai_driver *dai_drv, @@ -1812,20 +1817,29 @@ static int soc_tplg_dai_create(struct soc_tplg *tplg, if (dai_drv == NULL) return -ENOMEM; - if (strlen(pcm->dai_name)) + if (strlen(pcm->dai_name)) { dai_drv->name = kstrdup(pcm->dai_name, GFP_KERNEL); + if (!dai_drv->name) { + ret = -ENOMEM; + goto err; + } + } dai_drv->id = le32_to_cpu(pcm->dai_id); if (pcm->playback) { stream = &dai_drv->playback; caps = &pcm->caps[SND_SOC_TPLG_STREAM_PLAYBACK]; - set_stream_info(stream, caps); + ret = set_stream_info(stream, caps); + if (ret < 0) + goto err; } if (pcm->capture) { stream = &dai_drv->capture; caps = &pcm->caps[SND_SOC_TPLG_STREAM_CAPTURE]; - set_stream_info(stream, caps); + ret = set_stream_info(stream, caps); + if (ret < 0) + goto err; } if (pcm->compress) @@ -1835,11 +1849,7 @@ static int soc_tplg_dai_create(struct soc_tplg *tplg, ret = soc_tplg_dai_load(tplg, dai_drv, pcm, NULL); if (ret < 0) { dev_err(tplg->comp->dev, "ASoC: DAI loading failed\n"); - kfree(dai_drv->playback.stream_name); - kfree(dai_drv->capture.stream_name); - kfree(dai_drv->name); - kfree(dai_drv); - return ret; + goto err; } dai_drv->dobj.index = tplg->index; @@ -1857,9 +1867,17 @@ static int soc_tplg_dai_create(struct soc_tplg *tplg, if (ret != 0) { dev_err(dai->dev, "Failed to create DAI widgets %d\n", ret); snd_soc_unregister_dai(dai); - return ret; + goto err; } + return 0; + +err: + kfree(dai_drv->playback.stream_name); + kfree(dai_drv->capture.stream_name); + kfree(dai_drv->name); + kfree(dai_drv); + return ret; } @@ -1916,11 +1934,20 @@ static int soc_tplg_fe_link_create(struct soc_tplg *tplg, if (strlen(pcm->pcm_name)) { link->name = kstrdup(pcm->pcm_name, GFP_KERNEL); link->stream_name = kstrdup(pcm->pcm_name, GFP_KERNEL); + if (!link->name || !link->stream_name) { + ret = -ENOMEM; + goto err; + } } link->id = le32_to_cpu(pcm->pcm_id); - if (strlen(pcm->dai_name)) + if (strlen(pcm->dai_name)) { link->cpus->dai_name = kstrdup(pcm->dai_name, GFP_KERNEL); + if (!link->cpus->dai_name) { + ret = -ENOMEM; + goto err; + } + } link->codecs->name = "snd-soc-dummy"; link->codecs->dai_name = "snd-soc-dummy-dai"; @@ -2436,13 +2463,17 @@ static int soc_tplg_dai_config(struct soc_tplg *tplg, if (d->playback) { stream = &dai_drv->playback; caps = &d->caps[SND_SOC_TPLG_STREAM_PLAYBACK]; - set_stream_info(stream, caps); + ret = set_stream_info(stream, caps); + if (ret < 0) + goto err; } if (d->capture) { stream = &dai_drv->capture; caps = &d->caps[SND_SOC_TPLG_STREAM_CAPTURE]; - set_stream_info(stream, caps); + ret = set_stream_info(stream, caps); + if (ret < 0) + goto err; } if (d->flag_mask) @@ -2454,10 +2485,16 @@ static int soc_tplg_dai_config(struct soc_tplg *tplg, ret = soc_tplg_dai_load(tplg, dai_drv, NULL, dai); if (ret < 0) { dev_err(tplg->comp->dev, "ASoC: DAI loading failed\n"); - return ret; + goto err; } return 0; + +err: + kfree(dai_drv->playback.stream_name); + kfree(dai_drv->capture.stream_name); + + return ret; } /* load physical DAI elements */ -- 2.17.1

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