August 2023 - Alsa-devel - mailman.alsa-project.org

[PATCH] ALSA: hda/realtek: Add quirk for mute LEDs on HP ENVY x360 15-eu0xxx
by Fabian Vogt 25 Aug '23

25 Aug '23

The LED for the mic mute button is controlled by GPIO2. The mute button LED is slightly more complex, it's controlled by two bits in coeff 0x0b. Add a fixup for the mute LED and chain to an existing fixup for the mic mute LED. Signed-off-by: Fabian Vogt <fabian(a)ritter-vogt.de> --- Applies on top of 89bf6209cad6. I see that SungHwan Jung (CC'd) recently submitted the same fixup for a rather similar but different model as "ALSA: hda/realtek: Add quirk for HP Victus 16-d1xxx to enable mute LED" Does your HP Victus also need the ALC245_FIXUP_HP_GPIO_LED? Then we could combine the quirks. sound/pci/hda/patch_realtek.c | 26 ++++++++++++++++++++++++++ 1 file changed, 26 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index dc7b7a407638..cac207d7c0b9 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4639,6 +4639,24 @@ static void alc236_fixup_hp_mute_led_coefbit2(struct hda_codec *codec, } } +/* The mute LED is controlled by two bits in coef 0x0b: + * To turn on, set bit 3 but clear bit 2. + */ +static void alc245_fixup_hp_x360_mute_led(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + struct alc_spec *spec = codec->spec; + + if (action == HDA_FIXUP_ACT_PRE_PROBE) { + spec->mute_led_polarity = 0; + spec->mute_led_coef.idx = 0x0b; + spec->mute_led_coef.mask = 0xc; + spec->mute_led_coef.on = 0x8; + spec->mute_led_coef.off = 0x4; + snd_hda_gen_add_mute_led_cdev(codec, coef_mute_led_set); + } +} + /* turn on/off mic-mute LED per capture hook by coef bit */ static int coef_micmute_led_set(struct led_classdev *led_cdev, enum led_brightness brightness) @@ -7231,6 +7249,7 @@ enum { ALC295_FIXUP_DELL_INSPIRON_TOP_SPEAKERS, ALC236_FIXUP_DELL_DUAL_CODECS, ALC287_FIXUP_CS35L41_I2C_2_THINKPAD_ACPI, + ALC245_FIXUP_HP_X360_MUTE_LED, }; /* A special fixup for Lenovo C940 and Yoga Duet 7; @@ -9309,6 +9328,12 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_THINKPAD_ACPI, }, + [ALC245_FIXUP_HP_X360_MUTE_LED] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc245_fixup_hp_x360_mute_led, + .chained = true, + .chain_id = ALC245_FIXUP_HP_GPIO_LED + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -9551,6 +9576,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x8870, "HP ZBook Fury 15.6 Inch G8 Mobile Workstation PC", ALC285_FIXUP_HP_GPIO_AMP_INIT), SND_PCI_QUIRK(0x103c, 0x8873, "HP ZBook Studio 15.6 Inch G8 Mobile Workstation PC", ALC285_FIXUP_HP_GPIO_AMP_INIT), SND_PCI_QUIRK(0x103c, 0x887a, "HP Laptop 15s-eq2xxx", ALC236_FIXUP_HP_MUTE_LED_COEFBIT2), + SND_PCI_QUIRK(0x103c, 0x888a, "HP ENVY x360 Convertible 15-eu0xxx", ALC245_FIXUP_HP_X360_MUTE_LED), SND_PCI_QUIRK(0x103c, 0x888d, "HP ZBook Power 15.6 inch G8 Mobile Workstation PC", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8895, "HP EliteBook 855 G8 Notebook PC", ALC285_FIXUP_HP_SPEAKERS_MICMUTE_LED), SND_PCI_QUIRK(0x103c, 0x8896, "HP EliteBook 855 G8 Notebook PC", ALC285_FIXUP_HP_MUTE_LED), -- 2.41.0

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[GIT PULL] ASoC fixes for v6.5-rc7-2
by Mark Brown 25 Aug '23

25 Aug '23

The following changes since commit 37aba3190891d4de189bd5192ee95220e295f34d: ASoC: rt1308-sdw: fix random louder sound (2023-08-13 18:16:32 +0100) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.5-rc7-2 for you to fetch changes up to c008323fe361bd62a43d9fb29737dacd5c067fb7: ASoC: amd: yc: Fix a non-functional mic on Lenovo 82SJ (2023-08-24 13:15:33 +0100) ---------------------------------------------------------------- ASoC: Quirk for v6.5 One additional fix for v6.5, an additional quirk. As with the other fixes this could wait for the merge window. ---------------------------------------------------------------- BrenoRCBrito (1): ASoC: amd: yc: Add VivoBook Pro 15 to quirks list for acp6x Chao Song (1): ASoC: SOF: ipc4-pcm: fix possible null pointer deference Charles Keepax (1): ASoC: cs35l41: Correct amp_gain_tlv values Kevin-Lu (1): MAINTAINERS: Add entries for TEXAS INSTRUMENTS ASoC DRIVERS Maciej Strozek (1): ASoC: cs35l56: Read firmware uuid from a device property instead of _SUB Mario Limonciello (1): ASoC: amd: yc: Fix a non-functional mic on Lenovo 82SJ Mark Brown (1): ASoC: cs35l56: Update ACPI HID and property Shenghao Ding (1): ASoC: tas2781: fixed register access error when switching to other chips Simon Trimmer (1): ASoC: cs35l56: Add an ACPI match table MAINTAINERS | 33 +++++++++++++++++++++++++++++++++ sound/soc/amd/yc/acp6x-mach.c | 9 ++++++++- sound/soc/codecs/cs35l41.c | 2 +- sound/soc/codecs/cs35l56-i2c.c | 9 +++++++++ sound/soc/codecs/cs35l56-spi.c | 9 +++++++++ sound/soc/codecs/cs35l56.c | 31 ++++++++++++------------------- sound/soc/codecs/tas2781-comlib.c | 19 ++++++++++--------- sound/soc/sof/ipc4-pcm.c | 3 +++ 8 files changed, 85 insertions(+), 30 deletions(-)

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[PATCH] ASoC: soc-core.c: Do not error if a DAI link component is not found
by Cristian Ciocaltea 25 Aug '23

25 Aug '23

A recent cleanup of soc_dai_link_sanity_check() is responsible for generating lots of confusing errors before returning -EPROBE_DEFER: acp5x_mach acp5x_mach.0: ASoC: Component acp5x_i2s_dma.0 not found for link acp5x-8821-play [...] acp5x_mach acp5x_mach.0: ASoC: Component spi-VLV1776:00 not found for link acp5x-CS35L41-Stereo [...] acp5x_mach acp5x_mach.0: ASoC: Component spi-VLV1776:01 not found for link acp5x-CS35L41-Stereo Switch back to the initial behaviour of logging those messages on KERN_DEBUG level instead of KERN_ERR. While at it, use the correct form of the verb in 'component_not_find' label. Fixes: 0e66a2c69409 ("ASoC: soc-core.c: cleanup soc_dai_link_sanity_check()") Signed-off-by: Cristian Ciocaltea <cristian.ciocaltea(a)collabora.com> --- sound/soc/soc-core.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index a5b96c17633a..cc442c52cdea 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -942,7 +942,7 @@ static int soc_dai_link_sanity_check(struct snd_soc_card *card, * component list. */ if (!soc_find_component(dlc)) - goto component_not_find; + goto component_not_found; } /* Platform check */ @@ -963,7 +963,7 @@ static int soc_dai_link_sanity_check(struct snd_soc_card *card, * component list. */ if (!soc_find_component(dlc)) - goto component_not_find; + goto component_not_found; } /* CPU check */ @@ -988,7 +988,7 @@ static int soc_dai_link_sanity_check(struct snd_soc_card *card, * Defer card registration if Component is not added */ if (!soc_find_component(dlc)) - goto component_not_find; + goto component_not_found; } } @@ -1002,8 +1002,8 @@ static int soc_dai_link_sanity_check(struct snd_soc_card *card, dev_err(card->dev, "ASoC: Neither Component name/of_node are set for %s\n", link->name); return -EINVAL; -component_not_find: - dev_err(card->dev, "ASoC: Component %s not found for link %s\n", dlc->name, link->name); +component_not_found: + dev_dbg(card->dev, "ASoC: Component %s not found for link %s\n", dlc->name, link->name); return -EPROBE_DEFER; dai_empty: -- 2.41.0

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[PATCH next] ASoC: codecs: Fix error code in aw88261_i2c_probe()
by Harshit Mogalapalli 25 Aug '23

25 Aug '23

Passing zero to dev_err_probe is a success which is incorrect when i2c_check_functionality() fails. Fix this by passing -ENXIO instead of zero to dev_err_probe(). Fixes: 028a2ae25691 ("ASoC: codecs: Add aw88261 amplifier driver") Reported-by: kernel test robot <lkp(a)intel.com> Reported-by: Dan Carpenter <error27(a)gmail.com> Closes: https://lore.kernel.org/r/202308150315.CvOTIOKm-lkp@intel.com/ Signed-off-by: Harshit Mogalapalli <harshit.m.mogalapalli(a)oracle.com> --- This is only compile tested. --- sound/soc/codecs/aw88261.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/aw88261.c b/sound/soc/codecs/aw88261.c index 6e2266b71386..a697b5006b45 100644 --- a/sound/soc/codecs/aw88261.c +++ b/sound/soc/codecs/aw88261.c @@ -1245,7 +1245,7 @@ static int aw88261_i2c_probe(struct i2c_client *i2c) ret = i2c_check_functionality(i2c->adapter, I2C_FUNC_I2C); if (!ret) - return dev_err_probe(&i2c->dev, ret, "check_functionality failed"); + return dev_err_probe(&i2c->dev, -ENXIO, "check_functionality failed"); aw88261 = devm_kzalloc(&i2c->dev, sizeof(*aw88261), GFP_KERNEL); if (!aw88261) -- 2.39.3

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connecting second HDMI/DVI monitor mostly makes Samsung TV sound disappear
by GitHub issues - opened 25 Aug '23

25 Aug '23

alsa-project/alsa-lib issue #347 was opened from chemelnucfin: This is the same issue I posted in Pipewire: https://gitlab.freedesktop.org/pipewire/pipewire/-/issues/3420 But I think this issue might be relevant here also so wanted to try here also. A brief summary is that my Samsung TV loses sound when I connect either HDMI to HDMI or HDMI to DVI to any of 2 Acer and 1 Samsung monitor, but an LG monitor has no problem. The sound on the TV always comes back when I disconnect the (non-working) monitor. I have attached _wpctl status_ for both when I connect LG and Acer monitors, [0824_acer_mon_wpctl_status.txt](https://github.com/alsa-project/alsa-lib/fi… [0824_LG_mon_wpctl_status.txt](https://github.com/alsa-project/alsa-lib/file… and also _alsa_info.sh_ for both monitors. [0824_acer_mon_alsa.txt](https://github.com/alsa-project/alsa-lib/files/1243… [0824_LG_mon_alsa.txt](https://github.com/alsa-project/alsa-lib/files/124338… Please let me know if you need any other information or if there are better places for help to investigate the issue. A brief search of HDMI showed this https://github.com/alsa-project/alsa-lib/issues/95 and 3 other issues which seem less relevant https://github.com/alsa-project/alsa-lib/issues/233 https://github.com/alsa-project/alsa-lib/issues/209 https://github.com/alsa-project/alsa-lib/issues/112 Issue URL : https://github.com/alsa-project/alsa-lib/issues/347 Repository URL: https://github.com/alsa-project/alsa-lib

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[PATCH 00/17] ALSA/ASoC: hda: Address format selection limitations and ambiguity
by Cezary Rojewski 25 Aug '23

25 Aug '23

Patchset aims to address format selection restrictions present currently in the HDAudio library. Formats which we are concerned about are 20 and 24 valid bits per sample within 32 bit depth container. One may identify them as S20_LE and S24_LE except that those, according to comments found in include/uapi/sound/asound.h, are for LSB-aligned scenarios. HDAudio streams expect MSB-aligned data, no matter if we are speaking of HOST (SDxFMT) or LINK (PPLCxFMT) side - chapter 4.5.1 of the public HDAudio specification. In short, S20_LE and S24_LE are invalid options. Right now, given the implementation of snd_hdac_query_supported_pcm() within sound/hda/hdac_device.c, even if a codec responds with: "I support all possible formats specified within HDAudio specification", there will be no option to open a 20/32 or 24/32 stream. The kernel will force the stream to be opened using the highest available bit depth. After discussing subject initially with Jaroslav and Takashi, suggestion was made to utilize 'subformat' option to address the problem. The eye-opening discussion begun much earlier though, in 2019 [1]. Paired with PRs for alsa-utils [2] and alsa-lib [3]. Flow of changes: The very first patch adds MSBITS subformat options to allow for granular 20/32, 24/32 and 32/32 format selection. The next three make sure subformat is actually honored during runtime. Most of that code is based on format-related API. Follow up is upgrade to the hda stream-format interface - several functions are added to make the granular format selection simple in the HDAudio world. Core of the implementation is based on the existing snd_hdac_calc_stream_format(). The next ten patches are straightforward switch from one interface to another with cleanup of now-unsed function as a finishing touch. Last but not least - the avs-driver, on which the problem analyzed and debugged, is updated to no longer acknowledge S24_LE as a valid format option. Results with skylake-driver and snd_hda_intel show status quo on our RVPs. PR filed on SOF github shows promissing results too [4]. Changes in v1: - fixed UBSAN due to missing snd_pcm_subformat_names[] entries for new subformats - as HDMI stream capabilities are assigned on PCM open, patch 16/17 has been updated to ignore such codecs for now. A separate patchset will take care of this case - params_bps() reworded to snd_pcm_hw_params_bps() - fixed compilation issues in sof-driver, patch 13/17 [1]: https://lore.kernel.org/alsa-devel/20190905053302.9262-1-pawel.harlozinski@… [2]: https://github.com/alsa-project/alsa-utils/pull/228 [3]: https://github.com/alsa-project/alsa-lib/pull/342 [4]: https://github.com/thesofproject/linux/pull/4539 Cezary Rojewski (17): ALSA: pcm: Introduce MSBITS subformat interface ALSA: pcm: Honor subformat when configuring substream ALSA: hda: Honor subformat when querying PCMs ASoC: pcm: Honor subformat when configuring runtime ALSA: hda: Upgrade stream-format infrastructure ALSA: hda: Switch to new stream-format interface ALSA: hda/hdmi: Switch to new stream-format interface ALSA: hda/ca0132: Switch to new stream-format interface ASoC: codecs: hda: Switch to new stream-format interface ASoC: codecs: hdac_hda: Switch to new stream-format interface ASoC: codecs: hdac_hdmi: Switch to new stream-format interface ASoC: Intel Skylake: Switch to new stream-format interface ASoC: SOF: Intel: Switch to new stream-format interface ASoC: Intel: avs: Switch to new stream-format interface ALSA: hda: Drop snd_hdac_calc_stream_format() ASoC: Intel: avs: Kill S24_LE in HDAudio streaming ASoC: Intel: avs: Unhardcode HDAudio BE DAI drivers description include/sound/hda_codec.h | 5 +- include/sound/hdaudio.h | 12 +-- include/sound/pcm.h | 8 ++ include/sound/pcm_params.h | 2 + include/sound/soc.h | 1 + include/uapi/sound/asound.h | 5 +- sound/core/pcm.c | 3 + sound/core/pcm_lib.c | 30 ++++++ sound/core/pcm_misc.c | 23 +++++ sound/core/pcm_native.c | 5 +- sound/hda/hdac_device.c | 153 +++++++++++++++++++++--------- sound/pci/hda/hda_codec.c | 2 + sound/pci/hda/hda_controller.c | 10 +- sound/pci/hda/patch_ca0132.c | 3 +- sound/pci/hda/patch_hdmi.c | 6 +- sound/soc/codecs/hda-dai.c | 6 +- sound/soc/codecs/hda.c | 2 + sound/soc/codecs/hdac_hda.c | 8 +- sound/soc/codecs/hdac_hdmi.c | 10 +- sound/soc/intel/avs/loader.c | 4 +- sound/soc/intel/avs/path.c | 2 +- sound/soc/intel/avs/pcm.c | 50 ++++++++-- sound/soc/intel/avs/probes.c | 3 +- sound/soc/intel/avs/topology.c | 9 +- sound/soc/intel/skylake/skl-pcm.c | 11 ++- sound/soc/soc-pcm.c | 17 ++++ sound/soc/sof/intel/hda-dai-ops.c | 20 ++-- tools/include/uapi/sound/asound.h | 5 +- 28 files changed, 310 insertions(+), 105 deletions(-) base-commit: e69dadf638e625a950b2b91591a47c9ae32a9d5e -- 2.25.1

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[PATCH v2 0/2] A couple of ASoC core fixes
by Cristian Ciocaltea 25 Aug '23

25 Aug '23

This is just a split version of the initial patch [1], as requested by Mark. [1] https://lore.kernel.org/all/20230824193837.369761-1-cristian.ciocaltea@coll… Cristian Ciocaltea (2): ASoC: core: Do not error if a DAI link component is not found ASoC: core: Rename component_not_find label sound/soc/soc-core.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) -- 2.41.0

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[PATCH] ASoC: amd: yc: Fix a non-functional mic on Lenovo 82SJ
by Mario Limonciello 24 Aug '23

24 Aug '23

Lenovo 82SJ doesn't have DMIC connected like 82V2 does. Narrow the match down to only cover 82V2. Reported-by: prosenfeld(a)Yuhsbstudents.org Closes: https://bugzilla.kernel.org/show_bug.cgi?id=217063 Fixes: 2232b2dd8cd4 ("ASoC: amd: yc: Add Lenovo Yoga Slim 7 Pro X to quirks table") Signed-off-by: Mario Limonciello <mario.limonciello(a)amd.com> --- sound/soc/amd/yc/acp6x-mach.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/amd/yc/acp6x-mach.c b/sound/soc/amd/yc/acp6x-mach.c index a2fe3bd4f9a1..0ad1810cf29e 100644 --- a/sound/soc/amd/yc/acp6x-mach.c +++ b/sound/soc/amd/yc/acp6x-mach.c @@ -217,7 +217,7 @@ static const struct dmi_system_id yc_acp_quirk_table[] = { .driver_data = &acp6x_card, .matches = { DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"), - DMI_MATCH(DMI_PRODUCT_NAME, "82"), + DMI_MATCH(DMI_PRODUCT_NAME, "82V2"), } }, { -- 2.34.1

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[PATCH] ASoC: cs42l43: Use new-style PM runtime macros
by Charles Keepax 24 Aug '23

24 Aug '23

Update to the newer style PM runtime macros, no functional change. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- sound/soc/codecs/cs42l43.c | 7 +++---- 1 file changed, 3 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/cs42l43.c b/sound/soc/codecs/cs42l43.c index 18178ffa82f21..24e718e511746 100644 --- a/sound/soc/codecs/cs42l43.c +++ b/sound/soc/codecs/cs42l43.c @@ -2238,7 +2238,7 @@ static int cs42l43_codec_remove(struct platform_device *pdev) return 0; } -static int __maybe_unused cs42l43_codec_runtime_resume(struct device *dev) +static int cs42l43_codec_runtime_resume(struct device *dev) { struct cs42l43_codec *priv = dev_get_drvdata(dev); @@ -2250,9 +2250,8 @@ static int __maybe_unused cs42l43_codec_runtime_resume(struct device *dev) return 0; } -static const struct dev_pm_ops cs42l43_codec_pm_ops = { - SET_RUNTIME_PM_OPS(NULL, cs42l43_codec_runtime_resume, NULL) -}; +DEFINE_RUNTIME_DEV_PM_OPS(cs42l43_codec_pm_ops, NULL, + cs42l43_codec_runtime_resume, NULL); static const struct platform_device_id cs42l43_codec_id_table[] = { { "cs42l43-codec", }, -- 2.30.2

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[PATCH] ASoC: audio-graph-card.c: move audio_graph_parse_of()
by Kuninori Morimoto 24 Aug '23

24 Aug '23

Current audio-graph-card functions definition are like below (A)' static int graph_get_dais_count(); (B) int audio_graph_parse_of(...) { ... (A) ret = graph_get_dais_count(); ... } (A) static int graph_get_dais_count(...) { ... } (C) We don't need to have (A)' definition if audio_graph_parse_of() (B) was defined at (C). This patch moves (B) to (C). Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/generic/audio-graph-card.c | 139 +++++++++++++-------------- 1 file changed, 68 insertions(+), 71 deletions(-) diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c index 13693ef9c242..844a2ef15948 100644 --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -425,77 +425,6 @@ static int graph_for_each_link(struct asoc_simple_priv *priv, return ret; } -static int graph_get_dais_count(struct asoc_simple_priv *priv, - struct link_info *li); - -int audio_graph_parse_of(struct asoc_simple_priv *priv, struct device *dev) -{ - struct snd_soc_card *card = simple_priv_to_card(priv); - struct link_info *li; - int ret; - - li = devm_kzalloc(dev, sizeof(*li), GFP_KERNEL); - if (!li) - return -ENOMEM; - - card->owner = THIS_MODULE; - card->dev = dev; - - ret = graph_get_dais_count(priv, li); - if (ret < 0) - return ret; - - if (!li->link) - return -EINVAL; - - ret = asoc_simple_init_priv(priv, li); - if (ret < 0) - return ret; - - priv->pa_gpio = devm_gpiod_get_optional(dev, "pa", GPIOD_OUT_LOW); - if (IS_ERR(priv->pa_gpio)) { - ret = PTR_ERR(priv->pa_gpio); - dev_err(dev, "failed to get amplifier gpio: %d\n", ret); - return ret; - } - - ret = asoc_simple_parse_widgets(card, NULL); - if (ret < 0) - return ret; - - ret = asoc_simple_parse_routing(card, NULL); - if (ret < 0) - return ret; - - memset(li, 0, sizeof(*li)); - ret = graph_for_each_link(priv, li, - graph_dai_link_of, - graph_dai_link_of_dpcm); - if (ret < 0) - goto err; - - ret = asoc_simple_parse_card_name(card, NULL); - if (ret < 0) - goto err; - - snd_soc_card_set_drvdata(card, priv); - - asoc_simple_debug_info(priv); - - ret = devm_snd_soc_register_card(dev, card); - if (ret < 0) - goto err; - - devm_kfree(dev, li); - return 0; - -err: - asoc_simple_clean_reference(card); - - return dev_err_probe(dev, ret, "parse error\n"); -} -EXPORT_SYMBOL_GPL(audio_graph_parse_of); - static int graph_count_noml(struct asoc_simple_priv *priv, struct device_node *cpu_ep, struct device_node *codec_ep, @@ -612,6 +541,74 @@ static int graph_get_dais_count(struct asoc_simple_priv *priv, graph_count_dpcm); } +int audio_graph_parse_of(struct asoc_simple_priv *priv, struct device *dev) +{ + struct snd_soc_card *card = simple_priv_to_card(priv); + struct link_info *li; + int ret; + + li = devm_kzalloc(dev, sizeof(*li), GFP_KERNEL); + if (!li) + return -ENOMEM; + + card->owner = THIS_MODULE; + card->dev = dev; + + ret = graph_get_dais_count(priv, li); + if (ret < 0) + return ret; + + if (!li->link) + return -EINVAL; + + ret = asoc_simple_init_priv(priv, li); + if (ret < 0) + return ret; + + priv->pa_gpio = devm_gpiod_get_optional(dev, "pa", GPIOD_OUT_LOW); + if (IS_ERR(priv->pa_gpio)) { + ret = PTR_ERR(priv->pa_gpio); + dev_err(dev, "failed to get amplifier gpio: %d\n", ret); + return ret; + } + + ret = asoc_simple_parse_widgets(card, NULL); + if (ret < 0) + return ret; + + ret = asoc_simple_parse_routing(card, NULL); + if (ret < 0) + return ret; + + memset(li, 0, sizeof(*li)); + ret = graph_for_each_link(priv, li, + graph_dai_link_of, + graph_dai_link_of_dpcm); + if (ret < 0) + goto err; + + ret = asoc_simple_parse_card_name(card, NULL); + if (ret < 0) + goto err; + + snd_soc_card_set_drvdata(card, priv); + + asoc_simple_debug_info(priv); + + ret = devm_snd_soc_register_card(dev, card); + if (ret < 0) + goto err; + + devm_kfree(dev, li); + return 0; + +err: + asoc_simple_clean_reference(card); + + return dev_err_probe(dev, ret, "parse error\n"); +} +EXPORT_SYMBOL_GPL(audio_graph_parse_of); + static int graph_probe(struct platform_device *pdev) { struct asoc_simple_priv *priv; -- 2.25.1

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