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

[V2 PATCH 1/2] dt-bindings: Added device tree binding for max98390
by Steve Lee 12 May '20

12 May '20

Add documentation for DT binding of max98390 amplifier driver. Signed-off-by: Steve Lee <steves.lee(a)maximintegrated.com> --- .../devicetree/bindings/sound/max98390.txt | 26 +++++++++++++++++++ 1 file changed, 26 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/max98390.txt diff --git a/Documentation/devicetree/bindings/sound/max98390.txt b/Documentation/devicetree/bindings/sound/max98390.txt new file mode 100644 index 000000000000..0ddd4c6ae55e --- /dev/null +++ b/Documentation/devicetree/bindings/sound/max98390.txt @@ -0,0 +1,26 @@ +Maxim Integrated MAX98390 Speaker Amplifier + +This device supports I2C. + +Required properties: + + - compatible : "maxim,max98390" + + - reg : the I2C address of the device. + +Optional properties: + +- maxim,temperature_calib + u32. The calculated temperature data was measured while doing the calibration. Data : Temp / 100 * 2^12 + +- maxim,r0_calib + u32. This is r0 calibration data which was measured in factory mode. + +Example: + +codec: max98390@38 { + compatible = "maxim,max98390"; + reg = <0x38>; + maxim,temperature_calib = <1024>; + maxim,r0_calib = <100232>; +}; -- 2.17.1

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[PATCH] ALSA: hda/realtek - Add COEF workaround for ASUS ZenBook UX431DA
by Takashi Iwai 12 May '20

12 May '20

ASUS ZenBook UX431DA requires an additional COEF setup when booted from the recent Windows 10, otherwise it produces the noisy output. The quirk turns on COEF 0x1b bit 10 that has been cleared supposedly due to the pop noise reduction. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207553 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 13 +++++++++++++ 1 file changed, 13 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index f1cf2567d33d..4080b492ecc0 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6090,6 +6090,7 @@ enum { ALC294_FIXUP_ASUS_DUAL_SPK, ALC285_FIXUP_THINKPAD_HEADSET_JACK, ALC294_FIXUP_ASUS_HPE, + ALC294_FIXUP_ASUS_COEF_1B, ALC285_FIXUP_HP_GPIO_LED, ALC285_FIXUP_HP_MUTE_LED, ALC236_FIXUP_HP_MUTE_LED, @@ -7222,6 +7223,17 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC294_FIXUP_ASUS_HEADSET_MIC }, + [ALC294_FIXUP_ASUS_COEF_1B] = { + .type = HDA_FIXUP_VERBS, + .v.verbs = (const struct hda_verb[]) { + /* Set bit 10 to correct noisy output after reboot from + * Windows 10 (due to pop noise reduction?) + */ + { 0x20, AC_VERB_SET_COEF_INDEX, 0x1b }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x4e4b }, + { } + }, + }, [ALC285_FIXUP_HP_GPIO_LED] = { .type = HDA_FIXUP_FUNC, .v.func = alc285_fixup_hp_gpio_led, @@ -7420,6 +7432,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x19e1, "ASUS UX581LV", ALC295_FIXUP_ASUS_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1043, 0x1a13, "Asus G73Jw", ALC269_FIXUP_ASUS_G73JW), SND_PCI_QUIRK(0x1043, 0x1a30, "ASUS X705UD", ALC256_FIXUP_ASUS_MIC), + SND_PCI_QUIRK(0x1043, 0x1b11, "ASUS UX431DA", ALC294_FIXUP_ASUS_COEF_1B), SND_PCI_QUIRK(0x1043, 0x1b13, "Asus U41SV", ALC269_FIXUP_INV_DMIC), SND_PCI_QUIRK(0x1043, 0x1bbd, "ASUS Z550MA", ALC255_FIXUP_ASUS_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1043, 0x1c23, "Asus X55U", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), -- 2.16.4

1 0

[PATCH 1/3] ALSA: hda/realtek - Enable headset mic of ASUS GL503VM with ALC295
by Jian-Hong Pan 12 May '20

12 May '20

From: Chris Chiu <chiu(a)endlessm.com> The ASUS laptop GL503VM with ALC295 can't detect the headset microphone. The headset microphone does not work until pin 0x19 is enabled for it. Signed-off-by: Chris Chiu <chiu(a)endlessm.com> Signed-off-by: Daniel Drake <drake(a)endlessm.com> Signed-off-by: Jian-Hong Pan <jian-hong(a)endlessm.com> --- sound/pci/hda/patch_realtek.c | 14 ++++++++++++++ 1 file changed, 14 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index c16f63957c5a..6c996df16201 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6083,6 +6083,7 @@ enum { ALC285_FIXUP_HP_GPIO_LED, ALC285_FIXUP_HP_MUTE_LED, ALC236_FIXUP_HP_MUTE_LED, + ALC295_FIXUP_ASUS_MIC_NO_PRESENCE, }; static const struct hda_fixup alc269_fixups[] = { @@ -7216,6 +7217,15 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc236_fixup_hp_mute_led, }, + [ALC295_FIXUP_ASUS_MIC_NO_PRESENCE] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x19, 0x01a1913c }, /* use as headset mic, without its own jack detect */ + { } + }, + .chained = true, + .chain_id = ALC269_FIXUP_HEADSET_MODE + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -7993,6 +8003,10 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { {0x12, 0x90a60130}, {0x17, 0x90170110}, {0x21, 0x03211020}), + SND_HDA_PIN_QUIRK(0x10ec0295, 0x1043, "ASUS", ALC295_FIXUP_ASUS_MIC_NO_PRESENCE, + {0x12, 0x90a60130}, + {0x17, 0x90170110}, + {0x21, 0x03211020}), SND_HDA_PIN_QUIRK(0x10ec0295, 0x1028, "Dell", ALC269_FIXUP_DELL4_MIC_NO_PRESENCE, {0x14, 0x90170110}, {0x21, 0x04211020}), -- 2.26.2

2 5

[PATCH] ASoC: rsnd: add interrupt support for SSI BUSIF buffer
by Yongbo Zhang 12 May '20

12 May '20

SSI BUSIF buffer is possible to overflow or underflow, especially in a hypervisor environment. If there is no interrupt support, it will eventually lead to errors in pcm data. This patch adds overflow and underflow interrupt support for SSI BUSIF buffer. Cc: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Reported-by: Chen Li <licheng0822(a)thundersoft.com> Signed-off-by: Yongbo Zhang <giraffesnn123(a)gmail.com> Tested-by: Chen Li <licheng0822(a)thundersoft.com> --- sound/soc/sh/rcar/gen.c | 8 +++ sound/soc/sh/rcar/rsnd.h | 9 +++ sound/soc/sh/rcar/ssi.c | 142 +++++++++++++++++++++++++++++++++++++++ 3 files changed, 159 insertions(+) diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c index af19010b9d88..8bd49c8a9517 100644 --- a/sound/soc/sh/rcar/gen.c +++ b/sound/soc/sh/rcar/gen.c @@ -224,6 +224,14 @@ static int rsnd_gen2_probe(struct rsnd_priv *priv) RSND_GEN_S_REG(SSI_SYS_STATUS5, 0x884), RSND_GEN_S_REG(SSI_SYS_STATUS6, 0x888), RSND_GEN_S_REG(SSI_SYS_STATUS7, 0x88c), + RSND_GEN_S_REG(SSI_SYS_INT_ENABLE0, 0x850), + RSND_GEN_S_REG(SSI_SYS_INT_ENABLE1, 0x854), + RSND_GEN_S_REG(SSI_SYS_INT_ENABLE2, 0x858), + RSND_GEN_S_REG(SSI_SYS_INT_ENABLE3, 0x85c), + RSND_GEN_S_REG(SSI_SYS_INT_ENABLE4, 0x890), + RSND_GEN_S_REG(SSI_SYS_INT_ENABLE5, 0x894), + RSND_GEN_S_REG(SSI_SYS_INT_ENABLE6, 0x898), + RSND_GEN_S_REG(SSI_SYS_INT_ENABLE7, 0x89c), RSND_GEN_S_REG(HDMI0_SEL, 0x9e0), RSND_GEN_S_REG(HDMI1_SEL, 0x9e4), diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index ea6cbaa9743e..d47608ff5fac 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -189,6 +189,14 @@ enum rsnd_reg { SSI_SYS_STATUS5, SSI_SYS_STATUS6, SSI_SYS_STATUS7, + SSI_SYS_INT_ENABLE0, + SSI_SYS_INT_ENABLE1, + SSI_SYS_INT_ENABLE2, + SSI_SYS_INT_ENABLE3, + SSI_SYS_INT_ENABLE4, + SSI_SYS_INT_ENABLE5, + SSI_SYS_INT_ENABLE6, + SSI_SYS_INT_ENABLE7, HDMI0_SEL, HDMI1_SEL, SSI9_BUSIF0_MODE, @@ -237,6 +245,7 @@ enum rsnd_reg { #define SSI9_BUSIF_ADINR(i) (SSI9_BUSIF0_ADINR + (i)) #define SSI9_BUSIF_DALIGN(i) (SSI9_BUSIF0_DALIGN + (i)) #define SSI_SYS_STATUS(i) (SSI_SYS_STATUS0 + (i)) +#define SSI_SYS_INT_ENABLE(i) (SSI_SYS_INT_ENABLE0 + (i)) struct rsnd_priv; diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 4a7d3413917f..178f50a91b3e 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -372,6 +372,9 @@ static void rsnd_ssi_config_init(struct rsnd_mod *mod, u32 wsr = ssi->wsr; int width; int is_tdm, is_tdm_split; + int id = rsnd_mod_id(mod); + int i; + u32 sys_int_enable = 0; is_tdm = rsnd_runtime_is_tdm(io); is_tdm_split = rsnd_runtime_is_tdm_split(io); @@ -447,6 +450,38 @@ static void rsnd_ssi_config_init(struct rsnd_mod *mod, cr_mode = DIEN; /* PIO : enable Data interrupt */ } + /* enable busif buffer over/under run interrupt. */ + if (is_tdm || is_tdm_split) { + switch (id) { + case 0: + case 1: + case 2: + case 3: + case 4: + for (i = 0; i < 4; i++) { + sys_int_enable = rsnd_mod_read(mod, + SSI_SYS_INT_ENABLE(i * 2)); + sys_int_enable |= 0xf << (id * 4); + rsnd_mod_write(mod, + SSI_SYS_INT_ENABLE(i * 2), + sys_int_enable); + } + + break; + case 9: + for (i = 0; i < 4; i++) { + sys_int_enable = rsnd_mod_read(mod, + SSI_SYS_INT_ENABLE((i * 2) + 1)); + sys_int_enable |= 0xf << 4; + rsnd_mod_write(mod, + SSI_SYS_INT_ENABLE((i * 2) + 1), + sys_int_enable); + } + + break; + } + } + init_end: ssi->cr_own = cr_own; ssi->cr_mode = cr_mode; @@ -496,6 +531,13 @@ static int rsnd_ssi_quit(struct rsnd_mod *mod, { struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod); struct device *dev = rsnd_priv_to_dev(priv); + int is_tdm, is_tdm_split; + int id = rsnd_mod_id(mod); + int i; + u32 sys_int_enable = 0; + + is_tdm = rsnd_runtime_is_tdm(io); + is_tdm_split = rsnd_runtime_is_tdm_split(io); if (!rsnd_ssi_is_run_mods(mod, io)) return 0; @@ -517,6 +559,37 @@ static int rsnd_ssi_quit(struct rsnd_mod *mod, ssi->wsr = 0; } + /* disable busif buffer over/under run interrupt. */ + if (is_tdm || is_tdm_split) { + switch (id) { + case 0: + case 1: + case 2: + case 3: + case 4: + for (i = 0; i < 4; i++) { + sys_int_enable = rsnd_mod_read(mod, + SSI_SYS_INT_ENABLE(i * 2)); + sys_int_enable &= ~(0xf << (id * 4)); + rsnd_mod_write(mod, + SSI_SYS_INT_ENABLE(i * 2), + sys_int_enable); + } + + break; + case 9: + for (i = 0; i < 4; i++) { + sys_int_enable = rsnd_mod_read(mod, + SSI_SYS_INT_ENABLE((i * 2) + 1)); + sys_int_enable &= ~(0xf << 4); + rsnd_mod_write(mod, + SSI_SYS_INT_ENABLE((i * 2) + 1), + sys_int_enable); + } + + break; + } + return 0; } @@ -622,6 +695,11 @@ static int rsnd_ssi_irq(struct rsnd_mod *mod, int enable) { u32 val = 0; + int is_tdm, is_tdm_split; + int id = rsnd_mod_id(mod); + + is_tdm = rsnd_runtime_is_tdm(io); + is_tdm_split = rsnd_runtime_is_tdm_split(io); if (rsnd_is_gen1(priv)) return 0; @@ -635,6 +713,19 @@ static int rsnd_ssi_irq(struct rsnd_mod *mod, if (enable) val = rsnd_ssi_is_dma_mode(mod) ? 0x0e000000 : 0x0f000000; + if (is_tdm || is_tdm_split) { + switch (id) { + case 0: + case 1: + case 2: + case 3: + case 4: + case 9: + val |= 0xff00; + break; + } + } + rsnd_mod_write(mod, SSI_INT_ENABLE, val); return 0; @@ -651,6 +742,10 @@ static void __rsnd_ssi_interrupt(struct rsnd_mod *mod, u32 status; bool elapsed = false; bool stop = false; + int is_tdm, is_tdm_split; + + is_tdm = rsnd_runtime_is_tdm(io); + is_tdm_split = rsnd_runtime_is_tdm_split(io); spin_lock(&priv->lock); @@ -672,6 +767,53 @@ static void __rsnd_ssi_interrupt(struct rsnd_mod *mod, stop = true; } + status = 0; + + if (is_tdm || is_tdm_split) { + switch (id) { + case 0: + case 1: + case 2: + case 3: + case 4: + for (i = 0; i < 4; i++) { + status = rsnd_mod_read(mod, + SSI_SYS_STATUS(i * 2)); + status &= 0xf << (id * 4); + + if (status) { + rsnd_dbg_irq_status(dev, + "%s err status : 0x%08x\n", + rsnd_mod_name(mod), status); + rsnd_mod_write(mod, + SSI_SYS_STATUS(i * 2), + 0xf << (id * 4)); + stop = true; + break; + } + } + break; + case 9: + for (i = 0; i < 4; i++) { + status = rsnd_mod_write(mod, + SSI_SYS_STATUS((i * 2) + 1)); + status &= 0xf << 4; + + if (status) { + rsnd_dbg_irq_status(dev, + "%s err status : 0x%08x\n", + rsnd_mod_name(mod), status); + rsnd_mod_write(mod, + SSI_SYS_STATUS((i * 2) + 1), + 0xf << 4); + stop = true; + break; + } + } + break; + } + } + rsnd_ssi_status_clear(mod); rsnd_ssi_interrupt_out: spin_unlock(&priv->lock); -- 2.26.2

3 4

[PATCH 00/17] ASoC: cleanup DAI/Component activity
by Kuninori Morimoto 12 May '20

12 May '20

Hi Mark This patch-set cleanups DAI/Component activity operation. I believe these works correctly, but maybe need some tests or deep review. [17/17] patch can be apply after [03/17] patch. Kuninori Morimoto (17): 1 ASoC: soc-dai: add snd_soc_dai_activity() 2 ASoC: use snd_soc_dai/component_activity() 3 ASoC: soc-dai: add snd_soc_component_activity() 4 ASoC: atomel: use snd_soc_dai/component_activity() 5 ASoC: bcm: use snd_soc_dai/component_activity() 6 ASoC: cirrus: use snd_soc_dai/component_activity() 7 ASoC: codecs: use snd_soc_dai/component_activity() 8 ASoC: fsl: use snd_soc_dai/component_activity() 9 ASoC: intel: use snd_soc_dai/component_activity() 10 ASoC: jz4740: use snd_soc_dai/component_activity() 11 ASoC: mediatek: use snd_soc_dai/component_activity() 12 ASoC: meson: use snd_soc_dai/component_activity() 13 ASoC: pxa: use snd_soc_dai/component_activity() 14 ASoC: ti: use snd_soc_dai/component_activity() 15 ASoC: uniphier: use snd_soc_dai/component_activity() 16 ASoC: cleanup dai / component active code 17 ASoC: soc-dai: add snd_soc_dai_stream_activity() include/sound/soc-component.h | 7 +--- include/sound/soc-dai.h | 5 ++- sound/soc/atmel/atmel_ssc_dai.c | 4 +-- sound/soc/bcm/bcm2835-i2s.c | 6 ++-- sound/soc/bcm/cygnus-ssp.c | 4 +-- sound/soc/cirrus/ep93xx-i2s.c | 4 +-- sound/soc/codecs/adav80x.c | 4 +-- sound/soc/codecs/arizona.c | 2 +- sound/soc/codecs/cs4271.c | 4 +-- sound/soc/codecs/madera.c | 2 +- sound/soc/codecs/max98090.c | 6 ++-- sound/soc/codecs/tlv320aic23.c | 2 +- sound/soc/codecs/tlv320dac33.c | 2 +- sound/soc/codecs/uda1380.c | 2 +- sound/soc/codecs/wl1273.c | 2 +- sound/soc/codecs/wm8711.c | 2 +- sound/soc/codecs/wm8753.c | 4 +-- sound/soc/dwc/dwc-i2s.c | 2 +- sound/soc/fsl/fsl_esai.c | 2 +- sound/soc/fsl/fsl_spdif.c | 4 +-- sound/soc/intel/atom/sst-mfld-platform-pcm.c | 14 ++++---- sound/soc/jz4740/jz4740-i2s.c | 8 ++--- sound/soc/mediatek/mt8173/mt8173-afe-pcm.c | 8 ++--- sound/soc/meson/axg-tdm-interface.c | 2 +- sound/soc/pxa/pxa-ssp.c | 8 ++--- sound/soc/pxa/pxa2xx-i2s.c | 2 +- sound/soc/soc-compress.c | 4 +-- sound/soc/soc-core.c | 11 ++++--- sound/soc/soc-dai.c | 12 +++++++ sound/soc/soc-dapm.c | 8 ++--- sound/soc/soc-pcm.c | 34 ++++++++++---------- sound/soc/ti/davinci-mcasp.c | 2 +- sound/soc/ti/omap-dmic.c | 4 +-- sound/soc/ti/omap-mcbsp.c | 4 +-- sound/soc/ti/omap-mcpdm.c | 8 ++--- sound/soc/uniphier/aio-cpu.c | 4 +-- 36 files changed, 105 insertions(+), 98 deletions(-) -- 2.17.1

3 25

alsa-gobject: seq: user_client: remove useless argument of delete_port()
by GitHub pull_request - opened 12 May '20

12 May '20

alsa-project/alsa-gobject pull request #21 was opened from takaswie: In implementation of ALSA sequencer core, the port owned by client can be deleted by userspace applications which opens the client, therefore it's necessary and sufficient to have an argument just for the numerical ID of port. This commit removes the useless argument for port information. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> Request URL : https://github.com/alsa-project/alsa-gobject/pull/21 Patch URL : https://github.com/alsa-project/alsa-gobject/pull/21.patch Repository URL: https://github.com/alsa-project/alsa-gobject

1 0

[alsa-devel] [PATCH] sound/soc: only first codec is master in multicodec setup
by Johannes Krude 11 May '20

11 May '20

When using multiple codecs, at most one codec should generate the master clock. All codecs except the first are therefore configured for slave mode. Before this patch all codecs in a multicodec setup had to be slaves. This is needed when e.g., connecting multiple sound hats for simultaneous playback to the raspberry pi I2S output and one of the sound hats generates the I2S clocks (https://github.com/raspberrypi/linux/pull/3337). I checked the raspberry pi kernel tree for multicodec usage with `fgrep num_codecs -R sound/soc/` and verified that all existing multicodec drivers with hardcoded format have indeed configured all codecs for slave mode. Doing a similar check on the current for-5.5 tree is more difficult since .num_codecs is now hidden behind a preprocessor macro. Signed-off-by: Johannes Krude <johannes(a)krude.de> --- sound/soc/soc-core.c | 9 ++++++++- 1 file changed, 8 insertions(+), 1 deletion(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 058e038df..cb8952527 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1688,6 +1688,14 @@ int snd_soc_runtime_set_dai_fmt(struct snd_soc_pcm_runtime *rtd, for_each_rtd_codec_dai(rtd, i, codec_dai) { + unsigned int codec_dai_fmt = dai_fmt; + - ret = snd_soc_dai_set_fmt(codec_dai, dai_fmt); + // there can only be one master when using multiple codecs + if (i && (codec_dai_fmt & SND_SOC_DAIFMT_MASTER_MASK)) { + codec_dai_fmt &= ~SND_SOC_DAIFMT_MASTER_MASK; + codec_dai_fmt |= SND_SOC_DAIFMT_CBS_CFS; + } + + ret = snd_soc_dai_set_fmt(codec_dai, codec_dai_fmt); if (ret != 0 && ret != -ENOTSUPP) { dev_warn(codec_dai->dev, "ASoC: Failed to set DAI format: %d\n", ret); -- 2.17.1

2 2

[PATCH alsa-lib 0/3] Trivial compile warning fixes
by Takashi Iwai 11 May '20

11 May '20

Hi, here is a few patches to shut up trivial compile warnings that have been introduced recently. Takashi === Takashi Iwai (3): pcm: rate: Fix compile warning wrt bit ops and comparison pcm: rate: Fix uninitialized variable warning topology: Add missing ATTRIBUTE_UNUSED src/pcm/pcm_rate.c | 4 ++-- src/topology/pcm.c | 18 ++++++++++-------- 2 files changed, 12 insertions(+), 10 deletions(-) -- 2.16.4

2 4

[PATCH] ASoC: mediatek: mt8183-da7219: set headset button maps
by Tzung-Bi Shih 11 May '20

11 May '20

Sets headset button maps. Signed-off-by: Tzung-Bi Shih <tzungbi(a)google.com> --- .../soc/mediatek/mt8183/mt8183-da7219-max98357.c | 16 +++++++++++++--- 1 file changed, 13 insertions(+), 3 deletions(-) diff --git a/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c b/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c index 5b3dfa79b4ae..ffd7c931e7bb 100644 --- a/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c +++ b/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c @@ -6,11 +6,12 @@ // Copyright (c) 2018 MediaTek Inc. // Author: Shunli Wang <shunli.wang(a)mediatek.com> +#include <linux/input.h> #include <linux/module.h> +#include <linux/pinctrl/consumer.h> +#include <sound/jack.h> #include <sound/pcm_params.h> #include <sound/soc.h> -#include <sound/jack.h> -#include <linux/pinctrl/consumer.h> #include "mt8183-afe-common.h" #include "../../codecs/da7219-aad.h" @@ -471,9 +472,18 @@ mt8183_da7219_max98357_headset_init(struct snd_soc_component *component) if (ret) return ret; + snd_jack_set_key( + priv->headset_jack.jack, SND_JACK_BTN_0, KEY_PLAYPAUSE); + snd_jack_set_key( + priv->headset_jack.jack, SND_JACK_BTN_1, KEY_VOLUMEUP); + snd_jack_set_key( + priv->headset_jack.jack, SND_JACK_BTN_2, KEY_VOLUMEDOWN); + snd_jack_set_key( + priv->headset_jack.jack, SND_JACK_BTN_3, KEY_VOICECOMMAND); + da7219_aad_jack_det(component, &priv->headset_jack); - return ret; + return 0; } static int mt8183_da7219_max98357_dev_probe(struct platform_device *pdev) -- 2.26.2.645.ge9eca65c58-goog

2 1

[PATCH -next] ASoC: SOF: core: fix error return code in sof_probe_continue()
by Wei Yongjun 11 May '20

11 May '20

Fix to return negative error code -ENOMEM from the IPC init error handling case instead of 0, as done elsewhere in this function. Fixes: c16211d6226d ("ASoC: SOF: Add Sound Open Firmware driver core") Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: Wei Yongjun <weiyongjun1(a)huawei.com> --- sound/soc/sof/core.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/sof/core.c b/sound/soc/sof/core.c index 94a2cb58ab9a..ef9be4f45e27 100644 --- a/sound/soc/sof/core.c +++ b/sound/soc/sof/core.c @@ -176,6 +176,7 @@ static int sof_probe_continue(struct snd_sof_dev *sdev) /* init the IPC */ sdev->ipc = snd_sof_ipc_init(sdev); if (!sdev->ipc) { + ret = -ENOMEM; dev_err(sdev->dev, "error: failed to init DSP IPC %d\n", ret); goto ipc_err; }

2 1