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[PATCH AUTOSEL 5.15 09/41] ASoC: cs53l30: Correct number of volume levels on SX controls
by Sasha Levin 14 Jun '22

14 Jun '22

From: Charles Keepax <ckeepax(a)opensource.cirrus.com> [ Upstream commit 7fbd6dd68127927e844912a16741016d432a0737 ] This driver specified the maximum value rather than the number of volume levels on the SX controls, this is incorrect, so correct them. Reported-by: David Rhodes <david.rhodes(a)cirrus.com> Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220602162119.3393857-4-ckeepax@opensource.cirru… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/cs53l30.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/cs53l30.c b/sound/soc/codecs/cs53l30.c index f2087bd38dbc..c2912ad3851b 100644 --- a/sound/soc/codecs/cs53l30.c +++ b/sound/soc/codecs/cs53l30.c @@ -348,22 +348,22 @@ static const struct snd_kcontrol_new cs53l30_snd_controls[] = { SOC_ENUM("ADC2 NG Delay", adc2_ng_delay_enum), SOC_SINGLE_SX_TLV("ADC1A PGA Volume", - CS53L30_ADC1A_AFE_CTL, 0, 0x34, 0x18, pga_tlv), + CS53L30_ADC1A_AFE_CTL, 0, 0x34, 0x24, pga_tlv), SOC_SINGLE_SX_TLV("ADC1B PGA Volume", - CS53L30_ADC1B_AFE_CTL, 0, 0x34, 0x18, pga_tlv), + CS53L30_ADC1B_AFE_CTL, 0, 0x34, 0x24, pga_tlv), SOC_SINGLE_SX_TLV("ADC2A PGA Volume", - CS53L30_ADC2A_AFE_CTL, 0, 0x34, 0x18, pga_tlv), + CS53L30_ADC2A_AFE_CTL, 0, 0x34, 0x24, pga_tlv), SOC_SINGLE_SX_TLV("ADC2B PGA Volume", - CS53L30_ADC2B_AFE_CTL, 0, 0x34, 0x18, pga_tlv), + CS53L30_ADC2B_AFE_CTL, 0, 0x34, 0x24, pga_tlv), SOC_SINGLE_SX_TLV("ADC1A Digital Volume", - CS53L30_ADC1A_DIG_VOL, 0, 0xA0, 0x0C, dig_tlv), + CS53L30_ADC1A_DIG_VOL, 0, 0xA0, 0x6C, dig_tlv), SOC_SINGLE_SX_TLV("ADC1B Digital Volume", - CS53L30_ADC1B_DIG_VOL, 0, 0xA0, 0x0C, dig_tlv), + CS53L30_ADC1B_DIG_VOL, 0, 0xA0, 0x6C, dig_tlv), SOC_SINGLE_SX_TLV("ADC2A Digital Volume", - CS53L30_ADC2A_DIG_VOL, 0, 0xA0, 0x0C, dig_tlv), + CS53L30_ADC2A_DIG_VOL, 0, 0xA0, 0x6C, dig_tlv), SOC_SINGLE_SX_TLV("ADC2B Digital Volume", - CS53L30_ADC2B_DIG_VOL, 0, 0xA0, 0x0C, dig_tlv), + CS53L30_ADC2B_DIG_VOL, 0, 0xA0, 0x6C, dig_tlv), }; static const struct snd_soc_dapm_widget cs53l30_dapm_widgets[] = { -- 2.35.1

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[PATCH AUTOSEL 5.15 08/41] ASoC: cs35l36: Update digital volume TLV
by Sasha Levin 14 Jun '22

14 Jun '22

From: Charles Keepax <ckeepax(a)opensource.cirrus.com> [ Upstream commit 5005a2345825eb8346546d99bfe669f73111b5c5 ] The digital volume TLV specifies the step as 0.25dB but the actual step of the control is 0.125dB. Update the TLV to correct this. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220602162119.3393857-3-ckeepax@opensource.cirru… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/cs35l36.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs35l36.c b/sound/soc/codecs/cs35l36.c index d83c1b318c1c..0accdb45ed72 100644 --- a/sound/soc/codecs/cs35l36.c +++ b/sound/soc/codecs/cs35l36.c @@ -444,7 +444,8 @@ static bool cs35l36_volatile_reg(struct device *dev, unsigned int reg) } } -static DECLARE_TLV_DB_SCALE(dig_vol_tlv, -10200, 25, 0); +static const DECLARE_TLV_DB_RANGE(dig_vol_tlv, 0, 912, + TLV_DB_MINMAX_ITEM(-10200, 1200)); static DECLARE_TLV_DB_SCALE(amp_gain_tlv, 0, 1, 1); static const char * const cs35l36_pcm_sftramp_text[] = { -- 2.35.1

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[PATCH AUTOSEL 5.15 07/41] ASoC: cs42l52: Fix TLV scales for mixer controls
by Sasha Levin 14 Jun '22

14 Jun '22

From: Charles Keepax <ckeepax(a)opensource.cirrus.com> [ Upstream commit 8bf5aabf524eec61013e506f764a0b2652dc5665 ] The datasheet specifies the range of the mixer volumes as between -51.5dB and 12dB with a 0.5dB step. Update the TLVs for this. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220602162119.3393857-2-ckeepax@opensource.cirru… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/cs42l52.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index 80161151b3f2..785caba3f653 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -137,7 +137,7 @@ static DECLARE_TLV_DB_SCALE(mic_tlv, 1600, 100, 0); static DECLARE_TLV_DB_SCALE(pga_tlv, -600, 50, 0); -static DECLARE_TLV_DB_SCALE(mix_tlv, -50, 50, 0); +static DECLARE_TLV_DB_SCALE(mix_tlv, -5150, 50, 0); static DECLARE_TLV_DB_SCALE(beep_tlv, -56, 200, 0); @@ -364,7 +364,7 @@ static const struct snd_kcontrol_new cs42l52_snd_controls[] = { CS42L52_ADCB_VOL, 0, 0xA0, 0x78, ipd_tlv), SOC_DOUBLE_R_SX_TLV("ADC Mixer Volume", CS42L52_ADCA_MIXER_VOL, CS42L52_ADCB_MIXER_VOL, - 0, 0x19, 0x7F, ipd_tlv), + 0, 0x19, 0x7F, mix_tlv), SOC_DOUBLE("ADC Switch", CS42L52_ADC_MISC_CTL, 0, 1, 1, 0), -- 2.35.1

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[PATCH AUTOSEL 5.15 02/41] ASoC: nau8822: Add operation for internal PLL off and on
by Sasha Levin 14 Jun '22

14 Jun '22

From: Hui Wang <hui.wang(a)canonical.com> [ Upstream commit aeca8a3295022bcec46697f16e098140423d8463 ] We tried to enable the audio on an imx6sx EVB with the codec nau8822, after setting the internal PLL fractional parameters, the audio still couldn't work and the there was no sdma irq at all. After checking with the section "8.1.1 Phase Locked Loop (PLL) Design Example" of "NAU88C22 Datasheet Rev 0.6", we found we need to turn off the PLL before programming fractional parameters and turn on the PLL after programming. After this change, the audio driver could record and play sound and the sdma's irq is triggered when playing or recording. Cc: David Lin <ctlin0(a)nuvoton.com> Cc: John Hsu <kchsu0(a)nuvoton.com> Cc: Seven Li <wtli(a)nuvoton.com> Signed-off-by: Hui Wang <hui.wang(a)canonical.com> Link: https://lore.kernel.org/r/20220530040151.95221-2-hui.wang@canonical.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/nau8822.c | 4 ++++ sound/soc/codecs/nau8822.h | 3 +++ 2 files changed, 7 insertions(+) diff --git a/sound/soc/codecs/nau8822.c b/sound/soc/codecs/nau8822.c index 58123390c7a3..b436e532993d 100644 --- a/sound/soc/codecs/nau8822.c +++ b/sound/soc/codecs/nau8822.c @@ -740,6 +740,8 @@ static int nau8822_set_pll(struct snd_soc_dai *dai, int pll_id, int source, pll_param->pll_int, pll_param->pll_frac, pll_param->mclk_scaler, pll_param->pre_factor); + snd_soc_component_update_bits(component, + NAU8822_REG_POWER_MANAGEMENT_1, NAU8822_PLL_EN_MASK, NAU8822_PLL_OFF); snd_soc_component_update_bits(component, NAU8822_REG_PLL_N, NAU8822_PLLMCLK_DIV2 | NAU8822_PLLN_MASK, (pll_param->pre_factor ? NAU8822_PLLMCLK_DIV2 : 0) | @@ -757,6 +759,8 @@ static int nau8822_set_pll(struct snd_soc_dai *dai, int pll_id, int source, pll_param->mclk_scaler << NAU8822_MCLKSEL_SFT); snd_soc_component_update_bits(component, NAU8822_REG_CLOCKING, NAU8822_CLKM_MASK, NAU8822_CLKM_PLL); + snd_soc_component_update_bits(component, + NAU8822_REG_POWER_MANAGEMENT_1, NAU8822_PLL_EN_MASK, NAU8822_PLL_ON); return 0; } diff --git a/sound/soc/codecs/nau8822.h b/sound/soc/codecs/nau8822.h index 489191ff187e..b45d42c15de6 100644 --- a/sound/soc/codecs/nau8822.h +++ b/sound/soc/codecs/nau8822.h @@ -90,6 +90,9 @@ #define NAU8822_REFIMP_3K 0x3 #define NAU8822_IOBUF_EN (0x1 << 2) #define NAU8822_ABIAS_EN (0x1 << 3) +#define NAU8822_PLL_EN_MASK (0x1 << 5) +#define NAU8822_PLL_ON (0x1 << 5) +#define NAU8822_PLL_OFF (0x0 << 5) /* NAU8822_REG_AUDIO_INTERFACE (0x4) */ #define NAU8822_AIFMT_MASK (0x3 << 3) -- 2.35.1

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[PATCH AUTOSEL 5.17 27/43] ALSA: hda/realtek - Add HW8326 support
by Sasha Levin 14 Jun '22

14 Jun '22

From: huangwenhui <huangwenhuia(a)uniontech.com> [ Upstream commit 527f4643e03c298c1e3321cfa27866b1374a55e1 ] Added the support of new Huawei codec HW8326. The HW8326 is developed by Huawei with Realtek's IP Core, and it's compatible with ALC256. Signed-off-by: huangwenhui <huangwenhuia(a)uniontech.com> Link: https://lore.kernel.org/r/20220608082357.26898-1-huangwenhuia@uniontech.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/hda/hdac_device.c | 1 + sound/pci/hda/patch_realtek.c | 14 ++++++++++++++ 2 files changed, 15 insertions(+) diff --git a/sound/hda/hdac_device.c b/sound/hda/hdac_device.c index 3e9e9ac804f6..b7e5032b61c9 100644 --- a/sound/hda/hdac_device.c +++ b/sound/hda/hdac_device.c @@ -660,6 +660,7 @@ static const struct hda_vendor_id hda_vendor_ids[] = { { 0x14f1, "Conexant" }, { 0x17e8, "Chrontel" }, { 0x1854, "LG" }, + { 0x19e5, "Huawei" }, { 0x1aec, "Wolfson Microelectronics" }, { 0x1af4, "QEMU" }, { 0x434d, "C-Media" }, diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 112ecc256b14..7e7e517d32c6 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -443,6 +443,7 @@ static void alc_fill_eapd_coef(struct hda_codec *codec) case 0x10ec0245: case 0x10ec0255: case 0x10ec0256: + case 0x19e58326: case 0x10ec0257: case 0x10ec0282: case 0x10ec0283: @@ -580,6 +581,7 @@ static void alc_shutup_pins(struct hda_codec *codec) switch (codec->core.vendor_id) { case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: case 0x10ec0283: case 0x10ec0286: case 0x10ec0288: @@ -3247,6 +3249,7 @@ static void alc_disable_headset_jack_key(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_write_coef_idx(codec, 0x48, 0x0); alc_update_coef_idx(codec, 0x49, 0x0045, 0x0); break; @@ -3275,6 +3278,7 @@ static void alc_enable_headset_jack_key(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_write_coef_idx(codec, 0x48, 0xd011); alc_update_coef_idx(codec, 0x49, 0x007f, 0x0045); break; @@ -4910,6 +4914,7 @@ static void alc_headset_mode_unplugged(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_process_coef_fw(codec, coef0256); break; case 0x10ec0234: @@ -5025,6 +5030,7 @@ static void alc_headset_mode_mic_in(struct hda_codec *codec, hda_nid_t hp_pin, case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_write_coef_idx(codec, 0x45, 0xc489); snd_hda_set_pin_ctl_cache(codec, hp_pin, 0); alc_process_coef_fw(codec, coef0256); @@ -5175,6 +5181,7 @@ static void alc_headset_mode_default(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_write_coef_idx(codec, 0x1b, 0x0e4b); alc_write_coef_idx(codec, 0x45, 0xc089); msleep(50); @@ -5274,6 +5281,7 @@ static void alc_headset_mode_ctia(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_process_coef_fw(codec, coef0256); break; case 0x10ec0234: @@ -5388,6 +5396,7 @@ static void alc_headset_mode_omtp(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_process_coef_fw(codec, coef0256); break; case 0x10ec0234: @@ -5489,6 +5498,7 @@ static void alc_determine_headset_type(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_write_coef_idx(codec, 0x1b, 0x0e4b); alc_write_coef_idx(codec, 0x06, 0x6104); alc_write_coefex_idx(codec, 0x57, 0x3, 0x09a3); @@ -5783,6 +5793,7 @@ static void alc255_set_default_jack_type(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_process_coef_fw(codec, alc256fw); break; } @@ -6385,6 +6396,7 @@ static void alc_combo_jack_hp_jd_restart(struct hda_codec *codec) case 0x10ec0236: case 0x10ec0255: case 0x10ec0256: + case 0x19e58326: alc_update_coef_idx(codec, 0x1b, 0x8000, 1 << 15); /* Reset HP JD */ alc_update_coef_idx(codec, 0x1b, 0x8000, 0 << 15); break; @@ -10092,6 +10104,7 @@ static int patch_alc269(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: spec->codec_variant = ALC269_TYPE_ALC256; spec->shutup = alc256_shutup; spec->init_hook = alc256_init; @@ -11542,6 +11555,7 @@ static const struct hda_device_id snd_hda_id_realtek[] = { HDA_CODEC_ENTRY(0x10ec0b00, "ALCS1200A", patch_alc882), HDA_CODEC_ENTRY(0x10ec1168, "ALC1220", patch_alc882), HDA_CODEC_ENTRY(0x10ec1220, "ALC1220", patch_alc882), + HDA_CODEC_ENTRY(0x19e58326, "HW8326", patch_alc269), {} /* terminator */ }; MODULE_DEVICE_TABLE(hdaudio, snd_hda_id_realtek); -- 2.35.1

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[PATCH AUTOSEL 5.17 18/43] ASoC: wm_adsp: Fix event generation for wm_adsp_fw_put()
by Sasha Levin 14 Jun '22

14 Jun '22

From: Mark Brown <broonie(a)kernel.org> [ Upstream commit 2abdf9f80019e8244d3806ed0e1c9f725e50b452 ] Currently wm_adsp_fw_put() returns 0 rather than 1 when updating the value of the control, meaning that no event is generated to userspace. Fix this by setting the default return value to 1, the code already exits early with a return value of 0 if the value is unchanged. Signed-off-by: Mark Brown <broonie(a)kernel.org> Reviewed-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220603115003.3865834-1-broonie@kernel.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/wm_adsp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 0582585236a2..93677b9f3a68 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -328,7 +328,7 @@ int wm_adsp_fw_put(struct snd_kcontrol *kcontrol, struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; struct wm_adsp *dsp = snd_soc_component_get_drvdata(component); - int ret = 0; + int ret = 1; if (ucontrol->value.enumerated.item[0] == dsp[e->shift_l].fw) return 0; -- 2.35.1

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[PATCH AUTOSEL 5.17 17/43] ASoC: es8328: Fix event generation for deemphasis control
by Sasha Levin 14 Jun '22

14 Jun '22

From: Mark Brown <broonie(a)kernel.org> [ Upstream commit 8259610c2ec01c5cbfb61882ae176aabacac9c19 ] Currently the put() method for the deemphasis control returns 0 when a new value is written to the control even if the value changed, meaning events are not generated. Fix this, skip the work of updating the value when it is unchanged and then return 1 after having done so. Signed-off-by: Mark Brown <broonie(a)kernel.org> Link: https://lore.kernel.org/r/20220603123937.4013603-1-broonie@kernel.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/es8328.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/es8328.c b/sound/soc/codecs/es8328.c index 9632afc2d4d6..ca3b1c00fa78 100644 --- a/sound/soc/codecs/es8328.c +++ b/sound/soc/codecs/es8328.c @@ -161,13 +161,16 @@ static int es8328_put_deemph(struct snd_kcontrol *kcontrol, if (deemph > 1) return -EINVAL; + if (es8328->deemph == deemph) + return 0; + ret = es8328_set_deemph(component); if (ret < 0) return ret; es8328->deemph = deemph; - return 0; + return 1; } -- 2.35.1

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[PATCH AUTOSEL 5.17 16/43] ASoC: wm8962: Fix suspend while playing music
by Sasha Levin 14 Jun '22

14 Jun '22

From: Adam Ford <aford173(a)gmail.com> [ Upstream commit d1f5272c0f7d2e53c6f2480f46725442776f5f78 ] If the audio CODEC is playing sound when the system is suspended, it can be left in a state which throws the following error: wm8962 3-001a: ASoC: error at soc_component_read_no_lock on wm8962.3-001a: -16 Once this error has occurred, the audio will not work again until rebooted. Fix this by configuring SET_SYSTEM_SLEEP_PM_OPS. Signed-off-by: Adam Ford <aford173(a)gmail.com> Acked-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220526182129.538472-1-aford173@gmail.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/wm8962.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index a5584ba962dc..e97eaca7049f 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3858,6 +3858,7 @@ static int wm8962_runtime_suspend(struct device *dev) #endif static const struct dev_pm_ops wm8962_pm = { + SET_SYSTEM_SLEEP_PM_OPS(pm_runtime_force_suspend, pm_runtime_force_resume) SET_RUNTIME_PM_OPS(wm8962_runtime_suspend, wm8962_runtime_resume, NULL) }; -- 2.35.1

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[PATCH AUTOSEL 5.17 12/43] ASoC: cs42l51: Correct minimum value for SX volume control
by Sasha Levin 14 Jun '22

14 Jun '22

From: Charles Keepax <ckeepax(a)opensource.cirrus.com> [ Upstream commit fcb3b5a58926d16d9a338841b74af06d4c29be15 ] The minimum value for the PGA Volume is given as 0x1A, however the values from there to 0x19 are all the same volume and this is not represented in the TLV structure. The number of volumes given is correct so this leads to all the volumes being shifted. Move the minimum value up to 0x19 to fix this. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220602162119.3393857-7-ckeepax@opensource.cirru… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/cs42l51.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c index c61b17dc2af8..fc6a2bc311b4 100644 --- a/sound/soc/codecs/cs42l51.c +++ b/sound/soc/codecs/cs42l51.c @@ -146,7 +146,7 @@ static const struct snd_kcontrol_new cs42l51_snd_controls[] = { 0, 0xA0, 96, adc_att_tlv), SOC_DOUBLE_R_SX_TLV("PGA Volume", CS42L51_ALC_PGA_CTL, CS42L51_ALC_PGB_CTL, - 0, 0x1A, 30, pga_tlv), + 0, 0x19, 30, pga_tlv), SOC_SINGLE("Playback Deemphasis Switch", CS42L51_DAC_CTL, 3, 1, 0), SOC_SINGLE("Auto-Mute Switch", CS42L51_DAC_CTL, 2, 1, 0), SOC_SINGLE("Soft Ramp Switch", CS42L51_DAC_CTL, 1, 1, 0), -- 2.35.1

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[PATCH AUTOSEL 5.17 11/43] ASoC: cs42l56: Correct typo in minimum level for SX volume controls
by Sasha Levin 14 Jun '22

14 Jun '22

From: Charles Keepax <ckeepax(a)opensource.cirrus.com> [ Upstream commit a8928ada9b96944cadd8b65d191e33199fd38782 ] A couple of the SX volume controls specify 0x84 as the lowest volume value, however the correct value from the datasheet is 0x44. The datasheet don't include spaces in the value it displays as binary so this was almost certainly just a typo reading 1000100. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220602162119.3393857-6-ckeepax@opensource.cirru… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/cs42l56.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs42l56.c b/sound/soc/codecs/cs42l56.c index 3cf8a0b4478c..b39c25409c23 100644 --- a/sound/soc/codecs/cs42l56.c +++ b/sound/soc/codecs/cs42l56.c @@ -391,9 +391,9 @@ static const struct snd_kcontrol_new cs42l56_snd_controls[] = { SOC_DOUBLE("ADC Boost Switch", CS42L56_GAIN_BIAS_CTL, 3, 2, 1, 1), SOC_DOUBLE_R_SX_TLV("Headphone Volume", CS42L56_HPA_VOLUME, - CS42L56_HPB_VOLUME, 0, 0x84, 0x48, hl_tlv), + CS42L56_HPB_VOLUME, 0, 0x44, 0x48, hl_tlv), SOC_DOUBLE_R_SX_TLV("LineOut Volume", CS42L56_LOA_VOLUME, - CS42L56_LOB_VOLUME, 0, 0x84, 0x48, hl_tlv), + CS42L56_LOB_VOLUME, 0, 0x44, 0x48, hl_tlv), SOC_SINGLE_TLV("Bass Shelving Volume", CS42L56_TONE_CTL, 0, 0x00, 1, tone_tlv), -- 2.35.1

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[PATCH AUTOSEL 5.17 10/43] ASoC: cs42l52: Correct TLV for Bypass Volume
by Sasha Levin 14 Jun '22

14 Jun '22

From: Charles Keepax <ckeepax(a)opensource.cirrus.com> [ Upstream commit 91e90c712fade0b69cdff7cc6512f6099bd18ae5 ] The Bypass Volume is accidentally using a -6dB minimum TLV rather than the correct -60dB minimum. Add a new TLV to correct this. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220602162119.3393857-5-ckeepax@opensource.cirru… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/cs42l52.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index 785caba3f653..c19ad3c24702 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -137,6 +137,8 @@ static DECLARE_TLV_DB_SCALE(mic_tlv, 1600, 100, 0); static DECLARE_TLV_DB_SCALE(pga_tlv, -600, 50, 0); +static DECLARE_TLV_DB_SCALE(pass_tlv, -6000, 50, 0); + static DECLARE_TLV_DB_SCALE(mix_tlv, -5150, 50, 0); static DECLARE_TLV_DB_SCALE(beep_tlv, -56, 200, 0); @@ -351,7 +353,7 @@ static const struct snd_kcontrol_new cs42l52_snd_controls[] = { CS42L52_SPKB_VOL, 0, 0x40, 0xC0, hl_tlv), SOC_DOUBLE_R_SX_TLV("Bypass Volume", CS42L52_PASSTHRUA_VOL, - CS42L52_PASSTHRUB_VOL, 0, 0x88, 0x90, pga_tlv), + CS42L52_PASSTHRUB_VOL, 0, 0x88, 0x90, pass_tlv), SOC_DOUBLE("Bypass Mute", CS42L52_MISC_CTL, 4, 5, 1, 0), -- 2.35.1

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[PATCH AUTOSEL 5.17 09/43] ASoC: cs53l30: Correct number of volume levels on SX controls
by Sasha Levin 14 Jun '22

14 Jun '22

From: Charles Keepax <ckeepax(a)opensource.cirrus.com> [ Upstream commit 7fbd6dd68127927e844912a16741016d432a0737 ] This driver specified the maximum value rather than the number of volume levels on the SX controls, this is incorrect, so correct them. Reported-by: David Rhodes <david.rhodes(a)cirrus.com> Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220602162119.3393857-4-ckeepax@opensource.cirru… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/cs53l30.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/cs53l30.c b/sound/soc/codecs/cs53l30.c index f2087bd38dbc..c2912ad3851b 100644 --- a/sound/soc/codecs/cs53l30.c +++ b/sound/soc/codecs/cs53l30.c @@ -348,22 +348,22 @@ static const struct snd_kcontrol_new cs53l30_snd_controls[] = { SOC_ENUM("ADC2 NG Delay", adc2_ng_delay_enum), SOC_SINGLE_SX_TLV("ADC1A PGA Volume", - CS53L30_ADC1A_AFE_CTL, 0, 0x34, 0x18, pga_tlv), + CS53L30_ADC1A_AFE_CTL, 0, 0x34, 0x24, pga_tlv), SOC_SINGLE_SX_TLV("ADC1B PGA Volume", - CS53L30_ADC1B_AFE_CTL, 0, 0x34, 0x18, pga_tlv), + CS53L30_ADC1B_AFE_CTL, 0, 0x34, 0x24, pga_tlv), SOC_SINGLE_SX_TLV("ADC2A PGA Volume", - CS53L30_ADC2A_AFE_CTL, 0, 0x34, 0x18, pga_tlv), + CS53L30_ADC2A_AFE_CTL, 0, 0x34, 0x24, pga_tlv), SOC_SINGLE_SX_TLV("ADC2B PGA Volume", - CS53L30_ADC2B_AFE_CTL, 0, 0x34, 0x18, pga_tlv), + CS53L30_ADC2B_AFE_CTL, 0, 0x34, 0x24, pga_tlv), SOC_SINGLE_SX_TLV("ADC1A Digital Volume", - CS53L30_ADC1A_DIG_VOL, 0, 0xA0, 0x0C, dig_tlv), + CS53L30_ADC1A_DIG_VOL, 0, 0xA0, 0x6C, dig_tlv), SOC_SINGLE_SX_TLV("ADC1B Digital Volume", - CS53L30_ADC1B_DIG_VOL, 0, 0xA0, 0x0C, dig_tlv), + CS53L30_ADC1B_DIG_VOL, 0, 0xA0, 0x6C, dig_tlv), SOC_SINGLE_SX_TLV("ADC2A Digital Volume", - CS53L30_ADC2A_DIG_VOL, 0, 0xA0, 0x0C, dig_tlv), + CS53L30_ADC2A_DIG_VOL, 0, 0xA0, 0x6C, dig_tlv), SOC_SINGLE_SX_TLV("ADC2B Digital Volume", - CS53L30_ADC2B_DIG_VOL, 0, 0xA0, 0x0C, dig_tlv), + CS53L30_ADC2B_DIG_VOL, 0, 0xA0, 0x6C, dig_tlv), }; static const struct snd_soc_dapm_widget cs53l30_dapm_widgets[] = { -- 2.35.1

1 0

[PATCH AUTOSEL 5.17 08/43] ASoC: cs35l36: Update digital volume TLV
by Sasha Levin 14 Jun '22

14 Jun '22

From: Charles Keepax <ckeepax(a)opensource.cirrus.com> [ Upstream commit 5005a2345825eb8346546d99bfe669f73111b5c5 ] The digital volume TLV specifies the step as 0.25dB but the actual step of the control is 0.125dB. Update the TLV to correct this. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220602162119.3393857-3-ckeepax@opensource.cirru… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/cs35l36.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs35l36.c b/sound/soc/codecs/cs35l36.c index d83c1b318c1c..0accdb45ed72 100644 --- a/sound/soc/codecs/cs35l36.c +++ b/sound/soc/codecs/cs35l36.c @@ -444,7 +444,8 @@ static bool cs35l36_volatile_reg(struct device *dev, unsigned int reg) } } -static DECLARE_TLV_DB_SCALE(dig_vol_tlv, -10200, 25, 0); +static const DECLARE_TLV_DB_RANGE(dig_vol_tlv, 0, 912, + TLV_DB_MINMAX_ITEM(-10200, 1200)); static DECLARE_TLV_DB_SCALE(amp_gain_tlv, 0, 1, 1); static const char * const cs35l36_pcm_sftramp_text[] = { -- 2.35.1

1 0

[PATCH AUTOSEL 5.17 07/43] ASoC: cs42l52: Fix TLV scales for mixer controls
by Sasha Levin 14 Jun '22

14 Jun '22

From: Charles Keepax <ckeepax(a)opensource.cirrus.com> [ Upstream commit 8bf5aabf524eec61013e506f764a0b2652dc5665 ] The datasheet specifies the range of the mixer volumes as between -51.5dB and 12dB with a 0.5dB step. Update the TLVs for this. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220602162119.3393857-2-ckeepax@opensource.cirru… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/cs42l52.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index 80161151b3f2..785caba3f653 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -137,7 +137,7 @@ static DECLARE_TLV_DB_SCALE(mic_tlv, 1600, 100, 0); static DECLARE_TLV_DB_SCALE(pga_tlv, -600, 50, 0); -static DECLARE_TLV_DB_SCALE(mix_tlv, -50, 50, 0); +static DECLARE_TLV_DB_SCALE(mix_tlv, -5150, 50, 0); static DECLARE_TLV_DB_SCALE(beep_tlv, -56, 200, 0); @@ -364,7 +364,7 @@ static const struct snd_kcontrol_new cs42l52_snd_controls[] = { CS42L52_ADCB_VOL, 0, 0xA0, 0x78, ipd_tlv), SOC_DOUBLE_R_SX_TLV("ADC Mixer Volume", CS42L52_ADCA_MIXER_VOL, CS42L52_ADCB_MIXER_VOL, - 0, 0x19, 0x7F, ipd_tlv), + 0, 0x19, 0x7F, mix_tlv), SOC_DOUBLE("ADC Switch", CS42L52_ADC_MISC_CTL, 0, 1, 1, 0), -- 2.35.1

1 0

[PATCH AUTOSEL 5.17 02/43] ASoC: nau8822: Add operation for internal PLL off and on
by Sasha Levin 14 Jun '22

14 Jun '22

From: Hui Wang <hui.wang(a)canonical.com> [ Upstream commit aeca8a3295022bcec46697f16e098140423d8463 ] We tried to enable the audio on an imx6sx EVB with the codec nau8822, after setting the internal PLL fractional parameters, the audio still couldn't work and the there was no sdma irq at all. After checking with the section "8.1.1 Phase Locked Loop (PLL) Design Example" of "NAU88C22 Datasheet Rev 0.6", we found we need to turn off the PLL before programming fractional parameters and turn on the PLL after programming. After this change, the audio driver could record and play sound and the sdma's irq is triggered when playing or recording. Cc: David Lin <ctlin0(a)nuvoton.com> Cc: John Hsu <kchsu0(a)nuvoton.com> Cc: Seven Li <wtli(a)nuvoton.com> Signed-off-by: Hui Wang <hui.wang(a)canonical.com> Link: https://lore.kernel.org/r/20220530040151.95221-2-hui.wang@canonical.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/nau8822.c | 4 ++++ sound/soc/codecs/nau8822.h | 3 +++ 2 files changed, 7 insertions(+) diff --git a/sound/soc/codecs/nau8822.c b/sound/soc/codecs/nau8822.c index 58123390c7a3..b436e532993d 100644 --- a/sound/soc/codecs/nau8822.c +++ b/sound/soc/codecs/nau8822.c @@ -740,6 +740,8 @@ static int nau8822_set_pll(struct snd_soc_dai *dai, int pll_id, int source, pll_param->pll_int, pll_param->pll_frac, pll_param->mclk_scaler, pll_param->pre_factor); + snd_soc_component_update_bits(component, + NAU8822_REG_POWER_MANAGEMENT_1, NAU8822_PLL_EN_MASK, NAU8822_PLL_OFF); snd_soc_component_update_bits(component, NAU8822_REG_PLL_N, NAU8822_PLLMCLK_DIV2 | NAU8822_PLLN_MASK, (pll_param->pre_factor ? NAU8822_PLLMCLK_DIV2 : 0) | @@ -757,6 +759,8 @@ static int nau8822_set_pll(struct snd_soc_dai *dai, int pll_id, int source, pll_param->mclk_scaler << NAU8822_MCLKSEL_SFT); snd_soc_component_update_bits(component, NAU8822_REG_CLOCKING, NAU8822_CLKM_MASK, NAU8822_CLKM_PLL); + snd_soc_component_update_bits(component, + NAU8822_REG_POWER_MANAGEMENT_1, NAU8822_PLL_EN_MASK, NAU8822_PLL_ON); return 0; } diff --git a/sound/soc/codecs/nau8822.h b/sound/soc/codecs/nau8822.h index 489191ff187e..b45d42c15de6 100644 --- a/sound/soc/codecs/nau8822.h +++ b/sound/soc/codecs/nau8822.h @@ -90,6 +90,9 @@ #define NAU8822_REFIMP_3K 0x3 #define NAU8822_IOBUF_EN (0x1 << 2) #define NAU8822_ABIAS_EN (0x1 << 3) +#define NAU8822_PLL_EN_MASK (0x1 << 5) +#define NAU8822_PLL_ON (0x1 << 5) +#define NAU8822_PLL_OFF (0x0 << 5) /* NAU8822_REG_AUDIO_INTERFACE (0x4) */ #define NAU8822_AIFMT_MASK (0x3 << 3) -- 2.35.1

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[PATCH AUTOSEL 5.18 30/47] ALSA: hda/realtek - Add HW8326 support
by Sasha Levin 14 Jun '22

14 Jun '22

From: huangwenhui <huangwenhuia(a)uniontech.com> [ Upstream commit 527f4643e03c298c1e3321cfa27866b1374a55e1 ] Added the support of new Huawei codec HW8326. The HW8326 is developed by Huawei with Realtek's IP Core, and it's compatible with ALC256. Signed-off-by: huangwenhui <huangwenhuia(a)uniontech.com> Link: https://lore.kernel.org/r/20220608082357.26898-1-huangwenhuia@uniontech.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/hda/hdac_device.c | 1 + sound/pci/hda/patch_realtek.c | 14 ++++++++++++++ 2 files changed, 15 insertions(+) diff --git a/sound/hda/hdac_device.c b/sound/hda/hdac_device.c index 3e9e9ac804f6..b7e5032b61c9 100644 --- a/sound/hda/hdac_device.c +++ b/sound/hda/hdac_device.c @@ -660,6 +660,7 @@ static const struct hda_vendor_id hda_vendor_ids[] = { { 0x14f1, "Conexant" }, { 0x17e8, "Chrontel" }, { 0x1854, "LG" }, + { 0x19e5, "Huawei" }, { 0x1aec, "Wolfson Microelectronics" }, { 0x1af4, "QEMU" }, { 0x434d, "C-Media" }, diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 323c74a04268..87240f71892a 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -443,6 +443,7 @@ static void alc_fill_eapd_coef(struct hda_codec *codec) case 0x10ec0245: case 0x10ec0255: case 0x10ec0256: + case 0x19e58326: case 0x10ec0257: case 0x10ec0282: case 0x10ec0283: @@ -580,6 +581,7 @@ static void alc_shutup_pins(struct hda_codec *codec) switch (codec->core.vendor_id) { case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: case 0x10ec0283: case 0x10ec0286: case 0x10ec0288: @@ -3247,6 +3249,7 @@ static void alc_disable_headset_jack_key(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_write_coef_idx(codec, 0x48, 0x0); alc_update_coef_idx(codec, 0x49, 0x0045, 0x0); break; @@ -3275,6 +3278,7 @@ static void alc_enable_headset_jack_key(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_write_coef_idx(codec, 0x48, 0xd011); alc_update_coef_idx(codec, 0x49, 0x007f, 0x0045); break; @@ -4910,6 +4914,7 @@ static void alc_headset_mode_unplugged(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_process_coef_fw(codec, coef0256); break; case 0x10ec0234: @@ -5025,6 +5030,7 @@ static void alc_headset_mode_mic_in(struct hda_codec *codec, hda_nid_t hp_pin, case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_write_coef_idx(codec, 0x45, 0xc489); snd_hda_set_pin_ctl_cache(codec, hp_pin, 0); alc_process_coef_fw(codec, coef0256); @@ -5175,6 +5181,7 @@ static void alc_headset_mode_default(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_write_coef_idx(codec, 0x1b, 0x0e4b); alc_write_coef_idx(codec, 0x45, 0xc089); msleep(50); @@ -5274,6 +5281,7 @@ static void alc_headset_mode_ctia(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_process_coef_fw(codec, coef0256); break; case 0x10ec0234: @@ -5388,6 +5396,7 @@ static void alc_headset_mode_omtp(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_process_coef_fw(codec, coef0256); break; case 0x10ec0234: @@ -5489,6 +5498,7 @@ static void alc_determine_headset_type(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_write_coef_idx(codec, 0x1b, 0x0e4b); alc_write_coef_idx(codec, 0x06, 0x6104); alc_write_coefex_idx(codec, 0x57, 0x3, 0x09a3); @@ -5783,6 +5793,7 @@ static void alc255_set_default_jack_type(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_process_coef_fw(codec, alc256fw); break; } @@ -6385,6 +6396,7 @@ static void alc_combo_jack_hp_jd_restart(struct hda_codec *codec) case 0x10ec0236: case 0x10ec0255: case 0x10ec0256: + case 0x19e58326: alc_update_coef_idx(codec, 0x1b, 0x8000, 1 << 15); /* Reset HP JD */ alc_update_coef_idx(codec, 0x1b, 0x8000, 0 << 15); break; @@ -10147,6 +10159,7 @@ static int patch_alc269(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: spec->codec_variant = ALC269_TYPE_ALC256; spec->shutup = alc256_shutup; spec->init_hook = alc256_init; @@ -11597,6 +11610,7 @@ static const struct hda_device_id snd_hda_id_realtek[] = { HDA_CODEC_ENTRY(0x10ec0b00, "ALCS1200A", patch_alc882), HDA_CODEC_ENTRY(0x10ec1168, "ALC1220", patch_alc882), HDA_CODEC_ENTRY(0x10ec1220, "ALC1220", patch_alc882), + HDA_CODEC_ENTRY(0x19e58326, "HW8326", patch_alc269), {} /* terminator */ }; MODULE_DEVICE_TABLE(hdaudio, snd_hda_id_realtek); -- 2.35.1

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[PATCH AUTOSEL 5.18 21/47] ALSA: hda: MTL: add HD Audio PCI ID and HDMI codec vendor ID
by Sasha Levin 14 Jun '22

14 Jun '22

From: Yong Zhi <yong.zhi(a)intel.com> [ Upstream commit 2e45f2185283a2d927ef2cdbdc246cd65740c8df ] Add HD Audio PCI ID for Intel Meteorlake platform. [ corrected the hex number to lower letters by tiwai ] Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Signed-off-by: Yong Zhi <yong.zhi(a)intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20220606204232.144296-1-pierre-louis.bossart@linu… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/hda_intel.c | 3 +++ sound/pci/hda/patch_hdmi.c | 1 + 2 files changed, 4 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 0a83eb6b88b1..a77165bd92a9 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2525,6 +2525,9 @@ static const struct pci_device_id azx_ids[] = { .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, { PCI_DEVICE(0x8086, 0x51cf), .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, + /* Meteorlake-P */ + { PCI_DEVICE(0x8086, 0x7e28), + .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, /* Broxton-P(Apollolake) */ { PCI_DEVICE(0x8086, 0x5a98), .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_BROXTON }, diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 31fe41795571..6c209cd26c0c 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -4554,6 +4554,7 @@ HDA_CODEC_ENTRY(0x8086281a, "Jasperlake HDMI", patch_i915_icl_hdmi), HDA_CODEC_ENTRY(0x8086281b, "Elkhartlake HDMI", patch_i915_icl_hdmi), HDA_CODEC_ENTRY(0x8086281c, "Alderlake-P HDMI", patch_i915_adlp_hdmi), HDA_CODEC_ENTRY(0x8086281f, "Raptorlake-P HDMI", patch_i915_adlp_hdmi), +HDA_CODEC_ENTRY(0x8086281d, "Meteorlake HDMI", patch_i915_adlp_hdmi), HDA_CODEC_ENTRY(0x80862880, "CedarTrail HDMI", patch_generic_hdmi), HDA_CODEC_ENTRY(0x80862882, "Valleyview2 HDMI", patch_i915_byt_hdmi), HDA_CODEC_ENTRY(0x80862883, "Braswell HDMI", patch_i915_byt_hdmi), -- 2.35.1

1 0

[PATCH AUTOSEL 5.18 20/47] ASoC: wm_adsp: Fix event generation for wm_adsp_fw_put()
by Sasha Levin 14 Jun '22

14 Jun '22

From: Mark Brown <broonie(a)kernel.org> [ Upstream commit 2abdf9f80019e8244d3806ed0e1c9f725e50b452 ] Currently wm_adsp_fw_put() returns 0 rather than 1 when updating the value of the control, meaning that no event is generated to userspace. Fix this by setting the default return value to 1, the code already exits early with a return value of 0 if the value is unchanged. Signed-off-by: Mark Brown <broonie(a)kernel.org> Reviewed-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220603115003.3865834-1-broonie@kernel.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/wm_adsp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index e32c8ded181d..9cfd4f18493f 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -333,7 +333,7 @@ int wm_adsp_fw_put(struct snd_kcontrol *kcontrol, struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; struct wm_adsp *dsp = snd_soc_component_get_drvdata(component); - int ret = 0; + int ret = 1; if (ucontrol->value.enumerated.item[0] == dsp[e->shift_l].fw) return 0; -- 2.35.1

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[PATCH AUTOSEL 5.18 19/47] ASoC: es8328: Fix event generation for deemphasis control
by Sasha Levin 14 Jun '22

14 Jun '22

From: Mark Brown <broonie(a)kernel.org> [ Upstream commit 8259610c2ec01c5cbfb61882ae176aabacac9c19 ] Currently the put() method for the deemphasis control returns 0 when a new value is written to the control even if the value changed, meaning events are not generated. Fix this, skip the work of updating the value when it is unchanged and then return 1 after having done so. Signed-off-by: Mark Brown <broonie(a)kernel.org> Link: https://lore.kernel.org/r/20220603123937.4013603-1-broonie@kernel.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/es8328.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/es8328.c b/sound/soc/codecs/es8328.c index 3f00ead97006..dd53dfd87b04 100644 --- a/sound/soc/codecs/es8328.c +++ b/sound/soc/codecs/es8328.c @@ -161,13 +161,16 @@ static int es8328_put_deemph(struct snd_kcontrol *kcontrol, if (deemph > 1) return -EINVAL; + if (es8328->deemph == deemph) + return 0; + ret = es8328_set_deemph(component); if (ret < 0) return ret; es8328->deemph = deemph; - return 0; + return 1; } -- 2.35.1

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[PATCH AUTOSEL 5.18 18/47] ASoC: wm8962: Fix suspend while playing music
by Sasha Levin 14 Jun '22

14 Jun '22

From: Adam Ford <aford173(a)gmail.com> [ Upstream commit d1f5272c0f7d2e53c6f2480f46725442776f5f78 ] If the audio CODEC is playing sound when the system is suspended, it can be left in a state which throws the following error: wm8962 3-001a: ASoC: error at soc_component_read_no_lock on wm8962.3-001a: -16 Once this error has occurred, the audio will not work again until rebooted. Fix this by configuring SET_SYSTEM_SLEEP_PM_OPS. Signed-off-by: Adam Ford <aford173(a)gmail.com> Acked-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220526182129.538472-1-aford173@gmail.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/wm8962.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 2c41d31956aa..f622a6bbd2fb 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3871,6 +3871,7 @@ static int wm8962_runtime_suspend(struct device *dev) #endif static const struct dev_pm_ops wm8962_pm = { + SET_SYSTEM_SLEEP_PM_OPS(pm_runtime_force_suspend, pm_runtime_force_resume) SET_RUNTIME_PM_OPS(wm8962_runtime_suspend, wm8962_runtime_resume, NULL) }; -- 2.35.1

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[PATCH AUTOSEL 5.18 14/47] ASoC: cs42l51: Correct minimum value for SX volume control
by Sasha Levin 14 Jun '22

14 Jun '22

From: Charles Keepax <ckeepax(a)opensource.cirrus.com> [ Upstream commit fcb3b5a58926d16d9a338841b74af06d4c29be15 ] The minimum value for the PGA Volume is given as 0x1A, however the values from there to 0x19 are all the same volume and this is not represented in the TLV structure. The number of volumes given is correct so this leads to all the volumes being shifted. Move the minimum value up to 0x19 to fix this. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220602162119.3393857-7-ckeepax@opensource.cirru… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/cs42l51.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c index e9c3cb4e2bfc..b9c262a15edf 100644 --- a/sound/soc/codecs/cs42l51.c +++ b/sound/soc/codecs/cs42l51.c @@ -146,7 +146,7 @@ static const struct snd_kcontrol_new cs42l51_snd_controls[] = { 0, 0xA0, 96, adc_att_tlv), SOC_DOUBLE_R_SX_TLV("PGA Volume", CS42L51_ALC_PGA_CTL, CS42L51_ALC_PGB_CTL, - 0, 0x1A, 30, pga_tlv), + 0, 0x19, 30, pga_tlv), SOC_SINGLE("Playback Deemphasis Switch", CS42L51_DAC_CTL, 3, 1, 0), SOC_SINGLE("Auto-Mute Switch", CS42L51_DAC_CTL, 2, 1, 0), SOC_SINGLE("Soft Ramp Switch", CS42L51_DAC_CTL, 1, 1, 0), -- 2.35.1

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[PATCH AUTOSEL 5.18 13/47] ASoC: cs42l56: Correct typo in minimum level for SX volume controls
by Sasha Levin 14 Jun '22

14 Jun '22

From: Charles Keepax <ckeepax(a)opensource.cirrus.com> [ Upstream commit a8928ada9b96944cadd8b65d191e33199fd38782 ] A couple of the SX volume controls specify 0x84 as the lowest volume value, however the correct value from the datasheet is 0x44. The datasheet don't include spaces in the value it displays as binary so this was almost certainly just a typo reading 1000100. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220602162119.3393857-6-ckeepax@opensource.cirru… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/cs42l56.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs42l56.c b/sound/soc/codecs/cs42l56.c index 3cf8a0b4478c..b39c25409c23 100644 --- a/sound/soc/codecs/cs42l56.c +++ b/sound/soc/codecs/cs42l56.c @@ -391,9 +391,9 @@ static const struct snd_kcontrol_new cs42l56_snd_controls[] = { SOC_DOUBLE("ADC Boost Switch", CS42L56_GAIN_BIAS_CTL, 3, 2, 1, 1), SOC_DOUBLE_R_SX_TLV("Headphone Volume", CS42L56_HPA_VOLUME, - CS42L56_HPB_VOLUME, 0, 0x84, 0x48, hl_tlv), + CS42L56_HPB_VOLUME, 0, 0x44, 0x48, hl_tlv), SOC_DOUBLE_R_SX_TLV("LineOut Volume", CS42L56_LOA_VOLUME, - CS42L56_LOB_VOLUME, 0, 0x84, 0x48, hl_tlv), + CS42L56_LOB_VOLUME, 0, 0x44, 0x48, hl_tlv), SOC_SINGLE_TLV("Bass Shelving Volume", CS42L56_TONE_CTL, 0, 0x00, 1, tone_tlv), -- 2.35.1

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[PATCH AUTOSEL 5.18 12/47] ASoC: cs42l52: Correct TLV for Bypass Volume
by Sasha Levin 14 Jun '22

14 Jun '22

From: Charles Keepax <ckeepax(a)opensource.cirrus.com> [ Upstream commit 91e90c712fade0b69cdff7cc6512f6099bd18ae5 ] The Bypass Volume is accidentally using a -6dB minimum TLV rather than the correct -60dB minimum. Add a new TLV to correct this. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220602162119.3393857-5-ckeepax@opensource.cirru… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/cs42l52.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index 785caba3f653..c19ad3c24702 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -137,6 +137,8 @@ static DECLARE_TLV_DB_SCALE(mic_tlv, 1600, 100, 0); static DECLARE_TLV_DB_SCALE(pga_tlv, -600, 50, 0); +static DECLARE_TLV_DB_SCALE(pass_tlv, -6000, 50, 0); + static DECLARE_TLV_DB_SCALE(mix_tlv, -5150, 50, 0); static DECLARE_TLV_DB_SCALE(beep_tlv, -56, 200, 0); @@ -351,7 +353,7 @@ static const struct snd_kcontrol_new cs42l52_snd_controls[] = { CS42L52_SPKB_VOL, 0, 0x40, 0xC0, hl_tlv), SOC_DOUBLE_R_SX_TLV("Bypass Volume", CS42L52_PASSTHRUA_VOL, - CS42L52_PASSTHRUB_VOL, 0, 0x88, 0x90, pga_tlv), + CS42L52_PASSTHRUB_VOL, 0, 0x88, 0x90, pass_tlv), SOC_DOUBLE("Bypass Mute", CS42L52_MISC_CTL, 4, 5, 1, 0), -- 2.35.1

1 0

[PATCH AUTOSEL 5.18 11/47] ASoC: cs53l30: Correct number of volume levels on SX controls
by Sasha Levin 14 Jun '22

14 Jun '22

From: Charles Keepax <ckeepax(a)opensource.cirrus.com> [ Upstream commit 7fbd6dd68127927e844912a16741016d432a0737 ] This driver specified the maximum value rather than the number of volume levels on the SX controls, this is incorrect, so correct them. Reported-by: David Rhodes <david.rhodes(a)cirrus.com> Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220602162119.3393857-4-ckeepax@opensource.cirru… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/cs53l30.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/cs53l30.c b/sound/soc/codecs/cs53l30.c index f2087bd38dbc..c2912ad3851b 100644 --- a/sound/soc/codecs/cs53l30.c +++ b/sound/soc/codecs/cs53l30.c @@ -348,22 +348,22 @@ static const struct snd_kcontrol_new cs53l30_snd_controls[] = { SOC_ENUM("ADC2 NG Delay", adc2_ng_delay_enum), SOC_SINGLE_SX_TLV("ADC1A PGA Volume", - CS53L30_ADC1A_AFE_CTL, 0, 0x34, 0x18, pga_tlv), + CS53L30_ADC1A_AFE_CTL, 0, 0x34, 0x24, pga_tlv), SOC_SINGLE_SX_TLV("ADC1B PGA Volume", - CS53L30_ADC1B_AFE_CTL, 0, 0x34, 0x18, pga_tlv), + CS53L30_ADC1B_AFE_CTL, 0, 0x34, 0x24, pga_tlv), SOC_SINGLE_SX_TLV("ADC2A PGA Volume", - CS53L30_ADC2A_AFE_CTL, 0, 0x34, 0x18, pga_tlv), + CS53L30_ADC2A_AFE_CTL, 0, 0x34, 0x24, pga_tlv), SOC_SINGLE_SX_TLV("ADC2B PGA Volume", - CS53L30_ADC2B_AFE_CTL, 0, 0x34, 0x18, pga_tlv), + CS53L30_ADC2B_AFE_CTL, 0, 0x34, 0x24, pga_tlv), SOC_SINGLE_SX_TLV("ADC1A Digital Volume", - CS53L30_ADC1A_DIG_VOL, 0, 0xA0, 0x0C, dig_tlv), + CS53L30_ADC1A_DIG_VOL, 0, 0xA0, 0x6C, dig_tlv), SOC_SINGLE_SX_TLV("ADC1B Digital Volume", - CS53L30_ADC1B_DIG_VOL, 0, 0xA0, 0x0C, dig_tlv), + CS53L30_ADC1B_DIG_VOL, 0, 0xA0, 0x6C, dig_tlv), SOC_SINGLE_SX_TLV("ADC2A Digital Volume", - CS53L30_ADC2A_DIG_VOL, 0, 0xA0, 0x0C, dig_tlv), + CS53L30_ADC2A_DIG_VOL, 0, 0xA0, 0x6C, dig_tlv), SOC_SINGLE_SX_TLV("ADC2B Digital Volume", - CS53L30_ADC2B_DIG_VOL, 0, 0xA0, 0x0C, dig_tlv), + CS53L30_ADC2B_DIG_VOL, 0, 0xA0, 0x6C, dig_tlv), }; static const struct snd_soc_dapm_widget cs53l30_dapm_widgets[] = { -- 2.35.1

1 0

[PATCH AUTOSEL 5.18 10/47] ASoC: cs35l36: Update digital volume TLV
by Sasha Levin 14 Jun '22

14 Jun '22

From: Charles Keepax <ckeepax(a)opensource.cirrus.com> [ Upstream commit 5005a2345825eb8346546d99bfe669f73111b5c5 ] The digital volume TLV specifies the step as 0.25dB but the actual step of the control is 0.125dB. Update the TLV to correct this. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220602162119.3393857-3-ckeepax@opensource.cirru… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/cs35l36.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs35l36.c b/sound/soc/codecs/cs35l36.c index d83c1b318c1c..0accdb45ed72 100644 --- a/sound/soc/codecs/cs35l36.c +++ b/sound/soc/codecs/cs35l36.c @@ -444,7 +444,8 @@ static bool cs35l36_volatile_reg(struct device *dev, unsigned int reg) } } -static DECLARE_TLV_DB_SCALE(dig_vol_tlv, -10200, 25, 0); +static const DECLARE_TLV_DB_RANGE(dig_vol_tlv, 0, 912, + TLV_DB_MINMAX_ITEM(-10200, 1200)); static DECLARE_TLV_DB_SCALE(amp_gain_tlv, 0, 1, 1); static const char * const cs35l36_pcm_sftramp_text[] = { -- 2.35.1

1 0

[PATCH AUTOSEL 5.18 09/47] ASoC: cs42l52: Fix TLV scales for mixer controls
by Sasha Levin 14 Jun '22

14 Jun '22

From: Charles Keepax <ckeepax(a)opensource.cirrus.com> [ Upstream commit 8bf5aabf524eec61013e506f764a0b2652dc5665 ] The datasheet specifies the range of the mixer volumes as between -51.5dB and 12dB with a 0.5dB step. Update the TLVs for this. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220602162119.3393857-2-ckeepax@opensource.cirru… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/cs42l52.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index 80161151b3f2..785caba3f653 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -137,7 +137,7 @@ static DECLARE_TLV_DB_SCALE(mic_tlv, 1600, 100, 0); static DECLARE_TLV_DB_SCALE(pga_tlv, -600, 50, 0); -static DECLARE_TLV_DB_SCALE(mix_tlv, -50, 50, 0); +static DECLARE_TLV_DB_SCALE(mix_tlv, -5150, 50, 0); static DECLARE_TLV_DB_SCALE(beep_tlv, -56, 200, 0); @@ -364,7 +364,7 @@ static const struct snd_kcontrol_new cs42l52_snd_controls[] = { CS42L52_ADCB_VOL, 0, 0xA0, 0x78, ipd_tlv), SOC_DOUBLE_R_SX_TLV("ADC Mixer Volume", CS42L52_ADCA_MIXER_VOL, CS42L52_ADCB_MIXER_VOL, - 0, 0x19, 0x7F, ipd_tlv), + 0, 0x19, 0x7F, mix_tlv), SOC_DOUBLE("ADC Switch", CS42L52_ADC_MISC_CTL, 0, 1, 1, 0), -- 2.35.1

1 0

[PATCH AUTOSEL 5.18 08/47] ASoC: Intel: cirrus-common: fix incorrect channel mapping
by Sasha Levin 14 Jun '22

14 Jun '22

From: xliu <xiang.liu(a)cirrus.com> [ Upstream commit d69a155555c9d57463b788c400f6b452d976bacd ] The default mapping of ASPRX1 (DAC source) is slot 0. Change the slot mapping of right amplifiers (WR and TR) to slot 1 to receive right channel data. Also update the ACPI instance ID mapping according to HW configuration. Signed-off-by: xliu <xiang.liu(a)cirrus.com> Signed-off-by: Brent Lu <brent.lu(a)intel.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20220602051922.1232457-1-brent.lu@intel.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/sof_cirrus_common.c | 40 +++++++++++++++++++--- 1 file changed, 36 insertions(+), 4 deletions(-) diff --git a/sound/soc/intel/boards/sof_cirrus_common.c b/sound/soc/intel/boards/sof_cirrus_common.c index e71d74ec1b0b..f4192df962d6 100644 --- a/sound/soc/intel/boards/sof_cirrus_common.c +++ b/sound/soc/intel/boards/sof_cirrus_common.c @@ -54,22 +54,29 @@ static struct snd_soc_dai_link_component cs35l41_components[] = { }, }; +/* + * Mapping between ACPI instance id and speaker position. + * + * Four speakers: + * 0: Tweeter left, 1: Woofer left + * 2: Tweeter right, 3: Woofer right + */ static struct snd_soc_codec_conf cs35l41_codec_conf[] = { { .dlc = COMP_CODEC_CONF(CS35L41_DEV0_NAME), - .name_prefix = "WL", + .name_prefix = "TL", }, { .dlc = COMP_CODEC_CONF(CS35L41_DEV1_NAME), - .name_prefix = "WR", + .name_prefix = "WL", }, { .dlc = COMP_CODEC_CONF(CS35L41_DEV2_NAME), - .name_prefix = "TL", + .name_prefix = "TR", }, { .dlc = COMP_CODEC_CONF(CS35L41_DEV3_NAME), - .name_prefix = "TR", + .name_prefix = "WR", }, }; @@ -101,6 +108,21 @@ static int cs35l41_init(struct snd_soc_pcm_runtime *rtd) return ret; } +/* + * Channel map: + * + * TL/WL: ASPRX1 on slot 0, ASPRX2 on slot 1 (default) + * TR/WR: ASPRX1 on slot 1, ASPRX2 on slot 0 + */ +static const struct { + unsigned int rx[2]; +} cs35l41_channel_map[] = { + {.rx = {0, 1}}, /* TL */ + {.rx = {0, 1}}, /* WL */ + {.rx = {1, 0}}, /* TR */ + {.rx = {1, 0}}, /* WR */ +}; + static int cs35l41_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { @@ -134,6 +156,16 @@ static int cs35l41_hw_params(struct snd_pcm_substream *substream, ret); return ret; } + + /* setup channel map */ + ret = snd_soc_dai_set_channel_map(codec_dai, 0, NULL, + ARRAY_SIZE(cs35l41_channel_map[i].rx), + (unsigned int *)cs35l41_channel_map[i].rx); + if (ret < 0) { + dev_err(codec_dai->dev, "fail to set channel map, ret %d\n", + ret); + return ret; + } } return 0; -- 2.35.1

1 0

[PATCH AUTOSEL 5.18 03/47] ASoC: qcom: lpass-platform: Update VMA access permissions in mmap callback
by Sasha Levin 14 Jun '22

14 Jun '22

From: Srinivasa Rao Mandadapu <quic_srivasam(a)quicinc.com> [ Upstream commit ef8d89b83bf453ea9cc3c4873a84b50ff334f797 ] Replace page protection permissions from noncashed to writecombine, in lpass codec DMA path mmp callabck, to support 64 bit chromeOS. Avoid SIGBUS error in userspace caused by noncached permissions in 64 bit chromeOS. Signed-off-by: Srinivasa Rao Mandadapu <quic_srivasam(a)quicinc.com> Link: https://lore.kernel.org/r/1653660608-27245-1-git-send-email-quic_srivasam@q… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/qcom/lpass-platform.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/qcom/lpass-platform.c b/sound/soc/qcom/lpass-platform.c index 74d62f377dfd..ae2a7837e5cc 100644 --- a/sound/soc/qcom/lpass-platform.c +++ b/sound/soc/qcom/lpass-platform.c @@ -898,7 +898,7 @@ static int lpass_platform_cdc_dma_mmap(struct snd_pcm_substream *substream, struct snd_pcm_runtime *runtime = substream->runtime; unsigned long size, offset; - vma->vm_page_prot = pgprot_noncached(vma->vm_page_prot); + vma->vm_page_prot = pgprot_writecombine(vma->vm_page_prot); size = vma->vm_end - vma->vm_start; offset = vma->vm_pgoff << PAGE_SHIFT; return io_remap_pfn_range(vma, vma->vm_start, -- 2.35.1

1 0

[PATCH AUTOSEL 5.18 02/47] ASoC: nau8822: Add operation for internal PLL off and on
by Sasha Levin 14 Jun '22

14 Jun '22

From: Hui Wang <hui.wang(a)canonical.com> [ Upstream commit aeca8a3295022bcec46697f16e098140423d8463 ] We tried to enable the audio on an imx6sx EVB with the codec nau8822, after setting the internal PLL fractional parameters, the audio still couldn't work and the there was no sdma irq at all. After checking with the section "8.1.1 Phase Locked Loop (PLL) Design Example" of "NAU88C22 Datasheet Rev 0.6", we found we need to turn off the PLL before programming fractional parameters and turn on the PLL after programming. After this change, the audio driver could record and play sound and the sdma's irq is triggered when playing or recording. Cc: David Lin <ctlin0(a)nuvoton.com> Cc: John Hsu <kchsu0(a)nuvoton.com> Cc: Seven Li <wtli(a)nuvoton.com> Signed-off-by: Hui Wang <hui.wang(a)canonical.com> Link: https://lore.kernel.org/r/20220530040151.95221-2-hui.wang@canonical.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/nau8822.c | 4 ++++ sound/soc/codecs/nau8822.h | 3 +++ 2 files changed, 7 insertions(+) diff --git a/sound/soc/codecs/nau8822.c b/sound/soc/codecs/nau8822.c index 58123390c7a3..b436e532993d 100644 --- a/sound/soc/codecs/nau8822.c +++ b/sound/soc/codecs/nau8822.c @@ -740,6 +740,8 @@ static int nau8822_set_pll(struct snd_soc_dai *dai, int pll_id, int source, pll_param->pll_int, pll_param->pll_frac, pll_param->mclk_scaler, pll_param->pre_factor); + snd_soc_component_update_bits(component, + NAU8822_REG_POWER_MANAGEMENT_1, NAU8822_PLL_EN_MASK, NAU8822_PLL_OFF); snd_soc_component_update_bits(component, NAU8822_REG_PLL_N, NAU8822_PLLMCLK_DIV2 | NAU8822_PLLN_MASK, (pll_param->pre_factor ? NAU8822_PLLMCLK_DIV2 : 0) | @@ -757,6 +759,8 @@ static int nau8822_set_pll(struct snd_soc_dai *dai, int pll_id, int source, pll_param->mclk_scaler << NAU8822_MCLKSEL_SFT); snd_soc_component_update_bits(component, NAU8822_REG_CLOCKING, NAU8822_CLKM_MASK, NAU8822_CLKM_PLL); + snd_soc_component_update_bits(component, + NAU8822_REG_POWER_MANAGEMENT_1, NAU8822_PLL_EN_MASK, NAU8822_PLL_ON); return 0; } diff --git a/sound/soc/codecs/nau8822.h b/sound/soc/codecs/nau8822.h index 489191ff187e..b45d42c15de6 100644 --- a/sound/soc/codecs/nau8822.h +++ b/sound/soc/codecs/nau8822.h @@ -90,6 +90,9 @@ #define NAU8822_REFIMP_3K 0x3 #define NAU8822_IOBUF_EN (0x1 << 2) #define NAU8822_ABIAS_EN (0x1 << 3) +#define NAU8822_PLL_EN_MASK (0x1 << 5) +#define NAU8822_PLL_ON (0x1 << 5) +#define NAU8822_PLL_OFF (0x0 << 5) /* NAU8822_REG_AUDIO_INTERFACE (0x4) */ #define NAU8822_AIFMT_MASK (0x3 << 3) -- 2.35.1

1 0

[PATCH] MAINTAINERS: update ASoC/Intel/SOF maintainers
by Pierre-Louis Bossart 13 Jun '22

13 Jun '22

Keyon Jie was a key contributor to the Intel ASoC and SOF Intel drivers, but he's moved on to a different role within Intel. We wish him all the best in his new endeavors. Bard Liao, Kai Vehmanen, Ranjani Sridharan and Peter Ujfalusi have been involved in the Intel multi-maintainer team, it's time to update the MAINTAINERS entry to reflect their contributions and clarify their role. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Péter Ujfalusi <peter.ujfalusi(a)linux.intel.com> --- MAINTAINERS | 9 +++++++-- 1 file changed, 7 insertions(+), 2 deletions(-) diff --git a/MAINTAINERS b/MAINTAINERS index a6d3bd9d2a8d0..440f3d7c93b9e 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -9803,7 +9803,10 @@ INTEL ASoC DRIVERS M: Cezary Rojewski <cezary.rojewski(a)intel.com> M: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> M: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> -M: Jie Yang <yang.jie(a)linux.intel.com> +M: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> +M: Bard Liao <yung-chuan.liao(a)linux.intel.com> +M: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> +M: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) S: Supported F: sound/soc/intel/ @@ -18670,8 +18673,10 @@ F: sound/soc/ SOUND - SOUND OPEN FIRMWARE (SOF) DRIVERS M: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> M: Liam Girdwood <lgirdwood(a)gmail.com> +M: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> +M: Bard Liao <yung-chuan.liao(a)linux.intel.com> M: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> -M: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> +R: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> M: Daniel Baluta <daniel.baluta(a)nxp.com> L: sound-open-firmware(a)alsa-project.org (moderated for non-subscribers) S: Supported -- 2.34.1

2 1

[PATCH] ASoC: Intel: bytcr_wm5102: Fix GPIO related probe-ordering problem
by Hans de Goede 13 Jun '22

13 Jun '22

The "wlf,spkvdd-ena" GPIO needed by the bytcr_wm5102 driver is made available through a gpio-lookup table. This gpio-lookup table is registered by drivers/mfd/arizona-spi.c, which may get probed after the bytcr_wm5102 driver. If the gpio-lookup table has not registered yet then the gpiod_get() will return -ENOENT. Treat -ENOENT as -EPROBE_DEFER to still keep things working in this case. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- sound/soc/intel/boards/bytcr_wm5102.c | 13 +++++++++++-- 1 file changed, 11 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/boards/bytcr_wm5102.c b/sound/soc/intel/boards/bytcr_wm5102.c index 00384c6fbcaa..330c0ace1638 100644 --- a/sound/soc/intel/boards/bytcr_wm5102.c +++ b/sound/soc/intel/boards/bytcr_wm5102.c @@ -421,8 +421,17 @@ static int snd_byt_wm5102_mc_probe(struct platform_device *pdev) priv->spkvdd_en_gpio = gpiod_get(codec_dev, "wlf,spkvdd-ena", GPIOD_OUT_LOW); put_device(codec_dev); - if (IS_ERR(priv->spkvdd_en_gpio)) - return dev_err_probe(dev, PTR_ERR(priv->spkvdd_en_gpio), "getting spkvdd-GPIO\n"); + if (IS_ERR(priv->spkvdd_en_gpio)) { + ret = PTR_ERR(priv->spkvdd_en_gpio); + /* + * The spkvdd gpio-lookup is registered by: drivers/mfd/arizona-spi.c, + * so -ENOENT means that arizona-spi hasn't probed yet. + */ + if (ret == -ENOENT) + ret = -EPROBE_DEFER; + + return dev_err_probe(dev, ret, "getting spkvdd-GPIO\n"); + } /* override platform name, if required */ byt_wm5102_card.dev = dev; -- 2.36.0

3 2

[PATCH 00/23] ASoC: SOF: IPC4: Add topology, control and PCM ops
by Ranjani Sridharan 13 Jun '22

13 Jun '22

This set of patches includes changes to add the topology, control and PCM ops for IPC4. It also includes a couple of patches to set the IPC4 BE DAI trigger ops for SSP/DMIC/HDA type DAI's. Bard Liao (1): ASoC: SOF: IPC4: add sdw blob Ranjani Sridharan (22): ASoC: SOF: Add topology tokens for IPC4 ASoC: SOF: IPC4: Introduce topology ops ASoC: SOF: ipc4-topology: Add support for parsing AIF_IN/AIF_OUT widgets ASoC: SOF: ipc4-topology: Add support for parsing DAI_IN/DAI_OUT widgets ASoC: SOF: ipc4-topology: Add prepare op for AIF type widgets ASoC: SOF: ipc4-topology: Add prepare op for DAI type widgets ASoC: SOF: ipc4-topology: Add support for parsing and preparing pga widgets ASoC: SOF: ipc4-topology: Add support for parsing mixer widgets ASoC: SOF: ipc4-topology: Add control_setup op ASoC: SOF: ipc4-topology: Add control IO ops ASoC: SOF: IPC4: Add pcm ops ASoC: SOF: ipc4-topology: Add widget_setup/widget_free ops ASoC: SOF: ipc4-topology: Add route_setup/route_free ops ASoC: SOF: ipc4-topology: Add the dai_config op ASoC: SOF: ipc4-pcm: Expose sof_ipc4_set_pipeline_state() ASoC: SOF: IPC4: set the BE DAI ops ASoC: SOF: Add ops_free ASoC: SOF: Intel: hda: init NHLT for IPC4 ASoC: SOF: Add two new structures for topology manifest data ASoC: SOF: Add a new IPC op for parsing topology manifest ASoC: SOF: ipc4-topology: Add support for SSP/DMIC DAI's AsoC: SOF: ipc4-topology: Add dai_get_clk op include/sound/sof.h | 1 + include/uapi/sound/sof/abi.h | 2 + include/uapi/sound/sof/header.h | 30 + include/uapi/sound/sof/tokens.h | 40 + sound/soc/sof/Makefile | 2 +- sound/soc/sof/core.c | 7 +- sound/soc/sof/intel/hda-dai.c | 201 +++- sound/soc/sof/intel/hda.h | 1 + sound/soc/sof/intel/pci-apl.c | 1 + sound/soc/sof/intel/pci-cnl.c | 1 + sound/soc/sof/intel/pci-icl.c | 1 + sound/soc/sof/intel/pci-tgl.c | 1 + sound/soc/sof/ipc3-topology.c | 48 + sound/soc/sof/ipc4-control.c | 216 ++++ sound/soc/sof/ipc4-pcm.c | 230 +++++ sound/soc/sof/ipc4-priv.h | 7 + sound/soc/sof/ipc4-topology.c | 1702 +++++++++++++++++++++++++++++++ sound/soc/sof/ipc4-topology.h | 234 +++++ sound/soc/sof/ipc4.c | 2 + sound/soc/sof/ops.h | 6 + sound/soc/sof/sof-audio.h | 11 + sound/soc/sof/topology.c | 114 +-- 22 files changed, 2794 insertions(+), 64 deletions(-) create mode 100644 sound/soc/sof/ipc4-control.c create mode 100644 sound/soc/sof/ipc4-pcm.c create mode 100644 sound/soc/sof/ipc4-topology.c create mode 100644 sound/soc/sof/ipc4-topology.h -- 2.25.1

2 24

[PATCH] ASoC: hdmi-codec: Update to modern DAI terminology
by Mark Brown 13 Jun '22

13 Jun '22

As part of retiring the old defines used to specify DAI formats update the hdmi_codec driver to use the modern names, including the variables in the struct hdmi_codec_daifmt exported to the DRM drivers. In updating this I did note that the only use of this information in DRM drivers is to reject clock provider settings, thinking about what this hardware is doing I rather suspect that there might not be any hardware out there which needs the configuration so it may be worth considering just having hdmi-codec support only clock consumer. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- drivers/gpu/drm/bridge/sii902x.c | 5 +++-- .../drm/bridge/synopsys/dw-hdmi-i2s-audio.c | 2 +- drivers/gpu/drm/exynos/exynos_hdmi.c | 8 ++++---- drivers/gpu/drm/i2c/tda998x_drv.c | 6 +++--- drivers/gpu/drm/sti/sti_hdmi.c | 8 ++++---- include/sound/hdmi-codec.h | 4 ++-- sound/soc/codecs/hdmi-codec.c | 18 +++++++++--------- 7 files changed, 26 insertions(+), 25 deletions(-) diff --git a/drivers/gpu/drm/bridge/sii902x.c b/drivers/gpu/drm/bridge/sii902x.c index 65549fbfdc87..be9736f67542 100644 --- a/drivers/gpu/drm/bridge/sii902x.c +++ b/drivers/gpu/drm/bridge/sii902x.c @@ -549,8 +549,9 @@ static int sii902x_audio_hw_params(struct device *dev, void *data, unsigned long mclk_rate; int i, ret; - if (daifmt->bit_clk_master || daifmt->frame_clk_master) { - dev_dbg(dev, "%s: I2S master mode not supported\n", __func__); + if (daifmt->bit_clk_provider || daifmt->frame_clk_provider) { + dev_dbg(dev, "%s: I2S clock provider mode not supported\n", + __func__); return -EINVAL; } diff --git a/drivers/gpu/drm/bridge/synopsys/dw-hdmi-i2s-audio.c b/drivers/gpu/drm/bridge/synopsys/dw-hdmi-i2s-audio.c index f50b47ac11a8..a2f0860b20bb 100644 --- a/drivers/gpu/drm/bridge/synopsys/dw-hdmi-i2s-audio.c +++ b/drivers/gpu/drm/bridge/synopsys/dw-hdmi-i2s-audio.c @@ -45,7 +45,7 @@ static int dw_hdmi_i2s_hw_params(struct device *dev, void *data, u8 inputclkfs = 0; /* it cares I2S only */ - if (fmt->bit_clk_master | fmt->frame_clk_master) { + if (fmt->bit_clk_provider | fmt->frame_clk_provider) { dev_err(dev, "unsupported clock settings\n"); return -EINVAL; } diff --git a/drivers/gpu/drm/exynos/exynos_hdmi.c b/drivers/gpu/drm/exynos/exynos_hdmi.c index 7655142a4651..10b0036f8a2e 100644 --- a/drivers/gpu/drm/exynos/exynos_hdmi.c +++ b/drivers/gpu/drm/exynos/exynos_hdmi.c @@ -1594,12 +1594,12 @@ static int hdmi_audio_hw_params(struct device *dev, void *data, struct hdmi_context *hdata = dev_get_drvdata(dev); if (daifmt->fmt != HDMI_I2S || daifmt->bit_clk_inv || - daifmt->frame_clk_inv || daifmt->bit_clk_master || - daifmt->frame_clk_master) { + daifmt->frame_clk_inv || daifmt->bit_clk_provider || + daifmt->frame_clk_provider) { dev_err(dev, "%s: Bad flags %d %d %d %d\n", __func__, daifmt->bit_clk_inv, daifmt->frame_clk_inv, - daifmt->bit_clk_master, - daifmt->frame_clk_master); + daifmt->bit_clk_provider, + daifmt->frame_clk_provider); return -EINVAL; } diff --git a/drivers/gpu/drm/i2c/tda998x_drv.c b/drivers/gpu/drm/i2c/tda998x_drv.c index b7ec6c374fbd..c4fadaecbb2d 100644 --- a/drivers/gpu/drm/i2c/tda998x_drv.c +++ b/drivers/gpu/drm/i2c/tda998x_drv.c @@ -1095,11 +1095,11 @@ static int tda998x_audio_hw_params(struct device *dev, void *data, if (!spdif && (daifmt->bit_clk_inv || daifmt->frame_clk_inv || - daifmt->bit_clk_master || daifmt->frame_clk_master)) { + daifmt->bit_clk_provider || daifmt->frame_clk_provider)) { dev_err(dev, "%s: Bad flags %d %d %d %d\n", __func__, daifmt->bit_clk_inv, daifmt->frame_clk_inv, - daifmt->bit_clk_master, - daifmt->frame_clk_master); + daifmt->bit_clk_provider, + daifmt->frame_clk_provider); return -EINVAL; } diff --git a/drivers/gpu/drm/sti/sti_hdmi.c b/drivers/gpu/drm/sti/sti_hdmi.c index b3fbee7eac11..622d0dfe7bb9 100644 --- a/drivers/gpu/drm/sti/sti_hdmi.c +++ b/drivers/gpu/drm/sti/sti_hdmi.c @@ -1175,12 +1175,12 @@ static int hdmi_audio_hw_params(struct device *dev, DRM_DEBUG_DRIVER("\n"); if ((daifmt->fmt != HDMI_I2S) || daifmt->bit_clk_inv || - daifmt->frame_clk_inv || daifmt->bit_clk_master || - daifmt->frame_clk_master) { + daifmt->frame_clk_inv || daifmt->bit_clk_providre || + daifmt->frame_clk_provider) { dev_err(dev, "%s: Bad flags %d %d %d %d\n", __func__, daifmt->bit_clk_inv, daifmt->frame_clk_inv, - daifmt->bit_clk_master, - daifmt->frame_clk_master); + daifmt->bit_clk_provider, + daifmt->frame_clk_provider); return -EINVAL; } diff --git a/include/sound/hdmi-codec.h b/include/sound/hdmi-codec.h index 4fc733c8c570..48ad33aba393 100644 --- a/include/sound/hdmi-codec.h +++ b/include/sound/hdmi-codec.h @@ -32,8 +32,8 @@ struct hdmi_codec_daifmt { } fmt; unsigned int bit_clk_inv:1; unsigned int frame_clk_inv:1; - unsigned int bit_clk_master:1; - unsigned int frame_clk_master:1; + unsigned int bit_clk_provider:1; + unsigned int frame_clk_provider:1; /* bit_fmt could be standard PCM format or * IEC958 encoded format. ALSA IEC958 plugin will pass * IEC958_SUBFRAME format to the underneath driver. diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c index b773466619b2..7d1e351f863a 100644 --- a/sound/soc/codecs/hdmi-codec.c +++ b/sound/soc/codecs/hdmi-codec.c @@ -606,18 +606,18 @@ static int hdmi_codec_i2s_set_fmt(struct snd_soc_dai *dai, /* Reset daifmt */ memset(cf, 0, sizeof(*cf)); - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBM_CFM: - cf->bit_clk_master = 1; - cf->frame_clk_master = 1; + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_CBP_CFP: + cf->bit_clk_provider = 1; + cf->frame_clk_provider = 1; break; - case SND_SOC_DAIFMT_CBS_CFM: - cf->frame_clk_master = 1; + case SND_SOC_DAIFMT_CBC_CFP: + cf->frame_clk_provider = 1; break; - case SND_SOC_DAIFMT_CBM_CFS: - cf->bit_clk_master = 1; + case SND_SOC_DAIFMT_CBP_CFC: + cf->bit_clk_provider = 1; break; - case SND_SOC_DAIFMT_CBS_CFS: + case SND_SOC_DAIFMT_CBC_CFC: break; default: return -EINVAL; -- 2.30.2

1 1

[PATCH] ASoC: wcd9335: Remove RX channel from old list before adding it to a new one
by Yassine Oudjana 13 Jun '22

13 Jun '22

From: Yassine Oudjana <y.oudjana(a)protonmail.com> Currently in slim_rx_mux_put, an RX channel gets added to a new list even if it is already in one. This can mess up links and make either it, the new list head, or both, get linked to the wrong entries. This can cause an entry to link to itself which in turn ends up making list_for_each_entry in other functions loop infinitely. To avoid issues, always remove the RX channel from any list it's in before adding it to a new list. Signed-off-by: Yassine Oudjana <y.oudjana(a)protonmail.com> --- sound/soc/codecs/wcd9335.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/wcd9335.c b/sound/soc/codecs/wcd9335.c index 617a36a89dfe..597420679505 100644 --- a/sound/soc/codecs/wcd9335.c +++ b/sound/soc/codecs/wcd9335.c @@ -1289,9 +1289,12 @@ static int slim_rx_mux_put(struct snd_kcontrol *kc, wcd->rx_port_value[port_id] = ucontrol->value.enumerated.item[0]; + /* Remove channel from any list it's in before adding it to a new one */ + list_del_init(&wcd->rx_chs[port_id].list); + switch (wcd->rx_port_value[port_id]) { case 0: - list_del_init(&wcd->rx_chs[port_id].list); + /* Channel already removed from lists. Nothing to do here */ break; case 1: list_add_tail(&wcd->rx_chs[port_id].list, -- 2.36.1

2 1

[PATCH] ASoC: wcd9335: Fix spurious event generation
by Mark Brown 13 Jun '22

13 Jun '22

The slimbus mux put operation unconditionally reports a change in value which means that spurious events are generated. Fix this by exiting early in that case. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/wcd9335.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/codecs/wcd9335.c b/sound/soc/codecs/wcd9335.c index 617a36a89dfe..d25594a708e0 100644 --- a/sound/soc/codecs/wcd9335.c +++ b/sound/soc/codecs/wcd9335.c @@ -1287,6 +1287,9 @@ static int slim_rx_mux_put(struct snd_kcontrol *kc, struct snd_soc_dapm_update *update = NULL; u32 port_id = w->shift; + if (wcd->rx_port_value[port_id] == ucontrol->value.enumerated.item[0]) + return 0; + wcd->rx_port_value[port_id] = ucontrol->value.enumerated.item[0]; switch (wcd->rx_port_value[port_id]) { -- 2.30.2

1 1

[PATCH] ASoC: wcd938x: Fix event generation for some controls
by Mark Brown 13 Jun '22

13 Jun '22

Currently wcd938x_*_put() unconditionally report that the value of the control changed, resulting in spurious events being generated. Return 0 in that case instead as we should. There is still an issue in the compander control which is a bit more complex. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/wcd938x.c | 13 +++++++++++++ 1 file changed, 13 insertions(+) diff --git a/sound/soc/codecs/wcd938x.c b/sound/soc/codecs/wcd938x.c index c1b61b997f69..d956c58298c3 100644 --- a/sound/soc/codecs/wcd938x.c +++ b/sound/soc/codecs/wcd938x.c @@ -2519,6 +2519,9 @@ static int wcd938x_tx_mode_put(struct snd_kcontrol *kcontrol, struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; int path = e->shift_l; + if (wcd938x->tx_mode[path] == ucontrol->value.enumerated.item[0]) + return 0; + wcd938x->tx_mode[path] = ucontrol->value.enumerated.item[0]; return 1; @@ -2540,6 +2543,10 @@ static int wcd938x_rx_hph_mode_put(struct snd_kcontrol *kcontrol, { struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); struct wcd938x_priv *wcd938x = snd_soc_component_get_drvdata(component); + int change; + + if (wcd938x->hph_mode == ucontrol->value.enumerated.item[0]) + return 0; wcd938x->hph_mode = ucontrol->value.enumerated.item[0]; @@ -2632,6 +2639,9 @@ static int wcd938x_ldoh_put(struct snd_kcontrol *kcontrol, struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); struct wcd938x_priv *wcd938x = snd_soc_component_get_drvdata(component); + if (wcd938x->ldoh == ucontrol->value.integer.value[0]) + return 0; + wcd938x->ldoh = ucontrol->value.integer.value[0]; return 1; @@ -2654,6 +2664,9 @@ static int wcd938x_bcs_put(struct snd_kcontrol *kcontrol, struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); struct wcd938x_priv *wcd938x = snd_soc_component_get_drvdata(component); + if (wcd938x->bcs_dis == ucontrol->value.integer.value[0]) + return 0; + wcd938x->bcs_dis = ucontrol->value.integer.value[0]; return 1; -- 2.30.2

2 3

There is headset Mic issue on HP ALC897 Platform
by Kailang 13 Jun '22

13 Jun '22

Hi Takashi, HP ALC897 platform. It need to fill 0x19 verb table to enable Headset MIC. Attach was the supporting patch. BR, Kailang

2 1

[PATCH 0/2] Update external mclk name
by Srinivasa Rao Mandadapu 13 Jun '22

13 Jun '22

This patch set is to update external mclk name. Srinivasa Rao Mandadapu (2): ASoC: dt-bindings: lpass-cpu: Update external mclck0 name ASoC: qcom: lpass-sc7280: Update external mclk0 name Documentation/devicetree/bindings/sound/qcom,lpass-cpu.yaml | 2 +- sound/soc/qcom/lpass-sc7280.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) -- 2.7.4

3 5

[PATCH V0 0/1] asoc: msm: use hashtable to check kcontrol
by Raghu Bankapur 13 Jun '22

13 Jun '22

use hashtabe instead of linear list to check kcontrol before adding them for improving early audio KPI. With this changes we see 600ms improvement in start of audio Raghu Bankapur (1): asoc: msm: use hashtable to check kcontrol include/sound/control.h | 4 ++ include/sound/core.h | 12 +++++- sound/core/control.c | 92 +++++++++++++++++++++++++++++++++-------- sound/core/init.c | 3 ++ sound/soc/Kconfig | 9 ++++ 5 files changed, 101 insertions(+), 19 deletions(-) -- 2.17.1

5 10

[PATCH 00/56] Specify clock provider directly to CPU DAIs
by Charles Keepax 13 Jun '22

13 Jun '22

Currently the set_fmt callback always passes clock provider/consumer with respect to the CODEC. This made sense when the framework was directly broken down into platforms and CODECs. However, as things are now broken down into components which can be connected as either the CPU or CODEC side of a DAI link it simplifies things if each side of the link is just told if it is provider or consumer of the clocks. Making this change allows us to remove one of the last parts of the ASoC core that needs to know if a driver is a CODEC driver, where it flips the clock format specifier if a CODEC driver is used on the CPU side of a DAI link, as well as just being conceptually more consistent with componentisation. The basic idea of this patch chain is to change the set_fmt callback from specifying if the CODEC is provider/consumer into directly specifying if the component is provider/consumer. To do this we add some new defines, and then to preserve bisectability, the migration is done by adding a new callback, converting over all existing CPU side drivers, converting the core, and then finally reverting back to the old callback. Converting the platform drivers makes sense as the existing defines are from the perspective of the CODEC and there are more CODEC drivers than platform drivers. Obviously a fair amount of this patch chain I was only able to build test, so any testing that can be done would be greatly appreciated. Thanks, Charles Charles Keepax (56): ASoC: core: Add set_fmt_new callback that directly specifies provider ASoC: amd: vangogh: Update to use set_fmt_new callback ASoC: atmel: Update to use set_fmt_new callback ASoC: au1x: Update to use set_fmt_new callback ASoC: bcm: Update to use set_fmt_new callback ASoC: ep93xx: Update to use set_fmt_new callback ASoC: dwc: Update to use set_fmt_new callback ASoC: fsl: Update to use set_fmt_new callback ASoC: hisilicon: Update to use set_fmt_new callback ASoC: img: Update to use set_fmt_new callback ASoC: Intel: Update to use set_fmt_new callback ASoC: js4740-i2s: Update to use set_fmt_new callback ASoC: mediatek: Update to use set_fmt_new callback ASoC: meson: Update to use set_fmt_new callback ASoC: mxs-saif: Update to use set_fmt_new callback ASoC: pxa: Update to use set_fmt_new callback ASoC: qcom: Update to use set_fmt_new callback ASoC: rockchip: Update to use set_fmt_new callback ASoC: samsung: Update to use set_fmt_new callback ASoC: sh: Update to use set_fmt_new callback ASoC: stm: Update to use set_fmt_new callback ASoC: sunxi: Update to use set_fmt_new callback ASoC: tegra: Update to use set_fmt_new callback ASoC: test-component: Update to use set_fmt_new callback ASoC: ti: Update to use set_fmt_new callback ASoC: ux500: Update to use set_fmt_new callback ASoC: xtensa: Update to use set_fmt_new callback ASoC: core: Always send the CPU DAI a direct clock specifier ASoC: amd: vangogh: Rename set_fmt_new back to set_fmt ASoC: atmel: Rename set_fmt_new back to set_fmt ASoC: au1x: Rename set_fmt_new back to set_fmt ASoC: bcm: Rename set_fmt_new back to set_fmt ASoC: ep93xx: Rename set_fmt_new back to set_fmt ASoC: dwc: Rename set_fmt_new back to set_fmt ASoC: fsl: Rename set_fmt_new back to set_fmt ASoC: hisilicon: Rename set_fmt_new back to set_fmt ASoC: img: Rename set_fmt_new back to set_fmt ASoC: Intel: Rename set_fmt_new back to set_fmt ASoC: jz4740-i2s: Rename set_fmt_new back to set_fmt ASoC: mediatek: Rename set_fmt_new back to set_fmt ASoC: meson: Rename set_fmt_new back to set_fmt ASoC: mxs-saif: Rename set_fmt_new back to set_fmt ASoC: pxa: Rename set_fmt_new back to set_fmt ASoC: qcom: Rename set_fmt_new back to set_fmt ASoC: rockchip: Rename set_fmt_new back to set_fmt ASoC: samsung: Rename set_fmt_new back to set_fmt ASoC: sh: Rename set_fmt_new back to set_fmt ASoC: stm: Rename set_fmt_new back to set_fmt ASoC: sunxi: Rename set_fmt_new back to set_fmt ASoC: tegra: Rename set_fmt_new back to set_fmt ASoC: test-component: Rename set_fmt_new back to set_fmt ASoC: ti: Rename set_fmt_new back to set_fmt ASoC: ux500: Rename set_fmt_new back to set_fmt ASoC: xtensa: Rename set_fmt_new back to set_fmt ASoC: soc-dai: Remove set_fmt_new callback ASoC: simple-card-utils: Move snd_soc_component_is_codec to be local include/sound/soc-component.h | 5 --- include/sound/soc-dai.h | 6 ++++ sound/soc/amd/vangogh/acp5x-i2s.c | 4 +-- sound/soc/atmel/atmel-i2s.c | 4 +-- sound/soc/atmel/atmel_ssc_dai.c | 18 +++++----- sound/soc/atmel/mchp-i2s-mcc.c | 8 ++--- sound/soc/atmel/mchp-pdmc.c | 4 +-- sound/soc/au1x/i2sc.c | 2 +- sound/soc/au1x/psc-i2s.c | 4 +-- sound/soc/bcm/bcm2835-i2s.c | 20 +++++------ sound/soc/bcm/cygnus-ssp.c | 4 +-- sound/soc/cirrus/ep93xx-i2s.c | 4 +-- sound/soc/dwc/dwc-i2s.c | 8 ++--- sound/soc/fsl/fsl_audmix.c | 6 ++-- sound/soc/fsl/fsl_esai.c | 8 ++--- sound/soc/fsl/fsl_mqs.c | 2 +- sound/soc/fsl/fsl_sai.c | 8 ++--- sound/soc/fsl/fsl_ssi.c | 22 ++++++------ sound/soc/fsl/imx-audmix.c | 4 +-- sound/soc/fsl/imx-card.c | 2 +- sound/soc/generic/simple-card-utils.c | 7 +++- sound/soc/generic/test-component.c | 18 +++++----- sound/soc/hisilicon/hi6210-i2s.c | 18 +++++----- sound/soc/img/img-i2s-in.c | 4 +-- sound/soc/img/img-i2s-out.c | 6 ++-- sound/soc/intel/atom/sst-atom-controls.c | 4 +-- sound/soc/intel/boards/bytcht_cx2072x.c | 2 +- sound/soc/intel/boards/bytcht_da7213.c | 2 +- sound/soc/intel/boards/bytcht_es8316.c | 2 +- sound/soc/intel/boards/bytcht_nocodec.c | 2 +- sound/soc/intel/boards/bytcr_rt5640.c | 2 +- sound/soc/intel/boards/bytcr_rt5651.c | 2 +- sound/soc/intel/boards/bytcr_wm5102.c | 2 +- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 3 +- sound/soc/intel/boards/cht_bsw_rt5645.c | 6 ++-- sound/soc/intel/boards/cht_bsw_rt5672.c | 2 +- sound/soc/intel/keembay/kmb_platform.c | 4 +-- sound/soc/jz4740/jz4740-i2s.c | 10 +++--- sound/soc/mediatek/mt8195/mt8195-dai-etdm.c | 6 ++-- sound/soc/mediatek/mt8195/mt8195-dai-pcm.c | 6 ++-- sound/soc/meson/aiu-encoder-i2s.c | 2 +- sound/soc/meson/axg-tdm-interface.c | 14 ++++---- sound/soc/mxs/mxs-saif.c | 4 +-- sound/soc/pxa/magician.c | 8 ++--- sound/soc/pxa/mmp-sspa.c | 6 ++-- sound/soc/pxa/pxa-ssp.c | 22 ++++++------ sound/soc/pxa/pxa2xx-i2s.c | 6 ++-- sound/soc/qcom/apq8016_sbc.c | 2 +- sound/soc/qcom/qdsp6/audioreach.c | 4 +-- sound/soc/qcom/qdsp6/q6afe.c | 6 ++-- sound/soc/qcom/sc7180.c | 2 +- sound/soc/qcom/sdm845.c | 6 ++-- sound/soc/qcom/sm8250.c | 4 +-- sound/soc/rockchip/rockchip_i2s.c | 6 ++-- sound/soc/rockchip/rockchip_i2s_tdm.c | 6 ++-- sound/soc/samsung/i2s.c | 6 ++-- sound/soc/samsung/pcm.c | 4 +-- sound/soc/samsung/s3c-i2s-v2.c | 6 ++-- sound/soc/samsung/s3c24xx-i2s.c | 6 ++-- sound/soc/sh/fsi.c | 6 ++-- sound/soc/sh/rcar/core.c | 4 +-- sound/soc/sh/rz-ssi.c | 2 +- sound/soc/sh/ssi.c | 10 +++--- sound/soc/soc-core.c | 14 ++------ sound/soc/soc-dai.c | 3 +- sound/soc/stm/stm32_i2s.c | 8 ++--- sound/soc/stm/stm32_sai_sub.c | 8 ++--- sound/soc/sunxi/sun4i-i2s.c | 18 +++++----- sound/soc/sunxi/sun8i-codec.c | 6 ++-- sound/soc/tegra/tegra20_i2s.c | 6 ++-- sound/soc/tegra/tegra210_i2s.c | 6 ++-- sound/soc/tegra/tegra30_i2s.c | 6 ++-- sound/soc/ti/davinci-i2s.c | 32 ++++++++--------- sound/soc/ti/davinci-mcasp.c | 10 +++--- sound/soc/ti/omap-mcbsp.c | 12 +++---- sound/soc/ux500/ux500_msp_dai.c | 36 ++++++++++---------- sound/soc/xtensa/xtfpga-i2s.c | 4 +-- 77 files changed, 279 insertions(+), 283 deletions(-) -- 2.30.2

8 72

[PATCH 00/17] ASoC: Intel: haswell and broadwell boards update
by Cezary Rojewski 13 Jun '22

13 Jun '22

A number of patches improving overall quality and readability of haswell.c and broadwell.c source files found in sound/soc/intel/boards. Both files are first renamed and only then actual changes are being incrementally added. The respective names are: hsw_rt5640 and bdw_rt286 to match the pattern found in more recent boards. Most patches bring no functional change - the more impactful patches at are placed the end: Refactor of suspend/resume flow for the bdw_rt286 board by dropping dev->remove() in favour of card->remove() and adjust jack handling to reduce code size slightly by implementing card_set_jack(). The last patch is removing of FE DAI ops. Given the existence of platform FE DAI capabilities (either static declaration or through topology file), this code is redundant. Cezary Rojewski (17): ASoC: Intel: Rename haswell source file to hsw_rt5640 ASoC: Intel: hsw_rt5640: Reword prefixes of all driver members ASoC: Intel: hsw_rt5640: Reword driver name ASoC: Intel: hsw_rt5640: Update code indentation ASoC: Intel: hsw_rt5640: Update file comments ASoC: Intel: hsw_rt5640: Improve probe() function quality ASoC: Intel: hsw_rt5640: Improve hw_params() debug-ability ASoC: Intel: Rename broadwell source file to bdw_rt286 ASoC: Intel: bdw_rt286: Reword prefixes of all driver members ASoC: Intel: bdw_rt286: Reword driver name ASoC: Intel: bdw_rt286: Update code indentation ASoC: Intel: bdw_rt286: Update file comments ASoC: Intel: bdw_rt286: Improve probe() function quality ASoC: Intel: bdw_rt286: Improve hw_params() debug-ability ASoC: Intel: bdw_rt286: Improve codec_init() quality ASoC: Intel: bdw_rt286: Refactor suspend/resume ASoC: Intel: bdw_rt286: Remove FE DAI ops sound/soc/intel/boards/Kconfig | 4 +- sound/soc/intel/boards/Makefile | 4 +- sound/soc/intel/boards/bdw_rt286.c | 257 +++++++++++++ sound/soc/intel/boards/broadwell.c | 338 ------------------ sound/soc/intel/boards/haswell.c | 202 ----------- sound/soc/intel/boards/hsw_rt5640.c | 176 +++++++++ .../common/soc-acpi-intel-hsw-bdw-match.c | 6 +- 7 files changed, 440 insertions(+), 547 deletions(-) create mode 100644 sound/soc/intel/boards/bdw_rt286.c delete mode 100644 sound/soc/intel/boards/broadwell.c delete mode 100644 sound/soc/intel/boards/haswell.c create mode 100644 sound/soc/intel/boards/hsw_rt5640.c -- 2.25.1

2 23

[PATCH 00/23] Update Documentation/ cross-references
by Mauro Carvalho Chehab 13 Jun '22

13 Jun '22

Hi John, There were a number of DT binding conversions and other docs change that were not updated. Address them, in order to keep the cross-references on a sane state. Patch series is against v5.19-rc1 (and applies cleanly on the top of today's -next). Mauro Carvalho Chehab (23): dt-bindings: mfd: bd9571mwv: update rohm,bd9571mwv.yaml reference dt-bindings: interrupt-controller: update brcm,l2-intc.yaml reference dt-bindings: arm: update vexpress-config.yaml references dt-bindings: reset: update st,stih407-powerdown.yaml references dt-bindings: mfd: rk808: update rockchip,rk808.yaml reference dt-bindings: mmc: exynos-dw-mshc: update samsung,pinctrl.yaml reference docs: netdev: update maintainer-netdev.rst reference docs: filesystems: update netfs-api.rst reference Documentation: update watch_queue.rst references Documentation: KVM: update s390-pv.rst reference Documentation: KVM: update amd-memory-encryption.rst references Documentation: KVM: update msr.rst reference Documentation: KVM: update s390-diag.rst reference MAINTAINERS: update arm,hdlcd.yaml reference MAINTAINERS: update arm,komeda.yaml reference MAINTAINERS: update arm,malidp.yaml reference MAINTAINERS: update cortina,gemini-ethernet.yaml reference MAINTAINERS: update dongwoon,dw9807-vcm.yaml reference MAINTAINERS: update maxim,max77693.yaml reference MAINTAINERS: update snps,axs10x-reset.yaml reference objtool: update objtool.txt references ASoC: wm8731: update wlf,wm8731.yaml reference arch: m68k: q40: README: drop references to IDE driver .../ABI/testing/sysfs-driver-bd9571mwv-regulator | 2 +- Documentation/admin-guide/kernel-parameters.txt | 2 +- .../bindings/cpufreq/brcm,stb-avs-cpu-freq.txt | 2 +- .../devicetree/bindings/hwmon/vexpress.txt | 2 +- .../devicetree/bindings/mmc/exynos-dw-mshc.txt | 2 +- .../devicetree/bindings/phy/phy-stih407-usb.txt | 2 +- .../devicetree/bindings/pinctrl/pinctrl-rk805.txt | 2 +- .../devicetree/bindings/regulator/vexpress.txt | 2 +- .../bindings/sound/atmel-sam9x5-wm8731-audio.txt | 2 +- Documentation/devicetree/bindings/usb/dwc3-st.txt | 2 +- Documentation/devicetree/bindings/usb/ehci-st.txt | 2 +- Documentation/devicetree/bindings/usb/ohci-st.txt | 2 +- Documentation/security/keys/core.rst | 2 +- Documentation/security/secrets/coco.rst | 2 +- .../translations/it_IT/networking/netdev-FAQ.rst | 2 +- Documentation/virt/kvm/api.rst | 4 ++-- Documentation/virt/kvm/s390/s390-pv-boot.rst | 2 +- Documentation/virt/kvm/x86/hypercalls.rst | 2 +- Documentation/x86/orc-unwinder.rst | 2 +- MAINTAINERS | 14 +++++++------- arch/m68k/q40/README | 4 +--- include/linux/fscache.h | 2 +- include/linux/objtool.h | 2 +- include/linux/watch_queue.h | 2 +- init/Kconfig | 2 +- kernel/watch_queue.c | 2 +- lib/Kconfig.debug | 2 +- tools/include/linux/objtool.h | 2 +- tools/objtool/check.c | 2 +- 29 files changed, 36 insertions(+), 38 deletions(-) -- 2.36.1

4 5

ucm2: add ALC4080 ID for Asus Z690-I Gaming Wifi
by GitHub pull_request - opened 13 Jun '22

13 Jun '22

alsa-project/alsa-ucm-conf pull request #169 was opened from brenmous: ``` inxi -M Machine: Type: Desktop System: ASUS product: N/A v: N/A serial: <superuser required> Mobo: ASUSTeK model: ROG STRIX Z690-I GAMING WIFI v: Rev 1.xx serial: <superuser required> UEFI: American Megatrends v: 1403 date: 03/30/2022 ``` ``` aplay -l **** List of PLAYBACK Hardware Devices **** card 1: HDMI [HDA ATI HDMI], device 3: HDMI 0 [27GL850] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: HDMI [HDA ATI HDMI], device 7: HDMI 1 [HDMI 1] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: HDMI [HDA ATI HDMI], device 8: HDMI 2 [HDMI 2] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: HDMI [HDA ATI HDMI], device 9: HDMI 3 [HDMI 3] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: HDMI [HDA ATI HDMI], device 10: HDMI 4 [HDMI 4] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: HDMI [HDA ATI HDMI], device 11: HDMI 5 [HDMI 5] Subdevices: 1/1 Subdevice #0: subdevice #0 card 3: Audio [USB Audio], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0 card 3: Audio [USB Audio], device 1: USB Audio [USB Audio #1] Subdevices: 0/1 Subdevice #0: subdevice #0 card 3: Audio [USB Audio], device 2: USB Audio [USB Audio #2] Subdevices: 1/1 Subdevice #0: subdevice #0 card 3: Audio [USB Audio], device 3: USB Audio [USB Audio #3] Subdevices: 1/1 Subdevice #0: subdevice #0 ``` Card 3 is ALC4080 ``` udevadm info -a -p /sys/class/sound/card3 | grep idVendor -m1 ATTRS{idVendor}=="0b05" udevadm info -a -p /sys/class/sound/card3 | grep idProduct -m1 ATTRS{idProduct}=="1a20" ``` Rear panel line out and front panel headphone now work. Jack detection also working properly. Haven't tested any of the other inputs/outputs as I don't have suitable devices. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/169 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/169.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH] soc: fix typos in comment
by Ren Yu 13 Jun '22

13 Jun '22

change 'slection' to 'selection' Signed-off-by: Ren Yu <renyu(a)nfschina.com> --- sound/soc/codecs/lm49453.c | 2 +- sound/soc/codecs/rt5631.h | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/lm49453.c b/sound/soc/codecs/lm49453.c index bd0078e4499b..2eb43e93778e 100644 --- a/sound/soc/codecs/lm49453.c +++ b/sound/soc/codecs/lm49453.c @@ -1201,7 +1201,7 @@ static int lm49453_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id, case 12288000: case 26000000: case 19200000: - /* pll clk slection */ + /* pll clk selection */ pll_clk = 0; break; case 48000: diff --git a/sound/soc/codecs/rt5631.h b/sound/soc/codecs/rt5631.h index 8a6b99a48c7c..0caca2e1d31b 100644 --- a/sound/soc/codecs/rt5631.h +++ b/sound/soc/codecs/rt5631.h @@ -373,7 +373,7 @@ */ #define RT5631_SDP_DAC_DATA_L_R_SWAP (0x1 << 4) -/* Data Length Slection */ +/* Data Length Selection */ #define RT5631_SDP_I2S_DL_MASK (0x3 << 2) #define RT5631_SDP_I2S_DL_16 (0x0 << 2) #define RT5631_SDP_I2S_DL_20 (0x1 << 2) -- 2.11.0

1 0

[PATCH v7 0/8] ASoC: mediatek: Add support for MT8186 SoC
by Jiaxin Yu 13 Jun '22

13 Jun '22

This series of patches adds support for Mediatek AFE of MT8186 Soc. Patches are based on broonie tree "for-next" branch. Changes since v6: - The Makefile and Kconfig updates are moved into the driver patches so that can keep independence of each patch. Changes since v5: - fix build error about function snd_soc_card_jack_new that modified by: Link: https://lore.kernel.org/r/20220408041114.6024-1-akihiko.odaki@gmail.com - some have been accepted, details are in the link below: Link: https://lore.kernel.org/all/165459931885.399031.2621579592368573898.b4-ty@k… Changes since v4: - [v5 07/20] - remove unsusd controls - [v5 09/20] - correct indent error - [v5 10/20] - [v5 13/20] - [v5 14/20] - fix the return value if the value is different from the previous value in mixer controls - [v5 17/20] - [v5 19/20] - correct the compatible name with '_' instead of '-' - [v5 18/20] - [v5 20/20] - correct the yaml after 'pip3 install dtschema --upgrade' Changes since v3: - [v4 09/18] - remove DEBUG_COEFF debugging code - [v4 10/18] - simplify the logic of the code - [v4 13/18] - split out the MT6366 bits into mt8186-mt6366-commom.c - fix build error of "error: 'struct dev_pm_info' has no member named 'runtime_error'" - fix bug of adda dai driver - add route for pcm interface channel 2. - [v4 15/18] - [v4 17/18] - commonize the configuration of the codec - move MT6366 common bits into mt8186-mt6366-common.c Changes since v2: - add a new compatible string "mediatek,mt6366-sound" - modify the log level for simplicity - use dev_err_probe(...) instead of dev_err(...) in dev probe() - optimized the logic of some code - use BIT() and GENMASK() macros to descript the registers Thanks for AngeloGioacchino's careful reviews. Changes since v1: [v2 01/17] - add a new ID to the existing mt6358 codec driver [v2 03/17] - modify log level in DAPM events - use standard numeric control with name ending in Switch - return 1 when the value changed in mixer control's .get callback [v2 05/17] - ending in Switch to the standard on/off controls - change to "HW Gain 1 Volume" and "HW Gain 2 Volume" [v2 09/17] - return an error in the default case rather than just picking one of the behaviours when do .set_fmt - use the new defines that are _PROVIDER_MASK, _DAIFMT_CBP_CFP and _DAIFMT_CBC_CFC [v2 10/17] [v2 11/17] - the clock and gpio are aplit out into separate patches The source file's GPL comment use c++ style, and the header fils's GPL comment use c style. We have added "Switch" after the names of all the controls that just are simple on/off. Jiaxin Yu (8): dt-bindings: mediatek: mt6358: add new compatible for using mt6366 ASoC: mediatek: mt8186: add platform driver ASoC: mediatek: mt8186: add mt8186-mt6366 common driver dt-bindings: mediatek: mt8186: add audio afe document ASoC: mediatek: mt8186: add machine driver with mt6366, da7219 and max98357 dt-bindings: mediatek: mt8186: add mt8186-mt6366-da7219-max98357 document ASoC: mediatek: mt8186: add machine driver with mt6366, rt1019 and rt5682s dt-bindings: mediatek: mt8186: add mt8186-mt6366-rt1019-rt5682s document .../devicetree/bindings/sound/mt6358.txt | 4 +- .../bindings/sound/mt8186-afe-pcm.yaml | 175 + .../sound/mt8186-mt6366-da7219-max98357.yaml | 75 + .../sound/mt8186-mt6366-rt1019-rt5682s.yaml | 75 + sound/soc/mediatek/Kconfig | 44 + sound/soc/mediatek/Makefile | 1 + sound/soc/mediatek/mt8186/Makefile | 22 + sound/soc/mediatek/mt8186/mt8186-afe-common.h | 235 ++ .../soc/mediatek/mt8186/mt8186-afe-control.c | 261 ++ sound/soc/mediatek/mt8186/mt8186-afe-pcm.c | 3009 +++++++++++++++++ .../mediatek/mt8186/mt8186-mt6366-common.c | 59 + .../mediatek/mt8186/mt8186-mt6366-common.h | 17 + .../mt8186/mt8186-mt6366-da7219-max98357.c | 1002 ++++++ .../mt8186/mt8186-mt6366-rt1019-rt5682s.c | 978 ++++++ 14 files changed, 5956 insertions(+), 1 deletion(-) create mode 100644 Documentation/devicetree/bindings/sound/mt8186-afe-pcm.yaml create mode 100644 Documentation/devicetree/bindings/sound/mt8186-mt6366-da7219-max98357.yaml create mode 100644 Documentation/devicetree/bindings/sound/mt8186-mt6366-rt1019-rt5682s.yaml create mode 100644 sound/soc/mediatek/mt8186/Makefile create mode 100644 sound/soc/mediatek/mt8186/mt8186-afe-common.h create mode 100644 sound/soc/mediatek/mt8186/mt8186-afe-control.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-afe-pcm.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-mt6366-common.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-mt6366-common.h create mode 100644 sound/soc/mediatek/mt8186/mt8186-mt6366-da7219-max98357.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-mt6366-rt1019-rt5682s.c -- 2.18.0

2 12

[PATCH] soc: There are several spelling mistakes. Fix them
by Ren Yu 13 Jun '22

13 Jun '22

Signed-off-by: Ren Yu <renyu(a)nfschina.com> --- sound/soc/codecs/lm49453.c | 2 +- sound/soc/codecs/rt5631.h | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/lm49453.c b/sound/soc/codecs/lm49453.c index bd0078e4499b..2eb43e93778e 100644 --- a/sound/soc/codecs/lm49453.c +++ b/sound/soc/codecs/lm49453.c @@ -1201,7 +1201,7 @@ static int lm49453_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id, case 12288000: case 26000000: case 19200000: - /* pll clk slection */ + /* pll clk selection */ pll_clk = 0; break; case 48000: diff --git a/sound/soc/codecs/rt5631.h b/sound/soc/codecs/rt5631.h index 8a6b99a48c7c..0caca2e1d31b 100644 --- a/sound/soc/codecs/rt5631.h +++ b/sound/soc/codecs/rt5631.h @@ -373,7 +373,7 @@ */ #define RT5631_SDP_DAC_DATA_L_R_SWAP (0x1 << 4) -/* Data Length Slection */ +/* Data Length Selection */ #define RT5631_SDP_I2S_DL_MASK (0x3 << 2) #define RT5631_SDP_I2S_DL_16 (0x0 << 2) #define RT5631_SDP_I2S_DL_20 (0x1 << 2) -- 2.11.0

1 0

[PATCH] ALSA: usb-audio: US16x08: Move overflow check before array access
by Daniil Dementev 13 Jun '22

13 Jun '22

Buffer overflow could occur in the loop "while", due to accessing an array element before checking the index. Found by Linux Verification Center (linuxtesting.org) with SVACE. Signed-off-by: Daniil Dementev <d.dementev(a)ispras.ru> Reviewed-by: Alexey Khoroshilov <khoroshilov(a)ispras.ru> --- sound/usb/mixer_us16x08.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/usb/mixer_us16x08.c b/sound/usb/mixer_us16x08.c index b7b6f3834ed5..6eb7d93b358d 100644 --- a/sound/usb/mixer_us16x08.c +++ b/sound/usb/mixer_us16x08.c @@ -637,10 +637,10 @@ static int snd_get_meter_comp_index(struct snd_us16x08_meter_store *store) } } else { /* skip channels with no compressor active */ - while (!store->comp_store->val[ + while (store->comp_index <= SND_US16X08_MAX_CHANNELS + && !store->comp_store->val[ COMP_STORE_IDX(SND_US16X08_ID_COMP_SWITCH)] - [store->comp_index - 1] - && store->comp_index <= SND_US16X08_MAX_CHANNELS) { + [store->comp_index - 1]) { store->comp_index++; } ret = store->comp_index++; -- 2.35.1.windows.2

2 1

Asus Z690-E: No Profiles Available in alsa-lib 1.2.7-2
by GitHub issues - opened 11 Jun '22

11 Jun '22

alsa-project/alsa-ucm-conf issue #168 was opened from BrothersRM: Upgrading alsa-lib from 1.2.6.1-1 to 1.2.7-2 seems to have broken support for my motherboard's sound card. Downgrading the package did resolve the issue. Device is listed in `lsusb -v` as : `Bus 001 Device 002: ID 0b05:1996 ASUSTek Computer, Inc. USB Audio` My hardware configuration uses speakers attached to the front, surround and subwoofer ports on my motherboard. Some interesting things I noticed while it was broken: - Plugging headphones into the front-panel headphone port allowed me to turn on the rear speakers, though they didn't seem to use the correct profile. Max volume was far lower - Though the device was listed as unavailable, I could manually enable the rear speakers using pavucontrol when front headphones weren't plugged in - Trying to run speaker-test on the device failed with 'no valid configuration' Please let me know if I can provide more information / do any testing. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/168 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

1.2.7 aplay doesn't compile
by GitHub issues - opened 11 Jun '22

11 Jun '22

alsa-project/alsa-utils issue #153 was opened from LoneFox78: I'm getting this error: ``` Making all in aplay make[1]: Entering directory '/mnt/pkg/pkgmk_work/alsa-utils/src/alsa-utils-1.2.7/aplay' gcc -DHAVE_CONFIG_H -I. -I../include -I../include -s -Wl,-s -Wall -O2 -march=native -ftree-vectorize -pipe -MT aplay.o -MD -MP -MF .deps/aplay.Tpo -c -o aplay.o aplay.c aplay.c:44:10: fatal error: alsa/use-case.h: No such file or directory 44 | #include <alsa/use-case.h> | ^~~~~~~~~~~~~~~~~ compilation terminated. make[1]: *** [Makefile:495: aplay.o] Error 1 make[1]: Leaving directory '/mnt/pkg/pkgmk_work/alsa-utils/src/alsa-utils-1.2.7/aplay' make: *** [Makefile:461: all-recursive] Error 1 ``` The configure script already looks for that header, but I guess someone forgot to add the corresponding <code>#ifdef</code>s in this file. Issue URL : https://github.com/alsa-project/alsa-utils/issues/153 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[PATCH] ALSA: pcm: add 24KHz sample rate definition
by Patrick Lai 11 Jun '22

11 Jun '22

Add 24KHz sample rate definition as some hardware can support 24KHz sample rate PCM data Signed-off-by: Patrick Lai <quic_plai(a)quicinc.com> --- include/sound/pcm.h | 24 +++++++++++++----------- sound/core/pcm_native.c | 4 ++-- 2 files changed, 15 insertions(+), 13 deletions(-) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 6b99310b5b88..a761a4459091 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -111,23 +111,25 @@ struct snd_pcm_ops { #define SNDRV_PCM_RATE_11025 (1<<2) /* 11025Hz */ #define SNDRV_PCM_RATE_16000 (1<<3) /* 16000Hz */ #define SNDRV_PCM_RATE_22050 (1<<4) /* 22050Hz */ -#define SNDRV_PCM_RATE_32000 (1<<5) /* 32000Hz */ -#define SNDRV_PCM_RATE_44100 (1<<6) /* 44100Hz */ -#define SNDRV_PCM_RATE_48000 (1<<7) /* 48000Hz */ -#define SNDRV_PCM_RATE_64000 (1<<8) /* 64000Hz */ -#define SNDRV_PCM_RATE_88200 (1<<9) /* 88200Hz */ -#define SNDRV_PCM_RATE_96000 (1<<10) /* 96000Hz */ -#define SNDRV_PCM_RATE_176400 (1<<11) /* 176400Hz */ -#define SNDRV_PCM_RATE_192000 (1<<12) /* 192000Hz */ -#define SNDRV_PCM_RATE_352800 (1<<13) /* 352800Hz */ -#define SNDRV_PCM_RATE_384000 (1<<14) /* 384000Hz */ +#define SNDDV_PCM_RATE_24000 (1<<5) /* 24000Hz */ +#define SNDRV_PCM_RATE_32000 (1<<6) /* 32000Hz */ +#define SNDRV_PCM_RATE_44100 (1<<7) /* 44100Hz */ +#define SNDRV_PCM_RATE_48000 (1<<8) /* 48000Hz */ +#define SNDRV_PCM_RATE_64000 (1<<9) /* 64000Hz */ +#define SNDRV_PCM_RATE_88200 (1<<10) /* 88200Hz */ +#define SNDRV_PCM_RATE_96000 (1<<11) /* 96000Hz */ +#define SNDRV_PCM_RATE_176400 (1<<12) /* 176400Hz */ +#define SNDRV_PCM_RATE_192000 (1<<13) /* 192000Hz */ +#define SNDRV_PCM_RATE_352800 (1<<14) /* 352800Hz */ +#define SNDRV_PCM_RATE_384000 (1<<15) /* 384000Hz */ #define SNDRV_PCM_RATE_CONTINUOUS (1<<30) /* continuous range */ #define SNDRV_PCM_RATE_KNOT (1<<31) /* supports more non-continuos rates */ #define SNDRV_PCM_RATE_8000_44100 (SNDRV_PCM_RATE_8000|SNDRV_PCM_RATE_11025|\ SNDRV_PCM_RATE_16000|SNDRV_PCM_RATE_22050|\ - SNDRV_PCM_RATE_32000|SNDRV_PCM_RATE_44100) + SNDRV_PCM_RATE_24000|SNDRV_PCM_RATE_32000|\ + SNDRV_PCM_RATE_44100) #define SNDRV_PCM_RATE_8000_48000 (SNDRV_PCM_RATE_8000_44100|SNDRV_PCM_RATE_48000) #define SNDRV_PCM_RATE_8000_96000 (SNDRV_PCM_RATE_8000_48000|SNDRV_PCM_RATE_64000|\ SNDRV_PCM_RATE_88200|SNDRV_PCM_RATE_96000) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 4adaee62ef33..72e3eab09518 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -2442,12 +2442,12 @@ static int snd_pcm_hw_rule_sample_bits(struct snd_pcm_hw_params *params, return snd_interval_refine(hw_param_interval(params, rule->var), &t); } -#if SNDRV_PCM_RATE_5512 != 1 << 0 || SNDRV_PCM_RATE_192000 != 1 << 12 +#if SNDRV_PCM_RATE_5512 != 1 << 0 || SNDRV_PCM_RATE_192000 != 1 << 13 #error "Change this table" #endif static const unsigned int rates[] = { - 5512, 8000, 11025, 16000, 22050, 32000, 44100, + 5512, 8000, 11025, 16000, 22050, 24000, 32000, 44100, 48000, 64000, 88200, 96000, 176400, 192000, 352800, 384000 }; -- 2.36.1

2 1

[PATCH v3 0/2] [PATCH v3 0/2] Add pinctrl support adsp bypass platforms
by Srinivasa Rao Mandadapu 11 Jun '22

11 Jun '22

This patch set is to make clock voting optinal for adsp bypass sc7280 platforms. Changes Since V2: -- Remove redundant lpi pincontrol variant data structure and compatible entry. -- Add adsp bypass mode boolean param check. -- Remove compatible name in dt bindings. -- Update dt binding bypass mode boolean param. Changes Since V1: -- Update commit message. Srinivasa Rao Mandadapu (2): dt-bindings: pinctrl: qcom: sc7280: Add boolean param for ADSP bypass platforms pinctrl: qcom: sc7280: Add clock optional check for ADSP bypass targets .../bindings/pinctrl/qcom,sc7280-lpass-lpi-pinctrl.yaml | 9 ++++++++- drivers/pinctrl/qcom/pinctrl-lpass-lpi.c | 3 +++ drivers/pinctrl/qcom/pinctrl-sc7280-lpass-lpi.c | 2 +- 3 files changed, 12 insertions(+), 2 deletions(-) -- 2.7.4

2 5

[PATCH] ASoC: samsung: i2s: Check before clk_unregister() not needed
by Yihao Han 10 Jun '22

10 Jun '22

clk_unregister() already checks the clk ptr using !clk || WARN_ON_ONCE(IS_ERR(clk)) so there is no need to check it again before calling it. Signed-off-by: Yihao Han <hanyihao(a)vivo.com> --- sound/soc/samsung/i2s.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 70c827162be4..84e21f7fc179 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -1247,8 +1247,7 @@ static void i2s_unregister_clocks(struct samsung_i2s_priv *priv) int i; for (i = 0; i < priv->clk_data.clk_num; i++) { - if (!IS_ERR(priv->clk_table[i])) - clk_unregister(priv->clk_table[i]); + clk_unregister(priv->clk_table[i]); } } -- 2.17.1

3 3

[PATCH] ASoC: qdsp6: q6apm-dai: unprepare stream if its already prepared
by Srinivas Kandagatla 10 Jun '22

10 Jun '22

prepare callback can be called multiple times, so unprepare the stream if its already prepared. Without this DSP is not happy to setting the params on a already prepared graph. Fixes: 9b4fe0f1cd79 ("ASoC: qdsp6: audioreach: add q6apm-dai support") Reported-by: Srinivasa Rao Mandadapu <quic_srivasam(a)quicinc.com> Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- sound/soc/qcom/qdsp6/q6apm-dai.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/soc/qcom/qdsp6/q6apm-dai.c b/sound/soc/qcom/qdsp6/q6apm-dai.c index 19c4a90ec1ea..ee59ef36b85a 100644 --- a/sound/soc/qcom/qdsp6/q6apm-dai.c +++ b/sound/soc/qcom/qdsp6/q6apm-dai.c @@ -147,6 +147,12 @@ static int q6apm_dai_prepare(struct snd_soc_component *component, cfg.num_channels = runtime->channels; cfg.bit_width = prtd->bits_per_sample; + if (prtd->state) { + /* clear the previous setup if any */ + q6apm_graph_stop(prtd->graph); + q6apm_unmap_memory_regions(prtd->graph, substream->stream); + } + prtd->pcm_count = snd_pcm_lib_period_bytes(substream); prtd->pos = 0; /* rate and channels are sent to audio driver */ -- 2.25.1

2 1

[PATCH] ASoC: Remove unused hw_write_t type
by Amadeusz Sławiński 10 Jun '22

10 Jun '22

Commit 81da8a0b7975 ("ASoC: remove codec hw_write/control_data") removed use of hw_write_t in struct snd_soc_codec, but it left type definition. Fully clean it up. Fixes: 81da8a0b7975 ("ASoC: remove codec hw_write/control_data") Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> Reviewed-by: Cezary Rojewski <cezary.rojewski(a)intel.com> --- include/sound/soc.h | 2 -- 1 file changed, 2 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index f20f5f890794..b276dcb5d4e8 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -408,8 +408,6 @@ struct snd_soc_jack_pin; struct snd_soc_jack_gpio; -typedef int (*hw_write_t)(void *,const char* ,int); - enum snd_soc_pcm_subclass { SND_SOC_PCM_CLASS_PCM = 0, SND_SOC_PCM_CLASS_BE = 1, -- 2.25.1

2 1

[PATCH] ASoC: Intel: avs: Fix parsing UUIDs in topology
by Amadeusz Sławiński 10 Jun '22

10 Jun '22

Use correct type for parsing UUIDs, this eliminates warning present, when compiling with W=1. Fixes: 34ae2cd53673 ("ASoC: Intel: avs: Add topology parsing infrastructure") Reported-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> Reviewed-by: Cezary Rojewski <cezary.rojewski(a)intel.com> --- sound/soc/intel/avs/topology.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/avs/topology.c b/sound/soc/intel/avs/topology.c index 0d11cc8aab0b..6a06fe387d13 100644 --- a/sound/soc/intel/avs/topology.c +++ b/sound/soc/intel/avs/topology.c @@ -128,10 +128,10 @@ struct avs_tplg_token_parser { static int avs_parse_uuid_token(struct snd_soc_component *comp, void *elem, void *object, u32 offset) { - struct snd_soc_tplg_vendor_value_elem *tuple = elem; + struct snd_soc_tplg_vendor_uuid_elem *tuple = elem; guid_t *val = (guid_t *)((u8 *)object + offset); - guid_copy((guid_t *)val, (const guid_t *)&tuple->value); + guid_copy((guid_t *)val, (const guid_t *)&tuple->uuid); return 0; } -- 2.25.1

2 1

[PATCH 0/6] ASoC: SOF: Add support ctx_save with IPC4
by Peter Ujfalusi 10 Jun '22

10 Jun '22

Hi, The context save functionality with IPC4 is triggered by sending a message to the firmware about the pending power down of the primary core by the host. In order to have this functionality implemented in a clean way we need to introduce a new IPC level PM ops for core state management and use that instead of open coding IPC messages here and there. The first patch updates the ctx store/ctx_restore documentation to clarify that they are optional. Regards, Peter --- Peter Ujfalusi (6): ASoC: SOF: make ctx_store and ctx_restore as optional ASoC: SOF: sof_ipc_pm_ops: Add support for DSP core power management ASoC: SOF: ipc3: Add set_core_state pm_ops implementation ASoC: SOF: ipc4: Add set_core_state pm_ops implementation ASoC: SOF: Intel: Switch to use the generic pm_ops.set_core_state ASoC: SOF: ipc4: implement pm ctx_save callback include/sound/sof/ipc4/header.h | 8 +++++++ sound/soc/sof/intel/hda-dsp.c | 15 +++++------- sound/soc/sof/intel/tgl.c | 30 ++++++++---------------- sound/soc/sof/ipc3.c | 18 +++++++++++++++ sound/soc/sof/ipc4.c | 41 +++++++++++++++++++++++++++++++++ sound/soc/sof/sof-priv.h | 6 +++-- 6 files changed, 87 insertions(+), 31 deletions(-) -- 2.36.1

2 7

[PATCH 0/3] ASoC: SOF: ipc3-dtrace: Handle race during initialization
by Peter Ujfalusi 10 Jun '22

10 Jun '22

Hi, This series handles the race which can result missing the first position update after the trace is enabled. In short: the firmware might send the position update (if we have enough trace data generated) after the dma-trace is enabled by the TRACE_DMA_PARAMS_EXT message. Depending on scheduling, load, preemption on Linux side we have seen that occasionally this first position update got missed and we missed reading it out. A new state and more strict handling of host_offset can overcome this issue, making the dtrace more reliable. Regards, Peter --- Peter Ujfalusi (3): ASoC: SOF: ipc3-dtrace: Introduce SOF_DTRACE_INITIALIZING state ASoC: SOF: ipc3-dtrace: Add helper function to update the sdev->host_offset ASoC: SOF: ipc3-dtrace: Return from dtrace_read if there is no new data available sound/soc/sof/ipc3-dtrace.c | 47 ++++++++++++++++++++++++++++++------- 1 file changed, 38 insertions(+), 9 deletions(-) -- 2.36.1

2 4

[PATCH 0/2] ASoC: codecs: qualcomm move gain to S8_TLV
by Srinivas Kandagatla 10 Jun '22

10 Jun '22

Move all the digital and IIR gains form using SX_TLV to S8_TLV, these gains are actually 8 bit gains with 7th signed bit and ranges from -84dB to +40dB Tested on DB410c with Headset playback Thanks, Srini Srinivas Kandagatla (2): ASoC: codecs: msm8916-wcd-digital: move gains from SX_TLV to S8_TLV ASoC: codecs: wcd9335: move gains from SX_TLV to S8_TLV sound/soc/codecs/msm8916-wcd-digital.c | 46 +++++++-------- sound/soc/codecs/wcd9335.c | 81 ++++++++++++-------------- 2 files changed, 59 insertions(+), 68 deletions(-) -- 2.25.1

2 3

[PATCH] ASoC: SOF: debug: Clarify the IPC timeout handling path
by Peter Ujfalusi 10 Jun '22

10 Jun '22

The dmesg log message of "Firmware exception" causes lots of confusion as the snd_sof_handle_fw_exception() is only called in case of an IPC tx timeout, where such a message does not make much sense. To not limit the snd_sof_handle_fw_exception() handler to just one error case, add a parameter to allow the caller to specify a meaningful message to be printed. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Yaochun Hung <yc.hung(a)mediatek.com> --- sound/soc/sof/debug.c | 5 ++--- sound/soc/sof/ipc3.c | 2 +- sound/soc/sof/sof-priv.h | 2 +- 3 files changed, 4 insertions(+), 5 deletions(-) diff --git a/sound/soc/sof/debug.c b/sound/soc/sof/debug.c index cf1271eb29b2..c5d797e97c02 100644 --- a/sound/soc/sof/debug.c +++ b/sound/soc/sof/debug.c @@ -428,7 +428,7 @@ static void snd_sof_ipc_dump(struct snd_sof_dev *sdev) } } -void snd_sof_handle_fw_exception(struct snd_sof_dev *sdev) +void snd_sof_handle_fw_exception(struct snd_sof_dev *sdev, const char *msg) { if (IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG_RETAIN_DSP_CONTEXT) || sof_debug_check_flag(SOF_DBG_RETAIN_CTX)) { @@ -441,8 +441,7 @@ void snd_sof_handle_fw_exception(struct snd_sof_dev *sdev) /* dump vital information to the logs */ snd_sof_ipc_dump(sdev); - snd_sof_dsp_dbg_dump(sdev, "Firmware exception", - SOF_DBG_DUMP_REGS | SOF_DBG_DUMP_MBOX); + snd_sof_dsp_dbg_dump(sdev, msg, SOF_DBG_DUMP_REGS | SOF_DBG_DUMP_MBOX); sof_fw_trace_fw_crashed(sdev); } EXPORT_SYMBOL(snd_sof_handle_fw_exception); diff --git a/sound/soc/sof/ipc3.c b/sound/soc/sof/ipc3.c index dff5feaad370..ef8019e009b7 100644 --- a/sound/soc/sof/ipc3.c +++ b/sound/soc/sof/ipc3.c @@ -290,7 +290,7 @@ static int ipc3_wait_tx_done(struct snd_sof_ipc *ipc, void *reply_data) dev_err(sdev->dev, "ipc tx timed out for %#x (msg/reply size: %d/%zu)\n", hdr->cmd, hdr->size, msg->reply_size); - snd_sof_handle_fw_exception(ipc->sdev); + snd_sof_handle_fw_exception(ipc->sdev, "IPC timeout"); ret = -ETIMEDOUT; } else { ret = msg->reply_error; diff --git a/sound/soc/sof/sof-priv.h b/sound/soc/sof/sof-priv.h index 9d7f53ff9c70..32c152528f1d 100644 --- a/sound/soc/sof/sof-priv.h +++ b/sound/soc/sof/sof-priv.h @@ -655,7 +655,7 @@ void sof_print_oops_and_stack(struct snd_sof_dev *sdev, const char *level, u32 panic_code, u32 tracep_code, void *oops, struct sof_ipc_panic_info *panic_info, void *stack, size_t stack_words); -void snd_sof_handle_fw_exception(struct snd_sof_dev *sdev); +void snd_sof_handle_fw_exception(struct snd_sof_dev *sdev, const char *msg); int snd_sof_dbg_memory_info_init(struct snd_sof_dev *sdev); int snd_sof_debugfs_add_region_item_iomem(struct snd_sof_dev *sdev, enum snd_sof_fw_blk_type blk_type, u32 offset, size_t size, -- 2.36.1

2 1

[PATCH] ASoC: SOF: ipc3-dtrace: use pm_runtime_resume_and_get()
by Peter Ujfalusi 10 Jun '22

10 Jun '22

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Use pm_runtime_resume_and_get() to replace the pm_runtime_get_sync() and pm_runtime_put_noidle() pattern. No functional changes. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Péter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> --- sound/soc/sof/ipc3-dtrace.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/sof/ipc3-dtrace.c b/sound/soc/sof/ipc3-dtrace.c index b4e1343f9138..45bf9c5dc412 100644 --- a/sound/soc/sof/ipc3-dtrace.c +++ b/sound/soc/sof/ipc3-dtrace.c @@ -157,9 +157,8 @@ static int ipc3_trace_update_filter(struct snd_sof_dev *sdev, int num_elems, msg->elem_cnt = num_elems; memcpy(&msg->elems[0], elems, num_elems * sizeof(*elems)); - ret = pm_runtime_get_sync(sdev->dev); + ret = pm_runtime_resume_and_get(sdev->dev); if (ret < 0 && ret != -EACCES) { - pm_runtime_put_noidle(sdev->dev); dev_err(sdev->dev, "enabling device failed: %d\n", ret); goto error; } -- 2.36.1

2 1

[RFC PATCH v2 0/5] Apple Macs machine/platform ASoC driver
by Martin Povišer 10 Jun '22

10 Jun '22

Hi all, This is again RFC with a machine-level ASoC driver for recent Apple Macs with the M1 line of chips. This time I attached the platform driver too for good measure. What I am interested in the most is checking the overall approach, especially on two points (both in some ways already discussed in previous RFC [0]): - The way the platform/machine driver handles the fact that multiple I2S ports (now backend DAIs) can be driven by/connected to the same SERDES unit (now in effect a frontend DAI). After previous discussion I have transitioned to DPCM to model this. I took the opportunity of dynamic backend/frontend routing to support speakers/headphones runtime switching. More on this in comments at top of the machine and platform driver. - The way the machine driver deactivates some of the controls where suitable, and limits volume on others. I added a new ASoC card method to that end. Kind regards, Martin [0] https://lore.kernel.org/linux-devicetree/20220331000449.41062-1-povik+lin@c… Martin Povišer (5): dt-bindings: sound: Add Apple MCA I2S transceiver dt-bindings: sound: Add Apple Macs sound peripherals ASoC: apple: Add MCA platform driver for Apple SoCs ASoC: Introduce 'fixup_controls' card method ASoC: apple: Add macaudio machine driver .../bindings/sound/apple,macaudio.yaml | 157 +++ .../devicetree/bindings/sound/apple,mca.yaml | 102 ++ include/sound/soc-card.h | 1 + include/sound/soc.h | 1 + sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/apple/Kconfig | 25 + sound/soc/apple/Makefile | 5 + sound/soc/apple/macaudio.c | 1004 +++++++++++++++ sound/soc/apple/mca.c | 1122 +++++++++++++++++ sound/soc/soc-card.c | 6 + sound/soc/soc-core.c | 1 + 12 files changed, 2426 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/apple,macaudio.yaml create mode 100644 Documentation/devicetree/bindings/sound/apple,mca.yaml create mode 100644 sound/soc/apple/Kconfig create mode 100644 sound/soc/apple/Makefile create mode 100644 sound/soc/apple/macaudio.c create mode 100644 sound/soc/apple/mca.c -- 2.33.0

4 24

[PATCH] ASoC: SOF: ipc3-topology: Move and correct size checks in sof_ipc3_control_load_bytes()
by Peter Ujfalusi 10 Jun '22

10 Jun '22

Move the size checks prior to allocating memory as these checks do not need the data to be allocated and in case of an error we would not need to free the allocation. The max size must not be less than the size of struct sof_ipc_ctrl_data + struct sof_abi_hdr as the ABI header needs to be present under all circumstances. The check was incorrectly used or between the two size checks. Fixes: b5cee8feb1d4 ("ASoC: SOF: topology: Make control parsing IPC agnostic") Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> --- sound/soc/sof/ipc3-topology.c | 23 +++++++++++------------ 1 file changed, 11 insertions(+), 12 deletions(-) diff --git a/sound/soc/sof/ipc3-topology.c b/sound/soc/sof/ipc3-topology.c index 043554d7cb4a..10740c55294d 100644 --- a/sound/soc/sof/ipc3-topology.c +++ b/sound/soc/sof/ipc3-topology.c @@ -1577,24 +1577,23 @@ static int sof_ipc3_control_load_bytes(struct snd_sof_dev *sdev, struct snd_sof_ struct sof_ipc_ctrl_data *cdata; int ret; - scontrol->ipc_control_data = kzalloc(scontrol->max_size, GFP_KERNEL); - if (!scontrol->ipc_control_data) - return -ENOMEM; - - if (scontrol->max_size < sizeof(*cdata) || - scontrol->max_size < sizeof(struct sof_abi_hdr)) { - ret = -EINVAL; - goto err; + if (scontrol->max_size < (sizeof(*cdata) + sizeof(struct sof_abi_hdr))) { + dev_err(sdev->dev, "%s: insufficient size for a bytes control: %zu.\n", + __func__, scontrol->max_size); + return -EINVAL; } - /* init the get/put bytes data */ if (scontrol->priv_size > scontrol->max_size - sizeof(*cdata)) { - dev_err(sdev->dev, "err: bytes data size %zu exceeds max %zu.\n", + dev_err(sdev->dev, + "%s: bytes data size %zu exceeds max %zu.\n", __func__, scontrol->priv_size, scontrol->max_size - sizeof(*cdata)); - ret = -EINVAL; - goto err; + return -EINVAL; } + scontrol->ipc_control_data = kzalloc(scontrol->max_size, GFP_KERNEL); + if (!scontrol->ipc_control_data) + return -ENOMEM; + scontrol->size = sizeof(struct sof_ipc_ctrl_data) + scontrol->priv_size; cdata = scontrol->ipc_control_data; -- 2.36.1

2 1

[PATCH 1/3] ASoC: wm8940: Remove warning when no plat data
by Lukasz Majewski 10 Jun '22

10 Jun '22

The lack of platform data in the contemporary Linux shall not be the reason to display warnings to the kernel logs. Signed-off-by: Lukasz Majewski <lukma(a)denx.de> --- sound/soc/codecs/wm8940.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c index 440d048ef0c0..7cea54720436 100644 --- a/sound/soc/codecs/wm8940.c +++ b/sound/soc/codecs/wm8940.c @@ -709,9 +709,7 @@ static int wm8940_probe(struct snd_soc_component *component) if (ret < 0) return ret; - if (!pdata) - dev_warn(component->dev, "No platform data supplied\n"); - else { + if (pdata) { reg = snd_soc_component_read(component, WM8940_OUTPUTCTL); ret = snd_soc_component_write(component, WM8940_OUTPUTCTL, reg | pdata->vroi); if (ret < 0) -- 2.20.1

2 13

snd-usb-audio: probe of 1-1:1.0 failed with error -16
by Angelo Capone 10 Jun '22

10 Jun '22

Hi, my linux is not able to recognize usb headset, below you canfind dmesg output, while attached the alsa-info output. Can you help me? Thanks in advance. [10996.537749] usb 1-1: new full-speed USB device number 11 using xhci_hcd [10996.698320] usb 1-1: New USB device found, idVendor=0b0e, idProduct=0300, bcdDevice= 3.08 [10996.698337] usb 1-1: New USB device strings: Mfr=1, Product=2, SerialNumber=3 [10996.698346] usb 1-1: Product: Jabra EVOLVE 20 MS [10996.698354] usb 1-1: Manufacturer: GN Netcom A/S [10996.698362] usb 1-1: SerialNumber: 000D06866CEE0A [10996.699759] snd-usb-audio 1-1:1.0: cannot find the slot for index 0 (range 0-0), error: -16 [10996.699788] usb 1-1: cannot create card instance 0 [10996.699813] snd-usb-audio: probe of 1-1:1.0 failed with error -16 [10996.706056] input: GN Netcom A/S Jabra EVOLVE 20 MS as /devices/pci0000:00/0000:00:14.0/usb1/1-1/1-1:1.3/0003:0B0E:0300.0009/input/input38 [10996.766432] jabra 0003:0B0E:0300.0009: input,hiddev0,hidraw3: USB HID v1.00 Device [GN Netcom A/S Jabra EVOLVE 20 MS] on usb-0000:00:14.0-1/input3 -- Angelo

1 0

[PATCH v4 0/2] Add software clock gating requirement check
by Srinivasa Rao Mandadapu 10 Jun '22

10 Jun '22

This patch set is to add software clock gating requirement check Changes Since V3: -- Remove redundant check before reset control call -- Reorganiaze patches. -- Fix typos. Changes Since V2: -- Fix if check before reset control call Changes Since V1: -- Use boolean flag for bool variable initialization instead of hard coding. Srinivasa Rao Mandadapu (2): soundwire: qcom: Add flag for software clock gating check ASoC: qcom: soundwire: Enable software clock gating requirement flag drivers/soundwire/qcom.c | 22 +++++++++++++++++----- 1 file changed, 17 insertions(+), 5 deletions(-) -- 2.7.4

2 4

[PATCH v2 00/14] ASoC: Intel: avs: Topology and path management
by Cezary Rojewski 10 Jun '22

10 Jun '22

A continuation of avs-driver initial series [1]. This chapter covers path management and topology parsing part which was ealier path of the main series. The two patches that represented these two subjects in the initial series, have been split into many to allow for easier review and discussion. AVS topology is split into two major parts: dictionaries - found within ASoC topology manifest - and path templates. Dictionaries job is to reduce the total amount of memory occupied by topology elements. Rather than having every pipeline and module carry its own information, each refers to specific entry in specific dictionary by provided (from topology file) indexes. In consequence, most struct avs_tplg_xxx are made out of pointers. Path templates are similar to path descriptions found in skylake-driver and they describe how given path shall look like in runtime - number of modules and pipelines that shape it and how they are laid out. A single path template is tied either to FE or BE and thus at most to a single, user-visible endpoint when speaking of FE. Path is a software representation of its ADSP firmware equivalent. It's a logical container for pipelines which are themselves containers - this time for modules i.e. processing units. Depending on the number of audio formats supported, each path template may carry one or more descriptions of given path. During runtime, when audio format is known, description matching said format is selected and used when instantiating path on ADSP firmware side through IPCs. Changes v1 -> v2: - fixed 'set but not used' warning in patch 10/14 as mentioned by ikp builder Changes RFC -> v1: - Plenty of rewordings and spelling fixes as requested by Pierre - any sysfs-functionality-related left in RFC dropped, will be part of separate subject in the future - separated parsing of formatted strings into: "Support link_mask formatted string" patch, increasing series size to 14 patches [1]: https://lore.kernel.org/alsa-devel/20220207122108.3780926-1-cezary.rojewski… Cezary Rojewski (14): ASoC: Intel: avs: Declare vendor tokens ASoC: Intel: avs: Add topology parsing infrastructure ASoC: Intel: avs: Parse module-extension tuples ASoC: Intel: avs: Parse pplcfg and binding tuples ASoC: Intel: avs: Parse pipeline and module tuples ASoC: Intel: avs: Parse path and path templates tuples ASoC: Intel: avs: Add topology loading operations ASoC: Intel: avs: Support link_mask formatted string ASoC: Intel: avs: Declare path and its components ASoC: Intel: avs: Path creation and freeing ASoC: Intel: avs: Path state management ASoC: Intel: avs: Arm paths after creating them ASoC: Intel: avs: Prepare modules before bindings them ASoC: Intel: avs: Configure modules according to their type include/uapi/sound/intel/avs/tokens.h | 126 ++ sound/soc/intel/Kconfig | 2 + sound/soc/intel/avs/Makefile | 3 +- sound/soc/intel/avs/avs.h | 23 + sound/soc/intel/avs/path.c | 1005 ++++++++++++++++ sound/soc/intel/avs/path.h | 72 ++ sound/soc/intel/avs/topology.c | 1600 +++++++++++++++++++++++++ sound/soc/intel/avs/topology.h | 194 +++ 8 files changed, 3024 insertions(+), 1 deletion(-) create mode 100644 include/uapi/sound/intel/avs/tokens.h create mode 100644 sound/soc/intel/avs/path.c create mode 100644 sound/soc/intel/avs/path.h create mode 100644 sound/soc/intel/avs/topology.c create mode 100644 sound/soc/intel/avs/topology.h base-commit: bc55cfd5718c7c23e5524582e9fa70b4d10f2433 -- 2.25.1

3 19

[PATCH] ALSA: hda/realtek: Add mute LED quirk for HP Omen laptop
by contact＠sohamsen.me 10 Jun '22

10 Jun '22

From: Soham Sen <contact(a)sohamsen.me> The HP Omen 15 laptop needs a quirk to toggle the mute LED. It already is implemented for a different variant of the HP Omen laptop so a fixup entry is needed for this variant. Signed-off-by: Soham Sen <contact(a)sohamsen.me> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index f3ad454b3..495b919bf 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9010,6 +9010,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { ALC285_FIXUP_HP_GPIO_AMP_INIT), SND_PCI_QUIRK(0x103c, 0x8783, "HP ZBook Fury 15 G7 Mobile Workstation", ALC285_FIXUP_HP_GPIO_AMP_INIT), + SND_PCI_QUIRK(0x103c, 0x8787, "HP OMEN 15", ALC285_FIXUP_HP_MUTE_LED), SND_PCI_QUIRK(0x103c, 0x8788, "HP OMEN 15", ALC285_FIXUP_HP_MUTE_LED), SND_PCI_QUIRK(0x103c, 0x87c8, "HP", ALC287_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x87e5, "HP ProBook 440 G8 Notebook PC", ALC236_FIXUP_HP_GPIO_LED), -- 2.36.1

2 1

[PATCH v3] ALSA: control: Use xarray for faster lookups
by Takashi Iwai 10 Jun '22

10 Jun '22

The control elements are managed in a single linked list and we traverse the whole list for matching each numid or ctl id per every inquiry of a control element. This is OK-ish for a small number of elements but obviously it doesn't scale. Especially the matching with the ctl id takes time because it checks each field of the snd_ctl_id element, e.g. the name string is matched with strcmp(). This patch adds the hash tables with Xarray for improving the lookup speed of a control element. There are two xarray tables added to the card; one for numid and another for ctl id. For the numid, we use the numid as the index, while for the ctl id, we calculate a hash key. The lookup is done via a single xa_load() execution. As long as the given control element is found on the Xarray table, that's fine, we can give back a quick lookup result. The problem is when no entry hits on the table, and for this case, we have a slight optimization. Namely, the driver checks whether we had a collision on Xarray table, and do a fallback search (linear lookup of the full entries) only if a hash key collision happened beforehand. So, in theory, the inquiry for a non-existing element might take still time even with this patch in a worst case, but this must be pretty rare. The feature is enabled via CONFIG_SND_CTL_FAST_LOOKUP, which is turned on as default. For simplicity, the option can be turned off only when CONFIG_EXPERT is set ("You are expert? Then you manage 1000 knobs"). Link: https://lore.kernel.org/r/20211028130027.18764-1-tiwai@suse.de Link: https://lore.kernel.org/r/20220609180504.775-1-tiwai@suse.de Link: https://lore.kernel.org/all/cover.1653813866.git.quic_rbankapu@quicinc.com/ Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- v1->v2: optimize with collision detection, cleanups v2->v3: Fix missing dependency on CONFIG_XARRAY_MULTI include/sound/core.h | 6 ++ sound/core/Kconfig | 10 +++ sound/core/control.c | 180 +++++++++++++++++++++++++++++++++++-------- sound/core/init.c | 4 + 4 files changed, 168 insertions(+), 32 deletions(-) diff --git a/include/sound/core.h b/include/sound/core.h index 6d4cc49584c6..dd28de2343b8 100644 --- a/include/sound/core.h +++ b/include/sound/core.h @@ -14,6 +14,7 @@ #include <linux/pm.h> /* pm_message_t */ #include <linux/stringify.h> #include <linux/printk.h> +#include <linux/xarray.h> /* number of supported soundcards */ #ifdef CONFIG_SND_DYNAMIC_MINORS @@ -103,6 +104,11 @@ struct snd_card { size_t user_ctl_alloc_size; // current memory allocation by user controls. struct list_head controls; /* all controls for this card */ struct list_head ctl_files; /* active control files */ +#ifdef CONFIG_SND_CTL_FAST_LOOKUP + struct xarray ctl_numids; /* hash table for numids */ + struct xarray ctl_hash; /* hash table for ctl id matching */ + bool ctl_hash_collision; /* ctl_hash collision seen? */ +#endif struct snd_info_entry *proc_root; /* root for soundcard specific files */ struct proc_dir_entry *proc_root_link; /* number link to real id */ diff --git a/sound/core/Kconfig b/sound/core/Kconfig index dd7b40734723..25b2434e4556 100644 --- a/sound/core/Kconfig +++ b/sound/core/Kconfig @@ -154,6 +154,16 @@ config SND_VERBOSE_PRINTK You don't need this unless you're debugging ALSA. +config SND_CTL_FAST_LOOKUP + bool "Fast lookup of control elements" if EXPERT + default y + select XARRAY_MULTI + help + This option enables the faster lookup of control elements. + It will consume more memory because of the additional Xarray. + If you want to choose the memory footprint over the performance + inevitably, turn this off. + config SND_DEBUG bool "Debug" help diff --git a/sound/core/control.c b/sound/core/control.c index a25c0d64d104..6a8fd9933f06 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -364,6 +364,93 @@ static int snd_ctl_find_hole(struct snd_card *card, unsigned int count) return 0; } +/* check whether the given id is contained in the given kctl */ +static bool elem_id_matches(const struct snd_kcontrol *kctl, + const struct snd_ctl_elem_id *id) +{ + return kctl->id.iface == id->iface && + kctl->id.device == id->device && + kctl->id.subdevice == id->subdevice && + !strncmp(kctl->id.name, id->name, sizeof(kctl->id.name)) && + kctl->id.index <= id->index && + kctl->id.index + kctl->count > id->index; +} + +#ifdef CONFIG_SND_CTL_FAST_LOOKUP +/* Compute a hash key for the corresponding ctl id + * It's for the name lookup, hence the numid is excluded. + * The hash key is bound in LONG_MAX to be used for Xarray key. + */ +#define MULTIPLIER 37 +static unsigned long get_ctl_id_hash(const struct snd_ctl_elem_id *id) +{ + unsigned long h; + const unsigned char *p; + + h = id->iface; + h = MULTIPLIER * h + id->device; + h = MULTIPLIER * h + id->subdevice; + for (p = id->name; *p; p++) + h = MULTIPLIER * h + *p; + h = MULTIPLIER * h + id->index; + h &= LONG_MAX; + return h; +} + +/* add hash entries to numid and ctl xarray tables */ +static void add_hash_entries(struct snd_card *card, + struct snd_kcontrol *kcontrol) +{ + struct snd_ctl_elem_id id = kcontrol->id; + int i; + + xa_store_range(&card->ctl_numids, kcontrol->id.numid, + kcontrol->id.numid + kcontrol->count - 1, + kcontrol, GFP_KERNEL); + + for (i = 0; i < kcontrol->count; i++) { + id.index = kcontrol->id.index + i; + if (xa_insert(&card->ctl_hash, get_ctl_id_hash(&id), + kcontrol, GFP_KERNEL)) { + /* skip hash for this entry, noting we had collision */ + card->ctl_hash_collision = true; + dev_dbg(card->dev, "ctl_hash collision %d:%s:%d\n", + id.iface, id.name, id.index); + } + } +} + +/* remove hash entries that have been added */ +static void remove_hash_entries(struct snd_card *card, + struct snd_kcontrol *kcontrol) +{ + struct snd_ctl_elem_id id = kcontrol->id; + struct snd_kcontrol *matched; + unsigned long h; + int i; + + for (i = 0; i < kcontrol->count; i++) { + xa_erase(&card->ctl_numids, id.numid); + h = get_ctl_id_hash(&id); + matched = xa_load(&card->ctl_hash, h); + if (matched && (matched == kcontrol || + elem_id_matches(matched, &id))) + xa_erase(&card->ctl_hash, h); + id.index++; + id.numid++; + } +} +#else /* CONFIG_SND_CTL_FAST_LOOKUP */ +static inline void add_hash_entries(struct snd_card *card, + struct snd_kcontrol *kcontrol) +{ +} +static inline void remove_hash_entries(struct snd_card *card, + struct snd_kcontrol *kcontrol) +{ +} +#endif /* CONFIG_SND_CTL_FAST_LOOKUP */ + enum snd_ctl_add_mode { CTL_ADD_EXCLUSIVE, CTL_REPLACE, CTL_ADD_ON_REPLACE, }; @@ -408,6 +495,8 @@ static int __snd_ctl_add_replace(struct snd_card *card, kcontrol->id.numid = card->last_numid + 1; card->last_numid += kcontrol->count; + add_hash_entries(card, kcontrol); + for (idx = 0; idx < kcontrol->count; idx++) snd_ctl_notify_one(card, SNDRV_CTL_EVENT_MASK_ADD, kcontrol, idx); @@ -479,6 +568,26 @@ int snd_ctl_replace(struct snd_card *card, struct snd_kcontrol *kcontrol, } EXPORT_SYMBOL(snd_ctl_replace); +static int __snd_ctl_remove(struct snd_card *card, + struct snd_kcontrol *kcontrol, + bool remove_hash) +{ + unsigned int idx; + + if (snd_BUG_ON(!card || !kcontrol)) + return -EINVAL; + list_del(&kcontrol->list); + + if (remove_hash) + remove_hash_entries(card, kcontrol); + + card->controls_count -= kcontrol->count; + for (idx = 0; idx < kcontrol->count; idx++) + snd_ctl_notify_one(card, SNDRV_CTL_EVENT_MASK_REMOVE, kcontrol, idx); + snd_ctl_free_one(kcontrol); + return 0; +} + /** * snd_ctl_remove - remove the control from the card and release it * @card: the card instance @@ -492,16 +601,7 @@ EXPORT_SYMBOL(snd_ctl_replace); */ int snd_ctl_remove(struct snd_card *card, struct snd_kcontrol *kcontrol) { - unsigned int idx; - - if (snd_BUG_ON(!card || !kcontrol)) - return -EINVAL; - list_del(&kcontrol->list); - card->controls_count -= kcontrol->count; - for (idx = 0; idx < kcontrol->count; idx++) - snd_ctl_notify_one(card, SNDRV_CTL_EVENT_MASK_REMOVE, kcontrol, idx); - snd_ctl_free_one(kcontrol); - return 0; + return __snd_ctl_remove(card, kcontrol, true); } EXPORT_SYMBOL(snd_ctl_remove); @@ -642,14 +742,30 @@ int snd_ctl_rename_id(struct snd_card *card, struct snd_ctl_elem_id *src_id, up_write(&card->controls_rwsem); return -ENOENT; } + remove_hash_entries(card, kctl); kctl->id = *dst_id; kctl->id.numid = card->last_numid + 1; card->last_numid += kctl->count; + add_hash_entries(card, kctl); up_write(&card->controls_rwsem); return 0; } EXPORT_SYMBOL(snd_ctl_rename_id); +#ifndef CONFIG_SND_CTL_FAST_LOOKUP +static struct snd_kcontrol * +snd_ctl_find_numid_slow(struct snd_card *card, unsigned int numid) +{ + struct snd_kcontrol *kctl; + + list_for_each_entry(kctl, &card->controls, list) { + if (kctl->id.numid <= numid && kctl->id.numid + kctl->count > numid) + return kctl; + } + return NULL; +} +#endif /* !CONFIG_SND_CTL_FAST_LOOKUP */ + /** * snd_ctl_find_numid - find the control instance with the given number-id * @card: the card instance @@ -665,15 +781,13 @@ EXPORT_SYMBOL(snd_ctl_rename_id); */ struct snd_kcontrol *snd_ctl_find_numid(struct snd_card *card, unsigned int numid) { - struct snd_kcontrol *kctl; - if (snd_BUG_ON(!card || !numid)) return NULL; - list_for_each_entry(kctl, &card->controls, list) { - if (kctl->id.numid <= numid && kctl->id.numid + kctl->count > numid) - return kctl; - } - return NULL; +#ifdef CONFIG_SND_CTL_FAST_LOOKUP + return xa_load(&card->ctl_numids, numid); +#else + return snd_ctl_find_numid_slow(card, numid); +#endif } EXPORT_SYMBOL(snd_ctl_find_numid); @@ -699,21 +813,18 @@ struct snd_kcontrol *snd_ctl_find_id(struct snd_card *card, return NULL; if (id->numid != 0) return snd_ctl_find_numid(card, id->numid); - list_for_each_entry(kctl, &card->controls, list) { - if (kctl->id.iface != id->iface) - continue; - if (kctl->id.device != id->device) - continue; - if (kctl->id.subdevice != id->subdevice) - continue; - if (strncmp(kctl->id.name, id->name, sizeof(kctl->id.name))) - continue; - if (kctl->id.index > id->index) - continue; - if (kctl->id.index + kctl->count <= id->index) - continue; +#ifdef CONFIG_SND_CTL_FAST_LOOKUP + kctl = xa_load(&card->ctl_hash, get_ctl_id_hash(id)); + if (kctl && elem_id_matches(kctl, id)) return kctl; - } + if (!card->ctl_hash_collision) + return NULL; /* we can rely on only hash table */ +#endif + /* no matching in hash table - try all as the last resort */ + list_for_each_entry(kctl, &card->controls, list) + if (elem_id_matches(kctl, id)) + return kctl; + return NULL; } EXPORT_SYMBOL(snd_ctl_find_id); @@ -2195,8 +2306,13 @@ static int snd_ctl_dev_free(struct snd_device *device) down_write(&card->controls_rwsem); while (!list_empty(&card->controls)) { control = snd_kcontrol(card->controls.next); - snd_ctl_remove(card, control); + __snd_ctl_remove(card, control, false); } + +#ifdef CONFIG_SND_CTL_FAST_LOOKUP + xa_destroy(&card->ctl_numids); + xa_destroy(&card->ctl_hash); +#endif up_write(&card->controls_rwsem); put_device(&card->ctl_dev); return 0; diff --git a/sound/core/init.c b/sound/core/init.c index 726a8353201f..1870aee7b64f 100644 --- a/sound/core/init.c +++ b/sound/core/init.c @@ -310,6 +310,10 @@ static int snd_card_init(struct snd_card *card, struct device *parent, rwlock_init(&card->ctl_files_rwlock); INIT_LIST_HEAD(&card->controls); INIT_LIST_HEAD(&card->ctl_files); +#ifdef CONFIG_SND_CTL_FAST_LOOKUP + xa_init(&card->ctl_numids); + xa_init(&card->ctl_hash); +#endif spin_lock_init(&card->files_lock); INIT_LIST_HEAD(&card->files_list); mutex_init(&card->memory_mutex); -- 2.35.3

1 0

[PATCH v1 14/16] soundwire: Use acpi_dev_for_each_child()
by Rafael J. Wysocki 09 Jun '22

09 Jun '22

From: Rafael J. Wysocki <rafael.j.wysocki(a)intel.com> Instead of walking the list of children of an ACPI device directly, use acpi_dev_for_each_child() to carry out an action for all of the given ACPI device's children. This will help to eliminate the children list head from struct acpi_device as it is redundant and it is used in questionable ways in some places (in particular, locking is needed for walking the list pointed to it safely, but it is often missing). Signed-off-by: Rafael J. Wysocki <rafael.j.wysocki(a)intel.com> --- drivers/soundwire/slave.c | 115 ++++++++++++++++++++++++++-------------------- 1 file changed, 67 insertions(+), 48 deletions(-) Index: linux-pm/drivers/soundwire/slave.c =================================================================== --- linux-pm.orig/drivers/soundwire/slave.c +++ linux-pm/drivers/soundwire/slave.c @@ -127,6 +127,71 @@ static bool find_slave(struct sdw_bus *b return true; } +struct sdw_acpi_child_walk_data { + struct sdw_bus *bus; + struct acpi_device *adev; + struct sdw_slave_id id; + bool ignore_unique_id; +}; + +static int sdw_acpi_check_duplicate(struct acpi_device *adev, void *data) +{ + struct sdw_acpi_child_walk_data *cwd = data; + struct sdw_bus *bus = cwd->bus; + struct sdw_slave_id id; + + if (adev == cwd->adev) + return 0; + + if (!find_slave(bus, adev, &id)) + return 0; + + if (cwd->id.sdw_version != id.sdw_version || cwd->id.mfg_id != id.mfg_id || + cwd->id.part_id != id.part_id || cwd->id.class_id != id.class_id) + return 0; + + if (cwd->id.unique_id != id.unique_id) { + dev_dbg(bus->dev, + "Valid unique IDs 0x%x 0x%x for Slave mfg_id 0x%04x, part_id 0x%04x\n", + cwd->id.unique_id, id.unique_id, cwd->id.mfg_id, + cwd->id.part_id); + cwd->ignore_unique_id = false; + return 0; + } + + dev_err(bus->dev, + "Invalid unique IDs 0x%x 0x%x for Slave mfg_id 0x%04x, part_id 0x%04x\n", + cwd->id.unique_id, id.unique_id, cwd->id.mfg_id, cwd->id.part_id); + return -ENODEV; +} + +static int sdw_acpi_find_one(struct acpi_device *adev, void *data) +{ + struct sdw_bus *bus = data; + struct sdw_acpi_child_walk_data cwd = { + .bus = bus, + .adev = adev, + .ignore_unique_id = true, + }; + int ret; + + if (!find_slave(bus, adev, &cwd.id)) + return 0; + + /* Brute-force O(N^2) search for duplicates. */ + ret = acpi_dev_for_each_child(ACPI_COMPANION(bus->dev), + sdw_acpi_check_duplicate, &cwd); + if (ret) + return ret; + + if (cwd.ignore_unique_id) + cwd.id.unique_id = SDW_IGNORED_UNIQUE_ID; + + /* Ignore errors and continue. */ + sdw_slave_add(bus, &cwd.id, acpi_fwnode_handle(adev)); + return 0; +} + /* * sdw_acpi_find_slaves() - Find Slave devices in Master ACPI node * @bus: SDW bus instance @@ -135,8 +200,7 @@ static bool find_slave(struct sdw_bus *b */ int sdw_acpi_find_slaves(struct sdw_bus *bus) { - struct acpi_device *adev, *parent; - struct acpi_device *adev2, *parent2; + struct acpi_device *parent; parent = ACPI_COMPANION(bus->dev); if (!parent) { @@ -144,52 +208,7 @@ int sdw_acpi_find_slaves(struct sdw_bus return -ENODEV; } - list_for_each_entry(adev, &parent->children, node) { - struct sdw_slave_id id; - struct sdw_slave_id id2; - bool ignore_unique_id = true; - - if (!find_slave(bus, adev, &id)) - continue; - - /* brute-force O(N^2) search for duplicates */ - parent2 = parent; - list_for_each_entry(adev2, &parent2->children, node) { - - if (adev == adev2) - continue; - - if (!find_slave(bus, adev2, &id2)) - continue; - - if (id.sdw_version != id2.sdw_version || - id.mfg_id != id2.mfg_id || - id.part_id != id2.part_id || - id.class_id != id2.class_id) - continue; - - if (id.unique_id != id2.unique_id) { - dev_dbg(bus->dev, - "Valid unique IDs 0x%x 0x%x for Slave mfg_id 0x%04x, part_id 0x%04x\n", - id.unique_id, id2.unique_id, id.mfg_id, id.part_id); - ignore_unique_id = false; - } else { - dev_err(bus->dev, - "Invalid unique IDs 0x%x 0x%x for Slave mfg_id 0x%04x, part_id 0x%04x\n", - id.unique_id, id2.unique_id, id.mfg_id, id.part_id); - return -ENODEV; - } - } - - if (ignore_unique_id) - id.unique_id = SDW_IGNORED_UNIQUE_ID; - - /* - * don't error check for sdw_slave_add as we want to continue - * adding Slaves - */ - sdw_slave_add(bus, &id, acpi_fwnode_handle(adev)); - } + acpi_dev_for_each_child(parent, sdw_acpi_find_one, bus); return 0; }

3 5

[PATCH v2] ALSA: control: Use xarray for faster lookups
by Takashi Iwai 09 Jun '22

09 Jun '22

The control elements are managed in a single linked list and we traverse the whole list for matching each numid or ctl id per every inquiry of a control element. This is OK-ish for a small number of elements but obviously it doesn't scale. Especially the matching with the ctl id takes time because it checks each field of the snd_ctl_id element, e.g. the name string is matched with strcmp(). This patch adds the hash tables with Xarray for improving the lookup speed of a control element. There are two xarray tables added to the card; one for numid and another for ctl id. For the numid, we use the numid as the index, while for the ctl id, we calculate a hash key. The lookup is done via a single xa_load() execution. As long as the given control element is found on the Xarray table, that's fine, we can give back a quick lookup result. The problem is when no entry hits on the table, and for this case, we have a slight optimization. Namely, the driver checks whether we had a collision on Xarray table, and do a fallback search (linear lookup of the full entries) only if a hash key collision happened beforehand. So, in theory, the inquiry for a non-existing element might take still time even with this patch in a worst case, but this must be pretty rare. The feature is enabled via CONFIG_SND_CTL_FAST_LOOKUP, which is turned on as default. For simplicity, the option can be turned off only when CONFIG_EXPERT is set ("You are expert? Then you manage 1000 knobs"). Link: https://lore.kernel.org/r/20211028130027.18764-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- So this is a revised patch for the previous RFC (see the link above), as there seems real demands for the fast lookup of control elements. A minor improvement from the previous version is the avoidance of fallback in the case of hash non-collision. include/sound/core.h | 6 ++ sound/core/Kconfig | 9 +++ sound/core/control.c | 180 +++++++++++++++++++++++++++++++++++-------- sound/core/init.c | 4 + 4 files changed, 167 insertions(+), 32 deletions(-) diff --git a/include/sound/core.h b/include/sound/core.h index 6d4cc49584c6..dd28de2343b8 100644 --- a/include/sound/core.h +++ b/include/sound/core.h @@ -14,6 +14,7 @@ #include <linux/pm.h> /* pm_message_t */ #include <linux/stringify.h> #include <linux/printk.h> +#include <linux/xarray.h> /* number of supported soundcards */ #ifdef CONFIG_SND_DYNAMIC_MINORS @@ -103,6 +104,11 @@ struct snd_card { size_t user_ctl_alloc_size; // current memory allocation by user controls. struct list_head controls; /* all controls for this card */ struct list_head ctl_files; /* active control files */ +#ifdef CONFIG_SND_CTL_FAST_LOOKUP + struct xarray ctl_numids; /* hash table for numids */ + struct xarray ctl_hash; /* hash table for ctl id matching */ + bool ctl_hash_collision; /* ctl_hash collision seen? */ +#endif struct snd_info_entry *proc_root; /* root for soundcard specific files */ struct proc_dir_entry *proc_root_link; /* number link to real id */ diff --git a/sound/core/Kconfig b/sound/core/Kconfig index dd7b40734723..fa0f6faf821a 100644 --- a/sound/core/Kconfig +++ b/sound/core/Kconfig @@ -154,6 +154,15 @@ config SND_VERBOSE_PRINTK You don't need this unless you're debugging ALSA. +config SND_CTL_FAST_LOOKUP + bool "Fast lookup of control elements" if EXPERT + default y + help + This option enables the faster lookup of control elements. + It will consume more memory because of the additional Xarray. + If you want to choose the memory footprint over the performance + inevitably, turn this off. + config SND_DEBUG bool "Debug" help diff --git a/sound/core/control.c b/sound/core/control.c index a25c0d64d104..6a8fd9933f06 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -364,6 +364,93 @@ static int snd_ctl_find_hole(struct snd_card *card, unsigned int count) return 0; } +/* check whether the given id is contained in the given kctl */ +static bool elem_id_matches(const struct snd_kcontrol *kctl, + const struct snd_ctl_elem_id *id) +{ + return kctl->id.iface == id->iface && + kctl->id.device == id->device && + kctl->id.subdevice == id->subdevice && + !strncmp(kctl->id.name, id->name, sizeof(kctl->id.name)) && + kctl->id.index <= id->index && + kctl->id.index + kctl->count > id->index; +} + +#ifdef CONFIG_SND_CTL_FAST_LOOKUP +/* Compute a hash key for the corresponding ctl id + * It's for the name lookup, hence the numid is excluded. + * The hash key is bound in LONG_MAX to be used for Xarray key. + */ +#define MULTIPLIER 37 +static unsigned long get_ctl_id_hash(const struct snd_ctl_elem_id *id) +{ + unsigned long h; + const unsigned char *p; + + h = id->iface; + h = MULTIPLIER * h + id->device; + h = MULTIPLIER * h + id->subdevice; + for (p = id->name; *p; p++) + h = MULTIPLIER * h + *p; + h = MULTIPLIER * h + id->index; + h &= LONG_MAX; + return h; +} + +/* add hash entries to numid and ctl xarray tables */ +static void add_hash_entries(struct snd_card *card, + struct snd_kcontrol *kcontrol) +{ + struct snd_ctl_elem_id id = kcontrol->id; + int i; + + xa_store_range(&card->ctl_numids, kcontrol->id.numid, + kcontrol->id.numid + kcontrol->count - 1, + kcontrol, GFP_KERNEL); + + for (i = 0; i < kcontrol->count; i++) { + id.index = kcontrol->id.index + i; + if (xa_insert(&card->ctl_hash, get_ctl_id_hash(&id), + kcontrol, GFP_KERNEL)) { + /* skip hash for this entry, noting we had collision */ + card->ctl_hash_collision = true; + dev_dbg(card->dev, "ctl_hash collision %d:%s:%d\n", + id.iface, id.name, id.index); + } + } +} + +/* remove hash entries that have been added */ +static void remove_hash_entries(struct snd_card *card, + struct snd_kcontrol *kcontrol) +{ + struct snd_ctl_elem_id id = kcontrol->id; + struct snd_kcontrol *matched; + unsigned long h; + int i; + + for (i = 0; i < kcontrol->count; i++) { + xa_erase(&card->ctl_numids, id.numid); + h = get_ctl_id_hash(&id); + matched = xa_load(&card->ctl_hash, h); + if (matched && (matched == kcontrol || + elem_id_matches(matched, &id))) + xa_erase(&card->ctl_hash, h); + id.index++; + id.numid++; + } +} +#else /* CONFIG_SND_CTL_FAST_LOOKUP */ +static inline void add_hash_entries(struct snd_card *card, + struct snd_kcontrol *kcontrol) +{ +} +static inline void remove_hash_entries(struct snd_card *card, + struct snd_kcontrol *kcontrol) +{ +} +#endif /* CONFIG_SND_CTL_FAST_LOOKUP */ + enum snd_ctl_add_mode { CTL_ADD_EXCLUSIVE, CTL_REPLACE, CTL_ADD_ON_REPLACE, }; @@ -408,6 +495,8 @@ static int __snd_ctl_add_replace(struct snd_card *card, kcontrol->id.numid = card->last_numid + 1; card->last_numid += kcontrol->count; + add_hash_entries(card, kcontrol); + for (idx = 0; idx < kcontrol->count; idx++) snd_ctl_notify_one(card, SNDRV_CTL_EVENT_MASK_ADD, kcontrol, idx); @@ -479,6 +568,26 @@ int snd_ctl_replace(struct snd_card *card, struct snd_kcontrol *kcontrol, } EXPORT_SYMBOL(snd_ctl_replace); +static int __snd_ctl_remove(struct snd_card *card, + struct snd_kcontrol *kcontrol, + bool remove_hash) +{ + unsigned int idx; + + if (snd_BUG_ON(!card || !kcontrol)) + return -EINVAL; + list_del(&kcontrol->list); + + if (remove_hash) + remove_hash_entries(card, kcontrol); + + card->controls_count -= kcontrol->count; + for (idx = 0; idx < kcontrol->count; idx++) + snd_ctl_notify_one(card, SNDRV_CTL_EVENT_MASK_REMOVE, kcontrol, idx); + snd_ctl_free_one(kcontrol); + return 0; +} + /** * snd_ctl_remove - remove the control from the card and release it * @card: the card instance @@ -492,16 +601,7 @@ EXPORT_SYMBOL(snd_ctl_replace); */ int snd_ctl_remove(struct snd_card *card, struct snd_kcontrol *kcontrol) { - unsigned int idx; - - if (snd_BUG_ON(!card || !kcontrol)) - return -EINVAL; - list_del(&kcontrol->list); - card->controls_count -= kcontrol->count; - for (idx = 0; idx < kcontrol->count; idx++) - snd_ctl_notify_one(card, SNDRV_CTL_EVENT_MASK_REMOVE, kcontrol, idx); - snd_ctl_free_one(kcontrol); - return 0; + return __snd_ctl_remove(card, kcontrol, true); } EXPORT_SYMBOL(snd_ctl_remove); @@ -642,14 +742,30 @@ int snd_ctl_rename_id(struct snd_card *card, struct snd_ctl_elem_id *src_id, up_write(&card->controls_rwsem); return -ENOENT; } + remove_hash_entries(card, kctl); kctl->id = *dst_id; kctl->id.numid = card->last_numid + 1; card->last_numid += kctl->count; + add_hash_entries(card, kctl); up_write(&card->controls_rwsem); return 0; } EXPORT_SYMBOL(snd_ctl_rename_id); +#ifndef CONFIG_SND_CTL_FAST_LOOKUP +static struct snd_kcontrol * +snd_ctl_find_numid_slow(struct snd_card *card, unsigned int numid) +{ + struct snd_kcontrol *kctl; + + list_for_each_entry(kctl, &card->controls, list) { + if (kctl->id.numid <= numid && kctl->id.numid + kctl->count > numid) + return kctl; + } + return NULL; +} +#endif /* !CONFIG_SND_CTL_FAST_LOOKUP */ + /** * snd_ctl_find_numid - find the control instance with the given number-id * @card: the card instance @@ -665,15 +781,13 @@ EXPORT_SYMBOL(snd_ctl_rename_id); */ struct snd_kcontrol *snd_ctl_find_numid(struct snd_card *card, unsigned int numid) { - struct snd_kcontrol *kctl; - if (snd_BUG_ON(!card || !numid)) return NULL; - list_for_each_entry(kctl, &card->controls, list) { - if (kctl->id.numid <= numid && kctl->id.numid + kctl->count > numid) - return kctl; - } - return NULL; +#ifdef CONFIG_SND_CTL_FAST_LOOKUP + return xa_load(&card->ctl_numids, numid); +#else + return snd_ctl_find_numid_slow(card, numid); +#endif } EXPORT_SYMBOL(snd_ctl_find_numid); @@ -699,21 +813,18 @@ struct snd_kcontrol *snd_ctl_find_id(struct snd_card *card, return NULL; if (id->numid != 0) return snd_ctl_find_numid(card, id->numid); - list_for_each_entry(kctl, &card->controls, list) { - if (kctl->id.iface != id->iface) - continue; - if (kctl->id.device != id->device) - continue; - if (kctl->id.subdevice != id->subdevice) - continue; - if (strncmp(kctl->id.name, id->name, sizeof(kctl->id.name))) - continue; - if (kctl->id.index > id->index) - continue; - if (kctl->id.index + kctl->count <= id->index) - continue; +#ifdef CONFIG_SND_CTL_FAST_LOOKUP + kctl = xa_load(&card->ctl_hash, get_ctl_id_hash(id)); + if (kctl && elem_id_matches(kctl, id)) return kctl; - } + if (!card->ctl_hash_collision) + return NULL; /* we can rely on only hash table */ +#endif + /* no matching in hash table - try all as the last resort */ + list_for_each_entry(kctl, &card->controls, list) + if (elem_id_matches(kctl, id)) + return kctl; + return NULL; } EXPORT_SYMBOL(snd_ctl_find_id); @@ -2195,8 +2306,13 @@ static int snd_ctl_dev_free(struct snd_device *device) down_write(&card->controls_rwsem); while (!list_empty(&card->controls)) { control = snd_kcontrol(card->controls.next); - snd_ctl_remove(card, control); + __snd_ctl_remove(card, control, false); } + +#ifdef CONFIG_SND_CTL_FAST_LOOKUP + xa_destroy(&card->ctl_numids); + xa_destroy(&card->ctl_hash); +#endif up_write(&card->controls_rwsem); put_device(&card->ctl_dev); return 0; diff --git a/sound/core/init.c b/sound/core/init.c index 726a8353201f..1870aee7b64f 100644 --- a/sound/core/init.c +++ b/sound/core/init.c @@ -310,6 +310,10 @@ static int snd_card_init(struct snd_card *card, struct device *parent, rwlock_init(&card->ctl_files_rwlock); INIT_LIST_HEAD(&card->controls); INIT_LIST_HEAD(&card->ctl_files); +#ifdef CONFIG_SND_CTL_FAST_LOOKUP + xa_init(&card->ctl_numids); + xa_init(&card->ctl_hash); +#endif spin_lock_init(&card->files_lock); INIT_LIST_HEAD(&card->files_list); mutex_init(&card->memory_mutex); -- 2.35.3

1 0

Re: [RFC PATCH v2 5/5] ASoC: apple: Add macaudio machine driver
by Martin Povišer 09 Jun '22

09 Jun '22

> On 9. 6. 2022, at 17:50, Mark Brown <broonie(a)kernel.org> wrote: > > On Thu, Jun 09, 2022 at 05:24:49PM +0200, Martin Povišer wrote: >>> On 9. 6. 2022, at 17:03, Mark Brown <broonie(a)kernel.org> wrote: > > Why is this off list? By accident, added the CC list back with this reply (hopefully it still attaches to the thread when people receive it). >>> That's basically no userspaces at this point TBH. I'm not convinced >>> it's a good idea to be adding custom code for that use case. >> >> FWIW I know of at least one user of the WIP audio support on Macs who >> would welcome this feature. My preference is to keep it in, but in >> the end I guess it’s your call. > > I'd rather not have this open coded in individual drivers, we already > have an unfortunate abundance of jack detection interfaces. If we're > going to add anything I'd rather it were in core code and TBH I'm > struggling to be enthusiastic. Noted. > Can you say anything more about the use case? I can restate: The alleged use case is running userspace without sound server, but having playback switch transparently between speakers and headphones even mid-stream based on jack detection. >>>> I looked at the existing DAPM integration but I couldn’t figure out >>>> how to switch the demux with it. > >>> Yes, it won't do that. If you can't stream the same audio to both then >>> you'd need something else. > >> I don’t understand what’s meant by streaming the same audio here. > > Playing one audio stream from the host which appears on both speakers > and headphones - I don't know what the mixing and muxing capabilities of > the hardware are. > >> Taking a guess: The existing DAPM integration can enable the headphones >> path based on jack being plugged in, but it can’t disable the speakers >> path like the demux does? > > No, that works perfectly fine - you can enable or disable pins depending > on the jack state. Ah, I peeked into soc-jack.c. What about this then: If I understand what pins represent, they would be at the remote end of the DAPM paths. So if for the speakers I add something like Headphones Codec Out —> Jack pin +--> Always-on pin | Speaker Amp Out -> Mux | +--> Jack inverted pin and let userspace control the mux, it would in effect support the same use cases as what I attempted in the code so far. Sounds somewhat right?

2 1

[PATCH 0/3] ASoC: SOF: Intel: Fix IMR boot race caused by early core booting
by Peter Ujfalusi 09 Jun '22

09 Jun '22

Hi, After the revisited IMR sequence patch 2a68ff846164 ("ASoC: SOF: Intel: hda: Revisit IMR boot sequence") we have started to see random boot issues manifesting in ghost reply messages from the DSP for a never sent message. It turned out that an earlier commit: d416519982cb ("ASoC: SOF: hda: don't use the core op for power up/power down") replaced the core power up step with a full blown core power up + enable call. The result was that we have booted the core before we would place the message to the ROM code and based on luck and timing it got received by the (IMR) booted firmware or in between ROM and firmware boot. Mark, can you queue this series for 5.19 as the 2a68ff846164 is in rc1 to make sure that we fix this random issue? Regards, Peter --- Peter Ujfalusi (3): ASoC: SOF: Intel: hda-dsp: Expose hda_dsp_core_power_up() ASoC: SOF: Intel: hda-loader: Make sure that the fw load sequence is followed ASoC: SOF: Intel: hda-loader: Clarify the cl_dsp_init() flow sound/soc/sof/intel/hda-dsp.c | 10 +++++++++- sound/soc/sof/intel/hda-loader.c | 10 +++++----- sound/soc/sof/intel/hda.h | 1 + 3 files changed, 15 insertions(+), 6 deletions(-) -- 2.36.1

2 4

[PATCH V2] ASoC: imx-audmux: remove unnecessary check of clk_disable_unprepare/clk_prepare_enable
by cgel.zte＠gmail.com 09 Jun '22

09 Jun '22

From: Minghao Chi <chi.minghao(a)zte.com.cn> Because clk_disable_unprepare/clk_prepare_enable already checked NULL clock parameter, so the additional checks are unnecessary, just remove them. Reported-by: Zeal Robot <zealci(a)zte.com.cn> Signed-off-by: Minghao Chi <chi.minghao(a)zte.com.cn> --- v1->v2: remove the check of audmux_clk before "clk_prepare_enable" sound/soc/fsl/imx-audmux.c | 22 ++++++++-------------- 1 file changed, 8 insertions(+), 14 deletions(-) diff --git a/sound/soc/fsl/imx-audmux.c b/sound/soc/fsl/imx-audmux.c index dfa05d40b276..3ba82adace42 100644 --- a/sound/soc/fsl/imx-audmux.c +++ b/sound/soc/fsl/imx-audmux.c @@ -62,17 +62,14 @@ static ssize_t audmux_read_file(struct file *file, char __user *user_buf, uintptr_t port = (uintptr_t)file->private_data; u32 pdcr, ptcr; - if (audmux_clk) { - ret = clk_prepare_enable(audmux_clk); - if (ret) - return ret; - } + ret = clk_prepare_enable(audmux_clk); + if (ret) + return ret; ptcr = readl(audmux_base + IMX_AUDMUX_V2_PTCR(port)); pdcr = readl(audmux_base + IMX_AUDMUX_V2_PDCR(port)); - if (audmux_clk) - clk_disable_unprepare(audmux_clk); + clk_disable_unprepare(audmux_clk); buf = kmalloc(PAGE_SIZE, GFP_KERNEL); if (!buf) @@ -209,17 +206,14 @@ int imx_audmux_v2_configure_port(unsigned int port, unsigned int ptcr, if (!audmux_base) return -ENOSYS; - if (audmux_clk) { - ret = clk_prepare_enable(audmux_clk); - if (ret) - return ret; - } + ret = clk_prepare_enable(audmux_clk); + if (ret) + return ret; writel(ptcr, audmux_base + IMX_AUDMUX_V2_PTCR(port)); writel(pdcr, audmux_base + IMX_AUDMUX_V2_PDCR(port)); - if (audmux_clk) - clk_disable_unprepare(audmux_clk); + clk_disable_unprepare(audmux_clk); return 0; } -- 2.25.1

3 2

Audio mem2mem devices aka asymmetric sample rate converters
by Sascha Hauer 09 Jun '22

09 Jun '22

Hi All, I am currently looking into getting the asymmetric sample rate converters (ASRC) found on some i.MX SoCs to do something useful. The ASRC units are completely independent units, i.e. independent of the rest of the audio subsystem. They can read from memory using the SDMA engine, convert sample rates and/or audio formats and write back to memory also using the SDMA engine. The ASRC on the i.MX8MN has four contexts to convert up to four streams simultanously. I am not aware of any other non i.MX SoCs having such a unit, but I am pretty sure they exist on other SoCs as well. There are likely two usecases for such a unit. First would be to offload sample rate and format conversions to hardware. The other would be to synchronize different audio sources/sinks with different master clocks to each other when the master clocks drift away. How would such units be integrated into ASoC? I can think of two ways. First would be to create an separate audio card from them which records on one end and plays back with a different sample rate / format on the other end, in the v4l2 world that would be a classical mem2mem device. Is Alsa/ASoc prepared for something like this? Would it be feasible to go into such a direction? I haven't found any examples for this in the tree. The other way is to attach the ASRC to an existing audio card. That is done with the existing in-tree sound/soc/fsl/fsl_asrc.c and sound/soc/fsl/fsl_easrc.c drivers. This approach feels somehow limited as it's not possible to just do conversions without playing/recording something. OTOH userspace is unaffected which might be an advantage. What nags me with that approach is that it's currently not integrated into the simple-audio-card or audio-graph-card bindings. Currently the driver can only be used in conjunction with the fsl,imx-audio-* card driver. It seems backward to integrate such a generic ASRC unit into a special purpose audio card driver. The ASoC core is fully unaware of the ASRC with this approach currently which also doesn't look very appealing. OTOH I don't know if ASoC could handle this. Can ASoC handle for example a chain of DAIs when there are different sample rates and formats in that chain? Currently I don't really know how to proceed. It would be great if you could share some thoughts to this topic. Thanks, Sascha -- Pengutronix e.K. | | Steuerwalder Str. 21 | http://www.pengutronix.de/ | 31137 Hildesheim, Germany | Phone: +49-5121-206917-0 | Amtsgericht Hildesheim, HRA 2686 | Fax: +49-5121-206917-5555 |

2 3

[PATCH 0/4] ALSA: input validation for control element writes
by Takashi Iwai 09 Jun '22

09 Jun '22

Hi, this is a patch set to add a new feature to ALSA control interface, a validation for the write to control elements. So far we rely fully on the driver implementation to deal with the user inputs with invalid values (e.g. a value outside the defined range). This patch set allows ALSA core to verify the input values beforehand and returns the error immediately if the validation fails. The feature is enabled with a new Kconfig for now, as it brings a slight performance overhead, although this could be turned on as default in most cases, IMO. The patch set contains a few preliminary cleanup patches. The essential change is only the last one. Takashi === Takashi Iwai (4): ASoC: topology: Drop superfluous check of CONFIG_SND_CTL_VALIDATION ALSA: control: Rename CONFIG_SND_CTL_VALIDATION to CONFIG_SND_CTL_DEBUG ALSA: control: Drop superfluous ifdef CONFIG_SND_CTL_DEBUG ALSA: control: Add input validation include/sound/control.h | 2 +- sound/core/Kconfig | 27 ++++++++++--- sound/core/control.c | 87 +++++++++++++++++++++++----------------- sound/soc/soc-topology.c | 2 +- 4 files changed, 73 insertions(+), 45 deletions(-) -- 2.35.3

1 4

[PATCH v3 0/2] Add software clock gating requirement check
by Srinivasa Rao Mandadapu 09 Jun '22

09 Jun '22

This patch set is to add software clock gating requirement check Changes Since V1: -- Use boolean flag for bool variable initialization instead of hard coding. Changes Since V1: -- Fix if check before reset control call Srinivasa Rao Mandadapu (2): soundwire: qcom: Add flag for software clock gating check ASoC: qcom: soundwire: Add software clock gating requirement check drivers/soundwire/qcom.c | 28 +++++++++++++++++++++------- 1 file changed, 21 insertions(+), 7 deletions(-) -- 2.7.4

3 6

[PATCH v6 0/8] ASoC: mediatek: Add support for MT8186 SoC
by Jiaxin Yu 09 Jun '22

09 Jun '22

This series of patches adds support for Mediatek AFE of MT8186 Soc. Patches are based on broonie tree "for-next" branch. Changes since v5: - fix build error about function snd_soc_card_jack_new that modified by: Link: https://lore.kernel.org/r/20220408041114.6024-1-akihiko.odaki@gmail.com - some have been accepted, details are in the link below: Link: https://lore.kernel.org/all/165459931885.399031.2621579592368573898.b4-ty@k… Changes since v4: - [v5 07/20] - remove unsusd controls - [v5 09/20] - correct indent error - [v5 10/20] - [v5 13/20] - [v5 14/20] - fix the return value if the value is different from the previous value in mixer controls - [v5 17/20] - [v5 19/20] - correct the compatible name with '_' instead of '-' - [v5 18/20] - [v5 20/20] - correct the yaml after 'pip3 install dtschema --upgrade' Changes since v3: - [v4 09/18] - remove DEBUG_COEFF debugging code - [v4 10/18] - simplify the logic of the code - [v4 13/18] - split out the MT6366 bits into mt8186-mt6366-commom.c - fix build error of "error: 'struct dev_pm_info' has no member named 'runtime_error'" - fix bug of adda dai driver - add route for pcm interface channel 2. - [v4 15/18] - [v4 17/18] - commonize the configuration of the codec - move MT6366 common bits into mt8186-mt6366-common.c Changes since v2: - add a new compatible string "mediatek,mt6366-sound" - modify the log level for simplicity - use dev_err_probe(...) instead of dev_err(...) in dev probe() - optimized the logic of some code - use BIT() and GENMASK() macros to descript the registers Thanks for AngeloGioacchino's careful reviews. Changes since v1: [v2 01/17] - add a new ID to the existing mt6358 codec driver [v2 03/17] - modify log level in DAPM events - use standard numeric control with name ending in Switch - return 1 when the value changed in mixer control's .get callback [v2 05/17] - ending in Switch to the standard on/off controls - change to "HW Gain 1 Volume" and "HW Gain 2 Volume" [v2 09/17] - return an error in the default case rather than just picking one of the behaviours when do .set_fmt - use the new defines that are _PROVIDER_MASK, _DAIFMT_CBP_CFP and _DAIFMT_CBC_CFC [v2 10/17] [v2 11/17] - the clock and gpio are aplit out into separate patches The source file's GPL comment use c++ style, and the header fils's GPL comment use c style. We have added "Switch" after the names of all the controls that just are simple on/off. Jiaxin Yu (8): dt-bindings: mediatek: mt6358: add new compatible for using mt6366 ASoC: mediatek: mt8186: add platform driver ASoC: mediatek: mt8186: add mt8186-mt6366 common driver dt-bindings: mediatek: mt8186: add audio afe document ASoC: mediatek: mt8186: add machine driver with mt6366, da7219 and max98357 dt-bindings: mediatek: mt8186: add mt8186-mt6366-da7219-max98357 document ASoC: mediatek: mt8186: add machine driver with mt6366, rt1019 and rt5682s dt-bindings: mediatek: mt8186: add mt8186-mt6366-rt1019-rt5682s document .../devicetree/bindings/sound/mt6358.txt | 4 +- .../bindings/sound/mt8186-afe-pcm.yaml | 175 + .../sound/mt8186-mt6366-da7219-max98357.yaml | 75 + .../sound/mt8186-mt6366-rt1019-rt5682s.yaml | 75 + sound/soc/mediatek/Kconfig | 44 + sound/soc/mediatek/Makefile | 1 + sound/soc/mediatek/mt8186/Makefile | 22 + sound/soc/mediatek/mt8186/mt8186-afe-common.h | 235 ++ .../soc/mediatek/mt8186/mt8186-afe-control.c | 261 ++ sound/soc/mediatek/mt8186/mt8186-afe-pcm.c | 3009 +++++++++++++++++ .../mediatek/mt8186/mt8186-mt6366-common.c | 59 + .../mediatek/mt8186/mt8186-mt6366-common.h | 17 + .../mt8186/mt8186-mt6366-da7219-max98357.c | 1002 ++++++ .../mt8186/mt8186-mt6366-rt1019-rt5682s.c | 978 ++++++ 14 files changed, 5956 insertions(+), 1 deletion(-) create mode 100644 Documentation/devicetree/bindings/sound/mt8186-afe-pcm.yaml create mode 100644 Documentation/devicetree/bindings/sound/mt8186-mt6366-da7219-max98357.yaml create mode 100644 Documentation/devicetree/bindings/sound/mt8186-mt6366-rt1019-rt5682s.yaml create mode 100644 sound/soc/mediatek/mt8186/Makefile create mode 100644 sound/soc/mediatek/mt8186/mt8186-afe-common.h create mode 100644 sound/soc/mediatek/mt8186/mt8186-afe-control.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-afe-pcm.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-mt6366-common.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-mt6366-common.h create mode 100644 sound/soc/mediatek/mt8186/mt8186-mt6366-da7219-max98357.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-mt6366-rt1019-rt5682s.c -- 2.25.1

2 10

About Cleanup ASoC
by Kuninori Morimoto 09 Jun '22

09 Jun '22

Hi ASoC Cc Mark, Pierre-Louis, Cezary I have very interesting to clean up ASoC. I think it is still very complex, thus, non flexible. I'm thinking that we want to cleanup... - Component with multi Card connection - fixed playback vs capture - fixed CPU vs Codec - DPCM connection vs normal connection About first topic, I guess the biggest reason why we have limitation is its connections. Text based image is a little bit difficult, but if my understanding was correct, current connections are... card <-> rtd <-> rtd <-> rtd <-> ... | | +--------|-----+---------------+-----------+ (A) | | | | +-> component -> component -> component -> ... | (CPU) (Codec) (Platform) | | | | | +-> dai -> dai -> dai -> dai -> ... Card has "component list", and Component has pointer to Card (A). This makes the limitation which blocking multiple Card connections, I think. One note here is that "Platform" doesn't have "DAI". So if rtd have something other connector list, let's say "ep" list here for now, instead of "dai", and remove Card <-> Component connection, I think the connection can be simple, and the issue can be gone, but what do you think ? card <-> rtd <-> rtd <-> rtd <-> ... | | dai dai dai dai | | | | | +--> ep --> ep --> ep -> ep -> ep -> ... | | | | | component component component (CPU) (Codec) (Platform) Here, ep is like this struct rtd_entry_point { struct list_head list; // list for rtd struct struct snd_soc_pcm_runtime *rtd; struct snd_soc_component *component; struct snd_soc_dai *dai; }; Current for_each_xxx() can be like this for_each_rtd_components() = for_each_rtd_ep() + ep_to_component() for_each_rtd_dais() = for_each_rtd_ep() + ep_to_dai() for_each_card_components() = for_each_card_rtds() + for_each_rtd_components() I guess Component can have multiple Card connection on this idea, but what do you think ? Maybe I'm misunderstanding and/or missing something, because ASoC is complex enough :) Please let me know if I was misunderstanding or missing. Thank you for your help !! Best regards --- Kuninori Morimoto

5 23

[syzbot] general protection fault in snd_virmidi_output_trigger
by syzbot 09 Jun '22

09 Jun '22

Hello, syzbot found the following issue on: HEAD commit: 952923ddc011 Merge tag 'pull-18-rc1-work.namei' of git://g.. git tree: upstream console output: https://syzkaller.appspot.com/x/log.txt?x=10b95d0bf00000 kernel config: https://syzkaller.appspot.com/x/.config?x=3096247591885bfa dashboard link: https://syzkaller.appspot.com/bug?extid=8e407e6b1dba0a275884 compiler: gcc (Debian 10.2.1-6) 10.2.1 20210110, GNU ld (GNU Binutils for Debian) 2.35.2 Unfortunately, I don't have any reproducer for this issue yet. IMPORTANT: if you fix the issue, please add the following tag to the commit: Reported-by: syzbot+8e407e6b1dba0a275884(a)syzkaller.appspotmail.com general protection fault, probably for non-canonical address 0xdffffc0000000003: 0000 [#1] PREEMPT SMP KASAN KASAN: null-ptr-deref in range [0x0000000000000018-0x000000000000001f] CPU: 1 PID: 18021 Comm: syz-executor.1 Not tainted 5.18.0-syzkaller-13842-g952923ddc011 #0 Hardware name: Google Google Compute Engine/Google Compute Engine, BIOS Google 01/01/2011 RIP: 0010:snd_virmidi_output_trigger+0x82/0x130 sound/core/seq/seq_virmidi.c:169 Code: 8b ad 30 01 00 00 31 ff 89 de e8 d9 8b 4f fa 85 db 48 b8 00 00 00 00 00 fc ff df 41 0f 95 c4 48 8d 7d 1c 48 89 fa 48 c1 ea 03 <0f> b6 04 02 48 89 fa 83 e2 07 38 d0 7f 04 84 c0 75 60 31 ff 44 88 RSP: 0018:ffffc9000c6e7c48 EFLAGS: 00010207 RAX: dffffc0000000000 RBX: 0000000000000001 RCX: ffffffff872af747 RDX: 0000000000000003 RSI: 0000000000000000 RDI: 000000000000001c RBP: 0000000000000000 R08: 0000000000000005 R09: 0000000000000000 R10: 0000000000000001 R11: 0000000000000001 R12: ffff888078d6c801 R13: 000000007f012000 R14: ffff88814ab3c400 R15: 0000000000001000 FS: 00007fc5b5868700(0000) GS:ffff8880b9b00000(0000) knlGS:0000000000000000 CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 CR2: 0000000020fec000 CR3: 0000000046903000 CR4: 00000000003506e0 DR0: 0000000000000000 DR1: 0000000000000000 DR2: 0000000000000000 DR3: 0000000000000000 DR6: 00000000fffe0ff0 DR7: 0000000000000400 Call Trace: <TASK> snd_rawmidi_output_trigger sound/core/rawmidi.c:186 [inline] snd_rawmidi_output_trigger sound/core/rawmidi.c:182 [inline] snd_rawmidi_kernel_write1+0x548/0x870 sound/core/rawmidi.c:1502 snd_rawmidi_write+0x273/0xb90 sound/core/rawmidi.c:1555 vfs_write+0x269/0xac0 fs/read_write.c:589 ksys_write+0x1e8/0x250 fs/read_write.c:644 do_syscall_x64 arch/x86/entry/common.c:50 [inline] do_syscall_64+0x35/0xb0 arch/x86/entry/common.c:80 entry_SYSCALL_64_after_hwframe+0x46/0xb0 RIP: 0033:0x7fc5b4689109 Code: ff ff c3 66 2e 0f 1f 84 00 00 00 00 00 0f 1f 40 00 48 89 f8 48 89 f7 48 89 d6 48 89 ca 4d 89 c2 4d 89 c8 4c 8b 4c 24 08 0f 05 <48> 3d 01 f0 ff ff 73 01 c3 48 c7 c1 b8 ff ff ff f7 d8 64 89 01 48 RSP: 002b:00007fc5b5868168 EFLAGS: 00000246 ORIG_RAX: 0000000000000001 RAX: ffffffffffffffda RBX: 00007fc5b479bf60 RCX: 00007fc5b4689109 RDX: 00000000fffffd2c RSI: 0000000020000000 RDI: 0000000000000004 RBP: 00007fc5b46e308d R08: 0000000000000000 R09: 0000000000000000 R10: 0000000000000000 R11: 0000000000000246 R12: 0000000000000000 R13: 00007fc5b4ccfb1f R14: 00007fc5b5868300 R15: 0000000000022000 </TASK> Modules linked in: ---[ end trace 0000000000000000 ]--- RIP: 0010:snd_virmidi_output_trigger+0x82/0x130 sound/core/seq/seq_virmidi.c:169 Code: 8b ad 30 01 00 00 31 ff 89 de e8 d9 8b 4f fa 85 db 48 b8 00 00 00 00 00 fc ff df 41 0f 95 c4 48 8d 7d 1c 48 89 fa 48 c1 ea 03 <0f> b6 04 02 48 89 fa 83 e2 07 38 d0 7f 04 84 c0 75 60 31 ff 44 88 RSP: 0018:ffffc9000c6e7c48 EFLAGS: 00010207 RAX: dffffc0000000000 RBX: 0000000000000001 RCX: ffffffff872af747 RDX: 0000000000000003 RSI: 0000000000000000 RDI: 000000000000001c RBP: 0000000000000000 R08: 0000000000000005 R09: 0000000000000000 R10: 0000000000000001 R11: 0000000000000001 R12: ffff888078d6c801 R13: 000000007f012000 R14: ffff88814ab3c400 R15: 0000000000001000 FS: 00007fc5b5868700(0000) GS:ffff8880b9a00000(0000) knlGS:0000000000000000 CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 CR2: 0000001b2bc27000 CR3: 0000000046903000 CR4: 00000000003506f0 DR0: 0000000000000000 DR1: 0000000000000000 DR2: 0000000000000000 DR3: 0000000000000000 DR6: 00000000fffe0ff0 DR7: 0000000000000400 ---------------- Code disassembly (best guess): 0: 8b ad 30 01 00 00 mov 0x130(%rbp),%ebp 6: 31 ff xor %edi,%edi 8: 89 de mov %ebx,%esi a: e8 d9 8b 4f fa callq 0xfa4f8be8 f: 85 db test %ebx,%ebx 11: 48 b8 00 00 00 00 00 movabs $0xdffffc0000000000,%rax 18: fc ff df 1b: 41 0f 95 c4 setne %r12b 1f: 48 8d 7d 1c lea 0x1c(%rbp),%rdi 23: 48 89 fa mov %rdi,%rdx 26: 48 c1 ea 03 shr $0x3,%rdx * 2a: 0f b6 04 02 movzbl (%rdx,%rax,1),%eax <-- trapping instruction 2e: 48 89 fa mov %rdi,%rdx 31: 83 e2 07 and $0x7,%edx 34: 38 d0 cmp %dl,%al 36: 7f 04 jg 0x3c 38: 84 c0 test %al,%al 3a: 75 60 jne 0x9c 3c: 31 ff xor %edi,%edi 3e: 44 rex.R 3f: 88 .byte 0x88 --- This report is generated by a bot. It may contain errors. See https://goo.gl/tpsmEJ for more information about syzbot. syzbot engineers can be reached at syzkaller(a)googlegroups.com. syzbot will keep track of this issue. See: https://goo.gl/tpsmEJ#status for how to communicate with syzbot.

1 0

[PATCH v2 1/2] ASoC: dt-bindings: nxp,tfa989x: Add tfa9890 support
by Alexander Martinz 08 Jun '22

08 Jun '22

Document TFA9890 binding for tfa989x. Signed-off-by: Alexander Martinz <amartinz(a)shiftphones.com> Reviewed-by: Stephan Gerhold <stephan(a)gerhold.net> Acked-by: Rob Herring <robh(a)kernel.org> --- Documentation/devicetree/bindings/sound/nxp,tfa989x.yaml | 1 + 1 file changed, 1 insertion(+) diff --git a/Documentation/devicetree/bindings/sound/nxp,tfa989x.yaml b/Documentation/devicetree/bindings/sound/nxp,tfa989x.yaml index b9b1dba40856..7f2e68ff6d34 100644 --- a/Documentation/devicetree/bindings/sound/nxp,tfa989x.yaml +++ b/Documentation/devicetree/bindings/sound/nxp,tfa989x.yaml @@ -15,6 +15,7 @@ allOf: properties: compatible: enum: + - nxp,tfa9890 - nxp,tfa9895 - nxp,tfa9897 -- 2.36.1

2 2

no input from razer headset microphone when using RAZER BASE STATION V2 CHROMA
by GitHub issues - opened 08 Jun '22

08 Jun '22

alsa-project/alsa-lib issue #236 was opened from fiercebrake: Greetings everyone! So I currently have a razer "blacksharck v2 pro" headset that i connect to a razer "base station v2 chroma" The razer base station is properly detected/listed by (arch) linux. Problem is though the card and mic are detected and listed, the mic has no actual input at all. For the mic to start showing/taking any input I need to manually unplug and replug the headset from the base station. The jack in the base station is a 4-pin 3.5mm (TRRS) jack This article/fix below seems to alleviate the problem, say 50% of the times. So I am wondering if there is anything else I can do via ALSA to completely fix the problem "https://wiki.archlinux.org/title/Advanced_Linux_Sound_Architecture#Correctl…" Issue URL : https://github.com/alsa-project/alsa-lib/issues/236 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH] ALSA: hda/realtek: Add quirk for HP Dev One
by Tim Crawford 08 Jun '22

08 Jun '22

From: Jeremy Soller <jeremy(a)system76.com> Enables the audio mute LEDs and limits the mic boost to avoid picking up noise. Signed-off-by: Jeremy Soller <jeremy(a)system76.com> Signed-off-by: Tim Crawford <tcrawford(a)system76.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 7170e086f166..b0f954118e72 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9071,6 +9071,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x89c3, "Zbook Studio G9", ALC245_FIXUP_CS35L41_SPI_4_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x89c6, "Zbook Fury 17 G9", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x89ca, "HP", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), + SND_PCI_QUIRK(0x103c, 0x8a78, "HP Dev One", ALC285_FIXUP_HP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x1043, 0x103e, "ASUS X540SA", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x103f, "ASUS TX300", ALC282_FIXUP_ASUS_TX300), SND_PCI_QUIRK(0x1043, 0x106d, "Asus K53BE", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), -- 2.35.3

2 1

[PATCH][next] ASoC: mediatek: mt8186: Fix a handful of spelling mistakes
by Colin Ian King 08 Jun '22

08 Jun '22

There are several spelling mistakes in dev_err messages. Fix them. Signed-off-by: Colin Ian King <colin.i.king(a)gmail.com> --- sound/soc/mediatek/mt8186/mt8186-afe-gpio.c | 8 ++++---- sound/soc/mediatek/mt8186/mt8186-dai-adda.c | 2 +- 2 files changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/mediatek/mt8186/mt8186-afe-gpio.c b/sound/soc/mediatek/mt8186/mt8186-afe-gpio.c index 5ba28095b7da..255ffba637d3 100644 --- a/sound/soc/mediatek/mt8186/mt8186-afe-gpio.c +++ b/sound/soc/mediatek/mt8186/mt8186-afe-gpio.c @@ -137,25 +137,25 @@ static int mt8186_afe_gpio_adda_dl(struct device *dev, bool enable) if (enable) { ret = mt8186_afe_gpio_select(dev, MT8186_AFE_GPIO_CLK_MOSI_ON); if (ret) { - dev_err(dev, "%s(), MOSI CLK ON slect fail!\n", __func__); + dev_err(dev, "%s(), MOSI CLK ON select fail!\n", __func__); return ret; } ret = mt8186_afe_gpio_select(dev, MT8186_AFE_GPIO_DAT_MOSI_ON); if (ret) { - dev_err(dev, "%s(), MOSI DAT ON slect fail!\n", __func__); + dev_err(dev, "%s(), MOSI DAT ON select fail!\n", __func__); return ret; } } else { ret = mt8186_afe_gpio_select(dev, MT8186_AFE_GPIO_DAT_MOSI_OFF); if (ret) { - dev_err(dev, "%s(), MOSI DAT OFF slect fail!\n", __func__); + dev_err(dev, "%s(), MOSI DAT OFF select fail!\n", __func__); return ret; } ret = mt8186_afe_gpio_select(dev, MT8186_AFE_GPIO_CLK_MOSI_OFF); if (ret) { - dev_err(dev, "%s(), MOSI CLK ON slect fail!\n", __func__); + dev_err(dev, "%s(), MOSI CLK ON select fail!\n", __func__); return ret; } } diff --git a/sound/soc/mediatek/mt8186/mt8186-dai-adda.c b/sound/soc/mediatek/mt8186/mt8186-dai-adda.c index c66861fd197d..db71b032770d 100644 --- a/sound/soc/mediatek/mt8186/mt8186-dai-adda.c +++ b/sound/soc/mediatek/mt8186/mt8186-dai-adda.c @@ -341,7 +341,7 @@ static int mtk_adda_mtkaif_cfg_event(struct snd_soc_dapm_widget *w, if (afe_priv->mtkaif_chosen_phase[0] < 0 || afe_priv->mtkaif_chosen_phase[1] < 0) { dev_err(afe->dev, - "%s(), skip dealy setting mtkaif_chosen_phase[0/1]:%d/%d\n", + "%s(), skip delay setting mtkaif_chosen_phase[0/1]:%d/%d\n", __func__, afe_priv->mtkaif_chosen_phase[0], afe_priv->mtkaif_chosen_phase[1]); -- 2.35.3

2 1

[PATCH 0/7] ASoC: harden SoundWire codec/machine drivers used on Intel platforms
by Pierre-Louis Bossart 08 Jun '22

08 Jun '22

While testing fixes for SoundWire race conditions initially reported in [1], I found additional issues in codec drivers. When the order in which drivers are probed is changed, multiple errors are reported, from unbalanced pm_runtime_enable() calls to invalid mutex lock magic numbers, workqueues not initialized and missing resume sequences. In 'nominal' usages, there is no change of functionality, this is just a first step to test random device/driver bind/unbind sequences. Important note: these changes only touch Intel-based platforms, I don't have any background and ability to test on Qualcomm-based devices. [1] https://lore.kernel.org/alsa-devel/d0559e97-c4a0-b817-428c-d3e305390270@lin… Pierre-Louis Bossart (7): ASoC: Realtek/Maxim SoundWire codecs: disable pm_runtime on remove ASoC: rt711-sdca-sdw: fix calibrate mutex initialization ASoC: Intel: sof_sdw: handle errors on card registration ASoC: rt711: fix calibrate mutex initialization ASoC: rt7*-sdw: harden jack_detect_handler ASoC: codecs: rt700/rt711/rt711-sdca: initialize workqueues in probe ASoC: codecs: rt700/rt711/rt711-sdca: resume bus/codec in .set_jack_detect sound/soc/codecs/max98373-sdw.c | 12 +++++++- sound/soc/codecs/rt1308-sdw.c | 11 +++++++ sound/soc/codecs/rt1316-sdw.c | 11 +++++++ sound/soc/codecs/rt5682-sdw.c | 5 ++- sound/soc/codecs/rt700-sdw.c | 6 +++- sound/soc/codecs/rt700.c | 30 +++++++++++------- sound/soc/codecs/rt711-sdca-sdw.c | 9 +++++- sound/soc/codecs/rt711-sdca.c | 40 ++++++++++++------------ sound/soc/codecs/rt711-sdw.c | 9 +++++- sound/soc/codecs/rt711.c | 40 ++++++++++++------------ sound/soc/codecs/rt715-sdca-sdw.c | 12 ++++++++ sound/soc/codecs/rt715-sdw.c | 12 ++++++++ sound/soc/intel/boards/sof_sdw.c | 51 ++++++++++++++++++------------- 13 files changed, 169 insertions(+), 79 deletions(-) -- 2.34.1

2 8

[PATCH v2 0/2] Add software clock gating requirement check
by Srinivasa Rao Mandadapu 08 Jun '22

08 Jun '22

This patch set is to add software clock gating requirement check Changes Since V1: -- Fix if check before reset control call Srinivasa Rao Mandadapu (2): soundwire: qcom: Add flag for software clock gating check ASoC: qcom: soundwire: Add software clock gating requirement check drivers/soundwire/qcom.c | 28 +++++++++++++++++++++------- 1 file changed, 21 insertions(+), 7 deletions(-) -- 2.7.4

2 4

[PATCH] ASoC: ops: Fix off by one in range control validation
by Mark Brown 08 Jun '22

08 Jun '22

We currently report that range controls accept a range of 0..(max-min) but accept writes in the range 0..(max-min+1). Remove that extra +1. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/soc-ops.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c index 93e72a016b4d..e11109dcaa50 100644 --- a/sound/soc/soc-ops.c +++ b/sound/soc/soc-ops.c @@ -542,7 +542,7 @@ int snd_soc_put_volsw_range(struct snd_kcontrol *kcontrol, return -EINVAL; if (mc->platform_max && tmp > mc->platform_max) return -EINVAL; - if (tmp > mc->max - mc->min + 1) + if (tmp > mc->max - mc->min) return -EINVAL; if (invert) @@ -563,7 +563,7 @@ int snd_soc_put_volsw_range(struct snd_kcontrol *kcontrol, return -EINVAL; if (mc->platform_max && tmp > mc->platform_max) return -EINVAL; - if (tmp > mc->max - mc->min + 1) + if (tmp > mc->max - mc->min) return -EINVAL; if (invert) -- 2.30.2

1 1

[PATCH 00/20] ASoC: DAI clocking terminology modernisation
by Mark Brown 08 Jun '22

08 Jun '22

Update the last batch of CODEC drivers without specific maintainers to use the new defines for DAI clocking. Mark Brown (20): ASoC: cx2072x: Use modern ASoC DAI format terminology ASoC: max98090: Use modern ASoC DAI format terminology ASoC: rk3328: Use modern ASoC DAI format terminology ASoC: sta32x: Use modern ASoC DAI format terminology ASoC: sta350: Use modern ASoC DAI format terminology ASoC: sti-sas: Use modern ASoC DAI format terminology ASoC: tas2552: Use modern ASoC DAI format terminology ASoC: tas2770: Use modern ASoC DAI format terminology ASoC: tas5086: Use modern ASoC DAI format terminology ASoC: tas5720: Use modern ASoC DAI format terminology ASoC: tas6424: Use modern ASoC DAI format terminology ASoC: uda1334: Use modern ASoC DAI format terminology ASoC: tlv320adc3xxx: Use modern ASoC DAI format terminology ASoC: tlv320adcx140: Use modern ASoC DAI format terminology ASoC: tlv320aic23: Use modern ASoC DAI format terminology ASoC: tlv320aic26: Use modern ASoC DAI format terminology ASoC: tlv320aic31xx: Use modern ASoC DAI format terminology ASoC: tlv320aic32x4: Use modern ASoC DAI format terminology ASoC: tlv320aic33: Use modern ASoC DAI format terminology ASoC: tlv320dac3x: Use modern ASoC DAI format terminology sound/soc/codecs/cx2072x.c | 17 +++++++---------- sound/soc/codecs/max98090.c | 12 +++++------- sound/soc/codecs/rk3328_codec.c | 6 +++--- sound/soc/codecs/sta32x.c | 4 ++-- sound/soc/codecs/sta350.c | 4 ++-- sound/soc/codecs/sti-sas.c | 6 +++--- sound/soc/codecs/tas2552.c | 10 +++++----- sound/soc/codecs/tas2770.c | 6 +++--- sound/soc/codecs/tas5086.c | 2 +- sound/soc/codecs/tas5720.c | 4 ++-- sound/soc/codecs/tas6424.c | 6 +++--- sound/soc/codecs/tlv320adc3xxx.c | 3 +-- sound/soc/codecs/tlv320adcx140.c | 10 ++++------ sound/soc/codecs/tlv320aic23.c | 7 +++---- sound/soc/codecs/tlv320aic26.c | 13 ++++++------- sound/soc/codecs/tlv320aic31xx.c | 19 +++++++++---------- sound/soc/codecs/tlv320aic32x4.c | 9 ++++----- sound/soc/codecs/tlv320aic3x.c | 11 +++++------ sound/soc/codecs/tlv320dac33.c | 12 +++++------- sound/soc/codecs/uda1334.c | 2 +- 20 files changed, 74 insertions(+), 89 deletions(-) -- 2.30.2

2 23

[PATCH v4] ALSA: hda/realtek - Add HW8326 support
by huangwenhui 08 Jun '22

08 Jun '22

Added the support of new Huawei codec HW8326. The HW8326 is developed by Huawei with Realtek's IP Core, and it's compatible with ALC256. Signed-off-by: huangwenhui <huangwenhuia(a)uniontech.com> --- sound/hda/hdac_device.c | 1 + sound/pci/hda/patch_realtek.c | 14 ++++++++++++++ 2 files changed, 15 insertions(+) diff --git a/sound/hda/hdac_device.c b/sound/hda/hdac_device.c index 3e9e9ac804f6..b7e5032b61c9 100644 --- a/sound/hda/hdac_device.c +++ b/sound/hda/hdac_device.c @@ -660,6 +660,7 @@ static const struct hda_vendor_id hda_vendor_ids[] = { { 0x14f1, "Conexant" }, { 0x17e8, "Chrontel" }, { 0x1854, "LG" }, + { 0x19e5, "Huawei" }, { 0x1aec, "Wolfson Microelectronics" }, { 0x1af4, "QEMU" }, { 0x434d, "C-Media" }, diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index f3ad454b3fbf..cd1281691767 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -443,6 +443,7 @@ static void alc_fill_eapd_coef(struct hda_codec *codec) case 0x10ec0245: case 0x10ec0255: case 0x10ec0256: + case 0x19e58326: case 0x10ec0257: case 0x10ec0282: case 0x10ec0283: @@ -580,6 +581,7 @@ static void alc_shutup_pins(struct hda_codec *codec) switch (codec->core.vendor_id) { case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: case 0x10ec0283: case 0x10ec0286: case 0x10ec0288: @@ -3247,6 +3249,7 @@ static void alc_disable_headset_jack_key(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_write_coef_idx(codec, 0x48, 0x0); alc_update_coef_idx(codec, 0x49, 0x0045, 0x0); break; @@ -3275,6 +3278,7 @@ static void alc_enable_headset_jack_key(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_write_coef_idx(codec, 0x48, 0xd011); alc_update_coef_idx(codec, 0x49, 0x007f, 0x0045); break; @@ -4910,6 +4914,7 @@ static void alc_headset_mode_unplugged(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_process_coef_fw(codec, coef0256); break; case 0x10ec0234: @@ -5025,6 +5030,7 @@ static void alc_headset_mode_mic_in(struct hda_codec *codec, hda_nid_t hp_pin, case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_write_coef_idx(codec, 0x45, 0xc489); snd_hda_set_pin_ctl_cache(codec, hp_pin, 0); alc_process_coef_fw(codec, coef0256); @@ -5175,6 +5181,7 @@ static void alc_headset_mode_default(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_write_coef_idx(codec, 0x1b, 0x0e4b); alc_write_coef_idx(codec, 0x45, 0xc089); msleep(50); @@ -5274,6 +5281,7 @@ static void alc_headset_mode_ctia(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_process_coef_fw(codec, coef0256); break; case 0x10ec0234: @@ -5388,6 +5396,7 @@ static void alc_headset_mode_omtp(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_process_coef_fw(codec, coef0256); break; case 0x10ec0234: @@ -5489,6 +5498,7 @@ static void alc_determine_headset_type(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_write_coef_idx(codec, 0x1b, 0x0e4b); alc_write_coef_idx(codec, 0x06, 0x6104); alc_write_coefex_idx(codec, 0x57, 0x3, 0x09a3); @@ -5783,6 +5793,7 @@ static void alc255_set_default_jack_type(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_process_coef_fw(codec, alc256fw); break; } @@ -6385,6 +6396,7 @@ static void alc_combo_jack_hp_jd_restart(struct hda_codec *codec) case 0x10ec0236: case 0x10ec0255: case 0x10ec0256: + case 0x19e58326: alc_update_coef_idx(codec, 0x1b, 0x8000, 1 << 15); /* Reset HP JD */ alc_update_coef_idx(codec, 0x1b, 0x8000, 0 << 15); break; @@ -10095,6 +10107,7 @@ static int patch_alc269(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: spec->codec_variant = ALC269_TYPE_ALC256; spec->shutup = alc256_shutup; spec->init_hook = alc256_init; @@ -11545,6 +11558,7 @@ static const struct hda_device_id snd_hda_id_realtek[] = { HDA_CODEC_ENTRY(0x10ec0b00, "ALCS1200A", patch_alc882), HDA_CODEC_ENTRY(0x10ec1168, "ALC1220", patch_alc882), HDA_CODEC_ENTRY(0x10ec1220, "ALC1220", patch_alc882), + HDA_CODEC_ENTRY(0x19e58326, "HW8326", patch_alc269), {} /* terminator */ }; MODULE_DEVICE_TABLE(hdaudio, snd_hda_id_realtek); -- 2.20.1

2 1

[PATCH v3] ALSA: hda/realtek - Add HW8326 support
by huangwenhui 08 Jun '22

08 Jun '22

Added the support of new Huawei codec HW8326. The HW8326 is developed by Huawei with Realtek's IP Core, and it's compatible with ALC256. Signed-off-by: huangwenhui <huangwenhuia(a)uniontech.com> --- sound/hda/hdac_device.c | 1 + sound/pci/hda/patch_realtek.c | 14 ++++++++++++++ 2 files changed, 15 insertions(+) diff --git a/sound/hda/hdac_device.c b/sound/hda/hdac_device.c index 3e9e9ac804f6..775474f00141 100644 --- a/sound/hda/hdac_device.c +++ b/sound/hda/hdac_device.c @@ -651,6 +651,7 @@ static const struct hda_vendor_id hda_vendor_ids[] = { { 0x1095, "Silicon Image" }, { 0x10de, "Nvidia" }, { 0x10ec, "Realtek" }, + { 0x19e5, "Huawei" }, { 0x1102, "Creative" }, { 0x1106, "VIA" }, { 0x111d, "IDT" }, diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index f3ad454b3fbf..cd1281691767 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -443,6 +443,7 @@ static void alc_fill_eapd_coef(struct hda_codec *codec) case 0x10ec0245: case 0x10ec0255: case 0x10ec0256: + case 0x19e58326: case 0x10ec0257: case 0x10ec0282: case 0x10ec0283: @@ -580,6 +581,7 @@ static void alc_shutup_pins(struct hda_codec *codec) switch (codec->core.vendor_id) { case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: case 0x10ec0283: case 0x10ec0286: case 0x10ec0288: @@ -3247,6 +3249,7 @@ static void alc_disable_headset_jack_key(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_write_coef_idx(codec, 0x48, 0x0); alc_update_coef_idx(codec, 0x49, 0x0045, 0x0); break; @@ -3275,6 +3278,7 @@ static void alc_enable_headset_jack_key(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_write_coef_idx(codec, 0x48, 0xd011); alc_update_coef_idx(codec, 0x49, 0x007f, 0x0045); break; @@ -4910,6 +4914,7 @@ static void alc_headset_mode_unplugged(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_process_coef_fw(codec, coef0256); break; case 0x10ec0234: @@ -5025,6 +5030,7 @@ static void alc_headset_mode_mic_in(struct hda_codec *codec, hda_nid_t hp_pin, case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_write_coef_idx(codec, 0x45, 0xc489); snd_hda_set_pin_ctl_cache(codec, hp_pin, 0); alc_process_coef_fw(codec, coef0256); @@ -5175,6 +5181,7 @@ static void alc_headset_mode_default(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_write_coef_idx(codec, 0x1b, 0x0e4b); alc_write_coef_idx(codec, 0x45, 0xc089); msleep(50); @@ -5274,6 +5281,7 @@ static void alc_headset_mode_ctia(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_process_coef_fw(codec, coef0256); break; case 0x10ec0234: @@ -5388,6 +5396,7 @@ static void alc_headset_mode_omtp(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_process_coef_fw(codec, coef0256); break; case 0x10ec0234: @@ -5489,6 +5498,7 @@ static void alc_determine_headset_type(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_write_coef_idx(codec, 0x1b, 0x0e4b); alc_write_coef_idx(codec, 0x06, 0x6104); alc_write_coefex_idx(codec, 0x57, 0x3, 0x09a3); @@ -5783,6 +5793,7 @@ static void alc255_set_default_jack_type(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_process_coef_fw(codec, alc256fw); break; } @@ -6385,6 +6396,7 @@ static void alc_combo_jack_hp_jd_restart(struct hda_codec *codec) case 0x10ec0236: case 0x10ec0255: case 0x10ec0256: + case 0x19e58326: alc_update_coef_idx(codec, 0x1b, 0x8000, 1 << 15); /* Reset HP JD */ alc_update_coef_idx(codec, 0x1b, 0x8000, 0 << 15); break; @@ -10095,6 +10107,7 @@ static int patch_alc269(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: spec->codec_variant = ALC269_TYPE_ALC256; spec->shutup = alc256_shutup; spec->init_hook = alc256_init; @@ -11545,6 +11558,7 @@ static const struct hda_device_id snd_hda_id_realtek[] = { HDA_CODEC_ENTRY(0x10ec0b00, "ALCS1200A", patch_alc882), HDA_CODEC_ENTRY(0x10ec1168, "ALC1220", patch_alc882), HDA_CODEC_ENTRY(0x10ec1220, "ALC1220", patch_alc882), + HDA_CODEC_ENTRY(0x19e58326, "HW8326", patch_alc269), {} /* terminator */ }; MODULE_DEVICE_TABLE(hdaudio, snd_hda_id_realtek); -- 2.20.1

3 4

[PATCH] [PATCH v2]ALSA: hda/realtek - Add HW8326 support
by huangwenhui 08 Jun '22

08 Jun '22

Added the support of new Huawei codec HW8326.The HW8326 is developed by Huawei with Realtek's IP Core, and it's compatible with ALC256. Signed-off-by: huangwenhui <huangwenhuia(a)uniontech.com> --- sound/hda/hdac_device.c | 1 + sound/pci/hda/patch_realtek.c | 14 ++++++++++++++ 2 files changed, 15 insertions(+) diff --git a/sound/hda/hdac_device.c b/sound/hda/hdac_device.c index 3e9e9ac804f6..775474f00141 100644 --- a/sound/hda/hdac_device.c +++ b/sound/hda/hdac_device.c @@ -651,6 +651,7 @@ static const struct hda_vendor_id hda_vendor_ids[] = { { 0x1095, "Silicon Image" }, { 0x10de, "Nvidia" }, { 0x10ec, "Realtek" }, + { 0x19e5, "Huawei" }, { 0x1102, "Creative" }, { 0x1106, "VIA" }, { 0x111d, "IDT" }, diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index f3ad454b3fbf..cd1281691767 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -443,6 +443,7 @@ static void alc_fill_eapd_coef(struct hda_codec *codec) case 0x10ec0245: case 0x10ec0255: case 0x10ec0256: + case 0x19e58326: case 0x10ec0257: case 0x10ec0282: case 0x10ec0283: @@ -580,6 +581,7 @@ static void alc_shutup_pins(struct hda_codec *codec) switch (codec->core.vendor_id) { case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: case 0x10ec0283: case 0x10ec0286: case 0x10ec0288: @@ -3247,6 +3249,7 @@ static void alc_disable_headset_jack_key(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_write_coef_idx(codec, 0x48, 0x0); alc_update_coef_idx(codec, 0x49, 0x0045, 0x0); break; @@ -3275,6 +3278,7 @@ static void alc_enable_headset_jack_key(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_write_coef_idx(codec, 0x48, 0xd011); alc_update_coef_idx(codec, 0x49, 0x007f, 0x0045); break; @@ -4910,6 +4914,7 @@ static void alc_headset_mode_unplugged(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_process_coef_fw(codec, coef0256); break; case 0x10ec0234: @@ -5025,6 +5030,7 @@ static void alc_headset_mode_mic_in(struct hda_codec *codec, hda_nid_t hp_pin, case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_write_coef_idx(codec, 0x45, 0xc489); snd_hda_set_pin_ctl_cache(codec, hp_pin, 0); alc_process_coef_fw(codec, coef0256); @@ -5175,6 +5181,7 @@ static void alc_headset_mode_default(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_write_coef_idx(codec, 0x1b, 0x0e4b); alc_write_coef_idx(codec, 0x45, 0xc089); msleep(50); @@ -5274,6 +5281,7 @@ static void alc_headset_mode_ctia(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_process_coef_fw(codec, coef0256); break; case 0x10ec0234: @@ -5388,6 +5396,7 @@ static void alc_headset_mode_omtp(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_process_coef_fw(codec, coef0256); break; case 0x10ec0234: @@ -5489,6 +5498,7 @@ static void alc_determine_headset_type(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_write_coef_idx(codec, 0x1b, 0x0e4b); alc_write_coef_idx(codec, 0x06, 0x6104); alc_write_coefex_idx(codec, 0x57, 0x3, 0x09a3); @@ -5783,6 +5793,7 @@ static void alc255_set_default_jack_type(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_process_coef_fw(codec, alc256fw); break; } @@ -6385,6 +6396,7 @@ static void alc_combo_jack_hp_jd_restart(struct hda_codec *codec) case 0x10ec0236: case 0x10ec0255: case 0x10ec0256: + case 0x19e58326: alc_update_coef_idx(codec, 0x1b, 0x8000, 1 << 15); /* Reset HP JD */ alc_update_coef_idx(codec, 0x1b, 0x8000, 0 << 15); break; @@ -10095,6 +10107,7 @@ static int patch_alc269(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: spec->codec_variant = ALC269_TYPE_ALC256; spec->shutup = alc256_shutup; spec->init_hook = alc256_init; @@ -11545,6 +11558,7 @@ static const struct hda_device_id snd_hda_id_realtek[] = { HDA_CODEC_ENTRY(0x10ec0b00, "ALCS1200A", patch_alc882), HDA_CODEC_ENTRY(0x10ec1168, "ALC1220", patch_alc882), HDA_CODEC_ENTRY(0x10ec1220, "ALC1220", patch_alc882), + HDA_CODEC_ENTRY(0x19e58326, "HW8326", patch_alc269), {} /* terminator */ }; MODULE_DEVICE_TABLE(hdaudio, snd_hda_id_realtek); -- 2.20.1

2 1

Combined wait for pcm_wait and ctl_wait
by Pavel Hofman 08 Jun '22

08 Jun '22

Hi, Please is there any way to wait for nonblocking read/write at min_avail (like snd_pcm_wait) AND nonblocking subscribed ctl event (like snd_ctl_wait) in a single combined wait? The goal is to react as fast as possible to ctls in aloop and audio gadget devices which notify about stream stop and start (at new rate/format/channel count) on the other side. I guess some wizardry with file descriptors and polling multiple fds could do. Thanks a lot for any help or hints. Best regards, Pavel.

2 2

[PATCH] ALSA: hda/realtek - Add HW8326 support
by huangwenhui 08 Jun '22

08 Jun '22

Added the support of new Huawei codec HW8326. Signed-off-by: huangwenhui <huangwenhuia(a)uniontech.com> --- sound/hda/hdac_device.c | 1 + sound/pci/hda/patch_realtek.c | 14 ++++++++++++++ 2 files changed, 15 insertions(+) diff --git a/sound/hda/hdac_device.c b/sound/hda/hdac_device.c index 3e9e9ac804f6..775474f00141 100644 --- a/sound/hda/hdac_device.c +++ b/sound/hda/hdac_device.c @@ -651,6 +651,7 @@ static const struct hda_vendor_id hda_vendor_ids[] = { { 0x1095, "Silicon Image" }, { 0x10de, "Nvidia" }, { 0x10ec, "Realtek" }, + { 0x19e5, "Huawei" }, { 0x1102, "Creative" }, { 0x1106, "VIA" }, { 0x111d, "IDT" }, diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index f3ad454b3fbf..cd1281691767 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -443,6 +443,7 @@ static void alc_fill_eapd_coef(struct hda_codec *codec) case 0x10ec0245: case 0x10ec0255: case 0x10ec0256: + case 0x19e58326: case 0x10ec0257: case 0x10ec0282: case 0x10ec0283: @@ -580,6 +581,7 @@ static void alc_shutup_pins(struct hda_codec *codec) switch (codec->core.vendor_id) { case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: case 0x10ec0283: case 0x10ec0286: case 0x10ec0288: @@ -3247,6 +3249,7 @@ static void alc_disable_headset_jack_key(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_write_coef_idx(codec, 0x48, 0x0); alc_update_coef_idx(codec, 0x49, 0x0045, 0x0); break; @@ -3275,6 +3278,7 @@ static void alc_enable_headset_jack_key(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_write_coef_idx(codec, 0x48, 0xd011); alc_update_coef_idx(codec, 0x49, 0x007f, 0x0045); break; @@ -4910,6 +4914,7 @@ static void alc_headset_mode_unplugged(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_process_coef_fw(codec, coef0256); break; case 0x10ec0234: @@ -5025,6 +5030,7 @@ static void alc_headset_mode_mic_in(struct hda_codec *codec, hda_nid_t hp_pin, case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_write_coef_idx(codec, 0x45, 0xc489); snd_hda_set_pin_ctl_cache(codec, hp_pin, 0); alc_process_coef_fw(codec, coef0256); @@ -5175,6 +5181,7 @@ static void alc_headset_mode_default(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_write_coef_idx(codec, 0x1b, 0x0e4b); alc_write_coef_idx(codec, 0x45, 0xc089); msleep(50); @@ -5274,6 +5281,7 @@ static void alc_headset_mode_ctia(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_process_coef_fw(codec, coef0256); break; case 0x10ec0234: @@ -5388,6 +5396,7 @@ static void alc_headset_mode_omtp(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_process_coef_fw(codec, coef0256); break; case 0x10ec0234: @@ -5489,6 +5498,7 @@ static void alc_determine_headset_type(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_write_coef_idx(codec, 0x1b, 0x0e4b); alc_write_coef_idx(codec, 0x06, 0x6104); alc_write_coefex_idx(codec, 0x57, 0x3, 0x09a3); @@ -5783,6 +5793,7 @@ static void alc255_set_default_jack_type(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_process_coef_fw(codec, alc256fw); break; } @@ -6385,6 +6396,7 @@ static void alc_combo_jack_hp_jd_restart(struct hda_codec *codec) case 0x10ec0236: case 0x10ec0255: case 0x10ec0256: + case 0x19e58326: alc_update_coef_idx(codec, 0x1b, 0x8000, 1 << 15); /* Reset HP JD */ alc_update_coef_idx(codec, 0x1b, 0x8000, 0 << 15); break; @@ -10095,6 +10107,7 @@ static int patch_alc269(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: spec->codec_variant = ALC269_TYPE_ALC256; spec->shutup = alc256_shutup; spec->init_hook = alc256_init; @@ -11545,6 +11558,7 @@ static const struct hda_device_id snd_hda_id_realtek[] = { HDA_CODEC_ENTRY(0x10ec0b00, "ALCS1200A", patch_alc882), HDA_CODEC_ENTRY(0x10ec1168, "ALC1220", patch_alc882), HDA_CODEC_ENTRY(0x10ec1220, "ALC1220", patch_alc882), + HDA_CODEC_ENTRY(0x19e58326, "HW8326", patch_alc269), {} /* terminator */ }; MODULE_DEVICE_TABLE(hdaudio, snd_hda_id_realtek); -- 2.20.1

2 1

[PATCH 0/2] Add software clock gating requirement check
by Srinivasa Rao Mandadapu 08 Jun '22

08 Jun '22

This patch set is to add software clock gating requirement check Srinivasa Rao Mandadapu (2): soundwire: qcom: Add flag for software clock gating check ASoC: qcom: soundwire: Add software clock gating requirement check drivers/soundwire/qcom.c | 28 +++++++++++++++++++++------- 1 file changed, 21 insertions(+), 7 deletions(-) -- 2.7.4

2 3

[PATCH 0/6] ASoC: ak4613: add TDM256 test support
by Kuninori Morimoto 08 Jun '22

08 Jun '22

Renesas is the only user of ak4613 driver on upstream for now. It has STEREO/TDM512/TDM256/TDM128 mode, but STEREO only is used, because of Renesas board connection. I noticed that I can test first 2ch out of TDM256 mode 8ch Playback even in such a situation. [PATCH 1/6] - [PATCH 5/6] are prepare for TDM mode support. [PATCH 6/6] uses "ifdef style" to enable TDM because I can't test full TDM256 and/or other TDM mode. "ifdef style" has no effect to current supported STEREO mode. Kuninori Morimoto (6): 1) ASoC: ak4613: add missing mutex_lock() 2) ASoC: ak4613: tidyup ak4613_interface 3) ASoC: ak4613: return error if it was setup as clock provider 4) ASoC: ak4613: priv has ctrl1 instead of iface 5) ASoC: ak4613: rename constraint to constraint_rates 6) ASoC: ak4613: add TDM256 support sound/soc/codecs/ak4613.c | 367 +++++++++++++++++++++++++++++++------- 1 file changed, 300 insertions(+), 67 deletions(-) -- 2.25.1

3 9

[PATCH] ASoC: samsung: Fix error handling in aries_audio_probe
by Miaoqian Lin 07 Jun '22

07 Jun '22

of_get_child_by_name() returns a node pointer with refcount incremented, we should use of_node_put() on it when not need anymore. This function is missing of_node_put(cpu) in the error path. Fix this by goto out label. of_node_put() will check NULL pointer. Fixes: format:7a3a7671fa6c ("ASoC: samsung: Add driver for Aries boards") Signed-off-by: Miaoqian Lin <linmq006(a)gmail.com> --- sound/soc/samsung/aries_wm8994.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/samsung/aries_wm8994.c b/sound/soc/samsung/aries_wm8994.c index bb0cf4244e00..edee02d7f100 100644 --- a/sound/soc/samsung/aries_wm8994.c +++ b/sound/soc/samsung/aries_wm8994.c @@ -628,8 +628,10 @@ static int aries_audio_probe(struct platform_device *pdev) return -EINVAL; codec = of_get_child_by_name(dev->of_node, "codec"); - if (!codec) - return -EINVAL; + if (!codec) { + ret = -EINVAL; + goto out; + } for_each_card_prelinks(card, i, dai_link) { dai_link->codecs->of_node = of_parse_phandle(codec, -- 2.25.1

2 1

[PATCH] ASoC: topology: Avoid card NULL deref in snd_soc_tplg_component_remove()
by Dean Gehnert 07 Jun '22

07 Jun '22

Don't deference card in comp->card->snd_card before checking for NULL card. During the unloading of ASoC kernel modules, there is a kernel oops in snd_soc_tplg_component_remove() that happens because comp->card is set to NULL in soc_cleanup_component(). Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: alsa-devel(a)alsa-project.org Cc: linux-kernel(a)vger.kernel.org Cc: stable(a)vger.kernel.org Fixes: 7e567b5ae063 ("ASoC: topology: Add missing rwsem around snd_ctl_remove() calls") Signed-off-by: Dean Gehnert <deang(a)tpi.com> --- sound/soc/soc-topology.c | 10 +++++++--- 1 file changed, 7 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 3f9d314fba16..cf0efe1147c2 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -2613,15 +2613,18 @@ EXPORT_SYMBOL_GPL(snd_soc_tplg_component_load); /* remove dynamic controls from the component driver */ int snd_soc_tplg_component_remove(struct snd_soc_component *comp) { - struct snd_card *card = comp->card->snd_card; + struct snd_card *card; struct snd_soc_dobj *dobj, *next_dobj; int pass; /* process the header types from end to start */ for (pass = SOC_TPLG_PASS_END; pass >= SOC_TPLG_PASS_START; pass--) { + card = (comp->card) ? comp->card->snd_card : NULL; + /* remove mixer controls */ - down_write(&card->controls_rwsem); + if (card) + down_write(&card->controls_rwsem); list_for_each_entry_safe(dobj, next_dobj, &comp->dobj_list, list) { @@ -2660,7 +2663,8 @@ int snd_soc_tplg_component_remove(struct snd_soc_component *comp) break; } } - up_write(&card->controls_rwsem); + if (card) + up_write(&card->controls_rwsem); } /* let caller know if FW can be freed when no objects are left */ -- 2.17.1

2 2

[PATCH AUTOSEL 5.15 19/51] soundwire: qcom: adjust autoenumeration timeout
by Sasha Levin 07 Jun '22

07 Jun '22

From: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> [ Upstream commit 74da272400b46f2e898f115d1b1cd60828766919 ] Currently timeout for autoenumeration during probe and bus reset is set to 2 secs which is really a big value. This can have an adverse effect on boot time if the slave device is not ready/reset. This was the case with wcd938x which was not reset yet but we spent 2 secs waiting in the soundwire controller probe. Reduce this time to 1/10 of Hz which should be good enough time to finish autoenumeration if any slaves are available on the bus. Reported-by: Srinivasa Rao Mandadapu <quic_srivasam(a)quicinc.com> Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Link: https://lore.kernel.org/r/20220506084705.18525-1-srinivas.kandagatla@linaro… Signed-off-by: Vinod Koul <vkoul(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- drivers/soundwire/qcom.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/drivers/soundwire/qcom.c b/drivers/soundwire/qcom.c index 0ef79d60e88e..f5955826b152 100644 --- a/drivers/soundwire/qcom.c +++ b/drivers/soundwire/qcom.c @@ -97,7 +97,7 @@ #define SWRM_SPECIAL_CMD_ID 0xF #define MAX_FREQ_NUM 1 -#define TIMEOUT_MS (2 * HZ) +#define TIMEOUT_MS 100 #define QCOM_SWRM_MAX_RD_LEN 0x1 #define QCOM_SDW_MAX_PORTS 14 #define DEFAULT_CLK_FREQ 9600000 -- 2.35.1

1 0

[PATCH AUTOSEL 5.17 36/60] ASoC: rt5640: Do not manipulate pin "Platform Clock" if the "Platform Clock" is not in the DAPM
by Sasha Levin 07 Jun '22

07 Jun '22

From: Oder Chiou <oder_chiou(a)realtek.com> [ Upstream commit 832296804bc7171730884e78c761c29f6d258e13 ] The pin "Platform Clock" was only used by the Intel Byt CR platform. In the others, the error log will be informed. The patch will set the flag to avoid the pin "Platform Clock" manipulated by the other platforms. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> Reported-by: Sameer Pujar <spujar(a)nvidia.com> Link: https://lore.kernel.org/r/20220516103055.20003-1-oder_chiou@realtek.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt5640.c | 11 +++++++++-- sound/soc/codecs/rt5640.h | 2 ++ sound/soc/intel/boards/bytcr_rt5640.c | 2 ++ 3 files changed, 13 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index e7a82565b905..f078463346e8 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -2097,12 +2097,14 @@ EXPORT_SYMBOL_GPL(rt5640_sel_asrc_clk_src); void rt5640_enable_micbias1_for_ovcd(struct snd_soc_component *component) { struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component); + struct rt5640_priv *rt5640 = snd_soc_component_get_drvdata(component); snd_soc_dapm_mutex_lock(dapm); snd_soc_dapm_force_enable_pin_unlocked(dapm, "LDO2"); snd_soc_dapm_force_enable_pin_unlocked(dapm, "MICBIAS1"); /* OVCD is unreliable when used with RCCLK as sysclk-source */ - snd_soc_dapm_force_enable_pin_unlocked(dapm, "Platform Clock"); + if (rt5640->use_platform_clock) + snd_soc_dapm_force_enable_pin_unlocked(dapm, "Platform Clock"); snd_soc_dapm_sync_unlocked(dapm); snd_soc_dapm_mutex_unlock(dapm); } @@ -2111,9 +2113,11 @@ EXPORT_SYMBOL_GPL(rt5640_enable_micbias1_for_ovcd); void rt5640_disable_micbias1_for_ovcd(struct snd_soc_component *component) { struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component); + struct rt5640_priv *rt5640 = snd_soc_component_get_drvdata(component); snd_soc_dapm_mutex_lock(dapm); - snd_soc_dapm_disable_pin_unlocked(dapm, "Platform Clock"); + if (rt5640->use_platform_clock) + snd_soc_dapm_disable_pin_unlocked(dapm, "Platform Clock"); snd_soc_dapm_disable_pin_unlocked(dapm, "MICBIAS1"); snd_soc_dapm_disable_pin_unlocked(dapm, "LDO2"); snd_soc_dapm_sync_unlocked(dapm); @@ -2538,6 +2542,9 @@ static void rt5640_enable_jack_detect(struct snd_soc_component *component, rt5640->jd_gpio_irq_requested = true; } + if (jack_data && jack_data->use_platform_clock) + rt5640->use_platform_clock = jack_data->use_platform_clock; + ret = request_irq(rt5640->irq, rt5640_irq, IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING | IRQF_ONESHOT, "rt5640", rt5640); diff --git a/sound/soc/codecs/rt5640.h b/sound/soc/codecs/rt5640.h index 9e49b9a0ccaa..505c93514051 100644 --- a/sound/soc/codecs/rt5640.h +++ b/sound/soc/codecs/rt5640.h @@ -2155,11 +2155,13 @@ struct rt5640_priv { bool jd_inverted; unsigned int ovcd_th; unsigned int ovcd_sf; + bool use_platform_clock; }; struct rt5640_set_jack_data { int codec_irq_override; struct gpio_desc *jd_gpio; + bool use_platform_clock; }; int rt5640_dmic_enable(struct snd_soc_component *component, diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 2ace32c03ec9..cc5703ecf523 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -1179,12 +1179,14 @@ static int byt_rt5640_init(struct snd_soc_pcm_runtime *runtime) { struct snd_soc_card *card = runtime->card; struct byt_rt5640_private *priv = snd_soc_card_get_drvdata(card); + struct rt5640_set_jack_data *jack_data = &priv->jack_data; struct snd_soc_component *component = asoc_rtd_to_codec(runtime, 0)->component; const struct snd_soc_dapm_route *custom_map = NULL; int num_routes = 0; int ret; card->dapm.idle_bias_off = true; + jack_data->use_platform_clock = true; /* Start with RC clk for jack-detect (we disable MCLK below) */ if (byt_rt5640_quirk & BYT_RT5640_MCLK_EN) -- 2.35.1

1 0

[PATCH AUTOSEL 5.17 20/60] soundwire: qcom: adjust autoenumeration timeout
by Sasha Levin 07 Jun '22

07 Jun '22

From: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> [ Upstream commit 74da272400b46f2e898f115d1b1cd60828766919 ] Currently timeout for autoenumeration during probe and bus reset is set to 2 secs which is really a big value. This can have an adverse effect on boot time if the slave device is not ready/reset. This was the case with wcd938x which was not reset yet but we spent 2 secs waiting in the soundwire controller probe. Reduce this time to 1/10 of Hz which should be good enough time to finish autoenumeration if any slaves are available on the bus. Reported-by: Srinivasa Rao Mandadapu <quic_srivasam(a)quicinc.com> Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Link: https://lore.kernel.org/r/20220506084705.18525-1-srinivas.kandagatla@linaro… Signed-off-by: Vinod Koul <vkoul(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- drivers/soundwire/qcom.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/drivers/soundwire/qcom.c b/drivers/soundwire/qcom.c index 54813417ef8e..249f1b69ec94 100644 --- a/drivers/soundwire/qcom.c +++ b/drivers/soundwire/qcom.c @@ -99,7 +99,7 @@ #define SWRM_SPECIAL_CMD_ID 0xF #define MAX_FREQ_NUM 1 -#define TIMEOUT_MS (2 * HZ) +#define TIMEOUT_MS 100 #define QCOM_SWRM_MAX_RD_LEN 0x1 #define QCOM_SDW_MAX_PORTS 14 #define DEFAULT_CLK_FREQ 9600000 -- 2.35.1

1 0

[PATCH AUTOSEL 5.18 39/68] ASoC: rt5640: Do not manipulate pin "Platform Clock" if the "Platform Clock" is not in the DAPM
by Sasha Levin 07 Jun '22

07 Jun '22

From: Oder Chiou <oder_chiou(a)realtek.com> [ Upstream commit 832296804bc7171730884e78c761c29f6d258e13 ] The pin "Platform Clock" was only used by the Intel Byt CR platform. In the others, the error log will be informed. The patch will set the flag to avoid the pin "Platform Clock" manipulated by the other platforms. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> Reported-by: Sameer Pujar <spujar(a)nvidia.com> Link: https://lore.kernel.org/r/20220516103055.20003-1-oder_chiou@realtek.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt5640.c | 11 +++++++++-- sound/soc/codecs/rt5640.h | 2 ++ sound/soc/intel/boards/bytcr_rt5640.c | 2 ++ 3 files changed, 13 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 30c2e7cb7ed2..3559d9ecfa07 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -2094,12 +2094,14 @@ EXPORT_SYMBOL_GPL(rt5640_sel_asrc_clk_src); void rt5640_enable_micbias1_for_ovcd(struct snd_soc_component *component) { struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component); + struct rt5640_priv *rt5640 = snd_soc_component_get_drvdata(component); snd_soc_dapm_mutex_lock(dapm); snd_soc_dapm_force_enable_pin_unlocked(dapm, "LDO2"); snd_soc_dapm_force_enable_pin_unlocked(dapm, "MICBIAS1"); /* OVCD is unreliable when used with RCCLK as sysclk-source */ - snd_soc_dapm_force_enable_pin_unlocked(dapm, "Platform Clock"); + if (rt5640->use_platform_clock) + snd_soc_dapm_force_enable_pin_unlocked(dapm, "Platform Clock"); snd_soc_dapm_sync_unlocked(dapm); snd_soc_dapm_mutex_unlock(dapm); } @@ -2108,9 +2110,11 @@ EXPORT_SYMBOL_GPL(rt5640_enable_micbias1_for_ovcd); void rt5640_disable_micbias1_for_ovcd(struct snd_soc_component *component) { struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component); + struct rt5640_priv *rt5640 = snd_soc_component_get_drvdata(component); snd_soc_dapm_mutex_lock(dapm); - snd_soc_dapm_disable_pin_unlocked(dapm, "Platform Clock"); + if (rt5640->use_platform_clock) + snd_soc_dapm_disable_pin_unlocked(dapm, "Platform Clock"); snd_soc_dapm_disable_pin_unlocked(dapm, "MICBIAS1"); snd_soc_dapm_disable_pin_unlocked(dapm, "LDO2"); snd_soc_dapm_sync_unlocked(dapm); @@ -2535,6 +2539,9 @@ static void rt5640_enable_jack_detect(struct snd_soc_component *component, rt5640->jd_gpio_irq_requested = true; } + if (jack_data && jack_data->use_platform_clock) + rt5640->use_platform_clock = jack_data->use_platform_clock; + ret = request_irq(rt5640->irq, rt5640_irq, IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING | IRQF_ONESHOT, "rt5640", rt5640); diff --git a/sound/soc/codecs/rt5640.h b/sound/soc/codecs/rt5640.h index 9e49b9a0ccaa..505c93514051 100644 --- a/sound/soc/codecs/rt5640.h +++ b/sound/soc/codecs/rt5640.h @@ -2155,11 +2155,13 @@ struct rt5640_priv { bool jd_inverted; unsigned int ovcd_th; unsigned int ovcd_sf; + bool use_platform_clock; }; struct rt5640_set_jack_data { int codec_irq_override; struct gpio_desc *jd_gpio; + bool use_platform_clock; }; int rt5640_dmic_enable(struct snd_soc_component *component, diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index d76a505052fb..96a2ea92dd1e 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -1179,12 +1179,14 @@ static int byt_rt5640_init(struct snd_soc_pcm_runtime *runtime) { struct snd_soc_card *card = runtime->card; struct byt_rt5640_private *priv = snd_soc_card_get_drvdata(card); + struct rt5640_set_jack_data *jack_data = &priv->jack_data; struct snd_soc_component *component = asoc_rtd_to_codec(runtime, 0)->component; const struct snd_soc_dapm_route *custom_map = NULL; int num_routes = 0; int ret; card->dapm.idle_bias_off = true; + jack_data->use_platform_clock = true; /* Start with RC clk for jack-detect (we disable MCLK below) */ if (byt_rt5640_quirk & BYT_RT5640_MCLK_EN) -- 2.35.1

1 0

[PATCH AUTOSEL 5.18 37/68] ASoC: SOF: amd: Fixed Build error
by Sasha Levin 07 Jun '22

07 Jun '22

From: Syed Saba kareem <ssabakar(a)amd.com> [ Upstream commit 803a1f7272797faa15a7879cdc70f9adaf3fdcba ] Add linux/module.h in acp-pci.c to solve the below dependency All error/warnings (new ones prefixed by >>): >> sound/soc/amd/acp/acp-pci.c:148:1: warning: data definition has no type or storage class 148 | MODULE_DEVICE_TABLE(pci, acp_pci_ids); | ^~~~~~~~~~~~~~~~~~~ >> sound/soc/amd/acp/acp-pci.c:148:1: error: type defaults to 'int' in declaration of 'MODULE_DEVICE_TABLE' [-Werror=implicit-int] ... Reported-by: kernel test robot <lkp(a)intel.com> Signed-off-by: Syed Saba Kareem<ssabakar(a)amd.com> Link: https://lore.kernel.org/r/20220523112956.3087604-1-ssabakar@amd.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/amd/acp/acp-pci.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/amd/acp/acp-pci.c b/sound/soc/amd/acp/acp-pci.c index 340e39d7f420..c893963ee2d0 100644 --- a/sound/soc/amd/acp/acp-pci.c +++ b/sound/soc/amd/acp/acp-pci.c @@ -16,6 +16,7 @@ #include <linux/pci.h> #include <linux/platform_device.h> #include <linux/pm_runtime.h> +#include <linux/module.h> #include "amd.h" #include "../mach-config.h" -- 2.35.1

1 0

[PATCH AUTOSEL 5.18 22/68] soundwire: qcom: adjust autoenumeration timeout
by Sasha Levin 07 Jun '22

07 Jun '22

From: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> [ Upstream commit 74da272400b46f2e898f115d1b1cd60828766919 ] Currently timeout for autoenumeration during probe and bus reset is set to 2 secs which is really a big value. This can have an adverse effect on boot time if the slave device is not ready/reset. This was the case with wcd938x which was not reset yet but we spent 2 secs waiting in the soundwire controller probe. Reduce this time to 1/10 of Hz which should be good enough time to finish autoenumeration if any slaves are available on the bus. Reported-by: Srinivasa Rao Mandadapu <quic_srivasam(a)quicinc.com> Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Link: https://lore.kernel.org/r/20220506084705.18525-1-srinivas.kandagatla@linaro… Signed-off-by: Vinod Koul <vkoul(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- drivers/soundwire/qcom.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/drivers/soundwire/qcom.c b/drivers/soundwire/qcom.c index da1ad7ebb1aa..22947e8297ac 100644 --- a/drivers/soundwire/qcom.c +++ b/drivers/soundwire/qcom.c @@ -105,7 +105,7 @@ #define SWRM_SPECIAL_CMD_ID 0xF #define MAX_FREQ_NUM 1 -#define TIMEOUT_MS (2 * HZ) +#define TIMEOUT_MS 100 #define QCOM_SWRM_MAX_RD_LEN 0x1 #define QCOM_SDW_MAX_PORTS 14 #define DEFAULT_CLK_FREQ 9600000 -- 2.35.1

1 0

[PATCH v2] ASoC: dt-bindings: renesas,rz-ssi: Document RZ/G2UL SoC
by Biju Das 07 Jun '22

07 Jun '22

Document RZ/G2U2L SSI bindings. RZ/G2UL SSI is identical to one found on the RZ/G2L SoC. No driver changes are required as generic compatible string "renesas,rz-ssi" will be used as a fallback. While at it add a '.' at the end of dmas description for the first cell. Signed-off-by: Biju Das <biju.das.jz(a)bp.renesas.com> Acked-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> Reviewed-by: Geert Uytterhoeven <geert+renesas(a)glider.be> --- v1->v2: * Rebased to latest sound.git * Added Ack tag from Krzysztof Kozlowski * Added Rb tag from Geert. --- Documentation/devicetree/bindings/sound/renesas,rz-ssi.yaml | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/renesas,rz-ssi.yaml b/Documentation/devicetree/bindings/sound/renesas,rz-ssi.yaml index 7e8d252f7bca..0d9840375132 100644 --- a/Documentation/devicetree/bindings/sound/renesas,rz-ssi.yaml +++ b/Documentation/devicetree/bindings/sound/renesas,rz-ssi.yaml @@ -13,6 +13,7 @@ properties: compatible: items: - enum: + - renesas,r9a07g043-ssi # RZ/G2UL - renesas,r9a07g044-ssi # RZ/G2{L,LC} - renesas,r9a07g054-ssi # RZ/V2L - const: renesas,rz-ssi @@ -50,7 +51,7 @@ properties: minItems: 1 maxItems: 2 description: - The first cell represents a phandle to dmac + The first cell represents a phandle to dmac. The second cell specifies the encoded MID/RID values of the SSI port connected to the DMA client and the slave channel configuration parameters. -- 2.25.1

2 1

[syzbot] KASAN: use-after-free Write in snd_rawmidi_kernel_write1
by syzbot 07 Jun '22

07 Jun '22

Hello, syzbot found the following issue on: HEAD commit: 952923ddc011 Merge tag 'pull-18-rc1-work.namei' of git://g.. git tree: upstream console output: https://syzkaller.appspot.com/x/log.txt?x=158076dbf00000 kernel config: https://syzkaller.appspot.com/x/.config?x=3096247591885bfa dashboard link: https://syzkaller.appspot.com/bug?extid=8569b906285a4215278d compiler: gcc (Debian 10.2.1-6) 10.2.1 20210110, GNU ld (GNU Binutils for Debian) 2.35.2 Unfortunately, I don't have any reproducer for this issue yet. IMPORTANT: if you fix the issue, please add the following tag to the commit: Reported-by: syzbot+8569b906285a4215278d(a)syzkaller.appspotmail.com ================================================================== ================================================================== BUG: KASAN: use-after-free in instrument_copy_from_user include/linux/instrumented.h:135 [inline] BUG: KASAN: use-after-free in _copy_from_user+0x12a/0x170 lib/usercopy.c:15 Write of size 4096 at addr ffff8880280e2000 by task syz-executor.2/8284 CPU: 1 PID: 8284 Comm: syz-executor.2 Not tainted 5.18.0-syzkaller-13842-g952923ddc011 #0 Hardware name: Google Google Compute Engine/Google Compute Engine, BIOS Google 01/01/2011 Call Trace: <TASK> __dump_stack lib/dump_stack.c:88 [inline] dump_stack_lvl+0xcd/0x134 lib/dump_stack.c:106 print_address_description.constprop.0.cold+0xeb/0x495 mm/kasan/report.c:313 print_report mm/kasan/report.c:429 [inline] kasan_report.cold+0xf4/0x1c6 mm/kasan/report.c:491 check_region_inline mm/kasan/generic.c:183 [inline] kasan_check_range+0x13d/0x180 mm/kasan/generic.c:189 instrument_copy_from_user include/linux/instrumented.h:135 [inline] _copy_from_user+0x12a/0x170 lib/usercopy.c:15 copy_from_user include/linux/uaccess.h:152 [inline] snd_rawmidi_kernel_write1+0x361/0x870 sound/core/rawmidi.c:1486 snd_rawmidi_write+0x273/0xb90 sound/core/rawmidi.c:1555 vfs_write+0x269/0xac0 fs/read_write.c:589 ksys_write+0x1e8/0x250 fs/read_write.c:644 do_syscall_x64 arch/x86/entry/common.c:50 [inline] do_syscall_64+0x35/0xb0 arch/x86/entry/common.c:80 entry_SYSCALL_64_after_hwframe+0x46/0xb0 RIP: 0033:0x7fe46fa89109 Code: ff ff c3 66 2e 0f 1f 84 00 00 00 00 00 0f 1f 40 00 48 89 f8 48 89 f7 48 89 d6 48 89 ca 4d 89 c2 4d 89 c8 4c 8b 4c 24 08 0f 05 <48> 3d 01 f0 ff ff 73 01 c3 48 c7 c1 b8 ff ff ff f7 d8 64 89 01 48 RSP: 002b:00007fe470c0a168 EFLAGS: 00000246 ORIG_RAX: 0000000000000001 RAX: ffffffffffffffda RBX: 00007fe46fb9bf60 RCX: 00007fe46fa89109 RDX: 00000000fffffd2c RSI: 0000000020000000 RDI: 0000000000000004 RBP: 00007fe46fae308d R08: 0000000000000000 R09: 0000000000000000 R10: 0000000000000000 R11: 0000000000000246 R12: 0000000000000000 R13: 00007fe4700cfb1f R14: 00007fe470c0a300 R15: 0000000000022000 </TASK> Allocated by task 8284: kasan_save_stack+0x1e/0x40 mm/kasan/common.c:38 kasan_set_track mm/kasan/common.c:45 [inline] set_alloc_info mm/kasan/common.c:436 [inline] ____kasan_kmalloc mm/kasan/common.c:515 [inline] ____kasan_kmalloc mm/kasan/common.c:474 [inline] __kasan_kmalloc+0xa9/0xd0 mm/kasan/common.c:524 kmalloc_node include/linux/slab.h:623 [inline] kvmalloc_node+0x3e/0x190 mm/util.c:613 kvmalloc include/linux/slab.h:750 [inline] kvzalloc include/linux/slab.h:758 [inline] snd_rawmidi_runtime_create sound/core/rawmidi.c:162 [inline] open_substream+0x340/0x8b0 sound/core/rawmidi.c:306 rawmidi_open_priv+0x591/0x6f0 sound/core/rawmidi.c:357 snd_rawmidi_open+0x49f/0xb60 sound/core/rawmidi.c:455 snd_open+0x21f/0x460 sound/core/sound.c:169 chrdev_open+0x266/0x770 fs/char_dev.c:414 do_dentry_open+0x4a1/0x11f0 fs/open.c:848 do_open fs/namei.c:3520 [inline] path_openat+0x1c71/0x2910 fs/namei.c:3653 do_filp_open+0x1aa/0x400 fs/namei.c:3680 do_sys_openat2+0x16d/0x4c0 fs/open.c:1278 do_sys_open fs/open.c:1294 [inline] __do_sys_openat fs/open.c:1310 [inline] __se_sys_openat fs/open.c:1305 [inline] __x64_sys_openat+0x13f/0x1f0 fs/open.c:1305 do_syscall_x64 arch/x86/entry/common.c:50 [inline] do_syscall_64+0x35/0xb0 arch/x86/entry/common.c:80 entry_SYSCALL_64_after_hwframe+0x46/0xb0 Freed by task 8283: kasan_save_stack+0x1e/0x40 mm/kasan/common.c:38 kasan_set_track+0x21/0x30 mm/kasan/common.c:45 kasan_set_free_info+0x20/0x30 mm/kasan/generic.c:370 ____kasan_slab_free mm/kasan/common.c:366 [inline] ____kasan_slab_free+0x166/0x1a0 mm/kasan/common.c:328 kasan_slab_free include/linux/kasan.h:200 [inline] slab_free_hook mm/slub.c:1727 [inline] slab_free_freelist_hook+0x8b/0x1c0 mm/slub.c:1753 slab_free mm/slub.c:3507 [inline] kfree+0xd6/0x4d0 mm/slub.c:4555 kvfree+0x42/0x50 mm/util.c:655 snd_rawmidi_runtime_free sound/core/rawmidi.c:176 [inline] close_substream.part.0+0x185/0x720 sound/core/rawmidi.c:528 close_substream sound/core/rawmidi.c:507 [inline] rawmidi_release_priv+0x192/0x270 sound/core/rawmidi.c:547 snd_rawmidi_release+0x5e/0xf0 sound/core/rawmidi.c:578 __fput+0x277/0x9d0 fs/file_table.c:317 task_work_run+0xdd/0x1a0 kernel/task_work.c:177 resume_user_mode_work include/linux/resume_user_mode.h:49 [inline] exit_to_user_mode_loop kernel/entry/common.c:169 [inline] exit_to_user_mode_prepare+0x23c/0x250 kernel/entry/common.c:201 __syscall_exit_to_user_mode_work kernel/entry/common.c:283 [inline] syscall_exit_to_user_mode+0x19/0x50 kernel/entry/common.c:294 do_syscall_64+0x42/0xb0 arch/x86/entry/common.c:86 entry_SYSCALL_64_after_hwframe+0x46/0xb0 The buggy address belongs to the object at ffff8880280e2000 which belongs to the cache kmalloc-4k of size 4096 The buggy address is located 0 bytes inside of 4096-byte region [ffff8880280e2000, ffff8880280e3000) The buggy address belongs to the physical page: page:ffffea0000a03800 refcount:1 mapcount:0 mapping:0000000000000000 index:0x0 pfn:0x280e0 head:ffffea0000a03800 order:3 compound_mapcount:0 compound_pincount:0 flags: 0xfff00000010200(slab|head|node=0|zone=1|lastcpupid=0x7ff) raw: 00fff00000010200 dead000000000100 dead000000000122 ffff888011842140 raw: 0000000000000000 0000000000040004 00000001ffffffff 0000000000000000 page dumped because: kasan: bad access detected page_owner tracks the page as allocated page last allocated via order 3, migratetype Unmovable, gfp_mask 0x1d2040(__GFP_IO|__GFP_NOWARN|__GFP_NORETRY|__GFP_COMP|__GFP_NOMEMALLOC|__GFP_HARDWALL), pid 3912, tgid 3912 (udevd), ts 388242276639, free_ts 388232231340 prep_new_page mm/page_alloc.c:2456 [inline] get_page_from_freelist+0x1290/0x3b70 mm/page_alloc.c:4198 __alloc_pages+0x1c7/0x510 mm/page_alloc.c:5426 alloc_pages+0x1aa/0x310 mm/mempolicy.c:2272 alloc_slab_page mm/slub.c:1797 [inline] allocate_slab+0x26c/0x3c0 mm/slub.c:1942 new_slab mm/slub.c:2002 [inline] ___slab_alloc+0x985/0xd90 mm/slub.c:3002 __slab_alloc.constprop.0+0x4d/0xa0 mm/slub.c:3089 slab_alloc_node mm/slub.c:3180 [inline] slab_alloc mm/slub.c:3222 [inline] __kmalloc+0x318/0x350 mm/slub.c:4413 kmalloc include/linux/slab.h:605 [inline] tomoyo_realpath_from_path+0xc3/0x620 security/tomoyo/realpath.c:254 tomoyo_get_realpath security/tomoyo/file.c:151 [inline] tomoyo_check_open_permission+0x272/0x380 security/tomoyo/file.c:771 tomoyo_file_open security/tomoyo/tomoyo.c:320 [inline] tomoyo_file_open+0x9d/0xc0 security/tomoyo/tomoyo.c:315 security_file_open+0x45/0xb0 security/security.c:1645 do_dentry_open+0x349/0x11f0 fs/open.c:835 do_open fs/namei.c:3520 [inline] path_openat+0x1c71/0x2910 fs/namei.c:3653 do_filp_open+0x1aa/0x400 fs/namei.c:3680 do_sys_openat2+0x16d/0x4c0 fs/open.c:1278 do_sys_open fs/open.c:1294 [inline] __do_sys_openat fs/open.c:1310 [inline] __se_sys_openat fs/open.c:1305 [inline] __x64_sys_openat+0x13f/0x1f0 fs/open.c:1305 page last free stack trace: reset_page_owner include/linux/page_owner.h:24 [inline] free_pages_prepare mm/page_alloc.c:1371 [inline] free_pcp_prepare+0x549/0xd20 mm/page_alloc.c:1421 free_unref_page_prepare mm/page_alloc.c:3343 [inline] free_unref_page+0x19/0x6a0 mm/page_alloc.c:3438 __unfreeze_partials+0x17c/0x1a0 mm/slub.c:2521 qlink_free mm/kasan/quarantine.c:168 [inline] qlist_free_all+0x6a/0x170 mm/kasan/quarantine.c:187 kasan_quarantine_reduce+0x180/0x200 mm/kasan/quarantine.c:294 __kasan_slab_alloc+0xa2/0xc0 mm/kasan/common.c:446 kasan_slab_alloc include/linux/kasan.h:224 [inline] slab_post_alloc_hook mm/slab.h:750 [inline] slab_alloc_node mm/slub.c:3214 [inline] slab_alloc mm/slub.c:3222 [inline] __kmalloc+0x200/0x350 mm/slub.c:4413 kmalloc include/linux/slab.h:605 [inline] kzalloc include/linux/slab.h:733 [inline] tomoyo_encode2.part.0+0xe9/0x3a0 security/tomoyo/realpath.c:45 tomoyo_encode2 security/tomoyo/realpath.c:31 [inline] tomoyo_encode+0x28/0x50 security/tomoyo/realpath.c:80 tomoyo_realpath_from_path+0x186/0x620 security/tomoyo/realpath.c:288 tomoyo_get_realpath security/tomoyo/file.c:151 [inline] tomoyo_check_open_permission+0x272/0x380 security/tomoyo/file.c:771 tomoyo_file_open security/tomoyo/tomoyo.c:320 [inline] tomoyo_file_open+0x9d/0xc0 security/tomoyo/tomoyo.c:315 security_file_open+0x45/0xb0 security/security.c:1645 do_dentry_open+0x349/0x11f0 fs/open.c:835 do_open fs/namei.c:3520 [inline] path_openat+0x1c71/0x2910 fs/namei.c:3653 do_filp_open+0x1aa/0x400 fs/namei.c:3680 Memory state around the buggy address: ffff8880280e1f00: fc fc fc fc fc fc fc fc fc fc fc fc fc fc fc fc ffff8880280e1f80: fc fc fc fc fc fc fc fc fc fc fc fc fc fc fc fc >ffff8880280e2000: fa fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb ^ ffff8880280e2080: fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb ffff8880280e2100: fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb ================================================================== --- This report is generated by a bot. It may contain errors. See https://goo.gl/tpsmEJ for more information about syzbot. syzbot engineers can be reached at syzkaller(a)googlegroups.com. syzbot will keep track of this issue. See: https://goo.gl/tpsmEJ#status for how to communicate with syzbot.

1 0

[PATCH 0/3] ASoC: ssm: Use modern ASoC DAI format terminology
by Mark Brown 07 Jun '22

07 Jun '22

Trivial updates to modernise the various ssm* drivers to the new DAI clocking constants. Mark Brown (3): ASoC: ssm2518: Use modern ASoC DAI format terminology ASoC: ssm2602: Use modern ASoC DAI format terminology ASoC: ssm4567: Use modern ASoC DAI format terminology sound/soc/codecs/ssm2518.c | 4 ++-- sound/soc/codecs/ssm2602.c | 6 +++--- sound/soc/codecs/ssm4567.c | 4 ++-- 3 files changed, 7 insertions(+), 7 deletions(-) -- 2.30.2

1 4

[PATCH 0/5] ASoC: SOF: AMD/Mediatek updates for 5.20
by Pierre-Louis Bossart 07 Jun '22

07 Jun '22

Small patches reviewed on SOF GitHub. Ajit Kumar Pandey (1): ASoC: SOF: amd: Add SOF pm ops callback for Renoir YC Hung (4): ASoC: SOF: mediatek: revise mt8195 clock sequence ASoC: SOF: mediatek: Add shared_size for mediatek common chip information ASoC: SOF: mediatek: mt8195 modify dram type as non-cache ASoC: SOF: mediatek: mt8195 suspend check dsp idle sound/soc/sof/amd/acp-dsp-offset.h | 2 + sound/soc/sof/amd/acp.c | 36 ++++++++++++++++ sound/soc/sof/amd/acp.h | 4 ++ sound/soc/sof/amd/pci-rn.c | 4 ++ sound/soc/sof/amd/renoir.c | 4 ++ sound/soc/sof/mediatek/adsp_helper.h | 1 + sound/soc/sof/mediatek/mt8195/mt8195-clk.c | 7 +++ sound/soc/sof/mediatek/mt8195/mt8195.c | 50 +++++++++++++++------- sound/soc/sof/mediatek/mt8195/mt8195.h | 5 +++ 9 files changed, 97 insertions(+), 16 deletions(-) -- 2.34.1

2 6

[PATCH v2] ASoC: mediatek: mt8173-rt5650: Fix refcount leak in mt8173_rt5650_dev_probe
by Miaoqian Lin 07 Jun '22

07 Jun '22

of_parse_phandle() returns a node pointer with refcount incremented, we should use of_node_put() on it when not need anymore. Fix refcount leak in some error paths. Fixes: 0f83f9296d5c ("ASoC: mediatek: Add machine driver for ALC5650 codec") Signed-off-by: Miaoqian Lin <linmq006(a)gmail.com> --- - update fixes tag. v1 Link: https://lore.kernel.org/r/20220603081308.1332-1-linmq006@gmail.com --- sound/soc/mediatek/mt8173/mt8173-rt5650.c | 9 ++++++--- 1 file changed, 6 insertions(+), 3 deletions(-) diff --git a/sound/soc/mediatek/mt8173/mt8173-rt5650.c b/sound/soc/mediatek/mt8173/mt8173-rt5650.c index d1c94acb4516..e05f2b0231fe 100644 --- a/sound/soc/mediatek/mt8173/mt8173-rt5650.c +++ b/sound/soc/mediatek/mt8173/mt8173-rt5650.c @@ -280,7 +280,8 @@ static int mt8173_rt5650_dev_probe(struct platform_device *pdev) if (!mt8173_rt5650_dais[DAI_LINK_CODEC_I2S].codecs[0].of_node) { dev_err(&pdev->dev, "Property 'audio-codec' missing or invalid\n"); - return -EINVAL; + ret = -EINVAL; + goto put_platform_node; } mt8173_rt5650_dais[DAI_LINK_CODEC_I2S].codecs[1].of_node = mt8173_rt5650_dais[DAI_LINK_CODEC_I2S].codecs[0].of_node; @@ -293,7 +294,7 @@ static int mt8173_rt5650_dev_probe(struct platform_device *pdev) dev_err(&pdev->dev, "%s codec_capture_dai name fail %d\n", __func__, ret); - return ret; + goto put_platform_node; } mt8173_rt5650_dais[DAI_LINK_CODEC_I2S].codecs[1].dai_name = codec_capture_dai; @@ -315,12 +316,14 @@ static int mt8173_rt5650_dev_probe(struct platform_device *pdev) if (!mt8173_rt5650_dais[DAI_LINK_HDMI_I2S].codecs->of_node) { dev_err(&pdev->dev, "Property 'audio-codec' missing or invalid\n"); - return -EINVAL; + ret = -EINVAL; + goto put_platform_node; } card->dev = &pdev->dev; ret = devm_snd_soc_register_card(&pdev->dev, card); +put_platform_node: of_node_put(platform_node); return ret; } -- 2.25.1

2 1

[PATCH] ASoC: simple-card-utils: Make asoc_simple_clean_reference() return void
by Uwe Kleine-König 07 Jun '22

07 Jun '22

asoc_simple_clean_reference() returns zero unconditionally. Letting it return void instead makes it easier to see in the caller that there is no error to handle. This is a preparation for making platform remove callbacks return void. Signed-off-by: Uwe Kleine-König <u.kleine-koenig(a)pengutronix.de> --- include/sound/simple_card_utils.h | 2 +- sound/soc/generic/simple-card-utils.c | 7 ++++--- 2 files changed, 5 insertions(+), 4 deletions(-) diff --git a/include/sound/simple_card_utils.h b/include/sound/simple_card_utils.h index 8faa649f712b..fe2337fde1f4 100644 --- a/include/sound/simple_card_utils.h +++ b/include/sound/simple_card_utils.h @@ -173,7 +173,7 @@ void asoc_simple_canonicalize_platform(struct snd_soc_dai_link_component *platfo void asoc_simple_canonicalize_cpu(struct snd_soc_dai_link_component *cpus, int is_single_links); -int asoc_simple_clean_reference(struct snd_soc_card *card); +void asoc_simple_clean_reference(struct snd_soc_card *card); void asoc_simple_convert_fixup(struct asoc_simple_data *data, struct snd_pcm_hw_params *params); diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index da0c27828ce6..a542dec92ad5 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -609,7 +609,7 @@ void asoc_simple_canonicalize_cpu(struct snd_soc_dai_link_component *cpus, } EXPORT_SYMBOL_GPL(asoc_simple_canonicalize_cpu); -int asoc_simple_clean_reference(struct snd_soc_card *card) +void asoc_simple_clean_reference(struct snd_soc_card *card) { struct snd_soc_dai_link *dai_link; struct snd_soc_dai_link_component *cpu; @@ -622,7 +622,6 @@ int asoc_simple_clean_reference(struct snd_soc_card *card) for_each_link_codecs(dai_link, j, codec) of_node_put(codec->of_node); } - return 0; } EXPORT_SYMBOL_GPL(asoc_simple_clean_reference); @@ -878,7 +877,9 @@ int asoc_simple_remove(struct platform_device *pdev) { struct snd_soc_card *card = platform_get_drvdata(pdev); - return asoc_simple_clean_reference(card); + asoc_simple_clean_reference(card); + + return 0; } EXPORT_SYMBOL_GPL(asoc_simple_remove); base-commit: 4b0986a3613c92f4ec1bdc7f60ec66fea135991f -- 2.36.1

2 1

[PATCH] ASoC: Intel: broadwell: Make broadwell_disable_jack() return void
by Uwe Kleine-König 07 Jun '22

07 Jun '22

broadwell_disable_jack() returns zero unconditionally. Letting it return void instead makes it easier to see in the callers that there is no error to handle. This is a preparation for making platform remove callbacks return void. Signed-off-by: Uwe Kleine-König <u.kleine-koenig(a)pengutronix.de> --- sound/soc/intel/boards/broadwell.c | 12 +++++++----- 1 file changed, 7 insertions(+), 5 deletions(-) diff --git a/sound/soc/intel/boards/broadwell.c b/sound/soc/intel/boards/broadwell.c index d37c74fd1a3c..280e84bbe185 100644 --- a/sound/soc/intel/boards/broadwell.c +++ b/sound/soc/intel/boards/broadwell.c @@ -227,7 +227,7 @@ static struct snd_soc_dai_link broadwell_rt286_dais[] = { }, }; -static int broadwell_disable_jack(struct snd_soc_card *card) +static void broadwell_disable_jack(struct snd_soc_card *card) { struct snd_soc_component *component; @@ -239,13 +239,13 @@ static int broadwell_disable_jack(struct snd_soc_card *card) break; } } - - return 0; } static int broadwell_suspend(struct snd_soc_card *card) { - return broadwell_disable_jack(card); + broadwell_disable_jack(card); + + return 0; } static int broadwell_resume(struct snd_soc_card *card){ @@ -315,7 +315,9 @@ static int broadwell_audio_remove(struct platform_device *pdev) { struct snd_soc_card *card = platform_get_drvdata(pdev); - return broadwell_disable_jack(card); + broadwell_disable_jack(card); + + return 0; } static struct platform_driver broadwell_audio = { base-commit: 4b0986a3613c92f4ec1bdc7f60ec66fea135991f -- 2.36.1

3 2

[GIT PULL] ASoC fixes for v5.19-rc1
by Mark Brown 07 Jun '22

07 Jun '22

The following changes since commit ab0925ab728163b854ea17da09e299254a86076e: ASoC: da7219: Small fixes for jack detection and removal (2022-05-31 10:16:59 +0200) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v5.19-rc1 for you to fetch changes up to 2abdf9f80019e8244d3806ed0e1c9f725e50b452: ASoC: wm_adsp: Fix event generation for wm_adsp_fw_put() (2022-06-06 12:31:28 +0100) ---------------------------------------------------------------- ASoC: Fixes for v5.19 A few more fixes for v5.19 which came in during the second half of the merge window, again nothing that's really remarkable outside of the individual drivers. ---------------------------------------------------------------- Adam Ford (1): ASoC: wm8962: Fix suspend while playing music Amadeusz Sławiński (1): ASoC: SOF: Fix potential NULL pointer dereference Charles Keepax (6): ASoC: cs42l52: Fix TLV scales for mixer controls ASoC: cs35l36: Update digital volume TLV ASoC: cs53l30: Correct number of volume levels on SX controls ASoC: cs42l52: Correct TLV for Bypass Volume ASoC: cs42l56: Correct typo in minimum level for SX volume controls ASoC: cs42l51: Correct minimum value for SX volume control Dan Carpenter (2): ASoC: SOF: ipc-msg-injector: Propagate write errors correctly ASoC: SOF: ipc-msg-injector: Fix reversed if statement Hui Wang (1): ASoC: nau8822: Add operation for internal PLL off and on Mark Brown (3): ASoC: Fixup Cirrus SX control usage ASoC: es8328: Fix event generation for deemphasis control ASoC: wm_adsp: Fix event generation for wm_adsp_fw_put() Shengjiu Wang (1): ASoC: fsl_sai: Add support for i.MX8MN Srinivasa Rao Mandadapu (1): ASoC: qcom: lpass-platform: Update VMA access permissions in mmap callback xliu (1): ASoC: Intel: cirrus-common: fix incorrect channel mapping sound/soc/codecs/cs35l36.c | 3 ++- sound/soc/codecs/cs42l51.c | 2 +- sound/soc/codecs/cs42l52.c | 8 +++--- sound/soc/codecs/cs42l56.c | 4 +-- sound/soc/codecs/cs53l30.c | 16 ++++++------ sound/soc/codecs/es8328.c | 5 +++- sound/soc/codecs/nau8822.c | 4 +++ sound/soc/codecs/nau8822.h | 3 +++ sound/soc/codecs/wm8962.c | 1 + sound/soc/codecs/wm_adsp.c | 2 +- sound/soc/fsl/fsl_sai.c | 1 + sound/soc/intel/boards/sof_cirrus_common.c | 40 ++++++++++++++++++++++++++--- sound/soc/qcom/lpass-platform.c | 2 +- sound/soc/sof/sof-audio.c | 2 +- sound/soc/sof/sof-client-ipc-msg-injector.c | 28 +++++++++++--------- 15 files changed, 86 insertions(+), 35 deletions(-)

2 1

[PATCH] ASoC: dt-bindings: renesas,rz-ssi: Document RZ/G2UL SoC
by Biju Das 07 Jun '22

07 Jun '22

Document RZ/G2U2L SSI bindings. RZ/G2UL SSI is identical to one found on the RZ/G2L SoC. No driver changes are required as generic compatible string "renesas,rz-ssi" will be used as a fallback. While at it add a '.' at the end of dmas description for the first cell. Signed-off-by: Biju Das <biju.das.jz(a)bp.renesas.com> --- Documentation/devicetree/bindings/sound/renesas,rz-ssi.yaml | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/renesas,rz-ssi.yaml b/Documentation/devicetree/bindings/sound/renesas,rz-ssi.yaml index 7e8d252f7bca..0d9840375132 100644 --- a/Documentation/devicetree/bindings/sound/renesas,rz-ssi.yaml +++ b/Documentation/devicetree/bindings/sound/renesas,rz-ssi.yaml @@ -13,6 +13,7 @@ properties: compatible: items: - enum: + - renesas,r9a07g043-ssi # RZ/G2UL - renesas,r9a07g044-ssi # RZ/G2{L,LC} - renesas,r9a07g054-ssi # RZ/V2L - const: renesas,rz-ssi @@ -50,7 +51,7 @@ properties: minItems: 1 maxItems: 2 description: - The first cell represents a phandle to dmac + The first cell represents a phandle to dmac. The second cell specifies the encoded MID/RID values of the SSI port connected to the DMA client and the slave channel configuration parameters. -- 2.25.1

4 5

[PATCH 0/3] ASoC: Intel: soc-acpi and machine driver updates
by Pierre-Louis Bossart 07 Jun '22

07 Jun '22

Small updates to add initial tables for MeteorLake, SoundWire machine driver support for tests without HDMI and RT1019 for consistency on Chromebooks. Bard Liao (1): ASoC: SOF: Intel: add MeteorLake machines Pierre-Louis Bossart (1): ASoC: Intel: sof_sdw: allow HDaudio/HDMI disable Yong Zhi (1): ASoC: Intel: boards: rename RTL1019 compatible driver to rt1019p include/sound/soc-acpi-intel-match.h | 2 + sound/soc/intel/boards/Kconfig | 1 - sound/soc/intel/boards/sof_realtek_common.c | 24 +++++------ sound/soc/intel/boards/sof_realtek_common.h | 6 +-- sound/soc/intel/boards/sof_rt5682.c | 2 +- sound/soc/intel/boards/sof_sdw.c | 24 +++++++---- sound/soc/intel/common/Makefile | 1 + .../intel/common/soc-acpi-intel-mtl-match.c | 41 +++++++++++++++++++ 8 files changed, 75 insertions(+), 26 deletions(-) create mode 100644 sound/soc/intel/common/soc-acpi-intel-mtl-match.c -- 2.34.1

2 4

[PATCH v5 00/20] ASoC: mediatek: Add support for MT8186 SoC
by Jiaxin Yu 07 Jun '22

07 Jun '22

This series of patches adds support for Mediatek AFE of MT8186 Soc. Patches are based on broonie tree "for-next" branch. Changes since v4: - [v5 07/20] - remove unsusd controls - [v5 09/20] - correct indent error - [v5 10/20] - [v5 13/20] - [v5 14/20] - fix the return value if the value is different from the previous value in mixer controls - [v5 17/20] - [v5 19/20] - correct the compatible name with '_' instead of '-' - [v5 18/20] - [v5 20/20] - correct the yaml after 'pip3 install dtschema --upgrade' Changes since v3: - [v4 09/18] - remove DEBUG_COEFF debugging code - [v4 10/18] - simplify the logic of the code - [v4 13/18] - split out the MT6366 bits into mt8186-mt6366-commom.c - fix build error of "error: 'struct dev_pm_info' has no member named 'runtime_error'" - fix bug of adda dai driver - add route for pcm interface channel 2. - [v4 15/18] - [v4 17/18] - commonize the configuration of the codec - move MT6366 common bits into mt8186-mt6366-common.c Changes since v2: - add a new compatible string "mediatek,mt6366-sound" - modify the log level for simplicity - use dev_err_probe(...) instead of dev_err(...) in dev probe() - optimized the logic of some code - use BIT() and GENMASK() macros to descript the registers Thanks for AngeloGioacchino's careful reviews. Changes since v1: [v2 01/17] - add a new ID to the existing mt6358 codec driver [v2 03/17] - modify log level in DAPM events - use standard numeric control with name ending in Switch - return 1 when the value changed in mixer control's .get callback [v2 05/17] - ending in Switch to the standard on/off controls - change to "HW Gain 1 Volume" and "HW Gain 2 Volume" [v2 09/17] - return an error in the default case rather than just picking one of the behaviours when do .set_fmt - use the new defines that are _PROVIDER_MASK, _DAIFMT_CBP_CFP and _DAIFMT_CBC_CFC [v2 10/17] [v2 11/17] - the clock and gpio are aplit out into separate patches The source file's GPL comment use c++ style, and the header fils's GPL comment use c style. We have added "Switch" after the names of all the controls that just are simple on/off. Jiaxin Yu (20): ASoC: mediatek: mt6366: support for mt6366 codec dt-bindings: mediatek: mt6358: add new compatible for using mt6366 ASoC: mediatek: mt8186: support audsys clock control ASoC: mediatek: mt8186: support adda in platform driver ASoC: mediatek: mt8186: support hostless in platform driver ASoC: mediatek: mt8186: support hw gain in platform driver ASoC: mediatek: mt8186: support i2s in platform driver ASoC: mediatek: mt8186: support pcm in platform driver ASoC: mediatek: mt8186: support src in platform driver ASoC: mediatek: mt8186: support tdm in platform driver ASoC: mediatek: mt8186: support audio clock control in platform driver ASoC: mediatek: mt8186: support gpio control in platform driver ASoC: mediatek: mt8186: add misc driver and register definitions ASoC: mediatek: mt8186: add platform driver ASoC: mediatek: mt8186: add mt8186-mt6366 common driver dt-bindings: mediatek: mt8186: add audio afe document ASoC: mediatek: mt8186: add machine driver with mt6366, da7219 and max98357 dt-bindings: mediatek: mt8186: add mt8186-mt6366-da7219-max98357 document ASoC: mediatek: mt8186: add machine driver with mt6366, rt1019 and rt5682s dt-bindings: mediatek: mt8186: add mt8186-mt6366-rt1019-rt5682s document .../devicetree/bindings/sound/mt6358.txt | 4 +- .../bindings/sound/mt8186-afe-pcm.yaml | 175 + .../sound/mt8186-mt6366-da7219-max98357.yaml | 75 + .../sound/mt8186-mt6366-rt1019-rt5682s.yaml | 75 + sound/soc/codecs/mt6358.c | 1 + sound/soc/mediatek/Kconfig | 44 + sound/soc/mediatek/Makefile | 1 + sound/soc/mediatek/mt8186/Makefile | 22 + sound/soc/mediatek/mt8186/mt8186-afe-clk.c | 651 ++++ sound/soc/mediatek/mt8186/mt8186-afe-clk.h | 106 + sound/soc/mediatek/mt8186/mt8186-afe-common.h | 235 ++ .../soc/mediatek/mt8186/mt8186-afe-control.c | 261 ++ sound/soc/mediatek/mt8186/mt8186-afe-gpio.c | 244 ++ sound/soc/mediatek/mt8186/mt8186-afe-gpio.h | 19 + sound/soc/mediatek/mt8186/mt8186-afe-pcm.c | 3009 +++++++++++++++++ sound/soc/mediatek/mt8186/mt8186-audsys-clk.c | 150 + sound/soc/mediatek/mt8186/mt8186-audsys-clk.h | 15 + .../soc/mediatek/mt8186/mt8186-audsys-clkid.h | 45 + sound/soc/mediatek/mt8186/mt8186-dai-adda.c | 873 +++++ .../soc/mediatek/mt8186/mt8186-dai-hostless.c | 298 ++ .../soc/mediatek/mt8186/mt8186-dai-hw-gain.c | 236 ++ sound/soc/mediatek/mt8186/mt8186-dai-i2s.c | 1286 +++++++ sound/soc/mediatek/mt8186/mt8186-dai-pcm.c | 423 +++ sound/soc/mediatek/mt8186/mt8186-dai-src.c | 695 ++++ sound/soc/mediatek/mt8186/mt8186-dai-tdm.c | 698 ++++ .../mediatek/mt8186/mt8186-interconnection.h | 69 + .../soc/mediatek/mt8186/mt8186-misc-control.c | 252 ++ .../mediatek/mt8186/mt8186-mt6366-common.c | 59 + .../mediatek/mt8186/mt8186-mt6366-common.h | 17 + .../mt8186/mt8186-mt6366-da7219-max98357.c | 1003 ++++++ .../mt8186/mt8186-mt6366-rt1019-rt5682s.c | 979 ++++++ sound/soc/mediatek/mt8186/mt8186-reg.h | 2913 ++++++++++++++++ 32 files changed, 14932 insertions(+), 1 deletion(-) create mode 100644 Documentation/devicetree/bindings/sound/mt8186-afe-pcm.yaml create mode 100644 Documentation/devicetree/bindings/sound/mt8186-mt6366-da7219-max98357.yaml create mode 100644 Documentation/devicetree/bindings/sound/mt8186-mt6366-rt1019-rt5682s.yaml create mode 100644 sound/soc/mediatek/mt8186/Makefile create mode 100644 sound/soc/mediatek/mt8186/mt8186-afe-clk.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-afe-clk.h create mode 100644 sound/soc/mediatek/mt8186/mt8186-afe-common.h create mode 100644 sound/soc/mediatek/mt8186/mt8186-afe-control.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-afe-gpio.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-afe-gpio.h create mode 100644 sound/soc/mediatek/mt8186/mt8186-afe-pcm.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-audsys-clk.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-audsys-clk.h create mode 100644 sound/soc/mediatek/mt8186/mt8186-audsys-clkid.h create mode 100644 sound/soc/mediatek/mt8186/mt8186-dai-adda.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-dai-hostless.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-dai-hw-gain.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-dai-i2s.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-dai-pcm.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-dai-src.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-dai-tdm.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-interconnection.h create mode 100644 sound/soc/mediatek/mt8186/mt8186-misc-control.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-mt6366-common.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-mt6366-common.h create mode 100644 sound/soc/mediatek/mt8186/mt8186-mt6366-da7219-max98357.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-mt6366-rt1019-rt5682s.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-reg.h -- 2.18.0

3 24

[PATCH 0/2] Switch to use internal PLL for iMCLK
by Hui Wang 07 Jun '22

07 Jun '22

Hi David.Lin, Taking your advice and try to enable internal PLL to get a more accurate sample rate. And I also changed the fsl-asoc-card.c to support the nau8822 codec, now the sound quality is pretty good on my imx6sx EVB. Please help take a look at these 2 patches on codec driver. Thanks. Hui Wang (2): ASoC: nau8822: Add operation for internal PLL off and on ASoC: nau8822: Disable internal PLL if freq_out is zero sound/soc/codecs/nau8822.c | 11 +++++++++++ sound/soc/codecs/nau8822.h | 3 +++ 2 files changed, 14 insertions(+) -- 2.25.1

3 14

[PATCH v5 0/2]Add Machine driver support for nau8825, max98560 and rt5682s, rt1019
by V sujith kumar Reddy 07 Jun '22

07 Jun '22

This patch set depends on: --gpio patch --https://patchwork.kernel.org/project/alsa-devel/patch/20220516160619.17832-1-Vsujithkumar.Reddy@amd.com/ --https://patchwork.kernel.org/project/alsa-devel/patch/20220516160619.17832-2-Vsujithkumar.Reddy@amd.com/ v4 --> v5: Resolved: . Used separate firmware files. . Deleted useless initialization variables v3 --> v4: Resolved: . Assigning direct address without (void *) typecast. . Used snd_soc_card_jack_new instead of snd_soc_card_jack_new_pins as per suggestion. v2 --> v3: Resolved compilation error: /build/stage/linux/sound/soc/amd/acp/acp-mach-common.c: In function ‘acp_card_nau8825_init’: /build/stage/linux/sound/soc/amd/acp/acp-mach-common.c:479:15: error: too many arguments to function ‘snd_soc_card_jack_new’ 479 | ret = snd_soc_card_jack_new(card, "Headset Jack", | ^~~~~~~~~~~~~~~~~~~~~ In file included from /build/stage/linux/include/sound/soc.h:1362, from /build/stage/linux/sound/soc/amd/acp/acp-mach-common.c:20: /build/stage/linux/include/sound/soc-card.h:18:5: note: declared here 18 | int snd_soc_card_jack_new(struct snd_soc_card *card, const char *id, V sujith kumar Reddy (2): ASoC: amd: acp: Add support for nau8825 and max98360 card ASoC: amd: acp: Add support for rt5682s and rt1019 card with hs instance sound/soc/amd/acp-config.c | 30 ++++ sound/soc/amd/acp/Kconfig | 1 + sound/soc/amd/acp/acp-mach-common.c | 233 ++++++++++++++++++++++++++-- sound/soc/amd/acp/acp-mach.h | 3 + sound/soc/amd/acp/acp-sof-mach.c | 30 ++++ sound/soc/amd/mach-config.h | 1 + 6 files changed, 282 insertions(+), 16 deletions(-) -- 2.25.1

3 4

[PATCH v3 0/6] OPE support on Tegra210 and later
by Sameer Pujar 07 Jun '22

07 Jun '22

This series adds support for Output Prcoessing Module (OPE) module on Tegra210 and later generations of SoCs. OPE is a client of AHUB and it has sub blocks of PEQ (Parametric Equalizer) and MBDRC (Multi Band Dynamic Range Compressor) for data processing. An ASoC component is registered for OPE, which includes PEQ and MBDRC functions as well. This can be plugged in audio path using ALSA mixer controls. The series adds necessary binding documentaion, driver and DT binding patches to enable OPE module on Jetson platforms. Changelog ========= v2 -> v3: --------- * Drop "Device Tree Bindings" string from bindind doc titles for OPE, PEQ and MBDRC. v1 -> v2: --------- * Use generic node names for OPE, PEQ and MBDRC devices. Update binding doc and DT patches for this. * Remove redundant nodename rule enforcement from OPE, PEQ and MBDRC nodes. Update binding doc patch for this. * Fix spaces before binding doc examples and remove '|' from binding doc descriptions. Sameer Pujar (6): ASoC: tegra: Add binding doc for OPE module ASoC: tegra: Add Tegra210 based OPE driver ASoC: tegra: AHUB routes for OPE module arm64: defconfig: Build Tegra OPE module arm64: tegra: Add OPE device on Tegra210 and later arm64: tegra: Enable OPE on various platforms .../bindings/sound/nvidia,tegra210-ahub.yaml | 4 + .../bindings/sound/nvidia,tegra210-mbdrc.yaml | 47 + .../bindings/sound/nvidia,tegra210-ope.yaml | 87 ++ .../bindings/sound/nvidia,tegra210-peq.yaml | 48 + arch/arm64/boot/dts/nvidia/tegra186-p2771-0000.dts | 43 + arch/arm64/boot/dts/nvidia/tegra186.dtsi | 23 + arch/arm64/boot/dts/nvidia/tegra194-p2972-0000.dts | 43 + .../arm64/boot/dts/nvidia/tegra194-p3509-0000.dtsi | 43 + arch/arm64/boot/dts/nvidia/tegra194.dtsi | 23 + arch/arm64/boot/dts/nvidia/tegra210-p2371-2180.dts | 84 ++ arch/arm64/boot/dts/nvidia/tegra210-p3450-0000.dts | 84 ++ arch/arm64/boot/dts/nvidia/tegra210.dtsi | 40 + .../dts/nvidia/tegra234-p3737-0000+p3701-0000.dts | 43 + arch/arm64/boot/dts/nvidia/tegra234.dtsi | 23 + arch/arm64/configs/defconfig | 1 + sound/soc/tegra/Kconfig | 9 + sound/soc/tegra/Makefile | 2 + sound/soc/tegra/tegra210_ahub.c | 39 +- sound/soc/tegra/tegra210_mbdrc.c | 1012 ++++++++++++++++++++ sound/soc/tegra/tegra210_mbdrc.h | 215 +++++ sound/soc/tegra/tegra210_ope.c | 419 ++++++++ sound/soc/tegra/tegra210_ope.h | 90 ++ sound/soc/tegra/tegra210_peq.c | 434 +++++++++ sound/soc/tegra/tegra210_peq.h | 56 ++ 24 files changed, 2908 insertions(+), 4 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-mbdrc.yaml create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-ope.yaml create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-peq.yaml create mode 100644 sound/soc/tegra/tegra210_mbdrc.c create mode 100644 sound/soc/tegra/tegra210_mbdrc.h create mode 100644 sound/soc/tegra/tegra210_ope.c create mode 100644 sound/soc/tegra/tegra210_ope.h create mode 100644 sound/soc/tegra/tegra210_peq.c create mode 100644 sound/soc/tegra/tegra210_peq.h -- 2.7.4

3 10

[PATCH 0/4] ASoC: Drop some i2c noop remove callbacks
by Uwe Kleine-König 07 Jun '22

07 Jun '22

From: Uwe Kleine-König <uwe(a)kleine-koenig.org> Hello, I intended to send this after -rc1 was cut, but found a few spare minutes to prepare this series. All four patches were sent already before based on v5.18, but there were some conflicting changes added in the merge window. This series contains the four patches on top of current linus/master and so bases on a tree including the conflicting changes. Expecting no more sound changes in this merge window, this should apply cleanly on top of -rc1. Best regards Uwe Uwe Kleine-König (4): ASoC: ak4642: Drop no-op remove function ASoC: da7219: Drop no-op remove function ASoC: lm49453: Drop no-op remove function ASoC: da732x: Drop no-op remove function sound/soc/codecs/ak4613.c | 6 ------ sound/soc/codecs/da7219.c | 6 ------ sound/soc/codecs/da732x.c | 6 ------ sound/soc/codecs/lm49453.c | 6 ------ 4 files changed, 24 deletions(-) base-commit: 50fd82b3a9a9335df5d50c7ddcb81c81d358c4fc -- 2.36.1

2 5

[PATCH] ASoC: ak4642: Drop no-op remove function
by Uwe Kleine-König 07 Jun '22

07 Jun '22

A remove callback that just returns 0 is equivalent to no callback at all as can be seen in i2c_device_remove(). So simplify accordingly. Signed-off-by: Uwe Kleine-König <u.kleine-koenig(a)pengutronix.de> --- sound/soc/codecs/ak4613.c | 6 ------ 1 file changed, 6 deletions(-) diff --git a/sound/soc/codecs/ak4613.c b/sound/soc/codecs/ak4613.c index 034195c83bd7..20a85a81b2ad 100644 --- a/sound/soc/codecs/ak4613.c +++ b/sound/soc/codecs/ak4613.c @@ -684,18 +684,12 @@ static int ak4613_i2c_probe(struct i2c_client *i2c, &ak4613_dai, 1); } -static int ak4613_i2c_remove(struct i2c_client *client) -{ - return 0; -} - static struct i2c_driver ak4613_i2c_driver = { .driver = { .name = "ak4613-codec", .of_match_table = ak4613_of_match, }, .probe = ak4613_i2c_probe, - .remove = ak4613_i2c_remove, .id_table = ak4613_i2c_id, }; base-commit: 4b0986a3613c92f4ec1bdc7f60ec66fea135991f -- 2.36.1

2 1

[PATCH v4 0/2] Add Machine driver support for nau8825, max98560 and rt5682s, rt1019
by V sujith kumar Reddy 07 Jun '22

07 Jun '22

This patch set depends on: --gpio patch --https://patchwork.kernel.org/project/alsa-devel/patch/20220516160619.17832-1-Vsujithkumar.Reddy@amd.com/ --https://patchwork.kernel.org/project/alsa-devel/patch/20220516160619.17832-2-Vsujithkumar.Reddy@amd.com/ v3 --> v4: Resolved: . Assigning direct address without (void *) typecast. . Used snd_soc_card_jack_new instead of snd_soc_card_jack_new_pins as per suggestion. v2 --> v3: Resolved compilation error: /build/stage/linux/sound/soc/amd/acp/acp-mach-common.c: In function ‘acp_card_nau8825_init’: /build/stage/linux/sound/soc/amd/acp/acp-mach-common.c:479:15: error: too many arguments to function ‘snd_soc_card_jack_new’ 479 | ret = snd_soc_card_jack_new(card, "Headset Jack", | ^~~~~~~~~~~~~~~~~~~~~ In file included from /build/stage/linux/include/sound/soc.h:1362, from /build/stage/linux/sound/soc/amd/acp/acp-mach-common.c:20: /build/stage/linux/include/sound/soc-card.h:18:5: note: declared here 18 | int snd_soc_card_jack_new(struct snd_soc_card *card, const char *id, V sujith kumar Reddy (2): ASoC: amd: acp: Add support for nau8825 and max98360 card ASoC: amd: acp: Add support for rt5682s and rt1019 card with hs instance sound/soc/amd/acp-config.c | 30 ++++ sound/soc/amd/acp/Kconfig | 1 + sound/soc/amd/acp/acp-mach-common.c | 235 ++++++++++++++++++++++++++-- sound/soc/amd/acp/acp-mach.h | 3 + sound/soc/amd/acp/acp-sof-mach.c | 30 ++++ sound/soc/amd/mach-config.h | 1 + 6 files changed, 283 insertions(+), 17 deletions(-) -- 2.25.1

4 7

[PATCH v4 00/17] ALSA: hda: cirrus: Add initial DSP support and firmware loading
by Vitaly Rodionov 07 Jun '22

07 Jun '22

The CS35L41 Amplifier contains a DSP, capable of running firmware. The firmware can run algorithms such as Speaker Protection, to ensure that playback at high gains do not harm the speakers. Adding support for CS35L41 firmware into the CS35L41 HDA driver also allows us to support several extra features, such as hiberation and interrupts. The chain adds support in stages: - General fixes to improve generalization and code re-use inside the CS35L41 HDA driver. - Add support for interrupts into the driver, which is required for complete support of the firmware. - Refactor ASoC CS35L41 code which deals with firmware to allow for code re-use inside the CS35L41 HDA driver. - Add support for loading firmware and tuning files from file system, and creating alsa controls to control it. - Support firmware load paths for different hardware systems. - Support suspend/resume in the driver when using firmware. The firmware supports hibernation, which allows the CS35L41 to drop into a low power mode during suspend. - Support the ability to unload firmware, swap and reload the firmware. This is to allow different firmware to run during calibration. The intended use-case is to load the firmware once on boot, and the driver autmatically tries to load the firmware after it binds to the HDA driver. This behaviour can be switched off using a kconfig, if desired. Stefan Binding (16): ALSA: hda: hda_cs_dsp_ctl: Add Library to support CS_DSP ALSA controls ALSA: hda: hda_cs_dsp_ctl: Add apis to write the controls directly ALSA: hda: cs35l41: Save codec object inside component struct ALSA: hda: cs35l41: Save Subsystem ID inside CS35L41 Driver ALSA: hda: cs35l41: Support reading subsystem id from ACPI ALSA: hda: cs35l41: Support multiple load paths for firmware ALSA: hda: cs35l41: Support Speaker ID for laptops ASoC: cs35l41: Move cs35l41 exit hibernate function into shared code ASoC: cs35l41: Do not print error when waking from hibernation ASoC: cs35l41: Add common cs35l41 enter hibernate function ALSA: hda: cs35l41: Support Hibernation during Suspend ALSA: hda: cs35l41: Read Speaker Calibration data from UEFI variables ALSA: hda: hda_cs_dsp_ctl: Add fw id strings ALSA: hda: cs35l41: Add defaulted values into dsp bypass config sequence ALSA: hda: cs35l41: Support Firmware switching and reloading ALSA: hda: cs35l41: Add module parameter to control firmware load Vitaly Rodionov (1): ALSA: hda: cs35l41: Add initial DSP support and firmware loading MAINTAINERS | 1 + include/sound/cs35l41.h | 7 + sound/pci/hda/Kconfig | 8 + sound/pci/hda/Makefile | 2 + sound/pci/hda/cs35l41_hda.c | 853 +++++++++++++++++++++++++++++++- sound/pci/hda/cs35l41_hda.h | 37 ++ sound/pci/hda/cs35l41_hda_i2c.c | 1 + sound/pci/hda/cs35l41_hda_spi.c | 1 + sound/pci/hda/hda_component.h | 4 + sound/pci/hda/hda_cs_dsp_ctl.c | 302 +++++++++++ sound/pci/hda/hda_cs_dsp_ctl.h | 40 ++ sound/pci/hda/patch_realtek.c | 27 +- sound/soc/codecs/cs35l41-lib.c | 82 ++- sound/soc/codecs/cs35l41.c | 71 +-- 14 files changed, 1362 insertions(+), 74 deletions(-) create mode 100644 sound/pci/hda/hda_cs_dsp_ctl.c create mode 100644 sound/pci/hda/hda_cs_dsp_ctl.h -- 2.34.1

6 34

[PATCH 00/16] ASoC: trivial changes for cppcheck warnings
by Pierre-Louis Bossart 07 Jun '22

07 Jun '22

This can probably wait for the next merge window, I found a number of cppcheck warnings that I didn't see in my last checks. The irony is that the only really important issue found by cppcheck was on one of my previous DPCM changes (submitted separately as a fix). Pierre-Louis Bossart (16): ASoC: cs35l45: typo in argument definition ASoC: cs42l42: remove redundant test ASoC: wcd-mbhc-v2: remove useless initialization ASoC: wcd9335: remove redundant tests ASoC: Intel: atom: sst: remove useless initialization ASoC: Intel: atom: sst_ipc: remove redundant test ASoC: Intel: atom: sst_ipc: remove useless initializations ASoC: Intel: atom: controls: remove useless initializations ASoC: Intel: boards: reset acpi_chan_package ASoC: Intel: sof_pcm512x: remove unnecessary init ASoC: mediatek: mt8195: simplify error handling ASoC: qcom: q6dsp: q6adm: remove useless initializations ASoC: qcom: q6dsp: remove spurious space ASoC: rockchip: simplify error handling ASoC: samsung: snow: simplify error handling ASoC: meson: remove useless initialization sound/soc/codecs/cs35l45.h | 2 +- sound/soc/codecs/cs42l42.c | 3 +-- sound/soc/codecs/wcd-mbhc-v2.c | 2 +- sound/soc/codecs/wcd9335.c | 4 ++-- sound/soc/intel/atom/sst-atom-controls.c | 4 ++-- sound/soc/intel/atom/sst/sst.c | 2 +- sound/soc/intel/atom/sst/sst_ipc.c | 8 ++++---- sound/soc/intel/boards/bytcr_rt5640.c | 2 +- sound/soc/intel/boards/bytcr_rt5651.c | 2 +- sound/soc/intel/boards/cht_bsw_rt5645.c | 2 +- sound/soc/intel/boards/sof_pcm512x.c | 2 +- sound/soc/mediatek/mt8195/mt8195-afe-clk.c | 8 ++++---- sound/soc/meson/meson-codec-glue.c | 2 +- sound/soc/qcom/qdsp6/q6adm.c | 6 +++--- sound/soc/rockchip/rk3288_hdmi_analog.c | 4 ++-- sound/soc/samsung/snow.c | 2 +- 16 files changed, 27 insertions(+), 28 deletions(-) -- 2.30.2

2 17

[PATCH v3 0/2] Add Machine driver support for nau8825, max98560 and rt5682s, rt1019
by V sujith kumar Reddy 07 Jun '22

07 Jun '22

This patch set depends on: --gpio patch --https://patchwork.kernel.org/project/alsa-devel/patch/20220516160619.17832-1-Vsujithkumar.Reddy@amd.com/ --https://patchwork.kernel.org/project/alsa-devel/patch/20220516160619.17832-2-Vsujithkumar.Reddy@amd.com/ Resolved compilation error: /build/stage/linux/sound/soc/amd/acp/acp-mach-common.c: In function ‘acp_card_nau8825_init’: /build/stage/linux/sound/soc/amd/acp/acp-mach-common.c:479:15: error: too many arguments to function ‘snd_soc_card_jack_new’ 479 | ret = snd_soc_card_jack_new(card, "Headset Jack", | ^~~~~~~~~~~~~~~~~~~~~ In file included from /build/stage/linux/include/sound/soc.h:1362, from /build/stage/linux/sound/soc/amd/acp/acp-mach-common.c:20: /build/stage/linux/include/sound/soc-card.h:18:5: note: declared here 18 | int snd_soc_card_jack_new(struct snd_soc_card *card, const char *id, int type V sujith kumar Reddy (2): ASoC: amd: acp: Add support for nau8825 and max98360 card ASoC: amd: acp: Add support for rt5682s and rt1019 card with hs instance sound/soc/amd/acp-config.c | 30 ++++ sound/soc/amd/acp/Kconfig | 1 + sound/soc/amd/acp/acp-mach-common.c | 235 ++++++++++++++++++++++++++-- sound/soc/amd/acp/acp-mach.h | 3 + sound/soc/amd/acp/acp-sof-mach.c | 30 ++++ sound/soc/amd/mach-config.h | 1 + 6 files changed, 283 insertions(+), 17 deletions(-) -- 2.25.1

5 7

[PATCH v2 0/2] ASOC: amd: acp: Add Machine driver changes for
by V sujith kumar Reddy 07 Jun '22

07 Jun '22

Add Machine driver support for nau8825, max98560 and rt5682s,rt1019 combination This patch set depends on: --gpio patch --https://patchwork.kernel.org/project/alsa-devel/patch/20220516160619.17832-1-Vsujithkumar.Reddy@amd.com/ --https://patchwork.kernel.org/project/alsa-devel/patch/20220516160619.17832-2-Vsujithkumar.Reddy@amd.com/ V sujith kumar Reddy (2): ASoC: amd: acp: Add support for nau8825 and max98360 card ASoC: amd: acp: Add support for rt5682s and rt1019 card with hs instance sound/soc/amd/acp-config.c | 30 ++++ sound/soc/amd/acp/Kconfig | 1 + sound/soc/amd/acp/acp-mach-common.c | 228 +++++++++++++++++++++++++--- sound/soc/amd/acp/acp-mach.h | 3 + sound/soc/amd/acp/acp-sof-mach.c | 30 ++++ sound/soc/amd/mach-config.h | 1 + 6 files changed, 276 insertions(+), 17 deletions(-) -- 2.25.1

2 4

[PATCH v3 00/17] ALSA: hda: cirrus: Add initial DSP support and firmware loading
by Vitaly Rodionov 07 Jun '22

07 Jun '22

The CS35L41 Amplifier contains a DSP, capable of running firmware. The firmware can run algorithms such as Speaker Protection, to ensure that playback at high gains do not harm the speakers. Adding support for CS35L41 firmware into the CS35L41 HDA driver also allows us to support several extra features, such as hiberation and interrupts. The chain adds support in stages: - General fixes to improve generalization and code re-use inside the CS35L41 HDA driver. - Add support for interrupts into the driver, which is required for complete support of the firmware. - Refactor ASoC CS35L41 code which deals with firmware to allow for code re-use inside the CS35L41 HDA driver. - Add support for loading firmware and tuning files from file system, and creating alsa controls to control it. - Support firmware load paths for different hardware systems. - Support suspend/resume in the driver when using firmware. The firmware supports hibernation, which allows the CS35L41 to drop into a low power mode during suspend. - Support the ability to unload firmware, swap and reload the firmware. This is to allow different firmware to run during calibration. The intended use-case is to load the firmware once on boot, and the driver autmatically tries to load the firmware after it binds to the HDA driver. This behaviour can be switched off using a kconfig, if desired. Stefan Binding (16): ALSA: hda: hda_cs_dsp_ctl: Add Library to support CS_DSP ALSA controls ALSA: hda: hda_cs_dsp_ctl: Add apis to write the controls directly ALSA: hda: cs35l41: Save codec object inside component struct ALSA: hda: cs35l41: Save Subsystem ID inside CS35L41 Driver ALSA: hda: cs35l41: Support reading subsystem id from ACPI ALSA: hda: cs35l41: Support multiple load paths for firmware ALSA: hda: cs35l41: Support Speaker ID for laptops ASoC: cs35l41: Move cs35l41 exit hibernate function into shared code ASoC: cs35l41: Do not print error when waking from hibernation ASoC: cs35l41: Add common cs35l41 enter hibernate function ALSA: hda: cs35l41: Support Hibernation during Suspend ALSA: hda: cs35l41: Read Speaker Calibration data from UEFI variables ALSA: hda: hda_cs_dsp_ctl: Add fw id strings ALSA: hda: cs35l41: Add defaulted values into dsp bypass config sequence ALSA: hda: cs35l41: Support Firmware switching and reloading ALSA: hda: cs35l41: Add module parameter to control firmware load Vitaly Rodionov (1): ALSA: hda: cs35l41: Add initial DSP support and firmware loading MAINTAINERS | 1 + include/sound/cs35l41.h | 7 + sound/pci/hda/Kconfig | 8 + sound/pci/hda/Makefile | 2 + sound/pci/hda/cs35l41_hda.c | 815 +++++++++++++++++++++++++++++++- sound/pci/hda/cs35l41_hda.h | 37 ++ sound/pci/hda/cs35l41_hda_i2c.c | 1 + sound/pci/hda/cs35l41_hda_spi.c | 1 + sound/pci/hda/hda_component.h | 4 + sound/pci/hda/hda_cs_dsp_ctl.c | 302 ++++++++++++ sound/pci/hda/hda_cs_dsp_ctl.h | 40 ++ sound/pci/hda/patch_realtek.c | 27 +- sound/soc/codecs/cs35l41-lib.c | 82 +++- sound/soc/codecs/cs35l41.c | 71 +-- 14 files changed, 1324 insertions(+), 74 deletions(-) create mode 100644 sound/pci/hda/hda_cs_dsp_ctl.c create mode 100644 sound/pci/hda/hda_cs_dsp_ctl.h -- 2.34.1

3 23

[PATCH v2 00/26] *ALSA: hda: cirrus: Add initial DSP support and firmware loading
by Vitaly Rodionov 07 Jun '22

07 Jun '22

The CS35L41 Amplifier contains a DSP, capable of running firmware. The firmware can run algorithms such as Speaker Protection, to ensure that playback at high gains do not harm the speakers. Adding support for CS35L41 firmware into the CS35L41 HDA driver also allows us to support several extra features, such as hiberation and interrupts. The chain adds support in stages: - General fixes to improve generalization and code re-use inside the CS35L41 HDA driver. - Add support for interrupts into the driver, which is required for complete support of the firmware. - Refactor ASoC CS35L41 code which deals with firmware to allow for code re-use inside the CS35L41 HDA driver. - Add support for loading firmware and tuning files from file system, and creating alsa controls to control it. - Support firmware load paths for different hardware systems. - Support suspend/resume in the driver when using firmware. The firmware supports hibernation, which allows the CS35L41 to drop into a low power mode during suspend. - Support the ability to unload firmware, swap and reload the firmware. This is to allow different firmware to run during calibration. The intended use-case is to load the firmware once on boot, and the driver autmatically tries to load the firmware after it binds to the HDA driver. This behaviour can be switched off using a kconfig, if desired. Stefan Binding (25): ALSA: hda: cs35l41: Fix error in spi cs35l41 hda driver name ALSA: hda: cs35l41: Set Speaker Position for CLSA0100 Laptop ALSA: hda: cs35l41: Remove Set Channel Map api from binding ALSA: hda: cs35l41: Add Support for Interrupts ALSA: hda: cs35l41: Enable GPIO2 Interrupt for CLSA0100 laptops ASoC: cs35l41: Move cs35l41_set_cspl_mbox_cmd to shared code ASoC: cs35l41: Move cs35l41 fs errata into shared code ASoC: cs35l41: Move cs_dsp config struct into shared code ALSA: hda: cs35l41: Add Amp Name based on channel and index ALSA: hda: hda_cs_dsp_ctl: Add Library to support CS_DSP ALSA controls ALSA: hda: hda_cs_dsp_ctl: Add apis to write the controls directly ALSA: hda: cs35l41: Save codec object inside component struct ALSA: hda: cs35l41: Save Subsystem ID inside CS35L41 Driver ALSA: hda: cs35l41: Support reading subsystem id from ACPI ALSA: hda: cs35l41: Support multiple load paths for firmware ALSA: hda: cs35l41: Support Speaker ID for laptops ASoC: cs35l41: Move cs35l41 exit hibernate function into shared code ASoC: cs35l41: Do not print error when waking from hibernation ASoC: cs35l41: Add common cs35l41 enter hibernate function ALSA: hda: cs35l41: Support Hibernation during Suspend ALSA: hda: cs35l41: Read Speaker Calibration data from UEFI variables ALSA: hda: hda_cs_dsp_ctl: Add fw id strings ALSA: hda: cs35l41: Add defaulted values into dsp bypass config sequence ALSA: hda: cs35l41: Support Firmware switching and reloading ALSA: hda: cs35l41: Add kernel config to disable firmware autoload Vitaly Rodionov (1): ALSA: hda: cs35l41: Add initial DSP support and firmware loading MAINTAINERS | 1 + include/sound/cs35l41.h | 84 +++ sound/pci/hda/Kconfig | 17 + sound/pci/hda/Makefile | 2 + sound/pci/hda/cs35l41_hda.c | 954 +++++++++++++++++++++++++++++++- sound/pci/hda/cs35l41_hda.h | 41 ++ sound/pci/hda/cs35l41_hda_i2c.c | 1 + sound/pci/hda/cs35l41_hda_spi.c | 3 +- sound/pci/hda/hda_component.h | 6 +- sound/pci/hda/hda_cs_dsp_ctl.c | 424 ++++++++++++++ sound/pci/hda/hda_cs_dsp_ctl.h | 40 ++ sound/pci/hda/patch_realtek.c | 81 +-- sound/soc/codecs/cs35l41-lib.c | 193 +++++++ sound/soc/codecs/cs35l41.c | 186 +------ sound/soc/codecs/cs35l41.h | 18 - 15 files changed, 1797 insertions(+), 254 deletions(-) create mode 100644 sound/pci/hda/hda_cs_dsp_ctl.c create mode 100644 sound/pci/hda/hda_cs_dsp_ctl.h -- 2.34.1

4 34

[PATCH 00/26] ALSA: hda: cirrus: Add initial DSP support and firmware loading
by Vitaly Rodionov 07 Jun '22

07 Jun '22

The CS35L41 Amplifier contains a DSP, capable of running firmware. The firmware can run algorithms such as Speaker Protection, to ensure that playback at high gains do not harm the speakers. Adding support for CS35L41 firmware into the CS35L41 HDA driver also allows us to support several extra features, such as hiberation and interrupts. The chain adds support in stages: - General fixes to improve generalization and code re-use inside the CS35L41 HDA driver. - Add support for interrupts into the driver, which is required for complete support of the firmware. - Refactor ASoC CS35L41 code which deals with firmware to allow for code re-use inside the CS35L41 HDA driver. - Add support for loading firmware and tuning files from file system, and creating alsa controls to control it. - Support firmware load paths for different hardware systems. - Support suspend/resume in the driver when using firmware. The firmware supports hibernation, which allows the CS35L41 to drop into a low power mode during suspend. - Support the ability to unload firmware, swap and reload the firmware. This is to allow different firmware to run during calibration. The intended use-case is to load the firmware once on boot, and the driver autmatically tries to load the firmware after it binds to the HDA driver. This behaviour can be switched off using a kconfig, if desired. Stefan Binding (25): ALSA: hda: cs35l41: Fix error in spi cs35l41 hda driver name ALSA: hda: cs35l41: Set Speaker Position for CLSA0100 Laptop ALSA: hda: cs35l41: Remove Set Channel Map api from binding ALSA: hda: cs35l41: Add Support for Interrupts ALSA: hda: cs35l41: Enable GPIO2 Interrupt for CLSA0100 laptops ASoC: cs35l41: Move cs35l41_set_cspl_mbox_cmd to shared code ASoC: cs35l41: Move cs35l41 fs errata into shared code ASoC: cs35l41: Move cs_dsp config struct into shared code ALSA: hda: cs35l41: Add Amp Name based on channel and index ALSA: hda: hda_cs_dsp_ctl: Add Library to support CS_DSP ALSA controls ALSA: hda: hda_cs_dsp_ctl: Add apis to write the controls directly ALSA: hda: cs35l41: Save codec object inside component struct ALSA: hda: cs35l41: Save Subsystem ID inside CS35L41 Driver ALSA: hda: cs35l41: Support reading subsystem id from ACPI ALSA: hda: cs35l41: Support multiple load paths for firmware ALSA: hda: cs35l41: Support Speaker ID for laptops ASoC: cs35l41: Move cs35l41 exit hibernate function into shared code ALSA: cs35l41: Do not print error when waking from hibernation ASoC: cs35l41: Add common cs35l41 enter hibernate function ALSA: hda: cs35l41: Support Hibernation during Suspend ALSA: hda: cs35l41: Read Speaker Calibration data from UEFI variables ALSA: hda: hda_cs_dsp_ctl: Add fw id strings ALSA: hda: cs35l41: Add defaulted values into dsp bypass config sequence ALSA: hda: cs35l41: Support Firmware switching and reloading ALSA: hda: cs35l41: Add kernel config to disable firmware autoload Vitaly Rodionov (1): ALSA: hda: cs35l41: Add initial DSP support and firmware loading MAINTAINERS | 1 + include/sound/cs35l41.h | 84 +++ sound/pci/hda/Kconfig | 17 + sound/pci/hda/Makefile | 2 + sound/pci/hda/cs35l41_hda.c | 954 +++++++++++++++++++++++++++++++- sound/pci/hda/cs35l41_hda.h | 41 ++ sound/pci/hda/cs35l41_hda_i2c.c | 1 + sound/pci/hda/cs35l41_hda_spi.c | 3 +- sound/pci/hda/hda_component.h | 6 +- sound/pci/hda/hda_cs_dsp_ctl.c | 424 ++++++++++++++ sound/pci/hda/hda_cs_dsp_ctl.h | 40 ++ sound/pci/hda/patch_realtek.c | 81 +-- sound/soc/codecs/cs35l41-lib.c | 193 +++++++ sound/soc/codecs/cs35l41.c | 186 +------ sound/soc/codecs/cs35l41.h | 18 - 15 files changed, 1797 insertions(+), 254 deletions(-) create mode 100644 sound/pci/hda/hda_cs_dsp_ctl.c create mode 100644 sound/pci/hda/hda_cs_dsp_ctl.h -- 2.32.0

5 37

[PATCH] ASoC: es8328: Fix event generation for deemphasis control
by Mark Brown 07 Jun '22

07 Jun '22

Currently the put() method for the deemphasis control returns 0 when a new value is written to the control even if the value changed, meaning events are not generated. Fix this, skip the work of updating the value when it is unchanged and then return 1 after having done so. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/es8328.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/es8328.c b/sound/soc/codecs/es8328.c index 3f00ead97006..dd53dfd87b04 100644 --- a/sound/soc/codecs/es8328.c +++ b/sound/soc/codecs/es8328.c @@ -161,13 +161,16 @@ static int es8328_put_deemph(struct snd_kcontrol *kcontrol, if (deemph > 1) return -EINVAL; + if (es8328->deemph == deemph) + return 0; + ret = es8328_set_deemph(component); if (ret < 0) return ret; es8328->deemph = deemph; - return 0; + return 1; } -- 2.30.2

1 1

[PATCH] ASoC: wm8962: Fix suspend while playing music
by Adam Ford 07 Jun '22

07 Jun '22

If the audio CODEC is playing sound when the system is suspended, it can be left in a state which throws the following error: wm8962 3-001a: ASoC: error at soc_component_read_no_lock on wm8962.3-001a: -16 Once this error has occurred, the audio will not work again until rebooted. Fix this by configuring SET_SYSTEM_SLEEP_PM_OPS. Signed-off-by: Adam Ford <aford173(a)gmail.com> diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 34cd5a2a997c..5cca89364280 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3868,6 +3868,7 @@ static int wm8962_runtime_suspend(struct device *dev) #endif static const struct dev_pm_ops wm8962_pm = { + SET_SYSTEM_SLEEP_PM_OPS(pm_runtime_force_suspend, pm_runtime_force_resume) SET_RUNTIME_PM_OPS(wm8962_runtime_suspend, wm8962_runtime_resume, NULL) }; -- 2.25.1

3 2

[PATCH] ASoC: ab8500: Remove some leftover from the "Replace GPLv2 boilerplate/reference with SPDX" rules
by Christophe JAILLET 07 Jun '22

07 Jun '22

The "Replace GPLv2 boilerplate/reference with SPDX" has left some empty "License terms" paragraphs. Remove them as well. Signed-off-by: Christophe JAILLET <christophe.jaillet(a)wanadoo.fr> --- sound/soc/codecs/ab8500-codec.c | 2 -- sound/soc/codecs/ab8500-codec.h | 2 -- 2 files changed, 4 deletions(-) diff --git a/sound/soc/codecs/ab8500-codec.c b/sound/soc/codecs/ab8500-codec.c index aefafb0b7b97..cbd4a92cb06c 100644 --- a/sound/soc/codecs/ab8500-codec.c +++ b/sound/soc/codecs/ab8500-codec.c @@ -12,8 +12,6 @@ * Mikko Sarmanne <mikko.sarmanne(a)symbio.com>, * Jarmo K. Kuronen <jarmo.kuronen(a)symbio.com>, * for ST-Ericsson. - * - * License terms: */ #include <linux/kernel.h> diff --git a/sound/soc/codecs/ab8500-codec.h b/sound/soc/codecs/ab8500-codec.h index 0ac87d0446c2..2a6f6409f1f8 100644 --- a/sound/soc/codecs/ab8500-codec.h +++ b/sound/soc/codecs/ab8500-codec.h @@ -11,8 +11,6 @@ * Mikko J. Lehto <mikko.lehto(a)symbio.com>, * Mikko Sarmanne <mikko.sarmanne(a)symbio.com>, * for ST-Ericsson. - * - * License terms: */ #ifndef AB8500_CODEC_REGISTERS_H -- 2.34.1

2 1

[PATCH] ASoC: ux500: Remove some leftover from the "Replace GPLv2 boilerplate/reference with SPDX" rules
by Christophe JAILLET 07 Jun '22

07 Jun '22

The "Replace GPLv2 boilerplate/reference with SPDX" has left some empty "License terms" paragraphs. Remove them as well. Signed-off-by: Christophe JAILLET <christophe.jaillet(a)wanadoo.fr> --- sound/soc/ux500/mop500.c | 2 -- sound/soc/ux500/mop500_ab8500.c | 2 -- sound/soc/ux500/mop500_ab8500.h | 2 -- sound/soc/ux500/ux500_msp_dai.c | 2 -- sound/soc/ux500/ux500_msp_dai.h | 2 -- sound/soc/ux500/ux500_msp_i2s.c | 2 -- sound/soc/ux500/ux500_msp_i2s.h | 2 -- sound/soc/ux500/ux500_pcm.c | 2 -- sound/soc/ux500/ux500_pcm.h | 2 -- 9 files changed, 18 deletions(-) diff --git a/sound/soc/ux500/mop500.c b/sound/soc/ux500/mop500.c index 4f41bb0ab2b0..fdd55d772b8e 100644 --- a/sound/soc/ux500/mop500.c +++ b/sound/soc/ux500/mop500.c @@ -4,8 +4,6 @@ * * Author: Ola Lilja (ola.o.lilja(a)stericsson.com) * for ST-Ericsson. - * - * License terms: */ #include <asm/mach-types.h> diff --git a/sound/soc/ux500/mop500_ab8500.c b/sound/soc/ux500/mop500_ab8500.c index 1ea1729984a9..e5e73a2bd9fe 100644 --- a/sound/soc/ux500/mop500_ab8500.c +++ b/sound/soc/ux500/mop500_ab8500.c @@ -5,8 +5,6 @@ * Author: Ola Lilja <ola.o.lilja(a)stericsson.com>, * Kristoffer Karlsson <kristoffer.karlsson(a)stericsson.com> * for ST-Ericsson. - * - * License terms: */ #include <linux/module.h> diff --git a/sound/soc/ux500/mop500_ab8500.h b/sound/soc/ux500/mop500_ab8500.h index 087ef246d87d..98de80a9cc4f 100644 --- a/sound/soc/ux500/mop500_ab8500.h +++ b/sound/soc/ux500/mop500_ab8500.h @@ -4,8 +4,6 @@ * * Author: Ola Lilja <ola.o.lilja(a)stericsson.com> * for ST-Ericsson. - * - * License terms: */ #ifndef MOP500_AB8500_H diff --git a/sound/soc/ux500/ux500_msp_dai.c b/sound/soc/ux500/ux500_msp_dai.c index 21052378a32e..56532b62faf3 100644 --- a/sound/soc/ux500/ux500_msp_dai.c +++ b/sound/soc/ux500/ux500_msp_dai.c @@ -5,8 +5,6 @@ * Author: Ola Lilja <ola.o.lilja(a)stericsson.com>, * Roger Nilsson <roger.xr.nilsson(a)stericsson.com> * for ST-Ericsson. - * - * License terms: */ #include <linux/module.h> diff --git a/sound/soc/ux500/ux500_msp_dai.h b/sound/soc/ux500/ux500_msp_dai.h index fcd4b26f5d2d..30bf70838196 100644 --- a/sound/soc/ux500/ux500_msp_dai.h +++ b/sound/soc/ux500/ux500_msp_dai.h @@ -5,8 +5,6 @@ * Author: Ola Lilja <ola.o.lilja(a)stericsson.com>, * Roger Nilsson <roger.xr.nilsson(a)stericsson.com> * for ST-Ericsson. - * - * License terms: */ #ifndef UX500_msp_dai_H diff --git a/sound/soc/ux500/ux500_msp_i2s.c b/sound/soc/ux500/ux500_msp_i2s.c index fd0b88bb7921..d113411a19f8 100644 --- a/sound/soc/ux500/ux500_msp_i2s.c +++ b/sound/soc/ux500/ux500_msp_i2s.c @@ -6,8 +6,6 @@ * Roger Nilsson <roger.xr.nilsson(a)stericsson.com>, * Sandeep Kaushik <sandeep.kaushik(a)st.com> * for ST-Ericsson. - * - * License terms: */ #include <linux/module.h> diff --git a/sound/soc/ux500/ux500_msp_i2s.h b/sound/soc/ux500/ux500_msp_i2s.h index 756b3973af9a..d45b5e2831cc 100644 --- a/sound/soc/ux500/ux500_msp_i2s.h +++ b/sound/soc/ux500/ux500_msp_i2s.h @@ -4,8 +4,6 @@ * * Author: Ola Lilja <ola.o.lilja(a)stericsson.com>, * for ST-Ericsson. - * - * License terms: */ diff --git a/sound/soc/ux500/ux500_pcm.c b/sound/soc/ux500/ux500_pcm.c index 18191084b8b8..d3802e5ef196 100644 --- a/sound/soc/ux500/ux500_pcm.c +++ b/sound/soc/ux500/ux500_pcm.c @@ -5,8 +5,6 @@ * Author: Ola Lilja <ola.o.lilja(a)stericsson.com>, * Roger Nilsson <roger.xr.nilsson(a)stericsson.com> * for ST-Ericsson. - * - * License terms: */ #include <asm/page.h> diff --git a/sound/soc/ux500/ux500_pcm.h b/sound/soc/ux500/ux500_pcm.h index ff3ef7223db6..bd4348ebf9a1 100644 --- a/sound/soc/ux500/ux500_pcm.h +++ b/sound/soc/ux500/ux500_pcm.h @@ -5,8 +5,6 @@ * Author: Ola Lilja <ola.o.lilja(a)stericsson.com>, * Roger Nilsson <roger.xr.nilsson(a)stericsson.com> * for ST-Ericsson. - * - * License terms: */ #ifndef UX500_PCM_H #define UX500_PCM_H -- 2.34.1

2 1

[PATCH] ASoC: simple-card-utils: rename asoc_simple_init_dai_link_params() to asoc_simple_init_for_codec2codec()
by Kuninori Morimoto 07 Jun '22

07 Jun '22

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> commit 95cfc0a0aaf5752071 ("ASoC: simple-card: Add support for codec2codec DAI links") added the function asoc_simple_init_dai_link_params() to initialize dai_link "params". It is very straight naming, but difficult to noticed that it is for Codec2Codec support. Handling Codec2Codec is one of very tricky part on ALSA SoC, thus it is very important to clarify it. This patch renames the function name. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/generic/simple-card-utils.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index 539d7f081bd7..fa080f166345 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -513,7 +513,7 @@ static int asoc_simple_init_dai(struct snd_soc_dai *dai, return 0; } -static int asoc_simple_init_dai_link_params(struct snd_soc_pcm_runtime *rtd, +static int asoc_simple_init_for_codec2codec(struct snd_soc_pcm_runtime *rtd, struct simple_dai_props *dai_props) { struct snd_soc_dai_link *dai_link = rtd->dai_link; @@ -575,7 +575,7 @@ int asoc_simple_dai_init(struct snd_soc_pcm_runtime *rtd) return ret; } - ret = asoc_simple_init_dai_link_params(rtd, props); + ret = asoc_simple_init_for_codec2codec(rtd, props); if (ret < 0) return ret; -- 2.25.1

2 1

[PATCH] ASoC: fsl_sai: Add support for i.MX8MN
by Shengjiu Wang 07 Jun '22

07 Jun '22

The SAI module on i.MX8MN is almost same as i.MX8MP, So reuse same soc data as i.MX8MP. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/fsl_sai.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index dae16a14f177..59a91a6d91a9 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -1299,6 +1299,7 @@ static const struct of_device_id fsl_sai_ids[] = { { .compatible = "fsl,imx8mm-sai", .data = &fsl_sai_imx8mm_data }, { .compatible = "fsl,imx8mp-sai", .data = &fsl_sai_imx8mp_data }, { .compatible = "fsl,imx8ulp-sai", .data = &fsl_sai_imx8ulp_data }, + { .compatible = "fsl,imx8mn-sai", .data = &fsl_sai_imx8mp_data }, { /* sentinel */ } }; MODULE_DEVICE_TABLE(of, fsl_sai_ids); -- 2.17.1

2 1

[PATCH] ASoC: codecs: da7210: add check for i2c_add_driver
by Jiasheng Jiang 07 Jun '22

07 Jun '22

As i2c_add_driver could return error if fails, it should be better to check the return value. However, if the CONFIG_I2C and CONFIG_SPI_MASTER are both true, the return value of i2c_add_driver will be covered by spi_register_driver. Therefore, it is necessary to add check and return error if fails. Fixes: aa0e25caafb7 ("ASoC: da7210: Add support for spi regmap") Signed-off-by: Jiasheng Jiang <jiasheng(a)iscas.ac.cn> --- sound/soc/codecs/da7210.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index 3fa3042e4424..76a21976ccdd 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -1335,6 +1335,8 @@ static int __init da7210_modinit(void) int ret = 0; #if IS_ENABLED(CONFIG_I2C) ret = i2c_add_driver(&da7210_i2c_driver); + if (ret) + return ret; #endif #if defined(CONFIG_SPI_MASTER) ret = spi_register_driver(&da7210_spi_driver); -- 2.25.1

2 1

[PATCH 1/3] ASoC: fsl_sai: use local device pointer
by Marco Felsch 07 Jun '22

07 Jun '22

Use a local variable to dereference the device pointer once and use the local variable in further calls. No functional changes. Signed-off-by: Marco Felsch <m.felsch(a)pengutronix.de> --- sound/soc/fsl/fsl_sai.c | 53 +++++++++++++++++++++-------------------- 1 file changed, 27 insertions(+), 26 deletions(-) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index fa950dde5310..a7637d602f3c 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -1004,6 +1004,7 @@ static int fsl_sai_runtime_resume(struct device *dev); static int fsl_sai_probe(struct platform_device *pdev) { struct device_node *np = pdev->dev.of_node; + struct device *dev = &pdev->dev; struct fsl_sai *sai; struct regmap *gpr; struct resource *res; @@ -1012,12 +1013,12 @@ static int fsl_sai_probe(struct platform_device *pdev) int irq, ret, i; int index; - sai = devm_kzalloc(&pdev->dev, sizeof(*sai), GFP_KERNEL); + sai = devm_kzalloc(dev, sizeof(*sai), GFP_KERNEL); if (!sai) return -ENOMEM; sai->pdev = pdev; - sai->soc_data = of_device_get_match_data(&pdev->dev); + sai->soc_data = of_device_get_match_data(dev); sai->is_lsb_first = of_property_read_bool(np, "lsb-first"); @@ -1032,18 +1033,18 @@ static int fsl_sai_probe(struct platform_device *pdev) ARRAY_SIZE(fsl_sai_reg_defaults_ofs8); } - sai->regmap = devm_regmap_init_mmio(&pdev->dev, base, &fsl_sai_regmap_config); + sai->regmap = devm_regmap_init_mmio(dev, base, &fsl_sai_regmap_config); if (IS_ERR(sai->regmap)) { - dev_err(&pdev->dev, "regmap init failed\n"); + dev_err(dev, "regmap init failed\n"); return PTR_ERR(sai->regmap); } - sai->bus_clk = devm_clk_get(&pdev->dev, "bus"); + sai->bus_clk = devm_clk_get(dev, "bus"); /* Compatible with old DTB cases */ if (IS_ERR(sai->bus_clk) && PTR_ERR(sai->bus_clk) != -EPROBE_DEFER) - sai->bus_clk = devm_clk_get(&pdev->dev, "sai"); + sai->bus_clk = devm_clk_get(dev, "sai"); if (IS_ERR(sai->bus_clk)) { - dev_err(&pdev->dev, "failed to get bus clock: %ld\n", + dev_err(dev, "failed to get bus clock: %ld\n", PTR_ERR(sai->bus_clk)); /* -EPROBE_DEFER */ return PTR_ERR(sai->bus_clk); @@ -1051,9 +1052,9 @@ static int fsl_sai_probe(struct platform_device *pdev) for (i = 1; i < FSL_SAI_MCLK_MAX; i++) { sprintf(tmp, "mclk%d", i); - sai->mclk_clk[i] = devm_clk_get(&pdev->dev, tmp); + sai->mclk_clk[i] = devm_clk_get(dev, tmp); if (IS_ERR(sai->mclk_clk[i])) { - dev_err(&pdev->dev, "failed to get mclk%d clock: %ld\n", + dev_err(dev, "failed to get mclk%d clock: %ld\n", i + 1, PTR_ERR(sai->mclk_clk[i])); sai->mclk_clk[i] = NULL; } @@ -1068,10 +1069,10 @@ static int fsl_sai_probe(struct platform_device *pdev) if (irq < 0) return irq; - ret = devm_request_irq(&pdev->dev, irq, fsl_sai_isr, IRQF_SHARED, + ret = devm_request_irq(dev, irq, fsl_sai_isr, IRQF_SHARED, np->name, sai); if (ret) { - dev_err(&pdev->dev, "failed to claim irq %u\n", irq); + dev_err(dev, "failed to claim irq %u\n", irq); return ret; } @@ -1088,7 +1089,7 @@ static int fsl_sai_probe(struct platform_device *pdev) if (of_find_property(np, "fsl,sai-synchronous-rx", NULL) && of_find_property(np, "fsl,sai-asynchronous", NULL)) { /* error out if both synchronous and asynchronous are present */ - dev_err(&pdev->dev, "invalid binding for synchronous mode\n"); + dev_err(dev, "invalid binding for synchronous mode\n"); return -EINVAL; } @@ -1109,7 +1110,7 @@ static int fsl_sai_probe(struct platform_device *pdev) of_device_is_compatible(np, "fsl,imx6ul-sai")) { gpr = syscon_regmap_lookup_by_compatible("fsl,imx6ul-iomuxc-gpr"); if (IS_ERR(gpr)) { - dev_err(&pdev->dev, "cannot find iomuxc registers\n"); + dev_err(dev, "cannot find iomuxc registers\n"); return PTR_ERR(gpr); } @@ -1127,23 +1128,23 @@ static int fsl_sai_probe(struct platform_device *pdev) sai->dma_params_tx.maxburst = FSL_SAI_MAXBURST_TX; platform_set_drvdata(pdev, sai); - pm_runtime_enable(&pdev->dev); - if (!pm_runtime_enabled(&pdev->dev)) { - ret = fsl_sai_runtime_resume(&pdev->dev); + pm_runtime_enable(dev); + if (!pm_runtime_enabled(dev)) { + ret = fsl_sai_runtime_resume(dev); if (ret) goto err_pm_disable; } - ret = pm_runtime_get_sync(&pdev->dev); + ret = pm_runtime_get_sync(dev); if (ret < 0) { - pm_runtime_put_noidle(&pdev->dev); + pm_runtime_put_noidle(dev); goto err_pm_get_sync; } /* Get sai version */ - ret = fsl_sai_check_version(&pdev->dev); + ret = fsl_sai_check_version(dev); if (ret < 0) - dev_warn(&pdev->dev, "Error reading SAI version: %d\n", ret); + dev_warn(dev, "Error reading SAI version: %d\n", ret); /* Select MCLK direction */ if (of_find_property(np, "fsl,sai-mclk-direction-output", NULL) && @@ -1152,7 +1153,7 @@ static int fsl_sai_probe(struct platform_device *pdev) FSL_SAI_MCTL_MCLK_EN, FSL_SAI_MCTL_MCLK_EN); } - ret = pm_runtime_put_sync(&pdev->dev); + ret = pm_runtime_put_sync(dev); if (ret < 0) goto err_pm_get_sync; @@ -1165,12 +1166,12 @@ static int fsl_sai_probe(struct platform_device *pdev) if (ret) goto err_pm_get_sync; } else { - ret = devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0); + ret = devm_snd_dmaengine_pcm_register(dev, NULL, 0); if (ret) goto err_pm_get_sync; } - ret = devm_snd_soc_register_component(&pdev->dev, &fsl_component, + ret = devm_snd_soc_register_component(dev, &fsl_component, &sai->cpu_dai_drv, 1); if (ret) goto err_pm_get_sync; @@ -1178,10 +1179,10 @@ static int fsl_sai_probe(struct platform_device *pdev) return ret; err_pm_get_sync: - if (!pm_runtime_status_suspended(&pdev->dev)) - fsl_sai_runtime_suspend(&pdev->dev); + if (!pm_runtime_status_suspended(dev)) + fsl_sai_runtime_suspend(dev); err_pm_disable: - pm_runtime_disable(&pdev->dev); + pm_runtime_disable(dev); return ret; } -- 2.30.2

3 6

[PATCH] ASoC: mediatek: mt8173: Fix refcount leak in mt8173_rt5650_rt5676_dev_probe
by Miaoqian Lin 07 Jun '22

07 Jun '22

of_parse_phandle() returns a node pointer with refcount incremented, we should use of_node_put() on it when not need anymore. Fix missing of_node_put() in error paths. Fixes: 94319ba10eca ("ASoC: mediatek: Use platform_of_node for machine drivers") Signed-off-by: Miaoqian Lin <linmq006(a)gmail.com> --- sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c | 10 +++++++--- 1 file changed, 7 insertions(+), 3 deletions(-) diff --git a/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c b/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c index 70bf312e855f..8794720cea3a 100644 --- a/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c +++ b/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c @@ -256,14 +256,16 @@ static int mt8173_rt5650_rt5676_dev_probe(struct platform_device *pdev) if (!mt8173_rt5650_rt5676_dais[DAI_LINK_CODEC_I2S].codecs[0].of_node) { dev_err(&pdev->dev, "Property 'audio-codec' missing or invalid\n"); - return -EINVAL; + ret = -EINVAL; + goto put_node; } mt8173_rt5650_rt5676_dais[DAI_LINK_CODEC_I2S].codecs[1].of_node = of_parse_phandle(pdev->dev.of_node, "mediatek,audio-codec", 1); if (!mt8173_rt5650_rt5676_dais[DAI_LINK_CODEC_I2S].codecs[1].of_node) { dev_err(&pdev->dev, "Property 'audio-codec' missing or invalid\n"); - return -EINVAL; + ret = -EINVAL; + goto put_node; } mt8173_rt5650_rt5676_codec_conf[0].dlc.of_node = mt8173_rt5650_rt5676_dais[DAI_LINK_CODEC_I2S].codecs[1].of_node; @@ -276,13 +278,15 @@ static int mt8173_rt5650_rt5676_dev_probe(struct platform_device *pdev) if (!mt8173_rt5650_rt5676_dais[DAI_LINK_HDMI_I2S].codecs->of_node) { dev_err(&pdev->dev, "Property 'audio-codec' missing or invalid\n"); - return -EINVAL; + ret = -EINVAL; + goto put_node; } card->dev = &pdev->dev; ret = devm_snd_soc_register_card(&pdev->dev, card); +put_node: of_node_put(platform_node); return ret; } -- 2.25.1

2 1

[PATCH] ASoC: mediatek: remove unnecessary check of clk_disable_unprepare
by cgel.zte＠gmail.com 07 Jun '22

07 Jun '22

From: Minghao Chi <chi.minghao(a)zte.com.cn> Because clk_disable_unprepare already checked NULL clock parameter, so the additional checks are unnecessary, just remove them. Reported-by: Zeal Robot <zealci(a)zte.com.cn> Signed-off-by: Minghao Chi <chi.minghao(a)zte.com.cn> --- sound/soc/mediatek/mt8173/mt8173-afe-pcm.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) diff --git a/sound/soc/mediatek/mt8173/mt8173-afe-pcm.c b/sound/soc/mediatek/mt8173/mt8173-afe-pcm.c index 31494930433f..dcaeeeb8aac7 100644 --- a/sound/soc/mediatek/mt8173/mt8173-afe-pcm.c +++ b/sound/soc/mediatek/mt8173/mt8173-afe-pcm.c @@ -286,10 +286,8 @@ static int mt8173_afe_dais_set_clks(struct mtk_base_afe *afe, static void mt8173_afe_dais_disable_clks(struct mtk_base_afe *afe, struct clk *m_ck, struct clk *b_ck) { - if (m_ck) - clk_disable_unprepare(m_ck); - if (b_ck) - clk_disable_unprepare(b_ck); + clk_disable_unprepare(m_ck); + clk_disable_unprepare(b_ck); } static int mt8173_afe_i2s_startup(struct snd_pcm_substream *substream, -- 2.25.1

2 1

[PATCH V4] ASoC: fsl_xcvr:Fix unbalanced pm_runtime_enable in fsl_xcvr_probe
by zhangqilong 07 Jun '22

07 Jun '22

a) Add missing pm_runtime_disable() when probe error out. It could avoid pm_runtime implementation complains when removing and probing again the driver. b) Add remove for missing pm_runtime_disable(). Fix:c590fa80b3928 ("ASoC: fsl_xcvr: register platform component before registering cpu dai") Signed-off-by: Zhang Qilong <zhangqilong3(a)huawei.com> --- v2: - Add remove to put PM usage counter. v3: - Modify the commit message. --- sound/soc/fsl/fsl_xcvr.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/soc/fsl/fsl_xcvr.c b/sound/soc/fsl/fsl_xcvr.c index d0556c79fdb1..55e640cba87d 100644 --- a/sound/soc/fsl/fsl_xcvr.c +++ b/sound/soc/fsl/fsl_xcvr.c @@ -1228,6 +1228,7 @@ static int fsl_xcvr_probe(struct platform_device *pdev) */ ret = devm_snd_dmaengine_pcm_register(dev, NULL, 0); if (ret) { + pm_runtime_disable(dev); dev_err(dev, "failed to pcm register\n"); return ret; } @@ -1235,6 +1236,7 @@ static int fsl_xcvr_probe(struct platform_device *pdev) ret = devm_snd_soc_register_component(dev, &fsl_xcvr_comp, &fsl_xcvr_dai, 1); if (ret) { + pm_runtime_disable(dev); dev_err(dev, "failed to register component %s\n", fsl_xcvr_comp.name); } @@ -1242,6 +1244,12 @@ static int fsl_xcvr_probe(struct platform_device *pdev) return ret; } +static int fsl_xcvr_remove(struct platform_device *pdev) +{ + pm_runtime_disable(&pdev->dev); + return 0; +} + static __maybe_unused int fsl_xcvr_runtime_suspend(struct device *dev) { struct fsl_xcvr *xcvr = dev_get_drvdata(dev); @@ -1370,6 +1378,7 @@ static struct platform_driver fsl_xcvr_driver = { .pm = &fsl_xcvr_pm_ops, .of_match_table = fsl_xcvr_dt_ids, }, + .remove = fsl_xcvr_remove, }; module_platform_driver(fsl_xcvr_driver); -- 2.31.1

3 2

[PATCH 1/2] ASoC: SOF: ipc-msg-injector: Propagate write errors correctly
by Dan Carpenter 07 Jun '22

07 Jun '22

This code is supposed to propagate errors from simple_write_to_buffer() or return -EFAULT if "size != count". However "size" needs to be signed for the code to work correctly and the case where "size == 0" is not handled correctly. Fixes: 066c67624d8c ("ASoC: SOF: ipc-msg-injector: Add support for IPC4 messages") Fixes: 2f0b1b013bbc ("ASoC: SOF: debug: Add support for IPC message injection") Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- sound/soc/sof/sof-client-ipc-msg-injector.c | 16 +++++++++++----- 1 file changed, 11 insertions(+), 5 deletions(-) diff --git a/sound/soc/sof/sof-client-ipc-msg-injector.c b/sound/soc/sof/sof-client-ipc-msg-injector.c index 03490a4d4ae7..030cb97d7713 100644 --- a/sound/soc/sof/sof-client-ipc-msg-injector.c +++ b/sound/soc/sof/sof-client-ipc-msg-injector.c @@ -150,7 +150,7 @@ static ssize_t sof_msg_inject_dfs_write(struct file *file, const char __user *bu { struct sof_client_dev *cdev = file->private_data; struct sof_msg_inject_priv *priv = cdev->data; - size_t size; + ssize_t size; int ret; if (*ppos) @@ -158,8 +158,10 @@ static ssize_t sof_msg_inject_dfs_write(struct file *file, const char __user *bu size = simple_write_to_buffer(priv->tx_buffer, priv->max_msg_size, ppos, buffer, count); + if (size < 0) + return size; if (size != count) - return size > 0 ? -EFAULT : size; + return -EFAULT; memset(priv->rx_buffer, 0, priv->max_msg_size); @@ -179,7 +181,7 @@ static ssize_t sof_msg_inject_ipc4_dfs_write(struct file *file, struct sof_client_dev *cdev = file->private_data; struct sof_msg_inject_priv *priv = cdev->data; struct sof_ipc4_msg *ipc4_msg = priv->tx_buffer; - size_t size; + ssize_t size; int ret; if (*ppos) @@ -192,8 +194,10 @@ static ssize_t sof_msg_inject_ipc4_dfs_write(struct file *file, size = simple_write_to_buffer(&ipc4_msg->header_u64, sizeof(ipc4_msg->header_u64), ppos, buffer, count); + if (size < 0) + return size; if (size != sizeof(ipc4_msg->header_u64)) - return size > 0 ? -EFAULT : size; + return -EFAULT; count -= size; if (!count) { @@ -201,8 +205,10 @@ static ssize_t sof_msg_inject_ipc4_dfs_write(struct file *file, size = simple_write_to_buffer(ipc4_msg->data_ptr, priv->max_msg_size, ppos, buffer, count); + if (size < 0) + return size; if (size != count) - return size > 0 ? -EFAULT : size; + return -EFAULT; } ipc4_msg->data_size = count; -- 2.35.1

3 4

[PATCH] ASoC: ops: Remove unneeded delay.h inclusion
by Mark Brown 07 Jun '22

07 Jun '22

The ops code does not do any sleeps or delays so does not need delay.h. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/soc-ops.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c index 0d373435ceb8..2d39370ddeca 100644 --- a/sound/soc/soc-ops.c +++ b/sound/soc/soc-ops.c @@ -14,7 +14,6 @@ #include <linux/module.h> #include <linux/moduleparam.h> #include <linux/init.h> -#include <linux/delay.h> #include <linux/pm.h> #include <linux/bitops.h> #include <linux/ctype.h> -- 2.30.2

1 1

[PATCH] ASoC: ops: Clarify snd_soc_info_volsw_sx()
by Mark Brown 07 Jun '22

07 Jun '22

Currently snd_soc_info_volsw_sx() is implemented indirectly, wrapping snd_soc_info_volsw() and modifying the values it sets up rather than directly setting up the values reported to userspace. This makes it much harder to follow what the intended behaviour of these controls is. Let's rewrite the function to be self contained with a clarifying comment at the top in an effort to help maintainability. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- Build tested only. sound/soc/soc-ops.c | 22 ++++++++++++++++------ 1 file changed, 16 insertions(+), 6 deletions(-) diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c index a0ca58ba1627..03010577046c 100644 --- a/sound/soc/soc-ops.c +++ b/sound/soc/soc-ops.c @@ -203,7 +203,8 @@ EXPORT_SYMBOL_GPL(snd_soc_info_volsw); * Callback to provide information about a single mixer control, or a double * mixer control that spans 2 registers of the SX TLV type. SX TLV controls * have a range that represents both positive and negative values either side - * of zero but without a sign bit. + * of zero but without a sign bit. min is the minimum register value, max is + * the number of steps. * * Returns 0 for success. */ @@ -212,12 +213,21 @@ int snd_soc_info_volsw_sx(struct snd_kcontrol *kcontrol, { struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; + int max; - snd_soc_info_volsw(kcontrol, uinfo); - /* Max represents the number of levels in an SX control not the - * maximum value, so add the minimum value back on - */ - uinfo->value.integer.max += mc->min; + if (mc->platform_max) + max = mc->platform_max; + else + max = mc->max; + + if (max == 1 && !strstr(kcontrol->id.name, " Volume")) + uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; + else + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + + uinfo->count = snd_soc_volsw_is_stereo(mc) ? 2 : 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = max; return 0; } -- 2.30.2

2 3

[PATCH] ASoC: cros_ec_codec: Fix refcount leak in cros_ec_codec_platform_probe
by Miaoqian Lin 07 Jun '22

07 Jun '22

of_parse_phandle() returns a node pointer with refcount incremented, we should use of_node_put() on it when not need anymore. Add missing of_node_put() to avoid refcount leak. Fixes: format:b6bc07d4360d ("ASoC: cros_ec_codec: support WoV") Signed-off-by: Miaoqian Lin <linmq006(a)gmail.com> --- sound/soc/codecs/cros_ec_codec.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/cros_ec_codec.c b/sound/soc/codecs/cros_ec_codec.c index 8b0a9c788a26..11e7b3f6d410 100644 --- a/sound/soc/codecs/cros_ec_codec.c +++ b/sound/soc/codecs/cros_ec_codec.c @@ -995,6 +995,7 @@ static int cros_ec_codec_platform_probe(struct platform_device *pdev) dev_dbg(dev, "ap_shm_phys_addr=%#llx len=%#x\n", priv->ap_shm_phys_addr, priv->ap_shm_len); } + of_node_put(node); } #endif -- 2.25.1

2 1

[PATCH] ASoC: max9860: Use modern ASoC DAI format terminology
by Mark Brown 07 Jun '22

07 Jun '22

As part of moving to remove the old style defines for the bus clocks update the max9860 driver to use more modern terminology for clocking. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/max9860.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/max9860.c b/sound/soc/codecs/max9860.c index 82f20a8e27ad..a1d0179e12c7 100644 --- a/sound/soc/codecs/max9860.c +++ b/sound/soc/codecs/max9860.c @@ -448,9 +448,9 @@ static int max9860_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) struct snd_soc_component *component = dai->component; struct max9860_priv *max9860 = snd_soc_component_get_drvdata(component); - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBM_CFM: - case SND_SOC_DAIFMT_CBS_CFS: + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_CBP_CFP: + case SND_SOC_DAIFMT_CBC_CFC: max9860->fmt = fmt; return 0; -- 2.30.2

2 2

[PATCH] ASoC: tfa9879: Use modern ASoC DAI format terminology
by Mark Brown 07 Jun '22

07 Jun '22

As part of moving to remove the old style defines for the bus clocks update the tfa9879 driver to use more modern terminology for clocking. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/tfa9879.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/tfa9879.c b/sound/soc/codecs/tfa9879.c index 3d8e8c2276f0..41a9b1b76e62 100644 --- a/sound/soc/codecs/tfa9879.c +++ b/sound/soc/codecs/tfa9879.c @@ -111,8 +111,8 @@ static int tfa9879_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) int i2s_set; int sck_pol; - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBS_CFS: + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_CBC_CFC: break; default: return -EINVAL; -- 2.30.2

2 2

[PATCH] ASoC: SOF: Fix potential NULL pointer dereference
by Amadeusz Sławiński 07 Jun '22

07 Jun '22

Cleanup path for sof_prepare_widgets_in_path() should check if unprepare callback exists before calling it, instead it checks if it does not exist. Fix the check. Fixes: 370014c8197a ("ASoC: SOF: Add a prepare op to IPC topology widget ops") Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> --- sound/soc/sof/sof-audio.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sof/sof-audio.c b/sound/soc/sof/sof-audio.c index cd41d0599cef..789d0a1f1a77 100644 --- a/sound/soc/sof/sof-audio.c +++ b/sound/soc/sof/sof-audio.c @@ -323,7 +323,7 @@ sof_prepare_widgets_in_path(struct snd_sof_dev *sdev, struct snd_soc_dapm_widget p->walking = false; if (ret < 0) { /* unprepare the source widget */ - if (!widget_ops[widget->id].ipc_unprepare && swidget->prepared) { + if (widget_ops[widget->id].ipc_unprepare && swidget->prepared) { widget_ops[widget->id].ipc_unprepare(swidget); swidget->prepared = false; } -- 2.25.1

4 4

[PATCH 1/2] ASoC: codecs: tfa989x: Add support for tfa9890
by Alexander Martinz 07 Jun '22

07 Jun '22

The initialization sequence is taken from the version provided by the supplier [1]. This allows speakers using the TFA9890 amplifier to work, which are used by various mobile phones such as the SHIFT6mq. [1]: https://source.codeaurora.org/external/mas/tfa98xx/tree/src/tfa_init.c?id=d… Signed-off-by: Alexander Martinz <amartinz(a)shiftphones.com> --- sound/soc/codecs/tfa989x.c | 30 ++++++++++++++++++++++++++++++ 1 file changed, 30 insertions(+) diff --git a/sound/soc/codecs/tfa989x.c b/sound/soc/codecs/tfa989x.c index dc86852752c5..8ab2656de750 100644 --- a/sound/soc/codecs/tfa989x.c +++ b/sound/soc/codecs/tfa989x.c @@ -40,12 +40,14 @@ #define TFA989X_I2S_SEL_REG 0x0a #define TFA989X_I2S_SEL_REG_SPKR_MSK GENMASK(10, 9) /* speaker impedance */ #define TFA989X_I2S_SEL_REG_DCFG_MSK GENMASK(14, 11) /* DCDC compensation */ +#define TFA989X_HIDE_UNHIDE_KEY 0x40 #define TFA989X_PWM_CONTROL 0x41 #define TFA989X_CURRENTSENSE1 0x46 #define TFA989X_CURRENTSENSE2 0x47 #define TFA989X_CURRENTSENSE3 0x48 #define TFA989X_CURRENTSENSE4 0x49 +#define TFA9890_REVISION 0x80 #define TFA9895_REVISION 0x12 #define TFA9897_REVISION 0x97 @@ -188,6 +190,33 @@ static struct snd_soc_dai_driver tfa989x_dai = { .ops = &tfa989x_dai_ops, }; +static int tfa9890_init(struct regmap *regmap) +{ + int ret; + + /* unhide keys to allow updating them */ + ret = regmap_write(regmap, TFA989X_HIDE_UNHIDE_KEY, 0x5a6b); + if (ret) + return ret; + + /* update PLL registers */ + ret = regmap_set_bits(regmap, 0x59, 0x3); + if (ret) + return ret; + + /* hide keys again */ + ret = regmap_write(regmap, TFA989X_HIDE_UNHIDE_KEY, 0x0000); + if (ret) + return ret; + + return regmap_write(regmap, TFA989X_CURRENTSENSE2, 0x7BE1); +} + +static const struct tfa989x_rev tfa9890_rev = { + .rev = TFA9890_REVISION, + .init = tfa9890_init, +}; + static const struct reg_sequence tfa9895_reg_init[] = { /* some other registers must be set for optimal amplifier behaviour */ { TFA989X_BAT_PROT, 0x13ab }, @@ -376,6 +405,7 @@ static int tfa989x_i2c_probe(struct i2c_client *i2c) } static const struct of_device_id tfa989x_of_match[] = { + { .compatible = "nxp,tfa9890", .data = &tfa9890_rev }, { .compatible = "nxp,tfa9895", .data = &tfa9895_rev }, { .compatible = "nxp,tfa9897", .data = &tfa9897_rev }, { } -- 2.36.1

4 5

[PATCH] ASoC: mt6797-mt6351: Fix refcount leak in mt6797_mt6351_dev_probe
by Miaoqian Lin 07 Jun '22

07 Jun '22

of_parse_phandle() returns a node pointer with refcount incremented, we should use of_node_put() on it when not need anymore. Add missing of_node_put() to avoid refcount leak. Fixes: format:f0ab0bf250da ("ASoC: add mt6797-mt6351 driver and config option") Signed-off-by: Miaoqian Lin <linmq006(a)gmail.com> --- sound/soc/mediatek/mt6797/mt6797-mt6351.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/soc/mediatek/mt6797/mt6797-mt6351.c b/sound/soc/mediatek/mt6797/mt6797-mt6351.c index 496f32bcfb5e..d2f6213a6bfc 100644 --- a/sound/soc/mediatek/mt6797/mt6797-mt6351.c +++ b/sound/soc/mediatek/mt6797/mt6797-mt6351.c @@ -217,7 +217,8 @@ static int mt6797_mt6351_dev_probe(struct platform_device *pdev) if (!codec_node) { dev_err(&pdev->dev, "Property 'audio-codec' missing or invalid\n"); - return -EINVAL; + ret = -EINVAL; + goto put_platform_node; } for_each_card_prelinks(card, i, dai_link) { if (dai_link->codecs->name) @@ -230,6 +231,9 @@ static int mt6797_mt6351_dev_probe(struct platform_device *pdev) dev_err(&pdev->dev, "%s snd_soc_register_card fail %d\n", __func__, ret); + of_node_put(codec_node); +put_platform_node: + of_node_put(platform_node); return ret; } -- 2.25.1

2 1

[PATCH] ASoC: nau8822: Don't reconfigure PLL to the same values
by Mark Brown 07 Jun '22

07 Jun '22

When we configure the PLL record the input and output frequency, then if we get asked to configure the same values again just skip reprogramming the hardware. This makes things a bit easier to use for machine drivers since it means they don't need to keep track of if they've programmed the PLL so much. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/nau8822.c | 7 +++++++ sound/soc/codecs/nau8822.h | 2 ++ 2 files changed, 9 insertions(+) diff --git a/sound/soc/codecs/nau8822.c b/sound/soc/codecs/nau8822.c index 08f6c56dc387..f4f68b549e1a 100644 --- a/sound/soc/codecs/nau8822.c +++ b/sound/soc/codecs/nau8822.c @@ -726,6 +726,10 @@ static int nau8822_set_pll(struct snd_soc_dai *dai, int pll_id, int source, struct nau8822_pll *pll_param = &nau8822->pll; int ret, fs; + if (freq_in == pll_param->freq_in && + freq_out == pll_param->freq_out) + return 0; + fs = freq_out / 256; ret = nau8822_calc_pll(freq_in, fs, pll_param); @@ -762,6 +766,9 @@ static int nau8822_set_pll(struct snd_soc_dai *dai, int pll_id, int source, snd_soc_component_update_bits(component, NAU8822_REG_POWER_MANAGEMENT_1, NAU8822_PLL_EN_MASK, NAU8822_PLL_ON); + pll_param->freq_in = freq_in; + pll_param->freq_out = freq_out; + return 0; } diff --git a/sound/soc/codecs/nau8822.h b/sound/soc/codecs/nau8822.h index b45d42c15de6..547ec057f853 100644 --- a/sound/soc/codecs/nau8822.h +++ b/sound/soc/codecs/nau8822.h @@ -198,6 +198,8 @@ struct nau8822_pll { int mclk_scaler; int pll_frac; int pll_int; + int freq_in; + int freq_out; }; /* Codec Private Data */ -- 2.30.2

1 1

[PATCH v2] ASoC: samsung: Fix error handling in aries_audio_probe
by Miaoqian Lin 07 Jun '22

07 Jun '22

of_get_child_by_name() returns a node pointer with refcount incremented, we should use of_node_put() on it when not need anymore. This function is missing of_node_put(cpu) in the error path. Fix this by goto out label. of_node_put() will check NULL pointer. Fixes: 7a3a7671fa6c ("ASoC: samsung: Add driver for Aries boards") Signed-off-by: Miaoqian Lin <linmq006(a)gmail.com> --- changes in v2: - update Fixes tag. v1 Link: https://lore.kernel.org/r/20220602130218.14901-1-linmq006@gmail.com --- sound/soc/samsung/aries_wm8994.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/samsung/aries_wm8994.c b/sound/soc/samsung/aries_wm8994.c index bb0cf4244e00..edee02d7f100 100644 --- a/sound/soc/samsung/aries_wm8994.c +++ b/sound/soc/samsung/aries_wm8994.c @@ -628,8 +628,10 @@ static int aries_audio_probe(struct platform_device *pdev) return -EINVAL; codec = of_get_child_by_name(dev->of_node, "codec"); - if (!codec) - return -EINVAL; + if (!codec) { + ret = -EINVAL; + goto out; + } for_each_card_prelinks(card, i, dai_link) { dai_link->codecs->of_node = of_parse_phandle(codec, -- 2.25.1

3 2

[PATCH v2] ASoC: cros_ec_codec: Fix refcount leak in cros_ec_codec_platform_probe
by Miaoqian Lin 07 Jun '22

07 Jun '22

of_parse_phandle() returns a node pointer with refcount incremented, we should use of_node_put() on it when not need anymore. Add missing of_node_put() to avoid refcount leak. Fixes: b6bc07d4360d ("ASoC: cros_ec_codec: support WoV") Signed-off-by: Miaoqian Lin <linmq006(a)gmail.com> --- changes in v2: - update Fixes tag v1 Link: https://lore.kernel.org/r/20220602114529.6303-1-linmq006@gmail.com --- sound/soc/codecs/cros_ec_codec.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/cros_ec_codec.c b/sound/soc/codecs/cros_ec_codec.c index 8b0a9c788a26..11e7b3f6d410 100644 --- a/sound/soc/codecs/cros_ec_codec.c +++ b/sound/soc/codecs/cros_ec_codec.c @@ -995,6 +995,7 @@ static int cros_ec_codec_platform_probe(struct platform_device *pdev) dev_dbg(dev, "ap_shm_phys_addr=%#llx len=%#x\n", priv->ap_shm_phys_addr, priv->ap_shm_len); } + of_node_put(node); } #endif -- 2.25.1

4 3

[PATCH] ASoC: max98390: use linux/gpio/consumer.h to fix build
by Randy Dunlap 07 Jun '22

07 Jun '22

Change the header file to fix build errors in max98390.c: ../sound/soc/codecs/max98390.c: In function 'max98390_i2c_probe': ../sound/soc/codecs/max98390.c:1076:22: error: implicit declaration of function 'devm_gpiod_get_optional'; did you mean 'devm_regulator_get_optional'? [-Werror=implicit-function-declaration] 1076 | reset_gpio = devm_gpiod_get_optional(&i2c->dev, ../sound/soc/codecs/max98390.c:1077:55: error: 'GPIOD_OUT_HIGH' undeclared (first use in this function); did you mean 'GPIOF_INIT_HIGH'? 1077 | "reset", GPIOD_OUT_HIGH); ../sound/soc/codecs/max98390.c:1077:55: note: each undeclared identifier is reported only once for each function it appears in ../sound/soc/codecs/max98390.c:1083:17: error: implicit declaration of function 'gpiod_set_value_cansleep'; did you mean 'gpio_set_value_cansleep'? [-Werror=implicit-function-declaration] 1083 | gpiod_set_value_cansleep(reset_gpio, 0); Fixes: 397ff0249606 ("ASoC: max98390: Add reset gpio control") Signed-off-by: Randy Dunlap <rdunlap(a)infradead.org> Reported-by: kernel test robot <lkp(a)intel.com> Cc: Steve Lee <steve.lee.analog(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: alsa-devel(a)alsa-project.org --- sound/soc/codecs/max98390.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) --- a/sound/soc/codecs/max98390.c +++ b/sound/soc/codecs/max98390.c @@ -10,7 +10,7 @@ #include <linux/cdev.h> #include <linux/dmi.h> #include <linux/firmware.h> -#include <linux/gpio.h> +#include <linux/gpio/consumer.h> #include <linux/i2c.h> #include <linux/module.h> #include <linux/of_gpio.h>

2 1

[PATCH] ASoC: wm_adsp: Fix event generation for wm_adsp_fw_put()
by Mark Brown 07 Jun '22

07 Jun '22

Currently wm_adsp_fw_put() returns 0 rather than 1 when updating the value of the control, meaning that no event is generated to userspace. Fix this by setting the default return value to 1, the code already exits early with a return value of 0 if the value is unchanged. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/wm_adsp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 7973a75cac05..6d7fd88243aa 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -333,7 +333,7 @@ int wm_adsp_fw_put(struct snd_kcontrol *kcontrol, struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; struct wm_adsp *dsp = snd_soc_component_get_drvdata(component); - int ret = 0; + int ret = 1; if (ucontrol->value.enumerated.item[0] == dsp[e->shift_l].fw) return 0; -- 2.30.2

2 2

[PATCH] ASoC: imx-audmux: Silence a clang warning
by Fabio Estevam 07 Jun '22

07 Jun '22

Change the of_device_get_match_data() cast to (uintptr_t) to silence the following clang warning: sound/soc/fsl/imx-audmux.c:301:16: warning: cast to smaller integer type 'enum imx_audmux_type' from 'const void *' [-Wvoid-pointer-to-enum-cast] Reported-by: kernel test robot <lkp(a)intel.com> Fixes: 6a8b8b582db1 ("ASoC: imx-audmux: Remove unused .id_table") Signed-off-by: Fabio Estevam <festevam(a)gmail.com> --- sound/soc/fsl/imx-audmux.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/imx-audmux.c b/sound/soc/fsl/imx-audmux.c index dfa05d40b276..a8e5e0f57faf 100644 --- a/sound/soc/fsl/imx-audmux.c +++ b/sound/soc/fsl/imx-audmux.c @@ -298,7 +298,7 @@ static int imx_audmux_probe(struct platform_device *pdev) audmux_clk = NULL; } - audmux_type = (enum imx_audmux_type)of_device_get_match_data(&pdev->dev); + audmux_type = (uintptr_t)of_device_get_match_data(&pdev->dev); switch (audmux_type) { case IMX31_AUDMUX: -- 2.25.1

3 2

[PATCH] ASoC: fsl_mqs: simplify the code with adding fsl_mqs_soc_data
by Shengjiu Wang 07 Jun '22

07 Jun '22

Add soc specific data struct fsl_mqs_soc_data, move the definition of control register, each function bits to it, then the code can be simplified. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/fsl_mqs.c | 119 ++++++++++++++++++++++++---------------- 1 file changed, 71 insertions(+), 48 deletions(-) diff --git a/sound/soc/fsl/fsl_mqs.c b/sound/soc/fsl/fsl_mqs.c index ceaecbe3a25e..8a8d727319d6 100644 --- a/sound/soc/fsl/fsl_mqs.c +++ b/sound/soc/fsl/fsl_mqs.c @@ -10,6 +10,7 @@ #include <linux/moduleparam.h> #include <linux/mfd/syscon.h> #include <linux/mfd/syscon/imx6q-iomuxc-gpr.h> +#include <linux/of_device.h> #include <linux/pm_runtime.h> #include <linux/of.h> #include <linux/pm.h> @@ -29,15 +30,41 @@ #define MQS_CLK_DIV_MASK (0xFF << 0) #define MQS_CLK_DIV_SHIFT (0) +/** + * struct fsl_mqs_soc_data - soc specific data + * + * @use_gpr: control register is in General Purpose Register group + * @ctrl_off: control register offset + * @en_mask: enable bit mask + * @en_shift: enable bit shift + * @rst_mask: reset bit mask + * @rst_shift: reset bit shift + * @osr_mask: oversample bit mask + * @osr_shift: oversample bit shift + * @div_mask: clock divider mask + * @div_shift: clock divider bit shift + */ +struct fsl_mqs_soc_data { + bool use_gpr; + int ctrl_off; + int en_mask; + int en_shift; + int rst_mask; + int rst_shift; + int osr_mask; + int osr_shift; + int div_mask; + int div_shift; +}; + /* codec private data */ struct fsl_mqs { struct regmap *regmap; struct clk *mclk; struct clk *ipg; + const struct fsl_mqs_soc_data *soc; - unsigned int reg_iomuxc_gpr2; unsigned int reg_mqs_ctrl; - bool use_gpr; }; #define FSL_MQS_RATES (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000) @@ -65,19 +92,11 @@ static int fsl_mqs_hw_params(struct snd_pcm_substream *substream, res = mclk_rate % (32 * lrclk * 2 * 8); if (res == 0 && div > 0 && div <= 256) { - if (mqs_priv->use_gpr) { - regmap_update_bits(mqs_priv->regmap, IOMUXC_GPR2, - IMX6SX_GPR2_MQS_CLK_DIV_MASK, - (div - 1) << IMX6SX_GPR2_MQS_CLK_DIV_SHIFT); - regmap_update_bits(mqs_priv->regmap, IOMUXC_GPR2, - IMX6SX_GPR2_MQS_OVERSAMPLE_MASK, 0); - } else { - regmap_update_bits(mqs_priv->regmap, REG_MQS_CTRL, - MQS_CLK_DIV_MASK, - (div - 1) << MQS_CLK_DIV_SHIFT); - regmap_update_bits(mqs_priv->regmap, REG_MQS_CTRL, - MQS_OVERSAMPLE_MASK, 0); - } + regmap_update_bits(mqs_priv->regmap, mqs_priv->soc->ctrl_off, + mqs_priv->soc->div_mask, + (div - 1) << mqs_priv->soc->div_shift); + regmap_update_bits(mqs_priv->regmap, mqs_priv->soc->ctrl_off, + mqs_priv->soc->osr_mask, 0); } else { dev_err(component->dev, "can't get proper divider\n"); } @@ -118,14 +137,9 @@ static int fsl_mqs_startup(struct snd_pcm_substream *substream, struct snd_soc_component *component = dai->component; struct fsl_mqs *mqs_priv = snd_soc_component_get_drvdata(component); - if (mqs_priv->use_gpr) - regmap_update_bits(mqs_priv->regmap, IOMUXC_GPR2, - IMX6SX_GPR2_MQS_EN_MASK, - 1 << IMX6SX_GPR2_MQS_EN_SHIFT); - else - regmap_update_bits(mqs_priv->regmap, REG_MQS_CTRL, - MQS_EN_MASK, - 1 << MQS_EN_SHIFT); + regmap_update_bits(mqs_priv->regmap, mqs_priv->soc->ctrl_off, + mqs_priv->soc->en_mask, + 1 << mqs_priv->soc->en_shift); return 0; } @@ -135,12 +149,8 @@ static void fsl_mqs_shutdown(struct snd_pcm_substream *substream, struct snd_soc_component *component = dai->component; struct fsl_mqs *mqs_priv = snd_soc_component_get_drvdata(component); - if (mqs_priv->use_gpr) - regmap_update_bits(mqs_priv->regmap, IOMUXC_GPR2, - IMX6SX_GPR2_MQS_EN_MASK, 0); - else - regmap_update_bits(mqs_priv->regmap, REG_MQS_CTRL, - MQS_EN_MASK, 0); + regmap_update_bits(mqs_priv->regmap, mqs_priv->soc->ctrl_off, + mqs_priv->soc->en_mask, 0); } static const struct snd_soc_component_driver soc_codec_fsl_mqs = { @@ -191,12 +201,9 @@ static int fsl_mqs_probe(struct platform_device *pdev) * But in i.MX8QM/i.MX8QXP the control register is moved * to its own domain. */ - if (of_device_is_compatible(np, "fsl,imx8qm-mqs")) - mqs_priv->use_gpr = false; - else - mqs_priv->use_gpr = true; + mqs_priv->soc = of_device_get_match_data(&pdev->dev); - if (mqs_priv->use_gpr) { + if (mqs_priv->soc->use_gpr) { gpr_np = of_parse_phandle(np, "gpr", 0); if (!gpr_np) { dev_err(&pdev->dev, "failed to get gpr node by phandle\n"); @@ -280,12 +287,7 @@ static int fsl_mqs_runtime_resume(struct device *dev) return ret; } - if (mqs_priv->use_gpr) - regmap_write(mqs_priv->regmap, IOMUXC_GPR2, - mqs_priv->reg_iomuxc_gpr2); - else - regmap_write(mqs_priv->regmap, REG_MQS_CTRL, - mqs_priv->reg_mqs_ctrl); + regmap_write(mqs_priv->regmap, mqs_priv->soc->ctrl_off, mqs_priv->reg_mqs_ctrl); return 0; } @@ -293,12 +295,7 @@ static int fsl_mqs_runtime_suspend(struct device *dev) { struct fsl_mqs *mqs_priv = dev_get_drvdata(dev); - if (mqs_priv->use_gpr) - regmap_read(mqs_priv->regmap, IOMUXC_GPR2, - &mqs_priv->reg_iomuxc_gpr2); - else - regmap_read(mqs_priv->regmap, REG_MQS_CTRL, - &mqs_priv->reg_mqs_ctrl); + regmap_read(mqs_priv->regmap, mqs_priv->soc->ctrl_off, &mqs_priv->reg_mqs_ctrl); clk_disable_unprepare(mqs_priv->mclk); clk_disable_unprepare(mqs_priv->ipg); @@ -315,9 +312,35 @@ static const struct dev_pm_ops fsl_mqs_pm_ops = { pm_runtime_force_resume) }; +static const struct fsl_mqs_soc_data fsl_mqs_imx8qm_data = { + .use_gpr = false, + .ctrl_off = REG_MQS_CTRL, + .en_mask = MQS_EN_MASK, + .en_shift = MQS_EN_SHIFT, + .rst_mask = MQS_SW_RST_MASK, + .rst_shift = MQS_SW_RST_SHIFT, + .osr_mask = MQS_OVERSAMPLE_MASK, + .osr_shift = MQS_OVERSAMPLE_SHIFT, + .div_mask = MQS_CLK_DIV_MASK, + .div_shift = MQS_CLK_DIV_SHIFT, +}; + +static const struct fsl_mqs_soc_data fsl_mqs_imx6sx_data = { + .use_gpr = true, + .ctrl_off = IOMUXC_GPR2, + .en_mask = IMX6SX_GPR2_MQS_EN_MASK, + .en_shift = IMX6SX_GPR2_MQS_EN_SHIFT, + .rst_mask = IMX6SX_GPR2_MQS_SW_RST_MASK, + .rst_shift = IMX6SX_GPR2_MQS_SW_RST_SHIFT, + .osr_mask = IMX6SX_GPR2_MQS_OVERSAMPLE_MASK, + .osr_shift = IMX6SX_GPR2_MQS_OVERSAMPLE_SHIFT, + .div_mask = IMX6SX_GPR2_MQS_CLK_DIV_MASK, + .div_shift = IMX6SX_GPR2_MQS_CLK_DIV_SHIFT, +}; + static const struct of_device_id fsl_mqs_dt_ids[] = { - { .compatible = "fsl,imx8qm-mqs", }, - { .compatible = "fsl,imx6sx-mqs", }, + { .compatible = "fsl,imx8qm-mqs", .data = &fsl_mqs_imx8qm_data }, + { .compatible = "fsl,imx6sx-mqs", .data = &fsl_mqs_imx6sx_data }, {} }; MODULE_DEVICE_TABLE(of, fsl_mqs_dt_ids); -- 2.17.1

2 1

[PATCH] ASoC: cs35l45: Make bmc150_magn_remove() return void
by Uwe Kleine-König 07 Jun '22

07 Jun '22

cs35l45_remove() always returns zero. Make it return no value which makes it easier to see in the callers that there is no error to handle. Also the return value of i2c driver remove callbacks is ignored anyway. This prepares making i2c remove callbacks return void, too. Signed-off-by: Uwe Kleine-König <u.kleine-koenig(a)pengutronix.de> --- sound/soc/codecs/cs35l45-i2c.c | 4 +++- sound/soc/codecs/cs35l45.c | 4 +--- sound/soc/codecs/cs35l45.h | 2 +- 3 files changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/cs35l45-i2c.c b/sound/soc/codecs/cs35l45-i2c.c index 18ebf1e14b91..53a57320d0ab 100644 --- a/sound/soc/codecs/cs35l45-i2c.c +++ b/sound/soc/codecs/cs35l45-i2c.c @@ -40,7 +40,9 @@ static int cs35l45_i2c_remove(struct i2c_client *client) { struct cs35l45_private *cs35l45 = i2c_get_clientdata(client); - return cs35l45_remove(cs35l45); + cs35l45_remove(cs35l45); + + return 0; } static const struct of_device_id cs35l45_of_match[] = { diff --git a/sound/soc/codecs/cs35l45.c b/sound/soc/codecs/cs35l45.c index 7822626d3693..64a3abe14010 100644 --- a/sound/soc/codecs/cs35l45.c +++ b/sound/soc/codecs/cs35l45.c @@ -669,7 +669,7 @@ int cs35l45_probe(struct cs35l45_private *cs35l45) } EXPORT_SYMBOL_GPL(cs35l45_probe); -int cs35l45_remove(struct cs35l45_private *cs35l45) +void cs35l45_remove(struct cs35l45_private *cs35l45) { pm_runtime_disable(cs35l45->dev); @@ -677,8 +677,6 @@ int cs35l45_remove(struct cs35l45_private *cs35l45) regulator_disable(cs35l45->vdd_a); /* VDD_BATT must be the last to power-off */ regulator_disable(cs35l45->vdd_batt); - - return 0; } EXPORT_SYMBOL_GPL(cs35l45_remove); diff --git a/sound/soc/codecs/cs35l45.h b/sound/soc/codecs/cs35l45.h index 4e266d19cd1c..680891bcfce9 100644 --- a/sound/soc/codecs/cs35l45.h +++ b/sound/soc/codecs/cs35l45.h @@ -212,6 +212,6 @@ extern const struct regmap_config cs35l45_spi_regmap; int cs35l45_apply_patch(struct cs35l45_private *cs43l45); unsigned int cs35l45_get_clk_freq_id(unsigned int freq); int cs35l45_probe(struct cs35l45_private *cs35l45); -int cs35l45_remove(struct cs35l45_private *cs35l45); +void cs35l45_remove(struct cs35l45_private *cs35l45); #endif /* CS35L45_H */ base-commit: 0d463d016000d68d7e982720b5e4380b2d83409a -- 2.35.1

4 5

[PATCH] ASoC: amd: acp: fix typo in comment
by Julia Lawall 07 Jun '22

07 Jun '22

Spelling mistake (triple letters) in comment. Detected with the help of Coccinelle. Signed-off-by: Julia Lawall <Julia.Lawall(a)inria.fr> --- sound/soc/amd/acp/acp-pdm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/amd/acp/acp-pdm.c b/sound/soc/amd/acp/acp-pdm.c index 424c6e0bb9d6..7a0b26a30051 100644 --- a/sound/soc/amd/acp/acp-pdm.c +++ b/sound/soc/amd/acp/acp-pdm.c @@ -174,7 +174,7 @@ static void acp_dmic_dai_shutdown(struct snd_pcm_substream *substream, struct acp_dev_data *adata = dev_get_drvdata(dev); u32 ext_int_ctrl; - /* Disable DMIC interrrupts */ + /* Disable DMIC interrupts */ ext_int_ctrl = readl(adata->acp_base + ACP_EXTERNAL_INTR_CNTL); ext_int_ctrl |= ~PDM_DMA_INTR_MASK; writel(ext_int_ctrl, adata->acp_base + ACP_EXTERNAL_INTR_CNTL);

2 1

[PATCH] ASoC: stm32: dfsdm: fix typo in comment
by Julia Lawall 07 Jun '22

07 Jun '22

Spelling mistake (triple letters) in comment. Detected with the help of Coccinelle. Signed-off-by: Julia Lawall <Julia.Lawall(a)inria.fr> --- sound/soc/stm/stm32_adfsdm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/stm/stm32_adfsdm.c b/sound/soc/stm/stm32_adfsdm.c index 6ee714542b84..122805160e70 100644 --- a/sound/soc/stm/stm32_adfsdm.c +++ b/sound/soc/stm/stm32_adfsdm.c @@ -296,7 +296,7 @@ static int stm32_adfsdm_pcm_new(struct snd_soc_component *component, static int stm32_adfsdm_dummy_cb(const void *data, void *private) { /* - * This dummmy callback is requested by iio_channel_get_all_cb() API, + * This dummy callback is requested by iio_channel_get_all_cb() API, * but the stm32_dfsdm_get_buff_cb() API is used instead, to optimize * DMA transfers. */

2 1

[PATCH v2] ASoC: qcom: soundwire: Add support for controlling audio CGCR from HLOS
by Srinivasa Rao Mandadapu 07 Jun '22

07 Jun '22

Add support for controlling soundwire audio CGCR interface using clock framework to make hclk ungating with software. As per new hardware changes, software has to always ungate hclk if soundwire is operational and keep it running. This requirement is for latest LPASS chipsets for RX, TX and WSA path to work. Signed-off-by: Srinivasa Rao Mandadapu <quic_srivasam(a)quicinc.com> --- This patch set depends on: -- Clock driver patches for CGCR reset control support. --- https://patchwork.kernel.org/project/linux-arm-msm/list/?series=638002 --- https://patchwork.kernel.org/project/linux-arm-msm/list/?series=637998 Changes since v1: -- Add audio cgcr reset control in runtime PM resume handler. -- Update dependency list. drivers/soundwire/qcom.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/drivers/soundwire/qcom.c b/drivers/soundwire/qcom.c index da1ad7e..445e481 100644 --- a/drivers/soundwire/qcom.c +++ b/drivers/soundwire/qcom.c @@ -13,6 +13,7 @@ #include <linux/of_device.h> #include <linux/pm_runtime.h> #include <linux/regmap.h> +#include <linux/reset.h> #include <linux/slab.h> #include <linux/pm_wakeirq.h> #include <linux/slimbus.h> @@ -142,6 +143,7 @@ struct qcom_swrm_ctrl { struct device *dev; struct regmap *regmap; void __iomem *mmio; + struct reset_control *audio_cgcr; #ifdef CONFIG_DEBUG_FS struct dentry *debugfs; #endif @@ -656,6 +658,8 @@ static int qcom_swrm_init(struct qcom_swrm_ctrl *ctrl) val = FIELD_PREP(SWRM_MCP_FRAME_CTRL_BANK_ROW_CTRL_BMSK, ctrl->rows_index); val |= FIELD_PREP(SWRM_MCP_FRAME_CTRL_BANK_COL_CTRL_BMSK, ctrl->cols_index); + reset_control_reset(ctrl->audio_cgcr); + ctrl->reg_write(ctrl, SWRM_MCP_FRAME_CTRL_BANK_ADDR(0), val); /* Enable Auto enumeration */ @@ -1333,6 +1337,10 @@ static int qcom_swrm_probe(struct platform_device *pdev) ctrl->bus.compute_params = &qcom_swrm_compute_params; ctrl->bus.clk_stop_timeout = 300; + ctrl->audio_cgcr = devm_reset_control_get_exclusive(dev, "swr_audio_cgcr"); + if (IS_ERR(ctrl->audio_cgcr)) + dev_err(dev, "Failed to get audio_cgcr reset required for soundwire-v1.6.0\n"); + ret = qcom_swrm_get_port_config(ctrl); if (ret) goto err_clk; @@ -1486,6 +1494,8 @@ static int __maybe_unused swrm_runtime_resume(struct device *dev) qcom_swrm_get_device_status(ctrl); sdw_handle_slave_status(&ctrl->bus, ctrl->status); } else { + reset_control_reset(ctrl->audio_cgcr); + ctrl->reg_write(ctrl, SWRM_MCP_BUS_CTRL, SWRM_MCP_BUS_CLK_START); ctrl->reg_write(ctrl, SWRM_INTERRUPT_CLEAR, SWRM_INTERRUPT_STATUS_MASTER_CLASH_DET); -- 2.7.4

5 9

[PATCH] ASoC: fsl_asrc_dma: enable dual fifo for ASRC P2P
by Shengjiu Wang 07 Jun '22

07 Jun '22

The SSI and SPDIF has dual fifos, enhance P2P for these case with using the sdma_peripheral_config struct Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/fsl_asrc_dma.c | 14 ++++++++++++++ 1 file changed, 14 insertions(+) diff --git a/sound/soc/fsl/fsl_asrc_dma.c b/sound/soc/fsl/fsl_asrc_dma.c index 5038faf035cb..aaf7993935b7 100644 --- a/sound/soc/fsl/fsl_asrc_dma.c +++ b/sound/soc/fsl/fsl_asrc_dma.c @@ -129,6 +129,7 @@ static int fsl_asrc_dma_hw_params(struct snd_soc_component *component, struct snd_pcm_hw_params *params) { enum dma_slave_buswidth buswidth = DMA_SLAVE_BUSWIDTH_2_BYTES; + enum sdma_peripheral_type be_peripheral_type = IMX_DMATYPE_SSI; struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; struct snd_dmaengine_dai_dma_data *dma_params_fe = NULL; @@ -139,6 +140,7 @@ static int fsl_asrc_dma_hw_params(struct snd_soc_component *component, struct snd_soc_component *component_be = NULL; struct fsl_asrc *asrc = pair->asrc; struct dma_slave_config config_fe, config_be; + struct sdma_peripheral_config audio_config; enum asrc_pair_index index = pair->index; struct device *dev = component->dev; struct device_node *of_dma_node; @@ -221,6 +223,7 @@ static int fsl_asrc_dma_hw_params(struct snd_soc_component *component, /* Get DMA request of Back-End */ tmp_data = tmp_chan->private; pair->dma_data.dma_request = tmp_data->dma_request; + be_peripheral_type = tmp_data->peripheral_type; if (!be_chan) dma_release_channel(tmp_chan); @@ -268,6 +271,17 @@ static int fsl_asrc_dma_hw_params(struct snd_soc_component *component, config_be.dst_addr_width = buswidth; config_be.dst_maxburst = dma_params_be->maxburst; + memset(&audio_config, 0, sizeof(audio_config)); + config_be.peripheral_config = &audio_config; + config_be.peripheral_size = sizeof(audio_config); + + if (tx && (be_peripheral_type == IMX_DMATYPE_SSI_DUAL || + be_peripheral_type == IMX_DMATYPE_SPDIF)) + audio_config.n_fifos_dst = 2; + if (!tx && (be_peripheral_type == IMX_DMATYPE_SSI_DUAL || + be_peripheral_type == IMX_DMATYPE_SPDIF)) + audio_config.n_fifos_src = 2; + if (tx) { config_be.src_addr = asrc->paddr + asrc->get_fifo_addr(OUT, index); config_be.dst_addr = dma_params_be->addr; -- 2.17.1

2 1

[PATCH] ALSA: hda/realtek - Add HW8326 support
by huangwenhui 07 Jun '22

07 Jun '22

Added the support of new Huawei codec HW8326. Signed-off-by: huangwenhui <huangwenhuia(a)uniontech.com> --- sound/hda/hdac_device.c | 1 + sound/pci/hda/patch_realtek.c | 14 ++++++++++++++ 2 files changed, 15 insertions(+) diff --git a/sound/hda/hdac_device.c b/sound/hda/hdac_device.c index 3e9e9ac804f6..775474f00141 100644 --- a/sound/hda/hdac_device.c +++ b/sound/hda/hdac_device.c @@ -651,6 +651,7 @@ static const struct hda_vendor_id hda_vendor_ids[] = { { 0x1095, "Silicon Image" }, { 0x10de, "Nvidia" }, { 0x10ec, "Realtek" }, + { 0x19e5, "Huawei" }, { 0x1102, "Creative" }, { 0x1106, "VIA" }, { 0x111d, "IDT" }, diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index f3ad454b3fbf..0f32c0d17a3f 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -443,6 +443,7 @@ static void alc_fill_eapd_coef(struct hda_codec *codec) case 0x10ec0245: case 0x10ec0255: case 0x10ec0256: + case 0x19e58326: case 0x10ec0257: case 0x10ec0282: case 0x10ec0283: @@ -580,6 +581,7 @@ static void alc_shutup_pins(struct hda_codec *codec) switch (codec->core.vendor_id) { case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: case 0x10ec0283: case 0x10ec0286: case 0x10ec0288: @@ -3247,6 +3249,7 @@ static void alc_disable_headset_jack_key(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_write_coef_idx(codec, 0x48, 0x0); alc_update_coef_idx(codec, 0x49, 0x0045, 0x0); break; @@ -3275,6 +3278,7 @@ static void alc_enable_headset_jack_key(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_write_coef_idx(codec, 0x48, 0xd011); alc_update_coef_idx(codec, 0x49, 0x007f, 0x0045); break; @@ -4910,6 +4914,7 @@ static void alc_headset_mode_unplugged(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_process_coef_fw(codec, coef0256); break; case 0x10ec0234: @@ -5025,6 +5030,7 @@ static void alc_headset_mode_mic_in(struct hda_codec *codec, hda_nid_t hp_pin, case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_write_coef_idx(codec, 0x45, 0xc489); snd_hda_set_pin_ctl_cache(codec, hp_pin, 0); alc_process_coef_fw(codec, coef0256); @@ -5175,6 +5181,7 @@ static void alc_headset_mode_default(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_write_coef_idx(codec, 0x1b, 0x0e4b); alc_write_coef_idx(codec, 0x45, 0xc089); msleep(50); @@ -5274,6 +5281,7 @@ static void alc_headset_mode_ctia(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_process_coef_fw(codec, coef0256); break; case 0x10ec0234: @@ -5388,6 +5396,7 @@ static void alc_headset_mode_omtp(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_process_coef_fw(codec, coef0256); break; case 0x10ec0234: @@ -5489,6 +5498,7 @@ static void alc_determine_headset_type(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_write_coef_idx(codec, 0x1b, 0x0e4b); alc_write_coef_idx(codec, 0x06, 0x6104); alc_write_coefex_idx(codec, 0x57, 0x3, 0x09a3); @@ -5783,6 +5793,7 @@ static void alc255_set_default_jack_type(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_process_coef_fw(codec, alc256fw); break; } @@ -6385,6 +6396,7 @@ static void alc_combo_jack_hp_jd_restart(struct hda_codec *codec) case 0x10ec0236: case 0x10ec0255: case 0x10ec0256: + case 0x19e58326: alc_update_coef_idx(codec, 0x1b, 0x8000, 1 << 15); /* Reset HP JD */ alc_update_coef_idx(codec, 0x1b, 0x8000, 0 << 15); break; @@ -10095,6 +10107,7 @@ static int patch_alc269(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: spec->codec_variant = ALC269_TYPE_ALC256; spec->shutup = alc256_shutup; spec->init_hook = alc256_init; @@ -11479,6 +11492,7 @@ static const struct hda_device_id snd_hda_id_realtek[] = { HDA_CODEC_ENTRY(0x10ec0236, "ALC236", patch_alc269), HDA_CODEC_ENTRY(0x10ec0245, "ALC245", patch_alc269), HDA_CODEC_ENTRY(0x10ec0255, "ALC255", patch_alc269), + HDA_CODEC_ENTRY(0x19e58326, "HW8326", patch_alc269), HDA_CODEC_ENTRY(0x10ec0256, "ALC256", patch_alc269), HDA_CODEC_ENTRY(0x10ec0257, "ALC257", patch_alc269), HDA_CODEC_ENTRY(0x10ec0260, "ALC260", patch_alc260), -- 2.20.1

2 2

[PATCH] ALSA: hda/conexant - Fix loopback issue with CX20632
by huangwenhui 07 Jun '22

07 Jun '22

On a machine with CX20632,Alsamixer doesn't have 'Loopback Mixing' and 'Line'. Signed-off-by: huangwenhui <huangwenhuia(a)uniontech.com> --- sound/pci/hda/patch_conexant.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index aa360a0af284..1248d1a51cf0 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -1052,6 +1052,13 @@ static int patch_conexant_auto(struct hda_codec *codec) snd_hda_pick_fixup(codec, cxt5051_fixup_models, cxt5051_fixups, cxt_fixups); break; + case 0x14f15098: + codec->pin_amp_workaround = 1; + spec->gen.mixer_nid = 0x22; + spec->gen.add_stereo_mix_input = HDA_HINT_STEREO_MIX_AUTO; + snd_hda_pick_fixup(codec, cxt5066_fixup_models, + cxt5066_fixups, cxt_fixups); + break; case 0x14f150f2: codec->power_save_node = 1; fallthrough; -- 2.20.1

2 1

[PATCH] ALSA: hda: MTL: add HD Audio PCI ID and HDMI codec vendor ID
by Pierre-Louis Bossart 07 Jun '22

07 Jun '22

From: Yong Zhi <yong.zhi(a)intel.com> Add HD Audio PCI ID for Intel Meteorlake platform. Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Signed-off-by: Yong Zhi <yong.zhi(a)intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/pci/hda/hda_intel.c | 3 +++ sound/pci/hda/patch_hdmi.c | 1 + 2 files changed, 4 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 0a83eb6b88b1f..2c2257ec50929 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2525,6 +2525,9 @@ static const struct pci_device_id azx_ids[] = { .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, { PCI_DEVICE(0x8086, 0x51cf), .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, + /* Meteorlake-P */ + { PCI_DEVICE(0x8086, 0x7E28), + .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, /* Broxton-P(Apollolake) */ { PCI_DEVICE(0x8086, 0x5a98), .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_BROXTON }, diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 31fe417955712..6c209cd26c0ca 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -4554,6 +4554,7 @@ HDA_CODEC_ENTRY(0x8086281a, "Jasperlake HDMI", patch_i915_icl_hdmi), HDA_CODEC_ENTRY(0x8086281b, "Elkhartlake HDMI", patch_i915_icl_hdmi), HDA_CODEC_ENTRY(0x8086281c, "Alderlake-P HDMI", patch_i915_adlp_hdmi), HDA_CODEC_ENTRY(0x8086281f, "Raptorlake-P HDMI", patch_i915_adlp_hdmi), +HDA_CODEC_ENTRY(0x8086281d, "Meteorlake HDMI", patch_i915_adlp_hdmi), HDA_CODEC_ENTRY(0x80862880, "CedarTrail HDMI", patch_generic_hdmi), HDA_CODEC_ENTRY(0x80862882, "Valleyview2 HDMI", patch_i915_byt_hdmi), HDA_CODEC_ENTRY(0x80862883, "Braswell HDMI", patch_i915_byt_hdmi), -- 2.34.1

2 1

ASUS ROG Maximus Z690 Hero (ALC4082) - No audio from microphone, no audio on the front panel
by GitHub issues - edited 06 Jun '22

06 Jun '22

alsa-project/alsa-ucm-conf issue #165 was edited from Lastique: I'm having a problem with getting audio from the microphone connected to the rear panel jack. Also, if I connect the headset to the front panel headphone/mic jacks, I get neither audio out or in. My main concern is the rear panel jack though. The motherboard is ASUS ROG Maximus Z690 Hero, ALC4082 USB Audio chipset. I have tried this: 1. Installed linux kernel 5.18.0. 2. Copied `ucm2` from alsa-ucm-conf master (rev ef3b6513b6581705d36432428a7f6a4ae23b1276 which is after https://github.com/alsa-project/alsa-ucm-conf/pull/143 was merged) to `/usr/share/alsa`. 3. In `/usr/share/alsa/ucm2/USB-Audio/USB-Audio.conf` added hardware id of my USB Audio device (`0b05:1a27`) to the `realtek-alc4080` rule. Also added the hardware id in `/usr/share/alsa/ucm2/USB-Audio/Realtek/ALC4080-HiFi.conf`, in the `asus-rog-usb` block condition. 4. Rebooted. 5. The headset is connected to the rear panel (Front out and Mic in jacks; also tried Line in). There are three devices for the USB Audio card, and there are three volume controls in alsamixer: "Line", "Mic" and "Analog In". The problem is that there is no audio coming from the first two devices (which are presumably "Line" and "Mic"). I can record audio from the third device ("Analog In") when the microphone is connected to Mic jack. Unfortunately, "Analog In" cannot be selected in PulseAudio for some reason (I created a [bug](https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/1364) for PA), which means no audio can be recorded. I also tried to connect the headset to the front panel connectors, but got no audio output nor capture. This is not my intended configuration though, I normally connect the headset to the rear panel. `arecord` output: [arecord.txt](https://github.com/alsa-project/alsa-ucm-conf/files/8838239/ar… `lsusb` output: [lsusb.txt](https://github.com/alsa-project/alsa-ucm-conf/files/8838250/lsus… `alsa-info.sh` output: [alsa-info.txt](https://github.com/alsa-project/alsa-ucm-conf/files/8846120/… This is on Kubuntu 22.04. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/165 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH 0/2] ALSA: usb-audio: Fix regressions for Saffire 6
by Takashi Iwai 06 Jun '22

06 Jun '22

Hi, this is a small patch set to address the still remaining regressions for Focusrite Saffire 6. Takashi === Takashi Iwai (2): ALSA: usb-audio: Skip generic sync EP parse for secondary EP ALSA: usb-audio: Set up (implicit) sync for Saffire 6 sound/usb/pcm.c | 5 ++++- sound/usb/quirks-table.h | 7 ++++++- 2 files changed, 10 insertions(+), 2 deletions(-) -- 2.35.3

1 2

Add basic support for ALC4082 on ASUS ROG Maximus Z690 Hero
by GitHub pull_request - edited 06 Jun '22

06 Jun '22

alsa-project/alsa-ucm-conf pull request #166 was edited from Lastique: This enables Microphone In in the rear jack and Headphones Out in the front panel jack on ASUS ROG Maximus Z690 Hero with Realtek ALC4082. Things that still don't work: - Microphone In on the front panel - Side channel output on the Line In jack in 7.1 mode - Rear channels and Center/Sub are swapped Line In on the rear panel detects microphone plugged in. Actual input not tested (no line out device available; no input from microphone). S/PDIF not tested. Related to https://github.com/alsa-project/alsa-ucm-conf/issues/165. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/166 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/166.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Add basic support for ALC4082 on ASUS ROG Maximus Z690 Hero
by GitHub pull_request - opened 06 Jun '22

06 Jun '22

alsa-project/alsa-ucm-conf pull request #166 was opened from Lastique: This enables Microphone In in the rear jack and Headphones Out in the front panel jack on ASUS ROG Maximus Z690 Hero with Realtek ALC4082. Things that still don't work: - Microphone In on the front panel - Side channel output on the Line In jack in 7.1 mode - Rear channels and Center/Sub are swapped Line In on the rear panel detects microphone plugged in. Actual input not tested (no line out device available; no input from microphone). S/PDIF not tested. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/166 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/166.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[RESEND PATCH v3 0/6] OPE support on Tegra210 and later
by Sameer Pujar 06 Jun '22

06 Jun '22

This series adds support for Output Prcoessing Module (OPE) module on Tegra210 and later generations of SoCs. OPE is a client of AHUB and it has sub blocks of PEQ (Parametric Equalizer) and MBDRC (Multi Band Dynamic Range Compressor) for data processing. An ASoC component is registered for OPE, which includes PEQ and MBDRC functions as well. This can be plugged in audio path using ALSA mixer controls. The series adds necessary binding documentaion, driver and DT binding patches to enable OPE module on Jetson platforms. Changelog ========= v2 -> v3: --------- * Drop "Device Tree Bindings" string from bindind doc titles for OPE, PEQ and MBDRC. v1 -> v2: --------- * Use generic node names for OPE, PEQ and MBDRC devices. Update binding doc and DT patches for this. * Remove redundant nodename rule enforcement from OPE, PEQ and MBDRC nodes. Update binding doc patch for this. * Fix spaces before binding doc examples and remove '|' from binding doc descriptions. Sameer Pujar (6): ASoC: tegra: Add binding doc for OPE module ASoC: tegra: Add Tegra210 based OPE driver ASoC: tegra: AHUB routes for OPE module arm64: defconfig: Build Tegra OPE module arm64: tegra: Add OPE device on Tegra210 and later arm64: tegra: Enable OPE on various platforms .../bindings/sound/nvidia,tegra210-ahub.yaml | 4 + .../bindings/sound/nvidia,tegra210-mbdrc.yaml | 47 + .../bindings/sound/nvidia,tegra210-ope.yaml | 87 ++ .../bindings/sound/nvidia,tegra210-peq.yaml | 48 + arch/arm64/boot/dts/nvidia/tegra186-p2771-0000.dts | 43 + arch/arm64/boot/dts/nvidia/tegra186.dtsi | 23 + arch/arm64/boot/dts/nvidia/tegra194-p2972-0000.dts | 43 + .../arm64/boot/dts/nvidia/tegra194-p3509-0000.dtsi | 43 + arch/arm64/boot/dts/nvidia/tegra194.dtsi | 23 + arch/arm64/boot/dts/nvidia/tegra210-p2371-2180.dts | 84 ++ arch/arm64/boot/dts/nvidia/tegra210-p3450-0000.dts | 84 ++ arch/arm64/boot/dts/nvidia/tegra210.dtsi | 40 + .../dts/nvidia/tegra234-p3737-0000+p3701-0000.dts | 43 + arch/arm64/boot/dts/nvidia/tegra234.dtsi | 23 + arch/arm64/configs/defconfig | 1 + sound/soc/tegra/Kconfig | 9 + sound/soc/tegra/Makefile | 2 + sound/soc/tegra/tegra210_ahub.c | 39 +- sound/soc/tegra/tegra210_mbdrc.c | 1012 ++++++++++++++++++++ sound/soc/tegra/tegra210_mbdrc.h | 215 +++++ sound/soc/tegra/tegra210_ope.c | 419 ++++++++ sound/soc/tegra/tegra210_ope.h | 90 ++ sound/soc/tegra/tegra210_peq.c | 434 +++++++++ sound/soc/tegra/tegra210_peq.h | 56 ++ 24 files changed, 2908 insertions(+), 4 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-mbdrc.yaml create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-ope.yaml create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-peq.yaml create mode 100644 sound/soc/tegra/tegra210_mbdrc.c create mode 100644 sound/soc/tegra/tegra210_mbdrc.h create mode 100644 sound/soc/tegra/tegra210_ope.c create mode 100644 sound/soc/tegra/tegra210_ope.h create mode 100644 sound/soc/tegra/tegra210_peq.c create mode 100644 sound/soc/tegra/tegra210_peq.h -- 2.7.4

1 6

[PATCH v2] ASoC: SOF: Fix potential NULL pointer dereference
by Amadeusz Sławiński 06 Jun '22

06 Jun '22

Cleanup path for sof_prepare_widgets_in_path() should check if unprepare callback exists before calling it, instead it checks if it does not exist. Fix the check. Fixes: 66344c6d9211 ("ASoC: SOF: Add a prepare op to IPC topology widget ops") Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> --- v2 : Fix Fixes hash, apparently I looked at wrong tree --- sound/soc/sof/sof-audio.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sof/sof-audio.c b/sound/soc/sof/sof-audio.c index 8d740635a4bb..28976098a89e 100644 --- a/sound/soc/sof/sof-audio.c +++ b/sound/soc/sof/sof-audio.c @@ -318,7 +318,7 @@ sof_prepare_widgets_in_path(struct snd_sof_dev *sdev, struct snd_soc_dapm_widget p->walking = false; if (ret < 0) { /* unprepare the source widget */ - if (!widget_ops[widget->id].ipc_unprepare && swidget->prepared) { + if (widget_ops[widget->id].ipc_unprepare && swidget->prepared) { widget_ops[widget->id].ipc_unprepare(swidget); swidget->prepared = false; } -- 2.25.1

1 0

[PATCH] slimbus: messaging: fix typos in comments
by Julia Lawall 06 Jun '22

06 Jun '22

Spelling mistakes (triple letters) in comments. Detected with the help of Coccinelle. Signed-off-by: Julia Lawall <Julia.Lawall(a)inria.fr> --- drivers/slimbus/messaging.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/drivers/slimbus/messaging.c b/drivers/slimbus/messaging.c index e5ae26227bdb..4ce0cb61e481 100644 --- a/drivers/slimbus/messaging.c +++ b/drivers/slimbus/messaging.c @@ -79,7 +79,7 @@ int slim_alloc_txn_tid(struct slim_controller *ctrl, struct slim_msg_txn *txn) EXPORT_SYMBOL_GPL(slim_alloc_txn_tid); /** - * slim_free_txn_tid() - Freee tid of txn + * slim_free_txn_tid() - Free tid of txn * * @ctrl: Controller handle * @txn: transaction whose tid should be freed @@ -101,7 +101,7 @@ EXPORT_SYMBOL_GPL(slim_free_txn_tid); * @txn: Transaction to be sent over SLIMbus * * Called by controller to transmit messaging transactions not dealing with - * Interface/Value elements. (e.g. transmittting a message to assign logical + * Interface/Value elements. (e.g. transmitting a message to assign logical * address to a slave device * * Return: -ETIMEDOUT: If transmission of this message timed out

2 1

[PATCH] drivers: slimbus: Directly use ida_alloc()/free()
by keliu 06 Jun '22

06 Jun '22

Use ida_alloc()/ida_free() instead of deprecated ida_simple_get()/ida_simple_remove() . Signed-off-by: keliu <liuke94(a)huawei.com> --- drivers/slimbus/core.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/drivers/slimbus/core.c b/drivers/slimbus/core.c index 78480e332ab8..219483b79c09 100644 --- a/drivers/slimbus/core.c +++ b/drivers/slimbus/core.c @@ -250,7 +250,7 @@ int slim_register_controller(struct slim_controller *ctrl) { int id; - id = ida_simple_get(&ctrl_ida, 0, 0, GFP_KERNEL); + id = ida_alloc(&ctrl_ida, GFP_KERNEL); if (id < 0) return id; @@ -299,7 +299,7 @@ int slim_unregister_controller(struct slim_controller *ctrl) { /* Remove all clients */ device_for_each_child(ctrl->dev, NULL, slim_ctrl_remove_device); - ida_simple_remove(&ctrl_ida, ctrl->id); + ida_free(&ctrl_ida, ctrl->id); return 0; } @@ -323,7 +323,7 @@ void slim_report_absent(struct slim_device *sbdev) sbdev->is_laddr_valid = false; mutex_unlock(&ctrl->lock); if (!ctrl->get_laddr) - ida_simple_remove(&ctrl->laddr_ida, sbdev->laddr); + ida_free(&ctrl->laddr_ida, sbdev->laddr); slim_device_update_status(sbdev, SLIM_DEVICE_STATUS_DOWN); } EXPORT_SYMBOL_GPL(slim_report_absent); -- 2.25.1

2 1

[PATCH] ASoC: wm_adsp: Remove redundant NULL check before release_firmware() call
by cgel.zte＠gmail.com 06 Jun '22

06 Jun '22

From: Minghao Chi <chi.minghao(a)zte.com.cn> release_firmware() checks for NULL pointers internally so checking before calling it is redundant. Reported-by: Zeal Robot <zealci(a)zte.com.cn> Signed-off-by: Minghao Chi <chi.minghao(a)zte.com.cn> --- sound/soc/codecs/wm_adsp.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 7973a75cac05..4c4344b21c2e 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -739,12 +739,10 @@ static void wm_adsp_release_firmware_files(struct wm_adsp *dsp, const struct firmware *coeff_firmware, char *coeff_filename) { - if (wmfw_firmware) - release_firmware(wmfw_firmware); + release_firmware(wmfw_firmware); kfree(wmfw_filename); - if (coeff_firmware) - release_firmware(coeff_firmware); + release_firmware(coeff_firmware); kfree(coeff_filename); } -- 2.25.1

2 1

[PATCH] ALSA: hda/realtek: Fix for quirk to enable speaker output on the Lenovo Yoga DuetITL 2021
by Cameron Berkenpas 06 Jun '22

06 Jun '22

Enables the ALC287_FIXUP_YOGA7_14ITL_SPEAKERS quirk for the Lenovo Yoga DuetITL 2021 laptop to fix speaker output. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=208555 Signed-off-by: Cameron Berkenpas <cam(a)neo-zeon.de> Co-authored-by: Songine <donglingluoying(a)gmail.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index f3ad454b3fbf..2af5725dfca0 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9262,6 +9262,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x3818, "Lenovo C940", ALC298_FIXUP_LENOVO_SPK_VOLUME), SND_PCI_QUIRK(0x17aa, 0x3819, "Lenovo 13s Gen2 ITL", ALC287_FIXUP_13S_GEN2_SPEAKERS), SND_PCI_QUIRK(0x17aa, 0x3820, "Yoga Duet 7 13ITL6", ALC287_FIXUP_YOGA7_14ITL_SPEAKERS), + SND_PCI_QUIRK(0x17aa, 0x3802, "Lenovo Yoga DuetITL 2021", ALC287_FIXUP_YOGA7_14ITL_SPEAKERS), SND_PCI_QUIRK(0x17aa, 0x3824, "Legion Y9000X 2020", ALC285_FIXUP_LEGION_Y9000X_SPEAKERS), SND_PCI_QUIRK(0x17aa, 0x3827, "Ideapad S740", ALC285_FIXUP_IDEAPAD_S740_COEF), SND_PCI_QUIRK(0x17aa, 0x3834, "Lenovo IdeaPad Slim 9i 14ITL5", ALC287_FIXUP_YOGA7_14ITL_SPEAKERS), -- 2.34.1

2 1

[PATCH] dt-bindings: Fix properties without any type
by Rob Herring 06 Jun '22

06 Jun '22

Now that the schema tools can extract type information for all properties (in order to decode dtb files), finding properties missing any type definition is fairly trivial though not yet automated. Fix the various property schemas which are missing a type. Most of these tend to be device specific properties which don't have a vendor prefix. A vendor prefix is how we normally ensure a type is defined. Signed-off-by: Rob Herring <robh(a)kernel.org> --- .../arm/hisilicon/controller/hip04-bootwrapper.yaml | 5 +++-- .../bindings/display/bridge/toshiba,tc358768.yaml | 1 + .../devicetree/bindings/display/panel/panel-timing.yaml | 5 +++++ .../bindings/display/panel/raydium,rm67191.yaml | 1 + .../bindings/display/panel/samsung,s6e8aa0.yaml | 1 + .../devicetree/bindings/gpio/fairchild,74hc595.yaml | 1 + .../devicetree/bindings/input/google,cros-ec-keyb.yaml | 1 + .../devicetree/bindings/input/matrix-keymap.yaml | 4 ++++ Documentation/devicetree/bindings/media/i2c/adv7604.yaml | 3 ++- Documentation/devicetree/bindings/mux/reg-mux.yaml | 8 ++++++-- Documentation/devicetree/bindings/net/cdns,macb.yaml | 1 + Documentation/devicetree/bindings/net/ingenic,mac.yaml | 1 + .../devicetree/bindings/net/ti,davinci-mdio.yaml | 1 + .../devicetree/bindings/net/wireless/ti,wlcore.yaml | 2 ++ .../devicetree/bindings/pci/snps,dw-pcie-ep.yaml | 6 ++++-- Documentation/devicetree/bindings/pci/snps,dw-pcie.yaml | 2 ++ .../devicetree/bindings/pinctrl/canaan,k210-fpioa.yaml | 2 ++ Documentation/devicetree/bindings/power/avs/qcom,cpr.yaml | 1 + .../devicetree/bindings/power/supply/battery.yaml | 7 ++++++- .../devicetree/bindings/power/supply/charger-manager.yaml | 1 + Documentation/devicetree/bindings/rng/st,stm32-rng.yaml | 1 + Documentation/devicetree/bindings/serial/8250.yaml | 1 + .../devicetree/bindings/sound/audio-graph-card2.yaml | 3 +++ .../devicetree/bindings/sound/imx-audio-hdmi.yaml | 3 +++ Documentation/devicetree/bindings/usb/smsc,usb3503.yaml | 1 + 25 files changed, 55 insertions(+), 8 deletions(-) diff --git a/Documentation/devicetree/bindings/arm/hisilicon/controller/hip04-bootwrapper.yaml b/Documentation/devicetree/bindings/arm/hisilicon/controller/hip04-bootwrapper.yaml index 7378159e61df..483caf0ce25b 100644 --- a/Documentation/devicetree/bindings/arm/hisilicon/controller/hip04-bootwrapper.yaml +++ b/Documentation/devicetree/bindings/arm/hisilicon/controller/hip04-bootwrapper.yaml @@ -17,14 +17,15 @@ properties: - const: hisilicon,hip04-bootwrapper boot-method: + $ref: /schemas/types.yaml#/definitions/uint32-array description: | Address and size of boot method. [0]: bootwrapper physical address [1]: bootwrapper size [2]: relocation physical address [3]: relocation size - minItems: 1 - maxItems: 2 + minItems: 2 + maxItems: 4 required: - compatible diff --git a/Documentation/devicetree/bindings/display/bridge/toshiba,tc358768.yaml b/Documentation/devicetree/bindings/display/bridge/toshiba,tc358768.yaml index 3bd670b8e5cd..0b6f5bef120f 100644 --- a/Documentation/devicetree/bindings/display/bridge/toshiba,tc358768.yaml +++ b/Documentation/devicetree/bindings/display/bridge/toshiba,tc358768.yaml @@ -58,6 +58,7 @@ properties: properties: data-lines: + $ref: /schemas/types.yaml#/definitions/uint32 enum: [ 16, 18, 24 ] port@1: diff --git a/Documentation/devicetree/bindings/display/panel/panel-timing.yaml b/Documentation/devicetree/bindings/display/panel/panel-timing.yaml index 7749de95ee40..229e3b36ee29 100644 --- a/Documentation/devicetree/bindings/display/panel/panel-timing.yaml +++ b/Documentation/devicetree/bindings/display/panel/panel-timing.yaml @@ -146,6 +146,7 @@ properties: Horizontal sync pulse. 0 selects active low, 1 selects active high. If omitted then it is not used by the hardware + $ref: /schemas/types.yaml#/definitions/uint32 enum: [0, 1] vsync-active: @@ -153,6 +154,7 @@ properties: Vertical sync pulse. 0 selects active low, 1 selects active high. If omitted then it is not used by the hardware + $ref: /schemas/types.yaml#/definitions/uint32 enum: [0, 1] de-active: @@ -160,6 +162,7 @@ properties: Data enable. 0 selects active low, 1 selects active high. If omitted then it is not used by the hardware + $ref: /schemas/types.yaml#/definitions/uint32 enum: [0, 1] pixelclk-active: @@ -169,6 +172,7 @@ properties: sample data on rising edge. Use 1 to drive pixel data on rising edge and sample data on falling edge + $ref: /schemas/types.yaml#/definitions/uint32 enum: [0, 1] syncclk-active: @@ -179,6 +183,7 @@ properties: sample sync on rising edge of pixel clock. Use 1 to drive sync on rising edge and sample sync on falling edge of pixel clock + $ref: /schemas/types.yaml#/definitions/uint32 enum: [0, 1] interlaced: diff --git a/Documentation/devicetree/bindings/display/panel/raydium,rm67191.yaml b/Documentation/devicetree/bindings/display/panel/raydium,rm67191.yaml index 745dd247c409..617aa8c8c03a 100644 --- a/Documentation/devicetree/bindings/display/panel/raydium,rm67191.yaml +++ b/Documentation/devicetree/bindings/display/panel/raydium,rm67191.yaml @@ -24,6 +24,7 @@ properties: dsi-lanes: description: Number of DSI lanes to be used must be <3> or <4> + $ref: /schemas/types.yaml#/definitions/uint32 enum: [3, 4] v3p3-supply: diff --git a/Documentation/devicetree/bindings/display/panel/samsung,s6e8aa0.yaml b/Documentation/devicetree/bindings/display/panel/samsung,s6e8aa0.yaml index ca959451557e..1cdc91b3439f 100644 --- a/Documentation/devicetree/bindings/display/panel/samsung,s6e8aa0.yaml +++ b/Documentation/devicetree/bindings/display/panel/samsung,s6e8aa0.yaml @@ -36,6 +36,7 @@ properties: init-delay: description: delay after initialization sequence [ms] + $ref: /schemas/types.yaml#/definitions/uint32 panel-width-mm: description: physical panel width [mm] diff --git a/Documentation/devicetree/bindings/gpio/fairchild,74hc595.yaml b/Documentation/devicetree/bindings/gpio/fairchild,74hc595.yaml index 5fe19fa5f67c..a99e7842ca17 100644 --- a/Documentation/devicetree/bindings/gpio/fairchild,74hc595.yaml +++ b/Documentation/devicetree/bindings/gpio/fairchild,74hc595.yaml @@ -26,6 +26,7 @@ properties: const: 2 registers-number: + $ref: /schemas/types.yaml#/definitions/uint32 description: Number of daisy-chained shift registers enable-gpios: diff --git a/Documentation/devicetree/bindings/input/google,cros-ec-keyb.yaml b/Documentation/devicetree/bindings/input/google,cros-ec-keyb.yaml index e8f137abb03c..aa61fe64be63 100644 --- a/Documentation/devicetree/bindings/input/google,cros-ec-keyb.yaml +++ b/Documentation/devicetree/bindings/input/google,cros-ec-keyb.yaml @@ -31,6 +31,7 @@ properties: type: boolean function-row-physmap: + $ref: /schemas/types.yaml#/definitions/uint32-array minItems: 1 maxItems: 15 description: | diff --git a/Documentation/devicetree/bindings/input/matrix-keymap.yaml b/Documentation/devicetree/bindings/input/matrix-keymap.yaml index 6699d5e32dca..9f703bb51e12 100644 --- a/Documentation/devicetree/bindings/input/matrix-keymap.yaml +++ b/Documentation/devicetree/bindings/input/matrix-keymap.yaml @@ -27,6 +27,10 @@ properties: column and linux key-code. The 32-bit big endian cell is packed as: row << 24 | column << 16 | key-code + linux,no-autorepeat: + type: boolean + description: Disable keyrepeat + keypad,num-rows: $ref: /schemas/types.yaml#/definitions/uint32 description: Number of row lines connected to the keypad controller. diff --git a/Documentation/devicetree/bindings/media/i2c/adv7604.yaml b/Documentation/devicetree/bindings/media/i2c/adv7604.yaml index c19d8391e2d5..7589d377c686 100644 --- a/Documentation/devicetree/bindings/media/i2c/adv7604.yaml +++ b/Documentation/devicetree/bindings/media/i2c/adv7604.yaml @@ -60,7 +60,8 @@ properties: enables hot-plug detection. default-input: - maxItems: 1 + $ref: /schemas/types.yaml#/definitions/uint32 + enum: [ 0, 1 ] description: Select which input is selected after reset. diff --git a/Documentation/devicetree/bindings/mux/reg-mux.yaml b/Documentation/devicetree/bindings/mux/reg-mux.yaml index 60d5746eb39d..e2f6b11f1254 100644 --- a/Documentation/devicetree/bindings/mux/reg-mux.yaml +++ b/Documentation/devicetree/bindings/mux/reg-mux.yaml @@ -25,8 +25,12 @@ properties: const: 1 mux-reg-masks: - description: an array of register offset and pre-shifted bitfield mask - pairs, each describing a single mux control. + $ref: /schemas/types.yaml#/definitions/uint32-matrix + items: + items: + - description: register offset + - description: pre-shifted bitfield mask + description: Each entry describes a single mux control. idle-states: true diff --git a/Documentation/devicetree/bindings/net/cdns,macb.yaml b/Documentation/devicetree/bindings/net/cdns,macb.yaml index 6cd3d853dcba..59fe2789fa44 100644 --- a/Documentation/devicetree/bindings/net/cdns,macb.yaml +++ b/Documentation/devicetree/bindings/net/cdns,macb.yaml @@ -129,6 +129,7 @@ patternProperties: reset-gpios: true magic-packet: + type: boolean description: Indicates that the hardware supports waking up via magic packet. diff --git a/Documentation/devicetree/bindings/net/ingenic,mac.yaml b/Documentation/devicetree/bindings/net/ingenic,mac.yaml index 8e52b2e683b8..93b3e991d209 100644 --- a/Documentation/devicetree/bindings/net/ingenic,mac.yaml +++ b/Documentation/devicetree/bindings/net/ingenic,mac.yaml @@ -37,6 +37,7 @@ properties: const: stmmaceth mode-reg: + $ref: /schemas/types.yaml#/definitions/phandle description: An extra syscon register that control ethernet interface and timing delay rx-clk-delay-ps: diff --git a/Documentation/devicetree/bindings/net/ti,davinci-mdio.yaml b/Documentation/devicetree/bindings/net/ti,davinci-mdio.yaml index 6f44f9516c36..a339202c5e8e 100644 --- a/Documentation/devicetree/bindings/net/ti,davinci-mdio.yaml +++ b/Documentation/devicetree/bindings/net/ti,davinci-mdio.yaml @@ -34,6 +34,7 @@ properties: maxItems: 1 bus_freq: + $ref: /schemas/types.yaml#/definitions/uint32 maximum: 2500000 description: MDIO Bus frequency diff --git a/Documentation/devicetree/bindings/net/wireless/ti,wlcore.yaml b/Documentation/devicetree/bindings/net/wireless/ti,wlcore.yaml index 8dd164d10290..d68bb2ec1f7e 100644 --- a/Documentation/devicetree/bindings/net/wireless/ti,wlcore.yaml +++ b/Documentation/devicetree/bindings/net/wireless/ti,wlcore.yaml @@ -54,9 +54,11 @@ properties: ref-clock-frequency: + $ref: /schemas/types.yaml#/definitions/uint32 description: Reference clock frequency. tcxo-clock-frequency: + $ref: /schemas/types.yaml#/definitions/uint32 description: TCXO clock frequency. clock-xtal: diff --git a/Documentation/devicetree/bindings/pci/snps,dw-pcie-ep.yaml b/Documentation/devicetree/bindings/pci/snps,dw-pcie-ep.yaml index e59059ab5be0..b78535040f04 100644 --- a/Documentation/devicetree/bindings/pci/snps,dw-pcie-ep.yaml +++ b/Documentation/devicetree/bindings/pci/snps,dw-pcie-ep.yaml @@ -55,13 +55,15 @@ properties: Translation Unit) registers. num-ib-windows: + $ref: /schemas/types.yaml#/definitions/uint32 + maximum: 256 description: number of inbound address translation windows - maxItems: 1 deprecated: true num-ob-windows: + $ref: /schemas/types.yaml#/definitions/uint32 + maximum: 256 description: number of outbound address translation windows - maxItems: 1 deprecated: true required: diff --git a/Documentation/devicetree/bindings/pci/snps,dw-pcie.yaml b/Documentation/devicetree/bindings/pci/snps,dw-pcie.yaml index a5345c494744..c90e5e2d25f6 100644 --- a/Documentation/devicetree/bindings/pci/snps,dw-pcie.yaml +++ b/Documentation/devicetree/bindings/pci/snps,dw-pcie.yaml @@ -68,6 +68,8 @@ properties: Translation Unit) registers. num-viewport: + $ref: /schemas/types.yaml#/definitions/uint32 + maximum: 256 description: | number of view ports configured in hardware. If a platform does not specify it, the driver autodetects it. diff --git a/Documentation/devicetree/bindings/pinctrl/canaan,k210-fpioa.yaml b/Documentation/devicetree/bindings/pinctrl/canaan,k210-fpioa.yaml index 53e963e090f2..533b4cfe33d2 100644 --- a/Documentation/devicetree/bindings/pinctrl/canaan,k210-fpioa.yaml +++ b/Documentation/devicetree/bindings/pinctrl/canaan,k210-fpioa.yaml @@ -120,6 +120,7 @@ patternProperties: input-schmitt-disable: true input-polarity-invert: + type: boolean description: Enable or disable pin input polarity inversion. @@ -132,6 +133,7 @@ patternProperties: output-low: true output-polarity-invert: + type: boolean description: Enable or disable pin output polarity inversion. diff --git a/Documentation/devicetree/bindings/power/avs/qcom,cpr.yaml b/Documentation/devicetree/bindings/power/avs/qcom,cpr.yaml index 3301fa0c2653..301db7daf870 100644 --- a/Documentation/devicetree/bindings/power/avs/qcom,cpr.yaml +++ b/Documentation/devicetree/bindings/power/avs/qcom,cpr.yaml @@ -51,6 +51,7 @@ properties: supported by the CPR power domain. acc-syscon: + $ref: /schemas/types.yaml#/definitions/phandle description: A phandle to the syscon used for writing ACC settings. nvmem-cells: diff --git a/Documentation/devicetree/bindings/power/supply/battery.yaml b/Documentation/devicetree/bindings/power/supply/battery.yaml index d56ac484fec5..491488e7b970 100644 --- a/Documentation/devicetree/bindings/power/supply/battery.yaml +++ b/Documentation/devicetree/bindings/power/supply/battery.yaml @@ -85,8 +85,13 @@ properties: description: battery factory internal resistance resistance-temp-table: + $ref: /schemas/types.yaml#/definitions/uint32-matrix + items: + items: + - description: the temperature in degree Celsius + - description: battery internal resistance percent description: | - An array providing the temperature in degree Celsius + A table providing the temperature in degree Celsius and corresponding battery internal resistance percent, which is used to look up the resistance percent according to current temperature to get an accurate batterty internal resistance in different temperatures. diff --git a/Documentation/devicetree/bindings/power/supply/charger-manager.yaml b/Documentation/devicetree/bindings/power/supply/charger-manager.yaml index c863cfa67865..fbb2204769aa 100644 --- a/Documentation/devicetree/bindings/power/supply/charger-manager.yaml +++ b/Documentation/devicetree/bindings/power/supply/charger-manager.yaml @@ -36,6 +36,7 @@ properties: cm-poll-mode: description: polling mode + $ref: /schemas/types.yaml#/definitions/uint32 default: 0 enum: - 0 # disabled diff --git a/Documentation/devicetree/bindings/rng/st,stm32-rng.yaml b/Documentation/devicetree/bindings/rng/st,stm32-rng.yaml index 9a6e4eaf4d3c..fcd86f822a9c 100644 --- a/Documentation/devicetree/bindings/rng/st,stm32-rng.yaml +++ b/Documentation/devicetree/bindings/rng/st,stm32-rng.yaml @@ -27,6 +27,7 @@ properties: maxItems: 1 clock-error-detect: + type: boolean description: If set enable the clock detection management required: diff --git a/Documentation/devicetree/bindings/serial/8250.yaml b/Documentation/devicetree/bindings/serial/8250.yaml index 3bab2f27b970..5f6b113d378f 100644 --- a/Documentation/devicetree/bindings/serial/8250.yaml +++ b/Documentation/devicetree/bindings/serial/8250.yaml @@ -138,6 +138,7 @@ properties: description: The current active speed of the UART. reg-offset: + $ref: /schemas/types.yaml#/definitions/uint32 description: | Offset to apply to the mapbase from the start of the registers. diff --git a/Documentation/devicetree/bindings/sound/audio-graph-card2.yaml b/Documentation/devicetree/bindings/sound/audio-graph-card2.yaml index f7e94b1e0e4b..7416067c945e 100644 --- a/Documentation/devicetree/bindings/sound/audio-graph-card2.yaml +++ b/Documentation/devicetree/bindings/sound/audio-graph-card2.yaml @@ -24,10 +24,13 @@ properties: connection's sink, the second being the connection's source. $ref: /schemas/types.yaml#/definitions/non-unique-string-array multi: + type: object description: Multi-CPU/Codec node dpcm: + type: object description: DPCM node codec2codec: + type: object description: Codec to Codec node required: diff --git a/Documentation/devicetree/bindings/sound/imx-audio-hdmi.yaml b/Documentation/devicetree/bindings/sound/imx-audio-hdmi.yaml index d5474f83ac2c..e7e7bb65c366 100644 --- a/Documentation/devicetree/bindings/sound/imx-audio-hdmi.yaml +++ b/Documentation/devicetree/bindings/sound/imx-audio-hdmi.yaml @@ -20,9 +20,11 @@ properties: description: User specified audio sound card name audio-cpu: + $ref: /schemas/types.yaml#/definitions/phandle description: The phandle of an CPU DAI controller hdmi-out: + type: boolean description: | This is a boolean property. If present, the transmitting function of HDMI will be enabled, indicating there's a physical HDMI out @@ -30,6 +32,7 @@ properties: block, such as an HDMI encoder or display-controller. hdmi-in: + type: boolean description: | This is a boolean property. If present, the receiving function of HDMI will be enabled, indicating there is a physical HDMI in diff --git a/Documentation/devicetree/bindings/usb/smsc,usb3503.yaml b/Documentation/devicetree/bindings/usb/smsc,usb3503.yaml index b9e219829801..321b6f166197 100644 --- a/Documentation/devicetree/bindings/usb/smsc,usb3503.yaml +++ b/Documentation/devicetree/bindings/usb/smsc,usb3503.yaml @@ -45,6 +45,7 @@ properties: property if all ports have to be enabled. initial-mode: + $ref: /schemas/types.yaml#/definitions/uint32 enum: [1, 2] description: > Specifies initial mode. 1 for Hub mode, 2 for standby mode. -- 2.34.1

8 8

[PATCH] ASoC: imx-audmux: remove unnecessary check of clk_disable_unprepare
by cgel.zte＠gmail.com 06 Jun '22

06 Jun '22

From: Minghao Chi <chi.minghao(a)zte.com.cn> Because clk_disable_unprepare already checked NULL clock parameter, so the additional checks are unnecessary, just remove them. Reported-by: Zeal Robot <zealci(a)zte.com.cn> Signed-off-by: Minghao Chi <chi.minghao(a)zte.com.cn> --- sound/soc/fsl/imx-audmux.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) diff --git a/sound/soc/fsl/imx-audmux.c b/sound/soc/fsl/imx-audmux.c index dfa05d40b276..f434fa7decc1 100644 --- a/sound/soc/fsl/imx-audmux.c +++ b/sound/soc/fsl/imx-audmux.c @@ -71,8 +71,7 @@ static ssize_t audmux_read_file(struct file *file, char __user *user_buf, ptcr = readl(audmux_base + IMX_AUDMUX_V2_PTCR(port)); pdcr = readl(audmux_base + IMX_AUDMUX_V2_PDCR(port)); - if (audmux_clk) - clk_disable_unprepare(audmux_clk); + clk_disable_unprepare(audmux_clk); buf = kmalloc(PAGE_SIZE, GFP_KERNEL); if (!buf) @@ -218,8 +217,7 @@ int imx_audmux_v2_configure_port(unsigned int port, unsigned int ptcr, writel(ptcr, audmux_base + IMX_AUDMUX_V2_PTCR(port)); writel(pdcr, audmux_base + IMX_AUDMUX_V2_PDCR(port)); - if (audmux_clk) - clk_disable_unprepare(audmux_clk); + clk_disable_unprepare(audmux_clk); return 0; } -- 2.25.1

2 1

HDMI audio broken in 1.2.7 for old laptops
by GitHub issues - opened 05 Jun '22

05 Jun '22

alsa-project/alsa-lib issue #235 was opened from hwpfeil: Tag 1.2.7 breaks HDMI audio on HP laptops with AMD A6-5200 APU (video works). No HDMI option in System Settings|Audio Devices, both KDE and XFCE. Reverting to 1.2.6.1 restores the HDMI Audio Device. There should come a time when such dinosaurs are declared obsolete and no longer supported. They can keep running on 1.2.6.1. Not sure if it needs fixing. Issue URL : https://github.com/alsa-project/alsa-lib/issues/235 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH 0/6] timer: use explicit two types of signals to dispatch events
by GitHub pull_request - opened 05 Jun '22

05 Jun '22

alsa-project/alsa-gobject pull request #74 was opened from takaswie: In current implementation, ALSATimer.UserInstance dispatches one type of event for both tick and timestamp events. However, the type of event is decided before starting timer instance, therefore user application never processes both types of event. In this patchset, two types of signals are added to ALSATimer.UserInstance to dispatch the two types of event separately, then ALSATImer.Event is going to be removed. Additionally, some boxed objects and enumerations are renamed so that they have straight-forward names. ``` Takashi Sakamoto (6): timer: user_instance: use two signals to dispatch events explicitly timer: event: remove unused structure timer: enums: rename TstampEventType from EventType timer: enum: rename EventType from EventDataType timer: tick-event: rename from event-data-tick timer: tstamp-event: rename from event-data-tstamp samples/timer | 29 ++++---- src/timer/alsatimer-enum-types.h | 80 +++++++++++----------- src/timer/alsatimer.h | 5 +- src/timer/alsatimer.map | 30 ++++---- src/timer/event-data-tick.c | 51 -------------- src/timer/event-data-tick.h | 23 ------- src/timer/event-data-tstamp.c | 66 ------------------ src/timer/event-data-tstamp.h | 26 ------- src/timer/event.c | 64 ----------------- src/timer/event.h | 28 -------- src/timer/instance-params.c | 22 +++--- src/timer/instance-params.h | 4 +- src/timer/meson.build | 10 ++- src/timer/tick-event.c | 51 ++++++++++++++ src/timer/tick-event.h | 23 +++++++ src/timer/tstamp-event.c | 64 +++++++++++++++++ src/timer/tstamp-event.h | 24 +++++++ src/timer/user-instance.c | 114 ++++++++++++++++++++----------- src/timer/user-instance.h | 22 ++++-- tests/alsatimer-enums | 6 +- tests/alsatimer-user-instance | 8 ++- 21 files changed, 346 insertions(+), 404 deletions(-) delete mode 100644 src/timer/event-data-tick.c delete mode 100644 src/timer/event-data-tick.h delete mode 100644 src/timer/event-data-tstamp.c delete mode 100644 src/timer/event-data-tstamp.h delete mode 100644 src/timer/event.c delete mode 100644 src/timer/event.h create mode 100644 src/timer/tick-event.c create mode 100644 src/timer/tick-event.h create mode 100644 src/timer/tstamp-event.c create mode 100644 src/timer/tstamp-event.h ``` Request URL : https://github.com/alsa-project/alsa-gobject/pull/74 Patch URL : https://github.com/alsa-project/alsa-gobject/pull/74.patch Repository URL: https://github.com/alsa-project/alsa-gobject

1 0

ASUS ROG Maximus Z690 Hero (ALC4082) - No audio from microphone, no audio on the front panel
by GitHub issues - edited 04 Jun '22

04 Jun '22

alsa-project/alsa-ucm-conf issue #165 was edited from Lastique: I'm having a problem with getting audio from the microphone connected to the rear panel jack. Also, if I connect the headset to the front panel headphone/mic jacks, I get neither audio out or in. My main concern is the rear panel jack though. The motherboard is ASUS ROG Maximus Z690 Hero, ALC4082 USB Audio chipset. I have tried this: 1. Installed linux kernel 5.18.0. 2. Copied `ucm2` from alsa-ucm-conf master (rev ef3b6513b6581705d36432428a7f6a4ae23b1276 which is after https://github.com/alsa-project/alsa-ucm-conf/pull/143 was merged) to `/usr/share/alsa`. 3. In `/usr/share/alsa/ucm2/USB-Audio/USB-Audio.conf` added hardware id of my USB Audio device (`0b05:1a27`) to the `realtek-alc4080` rule. Also added the hardware id in `/usr/share/alsa/ucm2/USB-Audio/Realtek/ALC4080-HiFi.conf`, in the `asus-rog-usb` block condition. 4. Rebooted. 5. The headset is connected to the rear panel (Front out and Mic in jacks; also tried Line in). There are three devices for the USB Audio card, and there are three volume controls in alsamixer: "Line", "Mic" and "Analog In". The problem is that there is no audio coming from the first two devices (which are presumably "Line" and "Mic"). I can record audio from the third device ("Analog In") when the microphone is connected to Mic jack. Unfortunately, "Analog In" cannot be selected in PulseAudio for some reason (I created a [bug](https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/1364) for PA), which means no audio can be recorded. I also tried to connect the headset to the front panel connectors, but got no audio output nor capture. This is not my intended configuration though, I normally connect the headset to the rear panel. `arecord` output: [arecord.txt](https://github.com/alsa-project/alsa-ucm-conf/files/8838239/ar… `lsusb` output: [lsusb.txt](https://github.com/alsa-project/alsa-ucm-conf/files/8838250/lsus… This is on Kubuntu 22.04. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/165 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

ASUS ROG Maximus Z690 Hero (ALC4082) - No audio from microphone, no audio on the front panel
by GitHub issues - edited 04 Jun '22

04 Jun '22

alsa-project/alsa-ucm-conf issue #165 was edited from Lastique: I'm having a problem with getting audio from the microphone connected to the rear panel jack. Also, if I connect the headset to the front panel headphone/mic jacks, I get neither audio out or in. My main concern is the rear panel jack though. The motherboard is ASUS ROG Maximus Z690 Hero, ALC4082 USB Audio chipset. I have tried this: 1. Installed linux kernel 5.18.0. 2. Copied `ucm2` from alsa-ucm-conf master (rev ef3b6513b6581705d36432428a7f6a4ae23b1276 which is after https://github.com/alsa-project/alsa-ucm-conf/pull/143 was merged) to `/usr/share/alsa`. 3. In `/usr/share/alsa/ucm2/USB-Audio/USB-Audio.conf` added hardware id of my USB Audio device (`0b05:1a27`) to the `realtek-alc4080` rule. Also added the hardware id in `/usr/share/alsa/ucm2/USB-Audio/Realtek/ALC4080-HiFi.conf`, in the `asus-rog-usb` block condition. 4. Rebooted. 5. The headset is connected to the rear panel (Front out and Mic in jacks; also tried Line in). There are three devices for the USB Audio card, and there are three volume controls in alsamixer: "Line", "Mic" and "Analog In". The problem is that there is no audio coming from the first two devices (which are presumably "Line" and "Mic"). I can record audio from the third device ("Analog In") when the microphone is connected to Mic jack. Unfortunately, "Analog In" cannot be selected in PulseAudio for some reason (I created a [bug](https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/1364) for PA), which means no audio can be recorded. I also tried to connect the headset to the front panel connectors, but got no audio output nor capture. This is not my intended configuration though, I normally connect the headset to the rear panel. `arecord` output: [arecord.txt](https://github.com/alsa-project/alsa-ucm-conf/files/8838239/ar… `lsusb` output: [lsusb.txt](https://github.com/alsa-project/alsa-ucm-conf/files/8838241/lsus… This is on Kubuntu 22.04. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/165 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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New device naming breaks HDMI audio if ELD lookup fails
by GitHub issues - opened 04 Jun '22

04 Jun '22

alsa-project/alsa-lib issue #233 was opened from berolinux: The new device naming introduced by commit 859448f01033b40feebf9a0aab467ba57a655b5b breaks HDMI audio if the ELD lookup fails. This happens e.g. with an older Sharp Aquos TV (from 2008) connected to an AMD Radeon HD 7450 graphics card with Linux kernel 5.18.1. Adding some debug statements to control/eld.c shows it fails on the check for `l == 0`, with `l` being `0`, `cinfo.count` being `24`. A hexdump of the 24 bytes in eld is `10 00 05 00 00 10 00 01 00 00 00 00 00 00 00 00 00 00 00 00 09 07 01 00`. The EDID info from the graphics card in sysfs makes sense though: ``` $ edid-decode /sys/class/drm/card0-HDMI-A-1/edid edid-decode (hex): 00 ff ff ff ff ff ff 00 4d 10 22 10 00 00 00 00 ff 12 01 03 80 52 2e 78 2a 1b be a2 55 34 b3 26 14 4a 52 af ce 00 a9 40 90 40 81 80 01 01 01 01 01 01 01 01 01 01 02 3a 80 d0 72 38 2d 40 10 2c 45 80 34 cc 31 00 00 1a 66 21 50 b0 51 00 1b 30 40 70 36 00 00 00 00 00 00 1e 00 00 00 fc 00 53 48 41 52 50 20 48 44 4d 49 0a 20 20 00 00 00 fd 00 17 4c 0f 4b 11 00 0a 20 20 20 20 20 20 01 40 02 03 2e 72 50 9f 90 20 14 05 13 04 12 03 11 02 16 07 15 06 01 23 09 07 01 83 01 00 00 e3 05 03 01 6c 03 0c 00 20 00 80 22 c0 11 11 19 19 02 3a 80 18 71 38 2d 40 58 2c 45 00 34 cc 31 00 00 1e 01 1d 80 d0 72 1c 16 20 10 2c 25 80 34 cc 31 00 00 9e 01 1d 80 18 71 1c 16 20 58 2c 25 00 34 cc 31 00 00 9e 01 1d 00 bc 52 d0 1e 20 b8 28 55 40 34 cc 31 00 00 1e 00 00 00 00 00 00 00 00 00 63 ---------------- Block 0, Base EDID: EDID Structure Version & Revision: 1.3 Vendor & Product Identification: Manufacturer: SHP Model: 4130 Model year: 2008 Basic Display Parameters & Features: Digital display Maximum image size: 82 cm x 46 cm Gamma: 2.20 DPMS levels: Off RGB color display First detailed timing is the preferred timing Color Characteristics: Red : 0.6328, 0.3330 Green: 0.2050, 0.7021 Blue : 0.1503, 0.0810 White: 0.2919, 0.3222 Established Timings I & II: IBM : 720x400 70.082 Hz 9:5 31.467 kHz 28.320 MHz DMT 0x04: 640x480 59.940 Hz 4:3 31.469 kHz 25.175 MHz DMT 0x05: 640x480 72.809 Hz 4:3 37.861 kHz 31.500 MHz DMT 0x06: 640x480 75.000 Hz 4:3 37.500 kHz 31.500 MHz DMT 0x08: 800x600 56.250 Hz 4:3 35.156 kHz 36.000 MHz DMT 0x09: 800x600 60.317 Hz 4:3 37.879 kHz 40.000 MHz DMT 0x0a: 800x600 72.188 Hz 4:3 48.077 kHz 50.000 MHz DMT 0x0b: 800x600 75.000 Hz 4:3 46.875 kHz 49.500 MHz DMT 0x10: 1024x768 60.004 Hz 4:3 48.363 kHz 65.000 MHz DMT 0x11: 1024x768 70.069 Hz 4:3 56.476 kHz 75.000 MHz DMT 0x12: 1024x768 75.029 Hz 4:3 60.023 kHz 78.750 MHz Standard Timings: DMT 0x33: 1600x1200 60.000 Hz 4:3 75.000 kHz 162.000 MHz DMT 0x2a: 1400x1050 59.978 Hz 4:3 65.317 kHz 121.750 MHz DMT 0x23: 1280x1024 60.020 Hz 5:4 63.981 kHz 108.000 MHz Detailed Timing Descriptors: DTD 1: 1920x1080 50.000 Hz 16:9 56.250 kHz 148.500 MHz (820 mm x 460 mm) Hfront 528 Hsync 44 Hback 148 Hpol P Vfront 4 Vsync 5 Vback 36 Vpol N DTD 2: 1360x768 60.015 Hz 85:48 47.712 kHz 85.500 MHz Hfront 64 Hsync 112 Hback 256 Hpol P Vfront 3 Vsync 6 Vback 18 Vpol P Display Product Name: 'SHARP HDMI' Display Range Limits: Monitor ranges (GTF): 23-76 Hz V, 15-75 kHz H, max dotclock 170 MHz Extension blocks: 1 Checksum: 0x40 ---------------- Block 1, CTA-861 Extension Block: Revision: 3 Basic audio support Supports YCbCr 4:4:4 Supports YCbCr 4:2:2 Native detailed modes: 2 Video Data Block: VIC 31: 1920x1080 50.000 Hz 16:9 56.250 kHz 148.500 MHz (native) VIC 16: 1920x1080 60.000 Hz 16:9 67.500 kHz 148.500 MHz (native) VIC 32: 1920x1080 24.000 Hz 16:9 27.000 kHz 74.250 MHz VIC 20: 1920x1080i 50.000 Hz 16:9 28.125 kHz 74.250 MHz VIC 5: 1920x1080i 60.000 Hz 16:9 33.750 kHz 74.250 MHz VIC 19: 1280x720 50.000 Hz 16:9 37.500 kHz 74.250 MHz VIC 4: 1280x720 60.000 Hz 16:9 45.000 kHz 74.250 MHz VIC 18: 720x576 50.000 Hz 16:9 31.250 kHz 27.000 MHz VIC 3: 720x480 59.940 Hz 16:9 31.469 kHz 27.000 MHz VIC 17: 720x576 50.000 Hz 4:3 31.250 kHz 27.000 MHz VIC 2: 720x480 59.940 Hz 4:3 31.469 kHz 27.000 MHz VIC 22: 1440x576i 50.000 Hz 16:9 15.625 kHz 27.000 MHz VIC 7: 1440x480i 59.940 Hz 16:9 15.734 kHz 27.000 MHz VIC 21: 1440x576i 50.000 Hz 4:3 15.625 kHz 27.000 MHz VIC 6: 1440x480i 59.940 Hz 4:3 15.734 kHz 27.000 MHz VIC 1: 640x480 59.940 Hz 4:3 31.469 kHz 25.175 MHz Audio Data Block: Linear PCM: Max channels: 2 Supported sample rates (kHz): 48 44.1 32 Supported sample sizes (bits): 16 Speaker Allocation Data Block: FL/FR - Front Left/Right Colorimetry Data Block: xvYCC601 xvYCC709 Vendor-Specific Data Block (HDMI), OUI 00-0C-03: Source physical address: 2.0.0.0 Supports_AI Maximum TMDS clock: 170 MHz Video latency: 35 ms Audio latency: 35 ms Interlaced video latency: 51 ms Interlaced audio latency: 51 ms Detailed Timing Descriptors: DTD 3: 1920x1080 60.000 Hz 16:9 67.500 kHz 148.500 MHz (820 mm x 460 mm) Hfront 88 Hsync 44 Hback 148 Hpol P Vfront 4 Vsync 5 Vback 36 Vpol P DTD 4: 1920x1080i 50.000 Hz 16:9 28.125 kHz 74.250 MHz (820 mm x 460 mm) Hfront 528 Hsync 44 Hback 148 Hpol P Vfront 2 Vsync 5 Vback 15 Vpol P Vfront +0.5 Odd Field Vfront 2 Vsync 5 Vback 15 Vpol P Vback +0.5 Even Field DTD 5: 1920x1080i 60.000 Hz 16:9 33.750 kHz 74.250 MHz (820 mm x 460 mm) Hfront 88 Hsync 44 Hback 148 Hpol P Vfront 2 Vsync 5 Vback 15 Vpol P Vfront +0.5 Odd Field Vfront 2 Vsync 5 Vback 15 Vpol P Vback +0.5 Even Field DTD 6: 1280x720 50.000 Hz 16:9 37.500 kHz 74.250 MHz (820 mm x 460 mm) Hfront 440 Hsync 40 Hback 220 Hpol P Vfront 5 Vsync 5 Vback 20 Vpol P Checksum: 0x63 ``` Since an error code (`-EIO`) is passed on from `__snd_pcm_info_eld_fixup`, the HDMI audio output goes away altogether instead of just keeping the old name. Will send a PR to keep the audio working, but it would be nice to actually make the new naming work as well. Issue URL : https://github.com/alsa-project/alsa-lib/issues/233 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

ASUS ROG Maximus Z690 Hero (ALC4082) - No audio from microphone, no audio on the front panel
by GitHub issues - edited 04 Jun '22

04 Jun '22

alsa-project/alsa-ucm-conf issue #165 was edited from Lastique: I'm having a problem with getting audio from the microphone connected to the rear panel jack. Also, if I connect the headset to the front panel headphone/mic jacks, I get neither audio out or in. My main concern is the rear panel jack though. The motherboard is ASUS ROG Maximus Z690 Hero, ALC4082 USB Audio chipset. I have tried this: 1. Installed linux kernel 5.18.0. 2. Copied `ucm2` from alsa-ucm-conf master (rev ef3b6513b6581705d36432428a7f6a4ae23b1276 which is after https://github.com/alsa-project/alsa-ucm-conf/pull/143 was merged) to `/usr/share/alsa`. 3. In `/usr/share/alsa/ucm2/USB-Audio/USB-Audio.conf` added hardware id of my USB Audio device (`0b05:1a27`) to the `realtek-alc4080` rule. Also added the hardware id in `/usr/share/alsa/ucm2/USB-Audio/Realtek/ALC4080-HiFi.conf`, in the `asus-rog-usb` block condition. 4. Rebooted. 5. The headset is connected to the rear panel (Front out and Mic in jacks; also tried Line in). There are three devices for the USB Audio card, and there are three volume controls in alsamixer: "Line", "Mic" and "Analog In". The problem is that there is no audio coming from the first two devices (which are presumably "Line" and "Mic"). I can record audio from the third device ("Analog In") when the microphone is connected to Mic jack. Unfortunately, "Analog In" cannot be selected in PulseAudio for some reason (I created a [bug](https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/1364) for PA), which means no audio can be recorded. I also tried to connect the headset to the front panel connectors, but got no audio output nor capture. This is not my intended configuration though, I normally connect the headset to the rear panel. `arecord` output: [arecord.txt](https://github.com/alsa-project/alsa-ucm-conf/files/8838239/ar… `lsusb` output: [lsusb.txt](https://github.com/alsa-project/alsa-ucm-conf/files/8838241/lsus… Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/165 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

ASUS ROG Maximus Z690 Hero (ALC4082) - No audio from microphone, no audio on the front panel
by GitHub issues - opened 04 Jun '22

04 Jun '22

alsa-project/alsa-ucm-conf issue #165 was opened from Lastique: I'm having a problem with getting audio from the microphone connected to the rear panel jack. Also, if I connect the headset to the front panel headphone/mic jacks, I get neither audio out or in. My main concern is the rear panel jack though. The motherboard is ASUS ROG Maximus Z690 Hero, ALC4082 USB Audio chipset. I have tried this: 1. Installed linux kernel 5.18.0. 2. Copied `ucm2` from alsa-ucm-conf master (rev ef3b6513b6581705d36432428a7f6a4ae23b1276 which is after https://github.com/alsa-project/alsa-ucm-conf/pull/143 was merged) to `/usr/share/alsa`. 3. In `/usr/share/alsa/ucm2/USB-Audio/USB-Audio.conf` added hardware id of my USB Audio device (`0b05:1a27`) to the `realtek-alc4080` rule. Also added the hardware id in `/usr/share/alsa/ucm2/USB-Audio/Realtek/ALC4080-HiFi.conf`, in the `asus-rog-usb` block condition. 4. Rebooted. 5. The headset is connected to the rear panel (Front out and Mic in jacks; also tried Line in). There are three devices for the USB Audio card, and there are three volume controls in alsamixer: "Line", "Mic" and "Analog In". The problem is that there is no audio coming from the first two devices (which are presumably "Line" and "Mic"). I can record audio from the third device ("Analog In") when the microphone is connected to Mic jack. Unfortunately, "Analog In" cannot be selected in PulseAudio for some reason (I created a [bug](https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/1364) for PA), which means no audio can be recorded. `arecord` output: [arecord.txt](https://github.com/alsa-project/alsa-ucm-conf/files/8838239/ar… `lsusb` output: [lsusb.txt](https://github.com/alsa-project/alsa-ucm-conf/files/8838241/lsus… Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/165 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Fixes for veyron chromebooks
by GitHub pull_request - opened 04 Jun '22

04 Jun '22

alsa-project/alsa-ucm-conf pull request #164 was opened from jenneron: Add PlaybackVolume and JackControl. I didn't add JackControl for headset mic because internal mic breaks after switching to headset and back Add symlink to VEYRON-I2S, so ucm config will get probed Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/164 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/164.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Re: [Sound-open-firmware] jack detection via input/event on linux
by Pierre-Louis Bossart 03 Jun '22

03 Jun '22

[Adding alsa-devel, Mark and Takashi to this thread] On 6/3/22 08:33, Jaroslav Kysela wrote: > On 03. 06. 22 13:17, hede wrote: >> Am 03.06.2022 11:34, schrieb Jaroslav Kysela: >>> PA/PW use only ALSA controls at the moment - thus you should define >>> JackControl as defined in use-case.h. But it seems that the driver >>> does not create or publish the ALSA jack control (I see only >>> "iface=CARD,name='HDMI/DP,pcm=2 Jack'" jack controls for HDMI). >>> >>> Jaroslav >> >> Ah, thank you Jaroslav. That's the hint I was searching for atm. >> JackDev vs. JackControl. I had a vague guess on this but now you made it >> certain. >> >> Maybe I should take a look into the driver directly. > > It seems that many ASoC drivers do not create kctls - the argument pins > is zero in the snd_soc_card_jack_new() call. The SND_JACK_HEADSET should > be remapped to kctl in sound/soc/intel/boards/sof_rt5682.c, too. Did you mean something like the patch below Jaroslav? If yes, most of the Chromebook machine drivers are missing this remapping. That would also explain why the jack detection didn't work on my GLK test device, despite a UCM file that looked good... diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index 5d67a2c87a1d4..1db17ebd26f63 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -85,6 +85,17 @@ struct sof_card_private { bool idisp_codec; }; +static struct snd_soc_jack_pin sof_jack_pins[] = { + { + .pin = "Headphone", + .mask = SND_JACK_HEADPHONE, + }, + { + .pin = "Headset Mic", + .mask = SND_JACK_MICROPHONE, + }, +}; + static int sof_rt5682_quirk_cb(const struct dmi_system_id *id) { sof_rt5682_quirk = (unsigned long)id->driver_data; @@ -295,11 +306,13 @@ static int sof_rt5682_codec_init(struct snd_soc_pcm_runtime *rtd) * Headset buttons map to the google Reference headset. * These can be configured by userspace. */ - ret = snd_soc_card_jack_new(rtd->card, "Headset Jack", - SND_JACK_HEADSET | SND_JACK_BTN_0 | - SND_JACK_BTN_1 | SND_JACK_BTN_2 | - SND_JACK_BTN_3, - &ctx->sof_headset); + ret = snd_soc_card_jack_new_pins(rtd->card, "Headset Jack", + SND_JACK_HEADSET | SND_JACK_BTN_0 | + SND_JACK_BTN_1 | SND_JACK_BTN_2 | + SND_JACK_BTN_3, + &ctx->sof_headset, + sof_jack_pins, + ARRAY_SIZE(sof_jack_pins)); if (ret) { dev_err(rtd->dev, "Headset Jack creation failed: %d\n", ret); return ret; @@ -481,7 +494,7 @@ static int sof_card_late_probe(struct snd_soc_card *card) } static const struct snd_kcontrol_new sof_controls[] = { - SOC_DAPM_PIN_SWITCH("Headphone Jack"), + SOC_DAPM_PIN_SWITCH("Headphone"), SOC_DAPM_PIN_SWITCH("Headset Mic"), SOC_DAPM_PIN_SWITCH("Left Spk"), SOC_DAPM_PIN_SWITCH("Right Spk"), @@ -489,7 +502,7 @@ static const struct snd_kcontrol_new sof_controls[] = { }; static const struct snd_soc_dapm_widget sof_widgets[] = { - SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_HP("Headphone", NULL), SND_SOC_DAPM_MIC("Headset Mic", NULL), SND_SOC_DAPM_SPK("Left Spk", NULL), SND_SOC_DAPM_SPK("Right Spk", NULL), @@ -501,8 +514,8 @@ static const struct snd_soc_dapm_widget dmic_widgets[] = { static const struct snd_soc_dapm_route sof_map[] = { /* HP jack connectors - unknown if we have jack detection */ - { "Headphone Jack", NULL, "HPOL" }, - { "Headphone Jack", NULL, "HPOR" }, + { "Headphone", NULL, "HPOL" }, + { "Headphone", NULL, "HPOR" }, /* other jacks */ { "IN1P", NULL, "Headset Mic" },

3 3

[PATCH] ASoC: mt6797-mt6351: Fix refcount leak in mt6797_mt6351_dev_probe
by Miaoqian Lin 03 Jun '22

03 Jun '22

of_parse_phandle() returns a node pointer with refcount incremented, we should use of_node_put() on it when not need anymore. Add missing of_node_put() to avoid refcount leak. Fixes: f0ab0bf250da ("ASoC: add mt6797-mt6351 driver and config option") Signed-off-by: Miaoqian Lin <linmq006(a)gmail.com> --- changes in v2: - update Fixes tag v1 link: https://lore.kernel.org/r/20220603083417.9011-1-linmq006@gmail.com/ --- sound/soc/mediatek/mt6797/mt6797-mt6351.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/soc/mediatek/mt6797/mt6797-mt6351.c b/sound/soc/mediatek/mt6797/mt6797-mt6351.c index 496f32bcfb5e..d2f6213a6bfc 100644 --- a/sound/soc/mediatek/mt6797/mt6797-mt6351.c +++ b/sound/soc/mediatek/mt6797/mt6797-mt6351.c @@ -217,7 +217,8 @@ static int mt6797_mt6351_dev_probe(struct platform_device *pdev) if (!codec_node) { dev_err(&pdev->dev, "Property 'audio-codec' missing or invalid\n"); - return -EINVAL; + ret = -EINVAL; + goto put_platform_node; } for_each_card_prelinks(card, i, dai_link) { if (dai_link->codecs->name) @@ -230,6 +231,9 @@ static int mt6797_mt6351_dev_probe(struct platform_device *pdev) dev_err(&pdev->dev, "%s snd_soc_register_card fail %d\n", __func__, ret); + of_node_put(codec_node); +put_platform_node: + of_node_put(platform_node); return ret; } -- 2.25.1

1 0

[PATCH 0/6] Fixup Cirrus SX control usage
by Charles Keepax 03 Jun '22

03 Jun '22

Mostly the usage of the SX controls seems to match the lowest gain value + number of gain levels expected. The one notable exception there being cs53l30 as David noted. However, there are a couple of other places where the minimum value/TLVs are slightly incorrectly specified. There are two I couldn't quite work out yet, any input on these greatly appreciated: SOC_DOUBLE_R_SX_TLV("Speaker Volume", CS42L52_SPKA_VOL, CS42L52_SPKB_VOL, 0, 0x40, 0xC0, hl_tlv), 0.5dB step 0x40 - -96dB 0x00 - 0dB This one is weird, the datasheet says 0x40 is the minimum, which matches, but the datasheet also says 0.5dB steps to 0x00 being 0dB which doesn't match with 0x40 being the minimum, or 0xC0 being the number of levels. Some clarification will needed from the hardware guys. SOC_SINGLE_SX_TLV("Beep Volume", CS42L56_BEEP_FREQ_OFFTIME, 0, 0x07, 0x23, beep_tlv), 2dB step 0x07 - -50dB 0x00 - 0dB 0x06 - +6dB The minimum checks out at 0x07, but rest of this is weird. The values in the datasheet don't make sense, based on starting at -50dB with a 2dB step, 0x06 should equate to 12dB, over 31 volume levels, not 6dB over 35 volume levels as stated. The control specifies 0x23(35) levels, which takes us up to a final value of 0x0A at 20dB. Again some clarification from the hardware guys might be needed here. Finally it is worth noting I don't actually have hardware for most of these devices, so mostly this is just build tested. Thanks, Charles Charles Keepax (6): ASoC: cs42l52: Fix TLV scales for mixer controls ASoC: cs35l36: Update digital volume TLV ASoC: cs53l30: Correct number of volume levels on SX controls ASoC: cs42l52: Correct TLV for Bypass Volume ASoC: cs42l56: Correct typo in minimum level for SX volume controls ASoC: cs42l51: Correct minimum value for SX volume control sound/soc/codecs/cs35l36.c | 3 ++- sound/soc/codecs/cs42l51.c | 2 +- sound/soc/codecs/cs42l52.c | 8 +++++--- sound/soc/codecs/cs42l56.c | 4 ++-- sound/soc/codecs/cs53l30.c | 16 ++++++++-------- 5 files changed, 18 insertions(+), 15 deletions(-) -- 2.30.2

2 7

[PATCH v2] ASoC: Intel: cirrus-common: fix incorrect channel mapping
by Brent Lu 03 Jun '22

03 Jun '22

From: xliu <xiang.liu(a)cirrus.com> The default mapping of ASPRX1 (DAC source) is slot 0. Change the slot mapping of right amplifiers (WR and TR) to slot 1 to receive right channel data. Also update the ACPI instance ID mapping according to HW configuration. Signed-off-by: xliu <xiang.liu(a)cirrus.com> Signed-off-by: Brent Lu <brent.lu(a)intel.com> --- sound/soc/intel/boards/sof_cirrus_common.c | 40 +++++++++++++++++++--- 1 file changed, 36 insertions(+), 4 deletions(-) diff --git a/sound/soc/intel/boards/sof_cirrus_common.c b/sound/soc/intel/boards/sof_cirrus_common.c index e71d74ec1b0b..f4192df962d6 100644 --- a/sound/soc/intel/boards/sof_cirrus_common.c +++ b/sound/soc/intel/boards/sof_cirrus_common.c @@ -54,22 +54,29 @@ static struct snd_soc_dai_link_component cs35l41_components[] = { }, }; +/* + * Mapping between ACPI instance id and speaker position. + * + * Four speakers: + * 0: Tweeter left, 1: Woofer left + * 2: Tweeter right, 3: Woofer right + */ static struct snd_soc_codec_conf cs35l41_codec_conf[] = { { .dlc = COMP_CODEC_CONF(CS35L41_DEV0_NAME), - .name_prefix = "WL", + .name_prefix = "TL", }, { .dlc = COMP_CODEC_CONF(CS35L41_DEV1_NAME), - .name_prefix = "WR", + .name_prefix = "WL", }, { .dlc = COMP_CODEC_CONF(CS35L41_DEV2_NAME), - .name_prefix = "TL", + .name_prefix = "TR", }, { .dlc = COMP_CODEC_CONF(CS35L41_DEV3_NAME), - .name_prefix = "TR", + .name_prefix = "WR", }, }; @@ -101,6 +108,21 @@ static int cs35l41_init(struct snd_soc_pcm_runtime *rtd) return ret; } +/* + * Channel map: + * + * TL/WL: ASPRX1 on slot 0, ASPRX2 on slot 1 (default) + * TR/WR: ASPRX1 on slot 1, ASPRX2 on slot 0 + */ +static const struct { + unsigned int rx[2]; +} cs35l41_channel_map[] = { + {.rx = {0, 1}}, /* TL */ + {.rx = {0, 1}}, /* WL */ + {.rx = {1, 0}}, /* TR */ + {.rx = {1, 0}}, /* WR */ +}; + static int cs35l41_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { @@ -134,6 +156,16 @@ static int cs35l41_hw_params(struct snd_pcm_substream *substream, ret); return ret; } + + /* setup channel map */ + ret = snd_soc_dai_set_channel_map(codec_dai, 0, NULL, + ARRAY_SIZE(cs35l41_channel_map[i].rx), + (unsigned int *)cs35l41_channel_map[i].rx); + if (ret < 0) { + dev_err(codec_dai->dev, "fail to set channel map, ret %d\n", + ret); + return ret; + } } return 0; -- 2.25.1

4 3

[PATCH V0 1/1] asoc: msm: use hashtable to check kcontrol
by Raghu Bankapur 03 Jun '22

03 Jun '22

use hashtable instead of linear list to check kcontrol before adding them for improving early audio KPI. With this changes we see 600ms improvement in start of audio Change-Id: I1a95d88ef41b48430b57307a0e6a8e82788b4372 Signed-off-by: Raghu Bankapur <quic_rbankapu(a)quicinc.com> --- include/sound/control.h | 2 + include/sound/core.h | 5 +- sound/core/control.c | 106 +++++++++++++++++++++++----------------- sound/core/init.c | 1 + 4 files changed, 68 insertions(+), 46 deletions(-) diff --git a/include/sound/control.h b/include/sound/control.h index 985c51a8fb74..e50db5c45114 100644 --- a/include/sound/control.h +++ b/include/sound/control.h @@ -70,6 +70,8 @@ struct snd_kcontrol_volatile { struct snd_kcontrol { struct list_head list; /* list of controls */ struct snd_ctl_elem_id id; + struct hlist_node hnode; + unsigned int knametoint; /* kctl name to uint, hash key value */ unsigned int count; /* count of same elements */ snd_kcontrol_info_t *info; snd_kcontrol_get_t *get; diff --git a/include/sound/core.h b/include/sound/core.h index b7e9b58d3c78..90341d6f1573 100644 --- a/include/sound/core.h +++ b/include/sound/core.h @@ -14,6 +14,7 @@ #include <linux/pm.h> /* pm_message_t */ #include <linux/stringify.h> #include <linux/printk.h> +#include <linux/hashtable.h> /* number of supported soundcards */ #ifdef CONFIG_SND_DYNAMIC_MINORS @@ -24,6 +25,8 @@ #define CONFIG_SND_MAJOR 116 /* standard configuration */ +#define SND_CTL_HASH_TABLE_BITS 14 /* buckets numbers: 1 << 14 */ + /* forward declarations */ struct pci_dev; struct module; @@ -103,7 +106,7 @@ struct snd_card { size_t user_ctl_alloc_size; // current memory allocation by user controls. struct list_head controls; /* all controls for this card */ struct list_head ctl_files; /* active control files */ - + DECLARE_HASHTABLE(ctl_htable, SND_CTL_HASH_TABLE_BITS); struct snd_info_entry *proc_root; /* root for soundcard specific files */ struct proc_dir_entry *proc_root_link; /* number link to real id */ diff --git a/sound/core/control.c b/sound/core/control.c index a25c0d64d104..e00a02015837 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -331,43 +331,49 @@ void snd_ctl_free_one(struct snd_kcontrol *kcontrol) } EXPORT_SYMBOL(snd_ctl_free_one); -static bool snd_ctl_remove_numid_conflict(struct snd_card *card, - unsigned int count) +enum snd_ctl_add_mode { + CTL_ADD_EXCLUSIVE, CTL_REPLACE, CTL_ADD_ON_REPLACE, +}; + +char snd_ctl_string[50] = { '\0' }; + +/* Used to convert the string into int value -- BKDRHash */ +static unsigned int snd_ctl_strtoint(const char *s) { - struct snd_kcontrol *kctl; + unsigned int res = 0; - /* Make sure that the ids assigned to the control do not wrap around */ - if (card->last_numid >= UINT_MAX - count) - card->last_numid = 0; + while (*s) + res = (res << 5) - res + (*s++); - list_for_each_entry(kctl, &card->controls, list) { - if (kctl->id.numid < card->last_numid + 1 + count && - kctl->id.numid + kctl->count > card->last_numid + 1) { - card->last_numid = kctl->id.numid + kctl->count - 1; - return true; - } - } - return false; + return (res & 0x7FFFFFFF); } -static int snd_ctl_find_hole(struct snd_card *card, unsigned int count) +/** + * snd_ctl_hash_check - Check the duplicate enrty on snd hashtable + * @card: the card instance + * @nametoint: kctl name to uint + * + * Finds the control instance with the given nametoint from the card. + * + * Return: The pointer of the instance if found, or %NULL if not. + * + */ +static struct snd_kcontrol *snd_ctl_hash_check(struct snd_card *card, + unsigned int nametoint) { - unsigned int iter = 100000; + struct snd_kcontrol *kctl = NULL; - while (snd_ctl_remove_numid_conflict(card, count)) { - if (--iter == 0) { - /* this situation is very unlikely */ - dev_err(card->dev, "unable to allocate new control numid\n"); - return -ENOMEM; - } + if (snd_BUG_ON(!card)) + return NULL; + + hash_for_each_possible(card->ctl_htable, kctl, hnode, nametoint) { + if (kctl->knametoint != nametoint) + continue; + return kctl; } - return 0; + return NULL; } -enum snd_ctl_add_mode { - CTL_ADD_EXCLUSIVE, CTL_REPLACE, CTL_ADD_ON_REPLACE, -}; - /* add/replace a new kcontrol object; call with card->controls_rwsem locked */ static int __snd_ctl_add_replace(struct snd_card *card, struct snd_kcontrol *kcontrol, @@ -382,26 +388,34 @@ static int __snd_ctl_add_replace(struct snd_card *card, if (id.index > UINT_MAX - kcontrol->count) return -EINVAL; - old = snd_ctl_find_id(card, &id); - if (!old) { - if (mode == CTL_REPLACE) - return -EINVAL; - } else { - if (mode == CTL_ADD_EXCLUSIVE) { - dev_err(card->dev, - "control %i:%i:%i:%s:%i is already present\n", - id.iface, id.device, id.subdevice, id.name, - id.index); - return -EBUSY; - } + snprintf(snd_ctl_string, strlen(kcontrol->id.name) + 6, "%s%d%d%d", + kcontrol->id.name, kcontrol->id.iface, kcontrol->id.device, + kcontrol->id.subdevice); - err = snd_ctl_remove(card, old); - if (err < 0) - return err; - } + kcontrol->knametoint = snd_ctl_strtoint(snd_ctl_string); + if (kcontrol->knametoint < 0) + return -EINVAL; - if (snd_ctl_find_hole(card, kcontrol->count) < 0) - return -ENOMEM; + old = snd_ctl_hash_check(card, kcontrol->knametoint); + if (old) { + old = snd_ctl_find_id(card, &id); + if (!old) { + if (mode == CTL_REPLACE) + return -EINVAL; + } else { + if (mode == CTL_ADD_EXCLUSIVE) { + dev_err(card->dev, + "control %i:%i:%i:%s:%i is already present\n", + id.iface, id.device, id.subdevice, id.name, + id.index); + return -EBUSY; + } + + err = snd_ctl_remove(card, old); + if (err < 0) + return err; + } + } list_add_tail(&kcontrol->list, &card->controls); card->controls_count += kcontrol->count; @@ -411,6 +425,8 @@ static int __snd_ctl_add_replace(struct snd_card *card, for (idx = 0; idx < kcontrol->count; idx++) snd_ctl_notify_one(card, SNDRV_CTL_EVENT_MASK_ADD, kcontrol, idx); + hash_add(card->ctl_htable, &kcontrol->hnode, kcontrol->knametoint); + return 0; } diff --git a/sound/core/init.c b/sound/core/init.c index 31ba7024e3ad..24138902e5f2 100644 --- a/sound/core/init.c +++ b/sound/core/init.c @@ -284,6 +284,7 @@ static int snd_card_init(struct snd_card *card, struct device *parent, INIT_LIST_HEAD(&card->ctl_files); spin_lock_init(&card->files_lock); INIT_LIST_HEAD(&card->files_list); + hash_init(card->ctl_htable); mutex_init(&card->memory_mutex); #ifdef CONFIG_PM init_waitqueue_head(&card->power_sleep); -- 2.17.1

1 0

[PATCH V1 1/1] asoc: msm: use hashtable to check kcontrol
by Raghu Bankapur 03 Jun '22

03 Jun '22

use hashtabe instead of linear list to check kcontrol before adding them for improving early audio KPI. Change-Id: I7134816736e08e338c0f22a8ae283a0520aa847a Signed-off-by: Raghu Bankapur <quic_rbankapu(a)quicinc.com> --- include/sound/control.h | 2 + include/sound/core.h | 5 +- sound/core/control.c | 106 +++++++++++++++++++++++----------------- sound/core/init.c | 1 + 4 files changed, 68 insertions(+), 46 deletions(-) diff --git a/include/sound/control.h b/include/sound/control.h index 985c51a8fb74..e50db5c45114 100644 --- a/include/sound/control.h +++ b/include/sound/control.h @@ -70,6 +70,8 @@ struct snd_kcontrol_volatile { struct snd_kcontrol { struct list_head list; /* list of controls */ struct snd_ctl_elem_id id; + struct hlist_node hnode; + unsigned int knametoint; /* kctl name to uint, hash key value */ unsigned int count; /* count of same elements */ snd_kcontrol_info_t *info; snd_kcontrol_get_t *get; diff --git a/include/sound/core.h b/include/sound/core.h index b7e9b58d3c78..90341d6f1573 100644 --- a/include/sound/core.h +++ b/include/sound/core.h @@ -14,6 +14,7 @@ #include <linux/pm.h> /* pm_message_t */ #include <linux/stringify.h> #include <linux/printk.h> +#include <linux/hashtable.h> /* number of supported soundcards */ #ifdef CONFIG_SND_DYNAMIC_MINORS @@ -24,6 +25,8 @@ #define CONFIG_SND_MAJOR 116 /* standard configuration */ +#define SND_CTL_HASH_TABLE_BITS 14 /* buckets numbers: 1 << 14 */ + /* forward declarations */ struct pci_dev; struct module; @@ -103,7 +106,7 @@ struct snd_card { size_t user_ctl_alloc_size; // current memory allocation by user controls. struct list_head controls; /* all controls for this card */ struct list_head ctl_files; /* active control files */ - + DECLARE_HASHTABLE(ctl_htable, SND_CTL_HASH_TABLE_BITS); struct snd_info_entry *proc_root; /* root for soundcard specific files */ struct proc_dir_entry *proc_root_link; /* number link to real id */ diff --git a/sound/core/control.c b/sound/core/control.c index a25c0d64d104..e00a02015837 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -331,43 +331,49 @@ void snd_ctl_free_one(struct snd_kcontrol *kcontrol) } EXPORT_SYMBOL(snd_ctl_free_one); -static bool snd_ctl_remove_numid_conflict(struct snd_card *card, - unsigned int count) +enum snd_ctl_add_mode { + CTL_ADD_EXCLUSIVE, CTL_REPLACE, CTL_ADD_ON_REPLACE, +}; + +char snd_ctl_string[50] = { '\0' }; + +/* Used to convert the string into int value -- BKDRHash */ +static unsigned int snd_ctl_strtoint(const char *s) { - struct snd_kcontrol *kctl; + unsigned int res = 0; - /* Make sure that the ids assigned to the control do not wrap around */ - if (card->last_numid >= UINT_MAX - count) - card->last_numid = 0; + while (*s) + res = (res << 5) - res + (*s++); - list_for_each_entry(kctl, &card->controls, list) { - if (kctl->id.numid < card->last_numid + 1 + count && - kctl->id.numid + kctl->count > card->last_numid + 1) { - card->last_numid = kctl->id.numid + kctl->count - 1; - return true; - } - } - return false; + return (res & 0x7FFFFFFF); } -static int snd_ctl_find_hole(struct snd_card *card, unsigned int count) +/** + * snd_ctl_hash_check - Check the duplicate enrty on snd hashtable + * @card: the card instance + * @nametoint: kctl name to uint + * + * Finds the control instance with the given nametoint from the card. + * + * Return: The pointer of the instance if found, or %NULL if not. + * + */ +static struct snd_kcontrol *snd_ctl_hash_check(struct snd_card *card, + unsigned int nametoint) { - unsigned int iter = 100000; + struct snd_kcontrol *kctl = NULL; - while (snd_ctl_remove_numid_conflict(card, count)) { - if (--iter == 0) { - /* this situation is very unlikely */ - dev_err(card->dev, "unable to allocate new control numid\n"); - return -ENOMEM; - } + if (snd_BUG_ON(!card)) + return NULL; + + hash_for_each_possible(card->ctl_htable, kctl, hnode, nametoint) { + if (kctl->knametoint != nametoint) + continue; + return kctl; } - return 0; + return NULL; } -enum snd_ctl_add_mode { - CTL_ADD_EXCLUSIVE, CTL_REPLACE, CTL_ADD_ON_REPLACE, -}; - /* add/replace a new kcontrol object; call with card->controls_rwsem locked */ static int __snd_ctl_add_replace(struct snd_card *card, struct snd_kcontrol *kcontrol, @@ -382,26 +388,34 @@ static int __snd_ctl_add_replace(struct snd_card *card, if (id.index > UINT_MAX - kcontrol->count) return -EINVAL; - old = snd_ctl_find_id(card, &id); - if (!old) { - if (mode == CTL_REPLACE) - return -EINVAL; - } else { - if (mode == CTL_ADD_EXCLUSIVE) { - dev_err(card->dev, - "control %i:%i:%i:%s:%i is already present\n", - id.iface, id.device, id.subdevice, id.name, - id.index); - return -EBUSY; - } + snprintf(snd_ctl_string, strlen(kcontrol->id.name) + 6, "%s%d%d%d", + kcontrol->id.name, kcontrol->id.iface, kcontrol->id.device, + kcontrol->id.subdevice); - err = snd_ctl_remove(card, old); - if (err < 0) - return err; - } + kcontrol->knametoint = snd_ctl_strtoint(snd_ctl_string); + if (kcontrol->knametoint < 0) + return -EINVAL; - if (snd_ctl_find_hole(card, kcontrol->count) < 0) - return -ENOMEM; + old = snd_ctl_hash_check(card, kcontrol->knametoint); + if (old) { + old = snd_ctl_find_id(card, &id); + if (!old) { + if (mode == CTL_REPLACE) + return -EINVAL; + } else { + if (mode == CTL_ADD_EXCLUSIVE) { + dev_err(card->dev, + "control %i:%i:%i:%s:%i is already present\n", + id.iface, id.device, id.subdevice, id.name, + id.index); + return -EBUSY; + } + + err = snd_ctl_remove(card, old); + if (err < 0) + return err; + } + } list_add_tail(&kcontrol->list, &card->controls); card->controls_count += kcontrol->count; @@ -411,6 +425,8 @@ static int __snd_ctl_add_replace(struct snd_card *card, for (idx = 0; idx < kcontrol->count; idx++) snd_ctl_notify_one(card, SNDRV_CTL_EVENT_MASK_ADD, kcontrol, idx); + hash_add(card->ctl_htable, &kcontrol->hnode, kcontrol->knametoint); + return 0; } diff --git a/sound/core/init.c b/sound/core/init.c index 31ba7024e3ad..24138902e5f2 100644 --- a/sound/core/init.c +++ b/sound/core/init.c @@ -284,6 +284,7 @@ static int snd_card_init(struct snd_card *card, struct device *parent, INIT_LIST_HEAD(&card->ctl_files); spin_lock_init(&card->files_lock); INIT_LIST_HEAD(&card->files_list); + hash_init(card->ctl_htable); mutex_init(&card->memory_mutex); #ifdef CONFIG_PM init_waitqueue_head(&card->power_sleep); -- 2.17.1

2 1

[PATCH] ASoC: mediatek: mt8173-rt5650: Fix refcount leak in mt8173_rt5650_dev_probe
by Miaoqian Lin 03 Jun '22

03 Jun '22

of_parse_phandle() returns a node pointer with refcount incremented, we should use of_node_put() on it when not need anymore. Fix refcount leak in some error paths. Fixes: format:0f83f9296d5c ("ASoC: mediatek: Add machine driver for ALC5650 codec") Signed-off-by: Miaoqian Lin <linmq006(a)gmail.com> --- sound/soc/mediatek/mt8173/mt8173-rt5650.c | 9 ++++++--- 1 file changed, 6 insertions(+), 3 deletions(-) diff --git a/sound/soc/mediatek/mt8173/mt8173-rt5650.c b/sound/soc/mediatek/mt8173/mt8173-rt5650.c index d1c94acb4516..e05f2b0231fe 100644 --- a/sound/soc/mediatek/mt8173/mt8173-rt5650.c +++ b/sound/soc/mediatek/mt8173/mt8173-rt5650.c @@ -280,7 +280,8 @@ static int mt8173_rt5650_dev_probe(struct platform_device *pdev) if (!mt8173_rt5650_dais[DAI_LINK_CODEC_I2S].codecs[0].of_node) { dev_err(&pdev->dev, "Property 'audio-codec' missing or invalid\n"); - return -EINVAL; + ret = -EINVAL; + goto put_platform_node; } mt8173_rt5650_dais[DAI_LINK_CODEC_I2S].codecs[1].of_node = mt8173_rt5650_dais[DAI_LINK_CODEC_I2S].codecs[0].of_node; @@ -293,7 +294,7 @@ static int mt8173_rt5650_dev_probe(struct platform_device *pdev) dev_err(&pdev->dev, "%s codec_capture_dai name fail %d\n", __func__, ret); - return ret; + goto put_platform_node; } mt8173_rt5650_dais[DAI_LINK_CODEC_I2S].codecs[1].dai_name = codec_capture_dai; @@ -315,12 +316,14 @@ static int mt8173_rt5650_dev_probe(struct platform_device *pdev) if (!mt8173_rt5650_dais[DAI_LINK_HDMI_I2S].codecs->of_node) { dev_err(&pdev->dev, "Property 'audio-codec' missing or invalid\n"); - return -EINVAL; + ret = -EINVAL; + goto put_platform_node; } card->dev = &pdev->dev; ret = devm_snd_soc_register_card(&pdev->dev, card); +put_platform_node: of_node_put(platform_node); return ret; } -- 2.25.1

1 0

[bug report] ASoC: SOF: ipc-msg-injector: Separate the message sending
by Dan Carpenter 03 Jun '22

03 Jun '22

Hello Peter Ujfalusi, The patch a9aa3381e404: "ASoC: SOF: ipc-msg-injector: Separate the message sending" from May 6, 2022, leads to the following Smatch static checker warning: sound/soc/sof/sof-client-ipc-msg-injector.c:162 sof_msg_inject_dfs_write() warn: kernel error codes cast to unsigned 'size' sound/soc/sof/sof-client-ipc-msg-injector.c 148 static ssize_t sof_msg_inject_dfs_write(struct file *file, const char __user *buffer, 149 size_t count, loff_t *ppos) 150 { 151 struct sof_client_dev *cdev = file->private_data; 152 struct sof_msg_inject_priv *priv = cdev->data; 153 size_t size; 154 int ret; 155 156 if (*ppos) 157 return 0; I think there needs to be an "if (count != priv->max_msg_size)" check or something. Or another option would be the do a memset() memset(priv->tx_buffer, 0, priv->max_msg_size); before the simple_write_to_buffer(). Otherwise if count == 1 then we re-use stale data. 158 159 size = simple_write_to_buffer(priv->tx_buffer, priv->max_msg_size, 160 ppos, buffer, count); 161 if (size != count) --> 162 return size > 0 ? -EFAULT : size; 163 164 memset(priv->rx_buffer, 0, priv->max_msg_size); 165 166 ret = sof_msg_inject_send_message(cdev); 167 168 /* return the error code if test failed */ 169 if (ret < 0) 170 size = ret; 171 172 return size; 173 }; regards, dan carpenter

2 1

Issues with ALSA after update to Pop!_OS 22.04 LTS Jaunty Pipewire
by Mikhail Kravchenko 03 Jun '22

03 Jun '22

I updated the operating system running on my machine to 22.04 Jaunty. This update brings the system over to pipe-wire. After running certain programs that utilize alsa, the audio on my system becomes intermittent and runs only one application's audio at a time. Resetting the pipe-wire services and forcing alsa to reload its modules does not reliably mitigate the issue. This is the url generated by alsa-info. http://alsa-project.org/db/?f=4d643a3153ffabd5c72313e8ffa3c9f393e676ef On a power cycle, the audio seems to work fine, but as soon as I run the problematic program I get the output: ALSA lib pcm_dmix.c:1032:(snd_pcm_dmix_open) unable to open slave If the program's audio is successful, and it opens the slave. Audio to other applications is upset. sudo alsa force-reload Unloading ALSA sound driver modules: snd-seq-dummy snd-seq-midi snd-seq-midi-event snd-hda-codec-realtek snd-hda-codec-generic snd-hda-codec-hdmi snd-hda-intel snd-intel-dspcfg snd-intel-sdw-acpi snd-hda-codec snd-hda-core snd-hrtimer snd-usb-audio snd-usbmidi-lib snd-hwdep snd-rawmidi snd-seq snd-pcm snd-seq-device snd-timer (failed: modules still loaded: snd-hda-codec-realtek snd-hda-codec-generic snd-hda-codec-hdmi snd-hda-intel snd-intel-dspcfg snd-intel-sdw-acpi snd-hda-codec snd-hda-core snd-hrtimer snd-usb-audio snd-usbmidi-lib snd-hwdep snd-rawmidi snd-seq snd-pcm snd-seq-device snd-timer). Loading ALSA sound driver modules: snd-seq-dummy snd-seq-midi snd-seq-midi-event snd-hda-codec-realtek snd-hda-codec-generic snd-hda-codec-hdmi snd-hda-intel snd-intel-dspcfg snd-intel-sdw-acpi snd-hda-codec snd-hda-core snd-hrtimer snd-usb-audio snd-usbmidi-lib snd-hwdep snd-rawmidi snd-seq snd-pcm snd-seq-device snd-timer. MK

1 0

USB-Audio: Steinberg UR44 not getting expected results
by GitHub issues - opened 02 Jun '22

02 Jun '22

alsa-project/alsa-ucm-conf issue #163 was opened from CameronNemo: I am testing the following software: * alsa-lib: 1.2.7 * alsa-utils: 1.2.7 * alsa-ucm-conf: 1.2.7 Unfortunately the changes for the Steinberg UR44 do not seem to be working as intended. I am not seeing the expected mono inputs and stereo outputs that I see defined in `/usr/share/alsa/ucm2/USB-Audio/Steinberg/UR44-HiFi.conf`. I am seeing the following from `aplay -L`: ``` null Discard all samples (playback) or generate zero samples (capture) default Default Audio Device sysdefault Default Audio Device default:CARD=UR44 Steinberg UR44, USB Audio Default Audio Device sysdefault:CARD=UR44 Steinberg UR44, USB Audio Default Audio Device front:CARD=UR44,DEV=0 Steinberg UR44, USB Audio Front output / input surround21:CARD=UR44,DEV=0 Steinberg UR44, USB Audio 2.1 Surround output to Front and Subwoofer speakers surround40:CARD=UR44,DEV=0 Steinberg UR44, USB Audio 4.0 Surround output to Front and Rear speakers surround41:CARD=UR44,DEV=0 Steinberg UR44, USB Audio 4.1 Surround output to Front, Rear and Subwoofer speakers surround50:CARD=UR44,DEV=0 Steinberg UR44, USB Audio 5.0 Surround output to Front, Center and Rear speakers surround51:CARD=UR44,DEV=0 Steinberg UR44, USB Audio 5.1 Surround output to Front, Center, Rear and Subwoofer speakers surround71:CARD=UR44,DEV=0 Steinberg UR44, USB Audio 7.1 Surround output to Front, Center, Side, Rear and Woofer speakers ``` And the following from `arecord -L`: ``` null Discard all samples (playback) or generate zero samples (capture) default Default Audio Device sysdefault Default Audio Device default:CARD=UR44 Steinberg UR44, USB Audio Default Audio Device sysdefault:CARD=UR44 Steinberg UR44, USB Audio Default Audio Device front:CARD=UR44,DEV=0 Steinberg UR44, USB Audio Front output / input ``` `spa-acp-tool list`: ``` card 0: profiles:2 devices:2 ports:2 profile 0: name:"off" prio:0 (available: yes) * profile 1: name:"Direct" prio:1 (available: unknown) * port 0: name:"[Out] Direct" direction:playback prio:1000 (available: unknown) * port 1: name:"[In] Direct" direction:capture prio:1000 (available: unknown) * device 0: direction:playback name:"Direct: hw:UR44: sink" prio:1000 flags:00000009 devices: "hw:UR44" * device 1: direction:capture name:"Direct: hw:UR44: source" prio:1000 flags:00000009 devices: "hw:UR44" ``` `pactl list sources`: ``` Source #56 State: SUSPENDED Name: alsa_output.usb-Yamaha_Corporation_Steinberg_UR44-00.Direct__hw_UR44__sink.monitor Description: Monitor of Steinberg UR44 Direct Steinberg UR44 Driver: PipeWire Sample Specification: s24le 4ch 48000Hz Channel Map: front-left,front-right,rear-left,rear-right Owner Module: 4294967295 Mute: no Volume: front-left: 65536 / 100% / 0.00 dB, front-right: 65536 / 100% / 0.00 dB, rear-left: 65536 / 100% / 0.00 dB, rear-right: 65536 / 100% / 0.00 dB balance 0.00 Base Volume: 65536 / 100% / 0.00 dB Monitor of Sink: alsa_output.usb-Yamaha_Corporation_Steinberg_UR44-00.Direct__hw_UR44__sink Latency: 0 usec, configured 0 usec Flags: HARDWARE DECIBEL_VOLUME LATENCY Properties: object.path = "alsa:pcm:0:hw:UR44:playback" api.alsa.path = "hw:UR44" api.alsa.open.ucm = "true" api.alsa.pcm.card = "0" api.alsa.pcm.stream = "playback" audio.channels = "4" audio.position = "FL,FR,RL,RR" device.routes = "1" alsa.mixer_device = "_ucm0003.hw:UR44" alsa.resolution_bits = "24" device.api = "alsa" device.class = "monitor" alsa.class = "generic" alsa.subclass = "generic-mix" alsa.name = "USB Audio" alsa.id = "USB Audio" alsa.subdevice = "0" alsa.subdevice_name = "subdevice #0" alsa.device = "0" alsa.card = "0" alsa.card_name = "Steinberg UR44" alsa.long_card_name = "Yamaha Corporation Steinberg UR44 at usb-0000:04:00.4-1, high speed" alsa.driver_name = "snd_usb_audio" device.profile.name = "Direct: hw:UR44: sink" device.profile.description = "Direct Steinberg UR44" card.profile.device = "0" device.id = "54" factory.name = "api.alsa.pcm.sink" priority.driver = "1000" priority.session = "1000" media.class = "Audio/Sink" node.nick = "Steinberg UR44" node.name = "alsa_output.usb-Yamaha_Corporation_Steinberg_UR44-00.Direct__hw_UR44__sink" device.description = "Steinberg UR44 Direct Steinberg UR44" device.icon_name = "audio-card" device.bus = "usb" device.bus_path = "pci-0000:04:00.4-usb-0:1:1.0" node.pause-on-idle = "false" factory.id = "18" client.id = "34" clock.quantum-limit = "8192" node.driver = "true" factory.mode = "merge" audio.adapt.follower = "" library.name = "audioconvert/libspa-audioconvert" object.id = "56" object.serial = "56" Formats: pcm Source #57 State: SUSPENDED Name: alsa_input.usb-Yamaha_Corporation_Steinberg_UR44-00.Direct__hw_UR44__source Description: Steinberg UR44 Direct Steinberg UR44 Driver: PipeWire Sample Specification: s24le 6ch 48000Hz Channel Map: front-left,front-right,rear-left,rear-right,front-center,lfe Owner Module: 4294967295 Mute: no Volume: front-left: 48287 / 74% / -7.96 dB, front-right: 48287 / 74% / -7.96 dB, rear-left: 48287 / 74% / -7.96 dB, rear-right: 48287 / 74% / -7.96 dB, front-center: 48287 / 74% / -7.96 dB, lfe: 48287 / 74% / -7.96 dB balance 0.00 Base Volume: 65536 / 100% / 0.00 dB Monitor of Sink: n/a Latency: 0 usec, configured 0 usec Flags: HARDWARE DECIBEL_VOLUME LATENCY Properties: object.path = "alsa:pcm:0:hw:UR44:capture" api.alsa.path = "hw:UR44" api.alsa.open.ucm = "true" api.alsa.pcm.card = "0" api.alsa.pcm.stream = "capture" audio.channels = "6" audio.position = "FL,FR,RL,RR,FC,LFE" device.routes = "1" alsa.mixer_device = "_ucm0003.hw:UR44" alsa.resolution_bits = "24" device.api = "alsa" device.class = "sound" alsa.class = "generic" alsa.subclass = "generic-mix" alsa.name = "USB Audio" alsa.id = "USB Audio" alsa.subdevice = "0" alsa.subdevice_name = "subdevice #0" alsa.device = "0" alsa.card = "0" alsa.card_name = "Steinberg UR44" alsa.long_card_name = "Yamaha Corporation Steinberg UR44 at usb-0000:04:00.4-1, high speed" alsa.driver_name = "snd_usb_audio" device.profile.name = "Direct: hw:UR44: source" device.profile.description = "Direct Steinberg UR44" card.profile.device = "1" device.id = "54" factory.name = "api.alsa.pcm.source" priority.driver = "2000" priority.session = "2000" media.class = "Audio/Source" node.nick = "Steinberg UR44" node.name = "alsa_input.usb-Yamaha_Corporation_Steinberg_UR44-00.Direct__hw_UR44__source" device.description = "Steinberg UR44 Direct Steinberg UR44" device.icon_name = "audio-input-microphone" device.bus = "usb" device.bus_path = "pci-0000:04:00.4-usb-0:1:1.0" node.pause-on-idle = "false" factory.id = "18" client.id = "34" clock.quantum-limit = "8192" node.driver = "true" factory.mode = "split" audio.adapt.follower = "" library.name = "audioconvert/libspa-audioconvert" object.id = "57" object.serial = "57" Ports: [In] Direct: Direct Steinberg UR44 (type: Unknown, priority: 1000, availability unknown) Active Port: [In] Direct Formats: pcm ``` `pactl list sinks`: ``` Sink #56 State: SUSPENDED Name: alsa_output.usb-Yamaha_Corporation_Steinberg_UR44-00.Direct__hw_UR44__sink Description: Steinberg UR44 Direct Steinberg UR44 Driver: PipeWire Sample Specification: s24le 4ch 48000Hz Channel Map: front-left,front-right,rear-left,rear-right Owner Module: 4294967295 Mute: no Volume: front-left: 48287 / 74% / -7.96 dB, front-right: 48287 / 74% / -7.96 dB, rear-left: 48287 / 74% / -7.96 dB, rear-right: 48287 / 74% / -7.96 dB balance 0.00 Base Volume: 65536 / 100% / 0.00 dB Monitor Source: alsa_output.usb-Yamaha_Corporation_Steinberg_UR44-00.Direct__hw_UR44__sink.monitor Latency: 0 usec, configured 0 usec Flags: HARDWARE DECIBEL_VOLUME LATENCY Properties: object.path = "alsa:pcm:0:hw:UR44:playback" api.alsa.path = "hw:UR44" api.alsa.open.ucm = "true" api.alsa.pcm.card = "0" api.alsa.pcm.stream = "playback" audio.channels = "4" audio.position = "FL,FR,RL,RR" device.routes = "1" alsa.mixer_device = "_ucm0003.hw:UR44" alsa.resolution_bits = "24" device.api = "alsa" device.class = "sound" alsa.class = "generic" alsa.subclass = "generic-mix" alsa.name = "USB Audio" alsa.id = "USB Audio" alsa.subdevice = "0" alsa.subdevice_name = "subdevice #0" alsa.device = "0" alsa.card = "0" alsa.card_name = "Steinberg UR44" alsa.long_card_name = "Yamaha Corporation Steinberg UR44 at usb-0000:04:00.4-1, high speed" alsa.driver_name = "snd_usb_audio" device.profile.name = "Direct: hw:UR44: sink" device.profile.description = "Direct Steinberg UR44" card.profile.device = "0" device.id = "54" factory.name = "api.alsa.pcm.sink" priority.driver = "1000" priority.session = "1000" media.class = "Audio/Sink" node.nick = "Steinberg UR44" node.name = "alsa_output.usb-Yamaha_Corporation_Steinberg_UR44-00.Direct__hw_UR44__sink" device.description = "Steinberg UR44 Direct Steinberg UR44" device.icon_name = "audio-card" device.bus = "usb" device.bus_path = "pci-0000:04:00.4-usb-0:1:1.0" node.pause-on-idle = "false" factory.id = "18" client.id = "34" clock.quantum-limit = "8192" node.driver = "true" factory.mode = "merge" audio.adapt.follower = "" library.name = "audioconvert/libspa-audioconvert" object.id = "56" object.serial = "56" Ports: [Out] Direct: Direct Steinberg UR44 (type: Unknown, priority: 1000, availability unknown) Active Port: [Out] Direct Formats: pcm ``` Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/163 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH 0/4] hwdep: define interface for hwdep device
by GitHub pull_request - edited 02 Jun '22

02 Jun '22

alsa-project/alsa-gobject pull request #73 was edited from takaswie: The protocol in ALSA HwDep character device has common feature to each device. It's suitable for userspace implementation to have GObject interface for such features. This patchset adds the interface. The implementation of interface should have four methods; open, get_protocol_version, get_device_info, and create_source. Additionally, it should emit handle-disconnection signal when detecting ENODEV error in the interation of source. It requires to notify userspace application since the application should release character device immediately. ``` Takashi Sakamoto (4): hwdep: use explicit value for iface type enumeration hwdep: device-common: add interface for common feature of ALSA HwDep device hwdep: add error domain for implementators of DeviceCommon interface hwdep: device-info: allow to set property at instantiation src/hwdep/alsahwdep-enum-types.h | 89 +++++++++------- src/hwdep/alsahwdep.h | 2 + src/hwdep/alsahwdep.map | 10 ++ src/hwdep/device-common.c | 178 +++++++++++++++++++++++++++++++ src/hwdep/device-common.h | 106 ++++++++++++++++++ src/hwdep/device-info.c | 38 ++++++- src/hwdep/meson.build | 2 + tests/alsahwdep-enums | 10 ++ 8 files changed, 389 insertions(+), 46 deletions(-) create mode 100644 src/hwdep/device-common.c create mode 100644 src/hwdep/device-common.h ``` Request URL : https://github.com/alsa-project/alsa-gobject/pull/73 Patch URL : https://github.com/alsa-project/alsa-gobject/pull/73.patch Repository URL: https://github.com/alsa-project/alsa-gobject

1 0

[PATCH 0/4] hwdep: define interface for hwdep device
by GitHub pull_request - edited 02 Jun '22

02 Jun '22

alsa-project/alsa-gobject pull request #73 was edited from takaswie: The protocol in ALSA HwDep character device has common feature to each device. It's suitable for userspace implementation to have GObject interface for such features. This patchset adds the interface. The implementation of interface should have four methods; open, get_protocol_version, get_device_info, and create_source. Additionally, it should emit handle-disconnection signal when detecting ENODEV error in the interation of source. It requires to notify userspace application since the application should release character device immediately. ``` Takashi Sakamoto (3): hwdep: use explicit value for iface type enumeration hwdep: device-common: add interface for common feature of ALSA HwDep device hwdep: add error domain for implementators of DeviceCommon interface src/hwdep/alsahwdep-enum-types.h | 89 +++++++++------- src/hwdep/alsahwdep.h | 2 + src/hwdep/alsahwdep.map | 10 ++ src/hwdep/device-common.c | 178 +++++++++++++++++++++++++++++++ src/hwdep/device-common.h | 106 ++++++++++++++++++ src/hwdep/meson.build | 2 + tests/alsahwdep-enums | 10 ++ 7 files changed, 356 insertions(+), 41 deletions(-) create mode 100644 src/hwdep/device-common.c create mode 100644 src/hwdep/device-common.h ``` Request URL : https://github.com/alsa-project/alsa-gobject/pull/73 Patch URL : https://github.com/alsa-project/alsa-gobject/pull/73.patch Repository URL: https://github.com/alsa-project/alsa-gobject

1 0

[PATCH 0/3] hwdep: define interface for hwdep device
by GitHub pull_request - edited 02 Jun '22

02 Jun '22

alsa-project/alsa-gobject pull request #73 was edited from takaswie: The protocol in ALSA HwDep character device has common feature to each device. It's suitable for userspace implementation to have GObject interface for such features. This patchset adds the interface. The implementation of interface should have four methods; open, get_protocol_version, get_device_info, and create_source. Additionally, it should emit handle-disconnection signal when detecting ENODEV error in the interation of source. It requires to notify userspace application since the application should release character device immediately. ``` Takashi Sakamoto (3): hwdep: use explicit value for iface type enumeration hwdep: device-common: add interface for common feature of ALSA HwDep device hwdep: add error domain for implementators of DeviceCommon interface src/hwdep/alsahwdep-enum-types.h | 89 +++++++++------- src/hwdep/alsahwdep.h | 2 + src/hwdep/alsahwdep.map | 10 ++ src/hwdep/device-common.c | 178 +++++++++++++++++++++++++++++++ src/hwdep/device-common.h | 106 ++++++++++++++++++ src/hwdep/meson.build | 2 + tests/alsahwdep-enums | 10 ++ 7 files changed, 356 insertions(+), 41 deletions(-) create mode 100644 src/hwdep/device-common.c create mode 100644 src/hwdep/device-common.h ``` Request URL : https://github.com/alsa-project/alsa-gobject/pull/73 Patch URL : https://github.com/alsa-project/alsa-gobject/pull/73.patch Repository URL: https://github.com/alsa-project/alsa-gobject

1 0

[PATCH 0/2] hwdep: define interface for hwdep device
by GitHub pull_request - edited 02 Jun '22

02 Jun '22

alsa-project/alsa-gobject pull request #73 was edited from takaswie: The protocol in ALSA HwDep character device has common feature to each device. It's suitable for userspace implementation to have GObject interface for such features. This patchset adds the interface. The implementation of interface should have four methods; open, get_protocol_version, get_device_info, and create_source. Additionally, it should emit handle-disconnection signal when detecting ENODEV error in the interation of source. It requires to notify userspace application since the application should release character device immediately. ``` Takashi Sakamoto (3): hwdep: use explicit value for iface type enumeration hwdep: device-common: add interface for common feature of ALSA HwDep device hwdep: add error domain for implementators of DeviceCommon interface src/hwdep/alsahwdep-enum-types.h | 89 +++++++++------- src/hwdep/alsahwdep.h | 2 + src/hwdep/alsahwdep.map | 10 ++ src/hwdep/device-common.c | 178 +++++++++++++++++++++++++++++++ src/hwdep/device-common.h | 106 ++++++++++++++++++ src/hwdep/meson.build | 2 + tests/alsahwdep-enums | 10 ++ 7 files changed, 356 insertions(+), 41 deletions(-) create mode 100644 src/hwdep/device-common.c create mode 100644 src/hwdep/device-common.h ``` Request URL : https://github.com/alsa-project/alsa-gobject/pull/73 Patch URL : https://github.com/alsa-project/alsa-gobject/pull/73.patch Repository URL: https://github.com/alsa-project/alsa-gobject

1 0

[PATCH 0/2] hwdep: define interface for hwdep device
by GitHub pull_request - opened 02 Jun '22

02 Jun '22

alsa-project/alsa-gobject pull request #73 was opened from takaswie: The protocol in ALSA HwDep character device has common feature to each device. It's suitable for userspace implementation to have GObject interface for such features. This patchset adds the interface. The implementation of interface should have four methods; open, get_protocol_version, get_device_info, and create_source. Additionally, it should emit handle-disconnection signal when detecting ENODEV error in the interation of source. It requires to notify userspace application since the application should release character device immediately. ``` Takashi Sakamoto (2): hwdep: use explicit value for iface type enumeration hwdep: device-common: add interface for common feature of ALSA HwDep device src/hwdep/alsahwdep-enum-types.h | 67 +++++++--------- src/hwdep/alsahwdep.h | 2 + src/hwdep/alsahwdep.map | 6 ++ src/hwdep/device-common.c | 128 +++++++++++++++++++++++++++++++ src/hwdep/device-common.h | 102 ++++++++++++++++++++++++ src/hwdep/meson.build | 2 + 6 files changed, 267 insertions(+), 40 deletions(-) create mode 100644 src/hwdep/device-common.c create mode 100644 src/hwdep/device-common.h ``` Request URL : https://github.com/alsa-project/alsa-gobject/pull/73 Patch URL : https://github.com/alsa-project/alsa-gobject/pull/73.patch Repository URL: https://github.com/alsa-project/alsa-gobject

1 0

Semantics for _SX controls
by Mark Brown 02 Jun '22

02 Jun '22

Hi, The fixes for input validation have highlighed a bunch of problems with the _SX controls. These aren't widely used, it's just some Cirrus Logic and Qualcomm CODECs plus the obsolte ab8500 driver, but it has become apparent that at least some of the drivers were relying on the lack of validation we used to have to work. For the Cirrus case 34198710f55b5f ("ASoC: Add info callback for SX_TLV controls") says that the intended semantic is Currently every CODEC implementing these controls specifies the minimum as the non-zero value for the minimum and the maximum as the number of gain settings available. which was from Charles at Cirrus so hopefully that's the semantic used by Cirrus drivers. However Tan Nayir pointed at some out of tree patches from Qualcomm which change the info callback such that max is instead the maximum register value that can be written and reports that since we started validating written values properly they can no longer set some valid values in some controls on at least some Qualcomm devices (this was a wcd9340) which clearly indicates that there's some problems that need fixing. This isn't helped by the fact that snd_soc_info_volsw_sx() is implemented by calling snd_soc_info_volsw() and then applying an offset to the maximum value which makes things harder to follow - I think we should make snd_soc_info_volsw_sx() a standalone function, and we probably need some fixes in some combination of the shared functions and the drivers as well. Can people please check how the values in the current drivers correspond to what the actual bitfields in the registers have so we can get a handle on what's going on here? I've pasted a quick grep for _SX controls below: sound/soc/codecs/ab8500-codec.c: SOC_SINGLE_XR_SX("ANC Warp Delay Shift", sound/soc/codecs/ab8500-codec.c: SOC_SINGLE_XR_SX("ANC FIR Output Shift", sound/soc/codecs/ab8500-codec.c: SOC_SINGLE_XR_SX("ANC IIR Output Shift", sound/soc/codecs/ab8500-codec.c: SOC_SINGLE_XR_SX("ANC Warp Delay", sound/soc/codecs/cs35l33.c: SOC_SINGLE_SX_TLV("DAC Volume", CS35L33_DIG_VOL_CTL, sound/soc/codecs/cs35l34.c: SOC_SINGLE_SX_TLV("Digital Volume", CS35L34_AMP_DIG_VOL, sound/soc/codecs/cs35l35.c: SOC_SINGLE_SX_TLV("Digital Audio Volume", CS35L35_AMP_DIG_VOL, sound/soc/codecs/cs35l35.c: SOC_SINGLE_SX_TLV("Digital Advisory Volume", CS35L35_ADV_DIG_VOL, sound/soc/codecs/cs35l36.c: SOC_SINGLE_SX_TLV("Digital PCM Volume", CS35L36_AMP_DIG_VOL_CTRL, sound/soc/codecs/cs35l41.c: SOC_SINGLE_SX_TLV("Digital PCM Volume", CS35L41_AMP_DIG_VOL_CTRL, sound/soc/codecs/cs4265.c: SOC_DOUBLE_R_SX_TLV("PGA Volume", CS4265_CHA_PGA_CTL, sound/soc/codecs/cs42l51.c: SOC_DOUBLE_R_SX_TLV("PCM Playback Volume", sound/soc/codecs/cs42l51.c: SOC_DOUBLE_R_SX_TLV("Analog Playback Volume", sound/soc/codecs/cs42l51.c: SOC_DOUBLE_R_SX_TLV("ADC Mixer Volume", sound/soc/codecs/cs42l51.c: SOC_DOUBLE_R_SX_TLV("ADC Attenuator Volume", sound/soc/codecs/cs42l51.c: SOC_DOUBLE_R_SX_TLV("PGA Volume", sound/soc/codecs/cs42l52.c: SOC_DOUBLE_R_SX_TLV("Master Volume", CS42L52_MASTERA_VOL, sound/soc/codecs/cs42l52.c: SOC_DOUBLE_R_SX_TLV("Headphone Volume", CS42L52_HPA_VOL, sound/soc/codecs/cs42l52.c: SOC_DOUBLE_R_SX_TLV("Speaker Volume", CS42L52_SPKA_VOL, sound/soc/codecs/cs42l52.c: SOC_DOUBLE_R_SX_TLV("Bypass Volume", CS42L52_PASSTHRUA_VOL, sound/soc/codecs/cs42l52.c: SOC_DOUBLE_R_SX_TLV("ADC Volume", CS42L52_ADCA_VOL, sound/soc/codecs/cs42l52.c: SOC_DOUBLE_R_SX_TLV("ADC Mixer Volume", sound/soc/codecs/cs42l52.c: SOC_DOUBLE_R_SX_TLV("PGA Volume", CS42L52_PGAA_CTL, sound/soc/codecs/cs42l52.c: SOC_DOUBLE_R_SX_TLV("PCM Mixer Volume", sound/soc/codecs/cs42l52.c: SOC_SINGLE_SX_TLV("Beep Volume", CS42L52_BEEP_VOL, sound/soc/codecs/cs42l56.c: SOC_DOUBLE_R_SX_TLV("Master Volume", CS42L56_MASTER_A_VOLUME, sound/soc/codecs/cs42l56.c: SOC_DOUBLE_R_SX_TLV("ADC Mixer Volume", CS42L56_ADCA_MIX_VOLUME, sound/soc/codecs/cs42l56.c: SOC_DOUBLE_R_SX_TLV("PCM Mixer Volume", CS42L56_PCMA_MIX_VOLUME, sound/soc/codecs/cs42l56.c: SOC_DOUBLE_R_SX_TLV("PGA Volume", CS42L56_PGAA_MUX_VOLUME, sound/soc/codecs/cs42l56.c: SOC_DOUBLE_R_SX_TLV("Headphone Volume", CS42L56_HPA_VOLUME, sound/soc/codecs/cs42l56.c: SOC_DOUBLE_R_SX_TLV("LineOut Volume", CS42L56_LOA_VOLUME, sound/soc/codecs/cs42l56.c: SOC_SINGLE_SX_TLV("Beep Volume", CS42L56_BEEP_FREQ_OFFTIME, sound/soc/codecs/cs42l73.c: SOC_DOUBLE_R_SX_TLV("Headphone Analog Playback Volume", sound/soc/codecs/cs42l73.c: SOC_DOUBLE_R_SX_TLV("LineOut Analog Playback Volume", CS42L73_LOAAVOL, sound/soc/codecs/cs42l73.c: SOC_DOUBLE_R_SX_TLV("Input PGA Analog Volume", CS42L73_MICAPREPGAAVOL, sound/soc/codecs/cs42l73.c: SOC_DOUBLE_R_SX_TLV("Input Path Digital Volume", CS42L73_IPADVOL, sound/soc/codecs/cs42l73.c: SOC_DOUBLE_R_SX_TLV("HL Digital Playback Volume", sound/soc/codecs/cs42l73.c: SOC_SINGLE_SX_TLV("Speakerphone Digital Volume", sound/soc/codecs/cs42l73.c: SOC_SINGLE_SX_TLV("Ear Speaker Digital Volume", sound/soc/codecs/cs53l30.c: SOC_SINGLE_SX_TLV("ADC1A PGA Volume", sound/soc/codecs/cs53l30.c: SOC_SINGLE_SX_TLV("ADC1B PGA Volume", sound/soc/codecs/cs53l30.c: SOC_SINGLE_SX_TLV("ADC2A PGA Volume", sound/soc/codecs/cs53l30.c: SOC_SINGLE_SX_TLV("ADC2B PGA Volume", sound/soc/codecs/cs53l30.c: SOC_SINGLE_SX_TLV("ADC1A Digital Volume", sound/soc/codecs/cs53l30.c: SOC_SINGLE_SX_TLV("ADC1B Digital Volume", sound/soc/codecs/cs53l30.c: SOC_SINGLE_SX_TLV("ADC2A Digital Volume", sound/soc/codecs/cs53l30.c: SOC_SINGLE_SX_TLV("ADC2B Digital Volume", sound/soc/codecs/msm8916-wcd-digital.c: SOC_SINGLE_SX_TLV("IIR1 INP1 Volume", LPASS_CDC_IIR1_GAIN_B1_CTL, sound/soc/codecs/msm8916-wcd-digital.c: SOC_SINGLE_SX_TLV("IIR1 INP2 Volume", LPASS_CDC_IIR1_GAIN_B2_CTL, sound/soc/codecs/msm8916-wcd-digital.c: SOC_SINGLE_SX_TLV("IIR1 INP3 Volume", LPASS_CDC_IIR1_GAIN_B3_CTL, sound/soc/codecs/msm8916-wcd-digital.c: SOC_SINGLE_SX_TLV("IIR1 INP4 Volume", LPASS_CDC_IIR1_GAIN_B4_CTL, sound/soc/codecs/msm8916-wcd-digital.c: SOC_SINGLE_SX_TLV("IIR2 INP1 Volume", LPASS_CDC_IIR2_GAIN_B1_CTL, sound/soc/codecs/msm8916-wcd-digital.c: SOC_SINGLE_SX_TLV("IIR2 INP2 Volume", LPASS_CDC_IIR2_GAIN_B2_CTL, sound/soc/codecs/msm8916-wcd-digital.c: SOC_SINGLE_SX_TLV("IIR2 INP3 Volume", LPASS_CDC_IIR2_GAIN_B3_CTL, sound/soc/codecs/msm8916-wcd-digital.c: SOC_SINGLE_SX_TLV("IIR2 INP4 Volume", LPASS_CDC_IIR2_GAIN_B4_CTL, sound/soc/codecs/wcd9335.c: SOC_SINGLE_SX_TLV("RX0 Digital Volume", WCD9335_CDC_RX0_RX_VOL_CTL, sound/soc/codecs/wcd9335.c: SOC_SINGLE_SX_TLV("RX1 Digital Volume", WCD9335_CDC_RX1_RX_VOL_CTL, sound/soc/codecs/wcd9335.c: SOC_SINGLE_SX_TLV("RX2 Digital Volume", WCD9335_CDC_RX2_RX_VOL_CTL, sound/soc/codecs/wcd9335.c: SOC_SINGLE_SX_TLV("RX3 Digital Volume", WCD9335_CDC_RX3_RX_VOL_CTL, sound/soc/codecs/wcd9335.c: SOC_SINGLE_SX_TLV("RX4 Digital Volume", WCD9335_CDC_RX4_RX_VOL_CTL, sound/soc/codecs/wcd9335.c: SOC_SINGLE_SX_TLV("RX5 Digital Volume", WCD9335_CDC_RX5_RX_VOL_CTL, sound/soc/codecs/wcd9335.c: SOC_SINGLE_SX_TLV("RX6 Digital Volume", WCD9335_CDC_RX6_RX_VOL_CTL, sound/soc/codecs/wcd9335.c: SOC_SINGLE_SX_TLV("RX7 Digital Volume", WCD9335_CDC_RX7_RX_VOL_CTL, sound/soc/codecs/wcd9335.c: SOC_SINGLE_SX_TLV("RX8 Digital Volume", WCD9335_CDC_RX8_RX_VOL_CTL, sound/soc/codecs/wcd9335.c: SOC_SINGLE_SX_TLV("RX0 Mix Digital Volume", sound/soc/codecs/wcd9335.c: SOC_SINGLE_SX_TLV("RX1 Mix Digital Volume", sound/soc/codecs/wcd9335.c: SOC_SINGLE_SX_TLV("RX2 Mix Digital Volume", sound/soc/codecs/wcd9335.c: SOC_SINGLE_SX_TLV("RX3 Mix Digital Volume", sound/soc/codecs/wcd9335.c: SOC_SINGLE_SX_TLV("RX4 Mix Digital Volume", sound/soc/codecs/wcd9335.c: SOC_SINGLE_SX_TLV("RX5 Mix Digital Volume", sound/soc/codecs/wcd9335.c: SOC_SINGLE_SX_TLV("RX6 Mix Digital Volume", sound/soc/codecs/wcd9335.c: SOC_SINGLE_SX_TLV("RX7 Mix Digital Volume", sound/soc/codecs/wcd9335.c: SOC_SINGLE_SX_TLV("RX8 Mix Digital Volume", Thanks, Mark

3 4

"topology: nhlt plugin: second attempt to fix the .so linking" cause build static library failed
by GitHub issues - opened 02 Jun '22

02 Jun '22

alsa-project/alsa-utils issue #152 was opened from youling257: ./configure --prefix=/system/usr --with-alsa-inc-prefix=/system/usr/include --with-alsa-prefix=/system/usr/lib --enable-static=yes --enable-shared=no make LDFLAGS="--static" -j8 Revert "topology: nhlt plugin: second attempt to fix the .so linking" can build static library. Issue URL : https://github.com/alsa-project/alsa-utils/issues/152 Repository URL: https://github.com/alsa-project/alsa-utils

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[PATCH v3] ASoC: fsl_xcvr:Fix unbalanced pm_runtime_enable in fsl_xcvr_probe
by zhangqilong 02 Jun '22

02 Jun '22

a) Add missing pm_runtime_disable() when probe error out. It could avoid pm_runtime implementation complains when removing and probing again the driver. b) Add remove for missing pm_runtime_disable(). v2: --- Add remove to put PM usage counter. v3: --- Modify commit message. Fix:c590fa80b3928 ("ASoC: fsl_xcvr: register platform component before registering cpu dai") Signed-off-by: Zhang Qilong <zhangqilong3(a)huawei.com> --- sound/soc/fsl/fsl_xcvr.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/soc/fsl/fsl_xcvr.c b/sound/soc/fsl/fsl_xcvr.c index d0556c79fdb1..55e640cba87d 100644 --- a/sound/soc/fsl/fsl_xcvr.c +++ b/sound/soc/fsl/fsl_xcvr.c @@ -1228,6 +1228,7 @@ static int fsl_xcvr_probe(struct platform_device *pdev) */ ret = devm_snd_dmaengine_pcm_register(dev, NULL, 0); if (ret) { + pm_runtime_disable(dev); dev_err(dev, "failed to pcm register\n"); return ret; } @@ -1235,6 +1236,7 @@ static int fsl_xcvr_probe(struct platform_device *pdev) ret = devm_snd_soc_register_component(dev, &fsl_xcvr_comp, &fsl_xcvr_dai, 1); if (ret) { + pm_runtime_disable(dev); dev_err(dev, "failed to register component %s\n", fsl_xcvr_comp.name); } @@ -1242,6 +1244,12 @@ static int fsl_xcvr_probe(struct platform_device *pdev) return ret; } +static int fsl_xcvr_remove(struct platform_device *pdev) +{ + pm_runtime_disable(&pdev->dev); + return 0; +} + static __maybe_unused int fsl_xcvr_runtime_suspend(struct device *dev) { struct fsl_xcvr *xcvr = dev_get_drvdata(dev); @@ -1370,6 +1378,7 @@ static struct platform_driver fsl_xcvr_driver = { .pm = &fsl_xcvr_pm_ops, .of_match_table = fsl_xcvr_dt_ids, }, + .remove = fsl_xcvr_remove, }; module_platform_driver(fsl_xcvr_driver); -- 2.31.1

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[PATCH V2] ASoC: fsl_xcvr:Fix unbalanced pm_runtime_enable in fsl_xcvr_probe
by zhangqilong 02 Jun '22

02 Jun '22

a) Add missing pm_runtime_disable() when probe error out. It could avoid pm_runtime implementation complains when removing and probing again the driver. b) Add remove for missing pm_runtime_disable(). v2: add remove to put PM usage counter. Fix:c590fa80b3928 ("ASoC: fsl_xcvr: register platform component before registering cpu dai") Signed-off-by: Zhang Qilong <zhangqilong3(a)huawei.com> --- sound/soc/fsl/fsl_xcvr.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/soc/fsl/fsl_xcvr.c b/sound/soc/fsl/fsl_xcvr.c index d0556c79fdb1..55e640cba87d 100644 --- a/sound/soc/fsl/fsl_xcvr.c +++ b/sound/soc/fsl/fsl_xcvr.c @@ -1228,6 +1228,7 @@ static int fsl_xcvr_probe(struct platform_device *pdev) */ ret = devm_snd_dmaengine_pcm_register(dev, NULL, 0); if (ret) { + pm_runtime_disable(dev); dev_err(dev, "failed to pcm register\n"); return ret; } @@ -1235,6 +1236,7 @@ static int fsl_xcvr_probe(struct platform_device *pdev) ret = devm_snd_soc_register_component(dev, &fsl_xcvr_comp, &fsl_xcvr_dai, 1); if (ret) { + pm_runtime_disable(dev); dev_err(dev, "failed to register component %s\n", fsl_xcvr_comp.name); } @@ -1242,6 +1244,12 @@ static int fsl_xcvr_probe(struct platform_device *pdev) return ret; } +static int fsl_xcvr_remove(struct platform_device *pdev) +{ + pm_runtime_disable(&pdev->dev); + return 0; +} + static __maybe_unused int fsl_xcvr_runtime_suspend(struct device *dev) { struct fsl_xcvr *xcvr = dev_get_drvdata(dev); @@ -1370,6 +1378,7 @@ static struct platform_driver fsl_xcvr_driver = { .pm = &fsl_xcvr_pm_ops, .of_match_table = fsl_xcvr_dt_ids, }, + .remove = fsl_xcvr_remove, }; module_platform_driver(fsl_xcvr_driver); -- 2.31.1

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