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[PATCH v2 3/3] ASoC: rt715: modification for code simplicity
by Jack Yu 16 Apr '21

16 Apr '21

Modification for code simplicity. Signed-off-by: Jack Yu <jack.yu(a)realtek.com> --- sound/soc/codecs/rt715.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/rt715.c b/sound/soc/codecs/rt715.c index b33eabb9f740..1352869cc086 100644 --- a/sound/soc/codecs/rt715.c +++ b/sound/soc/codecs/rt715.c @@ -57,14 +57,14 @@ static void rt715_get_gain(struct rt715_priv *rt715, unsigned int addr_h, { int ret; /* R Channel */ - *r_val = (val_h << 8); + *r_val = val_h << 8; ret = regmap_read(rt715->regmap, addr_l, r_val); if (ret < 0) pr_err("Failed to get R channel gain.\n"); /* L Channel */ val_h |= 0x20; - *l_val = (val_h << 8); + *l_val = val_h << 8; ret = regmap_read(rt715->regmap, addr_h, l_val); if (ret < 0) pr_err("Failed to get L channel gain.\n"); @@ -183,8 +183,8 @@ static int rt715_set_amp_gain_get(struct snd_kcontrol *kcontrol, if (mc->invert) { /* for mute status */ - read_ll = !((read_ll & 0x80) >> RT715_MUTE_SFT); - read_rl = !((read_rl & 0x80) >> RT715_MUTE_SFT); + read_ll = !(read_ll & 0x80); + read_rl = !(read_rl & 0x80); } else { /* for gain */ read_ll = read_ll & 0x7f; -- 2.29.0

2 1

[PATCH v2 1/3] ASoC: rt715: add main capture switch and main capture volume
by Jack Yu 16 Apr '21

16 Apr '21

Add main capture switch and main capture volume control. Main capture control has its own channel value respectivelly. Signed-off-by: Jack Yu <jack.yu(a)realtek.com> --- sound/soc/codecs/rt715.c | 256 +++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/rt715.h | 2 + 2 files changed, 258 insertions(+) diff --git a/sound/soc/codecs/rt715.c b/sound/soc/codecs/rt715.c index f419d001a3fb..3cb8ab452bf1 100644 --- a/sound/soc/codecs/rt715.c +++ b/sound/soc/codecs/rt715.c @@ -201,9 +201,246 @@ static int rt715_set_amp_gain_get(struct snd_kcontrol *kcontrol, return 0; } +static int rt715_set_main_switch_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct snd_soc_dapm_context *dapm = + snd_soc_component_get_dapm(component); + struct rt715_priv *rt715 = snd_soc_component_get_drvdata(component); + unsigned int capture_reg_H[] = {RT715_SET_GAIN_MIC_ADC_H, + RT715_SET_GAIN_LINE_ADC_H, RT715_SET_GAIN_MIX_ADC_H, + RT715_SET_GAIN_MIX_ADC2_H}; + unsigned int capture_reg_L[] = {RT715_SET_GAIN_MIC_ADC_L, + RT715_SET_GAIN_LINE_ADC_L, RT715_SET_GAIN_MIX_ADC_L, + RT715_SET_GAIN_MIX_ADC2_L}; + unsigned int addr_h, addr_l, val_h = 0x0, val_ll, val_lr; + unsigned int k_shift = RT715_DIR_IN_SFT, k_changed = 0; + unsigned int read_ll, read_rl, i, j, loop_cnt = 4; + + for (i = 0; i < 8; i++) { + if (ucontrol->value.integer.value[i] != rt715->kctl_8ch_switch_ori[i]) + k_changed = 1; + } + + for (j = 0; j < loop_cnt; j++) { + /* Can't use update bit function, so read the original value first */ + addr_h = capture_reg_H[j]; + addr_l = capture_reg_L[j]; + rt715_get_gain(rt715, addr_h, addr_l, val_h, &read_rl, &read_ll); + + if (dapm->bias_level <= SND_SOC_BIAS_STANDBY) + regmap_write(rt715->regmap, + RT715_SET_AUDIO_POWER_STATE, AC_PWRST_D0); + + /* L Channel */ + /* for mute */ + rt715->kctl_8ch_switch_ori[j * 2] = + ucontrol->value.integer.value[j * 2]; + val_ll = (!ucontrol->value.integer.value[j * 2]) << 7; + /* keep gain */ + val_ll |= read_ll & 0x7f; + + /* R Channel */ + /* for mute */ + rt715->kctl_8ch_switch_ori[j * 2 + 1] = + ucontrol->value.integer.value[j * 2 + 1]; + val_lr = (!ucontrol->value.integer.value[j * 2 + 1]) << 7; + /* keep gain */ + val_lr |= read_rl & 0x7f; + + for (i = 0; i < 3; i++) { /* retry 3 times at most */ + + if (val_ll == val_lr) { + /* Set both L/R channels at the same time */ + val_h = (1 << k_shift) | (3 << 4); + regmap_write(rt715->regmap, addr_h, + (val_h << 8) | val_ll); + regmap_write(rt715->regmap, addr_l, + (val_h << 8) | val_ll); + } else { + /* Lch*/ + val_h = (1 << k_shift) | (1 << 5); + regmap_write(rt715->regmap, addr_h, + (val_h << 8) | val_ll); + /* Rch */ + val_h = (1 << k_shift) | (1 << 4); + regmap_write(rt715->regmap, addr_l, + (val_h << 8) | val_lr); + } + val_h = 0x0; + rt715_get_gain(rt715, addr_h, addr_l, val_h, + &read_rl, &read_ll); + if (read_rl == val_lr && read_ll == val_ll) + break; + } + } + + /* D0:power on state, D3: power saving mode */ + if (dapm->bias_level <= SND_SOC_BIAS_STANDBY) + regmap_write(rt715->regmap, + RT715_SET_AUDIO_POWER_STATE, AC_PWRST_D3); + return k_changed; +} + +static int rt715_set_main_switch_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct rt715_priv *rt715 = snd_soc_component_get_drvdata(component); + unsigned int capture_reg_H[] = {RT715_SET_GAIN_MIC_ADC_H, + RT715_SET_GAIN_LINE_ADC_H, RT715_SET_GAIN_MIX_ADC_H, + RT715_SET_GAIN_MIX_ADC2_H}; + unsigned int capture_reg_L[] = {RT715_SET_GAIN_MIC_ADC_L, + RT715_SET_GAIN_LINE_ADC_L, RT715_SET_GAIN_MIX_ADC_L, + RT715_SET_GAIN_MIX_ADC2_L}; + unsigned int addr_h, addr_l, val_h = 0x0, i, loop_cnt = 4; + unsigned int read_ll, read_rl; + + for (i = 0; i < loop_cnt; i++) { + addr_h = capture_reg_H[i]; + addr_l = capture_reg_L[i]; + rt715_get_gain(rt715, addr_h, addr_l, val_h, &read_rl, &read_ll); + + ucontrol->value.integer.value[i * 2] = !(read_ll & 0x80); + ucontrol->value.integer.value[i * 2 + 1] = !(read_rl & 0x80); + } + + return 0; +} + +static int rt715_set_main_vol_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct snd_soc_dapm_context *dapm = + snd_soc_component_get_dapm(component); + struct rt715_priv *rt715 = snd_soc_component_get_drvdata(component); + unsigned int capture_reg_H[] = {RT715_SET_GAIN_MIC_ADC_H, + RT715_SET_GAIN_LINE_ADC_H, RT715_SET_GAIN_MIX_ADC_H, + RT715_SET_GAIN_MIX_ADC2_H}; + unsigned int capture_reg_L[] = {RT715_SET_GAIN_MIC_ADC_L, + RT715_SET_GAIN_LINE_ADC_L, RT715_SET_GAIN_MIX_ADC_L, + RT715_SET_GAIN_MIX_ADC2_L}; + unsigned int addr_h, addr_l, val_h = 0x0, val_ll, val_lr; + unsigned int read_ll, read_rl, i, j, loop_cnt = 4, k_changed = 0; + unsigned int k_shift = RT715_DIR_IN_SFT, k_max = 0x3f; + + for (i = 0; i < 8; i++) { + if (ucontrol->value.integer.value[i] != rt715->kctl_8ch_vol_ori[i]) + k_changed = 1; + } + + for (j = 0; j < loop_cnt; j++) { + addr_h = capture_reg_H[j]; + addr_l = capture_reg_L[j]; + rt715_get_gain(rt715, addr_h, addr_l, val_h, &read_rl, &read_ll); + + if (dapm->bias_level <= SND_SOC_BIAS_STANDBY) + regmap_write(rt715->regmap, + RT715_SET_AUDIO_POWER_STATE, AC_PWRST_D0); + + /* L Channel */ + /* for gain */ + rt715->kctl_8ch_vol_ori[j * 2] = ucontrol->value.integer.value[j * 2]; + val_ll = ((ucontrol->value.integer.value[j * 2]) & 0x7f); + if (val_ll > k_max) + val_ll = k_max; + /* keep mute status */ + val_ll |= read_ll & 0x80; + + /* R Channel */ + /* for gain */ + rt715->kctl_8ch_vol_ori[j * 2 + 1] = + ucontrol->value.integer.value[j * 2 + 1]; + val_lr = ((ucontrol->value.integer.value[j * 2 + 1]) & 0x7f); + if (val_lr > k_max) + val_lr = k_max; + /* keep mute status */ + val_lr |= read_rl & 0x80; + + for (i = 0; i < 3; i++) { /* retry 3 times at most */ + if (val_ll == val_lr) { + /* Set both L/R channels at the same time */ + val_h = (1 << k_shift) | (3 << 4); + regmap_write(rt715->regmap, addr_h, + (val_h << 8) | val_ll); + regmap_write(rt715->regmap, addr_l, + (val_h << 8) | val_ll); + } else { + /* Lch*/ + val_h = (1 << k_shift) | (1 << 5); + regmap_write(rt715->regmap, addr_h, + (val_h << 8) | val_ll); + /* Rch */ + val_h = (1 << k_shift) | (1 << 4); + regmap_write(rt715->regmap, addr_l, + (val_h << 8) | val_lr); + } + val_h = 0x0; + rt715_get_gain(rt715, addr_h, addr_l, val_h, + &read_rl, &read_ll); + if (read_rl == val_lr && read_ll == val_ll) + break; + } + } + + /* D0:power on state, D3: power saving mode */ + if (dapm->bias_level <= SND_SOC_BIAS_STANDBY) + regmap_write(rt715->regmap, + RT715_SET_AUDIO_POWER_STATE, AC_PWRST_D3); + return k_changed; +} + +static int rt715_set_main_vol_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct rt715_priv *rt715 = snd_soc_component_get_drvdata(component); + unsigned int capture_reg_H[] = {RT715_SET_GAIN_MIC_ADC_H, + RT715_SET_GAIN_LINE_ADC_H, RT715_SET_GAIN_MIX_ADC_H, + RT715_SET_GAIN_MIX_ADC2_H}; + unsigned int capture_reg_L[] = {RT715_SET_GAIN_MIC_ADC_L, + RT715_SET_GAIN_LINE_ADC_L, RT715_SET_GAIN_MIX_ADC_L, + RT715_SET_GAIN_MIX_ADC2_L}; + unsigned int addr_h, addr_l, val_h = 0x0, i, loop_cnt = 4; + unsigned int read_ll, read_rl; + + for (i = 0; i < loop_cnt; i++) { + addr_h = capture_reg_H[i]; + addr_l = capture_reg_L[i]; + rt715_get_gain(rt715, addr_h, addr_l, val_h, &read_rl, &read_ll); + + ucontrol->value.integer.value[i * 2] = read_ll & 0x7f; + ucontrol->value.integer.value[i * 2 + 1] = read_rl & 0x7f; + } + + return 0; +} + static const DECLARE_TLV_DB_SCALE(in_vol_tlv, -1725, 75, 0); static const DECLARE_TLV_DB_SCALE(mic_vol_tlv, 0, 1000, 0); +static int rt715_switch_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; + uinfo->count = 8; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 1; + return 0; +} + +static int rt715_vol_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 8; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 0x3f; + return 0; +} + #define SOC_DOUBLE_R_EXT(xname, reg_left, reg_right, xshift, xmax, xinvert,\ xhandler_get, xhandler_put) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ @@ -212,6 +449,21 @@ static const DECLARE_TLV_DB_SCALE(mic_vol_tlv, 0, 1000, 0); .private_value = SOC_DOUBLE_R_VALUE(reg_left, reg_right, xshift, \ xmax, xinvert) } +#define RT715_MAIN_SWITCH_EXT(xname, xhandler_get, xhandler_put) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ + .info = rt715_switch_info, \ + .get = xhandler_get, .put = xhandler_put, \ +} + +#define RT715_MAIN_VOL_EXT_TLV(xname, xhandler_get, xhandler_put, tlv_array) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ + .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | \ + SNDRV_CTL_ELEM_ACCESS_READWRITE, \ + .tlv.p = (tlv_array), \ + .info = rt715_vol_info, \ + .get = xhandler_get, .put = xhandler_put, \ +} + static const struct snd_kcontrol_new rt715_snd_controls[] = { /* Capture switch */ SOC_DOUBLE_R_EXT("ADC 07 Capture Switch", RT715_SET_GAIN_MIC_ADC_H, @@ -226,6 +478,8 @@ static const struct snd_kcontrol_new rt715_snd_controls[] = { SOC_DOUBLE_R_EXT("ADC 27 Capture Switch", RT715_SET_GAIN_MIX_ADC2_H, RT715_SET_GAIN_MIX_ADC2_L, RT715_DIR_IN_SFT, 1, 1, rt715_set_amp_gain_get, rt715_set_amp_gain_put), + RT715_MAIN_SWITCH_EXT("Capture Switch", + rt715_set_main_switch_get, rt715_set_main_switch_put), /* Volume Control */ SOC_DOUBLE_R_EXT_TLV("ADC 07 Capture Volume", RT715_SET_GAIN_MIC_ADC_H, RT715_SET_GAIN_MIC_ADC_L, RT715_DIR_IN_SFT, 0x3f, 0, @@ -243,6 +497,8 @@ static const struct snd_kcontrol_new rt715_snd_controls[] = { RT715_SET_GAIN_MIX_ADC2_L, RT715_DIR_IN_SFT, 0x3f, 0, rt715_set_amp_gain_get, rt715_set_amp_gain_put, in_vol_tlv), + RT715_MAIN_VOL_EXT_TLV("Capture Volume", + rt715_set_main_vol_get, rt715_set_main_vol_put, in_vol_tlv), /* MIC Boost Control */ SOC_DOUBLE_R_EXT_TLV("DMIC1 Boost", RT715_SET_GAIN_DMIC1_H, RT715_SET_GAIN_DMIC1_L, RT715_DIR_IN_SFT, 3, 0, diff --git a/sound/soc/codecs/rt715.h b/sound/soc/codecs/rt715.h index 009a8266f606..72c320bd35cb 100644 --- a/sound/soc/codecs/rt715.h +++ b/sound/soc/codecs/rt715.h @@ -22,6 +22,8 @@ struct rt715_priv { struct sdw_bus_params params; bool hw_init; bool first_hw_init; + unsigned int kctl_8ch_switch_ori[8]; + unsigned int kctl_8ch_vol_ori[8]; }; struct sdw_stream_data { -- 2.29.0

2 1

[PATCH v2 2/3] ASoC: rt715: remove kcontrols which no longer be used
by Jack Yu 16 Apr '21

16 Apr '21

Using new kcontrols "Capture Switch" and "Capture Volume" instead, remove kcontrols which no longer be used. Signed-off-by: Jack Yu <jack.yu(a)realtek.com> --- sound/soc/codecs/rt715.c | 103 +++++++++++++-------------------------- sound/soc/codecs/rt715.h | 1 + 2 files changed, 36 insertions(+), 68 deletions(-) diff --git a/sound/soc/codecs/rt715.c b/sound/soc/codecs/rt715.c index 3cb8ab452bf1..b33eabb9f740 100644 --- a/sound/soc/codecs/rt715.c +++ b/sound/soc/codecs/rt715.c @@ -81,12 +81,20 @@ static int rt715_set_amp_gain_put(struct snd_kcontrol *kcontrol, (struct soc_mixer_control *)kcontrol->private_value; struct rt715_priv *rt715 = snd_soc_component_get_drvdata(component); unsigned int addr_h, addr_l, val_h, val_ll, val_lr; - unsigned int read_ll, read_rl; - int i; + unsigned int read_ll, read_rl, i; + unsigned int k_vol_changed = 0; + + for (i = 0; i < 2; i++) { + if (ucontrol->value.integer.value[i] != rt715->kctl_2ch_vol_ori[i]) { + k_vol_changed = 1; + break; + } + } /* Can't use update bit function, so read the original value first */ addr_h = mc->reg; addr_l = mc->rreg; + if (mc->shift == RT715_DIR_OUT_SFT) /* output */ val_h = 0x80; else /* input */ @@ -94,41 +102,27 @@ static int rt715_set_amp_gain_put(struct snd_kcontrol *kcontrol, rt715_get_gain(rt715, addr_h, addr_l, val_h, &read_rl, &read_ll); + if (dapm->bias_level <= SND_SOC_BIAS_STANDBY) + regmap_write(rt715->regmap, + RT715_SET_AUDIO_POWER_STATE, AC_PWRST_D0); + /* L Channel */ - if (mc->invert) { - /* for mute */ - val_ll = (mc->max - ucontrol->value.integer.value[0]) << 7; - /* keep gain */ - read_ll = read_ll & 0x7f; - val_ll |= read_ll; - } else { - /* for gain */ - val_ll = ((ucontrol->value.integer.value[0]) & 0x7f); - if (val_ll > mc->max) - val_ll = mc->max; - /* keep mute status */ - read_ll = read_ll & 0x80; - val_ll |= read_ll; - } + rt715->kctl_2ch_vol_ori[0] = ucontrol->value.integer.value[0]; + /* for gain */ + val_ll = ((ucontrol->value.integer.value[0]) & 0x7f); + if (val_ll > mc->max) + val_ll = mc->max; + /* keep mute status */ + val_ll |= read_ll & 0x80; /* R Channel */ - if (mc->invert) { - regmap_write(rt715->regmap, - RT715_SET_AUDIO_POWER_STATE, AC_PWRST_D0); - /* for mute */ - val_lr = (mc->max - ucontrol->value.integer.value[1]) << 7; - /* keep gain */ - read_rl = read_rl & 0x7f; - val_lr |= read_rl; - } else { - /* for gain */ - val_lr = ((ucontrol->value.integer.value[1]) & 0x7f); - if (val_lr > mc->max) - val_lr = mc->max; - /* keep mute status */ - read_rl = read_rl & 0x80; - val_lr |= read_rl; - } + rt715->kctl_2ch_vol_ori[1] = ucontrol->value.integer.value[1]; + /* for gain */ + val_lr = ((ucontrol->value.integer.value[1]) & 0x7f); + if (val_lr > mc->max) + val_lr = mc->max; + /* keep mute status */ + val_lr |= read_rl & 0x80; for (i = 0; i < 3; i++) { /* retry 3 times at most */ @@ -136,18 +130,18 @@ static int rt715_set_amp_gain_put(struct snd_kcontrol *kcontrol, /* Set both L/R channels at the same time */ val_h = (1 << mc->shift) | (3 << 4); regmap_write(rt715->regmap, addr_h, - (val_h << 8 | val_ll)); + (val_h << 8) | val_ll); regmap_write(rt715->regmap, addr_l, - (val_h << 8 | val_ll)); + (val_h << 8) | val_ll); } else { /* Lch*/ val_h = (1 << mc->shift) | (1 << 5); regmap_write(rt715->regmap, addr_h, - (val_h << 8 | val_ll)); + (val_h << 8) | val_ll); /* Rch */ val_h = (1 << mc->shift) | (1 << 4); regmap_write(rt715->regmap, addr_l, - (val_h << 8 | val_lr)); + (val_h << 8) | val_lr); } /* check result */ if (mc->shift == RT715_DIR_OUT_SFT) /* output */ @@ -156,15 +150,16 @@ static int rt715_set_amp_gain_put(struct snd_kcontrol *kcontrol, val_h = 0x0; rt715_get_gain(rt715, addr_h, addr_l, val_h, - &read_rl, &read_ll); + &read_rl, &read_ll); if (read_rl == val_lr && read_ll == val_ll) break; } + /* D0:power on state, D3: power saving mode */ if (dapm->bias_level <= SND_SOC_BIAS_STANDBY) regmap_write(rt715->regmap, RT715_SET_AUDIO_POWER_STATE, AC_PWRST_D3); - return 0; + return k_vol_changed; } static int rt715_set_amp_gain_get(struct snd_kcontrol *kcontrol, @@ -466,37 +461,9 @@ static int rt715_vol_info(struct snd_kcontrol *kcontrol, static const struct snd_kcontrol_new rt715_snd_controls[] = { /* Capture switch */ - SOC_DOUBLE_R_EXT("ADC 07 Capture Switch", RT715_SET_GAIN_MIC_ADC_H, - RT715_SET_GAIN_MIC_ADC_L, RT715_DIR_IN_SFT, 1, 1, - rt715_set_amp_gain_get, rt715_set_amp_gain_put), - SOC_DOUBLE_R_EXT("ADC 08 Capture Switch", RT715_SET_GAIN_LINE_ADC_H, - RT715_SET_GAIN_LINE_ADC_L, RT715_DIR_IN_SFT, 1, 1, - rt715_set_amp_gain_get, rt715_set_amp_gain_put), - SOC_DOUBLE_R_EXT("ADC 09 Capture Switch", RT715_SET_GAIN_MIX_ADC_H, - RT715_SET_GAIN_MIX_ADC_L, RT715_DIR_IN_SFT, 1, 1, - rt715_set_amp_gain_get, rt715_set_amp_gain_put), - SOC_DOUBLE_R_EXT("ADC 27 Capture Switch", RT715_SET_GAIN_MIX_ADC2_H, - RT715_SET_GAIN_MIX_ADC2_L, RT715_DIR_IN_SFT, 1, 1, - rt715_set_amp_gain_get, rt715_set_amp_gain_put), RT715_MAIN_SWITCH_EXT("Capture Switch", rt715_set_main_switch_get, rt715_set_main_switch_put), /* Volume Control */ - SOC_DOUBLE_R_EXT_TLV("ADC 07 Capture Volume", RT715_SET_GAIN_MIC_ADC_H, - RT715_SET_GAIN_MIC_ADC_L, RT715_DIR_IN_SFT, 0x3f, 0, - rt715_set_amp_gain_get, rt715_set_amp_gain_put, - in_vol_tlv), - SOC_DOUBLE_R_EXT_TLV("ADC 08 Capture Volume", RT715_SET_GAIN_LINE_ADC_H, - RT715_SET_GAIN_LINE_ADC_L, RT715_DIR_IN_SFT, 0x3f, 0, - rt715_set_amp_gain_get, rt715_set_amp_gain_put, - in_vol_tlv), - SOC_DOUBLE_R_EXT_TLV("ADC 09 Capture Volume", RT715_SET_GAIN_MIX_ADC_H, - RT715_SET_GAIN_MIX_ADC_L, RT715_DIR_IN_SFT, 0x3f, 0, - rt715_set_amp_gain_get, rt715_set_amp_gain_put, - in_vol_tlv), - SOC_DOUBLE_R_EXT_TLV("ADC 27 Capture Volume", RT715_SET_GAIN_MIX_ADC2_H, - RT715_SET_GAIN_MIX_ADC2_L, RT715_DIR_IN_SFT, 0x3f, 0, - rt715_set_amp_gain_get, rt715_set_amp_gain_put, - in_vol_tlv), RT715_MAIN_VOL_EXT_TLV("Capture Volume", rt715_set_main_vol_get, rt715_set_main_vol_put, in_vol_tlv), /* MIC Boost Control */ diff --git a/sound/soc/codecs/rt715.h b/sound/soc/codecs/rt715.h index 72c320bd35cb..25dba61f1760 100644 --- a/sound/soc/codecs/rt715.h +++ b/sound/soc/codecs/rt715.h @@ -22,6 +22,7 @@ struct rt715_priv { struct sdw_bus_params params; bool hw_init; bool first_hw_init; + unsigned int kctl_2ch_vol_ori[2]; unsigned int kctl_8ch_switch_ori[8]; unsigned int kctl_8ch_vol_ori[8]; }; -- 2.29.0

4 9

[PATCH] Revert "ASoC: mt6359: add ASoC mt6359 ASoC accdet jack document"
by Rob Herring 16 Apr '21

16 Apr '21

This reverts commit e61c589587c772c5f672b22683c3e0b38be20702. The binding document has lots of schema errors and there's been no effort to fix them, so let's remove it. Cc: Argus Lin <argus.lin(a)mediatek.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Matthias Brugger <matthias.bgg(a)gmail.com> Cc: alsa-devel(a)alsa-project.org Cc: linux-arm-kernel(a)lists.infradead.org Cc: linux-mediatek(a)lists.infradead.org Signed-off-by: Rob Herring <robh(a)kernel.org> --- .../bindings/sound/mt6359-accdet.yaml | 164 ------------------ 1 file changed, 164 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/mt6359-accdet.yaml diff --git a/Documentation/devicetree/bindings/sound/mt6359-accdet.yaml b/Documentation/devicetree/bindings/sound/mt6359-accdet.yaml deleted file mode 100644 index 7fb3e8dfb4c5..000000000000 --- a/Documentation/devicetree/bindings/sound/mt6359-accdet.yaml +++ /dev/null @@ -1,164 +0,0 @@ -# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) -%YAML 1.2 ---- -$id: http://devicetree.org/schemas/sound/mt6359-accdet.yaml# -$schema: http://devicetree.org/meta-schemas/core.yaml# - -title: Mediatek Technologies Inc. MT6359 ASoC accdet jack driver bindings - -maintainers: - - Chipeng Chang <chipeng.chang(a)mediatek.com> - -description: | - This binding describes the Mediatek Technologies MT6359 ASoC - accdet jack driver. - -properties: - mediatek,accdet-name: - maxItems: 1 - description: named as "mt6359-accdet" jack. - - mediatek,mic-vol: - maxItems: 1 - description: | - accdet micbias1 voltage. - - enum: - - 0x00 # 1.7v - micbias1 volage is 1.7v. - - 0x01 # 1.8v - micbias1 volage is 1.8v. - - 0x02 # 1.9v - micbias1 volage is 1.9v. - - 0x03 # 2.0v - micbias1 volage is 2.0v. - - 0x04 # 2.1v - micbias1 volage is 2.1v. - - 0x05 # 2.5v - micbias1 volage is 2.5v. - - 0x06 # 2.6v - micbias1 volage is 2.6v. - - 0x07 # 2.7v - micbias1 volage is 2.7v. - - 0x08 # 2.8v - micbias1 volage is 2.8v. - - 0x09 # 2.85v - micbias1 volage is 2.85v. - - mediatek,mic-mode: - maxItems: 1 - description: | - value as 1, 2, 6 to indicate ACC/DCC mode. default is DCC mode 2. - - enum: - - 0x01 # ACC - ACC mode. - - 0x02 # DCC - low cost without in bias. - - 0x06 # DCC - low cost with in bias. - - mediatek,pwm-deb-setting: - maxItems: 15 - description: | - headset-mode-setting : Indicates customized pwm, debounce setting - accdet pwm_width, pwm_thresh, fall_delay, rise_delay - debounce0, debounce1, debounce3, debounce4 - eint pwm_width, eint pwm_thresh - eint deb(debounce0, debounce1, debounce2, debounce3), inv_debounce. - - mediatek,eint-use-ap: - maxItems: 1 - description: | - indicates to use ap gpio. - - enum: - - 0x00 # MT6359 eint detection mode. - - 0x01 # SoC GPIO detection mode. - - mediatek,eint-detect-mode: - maxItems: 1 - description: | - indicates to use ap gpio. - - enum: - - 0x00 # detection mode with higher detection power. - - 0x01 # detection mode with lower detection power. - - 0x02 # detection mode with sw moisture detection mode. - - 0x03 # detection mode with hw moisture detection mode1. - - 0x04 # detection mode with hw moisture detection mode2. - - mediatek,eint-num: - maxItems: 1 - description: | - incicates pmic eint usage. - - enum: - - 0x0 # PMIC_EINT0 - use pmic eint0 to trigger plug interrupt. - - 0x1 # PMIC_EINT1 - use pmic eint1 to trigger plug interrupt. - - 0x2 # PMIC_BI_EINT - use pmic eint0/1 to trigger plug interrupt. - - mediatek,eint-trig-mode: - maxItems: 1 - description: | - incicates pmic eint trigger mode. - - enum: - - 0x0 # PMIC_GPIO - use pmic gpio to detect plug status. - - 0x1 # PMIC_INVERTER - use pmic inverter to detect plug status. - - mediatek,eint-use-ext-res: - maxItems: 1 - description: | - select HP_EINT pull up resistance, default 0 use internal resistance. - - enum: - - 0x0 # use internal resistor. - - 0x1 # use external resistor. - - mediatek,eint-comp-vth: - maxItems: 1 - description: | - indicates comparator detection voltage. - - enum: - - 0x00 # 2.8v. - - 0x01 # 2.4v. - - 0x02 # 2.0v. - - 0x03 # 0.7v. - - mediatek,key-mode: - maxItems: 1 - description: | - incicates key mode type. - - enum: - - 0x0 # THREE_KEY - support 3-key key mode. - - 0x1 # FOUR_KEY - support 4-key key mode. - - 0x2 # TRI_KEY_CDD - support 3-key google CDD key mode. - - mediatek,three-key-thr: - maxItems: 4 - description: | - 3 key device detection threshold: 0--MD_MAX--UP_MAX--DW_MAX. - - mediatek,tri-key-cdd-thr: - maxItems: 4 - description: | - 3 key CDD device detection threshold: 0--MD_MAX--UP_MAX--DW_MAX. - - mediatek,four-key-thr: - maxItems: 4 - description: | - 4 key device detection threshold: 0--MD_MAX--VOICE_MAX--UP_MAX--DW_MAX. - -additionalProperties: false - -examples: - - | - accdet: accdet { - mediatek,accdet-name = "mt6359-accdet"; - mediatek,mic-vol = <8>; - mediatek,mic-mode = <2>; - mediatek,pwm-deb-setting = <0x500 0x500 1 0x1f0 - 0x800 0x800 0x20 0x44 - 0x4 0x1 - 0x5 0x3 0x3 0x5 0xe>; - mediatek,eint-use-ap = <0>; - mediatek,eint-detect-mode = <4>; - mediatek,eint-num = <0>; - mediatek,eint-trig-mode = <1>; - mediatek,eint-use-ext-res = <0>; - mediatek,eint-comp-vth = <2>; - mediatek,key-mode = <0>; - mediatek,three-key-thr = <0 80 220 400>; - status = "okay"; - }; -... -- 2.27.0

2 1

[PATCH] ALSA: hda/realtek: Add quirk for Lenovo Ideapad S740
by Takashi Iwai 16 Apr '21

16 Apr '21

Lenovo Ideapad S740 requires quite a few COEF setups to make its speakers working. The verb table was provided from Ryan Prescott as the result of investigation via qemu: https://github.com/ryanprescott/realtek-verb-tools/wiki/How-to-sniff-verbs-… BugLink: https://github.com/thesofproject/linux/issues/2748 Tested-by: Ryan Prescott <ryan(a)cousinscomputers.net> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/ideapad_s740_helper.c | 492 ++++++++++++++++++++++++++++ sound/pci/hda/patch_realtek.c | 11 + 2 files changed, 503 insertions(+) create mode 100644 sound/pci/hda/ideapad_s740_helper.c diff --git a/sound/pci/hda/ideapad_s740_helper.c b/sound/pci/hda/ideapad_s740_helper.c new file mode 100644 index 000000000000..564b9086e52d --- /dev/null +++ b/sound/pci/hda/ideapad_s740_helper.c @@ -0,0 +1,492 @@ +// SPDX-License-Identifier: GPL-2.0 +/* Fixes for Lenovo Ideapad S740, to be included from codec driver */ + +static const struct hda_verb alc285_ideapad_s740_coefs[] = { +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x10 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0320 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x24 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0041 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x24 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0041 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x007f }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x007f }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0001 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0001 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0001 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0001 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x003c }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0011 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x003c }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0011 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x000c }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x001a }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x000c }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x001a }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x000f }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0042 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x000f }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0042 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0010 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0040 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0010 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0040 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0003 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0009 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0003 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0009 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x001c }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x004c }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x001c }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x004c }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x001d }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x004e }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x001d }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x004e }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x001b }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0001 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x001b }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0001 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0019 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0025 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0019 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0025 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0018 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0037 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0018 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0037 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x001a }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0040 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x001a }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0040 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0016 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0076 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0016 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0076 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0017 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0010 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0017 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0010 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0015 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0015 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0015 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0015 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0007 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0086 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0007 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0086 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0002 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0001 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0002 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0001 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0002 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0002 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x24 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0042 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x24 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0042 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x007f }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x007f }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0001 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0001 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0001 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0001 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x003c }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0011 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x003c }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0011 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x000c }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x002a }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x000c }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x002a }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x000f }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0046 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x000f }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0046 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0010 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0044 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0010 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0044 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0003 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0009 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0003 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0009 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x001c }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x004c }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x001c }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x004c }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x001b }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0001 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x001b }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0001 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0019 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0025 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0019 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0025 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0018 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0037 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0018 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0037 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x001a }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0040 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x001a }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0040 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0016 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0076 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0016 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0076 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0017 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0010 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0017 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0010 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0015 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0015 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0015 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0015 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0007 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0086 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0007 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0086 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0002 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0001 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0002 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0001 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0002 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{} +}; + +static void alc285_fixup_ideapad_s740_coef(struct hda_codec *codec, + const struct hda_fixup *fix, + int action) +{ + switch (action) { + case HDA_FIXUP_ACT_PRE_PROBE: + snd_hda_add_verbs(codec, alc285_ideapad_s740_coefs); + break; + } +} diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 50437ec1b9af..1ef40f3ca259 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6239,6 +6239,9 @@ static void alc_fixup_thinkpad_acpi(struct hda_codec *codec, /* for alc295_fixup_hp_top_speakers */ #include "hp_x360_helper.c" +/* for alc285_fixup_ideapad_s740_coef() */ +#include "ideapad_s740_helper.c" + enum { ALC269_FIXUP_GPIO2, ALC269_FIXUP_SONY_VAIO, @@ -6436,6 +6439,7 @@ enum { ALC282_FIXUP_ACER_DISABLE_LINEOUT, ALC255_FIXUP_ACER_LIMIT_INT_MIC_BOOST, ALC256_FIXUP_ACER_HEADSET_MIC, + ALC285_FIXUP_IDEAPAD_S740_COEF, }; static const struct hda_fixup alc269_fixups[] = { @@ -7920,6 +7924,12 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC }, + [ALC285_FIXUP_IDEAPAD_S740_COEF] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc285_fixup_ideapad_s740_coef, + .chained = true, + .chain_id = ALC269_FIXUP_THINKPAD_ACPI, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -8263,6 +8273,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x3176, "ThinkCentre Station", ALC283_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x17aa, 0x3178, "ThinkCentre Station", ALC283_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x17aa, 0x3818, "Lenovo C940", ALC298_FIXUP_LENOVO_SPK_VOLUME), + SND_PCI_QUIRK(0x17aa, 0x3827, "Ideapad S740", ALC285_FIXUP_IDEAPAD_S740_COEF), SND_PCI_QUIRK(0x17aa, 0x3902, "Lenovo E50-80", ALC269_FIXUP_DMIC_THINKPAD_ACPI), SND_PCI_QUIRK(0x17aa, 0x3977, "IdeaPad S210", ALC283_FIXUP_INT_MIC), SND_PCI_QUIRK(0x17aa, 0x3978, "Lenovo B50-70", ALC269_FIXUP_DMIC_THINKPAD_ACPI), -- 2.26.2

4 7

[PATCH] ASoC: ak4458: check reset control status
by Shengjiu Wang 16 Apr '21

16 Apr '21

From: Viorel Suman <viorel.suman(a)nxp.com> check the return value of ak4458_rstn_control. Signed-off-by: Viorel Suman <viorel.suman(a)nxp.com> Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/codecs/ak4458.c | 19 +++++++++++++++---- 1 file changed, 15 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/ak4458.c b/sound/soc/codecs/ak4458.c index 85a1d00894a9..0a94973889a1 100644 --- a/sound/soc/codecs/ak4458.c +++ b/sound/soc/codecs/ak4458.c @@ -419,8 +419,13 @@ static int ak4458_hw_params(struct snd_pcm_substream *substream, snd_soc_component_update_bits(component, AK4458_00_CONTROL1, AK4458_DIF_MASK, format); - ak4458_rstn_control(component, 0); - ak4458_rstn_control(component, 1); + ret = ak4458_rstn_control(component, 0); + if (ret) + return ret; + + ret = ak4458_rstn_control(component, 1); + if (ret) + return ret; return 0; } @@ -429,6 +434,7 @@ static int ak4458_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) { struct snd_soc_component *component = dai->component; struct ak4458_priv *ak4458 = snd_soc_component_get_drvdata(component); + int ret; switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBS_CFS: /* Slave Mode */ @@ -461,8 +467,13 @@ static int ak4458_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) ak4458->fmt == SND_SOC_DAIFMT_PDM ? AK4458_DP_MASK : 0); - ak4458_rstn_control(component, 0); - ak4458_rstn_control(component, 1); + ret = ak4458_rstn_control(component, 0); + if (ret) + return ret; + + ret = ak4458_rstn_control(component, 1); + if (ret) + return ret; return 0; } -- 2.27.0

2 2

Add UCM support for SDM845 based Lenovo yoga c360
by GitHub pull_request - opened 16 Apr '21

16 Apr '21

alsa-project/alsa-ucm-conf pull request #88 was opened from Srinivas-Kandagatla: One of the Lenovo YOGA C360 variant is based off Qualcomm SDM845 SoC. This PR adds support to On Board Speakers and Headset. This PR also has a small change to the codec Headphone sequences to be inline with other Sequences. thanks, srini Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/88 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/88.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Re: Implicit feedback on BOSS GT-1, the saga continues...
by Geraldo Nascimento 16 Apr '21

16 Apr '21

> > Mike, I just realize I missed reply-to-all once again. In case you were wondering, I was looking for a sync_ep event right in the start of the loading of snd-usb-audio. It happens inside pcm.c. If you see one of those specifically let me know. With my Pioneer device, the DDJ-SR2, snd-usb-audio recognized the device's implicit feedback EP as sync_ep and that prevented JACK from starting. Em Qua, 7 de abr de 2021 23:47, Mike Oliphant <oliphant(a)nostatic.org> > escreveu: > >> Hi Geraldo - I had the same thought and I have tried enabling playback >> feedback while turning off capture feedback without my endpoint patch. It >> doesn't work - at least not on the GT-1. Playback/capture fails to start. >> > > Would you mind sharing the dyndbg logs for the failure case from the point > you connect the device, and trying to start JACK? > > Sometimes failures elucidate more than success cases. > > >> Speaking of testing, if you (or anyone else following this discussion) >> have one of these Roland/BOSS devices, it would be great if you could try >> testing with playback feedback enabled and my endpoint.c workaround. >> > > I don't have the hardware, but I'm sure it works. The problem is that > endpoint.c code really shouldn't be disabled unless all the other options > fail, and even then, I'm not so sure the maintainers will be so willing to > disable it. So let's keep some other options on the table by debugging a > little further. > > Thank you, > Geraldo > > >> Mike >> >> On Wed, Apr 7, 2021 at 7:21 PM Geraldo Nascimento < >> geraldogabriel(a)gmail.com> wrote: >> >>> Thanks, Mike, one thing I can tell you right off the bat is that if you >>> turn implicit feedback sync on for both endpoints at the same time like you >>> mentioned, without your workaround inside endpoint.c, they most likely will >>> stall because they would both be waiting for each other to start. >>> >>> Have you tried engaging the generic implicit feedback sync just for the >>> playback, thereby commenting out the capture implicit feedback entries, and >>> re-enabling snd_usb_endpoint_implicit_feedback_sink(ep) inside endpoint.c? >>> Does playback still fail to start? >>> >>> If you choose to perform this simple test please share with us the >>> test's dyndbg logs right from the moment you turn on the device, not just >>> after you start jackd. >>> >>> Sorry for the endless do-this, do-that. Such is the pace of teledebugging >>> >>> >>> Em Qua, 7 de abr de 2021 21:38, Mike Oliphant <oliphant(a)nostatic.org> >>> escreveu: >>> >>>> Geraldo - here is the dmesg output when starting up jackd for >>>> capture/playback: >>>> >>>> With my patch: >>>> >>>> [ 136.734081] usb 3-1.2: Open EP 0x8e, iface=2:1, idx=0 >>>> [ 136.734088] usb 3-1.2: channels=2, rate=44100, format=S32_LE, >>>> period_bytes=256, periods=2, implicit_fb=0 >>>> [ 136.734092] usb 3-1.2: Setting usb interface 2:0 for EP 0x8e >>>> [ 136.734233] usb 3-1.2: Setting usb interface 2:1 for EP 0x8e >>>> [ 136.734381] usb 3-1.2: 2:1 Set sample rate 44100, clock 0 >>>> [ 136.734388] usb 3-1.2: Setting params for data EP 0x8e, pipe 0x70480 >>>> [ 136.734394] usb 3-1.2: Set up 12 URBS, ret=0 >>>> [ 136.734466] usb 3-1.2: Open EP 0xd, iface=1:1, idx=0 >>>> [ 136.734468] usb 3-1.2: channels=2, rate=44100, format=S32_LE, >>>> period_bytes=256, periods=2, implicit_fb=1 >>>> [ 136.734471] usb 3-1.2: Reopened EP 0x8e (count 1) >>>> [ 136.734473] usb 3-1.2: Setting usb interface 1:0 for EP 0xd >>>> [ 136.734605] usb 3-1.2: Setting usb interface 1:1 for EP 0xd >>>> [ 136.734762] usb 3-1.2: 1:1 Set sample rate 44100, clock 0 >>>> [ 136.734774] usb 3-1.2: Setting params for data EP 0xd, pipe 0x68400 >>>> [ 136.734781] usb 3-1.2: Set up 12 URBS, ret=0 >>>> [ 136.734830] usb 3-1.2: Starting data EP 0xd (running 0) >>>> [ 136.734873] usb 3-1.2: 12 URBs submitted for EP 0xd >>>> [ 136.734876] usb 3-1.2: Starting data EP 0x8e (running 0) >>>> [ 136.734906] usb 3-1.2: 12 URBs submitted for EP 0x8e >>>> [ 136.740627] usb 3-1.2: Starting data EP 0x8e (running 1) >>>> [ 136.740634] usb 3-1.2: 2:1 Start Capture PCM >>>> [ 136.740635] usb 3-1.2: 1:1 Start Playback PCM >>>> >>>> >>>> With the unpatched kernel: >>>> >>>> [ 344.813203] usb 3-1.2: Open EP 0x8e, iface=2:1, idx=0 >>>> [ 344.813210] usb 3-1.2: channels=2, rate=44100, format=S32_LE, >>>> period_bytes=256, periods=2, implicit_fb=1 >>>> [ 344.813213] usb 3-1.2: Open EP 0xd, iface=1:1, idx=0 >>>> [ 344.813215] usb 3-1.2: channels=2, rate=44100, format=S32_LE, >>>> period_bytes=256, periods=2, implicit_fb=0 >>>> [ 344.813217] usb 3-1.2: Setting usb interface 2:0 for EP 0x8e >>>> [ 344.813367] usb 3-1.2: Setting usb interface 2:1 for EP 0x8e >>>> [ 344.813494] usb 3-1.2: 2:1 Set sample rate 44100, clock 0 >>>> [ 344.813503] usb 3-1.2: Setting params for data EP 0x8e, pipe 0x70580 >>>> [ 344.813509] usb 3-1.2: Set up 12 URBS, ret=0 >>>> [ 344.813513] usb 3-1.2: Setting usb interface 1:0 for EP 0xd >>>> [ 344.813587] usb 3-1.2: Setting usb interface 1:1 for EP 0xd >>>> [ 344.813758] usb 3-1.2: 1:1 Set sample rate 44100, clock 0 >>>> [ 344.813765] usb 3-1.2: Setting params for data EP 0xd, pipe 0x68500 >>>> [ 344.813770] usb 3-1.2: Set up 2 URBS, ret=0 >>>> [ 344.813858] usb 3-1.2: Reopened EP 0xd (count 1) >>>> [ 344.813883] usb 3-1.2: Starting data EP 0xd (running 0) >>>> [ 344.813899] usb 3-1.2: 2 URBs submitted for EP 0xd >>>> [ 344.819499] usb 3-1.2: Starting data EP 0x8e (running 0) >>>> [ 344.819532] usb 3-1.2: 12 URBs submitted for EP 0x8e >>>> [ 344.819535] usb 3-1.2: Starting data EP 0xd (running 1) >>>> [ 344.819538] usb 3-1.2: 2:1 Start Capture PCM >>>> [ 344.819540] usb 3-1.2: 1:1 Start Playback PCM >>>> [ 344.820663] usb 3-1.2: Stopping data EP 0xd (running 2) >>>> [ 344.820688] usb 3-1.2: Stopping data EP 0x8e (running 1) >>>> [ 344.820693] usb 3-1.2: 2:1 Stop Capture PCM >>>> [ 344.820695] usb 3-1.2: Stopping data EP 0xd (running 1) >>>> [ 344.820697] usb 3-1.2: 1:1 Stop Playback PCM >>>> [ 344.833799] usb 3-1.2: Starting data EP 0xd (running 0) >>>> [ 344.833852] usb 3-1.2: 2 URBs submitted for EP 0xd >>>> [ 344.833867] usb 3-1.2: Starting data EP 0x8e (running 0) >>>> [ 344.833888] usb 3-1.2: 12 URBs submitted for EP 0x8e >>>> [ 344.833890] usb 3-1.2: Starting data EP 0xd (running 1) >>>> [ 344.833891] usb 3-1.2: 2:1 Start Capture PCM >>>> [ 344.833893] usb 3-1.2: 1:1 Start Playback PCM >>>> >>>> On Wed, Apr 7, 2021 at 1:30 PM Geraldo Nascimento < >>>> geraldogabriel(a)gmail.com> wrote: >>>> >>>>> Em Qua, 7 de abr de 2021 17:16, Mike Oliphant <oliphant(a)nostatic.org> >>>>> escreveu: >>>>> >>>>>> Hi Geraldo - I don't have that patch applied, but it shouldn't make >>>>>> any behavioral difference - it just seems to be simplifying the code. >>>>>> >>>>> >>>>> Point taken. >>>>> >>>>> >>>>>> The issue is that the BOSS GT-1 *does* need implicit feedback on >>>>>> playback to avoid clock timing issues, and the current behavior is >>>>>> disabling that feedback. >>>>>> >>>>> >>>>> Mike, would you mind posting some dyndbg logs for both the stock >>>>> behaviour and your endpoint.c/generic quirk modified behaviour? >>>>> >>>>> Just add snd_usb_audio.dyndbg=+p to your kernel options and please >>>>> share the logs. >>>>> >>>>> >>>>>> Mike >>>>>> >>>>>> On Wed, Apr 7, 2021 at 1:04 PM Geraldo Nascimento < >>>>>> geraldogabriel(a)gmail.com> wrote: >>>>>> >>>>>>> Hey Mike, did you catch the latest patch by Takashi Iwai for capture >>>>>>> quirky devices? >>>>>>> >>>>>>> You can find it here: >>>>>>> https://patchwork.kernel.org/project/alsa-devel/patch/20210406113837.32041-… >>>>>>> >>>>>>> Em Qua, 7 de abr de 2021 16:55, Mike Oliphant <oliphant(a)nostatic.org> >>>>>>> escreveu: >>>>>>> >>>>>>>> I had thought that the recent implicit feedback changes were fully >>>>>>>> working >>>>>>>> on the BOSS GT-1, but it turns out that I just hadn't tested well >>>>>>>> enough. >>>>>>>> >>>>>>>> Audio playback and capture works, but with periodic dropouts. I get >>>>>>>> the >>>>>>>> exact same behavior as I did with the quirk to completely disable >>>>>>>> implicit >>>>>>>> feedback. Without the implicit feedback, you get dropouts from >>>>>>>> clock drift >>>>>>>> - how bad probably varies from card to card. On mine it is every >>>>>>>> second or >>>>>>>> so. >>>>>>>> >>>>>>>> If I switch playback feedback for the GT-1 to generic by doing >>>>>>>> "IMPLICIT_FB_GENERIC_DEV(0x0582, 0x01d6)", I get the previous old >>>>>>>> behavior, >>>>>>>> which is that playback completely fails to start. >>>>>>>> >>>>>>>> With generic playback feedback, and using my previous patch to >>>>>>>> endpoint.c >>>>>>>> to avoid playback waiting on capture mentioned here: >>>>>>>> >>>>>>>> >>>>>>>> https://mailman.alsa-project.org/pipermail/alsa-devel/2020-January/161951.h… >>>>>>>> >>>>>>>> playback and capture work perfectly for me. >>>>>>>> >>>>>>>

4 34

[PATCH 0/4] ASoC: soc-pcm: ignore dummy-DAI at soc_pcm_params_symmetry()
by Kuninori Morimoto 16 Apr '21

16 Apr '21

Hi Mark I noticed if we have... 1) Sound Card used DPCM 2) It exchanges rate to 48kHz by using .be_hw_params_fixup() 3) Codec had symmetric_rate = 1 We will get below error. I didn't confirm, but maybe same things happen if it exchanged channels/sample_bits. # aplay 44100.wav # aplay 44100.wav => [kernel] be.ak4613-hifi: ASoC: unmatched rate symmetry: snd-soc-dummy-dai:44100 - soc_pcm_params_symmetry:48000 [kernel] be.ak4613-hifi: ASoC: hw_params BE failed -22 [kernel] fe.rsnd-dai.0: ASoC: hw_params BE failed -22 aplay: set_params:1407: Unable to install hw params: ACCESS: RW_INTERLEAVED FORMAT: S16_LE SUBFORMAT: STD SAMPLE_BITS: 16 FRAME_BITS: 32 CHANNELS: 2 RATE: 44100 PERIOD_TIME: (23219 23220) PERIOD_SIZE: 1024 PERIOD_BYTES: 4096 PERIODS: 4 BUFFER_TIME: (92879 92880) BUFFER_SIZE: 4096 BUFFER_BYTES: 16384 TICK_TIME: 0 This patch-set solves this issue. patch 1) - 3) are just cleanup patches. 4) is fot this issue. Link: https://lore.kernel.org/r/87a6q0z4xt.wl-kuninori.morimoto.gx@renesas.com Kuninori Morimoto (4): ASoC: soc-pcm: don't use "name" on __soc_pcm_params_symmetry() macro ASoC: soc-pcm: indicate DAI name if soc_pcm_params_symmetry() failed ASoC: soc-utils: add snd_soc_component_is_dummy() ASoC: soc-pcm: ignore dummy-DAI at soc_pcm_params_symmetry() include/sound/soc-component.h | 1 + sound/soc/soc-core.c | 2 +- sound/soc/soc-pcm.c | 14 ++++++++------ sound/soc/soc-utils.c | 6 ++++++ 4 files changed, 16 insertions(+), 7 deletions(-) -- 2.25.1

2 5

[RESEND PATCH v2] ASoC: core: Don't set platform name when of_node is set
by Daniel Baluta 16 Apr '21

16 Apr '21

From: Daniel Baluta <daniel.baluta(a)nxp.com> A DAI link has 3 components: * CPU * platform * codec(s) A component is specified via: * name * of_node * dai_name In order to avoid confusion when building a sound card we disallow matching by both name and of_node (1). soc_check_tplg_fes allows overriding certain BE links by overriding BE link name. This doesn't work well if BE link was specified via DT, because we end up with a link with both name and of_node specified which is conflicting with (1). In order to fix this we need to: * override of_node if component was specified via DT * override name, otherwise. Signed-off-by: Daniel Baluta <daniel.baluta(a)nxp.com> --- sound/soc/soc-core.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 522bae73640a..7a1c011a7fe0 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1648,7 +1648,11 @@ static void soc_check_tplg_fes(struct snd_soc_card *card) dev_err(card->dev, "init platform error"); continue; } - dai_link->platforms->name = component->name; + + if (component->dev->of_node) + dai_link->platforms->of_node = component->dev->of_node; + else + dai_link->platforms->name = component->name; /* convert non BE into BE */ if (!dai_link->no_pcm) { -- 2.27.0

2 1

[PATCH v2] ALSA: hda/hdmi: fix race in handling acomp ELD notification at resume
by Kai Vehmanen 16 Apr '21

16 Apr '21

When snd-hda-codec-hdmi is used with ASoC HDA controller like SOF (acomp used for ELD notifications), display connection change done during suspend, can be lost due to following sequence of events: 1. system in S3 suspend 2. DP/HDMI receiver connected 3. system resumed 4. HDA controller resumed, but card->deferred_resume_work not complete 5. acomp eld_notify callback 6. eld_notify ignored as power state is not CTL_POWER_D0 7. HDA resume deferred work completed, power state set to CTL_POWER_D0 This results in losing the notification, and the jack state reported to user-space is not correct. The check on step 6 was added in commit 8ae743e82f0b ("ALSA: hda - Skip ELD notification during system suspend"). It would seem with the deferred resume logic in ASoC core, this check is not safe. Fix the issue by modifying the check to use "dev.power.power_state.event" instead of ALSA specific card power state variable. BugLink: https://github.com/thesofproject/linux/issues/2825 Suggested-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> --- sound/pci/hda/patch_hdmi.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) Changes for V2: - use codec->core.dev.power.power_state.event instead of card state - verified on multiple systems and confirmed that checking for PM_EVENT_ON is not sufficient -- when resuming from S3, state can be PM_EVENT_RESUME at eld_notify, and if notification is ignored in this case, the original bugis hit (monitor state upon resume is lost) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 5de3666a7101..4b2cc8cb55c4 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -2654,7 +2654,7 @@ static void generic_acomp_pin_eld_notify(void *audio_ptr, int port, int dev_id) /* skip notification during system suspend (but not in runtime PM); * the state will be updated at resume */ - if (snd_power_get_state(codec->card) != SNDRV_CTL_POWER_D0) + if (codec->core.dev.power.power_state.event == PM_EVENT_SUSPEND) return; /* ditto during suspend/resume process itself */ if (snd_hdac_is_in_pm(&codec->core)) @@ -2840,7 +2840,7 @@ static void intel_pin_eld_notify(void *audio_ptr, int port, int pipe) /* skip notification during system suspend (but not in runtime PM); * the state will be updated at resume */ - if (snd_power_get_state(codec->card) != SNDRV_CTL_POWER_D0) + if (codec->core.dev.power.power_state.event == PM_EVENT_SUSPEND) return; /* ditto during suspend/resume process itself */ if (snd_hdac_is_in_pm(&codec->core)) base-commit: 1867068c322ea7a627fde8d947a9e40db32816a4 -- 2.31.0

2 1

[PATCH] ALSA: hda/realtek: fix mute/micmute LEDs for HP ProBook 445 G7
by Jonas Witschel 16 Apr '21

16 Apr '21

The HP ProBook 445 G7 (17T32ES) uses ALC236. Like ALC236_FIXUP_HP_GPIO_LED, COEF index 0x34 bit 5 is used to control the playback mute LED, but the microphone mute LED is controlled using pin VREF instead of a COEF index. AlsaInfo: https://alsa-project.org/db/?f=0d3f4d1af39cc359f9fea9b550727ee87e5cf45a Signed-off-by: Jonas Witschel <diabonas(a)archlinux.org> --- sound/pci/hda/patch_realtek.c | 25 +++++++++++++++++++++++++ 1 file changed, 25 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 00a15e7cfdb9..c4372d143051 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4445,6 +4445,25 @@ static void alc236_fixup_hp_mute_led(struct hda_codec *codec, alc236_fixup_hp_coef_micmute_led(codec, fix, action); } +static void alc236_fixup_hp_micmute_led_vref(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + struct alc_spec *spec = codec->spec; + + if (action == HDA_FIXUP_ACT_PRE_PROBE) { + spec->cap_mute_led_nid = 0x1a; + snd_hda_gen_add_micmute_led_cdev(codec, vref_micmute_led_set); + codec->power_filter = led_power_filter; + } +} + +static void alc236_fixup_hp_mute_led_micmute_vref(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + alc236_fixup_hp_mute_led_coefbit(codec, fix, action); + alc236_fixup_hp_micmute_led_vref(codec, fix, action); +} + #if IS_REACHABLE(CONFIG_INPUT) static void gpio2_mic_hotkey_event(struct hda_codec *codec, struct hda_jack_callback *event) @@ -6407,6 +6426,7 @@ enum { ALC285_FIXUP_HP_MUTE_LED, ALC236_FIXUP_HP_GPIO_LED, ALC236_FIXUP_HP_MUTE_LED, + ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF, ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET, ALC295_FIXUP_ASUS_MIC_NO_PRESENCE, ALC269VC_FIXUP_ACER_VCOPPERBOX_PINS, @@ -7653,6 +7673,10 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc236_fixup_hp_mute_led, }, + [ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc236_fixup_hp_mute_led_micmute_vref, + }, [ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET] = { .type = HDA_FIXUP_VERBS, .v.verbs = (const struct hda_verb[]) { @@ -8070,6 +8094,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x869d, "HP", ALC236_FIXUP_HP_MUTE_LED), SND_PCI_QUIRK(0x103c, 0x8724, "HP EliteBook 850 G7", ALC285_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8729, "HP", ALC285_FIXUP_HP_GPIO_LED), + SND_PCI_QUIRK(0x103c, 0x8730, "HP ProBook 445 G7", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), SND_PCI_QUIRK(0x103c, 0x8736, "HP", ALC285_FIXUP_HP_GPIO_AMP_INIT), SND_PCI_QUIRK(0x103c, 0x8760, "HP", ALC285_FIXUP_HP_MUTE_LED), SND_PCI_QUIRK(0x103c, 0x877a, "HP", ALC285_FIXUP_HP_MUTE_LED), -- 2.31.1

2 2

[PATCH] ALSA: isa: Fix inconsistent indenting
by Yang Li 16 Apr '21

16 Apr '21

Kernel test robot throws below warning -> smatch warnings: sound/isa/opti9xx/opti92x-ad1848.c:191 snd_opti9xx_init() warn: inconsistent indenting sound/isa/opti9xx/opti92x-ad1848.c:488 snd_opti9xx_init() warn: inconsistent indenting Reported-by: Abaci Robot <abaci(a)linux.alibaba.com> Signed-off-by: Yang Li <yang.lee(a)linux.alibaba.com> --- sound/isa/opti9xx/opti92x-ad1848.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/isa/opti9xx/opti92x-ad1848.c b/sound/isa/opti9xx/opti92x-ad1848.c index 08e61d9..234f023 100644 --- a/sound/isa/opti9xx/opti92x-ad1848.c +++ b/sound/isa/opti9xx/opti92x-ad1848.c @@ -188,7 +188,7 @@ static int snd_opti9xx_init(struct snd_opti9xx *chip, chip->mc_base_size = opti9xx_mc_size[hardware]; } #else - chip->mc_base_size = opti9xx_mc_size[hardware]; + chip->mc_base_size = opti9xx_mc_size[hardware]; #endif switch (hardware) { @@ -485,7 +485,7 @@ static int snd_opti9xx_configure(struct snd_opti9xx *chip, #endif /* CS4231 || OPTi93X */ #ifndef OPTi93X - outb(irq_bits << 3 | dma_bits, chip->wss_base); + outb(irq_bits << 3 | dma_bits, chip->wss_base); #else /* OPTi93X */ snd_opti9xx_write(chip, OPTi9XX_MC_REG(3), (irq_bits << 3 | dma_bits)); #endif /* OPTi93X */ -- 1.8.3.1

2 1

[PATCH 1/1] ALSA: hda/realtek: fix mic boost on Intel NUC 8
by Phil Calvin 16 Apr '21

16 Apr '21

Fix two bugs with the Intel HDA Realtek ALC233 sound codec present in Intel NUC NUC8i7BEH and probably a few other similar NUC models. These codecs advertise a 4-level microphone input boost amplifier on pin 0x19, but the highest two boost settings do not work correctly, and produce only low analog noise that does not seem to contain any discernible signal. There is an existing fixup for this exact problem but for a different PCI subsystem ID, so we re-use that logic. Changing the boost level also triggers a DC spike in the input signal that bleeds off over about a second and overwhelms any input during that time. Thankfully, the existing fixup has the side effect of making the boost control show up in userspace as a mute/unmute switch, and this keeps (e.g.) PulseAudio from fiddling with it during normal input volume adjustments. Finally, the NUC hardware has built-in inverted stereo mics. This patch also enables the usual fixup for this so the two channels cancel noise instead of the actual signal. Signed-off-by: Phil Calvin <phil(a)philcalvin.com> --- sound/pci/hda/patch_realtek.c | 13 +++++++++++++ 1 file changed, 13 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 58946d069ee5..e1fd4c81965a 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6405,6 +6405,8 @@ enum { ALC269_FIXUP_LEMOTE_A1802, ALC269_FIXUP_LEMOTE_A190X, ALC256_FIXUP_INTEL_NUC8_RUGGED, + ALC233_FIXUP_INTEL_NUC8_DMIC, + ALC233_FIXUP_INTEL_NUC8_BOOST, ALC256_FIXUP_INTEL_NUC10, ALC255_FIXUP_XIAOMI_HEADSET_MIC, ALC274_FIXUP_HP_MIC, @@ -7122,6 +7124,16 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc233_fixup_lenovo_line2_mic_hotkey, }, + [ALC233_FIXUP_INTEL_NUC8_DMIC] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc_fixup_inv_dmic, + .chained = true, + .chain_id = ALC233_FIXUP_INTEL_NUC8_BOOST, + }, + [ALC233_FIXUP_INTEL_NUC8_BOOST] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc269_fixup_limit_int_mic_boost + }, [ALC255_FIXUP_DELL_SPK_NOISE] = { .type = HDA_FIXUP_FUNC, .v.func = alc_fixup_disable_aamix, @@ -8265,6 +8277,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1c06, 0x2013, "Lemote A1802", ALC269_FIXUP_LEMOTE_A1802), SND_PCI_QUIRK(0x1c06, 0x2015, "Lemote A190X", ALC269_FIXUP_LEMOTE_A190X), SND_PCI_QUIRK(0x8086, 0x2080, "Intel NUC 8 Rugged", ALC256_FIXUP_INTEL_NUC8_RUGGED), + SND_PCI_QUIRK(0x8086, 0x2074, "Intel NUC 8", ALC233_FIXUP_INTEL_NUC8_DMIC), SND_PCI_QUIRK(0x8086, 0x2081, "Intel NUC 10", ALC256_FIXUP_INTEL_NUC10), #if 0 -- 2.30.2

2 1

[PATCH] ASoC: soc-pcm: fixup soc_pcm_params_symmetry()
by Kuninori Morimoto 16 Apr '21

16 Apr '21

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> commit 3a9067211122 ("ASoC: soc-pcm: cleanup soc_pcm_params_symmetry()") cleanups soc_pcm_params_symmetry() by addig new __soc_pcm_params_symmetry() macro. It checks symmetry first, and checks each DAI settings if symmetry was true. But original code checked symmetric_rate : DAI_Link / CPU (A) symmetric_channels : DAI_Link / CPU / Codec (B) symmetric_sample_bits : DAI_Link / CPU / Codec (B) (A) was using for_each_rtd_cpu_dais() (B) was using for_each_rtd_dais() Current code is using (B) for all symmetric_xxx. This is bug. One note is that we can use .be_hw_params_fixup in DPCM case. This means, BE settings might be fixuped/updated by FE. It can be happen not only for rate, but for channels/sample_bits too. And also, DPCM uses dummy-DAI which will be used for all DPCM connectoin. ALSA SoC will clean DAI params (a) if it was last user (b). But dummy-DAI is used from many place, it might not be cleaned. static int soc_pcm_hw_clean(...) { ... for_each_rtd_dais(rtd, i, dai) { ... (b) if (snd_soc_dai_active(dai) == 1) (a) soc_pcm_set_dai_params(dai, NULL); ... } ... } The solution maybe (A1) Symmetric checks CPU only (A2) Symmetric checks both CPU / Codec, but ignores dummy-DAI This patch selects A1. For example, if Sound Card is exchanging all rate to 48kHz by using .be_hw_params_fixup() on DPCM, below can be happen. It is using 44100 Hz, but has mismatch between dummy-DAI which is keeping 48kHz from 1st aplay. # aplay 44100.wav # aplay 44100.wav => [kernel] be.ak4613-hifi: ASoC: unmatched rate symmetry: 44100 - 48000 [kernel] be.ak4613-hifi: ASoC: hw_params BE failed -22 [kernel] fe.rsnd-dai.0: ASoC: hw_params BE failed -22 aplay: set_params:1407: Unable to install hw params: ACCESS: RW_INTERLEAVED FORMAT: S16_LE SUBFORMAT: STD SAMPLE_BITS: 16 FRAME_BITS: 32 CHANNELS: 2 RATE: 44100 PERIOD_TIME: (23219 23220) PERIOD_SIZE: 1024 PERIOD_BYTES: 4096 PERIODS: 4 BUFFER_TIME: (92879 92880) BUFFER_SIZE: 4096 BUFFER_BYTES: 16384 TICK_TIME: 0 Fixes: 3a9067211122 ("ASoC: soc-pcm: cleanup soc_pcm_params_symmetry()") Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- Hi Mark This patch is needed for latest linus tree (= for v5.12). Please let me know if you want (A2). sound/soc/soc-pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 02968a4e52b4..92e95e4aef9f 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -388,7 +388,7 @@ static int soc_pcm_params_symmetry(struct snd_pcm_substream *substream, #define __soc_pcm_params_symmetry(name) \ symmetry = rtd->dai_link->symmetric_##name; \ - for_each_rtd_dais(rtd, i, dai) \ + for_each_rtd_cpu_dais(rtd, i, dai) \ symmetry |= dai->driver->symmetric_##name; \ \ if (symmetry) \ -- 2.25.1

2 3

[PATCH 0/1] ALSA: hda/realtek: fix mic boost on Intel NUC 8
by Phil Calvin 16 Apr '21

16 Apr '21

I realized that I mistakenly sent this patch only to the listserv, and not directly to the relevant maintainers as I should have. I'm sorry for my error and the resulting list noise! A big thank you to everybody that helps maintain the Intel HDA subsystem. It is a well-documented and clear codebase to enter as a first-time kernel hacker. Thanks also to Dr. Iwai for `hda-emu`, which saved me a lot of rebooting while testing. I have tested the patch on this hardware both by recompiling the relevant modules and running them with the 5.10 kernel distributed in Debian `testing`, and atop the 5.12-rc6 kernel built with `make deb-pkg`. I could not decipher the ordering of the fixup tables, so I am of course happy to move these around to fit the organizational scheme as needed. I'm not an EE, but I am inclined to think the upper two boost values can probably be made to work by some kind of software workaround, given that they do produce analog noise coming from something. If you have any hunches about a possible workaround based on experience with other similar codecs, please point me to those fixups on- or off-list and I can try them out on my hardware. Phil Phil Calvin (1): ALSA: hda/realtek: fix mic boost on Intel NUC 8 sound/pci/hda/patch_realtek.c | 13 +++++++++++++ 1 file changed, 13 insertions(+) -- 2.30.2

1 0

[PATCH v3 0/2] clean up dt_binding_check and dtbs_check warnings of renesas, rsnd.yaml
by Zhen Lei 15 Apr '21

15 Apr '21

v2 --> v3: Modified based on Rob's review comments. For example, "ports" and "port" are not allowed at the same time. For the details, please refer: https://www.mail-archive.com/linux-kernel@vger.kernel.org/msg2539753.html v1 --> v2: The analysis was not clear in the previous version, "$ref: audio-graph.yaml#" is not required. It was then adjusted based on the following patch series: [1] https://lore.kernel.org/r/20210323163634.877511-1-robh@kernel.org/ v1: When I do dt_binding_check for all YAML files, below warning is reported: /root/mainline/Documentation/devicetree/bindings/sound/renesas,rsnd.example.dt.yaml: sound@ec500000: 'dais' is a required property From schema: /root/mainline/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml make dt_binding_check -j24 2>err.txt Before: cat err.txt | wc -l 2 After: cat err.txt | wc -l 0 make dtbs_check DT_SCHEMA_FILES=Documentation/devicetree/bindings/sound/renesas,rsnd.yaml -j24 2>err.txt Before: cat err.txt | wc -l 130 After: cat err.txt | wc -l 0 Zhen Lei (2): ASoC: dt-bindings: renesas, rsnd: Clear warning 'dais' is a required property ASoC: dt-bindings: renesas, rsnd: Clear warning 'ports' does not match any of the regexes .../devicetree/bindings/sound/renesas,rsnd.yaml | 13 ++++++++++++- 1 file changed, 12 insertions(+), 1 deletion(-) -- 2.26.0.106.g9fadedd

2 4

[PATCH v2] ASoC: SOF: use current DAI config during resume
by Pierre-Louis Bossart 15 Apr '21

15 Apr '21

From: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Recently, the sof_pcm_dai_link_fixup() function was updated to match SSP config with the PCM hw_params and set the current_config for the DAI widget. But the sof_restore_pipelines() function still chooses the default config for the DAI widget upon resuming. Fix this to use the last used config when setting up the DAI widget during resume. Fixes: c943a586f6e49 ("ASoC: SOF: match SSP config with pcm hw params") Reviewed-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Signed-off-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- v2: used SHA1 from Mark's tree, not the SOF one. Resending on behalf of Ranjani -> added Signed-off-by 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 0d0d47dc0246..510883cd9107 100644 --- a/sound/soc/sof/sof-audio.c +++ b/sound/soc/sof/sof-audio.c @@ -267,7 +267,7 @@ int sof_restore_pipelines(struct device *dev) /* restore dai links */ list_for_each_entry_reverse(dai, &sdev->dai_list, list) { struct sof_ipc_reply reply; - struct sof_ipc_dai_config *config = dai->dai_config; + struct sof_ipc_dai_config *config = &dai->dai_config[dai->current_config]; if (!config) { dev_err(dev, "error: no config for DAI %s\n", -- 2.25.1

2 1

[PATCH v1] ASoC: Intel: kbl_da7219_max98927: Fix kabylake_ssp_fixup function
by Lukasz Majczak 15 Apr '21

15 Apr '21

kabylake_ssp_fixup function uses snd_soc_dpcm to identify the codecs DAIs. The HW parameters are changed based on the codec DAI of the stream. The earlier approach to get snd_soc_dpcm was using container_of() macro on snd_pcm_hw_params. The structures have been modified over time and snd_soc_dpcm does not have snd_pcm_hw_params as a reference but as a copy. This causes the current driver to crash when used. This patch changes the way snd_soc_dpcm is extracted. snd_soc_pcm_runtime holds 2 dpcm instances (one for playback and one for capture). 2 codecs on the SSP are dmic (capture) and speakers (playback). Based on the stream direction, snd_soc_dpcm is extracted from snd_soc_pcm_runtime. Tested for all use cases of the driver. Based on similar fix in kbl_rt5663_rt5514_max98927.c from Harsha Priya <harshapriya.n(a)intel.com> and Vamshi Krishna Gopal <vamshi.krishna.gopal(a)intel.com> Cc: <stable(a)vger.kernel.org> # 5.4+ Signed-off-by: Lukasz Majczak <lma(a)semihalf.com> --- Hi, This is basically a cherry-pick of this change: https://patchwork.kernel.org/project/alsa-devel/patch/1595432147-11166-1-gi… just applied to the kbl_da7219_max98927. Best regards, Lukasz sound/soc/intel/boards/kbl_da7219_max98927.c | 38 +++++++++++++++----- 1 file changed, 30 insertions(+), 8 deletions(-) diff --git a/sound/soc/intel/boards/kbl_da7219_max98927.c b/sound/soc/intel/boards/kbl_da7219_max98927.c index 9dfe5bd9180d..4b7b4a044f81 100644 --- a/sound/soc/intel/boards/kbl_da7219_max98927.c +++ b/sound/soc/intel/boards/kbl_da7219_max98927.c @@ -284,11 +284,33 @@ static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd, struct snd_interval *chan = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); - struct snd_soc_dpcm *dpcm = container_of( - params, struct snd_soc_dpcm, hw_params); - struct snd_soc_dai_link *fe_dai_link = dpcm->fe->dai_link; - struct snd_soc_dai_link *be_dai_link = dpcm->be->dai_link; + struct snd_soc_dpcm *dpcm, *rtd_dpcm = NULL; + /* + * The following loop will be called only for playback stream + * In this platform, there is only one playback device on every SSP + */ + for_each_dpcm_fe(rtd, SNDRV_PCM_STREAM_PLAYBACK, dpcm) { + rtd_dpcm = dpcm; + break; + } + + /* + * This following loop will be called only for capture stream + * In this platform, there is only one capture device on every SSP + */ + for_each_dpcm_fe(rtd, SNDRV_PCM_STREAM_CAPTURE, dpcm) { + rtd_dpcm = dpcm; + break; + } + + if (!rtd_dpcm) + return -EINVAL; + + /* + * The above 2 loops are mutually exclusive based on the stream direction, + * thus rtd_dpcm variable will never be overwritten + */ /* * Topology for kblda7219m98373 & kblmax98373 supports only S24_LE, * where as kblda7219m98927 & kblmax98927 supports S16_LE by default. @@ -311,9 +333,9 @@ static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd, /* * The ADSP will convert the FE rate to 48k, stereo, 24 bit */ - if (!strcmp(fe_dai_link->name, "Kbl Audio Port") || - !strcmp(fe_dai_link->name, "Kbl Audio Headset Playback") || - !strcmp(fe_dai_link->name, "Kbl Audio Capture Port")) { + if (!strcmp(rtd_dpcm->fe->dai_link->name, "Kbl Audio Port") || + !strcmp(rtd_dpcm->fe->dai_link->name, "Kbl Audio Headset Playback") || + !strcmp(rtd_dpcm->fe->dai_link->name, "Kbl Audio Capture Port")) { rate->min = rate->max = 48000; chan->min = chan->max = 2; snd_mask_none(fmt); @@ -324,7 +346,7 @@ static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd, * The speaker on the SSP0 supports S16_LE and not S24_LE. * thus changing the mask here */ - if (!strcmp(be_dai_link->name, "SSP0-Codec")) + if (!strcmp(rtd_dpcm->be->dai_link->name, "SSP0-Codec")) snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S16_LE); return 0; -- 2.25.1

3 2

[PATCH 0/5] ASoC: simple-card/audio-graph: adjust to multi CPU/Codec
by Kuninori Morimoto 15 Apr '21

15 Apr '21

Hi Mark These patches adjusts to multi CPU/Codec on simple-card / audio-graph. This is part of prepare for new audio-graph-card2. Kuninori Morimoto (5): ASoC: simple-card: remove unused variable from simple_parse_of() ASoC: simple-card: use asoc_link_to_xxx() macro ASoC: simple-card: use simple_props_to_xxx() macro ASoC: audio-graph: use asoc_link_to_xxx() macro ASoC: audio-graph: use simple_props_to_xxx() macro sound/soc/generic/audio-graph-card.c | 43 +++++++++++++----------- sound/soc/generic/simple-card.c | 50 +++++++++++++--------------- 2 files changed, 47 insertions(+), 46 deletions(-) -- 2.25.1

2 6

[PATCH] ASoC: codecs: wcd934x: Fix missing IRQF_ONESHOT as only threaded handler
by zhuguangqing83＠gmail.com 15 Apr '21

15 Apr '21

From: Guangqing Zhu <zhuguangqing83(a)gmail.com> Coccinelle noticed: sound/soc/codecs/wcd934x.c:5041:7-32: ERROR: Threaded IRQ with no primary handler requested without IRQF_ONESHOT Signed-off-by: Guangqing Zhu <zhuguangqing83(a)gmail.com> --- sound/soc/codecs/wcd934x.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wcd934x.c b/sound/soc/codecs/wcd934x.c index 5fe403307b72..cddc49bbb7f6 100644 --- a/sound/soc/codecs/wcd934x.c +++ b/sound/soc/codecs/wcd934x.c @@ -5040,7 +5040,7 @@ static int wcd934x_codec_probe(struct platform_device *pdev) ret = devm_request_threaded_irq(dev, irq, NULL, wcd934x_slim_irq_handler, - IRQF_TRIGGER_RISING, + IRQF_TRIGGER_RISING | IRQF_ONESHOT, "slim", wcd); if (ret) { dev_err(dev, "Failed to request slimbus irq\n"); -- 2.17.1

3 2

[PATCH] ASoC: SOF: use current DAI config during resume
by Ranjani Sridharan 15 Apr '21

15 Apr '21

Recently, the sof_pcm_dai_link_fixup() function was updated to match SSP config with the PCM hw_params and set the current_config for the DAI widget. But the sof_restore_pipelines() function still chooses the default config for the DAI widget upon resuming. Fix this to use the last used config when setting up the DAI widget during resume. Fixes: a1e1f10a6ed6 ("ASoC: SOF: match SSP config with pcm hw params") Reviewed-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Ranjani Sridharan <ranjani.sridharan(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 0d0d47dc0246..510883cd9107 100644 --- a/sound/soc/sof/sof-audio.c +++ b/sound/soc/sof/sof-audio.c @@ -267,7 +267,7 @@ int sof_restore_pipelines(struct device *dev) /* restore dai links */ list_for_each_entry_reverse(dai, &sdev->dai_list, list) { struct sof_ipc_reply reply; - struct sof_ipc_dai_config *config = dai->dai_config; + struct sof_ipc_dai_config *config = &dai->dai_config[dai->current_config]; if (!config) { dev_err(dev, "error: no config for DAI %s\n", -- 2.25.1

3 4

[PATCH 00/14] ASoC: simple-card-utils: prepare for multi support
by Kuninori Morimoto 15 Apr '21

15 Apr '21

Hi Mark I want to add new audio-graph-card2 driver which can support not only DPCM, but also Multi-CPU/Codec, and Codec2Codec. And it is also supporting audio-graph-card2 base custom driver. But before supporting such driver, we need to cleanup existing simple-card / audio-graph, because these and new driver are sharing code. This patch-set are for Multi-CPU/Codec support, and some cleanups. Kuninori Morimoto (14): ASoC: simple-card-utils: enable flexible CPU/Codec/Platform ASoC: simple-card-utils: share dummy DAI and reduce memory ASoC: simple-card-utils: setup dai_props cpu_dai/codec_dai at initial timing ASoC: simple-card-utils: remove li->dais/li->conf ASoC: simple-card-utils: use for_each_prop_xxx() ASoC: simple-card-utils: remove asoc_simple_parse_xxx() ASoC: simple-card-utils: care multi DAI at asoc_simple_clean_reference() ASoC: simple-card-utils: indicate dai_fmt if exist ASoC: simple-card-utils: indicate missing CPU/Codec numbers for debug ASoC: simple-card-utils: add simple_props_to_xxx() macro ASoC: simple-card-utils: multi support at asoc_simple_canonicalize_cpu/platform() ASoC: simple-card-utils: tidyup debug info for clock ASoC: simple-card-utils: tidyup dev_dbg() to use 1 line ASoC: simple-card-utils: tidyup asoc_simple_parse_convert() include/sound/simple_card_utils.h | 116 +++++++---- sound/soc/generic/audio-graph-card.c | 90 ++++----- sound/soc/generic/simple-card-utils.c | 272 +++++++++++++++++--------- sound/soc/generic/simple-card.c | 104 +++++----- 4 files changed, 352 insertions(+), 230 deletions(-) -- 2.25.1

3 20

[RFC PATCH 0/2] ASoC: remove cppchecks warnings on lm49453 and da732x
by Pierre-Louis Bossart 15 Apr '21

15 Apr '21

There are the last two patches in the cleanups, this time I am not sure what the code does and what the proper fix might be. Feedback welcome. Pierre-Louis Bossart (2): ASoC: lm49453: fix useless assignment before return ASoC: da732x: simplify code sound/soc/codecs/da732x.c | 17 ++++++----------- sound/soc/codecs/da732x.h | 12 ++++-------- sound/soc/codecs/lm49453.c | 2 -- 3 files changed, 10 insertions(+), 21 deletions(-) -- 2.25.1

3 6

[PATCH 4.14 30/68] ASoC: sunxi: sun4i-codec: fill ASoC card owner
by Greg Kroah-Hartman 15 Apr '21

15 Apr '21

From: Bastian Germann <bage(a)linutronix.de> [ Upstream commit 7c0d6e482062eb5c06ecccfab340abc523bdca00 ] card->owner is a required property and since commit 81033c6b584b ("ALSA: core: Warn on empty module") a warning is issued if it is empty. Add it. This fixes following warning observed on Lamobo R1: WARNING: CPU: 1 PID: 190 at sound/core/init.c:207 snd_card_new+0x430/0x480 [snd] Modules linked in: sun4i_codec(E+) sun4i_backend(E+) snd_soc_core(E) ... CPU: 1 PID: 190 Comm: systemd-udevd Tainted: G C E 5.10.0-1-armmp #1 Debian 5.10.4-1 Hardware name: Allwinner sun7i (A20) Family Call trace: (snd_card_new [snd]) (snd_soc_bind_card [snd_soc_core]) (snd_soc_register_card [snd_soc_core]) (sun4i_codec_probe [sun4i_codec]) Fixes: 45fb6b6f2aa3 ("ASoC: sunxi: add support for the on-chip codec on early Allwinner SoCs") Related: commit 3c27ea23ffb4 ("ASoC: qcom: Set card->owner to avoid warnings") Related: commit ec653df2a0cb ("drm/vc4/vc4_hdmi: fill ASoC card owner") Cc: linux-arm-kernel(a)lists.infradead.org Cc: alsa-devel(a)alsa-project.org Signed-off-by: Bastian Germann <bage(a)linutronix.de> Link: https://lore.kernel.org/r/20210331151843.30583-1-bage@linutronix.de Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/sunxi/sun4i-codec.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/soc/sunxi/sun4i-codec.c b/sound/soc/sunxi/sun4i-codec.c index baa9007464ed..700779ca82d0 100644 --- a/sound/soc/sunxi/sun4i-codec.c +++ b/sound/soc/sunxi/sun4i-codec.c @@ -1199,6 +1199,7 @@ static struct snd_soc_card *sun4i_codec_create_card(struct device *dev) return ERR_PTR(-ENOMEM); card->dev = dev; + card->owner = THIS_MODULE; card->name = "sun4i-codec"; card->dapm_widgets = sun4i_codec_card_dapm_widgets; card->num_dapm_widgets = ARRAY_SIZE(sun4i_codec_card_dapm_widgets); @@ -1231,6 +1232,7 @@ static struct snd_soc_card *sun6i_codec_create_card(struct device *dev) return ERR_PTR(-ENOMEM); card->dev = dev; + card->owner = THIS_MODULE; card->name = "A31 Audio Codec"; card->dapm_widgets = sun6i_codec_card_dapm_widgets; card->num_dapm_widgets = ARRAY_SIZE(sun6i_codec_card_dapm_widgets); @@ -1284,6 +1286,7 @@ static struct snd_soc_card *sun8i_a23_codec_create_card(struct device *dev) return ERR_PTR(-ENOMEM); card->dev = dev; + card->owner = THIS_MODULE; card->name = "A23 Audio Codec"; card->dapm_widgets = sun6i_codec_card_dapm_widgets; card->num_dapm_widgets = ARRAY_SIZE(sun6i_codec_card_dapm_widgets); @@ -1322,6 +1325,7 @@ static struct snd_soc_card *sun8i_h3_codec_create_card(struct device *dev) return ERR_PTR(-ENOMEM); card->dev = dev; + card->owner = THIS_MODULE; card->name = "H3 Audio Codec"; card->dapm_widgets = sun6i_codec_card_dapm_widgets; card->num_dapm_widgets = ARRAY_SIZE(sun6i_codec_card_dapm_widgets); @@ -1360,6 +1364,7 @@ static struct snd_soc_card *sun8i_v3s_codec_create_card(struct device *dev) return ERR_PTR(-ENOMEM); card->dev = dev; + card->owner = THIS_MODULE; card->name = "V3s Audio Codec"; card->dapm_widgets = sun6i_codec_card_dapm_widgets; card->num_dapm_widgets = ARRAY_SIZE(sun6i_codec_card_dapm_widgets); -- 2.30.2

1 0

[Problem] A data race in snd_ctl_elem_add()
by Gong, Sishuai 15 Apr '21

15 Apr '21

Hi, We found a data race in sound/core/control.c in linux-5.12-rc3 and we are able to reproduce it under x86. In general, we found when 2 kernel threads are both running snd_ctl_elem_add(), one may read a stale value of card->user_ctl_count, as shown below. Currently, we haven’t found any explicit errors due to this data race, but it looks the reader threads may operate in an inconsistent state, where card->user_ctl_count + 1 is actually bigger than MAX_USER_CONTROLS, so we want to point it out. Thread 1 Thread 2 //snd_ctl_elem_add() //snd_ctl_elem_add() if (card->user_ctl_count + 1 > MAX_USER_CONTROLS) return -ENOMEM; card->user_ctl_count++; unlock: up_write(&card->controls_rwsem); return err; Thanks, Sishuai

2 2

SOF SoundWire: Improve dmic detection
by GitHub pull_request - edited 15 Apr '21

15 Apr '21

alsa-project/alsa-ucm-conf pull request #75 was edited from plbossart: DMIC support seems incomplete for HDaudio (not handling all possibilities for cfg-dmics) and missing completely for SoundWire. This is an untested draft just to get UCM3 feedback from @perexg and tests from @aigilea on an HP Spectre x360 device. Edit: adding @kv2019i @RanderWang @bardliao and @libinyang for reviews. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/75 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/75.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH] ALSA: control: Fix racy management of user ctl memory size account
by Takashi Iwai 15 Apr '21

15 Apr '21

We've got a report about the possible race in the user control element counts (card->user_ctl_count), and it was confirmed that the race wasn't serious in the old code up to 5.12. There, the value modification itself was exclusive and protected via a write semaphore, hence it's at most concurrent reads and evaluations before the increment. Since it's only about the soft-limit to avoid the exhausting memory usage, one-off isn't a big problem at all. Meanwhile, the relevant code has been largely modified recently, and now card->user_ctl_count was replaced with card->user_ctl_alloc_size, and a few more places were added to access this field. And, in this new code, it turned out to be more serious: the modifications are scattered in various places, and a few of them are without protection. It implies that it may lead to an inconsistent value by racy accesses. For addressing it, this patch extends the range covered by the card->controls_rwsem write lock at snd_ctl_elem_add() so that the all code paths that modify and refer to card->user_ctl_alloc_size are protected by the rwsem properly. The patch adds also comments in a couple of functions to indicate that they are under the rwsem lock. Fixes: 66c6d1ef86ff ("ALSA: control: Add memory consumption limit to user controls") Link: https://lore.kernel.org/r/FEEBF384-44BE-42CF-8FB3-93470933F64F@purdue.edu Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/control.c | 17 +++++++++++------ 1 file changed, 11 insertions(+), 6 deletions(-) diff --git a/sound/core/control.c b/sound/core/control.c index a076c08c21b6..498e3701514a 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -1337,6 +1337,7 @@ static int snd_ctl_elem_user_put(struct snd_kcontrol *kcontrol, return change; } +/* called in controls_rwsem write lock */ static int replace_user_tlv(struct snd_kcontrol *kctl, unsigned int __user *buf, unsigned int size) { @@ -1414,6 +1415,7 @@ static int snd_ctl_elem_user_tlv(struct snd_kcontrol *kctl, int op_flag, return read_user_tlv(kctl, buf, size); } +/* called in controls_rwsem write lock */ static int snd_ctl_elem_init_enum_names(struct user_element *ue) { char *names, *p; @@ -1529,8 +1531,11 @@ static int snd_ctl_elem_add(struct snd_ctl_file *file, private_size = value_sizes[info->type] * info->count; alloc_size = compute_user_elem_size(private_size, count); - if (check_user_elem_overflow(card, alloc_size)) - return -ENOMEM; + down_write(&card->controls_rwsem); + if (check_user_elem_overflow(card, alloc_size)) { + err = -ENOMEM; + goto unlock; + } /* * Keep memory object for this userspace control. After passing this @@ -1540,12 +1545,13 @@ static int snd_ctl_elem_add(struct snd_ctl_file *file, */ err = snd_ctl_new(&kctl, count, access, file); if (err < 0) - return err; + goto unlock; memcpy(&kctl->id, &info->id, sizeof(kctl->id)); ue = kzalloc(alloc_size, GFP_KERNEL); if (!ue) { kfree(kctl); - return -ENOMEM; + err = -ENOMEM; + goto unlock; } kctl->private_data = ue; kctl->private_free = snd_ctl_elem_user_free; @@ -1563,7 +1569,7 @@ static int snd_ctl_elem_add(struct snd_ctl_file *file, err = snd_ctl_elem_init_enum_names(ue); if (err < 0) { snd_ctl_free_one(kctl); - return err; + goto unlock; } } @@ -1580,7 +1586,6 @@ static int snd_ctl_elem_add(struct snd_ctl_file *file, kctl->tlv.c = snd_ctl_elem_user_tlv; /* This function manage to free the instance on failure. */ - down_write(&card->controls_rwsem); err = __snd_ctl_add_replace(card, kctl, CTL_ADD_EXCLUSIVE); if (err < 0) { snd_ctl_free_one(kctl); -- 2.26.2

1 0

[PATCH] ASoC: Intel: Boards: cml_da7219_max98390: set TDM format configuration
by mac.chiang＠intel.com 15 Apr '21

15 Apr '21

From: Mac Chiang <mac.chiang(a)intel.com> add tx/rx 2 channels, 32bits support Signed-off-by: Steve Lee <SteveS.Lee(a)maximintegrated.com> Signed-off-by: Mac Chiang <mac.chiang(a)intel.com> --- sound/soc/intel/boards/bxt_da7219_max98357a.c | 25 +++++++++++++++++++++++++ 1 file changed, 25 insertions(+) diff --git a/sound/soc/intel/boards/bxt_da7219_max98357a.c b/sound/soc/intel/boards/bxt_da7219_max98357a.c index 9ffef39..515379cd 100644 --- a/sound/soc/intel/boards/bxt_da7219_max98357a.c +++ b/sound/soc/intel/boards/bxt_da7219_max98357a.c @@ -253,6 +253,30 @@ static int broxton_da7219_codec_init(struct snd_soc_pcm_runtime *rtd) return ret; } +static int max98390_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai; + int j; + + for_each_rtd_codec_dais(rtd, j, codec_dai) { + if (!strcmp(codec_dai->component->name, MAX98390_DEV0_NAME)) { + /* DEV0 tdm slot configuration */ + snd_soc_dai_set_tdm_slot(codec_dai, 0x1, 3, 2, 32); + } + if (!strcmp(codec_dai->component->name, MAX98390_DEV1_NAME)) { + /* DEV1 tdm slot configuration */ + snd_soc_dai_set_tdm_slot(codec_dai, 0x2, 3, 2, 32); + } + } + return 0; +} + +struct snd_soc_ops max_98390_ops = { + .hw_params = max98390_hw_params, +}; + static int broxton_hdmi_init(struct snd_soc_pcm_runtime *rtd) { struct bxt_card_private *ctx = snd_soc_card_get_drvdata(rtd->card); @@ -813,6 +837,7 @@ static int broxton_audio_probe(struct platform_device *pdev) if (ctx->spkamp == SPKAMP_MAX98390) { broxton_dais[i].codecs = max98390_codec; broxton_dais[i].num_codecs = ARRAY_SIZE(max98390_codec); + broxton_dais[i].ops = &max_98390_ops; broxton_dais[i].dpcm_capture = 1; } } -- 2.7.4

2 1

Re: [2/2]ALSA: hda: add Zhaoxin HDAC non-snoop support: patch a design limitation
by Takashi Iwai 15 Apr '21

15 Apr '21

On Wed, 14 Apr 2021 14:00:23 +0200, Hans Hu(SH-RD) wrote: > > Hi Takashi, > > ZHAOXIN HDAC controller has one design limitation when it works in non-snoop mode. This design limitation is: hardware can't guarantee that the write CORB cycle always completes first before the write CORBWP cycle. Here is the error scene: > int snd_hdac_bus_send_cmd(struct hdac_bus *bus, unsigned int val) > { > ... > bus->corb.buf[wp] = cpu_to_le32(val); // cycle_1: write value to CORB > snd_hdac_chip_writew(bus, CORBWP, wp); // cycle_2: write wp to CORBWP > ... > } > Normally, after cycle_2, DMA engine will start working and get data from CORB and sent it out. But if cycle_1 haven’t finished yet at this time(limitation occurs), which means the value haven’t been written into CORB, then DMA engine will get unexpected value, then error occurred. (feels like one kind of CORB underrun). > > If we add one read CORB cycle between cycle_1 and cycle_2, cycle_1 and cycle_2 operations will be synchronized, this limitation will be fixed. We have written a draft patch based on this situation and hope to be accepted, the patch example is as follows: > diff --git a/include/sound/hdaudio.h b/include/sound/hdaudio.h > index 22af68b..c338db00 100644 > --- a/include/sound/hdaudio.h > +++ b/include/sound/hdaudio.h > @@ -339,6 +339,7 @@ struct hdac_bus { > bool sync_write:1; /* sync after verb write */ > bool use_posbuf:1; /* use position buffer */ > bool snoop:1; /* enable snooping */ > + bool no_snoop_corb_sync:1; > bool align_bdle_4k:1; /* BDLE align 4K boundary */ > bool reverse_assign:1; /* assign devices in reverse order */ > bool corbrp_self_clear:1; /* CORBRP clears itself after reset */ > diff --git a/sound/hda/hdac_controller.c b/sound/hda/hdac_controller.c > index 062da7a..6c90cdd 100644 > --- a/sound/hda/hdac_controller.c > +++ b/sound/hda/hdac_controller.c > @@ -167,6 +167,8 @@ int snd_hdac_bus_send_cmd(struct hdac_bus *bus, unsigned int val) > > bus->rirb.cmds[addr]++; > bus->corb.buf[wp] = cpu_to_le32(val); > + if (bus->no_snoop_corb_sync) > + val = bus->corb.buf[wp]; > snd_hdac_chip_writew(bus, CORBWP, wp); Just wondering whether using WRITE_ONCE() macro would be enough? e.g. WRITE_ONCE(bus->corb.buf[wp], cpu_to_le32(val)); Also, no_snoop_corb_sync is a bit confusing. What you do is rather sync of the written value, so it can be taken as if other way round. Maybe no_coherent_corb_write or such would be better understandable. thanks, Takashi

2 2

[PATCH] ALSA: usb-audio: Explicitly set up the clock selector
by Takashi Iwai 14 Apr '21

14 Apr '21

In the current code, we have some assumption that the audio clock selector has been set up implicitly and don't want to touch it unless it's really needed for the fallback autoclock setup. This works for most devices but some seem having a problem. Partially this was covered for the devices with a single connector at the initialization phase (commit 086b957cc17f "ALSA: usb-audio: Skip the clock selector inquiry for single connections"), but also there are cases where the wrong clock set up is kept silently. The latter seems to be the cause of the noises on Behringer devices. In this patch, we explicitly set up the audio clock selector whenever the appropriate node is found. Reported-by: Geraldo Nascimento <geraldogabriel(a)gmail.com> BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=199327 Link: https://lore.kernel.org/r/CAEsQvcvF7LnO8PxyyCxuRCx=7jNeSCvFAd-+dE0g_rd1rOxx… Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/clock.c | 18 ++++++++++++++---- 1 file changed, 14 insertions(+), 4 deletions(-) diff --git a/sound/usb/clock.c b/sound/usb/clock.c index a746802d0ac3..17bbde73d4d1 100644 --- a/sound/usb/clock.c +++ b/sound/usb/clock.c @@ -296,7 +296,7 @@ static int __uac_clock_find_source(struct snd_usb_audio *chip, selector = snd_usb_find_clock_selector(chip->ctrl_intf, entity_id); if (selector) { - int ret, i, cur; + int ret, i, cur, err; if (selector->bNrInPins == 1) { ret = 1; @@ -324,13 +324,17 @@ static int __uac_clock_find_source(struct snd_usb_audio *chip, ret = __uac_clock_find_source(chip, fmt, selector->baCSourceID[ret - 1], visited, validate); + if (ret > 0) { + err = uac_clock_selector_set_val(chip, entity_id, cur); + if (err < 0) + return err; + } + if (!validate || ret > 0 || !chip->autoclock) return ret; /* The current clock source is invalid, try others. */ for (i = 1; i <= selector->bNrInPins; i++) { - int err; - if (i == cur) continue; @@ -396,7 +400,7 @@ static int __uac3_clock_find_source(struct snd_usb_audio *chip, selector = snd_usb_find_clock_selector_v3(chip->ctrl_intf, entity_id); if (selector) { - int ret, i, cur; + int ret, i, cur, err; /* the entity ID we are looking for is a selector. * find out what it currently selects */ @@ -418,6 +422,12 @@ static int __uac3_clock_find_source(struct snd_usb_audio *chip, ret = __uac3_clock_find_source(chip, fmt, selector->baCSourceID[ret - 1], visited, validate); + if (ret > 0) { + err = uac_clock_selector_set_val(chip, entity_id, cur); + if (err < 0) + return err; + } + if (!validate || ret > 0 || !chip->autoclock) return ret; -- 2.26.2

2 1

[PATCH v2 -next 1/2] ASoC: ak5558: correct reset polarity
by Shengjiu Wang 14 Apr '21

14 Apr '21

Reset (aka power off) happens when the reset gpio is made active. The reset gpio is GPIO_ACTIVE_LOW Fixes: 920884777480 ("ASoC: ak5558: Add support for AK5558 ADC driver") Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- changes in v2: - split the patch to two patches. sound/soc/codecs/ak5558.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/ak5558.c b/sound/soc/codecs/ak5558.c index 8e4dca753f0b..f24d91b728dd 100644 --- a/sound/soc/codecs/ak5558.c +++ b/sound/soc/codecs/ak5558.c @@ -323,7 +323,7 @@ static void ak5558_power_off(struct ak5558_priv *ak5558) if (!ak5558->reset_gpiod) return; - gpiod_set_value_cansleep(ak5558->reset_gpiod, 0); + gpiod_set_value_cansleep(ak5558->reset_gpiod, 1); usleep_range(1000, 2000); } @@ -332,7 +332,7 @@ static void ak5558_power_on(struct ak5558_priv *ak5558) if (!ak5558->reset_gpiod) return; - gpiod_set_value_cansleep(ak5558->reset_gpiod, 1); + gpiod_set_value_cansleep(ak5558->reset_gpiod, 0); usleep_range(1000, 2000); } -- 2.27.0

2 2

[PATCH] ASoC: cs35l35: remove unused including <linux/version.h>
by Yang Li 14 Apr '21

14 Apr '21

Fix the following versioncheck warning: ./sound/soc/codecs/cs35l35.c: 12 linux/version.h not needed. Reported-by: Abaci Robot <abaci(a)linux.alibaba.com> Signed-off-by: Yang Li <yang.lee(a)linux.alibaba.com> --- sound/soc/codecs/cs35l35.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/cs35l35.c b/sound/soc/codecs/cs35l35.c index 55d529a..5d361c7 100644 --- a/sound/soc/codecs/cs35l35.c +++ b/sound/soc/codecs/cs35l35.c @@ -9,7 +9,6 @@ #include <linux/module.h> #include <linux/moduleparam.h> -#include <linux/version.h> #include <linux/kernel.h> #include <linux/init.h> #include <linux/delay.h> -- 1.8.3.1

2 1

[PATCH 0/5] ASoC: rsnd: tidyup Renesas sound
by Kuninori Morimoto 14 Apr '21

14 Apr '21

Hi Mark These are tidyup patches for Renesas sound drivers. Kuninori Morimoto (5): ASoC: rsnd: call rsnd_ssi_master_clk_start() from rsnd_ssi_init() ASoC: rsnd: check all BUSIF status when error ASoC: rsnd: add rsnd_ssi_busif_err_status_clear() ASoC: rsnd: add rsnd_ssi_busif_err_irq_enable/disable() ASoC: rsnd: add usage for SRC sound/soc/sh/rcar/src.c | 9 ++ sound/soc/sh/rcar/ssi.c | 225 ++++++++++++++++++---------------------- 2 files changed, 110 insertions(+), 124 deletions(-) -- 2.25.1

2 6

[PATCH v2 0/3] soundwire: add DMI quirks for overridind addr
by Bard Liao 14 Apr '21

14 Apr '21

Platform firmware may have incorrect _ADR values causing the driver probes to fail. Adding the override_ops allows people to override the addr values. v2: - Add the override_adr ops - Move DMI quirks to a new file Pierre-Louis Bossart (2): soundwire: Intel: introduce DMI quirks for HP Spectre x360 Convertible soundwire: Intel: add DMI quirk for Dell SKU 0A3E Vinod Koul (1): soundwire: add override addr ops drivers/soundwire/Makefile | 2 +- drivers/soundwire/bus.h | 2 + drivers/soundwire/dmi-quirks.c | 96 ++++++++++++++++++++++++++++++++++ drivers/soundwire/intel.c | 1 + drivers/soundwire/slave.c | 8 ++- include/linux/soundwire/sdw.h | 4 +- 6 files changed, 110 insertions(+), 3 deletions(-) create mode 100644 drivers/soundwire/dmi-quirks.c -- 2.17.1

4 7

[PATCH] ASoC: codec: remove unused variable
by Jiapeng Chong 14 Apr '21

14 Apr '21

Fix the following gcc warning: sound/soc/codecs/jz4760.c:201:6: warning: variable ‘ret’ set but not used [-Wunused-but-set-variable]. Reported-by: Abaci Robot <abaci(a)linux.alibaba.com> Signed-off-by: Jiapeng Chong <jiapeng.chong(a)linux.alibaba.com> --- sound/soc/codecs/jz4760.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/codecs/jz4760.c b/sound/soc/codecs/jz4760.c index e8f28cc..ad5e859 100644 --- a/sound/soc/codecs/jz4760.c +++ b/sound/soc/codecs/jz4760.c @@ -198,15 +198,13 @@ static int jz4760_codec_startup(struct snd_pcm_substream *substream, { struct snd_soc_component *codec = dai->component; struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(codec); - int ret; - /* * SYSCLK output from the codec to the AIC is required to keep the * DMA transfer going during playback when all audible outputs have * been disabled. */ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - ret = snd_soc_dapm_force_enable_pin(dapm, "SYSCLK"); + snd_soc_dapm_force_enable_pin(dapm, "SYSCLK"); return 0; } -- 1.8.3.1

2 1

[PATCH] ALSA: control_led - fix the stack usage (control element ops)
by Jaroslav Kysela 14 Apr '21

14 Apr '21

It's a bad idea to allocate big structures on the stack. Mark the variables as static and add a note for the locking. Fixes: 22d8de62f11b ("ALSA: control - add generic LED trigger module as the new control layer") Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> Cc: Nathan Chancellor <nathan(a)kernel.org> Cc: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> --- sound/core/control_led.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/core/control_led.c b/sound/core/control_led.c index 93b201063c7d..25f57c14f294 100644 --- a/sound/core/control_led.c +++ b/sound/core/control_led.c @@ -94,11 +94,15 @@ static struct snd_ctl_led *snd_ctl_led_get_by_access(unsigned int access) return &snd_ctl_leds[group]; } +/* + * A note for callers: + * The two static variables info and value are protected using snd_ctl_led_mutex. + */ static int snd_ctl_led_get(struct snd_ctl_led_ctl *lctl) { + static struct snd_ctl_elem_info info; + static struct snd_ctl_elem_value value; struct snd_kcontrol *kctl = lctl->kctl; - struct snd_ctl_elem_info info; - struct snd_ctl_elem_value value; unsigned int i; int result; -- 2.30.2

2 2

Re: [1/2]ALSA: hda: add Zhaoxin HDAC non-snoop support
by Takashi Iwai 14 Apr '21

14 Apr '21

On Wed, 14 Apr 2021 14:00:19 +0200, Hans Hu(SH-RD) wrote: > > Hi Takashi, > > Recently, our(AZX_DRIVER_ZHAOXIN) HDAC controller has added the feature of non-snoop mode, we have written a draft patch for it and hope to be accepted. The code change looks good. Please go ahead with the proper submission. thanks, Takashi > > diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index b59b0f3..420b728 100644 > --- a/sound/pci/hda/hda_intel.c > +++ b/sound/pci/hda/hda_intel.c > @@ -1702,6 +1702,14 @@ static void azx_check_snoop_available(struct azx *chip) > snoop = false; > } > > +if (azx_get_snoop_type(chip) == AZX_SNOOP_TYPE_NONE && > + chip->driver_type == AZX_DRIVER_ZHAOXIN) { > +u8 val; > +pci_read_config_byte(chip->pci, 0x42, &val); > +if (!(val & 0x80)) > +snoop = false; > +} > + > if (chip->driver_caps & AZX_DCAPS_SNOOP_OFF) > snoop = false; > > > Thanks, > 　Hans Hu :) > > > 保密声明： > 本邮件含有保密或专有信息，仅供指定收件人使用。严禁对本邮件或其内容做任何未经授权的查阅、使用、复制或转发。 > CONFIDENTIAL NOTE: > This email contains confidential or legally privileged information and is for the sole use of its intended recipient. Any unauthorized review, use, copying or forwarding of this email or the content of this email is strictly prohibited.

1 0

[PATCH] ALSA: control_led - fix the stack usage (control element ops)
by Jaroslav Kysela 14 Apr '21

14 Apr '21

It's a bad idea to allocate big structures on the stack. Allocate the required structures on demand and cache them in the led structure. Fixes: 22d8de62f11b ("ALSA: control - add generic LED trigger module as the new control layer") Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> Cc: Nathan Chancellor <nathan(a)kernel.org> Cc: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/core/control_led.c | 52 ++++++++++++++++++++++++---------------- 1 file changed, 32 insertions(+), 20 deletions(-) diff --git a/sound/core/control_led.c b/sound/core/control_led.c index 93b201063c7d..9d1612b1a6ff 100644 --- a/sound/core/control_led.c +++ b/sound/core/control_led.c @@ -37,6 +37,7 @@ struct snd_ctl_led { unsigned int group; enum led_audio trigger_type; enum snd_ctl_led_mode mode; + void *info_and_value; struct snd_ctl_led_card *cards[SNDRV_CARDS]; }; @@ -94,34 +95,41 @@ static struct snd_ctl_led *snd_ctl_led_get_by_access(unsigned int access) return &snd_ctl_leds[group]; } -static int snd_ctl_led_get(struct snd_ctl_led_ctl *lctl) +static int snd_ctl_led_get(struct snd_ctl_led *led, struct snd_ctl_led_ctl *lctl) { struct snd_kcontrol *kctl = lctl->kctl; - struct snd_ctl_elem_info info; - struct snd_ctl_elem_value value; + struct snd_ctl_elem_info *info = led->info_and_value; + struct snd_ctl_elem_value *value; unsigned int i; int result; - memset(&info, 0, sizeof(info)); - info.id = kctl->id; - info.id.index += lctl->index_offset; - info.id.numid += lctl->index_offset; - result = kctl->info(kctl, &info); + if (info == NULL) { + info = kmalloc(sizeof(*info) + sizeof(*value), GFP_KERNEL); + if (info == NULL) + return -1; + led->info_and_value = info; + } + value = (void *)(info + 1); + memset(info, 0, sizeof(*info)); + info->id = kctl->id; + info->id.index += lctl->index_offset; + info->id.numid += lctl->index_offset; + result = kctl->info(kctl, info); if (result < 0) return -1; - memset(&value, 0, sizeof(value)); - value.id = info.id; - result = kctl->get(kctl, &value); + memset(value, 0, sizeof(*value)); + value->id = info->id; + result = kctl->get(kctl, value); if (result < 0) return -1; - if (info.type == SNDRV_CTL_ELEM_TYPE_BOOLEAN || - info.type == SNDRV_CTL_ELEM_TYPE_INTEGER) { - for (i = 0; i < info.count; i++) - if (value.value.integer.value[i] != info.value.integer.min) + if (info->type == SNDRV_CTL_ELEM_TYPE_BOOLEAN || + info->type == SNDRV_CTL_ELEM_TYPE_INTEGER) { + for (i = 0; i < info->count; i++) + if (value->value.integer.value[i] != info->value.integer.min) return 1; - } else if (info.type == SNDRV_CTL_ELEM_TYPE_INTEGER64) { - for (i = 0; i < info.count; i++) - if (value.value.integer64.value[i] != info.value.integer64.min) + } else if (info->type == SNDRV_CTL_ELEM_TYPE_INTEGER64) { + for (i = 0; i < info->count; i++) + if (value->value.integer64.value[i] != info->value.integer64.min) return 1; } return 0; @@ -149,7 +157,7 @@ static void snd_ctl_led_set_state(struct snd_card *card, unsigned int access, list_for_each_entry(lctl, &led->controls, list) { if (lctl->kctl == kctl && lctl->index_offset == ioff) found = true; - UPDATE_ROUTE(route, snd_ctl_led_get(lctl)); + UPDATE_ROUTE(route, snd_ctl_led_get(led, lctl)); } if (!found && kctl && card) { lctl = kzalloc(sizeof(*lctl), GFP_KERNEL); @@ -159,7 +167,7 @@ static void snd_ctl_led_set_state(struct snd_card *card, unsigned int access, lctl->kctl = kctl; lctl->index_offset = ioff; list_add(&lctl->list, &led->controls); - UPDATE_ROUTE(route, snd_ctl_led_get(lctl)); + UPDATE_ROUTE(route, snd_ctl_led_get(led, lctl)); } } mutex_unlock(&snd_ctl_led_mutex); @@ -300,6 +308,10 @@ static void snd_ctl_led_clean(struct snd_card *card) snd_ctl_led_ctl_destroy(lctl); goto repeat; } + if (!card) { + kfree(led->info_and_value); + led->info_and_value = NULL; + } } } -- 2.30.2

2 4

[PATCH] ALSA: control_led: fix stack frame usage over 1024 bytes in snd_ctl_led_get()
by Takashi Sakamoto 14 Apr '21

14 Apr '21

GCC warns that snd_ctl_led_get() uses stack frame over 1024 bytes. control_led.c: In function ‘snd_ctl_led_get’: control_led.c:128:1: warning: the frame size of 1504 bytes is larger than 1024 bytes [-Wframe-larger-than=] When taking care of convension in Linux kernel development. it's preferable to suppress too large usage of kernel stack, when the function call is not in critical part. This commit fixes the bug, including some minor code refactoring relevant to variable names. Fixes: 22d8de62f11b ("ALSA: control - add generic LED trigger module as the new control layer") Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/core/control_led.c | 68 ++++++++++++++++++++++++++-------------- 1 file changed, 44 insertions(+), 24 deletions(-) diff --git a/sound/core/control_led.c b/sound/core/control_led.c index b97f118cd54e..1be854a861f0 100644 --- a/sound/core/control_led.c +++ b/sound/core/control_led.c @@ -94,34 +94,35 @@ static struct snd_ctl_led *snd_ctl_led_get_by_access(unsigned int access) return &snd_ctl_leds[group]; } -static int snd_ctl_led_get(struct snd_ctl_led_ctl *lctl) +static int snd_ctl_led_get(struct snd_ctl_led_ctl *lctl, struct snd_ctl_elem_info *elem_info, + struct snd_ctl_elem_value *elem_value) { struct snd_kcontrol *kctl = lctl->kctl; - struct snd_ctl_elem_info info; - struct snd_ctl_elem_value value; unsigned int i; - int result; - - memset(&info, 0, sizeof(info)); - info.id = kctl->id; - info.id.index += lctl->index_offset; - info.id.numid += lctl->index_offset; - result = kctl->info(kctl, &info); - if (result < 0) + int err; + + memset(elem_info, 0, sizeof(*elem_info)); + elem_info->id = kctl->id; + elem_info->id.index += lctl->index_offset; + elem_info->id.numid += lctl->index_offset; + err = kctl->info(kctl, elem_info); + if (err < 0) return -1; - memset(&value, 0, sizeof(value)); - value.id = info.id; - result = kctl->get(kctl, &value); - if (result < 0) + + memset(elem_value, 0, sizeof(*elem_value)); + elem_value->id = elem_info->id; + err = kctl->get(kctl, elem_value); + if (err < 0) return -1; - if (info.type == SNDRV_CTL_ELEM_TYPE_BOOLEAN || - info.type == SNDRV_CTL_ELEM_TYPE_INTEGER) { - for (i = 0; i < info.count; i++) - if (value.value.integer.value[i] != info.value.integer.min) + + if (elem_info->type == SNDRV_CTL_ELEM_TYPE_BOOLEAN || + elem_info->type == SNDRV_CTL_ELEM_TYPE_INTEGER) { + for (i = 0; i < elem_info->count; i++) + if (elem_value->value.integer.value[i] != elem_info->value.integer.min) return 1; - } else if (info.type == SNDRV_CTL_ELEM_TYPE_INTEGER64) { - for (i = 0; i < info.count; i++) - if (value.value.integer64.value[i] != info.value.integer64.min) + } else if (elem_info->type == SNDRV_CTL_ELEM_TYPE_INTEGER64) { + for (i = 0; i < elem_info->count; i++) + if (elem_value->value.integer64.value[i] != elem_info->value.integer64.min) return 1; } return 0; @@ -132,6 +133,8 @@ static void snd_ctl_led_set_state(struct snd_card *card, unsigned int access, { struct snd_ctl_led *led; struct snd_ctl_led_ctl *lctl; + struct snd_ctl_elem_info *elem_info; + struct snd_ctl_elem_value *elem_value; int route; bool found; @@ -146,10 +149,24 @@ static void snd_ctl_led_set_state(struct snd_card *card, unsigned int access, mutex_unlock(&snd_ctl_led_mutex); return; } + + elem_info = kmalloc(sizeof(*elem_info), GFP_KERNEL); + if (!elem_info) { + mutex_unlock(&snd_ctl_led_mutex); + return; + } + + elem_value = kmalloc(sizeof(*elem_value), GFP_KERNEL); + if (!elem_value) { + kfree(elem_info); + mutex_unlock(&snd_ctl_led_mutex); + return; + } + list_for_each_entry(lctl, &led->controls, list) { if (lctl->kctl == kctl && lctl->index_offset == ioff) found = true; - UPDATE_ROUTE(route, snd_ctl_led_get(lctl)); + UPDATE_ROUTE(route, snd_ctl_led_get(lctl, elem_info, elem_value)); } if (!found && kctl && card) { lctl = kzalloc(sizeof(*lctl), GFP_KERNEL); @@ -159,10 +176,13 @@ static void snd_ctl_led_set_state(struct snd_card *card, unsigned int access, lctl->kctl = kctl; lctl->index_offset = ioff; list_add(&lctl->list, &led->controls); - UPDATE_ROUTE(route, snd_ctl_led_get(lctl)); + UPDATE_ROUTE(route, snd_ctl_led_get(lctl, elem_info, elem_value)); } kfree(lctl); } + + kfree(elem_value); + kfree(elem_info); mutex_unlock(&snd_ctl_led_mutex); switch (led->mode) { case MODE_OFF: route = 1; break; -- 2.27.0

4 4

utils/alsa.m4: fix the brackets for real
by GitHub pull_request - opened 13 Apr '21

13 Apr '21

alsa-project/alsa-lib pull request #133 was opened from asavah: Even after https://github.com/alsa-project/alsa-lib/commit/f9ace404fde9058baf64d77eed9… building alsa-utils from git fails due to broken alsa.m4. ``` autoreconf: running: aclocal -I m4 --force -I m4 /home/asavah/kross/host/bin/m4:/home/asavah/kross/build/asusb450eg/rootfs/usr/share/aclocal/alsa.m4:20: ERROR: end of file in string autom4te2.71: error: /home/asavah/kross/host/bin/m4 failed with exit status: 1 aclocal: error: /home/asavah/kross/host/bin/autom4te2.71 failed with exit status: 1 autoreconf: error: aclocal failed with exit status: 1 ``` This PR should address missing brackets. Request URL : https://github.com/alsa-project/alsa-lib/pull/133 Patch URL : https://github.com/alsa-project/alsa-lib/pull/133.patch Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH] ASoC: cs35l35: Fix an error handling path in 'cs35l35_i2c_probe()'
by Christophe JAILLET 13 Apr '21

13 Apr '21

If 'devm_regmap_init_i2c()' fails, there is no need to goto err. We should return directly as already done by the surrounding error handling paths. Signed-off-by: Christophe JAILLET <christophe.jaillet(a)wanadoo.fr> --- sound/soc/codecs/cs35l35.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs35l35.c b/sound/soc/codecs/cs35l35.c index 55d529aa0011..7b9f5498f8a7 100644 --- a/sound/soc/codecs/cs35l35.c +++ b/sound/soc/codecs/cs35l35.c @@ -1488,7 +1488,7 @@ static int cs35l35_i2c_probe(struct i2c_client *i2c_client, if (IS_ERR(cs35l35->regmap)) { ret = PTR_ERR(cs35l35->regmap); dev_err(dev, "regmap_init() failed: %d\n", ret); - goto err; + return ret; } for (i = 0; i < ARRAY_SIZE(cs35l35_supplies); i++) -- 2.27.0

2 4

[PATCH v4] sound: rawmidi: Add framing mode
by David Henningsson 13 Apr '21

13 Apr '21

This commit adds a new framing mode that frames all MIDI data into 32-byte frames with a timestamp. The main benefit is that we can get accurate timestamps even if userspace wakeup and processing is not immediate. Testing on a Celeron N3150 with this mode has a max jitter of 2.8 ms, compared to the in-kernel seq implementation which has a max jitter of 5 ms during idle and much worse when running scheduler stress tests in parallel. Signed-off-by: David Henningsson <coding(a)diwic.se> --- include/sound/rawmidi.h | 2 ++ include/uapi/sound/asound.h | 26 ++++++++++++++-- sound/core/rawmidi.c | 60 +++++++++++++++++++++++++++++++++++-- 3 files changed, 84 insertions(+), 4 deletions(-) diff --git a/include/sound/rawmidi.h b/include/sound/rawmidi.h index 334842daa904..b0057a193c31 100644 --- a/include/sound/rawmidi.h +++ b/include/sound/rawmidi.h @@ -81,6 +81,8 @@ struct snd_rawmidi_substream { bool opened; /* open flag */ bool append; /* append flag (merge more streams) */ bool active_sensing; /* send active sensing when close */ + u8 framing; /* whether to frame input data */ + clockid_t clock_type; /* clock source to use for input framing */ int use_count; /* use counter (for output) */ size_t bytes; struct snd_rawmidi *rmidi; diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h index 535a7229e1d9..af8e60740218 100644 --- a/include/uapi/sound/asound.h +++ b/include/uapi/sound/asound.h @@ -710,7 +710,7 @@ enum { * Raw MIDI section - /dev/snd/midi?? */ -#define SNDRV_RAWMIDI_VERSION SNDRV_PROTOCOL_VERSION(2, 0, 1) +#define SNDRV_RAWMIDI_VERSION SNDRV_PROTOCOL_VERSION(2, 0, 2) enum { SNDRV_RAWMIDI_STREAM_OUTPUT = 0, @@ -736,12 +736,34 @@ struct snd_rawmidi_info { unsigned char reserved[64]; /* reserved for future use */ }; +enum { + SNDRV_RAWMIDI_FRAMING_NONE = 0, + SNDRV_RAWMIDI_FRAMING_TSTAMP, + SNDRV_RAWMIDI_FRAMING_LAST = SNDRV_RAWMIDI_FRAMING_TSTAMP, +}; + +#define SND_RAWMIDI_FRAMING_DATA_LENGTH 16 + +struct snd_rawmidi_framing_tstamp { + /* For now, frame_type is always 0. Midi 2.0 is expected to add new + * types here. Applications are expected to skip unknown frame types. + */ + unsigned char frame_type; + unsigned char length; /* number of valid bytes in data field */ + unsigned char reserved[2]; + unsigned int tv_nsec; /* nanoseconds */ + unsigned long long tv_sec; /* seconds */ + unsigned char data[SND_RAWMIDI_FRAMING_DATA_LENGTH]; +}; + struct snd_rawmidi_params { int stream; size_t buffer_size; /* queue size in bytes */ size_t avail_min; /* minimum avail bytes for wakeup */ unsigned int no_active_sensing: 1; /* do not send active sensing byte in close() */ - unsigned char reserved[16]; /* reserved for future use */ + unsigned char framing; /* For input data only, frame incoming data */ + unsigned char clock_type; /* Type of clock to use for framing, same as clockid_t */ + unsigned char reserved[14]; /* reserved for future use */ }; #ifndef __KERNEL__ diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c index aca00af93afe..d4b6b9b5c0e4 100644 --- a/sound/core/rawmidi.c +++ b/sound/core/rawmidi.c @@ -683,6 +683,8 @@ static int resize_runtime_buffer(struct snd_rawmidi_runtime *runtime, if (params->buffer_size < 32 || params->buffer_size > 1024L * 1024L) return -EINVAL; + if (params->framing == SNDRV_RAWMIDI_FRAMING_TSTAMP && params->buffer_size & 0x1f) + return -EINVAL; if (params->avail_min < 1 || params->avail_min > params->buffer_size) return -EINVAL; if (params->buffer_size != runtime->buffer_size) { @@ -720,7 +722,16 @@ EXPORT_SYMBOL(snd_rawmidi_output_params); int snd_rawmidi_input_params(struct snd_rawmidi_substream *substream, struct snd_rawmidi_params *params) { + if (params->framing) { + if (params->framing > SNDRV_RAWMIDI_FRAMING_LAST) + return -EINVAL; + /* framing requires a valid clock type */ + if (params->clock_type != CLOCK_MONOTONIC_RAW && params->clock_type != CLOCK_MONOTONIC) + return -EINVAL; + } snd_rawmidi_drain_input(substream); + substream->framing = params->framing; + substream->clock_type = params->clock_type; return resize_runtime_buffer(substream->runtime, params, true); } EXPORT_SYMBOL(snd_rawmidi_input_params); @@ -963,6 +974,42 @@ static int snd_rawmidi_control_ioctl(struct snd_card *card, return -ENOIOCTLCMD; } +static int receive_with_tstamp_framing(struct snd_rawmidi_substream *substream, + const unsigned char *buffer, int src_count, struct timespec64 *tstamp) +{ + struct snd_rawmidi_runtime *runtime = substream->runtime; + struct snd_rawmidi_framing_tstamp *dest_ptr; + struct snd_rawmidi_framing_tstamp frame = { .tv_sec = tstamp->tv_sec, .tv_nsec = tstamp->tv_nsec }; + + int dest_frames = 0; + int frame_size = sizeof(struct snd_rawmidi_framing_tstamp); + + if (snd_BUG_ON(runtime->hw_ptr & 0x1f || runtime->buffer_size & 0x1f || frame_size != 0x20)) + return -EINVAL; + while (src_count > 0) { + if ((int)(runtime->buffer_size - runtime->avail) < frame_size) { + runtime->xruns += src_count; + return dest_frames * frame_size; + } + if (src_count >= SND_RAWMIDI_FRAMING_DATA_LENGTH) + frame.length = SND_RAWMIDI_FRAMING_DATA_LENGTH; + else { + frame.length = src_count; + memset(frame.data, 0, SND_RAWMIDI_FRAMING_DATA_LENGTH); + } + memcpy(frame.data, buffer, frame.length); + buffer += frame.length; + src_count -= frame.length; + dest_ptr = (struct snd_rawmidi_framing_tstamp *) (runtime->buffer + runtime->hw_ptr); + *dest_ptr = frame; + runtime->avail += frame_size; + runtime->hw_ptr += frame_size; + runtime->hw_ptr %= runtime->buffer_size; + dest_frames++; + } + return dest_frames * frame_size; +} + /** * snd_rawmidi_receive - receive the input data from the device * @substream: the rawmidi substream @@ -977,6 +1024,7 @@ int snd_rawmidi_receive(struct snd_rawmidi_substream *substream, const unsigned char *buffer, int count) { unsigned long flags; + struct timespec64 ts64; int result = 0, count1; struct snd_rawmidi_runtime *runtime = substream->runtime; @@ -987,8 +1035,15 @@ int snd_rawmidi_receive(struct snd_rawmidi_substream *substream, "snd_rawmidi_receive: input is not active!!!\n"); return -EINVAL; } - spin_lock_irqsave(&runtime->lock, flags); - if (count == 1) { /* special case, faster code */ + if (substream->framing == SNDRV_RAWMIDI_FRAMING_TSTAMP) { + if (substream->clock_type == CLOCK_MONOTONIC_RAW) + ktime_get_raw_ts64(&ts64); + else + ktime_get_ts64(&ts64); + spin_lock_irqsave(&runtime->lock, flags); + result = receive_with_tstamp_framing(substream, buffer, count, &ts64); + } else if (count == 1) { /* special case, faster code */ + spin_lock_irqsave(&runtime->lock, flags); substream->bytes++; if (runtime->avail < runtime->buffer_size) { runtime->buffer[runtime->hw_ptr++] = buffer[0]; @@ -999,6 +1054,7 @@ int snd_rawmidi_receive(struct snd_rawmidi_substream *substream, runtime->xruns++; } } else { + spin_lock_irqsave(&runtime->lock, flags); substream->bytes += count; count1 = runtime->buffer_size - runtime->hw_ptr; if (count1 > count) -- 2.25.1

3 19

[PATCH v2 00/12] ASoC: simple-card-utils: prepare for multi support
by Kuninori Morimoto 13 Apr '21

13 Apr '21

Hi Mark This patch-set is for v2 of Multi-CPU/Codec support, and some cleanups. v1 had patch-conflict on simple-card / audio-graph with below. v2 was solved it. fa74c223b6fd78a5314b4c61b9abdbed3c2185b4 ("ASoC: simple-card: fix possible uninitialized single_cpu local variable") I want to add new audio-graph-card2 driver which can support not only DPCM, but also Multi-CPU/Codec, and Codec2Codec. And it is also supporting audio-graph-card2 base custom driver. But before supporting such driver, we need to cleanup existing simple-card / audio-graph, because these and new driver are sharing code. Link: https://lore.kernel.org/r/87wntmod33.wl-kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/20210408141639.GA39604@sirena.org.uk Kuninori Morimoto (12): ASoC: simple-card-utils: setup dai_props cpu_dai/codec_dai at initial timing ASoC: simple-card-utils: remove li->dais/li->conf ASoC: simple-card-utils: use for_each_prop_xxx() ASoC: simple-card-utils: remove asoc_simple_parse_xxx() ASoC: simple-card-utils: care multi DAI at asoc_simple_clean_reference() ASoC: simple-card-utils: indicate dai_fmt if exist ASoC: simple-card-utils: indicate missing CPU/Codec numbers for debug ASoC: simple-card-utils: add simple_props_to_xxx() macro ASoC: simple-card-utils: multi support at asoc_simple_canonicalize_cpu/platform() ASoC: simple-card-utils: tidyup debug info for clock ASoC: simple-card-utils: tidyup dev_dbg() to use 1 line ASoC: simple-card-utils: tidyup asoc_simple_parse_convert() include/sound/simple_card_utils.h | 107 ++++++++++------ sound/soc/generic/audio-graph-card.c | 64 ++++------ sound/soc/generic/simple-card-utils.c | 174 ++++++++++++++++---------- sound/soc/generic/simple-card.c | 70 ++++------- 4 files changed, 226 insertions(+), 189 deletions(-) -- 2.25.1

2 13

[PATCH -next] ASoC: sti: sti_uniperif: add missing MODULE_DEVICE_TABLE
by Chen Lifu 13 Apr '21

13 Apr '21

This patch adds missing MODULE_DEVICE_TABLE definition which generates correct modalias for automatic loading of this driver when it is built as an external module. Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: Chen Lifu <chenlifu(a)huawei.com> --- sound/soc/sti/sti_uniperif.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/sti/sti_uniperif.c b/sound/soc/sti/sti_uniperif.c index 67315d9b352d..e3561f00ed40 100644 --- a/sound/soc/sti/sti_uniperif.c +++ b/sound/soc/sti/sti_uniperif.c @@ -97,6 +97,7 @@ static const struct of_device_id snd_soc_sti_match[] = { }, {}, }; +MODULE_DEVICE_TABLE(of, snd_soc_sti_match); int sti_uniperiph_reset(struct uniperif *uni) {

3 2

[PATCH] ASoC: cs35l36: Fix an error handling path in 'cs35l36_i2c_probe()'
by Christophe JAILLET 13 Apr '21

13 Apr '21

If 'devm_regmap_init_i2c()' fails, there is no need to goto err. We should return directly as already done by the surrounding error handling paths. Signed-off-by: Christophe JAILLET <christophe.jaillet(a)wanadoo.fr> --- sound/soc/codecs/cs35l36.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs35l36.c b/sound/soc/codecs/cs35l36.c index 4451ca9f4916..a038bcec2d17 100644 --- a/sound/soc/codecs/cs35l36.c +++ b/sound/soc/codecs/cs35l36.c @@ -1721,7 +1721,7 @@ static int cs35l36_i2c_probe(struct i2c_client *i2c_client, if (IS_ERR(cs35l36->regmap)) { ret = PTR_ERR(cs35l36->regmap); dev_err(dev, "regmap_init() failed: %d\n", ret); - goto err; + return ret; } cs35l36->num_supplies = ARRAY_SIZE(cs35l36_supplies); -- 2.27.0

2 1

[PATCH] ASoC: meson: axg-frddr: fix fifo depth on g12 and sm1
by Jerome Brunet 13 Apr '21

13 Apr '21

Previous fifo depth patch was only tested on axg, not g12 or sm1. Of course, while adding hw_params dai callback for the axg, I forgot to do the same for g12 and sm1, leaving the depth unset and breaking playback on these SoCs. Add hw_params callback to the g12 dai_ops to fix the problem. Fixes: 6f68accaa864 ("ASoC: meson: axg-frddr: set fifo depth according to the period") Reported-by: Christian Hewitt <christianshewitt(a)gmail.com> Tested-by: Christian Hewitt <christianshewitt(a)gmail.com> Signed-off-by: Jerome Brunet <jbrunet(a)baylibre.com> --- sound/soc/meson/axg-frddr.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/meson/axg-frddr.c b/sound/soc/meson/axg-frddr.c index 8ed114de0bdf..37f4bb3469b5 100644 --- a/sound/soc/meson/axg-frddr.c +++ b/sound/soc/meson/axg-frddr.c @@ -171,6 +171,7 @@ static const struct axg_fifo_match_data axg_frddr_match_data = { static const struct snd_soc_dai_ops g12a_frddr_ops = { .prepare = g12a_frddr_dai_prepare, + .hw_params = axg_frddr_dai_hw_params, .startup = axg_frddr_dai_startup, .shutdown = axg_frddr_dai_shutdown, }; -- 2.30.2

2 1

rockchip/veyron sound level issues
by GitHub issues - opened 13 Apr '21

13 Apr '21

alsa-project/alsa-ucm-conf issue #87 was opened from kapouer: Hi, setup: > asus c201 (speedy veyron) > linux 5.10.28 or 5.11.13 > debian bullseye: alsa 1.2.4, asa-ucm-conf (and topology) installed, pulseaudio 14.2, gnome 3.38 on wayland what works: - jack input/output using headset problems: - speaker output is somewhat low even at 100% (feels like 50%) - microphone is so low that it cannot detect sound (smashing the mic hole with a stick is detected but barely audible) - last time i checked hdmi was working - i need to check again though What can i provide ? pulseaudio -vvvvvv ? Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/87 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH] ASoC: ak5558: correct reset polarity
by Shengjiu Wang 13 Apr '21

13 Apr '21

Reset (aka power off) happens when the reset gpio is made active. Change function name to ak5558_reset to match devicetree property "reset-gpios". Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/codecs/ak5558.c | 30 ++++++++++-------------------- 1 file changed, 10 insertions(+), 20 deletions(-) diff --git a/sound/soc/codecs/ak5558.c b/sound/soc/codecs/ak5558.c index 8e4dca753f0b..5c3f15827423 100644 --- a/sound/soc/codecs/ak5558.c +++ b/sound/soc/codecs/ak5558.c @@ -318,29 +318,19 @@ static struct snd_soc_dai_driver ak5552_dai = { .ops = &ak5558_dai_ops, }; -static void ak5558_power_off(struct ak5558_priv *ak5558) +static void ak5558_reset(struct ak5558_priv *ak5558, bool active) { - if (!ak5558->reset_gpiod) - return; - - gpiod_set_value_cansleep(ak5558->reset_gpiod, 0); - usleep_range(1000, 2000); -} - -static void ak5558_power_on(struct ak5558_priv *ak5558) -{ - if (!ak5558->reset_gpiod) - return; - - gpiod_set_value_cansleep(ak5558->reset_gpiod, 1); - usleep_range(1000, 2000); + if (ak5558->reset_gpiod) { + gpiod_set_value_cansleep(ak5558->reset_gpiod, active); + usleep_range(1000, 2000); + } } static int ak5558_probe(struct snd_soc_component *component) { struct ak5558_priv *ak5558 = snd_soc_component_get_drvdata(component); - ak5558_power_on(ak5558); + ak5558_reset(ak5558, false); return ak5558_set_mcki(component); } @@ -348,7 +338,7 @@ static void ak5558_remove(struct snd_soc_component *component) { struct ak5558_priv *ak5558 = snd_soc_component_get_drvdata(component); - ak5558_power_off(ak5558); + ak5558_reset(ak5558, true); } static int __maybe_unused ak5558_runtime_suspend(struct device *dev) @@ -356,7 +346,7 @@ static int __maybe_unused ak5558_runtime_suspend(struct device *dev) struct ak5558_priv *ak5558 = dev_get_drvdata(dev); regcache_cache_only(ak5558->regmap, true); - ak5558_power_off(ak5558); + ak5558_reset(ak5558, true); regulator_bulk_disable(ARRAY_SIZE(ak5558->supplies), ak5558->supplies); @@ -375,8 +365,8 @@ static int __maybe_unused ak5558_runtime_resume(struct device *dev) return ret; } - ak5558_power_off(ak5558); - ak5558_power_on(ak5558); + ak5558_reset(ak5558, true); + ak5558_reset(ak5558, false); regcache_cache_only(ak5558->regmap, false); regcache_mark_dirty(ak5558->regmap); -- 2.27.0

2 1

[PATCH 0/5] ASoC: clock provider clean-up
by Jerome Brunet 13 Apr '21

13 Apr '21

The purpose of this patchset it remove the use the clk member of 'struct clk_hw' in ASoC. 'struct clk' is a per-user reference to an actual clock. In the future, the clk member in 'struct clk_hw' may go away. The usage of this member by a clock provider usually falls into either of following categories: * Mis-usage of the clock consumer API by a clock provider. * Clock provider also being a user of its own clocks. In this case the provider should request a 'struct clk' through the appropriate API instead of poking in 'struct clk_hw' internals. Jerome Brunet (5): ASoC: stm32: properly get clk from the provider ASoC: wcd934x: use the clock provider API ASoC: rt5682: clock driver must use the clock provider API ASoC: lpass: use the clock provider API ASoC: da7219: properly get clk from the provider sound/soc/codecs/da7219.c | 5 ++++- sound/soc/codecs/lpass-va-macro.c | 2 +- sound/soc/codecs/lpass-wsa-macro.c | 9 +++------ sound/soc/codecs/rt5682.c | 6 +++--- sound/soc/codecs/wcd934x.c | 6 ++++-- sound/soc/stm/stm32_sai_sub.c | 5 ++++- 6 files changed, 19 insertions(+), 14 deletions(-) -- 2.30.2

3 13

[PATCH v2] Behringer UFX1604 / UFX1204: get rid of unneeded implicit feedback and pops and clicks while on 96000hz
by Geraldo Nascimento 13 Apr '21

13 Apr '21

More complete patch disabling unneeded implicit feedback and setting clock selector to default clock on rate change for UFX1604 After re-reading https://bugzilla.kernel.org/show_bug.cgi?id=199327 it is even more clear to me that implicit feedback for the UFX1604/UFX1204 needs to be disabled. This is a more complete patch that disables that and for the UFX1604 only sets the clock selector to its pin 1 default clock synced to the USB SOF upon rate change. This is needed because apparently the endpoints are hardwired to the clock selector and after we change the rate on the main USB SOF synced clock the clock selector is left in a halfway state in regards to the sampling rate. That's why the pops and clicks aren't evident at stock 48000Hz, become slightly audible at 44100Hz and detestable at 96000Hz. Seems the clock selector needs a nudge or it will screw up the sync. Unfortunately I don't have access to the lsusb -v of the UFX1204 soI'm waiting for someone to share it here in the list or in the bugzilla thread. This patch needs some more love from the community. --- This one has been bugging me for quite a while. I went deep hard in the guts of ALSA to try to solve it, and it turned out to be a minor thing apparently. The problem is old, and every UFX1604 Linux user can attest that it's impossible to use 96000hz in DUPLEX mode without annoying pops and clicks on the capture stream. The fix is simple: after we alter the CLOCK_SOURCE to match our sample rate, let's tell the CLOCK_SELECTOR we want CLOCK_SOURCE 212 (synced to USB SOF) on pin 1. Solves the problem for me, no more pops and clicks while on 96000hz. --- Signed-off-by: Geraldo Nascimento <geraldogabriel(a)gmail.com> --- implicit.c 2021-04-04 20:51:57.226754632 -0300 +++ implicit.c 2021-04-09 21:55:19.593994214 -0300 @@ -50,8 +50,6 @@ static const struct snd_usb_implicit_fb_ /* Fixed EP */ /* FIXME: check the availability of generic matching */ - IMPLICIT_FB_FIXED_DEV(0x1397, 0x0001, 0x81, 1), /* Behringer UFX1604 */ - IMPLICIT_FB_FIXED_DEV(0x1397, 0x0002, 0x81, 1), /* Behringer UFX1204 */ IMPLICIT_FB_FIXED_DEV(0x2466, 0x8010, 0x81, 2), /* Fractal Audio Axe-Fx III */ IMPLICIT_FB_FIXED_DEV(0x31e9, 0x0001, 0x81, 2), /* Solid State Logic SSL2 */ IMPLICIT_FB_FIXED_DEV(0x31e9, 0x0002, 0x81, 2), /* Solid State Logic SSL2+ */ --- clock.c 2021-04-09 22:01:08.234001611 -0300 +++ clock.c 2021-04-09 22:01:53.640669243 -0300 @@ -610,6 +610,14 @@ int snd_usb_set_sample_rate_v2v3(struct if (err < 0) return err; + if (chip->usb_id == USB_ID(0x1397, 0x0001)) { /* Behringer UFX1604 */ + printk(KERN_WARNING "Setting clock selector for UFX1604"); + err = uac_clock_selector_set_val(chip, 211, 1); + if (err < 0) + return err; + } + + return get_sample_rate_v2v3(chip, fmt->iface, fmt->altsetting, clock); }

2 4

[PATCH 0/2] ASoC: SOF: Intel: descriptor corrections for TGL and ADL
by Pierre-Louis Bossart 12 Apr '21

12 Apr '21

One missed property for TigerLake and need for separate descriptors between ADL-S and the other flavors. Libin Yang (1): ASoC: SOF: Intel: add missing use_acpi_target_states for TGL platforms Sathya Prakash M R (1): ASoC: SOF: Intel: Update ADL P to use its own descriptor sound/soc/sof/intel/pci-tgl.c | 20 +++++++++++++++++++- sound/soc/sof/intel/tgl.c | 2 +- 2 files changed, 20 insertions(+), 2 deletions(-) -- 2.25.1

2 3

[PATCH 0/6] ASoC: SOF: simplify nocodec mode
by Ranjani Sridharan 12 Apr '21

12 Apr '21

This set of patches simplify the implementation of nocodec mode in SOF. Pierre-Louis Bossart (6): ASoC: SOF: add Kconfig option for probe workqueues ASoC: soc-acpi: add new fields for mach_params ASoC: SOF: change signature of set_mach_params() callback ASoC: SOF: Intel: update set_mach_params() ASoC: SOF: pcm: export snd_pcm_dai_link_fixup ASOC: SOF: simplify nocodec mode include/sound/soc-acpi.h | 4 ++++ include/sound/sof.h | 3 --- sound/soc/sof/Kconfig | 11 +++++++++++ sound/soc/sof/intel/bdw.c | 8 ++++++-- sound/soc/sof/intel/byt.c | 22 +++++++++++++--------- sound/soc/sof/intel/hda.c | 8 ++++++-- sound/soc/sof/intel/hda.h | 2 +- sound/soc/sof/nocodec.c | 39 +++++++++++++++++++++------------------ sound/soc/sof/ops.h | 6 ++---- sound/soc/sof/pcm.c | 1 + sound/soc/sof/sof-audio.c | 34 +++++++++++++++------------------- sound/soc/sof/sof-priv.h | 2 +- 12 files changed, 81 insertions(+), 59 deletions(-) -- 2.25.1

2 7

[PATCH] ASoC: SOF: Kconfig: fix typo of SND_SOC_SOF_PCI
by Ranjani Sridharan 12 Apr '21

12 Apr '21

From: Libin Yang <libin.yang(a)intel.com> It should be 'endif ## SND_SOC_SOF_PCI' instead of 'endif ## SND_SOC_SOF_INTEL_PCI' Signed-off-by: Libin Yang <libin.yang(a)intel.com> Reviewed-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> --- sound/soc/sof/intel/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sof/intel/Kconfig b/sound/soc/sof/intel/Kconfig index da1c396f529d..4bce89b5ea40 100644 --- a/sound/soc/sof/intel/Kconfig +++ b/sound/soc/sof/intel/Kconfig @@ -293,6 +293,6 @@ config SND_SOC_SOF_INTEL_SOUNDWIRE Say Y if you want to enable SoundWire links with SOF. If unsure select "N". -endif ## SND_SOC_SOF_INTEL_PCI +endif ## SND_SOC_SOF_PCI endif ## SND_SOC_SOF_INTEL_TOPLEVEL -- 2.25.1

2 1

[PATCH 0/2] ASoC: SOF: simplify probe and report errors
by Ranjani Sridharan 12 Apr '21

12 Apr '21

These patches simplify the sof_probe_complete handling and address reporting of errors during probe. Peter Ujfalusi (2): ASoC: SOF: core: Add missing error prints to device probe operation ASoC: SOF: Simplify sof_probe_complete handling for acpi/pci/of sound/soc/sof/core.c | 9 +++++++-- sound/soc/sof/sof-acpi-dev.c | 18 +++--------------- sound/soc/sof/sof-of-dev.c | 18 +++--------------- sound/soc/sof/sof-pci-dev.c | 20 ++++---------------- 4 files changed, 17 insertions(+), 48 deletions(-) -- 2.25.1

2 3

[PATCH] ASoC: SOF: topology: remove useless code
by Ranjani Sridharan 12 Apr '21

12 Apr '21

The patch "ASoC: SOF: Intel: hda: assign link DMA channel at run-time" fixed the sof_link_hda_unload() to remove the call to the BE hw_free op but left the rest of code that become redundant. So, remove sof_link_hda_unload() along with the link_unload() op entirely as it is not longer needed. Reported-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> --- sound/soc/sof/topology.c | 64 ---------------------------------------- 1 file changed, 64 deletions(-) diff --git a/sound/soc/sof/topology.c b/sound/soc/sof/topology.c index 036202768b75..59abcfc9bd55 100644 --- a/sound/soc/sof/topology.c +++ b/sound/soc/sof/topology.c @@ -3371,69 +3371,6 @@ static int sof_link_load(struct snd_soc_component *scomp, int index, return ret; } -static int sof_link_hda_unload(struct snd_sof_dev *sdev, - struct snd_soc_dai_link *link) -{ - struct snd_soc_dai *dai; - - dai = snd_soc_find_dai(link->cpus); - if (!dai) { - dev_err(sdev->dev, "error: failed to find dai %s in %s", - link->cpus->dai_name, __func__); - return -EINVAL; - } - - return 0; -} - -static int sof_link_unload(struct snd_soc_component *scomp, - struct snd_soc_dobj *dobj) -{ - struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(scomp); - struct snd_soc_dai_link *link = - container_of(dobj, struct snd_soc_dai_link, dobj); - - struct snd_sof_dai *sof_dai; - int ret = 0; - - /* only BE link is loaded by sof */ - if (!link->no_pcm) - return 0; - - list_for_each_entry(sof_dai, &sdev->dai_list, list) { - if (!sof_dai->name) - continue; - - if (strcmp(link->name, sof_dai->name) == 0) - goto found; - } - - dev_err(scomp->dev, "error: failed to find dai %s in %s", - link->name, __func__); - return -EINVAL; -found: - - switch (sof_dai->dai_config->type) { - case SOF_DAI_INTEL_SSP: - case SOF_DAI_INTEL_DMIC: - case SOF_DAI_INTEL_ALH: - case SOF_DAI_IMX_SAI: - case SOF_DAI_IMX_ESAI: - /* no resource needs to be released for all cases above */ - break; - case SOF_DAI_INTEL_HDA: - ret = sof_link_hda_unload(sdev, link); - break; - default: - dev_err(scomp->dev, "error: invalid DAI type %d\n", - sof_dai->dai_config->type); - ret = -EINVAL; - break; - } - - return ret; -} - /* DAI link - used for any driver specific init */ static int sof_route_load(struct snd_soc_component *scomp, int index, struct snd_soc_dapm_route *route) @@ -3734,7 +3671,6 @@ static struct snd_soc_tplg_ops sof_tplg_ops = { /* DAI link - used for any driver specific init */ .link_load = sof_link_load, - .link_unload = sof_link_unload, /* completion - called at completion of firmware loading */ .complete = sof_complete, -- 2.25.1

2 1

[PATCH AUTOSEL 4.4 08/23] ASoC: fsl_esai: Fix TDM slot setup for I2S mode
by Sasha Levin 12 Apr '21

12 Apr '21

From: Alexander Shiyan <shc_work(a)mail.ru> [ Upstream commit e7a48c710defa0e0fef54d42b7d9e4ab596e2761 ] When using the driver in I2S TDM mode, the fsl_esai_startup() function rewrites the number of slots previously set by the fsl_esai_set_dai_tdm_slot() function to 2. To fix this, let's use the saved slot count value or, if TDM is not used and the number of slots is not set, the driver will use the default value (2), which is set by fsl_esai_probe(). Signed-off-by: Alexander Shiyan <shc_work(a)mail.ru> Acked-by: Nicolin Chen <nicoleotsuka(a)gmail.com> Link: https://lore.kernel.org/r/20210402081405.9892-1-shc_work@mail.ru Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/fsl/fsl_esai.c | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index 40075b9afb79..fad711a3f4b4 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -488,11 +488,13 @@ static int fsl_esai_startup(struct snd_pcm_substream *substream, ESAI_SAICR_SYNC, esai_priv->synchronous ? ESAI_SAICR_SYNC : 0); - /* Set a default slot number -- 2 */ + /* Set slots count */ regmap_update_bits(esai_priv->regmap, REG_ESAI_TCCR, - ESAI_xCCR_xDC_MASK, ESAI_xCCR_xDC(2)); + ESAI_xCCR_xDC_MASK, + ESAI_xCCR_xDC(esai_priv->slots)); regmap_update_bits(esai_priv->regmap, REG_ESAI_RCCR, - ESAI_xCCR_xDC_MASK, ESAI_xCCR_xDC(2)); + ESAI_xCCR_xDC_MASK, + ESAI_xCCR_xDC(esai_priv->slots)); } return 0; -- 2.30.2

1 0

[PATCH AUTOSEL 4.9 08/23] ASoC: fsl_esai: Fix TDM slot setup for I2S mode
by Sasha Levin 12 Apr '21

12 Apr '21

From: Alexander Shiyan <shc_work(a)mail.ru> [ Upstream commit e7a48c710defa0e0fef54d42b7d9e4ab596e2761 ] When using the driver in I2S TDM mode, the fsl_esai_startup() function rewrites the number of slots previously set by the fsl_esai_set_dai_tdm_slot() function to 2. To fix this, let's use the saved slot count value or, if TDM is not used and the number of slots is not set, the driver will use the default value (2), which is set by fsl_esai_probe(). Signed-off-by: Alexander Shiyan <shc_work(a)mail.ru> Acked-by: Nicolin Chen <nicoleotsuka(a)gmail.com> Link: https://lore.kernel.org/r/20210402081405.9892-1-shc_work@mail.ru Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/fsl/fsl_esai.c | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index fa64cc2b1729..94bf497092b2 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -495,11 +495,13 @@ static int fsl_esai_startup(struct snd_pcm_substream *substream, ESAI_SAICR_SYNC, esai_priv->synchronous ? ESAI_SAICR_SYNC : 0); - /* Set a default slot number -- 2 */ + /* Set slots count */ regmap_update_bits(esai_priv->regmap, REG_ESAI_TCCR, - ESAI_xCCR_xDC_MASK, ESAI_xCCR_xDC(2)); + ESAI_xCCR_xDC_MASK, + ESAI_xCCR_xDC(esai_priv->slots)); regmap_update_bits(esai_priv->regmap, REG_ESAI_RCCR, - ESAI_xCCR_xDC_MASK, ESAI_xCCR_xDC(2)); + ESAI_xCCR_xDC_MASK, + ESAI_xCCR_xDC(esai_priv->slots)); } return 0; -- 2.30.2

1 0

[PATCH AUTOSEL 4.14 08/25] ASoC: fsl_esai: Fix TDM slot setup for I2S mode
by Sasha Levin 12 Apr '21

12 Apr '21

From: Alexander Shiyan <shc_work(a)mail.ru> [ Upstream commit e7a48c710defa0e0fef54d42b7d9e4ab596e2761 ] When using the driver in I2S TDM mode, the fsl_esai_startup() function rewrites the number of slots previously set by the fsl_esai_set_dai_tdm_slot() function to 2. To fix this, let's use the saved slot count value or, if TDM is not used and the number of slots is not set, the driver will use the default value (2), which is set by fsl_esai_probe(). Signed-off-by: Alexander Shiyan <shc_work(a)mail.ru> Acked-by: Nicolin Chen <nicoleotsuka(a)gmail.com> Link: https://lore.kernel.org/r/20210402081405.9892-1-shc_work@mail.ru Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/fsl/fsl_esai.c | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index 6152ae24772b..3ac87f7843f6 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -494,11 +494,13 @@ static int fsl_esai_startup(struct snd_pcm_substream *substream, ESAI_SAICR_SYNC, esai_priv->synchronous ? ESAI_SAICR_SYNC : 0); - /* Set a default slot number -- 2 */ + /* Set slots count */ regmap_update_bits(esai_priv->regmap, REG_ESAI_TCCR, - ESAI_xCCR_xDC_MASK, ESAI_xCCR_xDC(2)); + ESAI_xCCR_xDC_MASK, + ESAI_xCCR_xDC(esai_priv->slots)); regmap_update_bits(esai_priv->regmap, REG_ESAI_RCCR, - ESAI_xCCR_xDC_MASK, ESAI_xCCR_xDC(2)); + ESAI_xCCR_xDC_MASK, + ESAI_xCCR_xDC(esai_priv->slots)); } return 0; -- 2.30.2

1 0

[PATCH AUTOSEL 4.19 11/28] ASoC: fsl_esai: Fix TDM slot setup for I2S mode
by Sasha Levin 12 Apr '21

12 Apr '21

From: Alexander Shiyan <shc_work(a)mail.ru> [ Upstream commit e7a48c710defa0e0fef54d42b7d9e4ab596e2761 ] When using the driver in I2S TDM mode, the fsl_esai_startup() function rewrites the number of slots previously set by the fsl_esai_set_dai_tdm_slot() function to 2. To fix this, let's use the saved slot count value or, if TDM is not used and the number of slots is not set, the driver will use the default value (2), which is set by fsl_esai_probe(). Signed-off-by: Alexander Shiyan <shc_work(a)mail.ru> Acked-by: Nicolin Chen <nicoleotsuka(a)gmail.com> Link: https://lore.kernel.org/r/20210402081405.9892-1-shc_work@mail.ru Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/fsl/fsl_esai.c | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index ff96db91f818..baa76337c33f 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -497,11 +497,13 @@ static int fsl_esai_startup(struct snd_pcm_substream *substream, ESAI_SAICR_SYNC, esai_priv->synchronous ? ESAI_SAICR_SYNC : 0); - /* Set a default slot number -- 2 */ + /* Set slots count */ regmap_update_bits(esai_priv->regmap, REG_ESAI_TCCR, - ESAI_xCCR_xDC_MASK, ESAI_xCCR_xDC(2)); + ESAI_xCCR_xDC_MASK, + ESAI_xCCR_xDC(esai_priv->slots)); regmap_update_bits(esai_priv->regmap, REG_ESAI_RCCR, - ESAI_xCCR_xDC_MASK, ESAI_xCCR_xDC(2)); + ESAI_xCCR_xDC_MASK, + ESAI_xCCR_xDC(esai_priv->slots)); } return 0; -- 2.30.2

1 0

[PATCH AUTOSEL 5.4 17/39] ASoC: fsl_esai: Fix TDM slot setup for I2S mode
by Sasha Levin 12 Apr '21

12 Apr '21

From: Alexander Shiyan <shc_work(a)mail.ru> [ Upstream commit e7a48c710defa0e0fef54d42b7d9e4ab596e2761 ] When using the driver in I2S TDM mode, the fsl_esai_startup() function rewrites the number of slots previously set by the fsl_esai_set_dai_tdm_slot() function to 2. To fix this, let's use the saved slot count value or, if TDM is not used and the number of slots is not set, the driver will use the default value (2), which is set by fsl_esai_probe(). Signed-off-by: Alexander Shiyan <shc_work(a)mail.ru> Acked-by: Nicolin Chen <nicoleotsuka(a)gmail.com> Link: https://lore.kernel.org/r/20210402081405.9892-1-shc_work@mail.ru Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/fsl/fsl_esai.c | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index 84290be778f0..33ade79fa032 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -494,11 +494,13 @@ static int fsl_esai_startup(struct snd_pcm_substream *substream, ESAI_SAICR_SYNC, esai_priv->synchronous ? ESAI_SAICR_SYNC : 0); - /* Set a default slot number -- 2 */ + /* Set slots count */ regmap_update_bits(esai_priv->regmap, REG_ESAI_TCCR, - ESAI_xCCR_xDC_MASK, ESAI_xCCR_xDC(2)); + ESAI_xCCR_xDC_MASK, + ESAI_xCCR_xDC(esai_priv->slots)); regmap_update_bits(esai_priv->regmap, REG_ESAI_RCCR, - ESAI_xCCR_xDC_MASK, ESAI_xCCR_xDC(2)); + ESAI_xCCR_xDC_MASK, + ESAI_xCCR_xDC(esai_priv->slots)); } return 0; -- 2.30.2

1 0

[PATCH AUTOSEL 5.4 10/39] ASoC: max98373: Added 30ms turn on/off time delay
by Sasha Levin 12 Apr '21

12 Apr '21

From: Ryan Lee <ryans.lee(a)maximintegrated.com> [ Upstream commit 3a27875e91fb9c29de436199d20b33f9413aea77 ] Amp requires 10 ~ 30ms for the power ON and OFF. Added 30ms delay for stability. Signed-off-by: Ryan Lee <ryans.lee(a)maximintegrated.com> Link: https://lore.kernel.org/r/20210325033555.29377-2-ryans.lee@maximintegrated.… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/max98373.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/max98373.c b/sound/soc/codecs/max98373.c index 96718e3a1ad0..16fbc9faed90 100644 --- a/sound/soc/codecs/max98373.c +++ b/sound/soc/codecs/max98373.c @@ -410,11 +410,13 @@ static int max98373_dac_event(struct snd_soc_dapm_widget *w, regmap_update_bits(max98373->regmap, MAX98373_R20FF_GLOBAL_SHDN, MAX98373_GLOBAL_EN_MASK, 1); + usleep_range(30000, 31000); break; case SND_SOC_DAPM_POST_PMD: regmap_update_bits(max98373->regmap, MAX98373_R20FF_GLOBAL_SHDN, MAX98373_GLOBAL_EN_MASK, 0); + usleep_range(30000, 31000); max98373->tdm_mode = false; break; default: -- 2.30.2

1 0

[PATCH AUTOSEL 5.10 21/46] ASoC: fsl_esai: Fix TDM slot setup for I2S mode
by Sasha Levin 12 Apr '21

12 Apr '21

From: Alexander Shiyan <shc_work(a)mail.ru> [ Upstream commit e7a48c710defa0e0fef54d42b7d9e4ab596e2761 ] When using the driver in I2S TDM mode, the fsl_esai_startup() function rewrites the number of slots previously set by the fsl_esai_set_dai_tdm_slot() function to 2. To fix this, let's use the saved slot count value or, if TDM is not used and the number of slots is not set, the driver will use the default value (2), which is set by fsl_esai_probe(). Signed-off-by: Alexander Shiyan <shc_work(a)mail.ru> Acked-by: Nicolin Chen <nicoleotsuka(a)gmail.com> Link: https://lore.kernel.org/r/20210402081405.9892-1-shc_work@mail.ru Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/fsl/fsl_esai.c | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index 39637ca78cdb..9f5f217a9607 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -524,11 +524,13 @@ static int fsl_esai_startup(struct snd_pcm_substream *substream, ESAI_SAICR_SYNC, esai_priv->synchronous ? ESAI_SAICR_SYNC : 0); - /* Set a default slot number -- 2 */ + /* Set slots count */ regmap_update_bits(esai_priv->regmap, REG_ESAI_TCCR, - ESAI_xCCR_xDC_MASK, ESAI_xCCR_xDC(2)); + ESAI_xCCR_xDC_MASK, + ESAI_xCCR_xDC(esai_priv->slots)); regmap_update_bits(esai_priv->regmap, REG_ESAI_RCCR, - ESAI_xCCR_xDC_MASK, ESAI_xCCR_xDC(2)); + ESAI_xCCR_xDC_MASK, + ESAI_xCCR_xDC(esai_priv->slots)); } return 0; -- 2.30.2

1 0

[PATCH AUTOSEL 5.10 13/46] ASoC: max98373: Added 30ms turn on/off time delay
by Sasha Levin 12 Apr '21

12 Apr '21

From: Ryan Lee <ryans.lee(a)maximintegrated.com> [ Upstream commit 3a27875e91fb9c29de436199d20b33f9413aea77 ] Amp requires 10 ~ 30ms for the power ON and OFF. Added 30ms delay for stability. Signed-off-by: Ryan Lee <ryans.lee(a)maximintegrated.com> Link: https://lore.kernel.org/r/20210325033555.29377-2-ryans.lee@maximintegrated.… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/max98373.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/max98373.c b/sound/soc/codecs/max98373.c index 929bb1798c43..1fd4dbbb4ecf 100644 --- a/sound/soc/codecs/max98373.c +++ b/sound/soc/codecs/max98373.c @@ -28,11 +28,13 @@ static int max98373_dac_event(struct snd_soc_dapm_widget *w, regmap_update_bits(max98373->regmap, MAX98373_R20FF_GLOBAL_SHDN, MAX98373_GLOBAL_EN_MASK, 1); + usleep_range(30000, 31000); break; case SND_SOC_DAPM_POST_PMD: regmap_update_bits(max98373->regmap, MAX98373_R20FF_GLOBAL_SHDN, MAX98373_GLOBAL_EN_MASK, 0); + usleep_range(30000, 31000); max98373->tdm_mode = false; break; default: -- 2.30.2

1 0

[PATCH AUTOSEL 5.10 12/46] ASoC: max98373: Changed amp shutdown register as volatile
by Sasha Levin 12 Apr '21

12 Apr '21

From: Ryan Lee <ryans.lee(a)maximintegrated.com> [ Upstream commit a23f9099ff1541f15704e96b784d3846d2a4483d ] 0x20FF(amp global enable) register was defined as non-volatile, but it is not. Overheating, overcurrent can cause amp shutdown in hardware. 'regmap_write' compare register readback value before writing to avoid same value writing. 'regmap_read' just read cache not actual hardware value for the non-volatile register. When amp is internally shutdown by some reason, next 'AMP ON' command can be ignored because regmap think amp is already ON. Signed-off-by: Ryan Lee <ryans.lee(a)maximintegrated.com> Link: https://lore.kernel.org/r/20210325033555.29377-1-ryans.lee@maximintegrated.… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/max98373-i2c.c | 1 + sound/soc/codecs/max98373-sdw.c | 1 + 2 files changed, 2 insertions(+) diff --git a/sound/soc/codecs/max98373-i2c.c b/sound/soc/codecs/max98373-i2c.c index 92921e34f948..32b0c1d98365 100644 --- a/sound/soc/codecs/max98373-i2c.c +++ b/sound/soc/codecs/max98373-i2c.c @@ -440,6 +440,7 @@ static bool max98373_volatile_reg(struct device *dev, unsigned int reg) case MAX98373_R2054_MEAS_ADC_PVDD_CH_READBACK: case MAX98373_R2055_MEAS_ADC_THERM_CH_READBACK: case MAX98373_R20B6_BDE_CUR_STATE_READBACK: + case MAX98373_R20FF_GLOBAL_SHDN: case MAX98373_R21FF_REV_ID: return true; default: diff --git a/sound/soc/codecs/max98373-sdw.c b/sound/soc/codecs/max98373-sdw.c index fa589d834f9a..14fd2f9a0bf3 100644 --- a/sound/soc/codecs/max98373-sdw.c +++ b/sound/soc/codecs/max98373-sdw.c @@ -214,6 +214,7 @@ static bool max98373_volatile_reg(struct device *dev, unsigned int reg) case MAX98373_R2054_MEAS_ADC_PVDD_CH_READBACK: case MAX98373_R2055_MEAS_ADC_THERM_CH_READBACK: case MAX98373_R20B6_BDE_CUR_STATE_READBACK: + case MAX98373_R20FF_GLOBAL_SHDN: case MAX98373_R21FF_REV_ID: /* SoundWire Control Port Registers */ case MAX98373_R0040_SCP_INIT_STAT_1 ... MAX98373_R0070_SCP_FRAME_CTLR: -- 2.30.2

1 0

[PATCH AUTOSEL 5.11 25/51] ASoC: fsl_esai: Fix TDM slot setup for I2S mode
by Sasha Levin 12 Apr '21

12 Apr '21

From: Alexander Shiyan <shc_work(a)mail.ru> [ Upstream commit e7a48c710defa0e0fef54d42b7d9e4ab596e2761 ] When using the driver in I2S TDM mode, the fsl_esai_startup() function rewrites the number of slots previously set by the fsl_esai_set_dai_tdm_slot() function to 2. To fix this, let's use the saved slot count value or, if TDM is not used and the number of slots is not set, the driver will use the default value (2), which is set by fsl_esai_probe(). Signed-off-by: Alexander Shiyan <shc_work(a)mail.ru> Acked-by: Nicolin Chen <nicoleotsuka(a)gmail.com> Link: https://lore.kernel.org/r/20210402081405.9892-1-shc_work@mail.ru Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/fsl/fsl_esai.c | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index 39637ca78cdb..9f5f217a9607 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -524,11 +524,13 @@ static int fsl_esai_startup(struct snd_pcm_substream *substream, ESAI_SAICR_SYNC, esai_priv->synchronous ? ESAI_SAICR_SYNC : 0); - /* Set a default slot number -- 2 */ + /* Set slots count */ regmap_update_bits(esai_priv->regmap, REG_ESAI_TCCR, - ESAI_xCCR_xDC_MASK, ESAI_xCCR_xDC(2)); + ESAI_xCCR_xDC_MASK, + ESAI_xCCR_xDC(esai_priv->slots)); regmap_update_bits(esai_priv->regmap, REG_ESAI_RCCR, - ESAI_xCCR_xDC_MASK, ESAI_xCCR_xDC(2)); + ESAI_xCCR_xDC_MASK, + ESAI_xCCR_xDC(esai_priv->slots)); } return 0; -- 2.30.2

1 0

[PATCH AUTOSEL 5.11 14/51] ASoC: max98373: Added 30ms turn on/off time delay
by Sasha Levin 12 Apr '21

12 Apr '21

From: Ryan Lee <ryans.lee(a)maximintegrated.com> [ Upstream commit 3a27875e91fb9c29de436199d20b33f9413aea77 ] Amp requires 10 ~ 30ms for the power ON and OFF. Added 30ms delay for stability. Signed-off-by: Ryan Lee <ryans.lee(a)maximintegrated.com> Link: https://lore.kernel.org/r/20210325033555.29377-2-ryans.lee@maximintegrated.… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/max98373.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/max98373.c b/sound/soc/codecs/max98373.c index 746c829312b8..1346a98ce8a1 100644 --- a/sound/soc/codecs/max98373.c +++ b/sound/soc/codecs/max98373.c @@ -28,11 +28,13 @@ static int max98373_dac_event(struct snd_soc_dapm_widget *w, regmap_update_bits(max98373->regmap, MAX98373_R20FF_GLOBAL_SHDN, MAX98373_GLOBAL_EN_MASK, 1); + usleep_range(30000, 31000); break; case SND_SOC_DAPM_POST_PMD: regmap_update_bits(max98373->regmap, MAX98373_R20FF_GLOBAL_SHDN, MAX98373_GLOBAL_EN_MASK, 0); + usleep_range(30000, 31000); max98373->tdm_mode = false; break; default: -- 2.30.2

1 0

[PATCH AUTOSEL 5.11 13/51] ASoC: max98373: Changed amp shutdown register as volatile
by Sasha Levin 12 Apr '21

12 Apr '21

From: Ryan Lee <ryans.lee(a)maximintegrated.com> [ Upstream commit a23f9099ff1541f15704e96b784d3846d2a4483d ] 0x20FF(amp global enable) register was defined as non-volatile, but it is not. Overheating, overcurrent can cause amp shutdown in hardware. 'regmap_write' compare register readback value before writing to avoid same value writing. 'regmap_read' just read cache not actual hardware value for the non-volatile register. When amp is internally shutdown by some reason, next 'AMP ON' command can be ignored because regmap think amp is already ON. Signed-off-by: Ryan Lee <ryans.lee(a)maximintegrated.com> Link: https://lore.kernel.org/r/20210325033555.29377-1-ryans.lee@maximintegrated.… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/max98373-i2c.c | 1 + sound/soc/codecs/max98373-sdw.c | 1 + 2 files changed, 2 insertions(+) diff --git a/sound/soc/codecs/max98373-i2c.c b/sound/soc/codecs/max98373-i2c.c index 85f6865019d4..ddb6436835d7 100644 --- a/sound/soc/codecs/max98373-i2c.c +++ b/sound/soc/codecs/max98373-i2c.c @@ -446,6 +446,7 @@ static bool max98373_volatile_reg(struct device *dev, unsigned int reg) case MAX98373_R2054_MEAS_ADC_PVDD_CH_READBACK: case MAX98373_R2055_MEAS_ADC_THERM_CH_READBACK: case MAX98373_R20B6_BDE_CUR_STATE_READBACK: + case MAX98373_R20FF_GLOBAL_SHDN: case MAX98373_R21FF_REV_ID: return true; default: diff --git a/sound/soc/codecs/max98373-sdw.c b/sound/soc/codecs/max98373-sdw.c index b8d471d79e93..1a1f97f24601 100644 --- a/sound/soc/codecs/max98373-sdw.c +++ b/sound/soc/codecs/max98373-sdw.c @@ -220,6 +220,7 @@ static bool max98373_volatile_reg(struct device *dev, unsigned int reg) case MAX98373_R2054_MEAS_ADC_PVDD_CH_READBACK: case MAX98373_R2055_MEAS_ADC_THERM_CH_READBACK: case MAX98373_R20B6_BDE_CUR_STATE_READBACK: + case MAX98373_R20FF_GLOBAL_SHDN: case MAX98373_R21FF_REV_ID: /* SoundWire Control Port Registers */ case MAX98373_R0040_SCP_INIT_STAT_1 ... MAX98373_R0070_SCP_FRAME_CTLR: -- 2.30.2

1 0

[PATCH v2] sound: rawmidi: Add framing mode
by David Henningsson 12 Apr '21

12 Apr '21

This commit adds a new framing mode that frames all MIDI data into 16-byte frames with a timestamp from the monotonic_raw clock. The main benefit is that we can get accurate timestamps even if userspace wakeup and processing is not immediate. Signed-off-by: David Henningsson <coding(a)diwic.se> --- v2: Fixed checkpatch errors. include/sound/rawmidi.h | 1 + include/uapi/sound/asound.h | 18 ++++++++++++++- sound/core/rawmidi.c | 45 ++++++++++++++++++++++++++++++++++++- 3 files changed, 62 insertions(+), 2 deletions(-) diff --git a/include/sound/rawmidi.h b/include/sound/rawmidi.h index 334842daa904..4ba5d2deec18 100644 --- a/include/sound/rawmidi.h +++ b/include/sound/rawmidi.h @@ -81,6 +81,7 @@ struct snd_rawmidi_substream { bool opened; /* open flag */ bool append; /* append flag (merge more streams) */ bool active_sensing; /* send active sensing when close */ + u8 framing; /* whether to frame data (for input) */ int use_count; /* use counter (for output) */ size_t bytes; struct snd_rawmidi *rmidi; diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h index 535a7229e1d9..f33076755025 100644 --- a/include/uapi/sound/asound.h +++ b/include/uapi/sound/asound.h @@ -736,12 +736,28 @@ struct snd_rawmidi_info { unsigned char reserved[64]; /* reserved for future use */ }; +enum { + SNDRV_RAWMIDI_FRAMING_NONE = 0, + SNDRV_RAWMIDI_FRAMING_TSTAMP_MONOTONIC_RAW, + SNDRV_RAWMIDI_FRAMING_LAST = SNDRV_RAWMIDI_FRAMING_TSTAMP_MONOTONIC_RAW, +}; + +#define SND_RAWMIDI_FRAMING_DATA_LENGTH 7 + +struct snd_rawmidi_framing_tstamp { + unsigned int tv_sec; /* seconds */ + unsigned int tv_nsec; /* nanoseconds */ + unsigned char length; + unsigned char data[SND_RAWMIDI_FRAMING_DATA_LENGTH]; +}; + struct snd_rawmidi_params { int stream; size_t buffer_size; /* queue size in bytes */ size_t avail_min; /* minimum avail bytes for wakeup */ unsigned int no_active_sensing: 1; /* do not send active sensing byte in close() */ - unsigned char reserved[16]; /* reserved for future use */ + unsigned char framing; /* For input data only, frame incoming data */ + unsigned char reserved[15]; /* reserved for future use */ }; #ifndef __KERNEL__ diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c index aca00af93afe..cd927ba178a6 100644 --- a/sound/core/rawmidi.c +++ b/sound/core/rawmidi.c @@ -721,6 +721,7 @@ int snd_rawmidi_input_params(struct snd_rawmidi_substream *substream, struct snd_rawmidi_params *params) { snd_rawmidi_drain_input(substream); + substream->framing = params->framing; return resize_runtime_buffer(substream->runtime, params, true); } EXPORT_SYMBOL(snd_rawmidi_input_params); @@ -963,6 +964,44 @@ static int snd_rawmidi_control_ioctl(struct snd_card *card, return -ENOIOCTLCMD; } +static int receive_with_tstamp_framing(struct snd_rawmidi_substream *substream, + const unsigned char *buffer, int src_count, struct timespec64 *tstamp) +{ + struct snd_rawmidi_runtime *runtime = substream->runtime; + struct snd_rawmidi_framing_tstamp frame; + struct snd_rawmidi_framing_tstamp *dest_ptr; + int dest_frames = 0; + int frame_size = sizeof(struct snd_rawmidi_framing_tstamp); + + frame.tv_sec = tstamp->tv_sec; + frame.tv_nsec = tstamp->tv_nsec; + if (snd_BUG_ON(runtime->hw_ptr & 15 || runtime->buffer_size & 15 || frame_size != 16)) + return -EINVAL; + + while (src_count > 0) { + if ((int)(runtime->buffer_size - runtime->avail) < frame_size) { + runtime->xruns += src_count; + return dest_frames * frame_size; + } + if (src_count >= SND_RAWMIDI_FRAMING_DATA_LENGTH) + frame.length = SND_RAWMIDI_FRAMING_DATA_LENGTH; + else { + frame.length = src_count; + memset(frame.data, 0, SND_RAWMIDI_FRAMING_DATA_LENGTH); + } + memcpy(frame.data, buffer, frame.length); + buffer += frame.length; + src_count -= frame.length; + dest_ptr = (struct snd_rawmidi_framing_tstamp *) (runtime->buffer + runtime->hw_ptr); + *dest_ptr = frame; + runtime->avail += frame_size; + runtime->hw_ptr += frame_size; + runtime->hw_ptr %= runtime->buffer_size; + dest_frames++; + } + return dest_frames * frame_size; +} + /** * snd_rawmidi_receive - receive the input data from the device * @substream: the rawmidi substream @@ -977,6 +1016,7 @@ int snd_rawmidi_receive(struct snd_rawmidi_substream *substream, const unsigned char *buffer, int count) { unsigned long flags; + struct timespec64 ts64; int result = 0, count1; struct snd_rawmidi_runtime *runtime = substream->runtime; @@ -988,7 +1028,10 @@ int snd_rawmidi_receive(struct snd_rawmidi_substream *substream, return -EINVAL; } spin_lock_irqsave(&runtime->lock, flags); - if (count == 1) { /* special case, faster code */ + if (substream->framing == SNDRV_RAWMIDI_FRAMING_TSTAMP_MONOTONIC_RAW) { + ktime_get_raw_ts64(&ts64); + result = receive_with_tstamp_framing(substream, buffer, count, &ts64); + } else if (count == 1) { /* special case, faster code */ substream->bytes++; if (runtime->avail < runtime->buffer_size) { runtime->buffer[runtime->hw_ptr++] = buffer[0]; -- 2.25.1

3 20

[PATCH] ASoC: fsl_sai: Don't use devm_regmap_init_mmio_clk
by Shengjiu Wang 12 Apr '21

12 Apr '21

When there is power domain bind with bus clock, The call flow: devm_regmap_init_mmio_clk - clk_prepare() - clk_pm_runtime_get() cause the power domain of clock always be enabled after regmap_init(). which impact the power consumption. So use devm_regmap_init_mmio instead of devm_regmap_init_mmio_clk, then explicitly enable clock when using by pm_runtime_get(), if CONFIG_PM=n, then fsl_sai_runtime_resume will be explicitly called. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> Signed-off-by: Viorel Suman <viorel.suman(a)nxp.com> --- sound/soc/fsl/fsl_sai.c | 48 +++++++++++++++++++++++++++-------------- 1 file changed, 32 insertions(+), 16 deletions(-) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 9e7893f91882..f2c70a31c7bb 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -987,6 +987,9 @@ static int fsl_sai_check_version(struct device *dev) return 0; } +static int fsl_sai_runtime_suspend(struct device *dev); +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; @@ -1019,24 +1022,21 @@ static int fsl_sai_probe(struct platform_device *pdev) ARRAY_SIZE(fsl_sai_reg_defaults_ofs8); } - sai->regmap = devm_regmap_init_mmio_clk(&pdev->dev, - "bus", base, &fsl_sai_regmap_config); - - /* Compatible with old DTB cases */ - if (IS_ERR(sai->regmap) && PTR_ERR(sai->regmap) != -EPROBE_DEFER) - sai->regmap = devm_regmap_init_mmio_clk(&pdev->dev, - "sai", base, &fsl_sai_regmap_config); + sai->regmap = devm_regmap_init_mmio(&pdev->dev, base, &fsl_sai_regmap_config); if (IS_ERR(sai->regmap)) { dev_err(&pdev->dev, "regmap init failed\n"); return PTR_ERR(sai->regmap); } - /* No error out for old DTB cases but only mark the clock NULL */ sai->bus_clk = devm_clk_get(&pdev->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"); if (IS_ERR(sai->bus_clk)) { dev_err(&pdev->dev, "failed to get bus clock: %ld\n", PTR_ERR(sai->bus_clk)); - sai->bus_clk = NULL; + /* -EPROBE_DEFER */ + return PTR_ERR(sai->bus_clk); } for (i = 1; i < FSL_SAI_MCLK_MAX; i++) { @@ -1117,6 +1117,18 @@ 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); + if (ret) + goto err_pm_disable; + } + + ret = pm_runtime_get_sync(&pdev->dev); + if (ret < 0) { + pm_runtime_put_noidle(&pdev->dev); + goto err_pm_get_sync; + } /* Get sai version */ ret = fsl_sai_check_version(&pdev->dev); @@ -1130,26 +1142,30 @@ static int fsl_sai_probe(struct platform_device *pdev) FSL_SAI_MCTL_MCLK_EN, FSL_SAI_MCTL_MCLK_EN); } - pm_runtime_enable(&pdev->dev); - regcache_cache_only(sai->regmap, true); + ret = pm_runtime_put_sync(&pdev->dev); + if (ret < 0) + goto err_pm_get_sync; ret = devm_snd_soc_register_component(&pdev->dev, &fsl_component, &sai->cpu_dai_drv, 1); if (ret) - goto err_pm_disable; + goto err_pm_get_sync; if (sai->soc_data->use_imx_pcm) { ret = imx_pcm_dma_init(pdev, IMX_SAI_DMABUF_SIZE); if (ret) - goto err_pm_disable; + goto err_pm_get_sync; } else { ret = devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0); if (ret) - goto err_pm_disable; + goto err_pm_get_sync; } return ret; +err_pm_get_sync: + if (!pm_runtime_status_suspended(&pdev->dev)) + fsl_sai_runtime_suspend(&pdev->dev); err_pm_disable: pm_runtime_disable(&pdev->dev); @@ -1159,6 +1175,8 @@ static int fsl_sai_probe(struct platform_device *pdev) static int fsl_sai_remove(struct platform_device *pdev) { pm_runtime_disable(&pdev->dev); + if (!pm_runtime_status_suspended(&pdev->dev)) + fsl_sai_runtime_suspend(&pdev->dev); return 0; } @@ -1219,7 +1237,6 @@ static const struct of_device_id fsl_sai_ids[] = { }; MODULE_DEVICE_TABLE(of, fsl_sai_ids); -#ifdef CONFIG_PM static int fsl_sai_runtime_suspend(struct device *dev) { struct fsl_sai *sai = dev_get_drvdata(dev); @@ -1292,7 +1309,6 @@ static int fsl_sai_runtime_resume(struct device *dev) return ret; } -#endif /* CONFIG_PM */ static const struct dev_pm_ops fsl_sai_pm_ops = { SET_RUNTIME_PM_OPS(fsl_sai_runtime_suspend, -- 2.27.0

4 4

wrong alsa time fix
by GitHub issues - edited 12 Apr '21

12 Apr '21

alsa-project/alsa-lib issue #132 was edited from circinusX1: @@ -42,7 +42,7 @@ long long timestamp2ns(snd_htimestamp_t t) { long long nsec; - nsec = t.tv_sec * 1000000000; + nsec = (long long)t.tv_sec * 1000000000; nsec += t.tv_nsec; Issue URL : https://github.com/alsa-project/alsa-lib/issues/132 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

wront alsa time
by GitHub issues - opened 12 Apr '21

12 Apr '21

alsa-project/alsa-lib issue #132 was opened from circinusX1: @@ -42,7 +42,7 @@ long long timestamp2ns(snd_htimestamp_t t) { long long nsec; - nsec = t.tv_sec * 1000000000; + nsec = (long long)t.tv_sec * 1000000000; nsec += t.tv_nsec; Issue URL : https://github.com/alsa-project/alsa-lib/issues/132 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH -next] ALSA: virtio: use module_virtio_driver() to simplify the code
by Chen Huang 12 Apr '21

12 Apr '21

module_virtio_driver() makes the code simpler by eliminating boilerplate code. Signed-off-by: Chen Huang <chenhuang5(a)huawei.com> --- sound/virtio/virtio_card.c | 12 +----------- 1 file changed, 1 insertion(+), 11 deletions(-) diff --git a/sound/virtio/virtio_card.c b/sound/virtio/virtio_card.c index ae9128063917..150ab3e37013 100644 --- a/sound/virtio/virtio_card.c +++ b/sound/virtio/virtio_card.c @@ -432,17 +432,7 @@ static struct virtio_driver virtsnd_driver = { #endif }; -static int __init init(void) -{ - return register_virtio_driver(&virtsnd_driver); -} -module_init(init); - -static void __exit fini(void) -{ - unregister_virtio_driver(&virtsnd_driver); -} -module_exit(fini); +module_virtio_driver(virtsnd_driver); MODULE_DEVICE_TABLE(virtio, id_table); MODULE_DESCRIPTION("Virtio sound card driver"); -- 2.17.1

3 2

[PATCH v7 2/2] ASoC: rt715:add micmute led state control supports
by Perry Yuan 12 Apr '21

12 Apr '21

From: Perry Yuan <perry_yuan(a)dell.com> Some new Dell system is going to support audio internal micphone privacy setting from hardware level with micmute led state changing When micmute hotkey pressed by user, soft mute will need to be enabled firstly in case of pop noise, and codec driver need to react to mic mute event to EC(embedded controller) notifying that SW mute is completed Then EC will do the hardware mute physically within the timeout reached This patch allow codec rt715 and rt715 sdca driver to change the local micmute led state. Dell privacy led trigger driver will ack EC when micmute key pressed through this micphone led control interface like hda_generic provided ACPI method defined in dell-privacy micmute led trigger will be called for notifying the EC that software mute has been completed, then hardware audio circuit solution controlled by EC will switch the audio input source off/on Signed-off-by: Perry Yuan <perry_yuan(a)dell.com> -------- v5 -> v6: * addresed review comments from Jaroslav * add quirks for micmute led control as short term solution to control micmute led state change * add comments for the invert checking v4 -> v5: * rebase to latest 5.12 rc4 upstream kernel v3 -> v4: * remove unused debug log * remove compile flag of DELL privacy * move the micmute_led to local from rt715_priv * when Jaroslav upstream his gerneric LED trigger driver,I will rebase this patch,please consider merge this at first https://lore.kernel.org/alsa-devel/20210211111400.1131020-1-perex@perex.cz/ v2 -> v3: * simplify the patch to reuse some val value * add more detail to the commit info v1 -> v2: * fix some format issue -------- --- sound/soc/codecs/rt715-sdca.c | 42 ++++++++++++++++++++++++++++++++++ sound/soc/codecs/rt715.c | 43 +++++++++++++++++++++++++++++++++++ 2 files changed, 85 insertions(+) diff --git a/sound/soc/codecs/rt715-sdca.c b/sound/soc/codecs/rt715-sdca.c index 20528afbdc57..47dc31f7f3e3 100644 --- a/sound/soc/codecs/rt715-sdca.c +++ b/sound/soc/codecs/rt715-sdca.c @@ -11,12 +11,14 @@ #include <linux/moduleparam.h> #include <linux/kernel.h> #include <linux/init.h> +#include <linux/leds.h> #include <linux/pm_runtime.h> #include <linux/pm.h> #include <linux/soundwire/sdw.h> #include <linux/regmap.h> #include <linux/slab.h> #include <linux/platform_device.h> +#include <linux/dmi.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -344,6 +346,32 @@ static int rt715_sdca_get_volsw(struct snd_kcontrol *kcontrol, return 0; } +static bool micmute_led_set; +static int dmi_matched(const struct dmi_system_id *dmi) +{ + micmute_led_set = 1; + return 1; +} + +/* Some systems will need to use this to trigger mic mute LED state changed */ +static const struct dmi_system_id micmute_led_dmi_table[] = { + { + .callback = dmi_matched, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc."), + DMI_MATCH(DMI_PRODUCT_SKU, "0A32"), + }, + }, + { + .callback = dmi_matched, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc."), + DMI_MATCH(DMI_PRODUCT_SKU, "0A3E"), + }, + }, + {}, +}; + static int rt715_sdca_put_volsw(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -358,6 +386,7 @@ static int rt715_sdca_put_volsw(struct snd_kcontrol *kcontrol, unsigned int mask = (1 << fls(max)) - 1; unsigned int invert = p->invert; int err; + bool micmute_led; for (i = 0; i < 4; i++) { if (ucontrol->value.integer.value[i] != rt715->kctl_switch_orig[i]) { @@ -394,6 +423,18 @@ static int rt715_sdca_put_volsw(struct snd_kcontrol *kcontrol, return err; } + /* Micmute LED state changed by muted/unmute switch + * to keep syncing with actual hardware mic mute led state + * invert will be checked to change the state switch + */ + if (invert && micmute_led_set) { + if (ucontrol->value.integer.value[0] || ucontrol->value.integer.value[1]) + micmute_led = LED_OFF; + else + micmute_led = LED_ON; + ledtrig_audio_set(LED_AUDIO_MICMUTE, micmute_led); + } + return k_changed; } @@ -1066,6 +1107,7 @@ int rt715_sdca_io_init(struct device *dev, struct sdw_slave *slave) pm_runtime_mark_last_busy(&slave->dev); pm_runtime_put_autosuspend(&slave->dev); + dmi_check_system(micmute_led_dmi_table); return 0; } diff --git a/sound/soc/codecs/rt715.c b/sound/soc/codecs/rt715.c index 34c3357e943b..59d05b28f52e 100644 --- a/sound/soc/codecs/rt715.c +++ b/sound/soc/codecs/rt715.c @@ -13,6 +13,7 @@ #include <linux/init.h> #include <linux/delay.h> #include <linux/i2c.h> +#include <linux/leds.h> #include <linux/pm_runtime.h> #include <linux/pm.h> #include <linux/soundwire/sdw.h> @@ -25,6 +26,7 @@ #include <linux/of.h> #include <linux/of_gpio.h> #include <linux/of_device.h> +#include <linux/dmi.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -70,6 +72,32 @@ static void rt715_get_gain(struct rt715_priv *rt715, unsigned int addr_h, pr_err("Failed to get L channel gain.\n"); } +static bool micmute_led_set; +static int dmi_matched(const struct dmi_system_id *dmi) +{ + micmute_led_set = 1; + return 1; +} + +/* Some systems will need to use this to trigger mic mute LED state changed */ +static const struct dmi_system_id micmute_led_dmi_table[] = { + { + .callback = dmi_matched, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc."), + DMI_MATCH(DMI_PRODUCT_SKU, "0A32"), + }, + }, + { + .callback = dmi_matched, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc."), + DMI_MATCH(DMI_PRODUCT_SKU, "0A3E"), + }, + }, + {}, +}; + /* For Verb-Set Amplifier Gain (Verb ID = 3h) */ static int rt715_set_amp_gain_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -88,6 +116,7 @@ static int rt715_set_amp_gain_put(struct snd_kcontrol *kcontrol, RT715_SET_GAIN_MIX_ADC2_L}; unsigned int addr_h, addr_l, val_h, val_ll, val_lr; unsigned int read_ll, read_rl, i, j, loop_cnt; + bool micmute_led; if (strstr(ucontrol->id.name, "Main Capture Switch") || strstr(ucontrol->id.name, "Main Capture Volume")) @@ -185,6 +214,19 @@ static int rt715_set_amp_gain_put(struct snd_kcontrol *kcontrol, if (dapm->bias_level <= SND_SOC_BIAS_STANDBY) regmap_write(rt715->regmap, RT715_SET_AUDIO_POWER_STATE, AC_PWRST_D3); + + /* Micmute LED state changed by muted/unmute switch + * to keep syncing with actual hardware mic mute led state + * invert will be checked to change the state switch + */ + if (mc->invert && micmute_led_set) { + if (ucontrol->value.integer.value[0] || ucontrol->value.integer.value[1]) + micmute_led = LED_OFF; + else + micmute_led = LED_ON; + ledtrig_audio_set(LED_AUDIO_MICMUTE, micmute_led); + } + return 0; } @@ -901,6 +943,7 @@ int rt715_io_init(struct device *dev, struct sdw_slave *slave) pm_runtime_mark_last_busy(&slave->dev); pm_runtime_put_autosuspend(&slave->dev); + dmi_check_system(micmute_led_dmi_table); return 0; } -- 2.25.1

2 1

[PATCH v6 2/2] ASoC: rt715:add micmute led state control supports
by Perry Yuan 12 Apr '21

12 Apr '21

From: Perry Yuan <perry_yuan(a)dell.com> Some new Dell system is going to support audio internal micphone privacy setting from hardware level with micmute led state changing When micmute hotkey pressed by user, soft mute will need to be enabled firstly in case of pop noise, and codec driver need to react to mic mute event to EC(embedded controller) notifying that SW mute is completed Then EC will do the hardware mute physically within the timeout reached This patch allow codec rt715 and rt715 sdca driver to change the local micmute led state. Dell privacy led trigger driver will ack EC when micmute key pressed through this micphone led control interface like hda_generic provided ACPI method defined in dell-privacy micmute led trigger will be called for notifying the EC that software mute has been completed, then hardware audio circuit solution controlled by EC will switch the audio input source off/on Signed-off-by: Perry Yuan <perry_yuan(a)dell.com> -------- v5 -> v6: * addresed review comments from Jaroslav * add quirks for micmute led control as short term solution to control micmute led state change v4 -> v5: * rebase to latest 5.12 rc4 upstream kernel v3 -> v4: * remove unused debug log * remove compile flag of DELL privacy * move the micmute_led to local from rt715_priv * when Jaroslav upstream his gerneric LED trigger driver,I will rebase this patch,please consider merge this at first https://lore.kernel.org/alsa-devel/20210211111400.1131020-1-perex@perex.cz/ v2 -> v3: * simplify the patch to reuse some val value * add more detail to the commit info v1 -> v2: * fix some format issue -------- --- sound/soc/codecs/rt715-sdca.c | 41 ++++++++++++++++++++++++++++++++++- sound/soc/codecs/rt715.c | 41 +++++++++++++++++++++++++++++++++++ 2 files changed, 81 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt715-sdca.c b/sound/soc/codecs/rt715-sdca.c index 20528afbdc57..7bd7ad0ba7d7 100644 --- a/sound/soc/codecs/rt715-sdca.c +++ b/sound/soc/codecs/rt715-sdca.c @@ -11,12 +11,14 @@ #include <linux/moduleparam.h> #include <linux/kernel.h> #include <linux/init.h> +#include <linux/leds.h> #include <linux/pm_runtime.h> #include <linux/pm.h> #include <linux/soundwire/sdw.h> #include <linux/regmap.h> #include <linux/slab.h> #include <linux/platform_device.h> +#include <linux/dmi.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -344,6 +346,34 @@ static int rt715_sdca_get_volsw(struct snd_kcontrol *kcontrol, return 0; } +static bool micmute_led_set; +static int dmi_matched(const struct dmi_system_id *dmi) +{ + micmute_led_set = 1; + return 1; +} + +/* Some systems will need to use this to trigger mic mute LED state changed */ +static const struct dmi_system_id micmute_led_dmi_table[] = { + { + .callback = dmi_matched, + .ident = "Dell Latitude 9420", + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc."), + DMI_MATCH(DMI_PRODUCT_NAME, "Latitude 9420"), + }, + }, + { + .callback = dmi_matched, + .ident = "Dell Latitude 9520", + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc."), + DMI_MATCH(DMI_PRODUCT_NAME, "Latitude 9520"), + }, + }, + {}, +}; + static int rt715_sdca_put_volsw(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -358,6 +388,7 @@ static int rt715_sdca_put_volsw(struct snd_kcontrol *kcontrol, unsigned int mask = (1 << fls(max)) - 1; unsigned int invert = p->invert; int err; + bool micmute_led; for (i = 0; i < 4; i++) { if (ucontrol->value.integer.value[i] != rt715->kctl_switch_orig[i]) { @@ -393,7 +424,15 @@ static int rt715_sdca_put_volsw(struct snd_kcontrol *kcontrol, if (err < 0) return err; } - + /* Micmute LED state changed by muted/unmute switch */ + dmi_check_system(micmute_led_dmi_table); + if (invert && micmute_led_set) { + if (ucontrol->value.integer.value[0] || ucontrol->value.integer.value[1]) + micmute_led = LED_OFF; + else + micmute_led = LED_ON; + ledtrig_audio_set(LED_AUDIO_MICMUTE, micmute_led); + } return k_changed; } diff --git a/sound/soc/codecs/rt715.c b/sound/soc/codecs/rt715.c index 34c3357e943b..490cf9e63c2d 100644 --- a/sound/soc/codecs/rt715.c +++ b/sound/soc/codecs/rt715.c @@ -13,6 +13,7 @@ #include <linux/init.h> #include <linux/delay.h> #include <linux/i2c.h> +#include <linux/leds.h> #include <linux/pm_runtime.h> #include <linux/pm.h> #include <linux/soundwire/sdw.h> @@ -25,6 +26,7 @@ #include <linux/of.h> #include <linux/of_gpio.h> #include <linux/of_device.h> +#include <linux/dmi.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -70,6 +72,34 @@ static void rt715_get_gain(struct rt715_priv *rt715, unsigned int addr_h, pr_err("Failed to get L channel gain.\n"); } +static bool micmute_led_set; +static int dmi_matched(const struct dmi_system_id *dmi) +{ + micmute_led_set = 1; + return 1; +} + +/* Some systems will need to use this to trigger mic mute LED state changed */ +static const struct dmi_system_id micmute_led_dmi_table[] = { + { + .callback = dmi_matched, + .ident = "Dell Latitude 9420", + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc."), + DMI_MATCH(DMI_PRODUCT_NAME, "Latitude 9420"), + }, + }, + { + .callback = dmi_matched, + .ident = "Dell Latitude 9520", + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc."), + DMI_MATCH(DMI_PRODUCT_NAME, "Latitude 9520"), + }, + }, + {}, +}; + /* For Verb-Set Amplifier Gain (Verb ID = 3h) */ static int rt715_set_amp_gain_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -88,6 +118,7 @@ static int rt715_set_amp_gain_put(struct snd_kcontrol *kcontrol, RT715_SET_GAIN_MIX_ADC2_L}; unsigned int addr_h, addr_l, val_h, val_ll, val_lr; unsigned int read_ll, read_rl, i, j, loop_cnt; + bool micmute_led; if (strstr(ucontrol->id.name, "Main Capture Switch") || strstr(ucontrol->id.name, "Main Capture Volume")) @@ -185,6 +216,16 @@ static int rt715_set_amp_gain_put(struct snd_kcontrol *kcontrol, if (dapm->bias_level <= SND_SOC_BIAS_STANDBY) regmap_write(rt715->regmap, RT715_SET_AUDIO_POWER_STATE, AC_PWRST_D3); + + /* Micmute LED state changed by muted/unmute switch */ + dmi_check_system(micmute_led_dmi_table); + if (mc->invert && micmute_led_set) { + if (ucontrol->value.integer.value[0] || ucontrol->value.integer.value[1]) + micmute_led = LED_OFF; + else + micmute_led = LED_ON; + ledtrig_audio_set(LED_AUDIO_MICMUTE, micmute_led); + } return 0; } -- 2.25.1

3 2

[PATCH v7 0/2] hardware-privacy-implementation-for-dell-laptop
by Perry Yuan 12 Apr '21

12 Apr '21

From: Perry Yuan <perry_yuan(a)dell.com> Hi All, This patch set is a new driver for Dell mobile platform that has the hardware privacy feature including micmute,camera mute. For micmute led control, the hotkey is Fn + F4, it is a hardware based mute state, and new privacy will prevent micphone void input from hardware layer, any application cannot get voice data when micmute activated. Camera mute use a new hardware design to control the camrea shutter. When video is muted, no OS application should be functionally able to capture images of the person/environment in front of the system Older history: [1]https://patchwork.kernel.org/project/platform-driver-x86/patch/2020122813… [2]https://patchwork.kernel.org/project/alsa-devel/patch/20201103125859.8759… [3]https://www.spinics.net/lists/alsa-devel/msg120537.html [4]https://github.com/thesofproject/linux/pull/2660 [5]https://github.com/thesofproject/linux/issues/2496 Perry Yuan (2): platform/x86: dell-privacy: Add support for Dell hardware privacy ASoC: rt715:add micmute led state control supports .../testing/sysfs-platform-dell-privacy-wmi | 55 +++ drivers/platform/x86/dell/Kconfig | 14 + drivers/platform/x86/dell/Makefile | 1 + drivers/platform/x86/dell/dell-laptop.c | 23 +- drivers/platform/x86/dell/dell-privacy-wmi.c | 394 ++++++++++++++++++ drivers/platform/x86/dell/dell-privacy-wmi.h | 23 + drivers/platform/x86/dell/dell-wmi.c | 8 +- sound/soc/codecs/rt715-sdca.c | 42 ++ sound/soc/codecs/rt715.c | 43 ++ 9 files changed, 596 insertions(+), 7 deletions(-) create mode 100644 Documentation/ABI/testing/sysfs-platform-dell-privacy-wmi create mode 100644 drivers/platform/x86/dell/dell-privacy-wmi.c create mode 100644 drivers/platform/x86/dell/dell-privacy-wmi.h -- 2.25.1

1 0

[PATCH 4.19 42/66] ASoC: sunxi: sun4i-codec: fill ASoC card owner
by Greg Kroah-Hartman 12 Apr '21

12 Apr '21

From: Bastian Germann <bage(a)linutronix.de> [ Upstream commit 7c0d6e482062eb5c06ecccfab340abc523bdca00 ] card->owner is a required property and since commit 81033c6b584b ("ALSA: core: Warn on empty module") a warning is issued if it is empty. Add it. This fixes following warning observed on Lamobo R1: WARNING: CPU: 1 PID: 190 at sound/core/init.c:207 snd_card_new+0x430/0x480 [snd] Modules linked in: sun4i_codec(E+) sun4i_backend(E+) snd_soc_core(E) ... CPU: 1 PID: 190 Comm: systemd-udevd Tainted: G C E 5.10.0-1-armmp #1 Debian 5.10.4-1 Hardware name: Allwinner sun7i (A20) Family Call trace: (snd_card_new [snd]) (snd_soc_bind_card [snd_soc_core]) (snd_soc_register_card [snd_soc_core]) (sun4i_codec_probe [sun4i_codec]) Fixes: 45fb6b6f2aa3 ("ASoC: sunxi: add support for the on-chip codec on early Allwinner SoCs") Related: commit 3c27ea23ffb4 ("ASoC: qcom: Set card->owner to avoid warnings") Related: commit ec653df2a0cb ("drm/vc4/vc4_hdmi: fill ASoC card owner") Cc: linux-arm-kernel(a)lists.infradead.org Cc: alsa-devel(a)alsa-project.org Signed-off-by: Bastian Germann <bage(a)linutronix.de> Link: https://lore.kernel.org/r/20210331151843.30583-1-bage@linutronix.de Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/sunxi/sun4i-codec.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/soc/sunxi/sun4i-codec.c b/sound/soc/sunxi/sun4i-codec.c index 9a3cb7704810..16e8f74288a5 100644 --- a/sound/soc/sunxi/sun4i-codec.c +++ b/sound/soc/sunxi/sun4i-codec.c @@ -1235,6 +1235,7 @@ static struct snd_soc_card *sun4i_codec_create_card(struct device *dev) return ERR_PTR(-ENOMEM); card->dev = dev; + card->owner = THIS_MODULE; card->name = "sun4i-codec"; card->dapm_widgets = sun4i_codec_card_dapm_widgets; card->num_dapm_widgets = ARRAY_SIZE(sun4i_codec_card_dapm_widgets); @@ -1267,6 +1268,7 @@ static struct snd_soc_card *sun6i_codec_create_card(struct device *dev) return ERR_PTR(-ENOMEM); card->dev = dev; + card->owner = THIS_MODULE; card->name = "A31 Audio Codec"; card->dapm_widgets = sun6i_codec_card_dapm_widgets; card->num_dapm_widgets = ARRAY_SIZE(sun6i_codec_card_dapm_widgets); @@ -1320,6 +1322,7 @@ static struct snd_soc_card *sun8i_a23_codec_create_card(struct device *dev) return ERR_PTR(-ENOMEM); card->dev = dev; + card->owner = THIS_MODULE; card->name = "A23 Audio Codec"; card->dapm_widgets = sun6i_codec_card_dapm_widgets; card->num_dapm_widgets = ARRAY_SIZE(sun6i_codec_card_dapm_widgets); @@ -1358,6 +1361,7 @@ static struct snd_soc_card *sun8i_h3_codec_create_card(struct device *dev) return ERR_PTR(-ENOMEM); card->dev = dev; + card->owner = THIS_MODULE; card->name = "H3 Audio Codec"; card->dapm_widgets = sun6i_codec_card_dapm_widgets; card->num_dapm_widgets = ARRAY_SIZE(sun6i_codec_card_dapm_widgets); @@ -1396,6 +1400,7 @@ static struct snd_soc_card *sun8i_v3s_codec_create_card(struct device *dev) return ERR_PTR(-ENOMEM); card->dev = dev; + card->owner = THIS_MODULE; card->name = "V3s Audio Codec"; card->dapm_widgets = sun6i_codec_card_dapm_widgets; card->num_dapm_widgets = ARRAY_SIZE(sun6i_codec_card_dapm_widgets); -- 2.30.2

1 0

[PATCH 5.11 142/210] ASoC: sunxi: sun4i-codec: fill ASoC card owner
by Greg Kroah-Hartman 12 Apr '21

12 Apr '21

From: Bastian Germann <bage(a)linutronix.de> [ Upstream commit 7c0d6e482062eb5c06ecccfab340abc523bdca00 ] card->owner is a required property and since commit 81033c6b584b ("ALSA: core: Warn on empty module") a warning is issued if it is empty. Add it. This fixes following warning observed on Lamobo R1: WARNING: CPU: 1 PID: 190 at sound/core/init.c:207 snd_card_new+0x430/0x480 [snd] Modules linked in: sun4i_codec(E+) sun4i_backend(E+) snd_soc_core(E) ... CPU: 1 PID: 190 Comm: systemd-udevd Tainted: G C E 5.10.0-1-armmp #1 Debian 5.10.4-1 Hardware name: Allwinner sun7i (A20) Family Call trace: (snd_card_new [snd]) (snd_soc_bind_card [snd_soc_core]) (snd_soc_register_card [snd_soc_core]) (sun4i_codec_probe [sun4i_codec]) Fixes: 45fb6b6f2aa3 ("ASoC: sunxi: add support for the on-chip codec on early Allwinner SoCs") Related: commit 3c27ea23ffb4 ("ASoC: qcom: Set card->owner to avoid warnings") Related: commit ec653df2a0cb ("drm/vc4/vc4_hdmi: fill ASoC card owner") Cc: linux-arm-kernel(a)lists.infradead.org Cc: alsa-devel(a)alsa-project.org Signed-off-by: Bastian Germann <bage(a)linutronix.de> Link: https://lore.kernel.org/r/20210331151843.30583-1-bage@linutronix.de Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/sunxi/sun4i-codec.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/soc/sunxi/sun4i-codec.c b/sound/soc/sunxi/sun4i-codec.c index 6c13cc84b3fb..2173991c13db 100644 --- a/sound/soc/sunxi/sun4i-codec.c +++ b/sound/soc/sunxi/sun4i-codec.c @@ -1364,6 +1364,7 @@ static struct snd_soc_card *sun4i_codec_create_card(struct device *dev) return ERR_PTR(-ENOMEM); card->dev = dev; + card->owner = THIS_MODULE; card->name = "sun4i-codec"; card->dapm_widgets = sun4i_codec_card_dapm_widgets; card->num_dapm_widgets = ARRAY_SIZE(sun4i_codec_card_dapm_widgets); @@ -1396,6 +1397,7 @@ static struct snd_soc_card *sun6i_codec_create_card(struct device *dev) return ERR_PTR(-ENOMEM); card->dev = dev; + card->owner = THIS_MODULE; card->name = "A31 Audio Codec"; card->dapm_widgets = sun6i_codec_card_dapm_widgets; card->num_dapm_widgets = ARRAY_SIZE(sun6i_codec_card_dapm_widgets); @@ -1449,6 +1451,7 @@ static struct snd_soc_card *sun8i_a23_codec_create_card(struct device *dev) return ERR_PTR(-ENOMEM); card->dev = dev; + card->owner = THIS_MODULE; card->name = "A23 Audio Codec"; card->dapm_widgets = sun6i_codec_card_dapm_widgets; card->num_dapm_widgets = ARRAY_SIZE(sun6i_codec_card_dapm_widgets); @@ -1487,6 +1490,7 @@ static struct snd_soc_card *sun8i_h3_codec_create_card(struct device *dev) return ERR_PTR(-ENOMEM); card->dev = dev; + card->owner = THIS_MODULE; card->name = "H3 Audio Codec"; card->dapm_widgets = sun6i_codec_card_dapm_widgets; card->num_dapm_widgets = ARRAY_SIZE(sun6i_codec_card_dapm_widgets); @@ -1525,6 +1529,7 @@ static struct snd_soc_card *sun8i_v3s_codec_create_card(struct device *dev) return ERR_PTR(-ENOMEM); card->dev = dev; + card->owner = THIS_MODULE; card->name = "V3s Audio Codec"; card->dapm_widgets = sun6i_codec_card_dapm_widgets; card->num_dapm_widgets = ARRAY_SIZE(sun6i_codec_card_dapm_widgets); -- 2.30.2

1 0

[PATCH 5.4 068/111] ASoC: sunxi: sun4i-codec: fill ASoC card owner
by Greg Kroah-Hartman 12 Apr '21

12 Apr '21

From: Bastian Germann <bage(a)linutronix.de> [ Upstream commit 7c0d6e482062eb5c06ecccfab340abc523bdca00 ] card->owner is a required property and since commit 81033c6b584b ("ALSA: core: Warn on empty module") a warning is issued if it is empty. Add it. This fixes following warning observed on Lamobo R1: WARNING: CPU: 1 PID: 190 at sound/core/init.c:207 snd_card_new+0x430/0x480 [snd] Modules linked in: sun4i_codec(E+) sun4i_backend(E+) snd_soc_core(E) ... CPU: 1 PID: 190 Comm: systemd-udevd Tainted: G C E 5.10.0-1-armmp #1 Debian 5.10.4-1 Hardware name: Allwinner sun7i (A20) Family Call trace: (snd_card_new [snd]) (snd_soc_bind_card [snd_soc_core]) (snd_soc_register_card [snd_soc_core]) (sun4i_codec_probe [sun4i_codec]) Fixes: 45fb6b6f2aa3 ("ASoC: sunxi: add support for the on-chip codec on early Allwinner SoCs") Related: commit 3c27ea23ffb4 ("ASoC: qcom: Set card->owner to avoid warnings") Related: commit ec653df2a0cb ("drm/vc4/vc4_hdmi: fill ASoC card owner") Cc: linux-arm-kernel(a)lists.infradead.org Cc: alsa-devel(a)alsa-project.org Signed-off-by: Bastian Germann <bage(a)linutronix.de> Link: https://lore.kernel.org/r/20210331151843.30583-1-bage@linutronix.de Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/sunxi/sun4i-codec.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/soc/sunxi/sun4i-codec.c b/sound/soc/sunxi/sun4i-codec.c index ee448d5e07a6..c4021d6ac9df 100644 --- a/sound/soc/sunxi/sun4i-codec.c +++ b/sound/soc/sunxi/sun4i-codec.c @@ -1364,6 +1364,7 @@ static struct snd_soc_card *sun4i_codec_create_card(struct device *dev) return ERR_PTR(-ENOMEM); card->dev = dev; + card->owner = THIS_MODULE; card->name = "sun4i-codec"; card->dapm_widgets = sun4i_codec_card_dapm_widgets; card->num_dapm_widgets = ARRAY_SIZE(sun4i_codec_card_dapm_widgets); @@ -1396,6 +1397,7 @@ static struct snd_soc_card *sun6i_codec_create_card(struct device *dev) return ERR_PTR(-ENOMEM); card->dev = dev; + card->owner = THIS_MODULE; card->name = "A31 Audio Codec"; card->dapm_widgets = sun6i_codec_card_dapm_widgets; card->num_dapm_widgets = ARRAY_SIZE(sun6i_codec_card_dapm_widgets); @@ -1449,6 +1451,7 @@ static struct snd_soc_card *sun8i_a23_codec_create_card(struct device *dev) return ERR_PTR(-ENOMEM); card->dev = dev; + card->owner = THIS_MODULE; card->name = "A23 Audio Codec"; card->dapm_widgets = sun6i_codec_card_dapm_widgets; card->num_dapm_widgets = ARRAY_SIZE(sun6i_codec_card_dapm_widgets); @@ -1487,6 +1490,7 @@ static struct snd_soc_card *sun8i_h3_codec_create_card(struct device *dev) return ERR_PTR(-ENOMEM); card->dev = dev; + card->owner = THIS_MODULE; card->name = "H3 Audio Codec"; card->dapm_widgets = sun6i_codec_card_dapm_widgets; card->num_dapm_widgets = ARRAY_SIZE(sun6i_codec_card_dapm_widgets); @@ -1525,6 +1529,7 @@ static struct snd_soc_card *sun8i_v3s_codec_create_card(struct device *dev) return ERR_PTR(-ENOMEM); card->dev = dev; + card->owner = THIS_MODULE; card->name = "V3s Audio Codec"; card->dapm_widgets = sun6i_codec_card_dapm_widgets; card->num_dapm_widgets = ARRAY_SIZE(sun6i_codec_card_dapm_widgets); -- 2.30.2

1 0

[PATCH 5.10 130/188] ASoC: sunxi: sun4i-codec: fill ASoC card owner
by Greg Kroah-Hartman 12 Apr '21

12 Apr '21

From: Bastian Germann <bage(a)linutronix.de> [ Upstream commit 7c0d6e482062eb5c06ecccfab340abc523bdca00 ] card->owner is a required property and since commit 81033c6b584b ("ALSA: core: Warn on empty module") a warning is issued if it is empty. Add it. This fixes following warning observed on Lamobo R1: WARNING: CPU: 1 PID: 190 at sound/core/init.c:207 snd_card_new+0x430/0x480 [snd] Modules linked in: sun4i_codec(E+) sun4i_backend(E+) snd_soc_core(E) ... CPU: 1 PID: 190 Comm: systemd-udevd Tainted: G C E 5.10.0-1-armmp #1 Debian 5.10.4-1 Hardware name: Allwinner sun7i (A20) Family Call trace: (snd_card_new [snd]) (snd_soc_bind_card [snd_soc_core]) (snd_soc_register_card [snd_soc_core]) (sun4i_codec_probe [sun4i_codec]) Fixes: 45fb6b6f2aa3 ("ASoC: sunxi: add support for the on-chip codec on early Allwinner SoCs") Related: commit 3c27ea23ffb4 ("ASoC: qcom: Set card->owner to avoid warnings") Related: commit ec653df2a0cb ("drm/vc4/vc4_hdmi: fill ASoC card owner") Cc: linux-arm-kernel(a)lists.infradead.org Cc: alsa-devel(a)alsa-project.org Signed-off-by: Bastian Germann <bage(a)linutronix.de> Link: https://lore.kernel.org/r/20210331151843.30583-1-bage@linutronix.de Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/sunxi/sun4i-codec.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/soc/sunxi/sun4i-codec.c b/sound/soc/sunxi/sun4i-codec.c index 6c13cc84b3fb..2173991c13db 100644 --- a/sound/soc/sunxi/sun4i-codec.c +++ b/sound/soc/sunxi/sun4i-codec.c @@ -1364,6 +1364,7 @@ static struct snd_soc_card *sun4i_codec_create_card(struct device *dev) return ERR_PTR(-ENOMEM); card->dev = dev; + card->owner = THIS_MODULE; card->name = "sun4i-codec"; card->dapm_widgets = sun4i_codec_card_dapm_widgets; card->num_dapm_widgets = ARRAY_SIZE(sun4i_codec_card_dapm_widgets); @@ -1396,6 +1397,7 @@ static struct snd_soc_card *sun6i_codec_create_card(struct device *dev) return ERR_PTR(-ENOMEM); card->dev = dev; + card->owner = THIS_MODULE; card->name = "A31 Audio Codec"; card->dapm_widgets = sun6i_codec_card_dapm_widgets; card->num_dapm_widgets = ARRAY_SIZE(sun6i_codec_card_dapm_widgets); @@ -1449,6 +1451,7 @@ static struct snd_soc_card *sun8i_a23_codec_create_card(struct device *dev) return ERR_PTR(-ENOMEM); card->dev = dev; + card->owner = THIS_MODULE; card->name = "A23 Audio Codec"; card->dapm_widgets = sun6i_codec_card_dapm_widgets; card->num_dapm_widgets = ARRAY_SIZE(sun6i_codec_card_dapm_widgets); @@ -1487,6 +1490,7 @@ static struct snd_soc_card *sun8i_h3_codec_create_card(struct device *dev) return ERR_PTR(-ENOMEM); card->dev = dev; + card->owner = THIS_MODULE; card->name = "H3 Audio Codec"; card->dapm_widgets = sun6i_codec_card_dapm_widgets; card->num_dapm_widgets = ARRAY_SIZE(sun6i_codec_card_dapm_widgets); @@ -1525,6 +1529,7 @@ static struct snd_soc_card *sun8i_v3s_codec_create_card(struct device *dev) return ERR_PTR(-ENOMEM); card->dev = dev; + card->owner = THIS_MODULE; card->name = "V3s Audio Codec"; card->dapm_widgets = sun6i_codec_card_dapm_widgets; card->num_dapm_widgets = ARRAY_SIZE(sun6i_codec_card_dapm_widgets); -- 2.30.2

1 0

[PATCH v2 0/2] clean up dt_binding_check and dtbs_check warnings of renesas, rsnd.yaml
by Zhen Lei 12 Apr '21

12 Apr '21

It was then adjusted based on the following patch series: [1] https://lore.kernel.org/r/20210323163634.877511-1-robh@kernel.org/ v1: When I do dt_binding_check for all YAML files, below warning is reported: /root/mainline/Documentation/devicetree/bindings/sound/renesas,rsnd.example.dt.yaml: sound@ec500000: 'dais' is a required property From schema: /root/mainline/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml make dt_binding_check -j24 2>err.txt Before: cat err.txt | wc -l 2 After: cat err.txt | wc -l 0 make dtbs_check DT_SCHEMA_FILES=Documentation/devicetree/bindings/sound/renesas,rsnd.yaml -j24 2>err.txt Before: cat err.txt | wc -l 130 After: cat err.txt | wc -l 0 Zhen Lei (2): ASoC: dt-bindings: renesas, rsnd: Clear warning 'dais' is a required property ASoC: dt-bindings: renesas, rsnd: Clear warning 'ports' does not match any of the regexes Documentation/devicetree/bindings/sound/renesas,rsnd.yaml | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) -- 1.8.3

3 7

[PATCH v6 1/2] platform/x86: dell-privacy: Add support for Dell hardware privacy
by Perry Yuan 12 Apr '21

12 Apr '21

From: Perry Yuan <perry_yuan(a)dell.com> add support for Dell privacy driver for the Dell units equipped hardware privacy design, which protect users privacy of audio and camera from hardware level. Once the audio or camera privacy mode activated, any applications will not get any audio or video stream when user pressed ctrl+F4 hotkey, audio privacy mode will be enabled, micmute led will be also changed accordingly The micmute led is fully controlled by hardware & EC(embedded controller) and camera mute hotkey is Ctrl+F9. Currently design only emits SW_CAMERA_LENS_COVER event while the camera lens shutter will be changed by EC & HW(hardware) control *The flow is like this: 1) User presses key. HW does stuff with this key (timeout timer is started) 2) WMI event is emitted from BIOS to kernel 3) WMI event is received by dell-privacy 4) KEY_MICMUTE emitted from dell-privacy 5) Userland picks up key and modifies kcontrol for SW mute 6) Codec kernel driver catches and calls ledtrig_audio_set, like this: ledtrig_audio_set(LED_AUDIO_MICMUTE, rt715->micmute_led ? LED_ON :LED_OFF); 7) If "LED" is set to on dell-privacy notifies EC, and timeout is cancelled, HW mic mute activated. If EC not notified, HW mic mute will also be activated when timeout used up, it is just later than active ack Signed-off-by: Perry Yuan <perry_yuan(a)dell.com> --- v5 -> v6: * addressed feedback from Hans * addressed feedback from Pierre * optimize some debug format with dev_dbg() * remove platform driver,combined privacy acpi driver into single wmi driver file * optimize sysfs interface with string added to be more clearly reading v4 -> v5: * addressed feedback from Randy Dunlap * addressed feedback from Pierre-Louis Bossart * rebase to latest 5.12 rc4 upstream kernel * fix some space alignment problem v3 -> v4: * fix format for Kconfig * add sysfs document * add flow comments to the privacy wmi/acpi driver * addressed feedback from Barnabás Pőcze[Thanks very much] * export privacy_valid to make the global state simpler to query * fix one issue which will block the dell-laptop driver to load when privacy driver invalid * addressed feedback from Pierre-Louis Bossart,remove the EC ID match v2 -> v3: * add sysfs attributes doc v1 -> v2: * query EC handle from EC driver directly. * fix some code style. * add KEY_END to keymap array. * clean platform device when cleanup called * use hexadecimal format for log print in dev_dbg * remove __set_bit for the report keys from probe. * fix keymap leak * add err_free_keymap in dell_privacy_wmi_probe * wmi driver will be unregistered if privacy_acpi_init() fails * add sysfs attribute files for user space query. * add leds micmute driver to privacy acpi * add more design info the commit info --- --- .../testing/sysfs-platform-dell-privacy-wmi | 55 +++ drivers/platform/x86/dell/Kconfig | 14 + drivers/platform/x86/dell/Makefile | 1 + drivers/platform/x86/dell/dell-laptop.c | 23 +- drivers/platform/x86/dell/dell-privacy-wmi.c | 441 ++++++++++++++++++ drivers/platform/x86/dell/dell-privacy-wmi.h | 32 ++ drivers/platform/x86/dell/dell-wmi.c | 13 +- 7 files changed, 572 insertions(+), 7 deletions(-) create mode 100644 Documentation/ABI/testing/sysfs-platform-dell-privacy-wmi create mode 100644 drivers/platform/x86/dell/dell-privacy-wmi.c create mode 100644 drivers/platform/x86/dell/dell-privacy-wmi.h diff --git a/Documentation/ABI/testing/sysfs-platform-dell-privacy-wmi b/Documentation/ABI/testing/sysfs-platform-dell-privacy-wmi new file mode 100644 index 000000000000..a57ddc6a221e --- /dev/null +++ b/Documentation/ABI/testing/sysfs-platform-dell-privacy-wmi @@ -0,0 +1,55 @@ +What: /sys/bus/wmi/devices/6932965F-1671-4CEB-B988-D3AB0A901919/dell_privacy_supported_type +Date: Apr 2021 +KernelVersion: 5.13 +Contact: "perry.yuan(a)dell.com>" +Description: + Display which dell hardware level privacy devices are supported + “Dell Privacy” is a set of HW, FW, and SW features to enhance + Dell’s commitment to platform privacy for MIC, Camera, and + ePrivacy screens. + The supported hardware privacy devices are: +Attributes: + Microphone Mute: + Identifies the local microphone can be muted by hardware, no applications + is available to capture system mic sound + + Camera Shutter: + Identifies camera shutter controlled by hardware, which is a micromechanical + shutter assembly that is built onto the camera module to block capturing images + from outside the laptop + + supported: + The privacy device is supported by this system + + unsupported: + The privacy device is not supported on this system + + For example to check which privacy devices are supported: + + # cat /sys/bus/wmi/drivers/dell-privacy/6932965F-1671-4CEB-B988-D3AB0A901919/dell_privacy_supported_type + [Microphone Mute] [supported] + [Camera Shutter] [supported] + [ePrivacy Screen] [unsupport] + +What: /sys/bus/wmi/devices/6932965F-1671-4CEB-B988-D3AB0A901919/dell_privacy_current_state +Date: Apr 2021 +KernelVersion: 5.13 +Contact: "perry.yuan(a)dell.com>" +Description: + Allow user space to check current dell privacy device state. + Describes the Device State class exposed by BIOS which can be + consumed by various applications interested in knowing the Privacy + feature capabilities +Attributes: + muted: + Identifies the privacy device is turned off and cannot send stream to OS applications + + unmuted: + Identifies the privacy device is turned on ,audio or camera driver can get + stream from mic and camera module to OS applications + + For example to check all supported current privacy device states: + + # cat /sys/bus/wmi/drivers/dell-privacy/6932965F-1671-4CEB-B988-D3AB0A901919/dell_privacy_current_state + [Microphone] [unmuted] + [Camera Shutter] [unmuted] diff --git a/drivers/platform/x86/dell/Kconfig b/drivers/platform/x86/dell/Kconfig index e0a55337f51a..05d124442b25 100644 --- a/drivers/platform/x86/dell/Kconfig +++ b/drivers/platform/x86/dell/Kconfig @@ -204,4 +204,18 @@ config DELL_WMI_SYSMAN To compile this driver as a module, choose M here: the module will be called dell-wmi-sysman. +config DELL_PRIVACY + tristate "Dell Hardware Privacy Support" + depends on ACPI + depends on ACPI_WMI + depends on INPUT + depends on DELL_LAPTOP + depends on LEDS_TRIGGER_AUDIO + select DELL_WMI + help + This driver provides integration with the "Dell Hardware Privacy" feature + of Dell laptops. + + To compile this driver as a module, choose M here: the module will + be called dell_privacy. endif # X86_PLATFORM_DRIVERS_DELL diff --git a/drivers/platform/x86/dell/Makefile b/drivers/platform/x86/dell/Makefile index d720a3e42ae3..7da0c33dfcca 100644 --- a/drivers/platform/x86/dell/Makefile +++ b/drivers/platform/x86/dell/Makefile @@ -19,3 +19,4 @@ obj-$(CONFIG_DELL_WMI_AIO) += dell-wmi-aio.o obj-$(CONFIG_DELL_WMI_DESCRIPTOR) += dell-wmi-descriptor.o obj-$(CONFIG_DELL_WMI_LED) += dell-wmi-led.o obj-$(CONFIG_DELL_WMI_SYSMAN) += dell-wmi-sysman/ +obj-$(CONFIG_DELL_PRIVACY) += dell-privacy-wmi.o diff --git a/drivers/platform/x86/dell/dell-laptop.c b/drivers/platform/x86/dell/dell-laptop.c index 70edc5bb3a14..a9b4cf13913c 100644 --- a/drivers/platform/x86/dell/dell-laptop.c +++ b/drivers/platform/x86/dell/dell-laptop.c @@ -31,6 +31,8 @@ #include "dell-rbtn.h" #include "dell-smbios.h" +#include "dell-privacy-wmi.h" + struct quirk_entry { bool touchpad_led; bool kbd_led_not_present; @@ -90,6 +92,7 @@ static struct rfkill *wifi_rfkill; static struct rfkill *bluetooth_rfkill; static struct rfkill *wwan_rfkill; static bool force_rfkill; +static bool has_privacy; module_param(force_rfkill, bool, 0444); MODULE_PARM_DESC(force_rfkill, "enable rfkill on non whitelisted models"); @@ -2206,10 +2209,16 @@ static int __init dell_init(void) if (dell_smbios_find_token(GLOBAL_MIC_MUTE_DISABLE) && dell_smbios_find_token(GLOBAL_MIC_MUTE_ENABLE)) { - micmute_led_cdev.brightness = ledtrig_audio_get(LED_AUDIO_MICMUTE); - ret = led_classdev_register(&platform_device->dev, &micmute_led_cdev); - if (ret < 0) - goto fail_led; + if (dell_privacy_has_micmute()) + has_privacy = true; + else + has_privacy = false; + if (!has_privacy) { + micmute_led_cdev.brightness = ledtrig_audio_get(LED_AUDIO_MICMUTE); + ret = led_classdev_register(&platform_device->dev, &micmute_led_cdev); + if (ret < 0) + goto fail_led; + } } if (acpi_video_get_backlight_type() != acpi_backlight_vendor) @@ -2257,7 +2266,8 @@ static int __init dell_init(void) fail_get_brightness: backlight_device_unregister(dell_backlight_device); fail_backlight: - led_classdev_unregister(&micmute_led_cdev); + if (!has_privacy) + led_classdev_unregister(&micmute_led_cdev); fail_led: dell_cleanup_rfkill(); fail_rfkill: @@ -2278,7 +2288,8 @@ static void __exit dell_exit(void) touchpad_led_exit(); kbd_led_exit(); backlight_device_unregister(dell_backlight_device); - led_classdev_unregister(&micmute_led_cdev); + if (!has_privacy) + led_classdev_unregister(&micmute_led_cdev); dell_cleanup_rfkill(); if (platform_device) { platform_device_unregister(platform_device); diff --git a/drivers/platform/x86/dell/dell-privacy-wmi.c b/drivers/platform/x86/dell/dell-privacy-wmi.c new file mode 100644 index 000000000000..3fae8dd077f1 --- /dev/null +++ b/drivers/platform/x86/dell/dell-privacy-wmi.c @@ -0,0 +1,441 @@ +// SPDX-License-Identifier: GPL-2.0-only +/* + * Dell privacy notification driver + * + * Copyright (C) 2021 Dell Inc. All Rights Reserved. + */ + +#define pr_fmt(fmt) KBUILD_MODNAME ": " fmt + +#include <linux/acpi.h> +#include <linux/bitops.h> +#include <linux/input.h> +#include <linux/input/sparse-keymap.h> +#include <linux/list.h> +#include <linux/leds.h> +#include <linux/module.h> +#include <linux/wmi.h> + +#include "dell-privacy-wmi.h" + +#define DELL_PRIVACY_GUID "6932965F-1671-4CEB-B988-D3AB0A901919" +#define MICROPHONE_STATUS BIT(0) +#define CAMERA_STATUS BIT(1) +#define DELL_PRIVACY_AUDIO_EVENT 0x1 +#define DELL_PRIVACY_CAMERA_EVENT 0x2 +#define cdev_to_led(c) container_of(c, struct privacy_wmi_data, cdev) + +/* + * The wmi_list is used to store the privacy_priv struct with mutex protecting + */ +static LIST_HEAD(wmi_list); +static DEFINE_MUTEX(list_mutex); + +struct privacy_wmi_data { + struct input_dev *input_dev; + struct wmi_device *wdev; + struct list_head list; + struct led_classdev cdev; + u32 features_present; + u32 last_status; +}; + +/* DELL Privacy Type */ +enum dell_hardware_privacy_type { + DELL_PRIVACY_TYPE_AUDIO = 0, + DELL_PRIVACY_TYPE_CAMERA, + DELL_PRIVACY_TYPE_SCREEN, + DELL_PRIVACY_TYPE_MAX, +}; + +static const char * const privacy_types[DELL_PRIVACY_TYPE_MAX] = { + [DELL_PRIVACY_TYPE_AUDIO] = "Microphone", + [DELL_PRIVACY_TYPE_CAMERA] = "Camera Shutter", + [DELL_PRIVACY_TYPE_SCREEN] = "ePrivacy Screen", +}; + +/* + * Keymap for WMI privacy events of type 0x0012 + */ +static const struct key_entry dell_wmi_keymap_type_0012[] = { + /* privacy mic mute */ + { KE_KEY, 0x0001, { KEY_MICMUTE } }, + /* privacy camera mute */ + { KE_SW, 0x0002, { SW_CAMERA_LENS_COVER } }, + { KE_END, 0}, +}; + +/* + * global privacy state for other modules to query if the privacy driver loaded successfully + * or no privacy WMI device was detected at initial probe phase + */ +bool dell_privacy_present(void) +{ + struct privacy_wmi_data *priv; + + mutex_lock(&list_mutex); + priv = list_first_entry_or_null(&wmi_list, + struct privacy_wmi_data, + list); + mutex_unlock(&list_mutex); + + return priv; +} +EXPORT_SYMBOL_GPL(dell_privacy_present); + +/* + * Allow other modules to check current dell privacy mic mute state + */ +int dell_privacy_has_micmute(void) +{ + struct privacy_wmi_data *priv; + int ret; + + mutex_lock(&list_mutex); + priv = list_first_entry_or_null(&wmi_list, + struct privacy_wmi_data, + list); + /* + * There is no privacy device node yet, so we need to wait + * until the privacy driver has created software node for the micmute device. + * This function will return -EPROBE_DEFER until the privacy device is registered. + */ + if (!priv) { + ret = -EPROBE_DEFER; + goto out; + } + ret = priv->features_present & MICROPHONE_STATUS; + if (!ret) { + dev_err(&priv->wdev->dev, "privacy mic mute feature is not supported: %d\n", ret); + ret = -ENODEV; + goto out; + } + dev_err(&priv->wdev->dev, "privacy mic mute feature is supported: %d\n", ret); + +out: + mutex_unlock(&list_mutex); + return ret; +} +EXPORT_SYMBOL_GPL(dell_privacy_has_micmute); + +/* + * The flow of privacy event: + * 1) User presses key. HW does stuff with this key (timeout is started) + * 2) WMI event is emitted from BIOS + * 3) WMI event is received by dell-privacy + * 4) KEY_MICMUTE emitted from dell-privacy + * 5) Userland picks up key and modifies kcontrol for SW mute + * 6) Codec kernel driver catches and calls ledtrig_audio_set defined by + * dell-privacy-acpi driver. Codec driver will call like this to switch micmute led state. + * ledtrig_audio_set(LED_AUDIO_MICMUTE, micmute_led ? LED_ON :LED_OFF); + * 7) If "LED" is set to on dell-privacy notifies EC,and timeout is cancelled, + * HW mic mute activated. + */ +bool dell_privacy_process_event(int type, int code, int status) +{ + struct privacy_wmi_data *priv; + const struct key_entry *key; + bool ret = false; + + mutex_lock(&list_mutex); + priv = list_first_entry_or_null(&wmi_list, + struct privacy_wmi_data, + list); + if (!priv) { + dev_err(&priv->wdev->dev, "priv data is NULL\n"); + goto error; + } + + key = sparse_keymap_entry_from_scancode(priv->input_dev, (type << 16) | code); + if (!key) { + dev_warn(&priv->wdev->dev, "Unknown key with type 0x%04x and code 0x%04x pressed\n", + type, code); + goto error; + } + dev_dbg(&priv->wdev->dev, "Key with type 0x%04x and code 0x%04x pressed\n", type, code); + + switch (code) { + case DELL_PRIVACY_AUDIO_EVENT: /* Mic mute */ + case DELL_PRIVACY_CAMERA_EVENT: /* Camera mute */ + priv->last_status = status; + sparse_keymap_report_entry(priv->input_dev, key, 1, true); + ret = true; + break; + default: + dev_dbg(&priv->wdev->dev, "unknown event type 0x%04x 0x%04x", type, code); + } + +error: + mutex_unlock(&list_mutex); + return ret; +} +EXPORT_SYMBOL_GPL(dell_privacy_process_event); + +static ssize_t dell_privacy_supported_type_show(struct device *dev, + struct device_attribute *attr, + char *buf) +{ + enum dell_hardware_privacy_type type; + struct privacy_wmi_data *priv = dev_get_drvdata(dev); + char *s = buf; + u32 privacy_list; + + privacy_list = priv->features_present; + for (type = DELL_PRIVACY_TYPE_AUDIO; type < DELL_PRIVACY_TYPE_MAX; type++) { + if (privacy_types[type]) { + if (privacy_list & BIT(type)) + s += sprintf(s, "[%s] [supported]\n", privacy_types[type]); + else + s += sprintf(s, "[%s] [unsupport]\n", privacy_types[type]); + } + } + + if (s != buf) + /* convert the last space to a newline */ + *(s-1) = '\n'; + return (s - buf); +} + +static ssize_t dell_privacy_current_state_show(struct device *dev, + struct device_attribute *attr, + char *buf) +{ + struct privacy_wmi_data *priv = dev_get_drvdata(dev); + enum dell_hardware_privacy_type type; + u32 privacy_state = priv->last_status; + u32 privacy_supported = priv->features_present; + char *s = buf; + + for (type = DELL_PRIVACY_TYPE_AUDIO; type < DELL_PRIVACY_TYPE_MAX; type++) { + if (privacy_supported & BIT(type)) { + if (privacy_state & BIT(type)) + s += sprintf(s, "[%s] [unmuted]\n", privacy_types[type]); + else + s += sprintf(s, "[%s] [muted]\n", privacy_types[type]); + } + } + + if (s != buf) + /* convert the last space to a newline */ + *(s-1) = '\n'; + return (s - buf); +} + +static DEVICE_ATTR_RO(dell_privacy_supported_type); +static DEVICE_ATTR_RO(dell_privacy_current_state); + +static struct attribute *privacy_attributes[] = { + &dev_attr_dell_privacy_supported_type.attr, + &dev_attr_dell_privacy_current_state.attr, + NULL, +}; + +static const struct attribute_group privacy_attribute_group = { + .attrs = privacy_attributes +}; + +/* + * Describes the Device State class exposed by BIOS which can be consumed by + * various applications interested in knowing the Privacy feature capabilities. + * class DeviceState + * { + * [key, read] string InstanceName; + * [read] boolean ReadOnly; + * [WmiDataId(1), read] uint32 DevicesSupported; + * 0 – None, 0x1 – Microphone, 0x2 – Camera, 0x4 -ePrivacy Screen + * [WmiDataId(2), read] uint32 CurrentState; + * 0:Off; 1:On. Bit0 – Microphone, Bit1 – Camera, Bit2 - ePrivacyScreen + * }; + */ +static int get_current_status(struct wmi_device *wdev) +{ + struct privacy_wmi_data *priv = dev_get_drvdata(&wdev->dev); + union acpi_object *obj_present; + u32 *buffer; + int ret = 0; + + if (!priv) { + pr_err("dell privacy priv is NULL\n"); + return -EINVAL; + } + /* check privacy support features and device states */ + obj_present = wmidev_block_query(wdev, 0); + if (!obj_present) { + dev_err(&wdev->dev, "failed to read Binary MOF\n"); + ret = -EIO; + return ret; + } + + if (obj_present->type != ACPI_TYPE_BUFFER) { + dev_err(&wdev->dev, "Binary MOF is not a buffer!\n"); + ret = -EIO; + goto obj_free; + } + /* Although it's not technically a failure, this would lead to + * unexpected behavior + */ + if (obj_present->buffer.length != 8) { + dev_err(&wdev->dev, "Dell privacy buffer has unexpected length (%d)!\n", + obj_present->buffer.length); + ret = -EINVAL; + goto obj_free; + } + buffer = (u32 *)obj_present->buffer.pointer; + priv->features_present = buffer[0]; + priv->last_status = buffer[1]; + +obj_free: + kfree(obj_present); + return ret; +} + +static int dell_privacy_micmute_led_set(struct led_classdev *led_cdev, + enum led_brightness brightness) +{ + struct privacy_wmi_data *priv = cdev_to_led(led_cdev); + static char *acpi_method = (char *)"ECAK"; + acpi_status status; + acpi_handle handle; + + handle = ec_get_handle(); + if (!handle) + return -EIO; + + if (!acpi_has_method(handle, acpi_method)) + return -EIO; + + status = acpi_evaluate_object(handle, acpi_method, NULL, NULL); + if (ACPI_FAILURE(status)) { + dev_err(&priv->wdev->dev, "Error setting privacy EC ack value: %s\n", + acpi_format_exception(status)); + return -EIO; + } + + return 0; +} + +/* + * Pressing the mute key activates a time delayed circuit to physically cut + * off the mute. The LED is in the same circuit, so it reflects the true + * state of the HW mute. The reason for the EC "ack" is so that software + * can first invoke a SW mute before the HW circuit is cut off. Without SW + * cutting this off first does not affect the time delayed muting or status + * of the LED but there is a possibility of a "popping" noise. + * + * If the EC receives the SW ack, the circuit will be activated before the + * delay completed. + * + * Exposing as an LED device allows the codec drivers notification path to + * EC ACK to work + */ +static int dell_privacy_leds_setup(struct device *dev) +{ + struct privacy_wmi_data *priv = dev_get_drvdata(dev); + int ret; + + priv->cdev.name = "dell-privacy::micmute"; + priv->cdev.max_brightness = 1; + priv->cdev.brightness_set_blocking = dell_privacy_micmute_led_set; + priv->cdev.default_trigger = "audio-micmute"; + priv->cdev.brightness = ledtrig_audio_get(LED_AUDIO_MICMUTE); + ret = devm_led_classdev_register(dev, &priv->cdev); + if (ret) + return ret; + return 0; +} + +static int dell_privacy_wmi_probe(struct wmi_device *wdev, const void *context) +{ + struct privacy_wmi_data *priv; + struct key_entry *keymap; + int ret, i; + + ret = wmi_has_guid(DELL_PRIVACY_GUID); + if (!ret) + pr_debug("Unable to detect available Dell privacy devices: %d\n", ret); + + priv = devm_kzalloc(&wdev->dev, sizeof(*priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + + dev_set_drvdata(&wdev->dev, priv); + priv->wdev = wdev; + /* create evdev passing interface */ + priv->input_dev = devm_input_allocate_device(&wdev->dev); + if (!priv->input_dev) + return -ENOMEM; + + /* remap the wmi keymap event to new keymap */ + keymap = kcalloc(ARRAY_SIZE(dell_wmi_keymap_type_0012), + sizeof(struct key_entry), GFP_KERNEL); + if (!keymap) + return -ENOMEM; + + /* remap the keymap code with Dell privacy key type 0x12 as prefix + * KEY_MICMUTE scancode will be reported as 0x120001 + */ + for (i = 0; i < ARRAY_SIZE(dell_wmi_keymap_type_0012); i++) { + keymap[i] = dell_wmi_keymap_type_0012[i]; + keymap[i].code |= (0x0012 << 16); + } + ret = sparse_keymap_setup(priv->input_dev, keymap, NULL); + if (ret) + return ret; + + priv->input_dev->dev.parent = &wdev->dev; + priv->input_dev->name = "Dell Privacy Driver"; + priv->input_dev->id.bustype = BUS_HOST; + + ret = input_register_device(priv->input_dev); + if (ret) + return ret; + + ret = get_current_status(priv->wdev); + if (ret) + return ret; + + ret = devm_device_add_group(&wdev->dev, &privacy_attribute_group); + if (ret) + return ret; + + ret = dell_privacy_leds_setup(&priv->wdev->dev); + if (ret) + return ret; + + mutex_lock(&list_mutex); + list_add_tail(&priv->list, &wmi_list); + mutex_unlock(&list_mutex); + kfree(keymap); + return 0; +} + +static int dell_privacy_wmi_remove(struct wmi_device *wdev) +{ + struct privacy_wmi_data *priv = dev_get_drvdata(&wdev->dev); + + mutex_lock(&list_mutex); + list_del(&priv->list); + mutex_unlock(&list_mutex); + return 0; +} + +static const struct wmi_device_id dell_wmi_privacy_wmi_id_table[] = { + { .guid_string = DELL_PRIVACY_GUID }, + { }, +}; + +static struct wmi_driver dell_privacy_wmi_driver = { + .driver = { + .name = "dell-privacy", + }, + .probe = dell_privacy_wmi_probe, + .remove = dell_privacy_wmi_remove, + .id_table = dell_wmi_privacy_wmi_id_table, +}; + +module_wmi_driver(dell_privacy_wmi_driver); + +MODULE_DEVICE_TABLE(wmi, dell_wmi_privacy_wmi_id_table); +MODULE_AUTHOR("Perry Yuan <perry_yuan(a)dell.com>"); +MODULE_DESCRIPTION("Dell Privacy WMI Driver"); +MODULE_LICENSE("GPL"); diff --git a/drivers/platform/x86/dell/dell-privacy-wmi.h b/drivers/platform/x86/dell/dell-privacy-wmi.h new file mode 100644 index 000000000000..a24893754286 --- /dev/null +++ b/drivers/platform/x86/dell/dell-privacy-wmi.h @@ -0,0 +1,32 @@ +/* SPDX-License-Identifier: GPL-2.0-only */ +/* + * Dell privacy notification driver + * + * Copyright (C) 2021 Dell Inc. All Rights Reserved. + */ + +#ifndef _DELL_PRIVACY_WMI_H_ +#define _DELL_PRIVACY_WMI_H_ + +#if IS_ENABLED(CONFIG_DELL_PRIVACY) +int dell_privacy_has_micmute(void); +bool dell_privacy_present(void); +bool dell_privacy_process_event(int type, int code, int status); +#else /* CONFIG_DELL_PRIVACY */ +static inline int dell_privacy_has_micmute(void) +{ + return -ENODEV; +} + +static inline int dell_privacy_present(void) +{ + return -ENODEV; +} + +static inline void dell_privacy_process_event(int type, int code, int status) +{} +#endif /* CONFIG_DELL_PRIVACY */ + +int dell_privacy_acpi_init(void); +void dell_privacy_acpi_exit(void); +#endif diff --git a/drivers/platform/x86/dell/dell-wmi.c b/drivers/platform/x86/dell/dell-wmi.c index bbdb3e860892..8ef9e22a538f 100644 --- a/drivers/platform/x86/dell/dell-wmi.c +++ b/drivers/platform/x86/dell/dell-wmi.c @@ -27,6 +27,7 @@ #include <acpi/video.h> #include "dell-smbios.h" #include "dell-wmi-descriptor.h" +#include "dell-privacy-wmi.h" MODULE_AUTHOR("Matthew Garrett <mjg(a)redhat.com>"); MODULE_AUTHOR("Pali Rohár <pali(a)kernel.org>"); @@ -381,6 +382,7 @@ static void dell_wmi_notify(struct wmi_device *wdev, u16 *buffer_entry, *buffer_end; acpi_size buffer_size; int len, i; + int ret; if (obj->type != ACPI_TYPE_BUFFER) { pr_warn("bad response type %x\n", obj->type); @@ -427,7 +429,6 @@ static void dell_wmi_notify(struct wmi_device *wdev, switch (buffer_entry[1]) { case 0x0000: /* One key pressed or event occurred */ - case 0x0012: /* Event with extended data occurred */ if (len > 2) dell_wmi_process_key(wdev, buffer_entry[1], buffer_entry[2]); @@ -439,6 +440,16 @@ static void dell_wmi_notify(struct wmi_device *wdev, dell_wmi_process_key(wdev, buffer_entry[1], buffer_entry[i]); break; + case 0x0012: + ret = dell_privacy_present(); + if ((ret) && (len > 2)) { + dell_privacy_process_event(buffer_entry[1], + buffer_entry[3], buffer_entry[4]); + } else if (len > 2) { + dell_wmi_process_key(wdev, buffer_entry[1], + buffer_entry[2]); + } + break; default: /* Unknown event */ pr_info("Unknown WMI event type 0x%x\n", (int)buffer_entry[1]); -- 2.25.1

5 4

advice for creating special ALSA driver or plugin
by Marc Lavallée 11 Apr '21

11 Apr '21

Hi ALSA developers. I'm completely new to ALSA developing, so I'm being lazy and ask for advices. I have a USB recording device for four 16bit/48Khz channels, "encoded" in a stereo 16bit/96Khz track; because it is a USB audio recording device, so no special recording software is required, but a (proprietary Windows) converting software is provided (that I can't use). Output channels are alternated at the sample level: L: 1313131313, R: 2424242424, etc... So basically the driver/plugin would alternates between input samples of a stereo track to output 4 channels (at half the input sample rate). Is there a code base I could study and adapt to start my simple project? Thanks for your attention. Marc

1 0

[PATCH] ALSA: core - add more card sysfs entries
by Jaroslav Kysela 10 Apr '21

10 Apr '21

There are several strings which are describing the card. As time goes, we have new universal drivers which probe components in a way, which is disassociated from the card structure (ASoC). Also, some drivers may require to select another firmware depending on the specific platform using udev. The new firmware may change the sound card behaviour. This patch allows flexible modifications of the card description from the user space to handle the specific boot / plug-in settings. Cc: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Cc: Mark Brown <broonie(a)kernel.org> Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> --- sound/core/init.c | 166 +++++++++++++++++++++++++++++++++++++++++++--- 1 file changed, 155 insertions(+), 11 deletions(-) diff --git a/sound/core/init.c b/sound/core/init.c index ef41f5b3a240..01b26912a4d0 100644 --- a/sound/core/init.c +++ b/sound/core/init.c @@ -662,6 +662,33 @@ void snd_card_set_id(struct snd_card *card, const char *nid) } EXPORT_SYMBOL(snd_card_set_id); +#define EXTRA_ID_CHARS "_-" +#define EXTRA_NAME_CHARS " _-.:" + +static bool safe_attr_strcpy(char *dst, size_t dst_count, + const char *src, size_t src_count, + const char *extra_characters) +{ + size_t idx, copy; + int c; + + copy = src_count >= dst_count ? dst_count - 1 : src_count; + for (idx = 0; idx < copy; idx++) { + c = src[idx]; + if (c < ' ') { + copy = idx; + break; + } + if (!isalnum(c) && !strchr(extra_characters, c)) + return false; + } + if (copy < 3) + return false; + memcpy(dst, src, copy); + dst[copy] = '\0'; + return true; +} + static ssize_t card_id_show_attr(struct device *dev, struct device_attribute *attr, char *buf) @@ -676,18 +703,10 @@ card_id_store_attr(struct device *dev, struct device_attribute *attr, { struct snd_card *card = container_of(dev, struct snd_card, card_dev); char buf1[sizeof(card->id)]; - size_t copy = count > sizeof(card->id) - 1 ? - sizeof(card->id) - 1 : count; - size_t idx; - int c; - for (idx = 0; idx < copy; idx++) { - c = buf[idx]; - if (!isalnum(c) && c != '_' && c != '-') - return -EINVAL; - } - memcpy(buf1, buf, copy); - buf1[copy] = '\0'; + if (!safe_attr_strcpy(buf1, sizeof(buf1), buf, count, EXTRA_ID_CHARS)) + return -EINVAL; + mutex_lock(&snd_card_mutex); if (!card_id_ok(NULL, buf1)) { mutex_unlock(&snd_card_mutex); @@ -712,9 +731,134 @@ card_number_show_attr(struct device *dev, static DEVICE_ATTR(number, 0444, card_number_show_attr, NULL); +static ssize_t +card_driver_show_attr(struct device *dev, struct device_attribute *attr, char *buf) +{ + struct snd_card *card = container_of(dev, struct snd_card, card_dev); + return scnprintf(buf, PAGE_SIZE, "%s\n", card->driver); +} + +static ssize_t +card_driver_store_attr(struct device *dev, struct device_attribute *attr, + const char *buf, size_t count) +{ + struct snd_card *card = container_of(dev, struct snd_card, card_dev); + char driver1[sizeof(card->driver)]; + + if (!safe_attr_strcpy(driver1, sizeof(driver1), buf, count, EXTRA_NAME_CHARS)) + return -EINVAL; + mutex_lock(&snd_card_mutex); + strcpy(card->driver, driver1); + mutex_unlock(&snd_card_mutex); + return count; +} + +static DEVICE_ATTR(driver, 0644, card_driver_show_attr, card_driver_store_attr); + +static ssize_t +card_name_show_attr(struct device *dev, struct device_attribute *attr, char *buf) +{ + struct snd_card *card = container_of(dev, struct snd_card, card_dev); + return scnprintf(buf, PAGE_SIZE, "%s\n", card->shortname); +} + +static ssize_t +card_name_store_attr(struct device *dev, struct device_attribute *attr, + const char *buf, size_t count) +{ + struct snd_card *card = container_of(dev, struct snd_card, card_dev); + char name1[sizeof(card->shortname)]; + + if (!safe_attr_strcpy(name1, sizeof(name1), buf, count, EXTRA_NAME_CHARS)) + return -EINVAL; + mutex_lock(&snd_card_mutex); + strcpy(card->shortname, name1); + mutex_unlock(&snd_card_mutex); + return count; +} + +static DEVICE_ATTR(name, 0644, card_name_show_attr, card_name_store_attr); + +static ssize_t +card_longname_show_attr(struct device *dev, struct device_attribute *attr, char *buf) +{ + struct snd_card *card = container_of(dev, struct snd_card, card_dev); + return scnprintf(buf, PAGE_SIZE, "%s\n", card->longname); +} + +static ssize_t +card_longname_store_attr(struct device *dev, struct device_attribute *attr, + const char *buf, size_t count) +{ + struct snd_card *card = container_of(dev, struct snd_card, card_dev); + char longname1[sizeof(card->longname)]; + + if (!safe_attr_strcpy(longname1, sizeof(longname1), buf, count, EXTRA_NAME_CHARS)) + return -EINVAL; + mutex_lock(&snd_card_mutex); + strcpy(card->longname, longname1); + mutex_unlock(&snd_card_mutex); + return count; +} + +static DEVICE_ATTR(longname, 0644, card_longname_show_attr, card_longname_store_attr); + +static ssize_t +card_mixername_show_attr(struct device *dev, struct device_attribute *attr, char *buf) +{ + struct snd_card *card = container_of(dev, struct snd_card, card_dev); + return scnprintf(buf, PAGE_SIZE, "%s\n", card->mixername); +} + +static ssize_t +card_mixername_store_attr(struct device *dev, struct device_attribute *attr, + const char *buf, size_t count) +{ + struct snd_card *card = container_of(dev, struct snd_card, card_dev); + char mixername1[sizeof(card->mixername)]; + + if (!safe_attr_strcpy(mixername1, sizeof(mixername1), buf, count, EXTRA_NAME_CHARS)) + return -EINVAL; + mutex_lock(&snd_card_mutex); + strcpy(card->mixername, mixername1); + mutex_unlock(&snd_card_mutex); + return count; +} + +static DEVICE_ATTR(mixername, 0644, card_mixername_show_attr, card_mixername_store_attr); + +static ssize_t +card_components_show_attr(struct device *dev, struct device_attribute *attr, char *buf) +{ + struct snd_card *card = container_of(dev, struct snd_card, card_dev); + return scnprintf(buf, PAGE_SIZE, "%s\n", card->components); +} + +static ssize_t +card_components_store_attr(struct device *dev, struct device_attribute *attr, + const char *buf, size_t count) +{ + struct snd_card *card = container_of(dev, struct snd_card, card_dev); + char components1[sizeof(card->components)]; + + if (!safe_attr_strcpy(components1, sizeof(components1), buf, count, EXTRA_NAME_CHARS)) + return -EINVAL; + mutex_lock(&snd_card_mutex); + strcpy(card->components, components1); + mutex_unlock(&snd_card_mutex); + return count; +} + +static DEVICE_ATTR(components, 0644, card_components_show_attr, card_components_store_attr); + static struct attribute *card_dev_attrs[] = { &dev_attr_id.attr, &dev_attr_number.attr, + &dev_attr_driver.attr, + &dev_attr_name.attr, + &dev_attr_longname.attr, + &dev_attr_mixername.attr, + &dev_attr_components.attr, NULL }; -- 2.30.2

7 34

[PATCH] ALSA: HDA: Add access description in __snd_hda_add_vmaster
by Ranjani Sridharan 10 Apr '21

10 Apr '21

Add description for access parameter in __snd_hda_add_vmaster() to prevent the compilation warning: warning: Function parameter or member 'access' not described in '__snd_hda_add_vmaster' Fixes: e65bf99718b5 ("ALSA: HDA - remove the custom implementation for the audio LED trigger") Signed-off-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> --- sound/pci/hda/hda_codec.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index e54e39b35709..a31009afc025 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -1938,6 +1938,7 @@ static int add_follower(struct hda_codec *codec, * @followers: follower control names (optional) * @suffix: suffix string to each follower name (optional) * @init_follower_vol: initialize followers to unmute/0dB + * @access: kcontrol access rights * @ctl_ret: store the vmaster kcontrol in return * * Create a virtual master control with the given name. The TLV data -- 2.25.1

2 1

[PATCH v2] ASoC: amd: Add support for ALC1015P codec in acp3x machine driver
by Vijendar Mukunda 09 Apr '21

09 Apr '21

Add ALC1015p codec support for acp3x machine driver. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- v1 -> v2: update ACPI HID as "10021015" for AMD platform with RT1015p support sound/soc/amd/Kconfig | 1 + sound/soc/amd/acp3x-rt5682-max9836.c | 46 +++++++++++++++++++++++++++++++++++- 2 files changed, 46 insertions(+), 1 deletion(-) diff --git a/sound/soc/amd/Kconfig b/sound/soc/amd/Kconfig index 256c192..ba5a85b 100644 --- a/sound/soc/amd/Kconfig +++ b/sound/soc/amd/Kconfig @@ -35,6 +35,7 @@ config SND_SOC_AMD_RV_RT5682_MACH select SND_SOC_CROS_EC_CODEC select I2C_CROS_EC_TUNNEL select SND_SOC_RT1015 + select SND_SOC_RT1015P depends on SND_SOC_AMD_ACP3x && I2C && CROS_EC help This option enables machine driver for RT5682 and MAX9835. diff --git a/sound/soc/amd/acp3x-rt5682-max9836.c b/sound/soc/amd/acp3x-rt5682-max9836.c index cea320a..d9980ab 100644 --- a/sound/soc/amd/acp3x-rt5682-max9836.c +++ b/sound/soc/amd/acp3x-rt5682-max9836.c @@ -275,6 +275,8 @@ SND_SOC_DAILINK_DEF(rt5682, DAILINK_COMP_ARRAY(COMP_CODEC("i2c-10EC5682:00", "rt5682-aif1"))); SND_SOC_DAILINK_DEF(max, DAILINK_COMP_ARRAY(COMP_CODEC("MX98357A:00", "HiFi"))); +SND_SOC_DAILINK_DEF(rt1015p, + DAILINK_COMP_ARRAY(COMP_CODEC("RTL1015:00", "HiFi"))); SND_SOC_DAILINK_DEF(rt1015, DAILINK_COMP_ARRAY(COMP_CODEC("i2c-10EC1015:00", "rt1015-aif"), COMP_CODEC("i2c-10EC1015:01", "rt1015-aif"))); @@ -419,6 +421,43 @@ static struct snd_soc_card acp3x_1015 = { .num_controls = ARRAY_SIZE(acp3x_mc_1015_controls), }; +static const struct snd_soc_dapm_widget acp3x_1015p_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_MUX("Dmic Mux", SND_SOC_NOPM, 0, 0, + &acp3x_dmic_mux_control), + SND_SOC_DAPM_SPK("Speakers", NULL), +}; + +static const struct snd_soc_dapm_route acp3x_1015p_route[] = { + {"Headphone Jack", NULL, "HPOL"}, + {"Headphone Jack", NULL, "HPOR"}, + {"IN1P", NULL, "Headset Mic"}, + {"Dmic Mux", "Front Mic", "DMIC"}, + {"Dmic Mux", "Rear Mic", "DMIC"}, + /* speaker */ + { "Speakers", NULL, "Speaker" }, +}; + +static const struct snd_kcontrol_new acp3x_mc_1015p_controls[] = { + SOC_DAPM_PIN_SWITCH("Speakers"), + SOC_DAPM_PIN_SWITCH("Headphone Jack"), + SOC_DAPM_PIN_SWITCH("Headset Mic"), +}; + +static struct snd_soc_card acp3x_1015p = { + .name = "acp3xalc56821015p", + .owner = THIS_MODULE, + .dai_link = acp3x_dai, + .num_links = ARRAY_SIZE(acp3x_dai), + .dapm_widgets = acp3x_1015p_widgets, + .num_dapm_widgets = ARRAY_SIZE(acp3x_1015p_widgets), + .dapm_routes = acp3x_1015p_route, + .num_dapm_routes = ARRAY_SIZE(acp3x_1015p_route), + .controls = acp3x_mc_1015p_controls, + .num_controls = ARRAY_SIZE(acp3x_mc_1015p_controls), +}; + void *soc_is_rltk_max(struct device *dev) { const struct acpi_device_id *match; @@ -435,6 +474,9 @@ static void card_spk_dai_link_present(struct snd_soc_dai_link *links, if (!strcmp(card_name, "acp3xalc56821015")) { links[1].codecs = rt1015; links[1].num_codecs = ARRAY_SIZE(rt1015); + } else if (!strcmp(card_name, "acp3xalc56821015p")) { + links[1].codecs = rt1015p; + links[1].num_codecs = ARRAY_SIZE(rt1015p); } else { links[1].codecs = max; links[1].num_codecs = ARRAY_SIZE(max); @@ -486,6 +528,7 @@ static int acp3x_probe(struct platform_device *pdev) static const struct acpi_device_id acp3x_audio_acpi_match[] = { { "AMDI5682", (unsigned long)&acp3x_5682}, { "AMDI1015", (unsigned long)&acp3x_1015}, + { "10021015", (unsigned long)&acp3x_1015p}, {}, }; MODULE_DEVICE_TABLE(acpi, acp3x_audio_acpi_match); @@ -503,5 +546,6 @@ module_platform_driver(acp3x_audio); MODULE_AUTHOR("akshu.agrawal(a)amd.com"); MODULE_AUTHOR("Vishnuvardhanrao.Ravulapati(a)amd.com"); -MODULE_DESCRIPTION("ALC5682 ALC1015 & MAX98357 audio support"); +MODULE_AUTHOR("Vijendar.Mukunda(a)amd.com"); +MODULE_DESCRIPTION("ALC5682 ALC1015, ALC1015P & MAX98357 audio support"); MODULE_LICENSE("GPL v2"); -- 2.7.4

2 1

[PATCH -next] ASoC: rt715-sdca: Constify static struct snd_soc_dai_ops
by Ye Bin 09 Apr '21

09 Apr '21

The snd_soc_dai_ops structures is only stored in the ops field of a snd_soc_dai_driver structure, so make the snd_soc_dai_ops structure const to allow the compiler to put it in read-only memory. Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: Ye Bin <yebin10(a)huawei.com> --- sound/soc/codecs/rt715-sdca.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt715-sdca.c b/sound/soc/codecs/rt715-sdca.c index 20528afbdc57..7db76c19e048 100644 --- a/sound/soc/codecs/rt715-sdca.c +++ b/sound/soc/codecs/rt715-sdca.c @@ -935,7 +935,7 @@ static int rt715_sdca_pcm_hw_free(struct snd_pcm_substream *substream, #define RT715_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S8) -static struct snd_soc_dai_ops rt715_sdca_ops = { +static const struct snd_soc_dai_ops rt715_sdca_ops = { .hw_params = rt715_sdca_pcm_hw_params, .hw_free = rt715_sdca_pcm_hw_free, .set_sdw_stream = rt715_sdca_set_sdw_stream,

2 1

[PATCH] ASoC: soc-compress: lock pcm_mutex to resolve lockdep error
by Gyeongtaek Lee 09 Apr '21

09 Apr '21

If panic_on_warn=1 is added in bootargs and compress offload playback with DPCM is started, kernel panic would be occurred because rtd->card->pcm_mutex isn't held in soc_compr_open_fe() and soc_compr_free_fe() and it generates lockdep warning in the following code. void snd_soc_runtime_action(struct snd_soc_pcm_runtime *rtd, int stream, int action) { struct snd_soc_dai *dai; int i; lockdep_assert_held(&rtd->card->pcm_mutex); To prevent lockdep warning but minimize side effect by adding mutex, pcm_mutex is held just before snd_soc_runtime_activate() and snd_soc_runtime_deactivate() and is released right after them. Signed-off-by: Gyeongtaek Lee <gt82.lee(a)samsung.com> --- sound/soc/soc-compress.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index 246a5e32e22a..b4810266f5e5 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -153,7 +153,9 @@ static int soc_compr_open_fe(struct snd_compr_stream *cstream) fe->dpcm[stream].state = SND_SOC_DPCM_STATE_OPEN; fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_NO; + mutex_lock_nested(&fe->card->pcm_mutex, fe->card->pcm_subclass); snd_soc_runtime_activate(fe, stream); + mutex_unlock(&fe->card->pcm_mutex); mutex_unlock(&fe->card->mutex); @@ -181,7 +183,9 @@ static int soc_compr_free_fe(struct snd_compr_stream *cstream) mutex_lock_nested(&fe->card->mutex, SND_SOC_CARD_CLASS_RUNTIME); + mutex_lock_nested(&fe->card->pcm_mutex, fe->card->pcm_subclass); snd_soc_runtime_deactivate(fe, stream); + mutex_unlock(&fe->card->pcm_mutex); fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_FE; base-commit: e49d033bddf5b565044e2abe4241353959bc9120 -- 2.21.0

2 1

[PATCH -next] ASoC: amd: raven: acp3x-i2s: Constify static struct snd_soc_dai_ops
by Ye Bin 09 Apr '21

09 Apr '21

The snd_soc_dai_ops structures is only stored in the ops field of a snd_soc_dai_driver structure, so make the snd_soc_dai_ops structure const to allow the compiler to put it in read-only memory. Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: Ye Bin <yebin10(a)huawei.com> --- sound/soc/amd/raven/acp3x-i2s.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/amd/raven/acp3x-i2s.c b/sound/soc/amd/raven/acp3x-i2s.c index 5bc028692fcf..396cb7d619e2 100644 --- a/sound/soc/amd/raven/acp3x-i2s.c +++ b/sound/soc/amd/raven/acp3x-i2s.c @@ -249,7 +249,7 @@ static int acp3x_i2s_trigger(struct snd_pcm_substream *substream, return ret; } -static struct snd_soc_dai_ops acp3x_i2s_dai_ops = { +static const struct snd_soc_dai_ops acp3x_i2s_dai_ops = { .hw_params = acp3x_i2s_hwparams, .trigger = acp3x_i2s_trigger, .set_fmt = acp3x_i2s_set_fmt,

2 1

[PATCH] ASoC: rt1019: remove registers to sync with rt1019 datasheet
by Jack Yu 09 Apr '21

09 Apr '21

Remove some registers to synchronize with the latest rt1019 datasheet. Signed-off-by: Jack Yu <jack.yu(a)realtek.com> --- sound/soc/codecs/rt1019.c | 334 +------------------------------------- sound/soc/codecs/rt1019.h | 162 ------------------ 2 files changed, 1 insertion(+), 495 deletions(-) diff --git a/sound/soc/codecs/rt1019.c b/sound/soc/codecs/rt1019.c index f64bb7302720..10656a5927f1 100644 --- a/sound/soc/codecs/rt1019.c +++ b/sound/soc/codecs/rt1019.c @@ -32,210 +32,40 @@ static const struct reg_default rt1019_reg[] = { { 0x0000, 0x00 }, - { 0x0002, 0x55 }, - { 0x0003, 0x55 }, - { 0x0005, 0x54 }, - { 0x0006, 0x05 }, - { 0x0007, 0x01 }, - { 0x0008, 0x70 }, - { 0x0009, 0x28 }, - { 0x000a, 0x7f }, { 0x0011, 0x04 }, { 0x0013, 0x00 }, - { 0x0015, 0x00 }, - { 0x0017, 0x00 }, { 0x0019, 0x30 }, { 0x001b, 0x01 }, - { 0x001d, 0x18 }, - { 0x001f, 0x29 }, - { 0x0021, 0x09 }, - { 0x0023, 0x02 }, - { 0x0025, 0x00 }, - { 0x0026, 0x00 }, - { 0x0028, 0x03 }, - { 0x0053, 0x00 }, - { 0x0055, 0x00 }, - { 0x0056, 0x00 }, - { 0x0057, 0x00 }, - { 0x0058, 0x00 }, - { 0x005a, 0x00 }, { 0x005c, 0x00 }, - { 0x005d, 0x00 }, { 0x005e, 0x10 }, { 0x005f, 0xec }, { 0x0061, 0x10 }, { 0x0062, 0x19 }, - { 0x0064, 0x00 }, { 0x0066, 0x08 }, - { 0x0068, 0x00 }, - { 0x006a, 0x00 }, - { 0x006c, 0x00 }, - { 0x006e, 0x00 }, { 0x0100, 0x80 }, { 0x0100, 0x51 }, { 0x0102, 0x23 }, - { 0x0102, 0x0f }, - { 0x0104, 0x6c }, - { 0x0105, 0xec }, - { 0x0106, 0x00 }, - { 0x0107, 0x00 }, - { 0x0108, 0x00 }, - { 0x0200, 0x40 }, - { 0x0201, 0x00 }, - { 0x0202, 0x00 }, - { 0x0203, 0x00 }, - { 0x0301, 0x02 }, - { 0x0302, 0xaa }, - { 0x0303, 0x2a }, - { 0x0304, 0x6a }, - { 0x0306, 0xb0 }, - { 0x0308, 0x48 }, - { 0x030a, 0x0a }, - { 0x030b, 0x4b }, - { 0x030d, 0x7d }, - { 0x030e, 0xef }, - { 0x030f, 0x5a }, { 0x0311, 0x00 }, { 0x0312, 0x3e }, { 0x0313, 0x86 }, - { 0x0315, 0xa8 }, - { 0x0318, 0x1b }, - { 0x031a, 0x3d }, - { 0x031c, 0x40 }, - { 0x031d, 0x40 }, - { 0x031e, 0x30 }, - { 0x031f, 0xbb }, - { 0x0320, 0xa5 }, - { 0x0321, 0xa5 }, - { 0x0323, 0x5a }, - { 0x0324, 0xaa }, - { 0x0325, 0x80 }, - { 0x0326, 0xaa }, - { 0x0327, 0x66 }, - { 0x0328, 0x94 }, - { 0x0329, 0x00 }, - { 0x0330, 0x00 }, - { 0x0331, 0x30 }, - { 0x0332, 0x05 }, { 0x0400, 0x03 }, { 0x0401, 0x02 }, { 0x0402, 0x01 }, - { 0x0403, 0x23 }, - { 0x0404, 0x45 }, - { 0x0405, 0x67 }, - { 0x0500, 0x80 }, - { 0x0501, 0x00 }, - { 0x0502, 0x00 }, - { 0x0503, 0x00 }, { 0x0504, 0xff }, { 0x0505, 0x24 }, - { 0x0600, 0x75 }, - { 0x0601, 0x41 }, - { 0x0602, 0x60 }, - { 0x0603, 0x7f }, - { 0x0604, 0x65 }, - { 0x0605, 0x43 }, - { 0x0606, 0x00 }, - { 0x0607, 0x00 }, - { 0x0608, 0x00 }, - { 0x0609, 0x00 }, - { 0x060a, 0x00 }, - { 0x060b, 0x00 }, - { 0x060c, 0x00 }, - { 0x060d, 0x00 }, - { 0x060e, 0x00 }, - { 0x060f, 0x00 }, - { 0x0700, 0x15 }, - { 0x0701, 0xc8 }, - { 0x0704, 0x02 }, - { 0x0705, 0x00 }, - { 0x0706, 0x00 }, - { 0x0707, 0x80 }, - { 0x0708, 0x08 }, - { 0x0709, 0x00 }, - { 0x0800, 0x00 }, - { 0x0801, 0x00 }, - { 0x0802, 0x09 }, - { 0x0803, 0x00 }, - { 0x0900, 0x87 }, - { 0x0a01, 0x99 }, - { 0x0a02, 0x40 }, - { 0x0a03, 0x10 }, { 0x0b00, 0x50 }, { 0x0b01, 0xc3 }, - { 0x0c00, 0x84 }, - { 0x0c01, 0x00 }, - { 0x0c02, 0xbb }, - { 0x0c03, 0x80 }, - { 0x0c04, 0x10 }, - { 0x0c05, 0x30 }, - { 0x0c06, 0x00 }, - { 0x0d00, 0x80 }, - { 0x0d01, 0xbb }, - { 0x0d02, 0x80 }, - { 0x0d03, 0x00 }, - { 0x0d04, 0x00 }, - { 0x0d05, 0x00 }, - { 0x0e00, 0x80 }, - { 0x0e01, 0xbb }, - { 0x0e02, 0x80 }, - { 0x0e03, 0x00 }, - { 0x0e04, 0x10 }, - { 0x0e05, 0x30 }, - { 0x0f00, 0x80 }, - { 0x0f01, 0xbb }, - { 0x0f02, 0x80 }, - { 0x0f03, 0x00 }, - { 0x0f04, 0x10 }, - { 0x0f05, 0x30 }, - { 0x0f06, 0x88 }, - { 0x0f07, 0x88 }, - { 0x0f08, 0x00 }, - { 0x0f09, 0x00 }, }; static bool rt1019_volatile_register(struct device *dev, unsigned int reg) { switch (reg) { - case RT1019_PWR_STRP_1: case RT1019_PWR_STRP_2: - case RT1019_SIL_DET_GAT: - case RT1019_PHASE_SYNC: - case RT1019_STAT_MACH_2: - case RT1019_FS_DET_1: - case RT1019_FS_DET_2: - case RT1019_FS_DET_3: - case RT1019_FS_DET_4: - case RT1019_FS_DET_5: - case RT1019_FS_DET_6: - case RT1019_FS_DET_7: - case RT1019_ANA_READ: case RT1019_VER_ID: - case RT1019_CUSTOM_ID: case RT1019_VEND_ID_1: case RT1019_VEND_ID_2: case RT1019_DEV_ID_1: case RT1019_DEV_ID_2: - case RT1019_CAL_TOP_3: - case RT1019_CAL_TOP_7: - case RT1019_CAL_TOP_17: - case RT1019_CAL_TOP_18: - case RT1019_CAL_TOP_19: - case RT1019_CAL_TOP_20: - case RT1019_CAL_TOP_21: - case RT1019_CAL_TOP_22: - case RT1019_MDRE_CTRL_2: - case RT1019_MDRE_CTRL_3: - case RT1019_MDRE_CTRL_4: - case RT1019_SIL_DET_2: - case RT1019_PWM_DC_DET_1: - case RT1019_PMC_8: - case RT1019_PMC_9: - case RT1019_SPKDRC_7: - case RT1019_HALF_FREQ_7: - case RT1019_CUR_CTRL_11: - case RT1019_CUR_CTRL_12: - case RT1019_CUR_CTRL_13: return true; default: @@ -247,191 +77,29 @@ static bool rt1019_readable_register(struct device *dev, unsigned int reg) { switch (reg) { case RT1019_RESET: - case RT1019_PAD_DRV_1: - case RT1019_PAD_DRV_2: - case RT1019_PAD_PULL_1: - case RT1019_PAD_PULL_2: - case RT1019_PAD_PULL_3: - case RT1019_I2C_CTRL_1: - case RT1019_I2C_CTRL_2: - case RT1019_I2C_CTRL_3: case RT1019_IDS_CTRL: case RT1019_ASEL_CTRL: - case RT1019_PLL_RESET: - case RT1019_PWR_STRP_1: case RT1019_PWR_STRP_2: case RT1019_BEEP_TONE: - case RT1019_SIL_DET_GAT: - case RT1019_CLASSD_TIME: - case RT1019_CLASSD_OCP: - case RT1019_PHASE_SYNC: - case RT1019_STAT_MACH_1: - case RT1019_STAT_MACH_2: - case RT1019_EFF_CTRL: - case RT1019_FS_DET_1: - case RT1019_FS_DET_2: - case RT1019_FS_DET_3: - case RT1019_FS_DET_4: - case RT1019_FS_DET_5: - case RT1019_FS_DET_6: - case RT1019_FS_DET_7: - case RT1019_ANA_CTRL: - case RT1019_DUMMY_A: - case RT1019_DUMMY_B: - case RT1019_DUMMY_C: - case RT1019_DUMMY_D: - case RT1019_ANA_READ: case RT1019_VER_ID: - case RT1019_CUSTOM_ID: case RT1019_VEND_ID_1: case RT1019_VEND_ID_2: case RT1019_DEV_ID_1: case RT1019_DEV_ID_2: - case RT1019_TEST_PAD: case RT1019_SDB_CTRL: - case RT1019_TEST_CTRL_1: - case RT1019_TEST_CTRL_2: - case RT1019_TEST_CTRL_3: - case RT1019_SCAN_MODE: case RT1019_CLK_TREE_1: case RT1019_CLK_TREE_2: case RT1019_CLK_TREE_3: - case RT1019_CLK_TREE_4: - case RT1019_CLK_TREE_5: - case RT1019_CLK_TREE_6: - case RT1019_CLK_TREE_7: - case RT1019_CLK_TREE_8: - case RT1019_CLK_TREE_9: - case RT1019_ASRC_1: - case RT1019_ASRC_2: - case RT1019_ASRC_3: - case RT1019_ASRC_4: - case RT1019_SYS_CLK: - case RT1019_BIAS_CUR_1: - case RT1019_BIAS_CUR_2: - case RT1019_BIAS_CUR_3: - case RT1019_BIAS_CUR_4: - case RT1019_CHOP_CLK_DAC: - case RT1019_CHOP_CLK_ADC: - case RT1019_LDO_CTRL_1: - case RT1019_LDO_CTRL_2: - case RT1019_PM_ANA_1: - case RT1019_PM_ANA_2: - case RT1019_PM_ANA_3: case RT1019_PLL_1: case RT1019_PLL_2: case RT1019_PLL_3: - case RT1019_PLL_INT_1: - case RT1019_PLL_INT_3: - case RT1019_MIXER: - case RT1019_CLD_OUT_1: - case RT1019_CLD_OUT_2: - case RT1019_CLD_OUT_3: - case RT1019_CLD_OUT_4: - case RT1019_CLD_OUT_5: - case RT1019_CLD_OUT_6: - case RT1019_CLS_INT_REG_1: - case RT1019_CLS_INT_REG_2: - case RT1019_CLS_INT_REG_3: - case RT1019_CLS_INT_REG_4: - case RT1019_CLS_INT_REG_5: - case RT1019_CLS_INT_REG_6: - case RT1019_CLS_INT_REG_7: - case RT1019_CLS_INT_REG_8: - case RT1019_CLS_INT_REG_9: - case RT1019_CLS_INT_REG_10: case RT1019_TDM_1: case RT1019_TDM_2: case RT1019_TDM_3: - case RT1019_TDM_4: - case RT1019_TDM_5: - case RT1019_TDM_6: - case RT1019_DVOL_1: - case RT1019_DVOL_2: - case RT1019_DVOL_3: - case RT1019_DVOL_4: case RT1019_DMIX_MONO_1: case RT1019_DMIX_MONO_2: - case RT1019_CAL_TOP_1: - case RT1019_CAL_TOP_2: - case RT1019_CAL_TOP_3: - case RT1019_CAL_TOP_4: - case RT1019_CAL_TOP_5: - case RT1019_CAL_TOP_6: - case RT1019_CAL_TOP_7: - case RT1019_CAL_TOP_8: - case RT1019_CAL_TOP_9: - case RT1019_CAL_TOP_10: - case RT1019_CAL_TOP_11: - case RT1019_CAL_TOP_12: - case RT1019_CAL_TOP_13: - case RT1019_CAL_TOP_14: - case RT1019_CAL_TOP_15: - case RT1019_CAL_TOP_16: - case RT1019_CAL_TOP_17: - case RT1019_CAL_TOP_18: - case RT1019_CAL_TOP_19: - case RT1019_CAL_TOP_20: - case RT1019_CAL_TOP_21: - case RT1019_CAL_TOP_22: - case RT1019_MDRE_CTRL_1: - case RT1019_MDRE_CTRL_2: - case RT1019_MDRE_CTRL_3: - case RT1019_MDRE_CTRL_4: - case RT1019_MDRE_CTRL_5: - case RT1019_MDRE_CTRL_6: - case RT1019_MDRE_CTRL_7: - case RT1019_MDRE_CTRL_8: - case RT1019_MDRE_CTRL_9: - case RT1019_MDRE_CTRL_10: - case RT1019_SCC_CTRL_1: - case RT1019_SCC_CTRL_2: - case RT1019_SCC_CTRL_3: - case RT1019_SCC_DUMMY: - case RT1019_SIL_DET_1: - case RT1019_SIL_DET_2: - case RT1019_PWM_DC_DET_1: - case RT1019_PWM_DC_DET_2: - case RT1019_PWM_DC_DET_3: - case RT1019_PWM_DC_DET_4: case RT1019_BEEP_1: case RT1019_BEEP_2: - case RT1019_PMC_1: - case RT1019_PMC_2: - case RT1019_PMC_3: - case RT1019_PMC_4: - case RT1019_PMC_5: - case RT1019_PMC_6: - case RT1019_PMC_7: - case RT1019_PMC_8: - case RT1019_PMC_9: - case RT1019_SPKDRC_1: - case RT1019_SPKDRC_2: - case RT1019_SPKDRC_3: - case RT1019_SPKDRC_4: - case RT1019_SPKDRC_5: - case RT1019_SPKDRC_6: - case RT1019_SPKDRC_7: - case RT1019_HALF_FREQ_1: - case RT1019_HALF_FREQ_2: - case RT1019_HALF_FREQ_3: - case RT1019_HALF_FREQ_4: - case RT1019_HALF_FREQ_5: - case RT1019_HALF_FREQ_6: - case RT1019_HALF_FREQ_7: - case RT1019_CUR_CTRL_1: - case RT1019_CUR_CTRL_2: - case RT1019_CUR_CTRL_3: - case RT1019_CUR_CTRL_4: - case RT1019_CUR_CTRL_5: - case RT1019_CUR_CTRL_6: - case RT1019_CUR_CTRL_7: - case RT1019_CUR_CTRL_8: - case RT1019_CUR_CTRL_9: - case RT1019_CUR_CTRL_10: - case RT1019_CUR_CTRL_11: - case RT1019_CUR_CTRL_12: - case RT1019_CUR_CTRL_13: return true; default: return false; @@ -861,7 +529,7 @@ static const struct regmap_config rt1019_regmap = { .val_bits = 8, .use_single_read = true, .use_single_write = true, - .max_register = RT1019_CUR_CTRL_13, + .max_register = RT1019_BEEP_2, .volatile_reg = rt1019_volatile_register, .readable_reg = rt1019_readable_register, .cache_type = REGCACHE_RBTREE, diff --git a/sound/soc/codecs/rt1019.h b/sound/soc/codecs/rt1019.h index 46973e71c963..64df831eeb72 100644 --- a/sound/soc/codecs/rt1019.h +++ b/sound/soc/codecs/rt1019.h @@ -12,191 +12,29 @@ #define RT1019_DEVICE_ID_VAL2 0x6731 #define RT1019_RESET 0x0000 -#define RT1019_PAD_DRV_1 0x0002 -#define RT1019_PAD_DRV_2 0x0003 -#define RT1019_PAD_PULL_1 0x0005 -#define RT1019_PAD_PULL_2 0x0006 -#define RT1019_PAD_PULL_3 0x0007 -#define RT1019_I2C_CTRL_1 0x0008 -#define RT1019_I2C_CTRL_2 0x0009 -#define RT1019_I2C_CTRL_3 0x000a #define RT1019_IDS_CTRL 0x0011 #define RT1019_ASEL_CTRL 0x0013 -#define RT1019_PLL_RESET 0x0015 -#define RT1019_PWR_STRP_1 0x0017 #define RT1019_PWR_STRP_2 0x0019 #define RT1019_BEEP_TONE 0x001b -#define RT1019_SIL_DET_GAT 0x001d -#define RT1019_CLASSD_TIME 0x001f -#define RT1019_CLASSD_OCP 0x0021 -#define RT1019_PHASE_SYNC 0x0023 -#define RT1019_STAT_MACH_1 0x0025 -#define RT1019_STAT_MACH_2 0x0026 -#define RT1019_EFF_CTRL 0x0028 -#define RT1019_FS_DET_1 0x002a -#define RT1019_FS_DET_2 0x002b -#define RT1019_FS_DET_3 0x002c -#define RT1019_FS_DET_4 0x002d -#define RT1019_FS_DET_5 0x002e -#define RT1019_FS_DET_6 0x002f -#define RT1019_FS_DET_7 0x0030 -#define RT1019_ANA_CTRL 0x0053 -#define RT1019_DUMMY_A 0x0055 -#define RT1019_DUMMY_B 0x0056 -#define RT1019_DUMMY_C 0x0057 -#define RT1019_DUMMY_D 0x0058 -#define RT1019_ANA_READ 0x005a #define RT1019_VER_ID 0x005c -#define RT1019_CUSTOM_ID 0x005d #define RT1019_VEND_ID_1 0x005e #define RT1019_VEND_ID_2 0x005f #define RT1019_DEV_ID_1 0x0061 #define RT1019_DEV_ID_2 0x0062 -#define RT1019_TEST_PAD 0x0064 #define RT1019_SDB_CTRL 0x0066 -#define RT1019_TEST_CTRL_1 0x0068 -#define RT1019_TEST_CTRL_2 0x006a -#define RT1019_TEST_CTRL_3 0x006c -#define RT1019_SCAN_MODE 0x006e #define RT1019_CLK_TREE_1 0x0100 #define RT1019_CLK_TREE_2 0x0101 #define RT1019_CLK_TREE_3 0x0102 -#define RT1019_CLK_TREE_4 0x0103 -#define RT1019_CLK_TREE_5 0x0104 -#define RT1019_CLK_TREE_6 0x0105 -#define RT1019_CLK_TREE_7 0x0106 -#define RT1019_CLK_TREE_8 0x0107 -#define RT1019_CLK_TREE_9 0x0108 -#define RT1019_ASRC_1 0x0200 -#define RT1019_ASRC_2 0x0201 -#define RT1019_ASRC_3 0x0202 -#define RT1019_ASRC_4 0x0203 -#define RT1019_SYS_CLK 0x0300 -#define RT1019_BIAS_CUR_1 0x0301 -#define RT1019_BIAS_CUR_2 0x0302 -#define RT1019_BIAS_CUR_3 0x0303 -#define RT1019_BIAS_CUR_4 0x0304 -#define RT1019_CHOP_CLK_DAC 0x0306 -#define RT1019_CHOP_CLK_ADC 0x0308 -#define RT1019_LDO_CTRL_1 0x030a -#define RT1019_LDO_CTRL_2 0x030b -#define RT1019_PM_ANA_1 0x030d -#define RT1019_PM_ANA_2 0x030e -#define RT1019_PM_ANA_3 0x030f #define RT1019_PLL_1 0x0311 #define RT1019_PLL_2 0x0312 #define RT1019_PLL_3 0x0313 -#define RT1019_PLL_INT_1 0x0315 -#define RT1019_PLL_INT_3 0x0318 -#define RT1019_MIXER 0x031a -#define RT1019_CLD_OUT_1 0x031c -#define RT1019_CLD_OUT_2 0x031d -#define RT1019_CLD_OUT_3 0x031e -#define RT1019_CLD_OUT_4 0x031f -#define RT1019_CLD_OUT_5 0x0320 -#define RT1019_CLD_OUT_6 0x0321 -#define RT1019_CLS_INT_REG_1 0x0323 -#define RT1019_CLS_INT_REG_2 0x0324 -#define RT1019_CLS_INT_REG_3 0x0325 -#define RT1019_CLS_INT_REG_4 0x0326 -#define RT1019_CLS_INT_REG_5 0x0327 -#define RT1019_CLS_INT_REG_6 0x0328 -#define RT1019_CLS_INT_REG_7 0x0329 -#define RT1019_CLS_INT_REG_8 0x0330 -#define RT1019_CLS_INT_REG_9 0x0331 -#define RT1019_CLS_INT_REG_10 0x0332 #define RT1019_TDM_1 0x0400 #define RT1019_TDM_2 0x0401 #define RT1019_TDM_3 0x0402 -#define RT1019_TDM_4 0x0403 -#define RT1019_TDM_5 0x0404 -#define RT1019_TDM_6 0x0405 -#define RT1019_DVOL_1 0x0500 -#define RT1019_DVOL_2 0x0501 -#define RT1019_DVOL_3 0x0502 -#define RT1019_DVOL_4 0x0503 #define RT1019_DMIX_MONO_1 0x0504 #define RT1019_DMIX_MONO_2 0x0505 -#define RT1019_CAL_TOP_1 0x0600 -#define RT1019_CAL_TOP_2 0x0601 -#define RT1019_CAL_TOP_3 0x0602 -#define RT1019_CAL_TOP_4 0x0603 -#define RT1019_CAL_TOP_5 0x0604 -#define RT1019_CAL_TOP_6 0x0605 -#define RT1019_CAL_TOP_7 0x0606 -#define RT1019_CAL_TOP_8 0x0607 -#define RT1019_CAL_TOP_9 0x0608 -#define RT1019_CAL_TOP_10 0x0609 -#define RT1019_CAL_TOP_11 0x060a -#define RT1019_CAL_TOP_12 0x060b -#define RT1019_CAL_TOP_13 0x060c -#define RT1019_CAL_TOP_14 0x060d -#define RT1019_CAL_TOP_15 0x060e -#define RT1019_CAL_TOP_16 0x060f -#define RT1019_CAL_TOP_17 0x0610 -#define RT1019_CAL_TOP_18 0x0611 -#define RT1019_CAL_TOP_19 0x0612 -#define RT1019_CAL_TOP_20 0x0613 -#define RT1019_CAL_TOP_21 0x0614 -#define RT1019_CAL_TOP_22 0x0615 -#define RT1019_MDRE_CTRL_1 0x0700 -#define RT1019_MDRE_CTRL_2 0x0701 -#define RT1019_MDRE_CTRL_3 0x0702 -#define RT1019_MDRE_CTRL_4 0x0703 -#define RT1019_MDRE_CTRL_5 0x0704 -#define RT1019_MDRE_CTRL_6 0x0705 -#define RT1019_MDRE_CTRL_7 0x0706 -#define RT1019_MDRE_CTRL_8 0x0707 -#define RT1019_MDRE_CTRL_9 0x0708 -#define RT1019_MDRE_CTRL_10 0x0709 -#define RT1019_SCC_CTRL_1 0x0800 -#define RT1019_SCC_CTRL_2 0x0801 -#define RT1019_SCC_CTRL_3 0x0802 -#define RT1019_SCC_DUMMY 0x0803 -#define RT1019_SIL_DET_1 0x0900 -#define RT1019_SIL_DET_2 0x0901 -#define RT1019_PWM_DC_DET_1 0x0a00 -#define RT1019_PWM_DC_DET_2 0x0a01 -#define RT1019_PWM_DC_DET_3 0x0a02 -#define RT1019_PWM_DC_DET_4 0x0a03 #define RT1019_BEEP_1 0x0b00 #define RT1019_BEEP_2 0x0b01 -#define RT1019_PMC_1 0x0c00 -#define RT1019_PMC_2 0x0c01 -#define RT1019_PMC_3 0x0c02 -#define RT1019_PMC_4 0x0c03 -#define RT1019_PMC_5 0x0c04 -#define RT1019_PMC_6 0x0c05 -#define RT1019_PMC_7 0x0c06 -#define RT1019_PMC_8 0x0c07 -#define RT1019_PMC_9 0x0c08 -#define RT1019_SPKDRC_1 0x0d00 -#define RT1019_SPKDRC_2 0x0d01 -#define RT1019_SPKDRC_3 0x0d02 -#define RT1019_SPKDRC_4 0x0d03 -#define RT1019_SPKDRC_5 0x0d04 -#define RT1019_SPKDRC_6 0x0d05 -#define RT1019_SPKDRC_7 0x0d06 -#define RT1019_HALF_FREQ_1 0x0e00 -#define RT1019_HALF_FREQ_2 0x0e01 -#define RT1019_HALF_FREQ_3 0x0e02 -#define RT1019_HALF_FREQ_4 0x0e03 -#define RT1019_HALF_FREQ_5 0x0e04 -#define RT1019_HALF_FREQ_6 0x0e05 -#define RT1019_HALF_FREQ_7 0x0e06 -#define RT1019_CUR_CTRL_1 0x0f00 -#define RT1019_CUR_CTRL_2 0x0f01 -#define RT1019_CUR_CTRL_3 0x0f02 -#define RT1019_CUR_CTRL_4 0x0f03 -#define RT1019_CUR_CTRL_5 0x0f04 -#define RT1019_CUR_CTRL_6 0x0f05 -#define RT1019_CUR_CTRL_7 0x0f06 -#define RT1019_CUR_CTRL_8 0x0f07 -#define RT1019_CUR_CTRL_9 0x0f08 -#define RT1019_CUR_CTRL_10 0x0f09 -#define RT1019_CUR_CTRL_11 0x0f0a -#define RT1019_CUR_CTRL_12 0x0f0b -#define RT1019_CUR_CTRL_13 0x0f0c /* 0x0019 Power On Strap Control-2 */ #define RT1019_AUTO_BITS_SEL_MASK (0x1 << 5) -- 2.29.0

2 1

[PATCH -next] ASoC: codecs: lpass-wsa-macro: constify static struct snd_soc_dai_ops
by Ye Bin 09 Apr '21

09 Apr '21

The snd_soc_dai_ops structures is only stored in the ops field of a snd_soc_dai_driver structure, so make the snd_soc_dai_ops structure const to allow the compiler to put it in read-only memory. Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: Ye Bin <yebin10(a)huawei.com> --- sound/soc/codecs/lpass-wsa-macro.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/lpass-wsa-macro.c b/sound/soc/codecs/lpass-wsa-macro.c index e37c8148c0de..e79a70386b4b 100644 --- a/sound/soc/codecs/lpass-wsa-macro.c +++ b/sound/soc/codecs/lpass-wsa-macro.c @@ -1031,7 +1031,7 @@ static int wsa_macro_get_channel_map(struct snd_soc_dai *dai, return 0; } -static struct snd_soc_dai_ops wsa_macro_dai_ops = { +static const struct snd_soc_dai_ops wsa_macro_dai_ops = { .hw_params = wsa_macro_hw_params, .get_channel_map = wsa_macro_get_channel_map, };

2 1

[PATCH -next] ASoC: codecs: lpass-rx-macro: constify static struct snd_soc_dai_ops
by Ye Bin 09 Apr '21

09 Apr '21

The snd_soc_dai_ops structures is only stored in the ops field of a snd_soc_dai_driver structure, so make the snd_soc_dai_ops structure const to allow the compiler to put it in read-only memory. Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: Ye Bin <yebin10(a)huawei.com> --- sound/soc/codecs/lpass-rx-macro.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/lpass-rx-macro.c b/sound/soc/codecs/lpass-rx-macro.c index 08fed40372ab..8e6a47f152ad 100644 --- a/sound/soc/codecs/lpass-rx-macro.c +++ b/sound/soc/codecs/lpass-rx-macro.c @@ -1767,7 +1767,7 @@ static int rx_macro_digital_mute(struct snd_soc_dai *dai, int mute, int stream) return 0; } -static struct snd_soc_dai_ops rx_macro_dai_ops = { +static const struct snd_soc_dai_ops rx_macro_dai_ops = { .hw_params = rx_macro_hw_params, .get_channel_map = rx_macro_get_channel_map, .mute_stream = rx_macro_digital_mute,

2 1

[PATCH -next] ASoC: codecs: lpass-tx-macro: constify static struct snd_soc_dai_ops
by Ye Bin 09 Apr '21

09 Apr '21

The snd_soc_dai_ops structures is only stored in the ops field of a snd_soc_dai_driver structure, so make the snd_soc_dai_ops structure const to allow the compiler to put it in read-only memory. Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: Ye Bin <yebin10(a)huawei.com> --- sound/soc/codecs/lpass-tx-macro.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/lpass-tx-macro.c b/sound/soc/codecs/lpass-tx-macro.c index e8c6c738bbaa..acd2fbc0ca7c 100644 --- a/sound/soc/codecs/lpass-tx-macro.c +++ b/sound/soc/codecs/lpass-tx-macro.c @@ -1124,7 +1124,7 @@ static int tx_macro_digital_mute(struct snd_soc_dai *dai, int mute, int stream) return 0; } -static struct snd_soc_dai_ops tx_macro_dai_ops = { +static const struct snd_soc_dai_ops tx_macro_dai_ops = { .hw_params = tx_macro_hw_params, .get_channel_map = tx_macro_get_channel_map, .mute_stream = tx_macro_digital_mute,

2 1

[PATCH -next] ASoC: Intel: KMB: Constify static struct snd_soc_dai_ops
by Ye Bin 09 Apr '21

09 Apr '21

The snd_soc_dai_ops structures is only stored in the ops field of a snd_soc_dai_driver structure, so make the snd_soc_dai_ops structure const to allow the compiler to put it in read-only memory. Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: Ye Bin <yebin10(a)huawei.com> --- sound/soc/intel/keembay/kmb_platform.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/keembay/kmb_platform.c b/sound/soc/intel/keembay/kmb_platform.c index 0fd1e8f62c89..ef73392f8cff 100644 --- a/sound/soc/intel/keembay/kmb_platform.c +++ b/sound/soc/intel/keembay/kmb_platform.c @@ -729,7 +729,7 @@ static int kmb_dai_hw_free(struct snd_pcm_substream *substream, return 0; } -static struct snd_soc_dai_ops kmb_dai_ops = { +static const struct snd_soc_dai_ops kmb_dai_ops = { .startup = kmb_dai_startup, .trigger = kmb_dai_trigger, .hw_params = kmb_dai_hw_params,

2 1

[PATCH -next] ASoC: amd: renoir: acp3x-pdm-dma: constify static struct snd_soc_dai_ops
by Ye Bin 09 Apr '21

09 Apr '21

The snd_soc_dai_ops structures is only stored in the ops field of a snd_soc_dai_driver structure, so make the snd_soc_dai_ops structure const to allow the compiler to put it in read-only memory. Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: Ye Bin <yebin10(a)huawei.com> --- sound/soc/amd/renoir/acp3x-pdm-dma.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/amd/renoir/acp3x-pdm-dma.c b/sound/soc/amd/renoir/acp3x-pdm-dma.c index 1acd20439399..4c2810e58dce 100644 --- a/sound/soc/amd/renoir/acp3x-pdm-dma.c +++ b/sound/soc/amd/renoir/acp3x-pdm-dma.c @@ -356,7 +356,7 @@ static int acp_pdm_dai_trigger(struct snd_pcm_substream *substream, return ret; } -static struct snd_soc_dai_ops acp_pdm_dai_ops = { +static const struct snd_soc_dai_ops acp_pdm_dai_ops = { .trigger = acp_pdm_dai_trigger, };

2 1

[PATCH -next] ASoC: wcd9335: constify static struct snd_soc_dai_ops
by Ye Bin 09 Apr '21

09 Apr '21

The snd_soc_dai_ops structures is only stored in the ops field of a snd_soc_dai_driver structure, so make the snd_soc_dai_ops structure const to allow the compiler to put it in read-only memory. Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: Ye Bin <yebin10(a)huawei.com> --- sound/soc/codecs/wcd9335.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wcd9335.c b/sound/soc/codecs/wcd9335.c index adb325fac930..86c92e03ea5d 100644 --- a/sound/soc/codecs/wcd9335.c +++ b/sound/soc/codecs/wcd9335.c @@ -2058,7 +2058,7 @@ static int wcd9335_get_channel_map(struct snd_soc_dai *dai, return 0; } -static struct snd_soc_dai_ops wcd9335_dai_ops = { +static const struct snd_soc_dai_ops wcd9335_dai_ops = { .hw_params = wcd9335_hw_params, .trigger = wcd9335_trigger, .set_channel_map = wcd9335_set_channel_map,

2 1

[PATCH -next] ASoC: rt711-sdca: Constify static struct snd_soc_dai_ops
by Ye Bin 09 Apr '21

09 Apr '21

The snd_soc_dai_ops structures is only stored in the ops field of a snd_soc_dai_driver structure, so make the snd_soc_dai_ops structure const to allow the compiler to put it in read-only memory. Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: Ye Bin <yebin10(a)huawei.com> --- sound/soc/codecs/rt711-sdca.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt711-sdca.c b/sound/soc/codecs/rt711-sdca.c index 381893b640c6..bfb7f1c8ec8f 100644 --- a/sound/soc/codecs/rt711-sdca.c +++ b/sound/soc/codecs/rt711-sdca.c @@ -1268,7 +1268,7 @@ static int rt711_sdca_pcm_hw_free(struct snd_pcm_substream *substream, #define RT711_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ SNDRV_PCM_FMTBIT_S24_LE) -static struct snd_soc_dai_ops rt711_sdca_ops = { +static const struct snd_soc_dai_ops rt711_sdca_ops = { .hw_params = rt711_sdca_pcm_hw_params, .hw_free = rt711_sdca_pcm_hw_free, .set_sdw_stream = rt711_sdca_set_sdw_stream,

2 1

[PATCH -next] ASoC: wcd934x: constify static struct snd_soc_dai_ops
by Ye Bin 09 Apr '21

09 Apr '21

The snd_soc_dai_ops structures is only stored in the ops field of a snd_soc_dai_driver structure, so make the snd_soc_dai_ops structure const to allow the compiler to put it in read-only memory. Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: Ye Bin <yebin10(a)huawei.com> --- sound/soc/codecs/wcd934x.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wcd934x.c b/sound/soc/codecs/wcd934x.c index 2c4818dc2e6d..5fe403307b72 100644 --- a/sound/soc/codecs/wcd934x.c +++ b/sound/soc/codecs/wcd934x.c @@ -1946,7 +1946,7 @@ static int wcd934x_get_channel_map(struct snd_soc_dai *dai, return 0; } -static struct snd_soc_dai_ops wcd934x_dai_ops = { +static const struct snd_soc_dai_ops wcd934x_dai_ops = { .hw_params = wcd934x_hw_params, .hw_free = wcd934x_hw_free, .trigger = wcd934x_trigger,

2 1

[PATCH V2 0/2] Add mediatek MT6359 ASoC accdet jack driver
by Argus Lin 09 Apr '21

09 Apr '21

All of 3-pole and 4-pole jack are supported. change since v2: - fixs missing blank at Kconfig. - fixs comment format and spelling mistake. - changes private structure mt6359_accdet to mt6359-accdet.h and uses this data as function parameter. - removes compatible string declaration. - uses regmap_read_poll_timeout as polling timer. - simplify jack detection and key detection report function. - adds mt6359_accdet_enable_jack_detect for sound card jack initialization. change since v1: - adds mt6359 accdet binding document - adds mt6359 accdet driver *** BLURB HERE *** Argus Lin (2): dt-bindings: mediatek: mt6359: add ASoC mt6359 ASoC accdet jack document ASoC: mediatek: mt6359: add MT6359 accdet jack driver .../bindings/sound/mt6359-accdet.yaml | 164 ++ sound/soc/codecs/Kconfig | 8 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/mt6359-accdet.c | 1080 ++++++++++ sound/soc/codecs/mt6359-accdet.h | 128 ++ sound/soc/codecs/mt6359.h | 1864 ++++++++++++++++- 6 files changed, 3139 insertions(+), 107 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/mt6359-accdet.yaml create mode 100644 sound/soc/codecs/mt6359-accdet.c create mode 100644 sound/soc/codecs/mt6359-accdet.h -- 2.18.0

3 5

[PATCH] ASoC: Intel: bytcr_wm5102: remove useless variable
by Jiapeng Chong 09 Apr '21

09 Apr '21

Fix the following gcc warning: sound/soc/intel/boards/bytcr_wm5102.c:216:40: warning: ‘byt_wm5102_dai_params’ defined but not used. Reported-by: Abaci Robot <abaci(a)linux.alibaba.com> Signed-off-by: Jiapeng Chong <jiapeng.chong(a)linux.alibaba.com> --- sound/soc/intel/boards/bytcr_wm5102.c | 8 -------- 1 file changed, 8 deletions(-) diff --git a/sound/soc/intel/boards/bytcr_wm5102.c b/sound/soc/intel/boards/bytcr_wm5102.c index f38850e..fd584e3 100644 --- a/sound/soc/intel/boards/bytcr_wm5102.c +++ b/sound/soc/intel/boards/bytcr_wm5102.c @@ -213,14 +213,6 @@ static int byt_wm5102_init(struct snd_soc_pcm_runtime *runtime) return 0; } -static const struct snd_soc_pcm_stream byt_wm5102_dai_params = { - .formats = SNDRV_PCM_FMTBIT_S16_LE, - .rate_min = 48000, - .rate_max = 48000, - .channels_min = 2, - .channels_max = 2, -}; - static int byt_wm5102_codec_fixup(struct snd_soc_pcm_runtime *rtd, struct snd_pcm_hw_params *params) { -- 1.8.3.1

2 1

[PATCH 1/2] ALSA: control - fix a leak in snd_ctl_led_init()
by Dan Carpenter 09 Apr '21

09 Apr '21

This unwind loop needs to free snd_ctl_leds[0] as well. Fixes: cb17fe0045aa ("ALSA: control - add sysfs support to the LED trigger module") Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- sound/core/control_led.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/core/control_led.c b/sound/core/control_led.c index d4fb8b873f34..202b475d0bf3 100644 --- a/sound/core/control_led.c +++ b/sound/core/control_led.c @@ -712,7 +712,7 @@ static struct snd_ctl_layer_ops snd_ctl_led_lops = { static int __init snd_ctl_led_init(void) { struct snd_ctl_led *led; - unsigned int group; + int group; device_initialize(&snd_ctl_led_dev); snd_ctl_led_dev.class = sound_class; @@ -730,7 +730,7 @@ static int __init snd_ctl_led_init(void) dev_set_name(&led->dev, led->name); if (device_add(&led->dev)) { put_device(&led->dev); - for (; group > 0; group--) { + for (; group >= 0; group--) { led = &snd_ctl_leds[group]; device_del(&led->dev); } -- 2.30.2

3 7

[PATCH -next] ASoC: sun4i-codec: remove redundant dev_err call in sun4i_codec_probe()
by Qiheng Lin 09 Apr '21

09 Apr '21

There is a error message within devm_ioremap_resource already, so remove the dev_err call to avoid redundant error message. Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: Qiheng Lin <linqiheng(a)huawei.com> --- sound/soc/sunxi/sun4i-codec.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/sunxi/sun4i-codec.c b/sound/soc/sunxi/sun4i-codec.c index 6c13cc84b3fb..00b73b7444b7 100644 --- a/sound/soc/sunxi/sun4i-codec.c +++ b/sound/soc/sunxi/sun4i-codec.c @@ -1706,10 +1706,8 @@ static int sun4i_codec_probe(struct platform_device *pdev) res = platform_get_resource(pdev, IORESOURCE_MEM, 0); base = devm_ioremap_resource(&pdev->dev, res); - if (IS_ERR(base)) { - dev_err(&pdev->dev, "Failed to map the registers\n"); + if (IS_ERR(base)) return PTR_ERR(base); - } quirks = of_device_get_match_data(&pdev->dev); if (quirks == NULL) {

1 0

[PATCH] ALSA: hda/realtek: fix mic boost on Intel NUC 8
by Phil Calvin 09 Apr '21

09 Apr '21

Just wanted to say thank you to everybody that helps maintain the Intel HDA subsystem. My first Linux sound experience was running `modprobe cs46xx` or so on a SuSE installation and hearing a promising noise from the speakers, and these days it seems audio works out of the box on most hardware. Thanks also to Takashi Iwai for `hda-emu`, which saved me a lot of rebooting while testing. I have tested the patch on this hardware both by recompiling the relevant modules and running them with the 5.10 kernel distributed in Debian `testing`, and atop the 5.12-rc6 kernel built with `make deb-pkg`. I could not decipher the ordering of the fixup tables, so I am of course happy to move these around to fit the organizational scheme as needed. I'm not an EE, but I am inclined to think the upper two boost values can probably be made to work by some kind of software workaround, given that they do produce analog noise coming from something. If you have any hunches about a possible workaround based on experience with other similar codecs, please point me to those fixups on- or off-list and I can try them out on my hardware. Phil >8------------------------------------------------------8< Fix two bugs with the Intel HDA Realtek ALC233 sound codec present in Intel NUC NUC8i7BEH and probably a few other similar NUC models. These codecs advertise a 4-level microphone input boost amplifier on pin 0x19, but the highest two boost settings do not work correctly, and produce only low analog noise that does not seem to contain any discernible signal. There is an existing fixup for this exact problem but for a different PCI subsystem ID, so we re-use that logic. Changing the boost level also triggers a DC spike in the input signal that bleeds off over about a second and overwhelms any input during that time. Thankfully, the existing fixup has the side effect of making the boost control show up in userspace as a mute/unmute switch, and this keeps (e.g.) PulseAudio from fiddling with it during normal input volume adjustments. Finally, the NUC hardware has built-in inverted stereo mics. This patch also enables the usual fixup for this so the two channels cancel noise instead of the actual signal. Signed-off-by: Phil Calvin <phil(a)philcalvin.com> --- sound/pci/hda/patch_realtek.c | 13 +++++++++++++ 1 file changed, 13 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 58946d069ee5..e1fd4c81965a 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6405,6 +6405,8 @@ enum { ALC269_FIXUP_LEMOTE_A1802, ALC269_FIXUP_LEMOTE_A190X, ALC256_FIXUP_INTEL_NUC8_RUGGED, + ALC233_FIXUP_INTEL_NUC8_DMIC, + ALC233_FIXUP_INTEL_NUC8_BOOST, ALC256_FIXUP_INTEL_NUC10, ALC255_FIXUP_XIAOMI_HEADSET_MIC, ALC274_FIXUP_HP_MIC, @@ -7122,6 +7124,16 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc233_fixup_lenovo_line2_mic_hotkey, }, + [ALC233_FIXUP_INTEL_NUC8_DMIC] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc_fixup_inv_dmic, + .chained = true, + .chain_id = ALC233_FIXUP_INTEL_NUC8_BOOST, + }, + [ALC233_FIXUP_INTEL_NUC8_BOOST] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc269_fixup_limit_int_mic_boost + }, [ALC255_FIXUP_DELL_SPK_NOISE] = { .type = HDA_FIXUP_FUNC, .v.func = alc_fixup_disable_aamix, @@ -8265,6 +8277,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1c06, 0x2013, "Lemote A1802", ALC269_FIXUP_LEMOTE_A1802), SND_PCI_QUIRK(0x1c06, 0x2015, "Lemote A190X", ALC269_FIXUP_LEMOTE_A190X), SND_PCI_QUIRK(0x8086, 0x2080, "Intel NUC 8 Rugged", ALC256_FIXUP_INTEL_NUC8_RUGGED), + SND_PCI_QUIRK(0x8086, 0x2074, "Intel NUC 8", ALC233_FIXUP_INTEL_NUC8_DMIC), SND_PCI_QUIRK(0x8086, 0x2081, "Intel NUC 10", ALC256_FIXUP_INTEL_NUC10), #if 0 -- 2.30.2

1 0

[PATCH -next] ASoC: rt1019: constify static struct snd_soc_dai_ops
by Ye Bin 08 Apr '21

08 Apr '21

The snd_soc_dai_ops structures is only stored in the ops field of a snd_soc_dai_driver structure, so make the snd_soc_dai_ops structure const to allow the compiler to put it in read-only memory. Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: Ye Bin <yebin10(a)huawei.com> --- sound/soc/codecs/rt1019.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt1019.c b/sound/soc/codecs/rt1019.c index 5138f028d9f2..f64bb7302720 100644 --- a/sound/soc/codecs/rt1019.c +++ b/sound/soc/codecs/rt1019.c @@ -823,7 +823,7 @@ static int rt1019_probe(struct snd_soc_component *component) #define RT1019_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S8) -static struct snd_soc_dai_ops rt1019_aif_dai_ops = { +static const struct snd_soc_dai_ops rt1019_aif_dai_ops = { .hw_params = rt1019_hw_params, .set_fmt = rt1019_set_dai_fmt, .set_sysclk = rt1019_set_dai_sysclk,

2 1

[PATCH] ASoC: rt1011: remove pack_id check in rt1011
by Jack Yu 08 Apr '21

08 Apr '21

For latest design, different package could use the same setting, therefore the check of pack_id will no longer be used. Signed-off-by: Jack Yu <jack.yu(a)realtek.com> --- sound/soc/codecs/rt1011.c | 16 ++-------------- sound/soc/codecs/rt1011.h | 1 - 2 files changed, 2 insertions(+), 15 deletions(-) diff --git a/sound/soc/codecs/rt1011.c b/sound/soc/codecs/rt1011.c index 238cb66f4c6a..faff2b558687 100644 --- a/sound/soc/codecs/rt1011.c +++ b/sound/soc/codecs/rt1011.c @@ -2236,18 +2236,9 @@ static int rt1011_calibrate(struct rt1011_priv *rt1011, unsigned char cali_flag) dc_offset |= (value & 0xffff); dev_info(dev, "Gain1 offset=0x%x\n", dc_offset); - /* check the package info. */ - regmap_read(rt1011->regmap, RT1011_EFUSE_MATCH_DONE, &value); - if (value & 0x4) - rt1011->pack_id = 1; - if (cali_flag) { - if (rt1011->pack_id) - regmap_write(rt1011->regmap, RT1011_ADC_SET_1, 0x292c); - else - regmap_write(rt1011->regmap, RT1011_ADC_SET_1, 0x2925); - + regmap_write(rt1011->regmap, RT1011_ADC_SET_1, 0x2925); /* Class D on */ regmap_write(rt1011->regmap, RT1011_CLASS_D_POS, 0x010e); regmap_write(rt1011->regmap, @@ -2373,10 +2364,7 @@ static void rt1011_calibration_work(struct work_struct *work) rt1011_r0_load(rt1011); } - if (rt1011->pack_id) - snd_soc_component_write(component, RT1011_ADC_SET_1, 0x292c); - else - snd_soc_component_write(component, RT1011_ADC_SET_1, 0x2925); + snd_soc_component_write(component, RT1011_ADC_SET_1, 0x2925); } static int rt1011_parse_dp(struct rt1011_priv *rt1011, struct device *dev) diff --git a/sound/soc/codecs/rt1011.h b/sound/soc/codecs/rt1011.h index f3a9a96640f1..68fadc15fa8c 100644 --- a/sound/soc/codecs/rt1011.h +++ b/sound/soc/codecs/rt1011.h @@ -692,7 +692,6 @@ struct rt1011_priv { unsigned int r0_reg, cali_done; unsigned int r0_calib, temperature_calib; int recv_spk_mode; - unsigned int pack_id; /* 0: WLCSP; 1: QFN */ }; #endif /* end of _RT1011_H_ */ -- 2.29.0

2 1

[PATCH -next] ASoC: codecs: wsa881x: constify static struct snd_soc_dai_ops
by Ye Bin 08 Apr '21

08 Apr '21

The snd_soc_dai_ops structures is only stored in the ops field of a snd_soc_dai_driver structure, so make the snd_soc_dai_ops structure const to allow the compiler to put it in read-only memory. Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: Ye Bin <yebin10(a)huawei.com> --- sound/soc/codecs/wsa881x.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wsa881x.c b/sound/soc/codecs/wsa881x.c index db87e07b11c9..2da4a5fa7a18 100644 --- a/sound/soc/codecs/wsa881x.c +++ b/sound/soc/codecs/wsa881x.c @@ -1014,7 +1014,7 @@ static int wsa881x_digital_mute(struct snd_soc_dai *dai, int mute, int stream) return 0; } -static struct snd_soc_dai_ops wsa881x_dai_ops = { +static const struct snd_soc_dai_ops wsa881x_dai_ops = { .hw_params = wsa881x_hw_params, .hw_free = wsa881x_hw_free, .mute_stream = wsa881x_digital_mute,

2 1

[PATCH] ASoC: fsl: sunxi: remove redundant dev_err call
by Muhammad Usama Anjum 08 Apr '21

08 Apr '21

devm_ioremap_resource() prints error message in itself. Remove the dev_err call to avoid redundant error message. Signed-off-by: Muhammad Usama Anjum <musamaanjum(a)gmail.com> --- sound/soc/fsl/fsl_aud2htx.c | 4 +--- sound/soc/fsl/fsl_easrc.c | 4 +--- sound/soc/sunxi/sun4i-codec.c | 4 +--- 3 files changed, 3 insertions(+), 9 deletions(-) diff --git a/sound/soc/fsl/fsl_aud2htx.c b/sound/soc/fsl/fsl_aud2htx.c index d70d5e75a30c..a328697511f7 100644 --- a/sound/soc/fsl/fsl_aud2htx.c +++ b/sound/soc/fsl/fsl_aud2htx.c @@ -198,10 +198,8 @@ static int fsl_aud2htx_probe(struct platform_device *pdev) res = platform_get_resource(pdev, IORESOURCE_MEM, 0); regs = devm_ioremap_resource(&pdev->dev, res); - if (IS_ERR(regs)) { - dev_err(&pdev->dev, "failed ioremap\n"); + if (IS_ERR(regs)) return PTR_ERR(regs); - } aud2htx->regmap = devm_regmap_init_mmio(&pdev->dev, regs, &fsl_aud2htx_regmap_config); diff --git a/sound/soc/fsl/fsl_easrc.c b/sound/soc/fsl/fsl_easrc.c index 5e33afe87c4a..b1765c7d3bcd 100644 --- a/sound/soc/fsl/fsl_easrc.c +++ b/sound/soc/fsl/fsl_easrc.c @@ -1889,10 +1889,8 @@ static int fsl_easrc_probe(struct platform_device *pdev) res = platform_get_resource(pdev, IORESOURCE_MEM, 0); regs = devm_ioremap_resource(dev, res); - if (IS_ERR(regs)) { - dev_err(&pdev->dev, "failed ioremap\n"); + if (IS_ERR(regs)) return PTR_ERR(regs); - } easrc->paddr = res->start; diff --git a/sound/soc/sunxi/sun4i-codec.c b/sound/soc/sunxi/sun4i-codec.c index 2173991c13db..6f3d9148a185 100644 --- a/sound/soc/sunxi/sun4i-codec.c +++ b/sound/soc/sunxi/sun4i-codec.c @@ -1711,10 +1711,8 @@ static int sun4i_codec_probe(struct platform_device *pdev) res = platform_get_resource(pdev, IORESOURCE_MEM, 0); base = devm_ioremap_resource(&pdev->dev, res); - if (IS_ERR(base)) { - dev_err(&pdev->dev, "Failed to map the registers\n"); + if (IS_ERR(base)) return PTR_ERR(base); - } quirks = of_device_get_match_data(&pdev->dev); if (quirks == NULL) { -- 2.25.1

2 1

[PATCH -next] ASoC: cx2072x: constify static struct snd_soc_dai_ops
by Ye Bin 08 Apr '21

08 Apr '21

The snd_soc_dai_ops structures is only stored in the ops field of a snd_soc_dai_driver structure, so make the snd_soc_dai_ops structure const to allow the compiler to put it in read-only memory. Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: Ye Bin <yebin10(a)huawei.com> --- sound/soc/codecs/cx2072x.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/cx2072x.c b/sound/soc/codecs/cx2072x.c index d924e3528029..1f5c57fab1d8 100644 --- a/sound/soc/codecs/cx2072x.c +++ b/sound/soc/codecs/cx2072x.c @@ -1532,7 +1532,7 @@ static const struct snd_soc_component_driver soc_codec_driver_cx2072x = { /* * DAI ops */ -static struct snd_soc_dai_ops cx2072x_dai_ops = { +static const struct snd_soc_dai_ops cx2072x_dai_ops = { .set_sysclk = cx2072x_set_dai_sysclk, .set_fmt = cx2072x_set_dai_fmt, .hw_params = cx2072x_hw_params,

2 1

[PATCH -next] ASoC: tas2770: Constify static struct snd_soc_dai_ops
by Ye Bin 08 Apr '21

08 Apr '21

The snd_soc_dai_ops structures is only stored in the ops field of a snd_soc_dai_driver structure, so make the snd_soc_dai_ops structure const to allow the compiler to put it in read-only memory. Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: Ye Bin <yebin10(a)huawei.com> --- sound/soc/codecs/tas2770.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/tas2770.c b/sound/soc/codecs/tas2770.c index 781bf9cc4faa..172e79cbe0da 100644 --- a/sound/soc/codecs/tas2770.c +++ b/sound/soc/codecs/tas2770.c @@ -464,7 +464,7 @@ static int tas2770_set_dai_tdm_slot(struct snd_soc_dai *dai, return 0; } -static struct snd_soc_dai_ops tas2770_dai_ops = { +static const struct snd_soc_dai_ops tas2770_dai_ops = { .mute_stream = tas2770_mute, .hw_params = tas2770_hw_params, .set_fmt = tas2770_set_fmt,

2 1

[PATCH -next] ASoC: tas2764: constify static struct snd_soc_dai_ops
by Ye Bin 08 Apr '21

08 Apr '21

The snd_soc_dai_ops structures is only stored in the ops field of a snd_soc_dai_driver structure, so make the snd_soc_dai_ops structure const to allow the compiler to put it in read-only memory. Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: Ye Bin <yebin10(a)huawei.com> --- sound/soc/codecs/tas2764.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/tas2764.c b/sound/soc/codecs/tas2764.c index 8ff4d9e8d568..9265af41c235 100644 --- a/sound/soc/codecs/tas2764.c +++ b/sound/soc/codecs/tas2764.c @@ -457,7 +457,7 @@ static int tas2764_set_dai_tdm_slot(struct snd_soc_dai *dai, return 0; } -static struct snd_soc_dai_ops tas2764_dai_ops = { +static const struct snd_soc_dai_ops tas2764_dai_ops = { .mute_stream = tas2764_mute, .hw_params = tas2764_hw_params, .set_fmt = tas2764_set_fmt,

2 1

[PATCH] [v2] ASoC: codecs: Fix runtime PM imbalance in tas2552_probe
by Dinghao Liu 08 Apr '21

08 Apr '21

There is a rumtime PM imbalance between the error handling path after devm_snd_soc_register_component() and all other error handling paths. Add a PM runtime increment to balance refcount. Signed-off-by: Dinghao Liu <dinghao.liu(a)zju.edu.cn> --- Changelog: v2: - Add a PM runtime increment to fix it instead of moving the PM related operations after the registration. --- sound/soc/codecs/tas2552.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/tas2552.c b/sound/soc/codecs/tas2552.c index bd00c35116cd..700baa6314aa 100644 --- a/sound/soc/codecs/tas2552.c +++ b/sound/soc/codecs/tas2552.c @@ -730,8 +730,10 @@ static int tas2552_probe(struct i2c_client *client, ret = devm_snd_soc_register_component(&client->dev, &soc_component_dev_tas2552, tas2552_dai, ARRAY_SIZE(tas2552_dai)); - if (ret < 0) + if (ret < 0) { dev_err(&client->dev, "Failed to register component: %d\n", ret); + pm_runtime_get_noresume(&client->dev); + } return ret; } -- 2.17.1

2 1

[PATCH] ASoC: codecs: tlv320aic3x: add AIC3106
by Jiri Prchal 08 Apr '21

08 Apr '21

In DT binding is mentioned that this driver is compatible with 3106. So added compatibility string and model number. Signed-off-by: Jiri Prchal <jiri.prchal(a)aksignal.cz> --- sound/soc/codecs/tlv320aic3x-i2c.c | 2 ++ sound/soc/codecs/tlv320aic3x-spi.c | 2 ++ sound/soc/codecs/tlv320aic3x.h | 1 + 3 files changed, 5 insertions(+) diff --git a/sound/soc/codecs/tlv320aic3x-i2c.c b/sound/soc/codecs/tlv320aic3x-i2c.c index 4535bb8dbadc..cd0558ed4dd4 100644 --- a/sound/soc/codecs/tlv320aic3x-i2c.c +++ b/sound/soc/codecs/tlv320aic3x-i2c.c @@ -40,6 +40,7 @@ static const struct i2c_device_id aic3x_i2c_id[] = { { "tlv320aic33", AIC3X_MODEL_33 }, { "tlv320aic3007", AIC3X_MODEL_3007 }, { "tlv320aic3104", AIC3X_MODEL_3104 }, + { "tlv320aic3106", AIC3X_MODEL_3106 }, { } }; MODULE_DEVICE_TABLE(i2c, aic3x_i2c_id); @@ -49,6 +50,7 @@ static const struct of_device_id aic3x_of_id[] = { { .compatible = "ti,tlv320aic33" }, { .compatible = "ti,tlv320aic3007" }, { .compatible = "ti,tlv320aic3104" }, + { .compatible = "ti,tlv320aic3106" }, {}, }; MODULE_DEVICE_TABLE(of, aic3x_of_id); diff --git a/sound/soc/codecs/tlv320aic3x-spi.c b/sound/soc/codecs/tlv320aic3x-spi.c index bd78ab0910da..8c7b6bb9223f 100644 --- a/sound/soc/codecs/tlv320aic3x-spi.c +++ b/sound/soc/codecs/tlv320aic3x-spi.c @@ -45,6 +45,7 @@ static const struct spi_device_id aic3x_spi_id[] = { { "tlv320aic33", AIC3X_MODEL_33 }, { "tlv320aic3007", AIC3X_MODEL_3007 }, { "tlv320aic3104", AIC3X_MODEL_3104 }, + { "tlv320aic3106", AIC3X_MODEL_3106 }, { } }; MODULE_DEVICE_TABLE(spi, aic3x_spi_id); @@ -54,6 +55,7 @@ static const struct of_device_id aic3x_of_id[] = { { .compatible = "ti,tlv320aic33" }, { .compatible = "ti,tlv320aic3007" }, { .compatible = "ti,tlv320aic3104" }, + { .compatible = "ti,tlv320aic3106" }, {}, }; MODULE_DEVICE_TABLE(of, aic3x_of_id); diff --git a/sound/soc/codecs/tlv320aic3x.h b/sound/soc/codecs/tlv320aic3x.h index 00908f1734a7..7e0063913017 100644 --- a/sound/soc/codecs/tlv320aic3x.h +++ b/sound/soc/codecs/tlv320aic3x.h @@ -20,6 +20,7 @@ int aic3x_remove(struct device *dev); #define AIC3X_MODEL_33 1 #define AIC3X_MODEL_3007 2 #define AIC3X_MODEL_3104 3 +#define AIC3X_MODEL_3106 4 /* AIC3X register space */ #define AIC3X_CACHEREGNUM 110 -- 2.25.1

2 1

[PATCH v2] ASoC: codecs: tlv320aic3x: add AIC3106
by Jiri Prchal 08 Apr '21

08 Apr '21

In DT binding is mentioned that this driver is compatible with 3106. So added compatibility string and model number. -v2: Also added in switches "case AIC3X_MODEL_3106:" to have right widgets etc. Signed-off-by: Jiri Prchal <jiri.prchal(a)aksignal.cz> --- sound/soc/codecs/tlv320aic3x-i2c.c | 2 ++ sound/soc/codecs/tlv320aic3x-spi.c | 2 ++ sound/soc/codecs/tlv320aic3x.c | 3 +++ sound/soc/codecs/tlv320aic3x.h | 1 + 4 files changed, 8 insertions(+) diff --git a/sound/soc/codecs/tlv320aic3x-i2c.c b/sound/soc/codecs/tlv320aic3x-i2c.c index 4535bb8dbadc..cd0558ed4dd4 100644 --- a/sound/soc/codecs/tlv320aic3x-i2c.c +++ b/sound/soc/codecs/tlv320aic3x-i2c.c @@ -40,6 +40,7 @@ static const struct i2c_device_id aic3x_i2c_id[] = { { "tlv320aic33", AIC3X_MODEL_33 }, { "tlv320aic3007", AIC3X_MODEL_3007 }, { "tlv320aic3104", AIC3X_MODEL_3104 }, + { "tlv320aic3106", AIC3X_MODEL_3106 }, { } }; MODULE_DEVICE_TABLE(i2c, aic3x_i2c_id); @@ -49,6 +50,7 @@ static const struct of_device_id aic3x_of_id[] = { { .compatible = "ti,tlv320aic33" }, { .compatible = "ti,tlv320aic3007" }, { .compatible = "ti,tlv320aic3104" }, + { .compatible = "ti,tlv320aic3106" }, {}, }; MODULE_DEVICE_TABLE(of, aic3x_of_id); diff --git a/sound/soc/codecs/tlv320aic3x-spi.c b/sound/soc/codecs/tlv320aic3x-spi.c index bd78ab0910da..8c7b6bb9223f 100644 --- a/sound/soc/codecs/tlv320aic3x-spi.c +++ b/sound/soc/codecs/tlv320aic3x-spi.c @@ -45,6 +45,7 @@ static const struct spi_device_id aic3x_spi_id[] = { { "tlv320aic33", AIC3X_MODEL_33 }, { "tlv320aic3007", AIC3X_MODEL_3007 }, { "tlv320aic3104", AIC3X_MODEL_3104 }, + { "tlv320aic3106", AIC3X_MODEL_3106 }, { } }; MODULE_DEVICE_TABLE(spi, aic3x_spi_id); @@ -54,6 +55,7 @@ static const struct of_device_id aic3x_of_id[] = { { .compatible = "ti,tlv320aic33" }, { .compatible = "ti,tlv320aic3007" }, { .compatible = "ti,tlv320aic3104" }, + { .compatible = "ti,tlv320aic3106" }, {}, }; MODULE_DEVICE_TABLE(of, aic3x_of_id); diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 2f351747b714..7731593a5509 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -1003,6 +1003,7 @@ static int aic3x_add_widgets(struct snd_soc_component *component) switch (aic3x->model) { case AIC3X_MODEL_3X: case AIC3X_MODEL_33: + case AIC3X_MODEL_3106: snd_soc_dapm_new_controls(dapm, aic3x_extra_dapm_widgets, ARRAY_SIZE(aic3x_extra_dapm_widgets)); snd_soc_dapm_add_routes(dapm, intercon_extra, @@ -1580,6 +1581,7 @@ static int aic3x_init(struct snd_soc_component *component) switch (aic3x->model) { case AIC3X_MODEL_3X: case AIC3X_MODEL_33: + case AIC3X_MODEL_3106: aic3x_mono_init(component); break; case AIC3X_MODEL_3007: @@ -1646,6 +1648,7 @@ static int aic3x_component_probe(struct snd_soc_component *component) switch (aic3x->model) { case AIC3X_MODEL_3X: case AIC3X_MODEL_33: + case AIC3X_MODEL_3106: snd_soc_add_component_controls(component, aic3x_extra_snd_controls, ARRAY_SIZE(aic3x_extra_snd_controls)); snd_soc_add_component_controls(component, aic3x_mono_controls, diff --git a/sound/soc/codecs/tlv320aic3x.h b/sound/soc/codecs/tlv320aic3x.h index 00908f1734a7..7e0063913017 100644 --- a/sound/soc/codecs/tlv320aic3x.h +++ b/sound/soc/codecs/tlv320aic3x.h @@ -20,6 +20,7 @@ int aic3x_remove(struct device *dev); #define AIC3X_MODEL_33 1 #define AIC3X_MODEL_3007 2 #define AIC3X_MODEL_3104 3 +#define AIC3X_MODEL_3106 4 /* AIC3X register space */ #define AIC3X_CACHEREGNUM 110 -- 2.25.1

2 2

[PATCH] ASoC: codecs: tlv320aic3x: add SPI to the DT binding
by Jiri Prchal 08 Apr '21

08 Apr '21

Added note and example of SPI support. Signed-off-by: Jiri Prchal <jiri.prchal(a)aksignal.cz> --- .../devicetree/bindings/sound/tlv320aic3x.txt | 21 +++++++++++++++++-- 1 file changed, 19 insertions(+), 2 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/tlv320aic3x.txt b/Documentation/devicetree/bindings/sound/tlv320aic3x.txt index 9796c4639262..20931a63fd64 100644 --- a/Documentation/devicetree/bindings/sound/tlv320aic3x.txt +++ b/Documentation/devicetree/bindings/sound/tlv320aic3x.txt @@ -1,6 +1,6 @@ Texas Instruments - tlv320aic3x Codec module -The tlv320aic3x serial control bus communicates through I2C protocols +The tlv320aic3x serial control bus communicates through both I2C and SPI bus protocols Required properties: @@ -63,7 +63,7 @@ CODEC input pins for other compatible codecs: The pins can be used in referring sound node's audio-routing property. -Example: +I2C example: #include <dt-bindings/gpio/gpio.h> @@ -78,3 +78,20 @@ tlv320aic3x: tlv320aic3x@1b { DRVDD-supply = <&regulator>; DVDD-supply = <&regulator>; }; + +SPI example: + +spi0: spi@f0000000 { + tlv320aic3x: codec@0 { + compatible = "ti,tlv320aic3x"; + reg = <0>; /* CS number */ + #sound-dai-cells = <0>; + spi-max-frequency = <1000000>; + + AVDD-supply = <&regulator>; + IOVDD-supply = <&regulator>; + DRVDD-supply = <&regulator>; + DVDD-supply = <&regulator>; + ai3x-ocmv = <0>; + }; +}; -- 2.25.1

2 1

[PATCH] ASoC: fsl_aud2htx: Remove redundant dev_err call in fsl_aud2htx_probe()
by yangerkun 08 Apr '21

08 Apr '21

There is a error message within devm_ioremap_resource already, so remove the dev_err call to avoid redundant error message. Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: yangerkun <yangerkun(a)huawei.com> --- sound/soc/fsl/fsl_aud2htx.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/fsl/fsl_aud2htx.c b/sound/soc/fsl/fsl_aud2htx.c index d70d5e75a30c..a328697511f7 100644 --- a/sound/soc/fsl/fsl_aud2htx.c +++ b/sound/soc/fsl/fsl_aud2htx.c @@ -198,10 +198,8 @@ static int fsl_aud2htx_probe(struct platform_device *pdev) res = platform_get_resource(pdev, IORESOURCE_MEM, 0); regs = devm_ioremap_resource(&pdev->dev, res); - if (IS_ERR(regs)) { - dev_err(&pdev->dev, "failed ioremap\n"); + if (IS_ERR(regs)) return PTR_ERR(regs); - } aud2htx->regmap = devm_regmap_init_mmio(&pdev->dev, regs, &fsl_aud2htx_regmap_config); -- 2.25.4

1 0

[RFC PATCH] ALSA: hda/hdmi: fix race in handling acomp ELD notification at resume
by Kai Vehmanen 08 Apr '21

08 Apr '21

When snd-hda-codec-hdmi is used with ASoC HDA controller like SOF (acomp used for ELD notifications), display connection change done during suspend, can be lost due to following sequence of events: 1. system in S3 suspend 2. DP/HDMI receiver connected 3. system resumed 4. HDA controller resumed, but card->deferred_resume_work not complete 5. acomp eld_notify callback 6. eld_notify ignored as power state is not CTL_POWER_D0 7. HDA resume deferred work completed, power state set to CTL_POWER_D0 This results in losing the notification, and the jack state reported to user-space is not correct. The check on step 6 was added in commit 8ae743e82f0b ("ALSA: hda - Skip ELD notification during system suspend"). It would seem with the deferred resume logic in ASoC core, this check is not safe. Fix the issue by modifying the check to only skip ELD notification processing if power state is D3 or deeper. This helps in the ASoC controller case as card power state is set to D2 at start of soc_resume_deferred(). BugLink: https://github.com/thesofproject/linux/issues/2825 Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> --- sound/pci/hda/patch_hdmi.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) NOTES: - I wonder if there is a better way to check for system suspend case than looking at snd_power_get_state() - 'chip->pm_prepared' is one option, but this is not directly available to codec drivers - storing PM target is hda_codec_pm_prepare() is perhaps one option diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 5de3666a7101..a43df036db1d 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -2654,7 +2654,7 @@ static void generic_acomp_pin_eld_notify(void *audio_ptr, int port, int dev_id) /* skip notification during system suspend (but not in runtime PM); * the state will be updated at resume */ - if (snd_power_get_state(codec->card) != SNDRV_CTL_POWER_D0) + if (snd_power_get_state(codec->card) >= SNDRV_CTL_POWER_D3) return; /* ditto during suspend/resume process itself */ if (snd_hdac_is_in_pm(&codec->core)) @@ -2840,7 +2840,7 @@ static void intel_pin_eld_notify(void *audio_ptr, int port, int pipe) /* skip notification during system suspend (but not in runtime PM); * the state will be updated at resume */ - if (snd_power_get_state(codec->card) != SNDRV_CTL_POWER_D0) + if (snd_power_get_state(codec->card) >= SNDRV_CTL_POWER_D3) return; /* ditto during suspend/resume process itself */ if (snd_hdac_is_in_pm(&codec->core)) base-commit: 7dc53a38e4ac00d68943bab91deadc67f07d4a0b -- 2.31.0

3 4

[PATCH] ALSA: usb-audio: Skip probe of UA-101 devices
by Takashi Iwai 08 Apr '21

08 Apr '21

UA-101 device and co are supported by another driver, snd-ua101, but the USB audio class driver (snd-usb-audio) catches all and this resulted in the lack of functionality like missing MIDI devices. This patch introduces a sort of deny-listing for those devices to just return -ENODEV at probe in snd-usb-audio driver, so that it falls back to the probe by snd-ua101. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=212477 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/card.c | 2 ++ sound/usb/quirks-table.h | 21 +++++++++++++++++++++ sound/usb/usbaudio.h | 1 + 3 files changed, 24 insertions(+) diff --git a/sound/usb/card.c b/sound/usb/card.c index b6f4c0848e66..a482287a2d7f 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -715,6 +715,8 @@ static int usb_audio_probe(struct usb_interface *intf, quirk = get_alias_quirk(dev, id); if (quirk && quirk->ifnum >= 0 && ifnum != quirk->ifnum) return -ENXIO; + if (quirk && quirk->ifnum == QUIRK_NODEV_INTERFACE) + return -ENODEV; err = snd_usb_apply_boot_quirk(dev, intf, quirk, id); if (err < 0) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 1165a5ac60f2..1f555d7b7752 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -1674,6 +1674,27 @@ YAMAHA_DEVICE(0x7010, "UB99"), } } }, + +/* UA101 and co are supported by another driver */ +{ + USB_DEVICE(0x0582, 0x0044), /* UA-1000 high speed */ + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + .ifnum = QUIRK_NODEV_INTERFACE + }, +}, +{ + USB_DEVICE(0x0582, 0x007d), /* UA-101 high speed */ + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + .ifnum = QUIRK_NODEV_INTERFACE + }, +}, +{ + USB_DEVICE(0x0582, 0x008d), /* UA-101 full speed */ + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + .ifnum = QUIRK_NODEV_INTERFACE + }, +}, + /* this catches most recent vendor-specific Roland devices */ { .match_flags = USB_DEVICE_ID_MATCH_VENDOR | diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index 60b9dd7df6bb..7f91ec1c5646 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -75,6 +75,7 @@ struct snd_usb_audio { */ /* special values for .ifnum */ +#define QUIRK_NODEV_INTERFACE -3 /* return -ENODEV */ #define QUIRK_NO_INTERFACE -2 #define QUIRK_ANY_INTERFACE -1 -- 2.26.2

1 0

[PATCH] slimbus: qcom-ngd-ctrl: Fix runtime PM imbalance in qcom_slim_ngd_enable
by Dinghao Liu 08 Apr '21

08 Apr '21

When slim_register_controller() fails, a pairing PM usage counter increment is needed to keep the counter balanced. Signed-off-by: Dinghao Liu <dinghao.liu(a)zju.edu.cn> --- drivers/slimbus/qcom-ngd-ctrl.c | 1 + 1 file changed, 1 insertion(+) diff --git a/drivers/slimbus/qcom-ngd-ctrl.c b/drivers/slimbus/qcom-ngd-ctrl.c index c054e83ab636..99cf2ab3d862 100644 --- a/drivers/slimbus/qcom-ngd-ctrl.c +++ b/drivers/slimbus/qcom-ngd-ctrl.c @@ -1268,6 +1268,7 @@ static int qcom_slim_ngd_enable(struct qcom_slim_ngd_ctrl *ctrl, bool enable) ret = slim_register_controller(&ctrl->ctrl); if (ret) { dev_err(ctrl->dev, "error adding slim controller\n"); + pm_runtime_get_noresume(ctrl->dev); return ret; } -- 2.17.1

1 0

[PATCH] ASoC: codecs: Fix rumtime PM imbalance in tas2552_probe
by Dinghao Liu 08 Apr '21

08 Apr '21

There is a rumtime PM imbalance between the error handling path after devm_snd_soc_register_component() and all other error handling paths. Fix this by moving PM runtime decrement to the end of the function. Signed-off-by: Dinghao Liu <dinghao.liu(a)zju.edu.cn> --- sound/soc/codecs/tas2552.c | 14 +++++++------- 1 file changed, 7 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/tas2552.c b/sound/soc/codecs/tas2552.c index bd00c35116cd..52de6f7b4227 100644 --- a/sound/soc/codecs/tas2552.c +++ b/sound/soc/codecs/tas2552.c @@ -718,13 +718,6 @@ static int tas2552_probe(struct i2c_client *client, return ret; } - pm_runtime_set_active(&client->dev); - pm_runtime_set_autosuspend_delay(&client->dev, 1000); - pm_runtime_use_autosuspend(&client->dev); - pm_runtime_enable(&client->dev); - pm_runtime_mark_last_busy(&client->dev); - pm_runtime_put_sync_autosuspend(&client->dev); - dev_set_drvdata(&client->dev, data); ret = devm_snd_soc_register_component(&client->dev, @@ -733,6 +726,13 @@ static int tas2552_probe(struct i2c_client *client, if (ret < 0) dev_err(&client->dev, "Failed to register component: %d\n", ret); + pm_runtime_set_active(&client->dev); + pm_runtime_set_autosuspend_delay(&client->dev, 1000); + pm_runtime_use_autosuspend(&client->dev); + pm_runtime_enable(&client->dev); + pm_runtime_mark_last_busy(&client->dev); + pm_runtime_put_sync_autosuspend(&client->dev); + return ret; } -- 2.17.1

3 2

Implicit feedback on BOSS GT-1, the saga continues...
by Mike Oliphant 08 Apr '21

08 Apr '21

I had thought that the recent implicit feedback changes were fully working on the BOSS GT-1, but it turns out that I just hadn't tested well enough. Audio playback and capture works, but with periodic dropouts. I get the exact same behavior as I did with the quirk to completely disable implicit feedback. Without the implicit feedback, you get dropouts from clock drift - how bad probably varies from card to card. On mine it is every second or so. If I switch playback feedback for the GT-1 to generic by doing "IMPLICIT_FB_GENERIC_DEV(0x0582, 0x01d6)", I get the previous old behavior, which is that playback completely fails to start. With generic playback feedback, and using my previous patch to endpoint.c to avoid playback waiting on capture mentioned here: https://mailman.alsa-project.org/pipermail/alsa-devel/2020-January/161951.h… playback and capture work perfectly for me.

3 12

[PATCH] sound:ppc: fix spelling typo of values
by caizhichao 08 Apr '21

08 Apr '21

From: caizhichao <caizhichao(a)yulong.com> vaules -> values Signed-off-by: caizhichao <caizhichao(a)yulong.com> --- sound/ppc/snd_ps3_reg.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/ppc/snd_ps3_reg.h b/sound/ppc/snd_ps3_reg.h index 566a318..e2212b7 100644 --- a/sound/ppc/snd_ps3_reg.h +++ b/sound/ppc/snd_ps3_reg.h @@ -308,7 +308,7 @@ each interrupt in this register. Writing 1b to a field containing 1b clears field and de-asserts interrupt. Writing 0b to a field has no effect. -Field vaules are the following: +Field values are the following: 0 - Interrupt hasn't occurred. 1 - Interrupt has occurred. -- 1.9.1

2 1

[PATCH] ASoC: simple-card: fix possible uninitialized single_cpu local variable
by Krzysztof Kozlowski 07 Apr '21

07 Apr '21

The 'single_cpu' local variable is assigned by asoc_simple_parse_dai() and later used in a asoc_simple_canonicalize_cpu() call, assuming the entire function did not exit on errors. However the first function returns 0 if passed device_node is NULL, thus leaving the variable uninitialized and reporting success. Addresses-Coverity: Uninitialized scalar variable Fixes: 8f7f298a3337 ("ASoC: simple-card-utils: separate asoc_simple_card_parse_dai()") Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)canonical.com> --- sound/soc/generic/audio-graph-card.c | 2 +- sound/soc/generic/simple-card.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c index ee1d924d68e5..76036ea377a9 100644 --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -367,7 +367,7 @@ static int graph_dai_link_of(struct asoc_simple_priv *priv, struct device_node *top = dev->of_node; struct asoc_simple_dai *cpu_dai; struct asoc_simple_dai *codec_dai; - int ret, single_cpu; + int ret, single_cpu = 0; dev_dbg(dev, "link_of (%pOF)\n", cpu_ep); diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 41aa40765a8d..9a05f44fc3a9 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -249,7 +249,7 @@ static int simple_dai_link_of(struct asoc_simple_priv *priv, struct device_node *plat = NULL; char prop[128]; char *prefix = ""; - int ret, single_cpu; + int ret, single_cpu = 0; cpu = np; node = of_get_parent(np); -- 2.25.1

3 2

[PATCH] ASoC: meson: axg-fifo: add NO_PERIOD_WAKEUP support
by Jerome Brunet 07 Apr '21

07 Apr '21

On the AXG family, the fifo irq is not necessary for the HW to operate. It is just used to notify that a period has elapsed. If userpace does not care for these wakeups (such as pipewire), we are just wasting CPU cycles. Add support for NO_PERIOD_WAKEUP and disable irq when they are no needed. Signed-off-by: Jerome Brunet <jbrunet(a)baylibre.com> --- sound/soc/meson/axg-fifo.c | 11 ++++++----- 1 file changed, 6 insertions(+), 5 deletions(-) diff --git a/sound/soc/meson/axg-fifo.c b/sound/soc/meson/axg-fifo.c index b2e867113226..b9af2d513e09 100644 --- a/sound/soc/meson/axg-fifo.c +++ b/sound/soc/meson/axg-fifo.c @@ -27,8 +27,8 @@ static struct snd_pcm_hardware axg_fifo_hw = { SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_BLOCK_TRANSFER | - SNDRV_PCM_INFO_PAUSE), - + SNDRV_PCM_INFO_PAUSE | + SNDRV_PCM_INFO_NO_PERIOD_WAKEUP), .formats = AXG_FIFO_FORMATS, .rate_min = 5512, .rate_max = 192000, @@ -113,7 +113,7 @@ int axg_fifo_pcm_hw_params(struct snd_soc_component *component, { struct snd_pcm_runtime *runtime = ss->runtime; struct axg_fifo *fifo = axg_fifo_data(ss); - unsigned int burst_num, period, threshold; + unsigned int burst_num, period, threshold, irq_en; dma_addr_t end_ptr; period = params_period_bytes(params); @@ -142,10 +142,11 @@ int axg_fifo_pcm_hw_params(struct snd_soc_component *component, regmap_field_write(fifo->field_threshold, threshold ? threshold - 1 : 0); - /* Enable block count irq */ + /* Enable irq if necessary */ + irq_en = runtime->no_period_wakeup ? 0 : FIFO_INT_COUNT_REPEAT; regmap_update_bits(fifo->map, FIFO_CTRL0, CTRL0_INT_EN(FIFO_INT_COUNT_REPEAT), - CTRL0_INT_EN(FIFO_INT_COUNT_REPEAT)); + CTRL0_INT_EN(irq_en)); return 0; } -- 2.30.2

2 1

[PATCH 0/4] Adds SPI support
by Jiri Prchal 07 Apr '21

07 Apr '21

Moved I2C stuff to separated file, shered thingk to header file. Added SPI in separeted file. Jiri Prchal (4): ASoC: codecs: tlv320aic3x: move model definitions ASoC: codecs: tlv320aic3x: rename probe function ASoC: codecs: tlv320aic3x: move I2C to separated file ASoC: codecs: tlv320aic3x: add SPI support sound/soc/codecs/Kconfig | 16 ++++- sound/soc/codecs/Makefile | 4 ++ sound/soc/codecs/tlv320aic3x-i2c.c | 70 +++++++++++++++++++ sound/soc/codecs/tlv320aic3x-spi.c | 76 ++++++++++++++++++++ sound/soc/codecs/tlv320aic3x.c | 108 ++++++++--------------------- sound/soc/codecs/tlv320aic3x.h | 12 ++++ 6 files changed, 204 insertions(+), 82 deletions(-) create mode 100644 sound/soc/codecs/tlv320aic3x-i2c.c create mode 100644 sound/soc/codecs/tlv320aic3x-spi.c -- 2.25.1

3 8

[PATCH -next] ASoC: codecs: lpass-va-macro: constify static struct snd_soc_dai_ops
by Ye Bin 07 Apr '21

07 Apr '21

The snd_soc_dai_ops structures is only stored in the ops field of a snd_soc_dai_driver structure, so make the snd_soc_dai_ops structure const to allow the compiler to put it in read-only memory. Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: Ye Bin <yebin10(a)huawei.com> --- sound/soc/codecs/lpass-va-macro.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/lpass-va-macro.c b/sound/soc/codecs/lpass-va-macro.c index 3d6976a3d9e4..5294c57b2cd4 100644 --- a/sound/soc/codecs/lpass-va-macro.c +++ b/sound/soc/codecs/lpass-va-macro.c @@ -894,7 +894,7 @@ static int va_macro_digital_mute(struct snd_soc_dai *dai, int mute, int stream) return 0; } -static struct snd_soc_dai_ops va_macro_dai_ops = { +static const struct snd_soc_dai_ops va_macro_dai_ops = { .hw_params = va_macro_hw_params, .get_channel_map = va_macro_get_channel_map, .mute_stream = va_macro_digital_mute,

2 1

[PATCH] ASoC: meson: axg-frddr: set fifo depth according to the period
by Jerome Brunet 07 Apr '21

07 Apr '21

When the period is small, using all the FRDDR fifo depth increases the latency of the playback because the following device won't start pulling data until the fifo reaches the depth set. We can adjust this depth so trim it down for small periods. Signed-off-by: Jerome Brunet <jbrunet(a)baylibre.com> --- sound/soc/meson/axg-frddr.c | 26 ++++++++++++++++++++------ 1 file changed, 20 insertions(+), 6 deletions(-) diff --git a/sound/soc/meson/axg-frddr.c b/sound/soc/meson/axg-frddr.c index c3ae8ac30745..8ed114de0bdf 100644 --- a/sound/soc/meson/axg-frddr.c +++ b/sound/soc/meson/axg-frddr.c @@ -11,6 +11,7 @@ #include <linux/regmap.h> #include <linux/module.h> #include <linux/of_platform.h> +#include <sound/pcm_params.h> #include <sound/soc.h> #include <sound/soc-dai.h> @@ -46,11 +47,28 @@ static int g12a_frddr_dai_prepare(struct snd_pcm_substream *substream, return 0; } +static int axg_frddr_dai_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct axg_fifo *fifo = snd_soc_dai_get_drvdata(dai); + unsigned int period, depth, val; + + period = params_period_bytes(params); + + /* Trim the FIFO depth if the period is small to improve latency */ + depth = min(period, fifo->depth); + val = (depth / AXG_FIFO_BURST) - 1; + regmap_update_bits(fifo->map, FIFO_CTRL1, CTRL1_FRDDR_DEPTH_MASK, + CTRL1_FRDDR_DEPTH(val)); + + return 0; +} + static int axg_frddr_dai_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct axg_fifo *fifo = snd_soc_dai_get_drvdata(dai); - unsigned int val; int ret; /* Enable pclk to access registers and clock the fifo ip */ @@ -61,11 +79,6 @@ static int axg_frddr_dai_startup(struct snd_pcm_substream *substream, /* Apply single buffer mode to the interface */ regmap_update_bits(fifo->map, FIFO_CTRL0, CTRL0_FRDDR_PP_MODE, 0); - /* Use all fifo depth */ - val = (fifo->depth / AXG_FIFO_BURST) - 1; - regmap_update_bits(fifo->map, FIFO_CTRL1, CTRL1_FRDDR_DEPTH_MASK, - CTRL1_FRDDR_DEPTH(val)); - return 0; } @@ -84,6 +97,7 @@ static int axg_frddr_pcm_new(struct snd_soc_pcm_runtime *rtd, } static const struct snd_soc_dai_ops axg_frddr_ops = { + .hw_params = axg_frddr_dai_hw_params, .startup = axg_frddr_dai_startup, .shutdown = axg_frddr_dai_shutdown, }; -- 2.30.2

2 1

[PATCH v5 2/4] ASoC: Add Rockchip rk817 audio CODEC support
by Chris Morgan 07 Apr '21

07 Apr '21

Add support for the Rockchip rk817 audio codec integrated into the rk817 PMIC. This is based on the sources provided by Rockchip from their BSP kernel. Changes from v4: Switched to using parent regmap instead of private regmap. Removed of_compatible strings. Removed register definitions and moved them to the MFD driver. Signed-off-by: Chris Morgan <macromorgan(a)hotmail.com> --- sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/rk817_codec.c | 560 +++++++++++++++++++++++++++++++++ 3 files changed, 566 insertions(+) create mode 100644 sound/soc/codecs/rk817_codec.c diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index e4cf14e66a51..d835145ad65e 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -155,6 +155,7 @@ config SND_SOC_ALL_CODECS imply SND_SOC_PCM512x_I2C imply SND_SOC_PCM512x_SPI imply SND_SOC_RK3328 + imply SND_SOC_RK817 imply SND_SOC_RT274 imply SND_SOC_RT286 imply SND_SOC_RT298 @@ -1059,6 +1060,11 @@ config SND_SOC_RK3328 tristate "Rockchip RK3328 audio CODEC" select REGMAP_MMIO +config SND_SOC_RK817 + tristate "Rockchip RK817 audio CODEC" + depends on MFD_RK808 + select REGMAP_I2C + config SND_SOC_RL6231 tristate default y if SND_SOC_RT5514=y diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 81357dc62ea0..4e5e9b28775c 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -165,6 +165,7 @@ snd-soc-pcm512x-objs := pcm512x.o snd-soc-pcm512x-i2c-objs := pcm512x-i2c.o snd-soc-pcm512x-spi-objs := pcm512x-spi.o snd-soc-rk3328-objs := rk3328_codec.o +snd-soc-rk817-objs := rk817_codec.o snd-soc-rl6231-objs := rl6231.o snd-soc-rl6347a-objs := rl6347a.o snd-soc-rt1011-objs := rt1011.o @@ -479,6 +480,7 @@ obj-$(CONFIG_SND_SOC_PCM512x) += snd-soc-pcm512x.o obj-$(CONFIG_SND_SOC_PCM512x_I2C) += snd-soc-pcm512x-i2c.o obj-$(CONFIG_SND_SOC_PCM512x_SPI) += snd-soc-pcm512x-spi.o obj-$(CONFIG_SND_SOC_RK3328) += snd-soc-rk3328.o +obj-$(CONFIG_SND_SOC_RK817) += snd-soc-rk817.o obj-$(CONFIG_SND_SOC_RL6231) += snd-soc-rl6231.o obj-$(CONFIG_SND_SOC_RL6347A) += snd-soc-rl6347a.o obj-$(CONFIG_SND_SOC_RT1011) += snd-soc-rt1011.o diff --git a/sound/soc/codecs/rk817_codec.c b/sound/soc/codecs/rk817_codec.c new file mode 100644 index 000000000000..0ad097db04bd --- /dev/null +++ b/sound/soc/codecs/rk817_codec.c @@ -0,0 +1,560 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// rk817 ALSA SoC Audio driver +// +// Copyright (c) 2018, Fuzhou Rockchip Electronics Co., Ltd All rights reserved. + +#include <linux/clk.h> +#include <linux/device.h> +#include <linux/delay.h> +#include <linux/mfd/rk808.h> +#include <linux/module.h> +#include <linux/of.h> +#include <linux/of_gpio.h> +#include <linux/platform_device.h> +#include <linux/regmap.h> +#include <sound/core.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/tlv.h> + +struct rk817_codec_priv { + struct snd_soc_component *component; + struct rk808 *rk808; + struct clk *mclk; + unsigned int stereo_sysclk; + bool mic_in_differential; +}; + +/* + * This sets the codec up with the values defined in the default implementation including the APLL + * from the Rockchip vendor kernel. I do not know if these values are universal despite differing + * from the default values defined above and taken from the datasheet, or implementation specific. + * I don't have another implementation to compare from the Rockchip sources. Hard-coding for now. + * Additionally, I do not know according to the documentation the units accepted for the clock + * values, so for the moment those are left unvalidated. + */ + +static int rk817_init(struct snd_soc_component *component) +{ + struct rk817_codec_priv *rk817 = snd_soc_component_get_drvdata(component); + + snd_soc_component_write(component, RK817_CODEC_DDAC_POPD_DACST, 0x02); + snd_soc_component_write(component, RK817_CODEC_DDAC_SR_LMT0, 0x02); + snd_soc_component_write(component, RK817_CODEC_DADC_SR_ACL0, 0x02); + snd_soc_component_write(component, RK817_CODEC_DTOP_VUCTIME, 0xf4); + if (rk817->mic_in_differential) { + snd_soc_component_update_bits(component, RK817_CODEC_AMIC_CFG0, MIC_DIFF_MASK, + MIC_DIFF_EN); + }; + return 0; +} + +static int rk817_set_component_pll(struct snd_soc_component *component, + int pll_id, int source, unsigned int freq_in, + unsigned int freq_out) +{ + /* Set resistor value and charge pump current for PLL. */ + snd_soc_component_write(component, RK817_CODEC_APLL_CFG1, 0x58); + /* Set the PLL feedback clock divide value (values not documented). */ + snd_soc_component_write(component, RK817_CODEC_APLL_CFG2, 0x2d); + /* Set the PLL pre-divide value (values not documented). */ + snd_soc_component_write(component, RK817_CODEC_APLL_CFG3, 0x0c); + /* Set the PLL VCO output clock divide and PLL divided ratio of PLL High Clk (values not + * documented). + */ + snd_soc_component_write(component, RK817_CODEC_APLL_CFG4, 0xa5); + + return 0; +} + +/* + * DDAC/DADC L/R volume setting + * 0db~-95db, 0.375db/step, for example: + * 0x00: 0dB + * 0xff: -95dB + */ + +static const DECLARE_TLV_DB_MINMAX(rk817_vol_tlv, -9500, 0); + +/* + * PGA GAIN L/R volume setting + * 27db~-18db, 3db/step, for example: + * 0x0: -18dB + * 0xf: 27dB + */ + +static const DECLARE_TLV_DB_MINMAX(rk817_gain_tlv, -1800, 2700); + +static const struct snd_kcontrol_new rk817_volume_controls[] = { + SOC_DOUBLE_R_RANGE_TLV("Master Playback Volume", RK817_CODEC_DDAC_VOLL, + RK817_CODEC_DDAC_VOLR, 0, 0x00, 0xff, 1, rk817_vol_tlv), + SOC_DOUBLE_R_RANGE_TLV("Master Capture Volume", RK817_CODEC_DADC_VOLL, + RK817_CODEC_DADC_VOLR, 0, 0x00, 0xff, 1, rk817_vol_tlv), + SOC_DOUBLE_TLV("Mic Capture Gain", RK817_CODEC_DMIC_PGA_GAIN, 4, 0, 0xf, 0, + rk817_gain_tlv), +}; + +/* Since the speaker output and L headphone pin are internally the same, make audio path mutually + * exclusive with a mux. + */ + +static const char *dac_mux_text[] = { + "HP", + "SPK", +}; + +static SOC_ENUM_SINGLE_VIRT_DECL(dac_enum, dac_mux_text); + +static const struct snd_kcontrol_new dac_mux = + SOC_DAPM_ENUM("Playback Mux", dac_enum); + +static const struct snd_soc_dapm_widget rk817_dapm_widgets[] = { + + /* capture/playback common */ + SND_SOC_DAPM_SUPPLY("LDO Regulator", RK817_CODEC_AREF_RTCFG1, 6, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("IBIAS Block", RK817_CODEC_AREF_RTCFG1, 2, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("VAvg Buffer", RK817_CODEC_AREF_RTCFG1, 1, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("PLL Power", RK817_CODEC_APLL_CFG5, 0, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("I2S TX1 Transfer Start", RK817_CODEC_DI2S_RXCMD_TSD, 5, 0, NULL, 0), + + /* capture path common */ + SND_SOC_DAPM_SUPPLY("ADC Clock", RK817_CODEC_DTOP_DIGEN_CLKE, 7, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("I2S TX Clock", RK817_CODEC_DTOP_DIGEN_CLKE, 6, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADC Channel Enable", RK817_CODEC_DTOP_DIGEN_CLKE, 5, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("I2S TX Channel Enable", RK817_CODEC_DTOP_DIGEN_CLKE, 4, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("MIC Power On", RK817_CODEC_AMIC_CFG0, 6, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("I2S TX3 Transfer Start", RK817_CODEC_DI2S_TXCR3_TXCMD, 7, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("I2S TX3 Right Justified", RK817_CODEC_DI2S_TXCR3_TXCMD, 3, 0, NULL, 0), + + /* capture path L */ + SND_SOC_DAPM_ADC("ADC L", "Capture", RK817_CODEC_AADC_CFG0, 7, 1), + SND_SOC_DAPM_SUPPLY("PGA L Power On", RK817_CODEC_AMIC_CFG0, 5, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("Mic Boost L1", RK817_CODEC_AMIC_CFG0, 3, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("Mic Boost L2", RK817_CODEC_AMIC_CFG0, 2, 0, NULL, 0), + + /* capture path R */ + SND_SOC_DAPM_ADC("ADC R", "Capture", RK817_CODEC_AADC_CFG0, 6, 1), + SND_SOC_DAPM_SUPPLY("PGA R Power On", RK817_CODEC_AMIC_CFG0, 4, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("Mic Boost R1", RK817_CODEC_AMIC_CFG0, 3, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("Mic Boost R2", RK817_CODEC_AMIC_CFG0, 3, 0, NULL, 0), + + /* playback path common */ + SND_SOC_DAPM_SUPPLY("DAC Clock", RK817_CODEC_DTOP_DIGEN_CLKE, 3, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("I2S RX Clock", RK817_CODEC_DTOP_DIGEN_CLKE, 2, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DAC Channel Enable", RK817_CODEC_DTOP_DIGEN_CLKE, 1, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("I2S RX Channel Enable", RK817_CODEC_DTOP_DIGEN_CLKE, 0, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DAC Bias", RK817_CODEC_ADAC_CFG1, 3, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("DAC Mute Off", RK817_CODEC_DDAC_MUTE_MIXCTL, 0, 1, NULL, 0), + + /* playback path speaker */ + SND_SOC_DAPM_SUPPLY("Class D Mode", RK817_CODEC_DDAC_MUTE_MIXCTL, 4, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("High Pass Filter", RK817_CODEC_DDAC_MUTE_MIXCTL, 7, 0, NULL, 0), + SND_SOC_DAPM_DAC("SPK DAC", "Playback", RK817_CODEC_ADAC_CFG1, 2, 1), + SND_SOC_DAPM_SUPPLY("Enable Class D", RK817_CODEC_ACLASSD_CFG1, 7, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("Disable Class D Mute Ramp", RK817_CODEC_ACLASSD_CFG1, 6, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("Class D Mute Rate 1", RK817_CODEC_ACLASSD_CFG1, 3, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("Class D Mute Rate 2", RK817_CODEC_ACLASSD_CFG1, 2, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("Class D OCPP 2", RK817_CODEC_ACLASSD_CFG2, 5, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("Class D OCPP 3", RK817_CODEC_ACLASSD_CFG2, 4, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("Class D OCPN 2", RK817_CODEC_ACLASSD_CFG2, 1, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("Class D OCPN 3", RK817_CODEC_ACLASSD_CFG2, 0, 0, NULL, 0), + + /* playback path headphones */ + SND_SOC_DAPM_SUPPLY("Headphone Charge Pump", RK817_CODEC_AHP_CP, 4, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("Headphone CP Discharge LDO", RK817_CODEC_AHP_CP, 3, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("Headphone OStage", RK817_CODEC_AHP_CFG0, 6, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("Headphone Pre Amp", RK817_CODEC_AHP_CFG0, 5, 1, NULL, 0), + SND_SOC_DAPM_DAC("DAC L", "Playback", RK817_CODEC_ADAC_CFG1, 1, 1), + SND_SOC_DAPM_DAC("DAC R", "Playback", RK817_CODEC_ADAC_CFG1, 0, 1), + + /* Mux for input/output path selection */ + SND_SOC_DAPM_MUX("Playback Mux", SND_SOC_NOPM, 1, 0, &dac_mux), + + /* Pins for Simple Card Bindings */ + SND_SOC_DAPM_INPUT("MICL"), + SND_SOC_DAPM_INPUT("MICR"), + SND_SOC_DAPM_OUTPUT("HPOL"), + SND_SOC_DAPM_OUTPUT("HPOR"), + SND_SOC_DAPM_OUTPUT("SPKO"), +}; + +static const struct snd_soc_dapm_route rk817_dapm_routes[] = { + + /* capture path */ + /* left mic */ + {"ADC L", NULL, "LDO Regulator"}, + {"ADC L", NULL, "IBIAS Block"}, + {"ADC L", NULL, "VAvg Buffer"}, + {"ADC L", NULL, "PLL Power"}, + {"ADC L", NULL, "ADC Clock"}, + {"ADC L", NULL, "I2S TX Clock"}, + {"ADC L", NULL, "ADC Channel Enable"}, + {"ADC L", NULL, "I2S TX Channel Enable"}, + {"ADC L", NULL, "I2S TX1 Transfer Start"}, + {"MICL", NULL, "MIC Power On"}, + {"MICL", NULL, "PGA L Power On"}, + {"MICL", NULL, "Mic Boost L1"}, + {"MICL", NULL, "Mic Boost L2"}, + {"MICL", NULL, "I2S TX3 Transfer Start"}, + {"MICL", NULL, "I2S TX3 Right Justified"}, + {"ADC L", NULL, "MICL"}, + + /* right mic */ + {"ADC R", NULL, "LDO Regulator"}, + {"ADC R", NULL, "IBIAS Block"}, + {"ADC R", NULL, "VAvg Buffer"}, + {"ADC R", NULL, "PLL Power"}, + {"ADC R", NULL, "ADC Clock"}, + {"ADC R", NULL, "I2S TX Clock"}, + {"ADC R", NULL, "ADC Channel Enable"}, + {"ADC R", NULL, "I2S TX Channel Enable"}, + {"ADC R", NULL, "I2S TX1 Transfer Start"}, + {"MICR", NULL, "MIC Power On"}, + {"MICR", NULL, "PGA R Power On"}, + {"MICR", NULL, "Mic Boost R1"}, + {"MICR", NULL, "Mic Boost R2"}, + {"MICR", NULL, "I2S TX3 Transfer Start"}, + {"MICR", NULL, "I2S TX3 Right Justified"}, + {"ADC R", NULL, "MICR"}, + + /* playback path */ + /* speaker path */ + {"SPK DAC", NULL, "LDO Regulator"}, + {"SPK DAC", NULL, "IBIAS Block"}, + {"SPK DAC", NULL, "VAvg Buffer"}, + {"SPK DAC", NULL, "PLL Power"}, + {"SPK DAC", NULL, "I2S TX1 Transfer Start"}, + {"SPK DAC", NULL, "DAC Clock"}, + {"SPK DAC", NULL, "I2S RX Clock"}, + {"SPK DAC", NULL, "DAC Channel Enable"}, + {"SPK DAC", NULL, "I2S RX Channel Enable"}, + {"SPK DAC", NULL, "Class D Mode"}, + {"SPK DAC", NULL, "DAC Bias"}, + {"SPK DAC", NULL, "DAC Mute Off"}, + {"SPK DAC", NULL, "Enable Class D"}, + {"SPK DAC", NULL, "Disable Class D Mute Ramp"}, + {"SPK DAC", NULL, "Class D Mute Rate 1"}, + {"SPK DAC", NULL, "Class D Mute Rate 2"}, + {"SPK DAC", NULL, "Class D OCPP 2"}, + {"SPK DAC", NULL, "Class D OCPP 3"}, + {"SPK DAC", NULL, "Class D OCPN 2"}, + {"SPK DAC", NULL, "Class D OCPN 3"}, + {"SPK DAC", NULL, "High Pass Filter"}, + + /* headphone path L */ + {"DAC L", NULL, "LDO Regulator"}, + {"DAC L", NULL, "IBIAS Block"}, + {"DAC L", NULL, "VAvg Buffer"}, + {"DAC L", NULL, "PLL Power"}, + {"DAC L", NULL, "I2S TX1 Transfer Start"}, + {"DAC L", NULL, "DAC Clock"}, + {"DAC L", NULL, "I2S RX Clock"}, + {"DAC L", NULL, "DAC Channel Enable"}, + {"DAC L", NULL, "I2S RX Channel Enable"}, + {"DAC L", NULL, "DAC Bias"}, + {"DAC L", NULL, "DAC Mute Off"}, + {"DAC L", NULL, "Headphone Charge Pump"}, + {"DAC L", NULL, "Headphone CP Discharge LDO"}, + {"DAC L", NULL, "Headphone OStage"}, + {"DAC L", NULL, "Headphone Pre Amp"}, + + /* headphone path R */ + {"DAC R", NULL, "LDO Regulator"}, + {"DAC R", NULL, "IBIAS Block"}, + {"DAC R", NULL, "VAvg Buffer"}, + {"DAC R", NULL, "PLL Power"}, + {"DAC R", NULL, "I2S TX1 Transfer Start"}, + {"DAC R", NULL, "DAC Clock"}, + {"DAC R", NULL, "I2S RX Clock"}, + {"DAC R", NULL, "DAC Channel Enable"}, + {"DAC R", NULL, "I2S RX Channel Enable"}, + {"DAC R", NULL, "DAC Bias"}, + {"DAC R", NULL, "DAC Mute Off"}, + {"DAC R", NULL, "Headphone Charge Pump"}, + {"DAC R", NULL, "Headphone CP Discharge LDO"}, + {"DAC R", NULL, "Headphone OStage"}, + {"DAC R", NULL, "Headphone Pre Amp"}, + + /* mux path for output selection */ + {"Playback Mux", "HP", "DAC L"}, + {"Playback Mux", "HP", "DAC R"}, + {"Playback Mux", "SPK", "SPK DAC"}, + {"SPKO", NULL, "Playback Mux"}, + {"HPOL", NULL, "Playback Mux"}, + {"HPOR", NULL, "Playback Mux"}, +}; + +static int rk817_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_component *component = codec_dai->component; + struct rk817_codec_priv *rk817 = snd_soc_component_get_drvdata(component); + + rk817->stereo_sysclk = freq; + + return 0; +} + +static int rk817_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_component *component = codec_dai->component; + unsigned int i2s_mst = 0; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + i2s_mst |= RK817_I2S_MODE_SLV; + break; + case SND_SOC_DAIFMT_CBM_CFM: + i2s_mst |= RK817_I2S_MODE_MST; + break; + default: + dev_err(component->dev, "%s : set master mask failed!\n", __func__); + return -EINVAL; + } + + snd_soc_component_update_bits(component, RK817_CODEC_DI2S_CKM, + RK817_I2S_MODE_MASK, i2s_mst); + + return 0; +} + +static int rk817_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + snd_soc_component_write(component, RK817_CODEC_DI2S_RXCR2, + VDW_RX_16BITS); + snd_soc_component_write(component, RK817_CODEC_DI2S_TXCR2, + VDW_TX_16BITS); + break; + case SNDRV_PCM_FORMAT_S24_LE: + case SNDRV_PCM_FORMAT_S32_LE: + snd_soc_component_write(component, RK817_CODEC_DI2S_RXCR2, + VDW_RX_24BITS); + snd_soc_component_write(component, RK817_CODEC_DI2S_TXCR2, + VDW_TX_24BITS); + break; + default: + return -EINVAL; + } + + return 0; +} + +static int rk817_digital_mute(struct snd_soc_dai *dai, int mute, int stream) +{ + struct snd_soc_component *component = dai->component; + + if (mute) + snd_soc_component_update_bits(component, + RK817_CODEC_DDAC_MUTE_MIXCTL, + DACMT_MASK, DACMT_ENABLE); + else + snd_soc_component_update_bits(component, + RK817_CODEC_DDAC_MUTE_MIXCTL, + DACMT_MASK, DACMT_DISABLE); + + return 0; +} + +#define RK817_PLAYBACK_RATES (SNDRV_PCM_RATE_8000 |\ + SNDRV_PCM_RATE_16000 | \ + SNDRV_PCM_RATE_32000 | \ + SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000 | \ + SNDRV_PCM_RATE_96000) + +#define RK817_CAPTURE_RATES (SNDRV_PCM_RATE_8000 |\ + SNDRV_PCM_RATE_16000 | \ + SNDRV_PCM_RATE_32000 | \ + SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000 | \ + SNDRV_PCM_RATE_96000) + +#define RK817_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\ + SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE |\ + SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_ops rk817_dai_ops = { + .hw_params = rk817_hw_params, + .set_fmt = rk817_set_dai_fmt, + .set_sysclk = rk817_set_dai_sysclk, + .mute_stream = rk817_digital_mute, + .no_capture_mute = 1, +}; + +static struct snd_soc_dai_driver rk817_dai[] = { + { + .name = "rk817-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 8, + .rates = RK817_PLAYBACK_RATES, + .formats = RK817_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = RK817_CAPTURE_RATES, + .formats = RK817_FORMATS, + }, + .ops = &rk817_dai_ops, + }, +}; + +static int rk817_probe(struct snd_soc_component *component) +{ + struct rk817_codec_priv *rk817 = snd_soc_component_get_drvdata(component); + struct rk808 *rk808 = dev_get_drvdata(component->dev->parent); + int ret; + + snd_soc_component_init_regmap(component, rk808->regmap); + rk817->component = component; + + ret = snd_soc_component_write(component, RK817_CODEC_DTOP_LPT_SRST, 0x40); + + rk817_init(component); + + /* setting initial pll values so that we can continue to leverage simple-audio-card. + * The values aren't important since no parameters are used. + */ + + snd_soc_component_set_pll(component, 0, 0, 0, 0); + + return 0; +} + +static void rk817_remove(struct snd_soc_component *component) +{ + snd_soc_component_exit_regmap(component); +} + +static const struct snd_soc_component_driver soc_codec_dev_rk817 = { + .probe = rk817_probe, + .remove = rk817_remove, + .idle_bias_on = 1, + .use_pmdown_time = 1, + .endianness = 1, + .non_legacy_dai_naming = 1, + .controls = rk817_volume_controls, + .num_controls = ARRAY_SIZE(rk817_volume_controls), + .dapm_routes = rk817_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(rk817_dapm_routes), + .dapm_widgets = rk817_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(rk817_dapm_widgets), + .set_pll = rk817_set_component_pll, +}; + +static int rk817_codec_parse_dt_property(struct device *dev, + struct rk817_codec_priv *rk817) +{ + struct device_node *node = dev->parent->of_node; + + if (!node) { + dev_err(dev, "%s() dev->parent->of_node is NULL\n", + __func__); + return -ENODEV; + } + + node = of_get_child_by_name(dev->parent->of_node, "codec"); + if (!node) { + dev_err(dev, "%s() Can not get child: codec\n", + __func__); + return -ENODEV; + } + + rk817->mic_in_differential = + of_property_read_bool(node, "mic-in-differential"); + + return 0; +} + +static int rk817_platform_probe(struct platform_device *pdev) +{ + struct rk808 *rk808 = dev_get_drvdata(pdev->dev.parent); + struct rk817_codec_priv *rk817_codec_data; + int ret; + + if (!rk808) { + dev_err(&pdev->dev, "%s : rk817 is NULL\n", __func__); + return -EINVAL; + } + + rk817_codec_data = devm_kzalloc(&pdev->dev, + sizeof(struct rk817_codec_priv), + GFP_KERNEL); + if (!rk817_codec_data) + return -ENOMEM; + + platform_set_drvdata(pdev, rk817_codec_data); + + rk817_codec_data->rk808 = rk808; + + ret = rk817_codec_parse_dt_property(&pdev->dev, rk817_codec_data); + if (ret < 0) { + dev_err(&pdev->dev, "%s() parse device tree property error %d\n", + __func__, ret); + goto err_; + } + + rk817_codec_data->mclk = clk_get(pdev->dev.parent, "mclk"); + if (IS_ERR(rk817_codec_data->mclk)) { + dev_err(&pdev->dev, "Unable to get mclk\n"); + ret = -ENXIO; + goto err_; + } + + ret = clk_prepare_enable(rk817_codec_data->mclk); + if (ret < 0) { + dev_err(&pdev->dev, "%s() clock prepare error %d\n", + __func__, ret); + goto err_; + } + + ret = devm_snd_soc_register_component(&pdev->dev, &soc_codec_dev_rk817, + rk817_dai, ARRAY_SIZE(rk817_dai)); + if (ret < 0) { + dev_err(&pdev->dev, "%s() register codec error %d\n", + __func__, ret); + goto err_; + } + + return 0; +err_: + + return ret; +} + +static int rk817_platform_remove(struct platform_device *pdev) +{ + struct rk817_codec_priv *rk817 = platform_get_drvdata(pdev); + + clk_disable_unprepare(rk817->mclk); + + return 0; +} + +MODULE_DEVICE_TABLE(of, rk817_codec_dt_ids); + +static struct platform_driver rk817_codec_driver = { + .driver = { + .name = "rk817-codec", + }, + .probe = rk817_platform_probe, + .remove = rk817_platform_remove, +}; + +module_platform_driver(rk817_codec_driver); + +MODULE_DESCRIPTION("ASoC RK817 codec driver"); +MODULE_AUTHOR("binyuan <kevan.lan(a)rock-chips.com>"); +MODULE_LICENSE("GPL v2"); -- 2.25.1

2 3

[pull requests] No SPDIF for Plantronics 3220, Dell AE515
by James Ettle 07 Apr '21

07 Apr '21

Hello, The Plantronics Blackwire 3220 Series headset and the Dell Professional Sound Bar AE515 are both USB-audio devices that show up with a digital S/PDIF output that physically doesn't exist. This can be fixed by adding them to the USB-Audio.pcm.iec958_device list in USB-Audio.conf, with the following names: "DELL PROFESSIONAL SOUND BAR AE5" "Plantronics Blackwire 3220 Seri" Patches for the respective devices are in the following alsa-lib pull requests: https://github.com/alsa-project/alsa-lib/pull/124 https://github.com/alsa-project/alsa-lib/pull/126 Many thanks, James.

1 0

[PATCH] ALSA: usb-audio: Add MIDI quirk for Vox ToneLab EX
by Takashi Iwai 07 Apr '21

07 Apr '21

ToneLab EX guitar pedal device requires the same quirk like ToneLab ST for supporting the MIDI. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=212593 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/quirks-table.h | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 9716a9f7c095..8a8fe2b980a1 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -2376,6 +2376,16 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, +{ + USB_DEVICE_VENDOR_SPEC(0x0944, 0x0204), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + .vendor_name = "KORG, Inc.", + /* .product_name = "ToneLab EX", */ + .ifnum = 3, + .type = QUIRK_MIDI_STANDARD_INTERFACE, + } +}, + /* AKAI devices */ { USB_DEVICE(0x09e8, 0x0062), -- 2.26.2

1 0

[GIT PULL] ASoC fixes for v5.12-rc6
by Mark Brown 07 Apr '21

07 Apr '21

The following changes since commit f9dc51cc6676e0360dfcb1836b3d426a3ccf3093: Merge series "Do not handle MCLK device clock in simple-card-utils" from Sameer Pujar <spujar(a)nvidia.com>: (2021-03-16 17:55:37 +0000) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v5.12-rc6 for you to fetch changes up to e7a48c710defa0e0fef54d42b7d9e4ab596e2761: ASoC: fsl_esai: Fix TDM slot setup for I2S mode (2021-04-02 16:24:16 +0100) ---------------------------------------------------------------- ASoC: Fixes for v5.12 A fairly small batch of driver specific fixes, mainly for various x86 systems with the biggest set being fixes to power down DSPs properly on x86 SOF systems. ---------------------------------------------------------------- Alexander Shiyan (1): ASoC: fsl_esai: Fix TDM slot setup for I2S mode Bastian Germann (1): ASoC: sunxi: sun4i-codec: fill ASoC card owner Guennadi Liakhovetski (1): ASoC: SOF: Intel: HDA: fix core status verification Hans de Goede (2): ASoC: intel: atom: Stop advertising non working S24LE support ASoC: intel: atom: Remove 44100 sample-rate from the media and deep-buffer DAI descriptions Libin Yang (5): ASoC: SOF: Intel: TGL: fix EHL ops ASoC: SOF: Intel: TGL: set shutdown callback to hda_dsp_shutdown ASoC: SOF: Intel: ICL: set shutdown callback to hda_dsp_shutdown ASoC: SOF: Intel: CNL: set shutdown callback to hda_dsp_shutdown ASoC: SOF: Intel: APL: set shutdown callback to hda_dsp_shutdown Mark Brown (1): Merge series "ASoC: SOF: Intel: shutdown and core handling corrections" from Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com>: Pierre-Louis Bossart (2): ASoC: SOF: core: harden shutdown helper ASoC: SOF: Intel: move ELH chip info Ryan Lee (2): ASoC: max98373: Changed amp shutdown register as volatile ASoC: max98373: Added 30ms turn on/off time delay Shengjiu Wang (1): ASoC: wm8960: Fix wrong bclk and lrclk with pll enabled for some chips Srinivas Kandagatla (2): ASoC: codecs: lpass-tx-macro: set npl clock rate correctly ASoC: codecs: lpass-rx-macro: set npl clock rate correctly kernel test robot (1): ASoC: cygnus: fix for_each_child.cocci warnings sound/soc/bcm/cygnus-ssp.c | 4 +++- sound/soc/codecs/lpass-rx-macro.c | 2 +- sound/soc/codecs/lpass-tx-macro.c | 2 +- sound/soc/codecs/max98373-i2c.c | 1 + sound/soc/codecs/max98373-sdw.c | 1 + sound/soc/codecs/max98373.c | 2 ++ sound/soc/codecs/wm8960.c | 8 +++++++- sound/soc/fsl/fsl_esai.c | 8 +++++--- sound/soc/intel/atom/sst-mfld-platform-pcm.c | 12 ++++++------ sound/soc/sof/core.c | 8 +++++++- sound/soc/sof/intel/apl.c | 3 ++- sound/soc/sof/intel/cnl.c | 19 ++----------------- sound/soc/sof/intel/hda-dsp.c | 21 +++++++++++++++++---- sound/soc/sof/intel/hda.h | 1 + sound/soc/sof/intel/icl.c | 3 ++- sound/soc/sof/intel/pci-tgl.c | 2 +- sound/soc/sof/intel/tgl.c | 18 +++++++++++++++++- sound/soc/sunxi/sun4i-codec.c | 5 +++++ 18 files changed, 81 insertions(+), 39 deletions(-)

2 1

[PATCH 32/32] pinctrl: update pin-control.rst references
by Mauro Carvalho Chehab 07 Apr '21

07 Apr '21

Changeset 5513b411ea5b ("Documentation: rename pinctl to pin-control") renamed: Documentation/driver-api/pinctl.rst to: Documentation/driver-api/pin-control.rst. Update the cross-references accordingly. Fixes: 5513b411ea5b ("Documentation: rename pinctl to pin-control") Signed-off-by: Mauro Carvalho Chehab <mchehab+huawei(a)kernel.org> --- include/linux/device.h | 2 +- include/linux/mfd/madera/pdata.h | 2 +- include/linux/pinctrl/pinconf-generic.h | 2 +- 3 files changed, 3 insertions(+), 3 deletions(-) diff --git a/include/linux/device.h b/include/linux/device.h index 38a2071cf776..d1183cfdc8fb 100644 --- a/include/linux/device.h +++ b/include/linux/device.h @@ -399,7 +399,7 @@ struct dev_links_info { * along with subsystem-level and driver-level callbacks. * @em_pd: device's energy model performance domain * @pins: For device pin management. - * See Documentation/driver-api/pinctl.rst for details. + * See Documentation/driver-api/pin-control.rst for details. * @msi_list: Hosts MSI descriptors * @msi_domain: The generic MSI domain this device is using. * @numa_node: NUMA node this device is close to. diff --git a/include/linux/mfd/madera/pdata.h b/include/linux/mfd/madera/pdata.h index 601cbbc10370..32e3470708ed 100644 --- a/include/linux/mfd/madera/pdata.h +++ b/include/linux/mfd/madera/pdata.h @@ -31,7 +31,7 @@ struct pinctrl_map; * @irq_flags: Mode for primary IRQ (defaults to active low) * @gpio_base: Base GPIO number * @gpio_configs: Array of GPIO configurations (See - * Documentation/driver-api/pinctl.rst) + * Documentation/driver-api/pin-control.rst) * @n_gpio_configs: Number of entries in gpio_configs * @gpsw: General purpose switch mode setting. Depends on the external * hardware connected to the switch. (See the SW1_MODE field diff --git a/include/linux/pinctrl/pinconf-generic.h b/include/linux/pinctrl/pinconf-generic.h index 6aeb711f7cd1..fdcd773adba3 100644 --- a/include/linux/pinctrl/pinconf-generic.h +++ b/include/linux/pinctrl/pinconf-generic.h @@ -88,7 +88,7 @@ struct pinctrl_map; * it. * @PIN_CONFIG_OUTPUT: this will configure the pin as an output and drive a * value on the line. Use argument 1 to indicate high level, argument 0 to - * indicate low level. (Please see Documentation/driver-api/pinctl.rst, + * indicate low level. (Please see Documentation/driver-api/pin-control.rst, * section "GPIO mode pitfalls" for a discussion around this parameter.) * @PIN_CONFIG_POWER_SOURCE: if the pin can select between different power * supplies, the argument to this parameter (on a custom format) tells -- 2.30.2

3 2

[PATCH] ALSA: hda/realtek: Fix speaker amp setup on Acer Aspire E1
by Takashi Iwai 07 Apr '21

07 Apr '21

We've got a report about Acer Aspire E1 (PCI SSID 1025:0840) that loses the speaker output after resume. With the comparison of COEF dumps, it was identified that the COEF 0x0d bits 0x6000 corresponds to the speaker amp. This patch adds the specific quirk for the device to restore the COEF bits at the codec (re-)initialization. BugLink: https://bugzilla.suse.com/show_bug.cgi?id=1183869 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 16 ++++++++++++++++ 1 file changed, 16 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 58946d069ee5..a7544b77d3f7 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3927,6 +3927,15 @@ static void alc271_fixup_dmic(struct hda_codec *codec, snd_hda_sequence_write(codec, verbs); } +/* Fix the speaker amp after resume, etc */ +static void alc269vb_fixup_aspire_e1_coef(struct hda_codec *codec, + const struct hda_fixup *fix, + int action) +{ + if (action == HDA_FIXUP_ACT_INIT) + alc_update_coef_idx(codec, 0x0d, 0x6000, 0x6000); +} + static void alc269_fixup_pcm_44k(struct hda_codec *codec, const struct hda_fixup *fix, int action) { @@ -6301,6 +6310,7 @@ enum { ALC283_FIXUP_HEADSET_MIC, ALC255_FIXUP_MIC_MUTE_LED, ALC282_FIXUP_ASPIRE_V5_PINS, + ALC269VB_FIXUP_ASPIRE_E1_COEF, ALC280_FIXUP_HP_GPIO4, ALC286_FIXUP_HP_GPIO_LED, ALC280_FIXUP_HP_GPIO2_MIC_HOTKEY, @@ -6979,6 +6989,10 @@ static const struct hda_fixup alc269_fixups[] = { { }, }, }, + [ALC269VB_FIXUP_ASPIRE_E1_COEF] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc269vb_fixup_aspire_e1_coef, + }, [ALC280_FIXUP_HP_GPIO4] = { .type = HDA_FIXUP_FUNC, .v.func = alc280_fixup_hp_gpio4, @@ -7901,6 +7915,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x0762, "Acer Aspire E1-472", ALC271_FIXUP_HP_GATE_MIC_JACK_E1_572), SND_PCI_QUIRK(0x1025, 0x0775, "Acer Aspire E1-572", ALC271_FIXUP_HP_GATE_MIC_JACK_E1_572), SND_PCI_QUIRK(0x1025, 0x079b, "Acer Aspire V5-573G", ALC282_FIXUP_ASPIRE_V5_PINS), + SND_PCI_QUIRK(0x1025, 0x0840, "Acer Aspire E1", ALC269VB_FIXUP_ASPIRE_E1_COEF), SND_PCI_QUIRK(0x1025, 0x101c, "Acer Veriton N2510G", ALC269_FIXUP_LIFEBOOK), SND_PCI_QUIRK(0x1025, 0x102b, "Acer Aspire C24-860", ALC286_FIXUP_ACER_AIO_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1025, 0x1065, "Acer Aspire C20-820", ALC269VC_FIXUP_ACER_HEADSET_MIC), @@ -8395,6 +8410,7 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {.id = ALC283_FIXUP_HEADSET_MIC, .name = "alc283-headset"}, {.id = ALC255_FIXUP_MIC_MUTE_LED, .name = "alc255-dell-mute"}, {.id = ALC282_FIXUP_ASPIRE_V5_PINS, .name = "aspire-v5"}, + {.id = ALC269VB_FIXUP_ASPIRE_E1_COEF, .name = "aspire-e1-coef"}, {.id = ALC280_FIXUP_HP_GPIO4, .name = "hp-gpio4"}, {.id = ALC286_FIXUP_HP_GPIO_LED, .name = "hp-gpio-led"}, {.id = ALC280_FIXUP_HP_GPIO2_MIC_HOTKEY, .name = "hp-gpio2-hotkey"}, -- 2.26.2

1 0

[PATCH] ALSA: aloop: Fix initialization of controls
by Jonas Holmberg 07 Apr '21

07 Apr '21

Add a control to the card before copying the id so that the numid field is initialized in the copy. Otherwise the numid field of active_id, format_id, rate_id and channels_id will be the same (0) and snd_ctl_notify() will not queue the events properly. Signed-off-by: Jonas Holmberg <jonashg(a)axis.com> --- sound/drivers/aloop.c | 11 ++++++++--- 1 file changed, 8 insertions(+), 3 deletions(-) diff --git a/sound/drivers/aloop.c b/sound/drivers/aloop.c index 52637180af33..80b814b9922a 100644 --- a/sound/drivers/aloop.c +++ b/sound/drivers/aloop.c @@ -1571,6 +1571,14 @@ static int loopback_mixer_new(struct loopback *loopback, int notify) return -ENOMEM; kctl->id.device = dev; kctl->id.subdevice = substr; + + /* Add the control before copying the id so that + * the numid field of the id is set in the copy. + */ + err = snd_ctl_add(card, kctl); + if (err < 0) + return err; + switch (idx) { case ACTIVE_IDX: setup->active_id = kctl->id; @@ -1587,9 +1595,6 @@ static int loopback_mixer_new(struct loopback *loopback, int notify) default: break; } - err = snd_ctl_add(card, kctl); - if (err < 0) - return err; } } } -- 2.26.3

3 2

[PATCH 0/5] SOC Codecs: TLV320AIC3X add SPI support
by Jiri Prchal 06 Apr '21

06 Apr '21

This patches add support of SPI to TLV320AIC3X codecs. Jiri Prchal (5): SOC Codecs: TLV320AIC3X add SPI: move thinks to header file SOC Codecs: TLV320AIC3X add SPI: main source code for I2C and SPI SOC Codecs: TLV320AIC3X add SPI: I2C code SOC Codecs: TLV320AIC3X add SPI: SPI code SOC Codecs: TLV320AIC3X add SPI: configuration and compile sound/soc/codecs/Kconfig | 11 +++ sound/soc/codecs/Makefile | 4 ++ sound/soc/codecs/tlv320aic3x-i2c.c | 72 ++++++++++++++++++++ sound/soc/codecs/tlv320aic3x-spi.c | 77 +++++++++++++++++++++ sound/soc/codecs/tlv320aic3x.c | 105 ++++++++--------------------- sound/soc/codecs/tlv320aic3x.h | 13 ++++ 6 files changed, 204 insertions(+), 78 deletions(-) create mode 100644 sound/soc/codecs/tlv320aic3x-i2c.c create mode 100644 sound/soc/codecs/tlv320aic3x-spi.c -- 2.25.1

3 15

[PATCH] ASoC: amd: Add support for ALC1015P codec in acp3x machine driver
by Vijendar Mukunda 06 Apr '21

06 Apr '21

Add ALC1015p codec support for acp3x machine driver. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- sound/soc/amd/Kconfig | 1 + sound/soc/amd/acp3x-rt5682-max9836.c | 46 +++++++++++++++++++++++++++++++++++- 2 files changed, 46 insertions(+), 1 deletion(-) diff --git a/sound/soc/amd/Kconfig b/sound/soc/amd/Kconfig index 256c192..ba5a85b 100644 --- a/sound/soc/amd/Kconfig +++ b/sound/soc/amd/Kconfig @@ -35,6 +35,7 @@ config SND_SOC_AMD_RV_RT5682_MACH select SND_SOC_CROS_EC_CODEC select I2C_CROS_EC_TUNNEL select SND_SOC_RT1015 + select SND_SOC_RT1015P depends on SND_SOC_AMD_ACP3x && I2C && CROS_EC help This option enables machine driver for RT5682 and MAX9835. diff --git a/sound/soc/amd/acp3x-rt5682-max9836.c b/sound/soc/amd/acp3x-rt5682-max9836.c index cea320a..8e11bb8 100644 --- a/sound/soc/amd/acp3x-rt5682-max9836.c +++ b/sound/soc/amd/acp3x-rt5682-max9836.c @@ -275,6 +275,8 @@ SND_SOC_DAILINK_DEF(rt5682, DAILINK_COMP_ARRAY(COMP_CODEC("i2c-10EC5682:00", "rt5682-aif1"))); SND_SOC_DAILINK_DEF(max, DAILINK_COMP_ARRAY(COMP_CODEC("MX98357A:00", "HiFi"))); +SND_SOC_DAILINK_DEF(rt1015p, + DAILINK_COMP_ARRAY(COMP_CODEC("RTL1015:00", "HiFi"))); SND_SOC_DAILINK_DEF(rt1015, DAILINK_COMP_ARRAY(COMP_CODEC("i2c-10EC1015:00", "rt1015-aif"), COMP_CODEC("i2c-10EC1015:01", "rt1015-aif"))); @@ -419,6 +421,43 @@ static struct snd_soc_card acp3x_1015 = { .num_controls = ARRAY_SIZE(acp3x_mc_1015_controls), }; +static const struct snd_soc_dapm_widget acp3x_1015p_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_MUX("Dmic Mux", SND_SOC_NOPM, 0, 0, + &acp3x_dmic_mux_control), + SND_SOC_DAPM_SPK("Speakers", NULL), +}; + +static const struct snd_soc_dapm_route acp3x_1015p_route[] = { + {"Headphone Jack", NULL, "HPOL"}, + {"Headphone Jack", NULL, "HPOR"}, + {"IN1P", NULL, "Headset Mic"}, + {"Dmic Mux", "Front Mic", "DMIC"}, + {"Dmic Mux", "Rear Mic", "DMIC"}, + /* speaker */ + { "Speakers", NULL, "Speaker" }, +}; + +static const struct snd_kcontrol_new acp3x_mc_1015p_controls[] = { + SOC_DAPM_PIN_SWITCH("Speakers"), + SOC_DAPM_PIN_SWITCH("Headphone Jack"), + SOC_DAPM_PIN_SWITCH("Headset Mic"), +}; + +static struct snd_soc_card acp3x_1015p = { + .name = "acp3xalc56821015p", + .owner = THIS_MODULE, + .dai_link = acp3x_dai, + .num_links = ARRAY_SIZE(acp3x_dai), + .dapm_widgets = acp3x_1015p_widgets, + .num_dapm_widgets = ARRAY_SIZE(acp3x_1015p_widgets), + .dapm_routes = acp3x_1015p_route, + .num_dapm_routes = ARRAY_SIZE(acp3x_1015p_route), + .controls = acp3x_mc_1015p_controls, + .num_controls = ARRAY_SIZE(acp3x_mc_1015p_controls), +}; + void *soc_is_rltk_max(struct device *dev) { const struct acpi_device_id *match; @@ -435,6 +474,9 @@ static void card_spk_dai_link_present(struct snd_soc_dai_link *links, if (!strcmp(card_name, "acp3xalc56821015")) { links[1].codecs = rt1015; links[1].num_codecs = ARRAY_SIZE(rt1015); + } else if (!strcmp(card_name, "acp3xalc56821015p")) { + links[1].codecs = rt1015p; + links[1].num_codecs = ARRAY_SIZE(rt1015p); } else { links[1].codecs = max; links[1].num_codecs = ARRAY_SIZE(max); @@ -486,6 +528,7 @@ static int acp3x_probe(struct platform_device *pdev) static const struct acpi_device_id acp3x_audio_acpi_match[] = { { "AMDI5682", (unsigned long)&acp3x_5682}, { "AMDI1015", (unsigned long)&acp3x_1015}, + { "AMDP1015", (unsigned long)&acp3x_1015p}, {}, }; MODULE_DEVICE_TABLE(acpi, acp3x_audio_acpi_match); @@ -503,5 +546,6 @@ module_platform_driver(acp3x_audio); MODULE_AUTHOR("akshu.agrawal(a)amd.com"); MODULE_AUTHOR("Vishnuvardhanrao.Ravulapati(a)amd.com"); -MODULE_DESCRIPTION("ALC5682 ALC1015 & MAX98357 audio support"); +MODULE_AUTHOR("Vijendar.Mukunda(a)amd.com"); +MODULE_DESCRIPTION("ALC5682 ALC1015, ALC1015P & MAX98357 audio support"); MODULE_LICENSE("GPL v2"); -- 2.7.4

4 7

[PATCH v2 0/2] ASoC: tlv320aic32x4: Bugfixes for tlv320aic32x4 driver
by Annaliese McDermond 06 Apr '21

06 Apr '21

These are a couple of bugfixes for the tlv320aic32x4 driver that we discovered after the Raspberry Pi Foundation's move to the 5.10 kernel. The first increases the maximum value in the regmap configuration so that the powerup command actually works. The second changes the ordering of the clock registration in the module probe function so that the clocks are properly registered before the component probe function runs. Changes in v2: - Corrected commit hashes in Fixes lines to be proper length - Including inadvertently omitted maintainers on To list. Annaliese McDermond (2): ASoC: tlv320aic32x4: Increase maximum register in regmap ASoC: tlv320aic32x4: Register clocks before registering component sound/soc/codecs/tlv320aic32x4.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) -- 2.27.0

2 1

[PATCH] ASoC: ak5558: Fix s/show/slow/ typo
by Shengjiu Wang 06 Apr '21

06 Apr '21

s/show/slow/ Fixes: 920884777480 ("ASoC: ak5558: Add support for AK5558 ADC driver") Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/codecs/ak5558.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/ak5558.c b/sound/soc/codecs/ak5558.c index 3ac62a281807..8e4dca753f0b 100644 --- a/sound/soc/codecs/ak5558.c +++ b/sound/soc/codecs/ak5558.c @@ -75,8 +75,8 @@ static const struct soc_enum ak5552_mono_enum[] = { }; static const char * const digfil_texts[] = { - "Sharp Roll-Off", "Show Roll-Off", - "Short Delay Sharp Roll-Off", "Short Delay Show Roll-Off", + "Sharp Roll-Off", "Slow Roll-Off", + "Short Delay Sharp Roll-Off", "Short Delay Slow Roll-Off", }; static const struct soc_enum ak5558_adcset_enum[] = { -- 2.27.0

2 1

[PATCH v3 0/2] kbl_da7219_max9357a machine changes for wov and MST
by vamshi.krishna.gopal＠intel.com 06 Apr '21

06 Apr '21

From: Vamshi Krishna Gopal <vamshi.krishna.gopal(a)intel.corp-partner.google.com> Hi all, This patch series about creating dailink for Wake on voice functionality and also adding MST route changes. changes in v3: -Remove DP from widgets -Add MST support in other routing table for kbl_rt5663_max98927.c v2 is here: https://patchwork.kernel.org/project/alsa-devel/cover/20210325174325.31802-… v1 is here: https://patchwork.kernel.org/project/alsa-devel/cover/20210324175200.44922-… Mac Chiang (1): ASoc: Intel: board: add BE DAI link for WoV Vamshi Krishna Gopal (1): ASoC: Intel: kbl: Add MST route change to kbl machine drivers sound/soc/intel/boards/kbl_da7219_max98357a.c | 58 +++++++++++++++++-- sound/soc/intel/boards/kbl_da7219_max98927.c | 10 ++-- sound/soc/intel/boards/kbl_rt5663_max98927.c | 14 +++-- 3 files changed, 70 insertions(+), 12 deletions(-) -- 2.17.1

3 4

[PATCH] ASoC: max98390: Add support for tx slot configuration.
by Steve Lee 06 Apr '21

06 Apr '21

Update voltage/current tx slot configuration support. Signed-off-by: Steve Lee <steves.lee(a)maximintegrated.com> --- sound/soc/codecs/max98390.c | 62 +++++++++++++++++++++++++++++++++++++ sound/soc/codecs/max98390.h | 2 ++ 2 files changed, 64 insertions(+) diff --git a/sound/soc/codecs/max98390.c b/sound/soc/codecs/max98390.c index bb736c44e68a..94773ccee9d5 100644 --- a/sound/soc/codecs/max98390.c +++ b/sound/soc/codecs/max98390.c @@ -856,6 +856,48 @@ static void max98390_init_regs(struct snd_soc_component *component) regmap_write(max98390->regmap, MAX98390_ENV_TRACK_VOUT_HEADROOM, 0x0e); regmap_write(max98390->regmap, MAX98390_BOOST_BYPASS1, 0x46); regmap_write(max98390->regmap, MAX98390_FET_SCALING3, 0x03); + + /* voltage, current slot configuration */ + regmap_write(max98390->regmap, + MAX98390_PCM_CH_SRC_2, + (max98390->i_l_slot << 4 | + max98390->v_l_slot)&0xFF); + + if (max98390->v_l_slot < 8) { + regmap_update_bits(max98390->regmap, + MAX98390_PCM_TX_HIZ_CTRL_A, + 1 << max98390->v_l_slot, 0); + regmap_update_bits(max98390->regmap, + MAX98390_PCM_TX_EN_A, + 1 << max98390->v_l_slot, + 1 << max98390->v_l_slot); + } else { + regmap_update_bits(max98390->regmap, + MAX98390_PCM_TX_HIZ_CTRL_B, + 1 << (max98390->v_l_slot - 8), 0); + regmap_update_bits(max98390->regmap, + MAX98390_PCM_TX_EN_B, + 1 << (max98390->v_l_slot - 8), + 1 << (max98390->v_l_slot - 8)); + } + + if (max98390->i_l_slot < 8) { + regmap_update_bits(max98390->regmap, + MAX98390_PCM_TX_HIZ_CTRL_A, + 1 << max98390->i_l_slot, 0); + regmap_update_bits(max98390->regmap, + MAX98390_PCM_TX_EN_A, + 1 << max98390->i_l_slot, + 1 << max98390->i_l_slot); + } else { + regmap_update_bits(max98390->regmap, + MAX98390_PCM_TX_HIZ_CTRL_B, + 1 << (max98390->i_l_slot - 8), 0); + regmap_update_bits(max98390->regmap, + MAX98390_PCM_TX_EN_B, + 1 << (max98390->i_l_slot - 8), + 1 << (max98390->i_l_slot - 8)); + } } static int max98390_probe(struct snd_soc_component *component) @@ -946,6 +988,23 @@ static const struct regmap_config max98390_regmap = { .cache_type = REGCACHE_RBTREE, }; +static void max98390_slot_config(struct i2c_client *i2c, + struct max98390_priv *max98390) +{ + int value; + struct device *dev = &i2c->dev; + + if (!device_property_read_u32(dev, "maxim,vmon-slot-no", &value)) + max98390->v_l_slot = value & 0xF; + else + max98390->v_l_slot = 0; + + if (!device_property_read_u32(dev, "maxim,imon-slot-no", &value)) + max98390->i_l_slot = value & 0xF; + else + max98390->i_l_slot = 1; +} + static int max98390_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -988,6 +1047,9 @@ static int max98390_i2c_probe(struct i2c_client *i2c, __func__, max98390->ref_rdc_value, max98390->ambient_temp_value); + /* voltage/current slot configuration */ + max98390_slot_config(i2c, max98390); + /* regmap initialization */ max98390->regmap = devm_regmap_init_i2c(i2c, &max98390_regmap); if (IS_ERR(max98390->regmap)) { diff --git a/sound/soc/codecs/max98390.h b/sound/soc/codecs/max98390.h index dff884f68e3e..e31516717d3b 100644 --- a/sound/soc/codecs/max98390.h +++ b/sound/soc/codecs/max98390.h @@ -658,6 +658,8 @@ struct max98390_priv { unsigned int sysclk; unsigned int master; unsigned int tdm_mode; + unsigned int v_l_slot; + unsigned int i_l_slot; unsigned int ref_rdc_value; unsigned int ambient_temp_value; }; -- 2.17.1

2 1

dsnoop/dshare/dmix: snd_pcm_htimestamp returns time past since last boot not clock time
by GitHub issues - edited 06 Apr '21

06 Apr '21

alsa-project/alsa-lib issue #131 was edited from RonaldAJ: While on a PC snd_pcm_htimestamp returns the clock time since the epoch, under Raspbian on an RPI it returns the time since the last boot. (A similar issue was described here: https://stackoverflow.com/questions/45841230/alsa-retrieving-audio-buffer-t…). The problem shows if we run a slightly modified version of recordtest.py using the latest pyalsaaudio versions. ``` #!/usr/bin/env python3 # -*- mode: python; indent-tabs-mode: t; c-basic-offset: 4; tab-width: 4 -*- ## recordtest.py ## ## This is an example of a simple sound capture script. ## ## The script opens an ALSA pcm device for sound capture, sets ## various attributes of the capture, and reads in a loop, ## writing the data to standard out. ## ## To test it out do the following: ## python recordtest.py out.raw # talk to the microphone ## aplay -r 8000 -f S16_LE -c 1 out.raw #!/usr/bin/env python from __future__ import print_function import sys import time import getopt import alsaaudio def usage(): print('usage: recordtest.py [-d <device>] <file>', file=sys.stderr) sys.exit(2) if __name__ == '__main__': device = 'default' opts, args = getopt.getopt(sys.argv[1:], 'd:') for o, a in opts: if o == '-d': device = a if not args: usage() f = open(args[0], 'wb') # Open the device in nonblocking capture mode in mono, with a sampling rate of 44100 Hz # and 16 bit little endian samples # The period size controls the internal number of frames per period. # The significance of this parameter is documented in the ALSA api. # For our purposes, it is suficcient to know that reads from the device # will return this many frames. Each frame being 2 bytes long. # This means that the reads below will return either 320 bytes of data # or 0 bytes of data. The latter is possible because we are in nonblocking # mode. inp = alsaaudio.PCM(alsaaudio.PCM_CAPTURE, alsaaudio.PCM_NONBLOCK, channels=1, rate=44100, format=alsaaudio.PCM_FORMAT_S16_LE, periodsize=160, device=device) print(inp.info()) # help(inp.htimestamp) loops = 100000 while loops > 0: loops -= 1 # Read data from device l, data = inp.read() if l: print(l, inp.htimestamp(), time.time()) f.write(data) time.sleep(.001) ``` When executed on an RPi it yields: ``` {'name': 'plugsnoop:1,0', 'state': 'PREPARED', 'access_type': 'RW_INTERLEAVED', ' (call value) type': 1, ' (call value) type_name': 'CAPTURE', ' (call value) mode': 1, ' (call value) mode_name': 'PCM_NONBLOCK', 'format': 2, 'format_name': 'S16_LE', 'format_description': 'Signed 16 bit Little Endian', 'subformat_name': 'STD', 'subformat_description': 'Standard', 'channels': 1, 'rate': 44100, 'period_time': 21333, 'period_size': 940, 'buffer_time': 0, 'buffer_size': 4704, 'get_periods': 0, 'rate_numden': (44100, 1), 'significant_bits': 16, 'is_batch': True, 'is_block_transfer': True, 'is_double': False, 'is_half_duplex': False, 'is_joint_duplex': False, 'can_overrange': False, 'can_mmap_sample_resolution': False, 'can_pause': False, 'can_resume': False, 'can_sync_start': False} 940 (5596, 851796145, 0) 1617702646.0637105 940 (5596, 872796030, 0) 1617702646.084714 940 (5596, 893795916, 0) 1617702646.105702 940 (5596, 915796158, 0) 1617702646.1277127 940 (5596, 936795314, 0) 1617702646.1486921 940 (5596, 957795199, 0) 1617702646.1697078 940 (5596, 979795858, 0) 1617702646.191694 940 (5597, 795171, 0) 1617702646.212712 ... ``` I read that for the timestamp function to work there should be hardware support. But here we have it half working. This behavior is shown if I use the following in my .asoundrc ``` pcm.plugsnoop { type plug @args [ CARD DEV ] @args.CARD { type string } @args.DEV { type integer } slave { pcm { @func concat strings [ "dsnoop:" $CARD "," $DEV ] } } } ``` and call it with: `python recordtest.py -d plugsnoop:1,0 test.raw` Using plughw all timestamps become zero. The attached microphone is an USB microphone: GoMic by SAMSON. Is there a fix or workaround for this behavior which gives me clocktime instead of seconds since last boot? Issue URL : https://github.com/alsa-project/alsa-lib/issues/131 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

dsnoop: snd_pcm_htimestamp returns time past since last boot not clock time
by GitHub issues - edited 06 Apr '21

06 Apr '21

alsa-project/alsa-lib issue #131 was edited from RonaldAJ: While on a PC snd_pcm_htimestamp returns the clock time since the epoch, under Raspbian on an RPI it returns the time since the last boot. (A similar issue was described here: https://stackoverflow.com/questions/45841230/alsa-retrieving-audio-buffer-t…). The problem shows if we run a slightly modified version of recordtest.py using the latest pyalsaaudio versions. ``` #!/usr/bin/env python3 # -*- mode: python; indent-tabs-mode: t; c-basic-offset: 4; tab-width: 4 -*- ## recordtest.py ## ## This is an example of a simple sound capture script. ## ## The script opens an ALSA pcm device for sound capture, sets ## various attributes of the capture, and reads in a loop, ## writing the data to standard out. ## ## To test it out do the following: ## python recordtest.py out.raw # talk to the microphone ## aplay -r 8000 -f S16_LE -c 1 out.raw #!/usr/bin/env python from __future__ import print_function import sys import time import getopt import alsaaudio def usage(): print('usage: recordtest.py [-d <device>] <file>', file=sys.stderr) sys.exit(2) if __name__ == '__main__': device = 'default' opts, args = getopt.getopt(sys.argv[1:], 'd:') for o, a in opts: if o == '-d': device = a if not args: usage() f = open(args[0], 'wb') # Open the device in nonblocking capture mode in mono, with a sampling rate of 44100 Hz # and 16 bit little endian samples # The period size controls the internal number of frames per period. # The significance of this parameter is documented in the ALSA api. # For our purposes, it is suficcient to know that reads from the device # will return this many frames. Each frame being 2 bytes long. # This means that the reads below will return either 320 bytes of data # or 0 bytes of data. The latter is possible because we are in nonblocking # mode. inp = alsaaudio.PCM(alsaaudio.PCM_CAPTURE, alsaaudio.PCM_NONBLOCK, channels=1, rate=44100, format=alsaaudio.PCM_FORMAT_S16_LE, periodsize=160, device=device) print(inp.info()) # help(inp.htimestamp) loops = 100000 while loops > 0: loops -= 1 # Read data from device l, data = inp.read() if l: print(l, inp.htimestamp(), time.time()) f.write(data) time.sleep(.001) ``` When executed on an RPi it yields: ``` {'name': 'plugsnoop:1,0', 'state': 'PREPARED', 'access_type': 'RW_INTERLEAVED', ' (call value) type': 1, ' (call value) type_name': 'CAPTURE', ' (call value) mode': 1, ' (call value) mode_name': 'PCM_NONBLOCK', 'format': 2, 'format_name': 'S16_LE', 'format_description': 'Signed 16 bit Little Endian', 'subformat_name': 'STD', 'subformat_description': 'Standard', 'channels': 1, 'rate': 44100, 'period_time': 21333, 'period_size': 940, 'buffer_time': 0, 'buffer_size': 4704, 'get_periods': 0, 'rate_numden': (44100, 1), 'significant_bits': 16, 'is_batch': True, 'is_block_transfer': True, 'is_double': False, 'is_half_duplex': False, 'is_joint_duplex': False, 'can_overrange': False, 'can_mmap_sample_resolution': False, 'can_pause': False, 'can_resume': False, 'can_sync_start': False} 940 (5596, 851796145, 0) 1617702646.0637105 940 (5596, 872796030, 0) 1617702646.084714 940 (5596, 893795916, 0) 1617702646.105702 940 (5596, 915796158, 0) 1617702646.1277127 940 (5596, 936795314, 0) 1617702646.1486921 940 (5596, 957795199, 0) 1617702646.1697078 940 (5596, 979795858, 0) 1617702646.191694 940 (5597, 795171, 0) 1617702646.212712 ... ``` I read that for the timestamp function to work there should be hardware support. But here we have it half working. This behavior is shown if I use the following in my .asoundrc ``` pcm.plugsnoop { type plug @args [ CARD DEV ] @args.CARD { type string } @args.DEV { type integer } slave { pcm { @func concat strings [ "dsnoop:" $CARD "," $DEV ] } } } ``` and call it with: `python recordtest.py -d plugsnoop:1,0 test.raw` Using plughw all timestamps become zero. The attached microphone is an USB microphone: GoMic by SAMSON. Is there a fix or workaround for this behavior which gives me clocktime instead of seconds since last boot? Issue URL : https://github.com/alsa-project/alsa-lib/issues/131 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

cplay: Add suport for AAC audio codec
by GitHub pull_request - opened 06 Apr '21

06 Apr '21

alsa-project/tinycompress pull request #10 was opened from TangleZ: Add parse_aac_header API to parse AAC streams, support ADTS and ADIF format. Signed-off-by: Zhang Peng <peng.zhang_8(a)nxp.com> Request URL : https://github.com/alsa-project/tinycompress/pull/10 Patch URL : https://github.com/alsa-project/tinycompress/pull/10.patch Repository URL: https://github.com/alsa-project/tinycompress

1 0

[PATCH] ALSA: usb-audio: Drop implicit fb quirk entries dubbed for capture
by Takashi Iwai 06 Apr '21

06 Apr '21

The implicit feedback quirk table contains the entries that also appear in the capture quirks, and those are all handled to be skipped. For the code simplicity, drop the duped entries in the playback quirk table, and check the match with the capture quirk table instead. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=212519 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/implicit.c | 15 +++++---------- 1 file changed, 5 insertions(+), 10 deletions(-) diff --git a/sound/usb/implicit.c b/sound/usb/implicit.c index 11a85e66aa96..19622c58b72b 100644 --- a/sound/usb/implicit.c +++ b/sound/usb/implicit.c @@ -70,16 +70,6 @@ static const struct snd_usb_implicit_fb_match playback_implicit_fb_quirks[] = { .type = IMPLICIT_FB_FIXED, .ep_num = 0x84, .iface = 0 }, /* MOTU MicroBook II */ - /* No quirk for playback but with capture quirk (see below) */ - IMPLICIT_FB_SKIP_DEV(0x0582, 0x0130), /* BOSS BR-80 */ - IMPLICIT_FB_SKIP_DEV(0x0582, 0x0171), /* BOSS RC-505 */ - IMPLICIT_FB_SKIP_DEV(0x0582, 0x0185), /* BOSS GP-10 */ - IMPLICIT_FB_SKIP_DEV(0x0582, 0x0189), /* BOSS GT-100v2 */ - IMPLICIT_FB_SKIP_DEV(0x0582, 0x01d6), /* BOSS GT-1 */ - IMPLICIT_FB_SKIP_DEV(0x0582, 0x01d8), /* BOSS Katana */ - IMPLICIT_FB_SKIP_DEV(0x0582, 0x01e5), /* BOSS GT-001 */ - IMPLICIT_FB_SKIP_DEV(0x0582, 0x0203), /* BOSS AD-10 */ - {} /* terminator */ }; @@ -278,6 +268,11 @@ static int audioformat_implicit_fb_quirk(struct snd_usb_audio *chip, } } + /* Don't apply playback quirks for the devices with capture quirk */ + p = find_implicit_fb_entry(chip, capture_implicit_fb_quirks, alts); + if (p && p->type == IMPLICIT_FB_FIXED) + return 0; /* no quirk */ + /* Generic UAC2 implicit feedback */ if (attr == USB_ENDPOINT_SYNC_ASYNC && alts->desc.bInterfaceClass == USB_CLASS_AUDIO && -- 2.26.2

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[PATCH] ALSA: usb-audio: Add error checks for usb_driver_claim_interface() calls
by Takashi Iwai 06 Apr '21

06 Apr '21

There are a few calls of usb_driver_claim_interface() but all of those miss the proper error checks, as reported by Coverity. This patch adds those missing checks. Along with it, replace the magic pointer with -1 with a constant USB_AUDIO_IFACE_UNUSED for better readability. Reported-by: coverity-bot <keescook+coverity-bot(a)chromium.org> Addresses-Coverity-ID: 1475943 ("Error handling issues") Addresses-Coverity-ID: 1475944 ("Error handling issues") Addresses-Coverity-ID: 1475945 ("Error handling issues") Fixes: b1ce7ba619d9 ("ALSA: usb-audio: claim autodetected PCM interfaces all at once") Fixes: e5779998bf8b ("ALSA: usb-audio: refactor code") Link: https://lore.kernel.org/r/202104051059.FB7F3016@keescook Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/card.c | 14 +++++++------- sound/usb/quirks.c | 16 ++++++++++++---- sound/usb/usbaudio.h | 2 ++ 3 files changed, 21 insertions(+), 11 deletions(-) diff --git a/sound/usb/card.c b/sound/usb/card.c index 0826a437f8fc..7b7526d3a56e 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -181,9 +181,8 @@ static int snd_usb_create_stream(struct snd_usb_audio *chip, int ctrlif, int int ctrlif, interface); return -EINVAL; } - usb_driver_claim_interface(&usb_audio_driver, iface, (void *)-1L); - - return 0; + return usb_driver_claim_interface(&usb_audio_driver, iface, + USB_AUDIO_IFACE_UNUSED); } if ((altsd->bInterfaceClass != USB_CLASS_AUDIO && @@ -203,7 +202,8 @@ static int snd_usb_create_stream(struct snd_usb_audio *chip, int ctrlif, int int if (! snd_usb_parse_audio_interface(chip, interface)) { usb_set_interface(dev, interface, 0); /* reset the current interface */ - usb_driver_claim_interface(&usb_audio_driver, iface, (void *)-1L); + return usb_driver_claim_interface(&usb_audio_driver, iface, + USB_AUDIO_IFACE_UNUSED); } return 0; @@ -862,7 +862,7 @@ static void usb_audio_disconnect(struct usb_interface *intf) struct snd_card *card; struct list_head *p; - if (chip == (void *)-1L) + if (chip == USB_AUDIO_IFACE_UNUSED) return; card = chip->card; @@ -992,7 +992,7 @@ static int usb_audio_suspend(struct usb_interface *intf, pm_message_t message) struct usb_mixer_interface *mixer; struct list_head *p; - if (chip == (void *)-1L) + if (chip == USB_AUDIO_IFACE_UNUSED) return 0; if (!chip->num_suspended_intf++) { @@ -1022,7 +1022,7 @@ static int __usb_audio_resume(struct usb_interface *intf, bool reset_resume) struct list_head *p; int err = 0; - if (chip == (void *)-1L) + if (chip == USB_AUDIO_IFACE_UNUSED) return 0; atomic_inc(&chip->active); /* avoid autopm */ diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index ae667cf77e61..63f27ce64bcb 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -55,8 +55,12 @@ static int create_composite_quirk(struct snd_usb_audio *chip, if (!iface) continue; if (quirk->ifnum != probed_ifnum && - !usb_interface_claimed(iface)) - usb_driver_claim_interface(driver, iface, (void *)-1L); + !usb_interface_claimed(iface)) { + err = usb_driver_claim_interface(driver, iface, + USB_AUDIO_IFACE_UNUSED); + if (err < 0) + return err; + } } return 0; @@ -426,8 +430,12 @@ static int create_autodetect_quirks(struct snd_usb_audio *chip, continue; err = create_autodetect_quirk(chip, iface, driver); - if (err >= 0) - usb_driver_claim_interface(driver, iface, (void *)-1L); + if (err >= 0) { + err = usb_driver_claim_interface(driver, iface, + USB_AUDIO_IFACE_UNUSED); + if (err < 0) + return err; + } } return 0; diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index 60b9dd7df6bb..8794c8658ab9 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -61,6 +61,8 @@ struct snd_usb_audio { struct media_intf_devnode *ctl_intf_media_devnode; }; +#define USB_AUDIO_IFACE_UNUSED ((void *)-1L) + #define usb_audio_err(chip, fmt, args...) \ dev_err(&(chip)->dev->dev, fmt, ##args) #define usb_audio_warn(chip, fmt, args...) \ -- 2.26.2

1 0

snd_pcm_htimestamp returns time past since last boot not clock time
by GitHub issues - opened 06 Apr '21

06 Apr '21

alsa-project/alsa-lib issue #131 was opened from RonaldAJ: While on a PC snd_pcm_htimestamp returns the clock time since the epoch, under Raspbian on an RPI it returns the time since the last boot. (A similar issue was described here: https://stackoverflow.com/questions/45841230/alsa-retrieving-audio-buffer-t…). The problem shows if we run a slightly modified version of recordtest.py using the latest pyalsaaudio versions. ``` #!/usr/bin/env python3 # -*- mode: python; indent-tabs-mode: t; c-basic-offset: 4; tab-width: 4 -*- ## recordtest.py ## ## This is an example of a simple sound capture script. ## ## The script opens an ALSA pcm device for sound capture, sets ## various attributes of the capture, and reads in a loop, ## writing the data to standard out. ## ## To test it out do the following: ## python recordtest.py out.raw # talk to the microphone ## aplay -r 8000 -f S16_LE -c 1 out.raw #!/usr/bin/env python from __future__ import print_function import sys import time import getopt import alsaaudio def usage(): print('usage: recordtest.py [-d <device>] <file>', file=sys.stderr) sys.exit(2) if __name__ == '__main__': device = 'default' opts, args = getopt.getopt(sys.argv[1:], 'd:') for o, a in opts: if o == '-d': device = a if not args: usage() f = open(args[0], 'wb') # Open the device in nonblocking capture mode in mono, with a sampling rate of 44100 Hz # and 16 bit little endian samples # The period size controls the internal number of frames per period. # The significance of this parameter is documented in the ALSA api. # For our purposes, it is suficcient to know that reads from the device # will return this many frames. Each frame being 2 bytes long. # This means that the reads below will return either 320 bytes of data # or 0 bytes of data. The latter is possible because we are in nonblocking # mode. inp = alsaaudio.PCM(alsaaudio.PCM_CAPTURE, alsaaudio.PCM_NONBLOCK, channels=1, rate=44100, format=alsaaudio.PCM_FORMAT_S16_LE, periodsize=160, device=device) print(inp.info()) # help(inp.htimestamp) loops = 100000 while loops > 0: loops -= 1 # Read data from device l, data = inp.read() if l: print(l, inp.htimestamp(), time.time()) f.write(data) time.sleep(.001) ``` When executed on an RPi it yields: ``` {'name': 'plugsnoop:1,0', 'state': 'PREPARED', 'access_type': 'RW_INTERLEAVED', ' (call value) type': 1, ' (call value) type_name': 'CAPTURE', ' (call value) mode': 1, ' (call value) mode_name': 'PCM_NONBLOCK', 'format': 2, 'format_name': 'S16_LE', 'format_description': 'Signed 16 bit Little Endian', 'subformat_name': 'STD', 'subformat_description': 'Standard', 'channels': 1, 'rate': 44100, 'period_time': 21333, 'period_size': 940, 'buffer_time': 0, 'buffer_size': 4704, 'get_periods': 0, 'rate_numden': (44100, 1), 'significant_bits': 16, 'is_batch': True, 'is_block_transfer': True, 'is_double': False, 'is_half_duplex': False, 'is_joint_duplex': False, 'can_overrange': False, 'can_mmap_sample_resolution': False, 'can_pause': False, 'can_resume': False, 'can_sync_start': False} 940 (5596, 851796145, 0) 1617702646.0637105 940 (5596, 872796030, 0) 1617702646.084714 940 (5596, 893795916, 0) 1617702646.105702 940 (5596, 915796158, 0) 1617702646.1277127 940 (5596, 936795314, 0) 1617702646.1486921 940 (5596, 957795199, 0) 1617702646.1697078 940 (5596, 979795858, 0) 1617702646.191694 940 (5597, 795171, 0) 1617702646.212712 ... ``` I read that for the timestamp function to work there should be hardware support. But here we have it half working. This behavior is shown if I use the following in my .asoundrc ``` pcm.plugsnoop { type plug @args [ CARD DEV ] @args.CARD { type string } @args.DEV { type integer } slave { pcm { @func concat strings [ "dsnoop:" $CARD "," $DEV ] } } } ``` and call it with: `python recordtest.py -d plugsnoop:1,0 test.raw` Using plughw all timestamps become zero. The attached microphone is an USB microphone: GoMic by SAMSON. Is there a fix or workaround for this behavior which gives me clocktime instead of seconds since last boot? Issue URL : https://github.com/alsa-project/alsa-lib/issues/131 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

RE: [PATCH v6 2/2] ASoC: rt715:add micmute led state control supportspobrn@protonmail.com
by Yuan, Perry 06 Apr '21

06 Apr '21

Hi Jaroslav: > -----Original Message----- > From: Yuan, Perry <Perry_Yuan(a)Dell.com> > Sent: Sunday, April 4, 2021 4:32 PM > To: pobrn(a)protonmail.com; pierre-louis.bossart(a)linux.intel.com; > oder_chiou(a)realtek.com; perex(a)perex.cz; tiwai(a)suse.com; > hdegoede(a)redhat.com; mgross(a)linux.intel.com; Limonciello, Mario > Cc: lgirdwood(a)gmail.com; broonie(a)kernel.org; alsa-devel(a)alsa-project.org; > linux-kernel(a)vger.kernel.org; platform-driver-x86(a)vger.kernel.org; Yuan, > Perry; mario.limonciello(a)outlook.com; Dell Client Kernel > Subject: [PATCH v6 2/2] ASoC: rt715:add micmute led state control supports > > From: Perry Yuan <perry_yuan(a)dell.com> > > Some new Dell system is going to support audio internal micphone privacy > setting from hardware level with micmute led state changing When micmute > hotkey pressed by user, soft mute will need to be enabled firstly in case of > pop noise, and codec driver need to react to mic mute event to > EC(embedded controller) notifying that SW mute is completed Then EC will > do the hardware mute physically within the timeout reached > > This patch allow codec rt715 and rt715 sdca driver to change the local > micmute led state. Dell privacy led trigger driver will ack EC when micmute > key pressed through this micphone led control interface like hda_generic > provided ACPI method defined in dell-privacy micmute led trigger will be > called for notifying the EC that software mute has been completed, then > hardware audio circuit solution controlled by EC will switch the audio input > source off/on > > Signed-off-by: Perry Yuan <perry_yuan(a)dell.com> > > -------- > v5 -> v6: > * addresed review comments from Jaroslav > * add quirks for micmute led control as short term solution to control > micmute led state change > v4 -> v5: > * rebase to latest 5.12 rc4 upstream kernel > v3 -> v4: > * remove unused debug log > * remove compile flag of DELL privacy > * move the micmute_led to local from rt715_priv > * when Jaroslav upstream his gerneric LED trigger driver,I will rebase > this patch,please consider merge this at first > https://lore.kernel.org/alsa-devel/20210211111400.1131020-1- > perex(a)perex.cz/ > v2 -> v3: > * simplify the patch to reuse some val value > * add more detail to the commit info > v1 -> v2: > * fix some format issue > -------- > --- > sound/soc/codecs/rt715-sdca.c | 41 > ++++++++++++++++++++++++++++++++++- > sound/soc/codecs/rt715.c | 41 > +++++++++++++++++++++++++++++++++++ > 2 files changed, 81 insertions(+), 1 deletion(-) > > diff --git a/sound/soc/codecs/rt715-sdca.c b/sound/soc/codecs/rt715-sdca.c > index 20528afbdc57..7bd7ad0ba7d7 100644 > --- a/sound/soc/codecs/rt715-sdca.c > +++ b/sound/soc/codecs/rt715-sdca.c > @@ -11,12 +11,14 @@ > #include <linux/moduleparam.h> > #include <linux/kernel.h> > #include <linux/init.h> > +#include <linux/leds.h> > #include <linux/pm_runtime.h> > #include <linux/pm.h> > #include <linux/soundwire/sdw.h> > #include <linux/regmap.h> > #include <linux/slab.h> > #include <linux/platform_device.h> > +#include <linux/dmi.h> > #include <sound/core.h> > #include <sound/pcm.h> > #include <sound/pcm_params.h> > @@ -344,6 +346,34 @@ static int rt715_sdca_get_volsw(struct snd_kcontrol > *kcontrol, > return 0; > } > > +static bool micmute_led_set; > +static int dmi_matched(const struct dmi_system_id *dmi) { > + micmute_led_set = 1; > + return 1; > +} > + > +/* Some systems will need to use this to trigger mic mute LED state > +changed */ static const struct dmi_system_id micmute_led_dmi_table[] = { > + { > + .callback = dmi_matched, > + .ident = "Dell Latitude 9420", > + .matches = { > + DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc."), > + DMI_MATCH(DMI_PRODUCT_NAME, "Latitude > 9420"), > + }, > + }, > + { > + .callback = dmi_matched, > + .ident = "Dell Latitude 9520", > + .matches = { > + DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc."), > + DMI_MATCH(DMI_PRODUCT_NAME, "Latitude > 9520"), > + }, > + }, > + {}, > +}; > + > static int rt715_sdca_put_volsw(struct snd_kcontrol *kcontrol, > struct snd_ctl_elem_value *ucontrol) > { > @@ -358,6 +388,7 @@ static int rt715_sdca_put_volsw(struct snd_kcontrol > *kcontrol, > unsigned int mask = (1 << fls(max)) - 1; > unsigned int invert = p->invert; > int err; > + bool micmute_led; > > for (i = 0; i < 4; i++) { > if (ucontrol->value.integer.value[i] != rt715- > >kctl_switch_orig[i]) { @@ -393,7 +424,15 @@ static int > rt715_sdca_put_volsw(struct snd_kcontrol *kcontrol, > if (err < 0) > return err; > } > - > + /* Micmute LED state changed by muted/unmute switch */ > + dmi_check_system(micmute_led_dmi_table); > + if (invert && micmute_led_set) { > + if (ucontrol->value.integer.value[0] || ucontrol- > >value.integer.value[1]) > + micmute_led = LED_OFF; > + else > + micmute_led = LED_ON; > + ledtrig_audio_set(LED_AUDIO_MICMUTE, micmute_led); > + } > return k_changed; > } > > diff --git a/sound/soc/codecs/rt715.c b/sound/soc/codecs/rt715.c index > 34c3357e943b..490cf9e63c2d 100644 > --- a/sound/soc/codecs/rt715.c > +++ b/sound/soc/codecs/rt715.c > @@ -13,6 +13,7 @@ > #include <linux/init.h> > #include <linux/delay.h> > #include <linux/i2c.h> > +#include <linux/leds.h> > #include <linux/pm_runtime.h> > #include <linux/pm.h> > #include <linux/soundwire/sdw.h> > @@ -25,6 +26,7 @@ > #include <linux/of.h> > #include <linux/of_gpio.h> > #include <linux/of_device.h> > +#include <linux/dmi.h> > #include <sound/core.h> > #include <sound/pcm.h> > #include <sound/pcm_params.h> > @@ -70,6 +72,34 @@ static void rt715_get_gain(struct rt715_priv *rt715, > unsigned int addr_h, > pr_err("Failed to get L channel gain.\n"); } > > +static bool micmute_led_set; > +static int dmi_matched(const struct dmi_system_id *dmi) { > + micmute_led_set = 1; > + return 1; > +} > + > +/* Some systems will need to use this to trigger mic mute LED state > +changed */ static const struct dmi_system_id micmute_led_dmi_table[] = { > + { > + .callback = dmi_matched, > + .ident = "Dell Latitude 9420", > + .matches = { > + DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc."), > + DMI_MATCH(DMI_PRODUCT_NAME, "Latitude > 9420"), > + }, > + }, > + { > + .callback = dmi_matched, > + .ident = "Dell Latitude 9520", > + .matches = { > + DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc."), > + DMI_MATCH(DMI_PRODUCT_NAME, "Latitude > 9520"), > + }, > + }, > + {}, > +}; > + > /* For Verb-Set Amplifier Gain (Verb ID = 3h) */ static int > rt715_set_amp_gain_put(struct snd_kcontrol *kcontrol, > struct snd_ctl_elem_value *ucontrol) > @@ -88,6 +118,7 @@ static int rt715_set_amp_gain_put(struct snd_kcontrol > *kcontrol, > RT715_SET_GAIN_MIX_ADC2_L}; > unsigned int addr_h, addr_l, val_h, val_ll, val_lr; > unsigned int read_ll, read_rl, i, j, loop_cnt; > + bool micmute_led; > > if (strstr(ucontrol->id.name, "Main Capture Switch") || > strstr(ucontrol->id.name, "Main Capture Volume")) @@ - > 185,6 +216,16 @@ static int rt715_set_amp_gain_put(struct snd_kcontrol > *kcontrol, > if (dapm->bias_level <= SND_SOC_BIAS_STANDBY) > regmap_write(rt715->regmap, > RT715_SET_AUDIO_POWER_STATE, > AC_PWRST_D3); > + > + /* Micmute LED state changed by muted/unmute switch */ > + dmi_check_system(micmute_led_dmi_table); > + if (mc->invert && micmute_led_set) { > + if (ucontrol->value.integer.value[0] || ucontrol- > >value.integer.value[1]) > + micmute_led = LED_OFF; > + else > + micmute_led = LED_ON; > + ledtrig_audio_set(LED_AUDIO_MICMUTE, micmute_led); > + } > return 0; > } > > -- > 2.25.1 Could you help to review if the patch is acceptable as short term solution? I added dmi checking when changing the led state for rt715 and rt715-sdca codec. Thank you! Perry

1 0

[ANNOUNCE] SOF v1.7 released
by Girdwood, Liam R 06 Apr '21

06 Apr '21

The SOF developers are pleased to announce that version 1.7 has been released and binaries available for Intel and NXP platforms[1]. The main focus of v1.7 was around adding support for new audio processing algorithms, continued Zephyr integration and general stabilisation. Release notes are here. https://github.com/thesofproject/sof/releases/tag/v1.7 Binaries are here. https://github.com/thesofproject/sof-bin Firmware issues can be reported here (and feature requests too). https://github.com/thesofproject/sof/issues and kernel issues here https://github.com/thesofproject/linux/issues The next version will be v1.8 due by the end of June. Thanks Liam [1] Preliminary support for other platforms can be enabled by using SOF as a Zephyr module (for Zephyr supported architectures and platforms). --------------------------------------------------------------------- Intel Corporation (UK) Limited Registered No. 1134945 (England) Registered Office: Pipers Way, Swindon SN3 1RJ VAT No: 860 2173 47 This e-mail and any attachments may contain confidential material for the sole use of the intended recipient(s). Any review or distribution by others is strictly prohibited. If you are not the intended recipient, please contact the sender and delete all copies.

2 1

[PATCH v2] soundwire: intel_init: test link->cdns
by Bard Liao 06 Apr '21

06 Apr '21

intel_link_probe() could return error and dev_get_drvdata() will return null in such case. So we have to test link->cdns after link->cdns = dev_get_drvdata(&ldev->auxdev.dev); Otherwise, we will meet the "kernel NULL pointer dereference" error. Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Rander Wang <rander.wang(a)intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- v2: - Rebase to latest code base. --- drivers/soundwire/intel_init.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/drivers/soundwire/intel_init.c b/drivers/soundwire/intel_init.c index 05b726cdfebc..30ce95ec2d70 100644 --- a/drivers/soundwire/intel_init.c +++ b/drivers/soundwire/intel_init.c @@ -178,6 +178,15 @@ static struct sdw_intel_ctx link->pdev = pdev; link->cdns = platform_get_drvdata(pdev); + if (!link->cdns) { + dev_err(&adev->dev, "failed to get link->cdns\n"); + /* + * 1 will be subtracted from i in the err label, but we need to call + * intel_link_dev_unregister for this ldev, so plus 1 now + */ + i++; + goto err; + } list_add_tail(&link->list, &ctx->link_list); bus = &link->cdns->bus; /* Calculate number of slaves */ -- 2.17.1

2 1

[PATCH] soundwire: qcom: handle return correctly in qcom_swrm_transport_params
by Srinivas Kandagatla 06 Apr '21

06 Apr '21

Looks like return from reg_write is set but not checked. Fix this by adding error return path. Reported-by: coverity-bot <keescook+coverity-bot(a)chromium.org> Addresses-Coverity-ID: 1503591 ("UNUSED_VALUE") Fixes: 128eaf937adb ("soundwire: qcom: add support to missing transport params") Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- drivers/soundwire/qcom.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/drivers/soundwire/qcom.c b/drivers/soundwire/qcom.c index 5fd4a99cc8ac..348d9a46f850 100644 --- a/drivers/soundwire/qcom.c +++ b/drivers/soundwire/qcom.c @@ -731,17 +731,23 @@ static int qcom_swrm_transport_params(struct sdw_bus *bus, value |= pcfg->si; ret = ctrl->reg_write(ctrl, reg, value); + if (ret) + goto err; if (pcfg->lane_control != SWR_INVALID_PARAM) { reg = SWRM_DP_PORT_CTRL_2_BANK(params->port_num, bank); value = pcfg->lane_control; ret = ctrl->reg_write(ctrl, reg, value); + if (ret) + goto err; } if (pcfg->blk_group_count != SWR_INVALID_PARAM) { reg = SWRM_DP_BLOCK_CTRL2_BANK(params->port_num, bank); value = pcfg->blk_group_count; ret = ctrl->reg_write(ctrl, reg, value); + if (ret) + goto err; } if (pcfg->hstart != SWR_INVALID_PARAM @@ -755,11 +761,15 @@ static int qcom_swrm_transport_params(struct sdw_bus *bus, ret = ctrl->reg_write(ctrl, reg, value); } + if (ret) + goto err; + if (pcfg->bp_mode != SWR_INVALID_PARAM) { reg = SWRM_DP_BLOCK_CTRL3_BANK(params->port_num, bank); ret = ctrl->reg_write(ctrl, reg, pcfg->bp_mode); } +err: return ret; } -- 2.21.0

3 2

[PATCH v2] soundwire: qcom: wait for fifo space to be available before read/write
by Srinivas Kandagatla 06 Apr '21

06 Apr '21

If we write registers very fast we can endup in a situation where some of the writes will be dropped without any notice. So wait for the fifo space to be available before reading/writing the soundwire registers. Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- Changes since v1: merged some of the loop code to make it simple as suggested by Pierre updated error code and comments as suggested by Vinod drivers/soundwire/qcom.c | 66 ++++++++++++++++++++++++++++++++++++++++ 1 file changed, 66 insertions(+) diff --git a/drivers/soundwire/qcom.c b/drivers/soundwire/qcom.c index 6affa3cd4039..5fd4a99cc8ac 100644 --- a/drivers/soundwire/qcom.c +++ b/drivers/soundwire/qcom.c @@ -24,6 +24,8 @@ #define SWRM_COMP_CFG_IRQ_LEVEL_OR_PULSE_MSK BIT(1) #define SWRM_COMP_CFG_ENABLE_MSK BIT(0) #define SWRM_COMP_PARAMS 0x100 +#define SWRM_COMP_PARAMS_WR_FIFO_DEPTH GENMASK(14, 10) +#define SWRM_COMP_PARAMS_RD_FIFO_DEPTH GENMASK(19, 15) #define SWRM_COMP_PARAMS_DOUT_PORTS_MASK GENMASK(4, 0) #define SWRM_COMP_PARAMS_DIN_PORTS_MASK GENMASK(9, 5) #define SWRM_INTERRUPT_STATUS 0x200 @@ -51,6 +53,8 @@ #define SWRM_CMD_FIFO_CMD 0x308 #define SWRM_CMD_FIFO_FLUSH 0x1 #define SWRM_CMD_FIFO_STATUS 0x30C +#define SWRM_RD_CMD_FIFO_CNT_MASK GENMASK(20, 16) +#define SWRM_WR_CMD_FIFO_CNT_MASK GENMASK(12, 8) #define SWRM_CMD_FIFO_CFG_ADDR 0x314 #define SWRM_CONTINUE_EXEC_ON_CMD_IGNORE BIT(31) #define SWRM_RD_WR_CMD_RETRIES 0x7 @@ -104,6 +108,7 @@ #define SWR_BROADCAST_CMD_ID 0x0F #define SWR_MAX_CMD_ID 14 #define MAX_FIFO_RD_RETRY 3 +#define SWR_OVERFLOW_RETRY_COUNT 30 struct qcom_swrm_port_config { u8 si; @@ -147,6 +152,8 @@ struct qcom_swrm_ctrl { int (*reg_read)(struct qcom_swrm_ctrl *ctrl, int reg, u32 *val); int (*reg_write)(struct qcom_swrm_ctrl *ctrl, int reg, int val); u32 slave_status; + u32 wr_fifo_depth; + u32 rd_fifo_depth; }; struct qcom_swrm_data { @@ -238,6 +245,55 @@ static u32 swrm_get_packed_reg_val(u8 *cmd_id, u8 cmd_data, return val; } +static int swrm_wait_for_rd_fifo_avail(struct qcom_swrm_ctrl *swrm) +{ + u32 fifo_outstanding_data, value; + int fifo_retry_count = SWR_OVERFLOW_RETRY_COUNT; + + do { + /* Check for fifo underflow during read */ + swrm->reg_read(swrm, SWRM_CMD_FIFO_STATUS, &value); + fifo_outstanding_data = FIELD_GET(SWRM_RD_CMD_FIFO_CNT_MASK, value); + + /* Check if read data is available in read fifo */ + if (fifo_outstanding_data > 0) + return 0; + + usleep_range(500, 510); + } while (fifo_retry_count--); + + if (fifo_outstanding_data == 0) { + dev_err_ratelimited(swrm->dev, "%s err read underflow\n", __func__); + return -EIO; + } + + return 0; +} + +static int swrm_wait_for_wr_fifo_avail(struct qcom_swrm_ctrl *swrm) +{ + u32 fifo_outstanding_cmds, value; + int fifo_retry_count = SWR_OVERFLOW_RETRY_COUNT; + + do { + /* Check for fifo overflow during write */ + swrm->reg_read(swrm, SWRM_CMD_FIFO_STATUS, &value); + fifo_outstanding_cmds = FIELD_GET(SWRM_WR_CMD_FIFO_CNT_MASK, value); + + /* Check for space in write fifo before writing */ + if (fifo_outstanding_cmds < swrm->wr_fifo_depth) + return 0; + + usleep_range(500, 510); + } while (fifo_retry_count--); + + if (fifo_outstanding_cmds == swrm->wr_fifo_depth) { + dev_err_ratelimited(swrm->dev, "%s err write overflow\n", __func__); + return -EIO; + } + + return 0; +} static int qcom_swrm_cmd_fifo_wr_cmd(struct qcom_swrm_ctrl *swrm, u8 cmd_data, u8 dev_addr, u16 reg_addr) @@ -256,6 +312,9 @@ static int qcom_swrm_cmd_fifo_wr_cmd(struct qcom_swrm_ctrl *swrm, u8 cmd_data, dev_addr, reg_addr); } + if (swrm_wait_for_wr_fifo_avail(swrm)) + return SDW_CMD_FAIL_OTHER; + /* Its assumed that write is okay as we do not get any status back */ swrm->reg_write(swrm, SWRM_CMD_FIFO_WR_CMD, val); @@ -295,6 +354,9 @@ static int qcom_swrm_cmd_fifo_rd_cmd(struct qcom_swrm_ctrl *swrm, /* wait for FIFO RD CMD complete to avoid overflow */ usleep_range(250, 255); + if (swrm_wait_for_rd_fifo_avail(swrm)) + return SDW_CMD_FAIL_OTHER; + do { swrm->reg_read(swrm, SWRM_CMD_FIFO_RD_FIFO_ADDR, &cmd_data); rval[0] = cmd_data & 0xFF; @@ -586,6 +648,10 @@ static int qcom_swrm_init(struct qcom_swrm_ctrl *ctrl) SWRM_INTERRUPT_STATUS_RMSK); } ctrl->slave_status = 0; + ctrl->reg_read(ctrl, SWRM_COMP_PARAMS, &val); + ctrl->rd_fifo_depth = FIELD_GET(SWRM_COMP_PARAMS_RD_FIFO_DEPTH, val); + ctrl->wr_fifo_depth = FIELD_GET(SWRM_COMP_PARAMS_WR_FIFO_DEPTH, val); + return 0; } -- 2.21.0

3 3

[PATCH] soundwire: qcom: cleanup internal port config indexing
by Srinivas Kandagatla 06 Apr '21

06 Apr '21

Internally used portconfig array for storing port bandwidth params starts from offset zero. However port zero is not really used and we also copy the bus parameters to offset zero. So basically we endup with a code which has to subtract 1 from port number to get to port parameters. This is bit confusing to the reader so, make this bit more obvious by only copying the parameters to offset 1 instead of zero. This will avoid doing -1 every time when we try to get port params. Similar thing has been recently done with din/dout_port_mask. Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- drivers/soundwire/qcom.c | 27 ++++++++++++++------------- 1 file changed, 14 insertions(+), 13 deletions(-) diff --git a/drivers/soundwire/qcom.c b/drivers/soundwire/qcom.c index 94fd58f9dda3..348d9a46f850 100644 --- a/drivers/soundwire/qcom.c +++ b/drivers/soundwire/qcom.c @@ -724,7 +724,7 @@ static int qcom_swrm_transport_params(struct sdw_bus *bus, int reg = SWRM_DP_PORT_CTRL_BANK((params->port_num), bank); int ret; - pcfg = &ctrl->pconfig[params->port_num - 1]; + pcfg = &ctrl->pconfig[params->port_num]; value = pcfg->off1 << SWRM_DP_PORT_CTRL_OFFSET1_SHFT; value |= pcfg->off2 << SWRM_DP_PORT_CTRL_OFFSET2_SHFT; @@ -811,11 +811,11 @@ static int qcom_swrm_compute_params(struct sdw_bus *bus) struct qcom_swrm_port_config *pcfg; struct sdw_slave *slave; unsigned int m_port; - int i = 0; + int i = 1; list_for_each_entry(m_rt, &bus->m_rt_list, bus_node) { list_for_each_entry(p_rt, &m_rt->port_list, port_node) { - pcfg = &ctrl->pconfig[p_rt->num - 1]; + pcfg = &ctrl->pconfig[p_rt->num]; p_rt->transport_params.port_num = p_rt->num; if (pcfg->word_length != SWR_INVALID_PARAM) { sdw_fill_port_params(&p_rt->port_params, @@ -832,7 +832,7 @@ static int qcom_swrm_compute_params(struct sdw_bus *bus) m_port = slave->m_port_map[p_rt->num]; /* port config starts at offset 0 so -1 from actual port number */ if (m_port) - pcfg = &ctrl->pconfig[m_port - 1]; + pcfg = &ctrl->pconfig[m_port]; else pcfg = &ctrl->pconfig[i]; p_rt->transport_params.port_num = p_rt->num; @@ -1167,15 +1167,16 @@ static int qcom_swrm_get_port_config(struct qcom_swrm_ctrl *ctrl) of_property_read_u8_array(np, "qcom,ports-lane-control", lane_control, nports); for (i = 0; i < nports; i++) { - ctrl->pconfig[i].si = si[i]; - ctrl->pconfig[i].off1 = off1[i]; - ctrl->pconfig[i].off2 = off2[i]; - ctrl->pconfig[i].bp_mode = bp_mode[i]; - ctrl->pconfig[i].hstart = hstart[i]; - ctrl->pconfig[i].hstop = hstop[i]; - ctrl->pconfig[i].word_length = word_length[i]; - ctrl->pconfig[i].blk_group_count = blk_group_count[i]; - ctrl->pconfig[i].lane_control = lane_control[i]; + /* Valid port number range is from 1-14 */ + ctrl->pconfig[i + 1].si = si[i]; + ctrl->pconfig[i + 1].off1 = off1[i]; + ctrl->pconfig[i + 1].off2 = off2[i]; + ctrl->pconfig[i + 1].bp_mode = bp_mode[i]; + ctrl->pconfig[i + 1].hstart = hstart[i]; + ctrl->pconfig[i + 1].hstop = hstop[i]; + ctrl->pconfig[i + 1].word_length = word_length[i]; + ctrl->pconfig[i + 1].blk_group_count = blk_group_count[i]; + ctrl->pconfig[i + 1].lane_control = lane_control[i]; } return 0; -- 2.21.0

2 1

[PATCH] ALSA: core: control_led: fix memory leak in snd_ctl_led_set_state()
by Takashi Sakamoto 06 Apr '21

06 Apr '21

When inquired control element is not in led group, snd_ctl_led_set_state() attempts to add the element into the group, according to function arguments. Although an memory object is allocated for led instance, it's left as is without being released. This commit fixes the memory leak. Fixes: 22d8de62f11b ("ALSA: control - add generic LED trigger module as the new control layer") Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/core/control_led.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/core/control_led.c b/sound/core/control_led.c index 788fd9e275e0..b97f118cd54e 100644 --- a/sound/core/control_led.c +++ b/sound/core/control_led.c @@ -161,6 +161,7 @@ static void snd_ctl_led_set_state(struct snd_card *card, unsigned int access, list_add(&lctl->list, &led->controls); UPDATE_ROUTE(route, snd_ctl_led_get(lctl)); } + kfree(lctl); } mutex_unlock(&snd_ctl_led_mutex); switch (led->mode) { -- 2.27.0

2 2

[PATCH v2 0/2] kbl_da7219_max9357a machine changes for wov and MST
by vamshi.krishna.gopal＠intel.com 05 Apr '21

05 Apr '21

From: Vamshi Krishna Gopal <vamshi.krishna.gopal(a)intel.corp-partner.google.com> Hi all, This patch series about creating dailink for Wake on voice functionality and also adding MST route changes. changes in v2 - Add MST route changes to kbl_da7219_max98927.c and kabl_rt5663_max98927.c - typo corrected in struct skylaye_refcap_ops to skylake_refcap_ops - Declare constraints_1600 struct and ch_mono variable v1 is here: https://patchwork.kernel.org/project/alsa-devel/cover/20210324175200.44922-… Mac Chiang (1): ASoc: Intel: board: add BE DAI link for WoV Vamshi Krishna Gopal (1): ASoC: Intel: kbl: Add MST route change to kbl machine drivers sound/soc/intel/boards/kbl_da7219_max98357a.c | 57 ++++++++++++++++++- sound/soc/intel/boards/kbl_da7219_max98927.c | 9 ++- sound/soc/intel/boards/kbl_rt5663_max98927.c | 4 ++ 3 files changed, 64 insertions(+), 6 deletions(-) -- 2.17.1

3 4

Front headphones jack not reporting state to Pulseaudio correctly (always unplugged)
by GitHub issues - opened 05 Apr '21

05 Apr '21

alsa-project/alsa-lib issue #130 was opened from jose1711: ALSA seems to always report state of front headphones jack as "unplugged" to PulseAudio (related bug report: https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/1180). [alsa-info.txt](https://github.com/alsa-project/alsa-lib/files/6255379/alsa-… Issue URL : https://github.com/alsa-project/alsa-lib/issues/130 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH] Pioneer devices: engage implicit feedback sync for playback
by Geraldo 05 Apr '21

05 Apr '21

Hello everybody, This is a highly experimental patch on top of 5.12-rc4 that's supposed to engage implicit feedback sync on the playback for Pioneer devices. Without implicit feedback sync the inputs started glitching due to clock drift in about an hour in my Pioneer DJ DDJ-SR2. If you own a Pioneer device please test this patch and verify in the dyndbg logs if your capture(input) EP is being opened twice and if implicit_fb = 1 for the playback EP. I can't ask of you all to test this for hours and hours but if you wanted to have an excuse to digitize vinyl or spin up DVS this is your perfect one. Help open source! I'd like to thank Takashi Iwai for his tireless support in the Pioneer regression and beyond. Hopefully this highly experimental feature's code can be polished and mainlined eventually. --- implicit.c.git 2021-03-21 22:40:48.363417245 -0300 +++ implicit.c 2021-03-22 01:03:05.726729481 -0300 @@ -177,30 +177,31 @@ ifnum, alts); } -/* Playback and capture EPs on Pioneer devices share the same iface/altset, - * but they don't seem working with the implicit fb mode well, hence we - * just return as if the sync were already set up. +/* Pioneer devices: playback and capture streams sharing the same iface/altset */ -static int skip_pioneer_sync_ep(struct snd_usb_audio *chip, - struct audioformat *fmt, - struct usb_host_interface *alts) -{ - struct usb_endpoint_descriptor *epd; - - if (alts->desc.bNumEndpoints != 2) - return 0; - - epd = get_endpoint(alts, 1); - if (!usb_endpoint_is_isoc_in(epd) || - (epd->bmAttributes & USB_ENDPOINT_SYNCTYPE) != USB_ENDPOINT_SYNC_ASYNC || - ((epd->bmAttributes & USB_ENDPOINT_USAGE_MASK) != - USB_ENDPOINT_USAGE_DATA && - (epd->bmAttributes & USB_ENDPOINT_USAGE_MASK) != - USB_ENDPOINT_USAGE_IMPLICIT_FB)) - return 0; - return 1; /* don't handle with the implicit fb, just skip sync EP */ +static int add_pioneer_implicit_fb(struct snd_usb_audio *chip, + struct audioformat *fmt, + struct usb_host_interface *alts) +{ + struct usb_endpoint_descriptor *epd; + + if (alts->desc.bNumEndpoints != 2) + return 0; + + epd = get_endpoint(alts, 1); + + if (!usb_endpoint_is_isoc_in(epd) || + (epd->bmAttributes & USB_ENDPOINT_SYNCTYPE) != USB_ENDPOINT_SYNC_ASYNC || + ((epd->bmAttributes & USB_ENDPOINT_USAGE_MASK) != + USB_ENDPOINT_USAGE_DATA && + (epd->bmAttributes & USB_ENDPOINT_USAGE_MASK) != + USB_ENDPOINT_USAGE_IMPLICIT_FB)) + return 0; + return add_implicit_fb_sync_ep(chip, fmt, epd->bEndpointAddress, 1, + alts->desc.bInterfaceNumber, alts); } + static int __add_generic_implicit_fb(struct snd_usb_audio *chip, struct audioformat *fmt, int iface, int altset) @@ -306,7 +307,7 @@ alts->desc.bInterfaceClass == USB_CLASS_VENDOR_SPEC && (USB_ID_VENDOR(chip->usb_id) == 0x2b73 || /* Pioneer */ USB_ID_VENDOR(chip->usb_id) == 0x08e4 /* Pioneer */)) { - if (skip_pioneer_sync_ep(chip, fmt, alts)) + if (add_pioneer_implicit_fb(chip, fmt, alts)) return 1; } @@ -339,8 +340,20 @@ struct audioformat *fmt, struct usb_host_interface *alts) { - if (fmt->endpoint & USB_DIR_IN) - return audioformat_capture_quirk(chip, fmt, alts); + bool isPioneer; + + if (alts->desc.bInterfaceClass == USB_CLASS_VENDOR_SPEC && + (USB_ID_VENDOR(chip->usb_id) == 0x2b73 || /* Pioneer */ + USB_ID_VENDOR(chip->usb_id) == 0x08e4 /* Pioneer */)) + isPioneer = true; + + if (fmt->endpoint & USB_DIR_IN) { + if (isPioneer == true) + return 1; + else + return audioformat_capture_quirk(chip, fmt, alts); + } + else return audioformat_implicit_fb_quirk(chip, fmt, alts); } --- endpoint.c.git 2021-03-21 22:40:54.800084101 -0300 +++ endpoint.c 2021-03-21 22:45:41.543425855 -0300 @@ -688,7 +688,7 @@ goto unlock; } - if (!ep->opened) { + if (ep->opened < 2) { if (is_sync_ep) { ep->iface = fp->sync_iface; ep->altsetting = fp->sync_altsetting;

1 3

ALC1220 and presence of ucm2/ucm.conf breaks audio
by GitHub issues - opened 04 Apr '21

04 Apr '21

alsa-project/alsa-ucm-conf issue #86 was opened from devnull0000: I'd want to make a better report but I am not an ALSA expert, so the issue is that after update I lose any sound and I use ALC1220 codec on MSI MEG X570 GODLIKE motherboard. Deep googling gave me the report: https://bugs.archlinux.org/task/67058 So indeed deletion of the file helped immediately. The bug is super confusing because it looks like the device is present at alsa level but absent on Pulse Audio and it is not shown in GUI at all. I am at alsa-ucm-conf-1.2.4, linux-5.11.11, pulseaudio 13.0, gentoo. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/86 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH v6 0/2] hardware-privacy-implementation-for-dell-laptop
by Perry Yuan 04 Apr '21

04 Apr '21

From: Perry Yuan <perry_yuan(a)dell.com> Hi All, This patch set is a new driver for Dell mobile platform that has the hardware privacy feature including micmute,camera mute. For micmute led control, the hotkey is Fn + F4, it is a hardware based mute state, and new privacy will prevent micphone void input from hardware layer, any application cannot get voice data when micmute activated. Camera mute use a new hardware design to control the camrea shutter. When video is muted, no OS application should be functionally able to capture images of the person/environment in front of the system Older history: [1]https://patchwork.kernel.org/project/platform-driver-x86/patch/2020122813… [2]https://patchwork.kernel.org/project/alsa-devel/patch/20201103125859.8759… [3]https://www.spinics.net/lists/alsa-devel/msg120537.html [4]https://github.com/thesofproject/linux/pull/2660 [5]https://github.com/thesofproject/linux/issues/2496 Perry Yuan (2): platform/x86: dell-privacy: Add support for Dell hardware privacy ASoC: rt715:add micmute led state control supports .../testing/sysfs-platform-dell-privacy-wmi | 55 +++ drivers/platform/x86/dell/Kconfig | 14 + drivers/platform/x86/dell/Makefile | 1 + drivers/platform/x86/dell/dell-laptop.c | 23 +- drivers/platform/x86/dell/dell-privacy-wmi.c | 441 ++++++++++++++++++ drivers/platform/x86/dell/dell-privacy-wmi.h | 32 ++ drivers/platform/x86/dell/dell-wmi.c | 13 +- sound/soc/codecs/rt715-sdca.c | 41 +- sound/soc/codecs/rt715.c | 41 ++ 9 files changed, 653 insertions(+), 8 deletions(-) create mode 100644 Documentation/ABI/testing/sysfs-platform-dell-privacy-wmi create mode 100644 drivers/platform/x86/dell/dell-privacy-wmi.c create mode 100644 drivers/platform/x86/dell/dell-privacy-wmi.h -- 2.25.1

1 0

[PATCH] ALSA: control_led: use helper macro to declare device attributes with specific permission
by Takashi Sakamoto 04 Apr '21

04 Apr '21

It's preferable to use DEVICE_ATTR_XXX() for device attributes with specific permission instead of DEVICE_ATTR() basis macro. Fixes: 22d8de62f11b ("ALSA: control - add generic LED trigger module as the new control layer") Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/core/control_led.c | 37 +++++++++++++++++-------------------- 1 file changed, 17 insertions(+), 20 deletions(-) diff --git a/sound/core/control_led.c b/sound/core/control_led.c index 788fd9e275e0..99c5d503638f 100644 --- a/sound/core/control_led.c +++ b/sound/core/control_led.c @@ -371,8 +371,7 @@ static void snd_ctl_led_disconnect(struct snd_card *card) * sysfs */ -static ssize_t show_mode(struct device *dev, - struct device_attribute *attr, char *buf) +static ssize_t mode_show(struct device *dev, struct device_attribute *attr, char *buf) { struct snd_ctl_led *led = container_of(dev, struct snd_ctl_led, dev); const char *str; @@ -386,8 +385,8 @@ static ssize_t show_mode(struct device *dev, return sprintf(buf, "%s\n", str); } -static ssize_t store_mode(struct device *dev, struct device_attribute *attr, - const char *buf, size_t count) +static ssize_t mode_store(struct device *dev, struct device_attribute *attr, const char *buf, + size_t count) { struct snd_ctl_led *led = container_of(dev, struct snd_ctl_led, dev); char _buf[16]; @@ -415,16 +414,15 @@ static ssize_t store_mode(struct device *dev, struct device_attribute *attr, return count; } -static ssize_t show_brightness(struct device *dev, - struct device_attribute *attr, char *buf) +static ssize_t brightness_show(struct device *dev, struct device_attribute *attr, char *buf) { struct snd_ctl_led *led = container_of(dev, struct snd_ctl_led, dev); return sprintf(buf, "%u\n", ledtrig_audio_get(led->trigger_type)); } -static DEVICE_ATTR(mode, 0644, show_mode, store_mode); -static DEVICE_ATTR(brightness, 0444, show_brightness, NULL); +static DEVICE_ATTR_RW(mode); +static DEVICE_ATTR_RO(brightness); static struct attribute *snd_ctl_led_dev_attrs[] = { &dev_attr_mode.attr, @@ -558,22 +556,22 @@ static ssize_t set_led_id(struct snd_ctl_led_card *led_card, const char *buf, si return count; } -static ssize_t parse_attach(struct device *dev, struct device_attribute *attr, - const char *buf, size_t count) +static ssize_t attach_store(struct device *dev, struct device_attribute *attr, const char *buf, + size_t count) { struct snd_ctl_led_card *led_card = container_of(dev, struct snd_ctl_led_card, dev); return set_led_id(led_card, buf, count, true); } -static ssize_t parse_detach(struct device *dev, struct device_attribute *attr, - const char *buf, size_t count) +static ssize_t detach_store(struct device *dev, struct device_attribute *attr, const char *buf, + size_t count) { struct snd_ctl_led_card *led_card = container_of(dev, struct snd_ctl_led_card, dev); return set_led_id(led_card, buf, count, false); } -static ssize_t ctl_reset(struct device *dev, struct device_attribute *attr, - const char *buf, size_t count) +static ssize_t reset_store(struct device *dev, struct device_attribute *attr, const char *buf, + size_t count) { struct snd_ctl_led_card *led_card = container_of(dev, struct snd_ctl_led_card, dev); int err; @@ -586,8 +584,7 @@ static ssize_t ctl_reset(struct device *dev, struct device_attribute *attr, return count; } -static ssize_t ctl_list(struct device *dev, - struct device_attribute *attr, char *buf) +static ssize_t list_show(struct device *dev, struct device_attribute *attr, char *buf) { struct snd_ctl_led_card *led_card = container_of(dev, struct snd_ctl_led_card, dev); struct snd_card *card; @@ -620,10 +617,10 @@ static ssize_t ctl_list(struct device *dev, return buf2 - buf; } -static DEVICE_ATTR(attach, 0200, NULL, parse_attach); -static DEVICE_ATTR(detach, 0200, NULL, parse_detach); -static DEVICE_ATTR(reset, 0200, NULL, ctl_reset); -static DEVICE_ATTR(list, 0444, ctl_list, NULL); +static DEVICE_ATTR_WO(attach); +static DEVICE_ATTR_WO(detach); +static DEVICE_ATTR_WO(reset); +static DEVICE_ATTR_RO(list); static struct attribute *snd_ctl_led_card_attrs[] = { &dev_attr_attach.attr, -- 2.27.0

1 0

[PATCH alsa-ucm-conf 0/1] cht-bsw-rt5672: Add support for the components string
by Hans de Goede 04 Apr '21

04 Apr '21

Hi All, Add support for using the card's components string to figure out the DMIC and speaker configuration so that we can stop duplicating the kernel's DMI quirks inside the UCM profile. The kernel side of this was just merged: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git/commit/?h… Regards, Hans Hans de Goede (1): cht-bsw-rt5672: Add support for the components string ucm2/cht-bsw-rt5672/HiFi.conf | 37 +++++++++++++++++++++++++++++++++-- 1 file changed, 35 insertions(+), 2 deletions(-) -- 2.30.2

1 1

[PATCH] ASoC: max98390: Add controls for tx path
by Steve Lee 03 Apr '21

03 Apr '21

Add controls for tx source. Signed-off-by: Steve Lee <steves.lee(a)maximintegrated.com> --- sound/soc/codecs/max98390.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/soc/codecs/max98390.c b/sound/soc/codecs/max98390.c index bb736c44e68a..163093959da8 100644 --- a/sound/soc/codecs/max98390.c +++ b/sound/soc/codecs/max98390.c @@ -656,6 +656,12 @@ static const struct snd_kcontrol_new max98390_snd_controls[] = { MAX98390_AMP_DSP_CFG_RMP_DN_SHIFT, 1, 0), SOC_SINGLE("Boost Clock Phase", MAX98390_BOOST_CTRL3, MAX98390_BOOST_CLK_PHASE_CFG_SHIFT, 3, 0), + SOC_SINGLE("Tx Enable Selection", MAX98390_PCM_TX_EN_A, + 0, 255, 0), + SOC_SINGLE("Tx Hiz Selection", MAX98390_PCM_TX_HIZ_CTRL_A, + 0, 255, 0), + SOC_SINGLE("Tx Source Selection", MAX98390_PCM_CH_SRC_2, + 0, 255, 0), SOC_ENUM("Boost Output Voltage", max98390_boost_voltage), SOC_ENUM("Current Limit", max98390_current_limit), SOC_SINGLE_EXT("DSM Rdc", SND_SOC_NOPM, 0, 0xffffff, 0, -- 2.17.1

2 2

[PATCH 0/6] ASoC: rt5670/Intel: Add support for Bay Trail CR / Dell Venue 10 Pro 5055 tablet
by Hans de Goede 02 Apr '21

02 Apr '21

Hi All, Here is a patch series to enable sound on the Dell Venue 10 Pro 5055 tablet this series also includes 2 generic improvements to rt5672 support: 1. Add support for boards using Bay Trail CR (SSP0) in combination with a rt5672 codec 2. Set the card's components string based on the DMI info, so that we don't need to duplicate the DMI quirks in userspace (in the UCM profile) I've a patch to add support for the components string to the UCM profile for these boards available here: https://github.com/jwrdegoede/alsa-ucm-conf/commit/fedacc2bdc530b6febac665d… I will submit this upstream once the kernel side has landed. Regards, Hans Hans de Goede (6): ASoC: Intel: Unify the thinkpad10 and aegex10 byt-match quirks ASoC: Intel: Baytrail: Add quirk for the Dell Venue 10 Pro 5055 tablet ASoC: Intel: cht_bsw_rt5672: Add support for Bay Trail CR / SSP0 ASoC: rt5670: Add a quirk for the Dell Venue 10 Pro 5055 ASoC: rt5670: Add a rt5670_components() helper ASoC: Intel: cht_bsw_rt5672: Set card.components string sound/soc/codecs/rt5670.c | 51 ++++++++++++ sound/soc/codecs/rt5670.h | 2 + sound/soc/intel/boards/cht_bsw_rt5672.c | 77 +++++++++++++++---- .../intel/common/soc-acpi-intel-byt-match.c | 40 +++++----- 4 files changed, 136 insertions(+), 34 deletions(-) -- 2.30.2

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[PATCH] ASoC: fsl_esai: Fix TDM slot setup for I2S mode
by Alexander Shiyan 02 Apr '21

02 Apr '21

When using the driver in I2S TDM mode, the fsl_esai_startup() function rewrites the number of slots previously set by the fsl_esai_set_dai_tdm_slot() function to 2. To fix this, let's use the saved slot count value or, if TDM is not used and the number of slots is not set, the driver will use the default value (2), which is set by fsl_esai_probe(). Signed-off-by: Alexander Shiyan <shc_work(a)mail.ru> --- sound/soc/fsl/fsl_esai.c | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index 08056fa0a0fa..a857a624864f 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -519,11 +519,13 @@ static int fsl_esai_startup(struct snd_pcm_substream *substream, ESAI_SAICR_SYNC, esai_priv->synchronous ? ESAI_SAICR_SYNC : 0); - /* Set a default slot number -- 2 */ + /* Set slots count */ regmap_update_bits(esai_priv->regmap, REG_ESAI_TCCR, - ESAI_xCCR_xDC_MASK, ESAI_xCCR_xDC(2)); + ESAI_xCCR_xDC_MASK, + ESAI_xCCR_xDC(esai_priv->slots)); regmap_update_bits(esai_priv->regmap, REG_ESAI_RCCR, - ESAI_xCCR_xDC_MASK, ESAI_xCCR_xDC(2)); + ESAI_xCCR_xDC_MASK, + ESAI_xCCR_xDC(esai_priv->slots)); } return 0; -- 2.26.2

3 2

[PATCH 0/2] soundwire: bus: handle errors in clock stop/start sequences
by Bard Liao 02 Apr '21

02 Apr '21

If a device lost sync and can no longer ACK a command, it may not be able to enter a lower-power state but it will still be able to resync when the clock restarts. In those cases, we want to continue with the clock stop sequence. This patch modifies the behavior when -ENODATA is received, with the error level demoted to a dev_dbg() since it's a recoverable issue. For consistency the log messages are also modified to be unique and self-explanatory, and missing logs are also added on clock stop exit. Pierre-Louis Bossart (2): soundwire: add macro to selectively change error levels soundwire: bus: handle errors in clock stop/start sequences drivers/soundwire/bus.c | 70 +++++++++++++++++++---------------------- drivers/soundwire/bus.h | 8 +++++ 2 files changed, 40 insertions(+), 38 deletions(-) -- 2.17.1

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[PATCH] ASoC: Intel: Boards: cml_da7219_max98390: add capture stream for echo reference
by mac.chiang＠intel.com 01 Apr '21

01 Apr '21

From: Mac Chiang <mac.chiang(a)intel.com> enable speaker capture dai link for feedback path Signed-off-by: Mac Chiang <mac.chiang(a)intel.com> --- sound/soc/intel/boards/bxt_da7219_max98357a.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/intel/boards/bxt_da7219_max98357a.c b/sound/soc/intel/boards/bxt_da7219_max98357a.c index 0c0a7178..9ffef39 100644 --- a/sound/soc/intel/boards/bxt_da7219_max98357a.c +++ b/sound/soc/intel/boards/bxt_da7219_max98357a.c @@ -813,6 +813,7 @@ static int broxton_audio_probe(struct platform_device *pdev) if (ctx->spkamp == SPKAMP_MAX98390) { broxton_dais[i].codecs = max98390_codec; broxton_dais[i].num_codecs = ARRAY_SIZE(max98390_codec); + broxton_dais[i].dpcm_capture = 1; } } /* DIALOG_CODEC is connected to SSP0 */ -- 2.7.4

3 2

[PATCH] Fixing most Roland-related devices' USB audio
by Lucas 01 Apr '21

01 Apr '21

At the risk of being redundant, I'm resending this with a simpler "[PATCH]" subject to reach a broader audience, since no comment has been made yet: TLDR? Please just glance at the patch at the end and include it in the kernel. :) Some days ago, I found this excellent informational post, and that Mike Oliphant's patch is in mainline now: http://blog.nostatic.org/2020/01/getting-boss-gt-1-effects-processor-to.html Since I was fairly certain it applies to the Roland devices I own, I added my USB IDs. Wonderfully, now USB audio capture seems to be working perfectly from the three devices I can test: the Boutique D-05, INTEGRA-7, & R-26! (The previously asserted VG-99 didn't need to be included in the patch for it to work.) I've also tested all three for playback capability, since that's implied when a device is using implicit feedback, which I hadn't realized the first time around. Anyway, they all seem to be working as expected for playback too. There are two caveats for playback, though: 1. The INTEGRA-7 won't output playback in any mode other than its "44.1 [kHz]" setting, though capture works with all of its Sample Rate possibilities. I think this is imposed by the hardware itself, and hence isn't a glitch. 2. The D-05's analog outputs aren't as clean as the other two devices, with little quiet occasional clicks sounding something like dust on an LP record's play. Possibly that can be due to it having only 1/8" mini-jack connections, but I suspect something more driver-related as It's not noticed during use as a synthesizer. Its firmware version is "1.02(1033)," so not up-to-date which may mean it's just the hardware itself, but I've decided not to risk any changes with it yet. If I knew I could revert it, that would be another story. Also its USB captures, play perfectly & cleanly on my usual sound device, the Edirol UA-4FX. You may know this, as the detection of these devices shows partial compatibility with the existing kernel, but here are the commands I've had to use for their sample formats: Boutique D-05: arecord -D hw:Boutique -f S32_LE -c 2 -r 96000 ./file.wav aplay -D hw:Boutique -f S32_LE -c 2 -r 96000 ./file.wav INTEGRA-7: arecord -D hw:INTEGRA7 -f S32_LE -c 2 -r 96000 ./file.wav (Then change MENU->SYSTEM->SOUND->Sampling Rate to 44.1 [kHz], otherwise its 96[kHz], and change MENU->SYSTEM->SOUND->Ext Part Source Select to USB AUDIO) arecord -D hw:INTEGRA7 -f S32_LE -c 2 -r 44100 ./file-44100.wav aplay -D hw:INTEGRA7 -f S32_LE -c 2 -r 44100 ./file-44100.wav R-26: (Through its built-in controls, first configure the MENU->AUDIO I/F SETUP->SAMPLE RATE to "96.0 kHz.") arecord -D hw:R26AUDIO -f S32_LE -r 96000 -c 2 ./file.wav aplay -D hw:R26AUDIO -f S32_LE -r 96000 -c 2 ./file.wav (Then via use of similar commands, & the "AUDIO I/F" tab on its main screen having both "LOOP BACK" & "DIRECT MONITOR" enabled, I was able to begin a new composite recording, voicing-over what I'd recorded earlier, by playing it back while interjecting - very fun!) The behavior on the vanilla kernel (whether or not "pasuspender -- cat" is running concurrently) follows. For each of these commands, dmesg is bombarded with lines like "[ 4681.162863] usb 2-1.2: Unable to change format on ep #8e: already in use" Boutique D-05: arecord -D hw:Boutique -f S32_LE -r 96000 -c 2 ./file.wav Recording WAVE './file.wav' : Signed 32 bit Little Endian, Rate 96000 Hz, Stereo (After a brief timeout, it says this & ends:) arecord: pcm_read:2153: read error: Input/output error aplay -D hw:Boutique -f S32_LE -r 96000 -c 2 ./valid-file.wav Playing WAVE './valid-file.wav' : Signed 32 bit Little Endian, Rate 96000 Hz, Stereo (After a brief timeout, it says this & ends:) aplay: pcm_write:2061: write error: Input/output error INTEGRA-7: arecord -D hw:INTEGRA7 -f S32_LE -r 96000 -c 2 ./file.wav Recording WAVE './file.wav' : Signed 32 bit Little Endian, Rate 96000 Hz, Stereo (After a brief timeout, it says this & ends:) arecord: pcm_read:2153: read error: Input/output error (Then change MENU->SYSTEM->SOUND->Sampling Rate to 44.1 [kHz], otherwise its 96[kHz], and change MENU->SYSTEM->SOUND->Ext Part Source Select to USB AUDIO) arecord -D hw:INTEGRA7 -f S32_LE -r 44100 -c 2 ./file.wav Recording WAVE './file.wav' : Signed 32 bit Little Endian, Rate 44100 Hz, Stereo (After a brief timeout, it says this & ends:) arecord: pcm_read:2153: read error: Input/output error aplay -D hw:INTEGRA7 -f S32_LE -r 44100 -c 2 ./valid-file.wav Playing WAVE './valid-file.wav' : Signed 32 bit Little Endian, Rate 44100 Hz, Stereo (After a brief timeout, it says this & ends:) aplay: pcm_write:2061: write error: Input/output error R-26: arecord -D hw:R26AUDIO -f S32_LE -r 96000 -c 2 ./file.wav Recording WAVE './file.wav' : Signed 32 bit Little Endian, Rate 96000 Hz, Stereo (After a brief timeout, it says this & ends:) arecord: pcm_read:2153: read error: Input/output error aplay -D hw:R26AUDIO -f S32_LE -r 96000 -c 2 ./valid-file.wav Playing WAVE './valid-file.wav' : Signed 32 bit Little Endian, Rate 96000 Hz, Stereo (After a brief timeout, it says this & ends:) aplay: pcm_write:2061: write error: Input/output error I searched a bit for other Roland-related devices people have had USB digital audio issues with over the years, and took the liberty of adding them to the patch too. Likely there are still more that people haven't tried or, at least, queried about issues. I hope this can be added to the mainline kernel, as this support should help many that have struggled. I, certainly, have been hoping somebody would figure this out for quite a while, and even bought a USB through/monitoring device to try to develop it myself (sadly, the software supporting that has its own bugs, so I probably should've just used wireshark). Suffice it to say, I'm quite happy to have this figured out without having had to delve into USB debugging stuff I lack experience with! :) Thanks for any comments or looking into adding this, Lucas Endres The patch follows (I removed my previous VG-99 additions since it already works): diff -Nurp linux-5.11.9.orig/sound/usb/implicit.c linux-5.11.9.roland/sound/usb/implicit.c --- linux-5.11.9.orig/sound/usb/implicit.c 2021-03-24 05:54:19.000000000 -0500 +++ linux-5.11.9.roland/sound/usb/implicit.c 2021-03-30 17:49:08.143196280 -0500 @@ -71,27 +71,87 @@ static const struct snd_usb_implicit_fb_ .ep_num = 0x84, .iface = 0 }, /* MOTU MicroBook II */ /* No quirk for playback but with capture quirk (see below) */ + IMPLICIT_FB_SKIP_DEV(0x0582, 0x00a6), /* Roland JUNO-G */ + IMPLICIT_FB_SKIP_DEV(0x0582, 0x00ad), /* Roland SH-201 */ + IMPLICIT_FB_SKIP_DEV(0x0582, 0x00c2), /* Roland SonicCell */ + IMPLICIT_FB_SKIP_DEV(0x0582, 0x00c4), /* Edirol M-16DX */ + IMPLICIT_FB_SKIP_DEV(0x0582, 0x00c7), /* Roland V-Synth GT */ + IMPLICIT_FB_SKIP_DEV(0x0582, 0x00da), /* BOSS GT-10 */ + IMPLICIT_FB_SKIP_DEV(0x0582, 0x00db), /* BOSS GT-10 Guitar Effects Processor */ + IMPLICIT_FB_SKIP_DEV(0x0582, 0x00dc), /* BOSS GT-10B */ + IMPLICIT_FB_SKIP_DEV(0x0582, 0x00de), /* Roland Fantom-G */ + IMPLICIT_FB_SKIP_DEV(0x0582, 0x00f8), /* Roland JUNO Series */ + IMPLICIT_FB_SKIP_DEV(0x0582, 0x0111), /* Roland GAIA SH-01 */ + IMPLICIT_FB_SKIP_DEV(0x0582, 0x0120), /* Roland OCTA-CAPTURE */ + IMPLICIT_FB_SKIP_DEV(0x0582, 0x0121), /* Roland OCTA-CAPTURE */ + IMPLICIT_FB_SKIP_DEV(0x0582, 0x0123), /* Roland JUNO-Gi */ + IMPLICIT_FB_SKIP_DEV(0x0582, 0x0127), /* Roland GR-55 */ + IMPLICIT_FB_SKIP_DEV(0x0582, 0x012b), /* Roland DUO-CAPTURE */ + IMPLICIT_FB_SKIP_DEV(0x0582, 0x012f), /* Roland QUAD-CAPTURE */ IMPLICIT_FB_SKIP_DEV(0x0582, 0x0130), /* BOSS BR-80 */ - IMPLICIT_FB_SKIP_DEV(0x0582, 0x0171), /* BOSS RC-505 */ + IMPLICIT_FB_SKIP_DEV(0x0582, 0x0132), /* Roland TRI-CAPTURE */ + IMPLICIT_FB_SKIP_DEV(0x0582, 0x0138), /* BOSS RC-300 */ + IMPLICIT_FB_SKIP_DEV(0x0582, 0x013e), /* Roland R-26 */ + IMPLICIT_FB_SKIP_DEV(0x0582, 0x014d), /* BOSS GT-100 */ + IMPLICIT_FB_SKIP_DEV(0x0582, 0x0150), /* Roland TD-15 */ + IMPLICIT_FB_SKIP_DEV(0x0582, 0x0151), /* Roland TD-11 */ + IMPLICIT_FB_SKIP_DEV(0x0582, 0x0158), /* Roland TD-30 */ + IMPLICIT_FB_SKIP_DEV(0x0582, 0x0159), /* Roland DUO-CAPTURE EX */ + IMPLICIT_FB_SKIP_DEV(0x0582, 0x015b), /* Roland INTEGRA-7 */ + IMPLICIT_FB_SKIP_DEV(0x0582, 0x015d), /* Roland R-88 */ + IMPLICIT_FB_SKIP_DEV(0x0582, 0x0171), /* BOSS RC-505 */ + IMPLICIT_FB_SKIP_DEV(0x0582, 0x017c), /* Roland TR-8 */ IMPLICIT_FB_SKIP_DEV(0x0582, 0x0185), /* BOSS GP-10 */ IMPLICIT_FB_SKIP_DEV(0x0582, 0x0189), /* BOSS GT-100v2 */ + IMPLICIT_FB_SKIP_DEV(0x0582, 0x01b5), /* Roland Boutique Series Synthesizer */ IMPLICIT_FB_SKIP_DEV(0x0582, 0x01d6), /* BOSS GT-1 */ IMPLICIT_FB_SKIP_DEV(0x0582, 0x01d8), /* BOSS Katana */ IMPLICIT_FB_SKIP_DEV(0x0582, 0x01e5), /* BOSS GT-001 */ - IMPLICIT_FB_SKIP_DEV(0x0582, 0x0203), /* BOSS AD-10 */ + IMPLICIT_FB_SKIP_DEV(0x0582, 0x01ff), /* Roland D-05 */ + IMPLICIT_FB_SKIP_DEV(0x0582, 0x0203), /* BOSS AD-10 */ {} /* terminator */ }; /* Implicit feedback quirk table for capture: only FIXED type */ static const struct snd_usb_implicit_fb_match capture_implicit_fb_quirks[] = { + IMPLICIT_FB_FIXED_DEV(0x0582, 0x00a6, 0x0d, 0x01), /* Roland JUNO-G */ + IMPLICIT_FB_FIXED_DEV(0x0582, 0x00ad, 0x0d, 0x01), /* Roland SH-201 */ + IMPLICIT_FB_FIXED_DEV(0x0582, 0x00c2, 0x0d, 0x01), /* Roland SonicCell */ + IMPLICIT_FB_FIXED_DEV(0x0582, 0x00c4, 0x0d, 0x01), /* Edirol M-16DX */ + IMPLICIT_FB_FIXED_DEV(0x0582, 0x00c7, 0x0d, 0x01), /* Roland V-Synth GT */ + IMPLICIT_FB_FIXED_DEV(0x0582, 0x00da, 0x0d, 0x01), /* BOSS GT-10 */ + IMPLICIT_FB_FIXED_DEV(0x0582, 0x00db, 0x0d, 0x01), /* BOSS GT-10 Guitar Effects Processor */ + IMPLICIT_FB_FIXED_DEV(0x0582, 0x00dc, 0x0d, 0x01), /* BOSS GT-10B */ + IMPLICIT_FB_FIXED_DEV(0x0582, 0x00de, 0x0d, 0x01), /* Roland Fantom-G */ + IMPLICIT_FB_FIXED_DEV(0x0582, 0x00f8, 0x0d, 0x01), /* Roland JUNO Series */ + IMPLICIT_FB_FIXED_DEV(0x0582, 0x0111, 0x0d, 0x01), /* Roland GAIA SH-01 */ + IMPLICIT_FB_FIXED_DEV(0x0582, 0x0120, 0x0d, 0x01), /* Roland OCTA-CAPTURE */ + IMPLICIT_FB_FIXED_DEV(0x0582, 0x0121, 0x0d, 0x01), /* Roland OCTA-CAPTURE */ + IMPLICIT_FB_FIXED_DEV(0x0582, 0x0123, 0x0d, 0x01), /* Roland JUNO-Gi */ + IMPLICIT_FB_FIXED_DEV(0x0582, 0x0127, 0x0d, 0x01), /* Roland GR-55 */ + IMPLICIT_FB_FIXED_DEV(0x0582, 0x012b, 0x0d, 0x01), /* Roland DUO-CAPTURE */ + IMPLICIT_FB_FIXED_DEV(0x0582, 0x012f, 0x0d, 0x01), /* Roland QUAD-CAPTURE */ IMPLICIT_FB_FIXED_DEV(0x0582, 0x0130, 0x0d, 0x01), /* BOSS BR-80 */ + IMPLICIT_FB_FIXED_DEV(0x0582, 0x0132, 0x0d, 0x01), /* Roland TRI-CAPTURE */ + IMPLICIT_FB_FIXED_DEV(0x0582, 0x0138, 0x0d, 0x01), /* BOSS RC-300 */ + IMPLICIT_FB_FIXED_DEV(0x0582, 0x013e, 0x0d, 0x01), /* Roland R-26 */ + IMPLICIT_FB_FIXED_DEV(0x0582, 0x014d, 0x0d, 0x01), /* BOSS GT-100 */ + IMPLICIT_FB_FIXED_DEV(0x0582, 0x0150, 0x0d, 0x01), /* Roland TD-15 */ + IMPLICIT_FB_FIXED_DEV(0x0582, 0x0151, 0x0d, 0x01), /* Roland TD-11 */ + IMPLICIT_FB_FIXED_DEV(0x0582, 0x0158, 0x0d, 0x01), /* Roland TD-30 */ + IMPLICIT_FB_FIXED_DEV(0x0582, 0x0159, 0x0d, 0x01), /* Roland DUO-CAPTURE EX */ + IMPLICIT_FB_FIXED_DEV(0x0582, 0x015b, 0x0d, 0x01), /* Roland INTEGRA-7 */ + IMPLICIT_FB_FIXED_DEV(0x0582, 0x015d, 0x0d, 0x01), /* Roland R-88 */ IMPLICIT_FB_FIXED_DEV(0x0582, 0x0171, 0x0d, 0x01), /* BOSS RC-505 */ + IMPLICIT_FB_FIXED_DEV(0x0582, 0x017c, 0x0d, 0x01), /* Roland TR-8 */ IMPLICIT_FB_FIXED_DEV(0x0582, 0x0185, 0x0d, 0x01), /* BOSS GP-10 */ IMPLICIT_FB_FIXED_DEV(0x0582, 0x0189, 0x0d, 0x01), /* BOSS GT-100v2 */ + IMPLICIT_FB_FIXED_DEV(0x0582, 0x01b5, 0x0d, 0x01), /* Roland Boutique Series Synthesizer */ IMPLICIT_FB_FIXED_DEV(0x0582, 0x01d6, 0x0d, 0x01), /* BOSS GT-1 */ IMPLICIT_FB_FIXED_DEV(0x0582, 0x01d8, 0x0d, 0x01), /* BOSS Katana */ IMPLICIT_FB_FIXED_DEV(0x0582, 0x01e5, 0x0d, 0x01), /* BOSS GT-001 */ + IMPLICIT_FB_FIXED_DEV(0x0582, 0x01ff, 0x0d, 0x01), /* Roland D-05 */ IMPLICIT_FB_FIXED_DEV(0x0582, 0x0203, 0x0d, 0x01), /* BOSS AD-10 */ {} /* terminator */

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[PATCH] ALSA: hda/conexant: Apply quirk for another HP ZBook G5 model
by Takashi Iwai 01 Apr '21

01 Apr '21

There is another HP ZBook G5 model with the PCI SSID 103c:844f that requires the same quirk for controlling the mute LED. Add the corresponding entry to the quirk table. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=212407 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_conexant.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index c20dad46a7c9..dfef9c17e140 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -944,6 +944,7 @@ static const struct snd_pci_quirk cxt5066_fixups[] = { SND_PCI_QUIRK(0x103c, 0x829a, "HP 800 G3 DM", CXT_FIXUP_HP_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x8402, "HP ProBook 645 G4", CXT_FIXUP_MUTE_LED_GPIO), SND_PCI_QUIRK(0x103c, 0x8427, "HP ZBook Studio G5", CXT_FIXUP_HP_ZBOOK_MUTE_LED), + SND_PCI_QUIRK(0x103c, 0x844f, "HP ZBook Studio G5", CXT_FIXUP_HP_ZBOOK_MUTE_LED), SND_PCI_QUIRK(0x103c, 0x8455, "HP Z2 G4", CXT_FIXUP_HP_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x8456, "HP Z2 G4 SFF", CXT_FIXUP_HP_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x8457, "HP Z2 G4 mini", CXT_FIXUP_HP_MIC_NO_PRESENCE), -- 2.26.2

1 0

[PATCH 1/2] ASoC: codecs: lpass-tx-macro: set npl clock rate correctly
by Srinivas Kandagatla 01 Apr '21

01 Apr '21

NPL clock rate is twice the MCLK rate, so set this correctly to avoid soundwire timeouts. Fixes: c39667ddcfc5 ("ASoC: codecs: lpass-tx-macro: add support for lpass tx macro") Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- sound/soc/codecs/lpass-tx-macro.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/lpass-tx-macro.c b/sound/soc/codecs/lpass-tx-macro.c index 36d7a6442cdb..e8c6c738bbaa 100644 --- a/sound/soc/codecs/lpass-tx-macro.c +++ b/sound/soc/codecs/lpass-tx-macro.c @@ -1811,7 +1811,7 @@ static int tx_macro_probe(struct platform_device *pdev) /* set MCLK and NPL rates */ clk_set_rate(tx->clks[2].clk, MCLK_FREQ); - clk_set_rate(tx->clks[3].clk, MCLK_FREQ); + clk_set_rate(tx->clks[3].clk, 2 * MCLK_FREQ); ret = clk_bulk_prepare_enable(TX_NUM_CLKS_MAX, tx->clks); if (ret) -- 2.21.0

2 2

[PATCH] ASoC: Intel: Fix a typo
by Bhaskar Chowdhury 01 Apr '21

01 Apr '21

s/struture/structure/ Signed-off-by: Bhaskar Chowdhury <unixbhaskar(a)gmail.com> --- sound/soc/intel/atom/sst/sst.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/atom/sst/sst.h b/sound/soc/intel/atom/sst/sst.h index 4d37d39fd8f4..978bf4255888 100644 --- a/sound/soc/intel/atom/sst/sst.h +++ b/sound/soc/intel/atom/sst/sst.h @@ -344,7 +344,7 @@ struct sst_fw_save { * @block_lock : spin lock to add block to block_list and assign pvt_id * @rx_msg_lock : spin lock to handle the rx messages from the DSP * @scard_ops : sst card ops - * @pci : sst pci device struture + * @pci : sst pci device structure * @dev : pointer to current device struct * @sst_lock : sst device lock * @pvt_id : sst private id -- 2.31.0

2 1

[PATCH 1/1] ASoC: intel, keembay-i2s: Fix a dt_binding_check warning
by Zhen Lei 01 Apr '21

01 Apr '21

The property "dmas" contains two items: DMA "TX" and "RX" channel, Therefore, its value also needs to be written in two parts. Otherwise, below YAML check warning is reported: Documentation/devicetree/bindings/sound/intel,keembay-i2s.example.dt.yaml:\ i2s@20140000: dmas: [[4294967295, 29, 4294967295, 33]] is too short Signed-off-by: Zhen Lei <thunder.leizhen(a)huawei.com> --- Documentation/devicetree/bindings/sound/intel,keembay-i2s.yaml | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/intel,keembay-i2s.yaml b/Documentation/devicetree/bindings/sound/intel,keembay-i2s.yaml index 6f71294909a5fb4..803627e984f6089 100644 --- a/Documentation/devicetree/bindings/sound/intel,keembay-i2s.yaml +++ b/Documentation/devicetree/bindings/sound/intel,keembay-i2s.yaml @@ -81,6 +81,6 @@ examples: interrupts = <GIC_SPI 120 IRQ_TYPE_LEVEL_HIGH>; clock-names = "osc", "apb_clk"; clocks = <&scmi_clk KEEM_BAY_PSS_AUX_I2S3>, <&scmi_clk KEEM_BAY_PSS_I2S3>; - dmas = <&axi_dma0 29 &axi_dma0 33>; + dmas = <&axi_dma0 29>, <&axi_dma0 33>; dma-names = "tx", "rx"; }; -- 1.8.3

3 2

[PATCH] soundwire: intel_init: test link->cdns
by Bard Liao 01 Apr '21

01 Apr '21

intel_link_probe() could return error and dev_get_drvdata() will return null in such case. So we have to test link->cdns after link->cdns = dev_get_drvdata(&ldev->auxdev.dev); Otherwise, we will meet the "kernel NULL pointer dereference" error. Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Rander Wang <rander.wang(a)intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- drivers/soundwire/intel_init.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/drivers/soundwire/intel_init.c b/drivers/soundwire/intel_init.c index 5b32a2ffd376..5ef5bd0defab 100644 --- a/drivers/soundwire/intel_init.c +++ b/drivers/soundwire/intel_init.c @@ -250,6 +250,15 @@ static struct sdw_intel_ctx link = &ldev->link_res; link->cdns = dev_get_drvdata(&ldev->auxdev.dev); + if (!link->cdns) { + dev_err(&adev->dev, "failed to get link->cdns\n"); + /* + * 1 will be subtracted from i in the err label, but we need to call + * intel_link_dev_unregister for this ldev, so plus 1 now + */ + i++; + goto err; + } list_add_tail(&link->list, &ctx->link_list); bus = &link->cdns->bus; /* Calculate number of slaves */ -- 2.17.1

3 2

Fwd: [Bug 212477] EDIROL UA-101: midi is not recognized
by nick83ola 01 Apr '21

01 Apr '21

Anyone can help to debug this? ---------- Forwarded message --------- From: <bugzilla-daemon(a)bugzilla.kernel.org> Date: Thu, 1 Apr 2021 at 07:34 Subject: [Bug 212477] EDIROL UA-101: midi is not recognized To: <nick83ola(a)gmail.com> https://bugzilla.kernel.org/show_bug.cgi?id=212477 --- Comment #2 from Lucas Endres (jaffa225man(a)gmail.com) --- I hadn't realized it was working previously as I use my other Edirol UA-4FX most of the time, but I do remember the UA-101's MIDI not working for me when I tried using it too. Sure, enough, that's the case on debian's 5.9.0-1-rt-amd64 kernel now: aconnect -l | grep -i ua client 28: 'UA-4FX' [type=kernel,card=3] 0 'UA-4FX MIDI 1 ' I would've thought the two Edirol devices would use the same MIDI kernel module (snd_seq_midi and/or snd_usbmidi_lib) lsmod | grep -i snd snd_seq_dummy 16384 0 snd_ua101 28672 0 snd_seq_midi 20480 0 snd_seq_midi_event 16384 1 snd_seq_midi snd_seq 90112 4 snd_seq_midi,snd_seq_midi_event,snd_seq_dummy snd_usb_audio 319488 3 snd_usbmidi_lib 40960 2 snd_usb_audio,snd_ua101 snd_rawmidi 45056 2 snd_seq_midi,snd_usbmidi_lib snd_seq_device 16384 3 snd_seq,snd_seq_midi,snd_rawmidi mc 61440 1 snd_usb_audio snd_hda_codec_realtek 139264 1 snd_hda_codec_generic 98304 1 snd_hda_codec_realtek ledtrig_audio 16384 1 snd_hda_codec_generic snd_hda_codec_hdmi 73728 1 snd_hda_intel 57344 4 snd_intel_dspcfg 24576 1 snd_hda_intel snd_hda_codec 163840 4 snd_hda_codec_generic,snd_hda_codec_hdmi,snd_hda_intel,snd_hda_codec_realtek snd_hda_core 106496 5 snd_hda_codec_generic,snd_hda_codec_hdmi,snd_hda_intel,snd_hda_codec,snd_hda_codec_realtek snd_hwdep 16384 2 snd_usb_audio,snd_hda_codec snd_pcm 135168 7 snd_hda_codec_hdmi,snd_hda_intel,snd_usb_audio,snd_hda_codec,snd_ua101,snd_hda_core snd_hrtimer 16384 1 snd_timer 45056 4 snd_seq,snd_hrtimer,snd_pcm snd 110592 27 snd_hda_codec_generic,snd_seq,snd_seq_device,snd_hda_codec_hdmi,snd_hwdep,snd_hda_intel,snd_usb_audio,snd_usbmidi_lib,snd_hda_codec,snd_hda_codec_realtek,snd_ua101,snd_timer,snd_pcm,snd_rawmidi soundcore 16384 1 snd usbcore 323584 10 xhci_hcd,ehci_pci,snd_usb_audio,usbhid,snd_usbmidi_lib,snd_ua101,usb_storage,ehci_hcd,xhci_pci,uas lsusb -vvv Bus 004 Device 003: ID 0582:007d Roland Corp. EDIROL UA-101 Couldn't open device, some information will be missing Device Descriptor: bLength 18 bDescriptorType 1 bcdUSB 2.00 bDeviceClass 255 Vendor Specific Class bDeviceSubClass 0 bDeviceProtocol 255 bMaxPacketSize0 64 idVendor 0x0582 Roland Corp. idProduct 0x007d EDIROL UA-101 bcdDevice 1.03 iManufacturer 1 EDIROL iProduct 2 UA-101 iSerial 3 AX85615 bNumConfigurations 1 Configuration Descriptor: bLength 9 bDescriptorType 2 wTotalLength 0x009b bNumInterfaces 3 bConfigurationValue 1 iConfiguration 0 bmAttributes 0xc0 Self Powered MaxPower 0mA Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 1 bInterfaceProtocol 2 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 1 bInterfaceProtocol 2 iInterface 0 ** UNRECOGNIZED: 07 24 01 01 00 01 00 ** UNRECOGNIZED: 0b 24 02 01 0a 04 18 01 80 bb 00 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x02 EP 2 OUT bmAttributes 5 Transfer Type Isochronous Synch Type Asynchronous Usage Type Data wMaxPacketSize 0x0118 1x 280 bytes bInterval 1 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 1 bInterfaceProtocol 2 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 1 bInterfaceProtocol 2 iInterface 0 ** UNRECOGNIZED: 07 24 01 07 00 01 00 ** UNRECOGNIZED: 0b 24 02 01 0c 04 18 01 80 bb 00 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x81 EP 1 IN bmAttributes 37 Transfer Type Isochronous Synch Type Asynchronous Usage Type Implicit feedback Data wMaxPacketSize 0x0150 1x 336 bytes bInterval 1 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 0 bNumEndpoints 2 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 1 bInterfaceProtocol 3 iInterface 0 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x04 EP 4 OUT bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0200 1x 512 bytes bInterval 1 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x83 EP 3 IN bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0200 1x 512 bytes bInterval 1 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 1 bNumEndpoints 2 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 1 bInterfaceProtocol 3 iInterface 0 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x04 EP 4 OUT bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0200 1x 512 bytes bInterval 1 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x83 EP 3 IN bmAttributes 3 Transfer Type Interrupt Synch Type None Usage Type Data wMaxPacketSize 0x0200 1x 512 bytes bInterval 1 Bus 002 Device 004: ID 0582:00a3 Roland Corp. EDIROL UA-4FX Couldn't open device, some information will be missing Device Descriptor: bLength 18 bDescriptorType 1 bcdUSB 1.10 bDeviceClass 255 Vendor Specific Class bDeviceSubClass 0 bDeviceProtocol 255 bMaxPacketSize0 8 idVendor 0x0582 Roland Corp. idProduct 0x00a3 EDIROL UA-4FX bcdDevice 1.00 iManufacturer 1 EDIROL iProduct 2 UA-4FX iSerial 0 bNumConfigurations 1 Configuration Descriptor: bLength 9 bDescriptorType 2 wTotalLength 0x00ca bNumInterfaces 3 bConfigurationValue 1 iConfiguration 0 bmAttributes 0x80 (Bus Powered) MaxPower 360mA Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 2 bInterfaceProtocol 2 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 2 bInterfaceProtocol 2 iInterface 0 ** UNRECOGNIZED: 07 24 01 01 00 01 00 ** UNRECOGNIZED: 0b 24 02 01 02 03 18 01 80 bb 00 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x01 EP 1 OUT bmAttributes 9 Transfer Type Isochronous Synch Type Adaptive Usage Type Data wMaxPacketSize 0x0140 1x 320 bytes bInterval 1 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 2 bNumEndpoints 1 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 2 bInterfaceProtocol 2 iInterface 0 ** UNRECOGNIZED: 07 24 01 02 00 01 20 ** UNRECOGNIZED: 0b 24 02 03 02 03 18 01 80 bb 00 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x01 EP 1 OUT bmAttributes 9 Transfer Type Isochronous Synch Type Adaptive Usage Type Data wMaxPacketSize 0x0138 1x 312 bytes bInterval 1 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 2 bInterfaceProtocol 1 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 2 bInterfaceProtocol 1 iInterface 0 ** UNRECOGNIZED: 07 24 01 07 00 01 00 ** UNRECOGNIZED: 0b 24 02 01 02 03 18 01 80 bb 00 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x82 EP 2 IN bmAttributes 5 Transfer Type Isochronous Synch Type Asynchronous Usage Type Data wMaxPacketSize 0x0140 1x 320 bytes bInterval 1 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 0 bNumEndpoints 2 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 3 bInterfaceProtocol 0 iInterface 0 ** UNRECOGNIZED: 06 24 f1 02 01 01 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x03 EP 3 OUT bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0020 1x 32 bytes bInterval 0 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x84 EP 4 IN bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0020 1x 32 bytes bInterval 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 1 bNumEndpoints 2 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 3 bInterfaceProtocol 0 iInterface 0 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x03 EP 3 OUT bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0020 1x 32 bytes bInterval 1 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x84 EP 4 IN bmAttributes 3 Transfer Type Interrupt Synch Type None Usage Type Data wMaxPacketSize 0x0020 1x 32 bytes bInterval 1 I, too, would be glad to help however possible, but don't know where to begin. I have the source for the latest mainline kernel (5.11.9), so I'd be happy to test proposed patches. -- You may reply to this email to add a comment. You are receiving this mail because: You reported the bug.

1 0

Re: [PATCH] rcar/rsnd.h: Remove repeated struct declaration
by Mark Brown 01 Apr '21

01 Apr '21

On Thu, Apr 01, 2021 at 06:37:58PM +0800, Wan Jiabing wrote: > struct rsnd_dai_stream is declared twice. One is declared > at 254th line. The blew one is not needed. Remove the duplicate. Please submit patches using subject lines reflecting the style for the subsystem, this makes it easier for people to identify relevant patches. Look at what existing commits in the area you're changing are doing and make sure your subject lines visually resemble what they're doing. There's no need to resubmit to fix this alone.

1 0

[PATCH 00/17] ASoC: remove cppcheck warnings for multiple SOCs
by Pierre-Louis Bossart 01 Apr '21

01 Apr '21

Trivial cleanups to make cppcheck less verbose. There should be no functionality change, except for the 'sti_uniperif' patch where an error check was added. Pierre-Louis Bossart (17): ASoC: amd: renoir: acp3x-pdm-dma: remove unnecessary assignments ASoC: atmel: fix shadowed variable ASoC: atmel: atmel-i2s: remove useless initialization ASoC: bcm: cygnus_ssp: remove useless initialization ASoC: meson: axg-tdmin: remove useless assignment ASoC: meson: axg-tdmout: remove useless assignment ASoC: pxa: remove useless assignment ASoC: sti: sti_uniperif: add missing error check ASoC: sti: uniperif: align function prototypes ASoC: stm: stm32_adfsdm: fix snprintf format string ASoC: sunxi: sun8i-codec: clarify expression ASoC: tegra: tegra20_das: clarify expression ASoC: tegra: tegra20_das: align function prototypes ASoC: ti: omap-abe-twl6040: remove useless assignment ASoC: ti: omap-mcsp: remove duplicate test ASoC: ux500: mop500: rename shadowing variable ASoC: ux500: mop500: align function prototype sound/soc/amd/renoir/acp3x-pdm-dma.c | 2 -- sound/soc/atmel/atmel-classd.c | 6 +++--- sound/soc/atmel/atmel-i2s.c | 2 +- sound/soc/bcm/cygnus-ssp.c | 2 +- sound/soc/meson/axg-tdmin.c | 2 +- sound/soc/meson/axg-tdmout.c | 2 +- sound/soc/pxa/mmp-pcm.c | 2 +- sound/soc/sti/sti_uniperif.c | 2 ++ sound/soc/sti/uniperif.h | 4 ++-- sound/soc/stm/stm32_adfsdm.c | 2 +- sound/soc/sunxi/sun8i-codec.c | 2 +- sound/soc/tegra/tegra20_das.c | 8 ++++---- sound/soc/tegra/tegra20_das.h | 6 +++--- sound/soc/ti/omap-abe-twl6040.c | 2 +- sound/soc/ti/omap-mcbsp.c | 3 +-- sound/soc/ux500/mop500.c | 6 +++--- sound/soc/ux500/mop500_ab8500.h | 2 +- 17 files changed, 27 insertions(+), 28 deletions(-) -- 2.25.1

7 27

[PATCH 0/3] ASoC: dt-bindings: Rework audio-graph-port schema
by Rob Herring 01 Apr '21

01 Apr '21

This series refactors the audio-graph-port.yaml schema moving the 'port' node out of the schema and updating to use graph.yaml schema. This allows users to define what each 'port' node is like other graph binding users. Rob Cc: alsa-devel(a)alsa-project.org Cc: linux-tegra(a)vger.kernel.org Cc: Jonathan Hunter <jonathanh(a)nvidia.com> Cc: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Lubomir Rintel <lkundrak(a)v3.sk> Cc: Mark Brown <broonie(a)kernel.org> Cc: Masahiro Yamada <yamada.masahiro(a)socionext.com> Cc: Sameer Pujar <spujar(a)nvidia.com> Cc: Thierry Reding <thierry.reding(a)gmail.com> Rob Herring (3): ASoC: dt-bindings: Move port/ports properties out of audio-graph-port.yaml ASoC: dt-bindings: Use OF graph schema ASoC: dt-bindings: socionext: Use audio-graph-port schema .../bindings/sound/audio-graph-port.yaml | 104 ++++++++---------- .../bindings/sound/marvell,mmp-sspa.yaml | 6 +- .../bindings/sound/nvidia,tegra186-dspk.yaml | 11 +- .../sound/nvidia,tegra210-admaif.yaml | 11 +- .../bindings/sound/nvidia,tegra210-ahub.yaml | 11 +- .../bindings/sound/nvidia,tegra210-dmic.yaml | 11 +- .../bindings/sound/nvidia,tegra210-i2s.yaml | 11 +- .../bindings/sound/renesas,rsnd.yaml | 5 +- .../sound/socionext,uniphier-aio.yaml | 8 +- .../sound/socionext,uniphier-evea.yaml | 8 +- 10 files changed, 89 insertions(+), 97 deletions(-) -- 2.27.0

2 6

[PATCH 0/2] ASoC: soc-core: tidyup error handling for rtd
by Kuninori Morimoto 01 Apr '21

01 Apr '21

Hi Mark These are small clanup patches for soc-core. [1/2] patch adds missing explanation, and [2/2] patch fixup error handling of rtd. Kuninori Morimoto (2): ASoC: soc-core: add comment for rtd freeing ASoC: soc-core: use device_unregister() if rtd allocation failed sound/soc/soc-core.c | 14 ++++++++++++-- 1 file changed, 12 insertions(+), 2 deletions(-) -- 2.25.1

2 3

[PATCH 0/6] ASoC: simple-card: cleanup and prepare for Multi CPU/Codec support
by Kuninori Morimoto 01 Apr '21

01 Apr '21

Hi Mark I want to add new audio-graph-card2 driver which can support not only DPCM, but also Multi-CPU/Codec, and Codec2Codec. And it is also supporting user customization. But before supporting such driver, we need to cleanup existing simple-card / audio-graph, because these and new driver are sharing code. This patch-set are cleanup and prepare for Multi-CPU/Codec support. Kuninori Morimoto (6): ASoC: simple-card-utils.c: remove old comment ASoC: simple-card-utils: alloc dai_link information for CPU/Codec/Platform ASoC: audio-graph: count DAI / link numbers as in order ASoC: audio-graph: cleanup graph_for_each_link() ASoC: simple-card: count DAI / link numbers as in order ASoC: simple-card: cleanup graph_for_each_link() include/sound/simple_card_utils.h | 7 +- sound/soc/generic/audio-graph-card.c | 112 +++++++++++++------------ sound/soc/generic/simple-card-utils.c | 20 +++-- sound/soc/generic/simple-card.c | 115 +++++++++++++++----------- 4 files changed, 143 insertions(+), 111 deletions(-) -- 2.25.1

2 7

Re: [PATCH] ASoC: sunxi: sun4i-codec: fill ASoC card owner
by Mark Brown 01 Apr '21

01 Apr '21

On Wed, 31 Mar 2021 16:59:15 +0200, bage(a)linutronix.de wrote: > card->owner is a required property and since commit 81033c6b584b ("ALSA: > core: Warn on empty module") a warning is issued if it is empty. Add it. > This fixes following warning observed on Lamobo R1: > > WARNING: CPU: 1 PID: 190 at sound/core/init.c:207 snd_card_new+0x430/0x480 [snd] > Modules linked in: sun4i_codec(E+) sun4i_backend(E+) snd_soc_core(E) ... > CPU: 1 PID: 190 Comm: systemd-udevd Tainted: G C E 5.10.0-1-armmp #1 Debian 5.10.4-1 > Hardware name: Allwinner sun7i (A20) Family > [<c0ce8544>] (dump_backtrace) from [<c0ce88f0>] (show_stack+0x20/0x24) > [<c0ce88d0>] (show_stack) from [<c0ceda20>] (dump_stack+0xc8/0xdc) > [<c0ced958>] (dump_stack) from [<c034d49c>] (__warn+0xfc/0x158) > [<c034d3a0>] (__warn) from [<c0ce966c>] (warn_slowpath_fmt+0x70/0xe4) > [<c0ce9600>] (warn_slowpath_fmt) from [<bf48f604>] (snd_card_new+0x430/0x480 [snd]) > [<bf48f1d4>] (snd_card_new [snd]) from [<bf53340c>] (snd_soc_bind_card+0x3a8/0xa30 [snd_soc_core]) > [<bf533064>] (snd_soc_bind_card [snd_soc_core]) from [<bf533b8c>] (snd_soc_register_card+0xf8/0x108 [snd_soc_core]) > [<bf533a94>] (snd_soc_register_card [snd_soc_core]) from [<bf55b888>] (sun4i_codec_probe+0x224/0x45c [sun4i_codec]) > [<bf55b664>] (sun4i_codec_probe [sun4i_codec]) from [<c09879b4>] (platform_drv_probe+0x58/0xa8) > [<c098795c>] (platform_drv_probe) from [<c0984dc8>] (really_probe+0x108/0x510) > [<c0984cc0>] (really_probe) from [<c0985674>] (driver_probe_device+0x100/0x1d0) > [<c0985574>] (driver_probe_device) from [<c0985a74>] (device_driver_attach+0xb8/0xc0) > [<c09859bc>] (device_driver_attach) from [<c0985b18>] (__driver_attach+0x9c/0x150) > [<c0985a7c>] (__driver_attach) from [<c09825c8>] (bus_for_each_dev+0x84/0xd0) > [<c0982544>] (bus_for_each_dev) from [<c09844bc>] (driver_attach+0x2c/0x30) > [<c0984490>] (driver_attach) from [<c0983e70>] (bus_add_driver+0x120/0x20c) > [<c0983d50>] (bus_add_driver) from [<c098670c>] (driver_register+0x98/0x128) > [<c0986674>] (driver_register) from [<c0987908>] (__platform_driver_register+0x50/0x58) > [<c09878b8>] (__platform_driver_register) from [<bf566024>] (sun4i_codec_driver_init+0x24/0x1000 [sun4i_codec]) > [<bf566000>] (sun4i_codec_driver_init [sun4i_codec]) from [<c030293c>] (do_one_initcall+0x50/0x274) > [<c03028ec>] (do_one_initcall) from [<c040d45c>] (do_init_module+0x70/0x294) > [<c040d3ec>] (do_init_module) from [<c040f9f4>] (load_module+0x2258/0x263c) > [<c040d79c>] (load_module) from [<c04100c0>] (sys_finit_module+0xc8/0x12c) > [<c040fff8>] (sys_finit_module) from [<c03002a4>] (__sys_trace_return+0x0/0x1c) Applied to https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-next Thanks! [1/1] ASoC: sunxi: sun4i-codec: fill ASoC card owner commit: 7c0d6e482062eb5c06ecccfab340abc523bdca00 All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark

1 0

[PATCH] ASoC: Intel: Fix a typo
by Bhaskar Chowdhury 01 Apr '21

01 Apr '21

s/defintions/definitions/ Signed-off-by: Bhaskar Chowdhury <unixbhaskar(a)gmail.com> --- sound/soc/intel/skylake/skl.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/skylake/skl.h b/sound/soc/intel/skylake/skl.h index 857ea17e3c9f..33ed274fc0cb 100644 --- a/sound/soc/intel/skylake/skl.h +++ b/sound/soc/intel/skylake/skl.h @@ -1,6 +1,6 @@ /* SPDX-License-Identifier: GPL-2.0-only */ /* - * skl.h - HD Audio skylake defintions. + * skl.h - HD Audio skylake definitions. * * Copyright (C) 2015 Intel Corp * Author: Jeeja KP <jeeja.kp(a)intel.com> -- 2.31.0

2 1

[PATCH] dt-bindings: sound: add address-cells and size-cells information
by Jack Yu 01 Apr '21

01 Apr '21

Add address-cells and size-cells information to fix warnings for rt1019.yaml. Signed-off-by: Jack Yu <jack.yu(a)realtek.com> --- Documentation/devicetree/bindings/sound/rt1019.yaml | 2 ++ 1 file changed, 2 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/rt1019.yaml b/Documentation/devicetree/bindings/sound/rt1019.yaml index c24c29eafa54..3d5a91a942f4 100644 --- a/Documentation/devicetree/bindings/sound/rt1019.yaml +++ b/Documentation/devicetree/bindings/sound/rt1019.yaml @@ -26,6 +26,8 @@ additionalProperties: false examples: - | i2c { + #address-cells = <1>; + #size-cells = <0>; rt1019: codec@28 { compatible = "realtek,rt1019"; reg = <0x28>; -- 2.29.0

2 2

[PATCH] ASoC: tegra: Set driver name explicitly
by Thierry Reding 01 Apr '21

01 Apr '21

From: Thierry Reding <treding(a)nvidia.com> The SoC sound core will generate a driver name by normalizing the card name. However, most of the time that name does not tell anything about the driver and is therefore useless for this purpose. Make the driver name more useful by setting it explicitly during card initialization. Signed-off-by: Thierry Reding <treding(a)nvidia.com> --- sound/soc/tegra/tegra_audio_graph_card.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/tegra/tegra_audio_graph_card.c b/sound/soc/tegra/tegra_audio_graph_card.c index ddedf18adde1..47b319504c8c 100644 --- a/sound/soc/tegra/tegra_audio_graph_card.c +++ b/sound/soc/tegra/tegra_audio_graph_card.c @@ -198,6 +198,7 @@ static int tegra_audio_graph_probe(struct platform_device *pdev) return -ENOMEM; card = simple_priv_to_card(&priv->simple); + card->driver_name = "tegra-ape"; card->probe = tegra_audio_graph_card_probe; -- 2.30.2

2 1

[PATCH v2 0/3] ASoC: dt-bindings: Rework audio-graph-port schema
by Rob Herring 01 Apr '21

01 Apr '21

This series refactors the audio-graph-port.yaml schema moving the 'port' node out of the schema and updating to use graph.yaml schema. This allows users to define what each 'port' node is like other graph binding users. v2: - Rebase on ASoC tree Rob Cc: alsa-devel(a)alsa-project.org Cc: linux-tegra(a)vger.kernel.org Cc: Jonathan Hunter <jonathanh(a)nvidia.com> Cc: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Lubomir Rintel <lkundrak(a)v3.sk> Cc: Mark Brown <broonie(a)kernel.org> Cc: Masahiro Yamada <yamada.masahiro(a)socionext.com> Cc: Sameer Pujar <spujar(a)nvidia.com> Cc: Thierry Reding <thierry.reding(a)gmail.com> Rob Herring (3): ASoC: dt-bindings: Move port/ports properties out of audio-graph-port.yaml ASoC: dt-bindings: Use OF graph schema ASoC: dt-bindings: socionext: Use audio-graph-port schema .../bindings/sound/audio-graph-port.yaml | 104 ++++++++---------- .../bindings/sound/marvell,mmp-sspa.yaml | 6 +- .../bindings/sound/nvidia,tegra186-dspk.yaml | 11 +- .../sound/nvidia,tegra210-admaif.yaml | 11 +- .../bindings/sound/nvidia,tegra210-ahub.yaml | 11 +- .../bindings/sound/nvidia,tegra210-dmic.yaml | 11 +- .../bindings/sound/nvidia,tegra210-i2s.yaml | 11 +- .../bindings/sound/renesas,rsnd.yaml | 5 +- .../sound/socionext,uniphier-aio.yaml | 8 +- .../sound/socionext,uniphier-evea.yaml | 8 +- 10 files changed, 89 insertions(+), 97 deletions(-) -- 2.27.0

2 4

Re: [PATCH v2] ASoC: sunxi: sun4i-codec: fill ASoC card owner
by Mark Brown 01 Apr '21

01 Apr '21

On Wed, 31 Mar 2021 17:18:43 +0200, bage(a)linutronix.de wrote: > card->owner is a required property and since commit 81033c6b584b ("ALSA: > core: Warn on empty module") a warning is issued if it is empty. Add it. > This fixes following warning observed on Lamobo R1: > > WARNING: CPU: 1 PID: 190 at sound/core/init.c:207 snd_card_new+0x430/0x480 [snd] > Modules linked in: sun4i_codec(E+) sun4i_backend(E+) snd_soc_core(E) ... > CPU: 1 PID: 190 Comm: systemd-udevd Tainted: G C E 5.10.0-1-armmp #1 Debian 5.10.4-1 > Hardware name: Allwinner sun7i (A20) Family > Call trace: > (snd_card_new [snd]) > (snd_soc_bind_card [snd_soc_core]) > (snd_soc_register_card [snd_soc_core]) > (sun4i_codec_probe [sun4i_codec]) Applied to https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-next Thanks! [1/1] ASoC: sunxi: sun4i-codec: fill ASoC card owner commit: 7c0d6e482062eb5c06ecccfab340abc523bdca00 All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark

1 0

[PATCH v2 1/2] ASoC: ak5558: Add support for ak5552
by Shengjiu Wang 01 Apr '21

01 Apr '21

From: Viorel Suman <viorel.suman(a)nxp.com> AK5552 is a 32-bit 2ch ADC and has the same register map as AK5558. Signed-off-by: Viorel Suman <viorel.suman(a)nxp.com> Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- changes in v2 - refine the control name. sound/soc/codecs/ak5558.c | 97 ++++++++++++++++++++++++++++++++++++--- 1 file changed, 90 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/ak5558.c b/sound/soc/codecs/ak5558.c index 85bdd0534180..3ac62a281807 100644 --- a/sound/soc/codecs/ak5558.c +++ b/sound/soc/codecs/ak5558.c @@ -9,6 +9,7 @@ #include <linux/gpio/consumer.h> #include <linux/i2c.h> #include <linux/module.h> +#include <linux/of_device.h> #include <linux/pm_runtime.h> #include <linux/regmap.h> #include <linux/regulator/consumer.h> @@ -23,6 +24,11 @@ #include "ak5558.h" +enum ak555x_type { + AK5558, + AK5552, +}; + #define AK5558_NUM_SUPPLIES 2 static const char *ak5558_supply_names[AK5558_NUM_SUPPLIES] = { "DVDD", @@ -59,6 +65,15 @@ static const struct soc_enum ak5558_mono_enum[] = { ARRAY_SIZE(mono_texts), mono_texts), }; +static const char * const mono_5552_texts[] = { + "2 Slot", "1 Slot (Fixed)", "2 Slot", "1 Slot (Optimal)", +}; + +static const struct soc_enum ak5552_mono_enum[] = { + SOC_ENUM_SINGLE(AK5558_01_POWER_MANAGEMENT2, 1, + ARRAY_SIZE(mono_5552_texts), mono_5552_texts), +}; + static const char * const digfil_texts[] = { "Sharp Roll-Off", "Show Roll-Off", "Short Delay Sharp Roll-Off", "Short Delay Show Roll-Off", @@ -70,8 +85,13 @@ static const struct soc_enum ak5558_adcset_enum[] = { }; static const struct snd_kcontrol_new ak5558_snd_controls[] = { - SOC_ENUM("AK5558 Monaural Mode", ak5558_mono_enum[0]), - SOC_ENUM("AK5558 Digital Filter", ak5558_adcset_enum[0]), + SOC_ENUM("Monaural Mode", ak5558_mono_enum[0]), + SOC_ENUM("Digital Filter", ak5558_adcset_enum[0]), +}; + +static const struct snd_kcontrol_new ak5552_snd_controls[] = { + SOC_ENUM("Monaural Mode", ak5552_mono_enum[0]), + SOC_ENUM("Digital Filter", ak5558_adcset_enum[0]), }; static const struct snd_soc_dapm_widget ak5558_dapm_widgets[] = { @@ -97,6 +117,17 @@ static const struct snd_soc_dapm_widget ak5558_dapm_widgets[] = { SND_SOC_DAPM_AIF_OUT("SDTO", "Capture", 0, SND_SOC_NOPM, 0, 0), }; +static const struct snd_soc_dapm_widget ak5552_dapm_widgets[] = { + /* Analog Input */ + SND_SOC_DAPM_INPUT("AIN1"), + SND_SOC_DAPM_INPUT("AIN2"), + + SND_SOC_DAPM_ADC("ADC Ch1", NULL, AK5558_00_POWER_MANAGEMENT1, 0, 0), + SND_SOC_DAPM_ADC("ADC Ch2", NULL, AK5558_00_POWER_MANAGEMENT1, 1, 0), + + SND_SOC_DAPM_AIF_OUT("SDTO", "Capture", 0, SND_SOC_NOPM, 0, 0), +}; + static const struct snd_soc_dapm_route ak5558_intercon[] = { {"ADC Ch1", NULL, "AIN1"}, {"SDTO", NULL, "ADC Ch1"}, @@ -123,6 +154,14 @@ static const struct snd_soc_dapm_route ak5558_intercon[] = { {"SDTO", NULL, "ADC Ch8"}, }; +static const struct snd_soc_dapm_route ak5552_intercon[] = { + {"ADC Ch1", NULL, "AIN1"}, + {"SDTO", NULL, "ADC Ch1"}, + + {"ADC Ch2", NULL, "AIN2"}, + {"SDTO", NULL, "ADC Ch2"}, +}; + static int ak5558_set_mcki(struct snd_soc_component *component) { return snd_soc_component_update_bits(component, AK5558_02_CONTROL1, AK5558_CKS, @@ -267,6 +306,18 @@ static struct snd_soc_dai_driver ak5558_dai = { .ops = &ak5558_dai_ops, }; +static struct snd_soc_dai_driver ak5552_dai = { + .name = "ak5558-aif", + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_KNOT, + .formats = AK5558_FORMATS, + }, + .ops = &ak5558_dai_ops, +}; + static void ak5558_power_off(struct ak5558_priv *ak5558) { if (!ak5558->reset_gpiod) @@ -354,6 +405,21 @@ static const struct snd_soc_component_driver soc_codec_dev_ak5558 = { .non_legacy_dai_naming = 1, }; +static const struct snd_soc_component_driver soc_codec_dev_ak5552 = { + .probe = ak5558_probe, + .remove = ak5558_remove, + .controls = ak5552_snd_controls, + .num_controls = ARRAY_SIZE(ak5552_snd_controls), + .dapm_widgets = ak5552_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(ak5552_dapm_widgets), + .dapm_routes = ak5552_intercon, + .num_dapm_routes = ARRAY_SIZE(ak5552_intercon), + .idle_bias_on = 1, + .use_pmdown_time = 1, + .endianness = 1, + .non_legacy_dai_naming = 1, +}; + static const struct regmap_config ak5558_regmap = { .reg_bits = 8, .val_bits = 8, @@ -368,6 +434,7 @@ static int ak5558_i2c_probe(struct i2c_client *i2c) { struct ak5558_priv *ak5558; int ret = 0; + int dev_id; int i; ak5558 = devm_kzalloc(&i2c->dev, sizeof(*ak5558), GFP_KERNEL); @@ -396,11 +463,26 @@ static int ak5558_i2c_probe(struct i2c_client *i2c) return ret; } - ret = devm_snd_soc_register_component(&i2c->dev, - &soc_codec_dev_ak5558, - &ak5558_dai, 1); - if (ret) + dev_id = (uintptr_t)of_device_get_match_data(&i2c->dev); + switch (dev_id) { + case AK5552: + ret = devm_snd_soc_register_component(&i2c->dev, + &soc_codec_dev_ak5552, + &ak5552_dai, 1); + break; + case AK5558: + ret = devm_snd_soc_register_component(&i2c->dev, + &soc_codec_dev_ak5558, + &ak5558_dai, 1); + break; + default: + dev_err(&i2c->dev, "unexpected device type\n"); + return -EINVAL; + } + if (ret < 0) { + dev_err(&i2c->dev, "failed to register component: %d\n", ret); return ret; + } pm_runtime_enable(&i2c->dev); regcache_cache_only(ak5558->regmap, true); @@ -416,7 +498,8 @@ static int ak5558_i2c_remove(struct i2c_client *i2c) } static const struct of_device_id ak5558_i2c_dt_ids[] __maybe_unused = { - { .compatible = "asahi-kasei,ak5558"}, + { .compatible = "asahi-kasei,ak5558", .data = (void *) AK5558 }, + { .compatible = "asahi-kasei,ak5552", .data = (void *) AK5552 }, { } }; MODULE_DEVICE_TABLE(of, ak5558_i2c_dt_ids); -- 2.27.0

2 2

[PATCH] ASoC: Intel: bytcr_rt5640: Add quirk for the Chuwi Hi8 tablet
by Hans de Goede 01 Apr '21

01 Apr '21

The Chuwi Hi8 tablet is using an analog mic on IN1 and has its jack-detect connected to JD2_IN4N, instead of using the default IN3 for its internal mic and JD1_IN4P for jack-detect. It also only has 1 speaker. Add a quirk applying the correct settings for this configuration. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- sound/soc/intel/boards/bytcr_rt5640.c | 17 +++++++++++++++++ 1 file changed, 17 insertions(+) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 59d6d47c8d82..df2f5d55e8ff 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -518,6 +518,23 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { BYT_RT5640_SSP0_AIF1 | BYT_RT5640_MCLK_EN), }, + { + /* Chuwi Hi8 (CWI509) */ + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "Hampoo"), + DMI_MATCH(DMI_BOARD_NAME, "BYT-PA03C"), + DMI_MATCH(DMI_SYS_VENDOR, "ilife"), + DMI_MATCH(DMI_PRODUCT_NAME, "S806"), + }, + .driver_data = (void *)(BYT_RT5640_IN1_MAP | + BYT_RT5640_JD_SRC_JD2_IN4N | + BYT_RT5640_OVCD_TH_2000UA | + BYT_RT5640_OVCD_SF_0P75 | + BYT_RT5640_MONO_SPEAKER | + BYT_RT5640_DIFF_MIC | + BYT_RT5640_SSP0_AIF1 | + BYT_RT5640_MCLK_EN), + }, { .matches = { DMI_MATCH(DMI_SYS_VENDOR, "Circuitco"), -- 2.30.2

2 1

[PATCH] ALSA: control led - improve the set_led_id() parser
by Jaroslav Kysela 01 Apr '21

01 Apr '21

It may be possible that the string pointer does not move when parsing. Add a code which detects this state and simply break the parser loop in this case. Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> --- sound/core/control_led.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/core/control_led.c b/sound/core/control_led.c index d4fb8b873f34..788fd9e275e0 100644 --- a/sound/core/control_led.c +++ b/sound/core/control_led.c @@ -506,7 +506,7 @@ static char *parse_iface(char *s, unsigned int *val) static ssize_t set_led_id(struct snd_ctl_led_card *led_card, const char *buf, size_t count, bool attach) { - char buf2[256], *s; + char buf2[256], *s, *os; size_t len = max(sizeof(s) - 1, count); struct snd_ctl_elem_id id; int err; @@ -517,6 +517,7 @@ static ssize_t set_led_id(struct snd_ctl_led_card *led_card, const char *buf, si id.iface = SNDRV_CTL_ELEM_IFACE_MIXER; s = buf2; while (*s) { + os = s; if (!strncasecmp(s, "numid=", 6)) { s = parse_uint(s + 6, &id.numid); } else if (!strncasecmp(s, "iface=", 6)) { @@ -546,6 +547,8 @@ static ssize_t set_led_id(struct snd_ctl_led_card *led_card, const char *buf, si } if (*s == ',') s++; + if (s == os) + break; } err = snd_ctl_led_set_id(led_card->number, &id, led_card->led->group, attach); -- 2.30.2

2 1

[PATCH] ALSA: control - add the missing prev_lops2 initialization
by Jaroslav Kysela 01 Apr '21

01 Apr '21

As static analysis reported, the prev_lops2 should contain the previous lops2 pointer in snd_ctl_disconnect_layer(). Link: https://lore.kernel.org/alsa-devel/96e9bd5c-c8db-0db8-b393-fbf4a047dc80@can… Fixes: 3f0638a0333b ("ALSA: control - add layer registration routines") Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> --- sound/core/control.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/core/control.c b/sound/core/control.c index 6825ca75daf5..20d707d4ef40 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -2074,7 +2074,7 @@ void snd_ctl_disconnect_layer(struct snd_ctl_layer_ops *lops) struct snd_ctl_layer_ops *lops2, *prev_lops2; down_write(&snd_ctl_layer_rwsem); - for (lops2 = snd_ctl_layer, prev_lops2 = NULL; lops2; lops2 = lops2->next) + for (lops2 = snd_ctl_layer, prev_lops2 = NULL; lops2; lops2 = lops2->next) { if (lops2 == lops) { if (!prev_lops2) snd_ctl_layer = lops->next; @@ -2082,6 +2082,8 @@ void snd_ctl_disconnect_layer(struct snd_ctl_layer_ops *lops) prev_lops2->next = lops->next; break; } + prev_lops2 = lops2; + } up_write(&snd_ctl_layer_rwsem); } EXPORT_SYMBOL_GPL(snd_ctl_disconnect_layer); -- 2.30.2

2 1

[PATCH v4 0/5] soundwire: add static port map support
by Srinivas Kandagatla 01 Apr '21

01 Apr '21

In some cases, SoundWire device ports are statically mapped to Controller ports during design, however there is no way to expose this information to the controller. Controllers like Qualcomm ones use this info to setup static bandwidth parameters for those ports. A generic port allocation is not possible in this cases! This patch adds a new member m_port_map to SoundWire device so that it can populate the static master port map and share it with controller to be able to setup correct bandwidth parameters. As a user of this feature this patchset also adds new bindings for wsa881x smart speaker which has 4 ports which are statically mapped to the 3 output and 1 input port of the controller. Tested it on DB845c and SM8250 MTP. thanks, srini Changes since v3: - updated kernel doc for more clarity on m_port_map Srinivas Kandagatla (5): soundwire: add static port mapping support soundwire: qcom: update port map allocation bit mask soundwire: qcom: add static port map support ASoC: dt-bindings: wsa881x: add bindings for port mapping ASoC: codecs: wsa881x: add static port map support .../bindings/sound/qcom,wsa881x.yaml | 9 ++++++ drivers/soundwire/qcom.c | 31 +++++++++++++++---- include/linux/soundwire/sdw.h | 2 ++ sound/soc/codecs/wsa881x.c | 7 +++++ 4 files changed, 43 insertions(+), 6 deletions(-) -- 2.21.0

4 9

[PATCH] soundwire: stream: fix memory leak in stream config error path
by Bard Liao 01 Apr '21

01 Apr '21

From: Rander Wang <rander.wang(a)intel.com> When stream config is failed, master runtime will release all slave runtime in the slave_rt_list, but slave runtime is not added to the list at this time. This patch frees slave runtime in the config error path to fix the memory leak. Fixes: bbe7379d8040a ("soundwire: Add support for SoundWire stream management") Signed-off-by: Rander Wang <rander.wang(a)intel.com> Reviewed-by: Keyon Jie <yang.jie(a)intel.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> --- drivers/soundwire/stream.c | 10 +++++++++- 1 file changed, 9 insertions(+), 1 deletion(-) diff --git a/drivers/soundwire/stream.c b/drivers/soundwire/stream.c index 9c064b672745..1eaedaaba094 100644 --- a/drivers/soundwire/stream.c +++ b/drivers/soundwire/stream.c @@ -1375,8 +1375,16 @@ int sdw_stream_add_slave(struct sdw_slave *slave, } ret = sdw_config_stream(&slave->dev, stream, stream_config, true); - if (ret) + if (ret) { + /* + * sdw_release_master_stream will release s_rt in slave_rt_list in + * stream_error case, but s_rt is only added to slave_rt_list + * when sdw_config_stream is successful, so free s_rt explicitly + * when sdw_config_stream is failed. + */ + kfree(s_rt); goto stream_error; + } list_add_tail(&s_rt->m_rt_node, &m_rt->slave_rt_list); -- 2.17.1

2 1

[PATCH] soundwire: qcom: wait for fifo space to be available before read/write
by Srinivas Kandagatla 01 Apr '21

01 Apr '21

If we write registers very fast we can endup in a situation where some of the writes will be dropped without any notice. So wait for the fifo space to be available before reading/writing the soundwire registers. Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- drivers/soundwire/qcom.c | 74 ++++++++++++++++++++++++++++++++++++++++ 1 file changed, 74 insertions(+) diff --git a/drivers/soundwire/qcom.c b/drivers/soundwire/qcom.c index 6affa3cd4039..9b45717577f2 100644 --- a/drivers/soundwire/qcom.c +++ b/drivers/soundwire/qcom.c @@ -24,6 +24,8 @@ #define SWRM_COMP_CFG_IRQ_LEVEL_OR_PULSE_MSK BIT(1) #define SWRM_COMP_CFG_ENABLE_MSK BIT(0) #define SWRM_COMP_PARAMS 0x100 +#define SWRM_COMP_PARAMS_WR_FIFO_DEPTH GENMASK(14, 10) +#define SWRM_COMP_PARAMS_RD_FIFO_DEPTH GENMASK(19, 15) #define SWRM_COMP_PARAMS_DOUT_PORTS_MASK GENMASK(4, 0) #define SWRM_COMP_PARAMS_DIN_PORTS_MASK GENMASK(9, 5) #define SWRM_INTERRUPT_STATUS 0x200 @@ -51,6 +53,8 @@ #define SWRM_CMD_FIFO_CMD 0x308 #define SWRM_CMD_FIFO_FLUSH 0x1 #define SWRM_CMD_FIFO_STATUS 0x30C +#define SWRM_RD_CMD_FIFO_CNT_MASK GENMASK(20, 16) +#define SWRM_WR_CMD_FIFO_CNT_MASK GENMASK(12, 8) #define SWRM_CMD_FIFO_CFG_ADDR 0x314 #define SWRM_CONTINUE_EXEC_ON_CMD_IGNORE BIT(31) #define SWRM_RD_WR_CMD_RETRIES 0x7 @@ -104,6 +108,7 @@ #define SWR_BROADCAST_CMD_ID 0x0F #define SWR_MAX_CMD_ID 14 #define MAX_FIFO_RD_RETRY 3 +#define SWR_OVERFLOW_RETRY_COUNT 30 struct qcom_swrm_port_config { u8 si; @@ -147,6 +152,8 @@ struct qcom_swrm_ctrl { int (*reg_read)(struct qcom_swrm_ctrl *ctrl, int reg, u32 *val); int (*reg_write)(struct qcom_swrm_ctrl *ctrl, int reg, int val); u32 slave_status; + u32 wr_fifo_depth; + u32 rd_fifo_depth; }; struct qcom_swrm_data { @@ -238,6 +245,63 @@ static u32 swrm_get_packed_reg_val(u8 *cmd_id, u8 cmd_data, return val; } +static int swrm_wait_for_rd_fifo_avail(struct qcom_swrm_ctrl *swrm) +{ + u32 fifo_outstanding_cmd, value; + u8 fifo_retry_count = SWR_OVERFLOW_RETRY_COUNT; + + /* Check for fifo underflow during read */ + swrm->reg_read(swrm, SWRM_CMD_FIFO_STATUS, &value); + fifo_outstanding_cmd = FIELD_GET(SWRM_RD_CMD_FIFO_CNT_MASK, value); + + /* Check number of outstanding commands in fifo before read */ + if (fifo_outstanding_cmd) + return 0; + + do { + usleep_range(500, 510); + swrm->reg_read(swrm, SWRM_CMD_FIFO_STATUS, &value); + fifo_outstanding_cmd = FIELD_GET(SWRM_RD_CMD_FIFO_CNT_MASK, value); + if (fifo_outstanding_cmd > 0) + break; + } while (fifo_retry_count--); + + if (fifo_outstanding_cmd == 0) { + dev_err_ratelimited(swrm->dev, "%s err read underflow\n", __func__); + return -ENOMEM; + } + + return 0; +} + +static int swrm_wait_for_wr_fifo_avail(struct qcom_swrm_ctrl *swrm) +{ + u32 fifo_outstanding_cmd, value; + u8 fifo_retry_count = SWR_OVERFLOW_RETRY_COUNT; + + /* Check for fifo overflow during write */ + swrm->reg_read(swrm, SWRM_CMD_FIFO_STATUS, &value); + fifo_outstanding_cmd = FIELD_GET(SWRM_WR_CMD_FIFO_CNT_MASK, value); + + /* Check number of outstanding commands in fifo before write */ + if (fifo_outstanding_cmd != swrm->wr_fifo_depth) + return 0; + + do { + usleep_range(500, 510); + swrm->reg_read(swrm, SWRM_CMD_FIFO_STATUS, &value); + fifo_outstanding_cmd = FIELD_GET(SWRM_WR_CMD_FIFO_CNT_MASK, value); + if (fifo_outstanding_cmd < swrm->wr_fifo_depth) + break; + } while (fifo_retry_count--); + + if (fifo_outstanding_cmd == swrm->wr_fifo_depth) { + dev_err_ratelimited(swrm->dev, "%s err write overflow\n", __func__); + return -ENOMEM; + } + + return 0; +} static int qcom_swrm_cmd_fifo_wr_cmd(struct qcom_swrm_ctrl *swrm, u8 cmd_data, u8 dev_addr, u16 reg_addr) @@ -256,6 +320,9 @@ static int qcom_swrm_cmd_fifo_wr_cmd(struct qcom_swrm_ctrl *swrm, u8 cmd_data, dev_addr, reg_addr); } + if (swrm_wait_for_wr_fifo_avail(swrm)) + return SDW_CMD_FAIL_OTHER; + /* Its assumed that write is okay as we do not get any status back */ swrm->reg_write(swrm, SWRM_CMD_FIFO_WR_CMD, val); @@ -295,6 +362,9 @@ static int qcom_swrm_cmd_fifo_rd_cmd(struct qcom_swrm_ctrl *swrm, /* wait for FIFO RD CMD complete to avoid overflow */ usleep_range(250, 255); + if (swrm_wait_for_rd_fifo_avail(swrm)) + return SDW_CMD_FAIL_OTHER; + do { swrm->reg_read(swrm, SWRM_CMD_FIFO_RD_FIFO_ADDR, &cmd_data); rval[0] = cmd_data & 0xFF; @@ -586,6 +656,10 @@ static int qcom_swrm_init(struct qcom_swrm_ctrl *ctrl) SWRM_INTERRUPT_STATUS_RMSK); } ctrl->slave_status = 0; + ctrl->reg_read(ctrl, SWRM_COMP_PARAMS, &val); + ctrl->rd_fifo_depth = FIELD_GET(SWRM_COMP_PARAMS_RD_FIFO_DEPTH, val); + ctrl->wr_fifo_depth = FIELD_GET(SWRM_COMP_PARAMS_WR_FIFO_DEPTH, val); + return 0; } -- 2.21.0

3 2

[PATCH 0/2] kbl_da7219_max9357a machine changes for wov and MST
by vamshi.krishna.gopal＠intel.com 31 Mar '21

31 Mar '21

From: Vamshi Krishna Gopal <vamshi.krishna.gopal(a)intel.com> Hello, This patch series about creating dailink for Wake on voice functionality and also adding MST route changes Mac Chiang (1): ASoc: Intel: board: add BE DAI link for WoV Vamshi Krishna Gopal (1): ASoC: Intel: kbl: Add MST route change to kbl machine drivers sound/soc/intel/boards/kbl_da7219_max98357a.c | 42 +++++++++++++++++-- 1 file changed, 38 insertions(+), 4 deletions(-) -- 2.17.1

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[PATCH][next] soundwire: qcom: Fix a u8 comparison with less than zero
by Colin King 31 Mar '21

31 Mar '21

From: Colin Ian King <colin.king(a)canonical.com> Variable devnum is being checked for a less than zero error return however the comparison will always be false because devnum is an 8 bit unsigned integer. Fix this by making devnum an int. Also there is no need to iniitialize devnum with zero as this value is no read, so remove the redundant assignment. Addresses-Coverity: ("Unsigned compared against 0") Fixes: c7d49c76d1d5 ("soundwire: qcom: add support to new interrupts") Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- 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 b08ecb9b418c..ec86c4e53fdb 100644 --- a/drivers/soundwire/qcom.c +++ b/drivers/soundwire/qcom.c @@ -428,7 +428,7 @@ static irqreturn_t qcom_swrm_irq_handler(int irq, void *dev_id) struct qcom_swrm_ctrl *swrm = dev_id; u32 value, intr_sts, intr_sts_masked, slave_status; u32 i; - u8 devnum = 0; + int devnum; int ret = IRQ_HANDLED; swrm->reg_read(swrm, SWRM_INTERRUPT_STATUS, &intr_sts); -- 2.30.2

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[PATCH v2 1/2] ASoC: tlv320aic32x4: Increase maximum register in regmap
by Annaliese McDermond 31 Mar '21

31 Mar '21

AIC32X4_REFPOWERUP was added as a register, but the maximum register value in the regmap and regmap range was not correspondingly increased. This caused an error when this register was attempted to be written. Fixes: ec96690de82c ("ASoC: tlv320aic32x4: Enable fast charge") Cc: stable(a)vger.kernel.org Signed-off-by: Annaliese McDermond <nh6z(a)nh6z.net> --- sound/soc/codecs/tlv320aic32x4.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index f04f88c8d425..1ac3b3b12dc6 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -577,12 +577,12 @@ static const struct regmap_range_cfg aic32x4_regmap_pages[] = { .window_start = 0, .window_len = 128, .range_min = 0, - .range_max = AIC32X4_RMICPGAVOL, + .range_max = AIC32X4_REFPOWERUP, }, }; const struct regmap_config aic32x4_regmap_config = { - .max_register = AIC32X4_RMICPGAVOL, + .max_register = AIC32X4_REFPOWERUP, .ranges = aic32x4_regmap_pages, .num_ranges = ARRAY_SIZE(aic32x4_regmap_pages), }; -- 2.27.0

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[PATCH v2 2/2] ASoC: tlv320aic32x4: Register clocks before registering component
by Annaliese McDermond 31 Mar '21

31 Mar '21

Clock registration must be performed before the component is registered. aic32x4_component_probe attempts to get all the clocks right off the bat. If the component is registered before the clocks there is a race condition where the clocks may not be registered by the time aic32x4_componet_probe actually runs. Fixes: d1c859d314d8 ("ASoC: codec: tlv3204: Increased maximum supported channels") Cc: stable(a)vger.kernel.org Signed-off-by: Annaliese McDermond <nh6z(a)nh6z.net> --- sound/soc/codecs/tlv320aic32x4.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index 1ac3b3b12dc6..b689f26fc4be 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -1243,6 +1243,10 @@ int aic32x4_probe(struct device *dev, struct regmap *regmap) if (ret) goto err_disable_regulators; + ret = aic32x4_register_clocks(dev, aic32x4->mclk_name); + if (ret) + goto err_disable_regulators; + ret = devm_snd_soc_register_component(dev, &soc_component_dev_aic32x4, &aic32x4_dai, 1); if (ret) { @@ -1250,10 +1254,6 @@ int aic32x4_probe(struct device *dev, struct regmap *regmap) goto err_disable_regulators; } - ret = aic32x4_register_clocks(dev, aic32x4->mclk_name); - if (ret) - goto err_disable_regulators; - return 0; err_disable_regulators: -- 2.27.0

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re: ALSA: control - add layer registration routines
by Colin Ian King 31 Mar '21

31 Mar '21

Hi, Static analysis on linux-next with Coverity has detected a potential issue in the following commit: commit 3f0638a0333bfdd0549985aa620f2ab69737af47 Author: Jaroslav Kysela <perex(a)perex.cz> Date: Wed Mar 17 18:29:41 2021 +0100 ALSA: control - add layer registration routines The static analysis is as follows: 2072 void snd_ctl_disconnect_layer(struct snd_ctl_layer_ops *lops) 2073 { 2074 struct snd_ctl_layer_ops *lops2, *prev_lops2; 2075 2076 down_write(&snd_ctl_layer_rwsem); assignment: Assigning: prev_lops2 = NULL. 2077 for (lops2 = snd_ctl_layer, prev_lops2 = NULL; lops2; lops2 = lops2->next) 2078 if (lops2 == lops) { null: At condition prev_lops2, the value of prev_lops2 must be NULL. dead_error_condition: The condition !prev_lops2 must be true. 2079 if (!prev_lops2) 2080 snd_ctl_layer = lops->next; 2081 else 'Constant' variable guards dead code (DEADCODE) dead_error_line: Execution cannot reach this statement: prev_lops2->next = lops->next;. Local variable prev_lops2 is assigned only once, to a constant value, making it effectively constant throughout its scope. If this is not the intent, examine the logic to see if there is a missing assignment that would make prev_lops2 not remain constant. 2082 prev_lops2->next = lops->next; 2083 break; 2084 } 2085 up_write(&snd_ctl_layer_rwsem); 2086 } I couldn't quite figure out the original intent of the prev_lops use, so I'd thought I'd report this issue as the code does look incorrect. Colin

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[PATCH V2] soundwire: qcom: use signed variable for error return
by Vinod Koul 31 Mar '21

31 Mar '21

We get warning of using a unsigned variable being compared to less than zero. The comparison is correct as it checks for errors from previous call to qcom_swrm_get_alert_slave_dev_num(), so we should use a signed variable here. While at it, drop the superfluous initialization as well drivers/soundwire/qcom.c: qcom_swrm_irq_handler() warn: impossible condition '(devnum < 0) => (0-255 < 0)' Reported-by: kernel test robot <lkp(a)intel.com> Signed-off-by: Vinod Koul <vkoul(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 b08ecb9b418c..ec86c4e53fdb 100644 --- a/drivers/soundwire/qcom.c +++ b/drivers/soundwire/qcom.c @@ -428,7 +428,7 @@ static irqreturn_t qcom_swrm_irq_handler(int irq, void *dev_id) struct qcom_swrm_ctrl *swrm = dev_id; u32 value, intr_sts, intr_sts_masked, slave_status; u32 i; - u8 devnum = 0; + int devnum; int ret = IRQ_HANDLED; swrm->reg_read(swrm, SWRM_INTERRUPT_STATUS, &intr_sts); -- 2.26.3

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[PATCH] soundwire: qcom: use signed variable for error return
by Vinod Koul 31 Mar '21

31 Mar '21

We get warning for using a unsigned variable being compared to less than zero. The comparison is correct as it checks for errors from previous call to qcom_swrm_get_alert_slave_dev_num(), so we should use a signed variable instead. drivers/soundwire/qcom.c: qcom_swrm_irq_handler() warn: impossible condition '(devnum < 0) => (0-255 < 0)' Reported-by: kernel test robot <lkp(a)intel.com> Signed-off-by: Vinod Koul <vkoul(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 b08ecb9b418c..55ed133c6704 100644 --- a/drivers/soundwire/qcom.c +++ b/drivers/soundwire/qcom.c @@ -428,7 +428,7 @@ static irqreturn_t qcom_swrm_irq_handler(int irq, void *dev_id) struct qcom_swrm_ctrl *swrm = dev_id; u32 value, intr_sts, intr_sts_masked, slave_status; u32 i; - u8 devnum = 0; + s8 devnum = 0; int ret = IRQ_HANDLED; swrm->reg_read(swrm, SWRM_INTERRUPT_STATUS, &intr_sts); -- 2.26.3

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Re: [PATCH] ASoC: sunxi: sun4i-codec: fill ASoC card owner
by Mark Brown 31 Mar '21

31 Mar '21

On Wed, Mar 31, 2021 at 04:59:15PM +0200, bage(a)linutronix.de wrote: > From: Bastian Germann <bage(a)linutronix.de> > > card->owner is a required property and since commit 81033c6b584b ("ALSA: > core: Warn on empty module") a warning is issued if it is empty. Add it. > This fixes following warning observed on Lamobo R1: > > WARNING: CPU: 1 PID: 190 at sound/core/init.c:207 snd_card_new+0x430/0x480 [snd] > Modules linked in: sun4i_codec(E+) sun4i_backend(E+) snd_soc_core(E) ... > CPU: 1 PID: 190 Comm: systemd-udevd Tainted: G C E 5.10.0-1-armmp #1 Debian 5.10.4-1 > Hardware name: Allwinner sun7i (A20) Family > [<c0ce8544>] (dump_backtrace) from [<c0ce88f0>] (show_stack+0x20/0x24) > [<c0ce88d0>] (show_stack) from [<c0ceda20>] (dump_stack+0xc8/0xdc) > [<c0ced958>] (dump_stack) from [<c034d49c>] (__warn+0xfc/0x158) > [<c034d3a0>] (__warn) from [<c0ce966c>] (warn_slowpath_fmt+0x70/0xe4) Please think hard before including complete backtraces in upstream reports, they are very large and contain almost no useful information relative to their size so often obscure the relevant content in your message. If part of the backtrace is usefully illustrative (it often is for search engines if nothing else) then it's usually better to pull out the relevant sections.

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Re: [PATCH -next] ASoC: cs35l35: Remove unused including <linux/version.h>
by Mark Brown 31 Mar '21

31 Mar '21

On Fri, Mar 26, 2021 at 02:13:31PM +0800, Zheng Yongjun wrote: > Remove including <linux/version.h> that don't need it. This doesn't apply against current code, please check and resend: Applying: ASoC: cs35l35: Remove unused including <linux/version.h> Using index info to reconstruct a base tree... error: patch failed: sound/soc/codecs/cs35l35.c:9 error: sound/soc/codecs/cs35l35.c: patch does not apply error: Did you hand edit your patch? It does not apply to blobs recorded in its index. Patch failed at 0006 ASoC: cs35l35: Remove unused including <linux/version.h>

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Re: [PATCH -next] ASoC: cs42l42: Remove unused including <linux/version.h>
by Mark Brown 31 Mar '21

31 Mar '21

On Fri, Mar 26, 2021 at 02:13:35PM +0800, Zheng Yongjun wrote: > Remove including <linux/version.h> that don't need it. This doesn't apply against current code, please check and resend: Applying: ASoC: cs42l42: Remove unused including <linux/version.h> Using index info to reconstruct a base tree... error: patch failed: sound/soc/codecs/cs42l42.c:11 error: sound/soc/codecs/cs42l42.c: patch does not apply error: Did you hand edit your patch? It does not apply to blobs recorded in its index. Patch failed at 0006 ASoC: cs42l42: Remove unused including <linux/version.h>

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Bus error with alsaloop
by GitHub issues - opened 31 Mar '21

31 Mar '21

alsa-project/alsa-utils issue #85 was opened from philipp-fischer: ## Behaviour alsaloop crashes with a bus error (SIGBUS) immediately ``` pi@pi-snapserver:~ $ alsaloop -C hw:2,0 -P hw:3,0 Bus error ``` ## Analysis and debugging Very verbose output: ``` pi@pi-snapserver:~ $ alsaloop -C hw:2,0 -P hw:3,0 -vvvvvvvvvvvvvvvvv !!!Scheduler set to Round Robin with priority 99 FAILED! playback hw:3,0: buffer_size=1920 playback hw:3,0: period_size=240 capture hw:2,0: buffer_size=1920 capture hw:2,0: period_size=240 playback hw:3,0: avail_min2=1560 capture hw:2,0: avail_min2=120 Hardware PCM card 3 'ICUSBAUDIO7D' device 0 subdevice 0 Its setup is: stream : PLAYBACK access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 48000 exact rate : 48000 (48000/1) msbits : 16 buffer_size : 1920 period_size : 240 period_time : 5000 tstamp_mode : NONE tstamp_type : MONOTONIC period_step : 1 avail_min : 1560 period_event : 0 start_threshold : 2147483647 stop_threshold : 1920 silence_threshold: 0 silence_size : 0 boundary : 2013265920 appl_ptr : 0 hw_ptr : 0 Hardware PCM card 2 'ICUSBAUDIO7D' device 0 subdevice 0 Its setup is: stream : CAPTURE access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 48000 exact rate : 48000 (48000/1) msbits : 16 buffer_size : 1920 period_size : 240 period_time : 5000 tstamp_mode : NONE tstamp_type : MONOTONIC period_step : 1 avail_min : 240 period_event : 0 start_threshold : 2147483647 stop_threshold : 1920 silence_threshold: 0 silence_size : 0 boundary : 2013265920 appl_ptr : 0 hw_ptr : 0 Latency 480 frames, 10000.000us, 10.000000ms (100.0000Hz) playback hw:3,0/capture hw:2,0 sync type: SAMPLERATE (SRC_SINC_FASTEST) playback hw:3,0/capture hw:2,0: capt->buffer_size = 3840, play->buffer_size = 3840 playback hw:3,0/capture hw:2,0: Samplerate src_ratio update1: 1.00000000 New pitch for playback hw:3,0/capture hw:2,0: 1.00000000 (min/max samples = 0/0) playback hw:3,0/capture hw:2,0: silence queued 480 samples pool took 3074us playback hw:3,0/capture hw:2,0: pollfds handle playback hw:3,0: delay 480 / 3840 / 0 capture hw:2,0: delay 0 / 3840 / 0 playback hw:3,0/capture hw:2,0: prevents = 0x4, crevents = 0x0 playback hw:3,0/capture hw:2,0: queued 480/0 samples playback hw:3,0: end delay 480 / 3840 / 0 capture hw:2,0: end delay 0 / 3840 / 0 playback hw:3,0/capture hw:2,0: processing time 59us pool took 13us [...] playback hw:3,0/capture hw:2,0: pollfds handle playback hw:3,0: delay 323 / 3840 / 0 capture hw:2,0: delay 0 / 3840 / 0 playback hw:3,0/capture hw:2,0: prevents = 0x4, crevents = 0x0 playback hw:3,0/capture hw:2,0: queued 323/0 samples playback hw:3,0: end delay 323 / 3840 / 0 capture hw:2,0: end delay 0 / 3840 / 0 playback hw:3,0/capture hw:2,0: processing time 212us pool took 5us playback hw:3,0/capture hw:2,0: pollfds handle playback hw:3,0: delay 323 / 3840 / 0 capture hw:2,0: delay 48 / 3840 / 0 playback hw:3,0/capture hw:2,0: prevents = 0x4, crevents = 0x0 Bus error ``` Debugging with backtrace (binary without dbg symbols): ``` [...] Program received signal SIGBUS, Bus error. 0xb6dbd394 in src_short_to_float_array () from /lib/arm-linux-gnueabihf/libsamplerate.so.0 (gdb) bt #0 0xb6dbd394 in src_short_to_float_array () from /lib/arm-linux-gnueabihf/libsamplerate.so.0 #1 0x00016a20 in ?? () Backtrace stopped: previous frame identical to this frame (corrupt stack?) (gdb) ``` So the error seems to happen in `libsamplerate.so` Switching the sync method to "SIMPLE" prevents the error from happening as alsaloop doesn't seem to make use of libsamplerate in this case. The following modes run without failing but don't give the desired results: ``` pi@pi-snapserver:~ $ alsaloop -C hw:2,0 -P hw:3,0 -S 0 # SYNC_TYPE_NONE pi@pi-snapserver:~ $ alsaloop -C hw:2,0 -P hw:3,0 -S 1 # SYNC_TYPE_SIMPLE pi@pi-snapserver:~ $ alsaloop -C hw:2,0 -P hw:3,0 -S 2 # SYNC_TYPE_CAPTRATESHIFT pi@pi-snapserver:~ $ alsaloop -C hw:2,0 -P hw:3,0 -S 3 # SYNC_TYPE_PLAYRATESHIFT ``` Only `SYNC_TYPE_SAMPLERATE` and `SYNC_TYPE_AUTO` (which selects SYNC_TYPE_SAMPLERATE) fail. The modes `SYNC_TYPE_NONE` and `SYNC_TYPE_SIMPLE` don't crash for me, but they don't work and result in increasing latency until the fifos reset. (Which is another issue I guess) Issue URL : https://github.com/alsa-project/alsa-utils/issues/85 Repository URL: https://github.com/alsa-project/alsa-utils

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