June 2017 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: rt5645: del btn_check_timer on remove
by Hans de Goede 28 Jun '17

28 Jun '17

The timer may still be running when rt5645_i2c_remove() gets called, call del_timer_sync() to make sure it is stopped. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- sound/soc/codecs/rt5645.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index d8e7487482c2..e6a3ad17437c 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3915,6 +3915,7 @@ static int rt5645_i2c_remove(struct i2c_client *i2c) cancel_delayed_work_sync(&rt5645->jack_detect_work); cancel_delayed_work_sync(&rt5645->rcclock_work); + del_timer_sync(&rt5645->btn_check_timer); snd_soc_unregister_codec(&i2c->dev); regulator_bulk_disable(ARRAY_SIZE(rt5645->supplies), rt5645->supplies); -- 2.13.0

2 1

[alsa-devel] [PATCH 1/3] ASoC: rt5670: fix wrong audio route
by Bard Liao 28 Jun '17

28 Jun '17

When we select "IF1_ADC4" for IF1 ADC Muxes, it is actually connected to "TxDP_ADC" widget. This patch fixes the audio route and remove the unexisting "IF1_ADC4" widget. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5670.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c index e27c5a4..d95d2e6 100644 --- a/sound/soc/codecs/rt5670.c +++ b/sound/soc/codecs/rt5670.c @@ -1717,7 +1717,6 @@ static const struct snd_soc_dapm_widget rt5670_dapm_widgets[] = { SND_SOC_DAPM_PGA("IF1_ADC1", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_PGA("IF1_ADC2", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_PGA("IF1_ADC3", SND_SOC_NOPM, 0, 0, NULL, 0), - SND_SOC_DAPM_PGA("IF1_ADC4", SND_SOC_NOPM, 0, 0, NULL, 0), /* DSP */ SND_SOC_DAPM_PGA("TxDP_ADC", SND_SOC_NOPM, 0, 0, NULL, 0), @@ -2086,13 +2085,13 @@ static const struct snd_soc_dapm_route rt5670_dapm_routes[] = { { "IF1 ADC1 IN1 Mux", "IF1_ADC3", "IF1_ADC3" }, { "IF1 ADC1 IN2 Mux", "IF1_ADC1_IN1", "IF1 ADC1 IN1 Mux" }, - { "IF1 ADC1 IN2 Mux", "IF1_ADC4", "IF1_ADC4" }, + { "IF1 ADC1 IN2 Mux", "IF1_ADC4", "TxDP_ADC" }, { "IF1 ADC2 IN Mux", "IF_ADC2", "IF_ADC2" }, { "IF1 ADC2 IN Mux", "VAD_ADC", "VAD_ADC" }, { "IF1 ADC2 IN1 Mux", "IF1_ADC2_IN", "IF1 ADC2 IN Mux" }, - { "IF1 ADC2 IN1 Mux", "IF1_ADC4", "IF1_ADC4" }, + { "IF1 ADC2 IN1 Mux", "IF1_ADC4", "TxDP_ADC" }, { "IF1_ADC1" , NULL, "IF1 ADC1 IN2 Mux" }, { "IF1_ADC2" , NULL, "IF1 ADC2 IN1 Mux" }, -- 2.7.4

2 5

[alsa-devel] [PATCH] ASoC: rt5651: remove unexisting Muxes
by Bard Liao 28 Jun '17

28 Jun '17

These MUXes are unexisting. So, remove them. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5651.c | 44 +------------------------------------------- 1 file changed, 1 insertion(+), 43 deletions(-) diff --git a/sound/soc/codecs/rt5651.c b/sound/soc/codecs/rt5651.c index f5d3415..db05b60 100644 --- a/sound/soc/codecs/rt5651.c +++ b/sound/soc/codecs/rt5651.c @@ -586,44 +586,6 @@ static const struct snd_kcontrol_new hpo_r_mute_control = SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT5651_HP_VOL, RT5651_R_MUTE_SFT, 1, 1); -/* INL/R source */ -static const char * const rt5651_inl_src[] = {"IN2P", "HPOVOLLP"}; - -static SOC_ENUM_SINGLE_DECL( - rt5651_inl_enum, RT5651_INL1_INR1_VOL, - RT5651_INL_SEL_SFT, rt5651_inl_src); - -static const struct snd_kcontrol_new rt5651_inl1_mux = - SOC_DAPM_ENUM("INL1 source", rt5651_inl_enum); - -static const char * const rt5651_inr1_src[] = {"IN2N", "HPOVOLRP"}; - -static SOC_ENUM_SINGLE_DECL( - rt5651_inr1_enum, RT5651_INL1_INR1_VOL, - RT5651_INR_SEL_SFT, rt5651_inr1_src); - -static const struct snd_kcontrol_new rt5651_inr1_mux = - SOC_DAPM_ENUM("INR1 source", rt5651_inr1_enum); - -static const char * const rt5651_inl2_src[] = {"IN3P", "OUTVOLLP"}; - -static SOC_ENUM_SINGLE_DECL( - rt5651_inl2_enum, RT5651_INL2_INR2_VOL, - RT5651_INL_SEL_SFT, rt5651_inl2_src); - -static const struct snd_kcontrol_new rt5651_inl2_mux = - SOC_DAPM_ENUM("INL2 source", rt5651_inl2_enum); - -static const char * const rt5651_inr2_src[] = {"IN3N", "OUTVOLRP"}; - -static SOC_ENUM_SINGLE_DECL( - rt5651_inr2_enum, RT5651_INL2_INR2_VOL, - RT5651_INR_SEL_SFT, rt5651_inr2_src); - -static const struct snd_kcontrol_new rt5651_inr2_mux = - SOC_DAPM_ENUM("INR2 source", rt5651_inr2_enum); - - /* Stereo ADC source */ static const char * const rt5651_stereo1_adc1_src[] = {"DD MIX", "ADC"}; @@ -955,11 +917,7 @@ static const struct snd_soc_dapm_widget rt5651_dapm_widgets[] = { RT5651_PWR_IN2_L_BIT, 0, NULL, 0), SND_SOC_DAPM_PGA("INR2 VOL", RT5651_PWR_VOL, RT5651_PWR_IN2_R_BIT, 0, NULL, 0), - /* IN Mux */ - SND_SOC_DAPM_MUX("INL1 Mux", SND_SOC_NOPM, 0, 0, &rt5651_inl1_mux), - SND_SOC_DAPM_MUX("INR1 Mux", SND_SOC_NOPM, 0, 0, &rt5651_inr1_mux), - SND_SOC_DAPM_MUX("INL2 Mux", SND_SOC_NOPM, 0, 0, &rt5651_inl2_mux), - SND_SOC_DAPM_MUX("INR2 Mux", SND_SOC_NOPM, 0, 0, &rt5651_inr2_mux), + /* REC Mixer */ SND_SOC_DAPM_MIXER("RECMIXL", RT5651_PWR_MIXER, RT5651_PWR_RM_L_BIT, 0, rt5651_rec_l_mix, ARRAY_SIZE(rt5651_rec_l_mix)), -- 2.7.4

2 1

[alsa-devel] [PATCH v2] ASoC: rt5645: enable speaker protection features
by Bard Liao 28 Jun '17

28 Jun '17

This patch is uploaded for enabling the speaker protection features of the audio codec. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- v2: * modify register value by regmap_update_bits instead of regmap_register_patch --- sound/soc/codecs/rt5645.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 162044d..627e801 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -59,7 +59,7 @@ static const struct regmap_range_cfg rt5645_ranges[] = { static const struct reg_sequence init_list[] = { {RT5645_PR_BASE + 0x3d, 0x3600}, - {RT5645_PR_BASE + 0x1c, 0xfd20}, + {RT5645_PR_BASE + 0x1c, 0xfd70}, {RT5645_PR_BASE + 0x20, 0x611f}, {RT5645_PR_BASE + 0x21, 0x4040}, {RT5645_PR_BASE + 0x23, 0x0004}, @@ -3730,6 +3730,8 @@ static int rt5645_i2c_probe(struct i2c_client *i2c, ret); } + regmap_update_bits(rt5645->regmap, RT5645_CLSD_OUT_CTRL, 0xc0, 0xc0); + if (rt5645->pdata.in2_diff) regmap_update_bits(rt5645->regmap, RT5645_IN2_CTRL, RT5645_IN_DF2, RT5645_IN_DF2); -- 2.7.4

2 1

[alsa-devel] [PATCH v13 00/14] ASoC: sun4i-codec: Add Line-In, FM-In, Mic 2, Capture Source, Differential Line-In
by Danny Milosavljevic 28 Jun '17

28 Jun '17

This patchset adds some mixer controls to sun4i-codec for the Allwinner A10 and the Allwinner A20. It also adds a mux for the capture source and the PGA for the MIC2 preamp. Where possible, it uses SOC_DAPM_DOUBLE in order to cut down on the number of distinct controls in alsamixer. v13 changes compared to v12 are: - Added my "Signed-off-by". - Clarified some commit message text. v12 changes compared to v11 are: - Split up patchset in another way. - Renamed "Mic1 Capture Volume" to "Mic1 Boost Volume". - Renamed "Mic2 Capture Volume" to "Mic2 Boost Volume". - Renamed "Line Capture Volume" to "Line Boost Volume". - Renamed "Differential Line Capture Switch" to "Differential Line Source". v11 changes compared to v10 are: - Split up patchset. - Fixed typo in Differential Line Capture Switch. - Renamed "Non-Differential" value to "Stereo". - Removed duplicate PA Volume mixer control. v10 changes compared to v9 are: - Use SOC_DAPM_DOUBLE where possible and it makes sense in order to cut down on the number of controls. v9 changes compared to v8 are: - added Line Differential Capture Switch. - split Capture Source into Left Capture Select, Right Capture Select. - added Line Capture Volume. - rename "sun4i_codec_widgets" to "sun4i_codec_controls" for consistency with the struct field it's used in. - rename "Line-In" to "Line". - rename "Power Amplifier Playback Volume" to "Headphone Playback Volume". v8 changes compared to v7 are: - fixed the routes for line and mic capturing. v7 changes compared to v6 are: - preparation for different A20, A10 controls is now in an extra patch. - all register definitions are now at the top. - sun7i-specific things (A20-specific things) are now less grouped-together. - rename "Power Amplifier Volume" to "Power Amplifier Playback Volume". v6 changes compared to v5 are: - Mic preamplifier special cases for A20 and A10 now are now not icky: There are two different _widget arrays and the probe() function now selects the right one to pass to snd_soc_register_codec() unmodified. - sun7i-specific things (A20-specific things) are now grouped together. v5 changes compared to v4 are: - Mic preamplifier controls have more common names now. - Mic preamplifier scale has a 0 dB entry as well now, as documented in the A20 user manual. - Mic preamplifier has special cases for A20 and A10 now. - Gain controls have "Gain" in the name now. v4 changes compared to v3 are: - names of the input are not uppercase anymore. - bit index constants are now named as in the A20 user manual v1.4. - added Mic1-In, Mac2-In. - added Mic1 and Mic2 Pre-Amplifiers. v3 changes compared to v2 are: - added DAPM routes. v2 changes compared to v1 are: - moved Line-In and FM-In playback switches to their respective sun4i_codec_*_mixer_controls. v1 changes: - added linein, fmin output volumes and switches. Danny Milosavljevic (14): sun4i-codec: Add MIC2 Pre-Amplifier, Mic2 input, Mic2 routes. sun4i-codec: Add Mic Playback Volume. sun4i-codec: Add support for extra controls to struct sun4i_codec_quirks and use them. sun4i-codec: Add Mic1 Boost Volume, Mic2 Boost Volume. sun4i-codec: Merge sun4i_codec_left_mixer_controls and sun4i_codec_right_mixer_controls into sun4i_codec_mixer_controls. sun4i-codec: Add Mic1 Playback Switch, Mic2 Playback Switch. sun4i-codec: Add FM Playback Volume. sun4i-codec: Add FM Left, FM Right, FM Playback Switch. sun4i-codec: Add Line Playback Volume. sun4i-codec: Add Line Boost Volume. sun4i-codec: Add Line Right, Line Left, Line Playback Switch. sun4i-codec: Add Differential Line Source. sun4i-codec: Add Left Capture Select, Right Capture Select. sun4i-codec: Add Capture Volume. sound/soc/sunxi/sun4i-codec.c | 233 +++++++++++++++++++++++++++++++++++++----- 1 file changed, 210 insertions(+), 23 deletions(-) -- 2.1.4

5 21

[alsa-devel] [PATCH] ASoC: arizona: Add DAC Volume Limit controls
by Charles Keepax 28 Jun '17

28 Jun '17

Add controls that allow the limiting of the volume coming from the DACs on the Arizona class devices, note that wm8998 does not have this feature. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- sound/soc/codecs/cs47l24.c | 8 ++++++++ sound/soc/codecs/wm5102.c | 17 +++++++++++++++++ sound/soc/codecs/wm5110.c | 20 ++++++++++++++++++++ sound/soc/codecs/wm8997.c | 14 ++++++++++++++ 4 files changed, 59 insertions(+) diff --git a/sound/soc/codecs/cs47l24.c b/sound/soc/codecs/cs47l24.c index 47e6fddef92b..a972e63bdda5 100644 --- a/sound/soc/codecs/cs47l24.c +++ b/sound/soc/codecs/cs47l24.c @@ -78,6 +78,7 @@ static int cs47l24_adsp_power_ev(struct snd_soc_dapm_widget *w, static DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0); static DECLARE_TLV_DB_SCALE(digital_tlv, -6400, 50, 0); +static DECLARE_TLV_DB_SCALE(vol_limit_tlv, -600, 50, 0); static DECLARE_TLV_DB_SCALE(noise_tlv, -13200, 600, 0); static DECLARE_TLV_DB_SCALE(ng_tlv, -10200, 600, 0); @@ -203,6 +204,13 @@ SOC_SINGLE_TLV("Speaker Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_4L, ARIZONA_OUT4L_VOL_SHIFT, 0xbf, 0, digital_tlv), +SOC_DOUBLE_R_RANGE_TLV("HPOUT1 Volume Limit", ARIZONA_DAC_VOLUME_LIMIT_1L, + ARIZONA_DAC_VOLUME_LIMIT_1R, ARIZONA_OUT1L_VOL_LIM_SHIFT, + 0x74, 0x8C, 0, vol_limit_tlv), +SOC_DOUBLE_R_RANGE_TLV("Speaker Volume Limit", ARIZONA_OUT_VOLUME_4L, + ARIZONA_OUT_VOLUME_4R, ARIZONA_OUT4L_VOL_LIM_SHIFT, + 0x74, 0x8C, 0, vol_limit_tlv), + SOC_ENUM("Output Ramp Up", arizona_out_vi_ramp), SOC_ENUM("Output Ramp Down", arizona_out_vd_ramp), diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index 1fe358e6be61..7a295146524b 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -42,6 +42,7 @@ struct wm5102_priv { static DECLARE_TLV_DB_SCALE(ana_tlv, 0, 100, 0); static DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0); static DECLARE_TLV_DB_SCALE(digital_tlv, -6400, 50, 0); +static DECLARE_TLV_DB_SCALE(vol_limit_tlv, -600, 50, 0); static DECLARE_TLV_DB_SCALE(noise_tlv, -13200, 600, 0); static DECLARE_TLV_DB_SCALE(ng_tlv, -10200, 600, 0); @@ -915,6 +916,22 @@ SOC_DOUBLE_R_TLV("SPKDAT1 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_5L, ARIZONA_DAC_DIGITAL_VOLUME_5R, ARIZONA_OUT5L_VOL_SHIFT, 0xbf, 0, digital_tlv), +SOC_DOUBLE_R_RANGE_TLV("HPOUT1 Volume Limit", ARIZONA_DAC_VOLUME_LIMIT_1L, + ARIZONA_DAC_VOLUME_LIMIT_1R, ARIZONA_OUT1L_VOL_LIM_SHIFT, + 0x74, 0x8C, 0, vol_limit_tlv), +SOC_DOUBLE_R_RANGE_TLV("HPOUT2 Volume Limit", ARIZONA_DAC_VOLUME_LIMIT_2L, + ARIZONA_DAC_VOLUME_LIMIT_2R, ARIZONA_OUT2L_VOL_LIM_SHIFT, + 0x74, 0x8C, 0, vol_limit_tlv), +SOC_DOUBLE_R_RANGE_TLV("EPOUT Volume Limit", ARIZONA_DAC_VOLUME_LIMIT_3L, + ARIZONA_DAC_VOLUME_LIMIT_3R, ARIZONA_OUT3L_VOL_LIM_SHIFT, + 0x74, 0x8C, 0, vol_limit_tlv), +SOC_DOUBLE_R_RANGE_TLV("Speaker Volume Limit", ARIZONA_OUT_VOLUME_4L, + ARIZONA_OUT_VOLUME_4R, ARIZONA_OUT4L_VOL_LIM_SHIFT, + 0x74, 0x8C, 0, vol_limit_tlv), +SOC_DOUBLE_R_RANGE_TLV("SPKDAT1 Volume Limit", ARIZONA_DAC_VOLUME_LIMIT_5L, + ARIZONA_DAC_VOLUME_LIMIT_5R, ARIZONA_OUT5L_VOL_LIM_SHIFT, + 0x74, 0x8C, 0, vol_limit_tlv), + SOC_ENUM("HPOUT1 OSR", wm5102_hpout_osr[0]), SOC_ENUM("HPOUT2 OSR", wm5102_hpout_osr[1]), SOC_ENUM("EPOUT OSR", wm5102_hpout_osr[2]), diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index 1bc942152eff..f7cf9f7dffa6 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -575,6 +575,7 @@ static int wm5110_in_ev(struct snd_soc_dapm_widget *w, static DECLARE_TLV_DB_SCALE(ana_tlv, 0, 100, 0); static DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0); static DECLARE_TLV_DB_SCALE(digital_tlv, -6400, 50, 0); +static DECLARE_TLV_DB_SCALE(vol_limit_tlv, -600, 50, 0); static DECLARE_TLV_DB_SCALE(noise_tlv, -13200, 600, 0); static DECLARE_TLV_DB_SCALE(ng_tlv, -10200, 600, 0); @@ -855,6 +856,25 @@ SOC_DOUBLE_R_TLV("SPKDAT2 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_6L, ARIZONA_DAC_DIGITAL_VOLUME_6R, ARIZONA_OUT6L_VOL_SHIFT, 0xbf, 0, digital_tlv), +SOC_DOUBLE_R_RANGE_TLV("HPOUT1 Volume Limit", ARIZONA_DAC_VOLUME_LIMIT_1L, + ARIZONA_DAC_VOLUME_LIMIT_1R, ARIZONA_OUT1L_VOL_LIM_SHIFT, + 0x74, 0x8C, 0, vol_limit_tlv), +SOC_DOUBLE_R_RANGE_TLV("HPOUT2 Volume Limit", ARIZONA_DAC_VOLUME_LIMIT_2L, + ARIZONA_DAC_VOLUME_LIMIT_2R, ARIZONA_OUT2L_VOL_LIM_SHIFT, + 0x74, 0x8C, 0, vol_limit_tlv), +SOC_DOUBLE_R_RANGE_TLV("HPOUT3 Volume Limit", ARIZONA_DAC_VOLUME_LIMIT_3L, + ARIZONA_DAC_VOLUME_LIMIT_3R, ARIZONA_OUT3L_VOL_LIM_SHIFT, + 0x74, 0x8C, 0, vol_limit_tlv), +SOC_DOUBLE_R_RANGE_TLV("Speaker Volume Limit", ARIZONA_OUT_VOLUME_4L, + ARIZONA_OUT_VOLUME_4R, ARIZONA_OUT4L_VOL_LIM_SHIFT, + 0x74, 0x8C, 0, vol_limit_tlv), +SOC_DOUBLE_R_RANGE_TLV("SPKDAT1 Volume Limit", ARIZONA_DAC_VOLUME_LIMIT_5L, + ARIZONA_DAC_VOLUME_LIMIT_5R, ARIZONA_OUT5L_VOL_LIM_SHIFT, + 0x74, 0x8C, 0, vol_limit_tlv), +SOC_DOUBLE_R_RANGE_TLV("SPKDAT2 Volume Limit", ARIZONA_DAC_VOLUME_LIMIT_6L, + ARIZONA_DAC_VOLUME_LIMIT_6R, ARIZONA_OUT6L_VOL_LIM_SHIFT, + 0x74, 0x8C, 0, vol_limit_tlv), + SOC_DOUBLE("SPKDAT1 Switch", ARIZONA_PDM_SPK1_CTRL_1, ARIZONA_SPK1L_MUTE_SHIFT, ARIZONA_SPK1R_MUTE_SHIFT, 1, 1), SOC_DOUBLE("SPKDAT2 Switch", ARIZONA_PDM_SPK2_CTRL_1, ARIZONA_SPK2L_MUTE_SHIFT, diff --git a/sound/soc/codecs/wm8997.c b/sound/soc/codecs/wm8997.c index 49401a8aae64..753a1c6a1484 100644 --- a/sound/soc/codecs/wm8997.c +++ b/sound/soc/codecs/wm8997.c @@ -40,6 +40,7 @@ struct wm8997_priv { static DECLARE_TLV_DB_SCALE(ana_tlv, 0, 100, 0); static DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0); static DECLARE_TLV_DB_SCALE(digital_tlv, -6400, 50, 0); +static DECLARE_TLV_DB_SCALE(vol_limit_tlv, -600, 50, 0); static DECLARE_TLV_DB_SCALE(noise_tlv, -13200, 600, 0); static DECLARE_TLV_DB_SCALE(ng_tlv, -10200, 600, 0); @@ -287,6 +288,19 @@ SOC_DOUBLE_R_TLV("SPKDAT1 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_5L, ARIZONA_DAC_DIGITAL_VOLUME_5R, ARIZONA_OUT5L_VOL_SHIFT, 0xbf, 0, digital_tlv), +SOC_DOUBLE_R_RANGE_TLV("HPOUT1 Volume Limit", ARIZONA_DAC_VOLUME_LIMIT_1L, + ARIZONA_DAC_VOLUME_LIMIT_1R, ARIZONA_OUT1L_VOL_LIM_SHIFT, + 0x74, 0x8C, 0, vol_limit_tlv), +SOC_DOUBLE_R_RANGE_TLV("EPOUT Volume Limit", ARIZONA_DAC_VOLUME_LIMIT_3L, + ARIZONA_DAC_VOLUME_LIMIT_3R, ARIZONA_OUT3L_VOL_LIM_SHIFT, + 0x74, 0x8C, 0, vol_limit_tlv), +SOC_DOUBLE_R_RANGE_TLV("Speaker Volume Limit", ARIZONA_OUT_VOLUME_4L, + ARIZONA_OUT_VOLUME_4R, ARIZONA_OUT4L_VOL_LIM_SHIFT, + 0x74, 0x8C, 0, vol_limit_tlv), +SOC_DOUBLE_R_RANGE_TLV("SPKDAT1 Volume Limit", ARIZONA_DAC_VOLUME_LIMIT_5L, + ARIZONA_DAC_VOLUME_LIMIT_5R, ARIZONA_OUT5L_VOL_LIM_SHIFT, + 0x74, 0x8C, 0, vol_limit_tlv), + SOC_ENUM("HPOUT1 OSR", wm8997_hpout_osr[0]), SOC_ENUM("EPOUT OSR", wm8997_hpout_osr[1]), -- 2.11.0

2 2

[alsa-devel] [PATCH 0/4] ALSA: HD-audio HDMI binding fixes
by Takashi Iwai 28 Jun '17

28 Jun '17

Hi, this is a patch serires to address the issue recently reported by Intel gfx people. The first patch is the critical one, and it'll go to for-linus branch, possibly for 4.12, while others are some following works and cleanups. Takashi === Takashi Iwai (4): ALSA: hda - Fix endless loop of codec configure ALSA: hda - Skip card registration when no codec is found ALSA: hda - Bind with i915 component before codec binding ALSA: hda - Minor code refactoring for Intel HDMI codec parsers sound/pci/hda/hda_codec.h | 2 + sound/pci/hda/hda_controller.c | 11 ++++- sound/pci/hda/hda_intel.c | 4 +- sound/pci/hda/patch_hdmi.c | 94 +++++++++++++++--------------------------- 4 files changed, 47 insertions(+), 64 deletions(-) -- 2.13.2

1 5

[alsa-devel] [PATCH] ASoC: stm32: sai: remove some stray tabs
by Dan Carpenter 28 Jun '17

28 Jun '17

This line was accidentally indented too far. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c index ba3fdc777ed8..90d439613899 100644 --- a/sound/soc/stm/stm32_sai_sub.c +++ b/sound/soc/stm/stm32_sai_sub.c @@ -631,7 +631,7 @@ static int stm32_sai_configure_clock(struct snd_soc_dai *cpu_dai, dev_dbg(cpu_dai->dev, "SAI clock %d, divider %d\n", sai_clk_rate, div); mask = SAI_XCR1_MCKDIV_MASK(SAI_XCR1_MCKDIV_WIDTH(version)); - cr1 = SAI_XCR1_MCKDIV_SET(div); + cr1 = SAI_XCR1_MCKDIV_SET(div); ret = regmap_update_bits(sai->regmap, STM_SAI_CR1_REGX, mask, cr1); if (ret < 0) { dev_err(cpu_dai->dev, "Failed to update CR1 register\n");

1 0

[alsa-devel] [PATCH] ALSA: Fix forgotten dependency fix for tristate OSS sequencer kconfig
by Takashi Iwai 28 Jun '17

28 Jun '17

In the commit 3d774d5ef066 ("ALSA: seq: Allow the tristate build of OSS emulation") we changed CONFIG_SND_SEQUENCER_OSS to tristate, but a couple of places were forgotten, namely, opl3 and emux Makefile. These contain the line like snd-opl3-synth-$(CONFIG_SND_SEQUENCER_OSS) += opl3_oss.o and this doesn't work any longer as expected because snd-opl3-synth can be built-in while CONFIG_SND_SEQUENCER_OSS=m. This patch fixes these places to build properly for the new kconfig dependency. In the end, we had to use ifneq() to satisfy the requirement. It's a bit ugly, but lesser evil. Fixes: 3d774d5ef066 ("ALSA: seq: Allow the tristate build of OSS emulation") Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/drivers/opl3/Makefile | 4 +++- sound/synth/emux/Makefile | 4 +++- 2 files changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/drivers/opl3/Makefile b/sound/drivers/opl3/Makefile index 7f2c2a10c4e5..d72b1e7b51c4 100644 --- a/sound/drivers/opl3/Makefile +++ b/sound/drivers/opl3/Makefile @@ -5,7 +5,9 @@ snd-opl3-lib-objs := opl3_lib.o opl3_synth.o snd-opl3-synth-y := opl3_seq.o opl3_midi.o opl3_drums.o -snd-opl3-synth-$(CONFIG_SND_SEQUENCER_OSS) += opl3_oss.o +ifneq ($(CONFIG_SND_SEQUENCER_OSS),) +snd-opl3-synth-y += opl3_oss.o +endif obj-$(CONFIG_SND_OPL3_LIB) += snd-opl3-lib.o obj-$(CONFIG_SND_OPL4_LIB) += snd-opl3-lib.o diff --git a/sound/synth/emux/Makefile b/sound/synth/emux/Makefile index 4599108452fd..d1bac923eb1b 100644 --- a/sound/synth/emux/Makefile +++ b/sound/synth/emux/Makefile @@ -6,6 +6,8 @@ snd-emux-synth-objs := emux.o emux_synth.o emux_seq.o emux_nrpn.o \ emux_effect.o emux_hwdep.o soundfont.o snd-emux-synth-$(CONFIG_SND_PROC_FS) += emux_proc.o -snd-emux-synth-$(CONFIG_SND_SEQUENCER_OSS) += emux_oss.o +ifneq ($(CONFIG_SND_SEQUENCER_OSS),) +snd-emux-synth-y += emux_oss.o +endif obj-$(CONFIG_SND_SYNTH_EMUX) += snd-emux-synth.o -- 2.13.2

1 0

[alsa-devel] ALSA core info race condition
by b_lkasam＠codeaurora.org 28 Jun '17

28 Jun '17

hi ALSA team, there is a race condition in below API when accessing list API. In file sound/core/info.c: Added below patch to avoid list access of same parent node by two threads at same time causing list_debug crash. diff --git a/sound/core/info.c b/sound/core/info.c index b5158b5..c1fd671 100644 --- a/sound/core/info.c +++ b/sound/core/info.c @@ -747,8 +747,11 @@ snd_info_create_entry(const char *name, struct snd_info_entry *parent) INIT_LIST_HEAD(&entry->children); INIT_LIST_HEAD(&entry->list); entry->parent = parent; - if (parent) + if (parent) { + mutex_lock(&parent->access); list_add_tail(&entry->list, &parent->children); + mutex_unlock(&parent->access); + } return entry; } Please check above logic looks fine, and help comment accordingly. Thanks Kasam

2 3