April 2014 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: rt5640: add rt5639 support
by Oder Chiou 08 Apr '14

08 Apr '14

This patch adds the rt5639 support Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5640.c | 286 ++++++++++++++++++++++++++++++++++------------ sound/soc/codecs/rt5640.h | 3 + 2 files changed, 213 insertions(+), 76 deletions(-) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 6b8778f..d672f44 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -399,18 +399,13 @@ static const struct snd_kcontrol_new rt5640_snd_controls[] = { RT5640_VOL_L_SFT, RT5640_VOL_R_SFT, 1, 1), SOC_DOUBLE_TLV("OUT Playback Volume", RT5640_OUTPUT, RT5640_L_VOL_SFT, RT5640_R_VOL_SFT, 39, 1, out_vol_tlv), - /* MONO Output Control */ - SOC_SINGLE("Mono Playback Switch", RT5640_MONO_OUT, - RT5640_L_MUTE_SFT, 1, 1), + /* DAC Digital Volume */ SOC_DOUBLE("DAC2 Playback Switch", RT5640_DAC2_CTRL, RT5640_M_DAC_L2_VOL_SFT, RT5640_M_DAC_R2_VOL_SFT, 1, 1), SOC_DOUBLE_TLV("DAC1 Playback Volume", RT5640_DAC1_DIG_VOL, RT5640_L_VOL_SFT, RT5640_R_VOL_SFT, 175, 0, dac_vol_tlv), - SOC_DOUBLE_TLV("Mono DAC Playback Volume", RT5640_DAC2_DIG_VOL, - RT5640_L_VOL_SFT, RT5640_R_VOL_SFT, - 175, 0, dac_vol_tlv), /* IN1/IN2 Control */ SOC_SINGLE_TLV("IN1 Boost", RT5640_IN1_IN2, RT5640_BST_SFT1, 8, 0, bst_tlv), @@ -442,6 +437,15 @@ static const struct snd_kcontrol_new rt5640_snd_controls[] = { SOC_ENUM("DAC IF2 Data Switch", rt5640_if2_dac_enum), }; +static const struct snd_kcontrol_new rt5640_specific_snd_controls[] = { + /* MONO Output Control */ + SOC_SINGLE("Mono Playback Switch", RT5640_MONO_OUT, RT5640_L_MUTE_SFT, + 1, 1), + + SOC_DOUBLE_TLV("Mono DAC Playback Volume", RT5640_DAC2_DIG_VOL, + RT5640_L_VOL_SFT, RT5640_R_VOL_SFT, 175, 0, dac_vol_tlv), +}; + /** * set_dmic_clk - Set parameter of dmic. * @@ -555,6 +559,20 @@ static const struct snd_kcontrol_new rt5640_sto_dac_r_mix[] = { RT5640_M_ANC_DAC_R_SFT, 1, 1), }; +static const struct snd_kcontrol_new rt5639_sto_dac_l_mix[] = { + SOC_DAPM_SINGLE("DAC L1 Switch", RT5640_STO_DAC_MIXER, + RT5640_M_DAC_L1_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC L2 Switch", RT5640_STO_DAC_MIXER, + RT5640_M_DAC_L2_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5639_sto_dac_r_mix[] = { + SOC_DAPM_SINGLE("DAC R1 Switch", RT5640_STO_DAC_MIXER, + RT5640_M_DAC_R1_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R2 Switch", RT5640_STO_DAC_MIXER, + RT5640_M_DAC_R2_SFT, 1, 1), +}; + static const struct snd_kcontrol_new rt5640_mono_dac_l_mix[] = { SOC_DAPM_SINGLE("DAC L1 Switch", RT5640_MONO_DAC_MIXER, RT5640_M_DAC_L1_MONO_L_SFT, 1, 1), @@ -677,6 +695,30 @@ static const struct snd_kcontrol_new rt5640_out_r_mix[] = { RT5640_M_DAC_R1_OM_R_SFT, 1, 1), }; +static const struct snd_kcontrol_new rt5639_out_l_mix[] = { + SOC_DAPM_SINGLE("BST1 Switch", RT5640_OUT_L3_MIXER, + RT5640_M_BST1_OM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("INL Switch", RT5640_OUT_L3_MIXER, + RT5640_M_IN_L_OM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("REC MIXL Switch", RT5640_OUT_L3_MIXER, + RT5640_M_RM_L_OM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC L1 Switch", RT5640_OUT_L3_MIXER, + RT5640_M_DAC_L1_OM_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5639_out_r_mix[] = { + SOC_DAPM_SINGLE("BST2 Switch", RT5640_OUT_R3_MIXER, + RT5640_M_BST4_OM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("BST1 Switch", RT5640_OUT_R3_MIXER, + RT5640_M_BST1_OM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("INR Switch", RT5640_OUT_R3_MIXER, + RT5640_M_IN_R_OM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("REC MIXR Switch", RT5640_OUT_R3_MIXER, + RT5640_M_RM_R_OM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R1 Switch", RT5640_OUT_R3_MIXER, + RT5640_M_DAC_R1_OM_R_SFT, 1, 1), +}; + static const struct snd_kcontrol_new rt5640_spo_l_mix[] = { SOC_DAPM_SINGLE("DAC R1 Switch", RT5640_SPO_L_MIXER, RT5640_M_DAC_R1_SPM_L_SFT, 1, 1), @@ -708,6 +750,13 @@ static const struct snd_kcontrol_new rt5640_hpo_mix[] = { RT5640_M_HPVOL_HM_SFT, 1, 1), }; +static const struct snd_kcontrol_new rt5639_hpo_mix[] = { + SOC_DAPM_SINGLE("HPO MIX DAC1 Switch", RT5640_HPO_MIXER, + RT5640_M_DAC1_HM_SFT, 1, 1), + SOC_DAPM_SINGLE("HPO MIX HPVOL Switch", RT5640_HPO_MIXER, + RT5640_M_HPVOL_HM_SFT, 1, 1), +}; + static const struct snd_kcontrol_new rt5640_lout_mix[] = { SOC_DAPM_SINGLE("DAC L1 Switch", RT5640_LOUT_MIXER, RT5640_M_DAC_L1_LM_SFT, 1, 1), @@ -1103,26 +1152,15 @@ static const struct snd_soc_dapm_widget rt5640_dapm_widgets[] = { SND_SOC_DAPM_AIF_OUT("AIF1TX", "AIF1 Capture", 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_IN("AIF2RX", "AIF2 Playback", 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_OUT("AIF2TX", "AIF2 Capture", 0, SND_SOC_NOPM, 0, 0), - /* Audio DSP */ - SND_SOC_DAPM_PGA("Audio DSP", SND_SOC_NOPM, 0, 0, NULL, 0), - /* ANC */ - SND_SOC_DAPM_PGA("ANC", SND_SOC_NOPM, 0, 0, NULL, 0), + /* Output Side */ /* DAC mixer before sound effect */ SND_SOC_DAPM_MIXER("DAC MIXL", SND_SOC_NOPM, 0, 0, rt5640_dac_l_mix, ARRAY_SIZE(rt5640_dac_l_mix)), SND_SOC_DAPM_MIXER("DAC MIXR", SND_SOC_NOPM, 0, 0, rt5640_dac_r_mix, ARRAY_SIZE(rt5640_dac_r_mix)), - /* DAC2 channel Mux */ - SND_SOC_DAPM_MUX("DAC L2 Mux", SND_SOC_NOPM, 0, 0, - &rt5640_dac_l2_mux), - SND_SOC_DAPM_MUX("DAC R2 Mux", SND_SOC_NOPM, 0, 0, - &rt5640_dac_r2_mux), + /* DAC Mixer */ - SND_SOC_DAPM_MIXER("Stereo DAC MIXL", SND_SOC_NOPM, 0, 0, - rt5640_sto_dac_l_mix, ARRAY_SIZE(rt5640_sto_dac_l_mix)), - SND_SOC_DAPM_MIXER("Stereo DAC MIXR", SND_SOC_NOPM, 0, 0, - rt5640_sto_dac_r_mix, ARRAY_SIZE(rt5640_sto_dac_r_mix)), SND_SOC_DAPM_MIXER("Mono DAC MIXL", SND_SOC_NOPM, 0, 0, rt5640_mono_dac_l_mix, ARRAY_SIZE(rt5640_mono_dac_l_mix)), SND_SOC_DAPM_MIXER("Mono DAC MIXR", SND_SOC_NOPM, 0, 0, @@ -1134,21 +1172,14 @@ static const struct snd_soc_dapm_widget rt5640_dapm_widgets[] = { /* DACs */ SND_SOC_DAPM_DAC("DAC L1", NULL, RT5640_PWR_DIG1, RT5640_PWR_DAC_L1_BIT, 0), - SND_SOC_DAPM_DAC("DAC L2", NULL, RT5640_PWR_DIG1, - RT5640_PWR_DAC_L2_BIT, 0), SND_SOC_DAPM_DAC("DAC R1", NULL, RT5640_PWR_DIG1, RT5640_PWR_DAC_R1_BIT, 0), - SND_SOC_DAPM_DAC("DAC R2", NULL, RT5640_PWR_DIG1, - RT5640_PWR_DAC_R2_BIT, 0), + /* SPK/OUT Mixer */ SND_SOC_DAPM_MIXER("SPK MIXL", RT5640_PWR_MIXER, RT5640_PWR_SM_L_BIT, 0, rt5640_spk_l_mix, ARRAY_SIZE(rt5640_spk_l_mix)), SND_SOC_DAPM_MIXER("SPK MIXR", RT5640_PWR_MIXER, RT5640_PWR_SM_R_BIT, 0, rt5640_spk_r_mix, ARRAY_SIZE(rt5640_spk_r_mix)), - SND_SOC_DAPM_MIXER("OUT MIXL", RT5640_PWR_MIXER, RT5640_PWR_OM_L_BIT, - 0, rt5640_out_l_mix, ARRAY_SIZE(rt5640_out_l_mix)), - SND_SOC_DAPM_MIXER("OUT MIXR", RT5640_PWR_MIXER, RT5640_PWR_OM_R_BIT, - 0, rt5640_out_r_mix, ARRAY_SIZE(rt5640_out_r_mix)), /* Ouput Volume */ SND_SOC_DAPM_PGA("SPKVOL L", RT5640_PWR_VOL, RT5640_PWR_SV_L_BIT, 0, NULL, 0), @@ -1167,16 +1198,8 @@ static const struct snd_soc_dapm_widget rt5640_dapm_widgets[] = { 0, rt5640_spo_l_mix, ARRAY_SIZE(rt5640_spo_l_mix)), SND_SOC_DAPM_MIXER("SPOR MIX", SND_SOC_NOPM, 0, 0, rt5640_spo_r_mix, ARRAY_SIZE(rt5640_spo_r_mix)), - SND_SOC_DAPM_MIXER("HPO MIX L", SND_SOC_NOPM, 0, 0, - rt5640_hpo_mix, ARRAY_SIZE(rt5640_hpo_mix)), - SND_SOC_DAPM_MIXER("HPO MIX R", SND_SOC_NOPM, 0, 0, - rt5640_hpo_mix, ARRAY_SIZE(rt5640_hpo_mix)), SND_SOC_DAPM_MIXER("LOUT MIX", RT5640_PWR_ANLG1, RT5640_PWR_LM_BIT, 0, rt5640_lout_mix, ARRAY_SIZE(rt5640_lout_mix)), - SND_SOC_DAPM_MIXER("Mono MIX", RT5640_PWR_ANLG1, RT5640_PWR_MM_BIT, 0, - rt5640_mono_mix, ARRAY_SIZE(rt5640_mono_mix)), - SND_SOC_DAPM_SUPPLY("Improve MONO Amp Drv", RT5640_PWR_ANLG1, - RT5640_PWR_MA_BIT, 0, NULL, 0), SND_SOC_DAPM_SUPPLY_S("Improve HP Amp Drv", 1, SND_SOC_NOPM, 0, 0, rt5640_hp_power_event, SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_PGA_S("HP Amp", 1, SND_SOC_NOPM, 0, 0, @@ -1208,10 +1231,69 @@ static const struct snd_soc_dapm_widget rt5640_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("HPOR"), SND_SOC_DAPM_OUTPUT("LOUTL"), SND_SOC_DAPM_OUTPUT("LOUTR"), +}; + +static const struct snd_soc_dapm_widget rt5640_specific_dapm_widgets[] = { + /* Audio DSP */ + SND_SOC_DAPM_PGA("Audio DSP", SND_SOC_NOPM, 0, 0, NULL, 0), + /* ANC */ + SND_SOC_DAPM_PGA("ANC", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* DAC2 channel Mux */ + SND_SOC_DAPM_MUX("DAC L2 Mux", SND_SOC_NOPM, 0, 0, &rt5640_dac_l2_mux), + SND_SOC_DAPM_MUX("DAC R2 Mux", SND_SOC_NOPM, 0, 0, &rt5640_dac_r2_mux), + + SND_SOC_DAPM_MIXER("Stereo DAC MIXL", SND_SOC_NOPM, 0, 0, + rt5640_sto_dac_l_mix, ARRAY_SIZE(rt5640_sto_dac_l_mix)), + SND_SOC_DAPM_MIXER("Stereo DAC MIXR", SND_SOC_NOPM, 0, 0, + rt5640_sto_dac_r_mix, ARRAY_SIZE(rt5640_sto_dac_r_mix)), + + SND_SOC_DAPM_DAC("DAC R2", NULL, RT5640_PWR_DIG1, RT5640_PWR_DAC_R2_BIT, + 0), + SND_SOC_DAPM_DAC("DAC L2", NULL, RT5640_PWR_DIG1, RT5640_PWR_DAC_L2_BIT, + 0), + + SND_SOC_DAPM_MIXER("OUT MIXL", RT5640_PWR_MIXER, RT5640_PWR_OM_L_BIT, + 0, rt5640_out_l_mix, ARRAY_SIZE(rt5640_out_l_mix)), + SND_SOC_DAPM_MIXER("OUT MIXR", RT5640_PWR_MIXER, RT5640_PWR_OM_R_BIT, + 0, rt5640_out_r_mix, ARRAY_SIZE(rt5640_out_r_mix)), + + SND_SOC_DAPM_MIXER("HPO MIX L", SND_SOC_NOPM, 0, 0, + rt5640_hpo_mix, ARRAY_SIZE(rt5640_hpo_mix)), + SND_SOC_DAPM_MIXER("HPO MIX R", SND_SOC_NOPM, 0, 0, + rt5640_hpo_mix, ARRAY_SIZE(rt5640_hpo_mix)), + + SND_SOC_DAPM_MIXER("Mono MIX", RT5640_PWR_ANLG1, RT5640_PWR_MM_BIT, 0, + rt5640_mono_mix, ARRAY_SIZE(rt5640_mono_mix)), + SND_SOC_DAPM_SUPPLY("Improve MONO Amp Drv", RT5640_PWR_ANLG1, + RT5640_PWR_MA_BIT, 0, NULL, 0), + SND_SOC_DAPM_OUTPUT("MONOP"), SND_SOC_DAPM_OUTPUT("MONON"), }; +static const struct snd_soc_dapm_widget rt5639_specific_dapm_widgets[] = { + SND_SOC_DAPM_MIXER("Stereo DAC MIXL", SND_SOC_NOPM, 0, 0, + rt5639_sto_dac_l_mix, ARRAY_SIZE(rt5639_sto_dac_l_mix)), + SND_SOC_DAPM_MIXER("Stereo DAC MIXR", SND_SOC_NOPM, 0, 0, + rt5639_sto_dac_r_mix, ARRAY_SIZE(rt5639_sto_dac_r_mix)), + + SND_SOC_DAPM_SUPPLY("DAC L2 Filter", RT5640_PWR_DIG1, + RT5640_PWR_DAC_L2_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DAC R2 Filter", RT5640_PWR_DIG1, + RT5640_PWR_DAC_R2_BIT, 0, NULL, 0), + + SND_SOC_DAPM_MIXER("OUT MIXL", RT5640_PWR_MIXER, RT5640_PWR_OM_L_BIT, + 0, rt5639_out_l_mix, ARRAY_SIZE(rt5639_out_l_mix)), + SND_SOC_DAPM_MIXER("OUT MIXR", RT5640_PWR_MIXER, RT5640_PWR_OM_R_BIT, + 0, rt5639_out_r_mix, ARRAY_SIZE(rt5639_out_r_mix)), + + SND_SOC_DAPM_MIXER("HPO MIX L", SND_SOC_NOPM, 0, 0, + rt5639_hpo_mix, ARRAY_SIZE(rt5639_hpo_mix)), + SND_SOC_DAPM_MIXER("HPO MIX R", SND_SOC_NOPM, 0, 0, + rt5639_hpo_mix, ARRAY_SIZE(rt5639_hpo_mix)), +}; + static const struct snd_soc_dapm_route rt5640_dapm_routes[] = { {"IN1P", NULL, "LDO2"}, {"IN2P", NULL, "LDO2"}, @@ -1353,71 +1435,38 @@ static const struct snd_soc_dapm_route rt5640_dapm_routes[] = { {"DAC MIXR", "Stereo ADC Switch", "Stereo ADC MIXR"}, {"DAC MIXR", "INF1 Switch", "IF1 DAC R"}, - {"ANC", NULL, "Stereo ADC MIXL"}, - {"ANC", NULL, "Stereo ADC MIXR"}, - - {"Audio DSP", NULL, "DAC MIXL"}, - {"Audio DSP", NULL, "DAC MIXR"}, - - {"DAC L2 Mux", "IF2", "IF2 DAC L"}, - {"DAC L2 Mux", "Base L/R", "Audio DSP"}, - - {"DAC R2 Mux", "IF2", "IF2 DAC R"}, - {"Stereo DAC MIXL", "DAC L1 Switch", "DAC MIXL"}, - {"Stereo DAC MIXL", "DAC L2 Switch", "DAC L2 Mux"}, - {"Stereo DAC MIXL", "ANC Switch", "ANC"}, {"Stereo DAC MIXR", "DAC R1 Switch", "DAC MIXR"}, - {"Stereo DAC MIXR", "DAC R2 Switch", "DAC R2 Mux"}, - {"Stereo DAC MIXR", "ANC Switch", "ANC"}, {"Mono DAC MIXL", "DAC L1 Switch", "DAC MIXL"}, - {"Mono DAC MIXL", "DAC L2 Switch", "DAC L2 Mux"}, - {"Mono DAC MIXL", "DAC R2 Switch", "DAC R2 Mux"}, {"Mono DAC MIXR", "DAC R1 Switch", "DAC MIXR"}, - {"Mono DAC MIXR", "DAC R2 Switch", "DAC R2 Mux"}, - {"Mono DAC MIXR", "DAC L2 Switch", "DAC L2 Mux"}, {"DIG MIXL", "DAC L1 Switch", "DAC MIXL"}, - {"DIG MIXL", "DAC L2 Switch", "DAC L2 Mux"}, {"DIG MIXR", "DAC R1 Switch", "DAC MIXR"}, - {"DIG MIXR", "DAC R2 Switch", "DAC R2 Mux"}, {"DAC L1", NULL, "Stereo DAC MIXL"}, {"DAC L1", NULL, "PLL1", is_sys_clk_from_pll}, {"DAC R1", NULL, "Stereo DAC MIXR"}, {"DAC R1", NULL, "PLL1", is_sys_clk_from_pll}, - {"DAC L2", NULL, "Mono DAC MIXL"}, - {"DAC L2", NULL, "PLL1", is_sys_clk_from_pll}, - {"DAC R2", NULL, "Mono DAC MIXR"}, - {"DAC R2", NULL, "PLL1", is_sys_clk_from_pll}, {"SPK MIXL", "REC MIXL Switch", "RECMIXL"}, {"SPK MIXL", "INL Switch", "INL VOL"}, {"SPK MIXL", "DAC L1 Switch", "DAC L1"}, - {"SPK MIXL", "DAC L2 Switch", "DAC L2"}, {"SPK MIXL", "OUT MIXL Switch", "OUT MIXL"}, {"SPK MIXR", "REC MIXR Switch", "RECMIXR"}, {"SPK MIXR", "INR Switch", "INR VOL"}, {"SPK MIXR", "DAC R1 Switch", "DAC R1"}, - {"SPK MIXR", "DAC R2 Switch", "DAC R2"}, {"SPK MIXR", "OUT MIXR Switch", "OUT MIXR"}, - {"OUT MIXL", "SPK MIXL Switch", "SPK MIXL"}, {"OUT MIXL", "BST1 Switch", "BST1"}, {"OUT MIXL", "INL Switch", "INL VOL"}, {"OUT MIXL", "REC MIXL Switch", "RECMIXL"}, - {"OUT MIXL", "DAC R2 Switch", "DAC R2"}, - {"OUT MIXL", "DAC L2 Switch", "DAC L2"}, {"OUT MIXL", "DAC L1 Switch", "DAC L1"}, - {"OUT MIXR", "SPK MIXR Switch", "SPK MIXR"}, {"OUT MIXR", "BST2 Switch", "BST2"}, {"OUT MIXR", "BST1 Switch", "BST1"}, {"OUT MIXR", "INR Switch", "INR VOL"}, {"OUT MIXR", "REC MIXR Switch", "RECMIXR"}, - {"OUT MIXR", "DAC L2 Switch", "DAC L2"}, - {"OUT MIXR", "DAC R2 Switch", "DAC R2"}, {"OUT MIXR", "DAC R1 Switch", "DAC R1"}, {"SPKVOL L", NULL, "SPK MIXL"}, @@ -1436,11 +1485,9 @@ static const struct snd_soc_dapm_route rt5640_dapm_routes[] = { {"SPOR MIX", "SPKVOL R Switch", "SPKVOL R"}, {"SPOR MIX", "BST1 Switch", "BST1"}, - {"HPO MIX L", "HPO MIX DAC2 Switch", "DAC L2"}, {"HPO MIX L", "HPO MIX DAC1 Switch", "DAC L1"}, {"HPO MIX L", "HPO MIX HPVOL Switch", "HPOVOL L"}, {"HPO MIX L", NULL, "HP L Amp"}, - {"HPO MIX R", "HPO MIX DAC2 Switch", "DAC R2"}, {"HPO MIX R", "HPO MIX DAC1 Switch", "DAC R1"}, {"HPO MIX R", "HPO MIX HPVOL Switch", "HPOVOL R"}, {"HPO MIX R", NULL, "HP R Amp"}, @@ -1450,12 +1497,6 @@ static const struct snd_soc_dapm_route rt5640_dapm_routes[] = { {"LOUT MIX", "OUTVOL L Switch", "OUTVOL L"}, {"LOUT MIX", "OUTVOL R Switch", "OUTVOL R"}, - {"Mono MIX", "DAC R2 Switch", "DAC R2"}, - {"Mono MIX", "DAC L2 Switch", "DAC L2"}, - {"Mono MIX", "OUTVOL R Switch", "OUTVOL R"}, - {"Mono MIX", "OUTVOL L Switch", "OUTVOL L"}, - {"Mono MIX", "BST1 Switch", "BST1"}, - {"HP Amp", NULL, "HPO MIX L"}, {"HP Amp", NULL, "HPO MIX R"}, @@ -1480,11 +1521,82 @@ static const struct snd_soc_dapm_route rt5640_dapm_routes[] = { {"HPOR", NULL, "HP R Playback"}, {"LOUTL", NULL, "LOUT MIX"}, {"LOUTR", NULL, "LOUT MIX"}, +}; + +static const struct snd_soc_dapm_route rt5640_specific_dapm_routes[] = { + {"ANC", NULL, "Stereo ADC MIXL"}, + {"ANC", NULL, "Stereo ADC MIXR"}, + + {"Audio DSP", NULL, "DAC MIXL"}, + {"Audio DSP", NULL, "DAC MIXR"}, + + {"DAC L2 Mux", "IF2", "IF2 DAC L"}, + {"DAC L2 Mux", "Base L/R", "Audio DSP"}, + + {"DAC R2 Mux", "IF2", "IF2 DAC R"}, + + {"Stereo DAC MIXL", "DAC L2 Switch", "DAC L2 Mux"}, + {"Stereo DAC MIXL", "ANC Switch", "ANC"}, + {"Stereo DAC MIXR", "DAC R2 Switch", "DAC R2 Mux"}, + {"Stereo DAC MIXR", "ANC Switch", "ANC"}, + + {"Mono DAC MIXL", "DAC L2 Switch", "DAC L2 Mux"}, + {"Mono DAC MIXL", "DAC R2 Switch", "DAC R2 Mux"}, + + {"Mono DAC MIXR", "DAC R2 Switch", "DAC R2 Mux"}, + {"Mono DAC MIXR", "DAC L2 Switch", "DAC L2 Mux"}, + + {"DIG MIXR", "DAC R2 Switch", "DAC R2 Mux"}, + {"DIG MIXL", "DAC L2 Switch", "DAC L2 Mux"}, + + {"DAC L2", NULL, "Mono DAC MIXL"}, + {"DAC L2", NULL, "PLL1", is_sys_clk_from_pll}, + {"DAC R2", NULL, "Mono DAC MIXR"}, + {"DAC R2", NULL, "PLL1", is_sys_clk_from_pll}, + + {"SPK MIXL", "DAC L2 Switch", "DAC L2"}, + {"SPK MIXR", "DAC R2 Switch", "DAC R2"}, + + {"OUT MIXL", "SPK MIXL Switch", "SPK MIXL"}, + {"OUT MIXR", "SPK MIXR Switch", "SPK MIXR"}, + + {"OUT MIXL", "DAC R2 Switch", "DAC R2"}, + {"OUT MIXL", "DAC L2 Switch", "DAC L2"}, + + {"OUT MIXR", "DAC L2 Switch", "DAC L2"}, + {"OUT MIXR", "DAC R2 Switch", "DAC R2"}, + + {"HPO MIX L", "HPO MIX DAC2 Switch", "DAC L2"}, + {"HPO MIX R", "HPO MIX DAC2 Switch", "DAC R2"}, + + {"Mono MIX", "DAC R2 Switch", "DAC R2"}, + {"Mono MIX", "DAC L2 Switch", "DAC L2"}, + {"Mono MIX", "OUTVOL R Switch", "OUTVOL R"}, + {"Mono MIX", "OUTVOL L Switch", "OUTVOL L"}, + {"Mono MIX", "BST1 Switch", "BST1"}, + {"MONOP", NULL, "Mono MIX"}, {"MONON", NULL, "Mono MIX"}, {"MONOP", NULL, "Improve MONO Amp Drv"}, }; +static const struct snd_soc_dapm_route rt5639_specific_dapm_routes[] = { + {"Stereo DAC MIXL", "DAC L2 Switch", "IF2 DAC L"}, + {"Stereo DAC MIXR", "DAC R2 Switch", "IF2 DAC R"}, + + {"Mono DAC MIXL", "DAC L2 Switch", "IF2 DAC L"}, + {"Mono DAC MIXL", "DAC R2 Switch", "IF2 DAC R"}, + + {"Mono DAC MIXR", "DAC R2 Switch", "IF2 DAC R"}, + {"Mono DAC MIXR", "DAC L2 Switch", "IF2 DAC L"}, + + {"DIG MIXL", "DAC L2 Switch", "IF2 DAC L"}, + {"DIG MIXR", "DAC R2 Switch", "IF2 DAC R"}, + + {"IF2 DAC L", NULL, "DAC L2 Filter"}, + {"IF2 DAC R", NULL, "DAC R2 Filter"}, +}; + static int get_sdp_info(struct snd_soc_codec *codec, int dai_id) { int ret = 0, val; @@ -1901,6 +2013,28 @@ static int rt5640_probe(struct snd_soc_codec *codec) snd_soc_update_bits(codec, RT5640_MICBIAS, 0x0030, 0x0030); snd_soc_update_bits(codec, RT5640_DSP_PATH2, 0xfc00, 0x0c00); + switch (snd_soc_read(codec, RT5640_RESET)) { + case RT5640_RESET_ID: + snd_soc_add_codec_controls(codec, + rt5640_specific_snd_controls, + ARRAY_SIZE(rt5640_specific_snd_controls)); + snd_soc_dapm_new_controls(&codec->dapm, + rt5640_specific_dapm_widgets, + ARRAY_SIZE(rt5640_specific_dapm_widgets)); + snd_soc_dapm_add_routes(&codec->dapm, + rt5640_specific_dapm_routes, + ARRAY_SIZE(rt5640_specific_dapm_routes)); + break; + case RT5639_RESET_ID: + snd_soc_dapm_new_controls(&codec->dapm, + rt5639_specific_dapm_widgets, + ARRAY_SIZE(rt5639_specific_dapm_widgets)); + snd_soc_dapm_add_routes(&codec->dapm, + rt5639_specific_dapm_routes, + ARRAY_SIZE(rt5639_specific_dapm_routes)); + break; + } + return 0; } diff --git a/sound/soc/codecs/rt5640.h b/sound/soc/codecs/rt5640.h index d7bd525..3b50459 100644 --- a/sound/soc/codecs/rt5640.h +++ b/sound/soc/codecs/rt5640.h @@ -14,6 +14,9 @@ #include <sound/rt5640.h> +#define RT5639_RESET_ID 0x0008 +#define RT5640_RESET_ID 0x000c + /* Info */ #define RT5640_RESET 0x00 #define RT5640_VENDOR_ID 0xfd -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] [PATCH 1/2] ALSA: hda/hdmi - Add Nvidia Tegra124 HDMI support
by Dylan Reid 08 Apr '14

08 Apr '14

From: Sumit Bhattacharya <sumitb(a)nvidia.com> Add the Tegra 12x HDMI controller id to patch_hdmi. Signed-off-by: Sumit Bhattacharya <sumitb(a)nvidia.com> Signed-off-by: Dylan Reid <dgreid(a)chromium.org> --- sound/pci/hda/patch_hdmi.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 0cb5b89..b5d1371 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -3322,6 +3322,7 @@ static const struct hda_codec_preset snd_hda_preset_hdmi[] = { { .id = 0x10de001a, .name = "GPU 1a HDMI/DP", .patch = patch_nvhdmi }, { .id = 0x10de001b, .name = "GPU 1b HDMI/DP", .patch = patch_nvhdmi }, { .id = 0x10de001c, .name = "GPU 1c HDMI/DP", .patch = patch_nvhdmi }, +{ .id = 0x10de0028, .name = "Tegra12x HDMI", .patch = patch_nvhdmi }, { .id = 0x10de0040, .name = "GPU 40 HDMI/DP", .patch = patch_nvhdmi }, { .id = 0x10de0041, .name = "GPU 41 HDMI/DP", .patch = patch_nvhdmi }, { .id = 0x10de0042, .name = "GPU 42 HDMI/DP", .patch = patch_nvhdmi }, @@ -3377,6 +3378,7 @@ MODULE_ALIAS("snd-hda-codec-id:10de0019"); MODULE_ALIAS("snd-hda-codec-id:10de001a"); MODULE_ALIAS("snd-hda-codec-id:10de001b"); MODULE_ALIAS("snd-hda-codec-id:10de001c"); +MODULE_ALIAS("snd-hda-codec-id:10de0028"); MODULE_ALIAS("snd-hda-codec-id:10de0040"); MODULE_ALIAS("snd-hda-codec-id:10de0041"); MODULE_ALIAS("snd-hda-codec-id:10de0042"); -- 1.8.1.3.605.g02339dd

1 0

[alsa-devel] [PATCH Resend 1/1] ASoC: Samsung: Fix build on multiplatform
by Sachin Kamat 08 Apr '14

08 Apr '14

PCM and S/PDIF drivers referenced mach headers for a trivial data structure. This caused build errors on multiplatform builds as machine headers are not accessible from driver files. Move the data structure definition to the driver header and remove the dependency. While at it rename the structure to avoid multiple definition errors as the same structure is also used by the platform code. Signed-off-by: Sachin Kamat <sachin.kamat(a)linaro.org> --- sound/soc/samsung/ac97.c | 6 +++--- sound/soc/samsung/dma.h | 6 +++++- sound/soc/samsung/i2s.c | 6 +++--- sound/soc/samsung/pcm.c | 5 ++--- sound/soc/samsung/s3c2412-i2s.c | 4 ++-- sound/soc/samsung/s3c24xx-i2s.c | 4 ++-- sound/soc/samsung/spdif.c | 3 +-- 7 files changed, 18 insertions(+), 16 deletions(-) diff --git a/sound/soc/samsung/ac97.c b/sound/soc/samsung/ac97.c index 4a88e36c82ec..76b072bd4ba2 100644 --- a/sound/soc/samsung/ac97.c +++ b/sound/soc/samsung/ac97.c @@ -39,15 +39,15 @@ struct s3c_ac97_info { }; static struct s3c_ac97_info s3c_ac97; -static struct s3c2410_dma_client s3c_dma_client_out = { +static struct s3c_dma_client s3c_dma_client_out = { .name = "AC97 PCMOut" }; -static struct s3c2410_dma_client s3c_dma_client_in = { +static struct s3c_dma_client s3c_dma_client_in = { .name = "AC97 PCMIn" }; -static struct s3c2410_dma_client s3c_dma_client_micin = { +static struct s3c_dma_client s3c_dma_client_micin = { .name = "AC97 MicIn" }; diff --git a/sound/soc/samsung/dma.h b/sound/soc/samsung/dma.h index 225e5378014e..ad7c0f04f00d 100644 --- a/sound/soc/samsung/dma.h +++ b/sound/soc/samsung/dma.h @@ -14,8 +14,12 @@ #include <sound/dmaengine_pcm.h> +struct s3c_dma_client { + char *name; +}; + struct s3c_dma_params { - struct s3c2410_dma_client *client; /* stream identifier */ + struct s3c_dma_client *client; /* stream identifier */ int channel; /* Channel ID */ dma_addr_t dma_addr; int dma_size; /* Size of the DMA transfer */ diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 0a9b44c940ce..048ead967199 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -1211,10 +1211,10 @@ static int samsung_i2s_probe(struct platform_device *pdev) pri_dai->dma_playback.dma_addr = regs_base + I2STXD; pri_dai->dma_capture.dma_addr = regs_base + I2SRXD; pri_dai->dma_playback.client = - (struct s3c2410_dma_client *)&pri_dai->dma_playback; + (struct s3c_dma_client *)&pri_dai->dma_playback; pri_dai->dma_playback.ch_name = "tx"; pri_dai->dma_capture.client = - (struct s3c2410_dma_client *)&pri_dai->dma_capture; + (struct s3c_dma_client *)&pri_dai->dma_capture; pri_dai->dma_capture.ch_name = "rx"; pri_dai->dma_playback.dma_size = 4; pri_dai->dma_capture.dma_size = 4; @@ -1233,7 +1233,7 @@ static int samsung_i2s_probe(struct platform_device *pdev) } sec_dai->dma_playback.dma_addr = regs_base + I2STXDS; sec_dai->dma_playback.client = - (struct s3c2410_dma_client *)&sec_dai->dma_playback; + (struct s3c_dma_client *)&sec_dai->dma_playback; sec_dai->dma_playback.ch_name = "tx-sec"; if (!np) { diff --git a/sound/soc/samsung/pcm.c b/sound/soc/samsung/pcm.c index 6a5e4bf6ac96..ab54e297957c 100644 --- a/sound/soc/samsung/pcm.c +++ b/sound/soc/samsung/pcm.c @@ -20,7 +20,6 @@ #include <sound/pcm_params.h> #include <linux/platform_data/asoc-s3c.h> -#include <mach/dma.h> #include "dma.h" #include "pcm.h" @@ -132,11 +131,11 @@ struct s3c_pcm_info { struct s3c_dma_params *dma_capture; }; -static struct s3c2410_dma_client s3c_pcm_dma_client_out = { +static struct s3c_dma_client s3c_pcm_dma_client_out = { .name = "PCM Stereo out" }; -static struct s3c2410_dma_client s3c_pcm_dma_client_in = { +static struct s3c_dma_client s3c_pcm_dma_client_in = { .name = "PCM Stereo in" }; diff --git a/sound/soc/samsung/s3c2412-i2s.c b/sound/soc/samsung/s3c2412-i2s.c index d0794458963a..e9bb5d7a71ee 100644 --- a/sound/soc/samsung/s3c2412-i2s.c +++ b/sound/soc/samsung/s3c2412-i2s.c @@ -33,11 +33,11 @@ #include "regs-i2s-v2.h" #include "s3c2412-i2s.h" -static struct s3c2410_dma_client s3c2412_dma_client_out = { +static struct s3c_dma_client s3c2412_dma_client_out = { .name = "I2S PCM Stereo out" }; -static struct s3c2410_dma_client s3c2412_dma_client_in = { +static struct s3c_dma_client s3c2412_dma_client_in = { .name = "I2S PCM Stereo in" }; diff --git a/sound/soc/samsung/s3c24xx-i2s.c b/sound/soc/samsung/s3c24xx-i2s.c index f31e916dd8c4..d7b8457b5650 100644 --- a/sound/soc/samsung/s3c24xx-i2s.c +++ b/sound/soc/samsung/s3c24xx-i2s.c @@ -31,11 +31,11 @@ #include "dma.h" #include "s3c24xx-i2s.h" -static struct s3c2410_dma_client s3c24xx_dma_client_out = { +static struct s3c_dma_client s3c24xx_dma_client_out = { .name = "I2S PCM Stereo out" }; -static struct s3c2410_dma_client s3c24xx_dma_client_in = { +static struct s3c_dma_client s3c24xx_dma_client_in = { .name = "I2S PCM Stereo in" }; diff --git a/sound/soc/samsung/spdif.c b/sound/soc/samsung/spdif.c index 28487dcc4538..cfe63b7bcc9f 100644 --- a/sound/soc/samsung/spdif.c +++ b/sound/soc/samsung/spdif.c @@ -18,7 +18,6 @@ #include <sound/pcm_params.h> #include <linux/platform_data/asoc-s3c.h> -#include <mach/dma.h> #include "dma.h" #include "spdif.h" @@ -94,7 +93,7 @@ struct samsung_spdif_info { struct s3c_dma_params *dma_playback; }; -static struct s3c2410_dma_client spdif_dma_client_out = { +static struct s3c_dma_client spdif_dma_client_out = { .name = "S/PDIF Stereo out", }; -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ALSA: ice1712: Fix boundary checks in PCM pointer ops
by Takashi Iwai 08 Apr '14

08 Apr '14

PCM pointer callbacks in ice1712 driver check the buffer size boundary wrongly between bytes and frames. This leads to PCM core warnings like: snd_pcm_update_hw_ptr0: 105 callbacks suppressed ALSA pcm_lib.c:352 BUG: pcmC3D0c:0, pos = 5461, buffer size = 5461, period size = 2730 This patch fixes these checks to be placed after the proper unit conversions. Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/ice1712/ice1712.c | 15 ++++++++++----- 1 file changed, 10 insertions(+), 5 deletions(-) diff --git a/sound/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c index da005493f060..d9b9e4595f17 100644 --- a/sound/pci/ice1712/ice1712.c +++ b/sound/pci/ice1712/ice1712.c @@ -685,9 +685,10 @@ static snd_pcm_uframes_t snd_ice1712_playback_pointer(struct snd_pcm_substream * if (!(snd_ice1712_read(ice, ICE1712_IREG_PBK_CTRL) & 1)) return 0; ptr = runtime->buffer_size - inw(ice->ddma_port + 4); + ptr = bytes_to_frames(substream->runtime, ptr); if (ptr == runtime->buffer_size) ptr = 0; - return bytes_to_frames(substream->runtime, ptr); + return ptr; } static snd_pcm_uframes_t snd_ice1712_playback_ds_pointer(struct snd_pcm_substream *substream) @@ -704,9 +705,10 @@ static snd_pcm_uframes_t snd_ice1712_playback_ds_pointer(struct snd_pcm_substrea addr = ICE1712_DSC_ADDR0; ptr = snd_ice1712_ds_read(ice, substream->number * 2, addr) - ice->playback_con_virt_addr[substream->number]; + ptr = bytes_to_frames(substream->runtime, ptr); if (ptr == substream->runtime->buffer_size) ptr = 0; - return bytes_to_frames(substream->runtime, ptr); + return ptr; } static snd_pcm_uframes_t snd_ice1712_capture_pointer(struct snd_pcm_substream *substream) @@ -717,9 +719,10 @@ static snd_pcm_uframes_t snd_ice1712_capture_pointer(struct snd_pcm_substream *s if (!(snd_ice1712_read(ice, ICE1712_IREG_CAP_CTRL) & 1)) return 0; ptr = inl(ICEREG(ice, CONCAP_ADDR)) - ice->capture_con_virt_addr; + ptr = bytes_to_frames(substream->runtime, ptr); if (ptr == substream->runtime->buffer_size) ptr = 0; - return bytes_to_frames(substream->runtime, ptr); + return ptr; } static const struct snd_pcm_hardware snd_ice1712_playback = { @@ -1116,9 +1119,10 @@ static snd_pcm_uframes_t snd_ice1712_playback_pro_pointer(struct snd_pcm_substre if (!(inl(ICEMT(ice, PLAYBACK_CONTROL)) & ICE1712_PLAYBACK_START)) return 0; ptr = ice->playback_pro_size - (inw(ICEMT(ice, PLAYBACK_SIZE)) << 2); + ptr = bytes_to_frames(substream->runtime, ptr); if (ptr == substream->runtime->buffer_size) ptr = 0; - return bytes_to_frames(substream->runtime, ptr); + return ptr; } static snd_pcm_uframes_t snd_ice1712_capture_pro_pointer(struct snd_pcm_substream *substream) @@ -1129,9 +1133,10 @@ static snd_pcm_uframes_t snd_ice1712_capture_pro_pointer(struct snd_pcm_substrea if (!(inl(ICEMT(ice, PLAYBACK_CONTROL)) & ICE1712_CAPTURE_START_SHADOW)) return 0; ptr = ice->capture_pro_size - (inw(ICEMT(ice, CAPTURE_SIZE)) << 2); + ptr = bytes_to_frames(substream->runtime, ptr); if (ptr == substream->runtime->buffer_size) ptr = 0; - return bytes_to_frames(substream->runtime, ptr); + return ptr; } static const struct snd_pcm_hardware snd_ice1712_playback_pro = { -- 1.9.1

1 0

[alsa-devel] (JOB) ALSA, ASoC driver development
by Adam Ćwiek 08 Apr '14

08 Apr '14

Hi, I am developing multi channel audio dsp platform for high level audio processing. Now I am searching for someone, who can deal with audio ALSA, ASoC driver development (of course paid offer). I am electronic engeneer, so I will take care of hardware. I am going to use audio codecs, like wolfsons WM8510, CS42448 from Cirrus Logic or PCM3168 from Texas. My goal is to be able to send / receive as many analog inputs/outputs as possible. As mainboard i want to use ready made modules as OLinuXino , Raspberry Pi, or BeagleBone. Looking forward to hearing from you. Adam Cwiek

1 0

[alsa-devel] [PATCH] ASoC: fsl_sai: Fix Bit Clock Polarity configurations
by Nicolin Chen 08 Apr '14

08 Apr '14

The BCP bit in TCR4/RCR4 register rules as followings: 0 Bit clock is active high with drive outputs on rising edge and sample inputs on falling edge. 1 Bit clock is active low with drive outputs on falling edge and sample inputs on rising edge. For all formats currently supported in the fsl_sai driver, they're exactly sending data on the falling edge and sampling on the rising edge. However, the driver clears this BCP bit for all of them which results click noise when working with SGTL5000 and big noise with WM8962. Thus this patch corrects the BCP settings for all the formats here to fix the nosie issue. Signed-off-by: Nicolin Chen <Guangyu.Chen(a)freescale.com> --- sound/soc/fsl/fsl_sai.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 99051c7..9bbebea 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -180,7 +180,7 @@ static int fsl_sai_set_dai_fmt_tr(struct snd_soc_dai *cpu_dai, * that is, together with the last bit of the previous * data word. */ - val_cr2 &= ~FSL_SAI_CR2_BCP; + val_cr2 |= FSL_SAI_CR2_BCP; val_cr4 |= FSL_SAI_CR4_FSE | FSL_SAI_CR4_FSP; break; case SND_SOC_DAIFMT_LEFT_J: @@ -188,7 +188,7 @@ static int fsl_sai_set_dai_fmt_tr(struct snd_soc_dai *cpu_dai, * Frame high, one word length for frame sync, * frame sync asserts with the first bit of the frame. */ - val_cr2 &= ~FSL_SAI_CR2_BCP; + val_cr2 |= FSL_SAI_CR2_BCP; val_cr4 &= ~(FSL_SAI_CR4_FSE | FSL_SAI_CR4_FSP); break; case SND_SOC_DAIFMT_DSP_A: @@ -198,7 +198,7 @@ static int fsl_sai_set_dai_fmt_tr(struct snd_soc_dai *cpu_dai, * that is, together with the last bit of the previous * data word. */ - val_cr2 &= ~FSL_SAI_CR2_BCP; + val_cr2 |= FSL_SAI_CR2_BCP; val_cr4 &= ~FSL_SAI_CR4_FSP; val_cr4 |= FSL_SAI_CR4_FSE; sai->is_dsp_mode = true; @@ -208,7 +208,7 @@ static int fsl_sai_set_dai_fmt_tr(struct snd_soc_dai *cpu_dai, * Frame high, one bit for frame sync, * frame sync asserts with the first bit of the frame. */ - val_cr2 &= ~FSL_SAI_CR2_BCP; + val_cr2 |= FSL_SAI_CR2_BCP; val_cr4 &= ~(FSL_SAI_CR4_FSE | FSL_SAI_CR4_FSP); sai->is_dsp_mode = true; break; -- 1.8.4

3 6

[alsa-devel] [PATCH] ASoC: rt5640: add rt5639 support
by Oder Chiou 08 Apr '14

08 Apr '14

This patch adds the rt5639 support Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5640.c | 365 +++++++++++++++++++++++++++++++--------------- sound/soc/codecs/rt5640.h | 3 + 2 files changed, 250 insertions(+), 118 deletions(-) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 6b8778f..f5b9ecc 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -361,24 +361,23 @@ static unsigned int bst_tlv[] = { static const char * const rt5640_data_select[] = { "Normal", "left copy to right", "right copy to left", "Swap"}; -static const SOC_ENUM_SINGLE_DECL(rt5640_if1_dac_enum, RT5640_DIG_INF_DATA, - RT5640_IF1_DAC_SEL_SFT, rt5640_data_select); +static SOC_ENUM_SINGLE_DECL(rt5640_if1_dac_enum, RT5640_DIG_INF_DATA, + RT5640_IF1_DAC_SEL_SFT, rt5640_data_select); -static const SOC_ENUM_SINGLE_DECL(rt5640_if1_adc_enum, RT5640_DIG_INF_DATA, - RT5640_IF1_ADC_SEL_SFT, rt5640_data_select); +static SOC_ENUM_SINGLE_DECL(rt5640_if1_adc_enum, RT5640_DIG_INF_DATA, + RT5640_IF1_ADC_SEL_SFT, rt5640_data_select); -static const SOC_ENUM_SINGLE_DECL(rt5640_if2_dac_enum, RT5640_DIG_INF_DATA, - RT5640_IF2_DAC_SEL_SFT, rt5640_data_select); +static SOC_ENUM_SINGLE_DECL(rt5640_if2_dac_enum, RT5640_DIG_INF_DATA, + RT5640_IF2_DAC_SEL_SFT, rt5640_data_select); -static const SOC_ENUM_SINGLE_DECL(rt5640_if2_adc_enum, RT5640_DIG_INF_DATA, - RT5640_IF2_ADC_SEL_SFT, rt5640_data_select); +static SOC_ENUM_SINGLE_DECL(rt5640_if2_adc_enum, RT5640_DIG_INF_DATA, + RT5640_IF2_ADC_SEL_SFT, rt5640_data_select); /* Class D speaker gain ratio */ static const char * const rt5640_clsd_spk_ratio[] = {"1.66x", "1.83x", "1.94x", "2x", "2.11x", "2.22x", "2.33x", "2.44x", "2.55x", "2.66x", "2.77x"}; -static const SOC_ENUM_SINGLE_DECL( - rt5640_clsd_spk_ratio_enum, RT5640_CLS_D_OUT, +static SOC_ENUM_SINGLE_DECL(rt5640_clsd_spk_ratio_enum, RT5640_CLS_D_OUT, RT5640_CLSD_RATIO_SFT, rt5640_clsd_spk_ratio); static const struct snd_kcontrol_new rt5640_snd_controls[] = { @@ -387,11 +386,13 @@ static const struct snd_kcontrol_new rt5640_snd_controls[] = { RT5640_VOL_L_SFT, RT5640_VOL_R_SFT, 1, 1), SOC_DOUBLE_TLV("Speaker Playback Volume", RT5640_SPK_VOL, RT5640_L_VOL_SFT, RT5640_R_VOL_SFT, 39, 1, out_vol_tlv), + /* Headphone Output Volume */ SOC_DOUBLE("HP Channel Switch", RT5640_HP_VOL, RT5640_VOL_L_SFT, RT5640_VOL_R_SFT, 1, 1), SOC_DOUBLE_TLV("HP Playback Volume", RT5640_HP_VOL, RT5640_L_VOL_SFT, RT5640_R_VOL_SFT, 39, 1, out_vol_tlv), + /* OUTPUT Control */ SOC_DOUBLE("OUT Playback Switch", RT5640_OUTPUT, RT5640_L_MUTE_SFT, RT5640_R_MUTE_SFT, 1, 1), @@ -399,49 +400,54 @@ static const struct snd_kcontrol_new rt5640_snd_controls[] = { RT5640_VOL_L_SFT, RT5640_VOL_R_SFT, 1, 1), SOC_DOUBLE_TLV("OUT Playback Volume", RT5640_OUTPUT, RT5640_L_VOL_SFT, RT5640_R_VOL_SFT, 39, 1, out_vol_tlv), - /* MONO Output Control */ - SOC_SINGLE("Mono Playback Switch", RT5640_MONO_OUT, - RT5640_L_MUTE_SFT, 1, 1), + /* DAC Digital Volume */ SOC_DOUBLE("DAC2 Playback Switch", RT5640_DAC2_CTRL, RT5640_M_DAC_L2_VOL_SFT, RT5640_M_DAC_R2_VOL_SFT, 1, 1), SOC_DOUBLE_TLV("DAC1 Playback Volume", RT5640_DAC1_DIG_VOL, - RT5640_L_VOL_SFT, RT5640_R_VOL_SFT, - 175, 0, dac_vol_tlv), - SOC_DOUBLE_TLV("Mono DAC Playback Volume", RT5640_DAC2_DIG_VOL, - RT5640_L_VOL_SFT, RT5640_R_VOL_SFT, - 175, 0, dac_vol_tlv), + RT5640_L_VOL_SFT, RT5640_R_VOL_SFT, 175, 0, dac_vol_tlv), + /* IN1/IN2 Control */ - SOC_SINGLE_TLV("IN1 Boost", RT5640_IN1_IN2, - RT5640_BST_SFT1, 8, 0, bst_tlv), - SOC_SINGLE_TLV("IN2 Boost", RT5640_IN3_IN4, - RT5640_BST_SFT2, 8, 0, bst_tlv), + SOC_SINGLE_TLV("IN1 Boost", RT5640_IN1_IN2, RT5640_BST_SFT1, 8, 0, + bst_tlv), + SOC_SINGLE_TLV("IN2 Boost", RT5640_IN3_IN4, RT5640_BST_SFT2, 8, 0, + bst_tlv), + /* INL/INR Volume Control */ SOC_DOUBLE_TLV("IN Capture Volume", RT5640_INL_INR_VOL, - RT5640_INL_VOL_SFT, RT5640_INR_VOL_SFT, - 31, 1, in_vol_tlv), + RT5640_INL_VOL_SFT, RT5640_INR_VOL_SFT, 31, 1, in_vol_tlv), + /* ADC Digital Volume Control */ SOC_DOUBLE("ADC Capture Switch", RT5640_ADC_DIG_VOL, RT5640_L_MUTE_SFT, RT5640_R_MUTE_SFT, 1, 1), SOC_DOUBLE_TLV("ADC Capture Volume", RT5640_ADC_DIG_VOL, - RT5640_L_VOL_SFT, RT5640_R_VOL_SFT, - 127, 0, adc_vol_tlv), + RT5640_L_VOL_SFT, RT5640_R_VOL_SFT, 127, 0, adc_vol_tlv), SOC_DOUBLE_TLV("Mono ADC Capture Volume", RT5640_ADC_DATA, - RT5640_L_VOL_SFT, RT5640_R_VOL_SFT, - 127, 0, adc_vol_tlv), + RT5640_L_VOL_SFT, RT5640_R_VOL_SFT, 127, 0, adc_vol_tlv), + /* ADC Boost Volume Control */ SOC_DOUBLE_TLV("ADC Boost Gain", RT5640_ADC_BST_VOL, - RT5640_ADC_L_BST_SFT, RT5640_ADC_R_BST_SFT, - 3, 0, adc_bst_tlv), + RT5640_ADC_L_BST_SFT, RT5640_ADC_R_BST_SFT, 3, 0, adc_bst_tlv), + /* Class D speaker gain ratio */ SOC_ENUM("Class D SPK Ratio Control", rt5640_clsd_spk_ratio_enum), + /* Data Swap */ SOC_ENUM("ADC IF1 Data Switch", rt5640_if1_adc_enum), SOC_ENUM("DAC IF1 Data Switch", rt5640_if1_dac_enum), SOC_ENUM("ADC IF2 Data Switch", rt5640_if2_adc_enum), SOC_ENUM("DAC IF2 Data Switch", rt5640_if2_dac_enum), }; +static const struct snd_kcontrol_new rt5640_specific_snd_controls[] = { + /* MONO Output Control */ + SOC_SINGLE("Mono Playback Switch", RT5640_MONO_OUT, RT5640_L_MUTE_SFT, + 1, 1), + + SOC_DOUBLE_TLV("Mono DAC Playback Volume", RT5640_DAC2_DIG_VOL, + RT5640_L_VOL_SFT, RT5640_R_VOL_SFT, 175, 0, dac_vol_tlv), +}; + /** * set_dmic_clk - Set parameter of dmic. * @@ -555,6 +561,20 @@ static const struct snd_kcontrol_new rt5640_sto_dac_r_mix[] = { RT5640_M_ANC_DAC_R_SFT, 1, 1), }; +static const struct snd_kcontrol_new rt5639_sto_dac_l_mix[] = { + SOC_DAPM_SINGLE("DAC L1 Switch", RT5640_STO_DAC_MIXER, + RT5640_M_DAC_L1_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC L2 Switch", RT5640_STO_DAC_MIXER, + RT5640_M_DAC_L2_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5639_sto_dac_r_mix[] = { + SOC_DAPM_SINGLE("DAC R1 Switch", RT5640_STO_DAC_MIXER, + RT5640_M_DAC_R1_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R2 Switch", RT5640_STO_DAC_MIXER, + RT5640_M_DAC_R2_SFT, 1, 1), +}; + static const struct snd_kcontrol_new rt5640_mono_dac_l_mix[] = { SOC_DAPM_SINGLE("DAC L1 Switch", RT5640_MONO_DAC_MIXER, RT5640_M_DAC_L1_MONO_L_SFT, 1, 1), @@ -677,6 +697,30 @@ static const struct snd_kcontrol_new rt5640_out_r_mix[] = { RT5640_M_DAC_R1_OM_R_SFT, 1, 1), }; +static const struct snd_kcontrol_new rt5639_out_l_mix[] = { + SOC_DAPM_SINGLE("BST1 Switch", RT5640_OUT_L3_MIXER, + RT5640_M_BST1_OM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("INL Switch", RT5640_OUT_L3_MIXER, + RT5640_M_IN_L_OM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("REC MIXL Switch", RT5640_OUT_L3_MIXER, + RT5640_M_RM_L_OM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC L1 Switch", RT5640_OUT_L3_MIXER, + RT5640_M_DAC_L1_OM_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5639_out_r_mix[] = { + SOC_DAPM_SINGLE("BST2 Switch", RT5640_OUT_R3_MIXER, + RT5640_M_BST4_OM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("BST1 Switch", RT5640_OUT_R3_MIXER, + RT5640_M_BST1_OM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("INR Switch", RT5640_OUT_R3_MIXER, + RT5640_M_IN_R_OM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("REC MIXR Switch", RT5640_OUT_R3_MIXER, + RT5640_M_RM_R_OM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R1 Switch", RT5640_OUT_R3_MIXER, + RT5640_M_DAC_R1_OM_R_SFT, 1, 1), +}; + static const struct snd_kcontrol_new rt5640_spo_l_mix[] = { SOC_DAPM_SINGLE("DAC R1 Switch", RT5640_SPO_L_MIXER, RT5640_M_DAC_R1_SPM_L_SFT, 1, 1), @@ -708,6 +752,13 @@ static const struct snd_kcontrol_new rt5640_hpo_mix[] = { RT5640_M_HPVOL_HM_SFT, 1, 1), }; +static const struct snd_kcontrol_new rt5639_hpo_mix[] = { + SOC_DAPM_SINGLE("HPO MIX DAC1 Switch", RT5640_HPO_MIXER, + RT5640_M_DAC1_HM_SFT, 1, 1), + SOC_DAPM_SINGLE("HPO MIX HPVOL Switch", RT5640_HPO_MIXER, + RT5640_M_HPVOL_HM_SFT, 1, 1), +}; + static const struct snd_kcontrol_new rt5640_lout_mix[] = { SOC_DAPM_SINGLE("DAC L1 Switch", RT5640_LOUT_MIXER, RT5640_M_DAC_L1_LM_SFT, 1, 1), @@ -753,8 +804,7 @@ static const char * const rt5640_stereo_adc1_src[] = { "DIG MIX", "ADC" }; -static const SOC_ENUM_SINGLE_DECL( - rt5640_stereo_adc1_enum, RT5640_STO_ADC_MIXER, +static SOC_ENUM_SINGLE_DECL(rt5640_stereo_adc1_enum, RT5640_STO_ADC_MIXER, RT5640_ADC_1_SRC_SFT, rt5640_stereo_adc1_src); static const struct snd_kcontrol_new rt5640_sto_adc_1_mux = @@ -764,8 +814,7 @@ static const char * const rt5640_stereo_adc2_src[] = { "DMIC1", "DMIC2", "DIG MIX" }; -static const SOC_ENUM_SINGLE_DECL( - rt5640_stereo_adc2_enum, RT5640_STO_ADC_MIXER, +static SOC_ENUM_SINGLE_DECL(rt5640_stereo_adc2_enum, RT5640_STO_ADC_MIXER, RT5640_ADC_2_SRC_SFT, rt5640_stereo_adc2_src); static const struct snd_kcontrol_new rt5640_sto_adc_2_mux = @@ -776,8 +825,7 @@ static const char * const rt5640_mono_adc_l1_src[] = { "Mono DAC MIXL", "ADCL" }; -static const SOC_ENUM_SINGLE_DECL( - rt5640_mono_adc_l1_enum, RT5640_MONO_ADC_MIXER, +static SOC_ENUM_SINGLE_DECL(rt5640_mono_adc_l1_enum, RT5640_MONO_ADC_MIXER, RT5640_MONO_ADC_L1_SRC_SFT, rt5640_mono_adc_l1_src); static const struct snd_kcontrol_new rt5640_mono_adc_l1_mux = @@ -787,8 +835,7 @@ static const char * const rt5640_mono_adc_l2_src[] = { "DMIC L1", "DMIC L2", "Mono DAC MIXL" }; -static const SOC_ENUM_SINGLE_DECL( - rt5640_mono_adc_l2_enum, RT5640_MONO_ADC_MIXER, +static SOC_ENUM_SINGLE_DECL(rt5640_mono_adc_l2_enum, RT5640_MONO_ADC_MIXER, RT5640_MONO_ADC_L2_SRC_SFT, rt5640_mono_adc_l2_src); static const struct snd_kcontrol_new rt5640_mono_adc_l2_mux = @@ -798,8 +845,7 @@ static const char * const rt5640_mono_adc_r1_src[] = { "Mono DAC MIXR", "ADCR" }; -static const SOC_ENUM_SINGLE_DECL( - rt5640_mono_adc_r1_enum, RT5640_MONO_ADC_MIXER, +static SOC_ENUM_SINGLE_DECL(rt5640_mono_adc_r1_enum, RT5640_MONO_ADC_MIXER, RT5640_MONO_ADC_R1_SRC_SFT, rt5640_mono_adc_r1_src); static const struct snd_kcontrol_new rt5640_mono_adc_r1_mux = @@ -809,8 +855,7 @@ static const char * const rt5640_mono_adc_r2_src[] = { "DMIC R1", "DMIC R2", "Mono DAC MIXR" }; -static const SOC_ENUM_SINGLE_DECL( - rt5640_mono_adc_r2_enum, RT5640_MONO_ADC_MIXER, +static SOC_ENUM_SINGLE_DECL(rt5640_mono_adc_r2_enum, RT5640_MONO_ADC_MIXER, RT5640_MONO_ADC_R2_SRC_SFT, rt5640_mono_adc_r2_src); static const struct snd_kcontrol_new rt5640_mono_adc_r2_mux = @@ -1103,26 +1148,15 @@ static const struct snd_soc_dapm_widget rt5640_dapm_widgets[] = { SND_SOC_DAPM_AIF_OUT("AIF1TX", "AIF1 Capture", 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_IN("AIF2RX", "AIF2 Playback", 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_OUT("AIF2TX", "AIF2 Capture", 0, SND_SOC_NOPM, 0, 0), - /* Audio DSP */ - SND_SOC_DAPM_PGA("Audio DSP", SND_SOC_NOPM, 0, 0, NULL, 0), - /* ANC */ - SND_SOC_DAPM_PGA("ANC", SND_SOC_NOPM, 0, 0, NULL, 0), + /* Output Side */ /* DAC mixer before sound effect */ SND_SOC_DAPM_MIXER("DAC MIXL", SND_SOC_NOPM, 0, 0, rt5640_dac_l_mix, ARRAY_SIZE(rt5640_dac_l_mix)), SND_SOC_DAPM_MIXER("DAC MIXR", SND_SOC_NOPM, 0, 0, rt5640_dac_r_mix, ARRAY_SIZE(rt5640_dac_r_mix)), - /* DAC2 channel Mux */ - SND_SOC_DAPM_MUX("DAC L2 Mux", SND_SOC_NOPM, 0, 0, - &rt5640_dac_l2_mux), - SND_SOC_DAPM_MUX("DAC R2 Mux", SND_SOC_NOPM, 0, 0, - &rt5640_dac_r2_mux), + /* DAC Mixer */ - SND_SOC_DAPM_MIXER("Stereo DAC MIXL", SND_SOC_NOPM, 0, 0, - rt5640_sto_dac_l_mix, ARRAY_SIZE(rt5640_sto_dac_l_mix)), - SND_SOC_DAPM_MIXER("Stereo DAC MIXR", SND_SOC_NOPM, 0, 0, - rt5640_sto_dac_r_mix, ARRAY_SIZE(rt5640_sto_dac_r_mix)), SND_SOC_DAPM_MIXER("Mono DAC MIXL", SND_SOC_NOPM, 0, 0, rt5640_mono_dac_l_mix, ARRAY_SIZE(rt5640_mono_dac_l_mix)), SND_SOC_DAPM_MIXER("Mono DAC MIXR", SND_SOC_NOPM, 0, 0, @@ -1132,23 +1166,15 @@ static const struct snd_soc_dapm_widget rt5640_dapm_widgets[] = { SND_SOC_DAPM_MIXER("DIG MIXR", SND_SOC_NOPM, 0, 0, rt5640_dig_r_mix, ARRAY_SIZE(rt5640_dig_r_mix)), /* DACs */ - SND_SOC_DAPM_DAC("DAC L1", NULL, RT5640_PWR_DIG1, - RT5640_PWR_DAC_L1_BIT, 0), - SND_SOC_DAPM_DAC("DAC L2", NULL, RT5640_PWR_DIG1, - RT5640_PWR_DAC_L2_BIT, 0), - SND_SOC_DAPM_DAC("DAC R1", NULL, RT5640_PWR_DIG1, - RT5640_PWR_DAC_R1_BIT, 0), - SND_SOC_DAPM_DAC("DAC R2", NULL, RT5640_PWR_DIG1, - RT5640_PWR_DAC_R2_BIT, 0), + SND_SOC_DAPM_DAC("DAC L1", NULL, RT5640_PWR_DIG1, RT5640_PWR_DAC_L1_BIT, + 0), + SND_SOC_DAPM_DAC("DAC R1", NULL, RT5640_PWR_DIG1, RT5640_PWR_DAC_R1_BIT, + 0), /* SPK/OUT Mixer */ SND_SOC_DAPM_MIXER("SPK MIXL", RT5640_PWR_MIXER, RT5640_PWR_SM_L_BIT, 0, rt5640_spk_l_mix, ARRAY_SIZE(rt5640_spk_l_mix)), SND_SOC_DAPM_MIXER("SPK MIXR", RT5640_PWR_MIXER, RT5640_PWR_SM_R_BIT, 0, rt5640_spk_r_mix, ARRAY_SIZE(rt5640_spk_r_mix)), - SND_SOC_DAPM_MIXER("OUT MIXL", RT5640_PWR_MIXER, RT5640_PWR_OM_L_BIT, - 0, rt5640_out_l_mix, ARRAY_SIZE(rt5640_out_l_mix)), - SND_SOC_DAPM_MIXER("OUT MIXR", RT5640_PWR_MIXER, RT5640_PWR_OM_R_BIT, - 0, rt5640_out_r_mix, ARRAY_SIZE(rt5640_out_r_mix)), /* Ouput Volume */ SND_SOC_DAPM_PGA("SPKVOL L", RT5640_PWR_VOL, RT5640_PWR_SV_L_BIT, 0, NULL, 0), @@ -1167,16 +1193,8 @@ static const struct snd_soc_dapm_widget rt5640_dapm_widgets[] = { 0, rt5640_spo_l_mix, ARRAY_SIZE(rt5640_spo_l_mix)), SND_SOC_DAPM_MIXER("SPOR MIX", SND_SOC_NOPM, 0, 0, rt5640_spo_r_mix, ARRAY_SIZE(rt5640_spo_r_mix)), - SND_SOC_DAPM_MIXER("HPO MIX L", SND_SOC_NOPM, 0, 0, - rt5640_hpo_mix, ARRAY_SIZE(rt5640_hpo_mix)), - SND_SOC_DAPM_MIXER("HPO MIX R", SND_SOC_NOPM, 0, 0, - rt5640_hpo_mix, ARRAY_SIZE(rt5640_hpo_mix)), SND_SOC_DAPM_MIXER("LOUT MIX", RT5640_PWR_ANLG1, RT5640_PWR_LM_BIT, 0, rt5640_lout_mix, ARRAY_SIZE(rt5640_lout_mix)), - SND_SOC_DAPM_MIXER("Mono MIX", RT5640_PWR_ANLG1, RT5640_PWR_MM_BIT, 0, - rt5640_mono_mix, ARRAY_SIZE(rt5640_mono_mix)), - SND_SOC_DAPM_SUPPLY("Improve MONO Amp Drv", RT5640_PWR_ANLG1, - RT5640_PWR_MA_BIT, 0, NULL, 0), SND_SOC_DAPM_SUPPLY_S("Improve HP Amp Drv", 1, SND_SOC_NOPM, 0, 0, rt5640_hp_power_event, SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_PGA_S("HP Amp", 1, SND_SOC_NOPM, 0, 0, @@ -1199,6 +1217,7 @@ static const struct snd_soc_dapm_widget rt5640_dapm_widgets[] = { SND_SOC_DAPM_SWITCH("HP R Playback", SND_SOC_NOPM, 0, 0, &hp_r_enable_control), SND_SOC_DAPM_POST("HP Post", rt5640_hp_post_event), + /* Output Lines */ SND_SOC_DAPM_OUTPUT("SPOLP"), SND_SOC_DAPM_OUTPUT("SPOLN"), @@ -1208,10 +1227,69 @@ static const struct snd_soc_dapm_widget rt5640_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("HPOR"), SND_SOC_DAPM_OUTPUT("LOUTL"), SND_SOC_DAPM_OUTPUT("LOUTR"), +}; + +static const struct snd_soc_dapm_widget rt5640_specific_dapm_widgets[] = { + /* Audio DSP */ + SND_SOC_DAPM_PGA("Audio DSP", SND_SOC_NOPM, 0, 0, NULL, 0), + /* ANC */ + SND_SOC_DAPM_PGA("ANC", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* DAC2 channel Mux */ + SND_SOC_DAPM_MUX("DAC L2 Mux", SND_SOC_NOPM, 0, 0, &rt5640_dac_l2_mux), + SND_SOC_DAPM_MUX("DAC R2 Mux", SND_SOC_NOPM, 0, 0, &rt5640_dac_r2_mux), + + SND_SOC_DAPM_MIXER("Stereo DAC MIXL", SND_SOC_NOPM, 0, 0, + rt5640_sto_dac_l_mix, ARRAY_SIZE(rt5640_sto_dac_l_mix)), + SND_SOC_DAPM_MIXER("Stereo DAC MIXR", SND_SOC_NOPM, 0, 0, + rt5640_sto_dac_r_mix, ARRAY_SIZE(rt5640_sto_dac_r_mix)), + + SND_SOC_DAPM_DAC("DAC R2", NULL, RT5640_PWR_DIG1, RT5640_PWR_DAC_R2_BIT, + 0), + SND_SOC_DAPM_DAC("DAC L2", NULL, RT5640_PWR_DIG1, RT5640_PWR_DAC_L2_BIT, + 0), + + SND_SOC_DAPM_MIXER("OUT MIXL", RT5640_PWR_MIXER, RT5640_PWR_OM_L_BIT, + 0, rt5640_out_l_mix, ARRAY_SIZE(rt5640_out_l_mix)), + SND_SOC_DAPM_MIXER("OUT MIXR", RT5640_PWR_MIXER, RT5640_PWR_OM_R_BIT, + 0, rt5640_out_r_mix, ARRAY_SIZE(rt5640_out_r_mix)), + + SND_SOC_DAPM_MIXER("HPO MIX L", SND_SOC_NOPM, 0, 0, + rt5640_hpo_mix, ARRAY_SIZE(rt5640_hpo_mix)), + SND_SOC_DAPM_MIXER("HPO MIX R", SND_SOC_NOPM, 0, 0, + rt5640_hpo_mix, ARRAY_SIZE(rt5640_hpo_mix)), + + SND_SOC_DAPM_MIXER("Mono MIX", RT5640_PWR_ANLG1, RT5640_PWR_MM_BIT, 0, + rt5640_mono_mix, ARRAY_SIZE(rt5640_mono_mix)), + SND_SOC_DAPM_SUPPLY("Improve MONO Amp Drv", RT5640_PWR_ANLG1, + RT5640_PWR_MA_BIT, 0, NULL, 0), + SND_SOC_DAPM_OUTPUT("MONOP"), SND_SOC_DAPM_OUTPUT("MONON"), }; +static const struct snd_soc_dapm_widget rt5639_specific_dapm_widgets[] = { + SND_SOC_DAPM_MIXER("Stereo DAC MIXL", SND_SOC_NOPM, 0, 0, + rt5639_sto_dac_l_mix, ARRAY_SIZE(rt5639_sto_dac_l_mix)), + SND_SOC_DAPM_MIXER("Stereo DAC MIXR", SND_SOC_NOPM, 0, 0, + rt5639_sto_dac_r_mix, ARRAY_SIZE(rt5639_sto_dac_r_mix)), + + SND_SOC_DAPM_SUPPLY("DAC L2 Filter", RT5640_PWR_DIG1, + RT5640_PWR_DAC_L2_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DAC R2 Filter", RT5640_PWR_DIG1, + RT5640_PWR_DAC_R2_BIT, 0, NULL, 0), + + SND_SOC_DAPM_MIXER("OUT MIXL", RT5640_PWR_MIXER, RT5640_PWR_OM_L_BIT, + 0, rt5639_out_l_mix, ARRAY_SIZE(rt5639_out_l_mix)), + SND_SOC_DAPM_MIXER("OUT MIXR", RT5640_PWR_MIXER, RT5640_PWR_OM_R_BIT, + 0, rt5639_out_r_mix, ARRAY_SIZE(rt5639_out_r_mix)), + + SND_SOC_DAPM_MIXER("HPO MIX L", SND_SOC_NOPM, 0, 0, + rt5639_hpo_mix, ARRAY_SIZE(rt5639_hpo_mix)), + SND_SOC_DAPM_MIXER("HPO MIX R", SND_SOC_NOPM, 0, 0, + rt5639_hpo_mix, ARRAY_SIZE(rt5639_hpo_mix)), +}; + static const struct snd_soc_dapm_route rt5640_dapm_routes[] = { {"IN1P", NULL, "LDO2"}, {"IN2P", NULL, "LDO2"}, @@ -1353,71 +1431,38 @@ static const struct snd_soc_dapm_route rt5640_dapm_routes[] = { {"DAC MIXR", "Stereo ADC Switch", "Stereo ADC MIXR"}, {"DAC MIXR", "INF1 Switch", "IF1 DAC R"}, - {"ANC", NULL, "Stereo ADC MIXL"}, - {"ANC", NULL, "Stereo ADC MIXR"}, - - {"Audio DSP", NULL, "DAC MIXL"}, - {"Audio DSP", NULL, "DAC MIXR"}, - - {"DAC L2 Mux", "IF2", "IF2 DAC L"}, - {"DAC L2 Mux", "Base L/R", "Audio DSP"}, - - {"DAC R2 Mux", "IF2", "IF2 DAC R"}, - {"Stereo DAC MIXL", "DAC L1 Switch", "DAC MIXL"}, - {"Stereo DAC MIXL", "DAC L2 Switch", "DAC L2 Mux"}, - {"Stereo DAC MIXL", "ANC Switch", "ANC"}, {"Stereo DAC MIXR", "DAC R1 Switch", "DAC MIXR"}, - {"Stereo DAC MIXR", "DAC R2 Switch", "DAC R2 Mux"}, - {"Stereo DAC MIXR", "ANC Switch", "ANC"}, {"Mono DAC MIXL", "DAC L1 Switch", "DAC MIXL"}, - {"Mono DAC MIXL", "DAC L2 Switch", "DAC L2 Mux"}, - {"Mono DAC MIXL", "DAC R2 Switch", "DAC R2 Mux"}, {"Mono DAC MIXR", "DAC R1 Switch", "DAC MIXR"}, - {"Mono DAC MIXR", "DAC R2 Switch", "DAC R2 Mux"}, - {"Mono DAC MIXR", "DAC L2 Switch", "DAC L2 Mux"}, {"DIG MIXL", "DAC L1 Switch", "DAC MIXL"}, - {"DIG MIXL", "DAC L2 Switch", "DAC L2 Mux"}, {"DIG MIXR", "DAC R1 Switch", "DAC MIXR"}, - {"DIG MIXR", "DAC R2 Switch", "DAC R2 Mux"}, {"DAC L1", NULL, "Stereo DAC MIXL"}, {"DAC L1", NULL, "PLL1", is_sys_clk_from_pll}, {"DAC R1", NULL, "Stereo DAC MIXR"}, {"DAC R1", NULL, "PLL1", is_sys_clk_from_pll}, - {"DAC L2", NULL, "Mono DAC MIXL"}, - {"DAC L2", NULL, "PLL1", is_sys_clk_from_pll}, - {"DAC R2", NULL, "Mono DAC MIXR"}, - {"DAC R2", NULL, "PLL1", is_sys_clk_from_pll}, {"SPK MIXL", "REC MIXL Switch", "RECMIXL"}, {"SPK MIXL", "INL Switch", "INL VOL"}, {"SPK MIXL", "DAC L1 Switch", "DAC L1"}, - {"SPK MIXL", "DAC L2 Switch", "DAC L2"}, {"SPK MIXL", "OUT MIXL Switch", "OUT MIXL"}, {"SPK MIXR", "REC MIXR Switch", "RECMIXR"}, {"SPK MIXR", "INR Switch", "INR VOL"}, {"SPK MIXR", "DAC R1 Switch", "DAC R1"}, - {"SPK MIXR", "DAC R2 Switch", "DAC R2"}, {"SPK MIXR", "OUT MIXR Switch", "OUT MIXR"}, - {"OUT MIXL", "SPK MIXL Switch", "SPK MIXL"}, {"OUT MIXL", "BST1 Switch", "BST1"}, {"OUT MIXL", "INL Switch", "INL VOL"}, {"OUT MIXL", "REC MIXL Switch", "RECMIXL"}, - {"OUT MIXL", "DAC R2 Switch", "DAC R2"}, - {"OUT MIXL", "DAC L2 Switch", "DAC L2"}, {"OUT MIXL", "DAC L1 Switch", "DAC L1"}, - {"OUT MIXR", "SPK MIXR Switch", "SPK MIXR"}, {"OUT MIXR", "BST2 Switch", "BST2"}, {"OUT MIXR", "BST1 Switch", "BST1"}, {"OUT MIXR", "INR Switch", "INR VOL"}, {"OUT MIXR", "REC MIXR Switch", "RECMIXR"}, - {"OUT MIXR", "DAC L2 Switch", "DAC L2"}, - {"OUT MIXR", "DAC R2 Switch", "DAC R2"}, {"OUT MIXR", "DAC R1 Switch", "DAC R1"}, {"SPKVOL L", NULL, "SPK MIXL"}, @@ -1436,11 +1481,9 @@ static const struct snd_soc_dapm_route rt5640_dapm_routes[] = { {"SPOR MIX", "SPKVOL R Switch", "SPKVOL R"}, {"SPOR MIX", "BST1 Switch", "BST1"}, - {"HPO MIX L", "HPO MIX DAC2 Switch", "DAC L2"}, {"HPO MIX L", "HPO MIX DAC1 Switch", "DAC L1"}, {"HPO MIX L", "HPO MIX HPVOL Switch", "HPOVOL L"}, {"HPO MIX L", NULL, "HP L Amp"}, - {"HPO MIX R", "HPO MIX DAC2 Switch", "DAC R2"}, {"HPO MIX R", "HPO MIX DAC1 Switch", "DAC R1"}, {"HPO MIX R", "HPO MIX HPVOL Switch", "HPOVOL R"}, {"HPO MIX R", NULL, "HP R Amp"}, @@ -1450,12 +1493,6 @@ static const struct snd_soc_dapm_route rt5640_dapm_routes[] = { {"LOUT MIX", "OUTVOL L Switch", "OUTVOL L"}, {"LOUT MIX", "OUTVOL R Switch", "OUTVOL R"}, - {"Mono MIX", "DAC R2 Switch", "DAC R2"}, - {"Mono MIX", "DAC L2 Switch", "DAC L2"}, - {"Mono MIX", "OUTVOL R Switch", "OUTVOL R"}, - {"Mono MIX", "OUTVOL L Switch", "OUTVOL L"}, - {"Mono MIX", "BST1 Switch", "BST1"}, - {"HP Amp", NULL, "HPO MIX L"}, {"HP Amp", NULL, "HPO MIX R"}, @@ -1480,11 +1517,82 @@ static const struct snd_soc_dapm_route rt5640_dapm_routes[] = { {"HPOR", NULL, "HP R Playback"}, {"LOUTL", NULL, "LOUT MIX"}, {"LOUTR", NULL, "LOUT MIX"}, +}; + +static const struct snd_soc_dapm_route rt5640_specific_dapm_routes[] = { + {"ANC", NULL, "Stereo ADC MIXL"}, + {"ANC", NULL, "Stereo ADC MIXR"}, + + {"Audio DSP", NULL, "DAC MIXL"}, + {"Audio DSP", NULL, "DAC MIXR"}, + + {"DAC L2 Mux", "IF2", "IF2 DAC L"}, + {"DAC L2 Mux", "Base L/R", "Audio DSP"}, + + {"DAC R2 Mux", "IF2", "IF2 DAC R"}, + + {"Stereo DAC MIXL", "DAC L2 Switch", "DAC L2 Mux"}, + {"Stereo DAC MIXL", "ANC Switch", "ANC"}, + {"Stereo DAC MIXR", "DAC R2 Switch", "DAC R2 Mux"}, + {"Stereo DAC MIXR", "ANC Switch", "ANC"}, + + {"Mono DAC MIXL", "DAC L2 Switch", "DAC L2 Mux"}, + {"Mono DAC MIXL", "DAC R2 Switch", "DAC R2 Mux"}, + + {"Mono DAC MIXR", "DAC R2 Switch", "DAC R2 Mux"}, + {"Mono DAC MIXR", "DAC L2 Switch", "DAC L2 Mux"}, + + {"DIG MIXR", "DAC R2 Switch", "DAC R2 Mux"}, + {"DIG MIXL", "DAC L2 Switch", "DAC L2 Mux"}, + + {"DAC L2", NULL, "Mono DAC MIXL"}, + {"DAC L2", NULL, "PLL1", is_sys_clk_from_pll}, + {"DAC R2", NULL, "Mono DAC MIXR"}, + {"DAC R2", NULL, "PLL1", is_sys_clk_from_pll}, + + {"SPK MIXL", "DAC L2 Switch", "DAC L2"}, + {"SPK MIXR", "DAC R2 Switch", "DAC R2"}, + + {"OUT MIXL", "SPK MIXL Switch", "SPK MIXL"}, + {"OUT MIXR", "SPK MIXR Switch", "SPK MIXR"}, + + {"OUT MIXL", "DAC R2 Switch", "DAC R2"}, + {"OUT MIXL", "DAC L2 Switch", "DAC L2"}, + + {"OUT MIXR", "DAC L2 Switch", "DAC L2"}, + {"OUT MIXR", "DAC R2 Switch", "DAC R2"}, + + {"HPO MIX L", "HPO MIX DAC2 Switch", "DAC L2"}, + {"HPO MIX R", "HPO MIX DAC2 Switch", "DAC R2"}, + + {"Mono MIX", "DAC R2 Switch", "DAC R2"}, + {"Mono MIX", "DAC L2 Switch", "DAC L2"}, + {"Mono MIX", "OUTVOL R Switch", "OUTVOL R"}, + {"Mono MIX", "OUTVOL L Switch", "OUTVOL L"}, + {"Mono MIX", "BST1 Switch", "BST1"}, + {"MONOP", NULL, "Mono MIX"}, {"MONON", NULL, "Mono MIX"}, {"MONOP", NULL, "Improve MONO Amp Drv"}, }; +static const struct snd_soc_dapm_route rt5639_specific_dapm_routes[] = { + {"Stereo DAC MIXL", "DAC L2 Switch", "IF2 DAC L"}, + {"Stereo DAC MIXR", "DAC R2 Switch", "IF2 DAC R"}, + + {"Mono DAC MIXL", "DAC L2 Switch", "IF2 DAC L"}, + {"Mono DAC MIXL", "DAC R2 Switch", "IF2 DAC R"}, + + {"Mono DAC MIXR", "DAC R2 Switch", "IF2 DAC R"}, + {"Mono DAC MIXR", "DAC L2 Switch", "IF2 DAC L"}, + + {"DIG MIXL", "DAC L2 Switch", "IF2 DAC L"}, + {"DIG MIXR", "DAC R2 Switch", "IF2 DAC R"}, + + {"IF2 DAC L", NULL, "DAC L2 Filter"}, + {"IF2 DAC R", NULL, "DAC R2 Filter"}, +}; + static int get_sdp_info(struct snd_soc_codec *codec, int dai_id) { int ret = 0, val; @@ -1551,8 +1659,7 @@ static int get_clk_info(int sclk, int rate) static int rt5640_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); unsigned int val_len = 0, val_clk, mask_clk; int dai_sel, pre_div, bclk_ms, frame_size; @@ -1901,6 +2008,28 @@ static int rt5640_probe(struct snd_soc_codec *codec) snd_soc_update_bits(codec, RT5640_MICBIAS, 0x0030, 0x0030); snd_soc_update_bits(codec, RT5640_DSP_PATH2, 0xfc00, 0x0c00); + switch (snd_soc_read(codec, RT5640_RESET)) { + case RT5640_RESET_ID: + snd_soc_add_codec_controls(codec, + rt5640_specific_snd_controls, + ARRAY_SIZE(rt5640_specific_snd_controls)); + snd_soc_dapm_new_controls(&codec->dapm, + rt5640_specific_dapm_widgets, + ARRAY_SIZE(rt5640_specific_dapm_widgets)); + snd_soc_dapm_add_routes(&codec->dapm, + rt5640_specific_dapm_routes, + ARRAY_SIZE(rt5640_specific_dapm_routes)); + break; + case RT5639_RESET_ID: + snd_soc_dapm_new_controls(&codec->dapm, + rt5639_specific_dapm_widgets, + ARRAY_SIZE(rt5639_specific_dapm_widgets)); + snd_soc_dapm_add_routes(&codec->dapm, + rt5639_specific_dapm_routes, + ARRAY_SIZE(rt5639_specific_dapm_routes)); + break; + } + return 0; } diff --git a/sound/soc/codecs/rt5640.h b/sound/soc/codecs/rt5640.h index d7bd525..3b50459 100644 --- a/sound/soc/codecs/rt5640.h +++ b/sound/soc/codecs/rt5640.h @@ -14,6 +14,9 @@ #include <sound/rt5640.h> +#define RT5639_RESET_ID 0x0008 +#define RT5640_RESET_ID 0x000c + /* Info */ #define RT5640_RESET 0x00 #define RT5640_VENDOR_ID 0xfd -- 1.8.1.1.439.g50a6b54

2 2

[alsa-devel] [PATCH] ASoC: core: Allow snd_soc_update_bits use 32 bits register
by Bard Liao 08 Apr '14

08 Apr '14

Change reg's type from unsigned short to unsigned int. So that we can use 32 bits reg value in snd_soc_update_bits. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- include/sound/soc.h | 6 +++--- sound/soc/soc-core.c | 8 ++++---- 2 files changed, 7 insertions(+), 7 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 5878410..b18cf27 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -469,12 +469,12 @@ static inline void snd_soc_jack_free_gpios(struct snd_soc_jack *jack, int count, #endif /* codec register bit access */ -int snd_soc_update_bits(struct snd_soc_codec *codec, unsigned short reg, +int snd_soc_update_bits(struct snd_soc_codec *codec, unsigned int reg, unsigned int mask, unsigned int value); int snd_soc_update_bits_locked(struct snd_soc_codec *codec, - unsigned short reg, unsigned int mask, + unsigned int reg, unsigned int mask, unsigned int value); -int snd_soc_test_bits(struct snd_soc_codec *codec, unsigned short reg, +int snd_soc_test_bits(struct snd_soc_codec *codec, unsigned int reg, unsigned int mask, unsigned int value); int snd_soc_new_ac97_codec(struct snd_soc_codec *codec, diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index cd52d52..7e2734d 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2316,7 +2316,7 @@ EXPORT_SYMBOL_GPL(snd_soc_write); * * Returns 1 for change, 0 for no change, or negative error code. */ -int snd_soc_update_bits(struct snd_soc_codec *codec, unsigned short reg, +int snd_soc_update_bits(struct snd_soc_codec *codec, unsigned int reg, unsigned int mask, unsigned int value) { bool change; @@ -2357,7 +2357,7 @@ EXPORT_SYMBOL_GPL(snd_soc_update_bits); * Returns 1 for change else 0. */ int snd_soc_update_bits_locked(struct snd_soc_codec *codec, - unsigned short reg, unsigned int mask, + unsigned int reg, unsigned int mask, unsigned int value) { int change; @@ -2382,7 +2382,7 @@ EXPORT_SYMBOL_GPL(snd_soc_update_bits_locked); * * Returns 1 for change else 0. */ -int snd_soc_test_bits(struct snd_soc_codec *codec, unsigned short reg, +int snd_soc_test_bits(struct snd_soc_codec *codec, unsigned int reg, unsigned int mask, unsigned int value) { int change; @@ -2897,7 +2897,7 @@ int snd_soc_put_volsw_sx(struct snd_kcontrol *kcontrol, int min = mc->min; int mask = (1 << (fls(min + max) - 1)) - 1; int err = 0; - unsigned short val, val_mask, val2 = 0; + unsigned int val, val_mask, val2 = 0; val_mask = mask << shift; val = (ucontrol->value.integer.value[0] + min) & mask; -- 1.8.1.1.439.g50a6b54

2 1

Re: [alsa-devel] Tenor TE8802, USB : clics and crackles during music play
by mick 08 Apr '14

08 Apr '14

Clemens, please help me out: are you asking me personally to test or is this a general call for the group? If you need me to test, what version of kernel sources do I patch? I am probably not running the latest. Forgive the questions, I do not routinely do these things. Also: Your patch looks quite different from the solution I pointed out. Why? Did you test it yourself? In particular the assertion: > + * The TEAC UD-H01 firmware sometimes changes the feedback value > + * by +/- 0x1.0000. is quite different from the assertion "the 3rd byte of the feadback value is getting corrupted and probably changing when it should not " from http://forums.slimdevices.com/showthread.php?93310-Async-USB-Dac-Experiment…. Also, you made it quite explicit for the TEAC UD-H01 but I think it should be more general. I have the Teac A-H01 and it also has the problem. As far as I gather from the various discussions on the Web it probably is a general Tenor 8802 issue. For instance with Nuforce DAC-9 here: http://www.spinics.net/linux/fedora/alsa-user/msg11387.html Mick On Mon, 07 Apr 2014 22:09:21 +0200, Clemens Ladisch <clemens(a)ladisch.de> wrote: > Please test this patch: > > --8<----------------------------------------------------------------->8-- > ALSA: usb-audio: work around corrupted TEAC UD-H01 feedback data > > The TEAC UD-H01 firmware sends wrong feedback frequency values, thus > causing the PC to send the samples at a wrong rate, which results in > clicks and crackles in the output. > > Add a workaround to detect and fix the corruption. > > Signed-off-by: Clemens Ladisch <clemens(a)ladisch.de> > Tested-by: Mick <mick37(a)gmx.de> > Cc: <stable(a)vger.kernel.org> > --- > sound/usb/card.h | 1 + > sound/usb/endpoint.c | 15 ++++++++++++++- > 2 files changed, 15 insertions(+), 1 deletion(-) > > diff --git a/sound/usb/card.h b/sound/usb/card.h > index 9867ab8..97acb90 100644 > --- a/sound/usb/card.h > +++ b/sound/usb/card.h > @@ -92,6 +92,7 @@ struct snd_usb_endpoint { > unsigned int curframesize; /* current packet size in frames (for > capture) */ > unsigned int syncmaxsize; /* sync endpoint packet size */ > unsigned int fill_max:1; /* fill max packet size always */ > + unsigned int udh01_fb_quirk:1; /* corrupted feedback data */ > unsigned int datainterval; /* log_2 of data packet interval */ > unsigned int syncinterval; /* P for adaptive mode, 0 otherwise */ > unsigned char silence_value; > diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c > index e70a87e..7b400eb 100644 > --- a/sound/usb/endpoint.c > +++ b/sound/usb/endpoint.c > @@ -471,6 +471,10 @@ struct snd_usb_endpoint > *snd_usb_add_endpoint(struct snd_usb_audio *chip, > ep->syncinterval = 3; > > ep->syncmaxsize = le16_to_cpu(get_endpoint(alts, 1)->wMaxPacketSize); > + > + if (chip->usb_id == USB_ID(0x0644, 0x8038) /* TEAC UD-H01 */ && > + ep->syncmaxsize == 4) > + ep->udh01_fb_quirk = 1; > } > > list_add_tail(&ep->list, &chip->ep_list); > @@ -1105,7 +1109,16 @@ void snd_usb_handle_sync_urb(struct > snd_usb_endpoint *ep, > if (f == 0) > return; > > - if (unlikely(ep->freqshift == INT_MIN)) { > + if (unlikely(ep->udh01_fb_quirk)) { > + /* > + * The TEAC UD-H01 firmware sometimes changes the feedback value > + * by +/- 0x1.0000. > + */ > + if (f < ep->freqn - 0x8000) > + f += 0x10000; > + else if (f > ep->freqn + 0x8000) > + f -= 0x10000; > + } else if (unlikely(ep->freqshift == INT_MIN)) { > /* > * The first time we see a feedback value, determine its format > * by shifting it left or right until it matches the nominal -- Erstellt mit Operas E-Mail-Modul: http://www.opera.com/mail/

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Re: [alsa-devel] [PATCH] ALSA: hda - Do not assign streams in reverse order
by Takashi Iwai 08 Apr '14

08 Apr '14

At Mon, 7 Apr 2014 22:36:38 +0300, Anssi Hannula wrote: > > Currently stream numbers are assigned in reverse order. > > Unfortunately commit 7546abfb8e1f9933b5 ("ALSA: hda - Increment > default stream numbers for AMD HDMI controllers") assumed this was not > the case (specifically, it had the "old cards had single device only" > => "extra unused stream numbers do not matter" assumption), causing > non-working audio regressions for AMD Radeon HDMI users on v3.14. > > Change the stream numbers to be assigned in increasing order instead. > > Bugzilla: https://bugs.freedesktop.org/show_bug.cgi?id=77002 > Reported-by: Christian Güdel <cg(a)dmesg.ch> > Signed-off-by: Anssi Hannula <anssi.hannula(a)iki.fi> > Tested-by: Christian Güdel <cg(a)dmesg.ch> > Cc: Alex Deucher <alexander.deucher(a)amd.com> > Cc: <stable(a)vger.kernel.org> # 3.14 > --- > > Takashi, please check I didn't miss anything (I'm not that familiar > with this low-level HDA code). :) In that case, let's revert the commit instead. There was a reason to behave in that way, although I don't remember well. That said, this change might lead to another regression. Thus, if a fix is needed for stable, the revert would be the best choice as a first aid. Of course, this change is good to have, so let's implement this for 3.15 and see whether it works well for all nasty controllers. Besides, this patch can't be applied any longer to the current code base since it's been rewritten a lot; most of codes have been moved to another file hda_controller.c. Could you rebase the patch for the latest tree? Meanwhile I'll prepare a revert that is applied only to 3.14.x. thanks, Takashi > > > sound/pci/hda/hda_intel.c | 9 +++++---- > 1 file changed, 5 insertions(+), 4 deletions(-) > > diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c > index e354ab1ec20f..7a7b7b10dddf 100644 > --- a/sound/pci/hda/hda_intel.c > +++ b/sound/pci/hda/hda_intel.c > @@ -1856,13 +1856,14 @@ azx_assign_device(struct azx *chip, struct snd_pcm_substream *substream) > struct azx_dev *azx_dev = &chip->azx_dev[dev]; > dsp_lock(azx_dev); > if (!azx_dev->opened && !dsp_is_locked(azx_dev)) { > - res = azx_dev; > - if (res->assigned_key == key) { > - res->opened = 1; > - res->assigned_key = key; > + if (azx_dev->assigned_key == key) { > + azx_dev->opened = 1; > + azx_dev->assigned_key = key; > dsp_unlock(azx_dev); > return azx_dev; > } > + if (!res) > + res = azx_dev; > } > dsp_unlock(azx_dev); > } > -- > 1.8.4.5 >

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