August 2013 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: rt5640: change widget sequence for depop
by bardliao＠realtek.com 21 Aug '13

21 Aug '13

From: Bard Liao <bardliao(a)realtek.com> Add mute/unmute control in widget event and modify the power on/off sequence to avoid pop noise. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5640.c | 200 +++++++++++++++++++++++++++++++++------------- sound/soc/codecs/rt5640.h | 11 +++ 2 files changed, 157 insertions(+), 54 deletions(-) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 4db7314..218aae2 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -50,8 +50,6 @@ static const struct regmap_range_cfg rt5640_ranges[] = { static struct reg_default init_list[] = { {RT5640_PR_BASE + 0x3d, 0x3600}, - {RT5640_PR_BASE + 0x1c, 0x0D21}, - {RT5640_PR_BASE + 0x1b, 0x0000}, {RT5640_PR_BASE + 0x12, 0x0aa8}, {RT5640_PR_BASE + 0x14, 0x0aaa}, {RT5640_PR_BASE + 0x20, 0x6110}, @@ -384,15 +382,11 @@ static const SOC_ENUM_SINGLE_DECL( static const struct snd_kcontrol_new rt5640_snd_controls[] = { /* Speaker Output Volume */ - SOC_DOUBLE("Speaker Playback Switch", RT5640_SPK_VOL, - RT5640_L_MUTE_SFT, RT5640_R_MUTE_SFT, 1, 1), SOC_DOUBLE("Speaker Channel Switch", RT5640_SPK_VOL, 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 Playback Switch", RT5640_HP_VOL, - RT5640_L_MUTE_SFT, RT5640_R_MUTE_SFT, 1, 1), 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, @@ -737,6 +731,22 @@ static const struct snd_kcontrol_new rt5640_mono_mix[] = { RT5640_M_BST1_MM_SFT, 1, 1), }; +static const struct snd_kcontrol_new spk_l_enable_control = + SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT5640_SPK_VOL, + RT5640_L_MUTE_SFT, 1, 1); + +static const struct snd_kcontrol_new spk_r_enable_control = + SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT5640_SPK_VOL, + RT5640_R_MUTE_SFT, 1, 1); + +static const struct snd_kcontrol_new hp_l_enable_control = + SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT5640_HP_VOL, + RT5640_L_MUTE_SFT, 1, 1); + +static const struct snd_kcontrol_new hp_r_enable_control = + SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT5640_HP_VOL, + RT5640_R_MUTE_SFT, 1, 1); + /* Stereo ADC source */ static const char * const rt5640_stereo_adc1_src[] = { "DIG MIX", "ADC" @@ -868,33 +878,6 @@ static const SOC_ENUM_SINGLE_DECL( static const struct snd_kcontrol_new rt5640_sdi_mux = SOC_DAPM_ENUM("SDI select", rt5640_sdi_sel_enum); -static int spk_event(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, int event) -{ - struct snd_soc_codec *codec = w->codec; - struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); - - switch (event) { - case SND_SOC_DAPM_POST_PMU: - regmap_update_bits(rt5640->regmap, RT5640_PWR_DIG1, - 0x0001, 0x0001); - regmap_update_bits(rt5640->regmap, RT5640_PR_BASE + 0x1c, - 0xf000, 0xf000); - break; - - case SND_SOC_DAPM_PRE_PMD: - regmap_update_bits(rt5640->regmap, RT5640_PR_BASE + 0x1c, - 0xf000, 0x0000); - regmap_update_bits(rt5640->regmap, RT5640_PWR_DIG1, - 0x0001, 0x0000); - break; - - default: - return 0; - } - return 0; -} - static int rt5640_set_dmic1_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -943,6 +926,103 @@ static int rt5640_set_dmic2_event(struct snd_soc_dapm_widget *w, return 0; } +void hp_amp_power_on(struct snd_soc_codec *codec) +{ + struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); + + /* depop parameters */ + regmap_update_bits(rt5640->regmap, RT5640_PR_BASE + + RT5640_CHPUMP_INT_REG1, 0x0700, 0x0200); + regmap_update_bits(rt5640->regmap, RT5640_DEPOP_M2, + RT5640_DEPOP_MASK, RT5640_DEPOP_MAN); + regmap_update_bits(rt5640->regmap, RT5640_DEPOP_M1, + RT5640_HP_CP_MASK | RT5640_HP_SG_MASK | RT5640_HP_CB_MASK, + RT5640_HP_CP_PU | RT5640_HP_SG_DIS | RT5640_HP_CB_PU); + regmap_write(rt5640->regmap, RT5640_PR_BASE + RT5640_HP_DCC_INT1, + 0x9f00); + /* headphone amp power on */ + regmap_update_bits(rt5640->regmap, RT5640_PWR_ANLG1, + RT5640_PWR_FV1 | RT5640_PWR_FV2, 0); + regmap_update_bits(rt5640->regmap, RT5640_PWR_ANLG1, + RT5640_PWR_HA, + RT5640_PWR_HA); + usleep_range(30000, 35000); + regmap_update_bits(rt5640->regmap, RT5640_PWR_ANLG1, + RT5640_PWR_FV1 | RT5640_PWR_FV2 , + RT5640_PWR_FV1 | RT5640_PWR_FV2); +} + +static void rt5640_pmu_depop(struct snd_soc_codec *codec) +{ + struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); + + regmap_update_bits(rt5640->regmap, RT5640_DEPOP_M2, + RT5640_DEPOP_MASK | RT5640_DIG_DP_MASK, + RT5640_DEPOP_AUTO | RT5640_DIG_DP_EN); + regmap_update_bits(rt5640->regmap, RT5640_CHARGE_PUMP, + RT5640_PM_HP_MASK, RT5640_PM_HP_HV); + + regmap_update_bits(rt5640->regmap, RT5640_DEPOP_M3, + RT5640_CP_FQ1_MASK | RT5640_CP_FQ2_MASK | RT5640_CP_FQ3_MASK, + (RT5640_CP_FQ_192_KHZ << RT5640_CP_FQ1_SFT) | + (RT5640_CP_FQ_12_KHZ << RT5640_CP_FQ2_SFT) | + (RT5640_CP_FQ_192_KHZ << RT5640_CP_FQ3_SFT)); + + regmap_write(rt5640->regmap, RT5640_PR_BASE + + RT5640_MAMP_INT_REG2, 0x1c00); + regmap_update_bits(rt5640->regmap, RT5640_DEPOP_M1, + RT5640_HP_CP_MASK | RT5640_HP_SG_MASK, + RT5640_HP_CP_PD | RT5640_HP_SG_EN); + regmap_update_bits(rt5640->regmap, RT5640_PR_BASE + + RT5640_CHPUMP_INT_REG1, 0x0700, 0x0400); +} + +static int rt5640_hp_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + unsigned int val = 0, pow; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + rt5640_pmu_depop(codec); + pow = snd_soc_read(codec, RT5640_PWR_ANLG1); + if (!(pow & RT5640_PWR_HP_L)) + val |= RT5640_L_MUTE; + if (!(pow & RT5640_PWR_HP_R)) + val |= RT5640_R_MUTE; + snd_soc_update_bits(codec, RT5640_HP_VOL, + RT5640_L_MUTE | RT5640_R_MUTE, val); + usleep_range(35000, 40000); + break; + + case SND_SOC_DAPM_PRE_PMD: + usleep_range(70000, 75000); + break; + + default: + return 0; + } + + return 0; +} + +static int rt5640_hp_power_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + hp_amp_power_on(codec); + break; + default: + return 0; + } + + return 0; +} + static const struct snd_soc_dapm_widget rt5640_dapm_widgets[] = { SND_SOC_DAPM_SUPPLY("PLL1", RT5640_PWR_ANLG2, RT5640_PWR_PLL_BIT, 0, NULL, 0), @@ -1132,15 +1212,27 @@ static const struct snd_soc_dapm_widget rt5640_dapm_widgets[] = { 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("Improve HP Amp Drv", RT5640_PWR_ANLG1, - SND_SOC_NOPM, 0, NULL, 0), - SND_SOC_DAPM_PGA("HP L Amp", RT5640_PWR_ANLG1, + 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, + rt5640_hp_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_SUPPLY("HP L Amp", RT5640_PWR_ANLG1, RT5640_PWR_HP_L_BIT, 0, NULL, 0), - SND_SOC_DAPM_PGA("HP R Amp", RT5640_PWR_ANLG1, + SND_SOC_DAPM_SUPPLY("HP R Amp", RT5640_PWR_ANLG1, RT5640_PWR_HP_R_BIT, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("Improve SPK Amp Drv", RT5640_PWR_DIG1, - SND_SOC_NOPM, 0, spk_event, - SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + RT5640_PWR_CLS_D_BIT, 0, NULL, 0), + + /* Output Switch */ + SND_SOC_DAPM_SWITCH("Speaker L Playback", SND_SOC_NOPM, 0, 0, + &spk_l_enable_control), + SND_SOC_DAPM_SWITCH("Speaker R Playback", SND_SOC_NOPM, 0, 0, + &spk_r_enable_control), + SND_SOC_DAPM_SWITCH("HP L Playback", SND_SOC_NOPM, 0, 0, + &hp_l_enable_control), + SND_SOC_DAPM_SWITCH("HP R Playback", SND_SOC_NOPM, 0, 0, + &hp_r_enable_control), /* Output Lines */ SND_SOC_DAPM_OUTPUT("SPOLP"), SND_SOC_DAPM_OUTPUT("SPOLN"), @@ -1381,9 +1473,11 @@ static const struct snd_soc_dapm_route rt5640_dapm_routes[] = { {"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"}, {"LOUT MIX", "DAC L1 Switch", "DAC L1"}, {"LOUT MIX", "DAC R1 Switch", "DAC R1"}, @@ -1396,13 +1490,15 @@ static const struct snd_soc_dapm_route rt5640_dapm_routes[] = { {"Mono MIX", "OUTVOL L Switch", "OUTVOL L"}, {"Mono MIX", "BST1 Switch", "BST1"}, - {"HP L Amp", NULL, "HPO MIX L"}, - {"HP R Amp", NULL, "HPO MIX R"}, + {"HP Amp", NULL, "HPO MIX L"}, + {"HP Amp", NULL, "HPO MIX R"}, - {"SPOLP", NULL, "SPOL MIX"}, - {"SPOLN", NULL, "SPOL MIX"}, - {"SPORP", NULL, "SPOR MIX"}, - {"SPORN", NULL, "SPOR MIX"}, + {"Speaker L Playback", "Switch", "SPOL MIX"}, + {"Speaker R Playback", "Switch", "SPOR MIX"}, + {"SPOLP", NULL, "Speaker L Playback"}, + {"SPOLN", NULL, "Speaker L Playback"}, + {"SPORP", NULL, "Speaker R Playback"}, + {"SPORN", NULL, "Speaker R Playback"}, {"SPOLP", NULL, "Improve SPK Amp Drv"}, {"SPOLN", NULL, "Improve SPK Amp Drv"}, @@ -1412,8 +1508,10 @@ static const struct snd_soc_dapm_route rt5640_dapm_routes[] = { {"HPOL", NULL, "Improve HP Amp Drv"}, {"HPOR", NULL, "Improve HP Amp Drv"}, - {"HPOL", NULL, "HP L Amp"}, - {"HPOR", NULL, "HP R Amp"}, + {"HP L Playback", "Switch", "HP Amp"}, + {"HP R Playback", "Switch", "HP Amp"}, + {"HPOL", NULL, "HP L Playback"}, + {"HPOR", NULL, "HP R Playback"}, {"LOUTL", NULL, "LOUT MIX"}, {"LOUTR", NULL, "LOUT MIX"}, {"MONOP", NULL, "Mono MIX"}, @@ -1792,17 +1890,13 @@ static int rt5640_set_bias_level(struct snd_soc_codec *codec, RT5640_PWR_BG | RT5640_PWR_VREF2, RT5640_PWR_VREF1 | RT5640_PWR_MB | RT5640_PWR_BG | RT5640_PWR_VREF2); - mdelay(10); + usleep_range(15000, 20000); snd_soc_update_bits(codec, RT5640_PWR_ANLG1, RT5640_PWR_FV1 | RT5640_PWR_FV2, RT5640_PWR_FV1 | RT5640_PWR_FV2); regcache_sync(rt5640->regmap); snd_soc_update_bits(codec, RT5640_DUMMY1, 0x0301, 0x0301); - snd_soc_update_bits(codec, RT5640_DEPOP_M1, - 0x001d, 0x0019); - snd_soc_update_bits(codec, RT5640_DEPOP_M2, - 0x2000, 0x2000); snd_soc_update_bits(codec, RT5640_MICBIAS, 0x0030, 0x0030); } @@ -1846,8 +1940,6 @@ static int rt5640_probe(struct snd_soc_codec *codec) rt5640_set_bias_level(codec, SND_SOC_BIAS_OFF); snd_soc_update_bits(codec, RT5640_DUMMY1, 0x0301, 0x0301); - snd_soc_update_bits(codec, RT5640_DEPOP_M1, 0x001d, 0x0019); - snd_soc_update_bits(codec, RT5640_DEPOP_M2, 0x2000, 0x2000); snd_soc_update_bits(codec, RT5640_MICBIAS, 0x0030, 0x0030); snd_soc_update_bits(codec, RT5640_DSP_PATH2, 0xfc00, 0x0c00); diff --git a/sound/soc/codecs/rt5640.h b/sound/soc/codecs/rt5640.h index c48286d..4d2cfe7 100644 --- a/sound/soc/codecs/rt5640.h +++ b/sound/soc/codecs/rt5640.h @@ -145,6 +145,8 @@ /* Index of Codec Private Register definition */ +#define RT5640_CHPUMP_INT_REG1 0x24 +#define RT5640_MAMP_INT_REG2 0x37 #define RT5640_3D_SPK 0x63 #define RT5640_WND_1 0x6c #define RT5640_WND_2 0x6d @@ -153,6 +155,7 @@ #define RT5640_WND_5 0x70 #define RT5640_WND_8 0x73 #define RT5640_DIP_SPK_INF 0x75 +#define RT5640_HP_DCC_INT1 0x77 #define RT5640_EQ_BW_LOP 0xa0 #define RT5640_EQ_GN_LOP 0xa1 #define RT5640_EQ_FC_BP1 0xa2 @@ -1201,6 +1204,14 @@ #define RT5640_CP_FQ2_SFT 4 #define RT5640_CP_FQ3_MASK (0x7) #define RT5640_CP_FQ3_SFT 0 +#define RT5640_CP_FQ_1_5_KHZ 0 +#define RT5640_CP_FQ_3_KHZ 1 +#define RT5640_CP_FQ_6_KHZ 2 +#define RT5640_CP_FQ_12_KHZ 3 +#define RT5640_CP_FQ_24_KHZ 4 +#define RT5640_CP_FQ_48_KHZ 5 +#define RT5640_CP_FQ_96_KHZ 6 +#define RT5640_CP_FQ_192_KHZ 7 /* HPOUT charge pump (0x91) */ #define RT5640_OSW_L_MASK (0x1 << 11) -- 1.8.1.1.439.g50a6b54

4 4

[alsa-devel] [PATCH v8 0/2] Add freescale S/PDIF CPU DAI and machine drivers
by Nicolin Chen 21 Aug '13

21 Aug '13

Changelog: v7->v8: * Revised DT binding doc. * Revised dev_err() error msg. * Check return value of clk_set_rate(). v6->v7: * Removed extra comma in array. * Revised some comments. * Added some dev_err(). * Check the return value of soft reset. * Use bitrev8() instead of extra reverse_bits(). * Use cache_bypass as default for regmap. v5->v6: * Sorted out rxtx clk source in DT binding. * Use devm_xxxx() functions. * Use platform_get_resource() instead. v4->v5: * Dropped rx/tx-clksrc-names DT bindings. * Use standard clock binding instead to pass the clock source list. * Update the compatible list by using "imx35", the first SoC that has spdif. v3->v4: * Use regmap for CPU DAI driver. * Use individual clock source for 32KHz, 44KHz, 48KHz playback. * Determine clock source configuration from 'clocks' entry. * Added imx53 to compatible list, merged imx6q and imx6dl in the list. * Improve the algorism of reverse_bits(). * Dropped the unneeded clk_put(). v2->v3: * Removed a wrong tag from the commit of patch-1. v1->v2: * Dropped one applied patch for spdif dummy codec drivers. * Use generic DMA DT binding. * Let spdif controller driver calculate the clock div. * Added one optional clock source for spdif tx. * Reivsed documentation accordingly. Nicolin Chen (2): ASoC: fsl: Add S/PDIF CPU DAI driver ASoC: fsl: Add S/PDIF machine driver .../devicetree/bindings/sound/fsl,spdif.txt | 54 + .../devicetree/bindings/sound/imx-audio-spdif.txt | 29 + sound/soc/fsl/Kconfig | 14 + sound/soc/fsl/Makefile | 4 + sound/soc/fsl/fsl_spdif.c | 1281 ++++++++++++++++++++ sound/soc/fsl/fsl_spdif.h | 224 ++++ sound/soc/fsl/imx-spdif.c | 134 ++ 7 files changed, 1740 insertions(+), 0 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/fsl,spdif.txt create mode 100644 Documentation/devicetree/bindings/sound/imx-audio-spdif.txt create mode 100644 sound/soc/fsl/fsl_spdif.c create mode 100644 sound/soc/fsl/fsl_spdif.h create mode 100644 sound/soc/fsl/imx-spdif.c

4 15

[alsa-devel] [PATCH v9 0/2] Add freescale S/PDIF CPU DAI and machine drivers
by Nicolin Chen 20 Aug '13

20 Aug '13

Changelog: v8->v9: * Use bool instead of atomic_t. * Use clk_prepare_enable() instead. * Dropped dumpregs(). * Dropped unnecessary clk_round_rate(). * Dropped unused clock source enum. * Revised dev_err() message. v7->v8: * Revised DT binding doc. * Revised dev_err() error msg. * Check return value of clk_set_rate(). v6->v7: * Removed extra comma in array. * Revised some comments. * Added some dev_err(). * Check the return value of soft reset. * Use bitrev8() instead of extra reverse_bits(). * Use cache_bypass as default for regmap. v5->v6: * Sorted out rxtx clk source in DT binding. * Use devm_xxxx() functions. * Use platform_get_resource() instead. v4->v5: * Dropped rx/tx-clksrc-names DT bindings. * Use standard clock binding instead to pass the clock source list. * Update the compatible list by using "imx35", the first SoC that has spdif. v3->v4: * Use regmap for CPU DAI driver. * Use individual clock source for 32KHz, 44KHz, 48KHz playback. * Determine clock source configuration from 'clocks' entry. * Added imx53 to compatible list, merged imx6q and imx6dl in the list. * Improve the algorism of reverse_bits(). * Dropped the unneeded clk_put(). v2->v3: * Removed a wrong tag from the commit of patch-1. v1->v2: * Dropped one applied patch for spdif dummy codec drivers. * Use generic DMA DT binding. * Let spdif controller driver calculate the clock div. * Added one optional clock source for spdif tx. * Reivsed documentation accordingly. Nicolin Chen (2): ASoC: fsl: Add S/PDIF CPU DAI driver ASoC: fsl: Add S/PDIF machine driver .../devicetree/bindings/sound/fsl,spdif.txt | 54 + .../devicetree/bindings/sound/imx-audio-spdif.txt | 29 + sound/soc/fsl/Kconfig | 14 + sound/soc/fsl/Makefile | 4 + sound/soc/fsl/fsl_spdif.c | 1236 ++++++++++++++++++++ sound/soc/fsl/fsl_spdif.h | 191 +++ sound/soc/fsl/imx-spdif.c | 134 +++ 7 files changed, 1662 insertions(+), 0 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/fsl,spdif.txt create mode 100644 Documentation/devicetree/bindings/sound/imx-audio-spdif.txt create mode 100644 sound/soc/fsl/fsl_spdif.c create mode 100644 sound/soc/fsl/fsl_spdif.h create mode 100644 sound/soc/fsl/imx-spdif.c

4 7

[alsa-devel] [PATCH v7 0/2] Add freescale S/PDIF CPU DAI and machine drivers
by Nicolin Chen 20 Aug '13

20 Aug '13

Changelog: v6->v7: * Removed extra comma in array. * Revised some comments. * Added some dev_err(). * Check the return value of soft reset. * Use bitrev8() instead of extra reverse_bits(). * Use cache_bypass as default for regmap. v5->v6: * Sorted out rxtx clk source in DT binding. * Use devm_xxxx() functions. * Use platform_get_resource() instead. v4->v5: * Dropped rx/tx-clksrc-names DT bindings. * Use standard clock binding instead to pass the clock source list. * Update the compatible list by using "imx35", the first SoC that has spdif. v3->v4: * Use regmap for CPU DAI driver. * Use individual clock source for 32KHz, 44KHz, 48KHz playback. * Determine clock source configuration from 'clocks' entry. * Added imx53 to compatible list, merged imx6q and imx6dl in the list. * Improve the algorism of reverse_bits(). * Dropped the unneeded clk_put(). v2->v3: * Removed a wrong tag from the commit of patch-1. v1->v2: * Dropped one applied patch for spdif dummy codec drivers. * Use generic DMA DT binding. * Let spdif controller driver calculate the clock div. * Added one optional clock source for spdif tx. * Reivsed documentation accordingly. Nicolin Chen (2): ASoC: fsl: Add S/PDIF CPU DAI driver ASoC: fsl: Add S/PDIF machine driver .../devicetree/bindings/sound/fsl,spdif.txt | 56 + .../devicetree/bindings/sound/imx-audio-spdif.txt | 29 + sound/soc/fsl/Kconfig | 14 + sound/soc/fsl/Makefile | 4 + sound/soc/fsl/fsl_spdif.c | 1277 ++++++++++++++++++++ sound/soc/fsl/fsl_spdif.h | 224 ++++ sound/soc/fsl/imx-spdif.c | 134 ++ 7 files changed, 1738 insertions(+), 0 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/fsl,spdif.txt create mode 100644 Documentation/devicetree/bindings/sound/imx-audio-spdif.txt create mode 100644 sound/soc/fsl/fsl_spdif.c create mode 100644 sound/soc/fsl/fsl_spdif.h create mode 100644 sound/soc/fsl/imx-spdif.c

6 21

[alsa-devel] [PATCH 0/6] simplify platform_get_resource_byname/devm_ioremap_resource
by Julia Lawall 20 Aug '13

20 Aug '13

devm_ioremap_resource often uses the result of a call to platform_get_resource_byname as its last argument. devm_ioremap_resource does appropriate error handling on this argument, so error handling can be removed from the call to platform_get_resource_byname. The semantic patch that makes this change is as follows: (http://coccinelle.lip6.fr/) // <smpl> @@ expression pdev,res,n,e,e1; expression ret != 0; identifier l,f; expression list es; @@ ( res = f(...); - if (res == NULL) { ... \(goto l;\|return ret;\) } e = devm_ioremap_resource(e1, res); | - res = f(es); ... when != res - if (res == NULL) { ... \(goto l;\|return ret;\) } ... when != res + res = f(es); e = devm_ioremap_resource(e1, res); ) // </smpl> In practice, f is always platform_get_resource_byname (or platform_get_resource, which was handled by a previous patch series). And the call to platform_get_resource_byname always immediately precedes the call to devm_ioremap_resource.

3 3

[alsa-devel] [PATCH] ASoC: fsl: fsl_ssi: Fix the order of resources removal
by Fabio Estevam 20 Aug '13

20 Aug '13

From: Fabio Estevam <fabio.estevam(a)freescale.com> In fsl_ssi_remove() we need to remove the resources in the opposite order that they were acquired in probe. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/fsl/fsl_ssi.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 0c072ff..c8aee4e 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -928,14 +928,14 @@ static int fsl_ssi_remove(struct platform_device *pdev) if (!ssi_private->new_binding) platform_device_unregister(ssi_private->pdev); - if (ssi_private->ssi_on_imx) { + if (ssi_private->ssi_on_imx) imx_pcm_dma_exit(pdev); - clk_disable_unprepare(ssi_private->clk); - } snd_soc_unregister_component(&pdev->dev); - device_remove_file(&pdev->dev, &ssi_private->dev_attr); - irq_dispose_mapping(ssi_private->irq); dev_set_drvdata(&pdev->dev, NULL); + device_remove_file(&pdev->dev, &ssi_private->dev_attr); + if (ssi_private->ssi_on_imx) + clk_disable_unprepare(ssi_private->clk); + irq_dispose_mapping(ssi_private->irq); return 0; } -- 1.8.1.2

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[alsa-devel] [PATCH] ASoC: samsung: Rename DMA platform registration functions
by Mark Brown 19 Aug '13

19 Aug '13

From: Mark Brown <broonie(a)linaro.org> The current naming with a simple asoc_ prefix is too generic for use in multiplatform kernels. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/samsung/ac97.c | 4 ++-- sound/soc/samsung/dma.c | 8 ++++---- sound/soc/samsung/dma.h | 4 ++-- sound/soc/samsung/i2s.c | 6 +++--- 4 files changed, 11 insertions(+), 11 deletions(-) diff --git a/sound/soc/samsung/ac97.c b/sound/soc/samsung/ac97.c index c732df9..2acf987 100644 --- a/sound/soc/samsung/ac97.c +++ b/sound/soc/samsung/ac97.c @@ -457,7 +457,7 @@ static int s3c_ac97_probe(struct platform_device *pdev) if (ret) goto err5; - ret = asoc_dma_platform_register(&pdev->dev); + ret = samsung_asoc_dma_platform_register(&pdev->dev); if (ret) { dev_err(&pdev->dev, "failed to get register DMA: %d\n", ret); goto err6; @@ -480,7 +480,7 @@ static int s3c_ac97_remove(struct platform_device *pdev) { struct resource *irq_res; - asoc_dma_platform_unregister(&pdev->dev); + samsung_asoc_dma_platform_unregister(&pdev->dev); snd_soc_unregister_component(&pdev->dev); irq_res = platform_get_resource(pdev, IORESOURCE_IRQ, 0); diff --git a/sound/soc/samsung/dma.c b/sound/soc/samsung/dma.c index d54df85..9338d11 100644 --- a/sound/soc/samsung/dma.c +++ b/sound/soc/samsung/dma.c @@ -444,17 +444,17 @@ static struct snd_soc_platform_driver samsung_asoc_platform = { .pcm_free = dma_free_dma_buffers, }; -int asoc_dma_platform_register(struct device *dev) +int samsung_asoc_dma_platform_register(struct device *dev) { return snd_soc_register_platform(dev, &samsung_asoc_platform); } -EXPORT_SYMBOL_GPL(asoc_dma_platform_register); +EXPORT_SYMBOL_GPL(samsung_asoc_dma_platform_register); -void asoc_dma_platform_unregister(struct device *dev) +void samsung_asoc_dma_platform_unregister(struct device *dev) { snd_soc_unregister_platform(dev); } -EXPORT_SYMBOL_GPL(asoc_dma_platform_unregister); +EXPORT_SYMBOL_GPL(samsung_asoc_dma_platform_unregister); MODULE_AUTHOR("Ben Dooks, <ben(a)simtec.co.uk>"); MODULE_DESCRIPTION("Samsung ASoC DMA Driver"); diff --git a/sound/soc/samsung/dma.h b/sound/soc/samsung/dma.h index 189a7a6..0e86315 100644 --- a/sound/soc/samsung/dma.h +++ b/sound/soc/samsung/dma.h @@ -22,7 +22,7 @@ struct s3c_dma_params { char *ch_name; }; -int asoc_dma_platform_register(struct device *dev); -void asoc_dma_platform_unregister(struct device *dev); +int samsung_asoc_dma_platform_register(struct device *dev); +void samsung_asoc_dma_platform_unregister(struct device *dev); #endif diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index ce487c2..b302f3b 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -1146,7 +1146,7 @@ static int samsung_i2s_probe(struct platform_device *pdev) snd_soc_register_component(&sec_dai->pdev->dev, &samsung_i2s_component, &sec_dai->i2s_dai_drv, 1); - asoc_dma_platform_register(&pdev->dev); + samsung_asoc_dma_platform_register(&pdev->dev); return 0; } @@ -1263,7 +1263,7 @@ static int samsung_i2s_probe(struct platform_device *pdev) pm_runtime_enable(&pdev->dev); - asoc_dma_platform_register(&pdev->dev); + samsung_asoc_dma_platform_register(&pdev->dev); return 0; err: @@ -1293,7 +1293,7 @@ static int samsung_i2s_remove(struct platform_device *pdev) i2s->pri_dai = NULL; i2s->sec_dai = NULL; - asoc_dma_platform_unregister(&pdev->dev); + samsung_asoc_dma_platform_unregister(&pdev->dev); snd_soc_unregister_component(&pdev->dev); return 0; -- 1.8.4.rc3

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[alsa-devel] [PATCH] ASoC: samsung: Check to see if we managed to allocate a channel
by Mark Brown 19 Aug '13

19 Aug '13

From: Mark Brown <broonie(a)linaro.org> Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/samsung/dma.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/samsung/dma.c b/sound/soc/samsung/dma.c index 6e2b2b4..d54df85 100644 --- a/sound/soc/samsung/dma.c +++ b/sound/soc/samsung/dma.c @@ -183,6 +183,10 @@ static int dma_hw_params(struct snd_pcm_substream *substream, prtd->params->ch = prtd->params->ops->request( prtd->params->channel, &req, rtd->cpu_dai->dev, prtd->params->ch_name); + if (!prtd->params->ch) { + pr_err("Failed to allocate DMA channel\n"); + return -ENXIO; + } prtd->params->ops->config(prtd->params->ch, &config); } -- 1.8.4.rc3

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[alsa-devel] [PATCH RFC 0/9] ALSA: usb-audio: fix playback/capture concurrent usage
by Eldad Zack 19 Aug '13

19 Aug '13

Hi, This patch series attempts to fix a long starting problem with the concurrent usage of playback and capture on implicit feedback devices. As Clemens found out, when implicit feedback is used, playback and capture cannot be used in the same time - unless started at the same time (e.g., with jack). While a substream is active, attempting to start a substream in the opposite direction will fail and also (on some devices?) cause the currently active stream to stop. There are multiple issues that contribute to this: 1. URBs are unconditionally deactivated in hw_free. On implicit feedback devices, the sync_endpoint (= data_endpoint of the capture) will be deactivated here. Fixed in patch #4. 2. Changing the alternate settings of an interface might break currently running streams. This happens on my device, but I suspect other devices too. Fixed in patch #7. 3. The endpoint use_count has a subtle issue (see patch #9). This causes jack to work fine, while using two separate programs (most probably) fails. In short: set_format is called from hw_params, but use_count is incremented only later (prepare or trigger). 4. Concurrent usage should be allowed but only if the hw_params match up (patch #8) One problem with this series is that I query the device for the altsettting. This might not be reliable for all devices, so do I plan to remove this. Of course, I've only tested this on my device, but it seems to work good and I couldn't break it so far, so hopefully I got this right. Can someone give it a try on other devices (also on non-implicit-feedback)? Cheers, Eldad Eldad Zack (9): ALSA: usb-audio: remove unused parameter from sync_ep_set_params ALSA: usb-audio: remove deactivate_endpoints() ALSA: usb-audio: prevent NULL dereference on stop trigger ALSA: usb-audio: don't deactivate URBs on in-use EP ALSA: usb-audio: void return type of snd_usb_endpoint_deactivate() ALSA: usb-audio: clear SUBSTREAM_FLAG_SYNC_EP_STARTED on error ALSA: usb-audio: conditional interface altsetting ALSA: usb-audio: conditional concurrent usage of endpoint ALSA: usb-audio: correct ep use_count semantics (add set_param flag) sound/usb/card.h | 1 + sound/usb/endpoint.c | 65 +++++++++++++++------ sound/usb/endpoint.h | 11 +++- sound/usb/pcm.c | 155 +++++++++++++++++++++++++++++++++++++++------------ 4 files changed, 176 insertions(+), 56 deletions(-) -- 1.8.1.5

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[alsa-devel] [PATCH 0/3] HDSPM: Fix LTC ioctl
by Adrian Knoth 19 Aug '13

19 Aug '13

Hi! This series fixes the broken LTC ioctl to query the TCO. Apparently, nobody ever attempted to use the code, otherwise, this rather obvious problem would have been spotted way earlier. This is maybe because not too many people happen to have a TV studio around to make use of this feature. ;) I'll also send a patch to fix alsa-lib. Cheers Adrian Knoth (3): ALSA: hdspm - Fix default value in SNDRV_HDSPM_IOCTL_GET_LTC ALSA: hdspm - Use enums in hdspm_tco_ltc_frames() ALSA: hdspm - Fix SNDRV_HDSPM_IOCTL_GET_LTC include/uapi/sound/hdspm.h | 2 +- sound/pci/rme9652/hdspm.c | 12 ++++++------ 2 files changed, 7 insertions(+), 7 deletions(-) -- 1.8.3.2

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