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-rw-r--r--sound/soc/cirrus/ep93xx-i2s.c2
-rw-r--r--sound/soc/codecs/dmic.c17
-rw-r--r--sound/soc/codecs/rt5640.c217
-rw-r--r--sound/soc/codecs/rt5640.h12
-rw-r--r--sound/soc/codecs/ssm2602.c3
-rw-r--r--sound/soc/codecs/tlv320aic32x4.c22
-rw-r--r--sound/soc/codecs/wm8904.c1
-rw-r--r--sound/soc/codecs/wm8962.c2
-rw-r--r--sound/soc/dwc/designware_i2s.c5
-rw-r--r--sound/soc/fsl/Kconfig11
-rw-r--r--sound/soc/fsl/Makefile2
-rw-r--r--sound/soc/fsl/fsl_spdif.c29
-rw-r--r--sound/soc/fsl/fsl_ssi.c1
-rw-r--r--sound/soc/fsl/imx-audmux.c3
-rw-r--r--sound/soc/fsl/imx-spdif.c148
-rw-r--r--sound/soc/generic/simple-card.c2
-rw-r--r--sound/soc/kirkwood/Kconfig4
-rw-r--r--sound/soc/kirkwood/kirkwood-i2s.c26
-rw-r--r--sound/soc/mxs/mxs-sgtl5000.c2
-rw-r--r--sound/soc/omap/mcbsp.c2
-rw-r--r--sound/soc/samsung/dma.c7
-rw-r--r--sound/soc/sh/fsi.c51
-rw-r--r--sound/soc/soc-core.c17
-rw-r--r--sound/soc/soc-dapm.c11
-rw-r--r--sound/soc/soc-jack.c2
-rw-r--r--sound/soc/soc-pcm.c10
26 files changed, 452 insertions, 157 deletions
diff --git a/sound/soc/cirrus/ep93xx-i2s.c b/sound/soc/cirrus/ep93xx-i2s.c
index f23f331e9a97..a57643d6402f 100644
--- a/sound/soc/cirrus/ep93xx-i2s.c
+++ b/sound/soc/cirrus/ep93xx-i2s.c
@@ -408,7 +408,6 @@ static int ep93xx_i2s_probe(struct platform_device *pdev)
return 0;
fail_put_lrclk:
- dev_set_drvdata(&pdev->dev, NULL);
clk_put(info->lrclk);
fail_put_sclk:
clk_put(info->sclk);
@@ -423,7 +422,6 @@ static int ep93xx_i2s_remove(struct platform_device *pdev)
struct ep93xx_i2s_info *info = dev_get_drvdata(&pdev->dev);
snd_soc_unregister_component(&pdev->dev);
- dev_set_drvdata(&pdev->dev, NULL);
clk_put(info->lrclk);
clk_put(info->sclk);
clk_put(info->mclk);
diff --git a/sound/soc/codecs/dmic.c b/sound/soc/codecs/dmic.c
index 66967ba6f757..b2090b2a5e2d 100644
--- a/sound/soc/codecs/dmic.c
+++ b/sound/soc/codecs/dmic.c
@@ -50,20 +50,11 @@ static const struct snd_soc_dapm_route intercon[] = {
{"DMIC AIF", NULL, "DMic"},
};
-static int dmic_probe(struct snd_soc_codec *codec)
-{
- struct snd_soc_dapm_context *dapm = &codec->dapm;
-
- snd_soc_dapm_new_controls(dapm, dmic_dapm_widgets,
- ARRAY_SIZE(dmic_dapm_widgets));
- snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon));
- snd_soc_dapm_new_widgets(dapm);
-
- return 0;
-}
-
static struct snd_soc_codec_driver soc_dmic = {
- .probe = dmic_probe,
+ .dapm_widgets = dmic_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(dmic_dapm_widgets),
+ .dapm_routes = intercon,
+ .num_dapm_routes = ARRAY_SIZE(intercon),
};
static int dmic_dev_probe(struct platform_device *pdev)
diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c
index 4db7314baabc..c26a8f814b18 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 +926,117 @@ 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(10000, 15000);
+ 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;
+ struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec);
+
+ switch (event) {
+ case SND_SOC_DAPM_POST_PMU:
+ rt5640_pmu_depop(codec);
+ rt5640->hp_mute = 0;
+ break;
+
+ case SND_SOC_DAPM_PRE_PMD:
+ rt5640->hp_mute = 1;
+ 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 int rt5640_hp_post_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:
+ if (!rt5640->hp_mute)
+ usleep_range(80000, 85000);
+
+ 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 +1226,28 @@ 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),
+ SND_SOC_DAPM_POST("HP Post", rt5640_hp_post_event),
/* Output Lines */
SND_SOC_DAPM_OUTPUT("SPOLP"),
SND_SOC_DAPM_OUTPUT("SPOLN"),
@@ -1381,9 +1488,11 @@ 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 +1505,15 @@ 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 +1523,10 @@ 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 +1905,13 @@ 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(10000, 15000);
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 +1955,6 @@ 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);
@@ -2069,6 +2176,8 @@ static int rt5640_i2c_probe(struct i2c_client *i2c,
regmap_update_bits(rt5640->regmap, RT5640_IN3_IN4,
RT5640_IN_DF2, RT5640_IN_DF2);
+ rt5640->hp_mute = 1;
+
ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5640,
rt5640_dai, ARRAY_SIZE(rt5640_dai));
if (ret < 0)
diff --git a/sound/soc/codecs/rt5640.h b/sound/soc/codecs/rt5640.h
index c48286d7118f..5e8df25a13f3 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)
@@ -2087,6 +2098,7 @@ struct rt5640_priv {
int pll_out;
int dmic_en;
+ bool hp_mute;
};
#endif
diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c
index f8d30e5f6371..492644e67ace 100644
--- a/sound/soc/codecs/ssm2602.c
+++ b/sound/soc/codecs/ssm2602.c
@@ -561,8 +561,9 @@ static int ssm2602_suspend(struct snd_soc_codec *codec)
static int ssm2602_resume(struct snd_soc_codec *codec)
{
- snd_soc_cache_sync(codec);
+ struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec);
+ regcache_sync(ssm2602->regmap);
ssm2602_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
return 0;
diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c
index 17df4e32feac..2ed57d4aa445 100644
--- a/sound/soc/codecs/tlv320aic32x4.c
+++ b/sound/soc/codecs/tlv320aic32x4.c
@@ -338,18 +338,6 @@ static inline int aic32x4_get_divs(int mclk, int rate)
return -EINVAL;
}
-static int aic32x4_add_widgets(struct snd_soc_codec *codec)
-{
- snd_soc_dapm_new_controls(&codec->dapm, aic32x4_dapm_widgets,
- ARRAY_SIZE(aic32x4_dapm_widgets));
-
- snd_soc_dapm_add_routes(&codec->dapm, aic32x4_dapm_routes,
- ARRAY_SIZE(aic32x4_dapm_routes));
-
- snd_soc_dapm_new_widgets(&codec->dapm);
- return 0;
-}
-
static int aic32x4_set_dai_sysclk(struct snd_soc_dai *codec_dai,
int clk_id, unsigned int freq, int dir)
{
@@ -683,9 +671,6 @@ static int aic32x4_probe(struct snd_soc_codec *codec)
}
aic32x4_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
- snd_soc_add_codec_controls(codec, aic32x4_snd_controls,
- ARRAY_SIZE(aic32x4_snd_controls));
- aic32x4_add_widgets(codec);
/*
* Workaround: for an unknown reason, the ADC needs to be powered up
@@ -714,6 +699,13 @@ static struct snd_soc_codec_driver soc_codec_dev_aic32x4 = {
.suspend = aic32x4_suspend,
.resume = aic32x4_resume,
.set_bias_level = aic32x4_set_bias_level,
+
+ .controls = aic32x4_snd_controls,
+ .num_controls = ARRAY_SIZE(aic32x4_snd_controls),
+ .dapm_widgets = aic32x4_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(aic32x4_dapm_widgets),
+ .dapm_routes = aic32x4_dapm_routes,
+ .num_dapm_routes = ARRAY_SIZE(aic32x4_dapm_routes),
};
static int aic32x4_i2c_probe(struct i2c_client *i2c,
diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c
index 91dfbfeda6f8..4dfa8dceeabf 100644
--- a/sound/soc/codecs/wm8904.c
+++ b/sound/soc/codecs/wm8904.c
@@ -1202,7 +1202,6 @@ static int wm8904_add_widgets(struct snd_soc_codec *codec)
break;
}
- snd_soc_dapm_new_widgets(dapm);
return 0;
}
diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c
index 36782f067cc5..11d80f3b6137 100644
--- a/sound/soc/codecs/wm8962.c
+++ b/sound/soc/codecs/wm8962.c
@@ -3174,7 +3174,7 @@ static ssize_t wm8962_beep_set(struct device *dev,
long int time;
int ret;
- ret = strict_strtol(buf, 10, &time);
+ ret = kstrtol(buf, 10, &time);
if (ret != 0)
return ret;
diff --git a/sound/soc/dwc/designware_i2s.c b/sound/soc/dwc/designware_i2s.c
index 70eb37a5dd16..25c31f1655f6 100644
--- a/sound/soc/dwc/designware_i2s.c
+++ b/sound/soc/dwc/designware_i2s.c
@@ -421,13 +421,11 @@ static int dw_i2s_probe(struct platform_device *pdev)
dw_i2s_dai, 1);
if (ret != 0) {
dev_err(&pdev->dev, "not able to register dai\n");
- goto err_set_drvdata;
+ goto err_clk_disable;
}
return 0;
-err_set_drvdata:
- dev_set_drvdata(&pdev->dev, NULL);
err_clk_disable:
clk_disable(dev->clk);
err_clk_put:
@@ -440,7 +438,6 @@ static int dw_i2s_remove(struct platform_device *pdev)
struct dw_i2s_dev *dev = dev_get_drvdata(&pdev->dev);
snd_soc_unregister_component(&pdev->dev);
- dev_set_drvdata(&pdev->dev, NULL);
clk_put(dev->clk);
diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig
index cd088cc8c866..b7ab71f2ccc1 100644
--- a/sound/soc/fsl/Kconfig
+++ b/sound/soc/fsl/Kconfig
@@ -193,6 +193,17 @@ config SND_SOC_IMX_SGTL5000
Say Y if you want to add support for SoC audio on an i.MX board with
a sgtl5000 codec.
+config SND_SOC_IMX_SPDIF
+ tristate "SoC Audio support for i.MX boards with S/PDIF"
+ select SND_SOC_IMX_PCM_DMA
+ select SND_SOC_FSL_SPDIF
+ select SND_SOC_SPDIF
+ select REGMAP_MMIO
+ help
+ SoC Audio support for i.MX boards with S/PDIF
+ Say Y if you want to add support for SoC audio on an i.MX board with
+ a S/DPDIF.
+
config SND_SOC_IMX_MC13783
tristate "SoC Audio support for I.MX boards with mc13783"
depends on MFD_MC13783 && ARM
diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile
index 4b5970e014dd..8db705b0fdf9 100644
--- a/sound/soc/fsl/Makefile
+++ b/sound/soc/fsl/Makefile
@@ -45,6 +45,7 @@ snd-soc-mx27vis-aic32x4-objs := mx27vis-aic32x4.o
snd-soc-wm1133-ev1-objs := wm1133-ev1.o
snd-soc-imx-sgtl5000-objs := imx-sgtl5000.o
snd-soc-imx-wm8962-objs := imx-wm8962.o
+snd-soc-imx-spdif-objs := imx-spdif.o
snd-soc-imx-mc13783-objs := imx-mc13783.o
obj-$(CONFIG_SND_SOC_EUKREA_TLV320) += snd-soc-eukrea-tlv320.o
@@ -53,4 +54,5 @@ obj-$(CONFIG_SND_SOC_MX27VIS_AIC32X4) += snd-soc-mx27vis-aic32x4.o
obj-$(CONFIG_SND_MXC_SOC_WM1133_EV1) += snd-soc-wm1133-ev1.o
obj-$(CONFIG_SND_SOC_IMX_SGTL5000) += snd-soc-imx-sgtl5000.o
obj-$(CONFIG_SND_SOC_IMX_WM8962) += snd-soc-imx-wm8962.o
+obj-$(CONFIG_SND_SOC_IMX_SPDIF) += snd-soc-imx-spdif.o
obj-$(CONFIG_SND_SOC_IMX_MC13783) += snd-soc-imx-mc13783.o
diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c
index 42a43820d993..3920c3e849ce 100644
--- a/sound/soc/fsl/fsl_spdif.c
+++ b/sound/soc/fsl/fsl_spdif.c
@@ -411,8 +411,8 @@ static int spdif_set_sample_rate(struct snd_pcm_substream *substream,
return 0;
}
-int fsl_spdif_startup(struct snd_pcm_substream *substream,
- struct snd_soc_dai *cpu_dai)
+static int fsl_spdif_startup(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *cpu_dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(rtd->cpu_dai);
@@ -546,7 +546,7 @@ static int fsl_spdif_trigger(struct snd_pcm_substream *substream,
return 0;
}
-struct snd_soc_dai_ops fsl_spdif_dai_ops = {
+static struct snd_soc_dai_ops fsl_spdif_dai_ops = {
.startup = fsl_spdif_startup,
.hw_params = fsl_spdif_hw_params,
.trigger = fsl_spdif_trigger,
@@ -555,7 +555,6 @@ struct snd_soc_dai_ops fsl_spdif_dai_ops = {
/*
- * ============================================
* FSL SPDIF IEC958 controller(mixer) functions
*
* Channel status get/put control
@@ -563,7 +562,6 @@ struct snd_soc_dai_ops fsl_spdif_dai_ops = {
* Valid bit value get control
* DPLL lock status get control
* User bit sync mode selection control
- * ============================================
*/
static int fsl_spdif_info(struct snd_kcontrol *kcontrol,
@@ -921,7 +919,7 @@ static int fsl_spdif_dai_probe(struct snd_soc_dai *dai)
return 0;
}
-struct snd_soc_dai_driver fsl_spdif_dai = {
+static struct snd_soc_dai_driver fsl_spdif_dai = {
.probe = &fsl_spdif_dai_probe,
.playback = {
.channels_min = 2,
@@ -942,11 +940,7 @@ static const struct snd_soc_component_driver fsl_spdif_component = {
.name = "fsl-spdif",
};
-/*
- * ================
- * FSL SPDIF REGMAP
- * ================
- */
+/* FSL SPDIF REGMAP */
static bool fsl_spdif_readable_reg(struct device *dev, unsigned int reg)
{
@@ -1077,9 +1071,9 @@ static int fsl_spdif_probe_txclk(struct fsl_spdif_priv *spdif_priv,
break;
}
- dev_dbg(&pdev->dev, "use rxtx%d as tx clock source for %dHz sample rate",
+ dev_dbg(&pdev->dev, "use rxtx%d as tx clock source for %dHz sample rate\n",
spdif_priv->txclk_src[index], rate[index]);
- dev_dbg(&pdev->dev, "use divisor %d for %dHz sample rate",
+ dev_dbg(&pdev->dev, "use divisor %d for %dHz sample rate\n",
spdif_priv->txclk_div[index], rate[index]);
return 0;
@@ -1119,10 +1113,8 @@ static int fsl_spdif_probe(struct platform_device *pdev)
}
regs = devm_ioremap_resource(&pdev->dev, res);
- if (IS_ERR(regs)) {
- dev_err(&pdev->dev, "could not map device resources\n");
+ if (IS_ERR(regs))
return PTR_ERR(regs);
- }
spdif_priv->regmap = devm_regmap_init_mmio_clk(&pdev->dev,
"core", regs, &fsl_spdif_regmap_config);
@@ -1184,7 +1176,7 @@ static int fsl_spdif_probe(struct platform_device *pdev)
&spdif_priv->cpu_dai_drv, 1);
if (ret) {
dev_err(&pdev->dev, "failed to register DAI: %d\n", ret);
- goto error_dev;
+ return ret;
}
ret = imx_pcm_dma_init(pdev);
@@ -1197,8 +1189,6 @@ static int fsl_spdif_probe(struct platform_device *pdev)
error_component:
snd_soc_unregister_component(&pdev->dev);
-error_dev:
- dev_set_drvdata(&pdev->dev, NULL);
return ret;
}
@@ -1207,7 +1197,6 @@ static int fsl_spdif_remove(struct platform_device *pdev)
{
imx_pcm_dma_exit(pdev);
snd_soc_unregister_component(&pdev->dev);
- dev_set_drvdata(&pdev->dev, NULL);
return 0;
}
diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index 5cf626c4dc96..c6b743978d5e 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -1114,7 +1114,6 @@ error_dai:
snd_soc_unregister_component(&pdev->dev);
error_dev:
- dev_set_drvdata(&pdev->dev, NULL);
device_remove_file(&pdev->dev, dev_attr);
error_clk:
diff --git a/sound/soc/fsl/imx-audmux.c b/sound/soc/fsl/imx-audmux.c
index ab17381cc981..d3bf71a0ec56 100644
--- a/sound/soc/fsl/imx-audmux.c
+++ b/sound/soc/fsl/imx-audmux.c
@@ -335,7 +335,8 @@ static int imx_audmux_probe(struct platform_device *pdev)
if (audmux_type == IMX31_AUDMUX)
audmux_debugfs_init();
- imx_audmux_parse_dt_defaults(pdev, pdev->dev.of_node);
+ if (of_id)
+ imx_audmux_parse_dt_defaults(pdev, pdev->dev.of_node);
return 0;
}
diff --git a/sound/soc/fsl/imx-spdif.c b/sound/soc/fsl/imx-spdif.c
new file mode 100644
index 000000000000..816013b0ebba
--- /dev/null
+++ b/sound/soc/fsl/imx-spdif.c
@@ -0,0 +1,148 @@
+/*
+ * Copyright (C) 2013 Freescale Semiconductor, Inc.
+ *
+ * The code contained herein is licensed under the GNU General Public
+ * License. You may obtain a copy of the GNU General Public License
+ * Version 2 or later at the following locations:
+ *
+ * http://www.opensource.org/licenses/gpl-license.html
+ * http://www.gnu.org/copyleft/gpl.html
+ */
+
+#include <linux/module.h>
+#include <linux/of_platform.h>
+#include <sound/soc.h>
+
+struct imx_spdif_data {
+ struct snd_soc_dai_link dai[2];
+ struct snd_soc_card card;
+ struct platform_device *txdev;
+ struct platform_device *rxdev;
+};
+
+static int imx_spdif_audio_probe(struct platform_device *pdev)
+{
+ struct device_node *spdif_np, *np = pdev->dev.of_node;
+ struct imx_spdif_data *data;
+ int ret = 0, num_links = 0;
+
+ spdif_np = of_parse_phandle(np, "spdif-controller", 0);
+ if (!spdif_np) {
+ dev_err(&pdev->dev, "failed to find spdif-controller\n");
+ ret = -EINVAL;
+ goto end;
+ }
+
+ data = devm_kzalloc(&pdev->dev, sizeof(*data), GFP_KERNEL);
+ if (!data) {
+ dev_err(&pdev->dev, "failed to allocate memory\n");
+ ret = -ENOMEM;
+ goto end;
+ }
+
+ if (of_property_read_bool(np, "spdif-out")) {
+ data->dai[num_links].name = "S/PDIF TX";
+ data->dai[num_links].stream_name = "S/PDIF PCM Playback";
+ data->dai[num_links].codec_dai_name = "dit-hifi";
+ data->dai[num_links].codec_name = "spdif-dit";
+ data->dai[num_links].cpu_of_node = spdif_np;
+ data->dai[num_links].platform_of_node = spdif_np;
+ num_links++;
+
+ data->txdev = platform_device_register_simple("spdif-dit", -1, NULL, 0);
+ if (IS_ERR(data->txdev)) {
+ ret = PTR_ERR(data->txdev);
+ dev_err(&pdev->dev, "register dit failed: %d\n", ret);
+ goto end;
+ }
+ }
+
+ if (of_property_read_bool(np, "spdif-in")) {
+ data->dai[num_links].name = "S/PDIF RX";
+ data->dai[num_links].stream_name = "S/PDIF PCM Capture";
+ data->dai[num_links].codec_dai_name = "dir-hifi";
+ data->dai[num_links].codec_name = "spdif-dir";
+ data->dai[num_links].cpu_of_node = spdif_np;
+ data->dai[num_links].platform_of_node = spdif_np;
+ num_links++;
+
+ data->rxdev = platform_device_register_simple("spdif-dir", -1, NULL, 0);
+ if (IS_ERR(data->rxdev)) {
+ ret = PTR_ERR(data->rxdev);
+ dev_err(&pdev->dev, "register dir failed: %d\n", ret);
+ goto error_dit;
+ }
+ }
+
+ if (!num_links) {
+ dev_err(&pdev->dev, "no enabled S/PDIF DAI link\n");
+ goto error_dir;
+ }
+
+ data->card.dev = &pdev->dev;
+ data->card.num_links = num_links;
+ data->card.dai_link = data->dai;
+
+ ret = snd_soc_of_parse_card_name(&data->card, "model");
+ if (ret)
+ goto error_dir;
+
+ ret = snd_soc_register_card(&data->card);
+ if (ret) {
+ dev_err(&pdev->dev, "snd_soc_register_card failed: %d\n", ret);
+ goto error_dir;
+ }
+
+ platform_set_drvdata(pdev, data);
+
+ goto end;
+
+error_dir:
+ if (data->rxdev)
+ platform_device_unregister(data->rxdev);
+error_dit:
+ if (data->txdev)
+ platform_device_unregister(data->txdev);
+end:
+ if (spdif_np)
+ of_node_put(spdif_np);
+
+ return ret;
+}
+
+static int imx_spdif_audio_remove(struct platform_device *pdev)
+{
+ struct imx_spdif_data *data = platform_get_drvdata(pdev);
+
+ if (data->rxdev)
+ platform_device_unregister(data->rxdev);
+ if (data->txdev)
+ platform_device_unregister(data->txdev);
+
+ snd_soc_unregister_card(&data->card);
+
+ return 0;
+}
+
+static const struct of_device_id imx_spdif_dt_ids[] = {
+ { .compatible = "fsl,imx-audio-spdif", },
+ { /* sentinel */ }
+};
+MODULE_DEVICE_TABLE(of, imx_spdif_dt_ids);
+
+static struct platform_driver imx_spdif_driver = {
+ .driver = {
+ .name = "imx-spdif",
+ .owner = THIS_MODULE,
+ .of_match_table = imx_spdif_dt_ids,
+ },
+ .probe = imx_spdif_audio_probe,
+ .remove = imx_spdif_audio_remove,
+};
+
+module_platform_driver(imx_spdif_driver);
+
+MODULE_AUTHOR("Freescale Semiconductor, Inc.");
+MODULE_DESCRIPTION("Freescale i.MX S/PDIF machine driver");
+MODULE_LICENSE("GPL v2");
+MODULE_ALIAS("platform:imx-spdif");
diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c
index 6cf8355a8542..8c49147db84c 100644
--- a/sound/soc/generic/simple-card.c
+++ b/sound/soc/generic/simple-card.c
@@ -105,6 +105,7 @@ static int asoc_simple_card_remove(struct platform_device *pdev)
static struct platform_driver asoc_simple_card = {
.driver = {
.name = "asoc-simple-card",
+ .owner = THIS_MODULE,
},
.probe = asoc_simple_card_probe,
.remove = asoc_simple_card_remove,
@@ -112,6 +113,7 @@ static struct platform_driver asoc_simple_card = {
module_platform_driver(asoc_simple_card);
+MODULE_ALIAS("platform:asoc-simple-card");
MODULE_LICENSE("GPL");
MODULE_DESCRIPTION("ASoC Simple Sound Card");
MODULE_AUTHOR("Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>");
diff --git a/sound/soc/kirkwood/Kconfig b/sound/soc/kirkwood/Kconfig
index 9e1970c44e86..78ed4a42ad21 100644
--- a/sound/soc/kirkwood/Kconfig
+++ b/sound/soc/kirkwood/Kconfig
@@ -1,6 +1,6 @@
config SND_KIRKWOOD_SOC
- tristate "SoC Audio for the Marvell Kirkwood chip"
- depends on ARCH_KIRKWOOD || COMPILE_TEST
+ tristate "SoC Audio for the Marvell Kirkwood and Dove chips"
+ depends on ARCH_KIRKWOOD || ARCH_DOVE || COMPILE_TEST
help
Say Y or M if you want to add support for codecs attached to
the Kirkwood I2S interface. You will also need to select the
diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c
index e5f3f7a9ea26..7fce340ab3ef 100644
--- a/sound/soc/kirkwood/kirkwood-i2s.c
+++ b/sound/soc/kirkwood/kirkwood-i2s.c
@@ -22,6 +22,8 @@
#include <sound/pcm_params.h>
#include <sound/soc.h>
#include <linux/platform_data/asoc-kirkwood.h>
+#include <linux/of.h>
+
#include "kirkwood.h"
#define DRV_NAME "mvebu-audio"
@@ -453,6 +455,7 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev)
struct snd_soc_dai_driver *soc_dai = &kirkwood_i2s_dai;
struct kirkwood_dma_data *priv;
struct resource *mem;
+ struct device_node *np = pdev->dev.of_node;
int err;
priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL);
@@ -473,14 +476,16 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev)
return -ENXIO;
}
- if (!data) {
- dev_err(&pdev->dev, "no platform data ?!\n");
+ if (np) {
+ priv->burst = 128; /* might be 32 or 128 */
+ } else if (data) {
+ priv->burst = data->burst;
+ } else {
+ dev_err(&pdev->dev, "no DT nor platform data ?!\n");
return -EINVAL;
}
- priv->burst = data->burst;
-
- priv->clk = devm_clk_get(&pdev->dev, NULL);
+ priv->clk = devm_clk_get(&pdev->dev, np ? "internal" : NULL);
if (IS_ERR(priv->clk)) {
dev_err(&pdev->dev, "no clock\n");
return PTR_ERR(priv->clk);
@@ -507,7 +512,7 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev)
priv->ctl_rec = KIRKWOOD_RECCTL_SIZE_24;
/* Select the burst size */
- if (data->burst == 32) {
+ if (priv->burst == 32) {
priv->ctl_play |= KIRKWOOD_PLAYCTL_BURST_32;
priv->ctl_rec |= KIRKWOOD_RECCTL_BURST_32;
} else {
@@ -552,12 +557,21 @@ static int kirkwood_i2s_dev_remove(struct platform_device *pdev)
return 0;
}
+#ifdef CONFIG_OF
+static struct of_device_id mvebu_audio_of_match[] = {
+ { .compatible = "marvell,mvebu-audio" },
+ { }
+};
+MODULE_DEVICE_TABLE(of, mvebu_audio_of_match);
+#endif
+
static struct platform_driver kirkwood_i2s_driver = {
.probe = kirkwood_i2s_dev_probe,
.remove = kirkwood_i2s_dev_remove,
.driver = {
.name = DRV_NAME,
.owner = THIS_MODULE,
+ .of_match_table = of_match_ptr(mvebu_audio_of_match),
},
};
diff --git a/sound/soc/mxs/mxs-sgtl5000.c b/sound/soc/mxs/mxs-sgtl5000.c
index ce084eb10c49..4bb273786ff3 100644
--- a/sound/soc/mxs/mxs-sgtl5000.c
+++ b/sound/soc/mxs/mxs-sgtl5000.c
@@ -105,11 +105,13 @@ static struct snd_soc_dai_link mxs_sgtl5000_dai[] = {
.stream_name = "HiFi Playback",
.codec_dai_name = "sgtl5000",
.ops = &mxs_sgtl5000_hifi_ops,
+ .playback_only = true,
}, {
.name = "HiFi Rx",
.stream_name = "HiFi Capture",
.codec_dai_name = "sgtl5000",
.ops = &mxs_sgtl5000_hifi_ops,
+ .capture_only = true,
},
};
diff --git a/sound/soc/omap/mcbsp.c b/sound/soc/omap/mcbsp.c
index 361e4c03646e..83433fdea32a 100644
--- a/sound/soc/omap/mcbsp.c
+++ b/sound/soc/omap/mcbsp.c
@@ -781,7 +781,7 @@ static ssize_t prop##_store(struct device *dev, \
unsigned long val; \
int status; \
\
- status = strict_strtoul(buf, 0, &val); \
+ status = kstrtoul(buf, 0, &val); \
if (status) \
return status; \
\
diff --git a/sound/soc/samsung/dma.c b/sound/soc/samsung/dma.c
index a0c67f60f594..9338d11e9216 100644
--- a/sound/soc/samsung/dma.c
+++ b/sound/soc/samsung/dma.c
@@ -90,6 +90,13 @@ static void dma_enqueue(struct snd_pcm_substream *substream)
dma_info.period = prtd->dma_period;
dma_info.len = prtd->dma_period*limit;
+ if (dma_info.cap == DMA_CYCLIC) {
+ dma_info.buf = pos;
+ prtd->params->ops->prepare(prtd->params->ch, &dma_info);
+ prtd->dma_loaded += limit;
+ return;
+ }
+
while (prtd->dma_loaded < limit) {
pr_debug("dma_loaded: %d\n", prtd->dma_loaded);
diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c
index 30390260bb67..b33ca7cd085b 100644
--- a/sound/soc/sh/fsi.c
+++ b/sound/soc/sh/fsi.c
@@ -235,6 +235,8 @@ struct fsi_stream {
struct sh_dmae_slave slave; /* see fsi_handler_init() */
struct work_struct work;
dma_addr_t dma;
+ int loop_cnt;
+ int additional_pos;
};
struct fsi_clk {
@@ -1289,6 +1291,8 @@ static int fsi_dma_init(struct fsi_priv *fsi, struct fsi_stream *io)
io->bus_option = BUSOP_SET(24, PACKAGE_24BITBUS_BACK) |
BUSOP_SET(16, PACKAGE_16BITBUS_STREAM);
+ io->loop_cnt = 2; /* push 1st, 2nd period first, then 3rd, 4th... */
+ io->additional_pos = 0;
io->dma = dma_map_single(dai->dev, runtime->dma_area,
snd_pcm_lib_buffer_bytes(io->substream), dir);
return 0;
@@ -1305,11 +1309,15 @@ static int fsi_dma_quit(struct fsi_priv *fsi, struct fsi_stream *io)
return 0;
}
-static dma_addr_t fsi_dma_get_area(struct fsi_stream *io)
+static dma_addr_t fsi_dma_get_area(struct fsi_stream *io, int additional)
{
struct snd_pcm_runtime *runtime = io->substream->runtime;
+ int period = io->period_pos + additional;
- return io->dma + samples_to_bytes(runtime, io->buff_sample_pos);
+ if (period >= runtime->periods)
+ period = 0;
+
+ return io->dma + samples_to_bytes(runtime, period * io->period_samples);
}
static void fsi_dma_complete(void *data)
@@ -1321,7 +1329,7 @@ static void fsi_dma_complete(void *data)
enum dma_data_direction dir = fsi_stream_is_play(fsi, io) ?
DMA_TO_DEVICE : DMA_FROM_DEVICE;
- dma_sync_single_for_cpu(dai->dev, fsi_dma_get_area(io),
+ dma_sync_single_for_cpu(dai->dev, fsi_dma_get_area(io, 0),
samples_to_bytes(runtime, io->period_samples), dir);
io->buff_sample_pos += io->period_samples;
@@ -1347,7 +1355,7 @@ static void fsi_dma_do_work(struct work_struct *work)
struct snd_pcm_runtime *runtime;
enum dma_data_direction dir;
int is_play = fsi_stream_is_play(fsi, io);
- int len;
+ int len, i;
dma_addr_t buf;
if (!fsi_stream_is_working(fsi, io))
@@ -1357,26 +1365,33 @@ static void fsi_dma_do_work(struct work_struct *work)
runtime = io->substream->runtime;
dir = is_play ? DMA_TO_DEVICE : DMA_FROM_DEVICE;
len = samples_to_bytes(runtime, io->period_samples);
- buf = fsi_dma_get_area(io);
- dma_sync_single_for_device(dai->dev, buf, len, dir);
+ for (i = 0; i < io->loop_cnt; i++) {
+ buf = fsi_dma_get_area(io, io->additional_pos);
- desc = dmaengine_prep_slave_single(io->chan, buf, len, dir,
- DMA_PREP_INTERRUPT | DMA_CTRL_ACK);
- if (!desc) {
- dev_err(dai->dev, "dmaengine_prep_slave_sg() fail\n");
- return;
- }
+ dma_sync_single_for_device(dai->dev, buf, len, dir);
- desc->callback = fsi_dma_complete;
- desc->callback_param = io;
+ desc = dmaengine_prep_slave_single(io->chan, buf, len, dir,
+ DMA_PREP_INTERRUPT | DMA_CTRL_ACK);
+ if (!desc) {
+ dev_err(dai->dev, "dmaengine_prep_slave_sg() fail\n");
+ return;
+ }
- if (dmaengine_submit(desc) < 0) {
- dev_err(dai->dev, "tx_submit() fail\n");
- return;
+ desc->callback = fsi_dma_complete;
+ desc->callback_param = io;
+
+ if (dmaengine_submit(desc) < 0) {
+ dev_err(dai->dev, "tx_submit() fail\n");
+ return;
+ }
+
+ dma_async_issue_pending(io->chan);
+
+ io->additional_pos = 1;
}
- dma_async_issue_pending(io->chan);
+ io->loop_cnt = 1;
/*
* FIXME
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 528f8708221d..4d0561312f3b 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -203,7 +203,7 @@ static ssize_t pmdown_time_set(struct device *dev,
struct snd_soc_pcm_runtime *rtd = dev_get_drvdata(dev);
int ret;
- ret = strict_strtol(buf, 10, &rtd->pmdown_time);
+ ret = kstrtol(buf, 10, &rtd->pmdown_time);
if (ret)
return ret;
@@ -248,6 +248,7 @@ static ssize_t codec_reg_write_file(struct file *file,
char *start = buf;
unsigned long reg, value;
struct snd_soc_codec *codec = file->private_data;
+ int ret;
buf_size = min(count, (sizeof(buf)-1));
if (copy_from_user(buf, user_buf, buf_size))
@@ -259,8 +260,9 @@ static ssize_t codec_reg_write_file(struct file *file,
reg = simple_strtoul(start, &start, 16);
while (*start == ' ')
start++;
- if (strict_strtoul(start, 16, &value))
- return -EINVAL;
+ ret = kstrtoul(start, 16, &value);
+ if (ret)
+ return ret;
/* Userspace has been fiddling around behind the kernel's back */
add_taint(TAINT_USER, LOCKDEP_NOW_UNRELIABLE);
@@ -1243,9 +1245,6 @@ static int soc_post_component_init(struct snd_soc_card *card,
}
rtd->card = card;
- /* Make sure all DAPM widgets are instantiated */
- snd_soc_dapm_new_widgets(&codec->dapm);
-
/* machine controls, routes and widgets are not prefixed */
temp = codec->name_prefix;
codec->name_prefix = NULL;
@@ -1741,8 +1740,6 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card)
snd_soc_dapm_add_routes(&card->dapm, card->dapm_routes,
card->num_dapm_routes);
- snd_soc_dapm_new_widgets(&card->dapm);
-
for (i = 0; i < card->num_links; i++) {
dai_link = &card->dai_link[i];
dai_fmt = dai_link->dai_fmt;
@@ -1821,12 +1818,12 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card)
}
}
- snd_soc_dapm_new_widgets(&card->dapm);
-
if (card->fully_routed)
list_for_each_entry(codec, &card->codec_dev_list, card_list)
snd_soc_dapm_auto_nc_codec_pins(codec);
+ snd_soc_dapm_new_widgets(card);
+
ret = snd_card_register(card->snd_card);
if (ret < 0) {
dev_err(card->dev, "ASoC: failed to register soundcard %d\n",
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index d84bd0f167b6..c17c14c394df 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -229,6 +229,8 @@ static int dapm_kcontrol_data_alloc(struct snd_soc_dapm_widget *widget,
template.id = snd_soc_dapm_kcontrol;
template.name = kcontrol->id.name;
+ data->value = template.on_val;
+
data->widget = snd_soc_dapm_new_control(widget->dapm,
&template);
if (!data->widget) {
@@ -2374,6 +2376,9 @@ static int snd_soc_dapm_add_path(struct snd_soc_dapm_context *dapm,
wsource->ext = 1;
}
+ dapm_mark_dirty(wsource, "Route added");
+ dapm_mark_dirty(wsink, "Route added");
+
/* connect static paths */
if (control == NULL) {
list_add(&path->list, &dapm->card->paths);
@@ -2436,9 +2441,6 @@ static int snd_soc_dapm_add_path(struct snd_soc_dapm_context *dapm,
return 0;
}
- dapm_mark_dirty(wsource, "Route added");
- dapm_mark_dirty(wsink, "Route added");
-
return 0;
err:
kfree(path);
@@ -2712,9 +2714,8 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_weak_routes);
*
* Returns 0 for success.
*/
-int snd_soc_dapm_new_widgets(struct snd_soc_dapm_context *dapm)
+int snd_soc_dapm_new_widgets(struct snd_soc_card *card)
{
- struct snd_soc_card *card = dapm->card;
struct snd_soc_dapm_widget *w;
unsigned int val;
diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c
index 7aa26b5178aa..71358e3b54d9 100644
--- a/sound/soc/soc-jack.c
+++ b/sound/soc/soc-jack.c
@@ -183,8 +183,6 @@ int snd_soc_jack_add_pins(struct snd_soc_jack *jack, int count,
list_add(&(pins[i].list), &jack->pins);
}
- snd_soc_dapm_new_widgets(&jack->codec->card->dapm);
-
/* Update to reflect the last reported status; canned jack
* implementations are likely to set their state before the
* card has an opportunity to associate pins.
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index fb70fbe26862..330c9a6b5cb5 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -2020,6 +2020,16 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num)
capture = 1;
}
+ if (rtd->dai_link->playback_only) {
+ playback = 1;
+ capture = 0;
+ }
+
+ if (rtd->dai_link->capture_only) {
+ playback = 0;
+ capture = 1;
+ }
+
/* create the PCM */
if (rtd->dai_link->no_pcm) {
snprintf(new_name, sizeof(new_name), "(%s)",
generated by cgit v1.2.3-58-ga151 (git 2.39.0) at 2024-12-19 20:20:43 +0000