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-rw-r--r--sound/soc/codecs/arizona.c9
-rw-r--r--sound/soc/codecs/arizona.h18
-rw-r--r--sound/soc/codecs/cs4271.c6
-rw-r--r--sound/soc/codecs/cs42l52.c4
-rw-r--r--sound/soc/codecs/cs42l73.c116
-rw-r--r--sound/soc/codecs/lm49453.c106
-rw-r--r--sound/soc/codecs/sgtl5000.c4
-rw-r--r--sound/soc/codecs/sigmadsp.c2
-rw-r--r--sound/soc/codecs/sta529.c9
-rw-r--r--sound/soc/codecs/tpa6130a2.c23
-rw-r--r--sound/soc/codecs/wm2000.c4
-rw-r--r--sound/soc/codecs/wm2200.c8
-rw-r--r--sound/soc/codecs/wm5100.c6
-rw-r--r--sound/soc/codecs/wm5102.c48
-rw-r--r--sound/soc/codecs/wm_adsp.c23
15 files changed, 229 insertions, 157 deletions
diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c
index adf397b9d0e6..1d8bb5917594 100644
--- a/sound/soc/codecs/arizona.c
+++ b/sound/soc/codecs/arizona.c
@@ -446,15 +446,9 @@ static int arizona_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
case SND_SOC_DAIFMT_DSP_A:
mode = 0;
break;
- case SND_SOC_DAIFMT_DSP_B:
- mode = 1;
- break;
case SND_SOC_DAIFMT_I2S:
mode = 2;
break;
- case SND_SOC_DAIFMT_LEFT_J:
- mode = 3;
- break;
default:
arizona_aif_err(dai, "Unsupported DAI format %d\n",
fmt & SND_SOC_DAIFMT_FORMAT_MASK);
@@ -714,7 +708,8 @@ static int arizona_hw_params(struct snd_pcm_substream *substream,
snd_soc_update_bits(codec, ARIZONA_ASYNC_SAMPLE_RATE_1,
ARIZONA_ASYNC_SAMPLE_RATE_MASK, sr_val);
snd_soc_update_bits(codec, base + ARIZONA_AIF_RATE_CTRL,
- ARIZONA_AIF1_RATE_MASK, 8);
+ ARIZONA_AIF1_RATE_MASK,
+ 8 << ARIZONA_AIF1_RATE_SHIFT);
break;
default:
arizona_aif_err(dai, "Invalid clock %d\n", dai_priv->clk);
diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h
index 41dae1ed3b71..4deebeb07177 100644
--- a/sound/soc/codecs/arizona.h
+++ b/sound/soc/codecs/arizona.h
@@ -34,15 +34,15 @@
#define ARIZONA_FLL_SRC_MCLK1 0
#define ARIZONA_FLL_SRC_MCLK2 1
-#define ARIZONA_FLL_SRC_SLIMCLK 2
-#define ARIZONA_FLL_SRC_FLL1 3
-#define ARIZONA_FLL_SRC_FLL2 4
-#define ARIZONA_FLL_SRC_AIF1BCLK 5
-#define ARIZONA_FLL_SRC_AIF2BCLK 6
-#define ARIZONA_FLL_SRC_AIF3BCLK 7
-#define ARIZONA_FLL_SRC_AIF1LRCLK 8
-#define ARIZONA_FLL_SRC_AIF2LRCLK 9
-#define ARIZONA_FLL_SRC_AIF3LRCLK 10
+#define ARIZONA_FLL_SRC_SLIMCLK 3
+#define ARIZONA_FLL_SRC_FLL1 4
+#define ARIZONA_FLL_SRC_FLL2 5
+#define ARIZONA_FLL_SRC_AIF1BCLK 8
+#define ARIZONA_FLL_SRC_AIF2BCLK 9
+#define ARIZONA_FLL_SRC_AIF3BCLK 10
+#define ARIZONA_FLL_SRC_AIF1LRCLK 12
+#define ARIZONA_FLL_SRC_AIF2LRCLK 13
+#define ARIZONA_FLL_SRC_AIF3LRCLK 14
#define ARIZONA_MIXER_VOL_MASK 0x00FE
#define ARIZONA_MIXER_VOL_SHIFT 1
diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c
index 4f1127935fdf..ac8742a1f25a 100644
--- a/sound/soc/codecs/cs4271.c
+++ b/sound/soc/codecs/cs4271.c
@@ -474,16 +474,16 @@ static int cs4271_probe(struct snd_soc_codec *codec)
struct cs4271_platform_data *cs4271plat = codec->dev->platform_data;
int ret;
int gpio_nreset = -EINVAL;
- int amutec_eq_bmutec = 0;
+ bool amutec_eq_bmutec = false;
#ifdef CONFIG_OF
if (of_match_device(cs4271_dt_ids, codec->dev)) {
gpio_nreset = of_get_named_gpio(codec->dev->of_node,
"reset-gpio", 0);
- if (!of_get_property(codec->dev->of_node,
+ if (of_get_property(codec->dev->of_node,
"cirrus,amutec-eq-bmutec", NULL))
- amutec_eq_bmutec = 1;
+ amutec_eq_bmutec = true;
}
#endif
diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c
index 99bb1c69499e..9811a5478c87 100644
--- a/sound/soc/codecs/cs42l52.c
+++ b/sound/soc/codecs/cs42l52.c
@@ -737,7 +737,7 @@ static const struct cs42l52_clk_para clk_map_table[] = {
static int cs42l52_get_clk(int mclk, int rate)
{
- int i, ret = 0;
+ int i, ret = -EINVAL;
u_int mclk1, mclk2 = 0;
for (i = 0; i < ARRAY_SIZE(clk_map_table); i++) {
@@ -749,8 +749,6 @@ static int cs42l52_get_clk(int mclk, int rate)
}
}
}
- if (ret > ARRAY_SIZE(clk_map_table))
- return -EINVAL;
return ret;
}
diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c
index a0791ecf6d95..6361dab48bd1 100644
--- a/sound/soc/codecs/cs42l73.c
+++ b/sound/soc/codecs/cs42l73.c
@@ -40,6 +40,7 @@ struct cs42l73_private {
u32 sysclk;
u8 mclksel;
u32 mclk;
+ int shutdwn_delay;
};
static const struct reg_default cs42l73_reg_defaults[] = {
@@ -588,7 +589,60 @@ static const struct snd_kcontrol_new cs42l73_snd_controls[] = {
SOC_ENUM("XSPOUT Mono/Stereo Select", xsp_output_mux_enum),
};
+static int cs42l73_spklo_spk_amp_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ struct snd_soc_codec *codec = w->codec;
+ struct cs42l73_private *priv = snd_soc_codec_get_drvdata(codec);
+ switch (event) {
+ case SND_SOC_DAPM_POST_PMD:
+ /* 150 ms delay between setting PDN and MCLKDIS */
+ priv->shutdwn_delay = 150;
+ break;
+ default:
+ pr_err("Invalid event = 0x%x\n", event);
+ }
+ return 0;
+}
+
+static int cs42l73_ear_amp_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ struct snd_soc_codec *codec = w->codec;
+ struct cs42l73_private *priv = snd_soc_codec_get_drvdata(codec);
+ switch (event) {
+ case SND_SOC_DAPM_POST_PMD:
+ /* 50 ms delay between setting PDN and MCLKDIS */
+ if (priv->shutdwn_delay < 50)
+ priv->shutdwn_delay = 50;
+ break;
+ default:
+ pr_err("Invalid event = 0x%x\n", event);
+ }
+ return 0;
+}
+
+
+static int cs42l73_hp_amp_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ struct snd_soc_codec *codec = w->codec;
+ struct cs42l73_private *priv = snd_soc_codec_get_drvdata(codec);
+ switch (event) {
+ case SND_SOC_DAPM_POST_PMD:
+ /* 30 ms delay between setting PDN and MCLKDIS */
+ if (priv->shutdwn_delay < 30)
+ priv->shutdwn_delay = 30;
+ break;
+ default:
+ pr_err("Invalid event = 0x%x\n", event);
+ }
+ return 0;
+}
+
static const struct snd_soc_dapm_widget cs42l73_dapm_widgets[] = {
+ SND_SOC_DAPM_INPUT("DMICA"),
+ SND_SOC_DAPM_INPUT("DMICB"),
SND_SOC_DAPM_INPUT("LINEINA"),
SND_SOC_DAPM_INPUT("LINEINB"),
SND_SOC_DAPM_INPUT("MIC1"),
@@ -604,9 +658,7 @@ static const struct snd_soc_dapm_widget cs42l73_dapm_widgets[] = {
CS42L73_PWRCTL2, 3, 1),
SND_SOC_DAPM_AIF_OUT("ASPOUTR", NULL, 0,
CS42L73_PWRCTL2, 3, 1),
- SND_SOC_DAPM_AIF_OUT("VSPOUTL", NULL, 0,
- CS42L73_PWRCTL2, 4, 1),
- SND_SOC_DAPM_AIF_OUT("VSPOUTR", NULL, 0,
+ SND_SOC_DAPM_AIF_OUT("VSPINOUT", NULL, 0,
CS42L73_PWRCTL2, 4, 1),
SND_SOC_DAPM_PGA("PGA Left", SND_SOC_NOPM, 0, 0, NULL, 0),
@@ -632,8 +684,7 @@ static const struct snd_soc_dapm_widget cs42l73_dapm_widgets[] = {
SND_SOC_DAPM_MIXER("ASPR Output Mixer", SND_SOC_NOPM, 0, 0, NULL, 0),
SND_SOC_DAPM_MIXER("XSPL Output Mixer", SND_SOC_NOPM, 0, 0, NULL, 0),
SND_SOC_DAPM_MIXER("XSPR Output Mixer", SND_SOC_NOPM, 0, 0, NULL, 0),
- SND_SOC_DAPM_MIXER("VSPL Output Mixer", SND_SOC_NOPM, 0, 0, NULL, 0),
- SND_SOC_DAPM_MIXER("VSPR Output Mixer", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_MIXER("VSP Output Mixer", SND_SOC_NOPM, 0, 0, NULL, 0),
SND_SOC_DAPM_AIF_IN("XSPINL", NULL, 0,
CS42L73_PWRCTL2, 0, 1),
@@ -649,7 +700,7 @@ static const struct snd_soc_dapm_widget cs42l73_dapm_widgets[] = {
SND_SOC_DAPM_AIF_IN("ASPINM", NULL, 0,
CS42L73_PWRCTL2, 2, 1),
- SND_SOC_DAPM_AIF_IN("VSPIN", NULL, 0,
+ SND_SOC_DAPM_AIF_IN("VSPINOUT", NULL, 0,
CS42L73_PWRCTL2, 4, 1),
SND_SOC_DAPM_MIXER("HL Left Mixer", SND_SOC_NOPM, 0, 0, NULL, 0),
@@ -674,16 +725,20 @@ static const struct snd_soc_dapm_widget cs42l73_dapm_widgets[] = {
SND_SOC_DAPM_PGA("SPK DAC", SND_SOC_NOPM, 0, 0, NULL, 0),
SND_SOC_DAPM_PGA("ESL DAC", SND_SOC_NOPM, 0, 0, NULL, 0),
- SND_SOC_DAPM_SWITCH("HP Amp", CS42L73_PWRCTL3, 0, 1,
- &hp_amp_ctl),
+ SND_SOC_DAPM_SWITCH_E("HP Amp", CS42L73_PWRCTL3, 0, 1,
+ &hp_amp_ctl, cs42l73_hp_amp_event,
+ SND_SOC_DAPM_POST_PMD),
SND_SOC_DAPM_SWITCH("LO Amp", CS42L73_PWRCTL3, 1, 1,
&lo_amp_ctl),
- SND_SOC_DAPM_SWITCH("SPK Amp", CS42L73_PWRCTL3, 2, 1,
- &spk_amp_ctl),
- SND_SOC_DAPM_SWITCH("EAR Amp", CS42L73_PWRCTL3, 3, 1,
- &ear_amp_ctl),
- SND_SOC_DAPM_SWITCH("SPKLO Amp", CS42L73_PWRCTL3, 4, 1,
- &spklo_amp_ctl),
+ SND_SOC_DAPM_SWITCH_E("SPK Amp", CS42L73_PWRCTL3, 2, 1,
+ &spk_amp_ctl, cs42l73_spklo_spk_amp_event,
+ SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_SWITCH_E("EAR Amp", CS42L73_PWRCTL3, 3, 1,
+ &ear_amp_ctl, cs42l73_ear_amp_event,
+ SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_SWITCH_E("SPKLO Amp", CS42L73_PWRCTL3, 4, 1,
+ &spklo_amp_ctl, cs42l73_spklo_spk_amp_event,
+ SND_SOC_DAPM_POST_PMD),
SND_SOC_DAPM_OUTPUT("HPOUTA"),
SND_SOC_DAPM_OUTPUT("HPOUTB"),
@@ -705,7 +760,7 @@ static const struct snd_soc_dapm_route cs42l73_audio_map[] = {
{"ESL DAC", "ESL-ASP Mono Volume", "ESL Mixer"},
{"ESL DAC", "ESL-XSP Mono Volume", "ESL Mixer"},
- {"ESL DAC", "ESL-VSP Mono Volume", "VSPIN"},
+ {"ESL DAC", "ESL-VSP Mono Volume", "VSPINOUT"},
/* Loopback */
{"ESL DAC", "ESL-IP Mono Volume", "Input Left Capture"},
{"ESL DAC", "ESL-IP Mono Volume", "Input Right Capture"},
@@ -727,7 +782,7 @@ static const struct snd_soc_dapm_route cs42l73_audio_map[] = {
{"SPK DAC", "SPK-ASP Mono Volume", "SPK Mixer"},
{"SPK DAC", "SPK-XSP Mono Volume", "SPK Mixer"},
- {"SPK DAC", "SPK-VSP Mono Volume", "VSPIN"},
+ {"SPK DAC", "SPK-VSP Mono Volume", "VSPINOUT"},
/* Loopback */
{"SPK DAC", "SPK-IP Mono Volume", "Input Left Capture"},
{"SPK DAC", "SPK-IP Mono Volume", "Input Right Capture"},
@@ -770,8 +825,8 @@ static const struct snd_soc_dapm_route cs42l73_audio_map[] = {
{"HL Right Mixer", NULL, "ASPINR"},
{"HL Left Mixer", NULL, "XSPINL"},
{"HL Right Mixer", NULL, "XSPINR"},
- {"HL Left Mixer", NULL, "VSPIN"},
- {"HL Right Mixer", NULL, "VSPIN"},
+ {"HL Left Mixer", NULL, "VSPINOUT"},
+ {"HL Right Mixer", NULL, "VSPINOUT"},
{"ASPINL", NULL, "ASP Playback"},
{"ASPINM", NULL, "ASP Playback"},
@@ -779,7 +834,7 @@ static const struct snd_soc_dapm_route cs42l73_audio_map[] = {
{"XSPINL", NULL, "XSP Playback"},
{"XSPINM", NULL, "XSP Playback"},
{"XSPINR", NULL, "XSP Playback"},
- {"VSPIN", NULL, "VSP Playback"},
+ {"VSPINOUT", NULL, "VSP Playback"},
/* Capture Paths */
{"MIC1", NULL, "MIC1 Bias"},
@@ -795,6 +850,8 @@ static const struct snd_soc_dapm_route cs42l73_audio_map[] = {
{"ADC Left", NULL, "PGA Left"},
{"ADC Right", NULL, "PGA Right"},
+ {"DMIC Left", NULL, "DMICA"},
+ {"DMIC Right", NULL, "DMICB"},
{"Input Left Capture", "ADC Left Input", "ADC Left"},
{"Input Right Capture", "ADC Right Input", "ADC Right"},
@@ -819,21 +876,18 @@ static const struct snd_soc_dapm_route cs42l73_audio_map[] = {
{"XSPOUTR", NULL, "XSPR Output Mixer"},
/* Voice Capture */
- {"VSPL Output Mixer", NULL, "Input Left Capture"},
- {"VSPR Output Mixer", NULL, "Input Left Capture"},
+ {"VSP Output Mixer", NULL, "Input Left Capture"},
+ {"VSP Output Mixer", NULL, "Input Right Capture"},
- {"VSPOUTL", "VSP-IP Volume", "VSPL Output Mixer"},
- {"VSPOUTR", "VSP-IP Volume", "VSPR Output Mixer"},
+ {"VSPINOUT", "VSP-IP Volume", "VSP Output Mixer"},
- {"VSPOUTL", NULL, "VSPL Output Mixer"},
- {"VSPOUTR", NULL, "VSPR Output Mixer"},
+ {"VSPINOUT", NULL, "VSP Output Mixer"},
{"ASP Capture", NULL, "ASPOUTL"},
{"ASP Capture", NULL, "ASPOUTR"},
{"XSP Capture", NULL, "XSPOUTL"},
{"XSP Capture", NULL, "XSPOUTR"},
- {"VSP Capture", NULL, "VSPOUTL"},
- {"VSP Capture", NULL, "VSPOUTR"},
+ {"VSP Capture", NULL, "VSPINOUT"},
};
struct cs42l73_mclk_div {
@@ -1167,6 +1221,14 @@ static int cs42l73_set_bias_level(struct snd_soc_codec *codec,
case SND_SOC_BIAS_OFF:
snd_soc_update_bits(codec, CS42L73_PWRCTL1, PDN, 1);
+ if (cs42l73->shutdwn_delay > 0) {
+ mdelay(cs42l73->shutdwn_delay);
+ cs42l73->shutdwn_delay = 0;
+ } else {
+ mdelay(15); /* Min amount of time requred to power
+ * down.
+ */
+ }
snd_soc_update_bits(codec, CS42L73_DMMCC, MCLKDIS, 1);
break;
}
diff --git a/sound/soc/codecs/lm49453.c b/sound/soc/codecs/lm49453.c
index d75257d40a49..e19490cfb3a8 100644
--- a/sound/soc/codecs/lm49453.c
+++ b/sound/soc/codecs/lm49453.c
@@ -111,9 +111,9 @@ static struct reg_default lm49453_reg_defs[] = {
{ 101, 0x00 },
{ 102, 0x00 },
{ 103, 0x01 },
- { 105, 0x01 },
- { 106, 0x00 },
- { 107, 0x01 },
+ { 104, 0x01 },
+ { 105, 0x00 },
+ { 106, 0x01 },
{ 107, 0x00 },
{ 108, 0x00 },
{ 109, 0x00 },
@@ -163,56 +163,25 @@ static struct reg_default lm49453_reg_defs[] = {
{ 184, 0x00 },
{ 185, 0x00 },
{ 186, 0x00 },
- { 189, 0x00 },
+ { 187, 0x00 },
{ 188, 0x00 },
- { 194, 0x00 },
- { 195, 0x00 },
- { 196, 0x00 },
- { 197, 0x00 },
- { 200, 0x00 },
- { 201, 0x00 },
- { 202, 0x00 },
- { 203, 0x00 },
- { 204, 0x00 },
- { 205, 0x00 },
- { 208, 0x00 },
+ { 189, 0x00 },
+ { 208, 0x06 },
{ 209, 0x00 },
- { 210, 0x00 },
- { 211, 0x00 },
- { 213, 0x00 },
- { 214, 0x00 },
- { 215, 0x00 },
- { 216, 0x00 },
- { 217, 0x00 },
- { 218, 0x00 },
- { 219, 0x00 },
+ { 210, 0x08 },
+ { 211, 0x54 },
+ { 212, 0x14 },
+ { 213, 0x0d },
+ { 214, 0x0d },
+ { 215, 0x14 },
+ { 216, 0x60 },
{ 221, 0x00 },
{ 222, 0x00 },
+ { 223, 0x00 },
{ 224, 0x00 },
- { 225, 0x00 },
- { 226, 0x00 },
- { 227, 0x00 },
- { 228, 0x00 },
- { 229, 0x00 },
- { 230, 0x13 },
- { 231, 0x00 },
- { 232, 0x80 },
- { 233, 0x0C },
- { 234, 0xDD },
- { 235, 0x00 },
- { 236, 0x04 },
- { 237, 0x00 },
- { 238, 0x00 },
- { 239, 0x00 },
- { 240, 0x00 },
- { 241, 0x00 },
- { 242, 0x00 },
- { 243, 0x00 },
- { 244, 0x00 },
- { 245, 0x00 },
{ 248, 0x00 },
{ 249, 0x00 },
- { 254, 0x00 },
+ { 250, 0x00 },
{ 255, 0x00 },
};
@@ -525,36 +494,41 @@ SOC_DAPM_SINGLE("Port2_2 Switch", LM49453_P0_PORT2_TX2_REG, 7, 1, 0),
};
/* TLV Declarations */
-static const DECLARE_TLV_DB_SCALE(digital_tlv, -7650, 150, 1);
-static const DECLARE_TLV_DB_SCALE(port_tlv, 0, 600, 0);
+static const DECLARE_TLV_DB_SCALE(adc_dac_tlv, -7650, 150, 1);
+static const DECLARE_TLV_DB_SCALE(mic_tlv, 0, 200, 1);
+static const DECLARE_TLV_DB_SCALE(port_tlv, -1800, 600, 0);
+static const DECLARE_TLV_DB_SCALE(stn_tlv, -7200, 150, 0);
static const struct snd_kcontrol_new lm49453_sidetone_mixer_controls[] = {
/* Sidetone supports mono only */
SOC_DAPM_SINGLE_TLV("Sidetone ADCL Volume", LM49453_P0_STN_VOL_ADCL_REG,
- 0, 0x3F, 0, digital_tlv),
+ 0, 0x3F, 0, stn_tlv),
SOC_DAPM_SINGLE_TLV("Sidetone ADCR Volume", LM49453_P0_STN_VOL_ADCR_REG,
- 0, 0x3F, 0, digital_tlv),
+ 0, 0x3F, 0, stn_tlv),
SOC_DAPM_SINGLE_TLV("Sidetone DMIC1L Volume", LM49453_P0_STN_VOL_DMIC1L_REG,
- 0, 0x3F, 0, digital_tlv),
+ 0, 0x3F, 0, stn_tlv),
SOC_DAPM_SINGLE_TLV("Sidetone DMIC1R Volume", LM49453_P0_STN_VOL_DMIC1R_REG,
- 0, 0x3F, 0, digital_tlv),
+ 0, 0x3F, 0, stn_tlv),
SOC_DAPM_SINGLE_TLV("Sidetone DMIC2L Volume", LM49453_P0_STN_VOL_DMIC2L_REG,
- 0, 0x3F, 0, digital_tlv),
+ 0, 0x3F, 0, stn_tlv),
SOC_DAPM_SINGLE_TLV("Sidetone DMIC2R Volume", LM49453_P0_STN_VOL_DMIC2R_REG,
- 0, 0x3F, 0, digital_tlv),
+ 0, 0x3F, 0, stn_tlv),
};
static const struct snd_kcontrol_new lm49453_snd_controls[] = {
/* mic1 and mic2 supports mono only */
- SOC_SINGLE_TLV("Mic1 Volume", LM49453_P0_ADC_LEVELL_REG, 0, 6,
- 0, digital_tlv),
- SOC_SINGLE_TLV("Mic2 Volume", LM49453_P0_ADC_LEVELR_REG, 0, 6,
- 0, digital_tlv),
+ SOC_SINGLE_TLV("Mic1 Volume", LM49453_P0_MICL_REG, 0, 15, 0, mic_tlv),
+ SOC_SINGLE_TLV("Mic2 Volume", LM49453_P0_MICR_REG, 0, 15, 0, mic_tlv),
+
+ SOC_SINGLE_TLV("ADCL Volume", LM49453_P0_ADC_LEVELL_REG, 0, 63,
+ 0, adc_dac_tlv),
+ SOC_SINGLE_TLV("ADCR Volume", LM49453_P0_ADC_LEVELR_REG, 0, 63,
+ 0, adc_dac_tlv),
SOC_DOUBLE_R_TLV("DMIC1 Volume", LM49453_P0_DMIC1_LEVELL_REG,
- LM49453_P0_DMIC1_LEVELR_REG, 0, 6, 0, digital_tlv),
+ LM49453_P0_DMIC1_LEVELR_REG, 0, 63, 0, adc_dac_tlv),
SOC_DOUBLE_R_TLV("DMIC2 Volume", LM49453_P0_DMIC2_LEVELL_REG,
- LM49453_P0_DMIC2_LEVELR_REG, 0, 6, 0, digital_tlv),
+ LM49453_P0_DMIC2_LEVELR_REG, 0, 63, 0, adc_dac_tlv),
SOC_DAPM_ENUM("Mic2Mode", lm49453_mic2mode_enum),
SOC_DAPM_ENUM("DMIC12 SRC", lm49453_dmic12_cfg_enum),
@@ -569,16 +543,16 @@ static const struct snd_kcontrol_new lm49453_snd_controls[] = {
2, 1, 0),
SOC_DOUBLE_R_TLV("DAC HP Volume", LM49453_P0_DAC_HP_LEVELL_REG,
- LM49453_P0_DAC_HP_LEVELR_REG, 0, 6, 0, digital_tlv),
+ LM49453_P0_DAC_HP_LEVELR_REG, 0, 63, 0, adc_dac_tlv),
SOC_DOUBLE_R_TLV("DAC LO Volume", LM49453_P0_DAC_LO_LEVELL_REG,
- LM49453_P0_DAC_LO_LEVELR_REG, 0, 6, 0, digital_tlv),
+ LM49453_P0_DAC_LO_LEVELR_REG, 0, 63, 0, adc_dac_tlv),
SOC_DOUBLE_R_TLV("DAC LS Volume", LM49453_P0_DAC_LS_LEVELL_REG,
- LM49453_P0_DAC_LS_LEVELR_REG, 0, 6, 0, digital_tlv),
+ LM49453_P0_DAC_LS_LEVELR_REG, 0, 63, 0, adc_dac_tlv),
SOC_DOUBLE_R_TLV("DAC HA Volume", LM49453_P0_DAC_HA_LEVELL_REG,
- LM49453_P0_DAC_HA_LEVELR_REG, 0, 6, 0, digital_tlv),
+ LM49453_P0_DAC_HA_LEVELR_REG, 0, 63, 0, adc_dac_tlv),
SOC_SINGLE_TLV("EP Volume", LM49453_P0_DAC_LS_LEVELL_REG,
- 0, 6, 0, digital_tlv),
+ 0, 63, 0, adc_dac_tlv),
SOC_SINGLE_TLV("PORT1_1_RX_LVL Volume", LM49453_P0_PORT1_RX_LVL1_REG,
0, 3, 0, port_tlv),
@@ -1218,7 +1192,7 @@ static int lm49453_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
}
snd_soc_update_bits(codec, LM49453_P0_AUDIO_PORT1_BASIC_REG,
- LM49453_AUDIO_PORT1_BASIC_FMT_MASK|BIT(1)|BIT(5),
+ LM49453_AUDIO_PORT1_BASIC_FMT_MASK|BIT(0)|BIT(5),
(aif_val | mode | clk_phase));
snd_soc_write(codec, LM49453_P0_AUDIO_PORT1_RX_MSB_REG, clk_shift);
diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c
index cb1675cd8e1c..92bbfec9b107 100644
--- a/sound/soc/codecs/sgtl5000.c
+++ b/sound/soc/codecs/sgtl5000.c
@@ -401,7 +401,7 @@ static const struct snd_kcontrol_new sgtl5000_snd_controls[] = {
5, 1, 0),
SOC_SINGLE_TLV("Mic Volume", SGTL5000_CHIP_MIC_CTRL,
- 0, 4, 0, mic_gain_tlv),
+ 0, 3, 0, mic_gain_tlv),
};
/* mute the codec used by alsa core */
@@ -1344,7 +1344,7 @@ static int sgtl5000_probe(struct snd_soc_codec *codec)
SGTL5000_HP_ZCD_EN |
SGTL5000_ADC_ZCD_EN);
- snd_soc_write(codec, SGTL5000_CHIP_MIC_CTRL, 0);
+ snd_soc_write(codec, SGTL5000_CHIP_MIC_CTRL, 2);
/*
* disable DAP
diff --git a/sound/soc/codecs/sigmadsp.c b/sound/soc/codecs/sigmadsp.c
index 5be42bf56996..4068f2491232 100644
--- a/sound/soc/codecs/sigmadsp.c
+++ b/sound/soc/codecs/sigmadsp.c
@@ -225,7 +225,7 @@ EXPORT_SYMBOL(process_sigma_firmware);
static int sigma_action_write_regmap(void *control_data,
const struct sigma_action *sa, size_t len)
{
- return regmap_raw_write(control_data, le16_to_cpu(sa->addr),
+ return regmap_raw_write(control_data, be16_to_cpu(sa->addr),
sa->payload, len - 2);
}
diff --git a/sound/soc/codecs/sta529.c b/sound/soc/codecs/sta529.c
index ab355c4f0b2d..40c07be9b581 100644
--- a/sound/soc/codecs/sta529.c
+++ b/sound/soc/codecs/sta529.c
@@ -74,9 +74,10 @@
SNDRV_PCM_FMTBIT_S32_LE)
#define S2PC_VALUE 0x98
#define CLOCK_OUT 0x60
-#define LEFT_J_DATA_FORMAT 0x10
-#define I2S_DATA_FORMAT 0x12
-#define RIGHT_J_DATA_FORMAT 0x14
+#define DATA_FORMAT_MSK 0x0E
+#define LEFT_J_DATA_FORMAT 0x00
+#define I2S_DATA_FORMAT 0x02
+#define RIGHT_J_DATA_FORMAT 0x04
#define CODEC_MUTE_VAL 0x80
#define POWER_CNTLMSAK 0x40
@@ -289,7 +290,7 @@ static int sta529_set_dai_fmt(struct snd_soc_dai *codec_dai, u32 fmt)
return -EINVAL;
}
- snd_soc_update_bits(codec, STA529_S2PCFG0, 0x0D, mode);
+ snd_soc_update_bits(codec, STA529_S2PCFG0, DATA_FORMAT_MSK, mode);
return 0;
}
diff --git a/sound/soc/codecs/tpa6130a2.c b/sound/soc/codecs/tpa6130a2.c
index 8d75aa152c8c..c58bee8346ce 100644
--- a/sound/soc/codecs/tpa6130a2.c
+++ b/sound/soc/codecs/tpa6130a2.c
@@ -398,7 +398,8 @@ static int tpa6130a2_probe(struct i2c_client *client,
TPA6130A2_MUTE_L;
if (data->power_gpio >= 0) {
- ret = gpio_request(data->power_gpio, "tpa6130a2 enable");
+ ret = devm_gpio_request(dev, data->power_gpio,
+ "tpa6130a2 enable");
if (ret < 0) {
dev_err(dev, "Failed to request power GPIO (%d)\n",
data->power_gpio);
@@ -419,16 +420,16 @@ static int tpa6130a2_probe(struct i2c_client *client,
break;
}
- data->supply = regulator_get(dev, regulator);
+ data->supply = devm_regulator_get(dev, regulator);
if (IS_ERR(data->supply)) {
ret = PTR_ERR(data->supply);
dev_err(dev, "Failed to request supply: %d\n", ret);
- goto err_regulator;
+ goto err_gpio;
}
ret = tpa6130a2_power(1);
if (ret != 0)
- goto err_power;
+ goto err_gpio;
/* Read version */
@@ -440,15 +441,10 @@ static int tpa6130a2_probe(struct i2c_client *client,
/* Disable the chip */
ret = tpa6130a2_power(0);
if (ret != 0)
- goto err_power;
+ goto err_gpio;
return 0;
-err_power:
- regulator_put(data->supply);
-err_regulator:
- if (data->power_gpio >= 0)
- gpio_free(data->power_gpio);
err_gpio:
tpa6130a2_client = NULL;
@@ -457,14 +453,7 @@ err_gpio:
static int tpa6130a2_remove(struct i2c_client *client)
{
- struct tpa6130a2_data *data = i2c_get_clientdata(client);
-
tpa6130a2_power(0);
-
- if (data->power_gpio >= 0)
- gpio_free(data->power_gpio);
-
- regulator_put(data->supply);
tpa6130a2_client = NULL;
return 0;
diff --git a/sound/soc/codecs/wm2000.c b/sound/soc/codecs/wm2000.c
index 1cbe88f01d63..12bcae63a7f0 100644
--- a/sound/soc/codecs/wm2000.c
+++ b/sound/soc/codecs/wm2000.c
@@ -209,9 +209,9 @@ static int wm2000_power_up(struct i2c_client *i2c, int analogue)
ret = wm2000_read(i2c, WM2000_REG_SPEECH_CLARITY);
if (wm2000->speech_clarity)
- ret &= ~WM2000_SPEECH_CLARITY;
- else
ret |= WM2000_SPEECH_CLARITY;
+ else
+ ret &= ~WM2000_SPEECH_CLARITY;
wm2000_write(i2c, WM2000_REG_SPEECH_CLARITY, ret);
wm2000_write(i2c, WM2000_REG_SYS_START0, 0x33);
diff --git a/sound/soc/codecs/wm2200.c b/sound/soc/codecs/wm2200.c
index afcf31df77e0..e6cefe1ac677 100644
--- a/sound/soc/codecs/wm2200.c
+++ b/sound/soc/codecs/wm2200.c
@@ -1566,15 +1566,9 @@ static int wm2200_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
case SND_SOC_DAIFMT_DSP_A:
fmt_val = 0;
break;
- case SND_SOC_DAIFMT_DSP_B:
- fmt_val = 1;
- break;
case SND_SOC_DAIFMT_I2S:
fmt_val = 2;
break;
- case SND_SOC_DAIFMT_LEFT_J:
- fmt_val = 3;
- break;
default:
dev_err(codec->dev, "Unsupported DAI format %d\n",
fmt & SND_SOC_DAIFMT_FORMAT_MASK);
@@ -1626,7 +1620,7 @@ static int wm2200_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
WM2200_AIF1TX_LRCLK_MSTR | WM2200_AIF1TX_LRCLK_INV,
lrclk);
snd_soc_update_bits(codec, WM2200_AUDIO_IF_1_5,
- WM2200_AIF1_FMT_MASK << 1, fmt_val << 1);
+ WM2200_AIF1_FMT_MASK, fmt_val);
return 0;
}
diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c
index 5a5f36936235..54397a508073 100644
--- a/sound/soc/codecs/wm5100.c
+++ b/sound/soc/codecs/wm5100.c
@@ -1279,15 +1279,9 @@ static int wm5100_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
case SND_SOC_DAIFMT_DSP_A:
mask = 0;
break;
- case SND_SOC_DAIFMT_DSP_B:
- mask = 1;
- break;
case SND_SOC_DAIFMT_I2S:
mask = 2;
break;
- case SND_SOC_DAIFMT_LEFT_J:
- mask = 3;
- break;
default:
dev_err(codec->dev, "Unsupported DAI format %d\n",
fmt & SND_SOC_DAIFMT_FORMAT_MASK);
diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c
index 688ade080589..7a9048dad1cd 100644
--- a/sound/soc/codecs/wm5102.c
+++ b/sound/soc/codecs/wm5102.c
@@ -36,6 +36,9 @@
struct wm5102_priv {
struct arizona_priv core;
struct arizona_fll fll[2];
+
+ unsigned int spk_ena:2;
+ unsigned int spk_ena_pending:1;
};
static DECLARE_TLV_DB_SCALE(ana_tlv, 0, 100, 0);
@@ -787,6 +790,47 @@ ARIZONA_MIXER_CONTROLS("AIF3TX1", ARIZONA_AIF3TX1MIX_INPUT_1_SOURCE),
ARIZONA_MIXER_CONTROLS("AIF3TX2", ARIZONA_AIF3TX2MIX_INPUT_1_SOURCE),
};
+static int wm5102_spk_ev(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol,
+ int event)
+{
+ struct snd_soc_codec *codec = w->codec;
+ struct arizona *arizona = dev_get_drvdata(codec->dev->parent);
+ struct wm5102_priv *wm5102 = snd_soc_codec_get_drvdata(codec);
+
+ if (arizona->rev < 1)
+ return 0;
+
+ switch (event) {
+ case SND_SOC_DAPM_PRE_PMU:
+ if (!wm5102->spk_ena) {
+ snd_soc_write(codec, 0x4f5, 0x25a);
+ wm5102->spk_ena_pending = true;
+ }
+ break;
+ case SND_SOC_DAPM_POST_PMU:
+ if (wm5102->spk_ena_pending) {
+ msleep(75);
+ snd_soc_write(codec, 0x4f5, 0xda);
+ wm5102->spk_ena_pending = false;
+ wm5102->spk_ena++;
+ }
+ break;
+ case SND_SOC_DAPM_PRE_PMD:
+ wm5102->spk_ena--;
+ if (!wm5102->spk_ena)
+ snd_soc_write(codec, 0x4f5, 0x25a);
+ break;
+ case SND_SOC_DAPM_POST_PMD:
+ if (!wm5102->spk_ena)
+ snd_soc_write(codec, 0x4f5, 0x0da);
+ break;
+ }
+
+ return 0;
+}
+
+
ARIZONA_MIXER_ENUMS(EQ1, ARIZONA_EQ1MIX_INPUT_1_SOURCE);
ARIZONA_MIXER_ENUMS(EQ2, ARIZONA_EQ2MIX_INPUT_1_SOURCE);
ARIZONA_MIXER_ENUMS(EQ3, ARIZONA_EQ3MIX_INPUT_1_SOURCE);
@@ -1034,10 +1078,10 @@ SND_SOC_DAPM_PGA_E("OUT3L", ARIZONA_OUTPUT_ENABLES_1,
ARIZONA_OUT3L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev,
SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
SND_SOC_DAPM_PGA_E("OUT4L", ARIZONA_OUTPUT_ENABLES_1,
- ARIZONA_OUT4L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev,
+ ARIZONA_OUT4L_ENA_SHIFT, 0, NULL, 0, wm5102_spk_ev,
SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
SND_SOC_DAPM_PGA_E("OUT4R", ARIZONA_OUTPUT_ENABLES_1,
- ARIZONA_OUT4R_ENA_SHIFT, 0, NULL, 0, arizona_out_ev,
+ ARIZONA_OUT4R_ENA_SHIFT, 0, NULL, 0, wm5102_spk_ev,
SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
SND_SOC_DAPM_PGA_E("OUT5L", ARIZONA_OUTPUT_ENABLES_1,
ARIZONA_OUT5L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev,
diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c
index ffc89fab96fb..7b198c38f3ef 100644
--- a/sound/soc/codecs/wm_adsp.c
+++ b/sound/soc/codecs/wm_adsp.c
@@ -169,6 +169,7 @@ static int wm_adsp_load(struct wm_adsp *dsp)
const struct wm_adsp_region *mem;
const char *region_name;
char *file, *text;
+ void *buf;
unsigned int reg;
int regions = 0;
int ret, offset, type, sizes;
@@ -322,8 +323,18 @@ static int wm_adsp_load(struct wm_adsp *dsp)
}
if (reg) {
- ret = regmap_raw_write(regmap, reg, region->data,
+ buf = kmemdup(region->data, le32_to_cpu(region->len),
+ GFP_KERNEL);
+ if (!buf) {
+ adsp_err(dsp, "Out of memory\n");
+ return -ENOMEM;
+ }
+
+ ret = regmap_raw_write(regmap, reg, buf,
le32_to_cpu(region->len));
+
+ kfree(buf);
+
if (ret != 0) {
adsp_err(dsp,
"%s.%d: Failed to write %d bytes at %d in %s: %d\n",
@@ -359,6 +370,7 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp)
const char *region_name;
int ret, pos, blocks, type, offset, reg;
char *file;
+ void *buf;
file = kzalloc(PAGE_SIZE, GFP_KERNEL);
if (file == NULL)
@@ -426,6 +438,13 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp)
}
if (reg) {
+ buf = kmemdup(blk->data, le32_to_cpu(blk->len),
+ GFP_KERNEL);
+ if (!buf) {
+ adsp_err(dsp, "Out of memory\n");
+ return -ENOMEM;
+ }
+
ret = regmap_raw_write(regmap, reg, blk->data,
le32_to_cpu(blk->len));
if (ret != 0) {
@@ -433,6 +452,8 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp)
"%s.%d: Failed to write to %x in %s\n",
file, blocks, reg, region_name);
}
+
+ kfree(buf);
}
pos += le32_to_cpu(blk->len) + sizeof(*blk);
generated by cgit v1.2.3 (git 2.39.1) at 2025-01-19 21:34:26 +0100