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-rw-r--r--sound/soc/codecs/88pm860x-codec.c12
-rw-r--r--sound/soc/codecs/ak4535.c10
-rw-r--r--sound/soc/codecs/alc5623.c10
-rw-r--r--sound/soc/codecs/wm8350.c9
-rw-r--r--sound/soc/codecs/wm8580.c18
-rw-r--r--sound/soc/codecs/wm8731.c88
-rw-r--r--sound/soc/codecs/wm8753.c21
-rw-r--r--sound/soc/codecs/wm8903.c182
-rw-r--r--sound/soc/codecs/wm8904.c3
-rw-r--r--sound/soc/codecs/wm8955.c3
-rw-r--r--sound/soc/codecs/wm8960.c3
-rw-r--r--sound/soc/codecs/wm8962.c2
-rw-r--r--sound/soc/codecs/wm8993.c6
-rw-r--r--sound/soc/codecs/wm8994.c193
-rw-r--r--sound/soc/codecs/wm9081.c4
-rw-r--r--sound/soc/codecs/wm_hubs.c2
16 files changed, 389 insertions, 177 deletions
diff --git a/sound/soc/codecs/88pm860x-codec.c b/sound/soc/codecs/88pm860x-codec.c
index 7e4d88007d4f..08e15dee9182 100644
--- a/sound/soc/codecs/88pm860x-codec.c
+++ b/sound/soc/codecs/88pm860x-codec.c
@@ -1358,7 +1358,7 @@ static int pm860x_probe(struct snd_soc_codec *codec)
pm860x->name[i], pm860x);
if (ret < 0) {
dev_err(codec->dev, "Failed to request IRQ!\n");
- goto out_irq;
+ goto out;
}
}
@@ -1369,7 +1369,7 @@ static int pm860x_probe(struct snd_soc_codec *codec)
if (ret < 0) {
dev_err(codec->dev, "Failed to fill register cache: %d\n",
ret);
- goto out_codec;
+ goto out;
}
snd_soc_add_controls(codec, pm860x_snd_controls,
@@ -1379,12 +1379,10 @@ static int pm860x_probe(struct snd_soc_codec *codec)
snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
return 0;
-out_codec:
- i = 3;
-out_irq:
- for (; i >= 0; i--)
+out:
+ while (--i >= 0)
free_irq(pm860x->irq[i], pm860x);
- return -EINVAL;
+ return ret;
}
static int pm860x_remove(struct snd_soc_codec *codec)
diff --git a/sound/soc/codecs/ak4535.c b/sound/soc/codecs/ak4535.c
index 73675458076d..8b38739c88f8 100644
--- a/sound/soc/codecs/ak4535.c
+++ b/sound/soc/codecs/ak4535.c
@@ -366,9 +366,9 @@ static int ak4535_set_dai_fmt(struct snd_soc_dai *codec_dai,
static int ak4535_mute(struct snd_soc_dai *dai, int mute)
{
struct snd_soc_codec *codec = dai->codec;
- u16 mute_reg = ak4535_read_reg_cache(codec, AK4535_DAC) & 0xffdf;
+ u16 mute_reg = ak4535_read_reg_cache(codec, AK4535_DAC);
if (!mute)
- ak4535_write(codec, AK4535_DAC, mute_reg);
+ ak4535_write(codec, AK4535_DAC, mute_reg & ~0x20);
else
ak4535_write(codec, AK4535_DAC, mute_reg | 0x20);
return 0;
@@ -381,11 +381,11 @@ static int ak4535_set_bias_level(struct snd_soc_codec *codec,
switch (level) {
case SND_SOC_BIAS_ON:
- mute_reg = ak4535_read_reg_cache(codec, AK4535_DAC) & 0xffdf;
- ak4535_write(codec, AK4535_DAC, mute_reg);
+ mute_reg = ak4535_read_reg_cache(codec, AK4535_DAC);
+ ak4535_write(codec, AK4535_DAC, mute_reg & ~0x20);
break;
case SND_SOC_BIAS_PREPARE:
- mute_reg = ak4535_read_reg_cache(codec, AK4535_DAC) & 0xffdf;
+ mute_reg = ak4535_read_reg_cache(codec, AK4535_DAC);
ak4535_write(codec, AK4535_DAC, mute_reg | 0x20);
break;
case SND_SOC_BIAS_STANDBY:
diff --git a/sound/soc/codecs/alc5623.c b/sound/soc/codecs/alc5623.c
index 9783e7e2eb93..4f377c9e868d 100644
--- a/sound/soc/codecs/alc5623.c
+++ b/sound/soc/codecs/alc5623.c
@@ -925,7 +925,6 @@ static int alc5623_probe(struct snd_soc_codec *codec)
}
switch (alc5623->id) {
- default:
case 0x21:
snd_soc_add_controls(codec, rt5621_vol_snd_controls,
ARRAY_SIZE(rt5621_vol_snd_controls));
@@ -938,6 +937,8 @@ static int alc5623_probe(struct snd_soc_codec *codec)
snd_soc_add_controls(codec, alc5623_vol_snd_controls,
ARRAY_SIZE(alc5623_vol_snd_controls));
break;
+ default:
+ return -EINVAL;
}
snd_soc_add_controls(codec, alc5623_snd_controls,
@@ -950,7 +951,6 @@ static int alc5623_probe(struct snd_soc_codec *codec)
snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon));
switch (alc5623->id) {
- default:
case 0x21:
case 0x22:
snd_soc_dapm_new_controls(dapm, alc5623_dapm_amp_widgets,
@@ -962,6 +962,8 @@ static int alc5623_probe(struct snd_soc_codec *codec)
snd_soc_dapm_add_routes(dapm, intercon_spk,
ARRAY_SIZE(intercon_spk));
break;
+ default:
+ return -EINVAL;
}
return ret;
@@ -1039,10 +1041,12 @@ static int alc5623_i2c_probe(struct i2c_client *client,
case 0x22:
alc5623_dai.name = "alc5622-hifi";
break;
- default:
case 0x23:
alc5623_dai.name = "alc5623-hifi";
break;
+ default:
+ kfree(alc5623);
+ return -EINVAL;
}
i2c_set_clientdata(client, alc5623);
diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c
index dc6912e9b667..07ba7e3f6a8c 100644
--- a/sound/soc/codecs/wm8350.c
+++ b/sound/soc/codecs/wm8350.c
@@ -1627,7 +1627,6 @@ static int wm8350_codec_remove(struct snd_soc_codec *codec)
{
struct wm8350_data *priv = snd_soc_codec_get_drvdata(codec);
struct wm8350 *wm8350 = dev_get_platdata(codec->dev);
- int ret;
wm8350_clear_bits(wm8350, WM8350_JACK_DETECT,
WM8350_JDL_ENA | WM8350_JDR_ENA);
@@ -1642,15 +1641,9 @@ static int wm8350_codec_remove(struct snd_soc_codec *codec)
priv->hpr.jack = NULL;
priv->mic.jack = NULL;
- /* cancel any work waiting to be queued. */
- ret = cancel_delayed_work(&codec->dapm.delayed_work);
-
/* if there was any work waiting then we run it now and
* wait for its completion */
- if (ret) {
- schedule_delayed_work(&codec->dapm.delayed_work, 0);
- flush_scheduled_work();
- }
+ flush_delayed_work_sync(&codec->dapm.delayed_work);
wm8350_set_bias_level(codec, SND_SOC_BIAS_OFF);
diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c
index e2a927684b48..0ebdecfd27a9 100644
--- a/sound/soc/codecs/wm8580.c
+++ b/sound/soc/codecs/wm8580.c
@@ -160,7 +160,7 @@
static const u16 wm8580_reg[] = {
0x0121, 0x017e, 0x007d, 0x0014, /*R3*/
0x0121, 0x017e, 0x007d, 0x0194, /*R7*/
- 0x001c, 0x0002, 0x0002, 0x00c2, /*R11*/
+ 0x0010, 0x0002, 0x0002, 0x00c2, /*R11*/
0x0182, 0x0082, 0x000a, 0x0024, /*R15*/
0x0009, 0x0000, 0x00ff, 0x0000, /*R19*/
0x00ff, 0x00ff, 0x00ff, 0x00ff, /*R23*/
@@ -491,29 +491,29 @@ static int wm8580_paif_hw_params(struct snd_pcm_substream *substream,
paifa |= 0x8;
break;
case SNDRV_PCM_FORMAT_S20_3LE:
- paifa |= 0x10;
+ paifa |= 0x0;
paifb |= WM8580_AIF_LENGTH_20;
break;
case SNDRV_PCM_FORMAT_S24_LE:
- paifa |= 0x10;
+ paifa |= 0x0;
paifb |= WM8580_AIF_LENGTH_24;
break;
case SNDRV_PCM_FORMAT_S32_LE:
- paifa |= 0x10;
- paifb |= WM8580_AIF_LENGTH_24;
+ paifa |= 0x0;
+ paifb |= WM8580_AIF_LENGTH_32;
break;
default:
return -EINVAL;
}
/* Look up the SYSCLK ratio; accept only exact matches */
- ratio = wm8580->sysclk[dai->id] / params_rate(params);
+ ratio = wm8580->sysclk[dai->driver->id] / params_rate(params);
for (i = 0; i < ARRAY_SIZE(wm8580_sysclk_ratios); i++)
if (ratio == wm8580_sysclk_ratios[i])
break;
if (i == ARRAY_SIZE(wm8580_sysclk_ratios)) {
dev_err(codec->dev, "Invalid clock ratio %d/%d\n",
- wm8580->sysclk[dai->id], params_rate(params));
+ wm8580->sysclk[dai->driver->id], params_rate(params));
return -EINVAL;
}
paifa |= i;
@@ -716,7 +716,7 @@ static int wm8580_set_sysclk(struct snd_soc_dai *dai, int clk_id,
switch (clk_id) {
case WM8580_CLKSRC_ADCMCLK:
- if (dai->id != WM8580_DAI_PAIFTX)
+ if (dai->driver->id != WM8580_DAI_PAIFTX)
return -EINVAL;
sel = 0 << sel_shift;
break;
@@ -735,7 +735,7 @@ static int wm8580_set_sysclk(struct snd_soc_dai *dai, int clk_id,
}
/* We really should validate PLL settings but not yet */
- wm8580->sysclk[dai->id] = freq;
+ wm8580->sysclk[dai->driver->id] = freq;
return snd_soc_update_bits(codec, WM8580_CLKSEL, sel_mask, sel);
}
diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c
index a1041450d9bb..71122dc36826 100644
--- a/sound/soc/codecs/wm8731.c
+++ b/sound/soc/codecs/wm8731.c
@@ -46,6 +46,8 @@ struct wm8731_priv {
u16 reg_cache[WM8731_CACHEREGNUM];
unsigned int sysclk;
int sysclk_type;
+ int playback_fs;
+ bool deemph;
};
@@ -64,16 +66,79 @@ static const u16 wm8731_reg[WM8731_CACHEREGNUM] = {
#define wm8731_reset(c) snd_soc_write(c, WM8731_RESET, 0)
static const char *wm8731_input_select[] = {"Line In", "Mic"};
-static const char *wm8731_deemph[] = {"None", "32Khz", "44.1Khz", "48Khz"};
-static const struct soc_enum wm8731_enum[] = {
- SOC_ENUM_SINGLE(WM8731_APANA, 2, 2, wm8731_input_select),
- SOC_ENUM_SINGLE(WM8731_APDIGI, 1, 4, wm8731_deemph),
-};
+static const struct soc_enum wm8731_insel_enum =
+ SOC_ENUM_SINGLE(WM8731_APANA, 2, 2, wm8731_input_select);
+
+static int wm8731_deemph[] = { 0, 32000, 44100, 48000 };
+
+static int wm8731_set_deemph(struct snd_soc_codec *codec)
+{
+ struct wm8731_priv *wm8731 = snd_soc_codec_get_drvdata(codec);
+ int val, i, best;
+
+ /* If we're using deemphasis select the nearest available sample
+ * rate.
+ */
+ if (wm8731->deemph) {
+ best = 1;
+ for (i = 2; i < ARRAY_SIZE(wm8731_deemph); i++) {
+ if (abs(wm8731_deemph[i] - wm8731->playback_fs) <
+ abs(wm8731_deemph[best] - wm8731->playback_fs))
+ best = i;
+ }
+
+ val = best << 1;
+ } else {
+ best = 0;
+ val = 0;
+ }
+
+ dev_dbg(codec->dev, "Set deemphasis %d (%dHz)\n",
+ best, wm8731_deemph[best]);
+
+ return snd_soc_update_bits(codec, WM8731_APDIGI, 0x6, val);
+}
+
+static int wm8731_get_deemph(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct wm8731_priv *wm8731 = snd_soc_codec_get_drvdata(codec);
+
+ ucontrol->value.enumerated.item[0] = wm8731->deemph;
+
+ return 0;
+}
+
+static int wm8731_put_deemph(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct wm8731_priv *wm8731 = snd_soc_codec_get_drvdata(codec);
+ int deemph = ucontrol->value.enumerated.item[0];
+ int ret = 0;
+
+ if (deemph > 1)
+ return -EINVAL;
+
+ mutex_lock(&codec->mutex);
+ if (wm8731->deemph != deemph) {
+ wm8731->deemph = deemph;
+
+ wm8731_set_deemph(codec);
+
+ ret = 1;
+ }
+ mutex_unlock(&codec->mutex);
+
+ return ret;
+}
static const DECLARE_TLV_DB_SCALE(in_tlv, -3450, 150, 0);
static const DECLARE_TLV_DB_SCALE(sidetone_tlv, -1500, 300, 0);
static const DECLARE_TLV_DB_SCALE(out_tlv, -12100, 100, 1);
+static const DECLARE_TLV_DB_SCALE(mic_tlv, 0, 2000, 0);
static const struct snd_kcontrol_new wm8731_snd_controls[] = {
@@ -86,7 +151,7 @@ SOC_DOUBLE_R_TLV("Capture Volume", WM8731_LINVOL, WM8731_RINVOL, 0, 31, 0,
in_tlv),
SOC_DOUBLE_R("Line Capture Switch", WM8731_LINVOL, WM8731_RINVOL, 7, 1, 1),
-SOC_SINGLE("Mic Boost (+20dB)", WM8731_APANA, 0, 1, 0),
+SOC_SINGLE_TLV("Mic Boost Volume", WM8731_APANA, 0, 1, 0, mic_tlv),
SOC_SINGLE("Mic Capture Switch", WM8731_APANA, 1, 1, 1),
SOC_SINGLE_TLV("Sidetone Playback Volume", WM8731_APANA, 6, 3, 1,
@@ -95,7 +160,8 @@ SOC_SINGLE_TLV("Sidetone Playback Volume", WM8731_APANA, 6, 3, 1,
SOC_SINGLE("ADC High Pass Filter Switch", WM8731_APDIGI, 0, 1, 1),
SOC_SINGLE("Store DC Offset Switch", WM8731_APDIGI, 4, 1, 0),
-SOC_ENUM("Playback De-emphasis", wm8731_enum[1]),
+SOC_SINGLE_BOOL_EXT("Playback Deemphasis Switch", 0,
+ wm8731_get_deemph, wm8731_put_deemph),
};
/* Output Mixer */
@@ -107,7 +173,7 @@ SOC_DAPM_SINGLE("HiFi Playback Switch", WM8731_APANA, 4, 1, 0),
/* Input mux */
static const struct snd_kcontrol_new wm8731_input_mux_controls =
-SOC_DAPM_ENUM("Input Select", wm8731_enum[0]);
+SOC_DAPM_ENUM("Input Select", wm8731_insel_enum);
static const struct snd_soc_dapm_widget wm8731_dapm_widgets[] = {
SND_SOC_DAPM_SUPPLY("OSC", WM8731_PWR, 5, 1, NULL, 0),
@@ -239,6 +305,8 @@ static int wm8731_hw_params(struct snd_pcm_substream *substream,
u16 srate = (coeff_div[i].sr << 2) |
(coeff_div[i].bosr << 1) | coeff_div[i].usb;
+ wm8731->playback_fs = params_rate(params);
+
snd_soc_write(codec, WM8731_SRATE, srate);
/* bit size */
@@ -253,6 +321,8 @@ static int wm8731_hw_params(struct snd_pcm_substream *substream,
break;
}
+ wm8731_set_deemph(codec);
+
snd_soc_write(codec, WM8731_IFACE, iface);
return 0;
}
@@ -526,7 +596,7 @@ static int wm8731_probe(struct snd_soc_codec *codec)
snd_soc_update_bits(codec, WM8731_RINVOL, 0x100, 0);
/* Disable bypass path by default */
- snd_soc_update_bits(codec, WM8731_APANA, 0x4, 0);
+ snd_soc_update_bits(codec, WM8731_APANA, 0x8, 0);
snd_soc_add_controls(codec, wm8731_snd_controls,
ARRAY_SIZE(wm8731_snd_controls));
diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c
index 57bf9468d39a..73507e71cb79 100644
--- a/sound/soc/codecs/wm8753.c
+++ b/sound/soc/codecs/wm8753.c
@@ -1528,25 +1528,6 @@ static int wm8753_resume(struct snd_soc_codec *codec)
return 0;
}
-/*
- * This function forces any delayed work to be queued and run.
- */
-static int run_delayed_work(struct delayed_work *dwork)
-{
- int ret;
-
- /* cancel any work waiting to be queued. */
- ret = cancel_delayed_work(dwork);
-
- /* if there was any work waiting then we run it now and
- * wait for it's completion */
- if (ret) {
- schedule_delayed_work(dwork, 0);
- flush_scheduled_work();
- }
- return ret;
-}
-
static int wm8753_probe(struct snd_soc_codec *codec)
{
struct wm8753_priv *wm8753 = snd_soc_codec_get_drvdata(codec);
@@ -1606,7 +1587,7 @@ static int wm8753_probe(struct snd_soc_codec *codec)
/* power down chip */
static int wm8753_remove(struct snd_soc_codec *codec)
{
- run_delayed_work(&codec->dapm.delayed_work);
+ flush_delayed_work_sync(&codec->dapm.delayed_work);
wm8753_set_bias_level(codec, SND_SOC_BIAS_OFF);
return 0;
diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c
index 620793e51666..d015745a886b 100644
--- a/sound/soc/codecs/wm8903.c
+++ b/sound/soc/codecs/wm8903.c
@@ -218,10 +218,11 @@ struct wm8903_priv {
int sysclk;
int irq;
- /* Reference counts */
+ int fs;
+ int deemph;
+
+ /* Reference count */
int class_w_users;
- int playback_active;
- int capture_active;
struct completion wseq;
@@ -230,9 +231,6 @@ struct wm8903_priv {
int mic_short;
int mic_last_report;
int mic_delay;
-
- struct snd_pcm_substream *master_substream;
- struct snd_pcm_substream *slave_substream;
};
static int wm8903_volatile_register(unsigned int reg)
@@ -462,6 +460,72 @@ static int wm8903_class_w_put(struct snd_kcontrol *kcontrol,
.private_value = SOC_SINGLE_VALUE(reg, shift, max, invert) }
+static int wm8903_deemph[] = { 0, 32000, 44100, 48000 };
+
+static int wm8903_set_deemph(struct snd_soc_codec *codec)
+{
+ struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec);
+ int val, i, best;
+
+ /* If we're using deemphasis select the nearest available sample
+ * rate.
+ */
+ if (wm8903->deemph) {
+ best = 1;
+ for (i = 2; i < ARRAY_SIZE(wm8903_deemph); i++) {
+ if (abs(wm8903_deemph[i] - wm8903->fs) <
+ abs(wm8903_deemph[best] - wm8903->fs))
+ best = i;
+ }
+
+ val = best << WM8903_DEEMPH_SHIFT;
+ } else {
+ best = 0;
+ val = 0;
+ }
+
+ dev_dbg(codec->dev, "Set deemphasis %d (%dHz)\n",
+ best, wm8903_deemph[best]);
+
+ return snd_soc_update_bits(codec, WM8903_DAC_DIGITAL_1,
+ WM8903_DEEMPH_MASK, val);
+}
+
+static int wm8903_get_deemph(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec);
+
+ ucontrol->value.enumerated.item[0] = wm8903->deemph;
+
+ return 0;
+}
+
+static int wm8903_put_deemph(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec);
+ int deemph = ucontrol->value.enumerated.item[0];
+ int ret = 0;
+
+ if (deemph > 1)
+ return -EINVAL;
+
+ mutex_lock(&codec->mutex);
+ if (wm8903->deemph != deemph) {
+ wm8903->deemph = deemph;
+
+ wm8903_set_deemph(codec);
+
+ ret = 1;
+ }
+ mutex_unlock(&codec->mutex);
+
+ return ret;
+}
+
/* ALSA can only do steps of .01dB */
static const DECLARE_TLV_DB_SCALE(digital_tlv, -7200, 75, 1);
@@ -474,6 +538,23 @@ static const DECLARE_TLV_DB_SCALE(drc_tlv_min, 0, 600, 0);
static const DECLARE_TLV_DB_SCALE(drc_tlv_max, 1200, 600, 0);
static const DECLARE_TLV_DB_SCALE(drc_tlv_startup, -300, 50, 0);
+static const char *hpf_mode_text[] = {
+ "Hi-fi", "Voice 1", "Voice 2", "Voice 3"
+};
+
+static const struct soc_enum hpf_mode =
+ SOC_ENUM_SINGLE(WM8903_ADC_DIGITAL_0, 5, 4, hpf_mode_text);
+
+static const char *osr_text[] = {
+ "Low power", "High performance"
+};
+
+static const struct soc_enum adc_osr =
+ SOC_ENUM_SINGLE(WM8903_ANALOGUE_ADC_0, 0, 2, osr_text);
+
+static const struct soc_enum dac_osr =
+ SOC_ENUM_SINGLE(WM8903_DAC_DIGITAL_1, 0, 2, osr_text);
+
static const char *drc_slope_text[] = {
"1", "1/2", "1/4", "1/8", "1/16", "0"
};
@@ -536,13 +617,6 @@ static const char *mute_mode_text[] = {
static const struct soc_enum mute_mode =
SOC_ENUM_SINGLE(WM8903_DAC_DIGITAL_1, 9, 2, mute_mode_text);
-static const char *dac_deemphasis_text[] = {
- "Disabled", "32kHz", "44.1kHz", "48kHz"
-};
-
-static const struct soc_enum dac_deemphasis =
- SOC_ENUM_SINGLE(WM8903_DAC_DIGITAL_1, 1, 4, dac_deemphasis_text);
-
static const char *companding_text[] = {
"ulaw", "alaw"
};
@@ -612,6 +686,9 @@ SOC_SINGLE("Right Input PGA Common Mode Switch", WM8903_ANALOGUE_RIGHT_INPUT_1,
6, 1, 0),
/* ADCs */
+SOC_ENUM("ADC OSR", adc_osr),
+SOC_SINGLE("HPF Switch", WM8903_ADC_DIGITAL_0, 4, 1, 0),
+SOC_ENUM("HPF Mode", hpf_mode),
SOC_SINGLE("DRC Switch", WM8903_DRC_0, 15, 1, 0),
SOC_ENUM("DRC Compressor Slope R0", drc_slope_r0),
SOC_ENUM("DRC Compressor Slope R1", drc_slope_r1),
@@ -641,14 +718,16 @@ SOC_DOUBLE_TLV("Digital Sidetone Volume", WM8903_DAC_DIGITAL_0, 4, 8,
12, 0, digital_sidetone_tlv),
/* DAC */
+SOC_ENUM("DAC OSR", dac_osr),
SOC_DOUBLE_R_TLV("Digital Playback Volume", WM8903_DAC_DIGITAL_VOLUME_LEFT,
WM8903_DAC_DIGITAL_VOLUME_RIGHT, 1, 120, 0, digital_tlv),
SOC_ENUM("DAC Soft Mute Rate", soft_mute),
SOC_ENUM("DAC Mute Mode", mute_mode),
SOC_SINGLE("DAC Mono Switch", WM8903_DAC_DIGITAL_1, 12, 1, 0),
-SOC_ENUM("DAC De-emphasis", dac_deemphasis),
SOC_ENUM("DAC Companding Mode", dac_companding),
SOC_SINGLE("DAC Companding Switch", WM8903_AUDIO_INTERFACE_0, 1, 1, 0),
+SOC_SINGLE_BOOL_EXT("Playback Deemphasis Switch", 0,
+ wm8903_get_deemph, wm8903_put_deemph),
/* Headphones */
SOC_DOUBLE_R("Headphone Switch",
@@ -1222,58 +1301,6 @@ static struct {
{ 0, 0 },
};
-static int wm8903_startup(struct snd_pcm_substream *substream,
- struct snd_soc_dai *dai)
-{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_codec *codec = rtd->codec;
- struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec);
- struct snd_pcm_runtime *master_runtime;
-
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
- wm8903->playback_active++;
- else
- wm8903->capture_active++;
-
- /* The DAI has shared clocks so if we already have a playback or
- * capture going then constrain this substream to match it.
- */
- if (wm8903->master_substream) {
- master_runtime = wm8903->master_substream->runtime;
-
- dev_dbg(codec->dev, "Constraining to %d bits\n",
- master_runtime->sample_bits);
-
- snd_pcm_hw_constraint_minmax(substream->runtime,
- SNDRV_PCM_HW_PARAM_SAMPLE_BITS,
- master_runtime->sample_bits,
- master_runtime->sample_bits);
-
- wm8903->slave_substream = substream;
- } else
- wm8903->master_substream = substream;
-
- return 0;
-}
-
-static void wm8903_shutdown(struct snd_pcm_substream *substream,
- struct snd_soc_dai *dai)
-{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_codec *codec = rtd->codec;
- struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec);
-
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
- wm8903->playback_active--;
- else
- wm8903->capture_active--;
-
- if (wm8903->master_substream == substream)
- wm8903->master_substream = wm8903->slave_substream;
-
- wm8903->slave_substream = NULL;
-}
-
static int wm8903_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params,
struct snd_soc_dai *dai)
@@ -1298,11 +1325,6 @@ static int wm8903_hw_params(struct snd_pcm_substream *substream,
u16 clock1 = snd_soc_read(codec, WM8903_CLOCK_RATES_1);
u16 dac_digital1 = snd_soc_read(codec, WM8903_DAC_DIGITAL_1);
- if (substream == wm8903->slave_substream) {
- dev_dbg(codec->dev, "Ignoring hw_params for slave substream\n");
- return 0;
- }
-
/* Enable sloping stopband filter for low sample rates */
if (fs <= 24000)
dac_digital1 |= WM8903_DAC_SB_FILT;
@@ -1320,19 +1342,6 @@ static int wm8903_hw_params(struct snd_pcm_substream *substream,
}
}
- /* Constraints should stop us hitting this but let's make sure */
- if (wm8903->capture_active)
- switch (sample_rates[dsp_config].rate) {
- case 88200:
- case 96000:
- dev_err(codec->dev, "%dHz unsupported by ADC\n",
- fs);
- return -EINVAL;
-
- default:
- break;
- }
-
dev_dbg(codec->dev, "DSP fs = %dHz\n", sample_rates[dsp_config].rate);
clock1 &= ~WM8903_SAMPLE_RATE_MASK;
clock1 |= sample_rates[dsp_config].value;
@@ -1428,6 +1437,9 @@ static int wm8903_hw_params(struct snd_pcm_substream *substream,
aif2 |= bclk_divs[bclk_div].div;
aif3 |= bclk / fs;
+ wm8903->fs = params_rate(params);
+ wm8903_set_deemph(codec);
+
snd_soc_write(codec, WM8903_CLOCK_RATES_0, clock0);
snd_soc_write(codec, WM8903_CLOCK_RATES_1, clock1);
snd_soc_write(codec, WM8903_AUDIO_INTERFACE_1, aif1);
@@ -1571,8 +1583,6 @@ static irqreturn_t wm8903_irq(int irq, void *data)
SNDRV_PCM_FMTBIT_S24_LE)
static struct snd_soc_dai_ops wm8903_dai_ops = {
- .startup = wm8903_startup,
- .shutdown = wm8903_shutdown,
.hw_params = wm8903_hw_params,
.digital_mute = wm8903_digital_mute,
.set_fmt = wm8903_set_dai_fmt,
diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c
index d0024666eaf9..8ba142abd8e6 100644
--- a/sound/soc/codecs/wm8904.c
+++ b/sound/soc/codecs/wm8904.c
@@ -817,7 +817,8 @@ static int wm8904_get_deemph(struct snd_kcontrol *kcontrol,
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
struct wm8904_priv *wm8904 = snd_soc_codec_get_drvdata(codec);
- return wm8904->deemph;
+ ucontrol->value.enumerated.item[0] = wm8904->deemph;
+ return 0;
}
static int wm8904_put_deemph(struct snd_kcontrol *kcontrol,
diff --git a/sound/soc/codecs/wm8955.c b/sound/soc/codecs/wm8955.c
index 6200beb082b2..ca0265f008d2 100644
--- a/sound/soc/codecs/wm8955.c
+++ b/sound/soc/codecs/wm8955.c
@@ -379,7 +379,8 @@ static int wm8955_get_deemph(struct snd_kcontrol *kcontrol,
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
struct wm8955_priv *wm8955 = snd_soc_codec_get_drvdata(codec);
- return wm8955->deemph;
+ ucontrol->value.enumerated.item[0] = wm8955->deemph;
+ return 0;
}
static int wm8955_put_deemph(struct snd_kcontrol *kcontrol,
diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c
index 0bcf36ee29bc..2c5712dce1d5 100644
--- a/sound/soc/codecs/wm8960.c
+++ b/sound/soc/codecs/wm8960.c
@@ -137,7 +137,8 @@ static int wm8960_get_deemph(struct snd_kcontrol *kcontrol,
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec);
- return wm8960->deemph;
+ ucontrol->value.enumerated.item[0] = wm8960->deemph;
+ return 0;
}
static int wm8960_put_deemph(struct snd_kcontrol *kcontrol,
diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c
index cf4b2722648a..f0c9d2691842 100644
--- a/sound/soc/codecs/wm8962.c
+++ b/sound/soc/codecs/wm8962.c
@@ -3339,7 +3339,7 @@ static irqreturn_t wm8962_irq(int irq, void *data)
int mask;
int active;
- mask = snd_soc_read(codec, WM8962_INTERRUPT_STATUS_2);
+ mask = snd_soc_read(codec, WM8962_INTERRUPT_STATUS_2_MASK);
active = snd_soc_read(codec, WM8962_INTERRUPT_STATUS_2);
active &= ~mask;
diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c
index 77b1d441c3d0..15f34a26debf 100644
--- a/sound/soc/codecs/wm8993.c
+++ b/sound/soc/codecs/wm8993.c
@@ -1030,6 +1030,12 @@ static int wm8993_set_bias_level(struct snd_soc_codec *codec,
WM8993_VMID_SEL_MASK | WM8993_BIAS_ENA,
0);
+ snd_soc_update_bits(codec, WM8993_ANTIPOP2,
+ WM8993_STARTUP_BIAS_ENA |
+ WM8993_VMID_BUF_ENA |
+ WM8993_VMID_RAMP_MASK |
+ WM8993_BIAS_SRC, 0);
+
#ifdef CONFIG_REGULATOR
/* Post 2.6.34 we will be able to get a callback when
* the regulators are disabled which we can use but
diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index bfd4cf4fde5f..af104acd75f8 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -92,6 +92,11 @@ struct wm8994_priv {
int retune_mobile_cfg[WM8994_NUM_EQ];
struct soc_enum retune_mobile_enum;
+ /* Platform dependant MBC configuration */
+ int mbc_cfg;
+ const char **mbc_texts;
+ struct soc_enum mbc_enum;
+
struct wm8994_micdet micdet[2];
wm8958_micdet_cb jack_cb;
@@ -305,6 +310,19 @@ static const char *sidetone_hpf_text[] = {
static const struct soc_enum sidetone_hpf =
SOC_ENUM_SINGLE(WM8994_SIDETONE, 7, 7, sidetone_hpf_text);
+static const char *adc_hpf_text[] = {
+ "HiFi", "Voice 1", "Voice 2", "Voice 3"
+};
+
+static const struct soc_enum aif1adc1_hpf =
+ SOC_ENUM_SINGLE(WM8994_AIF1_ADC1_FILTERS, 13, 4, adc_hpf_text);
+
+static const struct soc_enum aif1adc2_hpf =
+ SOC_ENUM_SINGLE(WM8994_AIF1_ADC2_FILTERS, 13, 4, adc_hpf_text);
+
+static const struct soc_enum aif2adc_hpf =
+ SOC_ENUM_SINGLE(WM8994_AIF2_ADC_FILTERS, 13, 4, adc_hpf_text);
+
static const DECLARE_TLV_DB_SCALE(aif_tlv, 0, 600, 0);
static const DECLARE_TLV_DB_SCALE(digital_tlv, -7200, 75, 1);
static const DECLARE_TLV_DB_SCALE(st_tlv, -3600, 300, 0);
@@ -540,11 +558,22 @@ static const struct soc_enum aif2dacl_src =
static const struct soc_enum aif2dacr_src =
SOC_ENUM_SINGLE(WM8994_AIF2_CONTROL_2, 14, 2, aif_chan_src_text);
+static const char *osr_text[] = {
+ "Low Power", "High Performance",
+};
+
+static const struct soc_enum dac_osr =
+ SOC_ENUM_SINGLE(WM8994_OVERSAMPLING, 0, 2, osr_text);
+
+static const struct soc_enum adc_osr =
+ SOC_ENUM_SINGLE(WM8994_OVERSAMPLING, 1, 2, osr_text);
+
static void wm8958_mbc_apply(struct snd_soc_codec *codec, int mbc, int start)
{
struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
+ struct wm8994_pdata *pdata = wm8994->pdata;
int pwr_reg = snd_soc_read(codec, WM8994_POWER_MANAGEMENT_5);
- int ena, reg, aif;
+ int ena, reg, aif, i;
switch (mbc) {
case 0:
@@ -587,7 +616,20 @@ static void wm8958_mbc_apply(struct snd_soc_codec *codec, int mbc, int start)
snd_soc_update_bits(codec, WM8958_DSP2_PROGRAM,
WM8958_DSP2_ENA, WM8958_DSP2_ENA);
- /* TODO: Apply any user specified MBC settings */
+ /* If we've got user supplied MBC settings use them */
+ if (pdata && pdata->num_mbc_cfgs) {
+ struct wm8958_mbc_cfg *cfg
+ = &pdata->mbc_cfgs[wm8994->mbc_cfg];
+
+ for (i = 0; i < ARRAY_SIZE(cfg->coeff_regs); i++)
+ snd_soc_write(codec, i + WM8958_MBC_BAND_1_K_1,
+ cfg->coeff_regs[i]);
+
+ for (i = 0; i < ARRAY_SIZE(cfg->cutoff_regs); i++)
+ snd_soc_write(codec,
+ i + WM8958_MBC_BAND_2_LOWER_CUTOFF_C1_1,
+ cfg->cutoff_regs[i]);
+ }
/* Run the DSP */
snd_soc_write(codec, WM8958_DSP2_EXECCONTROL,
@@ -648,6 +690,39 @@ static int wm8958_aif_ev(struct snd_soc_dapm_widget *w,
return 0;
}
+static int wm8958_put_mbc_enum(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
+ struct wm8994_pdata *pdata = wm8994->pdata;
+ int value = ucontrol->value.integer.value[0];
+ int reg;
+
+ /* Don't allow on the fly reconfiguration */
+ reg = snd_soc_read(codec, WM8994_CLOCKING_1);
+ if (reg < 0 || reg & WM8958_DSP2CLK_ENA)
+ return -EBUSY;
+
+ if (value >= pdata->num_mbc_cfgs)
+ return -EINVAL;
+
+ wm8994->mbc_cfg = value;
+
+ return 0;
+}
+
+static int wm8958_get_mbc_enum(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
+
+ ucontrol->value.enumerated.item[0] = wm8994->mbc_cfg;
+
+ return 0;
+}
+
static int wm8958_mbc_info(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
@@ -715,13 +790,13 @@ SOC_DOUBLE_R_TLV("AIF2ADC Volume", WM8994_AIF2_ADC_LEFT_VOLUME,
SOC_ENUM("AIF1ADCL Source", aif1adcl_src),
SOC_ENUM("AIF1ADCR Source", aif1adcr_src),
-SOC_ENUM("AIF2ADCL Source", aif1adcl_src),
-SOC_ENUM("AIF2ADCR Source", aif1adcr_src),
+SOC_ENUM("AIF2ADCL Source", aif2adcl_src),
+SOC_ENUM("AIF2ADCR Source", aif2adcr_src),
SOC_ENUM("AIF1DACL Source", aif1dacl_src),
SOC_ENUM("AIF1DACR Source", aif1dacr_src),
-SOC_ENUM("AIF2DACL Source", aif1dacl_src),
-SOC_ENUM("AIF2DACR Source", aif1dacr_src),
+SOC_ENUM("AIF2DACL Source", aif2dacl_src),
+SOC_ENUM("AIF2DACR Source", aif2dacr_src),
SOC_DOUBLE_R_TLV("AIF1DAC1 Volume", WM8994_AIF1_DAC1_LEFT_VOLUME,
WM8994_AIF1_DAC1_RIGHT_VOLUME, 1, 96, 0, digital_tlv),
@@ -760,6 +835,18 @@ SOC_SINGLE_TLV("DAC2 Left Sidetone Volume", WM8994_DAC2_MIXER_VOLUMES,
SOC_ENUM("Sidetone HPF Mux", sidetone_hpf),
SOC_SINGLE("Sidetone HPF Switch", WM8994_SIDETONE, 6, 1, 0),
+SOC_ENUM("AIF1ADC1 HPF Mode", aif1adc1_hpf),
+SOC_DOUBLE("AIF1ADC1 HPF Switch", WM8994_AIF1_ADC1_FILTERS, 12, 11, 1, 0),
+
+SOC_ENUM("AIF1ADC2 HPF Mode", aif1adc2_hpf),
+SOC_DOUBLE("AIF1ADC2 HPF Switch", WM8994_AIF1_ADC2_FILTERS, 12, 11, 1, 0),
+
+SOC_ENUM("AIF2ADC HPF Mode", aif2adc_hpf),
+SOC_DOUBLE("AIF2ADC HPF Switch", WM8994_AIF2_ADC_FILTERS, 12, 11, 1, 0),
+
+SOC_ENUM("ADC OSR", adc_osr),
+SOC_ENUM("DAC OSR", dac_osr),
+
SOC_DOUBLE_R_TLV("DAC1 Volume", WM8994_DAC1_LEFT_VOLUME,
WM8994_DAC1_RIGHT_VOLUME, 1, 96, 0, digital_tlv),
SOC_DOUBLE_R("DAC1 Switch", WM8994_DAC1_LEFT_VOLUME,
@@ -1205,10 +1292,10 @@ SND_SOC_DAPM_AIF_OUT("AIF1ADC1R", "AIF1 Capture",
0, WM8994_POWER_MANAGEMENT_4, 8, 0),
SND_SOC_DAPM_AIF_IN_E("AIF1DAC1L", NULL, 0,
WM8994_POWER_MANAGEMENT_5, 9, 0, wm8958_aif_ev,
- SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
+ SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
SND_SOC_DAPM_AIF_IN_E("AIF1DAC1R", NULL, 0,
WM8994_POWER_MANAGEMENT_5, 8, 0, wm8958_aif_ev,
- SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
+ SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
SND_SOC_DAPM_AIF_OUT("AIF1ADC2L", "AIF1 Capture",
0, WM8994_POWER_MANAGEMENT_4, 11, 0),
@@ -1216,10 +1303,10 @@ SND_SOC_DAPM_AIF_OUT("AIF1ADC2R", "AIF1 Capture",
0, WM8994_POWER_MANAGEMENT_4, 10, 0),
SND_SOC_DAPM_AIF_IN_E("AIF1DAC2L", NULL, 0,
WM8994_POWER_MANAGEMENT_5, 11, 0, wm8958_aif_ev,
- SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
+ SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
SND_SOC_DAPM_AIF_IN_E("AIF1DAC2R", NULL, 0,
WM8994_POWER_MANAGEMENT_5, 10, 0, wm8958_aif_ev,
- SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
+ SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
SND_SOC_DAPM_MIXER("AIF1ADC1L Mixer", SND_SOC_NOPM, 0, 0,
aif1adc1l_mix, ARRAY_SIZE(aif1adc1l_mix)),
@@ -1625,6 +1712,7 @@ static int _wm8994_set_fll(struct snd_soc_codec *codec, int id, int src,
/* Allow no source specification when stopping */
if (freq_out)
return -EINVAL;
+ src = wm8994->fll[id].src;
break;
case WM8994_FLL_SRC_MCLK1:
case WM8994_FLL_SRC_MCLK2:
@@ -1806,15 +1894,33 @@ static int wm8994_set_bias_level(struct snd_soc_codec *codec,
if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
pm_runtime_get_sync(codec->dev);
- /* Tweak DC servo and DSP configuration for
- * improved performance. */
- if (control->type == WM8994 && wm8994->revision < 4) {
- /* Tweak DC servo and DSP configuration for
- * improved performance. */
- snd_soc_write(codec, 0x102, 0x3);
- snd_soc_write(codec, 0x56, 0x3);
- snd_soc_write(codec, 0x817, 0);
- snd_soc_write(codec, 0x102, 0);
+ switch (control->type) {
+ case WM8994:
+ if (wm8994->revision < 4) {
+ /* Tweak DC servo and DSP
+ * configuration for improved
+ * performance. */
+ snd_soc_write(codec, 0x102, 0x3);
+ snd_soc_write(codec, 0x56, 0x3);
+ snd_soc_write(codec, 0x817, 0);
+ snd_soc_write(codec, 0x102, 0);
+ }
+ break;
+
+ case WM8958:
+ if (wm8994->revision == 0) {
+ /* Optimise performance for rev A */
+ snd_soc_write(codec, 0x102, 0x3);
+ snd_soc_write(codec, 0xcb, 0x81);
+ snd_soc_write(codec, 0x817, 0);
+ snd_soc_write(codec, 0x102, 0);
+
+ snd_soc_update_bits(codec,
+ WM8958_CHARGE_PUMP_2,
+ WM8958_CP_DISCH,
+ WM8958_CP_DISCH);
+ }
+ break;
}
/* Discharge LINEOUT1 & 2 */
@@ -2028,10 +2134,12 @@ static int wm8994_hw_params(struct snd_pcm_substream *substream,
struct wm8994 *control = codec->control_data;
struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
int aif1_reg;
+ int aif2_reg;
int bclk_reg;
int lrclk_reg;
int rate_reg;
int aif1 = 0;
+ int aif2 = 0;
int bclk = 0;
int lrclk = 0;
int rate_val = 0;
@@ -2042,6 +2150,7 @@ static int wm8994_hw_params(struct snd_pcm_substream *substream,
switch (dai->id) {
case 1:
aif1_reg = WM8994_AIF1_CONTROL_1;
+ aif2_reg = WM8994_AIF1_CONTROL_2;
bclk_reg = WM8994_AIF1_BCLK;
rate_reg = WM8994_AIF1_RATE;
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK ||
@@ -2054,6 +2163,7 @@ static int wm8994_hw_params(struct snd_pcm_substream *substream,
break;
case 2:
aif1_reg = WM8994_AIF2_CONTROL_1;
+ aif2_reg = WM8994_AIF2_CONTROL_2;
bclk_reg = WM8994_AIF2_BCLK;
rate_reg = WM8994_AIF2_RATE;
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK ||
@@ -2109,6 +2219,10 @@ static int wm8994_hw_params(struct snd_pcm_substream *substream,
dev_dbg(dai->dev, "AIF%dCLK is %dHz, target BCLK %dHz\n",
dai->id, wm8994->aifclk[id], bclk_rate);
+ if (params_channels(params) == 1 &&
+ (snd_soc_read(codec, aif1_reg) & 0x18) == 0x18)
+ aif2 |= WM8994_AIF1_MONO;
+
if (wm8994->aifclk[id] == 0) {
dev_err(dai->dev, "AIF%dCLK not configured\n", dai->id);
return -EINVAL;
@@ -2152,6 +2266,7 @@ static int wm8994_hw_params(struct snd_pcm_substream *substream,
lrclk, bclk_rate / lrclk);
snd_soc_update_bits(codec, aif1_reg, WM8994_AIF1_WL_MASK, aif1);
+ snd_soc_update_bits(codec, aif2_reg, WM8994_AIF1_MONO, aif2);
snd_soc_update_bits(codec, bclk_reg, WM8994_AIF1_BCLK_DIV_MASK, bclk);
snd_soc_update_bits(codec, lrclk_reg, WM8994_AIF1DAC_RATE_MASK,
lrclk);
@@ -2307,14 +2422,14 @@ static struct snd_soc_dai_driver wm8994_dai[] = {
.id = 1,
.playback = {
.stream_name = "AIF1 Playback",
- .channels_min = 2,
+ .channels_min = 1,
.channels_max = 2,
.rates = WM8994_RATES,
.formats = WM8994_FORMATS,
},
.capture = {
.stream_name = "AIF1 Capture",
- .channels_min = 2,
+ .channels_min = 1,
.channels_max = 2,
.rates = WM8994_RATES,
.formats = WM8994_FORMATS,
@@ -2326,14 +2441,14 @@ static struct snd_soc_dai_driver wm8994_dai[] = {
.id = 2,
.playback = {
.stream_name = "AIF2 Playback",
- .channels_min = 2,
+ .channels_min = 1,
.channels_max = 2,
.rates = WM8994_RATES,
.formats = WM8994_FORMATS,
},
.capture = {
.stream_name = "AIF2 Capture",
- .channels_min = 2,
+ .channels_min = 1,
.channels_max = 2,
.rates = WM8994_RATES,
.formats = WM8994_FORMATS,
@@ -2345,14 +2460,14 @@ static struct snd_soc_dai_driver wm8994_dai[] = {
.id = 3,
.playback = {
.stream_name = "AIF3 Playback",
- .channels_min = 2,
+ .channels_min = 1,
.channels_max = 2,
.rates = WM8994_RATES,
.formats = WM8994_FORMATS,
},
.capture = {
.stream_name = "AIF3 Capture",
- .channels_min = 2,
+ .channels_min = 1,
.channels_max = 2,
.rates = WM8994_RATES,
.formats = WM8994_FORMATS,
@@ -2539,6 +2654,34 @@ static void wm8994_handle_pdata(struct wm8994_priv *wm8994)
dev_dbg(codec->dev, "%d ReTune Mobile configurations\n",
pdata->num_retune_mobile_cfgs);
+ if (pdata->num_mbc_cfgs) {
+ struct snd_kcontrol_new control[] = {
+ SOC_ENUM_EXT("MBC Mode", wm8994->mbc_enum,
+ wm8958_get_mbc_enum, wm8958_put_mbc_enum),
+ };
+
+ /* We need an array of texts for the enum API */
+ wm8994->mbc_texts = kmalloc(sizeof(char *)
+ * pdata->num_mbc_cfgs, GFP_KERNEL);
+ if (!wm8994->mbc_texts) {
+ dev_err(wm8994->codec->dev,
+ "Failed to allocate %d MBC config texts\n",
+ pdata->num_mbc_cfgs);
+ return;
+ }
+
+ for (i = 0; i < pdata->num_mbc_cfgs; i++)
+ wm8994->mbc_texts[i] = pdata->mbc_cfgs[i].name;
+
+ wm8994->mbc_enum.max = pdata->num_mbc_cfgs;
+ wm8994->mbc_enum.texts = wm8994->mbc_texts;
+
+ ret = snd_soc_add_controls(wm8994->codec, control, 1);
+ if (ret != 0)
+ dev_err(wm8994->codec->dev,
+ "Failed to add MBC mode controls: %d\n", ret);
+ }
+
if (pdata->num_retune_mobile_cfgs)
wm8994_handle_retune_mobile_pdata(wm8994);
else
diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c
index c7060775e88b..e5055b28c638 100644
--- a/sound/soc/codecs/wm9081.c
+++ b/sound/soc/codecs/wm9081.c
@@ -590,6 +590,10 @@ static int wm9081_set_fll(struct snd_soc_codec *codec, int fll_id,
reg5 |= fll_div.fll_clk_ref_div << WM9081_FLL_CLK_REF_DIV_SHIFT;
snd_soc_write(codec, WM9081_FLL_CONTROL_5, reg5);
+ /* Set gain to the recommended value */
+ snd_soc_update_bits(codec, WM9081_FLL_CONTROL_4,
+ WM9081_FLL_GAIN_MASK, 0);
+
/* Enable the FLL */
snd_soc_write(codec, WM9081_FLL_CONTROL_1, reg1 | WM9081_FLL_ENA);
diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c
index b24ba9fa7ef7..c466982eed23 100644
--- a/sound/soc/codecs/wm_hubs.c
+++ b/sound/soc/codecs/wm_hubs.c
@@ -323,7 +323,7 @@ SOC_DOUBLE_R("Speaker Switch",
SOC_DOUBLE_R("Speaker ZC Switch",
WM8993_SPEAKER_VOLUME_LEFT, WM8993_SPEAKER_VOLUME_RIGHT,
7, 1, 0),
-SOC_DOUBLE_TLV("Speaker Boost Volume", WM8993_SPKOUT_BOOST, 0, 3, 7, 0,
+SOC_DOUBLE_TLV("Speaker Boost Volume", WM8993_SPKOUT_BOOST, 3, 0, 7, 0,
spkboost_tlv),
SOC_ENUM("Speaker Reference", speaker_ref),
SOC_ENUM("Speaker Mode", speaker_mode),