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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
-rw-r--r--sound/soc/ep93xx/ep93xx-i2s.c4
-rw-r--r--sound/soc/ep93xx/ep93xx-pcm.c4
-rw-r--r--sound/soc/nuc900/nuc900-pcm.c9
-rw-r--r--sound/soc/omap/Kconfig2
-rw-r--r--sound/soc/s6000/s6000-i2s.c2
-rw-r--r--sound/soc/s6000/s6000-pcm.c2
-rw-r--r--sound/soc/s6000/s6105-ipcam.c2
-rw-r--r--sound/soc/samsung/Kconfig8
-rw-r--r--sound/soc/samsung/Makefile2
-rw-r--r--sound/soc/samsung/h1940_uda1380.c296
-rw-r--r--sound/soc/samsung/smdk_spdif.c4
-rw-r--r--sound/soc/sh/fsi-ak4642.c140
-rw-r--r--sound/soc/sh/fsi.c230
-rw-r--r--sound/soc/soc-cache.c90
-rw-r--r--sound/soc/soc-core.c446
-rw-r--r--sound/soc/soc-dapm.c13
-rw-r--r--sound/soc/soc-jack.c7
33 files changed, 1336 insertions, 491 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),
diff --git a/sound/soc/ep93xx/ep93xx-i2s.c b/sound/soc/ep93xx/ep93xx-i2s.c
index 4f4873359613..9ac93f6b4f85 100644
--- a/sound/soc/ep93xx/ep93xx-i2s.c
+++ b/sound/soc/ep93xx/ep93xx-i2s.c
@@ -352,13 +352,13 @@ static struct snd_soc_dai_driver ep93xx_i2s_dai = {
.playback = {
.channels_min = 2,
.channels_max = 2,
- .rates = SNDRV_PCM_RATE_8000_48000,
+ .rates = SNDRV_PCM_RATE_8000_96000,
.formats = EP93XX_I2S_FORMATS,
},
.capture = {
.channels_min = 2,
.channels_max = 2,
- .rates = SNDRV_PCM_RATE_8000_48000,
+ .rates = SNDRV_PCM_RATE_8000_96000,
.formats = EP93XX_I2S_FORMATS,
},
.ops = &ep93xx_i2s_dai_ops,
diff --git a/sound/soc/ep93xx/ep93xx-pcm.c b/sound/soc/ep93xx/ep93xx-pcm.c
index 2f121ddbe4bb..06670776f649 100644
--- a/sound/soc/ep93xx/ep93xx-pcm.c
+++ b/sound/soc/ep93xx/ep93xx-pcm.c
@@ -35,9 +35,9 @@ static const struct snd_pcm_hardware ep93xx_pcm_hardware = {
SNDRV_PCM_INFO_INTERLEAVED |
SNDRV_PCM_INFO_BLOCK_TRANSFER),
- .rates = SNDRV_PCM_RATE_8000_48000,
+ .rates = SNDRV_PCM_RATE_8000_96000,
.rate_min = SNDRV_PCM_RATE_8000,
- .rate_max = SNDRV_PCM_RATE_48000,
+ .rate_max = SNDRV_PCM_RATE_96000,
.formats = (SNDRV_PCM_FMTBIT_S16_LE |
SNDRV_PCM_FMTBIT_S24_LE |
diff --git a/sound/soc/nuc900/nuc900-pcm.c b/sound/soc/nuc900/nuc900-pcm.c
index 2245f8b8edc1..8263f56dc665 100644
--- a/sound/soc/nuc900/nuc900-pcm.c
+++ b/sound/soc/nuc900/nuc900-pcm.c
@@ -50,12 +50,12 @@ static int nuc900_dma_hw_params(struct snd_pcm_substream *substream,
unsigned long flags;
int ret = 0;
- spin_lock_irqsave(&nuc900_audio->lock, flags);
-
ret = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params));
if (ret < 0)
return ret;
+ spin_lock_irqsave(&nuc900_audio->lock, flags);
+
nuc900_audio->substream = substream;
nuc900_audio->dma_addr[substream->stream] = runtime->dma_addr;
nuc900_audio->buffersize[substream->stream] =
@@ -169,6 +169,7 @@ static int nuc900_dma_prepare(struct snd_pcm_substream *substream)
struct snd_pcm_runtime *runtime = substream->runtime;
struct nuc900_audio *nuc900_audio = runtime->private_data;
unsigned long flags, val;
+ int ret = 0;
spin_lock_irqsave(&nuc900_audio->lock, flags);
@@ -197,10 +198,10 @@ static int nuc900_dma_prepare(struct snd_pcm_substream *substream)
AUDIO_WRITE(nuc900_audio->mmio + ACTL_RESET, val);
break;
default:
- return -EINVAL;
+ ret = -EINVAL;
}
spin_unlock_irqrestore(&nuc900_audio->lock, flags);
- return 0;
+ return ret;
}
static int nuc900_dma_trigger(struct snd_pcm_substream *substream, int cmd)
diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig
index d542ea2ff6be..a088db6d5091 100644
--- a/sound/soc/omap/Kconfig
+++ b/sound/soc/omap/Kconfig
@@ -12,8 +12,8 @@ config SND_OMAP_SOC_MCPDM
config SND_OMAP_SOC_N810
tristate "SoC Audio support for Nokia N810"
depends on SND_OMAP_SOC && MACH_NOKIA_N810 && I2C
+ depends on OMAP_MUX
select SND_OMAP_SOC_MCBSP
- select OMAP_MUX
select SND_SOC_TLV320AIC3X
help
Say Y if you want to add support for SoC audio on Nokia N810.
diff --git a/sound/soc/s6000/s6000-i2s.c b/sound/soc/s6000/s6000-i2s.c
index 8778faa174a6..3052f64b2403 100644
--- a/sound/soc/s6000/s6000-i2s.c
+++ b/sound/soc/s6000/s6000-i2s.c
@@ -434,7 +434,7 @@ static struct snd_soc_dai_driver s6000_i2s_dai = {
.rate_max = 1562500,
},
.ops = &s6000_i2s_dai_ops,
-}
+};
static int __devinit s6000_i2s_probe(struct platform_device *pdev)
{
diff --git a/sound/soc/s6000/s6000-pcm.c b/sound/soc/s6000/s6000-pcm.c
index 271fd222bf19..ab3ccaec72d2 100644
--- a/sound/soc/s6000/s6000-pcm.c
+++ b/sound/soc/s6000/s6000-pcm.c
@@ -473,7 +473,7 @@ static int s6000_pcm_new(struct snd_card *card,
}
res = request_irq(params->irq, s6000_pcm_irq, IRQF_SHARED,
- s6000_soc_platform.name, pcm);
+ "s6000-audio", pcm);
if (res) {
printk(KERN_ERR "s6000-pcm couldn't get IRQ\n");
return res;
diff --git a/sound/soc/s6000/s6105-ipcam.c b/sound/soc/s6000/s6105-ipcam.c
index 2e977e4d65e7..bf3f3f9165e4 100644
--- a/sound/soc/s6000/s6105-ipcam.c
+++ b/sound/soc/s6000/s6105-ipcam.c
@@ -168,7 +168,7 @@ static int s6105_aic3x_init(struct snd_soc_pcm_runtime *rtd)
snd_soc_dapm_sync(dapm);
- snd_ctl_add(codec->snd_card, snd_ctl_new1(&audio_out_mux, codec));
+ snd_ctl_add(codec->card->snd_card, snd_ctl_new1(&audio_out_mux, codec));
return 0;
}
diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig
index eb45cf90d4e2..67cdad4ee535 100644
--- a/sound/soc/samsung/Kconfig
+++ b/sound/soc/samsung/Kconfig
@@ -116,6 +116,14 @@ config ASOC_SAMSUNG_SIMTEC_HERMES
select SND_SOC_TLV320AIC3X
select ASOC_SAMSUNG_SIMTEC
+config ASOC_SAMSUNG_H1940_UDA1380
+ tristate "Audio support for the HP iPAQ H1940"
+ depends on ASOC_SAMSUNG && ARCH_H1940
+ select SND_S3C24XX_I2S
+ select SND_SOC_UDA1380
+ help
+ This driver provides audio support for HP iPAQ h1940 PDA.
+
config ASOC_SAMSUNG_RX1950_UDA1380
tristate "Audio support for the HP iPAQ RX1950"
depends on ASOC_SAMSUNG && MACH_RX1950
diff --git a/sound/soc/samsung/Makefile b/sound/soc/samsung/Makefile
index 0d24f95c8b1c..622e76eb9775 100644
--- a/sound/soc/samsung/Makefile
+++ b/sound/soc/samsung/Makefile
@@ -27,6 +27,7 @@ snd-soc-s3c24xx-uda134x-objs := s3c24xx_uda134x.o
snd-soc-s3c24xx-simtec-objs := s3c24xx_simtec.o
snd-soc-s3c24xx-simtec-hermes-objs := s3c24xx_simtec_hermes.o
snd-soc-s3c24xx-simtec-tlv320aic23-objs := s3c24xx_simtec_tlv320aic23.o
+snd-soc-h1940-uda1380-objs := h1940_uda1380.o
snd-soc-rx1950-uda1380-objs := rx1950_uda1380.o
snd-soc-smdk-wm8580-objs := smdk_wm8580.o
snd-soc-smdk-wm9713-objs := smdk_wm9713.o
@@ -43,6 +44,7 @@ obj-$(CONFIG_ASOC_SAMSUNG_S3C24XX_UDA134X) += snd-soc-s3c24xx-uda134x.o
obj-$(CONFIG_ASOC_SAMSUNG_SIMTEC) += snd-soc-s3c24xx-simtec.o
obj-$(CONFIG_ASOC_SAMSUNG_SIMTEC_HERMES) += snd-soc-s3c24xx-simtec-hermes.o
obj-$(CONFIG_ASOC_SAMSUNG_SIMTEC_TLV320AIC23) += snd-soc-s3c24xx-simtec-tlv320aic23.o
+obj-$(CONFIG_ASOC_SAMSUNG_H1940_UDA1380) += snd-soc-h1940-uda1380.o
obj-$(CONFIG_ASOC_SAMSUNG_RX1950_UDA1380) += snd-soc-rx1950-uda1380.o
obj-$(CONFIG_ASOC_SAMSUNG_SMDK_WM8580) += snd-soc-smdk-wm8580.o
obj-$(CONFIG_ASOC_SAMSUNG_SMDK_WM9713) += snd-soc-smdk-wm9713.o
diff --git a/sound/soc/samsung/h1940_uda1380.c b/sound/soc/samsung/h1940_uda1380.c
new file mode 100644
index 000000000000..c45f7ce14d61
--- /dev/null
+++ b/sound/soc/samsung/h1940_uda1380.c
@@ -0,0 +1,296 @@
+/*
+ * h1940-uda1380.c -- ALSA Soc Audio Layer
+ *
+ * Copyright (c) 2010 Arnaud Patard <arnaud.patard@rtp-net.org>
+ * Copyright (c) 2010 Vasily Khoruzhick <anarsoul@gmail.com>
+ *
+ * Based on version from Arnaud Patard <arnaud.patard@rtp-net.org>
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License as published by the
+ * Free Software Foundation; either version 2 of the License, or (at your
+ * option) any later version.
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/platform_device.h>
+#include <linux/i2c.h>
+#include <linux/gpio.h>
+
+#include <sound/soc.h>
+#include <sound/uda1380.h>
+#include <sound/jack.h>
+
+#include <plat/regs-iis.h>
+
+#include <mach/h1940-latch.h>
+
+#include <asm/mach-types.h>
+
+#include "dma.h"
+#include "s3c24xx-i2s.h"
+#include "../codecs/uda1380.h"
+
+static unsigned int rates[] = {
+ 11025,
+ 22050,
+ 44100,
+};
+
+static struct snd_pcm_hw_constraint_list hw_rates = {
+ .count = ARRAY_SIZE(rates),
+ .list = rates,
+ .mask = 0,
+};
+
+static struct snd_soc_jack hp_jack;
+
+static struct snd_soc_jack_pin hp_jack_pins[] = {
+ {
+ .pin = "Headphone Jack",
+ .mask = SND_JACK_HEADPHONE,
+ },
+ {
+ .pin = "Speaker",
+ .mask = SND_JACK_HEADPHONE,
+ .invert = 1,
+ },
+};
+
+static struct snd_soc_jack_gpio hp_jack_gpios[] = {
+ {
+ .gpio = S3C2410_GPG(4),
+ .name = "hp-gpio",
+ .report = SND_JACK_HEADPHONE,
+ .invert = 1,
+ .debounce_time = 200,
+ },
+};
+
+static int h1940_startup(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+
+ runtime->hw.rate_min = hw_rates.list[0];
+ runtime->hw.rate_max = hw_rates.list[hw_rates.count - 1];
+ runtime->hw.rates = SNDRV_PCM_RATE_KNOT;
+
+ return snd_pcm_hw_constraint_list(runtime, 0,
+ SNDRV_PCM_HW_PARAM_RATE,
+ &hw_rates);
+}
+
+static int h1940_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ int div;
+ int ret;
+ unsigned int rate = params_rate(params);
+
+ switch (rate) {
+ case 11025:
+ case 22050:
+ case 44100:
+ div = s3c24xx_i2s_get_clockrate() / (384 * rate);
+ if (s3c24xx_i2s_get_clockrate() % (384 * rate) > (192 * rate))
+ div++;
+ break;
+ default:
+ dev_err(&rtd->dev, "%s: rate %d is not supported\n",
+ __func__, rate);
+ return -EINVAL;
+ }
+
+ /* set codec DAI configuration */
+ ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
+ SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
+ if (ret < 0)
+ return ret;
+
+ /* set cpu DAI configuration */
+ ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
+ SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
+ if (ret < 0)
+ return ret;
+
+ /* select clock source */
+ ret = snd_soc_dai_set_sysclk(cpu_dai, S3C24XX_CLKSRC_PCLK, rate,
+ SND_SOC_CLOCK_OUT);
+ if (ret < 0)
+ return ret;
+
+ /* set MCLK division for sample rate */
+ ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_MCLK,
+ S3C2410_IISMOD_384FS);
+ if (ret < 0)
+ return ret;
+
+ /* set BCLK division for sample rate */
+ ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_BCLK,
+ S3C2410_IISMOD_32FS);
+ if (ret < 0)
+ return ret;
+
+ /* set prescaler division for sample rate */
+ ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_PRESCALER,
+ S3C24XX_PRESCALE(div, div));
+ if (ret < 0)
+ return ret;
+
+ return 0;
+}
+
+static struct snd_soc_ops h1940_ops = {
+ .startup = h1940_startup,
+ .hw_params = h1940_hw_params,
+};
+
+static int h1940_spk_power(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ if (SND_SOC_DAPM_EVENT_ON(event))
+ gpio_set_value(H1940_LATCH_AUDIO_POWER, 1);
+ else
+ gpio_set_value(H1940_LATCH_AUDIO_POWER, 0);
+
+ return 0;
+}
+
+/* h1940 machine dapm widgets */
+static const struct snd_soc_dapm_widget uda1380_dapm_widgets[] = {
+ SND_SOC_DAPM_HP("Headphone Jack", NULL),
+ SND_SOC_DAPM_MIC("Mic Jack", NULL),
+ SND_SOC_DAPM_SPK("Speaker", h1940_spk_power),
+};
+
+/* h1940 machine audio_map */
+static const struct snd_soc_dapm_route audio_map[] = {
+ /* headphone connected to VOUTLHP, VOUTRHP */
+ {"Headphone Jack", NULL, "VOUTLHP"},
+ {"Headphone Jack", NULL, "VOUTRHP"},
+
+ /* ext speaker connected to VOUTL, VOUTR */
+ {"Speaker", NULL, "VOUTL"},
+ {"Speaker", NULL, "VOUTR"},
+
+ /* mic is connected to VINM */
+ {"VINM", NULL, "Mic Jack"},
+};
+
+static struct platform_device *s3c24xx_snd_device;
+
+static int h1940_uda1380_init(struct snd_soc_pcm_runtime *rtd)
+{
+ struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
+ int err;
+
+ /* Add h1940 specific widgets */
+ err = snd_soc_dapm_new_controls(dapm, uda1380_dapm_widgets,
+ ARRAY_SIZE(uda1380_dapm_widgets));
+ if (err)
+ return err;
+
+ /* Set up h1940 specific audio path audio_mapnects */
+ err = snd_soc_dapm_add_routes(dapm, audio_map,
+ ARRAY_SIZE(audio_map));
+ if (err)
+ return err;
+
+ snd_soc_dapm_enable_pin(dapm, "Headphone Jack");
+ snd_soc_dapm_enable_pin(dapm, "Speaker");
+ snd_soc_dapm_enable_pin(dapm, "Mic Jack");
+
+ snd_soc_dapm_sync(dapm);
+
+ snd_soc_jack_new(codec, "Headphone Jack", SND_JACK_HEADPHONE,
+ &hp_jack);
+
+ snd_soc_jack_add_pins(&hp_jack, ARRAY_SIZE(hp_jack_pins),
+ hp_jack_pins);
+
+ snd_soc_jack_add_gpios(&hp_jack, ARRAY_SIZE(hp_jack_gpios),
+ hp_jack_gpios);
+
+ return 0;
+}
+
+/* s3c24xx digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link h1940_uda1380_dai[] = {
+ {
+ .name = "uda1380",
+ .stream_name = "UDA1380 Duplex",
+ .cpu_dai_name = "s3c24xx-iis",
+ .codec_dai_name = "uda1380-hifi",
+ .init = h1940_uda1380_init,
+ .platform_name = "samsung-audio",
+ .codec_name = "uda1380-codec.0-001a",
+ .ops = &h1940_ops,
+ },
+};
+
+static struct snd_soc_card h1940_asoc = {
+ .name = "h1940",
+ .dai_link = h1940_uda1380_dai,
+ .num_links = ARRAY_SIZE(h1940_uda1380_dai),
+};
+
+static int __init h1940_init(void)
+{
+ int ret;
+
+ if (!machine_is_h1940())
+ return -ENODEV;
+
+ /* configure some gpios */
+ ret = gpio_request(H1940_LATCH_AUDIO_POWER, "speaker-power");
+ if (ret)
+ goto err_out;
+
+ ret = gpio_direction_output(H1940_LATCH_AUDIO_POWER, 0);
+ if (ret)
+ goto err_gpio;
+
+ s3c24xx_snd_device = platform_device_alloc("soc-audio", -1);
+ if (!s3c24xx_snd_device) {
+ ret = -ENOMEM;
+ goto err_gpio;
+ }
+
+ platform_set_drvdata(s3c24xx_snd_device, &h1940_asoc);
+ ret = platform_device_add(s3c24xx_snd_device);
+
+ if (ret)
+ goto err_plat;
+
+ return 0;
+
+err_plat:
+ platform_device_put(s3c24xx_snd_device);
+err_gpio:
+ gpio_free(H1940_LATCH_AUDIO_POWER);
+
+err_out:
+ return ret;
+}
+
+static void __exit h1940_exit(void)
+{
+ platform_device_unregister(s3c24xx_snd_device);
+ snd_soc_jack_free_gpios(&hp_jack, ARRAY_SIZE(hp_jack_gpios),
+ hp_jack_gpios);
+ gpio_free(H1940_LATCH_AUDIO_POWER);
+}
+
+module_init(h1940_init);
+module_exit(h1940_exit);
+
+/* Module information */
+MODULE_AUTHOR("Arnaud Patard, Vasily Khoruzhick");
+MODULE_DESCRIPTION("ALSA SoC H1940");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/samsung/smdk_spdif.c b/sound/soc/samsung/smdk_spdif.c
index ca779ad1613c..cb2f4d04ab7e 100644
--- a/sound/soc/samsung/smdk_spdif.c
+++ b/sound/soc/samsung/smdk_spdif.c
@@ -28,7 +28,7 @@
static int set_audio_clock_heirachy(struct platform_device *pdev)
{
struct clk *fout_epll, *mout_epll, *sclk_audio0, *sclk_spdif;
- int ret;
+ int ret = 0;
fout_epll = clk_get(NULL, "fout_epll");
if (IS_ERR(fout_epll)) {
@@ -152,8 +152,6 @@ static struct snd_soc_ops smdk_spdif_ops = {
.hw_params = smdk_hw_params,
};
-static struct snd_soc_card smdk;
-
static struct snd_soc_dai_link smdk_dai = {
.name = "S/PDIF",
.stream_name = "S/PDIF PCM Playback",
diff --git a/sound/soc/sh/fsi-ak4642.c b/sound/soc/sh/fsi-ak4642.c
index d96602de71de..a14820ac9665 100644
--- a/sound/soc/sh/fsi-ak4642.c
+++ b/sound/soc/sh/fsi-ak4642.c
@@ -12,6 +12,14 @@
#include <linux/platform_device.h>
#include <sound/sh_fsi.h>
+struct fsi_ak4642_data {
+ const char *name;
+ const char *card;
+ const char *cpu_dai;
+ const char *codec;
+ const char *platform;
+};
+
static int fsi_ak4642_dai_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_dai *dai = rtd->codec_dai;
@@ -27,37 +35,42 @@ static int fsi_ak4642_dai_init(struct snd_soc_pcm_runtime *rtd)
}
static struct snd_soc_dai_link fsi_dai_link = {
- .name = "AK4642",
- .stream_name = "AK4642",
- .cpu_dai_name = "fsia-dai", /* fsi A */
.codec_dai_name = "ak4642-hifi",
-#ifdef CONFIG_MACH_AP4EVB
- .platform_name = "sh_fsi2",
- .codec_name = "ak4642-codec.0-0013",
-#else
- .platform_name = "sh_fsi.0",
- .codec_name = "ak4642-codec.0-0012",
-#endif
.init = fsi_ak4642_dai_init,
- .ops = NULL,
};
static struct snd_soc_card fsi_soc_card = {
- .name = "FSI (AK4642)",
.dai_link = &fsi_dai_link,
.num_links = 1,
};
static struct platform_device *fsi_snd_device;
-static int __init fsi_ak4642_init(void)
+static int fsi_ak4642_probe(struct platform_device *pdev)
{
int ret = -ENOMEM;
+ const struct platform_device_id *id_entry;
+ struct fsi_ak4642_data *pdata;
+
+ id_entry = pdev->id_entry;
+ if (!id_entry) {
+ dev_err(&pdev->dev, "unknown fsi ak4642\n");
+ return -ENODEV;
+ }
+
+ pdata = (struct fsi_ak4642_data *)id_entry->driver_data;
fsi_snd_device = platform_device_alloc("soc-audio", FSI_PORT_A);
if (!fsi_snd_device)
goto out;
+ fsi_dai_link.name = pdata->name;
+ fsi_dai_link.stream_name = pdata->name;
+ fsi_dai_link.cpu_dai_name = pdata->cpu_dai;
+ fsi_dai_link.platform_name = pdata->platform;
+ fsi_dai_link.codec_name = pdata->codec;
+ fsi_soc_card.name = pdata->card;
+
platform_set_drvdata(fsi_snd_device, &fsi_soc_card);
ret = platform_device_add(fsi_snd_device);
@@ -68,9 +81,108 @@ out:
return ret;
}
-static void __exit fsi_ak4642_exit(void)
+static int fsi_ak4642_remove(struct platform_device *pdev)
{
platform_device_unregister(fsi_snd_device);
+ return 0;
+}
+
+static struct fsi_ak4642_data fsi_a_ak4642 = {
+ .name = "AK4642",
+ .card = "FSIA (AK4642)",
+ .cpu_dai = "fsia-dai",
+ .codec = "ak4642-codec.0-0012",
+ .platform = "sh_fsi.0",
+};
+
+static struct fsi_ak4642_data fsi_b_ak4642 = {
+ .name = "AK4642",
+ .card = "FSIB (AK4642)",
+ .cpu_dai = "fsib-dai",
+ .codec = "ak4642-codec.0-0012",
+ .platform = "sh_fsi.0",
+};
+
+static struct fsi_ak4642_data fsi_a_ak4643 = {
+ .name = "AK4643",
+ .card = "FSIA (AK4643)",
+ .cpu_dai = "fsia-dai",
+ .codec = "ak4642-codec.0-0013",
+ .platform = "sh_fsi.0",
+};
+
+static struct fsi_ak4642_data fsi_b_ak4643 = {
+ .name = "AK4643",
+ .card = "FSIB (AK4643)",
+ .cpu_dai = "fsib-dai",
+ .codec = "ak4642-codec.0-0013",
+ .platform = "sh_fsi.0",
+};
+
+static struct fsi_ak4642_data fsi2_a_ak4642 = {
+ .name = "AK4642",
+ .card = "FSI2A (AK4642)",
+ .cpu_dai = "fsia-dai",
+ .codec = "ak4642-codec.0-0012",
+ .platform = "sh_fsi2",
+};
+
+static struct fsi_ak4642_data fsi2_b_ak4642 = {
+ .name = "AK4642",
+ .card = "FSI2B (AK4642)",
+ .cpu_dai = "fsib-dai",
+ .codec = "ak4642-codec.0-0012",
+ .platform = "sh_fsi2",
+};
+
+static struct fsi_ak4642_data fsi2_a_ak4643 = {
+ .name = "AK4643",
+ .card = "FSI2A (AK4643)",
+ .cpu_dai = "fsia-dai",
+ .codec = "ak4642-codec.0-0013",
+ .platform = "sh_fsi2",
+};
+
+static struct fsi_ak4642_data fsi2_b_ak4643 = {
+ .name = "AK4643",
+ .card = "FSI2B (AK4643)",
+ .cpu_dai = "fsib-dai",
+ .codec = "ak4642-codec.0-0013",
+ .platform = "sh_fsi2",
+};
+
+static struct platform_device_id fsi_id_table[] = {
+ /* FSI */
+ { "sh_fsi_a_ak4642", (kernel_ulong_t)&fsi_a_ak4642 },
+ { "sh_fsi_b_ak4642", (kernel_ulong_t)&fsi_b_ak4642 },
+ { "sh_fsi_a_ak4643", (kernel_ulong_t)&fsi_a_ak4643 },
+ { "sh_fsi_b_ak4643", (kernel_ulong_t)&fsi_b_ak4643 },
+
+ /* FSI 2 */
+ { "sh_fsi2_a_ak4642", (kernel_ulong_t)&fsi2_a_ak4642 },
+ { "sh_fsi2_b_ak4642", (kernel_ulong_t)&fsi2_b_ak4642 },
+ { "sh_fsi2_a_ak4643", (kernel_ulong_t)&fsi2_a_ak4643 },
+ { "sh_fsi2_b_ak4643", (kernel_ulong_t)&fsi2_b_ak4643 },
+ {},
+};
+
+static struct platform_driver fsi_ak4642 = {
+ .driver = {
+ .name = "fsi-ak4642-audio",
+ },
+ .probe = fsi_ak4642_probe,
+ .remove = fsi_ak4642_remove,
+ .id_table = fsi_id_table,
+};
+
+static int __init fsi_ak4642_init(void)
+{
+ return platform_driver_register(&fsi_ak4642);
+}
+
+static void __exit fsi_ak4642_exit(void)
+{
+ platform_driver_unregister(&fsi_ak4642);
}
module_init(fsi_ak4642_init);
diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c
index 507e709f2807..97c5394aa7d7 100644
--- a/sound/soc/sh/fsi.c
+++ b/sound/soc/sh/fsi.c
@@ -19,20 +19,26 @@
#include <sound/soc.h>
#include <sound/sh_fsi.h>
-#define DO_FMT 0x0000
-#define DOFF_CTL 0x0004
-#define DOFF_ST 0x0008
-#define DI_FMT 0x000C
-#define DIFF_CTL 0x0010
-#define DIFF_ST 0x0014
-#define CKG1 0x0018
-#define CKG2 0x001C
-#define DIDT 0x0020
-#define DODT 0x0024
-#define MUTE_ST 0x0028
-#define OUT_SEL 0x0030
-#define REG_END OUT_SEL
-
+/* PortA/PortB register */
+#define REG_DO_FMT 0x0000
+#define REG_DOFF_CTL 0x0004
+#define REG_DOFF_ST 0x0008
+#define REG_DI_FMT 0x000C
+#define REG_DIFF_CTL 0x0010
+#define REG_DIFF_ST 0x0014
+#define REG_CKG1 0x0018
+#define REG_CKG2 0x001C
+#define REG_DIDT 0x0020
+#define REG_DODT 0x0024
+#define REG_MUTE_ST 0x0028
+#define REG_OUT_SEL 0x0030
+
+/* master register */
+#define MST_CLK_RST 0x0210
+#define MST_SOFT_RST 0x0214
+#define MST_FIFO_SZ 0x0218
+
+/* core register (depend on FSI version) */
#define A_MST_CTLR 0x0180
#define B_MST_CTLR 0x01A0
#define CPU_INT_ST 0x01F4
@@ -41,22 +47,23 @@
#define INT_ST 0x0200
#define IEMSK 0x0204
#define IMSK 0x0208
-#define MUTE 0x020C
-#define CLK_RST 0x0210
-#define SOFT_RST 0x0214
-#define FIFO_SZ 0x0218
-#define MREG_START A_MST_CTLR
-#define MREG_END FIFO_SZ
/* DO_FMT */
/* DI_FMT */
+#define CR_BWS_24 (0x0 << 20) /* FSI2 */
+#define CR_BWS_16 (0x1 << 20) /* FSI2 */
+#define CR_BWS_20 (0x2 << 20) /* FSI2 */
+
+#define CR_DTMD_PCM (0x0 << 8) /* FSI2 */
+#define CR_DTMD_SPDIF_PCM (0x1 << 8) /* FSI2 */
+#define CR_DTMD_SPDIF_STREAM (0x2 << 8) /* FSI2 */
+
#define CR_MONO (0x0 << 4)
#define CR_MONO_D (0x1 << 4)
#define CR_PCM (0x2 << 4)
#define CR_I2S (0x3 << 4)
#define CR_TDM (0x4 << 4)
#define CR_TDM_D (0x5 << 4)
-#define CR_SPDIF 0x00100120
/* DOFF_CTL */
/* DIFF_CTL */
@@ -93,6 +100,10 @@
#define IR (1 << 4) /* Interrupt Reset */
#define FSISR (1 << 0) /* Software Reset */
+/* OUT_SEL (FSI2) */
+#define DMMD (1 << 4) /* SPDIF output timing 0: Biphase only */
+ /* 1: Biphase and serial */
+
/* FIFO_SZ */
#define FIFO_SZ_MASK 0x7
@@ -132,7 +143,7 @@ struct fsi_priv {
struct fsi_stream playback;
struct fsi_stream capture;
- u32 mst_ctrl;
+ long rate;
};
struct fsi_core {
@@ -141,6 +152,8 @@ struct fsi_core {
u32 int_st;
u32 iemsk;
u32 imsk;
+ u32 a_mclk;
+ u32 b_mclk;
};
struct fsi_master {
@@ -180,62 +193,22 @@ static void __fsi_reg_mask_set(u32 reg, u32 mask, u32 data)
__fsi_reg_write(reg, val);
}
-static void fsi_reg_write(struct fsi_priv *fsi, u32 reg, u32 data)
-{
- if (reg > REG_END) {
- pr_err("fsi: register access err (%s)\n", __func__);
- return;
- }
-
- __fsi_reg_write((u32)(fsi->base + reg), data);
-}
+#define fsi_reg_write(p, r, d)\
+ __fsi_reg_write((u32)(p->base + REG_##r), d)
-static u32 fsi_reg_read(struct fsi_priv *fsi, u32 reg)
-{
- if (reg > REG_END) {
- pr_err("fsi: register access err (%s)\n", __func__);
- return 0;
- }
+#define fsi_reg_read(p, r)\
+ __fsi_reg_read((u32)(p->base + REG_##r))
- return __fsi_reg_read((u32)(fsi->base + reg));
-}
+#define fsi_reg_mask_set(p, r, m, d)\
+ __fsi_reg_mask_set((u32)(p->base + REG_##r), m, d)
-static void fsi_reg_mask_set(struct fsi_priv *fsi, u32 reg, u32 mask, u32 data)
-{
- if (reg > REG_END) {
- pr_err("fsi: register access err (%s)\n", __func__);
- return;
- }
-
- __fsi_reg_mask_set((u32)(fsi->base + reg), mask, data);
-}
-
-static void fsi_master_write(struct fsi_master *master, u32 reg, u32 data)
-{
- unsigned long flags;
-
- if ((reg < MREG_START) ||
- (reg > MREG_END)) {
- pr_err("fsi: register access err (%s)\n", __func__);
- return;
- }
-
- spin_lock_irqsave(&master->lock, flags);
- __fsi_reg_write((u32)(master->base + reg), data);
- spin_unlock_irqrestore(&master->lock, flags);
-}
-
-static u32 fsi_master_read(struct fsi_master *master, u32 reg)
+#define fsi_master_read(p, r) _fsi_master_read(p, MST_##r)
+#define fsi_core_read(p, r) _fsi_master_read(p, p->core->r)
+static u32 _fsi_master_read(struct fsi_master *master, u32 reg)
{
u32 ret;
unsigned long flags;
- if ((reg < MREG_START) ||
- (reg > MREG_END)) {
- pr_err("fsi: register access err (%s)\n", __func__);
- return 0;
- }
-
spin_lock_irqsave(&master->lock, flags);
ret = __fsi_reg_read((u32)(master->base + reg));
spin_unlock_irqrestore(&master->lock, flags);
@@ -243,17 +216,13 @@ static u32 fsi_master_read(struct fsi_master *master, u32 reg)
return ret;
}
-static void fsi_master_mask_set(struct fsi_master *master,
+#define fsi_master_mask_set(p, r, m, d) _fsi_master_mask_set(p, MST_##r, m, d)
+#define fsi_core_mask_set(p, r, m, d) _fsi_master_mask_set(p, p->core->r, m, d)
+static void _fsi_master_mask_set(struct fsi_master *master,
u32 reg, u32 mask, u32 data)
{
unsigned long flags;
- if ((reg < MREG_START) ||
- (reg > MREG_END)) {
- pr_err("fsi: register access err (%s)\n", __func__);
- return;
- }
-
spin_lock_irqsave(&master->lock, flags);
__fsi_reg_mask_set((u32)(master->base + reg), mask, data);
spin_unlock_irqrestore(&master->lock, flags);
@@ -373,11 +342,13 @@ static void fsi_stream_pop(struct fsi_priv *fsi, int is_play)
static int fsi_get_fifo_data_num(struct fsi_priv *fsi, int is_play)
{
u32 status;
- u32 reg = is_play ? DOFF_ST : DIFF_ST;
struct fsi_stream *io = fsi_get_stream(fsi, is_play);
int data_num;
- status = fsi_reg_read(fsi, reg);
+ status = is_play ?
+ fsi_reg_read(fsi, DOFF_ST) :
+ fsi_reg_read(fsi, DIFF_ST);
+
data_num = 0x1ff & (status >> 8);
data_num *= io->chan_num;
@@ -471,8 +442,8 @@ static void fsi_irq_enable(struct fsi_priv *fsi, int is_play)
u32 data = AB_IO(1, fsi_get_port_shift(fsi, is_play));
struct fsi_master *master = fsi_get_master(fsi);
- fsi_master_mask_set(master, master->core->imsk, data, data);
- fsi_master_mask_set(master, master->core->iemsk, data, data);
+ fsi_core_mask_set(master, imsk, data, data);
+ fsi_core_mask_set(master, iemsk, data, data);
}
static void fsi_irq_disable(struct fsi_priv *fsi, int is_play)
@@ -480,18 +451,13 @@ static void fsi_irq_disable(struct fsi_priv *fsi, int is_play)
u32 data = AB_IO(1, fsi_get_port_shift(fsi, is_play));
struct fsi_master *master = fsi_get_master(fsi);
- fsi_master_mask_set(master, master->core->imsk, data, 0);
- fsi_master_mask_set(master, master->core->iemsk, data, 0);
+ fsi_core_mask_set(master, imsk, data, 0);
+ fsi_core_mask_set(master, iemsk, data, 0);
}
static u32 fsi_irq_get_status(struct fsi_master *master)
{
- return fsi_master_read(master, master->core->int_st);
-}
-
-static void fsi_irq_clear_all_status(struct fsi_master *master)
-{
- fsi_master_write(master, master->core->int_st, 0);
+ return fsi_core_read(master, int_st);
}
static void fsi_irq_clear_status(struct fsi_priv *fsi)
@@ -503,7 +469,7 @@ static void fsi_irq_clear_status(struct fsi_priv *fsi)
data |= AB_IO(1, fsi_get_port_shift(fsi, 1));
/* clear interrupt factor */
- fsi_master_mask_set(master, master->core->int_st, data, 0);
+ fsi_core_mask_set(master, int_st, data, 0);
}
/*
@@ -514,17 +480,19 @@ static void fsi_irq_clear_status(struct fsi_priv *fsi)
static void fsi_spdif_clk_ctrl(struct fsi_priv *fsi, int enable)
{
struct fsi_master *master = fsi_get_master(fsi);
- u32 val = BP | SE;
+ u32 mask, val;
if (master->core->ver < 2) {
pr_err("fsi: register access err (%s)\n", __func__);
return;
}
- if (enable)
- fsi_master_mask_set(master, fsi->mst_ctrl, val, val);
- else
- fsi_master_mask_set(master, fsi->mst_ctrl, val, 0);
+ mask = BP | SE;
+ val = enable ? mask : 0;
+
+ fsi_is_port_a(fsi) ?
+ fsi_core_mask_set(master, a_mclk, mask, val) :
+ fsi_core_mask_set(master, b_mclk, mask, val);
}
/*
@@ -548,7 +516,7 @@ static void fsi_fifo_init(struct fsi_priv *fsi,
{
struct fsi_master *master = fsi_get_master(fsi);
struct fsi_stream *io = fsi_get_stream(fsi, is_play);
- u32 ctrl, shift, i;
+ u32 shift, i;
/* get on-chip RAM capacity */
shift = fsi_master_read(master, FIFO_SZ);
@@ -581,13 +549,17 @@ static void fsi_fifo_init(struct fsi_priv *fsi,
dev_dbg(dai->dev, "%d channel %d store\n",
io->chan_num, io->fifo_max_num);
- ctrl = is_play ? DOFF_CTL : DIFF_CTL;
-
- /* set interrupt generation factor */
- fsi_reg_write(fsi, ctrl, IRQ_HALF);
-
- /* clear FIFO */
- fsi_reg_mask_set(fsi, ctrl, FIFO_CLR, FIFO_CLR);
+ /*
+ * set interrupt generation factor
+ * clear FIFO
+ */
+ if (is_play) {
+ fsi_reg_write(fsi, DOFF_CTL, IRQ_HALF);
+ fsi_reg_mask_set(fsi, DOFF_CTL, FIFO_CLR, FIFO_CLR);
+ } else {
+ fsi_reg_write(fsi, DIFF_CTL, IRQ_HALF);
+ fsi_reg_mask_set(fsi, DIFF_CTL, FIFO_CLR, FIFO_CLR);
+ }
}
static void fsi_soft_all_reset(struct fsi_master *master)
@@ -608,7 +580,6 @@ static int fsi_fifo_data_ctrl(struct fsi_priv *fsi, int startup, int stream)
struct snd_pcm_substream *substream = NULL;
int is_play = fsi_stream_is_play(stream);
struct fsi_stream *io = fsi_get_stream(fsi, is_play);
- u32 status_reg = is_play ? DOFF_ST : DIFF_ST;
int data_residue_num;
int data_num;
int data_num_max;
@@ -699,14 +670,19 @@ static int fsi_fifo_data_ctrl(struct fsi_priv *fsi, int startup, int stream)
/* check fifo status */
if (!startup) {
struct snd_soc_dai *dai = fsi_get_dai(substream);
- u32 status = fsi_reg_read(fsi, status_reg);
+ u32 status = is_play ?
+ fsi_reg_read(fsi, DOFF_ST) :
+ fsi_reg_read(fsi, DIFF_ST);
if (status & ERR_OVER)
dev_err(dai->dev, "over run\n");
if (status & ERR_UNDER)
dev_err(dai->dev, "under run\n");
}
- fsi_reg_write(fsi, status_reg, 0);
+
+ is_play ?
+ fsi_reg_write(fsi, DOFF_ST, 0) :
+ fsi_reg_write(fsi, DIFF_ST, 0);
/* re-enable irq */
fsi_irq_enable(fsi, is_play);
@@ -745,7 +721,8 @@ static irqreturn_t fsi_interrupt(int irq, void *data)
if (int_st & AB_IO(1, BI_SHIFT))
fsi_data_pop(&master->fsib, 0);
- fsi_irq_clear_all_status(master);
+ fsi_irq_clear_status(&master->fsia);
+ fsi_irq_clear_status(&master->fsib);
return IRQ_HANDLED;
}
@@ -762,7 +739,6 @@ static int fsi_dai_startup(struct snd_pcm_substream *substream,
struct fsi_stream *io;
u32 flags = fsi_get_info_flags(fsi);
u32 fmt;
- u32 reg;
u32 data;
int is_play = fsi_is_play(substream);
int is_master;
@@ -794,7 +770,6 @@ static int fsi_dai_startup(struct snd_pcm_substream *substream,
/* do fmt, di fmt */
data = 0;
- reg = is_play ? DO_FMT : DI_FMT;
fmt = is_play ? SH_FSI_GET_OFMT(flags) : SH_FSI_GET_IFMT(flags);
switch (fmt) {
case SH_FSI_FMT_MONO:
@@ -828,16 +803,18 @@ static int fsi_dai_startup(struct snd_pcm_substream *substream,
dev_err(dai->dev, "This FSI can not use SPDIF\n");
return -EINVAL;
}
- data = CR_SPDIF;
+ data = CR_BWS_16 | CR_DTMD_SPDIF_PCM | CR_PCM;
io->chan_num = 2;
fsi_spdif_clk_ctrl(fsi, 1);
- fsi_reg_mask_set(fsi, OUT_SEL, 0x0010, 0x0010);
+ fsi_reg_mask_set(fsi, OUT_SEL, DMMD, DMMD);
break;
default:
dev_err(dai->dev, "unknown format.\n");
return -EINVAL;
}
- fsi_reg_write(fsi, reg, data);
+ is_play ?
+ fsi_reg_write(fsi, DO_FMT, data) :
+ fsi_reg_write(fsi, DI_FMT, data);
/* irq clear */
fsi_irq_disable(fsi, is_play);
@@ -854,10 +831,17 @@ static void fsi_dai_shutdown(struct snd_pcm_substream *substream,
{
struct fsi_priv *fsi = fsi_get_priv(substream);
int is_play = fsi_is_play(substream);
+ struct fsi_master *master = fsi_get_master(fsi);
+ int (*set_rate)(struct device *dev, int is_porta, int rate, int enable);
fsi_irq_disable(fsi, is_play);
fsi_clk_ctrl(fsi, 0);
+ set_rate = master->info->set_rate;
+ if (set_rate && fsi->rate)
+ set_rate(dai->dev, fsi_is_port_a(fsi), fsi->rate, 0);
+ fsi->rate = 0;
+
pm_runtime_put_sync(dai->dev);
}
@@ -891,20 +875,20 @@ static int fsi_dai_hw_params(struct snd_pcm_substream *substream,
{
struct fsi_priv *fsi = fsi_get_priv(substream);
struct fsi_master *master = fsi_get_master(fsi);
- int (*set_rate)(int is_porta, int rate) = master->info->set_rate;
+ int (*set_rate)(struct device *dev, int is_porta, int rate, int enable);
int fsi_ver = master->core->ver;
- int is_play = fsi_is_play(substream);
+ long rate = params_rate(params);
int ret;
- /* if slave mode, set_rate is not needed */
- if (!fsi_is_master_mode(fsi, is_play))
+ set_rate = master->info->set_rate;
+ if (!set_rate)
return 0;
- /* it is error if no set_rate */
- if (!set_rate)
- return -EIO;
+ ret = set_rate(dai->dev, fsi_is_port_a(fsi), rate, 1);
+ if (ret < 0) /* error */
+ return ret;
- ret = set_rate(fsi_is_port_a(fsi), params_rate(params));
+ fsi->rate = rate;
if (ret > 0) {
u32 data = 0;
@@ -1165,12 +1149,10 @@ static int fsi_probe(struct platform_device *pdev)
/* FSI A setting */
master->fsia.base = master->base;
master->fsia.master = master;
- master->fsia.mst_ctrl = A_MST_CTLR;
/* FSI B setting */
master->fsib.base = master->base + 0x40;
master->fsib.master = master;
- master->fsib.mst_ctrl = B_MST_CTLR;
pm_runtime_enable(&pdev->dev);
pm_runtime_resume(&pdev->dev);
@@ -1257,6 +1239,8 @@ static struct fsi_core fsi2_core = {
.int_st = CPU_INT_ST,
.iemsk = CPU_IEMSK,
.imsk = CPU_IMSK,
+ .a_mclk = A_MST_CTLR,
+ .b_mclk = B_MST_CTLR,
};
static struct platform_device_id fsi_id_table[] = {
diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c
index 9b1ba33e6fe9..0e17b4050425 100644
--- a/sound/soc/soc-cache.c
+++ b/sound/soc/soc-cache.c
@@ -85,7 +85,7 @@ static int snd_soc_4_12_spi_write(void *control_data, const char *data,
msg[1] = data[0];
spi_message_init(&m);
- memset(&t, 0, (sizeof t));
+ memset(&t, 0, sizeof t);
t.tx_buf = &msg[0];
t.len = len;
@@ -166,7 +166,7 @@ static int snd_soc_7_9_spi_write(void *control_data, const char *data,
msg[1] = data[1];
spi_message_init(&m);
- memset(&t, 0, (sizeof t));
+ memset(&t, 0, sizeof t);
t.tx_buf = &msg[0];
t.len = len;
@@ -246,7 +246,7 @@ static int snd_soc_8_8_spi_write(void *control_data, const char *data,
msg[1] = data[1];
spi_message_init(&m);
- memset(&t, 0, (sizeof t));
+ memset(&t, 0, sizeof t);
t.tx_buf = &msg[0];
t.len = len;
@@ -326,7 +326,7 @@ static int snd_soc_8_16_spi_write(void *control_data, const char *data,
msg[2] = data[2];
spi_message_init(&m);
- memset(&t, 0, (sizeof t));
+ memset(&t, 0, sizeof t);
t.tx_buf = &msg[0];
t.len = len;
@@ -513,7 +513,7 @@ static int snd_soc_16_8_spi_write(void *control_data, const char *data,
msg[2] = data[2];
spi_message_init(&m);
- memset(&t, 0, (sizeof t));
+ memset(&t, 0, sizeof t);
t.tx_buf = &msg[0];
t.len = len;
@@ -633,7 +633,7 @@ static int snd_soc_16_16_spi_write(void *control_data, const char *data,
msg[3] = data[3];
spi_message_init(&m);
- memset(&t, 0, (sizeof t));
+ memset(&t, 0, sizeof t);
t.tx_buf = &msg[0];
t.len = len;
@@ -728,8 +728,8 @@ int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec,
return -EINVAL;
}
- codec->driver->write = io_types[i].write;
- codec->driver->read = io_types[i].read;
+ codec->write = io_types[i].write;
+ codec->read = io_types[i].read;
switch (control) {
case SND_SOC_CUSTOM:
@@ -933,7 +933,7 @@ static int snd_soc_rbtree_cache_init(struct snd_soc_codec *codec)
rbtree_ctx = codec->reg_cache;
rbtree_ctx->root = RB_ROOT;
- if (!codec->driver->reg_cache_default)
+ if (!codec->reg_def_copy)
return 0;
/*
@@ -951,7 +951,7 @@ static int snd_soc_rbtree_cache_init(struct snd_soc_codec *codec)
struct snd_soc_rbtree_node *rbtree_node; \
\
ret = 0; \
- cache = codec->driver->reg_cache_default; \
+ cache = codec->reg_def_copy; \
for (i = 0; i < codec->driver->reg_cache_size; ++i) { \
if (!cache[i]) \
continue; \
@@ -1078,7 +1078,7 @@ static int snd_soc_lzo_decompress_cache_block(struct snd_soc_codec *codec,
static inline int snd_soc_lzo_get_blkindex(struct snd_soc_codec *codec,
unsigned int reg)
{
- struct snd_soc_codec_driver *codec_drv;
+ const struct snd_soc_codec_driver *codec_drv;
size_t reg_size;
codec_drv = codec->driver;
@@ -1090,7 +1090,7 @@ static inline int snd_soc_lzo_get_blkindex(struct snd_soc_codec *codec,
static inline int snd_soc_lzo_get_blkpos(struct snd_soc_codec *codec,
unsigned int reg)
{
- struct snd_soc_codec_driver *codec_drv;
+ const struct snd_soc_codec_driver *codec_drv;
size_t reg_size;
codec_drv = codec->driver;
@@ -1101,7 +1101,7 @@ static inline int snd_soc_lzo_get_blkpos(struct snd_soc_codec *codec,
static inline int snd_soc_lzo_get_blksize(struct snd_soc_codec *codec)
{
- struct snd_soc_codec_driver *codec_drv;
+ const struct snd_soc_codec_driver *codec_drv;
size_t reg_size;
codec_drv = codec->driver;
@@ -1301,7 +1301,7 @@ static int snd_soc_lzo_cache_init(struct snd_soc_codec *codec)
{
struct snd_soc_lzo_ctx **lzo_blocks;
size_t reg_size, bmp_size;
- struct snd_soc_codec_driver *codec_drv;
+ const struct snd_soc_codec_driver *codec_drv;
int ret, tofree, i, blksize, blkcount;
const char *p, *end;
unsigned long *sync_bmp;
@@ -1316,13 +1316,13 @@ static int snd_soc_lzo_cache_init(struct snd_soc_codec *codec)
* and remember to free it afterwards.
*/
tofree = 0;
- if (!codec_drv->reg_cache_default)
+ if (!codec->reg_def_copy)
tofree = 1;
- if (!codec_drv->reg_cache_default) {
- codec_drv->reg_cache_default = kzalloc(reg_size,
+ if (!codec->reg_def_copy) {
+ codec->reg_def_copy = kzalloc(reg_size,
GFP_KERNEL);
- if (!codec_drv->reg_cache_default)
+ if (!codec->reg_def_copy)
return -ENOMEM;
}
@@ -1342,13 +1342,13 @@ static int snd_soc_lzo_cache_init(struct snd_soc_codec *codec)
* that register.
*/
bmp_size = codec_drv->reg_cache_size;
- sync_bmp = kmalloc(BITS_TO_LONGS(bmp_size) * sizeof (long),
+ sync_bmp = kmalloc(BITS_TO_LONGS(bmp_size) * sizeof(long),
GFP_KERNEL);
if (!sync_bmp) {
ret = -ENOMEM;
goto err;
}
- bitmap_zero(sync_bmp, reg_size);
+ bitmap_zero(sync_bmp, bmp_size);
/* allocate the lzo blocks and initialize them */
for (i = 0; i < blkcount; ++i) {
@@ -1368,8 +1368,8 @@ static int snd_soc_lzo_cache_init(struct snd_soc_codec *codec)
}
blksize = snd_soc_lzo_get_blksize(codec);
- p = codec_drv->reg_cache_default;
- end = codec_drv->reg_cache_default + reg_size;
+ p = codec->reg_def_copy;
+ end = codec->reg_def_copy + reg_size;
/* compress the register map and fill the lzo blocks */
for (i = 0; i < blkcount; ++i, p += blksize) {
lzo_blocks[i]->src = p;
@@ -1385,14 +1385,18 @@ static int snd_soc_lzo_cache_init(struct snd_soc_codec *codec)
lzo_blocks[i]->src_len;
}
- if (tofree)
- kfree(codec_drv->reg_cache_default);
+ if (tofree) {
+ kfree(codec->reg_def_copy);
+ codec->reg_def_copy = NULL;
+ }
return 0;
err:
snd_soc_cache_exit(codec);
err_tofree:
- if (tofree)
- kfree(codec_drv->reg_cache_default);
+ if (tofree) {
+ kfree(codec->reg_def_copy);
+ codec->reg_def_copy = NULL;
+ }
return ret;
}
@@ -1400,7 +1404,7 @@ static int snd_soc_flat_cache_sync(struct snd_soc_codec *codec)
{
int i;
int ret;
- struct snd_soc_codec_driver *codec_drv;
+ const struct snd_soc_codec_driver *codec_drv;
unsigned int val;
codec_drv = codec->driver;
@@ -1500,12 +1504,20 @@ static int snd_soc_flat_cache_exit(struct snd_soc_codec *codec)
static int snd_soc_flat_cache_init(struct snd_soc_codec *codec)
{
- struct snd_soc_codec_driver *codec_drv;
+ const struct snd_soc_codec_driver *codec_drv;
size_t reg_size;
codec_drv = codec->driver;
reg_size = codec_drv->reg_cache_size * codec_drv->reg_word_size;
+ /*
+ * for flat compression, we don't need to keep a copy of the
+ * original defaults register cache as it will definitely not
+ * be marked as __devinitconst
+ */
+ kfree(codec->reg_def_copy);
+ codec->reg_def_copy = NULL;
+
if (codec_drv->reg_cache_default)
codec->reg_cache = kmemdup(codec_drv->reg_cache_default,
reg_size, GFP_KERNEL);
@@ -1521,6 +1533,7 @@ static int snd_soc_flat_cache_init(struct snd_soc_codec *codec)
static const struct snd_soc_cache_ops cache_types[] = {
{
.id = SND_SOC_FLAT_COMPRESSION,
+ .name = "flat",
.init = snd_soc_flat_cache_init,
.exit = snd_soc_flat_cache_exit,
.read = snd_soc_flat_cache_read,
@@ -1529,6 +1542,7 @@ static const struct snd_soc_cache_ops cache_types[] = {
},
{
.id = SND_SOC_LZO_COMPRESSION,
+ .name = "LZO",
.init = snd_soc_lzo_cache_init,
.exit = snd_soc_lzo_cache_exit,
.read = snd_soc_lzo_cache_read,
@@ -1537,6 +1551,7 @@ static const struct snd_soc_cache_ops cache_types[] = {
},
{
.id = SND_SOC_RBTREE_COMPRESSION,
+ .name = "rbtree",
.init = snd_soc_rbtree_cache_init,
.exit = snd_soc_rbtree_cache_exit,
.read = snd_soc_rbtree_cache_read,
@@ -1550,19 +1565,23 @@ int snd_soc_cache_init(struct snd_soc_codec *codec)
int i;
for (i = 0; i < ARRAY_SIZE(cache_types); ++i)
- if (cache_types[i].id == codec->driver->compress_type)
+ if (cache_types[i].id == codec->compress_type)
break;
if (i == ARRAY_SIZE(cache_types)) {
dev_err(codec->dev, "Could not match compress type: %d\n",
- codec->driver->compress_type);
+ codec->compress_type);
return -EINVAL;
}
mutex_init(&codec->cache_rw_mutex);
codec->cache_ops = &cache_types[i];
- if (codec->cache_ops->init)
+ if (codec->cache_ops->init) {
+ if (codec->cache_ops->name)
+ dev_dbg(codec->dev, "Initializing %s cache for %s codec\n",
+ codec->cache_ops->name, codec->name);
return codec->cache_ops->init(codec);
+ }
return -EINVAL;
}
@@ -1572,8 +1591,12 @@ int snd_soc_cache_init(struct snd_soc_codec *codec)
*/
int snd_soc_cache_exit(struct snd_soc_codec *codec)
{
- if (codec->cache_ops && codec->cache_ops->exit)
+ if (codec->cache_ops && codec->cache_ops->exit) {
+ if (codec->cache_ops->name)
+ dev_dbg(codec->dev, "Destroying %s cache for %s codec\n",
+ codec->cache_ops->name, codec->name);
return codec->cache_ops->exit(codec);
+ }
return -EINVAL;
}
@@ -1645,6 +1668,9 @@ int snd_soc_cache_sync(struct snd_soc_codec *codec)
}
if (codec->cache_ops && codec->cache_ops->sync) {
+ if (codec->cache_ops->name)
+ dev_dbg(codec->dev, "Syncing %s cache for %s codec\n",
+ codec->cache_ops->name, codec->name);
ret = codec->cache_ops->sync(codec);
if (!ret)
codec->cache_sync = 0;
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index c8d8dde7f4dd..1dc4b11c1988 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -33,6 +33,7 @@
#include <linux/slab.h>
#include <sound/ac97_codec.h>
#include <sound/core.h>
+#include <sound/jack.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
#include <sound/soc.h>
@@ -69,25 +70,6 @@ static int pmdown_time = 5000;
module_param(pmdown_time, int, 0);
MODULE_PARM_DESC(pmdown_time, "DAPM stream powerdown time (msecs)");
-/*
- * 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;
-}
-
/* codec register dump */
static ssize_t soc_codec_reg_show(struct snd_soc_codec *codec, char *buf)
{
@@ -116,7 +98,7 @@ static ssize_t soc_codec_reg_show(struct snd_soc_codec *codec, char *buf)
* the register being volatile and the device being
* powered off.
*/
- ret = codec->driver->read(codec, i);
+ ret = snd_soc_read(codec, i);
if (ret >= 0)
count += snprintf(buf + count,
PAGE_SIZE - count,
@@ -227,7 +209,7 @@ static ssize_t codec_reg_write_file(struct file *file,
start++;
if (strict_strtoul(start, 16, &value))
return -EINVAL;
- codec->driver->write(codec, reg, value);
+ snd_soc_write(codec, reg, value);
return buf_size;
}
@@ -525,7 +507,7 @@ static int soc_pcm_open(struct snd_pcm_substream *substream)
}
}
- /* Check that the codec and cpu DAI's are compatible */
+ /* Check that the codec and cpu DAIs are compatible */
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
runtime->hw.rate_min =
max(codec_dai_drv->playback.rate_min,
@@ -874,7 +856,7 @@ codec_err:
}
/*
- * Free's resources allocated by hw_params, can be called multiple times
+ * Frees resources allocated by hw_params, can be called multiple times
*/
static int soc_pcm_hw_free(struct snd_pcm_substream *substream)
{
@@ -898,7 +880,7 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream)
if (platform->driver->ops->hw_free)
platform->driver->ops->hw_free(substream);
- /* now free hw params for the DAI's */
+ /* now free hw params for the DAIs */
if (codec_dai->driver->ops->hw_free)
codec_dai->driver->ops->hw_free(substream, codec_dai);
@@ -986,6 +968,7 @@ static int soc_suspend(struct device *dev)
{
struct platform_device *pdev = to_platform_device(dev);
struct snd_soc_card *card = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec;
int i;
/* If the initialization of this soc device failed, there is no codec
@@ -1004,7 +987,7 @@ static int soc_suspend(struct device *dev)
/* we're going to block userspace touching us until resume completes */
snd_power_change_state(card->snd_card, SNDRV_CTL_POWER_D3hot);
- /* mute any active DAC's */
+ /* mute any active DACs */
for (i = 0; i < card->num_rtd; i++) {
struct snd_soc_dai *dai = card->rtd[i].codec_dai;
struct snd_soc_dai_driver *drv = dai->driver;
@@ -1044,7 +1027,7 @@ static int soc_suspend(struct device *dev)
/* close any waiting streams and save state */
for (i = 0; i < card->num_rtd; i++) {
- run_delayed_work(&card->rtd[i].delayed_work);
+ flush_delayed_work_sync(&card->rtd[i].delayed_work);
card->rtd[i].codec->dapm.suspend_bias_level = card->rtd[i].codec->dapm.bias_level;
}
@@ -1064,8 +1047,7 @@ static int soc_suspend(struct device *dev)
}
/* suspend all CODECs */
- for (i = 0; i < card->num_rtd; i++) {
- struct snd_soc_codec *codec = card->rtd[i].codec;
+ list_for_each_entry(codec, &card->codec_dev_list, card_list) {
/* If there are paths active then the CODEC will be held with
* bias _ON and should not be suspended. */
if (!codec->suspended && codec->driver->suspend) {
@@ -1106,6 +1088,7 @@ static void soc_resume_deferred(struct work_struct *work)
struct snd_soc_card *card =
container_of(work, struct snd_soc_card, deferred_resume_work);
struct platform_device *pdev = to_platform_device(card->dev);
+ struct snd_soc_codec *codec;
int i;
/* our power state is still SNDRV_CTL_POWER_D3hot from suspend time,
@@ -1131,8 +1114,7 @@ static void soc_resume_deferred(struct work_struct *work)
cpu_dai->driver->resume(cpu_dai);
}
- for (i = 0; i < card->num_rtd; i++) {
- struct snd_soc_codec *codec = card->rtd[i].codec;
+ list_for_each_entry(codec, &card->codec_dev_list, card_list) {
/* If the CODEC was idle over suspend then it will have been
* left with bias OFF or STANDBY and suspended so we must now
* resume. Otherwise the suspend was suppressed.
@@ -1327,6 +1309,27 @@ out:
return 1;
}
+static void soc_remove_codec(struct snd_soc_codec *codec)
+{
+ int err;
+
+ if (codec->driver->remove) {
+ err = codec->driver->remove(codec);
+ if (err < 0)
+ dev_err(codec->dev,
+ "asoc: failed to remove %s: %d\n",
+ codec->name, err);
+ }
+
+ /* Make sure all DAPM widgets are freed */
+ snd_soc_dapm_free(&codec->dapm);
+
+ soc_cleanup_codec_debugfs(codec);
+ codec->probed = 0;
+ list_del(&codec->card_list);
+ module_put(codec->dev->driver->owner);
+}
+
static void soc_remove_dai_link(struct snd_soc_card *card, int num)
{
struct snd_soc_pcm_runtime *rtd = &card->rtd[num];
@@ -1338,6 +1341,7 @@ static void soc_remove_dai_link(struct snd_soc_card *card, int num)
/* unregister the rtd device */
if (rtd->dev_registered) {
device_remove_file(&rtd->dev, &dev_attr_pmdown_time);
+ device_remove_file(&rtd->dev, &dev_attr_codec_reg);
device_unregister(&rtd->dev);
rtd->dev_registered = 0;
}
@@ -1366,22 +1370,8 @@ static void soc_remove_dai_link(struct snd_soc_card *card, int num)
}
/* remove the CODEC */
- if (codec && codec->probed) {
- if (codec->driver->remove) {
- err = codec->driver->remove(codec);
- if (err < 0)
- printk(KERN_ERR "asoc: failed to remove %s\n", codec->name);
- }
-
- /* Make sure all DAPM widgets are freed */
- snd_soc_dapm_free(&codec->dapm);
-
- soc_cleanup_codec_debugfs(codec);
- device_remove_file(&rtd->dev, &dev_attr_codec_reg);
- codec->probed = 0;
- list_del(&codec->card_list);
- module_put(codec->dev->driver->owner);
- }
+ if (codec && codec->probed)
+ soc_remove_codec(codec);
/* remove the cpu_dai */
if (cpu_dai && cpu_dai->probed) {
@@ -1401,11 +1391,11 @@ static void soc_set_name_prefix(struct snd_soc_card *card,
{
int i;
- if (card->prefix_map == NULL)
+ if (card->codec_conf == NULL)
return;
- for (i = 0; i < card->num_prefixes; i++) {
- struct snd_soc_prefix_map *map = &card->prefix_map[i];
+ for (i = 0; i < card->num_configs; i++) {
+ struct snd_soc_codec_conf *map = &card->codec_conf[i];
if (map->dev_name && !strcmp(codec->name, map->dev_name)) {
codec->name_prefix = map->name_prefix;
break;
@@ -1413,8 +1403,105 @@ static void soc_set_name_prefix(struct snd_soc_card *card,
}
}
+static int soc_probe_codec(struct snd_soc_card *card,
+ struct snd_soc_codec *codec)
+{
+ int ret = 0;
+
+ codec->card = card;
+ codec->dapm.card = card;
+ soc_set_name_prefix(card, codec);
+
+ if (codec->driver->probe) {
+ ret = codec->driver->probe(codec);
+ if (ret < 0) {
+ dev_err(codec->dev,
+ "asoc: failed to probe CODEC %s: %d\n",
+ codec->name, ret);
+ return ret;
+ }
+ }
+
+ soc_init_codec_debugfs(codec);
+
+ /* mark codec as probed and add to card codec list */
+ codec->probed = 1;
+ list_add(&codec->card_list, &card->codec_dev_list);
+
+ return ret;
+}
+
static void rtd_release(struct device *dev) {}
+static int soc_post_component_init(struct snd_soc_card *card,
+ struct snd_soc_codec *codec,
+ int num, int dailess)
+{
+ struct snd_soc_dai_link *dai_link = NULL;
+ struct snd_soc_aux_dev *aux_dev = NULL;
+ struct snd_soc_pcm_runtime *rtd;
+ const char *temp, *name;
+ int ret = 0;
+
+ if (!dailess) {
+ dai_link = &card->dai_link[num];
+ rtd = &card->rtd[num];
+ name = dai_link->name;
+ } else {
+ aux_dev = &card->aux_dev[num];
+ rtd = &card->rtd_aux[num];
+ name = aux_dev->name;
+ }
+
+ /* machine controls, routes and widgets are not prefixed */
+ temp = codec->name_prefix;
+ codec->name_prefix = NULL;
+
+ /* do machine specific initialization */
+ if (!dailess && dai_link->init)
+ ret = dai_link->init(rtd);
+ else if (dailess && aux_dev->init)
+ ret = aux_dev->init(&codec->dapm);
+ if (ret < 0) {
+ dev_err(card->dev, "asoc: failed to init %s: %d\n", name, ret);
+ return ret;
+ }
+ codec->name_prefix = temp;
+
+ /* Make sure all DAPM widgets are instantiated */
+ snd_soc_dapm_new_widgets(&codec->dapm);
+ snd_soc_dapm_sync(&codec->dapm);
+
+ /* register the rtd device */
+ rtd->codec = codec;
+ rtd->card = card;
+ rtd->dev.parent = card->dev;
+ rtd->dev.release = rtd_release;
+ rtd->dev.init_name = name;
+ ret = device_register(&rtd->dev);
+ if (ret < 0) {
+ dev_err(card->dev,
+ "asoc: failed to register runtime device: %d\n", ret);
+ return ret;
+ }
+ rtd->dev_registered = 1;
+
+ /* add DAPM sysfs entries for this codec */
+ ret = snd_soc_dapm_sys_add(&rtd->dev);
+ if (ret < 0)
+ dev_err(codec->dev,
+ "asoc: failed to add codec dapm sysfs entries: %d\n",
+ ret);
+
+ /* add codec sysfs entries */
+ ret = device_create_file(&rtd->dev, &dev_attr_codec_reg);
+ if (ret < 0)
+ dev_err(codec->dev,
+ "asoc: failed to add codec sysfs files: %d\n", ret);
+
+ return 0;
+}
+
static int soc_probe_dai_link(struct snd_soc_card *card, int num)
{
struct snd_soc_dai_link *dai_link = &card->dai_link[num];
@@ -1422,17 +1509,13 @@ static int soc_probe_dai_link(struct snd_soc_card *card, int num)
struct snd_soc_codec *codec = rtd->codec;
struct snd_soc_platform *platform = rtd->platform;
struct snd_soc_dai *codec_dai = rtd->codec_dai, *cpu_dai = rtd->cpu_dai;
- const char *temp;
int ret;
dev_dbg(card->dev, "probe %s dai link %d\n", card->name, num);
/* config components */
codec_dai->codec = codec;
- codec->card = card;
cpu_dai->platform = platform;
- rtd->card = card;
- rtd->dev.parent = card->dev;
codec_dai->card = card;
cpu_dai->card = card;
@@ -1456,22 +1539,9 @@ static int soc_probe_dai_link(struct snd_soc_card *card, int num)
/* probe the CODEC */
if (!codec->probed) {
- codec->dapm.card = card;
- soc_set_name_prefix(card, codec);
- if (codec->driver->probe) {
- ret = codec->driver->probe(codec);
- if (ret < 0) {
- printk(KERN_ERR "asoc: failed to probe CODEC %s\n",
- codec->name);
- return ret;
- }
- }
-
- soc_init_codec_debugfs(codec);
-
- /* mark codec as probed and add to card codec list */
- codec->probed = 1;
- list_add(&codec->card_list, &card->codec_dev_list);
+ ret = soc_probe_codec(card, codec);
+ if (ret < 0)
+ return ret;
}
/* probe the platform */
@@ -1508,47 +1578,14 @@ static int soc_probe_dai_link(struct snd_soc_card *card, int num)
/* DAPM dai link stream work */
INIT_DELAYED_WORK(&rtd->delayed_work, close_delayed_work);
- /* now that all clients have probed, initialise the DAI link */
- if (dai_link->init) {
- /* machine controls, routes and widgets are not prefixed */
- temp = rtd->codec->name_prefix;
- rtd->codec->name_prefix = NULL;
- ret = dai_link->init(rtd);
- if (ret < 0) {
- printk(KERN_ERR "asoc: failed to init %s\n", dai_link->stream_name);
- return ret;
- }
- rtd->codec->name_prefix = temp;
- }
-
- /* Make sure all DAPM widgets are instantiated */
- snd_soc_dapm_new_widgets(&codec->dapm);
- snd_soc_dapm_sync(&codec->dapm);
-
- /* register the rtd device */
- rtd->dev.release = rtd_release;
- rtd->dev.init_name = dai_link->name;
- ret = device_register(&rtd->dev);
- if (ret < 0) {
- printk(KERN_ERR "asoc: failed to register DAI runtime device %d\n", ret);
+ ret = soc_post_component_init(card, codec, num, 0);
+ if (ret)
return ret;
- }
- rtd->dev_registered = 1;
ret = device_create_file(&rtd->dev, &dev_attr_pmdown_time);
if (ret < 0)
printk(KERN_WARNING "asoc: failed to add pmdown_time sysfs\n");
- /* add DAPM sysfs entries for this codec */
- ret = snd_soc_dapm_sys_add(&rtd->dev);
- if (ret < 0)
- printk(KERN_WARNING "asoc: failed to add codec dapm sysfs entries\n");
-
- /* add codec sysfs entries */
- ret = device_create_file(&rtd->dev, &dev_attr_codec_reg);
- if (ret < 0)
- printk(KERN_WARNING "asoc: failed to add codec sysfs files\n");
-
/* create the pcm */
ret = soc_new_pcm(rtd, num);
if (ret < 0) {
@@ -1603,9 +1640,85 @@ static void soc_unregister_ac97_dai_link(struct snd_soc_codec *codec)
}
#endif
+static int soc_probe_aux_dev(struct snd_soc_card *card, int num)
+{
+ struct snd_soc_aux_dev *aux_dev = &card->aux_dev[num];
+ struct snd_soc_codec *codec;
+ int ret = -ENODEV;
+
+ /* find CODEC from registered CODECs*/
+ list_for_each_entry(codec, &codec_list, list) {
+ if (!strcmp(codec->name, aux_dev->codec_name)) {
+ if (codec->probed) {
+ dev_err(codec->dev,
+ "asoc: codec already probed");
+ ret = -EBUSY;
+ goto out;
+ }
+ goto found;
+ }
+ }
+ /* codec not found */
+ dev_err(card->dev, "asoc: codec %s not found", aux_dev->codec_name);
+ goto out;
+
+found:
+ if (!try_module_get(codec->dev->driver->owner))
+ return -ENODEV;
+
+ ret = soc_probe_codec(card, codec);
+ if (ret < 0)
+ return ret;
+
+ ret = soc_post_component_init(card, codec, num, 1);
+
+out:
+ return ret;
+}
+
+static void soc_remove_aux_dev(struct snd_soc_card *card, int num)
+{
+ struct snd_soc_pcm_runtime *rtd = &card->rtd_aux[num];
+ struct snd_soc_codec *codec = rtd->codec;
+
+ /* unregister the rtd device */
+ if (rtd->dev_registered) {
+ device_remove_file(&rtd->dev, &dev_attr_codec_reg);
+ device_unregister(&rtd->dev);
+ rtd->dev_registered = 0;
+ }
+
+ if (codec && codec->probed)
+ soc_remove_codec(codec);
+}
+
+static int snd_soc_init_codec_cache(struct snd_soc_codec *codec,
+ enum snd_soc_compress_type compress_type)
+{
+ int ret;
+
+ if (codec->cache_init)
+ return 0;
+
+ /* override the compress_type if necessary */
+ if (compress_type && codec->compress_type != compress_type)
+ codec->compress_type = compress_type;
+ ret = snd_soc_cache_init(codec);
+ if (ret < 0) {
+ dev_err(codec->dev, "Failed to set cache compression type: %d\n",
+ ret);
+ return ret;
+ }
+ codec->cache_init = 1;
+ return 0;
+}
+
static void snd_soc_instantiate_card(struct snd_soc_card *card)
{
struct platform_device *pdev = to_platform_device(card->dev);
+ struct snd_soc_codec *codec;
+ struct snd_soc_codec_conf *codec_conf;
+ enum snd_soc_compress_type compress_type;
int ret, i;
mutex_lock(&card->mutex);
@@ -1625,6 +1738,39 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card)
return;
}
+ /* initialize the register cache for each available codec */
+ list_for_each_entry(codec, &codec_list, list) {
+ if (codec->cache_init)
+ continue;
+ /* check to see if we need to override the compress_type */
+ for (i = 0; i < card->num_configs; ++i) {
+ codec_conf = &card->codec_conf[i];
+ if (!strcmp(codec->name, codec_conf->dev_name)) {
+ compress_type = codec_conf->compress_type;
+ if (compress_type && compress_type
+ != codec->compress_type)
+ break;
+ }
+ }
+ if (i == card->num_configs) {
+ /* no need to override the compress_type so
+ * go ahead and do the standard thing */
+ ret = snd_soc_init_codec_cache(codec, 0);
+ if (ret < 0) {
+ mutex_unlock(&card->mutex);
+ return;
+ }
+ continue;
+ }
+ /* override the compress_type with the one supplied in
+ * the machine driver */
+ ret = snd_soc_init_codec_cache(codec, compress_type);
+ if (ret < 0) {
+ mutex_unlock(&card->mutex);
+ return;
+ }
+ }
+
/* card bind complete so register a sound card */
ret = snd_card_create(SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1,
card->owner, 0, &card->snd_card);
@@ -1657,6 +1803,15 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card)
}
}
+ for (i = 0; i < card->num_aux_devs; i++) {
+ ret = soc_probe_aux_dev(card, i);
+ if (ret < 0) {
+ pr_err("asoc: failed to add auxiliary devices %s: %d\n",
+ card->name, ret);
+ goto probe_aux_dev_err;
+ }
+ }
+
snprintf(card->snd_card->shortname, sizeof(card->snd_card->shortname),
"%s", card->name);
snprintf(card->snd_card->longname, sizeof(card->snd_card->longname),
@@ -1665,24 +1820,30 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card)
ret = snd_card_register(card->snd_card);
if (ret < 0) {
printk(KERN_ERR "asoc: failed to register soundcard for %s\n", card->name);
- goto probe_dai_err;
+ goto probe_aux_dev_err;
}
#ifdef CONFIG_SND_SOC_AC97_BUS
/* register any AC97 codecs */
for (i = 0; i < card->num_rtd; i++) {
- ret = soc_register_ac97_dai_link(&card->rtd[i]);
- if (ret < 0) {
- printk(KERN_ERR "asoc: failed to register AC97 %s\n", card->name);
- goto probe_dai_err;
- }
+ ret = soc_register_ac97_dai_link(&card->rtd[i]);
+ if (ret < 0) {
+ printk(KERN_ERR "asoc: failed to register AC97 %s\n", card->name);
+ while (--i >= 0)
+ soc_unregister_ac97_dai_link(card->rtd[i].codec);
+ goto probe_aux_dev_err;
}
+ }
#endif
card->instantiated = 1;
mutex_unlock(&card->mutex);
return;
+probe_aux_dev_err:
+ for (i = 0; i < card->num_aux_devs; i++)
+ soc_remove_aux_dev(card, i);
+
probe_dai_err:
for (i = 0; i < card->num_links; i++)
soc_remove_dai_link(card, i);
@@ -1741,9 +1902,13 @@ static int soc_remove(struct platform_device *pdev)
/* make sure any delayed work runs */
for (i = 0; i < card->num_rtd; i++) {
struct snd_soc_pcm_runtime *rtd = &card->rtd[i];
- run_delayed_work(&rtd->delayed_work);
+ flush_delayed_work_sync(&rtd->delayed_work);
}
+ /* remove auxiliary devices */
+ for (i = 0; i < card->num_aux_devs; i++)
+ soc_remove_aux_dev(card, i);
+
/* remove and free each DAI */
for (i = 0; i < card->num_rtd; i++)
soc_remove_dai_link(card, i);
@@ -1774,7 +1939,7 @@ static int soc_poweroff(struct device *dev)
* now, we're shutting down so no imminent restart. */
for (i = 0; i < card->num_rtd; i++) {
struct snd_soc_pcm_runtime *rtd = &card->rtd[i];
- run_delayed_work(&rtd->delayed_work);
+ flush_delayed_work_sync(&rtd->delayed_work);
}
snd_soc_dapm_shutdown(card);
@@ -1937,7 +2102,7 @@ unsigned int snd_soc_read(struct snd_soc_codec *codec, unsigned int reg)
{
unsigned int ret;
- ret = codec->driver->read(codec, reg);
+ ret = codec->read(codec, reg);
dev_dbg(codec->dev, "read %x => %x\n", reg, ret);
trace_snd_soc_reg_read(codec, reg, ret);
@@ -1950,7 +2115,7 @@ unsigned int snd_soc_write(struct snd_soc_codec *codec,
{
dev_dbg(codec->dev, "write %x = %x\n", reg, val);
trace_snd_soc_reg_write(codec, reg, val);
- return codec->driver->write(codec, reg, val);
+ return codec->write(codec, reg, val);
}
EXPORT_SYMBOL_GPL(snd_soc_write);
@@ -2946,10 +3111,12 @@ static int snd_soc_register_card(struct snd_soc_card *card)
if (!card->name || !card->dev)
return -EINVAL;
- card->rtd = kzalloc(sizeof(struct snd_soc_pcm_runtime) * card->num_links,
- GFP_KERNEL);
+ card->rtd = kzalloc(sizeof(struct snd_soc_pcm_runtime) *
+ (card->num_links + card->num_aux_devs),
+ GFP_KERNEL);
if (card->rtd == NULL)
return -ENOMEM;
+ card->rtd_aux = &card->rtd[card->num_links];
for (i = 0; i < card->num_links; i++)
card->rtd[i].dai_link = &card->dai_link[i];
@@ -2991,7 +3158,7 @@ static int snd_soc_unregister_card(struct snd_soc_card *card)
* Simplify DAI link configuration by removing ".-1" from device names
* and sanitizing names.
*/
-static inline char *fmt_single_name(struct device *dev, int *id)
+static char *fmt_single_name(struct device *dev, int *id)
{
char *found, name[NAME_SIZE];
int id1, id2;
@@ -2999,7 +3166,7 @@ static inline char *fmt_single_name(struct device *dev, int *id)
if (dev_name(dev) == NULL)
return NULL;
- strncpy(name, dev_name(dev), NAME_SIZE);
+ strlcpy(name, dev_name(dev), NAME_SIZE);
/* are we a "%s.%d" name (platform and SPI components) */
found = strstr(name, dev->driver->name);
@@ -3022,7 +3189,7 @@ static inline char *fmt_single_name(struct device *dev, int *id)
/* sanitize component name for DAI link creation */
snprintf(tmp, NAME_SIZE, "%s.%s", dev->driver->name, name);
- strncpy(name, tmp, NAME_SIZE);
+ strlcpy(name, tmp, NAME_SIZE);
} else
*id = 0;
}
@@ -3157,7 +3324,9 @@ int snd_soc_register_dais(struct device *dev,
pr_debug("Registered DAI '%s'\n", dai->name);
}
+ mutex_lock(&client_mutex);
snd_soc_instantiate_cards();
+ mutex_unlock(&client_mutex);
return 0;
err:
@@ -3285,9 +3454,11 @@ static void fixup_codec_formats(struct snd_soc_pcm_stream *stream)
* @codec: codec to register
*/
int snd_soc_register_codec(struct device *dev,
- struct snd_soc_codec_driver *codec_drv,
- struct snd_soc_dai_driver *dai_drv, int num_dai)
+ const struct snd_soc_codec_driver *codec_drv,
+ struct snd_soc_dai_driver *dai_drv,
+ int num_dai)
{
+ size_t reg_size;
struct snd_soc_codec *codec;
int ret, i;
@@ -3304,8 +3475,15 @@ int snd_soc_register_codec(struct device *dev,
return -ENOMEM;
}
+ if (codec_drv->compress_type)
+ codec->compress_type = codec_drv->compress_type;
+ else
+ codec->compress_type = SND_SOC_FLAT_COMPRESSION;
+
INIT_LIST_HEAD(&codec->dapm.widgets);
INIT_LIST_HEAD(&codec->dapm.paths);
+ codec->write = codec_drv->write;
+ codec->read = codec_drv->read;
codec->dapm.bias_level = SND_SOC_BIAS_OFF;
codec->dapm.dev = dev;
codec->dapm.codec = codec;
@@ -3316,11 +3494,18 @@ int snd_soc_register_codec(struct device *dev,
/* allocate CODEC register cache */
if (codec_drv->reg_cache_size && codec_drv->reg_word_size) {
- ret = snd_soc_cache_init(codec);
- if (ret < 0) {
- dev_err(codec->dev, "Failed to set cache compression type: %d\n",
- ret);
- goto error_cache;
+ reg_size = codec_drv->reg_cache_size * codec_drv->reg_word_size;
+ /* it is necessary to make a copy of the default register cache
+ * because in the case of using a compression type that requires
+ * the default register cache to be marked as __devinitconst the
+ * kernel might have freed the array by the time we initialize
+ * the cache.
+ */
+ codec->reg_def_copy = kmemdup(codec_drv->reg_cache_default,
+ reg_size, GFP_KERNEL);
+ if (!codec->reg_def_copy) {
+ ret = -ENOMEM;
+ goto fail;
}
}
@@ -3333,7 +3518,7 @@ int snd_soc_register_codec(struct device *dev,
if (num_dai) {
ret = snd_soc_register_dais(dev, dai_drv, num_dai);
if (ret < 0)
- goto error_dais;
+ goto fail;
}
mutex_lock(&client_mutex);
@@ -3344,9 +3529,9 @@ int snd_soc_register_codec(struct device *dev,
pr_debug("Registered codec '%s'\n", codec->name);
return 0;
-error_dais:
- snd_soc_cache_exit(codec);
-error_cache:
+fail:
+ kfree(codec->reg_def_copy);
+ codec->reg_def_copy = NULL;
kfree(codec->name);
kfree(codec);
return ret;
@@ -3381,6 +3566,7 @@ found:
pr_debug("Unregistered codec '%s'\n", codec->name);
snd_soc_cache_exit(codec);
+ kfree(codec->reg_def_copy);
kfree(codec->name);
kfree(codec);
}
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 6a29d59dabaf..3d310af28907 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -62,6 +62,7 @@ static int dapm_up_seq[] = {
[snd_soc_dapm_mixer_named_ctl] = 7,
[snd_soc_dapm_pga] = 8,
[snd_soc_dapm_adc] = 9,
+ [snd_soc_dapm_out_drv] = 10,
[snd_soc_dapm_hp] = 10,
[snd_soc_dapm_spk] = 10,
[snd_soc_dapm_post] = 11,
@@ -72,6 +73,7 @@ static int dapm_down_seq[] = {
[snd_soc_dapm_adc] = 1,
[snd_soc_dapm_hp] = 2,
[snd_soc_dapm_spk] = 2,
+ [snd_soc_dapm_out_drv] = 2,
[snd_soc_dapm_pga] = 4,
[snd_soc_dapm_mixer_named_ctl] = 5,
[snd_soc_dapm_mixer] = 5,
@@ -162,6 +164,10 @@ static int snd_soc_dapm_set_bias_level(struct snd_soc_card *card,
else
dapm->bias_level = level;
}
+ if (ret == 0) {
+ if (card && card->set_bias_level_post)
+ ret = card->set_bias_level_post(card, level);
+ }
trace_snd_soc_bias_level_done(card, level);
@@ -231,6 +237,7 @@ static void dapm_set_path_status(struct snd_soc_dapm_widget *w,
break;
/* does not effect routing - always connected */
case snd_soc_dapm_pga:
+ case snd_soc_dapm_out_drv:
case snd_soc_dapm_output:
case snd_soc_dapm_adc:
case snd_soc_dapm_input:
@@ -977,6 +984,9 @@ static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event)
case SND_SOC_DAPM_STREAM_RESUME:
sys_power = 1;
break;
+ case SND_SOC_DAPM_STREAM_STOP:
+ sys_power = !!dapm->codec->active;
+ break;
case SND_SOC_DAPM_STREAM_SUSPEND:
sys_power = 0;
break;
@@ -1243,6 +1253,7 @@ static ssize_t dapm_widget_show(struct device *dev,
case snd_soc_dapm_dac:
case snd_soc_dapm_adc:
case snd_soc_dapm_pga:
+ case snd_soc_dapm_out_drv:
case snd_soc_dapm_mixer:
case snd_soc_dapm_mixer_named_ctl:
case snd_soc_dapm_supply:
@@ -1421,6 +1432,7 @@ static int snd_soc_dapm_add_route(struct snd_soc_dapm_context *dapm,
case snd_soc_dapm_adc:
case snd_soc_dapm_dac:
case snd_soc_dapm_pga:
+ case snd_soc_dapm_out_drv:
case snd_soc_dapm_input:
case snd_soc_dapm_output:
case snd_soc_dapm_micbias:
@@ -1538,6 +1550,7 @@ int snd_soc_dapm_new_widgets(struct snd_soc_dapm_context *dapm)
w->power_check = dapm_dac_check_power;
break;
case snd_soc_dapm_pga:
+ case snd_soc_dapm_out_drv:
w->power_check = dapm_generic_check_power;
dapm_new_pga(dapm, w);
break;
diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c
index 0e9b0710928b..ac5a5bc7375a 100644
--- a/sound/soc/soc-jack.c
+++ b/sound/soc/soc-jack.c
@@ -17,6 +17,7 @@
#include <linux/interrupt.h>
#include <linux/workqueue.h>
#include <linux/delay.h>
+#include <trace/events/asoc.h>
/**
* snd_soc_jack_new - Create a new jack
@@ -64,6 +65,8 @@ void snd_soc_jack_report(struct snd_soc_jack *jack, int status, int mask)
int enable;
int oldstatus;
+ trace_snd_soc_jack_report(jack, mask, status);
+
if (!jack)
return;
@@ -82,6 +85,8 @@ void snd_soc_jack_report(struct snd_soc_jack *jack, int status, int mask)
if (mask && (jack->status == oldstatus))
goto out;
+ trace_snd_soc_jack_notify(jack, status);
+
list_for_each_entry(pin, &jack->pins, list) {
enable = pin->mask & jack->status;
@@ -210,6 +215,8 @@ static irqreturn_t gpio_handler(int irq, void *data)
struct snd_soc_jack_gpio *gpio = data;
struct device *dev = gpio->jack->codec->card->dev;
+ trace_snd_soc_jack_irq(gpio->name);
+
if (device_may_wakeup(dev))
pm_wakeup_event(dev, gpio->debounce_time + 50);
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