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authorMark Brown <broonie@opensource.wolfsonmicro.com>2010-01-12 12:59:05 +0000
committerMark Brown <broonie@opensource.wolfsonmicro.com>2010-01-12 12:59:05 +0000
commit163849ea9b4c5d50fbd324692461983d18faadad (patch)
tree7176a18fea53560eb4092aa103d94a266ab4e60d /sound/soc/codecs
parent2138301e1687bd4f22aa2b4df4829b6ffdae19bc (diff)
parent5ee518ecbcb5934e284ea51a19a939c891f5f7ea (diff)
Merge branch 'for-2.6.33' into for-2.6.34
Diffstat (limited to 'sound/soc/codecs')
-rw-r--r--sound/soc/codecs/ac97.c6
-rw-r--r--sound/soc/codecs/tlv320aic23.c2
-rw-r--r--sound/soc/codecs/twl4030.c10
-rw-r--r--sound/soc/codecs/uda134x.c4
-rw-r--r--sound/soc/codecs/wm8350.c27
-rw-r--r--sound/soc/codecs/wm8510.c14
-rw-r--r--sound/soc/codecs/wm8903.c6
-rw-r--r--sound/soc/codecs/wm8940.c14
-rw-r--r--sound/soc/codecs/wm8974.c14
-rw-r--r--sound/soc/codecs/wm8993.c4
-rw-r--r--sound/soc/codecs/wm9712.c3
-rw-r--r--sound/soc/codecs/wm9713.c22
12 files changed, 68 insertions, 58 deletions
diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c
index 69bd0acc81c..a1bbe16b7f9 100644
--- a/sound/soc/codecs/ac97.c
+++ b/sound/soc/codecs/ac97.c
@@ -102,6 +102,12 @@ static int ac97_soc_probe(struct platform_device *pdev)
INIT_LIST_HEAD(&codec->dapm_widgets);
INIT_LIST_HEAD(&codec->dapm_paths);
+ ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, 0);
+ if (ret < 0) {
+ printk(KERN_ERR "ASoC: failed to init gen ac97 glue\n");
+ goto err;
+ }
+
/* register pcms */
ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
if (ret < 0)
diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c
index 09d1af85b6f..a9dc5fb5477 100644
--- a/sound/soc/codecs/tlv320aic23.c
+++ b/sound/soc/codecs/tlv320aic23.c
@@ -85,7 +85,7 @@ static int tlv320aic23_write(struct snd_soc_codec *codec, unsigned int reg,
* of data into val
*/
- if ((reg < 0 || reg > 9) && (reg != 15)) {
+ if (reg > 9 && reg != 15) {
printk(KERN_WARNING "%s Invalid register R%u\n", __func__, reg);
return -1;
}
diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c
index 5f1681f6ca7..2a27f7b5672 100644
--- a/sound/soc/codecs/twl4030.c
+++ b/sound/soc/codecs/twl4030.c
@@ -26,7 +26,7 @@
#include <linux/pm.h>
#include <linux/i2c.h>
#include <linux/platform_device.h>
-#include <linux/i2c/twl4030.h>
+#include <linux/i2c/twl.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
@@ -175,7 +175,7 @@ static int twl4030_write(struct snd_soc_codec *codec,
{
twl4030_write_reg_cache(codec, reg, value);
if (likely(reg < TWL4030_REG_SW_SHADOW))
- return twl4030_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE, value,
+ return twl_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE, value,
reg);
else
return 0;
@@ -261,7 +261,7 @@ static void twl4030_power_up(struct snd_soc_codec *codec)
do {
/* this takes a little while, so don't slam i2c */
udelay(2000);
- twl4030_i2c_read_u8(TWL4030_MODULE_AUDIO_VOICE, &byte,
+ twl_i2c_read_u8(TWL4030_MODULE_AUDIO_VOICE, &byte,
TWL4030_REG_ANAMICL);
} while ((i++ < 100) &&
((byte & TWL4030_CNCL_OFFSET_START) ==
@@ -542,7 +542,7 @@ static int pin_name##pga_event(struct snd_soc_dapm_widget *w, \
break; \
case SND_SOC_DAPM_POST_PMD: \
reg_val = twl4030_read_reg_cache(w->codec, reg); \
- twl4030_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE, \
+ twl_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE, \
reg_val & (~mask), \
reg); \
break; \
@@ -679,7 +679,7 @@ static void headset_ramp(struct snd_soc_codec *codec, int ramp)
mdelay((ramp_base[(hs_pop & TWL4030_RAMP_DELAY) >> 2] /
twl4030->sysclk) + 1);
/* Bypass the reg_cache to mute the headset */
- twl4030_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE,
+ twl_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE,
hs_gain & (~0x0f),
TWL4030_REG_HS_GAIN_SET);
diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c
index aa40d985138..3e99fe5131d 100644
--- a/sound/soc/codecs/uda134x.c
+++ b/sound/soc/codecs/uda134x.c
@@ -101,7 +101,7 @@ static int uda134x_write(struct snd_soc_codec *codec, unsigned int reg,
pr_debug("%s reg: %02X, value:%02X\n", __func__, reg, value);
if (reg >= UDA134X_REGS_NUM) {
- printk(KERN_ERR "%s unkown register: reg: %u",
+ printk(KERN_ERR "%s unknown register: reg: %u",
__func__, reg);
return -EINVAL;
}
@@ -552,7 +552,7 @@ static int uda134x_soc_probe(struct platform_device *pdev)
ARRAY_SIZE(uda1341_snd_controls));
break;
default:
- printk(KERN_ERR "%s unkown codec type: %d",
+ printk(KERN_ERR "%s unknown codec type: %d",
__func__, pd->model);
return -EINVAL;
}
diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c
index f82125d9e85..718ef912e75 100644
--- a/sound/soc/codecs/wm8350.c
+++ b/sound/soc/codecs/wm8350.c
@@ -925,7 +925,7 @@ static int wm8350_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
iface |= 0x3 << 8;
break;
case SND_SOC_DAIFMT_DSP_B:
- iface |= 0x3 << 8; /* lg not sure which mode */
+ iface |= 0x3 << 8 | WM8350_AIF_LRCLK_INV;
break;
default:
return -EINVAL;
@@ -1340,9 +1340,10 @@ static int wm8350_resume(struct platform_device *pdev)
return 0;
}
-static void wm8350_hp_jack_handler(struct wm8350 *wm8350, int irq, void *data)
+static irqreturn_t wm8350_hp_jack_handler(int irq, void *data)
{
struct wm8350_data *priv = data;
+ struct wm8350 *wm8350 = priv->codec.control_data;
u16 reg;
int report;
int mask;
@@ -1365,7 +1366,7 @@ static void wm8350_hp_jack_handler(struct wm8350 *wm8350, int irq, void *data)
if (!jack->jack) {
dev_warn(wm8350->dev, "Jack interrupt called with no jack\n");
- return;
+ return IRQ_NONE;
}
/* Debounce */
@@ -1378,6 +1379,8 @@ static void wm8350_hp_jack_handler(struct wm8350 *wm8350, int irq, void *data)
report = 0;
snd_soc_jack_report(jack->jack, report, jack->report);
+
+ return IRQ_HANDLED;
}
/**
@@ -1421,9 +1424,7 @@ int wm8350_hp_jack_detect(struct snd_soc_codec *codec, enum wm8350_jack which,
wm8350_set_bits(wm8350, WM8350_JACK_DETECT, ena);
/* Sync status */
- wm8350_hp_jack_handler(wm8350, irq, priv);
-
- wm8350_unmask_irq(wm8350, irq);
+ wm8350_hp_jack_handler(irq, priv);
return 0;
}
@@ -1482,12 +1483,16 @@ static int wm8350_probe(struct platform_device *pdev)
wm8350_set_bits(wm8350, WM8350_ROUT2_VOLUME,
WM8350_OUT2_VU | WM8350_OUT2R_MUTE);
- wm8350_mask_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_L);
- wm8350_mask_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_R);
+ /* Make sure jack detect is disabled to start off with */
+ wm8350_clear_bits(wm8350, WM8350_JACK_DETECT,
+ WM8350_JDL_ENA | WM8350_JDR_ENA);
+
wm8350_register_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_L,
- wm8350_hp_jack_handler, priv);
+ wm8350_hp_jack_handler, 0, "Left jack detect",
+ priv);
wm8350_register_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_R,
- wm8350_hp_jack_handler, priv);
+ wm8350_hp_jack_handler, 0, "Right jack detect",
+ priv);
ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
if (ret < 0) {
@@ -1516,8 +1521,6 @@ static int wm8350_remove(struct platform_device *pdev)
WM8350_JDL_ENA | WM8350_JDR_ENA);
wm8350_clear_bits(wm8350, WM8350_POWER_MGMT_4, WM8350_TOCLK_ENA);
- wm8350_mask_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_L);
- wm8350_mask_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_R);
wm8350_free_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_L);
wm8350_free_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_R);
diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c
index 265e68c75df..af8cb6995a1 100644
--- a/sound/soc/codecs/wm8510.c
+++ b/sound/soc/codecs/wm8510.c
@@ -424,23 +424,23 @@ static int wm8510_pcm_hw_params(struct snd_pcm_substream *substream,
/* filter coefficient */
switch (params_rate(params)) {
- case SNDRV_PCM_RATE_8000:
+ case 8000:
adn |= 0x5 << 1;
break;
- case SNDRV_PCM_RATE_11025:
+ case 11025:
adn |= 0x4 << 1;
break;
- case SNDRV_PCM_RATE_16000:
+ case 16000:
adn |= 0x3 << 1;
break;
- case SNDRV_PCM_RATE_22050:
+ case 22050:
adn |= 0x2 << 1;
break;
- case SNDRV_PCM_RATE_32000:
+ case 32000:
adn |= 0x1 << 1;
break;
- case SNDRV_PCM_RATE_44100:
- case SNDRV_PCM_RATE_48000:
+ case 44100:
+ case 48000:
break;
}
diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c
index b8cae175864..ce5515e3f2b 100644
--- a/sound/soc/codecs/wm8903.c
+++ b/sound/soc/codecs/wm8903.c
@@ -607,7 +607,7 @@ SOC_SINGLE("Right Input PGA Common Mode Switch", WM8903_ANALOGUE_RIGHT_INPUT_1,
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),
-SOC_SINGLE_TLV("DRC Compressor Threashold Volume", WM8903_DRC_3, 5, 124, 1,
+SOC_SINGLE_TLV("DRC Compressor Threshold Volume", WM8903_DRC_3, 5, 124, 1,
drc_tlv_thresh),
SOC_SINGLE_TLV("DRC Volume", WM8903_DRC_3, 0, 30, 1, drc_tlv_amp),
SOC_SINGLE_TLV("DRC Minimum Gain Volume", WM8903_DRC_1, 2, 3, 1, drc_tlv_min),
@@ -617,11 +617,11 @@ SOC_ENUM("DRC Decay Rate", drc_decay),
SOC_ENUM("DRC FF Delay", drc_ff_delay),
SOC_SINGLE("DRC Anticlip Switch", WM8903_DRC_0, 1, 1, 0),
SOC_SINGLE("DRC QR Switch", WM8903_DRC_0, 2, 1, 0),
-SOC_SINGLE_TLV("DRC QR Threashold Volume", WM8903_DRC_0, 6, 3, 0, drc_tlv_max),
+SOC_SINGLE_TLV("DRC QR Threshold Volume", WM8903_DRC_0, 6, 3, 0, drc_tlv_max),
SOC_ENUM("DRC QR Decay Rate", drc_qr_decay),
SOC_SINGLE("DRC Smoothing Switch", WM8903_DRC_0, 3, 1, 0),
SOC_SINGLE("DRC Smoothing Hysteresis Switch", WM8903_DRC_0, 0, 1, 0),
-SOC_ENUM("DRC Smoothing Threashold", drc_smoothing),
+SOC_ENUM("DRC Smoothing Threshold", drc_smoothing),
SOC_SINGLE_TLV("DRC Startup Volume", WM8903_DRC_0, 6, 18, 0, drc_tlv_startup),
SOC_DOUBLE_R_TLV("Digital Capture Volume", WM8903_ADC_DIGITAL_VOLUME_LEFT,
diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c
index 3d850b97037..31e39ffd1d8 100644
--- a/sound/soc/codecs/wm8940.c
+++ b/sound/soc/codecs/wm8940.c
@@ -378,23 +378,23 @@ static int wm8940_i2s_hw_params(struct snd_pcm_substream *substream,
iface |= (1 << 9);
switch (params_rate(params)) {
- case SNDRV_PCM_RATE_8000:
+ case 8000:
addcntrl |= (0x5 << 1);
break;
- case SNDRV_PCM_RATE_11025:
+ case 11025:
addcntrl |= (0x4 << 1);
break;
- case SNDRV_PCM_RATE_16000:
+ case 16000:
addcntrl |= (0x3 << 1);
break;
- case SNDRV_PCM_RATE_22050:
+ case 22050:
addcntrl |= (0x2 << 1);
break;
- case SNDRV_PCM_RATE_32000:
+ case 32000:
addcntrl |= (0x1 << 1);
break;
- case SNDRV_PCM_RATE_44100:
- case SNDRV_PCM_RATE_48000:
+ case 44100:
+ case 48000:
break;
}
ret = snd_soc_write(codec, WM8940_ADDCNTRL, addcntrl);
diff --git a/sound/soc/codecs/wm8974.c b/sound/soc/codecs/wm8974.c
index a808675388f..8812751da8c 100644
--- a/sound/soc/codecs/wm8974.c
+++ b/sound/soc/codecs/wm8974.c
@@ -482,23 +482,23 @@ static int wm8974_pcm_hw_params(struct snd_pcm_substream *substream,
/* filter coefficient */
switch (params_rate(params)) {
- case SNDRV_PCM_RATE_8000:
+ case 8000:
adn |= 0x5 << 1;
break;
- case SNDRV_PCM_RATE_11025:
+ case 11025:
adn |= 0x4 << 1;
break;
- case SNDRV_PCM_RATE_16000:
+ case 16000:
adn |= 0x3 << 1;
break;
- case SNDRV_PCM_RATE_22050:
+ case 22050:
adn |= 0x2 << 1;
break;
- case SNDRV_PCM_RATE_32000:
+ case 32000:
adn |= 0x1 << 1;
break;
- case SNDRV_PCM_RATE_44100:
- case SNDRV_PCM_RATE_48000:
+ case 44100:
+ case 48000:
break;
}
diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c
index cd2bc05f78c..828d8174d5b 100644
--- a/sound/soc/codecs/wm8993.c
+++ b/sound/soc/codecs/wm8993.c
@@ -691,7 +691,7 @@ SOC_DOUBLE_TLV("Digital Sidetone Volume", WM8993_DIGITAL_SIDE_TONE,
SOC_SINGLE("DRC Switch", WM8993_DRC_CONTROL_1, 15, 1, 0),
SOC_ENUM("DRC Path", drc_path),
-SOC_SINGLE_TLV("DRC Compressor Threashold Volume", WM8993_DRC_CONTROL_2,
+SOC_SINGLE_TLV("DRC Compressor Threshold Volume", WM8993_DRC_CONTROL_2,
2, 60, 1, drc_comp_threash),
SOC_SINGLE_TLV("DRC Compressor Amplitude Volume", WM8993_DRC_CONTROL_3,
11, 30, 1, drc_comp_amp),
@@ -711,7 +711,7 @@ SOC_SINGLE_TLV("DRC Quick Release Volume", WM8993_DRC_CONTROL_3, 2, 3, 0,
SOC_ENUM("DRC Quick Release Rate", drc_qr_rate),
SOC_SINGLE("DRC Smoothing Switch", WM8993_DRC_CONTROL_1, 11, 1, 0),
SOC_SINGLE("DRC Smoothing Hysteresis Switch", WM8993_DRC_CONTROL_1, 8, 1, 0),
-SOC_ENUM("DRC Smoothing Hysteresis Threashold", drc_smooth),
+SOC_ENUM("DRC Smoothing Hysteresis Threshold", drc_smooth),
SOC_SINGLE_TLV("DRC Startup Volume", WM8993_DRC_CONTROL_4, 8, 18, 0,
drc_startup_tlv),
diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c
index 0ac1215dcd9..e237bf61512 100644
--- a/sound/soc/codecs/wm9712.c
+++ b/sound/soc/codecs/wm9712.c
@@ -463,7 +463,8 @@ static int ac97_write(struct snd_soc_codec *codec, unsigned int reg,
{
u16 *cache = codec->reg_cache;
- soc_ac97_ops.write(codec->ac97, reg, val);
+ if (reg < 0x7c)
+ soc_ac97_ops.write(codec->ac97, reg, val);
reg = reg >> 1;
if (reg < (ARRAY_SIZE(wm9712_reg)))
cache[reg] = val;
diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c
index 4d74ecb0e56..c58aab375ed 100644
--- a/sound/soc/codecs/wm9713.c
+++ b/sound/soc/codecs/wm9713.c
@@ -165,9 +165,9 @@ SOC_SINGLE("Mono Playback Switch", AC97_MASTER_TONE, 7, 1, 1),
SOC_SINGLE("Mono Playback ZC Switch", AC97_MASTER_TONE, 6, 1, 0),
SOC_SINGLE("Mono Playback Volume", AC97_MASTER_TONE, 0, 31, 1),
-SOC_SINGLE("PC Beep Playback Headphone Volume", AC97_AUX, 12, 7, 1),
-SOC_SINGLE("PC Beep Playback Speaker Volume", AC97_AUX, 8, 7, 1),
-SOC_SINGLE("PC Beep Playback Mono Volume", AC97_AUX, 4, 7, 1),
+SOC_SINGLE("Beep Playback Headphone Volume", AC97_AUX, 12, 7, 1),
+SOC_SINGLE("Beep Playback Speaker Volume", AC97_AUX, 8, 7, 1),
+SOC_SINGLE("Beep Playback Mono Volume", AC97_AUX, 4, 7, 1),
SOC_SINGLE("Voice Playback Headphone Volume", AC97_PCM, 12, 7, 1),
SOC_SINGLE("Voice Playback Master Volume", AC97_PCM, 8, 7, 1),
@@ -266,7 +266,7 @@ static int mixer_event(struct snd_soc_dapm_widget *w,
/* Left Headphone Mixers */
static const struct snd_kcontrol_new wm9713_hpl_mixer_controls[] = {
-SOC_DAPM_SINGLE("PC Beep Playback Switch", HPL_MIXER, 5, 1, 0),
+SOC_DAPM_SINGLE("Beep Playback Switch", HPL_MIXER, 5, 1, 0),
SOC_DAPM_SINGLE("Voice Playback Switch", HPL_MIXER, 4, 1, 0),
SOC_DAPM_SINGLE("Aux Playback Switch", HPL_MIXER, 3, 1, 0),
SOC_DAPM_SINGLE("PCM Playback Switch", HPL_MIXER, 2, 1, 0),
@@ -276,7 +276,7 @@ SOC_DAPM_SINGLE("Bypass Playback Switch", HPL_MIXER, 0, 1, 0),
/* Right Headphone Mixers */
static const struct snd_kcontrol_new wm9713_hpr_mixer_controls[] = {
-SOC_DAPM_SINGLE("PC Beep Playback Switch", HPR_MIXER, 5, 1, 0),
+SOC_DAPM_SINGLE("Beep Playback Switch", HPR_MIXER, 5, 1, 0),
SOC_DAPM_SINGLE("Voice Playback Switch", HPR_MIXER, 4, 1, 0),
SOC_DAPM_SINGLE("Aux Playback Switch", HPR_MIXER, 3, 1, 0),
SOC_DAPM_SINGLE("PCM Playback Switch", HPR_MIXER, 2, 1, 0),
@@ -294,7 +294,7 @@ SOC_DAPM_ENUM("Route", wm9713_enum[0]);
/* Speaker Mixer */
static const struct snd_kcontrol_new wm9713_speaker_mixer_controls[] = {
-SOC_DAPM_SINGLE("PC Beep Playback Switch", AC97_AUX, 11, 1, 1),
+SOC_DAPM_SINGLE("Beep Playback Switch", AC97_AUX, 11, 1, 1),
SOC_DAPM_SINGLE("Voice Playback Switch", AC97_PCM, 11, 1, 1),
SOC_DAPM_SINGLE("Aux Playback Switch", AC97_REC_SEL, 11, 1, 1),
SOC_DAPM_SINGLE("PCM Playback Switch", AC97_PHONE, 14, 1, 1),
@@ -304,7 +304,7 @@ SOC_DAPM_SINGLE("Bypass Playback Switch", AC97_PC_BEEP, 14, 1, 1),
/* Mono Mixer */
static const struct snd_kcontrol_new wm9713_mono_mixer_controls[] = {
-SOC_DAPM_SINGLE("PC Beep Playback Switch", AC97_AUX, 7, 1, 1),
+SOC_DAPM_SINGLE("Beep Playback Switch", AC97_AUX, 7, 1, 1),
SOC_DAPM_SINGLE("Voice Playback Switch", AC97_PCM, 7, 1, 1),
SOC_DAPM_SINGLE("Aux Playback Switch", AC97_REC_SEL, 7, 1, 1),
SOC_DAPM_SINGLE("PCM Playback Switch", AC97_PHONE, 13, 1, 1),
@@ -463,7 +463,7 @@ SND_SOC_DAPM_VMID("VMID"),
static const struct snd_soc_dapm_route audio_map[] = {
/* left HP mixer */
- {"Left HP Mixer", "PC Beep Playback Switch", "PCBEEP"},
+ {"Left HP Mixer", "Beep Playback Switch", "PCBEEP"},
{"Left HP Mixer", "Voice Playback Switch", "Voice DAC"},
{"Left HP Mixer", "Aux Playback Switch", "Aux DAC"},
{"Left HP Mixer", "Bypass Playback Switch", "Left Line In"},
@@ -472,7 +472,7 @@ static const struct snd_soc_dapm_route audio_map[] = {
{"Left HP Mixer", NULL, "Capture Headphone Mux"},
/* right HP mixer */
- {"Right HP Mixer", "PC Beep Playback Switch", "PCBEEP"},
+ {"Right HP Mixer", "Beep Playback Switch", "PCBEEP"},
{"Right HP Mixer", "Voice Playback Switch", "Voice DAC"},
{"Right HP Mixer", "Aux Playback Switch", "Aux DAC"},
{"Right HP Mixer", "Bypass Playback Switch", "Right Line In"},
@@ -491,7 +491,7 @@ static const struct snd_soc_dapm_route audio_map[] = {
{"Capture Mixer", NULL, "Right Capture Source"},
/* speaker mixer */
- {"Speaker Mixer", "PC Beep Playback Switch", "PCBEEP"},
+ {"Speaker Mixer", "Beep Playback Switch", "PCBEEP"},
{"Speaker Mixer", "Voice Playback Switch", "Voice DAC"},
{"Speaker Mixer", "Aux Playback Switch", "Aux DAC"},
{"Speaker Mixer", "Bypass Playback Switch", "Line Mixer"},
@@ -499,7 +499,7 @@ static const struct snd_soc_dapm_route audio_map[] = {
{"Speaker Mixer", "MonoIn Playback Switch", "Mono In"},
/* mono mixer */
- {"Mono Mixer", "PC Beep Playback Switch", "PCBEEP"},
+ {"Mono Mixer", "Beep Playback Switch", "PCBEEP"},
{"Mono Mixer", "Voice Playback Switch", "Voice DAC"},
{"Mono Mixer", "Aux Playback Switch", "Aux DAC"},
{"Mono Mixer", "Bypass Playback Switch", "Line Mixer"},