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-rw-r--r--sound/soc/codecs/Kconfig4
-rw-r--r--sound/soc/codecs/Makefile2
-rw-r--r--sound/soc/codecs/ak4641.c664
-rw-r--r--sound/soc/codecs/ak4641.h47
-rw-r--r--sound/soc/codecs/dmic.c26
-rw-r--r--sound/soc/codecs/max98088.c62
-rw-r--r--sound/soc/codecs/max98088.h13
-rw-r--r--sound/soc/codecs/max98095.c16
-rw-r--r--sound/soc/codecs/spdif_transciever.c8
-rw-r--r--sound/soc/codecs/ssm2602.c2
-rw-r--r--sound/soc/codecs/tlv320aic3x.c3
-rw-r--r--sound/soc/codecs/tlv320dac33.c4
-rw-r--r--sound/soc/codecs/tlv320dac33.h2
-rw-r--r--sound/soc/codecs/tpa6130a2.c4
-rw-r--r--sound/soc/codecs/tpa6130a2.h2
-rw-r--r--sound/soc/codecs/twl6040.c6
-rw-r--r--sound/soc/codecs/wm1250-ev1.c2
-rw-r--r--sound/soc/codecs/wm8731.c2
-rw-r--r--sound/soc/codecs/wm8903.c3
-rw-r--r--sound/soc/codecs/wm8915.c1
-rw-r--r--sound/soc/codecs/wm8958-dsp2.c29
-rw-r--r--sound/soc/codecs/wm8993.c3
-rw-r--r--sound/soc/codecs/wm8994.c31
-rw-r--r--sound/soc/codecs/wm8995.c4
-rw-r--r--sound/soc/codecs/wm_hubs.c32
25 files changed, 905 insertions, 67 deletions
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index 2a69718..98175a0 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -20,6 +20,7 @@ config SND_SOC_ALL_CODECS
select SND_SOC_ADS117X
select SND_SOC_AK4104 if SPI_MASTER
select SND_SOC_AK4535 if I2C
+ select SND_SOC_AK4641 if I2C
select SND_SOC_AK4642 if I2C
select SND_SOC_AK4671 if I2C
select SND_SOC_ALC5623 if I2C
@@ -139,6 +140,9 @@ config SND_SOC_AK4104
config SND_SOC_AK4535
tristate
+config SND_SOC_AK4641
+ tristate
+
config SND_SOC_AK4642
tristate
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index 4cb2f42..fd85584 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -7,6 +7,7 @@ snd-soc-ad73311-objs := ad73311.o
snd-soc-ads117x-objs := ads117x.o
snd-soc-ak4104-objs := ak4104.o
snd-soc-ak4535-objs := ak4535.o
+snd-soc-ak4641-objs := ak4641.o
snd-soc-ak4642-objs := ak4642.o
snd-soc-ak4671-objs := ak4671.o
snd-soc-cq93vc-objs := cq93vc.o
@@ -97,6 +98,7 @@ obj-$(CONFIG_SND_SOC_AD73311) += snd-soc-ad73311.o
obj-$(CONFIG_SND_SOC_ADS117X) += snd-soc-ads117x.o
obj-$(CONFIG_SND_SOC_AK4104) += snd-soc-ak4104.o
obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o
+obj-$(CONFIG_SND_SOC_AK4641) += snd-soc-ak4641.o
obj-$(CONFIG_SND_SOC_AK4642) += snd-soc-ak4642.o
obj-$(CONFIG_SND_SOC_AK4671) += snd-soc-ak4671.o
obj-$(CONFIG_SND_SOC_ALC5623) += snd-soc-alc5623.o
diff --git a/sound/soc/codecs/ak4641.c b/sound/soc/codecs/ak4641.c
new file mode 100644
index 0000000..ed96f247c
--- /dev/null
+++ b/sound/soc/codecs/ak4641.c
@@ -0,0 +1,664 @@
+/*
+ * ak4641.c -- AK4641 ALSA Soc Audio driver
+ *
+ * Copyright (C) 2008 Harald Welte <laforge@gnufiish.org>
+ * Copyright (C) 2011 Dmitry Artamonow <mad_soft@inbox.ru>
+ *
+ * Based on ak4535.c by Richard Purdie
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#include <linux/module.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/gpio.h>
+#include <linux/pm.h>
+#include <linux/i2c.h>
+#include <linux/platform_device.h>
+#include <linux/slab.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/initval.h>
+#include <sound/tlv.h>
+#include <sound/ak4641.h>
+
+#include "ak4641.h"
+
+/* codec private data */
+struct ak4641_priv {
+ struct snd_soc_codec *codec;
+ unsigned int sysclk;
+ int deemph;
+ int playback_fs;
+};
+
+/*
+ * ak4641 register cache
+ */
+static const u8 ak4641_reg[AK4641_CACHEREGNUM] = {
+ 0x00, 0x80, 0x00, 0x80,
+ 0x02, 0x00, 0x11, 0x05,
+ 0x00, 0x00, 0x36, 0x10,
+ 0x00, 0x00, 0x57, 0x00,
+ 0x88, 0x88, 0x08, 0x08
+};
+
+static const int deemph_settings[] = {44100, 0, 48000, 32000};
+
+static int ak4641_set_deemph(struct snd_soc_codec *codec)
+{
+ struct ak4641_priv *ak4641 = snd_soc_codec_get_drvdata(codec);
+ int i, best = 0;
+
+ for (i = 0 ; i < ARRAY_SIZE(deemph_settings); i++) {
+ /* if deemphasis is on, select the nearest available rate */
+ if (ak4641->deemph && deemph_settings[i] != 0 &&
+ abs(deemph_settings[i] - ak4641->playback_fs) <
+ abs(deemph_settings[best] - ak4641->playback_fs))
+ best = i;
+
+ if (!ak4641->deemph && deemph_settings[i] == 0)
+ best = i;
+ }
+
+ dev_dbg(codec->dev, "Set deemphasis %d\n", best);
+
+ return snd_soc_update_bits(codec, AK4641_DAC, 0x3, best);
+}
+
+static int ak4641_put_deemph(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct ak4641_priv *ak4641 = snd_soc_codec_get_drvdata(codec);
+ int deemph = ucontrol->value.enumerated.item[0];
+
+ if (deemph > 1)
+ return -EINVAL;
+
+ ak4641->deemph = deemph;
+
+ return ak4641_set_deemph(codec);
+}
+
+static int ak4641_get_deemph(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct ak4641_priv *ak4641 = snd_soc_codec_get_drvdata(codec);
+
+ ucontrol->value.enumerated.item[0] = ak4641->deemph;
+ return 0;
+};
+
+static const char *ak4641_mono_out[] = {"(L + R)/2", "Hi-Z"};
+static const char *ak4641_hp_out[] = {"Stereo", "Mono"};
+static const char *ak4641_mic_select[] = {"Internal", "External"};
+static const char *ak4641_mic_or_dac[] = {"Microphone", "Voice DAC"};
+
+
+static const DECLARE_TLV_DB_SCALE(mono_gain_tlv, -1700, 2300, 0);
+static const DECLARE_TLV_DB_SCALE(mic_boost_tlv, 0, 2000, 0);
+static const DECLARE_TLV_DB_SCALE(eq_tlv, -1050, 150, 0);
+static const DECLARE_TLV_DB_SCALE(master_tlv, -12750, 50, 0);
+static const DECLARE_TLV_DB_SCALE(mic_stereo_sidetone_tlv, -2700, 300, 0);
+static const DECLARE_TLV_DB_SCALE(mic_mono_sidetone_tlv, -400, 400, 0);
+static const DECLARE_TLV_DB_SCALE(capture_tlv, -800, 50, 0);
+static const DECLARE_TLV_DB_SCALE(alc_tlv, -800, 50, 0);
+static const DECLARE_TLV_DB_SCALE(aux_in_tlv, -2100, 300, 0);
+
+
+static const struct soc_enum ak4641_mono_out_enum =
+ SOC_ENUM_SINGLE(AK4641_SIG1, 6, 2, ak4641_mono_out);
+static const struct soc_enum ak4641_hp_out_enum =
+ SOC_ENUM_SINGLE(AK4641_MODE2, 2, 2, ak4641_hp_out);
+static const struct soc_enum ak4641_mic_select_enum =
+ SOC_ENUM_SINGLE(AK4641_MIC, 1, 2, ak4641_mic_select);
+static const struct soc_enum ak4641_mic_or_dac_enum =
+ SOC_ENUM_SINGLE(AK4641_BTIF, 4, 2, ak4641_mic_or_dac);
+
+static const struct snd_kcontrol_new ak4641_snd_controls[] = {
+ SOC_ENUM("Mono 1 Output", ak4641_mono_out_enum),
+ SOC_SINGLE_TLV("Mono 1 Gain Volume", AK4641_SIG1, 7, 1, 1,
+ mono_gain_tlv),
+ SOC_ENUM("Headphone Output", ak4641_hp_out_enum),
+ SOC_SINGLE_BOOL_EXT("Playback Deemphasis Switch", 0,
+ ak4641_get_deemph, ak4641_put_deemph),
+
+ SOC_SINGLE_TLV("Mic Boost Volume", AK4641_MIC, 0, 1, 0, mic_boost_tlv),
+
+ SOC_SINGLE("ALC Operation Time", AK4641_TIMER, 0, 3, 0),
+ SOC_SINGLE("ALC Recovery Time", AK4641_TIMER, 2, 3, 0),
+ SOC_SINGLE("ALC ZC Time", AK4641_TIMER, 4, 3, 0),
+
+ SOC_SINGLE("ALC 1 Switch", AK4641_ALC1, 5, 1, 0),
+
+ SOC_SINGLE_TLV("ALC Volume", AK4641_ALC2, 0, 71, 0, alc_tlv),
+ SOC_SINGLE("Left Out Enable Switch", AK4641_SIG2, 1, 1, 0),
+ SOC_SINGLE("Right Out Enable Switch", AK4641_SIG2, 0, 1, 0),
+
+ SOC_SINGLE_TLV("Capture Volume", AK4641_PGA, 0, 71, 0, capture_tlv),
+
+ SOC_DOUBLE_R_TLV("Master Playback Volume", AK4641_LATT,
+ AK4641_RATT, 0, 255, 1, master_tlv),
+
+ SOC_SINGLE_TLV("AUX In Volume", AK4641_VOL, 0, 15, 0, aux_in_tlv),
+
+ SOC_SINGLE("Equalizer Switch", AK4641_DAC, 2, 1, 0),
+ SOC_SINGLE_TLV("EQ1 100 Hz Volume", AK4641_EQLO, 0, 15, 1, eq_tlv),
+ SOC_SINGLE_TLV("EQ2 250 Hz Volume", AK4641_EQLO, 4, 15, 1, eq_tlv),
+ SOC_SINGLE_TLV("EQ3 1 kHz Volume", AK4641_EQMID, 0, 15, 1, eq_tlv),
+ SOC_SINGLE_TLV("EQ4 3.5 kHz Volume", AK4641_EQMID, 4, 15, 1, eq_tlv),
+ SOC_SINGLE_TLV("EQ5 10 kHz Volume", AK4641_EQHI, 0, 15, 1, eq_tlv),
+};
+
+/* Mono 1 Mixer */
+static const struct snd_kcontrol_new ak4641_mono1_mixer_controls[] = {
+ SOC_DAPM_SINGLE_TLV("Mic Mono Sidetone Volume", AK4641_VOL, 7, 1, 0,
+ mic_mono_sidetone_tlv),
+ SOC_DAPM_SINGLE("Mic Mono Sidetone Switch", AK4641_SIG1, 4, 1, 0),
+ SOC_DAPM_SINGLE("Mono Playback Switch", AK4641_SIG1, 5, 1, 0),
+};
+
+/* Stereo Mixer */
+static const struct snd_kcontrol_new ak4641_stereo_mixer_controls[] = {
+ SOC_DAPM_SINGLE_TLV("Mic Sidetone Volume", AK4641_VOL, 4, 7, 0,
+ mic_stereo_sidetone_tlv),
+ SOC_DAPM_SINGLE("Mic Sidetone Switch", AK4641_SIG2, 4, 1, 0),
+ SOC_DAPM_SINGLE("Playback Switch", AK4641_SIG2, 7, 1, 0),
+ SOC_DAPM_SINGLE("Aux Bypass Switch", AK4641_SIG2, 5, 1, 0),
+};
+
+/* Input Mixer */
+static const struct snd_kcontrol_new ak4641_input_mixer_controls[] = {
+ SOC_DAPM_SINGLE("Mic Capture Switch", AK4641_MIC, 2, 1, 0),
+ SOC_DAPM_SINGLE("Aux Capture Switch", AK4641_MIC, 5, 1, 0),
+};
+
+/* Mic mux */
+static const struct snd_kcontrol_new ak4641_mic_mux_control =
+ SOC_DAPM_ENUM("Mic Select", ak4641_mic_select_enum);
+
+/* Input mux */
+static const struct snd_kcontrol_new ak4641_input_mux_control =
+ SOC_DAPM_ENUM("Input Select", ak4641_mic_or_dac_enum);
+
+/* mono 2 switch */
+static const struct snd_kcontrol_new ak4641_mono2_control =
+ SOC_DAPM_SINGLE("Switch", AK4641_SIG1, 0, 1, 0);
+
+/* ak4641 dapm widgets */
+static const struct snd_soc_dapm_widget ak4641_dapm_widgets[] = {
+ SND_SOC_DAPM_MIXER("Stereo Mixer", SND_SOC_NOPM, 0, 0,
+ &ak4641_stereo_mixer_controls[0],
+ ARRAY_SIZE(ak4641_stereo_mixer_controls)),
+ SND_SOC_DAPM_MIXER("Mono1 Mixer", SND_SOC_NOPM, 0, 0,
+ &ak4641_mono1_mixer_controls[0],
+ ARRAY_SIZE(ak4641_mono1_mixer_controls)),
+ SND_SOC_DAPM_MIXER("Input Mixer", SND_SOC_NOPM, 0, 0,
+ &ak4641_input_mixer_controls[0],
+ ARRAY_SIZE(ak4641_input_mixer_controls)),
+ SND_SOC_DAPM_MUX("Mic Mux", SND_SOC_NOPM, 0, 0,
+ &ak4641_mic_mux_control),
+ SND_SOC_DAPM_MUX("Input Mux", SND_SOC_NOPM, 0, 0,
+ &ak4641_input_mux_control),
+ SND_SOC_DAPM_SWITCH("Mono 2 Enable", SND_SOC_NOPM, 0, 0,
+ &ak4641_mono2_control),
+
+ SND_SOC_DAPM_OUTPUT("LOUT"),
+ SND_SOC_DAPM_OUTPUT("ROUT"),
+ SND_SOC_DAPM_OUTPUT("MOUT1"),
+ SND_SOC_DAPM_OUTPUT("MOUT2"),
+ SND_SOC_DAPM_OUTPUT("MICOUT"),
+
+ SND_SOC_DAPM_ADC("ADC", "HiFi Capture", AK4641_PM1, 0, 0),
+ SND_SOC_DAPM_PGA("Mic", AK4641_PM1, 1, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("AUX In", AK4641_PM1, 2, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("Mono Out", AK4641_PM1, 3, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("Line Out", AK4641_PM1, 4, 0, NULL, 0),
+
+ SND_SOC_DAPM_DAC("DAC", "HiFi Playback", AK4641_PM2, 0, 0),
+ SND_SOC_DAPM_PGA("Mono Out 2", AK4641_PM2, 3, 0, NULL, 0),
+
+ SND_SOC_DAPM_ADC("Voice ADC", "Voice Capture", AK4641_BTIF, 0, 0),
+ SND_SOC_DAPM_ADC("Voice DAC", "Voice Playback", AK4641_BTIF, 1, 0),
+
+ SND_SOC_DAPM_MICBIAS("Mic Int Bias", AK4641_MIC, 3, 0),
+ SND_SOC_DAPM_MICBIAS("Mic Ext Bias", AK4641_MIC, 4, 0),
+
+ SND_SOC_DAPM_INPUT("MICIN"),
+ SND_SOC_DAPM_INPUT("MICEXT"),
+ SND_SOC_DAPM_INPUT("AUX"),
+ SND_SOC_DAPM_INPUT("AIN"),
+};
+
+static const struct snd_soc_dapm_route ak4641_audio_map[] = {
+ /* Stereo Mixer */
+ {"Stereo Mixer", "Playback Switch", "DAC"},
+ {"Stereo Mixer", "Mic Sidetone Switch", "Input Mux"},
+ {"Stereo Mixer", "Aux Bypass Switch", "AUX In"},
+
+ /* Mono 1 Mixer */
+ {"Mono1 Mixer", "Mic Mono Sidetone Switch", "Input Mux"},
+ {"Mono1 Mixer", "Mono Playback Switch", "DAC"},
+
+ /* Mic */
+ {"Mic", NULL, "AIN"},
+ {"Mic Mux", "Internal", "Mic Int Bias"},
+ {"Mic Mux", "External", "Mic Ext Bias"},
+ {"Mic Int Bias", NULL, "MICIN"},
+ {"Mic Ext Bias", NULL, "MICEXT"},
+ {"MICOUT", NULL, "Mic Mux"},
+
+ /* Input Mux */
+ {"Input Mux", "Microphone", "Mic"},
+ {"Input Mux", "Voice DAC", "Voice DAC"},
+
+ /* Line Out */
+ {"LOUT", NULL, "Line Out"},
+ {"ROUT", NULL, "Line Out"},
+ {"Line Out", NULL, "Stereo Mixer"},
+
+ /* Mono 1 Out */
+ {"MOUT1", NULL, "Mono Out"},
+ {"Mono Out", NULL, "Mono1 Mixer"},
+
+ /* Mono 2 Out */
+ {"MOUT2", NULL, "Mono 2 Enable"},
+ {"Mono 2 Enable", "Switch", "Mono Out 2"},
+ {"Mono Out 2", NULL, "Stereo Mixer"},
+
+ {"Voice ADC", NULL, "Mono 2 Enable"},
+
+ /* Aux In */
+ {"AUX In", NULL, "AUX"},
+
+ /* ADC */
+ {"ADC", NULL, "Input Mixer"},
+ {"Input Mixer", "Mic Capture Switch", "Mic"},
+ {"Input Mixer", "Aux Capture Switch", "AUX In"},
+};
+
+static int ak4641_set_dai_sysclk(struct snd_soc_dai *codec_dai,
+ int clk_id, unsigned int freq, int dir)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ struct ak4641_priv *ak4641 = snd_soc_codec_get_drvdata(codec);
+
+ ak4641->sysclk = freq;
+ return 0;
+}
+
+static int ak4641_i2s_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_codec *codec = rtd->codec;
+ struct ak4641_priv *ak4641 = snd_soc_codec_get_drvdata(codec);
+ int rate = params_rate(params), fs = 256;
+ u8 mode2;
+
+ if (rate)
+ fs = ak4641->sysclk / rate;
+ else
+ return -EINVAL;
+
+ /* set fs */
+ switch (fs) {
+ case 1024:
+ mode2 = (0x2 << 5);
+ break;
+ case 512:
+ mode2 = (0x1 << 5);
+ break;
+ case 256:
+ mode2 = (0x0 << 5);
+ break;
+ default:
+ dev_err(codec->dev, "Error: unsupported fs=%d\n", fs);
+ return -EINVAL;
+ }
+
+ snd_soc_update_bits(codec, AK4641_MODE2, (0x3 << 5), mode2);
+
+ /* Update de-emphasis filter for the new rate */
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ ak4641->playback_fs = rate;
+ ak4641_set_deemph(codec);
+ };
+
+ return 0;
+}
+
+static int ak4641_pcm_set_dai_fmt(struct snd_soc_dai *codec_dai,
+ unsigned int fmt)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ u8 btif;
+
+ /* interface format */
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ btif = (0x3 << 5);
+ break;
+ case SND_SOC_DAIFMT_LEFT_J:
+ btif = (0x2 << 5);
+ break;
+ case SND_SOC_DAIFMT_DSP_A: /* MSB after FRM */
+ btif = (0x0 << 5);
+ break;
+ case SND_SOC_DAIFMT_DSP_B: /* MSB during FRM */
+ btif = (0x1 << 5);
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ return snd_soc_update_bits(codec, AK4641_BTIF, (0x3 << 5), btif);
+}
+
+static int ak4641_i2s_set_dai_fmt(struct snd_soc_dai *codec_dai,
+ unsigned int fmt)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ u8 mode1 = 0;
+
+ /* interface format */
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ mode1 = 0x02;
+ break;
+ case SND_SOC_DAIFMT_LEFT_J:
+ mode1 = 0x01;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ return snd_soc_write(codec, AK4641_MODE1, mode1);
+}
+
+static int ak4641_mute(struct snd_soc_dai *dai, int mute)
+{
+ struct snd_soc_codec *codec = dai->codec;
+
+ return snd_soc_update_bits(codec, AK4641_DAC, 0x20, mute ? 0x20 : 0);
+}
+
+static int ak4641_set_bias_level(struct snd_soc_codec *codec,
+ enum snd_soc_bias_level level)
+{
+ struct ak4641_platform_data *pdata = codec->dev->platform_data;
+ int ret;
+
+ switch (level) {
+ case SND_SOC_BIAS_ON:
+ /* unmute */
+ snd_soc_update_bits(codec, AK4641_DAC, 0x20, 0);
+ break;
+ case SND_SOC_BIAS_PREPARE:
+ /* mute */
+ snd_soc_update_bits(codec, AK4641_DAC, 0x20, 0x20);
+ break;
+ case SND_SOC_BIAS_STANDBY:
+ if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
+ if (pdata && gpio_is_valid(pdata->gpio_power))
+ gpio_set_value(pdata->gpio_power, 1);
+ mdelay(1);
+ if (pdata && gpio_is_valid(pdata->gpio_npdn))
+ gpio_set_value(pdata->gpio_npdn, 1);
+ mdelay(1);
+
+ ret = snd_soc_cache_sync(codec);
+ if (ret) {
+ dev_err(codec->dev,
+ "Failed to sync cache: %d\n", ret);
+ return ret;
+ }
+ }
+ snd_soc_update_bits(codec, AK4641_PM1, 0x80, 0x80);
+ snd_soc_update_bits(codec, AK4641_PM2, 0x80, 0);
+ break;
+ case SND_SOC_BIAS_OFF:
+ snd_soc_update_bits(codec, AK4641_PM1, 0x80, 0);
+ if (pdata && gpio_is_valid(pdata->gpio_npdn))
+ gpio_set_value(pdata->gpio_npdn, 0);
+ if (pdata && gpio_is_valid(pdata->gpio_power))
+ gpio_set_value(pdata->gpio_power, 0);
+ codec->cache_sync = 1;
+ break;
+ }
+ codec->dapm.bias_level = level;
+ return 0;
+}
+
+#define AK4641_RATES (SNDRV_PCM_RATE_8000_48000)
+#define AK4641_RATES_BT (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\
+ SNDRV_PCM_RATE_16000)
+#define AK4641_FORMATS (SNDRV_PCM_FMTBIT_S16_LE)
+
+static struct snd_soc_dai_ops ak4641_i2s_dai_ops = {
+ .hw_params = ak4641_i2s_hw_params,
+ .set_fmt = ak4641_i2s_set_dai_fmt,
+ .digital_mute = ak4641_mute,
+ .set_sysclk = ak4641_set_dai_sysclk,
+};
+
+static struct snd_soc_dai_ops ak4641_pcm_dai_ops = {
+ .hw_params = NULL, /* rates are controlled by BT chip */
+ .set_fmt = ak4641_pcm_set_dai_fmt,
+ .digital_mute = ak4641_mute,
+ .set_sysclk = ak4641_set_dai_sysclk,
+};
+
+struct snd_soc_dai_driver ak4641_dai[] = {
+{
+ .name = "ak4641-hifi",
+ .id = 1,
+ .playback = {
+ .stream_name = "HiFi Playback",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = AK4641_RATES,
+ .formats = AK4641_FORMATS,
+ },
+ .capture = {
+ .stream_name = "HiFi Capture",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = AK4641_RATES,
+ .formats = AK4641_FORMATS,
+ },
+ .ops = &ak4641_i2s_dai_ops,
+ .symmetric_rates = 1,
+},
+{
+ .name = "ak4641-voice",
+ .id = 1,
+ .playback = {
+ .stream_name = "Voice Playback",
+ .channels_min = 1,
+ .channels_max = 1,
+ .rates = AK4641_RATES_BT,
+ .formats = AK4641_FORMATS,
+ },
+ .capture = {
+ .stream_name = "Voice Capture",
+ .channels_min = 1,
+ .channels_max = 1,
+ .rates = AK4641_RATES_BT,
+ .formats = AK4641_FORMATS,
+ },
+ .ops = &ak4641_pcm_dai_ops,
+ .symmetric_rates = 1,
+},
+};
+
+static int ak4641_suspend(struct snd_soc_codec *codec, pm_message_t state)
+{
+ ak4641_set_bias_level(codec, SND_SOC_BIAS_OFF);
+ return 0;
+}
+
+static int ak4641_resume(struct snd_soc_codec *codec)
+{
+ ak4641_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+ return 0;
+}
+
+static int ak4641_probe(struct snd_soc_codec *codec)
+{
+ struct ak4641_platform_data *pdata = codec->dev->platform_data;
+ int ret;
+
+
+ if (pdata) {
+ if (gpio_is_valid(pdata->gpio_power)) {
+ ret = gpio_request_one(pdata->gpio_power,
+ GPIOF_OUT_INIT_LOW, "ak4641 power");
+ if (ret)
+ goto err_out;
+ }
+ if (gpio_is_valid(pdata->gpio_npdn)) {
+ ret = gpio_request_one(pdata->gpio_npdn,
+ GPIOF_OUT_INIT_LOW, "ak4641 npdn");
+ if (ret)
+ goto err_gpio;
+
+ udelay(1); /* > 150 ns */
+ gpio_set_value(pdata->gpio_npdn, 1);
+ }
+ }
+
+ ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_I2C);
+ if (ret != 0) {
+ dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
+ goto err_register;
+ }
+
+ /* power on device */
+ ak4641_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+ return 0;
+
+err_register:
+ if (pdata) {
+ if (gpio_is_valid(pdata->gpio_power))
+ gpio_set_value(pdata->gpio_power, 0);
+ if (gpio_is_valid(pdata->gpio_npdn))
+ gpio_free(pdata->gpio_npdn);
+ }
+err_gpio:
+ if (pdata && gpio_is_valid(pdata->gpio_power))
+ gpio_free(pdata->gpio_power);
+err_out:
+ return ret;
+}
+
+static int ak4641_remove(struct snd_soc_codec *codec)
+{
+ struct ak4641_platform_data *pdata = codec->dev->platform_data;
+
+ ak4641_set_bias_level(codec, SND_SOC_BIAS_OFF);
+
+ if (pdata) {
+ if (gpio_is_valid(pdata->gpio_power)) {
+ gpio_set_value(pdata->gpio_power, 0);
+ gpio_free(pdata->gpio_power);
+ }
+ if (gpio_is_valid(pdata->gpio_npdn))
+ gpio_free(pdata->gpio_npdn);
+ }
+ return 0;
+}
+
+
+static struct snd_soc_codec_driver soc_codec_dev_ak4641 = {
+ .probe = ak4641_probe,
+ .remove = ak4641_remove,
+ .suspend = ak4641_suspend,
+ .resume = ak4641_resume,
+ .controls = ak4641_snd_controls,
+ .num_controls = ARRAY_SIZE(ak4641_snd_controls),
+ .dapm_widgets = ak4641_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(ak4641_dapm_widgets),
+ .dapm_routes = ak4641_audio_map,
+ .num_dapm_routes = ARRAY_SIZE(ak4641_audio_map),
+ .set_bias_level = ak4641_set_bias_level,
+ .reg_cache_size = ARRAY_SIZE(ak4641_reg),
+ .reg_word_size = sizeof(u8),
+ .reg_cache_default = ak4641_reg,
+ .reg_cache_step = 1,
+};
+
+
+static int __devinit ak4641_i2c_probe(struct i2c_client *i2c,
+ const struct i2c_device_id *id)
+{
+ struct ak4641_priv *ak4641;
+ int ret;
+
+ ak4641 = kzalloc(sizeof(struct ak4641_priv), GFP_KERNEL);
+ if (!ak4641)
+ return -ENOMEM;
+
+ i2c_set_clientdata(i2c, ak4641);
+
+ ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_ak4641,
+ ak4641_dai, ARRAY_SIZE(ak4641_dai));
+ if (ret < 0)
+ kfree(ak4641);
+
+ return ret;
+}
+
+static int __devexit ak4641_i2c_remove(struct i2c_client *i2c)
+{
+ snd_soc_unregister_codec(&i2c->dev);
+ kfree(i2c_get_clientdata(i2c));
+ return 0;
+}
+
+static const struct i2c_device_id ak4641_i2c_id[] = {
+ { "ak4641", 0 },
+ { }
+};
+MODULE_DEVICE_TABLE(i2c, ak4641_i2c_id);
+
+static struct i2c_driver ak4641_i2c_driver = {
+ .driver = {
+ .name = "ak4641",
+ .owner = THIS_MODULE,
+ },
+ .probe = ak4641_i2c_probe,
+ .remove = __devexit_p(ak4641_i2c_remove),
+ .id_table = ak4641_i2c_id,
+};
+
+static int __init ak4641_modinit(void)
+{
+ int ret;
+
+ ret = i2c_add_driver(&ak4641_i2c_driver);
+ if (ret != 0)
+ pr_err("Failed to register AK4641 I2C driver: %d\n", ret);
+
+ return ret;
+}
+module_init(ak4641_modinit);
+
+static void __exit ak4641_exit(void)
+{
+ i2c_del_driver(&ak4641_i2c_driver);
+}
+module_exit(ak4641_exit);
+
+MODULE_DESCRIPTION("SoC AK4641 driver");
+MODULE_AUTHOR("Harald Welte <laforge@gnufiish.org>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/ak4641.h b/sound/soc/codecs/ak4641.h
new file mode 100644
index 0000000..4a26324
--- /dev/null
+++ b/sound/soc/codecs/ak4641.h
@@ -0,0 +1,47 @@
+/*
+ * ak4641.h -- AK4641 SoC Audio driver
+ *
+ * Copyright 2008 Harald Welte <laforge@gnufiish.org>
+ *
+ * Based on ak4535.h
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef _AK4641_H
+#define _AK4641_H
+
+/* AK4641 register space */
+
+#define AK4641_PM1 0x00
+#define AK4641_PM2 0x01
+#define AK4641_SIG1 0x02
+#define AK4641_SIG2 0x03
+#define AK4641_MODE1 0x04
+#define AK4641_MODE2 0x05
+#define AK4641_DAC 0x06
+#define AK4641_MIC 0x07
+#define AK4641_TIMER 0x08
+#define AK4641_ALC1 0x09
+#define AK4641_ALC2 0x0a
+#define AK4641_PGA 0x0b
+#define AK4641_LATT 0x0c
+#define AK4641_RATT 0x0d
+#define AK4641_VOL 0x0e
+#define AK4641_STATUS 0x0f
+#define AK4641_EQLO 0x10
+#define AK4641_EQMID 0x11
+#define AK4641_EQHI 0x12
+#define AK4641_BTIF 0x13
+
+#define AK4641_CACHEREGNUM 0x14
+
+
+
+#define AK4641_DAI_HIFI 0
+#define AK4641_DAI_VOICE 1
+
+
+#endif
diff --git a/sound/soc/codecs/dmic.c b/sound/soc/codecs/dmic.c
index 57e9dac..f9a8773 100644
--- a/sound/soc/codecs/dmic.c
+++ b/sound/soc/codecs/dmic.c
@@ -39,7 +39,31 @@ static struct snd_soc_dai_driver dmic_dai = {
},
};
-static struct snd_soc_codec_driver soc_dmic = {};
+static const struct snd_soc_dapm_widget dmic_dapm_widgets[] = {
+ SND_SOC_DAPM_AIF_OUT("DMIC AIF", "Capture", 0,
+ SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_INPUT("DMic"),
+};
+
+static const struct snd_soc_dapm_route intercon[] = {
+ {"DMIC AIF", NULL, "DMic"},
+};
+
+static int dmic_probe(struct snd_soc_codec *codec)
+{
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
+
+ snd_soc_dapm_new_controls(dapm, dmic_dapm_widgets,
+ ARRAY_SIZE(dmic_dapm_widgets));
+ snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon));
+ snd_soc_dapm_new_widgets(dapm);
+
+ return 0;
+}
+
+static struct snd_soc_codec_driver soc_dmic = {
+ .probe = dmic_probe,
+};
static int __devinit dmic_dev_probe(struct platform_device *pdev)
{
diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c
index 93255ff..ac65a2d 100644
--- a/sound/soc/codecs/max98088.c
+++ b/sound/soc/codecs/max98088.c
@@ -656,8 +656,6 @@ static const struct soc_enum max98088_exmode_enum =
ARRAY_SIZE(max98088_exmode_texts),
max98088_exmode_texts,
max98088_exmode_values);
-static const struct snd_kcontrol_new max98088_exmode_controls =
- SOC_DAPM_VALUE_ENUM("Route", max98088_exmode_enum);
static const char *max98088_ex_thresh[] = { /* volts PP */
"0.6", "1.2", "1.8", "2.4", "3.0", "3.6", "4.2", "4.8"};
@@ -783,6 +781,7 @@ static const struct snd_kcontrol_new max98088_snd_controls[] = {
SOC_SINGLE("EQ1 Switch", M98088_REG_49_CFG_LEVEL, 0, 1, 0),
SOC_SINGLE("EQ2 Switch", M98088_REG_49_CFG_LEVEL, 1, 1, 0),
+ SOC_ENUM("EX Limiter Mode", max98088_exmode_enum),
SOC_ENUM("EX Limiter Threshold", max98088_ex_thresh_enum),
SOC_ENUM("DAI1 Filter Mode", max98088_filter_mode_enum),
@@ -808,10 +807,10 @@ static const struct snd_kcontrol_new max98088_snd_controls[] = {
/* Left speaker mixer switch */
static const struct snd_kcontrol_new max98088_left_speaker_mixer_controls[] = {
- SOC_DAPM_SINGLE("Left DAC1 Switch", M98088_REG_2B_MIX_SPK_LEFT, 7, 1, 0),
- SOC_DAPM_SINGLE("Right DAC1 Switch", M98088_REG_2B_MIX_SPK_LEFT, 0, 1, 0),
- SOC_DAPM_SINGLE("Left DAC2 Switch", M98088_REG_2B_MIX_SPK_LEFT, 7, 1, 0),
- SOC_DAPM_SINGLE("Right DAC2 Switch", M98088_REG_2B_MIX_SPK_LEFT, 0, 1, 0),
+ SOC_DAPM_SINGLE("Left DAC1 Switch", M98088_REG_2B_MIX_SPK_LEFT, 0, 1, 0),
+ SOC_DAPM_SINGLE("Right DAC1 Switch", M98088_REG_2B_MIX_SPK_LEFT, 7, 1, 0),
+ SOC_DAPM_SINGLE("Left DAC2 Switch", M98088_REG_2B_MIX_SPK_LEFT, 0, 1, 0),
+ SOC_DAPM_SINGLE("Right DAC2 Switch", M98088_REG_2B_MIX_SPK_LEFT, 7, 1, 0),
SOC_DAPM_SINGLE("MIC1 Switch", M98088_REG_2B_MIX_SPK_LEFT, 5, 1, 0),
SOC_DAPM_SINGLE("MIC2 Switch", M98088_REG_2B_MIX_SPK_LEFT, 6, 1, 0),
SOC_DAPM_SINGLE("INA1 Switch", M98088_REG_2B_MIX_SPK_LEFT, 1, 1, 0),
@@ -836,10 +835,10 @@ static const struct snd_kcontrol_new max98088_right_speaker_mixer_controls[] = {
/* Left headphone mixer switch */
static const struct snd_kcontrol_new max98088_left_hp_mixer_controls[] = {
- SOC_DAPM_SINGLE("Left DAC1 Switch", M98088_REG_25_MIX_HP_LEFT, 7, 1, 0),
- SOC_DAPM_SINGLE("Right DAC1 Switch", M98088_REG_25_MIX_HP_LEFT, 0, 1, 0),
- SOC_DAPM_SINGLE("Left DAC2 Switch", M98088_REG_25_MIX_HP_LEFT, 7, 1, 0),
- SOC_DAPM_SINGLE("Right DAC2 Switch", M98088_REG_25_MIX_HP_LEFT, 0, 1, 0),
+ SOC_DAPM_SINGLE("Left DAC1 Switch", M98088_REG_25_MIX_HP_LEFT, 0, 1, 0),
+ SOC_DAPM_SINGLE("Right DAC1 Switch", M98088_REG_25_MIX_HP_LEFT, 7, 1, 0),
+ SOC_DAPM_SINGLE("Left DAC2 Switch", M98088_REG_25_MIX_HP_LEFT, 0, 1, 0),
+ SOC_DAPM_SINGLE("Right DAC2 Switch", M98088_REG_25_MIX_HP_LEFT, 7, 1, 0),
SOC_DAPM_SINGLE("MIC1 Switch", M98088_REG_25_MIX_HP_LEFT, 5, 1, 0),
SOC_DAPM_SINGLE("MIC2 Switch", M98088_REG_25_MIX_HP_LEFT, 6, 1, 0),
SOC_DAPM_SINGLE("INA1 Switch", M98088_REG_25_MIX_HP_LEFT, 1, 1, 0),
@@ -864,10 +863,10 @@ static const struct snd_kcontrol_new max98088_right_hp_mixer_controls[] = {
/* Left earpiece/receiver mixer switch */
static const struct snd_kcontrol_new max98088_left_rec_mixer_controls[] = {
- SOC_DAPM_SINGLE("Left DAC1 Switch", M98088_REG_28_MIX_REC_LEFT, 7, 1, 0),
- SOC_DAPM_SINGLE("Right DAC1 Switch", M98088_REG_28_MIX_REC_LEFT, 0, 1, 0),
- SOC_DAPM_SINGLE("Left DAC2 Switch", M98088_REG_28_MIX_REC_LEFT, 7, 1, 0),
- SOC_DAPM_SINGLE("Right DAC2 Switch", M98088_REG_28_MIX_REC_LEFT, 0, 1, 0),
+ SOC_DAPM_SINGLE("Left DAC1 Switch", M98088_REG_28_MIX_REC_LEFT, 0, 1, 0),
+ SOC_DAPM_SINGLE("Right DAC1 Switch", M98088_REG_28_MIX_REC_LEFT, 7, 1, 0),
+ SOC_DAPM_SINGLE("Left DAC2 Switch", M98088_REG_28_MIX_REC_LEFT, 0, 1, 0),
+ SOC_DAPM_SINGLE("Right DAC2 Switch", M98088_REG_28_MIX_REC_LEFT, 7, 1, 0),
SOC_DAPM_SINGLE("MIC1 Switch", M98088_REG_28_MIX_REC_LEFT, 5, 1, 0),
SOC_DAPM_SINGLE("MIC2 Switch", M98088_REG_28_MIX_REC_LEFT, 6, 1, 0),
SOC_DAPM_SINGLE("INA1 Switch", M98088_REG_28_MIX_REC_LEFT, 1, 1, 0),
@@ -1094,9 +1093,6 @@ static const struct snd_soc_dapm_widget max98088_dapm_widgets[] = {
SND_SOC_DAPM_MICBIAS("MICBIAS", M98088_REG_4C_PWR_EN_IN, 3, 0),
- SND_SOC_DAPM_MUX("EX Limiter Mode", SND_SOC_NOPM, 0, 0,
- &max98088_exmode_controls),
-
SND_SOC_DAPM_OUTPUT("HPL"),
SND_SOC_DAPM_OUTPUT("HPR"),
SND_SOC_DAPM_OUTPUT("SPKL"),
@@ -1568,6 +1564,36 @@ static int max98088_dai2_set_fmt(struct snd_soc_dai *codec_dai,
return 0;
}
+static int max98088_dai1_digital_mute(struct snd_soc_dai *codec_dai, int mute)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ int reg;
+
+ if (mute)
+ reg = M98088_DAI_MUTE;
+ else
+ reg = 0;
+
+ snd_soc_update_bits(codec, M98088_REG_2F_LVL_DAI1_PLAY,
+ M98088_DAI_MUTE_MASK, reg);
+ return 0;
+}
+
+static int max98088_dai2_digital_mute(struct snd_soc_dai *codec_dai, int mute)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ int reg;
+
+ if (mute)
+ reg = M98088_DAI_MUTE;
+ else
+ reg = 0;
+
+ snd_soc_update_bits(codec, M98088_REG_31_LVL_DAI2_PLAY,
+ M98088_DAI_MUTE_MASK, reg);
+ return 0;
+}
+
static void max98088_sync_cache(struct snd_soc_codec *codec)
{
u16 *reg_cache = codec->reg_cache;
@@ -1629,12 +1655,14 @@ static struct snd_soc_dai_ops max98088_dai1_ops = {
.set_sysclk = max98088_dai_set_sysclk,
.set_fmt = max98088_dai1_set_fmt,
.hw_params = max98088_dai1_hw_params,
+ .digital_mute = max98088_dai1_digital_mute,
};
static struct snd_soc_dai_ops max98088_dai2_ops = {
.set_sysclk = max98088_dai_set_sysclk,
.set_fmt = max98088_dai2_set_fmt,
.hw_params = max98088_dai2_hw_params,
+ .digital_mute = max98088_dai2_digital_mute,
};
static struct snd_soc_dai_driver max98088_dai[] = {
diff --git a/sound/soc/codecs/max98088.h b/sound/soc/codecs/max98088.h
index 56554c7..be89a4f 100644
--- a/sound/soc/codecs/max98088.h
+++ b/sound/soc/codecs/max98088.h
@@ -133,6 +133,19 @@
#define M98088_REC_LINEMODE (1<<7)
#define M98088_REC_LINEMODE_MASK (1<<7)
+/* M98088_REG_2D_MIX_SPK_CNTL */
+ #define M98088_MIX_SPKR_GAIN_MASK (3<<2)
+ #define M98088_MIX_SPKR_GAIN_SHIFT 2
+ #define M98088_MIX_SPKL_GAIN_MASK (3<<0)
+ #define M98088_MIX_SPKL_GAIN_SHIFT 0
+
+/* M98088_REG_2F_LVL_DAI1_PLAY, M98088_REG_31_LVL_DAI2_PLAY */
+ #define M98088_DAI_MUTE (1<<7)
+ #define M98088_DAI_MUTE_MASK (1<<7)
+ #define M98088_DAI_VOICE_GAIN_MASK (3<<4)
+ #define M98088_DAI_ATTENUATION_MASK (0xF<<0)
+ #define M98088_DAI_ATTENUATION_SHIFT 0
+
/* M98088_REG_35_LVL_MIC1, M98088_REG_36_LVL_MIC2 */
#define M98088_MICPRE_MASK (3<<5)
#define M98088_MICPRE_SHIFT 5
diff --git a/sound/soc/codecs/max98095.c b/sound/soc/codecs/max98095.c
index fe19677..872a5fa 100644
--- a/sound/soc/codecs/max98095.c
+++ b/sound/soc/codecs/max98095.c
@@ -1870,16 +1870,14 @@ static int max98095_put_eq_enum(struct snd_kcontrol *kcontrol,
BUG_ON(channel > 1);
- cdata = &max98095->dai[channel];
+ if (!pdata || !max98095->eq_textcnt)
+ return 0;
if (sel >= pdata->eq_cfgcnt)
return -EINVAL;
+ cdata = &max98095->dai[channel];
cdata->eq_sel = sel;
-
- if (!pdata || !max98095->eq_textcnt)
- return 0;
-
fs = cdata->rate;
/* Find the selected configuration with nearest sample rate */
@@ -2018,16 +2016,14 @@ static int max98095_put_bq_enum(struct snd_kcontrol *kcontrol,
BUG_ON(channel > 1);
- cdata = &max98095->dai[channel];
+ if (!pdata || !max98095->bq_textcnt)
+ return 0;
if (sel >= pdata->bq_cfgcnt)
return -EINVAL;
+ cdata = &max98095->dai[channel];
cdata->bq_sel = sel;
-
- if (!pdata || !max98095->bq_textcnt)
- return 0;
-
fs = cdata->rate;
/* Find the selected configuration with nearest sample rate */
diff --git a/sound/soc/codecs/spdif_transciever.c b/sound/soc/codecs/spdif_transciever.c
index 4c32b54..6a1a7e7 100644
--- a/sound/soc/codecs/spdif_transciever.c
+++ b/sound/soc/codecs/spdif_transciever.c
@@ -21,7 +21,7 @@
#include <sound/pcm.h>
#include <sound/initval.h>
-MODULE_LICENSE("GPL");
+#define DRV_NAME "spdif-dit"
#define STUB_RATES SNDRV_PCM_RATE_8000_96000
#define STUB_FORMATS SNDRV_PCM_FMTBIT_S16_LE
@@ -56,7 +56,7 @@ static struct platform_driver spdif_dit_driver = {
.probe = spdif_dit_probe,
.remove = spdif_dit_remove,
.driver = {
- .name = "spdif-dit",
+ .name = DRV_NAME,
.owner = THIS_MODULE,
},
};
@@ -74,3 +74,7 @@ static void __exit dit_exit(void)
module_init(dit_modinit);
module_exit(dit_exit);
+MODULE_AUTHOR("Steve Chen <schen@mvista.com>");
+MODULE_DESCRIPTION("SPDIF dummy codec driver");
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:" DRV_NAME);
diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c
index 70099c9..84f4ad5 100644
--- a/sound/soc/codecs/ssm2602.c
+++ b/sound/soc/codecs/ssm2602.c
@@ -137,7 +137,7 @@ SND_SOC_DAPM_DAC("DAC", "HiFi Playback", SSM2602_PWR, 3, 1),
SND_SOC_DAPM_ADC("ADC", "HiFi Capture", SSM2602_PWR, 2, 1),
SND_SOC_DAPM_PGA("Line Input", SSM2602_PWR, 0, 1, NULL, 0),
-SND_SOC_DAPM_SUPPLY("Digital Core Power", SSM2602_ACTIVE, 0, 0, 0, 0),
+SND_SOC_DAPM_SUPPLY("Digital Core Power", SSM2602_ACTIVE, 0, 0, NULL, 0),
SND_SOC_DAPM_OUTPUT("LOUT"),
SND_SOC_DAPM_OUTPUT("ROUT"),
diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c
index 6c43c13..c3d96fc 100644
--- a/sound/soc/codecs/tlv320aic3x.c
+++ b/sound/soc/codecs/tlv320aic3x.c
@@ -157,7 +157,8 @@ static int aic3x_read(struct snd_soc_codec *codec, unsigned int reg,
static int snd_soc_dapm_put_volsw_aic3x(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
- struct snd_soc_dapm_widget *widget = snd_kcontrol_chip(kcontrol);
+ struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol);
+ struct snd_soc_dapm_widget *widget = wlist->widgets[0];
struct soc_mixer_control *mc =
(struct soc_mixer_control *)kcontrol->private_value;
unsigned int reg = mc->reg;
diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c
index 90c361e..faa5e9f 100644
--- a/sound/soc/codecs/tlv320dac33.c
+++ b/sound/soc/codecs/tlv320dac33.c
@@ -1,7 +1,7 @@
/*
* ALSA SoC Texas Instruments TLV320DAC33 codec driver
*
- * Author: Peter Ujfalusi <peter.ujfalusi@nokia.com>
+ * Author: Peter Ujfalusi <peter.ujfalusi@ti.com>
*
* Copyright: (C) 2009 Nokia Corporation
*
@@ -1658,5 +1658,5 @@ module_exit(dac33_module_exit);
MODULE_DESCRIPTION("ASoC TLV320DAC33 codec driver");
-MODULE_AUTHOR("Peter Ujfalusi <peter.ujfalusi@nokia.com>");
+MODULE_AUTHOR("Peter Ujfalusi <peter.ujfalusi@ti.com>");
MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/tlv320dac33.h b/sound/soc/codecs/tlv320dac33.h
index 7c318b5..ed69670 100644
--- a/sound/soc/codecs/tlv320dac33.h
+++ b/sound/soc/codecs/tlv320dac33.h
@@ -1,7 +1,7 @@
/*
* ALSA SoC Texas Instruments TLV320DAC33 codec driver
*
- * Author: Peter Ujfalusi <peter.ujfalusi@nokia.com>
+ * Author: Peter Ujfalusi <peter.ujfalusi@ti.com>
*
* Copyright: (C) 2009 Nokia Corporation
*
diff --git a/sound/soc/codecs/tpa6130a2.c b/sound/soc/codecs/tpa6130a2.c
index 1f1ac81..239e0c4 100644
--- a/sound/soc/codecs/tpa6130a2.c
+++ b/sound/soc/codecs/tpa6130a2.c
@@ -3,7 +3,7 @@
*
* Copyright (C) Nokia Corporation
*
- * Author: Peter Ujfalusi <peter.ujfalusi@nokia.com>
+ * Author: Peter Ujfalusi <peter.ujfalusi@ti.com>
*
* This program is free software; you can redistribute it and/or
* modify it under the terms of the GNU General Public License
@@ -495,7 +495,7 @@ static void __exit tpa6130a2_exit(void)
i2c_del_driver(&tpa6130a2_i2c_driver);
}
-MODULE_AUTHOR("Peter Ujfalusi");
+MODULE_AUTHOR("Peter Ujfalusi <peter.ujfalusi@ti.com>");
MODULE_DESCRIPTION("TPA6130A2 Headphone amplifier driver");
MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/tpa6130a2.h b/sound/soc/codecs/tpa6130a2.h
index 5df49c8..4174440 100644
--- a/sound/soc/codecs/tpa6130a2.h
+++ b/sound/soc/codecs/tpa6130a2.h
@@ -3,7 +3,7 @@
*
* Copyright (C) Nokia Corporation
*
- * Author: Peter Ujfalusi <peter.ujfalusi@nokia.com>
+ * Author: Peter Ujfalusi <peter.ujfalusi@ti.com>
*
* This program is free software; you can redistribute it and/or
* modify it under the terms of the GNU General Public License
diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c
index 255901c..4c33663 100644
--- a/sound/soc/codecs/twl6040.c
+++ b/sound/soc/codecs/twl6040.c
@@ -960,9 +960,9 @@ static DECLARE_TLV_DB_SCALE(mic_amp_tlv, -600, 600, 0);
/*
* AFMGAIN volume control:
- * from 18 to 24 dB in 6 dB steps
+ * from -18 to 24 dB in 6 dB steps
*/
-static DECLARE_TLV_DB_SCALE(afm_amp_tlv, 1800, 600, 0);
+static DECLARE_TLV_DB_SCALE(afm_amp_tlv, -1800, 600, 0);
/*
* HSGAIN volume control:
@@ -1049,7 +1049,7 @@ static const struct snd_kcontrol_new twl6040_snd_controls[] = {
/* AFM gains */
SOC_DOUBLE_TLV("Aux FM Volume",
- TWL6040_REG_LINEGAIN, 0, 4, 0xF, 0, afm_amp_tlv),
+ TWL6040_REG_LINEGAIN, 0, 3, 7, 0, afm_amp_tlv),
/* Playback gains */
SOC_TWL6040_DOUBLE_TLV("Headset Playback Volume",
diff --git a/sound/soc/codecs/wm1250-ev1.c b/sound/soc/codecs/wm1250-ev1.c
index 14d0716..bcc2089 100644
--- a/sound/soc/codecs/wm1250-ev1.c
+++ b/sound/soc/codecs/wm1250-ev1.c
@@ -22,7 +22,7 @@ SND_SOC_DAPM_ADC("ADC", "wm1250-ev1 Capture", SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_DAC("DAC", "wm1250-ev1 Playback", SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_INPUT("WM1250 Input"),
-SND_SOC_DAPM_INPUT("WM1250 Output"),
+SND_SOC_DAPM_OUTPUT("WM1250 Output"),
};
static const struct snd_soc_dapm_route wm1250_ev1_dapm_routes[] = {
diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c
index 6dec7ce..2dc964b 100644
--- a/sound/soc/codecs/wm8731.c
+++ b/sound/soc/codecs/wm8731.c
@@ -198,7 +198,7 @@ static int wm8731_check_osc(struct snd_soc_dapm_widget *source,
{
struct wm8731_priv *wm8731 = snd_soc_codec_get_drvdata(source->codec);
- return wm8731->sysclk_type == WM8731_SYSCLK_MCLK;
+ return wm8731->sysclk_type == WM8731_SYSCLK_XTAL;
}
static const struct snd_soc_dapm_route wm8731_intercon[] = {
diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c
index 957cd66..43e3d76 100644
--- a/sound/soc/codecs/wm8903.c
+++ b/sound/soc/codecs/wm8903.c
@@ -382,7 +382,8 @@ static void wm8903_seq_notifier(struct snd_soc_dapm_context *dapm,
static int wm8903_class_w_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
- struct snd_soc_dapm_widget *widget = snd_kcontrol_chip(kcontrol);
+ struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol);
+ struct snd_soc_dapm_widget *widget = wlist->widgets[0];
struct snd_soc_codec *codec = widget->codec;
struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec);
u16 reg;
diff --git a/sound/soc/codecs/wm8915.c b/sound/soc/codecs/wm8915.c
index ccc9bd8..a0b1a72 100644
--- a/sound/soc/codecs/wm8915.c
+++ b/sound/soc/codecs/wm8915.c
@@ -19,7 +19,6 @@
#include <linux/gcd.h>
#include <linux/gpio.h>
#include <linux/i2c.h>
-#include <linux/delay.h>
#include <linux/regulator/consumer.h>
#include <linux/slab.h>
#include <linux/workqueue.h>
diff --git a/sound/soc/codecs/wm8958-dsp2.c b/sound/soc/codecs/wm8958-dsp2.c
index 74983ee..0293763 100644
--- a/sound/soc/codecs/wm8958-dsp2.c
+++ b/sound/soc/codecs/wm8958-dsp2.c
@@ -99,7 +99,7 @@ static int wm8958_dsp2_fw(struct snd_soc_codec *codec, const char *name,
len = fw->size - len;
while (len) {
if (len < 12) {
- dev_err(codec->dev, "%s short data block of %d\n",
+ dev_err(codec->dev, "%s short data block of %zd\n",
name, len);
goto err;
}
@@ -107,7 +107,7 @@ static int wm8958_dsp2_fw(struct snd_soc_codec *codec, const char *name,
memcpy(&data32, data + 4, sizeof(data32));
block_len = be32_to_cpu(data32);
if (block_len + 8 > len) {
- dev_err(codec->dev, "%d byte block longer than file\n",
+ dev_err(codec->dev, "%zd byte block longer than file\n",
block_len);
goto err;
}
@@ -141,7 +141,7 @@ static int wm8958_dsp2_fw(struct snd_soc_codec *codec, const char *name,
case WM_FW_BLOCK_I:
case WM_FW_BLOCK_A:
case WM_FW_BLOCK_C:
- dev_dbg(codec->dev, "%s: %d bytes of %x@%x\n", name,
+ dev_dbg(codec->dev, "%s: %zd bytes of %x@%x\n", name,
block_len, (data32 >> 24) & 0xff,
data32 & 0xffffff);
@@ -362,6 +362,10 @@ static void wm8958_dsp_apply(struct snd_soc_codec *codec, int path, int start)
path, wm8994->dsp_active, start, pwr_reg, reg);
if (start && ena) {
+ /* If the DSP is already running then noop */
+ if (reg & WM8958_DSP2_ENA)
+ return;
+
/* If either AIFnCLK is not yet enabled postpone */
if (!(snd_soc_read(codec, WM8994_AIF1_CLOCKING_1)
& WM8994_AIF1CLK_ENA_MASK) &&
@@ -508,6 +512,9 @@ static int wm8958_mbc_put(struct snd_kcontrol *kcontrol,
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
+ if (wm8994->mbc_ena[mbc] == ucontrol->value.integer.value[0])
+ return 0;
+
if (ucontrol->value.integer.value[0] > 1)
return -EINVAL;
@@ -628,6 +635,9 @@ static int wm8958_vss_put(struct snd_kcontrol *kcontrol,
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
+ if (wm8994->vss_ena[vss] == ucontrol->value.integer.value[0])
+ return 0;
+
if (ucontrol->value.integer.value[0] > 1)
return -EINVAL;
@@ -689,6 +699,16 @@ static int wm8958_hpf_put(struct snd_kcontrol *kcontrol,
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
+ if (hpf < 3) {
+ if (wm8994->hpf1_ena[hpf % 3] ==
+ ucontrol->value.integer.value[0])
+ return 0;
+ } else {
+ if (wm8994->hpf2_ena[hpf % 3] ==
+ ucontrol->value.integer.value[0])
+ return 0;
+ }
+
if (ucontrol->value.integer.value[0] > 1)
return -EINVAL;
@@ -782,6 +802,9 @@ static int wm8958_enh_eq_put(struct snd_kcontrol *kcontrol,
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
+ if (wm8994->enh_eq_ena[eq] == ucontrol->value.integer.value[0])
+ return 0;
+
if (ucontrol->value.integer.value[0] > 1)
return -EINVAL;
diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c
index 056aef9..9e5ff78 100644
--- a/sound/soc/codecs/wm8993.c
+++ b/sound/soc/codecs/wm8993.c
@@ -718,7 +718,8 @@ static int clk_sys_event(struct snd_soc_dapm_widget *w,
static int class_w_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
- struct snd_soc_dapm_widget *widget = snd_kcontrol_chip(kcontrol);
+ struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol);
+ struct snd_soc_dapm_widget *widget = wlist->widgets[0];
struct snd_soc_codec *codec = widget->codec;
struct wm8993_priv *wm8993 = snd_soc_codec_get_drvdata(codec);
int ret;
diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index b6d47e7..970a95c 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -877,7 +877,8 @@ static const char *hp_mux_text[] = {
static int wm8994_put_hp_enum(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
- struct snd_soc_dapm_widget *w = snd_kcontrol_chip(kcontrol);
+ struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol);
+ struct snd_soc_dapm_widget *w = wlist->widgets[0];
struct snd_soc_codec *codec = w->codec;
int ret;
@@ -1004,7 +1005,8 @@ SOC_DAPM_SINGLE("AIF1.1 Switch", WM8994_DAC2_RIGHT_MIXER_ROUTING,
static int wm8994_put_class_w(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
- struct snd_soc_dapm_widget *w = snd_kcontrol_chip(kcontrol);
+ struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol);
+ struct snd_soc_dapm_widget *w = wlist->widgets[0];
struct snd_soc_codec *codec = w->codec;
int ret;
@@ -2416,8 +2418,19 @@ static struct snd_soc_dai_driver wm8994_dai[] = {
static int wm8994_suspend(struct snd_soc_codec *codec, pm_message_t state)
{
struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
+ struct wm8994 *control = codec->control_data;
int i, ret;
+ switch (control->type) {
+ case WM8994:
+ snd_soc_update_bits(codec, WM8994_MICBIAS, WM8994_MICD_ENA, 0);
+ break;
+ case WM8958:
+ snd_soc_update_bits(codec, WM8958_MIC_DETECT_1,
+ WM8958_MICD_ENA, 0);
+ break;
+ }
+
for (i = 0; i < ARRAY_SIZE(wm8994->fll); i++) {
memcpy(&wm8994->fll_suspend[i], &wm8994->fll[i],
sizeof(struct wm8994_fll_config));
@@ -2435,6 +2448,7 @@ static int wm8994_suspend(struct snd_soc_codec *codec, pm_message_t state)
static int wm8994_resume(struct snd_soc_codec *codec)
{
struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
+ struct wm8994 *control = codec->control_data;
int i, ret;
unsigned int val, mask;
@@ -2473,6 +2487,19 @@ static int wm8994_resume(struct snd_soc_codec *codec)
i + 1, ret);
}
+ switch (control->type) {
+ case WM8994:
+ if (wm8994->micdet[0].jack || wm8994->micdet[1].jack)
+ snd_soc_update_bits(codec, WM8994_MICBIAS,
+ WM8994_MICD_ENA, WM8994_MICD_ENA);
+ break;
+ case WM8958:
+ if (wm8994->jack_cb)
+ snd_soc_update_bits(codec, WM8958_MIC_DETECT_1,
+ WM8958_MICD_ENA, WM8958_MICD_ENA);
+ break;
+ }
+
return 0;
}
#else
diff --git a/sound/soc/codecs/wm8995.c b/sound/soc/codecs/wm8995.c
index 67eaaec..5ad873f 100644
--- a/sound/soc/codecs/wm8995.c
+++ b/sound/soc/codecs/wm8995.c
@@ -305,11 +305,11 @@ static int check_clk_sys(struct snd_soc_dapm_widget *source,
static int wm8995_put_class_w(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
- struct snd_soc_dapm_widget *w;
+ struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol);
+ struct snd_soc_dapm_widget *w = wlist->widgets[0];
struct snd_soc_codec *codec;
int ret;
- w = snd_kcontrol_chip(kcontrol);
codec = w->codec;
ret = snd_soc_dapm_put_volsw(kcontrol, ucontrol);
wm8995_update_class_w(codec);
diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c
index 4005e9a..9e370d1 100644
--- a/sound/soc/codecs/wm_hubs.c
+++ b/sound/soc/codecs/wm_hubs.c
@@ -215,23 +215,23 @@ static const struct snd_kcontrol_new analogue_snd_controls[] = {
SOC_SINGLE_TLV("IN1L Volume", WM8993_LEFT_LINE_INPUT_1_2_VOLUME, 0, 31, 0,
inpga_tlv),
SOC_SINGLE("IN1L Switch", WM8993_LEFT_LINE_INPUT_1_2_VOLUME, 7, 1, 1),
-SOC_SINGLE("IN1L ZC Switch", WM8993_LEFT_LINE_INPUT_1_2_VOLUME, 7, 1, 0),
+SOC_SINGLE("IN1L ZC Switch", WM8993_LEFT_LINE_INPUT_1_2_VOLUME, 6, 1, 0),
SOC_SINGLE_TLV("IN1R Volume", WM8993_RIGHT_LINE_INPUT_1_2_VOLUME, 0, 31, 0,
inpga_tlv),
SOC_SINGLE("IN1R Switch", WM8993_RIGHT_LINE_INPUT_1_2_VOLUME, 7, 1, 1),
-SOC_SINGLE("IN1R ZC Switch", WM8993_RIGHT_LINE_INPUT_1_2_VOLUME, 7, 1, 0),
+SOC_SINGLE("IN1R ZC Switch", WM8993_RIGHT_LINE_INPUT_1_2_VOLUME, 6, 1, 0),
SOC_SINGLE_TLV("IN2L Volume", WM8993_LEFT_LINE_INPUT_3_4_VOLUME, 0, 31, 0,
inpga_tlv),
SOC_SINGLE("IN2L Switch", WM8993_LEFT_LINE_INPUT_3_4_VOLUME, 7, 1, 1),
-SOC_SINGLE("IN2L ZC Switch", WM8993_LEFT_LINE_INPUT_3_4_VOLUME, 7, 1, 0),
+SOC_SINGLE("IN2L ZC Switch", WM8993_LEFT_LINE_INPUT_3_4_VOLUME, 6, 1, 0),
SOC_SINGLE_TLV("IN2R Volume", WM8993_RIGHT_LINE_INPUT_3_4_VOLUME, 0, 31, 0,
inpga_tlv),
SOC_SINGLE("IN2R Switch", WM8993_RIGHT_LINE_INPUT_3_4_VOLUME, 7, 1, 1),
-SOC_SINGLE("IN2R ZC Switch", WM8993_RIGHT_LINE_INPUT_3_4_VOLUME, 7, 1, 0),
+SOC_SINGLE("IN2R ZC Switch", WM8993_RIGHT_LINE_INPUT_3_4_VOLUME, 6, 1, 0),
SOC_SINGLE_TLV("MIXINL IN2L Volume", WM8993_INPUT_MIXER3, 7, 1, 0,
inmix_sw_tlv),
@@ -787,17 +787,17 @@ static const struct snd_soc_dapm_route analogue_routes[] = {
static const struct snd_soc_dapm_route lineout1_diff_routes[] = {
{ "LINEOUT1 Mixer", "IN1L Switch", "IN1L PGA" },
{ "LINEOUT1 Mixer", "IN1R Switch", "IN1R PGA" },
- { "LINEOUT1 Mixer", "Output Switch", "Left Output Mixer" },
+ { "LINEOUT1 Mixer", "Output Switch", "Left Output PGA" },
{ "LINEOUT1N Driver", NULL, "LINEOUT1 Mixer" },
{ "LINEOUT1P Driver", NULL, "LINEOUT1 Mixer" },
};
static const struct snd_soc_dapm_route lineout1_se_routes[] = {
- { "LINEOUT1N Mixer", "Left Output Switch", "Left Output Mixer" },
- { "LINEOUT1N Mixer", "Right Output Switch", "Left Output Mixer" },
+ { "LINEOUT1N Mixer", "Left Output Switch", "Left Output PGA" },
+ { "LINEOUT1N Mixer", "Right Output Switch", "Right Output PGA" },
- { "LINEOUT1P Mixer", "Left Output Switch", "Left Output Mixer" },
+ { "LINEOUT1P Mixer", "Left Output Switch", "Left Output PGA" },
{ "LINEOUT1N Driver", NULL, "LINEOUT1N Mixer" },
{ "LINEOUT1P Driver", NULL, "LINEOUT1P Mixer" },
@@ -806,17 +806,17 @@ static const struct snd_soc_dapm_route lineout1_se_routes[] = {
static const struct snd_soc_dapm_route lineout2_diff_routes[] = {
{ "LINEOUT2 Mixer", "IN2L Switch", "IN2L PGA" },
{ "LINEOUT2 Mixer", "IN2R Switch", "IN2R PGA" },
- { "LINEOUT2 Mixer", "Output Switch", "Right Output Mixer" },
+ { "LINEOUT2 Mixer", "Output Switch", "Right Output PGA" },
{ "LINEOUT2N Driver", NULL, "LINEOUT2 Mixer" },
{ "LINEOUT2P Driver", NULL, "LINEOUT2 Mixer" },
};
static const struct snd_soc_dapm_route lineout2_se_routes[] = {
- { "LINEOUT2N Mixer", "Left Output Switch", "Left Output Mixer" },
- { "LINEOUT2N Mixer", "Right Output Switch", "Left Output Mixer" },
+ { "LINEOUT2N Mixer", "Left Output Switch", "Left Output PGA" },
+ { "LINEOUT2N Mixer", "Right Output Switch", "Right Output PGA" },
- { "LINEOUT2P Mixer", "Right Output Switch", "Right Output Mixer" },
+ { "LINEOUT2P Mixer", "Right Output Switch", "Right Output PGA" },
{ "LINEOUT2N Driver", NULL, "LINEOUT2N Mixer" },
{ "LINEOUT2P Driver", NULL, "LINEOUT2P Mixer" },
@@ -836,17 +836,21 @@ int wm_hubs_add_analogue_controls(struct snd_soc_codec *codec)
snd_soc_update_bits(codec, WM8993_RIGHT_LINE_INPUT_3_4_VOLUME,
WM8993_IN2_VU, WM8993_IN2_VU);
+ snd_soc_update_bits(codec, WM8993_SPEAKER_VOLUME_LEFT,
+ WM8993_SPKOUT_VU, WM8993_SPKOUT_VU);
snd_soc_update_bits(codec, WM8993_SPEAKER_VOLUME_RIGHT,
WM8993_SPKOUT_VU, WM8993_SPKOUT_VU);
snd_soc_update_bits(codec, WM8993_LEFT_OUTPUT_VOLUME,
- WM8993_HPOUT1L_ZC, WM8993_HPOUT1L_ZC);
+ WM8993_HPOUT1_VU | WM8993_HPOUT1L_ZC,
+ WM8993_HPOUT1_VU | WM8993_HPOUT1L_ZC);
snd_soc_update_bits(codec, WM8993_RIGHT_OUTPUT_VOLUME,
WM8993_HPOUT1_VU | WM8993_HPOUT1R_ZC,
WM8993_HPOUT1_VU | WM8993_HPOUT1R_ZC);
snd_soc_update_bits(codec, WM8993_LEFT_OPGA_VOLUME,
- WM8993_MIXOUTL_ZC, WM8993_MIXOUTL_ZC);
+ WM8993_MIXOUTL_ZC | WM8993_MIXOUT_VU,
+ WM8993_MIXOUTL_ZC | WM8993_MIXOUT_VU);
snd_soc_update_bits(codec, WM8993_RIGHT_OPGA_VOLUME,
WM8993_MIXOUTR_ZC | WM8993_MIXOUT_VU,
WM8993_MIXOUTR_ZC | WM8993_MIXOUT_VU);
generated by cgit v0.10.2 at 2024-07-19 10:25:12 (GMT)