Merge tag 'v3.8-rc6' into next/soc

Linux 3.8-rc6

19 files changed, 189 insertions, 137 deletions

diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c
index adf397b9d0e6..ef62c435848e 100644
--- a/sound/soc/codecs/arizona.c
+++ b/sound/soc/codecs/arizona.c
@@ -446,15 +446,9 @@ static int arizona_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
         case SND_SOC_DAIFMT_DSP_A:
                 mode = 0;
                 break;
-        case SND_SOC_DAIFMT_DSP_B:
-                mode = 1;
-                break;
         case SND_SOC_DAIFMT_I2S:
                 mode = 2;
                 break;
-        case SND_SOC_DAIFMT_LEFT_J:
-                mode = 3;
-                break;
         default:
                 arizona_aif_err(dai, "Unsupported DAI format %d\n",
                                 fmt & SND_SOC_DAIFMT_FORMAT_MASK);
@@ -691,7 +685,7 @@ static int arizona_hw_params(struct snd_pcm_substream *substream,
         }
         sr_val = i;
 
-        lrclk = snd_soc_params_to_bclk(params) / params_rate(params);
+        lrclk = rates[bclk] / params_rate(params);
 
         arizona_aif_dbg(dai, "BCLK %dHz LRCLK %dHz\n",
                         rates[bclk], rates[bclk] / lrclk);
@@ -714,7 +708,8 @@ static int arizona_hw_params(struct snd_pcm_substream *substream,
                 snd_soc_update_bits(codec, ARIZONA_ASYNC_SAMPLE_RATE_1,
                                     ARIZONA_ASYNC_SAMPLE_RATE_MASK, sr_val);
                 snd_soc_update_bits(codec, base + ARIZONA_AIF_RATE_CTRL,
-                                    ARIZONA_AIF1_RATE_MASK, 8);
+                                    ARIZONA_AIF1_RATE_MASK,
+                                    8 << ARIZONA_AIF1_RATE_SHIFT);
                 break;
         default:
                 arizona_aif_err(dai, "Invalid clock %d\n", dai_priv->clk);
@@ -1087,6 +1082,9 @@ int arizona_init_fll(struct arizona *arizona, int id, int base, int lock_irq,
                         id, ret);
         }
 
+        regmap_update_bits(arizona->regmap, fll->base + 1,
+                           ARIZONA_FLL1_FREERUN, 0);
+
         return 0;
 }
 EXPORT_SYMBOL_GPL(arizona_init_fll);
diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h
index 41dae1ed3b71..4deebeb07177 100644
--- a/sound/soc/codecs/arizona.h
+++ b/sound/soc/codecs/arizona.h
@@ -34,15 +34,15 @@
 
 #define ARIZONA_FLL_SRC_MCLK1      0
 #define ARIZONA_FLL_SRC_MCLK2      1
-#define ARIZONA_FLL_SRC_SLIMCLK    2
-#define ARIZONA_FLL_SRC_FLL1       3
-#define ARIZONA_FLL_SRC_FLL2       4
-#define ARIZONA_FLL_SRC_AIF1BCLK   5
-#define ARIZONA_FLL_SRC_AIF2BCLK   6
-#define ARIZONA_FLL_SRC_AIF3BCLK   7
-#define ARIZONA_FLL_SRC_AIF1LRCLK  8
-#define ARIZONA_FLL_SRC_AIF2LRCLK  9
-#define ARIZONA_FLL_SRC_AIF3LRCLK 10
+#define ARIZONA_FLL_SRC_SLIMCLK    3
+#define ARIZONA_FLL_SRC_FLL1       4
+#define ARIZONA_FLL_SRC_FLL2       5
+#define ARIZONA_FLL_SRC_AIF1BCLK   8
+#define ARIZONA_FLL_SRC_AIF2BCLK   9
+#define ARIZONA_FLL_SRC_AIF3BCLK  10
+#define ARIZONA_FLL_SRC_AIF1LRCLK 12
+#define ARIZONA_FLL_SRC_AIF2LRCLK 13
+#define ARIZONA_FLL_SRC_AIF3LRCLK 14
 
 #define ARIZONA_MIXER_VOL_MASK             0x00FE
 #define ARIZONA_MIXER_VOL_SHIFT                 1
diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c
index 4f1127935fdf..ac8742a1f25a 100644
--- a/sound/soc/codecs/cs4271.c
+++ b/sound/soc/codecs/cs4271.c
@@ -474,16 +474,16 @@ static int cs4271_probe(struct snd_soc_codec *codec)
         struct cs4271_platform_data *cs4271plat = codec->dev->platform_data;
         int ret;
         int gpio_nreset = -EINVAL;
-        int amutec_eq_bmutec = 0;
+        bool amutec_eq_bmutec = false;
 
 #ifdef CONFIG_OF
         if (of_match_device(cs4271_dt_ids, codec->dev)) {
                 gpio_nreset = of_get_named_gpio(codec->dev->of_node,
                                                 "reset-gpio", 0);
 
-                if (!of_get_property(codec->dev->of_node,
+                if (of_get_property(codec->dev->of_node,
                                      "cirrus,amutec-eq-bmutec", NULL))
-                        amutec_eq_bmutec = 1;
+                        amutec_eq_bmutec = true;
         }
 #endif
 
diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c
index 99bb1c69499e..9811a5478c87 100644
--- a/sound/soc/codecs/cs42l52.c
+++ b/sound/soc/codecs/cs42l52.c
@@ -737,7 +737,7 @@ static const struct cs42l52_clk_para clk_map_table[] = {
 
 static int cs42l52_get_clk(int mclk, int rate)
 {
-        int i, ret = 0;
+        int i, ret = -EINVAL;
         u_int mclk1, mclk2 = 0;
 
         for (i = 0; i < ARRAY_SIZE(clk_map_table); i++) {
@@ -749,8 +749,6 @@ static int cs42l52_get_clk(int mclk, int rate)
                         }
                 }
         }
-        if (ret > ARRAY_SIZE(clk_map_table))
-                return -EINVAL;
         return ret;
 }
 
diff --git a/sound/soc/codecs/lm49453.c b/sound/soc/codecs/lm49453.c
index d75257d40a49..e19490cfb3a8 100644
--- a/sound/soc/codecs/lm49453.c
+++ b/sound/soc/codecs/lm49453.c
@@ -111,9 +111,9 @@ static struct reg_default lm49453_reg_defs[] = {
         { 101, 0x00 },
         { 102, 0x00 },
         { 103, 0x01 },
-        { 105, 0x01 },
-        { 106, 0x00 },
-        { 107, 0x01 },
+        { 104, 0x01 },
+        { 105, 0x00 },
+        { 106, 0x01 },
         { 107, 0x00 },
         { 108, 0x00 },
         { 109, 0x00 },
@@ -163,56 +163,25 @@ static struct reg_default lm49453_reg_defs[] = {
         { 184, 0x00 },
         { 185, 0x00 },
         { 186, 0x00 },
-        { 189, 0x00 },
+        { 187, 0x00 },
         { 188, 0x00 },
-        { 194, 0x00 },
-        { 195, 0x00 },
-        { 196, 0x00 },
-        { 197, 0x00 },
-        { 200, 0x00 },
-        { 201, 0x00 },
-        { 202, 0x00 },
-        { 203, 0x00 },
-        { 204, 0x00 },
-        { 205, 0x00 },
-        { 208, 0x00 },
+        { 189, 0x00 },
+        { 208, 0x06 },
         { 209, 0x00 },
-        { 210, 0x00 },
-        { 211, 0x00 },
-        { 213, 0x00 },
-        { 214, 0x00 },
-        { 215, 0x00 },
-        { 216, 0x00 },
-        { 217, 0x00 },
-        { 218, 0x00 },
-        { 219, 0x00 },
+        { 210, 0x08 },
+        { 211, 0x54 },
+        { 212, 0x14 },
+        { 213, 0x0d },
+        { 214, 0x0d },
+        { 215, 0x14 },
+        { 216, 0x60 },
         { 221, 0x00 },
         { 222, 0x00 },
+        { 223, 0x00 },
         { 224, 0x00 },
-        { 225, 0x00 },
-        { 226, 0x00 },
-        { 227, 0x00 },
-        { 228, 0x00 },
-        { 229, 0x00 },
-        { 230, 0x13 },
-        { 231, 0x00 },
-        { 232, 0x80 },
-        { 233, 0x0C },
-        { 234, 0xDD },
-        { 235, 0x00 },
-        { 236, 0x04 },
-        { 237, 0x00 },
-        { 238, 0x00 },
-        { 239, 0x00 },
-        { 240, 0x00 },
-        { 241, 0x00 },
-        { 242, 0x00 },
-        { 243, 0x00 },
-        { 244, 0x00 },
-        { 245, 0x00 },
         { 248, 0x00 },
         { 249, 0x00 },
-        { 254, 0x00 },
+        { 250, 0x00 },
         { 255, 0x00 },
 };
 
@@ -525,36 +494,41 @@ SOC_DAPM_SINGLE("Port2_2 Switch", LM49453_P0_PORT2_TX2_REG, 7, 1, 0),
 };
 
 /* TLV Declarations */
-static const DECLARE_TLV_DB_SCALE(digital_tlv, -7650, 150, 1);
-static const DECLARE_TLV_DB_SCALE(port_tlv, 0, 600, 0);
+static const DECLARE_TLV_DB_SCALE(adc_dac_tlv, -7650, 150, 1);
+static const DECLARE_TLV_DB_SCALE(mic_tlv, 0, 200, 1);
+static const DECLARE_TLV_DB_SCALE(port_tlv, -1800, 600, 0);
+static const DECLARE_TLV_DB_SCALE(stn_tlv, -7200, 150, 0);
 
 static const struct snd_kcontrol_new lm49453_sidetone_mixer_controls[] = {
 /* Sidetone supports mono only */
 SOC_DAPM_SINGLE_TLV("Sidetone ADCL Volume", LM49453_P0_STN_VOL_ADCL_REG,
-                     0, 0x3F, 0, digital_tlv),
+                     0, 0x3F, 0, stn_tlv),
 SOC_DAPM_SINGLE_TLV("Sidetone ADCR Volume", LM49453_P0_STN_VOL_ADCR_REG,
-                     0, 0x3F, 0, digital_tlv),
+                     0, 0x3F, 0, stn_tlv),
 SOC_DAPM_SINGLE_TLV("Sidetone DMIC1L Volume", LM49453_P0_STN_VOL_DMIC1L_REG,
-                     0, 0x3F, 0, digital_tlv),
+                     0, 0x3F, 0, stn_tlv),
 SOC_DAPM_SINGLE_TLV("Sidetone DMIC1R Volume", LM49453_P0_STN_VOL_DMIC1R_REG,
-                     0, 0x3F, 0, digital_tlv),
+                     0, 0x3F, 0, stn_tlv),
 SOC_DAPM_SINGLE_TLV("Sidetone DMIC2L Volume", LM49453_P0_STN_VOL_DMIC2L_REG,
-                     0, 0x3F, 0, digital_tlv),
+                     0, 0x3F, 0, stn_tlv),
 SOC_DAPM_SINGLE_TLV("Sidetone DMIC2R Volume", LM49453_P0_STN_VOL_DMIC2R_REG,
-                     0, 0x3F, 0, digital_tlv),
+                     0, 0x3F, 0, stn_tlv),
 };
 
 static const struct snd_kcontrol_new lm49453_snd_controls[] = {
         /* mic1 and mic2 supports mono only */
-        SOC_SINGLE_TLV("Mic1 Volume", LM49453_P0_ADC_LEVELL_REG, 0, 6,
-                        0, digital_tlv),
-        SOC_SINGLE_TLV("Mic2 Volume", LM49453_P0_ADC_LEVELR_REG, 0, 6,
-                        0, digital_tlv),
+        SOC_SINGLE_TLV("Mic1 Volume", LM49453_P0_MICL_REG, 0, 15, 0, mic_tlv),
+        SOC_SINGLE_TLV("Mic2 Volume", LM49453_P0_MICR_REG, 0, 15, 0, mic_tlv),
+
+        SOC_SINGLE_TLV("ADCL Volume", LM49453_P0_ADC_LEVELL_REG, 0, 63,
+                        0, adc_dac_tlv),
+        SOC_SINGLE_TLV("ADCR Volume", LM49453_P0_ADC_LEVELR_REG, 0, 63,
+                        0, adc_dac_tlv),
 
         SOC_DOUBLE_R_TLV("DMIC1 Volume", LM49453_P0_DMIC1_LEVELL_REG,
-                          LM49453_P0_DMIC1_LEVELR_REG, 0, 6, 0, digital_tlv),
+                          LM49453_P0_DMIC1_LEVELR_REG, 0, 63, 0, adc_dac_tlv),
         SOC_DOUBLE_R_TLV("DMIC2 Volume", LM49453_P0_DMIC2_LEVELL_REG,
-                          LM49453_P0_DMIC2_LEVELR_REG, 0, 6, 0, digital_tlv),
+                          LM49453_P0_DMIC2_LEVELR_REG, 0, 63, 0, adc_dac_tlv),
 
         SOC_DAPM_ENUM("Mic2Mode", lm49453_mic2mode_enum),
         SOC_DAPM_ENUM("DMIC12 SRC", lm49453_dmic12_cfg_enum),
@@ -569,16 +543,16 @@ static const struct snd_kcontrol_new lm49453_snd_controls[] = {
                                           2, 1, 0),
 
         SOC_DOUBLE_R_TLV("DAC HP Volume", LM49453_P0_DAC_HP_LEVELL_REG,
-                          LM49453_P0_DAC_HP_LEVELR_REG, 0, 6, 0, digital_tlv),
+                          LM49453_P0_DAC_HP_LEVELR_REG, 0, 63, 0, adc_dac_tlv),
         SOC_DOUBLE_R_TLV("DAC LO Volume", LM49453_P0_DAC_LO_LEVELL_REG,
-                          LM49453_P0_DAC_LO_LEVELR_REG, 0, 6, 0, digital_tlv),
+                          LM49453_P0_DAC_LO_LEVELR_REG, 0, 63, 0, adc_dac_tlv),
         SOC_DOUBLE_R_TLV("DAC LS Volume", LM49453_P0_DAC_LS_LEVELL_REG,
-                          LM49453_P0_DAC_LS_LEVELR_REG, 0, 6, 0, digital_tlv),
+                          LM49453_P0_DAC_LS_LEVELR_REG, 0, 63, 0, adc_dac_tlv),
         SOC_DOUBLE_R_TLV("DAC HA Volume", LM49453_P0_DAC_HA_LEVELL_REG,
-                          LM49453_P0_DAC_HA_LEVELR_REG, 0, 6, 0, digital_tlv),
+                          LM49453_P0_DAC_HA_LEVELR_REG, 0, 63, 0, adc_dac_tlv),
 
         SOC_SINGLE_TLV("EP Volume", LM49453_P0_DAC_LS_LEVELL_REG,
-                        0, 6, 0, digital_tlv),
+                        0, 63, 0, adc_dac_tlv),
 
         SOC_SINGLE_TLV("PORT1_1_RX_LVL Volume", LM49453_P0_PORT1_RX_LVL1_REG,
                         0, 3, 0, port_tlv),
@@ -1218,7 +1192,7 @@ static int lm49453_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
         }
 
         snd_soc_update_bits(codec, LM49453_P0_AUDIO_PORT1_BASIC_REG,
-                            LM49453_AUDIO_PORT1_BASIC_FMT_MASK|BIT(1)|BIT(5),
+                            LM49453_AUDIO_PORT1_BASIC_FMT_MASK|BIT(0)|BIT(5),
                             (aif_val | mode | clk_phase));
 
         snd_soc_write(codec, LM49453_P0_AUDIO_PORT1_RX_MSB_REG, clk_shift);
diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c
index cb1675cd8e1c..92bbfec9b107 100644
--- a/sound/soc/codecs/sgtl5000.c
+++ b/sound/soc/codecs/sgtl5000.c
@@ -401,7 +401,7 @@ static const struct snd_kcontrol_new sgtl5000_snd_controls[] = {
                         5, 1, 0),
 
         SOC_SINGLE_TLV("Mic Volume", SGTL5000_CHIP_MIC_CTRL,
-                        0, 4, 0, mic_gain_tlv),
+                        0, 3, 0, mic_gain_tlv),
 };
 
 /* mute the codec used by alsa core */
@@ -1344,7 +1344,7 @@ static int sgtl5000_probe(struct snd_soc_codec *codec)
                         SGTL5000_HP_ZCD_EN |
                         SGTL5000_ADC_ZCD_EN);
 
-        snd_soc_write(codec, SGTL5000_CHIP_MIC_CTRL, 0);
+        snd_soc_write(codec, SGTL5000_CHIP_MIC_CTRL, 2);
 
         /*
          * disable DAP
diff --git a/sound/soc/codecs/sta529.c b/sound/soc/codecs/sta529.c
index ab355c4f0b2d..40c07be9b581 100644
--- a/sound/soc/codecs/sta529.c
+++ b/sound/soc/codecs/sta529.c
@@ -74,9 +74,10 @@
                                 SNDRV_PCM_FMTBIT_S32_LE)
 #define S2PC_VALUE              0x98
 #define CLOCK_OUT               0x60
-#define LEFT_J_DATA_FORMAT      0x10
-#define I2S_DATA_FORMAT         0x12
-#define RIGHT_J_DATA_FORMAT     0x14
+#define DATA_FORMAT_MSK         0x0E
+#define LEFT_J_DATA_FORMAT      0x00
+#define I2S_DATA_FORMAT         0x02
+#define RIGHT_J_DATA_FORMAT     0x04
 #define CODEC_MUTE_VAL          0x80
 
 #define POWER_CNTLMSAK          0x40
@@ -289,7 +290,7 @@ static int sta529_set_dai_fmt(struct snd_soc_dai *codec_dai, u32 fmt)
                 return -EINVAL;
         }
 
-        snd_soc_update_bits(codec, STA529_S2PCFG0, 0x0D, mode);
+        snd_soc_update_bits(codec, STA529_S2PCFG0, DATA_FORMAT_MSK, mode);
 
         return 0;
 }
diff --git a/sound/soc/codecs/wm2000.c b/sound/soc/codecs/wm2000.c
index 1cbe88f01d63..12bcae63a7f0 100644
--- a/sound/soc/codecs/wm2000.c
+++ b/sound/soc/codecs/wm2000.c
@@ -209,9 +209,9 @@ static int wm2000_power_up(struct i2c_client *i2c, int analogue)
 
         ret = wm2000_read(i2c, WM2000_REG_SPEECH_CLARITY);
         if (wm2000->speech_clarity)
-                ret &= ~WM2000_SPEECH_CLARITY;
-        else
                 ret |= WM2000_SPEECH_CLARITY;
+        else
+                ret &= ~WM2000_SPEECH_CLARITY;
         wm2000_write(i2c, WM2000_REG_SPEECH_CLARITY, ret);
 
         wm2000_write(i2c, WM2000_REG_SYS_START0, 0x33);
diff --git a/sound/soc/codecs/wm2200.c b/sound/soc/codecs/wm2200.c
index afcf31df77e0..d8c65f574658 100644
--- a/sound/soc/codecs/wm2200.c
+++ b/sound/soc/codecs/wm2200.c
@@ -1019,8 +1019,6 @@ static const char *wm2200_mixer_texts[] = {
         "EQR",
         "LHPF1",
         "LHPF2",
-        "LHPF3",
-        "LHPF4",
         "DSP1.1",
         "DSP1.2",
         "DSP1.3",
@@ -1053,7 +1051,6 @@ static int wm2200_mixer_values[] = {
         0x25,
         0x50,   /* EQ */
         0x51,
-        0x52,
         0x60,   /* LHPF1 */
         0x61,   /* LHPF2 */
         0x68,   /* DSP1 */
@@ -1566,15 +1563,9 @@ static int wm2200_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
         case SND_SOC_DAIFMT_DSP_A:
                 fmt_val = 0;
                 break;
-        case SND_SOC_DAIFMT_DSP_B:
-                fmt_val = 1;
-                break;
         case SND_SOC_DAIFMT_I2S:
                 fmt_val = 2;
                 break;
-        case SND_SOC_DAIFMT_LEFT_J:
-                fmt_val = 3;
-                break;
         default:
                 dev_err(codec->dev, "Unsupported DAI format %d\n",
                         fmt & SND_SOC_DAIFMT_FORMAT_MASK);
@@ -1626,7 +1617,7 @@ static int wm2200_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
                             WM2200_AIF1TX_LRCLK_MSTR | WM2200_AIF1TX_LRCLK_INV,
                             lrclk);
         snd_soc_update_bits(codec, WM2200_AUDIO_IF_1_5,
-                            WM2200_AIF1_FMT_MASK << 1, fmt_val << 1);
+                            WM2200_AIF1_FMT_MASK, fmt_val);
 
         return 0;
 }
diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c
index 5a5f36936235..54397a508073 100644
--- a/sound/soc/codecs/wm5100.c
+++ b/sound/soc/codecs/wm5100.c
@@ -1279,15 +1279,9 @@ static int wm5100_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
         case SND_SOC_DAIFMT_DSP_A:
                 mask = 0;
                 break;
-        case SND_SOC_DAIFMT_DSP_B:
-                mask = 1;
-                break;
         case SND_SOC_DAIFMT_I2S:
                 mask = 2;
                 break;
-        case SND_SOC_DAIFMT_LEFT_J:
-                mask = 3;
-                break;
         default:
                 dev_err(codec->dev, "Unsupported DAI format %d\n",
                         fmt & SND_SOC_DAIFMT_FORMAT_MASK);
diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c
index 688ade080589..1440b3f9b7bb 100644
--- a/sound/soc/codecs/wm5102.c
+++ b/sound/soc/codecs/wm5102.c
@@ -36,6 +36,9 @@
 struct wm5102_priv {
         struct arizona_priv core;
         struct arizona_fll fll[2];
+
+        unsigned int spk_ena:2;
+        unsigned int spk_ena_pending:1;
 };
 
 static DECLARE_TLV_DB_SCALE(ana_tlv, 0, 100, 0);
@@ -787,6 +790,47 @@ ARIZONA_MIXER_CONTROLS("AIF3TX1", ARIZONA_AIF3TX1MIX_INPUT_1_SOURCE),
 ARIZONA_MIXER_CONTROLS("AIF3TX2", ARIZONA_AIF3TX2MIX_INPUT_1_SOURCE),
 };
 
+static int wm5102_spk_ev(struct snd_soc_dapm_widget *w,
+                         struct snd_kcontrol *kcontrol,
+                         int event)
+{
+        struct snd_soc_codec *codec = w->codec;
+        struct arizona *arizona = dev_get_drvdata(codec->dev->parent);
+        struct wm5102_priv *wm5102 = snd_soc_codec_get_drvdata(codec);
+
+        if (arizona->rev < 1)
+                return 0;
+
+        switch (event) {
+        case SND_SOC_DAPM_PRE_PMU:
+                if (!wm5102->spk_ena) {
+                        snd_soc_write(codec, 0x4f5, 0x25a);
+                        wm5102->spk_ena_pending = true;
+                }
+                break;
+        case SND_SOC_DAPM_POST_PMU:
+                if (wm5102->spk_ena_pending) {
+                        msleep(75);
+                        snd_soc_write(codec, 0x4f5, 0xda);
+                        wm5102->spk_ena_pending = false;
+                        wm5102->spk_ena++;
+                }
+                break;
+        case SND_SOC_DAPM_PRE_PMD:
+                wm5102->spk_ena--;
+                if (!wm5102->spk_ena)
+                        snd_soc_write(codec, 0x4f5, 0x25a);
+                break;
+        case SND_SOC_DAPM_POST_PMD:
+                if (!wm5102->spk_ena)
+                        snd_soc_write(codec, 0x4f5, 0x0da);
+                break;
+        }
+
+        return 0;
+}
+
+
 ARIZONA_MIXER_ENUMS(EQ1, ARIZONA_EQ1MIX_INPUT_1_SOURCE);
 ARIZONA_MIXER_ENUMS(EQ2, ARIZONA_EQ2MIX_INPUT_1_SOURCE);
 ARIZONA_MIXER_ENUMS(EQ3, ARIZONA_EQ3MIX_INPUT_1_SOURCE);
@@ -852,8 +896,7 @@ static const unsigned int wm5102_aec_loopback_values[] = {
 
 static const struct soc_enum wm5102_aec_loopback =
         SOC_VALUE_ENUM_SINGLE(ARIZONA_DAC_AEC_CONTROL_1,
-                              ARIZONA_AEC_LOOPBACK_SRC_SHIFT,
-                              ARIZONA_AEC_LOOPBACK_SRC_MASK,
+                              ARIZONA_AEC_LOOPBACK_SRC_SHIFT, 0xf,
                               ARRAY_SIZE(wm5102_aec_loopback_texts),
                               wm5102_aec_loopback_texts,
                               wm5102_aec_loopback_values);
@@ -1034,10 +1077,10 @@ SND_SOC_DAPM_PGA_E("OUT3L", ARIZONA_OUTPUT_ENABLES_1,
                    ARIZONA_OUT3L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev,
                    SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
 SND_SOC_DAPM_PGA_E("OUT4L", ARIZONA_OUTPUT_ENABLES_1,
-                   ARIZONA_OUT4L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev,
+                   ARIZONA_OUT4L_ENA_SHIFT, 0, NULL, 0, wm5102_spk_ev,
                    SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
 SND_SOC_DAPM_PGA_E("OUT4R", ARIZONA_OUTPUT_ENABLES_1,
-                   ARIZONA_OUT4R_ENA_SHIFT, 0, NULL, 0, arizona_out_ev,
+                   ARIZONA_OUT4R_ENA_SHIFT, 0, NULL, 0, wm5102_spk_ev,
                    SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
 SND_SOC_DAPM_PGA_E("OUT5L", ARIZONA_OUTPUT_ENABLES_1,
                    ARIZONA_OUT5L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev,
diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c
index ae80c8c28536..7a090968c4f7 100644
--- a/sound/soc/codecs/wm5110.c
+++ b/sound/soc/codecs/wm5110.c
@@ -344,8 +344,7 @@ static const unsigned int wm5110_aec_loopback_values[] = {
 
 static const struct soc_enum wm5110_aec_loopback =
         SOC_VALUE_ENUM_SINGLE(ARIZONA_DAC_AEC_CONTROL_1,
-                              ARIZONA_AEC_LOOPBACK_SRC_SHIFT,
-                              ARIZONA_AEC_LOOPBACK_SRC_MASK,
+                              ARIZONA_AEC_LOOPBACK_SRC_SHIFT, 0xf,
                               ARRAY_SIZE(wm5110_aec_loopback_texts),
                               wm5110_aec_loopback_texts,
                               wm5110_aec_loopback_values);
diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c
index ffc89fab96fb..b6b654837585 100644
--- a/sound/soc/codecs/wm_adsp.c
+++ b/sound/soc/codecs/wm_adsp.c
@@ -169,6 +169,7 @@ static int wm_adsp_load(struct wm_adsp *dsp)
         const struct wm_adsp_region *mem;
         const char *region_name;
         char *file, *text;
+        void *buf;
         unsigned int reg;
         int regions = 0;
         int ret, offset, type, sizes;
@@ -322,8 +323,18 @@ static int wm_adsp_load(struct wm_adsp *dsp)
                 }
 
                 if (reg) {
-                        ret = regmap_raw_write(regmap, reg, region->data,
+                        buf = kmemdup(region->data, le32_to_cpu(region->len),
+                                      GFP_KERNEL | GFP_DMA);
+                        if (!buf) {
+                                adsp_err(dsp, "Out of memory\n");
+                                return -ENOMEM;
+                        }
+
+                        ret = regmap_raw_write(regmap, reg, buf,
                                                le32_to_cpu(region->len));
+
+                        kfree(buf);
+
                         if (ret != 0) {
                                 adsp_err(dsp,
                                         "%s.%d: Failed to write %d bytes at %d in %s: %d\n",
@@ -359,6 +370,7 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp)
         const char *region_name;
         int ret, pos, blocks, type, offset, reg;
         char *file;
+        void *buf;
 
         file = kzalloc(PAGE_SIZE, GFP_KERNEL);
         if (file == NULL)
@@ -384,7 +396,7 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp)
         hdr = (void*)&firmware->data[0];
         if (memcmp(hdr->magic, "WMDR", 4) != 0) {
                 adsp_err(dsp, "%s: invalid magic\n", file);
-                return -EINVAL;
+                goto out_fw;
         }
 
         adsp_dbg(dsp, "%s: v%d.%d.%d\n", file,
@@ -426,6 +438,13 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp)
                 }
 
                 if (reg) {
+                        buf = kmemdup(blk->data, le32_to_cpu(blk->len),
+                                      GFP_KERNEL | GFP_DMA);
+                        if (!buf) {
+                                adsp_err(dsp, "Out of memory\n");
+                                return -ENOMEM;
+                        }
+
                         ret = regmap_raw_write(regmap, reg, blk->data,
                                                le32_to_cpu(blk->len));
                         if (ret != 0) {
@@ -433,6 +452,8 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp)
                                         "%s.%d: Failed to write to %x in %s\n",
                                         file, blocks, reg, region_name);
                         }
+
+                        kfree(buf);
                 }
 
                 pos += le32_to_cpu(blk->len) + sizeof(*blk);
diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig
index 3b98159d9645..a210c8d7b4bc 100644
--- a/sound/soc/fsl/Kconfig
+++ b/sound/soc/fsl/Kconfig
@@ -108,18 +108,13 @@ if SND_IMX_SOC
 config SND_SOC_IMX_SSI
         tristate
 
-config SND_SOC_IMX_PCM
-        tristate
-
 config SND_SOC_IMX_PCM_FIQ
-        bool
+        tristate
         select FIQ
-        select SND_SOC_IMX_PCM
 
 config SND_SOC_IMX_PCM_DMA
-        bool
+        tristate
         select SND_SOC_DMAENGINE_PCM
-        select SND_SOC_IMX_PCM
 
 config SND_SOC_IMX_AUDMUX
         tristate
diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile
index afd34794db53..ec1457915d7c 100644
--- a/sound/soc/fsl/Makefile
+++ b/sound/soc/fsl/Makefile
@@ -41,7 +41,10 @@ endif
 obj-$(CONFIG_SND_SOC_IMX_SSI) += snd-soc-imx-ssi.o
 obj-$(CONFIG_SND_SOC_IMX_AUDMUX) += snd-soc-imx-audmux.o
 
-obj-$(CONFIG_SND_SOC_IMX_PCM) += snd-soc-imx-pcm.o
+obj-$(CONFIG_SND_SOC_IMX_PCM_FIQ) += snd-soc-imx-pcm-fiq.o
+snd-soc-imx-pcm-fiq-y := imx-pcm-fiq.o imx-pcm.o
+obj-$(CONFIG_SND_SOC_IMX_PCM_DMA) += snd-soc-imx-pcm-dma.o
+snd-soc-imx-pcm-dma-y := imx-pcm-dma.o imx-pcm.o
 
 # i.MX Machine Support
 snd-soc-eukrea-tlv320-objs := eukrea-tlv320.o
diff --git a/sound/soc/fsl/imx-pcm.c b/sound/soc/fsl/imx-pcm.c
index d5cd9eff3b48..0c9f188ddc68 100644
--- a/sound/soc/fsl/imx-pcm.c
+++ b/sound/soc/fsl/imx-pcm.c
@@ -31,7 +31,6 @@ int snd_imx_pcm_mmap(struct snd_pcm_substream *substream,
                         runtime->dma_bytes);
         return ret;
 }
-EXPORT_SYMBOL_GPL(snd_imx_pcm_mmap);
 
 static int imx_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream)
 {
@@ -80,7 +79,6 @@ int imx_pcm_new(struct snd_soc_pcm_runtime *rtd)
 out:
         return ret;
 }
-EXPORT_SYMBOL_GPL(imx_pcm_new);
 
 void imx_pcm_free(struct snd_pcm *pcm)
 {
@@ -102,7 +100,6 @@ void imx_pcm_free(struct snd_pcm *pcm)
                 buf->area = NULL;
         }
 }
-EXPORT_SYMBOL_GPL(imx_pcm_free);
 
 MODULE_DESCRIPTION("Freescale i.MX PCM driver");
 MODULE_AUTHOR("Sascha Hauer <s.hauer@pengutronix.de>");
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 91d592ff67b7..2370063b5824 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -1255,6 +1255,8 @@ static int soc_post_component_init(struct snd_soc_card *card,
         INIT_LIST_HEAD(&rtd->dpcm[SNDRV_PCM_STREAM_CAPTURE].fe_clients);
         ret = device_add(rtd->dev);
         if (ret < 0) {
+                /* calling put_device() here to free the rtd->dev */
+                put_device(rtd->dev);
                 dev_err(card->dev,
                         "ASoC: failed to register runtime device: %d\n", ret);
                 return ret;
@@ -1554,7 +1556,7 @@ static void soc_remove_aux_dev(struct snd_soc_card *card, int num)
         /* unregister the rtd device */
         if (rtd->dev_registered) {
                 device_remove_file(rtd->dev, &dev_attr_codec_reg);
-                device_del(rtd->dev);
+                device_unregister(rtd->dev);
                 rtd->dev_registered = 0;
         }
 
@@ -2917,7 +2919,7 @@ int snd_soc_info_volsw_range(struct snd_kcontrol *kcontrol,
         platform_max = mc->platform_max;
 
         uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
-        uinfo->count = 1;
+        uinfo->count = snd_soc_volsw_is_stereo(mc) ? 2 : 1;
         uinfo->value.integer.min = 0;
         uinfo->value.integer.max = platform_max - min;
 
@@ -2941,12 +2943,14 @@ int snd_soc_put_volsw_range(struct snd_kcontrol *kcontrol,
                 (struct soc_mixer_control *)kcontrol->private_value;
         struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
         unsigned int reg = mc->reg;
+        unsigned int rreg = mc->rreg;
         unsigned int shift = mc->shift;
         int min = mc->min;
         int max = mc->max;
         unsigned int mask = (1 << fls(max)) - 1;
         unsigned int invert = mc->invert;
         unsigned int val, val_mask;
+        int ret;
 
         val = ((ucontrol->value.integer.value[0] + min) & mask);
         if (invert)
@@ -2954,7 +2958,21 @@ int snd_soc_put_volsw_range(struct snd_kcontrol *kcontrol,
         val_mask = mask << shift;
         val = val << shift;
 
-        return snd_soc_update_bits_locked(codec, reg, val_mask, val);
+        ret = snd_soc_update_bits_locked(codec, reg, val_mask, val);
+        if (ret != 0)
+                return ret;
+
+        if (snd_soc_volsw_is_stereo(mc)) {
+                val = ((ucontrol->value.integer.value[1] + min) & mask);
+                if (invert)
+                        val = max - val;
+                val_mask = mask << shift;
+                val = val << shift;
+
+                ret = snd_soc_update_bits_locked(codec, rreg, val_mask, val);
+        }
+
+        return ret;
 }
 EXPORT_SYMBOL_GPL(snd_soc_put_volsw_range);
 
@@ -2974,6 +2992,7 @@ int snd_soc_get_volsw_range(struct snd_kcontrol *kcontrol,
                 (struct soc_mixer_control *)kcontrol->private_value;
         struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
         unsigned int reg = mc->reg;
+        unsigned int rreg = mc->rreg;
         unsigned int shift = mc->shift;
         int min = mc->min;
         int max = mc->max;
@@ -2988,6 +3007,16 @@ int snd_soc_get_volsw_range(struct snd_kcontrol *kcontrol,
         ucontrol->value.integer.value[0] =
                 ucontrol->value.integer.value[0] - min;
 
+        if (snd_soc_volsw_is_stereo(mc)) {
+                ucontrol->value.integer.value[1] =
+                        (snd_soc_read(codec, rreg) >> shift) & mask;
+                if (invert)
+                        ucontrol->value.integer.value[1] =
+                                max - ucontrol->value.integer.value[1];
+                ucontrol->value.integer.value[1] =
+                        ucontrol->value.integer.value[1] - min;
+        }
+
         return 0;
 }
 EXPORT_SYMBOL_GPL(snd_soc_get_volsw_range);
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 1e36bc81e5af..258acadb9e7d 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -1023,7 +1023,7 @@ int dapm_regulator_event(struct snd_soc_dapm_widget *w,
 
         if (SND_SOC_DAPM_EVENT_ON(event)) {
                 if (w->invert & SND_SOC_DAPM_REGULATOR_BYPASS) {
-                        ret = regulator_allow_bypass(w->regulator, true);
+                        ret = regulator_allow_bypass(w->regulator, false);
                         if (ret != 0)
                                 dev_warn(w->dapm->dev,
                                          "ASoC: Failed to bypass %s: %d\n",
@@ -1033,7 +1033,7 @@ int dapm_regulator_event(struct snd_soc_dapm_widget *w,
                 return regulator_enable(w->regulator);
         } else {
                 if (w->invert & SND_SOC_DAPM_REGULATOR_BYPASS) {
-                        ret = regulator_allow_bypass(w->regulator, false);
+                        ret = regulator_allow_bypass(w->regulator, true);
                         if (ret != 0)
                                 dev_warn(w->dapm->dev,
                                          "ASoC: Failed to unbypass %s: %d\n",
@@ -3039,6 +3039,14 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm,
                                 w->name, ret);
                         return NULL;
                 }
+
+                if (w->invert & SND_SOC_DAPM_REGULATOR_BYPASS) {
+                        ret = regulator_allow_bypass(w->regulator, true);
+                        if (ret != 0)
+                                dev_warn(w->dapm->dev,
+                                         "ASoC: Failed to unbypass %s: %d\n",
+                                         w->name, ret);
+                }
                 break;
         case snd_soc_dapm_clock_supply:
 #ifdef CONFIG_CLKDEV_LOOKUP
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index d7711fce119b..cf191e6aebbe 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -1243,6 +1243,7 @@ static int dpcm_be_dai_hw_free(struct snd_soc_pcm_runtime *fe, int stream)
                 if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_HW_PARAMS) &&
                     (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PREPARE) &&
                     (be->dpcm[stream].state != SND_SOC_DPCM_STATE_HW_FREE) &&
+                    (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PAUSED) &&
                     (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP))
                         continue;