Merge remote-tracking branch 'asoc/fix/wm8994' into asoc-linus

diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c
index 0f6f481cec09cd941688ed790ecd31e10e207496..030f53c96ec0b2b0f14b86d4d428e5e1ec07f505 100644 (file)
--- a/sound/soc/codecs/cs42l52.c
+++ b/sound/soc/codecs/cs42l52.c
@@ -86,7 +86,7 @@ static const struct reg_default cs42l52_reg_defaults[] = {
        { CS42L52_BEEP_VOL, 0x00 },     /* r1D Beep Volume off Time */
        { CS42L52_BEEP_TONE_CTL, 0x00 },        /* r1E Beep Tone Cfg. */
        { CS42L52_TONE_CTL, 0x00 },     /* r1F Tone Ctl */
-       { CS42L52_MASTERA_VOL, 0x88 },  /* r20 Master A Volume */
+       { CS42L52_MASTERA_VOL, 0x00 },  /* r20 Master A Volume */
        { CS42L52_MASTERB_VOL, 0x00 },  /* r21 Master B Volume */
        { CS42L52_HPA_VOL, 0x00 },      /* r22 Headphone A Volume */
        { CS42L52_HPB_VOL, 0x00 },      /* r23 Headphone B Volume */
@@ -225,7 +225,7 @@ static const char * const mic_bias_level_text[] = {
 };
 
 static const struct soc_enum mic_bias_level_enum =
-       SOC_ENUM_SINGLE(CS42L52_IFACE_CTL1, 0,
+       SOC_ENUM_SINGLE(CS42L52_IFACE_CTL2, 0,
                        ARRAY_SIZE(mic_bias_level_text), mic_bias_level_text);
 
 static const char * const cs42l52_mic_text[] = { "Single", "Differential" };
@@ -413,7 +413,7 @@ static const struct snd_kcontrol_new cs42l52_snd_controls[] = {
        SOC_ENUM("Headphone Analog Gain", hp_gain_enum),
 
        SOC_DOUBLE_R_SX_TLV("Speaker Volume", CS42L52_SPKA_VOL,
-                             CS42L52_SPKB_VOL, 7, 0x1, 0xff, hl_tlv),
+                             CS42L52_SPKB_VOL, 0, 0x1, 0xff, hl_tlv),
 
        SOC_DOUBLE_R_SX_TLV("Bypass Volume", CS42L52_PASSTHRUA_VOL,
                              CS42L52_PASSTHRUB_VOL, 6, 0x18, 0x90, pga_tlv),
@@ -441,7 +441,7 @@ static const struct snd_kcontrol_new cs42l52_snd_controls[] = {
 
        SOC_DOUBLE_R_SX_TLV("PCM Mixer Volume",
                            CS42L52_PCMA_MIXER_VOL, CS42L52_PCMB_MIXER_VOL,
-                               6, 0x7f, 0x19, hl_tlv),
+                               0, 0x7f, 0x19, hl_tlv),
        SOC_DOUBLE_R("PCM Mixer Switch",
                     CS42L52_PCMA_MIXER_VOL, CS42L52_PCMB_MIXER_VOL, 7, 1, 1),
 

diff --git a/sound/soc/codecs/cs42l52.h b/sound/soc/codecs/cs42l52.h
index 60985c0590711e728fe2229fe459706d42d1ee57..4277012c471978ef1905a196c2d51cc43c636703 100644 (file)
--- a/sound/soc/codecs/cs42l52.h
+++ b/sound/soc/codecs/cs42l52.h
@@ -157,7 +157,7 @@
 #define CS42L52_PB_CTL1_INV_PCMA               (1 << 2)
 #define CS42L52_PB_CTL1_MSTB_MUTE              (1 << 1)
 #define CS42L52_PB_CTL1_MSTA_MUTE              (1 << 0)
-#define CS42L52_PB_CTL1_MUTE_MASK              0xFFFD
+#define CS42L52_PB_CTL1_MUTE_MASK              0x03
 #define CS42L52_PB_CTL1_MUTE                   3
 #define CS42L52_PB_CTL1_UNMUTE                 0
 

diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c
index ce0d36412c97d73eaa27c59de88c0ee44ce6066c..8d14a76c7249ade32bdc627fee92a5cefe6eaf91 100644 (file)
--- a/sound/soc/codecs/max98090.c
+++ b/sound/soc/codecs/max98090.c
@@ -2233,7 +2233,7 @@ static int max98090_probe(struct snd_soc_codec *codec)
        dev_dbg(codec->dev, "irq = %d\n", max98090->irq);
 
        ret = request_threaded_irq(max98090->irq, NULL,
-               max98090_interrupt, IRQF_TRIGGER_FALLING,
+               max98090_interrupt, IRQF_TRIGGER_FALLING | IRQF_ONESHOT,
                "max98090_interrupt", codec);
        if (ret < 0) {
                dev_err(codec->dev, "request_irq failed: %d\n",

diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c
index 731884e04776289449e0a135905eb883a217cb80..ba38f0679662784769aed8c0c6293f0da5d7e4ca 100644 (file)
--- a/sound/soc/codecs/wm5110.c
+++ b/sound/soc/codecs/wm5110.c
@@ -190,7 +190,7 @@ ARIZONA_MIXER_CONTROLS("DSP2R", ARIZONA_DSP2RMIX_INPUT_1_SOURCE),
 ARIZONA_MIXER_CONTROLS("DSP3L", ARIZONA_DSP3LMIX_INPUT_1_SOURCE),
 ARIZONA_MIXER_CONTROLS("DSP3R", ARIZONA_DSP3RMIX_INPUT_1_SOURCE),
 ARIZONA_MIXER_CONTROLS("DSP4L", ARIZONA_DSP4LMIX_INPUT_1_SOURCE),
-ARIZONA_MIXER_CONTROLS("DSP5R", ARIZONA_DSP4RMIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("DSP4R", ARIZONA_DSP4RMIX_INPUT_1_SOURCE),
 
 ARIZONA_MIXER_CONTROLS("Mic", ARIZONA_MICMIX_INPUT_1_SOURCE),
 ARIZONA_MIXER_CONTROLS("Noise", ARIZONA_NOISEMIX_INPUT_1_SOURCE),
@@ -976,6 +976,8 @@ static int wm5110_codec_probe(struct snd_soc_codec *codec)
        if (ret != 0)
                return ret;
 
+       arizona_init_spk(codec);
+
        snd_soc_dapm_disable_pin(&codec->dapm, "HAPTICS");
 
        priv->core.arizona->dapm = &codec->dapm;

diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c
index 56ecfc72f2e9500ebda81214d972b65f291cd518..81490febac6dc108decb890ac318ffbec9c2ab8b 100644 (file)
--- a/sound/soc/davinci/davinci-mcasp.c
+++ b/sound/soc/davinci/davinci-mcasp.c
@@ -631,7 +631,8 @@ static int davinci_config_channel_size(struct davinci_audio_dev *dev,
                                       int word_length)
 {
        u32 fmt;
-       u32 rotate = (word_length / 4) & 0x7;
+       u32 tx_rotate = (word_length / 4) & 0x7;
+       u32 rx_rotate = (32 - word_length) / 4;
        u32 mask = (1ULL << word_length) - 1;
 
        /*
@@ -655,9 +656,9 @@ static int davinci_config_channel_size(struct davinci_audio_dev *dev,
                mcasp_mod_bits(dev->base + DAVINCI_MCASP_TXFMT_REG,
                                TXSSZ(fmt), TXSSZ(0x0F));
                mcasp_mod_bits(dev->base + DAVINCI_MCASP_TXFMT_REG,
-                               TXROT(rotate), TXROT(7));
+                               TXROT(tx_rotate), TXROT(7));
                mcasp_mod_bits(dev->base + DAVINCI_MCASP_RXFMT_REG,
-                               RXROT(rotate), RXROT(7));
+                               RXROT(rx_rotate), RXROT(7));
                mcasp_set_reg(dev->base + DAVINCI_MCASP_RXMASK_REG,
                                mask);
        }

diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c
index 3853f7eb3f2843f35c8f908cf47caab47931952a..06a8000aa07bedd1c47beb401d26e9052512fc54 100644 (file)
--- a/sound/soc/soc-compress.c
+++ b/sound/soc/soc-compress.c
@@ -220,8 +220,12 @@ static int soc_compr_set_params(struct snd_compr_stream *cstream,
                        goto err;
        }
 
-       snd_soc_dapm_stream_event(rtd, SNDRV_PCM_STREAM_PLAYBACK,
-                               SND_SOC_DAPM_STREAM_START);
+       if (cstream->direction == SND_COMPRESS_PLAYBACK)
+               snd_soc_dapm_stream_event(rtd, SNDRV_PCM_STREAM_PLAYBACK,
+                                       SND_SOC_DAPM_STREAM_START);
+       else
+               snd_soc_dapm_stream_event(rtd, SNDRV_PCM_STREAM_CAPTURE,
+                                       SND_SOC_DAPM_STREAM_START);
 
        /* cancel any delayed stream shutdown that is pending */
        rtd->pop_wait = 0;