11 files changed, 61 insertions, 50 deletions

diff --git a/sound/soc/blackfin/bf5xx-i2s-pcm.c b/sound/soc/blackfin/bf5xx-i2s-pcm.c
index b5101efd1c87..f1fd95bb6416 100644
--- a/sound/soc/blackfin/bf5xx-i2s-pcm.c
+++ b/sound/soc/blackfin/bf5xx-i2s-pcm.c
@@ -138,11 +138,20 @@ static snd_pcm_uframes_t bf5xx_pcm_pointer(struct snd_pcm_substream *substream)
         pr_debug("%s enter\n", __func__);
         if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
                 diff = sport_curr_offset_tx(sport);
-                frames = bytes_to_frames(substream->runtime, diff);
         } else {
                 diff = sport_curr_offset_rx(sport);
-                frames = bytes_to_frames(substream->runtime, diff);
         }
+
+        /*
+         * TX at least can report one frame beyond the end of the
+         * buffer if we hit the wraparound case - clamp to within the
+         * buffer as the ALSA APIs require.
+         */
+        if (diff == snd_pcm_lib_buffer_bytes(substream))
+                diff = 0;
+
+        frames = bytes_to_frames(substream->runtime, diff);
+
         return frames;
 }
 
diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c
index 4be0570e3f1f..65f46047b1cb 100644
--- a/sound/soc/codecs/ak4642.c
+++ b/sound/soc/codecs/ak4642.c
@@ -357,7 +357,7 @@ static int ak4642_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
         default:
                 return -EINVAL;
         }
-        snd_soc_update_bits(codec, PW_MGMT2, MS, data);
+        snd_soc_update_bits(codec, PW_MGMT2, MS | MCKO | PMPLL, data);
         snd_soc_update_bits(codec, MD_CTL1, BCKO_MASK, bcko);
 
         /* format type */
diff --git a/sound/soc/codecs/tlv320aic26.c b/sound/soc/codecs/tlv320aic26.c
index e2a7608d3944..7859bdcc93db 100644
--- a/sound/soc/codecs/tlv320aic26.c
+++ b/sound/soc/codecs/tlv320aic26.c
@@ -161,10 +161,18 @@ static int aic26_hw_params(struct snd_pcm_substream *substream,
                 dev_dbg(&aic26->spi->dev, "bad format\n"); return -EINVAL;
         }
 
-        /* Configure PLL */
+        /**
+         * Configure PLL
+         * fsref = (mclk * PLLM) / 2048
+         * where PLLM = J.DDDD (DDDD register ranges from 0 to 9999, decimal)
+         */
         pval = 1;
-        jval = (fsref == 44100) ? 7 : 8;
-        dval = (fsref == 44100) ? 5264 : 1920;
+        /* compute J portion of multiplier */
+        jval = fsref / (aic26->mclk / 2048);
+        /* compute fractional DDDD component of multiplier */
+        dval = fsref - (jval * (aic26->mclk / 2048));
+        dval = (10000 * dval) / (aic26->mclk / 2048);
+        dev_dbg(&aic26->spi->dev, "Setting PLLM to %d.%04d\n", jval, dval);
         qval = 0;
         reg = 0x8000 | qval << 11 | pval << 8 | jval << 2;
         aic26_reg_write(codec, AIC26_REG_PLL_PROG1, reg);
diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c
index c3d96fc8c267..789453d44ec5 100644
--- a/sound/soc/codecs/tlv320aic3x.c
+++ b/sound/soc/codecs/tlv320aic3x.c
@@ -1114,12 +1114,19 @@ static int aic3x_set_power(struct snd_soc_codec *codec, int power)
 
                 /* Sync reg_cache with the hardware */
                 codec->cache_only = 0;
-                for (i = 0; i < ARRAY_SIZE(aic3x_reg); i++)
+                for (i = AIC3X_SAMPLE_RATE_SEL_REG; i < ARRAY_SIZE(aic3x_reg); i++)
                         snd_soc_write(codec, i, cache[i]);
                 if (aic3x->model == AIC3X_MODEL_3007)
                         aic3x_init_3007(codec);
                 codec->cache_sync = 0;
         } else {
+                /*
+                 * Do soft reset to this codec instance in order to clear
+                 * possible VDD leakage currents in case the supply regulators
+                 * remain on
+                 */
+                snd_soc_write(codec, AIC3X_RESET, SOFT_RESET);
+                codec->cache_sync = 1;
                 aic3x->power = 0;
                 /* HW writes are needless when bias is off */
                 codec->cache_only = 1;
diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c
index 2dc964b55e4f..76b4361e9b80 100644
--- a/sound/soc/codecs/wm8731.c
+++ b/sound/soc/codecs/wm8731.c
@@ -175,6 +175,7 @@ static const struct snd_kcontrol_new wm8731_input_mux_controls =
 SOC_DAPM_ENUM("Input Select", wm8731_insel_enum);
 
 static const struct snd_soc_dapm_widget wm8731_dapm_widgets[] = {
+SND_SOC_DAPM_SUPPLY("ACTIVE",WM8731_ACTIVE, 0, 0, NULL, 0),
 SND_SOC_DAPM_SUPPLY("OSC", WM8731_PWR, 5, 1, NULL, 0),
 SND_SOC_DAPM_MIXER("Output Mixer", WM8731_PWR, 4, 1,
         &wm8731_output_mixer_controls[0],
@@ -204,6 +205,8 @@ static int wm8731_check_osc(struct snd_soc_dapm_widget *source,
 static const struct snd_soc_dapm_route wm8731_intercon[] = {
         {"DAC", NULL, "OSC", wm8731_check_osc},
         {"ADC", NULL, "OSC", wm8731_check_osc},
+        {"DAC", NULL, "ACTIVE"},
+        {"ADC", NULL, "ACTIVE"},
 
         /* output mixer */
         {"Output Mixer", "Line Bypass Switch", "Line Input"},
@@ -315,29 +318,6 @@ static int wm8731_hw_params(struct snd_pcm_substream *substream,
         return 0;
 }
 
-static int wm8731_pcm_prepare(struct snd_pcm_substream *substream,
-                              struct snd_soc_dai *dai)
-{
-        struct snd_soc_codec *codec = dai->codec;
-
-        /* set active */
-        snd_soc_write(codec, WM8731_ACTIVE, 0x0001);
-
-        return 0;
-}
-
-static void wm8731_shutdown(struct snd_pcm_substream *substream,
-                            struct snd_soc_dai *dai)
-{
-        struct snd_soc_codec *codec = dai->codec;
-
-        /* deactivate */
-        if (!codec->active) {
-                udelay(50);
-                snd_soc_write(codec, WM8731_ACTIVE, 0x0);
-        }
-}
-
 static int wm8731_mute(struct snd_soc_dai *dai, int mute)
 {
         struct snd_soc_codec *codec = dai->codec;
@@ -480,7 +460,6 @@ static int wm8731_set_bias_level(struct snd_soc_codec *codec,
                 snd_soc_write(codec, WM8731_PWR, reg | 0x0040);
                 break;
         case SND_SOC_BIAS_OFF:
-                snd_soc_write(codec, WM8731_ACTIVE, 0x0);
                 snd_soc_write(codec, WM8731_PWR, 0xffff);
                 regulator_bulk_disable(ARRAY_SIZE(wm8731->supplies),
                                        wm8731->supplies);
@@ -496,9 +475,7 @@ static int wm8731_set_bias_level(struct snd_soc_codec *codec,
         SNDRV_PCM_FMTBIT_S24_LE)
 
 static struct snd_soc_dai_ops wm8731_dai_ops = {
-        .prepare        = wm8731_pcm_prepare,
         .hw_params      = wm8731_hw_params,
-        .shutdown       = wm8731_shutdown,
         .digital_mute   = wm8731_mute,
         .set_sysclk     = wm8731_set_dai_sysclk,
         .set_fmt        = wm8731_set_dai_fmt,
diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index 970a95c5360b..83014a7c2e14 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -1190,7 +1190,6 @@ SND_SOC_DAPM_INPUT("DMIC1DAT"),
 SND_SOC_DAPM_INPUT("DMIC2DAT"),
 SND_SOC_DAPM_INPUT("Clock"),
 
-SND_SOC_DAPM_MICBIAS("MICBIAS", WM8994_MICBIAS, 2, 0),
 SND_SOC_DAPM_SUPPLY_S("MICBIAS Supply", 1, SND_SOC_NOPM, 0, 0, micbias_ev,
                       SND_SOC_DAPM_PRE_PMU),
 
@@ -1509,8 +1508,10 @@ static const struct snd_soc_dapm_route wm8994_revd_intercon[] = {
         { "AIF2DACDAT", NULL, "AIF1DACDAT" },
         { "AIF1ADCDAT", NULL, "AIF2ADCDAT" },
         { "AIF2ADCDAT", NULL, "AIF1ADCDAT" },
-        { "MICBIAS", NULL, "CLK_SYS" },
-        { "MICBIAS", NULL, "MICBIAS Supply" },
+        { "MICBIAS1", NULL, "CLK_SYS" },
+        { "MICBIAS1", NULL, "MICBIAS Supply" },
+        { "MICBIAS2", NULL, "CLK_SYS" },
+        { "MICBIAS2", NULL, "MICBIAS Supply" },
 };
 
 static const struct snd_soc_dapm_route wm8994_intercon[] = {
@@ -1713,6 +1714,8 @@ static int _wm8994_set_fll(struct snd_soc_codec *codec, int id, int src,
                 snd_soc_update_bits(codec, WM8994_FLL1_CONTROL_1 + reg_offset,
                                     WM8994_FLL1_ENA | WM8994_FLL1_FRAC,
                                     reg);
+
+                msleep(5);
         }
 
         wm8994->fll[id].in = freq_in;
@@ -2761,7 +2764,7 @@ static void wm8958_default_micdet(u16 status, void *data)
         report = SND_JACK_MICROPHONE;
 
         /* Everything else is buttons; just assign slots */
-        if (status & 0x1c0)
+        if (status & 0x1c)
                 report |= SND_JACK_BTN_0;
 
 done:
diff --git a/sound/soc/sh/fsi-ak4642.c b/sound/soc/sh/fsi-ak4642.c
index d6f4703b3c07..770a71a15366 100644
--- a/sound/soc/sh/fsi-ak4642.c
+++ b/sound/soc/sh/fsi-ak4642.c
@@ -97,7 +97,7 @@ static int fsi_ak4642_remove(struct platform_device *pdev)
 
 static struct fsi_ak4642_data fsi_a_ak4642 = {
         .name           = "AK4642",
-        .card           = "FSIA (AK4642)",
+        .card           = "FSIA-AK4642",
         .cpu_dai        = "fsia-dai",
         .codec          = "ak4642-codec.0-0012",
         .platform       = "sh_fsi.0",
@@ -106,7 +106,7 @@ static struct fsi_ak4642_data fsi_a_ak4642 = {
 
 static struct fsi_ak4642_data fsi_b_ak4642 = {
         .name           = "AK4642",
-        .card           = "FSIB (AK4642)",
+        .card           = "FSIB-AK4642",
         .cpu_dai        = "fsib-dai",
         .codec          = "ak4642-codec.0-0012",
         .platform       = "sh_fsi.0",
@@ -115,7 +115,7 @@ static struct fsi_ak4642_data fsi_b_ak4642 = {
 
 static struct fsi_ak4642_data fsi_a_ak4643 = {
         .name           = "AK4643",
-        .card           = "FSIA (AK4643)",
+        .card           = "FSIA-AK4643",
         .cpu_dai        = "fsia-dai",
         .codec          = "ak4642-codec.0-0013",
         .platform       = "sh_fsi.0",
@@ -124,7 +124,7 @@ static struct fsi_ak4642_data fsi_a_ak4643 = {
 
 static struct fsi_ak4642_data fsi_b_ak4643 = {
         .name           = "AK4643",
-        .card           = "FSIB (AK4643)",
+        .card           = "FSIB-AK4643",
         .cpu_dai        = "fsib-dai",
         .codec          = "ak4642-codec.0-0013",
         .platform       = "sh_fsi.0",
@@ -133,7 +133,7 @@ static struct fsi_ak4642_data fsi_b_ak4643 = {
 
 static struct fsi_ak4642_data fsi2_a_ak4642 = {
         .name           = "AK4642",
-        .card           = "FSI2A (AK4642)",
+        .card           = "FSI2A-AK4642",
         .cpu_dai        = "fsia-dai",
         .codec          = "ak4642-codec.0-0012",
         .platform       = "sh_fsi2",
@@ -142,7 +142,7 @@ static struct fsi_ak4642_data fsi2_a_ak4642 = {
 
 static struct fsi_ak4642_data fsi2_b_ak4642 = {
         .name           = "AK4642",
-        .card           = "FSI2B (AK4642)",
+        .card           = "FSI2B-AK4642",
         .cpu_dai        = "fsib-dai",
         .codec          = "ak4642-codec.0-0012",
         .platform       = "sh_fsi2",
@@ -151,7 +151,7 @@ static struct fsi_ak4642_data fsi2_b_ak4642 = {
 
 static struct fsi_ak4642_data fsi2_a_ak4643 = {
         .name           = "AK4643",
-        .card           = "FSI2A (AK4643)",
+        .card           = "FSI2A-AK4643",
         .cpu_dai        = "fsia-dai",
         .codec          = "ak4642-codec.0-0013",
         .platform       = "sh_fsi2",
@@ -160,7 +160,7 @@ static struct fsi_ak4642_data fsi2_a_ak4643 = {
 
 static struct fsi_ak4642_data fsi2_b_ak4643 = {
         .name           = "AK4643",
-        .card           = "FSI2B (AK4643)",
+        .card           = "FSI2B-AK4643",
         .cpu_dai        = "fsib-dai",
         .codec          = "ak4642-codec.0-0013",
         .platform       = "sh_fsi2",
diff --git a/sound/soc/sh/fsi-da7210.c b/sound/soc/sh/fsi-da7210.c
index dbafd7ac5590..59553fd8c2fb 100644
--- a/sound/soc/sh/fsi-da7210.c
+++ b/sound/soc/sh/fsi-da7210.c
@@ -42,7 +42,7 @@ static struct snd_soc_dai_link fsi_da7210_dai = {
 };
 
 static struct snd_soc_card fsi_soc_card = {
-        .name           = "FSI (DA7210)",
+        .name           = "FSI-DA7210",
         .dai_link       = &fsi_da7210_dai,
         .num_links      = 1,
 };
diff --git a/sound/soc/sh/fsi-hdmi.c b/sound/soc/sh/fsi-hdmi.c
index 9719985eb82d..d3d9fd880680 100644
--- a/sound/soc/sh/fsi-hdmi.c
+++ b/sound/soc/sh/fsi-hdmi.c
@@ -83,13 +83,13 @@ static int fsi_hdmi_remove(struct platform_device *pdev)
 
 static struct fsi_hdmi_data fsi2_a_hdmi = {
         .cpu_dai        = "fsia-dai",
-        .card           = "FSI2A (SH MOBILE HDMI)",
+        .card           = "FSI2A-HDMI",
         .id             = FSI_PORT_A,
 };
 
 static struct fsi_hdmi_data fsi2_b_hdmi = {
         .cpu_dai        = "fsib-dai",
-        .card           = "FSI2B (SH MOBILE HDMI)",
+        .card           = "FSI2B-HDMI",
         .id             = FSI_PORT_B,
 };
 
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index d75043ed7fc0..b194be09e74d 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -1929,8 +1929,9 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card)
                  "%s", card->name);
         snprintf(card->snd_card->longname, sizeof(card->snd_card->longname),
                  "%s", card->long_name ? card->long_name : card->name);
-        snprintf(card->snd_card->driver, sizeof(card->snd_card->driver),
-                 "%s", card->driver_name ? card->driver_name : card->name);
+        if (card->driver_name)
+                strlcpy(card->snd_card->driver, card->driver_name,
+                        sizeof(card->snd_card->driver));
 
         if (card->late_probe) {
                 ret = card->late_probe(card);
diff --git a/sound/soc/tegra/tegra_i2s.c b/sound/soc/tegra/tegra_i2s.c
index 6b817e20548c..95f03c10b4f7 100644
--- a/sound/soc/tegra/tegra_i2s.c
+++ b/sound/soc/tegra/tegra_i2s.c
@@ -222,12 +222,18 @@ static int tegra_i2s_hw_params(struct snd_pcm_substream *substream,
         if (i2sclock % (2 * srate))
                 reg |= TEGRA_I2S_TIMING_NON_SYM_ENABLE;
 
+        if (!i2s->clk_refs)
+                clk_enable(i2s->clk_i2s);
+
         tegra_i2s_write(i2s, TEGRA_I2S_TIMING, reg);
 
         tegra_i2s_write(i2s, TEGRA_I2S_FIFO_SCR,
                 TEGRA_I2S_FIFO_SCR_FIFO2_ATN_LVL_FOUR_SLOTS |
                 TEGRA_I2S_FIFO_SCR_FIFO1_ATN_LVL_FOUR_SLOTS);
 
+        if (!i2s->clk_refs)
+                clk_disable(i2s->clk_i2s);
+
         return 0;
 }