Merge branch 'fixes-for-arnd' of git://git.pengutronix.de/git/imx/linux-2.6 into imx/fixes

14 files changed, 50 insertions, 49 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/wm8991.c b/sound/soc/codecs/wm8991.c
index 3c2ee1bb73cd..6af23d06870f 100644
--- a/sound/soc/codecs/wm8991.c
+++ b/sound/soc/codecs/wm8991.c
@@ -13,7 +13,6 @@
 
 #include <linux/module.h>
 #include <linux/moduleparam.h>
-#include <linux/version.h>
 #include <linux/kernel.h>
 #include <linux/init.h>
 #include <linux/delay.h>
diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index 970a95c5360b..c2fc0356c2a4 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -1713,6 +1713,8 @@ 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;
diff --git a/sound/soc/imx/Kconfig b/sound/soc/imx/Kconfig
index d8f130d39dd9..bb699bb55a50 100644
--- a/sound/soc/imx/Kconfig
+++ b/sound/soc/imx/Kconfig
@@ -11,9 +11,6 @@ menuconfig SND_IMX_SOC
 
 if SND_IMX_SOC
 
-config SND_MXC_SOC_SSI
-	tristate
-
 config SND_MXC_SOC_FIQ
 	tristate
 
@@ -24,7 +21,6 @@ config SND_MXC_SOC_WM1133_EV1
 	tristate "Audio on the the i.MX31ADS with WM1133-EV1 fitted"
 	depends on MACH_MX31ADS_WM1133_EV1 && EXPERIMENTAL
 	select SND_SOC_WM8350
-	select SND_MXC_SOC_SSI
 	select SND_MXC_SOC_FIQ
 	help
 	  Enable support for audio on the i.MX31ADS with the WM1133-EV1
@@ -34,7 +30,6 @@ config SND_SOC_MX27VIS_AIC32X4
 	tristate "SoC audio support for Visstrim M10 boards"
 	depends on MACH_IMX27_VISSTRIM_M10
 	select SND_SOC_TVL320AIC32X4
-	select SND_MXC_SOC_SSI
 	select SND_MXC_SOC_MX2
 	help
 	  Say Y if you want to add support for SoC audio on Visstrim SM10
@@ -44,7 +39,6 @@ config SND_SOC_PHYCORE_AC97
 	tristate "SoC Audio support for Phytec phyCORE (and phyCARD) boards"
 	depends on MACH_PCM043 || MACH_PCA100
 	select SND_SOC_WM9712
-	select SND_MXC_SOC_SSI
 	select SND_MXC_SOC_FIQ
 	help
 	  Say Y if you want to add support for SoC audio on Phytec phyCORE
@@ -57,7 +51,6 @@ config SND_SOC_EUKREA_TLV320
 		|| MACH_EUKREA_MBIMXSD35_BASEBOARD \
 		|| MACH_EUKREA_MBIMXSD51_BASEBOARD
 	select SND_SOC_TLV320AIC23
-	select SND_MXC_SOC_SSI
 	select SND_MXC_SOC_FIQ
 	help
 	  Enable I2S based access to the TLV320AIC23B codec attached
diff --git a/sound/soc/imx/imx-pcm-dma-mx2.c b/sound/soc/imx/imx-pcm-dma-mx2.c
index b2ed764fd896..43fdc24f7e8d 100644
--- a/sound/soc/imx/imx-pcm-dma-mx2.c
+++ b/sound/soc/imx/imx-pcm-dma-mx2.c
@@ -337,3 +337,5 @@ static void __exit snd_imx_pcm_exit(void)
 	platform_driver_unregister(&imx_pcm_driver);
 }
 module_exit(snd_imx_pcm_exit);
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:imx-pcm-audio");
diff --git a/sound/soc/imx/imx-ssi.c b/sound/soc/imx/imx-ssi.c
index 5b13feca7537..61fceb09cdb5 100644
--- a/sound/soc/imx/imx-ssi.c
+++ b/sound/soc/imx/imx-ssi.c
@@ -774,4 +774,4 @@ module_exit(imx_ssi_exit);
 MODULE_AUTHOR("Sascha Hauer, <s.hauer@pengutronix.de>");
 MODULE_DESCRIPTION("i.MX I2S/ac97 SoC Interface");
 MODULE_LICENSE("GPL");
-
+MODULE_ALIAS("platform:imx-ssi");
diff --git a/sound/soc/pxa/pxa2xx-pcm.c b/sound/soc/pxa/pxa2xx-pcm.c
index 2ce0b2d891d5..fab20a54e863 100644
--- a/sound/soc/pxa/pxa2xx-pcm.c
+++ b/sound/soc/pxa/pxa2xx-pcm.c
@@ -95,14 +95,14 @@ static int pxa2xx_soc_pcm_new(struct snd_card *card, struct snd_soc_dai *dai,
 	if (!card->dev->coherent_dma_mask)
 		card->dev->coherent_dma_mask = DMA_BIT_MASK(32);
 
-	if (dai->driver->playback.channels_min) {
+	if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) {
 		ret = pxa2xx_pcm_preallocate_dma_buffer(pcm,
 			SNDRV_PCM_STREAM_PLAYBACK);
 		if (ret)
 			goto out;
 	}
 
-	if (dai->driver->capture.channels_min) {
+	if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) {
 		ret = pxa2xx_pcm_preallocate_dma_buffer(pcm,
 			SNDRV_PCM_STREAM_CAPTURE);
 		if (ret)
diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c
index c005ceb70c9d..039b9532b270 100644
--- a/sound/soc/soc-cache.c
+++ b/sound/soc/soc-cache.c
@@ -409,9 +409,6 @@ int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec,
 	codec->bulk_write_raw = snd_soc_hw_bulk_write_raw;
 
 	switch (control) {
-	case SND_SOC_CUSTOM:
-		break;
-
 	case SND_SOC_I2C:
 #if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE))
 		codec->hw_write = (hw_write_t)i2c_master_send;
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;
 }