Merge branch 'fix/asoc' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

* 'fix/asoc' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
  ASoC: Wait for WM8993 FLL to stabilise
  ASoC: core - Free platform DAPM context at platform removal.
  ASoC: dapm - Fix check for codec context in dapm_power_widgets().
  ASoC: sgtl5000: update author email address
  ASoC: Fix DMA channel leak in imx-pcm-dma-mx2 driver.

5 files changed, 19 insertions, 7 deletions

diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c
index d7bd918..f8863eb 100644
--- a/sound/soc/codecs/sgtl5000.c
+++ b/sound/soc/codecs/sgtl5000.c
@@ -1457,5 +1457,5 @@ static void __exit sgtl5000_exit(void)
 module_exit(sgtl5000_exit);
 
 MODULE_DESCRIPTION("Freescale SGTL5000 ALSA SoC Codec Driver");
-MODULE_AUTHOR("Zeng Zhaoming <zhaoming.zeng@freescale.com>");
+MODULE_AUTHOR("Zeng Zhaoming <zengzm.kernel@gmail.com>");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c
index 2b40c93..7c7fd92 100644
--- a/sound/soc/codecs/wm8993.c
+++ b/sound/soc/codecs/wm8993.c
@@ -444,6 +444,12 @@ static int _wm8993_set_fll(struct snd_soc_codec *codec, int fll_id, int source,
 	/* Enable the FLL */
 	snd_soc_write(codec, WM8993_FLL_CONTROL_1, reg1 | WM8993_FLL_ENA);
 
+	/* Both overestimates */
+	if (Fref < 1000000)
+		msleep(3);
+	else
+		msleep(1);
+
 	dev_dbg(codec->dev, "FLL enabled at %dHz->%dHz\n", Fref, Fout);
 
 	wm8993->fll_fref = Fref;
diff --git a/sound/soc/imx/imx-pcm-dma-mx2.c b/sound/soc/imx/imx-pcm-dma-mx2.c
index aecdba9..5780c9b 100644
--- a/sound/soc/imx/imx-pcm-dma-mx2.c
+++ b/sound/soc/imx/imx-pcm-dma-mx2.c
@@ -88,11 +88,13 @@ static int imx_ssi_dma_alloc(struct snd_pcm_substream *substream,
 	iprtd->dma_data.dma_request = dma_params->dma;
 
 	/* Try to grab a DMA channel */
-	dma_cap_zero(mask);
-	dma_cap_set(DMA_SLAVE, mask);
-	iprtd->dma_chan = dma_request_channel(mask, filter, iprtd);
-	if (!iprtd->dma_chan)
-		return -EINVAL;
+	if (!iprtd->dma_chan) {
+		dma_cap_zero(mask);
+		dma_cap_set(DMA_SLAVE, mask);
+		iprtd->dma_chan = dma_request_channel(mask, filter, iprtd);
+		if (!iprtd->dma_chan)
+			return -EINVAL;
+	}
 
 	switch (params_format(params)) {
 	case SNDRV_PCM_FORMAT_S16_LE:
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 3986520..b5ecf6d 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -907,6 +907,10 @@ static void soc_remove_dai_link(struct snd_soc_card *card, int num, int order)
 			if (err < 0)
 				printk(KERN_ERR "asoc: failed to remove %s\n", platform->name);
 		}
+
+		/* Make sure all DAPM widgets are freed */
+		snd_soc_dapm_free(&platform->dapm);
+
 		platform->probed = 0;
 		list_del(&platform->card_list);
 		module_put(platform->dev->driver->owner);
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 3ad1f59..1f55ded 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -1426,7 +1426,7 @@ static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event)
 			dapm->target_bias_level = SND_SOC_BIAS_ON;
 			break;
 		case SND_SOC_DAPM_STREAM_STOP:
-			if (dapm->codec->active)
+			if (dapm->codec && dapm->codec->active)
 				dapm->target_bias_level = SND_SOC_BIAS_ON;
 			else
 				dapm->target_bias_level = SND_SOC_BIAS_STANDBY;