diff options
| -rw-r--r-- | sound/soc/codecs/sgtl5000.c | 2 | ||||
| -rw-r--r-- | sound/soc/codecs/wm8993.c | 6 | ||||
| -rw-r--r-- | sound/soc/imx/imx-pcm-dma-mx2.c | 12 | ||||
| -rw-r--r-- | sound/soc/soc-core.c | 4 | ||||
| -rw-r--r-- | sound/soc/soc-dapm.c | 2 |
5 files changed, 19 insertions, 7 deletions
diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index d7bd91831611..f8863ebb4304 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 2b40c93601ed..7c7fd925db8d 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 aecdba9f65a1..5780c9b9d569 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 3986520b4677..b5ecf6d23214 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 3ad1f59b8028..1f55ded4047f 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; |