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author | Linus Torvalds <torvalds@linux-foundation.org> | 2013-04-05 09:06:57 -0700 |
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committer | Linus Torvalds <torvalds@linux-foundation.org> | 2013-04-05 09:06:57 -0700 |
commit | 8f09aacfa6cf64c469fe60c05dfc1bd75e8615ed (patch) | |
tree | 46503c5bce589638d727bfd5415ba0dfb82b9a0e /sound/soc | |
parent | d08d528dc1848fb369a0b27cdb0749d8f6f38063 (diff) | |
parent | 868211db6df96ddae411fcd800502725beef8387 (diff) | |
download | talos-op-linux-8f09aacfa6cf64c469fe60c05dfc1bd75e8615ed.tar.gz talos-op-linux-8f09aacfa6cf64c469fe60c05dfc1bd75e8615ed.zip |
Merge tag 'sound-3.9' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"This contains slightly more volumes than usual at this stage, mostly
because of my vacation in the last week. Nothing to scare, all small
and/or trivial fixes:
- Fix loop path handling in ASoC DAPM
- Some memory handling fixes in ASoC core
- Fix spear_pcm to adapt to the updated API
- HD-audio HDMI ELD handling fixes
- Fix for CM6331 USB-audio SRC change bugs
- Revert power_save_controller option change due to user-space usage
- A few other small ASoC and HD-audio fixes"
* tag 'sound-3.9' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ALSA: hda/generic - fix uninitialized variable
Revert "ALSA: hda - Allow power_save_controller option override DCAPS"
ALSA: hda - fix typo in proc output
ALSA: hda - Enabling Realtek ALC 671 codec
ALSA: usb: Work around CM6631 sample rate change bug
ALSA: hda - bug fix on HDMI ELD debug message
ALSA: hda - bug fix on return value when getting HDMI ELD info
ASoC: dma-sh7760: Fix compile error
ASoC: core: fix invalid free of devm_ allocated data
ASoC: spear_pcm: Update to new pcm_new() API
ASoC:: max98090: Remove executable bit
ASoC: dapm: Fix pointer dereference in is_connected_output_ep()
ASoC: pcm030 audio fabric: remove __init from probe
ASoC: imx-ssi: Fix occasional AC97 reset failure
ASoC: core: fix possible memory leak in snd_soc_bytes_put()
ASoC: wm_adsp: fix possible memory leak in wm_adsp_load_coeff()
ASoC: dapm: Fix handling of loops
ASoC: si476x: Add missing break for SNDRV_PCM_FORMAT_S8 switch case
Diffstat (limited to 'sound/soc')
-rw-r--r--[-rwxr-xr-x] | sound/soc/codecs/max98090.c | 0 | ||||
-rw-r--r--[-rwxr-xr-x] | sound/soc/codecs/max98090.h | 0 | ||||
-rw-r--r-- | sound/soc/codecs/si476x.c | 1 | ||||
-rw-r--r-- | sound/soc/codecs/wm_adsp.c | 5 | ||||
-rw-r--r-- | sound/soc/fsl/imx-ssi.c | 5 | ||||
-rw-r--r-- | sound/soc/fsl/pcm030-audio-fabric.c | 2 | ||||
-rw-r--r-- | sound/soc/sh/dma-sh7760.c | 4 | ||||
-rw-r--r-- | sound/soc/soc-core.c | 8 | ||||
-rw-r--r-- | sound/soc/soc-dapm.c | 14 | ||||
-rw-r--r-- | sound/soc/spear/spear_pcm.c | 12 |
10 files changed, 36 insertions, 15 deletions
diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index fc176044994d..fc176044994d 100755..100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c diff --git a/sound/soc/codecs/max98090.h b/sound/soc/codecs/max98090.h index 7e103f249053..7e103f249053 100755..100644 --- a/sound/soc/codecs/max98090.h +++ b/sound/soc/codecs/max98090.h diff --git a/sound/soc/codecs/si476x.c b/sound/soc/codecs/si476x.c index f2d61a187830..566ea3256e2d 100644 --- a/sound/soc/codecs/si476x.c +++ b/sound/soc/codecs/si476x.c @@ -159,6 +159,7 @@ static int si476x_codec_hw_params(struct snd_pcm_substream *substream, switch (params_format(params)) { case SNDRV_PCM_FORMAT_S8: width = SI476X_PCM_FORMAT_S8; + break; case SNDRV_PCM_FORMAT_S16_LE: width = SI476X_PCM_FORMAT_S16_LE; break; diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index f3f7e75f8628..9af1bddc4c62 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -828,7 +828,8 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp) &buf_list); if (!buf) { adsp_err(dsp, "Out of memory\n"); - return -ENOMEM; + ret = -ENOMEM; + goto out_fw; } adsp_dbg(dsp, "%s.%d: Writing %d bytes at %x\n", @@ -865,7 +866,7 @@ out_fw: wm_adsp_buf_free(&buf_list); out: kfree(file); - return 0; + return ret; } int wm_adsp1_init(struct wm_adsp *adsp) diff --git a/sound/soc/fsl/imx-ssi.c b/sound/soc/fsl/imx-ssi.c index 55464a5b0706..810c7eeb7b03 100644 --- a/sound/soc/fsl/imx-ssi.c +++ b/sound/soc/fsl/imx-ssi.c @@ -496,6 +496,8 @@ static void imx_ssi_ac97_reset(struct snd_ac97 *ac97) if (imx_ssi->ac97_reset) imx_ssi->ac97_reset(ac97); + /* First read sometimes fails, do a dummy read */ + imx_ssi_ac97_read(ac97, 0); } static void imx_ssi_ac97_warm_reset(struct snd_ac97 *ac97) @@ -504,6 +506,9 @@ static void imx_ssi_ac97_warm_reset(struct snd_ac97 *ac97) if (imx_ssi->ac97_warm_reset) imx_ssi->ac97_warm_reset(ac97); + + /* First read sometimes fails, do a dummy read */ + imx_ssi_ac97_read(ac97, 0); } struct snd_ac97_bus_ops soc_ac97_ops = { diff --git a/sound/soc/fsl/pcm030-audio-fabric.c b/sound/soc/fsl/pcm030-audio-fabric.c index 8e52c1485df3..eb4373840bb6 100644 --- a/sound/soc/fsl/pcm030-audio-fabric.c +++ b/sound/soc/fsl/pcm030-audio-fabric.c @@ -51,7 +51,7 @@ static struct snd_soc_card pcm030_card = { .num_links = ARRAY_SIZE(pcm030_fabric_dai), }; -static int __init pcm030_fabric_probe(struct platform_device *op) +static int pcm030_fabric_probe(struct platform_device *op) { struct device_node *np = op->dev.of_node; struct device_node *platform_np; diff --git a/sound/soc/sh/dma-sh7760.c b/sound/soc/sh/dma-sh7760.c index 19eff8fc4fdd..1a8b03e4b41b 100644 --- a/sound/soc/sh/dma-sh7760.c +++ b/sound/soc/sh/dma-sh7760.c @@ -342,8 +342,8 @@ static int camelot_pcm_new(struct snd_soc_pcm_runtime *rtd) return 0; } -static struct snd_soc_platform sh7760_soc_platform = { - .pcm_ops = &camelot_pcm_ops, +static struct snd_soc_platform_driver sh7760_soc_platform = { + .ops = &camelot_pcm_ops, .pcm_new = camelot_pcm_new, .pcm_free = camelot_pcm_free, }; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index b7e84a7cd9ee..507d251916af 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3140,7 +3140,7 @@ int snd_soc_bytes_put(struct snd_kcontrol *kcontrol, if (params->mask) { ret = regmap_read(codec->control_data, params->base, &val); if (ret != 0) - return ret; + goto out; val &= params->mask; @@ -3158,13 +3158,15 @@ int snd_soc_bytes_put(struct snd_kcontrol *kcontrol, ((u32 *)data)[0] |= cpu_to_be32(val); break; default: - return -EINVAL; + ret = -EINVAL; + goto out; } } ret = regmap_raw_write(codec->control_data, params->base, data, len); +out: kfree(data); return ret; @@ -4197,7 +4199,6 @@ int snd_soc_of_parse_audio_routing(struct snd_soc_card *card, dev_err(card->dev, "ASoC: Property '%s' index %d could not be read: %d\n", propname, 2 * i, ret); - kfree(routes); return -EINVAL; } ret = of_property_read_string_index(np, propname, @@ -4206,7 +4207,6 @@ int snd_soc_of_parse_audio_routing(struct snd_soc_card *card, dev_err(card->dev, "ASoC: Property '%s' index %d could not be read: %d\n", propname, (2 * i) + 1, ret); - kfree(routes); return -EINVAL; } } diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 1d6a9b3ceb27..d6d9ba2e6916 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -831,6 +831,9 @@ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget, if (path->weak) continue; + if (path->walking) + return 1; + if (path->walked) continue; @@ -838,6 +841,7 @@ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget, if (path->sink && path->connect) { path->walked = 1; + path->walking = 1; /* do we need to add this widget to the list ? */ if (list) { @@ -847,11 +851,14 @@ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget, dev_err(widget->dapm->dev, "ASoC: could not add widget %s\n", widget->name); + path->walking = 0; return con; } } con += is_connected_output_ep(path->sink, list); + + path->walking = 0; } } @@ -931,6 +938,9 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget, if (path->weak) continue; + if (path->walking) + return 1; + if (path->walked) continue; @@ -938,6 +948,7 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget, if (path->source && path->connect) { path->walked = 1; + path->walking = 1; /* do we need to add this widget to the list ? */ if (list) { @@ -947,11 +958,14 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget, dev_err(widget->dapm->dev, "ASoC: could not add widget %s\n", widget->name); + path->walking = 0; return con; } } con += is_connected_input_ep(path->source, list); + + path->walking = 0; } } diff --git a/sound/soc/spear/spear_pcm.c b/sound/soc/spear/spear_pcm.c index 9b76cc5a1148..5e7aebe1e664 100644 --- a/sound/soc/spear/spear_pcm.c +++ b/sound/soc/spear/spear_pcm.c @@ -149,9 +149,9 @@ static void spear_pcm_free(struct snd_pcm *pcm) static u64 spear_pcm_dmamask = DMA_BIT_MASK(32); -static int spear_pcm_new(struct snd_card *card, - struct snd_soc_dai *dai, struct snd_pcm *pcm) +static int spear_pcm_new(struct snd_soc_pcm_runtime *rtd) { + struct snd_card *card = rtd->card->snd_card; int ret; if (!card->dev->dma_mask) @@ -159,16 +159,16 @@ static int spear_pcm_new(struct snd_card *card, if (!card->dev->coherent_dma_mask) card->dev->coherent_dma_mask = DMA_BIT_MASK(32); - if (dai->driver->playback.channels_min) { - ret = spear_pcm_preallocate_dma_buffer(pcm, + if (rtd->cpu_dai->driver->playback.channels_min) { + ret = spear_pcm_preallocate_dma_buffer(rtd->pcm, SNDRV_PCM_STREAM_PLAYBACK, spear_pcm_hardware.buffer_bytes_max); if (ret) return ret; } - if (dai->driver->capture.channels_min) { - ret = spear_pcm_preallocate_dma_buffer(pcm, + if (rtd->cpu_dai->driver->capture.channels_min) { + ret = spear_pcm_preallocate_dma_buffer(rtd->pcm, SNDRV_PCM_STREAM_CAPTURE, spear_pcm_hardware.buffer_bytes_max); if (ret) |