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Sound fixes for 3.9-rc6

Browse Source 
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
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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
This commit is contained in:
Linus Torvalds 2013-04-05 09:06:57 -07:00
commit 8f09aacfa6
20 changed files with 83 additions and 37 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

5
Documentation/sound/alsa/ALSA-Configuration.txt
View File

@ -890,9 +890,8 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
enable_msi - Enable Message Signaled Interrupt (MSI) (default = off)
power_save - Automatic power-saving timeout (in second, 0 =
disable)
power_save_controller - Support runtime D3 of HD-audio controller
(-1 = on for supported chip (default), false = off,
true = force to on even for unsupported hardware)
power_save_controller - Reset HD-audio controller in power-saving mode
(default = on)
align_buffer_size - Force rounding of buffer/period sizes to multiples
of 128 bytes. This is more efficient in terms of memory
access but isn't required by the HDA spec and prevents

0
include/sound/max98090.h Executable file → Normal file
View File

1
include/sound/soc-dapm.h
View File

@ -488,6 +488,7 @@ struct snd_soc_dapm_path {
/* status */
u32 connect:1; /* source and sink widgets are connected */
u32 walked:1; /* path has been walked */
u32 walking:1; /* path is in the process of being walked */
u32 weak:1; /* path ignored for power management */
int (*connected)(struct snd_soc_dapm_widget *source,

2
sound/pci/hda/hda_codec.c
View File

@ -173,7 +173,7 @@ const char *snd_hda_get_jack_type(u32 cfg)
"Line Out", "Speaker", "HP Out", "CD",
"SPDIF Out", "Digital Out", "Modem Line", "Modem Hand",
"Line In", "Aux", "Mic", "Telephony",
"SPDIF In", "Digitial In", "Reserved", "Other"
"SPDIF In", "Digital In", "Reserved", "Other"
};
return jack_types[(cfg & AC_DEFCFG_DEVICE)

2
sound/pci/hda/hda_eld.c
View File

@ -320,7 +320,7 @@ int snd_hdmi_get_eld(struct hda_codec *codec, hda_nid_t nid,
unsigned char *buf, int *eld_size)
{
int i;
int ret;
int ret = 0;
int size;
/*

2
sound/pci/hda/hda_generic.c
View File

@ -740,7 +740,7 @@ EXPORT_SYMBOL_HDA(snd_hda_activate_path);
static void path_power_down_sync(struct hda_codec *codec, struct nid_path *path)
{
struct hda_gen_spec *spec = codec->spec;
bool changed;
bool changed = false;
int i;
if (!spec->power_down_unused || path->active)

6
sound/pci/hda/hda_intel.c
View File

@ -134,8 +134,8 @@ MODULE_PARM_DESC(power_save, "Automatic power-saving timeout "
* this may give more power-saving, but will take longer time to
* wake up.
*/
static int power_save_controller = -1;
module_param(power_save_controller, bint, 0644);
static bool power_save_controller = 1;
module_param(power_save_controller, bool, 0644);
MODULE_PARM_DESC(power_save_controller, "Reset controller in power save mode.");
#endif /* CONFIG_PM */
@ -2931,8 +2931,6 @@ static int azx_runtime_idle(struct device *dev)
struct snd_card *card = dev_get_drvdata(dev);
struct azx *chip = card->private_data;
if (power_save_controller > 0)
return 0;
if (!power_save_controller ||
!(chip->driver_caps & AZX_DCAPS_PM_RUNTIME))
return -EBUSY;

2
sound/pci/hda/patch_hdmi.c
View File

@ -1196,7 +1196,7 @@ static void hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll)
_snd_printd(SND_PR_VERBOSE,
"HDMI status: Codec=%d Pin=%d Presence_Detect=%d ELD_Valid=%d\n",
codec->addr, pin_nid, eld->monitor_present, eld->eld_valid);
codec->addr, pin_nid, pin_eld->monitor_present, eld->eld_valid);
if (eld->eld_valid) {
if (snd_hdmi_get_eld(codec, pin_nid, eld->eld_buffer,

4
sound/pci/hda/patch_realtek.c
View File

@ -3440,7 +3440,8 @@ static int alc662_parse_auto_config(struct hda_codec *codec)
const hda_nid_t *ssids;
if (codec->vendor_id == 0x10ec0272 || codec->vendor_id == 0x10ec0663 ||
codec->vendor_id == 0x10ec0665 || codec->vendor_id == 0x10ec0670)
codec->vendor_id == 0x10ec0665 || codec->vendor_id == 0x10ec0670 ||
codec->vendor_id == 0x10ec0671)
ssids = alc663_ssids;
else
ssids = alc662_ssids;
@ -3894,6 +3895,7 @@ static const struct hda_codec_preset snd_hda_preset_realtek[] = {
{ .id = 0x10ec0665, .name = "ALC665", .patch = patch_alc662 },
{ .id = 0x10ec0668, .name = "ALC668", .patch = patch_alc662 },
{ .id = 0x10ec0670, .name = "ALC670", .patch = patch_alc662 },
{ .id = 0x10ec0671, .name = "ALC671", .patch = patch_alc662 },
{ .id = 0x10ec0680, .name = "ALC680", .patch = patch_alc680 },
{ .id = 0x10ec0880, .name = "ALC880", .patch = patch_alc880 },
{ .id = 0x10ec0882, .name = "ALC882", .patch = patch_alc882 },

0
sound/soc/codecs/max98090.c Executable file → Normal file
View File

0
sound/soc/codecs/max98090.h Executable file → Normal file
View File

1
sound/soc/codecs/si476x.c
View File

@ -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;

5
sound/soc/codecs/wm_adsp.c
View File

@ -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)

5
sound/soc/fsl/imx-ssi.c
View File

@ -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 = {

2
sound/soc/fsl/pcm030-audio-fabric.c
View File

@ -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;

4
sound/soc/sh/dma-sh7760.c
View File

@ -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,
};

8
sound/soc/soc-core.c
View File

@ -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;
}
}

14
sound/soc/soc-dapm.c
View File

@ -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;
}
}

12
sound/soc/spear/spear_pcm.c
View File

@ -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)

39
sound/usb/clock.c
View File

@ -253,7 +253,7 @@ static int set_sample_rate_v2(struct snd_usb_audio *chip, int iface,
{
struct usb_device *dev = chip->dev;
unsigned char data[4];
int err, crate;
int err, cur_rate, prev_rate;
int clock = snd_usb_clock_find_source(chip, fmt->clock);
if (clock < 0)
@ -266,6 +266,19 @@ static int set_sample_rate_v2(struct snd_usb_audio *chip, int iface,
return -ENXIO;
}
err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC2_CS_CUR,
USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN,
UAC2_CS_CONTROL_SAM_FREQ << 8,
snd_usb_ctrl_intf(chip) | (clock << 8),
data, sizeof(data));
if (err < 0) {
snd_printk(KERN_WARNING "%d:%d:%d: cannot get freq (v2)\n",
dev->devnum, iface, fmt->altsetting);
prev_rate = 0;
} else {
prev_rate = data[0] | (data[1] << 8) | (data[2] << 16) | (data[3] << 24);
}
data[0] = rate;
data[1] = rate >> 8;
data[2] = rate >> 16;
@ -280,19 +293,31 @@ static int set_sample_rate_v2(struct snd_usb_audio *chip, int iface,
return err;
}
if ((err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC2_CS_CUR,
err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC2_CS_CUR,
USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN,
UAC2_CS_CONTROL_SAM_FREQ << 8,
snd_usb_ctrl_intf(chip) | (clock << 8),
data, sizeof(data))) < 0) {
data, sizeof(data));
if (err < 0) {
snd_printk(KERN_WARNING "%d:%d:%d: cannot get freq (v2)\n",
dev->devnum, iface, fmt->altsetting);
return err;
cur_rate = 0;
} else {
cur_rate = data[0] | (data[1] << 8) | (data[2] << 16) | (data[3] << 24);
}
crate = data[0] | (data[1] << 8) | (data[2] << 16) | (data[3] << 24);
if (crate != rate)
snd_printd(KERN_WARNING "current rate %d is different from the runtime rate %d\n", crate, rate);
if (cur_rate != rate) {
snd_printd(KERN_WARNING
"current rate %d is different from the runtime rate %d\n",
cur_rate, rate);
}
/* Some devices doesn't respond to sample rate changes while the
* interface is active. */
if (rate != prev_rate) {
usb_set_interface(dev, iface, 0);
usb_set_interface(dev, iface, fmt->altsetting);
}
return 0;
}