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authorLinus Torvalds <torvalds@linux-foundation.org>2012-12-13 11:51:23 -0800
committerLinus Torvalds <torvalds@linux-foundation.org>2012-12-13 11:51:23 -0800
commit046e7d685bc370fd4c879ab6635ad3f69e6673d1 (patch)
tree36b981f8d1f2bfd348c1479acbe3a9426d35c377 /sound/usb/format.c
parentMerge tag 'boards2' of git://git.kernel.org/pub/scm/linux/kernel/git/arm/arm-soc (diff)
parentALSA: hda - Move runtime PM check to runtime_idle callback (diff)
downloadlinux-dev-046e7d685bc370fd4c879ab6635ad3f69e6673d1.tar.xz
linux-dev-046e7d685bc370fd4c879ab6635ad3f69e6673d1.zip
Merge tag 'sound-3.8' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound updates from Takashi Iwai: "This update contains a fairly wide range of changes all over in sound subdirectory, mainly because of UAPI header moves by David and __dev* annotation removals by Bill. Other highlights are: - Introduced the support for wallclock timestamps in ALSA PCM core - Add the poll loop implementation for HD-audio jack detection - Yet more VGA-switcheroo fixes for HD-audio - New VIA HD-audio codec support - More fixes on resource management in USB audio and MIDI drivers - More quirks for USB-audio ASUS Xonar U3, Reloop Play, Focusrite, Roland VG-99, etc - Add support for FastTrack C400 usb-audio - Clean ups in many drivers regarding firmware loading - Add PSC724 Ultiimate Edge support to ice1712 - A few hdspm driver updates - New Stanton SCS.1d/1m FireWire driver - Standardisation of the logging in ASoC codes - DT and dmaengine support for ASoC Atmel - Support for Wolfson ADSP cores - New drivers for Freescale/iVeia P1022 and Maxim MAX98090 - Lots of other ASoC driver fixes and developments" Fix up trivial conflicts. And go out on a limb and assume the dts file 'status' field of one of the conflicting things was supposed to be "disabled", not "disable" like in pretty much all other cases. * tag 'sound-3.8' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (341 commits) ALSA: hda - Move runtime PM check to runtime_idle callback ALSA: hda - Add stereo-dmic fixup for Acer Aspire One 522 ALSA: hda - Avoid doubly suspend after vga switcheroo ALSA: usb-audio: Enable S/PDIF on the ASUS Xonar U3 ALSA: hda - Check validity of CORB/RIRB WP reads ALSA: hda - use usleep_range in link reset and change timeout check ALSA: HDA: VIA: Add support for codec VT1808. ALSA: HDA: VIA Add support for codec VT1705CF. ASoC: codecs: remove __dev* attributes ASoC: utils: remove __dev* attributes ASoC: ux500: remove __dev* attributes ASoC: txx9: remove __dev* attributes ASoC: tegra: remove __dev* attributes ASoC: spear: remove __dev* attributes ASoC: sh: remove __dev* attributes ASoC: s6000: remove __dev* attributes ASoC: OMAP: remove __dev* attributes ASoC: nuc900: remove __dev* attributes ASoC: mxs: remove __dev* attributes ASoC: kirkwood: remove __dev* attributes ...
Diffstat (limited to 'sound/usb/format.c')
-rw-r--r--sound/usb/format.c10
1 files changed, 5 insertions, 5 deletions
diff --git a/sound/usb/format.c b/sound/usb/format.c
index ddfef57c4c9f..e831ee4238bb 100644
--- a/sound/usb/format.c
+++ b/sound/usb/format.c
@@ -155,7 +155,7 @@ static int parse_audio_format_rates_v1(struct snd_usb_audio *chip, struct audiof
if (fmt[0] < offset + 1 + 3 * (nr_rates ? nr_rates : 2)) {
snd_printk(KERN_ERR "%d:%u:%d : invalid UAC_FORMAT_TYPE desc\n",
chip->dev->devnum, fp->iface, fp->altsetting);
- return -1;
+ return -EINVAL;
}
if (nr_rates) {
@@ -167,7 +167,7 @@ static int parse_audio_format_rates_v1(struct snd_usb_audio *chip, struct audiof
fp->rate_table = kmalloc(sizeof(int) * nr_rates, GFP_KERNEL);
if (fp->rate_table == NULL) {
snd_printk(KERN_ERR "cannot malloc\n");
- return -1;
+ return -ENOMEM;
}
fp->nr_rates = 0;
@@ -198,7 +198,7 @@ static int parse_audio_format_rates_v1(struct snd_usb_audio *chip, struct audiof
}
if (!fp->nr_rates) {
hwc_debug("All rates were zero. Skipping format!\n");
- return -1;
+ return -EINVAL;
}
} else {
/* continuous rates */
@@ -383,7 +383,7 @@ static int parse_audio_format_i(struct snd_usb_audio *chip,
fp->formats = parse_audio_format_i_type(chip, fp, format,
fmt, protocol);
if (!fp->formats)
- return -1;
+ return -EINVAL;
}
/* gather possible sample rates */
@@ -409,7 +409,7 @@ static int parse_audio_format_i(struct snd_usb_audio *chip,
if (fp->channels < 1) {
snd_printk(KERN_ERR "%d:%u:%d : invalid channels %d\n",
chip->dev->devnum, fp->iface, fp->altsetting, fp->channels);
- return -1;
+ return -EINVAL;
}
return ret;