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authorLinus Torvalds <torvalds@linux-foundation.org>2012-12-13 23:51:23 +0400
committerLinus Torvalds <torvalds@linux-foundation.org>2012-12-13 23:51:23 +0400
commit046e7d685bc370fd4c879ab6635ad3f69e6673d1 (patch)
tree36b981f8d1f2bfd348c1479acbe3a9426d35c377 /sound/pci/ca0106/ca0106_mixer.c
parentfe504c5c745aeb767d978fbedeb94775fd4cb69c (diff)
parent6eb827d23577a4efec2b10a9c4cc9ded268a1d1c (diff)
downloadlinux-046e7d685bc370fd4c879ab6635ad3f69e6673d1.tar.xz
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/pci/ca0106/ca0106_mixer.c')
-rw-r--r--sound/pci/ca0106/ca0106_mixer.c26
1 files changed, 13 insertions, 13 deletions
diff --git a/sound/pci/ca0106/ca0106_mixer.c b/sound/pci/ca0106/ca0106_mixer.c
index 68eacf7002d6..27de0de90018 100644
--- a/sound/pci/ca0106/ca0106_mixer.c
+++ b/sound/pci/ca0106/ca0106_mixer.c
@@ -325,7 +325,7 @@ static int snd_ca0106_capture_mic_line_in_put(struct snd_kcontrol *kcontrol,
return change;
}
-static struct snd_kcontrol_new snd_ca0106_capture_mic_line_in __devinitdata =
+static struct snd_kcontrol_new snd_ca0106_capture_mic_line_in =
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.name = "Shared Mic/Line in Capture Switch",
@@ -334,7 +334,7 @@ static struct snd_kcontrol_new snd_ca0106_capture_mic_line_in __devinitdata =
.put = snd_ca0106_capture_mic_line_in_put
};
-static struct snd_kcontrol_new snd_ca0106_capture_line_in_side_out __devinitdata =
+static struct snd_kcontrol_new snd_ca0106_capture_line_in_side_out =
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.name = "Shared Line in/Side out Capture Switch",
@@ -588,7 +588,7 @@ static int spi_mute_put(struct snd_kcontrol *kcontrol,
.private_value = ((chid) << 8) | (reg) \
}
-static struct snd_kcontrol_new snd_ca0106_volume_ctls[] __devinitdata = {
+static struct snd_kcontrol_new snd_ca0106_volume_ctls[] = {
CA_VOLUME("Analog Front Playback Volume",
CONTROL_FRONT_CHANNEL, PLAYBACK_VOLUME2),
CA_VOLUME("Analog Rear Playback Volume",
@@ -669,7 +669,7 @@ static struct snd_kcontrol_new snd_ca0106_volume_ctls[] __devinitdata = {
.private_value = chid \
}
-static struct snd_kcontrol_new snd_ca0106_volume_i2c_adc_ctls[] __devinitdata = {
+static struct snd_kcontrol_new snd_ca0106_volume_i2c_adc_ctls[] = {
I2C_VOLUME("Phone Capture Volume", 0),
I2C_VOLUME("Mic Capture Volume", 1),
I2C_VOLUME("Line in Capture Volume", 2),
@@ -691,7 +691,7 @@ static const int spi_dmute_bit[] = {
SPI_DMUTE4_BIT,
};
-static struct snd_kcontrol_new __devinit
+static struct snd_kcontrol_new
snd_ca0106_volume_spi_dac_ctl(struct snd_ca0106_details *details,
int channel_id)
{
@@ -735,7 +735,7 @@ snd_ca0106_volume_spi_dac_ctl(struct snd_ca0106_details *details,
return spi_switch;
}
-static int __devinit remove_ctl(struct snd_card *card, const char *name)
+static int remove_ctl(struct snd_card *card, const char *name)
{
struct snd_ctl_elem_id id;
memset(&id, 0, sizeof(id));
@@ -744,7 +744,7 @@ static int __devinit remove_ctl(struct snd_card *card, const char *name)
return snd_ctl_remove_id(card, &id);
}
-static struct snd_kcontrol __devinit *ctl_find(struct snd_card *card, const char *name)
+static struct snd_kcontrol *ctl_find(struct snd_card *card, const char *name)
{
struct snd_ctl_elem_id sid;
memset(&sid, 0, sizeof(sid));
@@ -754,7 +754,7 @@ static struct snd_kcontrol __devinit *ctl_find(struct snd_card *card, const char
return snd_ctl_find_id(card, &sid);
}
-static int __devinit rename_ctl(struct snd_card *card, const char *src, const char *dst)
+static int rename_ctl(struct snd_card *card, const char *src, const char *dst)
{
struct snd_kcontrol *kctl = ctl_find(card, src);
if (kctl) {
@@ -774,10 +774,10 @@ static int __devinit rename_ctl(struct snd_card *card, const char *src, const ch
} \
} while (0)
-static __devinitdata
+static
DECLARE_TLV_DB_SCALE(snd_ca0106_master_db_scale, -6375, 25, 1);
-static char *slave_vols[] __devinitdata = {
+static char *slave_vols[] = {
"Analog Front Playback Volume",
"Analog Rear Playback Volume",
"Analog Center/LFE Playback Volume",
@@ -790,7 +790,7 @@ static char *slave_vols[] __devinitdata = {
NULL
};
-static char *slave_sws[] __devinitdata = {
+static char *slave_sws[] = {
"Analog Front Playback Switch",
"Analog Rear Playback Switch",
"Analog Center/LFE Playback Switch",
@@ -799,7 +799,7 @@ static char *slave_sws[] __devinitdata = {
NULL
};
-static void __devinit add_slaves(struct snd_card *card,
+static void add_slaves(struct snd_card *card,
struct snd_kcontrol *master, char **list)
{
for (; *list; list++) {
@@ -809,7 +809,7 @@ static void __devinit add_slaves(struct snd_card *card,
}
}
-int __devinit snd_ca0106_mixer(struct snd_ca0106 *emu)
+int snd_ca0106_mixer(struct snd_ca0106 *emu)
{
int err;
struct snd_card *card = emu->card;