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-rw-r--r--sound/core/init.c2
-rw-r--r--sound/core/pcm_native.c2
-rw-r--r--sound/isa/opl3sa2.c2
-rw-r--r--sound/pci/ca0106/ca0106.h2
-rw-r--r--sound/pci/ca0106/ca0106_main.c2
-rw-r--r--sound/pci/cs5535audio/cs5535audio_pm.c7
-rw-r--r--sound/pci/emu10k1/emu10k1x.c2
-rw-r--r--sound/pci/emu10k1/p16v.h2
-rw-r--r--sound/pci/es1968.c2
-rw-r--r--sound/pci/hda/patch_realtek.c26
-rw-r--r--sound/pci/rme9652/hdspm.c6
-rw-r--r--sound/ppc/snd_ps3.c2
-rw-r--r--sound/soc/codecs/max98088.c2
-rw-r--r--sound/soc/pxa/pxa-ssp.c2
-rw-r--r--sound/soc/samsung/smdk_spdif.c6
15 files changed, 35 insertions, 32 deletions
diff --git a/sound/core/init.c b/sound/core/init.c
index 57b792e2439a..3e65da21a08c 100644
--- a/sound/core/init.c
+++ b/sound/core/init.c
@@ -642,7 +642,7 @@ static struct device_attribute card_number_attrs =
* external accesses. Thus, you should call this function at the end
* of the initialization of the card.
*
- * Returns zero otherwise a negative error code if the registrain failed.
+ * Returns zero otherwise a negative error code if the registration failed.
*/
int snd_card_register(struct snd_card *card)
{
diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c
index 0db714e87a80..4be45e7be8ad 100644
--- a/sound/core/pcm_native.c
+++ b/sound/core/pcm_native.c
@@ -987,7 +987,7 @@ static int snd_pcm_do_pause(struct snd_pcm_substream *substream, int push)
if (push)
snd_pcm_update_hw_ptr(substream);
/* The jiffies check in snd_pcm_update_hw_ptr*() is done by
- * a delta betwen the current jiffies, this gives a large enough
+ * a delta between the current jiffies, this gives a large enough
* delta, effectively to skip the check once.
*/
substream->runtime->hw_ptr_jiffies = jiffies - HZ * 1000;
diff --git a/sound/isa/opl3sa2.c b/sound/isa/opl3sa2.c
index 265abcce9dba..9b915e27b5bd 100644
--- a/sound/isa/opl3sa2.c
+++ b/sound/isa/opl3sa2.c
@@ -264,7 +264,7 @@ static int __devinit snd_opl3sa2_detect(struct snd_card *card)
snd_printd("OPL3-SA [0x%lx] detect (1) = 0x%x (0x%x)\n", port, tmp, tmp1);
return -ENODEV;
}
- /* try if the MIC register is accesible */
+ /* try if the MIC register is accessible */
tmp = snd_opl3sa2_read(chip, OPL3SA2_MIC);
snd_opl3sa2_write(chip, OPL3SA2_MIC, 0x8a);
if (((tmp1 = snd_opl3sa2_read(chip, OPL3SA2_MIC)) & 0x9f) != 0x8a) {
diff --git a/sound/pci/ca0106/ca0106.h b/sound/pci/ca0106/ca0106.h
index f19c11077255..fc53b9bca26d 100644
--- a/sound/pci/ca0106/ca0106.h
+++ b/sound/pci/ca0106/ca0106.h
@@ -188,7 +188,7 @@
#define PLAYBACK_LIST_PTR 0x02 /* Pointer to the current period being played */
/* PTR[5:0], Default: 0x0 */
#define PLAYBACK_UNKNOWN3 0x03 /* Not used ?? */
-#define PLAYBACK_DMA_ADDR 0x04 /* Playback DMA addresss */
+#define PLAYBACK_DMA_ADDR 0x04 /* Playback DMA address */
/* DMA[31:0], Default: 0x0 */
#define PLAYBACK_PERIOD_SIZE 0x05 /* Playback period size. win2000 uses 0x04000000 */
/* SIZE[31:16], Default: 0x0 */
diff --git a/sound/pci/ca0106/ca0106_main.c b/sound/pci/ca0106/ca0106_main.c
index d2d12c08f937..01b49388fafd 100644
--- a/sound/pci/ca0106/ca0106_main.c
+++ b/sound/pci/ca0106/ca0106_main.c
@@ -1082,7 +1082,7 @@ snd_ca0106_pcm_pointer_capture(struct snd_pcm_substream *substream)
struct snd_pcm_runtime *runtime = substream->runtime;
struct snd_ca0106_pcm *epcm = runtime->private_data;
snd_pcm_uframes_t ptr, ptr1, ptr2 = 0;
- int channel = channel=epcm->channel_id;
+ int channel = epcm->channel_id;
if (!epcm->running)
return 0;
diff --git a/sound/pci/cs5535audio/cs5535audio_pm.c b/sound/pci/cs5535audio/cs5535audio_pm.c
index a3301cc4ab82..185b00088320 100644
--- a/sound/pci/cs5535audio/cs5535audio_pm.c
+++ b/sound/pci/cs5535audio/cs5535audio_pm.c
@@ -90,12 +90,7 @@ int snd_cs5535audio_resume(struct pci_dev *pci)
int i;
pci_set_power_state(pci, PCI_D0);
- if (pci_restore_state(pci) < 0) {
- printk(KERN_ERR "cs5535audio: pci_restore_state failed, "
- "disabling device\n");
- snd_card_disconnect(card);
- return -EIO;
- }
+ pci_restore_state(pci);
if (pci_enable_device(pci) < 0) {
printk(KERN_ERR "cs5535audio: pci_enable_device failed, "
"disabling device\n");
diff --git a/sound/pci/emu10k1/emu10k1x.c b/sound/pci/emu10k1/emu10k1x.c
index df47f738098d..0c701e4ec8a5 100644
--- a/sound/pci/emu10k1/emu10k1x.c
+++ b/sound/pci/emu10k1/emu10k1x.c
@@ -114,7 +114,7 @@ MODULE_PARM_DESC(enable, "Enable the EMU10K1X soundcard.");
*/
#define PLAYBACK_LIST_SIZE 0x01 /* Size of list in bytes << 16. E.g. 8 periods -> 0x00380000 */
#define PLAYBACK_LIST_PTR 0x02 /* Pointer to the current period being played */
-#define PLAYBACK_DMA_ADDR 0x04 /* Playback DMA addresss */
+#define PLAYBACK_DMA_ADDR 0x04 /* Playback DMA address */
#define PLAYBACK_PERIOD_SIZE 0x05 /* Playback period size */
#define PLAYBACK_POINTER 0x06 /* Playback period pointer. Sample currently in DAC */
#define PLAYBACK_UNKNOWN1 0x07
diff --git a/sound/pci/emu10k1/p16v.h b/sound/pci/emu10k1/p16v.h
index 153214940336..00f4817533b1 100644
--- a/sound/pci/emu10k1/p16v.h
+++ b/sound/pci/emu10k1/p16v.h
@@ -96,7 +96,7 @@
#define PLAYBACK_LIST_SIZE 0x01 /* Size of list in bytes << 16. E.g. 8 periods -> 0x00380000 */
#define PLAYBACK_LIST_PTR 0x02 /* Pointer to the current period being played */
#define PLAYBACK_UNKNOWN3 0x03 /* Not used */
-#define PLAYBACK_DMA_ADDR 0x04 /* Playback DMA addresss */
+#define PLAYBACK_DMA_ADDR 0x04 /* Playback DMA address */
#define PLAYBACK_PERIOD_SIZE 0x05 /* Playback period size. win2000 uses 0x04000000 */
#define PLAYBACK_POINTER 0x06 /* Playback period pointer. Used with PLAYBACK_LIST_PTR to determine buffer position currently in DAC */
#define PLAYBACK_FIFO_END_ADDRESS 0x07 /* Playback FIFO end address */
diff --git a/sound/pci/es1968.c b/sound/pci/es1968.c
index 23a58f0d6cb9..7c17f45d876d 100644
--- a/sound/pci/es1968.c
+++ b/sound/pci/es1968.c
@@ -220,7 +220,7 @@ MODULE_PARM_DESC(joystick, "Enable joystick.");
#define RINGB_EN_2CODEC 0x0020
#define RINGB_SING_BIT_DUAL 0x0040
-/* ****Port Adresses**** */
+/* ****Port Addresses**** */
/* Write & Read */
#define ESM_INDEX 0x02
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 269dbff70b92..be4df4c6fd56 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -1721,7 +1721,9 @@ static void alc_apply_fixup(struct hda_codec *codec, int action)
{
struct alc_spec *spec = codec->spec;
int id = spec->fixup_id;
+#ifdef CONFIG_SND_DEBUG_VERBOSE
const char *modelname = spec->fixup_name;
+#endif
int depth = 0;
if (!spec->fixup_list)
@@ -10930,9 +10932,6 @@ static int alc_auto_add_mic_boost(struct hda_codec *codec)
return 0;
}
-static int alc861vd_auto_create_multi_out_ctls(struct alc_spec *spec,
- const struct auto_pin_cfg *cfg);
-
/* almost identical with ALC880 parser... */
static int alc882_parse_auto_config(struct hda_codec *codec)
{
@@ -10950,10 +10949,7 @@ static int alc882_parse_auto_config(struct hda_codec *codec)
err = alc880_auto_fill_dac_nids(spec, &spec->autocfg);
if (err < 0)
return err;
- if (codec->vendor_id == 0x10ec0887)
- err = alc861vd_auto_create_multi_out_ctls(spec, &spec->autocfg);
- else
- err = alc880_auto_create_multi_out_ctls(spec, &spec->autocfg);
+ err = alc880_auto_create_multi_out_ctls(spec, &spec->autocfg);
if (err < 0)
return err;
err = alc880_auto_create_extra_out(spec, spec->autocfg.hp_pins[0],
@@ -12635,6 +12631,8 @@ static struct snd_pci_quirk alc262_cfg_tbl[] = {
ALC262_HP_BPC),
SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x1300, "HP xw series",
ALC262_HP_BPC),
+ SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x1500, "HP z series",
+ ALC262_HP_BPC),
SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x1700, "HP xw series",
ALC262_HP_BPC),
SND_PCI_QUIRK(0x103c, 0x2800, "HP D7000", ALC262_HP_BPC_D7000_WL),
@@ -14957,6 +14955,7 @@ static struct snd_pci_quirk alc269_fixup_tbl[] = {
SND_PCI_QUIRK_VENDOR(0x104d, "Sony VAIO", ALC269_FIXUP_SONY_VAIO),
SND_PCI_QUIRK(0x1028, 0x0470, "Dell M101z", ALC269_FIXUP_DELL_M101Z),
SND_PCI_QUIRK(0x17aa, 0x21b8, "Thinkpad Edge 14", ALC269_FIXUP_SKU_IGNORE),
+ SND_PCI_QUIRK(0x17aa, 0x21e9, "Thinkpad Edge 14", ALC269_FIXUP_SKU_IGNORE),
SND_PCI_QUIRK(0x17aa, 0x20f2, "Thinkpad SL410/510", ALC269_FIXUP_SKU_IGNORE),
SND_PCI_QUIRK(0x1043, 0x1a13, "Asus G73Jw", ALC269_FIXUP_ASUS_G73JW),
SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD),
@@ -17134,7 +17133,7 @@ static void alc861vd_auto_init_analog_input(struct hda_codec *codec)
#define alc861vd_idx_to_mixer_switch(nid) ((nid) + 0x0c)
/* add playback controls from the parsed DAC table */
-/* Based on ALC880 version. But ALC861VD and ALC887 have separate,
+/* Based on ALC880 version. But ALC861VD has separate,
* different NIDs for mute/unmute switch and volume control */
static int alc861vd_auto_create_multi_out_ctls(struct alc_spec *spec,
const struct auto_pin_cfg *cfg)
@@ -19461,6 +19460,7 @@ enum {
ALC662_FIXUP_ASPIRE,
ALC662_FIXUP_IDEAPAD,
ALC272_FIXUP_MARIO,
+ ALC662_FIXUP_CZC_P10T,
};
static const struct alc_fixup alc662_fixups[] = {
@@ -19481,7 +19481,14 @@ static const struct alc_fixup alc662_fixups[] = {
[ALC272_FIXUP_MARIO] = {
.type = ALC_FIXUP_FUNC,
.v.func = alc272_fixup_mario,
- }
+ },
+ [ALC662_FIXUP_CZC_P10T] = {
+ .type = ALC_FIXUP_VERBS,
+ .v.verbs = (const struct hda_verb[]) {
+ {0x14, AC_VERB_SET_EAPD_BTLENABLE, 0},
+ {}
+ }
+ },
};
static struct snd_pci_quirk alc662_fixup_tbl[] = {
@@ -19489,6 +19496,7 @@ static struct snd_pci_quirk alc662_fixup_tbl[] = {
SND_PCI_QUIRK(0x144d, 0xc051, "Samsung R720", ALC662_FIXUP_IDEAPAD),
SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo Ideapad Y550P", ALC662_FIXUP_IDEAPAD),
SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo Ideapad Y550", ALC662_FIXUP_IDEAPAD),
+ SND_PCI_QUIRK(0x1b35, 0x2206, "CZC P10T", ALC662_FIXUP_CZC_P10T),
{}
};
diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c
index 0c98ef9156d8..f5eadfc0672a 100644
--- a/sound/pci/rme9652/hdspm.c
+++ b/sound/pci/rme9652/hdspm.c
@@ -487,7 +487,7 @@ struct hdspm {
struct snd_kcontrol *playback_mixer_ctls[HDSPM_MAX_CHANNELS];
/* but input to much, so not used */
struct snd_kcontrol *input_mixer_ctls[HDSPM_MAX_CHANNELS];
- /* full mixer accessable over mixer ioctl or hwdep-device */
+ /* full mixer accessible over mixer ioctl or hwdep-device */
struct hdspm_mixer *mixer;
};
@@ -550,7 +550,7 @@ static inline int HDSPM_bit2freq(int n)
return bit2freq_tab[n];
}
-/* Write/read to/from HDSPM with Adresses in Bytes
+/* Write/read to/from HDSPM with Addresses in Bytes
not words but only 32Bit writes are allowed */
static inline void hdspm_write(struct hdspm * hdspm, unsigned int reg,
@@ -2908,7 +2908,7 @@ static int snd_hdspm_create_controls(struct snd_card *card, struct hdspm * hdspm
/* Channel playback mixer as default control
Note: the whole matrix would be 128*HDSPM_MIXER_CHANNELS Faders,
- thats too * big for any alsamixer they are accesible via special
+ thats too * big for any alsamixer they are accessible via special
IOCTL on hwdep and the mixer 2dimensional mixer control
*/
diff --git a/sound/ppc/snd_ps3.c b/sound/ppc/snd_ps3.c
index 581a670e8261..edce8a27e3ee 100644
--- a/sound/ppc/snd_ps3.c
+++ b/sound/ppc/snd_ps3.c
@@ -51,7 +51,7 @@ static struct snd_ps3_card_info the_card;
static int snd_ps3_start_delay = CONFIG_SND_PS3_DEFAULT_START_DELAY;
module_param_named(start_delay, snd_ps3_start_delay, uint, 0644);
-MODULE_PARM_DESC(start_delay, "time to insert silent data in milisec");
+MODULE_PARM_DESC(start_delay, "time to insert silent data in ms");
static int index = SNDRV_DEFAULT_IDX1;
static char *id = SNDRV_DEFAULT_STR1;
diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c
index 37133c40e762..89498f9ad2e5 100644
--- a/sound/soc/codecs/max98088.c
+++ b/sound/soc/codecs/max98088.c
@@ -1958,7 +1958,7 @@ static int max98088_probe(struct snd_soc_codec *codec)
return ret;
}
- /* initalize private data */
+ /* initialize private data */
max98088->sysclk = (unsigned)-1;
max98088->eq_textcnt = 0;
diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c
index b439eee462cb..8ad93ee2e92b 100644
--- a/sound/soc/pxa/pxa-ssp.c
+++ b/sound/soc/pxa/pxa-ssp.c
@@ -20,6 +20,7 @@
#include <linux/platform_device.h>
#include <linux/clk.h>
#include <linux/io.h>
+#include <linux/pxa2xx_ssp.h>
#include <asm/irq.h>
@@ -33,7 +34,6 @@
#include <mach/hardware.h>
#include <mach/dma.h>
#include <mach/audio.h>
-#include <plat/ssp.h>
#include "../../arm/pxa2xx-pcm.h"
#include "pxa-ssp.h"
diff --git a/sound/soc/samsung/smdk_spdif.c b/sound/soc/samsung/smdk_spdif.c
index cb2f4d04ab7e..b5c3fad01bb8 100644
--- a/sound/soc/samsung/smdk_spdif.c
+++ b/sound/soc/samsung/smdk_spdif.c
@@ -61,7 +61,7 @@ static int set_audio_clock_heirachy(struct platform_device *pdev)
goto out3;
}
- /* Set audio clock heirachy for S/PDIF */
+ /* Set audio clock hierarchy for S/PDIF */
clk_set_parent(mout_epll, fout_epll);
clk_set_parent(sclk_audio0, mout_epll);
clk_set_parent(sclk_spdif, sclk_audio0);
@@ -79,7 +79,7 @@ out1:
/* We should haved to set clock directly on this part because of clock
* scheme of Samsudng SoCs did not support to set rates from abstrct
- * clock of it's heirachy.
+ * clock of it's hierarchy.
*/
static int set_audio_clock_rate(unsigned long epll_rate,
unsigned long audio_rate)
@@ -195,7 +195,7 @@ static int __init smdk_init(void)
if (ret)
goto err3;
- /* Set audio clock heirachy manually */
+ /* Set audio clock hierarchy manually */
ret = set_audio_clock_heirachy(smdk_snd_spdif_device);
if (ret)
goto err4;
generated by cgit v1.2.3 (git 2.39.1) at 2024-11-25 02:40:02 +0000