aboutsummaryrefslogtreecommitdiff
path: root/sound/pci
diff options
context:
space:
mode:
authorTakashi Iwai <tiwai@suse.de>2009-12-21 11:21:15 +0100
committerTakashi Iwai <tiwai@suse.de>2009-12-21 11:21:15 +0100
commitde8853bc38ceab1fa7e7f723b21430d4aad60fea (patch)
tree5084ef51866fd1767324f8dc8eb36e97c55350f5 /sound/pci
parentf5de24b06aa46427500d0fdbe8616b73a71d8c28 (diff)
parent440b004cf953bec2bc8cd91c64ae707fd7e25327 (diff)
Merge remote branch 'alsa/fixes' into fix/hda
Diffstat (limited to 'sound/pci')
-rw-r--r--sound/pci/Kconfig1
-rw-r--r--sound/pci/ac97/ac97_codec.c2
-rw-r--r--sound/pci/ca0106/ca0106_proc.c2
-rw-r--r--sound/pci/cs46xx/imgs/cwcdma.asp9
-rw-r--r--sound/pci/emu10k1/emu10k1x.c2
-rw-r--r--sound/pci/fm801.c40
-rw-r--r--sound/pci/hda/patch_cirrus.c2
-rw-r--r--sound/pci/hda/patch_cmedia.c2
-rw-r--r--sound/pci/hda/patch_realtek.c4
-rw-r--r--sound/pci/ice1712/aureon.c31
-rw-r--r--sound/pci/ice1712/ice1712.h4
-rw-r--r--sound/pci/ice1712/juli.c34
-rw-r--r--sound/pci/ice1712/prodigy_hifi.c2
-rw-r--r--sound/pci/intel8x0.c12
-rw-r--r--sound/pci/rme9652/hdspm.c4
15 files changed, 96 insertions, 55 deletions
diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig
index 75c602b5b13..351654cf7b0 100644
--- a/sound/pci/Kconfig
+++ b/sound/pci/Kconfig
@@ -570,6 +570,7 @@ config SND_ICE1712
tristate "ICEnsemble ICE1712 (Envy24)"
select SND_MPU401_UART
select SND_AC97_CODEC
+ select BITREVERSE
help
Say Y here to include support for soundcards based on the
ICE1712 (Envy24) chip.
diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c
index 20cb60afb20..c1192062300 100644
--- a/sound/pci/ac97/ac97_codec.c
+++ b/sound/pci/ac97/ac97_codec.c
@@ -2122,7 +2122,7 @@ int snd_ac97_mixer(struct snd_ac97_bus *bus, struct snd_ac97_template *template,
}
/* nothing should be in powerdown mode */
snd_ac97_write_cache(ac97, AC97_GENERAL_PURPOSE, 0);
- end_time = jiffies + msecs_to_jiffies(120);
+ end_time = jiffies + msecs_to_jiffies(5000);
do {
if ((snd_ac97_read(ac97, AC97_POWERDOWN) & 0x0f) == 0x0f)
goto __ready_ok;
diff --git a/sound/pci/ca0106/ca0106_proc.c b/sound/pci/ca0106/ca0106_proc.c
index 15523e60351..0470461cc03 100644
--- a/sound/pci/ca0106/ca0106_proc.c
+++ b/sound/pci/ca0106/ca0106_proc.c
@@ -233,7 +233,7 @@ static void snd_ca0106_proc_dump_iec958( struct snd_info_buffer *buffer, u32 val
snd_iprintf(buffer, "user-defined\n");
break;
default:
- snd_iprintf(buffer, "unkown\n");
+ snd_iprintf(buffer, "unknown\n");
break;
}
snd_iprintf(buffer, "Sample Bits: ");
diff --git a/sound/pci/cs46xx/imgs/cwcdma.asp b/sound/pci/cs46xx/imgs/cwcdma.asp
index 09d24c76f03..a65e1193c89 100644
--- a/sound/pci/cs46xx/imgs/cwcdma.asp
+++ b/sound/pci/cs46xx/imgs/cwcdma.asp
@@ -26,10 +26,11 @@
//
//
// The purpose of this code is very simple: make it possible to tranfser
-// the samples 'as they are' with no alteration from a PCMreader SCB (DMA from host)
-// to any other SCB. This is useful for AC3 throug SPDIF. SRC (source rate converters)
-// task always alters the samples in some how, however it's from 48khz -> 48khz. The
-// alterations are not audible, but AC3 wont work.
+// the samples 'as they are' with no alteration from a PCMreader
+// SCB (DMA from host) to any other SCB. This is useful for AC3 through SPDIF.
+// SRC (source rate converters) task always alters the samples in somehow,
+// however it's from 48khz -> 48khz.
+// The alterations are not audible, but AC3 wont work.
//
// ...
// |
diff --git a/sound/pci/emu10k1/emu10k1x.c b/sound/pci/emu10k1/emu10k1x.c
index 6b8ae7b5cd5..1d369ff7380 100644
--- a/sound/pci/emu10k1/emu10k1x.c
+++ b/sound/pci/emu10k1/emu10k1x.c
@@ -184,7 +184,7 @@ MODULE_PARM_DESC(enable, "Enable the EMU10K1X soundcard.");
* The hardware has 3 channels for playback and 1 for capture.
* - channel 0 is the front channel
* - channel 1 is the rear channel
- * - channel 2 is the center/lfe chanel
+ * - channel 2 is the center/lfe channel
* Volume is controlled by the AC97 for the front and rear channels by
* the PCM Playback Volume, Sigmatel Surround Playback Volume and
* Surround Playback Volume. The Sigmatel 4-Speaker Stereo switch affects
diff --git a/sound/pci/fm801.c b/sound/pci/fm801.c
index 60cdb9e0b68..83508b3964f 100644
--- a/sound/pci/fm801.c
+++ b/sound/pci/fm801.c
@@ -55,7 +55,7 @@ static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card *
* 1 = MediaForte 256-PCS
* 2 = MediaForte 256-PCPR
* 3 = MediaForte 64-PCR
- * 16 = setup tuner only (this is additional bit), i.e. SF-64-PCR FM card
+ * 16 = setup tuner only (this is additional bit), i.e. SF64-PCR FM card
* High 16-bits are video (radio) device number + 1
*/
static int tea575x_tuner[SNDRV_CARDS];
@@ -67,7 +67,10 @@ MODULE_PARM_DESC(id, "ID string for the FM801 soundcard.");
module_param_array(enable, bool, NULL, 0444);
MODULE_PARM_DESC(enable, "Enable FM801 soundcard.");
module_param_array(tea575x_tuner, int, NULL, 0444);
-MODULE_PARM_DESC(tea575x_tuner, "Enable TEA575x tuner.");
+MODULE_PARM_DESC(tea575x_tuner, "TEA575x tuner access method (1 = SF256-PCS, 2=SF256-PCPR, 3=SF64-PCR, +16=tuner-only).");
+
+#define TUNER_ONLY (1<<4)
+#define TUNER_TYPE_MASK (~TUNER_ONLY & 0xFFFF)
/*
* Direct registers
@@ -160,7 +163,7 @@ struct fm801 {
unsigned int multichannel: 1, /* multichannel support */
secondary: 1; /* secondary codec */
unsigned char secondary_addr; /* address of the secondary codec */
- unsigned int tea575x_tuner; /* tuner flags */
+ unsigned int tea575x_tuner; /* tuner access method & flags */
unsigned short ply_ctrl; /* playback control */
unsigned short cap_ctrl; /* capture control */
@@ -1287,7 +1290,7 @@ static int snd_fm801_chip_init(struct fm801 *chip, int resume)
{
unsigned short cmdw;
- if (chip->tea575x_tuner & 0x0010)
+ if (chip->tea575x_tuner & TUNER_ONLY)
goto __ac97_ok;
/* codec cold reset + AC'97 warm reset */
@@ -1296,11 +1299,13 @@ static int snd_fm801_chip_init(struct fm801 *chip, int resume)
udelay(100);
outw(0, FM801_REG(chip, CODEC_CTRL));
- if (wait_for_codec(chip, 0, AC97_RESET, msecs_to_jiffies(750)) < 0) {
- snd_printk(KERN_ERR "Primary AC'97 codec not found\n");
- if (! resume)
- return -EIO;
- }
+ if (wait_for_codec(chip, 0, AC97_RESET, msecs_to_jiffies(750)) < 0)
+ if (!resume) {
+ snd_printk(KERN_INFO "Primary AC'97 codec not found, "
+ "assume SF64-PCR (tuner-only)\n");
+ chip->tea575x_tuner = 3 | TUNER_ONLY;
+ goto __ac97_ok;
+ }
if (chip->multichannel) {
if (chip->secondary_addr) {
@@ -1414,7 +1419,7 @@ static int __devinit snd_fm801_create(struct snd_card *card,
return err;
}
chip->port = pci_resource_start(pci, 0);
- if ((tea575x_tuner & 0x0010) == 0) {
+ if ((tea575x_tuner & TUNER_ONLY) == 0) {
if (request_irq(pci->irq, snd_fm801_interrupt, IRQF_SHARED,
"FM801", chip)) {
snd_printk(KERN_ERR "unable to grab IRQ %d\n", chip->irq);
@@ -1429,6 +1434,14 @@ static int __devinit snd_fm801_create(struct snd_card *card,
chip->multichannel = 1;
snd_fm801_chip_init(chip, 0);
+ /* init might set tuner access method */
+ tea575x_tuner = chip->tea575x_tuner;
+
+ if (chip->irq >= 0 && (tea575x_tuner & TUNER_ONLY)) {
+ pci_clear_master(pci);
+ free_irq(chip->irq, chip);
+ chip->irq = -1;
+ }
if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops)) < 0) {
snd_fm801_free(chip);
@@ -1438,12 +1451,13 @@ static int __devinit snd_fm801_create(struct snd_card *card,
snd_card_set_dev(card, &pci->dev);
#ifdef TEA575X_RADIO
- if (tea575x_tuner > 0 && (tea575x_tuner & 0x000f) < 4) {
+ if ((tea575x_tuner & TUNER_TYPE_MASK) > 0 &&
+ (tea575x_tuner & TUNER_TYPE_MASK) < 4) {
chip->tea.dev_nr = tea575x_tuner >> 16;
chip->tea.card = card;
chip->tea.freq_fixup = 10700;
chip->tea.private_data = chip;
- chip->tea.ops = &snd_fm801_tea_ops[(tea575x_tuner & 0x000f) - 1];
+ chip->tea.ops = &snd_fm801_tea_ops[(tea575x_tuner & TUNER_TYPE_MASK) - 1];
snd_tea575x_init(&chip->tea);
}
#endif
@@ -1483,7 +1497,7 @@ static int __devinit snd_card_fm801_probe(struct pci_dev *pci,
sprintf(card->longname, "%s at 0x%lx, irq %i",
card->shortname, chip->port, chip->irq);
- if (tea575x_tuner[dev] & 0x0010)
+ if (chip->tea575x_tuner & TUNER_ONLY)
goto __fm801_tuner_only;
if ((err = snd_fm801_pcm(chip, 0, NULL)) < 0) {
diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c
index 2439e84dcb2..4b200da1bd1 100644
--- a/sound/pci/hda/patch_cirrus.c
+++ b/sound/pci/hda/patch_cirrus.c
@@ -938,7 +938,7 @@ static void init_input(struct hda_codec *codec)
coef |= 0x0500; /* DMIC2 enable 2 channels, disable GPIO1 */
if (is_active_pin(codec, CS_DMIC1_PIN_NID))
coef |= 0x1800; /* DMIC1 enable 2 channels, disable GPIO0
- * No effect if SPDIF_OUT2 is slected in
+ * No effect if SPDIF_OUT2 is selected in
* IDX_SPDIF_CTL.
*/
cs_vendor_coef_set(codec, IDX_ADC_CFG, coef);
diff --git a/sound/pci/hda/patch_cmedia.c b/sound/pci/hda/patch_cmedia.c
index 85c81feb10c..a45c1169762 100644
--- a/sound/pci/hda/patch_cmedia.c
+++ b/sound/pci/hda/patch_cmedia.c
@@ -66,7 +66,7 @@ struct cmi_spec {
struct hda_pcm pcm_rec[2]; /* PCM information */
- /* pin deafault configuration */
+ /* pin default configuration */
hda_nid_t pin_nid[NUM_PINS];
unsigned int def_conf[NUM_PINS];
unsigned int pin_def_confs;
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index b3abe9ca826..0877bae5dae 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -6673,7 +6673,7 @@ static struct hda_input_mux alc889A_mb31_capture_source = {
/* Front Mic (0x01) unused */
{ "Line", 0x2 },
/* Line 2 (0x03) unused */
- /* CD (0x04) unsused? */
+ /* CD (0x04) unused? */
},
};
@@ -9287,8 +9287,6 @@ static struct alc_config_preset alc882_presets[] = {
.dac_nids = alc883_dac_nids,
.num_adc_nids = ARRAY_SIZE(alc889_adc_nids),
.adc_nids = alc889_adc_nids,
- .capsrc_nids = alc889_capsrc_nids,
- .capsrc_nids = alc889_capsrc_nids,
.dig_out_nid = ALC883_DIGOUT_NID,
.dig_in_nid = ALC883_DIGIN_NID,
.slave_dig_outs = alc883_slave_dig_outs,
diff --git a/sound/pci/ice1712/aureon.c b/sound/pci/ice1712/aureon.c
index 110d16e5273..765d7bd4c3d 100644
--- a/sound/pci/ice1712/aureon.c
+++ b/sound/pci/ice1712/aureon.c
@@ -689,32 +689,14 @@ static int aureon_ac97_mmute_put(struct snd_kcontrol *kcontrol, struct snd_ctl_e
return change;
}
-static const DECLARE_TLV_DB_SCALE(db_scale_wm_dac, -12700, 100, 1);
+static const DECLARE_TLV_DB_SCALE(db_scale_wm_dac, -10000, 100, 1);
static const DECLARE_TLV_DB_SCALE(db_scale_wm_pcm, -6400, 50, 1);
static const DECLARE_TLV_DB_SCALE(db_scale_wm_adc, -1200, 100, 0);
static const DECLARE_TLV_DB_SCALE(db_scale_ac97_master, -4650, 150, 0);
static const DECLARE_TLV_DB_SCALE(db_scale_ac97_gain, -3450, 150, 0);
-/*
- * Logarithmic volume values for WM8770
- * Computed as 20 * Log10(255 / x)
- */
-static const unsigned char wm_vol[256] = {
- 127, 48, 42, 39, 36, 34, 33, 31, 30, 29, 28, 27, 27, 26, 25, 25, 24, 24, 23,
- 23, 22, 22, 21, 21, 21, 20, 20, 20, 19, 19, 19, 18, 18, 18, 18, 17, 17, 17,
- 17, 16, 16, 16, 16, 15, 15, 15, 15, 15, 15, 14, 14, 14, 14, 14, 13, 13, 13,
- 13, 13, 13, 13, 12, 12, 12, 12, 12, 12, 12, 11, 11, 11, 11, 11, 11, 11, 11,
- 11, 10, 10, 10, 10, 10, 10, 10, 10, 10, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 8, 8,
- 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 6, 6, 6,
- 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5,
- 5, 5, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 3, 3, 3, 3, 3,
- 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2,
- 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1,
- 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0,
- 0, 0
-};
-
-#define WM_VOL_MAX (sizeof(wm_vol) - 1)
+#define WM_VOL_MAX 100
+#define WM_VOL_CNT 101 /* 0dB .. -100dB */
#define WM_VOL_MUTE 0x8000
static void wm_set_vol(struct snd_ice1712 *ice, unsigned int index, unsigned short vol, unsigned short master)
@@ -724,7 +706,8 @@ static void wm_set_vol(struct snd_ice1712 *ice, unsigned int index, unsigned sho
if ((master & WM_VOL_MUTE) || (vol & WM_VOL_MUTE))
nvol = 0;
else
- nvol = 127 - wm_vol[(((vol & ~WM_VOL_MUTE) * (master & ~WM_VOL_MUTE)) / 127) & WM_VOL_MAX];
+ nvol = ((vol % WM_VOL_CNT) * (master % WM_VOL_CNT)) /
+ WM_VOL_MAX;
wm_put(ice, index, nvol);
wm_put_nocache(ice, index, 0x180 | nvol);
@@ -820,7 +803,7 @@ static int wm_vol_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *
uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
uinfo->count = voices;
uinfo->value.integer.min = 0; /* mute (-101dB) */
- uinfo->value.integer.max = 0x7F; /* 0dB */
+ uinfo->value.integer.max = WM_VOL_MAX; /* 0dB */
return 0;
}
@@ -850,7 +833,7 @@ static int wm_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *
snd_ice1712_save_gpio_status(ice);
for (i = 0; i < voices; i++) {
unsigned int vol = ucontrol->value.integer.value[i];
- if (vol > 0x7f)
+ if (vol > WM_VOL_MAX)
continue;
vol |= spec->vol[ofs+i];
if (vol != spec->vol[ofs+i]) {
diff --git a/sound/pci/ice1712/ice1712.h b/sound/pci/ice1712/ice1712.h
index 4615bca39e1..0da778a69ef 100644
--- a/sound/pci/ice1712/ice1712.h
+++ b/sound/pci/ice1712/ice1712.h
@@ -387,8 +387,8 @@ struct snd_ice1712 {
#ifdef CONFIG_PM
int (*pm_suspend)(struct snd_ice1712 *);
int (*pm_resume)(struct snd_ice1712 *);
- int pm_suspend_enabled:1;
- int pm_saved_is_spdif_master:1;
+ unsigned int pm_suspend_enabled:1;
+ unsigned int pm_saved_is_spdif_master:1;
unsigned int pm_saved_spdif_ctrl;
unsigned char pm_saved_spdif_cfg;
unsigned int pm_saved_route;
diff --git a/sound/pci/ice1712/juli.c b/sound/pci/ice1712/juli.c
index 4bed9633a4c..98bc3b7681b 100644
--- a/sound/pci/ice1712/juli.c
+++ b/sound/pci/ice1712/juli.c
@@ -380,7 +380,7 @@ static struct snd_kcontrol_new juli_mute_controls[] __devinitdata = {
* inputs) are fed from Xilinx.
*
* I even checked traces on board and coded a support in driver for
- * an alternative possiblity - the unused I2S ICE output channels
+ * an alternative possibility - the unused I2S ICE output channels
* switched to HW-IN/SPDIF-IN and providing the monitoring signal to
* the DAC - to no avail. The I2S outputs seem to be unconnected.
*
@@ -483,6 +483,31 @@ static int __devinit juli_add_controls(struct snd_ice1712 *ice)
}
/*
+ * suspend/resume
+ * */
+
+#ifdef CONFIG_PM
+static int juli_resume(struct snd_ice1712 *ice)
+{
+ struct snd_akm4xxx *ak = ice->akm;
+ struct juli_spec *spec = ice->spec;
+ /* akm4358 un-reset, un-mute */
+ snd_akm4xxx_reset(ak, 0);
+ /* reinit ak4114 */
+ snd_ak4114_reinit(spec->ak4114);
+ return 0;
+}
+
+static int juli_suspend(struct snd_ice1712 *ice)
+{
+ struct snd_akm4xxx *ak = ice->akm;
+ /* akm4358 reset and soft-mute */
+ snd_akm4xxx_reset(ak, 1);
+ return 0;
+}
+#endif
+
+/*
* initialize the chip
*/
@@ -626,6 +651,13 @@ static int __devinit juli_init(struct snd_ice1712 *ice)
ice->set_spdif_clock = juli_set_spdif_clock;
ice->spdif.ops.open = juli_spdif_in_open;
+
+#ifdef CONFIG_PM
+ ice->pm_resume = juli_resume;
+ ice->pm_suspend = juli_suspend;
+ ice->pm_suspend_enabled = 1;
+#endif
+
return 0;
}
diff --git a/sound/pci/ice1712/prodigy_hifi.c b/sound/pci/ice1712/prodigy_hifi.c
index c75515f5be6..6a9fee3ee78 100644
--- a/sound/pci/ice1712/prodigy_hifi.c
+++ b/sound/pci/ice1712/prodigy_hifi.c
@@ -1100,7 +1100,7 @@ static void ak4396_init(struct snd_ice1712 *ice)
}
#ifdef CONFIG_PM
-static int __devinit prodigy_hd2_resume(struct snd_ice1712 *ice)
+static int prodigy_hd2_resume(struct snd_ice1712 *ice)
{
/* initialize ak4396 codec and restore previous mixer volumes */
struct prodigy_hifi_spec *spec = ice->spec;
diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c
index 754867ed478..b990143636f 100644
--- a/sound/pci/intel8x0.c
+++ b/sound/pci/intel8x0.c
@@ -1950,6 +1950,12 @@ static struct ac97_quirk ac97_quirks[] __devinitdata = {
},
{
.subvendor = 0x104d,
+ .subdevice = 0x8144,
+ .name = "Sony",
+ .type = AC97_TUNE_INV_EAPD
+ },
+ {
+ .subvendor = 0x104d,
.subdevice = 0x8197,
.name = "Sony S1XP",
.type = AC97_TUNE_INV_EAPD
@@ -2057,6 +2063,12 @@ static struct ac97_quirk ac97_quirks[] __devinitdata = {
.type = AC97_TUNE_HP_ONLY
},
{
+ .subvendor = 0x161f,
+ .subdevice = 0x203a,
+ .name = "Gateway 4525GZ", /* AD1981B */
+ .type = AC97_TUNE_INV_EAPD
+ },
+ {
.subvendor = 0x1734,
.subdevice = 0x0088,
.name = "Fujitsu-Siemens D1522", /* AD1981 */
diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c
index 0dce331a2a3..a1b10d1a384 100644
--- a/sound/pci/rme9652/hdspm.c
+++ b/sound/pci/rme9652/hdspm.c
@@ -3017,7 +3017,7 @@ snd_hdspm_proc_read_madi(struct snd_info_entry * entry,
insel = "Coaxial";
break;
default:
- insel = "Unkown";
+ insel = "Unknown";
}
switch (hdspm->control_register & HDSPM_SyncRefMask) {
@@ -3028,7 +3028,7 @@ snd_hdspm_proc_read_madi(struct snd_info_entry * entry,
syncref = "MADI";
break;
default:
- syncref = "Unkown";
+ syncref = "Unknown";
}
snd_iprintf(buffer, "Inputsel = %s, SyncRef = %s\n", insel,
syncref);