aboutsummaryrefslogtreecommitdiff
path: root/sound/pci
diff options
context:
space:
mode:
Diffstat (limited to 'sound/pci')
-rw-r--r--sound/pci/Kconfig111
-rw-r--r--sound/pci/Makefile2
-rw-r--r--sound/pci/ac97/Makefile2
-rw-r--r--sound/pci/ac97/ac97_codec.c40
-rw-r--r--sound/pci/ac97/ac97_id.h3
-rw-r--r--sound/pci/ac97/ac97_local.h2
-rw-r--r--sound/pci/ac97/ac97_patch.c162
-rw-r--r--sound/pci/ac97/ac97_patch.h2
-rw-r--r--sound/pci/ac97/ac97_pcm.c2
-rw-r--r--sound/pci/ac97/ac97_proc.c10
-rw-r--r--sound/pci/ac97/ak4531_codec.c4
-rw-r--r--sound/pci/ali5451/Makefile2
-rw-r--r--sound/pci/ali5451/ali5451.c10
-rw-r--r--sound/pci/als4000.c2
-rw-r--r--sound/pci/au88x0/au88x0.c1
-rw-r--r--sound/pci/au88x0/au88x0_eq.c10
-rw-r--r--sound/pci/au88x0/au88x0_mpu401.c2
-rw-r--r--sound/pci/au88x0/au88x0_synth.c4
-rw-r--r--sound/pci/bt87x.c217
-rw-r--r--sound/pci/ca0106/ca0106.h98
-rw-r--r--sound/pci/ca0106/ca0106_main.c103
-rw-r--r--sound/pci/ca0106/ca0106_mixer.c98
-rw-r--r--sound/pci/ca0106/ca_midi.c2
-rw-r--r--sound/pci/ca0106/ca_midi.h6
-rw-r--r--sound/pci/cmipci.c537
-rw-r--r--sound/pci/cs4281.c28
-rw-r--r--sound/pci/cs46xx/Makefile8
-rw-r--r--sound/pci/cs46xx/cs46xx.c4
-rw-r--r--sound/pci/cs46xx/cs46xx_lib.c12
-rw-r--r--sound/pci/cs46xx/cs46xx_lib.h2
-rw-r--r--sound/pci/cs46xx/dsp_spos.h2
-rw-r--r--sound/pci/cs46xx/dsp_spos_scb_lib.c2
-rw-r--r--sound/pci/cs5535audio/Makefile7
-rw-r--r--sound/pci/cs5535audio/cs5535audio.c24
-rw-r--r--sound/pci/cs5535audio/cs5535audio.h42
-rw-r--r--sound/pci/cs5535audio/cs5535audio_pcm.c10
-rw-r--r--sound/pci/cs5535audio/cs5535audio_pm.c26
-rw-r--r--sound/pci/echoaudio/echoaudio.c33
-rw-r--r--sound/pci/echoaudio/echoaudio_dsp.c4
-rw-r--r--sound/pci/echoaudio/echoaudio_dsp.h15
-rw-r--r--sound/pci/emu10k1/Makefile2
-rw-r--r--sound/pci/emu10k1/emu10k1.c4
-rw-r--r--sound/pci/emu10k1/emu10k1_main.c130
-rw-r--r--sound/pci/emu10k1/emu10k1x.c9
-rw-r--r--sound/pci/emu10k1/emufx.c251
-rw-r--r--sound/pci/emu10k1/emumixer.c86
-rw-r--r--sound/pci/emu10k1/emumpu401.c2
-rw-r--r--sound/pci/emu10k1/emupcm.c2
-rw-r--r--sound/pci/emu10k1/emuproc.c58
-rw-r--r--sound/pci/emu10k1/io.c12
-rw-r--r--sound/pci/emu10k1/irq.c2
-rw-r--r--sound/pci/emu10k1/memory.c2
-rw-r--r--sound/pci/emu10k1/p16v.c19
-rw-r--r--sound/pci/emu10k1/voice.c2
-rw-r--r--sound/pci/ens1370.c44
-rw-r--r--sound/pci/es1938.c22
-rw-r--r--sound/pci/es1968.c28
-rw-r--r--sound/pci/fm801.c4
-rw-r--r--sound/pci/hda/Makefile27
-rw-r--r--sound/pci/hda/hda_codec.c735
-rw-r--r--sound/pci/hda/hda_codec.h113
-rw-r--r--sound/pci/hda/hda_generic.c100
-rw-r--r--sound/pci/hda/hda_hwdep.c122
-rw-r--r--sound/pci/hda/hda_intel.c382
-rw-r--r--sound/pci/hda/hda_local.h193
-rw-r--r--sound/pci/hda/hda_patch.h16
-rw-r--r--sound/pci/hda/hda_proc.c30
-rw-r--r--sound/pci/hda/patch_analog.c524
-rw-r--r--sound/pci/hda/patch_atihdmi.c16
-rw-r--r--sound/pci/hda/patch_cmedia.c24
-rw-r--r--sound/pci/hda/patch_conexant.c156
-rw-r--r--sound/pci/hda/patch_realtek.c1840
-rw-r--r--sound/pci/hda/patch_si3054.c20
-rw-r--r--sound/pci/hda/patch_sigmatel.c1000
-rw-r--r--sound/pci/hda/patch_via.c80
-rw-r--r--sound/pci/ice1712/Makefile2
-rw-r--r--sound/pci/ice1712/ak4xxx.c4
-rw-r--r--sound/pci/ice1712/amp.c2
-rw-r--r--sound/pci/ice1712/amp.h2
-rw-r--r--sound/pci/ice1712/aureon.c45
-rw-r--r--sound/pci/ice1712/delta.c13
-rw-r--r--sound/pci/ice1712/delta.h2
-rw-r--r--sound/pci/ice1712/envy24ht.h2
-rw-r--r--sound/pci/ice1712/ews.c20
-rw-r--r--sound/pci/ice1712/ews.h2
-rw-r--r--sound/pci/ice1712/hoontech.c2
-rw-r--r--sound/pci/ice1712/hoontech.h2
-rw-r--r--sound/pci/ice1712/ice1712.c52
-rw-r--r--sound/pci/ice1712/ice1712.h5
-rw-r--r--sound/pci/ice1712/ice1724.c54
-rw-r--r--sound/pci/ice1712/juli.c2
-rw-r--r--sound/pci/ice1712/phase.c23
-rw-r--r--sound/pci/ice1712/pontis.c27
-rw-r--r--sound/pci/ice1712/prodigy192.c27
-rw-r--r--sound/pci/ice1712/wtm.c29
-rw-r--r--sound/pci/intel8x0.c4
-rw-r--r--sound/pci/intel8x0m.c4
-rw-r--r--sound/pci/korg1212/Makefile2
-rw-r--r--sound/pci/korg1212/korg1212.c4
-rw-r--r--sound/pci/maestro3.c2
-rw-r--r--sound/pci/mixart/Makefile2
-rw-r--r--sound/pci/mixart/mixart.c10
-rw-r--r--sound/pci/mixart/mixart_mixer.c9
-rw-r--r--sound/pci/nm256/Makefile2
-rw-r--r--sound/pci/nm256/nm256.c1
-rw-r--r--sound/pci/pcxhr/pcxhr.c5
-rw-r--r--sound/pci/pcxhr/pcxhr_mixer.c15
-rw-r--r--sound/pci/rme32.c33
-rw-r--r--sound/pci/rme96.c41
-rw-r--r--sound/pci/rme9652/Makefile2
-rw-r--r--sound/pci/rme9652/hdsp.c90
-rw-r--r--sound/pci/rme9652/hdspm.c723
-rw-r--r--sound/pci/rme9652/rme9652.c27
-rw-r--r--sound/pci/sonicvibes.c4
-rw-r--r--sound/pci/trident/Makefile2
-rw-r--r--sound/pci/trident/trident.c2
-rw-r--r--sound/pci/trident/trident_main.c22
-rw-r--r--sound/pci/trident/trident_memory.c2
-rw-r--r--sound/pci/via82xx.c19
-rw-r--r--sound/pci/via82xx_modem.c8
-rw-r--r--sound/pci/vx222/Makefile2
-rw-r--r--sound/pci/ymfpci/Makefile2
-rw-r--r--sound/pci/ymfpci/ymfpci.c4
-rw-r--r--sound/pci/ymfpci/ymfpci_main.c108
124 files changed, 5820 insertions, 3356 deletions
diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig
index c6b44102aa5b..356bf21a1506 100644
--- a/sound/pci/Kconfig
+++ b/sound/pci/Kconfig
@@ -170,14 +170,14 @@ config SND_CA0106
will be called snd-ca0106.
config SND_CMIPCI
- tristate "C-Media 8738, 8338"
+ tristate "C-Media 8338, 8738, 8768, 8770"
depends on SND
select SND_OPL3_LIB
select SND_MPU401_UART
select SND_PCM
help
- If you want to use soundcards based on C-Media CMI8338 or CMI8738
- chips, say Y here and read
+ If you want to use soundcards based on C-Media CMI8338, CMI8738,
+ CMI8768 or CMI8770 chips, say Y here and read
<file:Documentation/sound/alsa/CMIPCI.txt>.
To compile this driver as a module, choose M here: the module
@@ -500,6 +500,103 @@ config SND_HDA_INTEL
To compile this driver as a module, choose M here: the module
will be called snd-hda-intel.
+config SND_HDA_HWDEP
+ bool "Build hwdep interface for HD-audio driver"
+ depends on SND_HDA_INTEL
+ select SND_HWDEP
+ help
+ Say Y here to build a hwdep interface for HD-audio driver.
+ This interface can be used for out-of-band communication
+ with codecs for debugging purposes.
+
+config SND_HDA_CODEC_REALTEK
+ bool "Build Realtek HD-audio codec support"
+ depends on SND_HDA_INTEL
+ default y
+ help
+ Say Y here to include Realtek HD-audio codec support in
+ snd-hda-intel driver, such as ALC880.
+
+config SND_HDA_CODEC_ANALOG
+ bool "Build Analog Device HD-audio codec support"
+ depends on SND_HDA_INTEL
+ default y
+ help
+ Say Y here to include Analog Device HD-audio codec support in
+ snd-hda-intel driver, such as AD1986A.
+
+config SND_HDA_CODEC_SIGMATEL
+ bool "Build IDT/Sigmatel HD-audio codec support"
+ depends on SND_HDA_INTEL
+ default y
+ help
+ Say Y here to include IDT (Sigmatel) HD-audio codec support in
+ snd-hda-intel driver, such as STAC9200.
+
+config SND_HDA_CODEC_VIA
+ bool "Build VIA HD-audio codec support"
+ depends on SND_HDA_INTEL
+ default y
+ help
+ Say Y here to include VIA HD-audio codec support in
+ snd-hda-intel driver, such as VT1708.
+
+config SND_HDA_CODEC_ATIHDMI
+ bool "Build ATI HDMI HD-audio codec support"
+ depends on SND_HDA_INTEL
+ default y
+ help
+ Say Y here to include ATI HDMI HD-audio codec support in
+ snd-hda-intel driver, such as ATI RS600 HDMI.
+
+config SND_HDA_CODEC_CONEXANT
+ bool "Build Conexant HD-audio codec support"
+ depends on SND_HDA_INTEL
+ default y
+ help
+ Say Y here to include Conexant HD-audio codec support in
+ snd-hda-intel driver, such as CX20549.
+
+config SND_HDA_CODEC_CMEDIA
+ bool "Build C-Media HD-audio codec support"
+ depends on SND_HDA_INTEL
+ default y
+ help
+ Say Y here to include C-Media HD-audio codec support in
+ snd-hda-intel driver, such as CMI9880.
+
+config SND_HDA_CODEC_SI3054
+ bool "Build Silicon Labs 3054 HD-modem codec support"
+ depends on SND_HDA_INTEL
+ default y
+ help
+ Say Y here to include Silicon Labs 3054 HD-modem codec
+ (and compatibles) support in snd-hda-intel driver.
+
+config SND_HDA_GENERIC
+ bool "Enable generic HD-audio codec parser"
+ depends on SND_HDA_INTEL
+ default y
+ help
+ Say Y here to enable the generic HD-audio codec parser
+ in snd-hda-intel driver.
+
+config SND_HDA_POWER_SAVE
+ bool "Aggressive power-saving on HD-audio"
+ depends on SND_HDA_INTEL && EXPERIMENTAL
+ help
+ Say Y here to enable more aggressive power-saving mode on
+ HD-audio driver. The power-saving timeout can be configured
+ via power_save option or over sysfs on-the-fly.
+
+config SND_HDA_POWER_SAVE_DEFAULT
+ int "Default time-out for HD-audio power-save mode"
+ depends on SND_HDA_POWER_SAVE
+ default 0
+ help
+ The default time-out value in seconds for HD-audio automatic
+ power-save mode. 0 means to disable the power-save mode.
+
config SND_HDSP
tristate "RME Hammerfall DSP Audio"
depends on SND
@@ -799,4 +896,12 @@ config SND_AC97_POWER_SAVE
snd-ac97-codec driver. You can toggle it dynamically over
sysfs, too.
+config SND_AC97_POWER_SAVE_DEFAULT
+ int "Default time-out for AC97 power-save mode"
+ depends on SND_AC97_POWER_SAVE
+ default 0
+ help
+ The default time-out value in seconds for AC97 automatic
+ power-save mode. 0 means to disable the power-save mode.
+
endmenu
diff --git a/sound/pci/Makefile b/sound/pci/Makefile
index cd76e0293d06..09ddc82eeca2 100644
--- a/sound/pci/Makefile
+++ b/sound/pci/Makefile
@@ -1,6 +1,6 @@
#
# Makefile for ALSA
-# Copyright (c) 2001 by Jaroslav Kysela <perex@suse.cz>
+# Copyright (c) 2001 by Jaroslav Kysela <perex@perex.cz>
#
snd-ad1889-objs := ad1889.o
diff --git a/sound/pci/ac97/Makefile b/sound/pci/ac97/Makefile
index f5d471896b95..0be48b1a22d0 100644
--- a/sound/pci/ac97/Makefile
+++ b/sound/pci/ac97/Makefile
@@ -1,6 +1,6 @@
#
# Makefile for ALSA
-# Copyright (c) 2001 by Jaroslav Kysela <perex@suse.cz>
+# Copyright (c) 2001 by Jaroslav Kysela <perex@perex.cz>
#
snd-ac97-codec-objs := ac97_codec.o ac97_pcm.o
diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c
index bbed644bf9c5..6a9966df0cc9 100644
--- a/sound/pci/ac97/ac97_codec.c
+++ b/sound/pci/ac97/ac97_codec.c
@@ -1,5 +1,5 @@
/*
- * Copyright (c) by Jaroslav Kysela <perex@suse.cz>
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
* Universal interface for Audio Codec '97
*
* For more details look to AC '97 component specification revision 2.2
@@ -39,7 +39,7 @@
#include "ac97_patch.c"
-MODULE_AUTHOR("Jaroslav Kysela <perex@suse.cz>");
+MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
MODULE_DESCRIPTION("Universal interface for Audio Codec '97");
MODULE_LICENSE("GPL");
@@ -49,7 +49,7 @@ module_param(enable_loopback, bool, 0444);
MODULE_PARM_DESC(enable_loopback, "Enable AC97 ADC/DAC Loopback Control");
#ifdef CONFIG_SND_AC97_POWER_SAVE
-static int power_save;
+static int power_save = CONFIG_SND_AC97_POWER_SAVE_DEFAULT;
module_param(power_save, bool, 0644);
MODULE_PARM_DESC(power_save, "Enable AC97 power-saving control");
#endif
@@ -176,7 +176,7 @@ static const struct ac97_codec_id snd_ac97_codec_ids[] = {
{ 0x574d4C09, 0xffffffff, "WM9709", NULL, NULL},
{ 0x574d4C12, 0xffffffff, "WM9711,WM9712", patch_wolfson11, NULL},
{ 0x574d4c13, 0xffffffff, "WM9713,WM9714", patch_wolfson13, NULL, AC97_DEFAULT_POWER_OFF},
-{ 0x594d4800, 0xffffffff, "YMF743", NULL, NULL },
+{ 0x594d4800, 0xffffffff, "YMF743", patch_yamaha_ymf743, NULL },
{ 0x594d4802, 0xffffffff, "YMF752", NULL, NULL },
{ 0x594d4803, 0xffffffff, "YMF753", patch_yamaha_ymf753, NULL },
{ 0x83847600, 0xffffffff, "STAC9700,83,84", patch_sigmatel_stac9700, NULL },
@@ -779,6 +779,12 @@ static int snd_ac97_spdif_default_put(struct snd_kcontrol *kcontrol, struct snd_
change |= snd_ac97_update_bits_nolock(ac97, AC97_CXR_AUDIO_MISC,
AC97_CXR_SPDIF_MASK | AC97_CXR_COPYRGT,
v);
+ } else if (ac97->id == AC97_ID_YMF743) {
+ change |= snd_ac97_update_bits_nolock(ac97,
+ AC97_YMF7X3_DIT_CTRL,
+ 0xff38,
+ ((val << 4) & 0xff00) |
+ ((val << 2) & 0x0038));
} else {
unsigned short extst = snd_ac97_read_cache(ac97, AC97_EXTENDED_STATUS);
snd_ac97_update_bits_nolock(ac97, AC97_EXTENDED_STATUS, AC97_EA_SPDIF, 0); /* turn off */
@@ -1375,7 +1381,8 @@ static int snd_ac97_mixer_build(struct snd_ac97 * ac97)
for (idx = 0; idx < 2; idx++) {
if ((err = snd_ctl_add(card, kctl = snd_ac97_cnew(&snd_ac97_controls_tone[idx], ac97))) < 0)
return err;
- if (ac97->id == AC97_ID_YMF753) {
+ if (ac97->id == AC97_ID_YMF743 ||
+ ac97->id == AC97_ID_YMF753) {
kctl->private_value &= ~(0xff << 16);
kctl->private_value |= 7 << 16;
}
@@ -2036,11 +2043,12 @@ int snd_ac97_mixer(struct snd_ac97_bus *bus, struct snd_ac97_template *template,
else {
udelay(50);
if (ac97->scaps & AC97_SCAP_SKIP_AUDIO)
- err = ac97_reset_wait(ac97, HZ/2, 1);
+ err = ac97_reset_wait(ac97, msecs_to_jiffies(500), 1);
else {
- err = ac97_reset_wait(ac97, HZ/2, 0);
+ err = ac97_reset_wait(ac97, msecs_to_jiffies(500), 0);
if (err < 0)
- err = ac97_reset_wait(ac97, HZ/2, 1);
+ err = ac97_reset_wait(ac97,
+ msecs_to_jiffies(500), 1);
}
if (err < 0) {
snd_printk(KERN_WARNING "AC'97 %d does not respond - RESET\n", ac97->num);
@@ -2104,7 +2112,7 @@ 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 + (HZ / 10);
+ end_time = jiffies + msecs_to_jiffies(100);
do {
if ((snd_ac97_read(ac97, AC97_POWERDOWN) & 0x0f) == 0x0f)
goto __ready_ok;
@@ -2136,7 +2144,7 @@ int snd_ac97_mixer(struct snd_ac97_bus *bus, struct snd_ac97_template *template,
udelay(100);
/* nothing should be in powerdown mode */
snd_ac97_write_cache(ac97, AC97_EXTENDED_MSTATUS, 0);
- end_time = jiffies + (HZ / 10);
+ end_time = jiffies + msecs_to_jiffies(100);
do {
if ((snd_ac97_read(ac97, AC97_EXTENDED_MSTATUS) & tmp) == tmp)
goto __ready_ok;
@@ -2354,7 +2362,8 @@ int snd_ac97_update_power(struct snd_ac97 *ac97, int reg, int powerup)
* (for avoiding loud click noises for many (OSS) apps
* that open/close frequently)
*/
- schedule_delayed_work(&ac97->power_work, HZ*2);
+ schedule_delayed_work(&ac97->power_work,
+ msecs_to_jiffies(2000));
else {
cancel_delayed_work(&ac97->power_work);
update_power_regs(ac97);
@@ -2436,7 +2445,7 @@ EXPORT_SYMBOL(snd_ac97_suspend);
/*
* restore ac97 status
*/
-void snd_ac97_restore_status(struct snd_ac97 *ac97)
+static void snd_ac97_restore_status(struct snd_ac97 *ac97)
{
int i;
@@ -2457,7 +2466,7 @@ void snd_ac97_restore_status(struct snd_ac97 *ac97)
/*
* restore IEC958 status
*/
-void snd_ac97_restore_iec958(struct snd_ac97 *ac97)
+static void snd_ac97_restore_iec958(struct snd_ac97 *ac97)
{
if (ac97->ext_id & AC97_EI_SPDIF) {
if (ac97->regs[AC97_EXTENDED_STATUS] & AC97_EA_SPDIF) {
@@ -2494,7 +2503,10 @@ void snd_ac97_resume(struct snd_ac97 *ac97)
snd_ac97_write(ac97, AC97_POWERDOWN, 0);
if (! (ac97->flags & AC97_DEFAULT_POWER_OFF)) {
- snd_ac97_write(ac97, AC97_RESET, 0);
+ if (!(ac97->scaps & AC97_SCAP_SKIP_AUDIO))
+ snd_ac97_write(ac97, AC97_RESET, 0);
+ else if (!(ac97->scaps & AC97_SCAP_SKIP_MODEM))
+ snd_ac97_write(ac97, AC97_EXTENDED_MID, 0);
udelay(100);
snd_ac97_write(ac97, AC97_POWERDOWN, 0);
}
diff --git a/sound/pci/ac97/ac97_id.h b/sound/pci/ac97/ac97_id.h
index 6d73514dc49e..c129492c82b3 100644
--- a/sound/pci/ac97/ac97_id.h
+++ b/sound/pci/ac97/ac97_id.h
@@ -1,5 +1,5 @@
/*
- * Copyright (c) by Jaroslav Kysela <perex@suse.cz>
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
* Universal interface for Audio Codec '97
*
* For more details look to AC '97 component specification revision 2.2
@@ -54,6 +54,7 @@
#define AC97_ID_ALC658 0x414c4780
#define AC97_ID_ALC658D 0x414c4781
#define AC97_ID_ALC850 0x414c4790
+#define AC97_ID_YMF743 0x594d4800
#define AC97_ID_YMF753 0x594d4803
#define AC97_ID_VT1616 0x49434551
#define AC97_ID_CM9738 0x434d4941
diff --git a/sound/pci/ac97/ac97_local.h b/sound/pci/ac97/ac97_local.h
index 78745c5c6df8..c276a5e3f7ac 100644
--- a/sound/pci/ac97/ac97_local.h
+++ b/sound/pci/ac97/ac97_local.h
@@ -1,5 +1,5 @@
/*
- * Copyright (c) by Jaroslav Kysela <perex@suse.cz>
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
* Universal interface for Audio Codec '97
*
* For more details look to AC '97 component specification revision 2.2
diff --git a/sound/pci/ac97/ac97_patch.c b/sound/pci/ac97/ac97_patch.c
index 581ebba4d1a7..98c8b727b62b 100644
--- a/sound/pci/ac97/ac97_patch.c
+++ b/sound/pci/ac97/ac97_patch.c
@@ -1,5 +1,5 @@
/*
- * Copyright (c) by Jaroslav Kysela <perex@suse.cz>
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
* Universal interface for Audio Codec '97
*
* For more details look to AC '97 component specification revision 2.2
@@ -204,9 +204,13 @@ static inline int is_shared_micin(struct snd_ac97 *ac97)
/* The following snd_ac97_ymf753_... items added by David Shust (dshust@shustring.com) */
+/* Modified for YMF743 by Keita Maehara <maehara@debian.org> */
-/* It is possible to indicate to the Yamaha YMF753 the type of speakers being used. */
-static int snd_ac97_ymf753_info_speaker(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+/* It is possible to indicate to the Yamaha YMF7x3 the type of
+ speakers being used. */
+
+static int snd_ac97_ymf7x3_info_speaker(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
{
static char *texts[3] = {
"Standard", "Small", "Smaller"
@@ -221,12 +225,13 @@ static int snd_ac97_ymf753_info_speaker(struct snd_kcontrol *kcontrol, struct sn
return 0;
}
-static int snd_ac97_ymf753_get_speaker(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+static int snd_ac97_ymf7x3_get_speaker(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
{
struct snd_ac97 *ac97 = snd_kcontrol_chip(kcontrol);
unsigned short val;
- val = ac97->regs[AC97_YMF753_3D_MODE_SEL];
+ val = ac97->regs[AC97_YMF7X3_3D_MODE_SEL];
val = (val >> 10) & 3;
if (val > 0) /* 0 = invalid */
val--;
@@ -234,7 +239,8 @@ static int snd_ac97_ymf753_get_speaker(struct snd_kcontrol *kcontrol, struct snd
return 0;
}
-static int snd_ac97_ymf753_put_speaker(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+static int snd_ac97_ymf7x3_put_speaker(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
{
struct snd_ac97 *ac97 = snd_kcontrol_chip(kcontrol);
unsigned short val;
@@ -242,20 +248,22 @@ static int snd_ac97_ymf753_put_speaker(struct snd_kcontrol *kcontrol, struct snd
if (ucontrol->value.enumerated.item[0] > 2)
return -EINVAL;
val = (ucontrol->value.enumerated.item[0] + 1) << 10;
- return snd_ac97_update(ac97, AC97_YMF753_3D_MODE_SEL, val);
+ return snd_ac97_update(ac97, AC97_YMF7X3_3D_MODE_SEL, val);
}
-static const struct snd_kcontrol_new snd_ac97_ymf753_controls_speaker =
+static const struct snd_kcontrol_new snd_ac97_ymf7x3_controls_speaker =
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.name = "3D Control - Speaker",
- .info = snd_ac97_ymf753_info_speaker,
- .get = snd_ac97_ymf753_get_speaker,
- .put = snd_ac97_ymf753_put_speaker,
+ .info = snd_ac97_ymf7x3_info_speaker,
+ .get = snd_ac97_ymf7x3_get_speaker,
+ .put = snd_ac97_ymf7x3_put_speaker,
};
-/* It is possible to indicate to the Yamaha YMF753 the source to direct to the S/PDIF output. */
-static int snd_ac97_ymf753_spdif_source_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+/* It is possible to indicate to the Yamaha YMF7x3 the source to
+ direct to the S/PDIF output. */
+static int snd_ac97_ymf7x3_spdif_source_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
{
static char *texts[2] = { "AC-Link", "A/D Converter" };
@@ -268,17 +276,19 @@ static int snd_ac97_ymf753_spdif_source_info(struct snd_kcontrol *kcontrol, stru
return 0;
}
-static int snd_ac97_ymf753_spdif_source_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+static int snd_ac97_ymf7x3_spdif_source_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
{
struct snd_ac97 *ac97 = snd_kcontrol_chip(kcontrol);
unsigned short val;
- val = ac97->regs[AC97_YMF753_DIT_CTRL2];
+ val = ac97->regs[AC97_YMF7X3_DIT_CTRL];
ucontrol->value.enumerated.item[0] = (val >> 1) & 1;
return 0;
}
-static int snd_ac97_ymf753_spdif_source_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+static int snd_ac97_ymf7x3_spdif_source_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
{
struct snd_ac97 *ac97 = snd_kcontrol_chip(kcontrol);
unsigned short val;
@@ -286,7 +296,75 @@ static int snd_ac97_ymf753_spdif_source_put(struct snd_kcontrol *kcontrol, struc
if (ucontrol->value.enumerated.item[0] > 1)
return -EINVAL;
val = ucontrol->value.enumerated.item[0] << 1;
- return snd_ac97_update_bits(ac97, AC97_YMF753_DIT_CTRL2, 0x0002, val);
+ return snd_ac97_update_bits(ac97, AC97_YMF7X3_DIT_CTRL, 0x0002, val);
+}
+
+static int patch_yamaha_ymf7x3_3d(struct snd_ac97 *ac97)
+{
+ struct snd_kcontrol *kctl;
+ int err;
+
+ kctl = snd_ac97_cnew(&snd_ac97_controls_3d[0], ac97);
+ err = snd_ctl_add(ac97->bus->card, kctl);
+ if (err < 0)
+ return err;
+ strcpy(kctl->id.name, "3D Control - Wide");
+ kctl->private_value = AC97_SINGLE_VALUE(AC97_3D_CONTROL, 9, 7, 0);
+ snd_ac97_write_cache(ac97, AC97_3D_CONTROL, 0x0000);
+ err = snd_ctl_add(ac97->bus->card,
+ snd_ac97_cnew(&snd_ac97_ymf7x3_controls_speaker,
+ ac97));
+ if (err < 0)
+ return err;
+ snd_ac97_write_cache(ac97, AC97_YMF7X3_3D_MODE_SEL, 0x0c00);
+ return 0;
+}
+
+static const struct snd_kcontrol_new snd_ac97_yamaha_ymf743_controls_spdif[3] =
+{
+ AC97_SINGLE(SNDRV_CTL_NAME_IEC958("", PLAYBACK, SWITCH),
+ AC97_YMF7X3_DIT_CTRL, 0, 1, 0),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = SNDRV_CTL_NAME_IEC958("", PLAYBACK, NONE) "Source",
+ .info = snd_ac97_ymf7x3_spdif_source_info,
+ .get = snd_ac97_ymf7x3_spdif_source_get,
+ .put = snd_ac97_ymf7x3_spdif_source_put,
+ },
+ AC97_SINGLE(SNDRV_CTL_NAME_IEC958("", NONE, NONE) "Mute",
+ AC97_YMF7X3_DIT_CTRL, 2, 1, 1)
+};
+
+static int patch_yamaha_ymf743_build_spdif(struct snd_ac97 *ac97)
+{
+ int err;
+
+ err = patch_build_controls(ac97, &snd_ac97_controls_spdif[0], 3);
+ if (err < 0)
+ return err;
+ err = patch_build_controls(ac97,
+ snd_ac97_yamaha_ymf743_controls_spdif, 3);
+ if (err < 0)
+ return err;
+ /* set default PCM S/PDIF params */
+ /* PCM audio,no copyright,no preemphasis,PCM coder,original */
+ snd_ac97_write_cache(ac97, AC97_YMF7X3_DIT_CTRL, 0xa201);
+ return 0;
+}
+
+static struct snd_ac97_build_ops patch_yamaha_ymf743_ops = {
+ .build_spdif = patch_yamaha_ymf743_build_spdif,
+ .build_3d = patch_yamaha_ymf7x3_3d,
+};
+
+static int patch_yamaha_ymf743(struct snd_ac97 *ac97)
+{
+ ac97->build_ops = &patch_yamaha_ymf743_ops;
+ ac97->caps |= AC97_BC_BASS_TREBLE;
+ ac97->caps |= 0x04 << 10; /* Yamaha 3D enhancement */
+ ac97->rates[AC97_RATES_SPDIF] = SNDRV_PCM_RATE_48000; /* 48k only */
+ ac97->ext_id |= AC97_EI_SPDIF; /* force the detection of spdif */
+ return 0;
}
/* The AC'97 spec states that the S/PDIF signal is to be output at pin 48.
@@ -311,7 +389,7 @@ static int snd_ac97_ymf753_spdif_output_pin_get(struct snd_kcontrol *kcontrol, s
struct snd_ac97 *ac97 = snd_kcontrol_chip(kcontrol);
unsigned short val;
- val = ac97->regs[AC97_YMF753_DIT_CTRL2];
+ val = ac97->regs[AC97_YMF7X3_DIT_CTRL];
ucontrol->value.enumerated.item[0] = (val & 0x0008) ? 2 : (val & 0x0020) ? 1 : 0;
return 0;
}
@@ -325,7 +403,7 @@ static int snd_ac97_ymf753_spdif_output_pin_put(struct snd_kcontrol *kcontrol, s
return -EINVAL;
val = (ucontrol->value.enumerated.item[0] == 2) ? 0x0008 :
(ucontrol->value.enumerated.item[0] == 1) ? 0x0020 : 0;
- return snd_ac97_update_bits(ac97, AC97_YMF753_DIT_CTRL2, 0x0028, val);
+ return snd_ac97_update_bits(ac97, AC97_YMF7X3_DIT_CTRL, 0x0028, val);
/* The following can be used to direct S/PDIF output to pin 47 (EAPD).
snd_ac97_write_cache(ac97, 0x62, snd_ac97_read(ac97, 0x62) | 0x0008); */
}
@@ -334,9 +412,9 @@ static const struct snd_kcontrol_new snd_ac97_ymf753_controls_spdif[3] = {
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,NONE) "Source",
- .info = snd_ac97_ymf753_spdif_source_info,
- .get = snd_ac97_ymf753_spdif_source_get,
- .put = snd_ac97_ymf753_spdif_source_put,
+ .info = snd_ac97_ymf7x3_spdif_source_info,
+ .get = snd_ac97_ymf7x3_spdif_source_get,
+ .put = snd_ac97_ymf7x3_spdif_source_put,
},
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
@@ -345,25 +423,10 @@ static const struct snd_kcontrol_new snd_ac97_ymf753_controls_spdif[3] = {
.get = snd_ac97_ymf753_spdif_output_pin_get,
.put = snd_ac97_ymf753_spdif_output_pin_put,
},
- AC97_SINGLE(SNDRV_CTL_NAME_IEC958("",NONE,NONE) "Mute", AC97_YMF753_DIT_CTRL2, 2, 1, 1)
+ AC97_SINGLE(SNDRV_CTL_NAME_IEC958("", NONE, NONE) "Mute",
+ AC97_YMF7X3_DIT_CTRL, 2, 1, 1)
};
-static int patch_yamaha_ymf753_3d(struct snd_ac97 * ac97)
-{
- struct snd_kcontrol *kctl;
- int err;
-
- if ((err = snd_ctl_add(ac97->bus->card, kctl = snd_ac97_cnew(&snd_ac97_controls_3d[0], ac97))) < 0)
- return err;
- strcpy(kctl->id.name, "3D Control - Wide");
- kctl->private_value = AC97_SINGLE_VALUE(AC97_3D_CONTROL, 9, 7, 0);
- snd_ac97_write_cache(ac97, AC97_3D_CONTROL, 0x0000);
- if ((err = snd_ctl_add(ac97->bus->card, snd_ac97_cnew(&snd_ac97_ymf753_controls_speaker, ac97))) < 0)
- return err;
- snd_ac97_write_cache(ac97, AC97_YMF753_3D_MODE_SEL, 0x0c00);
- return 0;
-}
-
static int patch_yamaha_ymf753_post_spdif(struct snd_ac97 * ac97)
{
int err;
@@ -374,7 +437,7 @@ static int patch_yamaha_ymf753_post_spdif(struct snd_ac97 * ac97)
}
static struct snd_ac97_build_ops patch_yamaha_ymf753_ops = {
- .build_3d = patch_yamaha_ymf753_3d,
+ .build_3d = patch_yamaha_ymf7x3_3d,
.build_post_spdif = patch_yamaha_ymf753_post_spdif
};
@@ -1880,14 +1943,7 @@ static int patch_ad1981b(struct snd_ac97 *ac97)
return 0;
}
-static int snd_ac97_ad1888_lohpsel_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
-{
- uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
- uinfo->count = 1;
- uinfo->value.integer.min = 0;
- uinfo->value.integer.max = 1;
- return 0;
-}
+#define snd_ac97_ad1888_lohpsel_info snd_ctl_boolean_mono_info
static int snd_ac97_ad1888_lohpsel_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
{
@@ -2186,15 +2242,7 @@ static int patch_ad1985(struct snd_ac97 * ac97)
return 0;
}
-static int snd_ac97_ad1986_bool_info(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_info *uinfo)
-{
- uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
- uinfo->count = 1;
- uinfo->value.integer.min = 0;
- uinfo->value.integer.max = 1;
- return 0;
-}
+#define snd_ac97_ad1986_bool_info snd_ctl_boolean_mono_info
static int snd_ac97_ad1986_lososel_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
diff --git a/sound/pci/ac97/ac97_patch.h b/sound/pci/ac97/ac97_patch.h
index fd341ce63762..9cccc27ea1b5 100644
--- a/sound/pci/ac97/ac97_patch.h
+++ b/sound/pci/ac97/ac97_patch.h
@@ -1,5 +1,5 @@
/*
- * Copyright (c) by Jaroslav Kysela <perex@suse.cz>
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
* Universal interface for Audio Codec '97
*
* For more details look to AC '97 component specification revision 2.2
diff --git a/sound/pci/ac97/ac97_pcm.c b/sound/pci/ac97/ac97_pcm.c
index 4281e6d0c5b6..8cbc03332b01 100644
--- a/sound/pci/ac97/ac97_pcm.c
+++ b/sound/pci/ac97/ac97_pcm.c
@@ -1,5 +1,5 @@
/*
- * Copyright (c) by Jaroslav Kysela <perex@suse.cz>
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
* Universal interface for Audio Codec '97
*
* For more details look to AC '97 component specification revision 2.2
diff --git a/sound/pci/ac97/ac97_proc.c b/sound/pci/ac97/ac97_proc.c
index a3fdd7da911c..fed4a2c3d8a1 100644
--- a/sound/pci/ac97/ac97_proc.c
+++ b/sound/pci/ac97/ac97_proc.c
@@ -1,5 +1,5 @@
/*
- * Copyright (c) by Jaroslav Kysela <perex@suse.cz>
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
* Universal interface for Audio Codec '97
*
* For more details look to AC '97 component specification revision 2.2
@@ -236,10 +236,14 @@ static void snd_ac97_proc_read_main(struct snd_ac97 *ac97, struct snd_info_buffe
val = snd_ac97_read(ac97, AC97_PCM_MIC_ADC_RATE);
snd_iprintf(buffer, "PCM MIC ADC : %iHz\n", val);
}
- if ((ext & AC97_EI_SPDIF) || (ac97->flags & AC97_CS_SPDIF)) {
+ if ((ext & AC97_EI_SPDIF) || (ac97->flags & AC97_CS_SPDIF) ||
+ (ac97->id == AC97_ID_YMF743)) {
if (ac97->flags & AC97_CS_SPDIF)
val = snd_ac97_read(ac97, AC97_CSR_SPDIF);
- else
+ else if (ac97->id == AC97_ID_YMF743) {
+ val = snd_ac97_read(ac97, AC97_YMF7X3_DIT_CTRL);
+ val = 0x2000 | (val & 0xff00) >> 4 | (val & 0x38) >> 2;
+ } else
val = snd_ac97_read(ac97, AC97_SPDIF);
snd_iprintf(buffer, "SPDIF Control :%s%s%s%s Category=0x%x Generation=%i%s%s%s\n",
diff --git a/sound/pci/ac97/ak4531_codec.c b/sound/pci/ac97/ak4531_codec.c
index dc26820a03a5..722de451d15f 100644
--- a/sound/pci/ac97/ak4531_codec.c
+++ b/sound/pci/ac97/ak4531_codec.c
@@ -1,5 +1,5 @@
/*
- * Copyright (c) by Jaroslav Kysela <perex@suse.cz>
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
* Universal routines for AK4531 codec
*
*
@@ -29,7 +29,7 @@
#include <sound/ak4531_codec.h>
#include <sound/tlv.h>
-MODULE_AUTHOR("Jaroslav Kysela <perex@suse.cz>");
+MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
MODULE_DESCRIPTION("Universal routines for AK4531 codec");
MODULE_LICENSE("GPL");
diff --git a/sound/pci/ali5451/Makefile b/sound/pci/ali5451/Makefile
index 2e1831597474..713459c12d22 100644
--- a/sound/pci/ali5451/Makefile
+++ b/sound/pci/ali5451/Makefile
@@ -1,6 +1,6 @@
#
# Makefile for ALSA
-# Copyright (c) 2001 by Jaroslav Kysela <perex@suse.cz>
+# Copyright (c) 2001 by Jaroslav Kysela <perex@perex.cz>
#
snd-ali5451-objs := ali5451.o
diff --git a/sound/pci/ali5451/ali5451.c b/sound/pci/ali5451/ali5451.c
index 05b4c8696941..4c2bd7adf674 100644
--- a/sound/pci/ali5451/ali5451.c
+++ b/sound/pci/ali5451/ali5451.c
@@ -1804,15 +1804,7 @@ static int __devinit snd_ali_build_pcms(struct snd_ali *codec)
.info = snd_ali5451_spdif_info, .get = snd_ali5451_spdif_get, \
.put = snd_ali5451_spdif_put, .private_value = value}
-static int snd_ali5451_spdif_info(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_info *uinfo)
-{
- uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
- uinfo->count = 1;
- uinfo->value.integer.min = 0;
- uinfo->value.integer.max = 1;
- return 0;
-}
+#define snd_ali5451_spdif_info snd_ctl_boolean_mono_info
static int snd_ali5451_spdif_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
diff --git a/sound/pci/als4000.c b/sound/pci/als4000.c
index 8fb55d3b454b..1190ef366a41 100644
--- a/sound/pci/als4000.c
+++ b/sound/pci/als4000.c
@@ -1,7 +1,7 @@
/*
* card-als4000.c - driver for Avance Logic ALS4000 based soundcards.
* Copyright (C) 2000 by Bart Hartgers <bart@etpmod.phys.tue.nl>,
- * Jaroslav Kysela <perex@suse.cz>
+ * Jaroslav Kysela <perex@perex.cz>
* Copyright (C) 2002 by Andreas Mohr <hw7oshyuv3001@sneakemail.com>
*
* Framework borrowed from Massimo Piccioni's card-als100.c.
diff --git a/sound/pci/au88x0/au88x0.c b/sound/pci/au88x0/au88x0.c
index 5ec1b6fcd548..f70286a7364a 100644
--- a/sound/pci/au88x0/au88x0.c
+++ b/sound/pci/au88x0/au88x0.c
@@ -232,6 +232,7 @@ snd_vortex_create(struct snd_card *card, struct pci_dev *pci, vortex_t ** rchip)
pci_disable_device(chip->pci_dev);
//FIXME: this not the right place to unregister the gameport
vortex_gameport_unregister(chip);
+ kfree(chip);
return err;
}
diff --git a/sound/pci/au88x0/au88x0_eq.c b/sound/pci/au88x0/au88x0_eq.c
index 0c86a31c4336..38602b85874d 100644
--- a/sound/pci/au88x0/au88x0_eq.c
+++ b/sound/pci/au88x0/au88x0_eq.c
@@ -728,15 +728,7 @@ static void vortex_Eqlzr_shutdown(vortex_t * vortex)
/* ALSA interface */
/* Control interface */
-static int
-snd_vortex_eqtoggle_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
-{
- uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
- uinfo->count = 1;
- uinfo->value.integer.min = 0;
- uinfo->value.integer.max = 1;
- return 0;
-}
+#define snd_vortex_eqtoggle_info snd_ctl_boolean_mono_info
static int
snd_vortex_eqtoggle_get(struct snd_kcontrol *kcontrol,
diff --git a/sound/pci/au88x0/au88x0_mpu401.c b/sound/pci/au88x0/au88x0_mpu401.c
index c75d368ea087..8db3d3e6f7bb 100644
--- a/sound/pci/au88x0/au88x0_mpu401.c
+++ b/sound/pci/au88x0/au88x0_mpu401.c
@@ -1,5 +1,5 @@
/*
- * Copyright (c) by Jaroslav Kysela <perex@suse.cz>
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
* Routines for control of MPU-401 in UART mode
*
* Modified for the Aureal Vortex based Soundcards
diff --git a/sound/pci/au88x0/au88x0_synth.c b/sound/pci/au88x0/au88x0_synth.c
index d3e662a1285d..978b856f5621 100644
--- a/sound/pci/au88x0/au88x0_synth.c
+++ b/sound/pci/au88x0/au88x0_synth.c
@@ -370,8 +370,8 @@ static void vortex_wt_SetFrequency(vortex_t * vortex, int wt, unsigned int sr)
while ((edx & 0x80000000) == 0) {
edx <<= 1;
eax--;
- if (eax == 0) ;
- break;
+ if (eax == 0)
+ break;
}
if (eax)
edx <<= 1;
diff --git a/sound/pci/bt87x.c b/sound/pci/bt87x.c
index 131952f55857..91f9e6a112ff 100644
--- a/sound/pci/bt87x.c
+++ b/sound/pci/bt87x.c
@@ -147,15 +147,56 @@ MODULE_PARM_DESC(load_all, "Allow to load the non-whitelisted cards");
/* SYNC, one WRITE per line, one extra WRITE per page boundary, SYNC, JUMP */
#define MAX_RISC_SIZE ((1 + 255 + (PAGE_ALIGN(255 * 4092) / PAGE_SIZE - 1) + 1 + 1) * 8)
+/* Cards with configuration information */
+enum snd_bt87x_boardid {
+ SND_BT87X_BOARD_UNKNOWN,
+ SND_BT87X_BOARD_GENERIC, /* both an & dig interfaces, 32kHz */
+ SND_BT87X_BOARD_ANALOG, /* board with no external A/D */
+ SND_BT87X_BOARD_OSPREY2x0,
+ SND_BT87X_BOARD_OSPREY440,
+ SND_BT87X_BOARD_AVPHONE98,
+};
+
+/* Card configuration */
+struct snd_bt87x_board {
+ int dig_rate; /* Digital input sampling rate */
+ u32 digital_fmt; /* Register settings for digital input */
+ unsigned no_analog:1; /* No analog input */
+ unsigned no_digital:1; /* No digital input */
+};
+
+static const __devinitdata struct snd_bt87x_board snd_bt87x_boards[] = {
+ [SND_BT87X_BOARD_UNKNOWN] = {
+ .dig_rate = 32000, /* just a guess */
+ },
+ [SND_BT87X_BOARD_GENERIC] = {
+ .dig_rate = 32000,
+ },
+ [SND_BT87X_BOARD_ANALOG] = {
+ .no_digital = 1,
+ },
+ [SND_BT87X_BOARD_OSPREY2x0] = {
+ .dig_rate = 44100,
+ .digital_fmt = CTL_DA_LRI | (1 << CTL_DA_LRD_SHIFT),
+ },
+ [SND_BT87X_BOARD_OSPREY440] = {
+ .dig_rate = 32000,
+ .digital_fmt = CTL_DA_LRI | (1 << CTL_DA_LRD_SHIFT),
+ .no_analog = 1,
+ },
+ [SND_BT87X_BOARD_AVPHONE98] = {
+ .dig_rate = 48000,
+ },
+};
+
struct snd_bt87x {
struct snd_card *card;
struct pci_dev *pci;
+ struct snd_bt87x_board board;
void __iomem *mmio;
int irq;
- int dig_rate;
-
spinlock_t reg_lock;
unsigned long opened;
struct snd_pcm_substream *substream;
@@ -340,30 +381,11 @@ static struct snd_pcm_hardware snd_bt87x_analog_hw = {
static int snd_bt87x_set_digital_hw(struct snd_bt87x *chip, struct snd_pcm_runtime *runtime)
{
- static struct {
- int rate;
- unsigned int bit;
- } ratebits[] = {
- {8000, SNDRV_PCM_RATE_8000},
- {11025, SNDRV_PCM_RATE_11025},
- {16000, SNDRV_PCM_RATE_16000},
- {22050, SNDRV_PCM_RATE_22050},
- {32000, SNDRV_PCM_RATE_32000},
- {44100, SNDRV_PCM_RATE_44100},
- {48000, SNDRV_PCM_RATE_48000}
- };
- int i;
-
- chip->reg_control |= CTL_DA_IOM_DA;
+ chip->reg_control |= CTL_DA_IOM_DA | CTL_A_PWRDN;
runtime->hw = snd_bt87x_digital_hw;
- runtime->hw.rates = SNDRV_PCM_RATE_KNOT;
- for (i = 0; i < ARRAY_SIZE(ratebits); ++i)
- if (chip->dig_rate == ratebits[i].rate) {
- runtime->hw.rates = ratebits[i].bit;
- break;
- }
- runtime->hw.rate_min = chip->dig_rate;
- runtime->hw.rate_max = chip->dig_rate;
+ runtime->hw.rates = snd_pcm_rate_to_rate_bit(chip->board.dig_rate);
+ runtime->hw.rate_min = chip->board.dig_rate;
+ runtime->hw.rate_max = chip->board.dig_rate;
return 0;
}
@@ -380,7 +402,7 @@ static int snd_bt87x_set_analog_hw(struct snd_bt87x *chip, struct snd_pcm_runtim
.rats = &analog_clock
};
- chip->reg_control &= ~CTL_DA_IOM_DA;
+ chip->reg_control &= ~(CTL_DA_IOM_DA | CTL_A_PWRDN);
runtime->hw = snd_bt87x_analog_hw;
return snd_pcm_hw_constraint_ratnums(runtime, 0, SNDRV_PCM_HW_PARAM_RATE,
&constraint_rates);
@@ -419,6 +441,11 @@ static int snd_bt87x_close(struct snd_pcm_substream *substream)
{
struct snd_bt87x *chip = snd_pcm_substream_chip(substream);
+ spin_lock_irq(&chip->reg_lock);
+ chip->reg_control |= CTL_A_PWRDN;
+ snd_bt87x_writel(chip, REG_GPIO_DMA_CTL, chip->reg_control);
+ spin_unlock_irq(&chip->reg_lock);
+
chip->substream = NULL;
clear_bit(0, &chip->opened);
smp_mb__after_clear_bit();
@@ -569,15 +596,7 @@ static struct snd_kcontrol_new snd_bt87x_capture_volume = {
.put = snd_bt87x_capture_volume_put,
};
-static int snd_bt87x_capture_boost_info(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_info *info)
-{
- info->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
- info->count = 1;
- info->value.integer.min = 0;
- info->value.integer.max = 1;
- return 0;
-}
+#define snd_bt87x_capture_boost_info snd_ctl_boolean_mono_info
static int snd_bt87x_capture_boost_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *value)
@@ -736,61 +755,69 @@ static int __devinit snd_bt87x_create(struct snd_card *card,
chip->mmio = ioremap_nocache(pci_resource_start(pci, 0),
pci_resource_len(pci, 0));
if (!chip->mmio) {
- snd_bt87x_free(chip);
snd_printk(KERN_ERR "cannot remap io memory\n");
- return -ENOMEM;
+ err = -ENOMEM;
+ goto fail;
}
- chip->reg_control = CTL_DA_ES2 | CTL_PKTP_16 | (15 << CTL_DA_SDR_SHIFT);
+ chip->reg_control = CTL_A_PWRDN | CTL_DA_ES2 |
+ CTL_PKTP_16 | (15 << CTL_DA_SDR_SHIFT);
chip->interrupt_mask = MY_INTERRUPTS;
snd_bt87x_writel(chip, REG_GPIO_DMA_CTL, chip->reg_control);
snd_bt87x_writel(chip, REG_INT_MASK, 0);
snd_bt87x_writel(chip, REG_INT_STAT, MY_INTERRUPTS);
- if (request_irq(pci->irq, snd_bt87x_interrupt, IRQF_SHARED,
- "Bt87x audio", chip)) {
- snd_bt87x_free(chip);
- snd_printk(KERN_ERR "cannot grab irq\n");
- return -EBUSY;
+ err = request_irq(pci->irq, snd_bt87x_interrupt, IRQF_SHARED,
+ "Bt87x audio", chip);
+ if (err < 0) {
+ snd_printk(KERN_ERR "cannot grab irq %d\n", pci->irq);
+ goto fail;
}
chip->irq = pci->irq;
pci_set_master(pci);
synchronize_irq(chip->irq);
err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops);
- if (err < 0) {
- snd_bt87x_free(chip);
- return err;
- }
+ if (err < 0)
+ goto fail;
+
snd_card_set_dev(card, &pci->dev);
*rchip = chip;
return 0;
+
+fail:
+ snd_bt87x_free(chip);
+ return err;
}
-#define BT_DEVICE(chip, subvend, subdev, rate) \
+#define BT_DEVICE(chip, subvend, subdev, id) \
{ .vendor = PCI_VENDOR_ID_BROOKTREE, \
.device = chip, \
.subvendor = subvend, .subdevice = subdev, \
- .driver_data = rate }
+ .driver_data = SND_BT87X_BOARD_ ## id }
+/* driver_data is the card id for that device */
-/* driver_data is the default digital_rate value for that device */
static struct pci_device_id snd_bt87x_ids[] = {
/* Hauppauge WinTV series */
- BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, 0x0070, 0x13eb, 32000),
+ BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, 0x0070, 0x13eb, GENERIC),
/* Hauppauge WinTV series */
- BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_879, 0x0070, 0x13eb, 32000),
+ BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_879, 0x0070, 0x13eb, GENERIC),
/* Viewcast Osprey 200 */
- BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, 0x0070, 0xff01, 44100),
+ BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, 0x0070, 0xff01, OSPREY2x0),
/* Viewcast Osprey 440 (rate is configurable via gpio) */
- BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, 0x0070, 0xff07, 32000),
+ BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, 0x0070, 0xff07, OSPREY440),
/* ATI TV-Wonder */
- BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, 0x1002, 0x0001, 32000),
+ BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, 0x1002, 0x0001, GENERIC),
/* Leadtek Winfast tv 2000xp delux */
- BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, 0x107d, 0x6606, 32000),
+ BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, 0x107d, 0x6606, GENERIC),
/* Voodoo TV 200 */
- BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, 0x121a, 0x3000, 32000),
+ BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, 0x121a, 0x3000, GENERIC),
/* AVerMedia Studio No. 103, 203, ...? */
- BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, 0x1461, 0x0003, 48000),
+ BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, 0x1461, 0x0003, AVPHONE98),
+ /* Prolink PixelView PV-M4900 */
+ BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, 0x1554, 0x4011, GENERIC),
+ /* Pinnacle Studio PCTV rave */
+ BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, 0xbd11, 0x1200, GENERIC),
{ }
};
MODULE_DEVICE_TABLE(pci, snd_bt87x_ids);
@@ -815,7 +842,7 @@ static struct {
static struct pci_driver driver;
-/* return the rate of the card, or a negative value if it's blacklisted */
+/* return the id of the card, or a negative value if it's blacklisted */
static int __devinit snd_bt87x_detect_card(struct pci_dev *pci)
{
int i;
@@ -833,12 +860,12 @@ static int __devinit snd_bt87x_detect_card(struct pci_dev *pci)
return -EBUSY;
}
- snd_printk(KERN_INFO "unknown card %#04x-%#04x:%#04x, using default rate 32000\n",
- pci->device, pci->subsystem_vendor, pci->subsystem_device);
+ snd_printk(KERN_INFO "unknown card %#04x-%#04x:%#04x\n",
+ pci->device, pci->subsystem_vendor, pci->subsystem_device);
snd_printk(KERN_DEBUG "please mail id, board name, and, "
"if it works, the correct digital_rate option to "
"<alsa-devel@alsa-project.org>\n");
- return 32000; /* default rate */
+ return SND_BT87X_BOARD_UNKNOWN;
}
static int __devinit snd_bt87x_probe(struct pci_dev *pci,
@@ -847,12 +874,16 @@ static int __devinit snd_bt87x_probe(struct pci_dev *pci,
static int dev;
struct snd_card *card;
struct snd_bt87x *chip;
- int err, rate;
+ int err;
+ enum snd_bt87x_boardid boardid;
- rate = pci_id->driver_data;
- if (! rate)
- if ((rate = snd_bt87x_detect_card(pci)) <= 0)
+ if (!pci_id->driver_data) {
+ err = snd_bt87x_detect_card(pci);
+ if (err < 0)
return -ENODEV;
+ boardid = err;
+ } else
+ boardid = pci_id->driver_data;
if (dev >= SNDRV_CARDS)
return -ENODEV;
@@ -869,27 +900,39 @@ static int __devinit snd_bt87x_probe(struct pci_dev *pci,
if (err < 0)
goto _error;
- if (digital_rate[dev] > 0)
- chip->dig_rate = digital_rate[dev];
- else
- chip->dig_rate = rate;
+ memcpy(&chip->board, &snd_bt87x_boards[boardid], sizeof(chip->board));
- err = snd_bt87x_pcm(chip, DEVICE_DIGITAL, "Bt87x Digital");
- if (err < 0)
- goto _error;
- err = snd_bt87x_pcm(chip, DEVICE_ANALOG, "Bt87x Analog");
- if (err < 0)
- goto _error;
+ if (!chip->board.no_digital) {
+ if (digital_rate[dev] > 0)
+ chip->board.dig_rate = digital_rate[dev];
- err = snd_ctl_add(card, snd_ctl_new1(&snd_bt87x_capture_volume, chip));
- if (err < 0)
- goto _error;
- err = snd_ctl_add(card, snd_ctl_new1(&snd_bt87x_capture_boost, chip));
- if (err < 0)
- goto _error;
- err = snd_ctl_add(card, snd_ctl_new1(&snd_bt87x_capture_source, chip));
- if (err < 0)
- goto _error;
+ chip->reg_control |= chip->board.digital_fmt;
+
+ err = snd_bt87x_pcm(chip, DEVICE_DIGITAL, "Bt87x Digital");
+ if (err < 0)
+ goto _error;
+ }
+ if (!chip->board.no_analog) {
+ err = snd_bt87x_pcm(chip, DEVICE_ANALOG, "Bt87x Analog");
+ if (err < 0)
+ goto _error;
+ err = snd_ctl_add(card, snd_ctl_new1(
+ &snd_bt87x_capture_volume, chip));
+ if (err < 0)
+ goto _error;
+ err = snd_ctl_add(card, snd_ctl_new1(
+ &snd_bt87x_capture_boost, chip));
+ if (err < 0)
+ goto _error;
+ err = snd_ctl_add(card, snd_ctl_new1(
+ &snd_bt87x_capture_source, chip));
+ if (err < 0)
+ goto _error;
+ }
+ snd_printk(KERN_INFO "bt87x%d: Using board %d, %sanalog, %sdigital "
+ "(rate %d Hz)\n", dev, boardid,
+ chip->board.no_analog ? "no " : "",
+ chip->board.no_digital ? "no " : "", chip->board.dig_rate);
strcpy(card->driver, "Bt87x");
sprintf(card->shortname, "Brooktree Bt%x", pci->device);
@@ -920,8 +963,8 @@ static void __devexit snd_bt87x_remove(struct pci_dev *pci)
/* default entries for all Bt87x cards - it's not exported */
/* driver_data is set to 0 to call detection */
static struct pci_device_id snd_bt87x_default_ids[] __devinitdata = {
- BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, PCI_ANY_ID, PCI_ANY_ID, 0),
- BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_879, PCI_ANY_ID, PCI_ANY_ID, 0),
+ BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, PCI_ANY_ID, PCI_ANY_ID, UNKNOWN),
+ BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_879, PCI_ANY_ID, PCI_ANY_ID, UNKNOWN),
{ }
};
diff --git a/sound/pci/ca0106/ca0106.h b/sound/pci/ca0106/ca0106.h
index a0420bc63f0b..75da1746e758 100644
--- a/sound/pci/ca0106/ca0106.h
+++ b/sound/pci/ca0106/ca0106.h
@@ -1,7 +1,7 @@
/*
* Copyright (c) 2004 James Courtier-Dutton <James@superbug.demon.co.uk>
* Driver CA0106 chips. e.g. Sound Blaster Audigy LS and Live 24bit
- * Version: 0.0.21
+ * Version: 0.0.22
*
* FEATURES currently supported:
* See ca0106_main.c for features.
@@ -47,6 +47,8 @@
* Added GPIO info for SB Live 24bit.
* 0.0.21
* Implement support for Line-in capture on SB Live 24bit.
+ * 0.0.22
+ * Add support for mute control on SB Live 24bit (cards w/ SPI DAC)
*
*
* This code was initally based on code from ALSA's emu10k1x.c which is:
@@ -552,6 +554,95 @@
#define CONTROL_CENTER_LFE_CHANNEL 1
#define CONTROL_UNKNOWN_CHANNEL 2
+
+/* Based on WM8768 Datasheet Rev 4.2 page 32 */
+#define SPI_REG_MASK 0x1ff /* 16-bit SPI writes have a 7-bit address */
+#define SPI_REG_SHIFT 9 /* followed by 9 bits of data */
+
+#define SPI_LDA1_REG 0 /* digital attenuation */
+#define SPI_RDA1_REG 1
+#define SPI_LDA2_REG 4
+#define SPI_RDA2_REG 5
+#define SPI_LDA3_REG 6
+#define SPI_RDA3_REG 7
+#define SPI_LDA4_REG 13
+#define SPI_RDA4_REG 14
+#define SPI_MASTDA_REG 8
+
+#define SPI_DA_BIT_UPDATE (1<<8) /* update attenuation values */
+#define SPI_DA_BIT_0dB 0xff /* 0 dB */
+#define SPI_DA_BIT_infdB 0x00 /* inf dB attenuation (mute) */
+
+#define SPI_PL_REG 2
+#define SPI_PL_BIT_L_M (0<<5) /* left channel = mute */
+#define SPI_PL_BIT_L_L (1<<5) /* left channel = left */
+#define SPI_PL_BIT_L_R (2<<5) /* left channel = right */
+#define SPI_PL_BIT_L_C (3<<5) /* left channel = (L+R)/2 */
+#define SPI_PL_BIT_R_M (0<<7) /* right channel = mute */
+#define SPI_PL_BIT_R_L (1<<7) /* right channel = left */
+#define SPI_PL_BIT_R_R (2<<7) /* right channel = right */
+#define SPI_PL_BIT_R_C (3<<7) /* right channel = (L+R)/2 */
+#define SPI_IZD_REG 2
+#define SPI_IZD_BIT (1<<4) /* infinite zero detect */
+
+#define SPI_FMT_REG 3
+#define SPI_FMT_BIT_RJ (0<<0) /* right justified mode */
+#define SPI_FMT_BIT_LJ (1<<0) /* left justified mode */
+#define SPI_FMT_BIT_I2S (2<<0) /* I2S mode */
+#define SPI_FMT_BIT_DSP (3<<0) /* DSP Modes A or B */
+#define SPI_LRP_REG 3
+#define SPI_LRP_BIT (1<<2) /* invert LRCLK polarity */
+#define SPI_BCP_REG 3
+#define SPI_BCP_BIT (1<<3) /* invert BCLK polarity */
+#define SPI_IWL_REG 3
+#define SPI_IWL_BIT_16 (0<<4) /* 16-bit world length */
+#define SPI_IWL_BIT_20 (1<<4) /* 20-bit world length */
+#define SPI_IWL_BIT_24 (2<<4) /* 24-bit world length */
+#define SPI_IWL_BIT_32 (3<<4) /* 32-bit world length */
+
+#define SPI_MS_REG 10
+#define SPI_MS_BIT (1<<5) /* master mode */
+#define SPI_RATE_REG 10 /* only applies in master mode */
+#define SPI_RATE_BIT_128 (0<<6) /* MCLK = LRCLK * 128 */
+#define SPI_RATE_BIT_192 (1<<6)
+#define SPI_RATE_BIT_256 (2<<6)
+#define SPI_RATE_BIT_384 (3<<6)
+#define SPI_RATE_BIT_512 (4<<6)
+#define SPI_RATE_BIT_768 (5<<6)
+
+/* They really do label the bit for the 4th channel "4" and not "3" */
+#define SPI_DMUTE0_REG 9
+#define SPI_DMUTE1_REG 9
+#define SPI_DMUTE2_REG 9
+#define SPI_DMUTE4_REG 15
+#define SPI_DMUTE0_BIT (1<<3)
+#define SPI_DMUTE1_BIT (1<<4)
+#define SPI_DMUTE2_BIT (1<<5)
+#define SPI_DMUTE4_BIT (1<<2)
+
+#define SPI_PHASE0_REG 3
+#define SPI_PHASE1_REG 3
+#define SPI_PHASE2_REG 3
+#define SPI_PHASE4_REG 15
+#define SPI_PHASE0_BIT (1<<6)
+#define SPI_PHASE1_BIT (1<<7)
+#define SPI_PHASE2_BIT (1<<8)
+#define SPI_PHASE4_BIT (1<<3)
+
+#define SPI_PDWN_REG 2 /* power down all DACs */
+#define SPI_PDWN_BIT (1<<2)
+#define SPI_DACD0_REG 10 /* power down individual DACs */
+#define SPI_DACD1_REG 10
+#define SPI_DACD2_REG 10
+#define SPI_DACD4_REG 15
+#define SPI_DACD0_BIT (1<<1)
+#define SPI_DACD1_BIT (1<<2)
+#define SPI_DACD2_BIT (1<<3)
+#define SPI_DACD4_BIT (1<<0) /* datasheet error says it's 1 */
+
+#define SPI_PWRDNALL_REG 10 /* power down everything */
+#define SPI_PWRDNALL_BIT (1<<4)
+
#include "ca_midi.h"
struct snd_ca0106;
@@ -611,6 +702,8 @@ struct snd_ca0106 {
struct snd_ca_midi midi;
struct snd_ca_midi midi2;
+
+ u16 spi_dac_reg[16];
};
int snd_ca0106_mixer(struct snd_ca0106 *emu);
@@ -627,4 +720,5 @@ void snd_ca0106_ptr_write(struct snd_ca0106 *emu,
int snd_ca0106_i2c_write(struct snd_ca0106 *emu, u32 reg, u32 value);
-
+int snd_ca0106_spi_write(struct snd_ca0106 * emu,
+ unsigned int data);
diff --git a/sound/pci/ca0106/ca0106_main.c b/sound/pci/ca0106/ca0106_main.c
index fcab8fb97e38..31d8db9f7a4c 100644
--- a/sound/pci/ca0106/ca0106_main.c
+++ b/sound/pci/ca0106/ca0106_main.c
@@ -1,7 +1,7 @@
/*
* Copyright (c) 2004 James Courtier-Dutton <James@superbug.demon.co.uk>
* Driver CA0106 chips. e.g. Sound Blaster Audigy LS and Live 24bit
- * Version: 0.0.23
+ * Version: 0.0.25
*
* FEATURES currently supported:
* Front, Rear and Center/LFE.
@@ -79,6 +79,10 @@
* Add support for MSI K8N Diamond Motherboard with onboard SB Live 24bit without AC97. From kiksen, bug #901
* 0.0.23
* Implement support for Line-in capture on SB Live 24bit.
+ * 0.0.24
+ * Add support for mute control on SB Live 24bit (cards w/ SPI DAC)
+ * 0.0.25
+ * Powerdown SPI DAC channels when not in use
*
* BUGS:
* Some stability problems when unloading the snd-ca0106 kernel module.
@@ -170,6 +174,15 @@ MODULE_PARM_DESC(subsystem, "Force card subsystem model.");
static struct snd_ca0106_details ca0106_chip_details[] = {
/* Sound Blaster X-Fi Extreme Audio. This does not have an AC97. 53SB079000000 */
/* It is really just a normal SB Live 24bit. */
+ /* Tested:
+ * See ALSA bug#3251
+ */
+ { .serial = 0x10131102,
+ .name = "X-Fi Extreme Audio [SBxxxx]",
+ .gpio_type = 1,
+ .i2c_adc = 1 } ,
+ /* Sound Blaster X-Fi Extreme Audio. This does not have an AC97. 53SB079000000 */
+ /* It is really just a normal SB Live 24bit. */
/*
* CTRL:CA0111-WTLF
* ADC: WM8775SEDS
@@ -261,10 +274,11 @@ static struct snd_ca0106_details ca0106_chip_details[] = {
/* hardware definition */
static struct snd_pcm_hardware snd_ca0106_playback_hw = {
- .info = (SNDRV_PCM_INFO_MMAP |
- SNDRV_PCM_INFO_INTERLEAVED |
- SNDRV_PCM_INFO_BLOCK_TRANSFER |
- SNDRV_PCM_INFO_MMAP_VALID),
+ .info = SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_SYNC_START,
.formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE,
.rates = (SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000 |
SNDRV_PCM_RATE_192000),
@@ -447,6 +461,19 @@ static void snd_ca0106_pcm_free_substream(struct snd_pcm_runtime *runtime)
kfree(runtime->private_data);
}
+static const int spi_dacd_reg[] = {
+ [PCM_FRONT_CHANNEL] = SPI_DACD4_REG,
+ [PCM_REAR_CHANNEL] = SPI_DACD0_REG,
+ [PCM_CENTER_LFE_CHANNEL]= SPI_DACD2_REG,
+ [PCM_UNKNOWN_CHANNEL] = SPI_DACD1_REG,
+};
+static const int spi_dacd_bit[] = {
+ [PCM_FRONT_CHANNEL] = SPI_DACD4_BIT,
+ [PCM_REAR_CHANNEL] = SPI_DACD0_BIT,
+ [PCM_CENTER_LFE_CHANNEL]= SPI_DACD2_BIT,
+ [PCM_UNKNOWN_CHANNEL] = SPI_DACD1_BIT,
+};
+
/* open_playback callback */
static int snd_ca0106_pcm_open_playback_channel(struct snd_pcm_substream *substream,
int channel_id)
@@ -481,6 +508,17 @@ static int snd_ca0106_pcm_open_playback_channel(struct snd_pcm_substream *substr
return err;
if ((err = snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 64)) < 0)
return err;
+ snd_pcm_set_sync(substream);
+
+ if (chip->details->spi_dac && channel_id != PCM_FRONT_CHANNEL) {
+ const int reg = spi_dacd_reg[channel_id];
+
+ /* Power up dac */
+ chip->spi_dac_reg[reg] &= ~spi_dacd_bit[channel_id];
+ err = snd_ca0106_spi_write(chip, chip->spi_dac_reg[reg]);
+ if (err < 0)
+ return err;
+ }
return 0;
}
@@ -491,6 +529,14 @@ static int snd_ca0106_pcm_close_playback(struct snd_pcm_substream *substream)
struct snd_pcm_runtime *runtime = substream->runtime;
struct snd_ca0106_pcm *epcm = runtime->private_data;
chip->playback_channels[epcm->channel_id].use = 0;
+
+ if (chip->details->spi_dac && epcm->channel_id != PCM_FRONT_CHANNEL) {
+ const int reg = spi_dacd_reg[epcm->channel_id];
+
+ /* Power down DAC */
+ chip->spi_dac_reg[reg] |= spi_dacd_bit[epcm->channel_id];
+ snd_ca0106_spi_write(chip, chip->spi_dac_reg[reg]);
+ }
/* FIXME: maybe zero others */
return 0;
}
@@ -809,6 +855,9 @@ static int snd_ca0106_pcm_trigger_playback(struct snd_pcm_substream *substream,
break;
}
snd_pcm_group_for_each_entry(s, substream) {
+ if (snd_pcm_substream_chip(s) != emu ||
+ s->stream != SNDRV_PCM_STREAM_PLAYBACK)
+ continue;
runtime = s->runtime;
epcm = runtime->private_data;
channel = epcm->channel_id;
@@ -1214,28 +1263,23 @@ static int __devinit snd_ca0106_pcm(struct snd_ca0106 *emu, int device, struct s
return 0;
}
+#define SPI_REG(reg, value) (((reg) << SPI_REG_SHIFT) | (value))
static unsigned int spi_dac_init[] = {
- 0x00ff,
- 0x02ff,
- 0x0400,
- 0x0520,
- 0x0620, /* Set 24 bit. Was 0x0600 */
- 0x08ff,
- 0x0aff,
- 0x0cff,
- 0x0eff,
- 0x10ff,
- 0x1200,
- 0x1400,
- 0x1480,
- 0x1800,
- 0x1aff,
- 0x1cff,
- 0x1e00,
- 0x0530,
- 0x0602,
- 0x0622,
- 0x1400,
+ SPI_REG(SPI_LDA1_REG, SPI_DA_BIT_0dB), /* 0dB dig. attenuation */
+ SPI_REG(SPI_RDA1_REG, SPI_DA_BIT_0dB),
+ SPI_REG(SPI_PL_REG, SPI_PL_BIT_L_L | SPI_PL_BIT_R_R | SPI_IZD_BIT),
+ SPI_REG(SPI_FMT_REG, SPI_FMT_BIT_I2S | SPI_IWL_BIT_24),
+ SPI_REG(SPI_LDA2_REG, SPI_DA_BIT_0dB),
+ SPI_REG(SPI_RDA2_REG, SPI_DA_BIT_0dB),
+ SPI_REG(SPI_LDA3_REG, SPI_DA_BIT_0dB),
+ SPI_REG(SPI_RDA3_REG, SPI_DA_BIT_0dB),
+ SPI_REG(SPI_MASTDA_REG, SPI_DA_BIT_0dB),
+ SPI_REG(9, 0x00),
+ SPI_REG(SPI_MS_REG, SPI_DACD0_BIT | SPI_DACD1_BIT | SPI_DACD2_BIT),
+ SPI_REG(12, 0x00),
+ SPI_REG(SPI_LDA4_REG, SPI_DA_BIT_0dB),
+ SPI_REG(SPI_RDA4_REG, SPI_DA_BIT_0dB | SPI_DA_BIT_UPDATE),
+ SPI_REG(SPI_DACD4_REG, 0x00),
};
static unsigned int i2c_adc_init[][2] = {
@@ -1475,8 +1519,13 @@ static int __devinit snd_ca0106_create(int dev, struct snd_card *card,
int size, n;
size = ARRAY_SIZE(spi_dac_init);
- for (n=0; n < size; n++)
+ for (n = 0; n < size; n++) {
+ int reg = spi_dac_init[n] >> SPI_REG_SHIFT;
+
snd_ca0106_spi_write(chip, spi_dac_init[n]);
+ if (reg < ARRAY_SIZE(chip->spi_dac_reg))
+ chip->spi_dac_reg[reg] = spi_dac_init[n];
+ }
}
if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL,
diff --git a/sound/pci/ca0106/ca0106_mixer.c b/sound/pci/ca0106/ca0106_mixer.c
index 9c3a9c8d1dc2..be519a17dfa5 100644
--- a/sound/pci/ca0106/ca0106_mixer.c
+++ b/sound/pci/ca0106/ca0106_mixer.c
@@ -1,7 +1,7 @@
/*
* Copyright (c) 2004 James Courtier-Dutton <James@superbug.demon.co.uk>
* Driver CA0106 chips. e.g. Sound Blaster Audigy LS and Live 24bit
- * Version: 0.0.17
+ * Version: 0.0.18
*
* FEATURES currently supported:
* See ca0106_main.c for features.
@@ -39,6 +39,8 @@
* Modified Copyright message.
* 0.0.17
* Implement Mic and Line in Capture.
+ * 0.0.18
+ * Add support for mute control on SB Live 24bit (cards w/ SPI DAC)
*
* This code was initally based on code from ALSA's emu10k1x.c which is:
* Copyright (c) by Francisco Moraes <fmoraes@nc.rr.com>
@@ -77,15 +79,7 @@
static const DECLARE_TLV_DB_SCALE(snd_ca0106_db_scale1, -5175, 25, 1);
static const DECLARE_TLV_DB_SCALE(snd_ca0106_db_scale2, -10350, 50, 1);
-static int snd_ca0106_shared_spdif_info(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_info *uinfo)
-{
- uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
- uinfo->count = 1;
- uinfo->value.integer.min = 0;
- uinfo->value.integer.max = 1;
- return 0;
-}
+#define snd_ca0106_shared_spdif_info snd_ctl_boolean_mono_info
static int snd_ca0106_shared_spdif_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
@@ -470,6 +464,42 @@ static int snd_ca0106_i2c_volume_put(struct snd_kcontrol *kcontrol,
return change;
}
+#define spi_mute_info snd_ctl_boolean_mono_info
+
+static int spi_mute_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ca0106 *emu = snd_kcontrol_chip(kcontrol);
+ unsigned int reg = kcontrol->private_value >> SPI_REG_SHIFT;
+ unsigned int bit = kcontrol->private_value & SPI_REG_MASK;
+
+ ucontrol->value.integer.value[0] = !(emu->spi_dac_reg[reg] & bit);
+ return 0;
+}
+
+static int spi_mute_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_ca0106 *emu = snd_kcontrol_chip(kcontrol);
+ unsigned int reg = kcontrol->private_value >> SPI_REG_SHIFT;
+ unsigned int bit = kcontrol->private_value & SPI_REG_MASK;
+ int ret;
+
+ ret = emu->spi_dac_reg[reg] & bit;
+ if (ucontrol->value.integer.value[0]) {
+ if (!ret) /* bit already cleared, do nothing */
+ return 0;
+ emu->spi_dac_reg[reg] &= ~bit;
+ } else {
+ if (ret) /* bit already set, do nothing */
+ return 0;
+ emu->spi_dac_reg[reg] |= bit;
+ }
+
+ ret = snd_ca0106_spi_write(emu, emu->spi_dac_reg[reg]);
+ return ret ? -1 : 1;
+}
+
#define CA_VOLUME(xname,chid,reg) \
{ \
.iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
@@ -562,6 +592,28 @@ static struct snd_kcontrol_new snd_ca0106_volume_i2c_adc_ctls[] __devinitdata =
I2C_VOLUME("Aux Capture Volume", 3),
};
+#define SPI_SWITCH(xname,reg,bit) \
+{ \
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, \
+ .info = spi_mute_info, \
+ .get = spi_mute_get, \
+ .put = spi_mute_put, \
+ .private_value = (reg<<SPI_REG_SHIFT) | (bit) \
+}
+
+static struct snd_kcontrol_new snd_ca0106_volume_spi_dac_ctls[]
+__devinitdata = {
+ SPI_SWITCH("Analog Front Playback Switch",
+ SPI_DMUTE4_REG, SPI_DMUTE4_BIT),
+ SPI_SWITCH("Analog Rear Playback Switch",
+ SPI_DMUTE0_REG, SPI_DMUTE0_BIT),
+ SPI_SWITCH("Analog Center/LFE Playback Switch",
+ SPI_DMUTE2_REG, SPI_DMUTE2_BIT),
+ SPI_SWITCH("Analog Side Playback Switch",
+ SPI_DMUTE1_REG, SPI_DMUTE1_BIT),
+};
+
static int __devinit remove_ctl(struct snd_card *card, const char *name)
{
struct snd_ctl_elem_id id;
@@ -591,9 +643,19 @@ static int __devinit rename_ctl(struct snd_card *card, const char *src, const ch
return -ENOENT;
}
+#define ADD_CTLS(emu, ctls) \
+ do { \
+ int i, err; \
+ for (i = 0; i < ARRAY_SIZE(ctls); i++) { \
+ err = snd_ctl_add(card, snd_ctl_new1(&ctls[i], emu)); \
+ if (err < 0) \
+ return err; \
+ } \
+ } while (0)
+
int __devinit snd_ca0106_mixer(struct snd_ca0106 *emu)
{
- int i, err;
+ int err;
struct snd_card *card = emu->card;
char **c;
static char *ca0106_remove_ctls[] = {
@@ -640,17 +702,9 @@ int __devinit snd_ca0106_mixer(struct snd_ca0106 *emu)
rename_ctl(card, c[0], c[1]);
#endif
- for (i = 0; i < ARRAY_SIZE(snd_ca0106_volume_ctls); i++) {
- err = snd_ctl_add(card, snd_ctl_new1(&snd_ca0106_volume_ctls[i], emu));
- if (err < 0)
- return err;
- }
+ ADD_CTLS(emu, snd_ca0106_volume_ctls);
if (emu->details->i2c_adc == 1) {
- for (i = 0; i < ARRAY_SIZE(snd_ca0106_volume_i2c_adc_ctls); i++) {
- err = snd_ctl_add(card, snd_ctl_new1(&snd_ca0106_volume_i2c_adc_ctls[i], emu));
- if (err < 0)
- return err;
- }
+ ADD_CTLS(emu, snd_ca0106_volume_i2c_adc_ctls);
if (emu->details->gpio_type == 1)
err = snd_ctl_add(card, snd_ctl_new1(&snd_ca0106_capture_mic_line_in, emu));
else /* gpio_type == 2 */
@@ -658,6 +712,8 @@ int __devinit snd_ca0106_mixer(struct snd_ca0106 *emu)
if (err < 0)
return err;
}
+ if (emu->details->spi_dac == 1)
+ ADD_CTLS(emu, snd_ca0106_volume_spi_dac_ctls);
return 0;
}
diff --git a/sound/pci/ca0106/ca_midi.c b/sound/pci/ca0106/ca_midi.c
index 2e6eab1f1189..ad32eff2713f 100644
--- a/sound/pci/ca0106/ca_midi.c
+++ b/sound/pci/ca0106/ca_midi.c
@@ -6,7 +6,7 @@
* Changelog:
* Implementation is based on mpu401 and emu10k1x and
* tested with ca0106.
- * mpu401: Copyright (c) by Jaroslav Kysela <perex@suse.cz>
+ * mpu401: Copyright (c) by Jaroslav Kysela <perex@perex.cz>
* emu10k1x: Copyright (c) by Francisco Moraes <fmoraes@nc.rr.com>
*
* This program is free software; you can redistribute it and/or modify
diff --git a/sound/pci/ca0106/ca_midi.h b/sound/pci/ca0106/ca_midi.h
index b72c0933bd22..922ed3e3731e 100644
--- a/sound/pci/ca0106/ca_midi.h
+++ b/sound/pci/ca0106/ca_midi.h
@@ -22,9 +22,9 @@
*
*/
-#include<linux/spinlock.h>
-#include<sound/rawmidi.h>
-#include<sound/mpu401.h>
+#include <linux/spinlock.h>
+#include <sound/rawmidi.h>
+#include <sound/mpu401.h>
#define CA_MIDI_MODE_INPUT MPU401_MODE_INPUT
#define CA_MIDI_MODE_OUTPUT MPU401_MODE_OUTPUT
diff --git a/sound/pci/cmipci.c b/sound/pci/cmipci.c
index 7d3c5ee0005c..6832649879ce 100644
--- a/sound/pci/cmipci.c
+++ b/sound/pci/cmipci.c
@@ -95,30 +95,34 @@ MODULE_PARM_DESC(joystick_port, "Joystick port address.");
#define CM_CHADC0 0x00000001 /* ch0, 0:playback, 1:record */
#define CM_REG_FUNCTRL1 0x04
-#define CM_ASFC_MASK 0x0000E000 /* ADC sampling frequency */
-#define CM_ASFC_SHIFT 13
-#define CM_DSFC_MASK 0x00001C00 /* DAC sampling frequency */
-#define CM_DSFC_SHIFT 10
+#define CM_DSFC_MASK 0x0000E000 /* channel 1 (DAC?) sampling frequency */
+#define CM_DSFC_SHIFT 13
+#define CM_ASFC_MASK 0x00001C00 /* channel 0 (ADC?) sampling frequency */
+#define CM_ASFC_SHIFT 10
#define CM_SPDF_1 0x00000200 /* SPDIF IN/OUT at channel B */
#define CM_SPDF_0 0x00000100 /* SPDIF OUT only channel A */
-#define CM_SPDFLOOP 0x00000080 /* ext. SPDIIF/OUT -> IN loopback */
+#define CM_SPDFLOOP 0x00000080 /* ext. SPDIIF/IN -> OUT loopback */
#define CM_SPDO2DAC 0x00000040 /* SPDIF/OUT can be heard from internal DAC */
#define CM_INTRM 0x00000020 /* master control block (MCB) interrupt enabled */
#define CM_BREQ 0x00000010 /* bus master enabled */
#define CM_VOICE_EN 0x00000008 /* legacy voice (SB16,FM) */
-#define CM_UART_EN 0x00000004 /* UART */
-#define CM_JYSTK_EN 0x00000002 /* joy stick */
+#define CM_UART_EN 0x00000004 /* legacy UART */
+#define CM_JYSTK_EN 0x00000002 /* legacy joystick */
+#define CM_ZVPORT 0x00000001 /* ZVPORT */
#define CM_REG_CHFORMAT 0x08
#define CM_CHB3D5C 0x80000000 /* 5,6 channels */
+#define CM_FMOFFSET2 0x40000000 /* initial FM PCM offset 2 when Fmute=1 */
#define CM_CHB3D 0x20000000 /* 4 channels */
#define CM_CHIP_MASK1 0x1f000000
#define CM_CHIP_037 0x01000000
-
-#define CM_SPDIF_SELECT1 0x00080000 /* for model <= 037 ? */
+#define CM_SETLAT48 0x00800000 /* set latency timer 48h */
+#define CM_EDGEIRQ 0x00400000 /* emulated edge trigger legacy IRQ */
+#define CM_SPD24SEL39 0x00200000 /* 24-bit spdif: model 039 */
#define CM_AC3EN1 0x00100000 /* enable AC3: model 037 */
+#define CM_SPDIF_SELECT1 0x00080000 /* for model <= 037 ? */
#define CM_SPD24SEL 0x00020000 /* 24bit spdif: model 037 */
/* #define CM_SPDIF_INVERSE 0x00010000 */ /* ??? */
@@ -128,35 +132,45 @@ MODULE_PARM_DESC(joystick_port, "Joystick port address.");
#define CM_ADCBITLEN_14 0x00008000
#define CM_ADCBITLEN_13 0x0000C000
-#define CM_ADCDACLEN_MASK 0x00003000
+#define CM_ADCDACLEN_MASK 0x00003000 /* model 037 */
#define CM_ADCDACLEN_060 0x00000000
#define CM_ADCDACLEN_066 0x00001000
#define CM_ADCDACLEN_130 0x00002000
#define CM_ADCDACLEN_280 0x00003000
+#define CM_ADCDLEN_MASK 0x00003000 /* model 039 */
+#define CM_ADCDLEN_ORIGINAL 0x00000000
+#define CM_ADCDLEN_EXTRA 0x00001000
+#define CM_ADCDLEN_24K 0x00002000
+#define CM_ADCDLEN_WEIGHT 0x00003000
+
#define CM_CH1_SRATE_176K 0x00000800
+#define CM_CH1_SRATE_96K 0x00000800 /* model 055? */
#define CM_CH1_SRATE_88K 0x00000400
#define CM_CH0_SRATE_176K 0x00000200
+#define CM_CH0_SRATE_96K 0x00000200 /* model 055? */
#define CM_CH0_SRATE_88K 0x00000100
#define CM_SPDIF_INVERSE2 0x00000080 /* model 055? */
+#define CM_DBLSPDS 0x00000040 /* double SPDIF sample rate 88.2/96 */
+#define CM_POLVALID 0x00000020 /* inverse SPDIF/IN valid bit */
+#define CM_SPDLOCKED 0x00000010
-#define CM_CH1FMT_MASK 0x0000000C
+#define CM_CH1FMT_MASK 0x0000000C /* bit 3: 16 bits, bit 2: stereo */
#define CM_CH1FMT_SHIFT 2
-#define CM_CH0FMT_MASK 0x00000003
+#define CM_CH0FMT_MASK 0x00000003 /* bit 1: 16 bits, bit 0: stereo */
#define CM_CH0FMT_SHIFT 0
#define CM_REG_INT_HLDCLR 0x0C
#define CM_CHIP_MASK2 0xff000000
+#define CM_CHIP_8768 0x20000000
+#define CM_CHIP_055 0x08000000
#define CM_CHIP_039 0x04000000
#define CM_CHIP_039_6CH 0x01000000
-#define CM_CHIP_055 0x08000000
-#define CM_CHIP_8768 0x20000000
+#define CM_UNKNOWN_INT_EN 0x00080000 /* ? */
#define CM_TDMA_INT_EN 0x00040000
#define CM_CH1_INT_EN 0x00020000
#define CM_CH0_INT_EN 0x00010000
-#define CM_INT_HOLD 0x00000002
-#define CM_INT_CLEAR 0x00000001
#define CM_REG_INT_STATUS 0x10
#define CM_INTR 0x80000000
@@ -175,12 +189,13 @@ MODULE_PARM_DESC(joystick_port, "Joystick port address.");
#define CM_CHINT0 0x00000001
#define CM_REG_LEGACY_CTRL 0x14
-#define CM_NXCHG 0x80000000 /* h/w multi channels? */
+#define CM_NXCHG 0x80000000 /* don't map base reg dword->sample */
#define CM_VMPU_MASK 0x60000000 /* MPU401 i/o port address */
#define CM_VMPU_330 0x00000000
#define CM_VMPU_320 0x20000000
#define CM_VMPU_310 0x40000000
#define CM_VMPU_300 0x60000000
+#define CM_ENWR8237 0x10000000 /* enable bus master to write 8237 base reg */
#define CM_VSBSEL_MASK 0x0C000000 /* SB16 base address */
#define CM_VSBSEL_220 0x00000000
#define CM_VSBSEL_240 0x04000000
@@ -191,44 +206,74 @@ MODULE_PARM_DESC(joystick_port, "Joystick port address.");
#define CM_FMSEL_3C8 0x01000000
#define CM_FMSEL_3E0 0x02000000
#define CM_FMSEL_3E8 0x03000000
-#define CM_ENSPDOUT 0x00800000 /* enable XPDIF/OUT to I/O interface */
-#define CM_SPDCOPYRHT 0x00400000 /* set copyright spdif in/out */
+#define CM_ENSPDOUT 0x00800000 /* enable XSPDIF/OUT to I/O interface */
+#define CM_SPDCOPYRHT 0x00400000 /* spdif in/out copyright bit */
#define CM_DAC2SPDO 0x00200000 /* enable wave+fm_midi -> SPDIF/OUT */
-#define CM_SETRETRY 0x00010000 /* 0: legacy i/o wait (default), 1: legacy i/o bus retry */
+#define CM_INVIDWEN 0x00100000 /* internal vendor ID write enable, model 039? */
+#define CM_SETRETRY 0x00100000 /* 0: legacy i/o wait (default), 1: legacy i/o bus retry */
+#define CM_C_EEACCESS 0x00080000 /* direct programming eeprom regs */
+#define CM_C_EECS 0x00040000
+#define CM_C_EEDI46 0x00020000
+#define CM_C_EECK46 0x00010000
#define CM_CHB3D6C 0x00008000 /* 5.1 channels support */
-#define CM_LINE_AS_BASS 0x00006000 /* use line-in as bass */
+#define CM_CENTR2LIN 0x00004000 /* line-in as center out */
+#define CM_BASE2LIN 0x00002000 /* line-in as bass out */
+#define CM_EXBASEN 0x00001000 /* external bass input enable */
#define CM_REG_MISC_CTRL 0x18
-#define CM_PWD 0x80000000
+#define CM_PWD 0x80000000 /* power down */
#define CM_RESET 0x40000000
-#define CM_SFIL_MASK 0x30000000
-#define CM_TXVX 0x08000000
-#define CM_N4SPK3D 0x04000000 /* 4ch output */
+#define CM_SFIL_MASK 0x30000000 /* filter control at front end DAC, model 037? */
+#define CM_VMGAIN 0x10000000 /* analog master amp +6dB, model 039? */
+#define CM_TXVX 0x08000000 /* model 037? */
+#define CM_N4SPK3D 0x04000000 /* copy front to rear */
#define CM_SPDO5V 0x02000000 /* 5V spdif output (1 = 0.5v (coax)) */
#define CM_SPDIF48K 0x01000000 /* write */
#define CM_SPATUS48K 0x01000000 /* read */
-#define CM_ENDBDAC 0x00800000 /* enable dual dac */
+#define CM_ENDBDAC 0x00800000 /* enable double dac */
#define CM_XCHGDAC 0x00400000 /* 0: front=ch0, 1: front=ch1 */
#define CM_SPD32SEL 0x00200000 /* 0: 16bit SPDIF, 1: 32bit */
-#define CM_SPDFLOOPI 0x00100000 /* int. SPDIF-IN -> int. OUT */
-#define CM_FM_EN 0x00080000 /* enalbe FM */
+#define CM_SPDFLOOPI 0x00100000 /* int. SPDIF-OUT -> int. IN */
+#define CM_FM_EN 0x00080000 /* enable legacy FM */
#define CM_AC3EN2 0x00040000 /* enable AC3: model 039 */
-#define CM_VIDWPDSB 0x00010000
+#define CM_ENWRASID 0x00010000 /* choose writable internal SUBID (audio) */
+#define CM_VIDWPDSB 0x00010000 /* model 037? */
#define CM_SPDF_AC97 0x00008000 /* 0: SPDIF/OUT 44.1K, 1: 48K */
-#define CM_MASK_EN 0x00004000
-#define CM_VIDWPPRT 0x00002000
-#define CM_SFILENB 0x00001000
-#define CM_MMODE_MASK 0x00000E00
+#define CM_MASK_EN 0x00004000 /* activate channel mask on legacy DMA */
+#define CM_ENWRMSID 0x00002000 /* choose writable internal SUBID (modem) */
+#define CM_VIDWPPRT 0x00002000 /* model 037? */
+#define CM_SFILENB 0x00001000 /* filter stepping at front end DAC, model 037? */
+#define CM_MMODE_MASK 0x00000E00 /* model DAA interface mode */
#define CM_SPDIF_SELECT2 0x00000100 /* for model > 039 ? */
#define CM_ENCENTER 0x00000080
-#define CM_FLINKON 0x00000040
-#define CM_FLINKOFF 0x00000020
-#define CM_MIDSMP 0x00000010
-#define CM_UPDDMA_MASK 0x0000000C
-#define CM_TWAIT_MASK 0x00000003
+#define CM_FLINKON 0x00000080 /* force modem link detection on, model 037 */
+#define CM_MUTECH1 0x00000040 /* mute PCI ch1 to DAC */
+#define CM_FLINKOFF 0x00000040 /* force modem link detection off, model 037 */
+#define CM_UNKNOWN_18_5 0x00000020 /* ? */
+#define CM_MIDSMP 0x00000010 /* 1/2 interpolation at front end DAC */
+#define CM_UPDDMA_MASK 0x0000000C /* TDMA position update notification */
+#define CM_UPDDMA_2048 0x00000000
+#define CM_UPDDMA_1024 0x00000004
+#define CM_UPDDMA_512 0x00000008
+#define CM_UPDDMA_256 0x0000000C
+#define CM_TWAIT_MASK 0x00000003 /* model 037 */
+#define CM_TWAIT1 0x00000002 /* FM i/o cycle, 0: 48, 1: 64 PCICLKs */
+#define CM_TWAIT0 0x00000001 /* i/o cycle, 0: 4, 1: 6 PCICLKs */
+
+#define CM_REG_TDMA_POSITION 0x1C
+#define CM_TDMA_CNT_MASK 0xFFFF0000 /* current byte/word count */
+#define CM_TDMA_ADR_MASK 0x0000FFFF /* current address */
/* byte */
#define CM_REG_MIXER0 0x20
+#define CM_REG_SBVR 0x20 /* write: sb16 version */
+#define CM_REG_DEV 0x20 /* read: hardware device version */
+
+#define CM_REG_MIXER21 0x21
+#define CM_UNKNOWN_21_MASK 0x78 /* ? */
+#define CM_X_ADPCM 0x04 /* SB16 ADPCM enable */
+#define CM_PROINV 0x02 /* SBPro left/right channel switching */
+#define CM_X_SB16 0x01 /* SB16 compatible */
#define CM_REG_SB16_DATA 0x22
#define CM_REG_SB16_ADDR 0x23
@@ -243,8 +288,8 @@ MODULE_PARM_DESC(joystick_port, "Joystick port address.");
#define CM_FMMUTE_SHIFT 7
#define CM_WSMUTE 0x40 /* mute PCM */
#define CM_WSMUTE_SHIFT 6
-#define CM_SPK4 0x20 /* lin-in -> rear line out */
-#define CM_SPK4_SHIFT 5
+#define CM_REAR2LIN 0x20 /* lin-in -> rear line out */
+#define CM_REAR2LIN_SHIFT 5
#define CM_REAR2FRONT 0x10 /* exchange rear/front */
#define CM_REAR2FRONT_SHIFT 4
#define CM_WAVEINL 0x08 /* digital wave rec. left chan */
@@ -276,12 +321,13 @@ MODULE_PARM_DESC(joystick_port, "Joystick port address.");
#define CM_VAUXR_MASK 0x0f
#define CM_REG_MISC 0x27
+#define CM_UNKNOWN_27_MASK 0xd8 /* ? */
#define CM_XGPO1 0x20
// #define CM_XGPBIO 0x04
#define CM_MIC_CENTER_LFE 0x04 /* mic as center/lfe out? (model 039 or later?) */
#define CM_SPDIF_INVERSE 0x04 /* spdif input phase inverse (model 037) */
#define CM_SPDVALID 0x02 /* spdif input valid check */
-#define CM_DMAUTO 0x01
+#define CM_DMAUTO 0x01 /* SB16 DMA auto detect */
#define CM_REG_AC97 0x28 /* hmmm.. do we have ac97 link? */
/*
@@ -322,18 +368,20 @@ MODULE_PARM_DESC(joystick_port, "Joystick port address.");
/*
* extended registers
*/
-#define CM_REG_CH0_FRAME1 0x80 /* base address */
-#define CM_REG_CH0_FRAME2 0x84
+#define CM_REG_CH0_FRAME1 0x80 /* write: base address */
+#define CM_REG_CH0_FRAME2 0x84 /* read: current address */
#define CM_REG_CH1_FRAME1 0x88 /* 0-15: count of samples at bus master; buffer size */
#define CM_REG_CH1_FRAME2 0x8C /* 16-31: count of samples at codec; fragment size */
+
#define CM_REG_EXT_MISC 0x90
-#define CM_REG_MISC_CTRL_8768 0x92 /* reg. name the same as 0x18 */
-#define CM_CHB3D8C 0x20 /* 7.1 channels support */
-#define CM_SPD32FMT 0x10 /* SPDIF/IN 32k */
-#define CM_ADC2SPDIF 0x08 /* ADC output to SPDIF/OUT */
-#define CM_SHAREADC 0x04 /* DAC in ADC as Center/LFE */
-#define CM_REALTCMP 0x02 /* monitor the CMPL/CMPR of ADC */
-#define CM_INVLRCK 0x01 /* invert ZVPORT's LRCK */
+#define CM_ADC48K44K 0x10000000 /* ADC parameters group, 0: 44k, 1: 48k */
+#define CM_CHB3D8C 0x00200000 /* 7.1 channels support */
+#define CM_SPD32FMT 0x00100000 /* SPDIF/IN 32k sample rate */
+#define CM_ADC2SPDIF 0x00080000 /* ADC output to SPDIF/OUT */
+#define CM_SHAREADC 0x00040000 /* DAC in ADC as Center/LFE */
+#define CM_REALTCMP 0x00020000 /* monitor the CMPL/CMPR of ADC */
+#define CM_INVLRCK 0x00010000 /* invert ZVPORT's LRCK */
+#define CM_UNKNOWN_90_MASK 0x0000FFFF /* ? */
/*
* size of i/o region
@@ -383,15 +431,14 @@ MODULE_PARM_DESC(joystick_port, "Joystick port address.");
struct cmipci_pcm {
struct snd_pcm_substream *substream;
- int running; /* dac/adc running? */
+ u8 running; /* dac/adc running? */
+ u8 fmt; /* format bits */
+ u8 is_dac;
+ u8 needs_silencing;
unsigned int dma_size; /* in frames */
- unsigned int period_size; /* in frames */
+ unsigned int shift;
+ unsigned int ch; /* channel (0/1) */
unsigned int offset; /* physical address of the buffer */
- unsigned int fmt; /* format bits */
- int ch; /* channel (0/1) */
- unsigned int is_dac; /* is dac? */
- int bytes_per_frame;
- int shift;
};
/* mixer elements toggled/resumed during ac3 playback */
@@ -424,7 +471,6 @@ struct cmipci {
int chip_version;
int max_channels;
- unsigned int has_dual_dac: 1;
unsigned int can_ac3_sw: 1;
unsigned int can_ac3_hw: 1;
unsigned int can_multi_ch: 1;
@@ -557,6 +603,9 @@ static unsigned int rates[] = { 5512, 11025, 22050, 44100, 8000, 16000, 32000, 4
static unsigned int snd_cmipci_rate_freq(unsigned int rate)
{
unsigned int i;
+
+ if (rate > 48000)
+ rate /= 2;
for (i = 0; i < ARRAY_SIZE(rates); i++) {
if (rates[i] == rate)
return i;
@@ -671,19 +720,19 @@ static int snd_cmipci_hw_free(struct snd_pcm_substream *substream)
/*
*/
-static unsigned int hw_channels[] = {1, 2, 4, 5, 6, 8};
+static unsigned int hw_channels[] = {1, 2, 4, 6, 8};
static struct snd_pcm_hw_constraint_list hw_constraints_channels_4 = {
.count = 3,
.list = hw_channels,
.mask = 0,
};
static struct snd_pcm_hw_constraint_list hw_constraints_channels_6 = {
- .count = 5,
+ .count = 4,
.list = hw_channels,
.mask = 0,
};
static struct snd_pcm_hw_constraint_list hw_constraints_channels_8 = {
- .count = 6,
+ .count = 5,
.list = hw_channels,
.mask = 0,
};
@@ -691,48 +740,37 @@ static struct snd_pcm_hw_constraint_list hw_constraints_channels_8 = {
static int set_dac_channels(struct cmipci *cm, struct cmipci_pcm *rec, int channels)
{
if (channels > 2) {
- if (! cm->can_multi_ch)
+ if (!cm->can_multi_ch || !rec->ch)
return -EINVAL;
if (rec->fmt != 0x03) /* stereo 16bit only */
return -EINVAL;
+ }
+ if (cm->can_multi_ch) {
spin_lock_irq(&cm->reg_lock);
- snd_cmipci_set_bit(cm, CM_REG_LEGACY_CTRL, CM_NXCHG);
- snd_cmipci_set_bit(cm, CM_REG_MISC_CTRL, CM_XCHGDAC);
- if (channels > 4) {
- snd_cmipci_clear_bit(cm, CM_REG_CHFORMAT, CM_CHB3D);
- snd_cmipci_set_bit(cm, CM_REG_CHFORMAT, CM_CHB3D5C);
+ if (channels > 2) {
+ snd_cmipci_set_bit(cm, CM_REG_LEGACY_CTRL, CM_NXCHG);
+ snd_cmipci_set_bit(cm, CM_REG_MISC_CTRL, CM_XCHGDAC);
} else {
- snd_cmipci_clear_bit(cm, CM_REG_CHFORMAT, CM_CHB3D5C);
- snd_cmipci_set_bit(cm, CM_REG_CHFORMAT, CM_CHB3D);
+ snd_cmipci_clear_bit(cm, CM_REG_LEGACY_CTRL, CM_NXCHG);
+ snd_cmipci_clear_bit(cm, CM_REG_MISC_CTRL, CM_XCHGDAC);
}
- if (channels >= 6) {
+ if (channels == 8)
+ snd_cmipci_set_bit(cm, CM_REG_EXT_MISC, CM_CHB3D8C);
+ else
+ snd_cmipci_clear_bit(cm, CM_REG_EXT_MISC, CM_CHB3D8C);
+ if (channels == 6) {
+ snd_cmipci_set_bit(cm, CM_REG_CHFORMAT, CM_CHB3D5C);
snd_cmipci_set_bit(cm, CM_REG_LEGACY_CTRL, CM_CHB3D6C);
- snd_cmipci_set_bit(cm, CM_REG_MISC_CTRL, CM_ENCENTER);
} else {
- snd_cmipci_clear_bit(cm, CM_REG_LEGACY_CTRL, CM_CHB3D6C);
- snd_cmipci_clear_bit(cm, CM_REG_MISC_CTRL, CM_ENCENTER);
- }
- if (cm->chip_version == 68) {
- if (channels == 8) {
- snd_cmipci_set_bit(cm, CM_REG_MISC_CTRL_8768, CM_CHB3D8C);
- } else {
- snd_cmipci_clear_bit(cm, CM_REG_MISC_CTRL_8768, CM_CHB3D8C);
- }
- }
- spin_unlock_irq(&cm->reg_lock);
-
- } else {
- if (cm->can_multi_ch) {
- spin_lock_irq(&cm->reg_lock);
- snd_cmipci_clear_bit(cm, CM_REG_LEGACY_CTRL, CM_NXCHG);
- snd_cmipci_clear_bit(cm, CM_REG_CHFORMAT, CM_CHB3D);
snd_cmipci_clear_bit(cm, CM_REG_CHFORMAT, CM_CHB3D5C);
snd_cmipci_clear_bit(cm, CM_REG_LEGACY_CTRL, CM_CHB3D6C);
- snd_cmipci_clear_bit(cm, CM_REG_MISC_CTRL, CM_ENCENTER);
- snd_cmipci_clear_bit(cm, CM_REG_MISC_CTRL, CM_XCHGDAC);
- spin_unlock_irq(&cm->reg_lock);
}
+ if (channels == 4)
+ snd_cmipci_set_bit(cm, CM_REG_CHFORMAT, CM_CHB3D);
+ else
+ snd_cmipci_clear_bit(cm, CM_REG_CHFORMAT, CM_CHB3D);
+ spin_unlock_irq(&cm->reg_lock);
}
return 0;
}
@@ -746,6 +784,7 @@ static int snd_cmipci_pcm_prepare(struct cmipci *cm, struct cmipci_pcm *rec,
struct snd_pcm_substream *substream)
{
unsigned int reg, freq, val;
+ unsigned int period_size;
struct snd_pcm_runtime *runtime = substream->runtime;
rec->fmt = 0;
@@ -765,11 +804,11 @@ static int snd_cmipci_pcm_prepare(struct cmipci *cm, struct cmipci_pcm *rec,
rec->offset = runtime->dma_addr;
/* buffer and period sizes in frame */
rec->dma_size = runtime->buffer_size << rec->shift;
- rec->period_size = runtime->period_size << rec->shift;
+ period_size = runtime->period_size << rec->shift;
if (runtime->channels > 2) {
/* multi-channels */
rec->dma_size = (rec->dma_size * runtime->channels) / 2;
- rec->period_size = (rec->period_size * runtime->channels) / 2;
+ period_size = (period_size * runtime->channels) / 2;
}
spin_lock_irq(&cm->reg_lock);
@@ -780,7 +819,7 @@ static int snd_cmipci_pcm_prepare(struct cmipci *cm, struct cmipci_pcm *rec,
/* program sample counts */
reg = rec->ch ? CM_REG_CH1_FRAME2 : CM_REG_CH0_FRAME2;
snd_cmipci_write_w(cm, reg, rec->dma_size - 1);
- snd_cmipci_write_w(cm, reg + 2, rec->period_size - 1);
+ snd_cmipci_write_w(cm, reg + 2, period_size - 1);
/* set adc/dac flag */
val = rec->ch ? CM_CHADC1 : CM_CHADC0;
@@ -795,11 +834,11 @@ static int snd_cmipci_pcm_prepare(struct cmipci *cm, struct cmipci_pcm *rec,
freq = snd_cmipci_rate_freq(runtime->rate);
val = snd_cmipci_read(cm, CM_REG_FUNCTRL1);
if (rec->ch) {
- val &= ~CM_ASFC_MASK;
- val |= (freq << CM_ASFC_SHIFT) & CM_ASFC_MASK;
- } else {
val &= ~CM_DSFC_MASK;
val |= (freq << CM_DSFC_SHIFT) & CM_DSFC_MASK;
+ } else {
+ val &= ~CM_ASFC_MASK;
+ val |= (freq << CM_ASFC_SHIFT) & CM_ASFC_MASK;
}
snd_cmipci_write(cm, CM_REG_FUNCTRL1, val);
//snd_printd("cmipci: functrl1 = %08x\n", val);
@@ -813,6 +852,16 @@ static int snd_cmipci_pcm_prepare(struct cmipci *cm, struct cmipci_pcm *rec,
val &= ~CM_CH0FMT_MASK;
val |= rec->fmt << CM_CH0FMT_SHIFT;
}
+ if (cm->chip_version == 68) {
+ if (runtime->rate == 88200)
+ val |= CM_CH0_SRATE_88K << (rec->ch * 2);
+ else
+ val &= ~(CM_CH0_SRATE_88K << (rec->ch * 2));
+ if (runtime->rate == 96000)
+ val |= CM_CH0_SRATE_96K << (rec->ch * 2);
+ else
+ val &= ~(CM_CH0_SRATE_96K << (rec->ch * 2));
+ }
snd_cmipci_write(cm, CM_REG_CHFORMAT, val);
//snd_printd("cmipci: chformat = %08x\n", val);
@@ -826,7 +875,7 @@ static int snd_cmipci_pcm_prepare(struct cmipci *cm, struct cmipci_pcm *rec,
* PCM trigger/stop
*/
static int snd_cmipci_pcm_trigger(struct cmipci *cm, struct cmipci_pcm *rec,
- struct snd_pcm_substream *substream, int cmd)
+ int cmd)
{
unsigned int inthld, chen, reset, pause;
int result = 0;
@@ -855,6 +904,7 @@ static int snd_cmipci_pcm_trigger(struct cmipci *cm, struct cmipci_pcm *rec,
cm->ctrl &= ~chen;
snd_cmipci_write(cm, CM_REG_FUNCTRL0, cm->ctrl | reset);
snd_cmipci_write(cm, CM_REG_FUNCTRL0, cm->ctrl & ~reset);
+ rec->needs_silencing = rec->is_dac;
break;
case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
case SNDRV_PCM_TRIGGER_SUSPEND:
@@ -906,7 +956,7 @@ static int snd_cmipci_playback_trigger(struct snd_pcm_substream *substream,
int cmd)
{
struct cmipci *cm = snd_pcm_substream_chip(substream);
- return snd_cmipci_pcm_trigger(cm, &cm->channel[CM_CH_PLAY], substream, cmd);
+ return snd_cmipci_pcm_trigger(cm, &cm->channel[CM_CH_PLAY], cmd);
}
static snd_pcm_uframes_t snd_cmipci_playback_pointer(struct snd_pcm_substream *substream)
@@ -925,7 +975,7 @@ static int snd_cmipci_capture_trigger(struct snd_pcm_substream *substream,
int cmd)
{
struct cmipci *cm = snd_pcm_substream_chip(substream);
- return snd_cmipci_pcm_trigger(cm, &cm->channel[CM_CH_CAPT], substream, cmd);
+ return snd_cmipci_pcm_trigger(cm, &cm->channel[CM_CH_CAPT], cmd);
}
static snd_pcm_uframes_t snd_cmipci_capture_pointer(struct snd_pcm_substream *substream)
@@ -1199,15 +1249,19 @@ static int setup_spdif_playback(struct cmipci *cm, struct snd_pcm_substream *sub
snd_cmipci_set_bit(cm, CM_REG_FUNCTRL1, CM_PLAYBACK_SPDF);
setup_ac3(cm, subs, do_ac3, rate);
- if (rate == 48000)
+ if (rate == 48000 || rate == 96000)
snd_cmipci_set_bit(cm, CM_REG_MISC_CTRL, CM_SPDIF48K | CM_SPDF_AC97);
else
snd_cmipci_clear_bit(cm, CM_REG_MISC_CTRL, CM_SPDIF48K | CM_SPDF_AC97);
-
+ if (rate > 48000)
+ snd_cmipci_set_bit(cm, CM_REG_CHFORMAT, CM_DBLSPDS);
+ else
+ snd_cmipci_clear_bit(cm, CM_REG_CHFORMAT, CM_DBLSPDS);
} else {
/* they are controlled via "IEC958 Output Switch" */
/* snd_cmipci_clear_bit(cm, CM_REG_LEGACY_CTRL, CM_ENSPDOUT); */
/* snd_cmipci_clear_bit(cm, CM_REG_FUNCTRL1, CM_SPDO2DAC); */
+ snd_cmipci_clear_bit(cm, CM_REG_CHFORMAT, CM_DBLSPDS);
snd_cmipci_clear_bit(cm, CM_REG_FUNCTRL1, CM_PLAYBACK_SPDF);
setup_ac3(cm, subs, 0, 0);
}
@@ -1227,7 +1281,7 @@ static int snd_cmipci_playback_prepare(struct snd_pcm_substream *substream)
int rate = substream->runtime->rate;
int err, do_spdif, do_ac3 = 0;
- do_spdif = ((rate == 44100 || rate == 48000) &&
+ do_spdif = (rate >= 44100 &&
substream->runtime->format == SNDRV_PCM_FORMAT_S16_LE &&
substream->runtime->channels == 2);
if (do_spdif && cm->can_ac3_hw)
@@ -1252,11 +1306,75 @@ static int snd_cmipci_playback_spdif_prepare(struct snd_pcm_substream *substream
return snd_cmipci_pcm_prepare(cm, &cm->channel[CM_CH_PLAY], substream);
}
+/*
+ * Apparently, the samples last played on channel A stay in some buffer, even
+ * after the channel is reset, and get added to the data for the rear DACs when
+ * playing a multichannel stream on channel B. This is likely to generate
+ * wraparounds and thus distortions.
+ * To avoid this, we play at least one zero sample after the actual stream has
+ * stopped.
+ */
+static void snd_cmipci_silence_hack(struct cmipci *cm, struct cmipci_pcm *rec)
+{
+ struct snd_pcm_runtime *runtime = rec->substream->runtime;
+ unsigned int reg, val;
+
+ if (rec->needs_silencing && runtime && runtime->dma_area) {
+ /* set up a small silence buffer */
+ memset(runtime->dma_area, 0, PAGE_SIZE);
+ reg = rec->ch ? CM_REG_CH1_FRAME2 : CM_REG_CH0_FRAME2;
+ val = ((PAGE_SIZE / 4) - 1) | (((PAGE_SIZE / 4) / 2 - 1) << 16);
+ snd_cmipci_write(cm, reg, val);
+
+ /* configure for 16 bits, 2 channels, 8 kHz */
+ if (runtime->channels > 2)
+ set_dac_channels(cm, rec, 2);
+ spin_lock_irq(&cm->reg_lock);
+ val = snd_cmipci_read(cm, CM_REG_FUNCTRL1);
+ val &= ~(CM_ASFC_MASK << (rec->ch * 3));
+ val |= (4 << CM_ASFC_SHIFT) << (rec->ch * 3);
+ snd_cmipci_write(cm, CM_REG_FUNCTRL1, val);
+ val = snd_cmipci_read(cm, CM_REG_CHFORMAT);
+ val &= ~(CM_CH0FMT_MASK << (rec->ch * 2));
+ val |= (3 << CM_CH0FMT_SHIFT) << (rec->ch * 2);
+ if (cm->chip_version == 68) {
+ val &= ~(CM_CH0_SRATE_88K << (rec->ch * 2));
+ val &= ~(CM_CH0_SRATE_96K << (rec->ch * 2));
+ }
+ snd_cmipci_write(cm, CM_REG_CHFORMAT, val);
+
+ /* start stream (we don't need interrupts) */
+ cm->ctrl |= CM_CHEN0 << rec->ch;
+ snd_cmipci_write(cm, CM_REG_FUNCTRL0, cm->ctrl);
+ spin_unlock_irq(&cm->reg_lock);
+
+ msleep(1);
+
+ /* stop and reset stream */
+ spin_lock_irq(&cm->reg_lock);
+ cm->ctrl &= ~(CM_CHEN0 << rec->ch);
+ val = CM_RST_CH0 << rec->ch;
+ snd_cmipci_write(cm, CM_REG_FUNCTRL0, cm->ctrl | val);
+ snd_cmipci_write(cm, CM_REG_FUNCTRL0, cm->ctrl & ~val);
+ spin_unlock_irq(&cm->reg_lock);
+
+ rec->needs_silencing = 0;
+ }
+}
+
static int snd_cmipci_playback_hw_free(struct snd_pcm_substream *substream)
{
struct cmipci *cm = snd_pcm_substream_chip(substream);
setup_spdif_playback(cm, substream, 0, 0);
restore_mixer_state(cm);
+ snd_cmipci_silence_hack(cm, &cm->channel[0]);
+ return snd_cmipci_hw_free(substream);
+}
+
+static int snd_cmipci_playback2_hw_free(struct snd_pcm_substream *substream)
+{
+ struct cmipci *cm = snd_pcm_substream_chip(substream);
+ snd_cmipci_silence_hack(cm, &cm->channel[1]);
return snd_cmipci_hw_free(substream);
}
@@ -1515,7 +1633,11 @@ static int snd_cmipci_playback_open(struct snd_pcm_substream *substream)
if ((err = open_device_check(cm, CM_OPEN_PLAYBACK, substream)) < 0)
return err;
runtime->hw = snd_cmipci_playback;
- runtime->hw.channels_max = cm->max_channels;
+ if (cm->chip_version == 68) {
+ runtime->hw.rates |= SNDRV_PCM_RATE_88200 |
+ SNDRV_PCM_RATE_96000;
+ runtime->hw.rate_max = 96000;
+ }
snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, 0, 0x10000);
cm->dig_pcm_status = cm->dig_status;
return 0;
@@ -1558,9 +1680,14 @@ static int snd_cmipci_playback2_open(struct snd_pcm_substream *substream)
else if (cm->max_channels == 8)
snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, &hw_constraints_channels_8);
}
- snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, 0, 0x10000);
}
mutex_unlock(&cm->open_mutex);
+ if (cm->chip_version == 68) {
+ runtime->hw.rates |= SNDRV_PCM_RATE_88200 |
+ SNDRV_PCM_RATE_96000;
+ runtime->hw.rate_max = 96000;
+ }
+ snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, 0, 0x10000);
return 0;
}
@@ -1574,8 +1701,15 @@ static int snd_cmipci_playback_spdif_open(struct snd_pcm_substream *substream)
return err;
if (cm->can_ac3_hw) {
runtime->hw = snd_cmipci_playback_spdif;
- if (cm->chip_version >= 37)
+ if (cm->chip_version >= 37) {
runtime->hw.formats |= SNDRV_PCM_FMTBIT_S32_LE;
+ snd_pcm_hw_constraint_msbits(runtime, 0, 32, 24);
+ }
+ if (cm->chip_version == 68) {
+ runtime->hw.rates |= SNDRV_PCM_RATE_88200 |
+ SNDRV_PCM_RATE_96000;
+ runtime->hw.rate_max = 96000;
+ }
} else {
runtime->hw = snd_cmipci_playback_iec958_subframe;
}
@@ -1668,7 +1802,7 @@ static struct snd_pcm_ops snd_cmipci_playback2_ops = {
.close = snd_cmipci_playback2_close,
.ioctl = snd_pcm_lib_ioctl,
.hw_params = snd_cmipci_playback2_hw_params,
- .hw_free = snd_cmipci_hw_free,
+ .hw_free = snd_cmipci_playback2_hw_free,
.prepare = snd_cmipci_capture_prepare, /* channel B */
.trigger = snd_cmipci_capture_trigger, /* channel B */
.pointer = snd_cmipci_capture_pointer, /* channel B */
@@ -2139,15 +2273,7 @@ struct cmipci_switch_args {
*/
};
-static int snd_cmipci_uswitch_info(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_info *uinfo)
-{
- uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
- uinfo->count = 1;
- uinfo->value.integer.min = 0;
- uinfo->value.integer.max = 1;
- return 0;
-}
+#define snd_cmipci_uswitch_info snd_ctl_boolean_mono_info
static int _snd_cmipci_uswitch_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol,
@@ -2260,8 +2386,8 @@ DEFINE_SWITCH_ARG(exchange_dac, CM_REG_MISC_CTRL, CM_XCHGDAC, 0, 0, 0); /* rever
DEFINE_SWITCH_ARG(exchange_dac, CM_REG_MISC_CTRL, CM_XCHGDAC, CM_XCHGDAC, 0, 0);
#endif
DEFINE_BIT_SWITCH_ARG(fourch, CM_REG_MISC_CTRL, CM_N4SPK3D, 0, 0);
-// DEFINE_BIT_SWITCH_ARG(line_rear, CM_REG_MIXER1, CM_SPK4, 1, 0);
-// DEFINE_BIT_SWITCH_ARG(line_bass, CM_REG_LEGACY_CTRL, CM_LINE_AS_BASS, 0, 0);
+// DEFINE_BIT_SWITCH_ARG(line_rear, CM_REG_MIXER1, CM_REAR2LIN, 1, 0);
+// DEFINE_BIT_SWITCH_ARG(line_bass, CM_REG_LEGACY_CTRL, CM_CENTR2LIN|CM_BASE2LIN, 0, 0);
// DEFINE_BIT_SWITCH_ARG(joystick, CM_REG_FUNCTRL1, CM_JYSTK_EN, 0, 0); /* now module option */
DEFINE_SWITCH_ARG(modem, CM_REG_MISC_CTRL, CM_FLINKON|CM_FLINKOFF, CM_FLINKON, 0, 0);
@@ -2331,11 +2457,11 @@ static inline unsigned int get_line_in_mode(struct cmipci *cm)
unsigned int val;
if (cm->chip_version >= 39) {
val = snd_cmipci_read(cm, CM_REG_LEGACY_CTRL);
- if (val & CM_LINE_AS_BASS)
+ if (val & (CM_CENTR2LIN | CM_BASE2LIN))
return 2;
}
val = snd_cmipci_read_b(cm, CM_REG_MIXER1);
- if (val & CM_SPK4)
+ if (val & CM_REAR2LIN)
return 1;
return 0;
}
@@ -2359,13 +2485,13 @@ static int snd_cmipci_line_in_mode_put(struct snd_kcontrol *kcontrol,
spin_lock_irq(&cm->reg_lock);
if (ucontrol->value.enumerated.item[0] == 2)
- change = snd_cmipci_set_bit(cm, CM_REG_LEGACY_CTRL, CM_LINE_AS_BASS);
+ change = snd_cmipci_set_bit(cm, CM_REG_LEGACY_CTRL, CM_CENTR2LIN | CM_BASE2LIN);
else
- change = snd_cmipci_clear_bit(cm, CM_REG_LEGACY_CTRL, CM_LINE_AS_BASS);
+ change = snd_cmipci_clear_bit(cm, CM_REG_LEGACY_CTRL, CM_CENTR2LIN | CM_BASE2LIN);
if (ucontrol->value.enumerated.item[0] == 1)
- change |= snd_cmipci_set_bit_b(cm, CM_REG_MIXER1, CM_SPK4);
+ change |= snd_cmipci_set_bit_b(cm, CM_REG_MIXER1, CM_REAR2LIN);
else
- change |= snd_cmipci_clear_bit_b(cm, CM_REG_MIXER1, CM_SPK4);
+ change |= snd_cmipci_clear_bit_b(cm, CM_REG_MIXER1, CM_REAR2LIN);
spin_unlock_irq(&cm->reg_lock);
return change;
}
@@ -2583,19 +2709,18 @@ static void snd_cmipci_proc_read(struct snd_info_entry *entry,
struct snd_info_buffer *buffer)
{
struct cmipci *cm = entry->private_data;
- int i;
+ int i, v;
- snd_iprintf(buffer, "%s\n\n", cm->card->longname);
- for (i = 0; i < 0x40; i++) {
- int v = inb(cm->iobase + i);
+ snd_iprintf(buffer, "%s\n", cm->card->longname);
+ for (i = 0; i < 0x94; i++) {
+ if (i == 0x28)
+ i = 0x90;
+ v = inb(cm->iobase + i);
if (i % 4 == 0)
- snd_iprintf(buffer, "%02x: ", i);
- snd_iprintf(buffer, "%02x", v);
- if (i % 4 == 3)
- snd_iprintf(buffer, "\n");
- else
- snd_iprintf(buffer, " ");
+ snd_iprintf(buffer, "\n%02x:", i);
+ snd_iprintf(buffer, " %02x", v);
}
+ snd_iprintf(buffer, "\n");
}
static void __devinit snd_cmipci_proc_init(struct cmipci *cm)
@@ -2633,46 +2758,40 @@ static void __devinit query_chip(struct cmipci *cm)
if (! detect) {
/* check reg 08h, bit 24-28 */
detect = snd_cmipci_read(cm, CM_REG_CHFORMAT) & CM_CHIP_MASK1;
- if (! detect) {
+ switch (detect) {
+ case 0:
cm->chip_version = 33;
- cm->max_channels = 2;
if (cm->do_soft_ac3)
cm->can_ac3_sw = 1;
else
cm->can_ac3_hw = 1;
- cm->has_dual_dac = 1;
- } else {
+ break;
+ case CM_CHIP_037:
cm->chip_version = 37;
- cm->max_channels = 2;
cm->can_ac3_hw = 1;
- cm->has_dual_dac = 1;
+ break;
+ default:
+ cm->chip_version = 39;
+ cm->can_ac3_hw = 1;
+ break;
}
+ cm->max_channels = 2;
} else {
- /* check reg 0Ch, bit 26 */
- if (detect & CM_CHIP_8768) {
- cm->chip_version = 68;
- cm->max_channels = 8;
- cm->can_ac3_hw = 1;
- cm->has_dual_dac = 1;
- cm->can_multi_ch = 1;
- } else if (detect & CM_CHIP_055) {
- cm->chip_version = 55;
- cm->max_channels = 6;
- cm->can_ac3_hw = 1;
- cm->has_dual_dac = 1;
- cm->can_multi_ch = 1;
- } else if (detect & CM_CHIP_039) {
+ if (detect & CM_CHIP_039) {
cm->chip_version = 39;
if (detect & CM_CHIP_039_6CH) /* 4 or 6 channels */
cm->max_channels = 6;
else
cm->max_channels = 4;
- cm->can_ac3_hw = 1;
- cm->has_dual_dac = 1;
- cm->can_multi_ch = 1;
+ } else if (detect & CM_CHIP_8768) {
+ cm->chip_version = 68;
+ cm->max_channels = 8;
} else {
- printk(KERN_ERR "chip %x version not supported\n", detect);
+ cm->chip_version = 55;
+ cm->max_channels = 6;
}
+ cm->can_ac3_hw = 1;
+ cm->can_multi_ch = 1;
}
}
@@ -2782,10 +2901,14 @@ static int __devinit snd_cmipci_create_fm(struct cmipci *cm, long fm_port)
if (!fm_port)
goto disable_fm;
- /* first try FM regs in PCI port range */
- iosynth = cm->iobase + CM_REG_FM_PCI;
- err = snd_opl3_create(cm->card, iosynth, iosynth + 2,
- OPL3_HW_OPL3, 1, &opl3);
+ if (cm->chip_version >= 39) {
+ /* first try FM regs in PCI port range */
+ iosynth = cm->iobase + CM_REG_FM_PCI;
+ err = snd_opl3_create(cm->card, iosynth, iosynth + 2,
+ OPL3_HW_OPL3, 1, &opl3);
+ } else {
+ err = -EIO;
+ }
if (err < 0) {
/* then try legacy ports */
val = snd_cmipci_read(cm, CM_REG_LEGACY_CTRL) & ~CM_FMSEL_MASK;
@@ -2829,9 +2952,10 @@ static int __devinit snd_cmipci_create(struct snd_card *card, struct pci_dev *pc
static struct snd_device_ops ops = {
.dev_free = snd_cmipci_dev_free,
};
- unsigned int val = 0;
+ unsigned int val;
long iomidi;
- int integrated_midi;
+ int integrated_midi = 0;
+ char modelstr[16];
int pcm_index, pcm_spdif_index;
static struct pci_device_id intel_82437vx[] = {
{ PCI_DEVICE(PCI_VENDOR_ID_INTEL, PCI_DEVICE_ID_INTEL_82437VX) },
@@ -2904,6 +3028,8 @@ static int __devinit snd_cmipci_create(struct snd_card *card, struct pci_dev *pc
#endif
/* initialize codec registers */
+ snd_cmipci_set_bit(cm, CM_REG_MISC_CTRL, CM_RESET);
+ snd_cmipci_clear_bit(cm, CM_REG_MISC_CTRL, CM_RESET);
snd_cmipci_write(cm, CM_REG_INT_HLDCLR, 0); /* disable ints */
snd_cmipci_ch_reset(cm, CM_CH_PLAY);
snd_cmipci_ch_reset(cm, CM_CH_CAPT);
@@ -2917,6 +3043,10 @@ static int __devinit snd_cmipci_create(struct snd_card *card, struct pci_dev *pc
#else
snd_cmipci_clear_bit(cm, CM_REG_MISC_CTRL, CM_XCHGDAC);
#endif
+ if (cm->chip_version) {
+ snd_cmipci_write_b(cm, CM_REG_EXT_MISC, 0x20); /* magic */
+ snd_cmipci_write_b(cm, CM_REG_EXT_MISC + 1, 0x09); /* more magic */
+ }
/* Set Bus Master Request */
snd_cmipci_set_bit(cm, CM_REG_FUNCTRL1, CM_BREQ);
@@ -2931,15 +3061,55 @@ static int __devinit snd_cmipci_create(struct snd_card *card, struct pci_dev *pc
break;
}
+ if (cm->chip_version < 68) {
+ val = pci->device < 0x110 ? 8338 : 8738;
+ } else {
+ switch (snd_cmipci_read_b(cm, CM_REG_INT_HLDCLR + 3) & 0x03) {
+ case 0:
+ val = 8769;
+ break;
+ case 2:
+ val = 8762;
+ break;
+ default:
+ switch ((pci->subsystem_vendor << 16) |
+ pci->subsystem_device) {
+ case 0x13f69761:
+ case 0x584d3741:
+ case 0x584d3751:
+ case 0x584d3761:
+ case 0x584d3771:
+ case 0x72848384:
+ val = 8770;
+ break;
+ default:
+ val = 8768;
+ break;
+ }
+ }
+ }
+ sprintf(card->shortname, "C-Media CMI%d", val);
+ if (cm->chip_version < 68)
+ sprintf(modelstr, " (model %d)", cm->chip_version);
+ else
+ modelstr[0] = '\0';
+ sprintf(card->longname, "%s%s at %#lx, irq %i",
+ card->shortname, modelstr, cm->iobase, cm->irq);
+
if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, cm, &ops)) < 0) {
snd_cmipci_free(cm);
return err;
}
- integrated_midi = snd_cmipci_read_b(cm, CM_REG_MPU_PCI) != 0xff;
- if (integrated_midi && mpu_port[dev] == 1)
- iomidi = cm->iobase + CM_REG_MPU_PCI;
- else {
+ if (cm->chip_version >= 39) {
+ val = snd_cmipci_read_b(cm, CM_REG_MPU_PCI + 1);
+ if (val != 0x00 && val != 0xff) {
+ iomidi = cm->iobase + CM_REG_MPU_PCI;
+ integrated_midi = 1;
+ }
+ }
+ if (!integrated_midi) {
+ val = 0;
iomidi = mpu_port[dev];
switch (iomidi) {
case 0x320: val = CM_VMPU_320; break;
@@ -2953,11 +3123,21 @@ static int __devinit snd_cmipci_create(struct snd_card *card, struct pci_dev *pc
snd_cmipci_write(cm, CM_REG_LEGACY_CTRL, val);
/* enable UART */
snd_cmipci_set_bit(cm, CM_REG_FUNCTRL1, CM_UART_EN);
+ if (inb(iomidi + 1) == 0xff) {
+ snd_printk(KERN_ERR "cannot enable MPU-401 port"
+ " at %#lx\n", iomidi);
+ snd_cmipci_clear_bit(cm, CM_REG_FUNCTRL1,
+ CM_UART_EN);
+ iomidi = 0;
+ }
}
}
- if ((err = snd_cmipci_create_fm(cm, fm_port[dev])) < 0)
- return err;
+ if (cm->chip_version < 68) {
+ err = snd_cmipci_create_fm(cm, fm_port[dev]);
+ if (err < 0)
+ return err;
+ }
/* reset mixer */
snd_cmipci_mixer_write(cm, 0, 0);
@@ -2969,11 +3149,9 @@ static int __devinit snd_cmipci_create(struct snd_card *card, struct pci_dev *pc
if ((err = snd_cmipci_pcm_new(cm, pcm_index)) < 0)
return err;
pcm_index++;
- if (cm->has_dual_dac) {
- if ((err = snd_cmipci_pcm2_new(cm, pcm_index)) < 0)
- return err;
- pcm_index++;
- }
+ if ((err = snd_cmipci_pcm2_new(cm, pcm_index)) < 0)
+ return err;
+ pcm_index++;
if (cm->can_ac3_hw || cm->can_ac3_sw) {
pcm_spdif_index = pcm_index;
if ((err = snd_cmipci_pcm_spdif_new(cm, pcm_index)) < 0)
@@ -3057,15 +3235,6 @@ static int __devinit snd_cmipci_probe(struct pci_dev *pci,
}
card->private_data = cm;
- sprintf(card->shortname, "C-Media PCI %s", card->driver);
- sprintf(card->longname, "%s (model %d) at 0x%lx, irq %i",
- card->shortname,
- cm->chip_version,
- cm->iobase,
- cm->irq);
-
- //snd_printd("%s is detected\n", card->longname);
-
if ((err = snd_card_register(card)) < 0) {
snd_card_free(card);
return err;
diff --git a/sound/pci/cs4281.c b/sound/pci/cs4281.c
index 44cf54607647..9a55f4a9739b 100644
--- a/sound/pci/cs4281.c
+++ b/sound/pci/cs4281.c
@@ -1,6 +1,6 @@
/*
* Driver for Cirrus Logic CS4281 based PCI soundcard
- * Copyright (c) by Jaroslav Kysela <perex@suse.cz>,
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>,
*
*
* This program is free software; you can redistribute it and/or modify
@@ -38,7 +38,7 @@
#include <sound/initval.h>
-MODULE_AUTHOR("Jaroslav Kysela <perex@suse.cz>");
+MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
MODULE_DESCRIPTION("Cirrus Logic CS4281");
MODULE_LICENSE("GPL");
MODULE_SUPPORTED_DEVICE("{{Cirrus Logic,CS4281}}");
@@ -842,12 +842,11 @@ static snd_pcm_uframes_t snd_cs4281_pointer(struct snd_pcm_substream *substream)
static struct snd_pcm_hardware snd_cs4281_playback =
{
- .info = (SNDRV_PCM_INFO_MMAP |
- SNDRV_PCM_INFO_INTERLEAVED |
- SNDRV_PCM_INFO_MMAP_VALID |
- SNDRV_PCM_INFO_PAUSE |
- SNDRV_PCM_INFO_RESUME |
- SNDRV_PCM_INFO_SYNC_START),
+ .info = SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_PAUSE |
+ SNDRV_PCM_INFO_RESUME,
.formats = SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S8 |
SNDRV_PCM_FMTBIT_U16_LE | SNDRV_PCM_FMTBIT_S16_LE |
SNDRV_PCM_FMTBIT_U16_BE | SNDRV_PCM_FMTBIT_S16_BE |
@@ -868,12 +867,11 @@ static struct snd_pcm_hardware snd_cs4281_playback =
static struct snd_pcm_hardware snd_cs4281_capture =
{
- .info = (SNDRV_PCM_INFO_MMAP |
- SNDRV_PCM_INFO_INTERLEAVED |
- SNDRV_PCM_INFO_MMAP_VALID |
- SNDRV_PCM_INFO_PAUSE |
- SNDRV_PCM_INFO_RESUME |
- SNDRV_PCM_INFO_SYNC_START),
+ .info = SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_PAUSE |
+ SNDRV_PCM_INFO_RESUME,
.formats = SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S8 |
SNDRV_PCM_FMTBIT_U16_LE | SNDRV_PCM_FMTBIT_S16_LE |
SNDRV_PCM_FMTBIT_U16_BE | SNDRV_PCM_FMTBIT_S16_BE |
@@ -904,7 +902,6 @@ static int snd_cs4281_playback_open(struct snd_pcm_substream *substream)
dma->right_slot = 1;
runtime->private_data = dma;
runtime->hw = snd_cs4281_playback;
- snd_pcm_set_sync(substream);
/* should be detected from the AC'97 layer, but it seems
that although CS4297A rev B reports 18-bit ADC resolution,
samples are 20-bit */
@@ -924,7 +921,6 @@ static int snd_cs4281_capture_open(struct snd_pcm_substream *substream)
dma->right_slot = 11;
runtime->private_data = dma;
runtime->hw = snd_cs4281_capture;
- snd_pcm_set_sync(substream);
/* should be detected from the AC'97 layer, but it seems
that although CS4297A rev B reports 18-bit ADC resolution,
samples are 20-bit */
diff --git a/sound/pci/cs46xx/Makefile b/sound/pci/cs46xx/Makefile
index d8b77b89aec4..67e811ec8539 100644
--- a/sound/pci/cs46xx/Makefile
+++ b/sound/pci/cs46xx/Makefile
@@ -1,12 +1,10 @@
#
# Makefile for ALSA
-# Copyright (c) 2001 by Jaroslav Kysela <perex@suse.cz>
+# Copyright (c) 2001 by Jaroslav Kysela <perex@perex.cz>
#
-snd-cs46xx-objs := cs46xx.o cs46xx_lib.o
-ifeq ($(CONFIG_SND_CS46XX_NEW_DSP),y)
- snd-cs46xx-objs += dsp_spos.o dsp_spos_scb_lib.o
-endif
+snd-cs46xx-y := cs46xx.o cs46xx_lib.o
+snd-cs46xx-$(CONFIG_SND_CS46XX_NEW_DSP) += dsp_spos.o dsp_spos_scb_lib.o
# Toplevel Module Dependency
obj-$(CONFIG_SND_CS46XX) += snd-cs46xx.o
diff --git a/sound/pci/cs46xx/cs46xx.c b/sound/pci/cs46xx/cs46xx.c
index 8b6cd144d101..2699cb6c2cd6 100644
--- a/sound/pci/cs46xx/cs46xx.c
+++ b/sound/pci/cs46xx/cs46xx.c
@@ -1,6 +1,6 @@
/*
* The driver for the Cirrus Logic's Sound Fusion CS46XX based soundcards
- * Copyright (c) by Jaroslav Kysela <perex@suse.cz>
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
*
*
* This program is free software; you can redistribute it and/or modify
@@ -34,7 +34,7 @@
#include <sound/cs46xx.h>
#include <sound/initval.h>
-MODULE_AUTHOR("Jaroslav Kysela <perex@suse.cz>");
+MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
MODULE_DESCRIPTION("Cirrus Logic Sound Fusion CS46XX");
MODULE_LICENSE("GPL");
MODULE_SUPPORTED_DEVICE("{{Cirrus Logic,Sound Fusion (CS4280)},"
diff --git a/sound/pci/cs46xx/cs46xx_lib.c b/sound/pci/cs46xx/cs46xx_lib.c
index 71d7aab9d869..2c7bfc9fef61 100644
--- a/sound/pci/cs46xx/cs46xx_lib.c
+++ b/sound/pci/cs46xx/cs46xx_lib.c
@@ -1,5 +1,5 @@
/*
- * Copyright (c) by Jaroslav Kysela <perex@suse.cz>
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
* Abramo Bagnara <abramo@alsa-project.org>
* Cirrus Logic, Inc.
* Routines for control of Cirrus Logic CS461x chips
@@ -1818,15 +1818,7 @@ static int snd_cs46xx_vol_iec958_put(struct snd_kcontrol *kcontrol, struct snd_c
}
#endif
-static int snd_mixer_boolean_info(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_info *uinfo)
-{
- uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
- uinfo->count = 1;
- uinfo->value.integer.min = 0;
- uinfo->value.integer.max = 1;
- return 0;
-}
+#define snd_mixer_boolean_info snd_ctl_boolean_mono_info
static int snd_cs46xx_iec958_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
diff --git a/sound/pci/cs46xx/cs46xx_lib.h b/sound/pci/cs46xx/cs46xx_lib.h
index 20dcd72f06c1..018a7de56017 100644
--- a/sound/pci/cs46xx/cs46xx_lib.h
+++ b/sound/pci/cs46xx/cs46xx_lib.h
@@ -1,6 +1,6 @@
/*
* The driver for the Cirrus Logic's Sound Fusion CS46XX based soundcards
- * Copyright (c) by Jaroslav Kysela <perex@suse.cz>
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
*
*
* This program is free software; you can redistribute it and/or modify
diff --git a/sound/pci/cs46xx/dsp_spos.h b/sound/pci/cs46xx/dsp_spos.h
index 0d246bca4184..f9e169d33c03 100644
--- a/sound/pci/cs46xx/dsp_spos.h
+++ b/sound/pci/cs46xx/dsp_spos.h
@@ -1,6 +1,6 @@
/*
* The driver for the Cirrus Logic's Sound Fusion CS46XX based soundcards
- * Copyright (c) by Jaroslav Kysela <perex@suse.cz>
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
*
*
* This program is free software; you can redistribute it and/or modify
diff --git a/sound/pci/cs46xx/dsp_spos_scb_lib.c b/sound/pci/cs46xx/dsp_spos_scb_lib.c
index 57e357de1500..eded4dfeba12 100644
--- a/sound/pci/cs46xx/dsp_spos_scb_lib.c
+++ b/sound/pci/cs46xx/dsp_spos_scb_lib.c
@@ -1480,7 +1480,7 @@ void cs46xx_dsp_destroy_pcm_channel (struct snd_cs46xx * chip,
if (!pcm_channel->src_scb->ref_count) {
cs46xx_dsp_remove_scb(chip,pcm_channel->src_scb);
- snd_assert (pcm_channel->src_slot >= 0 && pcm_channel->src_slot <= DSP_MAX_SRC_NR,
+ snd_assert (pcm_channel->src_slot >= 0 && pcm_channel->src_slot < DSP_MAX_SRC_NR,
return );
ins->src_scb_slots[pcm_channel->src_slot] = 0;
diff --git a/sound/pci/cs5535audio/Makefile b/sound/pci/cs5535audio/Makefile
index ad947b4c04cc..bb3d57e6a3cb 100644
--- a/sound/pci/cs5535audio/Makefile
+++ b/sound/pci/cs5535audio/Makefile
@@ -2,11 +2,8 @@
# Makefile for cs5535audio
#
-snd-cs5535audio-objs := cs5535audio.o cs5535audio_pcm.o
-
-ifeq ($(CONFIG_PM),y)
-snd-cs5535audio-objs += cs5535audio_pm.o
-endif
+snd-cs5535audio-y := cs5535audio.o cs5535audio_pcm.o
+snd-cs5535audio-$(CONFIG_PM) += cs5535audio_pm.o
# Toplevel Module Dependency
obj-$(CONFIG_SND_CS5535AUDIO) += snd-cs5535audio.o
diff --git a/sound/pci/cs5535audio/cs5535audio.c b/sound/pci/cs5535audio/cs5535audio.c
index b8e75ef9c1e6..2b35889787be 100644
--- a/sound/pci/cs5535audio/cs5535audio.c
+++ b/sound/pci/cs5535audio/cs5535audio.c
@@ -206,7 +206,6 @@ static void process_bm1_irq(struct cs5535audio *cs5535au)
static irqreturn_t snd_cs5535audio_interrupt(int irq, void *dev_id)
{
u16 acc_irq_stat;
- u8 bm_stat;
unsigned char count;
struct cs5535audio *cs5535au = dev_id;
@@ -217,7 +216,7 @@ static irqreturn_t snd_cs5535audio_interrupt(int irq, void *dev_id)
if (!acc_irq_stat)
return IRQ_NONE;
- for (count = 0; count < 10; count++) {
+ for (count = 0; count < 4; count++) {
if (acc_irq_stat & (1 << count)) {
switch (count) {
case IRQ_STS:
@@ -232,26 +231,9 @@ static irqreturn_t snd_cs5535audio_interrupt(int irq, void *dev_id)
case BM1_IRQ_STS:
process_bm1_irq(cs5535au);
break;
- case BM2_IRQ_STS:
- bm_stat = cs_readb(cs5535au, ACC_BM2_STATUS);
- break;
- case BM3_IRQ_STS:
- bm_stat = cs_readb(cs5535au, ACC_BM3_STATUS);
- break;
- case BM4_IRQ_STS:
- bm_stat = cs_readb(cs5535au, ACC_BM4_STATUS);
- break;
- case BM5_IRQ_STS:
- bm_stat = cs_readb(cs5535au, ACC_BM5_STATUS);
- break;
- case BM6_IRQ_STS:
- bm_stat = cs_readb(cs5535au, ACC_BM6_STATUS);
- break;
- case BM7_IRQ_STS:
- bm_stat = cs_readb(cs5535au, ACC_BM7_STATUS);
- break;
default:
- snd_printk(KERN_ERR "Unexpected irq src\n");
+ snd_printk(KERN_ERR "Unexpected irq src: "
+ "0x%x\n", acc_irq_stat);
break;
}
}
diff --git a/sound/pci/cs5535audio/cs5535audio.h b/sound/pci/cs5535audio/cs5535audio.h
index 4fd1f31a6cf9..66bae7664193 100644
--- a/sound/pci/cs5535audio/cs5535audio.h
+++ b/sound/pci/cs5535audio/cs5535audio.h
@@ -16,57 +16,28 @@
#define ACC_IRQ_STATUS 0x12
#define ACC_BM0_CMD 0x20
#define ACC_BM1_CMD 0x28
-#define ACC_BM2_CMD 0x30
-#define ACC_BM3_CMD 0x38
-#define ACC_BM4_CMD 0x40
-#define ACC_BM5_CMD 0x48
-#define ACC_BM6_CMD 0x50
-#define ACC_BM7_CMD 0x58
#define ACC_BM0_PRD 0x24
#define ACC_BM1_PRD 0x2C
-#define ACC_BM2_PRD 0x34
-#define ACC_BM3_PRD 0x3C
-#define ACC_BM4_PRD 0x44
-#define ACC_BM5_PRD 0x4C
-#define ACC_BM6_PRD 0x54
-#define ACC_BM7_PRD 0x5C
#define ACC_BM0_STATUS 0x21
#define ACC_BM1_STATUS 0x29
-#define ACC_BM2_STATUS 0x31
-#define ACC_BM3_STATUS 0x39
-#define ACC_BM4_STATUS 0x41
-#define ACC_BM5_STATUS 0x49
-#define ACC_BM6_STATUS 0x51
-#define ACC_BM7_STATUS 0x59
#define ACC_BM0_PNTR 0x60
#define ACC_BM1_PNTR 0x64
-#define ACC_BM2_PNTR 0x68
-#define ACC_BM3_PNTR 0x6C
-#define ACC_BM4_PNTR 0x70
-#define ACC_BM5_PNTR 0x74
-#define ACC_BM6_PNTR 0x78
-#define ACC_BM7_PNTR 0x7C
+
/* acc_codec bar0 reg bits */
/* ACC_IRQ_STATUS */
#define IRQ_STS 0
#define WU_IRQ_STS 1
#define BM0_IRQ_STS 2
#define BM1_IRQ_STS 3
-#define BM2_IRQ_STS 4
-#define BM3_IRQ_STS 5
-#define BM4_IRQ_STS 6
-#define BM5_IRQ_STS 7
-#define BM6_IRQ_STS 8
-#define BM7_IRQ_STS 9
/* ACC_BMX_STATUS */
#define EOP (1<<0)
#define BM_EOP_ERR (1<<1)
/* ACC_BMX_CTL */
-#define BM_CTL_EN 0x00000001
-#define BM_CTL_PAUSE 0x00000011
-#define BM_CTL_DIS 0x00000000
-#define BM_CTL_BYTE_ORD_LE 0x00000000
-#define BM_CTL_BYTE_ORD_BE 0x00000100
+#define BM_CTL_EN 0x01
+#define BM_CTL_PAUSE 0x03
+#define BM_CTL_DIS 0x00
+#define BM_CTL_BYTE_ORD_LE 0x00
+#define BM_CTL_BYTE_ORD_BE 0x04
/* cs5535 specific ac97 codec register defines */
#define CMD_MASK 0xFF00FFFF
#define CMD_NEW 0x00010000
@@ -106,7 +77,6 @@ struct cs5535audio_dma {
struct snd_pcm_substream *substream;
unsigned int buf_addr, buf_bytes;
unsigned int period_bytes, periods;
- int suspended;
u32 saved_prd;
};
diff --git a/sound/pci/cs5535audio/cs5535audio_pcm.c b/sound/pci/cs5535audio/cs5535audio_pcm.c
index 5450a9e8f133..21df0634af32 100644
--- a/sound/pci/cs5535audio/cs5535audio_pcm.c
+++ b/sound/pci/cs5535audio/cs5535audio_pcm.c
@@ -43,7 +43,6 @@ static struct snd_pcm_hardware snd_cs5535audio_playback =
SNDRV_PCM_INFO_BLOCK_TRANSFER |
SNDRV_PCM_INFO_MMAP_VALID |
SNDRV_PCM_INFO_PAUSE |
- SNDRV_PCM_INFO_SYNC_START |
SNDRV_PCM_INFO_RESUME
),
.formats = (
@@ -71,8 +70,7 @@ static struct snd_pcm_hardware snd_cs5535audio_capture =
SNDRV_PCM_INFO_MMAP |
SNDRV_PCM_INFO_INTERLEAVED |
SNDRV_PCM_INFO_BLOCK_TRANSFER |
- SNDRV_PCM_INFO_MMAP_VALID |
- SNDRV_PCM_INFO_SYNC_START
+ SNDRV_PCM_INFO_MMAP_VALID
),
.formats = (
SNDRV_PCM_FMTBIT_S16_LE
@@ -102,7 +100,6 @@ static int snd_cs5535audio_playback_open(struct snd_pcm_substream *substream)
runtime->hw = snd_cs5535audio_playback;
cs5535au->playback_substream = substream;
runtime->private_data = &(cs5535au->dmas[CS5535AUDIO_DMA_PLAYBACK]);
- snd_pcm_set_sync(substream);
if ((err = snd_pcm_hw_constraint_integer(runtime,
SNDRV_PCM_HW_PARAM_PERIODS)) < 0)
return err;
@@ -164,6 +161,7 @@ static int cs5535audio_build_dma_packets(struct cs5535audio *cs5535au,
jmpprd_addr = cpu_to_le32(lastdesc->addr +
(sizeof(struct cs5535audio_dma_desc)*periods));
+ dma->substream = substream;
dma->period_bytes = period_bytes;
dma->periods = periods;
spin_lock_irq(&cs5535au->reg_lock);
@@ -241,6 +239,7 @@ static void cs5535audio_clear_dma_packets(struct cs5535audio *cs5535au,
{
snd_dma_free_pages(&dma->desc_buf);
dma->desc_buf.area = NULL;
+ dma->substream = NULL;
}
static int snd_cs5535audio_hw_params(struct snd_pcm_substream *substream,
@@ -298,14 +297,12 @@ static int snd_cs5535audio_trigger(struct snd_pcm_substream *substream, int cmd)
break;
case SNDRV_PCM_TRIGGER_RESUME:
dma->ops->enable_dma(cs5535au);
- dma->suspended = 0;
break;
case SNDRV_PCM_TRIGGER_STOP:
dma->ops->disable_dma(cs5535au);
break;
case SNDRV_PCM_TRIGGER_SUSPEND:
dma->ops->disable_dma(cs5535au);
- dma->suspended = 1;
break;
default:
snd_printk(KERN_ERR "unhandled trigger\n");
@@ -348,7 +345,6 @@ static int snd_cs5535audio_capture_open(struct snd_pcm_substream *substream)
runtime->hw = snd_cs5535audio_capture;
cs5535au->capture_substream = substream;
runtime->private_data = &(cs5535au->dmas[CS5535AUDIO_DMA_CAPTURE]);
- snd_pcm_set_sync(substream);
if ((err = snd_pcm_hw_constraint_integer(runtime,
SNDRV_PCM_HW_PARAM_PERIODS)) < 0)
return err;
diff --git a/sound/pci/cs5535audio/cs5535audio_pm.c b/sound/pci/cs5535audio/cs5535audio_pm.c
index 3e4d198a4502..838708f6d45e 100644
--- a/sound/pci/cs5535audio/cs5535audio_pm.c
+++ b/sound/pci/cs5535audio/cs5535audio_pm.c
@@ -64,18 +64,21 @@ int snd_cs5535audio_suspend(struct pci_dev *pci, pm_message_t state)
int i;
snd_power_change_state(card, SNDRV_CTL_POWER_D3hot);
+ snd_pcm_suspend_all(cs5535au->pcm);
+ snd_ac97_suspend(cs5535au->ac97);
for (i = 0; i < NUM_CS5535AUDIO_DMAS; i++) {
struct cs5535audio_dma *dma = &cs5535au->dmas[i];
- if (dma && dma->substream && !dma->suspended)
+ if (dma && dma->substream)
dma->saved_prd = dma->ops->read_prd(cs5535au);
}
- snd_pcm_suspend_all(cs5535au->pcm);
- snd_ac97_suspend(cs5535au->ac97);
/* save important regs, then disable aclink in hw */
snd_cs5535audio_stop_hardware(cs5535au);
+ if (pci_save_state(pci)) {
+ printk(KERN_ERR "cs5535audio: pci_save_state failed!\n");
+ return -EIO;
+ }
pci_disable_device(pci);
- pci_save_state(pci);
pci_set_power_state(pci, pci_choose_state(pci, state));
return 0;
}
@@ -89,7 +92,12 @@ int snd_cs5535audio_resume(struct pci_dev *pci)
int i;
pci_set_power_state(pci, PCI_D0);
- pci_restore_state(pci);
+ if (pci_restore_state(pci) < 0) {
+ printk(KERN_ERR "cs5535audio: pci_restore_state failed, "
+ "disabling device\n");
+ snd_card_disconnect(card);
+ return -EIO;
+ }
if (pci_enable_device(pci) < 0) {
printk(KERN_ERR "cs5535audio: pci_enable_device failed, "
"disabling device\n");
@@ -112,17 +120,17 @@ int snd_cs5535audio_resume(struct pci_dev *pci)
if (!timeout)
snd_printk(KERN_ERR "Failure getting AC Link ready\n");
- /* we depend on ac97 to perform the codec power up */
- snd_ac97_resume(cs5535au->ac97);
/* set up rate regs, dma. actual initiation is done in trig */
for (i = 0; i < NUM_CS5535AUDIO_DMAS; i++) {
struct cs5535audio_dma *dma = &cs5535au->dmas[i];
- if (dma && dma->substream && dma->suspended) {
+ if (dma && dma->substream) {
dma->substream->ops->prepare(dma->substream);
dma->ops->setup_prd(cs5535au, dma->saved_prd);
}
}
-
+
+ /* we depend on ac97 to perform the codec power up */
+ snd_ac97_resume(cs5535au->ac97);
snd_power_change_state(card, SNDRV_CTL_POWER_D0);
return 0;
diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c
index f27b6a733b96..499ee1a5319d 100644
--- a/sound/pci/echoaudio/echoaudio.c
+++ b/sound/pci/echoaudio/echoaudio.c
@@ -1595,15 +1595,7 @@ static struct snd_kcontrol_new snd_echo_clock_source_switch __devinitdata = {
#ifdef ECHOCARD_HAS_PHANTOM_POWER
/******************* Phantom power switch *******************/
-static int snd_echo_phantom_power_info(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_info *uinfo)
-{
- uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
- uinfo->count = 1;
- uinfo->value.integer.min = 0;
- uinfo->value.integer.max = 1;
- return 0;
-}
+#define snd_echo_phantom_power_info snd_ctl_boolean_mono_info
static int snd_echo_phantom_power_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
@@ -1646,15 +1638,7 @@ static struct snd_kcontrol_new snd_echo_phantom_power_switch __devinitdata = {
#ifdef ECHOCARD_HAS_DIGITAL_IN_AUTOMUTE
/******************* Digital input automute switch *******************/
-static int snd_echo_automute_info(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_info *uinfo)
-{
- uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
- uinfo->count = 1;
- uinfo->value.integer.min = 0;
- uinfo->value.integer.max = 1;
- return 0;
-}
+#define snd_echo_automute_info snd_ctl_boolean_mono_info
static int snd_echo_automute_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
@@ -1695,18 +1679,7 @@ static struct snd_kcontrol_new snd_echo_automute_switch __devinitdata = {
/******************* VU-meters switch *******************/
-static int snd_echo_vumeters_switch_info(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_info *uinfo)
-{
- struct echoaudio *chip;
-
- chip = snd_kcontrol_chip(kcontrol);
- uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
- uinfo->count = 1;
- uinfo->value.integer.min = 0;
- uinfo->value.integer.max = 1;
- return 0;
-}
+#define snd_echo_vumeters_switch_info snd_ctl_boolean_mono_info
static int snd_echo_vumeters_switch_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
diff --git a/sound/pci/echoaudio/echoaudio_dsp.c b/sound/pci/echoaudio/echoaudio_dsp.c
index 42afa837d9b4..e6c100770392 100644
--- a/sound/pci/echoaudio/echoaudio_dsp.c
+++ b/sound/pci/echoaudio/echoaudio_dsp.c
@@ -43,11 +43,11 @@ static int wait_handshake(struct echoaudio *chip)
{
int i;
- /* Wait up to 10ms for the handshake from the DSP */
+ /* Wait up to 20ms for the handshake from the DSP */
for (i = 0; i < HANDSHAKE_TIMEOUT; i++) {
/* Look for the handshake value */
+ barrier();
if (chip->comm_page->handshake) {
- /*if (i) DE_ACT(("Handshake time: %d\n", i));*/
return 0;
}
udelay(1);
diff --git a/sound/pci/echoaudio/echoaudio_dsp.h b/sound/pci/echoaudio/echoaudio_dsp.h
index e55ee00991ac..e352f3ae292c 100644
--- a/sound/pci/echoaudio/echoaudio_dsp.h
+++ b/sound/pci/echoaudio/echoaudio_dsp.h
@@ -642,18 +642,18 @@ struct comm_page { /* Base Length*/
u32 flags; /* See Appendix A below 0x004 4 */
u32 unused; /* Unused entry 0x008 4 */
u32 sample_rate; /* Card sample rate in Hz 0x00c 4 */
- volatile u32 handshake; /* DSP command handshake 0x010 4 */
+ u32 handshake; /* DSP command handshake 0x010 4 */
u32 cmd_start; /* Chs. to start mask 0x014 4 */
u32 cmd_stop; /* Chs. to stop mask 0x018 4 */
u32 cmd_reset; /* Chs. to reset mask 0x01c 4 */
u16 audio_format[DSP_MAXPIPES]; /* Chs. audio format 0x020 32*2 */
struct sg_entry sglist_addr[DSP_MAXPIPES];
/* Chs. Physical sglist addrs 0x060 32*8 */
- volatile u32 position[DSP_MAXPIPES];
+ u32 position[DSP_MAXPIPES];
/* Positions for ea. ch. 0x160 32*4 */
- volatile s8 vu_meter[DSP_MAXPIPES];
+ s8 vu_meter[DSP_MAXPIPES];
/* VU meters 0x1e0 32*1 */
- volatile s8 peak_meter[DSP_MAXPIPES];
+ s8 peak_meter[DSP_MAXPIPES];
/* Peak meters 0x200 32*1 */
s8 line_out_level[DSP_MAXAUDIOOUTPUTS];
/* Output gain 0x220 16*1 */
@@ -665,7 +665,7 @@ struct comm_page { /* Base Length*/
/* Gina/Darla play filters - obsolete 0x3c0 168*4 */
u32 rec_coeff[MAX_REC_TAPS];
/* Gina/Darla record filters - obsolete 0x660 192*4 */
- volatile u16 midi_input[MIDI_IN_BUFFER_SIZE];
+ u16 midi_input[MIDI_IN_BUFFER_SIZE];
/* MIDI input data transfer buffer 0x960 256*2 */
u8 gd_clock_state; /* Chg Gina/Darla clock state 0xb60 1 */
u8 gd_spdif_status; /* Chg. Gina/Darla S/PDIF state 0xb61 1 */
@@ -674,11 +674,10 @@ struct comm_page { /* Base Length*/
u32 nominal_level_mask; /* -10 level enable mask 0xb64 4 */
u16 input_clock; /* Chg. Input clock state 0xb68 2 */
u16 output_clock; /* Chg. Output clock state 0xb6a 2 */
- volatile u32 status_clocks;
- /* Current Input clock state 0xb6c 4 */
+ u32 status_clocks; /* Current Input clock state 0xb6c 4 */
u32 ext_box_status; /* External box status 0xb70 4 */
u32 cmd_add_buffer; /* Pipes to add (obsolete) 0xb74 4 */
- volatile u32 midi_out_free_count;
+ u32 midi_out_free_count;
/* # of bytes free in MIDI output FIFO 0xb78 4 */
u32 unused2; /* Cyclic pipes 0xb7c 4 */
u32 control_register;
diff --git a/sound/pci/emu10k1/Makefile b/sound/pci/emu10k1/Makefile
index e521c38cef45..cf2d5636d8be 100644
--- a/sound/pci/emu10k1/Makefile
+++ b/sound/pci/emu10k1/Makefile
@@ -1,6 +1,6 @@
#
# Makefile for ALSA
-# Copyright (c) 2001 by Jaroslav Kysela <perex@suse.cz>
+# Copyright (c) 2001 by Jaroslav Kysela <perex@perex.cz>
#
snd-emu10k1-objs := emu10k1.o emu10k1_main.o \
diff --git a/sound/pci/emu10k1/emu10k1.c b/sound/pci/emu10k1/emu10k1.c
index 55caf341933a..9680caff90c8 100644
--- a/sound/pci/emu10k1/emu10k1.c
+++ b/sound/pci/emu10k1/emu10k1.c
@@ -1,6 +1,6 @@
/*
* The driver for the EMU10K1 (SB Live!) based soundcards
- * Copyright (c) by Jaroslav Kysela <perex@suse.cz>
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
*
* Copyright (c) by James Courtier-Dutton <James@superbug.demon.co.uk>
* Added support for Audigy 2 Value.
@@ -32,7 +32,7 @@
#include <sound/emu10k1.h>
#include <sound/initval.h>
-MODULE_AUTHOR("Jaroslav Kysela <perex@suse.cz>");
+MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
MODULE_DESCRIPTION("EMU10K1");
MODULE_LICENSE("GPL");
MODULE_SUPPORTED_DEVICE("{{Creative Labs,SB Live!/PCI512/E-mu APS},"
diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c
index 404ae1be0a4b..97c41d72a255 100644
--- a/sound/pci/emu10k1/emu10k1_main.c
+++ b/sound/pci/emu10k1/emu10k1_main.c
@@ -1,5 +1,5 @@
/*
- * Copyright (c) by Jaroslav Kysela <perex@suse.cz>
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
* Creative Labs, Inc.
* Routines for control of EMU10K1 chips
*
@@ -31,6 +31,8 @@
*
*/
+#include <linux/sched.h>
+#include <linux/kthread.h>
#include <sound/driver.h>
#include <linux/delay.h>
#include <linux/init.h>
@@ -702,6 +704,65 @@ static int snd_emu1010_load_firmware(struct snd_emu10k1 * emu, const char * file
return 0;
}
+int emu1010_firmware_thread(void *data) {
+ struct snd_emu10k1 * emu = data;
+ int tmp,tmp2;
+ int reg;
+ int err;
+
+ for (;;) {
+ /* Delay to allow Audio Dock to settle */
+ msleep(1000);
+ if (kthread_should_stop())
+ break;
+ snd_emu1010_fpga_read(emu, EMU_HANA_IRQ_STATUS, &tmp ); /* IRQ Status */
+ snd_emu1010_fpga_read(emu, EMU_HANA_OPTION_CARDS, &reg ); /* OPTIONS: Which cards are attached to the EMU */
+ if (reg & EMU_HANA_OPTION_DOCK_OFFLINE) {
+ /* Audio Dock attached */
+ /* Return to Audio Dock programming mode */
+ snd_printk(KERN_INFO "emu1010: Loading Audio Dock Firmware\n");
+ snd_emu1010_fpga_write(emu, EMU_HANA_FPGA_CONFIG, EMU_HANA_FPGA_CONFIG_AUDIODOCK );
+ if (emu->card_capabilities->emu1010 == 1) {
+ if ((err = snd_emu1010_load_firmware(emu, DOCK_FILENAME)) != 0) {
+ return err;
+ }
+ } else if (emu->card_capabilities->emu1010 == 2) {
+ if ((err = snd_emu1010_load_firmware(emu, MICRO_DOCK_FILENAME)) != 0) {
+ return err;
+ }
+ } else if (emu->card_capabilities->emu1010 == 3) {
+ if ((err = snd_emu1010_load_firmware(emu, MICRO_DOCK_FILENAME)) != 0) {
+ return err;
+ }
+ }
+
+ snd_emu1010_fpga_write(emu, EMU_HANA_FPGA_CONFIG, 0 );
+ snd_emu1010_fpga_read(emu, EMU_HANA_IRQ_STATUS, &reg );
+ snd_printk(KERN_INFO "emu1010: EMU_HANA+DOCK_IRQ_STATUS=0x%x\n",reg);
+ /* ID, should read & 0x7f = 0x55 when FPGA programmed. */
+ snd_emu1010_fpga_read(emu, EMU_HANA_ID, &reg );
+ snd_printk(KERN_INFO "emu1010: EMU_HANA+DOCK_ID=0x%x\n",reg);
+ if ((reg & 0x1f) != 0x15) {
+ /* FPGA failed to be programmed */
+ snd_printk(KERN_INFO "emu1010: Loading Audio Dock Firmware file failed, reg=0x%x\n", reg);
+ return 0;
+ return -ENODEV;
+ }
+ snd_printk(KERN_INFO "emu1010: Audio Dock Firmware loaded\n");
+ snd_emu1010_fpga_read(emu, EMU_DOCK_MAJOR_REV, &tmp );
+ snd_emu1010_fpga_read(emu, EMU_DOCK_MINOR_REV, &tmp2 );
+ snd_printk("Audio Dock ver:%d.%d\n",tmp ,tmp2);
+ /* Sync clocking between 1010 and Dock */
+ /* Allow DLL to settle */
+ msleep(10);
+ /* Unmute all. Default is muted after a firmware load */
+ snd_emu1010_fpga_write(emu, EMU_HANA_UNMUTE, EMU_UNMUTE );
+ break;
+ }
+ }
+ return 0;
+}
+
/*
* EMU-1010 - details found out from this driver, official MS Win drivers,
* testing the card:
@@ -817,8 +878,16 @@ static int snd_emu10k1_emu1010_init(struct snd_emu10k1 * emu)
snd_emu1010_fpga_read(emu, EMU_HANA_OPTION_CARDS, &reg );
snd_printk(KERN_INFO "emu1010: Card options=0x%x\n",reg);
snd_emu1010_fpga_read(emu, EMU_HANA_OPTICAL_TYPE, &tmp );
- /* ADAT input. */
- snd_emu1010_fpga_write(emu, EMU_HANA_OPTICAL_TYPE, 0x01 );
+ /* Optical -> ADAT I/O */
+ /* 0 : SPDIF
+ * 1 : ADAT
+ */
+ emu->emu1010.optical_in = 1; /* IN_ADAT */
+ emu->emu1010.optical_out = 1; /* IN_ADAT */
+ tmp = 0;
+ tmp = (emu->emu1010.optical_in ? EMU_HANA_OPTICAL_IN_ADAT : 0) |
+ (emu->emu1010.optical_out ? EMU_HANA_OPTICAL_OUT_ADAT : 0);
+ snd_emu1010_fpga_write(emu, EMU_HANA_OPTICAL_TYPE, tmp );
snd_emu1010_fpga_read(emu, EMU_HANA_ADC_PADS, &tmp );
/* Set no attenuation on Audio Dock pads. */
snd_emu1010_fpga_write(emu, EMU_HANA_ADC_PADS, 0x00 );
@@ -1004,49 +1073,12 @@ static int snd_emu10k1_emu1010_init(struct snd_emu10k1 * emu)
snd_emu1010_fpga_read(emu, EMU_HANA_SPDIF_MODE, &tmp );
snd_emu1010_fpga_write(emu, EMU_HANA_SPDIF_MODE, 0x10 ); /* SPDIF Format spdif (or 0x11 for aes/ebu) */
- /* Delay to allow Audio Dock to settle */
- msleep(100);
- snd_emu1010_fpga_read(emu, EMU_HANA_IRQ_STATUS, &tmp ); /* IRQ Status */
- snd_emu1010_fpga_read(emu, EMU_HANA_OPTION_CARDS, &reg ); /* OPTIONS: Which cards are attached to the EMU */
- /* FIXME: The loading of this should be able to happen any time,
- * as the user can plug/unplug it at any time
- */
- if (reg & (EMU_HANA_OPTION_DOCK_ONLINE | EMU_HANA_OPTION_DOCK_OFFLINE) ) {
- /* Audio Dock attached */
- /* Return to Audio Dock programming mode */
- snd_printk(KERN_INFO "emu1010: Loading Audio Dock Firmware\n");
- snd_emu1010_fpga_write(emu, EMU_HANA_FPGA_CONFIG, EMU_HANA_FPGA_CONFIG_AUDIODOCK );
- if (emu->card_capabilities->emu1010 == 1) {
- if ((err = snd_emu1010_load_firmware(emu, DOCK_FILENAME)) != 0) {
- return err;
- }
- } else if (emu->card_capabilities->emu1010 == 2) {
- if ((err = snd_emu1010_load_firmware(emu, MICRO_DOCK_FILENAME)) != 0) {
- return err;
- }
- } else if (emu->card_capabilities->emu1010 == 3) {
- if ((err = snd_emu1010_load_firmware(emu, MICRO_DOCK_FILENAME)) != 0) {
- return err;
- }
- }
+ /* Start Micro/Audio Dock firmware loader thread */
+ emu->emu1010.firmware_thread = kthread_create(&emu1010_firmware_thread,
+ emu,
+ "emu1010_firmware");
+ wake_up_process(emu->emu1010.firmware_thread);
- snd_emu1010_fpga_write(emu, EMU_HANA_FPGA_CONFIG, 0 );
- snd_emu1010_fpga_read(emu, EMU_HANA_IRQ_STATUS, &reg );
- snd_printk(KERN_INFO "emu1010: EMU_HANA+DOCK_IRQ_STATUS=0x%x\n",reg);
- /* ID, should read & 0x7f = 0x55 when FPGA programmed. */
- snd_emu1010_fpga_read(emu, EMU_HANA_ID, &reg );
- snd_printk(KERN_INFO "emu1010: EMU_HANA+DOCK_ID=0x%x\n",reg);
- if ((reg & 0x3f) != 0x15) {
- /* FPGA failed to be programmed */
- snd_printk(KERN_INFO "emu1010: Loading Audio Dock Firmware file failed, reg=0x%x\n", reg);
- return 0;
- return -ENODEV;
- }
- snd_printk(KERN_INFO "emu1010: Audio Dock Firmware loaded\n");
- snd_emu1010_fpga_read(emu, EMU_DOCK_MAJOR_REV, &tmp );
- snd_emu1010_fpga_read(emu, EMU_DOCK_MINOR_REV, &tmp2 );
- snd_printk("Audio Dock ver:%d.%d\n",tmp ,tmp2);
- }
#if 0
snd_emu1010_fpga_link_dst_src_write(emu,
EMU_DST_HAMOA_DAC_LEFT1, EMU_SRC_ALICE_EMU32B + 2); /* ALICE2 bus 0xa2 */
@@ -1132,7 +1164,7 @@ static int snd_emu10k1_emu1010_init(struct snd_emu10k1 * emu)
emu->emu1010.output_source[23] = 28;
/* TEMP: Select SPDIF in/out */
- snd_emu1010_fpga_write(emu, EMU_HANA_OPTICAL_TYPE, 0x0); /* Output spdif */
+ //snd_emu1010_fpga_write(emu, EMU_HANA_OPTICAL_TYPE, 0x0); /* Output spdif */
/* TEMP: Select 48kHz SPDIF out */
snd_emu1010_fpga_write(emu, EMU_HANA_UNMUTE, 0x0); /* Mute all */
@@ -1173,6 +1205,7 @@ static int snd_emu10k1_free(struct snd_emu10k1 *emu)
if (emu->card_capabilities->emu1010) {
/* Disable 48Volt power to Audio Dock */
snd_emu1010_fpga_write(emu, EMU_HANA_DOCK_PWR, 0 );
+ kthread_stop(emu->emu1010.firmware_thread);
}
if (emu->memhdr)
snd_util_memhdr_free(emu->memhdr);
@@ -1722,8 +1755,9 @@ int __devinit snd_emu10k1_create(struct snd_card *card,
goto error;
}
- emu->page_ptr_table = (void **)vmalloc(emu->max_cache_pages * sizeof(void*));
- emu->page_addr_table = (unsigned long*)vmalloc(emu->max_cache_pages * sizeof(unsigned long));
+ emu->page_ptr_table = vmalloc(emu->max_cache_pages * sizeof(void *));
+ emu->page_addr_table = vmalloc(emu->max_cache_pages *
+ sizeof(unsigned long));
if (emu->page_ptr_table == NULL || emu->page_addr_table == NULL) {
err = -ENOMEM;
goto error;
diff --git a/sound/pci/emu10k1/emu10k1x.c b/sound/pci/emu10k1/emu10k1x.c
index e4af7a9b808c..1ec7ebaff9e9 100644
--- a/sound/pci/emu10k1/emu10k1x.c
+++ b/sound/pci/emu10k1/emu10k1x.c
@@ -1062,14 +1062,7 @@ static int __devinit snd_emu10k1x_proc_init(struct emu10k1x * emu)
return 0;
}
-static int snd_emu10k1x_shared_spdif_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
-{
- uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
- uinfo->count = 1;
- uinfo->value.integer.min = 0;
- uinfo->value.integer.max = 1;
- return 0;
-}
+#define snd_emu10k1x_shared_spdif_info snd_ctl_boolean_mono_info
static int snd_emu10k1x_shared_spdif_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
diff --git a/sound/pci/emu10k1/emufx.c b/sound/pci/emu10k1/emufx.c
index 7206c0fa06f2..9bf1cd592199 100644
--- a/sound/pci/emu10k1/emufx.c
+++ b/sound/pci/emu10k1/emufx.c
@@ -1,5 +1,5 @@
/*
- * Copyright (c) by Jaroslav Kysela <perex@suse.cz>
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
* Creative Labs, Inc.
* Routines for effect processor FX8010
*
@@ -642,10 +642,8 @@ snd_emu10k1_look_for_ctl(struct snd_emu10k1 *emu, struct snd_ctl_elem_id *id)
{
struct snd_emu10k1_fx8010_ctl *ctl;
struct snd_kcontrol *kcontrol;
- struct list_head *list;
-
- list_for_each(list, &emu->fx8010.gpr_ctl) {
- ctl = emu10k1_gpr_ctl(list);
+
+ list_for_each_entry(ctl, &emu->fx8010.gpr_ctl, list) {
kcontrol = ctl->kcontrol;
if (kcontrol->id.iface == id->iface &&
!strcmp(kcontrol->id.name, id->name) &&
@@ -895,14 +893,12 @@ static int snd_emu10k1_list_controls(struct snd_emu10k1 *emu,
struct snd_emu10k1_fx8010_control_gpr *gctl;
struct snd_emu10k1_fx8010_ctl *ctl;
struct snd_ctl_elem_id *id;
- struct list_head *list;
gctl = kmalloc(sizeof(*gctl), GFP_KERNEL);
if (! gctl)
return -ENOMEM;
- list_for_each(list, &emu->fx8010.gpr_ctl) {
- ctl = emu10k1_gpr_ctl(list);
+ list_for_each_entry(ctl, &emu->fx8010.gpr_ctl, list) {
total++;
if (icode->gpr_list_controls &&
i < icode->gpr_list_control_count) {
@@ -1207,7 +1203,7 @@ static int __devinit _snd_emu10k1_audigy_init_efx(struct snd_emu10k1 *emu)
A_OP(icode, &ptr, iMAC0, A_GPR(playback+1), A_C_00000000, A_GPR(gpr+1), A_FXBUS(FXBUS_PCM_RIGHT_FRONT));
snd_emu10k1_init_stereo_control(&controls[nctl++], "PCM Front Playback Volume", gpr, 100);
gpr += 2;
-
+
/* PCM Surround Playback (independent from stereo mix) */
A_OP(icode, &ptr, iMAC0, A_GPR(playback+2), A_C_00000000, A_GPR(gpr), A_FXBUS(FXBUS_PCM_LEFT_REAR));
A_OP(icode, &ptr, iMAC0, A_GPR(playback+3), A_C_00000000, A_GPR(gpr+1), A_FXBUS(FXBUS_PCM_RIGHT_REAR));
@@ -1267,8 +1263,16 @@ A_OP(icode, &ptr, iMAC0, A_GPR(var), A_GPR(var), A_GPR(vol), A_EXTIN(input))
/* emu1212 DSP 0 and DSP 1 Capture */
if (emu->card_capabilities->emu1010) {
- A_OP(icode, &ptr, iMAC0, A_GPR(capture+0), A_GPR(capture+0), A_GPR(gpr), A_P16VIN(0x0));
- A_OP(icode, &ptr, iMAC0, A_GPR(capture+1), A_GPR(capture+1), A_GPR(gpr+1), A_P16VIN(0x1));
+ if (emu->card_capabilities->ca0108_chip) {
+ /* Note:JCD:No longer bit shift lower 16bits to upper 16bits of 32bit value. */
+ A_OP(icode, &ptr, iMACINT0, A_GPR(tmp), A_C_00000000, A3_EMU32IN(0x0), A_C_00000001);
+ A_OP(icode, &ptr, iMAC0, A_GPR(capture+0), A_GPR(capture+0), A_GPR(gpr), A_GPR(tmp));
+ A_OP(icode, &ptr, iMACINT0, A_GPR(tmp), A_C_00000000, A3_EMU32IN(0x1), A_C_00000001);
+ A_OP(icode, &ptr, iMAC0, A_GPR(capture+1), A_GPR(capture+1), A_GPR(gpr), A_GPR(tmp));
+ } else {
+ A_OP(icode, &ptr, iMAC0, A_GPR(capture+0), A_GPR(capture+0), A_GPR(gpr), A_P16VIN(0x0));
+ A_OP(icode, &ptr, iMAC0, A_GPR(capture+1), A_GPR(capture+1), A_GPR(gpr+1), A_P16VIN(0x1));
+ }
snd_emu10k1_init_stereo_control(&controls[nctl++], "EMU Capture Volume", gpr, 0);
gpr += 2;
}
@@ -1516,7 +1520,11 @@ A_OP(icode, &ptr, iMAC0, A_GPR(var), A_GPR(var), A_GPR(vol), A_EXTIN(input))
/* EMU1010 Outputs from PCM Front, Rear, Center, LFE, Side */
snd_printk("EMU outputs on\n");
for (z = 0; z < 8; z++) {
- A_OP(icode, &ptr, iACC3, A_EMU32OUTL(z), A_GPR(playback + SND_EMU10K1_PLAYBACK_CHANNELS + z), A_C_00000000, A_C_00000000);
+ if (emu->card_capabilities->ca0108_chip) {
+ A_OP(icode, &ptr, iACC3, A3_EMU32OUT(z), A_GPR(playback + SND_EMU10K1_PLAYBACK_CHANNELS + z), A_C_00000000, A_C_00000000);
+ } else {
+ A_OP(icode, &ptr, iACC3, A_EMU32OUTL(z), A_GPR(playback + SND_EMU10K1_PLAYBACK_CHANNELS + z), A_C_00000000, A_C_00000000);
+ }
}
}
@@ -1557,106 +1565,116 @@ A_OP(icode, &ptr, iMAC0, A_GPR(var), A_GPR(var), A_GPR(vol), A_EXTIN(input))
#endif
if (emu->card_capabilities->emu1010) {
- snd_printk("EMU inputs on\n");
- /* Capture 16 (originally 8) channels of S32_LE sound */
-
- /* printk("emufx.c: gpr=0x%x, tmp=0x%x\n",gpr, tmp); */
- /* For the EMU1010: How to get 32bit values from the DSP. High 16bits into L, low 16bits into R. */
- /* A_P16VIN(0) is delayed by one sample,
- * so all other A_P16VIN channels will need to also be delayed
- */
- /* Left ADC in. 1 of 2 */
- snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_P16VIN(0x0), A_FXBUS2(0) );
- /* Right ADC in 1 of 2 */
- gpr_map[gpr++] = 0x00000000;
- /* Delaying by one sample: instead of copying the input
- * value A_P16VIN to output A_FXBUS2 as in the first channel,
- * we use an auxiliary register, delaying the value by one
- * sample
- */
- snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(2) );
- A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x1), A_C_00000000, A_C_00000000);
- gpr_map[gpr++] = 0x00000000;
- snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(4) );
- A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x2), A_C_00000000, A_C_00000000);
- gpr_map[gpr++] = 0x00000000;
- snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(6) );
- A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x3), A_C_00000000, A_C_00000000);
- /* For 96kHz mode */
- /* Left ADC in. 2 of 2 */
- gpr_map[gpr++] = 0x00000000;
- snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(0x8) );
- A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x4), A_C_00000000, A_C_00000000);
- /* Right ADC in 2 of 2 */
- gpr_map[gpr++] = 0x00000000;
- snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(0xa) );
- A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x5), A_C_00000000, A_C_00000000);
- gpr_map[gpr++] = 0x00000000;
- snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(0xc) );
- A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x6), A_C_00000000, A_C_00000000);
- gpr_map[gpr++] = 0x00000000;
- snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(0xe) );
- A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x7), A_C_00000000, A_C_00000000);
- /* Pavel Hofman - we still have voices, A_FXBUS2s, and
- * A_P16VINs available -
- * let's add 8 more capture channels - total of 16
- */
- gpr_map[gpr++] = 0x00000000;
- snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp,
- bit_shifter16,
- A_GPR(gpr - 1),
- A_FXBUS2(0x10));
- A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x8),
- A_C_00000000, A_C_00000000);
- gpr_map[gpr++] = 0x00000000;
- snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp,
- bit_shifter16,
- A_GPR(gpr - 1),
- A_FXBUS2(0x12));
- A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x9),
- A_C_00000000, A_C_00000000);
- gpr_map[gpr++] = 0x00000000;
- snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp,
- bit_shifter16,
- A_GPR(gpr - 1),
- A_FXBUS2(0x14));
- A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0xa),
- A_C_00000000, A_C_00000000);
- gpr_map[gpr++] = 0x00000000;
- snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp,
- bit_shifter16,
- A_GPR(gpr - 1),
- A_FXBUS2(0x16));
- A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0xb),
- A_C_00000000, A_C_00000000);
- gpr_map[gpr++] = 0x00000000;
- snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp,
- bit_shifter16,
- A_GPR(gpr - 1),
- A_FXBUS2(0x18));
- A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0xc),
- A_C_00000000, A_C_00000000);
- gpr_map[gpr++] = 0x00000000;
- snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp,
- bit_shifter16,
- A_GPR(gpr - 1),
- A_FXBUS2(0x1a));
- A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0xd),
- A_C_00000000, A_C_00000000);
- gpr_map[gpr++] = 0x00000000;
- snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp,
- bit_shifter16,
- A_GPR(gpr - 1),
- A_FXBUS2(0x1c));
- A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0xe),
- A_C_00000000, A_C_00000000);
- gpr_map[gpr++] = 0x00000000;
- snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp,
- bit_shifter16,
- A_GPR(gpr - 1),
- A_FXBUS2(0x1e));
- A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0xf),
- A_C_00000000, A_C_00000000);
+ if (emu->card_capabilities->ca0108_chip) {
+ snd_printk("EMU2 inputs on\n");
+ for (z = 0; z < 0x10; z++) {
+ snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp,
+ bit_shifter16,
+ A3_EMU32IN(z),
+ A_FXBUS2(z*2) );
+ }
+ } else {
+ snd_printk("EMU inputs on\n");
+ /* Capture 16 (originally 8) channels of S32_LE sound */
+
+ /* printk("emufx.c: gpr=0x%x, tmp=0x%x\n",gpr, tmp); */
+ /* For the EMU1010: How to get 32bit values from the DSP. High 16bits into L, low 16bits into R. */
+ /* A_P16VIN(0) is delayed by one sample,
+ * so all other A_P16VIN channels will need to also be delayed
+ */
+ /* Left ADC in. 1 of 2 */
+ snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_P16VIN(0x0), A_FXBUS2(0) );
+ /* Right ADC in 1 of 2 */
+ gpr_map[gpr++] = 0x00000000;
+ /* Delaying by one sample: instead of copying the input
+ * value A_P16VIN to output A_FXBUS2 as in the first channel,
+ * we use an auxiliary register, delaying the value by one
+ * sample
+ */
+ snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(2) );
+ A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x1), A_C_00000000, A_C_00000000);
+ gpr_map[gpr++] = 0x00000000;
+ snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(4) );
+ A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x2), A_C_00000000, A_C_00000000);
+ gpr_map[gpr++] = 0x00000000;
+ snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(6) );
+ A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x3), A_C_00000000, A_C_00000000);
+ /* For 96kHz mode */
+ /* Left ADC in. 2 of 2 */
+ gpr_map[gpr++] = 0x00000000;
+ snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(0x8) );
+ A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x4), A_C_00000000, A_C_00000000);
+ /* Right ADC in 2 of 2 */
+ gpr_map[gpr++] = 0x00000000;
+ snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(0xa) );
+ A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x5), A_C_00000000, A_C_00000000);
+ gpr_map[gpr++] = 0x00000000;
+ snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(0xc) );
+ A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x6), A_C_00000000, A_C_00000000);
+ gpr_map[gpr++] = 0x00000000;
+ snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(0xe) );
+ A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x7), A_C_00000000, A_C_00000000);
+ /* Pavel Hofman - we still have voices, A_FXBUS2s, and
+ * A_P16VINs available -
+ * let's add 8 more capture channels - total of 16
+ */
+ gpr_map[gpr++] = 0x00000000;
+ snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp,
+ bit_shifter16,
+ A_GPR(gpr - 1),
+ A_FXBUS2(0x10));
+ A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x8),
+ A_C_00000000, A_C_00000000);
+ gpr_map[gpr++] = 0x00000000;
+ snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp,
+ bit_shifter16,
+ A_GPR(gpr - 1),
+ A_FXBUS2(0x12));
+ A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x9),
+ A_C_00000000, A_C_00000000);
+ gpr_map[gpr++] = 0x00000000;
+ snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp,
+ bit_shifter16,
+ A_GPR(gpr - 1),
+ A_FXBUS2(0x14));
+ A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0xa),
+ A_C_00000000, A_C_00000000);
+ gpr_map[gpr++] = 0x00000000;
+ snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp,
+ bit_shifter16,
+ A_GPR(gpr - 1),
+ A_FXBUS2(0x16));
+ A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0xb),
+ A_C_00000000, A_C_00000000);
+ gpr_map[gpr++] = 0x00000000;
+ snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp,
+ bit_shifter16,
+ A_GPR(gpr - 1),
+ A_FXBUS2(0x18));
+ A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0xc),
+ A_C_00000000, A_C_00000000);
+ gpr_map[gpr++] = 0x00000000;
+ snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp,
+ bit_shifter16,
+ A_GPR(gpr - 1),
+ A_FXBUS2(0x1a));
+ A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0xd),
+ A_C_00000000, A_C_00000000);
+ gpr_map[gpr++] = 0x00000000;
+ snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp,
+ bit_shifter16,
+ A_GPR(gpr - 1),
+ A_FXBUS2(0x1c));
+ A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0xe),
+ A_C_00000000, A_C_00000000);
+ gpr_map[gpr++] = 0x00000000;
+ snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp,
+ bit_shifter16,
+ A_GPR(gpr - 1),
+ A_FXBUS2(0x1e));
+ A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0xf),
+ A_C_00000000, A_C_00000000);
+ }
#if 0
for (z = 4; z < 8; z++) {
@@ -2418,14 +2436,13 @@ static void copy_string(char *dst, char *src, char *null, int idx)
strcpy(dst, src);
}
-static int snd_emu10k1_fx8010_info(struct snd_emu10k1 *emu,
+static void snd_emu10k1_fx8010_info(struct snd_emu10k1 *emu,
struct snd_emu10k1_fx8010_info *info)
{
char **fxbus, **extin, **extout;
unsigned short fxbus_mask, extin_mask, extout_mask;
int res;
- memset(info, 0, sizeof(info));
info->internal_tram_size = emu->fx8010.itram_size;
info->external_tram_size = emu->fx8010.etram_pages.bytes / 2;
fxbus = fxbuses;
@@ -2442,7 +2459,6 @@ static int snd_emu10k1_fx8010_info(struct snd_emu10k1 *emu,
for (res = 16; res < 32; res++, extout++)
copy_string(info->extout_names[res], extout_mask & (1 << res) ? *extout : NULL, "Unused", res);
info->gpr_controls = emu->fx8010.gpr_count;
- return 0;
}
static int snd_emu10k1_fx8010_ioctl(struct snd_hwdep * hw, struct file *file, unsigned int cmd, unsigned long arg)
@@ -2463,10 +2479,7 @@ static int snd_emu10k1_fx8010_ioctl(struct snd_hwdep * hw, struct file *file, un
info = kmalloc(sizeof(*info), GFP_KERNEL);
if (!info)
return -ENOMEM;
- if ((res = snd_emu10k1_fx8010_info(emu, info)) < 0) {
- kfree(info);
- return res;
- }
+ snd_emu10k1_fx8010_info(emu, info);
if (copy_to_user(argp, info, sizeof(*info))) {
kfree(info);
return -EFAULT;
diff --git a/sound/pci/emu10k1/emumixer.c b/sound/pci/emu10k1/emumixer.c
index 7b2c1dcc5337..54a2034d8edd 100644
--- a/sound/pci/emu10k1/emumixer.c
+++ b/sound/pci/emu10k1/emumixer.c
@@ -1,5 +1,5 @@
/*
- * Copyright (c) by Jaroslav Kysela <perex@suse.cz>,
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>,
* Takashi Iwai <tiwai@suse.de>
* Creative Labs, Inc.
* Routines for control of EMU10K1 chips / mixer routines
@@ -400,15 +400,7 @@ static struct snd_kcontrol_new snd_emu1010_input_enum_ctls[] __devinitdata = {
-
-static int snd_emu1010_adc_pads_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
-{
- uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
- uinfo->count = 1;
- uinfo->value.integer.min = 0;
- uinfo->value.integer.max = 1;
- return 0;
-}
+#define snd_emu1010_adc_pads_info snd_ctl_boolean_mono_info
static int snd_emu1010_adc_pads_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
{
@@ -456,14 +448,7 @@ static struct snd_kcontrol_new snd_emu1010_adc_pads[] __devinitdata = {
EMU1010_ADC_PADS("ADC1 14dB PAD 0202 Capture Switch", EMU_HANA_0202_ADC_PAD1),
};
-static int snd_emu1010_dac_pads_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
-{
- uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
- uinfo->count = 1;
- uinfo->value.integer.min = 0;
- uinfo->value.integer.max = 1;
- return 0;
-}
+#define snd_emu1010_dac_pads_info snd_ctl_boolean_mono_info
static int snd_emu1010_dac_pads_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
{
@@ -516,17 +501,19 @@ static struct snd_kcontrol_new snd_emu1010_dac_pads[] __devinitdata = {
static int snd_emu1010_internal_clock_info(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
- static char *texts[2] = {
- "44100", "48000"
+ static char *texts[4] = {
+ "44100", "48000", "SPDIF", "ADAT"
};
-
+
uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
uinfo->count = 1;
- uinfo->value.enumerated.items = 2;
- if (uinfo->value.enumerated.item > 1)
- uinfo->value.enumerated.item = 1;
+ uinfo->value.enumerated.items = 4;
+ if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items)
+ uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1;
strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
return 0;
+
+
}
static int snd_emu1010_internal_clock_get(struct snd_kcontrol *kcontrol,
@@ -584,6 +571,44 @@ static int snd_emu1010_internal_clock_put(struct snd_kcontrol *kcontrol,
/* Unmute all */
snd_emu1010_fpga_write(emu, EMU_HANA_UNMUTE, EMU_UNMUTE );
break;
+
+ case 2: /* Take clock from S/PDIF IN */
+ /* Mute all */
+ snd_emu1010_fpga_write(emu, EMU_HANA_UNMUTE, EMU_MUTE );
+ /* Default fallback clock 48kHz */
+ snd_emu1010_fpga_write(emu, EMU_HANA_DEFCLOCK, EMU_HANA_DEFCLOCK_48K );
+ /* Word Clock source, sync to S/PDIF input */
+ snd_emu1010_fpga_write(emu, EMU_HANA_WCLOCK,
+ EMU_HANA_WCLOCK_HANA_SPDIF_IN | EMU_HANA_WCLOCK_1X );
+ /* Set LEDs on Audio Dock */
+ snd_emu1010_fpga_write(emu, EMU_HANA_DOCK_LEDS_2,
+ EMU_HANA_DOCK_LEDS_2_EXT | EMU_HANA_DOCK_LEDS_2_LOCK );
+ /* FIXME: We should set EMU_HANA_DOCK_LEDS_2_LOCK only when clock signal is present and valid */
+ /* Allow DLL to settle */
+ msleep(10);
+ /* Unmute all */
+ snd_emu1010_fpga_write(emu, EMU_HANA_UNMUTE, EMU_UNMUTE );
+ break;
+
+ case 3:
+ /* Take clock from ADAT IN */
+ /* Mute all */
+ snd_emu1010_fpga_write(emu, EMU_HANA_UNMUTE, EMU_MUTE );
+ /* Default fallback clock 48kHz */
+ snd_emu1010_fpga_write(emu, EMU_HANA_DEFCLOCK, EMU_HANA_DEFCLOCK_48K );
+ /* Word Clock source, sync to ADAT input */
+ snd_emu1010_fpga_write(emu, EMU_HANA_WCLOCK,
+ EMU_HANA_WCLOCK_HANA_ADAT_IN | EMU_HANA_WCLOCK_1X );
+ /* Set LEDs on Audio Dock */
+ snd_emu1010_fpga_write(emu, EMU_HANA_DOCK_LEDS_2, EMU_HANA_DOCK_LEDS_2_EXT | EMU_HANA_DOCK_LEDS_2_LOCK );
+ /* FIXME: We should set EMU_HANA_DOCK_LEDS_2_LOCK only when clock signal is present and valid */
+ /* Allow DLL to settle */
+ msleep(10);
+ /* Unmute all */
+ snd_emu1010_fpga_write(emu, EMU_HANA_UNMUTE, EMU_UNMUTE );
+
+
+ break;
}
}
return change;
@@ -871,7 +896,7 @@ static struct snd_kcontrol_new snd_emu10k1_spdif_mask_control =
.access = SNDRV_CTL_ELEM_ACCESS_READ,
.iface = SNDRV_CTL_ELEM_IFACE_PCM,
.name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,MASK),
- .count = 4,
+ .count = 3,
.info = snd_emu10k1_spdif_info,
.get = snd_emu10k1_spdif_get_mask
};
@@ -880,7 +905,7 @@ static struct snd_kcontrol_new snd_emu10k1_spdif_control =
{
.iface = SNDRV_CTL_ELEM_IFACE_PCM,
.name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,DEFAULT),
- .count = 4,
+ .count = 3,
.info = snd_emu10k1_spdif_info,
.get = snd_emu10k1_spdif_get,
.put = snd_emu10k1_spdif_put
@@ -1326,14 +1351,7 @@ static struct snd_kcontrol_new snd_emu10k1_efx_attn_control =
.put = snd_emu10k1_efx_attn_put
};
-static int snd_emu10k1_shared_spdif_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
-{
- uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
- uinfo->count = 1;
- uinfo->value.integer.min = 0;
- uinfo->value.integer.max = 1;
- return 0;
-}
+#define snd_emu10k1_shared_spdif_info snd_ctl_boolean_mono_info
static int snd_emu10k1_shared_spdif_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
diff --git a/sound/pci/emu10k1/emumpu401.c b/sound/pci/emu10k1/emumpu401.c
index 950c6bcd6b7d..04c7cf703531 100644
--- a/sound/pci/emu10k1/emumpu401.c
+++ b/sound/pci/emu10k1/emumpu401.c
@@ -1,5 +1,5 @@
/*
- * Copyright (c) by Jaroslav Kysela <perex@suse.cz>
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
* Routines for control of EMU10K1 MPU-401 in UART mode
*
*
diff --git a/sound/pci/emu10k1/emupcm.c b/sound/pci/emu10k1/emupcm.c
index eda5cb373ded..5ce5befc701b 100644
--- a/sound/pci/emu10k1/emupcm.c
+++ b/sound/pci/emu10k1/emupcm.c
@@ -1,5 +1,5 @@
/*
- * Copyright (c) by Jaroslav Kysela <perex@suse.cz>
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
* Creative Labs, Inc.
* Routines for control of EMU10K1 chips / PCM routines
* Multichannel PCM support Copyright (c) Lee Revell <rlrevell@joe-job.com>
diff --git a/sound/pci/emu10k1/emuproc.c b/sound/pci/emu10k1/emuproc.c
index 2c1585991bc8..c3fb10e81c9e 100644
--- a/sound/pci/emu10k1/emuproc.c
+++ b/sound/pci/emu10k1/emuproc.c
@@ -1,5 +1,5 @@
/*
- * Copyright (c) by Jaroslav Kysela <perex@suse.cz>
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
* Creative Labs, Inc.
* Routines for control of EMU10K1 chips / proc interface routines
*
@@ -240,8 +240,42 @@ static void snd_emu10k1_proc_spdif_read(struct snd_info_entry *entry,
struct snd_info_buffer *buffer)
{
struct snd_emu10k1 *emu = entry->private_data;
- snd_emu10k1_proc_spdif_status(emu, buffer, "CD-ROM S/PDIF In", CDCS, CDSRCS);
- snd_emu10k1_proc_spdif_status(emu, buffer, "Optical or Coax S/PDIF In", GPSCS, GPSRCS);
+ u32 value;
+ u32 value2;
+ unsigned long flags;
+ u32 rate;
+
+ if (emu->card_capabilities->emu1010) {
+ spin_lock_irqsave(&emu->emu_lock, flags);
+ snd_emu1010_fpga_read(emu, 0x38, &value);
+ spin_unlock_irqrestore(&emu->emu_lock, flags);
+ if ((value & 0x1) == 0) {
+ spin_lock_irqsave(&emu->emu_lock, flags);
+ snd_emu1010_fpga_read(emu, 0x2a, &value);
+ snd_emu1010_fpga_read(emu, 0x2b, &value2);
+ spin_unlock_irqrestore(&emu->emu_lock, flags);
+ rate = 0x1770000 / (((value << 5) | value2)+1);
+ snd_iprintf(buffer, "ADAT Locked : %u\n", rate);
+ } else {
+ snd_iprintf(buffer, "ADAT Unlocked\n");
+ }
+ spin_lock_irqsave(&emu->emu_lock, flags);
+ snd_emu1010_fpga_read(emu, 0x20, &value);
+ spin_unlock_irqrestore(&emu->emu_lock, flags);
+ if ((value & 0x4) == 0) {
+ spin_lock_irqsave(&emu->emu_lock, flags);
+ snd_emu1010_fpga_read(emu, 0x28, &value);
+ snd_emu1010_fpga_read(emu, 0x29, &value2);
+ spin_unlock_irqrestore(&emu->emu_lock, flags);
+ rate = 0x1770000 / (((value << 5) | value2)+1);
+ snd_iprintf(buffer, "SPDIF Locked : %d\n", rate);
+ } else {
+ snd_iprintf(buffer, "SPDIF Unlocked\n");
+ }
+ } else {
+ snd_emu10k1_proc_spdif_status(emu, buffer, "CD-ROM S/PDIF In", CDCS, CDSRCS);
+ snd_emu10k1_proc_spdif_status(emu, buffer, "Optical or Coax S/PDIF In", GPSCS, GPSRCS);
+ }
#if 0
val = snd_emu10k1_ptr_read(emu, ZVSRCS, 0);
snd_iprintf(buffer, "\nZoomed Video\n");
@@ -379,20 +413,16 @@ static void snd_emu_proc_emu1010_reg_read(struct snd_info_entry *entry,
struct snd_info_buffer *buffer)
{
struct snd_emu10k1 *emu = entry->private_data;
- unsigned long value;
+ int value;
unsigned long flags;
- unsigned long regs;
int i;
snd_iprintf(buffer, "EMU1010 Registers:\n\n");
- for(i = 0; i < 0x30; i+=1) {
+ for(i = 0; i < 0x40; i+=1) {
spin_lock_irqsave(&emu->emu_lock, flags);
- regs=i+0x40; /* 0x40 upwards are registers. */
- outl(regs, emu->port + A_IOCFG);
- outl(regs | 0x80, emu->port + A_IOCFG); /* High bit clocks the value into the fpga. */
- value = inl(emu->port + A_IOCFG);
+ snd_emu1010_fpga_read(emu, i, &value);
spin_unlock_irqrestore(&emu->emu_lock, flags);
- snd_iprintf(buffer, "%02X: %08lX, %02lX\n", i, value, (value >> 8) & 0x7f);
+ snd_iprintf(buffer, "%02X: %08X, %02X\n", i, value, (value >> 8) & 0x7f);
}
}
@@ -555,9 +585,9 @@ int __devinit snd_emu10k1_proc_init(struct snd_emu10k1 * emu)
{
struct snd_info_entry *entry;
#ifdef CONFIG_SND_DEBUG
- if ((emu->card_capabilities->emu1010) &&
- snd_card_proc_new(emu->card, "emu1010_regs", &entry)) {
- snd_info_set_text_ops(entry, emu, snd_emu_proc_emu1010_reg_read);
+ if (emu->card_capabilities->emu1010) {
+ if (! snd_card_proc_new(emu->card, "emu1010_regs", &entry))
+ snd_info_set_text_ops(entry, emu, snd_emu_proc_emu1010_reg_read);
}
if (! snd_card_proc_new(emu->card, "io_regs", &entry)) {
snd_info_set_text_ops(entry, emu, snd_emu_proc_io_reg_read);
diff --git a/sound/pci/emu10k1/io.c b/sound/pci/emu10k1/io.c
index 116e1c8d9361..6702c15fefa3 100644
--- a/sound/pci/emu10k1/io.c
+++ b/sound/pci/emu10k1/io.c
@@ -1,5 +1,5 @@
/*
- * Copyright (c) by Jaroslav Kysela <perex@suse.cz>
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
* Creative Labs, Inc.
* Routines for control of EMU10K1 chips
*
@@ -226,9 +226,9 @@ int snd_emu10k1_i2c_write(struct snd_emu10k1 *emu,
return 0;
}
-int snd_emu1010_fpga_write(struct snd_emu10k1 * emu, int reg, int value)
+int snd_emu1010_fpga_write(struct snd_emu10k1 * emu, u32 reg, u32 value)
{
- if (reg < 0 || reg > 0x3f)
+ if (reg > 0x3f)
return 1;
reg += 0x40; /* 0x40 upwards are registers. */
if (value < 0 || value > 0x3f) /* 0 to 0x3f are values */
@@ -244,9 +244,9 @@ int snd_emu1010_fpga_write(struct snd_emu10k1 * emu, int reg, int value)
return 0;
}
-int snd_emu1010_fpga_read(struct snd_emu10k1 * emu, int reg, int *value)
+int snd_emu1010_fpga_read(struct snd_emu10k1 * emu, u32 reg, u32 *value)
{
- if (reg < 0 || reg > 0x3f)
+ if (reg > 0x3f)
return 1;
reg += 0x40; /* 0x40 upwards are registers. */
outl(reg, emu->port + A_IOCFG);
@@ -261,7 +261,7 @@ int snd_emu1010_fpga_read(struct snd_emu10k1 * emu, int reg, int *value)
/* Each Destination has one and only one Source,
* but one Source can feed any number of Destinations simultaneously.
*/
-int snd_emu1010_fpga_link_dst_src_write(struct snd_emu10k1 * emu, int dst, int src)
+int snd_emu1010_fpga_link_dst_src_write(struct snd_emu10k1 * emu, u32 dst, u32 src)
{
snd_emu1010_fpga_write(emu, 0x00, ((dst >> 8) & 0x3f) );
snd_emu1010_fpga_write(emu, 0x01, (dst & 0x3f) );
diff --git a/sound/pci/emu10k1/irq.c b/sound/pci/emu10k1/irq.c
index 4f18f7e8bcfb..3c114b45e0b2 100644
--- a/sound/pci/emu10k1/irq.c
+++ b/sound/pci/emu10k1/irq.c
@@ -1,5 +1,5 @@
/*
- * Copyright (c) by Jaroslav Kysela <perex@suse.cz>
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
* Creative Labs, Inc.
* Routines for IRQ control of EMU10K1 chips
*
diff --git a/sound/pci/emu10k1/memory.c b/sound/pci/emu10k1/memory.c
index 4fcaefe5a3c5..48097c6bb15c 100644
--- a/sound/pci/emu10k1/memory.c
+++ b/sound/pci/emu10k1/memory.c
@@ -1,5 +1,5 @@
/*
- * Copyright (c) by Jaroslav Kysela <perex@suse.cz>
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
* Copyright (c) by Takashi Iwai <tiwai@suse.de>
*
* EMU10K1 memory page allocation (PTB area)
diff --git a/sound/pci/emu10k1/p16v.c b/sound/pci/emu10k1/p16v.c
index 7ee19c63c2c8..d619a3842cdd 100644
--- a/sound/pci/emu10k1/p16v.c
+++ b/sound/pci/emu10k1/p16v.c
@@ -124,11 +124,12 @@
/* hardware definition */
static struct snd_pcm_hardware snd_p16v_playback_hw = {
- .info = (SNDRV_PCM_INFO_MMAP |
- SNDRV_PCM_INFO_INTERLEAVED |
- SNDRV_PCM_INFO_BLOCK_TRANSFER |
- SNDRV_PCM_INFO_RESUME |
- SNDRV_PCM_INFO_MMAP_VALID),
+ .info = SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_RESUME |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_SYNC_START,
.formats = SNDRV_PCM_FMTBIT_S32_LE, /* Only supports 24-bit samples padded to 32 bits. */
.rates = SNDRV_PCM_RATE_192000 | SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_44100,
.rate_min = 44100,
@@ -207,6 +208,11 @@ static int snd_p16v_pcm_open_playback_channel(struct snd_pcm_substream *substrea
if ((err = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS)) < 0)
return err;
+ runtime->sync.id32[0] = substream->pcm->card->number;
+ runtime->sync.id32[1] = 'P';
+ runtime->sync.id32[2] = 16;
+ runtime->sync.id32[3] = 'V';
+
return 0;
}
/* open_capture callback */
@@ -448,6 +454,9 @@ static int snd_p16v_pcm_trigger_playback(struct snd_pcm_substream *substream,
break;
}
snd_pcm_group_for_each_entry(s, substream) {
+ if (snd_pcm_substream_chip(s) != emu ||
+ s->stream != SNDRV_PCM_STREAM_PLAYBACK)
+ continue;
runtime = s->runtime;
epcm = runtime->private_data;
channel = substream->pcm->device-emu->p16v_device_offset;
diff --git a/sound/pci/emu10k1/voice.c b/sound/pci/emu10k1/voice.c
index 1db50fe61475..04fa8492abb0 100644
--- a/sound/pci/emu10k1/voice.c
+++ b/sound/pci/emu10k1/voice.c
@@ -1,5 +1,5 @@
/*
- * Copyright (c) by Jaroslav Kysela <perex@suse.cz>
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
* Creative Labs, Inc.
* Lee Revell <rlrevell@joe-job.com>
* Routines for control of EMU10K1 chips - voice manager
diff --git a/sound/pci/ens1370.c b/sound/pci/ens1370.c
index 21cb4268a59b..b958f869cb13 100644
--- a/sound/pci/ens1370.c
+++ b/sound/pci/ens1370.c
@@ -1,6 +1,6 @@
/*
* Driver for Ensoniq ES1370/ES1371 AudioPCI soundcard
- * Copyright (c) by Jaroslav Kysela <perex@suse.cz>,
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>,
* Thomas Sailer <sailer@ife.ee.ethz.ch>
*
* This program is free software; you can redistribute it and/or modify
@@ -61,7 +61,7 @@
#endif
-MODULE_AUTHOR("Jaroslav Kysela <perex@suse.cz>, Thomas Sailer <sailer@ife.ee.ethz.ch>");
+MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>, Thomas Sailer <sailer@ife.ee.ethz.ch>");
MODULE_LICENSE("GPL");
#ifdef CHIP1370
MODULE_DESCRIPTION("Ensoniq AudioPCI ES1370");
@@ -1419,15 +1419,7 @@ static int snd_ens1373_spdif_stream_put(struct snd_kcontrol *kcontrol,
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .info = snd_es1371_spdif_info, \
.get = snd_es1371_spdif_get, .put = snd_es1371_spdif_put }
-static int snd_es1371_spdif_info(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_info *uinfo)
-{
- uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
- uinfo->count = 1;
- uinfo->value.integer.min = 0;
- uinfo->value.integer.max = 1;
- return 0;
-}
+#define snd_es1371_spdif_info snd_ctl_boolean_mono_info
static int snd_es1371_spdif_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
@@ -1489,15 +1481,7 @@ static struct snd_kcontrol_new snd_es1371_mixer_spdif[] __devinitdata = {
};
-static int snd_es1373_rear_info(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_info *uinfo)
-{
- uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
- uinfo->count = 1;
- uinfo->value.integer.min = 0;
- uinfo->value.integer.max = 1;
- return 0;
-}
+#define snd_es1373_rear_info snd_ctl_boolean_mono_info
static int snd_es1373_rear_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
@@ -1542,15 +1526,7 @@ static struct snd_kcontrol_new snd_ens1373_rear __devinitdata =
.put = snd_es1373_rear_put,
};
-static int snd_es1373_line_info(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_info *uinfo)
-{
- uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
- uinfo->count = 1;
- uinfo->value.integer.min = 0;
- uinfo->value.integer.max = 1;
- return 0;
-}
+#define snd_es1373_line_info snd_ctl_boolean_mono_info
static int snd_es1373_line_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
@@ -1707,15 +1683,7 @@ static int __devinit snd_ensoniq_1371_mixer(struct ensoniq *ensoniq,
.get = snd_ensoniq_control_get, .put = snd_ensoniq_control_put, \
.private_value = mask }
-static int snd_ensoniq_control_info(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_info *uinfo)
-{
- uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
- uinfo->count = 1;
- uinfo->value.integer.min = 0;
- uinfo->value.integer.max = 1;
- return 0;
-}
+#define snd_ensoniq_control_info snd_ctl_boolean_mono_info
static int snd_ensoniq_control_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
diff --git a/sound/pci/es1938.c b/sound/pci/es1938.c
index fec29a108945..fb25abe68a02 100644
--- a/sound/pci/es1938.c
+++ b/sound/pci/es1938.c
@@ -1,7 +1,7 @@
/*
* Driver for ESS Solo-1 (ES1938, ES1946, ES1969) soundcard
* Copyright (c) by Jaromir Koutek <miri@punknet.cz>,
- * Jaroslav Kysela <perex@suse.cz>,
+ * Jaroslav Kysela <perex@perex.cz>,
* Thomas Sailer <sailer@ife.ee.ethz.ch>,
* Abramo Bagnara <abramo@alsa-project.org>,
* Markus Gruber <gruber@eikon.tum.de>
@@ -1066,15 +1066,7 @@ static int snd_es1938_put_mux(struct snd_kcontrol *kcontrol,
return snd_es1938_mixer_bits(chip, 0x1c, 0x07, val) != val;
}
-static int snd_es1938_info_spatializer_enable(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_info *uinfo)
-{
- uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
- uinfo->count = 1;
- uinfo->value.integer.min = 0;
- uinfo->value.integer.max = 1;
- return 0;
-}
+#define snd_es1938_info_spatializer_enable snd_ctl_boolean_mono_info
static int snd_es1938_get_spatializer_enable(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
@@ -1120,15 +1112,7 @@ static int snd_es1938_get_hw_volume(struct snd_kcontrol *kcontrol,
return 0;
}
-static int snd_es1938_info_hw_switch(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_info *uinfo)
-{
- uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
- uinfo->count = 2;
- uinfo->value.integer.min = 0;
- uinfo->value.integer.max = 1;
- return 0;
-}
+#define snd_es1938_info_hw_switch snd_ctl_boolean_stereo_info
static int snd_es1938_get_hw_switch(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
diff --git a/sound/pci/es1968.c b/sound/pci/es1968.c
index 2faf009076bb..d69b11d1f993 100644
--- a/sound/pci/es1968.c
+++ b/sound/pci/es1968.c
@@ -843,10 +843,9 @@ static void snd_es1968_bob_dec(struct es1968 *chip)
snd_es1968_bob_stop(chip);
else if (chip->bob_freq > ESM_BOB_FREQ) {
/* check reduction of timer frequency */
- struct list_head *p;
int max_freq = ESM_BOB_FREQ;
- list_for_each(p, &chip->substream_list) {
- struct esschan *es = list_entry(p, struct esschan, list);
+ struct esschan *es;
+ list_for_each_entry(es, &chip->substream_list, list) {
if (max_freq < es->bob_freq)
max_freq = es->bob_freq;
}
@@ -1316,12 +1315,11 @@ static struct snd_pcm_hardware snd_es1968_capture = {
static int calc_available_memory_size(struct es1968 *chip)
{
- struct list_head *p;
int max_size = 0;
-
+ struct esm_memory *buf;
+
mutex_lock(&chip->memory_mutex);
- list_for_each(p, &chip->buf_list) {
- struct esm_memory *buf = list_entry(p, struct esm_memory, list);
+ list_for_each_entry(buf, &chip->buf_list, list) {
if (buf->empty && buf->buf.bytes > max_size)
max_size = buf->buf.bytes;
}
@@ -1335,12 +1333,10 @@ static int calc_available_memory_size(struct es1968 *chip)
static struct esm_memory *snd_es1968_new_memory(struct es1968 *chip, int size)
{
struct esm_memory *buf;
- struct list_head *p;
-
+
size = ALIGN(size, ESM_MEM_ALIGN);
mutex_lock(&chip->memory_mutex);
- list_for_each(p, &chip->buf_list) {
- buf = list_entry(p, struct esm_memory, list);
+ list_for_each_entry(buf, &chip->buf_list, list) {
if (buf->empty && buf->buf.bytes >= size)
goto __found;
}
@@ -1938,10 +1934,9 @@ static irqreturn_t snd_es1968_interrupt(int irq, void *dev_id)
}
if (event & ESM_SOUND_IRQ) {
- struct list_head *p;
+ struct esschan *es;
spin_lock(&chip->substream_lock);
- list_for_each(p, &chip->substream_list) {
- struct esschan *es = list_entry(p, struct esschan, list);
+ list_for_each_entry(es, &chip->substream_list, list) {
if (es->running)
snd_es1968_update_pcm(chip, es);
}
@@ -2345,7 +2340,7 @@ static int es1968_resume(struct pci_dev *pci)
{
struct snd_card *card = pci_get_drvdata(pci);
struct es1968 *chip = card->private_data;
- struct list_head *p;
+ struct esschan *es;
if (! chip->do_pm)
return 0;
@@ -2374,8 +2369,7 @@ static int es1968_resume(struct pci_dev *pci)
/* restore ac97 state */
snd_ac97_resume(chip->ac97);
- list_for_each(p, &chip->substream_list) {
- struct esschan *es = list_entry(p, struct esschan, list);
+ list_for_each_entry(es, &chip->substream_list, list) {
switch (es->mode) {
case ESM_MODE_PLAY:
snd_es1968_playback_setup(chip, es, es->substream->runtime);
diff --git a/sound/pci/fm801.c b/sound/pci/fm801.c
index 11015178e207..9939109f05a2 100644
--- a/sound/pci/fm801.c
+++ b/sound/pci/fm801.c
@@ -1,6 +1,6 @@
/*
* The driver for the ForteMedia FM801 based soundcards
- * Copyright (c) by Jaroslav Kysela <perex@suse.cz>
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
*
* Support FM only card by Andy Shevchenko <andy@smile.org.ua>
*
@@ -42,7 +42,7 @@
#define TEA575X_RADIO 1
#endif
-MODULE_AUTHOR("Jaroslav Kysela <perex@suse.cz>");
+MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
MODULE_DESCRIPTION("ForteMedia FM801");
MODULE_LICENSE("GPL");
MODULE_SUPPORTED_DEVICE("{{ForteMedia,FM801},"
diff --git a/sound/pci/hda/Makefile b/sound/pci/hda/Makefile
index b2484bbdcc1d..ab0c726d648e 100644
--- a/sound/pci/hda/Makefile
+++ b/sound/pci/hda/Makefile
@@ -1,19 +1,18 @@
-snd-hda-intel-objs := hda_intel.o
+snd-hda-intel-y := hda_intel.o
# since snd-hda-intel is the only driver using hda-codec,
# merge it into a single module although it was originally
# designed to be individual modules
-snd-hda-intel-objs += hda_codec.o \
- hda_generic.o \
- patch_realtek.o \
- patch_cmedia.o \
- patch_analog.o \
- patch_sigmatel.o \
- patch_si3054.o \
- patch_atihdmi.o \
- patch_conexant.o \
- patch_via.o
-ifdef CONFIG_PROC_FS
-snd-hda-intel-objs += hda_proc.o
-endif
+snd-hda-intel-y += hda_codec.o
+snd-hda-intel-$(CONFIG_PROC_FS) += hda_proc.o
+snd-hda-intel-$(CONFIG_SND_HDA_HWDEP) += hda_hwdep.o
+snd-hda-intel-$(CONFIG_SND_HDA_GENERIC) += hda_generic.o
+snd-hda-intel-$(CONFIG_SND_HDA_CODEC_REALTEK) += patch_realtek.o
+snd-hda-intel-$(CONFIG_SND_HDA_CODEC_CMEDIA) += patch_cmedia.o
+snd-hda-intel-$(CONFIG_SND_HDA_CODEC_ANALOG) += patch_analog.o
+snd-hda-intel-$(CONFIG_SND_HDA_CODEC_SIGMATEL) += patch_sigmatel.o
+snd-hda-intel-$(CONFIG_SND_HDA_CODEC_SI3054) += patch_si3054.o
+snd-hda-intel-$(CONFIG_SND_HDA_CODEC_ATIHDMI) += patch_atihdmi.o
+snd-hda-intel-$(CONFIG_SND_HDA_CODEC_CONEXANT) += patch_conexant.o
+snd-hda-intel-$(CONFIG_SND_HDA_CODEC_VIA) += patch_via.o
obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-intel.o
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index f87f8f088956..187533e477c6 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -31,7 +31,15 @@
#include <sound/tlv.h>
#include <sound/initval.h>
#include "hda_local.h"
-
+#include <sound/hda_hwdep.h>
+
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+/* define this option here to hide as static */
+static int power_save = CONFIG_SND_HDA_POWER_SAVE_DEFAULT;
+module_param(power_save, int, 0644);
+MODULE_PARM_DESC(power_save, "Automatic power-saving timeout "
+ "(in second, 0 = disable).");
+#endif
/*
* vendor / preset table
@@ -59,6 +67,13 @@ static struct hda_vendor_id hda_vendor_ids[] = {
#include "hda_patch.h"
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+static void hda_power_work(struct work_struct *work);
+static void hda_keep_power_on(struct hda_codec *codec);
+#else
+static inline void hda_keep_power_on(struct hda_codec *codec) {}
+#endif
+
/**
* snd_hda_codec_read - send a command and get the response
* @codec: the HDA codec
@@ -76,12 +91,14 @@ unsigned int snd_hda_codec_read(struct hda_codec *codec, hda_nid_t nid,
unsigned int verb, unsigned int parm)
{
unsigned int res;
+ snd_hda_power_up(codec);
mutex_lock(&codec->bus->cmd_mutex);
if (!codec->bus->ops.command(codec, nid, direct, verb, parm))
res = codec->bus->ops.get_response(codec);
else
res = (unsigned int)-1;
mutex_unlock(&codec->bus->cmd_mutex);
+ snd_hda_power_down(codec);
return res;
}
@@ -101,9 +118,11 @@ int snd_hda_codec_write(struct hda_codec *codec, hda_nid_t nid, int direct,
unsigned int verb, unsigned int parm)
{
int err;
+ snd_hda_power_up(codec);
mutex_lock(&codec->bus->cmd_mutex);
err = codec->bus->ops.command(codec, nid, direct, verb, parm);
mutex_unlock(&codec->bus->cmd_mutex);
+ snd_hda_power_down(codec);
return err;
}
@@ -136,6 +155,8 @@ int snd_hda_get_sub_nodes(struct hda_codec *codec, hda_nid_t nid,
unsigned int parm;
parm = snd_hda_param_read(codec, nid, AC_PAR_NODE_COUNT);
+ if (parm == -1)
+ return 0;
*start_id = (parm >> 16) & 0x7fff;
return (int)(parm & 0x7fff);
}
@@ -387,6 +408,13 @@ int __devinit snd_hda_bus_new(struct snd_card *card,
return 0;
}
+#ifdef CONFIG_SND_HDA_GENERIC
+#define is_generic_config(codec) \
+ (codec->bus->modelname && !strcmp(codec->bus->modelname, "generic"))
+#else
+#define is_generic_config(codec) 0
+#endif
+
/*
* find a matching codec preset
*/
@@ -395,7 +423,7 @@ find_codec_preset(struct hda_codec *codec)
{
const struct hda_codec_preset **tbl, *preset;
- if (codec->bus->modelname && !strcmp(codec->bus->modelname, "generic"))
+ if (is_generic_config(codec))
return NULL; /* use the generic parser */
for (tbl = hda_preset_tables; *tbl; tbl++) {
@@ -486,6 +514,10 @@ static int read_widget_caps(struct hda_codec *codec, hda_nid_t fg_node)
}
+static void init_hda_cache(struct hda_cache_rec *cache,
+ unsigned int record_size);
+static void free_hda_cache(struct hda_cache_rec *cache);
+
/*
* codec destructor
*/
@@ -493,17 +525,20 @@ static void snd_hda_codec_free(struct hda_codec *codec)
{
if (!codec)
return;
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ cancel_delayed_work(&codec->power_work);
+ flush_scheduled_work();
+#endif
list_del(&codec->list);
codec->bus->caddr_tbl[codec->addr] = NULL;
if (codec->patch_ops.free)
codec->patch_ops.free(codec);
- kfree(codec->amp_info);
+ free_hda_cache(&codec->amp_cache);
+ free_hda_cache(&codec->cmd_cache);
kfree(codec->wcaps);
kfree(codec);
}
-static void init_amp_hash(struct hda_codec *codec);
-
/**
* snd_hda_codec_new - create a HDA codec
* @bus: the bus to assign
@@ -537,7 +572,17 @@ int __devinit snd_hda_codec_new(struct hda_bus *bus, unsigned int codec_addr,
codec->bus = bus;
codec->addr = codec_addr;
mutex_init(&codec->spdif_mutex);
- init_amp_hash(codec);
+ init_hda_cache(&codec->amp_cache, sizeof(struct hda_amp_info));
+ init_hda_cache(&codec->cmd_cache, sizeof(struct hda_cache_head));
+
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ INIT_DELAYED_WORK(&codec->power_work, hda_power_work);
+ /* snd_hda_codec_new() marks the codec as power-up, and leave it as is.
+ * the caller has to power down appropriatley after initialization
+ * phase.
+ */
+ hda_keep_power_on(codec);
+#endif
list_add_tail(&codec->list, &bus->codec_list);
bus->caddr_tbl[codec_addr] = codec;
@@ -581,10 +626,26 @@ int __devinit snd_hda_codec_new(struct hda_bus *bus, unsigned int codec_addr,
snd_hda_get_codec_name(codec, bus->card->mixername,
sizeof(bus->card->mixername));
- if (codec->preset && codec->preset->patch)
- err = codec->preset->patch(codec);
- else
+#ifdef CONFIG_SND_HDA_GENERIC
+ if (is_generic_config(codec)) {
err = snd_hda_parse_generic_codec(codec);
+ goto patched;
+ }
+#endif
+ if (codec->preset && codec->preset->patch) {
+ err = codec->preset->patch(codec);
+ goto patched;
+ }
+
+ /* call the default parser */
+#ifdef CONFIG_SND_HDA_GENERIC
+ err = snd_hda_parse_generic_codec(codec);
+#else
+ printk(KERN_ERR "hda-codec: No codec parser is available\n");
+ err = -ENODEV;
+#endif
+
+ patched:
if (err < 0) {
snd_hda_codec_free(codec);
return err;
@@ -594,6 +655,9 @@ int __devinit snd_hda_codec_new(struct hda_bus *bus, unsigned int codec_addr,
init_unsol_queue(bus);
snd_hda_codec_proc_new(codec);
+#ifdef CONFIG_SND_HDA_HWDEP
+ snd_hda_create_hwdep(codec);
+#endif
sprintf(component, "HDA:%08x", codec->vendor_id);
snd_component_add(codec->bus->card, component);
@@ -637,59 +701,72 @@ void snd_hda_codec_setup_stream(struct hda_codec *codec, hda_nid_t nid,
#define INFO_AMP_VOL(ch) (1 << (1 + (ch)))
/* initialize the hash table */
-static void __devinit init_amp_hash(struct hda_codec *codec)
+static void __devinit init_hda_cache(struct hda_cache_rec *cache,
+ unsigned int record_size)
+{
+ memset(cache, 0, sizeof(*cache));
+ memset(cache->hash, 0xff, sizeof(cache->hash));
+ cache->record_size = record_size;
+}
+
+static void free_hda_cache(struct hda_cache_rec *cache)
{
- memset(codec->amp_hash, 0xff, sizeof(codec->amp_hash));
- codec->num_amp_entries = 0;
- codec->amp_info_size = 0;
- codec->amp_info = NULL;
+ kfree(cache->buffer);
}
/* query the hash. allocate an entry if not found. */
-static struct hda_amp_info *get_alloc_amp_hash(struct hda_codec *codec, u32 key)
+static struct hda_cache_head *get_alloc_hash(struct hda_cache_rec *cache,
+ u32 key)
{
- u16 idx = key % (u16)ARRAY_SIZE(codec->amp_hash);
- u16 cur = codec->amp_hash[idx];
- struct hda_amp_info *info;
+ u16 idx = key % (u16)ARRAY_SIZE(cache->hash);
+ u16 cur = cache->hash[idx];
+ struct hda_cache_head *info;
while (cur != 0xffff) {
- info = &codec->amp_info[cur];
+ info = (struct hda_cache_head *)(cache->buffer +
+ cur * cache->record_size);
if (info->key == key)
return info;
cur = info->next;
}
/* add a new hash entry */
- if (codec->num_amp_entries >= codec->amp_info_size) {
+ if (cache->num_entries >= cache->size) {
/* reallocate the array */
- int new_size = codec->amp_info_size + 64;
- struct hda_amp_info *new_info;
- new_info = kcalloc(new_size, sizeof(struct hda_amp_info),
- GFP_KERNEL);
- if (!new_info) {
+ unsigned int new_size = cache->size + 64;
+ void *new_buffer;
+ new_buffer = kcalloc(new_size, cache->record_size, GFP_KERNEL);
+ if (!new_buffer) {
snd_printk(KERN_ERR "hda_codec: "
"can't malloc amp_info\n");
return NULL;
}
- if (codec->amp_info) {
- memcpy(new_info, codec->amp_info,
- codec->amp_info_size *
- sizeof(struct hda_amp_info));
- kfree(codec->amp_info);
+ if (cache->buffer) {
+ memcpy(new_buffer, cache->buffer,
+ cache->size * cache->record_size);
+ kfree(cache->buffer);
}
- codec->amp_info_size = new_size;
- codec->amp_info = new_info;
+ cache->size = new_size;
+ cache->buffer = new_buffer;
}
- cur = codec->num_amp_entries++;
- info = &codec->amp_info[cur];
+ cur = cache->num_entries++;
+ info = (struct hda_cache_head *)(cache->buffer +
+ cur * cache->record_size);
info->key = key;
- info->status = 0; /* not initialized yet */
- info->next = codec->amp_hash[idx];
- codec->amp_hash[idx] = cur;
+ info->val = 0;
+ info->next = cache->hash[idx];
+ cache->hash[idx] = cur;
return info;
}
+/* query and allocate an amp hash entry */
+static inline struct hda_amp_info *
+get_alloc_amp_hash(struct hda_codec *codec, u32 key)
+{
+ return (struct hda_amp_info *)get_alloc_hash(&codec->amp_cache, key);
+}
+
/*
* query AMP capabilities for the given widget and direction
*/
@@ -700,7 +777,7 @@ static u32 query_amp_caps(struct hda_codec *codec, hda_nid_t nid, int direction)
info = get_alloc_amp_hash(codec, HDA_HASH_KEY(nid, direction, 0));
if (!info)
return 0;
- if (!(info->status & INFO_AMP_CAPS)) {
+ if (!(info->head.val & INFO_AMP_CAPS)) {
if (!(get_wcaps(codec, nid) & AC_WCAP_AMP_OVRD))
nid = codec->afg;
info->amp_caps = snd_hda_param_read(codec, nid,
@@ -708,7 +785,7 @@ static u32 query_amp_caps(struct hda_codec *codec, hda_nid_t nid, int direction)
AC_PAR_AMP_OUT_CAP :
AC_PAR_AMP_IN_CAP);
if (info->amp_caps)
- info->status |= INFO_AMP_CAPS;
+ info->head.val |= INFO_AMP_CAPS;
}
return info->amp_caps;
}
@@ -722,7 +799,7 @@ int snd_hda_override_amp_caps(struct hda_codec *codec, hda_nid_t nid, int dir,
if (!info)
return -EINVAL;
info->amp_caps = caps;
- info->status |= INFO_AMP_CAPS;
+ info->head.val |= INFO_AMP_CAPS;
return 0;
}
@@ -736,7 +813,7 @@ static unsigned int get_vol_mute(struct hda_codec *codec,
{
u32 val, parm;
- if (info->status & INFO_AMP_VOL(ch))
+ if (info->head.val & INFO_AMP_VOL(ch))
return info->vol[ch];
parm = ch ? AC_AMP_GET_RIGHT : AC_AMP_GET_LEFT;
@@ -745,7 +822,7 @@ static unsigned int get_vol_mute(struct hda_codec *codec,
val = snd_hda_codec_read(codec, nid, 0,
AC_VERB_GET_AMP_GAIN_MUTE, parm);
info->vol[ch] = val & 0xff;
- info->status |= INFO_AMP_VOL(ch);
+ info->head.val |= INFO_AMP_VOL(ch);
return info->vol[ch];
}
@@ -792,12 +869,50 @@ int snd_hda_codec_amp_update(struct hda_codec *codec, hda_nid_t nid, int ch,
return 0;
val &= mask;
val |= get_vol_mute(codec, info, nid, ch, direction, idx) & ~mask;
- if (info->vol[ch] == val && !codec->in_resume)
+ if (info->vol[ch] == val)
return 0;
put_vol_mute(codec, info, nid, ch, direction, idx, val);
return 1;
}
+/*
+ * update the AMP stereo with the same mask and value
+ */
+int snd_hda_codec_amp_stereo(struct hda_codec *codec, hda_nid_t nid,
+ int direction, int idx, int mask, int val)
+{
+ int ch, ret = 0;
+ for (ch = 0; ch < 2; ch++)
+ ret |= snd_hda_codec_amp_update(codec, nid, ch, direction,
+ idx, mask, val);
+ return ret;
+}
+
+#ifdef SND_HDA_NEEDS_RESUME
+/* resume the all amp commands from the cache */
+void snd_hda_codec_resume_amp(struct hda_codec *codec)
+{
+ struct hda_amp_info *buffer = codec->amp_cache.buffer;
+ int i;
+
+ for (i = 0; i < codec->amp_cache.size; i++, buffer++) {
+ u32 key = buffer->head.key;
+ hda_nid_t nid;
+ unsigned int idx, dir, ch;
+ if (!key)
+ continue;
+ nid = key & 0xff;
+ idx = (key >> 16) & 0xff;
+ dir = (key >> 24) & 0xff;
+ for (ch = 0; ch < 2; ch++) {
+ if (!(buffer->head.val & INFO_AMP_VOL(ch)))
+ continue;
+ put_vol_mute(codec, buffer, nid, ch, dir, idx,
+ buffer->vol[ch]);
+ }
+ }
+}
+#endif /* SND_HDA_NEEDS_RESUME */
/*
* AMP control callbacks
@@ -844,9 +959,11 @@ int snd_hda_mixer_amp_volume_get(struct snd_kcontrol *kcontrol,
long *valp = ucontrol->value.integer.value;
if (chs & 1)
- *valp++ = snd_hda_codec_amp_read(codec, nid, 0, dir, idx) & 0x7f;
+ *valp++ = snd_hda_codec_amp_read(codec, nid, 0, dir, idx)
+ & HDA_AMP_VOLMASK;
if (chs & 2)
- *valp = snd_hda_codec_amp_read(codec, nid, 1, dir, idx) & 0x7f;
+ *valp = snd_hda_codec_amp_read(codec, nid, 1, dir, idx)
+ & HDA_AMP_VOLMASK;
return 0;
}
@@ -861,6 +978,7 @@ int snd_hda_mixer_amp_volume_put(struct snd_kcontrol *kcontrol,
long *valp = ucontrol->value.integer.value;
int change = 0;
+ snd_hda_power_up(codec);
if (chs & 1) {
change = snd_hda_codec_amp_update(codec, nid, 0, dir, idx,
0x7f, *valp);
@@ -869,6 +987,7 @@ int snd_hda_mixer_amp_volume_put(struct snd_kcontrol *kcontrol,
if (chs & 2)
change |= snd_hda_codec_amp_update(codec, nid, 1, dir, idx,
0x7f, *valp);
+ snd_hda_power_down(codec);
return change;
}
@@ -923,10 +1042,10 @@ int snd_hda_mixer_amp_switch_get(struct snd_kcontrol *kcontrol,
if (chs & 1)
*valp++ = (snd_hda_codec_amp_read(codec, nid, 0, dir, idx) &
- 0x80) ? 0 : 1;
+ HDA_AMP_MUTE) ? 0 : 1;
if (chs & 2)
*valp = (snd_hda_codec_amp_read(codec, nid, 1, dir, idx) &
- 0x80) ? 0 : 1;
+ HDA_AMP_MUTE) ? 0 : 1;
return 0;
}
@@ -941,15 +1060,22 @@ int snd_hda_mixer_amp_switch_put(struct snd_kcontrol *kcontrol,
long *valp = ucontrol->value.integer.value;
int change = 0;
+ snd_hda_power_up(codec);
if (chs & 1) {
change = snd_hda_codec_amp_update(codec, nid, 0, dir, idx,
- 0x80, *valp ? 0 : 0x80);
+ HDA_AMP_MUTE,
+ *valp ? 0 : HDA_AMP_MUTE);
valp++;
}
if (chs & 2)
change |= snd_hda_codec_amp_update(codec, nid, 1, dir, idx,
- 0x80, *valp ? 0 : 0x80);
-
+ HDA_AMP_MUTE,
+ *valp ? 0 : HDA_AMP_MUTE);
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ if (codec->patch_ops.check_power_status)
+ codec->patch_ops.check_power_status(codec, nid);
+#endif
+ snd_hda_power_down(codec);
return change;
}
@@ -1002,6 +1128,93 @@ int snd_hda_mixer_bind_switch_put(struct snd_kcontrol *kcontrol,
}
/*
+ * generic bound volume/swtich controls
+ */
+int snd_hda_mixer_bind_ctls_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct hda_bind_ctls *c;
+ int err;
+
+ c = (struct hda_bind_ctls *)kcontrol->private_value;
+ mutex_lock(&codec->spdif_mutex); /* reuse spdif_mutex */
+ kcontrol->private_value = *c->values;
+ err = c->ops->info(kcontrol, uinfo);
+ kcontrol->private_value = (long)c;
+ mutex_unlock(&codec->spdif_mutex);
+ return err;
+}
+
+int snd_hda_mixer_bind_ctls_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct hda_bind_ctls *c;
+ int err;
+
+ c = (struct hda_bind_ctls *)kcontrol->private_value;
+ mutex_lock(&codec->spdif_mutex); /* reuse spdif_mutex */
+ kcontrol->private_value = *c->values;
+ err = c->ops->get(kcontrol, ucontrol);
+ kcontrol->private_value = (long)c;
+ mutex_unlock(&codec->spdif_mutex);
+ return err;
+}
+
+int snd_hda_mixer_bind_ctls_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct hda_bind_ctls *c;
+ unsigned long *vals;
+ int err = 0, change = 0;
+
+ c = (struct hda_bind_ctls *)kcontrol->private_value;
+ mutex_lock(&codec->spdif_mutex); /* reuse spdif_mutex */
+ for (vals = c->values; *vals; vals++) {
+ kcontrol->private_value = *vals;
+ err = c->ops->put(kcontrol, ucontrol);
+ if (err < 0)
+ break;
+ change |= err;
+ }
+ kcontrol->private_value = (long)c;
+ mutex_unlock(&codec->spdif_mutex);
+ return err < 0 ? err : change;
+}
+
+int snd_hda_mixer_bind_tlv(struct snd_kcontrol *kcontrol, int op_flag,
+ unsigned int size, unsigned int __user *tlv)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct hda_bind_ctls *c;
+ int err;
+
+ c = (struct hda_bind_ctls *)kcontrol->private_value;
+ mutex_lock(&codec->spdif_mutex); /* reuse spdif_mutex */
+ kcontrol->private_value = *c->values;
+ err = c->ops->tlv(kcontrol, op_flag, size, tlv);
+ kcontrol->private_value = (long)c;
+ mutex_unlock(&codec->spdif_mutex);
+ return err;
+}
+
+struct hda_ctl_ops snd_hda_bind_vol = {
+ .info = snd_hda_mixer_amp_volume_info,
+ .get = snd_hda_mixer_amp_volume_get,
+ .put = snd_hda_mixer_amp_volume_put,
+ .tlv = snd_hda_mixer_amp_tlv
+};
+
+struct hda_ctl_ops snd_hda_bind_sw = {
+ .info = snd_hda_mixer_amp_switch_info,
+ .get = snd_hda_mixer_amp_switch_get,
+ .put = snd_hda_mixer_amp_switch_put,
+ .tlv = snd_hda_mixer_amp_tlv
+};
+
+/*
* SPDIF out controls
*/
@@ -1118,26 +1331,20 @@ static int snd_hda_spdif_default_put(struct snd_kcontrol *kcontrol,
change = codec->spdif_ctls != val;
codec->spdif_ctls = val;
- if (change || codec->in_resume) {
- snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_DIGI_CONVERT_1,
- val & 0xff);
- snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_DIGI_CONVERT_2,
- val >> 8);
+ if (change) {
+ snd_hda_codec_write_cache(codec, nid, 0,
+ AC_VERB_SET_DIGI_CONVERT_1,
+ val & 0xff);
+ snd_hda_codec_write_cache(codec, nid, 0,
+ AC_VERB_SET_DIGI_CONVERT_2,
+ val >> 8);
}
mutex_unlock(&codec->spdif_mutex);
return change;
}
-static int snd_hda_spdif_out_switch_info(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_info *uinfo)
-{
- uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
- uinfo->count = 1;
- uinfo->value.integer.min = 0;
- uinfo->value.integer.max = 1;
- return 0;
-}
+#define snd_hda_spdif_out_switch_info snd_ctl_boolean_mono_info
static int snd_hda_spdif_out_switch_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
@@ -1161,17 +1368,16 @@ static int snd_hda_spdif_out_switch_put(struct snd_kcontrol *kcontrol,
if (ucontrol->value.integer.value[0])
val |= AC_DIG1_ENABLE;
change = codec->spdif_ctls != val;
- if (change || codec->in_resume) {
+ if (change) {
codec->spdif_ctls = val;
- snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_DIGI_CONVERT_1,
- val & 0xff);
+ snd_hda_codec_write_cache(codec, nid, 0,
+ AC_VERB_SET_DIGI_CONVERT_1,
+ val & 0xff);
/* unmute amp switch (if any) */
if ((get_wcaps(codec, nid) & AC_WCAP_OUT_AMP) &&
(val & AC_DIG1_ENABLE))
- snd_hda_codec_write(codec, nid, 0,
- AC_VERB_SET_AMP_GAIN_MUTE,
- AC_AMP_SET_RIGHT | AC_AMP_SET_LEFT |
- AC_AMP_SET_OUTPUT);
+ snd_hda_codec_amp_stereo(codec, nid, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, 0);
}
mutex_unlock(&codec->spdif_mutex);
return change;
@@ -1219,8 +1425,7 @@ static struct snd_kcontrol_new dig_mixes[] = {
*
* Returns 0 if successful, or a negative error code.
*/
-int __devinit snd_hda_create_spdif_out_ctls(struct hda_codec *codec,
- hda_nid_t nid)
+int snd_hda_create_spdif_out_ctls(struct hda_codec *codec, hda_nid_t nid)
{
int err;
struct snd_kcontrol *kctl;
@@ -1264,10 +1469,10 @@ static int snd_hda_spdif_in_switch_put(struct snd_kcontrol *kcontrol,
mutex_lock(&codec->spdif_mutex);
change = codec->spdif_in_enable != val;
- if (change || codec->in_resume) {
+ if (change) {
codec->spdif_in_enable = val;
- snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_DIGI_CONVERT_1,
- val);
+ snd_hda_codec_write_cache(codec, nid, 0,
+ AC_VERB_SET_DIGI_CONVERT_1, val);
}
mutex_unlock(&codec->spdif_mutex);
return change;
@@ -1318,8 +1523,7 @@ static struct snd_kcontrol_new dig_in_ctls[] = {
*
* Returns 0 if successful, or a negative error code.
*/
-int __devinit snd_hda_create_spdif_in_ctls(struct hda_codec *codec,
- hda_nid_t nid)
+int snd_hda_create_spdif_in_ctls(struct hda_codec *codec, hda_nid_t nid)
{
int err;
struct snd_kcontrol *kctl;
@@ -1338,6 +1542,79 @@ int __devinit snd_hda_create_spdif_in_ctls(struct hda_codec *codec,
return 0;
}
+#ifdef SND_HDA_NEEDS_RESUME
+/*
+ * command cache
+ */
+
+/* build a 32bit cache key with the widget id and the command parameter */
+#define build_cmd_cache_key(nid, verb) ((verb << 8) | nid)
+#define get_cmd_cache_nid(key) ((key) & 0xff)
+#define get_cmd_cache_cmd(key) (((key) >> 8) & 0xffff)
+
+/**
+ * snd_hda_codec_write_cache - send a single command with caching
+ * @codec: the HDA codec
+ * @nid: NID to send the command
+ * @direct: direct flag
+ * @verb: the verb to send
+ * @parm: the parameter for the verb
+ *
+ * Send a single command without waiting for response.
+ *
+ * Returns 0 if successful, or a negative error code.
+ */
+int snd_hda_codec_write_cache(struct hda_codec *codec, hda_nid_t nid,
+ int direct, unsigned int verb, unsigned int parm)
+{
+ int err;
+ snd_hda_power_up(codec);
+ mutex_lock(&codec->bus->cmd_mutex);
+ err = codec->bus->ops.command(codec, nid, direct, verb, parm);
+ if (!err) {
+ struct hda_cache_head *c;
+ u32 key = build_cmd_cache_key(nid, verb);
+ c = get_alloc_hash(&codec->cmd_cache, key);
+ if (c)
+ c->val = parm;
+ }
+ mutex_unlock(&codec->bus->cmd_mutex);
+ snd_hda_power_down(codec);
+ return err;
+}
+
+/* resume the all commands from the cache */
+void snd_hda_codec_resume_cache(struct hda_codec *codec)
+{
+ struct hda_cache_head *buffer = codec->cmd_cache.buffer;
+ int i;
+
+ for (i = 0; i < codec->cmd_cache.size; i++, buffer++) {
+ u32 key = buffer->key;
+ if (!key)
+ continue;
+ snd_hda_codec_write(codec, get_cmd_cache_nid(key), 0,
+ get_cmd_cache_cmd(key), buffer->val);
+ }
+}
+
+/**
+ * snd_hda_sequence_write_cache - sequence writes with caching
+ * @codec: the HDA codec
+ * @seq: VERB array to send
+ *
+ * Send the commands sequentially from the given array.
+ * Thte commands are recorded on cache for power-save and resume.
+ * The array must be terminated with NID=0.
+ */
+void snd_hda_sequence_write_cache(struct hda_codec *codec,
+ const struct hda_verb *seq)
+{
+ for (; seq->nid; seq++)
+ snd_hda_codec_write_cache(codec, seq->nid, 0, seq->verb,
+ seq->param);
+}
+#endif /* SND_HDA_NEEDS_RESUME */
/*
* set power state of the codec
@@ -1345,23 +1622,86 @@ int __devinit snd_hda_create_spdif_in_ctls(struct hda_codec *codec,
static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg,
unsigned int power_state)
{
- hda_nid_t nid, nid_start;
- int nodes;
+ hda_nid_t nid;
+ int i;
snd_hda_codec_write(codec, fg, 0, AC_VERB_SET_POWER_STATE,
power_state);
- nodes = snd_hda_get_sub_nodes(codec, fg, &nid_start);
- for (nid = nid_start; nid < nodes + nid_start; nid++) {
- if (get_wcaps(codec, nid) & AC_WCAP_POWER)
+ nid = codec->start_nid;
+ for (i = 0; i < codec->num_nodes; i++, nid++) {
+ if (get_wcaps(codec, nid) & AC_WCAP_POWER) {
+ unsigned int pincap;
+ /*
+ * don't power down the widget if it controls eapd
+ * and EAPD_BTLENABLE is set.
+ */
+ pincap = snd_hda_param_read(codec, nid, AC_PAR_PIN_CAP);
+ if (pincap & AC_PINCAP_EAPD) {
+ int eapd = snd_hda_codec_read(codec, nid,
+ 0, AC_VERB_GET_EAPD_BTLENABLE, 0);
+ eapd &= 0x02;
+ if (power_state == AC_PWRST_D3 && eapd)
+ continue;
+ }
snd_hda_codec_write(codec, nid, 0,
AC_VERB_SET_POWER_STATE,
power_state);
+ }
}
- if (power_state == AC_PWRST_D0)
+ if (power_state == AC_PWRST_D0) {
+ unsigned long end_time;
+ int state;
msleep(10);
+ /* wait until the codec reachs to D0 */
+ end_time = jiffies + msecs_to_jiffies(500);
+ do {
+ state = snd_hda_codec_read(codec, fg, 0,
+ AC_VERB_GET_POWER_STATE, 0);
+ if (state == power_state)
+ break;
+ msleep(1);
+ } while (time_after_eq(end_time, jiffies));
+ }
+}
+
+#ifdef SND_HDA_NEEDS_RESUME
+/*
+ * call suspend and power-down; used both from PM and power-save
+ */
+static void hda_call_codec_suspend(struct hda_codec *codec)
+{
+ if (codec->patch_ops.suspend)
+ codec->patch_ops.suspend(codec, PMSG_SUSPEND);
+ hda_set_power_state(codec,
+ codec->afg ? codec->afg : codec->mfg,
+ AC_PWRST_D3);
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ cancel_delayed_work(&codec->power_work);
+ codec->power_on = 0;
+ codec->power_transition = 0;
+#endif
+}
+
+/*
+ * kick up codec; used both from PM and power-save
+ */
+static void hda_call_codec_resume(struct hda_codec *codec)
+{
+ hda_set_power_state(codec,
+ codec->afg ? codec->afg : codec->mfg,
+ AC_PWRST_D0);
+ if (codec->patch_ops.resume)
+ codec->patch_ops.resume(codec);
+ else {
+ if (codec->patch_ops.init)
+ codec->patch_ops.init(codec);
+ snd_hda_codec_resume_amp(codec);
+ snd_hda_codec_resume_cache(codec);
+ }
}
+#endif /* SND_HDA_NEEDS_RESUME */
/**
@@ -1376,28 +1716,24 @@ int __devinit snd_hda_build_controls(struct hda_bus *bus)
{
struct hda_codec *codec;
- /* build controls */
list_for_each_entry(codec, &bus->codec_list, list) {
- int err;
- if (!codec->patch_ops.build_controls)
- continue;
- err = codec->patch_ops.build_controls(codec);
- if (err < 0)
- return err;
- }
-
- /* initialize */
- list_for_each_entry(codec, &bus->codec_list, list) {
- int err;
+ int err = 0;
+ /* fake as if already powered-on */
+ hda_keep_power_on(codec);
+ /* then fire up */
hda_set_power_state(codec,
codec->afg ? codec->afg : codec->mfg,
AC_PWRST_D0);
- if (!codec->patch_ops.init)
- continue;
- err = codec->patch_ops.init(codec);
+ /* continue to initialize... */
+ if (codec->patch_ops.init)
+ err = codec->patch_ops.init(codec);
+ if (!err && codec->patch_ops.build_controls)
+ err = codec->patch_ops.build_controls(codec);
+ snd_hda_power_down(codec);
if (err < 0)
return err;
}
+
return 0;
}
@@ -1789,9 +2125,9 @@ int __devinit snd_hda_build_pcms(struct hda_bus *bus)
*
* If no entries are matching, the function returns a negative value.
*/
-int __devinit snd_hda_check_board_config(struct hda_codec *codec,
- int num_configs, const char **models,
- const struct snd_pci_quirk *tbl)
+int snd_hda_check_board_config(struct hda_codec *codec,
+ int num_configs, const char **models,
+ const struct snd_pci_quirk *tbl)
{
if (codec->bus->modelname && models) {
int i;
@@ -1841,10 +2177,9 @@ int __devinit snd_hda_check_board_config(struct hda_codec *codec,
*
* Returns 0 if successful, or a negative error code.
*/
-int __devinit snd_hda_add_new_ctls(struct hda_codec *codec,
- struct snd_kcontrol_new *knew)
+int snd_hda_add_new_ctls(struct hda_codec *codec, struct snd_kcontrol_new *knew)
{
- int err;
+ int err;
for (; knew->name; knew++) {
struct snd_kcontrol *kctl;
@@ -1867,6 +2202,93 @@ int __devinit snd_hda_add_new_ctls(struct hda_codec *codec,
return 0;
}
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg,
+ unsigned int power_state);
+
+static void hda_power_work(struct work_struct *work)
+{
+ struct hda_codec *codec =
+ container_of(work, struct hda_codec, power_work.work);
+
+ if (!codec->power_on || codec->power_count) {
+ codec->power_transition = 0;
+ return;
+ }
+
+ hda_call_codec_suspend(codec);
+ if (codec->bus->ops.pm_notify)
+ codec->bus->ops.pm_notify(codec);
+}
+
+static void hda_keep_power_on(struct hda_codec *codec)
+{
+ codec->power_count++;
+ codec->power_on = 1;
+}
+
+void snd_hda_power_up(struct hda_codec *codec)
+{
+ codec->power_count++;
+ if (codec->power_on || codec->power_transition)
+ return;
+
+ codec->power_on = 1;
+ if (codec->bus->ops.pm_notify)
+ codec->bus->ops.pm_notify(codec);
+ hda_call_codec_resume(codec);
+ cancel_delayed_work(&codec->power_work);
+ codec->power_transition = 0;
+}
+
+void snd_hda_power_down(struct hda_codec *codec)
+{
+ --codec->power_count;
+ if (!codec->power_on || codec->power_count || codec->power_transition)
+ return;
+ if (power_save) {
+ codec->power_transition = 1; /* avoid reentrance */
+ schedule_delayed_work(&codec->power_work,
+ msecs_to_jiffies(power_save * 1000));
+ }
+}
+
+int snd_hda_check_amp_list_power(struct hda_codec *codec,
+ struct hda_loopback_check *check,
+ hda_nid_t nid)
+{
+ struct hda_amp_list *p;
+ int ch, v;
+
+ if (!check->amplist)
+ return 0;
+ for (p = check->amplist; p->nid; p++) {
+ if (p->nid == nid)
+ break;
+ }
+ if (!p->nid)
+ return 0; /* nothing changed */
+
+ for (p = check->amplist; p->nid; p++) {
+ for (ch = 0; ch < 2; ch++) {
+ v = snd_hda_codec_amp_read(codec, p->nid, ch, p->dir,
+ p->idx);
+ if (!(v & HDA_AMP_MUTE) && v > 0) {
+ if (!check->power_on) {
+ check->power_on = 1;
+ snd_hda_power_up(codec);
+ }
+ return 1;
+ }
+ }
+ }
+ if (check->power_on) {
+ check->power_on = 0;
+ snd_hda_power_down(codec);
+ }
+ return 0;
+}
+#endif
/*
* Channel mode helper
@@ -1913,12 +2335,12 @@ int snd_hda_ch_mode_put(struct hda_codec *codec,
mode = ucontrol->value.enumerated.item[0];
snd_assert(mode < num_chmodes, return -EINVAL);
- if (*max_channelsp == chmode[mode].channels && !codec->in_resume)
+ if (*max_channelsp == chmode[mode].channels)
return 0;
/* change the current channel setting */
*max_channelsp = chmode[mode].channels;
if (chmode[mode].sequence)
- snd_hda_sequence_write(codec, chmode[mode].sequence);
+ snd_hda_sequence_write_cache(codec, chmode[mode].sequence);
return 1;
}
@@ -1933,6 +2355,8 @@ int snd_hda_input_mux_info(const struct hda_input_mux *imux,
uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
uinfo->count = 1;
uinfo->value.enumerated.items = imux->num_items;
+ if (!imux->num_items)
+ return 0;
index = uinfo->value.enumerated.item;
if (index >= imux->num_items)
index = imux->num_items - 1;
@@ -1948,13 +2372,15 @@ int snd_hda_input_mux_put(struct hda_codec *codec,
{
unsigned int idx;
+ if (!imux->num_items)
+ return 0;
idx = ucontrol->value.enumerated.item[0];
if (idx >= imux->num_items)
idx = imux->num_items - 1;
- if (*cur_val == idx && !codec->in_resume)
+ if (*cur_val == idx)
return 0;
- snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CONNECT_SEL,
- imux->items[idx].index);
+ snd_hda_codec_write_cache(codec, nid, 0, AC_VERB_SET_CONNECT_SEL,
+ imux->items[idx].index);
*cur_val = idx;
return 1;
}
@@ -2118,7 +2544,7 @@ int snd_hda_multi_out_analog_cleanup(struct hda_codec *codec,
* Helper for automatic ping configuration
*/
-static int __devinit is_in_nid_list(hda_nid_t nid, hda_nid_t *list)
+static int is_in_nid_list(hda_nid_t nid, hda_nid_t *list)
{
for (; *list; list++)
if (*list == nid)
@@ -2169,9 +2595,9 @@ static void sort_pins_by_sequence(hda_nid_t * pins, short * sequences,
* The digital input/output pins are assigned to dig_in_pin and dig_out_pin,
* respectively.
*/
-int __devinit snd_hda_parse_pin_def_config(struct hda_codec *codec,
- struct auto_pin_cfg *cfg,
- hda_nid_t *ignore_nids)
+int snd_hda_parse_pin_def_config(struct hda_codec *codec,
+ struct auto_pin_cfg *cfg,
+ hda_nid_t *ignore_nids)
{
hda_nid_t nid, nid_start;
int nodes;
@@ -2371,13 +2797,12 @@ int snd_hda_suspend(struct hda_bus *bus, pm_message_t state)
{
struct hda_codec *codec;
- /* FIXME: should handle power widget capabilities */
list_for_each_entry(codec, &bus->codec_list, list) {
- if (codec->patch_ops.suspend)
- codec->patch_ops.suspend(codec, state);
- hda_set_power_state(codec,
- codec->afg ? codec->afg : codec->mfg,
- AC_PWRST_D3);
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ if (!codec->power_on)
+ continue;
+#endif
+ hda_call_codec_suspend(codec);
}
return 0;
}
@@ -2388,76 +2813,30 @@ int snd_hda_suspend(struct hda_bus *bus, pm_message_t state)
* @state: resume state
*
* Returns 0 if successful.
+ *
+ * This fucntion is defined only when POWER_SAVE isn't set.
+ * In the power-save mode, the codec is resumed dynamically.
*/
int snd_hda_resume(struct hda_bus *bus)
{
struct hda_codec *codec;
list_for_each_entry(codec, &bus->codec_list, list) {
- hda_set_power_state(codec,
- codec->afg ? codec->afg : codec->mfg,
- AC_PWRST_D0);
- if (codec->patch_ops.resume)
- codec->patch_ops.resume(codec);
+ if (snd_hda_codec_needs_resume(codec))
+ hda_call_codec_resume(codec);
}
return 0;
}
-
-/**
- * snd_hda_resume_ctls - resume controls in the new control list
- * @codec: the HDA codec
- * @knew: the array of struct snd_kcontrol_new
- *
- * This function resumes the mixer controls in the struct snd_kcontrol_new array,
- * originally for snd_hda_add_new_ctls().
- * The array must be terminated with an empty entry as terminator.
- */
-int snd_hda_resume_ctls(struct hda_codec *codec, struct snd_kcontrol_new *knew)
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+int snd_hda_codecs_inuse(struct hda_bus *bus)
{
- struct snd_ctl_elem_value *val;
+ struct hda_codec *codec;
- val = kmalloc(sizeof(*val), GFP_KERNEL);
- if (!val)
- return -ENOMEM;
- codec->in_resume = 1;
- for (; knew->name; knew++) {
- int i, count;
- count = knew->count ? knew->count : 1;
- for (i = 0; i < count; i++) {
- memset(val, 0, sizeof(*val));
- val->id.iface = knew->iface;
- val->id.device = knew->device;
- val->id.subdevice = knew->subdevice;
- strcpy(val->id.name, knew->name);
- val->id.index = knew->index ? knew->index : i;
- /* Assume that get callback reads only from cache,
- * not accessing to the real hardware
- */
- if (snd_ctl_elem_read(codec->bus->card, val) < 0)
- continue;
- snd_ctl_elem_write(codec->bus->card, NULL, val);
- }
+ list_for_each_entry(codec, &bus->codec_list, list) {
+ if (snd_hda_codec_needs_resume(codec))
+ return 1;
}
- codec->in_resume = 0;
- kfree(val);
return 0;
}
-
-/**
- * snd_hda_resume_spdif_out - resume the digital out
- * @codec: the HDA codec
- */
-int snd_hda_resume_spdif_out(struct hda_codec *codec)
-{
- return snd_hda_resume_ctls(codec, dig_mixes);
-}
-
-/**
- * snd_hda_resume_spdif_in - resume the digital in
- * @codec: the HDA codec
- */
-int snd_hda_resume_spdif_in(struct hda_codec *codec)
-{
- return snd_hda_resume_ctls(codec, dig_in_ctls);
-}
+#endif
#endif
diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h
index 56c26e7ccdf1..2bce925d84ef 100644
--- a/sound/pci/hda/hda_codec.h
+++ b/sound/pci/hda/hda_codec.h
@@ -24,6 +24,11 @@
#include <sound/info.h>
#include <sound/control.h>
#include <sound/pcm.h>
+#include <sound/hwdep.h>
+
+#if defined(CONFIG_PM) || defined(CONFIG_SND_HDA_POWER_SAVE)
+#define SND_HDA_NEEDS_RESUME /* resume control code is required */
+#endif
/*
* nodes
@@ -199,7 +204,9 @@ enum {
#define AC_AMPCAP_OFFSET_SHIFT 0
#define AC_AMPCAP_NUM_STEPS (0x7f<<8) /* number of steps */
#define AC_AMPCAP_NUM_STEPS_SHIFT 8
-#define AC_AMPCAP_STEP_SIZE (0x7f<<16) /* step size 0-32dB in 0.25dB */
+#define AC_AMPCAP_STEP_SIZE (0x7f<<16) /* step size 0-32dB
+ * in 0.25dB
+ */
#define AC_AMPCAP_STEP_SIZE_SHIFT 16
#define AC_AMPCAP_MUTE (1<<31) /* mute capable */
#define AC_AMPCAP_MUTE_SHIFT 31
@@ -409,6 +416,10 @@ struct hda_bus_ops {
unsigned int (*get_response)(struct hda_codec *codec);
/* free the private data */
void (*private_free)(struct hda_bus *);
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ /* notify power-up/down from codec to contoller */
+ void (*pm_notify)(struct hda_codec *codec);
+#endif
};
/* template to pass to the bus constructor */
@@ -436,7 +447,8 @@ struct hda_bus {
/* codec linked list */
struct list_head codec_list;
- struct hda_codec *caddr_tbl[HDA_MAX_CODEC_ADDRESS + 1]; /* caddr -> codec */
+ /* link caddr -> codec */
+ struct hda_codec *caddr_tbl[HDA_MAX_CODEC_ADDRESS + 1];
struct mutex cmd_mutex;
@@ -469,19 +481,34 @@ struct hda_codec_ops {
int (*init)(struct hda_codec *codec);
void (*free)(struct hda_codec *codec);
void (*unsol_event)(struct hda_codec *codec, unsigned int res);
-#ifdef CONFIG_PM
+#ifdef SND_HDA_NEEDS_RESUME
int (*suspend)(struct hda_codec *codec, pm_message_t state);
int (*resume)(struct hda_codec *codec);
#endif
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ int (*check_power_status)(struct hda_codec *codec, hda_nid_t nid);
+#endif
};
/* record for amp information cache */
-struct hda_amp_info {
+struct hda_cache_head {
u32 key; /* hash key */
+ u16 val; /* assigned value */
+ u16 next; /* next link; -1 = terminal */
+};
+
+struct hda_amp_info {
+ struct hda_cache_head head;
u32 amp_caps; /* amp capabilities */
u16 vol[2]; /* current volume & mute */
- u16 status; /* update flag */
- u16 next; /* next link */
+};
+
+struct hda_cache_rec {
+ u16 hash[64]; /* hash table for index */
+ unsigned int num_entries; /* number of assigned entries */
+ unsigned int size; /* allocated size */
+ unsigned int record_size; /* record size (including header) */
+ void *buffer; /* hash table entries */
};
/* PCM callbacks */
@@ -499,7 +526,7 @@ struct hda_pcm_ops {
/* PCM information for each substream */
struct hda_pcm_stream {
- unsigned int substreams; /* number of substreams, 0 = not exist */
+ unsigned int substreams; /* number of substreams, 0 = not exist*/
unsigned int channels_min; /* min. number of channels */
unsigned int channels_max; /* max. number of channels */
hda_nid_t nid; /* default NID to query rates/formats/bps, or set up */
@@ -536,11 +563,6 @@ struct hda_codec {
/* set by patch */
struct hda_codec_ops patch_ops;
- /* resume phase - all controls should update even if
- * the values are not changed
- */
- unsigned int in_resume;
-
/* PCM to create, set by patch_ops.build_pcms callback */
unsigned int num_pcms;
struct hda_pcm *pcm_info;
@@ -553,16 +575,22 @@ struct hda_codec {
hda_nid_t start_nid;
u32 *wcaps;
- /* hash for amp access */
- u16 amp_hash[32];
- int num_amp_entries;
- int amp_info_size;
- struct hda_amp_info *amp_info;
+ struct hda_cache_rec amp_cache; /* cache for amp access */
+ struct hda_cache_rec cmd_cache; /* cache for other commands */
struct mutex spdif_mutex;
unsigned int spdif_status; /* IEC958 status bits */
unsigned short spdif_ctls; /* SPDIF control bits */
unsigned int spdif_in_enable; /* SPDIF input enable? */
+
+ struct snd_hwdep *hwdep; /* assigned hwdep device */
+
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ unsigned int power_on :1; /* current (global) power-state */
+ unsigned int power_transition :1; /* power-state in transition */
+ int power_count; /* current (global) power refcount */
+ struct delayed_work power_work; /* delayed task for powerdown */
+#endif
};
/* direction */
@@ -582,13 +610,17 @@ int snd_hda_codec_new(struct hda_bus *bus, unsigned int codec_addr,
/*
* low level functions
*/
-unsigned int snd_hda_codec_read(struct hda_codec *codec, hda_nid_t nid, int direct,
+unsigned int snd_hda_codec_read(struct hda_codec *codec, hda_nid_t nid,
+ int direct,
unsigned int verb, unsigned int parm);
int snd_hda_codec_write(struct hda_codec *codec, hda_nid_t nid, int direct,
unsigned int verb, unsigned int parm);
-#define snd_hda_param_read(codec, nid, param) snd_hda_codec_read(codec, nid, 0, AC_VERB_PARAMETERS, param)
-int snd_hda_get_sub_nodes(struct hda_codec *codec, hda_nid_t nid, hda_nid_t *start_id);
-int snd_hda_get_connections(struct hda_codec *codec, hda_nid_t nid, hda_nid_t *conn_list, int max_conns);
+#define snd_hda_param_read(codec, nid, param) \
+ snd_hda_codec_read(codec, nid, 0, AC_VERB_PARAMETERS, param)
+int snd_hda_get_sub_nodes(struct hda_codec *codec, hda_nid_t nid,
+ hda_nid_t *start_id);
+int snd_hda_get_connections(struct hda_codec *codec, hda_nid_t nid,
+ hda_nid_t *conn_list, int max_conns);
struct hda_verb {
hda_nid_t nid;
@@ -596,11 +628,24 @@ struct hda_verb {
u32 param;
};
-void snd_hda_sequence_write(struct hda_codec *codec, const struct hda_verb *seq);
+void snd_hda_sequence_write(struct hda_codec *codec,
+ const struct hda_verb *seq);
/* unsolicited event */
int snd_hda_queue_unsol_event(struct hda_bus *bus, u32 res, u32 res_ex);
+/* cached write */
+#ifdef SND_HDA_NEEDS_RESUME
+int snd_hda_codec_write_cache(struct hda_codec *codec, hda_nid_t nid,
+ int direct, unsigned int verb, unsigned int parm);
+void snd_hda_sequence_write_cache(struct hda_codec *codec,
+ const struct hda_verb *seq);
+void snd_hda_codec_resume_cache(struct hda_codec *codec);
+#else
+#define snd_hda_codec_write_cache snd_hda_codec_write
+#define snd_hda_sequence_write_cache snd_hda_sequence_write
+#endif
+
/*
* Mixer
*/
@@ -610,10 +655,13 @@ int snd_hda_build_controls(struct hda_bus *bus);
* PCM
*/
int snd_hda_build_pcms(struct hda_bus *bus);
-void snd_hda_codec_setup_stream(struct hda_codec *codec, hda_nid_t nid, u32 stream_tag,
+void snd_hda_codec_setup_stream(struct hda_codec *codec, hda_nid_t nid,
+ u32 stream_tag,
int channel_id, int format);
-unsigned int snd_hda_calc_stream_format(unsigned int rate, unsigned int channels,
- unsigned int format, unsigned int maxbps);
+unsigned int snd_hda_calc_stream_format(unsigned int rate,
+ unsigned int channels,
+ unsigned int format,
+ unsigned int maxbps);
int snd_hda_query_supported_pcm(struct hda_codec *codec, hda_nid_t nid,
u32 *ratesp, u64 *formatsp, unsigned int *bpsp);
int snd_hda_is_supported_format(struct hda_codec *codec, hda_nid_t nid,
@@ -632,4 +680,19 @@ int snd_hda_suspend(struct hda_bus *bus, pm_message_t state);
int snd_hda_resume(struct hda_bus *bus);
#endif
+/*
+ * power saving
+ */
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+void snd_hda_power_up(struct hda_codec *codec);
+void snd_hda_power_down(struct hda_codec *codec);
+#define snd_hda_codec_needs_resume(codec) codec->power_count
+int snd_hda_codecs_inuse(struct hda_bus *bus);
+#else
+static inline void snd_hda_power_up(struct hda_codec *codec) {}
+static inline void snd_hda_power_down(struct hda_codec *codec) {}
+#define snd_hda_codec_needs_resume(codec) 1
+#define snd_hda_codecs_inuse(bus) 1
+#endif
+
#endif /* __SOUND_HDA_CODEC_H */
diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c
index 000287f7da43..c957eb58de5c 100644
--- a/sound/pci/hda/hda_generic.c
+++ b/sound/pci/hda/hda_generic.c
@@ -70,6 +70,13 @@ struct hda_gspec {
struct hda_pcm pcm_rec; /* PCM information */
struct list_head nid_list; /* list of widgets */
+
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+#define MAX_LOOPBACK_AMPS 7
+ struct hda_loopback_check loopback;
+ int num_loopbacks;
+ struct hda_amp_list loopback_list[MAX_LOOPBACK_AMPS + 1];
+#endif
};
/*
@@ -88,13 +95,12 @@ struct hda_gspec {
static void snd_hda_generic_free(struct hda_codec *codec)
{
struct hda_gspec *spec = codec->spec;
- struct list_head *p, *n;
+ struct hda_gnode *node, *n;
if (! spec)
return;
/* free all widgets */
- list_for_each_safe(p, n, &spec->nid_list) {
- struct hda_gnode *node = list_entry(p, struct hda_gnode, list);
+ list_for_each_entry_safe(node, n, &spec->nid_list, list) {
if (node->conn_list != node->slist)
kfree(node->conn_list);
kfree(node);
@@ -196,11 +202,9 @@ static int build_afg_tree(struct hda_codec *codec)
/* FIXME: should avoid the braindead linear search */
static struct hda_gnode *hda_get_node(struct hda_gspec *spec, hda_nid_t nid)
{
- struct list_head *p;
struct hda_gnode *node;
- list_for_each(p, &spec->nid_list) {
- node = list_entry(p, struct hda_gnode, list);
+ list_for_each_entry(node, &spec->nid_list, list) {
if (node->nid == nid)
return node;
}
@@ -218,9 +222,8 @@ static int unmute_output(struct hda_codec *codec, struct hda_gnode *node)
ofs = (node->amp_out_caps & AC_AMPCAP_OFFSET) >> AC_AMPCAP_OFFSET_SHIFT;
if (val >= ofs)
val -= ofs;
- val |= AC_AMP_SET_LEFT | AC_AMP_SET_RIGHT;
- val |= AC_AMP_SET_OUTPUT;
- return snd_hda_codec_write(codec, node->nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, val);
+ snd_hda_codec_amp_stereo(codec, node->nid, HDA_OUTPUT, 0, 0xff, val);
+ return 0;
}
/*
@@ -234,11 +237,8 @@ static int unmute_input(struct hda_codec *codec, struct hda_gnode *node, unsigne
ofs = (node->amp_in_caps & AC_AMPCAP_OFFSET) >> AC_AMPCAP_OFFSET_SHIFT;
if (val >= ofs)
val -= ofs;
- val |= AC_AMP_SET_LEFT | AC_AMP_SET_RIGHT;
- val |= AC_AMP_SET_INPUT;
- // awk added - fixed to allow unmuting of indexed amps
- val |= index << AC_AMP_SET_INDEX_SHIFT;
- return snd_hda_codec_write(codec, node->nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, val);
+ snd_hda_codec_amp_stereo(codec, node->nid, HDA_INPUT, index, 0xff, val);
+ return 0;
}
/*
@@ -248,7 +248,8 @@ static int select_input_connection(struct hda_codec *codec, struct hda_gnode *no
unsigned int index)
{
snd_printdd("CONNECT: NID=0x%x IDX=0x%x\n", node->nid, index);
- return snd_hda_codec_write(codec, node->nid, 0, AC_VERB_SET_CONNECT_SEL, index);
+ return snd_hda_codec_write_cache(codec, node->nid, 0,
+ AC_VERB_SET_CONNECT_SEL, index);
}
/*
@@ -256,11 +257,9 @@ static int select_input_connection(struct hda_codec *codec, struct hda_gnode *no
*/
static void clear_check_flags(struct hda_gspec *spec)
{
- struct list_head *p;
struct hda_gnode *node;
- list_for_each(p, &spec->nid_list) {
- node = list_entry(p, struct hda_gnode, list);
+ list_for_each_entry(node, &spec->nid_list, list) {
node->checked = 0;
}
}
@@ -343,12 +342,10 @@ static struct hda_gnode *parse_output_jack(struct hda_codec *codec,
struct hda_gspec *spec,
int jack_type)
{
- struct list_head *p;
struct hda_gnode *node;
int err;
- list_for_each(p, &spec->nid_list) {
- node = list_entry(p, struct hda_gnode, list);
+ list_for_each_entry(node, &spec->nid_list, list) {
if (node->type != AC_WID_PIN)
continue;
/* output capable? */
@@ -379,7 +376,7 @@ static struct hda_gnode *parse_output_jack(struct hda_codec *codec,
/* unmute the PIN output */
unmute_output(codec, node);
/* set PIN-Out enable */
- snd_hda_codec_write(codec, node->nid, 0,
+ snd_hda_codec_write_cache(codec, node->nid, 0,
AC_VERB_SET_PIN_WIDGET_CONTROL,
AC_PINCTL_OUT_EN |
((node->pin_caps & AC_PINCAP_HP_DRV) ?
@@ -570,7 +567,8 @@ static int parse_adc_sub_nodes(struct hda_codec *codec, struct hda_gspec *spec,
/* unmute the PIN external input */
unmute_input(codec, node, 0); /* index = 0? */
/* set PIN-In enable */
- snd_hda_codec_write(codec, node->nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, pinctl);
+ snd_hda_codec_write_cache(codec, node->nid, 0,
+ AC_VERB_SET_PIN_WIDGET_CONTROL, pinctl);
return 1; /* found */
}
@@ -659,7 +657,6 @@ static int parse_input_path(struct hda_codec *codec, struct hda_gnode *adc_node)
static int parse_input(struct hda_codec *codec)
{
struct hda_gspec *spec = codec->spec;
- struct list_head *p;
struct hda_gnode *node;
int err;
@@ -668,8 +665,7 @@ static int parse_input(struct hda_codec *codec)
* If it reaches to certain input PINs, we take it as the
* input path.
*/
- list_for_each(p, &spec->nid_list) {
- node = list_entry(p, struct hda_gnode, list);
+ list_for_each_entry(node, &spec->nid_list, list) {
if (node->wid_caps & AC_WCAP_DIGITAL)
continue; /* skip SPDIF */
if (node->type == AC_WID_AUD_IN) {
@@ -684,11 +680,33 @@ static int parse_input(struct hda_codec *codec)
return 0;
}
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+static void add_input_loopback(struct hda_codec *codec, hda_nid_t nid,
+ int dir, int idx)
+{
+ struct hda_gspec *spec = codec->spec;
+ struct hda_amp_list *p;
+
+ if (spec->num_loopbacks >= MAX_LOOPBACK_AMPS) {
+ snd_printk(KERN_ERR "hda_generic: Too many loopback ctls\n");
+ return;
+ }
+ p = &spec->loopback_list[spec->num_loopbacks++];
+ p->nid = nid;
+ p->dir = dir;
+ p->idx = idx;
+ spec->loopback.amplist = spec->loopback_list;
+}
+#else
+#define add_input_loopback(codec,nid,dir,idx)
+#endif
+
/*
* create mixer controls if possible
*/
static int create_mixer(struct hda_codec *codec, struct hda_gnode *node,
- unsigned int index, const char *type, const char *dir_sfx)
+ unsigned int index, const char *type,
+ const char *dir_sfx, int is_loopback)
{
char name[32];
int err;
@@ -702,6 +720,8 @@ static int create_mixer(struct hda_codec *codec, struct hda_gnode *node,
if ((node->wid_caps & AC_WCAP_IN_AMP) &&
(node->amp_in_caps & AC_AMPCAP_MUTE)) {
knew = (struct snd_kcontrol_new)HDA_CODEC_MUTE(name, node->nid, index, HDA_INPUT);
+ if (is_loopback)
+ add_input_loopback(codec, node->nid, HDA_INPUT, index);
snd_printdd("[%s] NID=0x%x, DIR=IN, IDX=0x%x\n", name, node->nid, index);
if ((err = snd_ctl_add(codec->bus->card, snd_ctl_new1(&knew, codec))) < 0)
return err;
@@ -709,6 +729,8 @@ static int create_mixer(struct hda_codec *codec, struct hda_gnode *node,
} else if ((node->wid_caps & AC_WCAP_OUT_AMP) &&
(node->amp_out_caps & AC_AMPCAP_MUTE)) {
knew = (struct snd_kcontrol_new)HDA_CODEC_MUTE(name, node->nid, 0, HDA_OUTPUT);
+ if (is_loopback)
+ add_input_loopback(codec, node->nid, HDA_OUTPUT, 0);
snd_printdd("[%s] NID=0x%x, DIR=OUT\n", name, node->nid);
if ((err = snd_ctl_add(codec->bus->card, snd_ctl_new1(&knew, codec))) < 0)
return err;
@@ -767,7 +789,7 @@ static int create_output_mixers(struct hda_codec *codec, const char **names)
for (i = 0; i < spec->pcm_vol_nodes; i++) {
err = create_mixer(codec, spec->pcm_vol[i].node,
spec->pcm_vol[i].index,
- names[i], "Playback");
+ names[i], "Playback", 0);
if (err < 0)
return err;
}
@@ -784,7 +806,7 @@ static int build_output_controls(struct hda_codec *codec)
case 1:
return create_mixer(codec, spec->pcm_vol[0].node,
spec->pcm_vol[0].index,
- "Master", "Playback");
+ "Master", "Playback", 0);
case 2:
if (defcfg_type(spec->out_pin_node[0]) == AC_JACK_SPEAKER)
return create_output_mixers(codec, types_speaker);
@@ -820,7 +842,7 @@ static int build_input_controls(struct hda_codec *codec)
if (spec->input_mux.num_items == 1) {
err = create_mixer(codec, adc_node,
spec->input_mux.items[0].index,
- NULL, "Capture");
+ NULL, "Capture", 0);
if (err < 0)
return err;
return 0;
@@ -886,7 +908,8 @@ static int parse_loopback_path(struct hda_codec *codec, struct hda_gspec *spec,
return err;
else if (err >= 1) {
if (err == 1) {
- err = create_mixer(codec, node, i, type, "Playback");
+ err = create_mixer(codec, node, i, type,
+ "Playback", 1);
if (err < 0)
return err;
if (err > 0)
@@ -911,7 +934,6 @@ static int parse_loopback_path(struct hda_codec *codec, struct hda_gspec *spec,
static int build_loopback_controls(struct hda_codec *codec)
{
struct hda_gspec *spec = codec->spec;
- struct list_head *p;
struct hda_gnode *node;
int err;
const char *type;
@@ -919,8 +941,7 @@ static int build_loopback_controls(struct hda_codec *codec)
if (! spec->out_pin_node[0])
return 0;
- list_for_each(p, &spec->nid_list) {
- node = list_entry(p, struct hda_gnode, list);
+ list_for_each_entry(node, &spec->nid_list, list) {
if (node->type != AC_WID_PIN)
continue;
/* input capable? */
@@ -1022,6 +1043,14 @@ static int build_generic_pcms(struct hda_codec *codec)
return 0;
}
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+static int generic_check_power_status(struct hda_codec *codec, hda_nid_t nid)
+{
+ struct hda_gspec *spec = codec->spec;
+ return snd_hda_check_amp_list_power(codec, &spec->loopback, nid);
+}
+#endif
+
/*
*/
@@ -1029,6 +1058,9 @@ static struct hda_codec_ops generic_patch_ops = {
.build_controls = build_generic_controls,
.build_pcms = build_generic_pcms,
.free = snd_hda_generic_free,
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ .check_power_status = generic_check_power_status,
+#endif
};
/*
diff --git a/sound/pci/hda/hda_hwdep.c b/sound/pci/hda/hda_hwdep.c
new file mode 100644
index 000000000000..bafb7b01f5a1
--- /dev/null
+++ b/sound/pci/hda/hda_hwdep.c
@@ -0,0 +1,122 @@
+/*
+ * HWDEP Interface for HD-audio codec
+ *
+ * Copyright (c) 2007 Takashi Iwai <tiwai@suse.de>
+ *
+ * This driver is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This driver is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#include <sound/driver.h>
+#include <linux/init.h>
+#include <linux/slab.h>
+#include <linux/pci.h>
+#include <linux/compat.h>
+#include <linux/mutex.h>
+#include <sound/core.h>
+#include "hda_codec.h"
+#include "hda_local.h"
+#include <sound/hda_hwdep.h>
+
+/*
+ * write/read an out-of-bound verb
+ */
+static int verb_write_ioctl(struct hda_codec *codec,
+ struct hda_verb_ioctl __user *arg)
+{
+ u32 verb, res;
+
+ if (get_user(verb, &arg->verb))
+ return -EFAULT;
+ res = snd_hda_codec_read(codec, verb >> 24, 0,
+ (verb >> 8) & 0xffff, verb & 0xff);
+ if (put_user(res, &arg->res))
+ return -EFAULT;
+ return 0;
+}
+
+static int get_wcap_ioctl(struct hda_codec *codec,
+ struct hda_verb_ioctl __user *arg)
+{
+ u32 verb, res;
+
+ if (get_user(verb, &arg->verb))
+ return -EFAULT;
+ res = get_wcaps(codec, verb >> 24);
+ if (put_user(res, &arg->res))
+ return -EFAULT;
+ return 0;
+}
+
+
+/*
+ */
+static int hda_hwdep_ioctl(struct snd_hwdep *hw, struct file *file,
+ unsigned int cmd, unsigned long arg)
+{
+ struct hda_codec *codec = hw->private_data;
+ void __user *argp = (void __user *)arg;
+
+ switch (cmd) {
+ case HDA_IOCTL_PVERSION:
+ return put_user(HDA_HWDEP_VERSION, (int __user *)argp);
+ case HDA_IOCTL_VERB_WRITE:
+ return verb_write_ioctl(codec, argp);
+ case HDA_IOCTL_GET_WCAP:
+ return get_wcap_ioctl(codec, argp);
+ }
+ return -ENOIOCTLCMD;
+}
+
+#ifdef CONFIG_COMPAT
+static int hda_hwdep_ioctl_compat(struct snd_hwdep *hw, struct file *file,
+ unsigned int cmd, unsigned long arg)
+{
+ return hda_hwdep_ioctl(hw, file, cmd, (unsigned long)compat_ptr(arg));
+}
+#endif
+
+static int hda_hwdep_open(struct snd_hwdep *hw, struct file *file)
+{
+#ifndef CONFIG_SND_DEBUG_DETECT
+ if (!capable(CAP_SYS_RAWIO))
+ return -EACCES;
+#endif
+ return 0;
+}
+
+int __devinit snd_hda_create_hwdep(struct hda_codec *codec)
+{
+ char hwname[16];
+ struct snd_hwdep *hwdep;
+ int err;
+
+ sprintf(hwname, "HDA Codec %d", codec->addr);
+ err = snd_hwdep_new(codec->bus->card, hwname, codec->addr, &hwdep);
+ if (err < 0)
+ return err;
+ codec->hwdep = hwdep;
+ sprintf(hwdep->name, "HDA Codec %d", codec->addr);
+ hwdep->iface = SNDRV_HWDEP_IFACE_HDA;
+ hwdep->private_data = codec;
+ hwdep->exclusive = 1;
+
+ hwdep->ops.open = hda_hwdep_open;
+ hwdep->ops.ioctl = hda_hwdep_ioctl;
+#ifdef CONFIG_COMPAT
+ hwdep->ops.ioctl_compat = hda_hwdep_ioctl_compat;
+#endif
+
+ return 0;
+}
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 92bc8b3fa2a0..3fa0f9704909 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -1,6 +1,7 @@
/*
*
- * hda_intel.c - Implementation of primary alsa driver code base for Intel HD Audio.
+ * hda_intel.c - Implementation of primary alsa driver code base
+ * for Intel HD Audio.
*
* Copyright(c) 2004 Intel Corporation. All rights reserved.
*
@@ -64,14 +65,27 @@ MODULE_PARM_DESC(id, "ID string for Intel HD audio interface.");
module_param(model, charp, 0444);
MODULE_PARM_DESC(model, "Use the given board model.");
module_param(position_fix, int, 0444);
-MODULE_PARM_DESC(position_fix, "Fix DMA pointer (0 = auto, 1 = none, 2 = POSBUF, 3 = FIFO size).");
+MODULE_PARM_DESC(position_fix, "Fix DMA pointer "
+ "(0 = auto, 1 = none, 2 = POSBUF, 3 = FIFO size).");
module_param(probe_mask, int, 0444);
MODULE_PARM_DESC(probe_mask, "Bitmask to probe codecs (default = -1).");
module_param(single_cmd, bool, 0444);
-MODULE_PARM_DESC(single_cmd, "Use single command to communicate with codecs (for debugging only).");
+MODULE_PARM_DESC(single_cmd, "Use single command to communicate with codecs "
+ "(for debugging only).");
module_param(enable_msi, int, 0);
MODULE_PARM_DESC(enable_msi, "Enable Message Signaled Interrupt (MSI)");
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+/* power_save option is defined in hda_codec.c */
+
+/* reset the HD-audio controller in power save mode.
+ * 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, bool, 0644);
+MODULE_PARM_DESC(power_save_controller, "Reset controller in power save mode.");
+#endif
/* just for backward compatibility */
static int enable;
@@ -98,6 +112,7 @@ MODULE_DESCRIPTION("Intel HDA driver");
#define SFX "hda-intel: "
+
/*
* registers
*/
@@ -213,15 +228,16 @@ enum { SDI0, SDI1, SDI2, SDI3, SDO0, SDO1, SDO2, SDO3 };
#define SD_INT_DESC_ERR 0x10 /* descriptor error interrupt */
#define SD_INT_FIFO_ERR 0x08 /* FIFO error interrupt */
#define SD_INT_COMPLETE 0x04 /* completion interrupt */
-#define SD_INT_MASK (SD_INT_DESC_ERR|SD_INT_FIFO_ERR|SD_INT_COMPLETE)
+#define SD_INT_MASK (SD_INT_DESC_ERR|SD_INT_FIFO_ERR|\
+ SD_INT_COMPLETE)
/* SD_STS */
#define SD_STS_FIFO_READY 0x20 /* FIFO ready */
/* INTCTL and INTSTS */
-#define ICH6_INT_ALL_STREAM 0xff /* all stream interrupts */
-#define ICH6_INT_CTRL_EN 0x40000000 /* controller interrupt enable bit */
-#define ICH6_INT_GLOBAL_EN 0x80000000 /* global interrupt enable bit */
+#define ICH6_INT_ALL_STREAM 0xff /* all stream interrupts */
+#define ICH6_INT_CTRL_EN 0x40000000 /* controller interrupt enable bit */
+#define ICH6_INT_GLOBAL_EN 0x80000000 /* global interrupt enable bit */
/* GCTL unsolicited response enable bit */
#define ICH6_GCTL_UREN (1<<8)
@@ -257,22 +273,26 @@ enum {
*/
struct azx_dev {
- u32 *bdl; /* virtual address of the BDL */
- dma_addr_t bdl_addr; /* physical address of the BDL */
- u32 *posbuf; /* position buffer pointer */
+ u32 *bdl; /* virtual address of the BDL */
+ dma_addr_t bdl_addr; /* physical address of the BDL */
+ u32 *posbuf; /* position buffer pointer */
- unsigned int bufsize; /* size of the play buffer in bytes */
- unsigned int fragsize; /* size of each period in bytes */
- unsigned int frags; /* number for period in the play buffer */
- unsigned int fifo_size; /* FIFO size */
+ unsigned int bufsize; /* size of the play buffer in bytes */
+ unsigned int fragsize; /* size of each period in bytes */
+ unsigned int frags; /* number for period in the play buffer */
+ unsigned int fifo_size; /* FIFO size */
- void __iomem *sd_addr; /* stream descriptor pointer */
+ void __iomem *sd_addr; /* stream descriptor pointer */
- u32 sd_int_sta_mask; /* stream int status mask */
+ u32 sd_int_sta_mask; /* stream int status mask */
/* pcm support */
- struct snd_pcm_substream *substream; /* assigned substream, set in PCM open */
- unsigned int format_val; /* format value to be set in the controller and the codec */
+ struct snd_pcm_substream *substream; /* assigned substream,
+ * set in PCM open
+ */
+ unsigned int format_val; /* format value to be set in the
+ * controller and the codec
+ */
unsigned char stream_tag; /* assigned stream */
unsigned char index; /* stream index */
/* for sanity check of position buffer */
@@ -337,6 +357,7 @@ struct azx {
/* flags */
int position_fix;
+ unsigned int running :1;
unsigned int initialized :1;
unsigned int single_cmd :1;
unsigned int polling_mode :1;
@@ -418,7 +439,8 @@ static int azx_alloc_cmd_io(struct azx *chip)
int err;
/* single page (at least 4096 bytes) must suffice for both ringbuffes */
- err = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(chip->pci),
+ err = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV,
+ snd_dma_pci_data(chip->pci),
PAGE_SIZE, &chip->rb);
if (err < 0) {
snd_printk(KERN_ERR SFX "cannot allocate CORB/RIRB\n");
@@ -531,9 +553,9 @@ static unsigned int azx_rirb_get_response(struct hda_codec *codec)
azx_update_rirb(chip);
spin_unlock_irq(&chip->reg_lock);
}
- if (! chip->rirb.cmds)
+ if (!chip->rirb.cmds)
return chip->rirb.res; /* the last value */
- schedule_timeout(1);
+ schedule_timeout_uninterruptible(1);
} while (time_after_eq(timeout, jiffies));
if (chip->msi) {
@@ -585,16 +607,19 @@ static int azx_single_send_cmd(struct hda_codec *codec, u32 val)
while (timeout--) {
/* check ICB busy bit */
- if (! (azx_readw(chip, IRS) & ICH6_IRS_BUSY)) {
+ if (!((azx_readw(chip, IRS) & ICH6_IRS_BUSY))) {
/* Clear IRV valid bit */
- azx_writew(chip, IRS, azx_readw(chip, IRS) | ICH6_IRS_VALID);
+ azx_writew(chip, IRS, azx_readw(chip, IRS) |
+ ICH6_IRS_VALID);
azx_writel(chip, IC, val);
- azx_writew(chip, IRS, azx_readw(chip, IRS) | ICH6_IRS_BUSY);
+ azx_writew(chip, IRS, azx_readw(chip, IRS) |
+ ICH6_IRS_BUSY);
return 0;
}
udelay(1);
}
- snd_printd(SFX "send_cmd timeout: IRS=0x%x, val=0x%x\n", azx_readw(chip, IRS), val);
+ snd_printd(SFX "send_cmd timeout: IRS=0x%x, val=0x%x\n",
+ azx_readw(chip, IRS), val);
return -EIO;
}
@@ -610,7 +635,8 @@ static unsigned int azx_single_get_response(struct hda_codec *codec)
return azx_readl(chip, IR);
udelay(1);
}
- snd_printd(SFX "get_response timeout: IRS=0x%x\n", azx_readw(chip, IRS));
+ snd_printd(SFX "get_response timeout: IRS=0x%x\n",
+ azx_readw(chip, IRS));
return (unsigned int)-1;
}
@@ -652,12 +678,18 @@ static unsigned int azx_get_response(struct hda_codec *codec)
return azx_rirb_get_response(codec);
}
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+static void azx_power_notify(struct hda_codec *codec);
+#endif
/* reset codec link */
static int azx_reset(struct azx *chip)
{
int count;
+ /* clear STATESTS */
+ azx_writeb(chip, STATESTS, STATESTS_INT_MASK);
+
/* reset controller */
azx_writel(chip, GCTL, azx_readl(chip, GCTL) & ~ICH6_GCTL_RESET);
@@ -777,18 +809,12 @@ static void azx_stream_stop(struct azx *chip, struct azx_dev *azx_dev)
/*
- * initialize the chip
+ * reset and start the controller registers
*/
static void azx_init_chip(struct azx *chip)
{
- unsigned char reg;
-
- /* Clear bits 0-2 of PCI register TCSEL (at offset 0x44)
- * TCSEL == Traffic Class Select Register, which sets PCI express QOS
- * Ensuring these bits are 0 clears playback static on some HD Audio codecs
- */
- pci_read_config_byte (chip->pci, ICH6_PCIREG_TCSEL, &reg);
- pci_write_config_byte(chip->pci, ICH6_PCIREG_TCSEL, reg & 0xf8);
+ if (chip->initialized)
+ return;
/* reset controller */
azx_reset(chip);
@@ -805,19 +831,45 @@ static void azx_init_chip(struct azx *chip)
azx_writel(chip, DPLBASE, (u32)chip->posbuf.addr);
azx_writel(chip, DPUBASE, upper_32bit(chip->posbuf.addr));
+ chip->initialized = 1;
+}
+
+/*
+ * initialize the PCI registers
+ */
+/* update bits in a PCI register byte */
+static void update_pci_byte(struct pci_dev *pci, unsigned int reg,
+ unsigned char mask, unsigned char val)
+{
+ unsigned char data;
+
+ pci_read_config_byte(pci, reg, &data);
+ data &= ~mask;
+ data |= (val & mask);
+ pci_write_config_byte(pci, reg, data);
+}
+
+static void azx_init_pci(struct azx *chip)
+{
+ /* Clear bits 0-2 of PCI register TCSEL (at offset 0x44)
+ * TCSEL == Traffic Class Select Register, which sets PCI express QOS
+ * Ensuring these bits are 0 clears playback static on some HD Audio
+ * codecs
+ */
+ update_pci_byte(chip->pci, ICH6_PCIREG_TCSEL, 0x07, 0);
+
switch (chip->driver_type) {
case AZX_DRIVER_ATI:
/* For ATI SB450 azalia HD audio, we need to enable snoop */
- pci_read_config_byte(chip->pci, ATI_SB450_HDAUDIO_MISC_CNTR2_ADDR,
- &reg);
- pci_write_config_byte(chip->pci, ATI_SB450_HDAUDIO_MISC_CNTR2_ADDR,
- (reg & 0xf8) | ATI_SB450_HDAUDIO_ENABLE_SNOOP);
+ update_pci_byte(chip->pci,
+ ATI_SB450_HDAUDIO_MISC_CNTR2_ADDR,
+ 0x07, ATI_SB450_HDAUDIO_ENABLE_SNOOP);
break;
case AZX_DRIVER_NVIDIA:
/* For NVIDIA HDA, enable snoop */
- pci_read_config_byte(chip->pci,NVIDIA_HDA_TRANSREG_ADDR, &reg);
- pci_write_config_byte(chip->pci,NVIDIA_HDA_TRANSREG_ADDR,
- (reg & 0xf0) | NVIDIA_HDA_ENABLE_COHBITS);
+ update_pci_byte(chip->pci,
+ NVIDIA_HDA_TRANSREG_ADDR,
+ 0x0f, NVIDIA_HDA_ENABLE_COHBITS);
break;
}
}
@@ -857,7 +909,7 @@ static irqreturn_t azx_interrupt(int irq, void *dev_id)
/* clear rirb int */
status = azx_readb(chip, RIRBSTS);
if (status & RIRB_INT_MASK) {
- if (! chip->single_cmd && (status & RIRB_INT_RESPONSE))
+ if (!chip->single_cmd && (status & RIRB_INT_RESPONSE))
azx_update_rirb(chip);
azx_writeb(chip, RIRBSTS, RIRB_INT_MASK);
}
@@ -911,9 +963,11 @@ static int azx_setup_controller(struct azx *chip, struct azx_dev *azx_dev)
int timeout;
/* make sure the run bit is zero for SD */
- azx_sd_writeb(azx_dev, SD_CTL, azx_sd_readb(azx_dev, SD_CTL) & ~SD_CTL_DMA_START);
+ azx_sd_writeb(azx_dev, SD_CTL, azx_sd_readb(azx_dev, SD_CTL) &
+ ~SD_CTL_DMA_START);
/* reset stream */
- azx_sd_writeb(azx_dev, SD_CTL, azx_sd_readb(azx_dev, SD_CTL) | SD_CTL_STREAM_RESET);
+ azx_sd_writeb(azx_dev, SD_CTL, azx_sd_readb(azx_dev, SD_CTL) |
+ SD_CTL_STREAM_RESET);
udelay(3);
timeout = 300;
while (!((val = azx_sd_readb(azx_dev, SD_CTL)) & SD_CTL_STREAM_RESET) &&
@@ -931,7 +985,7 @@ static int azx_setup_controller(struct azx *chip, struct azx_dev *azx_dev)
/* program the stream_tag */
azx_sd_writel(azx_dev, SD_CTL,
- (azx_sd_readl(azx_dev, SD_CTL) & ~SD_CTL_STREAM_TAG_MASK) |
+ (azx_sd_readl(azx_dev, SD_CTL) & ~SD_CTL_STREAM_TAG_MASK)|
(azx_dev->stream_tag << SD_CTL_STREAM_TAG_SHIFT));
/* program the length of samples in cyclic buffer */
@@ -951,11 +1005,13 @@ static int azx_setup_controller(struct azx *chip, struct azx_dev *azx_dev)
azx_sd_writel(azx_dev, SD_BDLPU, upper_32bit(azx_dev->bdl_addr));
/* enable the position buffer */
- if (! (azx_readl(chip, DPLBASE) & ICH6_DPLBASE_ENABLE))
- azx_writel(chip, DPLBASE, (u32)chip->posbuf.addr | ICH6_DPLBASE_ENABLE);
+ if (!(azx_readl(chip, DPLBASE) & ICH6_DPLBASE_ENABLE))
+ azx_writel(chip, DPLBASE,
+ (u32)chip->posbuf.addr |ICH6_DPLBASE_ENABLE);
/* set the interrupt enable bits in the descriptor control register */
- azx_sd_writel(azx_dev, SD_CTL, azx_sd_readl(azx_dev, SD_CTL) | SD_INT_MASK);
+ azx_sd_writel(azx_dev, SD_CTL,
+ azx_sd_readl(azx_dev, SD_CTL) | SD_INT_MASK);
return 0;
}
@@ -986,8 +1042,12 @@ static int __devinit azx_codec_create(struct azx *chip, const char *model)
bus_temp.pci = chip->pci;
bus_temp.ops.command = azx_send_cmd;
bus_temp.ops.get_response = azx_get_response;
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ bus_temp.ops.pm_notify = azx_power_notify;
+#endif
- if ((err = snd_hda_bus_new(chip->card, &bus_temp, &chip->bus)) < 0)
+ err = snd_hda_bus_new(chip->card, &bus_temp, &chip->bus);
+ if (err < 0)
return err;
codecs = audio_codecs = 0;
@@ -1038,7 +1098,7 @@ static inline struct azx_dev *azx_assign_device(struct azx *chip, int stream)
nums = chip->capture_streams;
}
for (i = 0; i < nums; i++, dev++)
- if (! chip->azx_dev[dev].opened) {
+ if (!chip->azx_dev[dev].opened) {
chip->azx_dev[dev].opened = 1;
return &chip->azx_dev[dev];
}
@@ -1052,7 +1112,8 @@ static inline void azx_release_device(struct azx_dev *azx_dev)
}
static struct snd_pcm_hardware azx_pcm_hw = {
- .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED |
+ .info = (SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_INTERLEAVED |
SNDRV_PCM_INFO_BLOCK_TRANSFER |
SNDRV_PCM_INFO_MMAP_VALID |
/* No full-resume yet implemented */
@@ -1105,8 +1166,11 @@ static int azx_pcm_open(struct snd_pcm_substream *substream)
128);
snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES,
128);
- if ((err = hinfo->ops.open(hinfo, apcm->codec, substream)) < 0) {
+ snd_hda_power_up(apcm->codec);
+ err = hinfo->ops.open(hinfo, apcm->codec, substream);
+ if (err < 0) {
azx_release_device(azx_dev);
+ snd_hda_power_down(apcm->codec);
mutex_unlock(&chip->open_mutex);
return err;
}
@@ -1135,13 +1199,16 @@ static int azx_pcm_close(struct snd_pcm_substream *substream)
spin_unlock_irqrestore(&chip->reg_lock, flags);
azx_release_device(azx_dev);
hinfo->ops.close(hinfo, apcm->codec, substream);
+ snd_hda_power_down(apcm->codec);
mutex_unlock(&chip->open_mutex);
return 0;
}
-static int azx_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *hw_params)
+static int azx_pcm_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *hw_params)
{
- return snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params));
+ return snd_pcm_lib_malloc_pages(substream,
+ params_buffer_bytes(hw_params));
}
static int azx_pcm_hw_free(struct snd_pcm_substream *substream)
@@ -1175,13 +1242,15 @@ static int azx_pcm_prepare(struct snd_pcm_substream *substream)
runtime->channels,
runtime->format,
hinfo->maxbps);
- if (! azx_dev->format_val) {
- snd_printk(KERN_ERR SFX "invalid format_val, rate=%d, ch=%d, format=%d\n",
+ if (!azx_dev->format_val) {
+ snd_printk(KERN_ERR SFX
+ "invalid format_val, rate=%d, ch=%d, format=%d\n",
runtime->rate, runtime->channels, runtime->format);
return -EINVAL;
}
- snd_printdd("azx_pcm_prepare: bufsize=0x%x, fragsize=0x%x, format=0x%x\n",
+ snd_printdd("azx_pcm_prepare: bufsize=0x%x, fragsize=0x%x, "
+ "format=0x%x\n",
azx_dev->bufsize, azx_dev->fragsize, azx_dev->format_val);
azx_setup_periods(azx_dev);
azx_setup_controller(chip, azx_dev);
@@ -1223,7 +1292,8 @@ static int azx_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
cmd == SNDRV_PCM_TRIGGER_SUSPEND ||
cmd == SNDRV_PCM_TRIGGER_STOP) {
int timeout = 5000;
- while (azx_sd_readb(azx_dev, SD_CTL) & SD_CTL_DMA_START && --timeout)
+ while ((azx_sd_readb(azx_dev, SD_CTL) & SD_CTL_DMA_START) &&
+ --timeout)
;
}
return err;
@@ -1241,7 +1311,7 @@ static snd_pcm_uframes_t azx_pcm_pointer(struct snd_pcm_substream *substream)
/* use the position buffer */
pos = le32_to_cpu(*azx_dev->posbuf);
if (chip->position_fix == POS_FIX_AUTO &&
- azx_dev->period_intr == 1 && ! pos) {
+ azx_dev->period_intr == 1 && !pos) {
printk(KERN_WARNING
"hda-intel: Invalid position buffer, "
"using LPIB read method instead.\n");
@@ -1292,7 +1362,8 @@ static int __devinit create_codec_pcm(struct azx *chip, struct hda_codec *codec,
snd_assert(cpcm->name, return -EINVAL);
err = snd_pcm_new(chip->card, cpcm->name, pcm_dev,
- cpcm->stream[0].substreams, cpcm->stream[1].substreams,
+ cpcm->stream[0].substreams,
+ cpcm->stream[1].substreams,
&pcm);
if (err < 0)
return err;
@@ -1322,26 +1393,27 @@ static int __devinit create_codec_pcm(struct azx *chip, struct hda_codec *codec,
static int __devinit azx_pcm_create(struct azx *chip)
{
- struct list_head *p;
struct hda_codec *codec;
int c, err;
int pcm_dev;
- if ((err = snd_hda_build_pcms(chip->bus)) < 0)
+ err = snd_hda_build_pcms(chip->bus);
+ if (err < 0)
return err;
/* create audio PCMs */
pcm_dev = 0;
- list_for_each(p, &chip->bus->codec_list) {
- codec = list_entry(p, struct hda_codec, list);
+ list_for_each_entry(codec, &chip->bus->codec_list, list) {
for (c = 0; c < codec->num_pcms; c++) {
if (codec->pcm_info[c].is_modem)
continue; /* create later */
if (pcm_dev >= AZX_MAX_AUDIO_PCMS) {
- snd_printk(KERN_ERR SFX "Too many audio PCMs\n");
+ snd_printk(KERN_ERR SFX
+ "Too many audio PCMs\n");
return -EINVAL;
}
- err = create_codec_pcm(chip, codec, &codec->pcm_info[c], pcm_dev);
+ err = create_codec_pcm(chip, codec,
+ &codec->pcm_info[c], pcm_dev);
if (err < 0)
return err;
pcm_dev++;
@@ -1350,16 +1422,17 @@ static int __devinit azx_pcm_create(struct azx *chip)
/* create modem PCMs */
pcm_dev = AZX_MAX_AUDIO_PCMS;
- list_for_each(p, &chip->bus->codec_list) {
- codec = list_entry(p, struct hda_codec, list);
+ list_for_each_entry(codec, &chip->bus->codec_list, list) {
for (c = 0; c < codec->num_pcms; c++) {
- if (! codec->pcm_info[c].is_modem)
+ if (!codec->pcm_info[c].is_modem)
continue; /* already created */
if (pcm_dev >= AZX_MAX_PCMS) {
- snd_printk(KERN_ERR SFX "Too many modem PCMs\n");
+ snd_printk(KERN_ERR SFX
+ "Too many modem PCMs\n");
return -EINVAL;
}
- err = create_codec_pcm(chip, codec, &codec->pcm_info[c], pcm_dev);
+ err = create_codec_pcm(chip, codec,
+ &codec->pcm_info[c], pcm_dev);
if (err < 0)
return err;
chip->pcm[pcm_dev]->dev_class = SNDRV_PCM_CLASS_MODEM;
@@ -1386,7 +1459,8 @@ static int __devinit azx_init_stream(struct azx *chip)
int i;
/* initialize each stream (aka device)
- * assign the starting bdl address to each stream (device) and initialize
+ * assign the starting bdl address to each stream (device)
+ * and initialize
*/
for (i = 0; i < chip->num_streams; i++) {
unsigned int off = sizeof(u32) * (i * AZX_MAX_FRAG * 4);
@@ -1423,6 +1497,46 @@ static int azx_acquire_irq(struct azx *chip, int do_disconnect)
}
+static void azx_stop_chip(struct azx *chip)
+{
+ if (!chip->initialized)
+ return;
+
+ /* disable interrupts */
+ azx_int_disable(chip);
+ azx_int_clear(chip);
+
+ /* disable CORB/RIRB */
+ azx_free_cmd_io(chip);
+
+ /* disable position buffer */
+ azx_writel(chip, DPLBASE, 0);
+ azx_writel(chip, DPUBASE, 0);
+
+ chip->initialized = 0;
+}
+
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+/* power-up/down the controller */
+static void azx_power_notify(struct hda_codec *codec)
+{
+ struct azx *chip = codec->bus->private_data;
+ struct hda_codec *c;
+ int power_on = 0;
+
+ list_for_each_entry(c, &codec->bus->codec_list, list) {
+ if (c->power_on) {
+ power_on = 1;
+ break;
+ }
+ }
+ if (power_on)
+ azx_init_chip(chip);
+ else if (chip->running && power_save_controller)
+ azx_stop_chip(chip);
+}
+#endif /* CONFIG_SND_HDA_POWER_SAVE */
+
#ifdef CONFIG_PM
/*
* power management
@@ -1436,8 +1550,9 @@ static int azx_suspend(struct pci_dev *pci, pm_message_t state)
snd_power_change_state(card, SNDRV_CTL_POWER_D3hot);
for (i = 0; i < chip->pcm_devs; i++)
snd_pcm_suspend_all(chip->pcm[i]);
- snd_hda_suspend(chip->bus, state);
- azx_free_cmd_io(chip);
+ if (chip->initialized)
+ snd_hda_suspend(chip->bus, state);
+ azx_stop_chip(chip);
if (chip->irq >= 0) {
synchronize_irq(chip->irq);
free_irq(chip->irq, chip);
@@ -1470,7 +1585,11 @@ static int azx_resume(struct pci_dev *pci)
chip->msi = 0;
if (azx_acquire_irq(chip, 1) < 0)
return -EIO;
- azx_init_chip(chip);
+ azx_init_pci(chip);
+
+ if (snd_hda_codecs_inuse(chip->bus))
+ azx_init_chip(chip);
+
snd_hda_resume(chip->bus);
snd_power_change_state(card, SNDRV_CTL_POWER_D0);
return 0;
@@ -1485,20 +1604,9 @@ static int azx_free(struct azx *chip)
{
if (chip->initialized) {
int i;
-
for (i = 0; i < chip->num_streams; i++)
azx_stream_stop(chip, &chip->azx_dev[i]);
-
- /* disable interrupts */
- azx_int_disable(chip);
- azx_int_clear(chip);
-
- /* disable CORB/RIRB */
- azx_free_cmd_io(chip);
-
- /* disable position buffer */
- azx_writel(chip, DPLBASE, 0);
- azx_writel(chip, DPUBASE, 0);
+ azx_stop_chip(chip);
}
if (chip->irq >= 0) {
@@ -1534,6 +1642,7 @@ static int azx_dev_free(struct snd_device *device)
*/
static struct snd_pci_quirk position_fix_list[] __devinitdata = {
SND_PCI_QUIRK(0x1028, 0x01cc, "Dell D820", POS_FIX_NONE),
+ SND_PCI_QUIRK(0x1028, 0x01de, "Dell Precision 390", POS_FIX_NONE),
{}
};
@@ -1544,7 +1653,7 @@ static int __devinit check_position_fix(struct azx *chip, int fix)
if (fix == POS_FIX_AUTO) {
q = snd_pci_quirk_lookup(chip->pci, position_fix_list);
if (q) {
- snd_printdd(KERN_INFO
+ printk(KERN_INFO
"hda_intel: position_fix set to %d "
"for device %04x:%04x\n",
q->value, q->subvendor, q->subdevice);
@@ -1555,6 +1664,36 @@ static int __devinit check_position_fix(struct azx *chip, int fix)
}
/*
+ * black-lists for probe_mask
+ */
+static struct snd_pci_quirk probe_mask_list[] __devinitdata = {
+ /* Thinkpad often breaks the controller communication when accessing
+ * to the non-working (or non-existing) modem codec slot.
+ */
+ SND_PCI_QUIRK(0x1014, 0x05b7, "Thinkpad Z60", 0x01),
+ SND_PCI_QUIRK(0x17aa, 0x2010, "Thinkpad X/T/R60", 0x01),
+ SND_PCI_QUIRK(0x17aa, 0x20ac, "Thinkpad X/T/R61", 0x01),
+ {}
+};
+
+static void __devinit check_probe_mask(struct azx *chip)
+{
+ const struct snd_pci_quirk *q;
+
+ if (probe_mask == -1) {
+ q = snd_pci_quirk_lookup(chip->pci, probe_mask_list);
+ if (q) {
+ printk(KERN_INFO
+ "hda_intel: probe_mask set to 0x%x "
+ "for device %04x:%04x\n",
+ q->value, q->subvendor, q->subdevice);
+ probe_mask = q->value;
+ }
+ }
+}
+
+
+/*
* constructor
*/
static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci,
@@ -1589,6 +1728,7 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci,
chip->msi = enable_msi;
chip->position_fix = check_position_fix(chip, position_fix);
+ check_probe_mask(chip);
chip->single_cmd = single_cmd;
@@ -1650,37 +1790,43 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci,
break;
}
chip->num_streams = chip->playback_streams + chip->capture_streams;
- chip->azx_dev = kcalloc(chip->num_streams, sizeof(*chip->azx_dev), GFP_KERNEL);
+ chip->azx_dev = kcalloc(chip->num_streams, sizeof(*chip->azx_dev),
+ GFP_KERNEL);
if (!chip->azx_dev) {
snd_printk(KERN_ERR "cannot malloc azx_dev\n");
goto errout;
}
/* allocate memory for the BDL for each stream */
- if ((err = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(chip->pci),
- BDL_SIZE, &chip->bdl)) < 0) {
+ err = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV,
+ snd_dma_pci_data(chip->pci),
+ BDL_SIZE, &chip->bdl);
+ if (err < 0) {
snd_printk(KERN_ERR SFX "cannot allocate BDL\n");
goto errout;
}
/* allocate memory for the position buffer */
- if ((err = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(chip->pci),
- chip->num_streams * 8, &chip->posbuf)) < 0) {
+ err = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV,
+ snd_dma_pci_data(chip->pci),
+ chip->num_streams * 8, &chip->posbuf);
+ if (err < 0) {
snd_printk(KERN_ERR SFX "cannot allocate posbuf\n");
goto errout;
}
/* allocate CORB/RIRB */
- if (! chip->single_cmd)
- if ((err = azx_alloc_cmd_io(chip)) < 0)
+ if (!chip->single_cmd) {
+ err = azx_alloc_cmd_io(chip);
+ if (err < 0)
goto errout;
+ }
/* initialize streams */
azx_init_stream(chip);
/* initialize chip */
+ azx_init_pci(chip);
azx_init_chip(chip);
- chip->initialized = 1;
-
/* codec detection */
if (!chip->codec_mask) {
snd_printk(KERN_ERR SFX "no codecs found!\n");
@@ -1688,14 +1834,16 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci,
goto errout;
}
- if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops)) <0) {
+ err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops);
+ if (err <0) {
snd_printk(KERN_ERR SFX "Error creating device [card]!\n");
goto errout;
}
strcpy(card->driver, "HDA-Intel");
strcpy(card->shortname, driver_short_names[chip->driver_type]);
- sprintf(card->longname, "%s at 0x%lx irq %i", card->shortname, chip->addr, chip->irq);
+ sprintf(card->longname, "%s at 0x%lx irq %i",
+ card->shortname, chip->addr, chip->irq);
*rchip = chip;
return 0;
@@ -1705,7 +1853,21 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci,
return err;
}
-static int __devinit azx_probe(struct pci_dev *pci, const struct pci_device_id *pci_id)
+static void power_down_all_codecs(struct azx *chip)
+{
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ /* The codecs were powered up in snd_hda_codec_new().
+ * Now all initialization done, so turn them down if possible
+ */
+ struct hda_codec *codec;
+ list_for_each_entry(codec, &chip->bus->codec_list, list) {
+ snd_hda_power_down(codec);
+ }
+#endif
+}
+
+static int __devinit azx_probe(struct pci_dev *pci,
+ const struct pci_device_id *pci_id)
{
struct snd_card *card;
struct azx *chip;
@@ -1725,31 +1887,37 @@ static int __devinit azx_probe(struct pci_dev *pci, const struct pci_device_id *
card->private_data = chip;
/* create codec instances */
- if ((err = azx_codec_create(chip, model)) < 0) {
+ err = azx_codec_create(chip, model);
+ if (err < 0) {
snd_card_free(card);
return err;
}
/* create PCM streams */
- if ((err = azx_pcm_create(chip)) < 0) {
+ err = azx_pcm_create(chip);
+ if (err < 0) {
snd_card_free(card);
return err;
}
/* create mixer controls */
- if ((err = azx_mixer_create(chip)) < 0) {
+ err = azx_mixer_create(chip);
+ if (err < 0) {
snd_card_free(card);
return err;
}
snd_card_set_dev(card, &pci->dev);
- if ((err = snd_card_register(card)) < 0) {
+ err = snd_card_register(card);
+ if (err < 0) {
snd_card_free(card);
return err;
}
pci_set_drvdata(pci, card);
+ chip->running = 1;
+ power_down_all_codecs(chip);
return err;
}
@@ -1791,6 +1959,10 @@ static struct pci_device_id azx_ids[] = {
{ 0x10de, 0x0775, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_NVIDIA }, /* NVIDIA MCP77 */
{ 0x10de, 0x0776, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_NVIDIA }, /* NVIDIA MCP77 */
{ 0x10de, 0x0777, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_NVIDIA }, /* NVIDIA MCP77 */
+ { 0x10de, 0x0ac0, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_NVIDIA }, /* NVIDIA MCP79 */
+ { 0x10de, 0x0ac1, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_NVIDIA }, /* NVIDIA MCP79 */
+ { 0x10de, 0x0ac2, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_NVIDIA }, /* NVIDIA MCP79 */
+ { 0x10de, 0x0ac3, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_NVIDIA }, /* NVIDIA MCP79 */
{ 0, }
};
MODULE_DEVICE_TABLE(pci, azx_ids);
diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h
index f91ea5ec9f6d..a79d0ed5469c 100644
--- a/sound/pci/hda/hda_local.h
+++ b/sound/pci/hda/hda_local.h
@@ -26,7 +26,8 @@
/*
* for mixer controls
*/
-#define HDA_COMPOSE_AMP_VAL(nid,chs,idx,dir) ((nid) | ((chs)<<16) | ((dir)<<18) | ((idx)<<19))
+#define HDA_COMPOSE_AMP_VAL(nid,chs,idx,dir) \
+ ((nid) | ((chs)<<16) | ((dir)<<18) | ((idx)<<19))
/* mono volume with index (index=0,1,...) (channel=1,2) */
#define HDA_CODEC_VOLUME_MONO_IDX(xname, xcidx, nid, channel, xindex, direction) \
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xcidx, \
@@ -64,18 +65,35 @@
#define HDA_CODEC_MUTE(xname, nid, xindex, direction) \
HDA_CODEC_MUTE_MONO(xname, nid, 3, xindex, direction)
-int snd_hda_mixer_amp_volume_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo);
-int snd_hda_mixer_amp_volume_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol);
-int snd_hda_mixer_amp_volume_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol);
-int snd_hda_mixer_amp_tlv(struct snd_kcontrol *kcontrol, int op_flag, unsigned int size, unsigned int __user *tlv);
-int snd_hda_mixer_amp_switch_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo);
-int snd_hda_mixer_amp_switch_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol);
-int snd_hda_mixer_amp_switch_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol);
+int snd_hda_mixer_amp_volume_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo);
+int snd_hda_mixer_amp_volume_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol);
+int snd_hda_mixer_amp_volume_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol);
+int snd_hda_mixer_amp_tlv(struct snd_kcontrol *kcontrol, int op_flag,
+ unsigned int size, unsigned int __user *tlv);
+int snd_hda_mixer_amp_switch_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo);
+int snd_hda_mixer_amp_switch_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol);
+int snd_hda_mixer_amp_switch_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol);
/* lowlevel accessor with caching; use carefully */
int snd_hda_codec_amp_read(struct hda_codec *codec, hda_nid_t nid, int ch,
int direction, int index);
int snd_hda_codec_amp_update(struct hda_codec *codec, hda_nid_t nid, int ch,
int direction, int idx, int mask, int val);
+int snd_hda_codec_amp_stereo(struct hda_codec *codec, hda_nid_t nid,
+ int dir, int idx, int mask, int val);
+#ifdef SND_HDA_NEEDS_RESUME
+void snd_hda_codec_resume_amp(struct hda_codec *codec);
+#endif
+
+/* amp value bits */
+#define HDA_AMP_MUTE 0x80
+#define HDA_AMP_UNMUTE 0x00
+#define HDA_AMP_VOLMASK 0x7f
/* mono switch binding multiple inputs */
#define HDA_BIND_MUTE_MONO(xname, nid, channel, indices, direction) \
@@ -86,11 +104,61 @@ int snd_hda_codec_amp_update(struct hda_codec *codec, hda_nid_t nid, int ch,
.private_value = HDA_COMPOSE_AMP_VAL(nid, channel, indices, direction) }
/* stereo switch binding multiple inputs */
-#define HDA_BIND_MUTE(xname,nid,indices,dir) HDA_BIND_MUTE_MONO(xname,nid,3,indices,dir)
+#define HDA_BIND_MUTE(xname,nid,indices,dir) \
+ HDA_BIND_MUTE_MONO(xname,nid,3,indices,dir)
+
+int snd_hda_mixer_bind_switch_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol);
+int snd_hda_mixer_bind_switch_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol);
+
+/* more generic bound controls */
+struct hda_ctl_ops {
+ snd_kcontrol_info_t *info;
+ snd_kcontrol_get_t *get;
+ snd_kcontrol_put_t *put;
+ snd_kcontrol_tlv_rw_t *tlv;
+};
-int snd_hda_mixer_bind_switch_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol);
-int snd_hda_mixer_bind_switch_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol);
+extern struct hda_ctl_ops snd_hda_bind_vol; /* for bind-volume with TLV */
+extern struct hda_ctl_ops snd_hda_bind_sw; /* for bind-switch */
+struct hda_bind_ctls {
+ struct hda_ctl_ops *ops;
+ long values[];
+};
+
+int snd_hda_mixer_bind_ctls_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo);
+int snd_hda_mixer_bind_ctls_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol);
+int snd_hda_mixer_bind_ctls_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol);
+int snd_hda_mixer_bind_tlv(struct snd_kcontrol *kcontrol, int op_flag,
+ unsigned int size, unsigned int __user *tlv);
+
+#define HDA_BIND_VOL(xname, bindrec) \
+ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
+ .name = xname, \
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE |\
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ |\
+ SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK,\
+ .info = snd_hda_mixer_bind_ctls_info,\
+ .get = snd_hda_mixer_bind_ctls_get,\
+ .put = snd_hda_mixer_bind_ctls_put,\
+ .tlv = { .c = snd_hda_mixer_bind_tlv },\
+ .private_value = (long) (bindrec) }
+#define HDA_BIND_SW(xname, bindrec) \
+ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER,\
+ .name = xname, \
+ .info = snd_hda_mixer_bind_ctls_info,\
+ .get = snd_hda_mixer_bind_ctls_get,\
+ .put = snd_hda_mixer_bind_ctls_put,\
+ .private_value = (long) (bindrec) }
+
+/*
+ * SPDIF I/O
+ */
int snd_hda_create_spdif_out_ctls(struct hda_codec *codec, hda_nid_t nid);
int snd_hda_create_spdif_in_ctls(struct hda_codec *codec, hda_nid_t nid);
@@ -107,8 +175,10 @@ struct hda_input_mux {
struct hda_input_mux_item items[HDA_MAX_NUM_INPUTS];
};
-int snd_hda_input_mux_info(const struct hda_input_mux *imux, struct snd_ctl_elem_info *uinfo);
-int snd_hda_input_mux_put(struct hda_codec *codec, const struct hda_input_mux *imux,
+int snd_hda_input_mux_info(const struct hda_input_mux *imux,
+ struct snd_ctl_elem_info *uinfo);
+int snd_hda_input_mux_put(struct hda_codec *codec,
+ const struct hda_input_mux *imux,
struct snd_ctl_elem_value *ucontrol, hda_nid_t nid,
unsigned int *cur_val);
@@ -120,13 +190,19 @@ struct hda_channel_mode {
const struct hda_verb *sequence;
};
-int snd_hda_ch_mode_info(struct hda_codec *codec, struct snd_ctl_elem_info *uinfo,
- const struct hda_channel_mode *chmode, int num_chmodes);
-int snd_hda_ch_mode_get(struct hda_codec *codec, struct snd_ctl_elem_value *ucontrol,
- const struct hda_channel_mode *chmode, int num_chmodes,
+int snd_hda_ch_mode_info(struct hda_codec *codec,
+ struct snd_ctl_elem_info *uinfo,
+ const struct hda_channel_mode *chmode,
+ int num_chmodes);
+int snd_hda_ch_mode_get(struct hda_codec *codec,
+ struct snd_ctl_elem_value *ucontrol,
+ const struct hda_channel_mode *chmode,
+ int num_chmodes,
int max_channels);
-int snd_hda_ch_mode_put(struct hda_codec *codec, struct snd_ctl_elem_value *ucontrol,
- const struct hda_channel_mode *chmode, int num_chmodes,
+int snd_hda_ch_mode_put(struct hda_codec *codec,
+ struct snd_ctl_elem_value *ucontrol,
+ const struct hda_channel_mode *chmode,
+ int num_chmodes,
int *max_channelsp);
/*
@@ -146,20 +222,25 @@ struct hda_multi_out {
int dig_out_used; /* current usage of digital out (HDA_DIG_XXX) */
};
-int snd_hda_multi_out_dig_open(struct hda_codec *codec, struct hda_multi_out *mout);
-int snd_hda_multi_out_dig_close(struct hda_codec *codec, struct hda_multi_out *mout);
+int snd_hda_multi_out_dig_open(struct hda_codec *codec,
+ struct hda_multi_out *mout);
+int snd_hda_multi_out_dig_close(struct hda_codec *codec,
+ struct hda_multi_out *mout);
int snd_hda_multi_out_dig_prepare(struct hda_codec *codec,
struct hda_multi_out *mout,
unsigned int stream_tag,
unsigned int format,
struct snd_pcm_substream *substream);
-int snd_hda_multi_out_analog_open(struct hda_codec *codec, struct hda_multi_out *mout,
+int snd_hda_multi_out_analog_open(struct hda_codec *codec,
+ struct hda_multi_out *mout,
struct snd_pcm_substream *substream);
-int snd_hda_multi_out_analog_prepare(struct hda_codec *codec, struct hda_multi_out *mout,
+int snd_hda_multi_out_analog_prepare(struct hda_codec *codec,
+ struct hda_multi_out *mout,
unsigned int stream_tag,
unsigned int format,
struct snd_pcm_substream *substream);
-int snd_hda_multi_out_analog_cleanup(struct hda_codec *codec, struct hda_multi_out *mout);
+int snd_hda_multi_out_analog_cleanup(struct hda_codec *codec,
+ struct hda_multi_out *mout);
/*
* generic codec parser
@@ -181,16 +262,8 @@ static inline int snd_hda_codec_proc_new(struct hda_codec *codec) { return 0; }
int snd_hda_check_board_config(struct hda_codec *codec, int num_configs,
const char **modelnames,
const struct snd_pci_quirk *pci_list);
-int snd_hda_add_new_ctls(struct hda_codec *codec, struct snd_kcontrol_new *knew);
-
-/*
- * power management
- */
-#ifdef CONFIG_PM
-int snd_hda_resume_ctls(struct hda_codec *codec, struct snd_kcontrol_new *knew);
-int snd_hda_resume_spdif_out(struct hda_codec *codec);
-int snd_hda_resume_spdif_in(struct hda_codec *codec);
-#endif
+int snd_hda_add_new_ctls(struct hda_codec *codec,
+ struct snd_kcontrol_new *knew);
/*
* unsolicited event handler
@@ -232,7 +305,9 @@ extern const char *auto_pin_cfg_labels[AUTO_PIN_LAST];
struct auto_pin_cfg {
int line_outs;
- hda_nid_t line_out_pins[5]; /* sorted in the order of Front/Surr/CLFE/Side */
+ hda_nid_t line_out_pins[5]; /* sorted in the order of
+ * Front/Surr/CLFE/Side
+ */
int speaker_outs;
hda_nid_t speaker_pins[5];
int hp_outs;
@@ -243,13 +318,19 @@ struct auto_pin_cfg {
hda_nid_t dig_in_pin;
};
-#define get_defcfg_connect(cfg) ((cfg & AC_DEFCFG_PORT_CONN) >> AC_DEFCFG_PORT_CONN_SHIFT)
-#define get_defcfg_association(cfg) ((cfg & AC_DEFCFG_DEF_ASSOC) >> AC_DEFCFG_ASSOC_SHIFT)
-#define get_defcfg_location(cfg) ((cfg & AC_DEFCFG_LOCATION) >> AC_DEFCFG_LOCATION_SHIFT)
-#define get_defcfg_sequence(cfg) (cfg & AC_DEFCFG_SEQUENCE)
-#define get_defcfg_device(cfg) ((cfg & AC_DEFCFG_DEVICE) >> AC_DEFCFG_DEVICE_SHIFT)
-
-int snd_hda_parse_pin_def_config(struct hda_codec *codec, struct auto_pin_cfg *cfg,
+#define get_defcfg_connect(cfg) \
+ ((cfg & AC_DEFCFG_PORT_CONN) >> AC_DEFCFG_PORT_CONN_SHIFT)
+#define get_defcfg_association(cfg) \
+ ((cfg & AC_DEFCFG_DEF_ASSOC) >> AC_DEFCFG_ASSOC_SHIFT)
+#define get_defcfg_location(cfg) \
+ ((cfg & AC_DEFCFG_LOCATION) >> AC_DEFCFG_LOCATION_SHIFT)
+#define get_defcfg_sequence(cfg) \
+ (cfg & AC_DEFCFG_SEQUENCE)
+#define get_defcfg_device(cfg) \
+ ((cfg & AC_DEFCFG_DEVICE) >> AC_DEFCFG_DEVICE_SHIFT)
+
+int snd_hda_parse_pin_def_config(struct hda_codec *codec,
+ struct auto_pin_cfg *cfg,
hda_nid_t *ignore_nids);
/* amp values */
@@ -280,4 +361,32 @@ static inline u32 get_wcaps(struct hda_codec *codec, hda_nid_t nid)
int snd_hda_override_amp_caps(struct hda_codec *codec, hda_nid_t nid, int dir,
unsigned int caps);
+/*
+ * hwdep interface
+ */
+int snd_hda_create_hwdep(struct hda_codec *codec);
+
+/*
+ * power-management
+ */
+
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+void snd_hda_schedule_power_save(struct hda_codec *codec);
+
+struct hda_amp_list {
+ hda_nid_t nid;
+ unsigned char dir;
+ unsigned char idx;
+};
+
+struct hda_loopback_check {
+ struct hda_amp_list *amplist;
+ int power_on;
+};
+
+int snd_hda_check_amp_list_power(struct hda_codec *codec,
+ struct hda_loopback_check *check,
+ hda_nid_t nid);
+#endif /* CONFIG_SND_HDA_POWER_SAVE */
+
#endif /* __SOUND_HDA_LOCAL_H */
diff --git a/sound/pci/hda/hda_patch.h b/sound/pci/hda/hda_patch.h
index 9f9e9ae44a9d..f5c23bb16d7e 100644
--- a/sound/pci/hda/hda_patch.h
+++ b/sound/pci/hda/hda_patch.h
@@ -20,13 +20,29 @@ extern struct hda_codec_preset snd_hda_preset_conexant[];
extern struct hda_codec_preset snd_hda_preset_via[];
static const struct hda_codec_preset *hda_preset_tables[] = {
+#ifdef CONFIG_SND_HDA_CODEC_REALTEK
snd_hda_preset_realtek,
+#endif
+#ifdef CONFIG_SND_HDA_CODEC_CMEDIA
snd_hda_preset_cmedia,
+#endif
+#ifdef CONFIG_SND_HDA_CODEC_ANALOG
snd_hda_preset_analog,
+#endif
+#ifdef CONFIG_SND_HDA_CODEC_SIGMATEL
snd_hda_preset_sigmatel,
+#endif
+#ifdef CONFIG_SND_HDA_CODEC_SI3054
snd_hda_preset_si3054,
+#endif
+#ifdef CONFIG_SND_HDA_CODEC_ATIHDMI
snd_hda_preset_atihdmi,
+#endif
+#ifdef CONFIG_SND_HDA_CODEC_CONEXANT
snd_hda_preset_conexant,
+#endif
+#ifdef CONFIG_SND_HDA_CODEC_VIA
snd_hda_preset_via,
+#endif
NULL
};
diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c
index ac15066fd300..e94944f34ffd 100644
--- a/sound/pci/hda/hda_proc.c
+++ b/sound/pci/hda/hda_proc.c
@@ -58,7 +58,8 @@ static void print_amp_caps(struct snd_info_buffer *buffer,
snd_iprintf(buffer, "N/A\n");
return;
}
- snd_iprintf(buffer, "ofs=0x%02x, nsteps=0x%02x, stepsize=0x%02x, mute=%x\n",
+ snd_iprintf(buffer, "ofs=0x%02x, nsteps=0x%02x, stepsize=0x%02x, "
+ "mute=%x\n",
caps & AC_AMPCAP_OFFSET,
(caps & AC_AMPCAP_NUM_STEPS) >> AC_AMPCAP_NUM_STEPS_SHIFT,
(caps & AC_AMPCAP_STEP_SIZE) >> AC_AMPCAP_STEP_SIZE_SHIFT,
@@ -76,11 +77,13 @@ static void print_amp_vals(struct snd_info_buffer *buffer,
for (i = 0; i < indices; i++) {
snd_iprintf(buffer, " [");
if (stereo) {
- val = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_AMP_GAIN_MUTE,
+ val = snd_hda_codec_read(codec, nid, 0,
+ AC_VERB_GET_AMP_GAIN_MUTE,
AC_AMP_GET_LEFT | dir | i);
snd_iprintf(buffer, "0x%02x ", val);
}
- val = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_AMP_GAIN_MUTE,
+ val = snd_hda_codec_read(codec, nid, 0,
+ AC_VERB_GET_AMP_GAIN_MUTE,
AC_AMP_GET_RIGHT | dir | i);
snd_iprintf(buffer, "0x%02x]", val);
}
@@ -237,7 +240,8 @@ static void print_pin_caps(struct snd_info_buffer *buffer,
}
-static void print_codec_info(struct snd_info_entry *entry, struct snd_info_buffer *buffer)
+static void print_codec_info(struct snd_info_entry *entry,
+ struct snd_info_buffer *buffer)
{
struct hda_codec *codec = entry->private_data;
char buf[32];
@@ -258,6 +262,7 @@ static void print_codec_info(struct snd_info_entry *entry, struct snd_info_buffe
if (! codec->afg)
return;
+ snd_hda_power_up(codec);
snd_iprintf(buffer, "Default PCM:\n");
print_pcm_caps(buffer, codec, codec->afg);
snd_iprintf(buffer, "Default Amp-In caps: ");
@@ -268,12 +273,15 @@ static void print_codec_info(struct snd_info_entry *entry, struct snd_info_buffe
nodes = snd_hda_get_sub_nodes(codec, codec->afg, &nid);
if (! nid || nodes < 0) {
snd_iprintf(buffer, "Invalid AFG subtree\n");
+ snd_hda_power_down(codec);
return;
}
for (i = 0; i < nodes; i++, nid++) {
- unsigned int wid_caps = snd_hda_param_read(codec, nid,
- AC_PAR_AUDIO_WIDGET_CAP);
- unsigned int wid_type = (wid_caps & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT;
+ unsigned int wid_caps =
+ snd_hda_param_read(codec, nid,
+ AC_PAR_AUDIO_WIDGET_CAP);
+ unsigned int wid_type =
+ (wid_caps & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT;
int conn_len = 0;
hda_nid_t conn[HDA_MAX_CONNECTIONS];
@@ -313,7 +321,9 @@ static void print_codec_info(struct snd_info_entry *entry, struct snd_info_buffe
if (wid_type == AC_WID_PIN) {
unsigned int pinctls;
print_pin_caps(buffer, codec, nid);
- pinctls = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
+ pinctls = snd_hda_codec_read(codec, nid, 0,
+ AC_VERB_GET_PIN_WIDGET_CONTROL,
+ 0);
snd_iprintf(buffer, " Pin-ctls: 0x%02x:", pinctls);
if (pinctls & AC_PINCTL_IN_EN)
snd_iprintf(buffer, " IN");
@@ -333,7 +343,8 @@ static void print_codec_info(struct snd_info_entry *entry, struct snd_info_buffe
if (wid_caps & AC_WCAP_POWER)
snd_iprintf(buffer, " Power: 0x%x\n",
snd_hda_codec_read(codec, nid, 0,
- AC_VERB_GET_POWER_STATE, 0));
+ AC_VERB_GET_POWER_STATE,
+ 0));
if (wid_caps & AC_WCAP_CONN_LIST) {
int c, curr = -1;
@@ -350,6 +361,7 @@ static void print_codec_info(struct snd_info_entry *entry, struct snd_info_buffe
snd_iprintf(buffer, "\n");
}
}
+ snd_hda_power_down(codec);
}
/*
diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index 4d7f8d11ad75..54cfd4526d20 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -73,6 +73,12 @@ struct ad198x_spec {
struct snd_kcontrol_new *kctl_alloc;
struct hda_input_mux private_imux;
hda_nid_t private_dac_nids[4];
+
+ unsigned int jack_present :1;
+
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ struct hda_loopback_check loopback;
+#endif
};
/*
@@ -144,6 +150,14 @@ static int ad198x_build_controls(struct hda_codec *codec)
return 0;
}
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+static int ad198x_check_power_status(struct hda_codec *codec, hda_nid_t nid)
+{
+ struct ad198x_spec *spec = codec->spec;
+ return snd_hda_check_amp_list_power(codec, &spec->loopback, nid);
+}
+#endif
+
/*
* Analog playback callbacks
*/
@@ -318,30 +332,13 @@ static void ad198x_free(struct hda_codec *codec)
kfree(codec->spec);
}
-#ifdef CONFIG_PM
-static int ad198x_resume(struct hda_codec *codec)
-{
- struct ad198x_spec *spec = codec->spec;
- int i;
-
- codec->patch_ops.init(codec);
- for (i = 0; i < spec->num_mixers; i++)
- snd_hda_resume_ctls(codec, spec->mixers[i]);
- if (spec->multiout.dig_out_nid)
- snd_hda_resume_spdif_out(codec);
- if (spec->dig_in_nid)
- snd_hda_resume_spdif_in(codec);
- return 0;
-}
-#endif
-
static struct hda_codec_ops ad198x_patch_ops = {
.build_controls = ad198x_build_controls,
.build_pcms = ad198x_build_pcms,
.init = ad198x_init,
.free = ad198x_free,
-#ifdef CONFIG_PM
- .resume = ad198x_resume,
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ .check_power_status = ad198x_check_power_status,
#endif
};
@@ -350,15 +347,7 @@ static struct hda_codec_ops ad198x_patch_ops = {
* EAPD control
* the private value = nid | (invert << 8)
*/
-static int ad198x_eapd_info(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_info *uinfo)
-{
- uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
- uinfo->count = 1;
- uinfo->value.integer.min = 0;
- uinfo->value.integer.max = 1;
- return 0;
-}
+#define ad198x_eapd_info snd_ctl_boolean_mono_info
static int ad198x_eapd_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
@@ -384,12 +373,12 @@ static int ad198x_eapd_put(struct snd_kcontrol *kcontrol,
eapd = ucontrol->value.integer.value[0];
if (invert)
eapd = !eapd;
- if (eapd == spec->cur_eapd && ! codec->in_resume)
+ if (eapd == spec->cur_eapd)
return 0;
spec->cur_eapd = eapd;
- snd_hda_codec_write(codec, nid,
- 0, AC_VERB_SET_EAPD_BTLENABLE,
- eapd ? 0x02 : 0x00);
+ snd_hda_codec_write_cache(codec, nid,
+ 0, AC_VERB_SET_EAPD_BTLENABLE,
+ eapd ? 0x02 : 0x00);
return 1;
}
@@ -430,94 +419,36 @@ static struct hda_input_mux ad1986a_capture_source = {
},
};
-/*
- * PCM control
- *
- * bind volumes/mutes of 3 DACs as a single PCM control for simplicity
- */
-
-#define ad1986a_pcm_amp_vol_info snd_hda_mixer_amp_volume_info
-
-static int ad1986a_pcm_amp_vol_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
-{
- struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
- struct ad198x_spec *ad = codec->spec;
-
- mutex_lock(&ad->amp_mutex);
- snd_hda_mixer_amp_volume_get(kcontrol, ucontrol);
- mutex_unlock(&ad->amp_mutex);
- return 0;
-}
-
-static int ad1986a_pcm_amp_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
-{
- struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
- struct ad198x_spec *ad = codec->spec;
- int i, change = 0;
-
- mutex_lock(&ad->amp_mutex);
- for (i = 0; i < ARRAY_SIZE(ad1986a_dac_nids); i++) {
- kcontrol->private_value = HDA_COMPOSE_AMP_VAL(ad1986a_dac_nids[i], 3, 0, HDA_OUTPUT);
- change |= snd_hda_mixer_amp_volume_put(kcontrol, ucontrol);
- }
- kcontrol->private_value = HDA_COMPOSE_AMP_VAL(AD1986A_FRONT_DAC, 3, 0, HDA_OUTPUT);
- mutex_unlock(&ad->amp_mutex);
- return change;
-}
-
-#define ad1986a_pcm_amp_sw_info snd_hda_mixer_amp_switch_info
-static int ad1986a_pcm_amp_sw_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
-{
- struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
- struct ad198x_spec *ad = codec->spec;
-
- mutex_lock(&ad->amp_mutex);
- snd_hda_mixer_amp_switch_get(kcontrol, ucontrol);
- mutex_unlock(&ad->amp_mutex);
- return 0;
-}
-
-static int ad1986a_pcm_amp_sw_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
-{
- struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
- struct ad198x_spec *ad = codec->spec;
- int i, change = 0;
+static struct hda_bind_ctls ad1986a_bind_pcm_vol = {
+ .ops = &snd_hda_bind_vol,
+ .values = {
+ HDA_COMPOSE_AMP_VAL(AD1986A_FRONT_DAC, 3, 0, HDA_OUTPUT),
+ HDA_COMPOSE_AMP_VAL(AD1986A_SURR_DAC, 3, 0, HDA_OUTPUT),
+ HDA_COMPOSE_AMP_VAL(AD1986A_CLFE_DAC, 3, 0, HDA_OUTPUT),
+ 0
+ },
+};
- mutex_lock(&ad->amp_mutex);
- for (i = 0; i < ARRAY_SIZE(ad1986a_dac_nids); i++) {
- kcontrol->private_value = HDA_COMPOSE_AMP_VAL(ad1986a_dac_nids[i], 3, 0, HDA_OUTPUT);
- change |= snd_hda_mixer_amp_switch_put(kcontrol, ucontrol);
- }
- kcontrol->private_value = HDA_COMPOSE_AMP_VAL(AD1986A_FRONT_DAC, 3, 0, HDA_OUTPUT);
- mutex_unlock(&ad->amp_mutex);
- return change;
-}
+static struct hda_bind_ctls ad1986a_bind_pcm_sw = {
+ .ops = &snd_hda_bind_sw,
+ .values = {
+ HDA_COMPOSE_AMP_VAL(AD1986A_FRONT_DAC, 3, 0, HDA_OUTPUT),
+ HDA_COMPOSE_AMP_VAL(AD1986A_SURR_DAC, 3, 0, HDA_OUTPUT),
+ HDA_COMPOSE_AMP_VAL(AD1986A_CLFE_DAC, 3, 0, HDA_OUTPUT),
+ 0
+ },
+};
/*
* mixers
*/
static struct snd_kcontrol_new ad1986a_mixers[] = {
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "PCM Playback Volume",
- .access = SNDRV_CTL_ELEM_ACCESS_READWRITE |
- SNDRV_CTL_ELEM_ACCESS_TLV_READ |
- SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK,
- .info = ad1986a_pcm_amp_vol_info,
- .get = ad1986a_pcm_amp_vol_get,
- .put = ad1986a_pcm_amp_vol_put,
- .tlv = { .c = snd_hda_mixer_amp_tlv },
- .private_value = HDA_COMPOSE_AMP_VAL(AD1986A_FRONT_DAC, 3, 0, HDA_OUTPUT)
- },
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "PCM Playback Switch",
- .info = ad1986a_pcm_amp_sw_info,
- .get = ad1986a_pcm_amp_sw_get,
- .put = ad1986a_pcm_amp_sw_put,
- .private_value = HDA_COMPOSE_AMP_VAL(AD1986A_FRONT_DAC, 3, 0, HDA_OUTPUT)
- },
+ /*
+ * bind volumes/mutes of 3 DACs as a single PCM control for simplicity
+ */
+ HDA_BIND_VOL("PCM Playback Volume", &ad1986a_bind_pcm_vol),
+ HDA_BIND_SW("PCM Playback Switch", &ad1986a_bind_pcm_sw),
HDA_CODEC_VOLUME("Front Playback Volume", 0x1b, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Front Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("Surround Playback Volume", 0x1c, 0x0, HDA_OUTPUT),
@@ -569,13 +500,30 @@ static struct snd_kcontrol_new ad1986a_3st_mixers[] = {
/* laptop model - 2ch only */
static hda_nid_t ad1986a_laptop_dac_nids[1] = { AD1986A_FRONT_DAC };
+/* master controls both pins 0x1a and 0x1b */
+static struct hda_bind_ctls ad1986a_laptop_master_vol = {
+ .ops = &snd_hda_bind_vol,
+ .values = {
+ HDA_COMPOSE_AMP_VAL(0x1a, 3, 0, HDA_OUTPUT),
+ HDA_COMPOSE_AMP_VAL(0x1b, 3, 0, HDA_OUTPUT),
+ 0,
+ },
+};
+
+static struct hda_bind_ctls ad1986a_laptop_master_sw = {
+ .ops = &snd_hda_bind_sw,
+ .values = {
+ HDA_COMPOSE_AMP_VAL(0x1a, 3, 0, HDA_OUTPUT),
+ HDA_COMPOSE_AMP_VAL(0x1b, 3, 0, HDA_OUTPUT),
+ 0,
+ },
+};
+
static struct snd_kcontrol_new ad1986a_laptop_mixers[] = {
HDA_CODEC_VOLUME("PCM Playback Volume", 0x03, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("PCM Playback Switch", 0x03, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Master Playback Volume", 0x1b, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Master Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
- /* HDA_CODEC_VOLUME("Headphone Playback Volume", 0x1a, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Headphone Playback Switch", 0x1a, 0x0, HDA_OUTPUT), */
+ HDA_BIND_VOL("Master Playback Volume", &ad1986a_laptop_master_vol),
+ HDA_BIND_SW("Master Playback Switch", &ad1986a_laptop_master_sw),
HDA_CODEC_VOLUME("CD Playback Volume", 0x15, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("CD Playback Switch", 0x15, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("Line Playback Volume", 0x17, 0x0, HDA_OUTPUT),
@@ -603,68 +551,114 @@ static struct snd_kcontrol_new ad1986a_laptop_mixers[] = {
/* laptop-eapd model - 2ch only */
-/* master controls both pins 0x1a and 0x1b */
-static int ad1986a_laptop_master_vol_put(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
+static struct hda_input_mux ad1986a_laptop_eapd_capture_source = {
+ .num_items = 3,
+ .items = {
+ { "Mic", 0x0 },
+ { "Internal Mic", 0x4 },
+ { "Mix", 0x5 },
+ },
+};
+
+static struct snd_kcontrol_new ad1986a_laptop_eapd_mixers[] = {
+ HDA_BIND_VOL("Master Playback Volume", &ad1986a_laptop_master_vol),
+ HDA_BIND_SW("Master Playback Switch", &ad1986a_laptop_master_sw),
+ HDA_CODEC_VOLUME("PCM Playback Volume", 0x03, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("PCM Playback Switch", 0x03, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x17, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x17, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x13, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x13, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Mic Boost", 0x0f, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Capture Volume", 0x12, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Capture Switch", 0x12, 0x0, HDA_OUTPUT),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Capture Source",
+ .info = ad198x_mux_enum_info,
+ .get = ad198x_mux_enum_get,
+ .put = ad198x_mux_enum_put,
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "External Amplifier",
+ .info = ad198x_eapd_info,
+ .get = ad198x_eapd_get,
+ .put = ad198x_eapd_put,
+ .private_value = 0x1b | (1 << 8), /* port-D, inversed */
+ },
+ { } /* end */
+};
+
+/* laptop-automute - 2ch only */
+
+static void ad1986a_update_hp(struct hda_codec *codec)
{
- struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
- long *valp = ucontrol->value.integer.value;
- int change;
+ struct ad198x_spec *spec = codec->spec;
+ unsigned int mute;
- change = snd_hda_codec_amp_update(codec, 0x1a, 0, HDA_OUTPUT, 0,
- 0x7f, valp[0] & 0x7f);
- change |= snd_hda_codec_amp_update(codec, 0x1a, 1, HDA_OUTPUT, 0,
- 0x7f, valp[1] & 0x7f);
- snd_hda_codec_amp_update(codec, 0x1b, 0, HDA_OUTPUT, 0,
- 0x7f, valp[0] & 0x7f);
- snd_hda_codec_amp_update(codec, 0x1b, 1, HDA_OUTPUT, 0,
- 0x7f, valp[1] & 0x7f);
- return change;
+ if (spec->jack_present)
+ mute = HDA_AMP_MUTE; /* mute internal speaker */
+ else
+ /* unmute internal speaker if necessary */
+ mute = snd_hda_codec_amp_read(codec, 0x1a, 0, HDA_OUTPUT, 0);
+ snd_hda_codec_amp_stereo(codec, 0x1b, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, mute);
}
-static int ad1986a_laptop_master_sw_put(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
+static void ad1986a_hp_automute(struct hda_codec *codec)
+{
+ struct ad198x_spec *spec = codec->spec;
+ unsigned int present;
+
+ present = snd_hda_codec_read(codec, 0x1a, 0, AC_VERB_GET_PIN_SENSE, 0);
+ spec->jack_present = (present & 0x80000000) != 0;
+ ad1986a_update_hp(codec);
+}
+
+#define AD1986A_HP_EVENT 0x37
+
+static void ad1986a_hp_unsol_event(struct hda_codec *codec, unsigned int res)
+{
+ if ((res >> 26) != AD1986A_HP_EVENT)
+ return;
+ ad1986a_hp_automute(codec);
+}
+
+static int ad1986a_hp_init(struct hda_codec *codec)
+{
+ ad198x_init(codec);
+ ad1986a_hp_automute(codec);
+ return 0;
+}
+
+/* bind hp and internal speaker mute (with plug check) */
+static int ad1986a_hp_master_sw_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
{
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
long *valp = ucontrol->value.integer.value;
int change;
change = snd_hda_codec_amp_update(codec, 0x1a, 0, HDA_OUTPUT, 0,
- 0x80, valp[0] ? 0 : 0x80);
+ HDA_AMP_MUTE,
+ valp[0] ? 0 : HDA_AMP_MUTE);
change |= snd_hda_codec_amp_update(codec, 0x1a, 1, HDA_OUTPUT, 0,
- 0x80, valp[1] ? 0 : 0x80);
- snd_hda_codec_amp_update(codec, 0x1b, 0, HDA_OUTPUT, 0,
- 0x80, valp[0] ? 0 : 0x80);
- snd_hda_codec_amp_update(codec, 0x1b, 1, HDA_OUTPUT, 0,
- 0x80, valp[1] ? 0 : 0x80);
+ HDA_AMP_MUTE,
+ valp[1] ? 0 : HDA_AMP_MUTE);
+ if (change)
+ ad1986a_update_hp(codec);
return change;
}
-static struct hda_input_mux ad1986a_laptop_eapd_capture_source = {
- .num_items = 3,
- .items = {
- { "Mic", 0x0 },
- { "Internal Mic", 0x4 },
- { "Mix", 0x5 },
- },
-};
-
-static struct snd_kcontrol_new ad1986a_laptop_eapd_mixers[] = {
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "Master Playback Volume",
- .info = snd_hda_mixer_amp_volume_info,
- .get = snd_hda_mixer_amp_volume_get,
- .put = ad1986a_laptop_master_vol_put,
- .tlv = { .c = snd_hda_mixer_amp_tlv },
- .private_value = HDA_COMPOSE_AMP_VAL(0x1a, 3, 0, HDA_OUTPUT),
- },
+static struct snd_kcontrol_new ad1986a_laptop_automute_mixers[] = {
+ HDA_BIND_VOL("Master Playback Volume", &ad1986a_laptop_master_vol),
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.name = "Master Playback Switch",
.info = snd_hda_mixer_amp_switch_info,
.get = snd_hda_mixer_amp_switch_get,
- .put = ad1986a_laptop_master_sw_put,
+ .put = ad1986a_hp_master_sw_put,
.private_value = HDA_COMPOSE_AMP_VAL(0x1a, 3, 0, HDA_OUTPUT),
},
HDA_CODEC_VOLUME("PCM Playback Volume", 0x03, 0x0, HDA_OUTPUT),
@@ -674,6 +668,8 @@ static struct snd_kcontrol_new ad1986a_laptop_eapd_mixers[] = {
HDA_CODEC_VOLUME("Mic Playback Volume", 0x13, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x13, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("Mic Boost", 0x0f, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Beep Playback Volume", 0x18, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Beep Playback Switch", 0x18, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("Capture Volume", 0x12, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Capture Switch", 0x12, 0x0, HDA_OUTPUT),
{
@@ -807,12 +803,20 @@ static struct hda_verb ad1986a_ultra_init[] = {
{ } /* end */
};
+/* pin sensing on HP jack */
+static struct hda_verb ad1986a_hp_init_verbs[] = {
+ {0x1a, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1986A_HP_EVENT},
+ {}
+};
+
+
/* models */
enum {
AD1986A_6STACK,
AD1986A_3STACK,
AD1986A_LAPTOP,
AD1986A_LAPTOP_EAPD,
+ AD1986A_LAPTOP_AUTOMUTE,
AD1986A_ULTRA,
AD1986A_MODELS
};
@@ -822,6 +826,7 @@ static const char *ad1986a_models[AD1986A_MODELS] = {
[AD1986A_3STACK] = "3stack",
[AD1986A_LAPTOP] = "laptop",
[AD1986A_LAPTOP_EAPD] = "laptop-eapd",
+ [AD1986A_LAPTOP_AUTOMUTE] = "laptop-automute",
[AD1986A_ULTRA] = "ultra",
};
@@ -850,11 +855,22 @@ static struct snd_pci_quirk ad1986a_cfg_tbl[] = {
SND_PCI_QUIRK(0x144d, 0xc027, "Samsung Q1", AD1986A_ULTRA),
SND_PCI_QUIRK(0x17aa, 0x1011, "Lenovo M55", AD1986A_LAPTOP),
SND_PCI_QUIRK(0x17aa, 0x1017, "Lenovo A60", AD1986A_3STACK),
- SND_PCI_QUIRK(0x17aa, 0x2066, "Lenovo N100", AD1986A_LAPTOP_EAPD),
+ SND_PCI_QUIRK(0x17aa, 0x2066, "Lenovo N100", AD1986A_LAPTOP_AUTOMUTE),
SND_PCI_QUIRK(0x17c0, 0x2017, "Samsung M50", AD1986A_LAPTOP),
{}
};
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+static struct hda_amp_list ad1986a_loopbacks[] = {
+ { 0x13, HDA_OUTPUT, 0 }, /* Mic */
+ { 0x14, HDA_OUTPUT, 0 }, /* Phone */
+ { 0x15, HDA_OUTPUT, 0 }, /* CD */
+ { 0x16, HDA_OUTPUT, 0 }, /* Aux */
+ { 0x17, HDA_OUTPUT, 0 }, /* Line */
+ { } /* end */
+};
+#endif
+
static int patch_ad1986a(struct hda_codec *codec)
{
struct ad198x_spec *spec;
@@ -864,7 +880,6 @@ static int patch_ad1986a(struct hda_codec *codec)
if (spec == NULL)
return -ENOMEM;
- mutex_init(&spec->amp_mutex);
codec->spec = spec;
spec->multiout.max_channels = 6;
@@ -879,6 +894,9 @@ static int patch_ad1986a(struct hda_codec *codec)
spec->mixers[0] = ad1986a_mixers;
spec->num_init_verbs = 1;
spec->init_verbs[0] = ad1986a_init_verbs;
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ spec->loopback.amplist = ad1986a_loopbacks;
+#endif
codec->patch_ops = ad198x_patch_ops;
@@ -914,6 +932,19 @@ static int patch_ad1986a(struct hda_codec *codec)
spec->multiout.dig_out_nid = 0;
spec->input_mux = &ad1986a_laptop_eapd_capture_source;
break;
+ case AD1986A_LAPTOP_AUTOMUTE:
+ spec->mixers[0] = ad1986a_laptop_automute_mixers;
+ spec->num_init_verbs = 3;
+ spec->init_verbs[1] = ad1986a_eapd_init_verbs;
+ spec->init_verbs[2] = ad1986a_hp_init_verbs;
+ spec->multiout.max_channels = 2;
+ spec->multiout.num_dacs = 1;
+ spec->multiout.dac_nids = ad1986a_laptop_dac_nids;
+ spec->multiout.dig_out_nid = 0;
+ spec->input_mux = &ad1986a_laptop_eapd_capture_source;
+ codec->patch_ops.unsol_event = ad1986a_hp_unsol_event;
+ codec->patch_ops.init = ad1986a_hp_init;
+ break;
case AD1986A_ULTRA:
spec->mixers[0] = ad1986a_laptop_eapd_mixers;
spec->num_init_verbs = 2;
@@ -982,8 +1013,9 @@ static int ad1983_spdif_route_put(struct snd_kcontrol *kcontrol, struct snd_ctl_
if (spec->spdif_route != ucontrol->value.enumerated.item[0]) {
spec->spdif_route = ucontrol->value.enumerated.item[0];
- snd_hda_codec_write(codec, spec->multiout.dig_out_nid, 0,
- AC_VERB_SET_CONNECT_SEL, spec->spdif_route);
+ snd_hda_codec_write_cache(codec, spec->multiout.dig_out_nid, 0,
+ AC_VERB_SET_CONNECT_SEL,
+ spec->spdif_route);
return 1;
}
return 0;
@@ -1063,6 +1095,13 @@ static struct hda_verb ad1983_init_verbs[] = {
{ } /* end */
};
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+static struct hda_amp_list ad1983_loopbacks[] = {
+ { 0x12, HDA_OUTPUT, 0 }, /* Mic */
+ { 0x13, HDA_OUTPUT, 0 }, /* Line */
+ { } /* end */
+};
+#endif
static int patch_ad1983(struct hda_codec *codec)
{
@@ -1072,7 +1111,6 @@ static int patch_ad1983(struct hda_codec *codec)
if (spec == NULL)
return -ENOMEM;
- mutex_init(&spec->amp_mutex);
codec->spec = spec;
spec->multiout.max_channels = 2;
@@ -1088,6 +1126,9 @@ static int patch_ad1983(struct hda_codec *codec)
spec->num_init_verbs = 1;
spec->init_verbs[0] = ad1983_init_verbs;
spec->spdif_route = 0;
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ spec->loopback.amplist = ad1983_loopbacks;
+#endif
codec->patch_ops = ad198x_patch_ops;
@@ -1211,6 +1252,17 @@ static struct hda_verb ad1981_init_verbs[] = {
{ } /* end */
};
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+static struct hda_amp_list ad1981_loopbacks[] = {
+ { 0x12, HDA_OUTPUT, 0 }, /* Front Mic */
+ { 0x13, HDA_OUTPUT, 0 }, /* Line */
+ { 0x1b, HDA_OUTPUT, 0 }, /* Aux */
+ { 0x1c, HDA_OUTPUT, 0 }, /* Mic */
+ { 0x1d, HDA_OUTPUT, 0 }, /* CD */
+ { } /* end */
+};
+#endif
+
/*
* Patch for HP nx6320
*
@@ -1240,31 +1292,21 @@ static int ad1981_hp_master_sw_put(struct snd_kcontrol *kcontrol,
return 0;
/* toggle HP mute appropriately */
- snd_hda_codec_amp_update(codec, 0x06, 0, HDA_OUTPUT, 0,
- 0x80, spec->cur_eapd ? 0 : 0x80);
- snd_hda_codec_amp_update(codec, 0x06, 1, HDA_OUTPUT, 0,
- 0x80, spec->cur_eapd ? 0 : 0x80);
+ snd_hda_codec_amp_stereo(codec, 0x06, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE,
+ spec->cur_eapd ? 0 : HDA_AMP_MUTE);
return 1;
}
/* bind volumes of both NID 0x05 and 0x06 */
-static int ad1981_hp_master_vol_put(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
- long *valp = ucontrol->value.integer.value;
- int change;
-
- change = snd_hda_codec_amp_update(codec, 0x05, 0, HDA_OUTPUT, 0,
- 0x7f, valp[0] & 0x7f);
- change |= snd_hda_codec_amp_update(codec, 0x05, 1, HDA_OUTPUT, 0,
- 0x7f, valp[1] & 0x7f);
- snd_hda_codec_amp_update(codec, 0x06, 0, HDA_OUTPUT, 0,
- 0x7f, valp[0] & 0x7f);
- snd_hda_codec_amp_update(codec, 0x06, 1, HDA_OUTPUT, 0,
- 0x7f, valp[1] & 0x7f);
- return change;
-}
+static struct hda_bind_ctls ad1981_hp_bind_master_vol = {
+ .ops = &snd_hda_bind_vol,
+ .values = {
+ HDA_COMPOSE_AMP_VAL(0x05, 3, 0, HDA_OUTPUT),
+ HDA_COMPOSE_AMP_VAL(0x06, 3, 0, HDA_OUTPUT),
+ 0
+ },
+};
/* mute internal speaker if HP is plugged */
static void ad1981_hp_automute(struct hda_codec *codec)
@@ -1273,10 +1315,8 @@ static void ad1981_hp_automute(struct hda_codec *codec)
present = snd_hda_codec_read(codec, 0x06, 0,
AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
- snd_hda_codec_amp_update(codec, 0x05, 0, HDA_OUTPUT, 0,
- 0x80, present ? 0x80 : 0);
- snd_hda_codec_amp_update(codec, 0x05, 1, HDA_OUTPUT, 0,
- 0x80, present ? 0x80 : 0);
+ snd_hda_codec_amp_stereo(codec, 0x05, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
}
/* toggle input of built-in and mic jack appropriately */
@@ -1327,14 +1367,7 @@ static struct hda_input_mux ad1981_hp_capture_source = {
};
static struct snd_kcontrol_new ad1981_hp_mixers[] = {
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "Master Playback Volume",
- .info = snd_hda_mixer_amp_volume_info,
- .get = snd_hda_mixer_amp_volume_get,
- .put = ad1981_hp_master_vol_put,
- .private_value = HDA_COMPOSE_AMP_VAL(0x05, 3, 0, HDA_OUTPUT),
- },
+ HDA_BIND_VOL("Master Playback Volume", &ad1981_hp_bind_master_vol),
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.name = "Master Playback Switch",
@@ -1474,7 +1507,6 @@ static int patch_ad1981(struct hda_codec *codec)
if (spec == NULL)
return -ENOMEM;
- mutex_init(&spec->amp_mutex);
codec->spec = spec;
spec->multiout.max_channels = 2;
@@ -1490,6 +1522,9 @@ static int patch_ad1981(struct hda_codec *codec)
spec->num_init_verbs = 1;
spec->init_verbs[0] = ad1981_init_verbs;
spec->spdif_route = 0;
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ spec->loopback.amplist = ad1981_loopbacks;
+#endif
codec->patch_ops = ad198x_patch_ops;
@@ -1897,16 +1932,19 @@ static int ad1988_spdif_playback_source_get(struct snd_kcontrol *kcontrol,
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
unsigned int sel;
- sel = snd_hda_codec_read(codec, 0x02, 0, AC_VERB_GET_CONNECT_SEL, 0);
- if (sel > 0) {
+ sel = snd_hda_codec_read(codec, 0x1d, 0, AC_VERB_GET_AMP_GAIN_MUTE,
+ AC_AMP_GET_INPUT);
+ if (!(sel & 0x80))
+ ucontrol->value.enumerated.item[0] = 0;
+ else {
sel = snd_hda_codec_read(codec, 0x0b, 0,
AC_VERB_GET_CONNECT_SEL, 0);
if (sel < 3)
sel++;
else
sel = 0;
+ ucontrol->value.enumerated.item[0] = sel;
}
- ucontrol->value.enumerated.item[0] = sel;
return 0;
}
@@ -1918,23 +1956,39 @@ static int ad1988_spdif_playback_source_put(struct snd_kcontrol *kcontrol,
int change;
val = ucontrol->value.enumerated.item[0];
- sel = snd_hda_codec_read(codec, 0x02, 0, AC_VERB_GET_CONNECT_SEL, 0);
if (!val) {
- change = sel != 0;
- if (change || codec->in_resume)
- snd_hda_codec_write(codec, 0x02, 0,
- AC_VERB_SET_CONNECT_SEL, 0);
+ sel = snd_hda_codec_read(codec, 0x1d, 0,
+ AC_VERB_GET_AMP_GAIN_MUTE,
+ AC_AMP_GET_INPUT);
+ change = sel & 0x80;
+ if (change) {
+ snd_hda_codec_write_cache(codec, 0x1d, 0,
+ AC_VERB_SET_AMP_GAIN_MUTE,
+ AMP_IN_UNMUTE(0));
+ snd_hda_codec_write_cache(codec, 0x1d, 0,
+ AC_VERB_SET_AMP_GAIN_MUTE,
+ AMP_IN_MUTE(1));
+ }
} else {
- change = sel == 0;
- if (change || codec->in_resume)
- snd_hda_codec_write(codec, 0x02, 0,
- AC_VERB_SET_CONNECT_SEL, 1);
+ sel = snd_hda_codec_read(codec, 0x1d, 0,
+ AC_VERB_GET_AMP_GAIN_MUTE,
+ AC_AMP_GET_INPUT | 0x01);
+ change = sel & 0x80;
+ if (change) {
+ snd_hda_codec_write_cache(codec, 0x1d, 0,
+ AC_VERB_SET_AMP_GAIN_MUTE,
+ AMP_IN_MUTE(0));
+ snd_hda_codec_write_cache(codec, 0x1d, 0,
+ AC_VERB_SET_AMP_GAIN_MUTE,
+ AMP_IN_UNMUTE(1));
+ }
sel = snd_hda_codec_read(codec, 0x0b, 0,
AC_VERB_GET_CONNECT_SEL, 0) + 1;
change |= sel != val;
- if (change || codec->in_resume)
- snd_hda_codec_write(codec, 0x0b, 0,
- AC_VERB_SET_CONNECT_SEL, val - 1);
+ if (change)
+ snd_hda_codec_write_cache(codec, 0x0b, 0,
+ AC_VERB_SET_CONNECT_SEL,
+ val - 1);
}
return change;
}
@@ -2047,10 +2101,9 @@ static struct hda_verb ad1988_spdif_init_verbs[] = {
{0x02, AC_VERB_SET_CONNECT_SEL, 0x0}, /* PCM */
{0x0b, AC_VERB_SET_CONNECT_SEL, 0x0}, /* ADC1 */
{0x1d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x1d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x1d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
/* SPDIF out pin */
{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x27}, /* 0dB */
- {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x17}, /* 0dB */
{ }
};
@@ -2225,6 +2278,15 @@ static void ad1988_laptop_unsol_event(struct hda_codec *codec, unsigned int res)
snd_hda_sequence_write(codec, ad1988_laptop_hp_off);
}
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+static struct hda_amp_list ad1988_loopbacks[] = {
+ { 0x20, HDA_INPUT, 0 }, /* Front Mic */
+ { 0x20, HDA_INPUT, 1 }, /* Line */
+ { 0x20, HDA_INPUT, 4 }, /* Mic */
+ { 0x20, HDA_INPUT, 6 }, /* CD */
+ { } /* end */
+};
+#endif
/*
* Automatic parse of I/O pins from the BIOS configuration
@@ -2663,7 +2725,6 @@ static int patch_ad1988(struct hda_codec *codec)
if (spec == NULL)
return -ENOMEM;
- mutex_init(&spec->amp_mutex);
codec->spec = spec;
if (is_rev2(codec))
@@ -2770,6 +2831,9 @@ static int patch_ad1988(struct hda_codec *codec)
codec->patch_ops.unsol_event = ad1988_laptop_unsol_event;
break;
}
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ spec->loopback.amplist = ad1988_loopbacks;
+#endif
return 0;
}
@@ -2926,6 +2990,16 @@ static struct hda_verb ad1884_init_verbs[] = {
{ } /* end */
};
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+static struct hda_amp_list ad1884_loopbacks[] = {
+ { 0x20, HDA_INPUT, 0 }, /* Front Mic */
+ { 0x20, HDA_INPUT, 1 }, /* Mic */
+ { 0x20, HDA_INPUT, 2 }, /* CD */
+ { 0x20, HDA_INPUT, 4 }, /* Docking */
+ { } /* end */
+};
+#endif
+
static int patch_ad1884(struct hda_codec *codec)
{
struct ad198x_spec *spec;
@@ -2950,6 +3024,9 @@ static int patch_ad1884(struct hda_codec *codec)
spec->num_init_verbs = 1;
spec->init_verbs[0] = ad1884_init_verbs;
spec->spdif_route = 0;
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ spec->loopback.amplist = ad1884_loopbacks;
+#endif
codec->patch_ops = ad198x_patch_ops;
@@ -3331,6 +3408,16 @@ static struct hda_verb ad1882_init_verbs[] = {
{ } /* end */
};
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+static struct hda_amp_list ad1882_loopbacks[] = {
+ { 0x20, HDA_INPUT, 0 }, /* Front Mic */
+ { 0x20, HDA_INPUT, 1 }, /* Mic */
+ { 0x20, HDA_INPUT, 4 }, /* Line */
+ { 0x20, HDA_INPUT, 6 }, /* CD */
+ { } /* end */
+};
+#endif
+
/* models */
enum {
AD1882_3STACK,
@@ -3369,6 +3456,9 @@ static int patch_ad1882(struct hda_codec *codec)
spec->num_init_verbs = 1;
spec->init_verbs[0] = ad1882_init_verbs;
spec->spdif_route = 0;
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ spec->loopback.amplist = ad1882_loopbacks;
+#endif
codec->patch_ops = ad198x_patch_ops;
diff --git a/sound/pci/hda/patch_atihdmi.c b/sound/pci/hda/patch_atihdmi.c
index 72d3ab9751ac..fbb8969dc559 100644
--- a/sound/pci/hda/patch_atihdmi.c
+++ b/sound/pci/hda/patch_atihdmi.c
@@ -62,19 +62,6 @@ static int atihdmi_init(struct hda_codec *codec)
return 0;
}
-#ifdef CONFIG_PM
-/*
- * resume
- */
-static int atihdmi_resume(struct hda_codec *codec)
-{
- atihdmi_init(codec);
- snd_hda_resume_spdif_out(codec);
-
- return 0;
-}
-#endif
-
/*
* Digital out
*/
@@ -141,9 +128,6 @@ static struct hda_codec_ops atihdmi_patch_ops = {
.build_pcms = atihdmi_build_pcms,
.init = atihdmi_init,
.free = atihdmi_free,
-#ifdef CONFIG_PM
- .resume = atihdmi_resume,
-#endif
};
static int patch_atihdmi(struct hda_codec *codec)
diff --git a/sound/pci/hda/patch_cmedia.c b/sound/pci/hda/patch_cmedia.c
index 3c722e667bc8..2468f3171222 100644
--- a/sound/pci/hda/patch_cmedia.c
+++ b/sound/pci/hda/patch_cmedia.c
@@ -427,27 +427,6 @@ static int cmi9880_init(struct hda_codec *codec)
return 0;
}
-#ifdef CONFIG_PM
-/*
- * resume
- */
-static int cmi9880_resume(struct hda_codec *codec)
-{
- struct cmi_spec *spec = codec->spec;
-
- cmi9880_init(codec);
- snd_hda_resume_ctls(codec, cmi9880_basic_mixer);
- if (spec->channel_modes)
- snd_hda_resume_ctls(codec, cmi9880_ch_mode_mixer);
- if (spec->multiout.dig_out_nid)
- snd_hda_resume_spdif_out(codec);
- if (spec->dig_in_nid)
- snd_hda_resume_spdif_in(codec);
-
- return 0;
-}
-#endif
-
/*
* Analog playback callbacks
*/
@@ -635,9 +614,6 @@ static struct hda_codec_ops cmi9880_patch_ops = {
.build_pcms = cmi9880_build_pcms,
.init = cmi9880_init,
.free = cmi9880_free,
-#ifdef CONFIG_PM
- .resume = cmi9880_resume,
-#endif
};
static int patch_cmi9880(struct hda_codec *codec)
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 4d8e8af5c819..080e3001d9c5 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -311,23 +311,6 @@ static void conexant_free(struct hda_codec *codec)
kfree(codec->spec);
}
-#ifdef CONFIG_PM
-static int conexant_resume(struct hda_codec *codec)
-{
- struct conexant_spec *spec = codec->spec;
- int i;
-
- codec->patch_ops.init(codec);
- for (i = 0; i < spec->num_mixers; i++)
- snd_hda_resume_ctls(codec, spec->mixers[i]);
- if (spec->multiout.dig_out_nid)
- snd_hda_resume_spdif_out(codec);
- if (spec->dig_in_nid)
- snd_hda_resume_spdif_in(codec);
- return 0;
-}
-#endif
-
static int conexant_build_controls(struct hda_codec *codec)
{
struct conexant_spec *spec = codec->spec;
@@ -358,9 +341,6 @@ static struct hda_codec_ops conexant_patch_ops = {
.build_pcms = conexant_build_pcms,
.init = conexant_init,
.free = conexant_free,
-#ifdef CONFIG_PM
- .resume = conexant_resume,
-#endif
};
/*
@@ -368,15 +348,7 @@ static struct hda_codec_ops conexant_patch_ops = {
* the private value = nid | (invert << 8)
*/
-static int cxt_eapd_info(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_info *uinfo)
-{
- uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
- uinfo->count = 1;
- uinfo->value.integer.min = 0;
- uinfo->value.integer.max = 1;
- return 0;
-}
+#define cxt_eapd_info snd_ctl_boolean_mono_info
static int cxt_eapd_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
@@ -404,13 +376,13 @@ static int cxt_eapd_put(struct snd_kcontrol *kcontrol,
eapd = ucontrol->value.integer.value[0];
if (invert)
eapd = !eapd;
- if (eapd == spec->cur_eapd && !codec->in_resume)
+ if (eapd == spec->cur_eapd)
return 0;
spec->cur_eapd = eapd;
- snd_hda_codec_write(codec, nid,
- 0, AC_VERB_SET_EAPD_BTLENABLE,
- eapd ? 0x02 : 0x00);
+ snd_hda_codec_write_cache(codec, nid,
+ 0, AC_VERB_SET_EAPD_BTLENABLE,
+ eapd ? 0x02 : 0x00);
return 1;
}
@@ -500,34 +472,25 @@ static int cxt5045_hp_master_sw_put(struct snd_kcontrol *kcontrol,
/* toggle internal speakers mute depending of presence of
* the headphone jack
*/
- bits = (!spec->hp_present && spec->cur_eapd) ? 0 : 0x80;
- snd_hda_codec_amp_update(codec, 0x10, 0, HDA_OUTPUT, 0, 0x80, bits);
- snd_hda_codec_amp_update(codec, 0x10, 1, HDA_OUTPUT, 0, 0x80, bits);
+ bits = (!spec->hp_present && spec->cur_eapd) ? 0 : HDA_AMP_MUTE;
+ snd_hda_codec_amp_stereo(codec, 0x10, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, bits);
- bits = spec->cur_eapd ? 0 : 0x80;
- snd_hda_codec_amp_update(codec, 0x11, 0, HDA_OUTPUT, 0, 0x80, bits);
- snd_hda_codec_amp_update(codec, 0x11, 1, HDA_OUTPUT, 0, 0x80, bits);
+ bits = spec->cur_eapd ? 0 : HDA_AMP_MUTE;
+ snd_hda_codec_amp_stereo(codec, 0x11, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, bits);
return 1;
}
/* bind volumes of both NID 0x10 and 0x11 */
-static int cxt5045_hp_master_vol_put(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
- long *valp = ucontrol->value.integer.value;
- int change;
-
- change = snd_hda_codec_amp_update(codec, 0x10, 0, HDA_OUTPUT, 0,
- 0x7f, valp[0] & 0x7f);
- change |= snd_hda_codec_amp_update(codec, 0x10, 1, HDA_OUTPUT, 0,
- 0x7f, valp[1] & 0x7f);
- snd_hda_codec_amp_update(codec, 0x11, 0, HDA_OUTPUT, 0,
- 0x7f, valp[0] & 0x7f);
- snd_hda_codec_amp_update(codec, 0x11, 1, HDA_OUTPUT, 0,
- 0x7f, valp[1] & 0x7f);
- return change;
-}
+static struct hda_bind_ctls cxt5045_hp_bind_master_vol = {
+ .ops = &snd_hda_bind_vol,
+ .values = {
+ HDA_COMPOSE_AMP_VAL(0x10, 3, 0, HDA_OUTPUT),
+ HDA_COMPOSE_AMP_VAL(0x11, 3, 0, HDA_OUTPUT),
+ 0
+ },
+};
/* toggle input of built-in and mic jack appropriately */
static void cxt5045_hp_automic(struct hda_codec *codec)
@@ -562,9 +525,9 @@ static void cxt5045_hp_automute(struct hda_codec *codec)
spec->hp_present = snd_hda_codec_read(codec, 0x11, 0,
AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
- bits = (spec->hp_present || !spec->cur_eapd) ? 0x80 : 0;
- snd_hda_codec_amp_update(codec, 0x10, 0, HDA_OUTPUT, 0, 0x80, bits);
- snd_hda_codec_amp_update(codec, 0x10, 1, HDA_OUTPUT, 0, 0x80, bits);
+ bits = (spec->hp_present || !spec->cur_eapd) ? HDA_AMP_MUTE : 0;
+ snd_hda_codec_amp_stereo(codec, 0x10, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, bits);
}
/* unsolicited event for HP jack sensing */
@@ -595,14 +558,7 @@ static struct snd_kcontrol_new cxt5045_mixers[] = {
HDA_CODEC_MUTE("Int Mic Switch", 0x1a, 0x01, HDA_INPUT),
HDA_CODEC_VOLUME("Ext Mic Volume", 0x1a, 0x02, HDA_INPUT),
HDA_CODEC_MUTE("Ext Mic Switch", 0x1a, 0x02, HDA_INPUT),
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "Master Playback Volume",
- .info = snd_hda_mixer_amp_volume_info,
- .get = snd_hda_mixer_amp_volume_get,
- .put = cxt5045_hp_master_vol_put,
- .private_value = HDA_COMPOSE_AMP_VAL(0x10, 3, 0, HDA_OUTPUT),
- },
+ HDA_BIND_VOL("Master Playback Volume", &cxt5045_hp_bind_master_vol),
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.name = "Master Playback Switch",
@@ -915,33 +871,24 @@ static int cxt5047_hp_master_sw_put(struct snd_kcontrol *kcontrol,
/* toggle internal speakers mute depending of presence of
* the headphone jack
*/
- bits = (!spec->hp_present && spec->cur_eapd) ? 0 : 0x80;
- snd_hda_codec_amp_update(codec, 0x1d, 0, HDA_OUTPUT, 0, 0x80, bits);
- snd_hda_codec_amp_update(codec, 0x1d, 1, HDA_OUTPUT, 0, 0x80, bits);
- bits = spec->cur_eapd ? 0 : 0x80;
- snd_hda_codec_amp_update(codec, 0x13, 0, HDA_OUTPUT, 0, 0x80, bits);
- snd_hda_codec_amp_update(codec, 0x13, 1, HDA_OUTPUT, 0, 0x80, bits);
+ bits = (!spec->hp_present && spec->cur_eapd) ? 0 : HDA_AMP_MUTE;
+ snd_hda_codec_amp_stereo(codec, 0x1d, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, bits);
+ bits = spec->cur_eapd ? 0 : HDA_AMP_MUTE;
+ snd_hda_codec_amp_stereo(codec, 0x13, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, bits);
return 1;
}
/* bind volumes of both NID 0x13 (Headphones) and 0x1d (Speakers) */
-static int cxt5047_hp_master_vol_put(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
- long *valp = ucontrol->value.integer.value;
- int change;
-
- change = snd_hda_codec_amp_update(codec, 0x1d, 0, HDA_OUTPUT, 0,
- 0x7f, valp[0] & 0x7f);
- change |= snd_hda_codec_amp_update(codec, 0x1d, 1, HDA_OUTPUT, 0,
- 0x7f, valp[1] & 0x7f);
- snd_hda_codec_amp_update(codec, 0x13, 0, HDA_OUTPUT, 0,
- 0x7f, valp[0] & 0x7f);
- snd_hda_codec_amp_update(codec, 0x13, 1, HDA_OUTPUT, 0,
- 0x7f, valp[1] & 0x7f);
- return change;
-}
+static struct hda_bind_ctls cxt5047_bind_master_vol = {
+ .ops = &snd_hda_bind_vol,
+ .values = {
+ HDA_COMPOSE_AMP_VAL(0x13, 3, 0, HDA_OUTPUT),
+ HDA_COMPOSE_AMP_VAL(0x1d, 3, 0, HDA_OUTPUT),
+ 0
+ },
+};
/* mute internal speaker if HP is plugged */
static void cxt5047_hp_automute(struct hda_codec *codec)
@@ -952,12 +899,12 @@ static void cxt5047_hp_automute(struct hda_codec *codec)
spec->hp_present = snd_hda_codec_read(codec, 0x13, 0,
AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
- bits = (spec->hp_present || !spec->cur_eapd) ? 0x80 : 0;
- snd_hda_codec_amp_update(codec, 0x1d, 0, HDA_OUTPUT, 0, 0x80, bits);
- snd_hda_codec_amp_update(codec, 0x1d, 1, HDA_OUTPUT, 0, 0x80, bits);
+ bits = (spec->hp_present || !spec->cur_eapd) ? HDA_AMP_MUTE : 0;
+ snd_hda_codec_amp_stereo(codec, 0x1d, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, bits);
/* Mute/Unmute PCM 2 for good measure - some systems need this */
- snd_hda_codec_amp_update(codec, 0x1c, 0, HDA_OUTPUT, 0, 0x80, bits);
- snd_hda_codec_amp_update(codec, 0x1c, 1, HDA_OUTPUT, 0, 0x80, bits);
+ snd_hda_codec_amp_stereo(codec, 0x1c, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, bits);
}
/* mute internal speaker if HP is plugged */
@@ -969,12 +916,12 @@ static void cxt5047_hp2_automute(struct hda_codec *codec)
spec->hp_present = snd_hda_codec_read(codec, 0x13, 0,
AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
- bits = spec->hp_present ? 0x80 : 0;
- snd_hda_codec_amp_update(codec, 0x1d, 0, HDA_OUTPUT, 0, 0x80, bits);
- snd_hda_codec_amp_update(codec, 0x1d, 1, HDA_OUTPUT, 0, 0x80, bits);
+ bits = spec->hp_present ? HDA_AMP_MUTE : 0;
+ snd_hda_codec_amp_stereo(codec, 0x1d, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, bits);
/* Mute/Unmute PCM 2 for good measure - some systems need this */
- snd_hda_codec_amp_update(codec, 0x1c, 0, HDA_OUTPUT, 0, 0x80, bits);
- snd_hda_codec_amp_update(codec, 0x1c, 1, HDA_OUTPUT, 0, 0x80, bits);
+ snd_hda_codec_amp_stereo(codec, 0x1c, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, bits);
}
/* toggle input of built-in and mic jack appropriately */
@@ -1063,14 +1010,7 @@ static struct snd_kcontrol_new cxt5047_toshiba_mixers[] = {
HDA_CODEC_MUTE("Capture Switch", 0x12, 0x03, HDA_INPUT),
HDA_CODEC_VOLUME("PCM Volume", 0x10, 0x00, HDA_OUTPUT),
HDA_CODEC_MUTE("PCM Switch", 0x10, 0x00, HDA_OUTPUT),
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "Master Playback Volume",
- .info = snd_hda_mixer_amp_volume_info,
- .get = snd_hda_mixer_amp_volume_get,
- .put = cxt5047_hp_master_vol_put,
- .private_value = HDA_COMPOSE_AMP_VAL(0x13, 3, 0, HDA_OUTPUT),
- },
+ HDA_BIND_VOL("Master Playback Volume", &cxt5047_bind_master_vol),
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.name = "Master Playback Switch",
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 9a47eec5a27b..53b0428abfc2 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -102,6 +102,8 @@ enum {
/* ALC268 models */
enum {
ALC268_3ST,
+ ALC268_TOSHIBA,
+ ALC268_ACER,
ALC268_AUTO,
ALC268_MODEL_LAST /* last tag */
};
@@ -129,6 +131,7 @@ enum {
ALC861VD_6ST_DIG,
ALC861VD_LENOVO,
ALC861VD_DALLAS,
+ ALC861VD_HP,
ALC861VD_AUTO,
ALC861VD_MODEL_LAST,
};
@@ -140,6 +143,7 @@ enum {
ALC662_3ST_6ch,
ALC662_5ST_DIG,
ALC662_LENOVO_101E,
+ ALC662_ASUS_EEEPC_P701,
ALC662_AUTO,
ALC662_MODEL_LAST,
};
@@ -152,7 +156,9 @@ enum {
ALC882_W2JC,
ALC882_TARGA,
ALC882_ASUS_A7J,
+ ALC882_ASUS_A7M,
ALC885_MACPRO,
+ ALC885_MBP3,
ALC885_IMAC24,
ALC882_AUTO,
ALC882_MODEL_LAST,
@@ -167,12 +173,14 @@ enum {
ALC883_TARGA_DIG,
ALC883_TARGA_2ch_DIG,
ALC883_ACER,
+ ALC883_ACER_ASPIRE,
ALC883_MEDION,
ALC883_MEDION_MD2,
ALC883_LAPTOP_EAPD,
ALC883_LENOVO_101E_2ch,
ALC883_LENOVO_NB0763,
- ALC888_LENOVO_MS7195_DIG,
+ ALC888_LENOVO_MS7195_DIG,
+ ALC883_HAIER_W66,
ALC888_6ST_HP,
ALC888_3ST_HP,
ALC883_AUTO,
@@ -239,6 +247,10 @@ struct alc_spec {
/* for pin sensing */
unsigned int sense_updated: 1;
unsigned int jack_present: 1;
+
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ struct hda_loopback_check loopback;
+#endif
};
/*
@@ -263,6 +275,9 @@ struct alc_config_preset {
const struct hda_input_mux *input_mux;
void (*unsol_event)(struct hda_codec *, unsigned int);
void (*init_hook)(struct hda_codec *);
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ struct hda_amp_list *loopbacks;
+#endif
};
@@ -441,8 +456,9 @@ static int alc_pin_mode_put(struct snd_kcontrol *kcontrol,
change = pinctl != alc_pin_mode_values[val];
if (change) {
/* Set pin mode to that requested */
- snd_hda_codec_write(codec,nid,0,AC_VERB_SET_PIN_WIDGET_CONTROL,
- alc_pin_mode_values[val]);
+ snd_hda_codec_write_cache(codec, nid, 0,
+ AC_VERB_SET_PIN_WIDGET_CONTROL,
+ alc_pin_mode_values[val]);
/* Also enable the retasking pin's input/output as required
* for the requested pin mode. Enum values of 2 or less are
@@ -455,19 +471,15 @@ static int alc_pin_mode_put(struct snd_kcontrol *kcontrol,
* this turns out to be necessary in the future.
*/
if (val <= 2) {
- snd_hda_codec_write(codec, nid, 0,
- AC_VERB_SET_AMP_GAIN_MUTE,
- AMP_OUT_MUTE);
- snd_hda_codec_write(codec, nid, 0,
- AC_VERB_SET_AMP_GAIN_MUTE,
- AMP_IN_UNMUTE(0));
+ snd_hda_codec_amp_stereo(codec, nid, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, HDA_AMP_MUTE);
+ snd_hda_codec_amp_stereo(codec, nid, HDA_INPUT, 0,
+ HDA_AMP_MUTE, 0);
} else {
- snd_hda_codec_write(codec, nid, 0,
- AC_VERB_SET_AMP_GAIN_MUTE,
- AMP_IN_MUTE(0));
- snd_hda_codec_write(codec, nid, 0,
- AC_VERB_SET_AMP_GAIN_MUTE,
- AMP_OUT_UNMUTE);
+ snd_hda_codec_amp_stereo(codec, nid, HDA_INPUT, 0,
+ HDA_AMP_MUTE, HDA_AMP_MUTE);
+ snd_hda_codec_amp_stereo(codec, nid, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, 0);
}
}
return change;
@@ -486,15 +498,7 @@ static int alc_pin_mode_put(struct snd_kcontrol *kcontrol,
* needed for any "production" models.
*/
#ifdef CONFIG_SND_DEBUG
-static int alc_gpio_data_info(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_info *uinfo)
-{
- uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
- uinfo->count = 1;
- uinfo->value.integer.min = 0;
- uinfo->value.integer.max = 1;
- return 0;
-}
+#define alc_gpio_data_info snd_ctl_boolean_mono_info
static int alc_gpio_data_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
@@ -527,7 +531,8 @@ static int alc_gpio_data_put(struct snd_kcontrol *kcontrol,
gpio_data &= ~mask;
else
gpio_data |= mask;
- snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_GPIO_DATA, gpio_data);
+ snd_hda_codec_write_cache(codec, nid, 0,
+ AC_VERB_SET_GPIO_DATA, gpio_data);
return change;
}
@@ -547,15 +552,7 @@ static int alc_gpio_data_put(struct snd_kcontrol *kcontrol,
* necessary.
*/
#ifdef CONFIG_SND_DEBUG
-static int alc_spdif_ctrl_info(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_info *uinfo)
-{
- uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
- uinfo->count = 1;
- uinfo->value.integer.min = 0;
- uinfo->value.integer.max = 1;
- return 0;
-}
+#define alc_spdif_ctrl_info snd_ctl_boolean_mono_info
static int alc_spdif_ctrl_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
@@ -588,8 +585,8 @@ static int alc_spdif_ctrl_put(struct snd_kcontrol *kcontrol,
ctrl_data &= ~mask;
else
ctrl_data |= mask;
- snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_DIGI_CONVERT_1,
- ctrl_data);
+ snd_hda_codec_write_cache(codec, nid, 0, AC_VERB_SET_DIGI_CONVERT_1,
+ ctrl_data);
return change;
}
@@ -638,6 +635,9 @@ static void setup_preset(struct alc_spec *spec,
spec->unsol_event = preset->unsol_event;
spec->init_hook = preset->init_hook;
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ spec->loopback.amplist = preset->loopbacks;
+#endif
}
/* Enable GPIO mask and set output */
@@ -662,6 +662,44 @@ static struct hda_verb alc_gpio3_init_verbs[] = {
{ }
};
+static void alc_sku_automute(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ unsigned int mute;
+ unsigned int present;
+ unsigned int hp_nid = spec->autocfg.hp_pins[0];
+ unsigned int sp_nid = spec->autocfg.speaker_pins[0];
+
+ /* need to execute and sync at first */
+ snd_hda_codec_read(codec, hp_nid, 0, AC_VERB_SET_PIN_SENSE, 0);
+ present = snd_hda_codec_read(codec, hp_nid, 0,
+ AC_VERB_GET_PIN_SENSE, 0);
+ spec->jack_present = (present & 0x80000000) != 0;
+ if (spec->jack_present) {
+ /* mute internal speaker */
+ snd_hda_codec_amp_stereo(codec, sp_nid, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, HDA_AMP_MUTE);
+ } else {
+ /* unmute internal speaker if necessary */
+ mute = snd_hda_codec_amp_read(codec, hp_nid, 0, HDA_OUTPUT, 0);
+ snd_hda_codec_amp_stereo(codec, sp_nid, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, mute);
+ }
+}
+
+/* unsolicited event for HP jack sensing */
+static void alc_sku_unsol_event(struct hda_codec *codec, unsigned int res)
+{
+ if (codec->vendor_id == 0x10ec0880)
+ res >>= 28;
+ else
+ res >>= 26;
+ if (res != ALC880_HP_EVENT)
+ return;
+
+ alc_sku_automute(codec);
+}
+
/* 32-bit subsystem ID for BIOS loading in HD Audio codec.
* 31 ~ 16 : Manufacture ID
* 15 ~ 8 : SKU ID
@@ -672,13 +710,48 @@ static void alc_subsystem_id(struct hda_codec *codec,
unsigned int porta, unsigned int porte,
unsigned int portd)
{
- unsigned int ass, tmp;
+ unsigned int ass, tmp, i;
+ unsigned nid;
+ struct alc_spec *spec = codec->spec;
- ass = codec->subsystem_id;
- if (!(ass & 1))
+ ass = codec->subsystem_id & 0xffff;
+ if ((ass != codec->bus->pci->subsystem_device) && (ass & 1))
+ goto do_sku;
+
+ /*
+ * 31~30 : port conetcivity
+ * 29~21 : reserve
+ * 20 : PCBEEP input
+ * 19~16 : Check sum (15:1)
+ * 15~1 : Custom
+ * 0 : override
+ */
+ nid = 0x1d;
+ if (codec->vendor_id == 0x10ec0260)
+ nid = 0x17;
+ ass = snd_hda_codec_read(codec, nid, 0,
+ AC_VERB_GET_CONFIG_DEFAULT, 0);
+ if (!(ass & 1) && !(ass & 0x100000))
+ return;
+ if ((ass >> 30) != 1) /* no physical connection */
return;
- /* Override */
+ /* check sum */
+ tmp = 0;
+ for (i = 1; i < 16; i++) {
+ if ((ass >> i) && 1)
+ tmp++;
+ }
+ if (((ass >> 16) & 0xf) != tmp)
+ return;
+do_sku:
+ /*
+ * 0 : override
+ * 1 : Swap Jack
+ * 2 : 0 --> Desktop, 1 --> Laptop
+ * 3~5 : External Amplifier control
+ * 7~6 : Reserved
+ */
tmp = (ass & 0x38) >> 3; /* external Amp control */
switch (tmp) {
case 1:
@@ -690,38 +763,108 @@ static void alc_subsystem_id(struct hda_codec *codec,
case 7:
snd_hda_sequence_write(codec, alc_gpio3_init_verbs);
break;
- case 5:
+ case 5: /* set EAPD output high */
switch (codec->vendor_id) {
- case 0x10ec0862:
- case 0x10ec0660:
- case 0x10ec0662:
+ case 0x10ec0260:
+ snd_hda_codec_write(codec, 0x0f, 0,
+ AC_VERB_SET_EAPD_BTLENABLE, 2);
+ snd_hda_codec_write(codec, 0x10, 0,
+ AC_VERB_SET_EAPD_BTLENABLE, 2);
+ break;
+ case 0x10ec0262:
case 0x10ec0267:
case 0x10ec0268:
+ case 0x10ec0269:
+ case 0x10ec0862:
+ case 0x10ec0662:
snd_hda_codec_write(codec, 0x14, 0,
AC_VERB_SET_EAPD_BTLENABLE, 2);
snd_hda_codec_write(codec, 0x15, 0,
AC_VERB_SET_EAPD_BTLENABLE, 2);
- return;
+ break;
}
- case 6:
- if (ass & 4) { /* bit 2 : 0 = Desktop, 1 = Laptop */
- hda_nid_t port = 0;
- tmp = (ass & 0x1800) >> 11;
- switch (tmp) {
- case 0: port = porta; break;
- case 1: port = porte; break;
- case 2: port = portd; break;
- }
- if (port)
- snd_hda_codec_write(codec, port, 0,
- AC_VERB_SET_EAPD_BTLENABLE,
- 2);
+ switch (codec->vendor_id) {
+ case 0x10ec0260:
+ snd_hda_codec_write(codec, 0x1a, 0,
+ AC_VERB_SET_COEF_INDEX, 7);
+ tmp = snd_hda_codec_read(codec, 0x1a, 0,
+ AC_VERB_GET_PROC_COEF, 0);
+ snd_hda_codec_write(codec, 0x1a, 0,
+ AC_VERB_SET_COEF_INDEX, 7);
+ snd_hda_codec_write(codec, 0x1a, 0,
+ AC_VERB_SET_PROC_COEF,
+ tmp | 0x2010);
+ break;
+ case 0x10ec0262:
+ case 0x10ec0880:
+ case 0x10ec0882:
+ case 0x10ec0883:
+ case 0x10ec0885:
+ case 0x10ec0888:
+ snd_hda_codec_write(codec, 0x20, 0,
+ AC_VERB_SET_COEF_INDEX, 7);
+ tmp = snd_hda_codec_read(codec, 0x20, 0,
+ AC_VERB_GET_PROC_COEF, 0);
+ snd_hda_codec_write(codec, 0x20, 0,
+ AC_VERB_SET_COEF_INDEX, 7);
+ snd_hda_codec_write(codec, 0x20, 0,
+ AC_VERB_SET_PROC_COEF,
+ tmp | 0x2010);
+ break;
+ case 0x10ec0267:
+ case 0x10ec0268:
+ snd_hda_codec_write(codec, 0x20, 0,
+ AC_VERB_SET_COEF_INDEX, 7);
+ tmp = snd_hda_codec_read(codec, 0x20, 0,
+ AC_VERB_GET_PROC_COEF, 0);
+ snd_hda_codec_write(codec, 0x20, 0,
+ AC_VERB_SET_COEF_INDEX, 7);
+ snd_hda_codec_write(codec, 0x20, 0,
+ AC_VERB_SET_PROC_COEF,
+ tmp | 0x3000);
+ break;
}
- snd_hda_codec_write(codec, 0x1a, 0, AC_VERB_SET_COEF_INDEX, 7);
- snd_hda_codec_write(codec, 0x1a, 0, AC_VERB_SET_PROC_COEF,
- (tmp == 5 ? 0x3040 : 0x3050));
+ default:
break;
}
+
+ /* is laptop and enable the function "Mute internal speaker
+ * when the external headphone out jack is plugged"
+ */
+ if (!(ass & 0x4) || !(ass & 0x8000))
+ return;
+ /*
+ * 10~8 : Jack location
+ * 12~11: Headphone out -> 00: PortA, 01: PortE, 02: PortD, 03: Resvered
+ * 14~13: Resvered
+ * 15 : 1 --> enable the function "Mute internal speaker
+ * when the external headphone out jack is plugged"
+ */
+ if (!spec->autocfg.speaker_pins[0]) {
+ if (spec->multiout.dac_nids[0])
+ spec->autocfg.speaker_pins[0] =
+ spec->multiout.dac_nids[0];
+ else
+ return;
+ }
+
+ if (!spec->autocfg.hp_pins[0]) {
+ tmp = (ass >> 11) & 0x3; /* HP to chassis */
+ if (tmp == 0)
+ spec->autocfg.hp_pins[0] = porta;
+ else if (tmp == 1)
+ spec->autocfg.hp_pins[0] = porte;
+ else if (tmp == 2)
+ spec->autocfg.hp_pins[0] = portd;
+ else
+ return;
+ }
+
+ snd_hda_codec_write(codec, spec->autocfg.hp_pins[0], 0,
+ AC_VERB_SET_UNSOLICITED_ENABLE,
+ AC_USRSP_EN | ALC880_HP_EVENT);
+ spec->unsol_event = alc_sku_unsol_event;
+ spec->init_hook = alc_sku_automute;
}
/*
@@ -1304,11 +1447,13 @@ static struct hda_verb alc880_volume_init_verbs[] = {
* panel mic (mic 2)
*/
/* Amp Indices: Mic1 = 0, Mic2 = 1, Line1 = 2, Line2 = 3, CD = 4 */
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)},
/*
* Set up output mixers (0x0c - 0x0f)
@@ -1568,15 +1713,11 @@ static void alc880_uniwill_hp_automute(struct hda_codec *codec)
present = snd_hda_codec_read(codec, 0x14, 0,
AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
- bits = present ? 0x80 : 0;
- snd_hda_codec_amp_update(codec, 0x15, 0, HDA_OUTPUT, 0,
- 0x80, bits);
- snd_hda_codec_amp_update(codec, 0x15, 1, HDA_OUTPUT, 0,
- 0x80, bits);
- snd_hda_codec_amp_update(codec, 0x16, 0, HDA_OUTPUT, 0,
- 0x80, bits);
- snd_hda_codec_amp_update(codec, 0x16, 1, HDA_OUTPUT, 0,
- 0x80, bits);
+ bits = present ? HDA_AMP_MUTE : 0;
+ snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, bits);
+ snd_hda_codec_amp_stereo(codec, 0x16, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, bits);
}
/* auto-toggle front mic */
@@ -1587,11 +1728,8 @@ static void alc880_uniwill_mic_automute(struct hda_codec *codec)
present = snd_hda_codec_read(codec, 0x18, 0,
AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
- bits = present ? 0x80 : 0;
- snd_hda_codec_amp_update(codec, 0x0b, 0, HDA_INPUT, 1,
- 0x80, bits);
- snd_hda_codec_amp_update(codec, 0x0b, 1, HDA_INPUT, 1,
- 0x80, bits);
+ bits = present ? HDA_AMP_MUTE : 0;
+ snd_hda_codec_amp_stereo(codec, 0x0b, HDA_INPUT, 1, HDA_AMP_MUTE, bits);
}
static void alc880_uniwill_automute(struct hda_codec *codec)
@@ -1623,11 +1761,8 @@ static void alc880_uniwill_p53_hp_automute(struct hda_codec *codec)
present = snd_hda_codec_read(codec, 0x14, 0,
AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
- bits = present ? 0x80 : 0;
- snd_hda_codec_amp_update(codec, 0x15, 0, HDA_INPUT, 0,
- 0x80, bits);
- snd_hda_codec_amp_update(codec, 0x15, 1, HDA_INPUT, 0,
- 0x80, bits);
+ bits = present ? HDA_AMP_MUTE : 0;
+ snd_hda_codec_amp_stereo(codec, 0x15, HDA_INPUT, 0, HDA_AMP_MUTE, bits);
}
static void alc880_uniwill_p53_dcvol_automute(struct hda_codec *codec)
@@ -1635,19 +1770,14 @@ static void alc880_uniwill_p53_dcvol_automute(struct hda_codec *codec)
unsigned int present;
present = snd_hda_codec_read(codec, 0x21, 0,
- AC_VERB_GET_VOLUME_KNOB_CONTROL, 0) & 0x7f;
-
- snd_hda_codec_amp_update(codec, 0x0c, 0, HDA_OUTPUT, 0,
- 0x7f, present);
- snd_hda_codec_amp_update(codec, 0x0c, 1, HDA_OUTPUT, 0,
- 0x7f, present);
-
- snd_hda_codec_amp_update(codec, 0x0d, 0, HDA_OUTPUT, 0,
- 0x7f, present);
- snd_hda_codec_amp_update(codec, 0x0d, 1, HDA_OUTPUT, 0,
- 0x7f, present);
-
+ AC_VERB_GET_VOLUME_KNOB_CONTROL, 0);
+ present &= HDA_AMP_VOLMASK;
+ snd_hda_codec_amp_stereo(codec, 0x0c, HDA_OUTPUT, 0,
+ HDA_AMP_VOLMASK, present);
+ snd_hda_codec_amp_stereo(codec, 0x0d, HDA_OUTPUT, 0,
+ HDA_AMP_VOLMASK, present);
}
+
static void alc880_uniwill_p53_unsol_event(struct hda_codec *codec,
unsigned int res)
{
@@ -1868,8 +1998,8 @@ static struct hda_verb alc880_lg_init_verbs[] = {
{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
/* mute all amp mixer inputs */
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(5)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(6)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(7)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)},
/* line-in to input */
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
@@ -1900,11 +2030,9 @@ static void alc880_lg_automute(struct hda_codec *codec)
present = snd_hda_codec_read(codec, 0x1b, 0,
AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
- bits = present ? 0x80 : 0;
- snd_hda_codec_amp_update(codec, 0x17, 0, HDA_OUTPUT, 0,
- 0x80, bits);
- snd_hda_codec_amp_update(codec, 0x17, 1, HDA_OUTPUT, 0,
- 0x80, bits);
+ bits = present ? HDA_AMP_MUTE : 0;
+ snd_hda_codec_amp_stereo(codec, 0x17, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, bits);
}
static void alc880_lg_unsol_event(struct hda_codec *codec, unsigned int res)
@@ -1973,7 +2101,7 @@ static struct hda_verb alc880_lg_lw_init_verbs[] = {
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(7)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)},
/* speaker-out */
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
@@ -1999,11 +2127,9 @@ static void alc880_lg_lw_automute(struct hda_codec *codec)
present = snd_hda_codec_read(codec, 0x1b, 0,
AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
- bits = present ? 0x80 : 0;
- snd_hda_codec_amp_update(codec, 0x14, 0, HDA_OUTPUT, 0,
- 0x80, bits);
- snd_hda_codec_amp_update(codec, 0x14, 1, HDA_OUTPUT, 0,
- 0x80, bits);
+ bits = present ? HDA_AMP_MUTE : 0;
+ snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, bits);
}
static void alc880_lg_lw_unsol_event(struct hda_codec *codec, unsigned int res)
@@ -2015,6 +2141,24 @@ static void alc880_lg_lw_unsol_event(struct hda_codec *codec, unsigned int res)
alc880_lg_lw_automute(codec);
}
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+static struct hda_amp_list alc880_loopbacks[] = {
+ { 0x0b, HDA_INPUT, 0 },
+ { 0x0b, HDA_INPUT, 1 },
+ { 0x0b, HDA_INPUT, 2 },
+ { 0x0b, HDA_INPUT, 3 },
+ { 0x0b, HDA_INPUT, 4 },
+ { } /* end */
+};
+
+static struct hda_amp_list alc880_lg_loopbacks[] = {
+ { 0x0b, HDA_INPUT, 1 },
+ { 0x0b, HDA_INPUT, 6 },
+ { 0x0b, HDA_INPUT, 7 },
+ { } /* end */
+};
+#endif
+
/*
* Common callbacks
*/
@@ -2041,24 +2185,11 @@ static void alc_unsol_event(struct hda_codec *codec, unsigned int res)
spec->unsol_event(codec, res);
}
-#ifdef CONFIG_PM
-/*
- * resume
- */
-static int alc_resume(struct hda_codec *codec)
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+static int alc_check_power_status(struct hda_codec *codec, hda_nid_t nid)
{
struct alc_spec *spec = codec->spec;
- int i;
-
- alc_init(codec);
- for (i = 0; i < spec->num_mixers; i++)
- snd_hda_resume_ctls(codec, spec->mixers[i]);
- if (spec->multiout.dig_out_nid)
- snd_hda_resume_spdif_out(codec);
- if (spec->dig_in_nid)
- snd_hda_resume_spdif_in(codec);
-
- return 0;
+ return snd_hda_check_amp_list_power(codec, &spec->loopback, nid);
}
#endif
@@ -2293,8 +2424,8 @@ static struct hda_codec_ops alc_patch_ops = {
.init = alc_init,
.free = alc_free,
.unsol_event = alc_unsol_event,
-#ifdef CONFIG_PM
- .resume = alc_resume,
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ .check_power_status = alc_check_power_status,
#endif
};
@@ -2392,11 +2523,14 @@ static int alc_test_pin_ctl_put(struct snd_kcontrol *kcontrol,
AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
new_ctl = ctls[ucontrol->value.enumerated.item[0]];
if (old_ctl != new_ctl) {
- snd_hda_codec_write(codec, nid, 0,
- AC_VERB_SET_PIN_WIDGET_CONTROL, new_ctl);
- snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE,
- (ucontrol->value.enumerated.item[0] >= 3 ?
- 0xb080 : 0xb000));
+ int val;
+ snd_hda_codec_write_cache(codec, nid, 0,
+ AC_VERB_SET_PIN_WIDGET_CONTROL,
+ new_ctl);
+ val = ucontrol->value.enumerated.item[0] >= 3 ?
+ HDA_AMP_MUTE : 0;
+ snd_hda_codec_amp_stereo(codec, nid, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, val);
return 1;
}
return 0;
@@ -2439,7 +2573,8 @@ static int alc_test_pin_src_put(struct snd_kcontrol *kcontrol,
sel = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONNECT_SEL, 0) & 3;
if (ucontrol->value.enumerated.item[0] != sel) {
sel = ucontrol->value.enumerated.item[0] & 3;
- snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CONNECT_SEL, sel);
+ snd_hda_codec_write_cache(codec, nid, 0,
+ AC_VERB_SET_CONNECT_SEL, sel);
return 1;
}
return 0;
@@ -2885,6 +3020,7 @@ static struct alc_config_preset alc880_presets[] = {
alc880_beep_init_verbs },
.num_dacs = ARRAY_SIZE(alc880_dac_nids),
.dac_nids = alc880_dac_nids,
+ .dig_out_nid = ALC880_DIGOUT_NID,
.num_channel_mode = ARRAY_SIZE(alc880_2_jack_modes),
.channel_mode = alc880_2_jack_modes,
.input_mux = &alc880_capture_source,
@@ -2916,6 +3052,9 @@ static struct alc_config_preset alc880_presets[] = {
.input_mux = &alc880_lg_capture_source,
.unsol_event = alc880_lg_unsol_event,
.init_hook = alc880_lg_automute,
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ .loopbacks = alc880_lg_loopbacks,
+#endif
},
[ALC880_LG_LW] = {
.mixers = { alc880_lg_lw_mixer },
@@ -3399,6 +3538,10 @@ static int patch_alc880(struct hda_codec *codec)
codec->patch_ops = alc_patch_ops;
if (board_config == ALC880_AUTO)
spec->init_hook = alc880_auto_init;
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ if (!spec->loopback.amplist)
+ spec->loopback.amplist = alc880_loopbacks;
+#endif
return 0;
}
@@ -3747,12 +3890,12 @@ static struct hda_verb alc260_init_verbs[] = {
/* Amp Indexes: CD = 0x04, Line In 1 = 0x02, Mic 1 = 0x00 &
* Line In 2 = 0x03
*/
- /* mute CD */
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)},
- /* mute Line In */
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
- /* mute Mic */
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ /* mute analog inputs */
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
/* Amp Indexes: DAC = 0x01 & mixer = 0x00 */
/* mute Front out path */
{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
@@ -3797,12 +3940,12 @@ static struct hda_verb alc260_hp_init_verbs[] = {
/* Amp Indexes: CD = 0x04, Line In 1 = 0x02, Mic 1 = 0x00 &
* Line In 2 = 0x03
*/
- /* unmute CD */
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x04 << 8))},
- /* unmute Line In */
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x02 << 8))},
- /* unmute Mic */
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
+ /* mute analog inputs */
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
/* Amp Indexes: DAC = 0x01 & mixer = 0x00 */
/* Unmute Front out path */
{0x08, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
@@ -3847,12 +3990,12 @@ static struct hda_verb alc260_hp_3013_init_verbs[] = {
/* Amp Indexes: CD = 0x04, Line In 1 = 0x02, Mic 1 = 0x00 &
* Line In 2 = 0x03
*/
- /* unmute CD */
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x04 << 8))},
- /* unmute Line In */
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x02 << 8))},
- /* unmute Mic */
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
+ /* mute analog inputs */
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
/* Amp Indexes: DAC = 0x01 & mixer = 0x00 */
/* Unmute Front out path */
{0x08, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
@@ -4069,13 +4212,17 @@ static void alc260_replacer_672v_automute(struct hda_codec *codec)
present = snd_hda_codec_read(codec, 0x0f, 0,
AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
if (present) {
- snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA, 1);
- snd_hda_codec_write(codec, 0x0f, 0,
- AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP);
+ snd_hda_codec_write_cache(codec, 0x01, 0,
+ AC_VERB_SET_GPIO_DATA, 1);
+ snd_hda_codec_write_cache(codec, 0x0f, 0,
+ AC_VERB_SET_PIN_WIDGET_CONTROL,
+ PIN_HP);
} else {
- snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA, 0);
- snd_hda_codec_write(codec, 0x0f, 0,
- AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT);
+ snd_hda_codec_write_cache(codec, 0x01, 0,
+ AC_VERB_SET_GPIO_DATA, 0);
+ snd_hda_codec_write_cache(codec, 0x0f, 0,
+ AC_VERB_SET_PIN_WIDGET_CONTROL,
+ PIN_OUT);
}
}
@@ -4470,11 +4617,12 @@ static struct hda_verb alc260_volume_init_verbs[] = {
* front panel mic (mic 2)
*/
/* Amp Indices: Mic1 = 0, Mic2 = 1, Line1 = 2, Line2 = 3, CD = 4 */
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)},
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)},
+ /* mute analog inputs */
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
/*
* Set up output mixers (0x08 - 0x0a)
@@ -4551,6 +4699,17 @@ static void alc260_auto_init(struct hda_codec *codec)
alc260_auto_init_analog_input(codec);
}
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+static struct hda_amp_list alc260_loopbacks[] = {
+ { 0x07, HDA_INPUT, 0 },
+ { 0x07, HDA_INPUT, 1 },
+ { 0x07, HDA_INPUT, 2 },
+ { 0x07, HDA_INPUT, 3 },
+ { 0x07, HDA_INPUT, 4 },
+ { } /* end */
+};
+#endif
+
/*
* ALC260 configurations
*/
@@ -4750,6 +4909,10 @@ static int patch_alc260(struct hda_codec *codec)
codec->patch_ops = alc_patch_ops;
if (board_config == ALC260_AUTO)
spec->init_hook = alc260_auto_init;
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ if (!spec->loopback.amplist)
+ spec->loopback.amplist = alc260_loopbacks;
+#endif
return 0;
}
@@ -4812,12 +4975,13 @@ static int alc882_mux_enum_put(struct snd_kcontrol *kcontrol,
idx = ucontrol->value.enumerated.item[0];
if (idx >= imux->num_items)
idx = imux->num_items - 1;
- if (*cur_val == idx && !codec->in_resume)
+ if (*cur_val == idx)
return 0;
for (i = 0; i < imux->num_items; i++) {
- unsigned int v = (i == idx) ? 0x7000 : 0x7080;
- snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE,
- v | (imux->items[i].index << 8));
+ unsigned int v = (i == idx) ? 0 : HDA_AMP_MUTE;
+ snd_hda_codec_amp_stereo(codec, nid, HDA_INPUT,
+ imux->items[i].index,
+ HDA_AMP_MUTE, v);
}
*cur_val = idx;
return 1;
@@ -4879,6 +5043,38 @@ static struct hda_channel_mode alc882_sixstack_modes[2] = {
{ 8, alc882_sixstack_ch8_init },
};
+/*
+ * macbook pro ALC885 can switch LineIn to LineOut without loosing Mic
+ */
+
+/*
+ * 2ch mode
+ */
+static struct hda_verb alc885_mbp_ch2_init[] = {
+ { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN },
+ { 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ { 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ { } /* end */
+};
+
+/*
+ * 6ch mode
+ */
+static struct hda_verb alc885_mbp_ch6_init[] = {
+ { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ { 0x1a, AC_VERB_SET_CONNECT_SEL, 0x01 },
+ { 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ { 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ { } /* end */
+};
+
+static struct hda_channel_mode alc885_mbp_6ch_modes[2] = {
+ { 2, alc885_mbp_ch2_init },
+ { 6, alc885_mbp_ch6_init },
+};
+
+
/* Pin assignment: Front=0x14, Rear=0x15, CLFE=0x16, Side=0x17
* Mic=0x18, Front Mic=0x19, Line-In=0x1a, HP=0x1b
*/
@@ -4909,6 +5105,19 @@ static struct snd_kcontrol_new alc882_base_mixer[] = {
{ } /* end */
};
+static struct snd_kcontrol_new alc885_mbp3_mixer[] = {
+ HDA_CODEC_VOLUME("Master Volume", 0x0c, 0x00, HDA_OUTPUT),
+ HDA_BIND_MUTE ("Master Switch", 0x0c, 0x02, HDA_INPUT),
+ HDA_CODEC_MUTE ("Speaker Switch", 0x14, 0x00, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Line Out Volume", 0x0d,0x00, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Line In Playback Volume", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_MUTE ("Line In Playback Switch", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x00, HDA_INPUT),
+ HDA_CODEC_MUTE ("Mic Playback Switch", 0x0b, 0x00, HDA_INPUT),
+ HDA_CODEC_VOLUME("Line In Boost", 0x1a, 0x00, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Boost", 0x18, 0x00, HDA_INPUT),
+ { } /* end */
+};
static struct snd_kcontrol_new alc882_w2jc_mixer[] = {
HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
@@ -4934,8 +5143,10 @@ static struct snd_kcontrol_new alc882_targa_mixer[] = {
HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT),
HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT),
{ } /* end */
};
@@ -4955,6 +5166,23 @@ static struct snd_kcontrol_new alc882_asus_a7j_mixer[] = {
HDA_CODEC_MUTE("Mobile Line Playback Switch", 0x0b, 0x03, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT),
+ { } /* end */
+};
+
+static struct snd_kcontrol_new alc882_asus_a7m_mixer[] = {
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT),
+ HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT),
{ } /* end */
};
@@ -5119,6 +5347,66 @@ static struct hda_verb alc882_macpro_init_verbs[] = {
{ }
};
+/* Macbook Pro rev3 */
+static struct hda_verb alc885_mbp3_init_verbs[] = {
+ /* Front mixer: unmute input/output amp left and right (volume = 0) */
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ /* Rear mixer */
+ {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ /* Front Pin: output 0 (0x0c) */
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x14, AC_VERB_SET_CONNECT_SEL, 0x00},
+ /* HP Pin: output 0 (0x0d) */
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc4},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ {0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
+ /* Mic (rear) pin: input vref at 80% */
+ {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ /* Front Mic pin: input vref at 80% */
+ {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ /* Line In pin: use output 1 when in LineOut mode */
+ {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ {0x1a, AC_VERB_SET_CONNECT_SEL, 0x01},
+
+ /* FIXME: use matrix-type input source selection */
+ /* Mixer elements: 0x18, 19, 1a, 1b, 1c, 1d, 14, 15, 16, 17, 0b */
+ /* Input mixer1: unmute Mic, F-Mic, Line, CD inputs */
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
+ /* Input mixer2 */
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
+ /* Input mixer3 */
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
+ /* ADC1: mute amp left and right */
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x07, AC_VERB_SET_CONNECT_SEL, 0x00},
+ /* ADC2: mute amp left and right */
+ {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x08, AC_VERB_SET_CONNECT_SEL, 0x00},
+ /* ADC3: mute amp left and right */
+ {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x09, AC_VERB_SET_CONNECT_SEL, 0x00},
+
+ { }
+};
+
/* iMac 24 mixer. */
static struct snd_kcontrol_new alc885_imac24_mixer[] = {
HDA_CODEC_VOLUME("Master Playback Volume", 0x0c, 0x00, HDA_OUTPUT),
@@ -5154,14 +5442,10 @@ static void alc885_imac24_automute(struct hda_codec *codec)
present = snd_hda_codec_read(codec, 0x14, 0,
AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
- snd_hda_codec_amp_update(codec, 0x18, 0, HDA_OUTPUT, 0,
- 0x80, present ? 0x80 : 0);
- snd_hda_codec_amp_update(codec, 0x18, 1, HDA_OUTPUT, 0,
- 0x80, present ? 0x80 : 0);
- snd_hda_codec_amp_update(codec, 0x1a, 0, HDA_OUTPUT, 0,
- 0x80, present ? 0x80 : 0);
- snd_hda_codec_amp_update(codec, 0x1a, 1, HDA_OUTPUT, 0,
- 0x80, present ? 0x80 : 0);
+ snd_hda_codec_amp_stereo(codec, 0x18, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
+ snd_hda_codec_amp_stereo(codec, 0x1a, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
}
/* Processes unsolicited events. */
@@ -5173,6 +5457,27 @@ static void alc885_imac24_unsol_event(struct hda_codec *codec,
alc885_imac24_automute(codec);
}
+static void alc885_mbp3_automute(struct hda_codec *codec)
+{
+ unsigned int present;
+
+ present = snd_hda_codec_read(codec, 0x15, 0,
+ AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
+ snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
+ snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, present ? 0 : HDA_AMP_MUTE);
+
+}
+static void alc885_mbp3_unsol_event(struct hda_codec *codec,
+ unsigned int res)
+{
+ /* Headphone insertion or removal. */
+ if ((res >> 26) == ALC880_HP_EVENT)
+ alc885_mbp3_automute(codec);
+}
+
+
static struct hda_verb alc882_targa_verbs[] = {
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
@@ -5198,11 +5503,10 @@ static void alc882_targa_automute(struct hda_codec *codec)
present = snd_hda_codec_read(codec, 0x14, 0,
AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
- snd_hda_codec_amp_update(codec, 0x1b, 0, HDA_OUTPUT, 0,
- 0x80, present ? 0x80 : 0);
- snd_hda_codec_amp_update(codec, 0x1b, 1, HDA_OUTPUT, 0,
- 0x80, present ? 0x80 : 0);
- snd_hda_codec_write(codec, 1, 0, AC_VERB_SET_GPIO_DATA, present ? 1 : 3);
+ snd_hda_codec_amp_stereo(codec, 0x1b, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
+ snd_hda_codec_write_cache(codec, 1, 0, AC_VERB_SET_GPIO_DATA,
+ present ? 1 : 3);
}
static void alc882_targa_unsol_event(struct hda_codec *codec, unsigned int res)
@@ -5233,6 +5537,24 @@ static struct hda_verb alc882_asus_a7j_verbs[] = {
{ } /* end */
};
+static struct hda_verb alc882_asus_a7m_verbs[] = {
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+
+ {0x14, AC_VERB_SET_CONNECT_SEL, 0x00}, /* Front */
+ {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP */
+ {0x16, AC_VERB_SET_CONNECT_SEL, 0x00}, /* Front */
+
+ {0x18, AC_VERB_SET_CONNECT_SEL, 0x02}, /* mic/clfe */
+ {0x1a, AC_VERB_SET_CONNECT_SEL, 0x01}, /* line/surround */
+ {0x1b, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP */
+ { } /* end */
+};
+
static void alc882_gpio_mute(struct hda_codec *codec, int pin, int muted)
{
unsigned int gpiostate, gpiomask, gpiodir;
@@ -5265,6 +5587,20 @@ static void alc882_gpio_mute(struct hda_codec *codec, int pin, int muted)
AC_VERB_SET_GPIO_DATA, gpiostate);
}
+/* set up GPIO at initialization */
+static void alc885_macpro_init_hook(struct hda_codec *codec)
+{
+ alc882_gpio_mute(codec, 0, 0);
+ alc882_gpio_mute(codec, 1, 0);
+}
+
+/* set up GPIO and update auto-muting at initialization */
+static void alc885_imac24_init_hook(struct hda_codec *codec)
+{
+ alc885_macpro_init_hook(codec);
+ alc885_imac24_automute(codec);
+}
+
/*
* generic initialization of ADC, input mixers and output mixers
*/
@@ -5279,17 +5615,17 @@ static struct hda_verb alc882_auto_init_verbs[] = {
{0x09, AC_VERB_SET_CONNECT_SEL, 0x00},
{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- /* Unmute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback
+ /* Mute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback
* mixer widget
* Note: PASD motherboards uses the Line In 2 as the input for
* front panel mic (mic 2)
*/
/* Amp Indices: Mic1 = 0, Mic2 = 1, Line1 = 2, Line2 = 3, CD = 4 */
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
/*
* Set up output mixers (0x0c - 0x0f)
@@ -5378,6 +5714,10 @@ static struct snd_kcontrol_new alc882_capture_mixer[] = {
{ } /* end */
};
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+#define alc882_loopbacks alc880_loopbacks
+#endif
+
/* pcm configuration: identiacal with ALC880 */
#define alc882_pcm_analog_playback alc880_pcm_analog_playback
#define alc882_pcm_analog_capture alc880_pcm_analog_capture
@@ -5392,7 +5732,11 @@ static const char *alc882_models[ALC882_MODEL_LAST] = {
[ALC882_6ST_DIG] = "6stack-dig",
[ALC882_ARIMA] = "arima",
[ALC882_W2JC] = "w2jc",
+ [ALC882_TARGA] = "targa",
+ [ALC882_ASUS_A7J] = "asus-a7j",
+ [ALC882_ASUS_A7M] = "asus-a7m",
[ALC885_MACPRO] = "macpro",
+ [ALC885_MBP3] = "mbp3",
[ALC885_IMAC24] = "imac24",
[ALC882_AUTO] = "auto",
};
@@ -5404,6 +5748,8 @@ static struct snd_pci_quirk alc882_cfg_tbl[] = {
SND_PCI_QUIRK(0x1462, 0x28fb, "Targa T8", ALC882_TARGA), /* MSI-1049 T8 */
SND_PCI_QUIRK(0x161f, 0x2054, "Arima W820", ALC882_ARIMA),
SND_PCI_QUIRK(0x1043, 0x060d, "Asus A7J", ALC882_ASUS_A7J),
+ SND_PCI_QUIRK(0x1043, 0x1243, "Asus A7J", ALC882_ASUS_A7J),
+ SND_PCI_QUIRK(0x1043, 0x13c2, "Asus A7M", ALC882_ASUS_A7M),
SND_PCI_QUIRK(0x1043, 0x817f, "Asus P5LD2", ALC882_6ST_DIG),
SND_PCI_QUIRK(0x1043, 0x81d8, "Asus P5WD", ALC882_6ST_DIG),
SND_PCI_QUIRK(0x1043, 0x1971, "Asus W2JC", ALC882_W2JC),
@@ -5455,6 +5801,20 @@ static struct alc_config_preset alc882_presets[] = {
.input_mux = &alc882_capture_source,
.dig_out_nid = ALC882_DIGOUT_NID,
},
+ [ALC885_MBP3] = {
+ .mixers = { alc885_mbp3_mixer, alc882_chmode_mixer },
+ .init_verbs = { alc885_mbp3_init_verbs,
+ alc880_gpio1_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc882_dac_nids),
+ .dac_nids = alc882_dac_nids,
+ .channel_mode = alc885_mbp_6ch_modes,
+ .num_channel_mode = ARRAY_SIZE(alc885_mbp_6ch_modes),
+ .input_mux = &alc882_capture_source,
+ .dig_out_nid = ALC882_DIGOUT_NID,
+ .dig_in_nid = ALC882_DIGIN_NID,
+ .unsol_event = alc885_mbp3_unsol_event,
+ .init_hook = alc885_mbp3_automute,
+ },
[ALC885_MACPRO] = {
.mixers = { alc882_macpro_mixer },
.init_verbs = { alc882_macpro_init_verbs },
@@ -5465,6 +5825,7 @@ static struct alc_config_preset alc882_presets[] = {
.num_channel_mode = ARRAY_SIZE(alc882_ch_modes),
.channel_mode = alc882_ch_modes,
.input_mux = &alc882_capture_source,
+ .init_hook = alc885_macpro_init_hook,
},
[ALC885_IMAC24] = {
.mixers = { alc885_imac24_mixer },
@@ -5477,7 +5838,7 @@ static struct alc_config_preset alc882_presets[] = {
.channel_mode = alc882_ch_modes,
.input_mux = &alc882_capture_source,
.unsol_event = alc885_imac24_unsol_event,
- .init_hook = alc885_imac24_automute,
+ .init_hook = alc885_imac24_init_hook,
},
[ALC882_TARGA] = {
.mixers = { alc882_targa_mixer, alc882_chmode_mixer,
@@ -5509,6 +5870,19 @@ static struct alc_config_preset alc882_presets[] = {
.need_dac_fix = 1,
.input_mux = &alc882_capture_source,
},
+ [ALC882_ASUS_A7M] = {
+ .mixers = { alc882_asus_a7m_mixer, alc882_chmode_mixer },
+ .init_verbs = { alc882_init_verbs, alc882_eapd_verbs,
+ alc880_gpio1_init_verbs,
+ alc882_asus_a7m_verbs },
+ .num_dacs = ARRAY_SIZE(alc882_dac_nids),
+ .dac_nids = alc882_dac_nids,
+ .dig_out_nid = ALC882_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc880_threestack_modes),
+ .channel_mode = alc880_threestack_modes,
+ .need_dac_fix = 1,
+ .input_mux = &alc882_capture_source,
+ },
};
@@ -5608,6 +5982,32 @@ static void alc882_auto_init_analog_input(struct hda_codec *codec)
}
}
+/* add mic boosts if needed */
+static int alc_auto_add_mic_boost(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ int err;
+ hda_nid_t nid;
+
+ nid = spec->autocfg.input_pins[AUTO_PIN_MIC];
+ if (nid) {
+ err = add_control(spec, ALC_CTL_WIDGET_VOL,
+ "Mic Boost",
+ HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_INPUT));
+ if (err < 0)
+ return err;
+ }
+ nid = spec->autocfg.input_pins[AUTO_PIN_FRONT_MIC];
+ if (nid) {
+ err = add_control(spec, ALC_CTL_WIDGET_VOL,
+ "Front Mic Boost",
+ HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_INPUT));
+ if (err < 0)
+ return err;
+ }
+ return 0;
+}
+
/* almost identical with ALC880 parser... */
static int alc882_parse_auto_config(struct hda_codec *codec)
{
@@ -5616,10 +6016,17 @@ static int alc882_parse_auto_config(struct hda_codec *codec)
if (err < 0)
return err;
- else if (err > 0)
- /* hack - override the init verbs */
- spec->init_verbs[0] = alc882_auto_init_verbs;
- return err;
+ else if (!err)
+ return 0; /* no config found */
+
+ err = alc_auto_add_mic_boost(codec);
+ if (err < 0)
+ return err;
+
+ /* hack - override the init verbs */
+ spec->init_verbs[0] = alc882_auto_init_verbs;
+
+ return 1; /* config found */
}
/* additional initialization for auto-configuration model */
@@ -5654,6 +6061,9 @@ static int patch_alc882(struct hda_codec *codec)
case 0x106b1000: /* iMac 24 */
board_config = ALC885_IMAC24;
break;
+ case 0x106b2c00: /* Macbook Pro rev3 */
+ board_config = ALC885_MBP3;
+ break;
default:
printk(KERN_INFO "hda_codec: Unknown model for ALC882, "
"trying auto-probe from BIOS...\n");
@@ -5680,11 +6090,6 @@ static int patch_alc882(struct hda_codec *codec)
if (board_config != ALC882_AUTO)
setup_preset(spec, &alc882_presets[board_config]);
- if (board_config == ALC885_MACPRO || board_config == ALC885_IMAC24) {
- alc882_gpio_mute(codec, 0, 0);
- alc882_gpio_mute(codec, 1, 0);
- }
-
spec->stream_name_analog = "ALC882 Analog";
spec->stream_analog_playback = &alc882_pcm_analog_playback;
spec->stream_analog_capture = &alc882_pcm_analog_capture;
@@ -5715,6 +6120,10 @@ static int patch_alc882(struct hda_codec *codec)
codec->patch_ops = alc_patch_ops;
if (board_config == ALC882_AUTO)
spec->init_hook = alc882_auto_init;
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ if (!spec->loopback.amplist)
+ spec->loopback.amplist = alc882_loopbacks;
+#endif
return 0;
}
@@ -5792,12 +6201,13 @@ static int alc883_mux_enum_put(struct snd_kcontrol *kcontrol,
idx = ucontrol->value.enumerated.item[0];
if (idx >= imux->num_items)
idx = imux->num_items - 1;
- if (*cur_val == idx && !codec->in_resume)
+ if (*cur_val == idx)
return 0;
for (i = 0; i < imux->num_items; i++) {
- unsigned int v = (i == idx) ? 0x7000 : 0x7080;
- snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE,
- v | (imux->items[i].index << 8));
+ unsigned int v = (i == idx) ? 0 : HDA_AMP_MUTE;
+ snd_hda_codec_amp_stereo(codec, nid, HDA_INPUT,
+ imux->items[i].index,
+ HDA_AMP_MUTE, v);
}
*cur_val = idx;
return 1;
@@ -5822,6 +6232,18 @@ static struct hda_verb alc883_3ST_ch2_init[] = {
};
/*
+ * 4ch mode
+ */
+static struct hda_verb alc883_3ST_ch4_init[] = {
+ { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
+ { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
+ { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+ { 0x1a, AC_VERB_SET_CONNECT_SEL, 0x01 },
+ { } /* end */
+};
+
+/*
* 6ch mode
*/
static struct hda_verb alc883_3ST_ch6_init[] = {
@@ -5834,8 +6256,9 @@ static struct hda_verb alc883_3ST_ch6_init[] = {
{ } /* end */
};
-static struct hda_channel_mode alc883_3ST_6ch_modes[2] = {
+static struct hda_channel_mode alc883_3ST_6ch_modes[3] = {
{ 2, alc883_3ST_ch2_init },
+ { 4, alc883_3ST_ch4_init },
{ 6, alc883_3ST_ch6_init },
};
@@ -6235,6 +6658,31 @@ static struct snd_kcontrol_new alc888_3st_hp_mixer[] = {
{ } /* end */
};
+static struct snd_kcontrol_new alc883_acer_aspire_mixer[] = {
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x14, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x09, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x09, 0x0, HDA_INPUT),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ /* .name = "Capture Source", */
+ .name = "Input Source",
+ .count = 2,
+ .info = alc883_mux_enum_info,
+ .get = alc883_mux_enum_get,
+ .put = alc883_mux_enum_put,
+ },
+ { } /* end */
+};
+
static struct snd_kcontrol_new alc883_chmode_mixer[] = {
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
@@ -6270,11 +6718,12 @@ static struct hda_verb alc883_init_verbs[] = {
{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)},
+ /* mute analog input loopbacks */
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
/* Front Pin: output 0 (0x0c) */
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
@@ -6366,6 +6815,19 @@ static struct hda_verb alc888_lenovo_ms7195_verbs[] = {
{ } /* end */
};
+static struct hda_verb alc883_haier_w66_verbs[] = {
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+
+ {0x1b, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
+ {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ { } /* end */
+};
+
static struct hda_verb alc888_6st_hp_verbs[] = {
{0x14, AC_VERB_SET_CONNECT_SEL, 0x00}, /* Front: output 0 (0x0c) */
{0x15, AC_VERB_SET_CONNECT_SEL, 0x02}, /* Rear : output 2 (0x0e) */
@@ -6409,15 +6871,10 @@ static void alc888_lenovo_ms7195_front_automute(struct hda_codec *codec)
present = snd_hda_codec_read(codec, 0x1b, 0,
AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
- snd_hda_codec_amp_update(codec, 0x14, 0, HDA_OUTPUT, 0,
- 0x80, present ? 0x80 : 0);
- snd_hda_codec_amp_update(codec, 0x14, 1, HDA_OUTPUT, 0,
- 0x80, present ? 0x80 : 0);
- snd_hda_codec_amp_update(codec, 0x15, 0, HDA_OUTPUT, 0,
- 0x80, present ? 0x80 : 0);
- snd_hda_codec_amp_update(codec, 0x15, 1, HDA_OUTPUT, 0,
- 0x80, present ? 0x80 : 0);
-
+ snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
+ snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
}
/* toggle RCA according to the front-jack state */
@@ -6427,12 +6884,10 @@ static void alc888_lenovo_ms7195_rca_automute(struct hda_codec *codec)
present = snd_hda_codec_read(codec, 0x14, 0,
AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
- snd_hda_codec_amp_update(codec, 0x15, 0, HDA_OUTPUT, 0,
- 0x80, present ? 0x80 : 0);
- snd_hda_codec_amp_update(codec, 0x15, 1, HDA_OUTPUT, 0,
- 0x80, present ? 0x80 : 0);
-
+ snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
}
+
static void alc883_lenovo_ms7195_unsol_event(struct hda_codec *codec,
unsigned int res)
{
@@ -6459,10 +6914,8 @@ static void alc883_medion_md2_automute(struct hda_codec *codec)
present = snd_hda_codec_read(codec, 0x14, 0,
AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
- snd_hda_codec_amp_update(codec, 0x15, 0, HDA_OUTPUT, 0,
- 0x80, present ? 0x80 : 0);
- snd_hda_codec_amp_update(codec, 0x15, 1, HDA_OUTPUT, 0,
- 0x80, present ? 0x80 : 0);
+ snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
}
static void alc883_medion_md2_unsol_event(struct hda_codec *codec,
@@ -6480,13 +6933,11 @@ static void alc883_tagra_automute(struct hda_codec *codec)
present = snd_hda_codec_read(codec, 0x14, 0,
AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
- bits = present ? 0x80 : 0;
- snd_hda_codec_amp_update(codec, 0x1b, 0, HDA_OUTPUT, 0,
- 0x80, bits);
- snd_hda_codec_amp_update(codec, 0x1b, 1, HDA_OUTPUT, 0,
- 0x80, bits);
- snd_hda_codec_write(codec, 1, 0, AC_VERB_SET_GPIO_DATA,
- present ? 1 : 3);
+ bits = present ? HDA_AMP_MUTE : 0;
+ snd_hda_codec_amp_stereo(codec, 0x1b, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, bits);
+ snd_hda_codec_write_cache(codec, 1, 0, AC_VERB_SET_GPIO_DATA,
+ present ? 1 : 3);
}
static void alc883_tagra_unsol_event(struct hda_codec *codec, unsigned int res)
@@ -6495,6 +6946,25 @@ static void alc883_tagra_unsol_event(struct hda_codec *codec, unsigned int res)
alc883_tagra_automute(codec);
}
+static void alc883_haier_w66_automute(struct hda_codec *codec)
+{
+ unsigned int present;
+ unsigned char bits;
+
+ present = snd_hda_codec_read(codec, 0x1b, 0,
+ AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
+ bits = present ? 0x80 : 0;
+ snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0,
+ 0x80, bits);
+}
+
+static void alc883_haier_w66_unsol_event(struct hda_codec *codec,
+ unsigned int res)
+{
+ if ((res >> 26) == ALC880_HP_EVENT)
+ alc883_haier_w66_automute(codec);
+}
+
static void alc883_lenovo_101e_ispeaker_automute(struct hda_codec *codec)
{
unsigned int present;
@@ -6502,11 +6972,9 @@ static void alc883_lenovo_101e_ispeaker_automute(struct hda_codec *codec)
present = snd_hda_codec_read(codec, 0x14, 0,
AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
- bits = present ? 0x80 : 0;
- snd_hda_codec_amp_update(codec, 0x15, 0, HDA_OUTPUT, 0,
- 0x80, bits);
- snd_hda_codec_amp_update(codec, 0x15, 1, HDA_OUTPUT, 0,
- 0x80, bits);
+ bits = present ? HDA_AMP_MUTE : 0;
+ snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, bits);
}
static void alc883_lenovo_101e_all_automute(struct hda_codec *codec)
@@ -6516,15 +6984,11 @@ static void alc883_lenovo_101e_all_automute(struct hda_codec *codec)
present = snd_hda_codec_read(codec, 0x1b, 0,
AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
- bits = present ? 0x80 : 0;
- snd_hda_codec_amp_update(codec, 0x15, 0, HDA_OUTPUT, 0,
- 0x80, bits);
- snd_hda_codec_amp_update(codec, 0x15, 1, HDA_OUTPUT, 0,
- 0x80, bits);
- snd_hda_codec_amp_update(codec, 0x14, 0, HDA_OUTPUT, 0,
- 0x80, bits);
- snd_hda_codec_amp_update(codec, 0x14, 1, HDA_OUTPUT, 0,
- 0x80, bits);
+ bits = present ? HDA_AMP_MUTE : 0;
+ snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, bits);
+ snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, bits);
}
static void alc883_lenovo_101e_unsol_event(struct hda_codec *codec,
@@ -6536,6 +7000,44 @@ static void alc883_lenovo_101e_unsol_event(struct hda_codec *codec,
alc883_lenovo_101e_ispeaker_automute(codec);
}
+/* toggle speaker-output according to the hp-jack state */
+static void alc883_acer_aspire_automute(struct hda_codec *codec)
+{
+ unsigned int present;
+
+ present = snd_hda_codec_read(codec, 0x14, 0,
+ AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
+ snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
+ snd_hda_codec_amp_stereo(codec, 0x16, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
+}
+
+static void alc883_acer_aspire_unsol_event(struct hda_codec *codec,
+ unsigned int res)
+{
+ if ((res >> 26) == ALC880_HP_EVENT)
+ alc883_acer_aspire_automute(codec);
+}
+
+static struct hda_verb alc883_acer_eapd_verbs[] = {
+ /* HP Pin: output 0 (0x0c) */
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x14, AC_VERB_SET_CONNECT_SEL, 0x00},
+ /* Front Pin: output 0 (0x0c) */
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x16, AC_VERB_SET_CONNECT_SEL, 0x00},
+ /* eanable EAPD on medion laptop */
+ {0x20, AC_VERB_SET_COEF_INDEX, 0x07},
+ {0x20, AC_VERB_SET_PROC_COEF, 0x3050},
+ /* enable unsolicited event */
+ {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
+ { }
+};
+
/*
* generic initialization of ADC, input mixers and output mixers
*/
@@ -6548,17 +7050,17 @@ static struct hda_verb alc883_auto_init_verbs[] = {
{0x09, AC_VERB_SET_CONNECT_SEL, 0x00},
{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- /* Unmute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback
+ /* Mute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback
* mixer widget
* Note: PASD motherboards uses the Line In 2 as the input for
* front panel mic (mic 2)
*/
/* Amp Indices: Mic1 = 0, Mic2 = 1, Line1 = 2, Line2 = 3, CD = 4 */
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
/*
* Set up output mixers (0x0c - 0x0f)
@@ -6621,6 +7123,10 @@ static struct snd_kcontrol_new alc883_capture_mixer[] = {
{ } /* end */
};
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+#define alc883_loopbacks alc880_loopbacks
+#endif
+
/* pcm configuration: identiacal with ALC880 */
#define alc883_pcm_analog_playback alc880_pcm_analog_playback
#define alc883_pcm_analog_capture alc880_pcm_analog_capture
@@ -6638,12 +7144,14 @@ static const char *alc883_models[ALC883_MODEL_LAST] = {
[ALC883_TARGA_DIG] = "targa-dig",
[ALC883_TARGA_2ch_DIG] = "targa-2ch-dig",
[ALC883_ACER] = "acer",
+ [ALC883_ACER_ASPIRE] = "acer-aspire",
[ALC883_MEDION] = "medion",
[ALC883_MEDION_MD2] = "medion-md2",
[ALC883_LAPTOP_EAPD] = "laptop-eapd",
[ALC883_LENOVO_101E_2ch] = "lenovo-101e",
[ALC883_LENOVO_NB0763] = "lenovo-nb0763",
[ALC888_LENOVO_MS7195_DIG] = "lenovo-ms7195-dig",
+ [ALC883_HAIER_W66] = "haier-w66",
[ALC888_6ST_HP] = "6stack-hp",
[ALC888_3ST_HP] = "3stack-hp",
[ALC883_AUTO] = "auto",
@@ -6669,10 +7177,14 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = {
SND_PCI_QUIRK(0x1462, 0x3fcc, "MSI", ALC883_TARGA_DIG),
SND_PCI_QUIRK(0x1462, 0x3fc1, "MSI", ALC883_TARGA_DIG),
SND_PCI_QUIRK(0x1462, 0x3fc3, "MSI", ALC883_TARGA_DIG),
+ SND_PCI_QUIRK(0x1462, 0x3fdf, "MSI", ALC883_TARGA_DIG),
SND_PCI_QUIRK(0x1462, 0x4314, "MSI", ALC883_TARGA_DIG),
SND_PCI_QUIRK(0x1462, 0x4319, "MSI", ALC883_TARGA_DIG),
SND_PCI_QUIRK(0x1462, 0x4324, "MSI", ALC883_TARGA_DIG),
SND_PCI_QUIRK(0x1462, 0xa422, "MSI", ALC883_TARGA_2ch_DIG),
+ SND_PCI_QUIRK(0x1025, 0x006c, "Acer Aspire 9810", ALC883_ACER_ASPIRE),
+ SND_PCI_QUIRK(0x1025, 0x0110, "Acer Aspire", ALC883_ACER_ASPIRE),
+ SND_PCI_QUIRK(0x1025, 0x0112, "Acer Aspire 9303", ALC883_ACER_ASPIRE),
SND_PCI_QUIRK(0x1025, 0, "Acer laptop", ALC883_ACER),
SND_PCI_QUIRK(0x15d9, 0x8780, "Supermicro PDSBA", ALC883_3ST_6ch),
SND_PCI_QUIRK(0x161f, 0x2054, "Medion laptop", ALC883_MEDION),
@@ -6685,6 +7197,10 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = {
SND_PCI_QUIRK(0x103c, 0x2a60, "HP Lucknow", ALC888_3ST_HP),
SND_PCI_QUIRK(0x103c, 0x2a4f, "HP Samba", ALC888_3ST_HP),
SND_PCI_QUIRK(0x17c0, 0x4071, "MEDION MD2", ALC883_MEDION_MD2),
+ SND_PCI_QUIRK(0x1991, 0x5625, "Haier W66", ALC883_HAIER_W66),
+ SND_PCI_QUIRK(0x17aa, 0x3bfc, "Lenovo NB0763", ALC883_LENOVO_NB0763),
+ SND_PCI_QUIRK(0x1043, 0x8249, "Asus M2A-VM HDMI", ALC883_3ST_6ch_DIG),
+ SND_PCI_QUIRK(0x147b, 0x1083, "Abit IP35-PRO", ALC883_6ST_DIG),
{}
};
@@ -6771,8 +7287,7 @@ static struct alc_config_preset alc883_presets[] = {
.init_hook = alc883_tagra_automute,
},
[ALC883_ACER] = {
- .mixers = { alc883_base_mixer,
- alc883_chmode_mixer },
+ .mixers = { alc883_base_mixer },
/* On TravelMate laptops, GPIO 0 enables the internal speaker
* and the headphone jack. Turn this on and rely on the
* standard mute methods whenever the user wants to turn
@@ -6787,6 +7302,20 @@ static struct alc_config_preset alc883_presets[] = {
.channel_mode = alc883_3ST_2ch_modes,
.input_mux = &alc883_capture_source,
},
+ [ALC883_ACER_ASPIRE] = {
+ .mixers = { alc883_acer_aspire_mixer },
+ .init_verbs = { alc883_init_verbs, alc883_acer_eapd_verbs },
+ .num_dacs = ARRAY_SIZE(alc883_dac_nids),
+ .dac_nids = alc883_dac_nids,
+ .dig_out_nid = ALC883_DIGOUT_NID,
+ .num_adc_nids = ARRAY_SIZE(alc883_adc_nids),
+ .adc_nids = alc883_adc_nids,
+ .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes),
+ .channel_mode = alc883_3ST_2ch_modes,
+ .input_mux = &alc883_capture_source,
+ .unsol_event = alc883_acer_aspire_unsol_event,
+ .init_hook = alc883_acer_aspire_automute,
+ },
[ALC883_MEDION] = {
.mixers = { alc883_fivestack_mixer,
alc883_chmode_mixer },
@@ -6815,8 +7344,7 @@ static struct alc_config_preset alc883_presets[] = {
.init_hook = alc883_medion_md2_automute,
},
[ALC883_LAPTOP_EAPD] = {
- .mixers = { alc883_base_mixer,
- alc883_chmode_mixer },
+ .mixers = { alc883_base_mixer },
.init_verbs = { alc883_init_verbs, alc882_eapd_verbs },
.num_dacs = ARRAY_SIZE(alc883_dac_nids),
.dac_nids = alc883_dac_nids,
@@ -6867,6 +7395,20 @@ static struct alc_config_preset alc883_presets[] = {
.input_mux = &alc883_capture_source,
.unsol_event = alc883_lenovo_ms7195_unsol_event,
.init_hook = alc888_lenovo_ms7195_front_automute,
+ },
+ [ALC883_HAIER_W66] = {
+ .mixers = { alc883_tagra_2ch_mixer},
+ .init_verbs = { alc883_init_verbs, alc883_haier_w66_verbs},
+ .num_dacs = ARRAY_SIZE(alc883_dac_nids),
+ .dac_nids = alc883_dac_nids,
+ .dig_out_nid = ALC883_DIGOUT_NID,
+ .num_adc_nids = ARRAY_SIZE(alc883_adc_nids),
+ .adc_nids = alc883_adc_nids,
+ .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes),
+ .channel_mode = alc883_3ST_2ch_modes,
+ .input_mux = &alc883_capture_source,
+ .unsol_event = alc883_haier_w66_unsol_event,
+ .init_hook = alc883_haier_w66_automute,
},
[ALC888_6ST_HP] = {
.mixers = { alc888_6st_hp_mixer, alc883_chmode_mixer },
@@ -6977,12 +7519,19 @@ static int alc883_parse_auto_config(struct hda_codec *codec)
if (err < 0)
return err;
- else if (err > 0)
- /* hack - override the init verbs */
- spec->init_verbs[0] = alc883_auto_init_verbs;
+ else if (!err)
+ return 0; /* no config found */
+
+ err = alc_auto_add_mic_boost(codec);
+ if (err < 0)
+ return err;
+
+ /* hack - override the init verbs */
+ spec->init_verbs[0] = alc883_auto_init_verbs;
spec->mixers[spec->num_mixers] = alc883_capture_mixer;
spec->num_mixers++;
- return err;
+
+ return 1; /* config found */
}
/* additional initialization for auto-configuration model */
@@ -7046,6 +7595,10 @@ static int patch_alc883(struct hda_codec *codec)
codec->patch_ops = alc_patch_ops;
if (board_config == ALC883_AUTO)
spec->init_hook = alc883_auto_init;
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ if (!spec->loopback.amplist)
+ spec->loopback.amplist = alc883_loopbacks;
+#endif
return 0;
}
@@ -7156,9 +7709,46 @@ static struct snd_kcontrol_new alc262_HP_BPC_WildWest_option_mixer[] = {
{ } /* end */
};
+/* bind hp and internal speaker mute (with plug check) */
+static int alc262_sony_master_sw_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ long *valp = ucontrol->value.integer.value;
+ int change;
+
+ /* change hp mute */
+ change = snd_hda_codec_amp_update(codec, 0x15, 0, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE,
+ valp[0] ? 0 : HDA_AMP_MUTE);
+ change |= snd_hda_codec_amp_update(codec, 0x15, 1, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE,
+ valp[1] ? 0 : HDA_AMP_MUTE);
+ if (change) {
+ /* change speaker according to HP jack state */
+ struct alc_spec *spec = codec->spec;
+ unsigned int mute;
+ if (spec->jack_present)
+ mute = HDA_AMP_MUTE;
+ else
+ mute = snd_hda_codec_amp_read(codec, 0x15, 0,
+ HDA_OUTPUT, 0);
+ snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, mute);
+ }
+ return change;
+}
+
static struct snd_kcontrol_new alc262_sony_mixer[] = {
- HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Front Playback Switch", 0x15, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Master Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Master Playback Switch",
+ .info = snd_hda_mixer_amp_switch_info,
+ .get = snd_hda_mixer_amp_switch_get,
+ .put = alc262_sony_master_sw_put,
+ .private_value = HDA_COMPOSE_AMP_VAL(0x15, 3, 0, HDA_OUTPUT),
+ },
HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME("ATAPI Mic Playback Volume", 0x0b, 0x01, HDA_INPUT),
@@ -7194,17 +7784,17 @@ static struct hda_verb alc262_init_verbs[] = {
{0x09, AC_VERB_SET_CONNECT_SEL, 0x00},
{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- /* Unmute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback
+ /* Mute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback
* mixer widget
* Note: PASD motherboards uses the Line In 2 as the input for
* front panel mic (mic 2)
*/
/* Amp Indices: Mic1 = 0, Mic2 = 1, Line1 = 2, Line2 = 3, CD = 4 */
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
/*
* Set up output mixers (0x0c - 0x0e)
@@ -7285,34 +7875,26 @@ static struct hda_verb alc262_sony_unsol_verbs[] = {
};
/* mute/unmute internal speaker according to the hp jack and mute state */
-static void alc262_hippo_automute(struct hda_codec *codec, int force)
+static void alc262_hippo_automute(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
unsigned int mute;
+ unsigned int present;
- if (force || !spec->sense_updated) {
- unsigned int present;
- /* need to execute and sync at first */
- snd_hda_codec_read(codec, 0x15, 0, AC_VERB_SET_PIN_SENSE, 0);
- present = snd_hda_codec_read(codec, 0x15, 0,
- AC_VERB_GET_PIN_SENSE, 0);
- spec->jack_present = (present & 0x80000000) != 0;
- spec->sense_updated = 1;
- }
+ /* need to execute and sync at first */
+ snd_hda_codec_read(codec, 0x15, 0, AC_VERB_SET_PIN_SENSE, 0);
+ present = snd_hda_codec_read(codec, 0x15, 0,
+ AC_VERB_GET_PIN_SENSE, 0);
+ spec->jack_present = (present & 0x80000000) != 0;
if (spec->jack_present) {
/* mute internal speaker */
- snd_hda_codec_amp_update(codec, 0x14, 0, HDA_OUTPUT, 0,
- 0x80, 0x80);
- snd_hda_codec_amp_update(codec, 0x14, 1, HDA_OUTPUT, 0,
- 0x80, 0x80);
+ snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, HDA_AMP_MUTE);
} else {
/* unmute internal speaker if necessary */
mute = snd_hda_codec_amp_read(codec, 0x15, 0, HDA_OUTPUT, 0);
- snd_hda_codec_amp_update(codec, 0x14, 0, HDA_OUTPUT, 0,
- 0x80, mute & 0x80);
- mute = snd_hda_codec_amp_read(codec, 0x15, 1, HDA_OUTPUT, 0);
- snd_hda_codec_amp_update(codec, 0x14, 1, HDA_OUTPUT, 0,
- 0x80, mute & 0x80);
+ snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, mute);
}
}
@@ -7322,37 +7904,27 @@ static void alc262_hippo_unsol_event(struct hda_codec *codec,
{
if ((res >> 26) != ALC880_HP_EVENT)
return;
- alc262_hippo_automute(codec, 1);
+ alc262_hippo_automute(codec);
}
-static void alc262_hippo1_automute(struct hda_codec *codec, int force)
+static void alc262_hippo1_automute(struct hda_codec *codec)
{
- struct alc_spec *spec = codec->spec;
unsigned int mute;
+ unsigned int present;
- if (force || !spec->sense_updated) {
- unsigned int present;
- /* need to execute and sync at first */
- snd_hda_codec_read(codec, 0x1b, 0, AC_VERB_SET_PIN_SENSE, 0);
- present = snd_hda_codec_read(codec, 0x1b, 0,
- AC_VERB_GET_PIN_SENSE, 0);
- spec->jack_present = (present & 0x80000000) != 0;
- spec->sense_updated = 1;
- }
- if (spec->jack_present) {
+ snd_hda_codec_read(codec, 0x1b, 0, AC_VERB_SET_PIN_SENSE, 0);
+ present = snd_hda_codec_read(codec, 0x1b, 0,
+ AC_VERB_GET_PIN_SENSE, 0);
+ present = (present & 0x80000000) != 0;
+ if (present) {
/* mute internal speaker */
- snd_hda_codec_amp_update(codec, 0x14, 0, HDA_OUTPUT, 0,
- 0x80, 0x80);
- snd_hda_codec_amp_update(codec, 0x14, 1, HDA_OUTPUT, 0,
- 0x80, 0x80);
+ snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, HDA_AMP_MUTE);
} else {
/* unmute internal speaker if necessary */
mute = snd_hda_codec_amp_read(codec, 0x1b, 0, HDA_OUTPUT, 0);
- snd_hda_codec_amp_update(codec, 0x14, 0, HDA_OUTPUT, 0,
- 0x80, mute & 0x80);
- mute = snd_hda_codec_amp_read(codec, 0x1b, 1, HDA_OUTPUT, 0);
- snd_hda_codec_amp_update(codec, 0x14, 1, HDA_OUTPUT, 0,
- 0x80, mute & 0x80);
+ snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, mute);
}
}
@@ -7362,7 +7934,7 @@ static void alc262_hippo1_unsol_event(struct hda_codec *codec,
{
if ((res >> 26) != ALC880_HP_EVENT)
return;
- alc262_hippo1_automute(codec, 1);
+ alc262_hippo1_automute(codec);
}
/*
@@ -7379,9 +7951,10 @@ static struct hda_verb alc262_fujitsu_unsol_verbs[] = {
};
static struct hda_input_mux alc262_fujitsu_capture_source = {
- .num_items = 2,
+ .num_items = 3,
.items = {
{ "Mic", 0x0 },
+ { "Int Mic", 0x1 },
{ "CD", 0x4 },
},
};
@@ -7390,13 +7963,23 @@ static struct hda_input_mux alc262_HP_capture_source = {
.num_items = 5,
.items = {
{ "Mic", 0x0 },
- { "Front Mic", 0x3 },
+ { "Front Mic", 0x1 },
{ "Line", 0x2 },
{ "CD", 0x4 },
{ "AUX IN", 0x6 },
},
};
+static struct hda_input_mux alc262_HP_D7000_capture_source = {
+ .num_items = 4,
+ .items = {
+ { "Mic", 0x0 },
+ { "Front Mic", 0x2 },
+ { "Line", 0x1 },
+ { "CD", 0x4 },
+ },
+};
+
/* mute/unmute internal speaker according to the hp jack and mute state */
static void alc262_fujitsu_automute(struct hda_codec *codec, int force)
{
@@ -7414,18 +7997,13 @@ static void alc262_fujitsu_automute(struct hda_codec *codec, int force)
}
if (spec->jack_present) {
/* mute internal speaker */
- snd_hda_codec_amp_update(codec, 0x15, 0, HDA_OUTPUT, 0,
- 0x80, 0x80);
- snd_hda_codec_amp_update(codec, 0x15, 1, HDA_OUTPUT, 0,
- 0x80, 0x80);
+ snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, HDA_AMP_MUTE);
} else {
/* unmute internal speaker if necessary */
mute = snd_hda_codec_amp_read(codec, 0x14, 0, HDA_OUTPUT, 0);
- snd_hda_codec_amp_update(codec, 0x15, 0, HDA_OUTPUT, 0,
- 0x80, mute & 0x80);
- mute = snd_hda_codec_amp_read(codec, 0x14, 1, HDA_OUTPUT, 0);
- snd_hda_codec_amp_update(codec, 0x15, 1, HDA_OUTPUT, 0,
- 0x80, mute & 0x80);
+ snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, mute);
}
}
@@ -7439,23 +8017,14 @@ static void alc262_fujitsu_unsol_event(struct hda_codec *codec,
}
/* bind volumes of both NID 0x0c and 0x0d */
-static int alc262_fujitsu_master_vol_put(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
- long *valp = ucontrol->value.integer.value;
- int change;
-
- change = snd_hda_codec_amp_update(codec, 0x0c, 0, HDA_OUTPUT, 0,
- 0x7f, valp[0] & 0x7f);
- change |= snd_hda_codec_amp_update(codec, 0x0c, 1, HDA_OUTPUT, 0,
- 0x7f, valp[1] & 0x7f);
- snd_hda_codec_amp_update(codec, 0x0d, 0, HDA_OUTPUT, 0,
- 0x7f, valp[0] & 0x7f);
- snd_hda_codec_amp_update(codec, 0x0d, 1, HDA_OUTPUT, 0,
- 0x7f, valp[1] & 0x7f);
- return change;
-}
+static struct hda_bind_ctls alc262_fujitsu_bind_master_vol = {
+ .ops = &snd_hda_bind_vol,
+ .values = {
+ HDA_COMPOSE_AMP_VAL(0x0c, 3, 0, HDA_OUTPUT),
+ HDA_COMPOSE_AMP_VAL(0x0d, 3, 0, HDA_OUTPUT),
+ 0
+ },
+};
/* bind hp and internal speaker mute (with plug check) */
static int alc262_fujitsu_master_sw_put(struct snd_kcontrol *kcontrol,
@@ -7466,24 +8035,18 @@ static int alc262_fujitsu_master_sw_put(struct snd_kcontrol *kcontrol,
int change;
change = snd_hda_codec_amp_update(codec, 0x14, 0, HDA_OUTPUT, 0,
- 0x80, valp[0] ? 0 : 0x80);
+ HDA_AMP_MUTE,
+ valp[0] ? 0 : HDA_AMP_MUTE);
change |= snd_hda_codec_amp_update(codec, 0x14, 1, HDA_OUTPUT, 0,
- 0x80, valp[1] ? 0 : 0x80);
- if (change || codec->in_resume)
- alc262_fujitsu_automute(codec, codec->in_resume);
+ HDA_AMP_MUTE,
+ valp[1] ? 0 : HDA_AMP_MUTE);
+ if (change)
+ alc262_fujitsu_automute(codec, 0);
return change;
}
static struct snd_kcontrol_new alc262_fujitsu_mixer[] = {
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "Master Playback Volume",
- .info = snd_hda_mixer_amp_volume_info,
- .get = snd_hda_mixer_amp_volume_get,
- .put = alc262_fujitsu_master_vol_put,
- .tlv = { .c = snd_hda_mixer_amp_tlv },
- .private_value = HDA_COMPOSE_AMP_VAL(0x0c, 3, 0, HDA_OUTPUT),
- },
+ HDA_BIND_VOL("Master Playback Volume", &alc262_fujitsu_bind_master_vol),
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.name = "Master Playback Switch",
@@ -7497,6 +8060,9 @@ static struct snd_kcontrol_new alc262_fujitsu_mixer[] = {
HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Int Mic Boost", 0x19, 0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Int Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_MUTE("Int Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
{ } /* end */
};
@@ -7611,17 +8177,17 @@ static struct hda_verb alc262_volume_init_verbs[] = {
{0x09, AC_VERB_SET_CONNECT_SEL, 0x00},
{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- /* Unmute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback
+ /* Mute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback
* mixer widget
* Note: PASD motherboards uses the Line In 2 as the input for
* front panel mic (mic 2)
*/
/* Amp Indices: Mic1 = 0, Mic2 = 1, Line1 = 2, Line2 = 3, CD = 4 */
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
/*
* Set up output mixers (0x0c - 0x0f)
@@ -7672,19 +8238,19 @@ static struct hda_verb alc262_HP_BPC_init_verbs[] = {
{0x09, AC_VERB_SET_CONNECT_SEL, 0x00},
{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- /* Unmute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback
+ /* Mute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback
* mixer widget
* Note: PASD motherboards uses the Line In 2 as the input for
* front panel mic (mic 2)
*/
/* Amp Indices: Mic1 = 0, Mic2 = 1, Line1 = 2, Line2 = 3, CD = 4 */
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(5)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(6)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)},
/*
* Set up output mixers (0x0c - 0x0e)
@@ -7759,20 +8325,20 @@ static struct hda_verb alc262_HP_BPC_WildWest_init_verbs[] = {
{0x09, AC_VERB_SET_CONNECT_SEL, 0x00},
{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- /* Unmute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback
+ /* Mute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback
* mixer widget
* Note: PASD motherboards uses the Line In 2 as the input for front
* panel mic (mic 2)
*/
/* Amp Indices: Mic1 = 0, Mic2 = 1, Line1 = 2, Line2 = 3, CD = 4 */
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(5)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(6)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(7)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)},
/*
* Set up output mixers (0x0c - 0x0e)
*/
@@ -7842,6 +8408,10 @@ static struct hda_verb alc262_HP_BPC_WildWest_init_verbs[] = {
{ }
};
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+#define alc262_loopbacks alc880_loopbacks
+#endif
+
/* pcm configuration: identiacal with ALC880 */
#define alc262_pcm_analog_playback alc880_pcm_analog_playback
#define alc262_pcm_analog_capture alc880_pcm_analog_capture
@@ -7884,6 +8454,10 @@ static int alc262_parse_auto_config(struct hda_codec *codec)
spec->num_mux_defs = 1;
spec->input_mux = &spec->private_imux;
+ err = alc_auto_add_mic_boost(codec);
+ if (err < 0)
+ return err;
+
return 1;
}
@@ -7939,6 +8513,7 @@ static struct snd_pci_quirk alc262_cfg_tbl[] = {
SND_PCI_QUIRK(0x17ff, 0x058f, "Benq Hippo", ALC262_HIPPO_1),
SND_PCI_QUIRK(0x17ff, 0x0560, "Benq ED8", ALC262_BENQ_ED8),
SND_PCI_QUIRK(0x17ff, 0x058d, "Benq T31-16", ALC262_BENQ_T31),
+ SND_PCI_QUIRK(0x104d, 0x820f, "Sony ASSAMD", ALC262_SONY_ASSAMD),
SND_PCI_QUIRK(0x104d, 0x9015, "Sony 0x9015", ALC262_SONY_ASSAMD),
SND_PCI_QUIRK(0x104d, 0x900e, "Sony ASSAMD", ALC262_SONY_ASSAMD),
SND_PCI_QUIRK(0x104d, 0x1f00, "Sony ASSAMD", ALC262_SONY_ASSAMD),
@@ -7967,6 +8542,7 @@ static struct alc_config_preset alc262_presets[] = {
.channel_mode = alc262_modes,
.input_mux = &alc262_capture_source,
.unsol_event = alc262_hippo_unsol_event,
+ .init_hook = alc262_hippo_automute,
},
[ALC262_HIPPO_1] = {
.mixers = { alc262_hippo1_mixer },
@@ -7979,10 +8555,12 @@ static struct alc_config_preset alc262_presets[] = {
.channel_mode = alc262_modes,
.input_mux = &alc262_capture_source,
.unsol_event = alc262_hippo1_unsol_event,
+ .init_hook = alc262_hippo1_automute,
},
[ALC262_FUJITSU] = {
.mixers = { alc262_fujitsu_mixer },
- .init_verbs = { alc262_init_verbs, alc262_fujitsu_unsol_verbs },
+ .init_verbs = { alc262_init_verbs, alc262_EAPD_verbs,
+ alc262_fujitsu_unsol_verbs },
.num_dacs = ARRAY_SIZE(alc262_dac_nids),
.dac_nids = alc262_dac_nids,
.hp_nid = 0x03,
@@ -8010,7 +8588,7 @@ static struct alc_config_preset alc262_presets[] = {
.hp_nid = 0x03,
.num_channel_mode = ARRAY_SIZE(alc262_modes),
.channel_mode = alc262_modes,
- .input_mux = &alc262_HP_capture_source,
+ .input_mux = &alc262_HP_D7000_capture_source,
},
[ALC262_HP_BPC_D7000_WL] = {
.mixers = { alc262_HP_BPC_WildWest_mixer,
@@ -8021,7 +8599,7 @@ static struct alc_config_preset alc262_presets[] = {
.hp_nid = 0x03,
.num_channel_mode = ARRAY_SIZE(alc262_modes),
.channel_mode = alc262_modes,
- .input_mux = &alc262_HP_capture_source,
+ .input_mux = &alc262_HP_D7000_capture_source,
},
[ALC262_BENQ_ED8] = {
.mixers = { alc262_base_mixer },
@@ -8043,6 +8621,7 @@ static struct alc_config_preset alc262_presets[] = {
.channel_mode = alc262_modes,
.input_mux = &alc262_capture_source,
.unsol_event = alc262_hippo_unsol_event,
+ .init_hook = alc262_hippo_automute,
},
[ALC262_BENQ_T31] = {
.mixers = { alc262_benq_t31_mixer },
@@ -8054,6 +8633,7 @@ static struct alc_config_preset alc262_presets[] = {
.channel_mode = alc262_modes,
.input_mux = &alc262_capture_source,
.unsol_event = alc262_hippo_unsol_event,
+ .init_hook = alc262_hippo_automute,
},
};
@@ -8139,6 +8719,10 @@ static int patch_alc262(struct hda_codec *codec)
codec->patch_ops = alc_patch_ops;
if (board_config == ALC262_AUTO)
spec->init_hook = alc262_auto_init;
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ if (!spec->loopback.amplist)
+ spec->loopback.amplist = alc262_loopbacks;
+#endif
return 0;
}
@@ -8170,9 +8754,125 @@ static struct snd_kcontrol_new alc268_base_mixer[] = {
HDA_CODEC_MUTE("Front Playback Switch", 0x14, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("Headphone Playback Volume", 0x3, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Line In Boost", 0x1a, 0, HDA_INPUT),
+ { }
+};
+
+static struct hda_verb alc268_eapd_verbs[] = {
+ {0x14, AC_VERB_SET_EAPD_BTLENABLE, 2},
+ {0x15, AC_VERB_SET_EAPD_BTLENABLE, 2},
+ { }
+};
+
+/* Toshiba specific */
+#define alc268_toshiba_automute alc262_hippo_automute
+
+static struct hda_verb alc268_toshiba_verbs[] = {
+ {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
+ { } /* end */
+};
+
+/* Acer specific */
+/* bind volumes of both NID 0x02 and 0x03 */
+static struct hda_bind_ctls alc268_acer_bind_master_vol = {
+ .ops = &snd_hda_bind_vol,
+ .values = {
+ HDA_COMPOSE_AMP_VAL(0x02, 3, 0, HDA_OUTPUT),
+ HDA_COMPOSE_AMP_VAL(0x03, 3, 0, HDA_OUTPUT),
+ 0
+ },
+};
+
+/* mute/unmute internal speaker according to the hp jack and mute state */
+static void alc268_acer_automute(struct hda_codec *codec, int force)
+{
+ struct alc_spec *spec = codec->spec;
+ unsigned int mute;
+
+ if (force || !spec->sense_updated) {
+ unsigned int present;
+ present = snd_hda_codec_read(codec, 0x14, 0,
+ AC_VERB_GET_PIN_SENSE, 0);
+ spec->jack_present = (present & 0x80000000) != 0;
+ spec->sense_updated = 1;
+ }
+ if (spec->jack_present)
+ mute = HDA_AMP_MUTE; /* mute internal speaker */
+ else /* unmute internal speaker if necessary */
+ mute = snd_hda_codec_amp_read(codec, 0x14, 0, HDA_OUTPUT, 0);
+ snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, mute);
+}
+
+
+/* bind hp and internal speaker mute (with plug check) */
+static int alc268_acer_master_sw_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ long *valp = ucontrol->value.integer.value;
+ int change;
+
+ change = snd_hda_codec_amp_update(codec, 0x14, 0, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE,
+ valp[0] ? 0 : HDA_AMP_MUTE);
+ change |= snd_hda_codec_amp_update(codec, 0x14, 1, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE,
+ valp[1] ? 0 : HDA_AMP_MUTE);
+ if (change)
+ alc268_acer_automute(codec, 0);
+ return change;
+}
+
+static struct snd_kcontrol_new alc268_acer_mixer[] = {
+ /* output mixer control */
+ HDA_BIND_VOL("Master Playback Volume", &alc268_acer_bind_master_vol),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Master Playback Switch",
+ .info = snd_hda_mixer_amp_switch_info,
+ .get = snd_hda_mixer_amp_switch_get,
+ .put = alc268_acer_master_sw_put,
+ .private_value = HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_OUTPUT),
+ },
+ HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Internal Mic Boost", 0x19, 0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Line In Boost", 0x1a, 0, HDA_INPUT),
{ }
};
+static struct hda_verb alc268_acer_verbs[] = {
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+
+ {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
+ { }
+};
+
+/* unsolicited event for HP jack sensing */
+static void alc268_toshiba_unsol_event(struct hda_codec *codec,
+ unsigned int res)
+{
+ if ((res >> 26) != ALC880_HP_EVENT)
+ return;
+ alc268_toshiba_automute(codec);
+}
+
+static void alc268_acer_unsol_event(struct hda_codec *codec,
+ unsigned int res)
+{
+ if ((res >> 26) != ALC880_HP_EVENT)
+ return;
+ alc268_acer_automute(codec, 1);
+}
+
+static void alc268_acer_init_hook(struct hda_codec *codec)
+{
+ alc268_acer_automute(codec, 1);
+}
+
/*
* generic initialization of ADC, input mixers and output mixers
*/
@@ -8282,14 +8982,16 @@ static int alc268_mux_enum_put(struct snd_kcontrol *kcontrol,
idx = ucontrol->value.enumerated.item[0];
if (idx >= imux->num_items)
idx = imux->num_items - 1;
- if (*cur_val == idx && !codec->in_resume)
+ if (*cur_val == idx)
return 0;
for (i = 0; i < imux->num_items; i++) {
- unsigned int v = (i == idx) ? 0x7000 : 0x7080;
- snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE,
- v | (imux->items[i].index << 8));
- snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CONNECT_SEL,
- idx );
+ unsigned int v = (i == idx) ? 0 : HDA_AMP_MUTE;
+ snd_hda_codec_amp_stereo(codec, nid, HDA_INPUT,
+ imux->items[i].index,
+ HDA_AMP_MUTE, v);
+ snd_hda_codec_write_cache(codec, nid, 0,
+ AC_VERB_SET_CONNECT_SEL,
+ idx );
}
*cur_val = idx;
return 1;
@@ -8530,6 +9232,10 @@ static int alc268_parse_auto_config(struct hda_codec *codec)
spec->num_mux_defs = 1;
spec->input_mux = &spec->private_imux;
+ err = alc_auto_add_mic_boost(codec);
+ if (err < 0)
+ return err;
+
return 1;
}
@@ -8551,11 +9257,19 @@ static void alc268_auto_init(struct hda_codec *codec)
*/
static const char *alc268_models[ALC268_MODEL_LAST] = {
[ALC268_3ST] = "3stack",
+ [ALC268_TOSHIBA] = "toshiba",
+ [ALC268_ACER] = "acer",
[ALC268_AUTO] = "auto",
};
static struct snd_pci_quirk alc268_cfg_tbl[] = {
SND_PCI_QUIRK(0x1043, 0x1205, "ASUS W7J", ALC268_3ST),
+ SND_PCI_QUIRK(0x1179, 0xff10, "TOSHIBA A205", ALC268_TOSHIBA),
+ SND_PCI_QUIRK(0x1179, 0xff50, "TOSHIBA A305", ALC268_TOSHIBA),
+ SND_PCI_QUIRK(0x103c, 0x30cc, "TOSHIBA", ALC268_TOSHIBA),
+ SND_PCI_QUIRK(0x1025, 0x0126, "Acer", ALC268_ACER),
+ SND_PCI_QUIRK(0x1025, 0x0130, "Acer Extensa 5210", ALC268_ACER),
+ SND_PCI_QUIRK(0x152d, 0x0763, "Diverse (CPR2000)", ALC268_ACER),
{}
};
@@ -8573,6 +9287,37 @@ static struct alc_config_preset alc268_presets[] = {
.channel_mode = alc268_modes,
.input_mux = &alc268_capture_source,
},
+ [ALC268_TOSHIBA] = {
+ .mixers = { alc268_base_mixer, alc268_capture_alt_mixer },
+ .init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
+ alc268_toshiba_verbs },
+ .num_dacs = ARRAY_SIZE(alc268_dac_nids),
+ .dac_nids = alc268_dac_nids,
+ .num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt),
+ .adc_nids = alc268_adc_nids_alt,
+ .hp_nid = 0x03,
+ .num_channel_mode = ARRAY_SIZE(alc268_modes),
+ .channel_mode = alc268_modes,
+ .input_mux = &alc268_capture_source,
+ .input_mux = &alc268_capture_source,
+ .unsol_event = alc268_toshiba_unsol_event,
+ .init_hook = alc268_toshiba_automute,
+ },
+ [ALC268_ACER] = {
+ .mixers = { alc268_acer_mixer, alc268_capture_alt_mixer },
+ .init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
+ alc268_acer_verbs },
+ .num_dacs = ARRAY_SIZE(alc268_dac_nids),
+ .dac_nids = alc268_dac_nids,
+ .num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt),
+ .adc_nids = alc268_adc_nids_alt,
+ .hp_nid = 0x02,
+ .num_channel_mode = ARRAY_SIZE(alc268_modes),
+ .channel_mode = alc268_modes,
+ .input_mux = &alc268_capture_source,
+ .unsol_event = alc268_acer_unsol_event,
+ .init_hook = alc268_acer_init_hook,
+ },
};
static int patch_alc268(struct hda_codec *codec)
@@ -9279,14 +10024,10 @@ static void alc861_toshiba_automute(struct hda_codec *codec)
present = snd_hda_codec_read(codec, 0x0f, 0,
AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
- snd_hda_codec_amp_update(codec, 0x16, 0, HDA_INPUT, 0,
- 0x80, present ? 0x80 : 0);
- snd_hda_codec_amp_update(codec, 0x16, 1, HDA_INPUT, 0,
- 0x80, present ? 0x80 : 0);
- snd_hda_codec_amp_update(codec, 0x1a, 0, HDA_INPUT, 3,
- 0x80, present ? 0 : 0x80);
- snd_hda_codec_amp_update(codec, 0x1a, 1, HDA_INPUT, 3,
- 0x80, present ? 0 : 0x80);
+ snd_hda_codec_amp_stereo(codec, 0x16, HDA_INPUT, 0,
+ HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
+ snd_hda_codec_amp_stereo(codec, 0x1a, HDA_INPUT, 3,
+ HDA_AMP_MUTE, present ? 0 : HDA_AMP_MUTE);
}
static void alc861_toshiba_unsol_event(struct hda_codec *codec,
@@ -9599,6 +10340,16 @@ static void alc861_auto_init(struct hda_codec *codec)
alc861_auto_init_analog_input(codec);
}
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+static struct hda_amp_list alc861_loopbacks[] = {
+ { 0x15, HDA_INPUT, 0 },
+ { 0x15, HDA_INPUT, 1 },
+ { 0x15, HDA_INPUT, 2 },
+ { 0x15, HDA_INPUT, 3 },
+ { } /* end */
+};
+#endif
+
/*
* configuration and preset
@@ -9796,6 +10547,10 @@ static int patch_alc861(struct hda_codec *codec)
codec->patch_ops = alc_patch_ops;
if (board_config == ALC861_AUTO)
spec->init_hook = alc861_auto_init;
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ if (!spec->loopback.amplist)
+ spec->loopback.amplist = alc861_loopbacks;
+#endif
return 0;
}
@@ -9852,6 +10607,14 @@ static struct hda_input_mux alc861vd_dallas_capture_source = {
},
};
+static struct hda_input_mux alc861vd_hp_capture_source = {
+ .num_items = 2,
+ .items = {
+ { "Front Mic", 0x0 },
+ { "ATAPI Mic", 0x1 },
+ },
+};
+
#define alc861vd_mux_enum_info alc_mux_enum_info
#define alc861vd_mux_enum_get alc_mux_enum_get
@@ -9870,12 +10633,13 @@ static int alc861vd_mux_enum_put(struct snd_kcontrol *kcontrol,
idx = ucontrol->value.enumerated.item[0];
if (idx >= imux->num_items)
idx = imux->num_items - 1;
- if (*cur_val == idx && !codec->in_resume)
+ if (*cur_val == idx)
return 0;
for (i = 0; i < imux->num_items; i++) {
- unsigned int v = (i == idx) ? 0x7000 : 0x7080;
- snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE,
- v | (imux->items[i].index << 8));
+ unsigned int v = (i == idx) ? 0 : HDA_AMP_MUTE;
+ snd_hda_codec_amp_stereo(codec, nid, HDA_INPUT,
+ imux->items[i].index,
+ HDA_AMP_MUTE, v);
}
*cur_val = idx;
return 1;
@@ -10049,17 +10813,22 @@ static struct snd_kcontrol_new alc861vd_dallas_mixer[] = {
HDA_CODEC_MUTE("ATAPI Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x05, HDA_INPUT),
HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x05, HDA_INPUT),
- HDA_CODEC_VOLUME("Capture Volume", 0x09, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("Capture Switch", 0x09, 0x0, HDA_INPUT),
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- /* .name = "Capture Source", */
- .name = "Input Source",
- .count = 1,
- .info = alc882_mux_enum_info,
- .get = alc882_mux_enum_get,
- .put = alc882_mux_enum_put,
- },
+ { } /* end */
+};
+
+/* Pin assignment: Speaker=0x14, Line-out = 0x15,
+ * Front Mic=0x18, ATAPI Mic = 0x19,
+ */
+static struct snd_kcontrol_new alc861vd_hp_mixer[] = {
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x02, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Headphone Playback Switch", 0x0d, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("ATAPI Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_MUTE("ATAPI Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+
{ } /* end */
};
@@ -10077,11 +10846,11 @@ static struct hda_verb alc861vd_volume_init_verbs[] = {
* the analog-loopback mixer widget
*/
/* Amp Indices: Mic1 = 0, Mic2 = 1, Line1 = 2, Line2 = 3, CD = 4 */
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
/* Capture mixer: unmute Mic, F-Mic, Line, CD inputs */
{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
@@ -10210,11 +10979,9 @@ static void alc861vd_lenovo_hp_automute(struct hda_codec *codec)
present = snd_hda_codec_read(codec, 0x1b, 0,
AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
- bits = present ? 0x80 : 0;
- snd_hda_codec_amp_update(codec, 0x14, 0, HDA_OUTPUT, 0,
- 0x80, bits);
- snd_hda_codec_amp_update(codec, 0x14, 1, HDA_OUTPUT, 0,
- 0x80, bits);
+ bits = present ? HDA_AMP_MUTE : 0;
+ snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, bits);
}
static void alc861vd_lenovo_mic_automute(struct hda_codec *codec)
@@ -10224,11 +10991,9 @@ static void alc861vd_lenovo_mic_automute(struct hda_codec *codec)
present = snd_hda_codec_read(codec, 0x18, 0,
AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
- bits = present ? 0x80 : 0;
- snd_hda_codec_amp_update(codec, 0x0b, 0, HDA_INPUT, 1,
- 0x80, bits);
- snd_hda_codec_amp_update(codec, 0x0b, 1, HDA_INPUT, 1,
- 0x80, bits);
+ bits = present ? HDA_AMP_MUTE : 0;
+ snd_hda_codec_amp_stereo(codec, 0x0b, HDA_INPUT, 1,
+ HDA_AMP_MUTE, bits);
}
static void alc861vd_lenovo_automute(struct hda_codec *codec)
@@ -10302,10 +11067,8 @@ static void alc861vd_dallas_automute(struct hda_codec *codec)
present = snd_hda_codec_read(codec, 0x15, 0,
AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
- snd_hda_codec_amp_update(codec, 0x14, 0, HDA_OUTPUT, 0,
- 0x80, present ? 0x80 : 0);
- snd_hda_codec_amp_update(codec, 0x14, 1, HDA_OUTPUT, 0,
- 0x80, present ? 0x80 : 0);
+ snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
}
static void alc861vd_dallas_unsol_event(struct hda_codec *codec, unsigned int res)
@@ -10314,6 +11077,10 @@ static void alc861vd_dallas_unsol_event(struct hda_codec *codec, unsigned int re
alc861vd_dallas_automute(codec);
}
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+#define alc861vd_loopbacks alc880_loopbacks
+#endif
+
/* pcm configuration: identiacal with ALC880 */
#define alc861vd_pcm_analog_playback alc880_pcm_analog_playback
#define alc861vd_pcm_analog_capture alc880_pcm_analog_capture
@@ -10325,12 +11092,13 @@ static void alc861vd_dallas_unsol_event(struct hda_codec *codec, unsigned int re
*/
static const char *alc861vd_models[ALC861VD_MODEL_LAST] = {
[ALC660VD_3ST] = "3stack-660",
- [ALC660VD_3ST_DIG]= "3stack-660-digout",
+ [ALC660VD_3ST_DIG] = "3stack-660-digout",
[ALC861VD_3ST] = "3stack",
[ALC861VD_3ST_DIG] = "3stack-digout",
[ALC861VD_6ST_DIG] = "6stack-digout",
[ALC861VD_LENOVO] = "lenovo",
[ALC861VD_DALLAS] = "dallas",
+ [ALC861VD_HP] = "hp",
[ALC861VD_AUTO] = "auto",
};
@@ -10341,11 +11109,15 @@ static struct snd_pci_quirk alc861vd_cfg_tbl[] = {
SND_PCI_QUIRK(0x10de, 0x03f0, "Realtek ALC660 demo", ALC660VD_3ST),
SND_PCI_QUIRK(0x1019, 0xa88d, "Realtek ALC660 demo", ALC660VD_3ST),
- SND_PCI_QUIRK(0x1179, 0xff00, "DALLAS", ALC861VD_DALLAS),
+ /*SND_PCI_QUIRK(0x1179, 0xff00, "DALLAS", ALC861VD_DALLAS),*/ /*lenovo*/
SND_PCI_QUIRK(0x1179, 0xff01, "DALLAS", ALC861VD_DALLAS),
SND_PCI_QUIRK(0x17aa, 0x3802, "Lenovo 3000 C200", ALC861VD_LENOVO),
SND_PCI_QUIRK(0x17aa, 0x2066, "Lenovo", ALC861VD_LENOVO),
SND_PCI_QUIRK(0x1179, 0xff00, "Toshiba A135", ALC861VD_LENOVO),
+ SND_PCI_QUIRK(0x1179, 0xff03, "Toshiba P205", ALC861VD_LENOVO),
+ SND_PCI_QUIRK(0x1565, 0x820d, "Biostar NF61S SE", ALC861VD_6ST_DIG),
+ SND_PCI_QUIRK(0x1849, 0x0862, "ASRock K8NF6G-VSTA", ALC861VD_6ST_DIG),
+ SND_PCI_QUIRK(0x103c, 0x30bf, "HP TX1000", ALC861VD_HP),
{}
};
@@ -10435,7 +11207,21 @@ static struct alc_config_preset alc861vd_presets[] = {
.input_mux = &alc861vd_dallas_capture_source,
.unsol_event = alc861vd_dallas_unsol_event,
.init_hook = alc861vd_dallas_automute,
- },
+ },
+ [ALC861VD_HP] = {
+ .mixers = { alc861vd_hp_mixer },
+ .init_verbs = { alc861vd_dallas_verbs, alc861vd_eapd_verbs },
+ .num_dacs = ARRAY_SIZE(alc861vd_dac_nids),
+ .dac_nids = alc861vd_dac_nids,
+ .num_adc_nids = ARRAY_SIZE(alc861vd_adc_nids),
+ .dig_out_nid = ALC861VD_DIGOUT_NID,
+ .adc_nids = alc861vd_adc_nids,
+ .num_channel_mode = ARRAY_SIZE(alc861vd_3stack_2ch_modes),
+ .channel_mode = alc861vd_3stack_2ch_modes,
+ .input_mux = &alc861vd_hp_capture_source,
+ .unsol_event = alc861vd_dallas_unsol_event,
+ .init_hook = alc861vd_dallas_automute,
+ },
};
/*
@@ -10668,6 +11454,10 @@ static int alc861vd_parse_auto_config(struct hda_codec *codec)
spec->num_mux_defs = 1;
spec->input_mux = &spec->private_imux;
+ err = alc_auto_add_mic_boost(codec);
+ if (err < 0)
+ return err;
+
return 1;
}
@@ -10735,6 +11525,10 @@ static int patch_alc861vd(struct hda_codec *codec)
if (board_config == ALC861VD_AUTO)
spec->init_hook = alc861vd_auto_init;
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ if (!spec->loopback.amplist)
+ spec->loopback.amplist = alc861vd_loopbacks;
+#endif
return 0;
}
@@ -10782,6 +11576,15 @@ static struct hda_input_mux alc662_lenovo_101e_capture_source = {
{ "Line", 0x2 },
},
};
+
+static struct hda_input_mux alc662_eeepc_capture_source = {
+ .num_items = 2,
+ .items = {
+ { "i-Mic", 0x1 },
+ { "e-Mic", 0x0 },
+ },
+};
+
#define alc662_mux_enum_info alc_mux_enum_info
#define alc662_mux_enum_get alc_mux_enum_get
@@ -10792,7 +11595,7 @@ static int alc662_mux_enum_put(struct snd_kcontrol *kcontrol,
struct alc_spec *spec = codec->spec;
const struct hda_input_mux *imux = spec->input_mux;
unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
- static hda_nid_t capture_mixers[3] = { 0x24, 0x23, 0x22 };
+ static hda_nid_t capture_mixers[2] = { 0x23, 0x22 };
hda_nid_t nid = capture_mixers[adc_idx];
unsigned int *cur_val = &spec->cur_mux[adc_idx];
unsigned int i, idx;
@@ -10800,12 +11603,13 @@ static int alc662_mux_enum_put(struct snd_kcontrol *kcontrol,
idx = ucontrol->value.enumerated.item[0];
if (idx >= imux->num_items)
idx = imux->num_items - 1;
- if (*cur_val == idx && !codec->in_resume)
+ if (*cur_val == idx)
return 0;
for (i = 0; i < imux->num_items; i++) {
- unsigned int v = (i == idx) ? 0x7000 : 0x7080;
- snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE,
- v | (imux->items[i].index << 8));
+ unsigned int v = (i == idx) ? 0 : HDA_AMP_MUTE;
+ snd_hda_codec_amp_stereo(codec, nid, HDA_INPUT,
+ imux->items[i].index,
+ HDA_AMP_MUTE, v);
}
*cur_val = idx;
return 1;
@@ -10997,6 +11801,22 @@ static struct snd_kcontrol_new alc662_lenovo_101e_mixer[] = {
{ } /* end */
};
+static struct snd_kcontrol_new alc662_eeepc_p701_mixer[] = {
+ HDA_CODEC_MUTE("iSpeaker Playback Switch", 0x14, 0x0, HDA_OUTPUT),
+
+ HDA_CODEC_VOLUME("LineOut Playback Volume", 0x02, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("LineOut Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
+
+ HDA_CODEC_VOLUME("e-Mic Boost", 0x18, 0, HDA_INPUT),
+ HDA_CODEC_VOLUME("e-Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("e-Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+
+ HDA_CODEC_VOLUME("i-Mic Boost", 0x19, 0, HDA_INPUT),
+ HDA_CODEC_VOLUME("i-Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_MUTE("i-Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+ { } /* end */
+};
+
static struct snd_kcontrol_new alc662_chmode_mixer[] = {
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
@@ -11014,18 +11834,18 @@ static struct hda_verb alc662_init_verbs[] = {
{0x09, AC_VERB_SET_CONNECT_SEL, 0x00},
/* Front mixer: unmute input/output amp left and right (volume = 0) */
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
- {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
/* Front Pin: output 0 (0x0c) */
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
@@ -11062,13 +11882,24 @@ static struct hda_verb alc662_init_verbs[] = {
{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)},
+
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)},
{ }
};
static struct hda_verb alc662_sue_init_verbs[] = {
{0x14, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|ALC880_FRONT_EVENT},
{0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|ALC880_HP_EVENT},
- {}
+ {}
+};
+
+static struct hda_verb alc662_eeepc_sue_init_verbs[] = {
+ {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_MIC_EVENT},
+ {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
+ {}
};
/*
@@ -11087,11 +11918,11 @@ static struct hda_verb alc662_auto_init_verbs[] = {
* panel mic (mic 2)
*/
/* Amp Indices: Mic1 = 0, Mic2 = 1, Line1 = 2, Line2 = 3, CD = 4 */
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
/*
* Set up output mixers (0x0c - 0x0f)
@@ -11103,23 +11934,19 @@ static struct hda_verb alc662_auto_init_verbs[] = {
/* set up input amps for analog loopback */
/* Amp Indices: DAC = 0, mixer = 1 */
- {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
/* FIXME: use matrix-type input source selection */
/* Mixer elements: 0x18, 19, 1a, 1b, 1c, 1d, 14, 15, 16, 17, 0b */
/* Input mixer */
{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
- /*{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)},*/
- {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)},
-
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{ }
};
@@ -11150,11 +11977,9 @@ static void alc662_lenovo_101e_ispeaker_automute(struct hda_codec *codec)
present = snd_hda_codec_read(codec, 0x14, 0,
AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
- bits = present ? 0x80 : 0;
- snd_hda_codec_amp_update(codec, 0x15, 0, HDA_OUTPUT, 0,
- 0x80, bits);
- snd_hda_codec_amp_update(codec, 0x15, 1, HDA_OUTPUT, 0,
- 0x80, bits);
+ bits = present ? HDA_AMP_MUTE : 0;
+ snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, bits);
}
static void alc662_lenovo_101e_all_automute(struct hda_codec *codec)
@@ -11164,15 +11989,11 @@ static void alc662_lenovo_101e_all_automute(struct hda_codec *codec)
present = snd_hda_codec_read(codec, 0x1b, 0,
AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
- bits = present ? 0x80 : 0;
- snd_hda_codec_amp_update(codec, 0x15, 0, HDA_OUTPUT, 0,
- 0x80, bits);
- snd_hda_codec_amp_update(codec, 0x15, 1, HDA_OUTPUT, 0,
- 0x80, bits);
- snd_hda_codec_amp_update(codec, 0x14, 0, HDA_OUTPUT, 0,
- 0x80, bits);
- snd_hda_codec_amp_update(codec, 0x14, 1, HDA_OUTPUT, 0,
- 0x80, bits);
+ bits = present ? HDA_AMP_MUTE : 0;
+ snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, bits);
+ snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, bits);
}
static void alc662_lenovo_101e_unsol_event(struct hda_codec *codec,
@@ -11184,6 +12005,43 @@ static void alc662_lenovo_101e_unsol_event(struct hda_codec *codec,
alc662_lenovo_101e_ispeaker_automute(codec);
}
+static void alc662_eeepc_mic_automute(struct hda_codec *codec)
+{
+ unsigned int present;
+
+ present = snd_hda_codec_read(codec, 0x18, 0,
+ AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
+ snd_hda_codec_write(codec, 0x22, 0, AC_VERB_SET_AMP_GAIN_MUTE,
+ 0x7000 | (0x00 << 8) | (present ? 0 : 0x80));
+ snd_hda_codec_write(codec, 0x23, 0, AC_VERB_SET_AMP_GAIN_MUTE,
+ 0x7000 | (0x00 << 8) | (present ? 0 : 0x80));
+ snd_hda_codec_write(codec, 0x22, 0, AC_VERB_SET_AMP_GAIN_MUTE,
+ 0x7000 | (0x01 << 8) | (present ? 0x80 : 0));
+ snd_hda_codec_write(codec, 0x23, 0, AC_VERB_SET_AMP_GAIN_MUTE,
+ 0x7000 | (0x01 << 8) | (present ? 0x80 : 0));
+}
+
+/* unsolicited event for HP jack sensing */
+static void alc662_eeepc_unsol_event(struct hda_codec *codec,
+ unsigned int res)
+{
+ if ((res >> 26) == ALC880_HP_EVENT)
+ alc262_hippo1_automute( codec );
+
+ if ((res >> 26) == ALC880_MIC_EVENT)
+ alc662_eeepc_mic_automute(codec);
+}
+
+static void alc662_eeepc_inithook(struct hda_codec *codec)
+{
+ alc262_hippo1_automute( codec );
+ alc662_eeepc_mic_automute(codec);
+}
+
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+#define alc662_loopbacks alc880_loopbacks
+#endif
+
/* pcm configuration: identiacal with ALC880 */
#define alc662_pcm_analog_playback alc880_pcm_analog_playback
@@ -11205,12 +12063,13 @@ static const char *alc662_models[ALC662_MODEL_LAST] = {
static struct snd_pci_quirk alc662_cfg_tbl[] = {
SND_PCI_QUIRK(0x17aa, 0x101e, "Lenovo", ALC662_LENOVO_101E),
+ SND_PCI_QUIRK(0x1043, 0x82a1, "ASUS Eeepc", ALC662_ASUS_EEEPC_P701),
{}
};
static struct alc_config_preset alc662_presets[] = {
[ALC662_3ST_2ch_DIG] = {
- .mixers = { alc662_3ST_2ch_mixer },
+ .mixers = { alc662_3ST_2ch_mixer, alc662_capture_mixer },
.init_verbs = { alc662_init_verbs },
.num_dacs = ARRAY_SIZE(alc662_dac_nids),
.dac_nids = alc662_dac_nids,
@@ -11223,7 +12082,8 @@ static struct alc_config_preset alc662_presets[] = {
.input_mux = &alc662_capture_source,
},
[ALC662_3ST_6ch_DIG] = {
- .mixers = { alc662_3ST_6ch_mixer, alc662_chmode_mixer },
+ .mixers = { alc662_3ST_6ch_mixer, alc662_chmode_mixer,
+ alc662_capture_mixer },
.init_verbs = { alc662_init_verbs },
.num_dacs = ARRAY_SIZE(alc662_dac_nids),
.dac_nids = alc662_dac_nids,
@@ -11237,7 +12097,8 @@ static struct alc_config_preset alc662_presets[] = {
.input_mux = &alc662_capture_source,
},
[ALC662_3ST_6ch] = {
- .mixers = { alc662_3ST_6ch_mixer, alc662_chmode_mixer },
+ .mixers = { alc662_3ST_6ch_mixer, alc662_chmode_mixer,
+ alc662_capture_mixer },
.init_verbs = { alc662_init_verbs },
.num_dacs = ARRAY_SIZE(alc662_dac_nids),
.dac_nids = alc662_dac_nids,
@@ -11249,7 +12110,8 @@ static struct alc_config_preset alc662_presets[] = {
.input_mux = &alc662_capture_source,
},
[ALC662_5ST_DIG] = {
- .mixers = { alc662_base_mixer, alc662_chmode_mixer },
+ .mixers = { alc662_base_mixer, alc662_chmode_mixer,
+ alc662_capture_mixer },
.init_verbs = { alc662_init_verbs },
.num_dacs = ARRAY_SIZE(alc662_dac_nids),
.dac_nids = alc662_dac_nids,
@@ -11262,7 +12124,7 @@ static struct alc_config_preset alc662_presets[] = {
.input_mux = &alc662_capture_source,
},
[ALC662_LENOVO_101E] = {
- .mixers = { alc662_lenovo_101e_mixer },
+ .mixers = { alc662_lenovo_101e_mixer, alc662_capture_mixer },
.init_verbs = { alc662_init_verbs, alc662_sue_init_verbs },
.num_dacs = ARRAY_SIZE(alc662_dac_nids),
.dac_nids = alc662_dac_nids,
@@ -11274,6 +12136,20 @@ static struct alc_config_preset alc662_presets[] = {
.unsol_event = alc662_lenovo_101e_unsol_event,
.init_hook = alc662_lenovo_101e_all_automute,
},
+ [ALC662_ASUS_EEEPC_P701] = {
+ .mixers = { alc662_eeepc_p701_mixer, alc662_capture_mixer },
+ .init_verbs = { alc662_init_verbs,
+ alc662_eeepc_sue_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc662_dac_nids),
+ .dac_nids = alc662_dac_nids,
+ .num_adc_nids = ARRAY_SIZE(alc861vd_adc_nids),
+ .adc_nids = alc662_adc_nids,
+ .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
+ .channel_mode = alc662_3ST_2ch_modes,
+ .input_mux = &alc662_eeepc_capture_source,
+ .unsol_event = alc662_eeepc_unsol_event,
+ .init_hook = alc662_eeepc_inithook,
+ },
};
@@ -11296,7 +12172,7 @@ static int alc662_auto_create_multi_out_ctls(struct alc_spec *spec,
for (i = 0; i < cfg->line_outs; i++) {
if (!spec->multiout.dac_nids[i])
continue;
- nid = alc880_idx_to_mixer(i);
+ nid = alc880_idx_to_dac(i);
if (i == 2) {
/* Center/LFE */
err = add_control(spec, ALC_CTL_WIDGET_VOL,
@@ -11586,6 +12462,10 @@ static int patch_alc662(struct hda_codec *codec)
codec->patch_ops = alc_patch_ops;
if (board_config == ALC662_AUTO)
spec->init_hook = alc662_auto_init;
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ if (!spec->loopback.amplist)
+ spec->loopback.amplist = alc662_loopbacks;
+#endif
return 0;
}
diff --git a/sound/pci/hda/patch_si3054.c b/sound/pci/hda/patch_si3054.c
index 6d2ecc38905c..2a4b9609aa5c 100644
--- a/sound/pci/hda/patch_si3054.c
+++ b/sound/pci/hda/patch_si3054.c
@@ -78,6 +78,8 @@
/* si3054 codec registers (nodes) access macros */
#define GET_REG(codec,reg) (snd_hda_codec_read(codec,reg,0,SI3054_VERB_READ_NODE,0))
#define SET_REG(codec,reg,val) (snd_hda_codec_write(codec,reg,0,SI3054_VERB_WRITE_NODE,val))
+#define SET_REG_CACHE(codec,reg,val) \
+ snd_hda_codec_write_cache(codec,reg,0,SI3054_VERB_WRITE_NODE,val)
struct si3054_spec {
@@ -94,15 +96,7 @@ struct si3054_spec {
#define PRIVATE_REG(val) ((val>>16)&0xffff)
#define PRIVATE_MASK(val) (val&0xffff)
-static int si3054_switch_info(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_info *uinfo)
-{
- uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
- uinfo->count = 1;
- uinfo->value.integer.min = 0;
- uinfo->value.integer.max = 1;
- return 0;
-}
+#define si3054_switch_info snd_ctl_boolean_mono_info
static int si3054_switch_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *uvalue)
@@ -121,9 +115,9 @@ static int si3054_switch_put(struct snd_kcontrol *kcontrol,
u16 reg = PRIVATE_REG(kcontrol->private_value);
u16 mask = PRIVATE_MASK(kcontrol->private_value);
if (uvalue->value.integer.value[0])
- SET_REG(codec, reg, (GET_REG(codec, reg)) | mask);
+ SET_REG_CACHE(codec, reg, (GET_REG(codec, reg)) | mask);
else
- SET_REG(codec, reg, (GET_REG(codec, reg)) & ~mask);
+ SET_REG_CACHE(codec, reg, (GET_REG(codec, reg)) & ~mask);
return 0;
}
@@ -275,10 +269,6 @@ static struct hda_codec_ops si3054_patch_ops = {
.build_pcms = si3054_build_pcms,
.init = si3054_init,
.free = si3054_free,
-#ifdef CONFIG_PM
- //.suspend = si3054_suspend,
- .resume = si3054_init,
-#endif
};
static int patch_si3054(struct hda_codec *codec)
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 3f25de72966b..bf950195107c 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -39,12 +39,25 @@
enum {
STAC_REF,
+ STAC_9200_DELL_D21,
+ STAC_9200_DELL_D22,
+ STAC_9200_DELL_D23,
+ STAC_9200_DELL_M21,
+ STAC_9200_DELL_M22,
+ STAC_9200_DELL_M23,
+ STAC_9200_DELL_M24,
+ STAC_9200_DELL_M25,
+ STAC_9200_DELL_M26,
+ STAC_9200_DELL_M27,
+ STAC_9200_GATEWAY,
STAC_9200_MODELS
};
enum {
STAC_9205_REF,
- STAC_M43xx,
+ STAC_9205_DELL_M42,
+ STAC_9205_DELL_M43,
+ STAC_9205_DELL_M44,
STAC_9205_MODELS
};
@@ -60,19 +73,22 @@ enum {
STAC_D945_REF,
STAC_D945GTP3,
STAC_D945GTP5,
- STAC_922X_DELL,
STAC_INTEL_MAC_V1,
STAC_INTEL_MAC_V2,
STAC_INTEL_MAC_V3,
STAC_INTEL_MAC_V4,
STAC_INTEL_MAC_V5,
- /* for backward compitability */
+ /* for backward compatibility */
STAC_MACMINI,
STAC_MACBOOK,
STAC_MACBOOK_PRO_V1,
STAC_MACBOOK_PRO_V2,
STAC_IMAC_INTEL,
STAC_IMAC_INTEL_20,
+ STAC_922X_DELL_D81,
+ STAC_922X_DELL_D82,
+ STAC_922X_DELL_M81,
+ STAC_922X_DELL_M82,
STAC_922X_MODELS
};
@@ -80,6 +96,7 @@ enum {
STAC_D965_REF,
STAC_D965_3ST,
STAC_D965_5ST,
+ STAC_DELL_3ST,
STAC_927X_MODELS
};
@@ -95,6 +112,8 @@ struct sigmatel_spec {
unsigned int hp_detect: 1;
unsigned int gpio_mute: 1;
+ unsigned int gpio_mask, gpio_data;
+
/* playback */
struct hda_multi_out multiout;
hda_nid_t dac_nids[5];
@@ -127,6 +146,8 @@ struct sigmatel_spec {
/* i/o switches */
unsigned int io_switch[2];
+ unsigned int clfe_swap;
+ unsigned int aloopback;
struct hda_pcm pcm_rec[2]; /* PCM information */
@@ -162,8 +183,9 @@ static hda_nid_t stac925x_dac_nids[1] = {
0x02,
};
-static hda_nid_t stac925x_dmic_nids[1] = {
- 0x15,
+#define STAC925X_NUM_DMICS 1
+static hda_nid_t stac925x_dmic_nids[STAC925X_NUM_DMICS + 1] = {
+ 0x15, 0
};
static hda_nid_t stac922x_adc_nids[2] = {
@@ -190,8 +212,9 @@ static hda_nid_t stac9205_mux_nids[2] = {
0x19, 0x1a
};
-static hda_nid_t stac9205_dmic_nids[2] = {
- 0x17, 0x18,
+#define STAC9205_NUM_DMICS 2
+static hda_nid_t stac9205_dmic_nids[STAC9205_NUM_DMICS + 1] = {
+ 0x17, 0x18, 0
};
static hda_nid_t stac9200_pin_nids[8] = {
@@ -276,12 +299,97 @@ static int stac92xx_mux_enum_put(struct snd_kcontrol *kcontrol, struct snd_ctl_e
spec->mux_nids[adc_idx], &spec->cur_mux[adc_idx]);
}
+#define stac92xx_aloopback_info snd_ctl_boolean_mono_info
+
+static int stac92xx_aloopback_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct sigmatel_spec *spec = codec->spec;
+
+ ucontrol->value.integer.value[0] = spec->aloopback;
+ return 0;
+}
+
+static int stac92xx_aloopback_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct sigmatel_spec *spec = codec->spec;
+ unsigned int dac_mode;
+
+ if (spec->aloopback == ucontrol->value.integer.value[0])
+ return 0;
+
+ spec->aloopback = ucontrol->value.integer.value[0];
+
+
+ dac_mode = snd_hda_codec_read(codec, codec->afg, 0,
+ kcontrol->private_value & 0xFFFF, 0x0);
+
+ if (spec->aloopback) {
+ snd_hda_power_up(codec);
+ dac_mode |= 0x40;
+ } else {
+ snd_hda_power_down(codec);
+ dac_mode &= ~0x40;
+ }
+
+ snd_hda_codec_write_cache(codec, codec->afg, 0,
+ kcontrol->private_value >> 16, dac_mode);
+
+ return 1;
+}
+
+static int stac92xx_volknob_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 1;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 127;
+ return 0;
+}
+
+static int stac92xx_volknob_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.integer.value[0] = kcontrol->private_value & 0xff;
+ return 0;
+}
+
+static int stac92xx_volknob_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ unsigned int val = kcontrol->private_value & 0xff;
+
+ if (val == ucontrol->value.integer.value[0])
+ return 0;
+
+ val = ucontrol->value.integer.value[0];
+ kcontrol->private_value &= ~0xff;
+ kcontrol->private_value |= val;
+
+ snd_hda_codec_write_cache(codec, kcontrol->private_value >> 16, 0,
+ AC_VERB_SET_VOLUME_KNOB_CONTROL, val | 0x80);
+ return 1;
+}
+
+
static struct hda_verb stac9200_core_init[] = {
/* set dac0mux for dac converter */
{ 0x07, AC_VERB_SET_CONNECT_SEL, 0x00},
{}
};
+static struct hda_verb stac9200_eapd_init[] = {
+ /* set dac0mux for dac converter */
+ {0x07, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x08, AC_VERB_SET_EAPD_BTLENABLE, 0x02},
+ {}
+};
+
static struct hda_verb stac925x_core_init[] = {
/* set dac0mux for dac converter */
{ 0x06, AC_VERB_SET_CONNECT_SEL, 0x00},
@@ -316,17 +424,43 @@ static struct hda_verb stac9205_core_init[] = {
{}
};
+#define STAC_INPUT_SOURCE(cnt) \
+ { \
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
+ .name = "Input Source", \
+ .count = cnt, \
+ .info = stac92xx_mux_enum_info, \
+ .get = stac92xx_mux_enum_get, \
+ .put = stac92xx_mux_enum_put, \
+ }
+
+#define STAC_ANALOG_LOOPBACK(verb_read,verb_write) \
+ { \
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
+ .name = "Analog Loopback", \
+ .count = 1, \
+ .info = stac92xx_aloopback_info, \
+ .get = stac92xx_aloopback_get, \
+ .put = stac92xx_aloopback_put, \
+ .private_value = verb_read | (verb_write << 16), \
+ }
+
+#define STAC_VOLKNOB(knob_nid) \
+ { \
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
+ .name = "Master Playback Volume", \
+ .count = 1, \
+ .info = stac92xx_volknob_info, \
+ .get = stac92xx_volknob_get, \
+ .put = stac92xx_volknob_put, \
+ .private_value = 127 | (knob_nid << 16), \
+ }
+
+
static struct snd_kcontrol_new stac9200_mixer[] = {
HDA_CODEC_VOLUME("Master Playback Volume", 0xb, 0, HDA_OUTPUT),
HDA_CODEC_MUTE("Master Playback Switch", 0xb, 0, HDA_OUTPUT),
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "Input Source",
- .count = 1,
- .info = stac92xx_mux_enum_info,
- .get = stac92xx_mux_enum_get,
- .put = stac92xx_mux_enum_put,
- },
+ STAC_INPUT_SOURCE(1),
HDA_CODEC_VOLUME("Capture Volume", 0x0a, 0, HDA_OUTPUT),
HDA_CODEC_MUTE("Capture Switch", 0x0a, 0, HDA_OUTPUT),
HDA_CODEC_VOLUME("Capture Mux Volume", 0x0c, 0, HDA_OUTPUT),
@@ -334,86 +468,68 @@ static struct snd_kcontrol_new stac9200_mixer[] = {
};
static struct snd_kcontrol_new stac925x_mixer[] = {
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "Input Source",
- .count = 1,
- .info = stac92xx_mux_enum_info,
- .get = stac92xx_mux_enum_get,
- .put = stac92xx_mux_enum_put,
- },
+ STAC_INPUT_SOURCE(1),
HDA_CODEC_VOLUME("Capture Volume", 0x09, 0, HDA_OUTPUT),
HDA_CODEC_MUTE("Capture Switch", 0x09, 0, HDA_OUTPUT),
HDA_CODEC_VOLUME("Capture Mux Volume", 0x0f, 0, HDA_OUTPUT),
{ } /* end */
};
-/* This needs to be generated dynamically based on sequence */
-static struct snd_kcontrol_new stac922x_mixer[] = {
+static struct snd_kcontrol_new stac9205_mixer[] = {
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "Input Source",
+ .name = "Digital Input Source",
.count = 1,
- .info = stac92xx_mux_enum_info,
- .get = stac92xx_mux_enum_get,
- .put = stac92xx_mux_enum_put,
+ .info = stac92xx_dmux_enum_info,
+ .get = stac92xx_dmux_enum_get,
+ .put = stac92xx_dmux_enum_put,
},
- HDA_CODEC_VOLUME("Capture Volume", 0x17, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("Capture Switch", 0x17, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Mux Capture Volume", 0x12, 0x0, HDA_OUTPUT),
+ STAC_INPUT_SOURCE(2),
+ STAC_ANALOG_LOOPBACK(0xFE0, 0x7E0),
+ STAC_VOLKNOB(0x24),
+
+ HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x1b, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x1d, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME_IDX("Mux Capture Volume", 0x0, 0x19, 0x0, HDA_OUTPUT),
+
+ HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x1c, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x1e, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME_IDX("Mux Capture Volume", 0x1, 0x1A, 0x0, HDA_OUTPUT),
+
{ } /* end */
};
/* This needs to be generated dynamically based on sequence */
-static struct snd_kcontrol_new stac9227_mixer[] = {
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "Input Source",
- .count = 1,
- .info = stac92xx_mux_enum_info,
- .get = stac92xx_mux_enum_get,
- .put = stac92xx_mux_enum_put,
- },
- HDA_CODEC_VOLUME("Capture Volume", 0x15, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Capture Switch", 0x1b, 0x0, HDA_OUTPUT),
+static struct snd_kcontrol_new stac922x_mixer[] = {
+ STAC_INPUT_SOURCE(2),
+ STAC_VOLKNOB(0x16),
+ HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x17, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x17, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME_IDX("Mux Capture Volume", 0x0, 0x12, 0x0, HDA_OUTPUT),
+
+ HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x18, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x18, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME_IDX("Mux Capture Volume", 0x1, 0x13, 0x0, HDA_OUTPUT),
{ } /* end */
};
+
static struct snd_kcontrol_new stac927x_mixer[] = {
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "Input Source",
- .count = 1,
- .info = stac92xx_mux_enum_info,
- .get = stac92xx_mux_enum_get,
- .put = stac92xx_mux_enum_put,
- },
- HDA_CODEC_VOLUME("InMux Capture Volume", 0x15, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("InVol Capture Volume", 0x18, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("ADCMux Capture Switch", 0x1b, 0x0, HDA_OUTPUT),
- { } /* end */
-};
+ STAC_INPUT_SOURCE(3),
+ STAC_VOLKNOB(0x24),
+ STAC_ANALOG_LOOPBACK(0xFEB, 0x7EB),
-static struct snd_kcontrol_new stac9205_mixer[] = {
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "Digital Input Source",
- .count = 1,
- .info = stac92xx_dmux_enum_info,
- .get = stac92xx_dmux_enum_get,
- .put = stac92xx_dmux_enum_put,
- },
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "Input Source",
- .count = 1,
- .info = stac92xx_mux_enum_info,
- .get = stac92xx_mux_enum_get,
- .put = stac92xx_mux_enum_put,
- },
- HDA_CODEC_VOLUME("InMux Capture Volume", 0x19, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("InVol Capture Volume", 0x1b, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("ADCMux Capture Switch", 0x1d, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x18, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x1b, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME_IDX("Mux Capture Volume", 0x0, 0x15, 0x0, HDA_OUTPUT),
+
+ HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x19, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x1c, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME_IDX("Mux Capture Volume", 0x1, 0x16, 0x0, HDA_OUTPUT),
+
+ HDA_CODEC_VOLUME_IDX("Capture Volume", 0x2, 0x1A, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE_IDX("Capture Switch", 0x2, 0x1d, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME_IDX("Mux Capture Volume", 0x2, 0x17, 0x0, HDA_OUTPUT),
{ } /* end */
};
@@ -451,12 +567,145 @@ static unsigned int ref9200_pin_configs[8] = {
0x02a19020, 0x01a19021, 0x90100140, 0x01813122,
};
+/*
+ STAC 9200 pin configs for
+ 102801A8
+ 102801DE
+ 102801E8
+*/
+static unsigned int dell9200_d21_pin_configs[8] = {
+ 0x400001f0, 0x400001f1, 0x02214030, 0x01014010,
+ 0x02a19020, 0x01a19021, 0x90100140, 0x01813122,
+};
+
+/*
+ STAC 9200 pin configs for
+ 102801C0
+ 102801C1
+*/
+static unsigned int dell9200_d22_pin_configs[8] = {
+ 0x400001f0, 0x400001f1, 0x0221401f, 0x01014010,
+ 0x01813020, 0x02a19021, 0x90100140, 0x400001f2,
+};
+
+/*
+ STAC 9200 pin configs for
+ 102801C4 (Dell Dimension E310)
+ 102801C5
+ 102801C7
+ 102801D9
+ 102801DA
+ 102801E3
+*/
+static unsigned int dell9200_d23_pin_configs[8] = {
+ 0x400001f0, 0x400001f1, 0x0221401f, 0x01014010,
+ 0x01813020, 0x01a19021, 0x90100140, 0x400001f2,
+};
+
+
+/*
+ STAC 9200-32 pin configs for
+ 102801B5 (Dell Inspiron 630m)
+ 102801D8 (Dell Inspiron 640m)
+*/
+static unsigned int dell9200_m21_pin_configs[8] = {
+ 0x40c003fa, 0x03441340, 0x0321121f, 0x90170310,
+ 0x408003fb, 0x03a11020, 0x401003fc, 0x403003fd,
+};
+
+/*
+ STAC 9200-32 pin configs for
+ 102801C2 (Dell Latitude D620)
+ 102801C8
+ 102801CC (Dell Latitude D820)
+ 102801D4
+ 102801D6
+*/
+static unsigned int dell9200_m22_pin_configs[8] = {
+ 0x40c003fa, 0x0144131f, 0x0321121f, 0x90170310,
+ 0x90a70321, 0x03a11020, 0x401003fb, 0x40f000fc,
+};
+
+/*
+ STAC 9200-32 pin configs for
+ 102801CE (Dell XPS M1710)
+ 102801CF (Dell Precision M90)
+*/
+static unsigned int dell9200_m23_pin_configs[8] = {
+ 0x40c003fa, 0x01441340, 0x0421421f, 0x90170310,
+ 0x408003fb, 0x04a1102e, 0x90170311, 0x403003fc,
+};
+
+/*
+ STAC 9200-32 pin configs for
+ 102801C9
+ 102801CA
+ 102801CB (Dell Latitude 120L)
+ 102801D3
+*/
+static unsigned int dell9200_m24_pin_configs[8] = {
+ 0x40c003fa, 0x404003fb, 0x0321121f, 0x90170310,
+ 0x408003fc, 0x03a11020, 0x401003fd, 0x403003fe,
+};
+
+/*
+ STAC 9200-32 pin configs for
+ 102801BD (Dell Inspiron E1505n)
+ 102801EE
+ 102801EF
+*/
+static unsigned int dell9200_m25_pin_configs[8] = {
+ 0x40c003fa, 0x01441340, 0x0421121f, 0x90170310,
+ 0x408003fb, 0x04a11020, 0x401003fc, 0x403003fd,
+};
+
+/*
+ STAC 9200-32 pin configs for
+ 102801F5 (Dell Inspiron 1501)
+ 102801F6
+*/
+static unsigned int dell9200_m26_pin_configs[8] = {
+ 0x40c003fa, 0x404003fb, 0x0421121f, 0x90170310,
+ 0x408003fc, 0x04a11020, 0x401003fd, 0x403003fe,
+};
+
+/*
+ STAC 9200-32
+ 102801CD (Dell Inspiron E1705/9400)
+*/
+static unsigned int dell9200_m27_pin_configs[8] = {
+ 0x40c003fa, 0x01441340, 0x0421121f, 0x90170310,
+ 0x90170310, 0x04a11020, 0x90170310, 0x40f003fc,
+};
+
+
static unsigned int *stac9200_brd_tbl[STAC_9200_MODELS] = {
[STAC_REF] = ref9200_pin_configs,
+ [STAC_9200_DELL_D21] = dell9200_d21_pin_configs,
+ [STAC_9200_DELL_D22] = dell9200_d22_pin_configs,
+ [STAC_9200_DELL_D23] = dell9200_d23_pin_configs,
+ [STAC_9200_DELL_M21] = dell9200_m21_pin_configs,
+ [STAC_9200_DELL_M22] = dell9200_m22_pin_configs,
+ [STAC_9200_DELL_M23] = dell9200_m23_pin_configs,
+ [STAC_9200_DELL_M24] = dell9200_m24_pin_configs,
+ [STAC_9200_DELL_M25] = dell9200_m25_pin_configs,
+ [STAC_9200_DELL_M26] = dell9200_m26_pin_configs,
+ [STAC_9200_DELL_M27] = dell9200_m27_pin_configs,
};
static const char *stac9200_models[STAC_9200_MODELS] = {
[STAC_REF] = "ref",
+ [STAC_9200_DELL_D21] = "dell-d21",
+ [STAC_9200_DELL_D22] = "dell-d22",
+ [STAC_9200_DELL_D23] = "dell-d23",
+ [STAC_9200_DELL_M21] = "dell-m21",
+ [STAC_9200_DELL_M22] = "dell-m22",
+ [STAC_9200_DELL_M23] = "dell-m23",
+ [STAC_9200_DELL_M24] = "dell-m24",
+ [STAC_9200_DELL_M25] = "dell-m25",
+ [STAC_9200_DELL_M26] = "dell-m26",
+ [STAC_9200_DELL_M27] = "dell-m27",
+ [STAC_9200_GATEWAY] = "gateway",
};
static struct snd_pci_quirk stac9200_cfg_tbl[] = {
@@ -464,30 +713,72 @@ static struct snd_pci_quirk stac9200_cfg_tbl[] = {
SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2668,
"DFI LanParty", STAC_REF),
/* Dell laptops have BIOS problem */
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01a8,
+ "unknown Dell", STAC_9200_DELL_D21),
SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01b5,
- "Dell Inspiron 630m", STAC_REF),
+ "Dell Inspiron 630m", STAC_9200_DELL_M21),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01bd,
+ "Dell Inspiron E1505n", STAC_9200_DELL_M25),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01c0,
+ "unknown Dell", STAC_9200_DELL_D22),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01c1,
+ "unknown Dell", STAC_9200_DELL_D22),
SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01c2,
- "Dell Latitude D620", STAC_REF),
+ "Dell Latitude D620", STAC_9200_DELL_M22),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01c5,
+ "unknown Dell", STAC_9200_DELL_D23),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01c7,
+ "unknown Dell", STAC_9200_DELL_D23),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01c8,
+ "unknown Dell", STAC_9200_DELL_M22),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01c9,
+ "unknown Dell", STAC_9200_DELL_M24),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01ca,
+ "unknown Dell", STAC_9200_DELL_M24),
SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01cb,
- "Dell Latitude 120L", STAC_REF),
+ "Dell Latitude 120L", STAC_9200_DELL_M24),
SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01cc,
- "Dell Latitude D820", STAC_REF),
+ "Dell Latitude D820", STAC_9200_DELL_M22),
SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01cd,
- "Dell Inspiron E1705/9400", STAC_REF),
+ "Dell Inspiron E1705/9400", STAC_9200_DELL_M27),
SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01ce,
- "Dell XPS M1710", STAC_REF),
+ "Dell XPS M1710", STAC_9200_DELL_M23),
SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01cf,
- "Dell Precision M90", STAC_REF),
+ "Dell Precision M90", STAC_9200_DELL_M23),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01d3,
+ "unknown Dell", STAC_9200_DELL_M22),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01d4,
+ "unknown Dell", STAC_9200_DELL_M22),
SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01d6,
- "unknown Dell", STAC_REF),
+ "unknown Dell", STAC_9200_DELL_M22),
SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01d8,
- "Dell Inspiron 640m", STAC_REF),
+ "Dell Inspiron 640m", STAC_9200_DELL_M21),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01d9,
+ "unknown Dell", STAC_9200_DELL_D23),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01da,
+ "unknown Dell", STAC_9200_DELL_D23),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01de,
+ "unknown Dell", STAC_9200_DELL_D21),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01e3,
+ "unknown Dell", STAC_9200_DELL_D23),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01e8,
+ "unknown Dell", STAC_9200_DELL_D21),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01ee,
+ "unknown Dell", STAC_9200_DELL_M25),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01ef,
+ "unknown Dell", STAC_9200_DELL_M25),
SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01f5,
- "Dell Inspiron 1501", STAC_REF),
-
+ "Dell Inspiron 1501", STAC_9200_DELL_M26),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01f6,
+ "unknown Dell", STAC_9200_DELL_M26),
/* Panasonic */
SND_PCI_QUIRK(0x10f7, 0x8338, "Panasonic CF-74", STAC_REF),
-
+ /* Gateway machines needs EAPD to be set on resume */
+ SND_PCI_QUIRK(0x107b, 0x0205, "Gateway S-7110M", STAC_9200_GATEWAY),
+ SND_PCI_QUIRK(0x107b, 0x0317, "Gateway MT3423, MX341*",
+ STAC_9200_GATEWAY),
+ SND_PCI_QUIRK(0x107b, 0x0318, "Gateway ML3019, MT3707",
+ STAC_9200_GATEWAY),
{} /* terminator */
};
@@ -543,6 +834,51 @@ static unsigned int ref922x_pin_configs[10] = {
0x40000100, 0x40000100,
};
+/*
+ STAC 922X pin configs for
+ 102801A7
+ 102801AB
+ 102801A9
+ 102801D1
+ 102801D2
+*/
+static unsigned int dell_922x_d81_pin_configs[10] = {
+ 0x02214030, 0x01a19021, 0x01111012, 0x01114010,
+ 0x02a19020, 0x01117011, 0x400001f0, 0x400001f1,
+ 0x01813122, 0x400001f2,
+};
+
+/*
+ STAC 922X pin configs for
+ 102801AC
+ 102801D0
+*/
+static unsigned int dell_922x_d82_pin_configs[10] = {
+ 0x02214030, 0x01a19021, 0x01111012, 0x01114010,
+ 0x02a19020, 0x01117011, 0x01451140, 0x400001f0,
+ 0x01813122, 0x400001f1,
+};
+
+/*
+ STAC 922X pin configs for
+ 102801BF
+*/
+static unsigned int dell_922x_m81_pin_configs[10] = {
+ 0x0321101f, 0x01112024, 0x01111222, 0x91174220,
+ 0x03a11050, 0x01116221, 0x90a70330, 0x01452340,
+ 0x40C003f1, 0x405003f0,
+};
+
+/*
+ STAC 9221 A1 pin configs for
+ 102801D7 (Dell XPS M1210)
+*/
+static unsigned int dell_922x_m82_pin_configs[10] = {
+ 0x0221121f, 0x408103ff, 0x02111212, 0x90100310,
+ 0x408003f1, 0x02111211, 0x03451340, 0x40c003f2,
+ 0x508003f3, 0x405003f4,
+};
+
static unsigned int d945gtp3_pin_configs[10] = {
0x0221401f, 0x01a19022, 0x01813021, 0x01014010,
0x40000100, 0x40000100, 0x40000100, 0x40000100,
@@ -585,48 +921,49 @@ static unsigned int intel_mac_v5_pin_configs[10] = {
0x400000fc, 0x400000fb,
};
-static unsigned int stac922x_dell_pin_configs[10] = {
- 0x0221121e, 0x408103ff, 0x02a1123e, 0x90100310,
- 0x408003f1, 0x0221122f, 0x03451340, 0x40c003f2,
- 0x50a003f3, 0x405003f4
-};
static unsigned int *stac922x_brd_tbl[STAC_922X_MODELS] = {
[STAC_D945_REF] = ref922x_pin_configs,
[STAC_D945GTP3] = d945gtp3_pin_configs,
[STAC_D945GTP5] = d945gtp5_pin_configs,
- [STAC_922X_DELL] = stac922x_dell_pin_configs,
[STAC_INTEL_MAC_V1] = intel_mac_v1_pin_configs,
[STAC_INTEL_MAC_V2] = intel_mac_v2_pin_configs,
[STAC_INTEL_MAC_V3] = intel_mac_v3_pin_configs,
[STAC_INTEL_MAC_V4] = intel_mac_v4_pin_configs,
[STAC_INTEL_MAC_V5] = intel_mac_v5_pin_configs,
- /* for backward compitability */
+ /* for backward compatibility */
[STAC_MACMINI] = intel_mac_v3_pin_configs,
[STAC_MACBOOK] = intel_mac_v5_pin_configs,
[STAC_MACBOOK_PRO_V1] = intel_mac_v3_pin_configs,
[STAC_MACBOOK_PRO_V2] = intel_mac_v3_pin_configs,
[STAC_IMAC_INTEL] = intel_mac_v2_pin_configs,
[STAC_IMAC_INTEL_20] = intel_mac_v3_pin_configs,
+ [STAC_922X_DELL_D81] = dell_922x_d81_pin_configs,
+ [STAC_922X_DELL_D82] = dell_922x_d82_pin_configs,
+ [STAC_922X_DELL_M81] = dell_922x_m81_pin_configs,
+ [STAC_922X_DELL_M82] = dell_922x_m82_pin_configs,
};
static const char *stac922x_models[STAC_922X_MODELS] = {
[STAC_D945_REF] = "ref",
[STAC_D945GTP5] = "5stack",
[STAC_D945GTP3] = "3stack",
- [STAC_922X_DELL] = "dell",
[STAC_INTEL_MAC_V1] = "intel-mac-v1",
[STAC_INTEL_MAC_V2] = "intel-mac-v2",
[STAC_INTEL_MAC_V3] = "intel-mac-v3",
[STAC_INTEL_MAC_V4] = "intel-mac-v4",
[STAC_INTEL_MAC_V5] = "intel-mac-v5",
- /* for backward compitability */
+ /* for backward compatibility */
[STAC_MACMINI] = "macmini",
[STAC_MACBOOK] = "macbook",
[STAC_MACBOOK_PRO_V1] = "macbook-pro-v1",
[STAC_MACBOOK_PRO_V2] = "macbook-pro",
[STAC_IMAC_INTEL] = "imac-intel",
[STAC_IMAC_INTEL_20] = "imac-intel-20",
+ [STAC_922X_DELL_D81] = "dell-d81",
+ [STAC_922X_DELL_D82] = "dell-d82",
+ [STAC_922X_DELL_M81] = "dell-m81",
+ [STAC_922X_DELL_M82] = "dell-m82",
};
static struct snd_pci_quirk stac922x_cfg_tbl[] = {
@@ -690,9 +1027,25 @@ static struct snd_pci_quirk stac922x_cfg_tbl[] = {
/* Apple Mac Mini (early 2006) */
SND_PCI_QUIRK(0x8384, 0x7680,
"Mac Mini", STAC_INTEL_MAC_V3),
- /* Dell */
- SND_PCI_QUIRK(0x1028, 0x01d7, "Dell XPS M1210", STAC_922X_DELL),
-
+ /* Dell systems */
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01a7,
+ "unknown Dell", STAC_922X_DELL_D81),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01a9,
+ "unknown Dell", STAC_922X_DELL_D81),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01ab,
+ "unknown Dell", STAC_922X_DELL_D81),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01ac,
+ "unknown Dell", STAC_922X_DELL_D82),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01bf,
+ "unknown Dell", STAC_922X_DELL_M81),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01d0,
+ "unknown Dell", STAC_922X_DELL_D82),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01d1,
+ "unknown Dell", STAC_922X_DELL_D81),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01d2,
+ "unknown Dell", STAC_922X_DELL_D81),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01d7,
+ "Dell XPS M1210", STAC_922X_DELL_M82),
{} /* terminator */
};
@@ -717,16 +1070,25 @@ static unsigned int d965_5st_pin_configs[14] = {
0x40000100, 0x40000100
};
+static unsigned int dell_3st_pin_configs[14] = {
+ 0x02211230, 0x02a11220, 0x01a19040, 0x01114210,
+ 0x01111212, 0x01116211, 0x01813050, 0x01112214,
+ 0x403003fa, 0x40000100, 0x40000100, 0x404003fb,
+ 0x40c003fc, 0x40000100
+};
+
static unsigned int *stac927x_brd_tbl[STAC_927X_MODELS] = {
[STAC_D965_REF] = ref927x_pin_configs,
[STAC_D965_3ST] = d965_3st_pin_configs,
[STAC_D965_5ST] = d965_5st_pin_configs,
+ [STAC_DELL_3ST] = dell_3st_pin_configs,
};
static const char *stac927x_models[STAC_927X_MODELS] = {
[STAC_D965_REF] = "ref",
[STAC_D965_3ST] = "3stack",
[STAC_D965_5ST] = "5stack",
+ [STAC_DELL_3ST] = "dell-3stack",
};
static struct snd_pci_quirk stac927x_cfg_tbl[] = {
@@ -753,7 +1115,13 @@ static struct snd_pci_quirk stac927x_cfg_tbl[] = {
SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2003, "Intel D965", STAC_D965_3ST),
SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2002, "Intel D965", STAC_D965_3ST),
SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2001, "Intel D965", STAC_D965_3ST),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01f3, "Dell Inspiron 1420", STAC_D965_3ST),
+ /* Dell 3 stack systems */
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01dd, "Dell Dimension E520", STAC_DELL_3ST),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01ed, "Dell ", STAC_DELL_3ST),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01f4, "Dell ", STAC_DELL_3ST),
/* 965 based 5 stack systems */
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0209, "Dell XPS 1330", STAC_D965_5ST),
SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2301, "Intel D965", STAC_D965_5ST),
SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2302, "Intel D965", STAC_D965_5ST),
SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2303, "Intel D965", STAC_D965_5ST),
@@ -772,23 +1140,97 @@ static unsigned int ref9205_pin_configs[12] = {
0x90a000f0, 0x90a000f0, 0x01441030, 0x01c41030
};
+/*
+ STAC 9205 pin configs for
+ 102801F1
+ 102801F2
+ 102801FC
+ 102801FD
+ 10280204
+ 1028021F
+*/
+static unsigned int dell_9205_m42_pin_configs[12] = {
+ 0x0321101F, 0x03A11020, 0x400003FA, 0x90170310,
+ 0x400003FB, 0x400003FC, 0x400003FD, 0x40F000F9,
+ 0x90A60330, 0x400003FF, 0x0144131F, 0x40C003FE,
+};
+
+/*
+ STAC 9205 pin configs for
+ 102801F9
+ 102801FA
+ 102801FE
+ 102801FF (Dell Precision M4300)
+ 10280206
+ 10280200
+ 10280201
+*/
+static unsigned int dell_9205_m43_pin_configs[12] = {
+ 0x0321101f, 0x03a11020, 0x90a70330, 0x90170310,
+ 0x400000fe, 0x400000ff, 0x400000fd, 0x40f000f9,
+ 0x400000fa, 0x400000fc, 0x0144131f, 0x40c003f8,
+};
+
+static unsigned int dell_9205_m44_pin_configs[12] = {
+ 0x0421101f, 0x04a11020, 0x400003fa, 0x90170310,
+ 0x400003fb, 0x400003fc, 0x400003fd, 0x400003f9,
+ 0x90a60330, 0x400003ff, 0x01441340, 0x40c003fe,
+};
+
static unsigned int *stac9205_brd_tbl[STAC_9205_MODELS] = {
- [STAC_REF] = ref9205_pin_configs,
- [STAC_M43xx] = NULL,
+ [STAC_9205_REF] = ref9205_pin_configs,
+ [STAC_9205_DELL_M42] = dell_9205_m42_pin_configs,
+ [STAC_9205_DELL_M43] = dell_9205_m43_pin_configs,
+ [STAC_9205_DELL_M44] = dell_9205_m44_pin_configs,
};
static const char *stac9205_models[STAC_9205_MODELS] = {
[STAC_9205_REF] = "ref",
+ [STAC_9205_DELL_M42] = "dell-m42",
+ [STAC_9205_DELL_M43] = "dell-m43",
+ [STAC_9205_DELL_M44] = "dell-m44",
};
static struct snd_pci_quirk stac9205_cfg_tbl[] = {
/* SigmaTel reference board */
SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2668,
"DFI LanParty", STAC_9205_REF),
- SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x01f8,
- "Dell Precision", STAC_M43xx),
- SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x01ff,
- "Dell Precision", STAC_M43xx),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01f1,
+ "unknown Dell", STAC_9205_DELL_M42),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01f2,
+ "unknown Dell", STAC_9205_DELL_M42),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01f8,
+ "Dell Precision", STAC_9205_DELL_M43),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x021c,
+ "Dell Precision", STAC_9205_DELL_M43),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01f9,
+ "Dell Precision", STAC_9205_DELL_M43),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x021b,
+ "Dell Precision", STAC_9205_DELL_M43),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01fa,
+ "Dell Precision", STAC_9205_DELL_M43),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01fc,
+ "unknown Dell", STAC_9205_DELL_M42),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01fd,
+ "unknown Dell", STAC_9205_DELL_M42),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01fe,
+ "Dell Precision", STAC_9205_DELL_M43),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01ff,
+ "Dell Precision M4300", STAC_9205_DELL_M43),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0206,
+ "Dell Precision", STAC_9205_DELL_M43),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01f1,
+ "Dell Inspiron", STAC_9205_DELL_M44),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01f2,
+ "Dell Inspiron", STAC_9205_DELL_M44),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01fc,
+ "Dell Inspiron", STAC_9205_DELL_M44),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01fd,
+ "Dell Inspiron", STAC_9205_DELL_M44),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0204,
+ "unknown Dell", STAC_9205_DELL_M42),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x021f,
+ "Dell Inspiron", STAC_9205_DELL_M44),
{} /* terminator */
};
@@ -854,20 +1296,20 @@ static void stac92xx_set_config_regs(struct hda_codec *codec)
spec->pin_configs[i]);
}
-static void stac92xx_enable_gpio_mask(struct hda_codec *codec,
- int gpio_mask, int gpio_data)
+static void stac92xx_enable_gpio_mask(struct hda_codec *codec)
{
+ struct sigmatel_spec *spec = codec->spec;
/* Configure GPIOx as output */
- snd_hda_codec_write(codec, codec->afg, 0,
- AC_VERB_SET_GPIO_DIRECTION, gpio_mask);
+ snd_hda_codec_write_cache(codec, codec->afg, 0,
+ AC_VERB_SET_GPIO_DIRECTION, spec->gpio_mask);
/* Configure GPIOx as CMOS */
- snd_hda_codec_write(codec, codec->afg, 0, 0x7e7, 0x00000000);
+ snd_hda_codec_write_cache(codec, codec->afg, 0, 0x7e7, 0x00000000);
/* Assert GPIOx */
- snd_hda_codec_write(codec, codec->afg, 0,
- AC_VERB_SET_GPIO_DATA, gpio_data);
+ snd_hda_codec_write_cache(codec, codec->afg, 0,
+ AC_VERB_SET_GPIO_DATA, spec->gpio_data);
/* Enable GPIOx */
- snd_hda_codec_write(codec, codec->afg, 0,
- AC_VERB_SET_GPIO_MASK, gpio_mask);
+ snd_hda_codec_write_cache(codec, codec->afg, 0,
+ AC_VERB_SET_GPIO_MASK, spec->gpio_mask);
}
/*
@@ -1000,10 +1442,9 @@ static struct hda_pcm_stream stac92xx_pcm_analog_alt_playback = {
};
static struct hda_pcm_stream stac92xx_pcm_analog_capture = {
- .substreams = 2,
.channels_min = 2,
.channels_max = 2,
- /* NID is set in stac92xx_build_pcms */
+ /* NID + .substreams is set in stac92xx_build_pcms */
.ops = {
.prepare = stac92xx_capture_pcm_prepare,
.cleanup = stac92xx_capture_pcm_cleanup
@@ -1022,6 +1463,7 @@ static int stac92xx_build_pcms(struct hda_codec *codec)
info->stream[SNDRV_PCM_STREAM_PLAYBACK] = stac92xx_pcm_analog_playback;
info->stream[SNDRV_PCM_STREAM_CAPTURE] = stac92xx_pcm_analog_capture;
info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->adc_nids[0];
+ info->stream[SNDRV_PCM_STREAM_CAPTURE].substreams = spec->num_adcs;
if (spec->alt_switch) {
codec->num_pcms++;
@@ -1066,17 +1508,11 @@ static unsigned int stac92xx_get_vref(struct hda_codec *codec, hda_nid_t nid)
static void stac92xx_auto_set_pinctl(struct hda_codec *codec, hda_nid_t nid, int pin_type)
{
- snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, pin_type);
+ snd_hda_codec_write_cache(codec, nid, 0,
+ AC_VERB_SET_PIN_WIDGET_CONTROL, pin_type);
}
-static int stac92xx_io_switch_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
-{
- uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
- uinfo->count = 1;
- uinfo->value.integer.min = 0;
- uinfo->value.integer.max = 1;
- return 0;
-}
+#define stac92xx_io_switch_info snd_ctl_boolean_mono_info
static int stac92xx_io_switch_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
{
@@ -1109,6 +1545,36 @@ static int stac92xx_io_switch_put(struct snd_kcontrol *kcontrol, struct snd_ctl_
return 1;
}
+#define stac92xx_clfe_switch_info snd_ctl_boolean_mono_info
+
+static int stac92xx_clfe_switch_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct sigmatel_spec *spec = codec->spec;
+
+ ucontrol->value.integer.value[0] = spec->clfe_swap;
+ return 0;
+}
+
+static int stac92xx_clfe_switch_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct sigmatel_spec *spec = codec->spec;
+ hda_nid_t nid = kcontrol->private_value & 0xff;
+
+ if (spec->clfe_swap == ucontrol->value.integer.value[0])
+ return 0;
+
+ spec->clfe_swap = ucontrol->value.integer.value[0];
+
+ snd_hda_codec_write_cache(codec, nid, 0, AC_VERB_SET_EAPD_BTLENABLE,
+ spec->clfe_swap ? 0x4 : 0x0);
+
+ return 1;
+}
+
#define STAC_CODEC_IO_SWITCH(xname, xpval) \
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
.name = xname, \
@@ -1119,17 +1585,28 @@ static int stac92xx_io_switch_put(struct snd_kcontrol *kcontrol, struct snd_ctl_
.private_value = xpval, \
}
+#define STAC_CODEC_CLFE_SWITCH(xname, xpval) \
+ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
+ .name = xname, \
+ .index = 0, \
+ .info = stac92xx_clfe_switch_info, \
+ .get = stac92xx_clfe_switch_get, \
+ .put = stac92xx_clfe_switch_put, \
+ .private_value = xpval, \
+ }
enum {
STAC_CTL_WIDGET_VOL,
STAC_CTL_WIDGET_MUTE,
STAC_CTL_WIDGET_IO_SWITCH,
+ STAC_CTL_WIDGET_CLFE_SWITCH
};
static struct snd_kcontrol_new stac92xx_control_templates[] = {
HDA_CODEC_VOLUME(NULL, 0, 0, 0),
HDA_CODEC_MUTE(NULL, 0, 0, 0),
STAC_CODEC_IO_SWITCH(NULL, 0),
+ STAC_CODEC_CLFE_SWITCH(NULL, 0),
};
/* add dynamic controls */
@@ -1182,7 +1659,8 @@ static int stac92xx_add_dyn_out_pins(struct hda_codec *codec, struct auto_pin_cf
case 3:
/* add line-in as side */
if (cfg->input_pins[AUTO_PIN_LINE] && num_dacs > 3) {
- cfg->line_out_pins[3] = cfg->input_pins[AUTO_PIN_LINE];
+ cfg->line_out_pins[cfg->line_outs] =
+ cfg->input_pins[AUTO_PIN_LINE];
spec->line_switch = 1;
cfg->line_outs++;
}
@@ -1190,12 +1668,14 @@ static int stac92xx_add_dyn_out_pins(struct hda_codec *codec, struct auto_pin_cf
case 2:
/* add line-in as clfe and mic as side */
if (cfg->input_pins[AUTO_PIN_LINE] && num_dacs > 2) {
- cfg->line_out_pins[2] = cfg->input_pins[AUTO_PIN_LINE];
+ cfg->line_out_pins[cfg->line_outs] =
+ cfg->input_pins[AUTO_PIN_LINE];
spec->line_switch = 1;
cfg->line_outs++;
}
if (cfg->input_pins[AUTO_PIN_MIC] && num_dacs > 3) {
- cfg->line_out_pins[3] = cfg->input_pins[AUTO_PIN_MIC];
+ cfg->line_out_pins[cfg->line_outs] =
+ cfg->input_pins[AUTO_PIN_MIC];
spec->mic_switch = 1;
cfg->line_outs++;
}
@@ -1203,12 +1683,14 @@ static int stac92xx_add_dyn_out_pins(struct hda_codec *codec, struct auto_pin_cf
case 1:
/* add line-in as surr and mic as clfe */
if (cfg->input_pins[AUTO_PIN_LINE] && num_dacs > 1) {
- cfg->line_out_pins[1] = cfg->input_pins[AUTO_PIN_LINE];
+ cfg->line_out_pins[cfg->line_outs] =
+ cfg->input_pins[AUTO_PIN_LINE];
spec->line_switch = 1;
cfg->line_outs++;
}
if (cfg->input_pins[AUTO_PIN_MIC] && num_dacs > 2) {
- cfg->line_out_pins[2] = cfg->input_pins[AUTO_PIN_MIC];
+ cfg->line_out_pins[cfg->line_outs] =
+ cfg->input_pins[AUTO_PIN_MIC];
spec->mic_switch = 1;
cfg->line_outs++;
}
@@ -1282,8 +1764,8 @@ static int stac92xx_auto_fill_dac_nids(struct hda_codec *codec,
spec->multiout.num_dacs++;
if (conn_len > 1) {
/* select this DAC in the pin's input mux */
- snd_hda_codec_write(codec, nid, 0,
- AC_VERB_SET_CONNECT_SEL, j);
+ snd_hda_codec_write_cache(codec, nid, 0,
+ AC_VERB_SET_CONNECT_SEL, j);
}
}
@@ -1318,7 +1800,7 @@ static int create_controls(struct sigmatel_spec *spec, const char *pfx, hda_nid_
}
/* add playback controls from the parsed DAC table */
-static int stac92xx_auto_create_multi_out_ctls(struct sigmatel_spec *spec,
+static int stac92xx_auto_create_multi_out_ctls(struct hda_codec *codec,
const struct auto_pin_cfg *cfg)
{
static const char *chname[4] = {
@@ -1327,6 +1809,10 @@ static int stac92xx_auto_create_multi_out_ctls(struct sigmatel_spec *spec,
hda_nid_t nid;
int i, err;
+ struct sigmatel_spec *spec = codec->spec;
+ unsigned int wid_caps;
+
+
for (i = 0; i < cfg->line_outs; i++) {
if (!spec->multiout.dac_nids[i])
continue;
@@ -1341,6 +1827,18 @@ static int stac92xx_auto_create_multi_out_ctls(struct sigmatel_spec *spec,
err = create_controls(spec, "LFE", nid, 2);
if (err < 0)
return err;
+
+ wid_caps = get_wcaps(codec, nid);
+
+ if (wid_caps & AC_WCAP_LR_SWAP) {
+ err = stac92xx_add_control(spec,
+ STAC_CTL_WIDGET_CLFE_SWITCH,
+ "Swap Center/LFE Playback Switch", nid);
+
+ if (err < 0)
+ return err;
+ }
+
} else {
err = create_controls(spec, chname[i], nid, 3);
if (err < 0)
@@ -1536,9 +2034,9 @@ static int stac92xx_auto_create_analog_input_ctls(struct hda_codec *codec, const
* NID lists. Hopefully this won't get confused.
*/
for (i = 0; i < spec->num_muxes; i++) {
- snd_hda_codec_write(codec, spec->mux_nids[i], 0,
- AC_VERB_SET_CONNECT_SEL,
- imux->items[0].index);
+ snd_hda_codec_write_cache(codec, spec->mux_nids[i], 0,
+ AC_VERB_SET_CONNECT_SEL,
+ imux->items[0].index);
}
}
@@ -1593,9 +2091,19 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out
if ((err = stac92xx_auto_fill_dac_nids(codec, &spec->autocfg)) < 0)
return err;
- if ((err = stac92xx_auto_create_multi_out_ctls(spec, &spec->autocfg)) < 0 ||
- (err = stac92xx_auto_create_hp_ctls(codec, &spec->autocfg)) < 0 ||
- (err = stac92xx_auto_create_analog_input_ctls(codec, &spec->autocfg)) < 0)
+ err = stac92xx_auto_create_multi_out_ctls(codec, &spec->autocfg);
+
+ if (err < 0)
+ return err;
+
+ err = stac92xx_auto_create_hp_ctls(codec, &spec->autocfg);
+
+ if (err < 0)
+ return err;
+
+ err = stac92xx_auto_create_analog_input_ctls(codec, &spec->autocfg);
+
+ if (err < 0)
return err;
if (spec->num_dmics > 0)
@@ -1764,9 +2272,9 @@ static void enable_pin_detect(struct hda_codec *codec, hda_nid_t nid,
unsigned int event)
{
if (get_wcaps(codec, nid) & AC_WCAP_UNSOL_CAP)
- snd_hda_codec_write(codec, nid, 0,
- AC_VERB_SET_UNSOLICITED_ENABLE,
- (AC_USRSP_EN | event));
+ snd_hda_codec_write_cache(codec, nid, 0,
+ AC_VERB_SET_UNSOLICITED_ENABLE,
+ (AC_USRSP_EN | event));
}
static int stac92xx_init(struct hda_codec *codec)
@@ -1870,7 +2378,7 @@ static void stac92xx_set_pinctl(struct hda_codec *codec, hda_nid_t nid,
if (flag & (AC_PINCTL_IN_EN | AC_PINCTL_OUT_EN))
pin_ctl &= ~(AC_PINCTL_IN_EN | AC_PINCTL_OUT_EN);
- snd_hda_codec_write(codec, nid, 0,
+ snd_hda_codec_write_cache(codec, nid, 0,
AC_VERB_SET_PIN_WIDGET_CONTROL,
pin_ctl | flag);
}
@@ -1880,7 +2388,7 @@ static void stac92xx_reset_pinctl(struct hda_codec *codec, hda_nid_t nid,
{
unsigned int pin_ctl = snd_hda_codec_read(codec, nid,
0, AC_VERB_GET_PIN_WIDGET_CONTROL, 0x00);
- snd_hda_codec_write(codec, nid, 0,
+ snd_hda_codec_write_cache(codec, nid, 0,
AC_VERB_SET_PIN_WIDGET_CONTROL,
pin_ctl & ~flag);
}
@@ -1936,22 +2444,22 @@ static void stac92xx_unsol_event(struct hda_codec *codec, unsigned int res)
}
}
-#ifdef CONFIG_PM
+#ifdef SND_HDA_NEEDS_RESUME
static int stac92xx_resume(struct hda_codec *codec)
{
struct sigmatel_spec *spec = codec->spec;
- int i;
- stac92xx_init(codec);
stac92xx_set_config_regs(codec);
- snd_hda_resume_ctls(codec, spec->mixer);
- for (i = 0; i < spec->num_mixers; i++)
- snd_hda_resume_ctls(codec, spec->mixers[i]);
- if (spec->multiout.dig_out_nid)
- snd_hda_resume_spdif_out(codec);
- if (spec->dig_in_nid)
- snd_hda_resume_spdif_in(codec);
-
+ snd_hda_sequence_write(codec, spec->init);
+ if (spec->gpio_mute) {
+ stac922x_gpio_mute(codec, 0, 0);
+ stac922x_gpio_mute(codec, 1, 0);
+ }
+ snd_hda_codec_resume_amp(codec);
+ snd_hda_codec_resume_cache(codec);
+ /* invoke unsolicited event to reset the HP state */
+ if (spec->hp_detect)
+ codec->patch_ops.unsol_event(codec, STAC_HP_EVENT << 26);
return 0;
}
#endif
@@ -1962,7 +2470,7 @@ static struct hda_codec_ops stac92xx_patch_ops = {
.init = stac92xx_init,
.free = stac92xx_free,
.unsol_event = stac92xx_unsol_event,
-#ifdef CONFIG_PM
+#ifdef SND_HDA_NEEDS_RESUME
.resume = stac92xx_resume,
#endif
};
@@ -2002,8 +2510,12 @@ static int patch_stac9200(struct hda_codec *codec)
spec->mux_nids = stac9200_mux_nids;
spec->num_muxes = 1;
spec->num_dmics = 0;
+ spec->num_adcs = 1;
- spec->init = stac9200_core_init;
+ if (spec->board_config == STAC_9200_GATEWAY)
+ spec->init = stac9200_eapd_init;
+ else
+ spec->init = stac9200_core_init;
spec->mixer = stac9200_mixer;
err = stac9200_parse_auto_config(codec);
@@ -2053,12 +2565,13 @@ static int patch_stac925x(struct hda_codec *codec)
spec->adc_nids = stac925x_adc_nids;
spec->mux_nids = stac925x_mux_nids;
spec->num_muxes = 1;
+ spec->num_adcs = 1;
switch (codec->vendor_id) {
case 0x83847632: /* STAC9202 */
case 0x83847633: /* STAC9202D */
case 0x83847636: /* STAC9251 */
case 0x83847637: /* STAC9251D */
- spec->num_dmics = 1;
+ spec->num_dmics = STAC925X_NUM_DMICS;
spec->dmic_nids = stac925x_dmic_nids;
break;
default:
@@ -2156,6 +2669,7 @@ static int patch_stac922x(struct hda_codec *codec)
spec->adc_nids = stac922x_adc_nids;
spec->mux_nids = stac922x_mux_nids;
spec->num_muxes = ARRAY_SIZE(stac922x_mux_nids);
+ spec->num_adcs = ARRAY_SIZE(stac922x_adc_nids);
spec->num_dmics = 0;
spec->init = stac922x_core_init;
@@ -2224,22 +2738,25 @@ static int patch_stac927x(struct hda_codec *codec)
spec->adc_nids = stac927x_adc_nids;
spec->mux_nids = stac927x_mux_nids;
spec->num_muxes = ARRAY_SIZE(stac927x_mux_nids);
+ spec->num_adcs = ARRAY_SIZE(stac927x_adc_nids);
spec->num_dmics = 0;
spec->init = d965_core_init;
- spec->mixer = stac9227_mixer;
+ spec->mixer = stac927x_mixer;
break;
case STAC_D965_5ST:
spec->adc_nids = stac927x_adc_nids;
spec->mux_nids = stac927x_mux_nids;
spec->num_muxes = ARRAY_SIZE(stac927x_mux_nids);
+ spec->num_adcs = ARRAY_SIZE(stac927x_adc_nids);
spec->num_dmics = 0;
spec->init = d965_core_init;
- spec->mixer = stac9227_mixer;
+ spec->mixer = stac927x_mixer;
break;
default:
spec->adc_nids = stac927x_adc_nids;
spec->mux_nids = stac927x_mux_nids;
spec->num_muxes = ARRAY_SIZE(stac927x_mux_nids);
+ spec->num_adcs = ARRAY_SIZE(stac927x_adc_nids);
spec->num_dmics = 0;
spec->init = stac927x_core_init;
spec->mixer = stac927x_mixer;
@@ -2247,7 +2764,8 @@ static int patch_stac927x(struct hda_codec *codec)
spec->multiout.dac_nids = spec->dac_nids;
/* GPIO0 High = Enable EAPD */
- stac92xx_enable_gpio_mask(codec, 0x00000001, 0x00000001);
+ spec->gpio_mask = spec->gpio_data = 0x00000001;
+ stac92xx_enable_gpio_mask(codec);
err = stac92xx_parse_auto_config(codec, 0x1e, 0x20);
if (!err) {
@@ -2272,7 +2790,7 @@ static int patch_stac927x(struct hda_codec *codec)
static int patch_stac9205(struct hda_codec *codec)
{
struct sigmatel_spec *spec;
- int err, gpio_mask, gpio_data;
+ int err;
spec = kzalloc(sizeof(*spec), GFP_KERNEL);
if (spec == NULL)
@@ -2299,10 +2817,11 @@ static int patch_stac9205(struct hda_codec *codec)
}
spec->adc_nids = stac9205_adc_nids;
+ spec->num_adcs = ARRAY_SIZE(stac9205_adc_nids);
spec->mux_nids = stac9205_mux_nids;
spec->num_muxes = ARRAY_SIZE(stac9205_mux_nids);
spec->dmic_nids = stac9205_dmic_nids;
- spec->num_dmics = ARRAY_SIZE(stac9205_dmic_nids);
+ spec->num_dmics = STAC9205_NUM_DMICS;
spec->dmux_nid = 0x1d;
spec->init = stac9205_core_init;
@@ -2310,20 +2829,25 @@ static int patch_stac9205(struct hda_codec *codec)
spec->multiout.dac_nids = spec->dac_nids;
- if (spec->board_config == STAC_M43xx) {
+ switch (spec->board_config){
+ case STAC_9205_DELL_M43:
/* Enable SPDIF in/out */
stac92xx_set_config_reg(codec, 0x1f, 0x01441030);
stac92xx_set_config_reg(codec, 0x20, 0x1c410030);
- gpio_mask = 0x00000007; /* GPIO0-2 */
+ spec->gpio_mask = 0x00000007; /* GPIO0-2 */
/* GPIO0 High = EAPD, GPIO1 Low = DRM,
* GPIO2 High = Headphone Mute
*/
- gpio_data = 0x00000005;
- } else
- gpio_mask = gpio_data = 0x00000001; /* GPIO0 High = EAPD */
+ spec->gpio_data = 0x00000005;
+ break;
+ default:
+ /* GPIO0 High = EAPD */
+ spec->gpio_mask = spec->gpio_data = 0x00000001;
+ break;
+ }
- stac92xx_enable_gpio_mask(codec, gpio_mask, gpio_data);
+ stac92xx_enable_gpio_mask(codec);
err = stac92xx_parse_auto_config(codec, 0x1f, 0x20);
if (!err) {
if (spec->board_config < 0) {
@@ -2355,7 +2879,7 @@ static hda_nid_t vaio_adcs[] = { 0x8 /*,0x6*/ };
static hda_nid_t vaio_mux_nids[] = { 0x15 };
static struct hda_input_mux vaio_mux = {
- .num_items = 2,
+ .num_items = 3,
.items = {
/* { "HP", 0x0 }, */
{ "Mic Jack", 0x1 },
@@ -2366,6 +2890,7 @@ static struct hda_input_mux vaio_mux = {
static struct hda_verb vaio_init[] = {
{0x0a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP }, /* HP <- 0x2 */
+ {0x0a, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | STAC_HP_EVENT},
{0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, /* Speaker <- 0x5 */
{0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, /* Mic? (<- 0x2) */
{0x0e, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN }, /* CD */
@@ -2397,61 +2922,28 @@ static struct hda_verb vaio_ar_init[] = {
};
/* bind volumes of both NID 0x02 and 0x05 */
-static int vaio_master_vol_put(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
- long *valp = ucontrol->value.integer.value;
- int change;
-
- change = snd_hda_codec_amp_update(codec, 0x02, 0, HDA_OUTPUT, 0,
- 0x7f, valp[0] & 0x7f);
- change |= snd_hda_codec_amp_update(codec, 0x02, 1, HDA_OUTPUT, 0,
- 0x7f, valp[1] & 0x7f);
- snd_hda_codec_amp_update(codec, 0x05, 0, HDA_OUTPUT, 0,
- 0x7f, valp[0] & 0x7f);
- snd_hda_codec_amp_update(codec, 0x05, 1, HDA_OUTPUT, 0,
- 0x7f, valp[1] & 0x7f);
- return change;
-}
+static struct hda_bind_ctls vaio_bind_master_vol = {
+ .ops = &snd_hda_bind_vol,
+ .values = {
+ HDA_COMPOSE_AMP_VAL(0x02, 3, 0, HDA_OUTPUT),
+ HDA_COMPOSE_AMP_VAL(0x05, 3, 0, HDA_OUTPUT),
+ 0
+ },
+};
/* bind volumes of both NID 0x02 and 0x05 */
-static int vaio_master_sw_put(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
- long *valp = ucontrol->value.integer.value;
- int change;
-
- change = snd_hda_codec_amp_update(codec, 0x02, 0, HDA_OUTPUT, 0,
- 0x80, (valp[0] ? 0 : 0x80));
- change |= snd_hda_codec_amp_update(codec, 0x02, 1, HDA_OUTPUT, 0,
- 0x80, (valp[1] ? 0 : 0x80));
- snd_hda_codec_amp_update(codec, 0x05, 0, HDA_OUTPUT, 0,
- 0x80, (valp[0] ? 0 : 0x80));
- snd_hda_codec_amp_update(codec, 0x05, 1, HDA_OUTPUT, 0,
- 0x80, (valp[1] ? 0 : 0x80));
- return change;
-}
+static struct hda_bind_ctls vaio_bind_master_sw = {
+ .ops = &snd_hda_bind_sw,
+ .values = {
+ HDA_COMPOSE_AMP_VAL(0x02, 3, 0, HDA_OUTPUT),
+ HDA_COMPOSE_AMP_VAL(0x05, 3, 0, HDA_OUTPUT),
+ 0,
+ },
+};
static struct snd_kcontrol_new vaio_mixer[] = {
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "Master Playback Volume",
- .info = snd_hda_mixer_amp_volume_info,
- .get = snd_hda_mixer_amp_volume_get,
- .put = vaio_master_vol_put,
- .tlv = { .c = snd_hda_mixer_amp_tlv },
- .private_value = HDA_COMPOSE_AMP_VAL(0x02, 3, 0, HDA_OUTPUT),
- },
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "Master Playback Switch",
- .info = snd_hda_mixer_amp_switch_info,
- .get = snd_hda_mixer_amp_switch_get,
- .put = vaio_master_sw_put,
- .private_value = HDA_COMPOSE_AMP_VAL(0x02, 3, 0, HDA_OUTPUT),
- },
+ HDA_BIND_VOL("Master Playback Volume", &vaio_bind_master_vol),
+ HDA_BIND_SW("Master Playback Switch", &vaio_bind_master_sw),
/* HDA_CODEC_VOLUME("CD Capture Volume", 0x07, 0, HDA_INPUT), */
HDA_CODEC_VOLUME("Capture Volume", 0x09, 0, HDA_INPUT),
HDA_CODEC_MUTE("Capture Switch", 0x09, 0, HDA_INPUT),
@@ -2467,22 +2959,8 @@ static struct snd_kcontrol_new vaio_mixer[] = {
};
static struct snd_kcontrol_new vaio_ar_mixer[] = {
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "Master Playback Volume",
- .info = snd_hda_mixer_amp_volume_info,
- .get = snd_hda_mixer_amp_volume_get,
- .put = vaio_master_vol_put,
- .private_value = HDA_COMPOSE_AMP_VAL(0x02, 3, 0, HDA_OUTPUT),
- },
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "Master Playback Switch",
- .info = snd_hda_mixer_amp_switch_info,
- .get = snd_hda_mixer_amp_switch_get,
- .put = vaio_master_sw_put,
- .private_value = HDA_COMPOSE_AMP_VAL(0x02, 3, 0, HDA_OUTPUT),
- },
+ HDA_BIND_VOL("Master Playback Volume", &vaio_bind_master_vol),
+ HDA_BIND_SW("Master Playback Switch", &vaio_bind_master_sw),
/* HDA_CODEC_VOLUME("CD Capture Volume", 0x07, 0, HDA_INPUT), */
HDA_CODEC_VOLUME("Capture Volume", 0x09, 0, HDA_INPUT),
HDA_CODEC_MUTE("Capture Switch", 0x09, 0, HDA_INPUT),
@@ -2504,6 +2982,49 @@ static struct hda_codec_ops stac9872_patch_ops = {
.build_pcms = stac92xx_build_pcms,
.init = stac92xx_init,
.free = stac92xx_free,
+#ifdef SND_HDA_NEEDS_RESUME
+ .resume = stac92xx_resume,
+#endif
+};
+
+static int stac9872_vaio_init(struct hda_codec *codec)
+{
+ int err;
+
+ err = stac92xx_init(codec);
+ if (err < 0)
+ return err;
+ if (codec->patch_ops.unsol_event)
+ codec->patch_ops.unsol_event(codec, STAC_HP_EVENT << 26);
+ return 0;
+}
+
+static void stac9872_vaio_hp_detect(struct hda_codec *codec, unsigned int res)
+{
+ if (get_pin_presence(codec, 0x0a)) {
+ stac92xx_reset_pinctl(codec, 0x0f, AC_PINCTL_OUT_EN);
+ stac92xx_set_pinctl(codec, 0x0a, AC_PINCTL_OUT_EN);
+ } else {
+ stac92xx_reset_pinctl(codec, 0x0a, AC_PINCTL_OUT_EN);
+ stac92xx_set_pinctl(codec, 0x0f, AC_PINCTL_OUT_EN);
+ }
+}
+
+static void stac9872_vaio_unsol_event(struct hda_codec *codec, unsigned int res)
+{
+ switch (res >> 26) {
+ case STAC_HP_EVENT:
+ stac9872_vaio_hp_detect(codec, res);
+ break;
+ }
+}
+
+static struct hda_codec_ops stac9872_vaio_patch_ops = {
+ .build_controls = stac92xx_build_controls,
+ .build_pcms = stac92xx_build_pcms,
+ .init = stac9872_vaio_init,
+ .free = stac92xx_free,
+ .unsol_event = stac9872_vaio_unsol_event,
#ifdef CONFIG_PM
.resume = stac92xx_resume,
#endif
@@ -2564,6 +3085,7 @@ static int patch_stac9872(struct hda_codec *codec)
spec->adc_nids = vaio_adcs;
spec->input_mux = &vaio_mux;
spec->mux_nids = vaio_mux_nids;
+ codec->patch_ops = stac9872_vaio_patch_ops;
break;
case CXD9872AKD_VAIO:
@@ -2577,10 +3099,10 @@ static int patch_stac9872(struct hda_codec *codec)
spec->adc_nids = vaio_adcs;
spec->input_mux = &vaio_mux;
spec->mux_nids = vaio_mux_nids;
+ codec->patch_ops = stac9872_patch_ops;
break;
}
- codec->patch_ops = stac9872_patch_ops;
return 0;
}
diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c
index ba32d1e52cb8..33b5e1ffa817 100644
--- a/sound/pci/hda/patch_via.c
+++ b/sound/pci/hda/patch_via.c
@@ -115,6 +115,10 @@ struct via_spec {
struct snd_kcontrol_new *kctl_alloc;
struct hda_input_mux private_imux;
hda_nid_t private_dac_nids[4];
+
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ struct hda_loopback_check loopback;
+#endif
};
static hda_nid_t vt1708_adc_nids[2] = {
@@ -305,15 +309,15 @@ static struct hda_verb vt1708_volume_init_verbs[] = {
{0x27, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- /* Unmute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback
+ /* Mute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback
* mixer widget
*/
/* Amp Indices: CD = 1, Mic1 = 2, Line = 3, Mic2 = 4 */
- {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
- {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)},
- {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)},
+ {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, /* master */
+ {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+ {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+ {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
/*
* Set up output mixers (0x19 - 0x1b)
@@ -543,24 +547,11 @@ static int via_init(struct hda_codec *codec)
return 0;
}
-#ifdef CONFIG_PM
-/*
- * resume
- */
-static int via_resume(struct hda_codec *codec)
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+static int via_check_power_status(struct hda_codec *codec, hda_nid_t nid)
{
struct via_spec *spec = codec->spec;
- int i;
-
- via_init(codec);
- for (i = 0; i < spec->num_mixers; i++)
- snd_hda_resume_ctls(codec, spec->mixers[i]);
- if (spec->multiout.dig_out_nid)
- snd_hda_resume_spdif_out(codec);
- if (spec->dig_in_nid)
- snd_hda_resume_spdif_in(codec);
-
- return 0;
+ return snd_hda_check_amp_list_power(codec, &spec->loopback, nid);
}
#endif
@@ -571,8 +562,8 @@ static struct hda_codec_ops via_patch_ops = {
.build_pcms = via_build_pcms,
.init = via_init,
.free = via_free,
-#ifdef CONFIG_PM
- .resume = via_resume,
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ .check_power_status = via_check_power_status,
#endif
};
@@ -762,6 +753,16 @@ static int vt1708_auto_create_analog_input_ctls(struct via_spec *spec,
return 0;
}
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+static struct hda_amp_list vt1708_loopbacks[] = {
+ { 0x17, HDA_INPUT, 1 },
+ { 0x17, HDA_INPUT, 2 },
+ { 0x17, HDA_INPUT, 3 },
+ { 0x17, HDA_INPUT, 4 },
+ { } /* end */
+};
+#endif
+
static int vt1708_parse_auto_config(struct hda_codec *codec)
{
struct via_spec *spec = codec->spec;
@@ -855,6 +856,9 @@ static int patch_vt1708(struct hda_codec *codec)
codec->patch_ops = via_patch_ops;
codec->patch_ops.init = via_auto_init;
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ spec->loopback.amplist = vt1708_loopbacks;
+#endif
return 0;
}
@@ -895,15 +899,15 @@ static struct hda_verb vt1709_10ch_volume_init_verbs[] = {
{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- /* Unmute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback
+ /* Mute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback
* mixer widget
*/
/* Amp Indices: AOW0=0, CD = 1, Mic1 = 2, Line = 3, Mic2 = 4 */
- {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
- {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)},
- {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)},
+ {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, /* unmute master */
+ {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+ {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+ {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
/*
* Set up output selector (0x1a, 0x1b, 0x29)
@@ -1251,6 +1255,16 @@ static int vt1709_parse_auto_config(struct hda_codec *codec)
return 1;
}
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+static struct hda_amp_list vt1709_loopbacks[] = {
+ { 0x18, HDA_INPUT, 1 },
+ { 0x18, HDA_INPUT, 2 },
+ { 0x18, HDA_INPUT, 3 },
+ { 0x18, HDA_INPUT, 4 },
+ { } /* end */
+};
+#endif
+
static int patch_vt1709_10ch(struct hda_codec *codec)
{
struct via_spec *spec;
@@ -1293,6 +1307,9 @@ static int patch_vt1709_10ch(struct hda_codec *codec)
codec->patch_ops = via_patch_ops;
codec->patch_ops.init = via_auto_init;
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ spec->loopback.amplist = vt1709_loopbacks;
+#endif
return 0;
}
@@ -1383,6 +1400,9 @@ static int patch_vt1709_6ch(struct hda_codec *codec)
codec->patch_ops = via_patch_ops;
codec->patch_ops.init = via_auto_init;
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ spec->loopback.amplist = vt1709_loopbacks;
+#endif
return 0;
}
diff --git a/sound/pci/ice1712/Makefile b/sound/pci/ice1712/Makefile
index 6efdd62f6837..65ce66adba5a 100644
--- a/sound/pci/ice1712/Makefile
+++ b/sound/pci/ice1712/Makefile
@@ -1,6 +1,6 @@
#
# Makefile for ALSA
-# Copyright (c) 2001 by Jaroslav Kysela <perex@suse.cz>
+# Copyright (c) 2001 by Jaroslav Kysela <perex@perex.cz>
#
snd-ice17xx-ak4xxx-objs := ak4xxx.o
diff --git a/sound/pci/ice1712/ak4xxx.c b/sound/pci/ice1712/ak4xxx.c
index ab00cce2c39f..a1aba0d7d0e4 100644
--- a/sound/pci/ice1712/ak4xxx.c
+++ b/sound/pci/ice1712/ak4xxx.c
@@ -3,7 +3,7 @@
*
* AK4524 / AK4528 / AK4529 / AK4355 / AK4381 interface
*
- * Copyright (c) 2000 Jaroslav Kysela <perex@suse.cz>
+ * Copyright (c) 2000 Jaroslav Kysela <perex@perex.cz>
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
@@ -30,7 +30,7 @@
#include <sound/initval.h>
#include "ice1712.h"
-MODULE_AUTHOR("Jaroslav Kysela <perex@suse.cz>");
+MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
MODULE_DESCRIPTION("ICEnsemble ICE17xx <-> AK4xxx AD/DA chip interface");
MODULE_LICENSE("GPL");
diff --git a/sound/pci/ice1712/amp.c b/sound/pci/ice1712/amp.c
index 44bbb630b949..6e13d758bb5d 100644
--- a/sound/pci/ice1712/amp.c
+++ b/sound/pci/ice1712/amp.c
@@ -3,7 +3,7 @@
*
* Lowlevel functions for Advanced Micro Peripherals Ltd AUDIO2000
*
- * Copyright (c) 2000 Jaroslav Kysela <perex@suse.cz>
+ * Copyright (c) 2000 Jaroslav Kysela <perex@perex.cz>
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
diff --git a/sound/pci/ice1712/amp.h b/sound/pci/ice1712/amp.h
index a0fc89b48122..bf81d30d9150 100644
--- a/sound/pci/ice1712/amp.h
+++ b/sound/pci/ice1712/amp.h
@@ -6,7 +6,7 @@
*
* Lowlevel functions for Advanced Micro Peripherals Ltd AUDIO2000
*
- * Copyright (c) 2000 Jaroslav Kysela <perex@suse.cz>
+ * Copyright (c) 2000 Jaroslav Kysela <perex@perex.cz>
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
diff --git a/sound/pci/ice1712/aureon.c b/sound/pci/ice1712/aureon.c
index 66bacde1ead3..ec0699c89952 100644
--- a/sound/pci/ice1712/aureon.c
+++ b/sound/pci/ice1712/aureon.c
@@ -394,7 +394,7 @@ static int aureon_ac97_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_ele
/*
* AC'97 mute controls
*/
-#define aureon_ac97_mute_info aureon_mono_bool_info
+#define aureon_ac97_mute_info snd_ctl_boolean_mono_info
static int aureon_ac97_mute_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
{
@@ -430,7 +430,7 @@ static int aureon_ac97_mute_put(struct snd_kcontrol *kcontrol, struct snd_ctl_el
/*
* AC'97 mute controls
*/
-#define aureon_ac97_micboost_info aureon_mono_bool_info
+#define aureon_ac97_micboost_info snd_ctl_boolean_mono_info
static int aureon_ac97_micboost_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
{
@@ -621,19 +621,12 @@ static void wm_put(struct snd_ice1712 *ice, int reg, unsigned short val)
/*
*/
-static int aureon_mono_bool_info(struct snd_kcontrol *k, struct snd_ctl_elem_info *uinfo)
-{
- uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
- uinfo->count = 1;
- uinfo->value.integer.min = 0;
- uinfo->value.integer.max = 1;
- return 0;
-}
+#define aureon_mono_bool_info snd_ctl_boolean_mono_info
/*
* AC'97 master playback mute controls (Mute on WM8770 chip)
*/
-#define aureon_ac97_mmute_info aureon_mono_bool_info
+#define aureon_ac97_mmute_info snd_ctl_boolean_mono_info
static int aureon_ac97_mmute_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
{
@@ -708,7 +701,7 @@ static void wm_set_vol(struct snd_ice1712 *ice, unsigned int index, unsigned sho
/*
* DAC mute control
*/
-#define wm_pcm_mute_info aureon_mono_bool_info
+#define wm_pcm_mute_info snd_ctl_boolean_mono_info
static int wm_pcm_mute_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
{
@@ -879,13 +872,7 @@ static int wm_mute_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value
/*
* WM8770 master mute control
*/
-static int wm_master_mute_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) {
- uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
- uinfo->count = 2;
- uinfo->value.integer.min = 0;
- uinfo->value.integer.max = 1;
- return 0;
-}
+#define wm_master_mute_info snd_ctl_boolean_stereo_info
static int wm_master_mute_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
{
@@ -969,14 +956,7 @@ static int wm_pcm_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_val
/*
* ADC mute control
*/
-static int wm_adc_mute_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
-{
- uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
- uinfo->count = 2;
- uinfo->value.integer.min = 0;
- uinfo->value.integer.max = 1;
- return 0;
-}
+#define wm_adc_mute_info snd_ctl_boolean_stereo_info
static int wm_adc_mute_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
{
@@ -1210,12 +1190,7 @@ static int aureon_cs8415_rate_get (struct snd_kcontrol *kcontrol, struct snd_ctl
/*
* CS8415A Mute
*/
-static int aureon_cs8415_mute_info (struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
-{
- uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
- uinfo->count = 1;
- return 0;
-}
+#define aureon_cs8415_mute_info snd_ctl_boolean_mono_info
static int aureon_cs8415_mute_get (struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
{
@@ -1316,7 +1291,7 @@ static int aureon_get_headphone_amp(struct snd_ice1712 *ice)
return ( tmp & AUREON_HP_SEL )!= 0;
}
-#define aureon_hpamp_info aureon_mono_bool_info
+#define aureon_hpamp_info snd_ctl_boolean_mono_info
static int aureon_hpamp_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
{
@@ -1338,7 +1313,7 @@ static int aureon_hpamp_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_v
* Deemphasis
*/
-#define aureon_deemp_info aureon_mono_bool_info
+#define aureon_deemp_info snd_ctl_boolean_mono_info
static int aureon_deemp_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
{
diff --git a/sound/pci/ice1712/delta.c b/sound/pci/ice1712/delta.c
index af659800c9b0..371f78461db4 100644
--- a/sound/pci/ice1712/delta.c
+++ b/sound/pci/ice1712/delta.c
@@ -4,7 +4,7 @@
* Lowlevel functions for M-Audio Delta 1010, 44, 66, Dio2496, Audiophile
* Digigram VX442
*
- * Copyright (c) 2000 Jaroslav Kysela <perex@suse.cz>
+ * Copyright (c) 2000 Jaroslav Kysela <perex@perex.cz>
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
@@ -393,15 +393,8 @@ static void delta_setup_spdif(struct snd_ice1712 *ice, int rate)
snd_ice1712_delta_cs8403_spdif_write(ice, tmp);
}
-static int snd_ice1712_delta1010lt_wordclock_status_info(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_info *uinfo)
-{
- uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
- uinfo->count = 1;
- uinfo->value.integer.min = 0;
- uinfo->value.integer.max = 1;
- return 0;
-}
+#define snd_ice1712_delta1010lt_wordclock_status_info \
+ snd_ctl_boolean_mono_info
static int snd_ice1712_delta1010lt_wordclock_status_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
diff --git a/sound/pci/ice1712/delta.h b/sound/pci/ice1712/delta.h
index 2697156607e4..26ea05a32f56 100644
--- a/sound/pci/ice1712/delta.h
+++ b/sound/pci/ice1712/delta.h
@@ -7,7 +7,7 @@
* Lowlevel functions for M-Audio Delta 1010, 44, 66, Dio2496, Audiophile
* Digigram VX442
*
- * Copyright (c) 2000 Jaroslav Kysela <perex@suse.cz>
+ * Copyright (c) 2000 Jaroslav Kysela <perex@perex.cz>
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
diff --git a/sound/pci/ice1712/envy24ht.h b/sound/pci/ice1712/envy24ht.h
index b58afcda9ed6..43b9e3e858be 100644
--- a/sound/pci/ice1712/envy24ht.h
+++ b/sound/pci/ice1712/envy24ht.h
@@ -4,7 +4,7 @@
/*
* ALSA driver for ICEnsemble VT1724 (Envy24)
*
- * Copyright (c) 2000 Jaroslav Kysela <perex@suse.cz>
+ * Copyright (c) 2000 Jaroslav Kysela <perex@perex.cz>
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
diff --git a/sound/pci/ice1712/ews.c b/sound/pci/ice1712/ews.c
index b135389fec6c..75e4e5e0f1e4 100644
--- a/sound/pci/ice1712/ews.c
+++ b/sound/pci/ice1712/ews.c
@@ -3,7 +3,7 @@
*
* Lowlevel functions for Terratec EWS88MT/D, EWX24/96, DMX 6Fire
*
- * Copyright (c) 2000 Jaroslav Kysela <perex@suse.cz>
+ * Copyright (c) 2000 Jaroslav Kysela <perex@perex.cz>
* 2002 Takashi Iwai <tiwai@suse.de>
*
* This program is free software; you can redistribute it and/or modify
@@ -700,14 +700,7 @@ static struct snd_kcontrol_new snd_ice1712_ews88mt_output_sense __devinitdata =
* EWS88D specific controls
*/
-static int snd_ice1712_ews88d_control_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
-{
- uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
- uinfo->count = 1;
- uinfo->value.integer.min = 0;
- uinfo->value.integer.max = 1;
- return 0;
-}
+#define snd_ice1712_ews88d_control_info snd_ctl_boolean_mono_info
static int snd_ice1712_ews88d_control_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
{
@@ -812,14 +805,7 @@ static int snd_ice1712_6fire_write_pca(struct snd_ice1712 *ice, unsigned char re
return 0;
}
-static int snd_ice1712_6fire_control_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
-{
- uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
- uinfo->count = 1;
- uinfo->value.integer.min = 0;
- uinfo->value.integer.max = 1;
- return 0;
-}
+#define snd_ice1712_6fire_control_info snd_ctl_boolean_mono_info
static int snd_ice1712_6fire_control_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
{
diff --git a/sound/pci/ice1712/ews.h b/sound/pci/ice1712/ews.h
index a12a0b053558..e4ed1b475b08 100644
--- a/sound/pci/ice1712/ews.h
+++ b/sound/pci/ice1712/ews.h
@@ -6,7 +6,7 @@
*
* Lowlevel functions for Terratec EWS88MT/D, EWX24/96, DMX 6Fire
*
- * Copyright (c) 2000 Jaroslav Kysela <perex@suse.cz>
+ * Copyright (c) 2000 Jaroslav Kysela <perex@perex.cz>
* 2002 Takashi Iwai <tiwai@suse.de>
*
* This program is free software; you can redistribute it and/or modify
diff --git a/sound/pci/ice1712/hoontech.c b/sound/pci/ice1712/hoontech.c
index 8203562ef7e7..abcfd1da6587 100644
--- a/sound/pci/ice1712/hoontech.c
+++ b/sound/pci/ice1712/hoontech.c
@@ -3,7 +3,7 @@
*
* Lowlevel functions for Hoontech STDSP24
*
- * Copyright (c) 2000 Jaroslav Kysela <perex@suse.cz>
+ * Copyright (c) 2000 Jaroslav Kysela <perex@perex.cz>
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
diff --git a/sound/pci/ice1712/hoontech.h b/sound/pci/ice1712/hoontech.h
index 1ee538b20fbf..cc1da1e69ad1 100644
--- a/sound/pci/ice1712/hoontech.h
+++ b/sound/pci/ice1712/hoontech.h
@@ -6,7 +6,7 @@
*
* Lowlevel functions for Hoontech STDSP24
*
- * Copyright (c) 2000 Jaroslav Kysela <perex@suse.cz>
+ * Copyright (c) 2000 Jaroslav Kysela <perex@perex.cz>
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
diff --git a/sound/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c
index 6630a0ae9527..052fc3cb3272 100644
--- a/sound/pci/ice1712/ice1712.c
+++ b/sound/pci/ice1712/ice1712.c
@@ -1,7 +1,7 @@
/*
* ALSA driver for ICEnsemble ICE1712 (Envy24)
*
- * Copyright (c) 2000 Jaroslav Kysela <perex@suse.cz>
+ * Copyright (c) 2000 Jaroslav Kysela <perex@perex.cz>
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
@@ -73,7 +73,7 @@
#include "ews.h"
#include "hoontech.h"
-MODULE_AUTHOR("Jaroslav Kysela <perex@suse.cz>");
+MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
MODULE_DESCRIPTION("ICEnsemble ICE1712 (Envy24)");
MODULE_LICENSE("GPL");
MODULE_SUPPORTED_DEVICE("{"
@@ -256,14 +256,7 @@ static unsigned short snd_ice1712_pro_ac97_read(struct snd_ac97 *ac97,
/*
* consumer ac97 digital mix
*/
-static int snd_ice1712_digmix_route_ac97_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
-{
- uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
- uinfo->count = 1;
- uinfo->value.integer.min = 0;
- uinfo->value.integer.max = 1;
- return 0;
-}
+#define snd_ice1712_digmix_route_ac97_info snd_ctl_boolean_mono_info
static int snd_ice1712_digmix_route_ac97_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
{
@@ -1300,14 +1293,7 @@ static void snd_ice1712_update_volume(struct snd_ice1712 *ice, int index)
outw(val, ICEMT(ice, MONITOR_VOLUME));
}
-static int snd_ice1712_pro_mixer_switch_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
-{
- uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
- uinfo->count = 2;
- uinfo->value.integer.min = 0;
- uinfo->value.integer.max = 1;
- return 0;
-}
+#define snd_ice1712_pro_mixer_switch_info snd_ctl_boolean_stereo_info
static int snd_ice1712_pro_mixer_switch_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
{
@@ -1759,16 +1745,6 @@ static struct snd_kcontrol_new snd_ice1712_spdif_stream __devinitdata =
.put = snd_ice1712_spdif_stream_put
};
-int snd_ice1712_gpio_info(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_info *uinfo)
-{
- uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
- uinfo->count = 1;
- uinfo->value.integer.min = 0;
- uinfo->value.integer.max = 1;
- return 0;
-}
-
int snd_ice1712_gpio_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
@@ -1968,15 +1944,7 @@ static struct snd_kcontrol_new snd_ice1712_pro_internal_clock_default __devinitd
.put = snd_ice1712_pro_internal_clock_default_put
};
-static int snd_ice1712_pro_rate_locking_info(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_info *uinfo)
-{
- uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
- uinfo->count = 1;
- uinfo->value.integer.min = 0;
- uinfo->value.integer.max = 1;
- return 0;
-}
+#define snd_ice1712_pro_rate_locking_info snd_ctl_boolean_mono_info
static int snd_ice1712_pro_rate_locking_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
@@ -2007,15 +1975,7 @@ static struct snd_kcontrol_new snd_ice1712_pro_rate_locking __devinitdata = {
.put = snd_ice1712_pro_rate_locking_put
};
-static int snd_ice1712_pro_rate_reset_info(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_info *uinfo)
-{
- uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
- uinfo->count = 1;
- uinfo->value.integer.min = 0;
- uinfo->value.integer.max = 1;
- return 0;
-}
+#define snd_ice1712_pro_rate_reset_info snd_ctl_boolean_mono_info
static int snd_ice1712_pro_rate_reset_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
diff --git a/sound/pci/ice1712/ice1712.h b/sound/pci/ice1712/ice1712.h
index 6ac486d9c138..58640afa5404 100644
--- a/sound/pci/ice1712/ice1712.h
+++ b/sound/pci/ice1712/ice1712.h
@@ -4,7 +4,7 @@
/*
* ALSA driver for ICEnsemble ICE1712 (Envy24)
*
- * Copyright (c) 2000 Jaroslav Kysela <perex@suse.cz>
+ * Copyright (c) 2000 Jaroslav Kysela <perex@perex.cz>
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
@@ -451,11 +451,10 @@ static inline void snd_ice1712_restore_gpio_status(struct snd_ice1712 *ice)
/* for bit controls */
#define ICE1712_GPIO(xiface, xname, xindex, mask, invert, xaccess) \
-{ .iface = xiface, .name = xname, .access = xaccess, .info = snd_ice1712_gpio_info, \
+{ .iface = xiface, .name = xname, .access = xaccess, .info = snd_ctl_boolean_mono_info, \
.get = snd_ice1712_gpio_get, .put = snd_ice1712_gpio_put, \
.private_value = mask | (invert << 24) }
-int snd_ice1712_gpio_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo);
int snd_ice1712_gpio_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol);
int snd_ice1712_gpio_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol);
diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c
index ee620dea7ef3..0b0bbb0d96b9 100644
--- a/sound/pci/ice1712/ice1724.c
+++ b/sound/pci/ice1712/ice1724.c
@@ -2,7 +2,7 @@
* ALSA driver for VT1724 ICEnsemble ICE1724 / VIA VT1724 (Envy24HT)
* VIA VT1720 (Envy24PT)
*
- * Copyright (c) 2000 Jaroslav Kysela <perex@suse.cz>
+ * Copyright (c) 2000 Jaroslav Kysela <perex@perex.cz>
* 2002 James Stafford <jstafford@ampltd.com>
* 2003 Takashi Iwai <tiwai@suse.de>
*
@@ -52,7 +52,7 @@
#include "phase.h"
#include "wtm.h"
-MODULE_AUTHOR("Jaroslav Kysela <perex@suse.cz>");
+MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
MODULE_DESCRIPTION("VIA ICEnsemble ICE1724/1720 (Envy24HT/PT)");
MODULE_LICENSE("GPL");
MODULE_SUPPORTED_DEVICE("{"
@@ -341,10 +341,12 @@ static int snd_vt1724_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
what = 0;
snd_pcm_group_for_each_entry(s, substream) {
- const struct vt1724_pcm_reg *reg;
- reg = s->runtime->private_data;
- what |= reg->start;
- snd_pcm_trigger_done(s, substream);
+ if (snd_pcm_substream_chip(s) == ice) {
+ const struct vt1724_pcm_reg *reg;
+ reg = s->runtime->private_data;
+ what |= reg->start;
+ snd_pcm_trigger_done(s, substream);
+ }
}
switch (cmd) {
@@ -1479,15 +1481,7 @@ static struct snd_kcontrol_new snd_vt1724_spdif_maskp __devinitdata =
.get = snd_vt1724_spdif_maskp_get,
};
-static int snd_vt1724_spdif_sw_info(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_info *uinfo)
-{
- uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
- uinfo->count = 1;
- uinfo->value.integer.min = 0;
- uinfo->value.integer.max = 1;
- return 0;
-}
+#define snd_vt1724_spdif_sw_info snd_ctl_boolean_mono_info
static int snd_vt1724_spdif_sw_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
@@ -1532,15 +1526,7 @@ static struct snd_kcontrol_new snd_vt1724_spdif_switch __devinitdata =
* GPIO access from extern
*/
-int snd_vt1724_gpio_info(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_info *uinfo)
-{
- uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
- uinfo->count = 1;
- uinfo->value.integer.min = 0;
- uinfo->value.integer.max = 1;
- return 0;
-}
+#define snd_vt1724_gpio_info snd_ctl_boolean_mono_info
int snd_vt1724_gpio_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
@@ -1706,15 +1692,7 @@ static struct snd_kcontrol_new snd_vt1724_pro_internal_clock __devinitdata = {
.put = snd_vt1724_pro_internal_clock_put
};
-static int snd_vt1724_pro_rate_locking_info(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_info *uinfo)
-{
- uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
- uinfo->count = 1;
- uinfo->value.integer.min = 0;
- uinfo->value.integer.max = 1;
- return 0;
-}
+#define snd_vt1724_pro_rate_locking_info snd_ctl_boolean_mono_info
static int snd_vt1724_pro_rate_locking_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
@@ -1745,15 +1723,7 @@ static struct snd_kcontrol_new snd_vt1724_pro_rate_locking __devinitdata = {
.put = snd_vt1724_pro_rate_locking_put
};
-static int snd_vt1724_pro_rate_reset_info(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_info *uinfo)
-{
- uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
- uinfo->count = 1;
- uinfo->value.integer.min = 0;
- uinfo->value.integer.max = 1;
- return 0;
-}
+#define snd_vt1724_pro_rate_reset_info snd_ctl_boolean_mono_info
static int snd_vt1724_pro_rate_reset_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
diff --git a/sound/pci/ice1712/juli.c b/sound/pci/ice1712/juli.c
index 3d8e74e493d7..1fbe3ef8e60a 100644
--- a/sound/pci/ice1712/juli.c
+++ b/sound/pci/ice1712/juli.c
@@ -3,7 +3,7 @@
*
* Lowlevel functions for ESI Juli@ cards
*
- * Copyright (c) 2004 Jaroslav Kysela <perex@suse.cz>
+ * Copyright (c) 2004 Jaroslav Kysela <perex@perex.cz>
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
diff --git a/sound/pci/ice1712/phase.c b/sound/pci/ice1712/phase.c
index 40a9098af777..3ac25058bb58 100644
--- a/sound/pci/ice1712/phase.c
+++ b/sound/pci/ice1712/phase.c
@@ -270,7 +270,7 @@ static void wm_set_vol(struct snd_ice1712 *ice, unsigned int index, unsigned sho
/*
* DAC mute control
*/
-#define wm_pcm_mute_info phase28_mono_bool_info
+#define wm_pcm_mute_info snd_ctl_boolean_mono_info
static int wm_pcm_mute_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
{
@@ -527,13 +527,7 @@ static int wm_mute_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value
/*
* WM8770 master mute control
*/
-static int wm_master_mute_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) {
- uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
- uinfo->count = 2;
- uinfo->value.integer.min = 0;
- uinfo->value.integer.max = 1;
- return 0;
-}
+#define wm_master_mute_info snd_ctl_boolean_stereo_info
static int wm_master_mute_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
{
@@ -615,20 +609,9 @@ static int wm_pcm_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_val
}
/*
- */
-static int phase28_mono_bool_info(struct snd_kcontrol *k, struct snd_ctl_elem_info *uinfo)
-{
- uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
- uinfo->count = 1;
- uinfo->value.integer.min = 0;
- uinfo->value.integer.max = 1;
- return 0;
-}
-
-/*
* Deemphasis
*/
-#define phase28_deemp_info phase28_mono_bool_info
+#define phase28_deemp_info snd_ctl_boolean_mono_info
static int phase28_deemp_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
{
diff --git a/sound/pci/ice1712/pontis.c b/sound/pci/ice1712/pontis.c
index 01c69453ddeb..faefd52c1b80 100644
--- a/sound/pci/ice1712/pontis.c
+++ b/sound/pci/ice1712/pontis.c
@@ -216,14 +216,7 @@ static int wm_adc_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_val
/*
* ADC input mux mixer control
*/
-static int wm_adc_mux_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
-{
- uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
- uinfo->count = 1;
- uinfo->value.integer.min = 0;
- uinfo->value.integer.max = 1;
- return 0;
-}
+#define wm_adc_mux_info snd_ctl_boolean_mono_info
static int wm_adc_mux_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
{
@@ -260,14 +253,7 @@ static int wm_adc_mux_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_val
/*
* Analog bypass (In -> Out)
*/
-static int wm_bypass_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
-{
- uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
- uinfo->count = 1;
- uinfo->value.integer.min = 0;
- uinfo->value.integer.max = 1;
- return 0;
-}
+#define wm_bypass_info snd_ctl_boolean_mono_info
static int wm_bypass_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
{
@@ -302,14 +288,7 @@ static int wm_bypass_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_valu
/*
* Left/Right swap
*/
-static int wm_chswap_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
-{
- uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
- uinfo->count = 1;
- uinfo->value.integer.min = 0;
- uinfo->value.integer.max = 1;
- return 0;
-}
+#define wm_chswap_info snd_ctl_boolean_mono_info
static int wm_chswap_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
{
diff --git a/sound/pci/ice1712/prodigy192.c b/sound/pci/ice1712/prodigy192.c
index 4bae7305a79b..4180f9739ecb 100644
--- a/sound/pci/ice1712/prodigy192.c
+++ b/sound/pci/ice1712/prodigy192.c
@@ -81,14 +81,7 @@ static inline unsigned char stac9460_get(struct snd_ice1712 *ice, int reg)
/*
* DAC mute control
*/
-static int stac9460_dac_mute_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
-{
- uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
- uinfo->count = 1;
- uinfo->value.integer.min = 0;
- uinfo->value.integer.max = 1;
- return 0;
-}
+#define stac9460_dac_mute_info snd_ctl_boolean_mono_info
static int stac9460_dac_mute_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
{
@@ -177,14 +170,7 @@ static int stac9460_dac_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_el
/*
* ADC mute control
*/
-static int stac9460_adc_mute_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
-{
- uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
- uinfo->count = 2;
- uinfo->value.integer.min = 0;
- uinfo->value.integer.max = 1;
- return 0;
-}
+#define stac9460_adc_mute_info snd_ctl_boolean_stereo_info
static int stac9460_adc_mute_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
{
@@ -292,14 +278,7 @@ static int aureon_get_headphone_amp(struct snd_ice1712 *ice)
return ( tmp & AUREON_HP_SEL )!= 0;
}
-static int aureon_bool_info(struct snd_kcontrol *k, struct snd_ctl_elem_info *uinfo)
-{
- uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
- uinfo->count = 1;
- uinfo->value.integer.min = 0;
- uinfo->value.integer.max = 1;
- return 0;
-}
+#define aureon_bool_info snd_ctl_boolean_mono_info
static int aureon_hpamp_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
{
diff --git a/sound/pci/ice1712/wtm.c b/sound/pci/ice1712/wtm.c
index 04e535c8542b..7fcce0a506d6 100644
--- a/sound/pci/ice1712/wtm.c
+++ b/sound/pci/ice1712/wtm.c
@@ -71,14 +71,7 @@ static inline unsigned char stac9460_2_get(struct snd_ice1712 *ice, int reg)
/*
* DAC mute control
*/
-static int stac9460_dac_mute_info(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_info *uinfo)
-{
- uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
- uinfo->count = 1;
- uinfo->value.integer.min = 0;
- return 0;
-}
+#define stac9460_dac_mute_info snd_ctl_boolean_mono_info
static int stac9460_dac_mute_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
@@ -218,15 +211,7 @@ static int stac9460_dac_vol_put(struct snd_kcontrol *kcontrol,
/*
* ADC mute control
*/
-static int stac9460_adc_mute_info(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_info *uinfo)
-{
- uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
- uinfo->count = 2;
- uinfo->value.integer.min = 0;
- uinfo->value.integer.max = 1;
- return 0;
-}
+#define stac9460_adc_mute_info snd_ctl_boolean_stereo_info
static int stac9460_adc_mute_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
@@ -357,15 +342,7 @@ static int stac9460_adc_vol_put(struct snd_kcontrol *kcontrol,
* MIC / LINE switch fonction
*/
-static int stac9460_mic_sw_info(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_info *uinfo)
-{
- uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
- uinfo->count = 1;
- uinfo->value.integer.min = 0;
- uinfo->value.integer.max = 1;
- return 0;
-}
+#define stac9460_mic_sw_info snd_ctl_boolean_mono_info
static int stac9460_mic_sw_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c
index da9734073dba..b4a38a3d855b 100644
--- a/sound/pci/intel8x0.c
+++ b/sound/pci/intel8x0.c
@@ -1,7 +1,7 @@
/*
* ALSA driver for Intel ICH (i8x0) chipsets
*
- * Copyright (c) 2000 Jaroslav Kysela <perex@suse.cz>
+ * Copyright (c) 2000 Jaroslav Kysela <perex@perex.cz>
*
*
* This code also contains alpha support for SiS 735 chipsets provided
@@ -43,7 +43,7 @@
#include <asm/pgtable.h>
#include <asm/cacheflush.h>
-MODULE_AUTHOR("Jaroslav Kysela <perex@suse.cz>");
+MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
MODULE_DESCRIPTION("Intel 82801AA,82901AB,i810,i820,i830,i840,i845,MX440; SiS 7012; Ali 5455");
MODULE_LICENSE("GPL");
MODULE_SUPPORTED_DEVICE("{{Intel,82801AA-ICH},"
diff --git a/sound/pci/intel8x0m.c b/sound/pci/intel8x0m.c
index c155e1f3a0e5..fad806e60f36 100644
--- a/sound/pci/intel8x0m.c
+++ b/sound/pci/intel8x0m.c
@@ -1,7 +1,7 @@
/*
* ALSA modem driver for Intel ICH (i8x0) chipsets
*
- * Copyright (c) 2000 Jaroslav Kysela <perex@suse.cz>
+ * Copyright (c) 2000 Jaroslav Kysela <perex@perex.cz>
*
* This is modified (by Sasha Khapyorsky <sashak@alsa-project.org>) version
* of ALSA ICH sound driver intel8x0.c .
@@ -37,7 +37,7 @@
#include <sound/info.h>
#include <sound/initval.h>
-MODULE_AUTHOR("Jaroslav Kysela <perex@suse.cz>");
+MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
MODULE_DESCRIPTION("Intel 82801AA,82901AB,i810,i820,i830,i840,i845,MX440; "
"SiS 7013; NVidia MCP/2/2S/3 modems");
MODULE_LICENSE("GPL");
diff --git a/sound/pci/korg1212/Makefile b/sound/pci/korg1212/Makefile
index 78c9dc6eeb2d..f11ce1b1b3d4 100644
--- a/sound/pci/korg1212/Makefile
+++ b/sound/pci/korg1212/Makefile
@@ -1,6 +1,6 @@
#
# Makefile for ALSA
-# Copyright (c) 2001 by Jaroslav Kysela <perex@suse.cz>
+# Copyright (c) 2001 by Jaroslav Kysela <perex@perex.cz>
#
snd-korg1212-objs := korg1212.o
diff --git a/sound/pci/korg1212/korg1212.c b/sound/pci/korg1212/korg1212.c
index 5338243fb035..c4af57fb5af1 100644
--- a/sound/pci/korg1212/korg1212.c
+++ b/sound/pci/korg1212/korg1212.c
@@ -1391,8 +1391,6 @@ static int snd_korg1212_playback_open(struct snd_pcm_substream *substream)
K1212_DEBUG_PRINTK("K1212_DEBUG: snd_korg1212_playback_open [%s]\n",
stateName[korg1212->cardState]);
- snd_pcm_set_sync(substream); // ???
-
snd_korg1212_OpenCard(korg1212);
runtime->hw = snd_korg1212_playback_info;
@@ -1422,8 +1420,6 @@ static int snd_korg1212_capture_open(struct snd_pcm_substream *substream)
K1212_DEBUG_PRINTK("K1212_DEBUG: snd_korg1212_capture_open [%s]\n",
stateName[korg1212->cardState]);
- snd_pcm_set_sync(substream);
-
snd_korg1212_OpenCard(korg1212);
runtime->hw = snd_korg1212_capture_info;
diff --git a/sound/pci/maestro3.c b/sound/pci/maestro3.c
index 8a5ff1cb5362..32245770595e 100644
--- a/sound/pci/maestro3.c
+++ b/sound/pci/maestro3.c
@@ -1821,7 +1821,6 @@ snd_m3_playback_open(struct snd_pcm_substream *subs)
return err;
runtime->hw = snd_m3_playback;
- snd_pcm_set_sync(subs);
return 0;
}
@@ -1846,7 +1845,6 @@ snd_m3_capture_open(struct snd_pcm_substream *subs)
return err;
runtime->hw = snd_m3_capture;
- snd_pcm_set_sync(subs);
return 0;
}
diff --git a/sound/pci/mixart/Makefile b/sound/pci/mixart/Makefile
index fe6ba0c4b567..cce159ec5624 100644
--- a/sound/pci/mixart/Makefile
+++ b/sound/pci/mixart/Makefile
@@ -1,6 +1,6 @@
#
# Makefile for ALSA
-# Copyright (c) 2001 by Jaroslav Kysela <perex@suse.cz>
+# Copyright (c) 2001 by Jaroslav Kysela <perex@perex.cz>
#
snd-mixart-objs := mixart.o mixart_core.o mixart_hwdep.o mixart_mixer.o
diff --git a/sound/pci/mixart/mixart.c b/sound/pci/mixart/mixart.c
index ac007cec0879..880b824e24cd 100644
--- a/sound/pci/mixart/mixart.c
+++ b/sound/pci/mixart/mixart.c
@@ -652,7 +652,7 @@ static int snd_mixart_hw_free(struct snd_pcm_substream *subs)
static struct snd_pcm_hardware snd_mixart_analog_caps =
{
.info = ( SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED |
- SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_SYNC_START |
+ SNDRV_PCM_INFO_MMAP_VALID |
SNDRV_PCM_INFO_PAUSE),
.formats = ( SNDRV_PCM_FMTBIT_U8 |
SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE |
@@ -673,7 +673,7 @@ static struct snd_pcm_hardware snd_mixart_analog_caps =
static struct snd_pcm_hardware snd_mixart_digital_caps =
{
.info = ( SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED |
- SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_SYNC_START |
+ SNDRV_PCM_INFO_MMAP_VALID |
SNDRV_PCM_INFO_PAUSE),
.formats = ( SNDRV_PCM_FMTBIT_U8 |
SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE |
@@ -1317,6 +1317,12 @@ static int __devinit snd_mixart_probe(struct pci_dev *pci,
mgr->mem[i].phys = pci_resource_start(pci, i);
mgr->mem[i].virt = ioremap_nocache(mgr->mem[i].phys,
pci_resource_len(pci, i));
+ if (!mgr->mem[i].virt) {
+ printk(KERN_ERR "unable to remap resource 0x%lx\n",
+ mgr->mem[i].phys);
+ snd_mixart_free(mgr);
+ return -EBUSY;
+ }
}
if (request_irq(pci->irq, snd_mixart_interrupt, IRQF_SHARED,
diff --git a/sound/pci/mixart/mixart_mixer.c b/sound/pci/mixart/mixart_mixer.c
index d7d15c036e02..0e16512d25f7 100644
--- a/sound/pci/mixart/mixart_mixer.c
+++ b/sound/pci/mixart/mixart_mixer.c
@@ -403,14 +403,7 @@ static struct snd_kcontrol_new mixart_control_analog_level = {
};
/* shared */
-static int mixart_sw_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
-{
- uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
- uinfo->count = 2;
- uinfo->value.integer.min = 0;
- uinfo->value.integer.max = 1;
- return 0;
-}
+#define mixart_sw_info snd_ctl_boolean_stereo_info
static int mixart_audio_sw_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
{
diff --git a/sound/pci/nm256/Makefile b/sound/pci/nm256/Makefile
index d91d8c519212..a1bd44ff850e 100644
--- a/sound/pci/nm256/Makefile
+++ b/sound/pci/nm256/Makefile
@@ -1,6 +1,6 @@
#
# Makefile for ALSA
-# Copyright (c) 2001 by Jaroslav Kysela <perex@suse.cz>
+# Copyright (c) 2001 by Jaroslav Kysela <perex@perex.cz>
#
snd-nm256-objs := nm256.o
diff --git a/sound/pci/nm256/nm256.c b/sound/pci/nm256/nm256.c
index c7621bd770a6..276c5763f0e5 100644
--- a/sound/pci/nm256/nm256.c
+++ b/sound/pci/nm256/nm256.c
@@ -842,7 +842,6 @@ static void snd_nm256_setup_stream(struct nm256 *chip, struct nm256_stream *s,
runtime->private_data = s;
s->substream = substream;
- snd_pcm_set_sync(substream);
snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE,
&constraints_rates);
}
diff --git a/sound/pci/pcxhr/pcxhr.c b/sound/pci/pcxhr/pcxhr.c
index f7f6a687f033..2d618bd7e62b 100644
--- a/sound/pci/pcxhr/pcxhr.c
+++ b/sound/pci/pcxhr/pcxhr.c
@@ -646,6 +646,8 @@ static int pcxhr_trigger(struct snd_pcm_substream *subs, int cmd)
if (snd_pcm_stream_linked(subs)) {
struct snd_pcxhr *chip = snd_pcm_substream_chip(subs);
snd_pcm_group_for_each_entry(s, subs) {
+ if (snd_pcm_substream_chip(s) != chip)
+ continue;
stream = s->runtime->private_data;
stream->status =
PCXHR_STREAM_STATUS_SCHEDULE_RUN;
@@ -662,6 +664,7 @@ static int pcxhr_trigger(struct snd_pcm_substream *subs, int cmd)
if (pcxhr_update_r_buffer(stream))
return -EINVAL;
+ stream->status = PCXHR_STREAM_STATUS_SCHEDULE_RUN;
if (pcxhr_set_stream_state(stream))
return -EINVAL;
stream->status = PCXHR_STREAM_STATUS_RUNNING;
@@ -902,6 +905,8 @@ static int pcxhr_open(struct snd_pcm_substream *subs)
snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, 4);
snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 4);
+ snd_pcm_set_sync(subs);
+
mgr->ref_count_rate++;
mutex_unlock(&mgr->setup_mutex);
diff --git a/sound/pci/pcxhr/pcxhr_mixer.c b/sound/pci/pcxhr/pcxhr_mixer.c
index d9cc8d2beb6d..5f8d42633b04 100644
--- a/sound/pci/pcxhr/pcxhr_mixer.c
+++ b/sound/pci/pcxhr/pcxhr_mixer.c
@@ -44,8 +44,8 @@
#define PCXHR_ANALOG_PLAYBACK_LEVEL_MAX 128 /* 0.0 dB */
#define PCXHR_ANALOG_PLAYBACK_ZERO_LEVEL 104 /* -24.0 dB ( 0.0 dB - fix level +24.0 dB ) */
-static const DECLARE_TLV_DB_SCALE(db_scale_analog_capture, -9600, 50, 0);
-static const DECLARE_TLV_DB_SCALE(db_scale_analog_playback, -12800, 100, 0);
+static const DECLARE_TLV_DB_SCALE(db_scale_analog_capture, -9600, 50, 3150);
+static const DECLARE_TLV_DB_SCALE(db_scale_analog_playback, -10400, 100, 2400);
static int pcxhr_update_analog_audio_level(struct snd_pcxhr *chip, int is_capture, int channel)
{
@@ -144,14 +144,7 @@ static struct snd_kcontrol_new pcxhr_control_analog_level = {
};
/* shared */
-static int pcxhr_sw_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
-{
- uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
- uinfo->count = 2;
- uinfo->value.integer.min = 0;
- uinfo->value.integer.max = 1;
- return 0;
-}
+#define pcxhr_sw_info snd_ctl_boolean_stereo_info
static int pcxhr_audio_sw_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
@@ -195,7 +188,7 @@ static struct snd_kcontrol_new pcxhr_control_output_switch = {
#define PCXHR_DIGITAL_LEVEL_MAX 0x1ff /* +18 dB */
#define PCXHR_DIGITAL_ZERO_LEVEL 0x1b7 /* 0 dB */
-static const DECLARE_TLV_DB_SCALE(db_scale_digital, -10950, 50, 0);
+static const DECLARE_TLV_DB_SCALE(db_scale_digital, -10975, 25, 1800);
#define MORE_THAN_ONE_STREAM_LEVEL 0x000001
#define VALID_STREAM_PAN_LEVEL_MASK 0x800000
diff --git a/sound/pci/rme32.c b/sound/pci/rme32.c
index 618653e22561..1475912588e9 100644
--- a/sound/pci/rme32.c
+++ b/sound/pci/rme32.c
@@ -258,19 +258,6 @@ static inline unsigned int snd_rme32_pcm_byteptr(struct rme32 * rme32)
& RME32_RCR_AUDIO_ADDR_MASK);
}
-static int snd_rme32_ratecode(int rate)
-{
- switch (rate) {
- case 32000: return SNDRV_PCM_RATE_32000;
- case 44100: return SNDRV_PCM_RATE_44100;
- case 48000: return SNDRV_PCM_RATE_48000;
- case 64000: return SNDRV_PCM_RATE_64000;
- case 88200: return SNDRV_PCM_RATE_88200;
- case 96000: return SNDRV_PCM_RATE_96000;
- }
- return 0;
-}
-
/* silence callback for halfduplex mode */
static int snd_rme32_playback_silence(struct snd_pcm_substream *substream, int channel, /* not used (interleaved data) */
snd_pcm_uframes_t pos,
@@ -887,7 +874,7 @@ static int snd_rme32_playback_spdif_open(struct snd_pcm_substream *substream)
if ((rme32->rcreg & RME32_RCR_KMODE) &&
(rate = snd_rme32_capture_getrate(rme32, &dummy)) > 0) {
/* AutoSync */
- runtime->hw.rates = snd_rme32_ratecode(rate);
+ runtime->hw.rates = snd_pcm_rate_to_rate_bit(rate);
runtime->hw.rate_min = rate;
runtime->hw.rate_max = rate;
}
@@ -929,7 +916,7 @@ static int snd_rme32_capture_spdif_open(struct snd_pcm_substream *substream)
if (isadat) {
return -EIO;
}
- runtime->hw.rates = snd_rme32_ratecode(rate);
+ runtime->hw.rates = snd_pcm_rate_to_rate_bit(rate);
runtime->hw.rate_min = rate;
runtime->hw.rate_max = rate;
}
@@ -965,7 +952,7 @@ snd_rme32_playback_adat_open(struct snd_pcm_substream *substream)
if ((rme32->rcreg & RME32_RCR_KMODE) &&
(rate = snd_rme32_capture_getrate(rme32, &dummy)) > 0) {
/* AutoSync */
- runtime->hw.rates = snd_rme32_ratecode(rate);
+ runtime->hw.rates = snd_pcm_rate_to_rate_bit(rate);
runtime->hw.rate_min = rate;
runtime->hw.rate_max = rate;
}
@@ -989,7 +976,7 @@ snd_rme32_capture_adat_open(struct snd_pcm_substream *substream)
if (!isadat) {
return -EIO;
}
- runtime->hw.rates = snd_rme32_ratecode(rate);
+ runtime->hw.rates = snd_pcm_rate_to_rate_bit(rate);
runtime->hw.rate_min = rate;
runtime->hw.rate_max = rate;
}
@@ -1582,16 +1569,8 @@ static void __devinit snd_rme32_proc_init(struct rme32 * rme32)
* control interface
*/
-static int
-snd_rme32_info_loopback_control(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_info *uinfo)
-{
- uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
- uinfo->count = 1;
- uinfo->value.integer.min = 0;
- uinfo->value.integer.max = 1;
- return 0;
-}
+#define snd_rme32_info_loopback_control snd_ctl_boolean_mono_info
+
static int
snd_rme32_get_loopback_control(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
diff --git a/sound/pci/rme96.c b/sound/pci/rme96.c
index e3304b7ccbcb..0b3c532c4014 100644
--- a/sound/pci/rme96.c
+++ b/sound/pci/rme96.c
@@ -301,20 +301,6 @@ snd_rme96_capture_ptr(struct rme96 *rme96)
}
static int
-snd_rme96_ratecode(int rate)
-{
- switch (rate) {
- case 32000: return SNDRV_PCM_RATE_32000;
- case 44100: return SNDRV_PCM_RATE_44100;
- case 48000: return SNDRV_PCM_RATE_48000;
- case 64000: return SNDRV_PCM_RATE_64000;
- case 88200: return SNDRV_PCM_RATE_88200;
- case 96000: return SNDRV_PCM_RATE_96000;
- }
- return 0;
-}
-
-static int
snd_rme96_playback_silence(struct snd_pcm_substream *substream,
int channel, /* not used (interleaved data) */
snd_pcm_uframes_t pos,
@@ -1176,8 +1162,6 @@ snd_rme96_playback_spdif_open(struct snd_pcm_substream *substream)
struct rme96 *rme96 = snd_pcm_substream_chip(substream);
struct snd_pcm_runtime *runtime = substream->runtime;
- snd_pcm_set_sync(substream);
-
spin_lock_irq(&rme96->lock);
if (rme96->playback_substream != NULL) {
spin_unlock_irq(&rme96->lock);
@@ -1194,7 +1178,7 @@ snd_rme96_playback_spdif_open(struct snd_pcm_substream *substream)
(rate = snd_rme96_capture_getrate(rme96, &dummy)) > 0)
{
/* slave clock */
- runtime->hw.rates = snd_rme96_ratecode(rate);
+ runtime->hw.rates = snd_pcm_rate_to_rate_bit(rate);
runtime->hw.rate_min = rate;
runtime->hw.rate_max = rate;
}
@@ -1214,8 +1198,6 @@ snd_rme96_capture_spdif_open(struct snd_pcm_substream *substream)
struct rme96 *rme96 = snd_pcm_substream_chip(substream);
struct snd_pcm_runtime *runtime = substream->runtime;
- snd_pcm_set_sync(substream);
-
runtime->hw = snd_rme96_capture_spdif_info;
if (snd_rme96_getinputtype(rme96) != RME96_INPUT_ANALOG &&
(rate = snd_rme96_capture_getrate(rme96, &isadat)) > 0)
@@ -1223,7 +1205,7 @@ snd_rme96_capture_spdif_open(struct snd_pcm_substream *substream)
if (isadat) {
return -EIO;
}
- runtime->hw.rates = snd_rme96_ratecode(rate);
+ runtime->hw.rates = snd_pcm_rate_to_rate_bit(rate);
runtime->hw.rate_min = rate;
runtime->hw.rate_max = rate;
}
@@ -1247,8 +1229,6 @@ snd_rme96_playback_adat_open(struct snd_pcm_substream *substream)
struct rme96 *rme96 = snd_pcm_substream_chip(substream);
struct snd_pcm_runtime *runtime = substream->runtime;
- snd_pcm_set_sync(substream);
-
spin_lock_irq(&rme96->lock);
if (rme96->playback_substream != NULL) {
spin_unlock_irq(&rme96->lock);
@@ -1265,7 +1245,7 @@ snd_rme96_playback_adat_open(struct snd_pcm_substream *substream)
(rate = snd_rme96_capture_getrate(rme96, &dummy)) > 0)
{
/* slave clock */
- runtime->hw.rates = snd_rme96_ratecode(rate);
+ runtime->hw.rates = snd_pcm_rate_to_rate_bit(rate);
runtime->hw.rate_min = rate;
runtime->hw.rate_max = rate;
}
@@ -1280,8 +1260,6 @@ snd_rme96_capture_adat_open(struct snd_pcm_substream *substream)
struct rme96 *rme96 = snd_pcm_substream_chip(substream);
struct snd_pcm_runtime *runtime = substream->runtime;
- snd_pcm_set_sync(substream);
-
runtime->hw = snd_rme96_capture_adat_info;
if (snd_rme96_getinputtype(rme96) == RME96_INPUT_ANALOG) {
/* makes no sense to use analog input. Note that analog
@@ -1292,7 +1270,7 @@ snd_rme96_capture_adat_open(struct snd_pcm_substream *substream)
if (!isadat) {
return -EIO;
}
- runtime->hw.rates = snd_rme96_ratecode(rate);
+ runtime->hw.rates = snd_pcm_rate_to_rate_bit(rate);
runtime->hw.rate_min = rate;
runtime->hw.rate_max = rate;
}
@@ -1826,15 +1804,8 @@ snd_rme96_proc_init(struct rme96 *rme96)
* control interface
*/
-static int
-snd_rme96_info_loopback_control(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
-{
- uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
- uinfo->count = 1;
- uinfo->value.integer.min = 0;
- uinfo->value.integer.max = 1;
- return 0;
-}
+#define snd_rme96_info_loopback_control snd_ctl_boolean_mono_info
+
static int
snd_rme96_get_loopback_control(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
{
diff --git a/sound/pci/rme9652/Makefile b/sound/pci/rme9652/Makefile
index d2c294e136f9..dcba56040205 100644
--- a/sound/pci/rme9652/Makefile
+++ b/sound/pci/rme9652/Makefile
@@ -1,6 +1,6 @@
#
# Makefile for ALSA
-# Copyright (c) 2001 by Jaroslav Kysela <perex@suse.cz>
+# Copyright (c) 2001 by Jaroslav Kysela <perex@perex.cz>
#
snd-rme9652-objs := rme9652.o
diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c
index 3b3ef657f73e..ff26a3672d40 100644
--- a/sound/pci/rme9652/hdsp.c
+++ b/sound/pci/rme9652/hdsp.c
@@ -606,28 +606,28 @@ static void snd_hdsp_9652_enable_mixer (struct hdsp *hdsp);
static int hdsp_playback_to_output_key (struct hdsp *hdsp, int in, int out)
{
- switch (hdsp->firmware_rev) {
- case 0xa:
+ switch (hdsp->io_type) {
+ case Multiface:
+ case Digiface:
+ default:
return (64 * out) + (32 + (in));
- case 0x96:
- case 0x97:
- case 0x98:
+ case H9632:
return (32 * out) + (16 + (in));
- default:
+ case H9652:
return (52 * out) + (26 + (in));
}
}
static int hdsp_input_to_output_key (struct hdsp *hdsp, int in, int out)
{
- switch (hdsp->firmware_rev) {
- case 0xa:
+ switch (hdsp->io_type) {
+ case Multiface:
+ case Digiface:
+ default:
return (64 * out) + in;
- case 0x96:
- case 0x97:
- case 0x98:
+ case H9632:
return (32 * out) + in;
- default:
+ case H9652:
return (52 * out) + in;
}
}
@@ -1623,14 +1623,7 @@ static int hdsp_set_spdif_output(struct hdsp *hdsp, int out)
return 0;
}
-static int snd_hdsp_info_spdif_bits(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
-{
- uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
- uinfo->count = 1;
- uinfo->value.integer.min = 0;
- uinfo->value.integer.max = 1;
- return 0;
-}
+#define snd_hdsp_info_spdif_bits snd_ctl_boolean_mono_info
static int snd_hdsp_get_spdif_out(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
{
@@ -2111,14 +2104,7 @@ static int snd_hdsp_put_clock_source(struct snd_kcontrol *kcontrol, struct snd_c
return change;
}
-static int snd_hdsp_info_clock_source_lock(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
-{
- uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
- uinfo->count = 1;
- uinfo->value.integer.min = 0;
- uinfo->value.integer.max = 1;
- return 0;
-}
+#define snd_hdsp_info_clock_source_lock snd_ctl_boolean_mono_info
static int snd_hdsp_get_clock_source_lock(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
{
@@ -2420,14 +2406,7 @@ static int hdsp_set_xlr_breakout_cable(struct hdsp *hdsp, int mode)
return 0;
}
-static int snd_hdsp_info_xlr_breakout_cable(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
-{
- uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
- uinfo->count = 1;
- uinfo->value.integer.min = 0;
- uinfo->value.integer.max = 1;
- return 0;
-}
+#define snd_hdsp_info_xlr_breakout_cable snd_ctl_boolean_mono_info
static int snd_hdsp_get_xlr_breakout_cable(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
{
@@ -2483,14 +2462,7 @@ static int hdsp_set_aeb(struct hdsp *hdsp, int mode)
return 0;
}
-static int snd_hdsp_info_aeb(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
-{
- uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
- uinfo->count = 1;
- uinfo->value.integer.min = 0;
- uinfo->value.integer.max = 1;
- return 0;
-}
+#define snd_hdsp_info_aeb snd_ctl_boolean_mono_info
static int snd_hdsp_get_aeb(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
{
@@ -2729,14 +2701,7 @@ static int hdsp_set_line_output(struct hdsp *hdsp, int out)
return 0;
}
-static int snd_hdsp_info_line_out(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
-{
- uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
- uinfo->count = 1;
- uinfo->value.integer.min = 0;
- uinfo->value.integer.max = 1;
- return 0;
-}
+#define snd_hdsp_info_line_out snd_ctl_boolean_mono_info
static int snd_hdsp_get_line_out(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
{
@@ -2782,14 +2747,7 @@ static int hdsp_set_precise_pointer(struct hdsp *hdsp, int precise)
return 0;
}
-static int snd_hdsp_info_precise_pointer(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
-{
- uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
- uinfo->count = 1;
- uinfo->value.integer.min = 0;
- uinfo->value.integer.max = 1;
- return 0;
-}
+#define snd_hdsp_info_precise_pointer snd_ctl_boolean_mono_info
static int snd_hdsp_get_precise_pointer(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
{
@@ -2835,14 +2793,7 @@ static int hdsp_set_use_midi_tasklet(struct hdsp *hdsp, int use_tasklet)
return 0;
}
-static int snd_hdsp_info_use_midi_tasklet(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
-{
- uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
- uinfo->count = 1;
- uinfo->value.integer.min = 0;
- uinfo->value.integer.max = 1;
- return 0;
-}
+#define snd_hdsp_info_use_midi_tasklet snd_ctl_boolean_mono_info
static int snd_hdsp_get_use_midi_tasklet(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
{
@@ -3108,6 +3059,9 @@ static int hdsp_dds_offset(struct hdsp *hdsp)
unsigned int dds_value = hdsp->dds_value;
int system_sample_rate = hdsp->system_sample_rate;
+ if (!dds_value)
+ return 0;
+
n = DDS_NUMERATOR;
/*
* dds_value = n / rate
diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c
index 143185e7e4dc..f1bdda6cbcff 100644
--- a/sound/pci/rme9652/hdspm.c
+++ b/sound/pci/rme9652/hdspm.c
@@ -1,5 +1,4 @@
-/* -*- linux-c -*-
- *
+/*
* ALSA driver for RME Hammerfall DSP MADI audio interface(s)
*
* Copyright (c) 2003 Winfried Ritsch (IEM)
@@ -78,7 +77,8 @@ MODULE_PARM_DESC(enable_monitor,
"Enable Analog Out on Channel 63/64 by default.");
MODULE_AUTHOR
- ("Winfried Ritsch <ritsch_AT_iem.at>, Paul Davis <paul@linuxaudiosystems.com>, "
+ ("Winfried Ritsch <ritsch_AT_iem.at>, "
+ "Paul Davis <paul@linuxaudiosystems.com>, "
"Marcus Andersson, Thomas Charbonnel <thomas@undata.org>, "
"Remy Bruno <remy.bruno@trinnov.com>");
MODULE_DESCRIPTION("RME HDSPM");
@@ -161,7 +161,9 @@ MODULE_SUPPORTED_DEVICE("{{RME HDSPM-MADI}}");
0=off, 1=on */ /* MADI ONLY */
#define HDSPM_Dolby (1<<11) /* Dolby = "NonAudio" ?? */ /* AES32 ONLY */
-#define HDSPM_InputSelect0 (1<<14) /* Input select 0= optical, 1=coax */ /* MADI ONLY*/
+#define HDSPM_InputSelect0 (1<<14) /* Input select 0= optical, 1=coax
+ * -- MADI ONLY
+ */
#define HDSPM_InputSelect1 (1<<15) /* should be 0 */
#define HDSPM_SyncRef0 (1<<16) /* 0=WOrd, 1=MADI */
@@ -189,11 +191,13 @@ MODULE_SUPPORTED_DEVICE("{{RME HDSPM-MADI}}");
/* --- bit helper defines */
#define HDSPM_LatencyMask (HDSPM_Latency0|HDSPM_Latency1|HDSPM_Latency2)
-#define HDSPM_FrequencyMask (HDSPM_Frequency0|HDSPM_Frequency1|HDSPM_DoubleSpeed|HDSPM_QuadSpeed)
+#define HDSPM_FrequencyMask (HDSPM_Frequency0|HDSPM_Frequency1|\
+ HDSPM_DoubleSpeed|HDSPM_QuadSpeed)
#define HDSPM_InputMask (HDSPM_InputSelect0|HDSPM_InputSelect1)
#define HDSPM_InputOptical 0
#define HDSPM_InputCoaxial (HDSPM_InputSelect0)
-#define HDSPM_SyncRefMask (HDSPM_SyncRef0|HDSPM_SyncRef1|HDSPM_SyncRef2|HDSPM_SyncRef3)
+#define HDSPM_SyncRefMask (HDSPM_SyncRef0|HDSPM_SyncRef1|\
+ HDSPM_SyncRef2|HDSPM_SyncRef3)
#define HDSPM_SyncRef_Word 0
#define HDSPM_SyncRef_MADI (HDSPM_SyncRef0)
@@ -205,10 +209,12 @@ MODULE_SUPPORTED_DEVICE("{{RME HDSPM-MADI}}");
#define HDSPM_Frequency48KHz (HDSPM_Frequency1|HDSPM_Frequency0)
#define HDSPM_Frequency64KHz (HDSPM_DoubleSpeed|HDSPM_Frequency0)
#define HDSPM_Frequency88_2KHz (HDSPM_DoubleSpeed|HDSPM_Frequency1)
-#define HDSPM_Frequency96KHz (HDSPM_DoubleSpeed|HDSPM_Frequency1|HDSPM_Frequency0)
+#define HDSPM_Frequency96KHz (HDSPM_DoubleSpeed|HDSPM_Frequency1|\
+ HDSPM_Frequency0)
#define HDSPM_Frequency128KHz (HDSPM_QuadSpeed|HDSPM_Frequency0)
#define HDSPM_Frequency176_4KHz (HDSPM_QuadSpeed|HDSPM_Frequency1)
-#define HDSPM_Frequency192KHz (HDSPM_QuadSpeed|HDSPM_Frequency1|HDSPM_Frequency0)
+#define HDSPM_Frequency192KHz (HDSPM_QuadSpeed|HDSPM_Frequency1|\
+ HDSPM_Frequency0)
/* --- for internal discrimination */
#define HDSPM_CLOCK_SOURCE_AUTOSYNC 0 /* Sample Clock Sources */
@@ -256,10 +262,14 @@ MODULE_SUPPORTED_DEVICE("{{RME HDSPM-MADI}}");
#define HDSPM_RD_MULTIPLE (1<<10)
/* --- Status Register bits --- */ /* MADI ONLY */ /* Bits defined here and
- that do not conflict with specific bits for AES32 seem to be valid also for the AES32 */
+ that do not conflict with specific bits for AES32 seem to be valid also
+ for the AES32
+ */
#define HDSPM_audioIRQPending (1<<0) /* IRQ is high and pending */
-#define HDSPM_RX_64ch (1<<1) /* Input 64chan. MODE=1, 56chn. MODE=0 */
-#define HDSPM_AB_int (1<<2) /* InputChannel Opt=0, Coax=1 (like inp0) */
+#define HDSPM_RX_64ch (1<<1) /* Input 64chan. MODE=1, 56chn MODE=0 */
+#define HDSPM_AB_int (1<<2) /* InputChannel Opt=0, Coax=1
+ * (like inp0)
+ */
#define HDSPM_madiLock (1<<3) /* MADI Locked =1, no=0 */
#define HDSPM_BufferPositionMask 0x000FFC0 /* Bit 6..15 : h/w buffer pointer */
@@ -274,12 +284,15 @@ MODULE_SUPPORTED_DEVICE("{{RME HDSPM-MADI}}");
#define HDSPM_madiFreq2 (1<<24) /* 4=64, 5=88.2 6=96 */
#define HDSPM_madiFreq3 (1<<25) /* 7=128, 8=176.4 9=192 */
-#define HDSPM_BufferID (1<<26) /* (Double)Buffer ID toggles with Interrupt */
+#define HDSPM_BufferID (1<<26) /* (Double)Buffer ID toggles with
+ * Interrupt
+ */
#define HDSPM_midi0IRQPending (1<<30) /* MIDI IRQ is pending */
#define HDSPM_midi1IRQPending (1<<31) /* and aktiv */
/* --- status bit helpers */
-#define HDSPM_madiFreqMask (HDSPM_madiFreq0|HDSPM_madiFreq1|HDSPM_madiFreq2|HDSPM_madiFreq3)
+#define HDSPM_madiFreqMask (HDSPM_madiFreq0|HDSPM_madiFreq1|\
+ HDSPM_madiFreq2|HDSPM_madiFreq3)
#define HDSPM_madiFreq32 (HDSPM_madiFreq0)
#define HDSPM_madiFreq44_1 (HDSPM_madiFreq1)
#define HDSPM_madiFreq48 (HDSPM_madiFreq0|HDSPM_madiFreq1)
@@ -319,10 +332,12 @@ MODULE_SUPPORTED_DEVICE("{{RME HDSPM-MADI}}");
#define HDSPM_wcFreq96 (HDSPM_wc_freq1|HDSPM_wc_freq2)
-#define HDSPM_SelSyncRefMask (HDSPM_SelSyncRef0|HDSPM_SelSyncRef1|HDSPM_SelSyncRef2)
+#define HDSPM_SelSyncRefMask (HDSPM_SelSyncRef0|HDSPM_SelSyncRef1|\
+ HDSPM_SelSyncRef2)
#define HDSPM_SelSyncRef_WORD 0
#define HDSPM_SelSyncRef_MADI (HDSPM_SelSyncRef0)
-#define HDSPM_SelSyncRef_NVALID (HDSPM_SelSyncRef0|HDSPM_SelSyncRef1|HDSPM_SelSyncRef2)
+#define HDSPM_SelSyncRef_NVALID (HDSPM_SelSyncRef0|HDSPM_SelSyncRef1|\
+ HDSPM_SelSyncRef2)
/*
For AES32, bits for status, status2 and timecode are different
@@ -344,7 +359,7 @@ MODULE_SUPPORTED_DEVICE("{{RME HDSPM-MADI}}");
#define HDSPM_AES32_AUTOSYNC_FROM_AES6 6
#define HDSPM_AES32_AUTOSYNC_FROM_AES7 7
#define HDSPM_AES32_AUTOSYNC_FROM_AES8 8
-#define HDSPM_AES32_AUTOSYNC_FROM_NONE -1
+#define HDSPM_AES32_AUTOSYNC_FROM_NONE 9
/* status2 */
/* HDSPM_LockAES_bit is given by HDSPM_LockAES >> (AES# - 1) */
@@ -398,6 +413,13 @@ MODULE_SUPPORTED_DEVICE("{{RME HDSPM-MADI}}");
/* revisions >= 230 indicate AES32 card */
#define HDSPM_AESREVISION 230
+/* speed factor modes */
+#define HDSPM_SPEED_SINGLE 0
+#define HDSPM_SPEED_DOUBLE 1
+#define HDSPM_SPEED_QUAD 2
+/* names for speed modes */
+static char *hdspm_speed_names[] = { "single", "double", "quad" };
+
struct hdspm_midi {
struct hdspm *hdspm;
int id;
@@ -412,8 +434,9 @@ struct hdspm_midi {
struct hdspm {
spinlock_t lock;
- struct snd_pcm_substream *capture_substream; /* only one playback */
- struct snd_pcm_substream *playback_substream; /* and/or capture stream */
+ /* only one playback and/or capture stream */
+ struct snd_pcm_substream *capture_substream;
+ struct snd_pcm_substream *playback_substream;
char *card_name; /* for procinfo */
unsigned short firmware_rev; /* dont know if relevant (yes if AES32)*/
@@ -460,9 +483,12 @@ struct hdspm {
struct pci_dev *pci; /* and an pci info */
/* Mixer vars */
- struct snd_kcontrol *playback_mixer_ctls[HDSPM_MAX_CHANNELS]; /* fast alsa mixer */
- struct snd_kcontrol *input_mixer_ctls[HDSPM_MAX_CHANNELS]; /* but input to much, so not used */
- struct hdspm_mixer *mixer; /* full mixer accessable over mixer ioctl or hwdep-device */
+ /* fast alsa mixer */
+ 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 */
+ struct hdspm_mixer *mixer;
};
@@ -616,13 +642,15 @@ static inline int hdspm_external_sample_rate(struct hdspm * hdspm)
if (hdspm->is_aes32) {
unsigned int status2 = hdspm_read(hdspm, HDSPM_statusRegister2);
unsigned int status = hdspm_read(hdspm, HDSPM_statusRegister);
- unsigned int timecode = hdspm_read(hdspm, HDSPM_timecodeRegister);
+ unsigned int timecode =
+ hdspm_read(hdspm, HDSPM_timecodeRegister);
int syncref = hdspm_autosync_ref(hdspm);
if (syncref == HDSPM_AES32_AUTOSYNC_FROM_WORD &&
status & HDSPM_AES32_wcLock)
- return HDSPM_bit2freq((status >> HDSPM_AES32_wcFreq_bit) & 0xF);
+ return HDSPM_bit2freq((status >> HDSPM_AES32_wcFreq_bit)
+ & 0xF);
if (syncref >= HDSPM_AES32_AUTOSYNC_FROM_AES1 &&
syncref <= HDSPM_AES32_AUTOSYNC_FROM_AES8 &&
status2 & (HDSPM_LockAES >>
@@ -668,7 +696,9 @@ static inline int hdspm_external_sample_rate(struct hdspm * hdspm)
}
}
- /* if rate detected and Syncref is Word than have it, word has priority to MADI */
+ /* if rate detected and Syncref is Word than have it,
+ * word has priority to MADI
+ */
if (rate != 0 &&
(status2 & HDSPM_SelSyncRefMask) == HDSPM_SelSyncRef_WORD)
return rate;
@@ -727,12 +757,12 @@ static snd_pcm_uframes_t hdspm_hw_pointer(struct hdspm * hdspm)
position = hdspm_read(hdspm, HDSPM_statusRegister);
- if (!hdspm->precise_ptr) {
- return (position & HDSPM_BufferID) ? (hdspm->period_bytes /
- 4) : 0;
- }
+ if (!hdspm->precise_ptr)
+ return (position & HDSPM_BufferID) ?
+ (hdspm->period_bytes / 4) : 0;
- /* hwpointer comes in bytes and is 64Bytes accurate (by docu since PCI Burst)
+ /* hwpointer comes in bytes and is 64Bytes accurate (by docu since
+ PCI Burst)
i have experimented that it is at most 64 Byte to much for playing
so substraction of 64 byte should be ok for ALSA, but use it only
for application where you know what you do since if you come to
@@ -811,7 +841,7 @@ static void hdspm_set_dds_value(struct hdspm *hdspm, int rate)
// return 104857600000000 / rate; // 100 MHz
return 110100480000000 / rate; // 105 MHz
*/
- //n = 104857600000000ULL; /* = 2^20 * 10^8 */
+ /* n = 104857600000000ULL; */ /* = 2^20 * 10^8 */
n = 110100480000000ULL; /* Value checked for AES32 and MADI */
div64_32(&n, rate, &r);
/* n should be less than 2^32 for being written to FREQ register */
@@ -822,11 +852,10 @@ static void hdspm_set_dds_value(struct hdspm *hdspm, int rate)
/* dummy set rate lets see what happens */
static int hdspm_set_rate(struct hdspm * hdspm, int rate, int called_internally)
{
- int reject_if_open = 0;
int current_rate;
int rate_bits;
int not_set = 0;
- int is_single, is_double, is_quad;
+ int current_speed, target_speed;
/* ASSUMPTION: hdspm->lock is either set, or there is no need for
it (e.g. during module initialization).
@@ -841,8 +870,9 @@ static int hdspm_set_rate(struct hdspm * hdspm, int rate, int called_internally)
just make a warning an remember setting
for future master mode switching */
- snd_printk
- (KERN_WARNING "HDSPM: Warning: device is not running as a clock master.\n");
+ snd_printk(KERN_WARNING "HDSPM: "
+ "Warning: device is not running "
+ "as a clock master.\n");
not_set = 1;
} else {
@@ -850,16 +880,18 @@ static int hdspm_set_rate(struct hdspm * hdspm, int rate, int called_internally)
int external_freq =
hdspm_external_sample_rate(hdspm);
- if ((hdspm_autosync_ref(hdspm) ==
- HDSPM_AUTOSYNC_FROM_NONE)) {
+ if (hdspm_autosync_ref(hdspm) ==
+ HDSPM_AUTOSYNC_FROM_NONE) {
- snd_printk(KERN_WARNING "HDSPM: Detected no Externel Sync \n");
+ snd_printk(KERN_WARNING "HDSPM: "
+ "Detected no Externel Sync \n");
not_set = 1;
} else if (rate != external_freq) {
- snd_printk
- (KERN_WARNING "HDSPM: Warning: No AutoSync source for requested rate\n");
+ snd_printk(KERN_WARNING "HDSPM: "
+ "Warning: No AutoSync source for "
+ "requested rate\n");
not_set = 1;
}
}
@@ -877,64 +909,60 @@ static int hdspm_set_rate(struct hdspm * hdspm, int rate, int called_internally)
changes in the read/write routines.
*/
- is_single = (current_rate <= 48000);
- is_double = (current_rate > 48000 && current_rate <= 96000);
- is_quad = (current_rate > 96000);
+ if (current_rate <= 48000)
+ current_speed = HDSPM_SPEED_SINGLE;
+ else if (current_rate <= 96000)
+ current_speed = HDSPM_SPEED_DOUBLE;
+ else
+ current_speed = HDSPM_SPEED_QUAD;
+
+ if (rate <= 48000)
+ target_speed = HDSPM_SPEED_SINGLE;
+ else if (rate <= 96000)
+ target_speed = HDSPM_SPEED_DOUBLE;
+ else
+ target_speed = HDSPM_SPEED_QUAD;
switch (rate) {
case 32000:
- if (!is_single)
- reject_if_open = 1;
rate_bits = HDSPM_Frequency32KHz;
break;
case 44100:
- if (!is_single)
- reject_if_open = 1;
rate_bits = HDSPM_Frequency44_1KHz;
break;
case 48000:
- if (!is_single)
- reject_if_open = 1;
rate_bits = HDSPM_Frequency48KHz;
break;
case 64000:
- if (!is_double)
- reject_if_open = 1;
rate_bits = HDSPM_Frequency64KHz;
break;
case 88200:
- if (!is_double)
- reject_if_open = 1;
rate_bits = HDSPM_Frequency88_2KHz;
break;
case 96000:
- if (!is_double)
- reject_if_open = 1;
rate_bits = HDSPM_Frequency96KHz;
break;
case 128000:
- if (!is_quad)
- reject_if_open = 1;
rate_bits = HDSPM_Frequency128KHz;
break;
case 176400:
- if (!is_quad)
- reject_if_open = 1;
rate_bits = HDSPM_Frequency176_4KHz;
break;
case 192000:
- if (!is_quad)
- reject_if_open = 1;
rate_bits = HDSPM_Frequency192KHz;
break;
default:
return -EINVAL;
}
- if (reject_if_open
+ if (current_speed != target_speed
&& (hdspm->capture_pid >= 0 || hdspm->playback_pid >= 0)) {
snd_printk
- (KERN_ERR "HDSPM: cannot change between single- and double-speed mode (capture PID = %d, playback PID = %d)\n",
+ (KERN_ERR "HDSPM: "
+ "cannot change from %s speed to %s speed mode "
+ "(capture PID = %d, playback PID = %d)\n",
+ hdspm_speed_names[current_speed],
+ hdspm_speed_names[target_speed],
hdspm->capture_pid, hdspm->playback_pid);
return -EBUSY;
}
@@ -966,8 +994,14 @@ static int hdspm_set_rate(struct hdspm * hdspm, int rate, int called_internally)
static void all_in_all_mixer(struct hdspm * hdspm, int sgain)
{
int i, j;
- unsigned int gain =
- (sgain > UNITY_GAIN) ? UNITY_GAIN : (sgain < 0) ? 0 : sgain;
+ unsigned int gain;
+
+ if (sgain > UNITY_GAIN)
+ gain = UNITY_GAIN;
+ else if (sgain < 0)
+ gain = 0;
+ else
+ gain = sgain;
for (i = 0; i < HDSPM_MIXER_CHANNELS; i++)
for (j = 0; j < HDSPM_MIXER_CHANNELS; j++) {
@@ -980,7 +1014,8 @@ static void all_in_all_mixer(struct hdspm * hdspm, int sgain)
MIDI
----------------------------------------------------------------------------*/
-static inline unsigned char snd_hdspm_midi_read_byte (struct hdspm *hdspm, int id)
+static inline unsigned char snd_hdspm_midi_read_byte (struct hdspm *hdspm,
+ int id)
{
/* the hardware already does the relevant bit-mask with 0xff */
if (id)
@@ -989,7 +1024,8 @@ static inline unsigned char snd_hdspm_midi_read_byte (struct hdspm *hdspm, int i
return hdspm_read(hdspm, HDSPM_midiDataIn0);
}
-static inline void snd_hdspm_midi_write_byte (struct hdspm *hdspm, int id, int val)
+static inline void snd_hdspm_midi_write_byte (struct hdspm *hdspm, int id,
+ int val)
{
/* the hardware already does the relevant bit-mask with 0xff */
if (id)
@@ -1011,9 +1047,10 @@ static inline int snd_hdspm_midi_output_possible (struct hdspm *hdspm, int id)
int fifo_bytes_used;
if (id)
- fifo_bytes_used = hdspm_read(hdspm, HDSPM_midiStatusOut1) & 0xff;
+ fifo_bytes_used = hdspm_read(hdspm, HDSPM_midiStatusOut1);
else
- fifo_bytes_used = hdspm_read(hdspm, HDSPM_midiStatusOut0) & 0xff;
+ fifo_bytes_used = hdspm_read(hdspm, HDSPM_midiStatusOut0);
+ fifo_bytes_used &= 0xff;
if (fifo_bytes_used < 128)
return 128 - fifo_bytes_used;
@@ -1038,16 +1075,21 @@ static int snd_hdspm_midi_output_write (struct hdspm_midi *hmidi)
/* Output is not interrupt driven */
spin_lock_irqsave (&hmidi->lock, flags);
- if (hmidi->output) {
- if (!snd_rawmidi_transmit_empty (hmidi->output)) {
- if ((n_pending = snd_hdspm_midi_output_possible (hmidi->hdspm, hmidi->id)) > 0) {
- if (n_pending > (int)sizeof (buf))
- n_pending = sizeof (buf);
-
- if ((to_write = snd_rawmidi_transmit (hmidi->output, buf, n_pending)) > 0) {
- for (i = 0; i < to_write; ++i)
- snd_hdspm_midi_write_byte (hmidi->hdspm, hmidi->id, buf[i]);
- }
+ if (hmidi->output &&
+ !snd_rawmidi_transmit_empty (hmidi->output)) {
+ n_pending = snd_hdspm_midi_output_possible (hmidi->hdspm,
+ hmidi->id);
+ if (n_pending > 0) {
+ if (n_pending > (int)sizeof (buf))
+ n_pending = sizeof (buf);
+
+ to_write = snd_rawmidi_transmit (hmidi->output, buf,
+ n_pending);
+ if (to_write > 0) {
+ for (i = 0; i < to_write; ++i)
+ snd_hdspm_midi_write_byte (hmidi->hdspm,
+ hmidi->id,
+ buf[i]);
}
}
}
@@ -1057,51 +1099,55 @@ static int snd_hdspm_midi_output_write (struct hdspm_midi *hmidi)
static int snd_hdspm_midi_input_read (struct hdspm_midi *hmidi)
{
- unsigned char buf[128]; /* this buffer is designed to match the MIDI input FIFO size */
+ unsigned char buf[128]; /* this buffer is designed to match the MIDI
+ * input FIFO size
+ */
unsigned long flags;
int n_pending;
int i;
spin_lock_irqsave (&hmidi->lock, flags);
- if ((n_pending = snd_hdspm_midi_input_available (hmidi->hdspm, hmidi->id)) > 0) {
+ n_pending = snd_hdspm_midi_input_available (hmidi->hdspm, hmidi->id);
+ if (n_pending > 0) {
if (hmidi->input) {
- if (n_pending > (int)sizeof (buf)) {
+ if (n_pending > (int)sizeof (buf))
n_pending = sizeof (buf);
- }
- for (i = 0; i < n_pending; ++i) {
- buf[i] = snd_hdspm_midi_read_byte (hmidi->hdspm, hmidi->id);
- }
- if (n_pending) {
- snd_rawmidi_receive (hmidi->input, buf, n_pending);
- }
+ for (i = 0; i < n_pending; ++i)
+ buf[i] = snd_hdspm_midi_read_byte (hmidi->hdspm,
+ hmidi->id);
+ if (n_pending)
+ snd_rawmidi_receive (hmidi->input, buf,
+ n_pending);
} else {
/* flush the MIDI input FIFO */
- while (n_pending--) {
- snd_hdspm_midi_read_byte (hmidi->hdspm, hmidi->id);
- }
+ while (n_pending--)
+ snd_hdspm_midi_read_byte (hmidi->hdspm,
+ hmidi->id);
}
}
hmidi->pending = 0;
- if (hmidi->id) {
+ if (hmidi->id)
hmidi->hdspm->control_register |= HDSPM_Midi1InterruptEnable;
- } else {
+ else
hmidi->hdspm->control_register |= HDSPM_Midi0InterruptEnable;
- }
- hdspm_write(hmidi->hdspm, HDSPM_controlRegister, hmidi->hdspm->control_register);
+ hdspm_write(hmidi->hdspm, HDSPM_controlRegister,
+ hmidi->hdspm->control_register);
spin_unlock_irqrestore (&hmidi->lock, flags);
return snd_hdspm_midi_output_write (hmidi);
}
-static void snd_hdspm_midi_input_trigger(struct snd_rawmidi_substream *substream, int up)
+static void
+snd_hdspm_midi_input_trigger(struct snd_rawmidi_substream *substream, int up)
{
struct hdspm *hdspm;
struct hdspm_midi *hmidi;
unsigned long flags;
u32 ie;
- hmidi = (struct hdspm_midi *) substream->rmidi->private_data;
+ hmidi = substream->rmidi->private_data;
hdspm = hmidi->hdspm;
- ie = hmidi->id ? HDSPM_Midi1InterruptEnable : HDSPM_Midi0InterruptEnable;
+ ie = hmidi->id ?
+ HDSPM_Midi1InterruptEnable : HDSPM_Midi0InterruptEnable;
spin_lock_irqsave (&hdspm->lock, flags);
if (up) {
if (!(hdspm->control_register & ie)) {
@@ -1138,12 +1184,13 @@ static void snd_hdspm_midi_output_timer(unsigned long data)
spin_unlock_irqrestore (&hmidi->lock, flags);
}
-static void snd_hdspm_midi_output_trigger(struct snd_rawmidi_substream *substream, int up)
+static void
+snd_hdspm_midi_output_trigger(struct snd_rawmidi_substream *substream, int up)
{
struct hdspm_midi *hmidi;
unsigned long flags;
- hmidi = (struct hdspm_midi *) substream->rmidi->private_data;
+ hmidi = substream->rmidi->private_data;
spin_lock_irqsave (&hmidi->lock, flags);
if (up) {
if (!hmidi->istimer) {
@@ -1155,9 +1202,8 @@ static void snd_hdspm_midi_output_trigger(struct snd_rawmidi_substream *substrea
hmidi->istimer++;
}
} else {
- if (hmidi->istimer && --hmidi->istimer <= 0) {
+ if (hmidi->istimer && --hmidi->istimer <= 0)
del_timer (&hmidi->timer);
- }
}
spin_unlock_irqrestore (&hmidi->lock, flags);
if (up)
@@ -1168,7 +1214,7 @@ static int snd_hdspm_midi_input_open(struct snd_rawmidi_substream *substream)
{
struct hdspm_midi *hmidi;
- hmidi = (struct hdspm_midi *) substream->rmidi->private_data;
+ hmidi = substream->rmidi->private_data;
spin_lock_irq (&hmidi->lock);
snd_hdspm_flush_midi_input (hmidi->hdspm, hmidi->id);
hmidi->input = substream;
@@ -1181,7 +1227,7 @@ static int snd_hdspm_midi_output_open(struct snd_rawmidi_substream *substream)
{
struct hdspm_midi *hmidi;
- hmidi = (struct hdspm_midi *) substream->rmidi->private_data;
+ hmidi = substream->rmidi->private_data;
spin_lock_irq (&hmidi->lock);
hmidi->output = substream;
spin_unlock_irq (&hmidi->lock);
@@ -1195,7 +1241,7 @@ static int snd_hdspm_midi_input_close(struct snd_rawmidi_substream *substream)
snd_hdspm_midi_input_trigger (substream, 0);
- hmidi = (struct hdspm_midi *) substream->rmidi->private_data;
+ hmidi = substream->rmidi->private_data;
spin_lock_irq (&hmidi->lock);
hmidi->input = NULL;
spin_unlock_irq (&hmidi->lock);
@@ -1209,7 +1255,7 @@ static int snd_hdspm_midi_output_close(struct snd_rawmidi_substream *substream)
snd_hdspm_midi_output_trigger (substream, 0);
- hmidi = (struct hdspm_midi *) substream->rmidi->private_data;
+ hmidi = substream->rmidi->private_data;
spin_lock_irq (&hmidi->lock);
hmidi->output = NULL;
spin_unlock_irq (&hmidi->lock);
@@ -1231,29 +1277,28 @@ static struct snd_rawmidi_ops snd_hdspm_midi_input =
.trigger = snd_hdspm_midi_input_trigger,
};
-static int __devinit snd_hdspm_create_midi (struct snd_card *card, struct hdspm *hdspm, int id)
+static int __devinit snd_hdspm_create_midi (struct snd_card *card,
+ struct hdspm *hdspm, int id)
{
int err;
char buf[32];
hdspm->midi[id].id = id;
- hdspm->midi[id].rmidi = NULL;
- hdspm->midi[id].input = NULL;
- hdspm->midi[id].output = NULL;
hdspm->midi[id].hdspm = hdspm;
- hdspm->midi[id].istimer = 0;
- hdspm->midi[id].pending = 0;
spin_lock_init (&hdspm->midi[id].lock);
sprintf (buf, "%s MIDI %d", card->shortname, id+1);
- if ((err = snd_rawmidi_new (card, buf, id, 1, 1, &hdspm->midi[id].rmidi)) < 0)
+ err = snd_rawmidi_new (card, buf, id, 1, 1, &hdspm->midi[id].rmidi);
+ if (err < 0)
return err;
sprintf (hdspm->midi[id].rmidi->name, "%s MIDI %d", card->id, id+1);
hdspm->midi[id].rmidi->private_data = &hdspm->midi[id];
- snd_rawmidi_set_ops (hdspm->midi[id].rmidi, SNDRV_RAWMIDI_STREAM_OUTPUT, &snd_hdspm_midi_output);
- snd_rawmidi_set_ops (hdspm->midi[id].rmidi, SNDRV_RAWMIDI_STREAM_INPUT, &snd_hdspm_midi_input);
+ snd_rawmidi_set_ops(hdspm->midi[id].rmidi, SNDRV_RAWMIDI_STREAM_OUTPUT,
+ &snd_hdspm_midi_output);
+ snd_rawmidi_set_ops(hdspm->midi[id].rmidi, SNDRV_RAWMIDI_STREAM_INPUT,
+ &snd_hdspm_midi_input);
hdspm->midi[id].rmidi->info_flags |= SNDRV_RAWMIDI_INFO_OUTPUT |
SNDRV_RAWMIDI_INFO_INPUT |
@@ -1558,8 +1603,8 @@ static int snd_hdspm_put_clock_source(struct snd_kcontrol *kcontrol,
val = ucontrol->value.enumerated.item[0];
if (val < 0)
val = 0;
- if (val > 6)
- val = 6;
+ if (val > 9)
+ val = 9;
spin_lock_irq(&hdspm->lock);
if (val != hdspm_clock_source(hdspm))
change = (hdspm_set_clock_source(hdspm, val) == 0) ? 1 : 0;
@@ -1637,7 +1682,8 @@ static int hdspm_set_pref_sync_ref(struct hdspm * hdspm, int pref)
hdspm->control_register |= HDSPM_SyncRef2+HDSPM_SyncRef1;
break;
case 7:
- hdspm->control_register |= HDSPM_SyncRef2+HDSPM_SyncRef1+HDSPM_SyncRef0;
+ hdspm->control_register |=
+ HDSPM_SyncRef2+HDSPM_SyncRef1+HDSPM_SyncRef0;
break;
case 8:
hdspm->control_register |= HDSPM_SyncRef3;
@@ -1675,7 +1721,8 @@ static int snd_hdspm_info_pref_sync_ref(struct snd_kcontrol *kcontrol,
uinfo->value.enumerated.items = 9;
- if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items)
+ if (uinfo->value.enumerated.item >=
+ uinfo->value.enumerated.items)
uinfo->value.enumerated.item =
uinfo->value.enumerated.items - 1;
strcpy(uinfo->value.enumerated.name,
@@ -1688,7 +1735,8 @@ static int snd_hdspm_info_pref_sync_ref(struct snd_kcontrol *kcontrol,
uinfo->value.enumerated.items = 2;
- if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items)
+ if (uinfo->value.enumerated.item >=
+ uinfo->value.enumerated.items)
uinfo->value.enumerated.item =
uinfo->value.enumerated.items - 1;
strcpy(uinfo->value.enumerated.name,
@@ -1740,7 +1788,8 @@ static int hdspm_autosync_ref(struct hdspm * hdspm)
{
if (hdspm->is_aes32) {
unsigned int status = hdspm_read(hdspm, HDSPM_statusRegister);
- unsigned int syncref = (status >> HDSPM_AES32_syncref_bit) & 0xF;
+ unsigned int syncref = (status >> HDSPM_AES32_syncref_bit) &
+ 0xF;
if (syncref == 0)
return HDSPM_AES32_AUTOSYNC_FROM_WORD;
if (syncref <= 8)
@@ -1777,20 +1826,20 @@ static int snd_hdspm_info_autosync_ref(struct snd_kcontrol *kcontrol,
uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
uinfo->count = 1;
uinfo->value.enumerated.items = 10;
- if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items)
+ if (uinfo->value.enumerated.item >=
+ uinfo->value.enumerated.items)
uinfo->value.enumerated.item =
uinfo->value.enumerated.items - 1;
strcpy(uinfo->value.enumerated.name,
texts[uinfo->value.enumerated.item]);
- }
- else
- {
+ } else {
static char *texts[] = { "WordClock", "MADI", "None" };
uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
uinfo->count = 1;
uinfo->value.enumerated.items = 3;
- if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items)
+ if (uinfo->value.enumerated.item >=
+ uinfo->value.enumerated.items)
uinfo->value.enumerated.item =
uinfo->value.enumerated.items - 1;
strcpy(uinfo->value.enumerated.name,
@@ -1804,7 +1853,7 @@ static int snd_hdspm_get_autosync_ref(struct snd_kcontrol *kcontrol,
{
struct hdspm *hdspm = snd_kcontrol_chip(kcontrol);
- ucontrol->value.enumerated.item[0] = hdspm_pref_sync_ref(hdspm);
+ ucontrol->value.enumerated.item[0] = hdspm_autosync_ref(hdspm);
return 0;
}
@@ -1834,15 +1883,7 @@ static int hdspm_set_line_output(struct hdspm * hdspm, int out)
return 0;
}
-static int snd_hdspm_info_line_out(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_info *uinfo)
-{
- uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
- uinfo->count = 1;
- uinfo->value.integer.min = 0;
- uinfo->value.integer.max = 1;
- return 0;
-}
+#define snd_hdspm_info_line_out snd_ctl_boolean_mono_info
static int snd_hdspm_get_line_out(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
@@ -1897,15 +1938,7 @@ static int hdspm_set_tx_64(struct hdspm * hdspm, int out)
return 0;
}
-static int snd_hdspm_info_tx_64(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_info *uinfo)
-{
- uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
- uinfo->count = 1;
- uinfo->value.integer.min = 0;
- uinfo->value.integer.max = 1;
- return 0;
-}
+#define snd_hdspm_info_tx_64 snd_ctl_boolean_mono_info
static int snd_hdspm_get_tx_64(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
@@ -1960,15 +1993,7 @@ static int hdspm_set_c_tms(struct hdspm * hdspm, int out)
return 0;
}
-static int snd_hdspm_info_c_tms(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_info *uinfo)
-{
- uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
- uinfo->count = 1;
- uinfo->value.integer.min = 0;
- uinfo->value.integer.max = 1;
- return 0;
-}
+#define snd_hdspm_info_c_tms snd_ctl_boolean_mono_info
static int snd_hdspm_get_c_tms(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
@@ -2023,15 +2048,7 @@ static int hdspm_set_safe_mode(struct hdspm * hdspm, int out)
return 0;
}
-static int snd_hdspm_info_safe_mode(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_info *uinfo)
-{
- uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
- uinfo->count = 1;
- uinfo->value.integer.min = 0;
- uinfo->value.integer.max = 1;
- return 0;
-}
+#define snd_hdspm_info_safe_mode snd_ctl_boolean_mono_info
static int snd_hdspm_get_safe_mode(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
@@ -2086,15 +2103,7 @@ static int hdspm_set_emphasis(struct hdspm * hdspm, int emp)
return 0;
}
-static int snd_hdspm_info_emphasis(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_info *uinfo)
-{
- uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
- uinfo->count = 1;
- uinfo->value.integer.min = 0;
- uinfo->value.integer.max = 1;
- return 0;
-}
+#define snd_hdspm_info_emphasis snd_ctl_boolean_mono_info
static int snd_hdspm_get_emphasis(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
@@ -2149,15 +2158,7 @@ static int hdspm_set_dolby(struct hdspm * hdspm, int dol)
return 0;
}
-static int snd_hdspm_info_dolby(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_info *uinfo)
-{
- uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
- uinfo->count = 1;
- uinfo->value.integer.min = 0;
- uinfo->value.integer.max = 1;
- return 0;
-}
+#define snd_hdspm_info_dolby snd_ctl_boolean_mono_info
static int snd_hdspm_get_dolby(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
@@ -2212,15 +2213,7 @@ static int hdspm_set_professional(struct hdspm * hdspm, int dol)
return 0;
}
-static int snd_hdspm_info_professional(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_info *uinfo)
-{
- uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
- uinfo->count = 1;
- uinfo->value.integer.min = 0;
- uinfo->value.integer.max = 1;
- return 0;
-}
+#define snd_hdspm_info_professional snd_ctl_boolean_mono_info
static int snd_hdspm_get_professional(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
@@ -2472,7 +2465,7 @@ static int snd_hdspm_put_qs_wire(struct snd_kcontrol *kcontrol,
if (val > 2)
val = 2;
spin_lock_irq(&hdspm->lock);
- change = (int) val != hdspm_qs_wire(hdspm);
+ change = val != hdspm_qs_wire(hdspm);
hdspm_set_qs_wire(hdspm, val);
spin_unlock_irq(&hdspm->lock);
return change;
@@ -2573,8 +2566,8 @@ static int snd_hdspm_put_mixer(struct snd_kcontrol *kcontrol,
source -
HDSPM_MAX_CHANNELS);
else
- change =
- gain != hdspm_read_in_gain(hdspm, destination, source);
+ change = gain != hdspm_read_in_gain(hdspm, destination,
+ source);
if (change) {
if (source >= HDSPM_MAX_CHANNELS)
@@ -2627,7 +2620,8 @@ static int snd_hdspm_get_playback_mixer(struct snd_kcontrol *kcontrol,
snd_assert(channel >= 0
|| channel < HDSPM_MAX_CHANNELS, return -EINVAL);
- if ((mapped_channel = hdspm->channel_map[channel]) < 0)
+ mapped_channel = hdspm->channel_map[channel];
+ if (mapped_channel < 0)
return -EINVAL;
spin_lock_irq(&hdspm->lock);
@@ -2635,10 +2629,12 @@ static int snd_hdspm_get_playback_mixer(struct snd_kcontrol *kcontrol,
hdspm_read_pb_gain(hdspm, mapped_channel, mapped_channel);
spin_unlock_irq(&hdspm->lock);
- /* snd_printdd("get pb mixer index %d, channel %d, mapped_channel %d, value %d\n",
- ucontrol->id.index, channel, mapped_channel, ucontrol->value.integer.value[0]);
- */
-
+ /*
+ snd_printdd("get pb mixer index %d, channel %d, mapped_channel %d, "
+ "value %d\n",
+ ucontrol->id.index, channel, mapped_channel,
+ ucontrol->value.integer.value[0]);
+ */
return 0;
}
@@ -2659,7 +2655,8 @@ static int snd_hdspm_put_playback_mixer(struct snd_kcontrol *kcontrol,
snd_assert(channel >= 0
|| channel < HDSPM_MAX_CHANNELS, return -EINVAL);
- if ((mapped_channel = hdspm->channel_map[channel]) < 0)
+ mapped_channel = hdspm->channel_map[channel];
+ if (mapped_channel < 0)
return -EINVAL;
gain = ucontrol->value.integer.value[0];
@@ -2909,28 +2906,26 @@ 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 IOCTL on hwdep
-and the mixer 2dimensional mixer control */
+ Note: the whole matrix would be 128*HDSPM_MIXER_CHANNELS Faders,
+ thats too * big for any alsamixer they are accesible via special
+ IOCTL on hwdep and the mixer 2dimensional mixer control
+ */
snd_hdspm_playback_mixer.name = "Chn";
limit = HDSPM_MAX_CHANNELS;
- /* The index values are one greater than the channel ID so that alsamixer
- will display them correctly. We want to use the index for fast lookup
- of the relevant channel, but if we use it at all, most ALSA software
- does the wrong thing with it ...
+ /* The index values are one greater than the channel ID so that
+ * alsamixer will display them correctly. We want to use the index
+ * for fast lookup of the relevant channel, but if we use it at all,
+ * most ALSA software does the wrong thing with it ...
*/
for (idx = 0; idx < limit; ++idx) {
snd_hdspm_playback_mixer.index = idx + 1;
- if ((err = snd_ctl_add(card,
- kctl =
- snd_ctl_new1
- (&snd_hdspm_playback_mixer,
- hdspm)))) {
+ kctl = snd_ctl_new1(&snd_hdspm_playback_mixer, hdspm);
+ err = snd_ctl_add(card, kctl);
+ if (err < 0)
return err;
- }
hdspm->playback_mixer_ctls[idx] = kctl;
}
@@ -2945,7 +2940,7 @@ static void
snd_hdspm_proc_read_madi(struct snd_info_entry * entry,
struct snd_info_buffer *buffer)
{
- struct hdspm *hdspm = (struct hdspm *) entry->private_data;
+ struct hdspm *hdspm = entry->private_data;
unsigned int status;
unsigned int status2;
char *pref_sync_ref;
@@ -2978,14 +2973,14 @@ snd_hdspm_proc_read_madi(struct snd_info_entry * entry,
(status & HDSPM_midi1IRQPending) ? 1 : 0,
hdspm->irq_count);
snd_iprintf(buffer,
- "HW pointer: id = %d, rawptr = %d (%d->%d) estimated= %ld (bytes)\n",
+ "HW pointer: id = %d, rawptr = %d (%d->%d) "
+ "estimated= %ld (bytes)\n",
((status & HDSPM_BufferID) ? 1 : 0),
(status & HDSPM_BufferPositionMask),
- (status & HDSPM_BufferPositionMask) % (2 *
- (int)hdspm->
- period_bytes),
- ((status & HDSPM_BufferPositionMask) -
- 64) % (2 * (int)hdspm->period_bytes),
+ (status & HDSPM_BufferPositionMask) %
+ (2 * (int)hdspm->period_bytes),
+ ((status & HDSPM_BufferPositionMask) - 64) %
+ (2 * (int)hdspm->period_bytes),
(long) hdspm_hw_pointer(hdspm) * 4);
snd_iprintf(buffer,
@@ -2995,24 +2990,22 @@ snd_hdspm_proc_read_madi(struct snd_info_entry * entry,
hdspm_read(hdspm, HDSPM_midiStatusIn0) & 0xFF,
hdspm_read(hdspm, HDSPM_midiStatusIn1) & 0xFF);
snd_iprintf(buffer,
- "Register: ctrl1=0x%x, ctrl2=0x%x, status1=0x%x, status2=0x%x\n",
+ "Register: ctrl1=0x%x, ctrl2=0x%x, status1=0x%x, "
+ "status2=0x%x\n",
hdspm->control_register, hdspm->control2_register,
status, status2);
snd_iprintf(buffer, "--- Settings ---\n");
- x = 1 << (6 +
- hdspm_decode_latency(hdspm->
- control_register &
- HDSPM_LatencyMask));
+ x = 1 << (6 + hdspm_decode_latency(hdspm->control_register &
+ HDSPM_LatencyMask));
snd_iprintf(buffer,
"Size (Latency): %d samples (2 periods of %lu bytes)\n",
x, (unsigned long) hdspm->period_bytes);
snd_iprintf(buffer, "Line out: %s, Precise Pointer: %s\n",
- (hdspm->
- control_register & HDSPM_LineOut) ? "on " : "off",
+ (hdspm->control_register & HDSPM_LineOut) ? "on " : "off",
(hdspm->precise_ptr) ? "on" : "off");
switch (hdspm->control_register & HDSPM_InputMask) {
@@ -3040,7 +3033,8 @@ snd_hdspm_proc_read_madi(struct snd_info_entry * entry,
syncref);
snd_iprintf(buffer,
- "ClearTrackMarker = %s, Transmit in %s Channel Mode, Auto Input %s\n",
+ "ClearTrackMarker = %s, Transmit in %s Channel Mode, "
+ "Auto Input %s\n",
(hdspm->
control_register & HDSPM_clr_tms) ? "on" : "off",
(hdspm->
@@ -3141,7 +3135,7 @@ static void
snd_hdspm_proc_read_aes32(struct snd_info_entry * entry,
struct snd_info_buffer *buffer)
{
- struct hdspm *hdspm = (struct hdspm *) entry->private_data;
+ struct hdspm *hdspm = entry->private_data;
unsigned int status;
unsigned int status2;
unsigned int timecode;
@@ -3171,14 +3165,14 @@ snd_hdspm_proc_read_aes32(struct snd_info_entry * entry,
(status & HDSPM_midi1IRQPending) ? 1 : 0,
hdspm->irq_count);
snd_iprintf(buffer,
- "HW pointer: id = %d, rawptr = %d (%d->%d) estimated= %ld (bytes)\n",
+ "HW pointer: id = %d, rawptr = %d (%d->%d) "
+ "estimated= %ld (bytes)\n",
((status & HDSPM_BufferID) ? 1 : 0),
(status & HDSPM_BufferPositionMask),
- (status & HDSPM_BufferPositionMask) % (2 *
- (int)hdspm->
- period_bytes),
- ((status & HDSPM_BufferPositionMask) -
- 64) % (2 * (int)hdspm->period_bytes),
+ (status & HDSPM_BufferPositionMask) %
+ (2 * (int)hdspm->period_bytes),
+ ((status & HDSPM_BufferPositionMask) - 64) %
+ (2 * (int)hdspm->period_bytes),
(long) hdspm_hw_pointer(hdspm) * 4);
snd_iprintf(buffer,
@@ -3188,16 +3182,15 @@ snd_hdspm_proc_read_aes32(struct snd_info_entry * entry,
hdspm_read(hdspm, HDSPM_midiStatusIn0) & 0xFF,
hdspm_read(hdspm, HDSPM_midiStatusIn1) & 0xFF);
snd_iprintf(buffer,
- "Register: ctrl1=0x%x, status1=0x%x, status2=0x%x, timecode=0x%x\n",
+ "Register: ctrl1=0x%x, status1=0x%x, status2=0x%x, "
+ "timecode=0x%x\n",
hdspm->control_register,
status, status2, timecode);
snd_iprintf(buffer, "--- Settings ---\n");
- x = 1 << (6 +
- hdspm_decode_latency(hdspm->
- control_register &
- HDSPM_LatencyMask));
+ x = 1 << (6 + hdspm_decode_latency(hdspm->control_register &
+ HDSPM_LatencyMask));
snd_iprintf(buffer,
"Size (Latency): %d samples (2 periods of %lu bytes)\n",
@@ -3280,14 +3273,15 @@ snd_hdspm_proc_read_aes32(struct snd_info_entry * entry,
snd_iprintf(buffer, "--- Status:\n");
snd_iprintf(buffer, "Word: %s Frequency: %d\n",
- (status & HDSPM_AES32_wcLock)? "Sync " : "No Lock",
- HDSPM_bit2freq((status >> HDSPM_AES32_wcFreq_bit) & 0xF));
+ (status & HDSPM_AES32_wcLock)? "Sync " : "No Lock",
+ HDSPM_bit2freq((status >> HDSPM_AES32_wcFreq_bit) & 0xF));
for (x = 0; x < 8; x++) {
snd_iprintf(buffer, "AES%d: %s Frequency: %d\n",
- x+1,
- (status2 & (HDSPM_LockAES >> x))? "Sync ": "No Lock",
- HDSPM_bit2freq((timecode >> (4*x)) & 0xF));
+ x+1,
+ (status2 & (HDSPM_LockAES >> x)) ?
+ "Sync ": "No Lock",
+ HDSPM_bit2freq((timecode >> (4*x)) & 0xF));
}
switch (hdspm_autosync_ref(hdspm)) {
@@ -3313,12 +3307,11 @@ static void
snd_hdspm_proc_read_debug(struct snd_info_entry * entry,
struct snd_info_buffer *buffer)
{
- struct hdspm *hdspm = (struct hdspm *)entry->private_data;
+ struct hdspm *hdspm = entry->private_data;
int j,i;
- for (i = 0; i < 256 /* 1024*64 */; i += j)
- {
+ for (i = 0; i < 256 /* 1024*64 */; i += j) {
snd_iprintf(buffer, "0x%08X: ", i);
for (j = 0; j < 16; j += 4)
snd_iprintf(buffer, "%08X ", hdspm_read(hdspm, i + j));
@@ -3361,14 +3354,20 @@ static int snd_hdspm_set_defaults(struct hdspm * hdspm)
/* set defaults: */
if (hdspm->is_aes32)
- hdspm->control_register = HDSPM_ClockModeMaster | /* Master Cloack Mode on */
- hdspm_encode_latency(7) | /* latency maximum = 8192 samples */
+ hdspm->control_register =
+ HDSPM_ClockModeMaster | /* Master Cloack Mode on */
+ hdspm_encode_latency(7) | /* latency maximum =
+ * 8192 samples
+ */
HDSPM_SyncRef0 | /* AES1 is syncclock */
HDSPM_LineOut | /* Analog output in */
HDSPM_Professional; /* Professional mode */
else
- hdspm->control_register = HDSPM_ClockModeMaster | /* Master Cloack Mode on */
- hdspm_encode_latency(7) | /* latency maximum = 8192 samples */
+ hdspm->control_register =
+ HDSPM_ClockModeMaster | /* Master Cloack Mode on */
+ hdspm_encode_latency(7) | /* latency maximum =
+ * 8192 samples
+ */
HDSPM_InputCoaxial | /* Input Coax not Optical */
HDSPM_SyncRef_MADI | /* Madi is syncclock */
HDSPM_LineOut | /* Analog output in */
@@ -3399,7 +3398,8 @@ static int snd_hdspm_set_defaults(struct hdspm * hdspm)
if (line_outs_monitor[hdspm->dev]) {
- snd_printk(KERN_INFO "HDSPM: sending all playback streams to line outs.\n");
+ snd_printk(KERN_INFO "HDSPM: "
+ "sending all playback streams to line outs.\n");
for (i = 0; i < HDSPM_MIXER_CHANNELS; i++) {
if (hdspm_write_pb_gain(hdspm, i, i, UNITY_GAIN))
@@ -3448,20 +3448,16 @@ static irqreturn_t snd_hdspm_interrupt(int irq, void *dev_id)
if (audio) {
if (hdspm->capture_substream)
- snd_pcm_period_elapsed(hdspm->pcm->
- streams
- [SNDRV_PCM_STREAM_CAPTURE].
- substream);
+ snd_pcm_period_elapsed(hdspm->capture_substream);
if (hdspm->playback_substream)
- snd_pcm_period_elapsed(hdspm->pcm->
- streams
- [SNDRV_PCM_STREAM_PLAYBACK].
- substream);
+ snd_pcm_period_elapsed(hdspm->playback_substream);
}
if (midi0 && midi0status) {
- /* we disable interrupts for this input until processing is done */
+ /* we disable interrupts for this input until processing
+ * is done
+ */
hdspm->control_register &= ~HDSPM_Midi0InterruptEnable;
hdspm_write(hdspm, HDSPM_controlRegister,
hdspm->control_register);
@@ -3469,7 +3465,9 @@ static irqreturn_t snd_hdspm_interrupt(int irq, void *dev_id)
schedule = 1;
}
if (midi1 && midi1status) {
- /* we disable interrupts for this input until processing is done */
+ /* we disable interrupts for this input until processing
+ * is done
+ */
hdspm->control_register &= ~HDSPM_Midi1InterruptEnable;
hdspm_write(hdspm, HDSPM_controlRegister,
hdspm->control_register);
@@ -3501,16 +3499,16 @@ static char *hdspm_channel_buffer_location(struct hdspm * hdspm,
snd_assert(channel >= 0
|| channel < HDSPM_MAX_CHANNELS, return NULL);
- if ((mapped_channel = hdspm->channel_map[channel]) < 0)
+ mapped_channel = hdspm->channel_map[channel];
+ if (mapped_channel < 0)
return NULL;
- if (stream == SNDRV_PCM_STREAM_CAPTURE) {
+ if (stream == SNDRV_PCM_STREAM_CAPTURE)
return hdspm->capture_buffer +
mapped_channel * HDSPM_CHANNEL_BUFFER_BYTES;
- } else {
+ else
return hdspm->playback_buffer +
mapped_channel * HDSPM_CHANNEL_BUFFER_BYTES;
- }
}
@@ -3525,9 +3523,9 @@ static int snd_hdspm_playback_copy(struct snd_pcm_substream *substream,
snd_assert(pos + count <= HDSPM_CHANNEL_BUFFER_BYTES / 4,
return -EINVAL);
- channel_buf = hdspm_channel_buffer_location(hdspm,
- substream->pstr->
- stream, channel);
+ channel_buf =
+ hdspm_channel_buffer_location(hdspm, substream->pstr->stream,
+ channel);
snd_assert(channel_buf != NULL, return -EIO);
@@ -3544,9 +3542,9 @@ static int snd_hdspm_capture_copy(struct snd_pcm_substream *substream,
snd_assert(pos + count <= HDSPM_CHANNEL_BUFFER_BYTES / 4,
return -EINVAL);
- channel_buf = hdspm_channel_buffer_location(hdspm,
- substream->pstr->
- stream, channel);
+ channel_buf =
+ hdspm_channel_buffer_location(hdspm, substream->pstr->stream,
+ channel);
snd_assert(channel_buf != NULL, return -EIO);
return copy_to_user(dst, channel_buf + pos * 4, count * 4);
}
@@ -3559,8 +3557,8 @@ static int snd_hdspm_hw_silence(struct snd_pcm_substream *substream,
char *channel_buf;
channel_buf =
- hdspm_channel_buffer_location(hdspm, substream->pstr->stream,
- channel);
+ hdspm_channel_buffer_location(hdspm, substream->pstr->stream,
+ channel);
snd_assert(channel_buf != NULL, return -EIO);
memset(channel_buf + pos * 4, 0, count * 4);
return 0;
@@ -3616,7 +3614,7 @@ static int snd_hdspm_hw_params(struct snd_pcm_substream *substream,
other_pid = hdspm->playback_pid;
}
- if ((other_pid > 0) && (this_pid != other_pid)) {
+ if (other_pid > 0 && this_pid != other_pid) {
/* The other stream is open, and not by the same
task as this one. Make sure that the parameters
@@ -3633,7 +3631,7 @@ static int snd_hdspm_hw_params(struct snd_pcm_substream *substream,
if (params_period_size(params) != hdspm->period_bytes / 4) {
spin_unlock_irq(&hdspm->lock);
_snd_pcm_hw_param_setempty(params,
- SNDRV_PCM_HW_PARAM_PERIOD_SIZE);
+ SNDRV_PCM_HW_PARAM_PERIOD_SIZE);
return -EBUSY;
}
@@ -3644,7 +3642,8 @@ static int snd_hdspm_hw_params(struct snd_pcm_substream *substream,
/* how to make sure that the rate matches an externally-set one ? */
spin_lock_irq(&hdspm->lock);
- if ((err = hdspm_set_rate(hdspm, params_rate(params), 0)) < 0) {
+ err = hdspm_set_rate(hdspm, params_rate(params), 0);
+ if (err < 0) {
spin_unlock_irq(&hdspm->lock);
_snd_pcm_hw_param_setempty(params,
SNDRV_PCM_HW_PARAM_RATE);
@@ -3652,16 +3651,17 @@ static int snd_hdspm_hw_params(struct snd_pcm_substream *substream,
}
spin_unlock_irq(&hdspm->lock);
- if ((err =
- hdspm_set_interrupt_interval(hdspm,
- params_period_size(params))) <
- 0) {
+ err = hdspm_set_interrupt_interval(hdspm,
+ params_period_size(params));
+ if (err < 0) {
_snd_pcm_hw_param_setempty(params,
SNDRV_PCM_HW_PARAM_PERIOD_SIZE);
return err;
}
- /* Memory allocation, takashi's method, dont know if we should spinlock */
+ /* Memory allocation, takashi's method, dont know if we should
+ * spinlock
+ */
/* malloc all buffer even if not enabled to get sure */
/* Update for MADI rev 204: we need to allocate for all channels,
* otherwise it doesn't work at 96kHz */
@@ -3746,7 +3746,8 @@ static int snd_hdspm_channel_info(struct snd_pcm_substream *substream,
snd_assert(info->channel < HDSPM_MAX_CHANNELS, return -EINVAL);
- if ((mapped_channel = hdspm->channel_map[info->channel]) < 0)
+ mapped_channel = hdspm->channel_map[info->channel];
+ if (mapped_channel < 0)
return -EINVAL;
info->offset = mapped_channel * HDSPM_CHANNEL_BUFFER_BYTES;
@@ -3760,15 +3761,13 @@ static int snd_hdspm_ioctl(struct snd_pcm_substream *substream,
{
switch (cmd) {
case SNDRV_PCM_IOCTL1_RESET:
- {
- return snd_hdspm_reset(substream);
- }
+ return snd_hdspm_reset(substream);
case SNDRV_PCM_IOCTL1_CHANNEL_INFO:
- {
- struct snd_pcm_channel_info *info = arg;
- return snd_hdspm_channel_info(substream, info);
- }
+ {
+ struct snd_pcm_channel_info *info = arg;
+ return snd_hdspm_channel_info(substream, info);
+ }
default:
break;
}
@@ -3979,9 +3978,12 @@ static int snd_hdspm_hw_rule_channels(struct snd_pcm_hw_params *params,
}
-static unsigned int hdspm_aes32_sample_rates[] = { 32000, 44100, 48000, 64000, 88200, 96000, 128000, 176400, 192000 };
+static unsigned int hdspm_aes32_sample_rates[] = {
+ 32000, 44100, 48000, 64000, 88200, 96000, 128000, 176400, 192000
+};
-static struct snd_pcm_hw_constraint_list hdspm_hw_constraints_aes32_sample_rates = {
+static struct snd_pcm_hw_constraint_list
+hdspm_hw_constraints_aes32_sample_rates = {
.count = ARRAY_SIZE(hdspm_aes32_sample_rates),
.list = hdspm_aes32_sample_rates,
.mask = 0
@@ -4107,7 +4109,7 @@ static int snd_hdspm_hwdep_dummy_op(struct snd_hwdep * hw, struct file *file)
static int snd_hdspm_hwdep_ioctl(struct snd_hwdep * hw, struct file *file,
unsigned int cmd, unsigned long arg)
{
- struct hdspm *hdspm = (struct hdspm *) hw->private_data;
+ struct hdspm *hdspm = hw->private_data;
struct hdspm_mixer_ioctl mixer;
struct hdspm_config_info info;
struct hdspm_version hdspm_version;
@@ -4115,11 +4117,12 @@ static int snd_hdspm_hwdep_ioctl(struct snd_hwdep * hw, struct file *file,
switch (cmd) {
-
case SNDRV_HDSPM_IOCTL_GET_PEAK_RMS:
if (copy_from_user(&rms, (void __user *)arg, sizeof(rms)))
return -EFAULT;
- /* maybe there is a chance to memorymap in future so dont touch just copy */
+ /* maybe there is a chance to memorymap in future
+ * so dont touch just copy
+ */
if(copy_to_user_fromio((void __user *)rms.peak,
hdspm->iobase+HDSPM_MADI_peakrmsbase,
sizeof(struct hdspm_peak_rms)) != 0 )
@@ -4131,21 +4134,16 @@ static int snd_hdspm_hwdep_ioctl(struct snd_hwdep * hw, struct file *file,
case SNDRV_HDSPM_IOCTL_GET_CONFIG_INFO:
spin_lock_irq(&hdspm->lock);
- info.pref_sync_ref =
- (unsigned char) hdspm_pref_sync_ref(hdspm);
- info.wordclock_sync_check =
- (unsigned char) hdspm_wc_sync_check(hdspm);
+ info.pref_sync_ref = hdspm_pref_sync_ref(hdspm);
+ info.wordclock_sync_check = hdspm_wc_sync_check(hdspm);
info.system_sample_rate = hdspm->system_sample_rate;
info.autosync_sample_rate =
hdspm_external_sample_rate(hdspm);
- info.system_clock_mode =
- (unsigned char) hdspm_system_clock_mode(hdspm);
- info.clock_source =
- (unsigned char) hdspm_clock_source(hdspm);
- info.autosync_ref =
- (unsigned char) hdspm_autosync_ref(hdspm);
- info.line_out = (unsigned char) hdspm_line_out(hdspm);
+ info.system_clock_mode = hdspm_system_clock_mode(hdspm);
+ info.clock_source = hdspm_clock_source(hdspm);
+ info.autosync_ref = hdspm_autosync_ref(hdspm);
+ info.line_out = hdspm_line_out(hdspm);
info.passthru = 0;
spin_unlock_irq(&hdspm->lock);
if (copy_to_user((void __user *) arg, &info, sizeof(info)))
@@ -4162,8 +4160,8 @@ static int snd_hdspm_hwdep_ioctl(struct snd_hwdep * hw, struct file *file,
case SNDRV_HDSPM_IOCTL_GET_MIXER:
if (copy_from_user(&mixer, (void __user *)arg, sizeof(mixer)))
return -EFAULT;
- if (copy_to_user
- ((void __user *)mixer.mixer, hdspm->mixer, sizeof(struct hdspm_mixer)))
+ if (copy_to_user((void __user *)mixer.mixer, hdspm->mixer,
+ sizeof(struct hdspm_mixer)))
return -EFAULT;
break;
@@ -4206,7 +4204,8 @@ static int __devinit snd_hdspm_create_hwdep(struct snd_card *card,
struct snd_hwdep *hw;
int err;
- if ((err = snd_hwdep_new(card, "HDSPM hwdep", 0, &hw)) < 0)
+ err = snd_hwdep_new(card, "HDSPM hwdep", 0, &hw);
+ if (err < 0)
return err;
hdspm->hwdep = hw;
@@ -4232,15 +4231,15 @@ static int __devinit snd_hdspm_preallocate_memory(struct hdspm * hdspm)
pcm = hdspm->pcm;
-/* wanted = HDSPM_DMA_AREA_BYTES + 4096;*/ /* dont know why, but it works */
wanted = HDSPM_DMA_AREA_BYTES;
- if ((err =
+ err =
snd_pcm_lib_preallocate_pages_for_all(pcm,
SNDRV_DMA_TYPE_DEV_SG,
snd_dma_pci_data(hdspm->pci),
wanted,
- wanted)) < 0) {
+ wanted);
+ if (err < 0) {
snd_printdd("Could not preallocate %zd Bytes\n", wanted);
return err;
@@ -4256,8 +4255,7 @@ static void hdspm_set_sgbuf(struct hdspm * hdspm, struct snd_sg_buf *sgbuf,
int i;
for (i = 0; i < (channels * 16); i++)
hdspm_write(hdspm, reg + 4 * i,
- snd_pcm_sgbuf_get_addr(sgbuf,
- (size_t) 4096 * i));
+ snd_pcm_sgbuf_get_addr(sgbuf, (size_t) 4096 * i));
}
/* ------------- ALSA Devices ---------------------------- */
@@ -4267,7 +4265,8 @@ static int __devinit snd_hdspm_create_pcm(struct snd_card *card,
struct snd_pcm *pcm;
int err;
- if ((err = snd_pcm_new(card, hdspm->card_name, 0, 1, 1, &pcm)) < 0)
+ err = snd_pcm_new(card, hdspm->card_name, 0, 1, 1, &pcm);
+ if (err < 0)
return err;
hdspm->pcm = pcm;
@@ -4281,7 +4280,8 @@ static int __devinit snd_hdspm_create_pcm(struct snd_card *card,
pcm->info_flags = SNDRV_PCM_INFO_JOINT_DUPLEX;
- if ((err = snd_hdspm_preallocate_memory(hdspm)) < 0)
+ err = snd_hdspm_preallocate_memory(hdspm);
+ if (err < 0)
return err;
return 0;
@@ -4299,19 +4299,24 @@ static int __devinit snd_hdspm_create_alsa_devices(struct snd_card *card,
int err;
snd_printdd("Create card...\n");
- if ((err = snd_hdspm_create_pcm(card, hdspm)) < 0)
+ err = snd_hdspm_create_pcm(card, hdspm);
+ if (err < 0)
return err;