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-rw-r--r--sound/aoa/codecs/tas.c9
-rw-r--r--sound/arm/aaci.c1
-rw-r--r--sound/core/isadma.c10
-rw-r--r--sound/core/oss/mixer_oss.c4
-rw-r--r--sound/core/pcm.c5
-rw-r--r--sound/drivers/dummy.c2
-rw-r--r--sound/drivers/opl3/opl3_midi.c28
-rw-r--r--sound/drivers/pcsp/pcsp.c32
-rw-r--r--sound/drivers/pcsp/pcsp.h2
-rw-r--r--sound/drivers/pcsp/pcsp_lib.c65
-rw-r--r--sound/drivers/pcsp/pcsp_mixer.c37
-rw-r--r--sound/isa/Kconfig12
-rw-r--r--sound/isa/cmi8330.c4
-rw-r--r--sound/isa/es1688/es1688_lib.c2
-rw-r--r--sound/isa/es18xx.c132
-rw-r--r--sound/isa/sb/sb_mixer.c4
-rw-r--r--sound/isa/sscape.c727
-rw-r--r--sound/isa/wss/wss_lib.c98
-rw-r--r--sound/mips/hal2.c2
-rw-r--r--sound/mips/sgio2audio.c2
-rw-r--r--sound/oss/Kconfig12
-rw-r--r--sound/oss/Makefile1
-rw-r--r--sound/oss/sh_dac_audio.c3
-rw-r--r--sound/oss/sscape.c1480
-rw-r--r--sound/parisc/harmony.c6
-rw-r--r--sound/pci/Kconfig1
-rw-r--r--sound/pci/ac97/ac97_codec.c6
-rw-r--r--sound/pci/ac97/ac97_patch.c12
-rw-r--r--sound/pci/ali5451/ali5451.c2
-rw-r--r--sound/pci/azt3328.c4
-rw-r--r--sound/pci/bt87x.c2
-rw-r--r--sound/pci/ca0106/ca0106_mixer.c4
-rw-r--r--sound/pci/ca0106/ca0106_proc.c4
-rw-r--r--sound/pci/cmipci.c4
-rw-r--r--sound/pci/ctxfi/ctatc.c6
-rw-r--r--sound/pci/echoaudio/echoaudio.c30
-rw-r--r--sound/pci/echoaudio/mia.c1
-rw-r--r--sound/pci/emu10k1/emu10k1x.c3
-rw-r--r--sound/pci/emu10k1/emumixer.c4
-rw-r--r--sound/pci/emu10k1/emuproc.c4
-rw-r--r--sound/pci/emu10k1/io.c2
-rw-r--r--sound/pci/es1938.c2
-rw-r--r--sound/pci/hda/Kconfig13
-rw-r--r--sound/pci/hda/hda_beep.c114
-rw-r--r--sound/pci/hda/hda_beep.h10
-rw-r--r--sound/pci/hda/hda_codec.c607
-rw-r--r--sound/pci/hda/hda_codec.h11
-rw-r--r--sound/pci/hda/hda_eld.c20
-rw-r--r--sound/pci/hda/hda_generic.c17
-rw-r--r--sound/pci/hda/hda_hwdep.c38
-rw-r--r--sound/pci/hda/hda_intel.c62
-rw-r--r--sound/pci/hda/hda_local.h69
-rw-r--r--sound/pci/hda/hda_proc.c70
-rw-r--r--sound/pci/hda/patch_analog.c192
-rw-r--r--sound/pci/hda/patch_ca0110.c4
-rw-r--r--sound/pci/hda/patch_cirrus.c31
-rw-r--r--sound/pci/hda/patch_cmedia.c4
-rw-r--r--sound/pci/hda/patch_conexant.c217
-rw-r--r--sound/pci/hda/patch_intelhdmi.c488
-rw-r--r--sound/pci/hda/patch_nvhdmi.c33
-rw-r--r--sound/pci/hda/patch_realtek.c731
-rw-r--r--sound/pci/hda/patch_sigmatel.c268
-rw-r--r--sound/pci/hda/patch_via.c3509
-rw-r--r--sound/pci/ice1712/amp.c8
-rw-r--r--sound/pci/ice1712/ice1712.c2
-rw-r--r--sound/pci/ice1712/ice1724.c8
-rw-r--r--sound/pci/intel8x0.c12
-rw-r--r--sound/pci/via82xx.c86
-rw-r--r--sound/ppc/awacs.c12
-rw-r--r--sound/ppc/burgundy.c8
-rw-r--r--sound/ppc/keywest.c14
-rw-r--r--sound/ppc/tumbler.c2
-rw-r--r--sound/sh/Kconfig8
-rw-r--r--sound/sh/Makefile2
-rw-r--r--sound/sh/sh_dac_audio.c453
-rw-r--r--sound/soc/blackfin/Kconfig98
-rw-r--r--sound/soc/blackfin/bf5xx-i2s.c8
-rw-r--r--sound/soc/blackfin/bf5xx-tdm.c8
-rw-r--r--sound/soc/codecs/tlv320aic23.c2
-rw-r--r--sound/soc/codecs/wm8350.c4
-rw-r--r--sound/soc/codecs/wm8940.c2
-rw-r--r--sound/soc/codecs/wm9713.c22
-rw-r--r--sound/soc/davinci/davinci-i2s.c37
-rw-r--r--sound/soc/davinci/davinci-mcasp.c80
-rw-r--r--sound/soc/davinci/davinci-mcasp.h7
-rw-r--r--sound/soc/davinci/davinci-pcm.c13
-rw-r--r--sound/soc/davinci/davinci-pcm.h1
-rw-r--r--sound/soc/imx/mxc-ssi.c8
-rw-r--r--sound/soc/pxa/Kconfig2
-rw-r--r--sound/soc/soc-dapm.c5
-rw-r--r--sound/usb/caiaq/audio.c16
-rw-r--r--sound/usb/caiaq/device.c2
-rw-r--r--sound/usb/usbmixer.c23
-rw-r--r--sound/usb/usx2y/us122l.c75
94 files changed, 6862 insertions, 3445 deletions
diff --git a/sound/aoa/codecs/tas.c b/sound/aoa/codecs/tas.c
index f0ebc971c68..1dd66ddffca 100644
--- a/sound/aoa/codecs/tas.c
+++ b/sound/aoa/codecs/tas.c
@@ -897,6 +897,15 @@ static int tas_create(struct i2c_adapter *adapter,
client = i2c_new_device(adapter, &info);
if (!client)
return -ENODEV;
+ /*
+ * We know the driver is already loaded, so the device should be
+ * already bound. If not it means binding failed, and then there
+ * is no point in keeping the device instantiated.
+ */
+ if (!client->driver) {
+ i2c_unregister_device(client);
+ return -ENODEV;
+ }
/*
* Let i2c-core delete that device on driver removal.
diff --git a/sound/arm/aaci.c b/sound/arm/aaci.c
index dc78272fc39..1f0f8213e2d 100644
--- a/sound/arm/aaci.c
+++ b/sound/arm/aaci.c
@@ -937,6 +937,7 @@ static int __devinit aaci_probe_ac97(struct aaci *aaci)
struct snd_ac97 *ac97;
int ret;
+ writel(0, aaci->base + AC97_POWERDOWN);
/*
* Assert AACIRESET for 2us
*/
diff --git a/sound/core/isadma.c b/sound/core/isadma.c
index 79f0f16af33..950e19ba91f 100644
--- a/sound/core/isadma.c
+++ b/sound/core/isadma.c
@@ -85,16 +85,24 @@ EXPORT_SYMBOL(snd_dma_disable);
unsigned int snd_dma_pointer(unsigned long dma, unsigned int size)
{
unsigned long flags;
- unsigned int result;
+ unsigned int result, result1;
flags = claim_dma_lock();
clear_dma_ff(dma);
if (!isa_dma_bridge_buggy)
disable_dma(dma);
result = get_dma_residue(dma);
+ /*
+ * HACK - read the counter again and choose higher value in order to
+ * avoid reading during counter lower byte roll over if the
+ * isa_dma_bridge_buggy is set.
+ */
+ result1 = get_dma_residue(dma);
if (!isa_dma_bridge_buggy)
enable_dma(dma);
release_dma_lock(flags);
+ if (unlikely(result < result1))
+ result = result1;
#ifdef CONFIG_SND_DEBUG
if (result > size)
snd_printk(KERN_ERR "pointer (0x%x) for DMA #%ld is greater than transfer size (0x%x)\n", result, dma, size);
diff --git a/sound/core/oss/mixer_oss.c b/sound/core/oss/mixer_oss.c
index 772423889eb..54e2eb56e4c 100644
--- a/sound/core/oss/mixer_oss.c
+++ b/sound/core/oss/mixer_oss.c
@@ -1251,7 +1251,9 @@ static void snd_mixer_oss_build(struct snd_mixer_oss *mixer)
{ SOUND_MIXER_SYNTH, "FM", 0 }, /* fallback */
{ SOUND_MIXER_SYNTH, "Music", 0 }, /* fallback */
{ SOUND_MIXER_PCM, "PCM", 0 },
- { SOUND_MIXER_SPEAKER, "PC Speaker", 0 },
+ { SOUND_MIXER_SPEAKER, "Beep", 0 },
+ { SOUND_MIXER_SPEAKER, "PC Speaker", 0 }, /* fallback */
+ { SOUND_MIXER_SPEAKER, "Speaker", 0 }, /* fallback */
{ SOUND_MIXER_LINE, "Line", 0 },
{ SOUND_MIXER_MIC, "Mic", 0 },
{ SOUND_MIXER_CD, "CD", 0 },
diff --git a/sound/core/pcm.c b/sound/core/pcm.c
index 0c1440121c2..c69c60b2a48 100644
--- a/sound/core/pcm.c
+++ b/sound/core/pcm.c
@@ -953,11 +953,12 @@ static int snd_pcm_dev_register(struct snd_device *device)
struct snd_pcm_substream *substream;
struct snd_pcm_notify *notify;
char str[16];
- struct snd_pcm *pcm = device->device_data;
+ struct snd_pcm *pcm;
struct device *dev;
- if (snd_BUG_ON(!pcm || !device))
+ if (snd_BUG_ON(!device || !device->device_data))
return -ENXIO;
+ pcm = device->device_data;
mutex_lock(&register_mutex);
err = snd_pcm_add(pcm);
if (err) {
diff --git a/sound/drivers/dummy.c b/sound/drivers/dummy.c
index 6ba066c41d2..146ef00f94a 100644
--- a/sound/drivers/dummy.c
+++ b/sound/drivers/dummy.c
@@ -808,8 +808,6 @@ static int __devinit snd_card_dummy_new_mixer(struct snd_dummy *dummy)
unsigned int idx;
int err;
- if (snd_BUG_ON(!dummy))
- return -EINVAL;
spin_lock_init(&dummy->mixer_lock);
strcpy(card->mixername, "Dummy Mixer");
diff --git a/sound/drivers/opl3/opl3_midi.c b/sound/drivers/opl3/opl3_midi.c
index 6e7d09ae0e8..7d722a025d0 100644
--- a/sound/drivers/opl3/opl3_midi.c
+++ b/sound/drivers/opl3/opl3_midi.c
@@ -29,6 +29,8 @@ extern char snd_opl3_regmap[MAX_OPL2_VOICES][4];
extern int use_internal_drums;
+static void snd_opl3_note_off_unsafe(void *p, int note, int vel,
+ struct snd_midi_channel *chan);
/*
* The next table looks magical, but it certainly is not. Its values have
* been calculated as table[i]=8*log(i/64)/log(2) with an obvious exception
@@ -242,16 +244,20 @@ void snd_opl3_timer_func(unsigned long data)
int again = 0;
int i;
- spin_lock_irqsave(&opl3->sys_timer_lock, flags);
+ spin_lock_irqsave(&opl3->voice_lock, flags);
for (i = 0; i < opl3->max_voices; i++) {
struct snd_opl3_voice *vp = &opl3->voices[i];
if (vp->state > 0 && vp->note_off_check) {
if (vp->note_off == jiffies)
- snd_opl3_note_off(opl3, vp->note, 0, vp->chan);
+ snd_opl3_note_off_unsafe(opl3, vp->note, 0,
+ vp->chan);
else
again++;
}
}
+ spin_unlock_irqrestore(&opl3->voice_lock, flags);
+
+ spin_lock_irqsave(&opl3->sys_timer_lock, flags);
if (again) {
opl3->tlist.expires = jiffies + 1; /* invoke again */
add_timer(&opl3->tlist);
@@ -658,15 +664,14 @@ static void snd_opl3_kill_voice(struct snd_opl3 *opl3, int voice)
/*
* Release a note in response to a midi note off.
*/
-void snd_opl3_note_off(void *p, int note, int vel, struct snd_midi_channel *chan)
+static void snd_opl3_note_off_unsafe(void *p, int note, int vel,
+ struct snd_midi_channel *chan)
{
struct snd_opl3 *opl3;
int voice;
struct snd_opl3_voice *vp;
- unsigned long flags;
-
opl3 = p;
#ifdef DEBUG_MIDI
@@ -674,12 +679,9 @@ void snd_opl3_note_off(void *p, int note, int vel, struct snd_midi_channel *chan
chan->number, chan->midi_program, note);
#endif
- spin_lock_irqsave(&opl3->voice_lock, flags);
-
if (opl3->synth_mode == SNDRV_OPL3_MODE_SEQ) {
if (chan->drum_channel && use_internal_drums) {
snd_opl3_drum_switch(opl3, note, vel, 0, chan);
- spin_unlock_irqrestore(&opl3->voice_lock, flags);
return;
}
/* this loop will hopefully kill all extra voices, because
@@ -697,6 +699,16 @@ void snd_opl3_note_off(void *p, int note, int vel, struct snd_midi_channel *chan
snd_opl3_kill_voice(opl3, voice);
}
}
+}
+
+void snd_opl3_note_off(void *p, int note, int vel,
+ struct snd_midi_channel *chan)
+{
+ struct snd_opl3 *opl3 = p;
+ unsigned long flags;
+
+ spin_lock_irqsave(&opl3->voice_lock, flags);
+ snd_opl3_note_off_unsafe(p, note, vel, chan);
spin_unlock_irqrestore(&opl3->voice_lock, flags);
}
diff --git a/sound/drivers/pcsp/pcsp.c b/sound/drivers/pcsp/pcsp.c
index b60cef257b5..f165c77d627 100644
--- a/sound/drivers/pcsp/pcsp.c
+++ b/sound/drivers/pcsp/pcsp.c
@@ -26,6 +26,7 @@ MODULE_ALIAS("platform:pcspkr");
static int index = SNDRV_DEFAULT_IDX1; /* Index 0-MAX */
static char *id = SNDRV_DEFAULT_STR1; /* ID for this card */
static int enable = SNDRV_DEFAULT_ENABLE1; /* Enable this card */
+static int nopcm; /* Disable PCM capability of the driver */
module_param(index, int, 0444);
MODULE_PARM_DESC(index, "Index value for pcsp soundcard.");
@@ -33,6 +34,8 @@ module_param(id, charp, 0444);
MODULE_PARM_DESC(id, "ID string for pcsp soundcard.");
module_param(enable, bool, 0444);
MODULE_PARM_DESC(enable, "Enable PC-Speaker sound.");
+module_param(nopcm, bool, 0444);
+MODULE_PARM_DESC(nopcm, "Disable PC-Speaker PCM sound. Only beeps remain.");
struct snd_pcsp pcsp_chip;
@@ -43,13 +46,16 @@ static int __devinit snd_pcsp_create(struct snd_card *card)
int err;
int div, min_div, order;
- hrtimer_get_res(CLOCK_MONOTONIC, &tp);
- if (tp.tv_sec || tp.tv_nsec > PCSP_MAX_PERIOD_NS) {
- printk(KERN_ERR "PCSP: Timer resolution is not sufficient "
- "(%linS)\n", tp.tv_nsec);
- printk(KERN_ERR "PCSP: Make sure you have HPET and ACPI "
- "enabled.\n");
- return -EIO;
+ if (!nopcm) {
+ hrtimer_get_res(CLOCK_MONOTONIC, &tp);
+ if (tp.tv_sec || tp.tv_nsec > PCSP_MAX_PERIOD_NS) {
+ printk(KERN_ERR "PCSP: Timer resolution is not sufficient "
+ "(%linS)\n", tp.tv_nsec);
+ printk(KERN_ERR "PCSP: Make sure you have HPET and ACPI "
+ "enabled.\n");
+ printk(KERN_ERR "PCSP: Turned into nopcm mode.\n");
+ nopcm = 1;
+ }
}
if (loops_per_jiffy >= PCSP_MIN_LPJ && tp.tv_nsec <= PCSP_MIN_PERIOD_NS)
@@ -107,12 +113,14 @@ static int __devinit snd_card_pcsp_probe(int devnum, struct device *dev)
snd_card_free(card);
return err;
}
- err = snd_pcsp_new_pcm(&pcsp_chip);
- if (err < 0) {
- snd_card_free(card);
- return err;
+ if (!nopcm) {
+ err = snd_pcsp_new_pcm(&pcsp_chip);
+ if (err < 0) {
+ snd_card_free(card);
+ return err;
+ }
}
- err = snd_pcsp_new_mixer(&pcsp_chip);
+ err = snd_pcsp_new_mixer(&pcsp_chip, nopcm);
if (err < 0) {
snd_card_free(card);
return err;
diff --git a/sound/drivers/pcsp/pcsp.h b/sound/drivers/pcsp/pcsp.h
index 174dd2ff0f2..1e123077923 100644
--- a/sound/drivers/pcsp/pcsp.h
+++ b/sound/drivers/pcsp/pcsp.h
@@ -83,6 +83,6 @@ extern enum hrtimer_restart pcsp_do_timer(struct hrtimer *handle);
extern void pcsp_sync_stop(struct snd_pcsp *chip);
extern int snd_pcsp_new_pcm(struct snd_pcsp *chip);
-extern int snd_pcsp_new_mixer(struct snd_pcsp *chip);
+extern int snd_pcsp_new_mixer(struct snd_pcsp *chip, int nopcm);
#endif
diff --git a/sound/drivers/pcsp/pcsp_lib.c b/sound/drivers/pcsp/pcsp_lib.c
index 84cc2658c05..e1145ac6e90 100644
--- a/sound/drivers/pcsp/pcsp_lib.c
+++ b/sound/drivers/pcsp/pcsp_lib.c
@@ -39,25 +39,20 @@ static DECLARE_TASKLET(pcsp_pcm_tasklet, pcsp_call_pcm_elapsed, 0);
/* write the port and returns the next expire time in ns;
* called at the trigger-start and in hrtimer callback
*/
-static unsigned long pcsp_timer_update(struct hrtimer *handle)
+static u64 pcsp_timer_update(struct snd_pcsp *chip)
{
unsigned char timer_cnt, val;
u64 ns;
struct snd_pcm_substream *substream;
struct snd_pcm_runtime *runtime;
- struct snd_pcsp *chip = container_of(handle, struct snd_pcsp, timer);
unsigned long flags;
if (chip->thalf) {
outb(chip->val61, 0x61);
chip->thalf = 0;
- if (!atomic_read(&chip->timer_active))
- return 0;
return chip->ns_rem;
}
- if (!atomic_read(&chip->timer_active))
- return 0;
substream = chip->playback_substream;
if (!substream)
return 0;
@@ -88,24 +83,17 @@ static unsigned long pcsp_timer_update(struct hrtimer *handle)
return ns;
}
-enum hrtimer_restart pcsp_do_timer(struct hrtimer *handle)
+static void pcsp_pointer_update(struct snd_pcsp *chip)
{
- struct snd_pcsp *chip = container_of(handle, struct snd_pcsp, timer);
struct snd_pcm_substream *substream;
- int periods_elapsed, pointer_update;
size_t period_bytes, buffer_bytes;
- unsigned long ns;
+ int periods_elapsed;
unsigned long flags;
- pointer_update = !chip->thalf;
- ns = pcsp_timer_update(handle);
- if (!ns)
- return HRTIMER_NORESTART;
-
/* update the playback position */
substream = chip->playback_substream;
if (!substream)
- return HRTIMER_NORESTART;
+ return;
period_bytes = snd_pcm_lib_period_bytes(substream);
buffer_bytes = snd_pcm_lib_buffer_bytes(substream);
@@ -134,6 +122,26 @@ enum hrtimer_restart pcsp_do_timer(struct hrtimer *handle)
if (periods_elapsed)
tasklet_schedule(&pcsp_pcm_tasklet);
+}
+
+enum hrtimer_restart pcsp_do_timer(struct hrtimer *handle)
+{
+ struct snd_pcsp *chip = container_of(handle, struct snd_pcsp, timer);
+ int pointer_update;
+ u64 ns;
+
+ if (!atomic_read(&chip->timer_active) || !chip->playback_substream)
+ return HRTIMER_NORESTART;
+
+ pointer_update = !chip->thalf;
+ ns = pcsp_timer_update(chip);
+ if (!ns) {
+ printk(KERN_WARNING "PCSP: unexpected stop\n");
+ return HRTIMER_NORESTART;
+ }
+
+ if (pointer_update)
+ pcsp_pointer_update(chip);
hrtimer_forward(handle, hrtimer_get_expires(handle), ns_to_ktime(ns));
@@ -142,8 +150,6 @@ enum hrtimer_restart pcsp_do_timer(struct hrtimer *handle)
static int pcsp_start_playing(struct snd_pcsp *chip)
{
- unsigned long ns;
-
#if PCSP_DEBUG
printk(KERN_INFO "PCSP: start_playing called\n");
#endif
@@ -159,11 +165,7 @@ static int pcsp_start_playing(struct snd_pcsp *chip)
atomic_set(&chip->timer_active, 1);
chip->thalf = 0;
- ns = pcsp_timer_update(&pcsp_chip.timer);
- if (!ns)
- return -EIO;
-
- hrtimer_start(&pcsp_chip.timer, ktime_set(0, ns), HRTIMER_MODE_REL);
+ hrtimer_start(&pcsp_chip.timer, ktime_set(0, 0), HRTIMER_MODE_REL);
return 0;
}
@@ -232,21 +234,22 @@ static int snd_pcsp_playback_hw_free(struct snd_pcm_substream *substream)
static int snd_pcsp_playback_prepare(struct snd_pcm_substream *substream)
{
struct snd_pcsp *chip = snd_pcm_substream_chip(substream);
+ pcsp_sync_stop(chip);
+ chip->playback_ptr = 0;
+ chip->period_ptr = 0;
+ chip->fmt_size =
+ snd_pcm_format_physical_width(substream->runtime->format) >> 3;
+ chip->is_signed = snd_pcm_format_signed(substream->runtime->format);
#if PCSP_DEBUG
printk(KERN_INFO "PCSP: prepare called, "
- "size=%zi psize=%zi f=%zi f1=%i\n",
+ "size=%zi psize=%zi f=%zi f1=%i fsize=%i\n",
snd_pcm_lib_buffer_bytes(substream),
snd_pcm_lib_period_bytes(substream),
snd_pcm_lib_buffer_bytes(substream) /
snd_pcm_lib_period_bytes(substream),
- substream->runtime->periods);
+ substream->runtime->periods,
+ chip->fmt_size);
#endif
- pcsp_sync_stop(chip);
- chip->playback_ptr = 0;
- chip->period_ptr = 0;
- chip->fmt_size =
- snd_pcm_format_physical_width(substream->runtime->format) >> 3;
- chip->is_signed = snd_pcm_format_signed(substream->runtime->format);
return 0;
}
diff --git a/sound/drivers/pcsp/pcsp_mixer.c b/sound/drivers/pcsp/pcsp_mixer.c
index 199b0337714..6f633f4f3b9 100644
--- a/sound/drivers/pcsp/pcsp_mixer.c
+++ b/sound/drivers/pcsp/pcsp_mixer.c
@@ -72,7 +72,7 @@ static int pcsp_treble_put(struct snd_kcontrol *kcontrol,
if (treble != chip->treble) {
chip->treble = treble;
#if PCSP_DEBUG
- printk(KERN_INFO "PCSP: rate set to %i\n", PCSP_RATE());
+ printk(KERN_INFO "PCSP: rate set to %li\n", PCSP_RATE());
#endif
changed = 1;
}
@@ -119,24 +119,43 @@ static int pcsp_pcspkr_put(struct snd_kcontrol *kcontrol,
.put = pcsp_##ctl_type##_put, \
}
-static struct snd_kcontrol_new __devinitdata snd_pcsp_controls[] = {
+static struct snd_kcontrol_new __devinitdata snd_pcsp_controls_pcm[] = {
PCSP_MIXER_CONTROL(enable, "Master Playback Switch"),
PCSP_MIXER_CONTROL(treble, "BaseFRQ Playback Volume"),
- PCSP_MIXER_CONTROL(pcspkr, "PC Speaker Playback Switch"),
};
-int __devinit snd_pcsp_new_mixer(struct snd_pcsp *chip)
+static struct snd_kcontrol_new __devinitdata snd_pcsp_controls_spkr[] = {
+ PCSP_MIXER_CONTROL(pcspkr, "Beep Playback Switch"),
+};
+
+static int __devinit snd_pcsp_ctls_add(struct snd_pcsp *chip,
+ struct snd_kcontrol_new *ctls, int num)
{
- struct snd_card *card = chip->card;
int i, err;
+ struct snd_card *card = chip->card;
+ for (i = 0; i < num; i++) {
+ err = snd_ctl_add(card, snd_ctl_new1(ctls + i, chip));
+ if (err < 0)
+ return err;
+ }
+ return 0;
+}
+
+int __devinit snd_pcsp_new_mixer(struct snd_pcsp *chip, int nopcm)
+{
+ int err;
+ struct snd_card *card = chip->card;
- for (i = 0; i < ARRAY_SIZE(snd_pcsp_controls); i++) {
- err = snd_ctl_add(card,
- snd_ctl_new1(snd_pcsp_controls + i,
- chip));
+ if (!nopcm) {
+ err = snd_pcsp_ctls_add(chip, snd_pcsp_controls_pcm,
+ ARRAY_SIZE(snd_pcsp_controls_pcm));
if (err < 0)
return err;
}
+ err = snd_pcsp_ctls_add(chip, snd_pcsp_controls_spkr,
+ ARRAY_SIZE(snd_pcsp_controls_spkr));
+ if (err < 0)
+ return err;
strcpy(card->mixername, "PC-Speaker");
diff --git a/sound/isa/Kconfig b/sound/isa/Kconfig
index 51a7e3777e1..02fe81ca88f 100644
--- a/sound/isa/Kconfig
+++ b/sound/isa/Kconfig
@@ -372,15 +372,21 @@ config SND_SGALAXY
config SND_SSCAPE
tristate "Ensoniq SoundScape driver"
- select SND_HWDEP
select SND_MPU401_UART
select SND_WSS_LIB
+ select FW_LOADER
help
Say Y here to include support for Ensoniq SoundScape
- soundcards.
+ and Ensoniq OEM soundcards.
The PCM audio is supported on SoundScape Classic, Elite, PnP
- and VIVO cards. The MIDI support is very experimental.
+ and VIVO cards. The supported OEM cards are SPEA Media FX and
+ Reveal SC-600.
+ The MIDI support is very experimental and requires binary
+ firmware files called "scope.cod" and "sndscape.co?" where the
+ ? is digit 0, 1, 2, 3 or 4. The firmware files can be found
+ in DOS or Windows driver packages. One has to put the firmware
+ files into the /lib/firmware directory.
To compile this driver as a module, choose M here: the module
will be called snd-sscape.
diff --git a/sound/isa/cmi8330.c b/sound/isa/cmi8330.c
index 02f79d25271..8246aae32ab 100644
--- a/sound/isa/cmi8330.c
+++ b/sound/isa/cmi8330.c
@@ -237,7 +237,7 @@ WSS_DOUBLE("Wavetable Capture Volume", 0,
CMI8330_WAVGAIN, CMI8330_WAVGAIN, 4, 0, 15, 0),
WSS_SINGLE("3D Control - Switch", 0,
CMI8330_RMUX3D, 5, 1, 1),
-WSS_SINGLE("PC Speaker Playback Volume", 0,
+WSS_SINGLE("Beep Playback Volume", 0,
CMI8330_OUTPUTVOL, 3, 3, 0),
WSS_DOUBLE("FM Playback Switch", 0,
CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 7, 7, 1, 1),
@@ -262,7 +262,7 @@ SB_DOUBLE("SB Line Playback Switch", SB_DSP4_OUTPUT_SW, SB_DSP4_OUTPUT_SW, 4, 3,
SB_DOUBLE("SB Line Playback Volume", SB_DSP4_LINE_DEV, (SB_DSP4_LINE_DEV + 1), 3, 3, 31),
SB_SINGLE("SB Mic Playback Switch", SB_DSP4_OUTPUT_SW, 0, 1),
SB_SINGLE("SB Mic Playback Volume", SB_DSP4_MIC_DEV, 3, 31),
-SB_SINGLE("SB PC Speaker Volume", SB_DSP4_SPEAKER_DEV, 6, 3),
+SB_SINGLE("SB Beep Volume", SB_DSP4_SPEAKER_DEV, 6, 3),
SB_DOUBLE("SB Capture Volume", SB_DSP4_IGAIN_DEV, (SB_DSP4_IGAIN_DEV + 1), 6, 6, 3),
SB_DOUBLE("SB Playback Volume", SB_DSP4_OGAIN_DEV, (SB_DSP4_OGAIN_DEV + 1), 6, 6, 3),
SB_SINGLE("SB Mic Auto Gain", SB_DSP4_MIC_AGC, 0, 1),
diff --git a/sound/isa/es1688/es1688_lib.c b/sound/isa/es1688/es1688_lib.c
index 4c6e14f87f2..c76bb00c9d1 100644
--- a/sound/isa/es1688/es1688_lib.c
+++ b/sound/isa/es1688/es1688_lib.c
@@ -982,7 +982,7 @@ ES1688_DOUBLE("CD Playback Volume", 0, ES1688_CD_DEV, ES1688_CD_DEV, 4, 0, 15, 0
ES1688_DOUBLE("FM Playback Volume", 0, ES1688_FM_DEV, ES1688_FM_DEV, 4, 0, 15, 0),
ES1688_DOUBLE("Mic Playback Volume", 0, ES1688_MIC_DEV, ES1688_MIC_DEV, 4, 0, 15, 0),
ES1688_DOUBLE("Aux Playback Volume", 0, ES1688_AUX_DEV, ES1688_AUX_DEV, 4, 0, 15, 0),
-ES1688_SINGLE("PC Speaker Playback Volume", 0, ES1688_SPEAKER_DEV, 0, 7, 0),
+ES1688_SINGLE("Beep Playback Volume", 0, ES1688_SPEAKER_DEV, 0, 7, 0),
ES1688_DOUBLE("Capture Volume", 0, ES1688_RECLEV_DEV, ES1688_RECLEV_DEV, 4, 0, 15, 0),
ES1688_SINGLE("Capture Switch", 0, ES1688_REC_DEV, 4, 1, 1),
{
diff --git a/sound/isa/es18xx.c b/sound/isa/es18xx.c
index 8cfbff73a83..e5bf3355d2c 100644
--- a/sound/isa/es18xx.c
+++ b/sound/isa/es18xx.c
@@ -121,7 +121,6 @@ struct snd_es18xx {
unsigned int dma1_shift;
unsigned int dma2_shift;
- struct snd_card *card;
struct snd_pcm *pcm;
struct snd_pcm_substream *playback_a_substream;
struct snd_pcm_substream *capture_a_substream;
@@ -140,10 +139,6 @@ struct snd_es18xx {
#ifdef CONFIG_PM
unsigned char pm_reg;
#endif
-};
-
-struct snd_audiodrive {
- struct snd_es18xx *chip;
#ifdef CONFIG_PNP
struct pnp_dev *dev;
struct pnp_dev *devc;
@@ -755,7 +750,8 @@ static int snd_es18xx_playback_trigger(struct snd_pcm_substream *substream,
static irqreturn_t snd_es18xx_interrupt(int irq, void *dev_id)
{
- struct snd_es18xx *chip = dev_id;
+ struct snd_card *card = dev_id;
+ struct snd_es18xx *chip = card->private_data;
unsigned char status;
if (chip->caps & ES18XX_CONTROL) {
@@ -805,12 +801,16 @@ static irqreturn_t snd_es18xx_interrupt(int irq, void *dev_id)
int split = 0;
if (chip->caps & ES18XX_HWV) {
split = snd_es18xx_mixer_read(chip, 0x64) & 0x80;
- snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_VALUE, &chip->hw_switch->id);
- snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_VALUE, &chip->hw_volume->id);
+ snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE,
+ &chip->hw_switch->id);
+ snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE,
+ &chip->hw_volume->id);
}
if (!split) {
- snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_VALUE, &chip->master_switch->id);
- snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_VALUE, &chip->master_volume->id);
+ snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE,
+ &chip->master_switch->id);
+ snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE,
+ &chip->master_volume->id);
}
/* ack interrupt */
snd_es18xx_mixer_write(chip, 0x66, 0x00);
@@ -1313,7 +1313,7 @@ ES18XX_DOUBLE("Aux Capture Volume", 0, 0x6c, 0x6c, 4, 0, 15, 0)
* The chipset specific mixer controls
*/
static struct snd_kcontrol_new snd_es18xx_opt_speaker =
- ES18XX_SINGLE("PC Speaker Playback Volume", 0, 0x3c, 0, 7, 0);
+ ES18XX_SINGLE("Beep Playback Volume", 0, 0x3c, 0, 7, 0);
static struct snd_kcontrol_new snd_es18xx_opt_1869[] = {
ES18XX_SINGLE("Capture Switch", 0, 0x1c, 4, 1, 1),
@@ -1691,8 +1691,10 @@ static struct snd_pcm_ops snd_es18xx_capture_ops = {
.pointer = snd_es18xx_capture_pointer,
};
-static int __devinit snd_es18xx_pcm(struct snd_es18xx *chip, int device, struct snd_pcm ** rpcm)
+static int __devinit snd_es18xx_pcm(struct snd_card *card, int device,
+ struct snd_pcm **rpcm)
{
+ struct snd_es18xx *chip = card->private_data;
struct snd_pcm *pcm;
char str[16];
int err;
@@ -1701,9 +1703,9 @@ static int __devinit snd_es18xx_pcm(struct snd_es18xx *chip, int device, struct
*rpcm = NULL;
sprintf(str, "ES%x", chip->version);
if (chip->caps & ES18XX_PCM2)
- err = snd_pcm_new(chip->card, str, device, 2, 1, &pcm);
+ err = snd_pcm_new(card, str, device, 2, 1, &pcm);
else
- err = snd_pcm_new(chip->card, str, device, 1, 1, &pcm);
+ err = snd_pcm_new(card, str, device, 1, 1, &pcm);
if (err < 0)
return err;
@@ -1734,10 +1736,9 @@ static int __devinit snd_es18xx_pcm(struct snd_es18xx *chip, int device, struct
#ifdef CONFIG_PM
static int snd_es18xx_suspend(struct snd_card *card, pm_message_t state)
{
- struct snd_audiodrive *acard = card->private_data;
- struct snd_es18xx *chip = acard->chip;
+ struct snd_es18xx *chip = card->private_data;
- snd_power_change_state(chip->card, SNDRV_CTL_POWER_D3hot);
+ snd_power_change_state(card, SNDRV_CTL_POWER_D3hot);
snd_pcm_suspend_all(chip->pcm);
@@ -1752,24 +1753,25 @@ static int snd_es18xx_suspend(struct snd_card *card, pm_message_t state)
static int snd_es18xx_resume(struct snd_card *card)
{
- struct snd_audiodrive *acard = card->private_data;
- struct snd_es18xx *chip = acard->chip;
+ struct snd_es18xx *chip = card->private_data;
/* restore PM register, we won't wake till (not 0x07) i/o activity though */
snd_es18xx_write(chip, ES18XX_PM, chip->pm_reg ^= ES18XX_PM_FM);
- snd_power_change_state(chip->card, SNDRV_CTL_POWER_D0);
+ snd_power_change_state(card, SNDRV_CTL_POWER_D0);
return 0;
}
#endif /* CONFIG_PM */
-static int snd_es18xx_free(struct snd_es18xx *chip)
+static int snd_es18xx_free(struct snd_card *card)
{
+ struct snd_es18xx *chip = card->private_data;
+
release_and_free_resource(chip->res_port);
release_and_free_resource(chip->res_ctrl_port);
release_and_free_resource(chip->res_mpu_port);
if (chip->irq >= 0)
- free_irq(chip->irq, (void *) chip);
+ free_irq(chip->irq, (void *) card);
if (chip->dma1 >= 0) {
disable_dma(chip->dma1);
free_dma(chip->dma1);
@@ -1778,37 +1780,29 @@ static int snd_es18xx_free(struct snd_es18xx *chip)
disable_dma(chip->dma2);
free_dma(chip->dma2);
}
- kfree(chip);
return 0;
}
static int snd_es18xx_dev_free(struct snd_device *device)
{
- struct snd_es18xx *chip = device->device_data;
- return snd_es18xx_free(chip);
+ return snd_es18xx_free(device->card);
}
static int __devinit snd_es18xx_new_device(struct snd_card *card,
unsigned long port,
unsigned long mpu_port,
unsigned long fm_port,
- int irq, int dma1, int dma2,
- struct snd_es18xx ** rchip)
+ int irq, int dma1, int dma2)
{
- struct snd_es18xx *chip;
+ struct snd_es18xx *chip = card->private_data;
static struct snd_device_ops ops = {
.dev_free = snd_es18xx_dev_free,
};
int err;
- *rchip = NULL;
- chip = kzalloc(sizeof(*chip), GFP_KERNEL);
- if (chip == NULL)
- return -ENOMEM;
spin_lock_init(&chip->reg_lock);
spin_lock_init(&chip->mixer_lock);
spin_lock_init(&chip->ctrl_lock);
- chip->card = card;
chip->port = port;
chip->mpu_port = mpu_port;
chip->fm_port = fm_port;
@@ -1818,53 +1812,53 @@ static int __devinit snd_es18xx_new_device(struct snd_card *card,
chip->audio2_vol = 0x00;
chip->active = 0;
- if ((chip->res_port = request_region(port, 16, "ES18xx")) == NULL) {
- snd_es18xx_free(chip);
+ chip->res_port = request_region(port, 16, "ES18xx");
+ if (chip->res_port == NULL) {
+ snd_es18xx_free(card);
snd_printk(KERN_ERR PFX "unable to grap ports 0x%lx-0x%lx\n", port, port + 16 - 1);
return -EBUSY;
}
- if (request_irq(irq, snd_es18xx_interrupt, IRQF_DISABLED, "ES18xx", (void *) chip)) {
- snd_es18xx_free(chip);
+ if (request_irq(irq, snd_es18xx_interrupt, IRQF_DISABLED, "ES18xx",
+ (void *) card)) {
+ snd_es18xx_free(card);
snd_printk(KERN_ERR PFX "unable to grap IRQ %d\n", irq);
return -EBUSY;
}
chip->irq = irq;
if (request_dma(dma1, "ES18xx DMA 1")) {
- snd_es18xx_free(chip);
+ snd_es18xx_free(card);
snd_printk(KERN_ERR PFX "unable to grap DMA1 %d\n", dma1);
return -EBUSY;
}
chip->dma1 = dma1;
if (dma2 != dma1 && request_dma(dma2, "ES18xx DMA 2")) {
- snd_es18xx_free(chip);
+ snd_es18xx_free(card);
snd_printk(KERN_ERR PFX "unable to grap DMA2 %d\n", dma2);
return -EBUSY;
}
chip->dma2 = dma2;
if (snd_es18xx_probe(chip) < 0) {
- snd_es18xx_free(chip);
+ snd_es18xx_free(card);
return -ENODEV;
}
- if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops)) < 0) {
- snd_es18xx_free(chip);
+ err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, NULL, &ops);
+ if (err < 0) {
+ snd_es18xx_free(card);
return err;
}
- *rchip = chip;
return 0;
}
-static int __devinit snd_es18xx_mixer(struct snd_es18xx *chip)
+static int __devinit snd_es18xx_mixer(struct snd_card *card)
{
- struct snd_card *card;
+ struct snd_es18xx *chip = card->private_data;
int err;
unsigned int idx;
- card = chip->card;
-
strcpy(card->mixername, chip->pcm->name);
for (idx = 0; idx < ARRAY_SIZE(snd_es18xx_base_controls); idx++) {
@@ -2063,11 +2057,11 @@ static int __devinit snd_audiodrive_pnp_init_main(int dev, struct pnp_dev *pdev)
return 0;
}
-static int __devinit snd_audiodrive_pnp(int dev, struct snd_audiodrive *acard,
+static int __devinit snd_audiodrive_pnp(int dev, struct snd_es18xx *chip,
struct pnp_dev *pdev)
{
- acard->dev = pdev;
- if (snd_audiodrive_pnp_init_main(dev, acard->dev) < 0)
+ chip->dev = pdev;
+ if (snd_audiodrive_pnp_init_main(dev, chip->dev) < 0)
return -EBUSY;
return 0;
}
@@ -2093,26 +2087,26 @@ static struct pnp_card_device_id snd_audiodrive_pnpids[] = {
MODULE_DEVICE_TABLE(pnp_card, snd_audiodrive_pnpids);
-static int __devinit snd_audiodrive_pnpc(int dev, struct snd_audiodrive *acard,
+static int __devinit snd_audiodrive_pnpc(int dev, struct snd_es18xx *chip,
struct pnp_card_link *card,
const struct pnp_card_device_id *id)
{
- acard->dev = pnp_request_card_device(card, id->devs[0].id, NULL);
- if (acard->dev == NULL)
+ chip->dev = pnp_request_card_device(card, id->devs[0].id, NULL);
+ if (chip->dev == NULL)
return -EBUSY;
- acard->devc = pnp_request_card_device(card, id->devs[1].id, NULL);
- if (acard->devc == NULL)
+ chip->devc = pnp_request_card_device(card, id->devs[1].id, NULL);
+ if (chip->devc == NULL)
return -EBUSY;
/* Control port initialization */
- if (pnp_activate_dev(acard->devc) < 0) {
+ if (pnp_activate_dev(chip->devc) < 0) {
snd_printk(KERN_ERR PFX "PnP control configure failure (out of resources?)\n");
return -EAGAIN;
}
snd_printdd("pnp: port=0x%llx\n",
- (unsigned long long)pnp_port_start(acard->devc, 0));
- if (snd_audiodrive_pnp_init_main(dev, acard->dev) < 0)
+ (unsigned long long)pnp_port_start(chip->devc, 0));
+ if (snd_audiodrive_pnp_init_main(dev, chip->dev) < 0)
return -EBUSY;
return 0;
@@ -2128,24 +2122,20 @@ static int __devinit snd_audiodrive_pnpc(int dev, struct snd_audiodrive *acard,
static int snd_es18xx_card_new(int dev, struct snd_card **cardp)
{
return snd_card_create(index[dev], id[dev], THIS_MODULE,
- sizeof(struct snd_audiodrive), cardp);
+ sizeof(struct snd_es18xx), cardp);
}
static int __devinit snd_audiodrive_probe(struct snd_card *card, int dev)
{
- struct snd_audiodrive *acard = card->private_data;
- struct snd_es18xx *chip;
+ struct snd_es18xx *chip = card->private_data;
struct snd_opl3 *opl3;
int err;
- if ((err = snd_es18xx_new_device(card,
- port[dev],
- mpu_port[dev],
- fm_port[dev],
- irq[dev], dma1[dev], dma2[dev],
- &chip)) < 0)
+ err = snd_es18xx_new_device(card,
+ port[dev], mpu_port[dev], fm_port[dev],
+ irq[dev], dma1[dev], dma2[dev]);
+ if (err < 0)
return err;
- acard->chip = chip;
sprintf(card->driver, "ES%x", chip->version);
@@ -2161,10 +2151,12 @@ static int __devinit snd_audiodrive_probe(struct snd_card *card, int dev)
chip->port,
irq[dev], dma1[dev]);
- if ((err = snd_es18xx_pcm(chip, 0, NULL)) < 0)
+ err = snd_es18xx_pcm(card, 0, NULL);
+ if (err < 0)
return err;
- if ((err = snd_es18xx_mixer(chip)) < 0)
+ err = snd_es18xx_mixer(card);
+ if (err < 0)
return err;
if (fm_port[dev] > 0 && fm_port[dev] != SNDRV_AUTO_PORT) {
diff --git a/sound/isa/sb/sb_mixer.c b/sound/isa/sb/sb_mixer.c
index 475220bbcc9..318ff0c823e 100644
--- a/sound/isa/sb/sb_mixer.c
+++ b/sound/isa/sb/sb_mixer.c
@@ -631,7 +631,7 @@ static struct sbmix_elem snd_sb16_ctl_mic_play_switch =
static struct sbmix_elem snd_sb16_ctl_mic_play_vol =
SB_SINGLE("Mic Playback Volume", SB_DSP4_MIC_DEV, 3, 31);
static struct sbmix_elem snd_sb16_ctl_pc_speaker_vol =
- SB_SINGLE("PC Speaker Volume", SB_DSP4_SPEAKER_DEV, 6, 3);
+ SB_SINGLE("Beep Volume", SB_DSP4_SPEAKER_DEV, 6, 3);
static struct sbmix_elem snd_sb16_ctl_capture_vol =
SB_DOUBLE("Capture Volume", SB_DSP4_IGAIN_DEV, (SB_DSP4_IGAIN_DEV + 1), 6, 6, 3);
static struct sbmix_elem snd_sb16_ctl_play_vol =
@@ -689,7 +689,7 @@ static struct sbmix_elem snd_dt019x_ctl_cd_play_vol =
static struct sbmix_elem snd_dt019x_ctl_mic_play_vol =
SB_SINGLE("Mic Playback Volume", SB_DT019X_MIC_DEV, 4, 7);
static struct sbmix_elem snd_dt019x_ctl_pc_speaker_vol =
- SB_SINGLE("PC Speaker Volume", SB_DT019X_SPKR_DEV, 0, 7);
+ SB_SINGLE("Beep Volume", SB_DT019X_SPKR_DEV, 0, 7);
static struct sbmix_elem snd_dt019x_ctl_line_play_vol =
SB_DOUBLE("Line Playback Volume", SB_DT019X_LINE_DEV, SB_DT019X_LINE_DEV, 4,0, 15);
static struct sbmix_elem snd_dt019x_ctl_pcm_play_switch =
diff --git a/sound/isa/sscape.c b/sound/isa/sscape.c
index 66187122377..e2d5d2d3ed9 100644
--- a/sound/isa/sscape.c
+++ b/sound/isa/sscape.c
@@ -1,5 +1,5 @@
/*
- * Low-level ALSA driver for the ENSONIQ SoundScape PnP
+ * Low-level ALSA driver for the ENSONIQ SoundScape
* Copyright (c) by Chris Rankin
*
* This driver was written in part using information obtained from
@@ -25,31 +25,36 @@
#include <linux/err.h>
#include <linux/isa.h>
#include <linux/delay.h>
+#include <linux/firmware.h>
#include <linux/pnp.h>
#include <linux/spinlock.h>
#include <linux/moduleparam.h>
#include <asm/dma.h>
#include <sound/core.h>
-#include <sound/hwdep.h>
#include <sound/wss.h>
#include <sound/mpu401.h>
#include <sound/initval.h>
-#include <sound/sscape_ioctl.h>
-
MODULE_AUTHOR("Chris Rankin");
-MODULE_DESCRIPTION("ENSONIQ SoundScape PnP driver");
+MODULE_DESCRIPTION("ENSONIQ SoundScape driver");
MODULE_LICENSE("GPL");
-
-static int index[SNDRV_CARDS] __devinitdata = SNDRV_DEFAULT_IDX;
-static char* id[SNDRV_CARDS] __devinitdata = SNDRV_DEFAULT_STR;
-static long port[SNDRV_CARDS] __devinitdata = SNDRV_DEFAULT_PORT;
-static long wss_port[SNDRV_CARDS] __devinitdata = SNDRV_DEFAULT_PORT;
-static int irq[SNDRV_CARDS] __devinitdata = SNDRV_DEFAULT_IRQ;
-static int mpu_irq[SNDRV_CARDS] __devinitdata = SNDRV_DEFAULT_IRQ;
-static int dma[SNDRV_CARDS] __devinitdata = SNDRV_DEFAULT_DMA;
-static int dma2[SNDRV_CARDS] __devinitdata = SNDRV_DEFAULT_DMA;
+MODULE_FIRMWARE("sndscape.co0");
+MODULE_FIRMWARE("sndscape.co1");
+MODULE_FIRMWARE("sndscape.co2");
+MODULE_FIRMWARE("sndscape.co3");
+MODULE_FIRMWARE("sndscape.co4");
+MODULE_FIRMWARE("scope.cod");
+
+static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX;
+static char* id[SNDRV_CARDS] = SNDRV_DEFAULT_STR;
+static long port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT;
+static long wss_port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT;
+static int irq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ;
+static int mpu_irq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ;
+static int dma[SNDRV_CARDS] = SNDRV_DEFAULT_DMA;
+static int dma2[SNDRV_CARDS] = SNDRV_DEFAULT_DMA;
+static bool joystick[SNDRV_CARDS];
module_param_array(index, int, NULL, 0444);
MODULE_PARM_DESC(index, "Index number for SoundScape soundcard");
@@ -75,6 +80,9 @@ MODULE_PARM_DESC(dma, "DMA # for SoundScape driver.");
module_param_array(dma2, int, NULL, 0444);
MODULE_PARM_DESC(dma2, "DMA2 # for SoundScape driver.");
+module_param_array(joystick, bool, NULL, 0444);
+MODULE_PARM_DESC(joystick, "Enable gameport.");
+
#ifdef CONFIG_PNP
static int isa_registered;
static int pnp_registered;
@@ -101,14 +109,14 @@ MODULE_DEVICE_TABLE(pnp_card, sscape_pnpids);
#define RX_READY 0x01
#define TX_READY 0x02
-#define CMD_ACK 0x80
-#define CMD_SET_MIDI_VOL 0x84
-#define CMD_GET_MIDI_VOL 0x85
-#define CMD_XXX_MIDI_VOL 0x86
-#define CMD_SET_EXTMIDI 0x8a
-#define CMD_GET_EXTMIDI 0x8b
-#define CMD_SET_MT32 0x8c
-#define CMD_GET_MT32 0x8d
+#define CMD_ACK 0x80
+#define CMD_SET_MIDI_VOL 0x84
+#define CMD_GET_MIDI_VOL 0x85
+#define CMD_XXX_MIDI_VOL 0x86
+#define CMD_SET_EXTMIDI 0x8a
+#define CMD_GET_EXTMIDI 0x8b
+#define CMD_SET_MT32 0x8c
+#define CMD_GET_MT32 0x8d
enum GA_REG {
GA_INTSTAT_REG = 0,
@@ -127,7 +135,8 @@ enum GA_REG {
enum card_type {
- SSCAPE,
+ MEDIA_FX, /* Sequoia S-1000 */
+ SSCAPE, /* Sequoia S-2000 */
SSCAPE_PNP,
SSCAPE_VIVO,
};
@@ -140,16 +149,7 @@ struct soundscape {
struct resource *io_res;
struct resource *wss_res;
struct snd_wss *chip;
- struct snd_mpu401 *mpu;
- struct snd_hwdep *hw;
- /*
- * The MIDI device won't work until we've loaded
- * its firmware via a hardware-dependent device IOCTL
- */
- spinlock_t fwlock;
- int hw_in_use;
- unsigned long midi_usage;
unsigned char midi_vol;
};
@@ -161,28 +161,21 @@ static inline struct soundscape *get_card_soundscape(struct snd_card *c)
return (struct soundscape *) (c->private_data);
}
-static inline struct soundscape *get_mpu401_soundscape(struct snd_mpu401 * mpu)
-{
- return (struct soundscape *) (mpu->private_data);
-}
-
-static inline struct soundscape *get_hwdep_soundscape(struct snd_hwdep * hw)
-{
- return (struct soundscape *) (hw->private_data);
-}
-
-
/*
* Allocates some kernel memory that we can use for DMA.
* I think this means that the memory has to map to
* contiguous pages of physical memory.
*/
-static struct snd_dma_buffer *get_dmabuf(struct snd_dma_buffer *buf, unsigned long size)
+static struct snd_dma_buffer *get_dmabuf(struct snd_dma_buffer *buf,
+ unsigned long size)
{
if (buf) {
- if (snd_dma_alloc_pages_fallback(SNDRV_DMA_TYPE_DEV, snd_dma_isa_data(),
+ if (snd_dma_alloc_pages_fallback(SNDRV_DMA_TYPE_DEV,
+ snd_dma_isa_data(),
size, buf) < 0) {
- snd_printk(KERN_ERR "sscape: Failed to allocate %lu bytes for DMA\n", size);
+ snd_printk(KERN_ERR "sscape: Failed to allocate "
+ "%lu bytes for DMA\n",
+ size);
return NULL;
}
}
@@ -199,13 +192,13 @@ static void free_dmabuf(struct snd_dma_buffer *buf)
snd_dma_free_pages(buf);
}
-
/*
* This function writes to the SoundScape's control registers,
* but doesn't do any locking. It's up to the caller to do that.
* This is why this function is "unsafe" ...
*/
-static inline void sscape_write_unsafe(unsigned io_base, enum GA_REG reg, unsigned char val)
+static inline void sscape_write_unsafe(unsigned io_base, enum GA_REG reg,
+ unsigned char val)
{
outb(reg, ODIE_ADDR_IO(io_base));
outb(val, ODIE_DATA_IO(io_base));
@@ -215,7 +208,8 @@ static inline void sscape_write_unsafe(unsigned io_base, enum GA_REG reg, unsign
* Write to the SoundScape's control registers, and do the
* necessary locking ...
*/
-static void sscape_write(struct soundscape *s, enum GA_REG reg, unsigned char val)
+static void sscape_write(struct soundscape *s, enum GA_REG reg,
+ unsigned char val)
{
unsigned long flags;
@@ -228,7 +222,8 @@ static void sscape_write(struct soundscape *s, enum GA_REG reg, unsigned char va
* Read from the SoundScape's control registers, but leave any
* locking to the caller. This is why the function is "unsafe" ...
*/
-static inline unsigned char sscape_read_unsafe(unsigned io_base, enum GA_REG reg)
+static inline unsigned char sscape_read_unsafe(unsigned io_base,
+ enum GA_REG reg)
{
outb(reg, ODIE_ADDR_IO(io_base));
return inb(ODIE_DATA_IO(io_base));
@@ -257,9 +252,8 @@ static inline void set_midi_mode_unsafe(unsigned io_base)
static inline int host_read_unsafe(unsigned io_base)
{
int data = -1;
- if ((inb(HOST_CTRL_IO(io_base)) & RX_READY) != 0) {
+ if ((inb(HOST_CTRL_IO(io_base)) & RX_READY) != 0)
data = inb(HOST_DATA_IO(io_base));
- }
return data;
}
@@ -301,7 +295,7 @@ static inline int host_write_unsafe(unsigned io_base, unsigned char data)
* Also leaves all locking-issues to the caller ...
*/
static int host_write_ctrl_unsafe(unsigned io_base, unsigned char data,
- unsigned timeout)
+ unsigned timeout)
{
int err;
@@ -320,7 +314,7 @@ static int host_write_ctrl_unsafe(unsigned io_base, unsigned char data,
*
* NOTE: This check is based upon observation, not documentation.
*/
-static inline int verify_mpu401(const struct snd_mpu401 * mpu)
+static inline int verify_mpu401(const struct snd_mpu401 *mpu)
{
return ((inb(MPU401C(mpu)) & 0xc0) == 0x80);
}
@@ -328,7 +322,7 @@ static inline int verify_mpu401(const struct snd_mpu401 * mpu)
/*
* This is apparently the standard way to initailise an MPU-401
*/
-static inline void initialise_mpu401(const struct snd_mpu401 * mpu)
+static inline void initialise_mpu401(const struct snd_mpu401 *mpu)
{
outb(0, MPU401D(mpu));
}
@@ -338,9 +332,10 @@ static inline void initialise_mpu401(const struct snd_mpu401 * mpu)
* The AD1845 detection fails if we *don't* do this, so I
* think that this is a good idea ...
*/
-static inline void activate_ad1845_unsafe(unsigned io_base)
+static void activate_ad1845_unsafe(unsigned io_base)
{
- sscape_write_unsafe(io_base, GA_HMCTL_REG, (sscape_read_unsafe(io_base, GA_HMCTL_REG) & 0xcf) | 0x10);
+ unsigned char val = sscape_read_unsafe(io_base, GA_HMCTL_REG);
+ sscape_write_unsafe(io_base, GA_HMCTL_REG, (val & 0xcf) | 0x10);
sscape_write_unsafe(io_base, GA_CDCFG_REG, 0x80);
}
@@ -359,24 +354,27 @@ static void soundscape_free(struct snd_card *c)
* Tell the SoundScape to begin a DMA tranfer using the given channel.
* All locking issues are left to the caller.
*/
-static inline void sscape_start_dma_unsafe(unsigned io_base, enum GA_REG reg)
+static void sscape_start_dma_unsafe(unsigned io_base, enum GA_REG reg)
{
- sscape_write_unsafe(io_base, reg, sscape_read_unsafe(io_base, reg) | 0x01);
- sscape_write_unsafe(io_base, reg, sscape_read_unsafe(io_base, reg) & 0xfe);
+ sscape_write_unsafe(io_base, reg,
+ sscape_read_unsafe(io_base, reg) | 0x01);
+ sscape_write_unsafe(io_base, reg,
+ sscape_read_unsafe(io_base, reg) & 0xfe);
}
/*
* Wait for a DMA transfer to complete. This is a "limited busy-wait",
* and all locking issues are left to the caller.
*/
-static int sscape_wait_dma_unsafe(unsigned io_base, enum GA_REG reg, unsigned timeout)
+static int sscape_wait_dma_unsafe(unsigned io_base, enum GA_REG reg,
+ unsigned timeout)
{
while (!(sscape_read_unsafe(io_base, reg) & 0x01) && (timeout != 0)) {
udelay(100);
--timeout;
} /* while */
- return (sscape_read_unsafe(io_base, reg) & 0x01);
+ return sscape_read_unsafe(io_base, reg) & 0x01;
}
/*
@@ -392,12 +390,12 @@ static int obp_startup_ack(struct soundscape *s, unsigned timeout)
do {
unsigned long flags;
- unsigned char x;
+ int x;
spin_lock_irqsave(&s->lock, flags);
- x = inb(HOST_DATA_IO(s->io_base));
+ x = host_read_unsafe(s->io_base);
spin_unlock_irqrestore(&s->lock, flags);
- if ((x & 0xfe) == 0xfe)
+ if (x == 0xfe || x == 0xff)
return 1;
msleep(10);
@@ -419,10 +417,10 @@ static int host_startup_ack(struct soundscape *s, unsigned timeout)
do {
unsigned long flags;
- unsigned char x;
+ int x;
spin_lock_irqsave(&s->lock, flags);
- x = inb(HOST_DATA_IO(s->io_base));
+ x = host_read_unsafe(s->io_base);
spin_unlock_irqrestore(&s->lock, flags);
if (x == 0xfe)
return 1;
@@ -436,15 +434,15 @@ static int host_startup_ack(struct soundscape *s, unsigned timeout)
/*
* Upload a byte-stream into the SoundScape using DMA channel A.
*/
-static int upload_dma_data(struct soundscape *s,
- const unsigned char __user *data,
- size_t size)
+static int upload_dma_data(struct soundscape *s, const unsigned char *data,
+ size_t size)
{
unsigned long flags;
struct snd_dma_buffer dma;
int ret;
+ unsigned char val;
- if (!get_dmabuf(&dma, PAGE_ALIGN(size)))
+ if (!get_dmabuf(&dma, PAGE_ALIGN(32 * 1024)))
return -ENOMEM;
spin_lock_irqsave(&s->lock, flags);
@@ -452,70 +450,57 @@ static int upload_dma_data(struct soundscape *s,
/*
* Reset the board ...
*/
- sscape_write_unsafe(s->io_base, GA_HMCTL_REG, sscape_read_unsafe(s->io_base, GA_HMCTL_REG) & 0x3f);
+ val = sscape_read_unsafe(s->io_base, GA_HMCTL_REG);
+ sscape_write_unsafe(s->io_base, GA_HMCTL_REG, val & 0x3f);
/*
* Enable the DMA channels and configure them ...
*/
- sscape_write_unsafe(s->io_base, GA_DMACFG_REG, 0x50);
- sscape_write_unsafe(s->io_base, GA_DMAA_REG, (s->chip->dma1 << 4) | DMA_8BIT);
+ val = (s->chip->dma1 << 4) | DMA_8BIT;
+ sscape_write_unsafe(s->io_base, GA_DMAA_REG, val);
sscape_write_unsafe(s->io_base, GA_DMAB_REG, 0x20);
/*
* Take the board out of reset ...
*/
- sscape_write_unsafe(s->io_base, GA_HMCTL_REG, sscape_read_unsafe(s->io_base, GA_HMCTL_REG) | 0x80);
+ val = sscape_read_unsafe(s->io_base, GA_HMCTL_REG);
+ sscape_write_unsafe(s->io_base, GA_HMCTL_REG, val | 0x80);
/*
- * Upload the user's data (firmware?) to the SoundScape
+ * Upload the firmware to the SoundScape
* board through the DMA channel ...
*/
while (size != 0) {
unsigned long len;
- /*
- * Apparently, copying to/from userspace can sleep.
- * We are therefore forbidden from holding any
- * spinlocks while we copy ...
- */
- spin_unlock_irqrestore(&s->lock, flags);
-
- /*
- * Remember that the data that we want to DMA
- * comes from USERSPACE. We have already verified
- * the userspace pointer ...
- */
len = min(size, dma.bytes);
- len -= __copy_from_user(dma.area, data, len);
+ memcpy(dma.area, data, len);
data += len;
size -= len;
- /*
- * Grab that spinlock again, now that we've
- * finished copying!
- */
- spin_lock_irqsave(&s->lock, flags);
-
snd_dma_program(s->chip->dma1, dma.addr, len, DMA_MODE_WRITE);
sscape_start_dma_unsafe(s->io_base, GA_DMAA_REG);
if (!sscape_wait_dma_unsafe(s->io_base, GA_DMAA_REG, 5000)) {
/*
- * Don't forget to release this spinlock we're holding ...
+ * Don't forget to release this spinlock we're holding
*/
spin_unlock_irqrestore(&s->lock, flags);
- snd_printk(KERN_ERR "sscape: DMA upload has timed out\n");
+ snd_printk(KERN_ERR
+ "sscape: DMA upload has timed out\n");
ret = -EAGAIN;
goto _release_dma;
}
} /* while */
set_host_mode_unsafe(s->io_base);
+ outb(0x0, s->io_base);
/*
* Boot the board ... (I think)
*/
- sscape_write_unsafe(s->io_base, GA_HMCTL_REG, sscape_read_unsafe(s->io_base, GA_HMCTL_REG) | 0x40);
+ val = sscape_read_unsafe(s->io_base, GA_HMCTL_REG);
+ sscape_write_unsafe(s->io_base, GA_HMCTL_REG, val | 0x40);
spin_unlock_irqrestore(&s->lock, flags);
/*
@@ -525,10 +510,12 @@ static int upload_dma_data(struct soundscape *s,
*/
ret = 0;
if (!obp_startup_ack(s, 5000)) {
- snd_printk(KERN_ERR "sscape: No response from on-board processor after upload\n");
+ snd_printk(KERN_ERR "sscape: No response "
+ "from on-board processor after upload\n");
ret = -EAGAIN;
} else if (!host_startup_ack(s, 5000)) {
- snd_printk(KERN_ERR "sscape: SoundScape failed to initialise\n");
+ snd_printk(KERN_ERR
+ "sscape: SoundScape failed to initialise\n");
ret = -EAGAIN;
}
@@ -536,7 +523,7 @@ _release_dma:
/*
* NOTE!!! We are NOT holding any spinlocks at this point !!!
*/
- sscape_write(s, GA_DMAA_REG, (s->ic_type == IC_ODIE ? 0x70 : 0x40));
+ sscape_write(s, GA_DMAA_REG, (s->ic_type == IC_OPUS ? 0x40 : 0x70));
free_dmabuf(&dma);
return ret;
@@ -546,167 +533,76 @@ _release_dma:
* Upload the bootblock(?) into the SoundScape. The only
* purpose of this block of code seems to be to tell
* us which version of the microcode we should be using.
- *
- * NOTE: The boot-block data resides in USER-SPACE!!!
- * However, we have already verified its memory
- * addresses by the time we get here.
*/
-static int sscape_upload_bootblock(struct soundscape *sscape, struct sscape_bootblock __user *bb)
+static int sscape_upload_bootblock(struct snd_card *card)
{
+ struct soundscape *sscape = get_card_soundscape(card);
unsigned long flags;
+ const struct firmware *init_fw = NULL;
int data = 0;
int ret;
- ret = upload_dma_data(sscape, bb->code, sizeof(bb->code));
-
- spin_lock_irqsave(&sscape->lock, flags);
- if (ret == 0) {
- data = host_read_ctrl_unsafe(sscape->io_base, 100);
- }
- set_midi_mode_unsafe(sscape->io_base);
- spin_unlock_irqrestore(&sscape->lock, flags);
-
- if (ret == 0) {
- if (data < 0) {
- snd_printk(KERN_ERR "sscape: timeout reading firmware version\n");
- ret = -EAGAIN;
- }
- else if (__copy_to_user(&bb->version, &data, sizeof(bb->version))) {
- ret = -EFAULT;
- }
+ ret = request_firmware(&init_fw, "scope.cod", card->dev);
+ if (ret < 0) {
+ snd_printk(KERN_ERR "sscape: Error loading scope.cod");
+ return ret;
}
+ ret = upload_dma_data(sscape, init_fw->data, init_fw->size);
- return ret;
-}
-
-/*
- * Upload the microcode into the SoundScape. The
- * microcode is 64K of data, and if we try to copy
- * it into a local variable then we will SMASH THE
- * KERNEL'S STACK! We therefore leave it in USER
- * SPACE, and save ourselves from copying it at all.
- */
-static int sscape_upload_microcode(struct soundscape *sscape,
- const struct sscape_microcode __user *mc)
-{
- unsigned long flags;
- char __user *code;
- int err;
+ release_firmware(init_fw);
- /*
- * We are going to have to copy this data into a special
- * DMA-able buffer before we can upload it. We shall therefore
- * just check that the data pointer is valid for now.
- *
- * NOTE: This buffer is 64K long! That's WAY too big to
- * copy into a stack-temporary anyway.
- */
- if ( get_user(code, &mc->code) ||
- !access_ok(VERIFY_READ, code, SSCAPE_MICROCODE_SIZE) )
- return -EFAULT;
+ spin_lock_irqsave(&sscape->lock, flags);
+ if (ret == 0)
+ data = host_read_ctrl_unsafe(sscape->io_base, 100);
- if ((err = upload_dma_data(sscape, code, SSCAPE_MICROCODE_SIZE)) == 0) {
- snd_printk(KERN_INFO "sscape: MIDI firmware loaded\n");
- }
+ if (data & 0x10)
+ sscape_write_unsafe(sscape->io_base, GA_SMCFGA_REG, 0x2f);
- spin_lock_irqsave(&sscape->lock, flags);
- set_midi_mode_unsafe(sscape->io_base);
spin_unlock_irqrestore(&sscape->lock, flags);
- initialise_mpu401(sscape->mpu);
+ data &= 0xf;
+ if (ret == 0 && data > 7) {
+ snd_printk(KERN_ERR
+ "sscape: timeout reading firmware version\n");
+ ret = -EAGAIN;
+ }
- return err;
+ return (ret == 0) ? data : ret;
}
/*
- * Hardware-specific device functions, to implement special
- * IOCTLs for the SoundScape card. This is how we upload
- * the microcode into the card, for example, and so we
- * must ensure that no two processes can open this device
- * simultaneously, and that we can't open it at all if
- * someone is using the MIDI device.
+ * Upload the microcode into the SoundScape.
*/
-static int sscape_hw_open(struct snd_hwdep * hw, struct file *file)
+static int sscape_upload_microcode(struct snd_card *card, int version)
{
- register struct soundscape *sscape = get_hwdep_soundscape(hw);
- unsigned long flags;
+ struct soundscape *sscape = get_card_soundscape(card);
+ const struct firmware *init_fw = NULL;
+ char name[14];
int err;
- spin_lock_irqsave(&sscape->fwlock, flags);
+ snprintf(name, sizeof(name), "sndscape.co%d", version);
- if ((sscape->midi_usage != 0) || sscape->hw_in_use) {
- err = -EBUSY;
- } else {
- sscape->hw_in_use = 1;
- err = 0;
+ err = request_firmware(&init_fw, name, card->dev);
+ if (err < 0) {
+ snd_printk(KERN_ERR "sscape: Error loading sndscape.co%d",
+ version);
+ return err;
}
+ err = upload_dma_data(sscape, init_fw->data, init_fw->size);
+ if (err == 0)
+ snd_printk(KERN_INFO "sscape: MIDI firmware loaded %d KBs\n",
+ init_fw->size >> 10);
- spin_unlock_irqrestore(&sscape->fwlock, flags);
- return err;
-}
-
-static int sscape_hw_release(struct snd_hwdep * hw, struct file *file)
-{
- register struct soundscape *sscape = get_hwdep_soundscape(hw);
- unsigned long flags;
-
- spin_lock_irqsave(&sscape->fwlock, flags);
- sscape->hw_in_use = 0;
- spin_unlock_irqrestore(&sscape->fwlock, flags);
- return 0;
-}
-
-static int sscape_hw_ioctl(struct snd_hwdep * hw, struct file *file,
- unsigned int cmd, unsigned long arg)
-{
- struct soundscape *sscape = get_hwdep_soundscape(hw);
- int err = -EBUSY;
-
- switch (cmd) {
- case SND_SSCAPE_LOAD_BOOTB:
- {
- register struct sscape_bootblock __user *bb = (struct sscape_bootblock __user *) arg;
-
- /*
- * We are going to have to copy this data into a special
- * DMA-able buffer before we can upload it. We shall therefore
- * just check that the data pointer is valid for now ...
- */
- if ( !access_ok(VERIFY_READ, bb->code, sizeof(bb->code)) )
- return -EFAULT;
-
- /*
- * Now check that we can write the firmware version number too...
- */
- if ( !access_ok(VERIFY_WRITE, &bb->version, sizeof(bb->version)) )
- return -EFAULT;
-
- err = sscape_upload_bootblock(sscape, bb);
- }
- break;
-
- case SND_SSCAPE_LOAD_MCODE:
- {
- register const struct sscape_microcode __user *mc = (const struct sscape_microcode __user *) arg;
-
- err = sscape_upload_microcode(sscape, mc);
- }
- break;
-
- default:
- err = -EINVAL;
- break;
- } /* switch */
+ release_firmware(init_fw);
return err;
}
-
/*
* Mixer control for the SoundScape's MIDI device.
*/
static int sscape_midi_info(struct snd_kcontrol *ctl,
- struct snd_ctl_elem_info *uinfo)
+ struct snd_ctl_elem_info *uinfo)
{
uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
uinfo->count = 1;
@@ -716,7 +612,7 @@ static int sscape_midi_info(struct snd_kcontrol *ctl,
}
static int sscape_midi_get(struct snd_kcontrol *kctl,
- struct snd_ctl_elem_value *uctl)
+ struct snd_ctl_elem_value *uctl)
{
struct snd_wss *chip = snd_kcontrol_chip(kctl);
struct snd_card *card = chip->card;
@@ -730,16 +626,18 @@ static int sscape_midi_get(struct snd_kcontrol *kctl,
}
static int sscape_midi_put(struct snd_kcontrol *kctl,
- struct snd_ctl_elem_value *uctl)
+ struct snd_ctl_elem_value *uctl)
{
struct snd_wss *chip = snd_kcontrol_chip(kctl);
struct snd_card *card = chip->card;
- register struct soundscape *s = get_card_soundscape(card);
+ struct soundscape *s = get_card_soundscape(card);
unsigned long flags;
int change;
+ unsigned char new_val;
spin_lock_irqsave(&s->lock, flags);
+ new_val = uctl->value.integer.value[0] & 127;
/*
* We need to put the board into HOST mode before we
* can send any volume-changing HOST commands ...
@@ -752,15 +650,16 @@ static int sscape_midi_put(struct snd_kcontrol *kctl,
* and then perform another volume-related command. Perhaps the
* first command is an "open" and the second command is a "close"?
*/
- if (s->midi_vol == ((unsigned char) uctl->value.integer. value[0] & 127)) {
+ if (s->midi_vol == new_val) {
change = 0;
goto __skip_change;
}
- change = (host_write_ctrl_unsafe(s->io_base, CMD_SET_MIDI_VOL, 100)
- && host_write_ctrl_unsafe(s->io_base, ((unsigned char) uctl->value.integer. value[0]) & 127, 100)
- && host_write_ctrl_unsafe(s->io_base, CMD_XXX_MIDI_VOL, 100));
- s->midi_vol = (unsigned char) uctl->value.integer.value[0] & 127;
- __skip_change:
+ change = host_write_ctrl_unsafe(s->io_base, CMD_SET_MIDI_VOL, 100)
+ && host_write_ctrl_unsafe(s->io_base, new_val, 100)
+ && host_write_ctrl_unsafe(s->io_base, CMD_XXX_MIDI_VOL, 100)
+ && host_write_ctrl_unsafe(s->io_base, new_val, 100);
+ s->midi_vol = new_val;
+__skip_change:
/*
* Take the board out of HOST mode and back into MIDI mode ...
@@ -784,20 +683,25 @@ static struct snd_kcontrol_new midi_mixer_ctl = {
* These IRQs are encoded as bit patterns so that they can be
* written to the control registers.
*/
-static unsigned __devinit get_irq_config(int irq)
+static unsigned __devinit get_irq_config(int sscape_type, int irq)
{
static const int valid_irq[] = { 9, 5, 7, 10 };
+ static const int old_irq[] = { 9, 7, 5, 15 };
unsigned cfg;
- for (cfg = 0; cfg < ARRAY_SIZE(valid_irq); ++cfg) {
- if (irq == valid_irq[cfg])
- return cfg;
- } /* for */
+ if (sscape_type == MEDIA_FX) {
+ for (cfg = 0; cfg < ARRAY_SIZE(old_irq); ++cfg)
+ if (irq == old_irq[cfg])
+ return cfg;
+ } else {
+ for (cfg = 0; cfg < ARRAY_SIZE(valid_irq); ++cfg)
+ if (irq == valid_irq[cfg])
+ return cfg;
+ }
return INVALID_IRQ;
}
-
/*
* Perform certain arcane port-checks to see whether there
* is a SoundScape board lurking behind the given ports.
@@ -842,11 +746,38 @@ static int __devinit detect_sscape(struct soundscape *s, long wss_io)
if (s->type != SSCAPE_VIVO && (d & 0x9f) != 0x0e)
goto _done;
- d = sscape_read_unsafe(s->io_base, GA_HMCTL_REG) & 0x3f;
- sscape_write_unsafe(s->io_base, GA_HMCTL_REG, d | 0xc0);
+ if (s->ic_type == IC_OPUS)
+ activate_ad1845_unsafe(s->io_base);
if (s->type == SSCAPE_VIVO)
wss_io += 4;
+
+ d = sscape_read_unsafe(s->io_base, GA_HMCTL_REG);
+ sscape_write_unsafe(s->io_base, GA_HMCTL_REG, d | 0xc0);
+
+ /* wait for WSS codec */
+ for (d = 0; d < 500; d++) {
+ if ((inb(wss_io) & 0x80) == 0)
+ break;
+ spin_unlock_irqrestore(&s->lock, flags);
+ msleep(1);
+ spin_lock_irqsave(&s->lock, flags);
+ }
+
+ if ((inb(wss_io) & 0x80) != 0)
+ goto _done;
+
+ if (inb(wss_io + 2) == 0xff)
+ goto _done;
+
+ d = sscape_read_unsafe(s->io_base, GA_HMCTL_REG) & 0x3f;
+ sscape_write_unsafe(s->io_base, GA_HMCTL_REG, d);
+
+ if ((inb(wss_io) & 0x80) != 0)
+ s->type = MEDIA_FX;
+
+ d = sscape_read_unsafe(s->io_base, GA_HMCTL_REG);
+ sscape_write_unsafe(s->io_base, GA_HMCTL_REG, d | 0xc0);
/* wait for WSS codec */
for (d = 0; d < 500; d++) {
if ((inb(wss_io) & 0x80) == 0)
@@ -855,14 +786,13 @@ static int __devinit detect_sscape(struct soundscape *s, long wss_io)
msleep(1);
spin_lock_irqsave(&s->lock, flags);
}
- snd_printd(KERN_INFO "init delay = %d ms\n", d);
/*
* SoundScape successfully detected!
*/
retval = 1;
- _done:
+_done:
spin_unlock_irqrestore(&s->lock, flags);
return retval;
}
@@ -873,63 +803,35 @@ static int __devinit detect_sscape(struct soundscape *s, long wss_io)
* to crash the machine. Also check that someone isn't using the hardware
* IOCTL device.
*/
-static int mpu401_open(struct snd_mpu401 * mpu)
+static int mpu401_open(struct snd_mpu401 *mpu)
{
- int err;
-
if (!verify_mpu401(mpu)) {
- snd_printk(KERN_ERR "sscape: MIDI disabled, please load firmware\n");
- err = -ENODEV;
- } else {
- register struct soundscape *sscape = get_mpu401_soundscape(mpu);
- unsigned long flags;
-
- spin_lock_irqsave(&sscape->fwlock, flags);
-
- if (sscape->hw_in_use || (sscape->midi_usage == ULONG_MAX)) {
- err = -EBUSY;
- } else {
- ++(sscape->midi_usage);
- err = 0;
- }
-
- spin_unlock_irqrestore(&sscape->fwlock, flags);
+ snd_printk(KERN_ERR "sscape: MIDI disabled, "
+ "please load firmware\n");
+ return -ENODEV;
}
- return err;
-}
-
-static void mpu401_close(struct snd_mpu401 * mpu)
-{
- register struct soundscape *sscape = get_mpu401_soundscape(mpu);
- unsigned long flags;
-
- spin_lock_irqsave(&sscape->fwlock, flags);
- --(sscape->midi_usage);
- spin_unlock_irqrestore(&sscape->fwlock, flags);
+ return 0;
}
/*
* Initialse an MPU-401 subdevice for MIDI support on the SoundScape.
*/
-static int __devinit create_mpu401(struct snd_card *card, int devnum, unsigned long port, int irq)
+static int __devinit create_mpu401(struct snd_card *card, int devnum,
+ unsigned long port, int irq)
{
struct soundscape *sscape = get_card_soundscape(card);
struct snd_rawmidi *rawmidi;
int err;
- if ((err = snd_mpu401_uart_new(card, devnum,
- MPU401_HW_MPU401,
- port, MPU401_INFO_INTEGRATED,
- irq, IRQF_DISABLED,
- &rawmidi)) == 0) {
- struct snd_mpu401 *mpu = (struct snd_mpu401 *) rawmidi->private_data;
+ err = snd_mpu401_uart_new(card, devnum, MPU401_HW_MPU401, port,
+ MPU401_INFO_INTEGRATED, irq, IRQF_DISABLED,
+ &rawmidi);
+ if (err == 0) {
+ struct snd_mpu401 *mpu = rawmidi->private_data;
mpu->open_input = mpu401_open;
mpu->open_output = mpu401_open;
- mpu->close_input = mpu401_close;
- mpu->close_output = mpu401_close;
mpu->private_data = sscape;
- sscape->mpu = mpu;
initialise_mpu401(mpu);
}
@@ -950,32 +852,34 @@ static int __devinit create_ad1845(struct snd_card *card, unsigned port,
register struct soundscape *sscape = get_card_soundscape(card);
struct snd_wss *chip;
int err;
+ int codec_type = WSS_HW_DETECT;
- if (sscape->type == SSCAPE_VIVO)
- port += 4;
+ switch (sscape->type) {
+ case MEDIA_FX:
+ case SSCAPE:
+ /*
+ * There are some freak examples of early Soundscape cards
+ * with CS4231 instead of AD1848/CS4248. Unfortunately, the
+ * CS4231 works only in CS4248 compatibility mode on
+ * these cards so force it.
+ */
+ if (sscape->ic_type != IC_OPUS)
+ codec_type = WSS_HW_AD1848;
+ break;
- if (dma1 == dma2)
- dma2 = -1;
+ case SSCAPE_VIVO:
+ port += 4;
+ break;
+ default:
+ break;
+ }
err = snd_wss_create(card, port, -1, irq, dma1, dma2,
- WSS_HW_DETECT, WSS_HWSHARE_DMA1, &chip);
+ codec_type, WSS_HWSHARE_DMA1, &chip);
if (!err) {
unsigned long flags;
struct snd_pcm *pcm;
-/*
- * It turns out that the PLAYBACK_ENABLE bit is set
- * by the lowlevel driver ...
- *
-#define AD1845_IFACE_CONFIG \
- (CS4231_AUTOCALIB | CS4231_RECORD_ENABLE | CS4231_PLAYBACK_ENABLE)
- snd_wss_mce_up(chip);
- spin_lock_irqsave(&chip->reg_lock, flags);
- snd_wss_out(chip, CS4231_IFACE_CTRL, AD1845_IFACE_CONFIG);
- spin_unlock_irqrestore(&chip->reg_lock, flags);
- snd_wss_mce_down(chip);
- */
-
if (sscape->type != SSCAPE_VIVO) {
/*
* The input clock frequency on the SoundScape must
@@ -1022,17 +926,10 @@ static int __devinit create_ad1845(struct snd_card *card, unsigned port,
}
}
- strcpy(card->driver, "SoundScape");
- strcpy(card->shortname, pcm->name);
- snprintf(card->longname, sizeof(card->longname),
- "%s at 0x%lx, IRQ %d, DMA1 %d, DMA2 %d\n",
- pcm->name, chip->port, chip->irq,
- chip->dma1, chip->dma2);
-
sscape->chip = chip;
}
- _error:
+_error:
return err;
}
@@ -1051,21 +948,8 @@ static int __devinit create_sscape(int dev, struct snd_card *card)
struct resource *wss_res;
unsigned long flags;
int err;
-
- /*
- * Check that the user didn't pass us garbage data ...
- */
- irq_cfg = get_irq_config(irq[dev]);
- if (irq_cfg == INVALID_IRQ) {
- snd_printk(KERN_ERR "sscape: Invalid IRQ %d\n", irq[dev]);
- return -ENXIO;
- }
-
- mpu_irq_cfg = get_irq_config(mpu_irq[dev]);
- if (mpu_irq_cfg == INVALID_IRQ) {
- printk(KERN_ERR "sscape: Invalid IRQ %d\n", mpu_irq[dev]);
- return -ENXIO;
- }
+ int val;
+ const char *name;
/*
* Grab IO ports that we will need to probe so that we
@@ -1098,41 +982,51 @@ static int __devinit create_sscape(int dev, struct snd_card *card)
}
spin_lock_init(&sscape->lock);
- spin_lock_init(&sscape->fwlock);
sscape->io_res = io_res;
sscape->wss_res = wss_res;
sscape->io_base = port[dev];
if (!detect_sscape(sscape, wss_port[dev])) {
- printk(KERN_ERR "sscape: hardware not detected at 0x%x\n", sscape->io_base);
+ printk(KERN_ERR "sscape: hardware not detected at 0x%x\n",
+ sscape->io_base);
err = -ENODEV;
goto _release_dma;
}
- printk(KERN_INFO "sscape: hardware detected at 0x%x, using IRQ %d, DMA %d\n",
- sscape->io_base, irq[dev], dma[dev]);
+ switch (sscape->type) {
+ case MEDIA_FX:
+ name = "MediaFX/SoundFX";
+ break;
+ case SSCAPE:
+ name = "Soundscape";
+ break;
+ case SSCAPE_PNP:
+ name = "Soundscape PnP";
+ break;
+ case SSCAPE_VIVO:
+ name = "Soundscape VIVO";
+ break;
+ default:
+ name = "unknown Soundscape";
+ break;
+ }
- if (sscape->type != SSCAPE_VIVO) {
- /*
- * Now create the hardware-specific device so that we can
- * load the microcode into the on-board processor.
- * We cannot use the MPU-401 MIDI system until this firmware
- * has been loaded into the card.
- */
- err = snd_hwdep_new(card, "MC68EC000", 0, &(sscape->hw));
- if (err < 0) {
- printk(KERN_ERR "sscape: Failed to create "
- "firmware device\n");
- goto _release_dma;
- }
- strlcpy(sscape->hw->name, "SoundScape M68K",
- sizeof(sscape->hw->name));
- sscape->hw->name[sizeof(sscape->hw->name) - 1] = '\0';
- sscape->hw->iface = SNDRV_HWDEP_IFACE_SSCAPE;
- sscape->hw->ops.open = sscape_hw_open;
- sscape->hw->ops.release = sscape_hw_release;
- sscape->hw->ops.ioctl = sscape_hw_ioctl;
- sscape->hw->private_data = sscape;
+ printk(KERN_INFO "sscape: %s card detected at 0x%x, using IRQ %d, DMA %d\n",
+ name, sscape->io_base, irq[dev], dma[dev]);
+
+ /*
+ * Check that the user didn't pass us garbage data ...
+ */
+ irq_cfg = get_irq_config(sscape->type, irq[dev]);
+ if (irq_cfg == INVALID_IRQ) {
+ snd_printk(KERN_ERR "sscape: Invalid IRQ %d\n", irq[dev]);
+ return -ENXIO;
+ }
+
+ mpu_irq_cfg = get_irq_config(sscape->type, mpu_irq[dev]);
+ if (mpu_irq_cfg == INVALID_IRQ) {
+ snd_printk(KERN_ERR "sscape: Invalid IRQ %d\n", mpu_irq[dev]);
+ return -ENXIO;
}
/*
@@ -1141,9 +1035,6 @@ static int __devinit create_sscape(int dev, struct snd_card *card)
*/
spin_lock_irqsave(&sscape->lock, flags);
- activate_ad1845_unsafe(sscape->io_base);
-
- sscape_write_unsafe(sscape->io_base, GA_INTENA_REG, 0x00); /* disable */
sscape_write_unsafe(sscape->io_base, GA_SMCFGA_REG, 0x2e);
sscape_write_unsafe(sscape->io_base, GA_SMCFGB_REG, 0x00);
@@ -1151,15 +1042,23 @@ static int __devinit create_sscape(int dev, struct snd_card *card)
* Enable and configure the DMA channels ...
*/
sscape_write_unsafe(sscape->io_base, GA_DMACFG_REG, 0x50);
- dma_cfg = (sscape->ic_type == IC_ODIE ? 0x70 : 0x40);
+ dma_cfg = (sscape->ic_type == IC_OPUS ? 0x40 : 0x70);
sscape_write_unsafe(sscape->io_base, GA_DMAA_REG, dma_cfg);
sscape_write_unsafe(sscape->io_base, GA_DMAB_REG, 0x20);
- sscape_write_unsafe(sscape->io_base,
- GA_INTCFG_REG, 0xf0 | (mpu_irq_cfg << 2) | mpu_irq_cfg);
+ mpu_irq_cfg |= mpu_irq_cfg << 2;
+ val = sscape_read_unsafe(sscape->io_base, GA_HMCTL_REG) & 0xF7;
+ if (joystick[dev])
+ val |= 8;
+ sscape_write_unsafe(sscape->io_base, GA_HMCTL_REG, val | 0x10);
+ sscape_write_unsafe(sscape->io_base, GA_INTCFG_REG, 0xf0 | mpu_irq_cfg);
sscape_write_unsafe(sscape->io_base,
GA_CDCFG_REG, 0x09 | DMA_8BIT
| (dma[dev] << 4) | (irq_cfg << 1));
+ /*
+ * Enable the master IRQ ...
+ */
+ sscape_write_unsafe(sscape->io_base, GA_INTENA_REG, 0x80);
spin_unlock_irqrestore(&sscape->lock, flags);
@@ -1170,32 +1069,56 @@ static int __devinit create_sscape(int dev, struct snd_card *card)
err = create_ad1845(card, wss_port[dev], irq[dev],
dma[dev], dma2[dev]);
if (err < 0) {
- printk(KERN_ERR "sscape: No AD1845 device at 0x%lx, IRQ %d\n",
- wss_port[dev], irq[dev]);
+ snd_printk(KERN_ERR
+ "sscape: No AD1845 device at 0x%lx, IRQ %d\n",
+ wss_port[dev], irq[dev]);
goto _release_dma;
}
+ strcpy(card->driver, "SoundScape");
+ strcpy(card->shortname, name);
+ snprintf(card->longname, sizeof(card->longname),
+ "%s at 0x%lx, IRQ %d, DMA1 %d, DMA2 %d\n",
+ name, sscape->chip->port, sscape->chip->irq,
+ sscape->chip->dma1, sscape->chip->dma2);
+
#define MIDI_DEVNUM 0
if (sscape->type != SSCAPE_VIVO) {
- err = create_mpu401(card, MIDI_DEVNUM, port[dev], mpu_irq[dev]);
- if (err < 0) {
- printk(KERN_ERR "sscape: Failed to create "
- "MPU-401 device at 0x%lx\n",
- port[dev]);
- goto _release_dma;
- }
+ err = sscape_upload_bootblock(card);
+ if (err >= 0)
+ err = sscape_upload_microcode(card, err);
- /*
- * Enable the master IRQ ...
- */
- sscape_write(sscape, GA_INTENA_REG, 0x80);
+ if (err == 0) {
+ err = create_mpu401(card, MIDI_DEVNUM, port[dev],
+ mpu_irq[dev]);
+ if (err < 0) {
+ snd_printk(KERN_ERR "sscape: Failed to create "
+ "MPU-401 device at 0x%lx\n",
+ port[dev]);
+ goto _release_dma;
+ }
- /*
- * Initialize mixer
- */
- sscape->midi_vol = 0;
- host_write_ctrl_unsafe(sscape->io_base, CMD_SET_MIDI_VOL, 100);
- host_write_ctrl_unsafe(sscape->io_base, 0, 100);
- host_write_ctrl_unsafe(sscape->io_base, CMD_XXX_MIDI_VOL, 100);
+ /*
+ * Initialize mixer
+ */
+ spin_lock_irqsave(&sscape->lock, flags);
+ sscape->midi_vol = 0;
+ host_write_ctrl_unsafe(sscape->io_base,
+ CMD_SET_MIDI_VOL, 100);
+ host_write_ctrl_unsafe(sscape->io_base,
+ sscape->midi_vol, 100);
+ host_write_ctrl_unsafe(sscape->io_base,
+ CMD_XXX_MIDI_VOL, 100);
+ host_write_ctrl_unsafe(sscape->io_base,
+ sscape->midi_vol, 100);
+ host_write_ctrl_unsafe(sscape->io_base,
+ CMD_SET_EXTMIDI, 100);
+ host_write_ctrl_unsafe(sscape->io_base,
+ 0, 100);
+ host_write_ctrl_unsafe(sscape->io_base, CMD_ACK, 100);
+
+ set_midi_mode_unsafe(sscape->io_base);
+ spin_unlock_irqrestore(&sscape->lock, flags);
+ }
}
/*
@@ -1231,7 +1154,8 @@ static int __devinit snd_sscape_match(struct device *pdev, unsigned int i)
mpu_irq[i] == SNDRV_AUTO_IRQ ||
dma[i] == SNDRV_AUTO_DMA) {
printk(KERN_INFO
- "sscape: insufficient parameters, need IO, IRQ, MPU-IRQ and DMA\n");
+ "sscape: insufficient parameters, "
+ "need IO, IRQ, MPU-IRQ and DMA\n");
return 0;
}
@@ -1253,13 +1177,15 @@ static int __devinit snd_sscape_probe(struct device *pdev, unsigned int dev)
sscape->type = SSCAPE;
dma[dev] &= 0x03;
+ snd_card_set_dev(card, pdev);
+
ret = create_sscape(dev, card);
if (ret < 0)
goto _release_card;
- snd_card_set_dev(card, pdev);
- if ((ret = snd_card_register(card)) < 0) {
- printk(KERN_ERR "sscape: Failed to register sound card\n");
+ ret = snd_card_register(card);
+ if (ret < 0) {
+ snd_printk(KERN_ERR "sscape: Failed to register sound card\n");
goto _release_card;
}
dev_set_drvdata(pdev, card);
@@ -1311,36 +1237,20 @@ static int __devinit sscape_pnp_detect(struct pnp_card_link *pcard,
* Allow this function to fail *quietly* if all the ISA PnP
* devices were configured using module parameters instead.
*/
- if ((idx = get_next_autoindex(idx)) >= SNDRV_CARDS)
+ idx = get_next_autoindex(idx);
+ if (idx >= SNDRV_CARDS)
return -ENOSPC;
/*
- * We have found a candidate ISA PnP card. Now we
- * have to check that it has the devices that we
- * expect it to have.
- *
- * We will NOT try and autoconfigure all of the resources
- * needed and then activate the card as we are assuming that
- * has already been done at boot-time using /proc/isapnp.
- * We shall simply try to give each active card the resources
- * that it wants. This is a sensible strategy for a modular
- * system where unused modules are unloaded regularly.
- *
- * This strategy is utterly useless if we compile the driver
- * into the kernel, of course.
- */
- // printk(KERN_INFO "sscape: %s\n", card->name);
-
- /*
* Check that we still have room for another sound card ...
*/
dev = pnp_request_card_device(pcard, pid->devs[0].id, NULL);
- if (! dev)
+ if (!dev)
return -ENODEV;
if (!pnp_is_active(dev)) {
if (pnp_activate_dev(dev) < 0) {
- printk(KERN_INFO "sscape: device is inactive\n");
+ snd_printk(KERN_INFO "sscape: device is inactive\n");
return -EBUSY;
}
}
@@ -1378,14 +1288,15 @@ static int __devinit sscape_pnp_detect(struct pnp_card_link *pcard,
wss_port[idx] = pnp_port_start(dev, 1);
dma2[idx] = pnp_dma(dev, 1);
}
+ snd_card_set_dev(card, &pcard->card->dev);
ret = create_sscape(idx, card);
if (ret < 0)
goto _release_card;
- snd_card_set_dev(card, &pcard->card->dev);
- if ((ret = snd_card_register(card)) < 0) {
- printk(KERN_ERR "sscape: Failed to register sound card\n");
+ ret = snd_card_register(card);
+ if (ret < 0) {
+ snd_printk(KERN_ERR "sscape: Failed to register sound card\n");
goto _release_card;
}
diff --git a/sound/isa/wss/wss_lib.c b/sound/isa/wss/wss_lib.c
index 5d2ba1b749a..2ba18978b41 100644
--- a/sound/isa/wss/wss_lib.c
+++ b/sound/isa/wss/wss_lib.c
@@ -2198,84 +2198,61 @@ EXPORT_SYMBOL(snd_wss_put_double);
static const DECLARE_TLV_DB_SCALE(db_scale_6bit, -9450, 150, 0);
static const DECLARE_TLV_DB_SCALE(db_scale_5bit_12db_max, -3450, 150, 0);
static const DECLARE_TLV_DB_SCALE(db_scale_rec_gain, 0, 150, 0);
+static const DECLARE_TLV_DB_SCALE(db_scale_4bit, -4500, 300, 0);
-static struct snd_kcontrol_new snd_ad1848_controls[] = {
-WSS_DOUBLE("PCM Playback Switch", 0, CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT,
- 7, 7, 1, 1),
+static struct snd_kcontrol_new snd_wss_controls[] = {
+WSS_DOUBLE("PCM Playback Switch", 0,
+ CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 7, 7, 1, 1),
WSS_DOUBLE_TLV("PCM Playback Volume", 0,
- CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 0, 0, 63, 1,
- db_scale_6bit),
+ CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 0, 0, 63, 1,
+ db_scale_6bit),
WSS_DOUBLE("Aux Playback Switch", 0,
- CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 7, 7, 1, 1),
+ CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 7, 7, 1, 1),
WSS_DOUBLE_TLV("Aux Playback Volume", 0,
- CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 0, 0, 31, 1,
- db_scale_5bit_12db_max),
+ CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 0, 0, 31, 1,
+ db_scale_5bit_12db_max),
WSS_DOUBLE("Aux Playback Switch", 1,
- CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 7, 7, 1, 1),
+ CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 7, 7, 1, 1),
WSS_DOUBLE_TLV("Aux Playback Volume", 1,
- CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 0, 0, 31, 1,
- db_scale_5bit_12db_max),
+ CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 0, 0, 31, 1,
+ db_scale_5bit_12db_max),
WSS_DOUBLE_TLV("Capture Volume", 0, CS4231_LEFT_INPUT, CS4231_RIGHT_INPUT,
0, 0, 15, 0, db_scale_rec_gain),
{
- .name = "Capture Source",
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Capture Source",
.info = snd_wss_info_mux,
.get = snd_wss_get_mux,
.put = snd_wss_put_mux,
},
-WSS_SINGLE("Loopback Capture Switch", 0, CS4231_LOOPBACK, 0, 1, 0),
-WSS_SINGLE_TLV("Loopback Capture Volume", 0, CS4231_LOOPBACK, 1, 63, 0,
- db_scale_6bit),
-};
-
-static struct snd_kcontrol_new snd_wss_controls[] = {
-WSS_DOUBLE("PCM Playback Switch", 0,
- CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 7, 7, 1, 1),
-WSS_DOUBLE("PCM Playback Volume", 0,
- CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 0, 0, 63, 1),
+WSS_DOUBLE("Mic Boost", 0,
+ CS4231_LEFT_INPUT, CS4231_RIGHT_INPUT, 5, 5, 1, 0),
+WSS_SINGLE("Loopback Capture Switch", 0,
+ CS4231_LOOPBACK, 0, 1, 0),
+WSS_SINGLE_TLV("Loopback Capture Volume", 0, CS4231_LOOPBACK, 2, 63, 1,
+ db_scale_6bit),
WSS_DOUBLE("Line Playback Switch", 0,
CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 7, 7, 1, 1),
-WSS_DOUBLE("Line Playback Volume", 0,
- CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 0, 0, 31, 1),
-WSS_DOUBLE("Aux Playback Switch", 0,
- CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 7, 7, 1, 1),
-WSS_DOUBLE("Aux Playback Volume", 0,
- CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 0, 0, 31, 1),
-WSS_DOUBLE("Aux Playback Switch", 1,
- CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 7, 7, 1, 1),
-WSS_DOUBLE("Aux Playback Volume", 1,
- CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 0, 0, 31, 1),
+WSS_DOUBLE_TLV("Line Playback Volume", 0,
+ CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 0, 0, 31, 1,
+ db_scale_5bit_12db_max),
WSS_SINGLE("Mono Playback Switch", 0,
CS4231_MONO_CTRL, 7, 1, 1),
-WSS_SINGLE("Mono Playback Volume", 0,
- CS4231_MONO_CTRL, 0, 15, 1),
+WSS_SINGLE_TLV("Mono Playback Volume", 0,
+ CS4231_MONO_CTRL, 0, 15, 1,
+ db_scale_4bit),
WSS_SINGLE("Mono Output Playback Switch", 0,
CS4231_MONO_CTRL, 6, 1, 1),
WSS_SINGLE("Mono Output Playback Bypass", 0,
CS4231_MONO_CTRL, 5, 1, 0),
-WSS_DOUBLE("Capture Volume", 0,
- CS4231_LEFT_INPUT, CS4231_RIGHT_INPUT, 0, 0, 15, 0),
-{
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "Capture Source",
- .info = snd_wss_info_mux,
- .get = snd_wss_get_mux,
- .put = snd_wss_put_mux,
-},
-WSS_DOUBLE("Mic Boost", 0,
- CS4231_LEFT_INPUT, CS4231_RIGHT_INPUT, 5, 5, 1, 0),
-WSS_SINGLE("Loopback Capture Switch", 0,
- CS4231_LOOPBACK, 0, 1, 0),
-WSS_SINGLE("Loopback Capture Volume", 0,
- CS4231_LOOPBACK, 2, 63, 1)
};
static struct snd_kcontrol_new snd_opti93x_controls[] = {
WSS_DOUBLE("Master Playback Switch", 0,
OPTi93X_OUT_LEFT, OPTi93X_OUT_RIGHT, 7, 7, 1, 1),
-WSS_DOUBLE("Master Playback Volume", 0,
- OPTi93X_OUT_LEFT, OPTi93X_OUT_RIGHT, 1, 1, 31, 1),
+WSS_DOUBLE_TLV("Master Playback Volume", 0,
+ OPTi93X_OUT_LEFT, OPTi93X_OUT_RIGHT, 1, 1, 31, 1,
+ db_scale_6bit),
WSS_DOUBLE("PCM Playback Switch", 0,
CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 7, 7, 1, 1),
WSS_DOUBLE("PCM Playback Volume", 0,
@@ -2334,22 +2311,21 @@ int snd_wss_mixer(struct snd_wss *chip)
if (err < 0)
return err;
}
- else if (chip->hardware & WSS_HW_AD1848_MASK)
- for (idx = 0; idx < ARRAY_SIZE(snd_ad1848_controls); idx++) {
- err = snd_ctl_add(card,
- snd_ctl_new1(&snd_ad1848_controls[idx],
- chip));
- if (err < 0)
- return err;
- }
- else
- for (idx = 0; idx < ARRAY_SIZE(snd_wss_controls); idx++) {
+ else {
+ int count = ARRAY_SIZE(snd_wss_controls);
+
+ /* Use only the first 11 entries on AD1848 */
+ if (chip->hardware & WSS_HW_AD1848_MASK)
+ count = 11;
+
+ for (idx = 0; idx < count; idx++) {
err = snd_ctl_add(card,
snd_ctl_new1(&snd_wss_controls[idx],
chip));
if (err < 0)
return err;
}
+ }
return 0;
}
EXPORT_SYMBOL(snd_wss_mixer);
diff --git a/sound/mips/hal2.c b/sound/mips/hal2.c
index c52691c2fc4..9a88cdfd952 100644
--- a/sound/mips/hal2.c
+++ b/sound/mips/hal2.c
@@ -915,7 +915,7 @@ static int __devinit hal2_probe(struct platform_device *pdev)
return 0;
}
-static int __exit hal2_remove(struct platform_device *pdev)
+static int __devexit hal2_remove(struct platform_device *pdev)
{
struct snd_card *card = platform_get_drvdata(pdev);
diff --git a/sound/mips/sgio2audio.c b/sound/mips/sgio2audio.c
index e497525bc11..8691f4cf619 100644
--- a/sound/mips/sgio2audio.c
+++ b/sound/mips/sgio2audio.c
@@ -973,7 +973,7 @@ static int __devinit snd_sgio2audio_probe(struct platform_device *pdev)
return 0;
}
-static int __exit snd_sgio2audio_remove(struct platform_device *pdev)
+static int __devexit snd_sgio2audio_remove(struct platform_device *pdev)
{
struct snd_card *card = platform_get_drvdata(pdev);
diff --git a/sound/oss/Kconfig b/sound/oss/Kconfig
index bcf2a0698d5..135a2b77cc4 100644
--- a/sound/oss/Kconfig
+++ b/sound/oss/Kconfig
@@ -287,18 +287,6 @@ config SOUND_DMAP
Say Y unless you have 16MB or more RAM or a PCI sound card.
-config SOUND_SSCAPE
- tristate "Ensoniq SoundScape support"
- help
- Answer Y if you have a sound card based on the Ensoniq SoundScape
- chipset. Such cards are being manufactured at least by Ensoniq, Spea
- and Reveal (Reveal makes also other cards).
-
- If you compile the driver into the kernel, you have to add
- "sscape=<io>,<irq>,<dma>,<mpuio>,<mpuirq>" to the kernel command
- line.
-
-
config SOUND_VMIDI
tristate "Loopback MIDI device support"
help
diff --git a/sound/oss/Makefile b/sound/oss/Makefile
index e0ae4d4d6a5..567b8a74178 100644
--- a/sound/oss/Makefile
+++ b/sound/oss/Makefile
@@ -13,7 +13,6 @@ obj-$(CONFIG_SOUND_SH_DAC_AUDIO) += sh_dac_audio.o
obj-$(CONFIG_SOUND_AEDSP16) += aedsp16.o
obj-$(CONFIG_SOUND_PSS) += pss.o ad1848.o mpu401.o
obj-$(CONFIG_SOUND_TRIX) += trix.o ad1848.o sb_lib.o uart401.o
-obj-$(CONFIG_SOUND_SSCAPE) += sscape.o ad1848.o mpu401.o
obj-$(CONFIG_SOUND_MSS) += ad1848.o
obj-$(CONFIG_SOUND_PAS) += pas2.o sb.o sb_lib.o uart401.o
obj-$(CONFIG_SOUND_SB) += sb.o sb_lib.o uart401.o
diff --git a/sound/oss/sh_dac_audio.c b/sound/oss/sh_dac_audio.c
index b2ed8757542..4153752507e 100644
--- a/sound/oss/sh_dac_audio.c
+++ b/sound/oss/sh_dac_audio.c
@@ -164,9 +164,6 @@ static ssize_t dac_audio_write(struct file *file, const char *buf, size_t count,
int free;
int nbytes;
- if (count < 0)
- return -EINVAL;
-
if (!count) {
dac_audio_sync();
return 0;
diff --git a/sound/oss/sscape.c b/sound/oss/sscape.c
deleted file mode 100644
index 30c36d1f35d..00000000000
--- a/sound/oss/sscape.c
+++ /dev/null
@@ -1,1480 +0,0 @@
-/*
- * sound/oss/sscape.c
- *
- * Low level driver for Ensoniq SoundScape
- *
- *
- * Copyright (C) by Hannu Savolainen 1993-1997
- *
- * OSS/Free for Linux is distributed under the GNU GENERAL PUBLIC LICENSE (GPL)
- * Version 2 (June 1991). See the "COPYING" file distributed with this software
- * for more info.
- *
- *
- * Thomas Sailer : ioctl code reworked (vmalloc/vfree removed)
- * Sergey Smitienko : ensoniq p'n'p support
- * Christoph Hellwig : adapted to module_init/module_exit
- * Bartlomiej Zolnierkiewicz : added __init to attach_sscape()
- * Chris Rankin : Specify that this module owns the coprocessor
- * Arnaldo C. de Melo : added missing restore_flags in sscape_pnp_upload_file
- */
-
-#include <linux/init.h>
-#include <linux/module.h>
-
-#include "sound_config.h"
-#include "sound_firmware.h"
-
-#include <linux/types.h>
-#include <linux/errno.h>
-#include <linux/signal.h>
-#include <linux/fcntl.h>
-#include <linux/ctype.h>
-#include <linux/stddef.h>
-#include <linux/kmod.h>
-#include <asm/dma.h>
-#include <asm/io.h>
-#include <linux/wait.h>
-#include <linux/slab.h>
-#include <linux/ioport.h>
-#include <linux/delay.h>
-#include <linux/proc_fs.h>
-#include <linux/mm.h>
-#include <linux/spinlock.h>
-
-#include "coproc.h"
-
-#include "ad1848.h"
-#include "mpu401.h"
-
-/*
- * I/O ports
- */
-#define MIDI_DATA 0
-#define MIDI_CTRL 1
-#define HOST_CTRL 2
-#define TX_READY 0x02
-#define RX_READY 0x01
-#define HOST_DATA 3
-#define ODIE_ADDR 4
-#define ODIE_DATA 5
-
-/*
- * Indirect registers
- */
-
-#define GA_INTSTAT_REG 0
-#define GA_INTENA_REG 1
-#define GA_DMAA_REG 2
-#define GA_DMAB_REG 3
-#define GA_INTCFG_REG 4
-#define GA_DMACFG_REG 5
-#define GA_CDCFG_REG 6
-#define GA_SMCFGA_REG 7
-#define GA_SMCFGB_REG 8
-#define GA_HMCTL_REG 9
-
-/*
- * DMA channel identifiers (A and B)
- */
-
-#define SSCAPE_DMA_A 0
-#define SSCAPE_DMA_B 1
-
-#define PORT(name) (devc->base+name)
-
-/*
- * Host commands recognized by the OBP microcode
- */
-
-#define CMD_GEN_HOST_ACK 0x80
-#define CMD_GEN_MPU_ACK 0x81
-#define CMD_GET_BOARD_TYPE 0x82
-#define CMD_SET_CONTROL 0x88 /* Old firmware only */
-#define CMD_GET_CONTROL 0x89 /* Old firmware only */
-#define CTL_MASTER_VOL 0
-#define CTL_MIC_MODE 2
-#define CTL_SYNTH_VOL 4
-#define CTL_WAVE_VOL 7
-#define CMD_SET_EXTMIDI 0x8a
-#define CMD_GET_EXTMIDI 0x8b
-#define CMD_SET_MT32 0x8c
-#define CMD_GET_MT32 0x8d
-
-#define CMD_ACK 0x80
-
-#define IC_ODIE 1
-#define IC_OPUS 2
-
-typedef struct sscape_info
-{
- int base, irq, dma;
-
- int codec, codec_irq; /* required to setup pnp cards*/
- int codec_type;
- int ic_type;
- char* raw_buf;
- unsigned long raw_buf_phys;
- int buffsize; /* -------------------------- */
- spinlock_t lock;
- int ok; /* Properly detected */
- int failed;
- int dma_allocated;
- int codec_audiodev;
- int opened;
- int *osp;
- int my_audiodev;
-} sscape_info;
-
-static struct sscape_info adev_info = {
- 0
-};
-
-static struct sscape_info *devc = &adev_info;
-static int sscape_mididev = -1;
-
-/* Some older cards have assigned interrupt bits differently than new ones */
-static char valid_interrupts_old[] = {
- 9, 7, 5, 15
-};
-
-static char valid_interrupts_new[] = {
- 9, 5, 7, 10
-};
-
-static char *valid_interrupts = valid_interrupts_new;
-
-/*
- * See the bottom of the driver. This can be set by spea =0/1.
- */
-
-#ifdef REVEAL_SPEA
-static char old_hardware = 1;
-#else
-static char old_hardware;
-#endif
-
-static void sleep(unsigned howlong)
-{
- current->state = TASK_INTERRUPTIBLE;
- schedule_timeout(howlong);
-}
-
-static unsigned char sscape_read(struct sscape_info *devc, int reg)
-{
- unsigned long flags;
- unsigned char val;
-
- spin_lock_irqsave(&devc->lock,flags);
- outb(reg, PORT(ODIE_ADDR));
- val = inb(PORT(ODIE_DATA));
- spin_unlock_irqrestore(&devc->lock,flags);
- return val;
-}
-
-static void __sscape_write(int reg, int data)
-{
- outb(reg, PORT(ODIE_ADDR));
- outb(data, PORT(ODIE_DATA));
-}
-
-static void sscape_write(struct sscape_info *devc, int reg, int data)
-{
- unsigned long flags;
-
- spin_lock_irqsave(&devc->lock,flags);
- __sscape_write(reg, data);
- spin_unlock_irqrestore(&devc->lock,flags);
-}
-
-static unsigned char sscape_pnp_read_codec(sscape_info* devc, unsigned char reg)
-{
- unsigned char res;
- unsigned long flags;
-
- spin_lock_irqsave(&devc->lock,flags);
- outb( reg, devc -> codec);
- res = inb (devc -> codec + 1);
- spin_unlock_irqrestore(&devc->lock,flags);
- return res;
-
-}
-
-static void sscape_pnp_write_codec(sscape_info* devc, unsigned char reg, unsigned char data)
-{
- unsigned long flags;
-
- spin_lock_irqsave(&devc->lock,flags);
- outb( reg, devc -> codec);
- outb( data, devc -> codec + 1);
- spin_unlock_irqrestore(&devc->lock,flags);
-}
-
-static void host_open(struct sscape_info *devc)
-{
- outb((0x00), PORT(HOST_CTRL)); /* Put the board to the host mode */
-}
-
-static void host_close(struct sscape_info *devc)
-{
- outb((0x03), PORT(HOST_CTRL)); /* Put the board to the MIDI mode */
-}
-
-static int host_write(struct sscape_info *devc, unsigned char *data, int count)
-{
- unsigned long flags;
- int i, timeout_val;
-
- spin_lock_irqsave(&devc->lock,flags);
- /*
- * Send the command and data bytes
- */
-
- for (i = 0; i < count; i++)
- {
- for (timeout_val = 10000; timeout_val > 0; timeout_val--)
- if (inb(PORT(HOST_CTRL)) & TX_READY)
- break;
-
- if (timeout_val <= 0)
- {
- spin_unlock_irqrestore(&devc->lock,flags);
- return 0;
- }
- outb(data[i], PORT(HOST_DATA));
- }
- spin_unlock_irqrestore(&devc->lock,flags);
- return 1;
-}
-
-static int host_read(struct sscape_info *devc)
-{
- unsigned long flags;
- int timeout_val;
- unsigned char data;
-
- spin_lock_irqsave(&devc->lock,flags);
- /*
- * Read a byte
- */
-
- for (timeout_val = 10000; timeout_val > 0; timeout_val--)
- if (inb(PORT(HOST_CTRL)) & RX_READY)
- break;
-
- if (timeout_val <= 0)
- {
- spin_unlock_irqrestore(&devc->lock,flags);
- return -1;
- }
- data = inb(PORT(HOST_DATA));
- spin_unlock_irqrestore(&devc->lock,flags);
- return data;
-}
-
-#if 0 /* unused */
-static int host_command1(struct sscape_info *devc, int cmd)
-{
- unsigned char buf[10];
- buf[0] = (unsigned char) (cmd & 0xff);
- return host_write(devc, buf, 1);
-}
-#endif /* unused */
-
-
-static int host_command2(struct sscape_info *devc, int cmd, int parm1)
-{
- unsigned char buf[10];
-
- buf[0] = (unsigned char) (cmd & 0xff);
- buf[1] = (unsigned char) (parm1 & 0xff);
-
- return host_write(devc, buf, 2);
-}
-
-static int host_command3(struct sscape_info *devc, int cmd, int parm1, int parm2)
-{
- unsigned char buf[10];
-
- buf[0] = (unsigned char) (cmd & 0xff);
- buf[1] = (unsigned char) (parm1 & 0xff);
- buf[2] = (unsigned char) (parm2 & 0xff);
- return host_write(devc, buf, 3);
-}
-
-static void set_mt32(struct sscape_info *devc, int value)
-{
- host_open(devc);
- host_command2(devc, CMD_SET_MT32, value ? 1 : 0);
- if (host_read(devc) != CMD_ACK)
- {
- /* printk( "SNDSCAPE: Setting MT32 mode failed\n"); */
- }
- host_close(devc);
-}
-
-static void set_control(struct sscape_info *devc, int ctrl, int value)
-{
- host_open(devc);
- host_command3(devc, CMD_SET_CONTROL, ctrl, value);
- if (host_read(devc) != CMD_ACK)
- {
- /* printk( "SNDSCAPE: Setting control (%d) failed\n", ctrl); */
- }
- host_close(devc);
-}
-
-static void do_dma(struct sscape_info *devc, int dma_chan, unsigned long buf, int blk_size, int mode)
-{
- unsigned char temp;
-
- if (dma_chan != SSCAPE_DMA_A)
- {
- printk(KERN_WARNING "soundscape: Tried to use DMA channel != A. Why?\n");
- return;
- }
- audio_devs[devc->codec_audiodev]->flags &= ~DMA_AUTOMODE;
- DMAbuf_start_dma(devc->codec_audiodev, buf, blk_size, mode);
- audio_devs[devc->codec_audiodev]->flags |= DMA_AUTOMODE;
-
- temp = devc->dma << 4; /* Setup DMA channel select bits */
- if (devc->dma <= 3)
- temp |= 0x80; /* 8 bit DMA channel */
-
- temp |= 1; /* Trigger DMA */
- sscape_write(devc, GA_DMAA_REG, temp);
- temp &= 0xfe; /* Clear DMA trigger */
- sscape_write(devc, GA_DMAA_REG, temp);
-}
-
-static int verify_mpu(struct sscape_info *devc)
-{
- /*
- * The SoundScape board could be in three modes (MPU, 8250 and host).
- * If the card is not in the MPU mode, enabling the MPU driver will
- * cause infinite loop (the driver believes that there is always some
- * received data in the buffer.
- *
- * Detect this by looking if there are more than 10 received MIDI bytes
- * (0x00) in the buffer.
- */
-
- int i;
-
- for (i = 0; i < 10; i++)
- {
- if (inb(devc->base + HOST_CTRL) & 0x80)
- return 1;
-
- if (inb(devc->base) != 0x00)
- return 1;
- }
- printk(KERN_WARNING "SoundScape: The device is not in the MPU-401 mode\n");
- return 0;
-}
-
-static int sscape_coproc_open(void *dev_info, int sub_device)
-{
- if (sub_device == COPR_MIDI)
- {
- set_mt32(devc, 0);
- if (!verify_mpu(devc))
- return -EIO;
- }
- return 0;
-}
-
-static void sscape_coproc_close(void *dev_info, int sub_device)
-{
- struct sscape_info *devc = dev_info;
- unsigned long flags;
-
- spin_lock_irqsave(&devc->lock,flags);
- if (devc->dma_allocated)
- {
- __sscape_write(GA_DMAA_REG, 0x20); /* DMA channel disabled */
- devc->dma_allocated = 0;
- }
- spin_unlock_irqrestore(&devc->lock,flags);
- return;
-}
-
-static void sscape_coproc_reset(void *dev_info)
-{
-}
-
-static int sscape_download_boot(struct sscape_info *devc, unsigned char *block, int size, int flag)
-{
- unsigned long flags;
- unsigned char temp;
- volatile int done, timeout_val;
- static unsigned char codec_dma_bits;
-
- if (flag & CPF_FIRST)
- {
- /*
- * First block. Have to allocate DMA and to reset the board
- * before continuing.
- */
-
- spin_lock_irqsave(&devc->lock,flags);
- codec_dma_bits = sscape_read(devc, GA_CDCFG_REG);
-
- if (devc->dma_allocated == 0)
- devc->dma_allocated = 1;
-
- spin_unlock_irqrestore(&devc->lock,flags);
-
- sscape_write(devc, GA_HMCTL_REG,
- (temp = sscape_read(devc, GA_HMCTL_REG)) & 0x3f); /*Reset */
-
- for (timeout_val = 10000; timeout_val > 0; timeout_val--)
- sscape_read(devc, GA_HMCTL_REG); /* Delay */
-
- /* Take board out of reset */
- sscape_write(devc, GA_HMCTL_REG,
- (temp = sscape_read(devc, GA_HMCTL_REG)) | 0x80);
- }
- /*
- * Transfer one code block using DMA
- */
- if (audio_devs[devc->codec_audiodev]->dmap_out->raw_buf == NULL)
- {
- printk(KERN_WARNING "soundscape: DMA buffer not available\n");
- return 0;
- }
- memcpy(audio_devs[devc->codec_audiodev]->dmap_out->raw_buf, block, size);
-
- spin_lock_irqsave(&devc->lock,flags);
-
- /******** INTERRUPTS DISABLED NOW ********/
-
- do_dma(devc, SSCAPE_DMA_A,
- audio_devs[devc->codec_audiodev]->dmap_out->raw_buf_phys,
- size, DMA_MODE_WRITE);
-
- /*
- * Wait until transfer completes.
- */
-
- done = 0;
- timeout_val = 30;
- while (!done && timeout_val-- > 0)
- {
- int resid;
-
- if (HZ / 50)
- sleep(HZ / 50);
- clear_dma_ff(devc->dma);
- if ((resid = get_dma_residue(devc->dma)) == 0)
- done = 1;
- }
-
- spin_unlock_irqrestore(&devc->lock,flags);
- if (!done)
- return 0;
-
- if (flag & CPF_LAST)
- {
- /*
- * Take the board out of reset
- */
- outb((0x00), PORT(HOST_CTRL));
- outb((0x00), PORT(MIDI_CTRL));
-
- temp = sscape_read(devc, GA_HMCTL_REG);
- temp |= 0x40;
- sscape_write(devc, GA_HMCTL_REG, temp); /* Kickstart the board */
-
- /*
- * Wait until the ODB wakes up
- */
- spin_lock_irqsave(&devc->lock,flags);
- done = 0;
- timeout_val = 5 * HZ;
- while (!done && timeout_val-- > 0)
- {
- unsigned char x;
-
- sleep(1);
- x = inb(PORT(HOST_DATA));
- if (x == 0xff || x == 0xfe) /* OBP startup acknowledge */
- {
- DDB(printk("Soundscape: Acknowledge = %x\n", x));
- done = 1;
- }
- }
- sscape_write(devc, GA_CDCFG_REG, codec_dma_bits);
-
- spin_unlock_irqrestore(&devc->lock,flags);
- if (!done)
- {
- printk(KERN_ERR "soundscape: The OBP didn't respond after code download\n");
- return 0;
- }
- spin_lock_irqsave(&devc->lock,flags);
- done = 0;
- timeout_val = 5 * HZ;
- while (!done && timeout_val-- > 0)
- {
- sleep(1);
- if (inb(PORT(HOST_DATA)) == 0xfe) /* Host startup acknowledge */
- done = 1;
- }
- spin_unlock_irqrestore(&devc->lock,flags);
- if (!done)
- {
- printk(KERN_ERR "soundscape: OBP Initialization failed.\n");
- return 0;
- }
- printk(KERN_INFO "SoundScape board initialized OK\n");
- set_control(devc, CTL_MASTER_VOL, 100);
- set_control(devc, CTL_SYNTH_VOL, 100);
-
-#ifdef SSCAPE_DEBUG3
- /*
- * Temporary debugging aid. Print contents of the registers after
- * downloading the code.
- */
- {
- int i;
-
- for (i = 0; i < 13; i++)
- printk("I%d = %02x (new value)\n", i, sscape_read(devc, i));
- }
-#endif
-
- }
- return 1;
-}
-
-static int download_boot_block(void *dev_info, copr_buffer * buf)
-{
- if (buf->len <= 0 || buf->len > sizeof(buf->data))
- return -EINVAL;
-
- if (!sscape_download_boot(devc, buf->data, buf->len, buf->flags))
- {
- printk(KERN_ERR "soundscape: Unable to load microcode block to the OBP.\n");
- return -EIO;
- }
- return 0;
-}
-
-static int sscape_coproc_ioctl(void *dev_info, unsigned int cmd, void __user *arg, int local)
-{
- copr_buffer *buf;
- int err;
-
- switch (cmd)
- {
- case SNDCTL_COPR_RESET:
- sscape_coproc_reset(dev_info);
- return 0;
-
- case SNDCTL_COPR_LOAD:
- buf = (copr_buffer *) vmalloc(sizeof(copr_buffer));
- if (buf == NULL)
- return -ENOSPC;
- if (copy_from_user(buf, arg, sizeof(copr_buffer)))
- {
- vfree(buf);
- return -EFAULT;
- }
- err = download_boot_block(dev_info, buf);
- vfree(buf);
- return err;
-
- default:
- return -EINVAL;
- }
-}
-
-static coproc_operations sscape_coproc_operations =
-{
- "SoundScape M68K",
- THIS_MODULE,
- sscape_coproc_open,
- sscape_coproc_close,
- sscape_coproc_ioctl,
- sscape_coproc_reset,
- &adev_info
-};
-
-static struct resource *sscape_ports;
-static int sscape_is_pnp;
-
-static void __init attach_sscape(struct address_info *hw_config)
-{
-#ifndef SSCAPE_REGS
- /*
- * Config register values for Spea/V7 Media FX and Ensoniq S-2000.
- * These values are card
- * dependent. If you have another SoundScape based card, you have to
- * find the correct values. Do the following:
- * - Compile this driver with SSCAPE_DEBUG1 defined.
- * - Shut down and power off your machine.
- * - Boot with DOS so that the SSINIT.EXE program is run.
- * - Warm boot to {Linux|SYSV|BSD} and write down the lines displayed
- * when detecting the SoundScape.
- * - Modify the following list to use the values printed during boot.
- * Undefine the SSCAPE_DEBUG1
- */
-#define SSCAPE_REGS { \
-/* I0 */ 0x00, \
-/* I1 */ 0xf0, /* Note! Ignored. Set always to 0xf0 */ \
-/* I2 */ 0x20, /* Note! Ignored. Set always to 0x20 */ \
-/* I3 */ 0x20, /* Note! Ignored. Set always to 0x20 */ \
-/* I4 */ 0xf5, /* Ignored */ \
-/* I5 */ 0x10, \
-/* I6 */ 0x00, \
-/* I7 */ 0x2e, /* I7 MEM config A. Likely to vary between models */ \
-/* I8 */ 0x00, /* I8 MEM config B. Likely to vary between models */ \
-/* I9 */ 0x40 /* Ignored */ \
- }
-#endif
-
- unsigned long flags;
- static unsigned char regs[10] = SSCAPE_REGS;
-
- int i, irq_bits = 0xff;
-
- if (old_hardware)
- {
- valid_interrupts = valid_interrupts_old;
- conf_printf("Ensoniq SoundScape (old)", hw_config);
- }
- else
- conf_printf("Ensoniq SoundScape", hw_config);
-
- for (i = 0; i < 4; i++)
- {
- if (hw_config->irq == valid_interrupts[i])
- {
- irq_bits = i;
- break;
- }
- }
- if (hw_config->irq > 15 || (regs[4] = irq_bits == 0xff))
- {
- printk(KERN_ERR "Invalid IRQ%d\n", hw_config->irq);
- release_region(devc->base, 2);
- release_region(devc->base + 2, 6);
- if (sscape_is_pnp)
- release_region(devc->codec, 2);
- return;
- }
-
- if (!sscape_is_pnp) {
-
- spin_lock_irqsave(&devc->lock,flags);
- /* Host interrupt enable */
- sscape_write(devc, 1, 0xf0); /* All interrupts enabled */
- /* DMA A status/trigger register */
- sscape_write(devc, 2, 0x20); /* DMA channel disabled */
- /* DMA B status/trigger register */
- sscape_write(devc, 3, 0x20); /* DMA channel disabled */
- /* Host interrupt config reg */
- sscape_write(devc, 4, 0xf0 | (irq_bits << 2) | irq_bits);
- /* Don't destroy CD-ROM DMA config bits (0xc0) */
- sscape_write(devc, 5, (regs[5] & 0x3f) | (sscape_read(devc, 5) & 0xc0));
- /* CD-ROM config (WSS codec actually) */
- sscape_write(devc, 6, regs[6]);
- sscape_write(devc, 7, regs[7]);
- sscape_write(devc, 8, regs[8]);
- /* Master control reg. Don't modify CR-ROM bits. Disable SB emul */
- sscape_write(devc, 9, (sscape_read(devc, 9) & 0xf0) | 0x08);
- spin_unlock_irqrestore(&devc->lock,flags);
- }
-#ifdef SSCAPE_DEBUG2
- /*
- * Temporary debugging aid. Print contents of the registers after
- * changing them.
- */
- {
- int i;
-
- for (i = 0; i < 13; i++)
- printk("I%d = %02x (new value)\n", i, sscape_read(devc, i));
- }
-#endif
-
- if (probe_mpu401(hw_config, sscape_ports))
- hw_config->always_detect = 1;
- hw_config->name = "SoundScape";
-
- hw_config->irq *= -1; /* Negative value signals IRQ sharing */
- attach_mpu401(hw_config, THIS_MODULE);
- hw_config->irq *= -1; /* Restore it */
-
- if (hw_config->slots[1] != -1) /* The MPU driver installed itself */
- {
- sscape_mididev = hw_config->slots[1];
- midi_devs[hw_config->slots[1]]->coproc = &sscape_coproc_operations;
- }
- sscape_write(devc, GA_INTENA_REG, 0x80); /* Master IRQ enable */
- devc->ok = 1;
- devc->failed = 0;
-}
-
-static int detect_ga(sscape_info * devc)
-{
- unsigned char save;
-
- DDB(printk("Entered Soundscape detect_ga(%x)\n", devc->base));
-
- /*
- * First check that the address register of "ODIE" is
- * there and that it has exactly 4 writable bits.
- * First 4 bits
- */
-
- if ((save = inb(PORT(ODIE_ADDR))) & 0xf0)
- {
- DDB(printk("soundscape: Detect error A\n"));
- return 0;
- }
- outb((0x00), PORT(ODIE_ADDR));
- if (inb(PORT(ODIE_ADDR)) != 0x00)
- {
- DDB(printk("soundscape: Detect error B\n"));
- return 0;
- }
- outb((0xff), PORT(ODIE_ADDR));
- if (inb(PORT(ODIE_ADDR)) != 0x0f)
- {
- DDB(printk("soundscape: Detect error C\n"));
- return 0;
- }
- outb((save), PORT(ODIE_ADDR));
-
- /*
- * Now verify that some indirect registers return zero on some bits.
- * This may break the driver with some future revisions of "ODIE" but...
- */
-
- if (sscape_read(devc, 0) & 0x0c)
- {
- DDB(printk("soundscape: Detect error D (%x)\n", sscape_read(devc, 0)));
- return 0;
- }
- if (sscape_read(devc, 1) & 0x0f)
- {
- DDB(printk("soundscape: Detect error E\n"));
- return 0;
- }
- if (sscape_read(devc, 5) & 0x0f)
- {
- DDB(printk("soundscape: Detect error F\n"));
- return 0;
- }
- return 1;
-}
-
-static int sscape_read_host_ctrl(sscape_info* devc)
-{
- return host_read(devc);
-}
-
-static void sscape_write_host_ctrl2(sscape_info *devc, int a, int b)
-{
- host_command2(devc, a, b);
-}
-
-static int sscape_alloc_dma(sscape_info *devc)
-{
- char *start_addr, *end_addr;
- int dma_pagesize;
- int sz, size;
- struct page *page;
-
- if (devc->raw_buf != NULL) return 0; /* Already done */
- dma_pagesize = (devc->dma < 4) ? (64 * 1024) : (128 * 1024);
- devc->raw_buf = NULL;
- devc->buffsize = 8192*4;
- if (devc->buffsize > dma_pagesize) devc->buffsize = dma_pagesize;
- start_addr = NULL;
- /*
- * Now loop until we get a free buffer. Try to get smaller buffer if
- * it fails. Don't accept smaller than 8k buffer for performance
- * reasons.
- */
- while (start_addr == NULL && devc->buffsize > PAGE_SIZE) {
- for (sz = 0, size = PAGE_SIZE; size < devc->buffsize; sz++, size <<= 1);
- devc->buffsize = PAGE_SIZE * (1 << sz);
- start_addr = (char *) __get_free_pages(GFP_ATOMIC|GFP_DMA, sz);
- if (start_addr == NULL) devc->buffsize /= 2;
- }
-
- if (start_addr == NULL) {
- printk(KERN_ERR "sscape pnp init error: Couldn't allocate DMA buffer\n");
- return 0;
- } else {
- /* make some checks */
- end_addr = start_addr + devc->buffsize - 1;
- /* now check if it fits into the same dma-pagesize */
-
- if (((long) start_addr & ~(dma_pagesize - 1)) != ((long) end_addr & ~(dma_pagesize - 1))
- || end_addr >= (char *) (MAX_DMA_ADDRESS)) {
- printk(KERN_ERR "sscape pnp: Got invalid address 0x%lx for %db DMA-buffer\n", (long) start_addr, devc->buffsize);
- return 0;
- }
- }
- devc->raw_buf = start_addr;
- devc->raw_buf_phys = virt_to_bus(start_addr);
-
- for (page = virt_to_page(start_addr); page <= virt_to_page(end_addr); page++)
- SetPageReserved(page);
- return 1;
-}
-
-static void sscape_free_dma(sscape_info *devc)
-{
- int sz, size;
- unsigned long start_addr, end_addr;
- struct page *page;
-
- if (devc->raw_buf == NULL) return;
- for (sz = 0, size = PAGE_SIZE; size < devc->buffsize; sz++, size <<= 1);
- start_addr = (unsigned long) devc->raw_buf;
- end_addr = start_addr + devc->buffsize;
-
- for (page = virt_to_page(start_addr); page <= virt_to_page(end_addr); page++)
- ClearPageReserved(page);
-
- free_pages((unsigned long) devc->raw_buf, sz);
- devc->raw_buf = NULL;
-}
-
-/* Intel version !!!!!!!!! */
-
-static int sscape_start_dma(int chan, unsigned long physaddr, int count, int dma_mode)
-{
- unsigned long flags;
-
- flags = claim_dma_lock();
- disable_dma(chan);
- clear_dma_ff(chan);
- set_dma_mode(chan, dma_mode);
- set_dma_addr(chan, physaddr);
- set_dma_count(chan, count);
- enable_dma(chan);
- release_dma_lock(flags);
- return 0;
-}
-
-static void sscape_pnp_start_dma(sscape_info* devc, int arg )
-{
- int reg;
- if (arg == 0) reg = 2;
- else reg = 3;
-
- sscape_write(devc, reg, sscape_read( devc, reg) | 0x01);
- sscape_write(devc, reg, sscape_read( devc, reg) & 0xFE);
-}
-
-static int sscape_pnp_wait_dma (sscape_info* devc, int arg )
-{
- int reg;
- unsigned long i;
- unsigned char d;
-
- if (arg == 0) reg = 2;
- else reg = 3;
-
- sleep ( 1 );
- i = 0;
- do {
- d = sscape_read(devc, reg) & 1;
- if ( d == 1) break;
- i++;
- } while (i < 500000);
- d = sscape_read(devc, reg) & 1;
- return d;
-}
-
-static int sscape_pnp_alloc_dma(sscape_info* devc)
-{
- /* printk(KERN_INFO "sscape: requesting dma\n"); */
- if (request_dma(devc -> dma, "sscape")) return 0;
- /* printk(KERN_INFO "sscape: dma channel allocated\n"); */
- if (!sscape_alloc_dma(devc)) {
- free_dma(devc -> dma);
- return 0;
- };
- return 1;
-}
-
-static void sscape_pnp_free_dma(sscape_info* devc)
-{
- sscape_free_dma( devc);
- free_dma(devc -> dma );
- /* printk(KERN_INFO "sscape: dma released\n"); */
-}
-
-static int sscape_pnp_upload_file(sscape_info* devc, char* fn)
-{
- int done = 0;
- int timeout_val;
- char* data,*dt;
- int len,l;
- unsigned long flags;
-
- sscape_write( devc, 9, sscape_read(devc, 9 ) & 0x3F );
- sscape_write( devc, 2, (devc -> dma << 4) | 0x80 );
- sscape_write( devc, 3, 0x20 );
- sscape_write( devc, 9, sscape_read( devc, 9 ) | 0x80 );
-
- len = mod_firmware_load(fn, &data);
- if (len == 0) {
- printk(KERN_ERR "sscape: file not found: %s\n", fn);
- return 0;
- }
- dt = data;
- spin_lock_irqsave(&devc->lock,flags);
- while ( len > 0 ) {
- if (len > devc -> buffsize) l = devc->buffsize;
- else l = len;
- len -= l;
- memcpy(devc->raw_buf, dt, l); dt += l;
- sscape_start_dma(devc->dma, devc->raw_buf_phys, l, 0x48);
- sscape_pnp_start_dma ( devc, 0 );
- if (sscape_pnp_wait_dma ( devc, 0 ) == 0) {
- spin_unlock_irqrestore(&devc->lock,flags);
- return 0;
- }
- }
-
- spin_unlock_irqrestore(&devc->lock,flags);
- vfree(data);
-
- outb(0, devc -> base + 2);
- outb(0, devc -> base);
-
- sscape_write ( devc, 9, sscape_read( devc, 9 ) | 0x40);
-
- timeout_val = 5 * HZ;
- while (!done && timeout_val-- > 0)
- {
- unsigned char x;
- sleep(1);
- x = inb( devc -> base + 3);
- if (x == 0xff || x == 0xfe) /* OBP startup acknowledge */
- {
- //printk(KERN_ERR "Soundscape: Acknowledge = %x\n", x);
- done = 1;
- }
- }
- timeout_val = 5 * HZ;
- done = 0;
- while (!done && timeout_val-- > 0)
- {
- unsigned char x;
- sleep(1);
- x = inb( devc -> base + 3);
- if (x == 0xfe) /* OBP startup acknowledge */
- {
- //printk(KERN_ERR "Soundscape: Acknowledge = %x\n", x);
- done = 1;
- }
- }
-
- if ( !done ) printk(KERN_ERR "soundscape: OBP Initialization failed.\n");
-
- sscape_write( devc, 2, devc->ic_type == IC_ODIE ? 0x70 : 0x40);
- sscape_write( devc, 3, (devc -> dma << 4) + 0x80);
- return 1;
-}
-
-static void __init sscape_pnp_init_hw(sscape_info* devc)
-{
- unsigned char midi_irq = 0, sb_irq = 0;
- unsigned i;
- static char code_file_name[23] = "/sndscape/sndscape.cox";
-
- int sscape_joystic_enable = 0x7f;
- int sscape_mic_enable = 0;
- int sscape_ext_midi = 0;
-
- if ( !sscape_pnp_alloc_dma(devc) ) {
- printk(KERN_ERR "sscape: faild to allocate dma\n");
- return;
- }
-
- for (i = 0; i < 4; i++) {
- if ( devc -> irq == valid_interrupts[i] )
- midi_irq = i;
- if ( devc -> codec_irq == valid_interrupts[i] )
- sb_irq = i;
- }
-
- sscape_write( devc, 5, 0x50);
- sscape_write( devc, 7, 0x2e);
- sscape_write( devc, 8, 0x00);
-
- sscape_write( devc, 2, devc->ic_type == IC_ODIE ? 0x70 : 0x40);
- sscape_write( devc, 3, ( devc -> dma << 4) | 0x80);
-
- sscape_write (devc, 4, 0xF0 | (midi_irq<<2) | midi_irq);
-
- i = 0x10; //sscape_read(devc, 9) & (devc->ic_type == IC_ODIE ? 0xf0 : 0xc0);
- if (sscape_joystic_enable) i |= 8;
-
- sscape_write (devc, 9, i);
- sscape_write (devc, 6, 0x80);
- sscape_write (devc, 1, 0x80);
-
- if (devc -> codec_type == 2) {
- sscape_pnp_write_codec( devc, 0x0C, 0x50);
- sscape_pnp_write_codec( devc, 0x10, sscape_pnp_read_codec( devc, 0x10) & 0x3F);
- sscape_pnp_write_codec( devc, 0x11, sscape_pnp_read_codec( devc, 0x11) | 0xC0);
- sscape_pnp_write_codec( devc, 29, 0x20);
- }
-
- if (sscape_pnp_upload_file(devc, "/sndscape/scope.cod") == 0 ) {
- printk(KERN_ERR "sscape: faild to upload file /sndscape/scope.cod\n");
- sscape_pnp_free_dma(devc);
- return;
- }
-
- i = sscape_read_host_ctrl( devc );
-
- if ( (i & 0x0F) > 7 ) {
- printk(KERN_ERR "sscape: scope.cod faild\n");
- sscape_pnp_free_dma(devc);
- return;
- }
- if ( i & 0x10 ) sscape_write( devc, 7, 0x2F);
- code_file_name[21] = (char) ( i & 0x0F) + 0x30;
- if (sscape_pnp_upload_file( devc, code_file_name) == 0) {
- printk(KERN_ERR "sscape: faild to upload file %s\n", code_file_name);
- sscape_pnp_free_dma(devc);
- return;
- }
-
- if (devc->ic_type != IC_ODIE) {
- sscape_pnp_write_codec( devc, 10, (sscape_pnp_read_codec(devc, 10) & 0x7f) |
- ( sscape_mic_enable == 0 ? 0x00 : 0x80) );
- }
- sscape_write_host_ctrl2( devc, 0x84, 0x64 ); /* MIDI volume */
- sscape_write_host_ctrl2( devc, 0x86, 0x64 ); /* MIDI volume?? */
- sscape_write_host_ctrl2( devc, 0x8A, sscape_ext_midi);
-
- sscape_pnp_write_codec ( devc, 6, 0x3f ); //WAV_VOL
- sscape_pnp_write_codec ( devc, 7, 0x3f ); //WAV_VOL
- sscape_pnp_write_codec ( devc, 2, 0x1F ); //WD_CDXVOLL
- sscape_pnp_write_codec ( devc, 3, 0x1F ); //WD_CDXVOLR
-
- if (devc -> codec_type == 1) {
- sscape_pnp_write_codec ( devc, 4, 0x1F );
- sscape_pnp_write_codec ( devc, 5, 0x1F );
- sscape_write_host_ctrl2( devc, 0x88, sscape_mic_enable);
- } else {
- int t;
- sscape_pnp_write_codec ( devc, 0x10, 0x1F << 1);
- sscape_pnp_write_codec ( devc, 0x11, 0xC0 | (0x1F << 1));
-
- t = sscape_pnp_read_codec( devc, 0x00) & 0xDF;
- if ( (sscape_mic_enable == 0)) t |= 0;
- else t |= 0x20;
- sscape_pnp_write_codec ( devc, 0x00, t);
- t = sscape_pnp_read_codec( devc, 0x01) & 0xDF;
- if ( (sscape_mic_enable == 0) ) t |= 0;
- else t |= 0x20;
- sscape_pnp_write_codec ( devc, 0x01, t);
- sscape_pnp_write_codec ( devc, 0x40 | 29 , 0x20);
- outb(0, devc -> codec);
- }
- if (devc -> ic_type == IC_OPUS ) {
- int i = sscape_read( devc, 9 );
- sscape_write( devc, 9, i | 3 );
- sscape_write( devc, 3, 0x40);
-
- if (request_region(0x228, 1, "sscape setup junk")) {
- outb(0, 0x228);
- release_region(0x228,1);
- }
- sscape_write( devc, 3, (devc -> dma << 4) | 0x80);
- sscape_write( devc, 9, i );
- }
-
- host_close ( devc );
- sscape_pnp_free_dma(devc);
-}
-
-static int __init detect_sscape_pnp(sscape_info* devc)
-{
- long i, irq_bits = 0xff;
- unsigned int d;
-
- DDB(printk("Entered detect_sscape_pnp(%x)\n", devc->base));
-
- if (!request_region(devc->codec, 2, "sscape codec")) {
- printk(KERN_ERR "detect_sscape_pnp: port %x is not free\n", devc->codec);
- return 0;
- }
-
- if ((inb(devc->base + 2) & 0x78) != 0)
- goto fail;
-
- d = inb ( devc -> base + 4) & 0xF0;
- if (d & 0x80)
- goto fail;
-
- if (d == 0) {
- devc->codec_type = 1;
- devc->ic_type = IC_ODIE;
- } else if ( (d & 0x60) != 0) {
- devc->codec_type = 2;
- devc->ic_type = IC_OPUS;
- } else if ( (d & 0x40) != 0) { /* WTF? */
- devc->codec_type = 2;
- devc->ic_type = IC_ODIE;
- } else
- goto fail;
-
- sscape_is_pnp = 1;
-
- outb(0xFA, devc -> base+4);
- if ((inb( devc -> base+4) & 0x9F) != 0x0A)
- goto fail;
- outb(0xFE, devc -> base+4);
- if ( (inb(devc -> base+4) & 0x9F) != 0x0E)
- goto fail;
- if ( (inb(devc -> base+5) & 0x9F) != 0x0E)
- goto fail;
-
- if (devc->codec_type == 2) {
- if (devc->codec != devc->base + 8) {
- printk("soundscape warning: incorrect codec port specified\n");
- goto fail;
- }
- d = 0x10 | (sscape_read(devc, 9) & 0xCF);
- sscape_write(devc, 9, d);
- sscape_write(devc, 6, 0x80);
- } else {
- //todo: check codec is not base + 8
- }
-
- d = (sscape_read(devc, 9) & 0x3F) | 0xC0;
- sscape_write(devc, 9, d);
-
- for (i = 0; i < 550000; i++)
- if ( !(inb(devc -> codec) & 0x80) ) break;
-
- d = inb(devc -> codec);
- if (d & 0x80)
- goto fail;
- if ( inb(devc -> codec + 2) == 0xFF)
- goto fail;
-
- sscape_write(devc, 9, sscape_read(devc, 9) & 0x3F );
-
- d = inb(devc -> codec) & 0x80;
- if ( d == 0) {
- printk(KERN_INFO "soundscape: hardware detected\n");
- valid_interrupts = valid_interrupts_new;
- } else {
- printk(KERN_INFO "soundscape: board looks like media fx\n");
- valid_interrupts = valid_interrupts_old;
- old_hardware = 1;
- }
-
- sscape_write( devc, 9, 0xC0 | (sscape_read(devc, 9) & 0x3F) );
-
- for (i = 0; i < 550000; i++)
- if ( !(inb(devc -> codec) & 0x80))
- break;
-
- sscape_pnp_init_hw(devc);
-
- for (i = 0; i < 4; i++)
- {
- if (devc->codec_irq == valid_interrupts[i]) {
- irq_bits = i;
- break;
- }
- }
- sscape_write(devc, GA_INTENA_REG, 0x00);
- sscape_write(devc, GA_DMACFG_REG, 0x50);
- sscape_write(devc, GA_DMAA_REG, 0x70);
- sscape_write(devc, GA_DMAB_REG, 0x20);
- sscape_write(devc, GA_INTCFG_REG, 0xf0);
- sscape_write(devc, GA_CDCFG_REG, 0x89 | (devc->dma << 4) | (irq_bits << 1));
-
- sscape_pnp_write_codec( devc, 0, sscape_pnp_read_codec( devc, 0) | 0x20);
- sscape_pnp_write_codec( devc, 0, sscape_pnp_read_codec( devc, 1) | 0x20);
-
- return 1;
-fail:
- release_region(devc->codec, 2);
- return 0;
-}
-
-static int __init probe_sscape(struct address_info *hw_config)
-{
- devc->base = hw_config->io_base;
- devc->irq = hw_config->irq;
- devc->dma = hw_config->dma;
- devc->osp = hw_config->osp;
-
-#ifdef SSCAPE_DEBUG1
- /*
- * Temporary debugging aid. Print contents of the registers before
- * changing them.
- */
- {
- int i;
-
- for (i = 0; i < 13; i++)
- printk("I%d = %02x (old value)\n", i, sscape_read(devc, i));
- }
-#endif
- devc->failed = 1;
-
- sscape_ports = request_region(devc->base, 2, "mpu401");
- if (!sscape_ports)
- return 0;
-
- if (!request_region(devc->base + 2, 6, "SoundScape")) {
- release_region(devc->base, 2);
- return 0;
- }
-
- if (!detect_ga(devc)) {
- if (detect_sscape_pnp(devc))
- return 1;
- release_region(devc->base, 2);
- release_region(devc->base + 2, 6);
- return 0;
- }
-
- if (old_hardware) /* Check that it's really an old Spea/Reveal card. */
- {
- unsigned char tmp;
- int cc;
-
- if (!((tmp = sscape_read(devc, GA_HMCTL_REG)) & 0xc0))
- {
- sscape_write(devc, GA_HMCTL_REG, tmp | 0x80);
- for (cc = 0; cc < 200000; ++cc)
- inb(devc->base + ODIE_ADDR);
- }
- }
- return 1;
-}
-
-static int __init init_ss_ms_sound(struct address_info *hw_config)
-{
- int i, irq_bits = 0xff;
- int ad_flags = 0;
- struct resource *ports;
-
- if (devc->failed)
- {
- printk(KERN_ERR "soundscape: Card not detected\n");
- return 0;
- }
- if (devc->ok == 0)
- {
- printk(KERN_ERR "soundscape: Invalid initialization order.\n");
- return 0;
- }
- for (i = 0; i < 4; i++)
- {
- if (hw_config->irq == valid_interrupts[i])
- {
- irq_bits = i;
- break;
- }
- }
- if (irq_bits == 0xff) {
- printk(KERN_ERR "soundscape: Invalid MSS IRQ%d\n", hw_config->irq);
- return 0;
- }
-
- if (old_hardware)
- ad_flags = 0x12345677; /* Tell that we may have a CS4248 chip (Spea-V7 Media FX) */
- else if (sscape_is_pnp)
- ad_flags = 0x87654321; /* Tell that we have a soundscape pnp with 1845 chip */
-
- ports = request_region(hw_config->io_base, 4, "ad1848");
- if (!ports) {
- printk(KERN_ERR "soundscape: ports busy\n");
- return 0;
- }
-
- if (!ad1848_detect(ports, &ad_flags, hw_config->osp)) {
- release_region(hw_config->io_base, 4);
- return 0;
- }
-
- if (!sscape_is_pnp) /*pnp is already setup*/
- {
- /*
- * Setup the DMA polarity.
- */
- sscape_write(devc, GA_DMACFG_REG, 0x50);
-
- /*
- * Take the gate-array off of the DMA channel.
- */
- sscape_write(devc, GA_DMAB_REG, 0x20);
-
- /*
- * Init the AD1848 (CD-ROM) config reg.
- */
- sscape_write(devc, GA_CDCFG_REG, 0x89 | (hw_config->dma << 4) | (irq_bits << 1));
- }
-
- if (hw_config->irq == devc->irq)
- printk(KERN_WARNING "soundscape: Warning! The WSS mode can't share IRQ with MIDI\n");
-
- hw_config->slots[0] = ad1848_init(
- sscape_is_pnp ? "SoundScape" : "SoundScape PNP",
- ports,
- hw_config->irq,
- hw_config->dma,
- hw_config->dma,
- 0,
- devc->osp,
- THIS_MODULE);
-
-
- if (hw_config->slots[0] != -1) /* The AD1848 driver installed itself */
- {
- audio_devs[hw_config->slots[0]]->coproc = &sscape_coproc_operations;
- devc->codec_audiodev = hw_config->slots[0];
- devc->my_audiodev = hw_config->slots[0];
-
- /* Set proper routings here (what are they) */
- AD1848_REROUTE(SOUND_MIXER_LINE1, SOUND_MIXER_LINE);
- }
-
-#ifdef SSCAPE_DEBUG5
- /*
- * Temporary debugging aid. Print contents of the registers
- * after the AD1848 device has been initialized.
- */
- {
- int i;
-
- for (i = 0; i < 13; i++)
- printk("I%d = %02x\n", i, sscape_read(devc, i));
- }
-#endif
- return 1;
-}
-
-static void __exit unload_sscape(struct address_info *hw_config)
-{
- release_region(devc->base + 2, 6);
- unload_mpu401(hw_config);
- if (sscape_is_pnp)
- release_region(devc->codec, 2);
-}
-
-static void __exit unload_ss_ms_sound(struct address_info *hw_config)
-{
- ad1848_unload(hw_config->io_base,
- hw_config->irq,
- devc->dma,
- devc->dma,
- 0);
- sound_unload_audiodev(hw_config->slots[0]);
-}
-
-static struct address_info cfg;
-static struct address_info cfg_mpu;
-
-static int __initdata spea = -1;
-static int mss = 0;
-static int __initdata dma = -1;
-static int __initdata irq = -1;
-static int __initdata io = -1;
-static int __initdata mpu_irq = -1;
-static int __initdata mpu_io = -1;
-
-module_param(dma, int, 0);
-module_param(irq, int, 0);
-module_param(io, int, 0);
-module_param(spea, int, 0); /* spea=0/1 set the old_hardware */
-module_param(mpu_irq, int, 0);
-module_param(mpu_io, int, 0);
-module_param(mss, int, 0);
-
-static int __init init_sscape(void)
-{
- printk(KERN_INFO "Soundscape driver Copyright (C) by Hannu Savolainen 1993-1996\n");
-
- cfg.irq = irq;
- cfg.dma = dma;
- cfg.io_base = io;
-
- cfg_mpu.irq = mpu_irq;
- cfg_mpu.io_base = mpu_io;
- /* WEH - Try to get right dma channel */
- cfg_mpu.dma = dma;
-
- devc->codec = cfg.io_base;
- devc->codec_irq = cfg.irq;
- devc->codec_type = 0;
- devc->ic_type = 0;
- devc->raw_buf = NULL;
- spin_lock_init(&devc->lock);
-
- if (cfg.dma == -1 || cfg.irq == -1 || cfg.io_base == -1) {
- printk(KERN_ERR "DMA, IRQ, and IO port must be specified.\n");
- return -EINVAL;
- }
-
- if (cfg_mpu.irq == -1 && cfg_mpu.io_base != -1) {
- printk(KERN_ERR "MPU_IRQ must be specified if MPU_IO is set.\n");
- return -EINVAL;
- }
-
- if(spea != -1) {
- old_hardware = spea;
- printk(KERN_INFO "Forcing %s hardware support.\n",
- spea?"new":"old");
- }
- if (probe_sscape(&cfg_mpu) == 0)
- return -ENODEV;
-
- attach_sscape(&cfg_mpu);
-
- mss = init_ss_ms_sound(&cfg);
-
- return 0;
-}
-
-static void __exit cleanup_sscape(void)
-{
- if (mss)
- unload_ss_ms_sound(&cfg);
- unload_sscape(&cfg_mpu);
-}
-
-module_init(init_sscape);
-module_exit(cleanup_sscape);
-
-#ifndef MODULE
-static int __init setup_sscape(char *str)
-{
- /* io, irq, dma, mpu_io, mpu_irq */
- int ints[6];
-
- str = get_options(str, ARRAY_SIZE(ints), ints);
-
- io = ints[1];
- irq = ints[2];
- dma = ints[3];
- mpu_io = ints[4];
- mpu_irq = ints[5];
-
- return 1;
-}
-
-__setup("sscape=", setup_sscape);
-#endif
-MODULE_LICENSE("GPL");
diff --git a/sound/parisc/harmony.c b/sound/parisc/harmony.c
index e924492df21..f47f9e226b0 100644
--- a/sound/parisc/harmony.c
+++ b/sound/parisc/harmony.c
@@ -624,6 +624,9 @@ snd_harmony_pcm_init(struct snd_harmony *h)
struct snd_pcm *pcm;
int err;
+ if (snd_BUG_ON(!h))
+ return -EINVAL;
+
harmony_disable_interrupts(h);
err = snd_pcm_new(h->card, "harmony", 0, 1, 1, &pcm);
@@ -865,11 +868,12 @@ snd_harmony_mixer_reset(struct snd_harmony *h)
static int __devinit
snd_harmony_mixer_init(struct snd_harmony *h)
{
- struct snd_card *card = h->card;
+ struct snd_card *card;
int idx, err;
if (snd_BUG_ON(!h))
return -EINVAL;
+ card = h->card;
strcpy(card->mixername, "Harmony Gain control interface");
for (idx = 0; idx < HARMONY_CONTROLS; idx++) {
diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig
index fb5ee3cc396..75c602b5b13 100644
--- a/sound/pci/Kconfig
+++ b/sound/pci/Kconfig
@@ -259,7 +259,6 @@ config SND_CS5530
config SND_CS5535AUDIO
tristate "CS5535/CS5536 Audio"
- depends on X86 && !X86_64
select SND_PCM
select SND_AC97_CODEC
help
diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c
index 78288dbfc17..20cb60afb20 100644
--- a/sound/pci/ac97/ac97_codec.c
+++ b/sound/pci/ac97/ac97_codec.c
@@ -603,8 +603,8 @@ AC97_SINGLE("Tone Control - Treble", AC97_MASTER_TONE, 0, 15, 1)
};
static const struct snd_kcontrol_new snd_ac97_controls_pc_beep[2] = {
-AC97_SINGLE("PC Speaker Playback Switch", AC97_PC_BEEP, 15, 1, 1),
-AC97_SINGLE("PC Speaker Playback Volume", AC97_PC_BEEP, 1, 15, 1)
+AC97_SINGLE("Beep Playback Switch", AC97_PC_BEEP, 15, 1, 1),
+AC97_SINGLE("Beep Playback Volume", AC97_PC_BEEP, 1, 15, 1)
};
static const struct snd_kcontrol_new snd_ac97_controls_mic_boost =
@@ -1393,7 +1393,7 @@ static int snd_ac97_mixer_build(struct snd_ac97 * ac97)
}
}
- /* build PC Speaker controls */
+ /* build Beep controls */
if (!(ac97->flags & AC97_HAS_NO_PC_BEEP) &&
((ac97->flags & AC97_HAS_PC_BEEP) ||
snd_ac97_try_volume_mix(ac97, AC97_PC_BEEP))) {
diff --git a/sound/pci/ac97/ac97_patch.c b/sound/pci/ac97/ac97_patch.c
index 7337abdbe4e..139cf3b2b9d 100644
--- a/sound/pci/ac97/ac97_patch.c
+++ b/sound/pci/ac97/ac97_patch.c
@@ -800,12 +800,12 @@ AC97_SINGLE("Mono Switch", AC97_MASTER_TONE, 7, 1, 1),
AC97_SINGLE("Mono ZC Switch", AC97_MASTER_TONE, 6, 1, 0),
AC97_SINGLE("Mono Volume", AC97_MASTER_TONE, 0, 31, 1),
-AC97_SINGLE("PC Beep to Headphone Switch", AC97_AUX, 15, 1, 1),
-AC97_SINGLE("PC Beep to Headphone Volume", AC97_AUX, 12, 7, 1),
-AC97_SINGLE("PC Beep to Master Switch", AC97_AUX, 11, 1, 1),
-AC97_SINGLE("PC Beep to Master Volume", AC97_AUX, 8, 7, 1),
-AC97_SINGLE("PC Beep to Mono Switch", AC97_AUX, 7, 1, 1),
-AC97_SINGLE("PC Beep to Mono Volume", AC97_AUX, 4, 7, 1),
+AC97_SINGLE("Beep to Headphone Switch", AC97_AUX, 15, 1, 1),
+AC97_SINGLE("Beep to Headphone Volume", AC97_AUX, 12, 7, 1),
+AC97_SINGLE("Beep to Master Switch", AC97_AUX, 11, 1, 1),
+AC97_SINGLE("Beep to Master Volume", AC97_AUX, 8, 7, 1),
+AC97_SINGLE("Beep to Mono Switch", AC97_AUX, 7, 1, 1),
+AC97_SINGLE("Beep to Mono Volume", AC97_AUX, 4, 7, 1),
AC97_SINGLE("Voice to Headphone Switch", AC97_PCM, 15, 1, 1),
AC97_SINGLE("Voice to Headphone Volume", AC97_PCM, 12, 7, 1),
diff --git a/sound/pci/ali5451/ali5451.c b/sound/pci/ali5451/ali5451.c
index b458d208720..aaf4da68969 100644
--- a/sound/pci/ali5451/ali5451.c
+++ b/sound/pci/ali5451/ali5451.c
@@ -973,7 +973,7 @@ static void snd_ali_free_voice(struct snd_ali * codec,
void *private_data;
snd_ali_printk("free_voice: channel=%d\n",pvoice->number);
- if (pvoice == NULL || !pvoice->use)
+ if (!pvoice->use)
return;
snd_ali_clear_voices(codec, pvoice->number, pvoice->number);
spin_lock_irq(&codec->voice_alloc);
diff --git a/sound/pci/azt3328.c b/sound/pci/azt3328.c
index 8451a0169f3..69867ace786 100644
--- a/sound/pci/azt3328.c
+++ b/sound/pci/azt3328.c
@@ -830,8 +830,8 @@ static struct snd_kcontrol_new snd_azf3328_mixer_controls[] __devinitdata = {
AZF3328_MIXER_SWITCH("Mic Boost (+20dB)", IDX_MIXER_MIC, 6, 0),
AZF3328_MIXER_SWITCH("Line Playback Switch", IDX_MIXER_LINEIN, 15, 1),
AZF3328_MIXER_VOL_STEREO("Line Playback Volume", IDX_MIXER_LINEIN, 0x1f, 1),
- AZF3328_MIXER_SWITCH("PC Speaker Playback Switch", IDX_MIXER_PCBEEP, 15, 1),
- AZF3328_MIXER_VOL_SPECIAL("PC Speaker Playback Volume", IDX_MIXER_PCBEEP, 0x0f, 1, 1),
+ AZF3328_MIXER_SWITCH("Beep Playback Switch", IDX_MIXER_PCBEEP, 15, 1),
+ AZF3328_MIXER_VOL_SPECIAL("Beep Playback Volume", IDX_MIXER_PCBEEP, 0x0f, 1, 1),
AZF3328_MIXER_SWITCH("Video Playback Switch", IDX_MIXER_VIDEO, 15, 1),
AZF3328_MIXER_VOL_STEREO("Video Playback Volume", IDX_MIXER_VIDEO, 0x1f, 1),
AZF3328_MIXER_SWITCH("Aux Playback Switch", IDX_MIXER_AUX, 15, 1),
diff --git a/sound/pci/bt87x.c b/sound/pci/bt87x.c
index 24585c6c6d0..4e2b925a94c 100644
--- a/sound/pci/bt87x.c
+++ b/sound/pci/bt87x.c
@@ -808,6 +808,8 @@ static struct pci_device_id snd_bt87x_ids[] = {
BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, 0x1002, 0x0001, GENERIC),
/* Leadtek Winfast tv 2000xp delux */
BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, 0x107d, 0x6606, GENERIC),
+ /* Pinnacle PCTV */
+ BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, 0x11bd, 0x0012, GENERIC),
/* Voodoo TV 200 */
BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, 0x121a, 0x3000, GENERIC),
/* Askey Computer Corp. MagicTView'99 */
diff --git a/sound/pci/ca0106/ca0106_mixer.c b/sound/pci/ca0106/ca0106_mixer.c
index c8c6f437f5b..8f443a9d61e 100644
--- a/sound/pci/ca0106/ca0106_mixer.c
+++ b/sound/pci/ca0106/ca0106_mixer.c
@@ -792,8 +792,8 @@ int __devinit snd_ca0106_mixer(struct snd_ca0106 *emu)
"Phone Playback Volume",
"Video Playback Switch",
"Video Playback Volume",
- "PC Speaker Playback Switch",
- "PC Speaker Playback Volume",
+ "Beep Playback Switch",
+ "Beep Playback Volume",
"Mono Output Select",
"Capture Source",
"Capture Switch",
diff --git a/sound/pci/ca0106/ca0106_proc.c b/sound/pci/ca0106/ca0106_proc.c
index c62b7d10ec6..15523e60351 100644
--- a/sound/pci/ca0106/ca0106_proc.c
+++ b/sound/pci/ca0106/ca0106_proc.c
@@ -304,7 +304,7 @@ static void snd_ca0106_proc_reg_write32(struct snd_info_entry *entry,
while (!snd_info_get_line(buffer, line, sizeof(line))) {
if (sscanf(line, "%x %x", &reg, &val) != 2)
continue;
- if ((reg < 0x40) && (reg >=0) && (val <= 0xffffffff) ) {
+ if (reg < 0x40 && val <= 0xffffffff) {
spin_lock_irqsave(&emu->emu_lock, flags);
outl(val, emu->port + (reg & 0xfffffffc));
spin_unlock_irqrestore(&emu->emu_lock, flags);
@@ -405,7 +405,7 @@ static void snd_ca0106_proc_reg_write(struct snd_info_entry *entry,
while (!snd_info_get_line(buffer, line, sizeof(line))) {
if (sscanf(line, "%x %x %x", &reg, &channel_id, &val) != 3)
continue;
- if ((reg < 0x80) && (reg >=0) && (val <= 0xffffffff) && (channel_id >=0) && (channel_id <= 3) )
+ if (reg < 0x80 && val <= 0xffffffff && channel_id <= 3)
snd_ca0106_ptr_write(emu, reg, channel_id, val);
}
}
diff --git a/sound/pci/cmipci.c b/sound/pci/cmipci.c
index ddcd4a9fd7e..a312bae08f5 100644
--- a/sound/pci/cmipci.c
+++ b/sound/pci/cmipci.c
@@ -2302,7 +2302,7 @@ static struct snd_kcontrol_new snd_cmipci_mixers[] __devinitdata = {
CMIPCI_SB_VOL_MONO("Mic Playback Volume", SB_DSP4_MIC_DEV, 3, 31),
CMIPCI_SB_SW_MONO("Mic Playback Switch", 0),
CMIPCI_DOUBLE("Mic Capture Switch", SB_DSP4_INPUT_LEFT, SB_DSP4_INPUT_RIGHT, 0, 0, 1, 0, 0),
- CMIPCI_SB_VOL_MONO("PC Speaker Playback Volume", SB_DSP4_SPEAKER_DEV, 6, 3),
+ CMIPCI_SB_VOL_MONO("Beep Playback Volume", SB_DSP4_SPEAKER_DEV, 6, 3),
CMIPCI_MIXER_VOL_STEREO("Aux Playback Volume", CM_REG_AUX_VOL, 4, 0, 15),
CMIPCI_MIXER_SW_STEREO("Aux Playback Switch", CM_REG_MIXER2, CM_VAUXLM_SHIFT, CM_VAUXRM_SHIFT, 0),
CMIPCI_MIXER_SW_STEREO("Aux Capture Switch", CM_REG_MIXER2, CM_RAUXLEN_SHIFT, CM_RAUXREN_SHIFT, 0),
@@ -2310,7 +2310,7 @@ static struct snd_kcontrol_new snd_cmipci_mixers[] __devinitdata = {
CMIPCI_MIXER_VOL_MONO("Mic Capture Volume", CM_REG_MIXER2, CM_VADMIC_SHIFT, 7),
CMIPCI_SB_VOL_MONO("Phone Playback Volume", CM_REG_EXTENT_IND, 5, 7),
CMIPCI_DOUBLE("Phone Playback Switch", CM_REG_EXTENT_IND, CM_REG_EXTENT_IND, 4, 4, 1, 0, 0),
- CMIPCI_DOUBLE("PC Speaker Playback Switch", CM_REG_EXTENT_IND, CM_REG_EXTENT_IND, 3, 3, 1, 0, 0),
+ CMIPCI_DOUBLE("Beep Playback Switch", CM_REG_EXTENT_IND, CM_REG_EXTENT_IND, 3, 3, 1, 0, 0),
CMIPCI_DOUBLE("Mic Boost Capture Switch", CM_REG_EXTENT_IND, CM_REG_EXTENT_IND, 0, 0, 1, 0, 0),
};
diff --git a/sound/pci/ctxfi/ctatc.c b/sound/pci/ctxfi/ctatc.c
index b1b3a644f73..cb65bd0dd35 100644
--- a/sound/pci/ctxfi/ctatc.c
+++ b/sound/pci/ctxfi/ctatc.c
@@ -240,7 +240,7 @@ static int select_rom(unsigned int pitch)
} else if (pitch == 0x02000000) {
/* pitch == 2 */
return 3;
- } else if (pitch >= 0x0 && pitch <= 0x08000000) {
+ } else if (pitch <= 0x08000000) {
/* 0 <= pitch <= 8 */
return 0;
} else {
@@ -1037,7 +1037,7 @@ static int atc_line_front_unmute(struct ct_atc *atc, unsigned char state)
static int atc_line_surround_unmute(struct ct_atc *atc, unsigned char state)
{
- return atc_daio_unmute(atc, state, LINEO4);
+ return atc_daio_unmute(atc, state, LINEO2);
}
static int atc_line_clfe_unmute(struct ct_atc *atc, unsigned char state)
@@ -1047,7 +1047,7 @@ static int atc_line_clfe_unmute(struct ct_atc *atc, unsigned char state)
static int atc_line_rear_unmute(struct ct_atc *atc, unsigned char state)
{
- return atc_daio_unmute(atc, state, LINEO2);
+ return atc_daio_unmute(atc, state, LINEO4);
}
static int atc_line_in_unmute(struct ct_atc *atc, unsigned char state)
diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c
index da2065cd2c0..1305f7ca02c 100644
--- a/sound/pci/echoaudio/echoaudio.c
+++ b/sound/pci/echoaudio/echoaudio.c
@@ -950,7 +950,7 @@ static int __devinit snd_echo_new_pcm(struct echoaudio *chip)
Control interface
******************************************************************************/
-#ifndef ECHOCARD_HAS_VMIXER
+#if !defined(ECHOCARD_HAS_VMIXER) || defined(ECHOCARD_HAS_LINE_OUT_GAIN)
/******************* PCM output volume *******************/
static int snd_echo_output_gain_info(struct snd_kcontrol *kcontrol,
@@ -1003,6 +1003,19 @@ static int snd_echo_output_gain_put(struct snd_kcontrol *kcontrol,
return changed;
}
+#ifdef ECHOCARD_HAS_LINE_OUT_GAIN
+/* On the Mia this one controls the line-out volume */
+static struct snd_kcontrol_new snd_echo_line_output_gain __devinitdata = {
+ .name = "Line Playback Volume",
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE |
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ,
+ .info = snd_echo_output_gain_info,
+ .get = snd_echo_output_gain_get,
+ .put = snd_echo_output_gain_put,
+ .tlv = {.p = db_scale_output_gain},
+};
+#else
static struct snd_kcontrol_new snd_echo_pcm_output_gain __devinitdata = {
.name = "PCM Playback Volume",
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
@@ -1012,9 +1025,10 @@ static struct snd_kcontrol_new snd_echo_pcm_output_gain __devinitdata = {
.put = snd_echo_output_gain_put,
.tlv = {.p = db_scale_output_gain},
};
-
#endif
+#endif /* !ECHOCARD_HAS_VMIXER || ECHOCARD_HAS_LINE_OUT_GAIN */
+
#ifdef ECHOCARD_HAS_INPUT_GAIN
@@ -2030,10 +2044,18 @@ static int __devinit snd_echo_probe(struct pci_dev *pci,
snd_echo_vmixer.count = num_pipes_out(chip) * num_busses_out(chip);
if ((err = snd_ctl_add(chip->card, snd_ctl_new1(&snd_echo_vmixer, chip))) < 0)
goto ctl_error;
-#else
- if ((err = snd_ctl_add(chip->card, snd_ctl_new1(&snd_echo_pcm_output_gain, chip))) < 0)
+#ifdef ECHOCARD_HAS_LINE_OUT_GAIN
+ err = snd_ctl_add(chip->card,
+ snd_ctl_new1(&snd_echo_line_output_gain, chip));
+ if (err < 0)
goto ctl_error;
#endif
+#else /* ECHOCARD_HAS_VMIXER */
+ err = snd_ctl_add(chip->card,
+ snd_ctl_new1(&snd_echo_pcm_output_gain, chip));
+ if (err < 0)
+ goto ctl_error;
+#endif /* ECHOCARD_HAS_VMIXER */
#ifdef ECHOCARD_HAS_INPUT_GAIN
if ((err = snd_ctl_add(chip->card, snd_ctl_new1(&snd_echo_line_input_gain, chip))) < 0)
diff --git a/sound/pci/echoaudio/mia.c b/sound/pci/echoaudio/mia.c
index f3b9b45c9c1..f05c8c097aa 100644
--- a/sound/pci/echoaudio/mia.c
+++ b/sound/pci/echoaudio/mia.c
@@ -29,6 +29,7 @@
#define ECHOCARD_HAS_ADAT FALSE
#define ECHOCARD_HAS_STEREO_BIG_ENDIAN32
#define ECHOCARD_HAS_MIDI
+#define ECHOCARD_HAS_LINE_OUT_GAIN
/* Pipe indexes */
#define PX_ANALOG_OUT 0 /* 8 */
diff --git a/sound/pci/emu10k1/emu10k1x.c b/sound/pci/emu10k1/emu10k1x.c
index 36e08bd2b3c..6b8ae7b5cd5 100644
--- a/sound/pci/emu10k1/emu10k1x.c
+++ b/sound/pci/emu10k1/emu10k1x.c
@@ -1040,8 +1040,7 @@ static void snd_emu10k1x_proc_reg_write(struct snd_info_entry *entry,
if (sscanf(line, "%x %x %x", &reg, &channel_id, &val) != 3)
continue;
- if ((reg < 0x49) && (reg >= 0) && (val <= 0xffffffff)
- && (channel_id >= 0) && (channel_id <= 2) )
+ if (reg < 0x49 && val <= 0xffffffff && channel_id <= 2)
snd_emu10k1x_ptr_write(emu, reg, channel_id, val);
}
}
diff --git a/sound/pci/emu10k1/emumixer.c b/sound/pci/emu10k1/emumixer.c
index b0fb6c917c3..05afe06e353 100644
--- a/sound/pci/emu10k1/emumixer.c
+++ b/sound/pci/emu10k1/emumixer.c
@@ -1818,8 +1818,8 @@ int __devinit snd_emu10k1_mixer(struct snd_emu10k1 *emu,
"Master Playback Switch", "Master Capture Switch",
"Master Playback Volume", "Master Capture Volume",
"Wave Master Playback Volume", "Master Playback Volume",
- "PC Speaker Playback Switch", "PC Speaker Capture Switch",
- "PC Speaker Playback Volume", "PC Speaker Capture Volume",
+ "Beep Playback Switch", "Beep Capture Switch",
+ "Beep Playback Volume", "Beep Capture Volume",
"Phone Playback Switch", "Phone Capture Switch",
"Phone Playback Volume", "Phone Capture Volume",
"Mic Playback Switch", "Mic Capture Switch",
diff --git a/sound/pci/emu10k1/emuproc.c b/sound/pci/emu10k1/emuproc.c
index 216f9748aff..baa7cd508cd 100644
--- a/sound/pci/emu10k1/emuproc.c
+++ b/sound/pci/emu10k1/emuproc.c
@@ -451,7 +451,7 @@ static void snd_emu_proc_io_reg_write(struct snd_info_entry *entry,
while (!snd_info_get_line(buffer, line, sizeof(line))) {
if (sscanf(line, "%x %x", &reg, &val) != 2)
continue;
- if ((reg < 0x40) && (reg >= 0) && (val <= 0xffffffff) ) {
+ if (reg < 0x40 && val <= 0xffffffff) {
spin_lock_irqsave(&emu->emu_lock, flags);
outl(val, emu->port + (reg & 0xfffffffc));
spin_unlock_irqrestore(&emu->emu_lock, flags);
@@ -527,7 +527,7 @@ static void snd_emu_proc_ptr_reg_write(struct snd_info_entry *entry,
while (!snd_info_get_line(buffer, line, sizeof(line))) {
if (sscanf(line, "%x %x %x", &reg, &channel_id, &val) != 3)
continue;
- if ((reg < 0xa0) && (reg >= 0) && (val <= 0xffffffff) && (channel_id >= 0) && (channel_id <= 3) )
+ if (reg < 0xa0 && val <= 0xffffffff && channel_id <= 3)
snd_ptr_write(emu, iobase, reg, channel_id, val);
}
}
diff --git a/sound/pci/emu10k1/io.c b/sound/pci/emu10k1/io.c
index c1a5aa15af8..5ef7080e14d 100644
--- a/sound/pci/emu10k1/io.c
+++ b/sound/pci/emu10k1/io.c
@@ -256,7 +256,7 @@ int snd_emu1010_fpga_write(struct snd_emu10k1 * emu, u32 reg, u32 value)
if (reg > 0x3f)
return 1;
reg += 0x40; /* 0x40 upwards are registers. */
- if (value < 0 || value > 0x3f) /* 0 to 0x3f are values */
+ if (value > 0x3f) /* 0 to 0x3f are values */
return 1;
spin_lock_irqsave(&emu->emu_lock, flags);
outl(reg, emu->port + A_IOCFG);
diff --git a/sound/pci/es1938.c b/sound/pci/es1938.c
index 820318ee62c..fb83e1ffa5c 100644
--- a/sound/pci/es1938.c
+++ b/sound/pci/es1938.c
@@ -1387,7 +1387,7 @@ ES1938_DOUBLE_TLV("Aux Playback Volume", 0, 0x3a, 0x3a, 4, 0, 15, 0,
db_scale_line),
ES1938_DOUBLE_TLV("Capture Volume", 0, 0xb4, 0xb4, 4, 0, 15, 0,
db_scale_capture),
-ES1938_SINGLE("PC Speaker Volume", 0, 0x3c, 0, 7, 0),
+ES1938_SINGLE("Beep Volume", 0, 0x3c, 0, 7, 0),
ES1938_SINGLE("Record Monitor", 0, 0xa8, 3, 1, 0),
ES1938_SINGLE("Capture Switch", 0, 0x1c, 4, 1, 1),
{
diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig
index 55545e0818b..556cff937be 100644
--- a/sound/pci/hda/Kconfig
+++ b/sound/pci/hda/Kconfig
@@ -38,9 +38,20 @@ config SND_HDA_INPUT_BEEP
Say Y here to build a digital beep interface for HD-audio
driver. This interface is used to generate digital beeps.
+config SND_HDA_INPUT_BEEP_MODE
+ int "Digital beep registration mode (0=off, 1=on, 2=mute sw on/off)"
+ depends on SND_HDA_INPUT_BEEP=y
+ default "1"
+ range 0 2
+ help
+ Set 0 to disable the digital beep interface for HD-audio by default.
+ Set 1 to always enable the digital beep interface for HD-audio by
+ default. Set 2 to control the beep device registration to input
+ layer using a "Beep Switch" in mixer applications.
+
config SND_HDA_INPUT_JACK
bool "Support jack plugging notification via input layer"
- depends on INPUT=y || INPUT=SND_HDA_INTEL
+ depends on INPUT=y || INPUT=SND
select SND_JACK
help
Say Y here to enable the jack plugging notification via
diff --git a/sound/pci/hda/hda_beep.c b/sound/pci/hda/hda_beep.c
index 3f51a981e60..5fe34a8d8c8 100644
--- a/sound/pci/hda/hda_beep.c
+++ b/sound/pci/hda/hda_beep.c
@@ -113,23 +113,25 @@ static int snd_hda_beep_event(struct input_dev *dev, unsigned int type,
return 0;
}
-int snd_hda_attach_beep_device(struct hda_codec *codec, int nid)
+static void snd_hda_do_detach(struct hda_beep *beep)
+{
+ input_unregister_device(beep->dev);
+ beep->dev = NULL;
+ cancel_work_sync(&beep->beep_work);
+ /* turn off beep for sure */
+ snd_hda_codec_write_cache(beep->codec, beep->nid, 0,
+ AC_VERB_SET_BEEP_CONTROL, 0);
+}
+
+static int snd_hda_do_attach(struct hda_beep *beep)
{
struct input_dev *input_dev;
- struct hda_beep *beep;
+ struct hda_codec *codec = beep->codec;
int err;
- if (!snd_hda_get_bool_hint(codec, "beep"))
- return 0; /* disabled explicitly */
-
- beep = kzalloc(sizeof(*beep), GFP_KERNEL);
- if (beep == NULL)
- return -ENOMEM;
- snprintf(beep->phys, sizeof(beep->phys),
- "card%d/codec#%d/beep0", codec->bus->card->number, codec->addr);
input_dev = input_allocate_device();
if (!input_dev) {
- kfree(beep);
+ printk(KERN_INFO "hda_beep: unable to allocate input device\n");
return -ENOMEM;
}
@@ -151,21 +153,96 @@ int snd_hda_attach_beep_device(struct hda_codec *codec, int nid)
err = input_register_device(input_dev);
if (err < 0) {
input_free_device(input_dev);
- kfree(beep);
+ printk(KERN_INFO "hda_beep: unable to register input device\n");
return err;
}
+ beep->dev = input_dev;
+ return 0;
+}
+
+static void snd_hda_do_register(struct work_struct *work)
+{
+ struct hda_beep *beep =
+ container_of(work, struct hda_beep, register_work);
+
+ mutex_lock(&beep->mutex);
+ if (beep->enabled && !beep->dev)
+ snd_hda_do_attach(beep);
+ mutex_unlock(&beep->mutex);
+}
+
+static void snd_hda_do_unregister(struct work_struct *work)
+{
+ struct hda_beep *beep =
+ container_of(work, struct hda_beep, unregister_work.work);
+
+ mutex_lock(&beep->mutex);
+ if (!beep->enabled && beep->dev)
+ snd_hda_do_detach(beep);
+ mutex_unlock(&beep->mutex);
+}
+int snd_hda_enable_beep_device(struct hda_codec *codec, int enable)
+{
+ struct hda_beep *beep = codec->beep;
+ enable = !!enable;
+ if (beep == NULL)
+ return 0;
+ if (beep->enabled != enable) {
+ beep->enabled = enable;
+ if (!enable) {
+ /* turn off beep */
+ snd_hda_codec_write_cache(beep->codec, beep->nid, 0,
+ AC_VERB_SET_BEEP_CONTROL, 0);
+ }
+ if (beep->mode == HDA_BEEP_MODE_SWREG) {
+ if (enable) {
+ cancel_delayed_work(&beep->unregister_work);
+ schedule_work(&beep->register_work);
+ } else {
+ schedule_delayed_work(&beep->unregister_work,
+ HZ);
+ }
+ }
+ return 1;
+ }
+ return 0;
+}
+EXPORT_SYMBOL_HDA(snd_hda_enable_beep_device);
+
+int snd_hda_attach_beep_device(struct hda_codec *codec, int nid)
+{
+ struct hda_beep *beep;
+
+ if (!snd_hda_get_bool_hint(codec, "beep"))
+ return 0; /* disabled explicitly by hints */
+ if (codec->beep_mode == HDA_BEEP_MODE_OFF)
+ return 0; /* disabled by module option */
+
+ beep = kzalloc(sizeof(*beep), GFP_KERNEL);
+ if (beep == NULL)
+ return -ENOMEM;
+ snprintf(beep->phys, sizeof(beep->phys),
+ "card%d/codec#%d/beep0", codec->bus->card->number, codec->addr);
/* enable linear scale */
snd_hda_codec_write(codec, nid, 0,
AC_VERB_SET_DIGI_CONVERT_2, 0x01);
beep->nid = nid;
- beep->dev = input_dev;
beep->codec = codec;
- beep->enabled = 1;
+ beep->mode = codec->beep_mode;
codec->beep = beep;
+ INIT_WORK(&beep->register_work, &snd_hda_do_register);
+ INIT_DELAYED_WORK(&beep->unregister_work, &snd_hda_do_unregister);
INIT_WORK(&beep->beep_work, &snd_hda_generate_beep);
+ mutex_init(&beep->mutex);
+
+ if (beep->mode == HDA_BEEP_MODE_ON) {
+ beep->enabled = 1;
+ snd_hda_do_register(&beep->register_work);
+ }
+
return 0;
}
EXPORT_SYMBOL_HDA(snd_hda_attach_beep_device);
@@ -174,11 +251,12 @@ void snd_hda_detach_beep_device(struct hda_codec *codec)
{
struct hda_beep *beep = codec->beep;
if (beep) {
- cancel_work_sync(&beep->beep_work);
-
- input_unregister_device(beep->dev);
- kfree(beep);
+ cancel_work_sync(&beep->register_work);
+ cancel_delayed_work(&beep->unregister_work);
+ if (beep->enabled)
+ snd_hda_do_detach(beep);
codec->beep = NULL;
+ kfree(beep);
}
}
EXPORT_SYMBOL_HDA(snd_hda_detach_beep_device);
diff --git a/sound/pci/hda/hda_beep.h b/sound/pci/hda/hda_beep.h
index 0c3de787c71..f1de1bac042 100644
--- a/sound/pci/hda/hda_beep.h
+++ b/sound/pci/hda/hda_beep.h
@@ -24,19 +24,29 @@
#include "hda_codec.h"
+#define HDA_BEEP_MODE_OFF 0
+#define HDA_BEEP_MODE_ON 1
+#define HDA_BEEP_MODE_SWREG 2
+
/* beep information */
struct hda_beep {
struct input_dev *dev;
struct hda_codec *codec;
+ unsigned int mode;
char phys[32];
int tone;
hda_nid_t nid;
unsigned int enabled:1;
+ unsigned int request_enable:1;
unsigned int linear_tone:1; /* linear tone for IDT/STAC codec */
+ struct work_struct register_work; /* registration work */
+ struct delayed_work unregister_work; /* unregistration work */
struct work_struct beep_work; /* scheduled task for beep event */
+ struct mutex mutex;
};
#ifdef CONFIG_SND_HDA_INPUT_BEEP
+int snd_hda_enable_beep_device(struct hda_codec *codec, int enable);
int snd_hda_attach_beep_device(struct hda_codec *codec, int nid);
void snd_hda_detach_beep_device(struct hda_codec *codec);
#else
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index af989f660cc..9cfdb771928 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -30,6 +30,7 @@
#include <sound/tlv.h>
#include <sound/initval.h>
#include "hda_local.h"
+#include "hda_beep.h"
#include <sound/hda_hwdep.h>
/*
@@ -93,6 +94,13 @@ static void hda_keep_power_on(struct hda_codec *codec);
static inline void hda_keep_power_on(struct hda_codec *codec) {}
#endif
+/**
+ * snd_hda_get_jack_location - Give a location string of the jack
+ * @cfg: pin default config value
+ *
+ * Parse the pin default config value and returns the string of the
+ * jack location, e.g. "Rear", "Front", etc.
+ */
const char *snd_hda_get_jack_location(u32 cfg)
{
static char *bases[7] = {
@@ -120,6 +128,13 @@ const char *snd_hda_get_jack_location(u32 cfg)
}
EXPORT_SYMBOL_HDA(snd_hda_get_jack_location);
+/**
+ * snd_hda_get_jack_connectivity - Give a connectivity string of the jack
+ * @cfg: pin default config value
+ *
+ * Parse the pin default config value and returns the string of the
+ * jack connectivity, i.e. external or internal connection.
+ */
const char *snd_hda_get_jack_connectivity(u32 cfg)
{
static char *jack_locations[4] = { "Ext", "Int", "Sep", "Oth" };
@@ -128,6 +143,13 @@ const char *snd_hda_get_jack_connectivity(u32 cfg)
}
EXPORT_SYMBOL_HDA(snd_hda_get_jack_connectivity);
+/**
+ * snd_hda_get_jack_type - Give a type string of the jack
+ * @cfg: pin default config value
+ *
+ * Parse the pin default config value and returns the string of the
+ * jack type, i.e. the purpose of the jack, such as Line-Out or CD.
+ */
const char *snd_hda_get_jack_type(u32 cfg)
{
static char *jack_types[16] = {
@@ -515,6 +537,7 @@ static int snd_hda_bus_dev_register(struct snd_device *device)
struct hda_codec *codec;
list_for_each_entry(codec, &bus->codec_list, list) {
snd_hda_hwdep_add_sysfs(codec);
+ snd_hda_hwdep_add_power_sysfs(codec);
}
return 0;
}
@@ -820,6 +843,16 @@ int snd_hda_add_pincfg(struct hda_codec *codec, struct snd_array *list,
return 0;
}
+/**
+ * snd_hda_codec_set_pincfg - Override a pin default configuration
+ * @codec: the HDA codec
+ * @nid: NID to set the pin config
+ * @cfg: the pin default config value
+ *
+ * Override a pin default configuration value in the cache.
+ * This value can be read by snd_hda_codec_get_pincfg() in a higher
+ * priority than the real hardware value.
+ */
int snd_hda_codec_set_pincfg(struct hda_codec *codec,
hda_nid_t nid, unsigned int cfg)
{
@@ -827,7 +860,15 @@ int snd_hda_codec_set_pincfg(struct hda_codec *codec,
}
EXPORT_SYMBOL_HDA(snd_hda_codec_set_pincfg);
-/* get the current pin config value of the given pin NID */
+/**
+ * snd_hda_codec_get_pincfg - Obtain a pin-default configuration
+ * @codec: the HDA codec
+ * @nid: NID to get the pin config
+ *
+ * Get the current pin config value of the given pin NID.
+ * If the pincfg value is cached or overridden via sysfs or driver,
+ * returns the cached value.
+ */
unsigned int snd_hda_codec_get_pincfg(struct hda_codec *codec, hda_nid_t nid)
{
struct hda_pincfg *pin;
@@ -944,7 +985,7 @@ int /*__devinit*/ snd_hda_codec_new(struct hda_bus *bus, unsigned int codec_addr
mutex_init(&codec->control_mutex);
init_hda_cache(&codec->amp_cache, sizeof(struct hda_amp_info));
init_hda_cache(&codec->cmd_cache, sizeof(struct hda_cache_head));
- snd_array_init(&codec->mixers, sizeof(struct snd_kcontrol *), 32);
+ snd_array_init(&codec->mixers, sizeof(struct hda_nid_item), 60);
snd_array_init(&codec->init_pins, sizeof(struct hda_pincfg), 16);
snd_array_init(&codec->driver_pins, sizeof(struct hda_pincfg), 16);
if (codec->bus->modelname) {
@@ -1026,6 +1067,15 @@ int /*__devinit*/ snd_hda_codec_new(struct hda_bus *bus, unsigned int codec_addr
}
EXPORT_SYMBOL_HDA(snd_hda_codec_new);
+/**
+ * snd_hda_codec_configure - (Re-)configure the HD-audio codec
+ * @codec: the HDA codec
+ *
+ * Start parsing of the given codec tree and (re-)initialize the whole
+ * patch instance.
+ *
+ * Returns 0 if successful or a negative error code.
+ */
int snd_hda_codec_configure(struct hda_codec *codec)
{
int err;
@@ -1088,6 +1138,11 @@ void snd_hda_codec_setup_stream(struct hda_codec *codec, hda_nid_t nid,
}
EXPORT_SYMBOL_HDA(snd_hda_codec_setup_stream);
+/**
+ * snd_hda_codec_cleanup_stream - clean up the codec for closing
+ * @codec: the CODEC to clean up
+ * @nid: the NID to clean up
+ */
void snd_hda_codec_cleanup_stream(struct hda_codec *codec, hda_nid_t nid)
{
if (!nid)
@@ -1163,8 +1218,17 @@ 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
+/**
+ * query_amp_caps - query AMP capabilities
+ * @codec: the HD-auio codec
+ * @nid: the NID to query
+ * @direction: either #HDA_INPUT or #HDA_OUTPUT
+ *
+ * Query AMP capabilities for the given widget and direction.
+ * Returns the obtained capability bits.
+ *
+ * When cap bits have been already read, this doesn't read again but
+ * returns the cached value.
*/
u32 query_amp_caps(struct hda_codec *codec, hda_nid_t nid, int direction)
{
@@ -1187,6 +1251,19 @@ u32 query_amp_caps(struct hda_codec *codec, hda_nid_t nid, int direction)
}
EXPORT_SYMBOL_HDA(query_amp_caps);
+/**
+ * snd_hda_override_amp_caps - Override the AMP capabilities
+ * @codec: the CODEC to clean up
+ * @nid: the NID to clean up
+ * @direction: either #HDA_INPUT or #HDA_OUTPUT
+ * @caps: the capability bits to set
+ *
+ * Override the cached AMP caps bits value by the given one.
+ * This function is useful if the driver needs to adjust the AMP ranges,
+ * e.g. limit to 0dB, etc.
+ *
+ * Returns zero if successful or a negative error code.
+ */
int snd_hda_override_amp_caps(struct hda_codec *codec, hda_nid_t nid, int dir,
unsigned int caps)
{
@@ -1222,6 +1299,17 @@ static unsigned int read_pin_cap(struct hda_codec *codec, hda_nid_t nid)
return snd_hda_param_read(codec, nid, AC_PAR_PIN_CAP);
}
+/**
+ * snd_hda_query_pin_caps - Query PIN capabilities
+ * @codec: the HD-auio codec
+ * @nid: the NID to query
+ *
+ * Query PIN capabilities for the given widget.
+ * Returns the obtained capability bits.
+ *
+ * When cap bits have been already read, this doesn't read again but
+ * returns the cached value.
+ */
u32 snd_hda_query_pin_caps(struct hda_codec *codec, hda_nid_t nid)
{
return query_caps_hash(codec, nid, HDA_HASH_PINCAP_KEY(nid),
@@ -1229,6 +1317,40 @@ u32 snd_hda_query_pin_caps(struct hda_codec *codec, hda_nid_t nid)
}
EXPORT_SYMBOL_HDA(snd_hda_query_pin_caps);
+/**
+ * snd_hda_pin_sense - execute pin sense measurement
+ * @codec: the CODEC to sense
+ * @nid: the pin NID to sense
+ *
+ * Execute necessary pin sense measurement and return its Presence Detect,
+ * Impedance, ELD Valid etc. status bits.
+ */
+u32 snd_hda_pin_sense(struct hda_codec *codec, hda_nid_t nid)
+{
+ u32 pincap = snd_hda_query_pin_caps(codec, nid);
+
+ if (pincap & AC_PINCAP_TRIG_REQ) /* need trigger? */
+ snd_hda_codec_read(codec, nid, 0, AC_VERB_SET_PIN_SENSE, 0);
+
+ return snd_hda_codec_read(codec, nid, 0,
+ AC_VERB_GET_PIN_SENSE, 0);
+}
+EXPORT_SYMBOL_HDA(snd_hda_pin_sense);
+
+/**
+ * snd_hda_jack_detect - query pin Presence Detect status
+ * @codec: the CODEC to sense
+ * @nid: the pin NID to sense
+ *
+ * Query and return the pin's Presence Detect status.
+ */
+int snd_hda_jack_detect(struct hda_codec *codec, hda_nid_t nid)
+{
+ u32 sense = snd_hda_pin_sense(codec, nid);
+ return !!(sense & AC_PINSENSE_PRESENCE);
+}
+EXPORT_SYMBOL_HDA(snd_hda_jack_detect);
+
/*
* read the current volume to info
* if the cache exists, read the cache value.
@@ -1269,8 +1391,15 @@ static void put_vol_mute(struct hda_codec *codec, struct hda_amp_info *info,
info->vol[ch] = val;
}
-/*
- * read AMP value. The volume is between 0 to 0x7f, 0x80 = mute bit.
+/**
+ * snd_hda_codec_amp_read - Read AMP value
+ * @codec: HD-audio codec
+ * @nid: NID to read the AMP value
+ * @ch: channel (left=0 or right=1)
+ * @direction: #HDA_INPUT or #HDA_OUTPUT
+ * @index: the index value (only for input direction)
+ *
+ * Read AMP value. The volume is between 0 to 0x7f, 0x80 = mute bit.
*/
int snd_hda_codec_amp_read(struct hda_codec *codec, hda_nid_t nid, int ch,
int direction, int index)
@@ -1283,8 +1412,18 @@ int snd_hda_codec_amp_read(struct hda_codec *codec, hda_nid_t nid, int ch,
}
EXPORT_SYMBOL_HDA(snd_hda_codec_amp_read);
-/*
- * update the AMP value, mask = bit mask to set, val = the value
+/**
+ * snd_hda_codec_amp_update - update the AMP value
+ * @codec: HD-audio codec
+ * @nid: NID to read the AMP value
+ * @ch: channel (left=0 or right=1)
+ * @direction: #HDA_INPUT or #HDA_OUTPUT
+ * @idx: the index value (only for input direction)
+ * @mask: bit mask to set
+ * @val: the bits value to set
+ *
+ * Update the AMP value with a bit mask.
+ * Returns 0 if the value is unchanged, 1 if changed.
*/
int snd_hda_codec_amp_update(struct hda_codec *codec, hda_nid_t nid, int ch,
int direction, int idx, int mask, int val)
@@ -1303,8 +1442,17 @@ int snd_hda_codec_amp_update(struct hda_codec *codec, hda_nid_t nid, int ch,
}
EXPORT_SYMBOL_HDA(snd_hda_codec_amp_update);
-/*
- * update the AMP stereo with the same mask and value
+/**
+ * snd_hda_codec_amp_stereo - update the AMP stereo values
+ * @codec: HD-audio codec
+ * @nid: NID to read the AMP value
+ * @direction: #HDA_INPUT or #HDA_OUTPUT
+ * @idx: the index value (only for input direction)
+ * @mask: bit mask to set
+ * @val: the bits value to set
+ *
+ * Update the AMP values like snd_hda_codec_amp_update(), but for a
+ * stereo widget 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)
@@ -1318,7 +1466,12 @@ int snd_hda_codec_amp_stereo(struct hda_codec *codec, hda_nid_t nid,
EXPORT_SYMBOL_HDA(snd_hda_codec_amp_stereo);
#ifdef SND_HDA_NEEDS_RESUME
-/* resume the all amp commands from the cache */
+/**
+ * snd_hda_codec_resume_amp - Resume all AMP commands from the cache
+ * @codec: HD-audio codec
+ *
+ * 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.buf.list;
@@ -1344,7 +1497,12 @@ void snd_hda_codec_resume_amp(struct hda_codec *codec)
EXPORT_SYMBOL_HDA(snd_hda_codec_resume_amp);
#endif /* SND_HDA_NEEDS_RESUME */
-/* volume */
+/**
+ * snd_hda_mixer_amp_volume_info - Info callback for a standard AMP mixer
+ *
+ * The control element is supposed to have the private_value field
+ * set up via HDA_COMPOSE_AMP_VAL*() or related macros.
+ */
int snd_hda_mixer_amp_volume_info(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
@@ -1400,6 +1558,12 @@ update_amp_value(struct hda_codec *codec, hda_nid_t nid,
HDA_AMP_VOLMASK, val);
}
+/**
+ * snd_hda_mixer_amp_volume_get - Get callback for a standard AMP mixer volume
+ *
+ * The control element is supposed to have the private_value field
+ * set up via HDA_COMPOSE_AMP_VAL*() or related macros.
+ */
int snd_hda_mixer_amp_volume_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
@@ -1419,6 +1583,12 @@ int snd_hda_mixer_amp_volume_get(struct snd_kcontrol *kcontrol,
}
EXPORT_SYMBOL_HDA(snd_hda_mixer_amp_volume_get);
+/**
+ * snd_hda_mixer_amp_volume_put - Put callback for a standard AMP mixer volume
+ *
+ * The control element is supposed to have the private_value field
+ * set up via HDA_COMPOSE_AMP_VAL*() or related macros.
+ */
int snd_hda_mixer_amp_volume_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
@@ -1443,6 +1613,12 @@ int snd_hda_mixer_amp_volume_put(struct snd_kcontrol *kcontrol,
}
EXPORT_SYMBOL_HDA(snd_hda_mixer_amp_volume_put);
+/**
+ * snd_hda_mixer_amp_volume_put - TLV callback for a standard AMP mixer volume
+ *
+ * The control element is supposed to have the private_value field
+ * set up via HDA_COMPOSE_AMP_VAL*() or related macros.
+ */
int snd_hda_mixer_amp_tlv(struct snd_kcontrol *kcontrol, int op_flag,
unsigned int size, unsigned int __user *_tlv)
{
@@ -1472,8 +1648,16 @@ int snd_hda_mixer_amp_tlv(struct snd_kcontrol *kcontrol, int op_flag,
}
EXPORT_SYMBOL_HDA(snd_hda_mixer_amp_tlv);
-/*
- * set (static) TLV for virtual master volume; recalculated as max 0dB
+/**
+ * snd_hda_set_vmaster_tlv - Set TLV for a virtual master control
+ * @codec: HD-audio codec
+ * @nid: NID of a reference widget
+ * @dir: #HDA_INPUT or #HDA_OUTPUT
+ * @tlv: TLV data to be stored, at least 4 elements
+ *
+ * Set (static) TLV data for a virtual master volume using the AMP caps
+ * obtained from the reference NID.
+ * The volume range is recalculated as if the max volume is 0dB.
*/
void snd_hda_set_vmaster_tlv(struct hda_codec *codec, hda_nid_t nid, int dir,
unsigned int *tlv)
@@ -1507,6 +1691,13 @@ _snd_hda_find_mixer_ctl(struct hda_codec *codec,
return snd_ctl_find_id(codec->bus->card, &id);
}
+/**
+ * snd_hda_find_mixer_ctl - Find a mixer control element with the given name
+ * @codec: HD-audio codec
+ * @name: ctl id name string
+ *
+ * Get the control element with the given id string and IFACE_MIXER.
+ */
struct snd_kcontrol *snd_hda_find_mixer_ctl(struct hda_codec *codec,
const char *name)
{
@@ -1514,30 +1705,57 @@ struct snd_kcontrol *snd_hda_find_mixer_ctl(struct hda_codec *codec,
}
EXPORT_SYMBOL_HDA(snd_hda_find_mixer_ctl);
-/* Add a control element and assign to the codec */
-int snd_hda_ctl_add(struct hda_codec *codec, struct snd_kcontrol *kctl)
+/**
+ * snd_hda_ctl-add - Add a control element and assign to the codec
+ * @codec: HD-audio codec
+ * @nid: corresponding NID (optional)
+ * @kctl: the control element to assign
+ *
+ * Add the given control element to an array inside the codec instance.
+ * All control elements belonging to a codec are supposed to be added
+ * by this function so that a proper clean-up works at the free or
+ * reconfiguration time.
+ *
+ * If non-zero @nid is passed, the NID is assigned to the control element.
+ * The assignment is shown in the codec proc file.
+ *
+ * snd_hda_ctl_add() checks the control subdev id field whether
+ * #HDA_SUBDEV_NID_FLAG bit is set. If set (and @nid is zero), the lower
+ * bits value is taken as the NID to assign.
+ */
+int snd_hda_ctl_add(struct hda_codec *codec, hda_nid_t nid,
+ struct snd_kcontrol *kctl)
{
int err;
- struct snd_kcontrol **knewp;
+ struct hda_nid_item *item;
+ if (kctl->id.subdevice & HDA_SUBDEV_NID_FLAG) {
+ if (nid == 0)
+ nid = kctl->id.subdevice & 0xffff;
+ kctl->id.subdevice = 0;
+ }
err = snd_ctl_add(codec->bus->card, kctl);
if (err < 0)
return err;
- knewp = snd_array_new(&codec->mixers);
- if (!knewp)
+ item = snd_array_new(&codec->mixers);
+ if (!item)
return -ENOMEM;
- *knewp = kctl;
+ item->kctl = kctl;
+ item->nid = nid;
return 0;
}
EXPORT_SYMBOL_HDA(snd_hda_ctl_add);
-/* Clear all controls assigned to the given codec */
+/**
+ * snd_hda_ctls_clear - Clear all controls assigned to the given codec
+ * @codec: HD-audio codec
+ */
void snd_hda_ctls_clear(struct hda_codec *codec)
{
int i;
- struct snd_kcontrol **kctls = codec->mixers.list;
+ struct hda_nid_item *items = codec->mixers.list;
for (i = 0; i < codec->mixers.used; i++)
- snd_ctl_remove(codec->bus->card, kctls[i]);
+ snd_ctl_remove(codec->bus->card, items[i].kctl);
snd_array_free(&codec->mixers);
}
@@ -1563,6 +1781,16 @@ static void hda_unlock_devices(struct snd_card *card)
spin_unlock(&card->files_lock);
}
+/**
+ * snd_hda_codec_reset - Clear all objects assigned to the codec
+ * @codec: HD-audio codec
+ *
+ * This frees the all PCM and control elements assigned to the codec, and
+ * clears the caches and restores the pin default configurations.
+ *
+ * When a device is being used, it returns -EBSY. If successfully freed,
+ * returns zero.
+ */
int snd_hda_codec_reset(struct hda_codec *codec)
{
struct snd_card *card = codec->bus->card;
@@ -1626,7 +1854,22 @@ int snd_hda_codec_reset(struct hda_codec *codec)
return 0;
}
-/* create a virtual master control and add slaves */
+/**
+ * snd_hda_add_vmaster - create a virtual master control and add slaves
+ * @codec: HD-audio codec
+ * @name: vmaster control name
+ * @tlv: TLV data (optional)
+ * @slaves: slave control names (optional)
+ *
+ * Create a virtual master control with the given name. The TLV data
+ * must be either NULL or a valid data.
+ *
+ * @slaves is a NULL-terminated array of strings, each of which is a
+ * slave control name. All controls with these names are assigned to
+ * the new virtual master control.
+ *
+ * This function returns zero if successful or a negative error code.
+ */
int snd_hda_add_vmaster(struct hda_codec *codec, char *name,
unsigned int *tlv, const char **slaves)
{
@@ -1643,7 +1886,7 @@ int snd_hda_add_vmaster(struct hda_codec *codec, char *name,
kctl = snd_ctl_make_virtual_master(name, tlv);
if (!kctl)
return -ENOMEM;
- err = snd_hda_ctl_add(codec, kctl);
+ err = snd_hda_ctl_add(codec, 0, kctl);
if (err < 0)
return err;
@@ -1668,7 +1911,12 @@ int snd_hda_add_vmaster(struct hda_codec *codec, char *name,
}
EXPORT_SYMBOL_HDA(snd_hda_add_vmaster);
-/* switch */
+/**
+ * snd_hda_mixer_amp_switch_info - Info callback for a standard AMP mixer switch
+ *
+ * The control element is supposed to have the private_value field
+ * set up via HDA_COMPOSE_AMP_VAL*() or related macros.
+ */
int snd_hda_mixer_amp_switch_info(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
@@ -1682,6 +1930,12 @@ int snd_hda_mixer_amp_switch_info(struct snd_kcontrol *kcontrol,
}
EXPORT_SYMBOL_HDA(snd_hda_mixer_amp_switch_info);
+/**
+ * snd_hda_mixer_amp_switch_get - Get callback for a standard AMP mixer switch
+ *
+ * The control element is supposed to have the private_value field
+ * set up via HDA_COMPOSE_AMP_VAL*() or related macros.
+ */
int snd_hda_mixer_amp_switch_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
@@ -1702,6 +1956,12 @@ int snd_hda_mixer_amp_switch_get(struct snd_kcontrol *kcontrol,
}
EXPORT_SYMBOL_HDA(snd_hda_mixer_amp_switch_get);
+/**
+ * snd_hda_mixer_amp_switch_put - Put callback for a standard AMP mixer switch
+ *
+ * The control element is supposed to have the private_value field
+ * set up via HDA_COMPOSE_AMP_VAL*() or related macros.
+ */
int snd_hda_mixer_amp_switch_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
@@ -1733,6 +1993,25 @@ int snd_hda_mixer_amp_switch_put(struct snd_kcontrol *kcontrol,
}
EXPORT_SYMBOL_HDA(snd_hda_mixer_amp_switch_put);
+#ifdef CONFIG_SND_HDA_INPUT_BEEP
+/**
+ * snd_hda_mixer_amp_switch_put_beep - Put callback for a beep AMP switch
+ *
+ * This function calls snd_hda_enable_beep_device(), which behaves differently
+ * depending on beep_mode option.
+ */
+int snd_hda_mixer_amp_switch_put_beep(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ long *valp = ucontrol->value.integer.value;
+
+ snd_hda_enable_beep_device(codec, *valp);
+ return snd_hda_mixer_amp_switch_put(kcontrol, ucontrol);
+}
+EXPORT_SYMBOL_HDA(snd_hda_mixer_amp_switch_put_beep);
+#endif /* CONFIG_SND_HDA_INPUT_BEEP */
+
/*
* bound volume controls
*
@@ -1742,6 +2021,12 @@ EXPORT_SYMBOL_HDA(snd_hda_mixer_amp_switch_put);
#define AMP_VAL_IDX_SHIFT 19
#define AMP_VAL_IDX_MASK (0x0f<<19)
+/**
+ * snd_hda_mixer_bind_switch_get - Get callback for a bound volume control
+ *
+ * The control element is supposed to have the private_value field
+ * set up via HDA_BIND_MUTE*() macros.
+ */
int snd_hda_mixer_bind_switch_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
@@ -1759,6 +2044,12 @@ int snd_hda_mixer_bind_switch_get(struct snd_kcontrol *kcontrol,
}
EXPORT_SYMBOL_HDA(snd_hda_mixer_bind_switch_get);
+/**
+ * snd_hda_mixer_bind_switch_put - Put callback for a bound volume control
+ *
+ * The control element is supposed to have the private_value field
+ * set up via HDA_BIND_MUTE*() macros.
+ */
int snd_hda_mixer_bind_switch_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
@@ -1783,8 +2074,11 @@ int snd_hda_mixer_bind_switch_put(struct snd_kcontrol *kcontrol,
}
EXPORT_SYMBOL_HDA(snd_hda_mixer_bind_switch_put);
-/*
- * generic bound volume/swtich controls
+/**
+ * snd_hda_mixer_bind_ctls_info - Info callback for a generic bound control
+ *
+ * The control element is supposed to have the private_value field
+ * set up via HDA_BIND_VOL() or HDA_BIND_SW() macros.
*/
int snd_hda_mixer_bind_ctls_info(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
@@ -1803,6 +2097,12 @@ int snd_hda_mixer_bind_ctls_info(struct snd_kcontrol *kcontrol,
}
EXPORT_SYMBOL_HDA(snd_hda_mixer_bind_ctls_info);
+/**
+ * snd_hda_mixer_bind_ctls_get - Get callback for a generic bound control
+ *
+ * The control element is supposed to have the private_value field
+ * set up via HDA_BIND_VOL() or HDA_BIND_SW() macros.
+ */
int snd_hda_mixer_bind_ctls_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
@@ -1820,6 +2120,12 @@ int snd_hda_mixer_bind_ctls_get(struct snd_kcontrol *kcontrol,
}
EXPORT_SYMBOL_HDA(snd_hda_mixer_bind_ctls_get);
+/**
+ * snd_hda_mixer_bind_ctls_put - Put callback for a generic bound control
+ *
+ * The control element is supposed to have the private_value field
+ * set up via HDA_BIND_VOL() or HDA_BIND_SW() macros.
+ */
int snd_hda_mixer_bind_ctls_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
@@ -1843,6 +2149,12 @@ int snd_hda_mixer_bind_ctls_put(struct snd_kcontrol *kcontrol,
}
EXPORT_SYMBOL_HDA(snd_hda_mixer_bind_ctls_put);
+/**
+ * snd_hda_mixer_bind_tlv - TLV callback for a generic bound control
+ *
+ * The control element is supposed to have the private_value field
+ * set up via HDA_BIND_VOL() macro.
+ */
int snd_hda_mixer_bind_tlv(struct snd_kcontrol *kcontrol, int op_flag,
unsigned int size, unsigned int __user *tlv)
{
@@ -2126,7 +2438,7 @@ int snd_hda_create_spdif_out_ctls(struct hda_codec *codec, hda_nid_t nid)
return -ENOMEM;
kctl->id.index = idx;
kctl->private_value = nid;
- err = snd_hda_ctl_add(codec, kctl);
+ err = snd_hda_ctl_add(codec, nid, kctl);
if (err < 0)
return err;
}
@@ -2165,14 +2477,19 @@ static struct snd_kcontrol_new spdif_share_sw = {
.put = spdif_share_sw_put,
};
+/**
+ * snd_hda_create_spdif_share_sw - create Default PCM switch
+ * @codec: the HDA codec
+ * @mout: multi-out instance
+ */
int snd_hda_create_spdif_share_sw(struct hda_codec *codec,
struct hda_multi_out *mout)
{
if (!mout->dig_out_nid)
return 0;
/* ATTENTION: here mout is passed as private_data, instead of codec */
- return snd_hda_ctl_add(codec,
- snd_ctl_new1(&spdif_share_sw, mout));
+ return snd_hda_ctl_add(codec, mout->dig_out_nid,
+ snd_ctl_new1(&spdif_share_sw, mout));
}
EXPORT_SYMBOL_HDA(snd_hda_create_spdif_share_sw);
@@ -2276,7 +2593,7 @@ int snd_hda_create_spdif_in_ctls(struct hda_codec *codec, hda_nid_t nid)
if (!kctl)
return -ENOMEM;
kctl->private_value = nid;
- err = snd_hda_ctl_add(codec, kctl);
+ err = snd_hda_ctl_add(codec, nid, kctl);
if (err < 0)
return err;
}
@@ -2332,7 +2649,12 @@ int snd_hda_codec_write_cache(struct hda_codec *codec, hda_nid_t nid,
}
EXPORT_SYMBOL_HDA(snd_hda_codec_write_cache);
-/* resume the all commands from the cache */
+/**
+ * snd_hda_codec_resume_cache - Resume the all commands from the cache
+ * @codec: HD-audio codec
+ *
+ * Execute all verbs recorded in the command caches to resume.
+ */
void snd_hda_codec_resume_cache(struct hda_codec *codec)
{
struct hda_cache_head *buffer = codec->cmd_cache.buf.list;
@@ -2452,9 +2774,11 @@ static void hda_call_codec_suspend(struct hda_codec *codec)
codec->afg ? codec->afg : codec->mfg,
AC_PWRST_D3);
#ifdef CONFIG_SND_HDA_POWER_SAVE
+ snd_hda_update_power_acct(codec);
cancel_delayed_work(&codec->power_work);
codec->power_on = 0;
codec->power_transition = 0;
+ codec->power_jiffies = jiffies;
#endif
}
@@ -2756,8 +3080,12 @@ static int snd_hda_query_supported_pcm(struct hda_codec *codec, hda_nid_t nid,
}
/**
- * snd_hda_is_supported_format - check whether the given node supports
- * the format val
+ * snd_hda_is_supported_format - Check the validity of the format
+ * @codec: HD-audio codec
+ * @nid: NID to check
+ * @format: the HD-audio format value to check
+ *
+ * Check whether the given node supports the format value.
*
* Returns 1 if supported, 0 if not.
*/
@@ -2877,51 +3205,36 @@ static int set_pcm_default_values(struct hda_codec *codec,
return 0;
}
+/* global */
+const char *snd_hda_pcm_type_name[HDA_PCM_NTYPES] = {
+ "Audio", "SPDIF", "HDMI", "Modem"
+};
+
/*
* get the empty PCM device number to assign
*/
static int get_empty_pcm_device(struct hda_bus *bus, int type)
{
- static const char *dev_name[HDA_PCM_NTYPES] = {
- "Audio", "SPDIF", "HDMI", "Modem"
- };
- /* starting device index for each PCM type */
- static int dev_idx[HDA_PCM_NTYPES] = {
- [HDA_PCM_TYPE_AUDIO] = 0,
- [HDA_PCM_TYPE_SPDIF] = 1,
- [HDA_PCM_TYPE_HDMI] = 3,
- [HDA_PCM_TYPE_MODEM] = 6
+ /* audio device indices; not linear to keep compatibility */
+ static int audio_idx[HDA_PCM_NTYPES][5] = {
+ [HDA_PCM_TYPE_AUDIO] = { 0, 2, 4, 5, -1 },
+ [HDA_PCM_TYPE_SPDIF] = { 1, -1 },
+ [HDA_PCM_TYPE_HDMI] = { 3, 7, 8, 9, -1 },
+ [HDA_PCM_TYPE_MODEM] = { 6, -1 },
};
- /* normal audio device indices; not linear to keep compatibility */
- static int audio_idx[4] = { 0, 2, 4, 5 };
- int i, dev;
-
- switch (type) {
- case HDA_PCM_TYPE_AUDIO:
- for (i = 0; i < ARRAY_SIZE(audio_idx); i++) {
- dev = audio_idx[i];
- if (!test_bit(dev, bus->pcm_dev_bits))
- goto ok;
- }
- snd_printk(KERN_WARNING "Too many audio devices\n");
- return -EAGAIN;
- case HDA_PCM_TYPE_SPDIF:
- case HDA_PCM_TYPE_HDMI:
- case HDA_PCM_TYPE_MODEM:
- dev = dev_idx[type];
- if (test_bit(dev, bus->pcm_dev_bits)) {
- snd_printk(KERN_WARNING "%s already defined\n",
- dev_name[type]);
- return -EAGAIN;
- }
- break;
- default:
+ int i;
+
+ if (type >= HDA_PCM_NTYPES) {
snd_printk(KERN_WARNING "Invalid PCM type %d\n", type);
return -EINVAL;
}
- ok:
- set_bit(dev, bus->pcm_dev_bits);
- return dev;
+
+ for (i = 0; audio_idx[type][i] >= 0 ; i++)
+ if (!test_and_set_bit(audio_idx[type][i], bus->pcm_dev_bits))
+ return audio_idx[type][i];
+
+ snd_printk(KERN_WARNING "Too many %s devices\n", snd_hda_pcm_type_name[type]);
+ return -EAGAIN;
}
/*
@@ -3159,14 +3472,14 @@ EXPORT_SYMBOL_HDA(snd_hda_check_board_codec_sid_config);
*/
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;
kctl = snd_ctl_new1(knew, codec);
if (!kctl)
return -ENOMEM;
- err = snd_hda_ctl_add(codec, kctl);
+ err = snd_hda_ctl_add(codec, 0, kctl);
if (err < 0) {
if (!codec->addr)
return err;
@@ -3174,7 +3487,7 @@ int snd_hda_add_new_ctls(struct hda_codec *codec, struct snd_kcontrol_new *knew)
if (!kctl)
return -ENOMEM;
kctl->id.device = codec->addr;
- err = snd_hda_ctl_add(codec, kctl);
+ err = snd_hda_ctl_add(codec, 0, kctl);
if (err < 0)
return err;
}
@@ -3207,8 +3520,27 @@ static void hda_keep_power_on(struct hda_codec *codec)
{
codec->power_count++;
codec->power_on = 1;
+ codec->power_jiffies = jiffies;
}
+/* update the power on/off account with the current jiffies */
+void snd_hda_update_power_acct(struct hda_codec *codec)
+{
+ unsigned long delta = jiffies - codec->power_jiffies;
+ if (codec->power_on)
+ codec->power_on_acct += delta;
+ else
+ codec->power_off_acct += delta;
+ codec->power_jiffies += delta;
+}
+
+/**
+ * snd_hda_power_up - Power-up the codec
+ * @codec: HD-audio codec
+ *
+ * Increment the power-up counter and power up the hardware really when
+ * not turned on yet.
+ */
void snd_hda_power_up(struct hda_codec *codec)
{
struct hda_bus *bus = codec->bus;
@@ -3217,7 +3549,9 @@ void snd_hda_power_up(struct hda_codec *codec)
if (codec->power_on || codec->power_transition)
return;
+ snd_hda_update_power_acct(codec);
codec->power_on = 1;
+ codec->power_jiffies = jiffies;
if (bus->ops.pm_notify)
bus->ops.pm_notify(bus);
hda_call_codec_resume(codec);
@@ -3229,9 +3563,13 @@ EXPORT_SYMBOL_HDA(snd_hda_power_up);
#define power_save(codec) \
((codec)->bus->power_save ? *(codec)->bus->power_save : 0)
-#define power_save(codec) \
- ((codec)->bus->power_save ? *(codec)->bus->power_save : 0)
-
+/**
+ * snd_hda_power_down - Power-down the codec
+ * @codec: HD-audio codec
+ *
+ * Decrement the power-up counter and schedules the power-off work if
+ * the counter rearches to zero.
+ */
void snd_hda_power_down(struct hda_codec *codec)
{
--codec->power_count;
@@ -3245,6 +3583,19 @@ void snd_hda_power_down(struct hda_codec *codec)
}
EXPORT_SYMBOL_HDA(snd_hda_power_down);
+/**
+ * snd_hda_check_amp_list_power - Check the amp list and update the power
+ * @codec: HD-audio codec
+ * @check: the object containing an AMP list and the status
+ * @nid: NID to check / update
+ *
+ * Check whether the given NID is in the amp list. If it's in the list,
+ * check the current AMP status, and update the the power-status according
+ * to the mute status.
+ *
+ * This function is supposed to be set or called from the check_power_status
+ * patch ops.
+ */
int snd_hda_check_amp_list_power(struct hda_codec *codec,
struct hda_loopback_check *check,
hda_nid_t nid)
@@ -3286,6 +3637,10 @@ EXPORT_SYMBOL_HDA(snd_hda_check_amp_list_power);
/*
* Channel mode helper
*/
+
+/**
+ * snd_hda_ch_mode_info - Info callback helper for the channel mode enum
+ */
int snd_hda_ch_mode_info(struct hda_codec *codec,
struct snd_ctl_elem_info *uinfo,
const struct hda_channel_mode *chmode,
@@ -3302,6 +3657,9 @@ int snd_hda_ch_mode_info(struct hda_codec *codec,
}
EXPORT_SYMBOL_HDA(snd_hda_ch_mode_info);
+/**
+ * snd_hda_ch_mode_get - Get callback helper for the channel mode enum
+ */
int snd_hda_ch_mode_get(struct hda_codec *codec,
struct snd_ctl_elem_value *ucontrol,
const struct hda_channel_mode *chmode,
@@ -3320,6 +3678,9 @@ int snd_hda_ch_mode_get(struct hda_codec *codec,
}
EXPORT_SYMBOL_HDA(snd_hda_ch_mode_get);
+/**
+ * snd_hda_ch_mode_put - Put callback helper for the channel mode enum
+ */
int snd_hda_ch_mode_put(struct hda_codec *codec,
struct snd_ctl_elem_value *ucontrol,
const struct hda_channel_mode *chmode,
@@ -3344,6 +3705,10 @@ EXPORT_SYMBOL_HDA(snd_hda_ch_mode_put);
/*
* input MUX helper
*/
+
+/**
+ * snd_hda_input_mux_info_info - Info callback helper for the input-mux enum
+ */
int snd_hda_input_mux_info(const struct hda_input_mux *imux,
struct snd_ctl_elem_info *uinfo)
{
@@ -3362,6 +3727,9 @@ int snd_hda_input_mux_info(const struct hda_input_mux *imux,
}
EXPORT_SYMBOL_HDA(snd_hda_input_mux_info);
+/**
+ * snd_hda_input_mux_info_put - Put callback helper for the input-mux enum
+ */
int snd_hda_input_mux_put(struct hda_codec *codec,
const struct hda_input_mux *imux,
struct snd_ctl_elem_value *ucontrol,
@@ -3421,8 +3789,29 @@ static void cleanup_dig_out_stream(struct hda_codec *codec, hda_nid_t nid)
}
}
-/*
- * open the digital out in the exclusive mode
+/**
+ * snd_hda_bus_reboot_notify - call the reboot notifier of each codec
+ * @bus: HD-audio bus
+ */
+void snd_hda_bus_reboot_notify(struct hda_bus *bus)
+{
+ struct hda_codec *codec;
+
+ if (!bus)
+ return;
+ list_for_each_entry(codec, &bus->codec_list, list) {
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ if (!codec->power_on)
+ continue;
+#endif
+ if (codec->patch_ops.reboot_notify)
+ codec->patch_ops.reboot_notify(codec);
+ }
+}
+EXPORT_SYMBOL_HDA(snd_hda_bus_reboot_notify);
+
+/**
+ * snd_hda_multi_out_dig_open - open the digital out in the exclusive mode
*/
int snd_hda_multi_out_dig_open(struct hda_codec *codec,
struct hda_multi_out *mout)
@@ -3437,6 +3826,9 @@ int snd_hda_multi_out_dig_open(struct hda_codec *codec,
}
EXPORT_SYMBOL_HDA(snd_hda_multi_out_dig_open);
+/**
+ * snd_hda_multi_out_dig_prepare - prepare the digital out stream
+ */
int snd_hda_multi_out_dig_prepare(struct hda_codec *codec,
struct hda_multi_out *mout,
unsigned int stream_tag,
@@ -3450,6 +3842,9 @@ int snd_hda_multi_out_dig_prepare(struct hda_codec *codec,
}
EXPORT_SYMBOL_HDA(snd_hda_multi_out_dig_prepare);
+/**
+ * snd_hda_multi_out_dig_cleanup - clean-up the digital out stream
+ */
int snd_hda_multi_out_dig_cleanup(struct hda_codec *codec,
struct hda_multi_out *mout)
{
@@ -3460,8 +3855,8 @@ int snd_hda_multi_out_dig_cleanup(struct hda_codec *codec,
}
EXPORT_SYMBOL_HDA(snd_hda_multi_out_dig_cleanup);
-/*
- * release the digital out
+/**
+ * snd_hda_multi_out_dig_close - release the digital out stream
*/
int snd_hda_multi_out_dig_close(struct hda_codec *codec,
struct hda_multi_out *mout)
@@ -3473,8 +3868,12 @@ int snd_hda_multi_out_dig_close(struct hda_codec *codec,
}
EXPORT_SYMBOL_HDA(snd_hda_multi_out_dig_close);
-/*
- * set up more restrictions for analog out
+/**
+ * snd_hda_multi_out_analog_open - open analog outputs
+ *
+ * Open analog outputs and set up the hw-constraints.
+ * If the digital outputs can be opened as slave, open the digital
+ * outputs, too.
*/
int snd_hda_multi_out_analog_open(struct hda_codec *codec,
struct hda_multi_out *mout,
@@ -3519,9 +3918,11 @@ int snd_hda_multi_out_analog_open(struct hda_codec *codec,
}
EXPORT_SYMBOL_HDA(snd_hda_multi_out_analog_open);
-/*
- * set up the i/o for analog out
- * when the digital out is available, copy the front out to digital out, too.
+/**
+ * snd_hda_multi_out_analog_prepare - Preapre the analog outputs.
+ *
+ * Set up the i/o for analog out.
+ * When the digital out is available, copy the front out to digital out, too.
*/
int snd_hda_multi_out_analog_prepare(struct hda_codec *codec,
struct hda_multi_out *mout,
@@ -3578,8 +3979,8 @@ int snd_hda_multi_out_analog_prepare(struct hda_codec *codec,
}
EXPORT_SYMBOL_HDA(snd_hda_multi_out_analog_prepare);
-/*
- * clean up the setting for analog out
+/**
+ * snd_hda_multi_out_analog_cleanup - clean up the setting for analog out
*/
int snd_hda_multi_out_analog_cleanup(struct hda_codec *codec,
struct hda_multi_out *mout)
@@ -3965,8 +4366,14 @@ EXPORT_SYMBOL_HDA(snd_hda_resume);
* generic arrays
*/
-/* get a new element from the given array
- * if it exceeds the pre-allocated array size, re-allocate the array
+/**
+ * snd_array_new - get a new element from the given array
+ * @array: the array object
+ *
+ * Get a new element from the given array. If it exceeds the
+ * pre-allocated array size, re-allocate the array.
+ *
+ * Returns NULL if allocation failed.
*/
void *snd_array_new(struct snd_array *array)
{
@@ -3990,7 +4397,10 @@ void *snd_array_new(struct snd_array *array)
}
EXPORT_SYMBOL_HDA(snd_array_new);
-/* free the given array elements */
+/**
+ * snd_array_free - free the given array elements
+ * @array: the array object
+ */
void snd_array_free(struct snd_array *array)
{
kfree(array->list);
@@ -4000,7 +4410,12 @@ void snd_array_free(struct snd_array *array)
}
EXPORT_SYMBOL_HDA(snd_array_free);
-/*
+/**
+ * snd_print_pcm_rates - Print the supported PCM rates to the string buffer
+ * @pcm: PCM caps bits
+ * @buf: the string buffer to write
+ * @buflen: the max buffer length
+ *
* used by hda_proc.c and hda_eld.c
*/
void snd_print_pcm_rates(int pcm, char *buf, int buflen)
@@ -4019,6 +4434,14 @@ void snd_print_pcm_rates(int pcm, char *buf, int buflen)
}
EXPORT_SYMBOL_HDA(snd_print_pcm_rates);
+/**
+ * snd_print_pcm_bits - Print the supported PCM fmt bits to the string buffer
+ * @pcm: PCM caps bits
+ * @buf: the string buffer to write
+ * @buflen: the max buffer length
+ *
+ * used by hda_proc.c and hda_eld.c
+ */
void snd_print_pcm_bits(int pcm, char *buf, int buflen)
{
static unsigned int bits[] = { 8, 16, 20, 24, 32 };
diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h
index 99552fb5f75..2d627613aea 100644
--- a/sound/pci/hda/hda_codec.h
+++ b/sound/pci/hda/hda_codec.h
@@ -286,6 +286,10 @@ enum {
#define AC_PWRST_D1SUP (1<<1)
#define AC_PWRST_D2SUP (1<<2)
#define AC_PWRST_D3SUP (1<<3)
+#define AC_PWRST_D3COLDSUP (1<<4)
+#define AC_PWRST_S3D3COLDSUP (1<<29)
+#define AC_PWRST_CLKSTOP (1<<30)
+#define AC_PWRST_EPSS (1U<<31)
/* Power state values */
#define AC_PWRST_SETTING (0xf<<0)
@@ -674,6 +678,7 @@ struct hda_codec_ops {
#ifdef CONFIG_SND_HDA_POWER_SAVE
int (*check_power_status)(struct hda_codec *codec, hda_nid_t nid);
#endif
+ void (*reboot_notify)(struct hda_codec *codec);
};
/* record for amp information cache */
@@ -771,6 +776,7 @@ struct hda_codec {
/* beep device */
struct hda_beep *beep;
+ unsigned int beep_mode;
/* widget capabilities cache */
unsigned int num_nodes;
@@ -811,6 +817,9 @@ struct hda_codec {
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 */
+ unsigned long power_on_acct;
+ unsigned long power_off_acct;
+ unsigned long power_jiffies;
#endif
/* codec-specific additional proc output */
@@ -910,6 +919,7 @@ int snd_hda_is_supported_format(struct hda_codec *codec, hda_nid_t nid,
* Misc
*/
void snd_hda_get_codec_name(struct hda_codec *codec, char *name, int namelen);
+void snd_hda_bus_reboot_notify(struct hda_bus *bus);
/*
* power management
@@ -933,6 +943,7 @@ const char *snd_hda_get_jack_location(u32 cfg);
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
+void snd_hda_update_power_acct(struct hda_codec *codec);
#else
static inline void snd_hda_power_up(struct hda_codec *codec) {}
static inline void snd_hda_power_down(struct hda_codec *codec) {}
diff --git a/sound/pci/hda/hda_eld.c b/sound/pci/hda/hda_eld.c
index 9446a5abea1..4228f2fe595 100644
--- a/sound/pci/hda/hda_eld.c
+++ b/sound/pci/hda/hda_eld.c
@@ -309,17 +309,12 @@ out_fail:
return -EINVAL;
}
-static int hdmi_present_sense(struct hda_codec *codec, hda_nid_t nid)
-{
- return snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_PIN_SENSE, 0);
-}
-
static int hdmi_eld_valid(struct hda_codec *codec, hda_nid_t nid)
{
int eldv;
int present;
- present = hdmi_present_sense(codec, nid);
+ present = snd_hda_pin_sense(codec, nid);
eldv = (present & AC_PINSENSE_ELDV);
present = (present & AC_PINSENSE_PRESENCE);
@@ -477,6 +472,8 @@ static void hdmi_print_eld_info(struct snd_info_entry *entry,
[4 ... 7] = "reserved"
};
+ snd_iprintf(buffer, "monitor_present\t\t%d\n", e->monitor_present);
+ snd_iprintf(buffer, "eld_valid\t\t%d\n", e->eld_valid);
snd_iprintf(buffer, "monitor_name\t\t%s\n", e->monitor_name);
snd_iprintf(buffer, "connection_type\t\t%s\n",
eld_connection_type_names[e->conn_type]);
@@ -518,7 +515,11 @@ static void hdmi_write_eld_info(struct snd_info_entry *entry,
* monitor_name manufacture_id product_id
* eld_version edid_version
*/
- if (!strcmp(name, "connection_type"))
+ if (!strcmp(name, "monitor_present"))
+ e->monitor_present = val;
+ else if (!strcmp(name, "eld_valid"))
+ e->eld_valid = val;
+ else if (!strcmp(name, "connection_type"))
e->conn_type = val;
else if (!strcmp(name, "port_id"))
e->port_id = val;
@@ -560,13 +561,14 @@ static void hdmi_write_eld_info(struct snd_info_entry *entry,
}
-int snd_hda_eld_proc_new(struct hda_codec *codec, struct hdmi_eld *eld)
+int snd_hda_eld_proc_new(struct hda_codec *codec, struct hdmi_eld *eld,
+ int index)
{
char name[32];
struct snd_info_entry *entry;
int err;
- snprintf(name, sizeof(name), "eld#%d", codec->addr);
+ snprintf(name, sizeof(name), "eld#%d.%d", codec->addr, index);
err = snd_card_proc_new(codec->bus->card, name, &entry);
if (err < 0)
return err;
diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c
index b36f6c5a92d..092c6a7c2ff 100644
--- a/sound/pci/hda/hda_generic.c
+++ b/sound/pci/hda/hda_generic.c
@@ -727,7 +727,8 @@ static int create_mixer(struct hda_codec *codec, struct hda_gnode *node,
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);
- err = snd_hda_ctl_add(codec, snd_ctl_new1(&knew, codec));
+ err = snd_hda_ctl_add(codec, node->nid,
+ snd_ctl_new1(&knew, codec));
if (err < 0)
return err;
created = 1;
@@ -737,7 +738,8 @@ static int create_mixer(struct hda_codec *codec, struct hda_gnode *node,
if (is_loopback)
add_input_loopback(codec, node->nid, HDA_OUTPUT, 0);
snd_printdd("[%s] NID=0x%x, DIR=OUT\n", name, node->nid);
- err = snd_hda_ctl_add(codec, snd_ctl_new1(&knew, codec));
+ err = snd_hda_ctl_add(codec, node->nid,
+ snd_ctl_new1(&knew, codec));
if (err < 0)
return err;
created = 1;
@@ -751,7 +753,8 @@ static int create_mixer(struct hda_codec *codec, struct hda_gnode *node,
(node->amp_in_caps & AC_AMPCAP_NUM_STEPS)) {
knew = (struct snd_kcontrol_new)HDA_CODEC_VOLUME(name, node->nid, index, HDA_INPUT);
snd_printdd("[%s] NID=0x%x, DIR=IN, IDX=0x%x\n", name, node->nid, index);
- err = snd_hda_ctl_add(codec, snd_ctl_new1(&knew, codec));
+ err = snd_hda_ctl_add(codec, node->nid,
+ snd_ctl_new1(&knew, codec));
if (err < 0)
return err;
created = 1;
@@ -759,7 +762,8 @@ static int create_mixer(struct hda_codec *codec, struct hda_gnode *node,
(node->amp_out_caps & AC_AMPCAP_NUM_STEPS)) {
knew = (struct snd_kcontrol_new)HDA_CODEC_VOLUME(name, node->nid, 0, HDA_OUTPUT);
snd_printdd("[%s] NID=0x%x, DIR=OUT\n", name, node->nid);
- err = snd_hda_ctl_add(codec, snd_ctl_new1(&knew, codec));
+ err = snd_hda_ctl_add(codec, node->nid,
+ snd_ctl_new1(&knew, codec));
if (err < 0)
return err;
created = 1;
@@ -857,7 +861,7 @@ static int build_input_controls(struct hda_codec *codec)
}
/* create input MUX if multiple sources are available */
- err = snd_hda_ctl_add(codec, snd_ctl_new1(&cap_sel, codec));
+ err = snd_hda_ctl_add(codec, 0, snd_ctl_new1(&cap_sel, codec));
if (err < 0)
return err;
@@ -875,7 +879,8 @@ static int build_input_controls(struct hda_codec *codec)
HDA_CODEC_VOLUME(name, adc_node->nid,
spec->input_mux.items[i].index,
HDA_INPUT);
- err = snd_hda_ctl_add(codec, snd_ctl_new1(&knew, codec));
+ err = snd_hda_ctl_add(codec, adc_node->nid,
+ snd_ctl_new1(&knew, codec));
if (err < 0)
return err;
}
diff --git a/sound/pci/hda/hda_hwdep.c b/sound/pci/hda/hda_hwdep.c
index cc24e6721d7..d24328661c6 100644
--- a/sound/pci/hda/hda_hwdep.c
+++ b/sound/pci/hda/hda_hwdep.c
@@ -154,6 +154,44 @@ int /*__devinit*/ snd_hda_create_hwdep(struct hda_codec *codec)
return 0;
}
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+static ssize_t power_on_acct_show(struct device *dev,
+ struct device_attribute *attr,
+ char *buf)
+{
+ struct snd_hwdep *hwdep = dev_get_drvdata(dev);
+ struct hda_codec *codec = hwdep->private_data;
+ snd_hda_update_power_acct(codec);
+ return sprintf(buf, "%u\n", jiffies_to_msecs(codec->power_on_acct));
+}
+
+static ssize_t power_off_acct_show(struct device *dev,
+ struct device_attribute *attr,
+ char *buf)
+{
+ struct snd_hwdep *hwdep = dev_get_drvdata(dev);
+ struct hda_codec *codec = hwdep->private_data;
+ snd_hda_update_power_acct(codec);
+ return sprintf(buf, "%u\n", jiffies_to_msecs(codec->power_off_acct));
+}
+
+static struct device_attribute power_attrs[] = {
+ __ATTR_RO(power_on_acct),
+ __ATTR_RO(power_off_acct),
+};
+
+int snd_hda_hwdep_add_power_sysfs(struct hda_codec *codec)
+{
+ struct snd_hwdep *hwdep = codec->hwdep;
+ int i;
+
+ for (i = 0; i < ARRAY_SIZE(power_attrs); i++)
+ snd_add_device_sysfs_file(SNDRV_DEVICE_TYPE_HWDEP, hwdep->card,
+ hwdep->device, &power_attrs[i]);
+ return 0;
+}
+#endif /* CONFIG_SND_HDA_POWER_SAVE */
+
#ifdef CONFIG_SND_HDA_RECONFIG
/*
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 20a66f85f0a..d822bfc6cad 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -60,10 +60,14 @@ static int bdl_pos_adj[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS-1)] = -1};
static int probe_mask[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS-1)] = -1};
static int probe_only[SNDRV_CARDS];
static int single_cmd;
-static int enable_msi;
+static int enable_msi = -1;
#ifdef CONFIG_SND_HDA_PATCH_LOADER
static char *patch[SNDRV_CARDS];
#endif
+#ifdef CONFIG_SND_HDA_INPUT_BEEP
+static int beep_mode[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS-1)] =
+ CONFIG_SND_HDA_INPUT_BEEP_MODE};
+#endif
module_param_array(index, int, NULL, 0444);
MODULE_PARM_DESC(index, "Index value for Intel HD audio interface.");
@@ -91,6 +95,11 @@ MODULE_PARM_DESC(enable_msi, "Enable Message Signaled Interrupt (MSI)");
module_param_array(patch, charp, NULL, 0444);
MODULE_PARM_DESC(patch, "Patch file for Intel HD audio interface.");
#endif
+#ifdef CONFIG_SND_HDA_INPUT_BEEP
+module_param_array(beep_mode, int, NULL, 0444);
+MODULE_PARM_DESC(beep_mode, "Select HDA Beep registration mode "
+ "(0=off, 1=on, 2=mute switch on/off) (default=1).");
+#endif
#ifdef CONFIG_SND_HDA_POWER_SAVE
static int power_save = CONFIG_SND_HDA_POWER_SAVE_DEFAULT;
@@ -404,6 +413,7 @@ struct azx {
unsigned short codec_mask;
int codec_probe_mask; /* copied from probe_mask option */
struct hda_bus *bus;
+ unsigned int beep_mode;
/* CORB/RIRB */
struct azx_rb corb;
@@ -677,6 +687,14 @@ static unsigned int azx_rirb_get_response(struct hda_bus *bus,
}
}
+ if (!chip->polling_mode) {
+ snd_printk(KERN_WARNING SFX "azx_get_response timeout, "
+ "switching to polling mode: last cmd=0x%08x\n",
+ chip->last_cmd[addr]);
+ chip->polling_mode = 1;
+ goto again;
+ }
+
if (chip->msi) {
snd_printk(KERN_WARNING SFX "No response from codec, "
"disabling MSI: last cmd=0x%08x\n",
@@ -692,14 +710,6 @@ static unsigned int azx_rirb_get_response(struct hda_bus *bus,
goto again;
}
- if (!chip->polling_mode) {
- snd_printk(KERN_WARNING SFX "azx_get_response timeout, "
- "switching to polling mode: last cmd=0x%08x\n",
- chip->last_cmd[addr]);
- chip->polling_mode = 1;
- goto again;
- }
-
if (chip->probing) {
/* If this critical timeout happens during the codec probing
* phase, this is likely an access to a non-existing codec
@@ -722,9 +732,10 @@ static unsigned int azx_rirb_get_response(struct hda_bus *bus,
chip->last_cmd[addr]);
chip->single_cmd = 1;
bus->response_reset = 0;
- /* re-initialize CORB/RIRB */
+ /* release CORB/RIRB */
azx_free_cmd_io(chip);
- azx_init_cmd_io(chip);
+ /* disable unsolicited responses */
+ azx_writel(chip, GCTL, azx_readl(chip, GCTL) & ~ICH6_GCTL_UNSOL);
return -1;
}
@@ -865,7 +876,9 @@ static int azx_reset(struct azx *chip)
}
/* Accept unsolicited responses */
- azx_writel(chip, GCTL, azx_readl(chip, GCTL) | ICH6_GCTL_UNSOL);
+ if (!chip->single_cmd)
+ azx_writel(chip, GCTL, azx_readl(chip, GCTL) |
+ ICH6_GCTL_UNSOL);
/* detect codecs */
if (!chip->codec_mask) {
@@ -980,7 +993,8 @@ static void azx_init_chip(struct azx *chip)
azx_int_enable(chip);
/* initialize the codec command I/O */
- azx_init_cmd_io(chip);
+ if (!chip->single_cmd)
+ azx_init_cmd_io(chip);
/* program the position buffer */
azx_writel(chip, DPLBASE, (u32)chip->posbuf.addr);
@@ -1400,6 +1414,7 @@ static int __devinit azx_codec_create(struct azx *chip, const char *model)
err = snd_hda_codec_new(chip->bus, c, &codec);
if (err < 0)
continue;
+ codec->beep_mode = chip->beep_mode;
codecs++;
}
}
@@ -2150,6 +2165,7 @@ static int azx_resume(struct pci_dev *pci)
static int azx_halt(struct notifier_block *nb, unsigned long event, void *buf)
{
struct azx *chip = container_of(nb, struct azx, reboot_notifier);
+ snd_hda_bus_reboot_notify(chip->bus);
azx_stop_chip(chip);
return NOTIFY_OK;
}
@@ -2217,7 +2233,9 @@ 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_LPIB),
SND_PCI_QUIRK(0x1028, 0x01de, "Dell Precision 390", POS_FIX_LPIB),
+ SND_PCI_QUIRK(0x103c, 0x306d, "HP dv3", POS_FIX_LPIB),
SND_PCI_QUIRK(0x1043, 0x813d, "ASUS P5AD2", POS_FIX_LPIB),
+ SND_PCI_QUIRK(0x1462, 0x1002, "MSI Wind U115", POS_FIX_LPIB),
{}
};
@@ -2300,10 +2318,9 @@ static void __devinit check_probe_mask(struct azx *chip, int dev)
}
/*
- * white-list for enable_msi
+ * white/black-list for enable_msi
*/
-static struct snd_pci_quirk msi_white_list[] __devinitdata = {
- SND_PCI_QUIRK(0x103c, 0x3607, "HP Compa CQ40", 1),
+static struct snd_pci_quirk msi_black_list[] __devinitdata = {
{}
};
@@ -2311,10 +2328,12 @@ static void __devinit check_msi(struct azx *chip)
{
const struct snd_pci_quirk *q;
- chip->msi = enable_msi;
- if (chip->msi)
+ if (enable_msi >= 0) {
+ chip->msi = !!enable_msi;
return;
- q = snd_pci_quirk_lookup(chip->pci, msi_white_list);
+ }
+ chip->msi = 1; /* enable MSI as default */
+ q = snd_pci_quirk_lookup(chip->pci, msi_black_list);
if (q) {
printk(KERN_INFO
"hda_intel: msi for device %04x:%04x set to %d\n",
@@ -2573,6 +2592,10 @@ static int __devinit azx_probe(struct pci_dev *pci,
goto out_free;
card->private_data = chip;
+#ifdef CONFIG_SND_HDA_INPUT_BEEP
+ chip->beep_mode = beep_mode[dev];
+#endif
+
/* create codec instances */
err = azx_codec_create(chip, model[dev]);
if (err < 0)
@@ -2673,6 +2696,7 @@ static struct pci_device_id azx_ids[] = {
{ PCI_DEVICE(0x10de, 0x044b), .driver_data = AZX_DRIVER_NVIDIA },
{ PCI_DEVICE(0x10de, 0x055c), .driver_data = AZX_DRIVER_NVIDIA },
{ PCI_DEVICE(0x10de, 0x055d), .driver_data = AZX_DRIVER_NVIDIA },
+ { PCI_DEVICE(0x10de, 0x0590), .driver_data = AZX_DRIVER_NVIDIA },
{ PCI_DEVICE(0x10de, 0x0774), .driver_data = AZX_DRIVER_NVIDIA },
{ PCI_DEVICE(0x10de, 0x0775), .driver_data = AZX_DRIVER_NVIDIA },
{ PCI_DEVICE(0x10de, 0x0776), .driver_data = AZX_DRIVER_NVIDIA },
diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h
index 5f1dcc59002..5778ae882b8 100644
--- a/sound/pci/hda/hda_local.h
+++ b/sound/pci/hda/hda_local.h
@@ -23,6 +23,15 @@
#ifndef __SOUND_HDA_LOCAL_H
#define __SOUND_HDA_LOCAL_H
+/* We abuse kcontrol_new.subdev field to pass the NID corresponding to
+ * the given new control. If id.subdev has a bit flag HDA_SUBDEV_NID_FLAG,
+ * snd_hda_ctl_add() takes the lower-bit subdev value as a valid NID.
+ *
+ * Note that the subdevice field is cleared again before the real registration
+ * in snd_hda_ctl_add(), so that this value won't appear in the outside.
+ */
+#define HDA_SUBDEV_NID_FLAG (1U << 31)
+
/*
* for mixer controls
*/
@@ -33,6 +42,7 @@
/* 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, \
+ .subdevice = HDA_SUBDEV_NID_FLAG | (nid), \
.access = SNDRV_CTL_ELEM_ACCESS_READWRITE | \
SNDRV_CTL_ELEM_ACCESS_TLV_READ | \
SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK, \
@@ -53,6 +63,7 @@
/* mono mute switch with index (index=0,1,...) (channel=1,2) */
#define HDA_CODEC_MUTE_MONO_IDX(xname, xcidx, nid, channel, xindex, direction) \
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xcidx, \
+ .subdevice = HDA_SUBDEV_NID_FLAG | (nid), \
.info = snd_hda_mixer_amp_switch_info, \
.get = snd_hda_mixer_amp_switch_get, \
.put = snd_hda_mixer_amp_switch_put, \
@@ -66,6 +77,28 @@
/* stereo mute switch */
#define HDA_CODEC_MUTE(xname, nid, xindex, direction) \
HDA_CODEC_MUTE_MONO(xname, nid, 3, xindex, direction)
+#ifdef CONFIG_SND_HDA_INPUT_BEEP
+/* special beep mono mute switch with index (index=0,1,...) (channel=1,2) */
+#define HDA_CODEC_MUTE_BEEP_MONO_IDX(xname, xcidx, nid, channel, xindex, direction) \
+ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xcidx, \
+ .subdevice = HDA_SUBDEV_NID_FLAG | (nid), \
+ .info = snd_hda_mixer_amp_switch_info, \
+ .get = snd_hda_mixer_amp_switch_get, \
+ .put = snd_hda_mixer_amp_switch_put_beep, \
+ .private_value = HDA_COMPOSE_AMP_VAL(nid, channel, xindex, direction) }
+#else
+/* no digital beep - just the standard one */
+#define HDA_CODEC_MUTE_BEEP_MONO_IDX(xname, xcidx, nid, ch, xidx, dir) \
+ HDA_CODEC_MUTE_MONO_IDX(xname, xcidx, nid, ch, xidx, dir)
+#endif /* CONFIG_SND_HDA_INPUT_BEEP */
+/* special beep mono mute switch */
+#define HDA_CODEC_MUTE_BEEP_MONO(xname, nid, channel, xindex, direction) \
+ HDA_CODEC_MUTE_BEEP_MONO_IDX(xname, 0, nid, channel, xindex, direction)
+/* special beep stereo mute switch */
+#define HDA_CODEC_MUTE_BEEP(xname, nid, xindex, direction) \
+ HDA_CODEC_MUTE_BEEP_MONO(xname, nid, 3, xindex, direction)
+
+extern const char *snd_hda_pcm_type_name[];
int snd_hda_mixer_amp_volume_info(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo);
@@ -81,6 +114,10 @@ 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);
+#ifdef CONFIG_SND_HDA_INPUT_BEEP
+int snd_hda_mixer_amp_switch_put_beep(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol);
+#endif
/* 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);
@@ -424,8 +461,16 @@ u32 query_amp_caps(struct hda_codec *codec, hda_nid_t nid, int direction);
int snd_hda_override_amp_caps(struct hda_codec *codec, hda_nid_t nid, int dir,
unsigned int caps);
u32 snd_hda_query_pin_caps(struct hda_codec *codec, hda_nid_t nid);
+u32 snd_hda_pin_sense(struct hda_codec *codec, hda_nid_t nid);
+int snd_hda_jack_detect(struct hda_codec *codec, hda_nid_t nid);
-int snd_hda_ctl_add(struct hda_codec *codec, struct snd_kcontrol *kctl);
+struct hda_nid_item {
+ struct snd_kcontrol *kctl;
+ hda_nid_t nid;
+};
+
+int snd_hda_ctl_add(struct hda_codec *codec, hda_nid_t nid,
+ struct snd_kcontrol *kctl);
void snd_hda_ctls_clear(struct hda_codec *codec);
/*
@@ -437,6 +482,15 @@ int snd_hda_create_hwdep(struct hda_codec *codec);
static inline int snd_hda_create_hwdep(struct hda_codec *codec) { return 0; }
#endif
+#if defined(CONFIG_SND_HDA_POWER_SAVE) && defined(CONFIG_SND_HDA_HWDEP)
+int snd_hda_hwdep_add_power_sysfs(struct hda_codec *codec);
+#else
+static inline int snd_hda_hwdep_add_power_sysfs(struct hda_codec *codec)
+{
+ return 0;
+}
+#endif
+
#ifdef CONFIG_SND_HDA_RECONFIG
int snd_hda_hwdep_add_sysfs(struct hda_codec *codec);
#else
@@ -490,7 +544,8 @@ int snd_hda_check_amp_list_power(struct hda_codec *codec,
* AMP control callbacks
*/
/* retrieve parameters from private_value */
-#define get_amp_nid(kc) ((kc)->private_value & 0xffff)
+#define get_amp_nid_(pv) ((pv) & 0xffff)
+#define get_amp_nid(kc) get_amp_nid_((kc)->private_value)
#define get_amp_channels(kc) (((kc)->private_value >> 16) & 0x3)
#define get_amp_direction(kc) (((kc)->private_value >> 18) & 0x1)
#define get_amp_index(kc) (((kc)->private_value >> 19) & 0xf)
@@ -516,9 +571,11 @@ struct cea_sad {
* ELD: EDID Like Data
*/
struct hdmi_eld {
+ bool monitor_present;
+ bool eld_valid;
int eld_size;
int baseline_len;
- int eld_ver; /* (eld_ver == 0) indicates invalid ELD */
+ int eld_ver;
int cea_edid_ver;
char monitor_name[ELD_MAX_MNL + 1];
int manufacture_id;
@@ -541,11 +598,13 @@ int snd_hdmi_get_eld(struct hdmi_eld *, struct hda_codec *, hda_nid_t);
void snd_hdmi_show_eld(struct hdmi_eld *eld);
#ifdef CONFIG_PROC_FS
-int snd_hda_eld_proc_new(struct hda_codec *codec, struct hdmi_eld *eld);
+int snd_hda_eld_proc_new(struct hda_codec *codec, struct hdmi_eld *eld,
+ int index);
void snd_hda_eld_proc_free(struct hda_codec *codec, struct hdmi_eld *eld);
#else
static inline int snd_hda_eld_proc_new(struct hda_codec *codec,
- struct hdmi_eld *eld)
+ struct hdmi_eld *eld,
+ int index)
{
return 0;
}
diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c
index 95f24e4729f..09476fc1ab6 100644
--- a/sound/pci/hda/hda_proc.c
+++ b/sound/pci/hda/hda_proc.c
@@ -26,6 +26,21 @@
#include "hda_codec.h"
#include "hda_local.h"
+static char *bits_names(unsigned int bits, char *names[], int size)
+{
+ int i, n;
+ static char buf[128];
+
+ for (i = 0, n = 0; i < size; i++) {
+ if (bits & (1U<<i) && names[i])
+ n += snprintf(buf + n, sizeof(buf) - n, " %s",
+ names[i]);
+ }
+ buf[n] = '\0';
+
+ return buf;
+}
+
static const char *get_wid_type_name(unsigned int wid_value)
{
static char *names[16] = {
@@ -46,6 +61,41 @@ static const char *get_wid_type_name(unsigned int wid_value)
return "UNKNOWN Widget";
}
+static void print_nid_mixers(struct snd_info_buffer *buffer,
+ struct hda_codec *codec, hda_nid_t nid)
+{
+ int i;
+ struct hda_nid_item *items = codec->mixers.list;
+ struct snd_kcontrol *kctl;
+ for (i = 0; i < codec->mixers.used; i++) {
+ if (items[i].nid == nid) {
+ kctl = items[i].kctl;
+ snd_iprintf(buffer,
+ " Control: name=\"%s\", index=%i, device=%i\n",
+ kctl->id.name, kctl->id.index, kctl->id.device);
+ }
+ }
+}
+
+static void print_nid_pcms(struct snd_info_buffer *buffer,
+ struct hda_codec *codec, hda_nid_t nid)
+{
+ int pcm, type;
+ struct hda_pcm *cpcm;
+ for (pcm = 0; pcm < codec->num_pcms; pcm++) {
+ cpcm = &codec->pcm_info[pcm];
+ for (type = 0; type < 2; type++) {
+ if (cpcm->stream[type].nid != nid || cpcm->pcm == NULL)
+ continue;
+ snd_iprintf(buffer, " Device: name=\"%s\", "
+ "type=\"%s\", device=%i\n",
+ cpcm->name,
+ snd_hda_pcm_type_name[cpcm->pcm_type],
+ cpcm->pcm->device);
+ }
+ }
+}
+
static void print_amp_caps(struct snd_info_buffer *buffer,
struct hda_codec *codec, hda_nid_t nid, int dir)
{
@@ -363,8 +413,24 @@ static const char *get_pwr_state(u32 state)
static void print_power_state(struct snd_info_buffer *buffer,
struct hda_codec *codec, hda_nid_t nid)
{
+ static char *names[] = {
+ [ilog2(AC_PWRST_D0SUP)] = "D0",
+ [ilog2(AC_PWRST_D1SUP)] = "D1",
+ [ilog2(AC_PWRST_D2SUP)] = "D2",
+ [ilog2(AC_PWRST_D3SUP)] = "D3",
+ [ilog2(AC_PWRST_D3COLDSUP)] = "D3cold",
+ [ilog2(AC_PWRST_S3D3COLDSUP)] = "S3D3cold",
+ [ilog2(AC_PWRST_CLKSTOP)] = "CLKSTOP",
+ [ilog2(AC_PWRST_EPSS)] = "EPSS",
+ };
+
+ int sup = snd_hda_param_read(codec, nid, AC_PAR_POWER_STATE);
int pwr = snd_hda_codec_read(codec, nid, 0,
AC_VERB_GET_POWER_STATE, 0);
+ if (sup)
+ snd_iprintf(buffer, " Power states: %s\n",
+ bits_names(sup, names, ARRAY_SIZE(names)));
+
snd_iprintf(buffer, " Power: setting=%s, actual=%s\n",
get_pwr_state(pwr & AC_PWRST_SETTING),
get_pwr_state((pwr & AC_PWRST_ACTUAL) >>
@@ -457,6 +523,7 @@ static void print_gpio(struct snd_info_buffer *buffer,
(data & (1<<i)) ? 1 : 0,
(unsol & (1<<i)) ? 1 : 0);
/* FIXME: add GPO and GPI pin information */
+ print_nid_mixers(buffer, codec, nid);
}
static void print_codec_info(struct snd_info_entry *entry,
@@ -536,6 +603,9 @@ static void print_codec_info(struct snd_info_entry *entry,
snd_iprintf(buffer, " CP");
snd_iprintf(buffer, "\n");
+ print_nid_mixers(buffer, codec, nid);
+ print_nid_pcms(buffer, codec, nid);
+
/* volume knob is a special widget that always have connection
* list
*/
diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index 215e72a8711..455a0494f90 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -156,15 +156,19 @@ static const char *ad_slave_sws[] = {
static void ad198x_free_kctls(struct hda_codec *codec);
+#ifdef CONFIG_SND_HDA_INPUT_BEEP
/* additional beep mixers; the actual parameters are overwritten at build */
static struct snd_kcontrol_new ad_beep_mixer[] = {
HDA_CODEC_VOLUME("Beep Playback Volume", 0, 0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Beep Playback Switch", 0, 0, HDA_OUTPUT),
+ HDA_CODEC_MUTE_BEEP("Beep Playback Switch", 0, 0, HDA_OUTPUT),
{ } /* end */
};
#define set_beep_amp(spec, nid, idx, dir) \
((spec)->beep_amp = HDA_COMPOSE_AMP_VAL(nid, 1, idx, dir)) /* mono */
+#else
+#define set_beep_amp(spec, nid, idx, dir) /* NOP */
+#endif
static int ad198x_build_controls(struct hda_codec *codec)
{
@@ -194,6 +198,7 @@ static int ad198x_build_controls(struct hda_codec *codec)
}
/* create beep controls if needed */
+#ifdef CONFIG_SND_HDA_INPUT_BEEP
if (spec->beep_amp) {
struct snd_kcontrol_new *knew;
for (knew = ad_beep_mixer; knew->name; knew++) {
@@ -202,11 +207,14 @@ static int ad198x_build_controls(struct hda_codec *codec)
if (!kctl)
return -ENOMEM;
kctl->private_value = spec->beep_amp;
- err = snd_hda_ctl_add(codec, kctl);
+ err = snd_hda_ctl_add(codec,
+ get_amp_nid_(spec->beep_amp),
+ kctl);
if (err < 0)
return err;
}
}
+#endif
/* if we have no master control, let's create it */
if (!snd_hda_find_mixer_ctl(codec, "Master Playback Volume")) {
@@ -712,10 +720,10 @@ static struct snd_kcontrol_new ad1986a_laptop_intmic_mixers[] = {
static void ad1986a_automic(struct hda_codec *codec)
{
unsigned int present;
- present = snd_hda_codec_read(codec, 0x1f, 0, AC_VERB_GET_PIN_SENSE, 0);
+ present = snd_hda_jack_detect(codec, 0x1f);
/* 0 = 0x1f, 2 = 0x1d, 4 = mixed */
snd_hda_codec_write(codec, 0x0f, 0, AC_VERB_SET_CONNECT_SEL,
- (present & AC_PINSENSE_PRESENCE) ? 0 : 2);
+ present ? 0 : 2);
}
#define AD1986A_MIC_EVENT 0x36
@@ -754,10 +762,8 @@ static void ad1986a_update_hp(struct hda_codec *codec)
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);
+ spec->jack_present = snd_hda_jack_detect(codec, 0x1a);
if (spec->inv_jack_detect)
spec->jack_present = !spec->jack_present;
ad1986a_update_hp(codec);
@@ -1547,8 +1553,7 @@ static void ad1981_hp_automute(struct hda_codec *codec)
{
unsigned int present;
- present = snd_hda_codec_read(codec, 0x06, 0,
- AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
+ present = snd_hda_jack_detect(codec, 0x06);
snd_hda_codec_amp_stereo(codec, 0x05, HDA_OUTPUT, 0,
HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
}
@@ -1568,8 +1573,7 @@ static void ad1981_hp_automic(struct hda_codec *codec)
};
unsigned int present;
- present = snd_hda_codec_read(codec, 0x08, 0,
- AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
+ present = snd_hda_jack_detect(codec, 0x08);
if (present)
snd_hda_sequence_write(codec, mic_jack_on);
else
@@ -2524,7 +2528,7 @@ static void ad1988_laptop_unsol_event(struct hda_codec *codec, unsigned int res)
{
if ((res >> 26) != AD1988_HP_EVENT)
return;
- if (snd_hda_codec_read(codec, 0x11, 0, AC_VERB_GET_PIN_SENSE, 0) & (1 << 31))
+ if (snd_hda_jack_detect(codec, 0x11))
snd_hda_sequence_write(codec, ad1988_laptop_hp_on);
else
snd_hda_sequence_write(codec, ad1988_laptop_hp_off);
@@ -2569,6 +2573,8 @@ static int add_control(struct ad198x_spec *spec, int type, const char *name,
knew->name = kstrdup(name, GFP_KERNEL);
if (! knew->name)
return -ENOMEM;
+ if (get_amp_nid_(val))
+ knew->subdevice = HDA_SUBDEV_NID_FLAG | get_amp_nid_(val);
knew->private_value = val;
return 0;
}
@@ -3768,8 +3774,7 @@ static void ad1884a_hp_automute(struct hda_codec *codec)
{
unsigned int present;
- present = snd_hda_codec_read(codec, 0x11, 0,
- AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
+ present = snd_hda_jack_detect(codec, 0x11);
snd_hda_codec_amp_stereo(codec, 0x16, HDA_OUTPUT, 0,
HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
snd_hda_codec_write(codec, 0x16, 0, AC_VERB_SET_EAPD_BTLENABLE,
@@ -3781,8 +3786,7 @@ static void ad1884a_hp_automic(struct hda_codec *codec)
{
unsigned int present;
- present = snd_hda_codec_read(codec, 0x14, 0,
- AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
+ present = snd_hda_jack_detect(codec, 0x14);
snd_hda_codec_write(codec, 0x0c, 0, AC_VERB_SET_CONNECT_SEL,
present ? 0 : 1);
}
@@ -3817,13 +3821,9 @@ static void ad1884a_laptop_automute(struct hda_codec *codec)
{
unsigned int present;
- present = snd_hda_codec_read(codec, 0x11, 0, AC_VERB_GET_PIN_SENSE, 0);
- present &= AC_PINSENSE_PRESENCE;
- if (!present) {
- present = snd_hda_codec_read(codec, 0x12, 0,
- AC_VERB_GET_PIN_SENSE, 0);
- present &= AC_PINSENSE_PRESENCE;
- }
+ present = snd_hda_jack_detect(codec, 0x11);
+ if (!present)
+ present = snd_hda_jack_detect(codec, 0x12);
snd_hda_codec_amp_stereo(codec, 0x16, HDA_OUTPUT, 0,
HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
snd_hda_codec_write(codec, 0x16, 0, AC_VERB_SET_EAPD_BTLENABLE,
@@ -3835,11 +3835,9 @@ static void ad1884a_laptop_automic(struct hda_codec *codec)
{
unsigned int idx;
- if (snd_hda_codec_read(codec, 0x14, 0, AC_VERB_GET_PIN_SENSE, 0) &
- AC_PINSENSE_PRESENCE)
+ if (snd_hda_jack_detect(codec, 0x14))
idx = 0;
- else if (snd_hda_codec_read(codec, 0x1c, 0, AC_VERB_GET_PIN_SENSE, 0) &
- AC_PINSENSE_PRESENCE)
+ else if (snd_hda_jack_detect(codec, 0x1c))
idx = 4;
else
idx = 1;
@@ -4008,8 +4006,7 @@ static void ad1984a_thinkpad_automute(struct hda_codec *codec)
{
unsigned int present;
- present = snd_hda_codec_read(codec, 0x11, 0, AC_VERB_GET_PIN_SENSE, 0)
- & AC_PINSENSE_PRESENCE;
+ present = snd_hda_jack_detect(codec, 0x11);
snd_hda_codec_amp_stereo(codec, 0x12, HDA_OUTPUT, 0,
HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
}
@@ -4032,6 +4029,125 @@ static int ad1984a_thinkpad_init(struct hda_codec *codec)
}
/*
+ * HP Touchsmart
+ * port-A (0x11) - front hp-out
+ * port-B (0x14) - unused
+ * port-C (0x15) - unused
+ * port-D (0x12) - rear line out
+ * port-E (0x1c) - front mic-in
+ * port-F (0x16) - Internal speakers
+ * digital-mic (0x17) - Internal mic
+ */
+
+static struct hda_verb ad1984a_touchsmart_verbs[] = {
+ /* DACs; unmute as default */
+ {0x03, AC_VERB_SET_AMP_GAIN_MUTE, 0x27}, /* 0dB */
+ {0x04, AC_VERB_SET_AMP_GAIN_MUTE, 0x27}, /* 0dB */
+ /* Port-A (HP) mixer - route only from analog mixer */
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ /* Port-A pin */
+ {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ /* Port-A (HP) pin - always unmuted */
+ {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ /* Port-E (int speaker) mixer - route only from analog mixer */
+ {0x25, AC_VERB_SET_AMP_GAIN_MUTE, 0x03},
+ /* Port-E pin */
+ {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ /* Port-F (int speaker) mixer - route only from analog mixer */
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ /* Port-F pin */
+ {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ /* Analog mixer; mute as default */
+ {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+ {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+ {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
+ {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)},
+ /* Analog Mix output amp */
+ {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ /* capture sources */
+ /* {0x0c, AC_VERB_SET_CONNECT_SEL, 0x0}, */ /* set via unsol */
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ {0x0d, AC_VERB_SET_CONNECT_SEL, 0x0},
+ {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ /* unsolicited event for pin-sense */
+ {0x11, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_HP_EVENT},
+ {0x1c, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_MIC_EVENT},
+ /* allow to touch GPIO1 (for mute control) */
+ {0x01, AC_VERB_SET_GPIO_MASK, 0x02},
+ {0x01, AC_VERB_SET_GPIO_DIRECTION, 0x02},
+ {0x01, AC_VERB_SET_GPIO_DATA, 0x02}, /* first muted */
+ /* internal mic - dmic */
+ {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ /* set magic COEFs for dmic */
+ {0x01, AC_VERB_SET_COEF_INDEX, 0x13f7},
+ {0x01, AC_VERB_SET_PROC_COEF, 0x08},
+ { } /* end */
+};
+
+static struct snd_kcontrol_new ad1984a_touchsmart_mixers[] = {
+ HDA_CODEC_VOLUME("Master Playback Volume", 0x21, 0x0, HDA_OUTPUT),
+/* HDA_CODEC_MUTE("Master Playback Switch", 0x21, 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 = ad1884a_mobile_master_sw_put,
+ .private_value = HDA_COMPOSE_AMP_VAL(0x21, 3, 0, HDA_OUTPUT),
+ },
+ HDA_CODEC_VOLUME("PCM Playback Volume", 0x20, 0x5, HDA_INPUT),
+ HDA_CODEC_MUTE("PCM Playback Switch", 0x20, 0x5, HDA_INPUT),
+ HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Mic Boost", 0x25, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Internal Mic Boost", 0x17, 0x0, HDA_INPUT),
+ { } /* end */
+};
+
+/* switch to external mic if plugged */
+static void ad1984a_touchsmart_automic(struct hda_codec *codec)
+{
+ if (snd_hda_jack_detect(codec, 0x1c))
+ snd_hda_codec_write(codec, 0x0c, 0,
+ AC_VERB_SET_CONNECT_SEL, 0x4);
+ else
+ snd_hda_codec_write(codec, 0x0c, 0,
+ AC_VERB_SET_CONNECT_SEL, 0x5);
+}
+
+
+/* unsolicited event for HP jack sensing */
+static void ad1984a_touchsmart_unsol_event(struct hda_codec *codec,
+ unsigned int res)
+{
+ switch (res >> 26) {
+ case AD1884A_HP_EVENT:
+ ad1884a_hp_automute(codec);
+ break;
+ case AD1884A_MIC_EVENT:
+ ad1984a_touchsmart_automic(codec);
+ break;
+ }
+}
+
+/* initialize jack-sensing, too */
+static int ad1984a_touchsmart_init(struct hda_codec *codec)
+{
+ ad198x_init(codec);
+ ad1884a_hp_automute(codec);
+ ad1984a_touchsmart_automic(codec);
+ return 0;
+}
+
+
+/*
*/
enum {
@@ -4039,6 +4155,7 @@ enum {
AD1884A_LAPTOP,
AD1884A_MOBILE,
AD1884A_THINKPAD,
+ AD1984A_TOUCHSMART,
AD1884A_MODELS
};
@@ -4047,6 +4164,7 @@ static const char *ad1884a_models[AD1884A_MODELS] = {
[AD1884A_LAPTOP] = "laptop",
[AD1884A_MOBILE] = "mobile",
[AD1884A_THINKPAD] = "thinkpad",
+ [AD1984A_TOUCHSMART] = "touchsmart",
};
static struct snd_pci_quirk ad1884a_cfg_tbl[] = {
@@ -4059,6 +4177,7 @@ static struct snd_pci_quirk ad1884a_cfg_tbl[] = {
SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x3600, "HP laptop", AD1884A_LAPTOP),
SND_PCI_QUIRK_MASK(0x103c, 0xfff0, 0x7010, "HP laptop", AD1884A_MOBILE),
SND_PCI_QUIRK(0x17aa, 0x20ac, "Thinkpad X300", AD1884A_THINKPAD),
+ SND_PCI_QUIRK(0x103c, 0x2a82, "Touchsmart", AD1984A_TOUCHSMART),
{}
};
@@ -4142,6 +4261,21 @@ static int patch_ad1884a(struct hda_codec *codec)
codec->patch_ops.unsol_event = ad1984a_thinkpad_unsol_event;
codec->patch_ops.init = ad1984a_thinkpad_init;
break;
+ case AD1984A_TOUCHSMART:
+ spec->mixers[0] = ad1984a_touchsmart_mixers;
+ spec->init_verbs[0] = ad1984a_touchsmart_verbs;
+ spec->multiout.dig_out_nid = 0;
+ codec->patch_ops.unsol_event = ad1984a_touchsmart_unsol_event;
+ codec->patch_ops.init = ad1984a_touchsmart_init;
+ /* set the upper-limit for mixer amp to 0dB for avoiding the
+ * possible damage by overloading
+ */
+ snd_hda_override_amp_caps(codec, 0x20, HDA_INPUT,
+ (0x17 << AC_AMPCAP_OFFSET_SHIFT) |
+ (0x17 << AC_AMPCAP_NUM_STEPS_SHIFT) |
+ (0x05 << AC_AMPCAP_STEP_SIZE_SHIFT) |
+ (1 << AC_AMPCAP_MUTE_SHIFT));
+ break;
}
return 0;
diff --git a/sound/pci/hda/patch_ca0110.c b/sound/pci/hda/patch_ca0110.c
index d08353d3bb7..af478019088 100644
--- a/sound/pci/hda/patch_ca0110.c
+++ b/sound/pci/hda/patch_ca0110.c
@@ -144,7 +144,7 @@ static int _add_switch(struct hda_codec *codec, hda_nid_t nid, const char *pfx,
struct snd_kcontrol_new knew =
HDA_CODEC_MUTE_MONO(namestr, nid, chan, 0, type);
sprintf(namestr, "%s %s Switch", pfx, dirstr[dir]);
- return snd_hda_ctl_add(codec, snd_ctl_new1(&knew, codec));
+ return snd_hda_ctl_add(codec, nid, snd_ctl_new1(&knew, codec));
}
static int _add_volume(struct hda_codec *codec, hda_nid_t nid, const char *pfx,
@@ -155,7 +155,7 @@ static int _add_volume(struct hda_codec *codec, hda_nid_t nid, const char *pfx,
struct snd_kcontrol_new knew =
HDA_CODEC_VOLUME_MONO(namestr, nid, chan, 0, type);
sprintf(namestr, "%s %s Volume", pfx, dirstr[dir]);
- return snd_hda_ctl_add(codec, snd_ctl_new1(&knew, codec));
+ return snd_hda_ctl_add(codec, nid, snd_ctl_new1(&knew, codec));
}
#define add_out_switch(codec, nid, pfx) _add_switch(codec, nid, pfx, 3, 0)
diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c
index 8ba306856d3..2439e84dcb2 100644
--- a/sound/pci/hda/patch_cirrus.c
+++ b/sound/pci/hda/patch_cirrus.c
@@ -500,7 +500,7 @@ static int add_mute(struct hda_codec *codec, const char *name, int index,
knew.private_value = pval;
snprintf(tmp, sizeof(tmp), "%s %s Switch", name, dir_sfx[dir]);
*kctlp = snd_ctl_new1(&knew, codec);
- return snd_hda_ctl_add(codec, *kctlp);
+ return snd_hda_ctl_add(codec, get_amp_nid_(pval), *kctlp);
}
static int add_volume(struct hda_codec *codec, const char *name,
@@ -513,7 +513,7 @@ static int add_volume(struct hda_codec *codec, const char *name,
knew.private_value = pval;
snprintf(tmp, sizeof(tmp), "%s %s Volume", name, dir_sfx[dir]);
*kctlp = snd_ctl_new1(&knew, codec);
- return snd_hda_ctl_add(codec, *kctlp);
+ return snd_hda_ctl_add(codec, get_amp_nid_(pval), *kctlp);
}
static void fix_volume_caps(struct hda_codec *codec, hda_nid_t dac)
@@ -536,14 +536,14 @@ static int add_vmaster(struct hda_codec *codec, hda_nid_t dac)
spec->vmaster_sw =
snd_ctl_make_virtual_master("Master Playback Switch", NULL);
- err = snd_hda_ctl_add(codec, spec->vmaster_sw);
+ err = snd_hda_ctl_add(codec, dac, spec->vmaster_sw);
if (err < 0)
return err;
snd_hda_set_vmaster_tlv(codec, dac, HDA_OUTPUT, tlv);
spec->vmaster_vol =
snd_ctl_make_virtual_master("Master Playback Volume", tlv);
- err = snd_hda_ctl_add(codec, spec->vmaster_vol);
+ err = snd_hda_ctl_add(codec, dac, spec->vmaster_vol);
if (err < 0)
return err;
return 0;
@@ -756,13 +756,13 @@ static int build_input(struct hda_codec *codec)
if (!kctl)
return -ENOMEM;
kctl->private_value = (long)spec->capture_bind[i];
- err = snd_hda_ctl_add(codec, kctl);
+ err = snd_hda_ctl_add(codec, 0, kctl);
if (err < 0)
return err;
}
if (spec->num_inputs > 1 && !spec->mic_detect) {
- err = snd_hda_ctl_add(codec,
+ err = snd_hda_ctl_add(codec, 0,
snd_ctl_new1(&cs_capture_source, codec));
if (err < 0)
return err;
@@ -807,7 +807,7 @@ static void cs_automute(struct hda_codec *codec)
{
struct cs_spec *spec = codec->spec;
struct auto_pin_cfg *cfg = &spec->autocfg;
- unsigned int caps, present, hp_present;
+ unsigned int caps, hp_present;
hda_nid_t nid;
int i;
@@ -817,12 +817,7 @@ static void cs_automute(struct hda_codec *codec)
caps = snd_hda_query_pin_caps(codec, nid);
if (!(caps & AC_PINCAP_PRES_DETECT))
continue;
- if (caps & AC_PINCAP_TRIG_REQ)
- snd_hda_codec_read(codec, nid, 0,
- AC_VERB_SET_PIN_SENSE, 0);
- present = snd_hda_codec_read(codec, nid, 0,
- AC_VERB_GET_PIN_SENSE, 0);
- hp_present |= (present & AC_PINSENSE_PRESENCE) != 0;
+ hp_present = snd_hda_jack_detect(codec, nid);
if (hp_present)
break;
}
@@ -844,15 +839,11 @@ static void cs_automic(struct hda_codec *codec)
struct cs_spec *spec = codec->spec;
struct auto_pin_cfg *cfg = &spec->autocfg;
hda_nid_t nid;
- unsigned int caps, present;
+ unsigned int present;
nid = cfg->input_pins[spec->automic_idx];
- caps = snd_hda_query_pin_caps(codec, nid);
- if (caps & AC_PINCAP_TRIG_REQ)
- snd_hda_codec_read(codec, nid, 0, AC_VERB_SET_PIN_SENSE, 0);
- present = snd_hda_codec_read(codec, nid, 0,
- AC_VERB_GET_PIN_SENSE, 0);
- if (present & AC_PINSENSE_PRESENCE)
+ present = snd_hda_jack_detect(codec, nid);
+ if (present)
change_cur_input(codec, spec->automic_idx, 0);
else {
unsigned int imic = (spec->automic_idx == AUTO_PIN_MIC) ?
diff --git a/sound/pci/hda/patch_cmedia.c b/sound/pci/hda/patch_cmedia.c
index 780e1a72114..85c81feb10c 100644
--- a/sound/pci/hda/patch_cmedia.c
+++ b/sound/pci/hda/patch_cmedia.c
@@ -197,8 +197,8 @@ static struct snd_kcontrol_new cmi9880_basic_mixer[] = {
HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x09, 0, HDA_INPUT),
HDA_CODEC_MUTE("Capture Switch", 0x08, 0, HDA_INPUT),
HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x09, 0, HDA_INPUT),
- HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x23, 0, HDA_OUTPUT),
- HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x23, 0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Beep Playback Volume", 0x23, 0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Beep Playback Switch", 0x23, 0, HDA_OUTPUT),
{ } /* end */
};
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 9d899eda44d..a09c03c3f62 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -110,6 +110,7 @@ struct conexant_spec {
unsigned int dell_automute;
unsigned int port_d_mode;
+ unsigned char ext_mic_bias;
};
static int conexant_playback_pcm_open(struct hda_pcm_stream *hinfo,
@@ -396,9 +397,7 @@ static void conexant_report_jack(struct hda_codec *codec, hda_nid_t nid)
for (i = 0; i < spec->jacks.used; i++) {
if (jacks->nid == nid) {
unsigned int present;
- present = snd_hda_codec_read(codec, nid, 0,
- AC_VERB_GET_PIN_SENSE, 0) &
- AC_PINSENSE_PRESENCE;
+ present = snd_hda_jack_detect(codec, nid);
present = (present) ? jacks->type : 0 ;
@@ -682,11 +681,13 @@ static struct hda_input_mux cxt5045_capture_source = {
};
static struct hda_input_mux cxt5045_capture_source_benq = {
- .num_items = 3,
+ .num_items = 5,
.items = {
{ "IntMic", 0x1 },
{ "ExtMic", 0x2 },
{ "LineIn", 0x3 },
+ { "CD", 0x4 },
+ { "Mixer", 0x0 },
}
};
@@ -747,8 +748,7 @@ static void cxt5045_hp_automic(struct hda_codec *codec)
};
unsigned int present;
- present = snd_hda_codec_read(codec, 0x12, 0,
- AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
+ present = snd_hda_jack_detect(codec, 0x12);
if (present)
snd_hda_sequence_write(codec, mic_jack_on);
else
@@ -762,8 +762,7 @@ static void cxt5045_hp_automute(struct hda_codec *codec)
struct conexant_spec *spec = codec->spec;
unsigned int bits;
- spec->hp_present = snd_hda_codec_read(codec, 0x11, 0,
- AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
+ spec->hp_present = snd_hda_jack_detect(codec, 0x11);
bits = (spec->hp_present || !spec->cur_eapd) ? HDA_AMP_MUTE : 0;
snd_hda_codec_amp_stereo(codec, 0x10, HDA_OUTPUT, 0,
@@ -811,11 +810,19 @@ static struct snd_kcontrol_new cxt5045_mixers[] = {
};
static struct snd_kcontrol_new cxt5045_benq_mixers[] = {
+ HDA_CODEC_VOLUME("CD Capture Volume", 0x1a, 0x04, HDA_INPUT),
+ HDA_CODEC_MUTE("CD Capture Switch", 0x1a, 0x04, HDA_INPUT),
+ HDA_CODEC_VOLUME("CD Playback Volume", 0x17, 0x4, HDA_INPUT),
+ HDA_CODEC_MUTE("CD Playback Switch", 0x17, 0x4, HDA_INPUT),
+
HDA_CODEC_VOLUME("Line In Capture Volume", 0x1a, 0x03, HDA_INPUT),
HDA_CODEC_MUTE("Line In Capture Switch", 0x1a, 0x03, HDA_INPUT),
HDA_CODEC_VOLUME("Line In Playback Volume", 0x17, 0x3, HDA_INPUT),
HDA_CODEC_MUTE("Line In Playback Switch", 0x17, 0x3, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mixer Capture Volume", 0x1a, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mixer Capture Switch", 0x1a, 0x0, HDA_INPUT),
+
{}
};
@@ -1164,9 +1171,10 @@ static int patch_cxt5045(struct hda_codec *codec)
switch (codec->subsystem_id >> 16) {
case 0x103c:
- /* HP laptop has a really bad sound over 0dB on NID 0x17.
- * Fix max PCM level to 0 dB
- * (originall it has 0x2b steps with 0dB offset 0x14)
+ case 0x1734:
+ /* HP & Fujitsu-Siemens laptops have really bad sound over 0dB
+ * on NID 0x17. Fix max PCM level to 0 dB
+ * (originally it has 0x2b steps with 0dB offset 0x14)
*/
snd_hda_override_amp_caps(codec, 0x17, HDA_INPUT,
(0x14 << AC_AMPCAP_OFFSET_SHIFT) |
@@ -1232,8 +1240,7 @@ static void cxt5047_hp_automute(struct hda_codec *codec)
struct conexant_spec *spec = codec->spec;
unsigned int bits;
- spec->hp_present = snd_hda_codec_read(codec, 0x13, 0,
- AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
+ spec->hp_present = snd_hda_jack_detect(codec, 0x13);
bits = (spec->hp_present || !spec->cur_eapd) ? HDA_AMP_MUTE : 0;
/* See the note in cxt5047_hp_master_sw_put */
@@ -1256,8 +1263,7 @@ static void cxt5047_hp_automic(struct hda_codec *codec)
};
unsigned int present;
- present = snd_hda_codec_read(codec, 0x15, 0,
- AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
+ present = snd_hda_jack_detect(codec, 0x15);
if (present)
snd_hda_sequence_write(codec, mic_jack_on);
else
@@ -1404,16 +1410,7 @@ static struct snd_kcontrol_new cxt5047_test_mixer[] = {
.get = conexant_mux_enum_get,
.put = conexant_mux_enum_put,
},
- HDA_CODEC_VOLUME("Input-1 Volume", 0x1a, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("Input-1 Switch", 0x1a, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Input-2 Volume", 0x1a, 0x1, HDA_INPUT),
- HDA_CODEC_MUTE("Input-2 Switch", 0x1a, 0x1, HDA_INPUT),
- HDA_CODEC_VOLUME("Input-3 Volume", 0x1a, 0x2, HDA_INPUT),
- HDA_CODEC_MUTE("Input-3 Switch", 0x1a, 0x2, HDA_INPUT),
- HDA_CODEC_VOLUME("Input-4 Volume", 0x1a, 0x3, HDA_INPUT),
- HDA_CODEC_MUTE("Input-4 Switch", 0x1a, 0x3, HDA_INPUT),
- HDA_CODEC_VOLUME("Input-5 Volume", 0x1a, 0x4, HDA_INPUT),
- HDA_CODEC_MUTE("Input-5 Switch", 0x1a, 0x4, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Boost Volume", 0x1a, 0x0, HDA_OUTPUT),
{ } /* end */
};
@@ -1610,9 +1607,7 @@ static void cxt5051_portb_automic(struct hda_codec *codec)
if (spec->no_auto_mic)
return;
- present = snd_hda_codec_read(codec, 0x17, 0,
- AC_VERB_GET_PIN_SENSE, 0) &
- AC_PINSENSE_PRESENCE;
+ present = snd_hda_jack_detect(codec, 0x17);
snd_hda_codec_write(codec, 0x14, 0,
AC_VERB_SET_CONNECT_SEL,
present ? 0x01 : 0x00);
@@ -1627,9 +1622,7 @@ static void cxt5051_portc_automic(struct hda_codec *codec)
if (spec->no_auto_mic)
return;
- present = snd_hda_codec_read(codec, 0x18, 0,
- AC_VERB_GET_PIN_SENSE, 0) &
- AC_PINSENSE_PRESENCE;
+ present = snd_hda_jack_detect(codec, 0x18);
if (present)
spec->cur_adc_idx = 1;
else
@@ -1650,9 +1643,7 @@ static void cxt5051_hp_automute(struct hda_codec *codec)
{
struct conexant_spec *spec = codec->spec;
- spec->hp_present = snd_hda_codec_read(codec, 0x16, 0,
- AC_VERB_GET_PIN_SENSE, 0) &
- AC_PINSENSE_PRESENCE;
+ spec->hp_present = snd_hda_jack_detect(codec, 0x16);
cxt5051_update_speaker(codec);
}
@@ -1917,6 +1908,11 @@ static hda_nid_t cxt5066_adc_nids[3] = { 0x14, 0x15, 0x16 };
static hda_nid_t cxt5066_capsrc_nids[1] = { 0x17 };
#define CXT5066_SPDIF_OUT 0x21
+/* OLPC's microphone port is DC coupled for use with external sensors,
+ * therefore we use a 50% mic bias in order to center the input signal with
+ * the DC input range of the codec. */
+#define CXT5066_OLPC_EXT_MIC_BIAS PIN_VREF50
+
static struct hda_channel_mode cxt5066_modes[1] = {
{ 2, NULL },
};
@@ -1970,9 +1966,10 @@ static int cxt5066_hp_master_sw_put(struct snd_kcontrol *kcontrol,
/* toggle input of built-in and mic jack appropriately */
static void cxt5066_automic(struct hda_codec *codec)
{
- static struct hda_verb ext_mic_present[] = {
+ struct conexant_spec *spec = codec->spec;
+ struct hda_verb ext_mic_present[] = {
/* enable external mic, port B */
- {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, spec->ext_mic_bias},
/* switch to external mic input */
{0x17, AC_VERB_SET_CONNECT_SEL, 0},
@@ -1994,8 +1991,47 @@ static void cxt5066_automic(struct hda_codec *codec)
};
unsigned int present;
- present = snd_hda_codec_read(codec, 0x1a, 0,
- AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
+ present = snd_hda_jack_detect(codec, 0x1a);
+ if (present) {
+ snd_printdd("CXT5066: external microphone detected\n");
+ snd_hda_sequence_write(codec, ext_mic_present);
+ } else {
+ snd_printdd("CXT5066: external microphone absent\n");
+ snd_hda_sequence_write(codec, ext_mic_absent);
+ }
+}
+
+/* toggle input of built-in digital mic and mic jack appropriately */
+static void cxt5066_vostro_automic(struct hda_codec *codec)
+{
+ struct conexant_spec *spec = codec->spec;
+ unsigned int present;
+
+ struct hda_verb ext_mic_present[] = {
+ /* enable external mic, port B */
+ {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, spec->ext_mic_bias},
+
+ /* switch to external mic input */
+ {0x17, AC_VERB_SET_CONNECT_SEL, 0},
+ {0x14, AC_VERB_SET_CONNECT_SEL, 0},
+
+ /* disable internal digital mic */
+ {0x23, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
+ {}
+ };
+ static struct hda_verb ext_mic_absent[] = {
+ /* enable internal mic, port C */
+ {0x23, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+
+ /* switch to internal mic input */
+ {0x14, AC_VERB_SET_CONNECT_SEL, 2},
+
+ /* disable external mic, port B */
+ {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
+ {}
+ };
+
+ present = snd_hda_jack_detect(codec, 0x1a);
if (present) {
snd_printdd("CXT5066: external microphone detected\n");
snd_hda_sequence_write(codec, ext_mic_present);
@@ -2012,12 +2048,10 @@ static void cxt5066_hp_automute(struct hda_codec *codec)
unsigned int portA, portD;
/* Port A */
- portA = snd_hda_codec_read(codec, 0x19, 0, AC_VERB_GET_PIN_SENSE, 0)
- & AC_PINSENSE_PRESENCE;
+ portA = snd_hda_jack_detect(codec, 0x19);
/* Port D */
- portD = (snd_hda_codec_read(codec, 0x1c, 0, AC_VERB_GET_PIN_SENSE, 0)
- & AC_PINSENSE_PRESENCE) << 1;
+ portD = snd_hda_jack_detect(codec, 0x1c);
spec->hp_present = !!(portA | portD);
snd_printdd("CXT5066: hp automute portA=%x portD=%x present=%d\n",
@@ -2039,6 +2073,20 @@ static void cxt5066_unsol_event(struct hda_codec *codec, unsigned int res)
}
}
+/* unsolicited event for jack sensing */
+static void cxt5066_vostro_event(struct hda_codec *codec, unsigned int res)
+{
+ snd_printdd("CXT5066_vostro: unsol event %x (%x)\n", res, res >> 26);
+ switch (res >> 26) {
+ case CONEXANT_HP_EVENT:
+ cxt5066_hp_automute(codec);
+ break;
+ case CONEXANT_MIC_EVENT:
+ cxt5066_vostro_automic(codec);
+ break;
+ }
+}
+
static const struct hda_input_mux cxt5066_analog_mic_boost = {
.num_items = 5,
.items = {
@@ -2225,7 +2273,7 @@ static struct hda_verb cxt5066_init_verbs_olpc[] = {
{0x19, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC1 */
/* Port B: external microphone */
- {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, CXT5066_OLPC_EXT_MIC_BIAS},
/* Port C: internal microphone */
{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
@@ -2280,6 +2328,67 @@ static struct hda_verb cxt5066_init_verbs_olpc[] = {
{ } /* end */
};
+static struct hda_verb cxt5066_init_verbs_vostro[] = {
+ /* Port A: headphones */
+ {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
+ {0x19, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC1 */
+
+ /* Port B: external microphone */
+ {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
+
+ /* Port C: unused */
+ {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
+
+ /* Port D: unused */
+ {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
+
+ /* Port E: unused, but has primary EAPD */
+ {0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
+ {0x1d, AC_VERB_SET_EAPD_BTLENABLE, 0x2}, /* default on */
+
+ /* Port F: unused */
+ {0x1e, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
+
+ /* Port G: internal speakers */
+ {0x1f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x1f, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC1 */
+
+ /* DAC1 */
+ {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+
+ /* DAC2: unused */
+ {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+ {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+ {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+
+ /* Digital microphone port */
+ {0x23, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+
+ /* Audio input selectors */
+ {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x3},
+ {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
+
+ /* Disable SPDIF */
+ {0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
+ {0x22, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
+
+ /* enable unsolicited events for Port A and B */
+ {0x19, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_HP_EVENT},
+ {0x1a, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_MIC_EVENT},
+ { } /* end */
+};
+
static struct hda_verb cxt5066_init_verbs_portd_lo[] = {
{0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{ } /* end */
@@ -2301,6 +2410,7 @@ enum {
CXT5066_LAPTOP, /* Laptops w/ EAPD support */
CXT5066_DELL_LAPTOP, /* Dell Laptop */
CXT5066_OLPC_XO_1_5, /* OLPC XO 1.5 */
+ CXT5066_DELL_VOSTO, /* Dell Vostro 1015i */
CXT5066_MODELS
};
@@ -2308,6 +2418,7 @@ static const char *cxt5066_models[CXT5066_MODELS] = {
[CXT5066_LAPTOP] = "laptop",
[CXT5066_DELL_LAPTOP] = "dell-laptop",
[CXT5066_OLPC_XO_1_5] = "olpc-xo-1_5",
+ [CXT5066_DELL_VOSTO] = "dell-vostro"
};
static struct snd_pci_quirk cxt5066_cfg_tbl[] = {
@@ -2315,6 +2426,8 @@ static struct snd_pci_quirk cxt5066_cfg_tbl[] = {
CXT5066_LAPTOP),
SND_PCI_QUIRK(0x1028, 0x02f5, "Dell",
CXT5066_DELL_LAPTOP),
+ SND_PCI_QUIRK(0x152d, 0x0833, "OLPC XO-1.5", CXT5066_OLPC_XO_1_5),
+ SND_PCI_QUIRK(0x1028, 0x0402, "Dell Vostro", CXT5066_DELL_VOSTO),
{}
};
@@ -2342,6 +2455,7 @@ static int patch_cxt5066(struct hda_codec *codec)
spec->input_mux = &cxt5066_capture_source;
spec->port_d_mode = PIN_HP;
+ spec->ext_mic_bias = PIN_VREF80;
spec->num_init_verbs = 1;
spec->init_verbs[0] = cxt5066_init_verbs;
@@ -2373,6 +2487,20 @@ static int patch_cxt5066(struct hda_codec *codec)
spec->mixers[spec->num_mixers++] = cxt5066_mixer_master_olpc;
spec->mixers[spec->num_mixers++] = cxt5066_mixers;
spec->port_d_mode = 0;
+ spec->ext_mic_bias = CXT5066_OLPC_EXT_MIC_BIAS;
+
+ /* no S/PDIF out */
+ spec->multiout.dig_out_nid = 0;
+
+ /* input source automatically selected */
+ spec->input_mux = NULL;
+ break;
+ case CXT5066_DELL_VOSTO:
+ codec->patch_ops.unsol_event = cxt5066_vostro_event;
+ spec->init_verbs[0] = cxt5066_init_verbs_vostro;
+ spec->mixers[spec->num_mixers++] = cxt5066_mixer_master_olpc;
+ spec->mixers[spec->num_mixers++] = cxt5066_mixers;
+ spec->port_d_mode = 0;
/* no S/PDIF out */
spec->multiout.dig_out_nid = 0;
@@ -2397,6 +2525,8 @@ static struct hda_codec_preset snd_hda_preset_conexant[] = {
.patch = patch_cxt5051 },
{ .id = 0x14f15066, .name = "CX20582 (Pebble)",
.patch = patch_cxt5066 },
+ { .id = 0x14f15067, .name = "CX20583 (Pebble HSF)",
+ .patch = patch_cxt5066 },
{} /* terminator */
};
@@ -2404,6 +2534,7 @@ MODULE_ALIAS("snd-hda-codec-id:14f15045");
MODULE_ALIAS("snd-hda-codec-id:14f15047");
MODULE_ALIAS("snd-hda-codec-id:14f15051");
MODULE_ALIAS("snd-hda-codec-id:14f15066");
+MODULE_ALIAS("snd-hda-codec-id:14f15067");
MODULE_LICENSE("GPL");
MODULE_DESCRIPTION("Conexant HD-audio codec");
diff --git a/sound/pci/hda/patch_intelhdmi.c b/sound/pci/hda/patch_intelhdmi.c
index 01a18ed475a..928df59be5d 100644
--- a/sound/pci/hda/patch_intelhdmi.c
+++ b/sound/pci/hda/patch_intelhdmi.c
@@ -33,15 +33,41 @@
#include "hda_codec.h"
#include "hda_local.h"
-static hda_nid_t cvt_nid; /* audio converter */
-static hda_nid_t pin_nid; /* HDMI output pin */
+/*
+ * The HDMI/DisplayPort configuration can be highly dynamic. A graphics device
+ * could support two independent pipes, each of them can be connected to one or
+ * more ports (DVI, HDMI or DisplayPort).
+ *
+ * The HDA correspondence of pipes/ports are converter/pin nodes.
+ */
+#define INTEL_HDMI_CVTS 2
+#define INTEL_HDMI_PINS 3
-#define INTEL_HDMI_EVENT_TAG 0x08
+static char *intel_hdmi_pcm_names[INTEL_HDMI_CVTS] = {
+ "INTEL HDMI 0",
+ "INTEL HDMI 1",
+};
struct intel_hdmi_spec {
- struct hda_multi_out multiout;
- struct hda_pcm pcm_rec;
- struct hdmi_eld sink_eld;
+ int num_cvts;
+ int num_pins;
+ hda_nid_t cvt[INTEL_HDMI_CVTS+1]; /* audio sources */
+ hda_nid_t pin[INTEL_HDMI_PINS+1]; /* audio sinks */
+
+ /*
+ * source connection for each pin
+ */
+ hda_nid_t pin_cvt[INTEL_HDMI_PINS+1];
+
+ /*
+ * HDMI sink attached to each pin
+ */
+ struct hdmi_eld sink_eld[INTEL_HDMI_PINS];
+
+ /*
+ * export one pcm per pipe
+ */
+ struct hda_pcm pcm_rec[INTEL_HDMI_CVTS];
};
struct hdmi_audio_infoframe {
@@ -184,40 +210,186 @@ static struct cea_channel_speaker_allocation channel_allocations[] = {
{ .ca_index = 0x31, .speakers = { FRW, FLW, RR, RL, FC, LFE, FR, FL } },
};
+
+/*
+ * HDA/HDMI auto parsing
+ */
+
+static int hda_node_index(hda_nid_t *nids, hda_nid_t nid)
+{
+ int i;
+
+ for (i = 0; nids[i]; i++)
+ if (nids[i] == nid)
+ return i;
+
+ snd_printk(KERN_WARNING "HDMI: nid %d not registered\n", nid);
+ return -EINVAL;
+}
+
+static int intel_hdmi_read_pin_conn(struct hda_codec *codec, hda_nid_t pin_nid)
+{
+ struct intel_hdmi_spec *spec = codec->spec;
+ hda_nid_t conn_list[HDA_MAX_CONNECTIONS];
+ int conn_len, curr;
+ int index;
+
+ if (!(get_wcaps(codec, pin_nid) & AC_WCAP_CONN_LIST)) {
+ snd_printk(KERN_WARNING
+ "HDMI: pin %d wcaps %#x "
+ "does not support connection list\n",
+ pin_nid, get_wcaps(codec, pin_nid));
+ return -EINVAL;
+ }
+
+ conn_len = snd_hda_get_connections(codec, pin_nid, conn_list,
+ HDA_MAX_CONNECTIONS);
+ if (conn_len > 1)
+ curr = snd_hda_codec_read(codec, pin_nid, 0,
+ AC_VERB_GET_CONNECT_SEL, 0);
+ else
+ curr = 0;
+
+ index = hda_node_index(spec->pin, pin_nid);
+ if (index < 0)
+ return -EINVAL;
+
+ spec->pin_cvt[index] = conn_list[curr];
+
+ return 0;
+}
+
+static void hdmi_get_show_eld(struct hda_codec *codec, hda_nid_t pin_nid,
+ struct hdmi_eld *eld)
+{
+ if (!snd_hdmi_get_eld(eld, codec, pin_nid))
+ snd_hdmi_show_eld(eld);
+}
+
+static void hdmi_present_sense(struct hda_codec *codec, hda_nid_t pin_nid,
+ struct hdmi_eld *eld)
+{
+ int present = snd_hda_pin_sense(codec, pin_nid);
+
+ eld->monitor_present = !!(present & AC_PINSENSE_PRESENCE);
+ eld->eld_valid = !!(present & AC_PINSENSE_ELDV);
+
+ if (present & AC_PINSENSE_ELDV)
+ hdmi_get_show_eld(codec, pin_nid, eld);
+}
+
+static int intel_hdmi_add_pin(struct hda_codec *codec, hda_nid_t pin_nid)
+{
+ struct intel_hdmi_spec *spec = codec->spec;
+
+ if (spec->num_pins >= INTEL_HDMI_PINS) {
+ snd_printk(KERN_WARNING
+ "HDMI: no space for pin %d \n", pin_nid);
+ return -EINVAL;
+ }
+
+ hdmi_present_sense(codec, pin_nid, &spec->sink_eld[spec->num_pins]);
+
+ spec->pin[spec->num_pins] = pin_nid;
+ spec->num_pins++;
+
+ /*
+ * It is assumed that converter nodes come first in the node list and
+ * hence have been registered and usable now.
+ */
+ return intel_hdmi_read_pin_conn(codec, pin_nid);
+}
+
+static int intel_hdmi_add_cvt(struct hda_codec *codec, hda_nid_t nid)
+{
+ struct intel_hdmi_spec *spec = codec->spec;
+
+ if (spec->num_cvts >= INTEL_HDMI_CVTS) {
+ snd_printk(KERN_WARNING
+ "HDMI: no space for converter %d \n", nid);
+ return -EINVAL;
+ }
+
+ spec->cvt[spec->num_cvts] = nid;
+ spec->num_cvts++;
+
+ return 0;
+}
+
+static int intel_hdmi_parse_codec(struct hda_codec *codec)
+{
+ hda_nid_t nid;
+ int i, nodes;
+
+ nodes = snd_hda_get_sub_nodes(codec, codec->afg, &nid);
+ if (!nid || nodes < 0) {
+ snd_printk(KERN_WARNING "HDMI: failed to get afg sub nodes\n");
+ return -EINVAL;
+ }
+
+ for (i = 0; i < nodes; i++, nid++) {
+ unsigned int caps;
+ unsigned int type;
+
+ caps = snd_hda_param_read(codec, nid, AC_PAR_AUDIO_WIDGET_CAP);
+ type = get_wcaps_type(caps);
+
+ if (!(caps & AC_WCAP_DIGITAL))
+ continue;
+
+ switch (type) {
+ case AC_WID_AUD_OUT:
+ if (intel_hdmi_add_cvt(codec, nid) < 0)
+ return -EINVAL;
+ break;
+ case AC_WID_PIN:
+ caps = snd_hda_param_read(codec, nid, AC_PAR_PIN_CAP);
+ if (!(caps & AC_PINCAP_HDMI))
+ continue;
+ if (intel_hdmi_add_pin(codec, nid) < 0)
+ return -EINVAL;
+ break;
+ }
+ }
+
+ return 0;
+}
+
/*
* HDMI routines
*/
#ifdef BE_PARANOID
-static void hdmi_get_dip_index(struct hda_codec *codec, hda_nid_t nid,
+static void hdmi_get_dip_index(struct hda_codec *codec, hda_nid_t pin_nid,
int *packet_index, int *byte_index)
{
int val;
- val = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_HDMI_DIP_INDEX, 0);
+ val = snd_hda_codec_read(codec, pin_nid, 0,
+ AC_VERB_GET_HDMI_DIP_INDEX, 0);
*packet_index = val >> 5;
*byte_index = val & 0x1f;
}
#endif
-static void hdmi_set_dip_index(struct hda_codec *codec, hda_nid_t nid,
+static void hdmi_set_dip_index(struct hda_codec *codec, hda_nid_t pin_nid,
int packet_index, int byte_index)
{
int val;
val = (packet_index << 5) | (byte_index & 0x1f);
- snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_HDMI_DIP_INDEX, val);
+ snd_hda_codec_write(codec, pin_nid, 0, AC_VERB_SET_HDMI_DIP_INDEX, val);
}
-static void hdmi_write_dip_byte(struct hda_codec *codec, hda_nid_t nid,
+static void hdmi_write_dip_byte(struct hda_codec *codec, hda_nid_t pin_nid,
unsigned char val)
{
- snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_HDMI_DIP_DATA, val);
+ snd_hda_codec_write(codec, pin_nid, 0, AC_VERB_SET_HDMI_DIP_DATA, val);
}
-static void hdmi_enable_output(struct hda_codec *codec)
+static void hdmi_enable_output(struct hda_codec *codec, hda_nid_t pin_nid)
{
/* Unmute */
if (get_wcaps(codec, pin_nid) & AC_WCAP_OUT_AMP)
@@ -231,7 +403,8 @@ static void hdmi_enable_output(struct hda_codec *codec)
/*
* Enable Audio InfoFrame Transmission
*/
-static void hdmi_start_infoframe_trans(struct hda_codec *codec)
+static void hdmi_start_infoframe_trans(struct hda_codec *codec,
+ hda_nid_t pin_nid)
{
hdmi_set_dip_index(codec, pin_nid, 0x0, 0x0);
snd_hda_codec_write(codec, pin_nid, 0, AC_VERB_SET_HDMI_DIP_XMIT,
@@ -241,59 +414,49 @@ static void hdmi_start_infoframe_trans(struct hda_codec *codec)
/*
* Disable Audio InfoFrame Transmission
*/
-static void hdmi_stop_infoframe_trans(struct hda_codec *codec)
+static void hdmi_stop_infoframe_trans(struct hda_codec *codec,
+ hda_nid_t pin_nid)
{
hdmi_set_dip_index(codec, pin_nid, 0x0, 0x0);
snd_hda_codec_write(codec, pin_nid, 0, AC_VERB_SET_HDMI_DIP_XMIT,
AC_DIPXMIT_DISABLE);
}
-static int hdmi_get_channel_count(struct hda_codec *codec)
+static int hdmi_get_channel_count(struct hda_codec *codec, hda_nid_t nid)
{
- return 1 + snd_hda_codec_read(codec, cvt_nid, 0,
+ return 1 + snd_hda_codec_read(codec, nid, 0,
AC_VERB_GET_CVT_CHAN_COUNT, 0);
}
-static void hdmi_set_channel_count(struct hda_codec *codec, int chs)
+static void hdmi_set_channel_count(struct hda_codec *codec,
+ hda_nid_t nid, int chs)
{
- snd_hda_codec_write(codec, cvt_nid, 0,
- AC_VERB_SET_CVT_CHAN_COUNT, chs - 1);
-
- if (chs != hdmi_get_channel_count(codec))
- snd_printd(KERN_INFO "HDMI channel count: expect %d, get %d\n",
- chs, hdmi_get_channel_count(codec));
+ if (chs != hdmi_get_channel_count(codec, nid))
+ snd_hda_codec_write(codec, nid, 0,
+ AC_VERB_SET_CVT_CHAN_COUNT, chs - 1);
}
-static void hdmi_debug_channel_mapping(struct hda_codec *codec)
+static void hdmi_debug_channel_mapping(struct hda_codec *codec, hda_nid_t nid)
{
#ifdef CONFIG_SND_DEBUG_VERBOSE
int i;
int slot;
for (i = 0; i < 8; i++) {
- slot = snd_hda_codec_read(codec, cvt_nid, 0,
+ slot = snd_hda_codec_read(codec, nid, 0,
AC_VERB_GET_HDMI_CHAN_SLOT, i);
printk(KERN_DEBUG "HDMI: ASP channel %d => slot %d\n",
- slot >> 4, slot & 0x7);
+ slot >> 4, slot & 0xf);
}
#endif
}
-static void hdmi_parse_eld(struct hda_codec *codec)
-{
- struct intel_hdmi_spec *spec = codec->spec;
- struct hdmi_eld *eld = &spec->sink_eld;
-
- if (!snd_hdmi_get_eld(eld, codec, pin_nid))
- snd_hdmi_show_eld(eld);
-}
-
/*
* Audio InfoFrame routines
*/
-static void hdmi_debug_dip_size(struct hda_codec *codec)
+static void hdmi_debug_dip_size(struct hda_codec *codec, hda_nid_t pin_nid)
{
#ifdef CONFIG_SND_DEBUG_VERBOSE
int i;
@@ -310,7 +473,7 @@ static void hdmi_debug_dip_size(struct hda_codec *codec)
#endif
}
-static void hdmi_clear_dip_buffers(struct hda_codec *codec)
+static void hdmi_clear_dip_buffers(struct hda_codec *codec, hda_nid_t pin_nid)
{
#ifdef BE_PARANOID
int i, j;
@@ -339,23 +502,35 @@ static void hdmi_clear_dip_buffers(struct hda_codec *codec)
#endif
}
-static void hdmi_fill_audio_infoframe(struct hda_codec *codec,
- struct hdmi_audio_infoframe *ai)
+static void hdmi_checksum_audio_infoframe(struct hdmi_audio_infoframe *ai)
{
- u8 *params = (u8 *)ai;
+ u8 *bytes = (u8 *)ai;
u8 sum = 0;
int i;
- hdmi_debug_dip_size(codec);
- hdmi_clear_dip_buffers(codec); /* be paranoid */
+ ai->checksum = 0;
+
+ for (i = 0; i < sizeof(*ai); i++)
+ sum += bytes[i];
- for (i = 0; i < sizeof(ai); i++)
- sum += params[i];
ai->checksum = - sum;
+}
+
+static void hdmi_fill_audio_infoframe(struct hda_codec *codec,
+ hda_nid_t pin_nid,
+ struct hdmi_audio_infoframe *ai)
+{
+ u8 *bytes = (u8 *)ai;
+ int i;
+
+ hdmi_debug_dip_size(codec, pin_nid);
+ hdmi_clear_dip_buffers(codec, pin_nid); /* be paranoid */
+
+ hdmi_checksum_audio_infoframe(ai);
hdmi_set_dip_index(codec, pin_nid, 0x0, 0x0);
- for (i = 0; i < sizeof(ai); i++)
- hdmi_write_dip_byte(codec, pin_nid, params[i]);
+ for (i = 0; i < sizeof(*ai); i++)
+ hdmi_write_dip_byte(codec, pin_nid, bytes[i]);
}
/*
@@ -386,11 +561,11 @@ static void init_channel_allocations(void)
*
* TODO: it could select the wrong CA from multiple candidates.
*/
-static int hdmi_setup_channel_allocation(struct hda_codec *codec,
+static int hdmi_setup_channel_allocation(struct hda_codec *codec, hda_nid_t nid,
struct hdmi_audio_infoframe *ai)
{
struct intel_hdmi_spec *spec = codec->spec;
- struct hdmi_eld *eld = &spec->sink_eld;
+ struct hdmi_eld *eld;
int i;
int spk_mask = 0;
int channels = 1 + (ai->CC02_CT47 & 0x7);
@@ -402,6 +577,11 @@ static int hdmi_setup_channel_allocation(struct hda_codec *codec,
if (channels <= 2)
return 0;
+ i = hda_node_index(spec->pin_cvt, nid);
+ if (i < 0)
+ return 0;
+ eld = &spec->sink_eld[i];
+
/*
* HDMI sink's ELD info cannot always be retrieved for now, e.g.
* in console or for audio devices. Assume the highest speakers
@@ -439,8 +619,8 @@ static int hdmi_setup_channel_allocation(struct hda_codec *codec,
return ai->CA;
}
-static void hdmi_setup_channel_mapping(struct hda_codec *codec,
- struct hdmi_audio_infoframe *ai)
+static void hdmi_setup_channel_mapping(struct hda_codec *codec, hda_nid_t nid,
+ struct hdmi_audio_infoframe *ai)
{
int i;
@@ -453,17 +633,41 @@ static void hdmi_setup_channel_mapping(struct hda_codec *codec,
*/
for (i = 0; i < 8; i++)
- snd_hda_codec_write(codec, cvt_nid, 0,
+ snd_hda_codec_write(codec, nid, 0,
AC_VERB_SET_HDMI_CHAN_SLOT,
(i << 4) | i);
- hdmi_debug_channel_mapping(codec);
+ hdmi_debug_channel_mapping(codec, nid);
}
+static bool hdmi_infoframe_uptodate(struct hda_codec *codec, hda_nid_t pin_nid,
+ struct hdmi_audio_infoframe *ai)
+{
+ u8 *bytes = (u8 *)ai;
+ u8 val;
+ int i;
+
+ if (snd_hda_codec_read(codec, pin_nid, 0, AC_VERB_GET_HDMI_DIP_XMIT, 0)
+ != AC_DIPXMIT_BEST)
+ return false;
+
+ hdmi_set_dip_index(codec, pin_nid, 0x0, 0x0);
+ for (i = 0; i < sizeof(*ai); i++) {
+ val = snd_hda_codec_read(codec, pin_nid, 0,
+ AC_VERB_GET_HDMI_DIP_DATA, 0);
+ if (val != bytes[i])
+ return false;
+ }
-static void hdmi_setup_audio_infoframe(struct hda_codec *codec,
+ return true;
+}
+
+static void hdmi_setup_audio_infoframe(struct hda_codec *codec, hda_nid_t nid,
struct snd_pcm_substream *substream)
{
+ struct intel_hdmi_spec *spec = codec->spec;
+ hda_nid_t pin_nid;
+ int i;
struct hdmi_audio_infoframe ai = {
.type = 0x84,
.ver = 0x01,
@@ -471,11 +675,22 @@ static void hdmi_setup_audio_infoframe(struct hda_codec *codec,
.CC02_CT47 = substream->runtime->channels - 1,
};
- hdmi_setup_channel_allocation(codec, &ai);
- hdmi_setup_channel_mapping(codec, &ai);
+ hdmi_setup_channel_allocation(codec, nid, &ai);
+ hdmi_setup_channel_mapping(codec, nid, &ai);
- hdmi_fill_audio_infoframe(codec, &ai);
- hdmi_start_infoframe_trans(codec);
+ for (i = 0; i < spec->num_pins; i++) {
+ if (spec->pin_cvt[i] != nid)
+ continue;
+ if (!spec->sink_eld[i].monitor_present)
+ continue;
+
+ pin_nid = spec->pin[i];
+ if (!hdmi_infoframe_uptodate(codec, pin_nid, &ai)) {
+ hdmi_stop_infoframe_trans(codec, pin_nid);
+ hdmi_fill_audio_infoframe(codec, pin_nid, &ai);
+ hdmi_start_infoframe_trans(codec, pin_nid);
+ }
+ }
}
@@ -485,27 +700,39 @@ static void hdmi_setup_audio_infoframe(struct hda_codec *codec,
static void hdmi_intrinsic_event(struct hda_codec *codec, unsigned int res)
{
+ struct intel_hdmi_spec *spec = codec->spec;
+ int tag = res >> AC_UNSOL_RES_TAG_SHIFT;
int pind = !!(res & AC_UNSOL_RES_PD);
int eldv = !!(res & AC_UNSOL_RES_ELDV);
+ int index;
printk(KERN_INFO
- "HDMI hot plug event: Presence_Detect=%d ELD_Valid=%d\n",
- pind, eldv);
+ "HDMI hot plug event: Pin=%d Presence_Detect=%d ELD_Valid=%d\n",
+ tag, pind, eldv);
+
+ index = hda_node_index(spec->pin, tag);
+ if (index < 0)
+ return;
+
+ spec->sink_eld[index].monitor_present = pind;
+ spec->sink_eld[index].eld_valid = eldv;
if (pind && eldv) {
- hdmi_parse_eld(codec);
+ hdmi_get_show_eld(codec, spec->pin[index], &spec->sink_eld[index]);
/* TODO: do real things about ELD */
}
}
static void hdmi_non_intrinsic_event(struct hda_codec *codec, unsigned int res)
{
+ int tag = res >> AC_UNSOL_RES_TAG_SHIFT;
int subtag = (res & AC_UNSOL_RES_SUBTAG) >> AC_UNSOL_RES_SUBTAG_SHIFT;
int cp_state = !!(res & AC_UNSOL_RES_CP_STATE);
int cp_ready = !!(res & AC_UNSOL_RES_CP_READY);
printk(KERN_INFO
- "HDMI content protection event: SUBTAG=0x%x CP_STATE=%d CP_READY=%d\n",
+ "HDMI CP event: PIN=%d SUBTAG=0x%x CP_STATE=%d CP_READY=%d\n",
+ tag,
subtag,
cp_state,
cp_ready);
@@ -520,10 +747,11 @@ static void hdmi_non_intrinsic_event(struct hda_codec *codec, unsigned int res)
static void intel_hdmi_unsol_event(struct hda_codec *codec, unsigned int res)
{
+ struct intel_hdmi_spec *spec = codec->spec;
int tag = res >> AC_UNSOL_RES_TAG_SHIFT;
int subtag = (res & AC_UNSOL_RES_SUBTAG) >> AC_UNSOL_RES_SUBTAG_SHIFT;
- if (tag != INTEL_HDMI_EVENT_TAG) {
+ if (hda_node_index(spec->pin, tag) < 0) {
snd_printd(KERN_INFO "Unexpected HDMI event tag 0x%x\n", tag);
return;
}
@@ -538,24 +766,29 @@ static void intel_hdmi_unsol_event(struct hda_codec *codec, unsigned int res)
* Callbacks
*/
-static int intel_hdmi_playback_pcm_open(struct hda_pcm_stream *hinfo,
- struct hda_codec *codec,
- struct snd_pcm_substream *substream)
+static void hdmi_setup_stream(struct hda_codec *codec, hda_nid_t nid,
+ u32 stream_tag, int format)
{
- struct intel_hdmi_spec *spec = codec->spec;
-
- return snd_hda_multi_out_dig_open(codec, &spec->multiout);
-}
+ int tag;
+ int fmt;
-static int intel_hdmi_playback_pcm_close(struct hda_pcm_stream *hinfo,
- struct hda_codec *codec,
- struct snd_pcm_substream *substream)
-{
- struct intel_hdmi_spec *spec = codec->spec;
+ tag = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONV, 0) >> 4;
+ fmt = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_STREAM_FORMAT, 0);
- hdmi_stop_infoframe_trans(codec);
+ snd_printdd("hdmi_setup_stream: "
+ "NID=0x%x, %sstream=0x%x, %sformat=0x%x\n",
+ nid,
+ tag == stream_tag ? "" : "new-",
+ stream_tag,
+ fmt == format ? "" : "new-",
+ format);
- return snd_hda_multi_out_dig_close(codec, &spec->multiout);
+ if (tag != stream_tag)
+ snd_hda_codec_write(codec, nid, 0,
+ AC_VERB_SET_CHANNEL_STREAMID, stream_tag << 4);
+ if (fmt != format)
+ snd_hda_codec_write(codec, nid, 0,
+ AC_VERB_SET_STREAM_FORMAT, format);
}
static int intel_hdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
@@ -564,43 +797,53 @@ static int intel_hdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
unsigned int format,
struct snd_pcm_substream *substream)
{
- struct intel_hdmi_spec *spec = codec->spec;
-
- snd_hda_multi_out_dig_prepare(codec, &spec->multiout, stream_tag,
- format, substream);
+ hdmi_set_channel_count(codec, hinfo->nid,
+ substream->runtime->channels);
- hdmi_set_channel_count(codec, substream->runtime->channels);
+ hdmi_setup_audio_infoframe(codec, hinfo->nid, substream);
- hdmi_setup_audio_infoframe(codec, substream);
+ hdmi_setup_stream(codec, hinfo->nid, stream_tag, format);
+ return 0;
+}
+static int intel_hdmi_playback_pcm_cleanup(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ struct snd_pcm_substream *substream)
+{
return 0;
}
static struct hda_pcm_stream intel_hdmi_pcm_playback = {
.substreams = 1,
.channels_min = 2,
- .channels_max = 8,
.ops = {
- .open = intel_hdmi_playback_pcm_open,
- .close = intel_hdmi_playback_pcm_close,
- .prepare = intel_hdmi_playback_pcm_prepare
+ .prepare = intel_hdmi_playback_pcm_prepare,
+ .cleanup = intel_hdmi_playback_pcm_cleanup,
},
};
static int intel_hdmi_build_pcms(struct hda_codec *codec)
{
struct intel_hdmi_spec *spec = codec->spec;
- struct hda_pcm *info = &spec->pcm_rec;
+ struct hda_pcm *info = spec->pcm_rec;
+ int i;
- codec->num_pcms = 1;
+ codec->num_pcms = spec->num_cvts;
codec->pcm_info = info;
- /* NID to query formats and rates and setup streams */
- intel_hdmi_pcm_playback.nid = cvt_nid;
+ for (i = 0; i < codec->num_pcms; i++, info++) {
+ unsigned int chans;
- info->name = "INTEL HDMI";
- info->pcm_type = HDA_PCM_TYPE_HDMI;
- info->stream[SNDRV_PCM_STREAM_PLAYBACK] = intel_hdmi_pcm_playback;
+ chans = get_wcaps(codec, spec->cvt[i]);
+ chans = get_wcaps_channels(chans);
+
+ info->name = intel_hdmi_pcm_names[i];
+ info->pcm_type = HDA_PCM_TYPE_HDMI;
+ info->stream[SNDRV_PCM_STREAM_PLAYBACK] =
+ intel_hdmi_pcm_playback;
+ info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = spec->cvt[i];
+ info->stream[SNDRV_PCM_STREAM_PLAYBACK].channels_max = chans;
+ }
return 0;
}
@@ -609,29 +852,39 @@ static int intel_hdmi_build_controls(struct hda_codec *codec)
{
struct intel_hdmi_spec *spec = codec->spec;
int err;
+ int i;
- err = snd_hda_create_spdif_out_ctls(codec, spec->multiout.dig_out_nid);
- if (err < 0)
- return err;
+ for (i = 0; i < codec->num_pcms; i++) {
+ err = snd_hda_create_spdif_out_ctls(codec, spec->cvt[i]);
+ if (err < 0)
+ return err;
+ }
return 0;
}
static int intel_hdmi_init(struct hda_codec *codec)
{
- hdmi_enable_output(codec);
+ struct intel_hdmi_spec *spec = codec->spec;
+ int i;
- snd_hda_codec_write(codec, pin_nid, 0,
- AC_VERB_SET_UNSOLICITED_ENABLE,
- AC_USRSP_EN | INTEL_HDMI_EVENT_TAG);
+ for (i = 0; spec->pin[i]; i++) {
+ hdmi_enable_output(codec, spec->pin[i]);
+ snd_hda_codec_write(codec, spec->pin[i], 0,
+ AC_VERB_SET_UNSOLICITED_ENABLE,
+ AC_USRSP_EN | spec->pin[i]);
+ }
return 0;
}
static void intel_hdmi_free(struct hda_codec *codec)
{
struct intel_hdmi_spec *spec = codec->spec;
+ int i;
+
+ for (i = 0; i < spec->num_pins; i++)
+ snd_hda_eld_proc_free(codec, &spec->sink_eld[i]);
- snd_hda_eld_proc_free(codec, &spec->sink_eld);
kfree(spec);
}
@@ -643,49 +896,38 @@ static struct hda_codec_ops intel_hdmi_patch_ops = {
.unsol_event = intel_hdmi_unsol_event,
};
-static int do_patch_intel_hdmi(struct hda_codec *codec)
+static int patch_intel_hdmi(struct hda_codec *codec)
{
struct intel_hdmi_spec *spec;
+ int i;
spec = kzalloc(sizeof(*spec), GFP_KERNEL);
if (spec == NULL)
return -ENOMEM;
- spec->multiout.num_dacs = 0; /* no analog */
- spec->multiout.max_channels = 8;
- spec->multiout.dig_out_nid = cvt_nid;
-
codec->spec = spec;
+ if (intel_hdmi_parse_codec(codec) < 0) {
+ codec->spec = NULL;
+ kfree(spec);
+ return -EINVAL;
+ }
codec->patch_ops = intel_hdmi_patch_ops;
- snd_hda_eld_proc_new(codec, &spec->sink_eld);
+ for (i = 0; i < spec->num_pins; i++)
+ snd_hda_eld_proc_new(codec, &spec->sink_eld[i], i);
init_channel_allocations();
return 0;
}
-static int patch_intel_hdmi(struct hda_codec *codec)
-{
- cvt_nid = 0x02;
- pin_nid = 0x03;
- return do_patch_intel_hdmi(codec);
-}
-
-static int patch_intel_hdmi_ibexpeak(struct hda_codec *codec)
-{
- cvt_nid = 0x02;
- pin_nid = 0x04;
- return do_patch_intel_hdmi(codec);
-}
-
static struct hda_codec_preset snd_hda_preset_intelhdmi[] = {
{ .id = 0x808629fb, .name = "G45 DEVCL", .patch = patch_intel_hdmi },
{ .id = 0x80862801, .name = "G45 DEVBLC", .patch = patch_intel_hdmi },
{ .id = 0x80862802, .name = "G45 DEVCTG", .patch = patch_intel_hdmi },
{ .id = 0x80862803, .name = "G45 DEVELK", .patch = patch_intel_hdmi },
{ .id = 0x80862804, .name = "G45 DEVIBX", .patch = patch_intel_hdmi },
- { .id = 0x80860054, .name = "Q57 DEVIBX", .patch = patch_intel_hdmi_ibexpeak },
+ { .id = 0x80860054, .name = "Q57 DEVIBX", .patch = patch_intel_hdmi },
{ .id = 0x10951392, .name = "SiI1392 HDMI", .patch = patch_intel_hdmi },
{} /* terminator */
};
diff --git a/sound/pci/hda/patch_nvhdmi.c b/sound/pci/hda/patch_nvhdmi.c
index c8435c9a97f..6afdab09bab 100644
--- a/sound/pci/hda/patch_nvhdmi.c
+++ b/sound/pci/hda/patch_nvhdmi.c
@@ -29,6 +29,9 @@
#include "hda_codec.h"
#include "hda_local.h"
+/* define below to restrict the supported rates and formats */
+/* #define LIMITED_RATE_FMT_SUPPORT */
+
struct nvhdmi_spec {
struct hda_multi_out multiout;
@@ -60,6 +63,22 @@ static struct hda_verb nvhdmi_basic_init[] = {
{} /* terminator */
};
+#ifdef LIMITED_RATE_FMT_SUPPORT
+/* support only the safe format and rate */
+#define SUPPORTED_RATES SNDRV_PCM_RATE_48000
+#define SUPPORTED_MAXBPS 16
+#define SUPPORTED_FORMATS SNDRV_PCM_FMTBIT_S16_LE
+#else
+/* support all rates and formats */
+#define SUPPORTED_RATES \
+ (SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |\
+ SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_176400 |\
+ SNDRV_PCM_RATE_192000)
+#define SUPPORTED_MAXBPS 24
+#define SUPPORTED_FORMATS \
+ (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE)
+#endif
+
/*
* Controls
*/
@@ -258,9 +277,9 @@ static struct hda_pcm_stream nvhdmi_pcm_digital_playback_8ch = {
.channels_min = 2,
.channels_max = 8,
.nid = Nv_Master_Convert_nid,
- .rates = SNDRV_PCM_RATE_48000,
- .maxbps = 16,
- .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .rates = SUPPORTED_RATES,
+ .maxbps = SUPPORTED_MAXBPS,
+ .formats = SUPPORTED_FORMATS,
.ops = {
.open = nvhdmi_dig_playback_pcm_open,
.close = nvhdmi_dig_playback_pcm_close_8ch,
@@ -273,9 +292,9 @@ static struct hda_pcm_stream nvhdmi_pcm_digital_playback_2ch = {
.channels_min = 2,
.channels_max = 2,
.nid = Nv_Master_Convert_nid,
- .rates = SNDRV_PCM_RATE_48000,
- .maxbps = 16,
- .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .rates = SUPPORTED_RATES,
+ .maxbps = SUPPORTED_MAXBPS,
+ .formats = SUPPORTED_FORMATS,
.ops = {
.open = nvhdmi_dig_playback_pcm_open,
.close = nvhdmi_dig_playback_pcm_close_2ch,
@@ -378,6 +397,7 @@ static int patch_nvhdmi_2ch(struct hda_codec *codec)
static struct hda_codec_preset snd_hda_preset_nvhdmi[] = {
{ .id = 0x10de0002, .name = "MCP78 HDMI", .patch = patch_nvhdmi_8ch },
{ .id = 0x10de0003, .name = "MCP78 HDMI", .patch = patch_nvhdmi_8ch },
+ { .id = 0x10de0005, .name = "MCP78 HDMI", .patch = patch_nvhdmi_8ch },
{ .id = 0x10de0006, .name = "MCP78 HDMI", .patch = patch_nvhdmi_8ch },
{ .id = 0x10de0007, .name = "MCP7A HDMI", .patch = patch_nvhdmi_8ch },
{ .id = 0x10de0067, .name = "MCP67 HDMI", .patch = patch_nvhdmi_2ch },
@@ -387,6 +407,7 @@ static struct hda_codec_preset snd_hda_preset_nvhdmi[] = {
MODULE_ALIAS("snd-hda-codec-id:10de0002");
MODULE_ALIAS("snd-hda-codec-id:10de0003");
+MODULE_ALIAS("snd-hda-codec-id:10de0005");
MODULE_ALIAS("snd-hda-codec-id:10de0006");
MODULE_ALIAS("snd-hda-codec-id:10de0007");
MODULE_ALIAS("snd-hda-codec-id:10de0067");
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 12960581956..a38a81e5386 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -275,7 +275,7 @@ struct alc_spec {
struct snd_kcontrol_new *cap_mixer; /* capture mixer */
unsigned int beep_amp; /* beep amp value, set via set_beep_amp() */
- const struct hda_verb *init_verbs[5]; /* initialization verbs
+ const struct hda_verb *init_verbs[10]; /* initialization verbs
* don't forget NULL
* termination!
*/
@@ -961,16 +961,12 @@ static void alc_fix_pll_init(struct hda_codec *codec, hda_nid_t nid,
static void alc_automute_pin(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
- unsigned int present, pincap;
unsigned int nid = spec->autocfg.hp_pins[0];
int i;
- pincap = snd_hda_query_pin_caps(codec, nid);
- if (pincap & AC_PINCAP_TRIG_REQ) /* need trigger? */
- snd_hda_codec_read(codec, nid, 0, AC_VERB_SET_PIN_SENSE, 0);
- present = snd_hda_codec_read(codec, nid, 0,
- AC_VERB_GET_PIN_SENSE, 0);
- spec->jack_present = (present & AC_PINSENSE_PRESENCE) != 0;
+ if (!nid)
+ return;
+ spec->jack_present = snd_hda_jack_detect(codec, nid);
for (i = 0; i < ARRAY_SIZE(spec->autocfg.speaker_pins); i++) {
nid = spec->autocfg.speaker_pins[i];
if (!nid)
@@ -1010,9 +1006,7 @@ static void alc_mic_automute(struct hda_codec *codec)
cap_nid = spec->capsrc_nids ? spec->capsrc_nids[0] : spec->adc_nids[0];
- present = snd_hda_codec_read(codec, spec->ext_mic.pin, 0,
- AC_VERB_GET_PIN_SENSE, 0);
- present &= AC_PINSENSE_PRESENCE;
+ present = snd_hda_jack_detect(codec, spec->ext_mic.pin);
if (present) {
alive = &spec->ext_mic;
dead = &spec->int_mic;
@@ -1332,15 +1326,20 @@ do_sku:
* when the external headphone out jack is plugged"
*/
if (!spec->autocfg.hp_pins[0]) {
+ hda_nid_t nid;
tmp = (ass >> 11) & 0x3; /* HP to chassis */
if (tmp == 0)
- spec->autocfg.hp_pins[0] = porta;
+ nid = porta;
else if (tmp == 1)
- spec->autocfg.hp_pins[0] = porte;
+ nid = porte;
else if (tmp == 2)
- spec->autocfg.hp_pins[0] = portd;
+ nid = portd;
else
return 1;
+ for (i = 0; i < spec->autocfg.line_outs; i++)
+ if (spec->autocfg.line_out_pins[i] == nid)
+ return 1;
+ spec->autocfg.hp_pins[0] = nid;
}
alc_init_auto_hp(codec);
@@ -1362,7 +1361,7 @@ static void alc_ssid_check(struct hda_codec *codec,
}
/*
- * Fix-up pin default configurations
+ * Fix-up pin default configurations and add default verbs
*/
struct alc_pincfg {
@@ -1370,9 +1369,14 @@ struct alc_pincfg {
u32 val;
};
-static void alc_fix_pincfg(struct hda_codec *codec,
+struct alc_fixup {
+ const struct alc_pincfg *pins;
+ const struct hda_verb *verbs;
+};
+
+static void alc_pick_fixup(struct hda_codec *codec,
const struct snd_pci_quirk *quirk,
- const struct alc_pincfg **pinfix)
+ const struct alc_fixup *fix)
{
const struct alc_pincfg *cfg;
@@ -1380,9 +1384,14 @@ static void alc_fix_pincfg(struct hda_codec *codec,
if (!quirk)
return;
- cfg = pinfix[quirk->value];
- for (; cfg->nid; cfg++)
- snd_hda_codec_set_pincfg(codec, cfg->nid, cfg->val);
+ fix += quirk->value;
+ cfg = fix->pins;
+ if (cfg) {
+ for (; cfg->nid; cfg++)
+ snd_hda_codec_set_pincfg(codec, cfg->nid, cfg->val);
+ }
+ if (fix->verbs)
+ add_verb(codec->spec, fix->verbs);
}
/*
@@ -1496,7 +1505,7 @@ static struct hda_verb alc888_fujitsu_xa3530_verbs[] = {
static void alc_automute_amp(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
- unsigned int val, mute, pincap;
+ unsigned int mute;
hda_nid_t nid;
int i;
@@ -1505,13 +1514,7 @@ static void alc_automute_amp(struct hda_codec *codec)
nid = spec->autocfg.hp_pins[i];
if (!nid)
break;
- pincap = snd_hda_query_pin_caps(codec, nid);
- if (pincap & AC_PINCAP_TRIG_REQ) /* need trigger? */
- snd_hda_codec_read(codec, nid, 0,
- AC_VERB_SET_PIN_SENSE, 0);
- val = snd_hda_codec_read(codec, nid, 0,
- AC_VERB_GET_PIN_SENSE, 0);
- if (val & AC_PINSENSE_PRESENCE) {
+ if (snd_hda_jack_detect(codec, nid)) {
spec->jack_present = 1;
break;
}
@@ -1769,6 +1772,8 @@ static void alc888_acer_aspire_4930g_setup(struct hda_codec *codec)
spec->autocfg.hp_pins[0] = 0x15;
spec->autocfg.speaker_pins[0] = 0x14;
+ spec->autocfg.speaker_pins[1] = 0x16;
+ spec->autocfg.speaker_pins[2] = 0x17;
}
static void alc888_acer_aspire_6530g_setup(struct hda_codec *codec)
@@ -2393,12 +2398,14 @@ static const char *alc_slave_sws[] = {
static void alc_free_kctls(struct hda_codec *codec);
+#ifdef CONFIG_SND_HDA_INPUT_BEEP
/* additional beep mixers; the actual parameters are overwritten at build */
static struct snd_kcontrol_new alc_beep_mixer[] = {
HDA_CODEC_VOLUME("Beep Playback Volume", 0, 0, HDA_INPUT),
- HDA_CODEC_MUTE("Beep Playback Switch", 0, 0, HDA_INPUT),
+ HDA_CODEC_MUTE_BEEP("Beep Playback Switch", 0, 0, HDA_INPUT),
{ } /* end */
};
+#endif
static int alc_build_controls(struct hda_codec *codec)
{
@@ -2435,6 +2442,7 @@ static int alc_build_controls(struct hda_codec *codec)
return err;
}
+#ifdef CONFIG_SND_HDA_INPUT_BEEP
/* create beep controls if needed */
if (spec->beep_amp) {
struct snd_kcontrol_new *knew;
@@ -2444,11 +2452,13 @@ static int alc_build_controls(struct hda_codec *codec)
if (!kctl)
return -ENOMEM;
kctl->private_value = spec->beep_amp;
- err = snd_hda_ctl_add(codec, kctl);
+ err = snd_hda_ctl_add(codec,
+ get_amp_nid_(spec->beep_amp), kctl);
if (err < 0)
return err;
}
}
+#endif
/* if we have no master control, let's create it */
if (!spec->no_analog &&
@@ -2762,8 +2772,7 @@ static void alc880_uniwill_mic_automute(struct hda_codec *codec)
unsigned int present;
unsigned char bits;
- present = snd_hda_codec_read(codec, 0x18, 0,
- AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
+ present = snd_hda_jack_detect(codec, 0x18);
bits = present ? HDA_AMP_MUTE : 0;
snd_hda_codec_amp_stereo(codec, 0x0b, HDA_INPUT, 1, HDA_AMP_MUTE, bits);
}
@@ -4305,10 +4314,26 @@ static int add_control(struct alc_spec *spec, int type, const char *name,
knew->name = kstrdup(name, GFP_KERNEL);
if (!knew->name)
return -ENOMEM;
+ if (get_amp_nid_(val))
+ knew->subdevice = HDA_SUBDEV_NID_FLAG | get_amp_nid_(val);
knew->private_value = val;
return 0;
}
+static int add_control_with_pfx(struct alc_spec *spec, int type,
+ const char *pfx, const char *dir,
+ const char *sfx, unsigned long val)
+{
+ char name[32];
+ snprintf(name, sizeof(name), "%s %s %s", pfx, dir, sfx);
+ return add_control(spec, type, name, val);
+}
+
+#define add_pb_vol_ctrl(spec, type, pfx, val) \
+ add_control_with_pfx(spec, type, pfx, "Playback", "Volume", val)
+#define add_pb_sw_ctrl(spec, type, pfx, val) \
+ add_control_with_pfx(spec, type, pfx, "Playback", "Switch", val)
+
#define alc880_is_fixed_pin(nid) ((nid) >= 0x14 && (nid) <= 0x17)
#define alc880_fixed_pin_idx(nid) ((nid) - 0x14)
#define alc880_is_multi_pin(nid) ((nid) >= 0x18)
@@ -4362,7 +4387,6 @@ static int alc880_auto_fill_dac_nids(struct alc_spec *spec,
static int alc880_auto_create_multi_out_ctls(struct alc_spec *spec,
const struct auto_pin_cfg *cfg)
{
- char name[32];
static const char *chname[4] = {
"Front", "Surround", NULL /*CLFE*/, "Side"
};
@@ -4375,26 +4399,26 @@ static int alc880_auto_create_multi_out_ctls(struct alc_spec *spec,
nid = alc880_idx_to_mixer(alc880_dac_to_idx(spec->multiout.dac_nids[i]));
if (i == 2) {
/* Center/LFE */
- err = add_control(spec, ALC_CTL_WIDGET_VOL,
- "Center Playback Volume",
+ err = add_pb_vol_ctrl(spec, ALC_CTL_WIDGET_VOL,
+ "Center",
HDA_COMPOSE_AMP_VAL(nid, 1, 0,
HDA_OUTPUT));
if (err < 0)
return err;
- err = add_control(spec, ALC_CTL_WIDGET_VOL,
- "LFE Playback Volume",
+ err = add_pb_vol_ctrl(spec, ALC_CTL_WIDGET_VOL,
+ "LFE",
HDA_COMPOSE_AMP_VAL(nid, 2, 0,
HDA_OUTPUT));
if (err < 0)
return err;
- err = add_control(spec, ALC_CTL_BIND_MUTE,
- "Center Playback Switch",
+ err = add_pb_sw_ctrl(spec, ALC_CTL_BIND_MUTE,
+ "Center",
HDA_COMPOSE_AMP_VAL(nid, 1, 2,
HDA_INPUT));
if (err < 0)
return err;
- err = add_control(spec, ALC_CTL_BIND_MUTE,
- "LFE Playback Switch",
+ err = add_pb_sw_ctrl(spec, ALC_CTL_BIND_MUTE,
+ "LFE",
HDA_COMPOSE_AMP_VAL(nid, 2, 2,
HDA_INPUT));
if (err < 0)
@@ -4406,14 +4430,12 @@ static int alc880_auto_create_multi_out_ctls(struct alc_spec *spec,
pfx = "Speaker";
else
pfx = chname[i];
- sprintf(name, "%s Playback Volume", pfx);
- err = add_control(spec, ALC_CTL_WIDGET_VOL, name,
+ err = add_pb_vol_ctrl(spec, ALC_CTL_WIDGET_VOL, pfx,
HDA_COMPOSE_AMP_VAL(nid, 3, 0,
HDA_OUTPUT));
if (err < 0)
return err;
- sprintf(name, "%s Playback Switch", pfx);
- err = add_control(spec, ALC_CTL_BIND_MUTE, name,
+ err = add_pb_sw_ctrl(spec, ALC_CTL_BIND_MUTE, pfx,
HDA_COMPOSE_AMP_VAL(nid, 3, 2,
HDA_INPUT));
if (err < 0)
@@ -4429,7 +4451,6 @@ static int alc880_auto_create_extra_out(struct alc_spec *spec, hda_nid_t pin,
{
hda_nid_t nid;
int err;
- char name[32];
if (!pin)
return 0;
@@ -4443,21 +4464,18 @@ static int alc880_auto_create_extra_out(struct alc_spec *spec, hda_nid_t pin,
spec->multiout.extra_out_nid[0] = nid;
/* control HP volume/switch on the output mixer amp */
nid = alc880_idx_to_mixer(alc880_fixed_pin_idx(pin));
- sprintf(name, "%s Playback Volume", pfx);
- err = add_control(spec, ALC_CTL_WIDGET_VOL, name,
+ err = add_pb_vol_ctrl(spec, ALC_CTL_WIDGET_VOL, pfx,
HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT));
if (err < 0)
return err;
- sprintf(name, "%s Playback Switch", pfx);
- err = add_control(spec, ALC_CTL_BIND_MUTE, name,
+ err = add_pb_sw_ctrl(spec, ALC_CTL_BIND_MUTE, pfx,
HDA_COMPOSE_AMP_VAL(nid, 3, 2, HDA_INPUT));
if (err < 0)
return err;
} else if (alc880_is_multi_pin(pin)) {
/* set manual connection */
/* we have only a switch on HP-out PIN */
- sprintf(name, "%s Playback Switch", pfx);
- err = add_control(spec, ALC_CTL_WIDGET_MUTE, name,
+ err = add_pb_sw_ctrl(spec, ALC_CTL_WIDGET_MUTE, pfx,
HDA_COMPOSE_AMP_VAL(pin, 3, 0, HDA_OUTPUT));
if (err < 0)
return err;
@@ -4470,16 +4488,13 @@ static int new_analog_input(struct alc_spec *spec, hda_nid_t pin,
const char *ctlname,
int idx, hda_nid_t mix_nid)
{
- char name[32];
int err;
- sprintf(name, "%s Playback Volume", ctlname);
- err = add_control(spec, ALC_CTL_WIDGET_VOL, name,
+ err = add_pb_vol_ctrl(spec, ALC_CTL_WIDGET_VOL, ctlname,
HDA_COMPOSE_AMP_VAL(mix_nid, 3, idx, HDA_INPUT));
if (err < 0)
return err;
- sprintf(name, "%s Playback Switch", ctlname);
- err = add_control(spec, ALC_CTL_WIDGET_MUTE, name,
+ err = add_pb_sw_ctrl(spec, ALC_CTL_WIDGET_MUTE, ctlname,
HDA_COMPOSE_AMP_VAL(mix_nid, 3, idx, HDA_INPUT));
if (err < 0)
return err;
@@ -4667,9 +4682,9 @@ static int alc880_parse_auto_config(struct hda_codec *codec)
spec->multiout.dig_out_nid = dig_nid;
else {
spec->multiout.slave_dig_outs = spec->slave_dig_outs;
- spec->slave_dig_outs[i - 1] = dig_nid;
- if (i == ARRAY_SIZE(spec->slave_dig_outs) - 1)
+ if (i >= ARRAY_SIZE(spec->slave_dig_outs) - 1)
break;
+ spec->slave_dig_outs[i - 1] = dig_nid;
}
}
if (spec->autocfg.dig_in_pin)
@@ -4756,8 +4771,12 @@ static void set_capture_mixer(struct hda_codec *codec)
}
}
+#ifdef CONFIG_SND_HDA_INPUT_BEEP
#define set_beep_amp(spec, nid, idx, dir) \
((spec)->beep_amp = HDA_COMPOSE_AMP_VAL(nid, 3, idx, dir))
+#else
+#define set_beep_amp(spec, nid, idx, dir) /* NOP */
+#endif
/*
* OK, here we have finally the patch for ALC880
@@ -5070,11 +5089,8 @@ static struct hda_verb alc260_hp_unsol_verbs[] = {
static void alc260_hp_automute(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
- unsigned int present;
- present = snd_hda_codec_read(codec, 0x10, 0,
- AC_VERB_GET_PIN_SENSE, 0);
- spec->jack_present = (present & AC_PINSENSE_PRESENCE) != 0;
+ spec->jack_present = snd_hda_jack_detect(codec, 0x10);
alc260_hp_master_update(codec, 0x0f, 0x10, 0x11);
}
@@ -5139,11 +5155,8 @@ static struct hda_verb alc260_hp_3013_unsol_verbs[] = {
static void alc260_hp_3013_automute(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
- unsigned int present;
- present = snd_hda_codec_read(codec, 0x15, 0,
- AC_VERB_GET_PIN_SENSE, 0);
- spec->jack_present = (present & AC_PINSENSE_PRESENCE) != 0;
+ spec->jack_present = snd_hda_jack_detect(codec, 0x15);
alc260_hp_master_update(codec, 0x15, 0x10, 0x11);
}
@@ -5156,12 +5169,8 @@ static void alc260_hp_3013_unsol_event(struct hda_codec *codec,
static void alc260_hp_3012_automute(struct hda_codec *codec)
{
- unsigned int present, bits;
+ unsigned int bits = snd_hda_jack_detect(codec, 0x10) ? 0 : PIN_OUT;
- present = snd_hda_codec_read(codec, 0x10, 0,
- AC_VERB_GET_PIN_SENSE, 0) & AC_PINSENSE_PRESENCE;
-
- bits = present ? 0 : PIN_OUT;
snd_hda_codec_write(codec, 0x0f, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
bits);
snd_hda_codec_write(codec, 0x11, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
@@ -5731,8 +5740,7 @@ static void alc260_replacer_672v_automute(struct hda_codec *codec)
unsigned int present;
/* speaker --> GPIO Data 0, hp or spdif --> GPIO data 1 */
- present = snd_hda_codec_read(codec, 0x0f, 0,
- AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
+ present = snd_hda_jack_detect(codec, 0x0f);
if (present) {
snd_hda_codec_write_cache(codec, 0x01, 0,
AC_VERB_SET_GPIO_DATA, 1);
@@ -5972,7 +5980,6 @@ static int alc260_add_playback_controls(struct alc_spec *spec, hda_nid_t nid,
{
hda_nid_t nid_vol;
unsigned long vol_val, sw_val;
- char name[32];
int err;
if (nid >= 0x0f && nid < 0x11) {
@@ -5992,14 +5999,12 @@ static int alc260_add_playback_controls(struct alc_spec *spec, hda_nid_t nid,
if (!(*vol_bits & (1 << nid_vol))) {
/* first control for the volume widget */
- snprintf(name, sizeof(name), "%s Playback Volume", pfx);
- err = add_control(spec, ALC_CTL_WIDGET_VOL, name, vol_val);
+ err = add_pb_vol_ctrl(spec, ALC_CTL_WIDGET_VOL, pfx, vol_val);
if (err < 0)
return err;
*vol_bits |= (1 << nid_vol);
}
- snprintf(name, sizeof(name), "%s Playback Switch", pfx);
- err = add_control(spec, ALC_CTL_WIDGET_MUTE, name, sw_val);
+ err = add_pb_sw_ctrl(spec, ALC_CTL_WIDGET_MUTE, pfx, sw_val);
if (err < 0)
return err;
return 1;
@@ -6232,7 +6237,7 @@ static struct snd_pci_quirk alc260_cfg_tbl[] = {
SND_PCI_QUIRK(0x1025, 0x008f, "Acer", ALC260_ACER),
SND_PCI_QUIRK(0x1509, 0x4540, "Favorit 100XS", ALC260_FAVORIT100),
SND_PCI_QUIRK(0x103c, 0x2808, "HP d5700", ALC260_HP_3013),
- SND_PCI_QUIRK(0x103c, 0x280a, "HP d5750", ALC260_HP_3013),
+ SND_PCI_QUIRK(0x103c, 0x280a, "HP d5750", ALC260_AUTO), /* no quirk */
SND_PCI_QUIRK(0x103c, 0x3010, "HP", ALC260_HP_3013),
SND_PCI_QUIRK(0x103c, 0x3011, "HP", ALC260_HP_3013),
SND_PCI_QUIRK(0x103c, 0x3012, "HP", ALC260_HP_DC7600),
@@ -7319,8 +7324,8 @@ static struct snd_kcontrol_new alc882_macpro_mixer[] = {
HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x01, HDA_INPUT),
HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x01, HDA_INPUT),
/* FIXME: this looks suspicious...
- HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x02, HDA_INPUT),
- HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_VOLUME("Beep Playback Volume", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_MUTE("Beep Playback Switch", 0x0b, 0x02, HDA_INPUT),
*/
{ } /* end */
};
@@ -8167,12 +8172,8 @@ static void alc883_mitac_setup(struct hda_codec *codec)
/*
static void alc883_mitac_mic_automute(struct hda_codec *codec)
{
- unsigned int present;
- unsigned char bits;
+ unsigned char bits = snd_hda_jack_detect(codec, 0x18) ? HDA_AMP_MUTE : 0;
- present = snd_hda_codec_read(codec, 0x18, 0,
- AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
- bits = present ? HDA_AMP_MUTE : 0;
snd_hda_codec_amp_stereo(codec, 0x0b, HDA_INPUT, 1, HDA_AMP_MUTE, bits);
}
*/
@@ -8394,10 +8395,8 @@ static struct hda_channel_mode alc888_3st_hp_modes[3] = {
/* toggle front-jack and RCA according to the hp-jack state */
static void alc888_lenovo_ms7195_front_automute(struct hda_codec *codec)
{
- unsigned int present;
+ unsigned int present = snd_hda_jack_detect(codec, 0x1b);
- present = snd_hda_codec_read(codec, 0x1b, 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,
@@ -8407,10 +8406,8 @@ static void alc888_lenovo_ms7195_front_automute(struct hda_codec *codec)
/* toggle RCA according to the front-jack state */
static void alc888_lenovo_ms7195_rca_automute(struct hda_codec *codec)
{
- unsigned int present;
+ unsigned int present = snd_hda_jack_detect(codec, 0x14);
- 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);
}
@@ -8451,8 +8448,7 @@ static void alc883_clevo_m720_mic_automute(struct hda_codec *codec)
{
unsigned int present;
- present = snd_hda_codec_read(codec, 0x18, 0,
- AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
+ present = snd_hda_jack_detect(codec, 0x18);
snd_hda_codec_amp_stereo(codec, 0x0b, HDA_INPUT, 1,
HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
}
@@ -8503,24 +8499,16 @@ static void alc883_haier_w66_setup(struct hda_codec *codec)
static void alc883_lenovo_101e_ispeaker_automute(struct hda_codec *codec)
{
- unsigned int present;
- unsigned char bits;
+ int bits = snd_hda_jack_detect(codec, 0x14) ? HDA_AMP_MUTE : 0;
- present = snd_hda_codec_read(codec, 0x14, 0, AC_VERB_GET_PIN_SENSE, 0)
- & AC_PINSENSE_PRESENCE;
- 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)
{
- unsigned int present;
- unsigned char bits;
+ int bits = snd_hda_jack_detect(codec, 0x1b) ? HDA_AMP_MUTE : 0;
- present = snd_hda_codec_read(codec, 0x1b, 0,
- AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
- 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,
@@ -8671,8 +8659,7 @@ static void alc889A_mb31_automute(struct hda_codec *codec)
/* Mute only in 2ch or 4ch mode */
if (snd_hda_codec_read(codec, 0x15, 0, AC_VERB_GET_CONNECT_SEL, 0)
== 0x00) {
- present = snd_hda_codec_read(codec, 0x15, 0,
- AC_VERB_GET_PIN_SENSE, 0) & AC_PINSENSE_PRESENCE;
+ present = snd_hda_jack_detect(codec, 0x15);
snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0,
HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
snd_hda_codec_amp_stereo(codec, 0x16, HDA_OUTPUT, 0,
@@ -8894,10 +8881,11 @@ static struct snd_pci_quirk alc882_ssid_cfg_tbl[] = {
SND_PCI_QUIRK(0x106b, 0x3800, "MacbookPro 4,1", ALC885_MBP3),
SND_PCI_QUIRK(0x106b, 0x3e00, "iMac 24 Aluminum", ALC885_IMAC24),
SND_PCI_QUIRK(0x106b, 0x3f00, "Macbook 5,1", ALC885_MB5),
- /* FIXME: HP jack sense seems not working for MBP 5,1, so apparently
- * no perfect solution yet
+ /* FIXME: HP jack sense seems not working for MBP 5,1 or 5,2,
+ * so apparently no perfect solution yet
*/
SND_PCI_QUIRK(0x106b, 0x4000, "MacbookPro 5,1", ALC885_MB5),
+ SND_PCI_QUIRK(0x106b, 0x4600, "MacbookPro 5,2", ALC885_MB5),
{} /* terminator */
};
@@ -9593,11 +9581,13 @@ static struct alc_pincfg alc882_abit_aw9d_pinfix[] = {
{ }
};
-static const struct alc_pincfg *alc882_pin_fixes[] = {
- [PINFIX_ABIT_AW9D_MAX] = alc882_abit_aw9d_pinfix,
+static const struct alc_fixup alc882_fixups[] = {
+ [PINFIX_ABIT_AW9D_MAX] = {
+ .pins = alc882_abit_aw9d_pinfix
+ },
};
-static struct snd_pci_quirk alc882_pinfix_tbl[] = {
+static struct snd_pci_quirk alc882_fixup_tbl[] = {
SND_PCI_QUIRK(0x147b, 0x107a, "Abit AW9D-MAX", PINFIX_ABIT_AW9D_MAX),
{}
};
@@ -9794,9 +9784,9 @@ static int alc882_parse_auto_config(struct hda_codec *codec)
spec->multiout.dig_out_nid = dig_nid;
else {
spec->multiout.slave_dig_outs = spec->slave_dig_outs;
- spec->slave_dig_outs[i - 1] = dig_nid;
- if (i == ARRAY_SIZE(spec->slave_dig_outs) - 1)
+ if (i >= ARRAY_SIZE(spec->slave_dig_outs) - 1)
break;
+ spec->slave_dig_outs[i - 1] = dig_nid;
}
}
if (spec->autocfg.dig_in_pin)
@@ -9869,7 +9859,7 @@ static int patch_alc882(struct hda_codec *codec)
board_config = ALC882_AUTO;
}
- alc_fix_pincfg(codec, alc882_pinfix_tbl, alc882_pin_fixes);
+ alc_pick_fixup(codec, alc882_fixup_tbl, alc882_fixups);
if (board_config == ALC882_AUTO) {
/* automatic parse from the BIOS config */
@@ -10012,10 +10002,8 @@ static void alc262_hp_master_update(struct hda_codec *codec)
static void alc262_hp_bpc_automute(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
- unsigned int presence;
- presence = snd_hda_codec_read(codec, 0x1b, 0,
- AC_VERB_GET_PIN_SENSE, 0);
- spec->jack_present = !!(presence & AC_PINSENSE_PRESENCE);
+
+ spec->jack_present = snd_hda_jack_detect(codec, 0x1b);
alc262_hp_master_update(codec);
}
@@ -10029,10 +10017,8 @@ static void alc262_hp_bpc_unsol_event(struct hda_codec *codec, unsigned int res)
static void alc262_hp_wildwest_automute(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
- unsigned int presence;
- presence = snd_hda_codec_read(codec, 0x15, 0,
- AC_VERB_GET_PIN_SENSE, 0);
- spec->jack_present = !!(presence & AC_PINSENSE_PRESENCE);
+
+ spec->jack_present = snd_hda_jack_detect(codec, 0x15);
alc262_hp_master_update(codec);
}
@@ -10266,13 +10252,8 @@ static void alc262_hippo_automute(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
hda_nid_t hp_nid = spec->autocfg.hp_pins[0];
- unsigned int present;
- /* 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;
+ spec->jack_present = snd_hda_jack_detect(codec, hp_nid);
alc262_hippo_master_update(codec);
}
@@ -10598,21 +10579,8 @@ static void alc262_fujitsu_automute(struct hda_codec *codec, int force)
unsigned int mute;
if (force || !spec->sense_updated) {
- unsigned int present;
- /* need to execute and sync at first */
- snd_hda_codec_read(codec, 0x14, 0, AC_VERB_SET_PIN_SENSE, 0);
- /* check laptop HP jack */
- present = snd_hda_codec_read(codec, 0x14, 0,
- AC_VERB_GET_PIN_SENSE, 0);
- /* need to execute and sync at first */
- snd_hda_codec_read(codec, 0x1b, 0, AC_VERB_SET_PIN_SENSE, 0);
- /* check docking HP jack */
- present |= snd_hda_codec_read(codec, 0x1b, 0,
- AC_VERB_GET_PIN_SENSE, 0);
- if (present & AC_PINSENSE_PRESENCE)
- spec->jack_present = 1;
- else
- spec->jack_present = 0;
+ spec->jack_present = snd_hda_jack_detect(codec, 0x14) ||
+ snd_hda_jack_detect(codec, 0x1b);
spec->sense_updated = 1;
}
/* unmute internal speaker only if both HPs are unplugged and
@@ -10657,12 +10625,7 @@ static void alc262_lenovo_3000_automute(struct hda_codec *codec, int force)
unsigned int mute;
if (force || !spec->sense_updated) {
- unsigned int present_int_hp;
- /* need to execute and sync at first */
- snd_hda_codec_read(codec, 0x1b, 0, AC_VERB_SET_PIN_SENSE, 0);
- present_int_hp = snd_hda_codec_read(codec, 0x1b, 0,
- AC_VERB_GET_PIN_SENSE, 0);
- spec->jack_present = (present_int_hp & 0x80000000) != 0;
+ spec->jack_present = snd_hda_jack_detect(codec, 0x1b);
spec->sense_updated = 1;
}
if (spec->jack_present) {
@@ -10854,12 +10817,7 @@ static void alc262_ultra_automute(struct hda_codec *codec)
mute = 0;
/* auto-mute only when HP is used as HP */
if (!spec->cur_mux[0]) {
- 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 & AC_PINSENSE_PRESENCE) != 0;
+ spec->jack_present = snd_hda_jack_detect(codec, 0x15);
if (spec->jack_present)
mute = HDA_AMP_MUTE;
}
@@ -10936,7 +10894,6 @@ static int alc262_check_volbit(hda_nid_t nid)
static int alc262_add_out_vol_ctl(struct alc_spec *spec, hda_nid_t nid,
const char *pfx, int *vbits)
{
- char name[32];
unsigned long val;
int vbit;
@@ -10946,28 +10903,25 @@ static int alc262_add_out_vol_ctl(struct alc_spec *spec, hda_nid_t nid,
if (*vbits & vbit) /* a volume control for this mixer already there */
return 0;
*vbits |= vbit;
- snprintf(name, sizeof(name), "%s Playback Volume", pfx);
if (vbit == 2)
val = HDA_COMPOSE_AMP_VAL(0x0e, 2, 0, HDA_OUTPUT);
else
val = HDA_COMPOSE_AMP_VAL(0x0c, 3, 0, HDA_OUTPUT);
- return add_control(spec, ALC_CTL_WIDGET_VOL, name, val);
+ return add_pb_vol_ctrl(spec, ALC_CTL_WIDGET_VOL, pfx, val);
}
static int alc262_add_out_sw_ctl(struct alc_spec *spec, hda_nid_t nid,
const char *pfx)
{
- char name[32];
unsigned long val;
if (!nid)
return 0;
- snprintf(name, sizeof(name), "%s Playback Switch", pfx);
if (nid == 0x16)
val = HDA_COMPOSE_AMP_VAL(nid, 2, 0, HDA_OUTPUT);
else
val = HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT);
- return add_control(spec, ALC_CTL_WIDGET_MUTE, name, val);
+ return add_pb_sw_ctrl(spec, ALC_CTL_WIDGET_MUTE, pfx, val);
}
/* add playback controls from the parsed DAC table */
@@ -11441,8 +11395,12 @@ static struct snd_pci_quirk alc262_cfg_tbl[] = {
SND_PCI_QUIRK(0x104d, 0x820f, "Sony ASSAMD", ALC262_SONY_ASSAMD),
SND_PCI_QUIRK(0x104d, 0x9016, "Sony VAIO", ALC262_AUTO), /* dig-only */
SND_PCI_QUIRK(0x104d, 0x9025, "Sony VAIO Z21MN", ALC262_TOSHIBA_S06),
+ SND_PCI_QUIRK(0x104d, 0x9035, "Sony VAIO VGN-FW170J", ALC262_AUTO),
+ SND_PCI_QUIRK(0x104d, 0x9047, "Sony VAIO Type G", ALC262_AUTO),
+#if 0 /* disable the quirk since model=auto works better in recent versions */
SND_PCI_QUIRK_MASK(0x104d, 0xff00, 0x9000, "Sony VAIO",
ALC262_SONY_ASSAMD),
+#endif
SND_PCI_QUIRK(0x1179, 0x0001, "Toshiba dynabook SS RX1",
ALC262_TOSHIBA_RX1),
SND_PCI_QUIRK(0x1179, 0xff7b, "Toshiba S06", ALC262_TOSHIBA_S06),
@@ -11901,10 +11859,7 @@ static void alc268_acer_automute(struct hda_codec *codec, int force)
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->jack_present = snd_hda_jack_detect(codec, 0x14);
spec->sense_updated = 1;
}
if (spec->jack_present)
@@ -12023,8 +11978,7 @@ static void alc268_aspire_one_speaker_automute(struct hda_codec *codec)
unsigned int present;
unsigned char bits;
- present = snd_hda_codec_read(codec, 0x15, 0,
- AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
+ present = snd_hda_jack_detect(codec, 0x15);
bits = present ? AMP_IN_MUTE(0) : 0;
snd_hda_codec_amp_stereo(codec, 0x0f, HDA_INPUT, 0,
AMP_IN_MUTE(0), bits);
@@ -12305,11 +12259,9 @@ static struct snd_kcontrol_new alc268_test_mixer[] = {
static int alc268_new_analog_output(struct alc_spec *spec, hda_nid_t nid,
const char *ctlname, int idx)
{
- char name[32];
hda_nid_t dac;
int err;
- sprintf(name, "%s Playback Volume", ctlname);
switch (nid) {
case 0x14:
case 0x16:
@@ -12323,7 +12275,7 @@ static int alc268_new_analog_output(struct alc_spec *spec, hda_nid_t nid,
}
if (spec->multiout.dac_nids[0] != dac &&
spec->multiout.dac_nids[1] != dac) {
- err = add_control(spec, ALC_CTL_WIDGET_VOL, name,
+ err = add_pb_vol_ctrl(spec, ALC_CTL_WIDGET_VOL, ctlname,
HDA_COMPOSE_AMP_VAL(dac, 3, idx,
HDA_OUTPUT));
if (err < 0)
@@ -12331,12 +12283,11 @@ static int alc268_new_analog_output(struct alc_spec *spec, hda_nid_t nid,
spec->multiout.dac_nids[spec->multiout.num_dacs++] = dac;
}
- sprintf(name, "%s Playback Switch", ctlname);
if (nid != 0x16)
- err = add_control(spec, ALC_CTL_WIDGET_MUTE, name,
+ err = add_pb_sw_ctrl(spec, ALC_CTL_WIDGET_MUTE, ctlname,
HDA_COMPOSE_AMP_VAL(nid, 3, idx, HDA_OUTPUT));
else /* mono */
- err = add_control(spec, ALC_CTL_WIDGET_MUTE, name,
+ err = add_pb_sw_ctrl(spec, ALC_CTL_WIDGET_MUTE, ctlname,
HDA_COMPOSE_AMP_VAL(nid, 2, idx, HDA_OUTPUT));
if (err < 0)
return err;
@@ -12366,8 +12317,7 @@ static int alc268_auto_create_multi_out_ctls(struct alc_spec *spec,
nid = cfg->speaker_pins[0];
if (nid == 0x1d) {
- err = add_control(spec, ALC_CTL_WIDGET_VOL,
- "Speaker Playback Volume",
+ err = add_pb_vol_ctrl(spec, ALC_CTL_WIDGET_VOL, "Speaker",
HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_INPUT));
if (err < 0)
return err;
@@ -12385,8 +12335,7 @@ static int alc268_auto_create_multi_out_ctls(struct alc_spec *spec,
nid = cfg->line_out_pins[1] | cfg->line_out_pins[2];
if (nid == 0x16) {
- err = add_control(spec, ALC_CTL_WIDGET_MUTE,
- "Mono Playback Switch",
+ err = add_pb_sw_ctrl(spec, ALC_CTL_WIDGET_MUTE, "Mono",
HDA_COMPOSE_AMP_VAL(nid, 2, 0, HDA_OUTPUT));
if (err < 0)
return err;
@@ -12585,7 +12534,8 @@ static struct snd_pci_quirk alc268_cfg_tbl[] = {
SND_PCI_QUIRK(0x1025, 0x015b, "Acer Aspire One",
ALC268_ACER_ASPIRE_ONE),
SND_PCI_QUIRK(0x1028, 0x0253, "Dell OEM", ALC268_DELL),
- SND_PCI_QUIRK(0x1028, 0x02b0, "Dell Inspiron Mini9", ALC268_DELL),
+ SND_PCI_QUIRK_MASK(0x1028, 0xfff0, 0x02b0,
+ "Dell Inspiron Mini9/Vostro A90", ALC268_DELL),
/* almost compatible with toshiba but with optional digital outs;
* auto-probing seems working fine
*/
@@ -12660,7 +12610,7 @@ static struct alc_config_preset alc268_presets[] = {
.init_hook = alc268_toshiba_automute,
},
[ALC268_ACER] = {
- .mixers = { alc268_acer_mixer, alc268_capture_nosrc_mixer,
+ .mixers = { alc268_acer_mixer, alc268_capture_alt_mixer,
alc268_beep_mixer },
.init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
alc268_acer_verbs },
@@ -12842,12 +12792,15 @@ static int patch_alc268(struct hda_codec *codec)
unsigned int wcap = get_wcaps(codec, 0x07);
int i;
+ spec->capsrc_nids = alc268_capsrc_nids;
/* get type */
wcap = get_wcaps_type(wcap);
if (spec->auto_mic ||
wcap != AC_WID_AUD_IN || spec->input_mux->num_items == 1) {
spec->adc_nids = alc268_adc_nids_alt;
spec->num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt);
+ if (spec->auto_mic)
+ fixup_automic_adc(codec);
if (spec->auto_mic || spec->input_mux->num_items == 1)
add_mixer(spec, alc268_capture_nosrc_mixer);
else
@@ -12857,7 +12810,6 @@ static int patch_alc268(struct hda_codec *codec)
spec->num_adc_nids = ARRAY_SIZE(alc268_adc_nids);
add_mixer(spec, alc268_capture_mixer);
}
- spec->capsrc_nids = alc268_capsrc_nids;
/* set default input source */
for (i = 0; i < spec->num_adc_nids; i++)
snd_hda_codec_write_cache(codec, alc268_capsrc_nids[i],
@@ -13009,8 +12961,7 @@ static void alc269_quanta_fl1_speaker_automute(struct hda_codec *codec)
unsigned int present;
unsigned char bits;
- present = snd_hda_codec_read(codec, 0x15, 0,
- AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
+ present = snd_hda_jack_detect(codec, 0x15);
bits = present ? AMP_IN_MUTE(0) : 0;
snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0,
AMP_IN_MUTE(0), bits);
@@ -13035,12 +12986,10 @@ static void alc269_lifebook_speaker_automute(struct hda_codec *codec)
unsigned char bits;
/* Check laptop headphone socket */
- present = snd_hda_codec_read(codec, 0x15, 0,
- AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
+ present = snd_hda_jack_detect(codec, 0x15);
/* Check port replicator headphone socket */
- present |= snd_hda_codec_read(codec, 0x1a, 0,
- AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
+ present |= snd_hda_jack_detect(codec, 0x1a);
bits = present ? AMP_IN_MUTE(0) : 0;
snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0,
@@ -13064,11 +13013,8 @@ static void alc269_lifebook_mic_autoswitch(struct hda_codec *codec)
unsigned int present_laptop;
unsigned int present_dock;
- present_laptop = snd_hda_codec_read(codec, 0x18, 0,
- AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
-
- present_dock = snd_hda_codec_read(codec, 0x1b, 0,
- AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
+ present_laptop = snd_hda_jack_detect(codec, 0x18);
+ present_dock = snd_hda_jack_detect(codec, 0x1b);
/* Laptop mic port overrides dock mic port, design decision */
if (present_dock)
@@ -13153,8 +13099,7 @@ static void alc269_speaker_automute(struct hda_codec *codec)
unsigned int present;
unsigned char bits;
- present = snd_hda_codec_read(codec, 0x15, 0,
- AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
+ present = snd_hda_jack_detect(codec, 0x15);
bits = present ? AMP_IN_MUTE(0) : 0;
snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0,
AMP_IN_MUTE(0), bits);
@@ -14132,10 +14077,8 @@ static struct hda_verb alc861_toshiba_init_verbs[] = {
/* toggle speaker-output according to the hp-jack state */
static void alc861_toshiba_automute(struct hda_codec *codec)
{
- unsigned int present;
+ unsigned int present = snd_hda_jack_detect(codec, 0x0f);
- present = snd_hda_codec_read(codec, 0x0f, 0,
- AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
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,
@@ -14235,9 +14178,7 @@ static int alc861_auto_fill_dac_nids(struct hda_codec *codec,
static int alc861_create_out_sw(struct hda_codec *codec, const char *pfx,
hda_nid_t nid, unsigned int chs)
{
- char name[32];
- snprintf(name, sizeof(name), "%s Playback Switch", pfx);
- return add_control(codec->spec, ALC_CTL_WIDGET_MUTE, name,
+ return add_pb_sw_ctrl(codec->spec, ALC_CTL_WIDGET_MUTE, pfx,
HDA_COMPOSE_AMP_VAL(nid, chs, 0, HDA_OUTPUT));
}
@@ -14357,15 +14298,16 @@ static void alc861_auto_init_multi_out(struct hda_codec *codec)
static void alc861_auto_init_hp_out(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
- hda_nid_t pin;
- pin = spec->autocfg.hp_pins[0];
- if (pin)
- alc861_auto_set_output_and_unmute(codec, pin, PIN_HP,
+ if (spec->autocfg.hp_outs)
+ alc861_auto_set_output_and_unmute(codec,
+ spec->autocfg.hp_pins[0],
+ PIN_HP,
spec->multiout.hp_nid);
- pin = spec->autocfg.speaker_pins[0];
- if (pin)
- alc861_auto_set_output_and_unmute(codec, pin, PIN_OUT,
+ if (spec->autocfg.speaker_outs)
+ alc861_auto_set_output_and_unmute(codec,
+ spec->autocfg.speaker_pins[0],
+ PIN_OUT,
spec->multiout.dac_nids[0]);
}
@@ -14601,6 +14543,27 @@ static struct alc_config_preset alc861_presets[] = {
},
};
+/* Pin config fixes */
+enum {
+ PINFIX_FSC_AMILO_PI1505,
+};
+
+static struct alc_pincfg alc861_fsc_amilo_pi1505_pinfix[] = {
+ { 0x0b, 0x0221101f }, /* HP */
+ { 0x0f, 0x90170310 }, /* speaker */
+ { }
+};
+
+static const struct alc_fixup alc861_fixups[] = {
+ [PINFIX_FSC_AMILO_PI1505] = {
+ .pins = alc861_fsc_amilo_pi1505_pinfix
+ },
+};
+
+static struct snd_pci_quirk alc861_fixup_tbl[] = {
+ SND_PCI_QUIRK(0x1734, 0x10c7, "FSC Amilo Pi1505", PINFIX_FSC_AMILO_PI1505),
+ {}
+};
static int patch_alc861(struct hda_codec *codec)
{
@@ -14624,6 +14587,8 @@ static int patch_alc861(struct hda_codec *codec)
board_config = ALC861_AUTO;
}
+ alc_pick_fixup(codec, alc861_fixup_tbl, alc861_fixups);
+
if (board_config == ALC861_AUTO) {
/* automatic parse from the BIOS config */
err = alc861_parse_auto_config(codec);
@@ -15041,9 +15006,9 @@ static void alc861vd_lenovo_mic_automute(struct hda_codec *codec)
unsigned int present;
unsigned char bits;
- present = snd_hda_codec_read(codec, 0x18, 0,
- AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
+ present = snd_hda_jack_detect(codec, 0x18);
bits = present ? HDA_AMP_MUTE : 0;
+
snd_hda_codec_amp_stereo(codec, 0x0b, HDA_INPUT, 1,
HDA_AMP_MUTE, bits);
}
@@ -15158,7 +15123,7 @@ static struct snd_pci_quirk alc861vd_cfg_tbl[] = {
SND_PCI_QUIRK(0x1019, 0xa88d, "Realtek ALC660 demo", ALC660VD_3ST),
SND_PCI_QUIRK(0x103c, 0x30bf, "HP TX1000", ALC861VD_HP),
SND_PCI_QUIRK(0x1043, 0x12e2, "Asus z35m", ALC660VD_3ST),
- SND_PCI_QUIRK(0x1043, 0x1339, "Asus G1", ALC660VD_3ST),
+ /*SND_PCI_QUIRK(0x1043, 0x1339, "Asus G1", ALC660VD_3ST),*/ /* auto */
SND_PCI_QUIRK(0x1043, 0x1633, "Asus V1Sn", ALC660VD_ASUS_V1S),
SND_PCI_QUIRK(0x1043, 0x81e7, "ASUS", ALC660VD_3ST_DIG),
SND_PCI_QUIRK(0x10de, 0x03f0, "Realtek ALC660 demo", ALC660VD_3ST),
@@ -15360,7 +15325,6 @@ static void alc861vd_auto_init_analog_input(struct hda_codec *codec)
static int alc861vd_auto_create_multi_out_ctls(struct alc_spec *spec,
const struct auto_pin_cfg *cfg)
{
- char name[32];
static const char *chname[4] = {"Front", "Surround", "CLFE", "Side"};
hda_nid_t nid_v, nid_s;
int i, err;
@@ -15377,26 +15341,26 @@ static int alc861vd_auto_create_multi_out_ctls(struct alc_spec *spec,
if (i == 2) {
/* Center/LFE */
- err = add_control(spec, ALC_CTL_WIDGET_VOL,
- "Center Playback Volume",
+ err = add_pb_vol_ctrl(spec, ALC_CTL_WIDGET_VOL,
+ "Center",
HDA_COMPOSE_AMP_VAL(nid_v, 1, 0,
HDA_OUTPUT));
if (err < 0)
return err;
- err = add_control(spec, ALC_CTL_WIDGET_VOL,
- "LFE Playback Volume",
+ err = add_pb_vol_ctrl(spec, ALC_CTL_WIDGET_VOL,
+ "LFE",
HDA_COMPOSE_AMP_VAL(nid_v, 2, 0,
HDA_OUTPUT));
if (err < 0)
return err;
- err = add_control(spec, ALC_CTL_BIND_MUTE,
- "Center Playback Switch",
+ err = add_pb_sw_ctrl(spec, ALC_CTL_BIND_MUTE,
+ "Center",
HDA_COMPOSE_AMP_VAL(nid_s, 1, 2,
HDA_INPUT));
if (err < 0)
return err;
- err = add_control(spec, ALC_CTL_BIND_MUTE,
- "LFE Playback Switch",
+ err = add_pb_sw_ctrl(spec, ALC_CTL_BIND_MUTE,
+ "LFE",
HDA_COMPOSE_AMP_VAL(nid_s, 2, 2,
HDA_INPUT));
if (err < 0)
@@ -15411,8 +15375,7 @@ static int alc861vd_auto_create_multi_out_ctls(struct alc_spec *spec,
pfx = "PCM";
} else
pfx = chname[i];
- sprintf(name, "%s Playback Volume", pfx);
- err = add_control(spec, ALC_CTL_WIDGET_VOL, name,
+ err = add_pb_vol_ctrl(spec, ALC_CTL_WIDGET_VOL, pfx,
HDA_COMPOSE_AMP_VAL(nid_v, 3, 0,
HDA_OUTPUT));
if (err < 0)
@@ -15420,8 +15383,7 @@ static int alc861vd_auto_create_multi_out_ctls(struct alc_spec *spec,
if (cfg->line_outs == 1 &&
cfg->line_out_type == AUTO_PIN_SPEAKER_OUT)
pfx = "Speaker";
- sprintf(name, "%s Playback Switch", pfx);
- err = add_control(spec, ALC_CTL_BIND_MUTE, name,
+ err = add_pb_sw_ctrl(spec, ALC_CTL_BIND_MUTE, pfx,
HDA_COMPOSE_AMP_VAL(nid_s, 3, 2,
HDA_INPUT));
if (err < 0)
@@ -15439,7 +15401,6 @@ static int alc861vd_auto_create_extra_out(struct alc_spec *spec,
{
hda_nid_t nid_v, nid_s;
int err;
- char name[32];
if (!pin)
return 0;
@@ -15457,21 +15418,18 @@ static int alc861vd_auto_create_extra_out(struct alc_spec *spec,
nid_s = alc861vd_idx_to_mixer_switch(
alc880_fixed_pin_idx(pin));
- sprintf(name, "%s Playback Volume", pfx);
- err = add_control(spec, ALC_CTL_WIDGET_VOL, name,
+ err = add_pb_vol_ctrl(spec, ALC_CTL_WIDGET_VOL, pfx,
HDA_COMPOSE_AMP_VAL(nid_v, 3, 0, HDA_OUTPUT));
if (err < 0)
return err;
- sprintf(name, "%s Playback Switch", pfx);
- err = add_control(spec, ALC_CTL_BIND_MUTE, name,
+ err = add_pb_sw_ctrl(spec, ALC_CTL_BIND_MUTE, pfx,
HDA_COMPOSE_AMP_VAL(nid_s, 3, 2, HDA_INPUT));
if (err < 0)
return err;
} else if (alc880_is_multi_pin(pin)) {
/* set manual connection */
/* we have only a switch on HP-out PIN */
- sprintf(name, "%s Playback Switch", pfx);
- err = add_control(spec, ALC_CTL_WIDGET_MUTE, name,
+ err = add_pb_sw_ctrl(spec, ALC_CTL_WIDGET_MUTE, pfx,
HDA_COMPOSE_AMP_VAL(pin, 3, 0, HDA_OUTPUT));
if (err < 0)
return err;
@@ -15551,6 +15509,29 @@ static void alc861vd_auto_init(struct hda_codec *codec)
alc_inithook(codec);
}
+enum {
+ ALC660VD_FIX_ASUS_GPIO1
+};
+
+/* reset GPIO1 */
+static const struct hda_verb alc660vd_fix_asus_gpio1_verbs[] = {
+ {0x01, AC_VERB_SET_GPIO_MASK, 0x03},
+ {0x01, AC_VERB_SET_GPIO_DIRECTION, 0x01},
+ {0x01, AC_VERB_SET_GPIO_DATA, 0x01},
+ { }
+};
+
+static const struct alc_fixup alc861vd_fixups[] = {
+ [ALC660VD_FIX_ASUS_GPIO1] = {
+ .verbs = alc660vd_fix_asus_gpio1_verbs,
+ },
+};
+
+static struct snd_pci_quirk alc861vd_fixup_tbl[] = {
+ SND_PCI_QUIRK(0x1043, 0x1339, "ASUS A7-K", ALC660VD_FIX_ASUS_GPIO1),
+ {}
+};
+
static int patch_alc861vd(struct hda_codec *codec)
{
struct alc_spec *spec;
@@ -15572,6 +15553,8 @@ static int patch_alc861vd(struct hda_codec *codec)
board_config = ALC861VD_AUTO;
}
+ alc_pick_fixup(codec, alc861vd_fixup_tbl, alc861vd_fixups);
+
if (board_config == ALC861VD_AUTO) {
/* automatic parse from the BIOS config */
err = alc861vd_parse_auto_config(codec);
@@ -16336,9 +16319,9 @@ static void alc662_lenovo_101e_ispeaker_automute(struct hda_codec *codec)
unsigned int present;
unsigned char bits;
- present = snd_hda_codec_read(codec, 0x14, 0,
- AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
+ present = snd_hda_jack_detect(codec, 0x14);
bits = present ? HDA_AMP_MUTE : 0;
+
snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0,
HDA_AMP_MUTE, bits);
}
@@ -16348,9 +16331,9 @@ static void alc662_lenovo_101e_all_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;
+ present = snd_hda_jack_detect(codec, 0x1b);
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,
@@ -16409,9 +16392,7 @@ static void alc663_m51va_speaker_automute(struct hda_codec *codec)
unsigned int present;
unsigned char bits;
- present = snd_hda_codec_read(codec, 0x21, 0,
- AC_VERB_GET_PIN_SENSE, 0)
- & AC_PINSENSE_PRESENCE;
+ present = snd_hda_jack_detect(codec, 0x21);
bits = present ? HDA_AMP_MUTE : 0;
snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0,
AMP_IN_MUTE(0), bits);
@@ -16424,9 +16405,7 @@ static void alc663_21jd_two_speaker_automute(struct hda_codec *codec)
unsigned int present;
unsigned char bits;
- present = snd_hda_codec_read(codec, 0x21, 0,
- AC_VERB_GET_PIN_SENSE, 0)
- & AC_PINSENSE_PRESENCE;
+ present = snd_hda_jack_detect(codec, 0x21);
bits = present ? HDA_AMP_MUTE : 0;
snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0,
AMP_IN_MUTE(0), bits);
@@ -16443,9 +16422,7 @@ static void alc663_15jd_two_speaker_automute(struct hda_codec *codec)
unsigned int present;
unsigned char bits;
- present = snd_hda_codec_read(codec, 0x15, 0,
- AC_VERB_GET_PIN_SENSE, 0)
- & AC_PINSENSE_PRESENCE;
+ present = snd_hda_jack_detect(codec, 0x15);
bits = present ? HDA_AMP_MUTE : 0;
snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0,
AMP_IN_MUTE(0), bits);
@@ -16462,9 +16439,7 @@ static void alc662_f5z_speaker_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)
- & AC_PINSENSE_PRESENCE;
+ present = snd_hda_jack_detect(codec, 0x1b);
bits = present ? 0 : PIN_OUT;
snd_hda_codec_write(codec, 0x14, 0,
AC_VERB_SET_PIN_WIDGET_CONTROL, bits);
@@ -16474,12 +16449,8 @@ static void alc663_two_hp_m1_speaker_automute(struct hda_codec *codec)
{
unsigned int present1, present2;
- present1 = snd_hda_codec_read(codec, 0x21, 0,
- AC_VERB_GET_PIN_SENSE, 0)
- & AC_PINSENSE_PRESENCE;
- present2 = snd_hda_codec_read(codec, 0x15, 0,
- AC_VERB_GET_PIN_SENSE, 0)
- & AC_PINSENSE_PRESENCE;
+ present1 = snd_hda_jack_detect(codec, 0x21);
+ present2 = snd_hda_jack_detect(codec, 0x15);
if (present1 || present2) {
snd_hda_codec_write_cache(codec, 0x14, 0,
@@ -16494,12 +16465,8 @@ static void alc663_two_hp_m2_speaker_automute(struct hda_codec *codec)
{
unsigned int present1, present2;
- present1 = snd_hda_codec_read(codec, 0x1b, 0,
- AC_VERB_GET_PIN_SENSE, 0)
- & AC_PINSENSE_PRESENCE;
- present2 = snd_hda_codec_read(codec, 0x15, 0,
- AC_VERB_GET_PIN_SENSE, 0)
- & AC_PINSENSE_PRESENCE;
+ present1 = snd_hda_jack_detect(codec, 0x1b);
+ present2 = snd_hda_jack_detect(codec, 0x15);
if (present1 || present2) {
snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0,
@@ -16659,9 +16626,7 @@ static void alc663_g71v_hp_automute(struct hda_codec *codec)
unsigned int present;
unsigned char bits;
- present = snd_hda_codec_read(codec, 0x21, 0,
- AC_VERB_GET_PIN_SENSE, 0)
- & AC_PINSENSE_PRESENCE;
+ present = snd_hda_jack_detect(codec, 0x21);
bits = present ? HDA_AMP_MUTE : 0;
snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0,
HDA_AMP_MUTE, bits);
@@ -16674,9 +16639,7 @@ static void alc663_g71v_front_automute(struct hda_codec *codec)
unsigned int present;
unsigned char bits;
- present = snd_hda_codec_read(codec, 0x15, 0,
- AC_VERB_GET_PIN_SENSE, 0)
- & AC_PINSENSE_PRESENCE;
+ present = snd_hda_jack_detect(codec, 0x15);
bits = present ? HDA_AMP_MUTE : 0;
snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0,
HDA_AMP_MUTE, bits);
@@ -16852,6 +16815,7 @@ static struct snd_pci_quirk alc662_cfg_tbl[] = {
SND_PCI_QUIRK(0x105b, 0x0cd6, "Foxconn", ALC662_ECS),
SND_PCI_QUIRK(0x105b, 0x0d47, "Foxconn 45CMX/45GMX/45CMX-K",
ALC662_3ST_6ch_DIG),
+ SND_PCI_QUIRK(0x1179, 0xff6e, "Toshiba NB200", ALC663_ASUS_MODE4),
SND_PCI_QUIRK(0x144d, 0xca00, "Samsung NC10", ALC272_SAMSUNG_NC10),
SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte 945GCM-S2L",
ALC662_3ST_6ch_DIG),
@@ -17145,70 +17109,141 @@ static struct alc_config_preset alc662_presets[] = {
* BIOS auto configuration
*/
+/* convert from MIX nid to DAC */
+static inline hda_nid_t alc662_mix_to_dac(hda_nid_t nid)
+{
+ if (nid == 0x0f)
+ return 0x02;
+ else if (nid >= 0x0c && nid <= 0x0e)
+ return nid - 0x0c + 0x02;
+ else
+ return 0;
+}
+
+/* get MIX nid connected to the given pin targeted to DAC */
+static hda_nid_t alc662_dac_to_mix(struct hda_codec *codec, hda_nid_t pin,
+ hda_nid_t dac)
+{
+ hda_nid_t mix[4];
+ int i, num;
+
+ num = snd_hda_get_connections(codec, pin, mix, ARRAY_SIZE(mix));
+ for (i = 0; i < num; i++) {
+ if (alc662_mix_to_dac(mix[i]) == dac)
+ return mix[i];
+ }
+ return 0;
+}
+
+/* look for an empty DAC slot */
+static hda_nid_t alc662_look_for_dac(struct hda_codec *codec, hda_nid_t pin)
+{
+ struct alc_spec *spec = codec->spec;
+ hda_nid_t srcs[5];
+ int i, j, num;
+
+ num = snd_hda_get_connections(codec, pin, srcs, ARRAY_SIZE(srcs));
+ if (num < 0)
+ return 0;
+ for (i = 0; i < num; i++) {
+ hda_nid_t nid = alc662_mix_to_dac(srcs[i]);
+ if (!nid)
+ continue;
+ for (j = 0; j < spec->multiout.num_dacs; j++)
+ if (spec->multiout.dac_nids[j] == nid)
+ break;
+ if (j >= spec->multiout.num_dacs)
+ return nid;
+ }
+ return 0;
+}
+
+/* fill in the dac_nids table from the parsed pin configuration */
+static int alc662_auto_fill_dac_nids(struct hda_codec *codec,
+ const struct auto_pin_cfg *cfg)
+{
+ struct alc_spec *spec = codec->spec;
+ int i;
+ hda_nid_t dac;
+
+ spec->multiout.dac_nids = spec->private_dac_nids;
+ for (i = 0; i < cfg->line_outs; i++) {
+ dac = alc662_look_for_dac(codec, cfg->line_out_pins[i]);
+ if (!dac)
+ continue;
+ spec->multiout.dac_nids[spec->multiout.num_dacs++] = dac;
+ }
+ return 0;
+}
+
+static inline int alc662_add_vol_ctl(struct alc_spec *spec, const char *pfx,
+ hda_nid_t nid, unsigned int chs)
+{
+ return add_pb_vol_ctrl(spec, ALC_CTL_WIDGET_VOL, pfx,
+ HDA_COMPOSE_AMP_VAL(nid, chs, 0, HDA_OUTPUT));
+}
+
+static inline int alc662_add_sw_ctl(struct alc_spec *spec, const char *pfx,
+ hda_nid_t nid, unsigned int chs)
+{
+ return add_pb_sw_ctrl(spec, ALC_CTL_WIDGET_MUTE, pfx,
+ HDA_COMPOSE_AMP_VAL(nid, chs, 0, HDA_INPUT));
+}
+
+#define alc662_add_stereo_vol(spec, pfx, nid) \
+ alc662_add_vol_ctl(spec, pfx, nid, 3)
+#define alc662_add_stereo_sw(spec, pfx, nid) \
+ alc662_add_sw_ctl(spec, pfx, nid, 3)
+
/* add playback controls from the parsed DAC table */
-static int alc662_auto_create_multi_out_ctls(struct alc_spec *spec,
+static int alc662_auto_create_multi_out_ctls(struct hda_codec *codec,
const struct auto_pin_cfg *cfg)
{
- char name[32];
+ struct alc_spec *spec = codec->spec;
static const char *chname[4] = {
"Front", "Surround", NULL /*CLFE*/, "Side"
};
- hda_nid_t nid;
+ hda_nid_t nid, mix;
int i, err;
for (i = 0; i < cfg->line_outs; i++) {
- if (!spec->multiout.dac_nids[i])
+ nid = spec->multiout.dac_nids[i];
+ if (!nid)
+ continue;
+ mix = alc662_dac_to_mix(codec, cfg->line_out_pins[i], nid);
+ if (!mix)
continue;
- nid = alc880_idx_to_dac(i);
if (i == 2) {
/* Center/LFE */
- err = add_control(spec, ALC_CTL_WIDGET_VOL,
- "Center Playback Volume",
- HDA_COMPOSE_AMP_VAL(nid, 1, 0,
- HDA_OUTPUT));
+ err = alc662_add_vol_ctl(spec, "Center", nid, 1);
if (err < 0)
return err;
- err = add_control(spec, ALC_CTL_WIDGET_VOL,
- "LFE Playback Volume",
- HDA_COMPOSE_AMP_VAL(nid, 2, 0,
- HDA_OUTPUT));
+ err = alc662_add_vol_ctl(spec, "LFE", nid, 2);
if (err < 0)
return err;
- err = add_control(spec, ALC_CTL_WIDGET_MUTE,
- "Center Playback Switch",
- HDA_COMPOSE_AMP_VAL(0x0e, 1, 0,
- HDA_INPUT));
+ err = alc662_add_sw_ctl(spec, "Center", mix, 1);
if (err < 0)
return err;
- err = add_control(spec, ALC_CTL_WIDGET_MUTE,
- "LFE Playback Switch",
- HDA_COMPOSE_AMP_VAL(0x0e, 2, 0,
- HDA_INPUT));
+ err = alc662_add_sw_ctl(spec, "LFE", mix, 2);
if (err < 0)
return err;
} else {
const char *pfx;
if (cfg->line_outs == 1 &&
cfg->line_out_type == AUTO_PIN_SPEAKER_OUT) {
- if (!cfg->hp_pins)
+ if (cfg->hp_outs)
pfx = "Speaker";
else
pfx = "PCM";
} else
pfx = chname[i];
- sprintf(name, "%s Playback Volume", pfx);
- err = add_control(spec, ALC_CTL_WIDGET_VOL, name,
- HDA_COMPOSE_AMP_VAL(nid, 3, 0,
- HDA_OUTPUT));
+ err = alc662_add_vol_ctl(spec, pfx, nid, 3);
if (err < 0)
return err;
if (cfg->line_outs == 1 &&
cfg->line_out_type == AUTO_PIN_SPEAKER_OUT)
pfx = "Speaker";
- sprintf(name, "%s Playback Switch", pfx);
- err = add_control(spec, ALC_CTL_WIDGET_MUTE, name,
- HDA_COMPOSE_AMP_VAL(alc880_idx_to_mixer(i),
- 3, 0, HDA_INPUT));
+ err = alc662_add_sw_ctl(spec, pfx, mix, 3);
if (err < 0)
return err;
}
@@ -17217,86 +17252,73 @@ static int alc662_auto_create_multi_out_ctls(struct alc_spec *spec,
}
/* add playback controls for speaker and HP outputs */
-static int alc662_auto_create_extra_out(struct alc_spec *spec, hda_nid_t pin,
+/* return DAC nid if any new DAC is assigned */
+static int alc662_auto_create_extra_out(struct hda_codec *codec, hda_nid_t pin,
const char *pfx)
{
- hda_nid_t nid;
+ struct alc_spec *spec = codec->spec;
+ hda_nid_t nid, mix;
int err;
- char name[32];
if (!pin)
return 0;
-
- if (pin == 0x17) {
- /* ALC663 has a mono output pin on 0x17 */
- sprintf(name, "%s Playback Switch", pfx);
- err = add_control(spec, ALC_CTL_WIDGET_MUTE, name,
- HDA_COMPOSE_AMP_VAL(pin, 2, 0, HDA_OUTPUT));
- return err;
+ nid = alc662_look_for_dac(codec, pin);
+ if (!nid) {
+ /* the corresponding DAC is already occupied */
+ if (!(get_wcaps(codec, pin) & AC_WCAP_OUT_AMP))
+ return 0; /* no way */
+ /* create a switch only */
+ return add_pb_sw_ctrl(spec, ALC_CTL_WIDGET_MUTE, pfx,
+ HDA_COMPOSE_AMP_VAL(pin, 3, 0, HDA_OUTPUT));
}
- if (alc880_is_fixed_pin(pin)) {
- nid = alc880_idx_to_dac(alc880_fixed_pin_idx(pin));
- /* printk(KERN_DEBUG "DAC nid=%x\n",nid); */
- /* specify the DAC as the extra output */
- if (!spec->multiout.hp_nid)
- spec->multiout.hp_nid = nid;
- else
- spec->multiout.extra_out_nid[0] = nid;
- /* control HP volume/switch on the output mixer amp */
- nid = alc880_idx_to_dac(alc880_fixed_pin_idx(pin));
- sprintf(name, "%s Playback Volume", pfx);
- err = add_control(spec, ALC_CTL_WIDGET_VOL, name,
- HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT));
- if (err < 0)
- return err;
- sprintf(name, "%s Playback Switch", pfx);
- err = add_control(spec, ALC_CTL_BIND_MUTE, name,
- HDA_COMPOSE_AMP_VAL(nid, 3, 2, HDA_INPUT));
- if (err < 0)
- return err;
- } else if (alc880_is_multi_pin(pin)) {
- /* set manual connection */
- /* we have only a switch on HP-out PIN */
- sprintf(name, "%s Playback Switch", pfx);
- err = add_control(spec, ALC_CTL_WIDGET_MUTE, name,
- HDA_COMPOSE_AMP_VAL(pin, 3, 0, HDA_OUTPUT));
- if (err < 0)
- return err;
- }
- return 0;
+ mix = alc662_dac_to_mix(codec, pin, nid);
+ if (!mix)
+ return 0;
+ err = alc662_add_vol_ctl(spec, pfx, nid, 3);
+ if (err < 0)
+ return err;
+ err = alc662_add_sw_ctl(spec, pfx, mix, 3);
+ if (err < 0)
+ return err;
+ return nid;
}
/* create playback/capture controls for input pins */
#define alc662_auto_create_input_ctls \
- alc880_auto_create_input_ctls
+ alc882_auto_create_input_ctls
static void alc662_auto_set_output_and_unmute(struct hda_codec *codec,
hda_nid_t nid, int pin_type,
- int dac_idx)
+ hda_nid_t dac)
{
+ int i, num;
+ hda_nid_t srcs[4];
+
alc_set_pin_output(codec, nid, pin_type);
/* need the manual connection? */
- if (alc880_is_multi_pin(nid)) {
- struct alc_spec *spec = codec->spec;
- int idx = alc880_multi_pin_idx(nid);
- snd_hda_codec_write(codec, alc880_idx_to_selector(idx), 0,
- AC_VERB_SET_CONNECT_SEL,
- alc880_dac_to_idx(spec->multiout.dac_nids[dac_idx]));
+ num = snd_hda_get_connections(codec, nid, srcs, ARRAY_SIZE(srcs));
+ if (num <= 1)
+ return;
+ for (i = 0; i < num; i++) {
+ if (alc662_mix_to_dac(srcs[i]) != dac)
+ continue;
+ snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CONNECT_SEL, i);
+ return;
}
}
static void alc662_auto_init_multi_out(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
+ int pin_type = get_pin_type(spec->autocfg.line_out_type);
int i;
for (i = 0; i <= HDA_SIDE; i++) {
hda_nid_t nid = spec->autocfg.line_out_pins[i];
- int pin_type = get_pin_type(spec->autocfg.line_out_type);
if (nid)
alc662_auto_set_output_and_unmute(codec, nid, pin_type,
- i);
+ spec->multiout.dac_nids[i]);
}
}
@@ -17306,12 +17328,13 @@ static void alc662_auto_init_hp_out(struct hda_codec *codec)
hda_nid_t pin;
pin = spec->autocfg.hp_pins[0];
- if (pin) /* connect to front */
- /* use dac 0 */
- alc662_auto_set_output_and_unmute(codec, pin, PIN_HP, 0);
+ if (pin)
+ alc662_auto_set_output_and_unmute(codec, pin, PIN_HP,
+ spec->multiout.hp_nid);
pin = spec->autocfg.speaker_pins[0];
if (pin)
- alc662_auto_set_output_and_unmute(codec, pin, PIN_OUT, 0);
+ alc662_auto_set_output_and_unmute(codec, pin, PIN_OUT,
+ spec->multiout.extra_out_nid[0]);
}
#define ALC662_PIN_CD_NID ALC880_PIN_CD_NID
@@ -17349,21 +17372,25 @@ static int alc662_parse_auto_config(struct hda_codec *codec)
if (!spec->autocfg.line_outs)
return 0; /* can't find valid BIOS pin config */
- err = alc880_auto_fill_dac_nids(spec, &spec->autocfg);
+ err = alc662_auto_fill_dac_nids(codec, &spec->autocfg);
if (err < 0)
return err;
- err = alc662_auto_create_multi_out_ctls(spec, &spec->autocfg);
+ err = alc662_auto_create_multi_out_ctls(codec, &spec->autocfg);
if (err < 0)
return err;
- err = alc662_auto_create_extra_out(spec,
+ err = alc662_auto_create_extra_out(codec,
spec->autocfg.speaker_pins[0],
"Speaker");
if (err < 0)
return err;
- err = alc662_auto_create_extra_out(spec, spec->autocfg.hp_pins[0],
+ if (err)
+ spec->multiout.extra_out_nid[0] = err;
+ err = alc662_auto_create_extra_out(codec, spec->autocfg.hp_pins[0],
"Headphone");
if (err < 0)
return err;
+ if (err)
+ spec->multiout.hp_nid = err;
err = alc662_auto_create_input_ctls(codec, &spec->autocfg);
if (err < 0)
return err;
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 826137ec300..6b0bc040c3b 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -28,6 +28,7 @@
#include <linux/delay.h>
#include <linux/slab.h>
#include <linux/pci.h>
+#include <linux/dmi.h>
#include <sound/core.h>
#include <sound/asoundef.h>
#include <sound/jack.h>
@@ -92,6 +93,7 @@ enum {
STAC_92HD83XXX_REF,
STAC_92HD83XXX_PWR_REF,
STAC_DELL_S14,
+ STAC_92HD83XXX_HP,
STAC_92HD83XXX_MODELS
};
@@ -158,6 +160,7 @@ enum {
STAC_D965_5ST_NO_FP,
STAC_DELL_3ST,
STAC_DELL_BIOS,
+ STAC_927X_VOLKNOB,
STAC_927X_MODELS
};
@@ -182,8 +185,8 @@ struct sigmatel_jack {
struct sigmatel_mic_route {
hda_nid_t pin;
- unsigned char mux_idx;
- unsigned char dmux_idx;
+ signed char mux_idx;
+ signed char dmux_idx;
};
struct sigmatel_spec {
@@ -907,6 +910,16 @@ static struct hda_verb d965_core_init[] = {
{}
};
+static struct hda_verb dell_3st_core_init[] = {
+ /* don't set delta bit */
+ {0x24, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0x7f},
+ /* unmute node 0x1b */
+ {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
+ /* select node 0x03 as DAC */
+ {0x0b, AC_VERB_SET_CONNECT_SEL, 0x01},
+ {}
+};
+
static struct hda_verb stac927x_core_init[] = {
/* set master volume and direct control */
{ 0x24, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff},
@@ -915,6 +928,14 @@ static struct hda_verb stac927x_core_init[] = {
{}
};
+static struct hda_verb stac927x_volknob_core_init[] = {
+ /* don't set delta bit */
+ {0x24, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0x7f},
+ /* enable analog pc beep path */
+ {0x01, AC_VERB_SET_DIGI_CONVERT_2, 1 << 5},
+ {}
+};
+
static struct hda_verb stac9205_core_init[] = {
/* set master volume and direct control */
{ 0x24, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff},
@@ -1065,7 +1086,7 @@ static int stac92xx_build_controls(struct hda_codec *codec)
if (!spec->auto_mic && spec->num_dmuxes > 0 &&
snd_hda_get_bool_hint(codec, "separate_dmux") == 1) {
stac_dmux_mixer.count = spec->num_dmuxes;
- err = snd_hda_ctl_add(codec,
+ err = snd_hda_ctl_add(codec, 0,
snd_ctl_new1(&stac_dmux_mixer, codec));
if (err < 0)
return err;
@@ -1081,7 +1102,7 @@ static int stac92xx_build_controls(struct hda_codec *codec)
spec->spdif_mute = 1;
}
stac_smux_mixer.count = spec->num_smuxes;
- err = snd_hda_ctl_add(codec,
+ err = snd_hda_ctl_add(codec, 0,
snd_ctl_new1(&stac_smux_mixer, codec));
if (err < 0)
return err;
@@ -1570,6 +1591,8 @@ static struct snd_pci_quirk stac92hd73xx_cfg_tbl[] = {
"Dell Studio 17", STAC_DELL_M6_DMIC),
SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x02be,
"Dell Studio 1555", STAC_DELL_M6_DMIC),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x02bd,
+ "Dell Studio 1557", STAC_DELL_M6_DMIC),
{} /* terminator */
};
@@ -1602,6 +1625,7 @@ static const char *stac92hd83xxx_models[STAC_92HD83XXX_MODELS] = {
[STAC_92HD83XXX_REF] = "ref",
[STAC_92HD83XXX_PWR_REF] = "mic-ref",
[STAC_DELL_S14] = "dell-s14",
+ [STAC_92HD83XXX_HP] = "hp",
};
static struct snd_pci_quirk stac92hd83xxx_cfg_tbl[] = {
@@ -1612,6 +1636,8 @@ static struct snd_pci_quirk stac92hd83xxx_cfg_tbl[] = {
"DFI LanParty", STAC_92HD83XXX_REF),
SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x02ba,
"unknown Dell", STAC_DELL_S14),
+ SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_HP, 0xff00, 0x3600,
+ "HP", STAC_92HD83XXX_HP),
{} /* terminator */
};
@@ -1674,6 +1700,8 @@ static struct snd_pci_quirk stac92hd71bxx_cfg_tbl[] = {
"DFI LanParty", STAC_92HD71BXX_REF),
SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x30fb,
"HP dv4-1222nr", STAC_HP_DV4_1222NR),
+ SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_HP, 0xfff0, 0x1720,
+ "HP", STAC_HP_DV5),
SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_HP, 0xfff0, 0x3080,
"HP", STAC_HP_DV5),
SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_HP, 0xfff0, 0x30f0,
@@ -1999,6 +2027,7 @@ static unsigned int *stac927x_brd_tbl[STAC_927X_MODELS] = {
[STAC_D965_5ST_NO_FP] = d965_5st_no_fp_pin_configs,
[STAC_DELL_3ST] = dell_3st_pin_configs,
[STAC_DELL_BIOS] = NULL,
+ [STAC_927X_VOLKNOB] = NULL,
};
static const char *stac927x_models[STAC_927X_MODELS] = {
@@ -2010,6 +2039,7 @@ static const char *stac927x_models[STAC_927X_MODELS] = {
[STAC_D965_5ST_NO_FP] = "5stack-no-fp",
[STAC_DELL_3ST] = "dell-3stack",
[STAC_DELL_BIOS] = "dell-bios",
+ [STAC_927X_VOLKNOB] = "volknob",
};
static struct snd_pci_quirk stac927x_cfg_tbl[] = {
@@ -2045,6 +2075,8 @@ static struct snd_pci_quirk stac927x_cfg_tbl[] = {
"Intel D965", STAC_D965_5ST),
SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_INTEL, 0xff00, 0x2500,
"Intel D965", STAC_D965_5ST),
+ /* volume-knob fixes */
+ SND_PCI_QUIRK_VENDOR(0x10cf, "FSC", STAC_927X_VOLKNOB),
{} /* terminator */
};
@@ -2620,6 +2652,7 @@ static int stac92xx_clfe_switch_put(struct snd_kcontrol *kcontrol,
enum {
STAC_CTL_WIDGET_VOL,
STAC_CTL_WIDGET_MUTE,
+ STAC_CTL_WIDGET_MUTE_BEEP,
STAC_CTL_WIDGET_MONO_MUX,
STAC_CTL_WIDGET_HP_SWITCH,
STAC_CTL_WIDGET_IO_SWITCH,
@@ -2630,6 +2663,7 @@ enum {
static struct snd_kcontrol_new stac92xx_control_templates[] = {
HDA_CODEC_VOLUME(NULL, 0, 0, 0),
HDA_CODEC_MUTE(NULL, 0, 0, 0),
+ HDA_CODEC_MUTE_BEEP(NULL, 0, 0, 0),
STAC_MONO_MUX,
STAC_CODEC_HP_SWITCH(NULL),
STAC_CODEC_IO_SWITCH(NULL, 0),
@@ -2641,7 +2675,8 @@ static struct snd_kcontrol_new stac92xx_control_templates[] = {
static struct snd_kcontrol_new *
stac_control_new(struct sigmatel_spec *spec,
struct snd_kcontrol_new *ktemp,
- const char *name)
+ const char *name,
+ hda_nid_t nid)
{
struct snd_kcontrol_new *knew;
@@ -2657,6 +2692,8 @@ stac_control_new(struct sigmatel_spec *spec,
spec->kctls.alloced--;
return NULL;
}
+ if (nid)
+ knew->subdevice = HDA_SUBDEV_NID_FLAG | nid;
return knew;
}
@@ -2665,7 +2702,8 @@ static int stac92xx_add_control_temp(struct sigmatel_spec *spec,
int idx, const char *name,
unsigned long val)
{
- struct snd_kcontrol_new *knew = stac_control_new(spec, ktemp, name);
+ struct snd_kcontrol_new *knew = stac_control_new(spec, ktemp, name,
+ get_amp_nid_(val));
if (!knew)
return -ENOMEM;
knew->index = idx;
@@ -2736,7 +2774,7 @@ static int stac92xx_add_input_source(struct sigmatel_spec *spec)
if (!spec->num_adcs || imux->num_items <= 1)
return 0; /* no need for input source control */
knew = stac_control_new(spec, &stac_input_src_temp,
- stac_input_src_temp.name);
+ stac_input_src_temp.name, 0);
if (!knew)
return -ENOMEM;
knew->count = spec->num_adcs;
@@ -3193,12 +3231,15 @@ static int stac92xx_auto_create_beep_ctls(struct hda_codec *codec,
{
struct sigmatel_spec *spec = codec->spec;
u32 caps = query_amp_caps(codec, nid, HDA_OUTPUT);
- int err;
+ int err, type = STAC_CTL_WIDGET_MUTE_BEEP;
+
+ if (spec->anabeep_nid == nid)
+ type = STAC_CTL_WIDGET_MUTE;
/* check for mute support for the the amp */
if ((caps & AC_AMPCAP_MUTE) >> AC_AMPCAP_MUTE_SHIFT) {
- err = stac92xx_add_control(spec, STAC_CTL_WIDGET_MUTE,
- "PC Beep Playback Switch",
+ err = stac92xx_add_control(spec, type,
+ "Beep Playback Switch",
HDA_COMPOSE_AMP_VAL(nid, 1, 0, HDA_OUTPUT));
if (err < 0)
return err;
@@ -3207,7 +3248,7 @@ static int stac92xx_auto_create_beep_ctls(struct hda_codec *codec,
/* check to see if there is volume support for the amp */
if ((caps & AC_AMPCAP_NUM_STEPS) >> AC_AMPCAP_NUM_STEPS_SHIFT) {
err = stac92xx_add_control(spec, STAC_CTL_WIDGET_VOL,
- "PC Beep Playback Volume",
+ "Beep Playback Volume",
HDA_COMPOSE_AMP_VAL(nid, 1, 0, HDA_OUTPUT));
if (err < 0)
return err;
@@ -3230,12 +3271,7 @@ static int stac92xx_dig_beep_switch_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
- int enabled = !!ucontrol->value.integer.value[0];
- if (codec->beep->enabled != enabled) {
- codec->beep->enabled = enabled;
- return 1;
- }
- return 0;
+ return snd_hda_enable_beep_device(codec, ucontrol->value.integer.value[0]);
}
static struct snd_kcontrol_new stac92xx_dig_beep_ctrl = {
@@ -3248,7 +3284,7 @@ static struct snd_kcontrol_new stac92xx_dig_beep_ctrl = {
static int stac92xx_beep_switch_ctl(struct hda_codec *codec)
{
return stac92xx_add_control_temp(codec->spec, &stac92xx_dig_beep_ctrl,
- 0, "PC Beep Playback Switch", 0);
+ 0, "Beep Playback Switch", 0);
}
#endif
@@ -3469,18 +3505,26 @@ static int set_mic_route(struct hda_codec *codec,
break;
if (i <= AUTO_PIN_FRONT_MIC) {
/* analog pin */
- mic->dmux_idx = 0;
i = get_connection_index(codec, spec->mux_nids[0], pin);
if (i < 0)
return -1;
mic->mux_idx = i;
+ mic->dmux_idx = -1;
+ if (spec->dmux_nids)
+ mic->dmux_idx = get_connection_index(codec,
+ spec->dmux_nids[0],
+ spec->mux_nids[0]);
} else if (spec->dmux_nids) {
/* digital pin */
- mic->mux_idx = 0;
i = get_connection_index(codec, spec->dmux_nids[0], pin);
if (i < 0)
return -1;
mic->dmux_idx = i;
+ mic->mux_idx = -1;
+ if (spec->mux_nids)
+ mic->mux_idx = get_connection_index(codec,
+ spec->mux_nids[0],
+ spec->dmux_nids[0]);
}
return 0;
}
@@ -3595,6 +3639,26 @@ static void stac92xx_auto_init_hp_out(struct hda_codec *codec)
}
}
+static int is_dual_headphones(struct hda_codec *codec)
+{
+ struct sigmatel_spec *spec = codec->spec;
+ int i, valid_hps;
+
+ if (spec->autocfg.line_out_type != AUTO_PIN_SPEAKER_OUT ||
+ spec->autocfg.hp_outs <= 1)
+ return 0;
+ valid_hps = 0;
+ for (i = 0; i < spec->autocfg.hp_outs; i++) {
+ hda_nid_t nid = spec->autocfg.hp_pins[i];
+ unsigned int cfg = snd_hda_codec_get_pincfg(codec, nid);
+ if (get_defcfg_location(cfg) & AC_JACK_LOC_SEPARATE)
+ continue;
+ valid_hps++;
+ }
+ return (valid_hps > 1);
+}
+
+
static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out, hda_nid_t dig_in)
{
struct sigmatel_spec *spec = codec->spec;
@@ -3611,8 +3675,7 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out
/* If we have no real line-out pin and multiple hp-outs, HPs should
* be set up as multi-channel outputs.
*/
- if (spec->autocfg.line_out_type == AUTO_PIN_SPEAKER_OUT &&
- spec->autocfg.hp_outs > 1) {
+ if (is_dual_headphones(codec)) {
/* Copy hp_outs to line_outs, backup line_outs in
* speaker_outs so that the following routines can handle
* HP pins as primary outputs.
@@ -4293,6 +4356,28 @@ static void stac92xx_free_kctls(struct hda_codec *codec)
snd_array_free(&spec->kctls);
}
+static void stac92xx_shutup(struct hda_codec *codec)
+{
+ struct sigmatel_spec *spec = codec->spec;
+ int i;
+ hda_nid_t nid;
+
+ /* reset each pin before powering down DAC/ADC to avoid click noise */
+ nid = codec->start_nid;
+ for (i = 0; i < codec->num_nodes; i++, nid++) {
+ unsigned int wcaps = get_wcaps(codec, nid);
+ unsigned int wid_type = get_wcaps_type(wcaps);
+ if (wid_type == AC_WID_PIN)
+ snd_hda_codec_read(codec, nid, 0,
+ AC_VERB_SET_PIN_WIDGET_CONTROL, 0);
+ }
+
+ if (spec->eapd_mask)
+ stac_gpio_set(codec, spec->gpio_mask,
+ spec->gpio_dir, spec->gpio_data &
+ ~spec->eapd_mask);
+}
+
static void stac92xx_free(struct hda_codec *codec)
{
struct sigmatel_spec *spec = codec->spec;
@@ -4300,6 +4385,7 @@ static void stac92xx_free(struct hda_codec *codec)
if (! spec)
return;
+ stac92xx_shutup(codec);
stac92xx_free_jacks(codec);
snd_array_free(&spec->events);
@@ -4350,12 +4436,16 @@ static void stac92xx_reset_pinctl(struct hda_codec *codec, hda_nid_t nid,
pin_ctl & ~flag);
}
-static int get_pin_presence(struct hda_codec *codec, hda_nid_t nid)
+static inline int get_pin_presence(struct hda_codec *codec, hda_nid_t nid)
{
if (!nid)
return 0;
- if (snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_PIN_SENSE, 0x00)
- & (1 << 31))
+ /* NOTE: we can't use snd_hda_jack_detect() here because STAC/IDT
+ * codecs behave wrongly when SET_PIN_SENSE is triggered, although
+ * the pincap gives TRIG_REQ bit.
+ */
+ if (snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_PIN_SENSE, 0) &
+ AC_PINSENSE_PRESENCE)
return 1;
return 0;
}
@@ -4557,11 +4647,11 @@ static void stac92xx_mic_detect(struct hda_codec *codec)
mic = &spec->ext_mic;
else
mic = &spec->int_mic;
- if (mic->dmux_idx)
+ if (mic->dmux_idx >= 0)
snd_hda_codec_write_cache(codec, spec->dmux_nids[0], 0,
AC_VERB_SET_CONNECT_SEL,
mic->dmux_idx);
- else
+ if (mic->mux_idx >= 0)
snd_hda_codec_write_cache(codec, spec->mux_nids[0], 0,
AC_VERB_SET_CONNECT_SEL,
mic->mux_idx);
@@ -4634,6 +4724,26 @@ static void stac92xx_unsol_event(struct hda_codec *codec, unsigned int res)
}
}
+static int hp_bseries_system(u32 subsystem_id)
+{
+ switch (subsystem_id) {
+ case 0x103c307e:
+ case 0x103c307f:
+ case 0x103c3080:
+ case 0x103c3081:
+ case 0x103c1722:
+ case 0x103c1723:
+ case 0x103c1724:
+ case 0x103c1725:
+ case 0x103c1726:
+ case 0x103c1727:
+ case 0x103c1728:
+ case 0x103c1729:
+ return 1;
+ }
+ return 0;
+}
+
#ifdef CONFIG_PROC_FS
static void stac92hd_proc_hook(struct snd_info_buffer *buffer,
struct hda_codec *codec, hda_nid_t nid)
@@ -4723,6 +4833,11 @@ static int stac92xx_hp_check_power_status(struct hda_codec *codec,
else
spec->gpio_data |= spec->gpio_led; /* white */
+ if (hp_bseries_system(codec->subsystem_id)) {
+ /* LED state is inverted on these systems */
+ spec->gpio_data ^= spec->gpio_led;
+ }
+
stac_gpio_set(codec, spec->gpio_mask,
spec->gpio_dir,
spec->gpio_data);
@@ -4730,28 +4845,28 @@ static int stac92xx_hp_check_power_status(struct hda_codec *codec,
return 0;
}
-#endif
-static int stac92xx_suspend(struct hda_codec *codec, pm_message_t state)
+static int idt92hd83xxx_hp_check_power_status(struct hda_codec *codec,
+ hda_nid_t nid)
{
struct sigmatel_spec *spec = codec->spec;
- int i;
- hda_nid_t nid;
- /* reset each pin before powering down DAC/ADC to avoid click noise */
- nid = codec->start_nid;
- for (i = 0; i < codec->num_nodes; i++, nid++) {
- unsigned int wcaps = get_wcaps(codec, nid);
- unsigned int wid_type = get_wcaps_type(wcaps);
- if (wid_type == AC_WID_PIN)
- snd_hda_codec_read(codec, nid, 0,
- AC_VERB_SET_PIN_WIDGET_CONTROL, 0);
- }
+ if (nid != 0x13)
+ return 0;
+ if (snd_hda_codec_amp_read(codec, nid, 0, HDA_OUTPUT, 0) & HDA_AMP_MUTE)
+ spec->gpio_data |= spec->gpio_led; /* mute LED on */
+ else
+ spec->gpio_data &= ~spec->gpio_led; /* mute LED off */
+ stac_gpio_set(codec, spec->gpio_mask, spec->gpio_dir, spec->gpio_data);
- if (spec->eapd_mask)
- stac_gpio_set(codec, spec->gpio_mask,
- spec->gpio_dir, spec->gpio_data &
- ~spec->eapd_mask);
+ return 0;
+}
+
+#endif
+
+static int stac92xx_suspend(struct hda_codec *codec, pm_message_t state)
+{
+ stac92xx_shutup(codec);
return 0;
}
#endif
@@ -4766,6 +4881,7 @@ static struct hda_codec_ops stac92xx_patch_ops = {
.suspend = stac92xx_suspend,
.resume = stac92xx_resume,
#endif
+ .reboot_notify = stac92xx_shutup,
};
static int patch_stac9200(struct hda_codec *codec)
@@ -5111,6 +5227,22 @@ again:
break;
}
+ codec->patch_ops = stac92xx_patch_ops;
+
+ if (spec->board_config == STAC_92HD83XXX_HP)
+ spec->gpio_led = 0x01;
+
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ if (spec->gpio_led) {
+ spec->gpio_mask |= spec->gpio_led;
+ spec->gpio_dir |= spec->gpio_led;
+ spec->gpio_data |= spec->gpio_led;
+ /* register check_power_status callback. */
+ codec->patch_ops.check_power_status =
+ idt92hd83xxx_hp_check_power_status;
+ }
+#endif
+
err = stac92xx_parse_auto_config(codec, 0x1d, 0);
if (!err) {
if (spec->board_config < 0) {
@@ -5146,8 +5278,6 @@ again:
snd_hda_codec_write_cache(codec, nid, 0,
AC_VERB_SET_CONNECT_SEL, num_dacs);
- codec->patch_ops = stac92xx_patch_ops;
-
codec->proc_widget_hook = stac92hd_proc_hook;
return 0;
@@ -5212,6 +5342,7 @@ static int patch_stac92hd71bxx(struct hda_codec *codec)
{
struct sigmatel_spec *spec;
struct hda_verb *unmute_init = stac92hd71bxx_unmute_core_init;
+ unsigned int pin_cfg;
int err = 0;
spec = kzalloc(sizeof(*spec), GFP_KERNEL);
@@ -5395,6 +5526,45 @@ again:
break;
}
+ if (hp_bseries_system(codec->subsystem_id)) {
+ pin_cfg = snd_hda_codec_get_pincfg(codec, 0x0f);
+ if (get_defcfg_device(pin_cfg) == AC_JACK_LINE_OUT ||
+ get_defcfg_device(pin_cfg) == AC_JACK_SPEAKER ||
+ get_defcfg_device(pin_cfg) == AC_JACK_HP_OUT) {
+ /* It was changed in the BIOS to just satisfy MS DTM.
+ * Lets turn it back into slaved HP
+ */
+ pin_cfg = (pin_cfg & (~AC_DEFCFG_DEVICE))
+ | (AC_JACK_HP_OUT <<
+ AC_DEFCFG_DEVICE_SHIFT);
+ pin_cfg = (pin_cfg & (~(AC_DEFCFG_DEF_ASSOC
+ | AC_DEFCFG_SEQUENCE)))
+ | 0x1f;
+ snd_hda_codec_set_pincfg(codec, 0x0f, pin_cfg);
+ }
+ }
+
+ if ((codec->subsystem_id >> 16) == PCI_VENDOR_ID_HP) {
+ const struct dmi_device *dev = NULL;
+ while ((dev = dmi_find_device(DMI_DEV_TYPE_OEM_STRING,
+ NULL, dev))) {
+ if (strcmp(dev->name, "HP_Mute_LED_1")) {
+ switch (codec->vendor_id) {
+ case 0x111d7608:
+ spec->gpio_led = 0x01;
+ break;
+ case 0x111d7600:
+ case 0x111d7601:
+ case 0x111d7602:
+ case 0x111d7603:
+ spec->gpio_led = 0x08;
+ break;
+ }
+ break;
+ }
+ }
+ }
+
#ifdef CONFIG_SND_HDA_POWER_SAVE
if (spec->gpio_led) {
spec->gpio_mask |= spec->gpio_led;
@@ -5604,10 +5774,14 @@ static int patch_stac927x(struct hda_codec *codec)
spec->dmic_nids = stac927x_dmic_nids;
spec->num_dmics = STAC927X_NUM_DMICS;
- spec->init = d965_core_init;
+ spec->init = dell_3st_core_init;
spec->dmux_nids = stac927x_dmux_nids;
spec->num_dmuxes = ARRAY_SIZE(stac927x_dmux_nids);
break;
+ case STAC_927X_VOLKNOB:
+ spec->num_dmics = 0;
+ spec->init = stac927x_volknob_core_init;
+ break;
default:
spec->num_dmics = 0;
spec->init = stac927x_core_init;
diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c
index ee89db90c9b..b70e26ad263 100644
--- a/sound/pci/hda/patch_via.c
+++ b/sound/pci/hda/patch_via.c
@@ -1,10 +1,10 @@
/*
* Universal Interface for Intel High Definition Audio Codec
*
- * HD audio interface patch for VIA VT1702/VT1708/VT1709 codec
+ * HD audio interface patch for VIA VT17xx/VT18xx/VT20xx codec
*
- * Copyright (c) 2006-2008 Lydia Wang <lydiawang@viatech.com>
- * Takashi Iwai <tiwai@suse.de>
+ * (C) 2006-2009 VIA Technology, Inc.
+ * (C) 2006-2008 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
@@ -22,21 +22,27 @@
*/
/* * * * * * * * * * * * * * Release History * * * * * * * * * * * * * * * * */
-/* */
+/* */
/* 2006-03-03 Lydia Wang Create the basic patch to support VT1708 codec */
-/* 2006-03-14 Lydia Wang Modify hard code for some pin widget nid */
-/* 2006-08-02 Lydia Wang Add support to VT1709 codec */
+/* 2006-03-14 Lydia Wang Modify hard code for some pin widget nid */
+/* 2006-08-02 Lydia Wang Add support to VT1709 codec */
/* 2006-09-08 Lydia Wang Fix internal loopback recording source select bug */
-/* 2007-09-12 Lydia Wang Add EAPD enable during driver initialization */
-/* 2007-09-17 Lydia Wang Add VT1708B codec support */
+/* 2007-09-12 Lydia Wang Add EAPD enable during driver initialization */
+/* 2007-09-17 Lydia Wang Add VT1708B codec support */
/* 2007-11-14 Lydia Wang Add VT1708A codec HP and CD pin connect config */
/* 2008-02-03 Lydia Wang Fix Rear channels and Back channels inverse issue */
-/* 2008-03-06 Lydia Wang Add VT1702 codec and VT1708S codec support */
-/* 2008-04-09 Lydia Wang Add mute front speaker when HP plugin */
-/* 2008-04-09 Lydia Wang Add Independent HP feature */
+/* 2008-03-06 Lydia Wang Add VT1702 codec and VT1708S codec support */
+/* 2008-04-09 Lydia Wang Add mute front speaker when HP plugin */
+/* 2008-04-09 Lydia Wang Add Independent HP feature */
/* 2008-05-28 Lydia Wang Add second S/PDIF Out support for VT1702 */
-/* 2008-09-15 Logan Li Add VT1708S Mic Boost workaround/backdoor */
-/* */
+/* 2008-09-15 Logan Li Add VT1708S Mic Boost workaround/backdoor */
+/* 2009-02-16 Logan Li Add support for VT1718S */
+/* 2009-03-13 Logan Li Add support for VT1716S */
+/* 2009-04-14 Lydai Wang Add support for VT1828S and VT2020 */
+/* 2009-07-08 Lydia Wang Add support for VT2002P */
+/* 2009-07-21 Lydia Wang Add support for VT1812 */
+/* 2009-09-19 Lydia Wang Add support for VT1818S */
+/* */
/* * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * */
@@ -76,14 +82,6 @@
#define VT1702_HP_NID 0x17
#define VT1702_DIGOUT_NID 0x11
-#define IS_VT1708_VENDORID(x) ((x) >= 0x11061708 && (x) <= 0x1106170b)
-#define IS_VT1709_10CH_VENDORID(x) ((x) >= 0x1106e710 && (x) <= 0x1106e713)
-#define IS_VT1709_6CH_VENDORID(x) ((x) >= 0x1106e714 && (x) <= 0x1106e717)
-#define IS_VT1708B_8CH_VENDORID(x) ((x) >= 0x1106e720 && (x) <= 0x1106e723)
-#define IS_VT1708B_4CH_VENDORID(x) ((x) >= 0x1106e724 && (x) <= 0x1106e727)
-#define IS_VT1708S_VENDORID(x) ((x) >= 0x11060397 && (x) <= 0x11067397)
-#define IS_VT1702_VENDORID(x) ((x) >= 0x11060398 && (x) <= 0x11067398)
-
enum VIA_HDA_CODEC {
UNKNOWN = -1,
VT1708,
@@ -92,12 +90,76 @@ enum VIA_HDA_CODEC {
VT1708B_8CH,
VT1708B_4CH,
VT1708S,
+ VT1708BCE,
VT1702,
+ VT1718S,
+ VT1716S,
+ VT2002P,
+ VT1812,
CODEC_TYPES,
};
-static enum VIA_HDA_CODEC get_codec_type(u32 vendor_id)
+struct via_spec {
+ /* codec parameterization */
+ struct snd_kcontrol_new *mixers[6];
+ unsigned int num_mixers;
+
+ struct hda_verb *init_verbs[5];
+ unsigned int num_iverbs;
+
+ char *stream_name_analog;
+ struct hda_pcm_stream *stream_analog_playback;
+ struct hda_pcm_stream *stream_analog_capture;
+
+ char *stream_name_digital;
+ struct hda_pcm_stream *stream_digital_playback;
+ struct hda_pcm_stream *stream_digital_capture;
+
+ /* playback */
+ struct hda_multi_out multiout;
+ hda_nid_t slave_dig_outs[2];
+
+ /* capture */
+ unsigned int num_adc_nids;
+ hda_nid_t *adc_nids;
+ hda_nid_t mux_nids[3];
+ hda_nid_t dig_in_nid;
+ hda_nid_t dig_in_pin;
+
+ /* capture source */
+ const struct hda_input_mux *input_mux;
+ unsigned int cur_mux[3];
+
+ /* PCM information */
+ struct hda_pcm pcm_rec[3];
+
+ /* dynamic controls, init_verbs and input_mux */
+ struct auto_pin_cfg autocfg;
+ struct snd_array kctls;
+ struct hda_input_mux private_imux[2];
+ hda_nid_t private_dac_nids[AUTO_CFG_MAX_OUTS];
+
+ /* HP mode source */
+ const struct hda_input_mux *hp_mux;
+ unsigned int hp_independent_mode;
+ unsigned int hp_independent_mode_index;
+ unsigned int smart51_enabled;
+ unsigned int dmic_enabled;
+ enum VIA_HDA_CODEC codec_type;
+
+ /* work to check hp jack state */
+ struct hda_codec *codec;
+ struct delayed_work vt1708_hp_work;
+ int vt1708_jack_detectect;
+ int vt1708_hp_present;
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ struct hda_loopback_check loopback;
+#endif
+};
+
+static enum VIA_HDA_CODEC get_codec_type(struct hda_codec *codec)
{
+ u32 vendor_id = codec->vendor_id;
u16 ven_id = vendor_id >> 16;
u16 dev_id = vendor_id & 0xffff;
enum VIA_HDA_CODEC codec_type;
@@ -111,9 +173,11 @@ static enum VIA_HDA_CODEC get_codec_type(u32 vendor_id)
codec_type = VT1709_10CH;
else if (dev_id >= 0xe714 && dev_id <= 0xe717)
codec_type = VT1709_6CH;
- else if (dev_id >= 0xe720 && dev_id <= 0xe723)
+ else if (dev_id >= 0xe720 && dev_id <= 0xe723) {
codec_type = VT1708B_8CH;
- else if (dev_id >= 0xe724 && dev_id <= 0xe727)
+ if (snd_hda_param_read(codec, 0x16, AC_PAR_CONNLIST_LEN) == 0x7)
+ codec_type = VT1708BCE;
+ } else if (dev_id >= 0xe724 && dev_id <= 0xe727)
codec_type = VT1708B_4CH;
else if ((dev_id & 0xfff) == 0x397
&& (dev_id >> 12) < 8)
@@ -121,6 +185,19 @@ static enum VIA_HDA_CODEC get_codec_type(u32 vendor_id)
else if ((dev_id & 0xfff) == 0x398
&& (dev_id >> 12) < 8)
codec_type = VT1702;
+ else if ((dev_id & 0xfff) == 0x428
+ && (dev_id >> 12) < 8)
+ codec_type = VT1718S;
+ else if (dev_id == 0x0433 || dev_id == 0xa721)
+ codec_type = VT1716S;
+ else if (dev_id == 0x0441 || dev_id == 0x4441)
+ codec_type = VT1718S;
+ else if (dev_id == 0x0438 || dev_id == 0x4438)
+ codec_type = VT2002P;
+ else if (dev_id == 0x0448)
+ codec_type = VT1812;
+ else if (dev_id == 0x0440)
+ codec_type = VT1708S;
else
codec_type = UNKNOWN;
return codec_type;
@@ -128,10 +205,16 @@ static enum VIA_HDA_CODEC get_codec_type(u32 vendor_id)
#define VIA_HP_EVENT 0x01
#define VIA_GPIO_EVENT 0x02
+#define VIA_JACK_EVENT 0x04
+#define VIA_MONO_EVENT 0x08
+#define VIA_SPEAKER_EVENT 0x10
+#define VIA_BIND_HP_EVENT 0x20
enum {
VIA_CTL_WIDGET_VOL,
VIA_CTL_WIDGET_MUTE,
+ VIA_CTL_WIDGET_ANALOG_MUTE,
+ VIA_CTL_WIDGET_BIND_PIN_MUTE,
};
enum {
@@ -141,99 +224,162 @@ enum {
AUTO_SEQ_SIDE
};
-/* Some VT1708S based boards gets the micboost setting wrong, so we have
- * to apply some brute-force and re-write the TLV's by software. */
-static int mic_boost_tlv(struct snd_kcontrol *kcontrol, int op_flag,
- unsigned int size, unsigned int __user *_tlv)
+static void analog_low_current_mode(struct hda_codec *codec, int stream_idle);
+static void set_jack_power_state(struct hda_codec *codec);
+static int is_aa_path_mute(struct hda_codec *codec);
+
+static void vt1708_start_hp_work(struct via_spec *spec)
{
- struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
- hda_nid_t nid = get_amp_nid(kcontrol);
+ if (spec->codec_type != VT1708 || spec->autocfg.hp_pins[0] == 0)
+ return;
+ snd_hda_codec_write(spec->codec, 0x1, 0, 0xf81,
+ !spec->vt1708_jack_detectect);
+ if (!delayed_work_pending(&spec->vt1708_hp_work))
+ schedule_delayed_work(&spec->vt1708_hp_work,
+ msecs_to_jiffies(100));
+}
- if (get_codec_type(codec->vendor_id) == VT1708S
- && (nid == 0x1a || nid == 0x1e)) {
- if (size < 4 * sizeof(unsigned int))
- return -ENOMEM;
- if (put_user(1, _tlv)) /* SNDRV_CTL_TLVT_DB_SCALE */
- return -EFAULT;
- if (put_user(2 * sizeof(unsigned int), _tlv + 1))
- return -EFAULT;
- if (put_user(0, _tlv + 2)) /* offset = 0 */
- return -EFAULT;
- if (put_user(1000, _tlv + 3)) /* step size = 10 dB */
- return -EFAULT;
- }
- return 0;
+static void vt1708_stop_hp_work(struct via_spec *spec)
+{
+ if (spec->codec_type != VT1708 || spec->autocfg.hp_pins[0] == 0)
+ return;
+ if (snd_hda_get_bool_hint(spec->codec, "analog_loopback_hp_detect") == 1
+ && !is_aa_path_mute(spec->codec))
+ return;
+ snd_hda_codec_write(spec->codec, 0x1, 0, 0xf81,
+ !spec->vt1708_jack_detectect);
+ cancel_delayed_work(&spec->vt1708_hp_work);
+ flush_scheduled_work();
}
-static int mic_boost_volume_info(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_info *uinfo)
+
+static int analog_input_switch_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
{
+ int change = snd_hda_mixer_amp_switch_put(kcontrol, ucontrol);
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
- hda_nid_t nid = get_amp_nid(kcontrol);
- if (get_codec_type(codec->vendor_id) == VT1708S
- && (nid == 0x1a || nid == 0x1e)) {
- uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
- uinfo->count = 2;
- uinfo->value.integer.min = 0;
- uinfo->value.integer.max = 3;
+ set_jack_power_state(codec);
+ analog_low_current_mode(snd_kcontrol_chip(kcontrol), -1);
+ if (snd_hda_get_bool_hint(codec, "analog_loopback_hp_detect") == 1) {
+ if (is_aa_path_mute(codec))
+ vt1708_start_hp_work(codec->spec);
+ else
+ vt1708_stop_hp_work(codec->spec);
}
- return 0;
+ return change;
}
-static struct snd_kcontrol_new vt1708_control_templates[] = {
- HDA_CODEC_VOLUME(NULL, 0, 0, 0),
- HDA_CODEC_MUTE(NULL, 0, 0, 0),
-};
-
-
-struct via_spec {
- /* codec parameterization */
- struct snd_kcontrol_new *mixers[3];
- unsigned int num_mixers;
+/* modify .put = snd_hda_mixer_amp_switch_put */
+#define ANALOG_INPUT_MUTE \
+ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
+ .name = NULL, \
+ .index = 0, \
+ .info = snd_hda_mixer_amp_switch_info, \
+ .get = snd_hda_mixer_amp_switch_get, \
+ .put = analog_input_switch_put, \
+ .private_value = HDA_COMPOSE_AMP_VAL(0, 3, 0, 0) }
- struct hda_verb *init_verbs[5];
- unsigned int num_iverbs;
+static void via_hp_bind_automute(struct hda_codec *codec);
- char *stream_name_analog;
- struct hda_pcm_stream *stream_analog_playback;
- struct hda_pcm_stream *stream_analog_capture;
-
- char *stream_name_digital;
- struct hda_pcm_stream *stream_digital_playback;
- struct hda_pcm_stream *stream_digital_capture;
-
- /* playback */
- struct hda_multi_out multiout;
- hda_nid_t slave_dig_outs[2];
-
- /* capture */
- unsigned int num_adc_nids;
- hda_nid_t *adc_nids;
- hda_nid_t mux_nids[3];
- hda_nid_t dig_in_nid;
- hda_nid_t dig_in_pin;
+static int bind_pin_switch_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct via_spec *spec = codec->spec;
+ int i;
+ int change = 0;
- /* capture source */
- const struct hda_input_mux *input_mux;
- unsigned int cur_mux[3];
+ long *valp = ucontrol->value.integer.value;
+ int lmute, rmute;
+ if (strstr(kcontrol->id.name, "Switch") == NULL) {
+ snd_printd("Invalid control!\n");
+ return change;
+ }
+ change = snd_hda_mixer_amp_switch_put(kcontrol,
+ ucontrol);
+ /* Get mute value */
+ lmute = *valp ? 0 : HDA_AMP_MUTE;
+ valp++;
+ rmute = *valp ? 0 : HDA_AMP_MUTE;
+
+ /* Set hp pins */
+ if (!spec->hp_independent_mode) {
+ for (i = 0; i < spec->autocfg.hp_outs; i++) {
+ snd_hda_codec_amp_update(
+ codec, spec->autocfg.hp_pins[i],
+ 0, HDA_OUTPUT, 0, HDA_AMP_MUTE,
+ lmute);
+ snd_hda_codec_amp_update(
+ codec, spec->autocfg.hp_pins[i],
+ 1, HDA_OUTPUT, 0, HDA_AMP_MUTE,
+ rmute);
+ }
+ }
- /* PCM information */
- struct hda_pcm pcm_rec[3];
+ if (!lmute && !rmute) {
+ /* Line Outs */
+ for (i = 0; i < spec->autocfg.line_outs; i++)
+ snd_hda_codec_amp_stereo(
+ codec, spec->autocfg.line_out_pins[i],
+ HDA_OUTPUT, 0, HDA_AMP_MUTE, 0);
+ /* Speakers */
+ for (i = 0; i < spec->autocfg.speaker_outs; i++)
+ snd_hda_codec_amp_stereo(
+ codec, spec->autocfg.speaker_pins[i],
+ HDA_OUTPUT, 0, HDA_AMP_MUTE, 0);
+ /* unmute */
+ via_hp_bind_automute(codec);
- /* dynamic controls, init_verbs and input_mux */
- struct auto_pin_cfg autocfg;
- struct snd_array kctls;
- struct hda_input_mux private_imux[2];
- hda_nid_t private_dac_nids[AUTO_CFG_MAX_OUTS];
+ } else {
+ if (lmute) {
+ /* Mute all left channels */
+ for (i = 1; i < spec->autocfg.line_outs; i++)
+ snd_hda_codec_amp_update(
+ codec,
+ spec->autocfg.line_out_pins[i],
+ 0, HDA_OUTPUT, 0, HDA_AMP_MUTE,
+ lmute);
+ for (i = 0; i < spec->autocfg.speaker_outs; i++)
+ snd_hda_codec_amp_update(
+ codec,
+ spec->autocfg.speaker_pins[i],
+ 0, HDA_OUTPUT, 0, HDA_AMP_MUTE,
+ lmute);
+ }
+ if (rmute) {
+ /* mute all right channels */
+ for (i = 1; i < spec->autocfg.line_outs; i++)
+ snd_hda_codec_amp_update(
+ codec,
+ spec->autocfg.line_out_pins[i],
+ 1, HDA_OUTPUT, 0, HDA_AMP_MUTE,
+ rmute);
+ for (i = 0; i < spec->autocfg.speaker_outs; i++)
+ snd_hda_codec_amp_update(
+ codec,
+ spec->autocfg.speaker_pins[i],
+ 1, HDA_OUTPUT, 0, HDA_AMP_MUTE,
+ rmute);
+ }
+ }
+ return change;
+}
- /* HP mode source */
- const struct hda_input_mux *hp_mux;
- unsigned int hp_independent_mode;
+#define BIND_PIN_MUTE \
+ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
+ .name = NULL, \
+ .index = 0, \
+ .info = snd_hda_mixer_amp_switch_info, \
+ .get = snd_hda_mixer_amp_switch_get, \
+ .put = bind_pin_switch_put, \
+ .private_value = HDA_COMPOSE_AMP_VAL(0, 3, 0, 0) }
-#ifdef CONFIG_SND_HDA_POWER_SAVE
- struct hda_loopback_check loopback;
-#endif
+static struct snd_kcontrol_new via_control_templates[] = {
+ HDA_CODEC_VOLUME(NULL, 0, 0, 0),
+ HDA_CODEC_MUTE(NULL, 0, 0, 0),
+ ANALOG_INPUT_MUTE,
+ BIND_PIN_MUTE,
};
static hda_nid_t vt1708_adc_nids[2] = {
@@ -261,6 +407,27 @@ static hda_nid_t vt1702_adc_nids[3] = {
0x12, 0x20, 0x1F
};
+static hda_nid_t vt1718S_adc_nids[2] = {
+ /* ADC1-2 */
+ 0x10, 0x11
+};
+
+static hda_nid_t vt1716S_adc_nids[2] = {
+ /* ADC1-2 */
+ 0x13, 0x14
+};
+
+static hda_nid_t vt2002P_adc_nids[2] = {
+ /* ADC1-2 */
+ 0x10, 0x11
+};
+
+static hda_nid_t vt1812_adc_nids[2] = {
+ /* ADC1-2 */
+ 0x10, 0x11
+};
+
+
/* add dynamic controls */
static int via_add_control(struct via_spec *spec, int type, const char *name,
unsigned long val)
@@ -271,10 +438,12 @@ static int via_add_control(struct via_spec *spec, int type, const char *name,
knew = snd_array_new(&spec->kctls);
if (!knew)
return -ENOMEM;
- *knew = vt1708_control_templates[type];
+ *knew = via_control_templates[type];
knew->name = kstrdup(name, GFP_KERNEL);
if (!knew->name)
return -ENOMEM;
+ if (get_amp_nid_(val))
+ knew->subdevice = HDA_SUBDEV_NID_FLAG | get_amp_nid_(val);
knew->private_value = val;
return 0;
}
@@ -293,8 +462,8 @@ static void via_free_kctls(struct hda_codec *codec)
}
/* create input playback/capture controls for the given pin */
-static int via_new_analog_input(struct via_spec *spec, hda_nid_t pin,
- const char *ctlname, int idx, int mix_nid)
+static int via_new_analog_input(struct via_spec *spec, const char *ctlname,
+ int idx, int mix_nid)
{
char name[32];
int err;
@@ -305,7 +474,7 @@ static int via_new_analog_input(struct via_spec *spec, hda_nid_t pin,
if (err < 0)
return err;
sprintf(name, "%s Playback Switch", ctlname);
- err = via_add_control(spec, VIA_CTL_WIDGET_MUTE, name,
+ err = via_add_control(spec, VIA_CTL_WIDGET_ANALOG_MUTE, name,
HDA_COMPOSE_AMP_VAL(mix_nid, 3, idx, HDA_INPUT));
if (err < 0)
return err;
@@ -322,7 +491,7 @@ static void via_auto_set_output_and_unmute(struct hda_codec *codec,
snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE,
AMP_OUT_UNMUTE);
if (snd_hda_query_pin_caps(codec, nid) & AC_PINCAP_EAPD)
- snd_hda_codec_write(codec, nid, 0,
+ snd_hda_codec_write(codec, nid, 0,
AC_VERB_SET_EAPD_BTLENABLE, 0x02);
}
@@ -343,10 +512,13 @@ static void via_auto_init_hp_out(struct hda_codec *codec)
{
struct via_spec *spec = codec->spec;
hda_nid_t pin;
+ int i;
- pin = spec->autocfg.hp_pins[0];
- if (pin) /* connect to front */
- via_auto_set_output_and_unmute(codec, pin, PIN_HP, 0);
+ for (i = 0; i < spec->autocfg.hp_outs; i++) {
+ pin = spec->autocfg.hp_pins[i];
+ if (pin) /* connect to front */
+ via_auto_set_output_and_unmute(codec, pin, PIN_HP, 0);
+ }
}
static void via_auto_init_analog_input(struct hda_codec *codec)
@@ -364,6 +536,502 @@ static void via_auto_init_analog_input(struct hda_codec *codec)
}
}
+
+static int is_smart51_pins(struct via_spec *spec, hda_nid_t pin);
+
+static void set_pin_power_state(struct hda_codec *codec, hda_nid_t nid,
+ unsigned int *affected_parm)
+{
+ unsigned parm;
+ unsigned def_conf = snd_hda_codec_get_pincfg(codec, nid);
+ unsigned no_presence = (def_conf & AC_DEFCFG_MISC)
+ >> AC_DEFCFG_MISC_SHIFT
+ & AC_DEFCFG_MISC_NO_PRESENCE; /* do not support pin sense */
+ unsigned present = snd_hda_jack_detect(codec, nid);
+ struct via_spec *spec = codec->spec;
+ if ((spec->smart51_enabled && is_smart51_pins(spec, nid))
+ || ((no_presence || present)
+ && get_defcfg_connect(def_conf) != AC_JACK_PORT_NONE)) {
+ *affected_parm = AC_PWRST_D0; /* if it's connected */
+ parm = AC_PWRST_D0;
+ } else
+ parm = AC_PWRST_D3;
+
+ snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_POWER_STATE, parm);
+}
+
+static void set_jack_power_state(struct hda_codec *codec)
+{
+ struct via_spec *spec = codec->spec;
+ int imux_is_smixer;
+ unsigned int parm;
+
+ if (spec->codec_type == VT1702) {
+ imux_is_smixer = snd_hda_codec_read(
+ codec, 0x13, 0, AC_VERB_GET_CONNECT_SEL, 0x00) == 3;
+ /* inputs */
+ /* PW 1/2/5 (14h/15h/18h) */
+ parm = AC_PWRST_D3;
+ set_pin_power_state(codec, 0x14, &parm);
+ set_pin_power_state(codec, 0x15, &parm);
+ set_pin_power_state(codec, 0x18, &parm);
+ if (imux_is_smixer)
+ parm = AC_PWRST_D0; /* SW0 = stereo mixer (idx 3) */
+ /* SW0 (13h), AIW 0/1/2 (12h/1fh/20h) */
+ snd_hda_codec_write(codec, 0x13, 0, AC_VERB_SET_POWER_STATE,
+ parm);
+ snd_hda_codec_write(codec, 0x12, 0, AC_VERB_SET_POWER_STATE,
+ parm);
+ snd_hda_codec_write(codec, 0x1f, 0, AC_VERB_SET_POWER_STATE,
+ parm);
+ snd_hda_codec_write(codec, 0x20, 0, AC_VERB_SET_POWER_STATE,
+ parm);
+
+ /* outputs */
+ /* PW 3/4 (16h/17h) */
+ parm = AC_PWRST_D3;
+ set_pin_power_state(codec, 0x16, &parm);
+ set_pin_power_state(codec, 0x17, &parm);
+ /* MW0 (1ah), AOW 0/1 (10h/1dh) */
+ snd_hda_codec_write(codec, 0x1a, 0, AC_VERB_SET_POWER_STATE,
+ imux_is_smixer ? AC_PWRST_D0 : parm);
+ snd_hda_codec_write(codec, 0x10, 0, AC_VERB_SET_POWER_STATE,
+ parm);
+ snd_hda_codec_write(codec, 0x1d, 0, AC_VERB_SET_POWER_STATE,
+ parm);
+ } else if (spec->codec_type == VT1708B_8CH
+ || spec->codec_type == VT1708B_4CH
+ || spec->codec_type == VT1708S) {
+ /* SW0 (17h) = stereo mixer */
+ int is_8ch = spec->codec_type != VT1708B_4CH;
+ imux_is_smixer = snd_hda_codec_read(
+ codec, 0x17, 0, AC_VERB_GET_CONNECT_SEL, 0x00)
+ == ((spec->codec_type == VT1708S) ? 5 : 0);
+ /* inputs */
+ /* PW 1/2/5 (1ah/1bh/1eh) */
+ parm = AC_PWRST_D3;
+ set_pin_power_state(codec, 0x1a, &parm);
+ set_pin_power_state(codec, 0x1b, &parm);
+ set_pin_power_state(codec, 0x1e, &parm);
+ if (imux_is_smixer)
+ parm = AC_PWRST_D0;
+ /* SW0 (17h), AIW 0/1 (13h/14h) */
+ snd_hda_codec_write(codec, 0x17, 0, AC_VERB_SET_POWER_STATE,
+ parm);
+ snd_hda_codec_write(codec, 0x13, 0, AC_VERB_SET_POWER_STATE,
+ parm);
+ snd_hda_codec_write(codec, 0x14, 0, AC_VERB_SET_POWER_STATE,
+ parm);
+
+ /* outputs */
+ /* PW0 (19h), SW1 (18h), AOW1 (11h) */
+ parm = AC_PWRST_D3;
+ set_pin_power_state(codec, 0x19, &parm);
+ snd_hda_codec_write(codec, 0x18, 0, AC_VERB_SET_POWER_STATE,
+ parm);
+ snd_hda_codec_write(codec, 0x11, 0, AC_VERB_SET_POWER_STATE,
+ parm);
+
+ /* PW6 (22h), SW2 (26h), AOW2 (24h) */
+ if (is_8ch) {
+ parm = AC_PWRST_D3;
+ set_pin_power_state(codec, 0x22, &parm);
+ snd_hda_codec_write(codec, 0x26, 0,
+ AC_VERB_SET_POWER_STATE, parm);
+ snd_hda_codec_write(codec, 0x24, 0,
+ AC_VERB_SET_POWER_STATE, parm);
+ }
+
+ /* PW 3/4/7 (1ch/1dh/23h) */
+ parm = AC_PWRST_D3;
+ /* force to D0 for internal Speaker */
+ set_pin_power_state(codec, 0x1c, &parm);
+ set_pin_power_state(codec, 0x1d, &parm);
+ if (is_8ch)
+ set_pin_power_state(codec, 0x23, &parm);
+ /* MW0 (16h), Sw3 (27h), AOW 0/3 (10h/25h) */
+ snd_hda_codec_write(codec, 0x16, 0, AC_VERB_SET_POWER_STATE,
+ imux_is_smixer ? AC_PWRST_D0 : parm);
+ snd_hda_codec_write(codec, 0x10, 0, AC_VERB_SET_POWER_STATE,
+ parm);
+ if (is_8ch) {
+ snd_hda_codec_write(codec, 0x25, 0,
+ AC_VERB_SET_POWER_STATE, parm);
+ snd_hda_codec_write(codec, 0x27, 0,
+ AC_VERB_SET_POWER_STATE, parm);
+ }
+ } else if (spec->codec_type == VT1718S) {
+ /* MUX6 (1eh) = stereo mixer */
+ imux_is_smixer = snd_hda_codec_read(
+ codec, 0x1e, 0, AC_VERB_GET_CONNECT_SEL, 0x00) == 5;
+ /* inputs */
+ /* PW 5/6/7 (29h/2ah/2bh) */
+ parm = AC_PWRST_D3;
+ set_pin_power_state(codec, 0x29, &parm);
+ set_pin_power_state(codec, 0x2a, &parm);
+ set_pin_power_state(codec, 0x2b, &parm);
+ if (imux_is_smixer)
+ parm = AC_PWRST_D0;
+ /* MUX6/7 (1eh/1fh), AIW 0/1 (10h/11h) */
+ snd_hda_codec_write(codec, 0x1e, 0, AC_VERB_SET_POWER_STATE,
+ parm);
+ snd_hda_codec_write(codec, 0x1f, 0, AC_VERB_SET_POWER_STATE,
+ parm);
+ snd_hda_codec_write(codec, 0x10, 0, AC_VERB_SET_POWER_STATE,
+ parm);
+ snd_hda_codec_write(codec, 0x11, 0, AC_VERB_SET_POWER_STATE,
+ parm);
+
+ /* outputs */
+ /* PW3 (27h), MW2 (1ah), AOW3 (bh) */
+ parm = AC_PWRST_D3;
+ set_pin_power_state(codec, 0x27, &parm);
+ snd_hda_codec_write(codec, 0x1a, 0, AC_VERB_SET_POWER_STATE,
+ parm);
+ snd_hda_codec_write(codec, 0xb, 0, AC_VERB_SET_POWER_STATE,
+ parm);
+
+ /* PW2 (26h), AOW2 (ah) */
+ parm = AC_PWRST_D3;
+ set_pin_power_state(codec, 0x26, &parm);
+ snd_hda_codec_write(codec, 0xa, 0, AC_VERB_SET_POWER_STATE,
+ parm);
+
+ /* PW0/1 (24h/25h) */
+ parm = AC_PWRST_D3;
+ set_pin_power_state(codec, 0x24, &parm);
+ set_pin_power_state(codec, 0x25, &parm);
+ if (!spec->hp_independent_mode) /* check for redirected HP */
+ set_pin_power_state(codec, 0x28, &parm);
+ snd_hda_codec_write(codec, 0x8, 0, AC_VERB_SET_POWER_STATE,
+ parm);
+ snd_hda_codec_write(codec, 0x9, 0, AC_VERB_SET_POWER_STATE,
+ parm);
+ /* MW9 (21h), Mw2 (1ah), AOW0 (8h) */
+ snd_hda_codec_write(codec, 0x21, 0, AC_VERB_SET_POWER_STATE,
+ imux_is_smixer ? AC_PWRST_D0 : parm);
+ if (spec->hp_independent_mode) {
+ /* PW4 (28h), MW3 (1bh), MUX1(34h), AOW4 (ch) */
+ parm = AC_PWRST_D3;
+ set_pin_power_state(codec, 0x28, &parm);
+ snd_hda_codec_write(codec, 0x1b, 0,
+ AC_VERB_SET_POWER_STATE, parm);
+ snd_hda_codec_write(codec, 0x34, 0,
+ AC_VERB_SET_POWER_STATE, parm);
+ snd_hda_codec_write(codec, 0xc, 0,
+ AC_VERB_SET_POWER_STATE, parm);
+ }
+ } else if (spec->codec_type == VT1716S) {
+ unsigned int mono_out, present;
+ /* SW0 (17h) = stereo mixer */
+ imux_is_smixer = snd_hda_codec_read(
+ codec, 0x17, 0, AC_VERB_GET_CONNECT_SEL, 0x00) == 5;
+ /* inputs */
+ /* PW 1/2/5 (1ah/1bh/1eh) */
+ parm = AC_PWRST_D3;
+ set_pin_power_state(codec, 0x1a, &parm);
+ set_pin_power_state(codec, 0x1b, &parm);
+ set_pin_power_state(codec, 0x1e, &parm);
+ if (imux_is_smixer)
+ parm = AC_PWRST_D0;
+ /* SW0 (17h), AIW0(13h) */
+ snd_hda_codec_write(codec, 0x17, 0, AC_VERB_SET_POWER_STATE,
+ parm);
+ snd_hda_codec_write(codec, 0x13, 0, AC_VERB_SET_POWER_STATE,
+ parm);
+
+ parm = AC_PWRST_D3;
+ set_pin_power_state(codec, 0x1e, &parm);
+ /* PW11 (22h) */
+ if (spec->dmic_enabled)
+ set_pin_power_state(codec, 0x22, &parm);
+ else
+ snd_hda_codec_write(
+ codec, 0x22, 0,
+ AC_VERB_SET_POWER_STATE, AC_PWRST_D3);
+
+ /* SW2(26h), AIW1(14h) */
+ snd_hda_codec_write(codec, 0x26, 0, AC_VERB_SET_POWER_STATE,
+ parm);
+ snd_hda_codec_write(codec, 0x14, 0, AC_VERB_SET_POWER_STATE,
+ parm);
+
+ /* outputs */
+ /* PW0 (19h), SW1 (18h), AOW1 (11h) */
+ parm = AC_PWRST_D3;
+ set_pin_power_state(codec, 0x19, &parm);
+ /* Smart 5.1 PW2(1bh) */
+ if (spec->smart51_enabled)
+ set_pin_power_state(codec, 0x1b, &parm);
+ snd_hda_codec_write(codec, 0x18, 0, AC_VERB_SET_POWER_STATE,
+ parm);
+ snd_hda_codec_write(codec, 0x11, 0, AC_VERB_SET_POWER_STATE,
+ parm);
+
+ /* PW7 (23h), SW3 (27h), AOW3 (25h) */
+ parm = AC_PWRST_D3;
+ set_pin_power_state(codec, 0x23, &parm);
+ /* Smart 5.1 PW1(1ah) */
+ if (spec->smart51_enabled)
+ set_pin_power_state(codec, 0x1a, &parm);
+ snd_hda_codec_write(codec, 0x27, 0, AC_VERB_SET_POWER_STATE,
+ parm);
+
+ /* Smart 5.1 PW5(1eh) */
+ if (spec->smart51_enabled)
+ set_pin_power_state(codec, 0x1e, &parm);
+ snd_hda_codec_write(codec, 0x25, 0, AC_VERB_SET_POWER_STATE,
+ parm);
+
+ /* Mono out */
+ /* SW4(28h)->MW1(29h)-> PW12 (2ah)*/
+ present = snd_hda_jack_detect(codec, 0x1c);
+ if (present)
+ mono_out = 0;
+ else {
+ present = snd_hda_jack_detect(codec, 0x1d);
+ if (!spec->hp_independent_mode && present)
+ mono_out = 0;
+ else
+ mono_out = 1;
+ }
+ parm = mono_out ? AC_PWRST_D0 : AC_PWRST_D3;
+ snd_hda_codec_write(codec, 0x28, 0, AC_VERB_SET_POWER_STATE,
+ parm);
+ snd_hda_codec_write(codec, 0x29, 0, AC_VERB_SET_POWER_STATE,
+ parm);
+ snd_hda_codec_write(codec, 0x2a, 0, AC_VERB_SET_POWER_STATE,
+ parm);
+
+ /* PW 3/4 (1ch/1dh) */
+ parm = AC_PWRST_D3;
+ set_pin_power_state(codec, 0x1c, &parm);
+ set_pin_power_state(codec, 0x1d, &parm);
+ /* HP Independent Mode, power on AOW3 */
+ if (spec->hp_independent_mode)
+ snd_hda_codec_write(codec, 0x25, 0,
+ AC_VERB_SET_POWER_STATE, parm);
+
+ /* force to D0 for internal Speaker */
+ /* MW0 (16h), AOW0 (10h) */
+ snd_hda_codec_write(codec, 0x16, 0, AC_VERB_SET_POWER_STATE,
+ imux_is_smixer ? AC_PWRST_D0 : parm);
+ snd_hda_codec_write(codec, 0x10, 0, AC_VERB_SET_POWER_STATE,
+ mono_out ? AC_PWRST_D0 : parm);
+ } else if (spec->codec_type == VT2002P) {
+ unsigned int present;
+ /* MUX9 (1eh) = stereo mixer */
+ imux_is_smixer = snd_hda_codec_read(
+ codec, 0x1e, 0, AC_VERB_GET_CONNECT_SEL, 0x00) == 3;
+ /* inputs */
+ /* PW 5/6/7 (29h/2ah/2bh) */
+ parm = AC_PWRST_D3;
+ set_pin_power_state(codec, 0x29, &parm);
+ set_pin_power_state(codec, 0x2a, &parm);
+ set_pin_power_state(codec, 0x2b, &parm);
+ if (imux_is_smixer)
+ parm = AC_PWRST_D0;
+ /* MUX9/10 (1eh/1fh), AIW 0/1 (10h/11h) */
+ snd_hda_codec_write(codec, 0x1e, 0,
+ AC_VERB_SET_POWER_STATE, parm);
+ snd_hda_codec_write(codec, 0x1f, 0,
+ AC_VERB_SET_POWER_STATE, parm);
+ snd_hda_codec_write(codec, 0x10, 0,
+ AC_VERB_SET_POWER_STATE, parm);
+ snd_hda_codec_write(codec, 0x11, 0,
+ AC_VERB_SET_POWER_STATE, parm);
+
+ /* outputs */
+ /* AOW0 (8h)*/
+ snd_hda_codec_write(codec, 0x8, 0,
+ AC_VERB_SET_POWER_STATE, AC_PWRST_D0);
+
+ /* PW4 (26h), MW4 (1ch), MUX4(37h) */
+ parm = AC_PWRST_D3;
+ set_pin_power_state(codec, 0x26, &parm);
+ snd_hda_codec_write(codec, 0x1c, 0,
+ AC_VERB_SET_POWER_STATE, parm);
+ snd_hda_codec_write(codec, 0x37,
+ 0, AC_VERB_SET_POWER_STATE, parm);
+
+ /* PW1 (25h), MW1 (19h), MUX1(35h), AOW1 (9h) */
+ parm = AC_PWRST_D3;
+ set_pin_power_state(codec, 0x25, &parm);
+ snd_hda_codec_write(codec, 0x19, 0,
+ AC_VERB_SET_POWER_STATE, parm);
+ snd_hda_codec_write(codec, 0x35, 0,
+ AC_VERB_SET_POWER_STATE, parm);
+ if (spec->hp_independent_mode) {
+ snd_hda_codec_write(codec, 0x9, 0,
+ AC_VERB_SET_POWER_STATE, parm);
+ }
+
+ /* Class-D */
+ /* PW0 (24h), MW0(18h), MUX0(34h) */
+ present = snd_hda_jack_detect(codec, 0x25);
+ parm = AC_PWRST_D3;
+ set_pin_power_state(codec, 0x24, &parm);
+ if (present) {
+ snd_hda_codec_write(
+ codec, 0x18, 0,
+ AC_VERB_SET_POWER_STATE, AC_PWRST_D3);
+ snd_hda_codec_write(
+ codec, 0x34, 0,
+ AC_VERB_SET_POWER_STATE, AC_PWRST_D3);
+ } else {
+ snd_hda_codec_write(
+ codec, 0x18, 0,
+ AC_VERB_SET_POWER_STATE, AC_PWRST_D0);
+ snd_hda_codec_write(
+ codec, 0x34, 0,
+ AC_VERB_SET_POWER_STATE, AC_PWRST_D0);
+ }
+
+ /* Mono Out */
+ /* PW15 (31h), MW8(17h), MUX8(3bh) */
+ present = snd_hda_jack_detect(codec, 0x26);
+ parm = AC_PWRST_D3;
+ set_pin_power_state(codec, 0x31, &parm);
+ if (present) {
+ snd_hda_codec_write(
+ codec, 0x17, 0,
+ AC_VERB_SET_POWER_STATE, AC_PWRST_D3);
+ snd_hda_codec_write(
+ codec, 0x3b, 0,
+ AC_VERB_SET_POWER_STATE, AC_PWRST_D3);
+ } else {
+ snd_hda_codec_write(
+ codec, 0x17, 0,
+ AC_VERB_SET_POWER_STATE, AC_PWRST_D0);
+ snd_hda_codec_write(
+ codec, 0x3b, 0,
+ AC_VERB_SET_POWER_STATE, AC_PWRST_D0);
+ }
+
+ /* MW9 (21h) */
+ if (imux_is_smixer || !is_aa_path_mute(codec))
+ snd_hda_codec_write(
+ codec, 0x21, 0,
+ AC_VERB_SET_POWER_STATE, AC_PWRST_D0);
+ else
+ snd_hda_codec_write(
+ codec, 0x21, 0,
+ AC_VERB_SET_POWER_STATE, AC_PWRST_D3);
+ } else if (spec->codec_type == VT1812) {
+ unsigned int present;
+ /* MUX10 (1eh) = stereo mixer */
+ imux_is_smixer = snd_hda_codec_read(
+ codec, 0x1e, 0, AC_VERB_GET_CONNECT_SEL, 0x00) == 5;
+ /* inputs */
+ /* PW 5/6/7 (29h/2ah/2bh) */
+ parm = AC_PWRST_D3;
+ set_pin_power_state(codec, 0x29, &parm);
+ set_pin_power_state(codec, 0x2a, &parm);
+ set_pin_power_state(codec, 0x2b, &parm);
+ if (imux_is_smixer)
+ parm = AC_PWRST_D0;
+ /* MUX10/11 (1eh/1fh), AIW 0/1 (10h/11h) */
+ snd_hda_codec_write(codec, 0x1e, 0,
+ AC_VERB_SET_POWER_STATE, parm);
+ snd_hda_codec_write(codec, 0x1f, 0,
+ AC_VERB_SET_POWER_STATE, parm);
+ snd_hda_codec_write(codec, 0x10, 0,
+ AC_VERB_SET_POWER_STATE, parm);
+ snd_hda_codec_write(codec, 0x11, 0,
+ AC_VERB_SET_POWER_STATE, parm);
+
+ /* outputs */
+ /* AOW0 (8h)*/
+ snd_hda_codec_write(codec, 0x8, 0,
+ AC_VERB_SET_POWER_STATE, AC_PWRST_D0);
+
+ /* PW4 (28h), MW4 (18h), MUX4(38h) */
+ parm = AC_PWRST_D3;
+ set_pin_power_state(codec, 0x28, &parm);
+ snd_hda_codec_write(codec, 0x18, 0,
+ AC_VERB_SET_POWER_STATE, parm);
+ snd_hda_codec_write(codec, 0x38, 0,
+ AC_VERB_SET_POWER_STATE, parm);
+
+ /* PW1 (25h), MW1 (15h), MUX1(35h), AOW1 (9h) */
+ parm = AC_PWRST_D3;
+ set_pin_power_state(codec, 0x25, &parm);
+ snd_hda_codec_write(codec, 0x15, 0,
+ AC_VERB_SET_POWER_STATE, parm);
+ snd_hda_codec_write(codec, 0x35, 0,
+ AC_VERB_SET_POWER_STATE, parm);
+ if (spec->hp_independent_mode) {
+ snd_hda_codec_write(codec, 0x9, 0,
+ AC_VERB_SET_POWER_STATE, parm);
+ }
+
+ /* Internal Speaker */
+ /* PW0 (24h), MW0(14h), MUX0(34h) */
+ present = snd_hda_jack_detect(codec, 0x25);
+ parm = AC_PWRST_D3;
+ set_pin_power_state(codec, 0x24, &parm);
+ if (present) {
+ snd_hda_codec_write(codec, 0x14, 0,
+ AC_VERB_SET_POWER_STATE,
+ AC_PWRST_D3);
+ snd_hda_codec_write(codec, 0x34, 0,
+ AC_VERB_SET_POWER_STATE,
+ AC_PWRST_D3);
+ } else {
+ snd_hda_codec_write(codec, 0x14, 0,
+ AC_VERB_SET_POWER_STATE,
+ AC_PWRST_D0);
+ snd_hda_codec_write(codec, 0x34, 0,
+ AC_VERB_SET_POWER_STATE,
+ AC_PWRST_D0);
+ }
+ /* Mono Out */
+ /* PW13 (31h), MW13(1ch), MUX13(3ch), MW14(3eh) */
+ present = snd_hda_jack_detect(codec, 0x28);
+ parm = AC_PWRST_D3;
+ set_pin_power_state(codec, 0x31, &parm);
+ if (present) {
+ snd_hda_codec_write(codec, 0x1c, 0,
+ AC_VERB_SET_POWER_STATE,
+ AC_PWRST_D3);
+ snd_hda_codec_write(codec, 0x3c, 0,
+ AC_VERB_SET_POWER_STATE,
+ AC_PWRST_D3);
+ snd_hda_codec_write(codec, 0x3e, 0,
+ AC_VERB_SET_POWER_STATE,
+ AC_PWRST_D3);
+ } else {
+ snd_hda_codec_write(codec, 0x1c, 0,
+ AC_VERB_SET_POWER_STATE,
+ AC_PWRST_D0);
+ snd_hda_codec_write(codec, 0x3c, 0,
+ AC_VERB_SET_POWER_STATE,
+ AC_PWRST_D0);
+ snd_hda_codec_write(codec, 0x3e, 0,
+ AC_VERB_SET_POWER_STATE,
+ AC_PWRST_D0);
+ }
+
+ /* PW15 (33h), MW15 (1dh), MUX15(3dh) */
+ parm = AC_PWRST_D3;
+ set_pin_power_state(codec, 0x33, &parm);
+ snd_hda_codec_write(codec, 0x1d, 0,
+ AC_VERB_SET_POWER_STATE, parm);
+ snd_hda_codec_write(codec, 0x3d, 0,
+ AC_VERB_SET_POWER_STATE, parm);
+
+ /* MW9 (21h) */
+ if (imux_is_smixer || !is_aa_path_mute(codec))
+ snd_hda_codec_write(
+ codec, 0x21, 0,
+ AC_VERB_SET_POWER_STATE, AC_PWRST_D0);
+ else
+ snd_hda_codec_write(
+ codec, 0x21, 0,
+ AC_VERB_SET_POWER_STATE, AC_PWRST_D3);
+ }
+}
+
/*
* input MUX handling
*/
@@ -395,6 +1063,14 @@ static int via_mux_enum_put(struct snd_kcontrol *kcontrol,
if (!spec->mux_nids[adc_idx])
return -EINVAL;
+ /* switch to D0 beofre change index */
+ if (snd_hda_codec_read(codec, spec->mux_nids[adc_idx], 0,
+ AC_VERB_GET_POWER_STATE, 0x00) != AC_PWRST_D0)
+ snd_hda_codec_write(codec, spec->mux_nids[adc_idx], 0,
+ AC_VERB_SET_POWER_STATE, AC_PWRST_D0);
+ /* update jack power state */
+ set_jack_power_state(codec);
+
return snd_hda_input_mux_put(codec, spec->input_mux, ucontrol,
spec->mux_nids[adc_idx],
&spec->cur_mux[adc_idx]);
@@ -413,16 +1089,74 @@ static int via_independent_hp_get(struct snd_kcontrol *kcontrol,
{
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
struct via_spec *spec = codec->spec;
- hda_nid_t nid = spec->autocfg.hp_pins[0];
- unsigned int pinsel = snd_hda_codec_read(codec, nid, 0,
- AC_VERB_GET_CONNECT_SEL,
- 0x00);
-
+ hda_nid_t nid;
+ unsigned int pinsel;
+
+ switch (spec->codec_type) {
+ case VT1718S:
+ nid = 0x34;
+ break;
+ case VT2002P:
+ nid = 0x35;
+ break;
+ case VT1812:
+ nid = 0x3d;
+ break;
+ default:
+ nid = spec->autocfg.hp_pins[0];
+ break;
+ }
+ /* use !! to translate conn sel 2 for VT1718S */
+ pinsel = !!snd_hda_codec_read(codec, nid, 0,
+ AC_VERB_GET_CONNECT_SEL,
+ 0x00);
ucontrol->value.enumerated.item[0] = pinsel;
return 0;
}
+static void activate_ctl(struct hda_codec *codec, const char *name, int active)
+{
+ struct snd_kcontrol *ctl = snd_hda_find_mixer_ctl(codec, name);
+ if (ctl) {
+ ctl->vd[0].access &= ~SNDRV_CTL_ELEM_ACCESS_INACTIVE;
+ ctl->vd[0].access |= active
+ ? 0 : SNDRV_CTL_ELEM_ACCESS_INACTIVE;
+ snd_ctl_notify(codec->bus->card,
+ SNDRV_CTL_EVENT_MASK_VALUE, &ctl->id);
+ }
+}
+
+static int update_side_mute_status(struct hda_codec *codec)
+{
+ /* mute side channel */
+ struct via_spec *spec = codec->spec;
+ unsigned int parm = spec->hp_independent_mode
+ ? AMP_OUT_MUTE : AMP_OUT_UNMUTE;
+ hda_nid_t sw3;
+
+ switch (spec->codec_type) {
+ case VT1708:
+ sw3 = 0x1b;
+ break;
+ case VT1709_10CH:
+ sw3 = 0x29;
+ break;
+ case VT1708B_8CH:
+ case VT1708S:
+ sw3 = 0x27;
+ break;
+ default:
+ sw3 = 0;
+ break;
+ }
+
+ if (sw3)
+ snd_hda_codec_write(codec, sw3, 0, AC_VERB_SET_AMP_GAIN_MUTE,
+ parm);
+ return 0;
+}
+
static int via_independent_hp_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
@@ -430,47 +1164,46 @@ static int via_independent_hp_put(struct snd_kcontrol *kcontrol,
struct via_spec *spec = codec->spec;
hda_nid_t nid = spec->autocfg.hp_pins[0];
unsigned int pinsel = ucontrol->value.enumerated.item[0];
- unsigned int con_nid = snd_hda_codec_read(codec, nid, 0,
- AC_VERB_GET_CONNECT_LIST, 0) & 0xff;
-
- if (con_nid == spec->multiout.hp_nid) {
- if (pinsel == 0) {
- if (!spec->hp_independent_mode) {
- if (spec->multiout.num_dacs > 1)
- spec->multiout.num_dacs -= 1;
- spec->hp_independent_mode = 1;
- }
- } else if (pinsel == 1) {
- if (spec->hp_independent_mode) {
- if (spec->multiout.num_dacs > 1)
- spec->multiout.num_dacs += 1;
- spec->hp_independent_mode = 0;
- }
- }
- } else {
- if (pinsel == 0) {
- if (spec->hp_independent_mode) {
- if (spec->multiout.num_dacs > 1)
- spec->multiout.num_dacs += 1;
- spec->hp_independent_mode = 0;
- }
- } else if (pinsel == 1) {
- if (!spec->hp_independent_mode) {
- if (spec->multiout.num_dacs > 1)
- spec->multiout.num_dacs -= 1;
- spec->hp_independent_mode = 1;
- }
- }
+ /* Get Independent Mode index of headphone pin widget */
+ spec->hp_independent_mode = spec->hp_independent_mode_index == pinsel
+ ? 1 : 0;
+
+ switch (spec->codec_type) {
+ case VT1718S:
+ nid = 0x34;
+ pinsel = pinsel ? 2 : 0; /* indep HP use AOW4 (index 2) */
+ spec->multiout.num_dacs = 4;
+ break;
+ case VT2002P:
+ nid = 0x35;
+ break;
+ case VT1812:
+ nid = 0x3d;
+ break;
+ default:
+ nid = spec->autocfg.hp_pins[0];
+ break;
+ }
+ snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CONNECT_SEL, pinsel);
+
+ if (spec->multiout.hp_nid && spec->multiout.hp_nid
+ != spec->multiout.dac_nids[HDA_FRONT])
+ snd_hda_codec_setup_stream(codec, spec->multiout.hp_nid,
+ 0, 0, 0);
+
+ update_side_mute_status(codec);
+ /* update HP volume/swtich active state */
+ if (spec->codec_type == VT1708S
+ || spec->codec_type == VT1702
+ || spec->codec_type == VT1718S
+ || spec->codec_type == VT1716S
+ || spec->codec_type == VT2002P
+ || spec->codec_type == VT1812) {
+ activate_ctl(codec, "Headphone Playback Volume",
+ spec->hp_independent_mode);
+ activate_ctl(codec, "Headphone Playback Switch",
+ spec->hp_independent_mode);
}
- snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CONNECT_SEL,
- pinsel);
-
- if (spec->multiout.hp_nid &&
- spec->multiout.hp_nid != spec->multiout.dac_nids[HDA_FRONT])
- snd_hda_codec_setup_stream(codec,
- spec->multiout.hp_nid,
- 0, 0, 0);
-
return 0;
}
@@ -486,6 +1219,175 @@ static struct snd_kcontrol_new via_hp_mixer[] = {
{ } /* end */
};
+static void notify_aa_path_ctls(struct hda_codec *codec)
+{
+ int i;
+ struct snd_ctl_elem_id id;
+ const char *labels[] = {"Mic", "Front Mic", "Line"};
+
+ memset(&id, 0, sizeof(id));
+ id.iface = SNDRV_CTL_ELEM_IFACE_MIXER;
+ for (i = 0; i < ARRAY_SIZE(labels); i++) {
+ sprintf(id.name, "%s Playback Volume", labels[i]);
+ snd_ctl_notify(codec->bus->card, SNDRV_CTL_EVENT_MASK_VALUE,
+ &id);
+ }
+}
+
+static void mute_aa_path(struct hda_codec *codec, int mute)
+{
+ struct via_spec *spec = codec->spec;
+ hda_nid_t nid_mixer;
+ int start_idx;
+ int end_idx;
+ int i;
+ /* get nid of MW0 and start & end index */
+ switch (spec->codec_type) {
+ case VT1708:
+ nid_mixer = 0x17;
+ start_idx = 2;
+ end_idx = 4;
+ break;
+ case VT1709_10CH:
+ case VT1709_6CH:
+ nid_mixer = 0x18;
+ start_idx = 2;
+ end_idx = 4;
+ break;
+ case VT1708B_8CH:
+ case VT1708B_4CH:
+ case VT1708S:
+ case VT1716S:
+ nid_mixer = 0x16;
+ start_idx = 2;
+ end_idx = 4;
+ break;
+ default:
+ return;
+ }
+ /* check AA path's mute status */
+ for (i = start_idx; i <= end_idx; i++) {
+ int val = mute ? HDA_AMP_MUTE : HDA_AMP_UNMUTE;
+ snd_hda_codec_amp_stereo(codec, nid_mixer, HDA_INPUT, i,
+ HDA_AMP_MUTE, val);
+ }
+}
+static int is_smart51_pins(struct via_spec *spec, hda_nid_t pin)
+{
+ int res = 0;
+ int index;
+ for (index = AUTO_PIN_MIC; index < AUTO_PIN_FRONT_LINE; index++) {
+ if (pin == spec->autocfg.input_pins[index]) {
+ res = 1;
+ break;
+ }
+ }
+ return res;
+}
+
+static int via_smart51_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;
+}
+
+static int via_smart51_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct via_spec *spec = codec->spec;
+ int index[] = { AUTO_PIN_MIC, AUTO_PIN_FRONT_MIC, AUTO_PIN_LINE };
+ int on = 1;
+ int i;
+
+ for (i = 0; i < ARRAY_SIZE(index); i++) {
+ hda_nid_t nid = spec->autocfg.input_pins[index[i]];
+ if (nid) {
+ int ctl =
+ snd_hda_codec_read(codec, nid, 0,
+ AC_VERB_GET_PIN_WIDGET_CONTROL,
+ 0);
+ if (i == AUTO_PIN_FRONT_MIC
+ && spec->hp_independent_mode
+ && spec->codec_type != VT1718S)
+ continue; /* ignore FMic for independent HP */
+ if (ctl & AC_PINCTL_IN_EN
+ && !(ctl & AC_PINCTL_OUT_EN))
+ on = 0;
+ }
+ }
+ *ucontrol->value.integer.value = on;
+ return 0;
+}
+
+static int via_smart51_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct via_spec *spec = codec->spec;
+ int out_in = *ucontrol->value.integer.value
+ ? AC_PINCTL_OUT_EN : AC_PINCTL_IN_EN;
+ int index[] = { AUTO_PIN_MIC, AUTO_PIN_FRONT_MIC, AUTO_PIN_LINE };
+ int i;
+
+ for (i = 0; i < ARRAY_SIZE(index); i++) {
+ hda_nid_t nid = spec->autocfg.input_pins[index[i]];
+ if (i == AUTO_PIN_FRONT_MIC
+ && spec->hp_independent_mode
+ && spec->codec_type != VT1718S)
+ continue; /* don't retask FMic for independent HP */
+ if (nid) {
+ unsigned int parm = snd_hda_codec_read(
+ codec, nid, 0,
+ AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
+ parm &= ~(AC_PINCTL_IN_EN | AC_PINCTL_OUT_EN);
+ parm |= out_in;
+ snd_hda_codec_write(codec, nid, 0,
+ AC_VERB_SET_PIN_WIDGET_CONTROL,
+ parm);
+ if (out_in == AC_PINCTL_OUT_EN) {
+ mute_aa_path(codec, 1);
+ notify_aa_path_ctls(codec);
+ }
+ if (spec->codec_type == VT1718S)
+ snd_hda_codec_amp_stereo(
+ codec, nid, HDA_OUTPUT, 0, HDA_AMP_MUTE,
+ HDA_AMP_UNMUTE);
+ }
+ if (i == AUTO_PIN_FRONT_MIC) {
+ if (spec->codec_type == VT1708S
+ || spec->codec_type == VT1716S) {
+ /* input = index 1 (AOW3) */
+ snd_hda_codec_write(
+ codec, nid, 0,
+ AC_VERB_SET_CONNECT_SEL, 1);
+ snd_hda_codec_amp_stereo(
+ codec, nid, HDA_OUTPUT,
+ 0, HDA_AMP_MUTE, HDA_AMP_UNMUTE);
+ }
+ }
+ }
+ spec->smart51_enabled = *ucontrol->value.integer.value;
+ set_jack_power_state(codec);
+ return 1;
+}
+
+static struct snd_kcontrol_new via_smart51_mixer[] = {
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Smart 5.1",
+ .count = 1,
+ .info = via_smart51_info,
+ .get = via_smart51_get,
+ .put = via_smart51_put,
+ },
+ {} /* end */
+};
+
/* capture mixer elements */
static struct snd_kcontrol_new vt1708_capture_mixer[] = {
HDA_CODEC_VOLUME("Capture Volume", 0x15, 0x0, HDA_INPUT),
@@ -506,6 +1408,112 @@ static struct snd_kcontrol_new vt1708_capture_mixer[] = {
},
{ } /* end */
};
+
+/* check AA path's mute statue */
+static int is_aa_path_mute(struct hda_codec *codec)
+{
+ int mute = 1;
+ hda_nid_t nid_mixer;
+ int start_idx;
+ int end_idx;
+ int i;
+ struct via_spec *spec = codec->spec;
+ /* get nid of MW0 and start & end index */
+ switch (spec->codec_type) {
+ case VT1708B_8CH:
+ case VT1708B_4CH:
+ case VT1708S:
+ case VT1716S:
+ nid_mixer = 0x16;
+ start_idx = 2;
+ end_idx = 4;
+ break;
+ case VT1702:
+ nid_mixer = 0x1a;
+ start_idx = 1;
+ end_idx = 3;
+ break;
+ case VT1718S:
+ nid_mixer = 0x21;
+ start_idx = 1;
+ end_idx = 3;
+ break;
+ case VT2002P:
+ case VT1812:
+ nid_mixer = 0x21;
+ start_idx = 0;
+ end_idx = 2;
+ break;
+ default:
+ return 0;
+ }
+ /* check AA path's mute status */
+ for (i = start_idx; i <= end_idx; i++) {
+ unsigned int con_list = snd_hda_codec_read(
+ codec, nid_mixer, 0, AC_VERB_GET_CONNECT_LIST, i/4*4);
+ int shift = 8 * (i % 4);
+ hda_nid_t nid_pin = (con_list & (0xff << shift)) >> shift;
+ unsigned int defconf = snd_hda_codec_get_pincfg(codec, nid_pin);
+ if (get_defcfg_connect(defconf) == AC_JACK_PORT_COMPLEX) {
+ /* check mute status while the pin is connected */
+ int mute_l = snd_hda_codec_amp_read(codec, nid_mixer, 0,
+ HDA_INPUT, i) >> 7;
+ int mute_r = snd_hda_codec_amp_read(codec, nid_mixer, 1,
+ HDA_INPUT, i) >> 7;
+ if (!mute_l || !mute_r) {
+ mute = 0;
+ break;
+ }
+ }
+ }
+ return mute;
+}
+
+/* enter/exit analog low-current mode */
+static void analog_low_current_mode(struct hda_codec *codec, int stream_idle)
+{
+ struct via_spec *spec = codec->spec;
+ static int saved_stream_idle = 1; /* saved stream idle status */
+ int enable = is_aa_path_mute(codec);
+ unsigned int verb = 0;
+ unsigned int parm = 0;
+
+ if (stream_idle == -1) /* stream status did not change */
+ enable = enable && saved_stream_idle;
+ else {
+ enable = enable && stream_idle;
+ saved_stream_idle = stream_idle;
+ }
+
+ /* decide low current mode's verb & parameter */
+ switch (spec->codec_type) {
+ case VT1708B_8CH:
+ case VT1708B_4CH:
+ verb = 0xf70;
+ parm = enable ? 0x02 : 0x00; /* 0x02: 2/3x, 0x00: 1x */
+ break;
+ case VT1708S:
+ case VT1718S:
+ case VT1716S:
+ verb = 0xf73;
+ parm = enable ? 0x51 : 0xe1; /* 0x51: 4/28x, 0xe1: 1x */
+ break;
+ case VT1702:
+ verb = 0xf73;
+ parm = enable ? 0x01 : 0x1d; /* 0x01: 4/40x, 0x1d: 1x */
+ break;
+ case VT2002P:
+ case VT1812:
+ verb = 0xf93;
+ parm = enable ? 0x00 : 0xe0; /* 0x00: 4/40x, 0xe0: 1x */
+ break;
+ default:
+ return; /* other codecs are not supported */
+ }
+ /* send verb */
+ snd_hda_codec_write(codec, codec->afg, 0, verb, parm);
+}
+
/*
* generic initialization of ADC, input mixers and output mixers
*/
@@ -534,9 +1542,9 @@ static struct hda_verb vt1708_volume_init_verbs[] = {
{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
-
- /* Setup default input to PW4 */
- {0x20, AC_VERB_SET_CONNECT_SEL, 0x1},
+
+ /* Setup default input MW0 to PW4 */
+ {0x20, AC_VERB_SET_CONNECT_SEL, 0},
/* PW9 Output enable */
{0x25, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
{ }
@@ -547,30 +1555,13 @@ static int via_playback_pcm_open(struct hda_pcm_stream *hinfo,
struct snd_pcm_substream *substream)
{
struct via_spec *spec = codec->spec;
+ int idle = substream->pstr->substream_opened == 1
+ && substream->ref_count == 0;
+ analog_low_current_mode(codec, idle);
return snd_hda_multi_out_analog_open(codec, &spec->multiout, substream,
hinfo);
}
-static int via_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
- struct hda_codec *codec,
- unsigned int stream_tag,
- unsigned int format,
- struct snd_pcm_substream *substream)
-{
- struct via_spec *spec = codec->spec;
- return snd_hda_multi_out_analog_prepare(codec, &spec->multiout,
- stream_tag, format, substream);
-}
-
-static int via_playback_pcm_cleanup(struct hda_pcm_stream *hinfo,
- struct hda_codec *codec,
- struct snd_pcm_substream *substream)
-{
- struct via_spec *spec = codec->spec;
- return snd_hda_multi_out_analog_cleanup(codec, &spec->multiout);
-}
-
-
static void playback_multi_pcm_prep_0(struct hda_codec *codec,
unsigned int stream_tag,
unsigned int format,
@@ -615,8 +1606,8 @@ static void playback_multi_pcm_prep_0(struct hda_codec *codec,
snd_hda_codec_setup_stream(codec, nids[HDA_FRONT], stream_tag,
0, format);
- if (mout->hp_nid && mout->hp_nid != nids[HDA_FRONT] &&
- !spec->hp_independent_mode)
+ if (mout->hp_nid && mout->hp_nid != nids[HDA_FRONT]
+ && !spec->hp_independent_mode)
/* headphone out will just decode front left/right (stereo) */
snd_hda_codec_setup_stream(codec, mout->hp_nid, stream_tag,
0, format);
@@ -658,7 +1649,7 @@ static int via_playback_multi_pcm_prepare(struct hda_pcm_stream *hinfo,
snd_hda_codec_setup_stream(codec, mout->hp_nid,
stream_tag, 0, format);
}
-
+ vt1708_start_hp_work(spec);
return 0;
}
@@ -698,7 +1689,7 @@ static int via_playback_multi_pcm_cleanup(struct hda_pcm_stream *hinfo,
snd_hda_codec_setup_stream(codec, mout->hp_nid,
0, 0, 0);
}
-
+ vt1708_stop_hp_work(spec);
return 0;
}
@@ -779,7 +1770,7 @@ static struct hda_pcm_stream vt1708_pcm_analog_playback = {
};
static struct hda_pcm_stream vt1708_pcm_analog_s16_playback = {
- .substreams = 1,
+ .substreams = 2,
.channels_min = 2,
.channels_max = 8,
.nid = 0x10, /* NID to query formats and rates */
@@ -790,8 +1781,8 @@ static struct hda_pcm_stream vt1708_pcm_analog_s16_playback = {
.formats = SNDRV_PCM_FMTBIT_S16_LE,
.ops = {
.open = via_playback_pcm_open,
- .prepare = via_playback_pcm_prepare,
- .cleanup = via_playback_pcm_cleanup
+ .prepare = via_playback_multi_pcm_prepare,
+ .cleanup = via_playback_multi_pcm_cleanup
},
};
@@ -853,6 +1844,11 @@ static int via_build_controls(struct hda_codec *codec)
if (err < 0)
return err;
}
+
+ /* init power states */
+ set_jack_power_state(codec);
+ analog_low_current_mode(codec, 1);
+
via_free_kctls(codec); /* no longer needed */
return 0;
}
@@ -866,8 +1862,10 @@ static int via_build_pcms(struct hda_codec *codec)
codec->pcm_info = info;
info->name = spec->stream_name_analog;
- info->stream[SNDRV_PCM_STREAM_PLAYBACK] = *(spec->stream_analog_playback);
- info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = spec->multiout.dac_nids[0];
+ info->stream[SNDRV_PCM_STREAM_PLAYBACK] =
+ *(spec->stream_analog_playback);
+ info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid =
+ spec->multiout.dac_nids[0];
info->stream[SNDRV_PCM_STREAM_CAPTURE] = *(spec->stream_analog_capture);
info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->adc_nids[0];
@@ -904,20 +1902,58 @@ static void via_free(struct hda_codec *codec)
return;
via_free_kctls(codec);
+ vt1708_stop_hp_work(spec);
kfree(codec->spec);
}
/* mute internal speaker if HP is plugged */
static void via_hp_automute(struct hda_codec *codec)
{
- unsigned int present;
+ unsigned int present = 0;
struct via_spec *spec = codec->spec;
- present = snd_hda_codec_read(codec, spec->autocfg.hp_pins[0], 0,
- AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
- snd_hda_codec_amp_stereo(codec, spec->autocfg.line_out_pins[0],
- HDA_OUTPUT, 0, HDA_AMP_MUTE,
- present ? HDA_AMP_MUTE : 0);
+ present = snd_hda_jack_detect(codec, spec->autocfg.hp_pins[0]);
+
+ if (!spec->hp_independent_mode) {
+ struct snd_ctl_elem_id id;
+ /* auto mute */
+ snd_hda_codec_amp_stereo(
+ codec, spec->autocfg.line_out_pins[0], HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
+ /* notify change */
+ memset(&id, 0, sizeof(id));
+ id.iface = SNDRV_CTL_ELEM_IFACE_MIXER;
+ strcpy(id.name, "Front Playback Switch");
+ snd_ctl_notify(codec->bus->card, SNDRV_CTL_EVENT_MASK_VALUE,
+ &id);
+ }
+}
+
+/* mute mono out if HP or Line out is plugged */
+static void via_mono_automute(struct hda_codec *codec)
+{
+ unsigned int hp_present, lineout_present;
+ struct via_spec *spec = codec->spec;
+
+ if (spec->codec_type != VT1716S)
+ return;
+
+ lineout_present = snd_hda_jack_detect(codec,
+ spec->autocfg.line_out_pins[0]);
+
+ /* Mute Mono Out if Line Out is plugged */
+ if (lineout_present) {
+ snd_hda_codec_amp_stereo(
+ codec, 0x2A, HDA_OUTPUT, 0, HDA_AMP_MUTE, HDA_AMP_MUTE);
+ return;
+ }
+
+ hp_present = snd_hda_jack_detect(codec, spec->autocfg.hp_pins[0]);
+
+ if (!spec->hp_independent_mode)
+ snd_hda_codec_amp_stereo(
+ codec, 0x2A, HDA_OUTPUT, 0, HDA_AMP_MUTE,
+ hp_present ? HDA_AMP_MUTE : 0);
}
static void via_gpio_control(struct hda_codec *codec)
@@ -968,15 +2004,83 @@ static void via_gpio_control(struct hda_codec *codec)
}
}
+/* mute Internal-Speaker if HP is plugged */
+static void via_speaker_automute(struct hda_codec *codec)
+{
+ unsigned int hp_present;
+ struct via_spec *spec = codec->spec;
+
+ if (spec->codec_type != VT2002P && spec->codec_type != VT1812)
+ return;
+
+ hp_present = snd_hda_jack_detect(codec, spec->autocfg.hp_pins[0]);
+
+ if (!spec->hp_independent_mode) {
+ struct snd_ctl_elem_id id;
+ snd_hda_codec_amp_stereo(
+ codec, spec->autocfg.speaker_pins[0], HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, hp_present ? HDA_AMP_MUTE : 0);
+ /* notify change */
+ memset(&id, 0, sizeof(id));
+ id.iface = SNDRV_CTL_ELEM_IFACE_MIXER;
+ strcpy(id.name, "Speaker Playback Switch");
+ snd_ctl_notify(codec->bus->card, SNDRV_CTL_EVENT_MASK_VALUE,
+ &id);
+ }
+}
+
+/* mute line-out and internal speaker if HP is plugged */
+static void via_hp_bind_automute(struct hda_codec *codec)
+{
+ /* use long instead of int below just to avoid an internal compiler
+ * error with gcc 4.0.x
+ */
+ unsigned long hp_present, present = 0;
+ struct via_spec *spec = codec->spec;
+ int i;
+
+ if (!spec->autocfg.hp_pins[0] || !spec->autocfg.line_out_pins[0])
+ return;
+
+ hp_present = snd_hda_jack_detect(codec, spec->autocfg.hp_pins[0]);
+
+ present = snd_hda_jack_detect(codec, spec->autocfg.line_out_pins[0]);
+
+ if (!spec->hp_independent_mode) {
+ /* Mute Line-Outs */
+ for (i = 0; i < spec->autocfg.line_outs; i++)
+ snd_hda_codec_amp_stereo(
+ codec, spec->autocfg.line_out_pins[i],
+ HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, hp_present ? HDA_AMP_MUTE : 0);
+ if (hp_present)
+ present = hp_present;
+ }
+ /* Speakers */
+ for (i = 0; i < spec->autocfg.speaker_outs; i++)
+ snd_hda_codec_amp_stereo(
+ codec, spec->autocfg.speaker_pins[i], HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
+}
+
+
/* unsolicited event for jack sensing */
static void via_unsol_event(struct hda_codec *codec,
unsigned int res)
{
res >>= 26;
- if (res == VIA_HP_EVENT)
+ if (res & VIA_HP_EVENT)
via_hp_automute(codec);
- else if (res == VIA_GPIO_EVENT)
+ if (res & VIA_GPIO_EVENT)
via_gpio_control(codec);
+ if (res & VIA_JACK_EVENT)
+ set_jack_power_state(codec);
+ if (res & VIA_MONO_EVENT)
+ via_mono_automute(codec);
+ if (res & VIA_SPEAKER_EVENT)
+ via_speaker_automute(codec);
+ if (res & VIA_BIND_HP_EVENT)
+ via_hp_bind_automute(codec);
}
static int via_init(struct hda_codec *codec)
@@ -986,6 +2090,10 @@ static int via_init(struct hda_codec *codec)
for (i = 0; i < spec->num_iverbs; i++)
snd_hda_sequence_write(codec, spec->init_verbs[i]);
+ spec->codec_type = get_codec_type(codec);
+ if (spec->codec_type == VT1708BCE)
+ spec->codec_type = VT1708S; /* VT1708BCE & VT1708S are almost
+ same */
/* Lydia Add for EAPD enable */
if (!spec->dig_in_nid) { /* No Digital In connection */
if (spec->dig_in_pin) {
@@ -1003,8 +2111,17 @@ static int via_init(struct hda_codec *codec)
if (spec->slave_dig_outs[0])
codec->slave_dig_outs = spec->slave_dig_outs;
- return 0;
+ return 0;
+}
+
+#ifdef SND_HDA_NEEDS_RESUME
+static int via_suspend(struct hda_codec *codec, pm_message_t state)
+{
+ struct via_spec *spec = codec->spec;
+ vt1708_stop_hp_work(spec);
+ return 0;
}
+#endif
#ifdef CONFIG_SND_HDA_POWER_SAVE
static int via_check_power_status(struct hda_codec *codec, hda_nid_t nid)
@@ -1021,6 +2138,9 @@ static struct hda_codec_ops via_patch_ops = {
.build_pcms = via_build_pcms,
.init = via_init,
.free = via_free,
+#ifdef SND_HDA_NEEDS_RESUME
+ .suspend = via_suspend,
+#endif
#ifdef CONFIG_SND_HDA_POWER_SAVE
.check_power_status = via_check_power_status,
#endif
@@ -1036,8 +2156,8 @@ static int vt1708_auto_fill_dac_nids(struct via_spec *spec,
spec->multiout.num_dacs = cfg->line_outs;
spec->multiout.dac_nids = spec->private_dac_nids;
-
- for(i = 0; i < 4; i++) {
+
+ for (i = 0; i < 4; i++) {
nid = cfg->line_out_pins[i];
if (nid) {
/* config dac list */
@@ -1067,7 +2187,7 @@ static int vt1708_auto_create_multi_out_ctls(struct via_spec *spec,
{
char name[32];
static const char *chname[4] = { "Front", "Surround", "C/LFE", "Side" };
- hda_nid_t nid, nid_vol = 0;
+ hda_nid_t nid, nid_vol, nid_vols[] = {0x17, 0x19, 0x1a, 0x1b};
int i, err;
for (i = 0; i <= AUTO_SEQ_SIDE; i++) {
@@ -1075,9 +2195,8 @@ static int vt1708_auto_create_multi_out_ctls(struct via_spec *spec,
if (!nid)
continue;
-
- if (i != AUTO_SEQ_FRONT)
- nid_vol = 0x18 + i;
+
+ nid_vol = nid_vols[i];
if (i == AUTO_SEQ_CENLFE) {
/* Center/LFE */
@@ -1105,21 +2224,21 @@ static int vt1708_auto_create_multi_out_ctls(struct via_spec *spec,
HDA_OUTPUT));
if (err < 0)
return err;
- } else if (i == AUTO_SEQ_FRONT){
+ } else if (i == AUTO_SEQ_FRONT) {
/* add control to mixer index 0 */
err = via_add_control(spec, VIA_CTL_WIDGET_VOL,
"Master Front Playback Volume",
- HDA_COMPOSE_AMP_VAL(0x17, 3, 0,
+ HDA_COMPOSE_AMP_VAL(nid_vol, 3, 0,
HDA_INPUT));
if (err < 0)
return err;
err = via_add_control(spec, VIA_CTL_WIDGET_MUTE,
"Master Front Playback Switch",
- HDA_COMPOSE_AMP_VAL(0x17, 3, 0,
+ HDA_COMPOSE_AMP_VAL(nid_vol, 3, 0,
HDA_INPUT));
if (err < 0)
return err;
-
+
/* add control to PW3 */
sprintf(name, "%s Playback Volume", chname[i]);
err = via_add_control(spec, VIA_CTL_WIDGET_VOL, name,
@@ -1178,6 +2297,7 @@ static int vt1708_auto_create_hp_ctls(struct via_spec *spec, hda_nid_t pin)
return 0;
spec->multiout.hp_nid = VT1708_HP_NID; /* AOW3 */
+ spec->hp_independent_mode_index = 1;
err = via_add_control(spec, VIA_CTL_WIDGET_VOL,
"Headphone Playback Volume",
@@ -1218,7 +2338,7 @@ static int vt1708_auto_create_analog_input_ctls(struct via_spec *spec,
case 0x1d: /* Mic */
idx = 2;
break;
-
+
case 0x1e: /* Line In */
idx = 3;
break;
@@ -1231,8 +2351,7 @@ static int vt1708_auto_create_analog_input_ctls(struct via_spec *spec,
idx = 1;
break;
}
- err = via_new_analog_input(spec, cfg->input_pins[i], labels[i],
- idx, 0x17);
+ err = via_new_analog_input(spec, labels[i], idx, 0x17);
if (err < 0)
return err;
imux->items[imux->num_items].label = labels[i];
@@ -1260,16 +2379,60 @@ static void vt1708_set_pinconfig_connect(struct hda_codec *codec, hda_nid_t nid)
def_conf = snd_hda_codec_get_pincfg(codec, nid);
seqassoc = (unsigned char) get_defcfg_association(def_conf);
seqassoc = (seqassoc << 4) | get_defcfg_sequence(def_conf);
- if (get_defcfg_connect(def_conf) == AC_JACK_PORT_NONE) {
- if (seqassoc == 0xff) {
- def_conf = def_conf & (~(AC_JACK_PORT_BOTH << 30));
- snd_hda_codec_set_pincfg(codec, nid, def_conf);
- }
+ if (get_defcfg_connect(def_conf) == AC_JACK_PORT_NONE
+ && (seqassoc == 0xf0 || seqassoc == 0xff)) {
+ def_conf = def_conf & (~(AC_JACK_PORT_BOTH << 30));
+ snd_hda_codec_set_pincfg(codec, nid, def_conf);
}
return;
}
+static int vt1708_jack_detectect_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct via_spec *spec = codec->spec;
+
+ if (spec->codec_type != VT1708)
+ return 0;
+ spec->vt1708_jack_detectect =
+ !((snd_hda_codec_read(codec, 0x1, 0, 0xf84, 0) >> 8) & 0x1);
+ ucontrol->value.integer.value[0] = spec->vt1708_jack_detectect;
+ return 0;
+}
+
+static int vt1708_jack_detectect_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct via_spec *spec = codec->spec;
+ int change;
+
+ if (spec->codec_type != VT1708)
+ return 0;
+ spec->vt1708_jack_detectect = ucontrol->value.integer.value[0];
+ change = (0x1 & (snd_hda_codec_read(codec, 0x1, 0, 0xf84, 0) >> 8))
+ == !spec->vt1708_jack_detectect;
+ if (spec->vt1708_jack_detectect) {
+ mute_aa_path(codec, 1);
+ notify_aa_path_ctls(codec);
+ }
+ return change;
+}
+
+static struct snd_kcontrol_new vt1708_jack_detectect[] = {
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Jack Detect",
+ .count = 1,
+ .info = snd_ctl_boolean_mono_info,
+ .get = vt1708_jack_detectect_get,
+ .put = vt1708_jack_detectect_put,
+ },
+ {} /* end */
+};
+
static int vt1708_parse_auto_config(struct hda_codec *codec)
{
struct via_spec *spec = codec->spec;
@@ -1297,6 +2460,10 @@ static int vt1708_parse_auto_config(struct hda_codec *codec)
err = vt1708_auto_create_analog_input_ctls(spec, &spec->autocfg);
if (err < 0)
return err;
+ /* add jack detect on/off control */
+ err = snd_hda_add_new_ctls(codec, vt1708_jack_detectect);
+ if (err < 0)
+ return err;
spec->multiout.max_channels = spec->multiout.num_dacs * 2;
@@ -1316,19 +2483,44 @@ static int vt1708_parse_auto_config(struct hda_codec *codec)
if (spec->hp_mux)
spec->mixers[spec->num_mixers++] = via_hp_mixer;
+ spec->mixers[spec->num_mixers++] = via_smart51_mixer;
return 1;
}
/* init callback for auto-configuration model -- overriding the default init */
static int via_auto_init(struct hda_codec *codec)
{
+ struct via_spec *spec = codec->spec;
+
via_init(codec);
via_auto_init_multi_out(codec);
via_auto_init_hp_out(codec);
via_auto_init_analog_input(codec);
+ if (spec->codec_type == VT2002P || spec->codec_type == VT1812) {
+ via_hp_bind_automute(codec);
+ } else {
+ via_hp_automute(codec);
+ via_speaker_automute(codec);
+ }
+
return 0;
}
+static void vt1708_update_hp_jack_state(struct work_struct *work)
+{
+ struct via_spec *spec = container_of(work, struct via_spec,
+ vt1708_hp_work.work);
+ if (spec->codec_type != VT1708)
+ return;
+ /* if jack state toggled */
+ if (spec->vt1708_hp_present
+ != snd_hda_jack_detect(spec->codec, spec->autocfg.hp_pins[0])) {
+ spec->vt1708_hp_present ^= 1;
+ via_hp_automute(spec->codec);
+ }
+ vt1708_start_hp_work(spec);
+}
+
static int get_mux_nids(struct hda_codec *codec)
{
struct via_spec *spec = codec->spec;
@@ -1378,7 +2570,7 @@ static int patch_vt1708(struct hda_codec *codec)
"from BIOS. Using genenic mode...\n");
}
-
+
spec->stream_name_analog = "VT1708 Analog";
spec->stream_analog_playback = &vt1708_pcm_analog_playback;
/* disable 32bit format on VT1708 */
@@ -1390,7 +2582,7 @@ static int patch_vt1708(struct hda_codec *codec)
spec->stream_digital_playback = &vt1708_pcm_digital_playback;
spec->stream_digital_capture = &vt1708_pcm_digital_capture;
-
+
if (!spec->adc_nids && spec->input_mux) {
spec->adc_nids = vt1708_adc_nids;
spec->num_adc_nids = ARRAY_SIZE(vt1708_adc_nids);
@@ -1405,7 +2597,8 @@ static int patch_vt1708(struct hda_codec *codec)
#ifdef CONFIG_SND_HDA_POWER_SAVE
spec->loopback.amplist = vt1708_loopbacks;
#endif
-
+ spec->codec = codec;
+ INIT_DELAYED_WORK(&spec->vt1708_hp_work, vt1708_update_hp_jack_state);
return 0;
}
@@ -1433,7 +2626,8 @@ static struct snd_kcontrol_new vt1709_capture_mixer[] = {
};
static struct hda_verb vt1709_uniwill_init_verbs[] = {
- {0x20, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_HP_EVENT},
+ {0x20, AC_VERB_SET_UNSOLICITED_ENABLE,
+ AC_USRSP_EN | VIA_HP_EVENT | VIA_JACK_EVENT},
{ }
};
@@ -1473,8 +2667,8 @@ static struct hda_verb vt1709_10ch_volume_init_verbs[] = {
{0x1f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- /* Set input of PW4 as AOW4 */
- {0x20, AC_VERB_SET_CONNECT_SEL, 0x1},
+ /* Set input of PW4 as MW0 */
+ {0x20, AC_VERB_SET_CONNECT_SEL, 0},
/* PW9 Output enable */
{0x24, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
{ }
@@ -1487,8 +2681,8 @@ static struct hda_pcm_stream vt1709_10ch_pcm_analog_playback = {
.nid = 0x10, /* NID to query formats and rates */
.ops = {
.open = via_playback_pcm_open,
- .prepare = via_playback_pcm_prepare,
- .cleanup = via_playback_pcm_cleanup
+ .prepare = via_playback_multi_pcm_prepare,
+ .cleanup = via_playback_multi_pcm_cleanup,
},
};
@@ -1499,8 +2693,8 @@ static struct hda_pcm_stream vt1709_6ch_pcm_analog_playback = {
.nid = 0x10, /* NID to query formats and rates */
.ops = {
.open = via_playback_pcm_open,
- .prepare = via_playback_pcm_prepare,
- .cleanup = via_playback_pcm_cleanup
+ .prepare = via_playback_multi_pcm_prepare,
+ .cleanup = via_playback_multi_pcm_cleanup,
},
};
@@ -1575,11 +2769,11 @@ static int vt1709_auto_fill_dac_nids(struct via_spec *spec,
spec->multiout.dac_nids[cfg->line_outs] = 0x28; /* AOW4 */
} else if (cfg->line_outs == 3) { /* 6 channels */
- for(i = 0; i < cfg->line_outs; i++) {
+ for (i = 0; i < cfg->line_outs; i++) {
nid = cfg->line_out_pins[i];
if (nid) {
/* config dac list */
- switch(i) {
+ switch (i) {
case AUTO_SEQ_FRONT:
/* AOW0 */
spec->multiout.dac_nids[i] = 0x10;
@@ -1608,56 +2802,58 @@ static int vt1709_auto_create_multi_out_ctls(struct via_spec *spec,
{
char name[32];
static const char *chname[4] = { "Front", "Surround", "C/LFE", "Side" };
- hda_nid_t nid = 0;
+ hda_nid_t nid, nid_vol, nid_vols[] = {0x18, 0x1a, 0x1b, 0x29};
int i, err;
for (i = 0; i <= AUTO_SEQ_SIDE; i++) {
nid = cfg->line_out_pins[i];
- if (!nid)
+ if (!nid)
continue;
+ nid_vol = nid_vols[i];
+
if (i == AUTO_SEQ_CENLFE) {
/* Center/LFE */
err = via_add_control(spec, VIA_CTL_WIDGET_VOL,
"Center Playback Volume",
- HDA_COMPOSE_AMP_VAL(0x1b, 1, 0,
+ HDA_COMPOSE_AMP_VAL(nid_vol, 1, 0,
HDA_OUTPUT));
if (err < 0)
return err;
err = via_add_control(spec, VIA_CTL_WIDGET_VOL,
"LFE Playback Volume",
- HDA_COMPOSE_AMP_VAL(0x1b, 2, 0,
+ HDA_COMPOSE_AMP_VAL(nid_vol, 2, 0,
HDA_OUTPUT));
if (err < 0)
return err;
err = via_add_control(spec, VIA_CTL_WIDGET_MUTE,
"Center Playback Switch",
- HDA_COMPOSE_AMP_VAL(0x1b, 1, 0,
+ HDA_COMPOSE_AMP_VAL(nid_vol, 1, 0,
HDA_OUTPUT));
if (err < 0)
return err;
err = via_add_control(spec, VIA_CTL_WIDGET_MUTE,
"LFE Playback Switch",
- HDA_COMPOSE_AMP_VAL(0x1b, 2, 0,
+ HDA_COMPOSE_AMP_VAL(nid_vol, 2, 0,
HDA_OUTPUT));
if (err < 0)
return err;
- } else if (i == AUTO_SEQ_FRONT){
- /* add control to mixer index 0 */
+ } else if (i == AUTO_SEQ_FRONT) {
+ /* ADD control to mixer index 0 */
err = via_add_control(spec, VIA_CTL_WIDGET_VOL,
"Master Front Playback Volume",
- HDA_COMPOSE_AMP_VAL(0x18, 3, 0,
+ HDA_COMPOSE_AMP_VAL(nid_vol, 3, 0,
HDA_INPUT));
if (err < 0)
return err;
err = via_add_control(spec, VIA_CTL_WIDGET_MUTE,
"Master Front Playback Switch",
- HDA_COMPOSE_AMP_VAL(0x18, 3, 0,
+ HDA_COMPOSE_AMP_VAL(nid_vol, 3, 0,
HDA_INPUT));
if (err < 0)
return err;
-
+
/* add control to PW3 */
sprintf(name, "%s Playback Volume", chname[i]);
err = via_add_control(spec, VIA_CTL_WIDGET_VOL, name,
@@ -1674,26 +2870,26 @@ static int vt1709_auto_create_multi_out_ctls(struct via_spec *spec,
} else if (i == AUTO_SEQ_SURROUND) {
sprintf(name, "%s Playback Volume", chname[i]);
err = via_add_control(spec, VIA_CTL_WIDGET_VOL, name,
- HDA_COMPOSE_AMP_VAL(0x1a, 3, 0,
+ HDA_COMPOSE_AMP_VAL(nid_vol, 3, 0,
HDA_OUTPUT));
if (err < 0)
return err;
sprintf(name, "%s Playback Switch", chname[i]);
err = via_add_control(spec, VIA_CTL_WIDGET_MUTE, name,
- HDA_COMPOSE_AMP_VAL(0x1a, 3, 0,
+ HDA_COMPOSE_AMP_VAL(nid_vol, 3, 0,
HDA_OUTPUT));
if (err < 0)
return err;
} else if (i == AUTO_SEQ_SIDE) {
sprintf(name, "%s Playback Volume", chname[i]);
err = via_add_control(spec, VIA_CTL_WIDGET_VOL, name,
- HDA_COMPOSE_AMP_VAL(0x29, 3, 0,
+ HDA_COMPOSE_AMP_VAL(nid_vol, 3, 0,
HDA_OUTPUT));
if (err < 0)
return err;
sprintf(name, "%s Playback Switch", chname[i]);
err = via_add_control(spec, VIA_CTL_WIDGET_MUTE, name,
- HDA_COMPOSE_AMP_VAL(0x29, 3, 0,
+ HDA_COMPOSE_AMP_VAL(nid_vol, 3, 0,
HDA_OUTPUT));
if (err < 0)
return err;
@@ -1714,6 +2910,7 @@ static int vt1709_auto_create_hp_ctls(struct via_spec *spec, hda_nid_t pin)
spec->multiout.hp_nid = VT1709_HP_DAC_NID;
else if (spec->multiout.num_dacs == 3) /* 6 channels */
spec->multiout.hp_nid = 0;
+ spec->hp_independent_mode_index = 1;
err = via_add_control(spec, VIA_CTL_WIDGET_VOL,
"Headphone Playback Volume",
@@ -1752,7 +2949,7 @@ static int vt1709_auto_create_analog_input_ctls(struct via_spec *spec,
case 0x1d: /* Mic */
idx = 2;
break;
-
+
case 0x1e: /* Line In */
idx = 3;
break;
@@ -1765,8 +2962,7 @@ static int vt1709_auto_create_analog_input_ctls(struct via_spec *spec,
idx = 1;
break;
}
- err = via_new_analog_input(spec, cfg->input_pins[i], labels[i],
- idx, 0x18);
+ err = via_new_analog_input(spec, labels[i], idx, 0x18);
if (err < 0)
return err;
imux->items[imux->num_items].label = labels[i];
@@ -1816,6 +3012,7 @@ static int vt1709_parse_auto_config(struct hda_codec *codec)
if (spec->hp_mux)
spec->mixers[spec->num_mixers++] = via_hp_mixer;
+ spec->mixers[spec->num_mixers++] = via_smart51_mixer;
return 1;
}
@@ -1861,7 +3058,7 @@ static int patch_vt1709_10ch(struct hda_codec *codec)
spec->stream_digital_playback = &vt1709_pcm_digital_playback;
spec->stream_digital_capture = &vt1709_pcm_digital_capture;
-
+
if (!spec->adc_nids && spec->input_mux) {
spec->adc_nids = vt1709_adc_nids;
spec->num_adc_nids = ARRAY_SIZE(vt1709_adc_nids);
@@ -1955,7 +3152,7 @@ static int patch_vt1709_6ch(struct hda_codec *codec)
spec->stream_digital_playback = &vt1709_pcm_digital_playback;
spec->stream_digital_capture = &vt1709_pcm_digital_capture;
-
+
if (!spec->adc_nids && spec->input_mux) {
spec->adc_nids = vt1709_adc_nids;
spec->num_adc_nids = ARRAY_SIZE(vt1709_adc_nids);
@@ -2024,7 +3221,7 @@ static struct hda_verb vt1708B_8ch_volume_init_verbs[] = {
{0x27, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
/* Setup default input to PW4 */
- {0x1d, AC_VERB_SET_CONNECT_SEL, 0x1},
+ {0x1d, AC_VERB_SET_CONNECT_SEL, 0},
/* PW9 Output enable */
{0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
/* PW10 Input enable */
@@ -2068,10 +3265,29 @@ static struct hda_verb vt1708B_4ch_volume_init_verbs[] = {
};
static struct hda_verb vt1708B_uniwill_init_verbs[] = {
- {0x1D, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_HP_EVENT},
+ {0x1d, AC_VERB_SET_UNSOLICITED_ENABLE,
+ AC_USRSP_EN | VIA_HP_EVENT | VIA_JACK_EVENT},
+ {0x19, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT},
+ {0x1a, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT},
+ {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT},
+ {0x1c, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT},
+ {0x1e, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT},
+ {0x22, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT},
+ {0x23, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT},
{ }
};
+static int via_pcm_open_close(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ struct snd_pcm_substream *substream)
+{
+ int idle = substream->pstr->substream_opened == 1
+ && substream->ref_count == 0;
+
+ analog_low_current_mode(codec, idle);
+ return 0;
+}
+
static struct hda_pcm_stream vt1708B_8ch_pcm_analog_playback = {
.substreams = 2,
.channels_min = 2,
@@ -2080,7 +3296,8 @@ static struct hda_pcm_stream vt1708B_8ch_pcm_analog_playback = {
.ops = {
.open = via_playback_pcm_open,
.prepare = via_playback_multi_pcm_prepare,
- .cleanup = via_playback_multi_pcm_cleanup
+ .cleanup = via_playback_multi_pcm_cleanup,
+ .close = via_pcm_open_close
},
};
@@ -2102,8 +3319,10 @@ static struct hda_pcm_stream vt1708B_pcm_analog_capture = {
.channels_max = 2,
.nid = 0x13, /* NID to query formats and rates */
.ops = {
+ .open = via_pcm_open_close,
.prepare = via_capture_pcm_prepare,
- .cleanup = via_capture_pcm_cleanup
+ .cleanup = via_capture_pcm_cleanup,
+ .close = via_pcm_open_close
},
};
@@ -2260,6 +3479,7 @@ static int vt1708B_auto_create_hp_ctls(struct via_spec *spec, hda_nid_t pin)
return 0;
spec->multiout.hp_nid = VT1708B_HP_NID; /* AOW3 */
+ spec->hp_independent_mode_index = 1;
err = via_add_control(spec, VIA_CTL_WIDGET_VOL,
"Headphone Playback Volume",
@@ -2313,8 +3533,7 @@ static int vt1708B_auto_create_analog_input_ctls(struct via_spec *spec,
idx = 1;
break;
}
- err = via_new_analog_input(spec, cfg->input_pins[i], labels[i],
- idx, 0x16);
+ err = via_new_analog_input(spec, labels[i], idx, 0x16);
if (err < 0)
return err;
imux->items[imux->num_items].label = labels[i];
@@ -2364,6 +3583,7 @@ static int vt1708B_parse_auto_config(struct hda_codec *codec)
if (spec->hp_mux)
spec->mixers[spec->num_mixers++] = via_hp_mixer;
+ spec->mixers[spec->num_mixers++] = via_smart51_mixer;
return 1;
}
@@ -2376,12 +3596,14 @@ static struct hda_amp_list vt1708B_loopbacks[] = {
{ } /* end */
};
#endif
-
+static int patch_vt1708S(struct hda_codec *codec);
static int patch_vt1708B_8ch(struct hda_codec *codec)
{
struct via_spec *spec;
int err;
+ if (get_codec_type(codec) == VT1708BCE)
+ return patch_vt1708S(codec);
/* create a codec specific record */
spec = kzalloc(sizeof(*spec), GFP_KERNEL);
if (spec == NULL)
@@ -2483,29 +3705,15 @@ static int patch_vt1708B_4ch(struct hda_codec *codec)
/* Patch for VT1708S */
-/* VT1708S software backdoor based override for buggy hardware micboost
- * setting */
-#define MIC_BOOST_VOLUME(xname, nid) { \
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
- .name = xname, \
- .index = 0, \
- .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | \
- SNDRV_CTL_ELEM_ACCESS_TLV_READ | \
- SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK, \
- .info = mic_boost_volume_info, \
- .get = snd_hda_mixer_amp_volume_get, \
- .put = snd_hda_mixer_amp_volume_put, \
- .tlv = { .c = mic_boost_tlv }, \
- .private_value = HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_INPUT) }
-
/* capture mixer elements */
static struct snd_kcontrol_new vt1708S_capture_mixer[] = {
HDA_CODEC_VOLUME("Capture Volume", 0x13, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Capture Switch", 0x13, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x14, 0x0, HDA_INPUT),
HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x14, 0x0, HDA_INPUT),
- MIC_BOOST_VOLUME("Mic Boost Capture Volume", 0x1A),
- MIC_BOOST_VOLUME("Front Mic Boost Capture Volume", 0x1E),
+ HDA_CODEC_VOLUME("Mic Boost Capture Volume", 0x1A, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Front Mic Boost Capture Volume", 0x1E, 0x0,
+ HDA_INPUT),
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
/* The multiple "Capture Source" controls confuse alsamixer
@@ -2542,11 +3750,21 @@ static struct hda_verb vt1708S_volume_init_verbs[] = {
{0x21, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
/* Enable Mic Boost Volume backdoor */
{0x1, 0xf98, 0x1},
+ /* don't bybass mixer */
+ {0x1, 0xf88, 0xc0},
{ }
};
static struct hda_verb vt1708S_uniwill_init_verbs[] = {
- {0x1D, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_HP_EVENT},
+ {0x1d, AC_VERB_SET_UNSOLICITED_ENABLE,
+ AC_USRSP_EN | VIA_HP_EVENT | VIA_JACK_EVENT},
+ {0x19, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT},
+ {0x1a, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT},
+ {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT},
+ {0x1c, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT},
+ {0x1e, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT},
+ {0x22, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT},
+ {0x23, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT},
{ }
};
@@ -2557,8 +3775,9 @@ static struct hda_pcm_stream vt1708S_pcm_analog_playback = {
.nid = 0x10, /* NID to query formats and rates */
.ops = {
.open = via_playback_pcm_open,
- .prepare = via_playback_pcm_prepare,
- .cleanup = via_playback_pcm_cleanup
+ .prepare = via_playback_multi_pcm_prepare,
+ .cleanup = via_playback_multi_pcm_cleanup,
+ .close = via_pcm_open_close
},
};
@@ -2568,8 +3787,10 @@ static struct hda_pcm_stream vt1708S_pcm_analog_capture = {
.channels_max = 2,
.nid = 0x13, /* NID to query formats and rates */
.ops = {
+ .open = via_pcm_open_close,
.prepare = via_capture_pcm_prepare,
- .cleanup = via_capture_pcm_cleanup
+ .cleanup = via_capture_pcm_cleanup,
+ .close = via_pcm_open_close
},
};
@@ -2726,6 +3947,7 @@ static int vt1708S_auto_create_hp_ctls(struct via_spec *spec, hda_nid_t pin)
return 0;
spec->multiout.hp_nid = VT1708S_HP_NID; /* AOW3 */
+ spec->hp_independent_mode_index = 1;
err = via_add_control(spec, VIA_CTL_WIDGET_VOL,
"Headphone Playback Volume",
@@ -2780,8 +4002,7 @@ static int vt1708S_auto_create_analog_input_ctls(struct via_spec *spec,
idx = 1;
break;
}
- err = via_new_analog_input(spec, cfg->input_pins[i], labels[i],
- idx, 0x16);
+ err = via_new_analog_input(spec, labels[i], idx, 0x16);
if (err < 0)
return err;
imux->items[imux->num_items].label = labels[i];
@@ -2852,6 +4073,7 @@ static int vt1708S_parse_auto_config(struct hda_codec *codec)
if (spec->hp_mux)
spec->mixers[spec->num_mixers++] = via_hp_mixer;
+ spec->mixers[spec->num_mixers++] = via_smart51_mixer;
return 1;
}
@@ -2865,6 +4087,16 @@ static struct hda_amp_list vt1708S_loopbacks[] = {
};
#endif
+static void override_mic_boost(struct hda_codec *codec, hda_nid_t pin,
+ int offset, int num_steps, int step_size)
+{
+ snd_hda_override_amp_caps(codec, pin, HDA_INPUT,
+ (offset << AC_AMPCAP_OFFSET_SHIFT) |
+ (num_steps << AC_AMPCAP_NUM_STEPS_SHIFT) |
+ (step_size << AC_AMPCAP_STEP_SIZE_SHIFT) |
+ (0 << AC_AMPCAP_MUTE_SHIFT));
+}
+
static int patch_vt1708S(struct hda_codec *codec)
{
struct via_spec *spec;
@@ -2890,17 +4122,25 @@ static int patch_vt1708S(struct hda_codec *codec)
spec->init_verbs[spec->num_iverbs++] = vt1708S_volume_init_verbs;
spec->init_verbs[spec->num_iverbs++] = vt1708S_uniwill_init_verbs;
- spec->stream_name_analog = "VT1708S Analog";
+ if (codec->vendor_id == 0x11060440)
+ spec->stream_name_analog = "VT1818S Analog";
+ else
+ spec->stream_name_analog = "VT1708S Analog";
spec->stream_analog_playback = &vt1708S_pcm_analog_playback;
spec->stream_analog_capture = &vt1708S_pcm_analog_capture;
- spec->stream_name_digital = "VT1708S Digital";
+ if (codec->vendor_id == 0x11060440)
+ spec->stream_name_digital = "VT1818S Digital";
+ else
+ spec->stream_name_digital = "VT1708S Digital";
spec->stream_digital_playback = &vt1708S_pcm_digital_playback;
if (!spec->adc_nids && spec->input_mux) {
spec->adc_nids = vt1708S_adc_nids;
spec->num_adc_nids = ARRAY_SIZE(vt1708S_adc_nids);
get_mux_nids(codec);
+ override_mic_boost(codec, 0x1a, 0, 3, 40);
+ override_mic_boost(codec, 0x1e, 0, 3, 40);
spec->mixers[spec->num_mixers] = vt1708S_capture_mixer;
spec->num_mixers++;
}
@@ -2913,6 +4153,16 @@ static int patch_vt1708S(struct hda_codec *codec)
spec->loopback.amplist = vt1708S_loopbacks;
#endif
+ /* correct names for VT1708BCE */
+ if (get_codec_type(codec) == VT1708BCE) {
+ kfree(codec->chip_name);
+ codec->chip_name = kstrdup("VT1708BCE", GFP_KERNEL);
+ snprintf(codec->bus->card->mixername,
+ sizeof(codec->bus->card->mixername),
+ "%s %s", codec->vendor_name, codec->chip_name);
+ spec->stream_name_analog = "VT1708BCE Analog";
+ spec->stream_name_digital = "VT1708BCE Digital";
+ }
return 0;
}
@@ -2967,12 +4217,20 @@ static struct hda_verb vt1702_volume_init_verbs[] = {
/* PW6 PW7 Output enable */
{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
{0x1C, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
+ /* mixer enable */
+ {0x1, 0xF88, 0x3},
+ /* GPIO 0~2 */
+ {0x1, 0xF82, 0x3F},
{ }
};
static struct hda_verb vt1702_uniwill_init_verbs[] = {
- {0x01, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_GPIO_EVENT},
- {0x17, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_HP_EVENT},
+ {0x17, AC_VERB_SET_UNSOLICITED_ENABLE,
+ AC_USRSP_EN | VIA_HP_EVENT | VIA_JACK_EVENT},
+ {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT},
+ {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT},
+ {0x16, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT},
+ {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT},
{ }
};
@@ -2984,7 +4242,8 @@ static struct hda_pcm_stream vt1702_pcm_analog_playback = {
.ops = {
.open = via_playback_pcm_open,
.prepare = via_playback_multi_pcm_prepare,
- .cleanup = via_playback_multi_pcm_cleanup
+ .cleanup = via_playback_multi_pcm_cleanup,
+ .close = via_pcm_open_close
},
};
@@ -2994,8 +4253,10 @@ static struct hda_pcm_stream vt1702_pcm_analog_capture = {
.channels_max = 2,
.nid = 0x12, /* NID to query formats and rates */
.ops = {
+ .open = via_pcm_open_close,
.prepare = via_capture_pcm_prepare,
- .cleanup = via_capture_pcm_cleanup
+ .cleanup = via_capture_pcm_cleanup,
+ .close = via_pcm_open_close
},
};
@@ -3065,12 +4326,13 @@ static int vt1702_auto_create_line_out_ctls(struct via_spec *spec,
static int vt1702_auto_create_hp_ctls(struct via_spec *spec, hda_nid_t pin)
{
- int err;
-
+ int err, i;
+ struct hda_input_mux *imux;
+ static const char *texts[] = { "ON", "OFF", NULL};
if (!pin)
return 0;
-
spec->multiout.hp_nid = 0x1D;
+ spec->hp_independent_mode_index = 0;
err = via_add_control(spec, VIA_CTL_WIDGET_VOL,
"Headphone Playback Volume",
@@ -3084,8 +4346,18 @@ static int vt1702_auto_create_hp_ctls(struct via_spec *spec, hda_nid_t pin)
if (err < 0)
return err;
- create_hp_imux(spec);
+ imux = &spec->private_imux[1];
+ /* for hp mode select */
+ i = 0;
+ while (texts[i] != NULL) {
+ imux->items[imux->num_items].label = texts[i];
+ imux->items[imux->num_items].index = i;
+ imux->num_items++;
+ i++;
+ }
+
+ spec->hp_mux = &spec->private_imux[1];
return 0;
}
@@ -3121,8 +4393,7 @@ static int vt1702_auto_create_analog_input_ctls(struct via_spec *spec,
idx = 3;
break;
}
- err = via_new_analog_input(spec, cfg->input_pins[i],
- labels[i], idx, 0x1A);
+ err = via_new_analog_input(spec, labels[i], idx, 0x1A);
if (err < 0)
return err;
imux->items[imux->num_items].label = labels[i];
@@ -3152,6 +4423,12 @@ static int vt1702_parse_auto_config(struct hda_codec *codec)
err = vt1702_auto_create_hp_ctls(spec, spec->autocfg.hp_pins[0]);
if (err < 0)
return err;
+ /* limit AA path volume to 0 dB */
+ snd_hda_override_amp_caps(codec, 0x1A, HDA_INPUT,
+ (0x17 << AC_AMPCAP_OFFSET_SHIFT) |
+ (0x17 << AC_AMPCAP_NUM_STEPS_SHIFT) |
+ (0x5 << AC_AMPCAP_STEP_SIZE_SHIFT) |
+ (1 << AC_AMPCAP_MUTE_SHIFT));
err = vt1702_auto_create_analog_input_ctls(spec, &spec->autocfg);
if (err < 0)
return err;
@@ -3185,8 +4462,6 @@ static int patch_vt1702(struct hda_codec *codec)
{
struct via_spec *spec;
int err;
- unsigned int response;
- unsigned char control;
/* create a codec specific record */
spec = kzalloc(sizeof(*spec), GFP_KERNEL);
@@ -3231,17 +4506,1638 @@ static int patch_vt1702(struct hda_codec *codec)
spec->loopback.amplist = vt1702_loopbacks;
#endif
- /* Open backdoor */
- response = snd_hda_codec_read(codec, codec->afg, 0, 0xF8C, 0);
- control = (unsigned char)(response & 0xff);
- control |= 0x3;
- snd_hda_codec_write(codec, codec->afg, 0, 0xF88, control);
+ return 0;
+}
+
+/* Patch for VT1718S */
+
+/* capture mixer elements */
+static struct snd_kcontrol_new vt1718S_capture_mixer[] = {
+ HDA_CODEC_VOLUME("Capture Volume", 0x10, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Capture Switch", 0x10, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x11, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x11, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Boost Capture Volume", 0x2b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Front Mic Boost Capture Volume", 0x29, 0x0,
+ HDA_INPUT),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ /* The multiple "Capture Source" controls confuse alsamixer
+ * So call somewhat different..
+ */
+ .name = "Input Source",
+ .count = 2,
+ .info = via_mux_enum_info,
+ .get = via_mux_enum_get,
+ .put = via_mux_enum_put,
+ },
+ { } /* end */
+};
+
+static struct hda_verb vt1718S_volume_init_verbs[] = {
+ /*
+ * Unmute ADC0-1 and set the default input to mic-in
+ */
+ {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+
+
+ /* 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 */
+ {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+ {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+ {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)},
+
+ /* Setup default input of Front HP to MW9 */
+ {0x28, AC_VERB_SET_CONNECT_SEL, 0x1},
+ /* PW9 PW10 Output enable */
+ {0x2d, AC_VERB_SET_PIN_WIDGET_CONTROL, AC_PINCTL_OUT_EN},
+ {0x2e, AC_VERB_SET_PIN_WIDGET_CONTROL, AC_PINCTL_OUT_EN},
+ /* PW11 Input enable */
+ {0x2f, AC_VERB_SET_PIN_WIDGET_CONTROL, AC_PINCTL_IN_EN},
+ /* Enable Boost Volume backdoor */
+ {0x1, 0xf88, 0x8},
+ /* MW0/1/2/3/4: un-mute index 0 (AOWx), mute index 1 (MW9) */
+ {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ /* set MUX1 = 2 (AOW4), MUX2 = 1 (AOW3) */
+ {0x34, AC_VERB_SET_CONNECT_SEL, 0x2},
+ {0x35, AC_VERB_SET_CONNECT_SEL, 0x1},
+ /* Unmute MW4's index 0 */
+ {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ { }
+};
+
+
+static struct hda_verb vt1718S_uniwill_init_verbs[] = {
+ {0x28, AC_VERB_SET_UNSOLICITED_ENABLE,
+ AC_USRSP_EN | VIA_HP_EVENT | VIA_JACK_EVENT},
+ {0x24, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT},
+ {0x25, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT},
+ {0x26, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT},
+ {0x27, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT},
+ {0x29, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT},
+ {0x2a, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT},
+ {0x2b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT},
+ { }
+};
+
+static struct hda_pcm_stream vt1718S_pcm_analog_playback = {
+ .substreams = 2,
+ .channels_min = 2,
+ .channels_max = 10,
+ .nid = 0x8, /* NID to query formats and rates */
+ .ops = {
+ .open = via_playback_pcm_open,
+ .prepare = via_playback_multi_pcm_prepare,
+ .cleanup = via_playback_multi_pcm_cleanup,
+ .close = via_pcm_open_close,
+ },
+};
+
+static struct hda_pcm_stream vt1718S_pcm_analog_capture = {
+ .substreams = 2,
+ .channels_min = 2,
+ .channels_max = 2,
+ .nid = 0x10, /* NID to query formats and rates */
+ .ops = {
+ .open = via_pcm_open_close,
+ .prepare = via_capture_pcm_prepare,
+ .cleanup = via_capture_pcm_cleanup,
+ .close = via_pcm_open_close,
+ },
+};
+
+static struct hda_pcm_stream vt1718S_pcm_digital_playback = {
+ .substreams = 2,
+ .channels_min = 2,
+ .channels_max = 2,
+ /* NID is set in via_build_pcms */
+ .ops = {
+ .open = via_dig_playback_pcm_open,
+ .close = via_dig_playback_pcm_close,
+ .prepare = via_dig_playback_pcm_prepare,
+ .cleanup = via_dig_playback_pcm_cleanup
+ },
+};
+
+static struct hda_pcm_stream vt1718S_pcm_digital_capture = {
+ .substreams = 1,
+ .channels_min = 2,
+ .channels_max = 2,
+};
+
+/* fill in the dac_nids table from the parsed pin configuration */
+static int vt1718S_auto_fill_dac_nids(struct via_spec *spec,
+ const struct auto_pin_cfg *cfg)
+{
+ int i;
+ hda_nid_t nid;
+
+ spec->multiout.num_dacs = cfg->line_outs;
+
+ spec->multiout.dac_nids = spec->private_dac_nids;
+
+ for (i = 0; i < 4; i++) {
+ nid = cfg->line_out_pins[i];
+ if (nid) {
+ /* config dac list */
+ switch (i) {
+ case AUTO_SEQ_FRONT:
+ spec->multiout.dac_nids[i] = 0x8;
+ break;
+ case AUTO_SEQ_CENLFE:
+ spec->multiout.dac_nids[i] = 0xa;
+ break;
+ case AUTO_SEQ_SURROUND:
+ spec->multiout.dac_nids[i] = 0x9;
+ break;
+ case AUTO_SEQ_SIDE:
+ spec->multiout.dac_nids[i] = 0xb;
+ break;
+ }
+ }
+ }
+
+ return 0;
+}
+
+/* add playback controls from the parsed DAC table */
+static int vt1718S_auto_create_multi_out_ctls(struct via_spec *spec,
+ const struct auto_pin_cfg *cfg)
+{
+ char name[32];
+ static const char *chname[4] = { "Front", "Surround", "C/LFE", "Side" };
+ hda_nid_t nid_vols[] = {0x8, 0x9, 0xa, 0xb};
+ hda_nid_t nid_mutes[] = {0x24, 0x25, 0x26, 0x27};
+ hda_nid_t nid, nid_vol, nid_mute = 0;
+ int i, err;
+
+ for (i = 0; i <= AUTO_SEQ_SIDE; i++) {
+ nid = cfg->line_out_pins[i];
+
+ if (!nid)
+ continue;
+ nid_vol = nid_vols[i];
+ nid_mute = nid_mutes[i];
+
+ if (i == AUTO_SEQ_CENLFE) {
+ /* Center/LFE */
+ err = via_add_control(spec, VIA_CTL_WIDGET_VOL,
+ "Center Playback Volume",
+ HDA_COMPOSE_AMP_VAL(nid_vol, 1, 0,
+ HDA_OUTPUT));
+ if (err < 0)
+ return err;
+ err = via_add_control(spec, VIA_CTL_WIDGET_VOL,
+ "LFE Playback Volume",
+ HDA_COMPOSE_AMP_VAL(nid_vol, 2, 0,
+ HDA_OUTPUT));
+ if (err < 0)
+ return err;
+ err = via_add_control(
+ spec, VIA_CTL_WIDGET_MUTE,
+ "Center Playback Switch",
+ HDA_COMPOSE_AMP_VAL(nid_mute, 1, 0,
+ HDA_OUTPUT));
+ if (err < 0)
+ return err;
+ err = via_add_control(
+ spec, VIA_CTL_WIDGET_MUTE,
+ "LFE Playback Switch",
+ HDA_COMPOSE_AMP_VAL(nid_mute, 2, 0,
+ HDA_OUTPUT));
+ if (err < 0)
+ return err;
+ } else if (i == AUTO_SEQ_FRONT) {
+ /* Front */
+ sprintf(name, "%s Playback Volume", chname[i]);
+ err = via_add_control(
+ spec, VIA_CTL_WIDGET_VOL, name,
+ HDA_COMPOSE_AMP_VAL(nid_vol, 3, 0, HDA_OUTPUT));
+ if (err < 0)
+ return err;
+ sprintf(name, "%s Playback Switch", chname[i]);
+ err = via_add_control(
+ spec, VIA_CTL_WIDGET_MUTE, name,
+ HDA_COMPOSE_AMP_VAL(nid_mute, 3, 0,
+ HDA_OUTPUT));
+ if (err < 0)
+ return err;
+ } else {
+ sprintf(name, "%s Playback Volume", chname[i]);
+ err = via_add_control(
+ spec, VIA_CTL_WIDGET_VOL, name,
+ HDA_COMPOSE_AMP_VAL(nid_vol, 3, 0, HDA_OUTPUT));
+ if (err < 0)
+ return err;
+ sprintf(name, "%s Playback Switch", chname[i]);
+ err = via_add_control(
+ spec, VIA_CTL_WIDGET_MUTE, name,
+ HDA_COMPOSE_AMP_VAL(nid_mute, 3, 0,
+ HDA_OUTPUT));
+ if (err < 0)
+ return err;
+ }
+ }
+ return 0;
+}
+
+static int vt1718S_auto_create_hp_ctls(struct via_spec *spec, hda_nid_t pin)
+{
+ int err;
+
+ if (!pin)
+ return 0;
+
+ spec->multiout.hp_nid = 0xc; /* AOW4 */
+ spec->hp_independent_mode_index = 1;
+
+ err = via_add_control(spec, VIA_CTL_WIDGET_VOL,
+ "Headphone Playback Volume",
+ HDA_COMPOSE_AMP_VAL(0xc, 3, 0, HDA_OUTPUT));
+ if (err < 0)
+ return err;
+
+ err = via_add_control(spec, VIA_CTL_WIDGET_MUTE,
+ "Headphone Playback Switch",
+ HDA_COMPOSE_AMP_VAL(pin, 3, 0, HDA_OUTPUT));
+ if (err < 0)
+ return err;
+
+ create_hp_imux(spec);
+ return 0;
+}
+
+/* create playback/capture controls for input pins */
+static int vt1718S_auto_create_analog_input_ctls(struct via_spec *spec,
+ const struct auto_pin_cfg *cfg)
+{
+ static char *labels[] = {
+ "Mic", "Front Mic", "Line", "Front Line", "CD", "Aux", NULL
+ };
+ struct hda_input_mux *imux = &spec->private_imux[0];
+ int i, err, idx = 0;
+
+ /* for internal loopback recording select */
+ imux->items[imux->num_items].label = "Stereo Mixer";
+ imux->items[imux->num_items].index = 5;
+ imux->num_items++;
+
+ for (i = 0; i < AUTO_PIN_LAST; i++) {
+ if (!cfg->input_pins[i])
+ continue;
+
+ switch (cfg->input_pins[i]) {
+ case 0x2b: /* Mic */
+ idx = 1;
+ break;
+
+ case 0x2a: /* Line In */
+ idx = 2;
+ break;
+
+ case 0x29: /* Front Mic */
+ idx = 3;
+ break;
+
+ case 0x2c: /* CD */
+ idx = 0;
+ break;
+ }
+ err = via_new_analog_input(spec, labels[i], idx, 0x21);
+ if (err < 0)
+ return err;
+ imux->items[imux->num_items].label = labels[i];
+ imux->items[imux->num_items].index = idx;
+ imux->num_items++;
+ }
+ return 0;
+}
+
+static int vt1718S_parse_auto_config(struct hda_codec *codec)
+{
+ struct via_spec *spec = codec->spec;
+ int err;
+
+ err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, NULL);
+
+ if (err < 0)
+ return err;
+ err = vt1718S_auto_fill_dac_nids(spec, &spec->autocfg);
+ if (err < 0)
+ return err;
+ if (!spec->autocfg.line_outs && !spec->autocfg.hp_pins[0])
+ return 0; /* can't find valid BIOS pin config */
+
+ err = vt1718S_auto_create_multi_out_ctls(spec, &spec->autocfg);
+ if (err < 0)
+ return err;
+ err = vt1718S_auto_create_hp_ctls(spec, spec->autocfg.hp_pins[0]);
+ if (err < 0)
+ return err;
+ err = vt1718S_auto_create_analog_input_ctls(spec, &spec->autocfg);
+ if (err < 0)
+ return err;
+
+ spec->multiout.max_channels = spec->multiout.num_dacs * 2;
+
+ fill_dig_outs(codec);
+
+ if (spec->autocfg.dig_in_pin && codec->vendor_id == 0x11060428)
+ spec->dig_in_nid = 0x13;
+
+ if (spec->kctls.list)
+ spec->mixers[spec->num_mixers++] = spec->kctls.list;
+
+ spec->input_mux = &spec->private_imux[0];
+
+ if (spec->hp_mux)
+ spec->mixers[spec->num_mixers++] = via_hp_mixer;
+
+ spec->mixers[spec->num_mixers++] = via_smart51_mixer;
+
+ return 1;
+}
+
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+static struct hda_amp_list vt1718S_loopbacks[] = {
+ { 0x21, HDA_INPUT, 1 },
+ { 0x21, HDA_INPUT, 2 },
+ { 0x21, HDA_INPUT, 3 },
+ { 0x21, HDA_INPUT, 4 },
+ { } /* end */
+};
+#endif
+
+static int patch_vt1718S(struct hda_codec *codec)
+{
+ struct via_spec *spec;
+ int err;
- /* Enable GPIO 0&1 for volume&mute control */
- /* Enable GPIO 2 for DMIC-DATA */
- response = snd_hda_codec_read(codec, codec->afg, 0, 0xF84, 0);
- control = (unsigned char)((response >> 16) & 0x3f);
- snd_hda_codec_write(codec, codec->afg, 0, 0xF82, control);
+ /* create a codec specific record */
+ spec = kzalloc(sizeof(*spec), GFP_KERNEL);
+ if (spec == NULL)
+ return -ENOMEM;
+
+ codec->spec = spec;
+
+ /* automatic parse from the BIOS config */
+ err = vt1718S_parse_auto_config(codec);
+ if (err < 0) {
+ via_free(codec);
+ return err;
+ } else if (!err) {
+ printk(KERN_INFO "hda_codec: Cannot set up configuration "
+ "from BIOS. Using genenic mode...\n");
+ }
+
+ spec->init_verbs[spec->num_iverbs++] = vt1718S_volume_init_verbs;
+ spec->init_verbs[spec->num_iverbs++] = vt1718S_uniwill_init_verbs;
+
+ if (codec->vendor_id == 0x11060441)
+ spec->stream_name_analog = "VT2020 Analog";
+ else if (codec->vendor_id == 0x11064441)
+ spec->stream_name_analog = "VT1828S Analog";
+ else
+ spec->stream_name_analog = "VT1718S Analog";
+ spec->stream_analog_playback = &vt1718S_pcm_analog_playback;
+ spec->stream_analog_capture = &vt1718S_pcm_analog_capture;
+
+ if (codec->vendor_id == 0x11060441)
+ spec->stream_name_digital = "VT2020 Digital";
+ else if (codec->vendor_id == 0x11064441)
+ spec->stream_name_digital = "VT1828S Digital";
+ else
+ spec->stream_name_digital = "VT1718S Digital";
+ spec->stream_digital_playback = &vt1718S_pcm_digital_playback;
+ if (codec->vendor_id == 0x11060428 || codec->vendor_id == 0x11060441)
+ spec->stream_digital_capture = &vt1718S_pcm_digital_capture;
+
+ if (!spec->adc_nids && spec->input_mux) {
+ spec->adc_nids = vt1718S_adc_nids;
+ spec->num_adc_nids = ARRAY_SIZE(vt1718S_adc_nids);
+ get_mux_nids(codec);
+ override_mic_boost(codec, 0x2b, 0, 3, 40);
+ override_mic_boost(codec, 0x29, 0, 3, 40);
+ spec->mixers[spec->num_mixers] = vt1718S_capture_mixer;
+ spec->num_mixers++;
+ }
+
+ codec->patch_ops = via_patch_ops;
+
+ codec->patch_ops.init = via_auto_init;
+ codec->patch_ops.unsol_event = via_unsol_event;
+
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ spec->loopback.amplist = vt1718S_loopbacks;
+#endif
+
+ return 0;
+}
+
+/* Patch for VT1716S */
+
+static int vt1716s_dmic_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;
+}
+
+static int vt1716s_dmic_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ int index = 0;
+
+ index = snd_hda_codec_read(codec, 0x26, 0,
+ AC_VERB_GET_CONNECT_SEL, 0);
+ if (index != -1)
+ *ucontrol->value.integer.value = index;
+
+ return 0;
+}
+
+static int vt1716s_dmic_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct via_spec *spec = codec->spec;
+ int index = *ucontrol->value.integer.value;
+
+ snd_hda_codec_write(codec, 0x26, 0,
+ AC_VERB_SET_CONNECT_SEL, index);
+ spec->dmic_enabled = index;
+ set_jack_power_state(codec);
+
+ return 1;
+}
+
+/* capture mixer elements */
+static struct snd_kcontrol_new vt1716S_capture_mixer[] = {
+ HDA_CODEC_VOLUME("Capture Volume", 0x13, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Capture Switch", 0x13, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x14, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x14, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Boost Capture Volume", 0x1A, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Front Mic Boost Capture Volume", 0x1E, 0x0,
+ HDA_INPUT),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Input Source",
+ .count = 1,
+ .info = via_mux_enum_info,
+ .get = via_mux_enum_get,
+ .put = via_mux_enum_put,
+ },
+ { } /* end */
+};
+
+static struct snd_kcontrol_new vt1716s_dmic_mixer[] = {
+ HDA_CODEC_VOLUME("Digital Mic Capture Volume", 0x22, 0x0, HDA_INPUT),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Digital Mic Capture Switch",
+ .count = 1,
+ .info = vt1716s_dmic_info,
+ .get = vt1716s_dmic_get,
+ .put = vt1716s_dmic_put,
+ },
+ {} /* end */
+};
+
+
+/* mono-out mixer elements */
+static struct snd_kcontrol_new vt1716S_mono_out_mixer[] = {
+ HDA_CODEC_MUTE("Mono Playback Switch", 0x2a, 0x0, HDA_OUTPUT),
+ { } /* end */
+};
+
+static struct hda_verb vt1716S_volume_init_verbs[] = {
+ /*
+ * Unmute ADC0-1 and set the default input to mic-in
+ */
+ {0x13, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+
+
+ /* 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 */
+ {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+ {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+ {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
+
+ /* MUX Indices: Stereo Mixer = 5 */
+ {0x17, AC_VERB_SET_CONNECT_SEL, 0x5},
+
+ /* Setup default input of PW4 to MW0 */
+ {0x1d, AC_VERB_SET_CONNECT_SEL, 0x0},
+
+ /* Setup default input of SW1 as MW0 */
+ {0x18, AC_VERB_SET_CONNECT_SEL, 0x1},
+
+ /* Setup default input of SW4 as AOW0 */
+ {0x28, AC_VERB_SET_CONNECT_SEL, 0x1},
+
+ /* PW9 PW10 Output enable */
+ {0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
+ {0x21, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
+
+ /* Unmute SW1, PW12 */
+ {0x29, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x2a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ /* PW12 Output enable */
+ {0x2a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
+ /* Enable Boost Volume backdoor */
+ {0x1, 0xf8a, 0x80},
+ /* don't bybass mixer */
+ {0x1, 0xf88, 0xc0},
+ /* Enable mono output */
+ {0x1, 0xf90, 0x08},
+ { }
+};
+
+
+static struct hda_verb vt1716S_uniwill_init_verbs[] = {
+ {0x1d, AC_VERB_SET_UNSOLICITED_ENABLE,
+ AC_USRSP_EN | VIA_HP_EVENT | VIA_JACK_EVENT},
+ {0x19, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT},
+ {0x1a, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT},
+ {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT},
+ {0x1c, AC_VERB_SET_UNSOLICITED_ENABLE,
+ AC_USRSP_EN | VIA_MONO_EVENT | VIA_JACK_EVENT},
+ {0x1e, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT},
+ {0x23, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT},
+ { }
+};
+
+static struct hda_pcm_stream vt1716S_pcm_analog_playback = {
+ .substreams = 2,
+ .channels_min = 2,
+ .channels_max = 6,
+ .nid = 0x10, /* NID to query formats and rates */
+ .ops = {
+ .open = via_playback_pcm_open,
+ .prepare = via_playback_multi_pcm_prepare,
+ .cleanup = via_playback_multi_pcm_cleanup,
+ .close = via_pcm_open_close,
+ },
+};
+
+static struct hda_pcm_stream vt1716S_pcm_analog_capture = {
+ .substreams = 2,
+ .channels_min = 2,
+ .channels_max = 2,
+ .nid = 0x13, /* NID to query formats and rates */
+ .ops = {
+ .open = via_pcm_open_close,
+ .prepare = via_capture_pcm_prepare,
+ .cleanup = via_capture_pcm_cleanup,
+ .close = via_pcm_open_close,
+ },
+};
+
+static struct hda_pcm_stream vt1716S_pcm_digital_playback = {
+ .substreams = 2,
+ .channels_min = 2,
+ .channels_max = 2,
+ /* NID is set in via_build_pcms */
+ .ops = {
+ .open = via_dig_playback_pcm_open,
+ .close = via_dig_playback_pcm_close,
+ .prepare = via_dig_playback_pcm_prepare,
+ .cleanup = via_dig_playback_pcm_cleanup
+ },
+};
+
+/* fill in the dac_nids table from the parsed pin configuration */
+static int vt1716S_auto_fill_dac_nids(struct via_spec *spec,
+ const struct auto_pin_cfg *cfg)
+{ int i;
+ hda_nid_t nid;
+
+ spec->multiout.num_dacs = cfg->line_outs;
+
+ spec->multiout.dac_nids = spec->private_dac_nids;
+
+ for (i = 0; i < 3; i++) {
+ nid = cfg->line_out_pins[i];
+ if (nid) {
+ /* config dac list */
+ switch (i) {
+ case AUTO_SEQ_FRONT:
+ spec->multiout.dac_nids[i] = 0x10;
+ break;
+ case AUTO_SEQ_CENLFE:
+ spec->multiout.dac_nids[i] = 0x25;
+ break;
+ case AUTO_SEQ_SURROUND:
+ spec->multiout.dac_nids[i] = 0x11;
+ break;
+ }
+ }
+ }
+
+ return 0;
+}
+
+/* add playback controls from the parsed DAC table */
+static int vt1716S_auto_create_multi_out_ctls(struct via_spec *spec,
+ const struct auto_pin_cfg *cfg)
+{
+ char name[32];
+ static const char *chname[3] = { "Front", "Surround", "C/LFE" };
+ hda_nid_t nid_vols[] = {0x10, 0x11, 0x25};
+ hda_nid_t nid_mutes[] = {0x1C, 0x18, 0x27};
+ hda_nid_t nid, nid_vol, nid_mute;
+ int i, err;
+
+ for (i = 0; i <= AUTO_SEQ_CENLFE; i++) {
+ nid = cfg->line_out_pins[i];
+
+ if (!nid)
+ continue;
+
+ nid_vol = nid_vols[i];
+ nid_mute = nid_mutes[i];
+
+ if (i == AUTO_SEQ_CENLFE) {
+ err = via_add_control(
+ spec, VIA_CTL_WIDGET_VOL,
+ "Center Playback Volume",
+ HDA_COMPOSE_AMP_VAL(nid_vol, 1, 0, HDA_OUTPUT));
+ if (err < 0)
+ return err;
+ err = via_add_control(
+ spec, VIA_CTL_WIDGET_VOL,
+ "LFE Playback Volume",
+ HDA_COMPOSE_AMP_VAL(nid_vol, 2, 0, HDA_OUTPUT));
+ if (err < 0)
+ return err;
+ err = via_add_control(
+ spec, VIA_CTL_WIDGET_MUTE,
+ "Center Playback Switch",
+ HDA_COMPOSE_AMP_VAL(nid_mute, 1, 0,
+ HDA_OUTPUT));
+ if (err < 0)
+ return err;
+ err = via_add_control(
+ spec, VIA_CTL_WIDGET_MUTE,
+ "LFE Playback Switch",
+ HDA_COMPOSE_AMP_VAL(nid_mute, 2, 0,
+ HDA_OUTPUT));
+ if (err < 0)
+ return err;
+ } else if (i == AUTO_SEQ_FRONT) {
+
+ err = via_add_control(
+ spec, VIA_CTL_WIDGET_VOL,
+ "Master Front Playback Volume",
+ HDA_COMPOSE_AMP_VAL(0x16, 3, 0, HDA_INPUT));
+ if (err < 0)
+ return err;
+ err = via_add_control(
+ spec, VIA_CTL_WIDGET_MUTE,
+ "Master Front Playback Switch",
+ HDA_COMPOSE_AMP_VAL(0x16, 3, 0, HDA_INPUT));
+ if (err < 0)
+ return err;
+
+ sprintf(name, "%s Playback Volume", chname[i]);
+ err = via_add_control(
+ spec, VIA_CTL_WIDGET_VOL, name,
+ HDA_COMPOSE_AMP_VAL(nid_vol, 3, 0, HDA_OUTPUT));
+ if (err < 0)
+ return err;
+ sprintf(name, "%s Playback Switch", chname[i]);
+ err = via_add_control(
+ spec, VIA_CTL_WIDGET_MUTE, name,
+ HDA_COMPOSE_AMP_VAL(nid_mute, 3, 0,
+ HDA_OUTPUT));
+ if (err < 0)
+ return err;
+ } else {
+ sprintf(name, "%s Playback Volume", chname[i]);
+ err = via_add_control(
+ spec, VIA_CTL_WIDGET_VOL, name,
+ HDA_COMPOSE_AMP_VAL(nid_vol, 3, 0, HDA_OUTPUT));
+ if (err < 0)
+ return err;
+ sprintf(name, "%s Playback Switch", chname[i]);
+ err = via_add_control(
+ spec, VIA_CTL_WIDGET_MUTE, name,
+ HDA_COMPOSE_AMP_VAL(nid_mute, 3, 0,
+ HDA_OUTPUT));
+ if (err < 0)
+ return err;
+ }
+ }
+ return 0;
+}
+
+static int vt1716S_auto_create_hp_ctls(struct via_spec *spec, hda_nid_t pin)
+{
+ int err;
+
+ if (!pin)
+ return 0;
+
+ spec->multiout.hp_nid = 0x25; /* AOW3 */
+ spec->hp_independent_mode_index = 1;
+
+ err = via_add_control(spec, VIA_CTL_WIDGET_VOL,
+ "Headphone Playback Volume",
+ HDA_COMPOSE_AMP_VAL(0x25, 3, 0, HDA_OUTPUT));
+ if (err < 0)
+ return err;
+
+ err = via_add_control(spec, VIA_CTL_WIDGET_MUTE,
+ "Headphone Playback Switch",
+ HDA_COMPOSE_AMP_VAL(pin, 3, 0, HDA_OUTPUT));
+ if (err < 0)
+ return err;
+
+ create_hp_imux(spec);
+ return 0;
+}
+
+/* create playback/capture controls for input pins */
+static int vt1716S_auto_create_analog_input_ctls(struct via_spec *spec,
+ const struct auto_pin_cfg *cfg)
+{
+ static char *labels[] = {
+ "Mic", "Front Mic", "Line", "Front Line", "CD", "Aux", NULL
+ };
+ struct hda_input_mux *imux = &spec->private_imux[0];
+ int i, err, idx = 0;
+
+ /* for internal loopback recording select */
+ imux->items[imux->num_items].label = "Stereo Mixer";
+ imux->items[imux->num_items].index = 5;
+ imux->num_items++;
+
+ for (i = 0; i < AUTO_PIN_LAST; i++) {
+ if (!cfg->input_pins[i])
+ continue;
+
+ switch (cfg->input_pins[i]) {
+ case 0x1a: /* Mic */
+ idx = 2;
+ break;
+
+ case 0x1b: /* Line In */
+ idx = 3;
+ break;
+
+ case 0x1e: /* Front Mic */
+ idx = 4;
+ break;
+
+ case 0x1f: /* CD */
+ idx = 1;
+ break;
+ }
+ err = via_new_analog_input(spec, labels[i], idx, 0x16);
+ if (err < 0)
+ return err;
+ imux->items[imux->num_items].label = labels[i];
+ imux->items[imux->num_items].index = idx-1;
+ imux->num_items++;
+ }
+ return 0;
+}
+
+static int vt1716S_parse_auto_config(struct hda_codec *codec)
+{
+ struct via_spec *spec = codec->spec;
+ int err;
+
+ err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, NULL);
+ if (err < 0)
+ return err;
+ err = vt1716S_auto_fill_dac_nids(spec, &spec->autocfg);
+ if (err < 0)
+ return err;
+ if (!spec->autocfg.line_outs && !spec->autocfg.hp_pins[0])
+ return 0; /* can't find valid BIOS pin config */
+
+ err = vt1716S_auto_create_multi_out_ctls(spec, &spec->autocfg);
+ if (err < 0)
+ return err;
+ err = vt1716S_auto_create_hp_ctls(spec, spec->autocfg.hp_pins[0]);
+ if (err < 0)
+ return err;
+ err = vt1716S_auto_create_analog_input_ctls(spec, &spec->autocfg);
+ if (err < 0)
+ return err;
+
+ spec->multiout.max_channels = spec->multiout.num_dacs * 2;
+
+ fill_dig_outs(codec);
+
+ if (spec->kctls.list)
+ spec->mixers[spec->num_mixers++] = spec->kctls.list;
+
+ spec->input_mux = &spec->private_imux[0];
+
+ if (spec->hp_mux)
+ spec->mixers[spec->num_mixers++] = via_hp_mixer;
+
+ spec->mixers[spec->num_mixers++] = via_smart51_mixer;
+
+ return 1;
+}
+
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+static struct hda_amp_list vt1716S_loopbacks[] = {
+ { 0x16, HDA_INPUT, 1 },
+ { 0x16, HDA_INPUT, 2 },
+ { 0x16, HDA_INPUT, 3 },
+ { 0x16, HDA_INPUT, 4 },
+ { } /* end */
+};
+#endif
+
+static int patch_vt1716S(struct hda_codec *codec)
+{
+ struct via_spec *spec;
+ int err;
+
+ /* create a codec specific record */
+ spec = kzalloc(sizeof(*spec), GFP_KERNEL);
+ if (spec == NULL)
+ return -ENOMEM;
+
+ codec->spec = spec;
+
+ /* automatic parse from the BIOS config */
+ err = vt1716S_parse_auto_config(codec);
+ if (err < 0) {
+ via_free(codec);
+ return err;
+ } else if (!err) {
+ printk(KERN_INFO "hda_codec: Cannot set up configuration "
+ "from BIOS. Using genenic mode...\n");
+ }
+
+ spec->init_verbs[spec->num_iverbs++] = vt1716S_volume_init_verbs;
+ spec->init_verbs[spec->num_iverbs++] = vt1716S_uniwill_init_verbs;
+
+ spec->stream_name_analog = "VT1716S Analog";
+ spec->stream_analog_playback = &vt1716S_pcm_analog_playback;
+ spec->stream_analog_capture = &vt1716S_pcm_analog_capture;
+
+ spec->stream_name_digital = "VT1716S Digital";
+ spec->stream_digital_playback = &vt1716S_pcm_digital_playback;
+
+ if (!spec->adc_nids && spec->input_mux) {
+ spec->adc_nids = vt1716S_adc_nids;
+ spec->num_adc_nids = ARRAY_SIZE(vt1716S_adc_nids);
+ get_mux_nids(codec);
+ override_mic_boost(codec, 0x1a, 0, 3, 40);
+ override_mic_boost(codec, 0x1e, 0, 3, 40);
+ spec->mixers[spec->num_mixers] = vt1716S_capture_mixer;
+ spec->num_mixers++;
+ }
+
+ spec->mixers[spec->num_mixers] = vt1716s_dmic_mixer;
+ spec->num_mixers++;
+
+ spec->mixers[spec->num_mixers++] = vt1716S_mono_out_mixer;
+
+ codec->patch_ops = via_patch_ops;
+
+ codec->patch_ops.init = via_auto_init;
+ codec->patch_ops.unsol_event = via_unsol_event;
+
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ spec->loopback.amplist = vt1716S_loopbacks;
+#endif
+
+ return 0;
+}
+
+/* for vt2002P */
+
+/* capture mixer elements */
+static struct snd_kcontrol_new vt2002P_capture_mixer[] = {
+ HDA_CODEC_VOLUME("Capture Volume", 0x10, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Capture Switch", 0x10, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x11, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x11, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Boost Capture Volume", 0x2b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Front Mic Boost Capture Volume", 0x29, 0x0,
+ HDA_INPUT),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ /* The multiple "Capture Source" controls confuse alsamixer
+ * So call somewhat different..
+ */
+ /* .name = "Capture Source", */
+ .name = "Input Source",
+ .count = 2,
+ .info = via_mux_enum_info,
+ .get = via_mux_enum_get,
+ .put = via_mux_enum_put,
+ },
+ { } /* end */
+};
+
+static struct hda_verb vt2002P_volume_init_verbs[] = {
+ /*
+ * Unmute ADC0-1 and set the default input to mic-in
+ */
+ {0x8, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x9, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+
+
+ /* 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 */
+ {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+ {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+ {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
+
+ /* MUX Indices: Mic = 0 */
+ {0x1e, AC_VERB_SET_CONNECT_SEL, 0},
+ {0x1f, AC_VERB_SET_CONNECT_SEL, 0},
+
+ /* PW9 Output enable */
+ {0x2d, AC_VERB_SET_PIN_WIDGET_CONTROL, AC_PINCTL_OUT_EN},
+
+ /* Enable Boost Volume backdoor */
+ {0x1, 0xfb9, 0x24},
+
+ /* MW0/1/4/8: un-mute index 0 (MUXx), un-mute index 1 (MW9) */
+ {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+
+ /* set MUX0/1/4/8 = 0 (AOW0) */
+ {0x34, AC_VERB_SET_CONNECT_SEL, 0},
+ {0x35, AC_VERB_SET_CONNECT_SEL, 0},
+ {0x37, AC_VERB_SET_CONNECT_SEL, 0},
+ {0x3b, AC_VERB_SET_CONNECT_SEL, 0},
+
+ /* set PW0 index=0 (MW0) */
+ {0x24, AC_VERB_SET_CONNECT_SEL, 0},
+
+ /* Enable AOW0 to MW9 */
+ {0x1, 0xfb8, 0x88},
+ { }
+};
+
+
+static struct hda_verb vt2002P_uniwill_init_verbs[] = {
+ {0x25, AC_VERB_SET_UNSOLICITED_ENABLE,
+ AC_USRSP_EN | VIA_JACK_EVENT | VIA_BIND_HP_EVENT},
+ {0x26, AC_VERB_SET_UNSOLICITED_ENABLE,
+ AC_USRSP_EN | VIA_JACK_EVENT | VIA_BIND_HP_EVENT},
+ {0x29, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT},
+ {0x2a, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT},
+ {0x2b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT},
+ { }
+};
+
+static struct hda_pcm_stream vt2002P_pcm_analog_playback = {
+ .substreams = 2,
+ .channels_min = 2,
+ .channels_max = 2,
+ .nid = 0x8, /* NID to query formats and rates */
+ .ops = {
+ .open = via_playback_pcm_open,
+ .prepare = via_playback_multi_pcm_prepare,
+ .cleanup = via_playback_multi_pcm_cleanup,
+ .close = via_pcm_open_close,
+ },
+};
+
+static struct hda_pcm_stream vt2002P_pcm_analog_capture = {
+ .substreams = 2,
+ .channels_min = 2,
+ .channels_max = 2,
+ .nid = 0x10, /* NID to query formats and rates */
+ .ops = {
+ .open = via_pcm_open_close,
+ .prepare = via_capture_pcm_prepare,
+ .cleanup = via_capture_pcm_cleanup,
+ .close = via_pcm_open_close,
+ },
+};
+
+static struct hda_pcm_stream vt2002P_pcm_digital_playback = {
+ .substreams = 1,
+ .channels_min = 2,
+ .channels_max = 2,
+ /* NID is set in via_build_pcms */
+ .ops = {
+ .open = via_dig_playback_pcm_open,
+ .close = via_dig_playback_pcm_close,
+ .prepare = via_dig_playback_pcm_prepare,
+ .cleanup = via_dig_playback_pcm_cleanup
+ },
+};
+
+/* fill in the dac_nids table from the parsed pin configuration */
+static int vt2002P_auto_fill_dac_nids(struct via_spec *spec,
+ const struct auto_pin_cfg *cfg)
+{
+ spec->multiout.num_dacs = 1;
+ spec->multiout.dac_nids = spec->private_dac_nids;
+ if (cfg->line_out_pins[0])
+ spec->multiout.dac_nids[0] = 0x8;
+ return 0;
+}
+
+/* add playback controls from the parsed DAC table */
+static int vt2002P_auto_create_multi_out_ctls(struct via_spec *spec,
+ const struct auto_pin_cfg *cfg)
+{
+ int err;
+
+ if (!cfg->line_out_pins[0])
+ return -1;
+
+
+ /* Line-Out: PortE */
+ err = via_add_control(spec, VIA_CTL_WIDGET_VOL,
+ "Master Front Playback Volume",
+ HDA_COMPOSE_AMP_VAL(0x8, 3, 0, HDA_OUTPUT));
+ if (err < 0)
+ return err;
+ err = via_add_control(spec, VIA_CTL_WIDGET_BIND_PIN_MUTE,
+ "Master Front Playback Switch",
+ HDA_COMPOSE_AMP_VAL(0x26, 3, 0, HDA_OUTPUT));
+ if (err < 0)
+ return err;
+
+ return 0;
+}
+
+static int vt2002P_auto_create_hp_ctls(struct via_spec *spec, hda_nid_t pin)
+{
+ int err;
+
+ if (!pin)
+ return 0;
+
+ spec->multiout.hp_nid = 0x9;
+ spec->hp_independent_mode_index = 1;
+
+ err = via_add_control(spec, VIA_CTL_WIDGET_VOL,
+ "Headphone Playback Volume",
+ HDA_COMPOSE_AMP_VAL(
+ spec->multiout.hp_nid, 3, 0, HDA_OUTPUT));
+ if (err < 0)
+ return err;
+
+ err = via_add_control(spec, VIA_CTL_WIDGET_MUTE,
+ "Headphone Playback Switch",
+ HDA_COMPOSE_AMP_VAL(0x25, 3, 0, HDA_OUTPUT));
+ if (err < 0)
+ return err;
+
+ create_hp_imux(spec);
+ return 0;
+}
+
+/* create playback/capture controls for input pins */
+static int vt2002P_auto_create_analog_input_ctls(struct via_spec *spec,
+ const struct auto_pin_cfg *cfg)
+{
+ static char *labels[] = {
+ "Mic", "Front Mic", "Line", "Front Line", "CD", "Aux", NULL
+ };
+ struct hda_input_mux *imux = &spec->private_imux[0];
+ int i, err, idx = 0;
+
+ for (i = 0; i < AUTO_PIN_LAST; i++) {
+ if (!cfg->input_pins[i])
+ continue;
+
+ switch (cfg->input_pins[i]) {
+ case 0x2b: /* Mic */
+ idx = 0;
+ break;
+
+ case 0x2a: /* Line In */
+ idx = 1;
+ break;
+
+ case 0x29: /* Front Mic */
+ idx = 2;
+ break;
+ }
+ err = via_new_analog_input(spec, labels[i], idx, 0x21);
+ if (err < 0)
+ return err;
+ imux->items[imux->num_items].label = labels[i];
+ imux->items[imux->num_items].index = idx;
+ imux->num_items++;
+ }
+
+ /* build volume/mute control of loopback */
+ err = via_new_analog_input(spec, "Stereo Mixer", 3, 0x21);
+ if (err < 0)
+ return err;
+
+ /* for internal loopback recording select */
+ imux->items[imux->num_items].label = "Stereo Mixer";
+ imux->items[imux->num_items].index = 3;
+ imux->num_items++;
+
+ /* for digital mic select */
+ imux->items[imux->num_items].label = "Digital Mic";
+ imux->items[imux->num_items].index = 4;
+ imux->num_items++;
+
+ return 0;
+}
+
+static int vt2002P_parse_auto_config(struct hda_codec *codec)
+{
+ struct via_spec *spec = codec->spec;
+ int err;
+
+
+ err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, NULL);
+ if (err < 0)
+ return err;
+
+ err = vt2002P_auto_fill_dac_nids(spec, &spec->autocfg);
+ if (err < 0)
+ return err;
+
+ if (!spec->autocfg.line_outs && !spec->autocfg.hp_pins[0])
+ return 0; /* can't find valid BIOS pin config */
+
+ err = vt2002P_auto_create_multi_out_ctls(spec, &spec->autocfg);
+ if (err < 0)
+ return err;
+ err = vt2002P_auto_create_hp_ctls(spec, spec->autocfg.hp_pins[0]);
+ if (err < 0)
+ return err;
+ err = vt2002P_auto_create_analog_input_ctls(spec, &spec->autocfg);
+ if (err < 0)
+ return err;
+
+ spec->multiout.max_channels = spec->multiout.num_dacs * 2;
+
+ fill_dig_outs(codec);
+
+ if (spec->kctls.list)
+ spec->mixers[spec->num_mixers++] = spec->kctls.list;
+
+ spec->input_mux = &spec->private_imux[0];
+
+ if (spec->hp_mux)
+ spec->mixers[spec->num_mixers++] = via_hp_mixer;
+
+ return 1;
+}
+
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+static struct hda_amp_list vt2002P_loopbacks[] = {
+ { 0x21, HDA_INPUT, 0 },
+ { 0x21, HDA_INPUT, 1 },
+ { 0x21, HDA_INPUT, 2 },
+ { } /* end */
+};
+#endif
+
+
+/* patch for vt2002P */
+static int patch_vt2002P(struct hda_codec *codec)
+{
+ struct via_spec *spec;
+ int err;
+
+ /* create a codec specific record */
+ spec = kzalloc(sizeof(*spec), GFP_KERNEL);
+ if (spec == NULL)
+ return -ENOMEM;
+
+ codec->spec = spec;
+
+ /* automatic parse from the BIOS config */
+ err = vt2002P_parse_auto_config(codec);
+ if (err < 0) {
+ via_free(codec);
+ return err;
+ } else if (!err) {
+ printk(KERN_INFO "hda_codec: Cannot set up configuration "
+ "from BIOS. Using genenic mode...\n");
+ }
+
+ spec->init_verbs[spec->num_iverbs++] = vt2002P_volume_init_verbs;
+ spec->init_verbs[spec->num_iverbs++] = vt2002P_uniwill_init_verbs;
+
+ spec->stream_name_analog = "VT2002P Analog";
+ spec->stream_analog_playback = &vt2002P_pcm_analog_playback;
+ spec->stream_analog_capture = &vt2002P_pcm_analog_capture;
+
+ spec->stream_name_digital = "VT2002P Digital";
+ spec->stream_digital_playback = &vt2002P_pcm_digital_playback;
+
+ if (!spec->adc_nids && spec->input_mux) {
+ spec->adc_nids = vt2002P_adc_nids;
+ spec->num_adc_nids = ARRAY_SIZE(vt2002P_adc_nids);
+ get_mux_nids(codec);
+ override_mic_boost(codec, 0x2b, 0, 3, 40);
+ override_mic_boost(codec, 0x29, 0, 3, 40);
+ spec->mixers[spec->num_mixers] = vt2002P_capture_mixer;
+ spec->num_mixers++;
+ }
+
+ codec->patch_ops = via_patch_ops;
+
+ codec->patch_ops.init = via_auto_init;
+ codec->patch_ops.unsol_event = via_unsol_event;
+
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ spec->loopback.amplist = vt2002P_loopbacks;
+#endif
+
+ return 0;
+}
+
+/* for vt1812 */
+
+/* capture mixer elements */
+static struct snd_kcontrol_new vt1812_capture_mixer[] = {
+ HDA_CODEC_VOLUME("Capture Volume", 0x10, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Capture Switch", 0x10, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x11, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x11, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Boost Capture Volume", 0x2b, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Front Mic Boost Capture Volume", 0x29, 0x0,
+ HDA_INPUT),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ /* The multiple "Capture Source" controls confuse alsamixer
+ * So call somewhat different..
+ */
+ .name = "Input Source",
+ .count = 2,
+ .info = via_mux_enum_info,
+ .get = via_mux_enum_get,
+ .put = via_mux_enum_put,
+ },
+ { } /* end */
+};
+
+static struct hda_verb vt1812_volume_init_verbs[] = {
+ /*
+ * Unmute ADC0-1 and set the default input to mic-in
+ */
+ {0x8, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x9, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+
+
+ /* 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 */
+ {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+ {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+ {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
+
+ /* MUX Indices: Mic = 0 */
+ {0x1e, AC_VERB_SET_CONNECT_SEL, 0},
+ {0x1f, AC_VERB_SET_CONNECT_SEL, 0},
+
+ /* PW9 Output enable */
+ {0x2d, AC_VERB_SET_PIN_WIDGET_CONTROL, AC_PINCTL_OUT_EN},
+
+ /* Enable Boost Volume backdoor */
+ {0x1, 0xfb9, 0x24},
+
+ /* MW0/1/4/13/15: un-mute index 0 (MUXx), un-mute index 1 (MW9) */
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x1d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x1d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+
+ /* set MUX0/1/4/13/15 = 0 (AOW0) */
+ {0x34, AC_VERB_SET_CONNECT_SEL, 0},
+ {0x35, AC_VERB_SET_CONNECT_SEL, 0},
+ {0x38, AC_VERB_SET_CONNECT_SEL, 0},
+ {0x3c, AC_VERB_SET_CONNECT_SEL, 0},
+ {0x3d, AC_VERB_SET_CONNECT_SEL, 0},
+
+ /* Enable AOW0 to MW9 */
+ {0x1, 0xfb8, 0xa8},
+ { }
+};
+
+
+static struct hda_verb vt1812_uniwill_init_verbs[] = {
+ {0x33, AC_VERB_SET_UNSOLICITED_ENABLE,
+ AC_USRSP_EN | VIA_JACK_EVENT | VIA_BIND_HP_EVENT},
+ {0x25, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT },
+ {0x28, AC_VERB_SET_UNSOLICITED_ENABLE,
+ AC_USRSP_EN | VIA_JACK_EVENT | VIA_BIND_HP_EVENT},
+ {0x29, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT},
+ {0x2a, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT},
+ {0x2b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT},
+ { }
+};
+
+static struct hda_pcm_stream vt1812_pcm_analog_playback = {
+ .substreams = 2,
+ .channels_min = 2,
+ .channels_max = 2,
+ .nid = 0x8, /* NID to query formats and rates */
+ .ops = {
+ .open = via_playback_pcm_open,
+ .prepare = via_playback_multi_pcm_prepare,
+ .cleanup = via_playback_multi_pcm_cleanup,
+ .close = via_pcm_open_close,
+ },
+};
+
+static struct hda_pcm_stream vt1812_pcm_analog_capture = {
+ .substreams = 2,
+ .channels_min = 2,
+ .channels_max = 2,
+ .nid = 0x10, /* NID to query formats and rates */
+ .ops = {
+ .open = via_pcm_open_close,
+ .prepare = via_capture_pcm_prepare,
+ .cleanup = via_capture_pcm_cleanup,
+ .close = via_pcm_open_close,
+ },
+};
+
+static struct hda_pcm_stream vt1812_pcm_digital_playback = {
+ .substreams = 1,
+ .channels_min = 2,
+ .channels_max = 2,
+ /* NID is set in via_build_pcms */
+ .ops = {
+ .open = via_dig_playback_pcm_open,
+ .close = via_dig_playback_pcm_close,
+ .prepare = via_dig_playback_pcm_prepare,
+ .cleanup = via_dig_playback_pcm_cleanup
+ },
+};
+/* fill in the dac_nids table from the parsed pin configuration */
+static int vt1812_auto_fill_dac_nids(struct via_spec *spec,
+ const struct auto_pin_cfg *cfg)
+{
+ spec->multiout.num_dacs = 1;
+ spec->multiout.dac_nids = spec->private_dac_nids;
+ if (cfg->line_out_pins[0])
+ spec->multiout.dac_nids[0] = 0x8;
+ return 0;
+}
+
+
+/* add playback controls from the parsed DAC table */
+static int vt1812_auto_create_multi_out_ctls(struct via_spec *spec,
+ const struct auto_pin_cfg *cfg)
+{
+ int err;
+
+ if (!cfg->line_out_pins[0])
+ return -1;
+
+ /* Line-Out: PortE */
+ err = via_add_control(spec, VIA_CTL_WIDGET_VOL,
+ "Master Front Playback Volume",
+ HDA_COMPOSE_AMP_VAL(0x8, 3, 0, HDA_OUTPUT));
+ if (err < 0)
+ return err;
+ err = via_add_control(spec, VIA_CTL_WIDGET_BIND_PIN_MUTE,
+ "Master Front Playback Switch",
+ HDA_COMPOSE_AMP_VAL(0x28, 3, 0, HDA_OUTPUT));
+ if (err < 0)
+ return err;
+
+ return 0;
+}
+
+static int vt1812_auto_create_hp_ctls(struct via_spec *spec, hda_nid_t pin)
+{
+ int err;
+
+ if (!pin)
+ return 0;
+
+ spec->multiout.hp_nid = 0x9;
+ spec->hp_independent_mode_index = 1;
+
+
+ err = via_add_control(spec, VIA_CTL_WIDGET_VOL,
+ "Headphone Playback Volume",
+ HDA_COMPOSE_AMP_VAL(
+ spec->multiout.hp_nid, 3, 0, HDA_OUTPUT));
+ if (err < 0)
+ return err;
+
+ err = via_add_control(spec, VIA_CTL_WIDGET_MUTE,
+ "Headphone Playback Switch",
+ HDA_COMPOSE_AMP_VAL(pin, 3, 0, HDA_OUTPUT));
+ if (err < 0)
+ return err;
+
+ create_hp_imux(spec);
+ return 0;
+}
+
+/* create playback/capture controls for input pins */
+static int vt1812_auto_create_analog_input_ctls(struct via_spec *spec,
+ const struct auto_pin_cfg *cfg)
+{
+ static char *labels[] = {
+ "Mic", "Front Mic", "Line", "Front Line", "CD", "Aux", NULL
+ };
+ struct hda_input_mux *imux = &spec->private_imux[0];
+ int i, err, idx = 0;
+
+ for (i = 0; i < AUTO_PIN_LAST; i++) {
+ if (!cfg->input_pins[i])
+ continue;
+
+ switch (cfg->input_pins[i]) {
+ case 0x2b: /* Mic */
+ idx = 0;
+ break;
+
+ case 0x2a: /* Line In */
+ idx = 1;
+ break;
+
+ case 0x29: /* Front Mic */
+ idx = 2;
+ break;
+ }
+ err = via_new_analog_input(spec, labels[i], idx, 0x21);
+ if (err < 0)
+ return err;
+ imux->items[imux->num_items].label = labels[i];
+ imux->items[imux->num_items].index = idx;
+ imux->num_items++;
+ }
+ /* build volume/mute control of loopback */
+ err = via_new_analog_input(spec, "Stereo Mixer", 5, 0x21);
+ if (err < 0)
+ return err;
+
+ /* for internal loopback recording select */
+ imux->items[imux->num_items].label = "Stereo Mixer";
+ imux->items[imux->num_items].index = 5;
+ imux->num_items++;
+
+ /* for digital mic select */
+ imux->items[imux->num_items].label = "Digital Mic";
+ imux->items[imux->num_items].index = 6;
+ imux->num_items++;
+
+ return 0;
+}
+
+static int vt1812_parse_auto_config(struct hda_codec *codec)
+{
+ struct via_spec *spec = codec->spec;
+ int err;
+
+
+ err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, NULL);
+ if (err < 0)
+ return err;
+ fill_dig_outs(codec);
+ err = vt1812_auto_fill_dac_nids(spec, &spec->autocfg);
+ if (err < 0)
+ return err;
+
+ if (!spec->autocfg.line_outs && !spec->autocfg.hp_outs)
+ return 0; /* can't find valid BIOS pin config */
+
+ err = vt1812_auto_create_multi_out_ctls(spec, &spec->autocfg);
+ if (err < 0)
+ return err;
+ err = vt1812_auto_create_hp_ctls(spec, spec->autocfg.hp_pins[0]);
+ if (err < 0)
+ return err;
+ err = vt1812_auto_create_analog_input_ctls(spec, &spec->autocfg);
+ if (err < 0)
+ return err;
+
+ spec->multiout.max_channels = spec->multiout.num_dacs * 2;
+
+ fill_dig_outs(codec);
+
+ if (spec->kctls.list)
+ spec->mixers[spec->num_mixers++] = spec->kctls.list;
+
+ spec->input_mux = &spec->private_imux[0];
+
+ if (spec->hp_mux)
+ spec->mixers[spec->num_mixers++] = via_hp_mixer;
+
+ return 1;
+}
+
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+static struct hda_amp_list vt1812_loopbacks[] = {
+ { 0x21, HDA_INPUT, 0 },
+ { 0x21, HDA_INPUT, 1 },
+ { 0x21, HDA_INPUT, 2 },
+ { } /* end */
+};
+#endif
+
+
+/* patch for vt1812 */
+static int patch_vt1812(struct hda_codec *codec)
+{
+ struct via_spec *spec;
+ int err;
+
+ /* create a codec specific record */
+ spec = kzalloc(sizeof(*spec), GFP_KERNEL);
+ if (spec == NULL)
+ return -ENOMEM;
+
+ codec->spec = spec;
+
+ /* automatic parse from the BIOS config */
+ err = vt1812_parse_auto_config(codec);
+ if (err < 0) {
+ via_free(codec);
+ return err;
+ } else if (!err) {
+ printk(KERN_INFO "hda_codec: Cannot set up configuration "
+ "from BIOS. Using genenic mode...\n");
+ }
+
+
+ spec->init_verbs[spec->num_iverbs++] = vt1812_volume_init_verbs;
+ spec->init_verbs[spec->num_iverbs++] = vt1812_uniwill_init_verbs;
+
+ spec->stream_name_analog = "VT1812 Analog";
+ spec->stream_analog_playback = &vt1812_pcm_analog_playback;
+ spec->stream_analog_capture = &vt1812_pcm_analog_capture;
+
+ spec->stream_name_digital = "VT1812 Digital";
+ spec->stream_digital_playback = &vt1812_pcm_digital_playback;
+
+
+ if (!spec->adc_nids && spec->input_mux) {
+ spec->adc_nids = vt1812_adc_nids;
+ spec->num_adc_nids = ARRAY_SIZE(vt1812_adc_nids);
+ get_mux_nids(codec);
+ override_mic_boost(codec, 0x2b, 0, 3, 40);
+ override_mic_boost(codec, 0x29, 0, 3, 40);
+ spec->mixers[spec->num_mixers] = vt1812_capture_mixer;
+ spec->num_mixers++;
+ }
+
+ codec->patch_ops = via_patch_ops;
+
+ codec->patch_ops.init = via_auto_init;
+ codec->patch_ops.unsol_event = via_unsol_event;
+
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ spec->loopback.amplist = vt1812_loopbacks;
+#endif
return 0;
}
@@ -3318,6 +6214,23 @@ static struct hda_codec_preset snd_hda_preset_via[] = {
.patch = patch_vt1702},
{ .id = 0x11067398, .name = "VT1702",
.patch = patch_vt1702},
+ { .id = 0x11060428, .name = "VT1718S",
+ .patch = patch_vt1718S},
+ { .id = 0x11064428, .name = "VT1718S",
+ .patch = patch_vt1718S},
+ { .id = 0x11060441, .name = "VT2020",
+ .patch = patch_vt1718S},
+ { .id = 0x11064441, .name = "VT1828S",
+ .patch = patch_vt1718S},
+ { .id = 0x11060433, .name = "VT1716S",
+ .patch = patch_vt1716S},
+ { .id = 0x1106a721, .name = "VT1716S",
+ .patch = patch_vt1716S},
+ { .id = 0x11060438, .name = "VT2002P", .patch = patch_vt2002P},
+ { .id = 0x11064438, .name = "VT2002P", .patch = patch_vt2002P},
+ { .id = 0x11060448, .name = "VT1812", .patch = patch_vt1812},
+ { .id = 0x11060440, .name = "VT1818S",
+ .patch = patch_vt1708S},
{} /* terminator */
};
diff --git a/sound/pci/ice1712/amp.c b/sound/pci/ice1712/amp.c
index 37564300b50..6da21a2bcad 100644
--- a/sound/pci/ice1712/amp.c
+++ b/sound/pci/ice1712/amp.c
@@ -52,11 +52,13 @@ static int __devinit snd_vt1724_amp_init(struct snd_ice1712 *ice)
/* only use basic functionality for now */
- ice->num_total_dacs = 2; /* only PSDOUT0 is connected */
+ /* VT1616 6ch codec connected to PSDOUT0 using packed mode */
+ ice->num_total_dacs = 6;
ice->num_total_adcs = 2;
- /* Chaintech AV-710 has another codecs, which need initialization */
- /* initialize WM8728 codec */
+ /* Chaintech AV-710 has another WM8728 codec connected to PSDOUT4
+ (shared with the SPDIF output). Mixer control for this codec
+ is not yet supported. */
if (ice->eeprom.subvendor == VT1724_SUBDEVICE_AV710) {
for (i = 0; i < ARRAY_SIZE(wm_inits); i += 2)
wm_put(ice, wm_inits[i], wm_inits[i+1]);
diff --git a/sound/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c
index cecf1ffeeaa..d74033a2cfb 100644
--- a/sound/pci/ice1712/ice1712.c
+++ b/sound/pci/ice1712/ice1712.c
@@ -2259,7 +2259,7 @@ static int snd_ice1712_pro_peak_get(struct snd_kcontrol *kcontrol,
}
static struct snd_kcontrol_new snd_ice1712_mixer_pro_peak __devinitdata = {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
.name = "Multi Track Peak",
.access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE,
.info = snd_ice1712_pro_peak_info,
diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c
index af6e0014862..10fc92c0557 100644
--- a/sound/pci/ice1712/ice1724.c
+++ b/sound/pci/ice1712/ice1724.c
@@ -648,7 +648,7 @@ static int snd_vt1724_set_pro_rate(struct snd_ice1712 *ice, unsigned int rate,
(inb(ICEMT1724(ice, DMA_PAUSE)) & DMA_PAUSES)) {
/* running? we cannot change the rate now... */
spin_unlock_irqrestore(&ice->reg_lock, flags);
- return -EBUSY;
+ return ((rate == ice->cur_rate) && !force) ? 0 : -EBUSY;
}
if (!force && is_pro_rate_locked(ice)) {
spin_unlock_irqrestore(&ice->reg_lock, flags);
@@ -1294,7 +1294,7 @@ static int __devinit snd_vt1724_pcm_spdif(struct snd_ice1712 *ice, int device)
snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV,
snd_dma_pci_data(ice->pci),
- 64*1024, 64*1024);
+ 256*1024, 256*1024);
ice->pcm = pcm;
@@ -1408,7 +1408,7 @@ static int __devinit snd_vt1724_pcm_indep(struct snd_ice1712 *ice, int device)
snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV,
snd_dma_pci_data(ice->pci),
- 64*1024, 64*1024);
+ 256*1024, 256*1024);
ice->pcm_ds = pcm;
@@ -2110,7 +2110,7 @@ static int snd_vt1724_pro_peak_get(struct snd_kcontrol *kcontrol,
}
static struct snd_kcontrol_new snd_vt1724_mixer_pro_peak __devinitdata = {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
.name = "Multi Track Peak",
.access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE,
.info = snd_vt1724_pro_peak_info,
diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c
index 171ada53520..754867ed478 100644
--- a/sound/pci/intel8x0.c
+++ b/sound/pci/intel8x0.c
@@ -1954,6 +1954,18 @@ static struct ac97_quirk ac97_quirks[] __devinitdata = {
.name = "Sony S1XP",
.type = AC97_TUNE_INV_EAPD
},
+ {
+ .subvendor = 0x104d,
+ .subdevice = 0x81c0,
+ .name = "Sony VAIO VGN-T350P", /*AD1981B*/
+ .type = AC97_TUNE_INV_EAPD
+ },
+ {
+ .subvendor = 0x104d,
+ .subdevice = 0x81c5,
+ .name = "Sony VAIO VGN-B1VP", /*AD1981B*/
+ .type = AC97_TUNE_INV_EAPD
+ },
{
.subvendor = 0x1043,
.subdevice = 0x80f3,
diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c
index acfa4760da4..8a332d2f615 100644
--- a/sound/pci/via82xx.c
+++ b/sound/pci/via82xx.c
@@ -386,6 +386,7 @@ struct via82xx {
struct snd_pcm *pcms[2];
struct snd_rawmidi *rmidi;
+ struct snd_kcontrol *dxs_controls[4];
struct snd_ac97_bus *ac97_bus;
struct snd_ac97 *ac97;
@@ -1216,9 +1217,9 @@ static int snd_via82xx_pcm_open(struct via82xx *chip, struct viadev *viadev,
/*
- * open callback for playback on via686 and via823x DSX
+ * open callback for playback on via686
*/
-static int snd_via82xx_playback_open(struct snd_pcm_substream *substream)
+static int snd_via686_playback_open(struct snd_pcm_substream *substream)
{
struct via82xx *chip = snd_pcm_substream_chip(substream);
struct viadev *viadev = &chip->devs[chip->playback_devno + substream->number];
@@ -1230,6 +1231,32 @@ static int snd_via82xx_playback_open(struct snd_pcm_substream *substream)
}
/*
+ * open callback for playback on via823x DXS
+ */
+static int snd_via8233_playback_open(struct snd_pcm_substream *substream)
+{
+ struct via82xx *chip = snd_pcm_substream_chip(substream);
+ struct viadev *viadev;
+ unsigned int stream;
+ int err;
+
+ viadev = &chip->devs[chip->playback_devno + substream->number];
+ if ((err = snd_via82xx_pcm_open(chip, viadev, substream)) < 0)
+ return err;
+ stream = viadev->reg_offset / 0x10;
+ if (chip->dxs_controls[stream]) {
+ chip->playback_volume[stream][0] = 0;
+ chip->playback_volume[stream][1] = 0;
+ chip->dxs_controls[stream]->vd[0].access &=
+ ~SNDRV_CTL_ELEM_ACCESS_INACTIVE;
+ snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_VALUE |
+ SNDRV_CTL_EVENT_MASK_INFO,
+ &chip->dxs_controls[stream]->id);
+ }
+ return 0;
+}
+
+/*
* open callback for playback on via823x multi-channel
*/
static int snd_via8233_multi_open(struct snd_pcm_substream *substream)
@@ -1302,10 +1329,26 @@ static int snd_via82xx_pcm_close(struct snd_pcm_substream *substream)
return 0;
}
+static int snd_via8233_playback_close(struct snd_pcm_substream *substream)
+{
+ struct via82xx *chip = snd_pcm_substream_chip(substream);
+ struct viadev *viadev = substream->runtime->private_data;
+ unsigned int stream;
+
+ stream = viadev->reg_offset / 0x10;
+ if (chip->dxs_controls[stream]) {
+ chip->dxs_controls[stream]->vd[0].access |=
+ SNDRV_CTL_ELEM_ACCESS_INACTIVE;
+ snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_INFO,
+ &chip->dxs_controls[stream]->id);
+ }
+ return snd_via82xx_pcm_close(substream);
+}
+
/* via686 playback callbacks */
static struct snd_pcm_ops snd_via686_playback_ops = {
- .open = snd_via82xx_playback_open,
+ .open = snd_via686_playback_open,
.close = snd_via82xx_pcm_close,
.ioctl = snd_pcm_lib_ioctl,
.hw_params = snd_via82xx_hw_params,
@@ -1331,8 +1374,8 @@ static struct snd_pcm_ops snd_via686_capture_ops = {
/* via823x DSX playback callbacks */
static struct snd_pcm_ops snd_via8233_playback_ops = {
- .open = snd_via82xx_playback_open,
- .close = snd_via82xx_pcm_close,
+ .open = snd_via8233_playback_open,
+ .close = snd_via8233_playback_close,
.ioctl = snd_pcm_lib_ioctl,
.hw_params = snd_via82xx_hw_params,
.hw_free = snd_via82xx_hw_free,
@@ -1626,7 +1669,7 @@ static int snd_via8233_dxs_volume_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct via82xx *chip = snd_kcontrol_chip(kcontrol);
- unsigned int idx = snd_ctl_get_ioff(kcontrol, &ucontrol->id);
+ unsigned int idx = kcontrol->id.subdevice;
ucontrol->value.integer.value[0] = VIA_DXS_MAX_VOLUME - chip->playback_volume[idx][0];
ucontrol->value.integer.value[1] = VIA_DXS_MAX_VOLUME - chip->playback_volume[idx][1];
@@ -1646,7 +1689,7 @@ static int snd_via8233_dxs_volume_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct via82xx *chip = snd_kcontrol_chip(kcontrol);
- unsigned int idx = snd_ctl_get_ioff(kcontrol, &ucontrol->id);
+ unsigned int idx = kcontrol->id.subdevice;
unsigned long port = chip->port + 0x10 * idx;
unsigned char val;
int i, change = 0;
@@ -1705,11 +1748,13 @@ static struct snd_kcontrol_new snd_via8233_pcmdxs_volume_control __devinitdata =
};
static struct snd_kcontrol_new snd_via8233_dxs_volume_control __devinitdata = {
- .name = "VIA DXS Playback Volume",
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE |
- SNDRV_CTL_ELEM_ACCESS_TLV_READ),
- .count = 4,
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .device = 0,
+ /* .subdevice set later */
+ .name = "PCM Playback Volume",
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE |
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ |
+ SNDRV_CTL_ELEM_ACCESS_INACTIVE,
.info = snd_via8233_dxs_volume_info,
.get = snd_via8233_dxs_volume_get,
.put = snd_via8233_dxs_volume_put,
@@ -1936,10 +1981,19 @@ static int __devinit snd_via8233_init_misc(struct via82xx *chip)
}
else /* Using DXS when PCM emulation is enabled is really weird */
{
- /* Standalone DXS controls */
- err = snd_ctl_add(chip->card, snd_ctl_new1(&snd_via8233_dxs_volume_control, chip));
- if (err < 0)
- return err;
+ for (i = 0; i < 4; ++i) {
+ struct snd_kcontrol *kctl;
+
+ kctl = snd_ctl_new1(
+ &snd_via8233_dxs_volume_control, chip);
+ if (!kctl)
+ return -ENOMEM;
+ kctl->id.subdevice = i;
+ err = snd_ctl_add(chip->card, kctl);
+ if (err < 0)
+ return err;
+ chip->dxs_controls[i] = kctl;
+ }
}
}
/* select spdif data slot 10/11 */
diff --git a/sound/ppc/awacs.c b/sound/ppc/awacs.c
index 2cc0eda4f20..2e156467b81 100644
--- a/sound/ppc/awacs.c
+++ b/sound/ppc/awacs.c
@@ -479,7 +479,7 @@ static int snd_pmac_awacs_put_master_amp(struct snd_kcontrol *kcontrol,
static struct snd_kcontrol_new snd_pmac_awacs_amp_vol[] __devinitdata = {
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "PC Speaker Playback Volume",
+ .name = "Speaker Playback Volume",
.info = snd_pmac_awacs_info_volume_amp,
.get = snd_pmac_awacs_get_volume_amp,
.put = snd_pmac_awacs_put_volume_amp,
@@ -525,7 +525,7 @@ static struct snd_kcontrol_new snd_pmac_awacs_amp_hp_sw __devinitdata = {
static struct snd_kcontrol_new snd_pmac_awacs_amp_spk_sw __devinitdata = {
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "PC Speaker Playback Switch",
+ .name = "Speaker Playback Switch",
.info = snd_pmac_boolean_stereo_info,
.get = snd_pmac_awacs_get_switch_amp,
.put = snd_pmac_awacs_put_switch_amp,
@@ -696,17 +696,17 @@ static struct snd_kcontrol_new snd_pmac_screamer_mic_boost_imac[] __devinitdata
};
static struct snd_kcontrol_new snd_pmac_awacs_speaker_vol[] __devinitdata = {
- AWACS_VOLUME("PC Speaker Playback Volume", 4, 6, 1),
+ AWACS_VOLUME("Speaker Playback Volume", 4, 6, 1),
};
static struct snd_kcontrol_new snd_pmac_awacs_speaker_sw __devinitdata =
-AWACS_SWITCH("PC Speaker Playback Switch", 1, SHIFT_SPKMUTE, 1);
+AWACS_SWITCH("Speaker Playback Switch", 1, SHIFT_SPKMUTE, 1);
static struct snd_kcontrol_new snd_pmac_awacs_speaker_sw_imac1 __devinitdata =
-AWACS_SWITCH("PC Speaker Playback Switch", 1, SHIFT_PAROUT1, 1);
+AWACS_SWITCH("Speaker Playback Switch", 1, SHIFT_PAROUT1, 1);
static struct snd_kcontrol_new snd_pmac_awacs_speaker_sw_imac2 __devinitdata =
-AWACS_SWITCH("PC Speaker Playback Switch", 1, SHIFT_PAROUT1, 0);
+AWACS_SWITCH("Speaker Playback Switch", 1, SHIFT_PAROUT1, 0);
/*
diff --git a/sound/ppc/burgundy.c b/sound/ppc/burgundy.c
index 16ed240e423..0accfe49735 100644
--- a/sound/ppc/burgundy.c
+++ b/sound/ppc/burgundy.c
@@ -505,7 +505,7 @@ static struct snd_kcontrol_new snd_pmac_burgundy_mixers_imac[] __devinitdata = {
MASK_ADDR_BURGUNDY_GAINLINE, 1, 0),
BURGUNDY_VOLUME_B("Mic Gain Capture Volume", 0,
MASK_ADDR_BURGUNDY_GAINMIC, 1, 0),
- BURGUNDY_VOLUME_B("PC Speaker Playback Volume", 0,
+ BURGUNDY_VOLUME_B("Speaker Playback Volume", 0,
MASK_ADDR_BURGUNDY_ATTENSPEAKER, 1, 1),
BURGUNDY_VOLUME_B("Line out Playback Volume", 0,
MASK_ADDR_BURGUNDY_ATTENLINEOUT, 1, 1),
@@ -527,7 +527,7 @@ static struct snd_kcontrol_new snd_pmac_burgundy_mixers_pmac[] __devinitdata = {
MASK_ADDR_BURGUNDY_VOLMIC, 16),
BURGUNDY_VOLUME_B("Line in Gain Capture Volume", 0,
MASK_ADDR_BURGUNDY_GAINMIC, 1, 0),
- BURGUNDY_VOLUME_B("PC Speaker Playback Volume", 0,
+ BURGUNDY_VOLUME_B("Speaker Playback Volume", 0,
MASK_ADDR_BURGUNDY_ATTENMONO, 0, 1),
BURGUNDY_VOLUME_B("Line out Playback Volume", 0,
MASK_ADDR_BURGUNDY_ATTENSPEAKER, 1, 1),
@@ -549,11 +549,11 @@ BURGUNDY_SWITCH_B("Master Playback Switch", 0,
BURGUNDY_OUTPUT_INTERN
| BURGUNDY_OUTPUT_LEFT, BURGUNDY_OUTPUT_RIGHT, 1);
static struct snd_kcontrol_new snd_pmac_burgundy_speaker_sw_imac __devinitdata =
-BURGUNDY_SWITCH_B("PC Speaker Playback Switch", 0,
+BURGUNDY_SWITCH_B("Speaker Playback Switch", 0,
MASK_ADDR_BURGUNDY_MORE_OUTPUTENABLES,
BURGUNDY_OUTPUT_LEFT, BURGUNDY_OUTPUT_RIGHT, 1);
static struct snd_kcontrol_new snd_pmac_burgundy_speaker_sw_pmac __devinitdata =
-BURGUNDY_SWITCH_B("PC Speaker Playback Switch", 0,
+BURGUNDY_SWITCH_B("Speaker Playback Switch", 0,
MASK_ADDR_BURGUNDY_MORE_OUTPUTENABLES,
BURGUNDY_OUTPUT_INTERN, 0, 0);
static struct snd_kcontrol_new snd_pmac_burgundy_line_sw_imac __devinitdata =
diff --git a/sound/ppc/keywest.c b/sound/ppc/keywest.c
index 835fa19ed46..d06f780bd7e 100644
--- a/sound/ppc/keywest.c
+++ b/sound/ppc/keywest.c
@@ -59,6 +59,18 @@ static int keywest_attach_adapter(struct i2c_adapter *adapter)
strlcpy(info.type, "keywest", I2C_NAME_SIZE);
info.addr = keywest_ctx->addr;
keywest_ctx->client = i2c_new_device(adapter, &info);
+ if (!keywest_ctx->client)
+ return -ENODEV;
+ /*
+ * We know the driver is already loaded, so the device should be
+ * already bound. If not it means binding failed, and then there
+ * is no point in keeping the device instantiated.
+ */
+ if (!keywest_ctx->client->driver) {
+ i2c_unregister_device(keywest_ctx->client);
+ keywest_ctx->client = NULL;
+ return -ENODEV;
+ }
/*
* Let i2c-core delete that device on driver removal.
@@ -86,7 +98,7 @@ static const struct i2c_device_id keywest_i2c_id[] = {
{ }
};
-struct i2c_driver keywest_driver = {
+static struct i2c_driver keywest_driver = {
.driver = {
.name = "PMac Keywest Audio",
},
diff --git a/sound/ppc/tumbler.c b/sound/ppc/tumbler.c
index 08e584d1453..789f44f4ac7 100644
--- a/sound/ppc/tumbler.c
+++ b/sound/ppc/tumbler.c
@@ -905,7 +905,7 @@ static struct snd_kcontrol_new tumbler_hp_sw __devinitdata = {
};
static struct snd_kcontrol_new tumbler_speaker_sw __devinitdata = {
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "PC Speaker Playback Switch",
+ .name = "Speaker Playback Switch",
.info = snd_pmac_boolean_mono_info,
.get = tumbler_get_mute_switch,
.put = tumbler_put_mute_switch,
diff --git a/sound/sh/Kconfig b/sound/sh/Kconfig
index aed0f90c391..61139f3c161 100644
--- a/sound/sh/Kconfig
+++ b/sound/sh/Kconfig
@@ -19,5 +19,13 @@ config SND_AICA
help
ALSA Sound driver for the SEGA Dreamcast console.
+config SND_SH_DAC_AUDIO
+ tristate "SuperH DAC audio support"
+ depends on SND
+ depends on CPU_SH3 && HIGH_RES_TIMERS
+ select SND_PCM
+ help
+ Say Y here to include support for the on-chip DAC.
+
endif # SND_SUPERH
diff --git a/sound/sh/Makefile b/sound/sh/Makefile
index 8fdcb6e26f0..7d09b5188cf 100644
--- a/sound/sh/Makefile
+++ b/sound/sh/Makefile
@@ -3,6 +3,8 @@
#
snd-aica-objs := aica.o
+snd-sh_dac_audio-objs := sh_dac_audio.o
# Toplevel Module Dependency
obj-$(CONFIG_SND_AICA) += snd-aica.o
+obj-$(CONFIG_SND_SH_DAC_AUDIO) += snd-sh_dac_audio.o
diff --git a/sound/sh/sh_dac_audio.c b/sound/sh/sh_dac_audio.c
new file mode 100644
index 00000000000..76d9ad27d91
--- /dev/null
+++ b/sound/sh/sh_dac_audio.c
@@ -0,0 +1,453 @@
+/*
+ * sh_dac_audio.c - SuperH DAC audio driver for ALSA
+ *
+ * Copyright (c) 2009 by Rafael Ignacio Zurita <rizurita@yahoo.com>
+ *
+ *
+ * Based on sh_dac_audio.c (Copyright (C) 2004, 2005 by Andriy Skulysh)
+ *
+ * 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
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program 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 <linux/hrtimer.h>
+#include <linux/interrupt.h>
+#include <linux/io.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/initval.h>
+#include <sound/pcm.h>
+#include <sound/sh_dac_audio.h>
+#include <asm/clock.h>
+#include <asm/hd64461.h>
+#include <mach/hp6xx.h>
+#include <cpu/dac.h>
+
+MODULE_AUTHOR("Rafael Ignacio Zurita <rizurita@yahoo.com>");
+MODULE_DESCRIPTION("SuperH DAC audio driver");
+MODULE_LICENSE("GPL");
+MODULE_SUPPORTED_DEVICE("{{SuperH DAC audio support}}");
+
+/* Module Parameters */
+static int index = SNDRV_DEFAULT_IDX1;
+static char *id = SNDRV_DEFAULT_STR1;
+module_param(index, int, 0444);
+MODULE_PARM_DESC(index, "Index value for SuperH DAC audio.");
+module_param(id, charp, 0444);
+MODULE_PARM_DESC(id, "ID string for SuperH DAC audio.");
+
+/* main struct */
+struct snd_sh_dac {
+ struct snd_card *card;
+ struct snd_pcm_substream *substream;
+ struct hrtimer hrtimer;
+ ktime_t wakeups_per_second;
+
+ int rate;
+ int empty;
+ char *data_buffer, *buffer_begin, *buffer_end;
+ int processed; /* bytes proccesed, to compare with period_size */
+ int buffer_size;
+ struct dac_audio_pdata *pdata;
+};
+
+
+static void dac_audio_start_timer(struct snd_sh_dac *chip)
+{
+ hrtimer_start(&chip->hrtimer, chip->wakeups_per_second,
+ HRTIMER_MODE_REL);
+}
+
+static void dac_audio_stop_timer(struct snd_sh_dac *chip)
+{
+ hrtimer_cancel(&chip->hrtimer);
+}
+
+static void dac_audio_reset(struct snd_sh_dac *chip)
+{
+ dac_audio_stop_timer(chip);
+ chip->buffer_begin = chip->buffer_end = chip->data_buffer;
+ chip->processed = 0;
+ chip->empty = 1;
+}
+
+static void dac_audio_set_rate(struct snd_sh_dac *chip)
+{
+ chip->wakeups_per_second = ktime_set(0, 1000000000 / chip->rate);
+}
+
+
+/* PCM INTERFACE */
+
+static struct snd_pcm_hardware snd_sh_dac_pcm_hw = {
+ .info = (SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_HALF_DUPLEX),
+ .formats = SNDRV_PCM_FMTBIT_U8,
+ .rates = SNDRV_PCM_RATE_8000,
+ .rate_min = 8000,
+ .rate_max = 8000,
+ .channels_min = 1,
+ .channels_max = 1,
+ .buffer_bytes_max = (48*1024),
+ .period_bytes_min = 1,
+ .period_bytes_max = (48*1024),
+ .periods_min = 1,
+ .periods_max = 1024,
+};
+
+static int snd_sh_dac_pcm_open(struct snd_pcm_substream *substream)
+{
+ struct snd_sh_dac *chip = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+
+ runtime->hw = snd_sh_dac_pcm_hw;
+
+ chip->substream = substream;
+ chip->buffer_begin = chip->buffer_end = chip->data_buffer;
+ chip->processed = 0;
+ chip->empty = 1;
+
+ chip->pdata->start(chip->pdata);
+
+ return 0;
+}
+
+static int snd_sh_dac_pcm_close(struct snd_pcm_substream *substream)
+{
+ struct snd_sh_dac *chip = snd_pcm_substream_chip(substream);
+
+ chip->substream = NULL;
+
+ dac_audio_stop_timer(chip);
+ chip->pdata->stop(chip->pdata);
+
+ return 0;
+}
+
+static int snd_sh_dac_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));
+}
+
+static int snd_sh_dac_pcm_hw_free(struct snd_pcm_substream *substream)
+{
+ return snd_pcm_lib_free_pages(substream);
+}
+
+static int snd_sh_dac_pcm_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_sh_dac *chip = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = chip->substream->runtime;
+
+ chip->buffer_size = runtime->buffer_size;
+ memset(chip->data_buffer, 0, chip->pdata->buffer_size);
+
+ return 0;
+}
+
+static int snd_sh_dac_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ struct snd_sh_dac *chip = snd_pcm_substream_chip(substream);
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ dac_audio_start_timer(chip);
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ chip->buffer_begin = chip->buffer_end = chip->data_buffer;
+ chip->processed = 0;
+ chip->empty = 1;
+ dac_audio_stop_timer(chip);
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+static int snd_sh_dac_pcm_copy(struct snd_pcm_substream *substream, int channel,
+ snd_pcm_uframes_t pos, void __user *src, snd_pcm_uframes_t count)
+{
+ /* channel is not used (interleaved data) */
+ struct snd_sh_dac *chip = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ ssize_t b_count = frames_to_bytes(runtime , count);
+ ssize_t b_pos = frames_to_bytes(runtime , pos);
+
+ if (count < 0)
+ return -EINVAL;
+
+ if (!count)
+ return 0;
+
+ memcpy_toio(chip->data_buffer + b_pos, src, b_count);
+ chip->buffer_end = chip->data_buffer + b_pos + b_count;
+
+ if (chip->empty) {
+ chip->empty = 0;
+ dac_audio_start_timer(chip);
+ }
+
+ return 0;
+}
+
+static int snd_sh_dac_pcm_silence(struct snd_pcm_substream *substream,
+ int channel, snd_pcm_uframes_t pos,
+ snd_pcm_uframes_t count)
+{
+ /* channel is not used (interleaved data) */
+ struct snd_sh_dac *chip = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ ssize_t b_count = frames_to_bytes(runtime , count);
+ ssize_t b_pos = frames_to_bytes(runtime , pos);
+
+ if (count < 0)
+ return -EINVAL;
+
+ if (!count)
+ return 0;
+
+ memset_io(chip->data_buffer + b_pos, 0, b_count);
+ chip->buffer_end = chip->data_buffer + b_pos + b_count;
+
+ if (chip->empty) {
+ chip->empty = 0;
+ dac_audio_start_timer(chip);
+ }
+
+ return 0;
+}
+
+static
+snd_pcm_uframes_t snd_sh_dac_pcm_pointer(struct snd_pcm_substream *substream)
+{
+ struct snd_sh_dac *chip = snd_pcm_substream_chip(substream);
+ int pointer = chip->buffer_begin - chip->data_buffer;
+
+ return pointer;
+}
+
+/* pcm ops */
+static struct snd_pcm_ops snd_sh_dac_pcm_ops = {
+ .open = snd_sh_dac_pcm_open,
+ .close = snd_sh_dac_pcm_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_sh_dac_pcm_hw_params,
+ .hw_free = snd_sh_dac_pcm_hw_free,
+ .prepare = snd_sh_dac_pcm_prepare,
+ .trigger = snd_sh_dac_pcm_trigger,
+ .pointer = snd_sh_dac_pcm_pointer,
+ .copy = snd_sh_dac_pcm_copy,
+ .silence = snd_sh_dac_pcm_silence,
+ .mmap = snd_pcm_lib_mmap_iomem,
+};
+
+static int __devinit snd_sh_dac_pcm(struct snd_sh_dac *chip, int device)
+{
+ int err;
+ struct snd_pcm *pcm;
+
+ /* device should be always 0 for us */
+ err = snd_pcm_new(chip->card, "SH_DAC PCM", device, 1, 0, &pcm);
+ if (err < 0)
+ return err;
+
+ pcm->private_data = chip;
+ strcpy(pcm->name, "SH_DAC PCM");
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_sh_dac_pcm_ops);
+
+ /* buffer size=48K */
+ snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_CONTINUOUS,
+ snd_dma_continuous_data(GFP_KERNEL),
+ 48 * 1024,
+ 48 * 1024);
+
+ return 0;
+}
+/* END OF PCM INTERFACE */
+
+
+/* driver .remove -- destructor */
+static int snd_sh_dac_remove(struct platform_device *devptr)
+{
+ snd_card_free(platform_get_drvdata(devptr));
+ platform_set_drvdata(devptr, NULL);
+
+ return 0;
+}
+
+/* free -- it has been defined by create */
+static int snd_sh_dac_free(struct snd_sh_dac *chip)
+{
+ /* release the data */
+ kfree(chip->data_buffer);
+ kfree(chip);
+
+ return 0;
+}
+
+static int snd_sh_dac_dev_free(struct snd_device *device)
+{
+ struct snd_sh_dac *chip = device->device_data;
+
+ return snd_sh_dac_free(chip);
+}
+
+static enum hrtimer_restart sh_dac_audio_timer(struct hrtimer *handle)
+{
+ struct snd_sh_dac *chip = container_of(handle, struct snd_sh_dac,
+ hrtimer);
+ struct snd_pcm_runtime *runtime = chip->substream->runtime;
+ ssize_t b_ps = frames_to_bytes(runtime, runtime->period_size);
+
+ if (!chip->empty) {
+ sh_dac_output(*chip->buffer_begin, chip->pdata->channel);
+ chip->buffer_begin++;
+
+ chip->processed++;
+ if (chip->processed >= b_ps) {
+ chip->processed -= b_ps;
+ snd_pcm_period_elapsed(chip->substream);
+ }
+
+ if (chip->buffer_begin == (chip->data_buffer +
+ chip->buffer_size - 1))
+ chip->buffer_begin = chip->data_buffer;
+
+ if (chip->buffer_begin == chip->buffer_end)
+ chip->empty = 1;
+
+ }
+
+ if (!chip->empty)
+ hrtimer_start(&chip->hrtimer, chip->wakeups_per_second,
+ HRTIMER_MODE_REL);
+
+ return HRTIMER_NORESTART;
+}
+
+/* create -- chip-specific constructor for the cards components */
+static int __devinit snd_sh_dac_create(struct snd_card *card,
+ struct platform_device *devptr,
+ struct snd_sh_dac **rchip)
+{
+ struct snd_sh_dac *chip;
+ int err;
+
+ static struct snd_device_ops ops = {
+ .dev_free = snd_sh_dac_dev_free,
+ };
+
+ *rchip = NULL;
+
+ chip = kzalloc(sizeof(*chip), GFP_KERNEL);
+ if (chip == NULL)
+ return -ENOMEM;
+
+ chip->card = card;
+
+ hrtimer_init(&chip->hrtimer, CLOCK_MONOTONIC, HRTIMER_MODE_REL);
+ chip->hrtimer.function = sh_dac_audio_timer;
+
+ dac_audio_reset(chip);
+ chip->rate = 8000;
+ dac_audio_set_rate(chip);
+
+ chip->pdata = devptr->dev.platform_data;
+
+ chip->data_buffer = kmalloc(chip->pdata->buffer_size, GFP_KERNEL);
+ if (chip->data_buffer == NULL) {
+ kfree(chip);
+ return -ENOMEM;
+ }
+
+ err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops);
+ if (err < 0) {
+ snd_sh_dac_free(chip);
+ return err;
+ }
+
+ *rchip = chip;
+
+ return 0;
+}
+
+/* driver .probe -- constructor */
+static int __devinit snd_sh_dac_probe(struct platform_device *devptr)
+{
+ struct snd_sh_dac *chip;
+ struct snd_card *card;
+ int err;
+
+ err = snd_card_create(index, id, THIS_MODULE, 0, &card);
+ if (err < 0) {
+ snd_printk(KERN_ERR "cannot allocate the card\n");
+ return err;
+ }
+
+ err = snd_sh_dac_create(card, devptr, &chip);
+ if (err < 0)
+ goto probe_error;
+
+ err = snd_sh_dac_pcm(chip, 0);
+ if (err < 0)
+ goto probe_error;
+
+ strcpy(card->driver, "snd_sh_dac");
+ strcpy(card->shortname, "SuperH DAC audio driver");
+ printk(KERN_INFO "%s %s", card->longname, card->shortname);
+
+ err = snd_card_register(card);
+ if (err < 0)
+ goto probe_error;
+
+ snd_printk("ALSA driver for SuperH DAC audio");
+
+ platform_set_drvdata(devptr, card);
+ return 0;
+
+probe_error:
+ snd_card_free(card);
+ return err;
+}
+
+/*
+ * "driver" definition
+ */
+static struct platform_driver driver = {
+ .probe = snd_sh_dac_probe,
+ .remove = snd_sh_dac_remove,
+ .driver = {
+ .name = "dac_audio",
+ },
+};
+
+static int __init sh_dac_init(void)
+{
+ return platform_driver_register(&driver);
+}
+
+static void __exit sh_dac_exit(void)
+{
+ platform_driver_unregister(&driver);
+}
+
+module_init(sh_dac_init);
+module_exit(sh_dac_exit);
diff --git a/sound/soc/blackfin/Kconfig b/sound/soc/blackfin/Kconfig
index ac927ffdc96..97f1a251e44 100644
--- a/sound/soc/blackfin/Kconfig
+++ b/sound/soc/blackfin/Kconfig
@@ -7,15 +7,6 @@ config SND_BF5XX_I2S
mode (supports single stereo In/Out).
You will also need to select the audio interfaces to support below.
-config SND_BF5XX_TDM
- tristate "SoC I2S(TDM mode) Audio for the ADI BF5xx chip"
- depends on (BLACKFIN && SND_SOC)
- help
- Say Y or M if you want to add support for codecs attached to
- the Blackfin SPORT (synchronous serial ports) interface in TDM
- mode.
- You will also need to select the audio interfaces to support below.
-
config SND_BF5XX_SOC_SSM2602
tristate "SoC SSM2602 Audio support for BF52x ezkit"
depends on SND_BF5XX_I2S
@@ -41,6 +32,31 @@ config SND_BFIN_AD73311_SE
Enter the GPIO used to control AD73311's SE pin. Acceptable
values are 0 to 7
+config SND_BF5XX_TDM
+ tristate "SoC I2S(TDM mode) Audio for the ADI BF5xx chip"
+ depends on (BLACKFIN && SND_SOC)
+ help
+ Say Y or M if you want to add support for codecs attached to
+ the Blackfin SPORT (synchronous serial ports) interface in TDM
+ mode.
+ You will also need to select the audio interfaces to support below.
+
+config SND_BF5XX_SOC_AD1836
+ tristate "SoC AD1836 Audio support for BF5xx"
+ depends on SND_BF5XX_TDM
+ select SND_BF5XX_SOC_TDM
+ select SND_SOC_AD1836
+ help
+ Say Y if you want to add support for SoC audio on BF5xx STAMP/EZKIT.
+
+config SND_BF5XX_SOC_AD1938
+ tristate "SoC AD1938 Audio support for Blackfin"
+ depends on SND_BF5XX_TDM
+ select SND_BF5XX_SOC_TDM
+ select SND_SOC_AD1938
+ help
+ Say Y if you want to add support for AD1938 codec on Blackfin.
+
config SND_BF5XX_AC97
tristate "SoC AC97 Audio for the ADI BF5xx chip"
depends on BLACKFIN
@@ -71,6 +87,30 @@ config SND_BF5XX_MULTICHAN_SUPPORT
Say y if you want AC97 driver to support up to 5.1 channel audio.
this mode will consume much more memory for DMA.
+config SND_BF5XX_HAVE_COLD_RESET
+ bool "BOARD has COLD Reset GPIO"
+ depends on SND_BF5XX_AC97
+ default y if BFIN548_EZKIT
+ default n if !BFIN548_EZKIT
+
+config SND_BF5XX_RESET_GPIO_NUM
+ int "Set a GPIO for cold reset"
+ depends on SND_BF5XX_HAVE_COLD_RESET
+ range 0 159
+ default 19 if BFIN548_EZKIT
+ default 5 if BFIN537_STAMP
+ default 0
+ help
+ Set the correct GPIO for RESET the sound chip.
+
+config SND_BF5XX_SOC_AD1980
+ tristate "SoC AD1980/1 Audio support for BF5xx"
+ depends on SND_BF5XX_AC97
+ select SND_BF5XX_SOC_AC97
+ select SND_SOC_AD1980
+ help
+ Say Y if you want to add support for SoC audio on BF5xx STAMP/EZKIT.
+
config SND_BF5XX_SOC_SPORT
tristate
@@ -88,30 +128,6 @@ config SND_BF5XX_SOC_AC97
select SND_SOC_AC97_BUS
select SND_BF5XX_SOC_SPORT
-config SND_BF5XX_SOC_AD1836
- tristate "SoC AD1836 Audio support for BF5xx"
- depends on SND_BF5XX_TDM
- select SND_BF5XX_SOC_TDM
- select SND_SOC_AD1836
- help
- Say Y if you want to add support for SoC audio on BF5xx STAMP/EZKIT.
-
-config SND_BF5XX_SOC_AD1980
- tristate "SoC AD1980/1 Audio support for BF5xx"
- depends on SND_BF5XX_AC97
- select SND_BF5XX_SOC_AC97
- select SND_SOC_AD1980
- help
- Say Y if you want to add support for SoC audio on BF5xx STAMP/EZKIT.
-
-config SND_BF5XX_SOC_AD1938
- tristate "SoC AD1938 Audio support for Blackfin"
- depends on SND_BF5XX_TDM
- select SND_BF5XX_SOC_TDM
- select SND_SOC_AD1938
- help
- Say Y if you want to add support for AD1938 codec on Blackfin.
-
config SND_BF5XX_SPORT_NUM
int "Set a SPORT for Sound chip"
depends on (SND_BF5XX_I2S || SND_BF5XX_AC97 || SND_BF5XX_TDM)
@@ -120,19 +136,3 @@ config SND_BF5XX_SPORT_NUM
default 0
help
Set the correct SPORT for sound chip.
-
-config SND_BF5XX_HAVE_COLD_RESET
- bool "BOARD has COLD Reset GPIO"
- depends on SND_BF5XX_AC97
- default y if BFIN548_EZKIT
- default n if !BFIN548_EZKIT
-
-config SND_BF5XX_RESET_GPIO_NUM
- int "Set a GPIO for cold reset"
- depends on SND_BF5XX_HAVE_COLD_RESET
- range 0 159
- default 19 if BFIN548_EZKIT
- default 5 if BFIN537_STAMP
- default 0
- help
- Set the correct GPIO for RESET the sound chip.
diff --git a/sound/soc/blackfin/bf5xx-i2s.c b/sound/soc/blackfin/bf5xx-i2s.c
index 1e9d161c76c..084b68884ad 100644
--- a/sound/soc/blackfin/bf5xx-i2s.c
+++ b/sound/soc/blackfin/bf5xx-i2s.c
@@ -77,12 +77,12 @@ static struct sport_param sport_params[2] = {
* TFS. When Port G is selected and EMAC then there is a conflict between
* the PHY interrupt line and TFS. Current settings prevent the conflict
* by ignoring the TFS pin when Port G is selected. This allows both
- * ssm2602 using Port G and EMAC concurrently.
+ * codecs and EMAC using Port G concurrently.
*/
-#ifdef CONFIG_BF527_SPORT0_PORTF
-#define LOCAL_SPORT0_TFS (P_SPORT0_TFS)
-#else
+#ifdef CONFIG_BF527_SPORT0_PORTG
#define LOCAL_SPORT0_TFS (0)
+#else
+#define LOCAL_SPORT0_TFS (P_SPORT0_TFS)
#endif
static u16 sport_req[][7] = { {P_SPORT0_DTPRI, P_SPORT0_TSCLK, P_SPORT0_RFS,
diff --git a/sound/soc/blackfin/bf5xx-tdm.c b/sound/soc/blackfin/bf5xx-tdm.c
index 3096badf09a..ff546e91a22 100644
--- a/sound/soc/blackfin/bf5xx-tdm.c
+++ b/sound/soc/blackfin/bf5xx-tdm.c
@@ -78,12 +78,12 @@ static struct sport_param sport_params[2] = {
* TFS. When Port G is selected and EMAC then there is a conflict between
* the PHY interrupt line and TFS. Current settings prevent the conflict
* by ignoring the TFS pin when Port G is selected. This allows both
- * ssm2602 using Port G and EMAC concurrently.
+ * codecs and EMAC using Port G concurrently.
*/
-#ifdef CONFIG_BF527_SPORT0_PORTF
-#define LOCAL_SPORT0_TFS (P_SPORT0_TFS)
-#else
+#ifdef CONFIG_BF527_SPORT0_PORTG
#define LOCAL_SPORT0_TFS (0)
+#else
+#define LOCAL_SPORT0_TFS (P_SPORT0_TFS)
#endif
static u16 sport_req[][7] = { {P_SPORT0_DTPRI, P_SPORT0_TSCLK, P_SPORT0_RFS,
diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c
index 0b8dcb5cd72..35606ae6086 100644
--- a/sound/soc/codecs/tlv320aic23.c
+++ b/sound/soc/codecs/tlv320aic23.c
@@ -85,7 +85,7 @@ static int tlv320aic23_write(struct snd_soc_codec *codec, unsigned int reg,
* of data into val
*/
- if ((reg < 0 || reg > 9) && (reg != 15)) {
+ if (reg > 9 && reg != 15) {
printk(KERN_WARNING "%s Invalid register R%u\n", __func__, reg);
return -1;
}
diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c
index 3ff0373dff8..593d5b9c9f0 100644
--- a/sound/soc/codecs/wm8350.c
+++ b/sound/soc/codecs/wm8350.c
@@ -579,7 +579,7 @@ static const struct snd_kcontrol_new wm8350_left_capt_mixer_controls[] = {
SOC_DAPM_SINGLE_TLV("L3 Capture Volume",
WM8350_INPUT_MIXER_VOLUME_L, 9, 7, 0, out_mix_tlv),
SOC_DAPM_SINGLE("PGA Capture Switch",
- WM8350_LEFT_INPUT_VOLUME, 14, 1, 0),
+ WM8350_LEFT_INPUT_VOLUME, 14, 1, 1),
};
/* Right Input Mixer */
@@ -589,7 +589,7 @@ static const struct snd_kcontrol_new wm8350_right_capt_mixer_controls[] = {
SOC_DAPM_SINGLE_TLV("L3 Capture Volume",
WM8350_INPUT_MIXER_VOLUME_R, 13, 7, 0, out_mix_tlv),
SOC_DAPM_SINGLE("PGA Capture Switch",
- WM8350_RIGHT_INPUT_VOLUME, 14, 1, 0),
+ WM8350_RIGHT_INPUT_VOLUME, 14, 1, 1),
};
/* Left Mic Mixer */
diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c
index da97aae475a..1ef2454c520 100644
--- a/sound/soc/codecs/wm8940.c
+++ b/sound/soc/codecs/wm8940.c
@@ -790,7 +790,7 @@ static int wm8940_register(struct wm8940_priv *wm8940,
codec->reg_cache = &wm8940->reg_cache;
ret = snd_soc_codec_set_cache_io(codec, 8, 16, control);
- if (ret == 0) {
+ if (ret < 0) {
dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
return ret;
}
diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c
index abed37acf78..60e360b1046 100644
--- a/sound/soc/codecs/wm9713.c
+++ b/sound/soc/codecs/wm9713.c
@@ -165,9 +165,9 @@ SOC_SINGLE("Mono Playback Switch", AC97_MASTER_TONE, 7, 1, 1),
SOC_SINGLE("Mono Playback ZC Switch", AC97_MASTER_TONE, 6, 1, 0),
SOC_SINGLE("Mono Playback Volume", AC97_MASTER_TONE, 0, 31, 1),
-SOC_SINGLE("PC Beep Playback Headphone Volume", AC97_AUX, 12, 7, 1),
-SOC_SINGLE("PC Beep Playback Speaker Volume", AC97_AUX, 8, 7, 1),
-SOC_SINGLE("PC Beep Playback Mono Volume", AC97_AUX, 4, 7, 1),
+SOC_SINGLE("Beep Playback Headphone Volume", AC97_AUX, 12, 7, 1),
+SOC_SINGLE("Beep Playback Speaker Volume", AC97_AUX, 8, 7, 1),
+SOC_SINGLE("Beep Playback Mono Volume", AC97_AUX, 4, 7, 1),
SOC_SINGLE("Voice Playback Headphone Volume", AC97_PCM, 12, 7, 1),
SOC_SINGLE("Voice Playback Master Volume", AC97_PCM, 8, 7, 1),
@@ -266,7 +266,7 @@ static int mixer_event(struct snd_soc_dapm_widget *w,
/* Left Headphone Mixers */
static const struct snd_kcontrol_new wm9713_hpl_mixer_controls[] = {
-SOC_DAPM_SINGLE("PC Beep Playback Switch", HPL_MIXER, 5, 1, 0),
+SOC_DAPM_SINGLE("Beep Playback Switch", HPL_MIXER, 5, 1, 0),
SOC_DAPM_SINGLE("Voice Playback Switch", HPL_MIXER, 4, 1, 0),
SOC_DAPM_SINGLE("Aux Playback Switch", HPL_MIXER, 3, 1, 0),
SOC_DAPM_SINGLE("PCM Playback Switch", HPL_MIXER, 2, 1, 0),
@@ -276,7 +276,7 @@ SOC_DAPM_SINGLE("Bypass Playback Switch", HPL_MIXER, 0, 1, 0),
/* Right Headphone Mixers */
static const struct snd_kcontrol_new wm9713_hpr_mixer_controls[] = {
-SOC_DAPM_SINGLE("PC Beep Playback Switch", HPR_MIXER, 5, 1, 0),
+SOC_DAPM_SINGLE("Beep Playback Switch", HPR_MIXER, 5, 1, 0),
SOC_DAPM_SINGLE("Voice Playback Switch", HPR_MIXER, 4, 1, 0),
SOC_DAPM_SINGLE("Aux Playback Switch", HPR_MIXER, 3, 1, 0),
SOC_DAPM_SINGLE("PCM Playback Switch", HPR_MIXER, 2, 1, 0),
@@ -294,7 +294,7 @@ SOC_DAPM_ENUM("Route", wm9713_enum[0]);
/* Speaker Mixer */
static const struct snd_kcontrol_new wm9713_speaker_mixer_controls[] = {
-SOC_DAPM_SINGLE("PC Beep Playback Switch", AC97_AUX, 11, 1, 1),
+SOC_DAPM_SINGLE("Beep Playback Switch", AC97_AUX, 11, 1, 1),
SOC_DAPM_SINGLE("Voice Playback Switch", AC97_PCM, 11, 1, 1),
SOC_DAPM_SINGLE("Aux Playback Switch", AC97_REC_SEL, 11, 1, 1),
SOC_DAPM_SINGLE("PCM Playback Switch", AC97_PHONE, 14, 1, 1),
@@ -304,7 +304,7 @@ SOC_DAPM_SINGLE("Bypass Playback Switch", AC97_PC_BEEP, 14, 1, 1),
/* Mono Mixer */
static const struct snd_kcontrol_new wm9713_mono_mixer_controls[] = {
-SOC_DAPM_SINGLE("PC Beep Playback Switch", AC97_AUX, 7, 1, 1),
+SOC_DAPM_SINGLE("Beep Playback Switch", AC97_AUX, 7, 1, 1),
SOC_DAPM_SINGLE("Voice Playback Switch", AC97_PCM, 7, 1, 1),
SOC_DAPM_SINGLE("Aux Playback Switch", AC97_REC_SEL, 7, 1, 1),
SOC_DAPM_SINGLE("PCM Playback Switch", AC97_PHONE, 13, 1, 1),
@@ -463,7 +463,7 @@ SND_SOC_DAPM_VMID("VMID"),
static const struct snd_soc_dapm_route audio_map[] = {
/* left HP mixer */
- {"Left HP Mixer", "PC Beep Playback Switch", "PCBEEP"},
+ {"Left HP Mixer", "Beep Playback Switch", "PCBEEP"},
{"Left HP Mixer", "Voice Playback Switch", "Voice DAC"},
{"Left HP Mixer", "Aux Playback Switch", "Aux DAC"},
{"Left HP Mixer", "Bypass Playback Switch", "Left Line In"},
@@ -472,7 +472,7 @@ static const struct snd_soc_dapm_route audio_map[] = {
{"Left HP Mixer", NULL, "Capture Headphone Mux"},
/* right HP mixer */
- {"Right HP Mixer", "PC Beep Playback Switch", "PCBEEP"},
+ {"Right HP Mixer", "Beep Playback Switch", "PCBEEP"},
{"Right HP Mixer", "Voice Playback Switch", "Voice DAC"},
{"Right HP Mixer", "Aux Playback Switch", "Aux DAC"},
{"Right HP Mixer", "Bypass Playback Switch", "Right Line In"},
@@ -491,7 +491,7 @@ static const struct snd_soc_dapm_route audio_map[] = {
{"Capture Mixer", NULL, "Right Capture Source"},
/* speaker mixer */
- {"Speaker Mixer", "PC Beep Playback Switch", "PCBEEP"},
+ {"Speaker Mixer", "Beep Playback Switch", "PCBEEP"},
{"Speaker Mixer", "Voice Playback Switch", "Voice DAC"},
{"Speaker Mixer", "Aux Playback Switch", "Aux DAC"},
{"Speaker Mixer", "Bypass Playback Switch", "Line Mixer"},
@@ -499,7 +499,7 @@ static const struct snd_soc_dapm_route audio_map[] = {
{"Speaker Mixer", "MonoIn Playback Switch", "Mono In"},
/* mono mixer */
- {"Mono Mixer", "PC Beep Playback Switch", "PCBEEP"},
+ {"Mono Mixer", "Beep Playback Switch", "PCBEEP"},
{"Mono Mixer", "Voice Playback Switch", "Voice DAC"},
{"Mono Mixer", "Aux Playback Switch", "Aux DAC"},
{"Mono Mixer", "Bypass Playback Switch", "Line Mixer"},
diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c
index 12a6c549ee6..4ae70704802 100644
--- a/sound/soc/davinci/davinci-i2s.c
+++ b/sound/soc/davinci/davinci-i2s.c
@@ -97,22 +97,19 @@ enum {
DAVINCI_MCBSP_WORD_32,
};
-static struct davinci_pcm_dma_params davinci_i2s_pcm_out = {
- .name = "I2S PCM Stereo out",
-};
-
-static struct davinci_pcm_dma_params davinci_i2s_pcm_in = {
- .name = "I2S PCM Stereo in",
-};
-
struct davinci_mcbsp_dev {
+ /*
+ * dma_params must be first because rtd->dai->cpu_dai->private_data
+ * is cast to a pointer of an array of struct davinci_pcm_dma_params in
+ * davinci_pcm_open.
+ */
+ struct davinci_pcm_dma_params dma_params[2];
void __iomem *base;
#define MOD_DSP_A 0
#define MOD_DSP_B 1
int mode;
u32 pcr;
struct clk *clk;
- struct davinci_pcm_dma_params *dma_params[2];
};
static inline void davinci_mcbsp_write_reg(struct davinci_mcbsp_dev *dev,
@@ -215,14 +212,6 @@ static void davinci_mcbsp_stop(struct davinci_mcbsp_dev *dev, int playback)
toggle_clock(dev, playback);
}
-static int davinci_i2s_startup(struct snd_pcm_substream *substream,
- struct snd_soc_dai *cpu_dai)
-{
- struct davinci_mcbsp_dev *dev = cpu_dai->private_data;
- cpu_dai->dma_data = dev->dma_params[substream->stream];
- return 0;
-}
-
#define DEFAULT_BITPERSAMPLE 16
static int davinci_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai,
@@ -353,8 +342,9 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params,
struct snd_soc_dai *dai)
{
- struct davinci_pcm_dma_params *dma_params = dai->dma_data;
struct davinci_mcbsp_dev *dev = dai->private_data;
+ struct davinci_pcm_dma_params *dma_params =
+ &dev->dma_params[substream->stream];
struct snd_interval *i = NULL;
int mcbsp_word_length;
unsigned int rcr, xcr, srgr;
@@ -472,7 +462,6 @@ static void davinci_i2s_shutdown(struct snd_pcm_substream *substream,
#define DAVINCI_I2S_RATES SNDRV_PCM_RATE_8000_96000
static struct snd_soc_dai_ops davinci_i2s_dai_ops = {
- .startup = davinci_i2s_startup,
.shutdown = davinci_i2s_shutdown,
.prepare = davinci_i2s_prepare,
.trigger = davinci_i2s_trigger,
@@ -534,12 +523,10 @@ static int davinci_i2s_probe(struct platform_device *pdev)
dev->base = (void __iomem *)IO_ADDRESS(mem->start);
- dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK] = &davinci_i2s_pcm_out;
- dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK]->dma_addr =
+ dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK].dma_addr =
(dma_addr_t)(io_v2p(dev->base) + DAVINCI_MCBSP_DXR_REG);
- dev->dma_params[SNDRV_PCM_STREAM_CAPTURE] = &davinci_i2s_pcm_in;
- dev->dma_params[SNDRV_PCM_STREAM_CAPTURE]->dma_addr =
+ dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].dma_addr =
(dma_addr_t)(io_v2p(dev->base) + DAVINCI_MCBSP_DRR_REG);
/* first TX, then RX */
@@ -549,7 +536,7 @@ static int davinci_i2s_probe(struct platform_device *pdev)
ret = -ENXIO;
goto err_free_mem;
}
- dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK]->channel = res->start;
+ dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK].channel = res->start;
res = platform_get_resource(pdev, IORESOURCE_DMA, 1);
if (!res) {
@@ -557,7 +544,7 @@ static int davinci_i2s_probe(struct platform_device *pdev)
ret = -ENXIO;
goto err_free_mem;
}
- dev->dma_params[SNDRV_PCM_STREAM_CAPTURE]->channel = res->start;
+ dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].channel = res->start;
davinci_i2s_dai.private_data = dev;
ret = snd_soc_register_dai(&davinci_i2s_dai);
diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c
index 7a06c0a8666..5d1f98a4c97 100644
--- a/sound/soc/davinci/davinci-mcasp.c
+++ b/sound/soc/davinci/davinci-mcasp.c
@@ -332,14 +332,6 @@ static inline void mcasp_set_ctl_reg(void __iomem *regs, u32 val)
printk(KERN_ERR "GBLCTL write error\n");
}
-static int davinci_mcasp_startup(struct snd_pcm_substream *substream,
- struct snd_soc_dai *cpu_dai)
-{
- struct davinci_audio_dev *dev = cpu_dai->private_data;
- cpu_dai->dma_data = dev->dma_params[substream->stream];
- return 0;
-}
-
static void mcasp_start_rx(struct davinci_audio_dev *dev)
{
mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, RXHCLKRST);
@@ -386,17 +378,17 @@ static void mcasp_start_tx(struct davinci_audio_dev *dev)
static void davinci_mcasp_start(struct davinci_audio_dev *dev, int stream)
{
- if (stream == SNDRV_PCM_STREAM_PLAYBACK)
+ if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ if (dev->txnumevt) /* enable FIFO */
+ mcasp_set_bits(dev->base + DAVINCI_MCASP_WFIFOCTL,
+ FIFO_ENABLE);
mcasp_start_tx(dev);
- else
+ } else {
+ if (dev->rxnumevt) /* enable FIFO */
+ mcasp_set_bits(dev->base + DAVINCI_MCASP_RFIFOCTL,
+ FIFO_ENABLE);
mcasp_start_rx(dev);
-
- /* enable FIFO */
- if (dev->txnumevt)
- mcasp_set_bits(dev->base + DAVINCI_MCASP_WFIFOCTL, FIFO_ENABLE);
-
- if (dev->rxnumevt)
- mcasp_set_bits(dev->base + DAVINCI_MCASP_RFIFOCTL, FIFO_ENABLE);
+ }
}
static void mcasp_stop_rx(struct davinci_audio_dev *dev)
@@ -413,17 +405,17 @@ static void mcasp_stop_tx(struct davinci_audio_dev *dev)
static void davinci_mcasp_stop(struct davinci_audio_dev *dev, int stream)
{
- if (stream == SNDRV_PCM_STREAM_PLAYBACK)
+ if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ if (dev->txnumevt) /* disable FIFO */
+ mcasp_clr_bits(dev->base + DAVINCI_MCASP_WFIFOCTL,
+ FIFO_ENABLE);
mcasp_stop_tx(dev);
- else
+ } else {
+ if (dev->rxnumevt) /* disable FIFO */
+ mcasp_clr_bits(dev->base + DAVINCI_MCASP_RFIFOCTL,
+ FIFO_ENABLE);
mcasp_stop_rx(dev);
-
- /* disable FIFO */
- if (dev->txnumevt)
- mcasp_clr_bits(dev->base + DAVINCI_MCASP_WFIFOCTL, FIFO_ENABLE);
-
- if (dev->rxnumevt)
- mcasp_clr_bits(dev->base + DAVINCI_MCASP_RFIFOCTL, FIFO_ENABLE);
+ }
}
static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai,
@@ -720,7 +712,7 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream,
{
struct davinci_audio_dev *dev = cpu_dai->private_data;
struct davinci_pcm_dma_params *dma_params =
- dev->dma_params[substream->stream];
+ &dev->dma_params[substream->stream];
int word_length;
u8 numevt;
@@ -798,7 +790,6 @@ static int davinci_mcasp_trigger(struct snd_pcm_substream *substream,
}
static struct snd_soc_dai_ops davinci_mcasp_dai_ops = {
- .startup = davinci_mcasp_startup,
.trigger = davinci_mcasp_trigger,
.hw_params = davinci_mcasp_hw_params,
.set_fmt = davinci_mcasp_set_dai_fmt,
@@ -849,20 +840,12 @@ static int davinci_mcasp_probe(struct platform_device *pdev)
struct resource *mem, *ioarea, *res;
struct snd_platform_data *pdata;
struct davinci_audio_dev *dev;
- int count = 0;
int ret = 0;
dev = kzalloc(sizeof(struct davinci_audio_dev), GFP_KERNEL);
if (!dev)
return -ENOMEM;
- dma_data = kzalloc(sizeof(struct davinci_pcm_dma_params) * 2,
- GFP_KERNEL);
- if (!dma_data) {
- ret = -ENOMEM;
- goto err_release_dev;
- }
-
mem = platform_get_resource(pdev, IORESOURCE_MEM, 0);
if (!mem) {
dev_err(&pdev->dev, "no mem resource?\n");
@@ -897,11 +880,10 @@ static int davinci_mcasp_probe(struct platform_device *pdev)
dev->txnumevt = pdata->txnumevt;
dev->rxnumevt = pdata->rxnumevt;
- dma_data[count].name = "I2S PCM Stereo out";
- dma_data[count].eventq_no = pdata->eventq_no;
- dma_data[count].dma_addr = (dma_addr_t) (pdata->tx_dma_offset +
+ dma_data = &dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK];
+ dma_data->eventq_no = pdata->eventq_no;
+ dma_data->dma_addr = (dma_addr_t) (pdata->tx_dma_offset +
io_v2p(dev->base));
- dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK] = &dma_data[count];
/* first TX, then RX */
res = platform_get_resource(pdev, IORESOURCE_DMA, 0);
@@ -910,13 +892,12 @@ static int davinci_mcasp_probe(struct platform_device *pdev)
goto err_release_region;
}
- dma_data[count].channel = res->start;
- count++;
- dma_data[count].name = "I2S PCM Stereo in";
- dma_data[count].eventq_no = pdata->eventq_no;
- dma_data[count].dma_addr = (dma_addr_t)(pdata->rx_dma_offset +
+ dma_data->channel = res->start;
+
+ dma_data = &dev->dma_params[SNDRV_PCM_STREAM_CAPTURE];
+ dma_data->eventq_no = pdata->eventq_no;
+ dma_data->dma_addr = (dma_addr_t)(pdata->rx_dma_offset +
io_v2p(dev->base));
- dev->dma_params[SNDRV_PCM_STREAM_CAPTURE] = &dma_data[count];
res = platform_get_resource(pdev, IORESOURCE_DMA, 1);
if (!res) {
@@ -924,7 +905,7 @@ static int davinci_mcasp_probe(struct platform_device *pdev)
goto err_release_region;
}
- dma_data[count].channel = res->start;
+ dma_data->channel = res->start;
davinci_mcasp_dai[pdata->op_mode].private_data = dev;
davinci_mcasp_dai[pdata->op_mode].dev = &pdev->dev;
ret = snd_soc_register_dai(&davinci_mcasp_dai[pdata->op_mode]);
@@ -936,8 +917,6 @@ static int davinci_mcasp_probe(struct platform_device *pdev)
err_release_region:
release_mem_region(mem->start, (mem->end - mem->start) + 1);
err_release_data:
- kfree(dma_data);
-err_release_dev:
kfree(dev);
return ret;
@@ -946,7 +925,6 @@ err_release_dev:
static int davinci_mcasp_remove(struct platform_device *pdev)
{
struct snd_platform_data *pdata = pdev->dev.platform_data;
- struct davinci_pcm_dma_params *dma_data;
struct davinci_audio_dev *dev;
struct resource *mem;
@@ -959,8 +937,6 @@ static int davinci_mcasp_remove(struct platform_device *pdev)
mem = platform_get_resource(pdev, IORESOURCE_MEM, 0);
release_mem_region(mem->start, (mem->end - mem->start) + 1);
- dma_data = dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK];
- kfree(dma_data);
kfree(dev);
return 0;
diff --git a/sound/soc/davinci/davinci-mcasp.h b/sound/soc/davinci/davinci-mcasp.h
index 554354c1cc2..9d179cc88f7 100644
--- a/sound/soc/davinci/davinci-mcasp.h
+++ b/sound/soc/davinci/davinci-mcasp.h
@@ -39,10 +39,15 @@ enum {
};
struct davinci_audio_dev {
+ /*
+ * dma_params must be first because rtd->dai->cpu_dai->private_data
+ * is cast to a pointer of an array of struct davinci_pcm_dma_params in
+ * davinci_pcm_open.
+ */
+ struct davinci_pcm_dma_params dma_params[2];
void __iomem *base;
int sample_rate;
struct clk *clk;
- struct davinci_pcm_dma_params *dma_params[2];
unsigned int codec_fmt;
/* McASP specific data */
diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c
index 2f7da49ed34..c73a915f233 100644
--- a/sound/soc/davinci/davinci-pcm.c
+++ b/sound/soc/davinci/davinci-pcm.c
@@ -126,16 +126,9 @@ static void davinci_pcm_dma_irq(unsigned lch, u16 ch_status, void *data)
static int davinci_pcm_dma_request(struct snd_pcm_substream *substream)
{
struct davinci_runtime_data *prtd = substream->runtime->private_data;
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct davinci_pcm_dma_params *dma_data = rtd->dai->cpu_dai->dma_data;
struct edmacc_param p_ram;
int ret;
- if (!dma_data)
- return -ENODEV;
-
- prtd->params = dma_data;
-
/* Request master DMA channel */
ret = edma_alloc_channel(prtd->params->channel,
davinci_pcm_dma_irq, substream,
@@ -244,6 +237,11 @@ static int davinci_pcm_open(struct snd_pcm_substream *substream)
struct snd_pcm_runtime *runtime = substream->runtime;
struct davinci_runtime_data *prtd;
int ret = 0;
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct davinci_pcm_dma_params *pa = rtd->dai->cpu_dai->private_data;
+ struct davinci_pcm_dma_params *params = &pa[substream->stream];
+ if (!params)
+ return -ENODEV;
snd_soc_set_runtime_hwparams(substream, &davinci_pcm_hardware);
/* ensure that buffer size is a multiple of period size */
@@ -257,6 +255,7 @@ static int davinci_pcm_open(struct snd_pcm_substream *substream)
return -ENOMEM;
spin_lock_init(&prtd->lock);
+ prtd->params = params;
runtime->private_data = prtd;
diff --git a/sound/soc/davinci/davinci-pcm.h b/sound/soc/davinci/davinci-pcm.h
index 63d96253c73..8746606efc8 100644
--- a/sound/soc/davinci/davinci-pcm.h
+++ b/sound/soc/davinci/davinci-pcm.h
@@ -17,7 +17,6 @@
struct davinci_pcm_dma_params {
- char *name; /* stream identifier */
int channel; /* sync dma channel ID */
unsigned short acnt;
dma_addr_t dma_addr; /* device physical address for DMA */
diff --git a/sound/soc/imx/mxc-ssi.c b/sound/soc/imx/mxc-ssi.c
index 3806ff2c0cd..ccdefe60e75 100644
--- a/sound/soc/imx/mxc-ssi.c
+++ b/sound/soc/imx/mxc-ssi.c
@@ -397,14 +397,6 @@ static int imx_ssi_set_dai_fmt(struct snd_soc_dai *cpu_dai,
break;
}
- /* sync */
- if (!(fmt & SND_SOC_DAIFMT_ASYNC))
- scr |= SSI_SCR_SYN;
-
- /* tdm - only for stereo atm */
- if (fmt & SND_SOC_DAIFMT_TDM)
- scr |= SSI_SCR_NET;
-
if (cpu_dai->id == IMX_DAI_SSI0 || cpu_dai->id == IMX_DAI_SSI2) {
SSI1_STCR = stcr;
SSI1_SRCR = srcr;
diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig
index 6375b4ea525..dcb3181bb34 100644
--- a/sound/soc/pxa/Kconfig
+++ b/sound/soc/pxa/Kconfig
@@ -138,7 +138,7 @@ config SND_PXA2XX_SOC_MIOA701
config SND_PXA2XX_SOC_IMOTE2
tristate "SoC Audio support for IMote 2"
- depends on SND_PXA2XX_SOC && MACH_INTELMOTE2
+ depends on SND_PXA2XX_SOC && MACH_INTELMOTE2 && I2C
select SND_PXA2XX_SOC_I2S
select SND_SOC_WM8940
help
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index f79711b9fa5..8de6f9dec4a 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -524,7 +524,7 @@ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget)
/* connected jack or spk ? */
if (widget->id == snd_soc_dapm_hp || widget->id == snd_soc_dapm_spk ||
- widget->id == snd_soc_dapm_line)
+ (widget->id == snd_soc_dapm_line && !list_empty(&widget->sources)))
return 1;
}
@@ -573,7 +573,8 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget)
return 1;
/* connected jack ? */
- if (widget->id == snd_soc_dapm_mic || widget->id == snd_soc_dapm_line)
+ if (widget->id == snd_soc_dapm_mic ||
+ (widget->id == snd_soc_dapm_line && !list_empty(&widget->sinks)))
return 1;
}
diff --git a/sound/usb/caiaq/audio.c b/sound/usb/caiaq/audio.c
index 121af0644fd..86b2c3b92df 100644
--- a/sound/usb/caiaq/audio.c
+++ b/sound/usb/caiaq/audio.c
@@ -62,10 +62,14 @@ static void
activate_substream(struct snd_usb_caiaqdev *dev,
struct snd_pcm_substream *sub)
{
+ spin_lock(&dev->spinlock);
+
if (sub->stream == SNDRV_PCM_STREAM_PLAYBACK)
dev->sub_playback[sub->number] = sub;
else
dev->sub_capture[sub->number] = sub;
+
+ spin_unlock(&dev->spinlock);
}
static void
@@ -269,16 +273,22 @@ snd_usb_caiaq_pcm_pointer(struct snd_pcm_substream *sub)
{
int index = sub->number;
struct snd_usb_caiaqdev *dev = snd_pcm_substream_chip(sub);
+ snd_pcm_uframes_t ptr;
+
+ spin_lock(&dev->spinlock);
if (dev->input_panic || dev->output_panic)
- return SNDRV_PCM_POS_XRUN;
+ ptr = SNDRV_PCM_POS_XRUN;
if (sub->stream == SNDRV_PCM_STREAM_PLAYBACK)
- return bytes_to_frames(sub->runtime,
+ ptr = bytes_to_frames(sub->runtime,
dev->audio_out_buf_pos[index]);
else
- return bytes_to_frames(sub->runtime,
+ ptr = bytes_to_frames(sub->runtime,
dev->audio_in_buf_pos[index]);
+
+ spin_unlock(&dev->spinlock);
+ return ptr;
}
/* operators for both playback and capture */
diff --git a/sound/usb/caiaq/device.c b/sound/usb/caiaq/device.c
index 83e6c1312d4..a3f02dd9744 100644
--- a/sound/usb/caiaq/device.c
+++ b/sound/usb/caiaq/device.c
@@ -35,7 +35,7 @@
#include "input.h"
MODULE_AUTHOR("Daniel Mack <daniel@caiaq.de>");
-MODULE_DESCRIPTION("caiaq USB audio, version 1.3.19");
+MODULE_DESCRIPTION("caiaq USB audio, version 1.3.20");
MODULE_LICENSE("GPL");
MODULE_SUPPORTED_DEVICE("{{Native Instruments, RigKontrol2},"
"{Native Instruments, RigKontrol3},"
diff --git a/sound/usb/usbmixer.c b/sound/usb/usbmixer.c
index ab5a3ac2ac4..9efcfd08d74 100644
--- a/sound/usb/usbmixer.c
+++ b/sound/usb/usbmixer.c
@@ -898,6 +898,11 @@ static struct snd_kcontrol_new usb_feature_unit_ctl = {
* build a feature control
*/
+static size_t append_ctl_name(struct snd_kcontrol *kctl, const char *str)
+{
+ return strlcat(kctl->id.name, str, sizeof(kctl->id.name));
+}
+
static void build_feature_ctl(struct mixer_build *state, unsigned char *desc,
unsigned int ctl_mask, int control,
struct usb_audio_term *iterm, int unitid)
@@ -978,13 +983,13 @@ static void build_feature_ctl(struct mixer_build *state, unsigned char *desc,
*/
if (! mapped_name && ! (state->oterm.type >> 16)) {
if ((state->oterm.type & 0xff00) == 0x0100) {
- len = strlcat(kctl->id.name, " Capture", sizeof(kctl->id.name));
+ len = append_ctl_name(kctl, " Capture");
} else {
- len = strlcat(kctl->id.name + len, " Playback", sizeof(kctl->id.name));
+ len = append_ctl_name(kctl, " Playback");
}
}
- strlcat(kctl->id.name + len, control == USB_FEATURE_MUTE ? " Switch" : " Volume",
- sizeof(kctl->id.name));
+ append_ctl_name(kctl, control == USB_FEATURE_MUTE ?
+ " Switch" : " Volume");
if (control == USB_FEATURE_VOLUME) {
kctl->tlv.c = mixer_vol_tlv;
kctl->vd[0].access |=
@@ -1143,7 +1148,7 @@ static void build_mixer_unit_ctl(struct mixer_build *state, unsigned char *desc,
len = get_term_name(state, iterm, kctl->id.name, sizeof(kctl->id.name), 0);
if (! len)
len = sprintf(kctl->id.name, "Mixer Source %d", in_ch + 1);
- strlcat(kctl->id.name + len, " Volume", sizeof(kctl->id.name));
+ append_ctl_name(kctl, " Volume");
snd_printdd(KERN_INFO "[%d] MU [%s] ch = %d, val = %d/%d\n",
cval->id, kctl->id.name, cval->channels, cval->min, cval->max);
@@ -1400,8 +1405,8 @@ static int build_audio_procunit(struct mixer_build *state, int unitid, unsigned
if (! len)
strlcpy(kctl->id.name, name, sizeof(kctl->id.name));
}
- strlcat(kctl->id.name, " ", sizeof(kctl->id.name));
- strlcat(kctl->id.name, valinfo->suffix, sizeof(kctl->id.name));
+ append_ctl_name(kctl, " ");
+ append_ctl_name(kctl, valinfo->suffix);
snd_printdd(KERN_INFO "[%d] PU [%s] ch = %d, val = %d/%d\n",
cval->id, kctl->id.name, cval->channels, cval->min, cval->max);
@@ -1610,9 +1615,9 @@ static int parse_audio_selector_unit(struct mixer_build *state, int unitid, unsi
strlcpy(kctl->id.name, "USB", sizeof(kctl->id.name));
if ((state->oterm.type & 0xff00) == 0x0100)
- strlcat(kctl->id.name, " Capture Source", sizeof(kctl->id.name));
+ append_ctl_name(kctl, " Capture Source");
else
- strlcat(kctl->id.name, " Playback Source", sizeof(kctl->id.name));
+ append_ctl_name(kctl, " Playback Source");
}
snd_printdd(KERN_INFO "[%d] SU [%s] items = %d\n",
diff --git a/sound/usb/usx2y/us122l.c b/sound/usb/usx2y/us122l.c
index 99f33766cd5..00cd54c236b 100644
--- a/sound/usb/usx2y/us122l.c
+++ b/sound/usb/usx2y/us122l.c
@@ -66,6 +66,28 @@ static int us122l_create_usbmidi(struct snd_card *card)
iface, &quirk);
}
+static int us144_create_usbmidi(struct snd_card *card)
+{
+ static struct snd_usb_midi_endpoint_info quirk_data = {
+ .out_ep = 4,
+ .in_ep = 3,
+ .out_cables = 0x001,
+ .in_cables = 0x001
+ };
+ static struct snd_usb_audio_quirk quirk = {
+ .vendor_name = "US144",
+ .product_name = NAME_ALLCAPS,
+ .ifnum = 0,
+ .type = QUIRK_MIDI_US122L,
+ .data = &quirk_data
+ };
+ struct usb_device *dev = US122L(card)->chip.dev;
+ struct usb_interface *iface = usb_ifnum_to_if(dev, 0);
+
+ return snd_usb_create_midi_interface(&US122L(card)->chip,
+ iface, &quirk);
+}
+
/*
* Wrapper for usb_control_msg().
* Allocates a temp buffer to prevent dmaing from/to the stack.
@@ -171,6 +193,11 @@ static int usb_stream_hwdep_open(struct snd_hwdep *hw, struct file *file)
if (!us122l->first)
us122l->first = file;
+
+ if (us122l->chip.dev->descriptor.idProduct == USB_ID_US144) {
+ iface = usb_ifnum_to_if(us122l->chip.dev, 0);
+ usb_autopm_get_interface(iface);
+ }
iface = usb_ifnum_to_if(us122l->chip.dev, 1);
usb_autopm_get_interface(iface);
return 0;
@@ -179,8 +206,14 @@ static int usb_stream_hwdep_open(struct snd_hwdep *hw, struct file *file)
static int usb_stream_hwdep_release(struct snd_hwdep *hw, struct file *file)
{
struct us122l *us122l = hw->private_data;
- struct usb_interface *iface = usb_ifnum_to_if(us122l->chip.dev, 1);
+ struct usb_interface *iface;
snd_printdd(KERN_DEBUG "%p %p\n", hw, file);
+
+ if (us122l->chip.dev->descriptor.idProduct == USB_ID_US144) {
+ iface = usb_ifnum_to_if(us122l->chip.dev, 0);
+ usb_autopm_put_interface(iface);
+ }
+ iface = usb_ifnum_to_if(us122l->chip.dev, 1);
usb_autopm_put_interface(iface);
if (us122l->first == file)
us122l->first = NULL;
@@ -443,6 +476,13 @@ static bool us122l_create_card(struct snd_card *card)
int err;
struct us122l *us122l = US122L(card);
+ if (us122l->chip.dev->descriptor.idProduct == USB_ID_US144) {
+ err = usb_set_interface(us122l->chip.dev, 0, 1);
+ if (err) {
+ snd_printk(KERN_ERR "usb_set_interface error \n");
+ return false;
+ }
+ }
err = usb_set_interface(us122l->chip.dev, 1, 1);
if (err) {
snd_printk(KERN_ERR "usb_set_interface error \n");
@@ -455,7 +495,10 @@ static bool us122l_create_card(struct snd_card *card)
if (!us122l_start(us122l, 44100, 256))
return false;
- err = us122l_create_usbmidi(card);
+ if (us122l->chip.dev->descriptor.idProduct == USB_ID_US144)
+ err = us144_create_usbmidi(card);
+ else
+ err = us122l_create_usbmidi(card);
if (err < 0) {
snd_printk(KERN_ERR "us122l_create_usbmidi error %i \n", err);
us122l_stop(us122l);
@@ -542,6 +585,7 @@ static int us122l_usb_probe(struct usb_interface *intf,
return err;
}
+ usb_get_intf(usb_ifnum_to_if(device, 0));
usb_get_dev(device);
*cardp = card;
return 0;
@@ -550,9 +594,16 @@ static int us122l_usb_probe(struct usb_interface *intf,
static int snd_us122l_probe(struct usb_interface *intf,
const struct usb_device_id *id)
{
+ struct usb_device *device = interface_to_usbdev(intf);
struct snd_card *card;
int err;
+ if (device->descriptor.idProduct == USB_ID_US144
+ && device->speed == USB_SPEED_HIGH) {
+ snd_printk(KERN_ERR "disable ehci-hcd to run US-144 \n");
+ return -ENODEV;
+ }
+
snd_printdd(KERN_DEBUG"%p:%i\n",
intf, intf->cur_altsetting->desc.bInterfaceNumber);
if (intf->cur_altsetting->desc.bInterfaceNumber != 1)
@@ -591,7 +642,8 @@ static void snd_us122l_disconnect(struct usb_interface *intf)
snd_usbmidi_disconnect(p);
}
- usb_put_intf(intf);
+ usb_put_intf(usb_ifnum_to_if(us122l->chip.dev, 0));
+ usb_put_intf(usb_ifnum_to_if(us122l->chip.dev, 1));
usb_put_dev(us122l->chip.dev);
while (atomic_read(&us122l->mmap_count))
@@ -642,6 +694,13 @@ static int snd_us122l_resume(struct usb_interface *intf)
mutex_lock(&us122l->mutex);
/* needed, doesn't restart without: */
+ if (us122l->chip.dev->descriptor.idProduct == USB_ID_US144) {
+ err = usb_set_interface(us122l->chip.dev, 0, 1);
+ if (err) {
+ snd_printk(KERN_ERR "usb_set_interface error \n");
+ goto unlock;
+ }
+ }
err = usb_set_interface(us122l->chip.dev, 1, 1);
if (err) {
snd_printk(KERN_ERR "usb_set_interface error \n");
@@ -675,11 +734,11 @@ static struct usb_device_id snd_us122l_usb_id_table[] = {
.idVendor = 0x0644,
.idProduct = USB_ID_US122L
},
-/* { */ /* US-144 maybe works when @USB1.1. Untested. */
-/* .match_flags = USB_DEVICE_ID_MATCH_DEVICE, */
-/* .idVendor = 0x0644, */
-/* .idProduct = USB_ID_US144 */
-/* }, */
+ { /* US-144 only works at USB1.1! Disable module ehci-hcd. */
+ .match_flags = USB_DEVICE_ID_MATCH_DEVICE,
+ .idVendor = 0x0644,
+ .idProduct = USB_ID_US144
+ },
{ /* terminator */ }
};