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Date: Thu, 08 Oct 2009 13:08:21 +0200
From: Takashi Iwai <tiwai@...e.de>
To: Linus Torvalds <torvalds@...ux-foundation.org>
Cc: Andrew Morton <akpm@...ux-foundation.org>,
linux-kernel@...r.kernel.org
Subject: [GIT PULL] sound fixes for 2.6.32-rc4
Linus,
please pull fixes for sound subsystem for v2.6.32-rc4 from:
git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git for-linus
The shortlogs are below.
Thanks!
Takashi
===
Clemens Ladisch (1):
sound: via82xx: move DXS volume controls to PCM interface
Jonathan Cameron (1):
ASoC: wm8940: Fix check on error code form snd_soc_codec_set_cache_io
Krzysztof Helt (1):
ALSA: opl3: circular locking in the snd_opl3_note_on() and snd_opl3_note_off()
Mark Brown (2):
ASoC: Remove absent SYNC and TDM DAI format options from i.MX SSI
ASoC: WM8350 capture PGA mutes are inverted
Pavel Hofman (1):
ALSA: ICE1712/24 - Change the Multi Track Peak control (level meters) from MIXER to PCM type
Peter Ujfalusi (1):
ASoC: Fix SND_SOC_DAPM_LINE handling
Robert Hancock (1):
ALSA: ice1724: increase SPDIF and independent stereo buffer sizes
Takashi Iwai (4):
ALSA: hda - Fix invalid initializations for ALC861 auto mode
ALSA: hda - Add a workaround for ASUS A7K
ALSA: hda - Don't pick up invalid HP pins in alc_subsystem_id()
ALSA: hda - Fix yet another auto-mic bug in ALC268
---
sound/drivers/opl3/opl3_midi.c | 28 +++++++++----
sound/pci/hda/patch_realtek.c | 89 ++++++++++++++++++++++++++++++----------
sound/pci/ice1712/ice1712.c | 2 +-
sound/pci/ice1712/ice1724.c | 6 +-
sound/pci/via82xx.c | 27 ++++++++----
sound/soc/codecs/wm8350.c | 4 +-
sound/soc/codecs/wm8940.c | 2 +-
sound/soc/imx/mxc-ssi.c | 8 ----
sound/soc/soc-dapm.c | 5 +-
9 files changed, 115 insertions(+), 56 deletions(-)
diff --git a/sound/drivers/opl3/opl3_midi.c b/sound/drivers/opl3/opl3_midi.c
index 6e7d09a..7d722a0 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/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 7810d3d..470fd74 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -1332,15 +1332,20 @@ 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 +1367,7 @@ 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 +1375,14 @@ 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 +1390,14 @@ 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);
}
/*
@@ -9593,11 +9608,13 @@ 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),
{}
};
@@ -9869,7 +9886,7 @@ 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 */
@@ -12842,12 +12859,15 @@ 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 +12877,6 @@ 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],
@@ -14357,15 +14376,16 @@ 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]);
}
@@ -15158,7 +15178,7 @@ 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),
@@ -15551,6 +15571,29 @@ 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 +15615,8 @@ 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);
diff --git a/sound/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c
index cecf1ff..d74033a 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 af6e001..76b717d 100644
--- a/sound/pci/ice1712/ice1724.c
+++ b/sound/pci/ice1712/ice1724.c
@@ -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/via82xx.c b/sound/pci/via82xx.c
index acfa476..91683a3 100644
--- a/sound/pci/via82xx.c
+++ b/sound/pci/via82xx.c
@@ -1626,7 +1626,7 @@ 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 +1646,7 @@ 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 +1705,12 @@ 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,
+ .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),
- .count = 4,
.info = snd_via8233_dxs_volume_info,
.get = snd_via8233_dxs_volume_get,
.put = snd_via8233_dxs_volume_put,
@@ -1936,10 +1937,18 @@ 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;
+ }
}
}
/* select spdif data slot 10/11 */
diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c
index 3ff0373..593d5b9 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 da97aae..1ef2454 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/imx/mxc-ssi.c b/sound/soc/imx/mxc-ssi.c
index 3806ff2..ccdefe6 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/soc-dapm.c b/sound/soc/soc-dapm.c
index f79711b..8de6f9d 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;
}
--
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