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Merge branch 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6

Browse source 
* 'for-linus' of ssh://master.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6: (21 commits)
  ALSA: usb - Use strlcat() correctly
  ALSA: Fix invalid __exit in sound/mips/*.c
  ALSA: hda - Fix / improve ALC66x parser
  ALSA: ctxfi: Swapped SURROUND-SIDE mute
  sound: Make keywest_driver static
  ALSA: intel8x0 - Mute External Amplifier by default for Sony VAIO VGN-B1VP
  ALSA: hda - Fix digita/analog mic auto-switching with IDT codecs
  ASoC: fix kconfig order of Blackfin drivers
  ALSA: hda - Added quirk to enable sound on Toshiba NB200
  ASoC: Fix dependency of CONFIG_SND_PXA2XX_SOC_IMOTE2
  ALSA: Don't assume i2c device probing always succeeds
  ALSA: intel8x0 - Mute External Amplifier by default for Sony VAIO VGN-T350P
  ALSA: echoaudio - Re-enable the line-out control for the Mia card
  ALSA: hda - Resurrect input-source mixer of ALC268 model=acer
  ALSA: hda - Analog Devices AD1984A add HP Touchsmart model
  ALSA: hda - Add HP Pavilion dv4t-1300 to MSI whitelist
  ALSA: hda - CD-audio sound for hda-intel conexant benq laptop
  ASoC: DaVinci: Correct McASP FIFO initialization
  ASoC: Davinci: Fix race with cpu_dai->dma_data
  ASoC: DaVinci: Fix divide by zero error during 1st execution
  ...
This commit is contained in:
Linus Torvalds 2009-10-03 11:25:30 -07:00
commit f0a221ef47
24 changed files with 514 additions and 258 deletions
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1
Documentation/sound/alsa/HD-Audio-Models.txt
View file

@ -209,6 +209,7 @@ AD1884A / AD1883 / AD1984A / AD1984B
laptop laptop with HP jack sensing
mobile mobile devices with HP jack sensing
thinkpad Lenovo Thinkpad X300
touchsmart HP Touchsmart
AD1884
======

9
sound/aoa/codecs/tas.c
View file

@ -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.

2
sound/mips/hal2.c
View file

@ -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);

2
sound/mips/sgio2audio.c
View file

@ -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);

4
sound/pci/ctxfi/ctatc.c
View file

@ -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)

30
sound/pci/echoaudio/echoaudio.c
View file

@ -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)

1
sound/pci/echoaudio/mia.c
View file

@ -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 */

1
sound/pci/hda/hda_intel.c
View file

@ -2303,6 +2303,7 @@ static void __devinit check_probe_mask(struct azx *chip, int dev)
* white-list for enable_msi
*/
static struct snd_pci_quirk msi_white_list[] __devinitdata = {
SND_PCI_QUIRK(0x103c, 0x30f7, "HP Pavilion dv4t-1300", 1),
SND_PCI_QUIRK(0x103c, 0x3607, "HP Compa CQ40", 1),
{}
};

139
sound/pci/hda/patch_analog.c
View file

@ -4031,6 +4031,127 @@ static int ad1984a_thinkpad_init(struct hda_codec *codec)
return 0;
}
/*
* 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_codec_read(codec, 0x1c, 0,
AC_VERB_GET_PIN_SENSE, 0) & 0x80000000) {
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;
}
/*
*/
@ -4039,6 +4160,7 @@ enum {
AD1884A_LAPTOP,
AD1884A_MOBILE,
AD1884A_THINKPAD,
AD1984A_TOUCHSMART,
AD1884A_MODELS
};
@ -4047,6 +4169,7 @@ 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 +4182,7 @@ 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 +4266,21 @@ 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;

12
sound/pci/hda/patch_conexant.c
View file

@ -682,11 +682,13 @@ 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 },
}
};
@ -811,11 +813,19 @@ 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),
{}
};

248
sound/pci/hda/patch_realtek.c
View file

@ -12660,7 +12660,7 @@ 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 },
@ -16852,6 +16852,7 @@ 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 +17146,145 @@ static struct alc_config_preset alc662_presets[] = {
* BIOS auto configuration
*/
/* add playback controls from the parsed DAC table */
static int alc662_auto_create_multi_out_ctls(struct alc_spec *spec,
const struct auto_pin_cfg *cfg)
/* 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 int alc662_add_vol_ctl(struct alc_spec *spec, const char *pfx,
hda_nid_t nid, unsigned int chs)
{
char name[32];
sprintf(name, "%s Playback Volume", pfx);
return add_control(spec, ALC_CTL_WIDGET_VOL, name,
HDA_COMPOSE_AMP_VAL(nid, chs, 0, HDA_OUTPUT));
}
static int alc662_add_sw_ctl(struct alc_spec *spec, const char *pfx,
hda_nid_t nid, unsigned int chs)
{
char name[32];
sprintf(name, "%s Playback Switch", pfx);
return add_control(spec, ALC_CTL_WIDGET_MUTE, name,
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 hda_codec *codec,
const struct auto_pin_cfg *cfg)
{
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;
}
@ -17217,54 +17293,38 @@ 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 */
nid = alc662_look_for_dac(codec, pin);
if (!nid) {
char name[32];
/* the corresponding DAC is already occupied */
if (!(get_wcaps(codec, pin) & AC_WCAP_OUT_AMP))
return 0; /* no way */
/* create a switch only */
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 add_control(spec, ALC_CTL_WIDGET_MUTE, name,
HDA_COMPOSE_AMP_VAL(pin, 3, 0, HDA_OUTPUT));
}
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;
}
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;
err = alc662_add_sw_ctl(spec, pfx, mix, 3);
if (err < 0)
return err;
return nid;
}
/* create playback/capture controls for input pins */
@ -17273,30 +17333,35 @@ static int alc662_auto_create_extra_out(struct alc_spec *spec, hda_nid_t pin,
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 +17371,13 @@ 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 +17415,25 @@ 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;

20
sound/pci/hda/patch_sigmatel.c
View file

@ -182,8 +182,8 @@ 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 {
@ -3469,18 +3469,26 @@ 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;
}
@ -4557,11 +4565,11 @@ 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);

12
sound/pci/intel8x0.c
View file

@ -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,

14
sound/ppc/keywest.c
View file

@ -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",
},

98
sound/soc/blackfin/Kconfig
View file

@ -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.

8
sound/soc/blackfin/bf5xx-i2s.c
View file

@ -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,

8
sound/soc/blackfin/bf5xx-tdm.c
View file

@ -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,

37
sound/soc/davinci/davinci-i2s.c
View file

@ -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);

80
sound/soc/davinci/davinci-mcasp.c
View file

@ -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;

7
sound/soc/davinci/davinci-mcasp.h
View file

@ -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 */

13
sound/soc/davinci/davinci-pcm.c
View file

@ -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;

1
sound/soc/davinci/davinci-pcm.h
View file

@ -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 */

2
sound/soc/pxa/Kconfig
View file

@ -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

23
sound/usb/usbmixer.c
View file

@ -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",