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

* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6: (33 commits)
  ALSA: ASoC codec: remove unused #include <version.h>
  ALSA: ASoC: update email address for Liam Girdwood
  ALSA: hda: corrected invalid mixer values
  ALSA: hda: add mixers for analog mixer on 92hd75xx codecs
  ALSA: ASoC: Add destination and source port for DMA on OMAP1
  ALSA: ASoC: Drop device registration from GTA01 lm4857 driver
  ALSA: ASoC: Fix build of GTA01 audio driver
  ALSA: ASoC: Add widgets before setting endpoints on GTA01
  ALSA: ASoC: Fix inverted input PGA mute bits in WM8903
  ALSA: ASoC: OMAP: Set DMA stream name at runtime in McBSP DAI driver
  ALSA: ASoC: OMAP: Add support for OMAP2430 and OMAP34xx in McBSP DAI driver
  ALSA: ASoC: OMAP: Add multilink support to McBSP DAI driver
  ALSA: ASoC: Make TLV320AIC26 user-visible
  ALSA: ASoC - clean up Kconfig for TLV320AIC2
  ALSA: ASoC: Make WM8510 microphone input a DAPM mixer
  ALSA: ASoC: Implement WM8510 bias level control
  ALSA: ASoC: Remove unused AUDIO_NAME define from codec drivers
  ALSA: ASoC: tlv320aic3x: Use uniform tlv320aic naming
  ALSA: ASoC: Add WM8510 SPI support
  ALSA: ASoC: Add WM8753 SPI support
  ...
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Linus Torvalds committed Oct 13, 2008
2 parents 20272c8 + 7dc8507 commit be3bfbb
Showing 56 changed files with 2,037 additions and 601 deletions.
1 change: 1 addition & 0 deletions include/sound/soc-dapm.h
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Original file line number Diff line number Diff line change
@@ -240,6 +240,7 @@ int snd_soc_dapm_sys_add(struct device *dev);
/* dapm audio pin control and status */
int snd_soc_dapm_enable_pin(struct snd_soc_codec *codec, char *pin);
int snd_soc_dapm_disable_pin(struct snd_soc_codec *codec, char *pin);
int snd_soc_dapm_nc_pin(struct snd_soc_codec *codec, char *pin);
int snd_soc_dapm_get_pin_status(struct snd_soc_codec *codec, char *pin);
int snd_soc_dapm_sync(struct snd_soc_codec *codec);

2 changes: 1 addition & 1 deletion sound/oss/ac97_codec.c
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Original file line number Diff line number Diff line change
@@ -30,7 +30,7 @@
**************************************************************************
*
* History
* May 02, 2003 Liam Girdwood <liam.girdwood@wolfsonmicro.com>
* May 02, 2003 Liam Girdwood <lrg@slimlogic.co.uk>
* Removed non existant WM9700
* Added support for WM9705, WM9708, WM9709, WM9710, WM9711
* WM9712 and WM9717
2 changes: 1 addition & 1 deletion sound/pci/ac97/ac97_patch.c
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Original file line number Diff line number Diff line change
@@ -476,7 +476,7 @@ static int patch_yamaha_ymf753(struct snd_ac97 * ac97)
}

/*
* May 2, 2003 Liam Girdwood <liam.girdwood@wolfsonmicro.com>
* May 2, 2003 Liam Girdwood <lrg@slimlogic.co.uk>
* removed broken wolfson00 patch.
* added support for WM9705,WM9708,WM9709,WM9710,WM9711,WM9712 and WM9717.
*/
50 changes: 36 additions & 14 deletions sound/pci/hda/patch_sigmatel.c
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Original file line number Diff line number Diff line change
@@ -322,8 +322,8 @@ static hda_nid_t stac92hd71bxx_mux_nids[2] = {
0x1a, 0x1b
};

static hda_nid_t stac92hd71bxx_dmux_nids[1] = {
0x1c,
static hda_nid_t stac92hd71bxx_dmux_nids[2] = {
0x1c, 0x1d,
};

static hda_nid_t stac92hd71bxx_smux_nids[2] = {
@@ -861,20 +861,18 @@ static struct hda_verb stac92hd71bxx_core_init[] = {
{ 0x28, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff},
/* connect headphone jack to dac1 */
{ 0x0a, AC_VERB_SET_CONNECT_SEL, 0x01},
{ 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, /* Speaker */
/* unmute right and left channels for nodes 0x0a, 0xd, 0x0f */
{ 0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{ 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{ 0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
};

#define HD_DISABLE_PORTF 3
#define HD_DISABLE_PORTF 2
static struct hda_verb stac92hd71bxx_analog_core_init[] = {
/* start of config #1 */

/* connect port 0f to audio mixer */
{ 0x0f, AC_VERB_SET_CONNECT_SEL, 0x2},
{ 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, /* Speaker */
/* unmute right and left channels for node 0x0f */
{ 0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
/* start of config #2 */
@@ -883,10 +881,6 @@ static struct hda_verb stac92hd71bxx_analog_core_init[] = {
{ 0x28, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff},
/* connect headphone jack to dac1 */
{ 0x0a, AC_VERB_SET_CONNECT_SEL, 0x01},
/* connect port 0d to audio mixer */
{ 0x0d, AC_VERB_SET_CONNECT_SEL, 0x2},
/* unmute dac0 input in audio mixer */
{ 0x17, AC_VERB_SET_AMP_GAIN_MUTE, 0x701f},
/* unmute right and left channels for nodes 0x0a, 0xd */
{ 0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{ 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
@@ -1107,6 +1101,7 @@ static struct snd_kcontrol_new stac92hd83xxx_mixer[] = {

static struct snd_kcontrol_new stac92hd71bxx_analog_mixer[] = {
STAC_INPUT_SOURCE(2),
STAC_ANALOG_LOOPBACK(0xFA0, 0x7A0, 2),

HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x1c, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x1c, 0x0, HDA_OUTPUT),
@@ -1119,8 +1114,17 @@ static struct snd_kcontrol_new stac92hd71bxx_analog_mixer[] = {
HDA_CODEC_MUTE("PC Beep Switch", 0x17, 0x2, HDA_INPUT),
*/

HDA_CODEC_MUTE("Analog Loopback 1", 0x17, 0x3, HDA_INPUT),
HDA_CODEC_MUTE("Analog Loopback 2", 0x17, 0x4, HDA_INPUT),
HDA_CODEC_MUTE("Import0 Mux Capture Switch", 0x17, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME("Import0 Mux Capture Volume", 0x17, 0x0, HDA_INPUT),

HDA_CODEC_MUTE("Import1 Mux Capture Switch", 0x17, 0x1, HDA_INPUT),
HDA_CODEC_VOLUME("Import1 Mux Capture Volume", 0x17, 0x1, HDA_INPUT),

HDA_CODEC_MUTE("DAC0 Capture Switch", 0x17, 0x3, HDA_INPUT),
HDA_CODEC_VOLUME("DAC0 Capture Volume", 0x17, 0x3, HDA_INPUT),

HDA_CODEC_MUTE("DAC1 Capture Switch", 0x17, 0x4, HDA_INPUT),
HDA_CODEC_VOLUME("DAC1 Capture Volume", 0x17, 0x4, HDA_INPUT),
{ } /* end */
};

@@ -1649,7 +1653,7 @@ static struct snd_pci_quirk stac92hd83xxx_cfg_tbl[] = {

static unsigned int ref92hd71bxx_pin_configs[11] = {
0x02214030, 0x02a19040, 0x01a19020, 0x01014010,
0x0181302e, 0x01114010, 0x01019020, 0x90a000f0,
0x0181302e, 0x01014010, 0x01019020, 0x90a000f0,
0x90a000f0, 0x01452050, 0x01452050,
};

@@ -3000,7 +3004,7 @@ static int stac92xx_auto_create_mono_output_ctls(struct hda_codec *codec)

/* labels for amp mux outputs */
static const char *stac92xx_amp_labels[3] = {
"Front Microphone", "Microphone", "Line In"
"Front Microphone", "Microphone", "Line In",
};

/* create amp out controls mux on capable codecs */
@@ -4327,6 +4331,16 @@ static struct hda_codec_ops stac92hd71bxx_patch_ops = {
#endif
};

static struct hda_input_mux stac92hd71bxx_dmux = {
.num_items = 4,
.items = {
{ "Analog Inputs", 0x00 },
{ "Mixer", 0x01 },
{ "Digital Mic 1", 0x02 },
{ "Digital Mic 2", 0x03 },
}
};

static int patch_stac92hd71bxx(struct hda_codec *codec)
{
struct sigmatel_spec *spec;
@@ -4341,6 +4355,8 @@ static int patch_stac92hd71bxx(struct hda_codec *codec)
spec->num_pins = ARRAY_SIZE(stac92hd71bxx_pin_nids);
spec->num_pwrs = ARRAY_SIZE(stac92hd71bxx_pwr_nids);
spec->pin_nids = stac92hd71bxx_pin_nids;
memcpy(&spec->private_dimux, &stac92hd71bxx_dmux,
sizeof(stac92hd71bxx_dmux));
spec->board_config = snd_hda_check_board_config(codec,
STAC_92HD71BXX_MODELS,
stac92hd71bxx_models,
@@ -4392,6 +4408,7 @@ static int patch_stac92hd71bxx(struct hda_codec *codec)
/* no output amps */
spec->num_pwrs = 0;
spec->mixer = stac92hd71bxx_analog_mixer;
spec->dinput_mux = &spec->private_dimux;

/* disable VSW */
spec->init = &stac92hd71bxx_analog_core_init[HD_DISABLE_PORTF];
@@ -4409,12 +4426,13 @@ static int patch_stac92hd71bxx(struct hda_codec *codec)
spec->num_pwrs = 0;
/* fallthru */
default:
spec->dinput_mux = &spec->private_dimux;
spec->mixer = stac92hd71bxx_analog_mixer;
spec->init = stac92hd71bxx_analog_core_init;
codec->slave_dig_outs = stac92hd71bxx_slave_dig_outs;
}

spec->aloopback_mask = 0x20;
spec->aloopback_mask = 0x50;
spec->aloopback_shift = 0;

if (spec->board_config > STAC_92HD71BXX_REF) {
@@ -4456,6 +4474,10 @@ static int patch_stac92hd71bxx(struct hda_codec *codec)
spec->multiout.num_dacs = 1;
spec->multiout.hp_nid = 0x11;
spec->multiout.dac_nids = stac92hd71bxx_dac_nids;
if (spec->dinput_mux)
spec->private_dimux.num_items +=
spec->num_dmics -
(ARRAY_SIZE(stac92hd71bxx_dmic_nids) - 1);

err = stac92xx_parse_auto_config(codec, 0x21, 0x23);
if (!err) {
17 changes: 0 additions & 17 deletions sound/soc/at91/Kconfig
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Original file line number Diff line number Diff line change
@@ -8,20 +8,3 @@ config SND_AT91_SOC

config SND_AT91_SOC_SSC
tristate

config SND_AT91_SOC_ETI_B1_WM8731
tristate "SoC Audio support for WM8731-based Endrelia ETI-B1 boards"
depends on SND_AT91_SOC && (MACH_ETI_B1 || MACH_ETI_C1)
select SND_AT91_SOC_SSC
select SND_SOC_WM8731
help
Say Y if you want to add support for SoC audio on WM8731-based
Endrelia Technologies Inc ETI-B1 or ETI-C1 boards.

config SND_AT91_SOC_ETI_SLAVE
bool "Run codec in slave Mode on Endrelia boards"
depends on SND_AT91_SOC_ETI_B1_WM8731
default n
help
Say Y if you want to run with the AT91 SSC generating the BCLK
and LRC signals on Endrelia boards.
5 changes: 0 additions & 5 deletions sound/soc/at91/Makefile
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Original file line number Diff line number Diff line change
@@ -4,8 +4,3 @@ snd-soc-at91-ssc-objs := at91-ssc.o

obj-$(CONFIG_SND_AT91_SOC) += snd-soc-at91.o
obj-$(CONFIG_SND_AT91_SOC_SSC) += snd-soc-at91-ssc.o

# AT91 Machine Support
snd-soc-eti-b1-wm8731-objs := eti_b1_wm8731.o

obj-$(CONFIG_SND_AT91_SOC_ETI_B1_WM8731) += snd-soc-eti-b1-wm8731.o
2 changes: 1 addition & 1 deletion sound/soc/at91/at91-ssc.c
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Original file line number Diff line number Diff line change
@@ -5,7 +5,7 @@
* Endrelia Technologies Inc.
*
* Based on pxa2xx Platform drivers by
* Liam Girdwood <liam.girdwood@wolfsonmicro.com>
* Liam Girdwood <lrg@slimlogic.co.uk>
*
* 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
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