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Commit be3bfbba authored by Linus Torvalds's avatar Linus Torvalds
Browse files

Merge branch 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/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
  ...
parents 20272c89 7dc85076

include/sound/soc-dapm.h

+1 −0
Original line number Diff line number Diff line
@@ -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);

sound/oss/ac97_codec.c

+1 −1
Original line number Diff line number Diff line
@@ -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

sound/pci/ac97/ac97_patch.c

+1 −1
Original line number Diff line number Diff line
@@ -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.

 */

sound/pci/hda/patch_sigmatel.c

+36 −14
Original line number Diff line number Diff line
@@ -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) {

sound/soc/at91/Kconfig

+0 −17
Original line number Diff line number Diff line
@@ -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.