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Commit 0200971d authored by Linus Torvalds's avatar Linus Torvalds
Browse files

Merge tag 'sound-fixes' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound 

sound fixes for 3.3-rc5

Just a collection of boring small fixes for ASoC, HD-audio Realtek
and USB-audio drivers.

* tag 'sound-fixes' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
  ALSA: snd-usb-caiaq: Fix the return of XRUN
  ASoC: ak4642: fixup HeadPhone L/R dapm settings
  ALSA: hda/realtek - Fix surround output regression on Acer Aspire 5935
  ALSA: hda/realtek - Fix overflow of vol/sw check bitmap
  ALSA: usb-audio: avoid integer overflow in create_fixed_stream_quirk()
  ASoC: wm8962: Fix sidetone enumeration texts
parents 45196cee cb74eb15

sound/pci/hda/patch_realtek.c

+16 −3
Original line number Original line Diff line number Diff line
@@ -80,6 +80,8 @@ enum { 
	ALC_AUTOMUTE_MIXER,	/* mute/unmute mixer widget AMP */

	ALC_AUTOMUTE_MIXER,	/* mute/unmute mixer widget AMP */

};

};





#define MAX_VOL_NIDS	0x40



struct alc_spec {

struct alc_spec {

	/* codec parameterization */

	/* codec parameterization */

	const struct snd_kcontrol_new *mixers[5];	/* mixer arrays */

	const struct snd_kcontrol_new *mixers[5];	/* mixer arrays */
@@ -118,8 +120,8 @@ struct alc_spec { 
	const hda_nid_t *capsrc_nids;

	const hda_nid_t *capsrc_nids;

	hda_nid_t dig_in_nid;		/* digital-in NID; optional */

	hda_nid_t dig_in_nid;		/* digital-in NID; optional */

	hda_nid_t mixer_nid;		/* analog-mixer NID */

	hda_nid_t mixer_nid;		/* analog-mixer NID */

	DECLARE_BITMAP(vol_ctls, 0x20 << 1);

	DECLARE_BITMAP(vol_ctls, MAX_VOL_NIDS << 1);

	DECLARE_BITMAP(sw_ctls, 0x20 << 1);

	DECLARE_BITMAP(sw_ctls, MAX_VOL_NIDS << 1);





	/* capture setup for dynamic dual-adc switch */

	/* capture setup for dynamic dual-adc switch */

	hda_nid_t cur_adc;

	hda_nid_t cur_adc;
@@ -3149,7 +3151,10 @@ static int alc_auto_fill_dac_nids(struct hda_codec *codec) 
static inline unsigned int get_ctl_pos(unsigned int data)

static inline unsigned int get_ctl_pos(unsigned int data)

{

{

	hda_nid_t nid = get_amp_nid_(data);

	hda_nid_t nid = get_amp_nid_(data);

	unsigned int dir = get_amp_direction_(data);

	unsigned int dir;

	if (snd_BUG_ON(nid >= MAX_VOL_NIDS))

		return 0;

	dir = get_amp_direction_(data);

	return (nid << 1) | dir;

	return (nid << 1) | dir;

}

}
@@ -4436,12 +4441,20 @@ static void alc889_fixup_dac_route(struct hda_codec *codec, 
				   const struct alc_fixup *fix, int action)

				   const struct alc_fixup *fix, int action)

{

{

	if (action == ALC_FIXUP_ACT_PRE_PROBE) {

	if (action == ALC_FIXUP_ACT_PRE_PROBE) {

		/* fake the connections during parsing the tree */

		hda_nid_t conn1[2] = { 0x0c, 0x0d };

		hda_nid_t conn1[2] = { 0x0c, 0x0d };

		hda_nid_t conn2[2] = { 0x0e, 0x0f };

		hda_nid_t conn2[2] = { 0x0e, 0x0f };

		snd_hda_override_conn_list(codec, 0x14, 2, conn1);

		snd_hda_override_conn_list(codec, 0x14, 2, conn1);

		snd_hda_override_conn_list(codec, 0x15, 2, conn1);

		snd_hda_override_conn_list(codec, 0x15, 2, conn1);

		snd_hda_override_conn_list(codec, 0x18, 2, conn2);

		snd_hda_override_conn_list(codec, 0x18, 2, conn2);

		snd_hda_override_conn_list(codec, 0x1a, 2, conn2);

		snd_hda_override_conn_list(codec, 0x1a, 2, conn2);

	} else if (action == ALC_FIXUP_ACT_PROBE) {

		/* restore the connections */

		hda_nid_t conn[5] = { 0x0c, 0x0d, 0x0e, 0x0f, 0x26 };

		snd_hda_override_conn_list(codec, 0x14, 5, conn);

		snd_hda_override_conn_list(codec, 0x15, 5, conn);

		snd_hda_override_conn_list(codec, 0x18, 5, conn);

		snd_hda_override_conn_list(codec, 0x1a, 5, conn);

	}

	}

}

}

sound/soc/codecs/ak4642.c

+16 −15
Original line number Original line Diff line number Diff line
@@ -146,13 +146,10 @@ static const struct snd_kcontrol_new ak4642_snd_controls[] = { 




	SOC_DOUBLE_R_TLV("Digital Playback Volume", L_DVC, R_DVC,

	SOC_DOUBLE_R_TLV("Digital Playback Volume", L_DVC, R_DVC,

			 0, 0xFF, 1, out_tlv),

			 0, 0xFF, 1, out_tlv),



	SOC_SINGLE("Headphone Switch", PW_MGMT2, 6, 1, 0),

};

};





static const struct snd_kcontrol_new ak4642_hpout_mixer_controls[] = {

static const struct snd_kcontrol_new ak4642_headphone_control =

	SOC_DAPM_SINGLE("DACH", MD_CTL4, 0, 1, 0),

	SOC_DAPM_SINGLE("Switch", PW_MGMT2, 6, 1, 0);

};





static const struct snd_kcontrol_new ak4642_lout_mixer_controls[] = {

static const struct snd_kcontrol_new ak4642_lout_mixer_controls[] = {

	SOC_DAPM_SINGLE("DACL", SG_SL1, 4, 1, 0),

	SOC_DAPM_SINGLE("DACL", SG_SL1, 4, 1, 0),
@@ -165,13 +162,12 @@ static const struct snd_soc_dapm_widget ak4642_dapm_widgets[] = { 
	SND_SOC_DAPM_OUTPUT("HPOUTR"),

	SND_SOC_DAPM_OUTPUT("HPOUTR"),

	SND_SOC_DAPM_OUTPUT("LINEOUT"),

	SND_SOC_DAPM_OUTPUT("LINEOUT"),





	SND_SOC_DAPM_MIXER("HPOUTL Mixer", PW_MGMT2, 5, 0,

	SND_SOC_DAPM_PGA("HPL Out", PW_MGMT2, 5, 0, NULL, 0),

			   &ak4642_hpout_mixer_controls[0],

	SND_SOC_DAPM_PGA("HPR Out", PW_MGMT2, 4, 0, NULL, 0),

			   ARRAY_SIZE(ak4642_hpout_mixer_controls)),

	SND_SOC_DAPM_SWITCH("Headphone Enable", SND_SOC_NOPM, 0, 0,

			    &ak4642_headphone_control),





	SND_SOC_DAPM_MIXER("HPOUTR Mixer", PW_MGMT2, 4, 0,

	SND_SOC_DAPM_PGA("DACH", MD_CTL4, 0, 0, NULL, 0),

			   &ak4642_hpout_mixer_controls[0],

			   ARRAY_SIZE(ak4642_hpout_mixer_controls)),





	SND_SOC_DAPM_MIXER("LINEOUT Mixer", PW_MGMT1, 3, 0,

	SND_SOC_DAPM_MIXER("LINEOUT Mixer", PW_MGMT1, 3, 0,

			   &ak4642_lout_mixer_controls[0],

			   &ak4642_lout_mixer_controls[0],
@@ -184,12 +180,17 @@ static const struct snd_soc_dapm_widget ak4642_dapm_widgets[] = { 
static const struct snd_soc_dapm_route ak4642_intercon[] = {

static const struct snd_soc_dapm_route ak4642_intercon[] = {





	/* Outputs */

	/* Outputs */

	{"HPOUTL", NULL, "HPOUTL Mixer"},

	{"HPOUTL", NULL, "HPL Out"},

	{"HPOUTR", NULL, "HPOUTR Mixer"},

	{"HPOUTR", NULL, "HPR Out"},

	{"LINEOUT", NULL, "LINEOUT Mixer"},

	{"LINEOUT", NULL, "LINEOUT Mixer"},





	{"HPOUTL Mixer", "DACH", "DAC"},

	{"HPL Out", NULL, "Headphone Enable"},

	{"HPOUTR Mixer", "DACH", "DAC"},

	{"HPR Out", NULL, "Headphone Enable"},



	{"Headphone Enable", "Switch", "DACH"},



	{"DACH", NULL, "DAC"},



	{"LINEOUT Mixer", "DACL", "DAC"},

	{"LINEOUT Mixer", "DACL", "DAC"},

};

};

sound/soc/codecs/wm8962.c

+1 −1
Original line number Original line Diff line number Diff line
@@ -2564,7 +2564,7 @@ static int dsp2_event(struct snd_soc_dapm_widget *w, 
	return 0;

	return 0;

}

}





static const char *st_text[] = { "None", "Right", "Left" };

static const char *st_text[] = { "None", "Left", "Right" };





static const struct soc_enum str_enum =

static const struct soc_enum str_enum =

	SOC_ENUM_SINGLE(WM8962_DAC_DSP_MIXING_1, 2, 3, st_text);

	SOC_ENUM_SINGLE(WM8962_DAC_DSP_MIXING_1, 2, 3, st_text);

sound/usb/caiaq/audio.c

+4 −1
Original line number Original line Diff line number Diff line
@@ -311,8 +311,10 @@ snd_usb_caiaq_pcm_pointer(struct snd_pcm_substream *sub) 




	spin_lock(&dev->spinlock);

	spin_lock(&dev->spinlock);





	if (dev->input_panic || dev->output_panic)

	if (dev->input_panic || dev->output_panic) {

		ptr = SNDRV_PCM_POS_XRUN;

		ptr = SNDRV_PCM_POS_XRUN;

		goto unlock;

	}





	if (sub->stream == SNDRV_PCM_STREAM_PLAYBACK)

	if (sub->stream == SNDRV_PCM_STREAM_PLAYBACK)

		ptr = bytes_to_frames(sub->runtime,

		ptr = bytes_to_frames(sub->runtime,
@@ -321,6 +323,7 @@ snd_usb_caiaq_pcm_pointer(struct snd_pcm_substream *sub) 
		ptr = bytes_to_frames(sub->runtime,

		ptr = bytes_to_frames(sub->runtime,

					dev->audio_in_buf_pos[index]);

					dev->audio_in_buf_pos[index]);





unlock:

	spin_unlock(&dev->spinlock);

	spin_unlock(&dev->spinlock);

	return ptr;

	return ptr;

}

}

sound/usb/card.h

+1 −0
Original line number Original line Diff line number Diff line 
#ifndef __USBAUDIO_CARD_H

#ifndef __USBAUDIO_CARD_H

#define __USBAUDIO_CARD_H

#define __USBAUDIO_CARD_H





#define MAX_NR_RATES	1024

#define MAX_PACKS	20

#define MAX_PACKS	20

#define MAX_PACKS_HS	(MAX_PACKS * 8)	/* in high speed mode */

#define MAX_PACKS_HS	(MAX_PACKS * 8)	/* in high speed mode */

#define MAX_URBS	8

#define MAX_URBS	8