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

* '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
...

Documentation/sound/alsa/HD-Audio-Models.txt

···
   laptop	laptop with HP jack sensing
   mobile	mobile devices with HP jack sensing
   thinkpad	Lenovo Thinkpad X300
+  touchsmart	HP Touchsmart
 
 AD1884
 ======

sound/aoa/codecs/tas.c

···
 	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.

sound/mips/hal2.c

···
 	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);
 

sound/mips/sgio2audio.c

···
 	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);
 

sound/pci/ctxfi/ctatc.c

···
 
 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)
···
 
 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)

sound/pci/echoaudio/echoaudio.c

···
 	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,
···
 	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,
···
 	.put = snd_echo_output_gain_put,
 	.tlv = {.p = db_scale_output_gain},
 };
-
 #endif
+
+#endif /* !ECHOCARD_HAS_VMIXER || ECHOCARD_HAS_LINE_OUT_GAIN */
 
 
 
···
 	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)

sound/pci/echoaudio/mia.c

···
 #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 */

sound/pci/hda/hda_intel.c

···
  * 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),
 	{}
 };

sound/pci/hda/patch_analog.c

···
 }
 
 /*
+ * 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;
+}
+
+
+/*
  */
 
 enum {
···
 	AD1884A_LAPTOP,
 	AD1884A_MOBILE,
 	AD1884A_THINKPAD,
+	AD1984A_TOUCHSMART,
 	AD1884A_MODELS
 };
 
···
 	[AD1884A_LAPTOP]	= "laptop",
 	[AD1884A_MOBILE]	= "mobile",
 	[AD1884A_THINKPAD]	= "thinkpad",
+	[AD1984A_TOUCHSMART]	= "touchsmart",
 };
 
 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),
 	{}
 };
 
···
 		spec->input_mux = &ad1984a_thinkpad_capture_source;
 		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;
 	}
 

sound/pci/hda/patch_conexant.c

···
 };
 
 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 },
 	}
 };
 
···
 };
 
 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),
 
 	{}
 };

sound/pci/hda/patch_realtek.c

···
 		.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 },
···
 	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),
···
  * 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;
-		nid = alc880_idx_to_dac(i);
+		mix = alc662_dac_to_mix(codec, cfg->line_out_pins[i], nid);
+		if (!mix)
+			continue;
 		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;
 		}
···
 }
 
 /* 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 */
···
 
 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]);
 	}
 }
 
···
 	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
···
 	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;

sound/pci/hda/patch_sigmatel.c

···
 
 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 {
···
 			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;
 }
···
 		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);

sound/pci/intel8x0.c

···
 		.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,

sound/ppc/keywest.c

···
 	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.
···
 	{ }
 };
 
-struct i2c_driver keywest_driver = {
+static struct i2c_driver keywest_driver = {
 	.driver = {
 		.name = "PMac Keywest Audio",
 	},

sound/soc/blackfin/Kconfig

···
 	  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
···
 	help
 	  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"
···
 	  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
 
···
 	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)
···
 	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.

sound/soc/blackfin/bf5xx-i2s.c

···
  * 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,

sound/soc/blackfin/bf5xx-tdm.c

···
  * 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,

sound/soc/davinci/davinci-i2s.c

···
 	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,
···
 	spcr &= playback ? ~DAVINCI_MCBSP_SPCR_XRST : ~DAVINCI_MCBSP_SPCR_RRST;
 	davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, spcr);
 	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
···
 				 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;
···
 #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,
···
 
 	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 */
···
 		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) {
···
 		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);

sound/soc/davinci/davinci-mcasp.c

···
 		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);
···
 
 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)
···
 
 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,
···
 {
 	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;
 
···
 }
 
 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,
···
 	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->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);
···
 		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) {
···
 		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]);
···
 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;
···
 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;
 
···
 	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;

sound/soc/davinci/davinci-mcasp.h

···
 };
 
 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 */

sound/soc/davinci/davinci-pcm.c

···
 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,
···
 	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 */
···
 		return -ENOMEM;
 
 	spin_lock_init(&prtd->lock);
+	prtd->params = params;
 
 	runtime->private_data = prtd;
 

sound/soc/davinci/davinci-pcm.h

···
 
 
 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 */

sound/soc/pxa/Kconfig

···
 
 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

sound/usb/usbmixer.c

···
  * 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)
···
 		 */
 		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 |= 
···
 		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);
···
 			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);
···
 			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",