Merge commit 'v3.0-rc5' into perf/core

Merge reason: Pick up the latest fixes.

Signed-off-by: Ingo Molnar <mingo@elte.hu>

16 files changed, 76 insertions, 39 deletions

diff --git a/sound/firewire/isight.c b/sound/firewire/isight.c
index 86ee16ca365e..440030818db7 100644
--- a/sound/firewire/isight.c
+++ b/sound/firewire/isight.c
@@ -209,6 +209,7 @@ static void isight_packet(struct fw_iso_context *context, u32 cycle,
 		isight->packet_index = -1;
 		return;
 	}
+	fw_iso_context_queue_flush(isight->context);
 
 	if (++index >= QUEUE_LENGTH)
 		index = 0;
diff --git a/sound/pci/asihpi/asihpi.c b/sound/pci/asihpi/asihpi.c
index 2ca6f4f85b41..e3569bdd3b64 100644
--- a/sound/pci/asihpi/asihpi.c
+++ b/sound/pci/asihpi/asihpi.c
@@ -27,7 +27,6 @@
 #include "hpioctl.h"
 
 #include <linux/pci.h>
-#include <linux/version.h>
 #include <linux/init.h>
 #include <linux/jiffies.h>
 #include <linux/slab.h>
diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c
index 5e619a84da06..15f0161ce4a2 100644
--- a/sound/pci/emu10k1/emu10k1_main.c
+++ b/sound/pci/emu10k1/emu10k1_main.c
@@ -1440,6 +1440,14 @@ static struct snd_emu_chip_details emu_chip_details[] = {
 	 .ca0102_chip = 1,
 	 .spk71 = 1,
 	 .emu_model = EMU_MODEL_EMU0404}, /* EMU 0404 */
+	/* EMU0404 PCIe */
+	{.vendor = 0x1102, .device = 0x0008, .subsystem = 0x40051102,
+	 .driver = "Audigy2", .name = "E-mu 0404 PCIe [MAEM8984]",
+	 .id = "EMU0404",
+	 .emu10k2_chip = 1,
+	 .ca0108_chip = 1,
+	 .spk71 = 1,
+	 .emu_model = EMU_MODEL_EMU0404}, /* EMU 0404 PCIe ver_03 */
 	/* Note that all E-mu cards require kernel 2.6 or newer. */
 	{.vendor = 0x1102, .device = 0x0008,
 	 .driver = "Audigy2", .name = "SB Audigy 2 Value [Unknown]",
diff --git a/sound/pci/hda/hda_beep.h b/sound/pci/hda/hda_beep.h
index f1de1bac042c..55f0647458c7 100644
--- a/sound/pci/hda/hda_beep.h
+++ b/sound/pci/hda/hda_beep.h
@@ -50,7 +50,12 @@ int snd_hda_enable_beep_device(struct hda_codec *codec, int enable);
 int snd_hda_attach_beep_device(struct hda_codec *codec, int nid);
 void snd_hda_detach_beep_device(struct hda_codec *codec);
 #else
-#define snd_hda_attach_beep_device(...)		0
-#define snd_hda_detach_beep_device(...)
+static inline int snd_hda_attach_beep_device(struct hda_codec *codec, int nid)
+{
+	return 0;
+}
+static inline void snd_hda_detach_beep_device(struct hda_codec *codec)
+{
+}
 #endif
 #endif
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 43fcfbd32847..d21191dcfe88 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -4883,7 +4883,6 @@ static const struct snd_pci_quirk alc880_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x1025, 0xe309, "ULI", ALC880_3ST_DIG),
 	SND_PCI_QUIRK(0x1025, 0xe310, "ULI", ALC880_3ST),
 	SND_PCI_QUIRK(0x1039, 0x1234, NULL, ALC880_6ST_DIG),
-	SND_PCI_QUIRK(0x103c, 0x2a09, "HP", ALC880_5ST),
 	SND_PCI_QUIRK(0x1043, 0x10b3, "ASUS W1V", ALC880_ASUS_W1V),
 	SND_PCI_QUIRK(0x1043, 0x10c2, "ASUS W6A", ALC880_ASUS_DIG),
 	SND_PCI_QUIRK(0x1043, 0x10c3, "ASUS Wxx", ALC880_ASUS_DIG),
@@ -12600,6 +12599,7 @@ static const struct hda_verb alc262_toshiba_rx1_unsol_verbs[] = {
  */
 enum {
 	PINFIX_FSC_H270,
+	PINFIX_HP_Z200,
 };
 
 static const struct alc_fixup alc262_fixups[] = {
@@ -12612,9 +12612,17 @@ static const struct alc_fixup alc262_fixups[] = {
 			{ }
 		}
 	},
+	[PINFIX_HP_Z200] = {
+		.type = ALC_FIXUP_PINS,
+		.v.pins = (const struct alc_pincfg[]) {
+			{ 0x16, 0x99130120 }, /* internal speaker */
+			{ }
+		}
+	},
 };
 
 static const struct snd_pci_quirk alc262_fixup_tbl[] = {
+	SND_PCI_QUIRK(0x103c, 0x170b, "HP Z200", PINFIX_HP_Z200),
 	SND_PCI_QUIRK(0x1734, 0x1147, "FSC Celsius H270", PINFIX_FSC_H270),
 	{}
 };
@@ -12731,6 +12739,8 @@ static const struct snd_pci_quirk alc262_cfg_tbl[] = {
 			   ALC262_HP_BPC),
 	SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x1500, "HP z series",
 			   ALC262_HP_BPC),
+	SND_PCI_QUIRK(0x103c, 0x170b, "HP Z200",
+			   ALC262_AUTO),
 	SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x1700, "HP xw series",
 			   ALC262_HP_BPC),
 	SND_PCI_QUIRK(0x103c, 0x2800, "HP D7000", ALC262_HP_BPC_D7000_WL),
@@ -13316,9 +13326,8 @@ static void alc268_acer_lc_setup(struct hda_codec *codec)
 	struct alc_spec *spec = codec->spec;
 	spec->autocfg.hp_pins[0] = 0x15;
 	spec->autocfg.speaker_pins[0] = 0x14;
-	spec->automute_mixer_nid[0] = 0x0f;
 	spec->automute = 1;
-	spec->automute_mode = ALC_AUTOMUTE_MIXER;
+	spec->automute_mode = ALC_AUTOMUTE_AMP;
 	spec->ext_mic.pin = 0x18;
 	spec->ext_mic.mux_idx = 0;
 	spec->int_mic.pin = 0x12;
@@ -13873,7 +13882,6 @@ static const struct snd_pci_quirk alc268_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x1043, 0x1205, "ASUS W7J", ALC268_3ST),
 	SND_PCI_QUIRK(0x1170, 0x0040, "ZEPTO", ALC268_ZEPTO),
 	SND_PCI_QUIRK(0x14c0, 0x0025, "COMPAL IFL90/JFL-92", ALC268_TOSHIBA),
-	SND_PCI_QUIRK(0x152d, 0x0763, "Diverse (CPR2000)", ALC268_ACER),
 	SND_PCI_QUIRK(0x152d, 0x0771, "Quanta IL1", ALC267_QUANTA_IL1),
 	{}
 };
diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c
index 605c99e1e520..f43bb0eaed8b 100644
--- a/sound/pci/hda/patch_via.c
+++ b/sound/pci/hda/patch_via.c
@@ -745,12 +745,23 @@ static int via_independent_hp_put(struct snd_kcontrol *kcontrol,
 	struct via_spec *spec = codec->spec;
 	hda_nid_t nid = kcontrol->private_value;
 	unsigned int pinsel = ucontrol->value.enumerated.item[0];
+	unsigned int parm0, parm1;
 	/* Get Independent Mode index of headphone pin widget */
 	spec->hp_independent_mode = spec->hp_independent_mode_index == pinsel
 		? 1 : 0;
-	if (spec->codec_type == VT1718S)
+	if (spec->codec_type == VT1718S) {
 		snd_hda_codec_write(codec, nid, 0,
 				    AC_VERB_SET_CONNECT_SEL, pinsel ? 2 : 0);
+		/* Set correct mute switch for MW3 */
+		parm0 = spec->hp_independent_mode ?
+			       AMP_IN_UNMUTE(0) : AMP_IN_MUTE(0);
+		parm1 = spec->hp_independent_mode ?
+			       AMP_IN_MUTE(1) : AMP_IN_UNMUTE(1);
+		snd_hda_codec_write(codec, 0x1b, 0,
+				    AC_VERB_SET_AMP_GAIN_MUTE, parm0);
+		snd_hda_codec_write(codec, 0x1b, 0,
+				    AC_VERB_SET_AMP_GAIN_MUTE, parm1);
+	}
 	else
 		snd_hda_codec_write(codec, nid, 0,
 				    AC_VERB_SET_CONNECT_SEL, pinsel);
@@ -832,10 +843,13 @@ static int via_hp_build(struct hda_codec *codec)
 	knew->subdevice = HDA_SUBDEV_NID_FLAG | nid;
 	knew->private_value = nid;
 
-	knew = via_clone_control(spec, &via_hp_mixer[1]);
-	if (knew == NULL)
-		return -ENOMEM;
-	knew->subdevice = side_mute_channel(spec);
+	nid = side_mute_channel(spec);
+	if (nid) {
+		knew = via_clone_control(spec, &via_hp_mixer[1]);
+		if (knew == NULL)
+			return -ENOMEM;
+		knew->subdevice = nid;
+	}
 
 	return 0;
 }
@@ -4280,9 +4294,6 @@ static const struct hda_verb vt1718S_volume_init_verbs[] = {
 	{0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
 	{0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
 	{0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(5)},
-
-	/* Setup default input of Front HP to MW9 */
-	{0x28, AC_VERB_SET_CONNECT_SEL, 0x1},
 	/* PW9 PW10 Output enable */
 	{0x2d, AC_VERB_SET_PIN_WIDGET_CONTROL, AC_PINCTL_OUT_EN},
 	{0x2e, AC_VERB_SET_PIN_WIDGET_CONTROL, AC_PINCTL_OUT_EN},
@@ -4291,10 +4302,10 @@ static const struct hda_verb vt1718S_volume_init_verbs[] = {
 	/* Enable Boost Volume backdoor */
 	{0x1, 0xf88, 0x8},
 	/* MW0/1/2/3/4: un-mute index 0 (AOWx), mute index 1 (MW9) */
-	{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+	{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
 	{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
 	{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-	{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+	{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
 	{0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
 	{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
 	{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
@@ -4304,8 +4315,6 @@ static const struct hda_verb vt1718S_volume_init_verbs[] = {
 	/* set MUX1 = 2 (AOW4), MUX2 = 1 (AOW3) */
 	{0x34, AC_VERB_SET_CONNECT_SEL, 0x2},
 	{0x35, AC_VERB_SET_CONNECT_SEL, 0x1},
-	/* Unmute MW4's index 0 */
-	{0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
 	{ }
 };
 
@@ -4453,6 +4462,19 @@ static int vt1718S_auto_create_multi_out_ctls(struct via_spec *spec,
 			if (err < 0)
 				return err;
 		} else if (i == AUTO_SEQ_FRONT) {
+			/* add control to mixer index 0 */
+			err = via_add_control(spec, VIA_CTL_WIDGET_VOL,
+					      "Master Front Playback Volume",
+					      HDA_COMPOSE_AMP_VAL(0x21, 3, 5,
+								  HDA_INPUT));
+			if (err < 0)
+				return err;
+			err = via_add_control(spec, VIA_CTL_WIDGET_MUTE,
+					      "Master Front Playback Switch",
+					      HDA_COMPOSE_AMP_VAL(0x21, 3, 5,
+								  HDA_INPUT));
+			if (err < 0)
+				return err;
 			/* Front */
 			sprintf(name, "%s Playback Volume", chname[i]);
 			err = via_add_control(
diff --git a/sound/pci/lola/lola.c b/sound/pci/lola/lola.c
index 34b24286d279..2692e5ae5f2d 100644
--- a/sound/pci/lola/lola.c
+++ b/sound/pci/lola/lola.c
@@ -445,7 +445,7 @@ static void lola_reset_setups(struct lola *chip)
 	lola_setup_all_analog_gains(chip, PLAY, false); /* output, update */
 }
 
-static int lola_parse_tree(struct lola *chip)
+static int __devinit lola_parse_tree(struct lola *chip)
 {
 	unsigned int val;
 	int nid, err;
diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c
index 949691a876d3..3f08afc0f0d3 100644
--- a/sound/pci/rme9652/hdspm.c
+++ b/sound/pci/rme9652/hdspm.c
@@ -521,6 +521,7 @@ MODULE_SUPPORTED_DEVICE("{{RME HDSPM-MADI}}");
 #define HDSPM_DMA_AREA_KILOBYTES (HDSPM_DMA_AREA_BYTES/1024)
 
 /* revisions >= 230 indicate AES32 card */
+#define HDSPM_MADI_OLD_REV	207
 #define HDSPM_MADI_REV		210
 #define HDSPM_RAYDAT_REV	211
 #define HDSPM_AIO_REV		212
@@ -1143,7 +1144,7 @@ static int hdspm_external_sample_rate(struct hdspm *hdspm)
 
 		/* if wordclock has synced freq and wordclock is valid */
 		if ((status2 & HDSPM_wcLock) != 0 &&
-				(status & HDSPM_SelSyncRef0) == 0) {
+				(status2 & HDSPM_SelSyncRef0) == 0) {
 
 			rate_bits = status2 & HDSPM_wcFreqMask;
 
@@ -1639,12 +1640,14 @@ static int snd_hdspm_midi_input_read (struct hdspm_midi *hmidi)
 		}
 	}
 	hmidi->pending = 0;
+	spin_unlock_irqrestore(&hmidi->lock, flags);
 
+	spin_lock_irqsave(&hmidi->hdspm->lock, flags);
 	hmidi->hdspm->control_register |= hmidi->ie;
 	hdspm_write(hmidi->hdspm, HDSPM_controlRegister,
 		    hmidi->hdspm->control_register);
+	spin_unlock_irqrestore(&hmidi->hdspm->lock, flags);
 
-	spin_unlock_irqrestore (&hmidi->lock, flags);
 	return snd_hdspm_midi_output_write (hmidi);
 }
 
@@ -6377,6 +6380,7 @@ static int __devinit snd_hdspm_create(struct snd_card *card,
 
 	switch (hdspm->firmware_rev) {
 	case HDSPM_MADI_REV:
+	case HDSPM_MADI_OLD_REV:
 		hdspm->io_type = MADI;
 		hdspm->card_name = "RME MADI";
 		hdspm->midiPorts = 3;
diff --git a/sound/soc/codecs/wm8991.c b/sound/soc/codecs/wm8991.c
index 3c2ee1bb73cd..6af23d06870f 100644
--- a/sound/soc/codecs/wm8991.c
+++ b/sound/soc/codecs/wm8991.c
@@ -13,7 +13,6 @@
 
 #include <linux/module.h>
 #include <linux/moduleparam.h>
-#include <linux/version.h>
 #include <linux/kernel.h>
 #include <linux/init.h>
 #include <linux/delay.h>
diff --git a/sound/soc/imx/Kconfig b/sound/soc/imx/Kconfig
index d8f130d39dd9..bb699bb55a50 100644
--- a/sound/soc/imx/Kconfig
+++ b/sound/soc/imx/Kconfig
@@ -11,9 +11,6 @@ menuconfig SND_IMX_SOC
 
 if SND_IMX_SOC
 
-config SND_MXC_SOC_SSI
-	tristate
-
 config SND_MXC_SOC_FIQ
 	tristate
 
@@ -24,7 +21,6 @@ config SND_MXC_SOC_WM1133_EV1
 	tristate "Audio on the the i.MX31ADS with WM1133-EV1 fitted"
 	depends on MACH_MX31ADS_WM1133_EV1 && EXPERIMENTAL
 	select SND_SOC_WM8350
-	select SND_MXC_SOC_SSI
 	select SND_MXC_SOC_FIQ
 	help
 	  Enable support for audio on the i.MX31ADS with the WM1133-EV1
@@ -34,7 +30,6 @@ config SND_SOC_MX27VIS_AIC32X4
 	tristate "SoC audio support for Visstrim M10 boards"
 	depends on MACH_IMX27_VISSTRIM_M10
 	select SND_SOC_TVL320AIC32X4
-	select SND_MXC_SOC_SSI
 	select SND_MXC_SOC_MX2
 	help
 	  Say Y if you want to add support for SoC audio on Visstrim SM10
@@ -44,7 +39,6 @@ config SND_SOC_PHYCORE_AC97
 	tristate "SoC Audio support for Phytec phyCORE (and phyCARD) boards"
 	depends on MACH_PCM043 || MACH_PCA100
 	select SND_SOC_WM9712
-	select SND_MXC_SOC_SSI
 	select SND_MXC_SOC_FIQ
 	help
 	  Say Y if you want to add support for SoC audio on Phytec phyCORE
@@ -57,7 +51,6 @@ config SND_SOC_EUKREA_TLV320
 		|| MACH_EUKREA_MBIMXSD35_BASEBOARD \
 		|| MACH_EUKREA_MBIMXSD51_BASEBOARD
 	select SND_SOC_TLV320AIC23
-	select SND_MXC_SOC_SSI
 	select SND_MXC_SOC_FIQ
 	help
 	  Enable I2S based access to the TLV320AIC23B codec attached
diff --git a/sound/soc/imx/imx-pcm-dma-mx2.c b/sound/soc/imx/imx-pcm-dma-mx2.c
index aab7765f401a..4173b3d87f97 100644
--- a/sound/soc/imx/imx-pcm-dma-mx2.c
+++ b/sound/soc/imx/imx-pcm-dma-mx2.c
@@ -337,3 +337,5 @@ static void __exit snd_imx_pcm_exit(void)
 	platform_driver_unregister(&imx_pcm_driver);
 }
 module_exit(snd_imx_pcm_exit);
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:imx-pcm-audio");
diff --git a/sound/soc/imx/imx-ssi.c b/sound/soc/imx/imx-ssi.c
index 5b13feca7537..61fceb09cdb5 100644
--- a/sound/soc/imx/imx-ssi.c
+++ b/sound/soc/imx/imx-ssi.c
@@ -774,4 +774,4 @@ module_exit(imx_ssi_exit);
 MODULE_AUTHOR("Sascha Hauer, <s.hauer@pengutronix.de>");
 MODULE_DESCRIPTION("i.MX I2S/ac97 SoC Interface");
 MODULE_LICENSE("GPL");
-
+MODULE_ALIAS("platform:imx-ssi");
diff --git a/sound/soc/pxa/pxa2xx-pcm.c b/sound/soc/pxa/pxa2xx-pcm.c
index 2ce0b2d891d5..fab20a54e863 100644
--- a/sound/soc/pxa/pxa2xx-pcm.c
+++ b/sound/soc/pxa/pxa2xx-pcm.c
@@ -95,14 +95,14 @@ static int pxa2xx_soc_pcm_new(struct snd_card *card, struct snd_soc_dai *dai,
 	if (!card->dev->coherent_dma_mask)
 		card->dev->coherent_dma_mask = DMA_BIT_MASK(32);
 
-	if (dai->driver->playback.channels_min) {
+	if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) {
 		ret = pxa2xx_pcm_preallocate_dma_buffer(pcm,
 			SNDRV_PCM_STREAM_PLAYBACK);
 		if (ret)
 			goto out;
 	}
 
-	if (dai->driver->capture.channels_min) {
+	if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) {
 		ret = pxa2xx_pcm_preallocate_dma_buffer(pcm,
 			SNDRV_PCM_STREAM_CAPTURE);
 		if (ret)
diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c
index c005ceb70c9d..039b9532b270 100644
--- a/sound/soc/soc-cache.c
+++ b/sound/soc/soc-cache.c
@@ -409,9 +409,6 @@ int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec,
 	codec->bulk_write_raw = snd_soc_hw_bulk_write_raw;
 
 	switch (control) {
-	case SND_SOC_CUSTOM:
-		break;
-
 	case SND_SOC_I2C:
 #if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE))
 		codec->hw_write = (hw_write_t)i2c_master_send;
diff --git a/sound/usb/6fire/firmware.c b/sound/usb/6fire/firmware.c
index a91719d5918b..1e3ae3327dd3 100644
--- a/sound/usb/6fire/firmware.c
+++ b/sound/usb/6fire/firmware.c
@@ -270,7 +270,6 @@ static int usb6fire_fw_ezusb_upload(
 	data = 0x00; /* resume ezusb cpu */
 	ret = usb6fire_fw_ezusb_write(device, 0xa0, 0xe600, &data, 1);
 	if (ret < 0) {
-		release_firmware(fw);
 		snd_printk(KERN_ERR PREFIX "unable to upload ezusb "
 				"firmware %s: end message.\n", fwname);
 		return ret;
diff --git a/sound/usb/6fire/pcm.c b/sound/usb/6fire/pcm.c
index b137b25865cc..d144cdb2f159 100644
--- a/sound/usb/6fire/pcm.c
+++ b/sound/usb/6fire/pcm.c
@@ -395,12 +395,12 @@ static int usb6fire_pcm_open(struct snd_pcm_substream *alsa_sub)
 	alsa_rt->hw = pcm_hw;
 
 	if (alsa_sub->stream == SNDRV_PCM_STREAM_PLAYBACK) {
-		if (rt->rate >= 0)
+		if (rt->rate < ARRAY_SIZE(rates))
 			alsa_rt->hw.rates = rates_alsaid[rt->rate];
 		alsa_rt->hw.channels_max = OUT_N_CHANNELS;
 		sub = &rt->playback;
 	} else if (alsa_sub->stream == SNDRV_PCM_STREAM_CAPTURE) {
-		if (rt->rate >= 0)
+		if (rt->rate < ARRAY_SIZE(rates))
 			alsa_rt->hw.rates = rates_alsaid[rt->rate];
 		alsa_rt->hw.channels_max = IN_N_CHANNELS;
 		sub = &rt->capture;