Merge tag 'sound-3.11' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound fixes from Takashi Iwai:
"This batch contains a few USB audio fixes, a couple of HD-audio
quirks, various small ASoC driver fixes in addition to an ASoC core
fix that may lead to memory corruption.

Unfortunately slightly more volume than the previous pull request, but
all are reasonable regression fixes"

* tag 'sound-3.11' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ALSA: hda - Add a fixup for Gateway LT27
ASoC: tegra: fix Tegra30 I2S capture parameter setup
ALSA: usb-audio: Fix invalid volume resolution for Logitech HD Webcam C525
ALSA: hda - Fix missing mute controls for CX5051
ALSA: usb-audio: fix automatic Roland/Yamaha MIDI detection
ALSA: 6fire: make buffers DMA-able (midi)
ALSA: 6fire: make buffers DMA-able (pcm)
ALSA: hda - Add pinfix for LG LW25 laptop
ASoC: cs42l52: Add new TLV for Beep Volume
ASoC: cs42l52: Reorder Min/Max and update to SX_TLV for Beep Volume
ASoC: dapm: Fix empty list check in dapm_new_mux()
ASoC: sgtl5000: fix buggy 'Capture Attenuate Switch' control
ASoC: sgtl5000: prevent playback to be muted when terminating concurrent capture

sound/pci/hda/hda_generic.c

···
 }
 
 #define nid_has_mute(codec, nid, dir) \
-	check_amp_caps(codec, nid, dir, AC_AMPCAP_MUTE)
+	check_amp_caps(codec, nid, dir, (AC_AMPCAP_MUTE | AC_AMPCAP_MIN_MUTE))
 #define nid_has_volume(codec, nid, dir) \
 	check_amp_caps(codec, nid, dir, AC_AMPCAP_NUM_STEPS)
 
···
 		if (enable)
 			val = (caps & AC_AMPCAP_OFFSET) >> AC_AMPCAP_OFFSET_SHIFT;
 	}
-	if (caps & AC_AMPCAP_MUTE) {
+	if (caps & (AC_AMPCAP_MUTE | AC_AMPCAP_MIN_MUTE)) {
 		if (!enable)
 			val |= HDA_AMP_MUTE;
 	}
···
 {
 	unsigned int mask = 0xff;
 
-	if (caps & AC_AMPCAP_MUTE) {
+	if (caps & (AC_AMPCAP_MUTE | AC_AMPCAP_MIN_MUTE)) {
 		if (is_ctl_associated(codec, nid, dir, idx, NID_PATH_MUTE_CTL))
 			mask &= ~0x80;
 	}

sound/pci/hda/patch_realtek.c

···
 	ALC880_FIXUP_GPIO2,
 	ALC880_FIXUP_MEDION_RIM,
 	ALC880_FIXUP_LG,
+	ALC880_FIXUP_LG_LW25,
 	ALC880_FIXUP_W810,
 	ALC880_FIXUP_EAPD_COEF,
 	ALC880_FIXUP_TCL_S700,
···
 			{ 0x16, 0x411111f0 },
 			{ 0x18, 0x411111f0 },
 			{ 0x1a, 0x411111f0 },
+			{ }
+		}
+	},
+	[ALC880_FIXUP_LG_LW25] = {
+		.type = HDA_FIXUP_PINS,
+		.v.pins = (const struct hda_pintbl[]) {
+			{ 0x1a, 0x0181344f }, /* line-in */
+			{ 0x1b, 0x0321403f }, /* headphone */
 			{ }
 		}
 	},
···
 	SND_PCI_QUIRK(0x1854, 0x003b, "LG", ALC880_FIXUP_LG),
 	SND_PCI_QUIRK(0x1854, 0x005f, "LG P1 Express", ALC880_FIXUP_LG),
 	SND_PCI_QUIRK(0x1854, 0x0068, "LG w1", ALC880_FIXUP_LG),
+	SND_PCI_QUIRK(0x1854, 0x0077, "LG LW25", ALC880_FIXUP_LG_LW25),
 	SND_PCI_QUIRK(0x19db, 0x4188, "TCL S700", ALC880_FIXUP_TCL_S700),
 
 	/* Below is the copied entries from alc880_quirks.c.
···
 	SND_PCI_QUIRK(0x1025, 0x0308, "Acer Aspire 8942G", ALC662_FIXUP_ASPIRE),
 	SND_PCI_QUIRK(0x1025, 0x031c, "Gateway NV79", ALC662_FIXUP_SKU_IGNORE),
 	SND_PCI_QUIRK(0x1025, 0x0349, "eMachines eM250", ALC662_FIXUP_INV_DMIC),
+	SND_PCI_QUIRK(0x1025, 0x034a, "Gateway LT27", ALC662_FIXUP_INV_DMIC),
 	SND_PCI_QUIRK(0x1025, 0x038b, "Acer Aspire 8943G", ALC662_FIXUP_ASPIRE),
 	SND_PCI_QUIRK(0x1028, 0x05d8, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1028, 0x05db, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE),

sound/soc/codecs/cs42l52.c

···
 
 static DECLARE_TLV_DB_SCALE(mix_tlv, -50, 50, 0);
 
+static DECLARE_TLV_DB_SCALE(beep_tlv, -56, 200, 0);
+
 static const unsigned int limiter_tlv[] = {
 	TLV_DB_RANGE_HEAD(2),
 	0, 2, TLV_DB_SCALE_ITEM(-3000, 600, 0),
···
 	SOC_ENUM("Beep Pitch", beep_pitch_enum),
 	SOC_ENUM("Beep on Time", beep_ontime_enum),
 	SOC_ENUM("Beep off Time", beep_offtime_enum),
-	SOC_SINGLE_TLV("Beep Volume", CS42L52_BEEP_VOL, 0, 0x1f, 0x07, hl_tlv),
+	SOC_SINGLE_SX_TLV("Beep Volume", CS42L52_BEEP_VOL,
+			0, 0x07, 0x1f, beep_tlv),
 	SOC_SINGLE("Beep Mixer Switch", CS42L52_BEEP_TONE_CTL, 5, 1, 1),
 	SOC_ENUM("Beep Treble Corner Freq", beep_treble_enum),
 	SOC_ENUM("Beep Bass Corner Freq", beep_bass_enum),

sound/soc/codecs/sgtl5000.c

···
 static int power_vag_event(struct snd_soc_dapm_widget *w,
 	struct snd_kcontrol *kcontrol, int event)
 {
+	const u32 mask = SGTL5000_DAC_POWERUP | SGTL5000_ADC_POWERUP;
+
 	switch (event) {
 	case SND_SOC_DAPM_POST_PMU:
 		snd_soc_update_bits(w->codec, SGTL5000_CHIP_ANA_POWER,
···
 		break;
 
 	case SND_SOC_DAPM_PRE_PMD:
-		snd_soc_update_bits(w->codec, SGTL5000_CHIP_ANA_POWER,
-			SGTL5000_VAG_POWERUP, 0);
-		msleep(400);
+		/*
+		 * Don't clear VAG_POWERUP, when both DAC and ADC are
+		 * operational to prevent inadvertently starving the
+		 * other one of them.
+		 */
+		if ((snd_soc_read(w->codec, SGTL5000_CHIP_ANA_POWER) &
+				mask) != mask) {
+			snd_soc_update_bits(w->codec, SGTL5000_CHIP_ANA_POWER,
+				SGTL5000_VAG_POWERUP, 0);
+			msleep(400);
+		}
 		break;
 	default:
 		break;
···
 	SOC_DOUBLE("Capture Volume", SGTL5000_CHIP_ANA_ADC_CTRL, 0, 4, 0xf, 0),
 	SOC_SINGLE_TLV("Capture Attenuate Switch (-6dB)",
 			SGTL5000_CHIP_ANA_ADC_CTRL,
-			8, 2, 0, capture_6db_attenuate),
+			8, 1, 0, capture_6db_attenuate),
 	SOC_SINGLE("Capture ZC Switch", SGTL5000_CHIP_ANA_CTRL, 1, 1, 0),
 
 	SOC_DOUBLE_TLV("Headphone Playback Volume",

sound/soc/soc-dapm.c

···
 		return -EINVAL;
 	}
 
-	path = list_first_entry(&w->sources, struct snd_soc_dapm_path,
-				list_sink);
-	if (!path) {
+	if (list_empty(&w->sources)) {
 		dev_err(dapm->dev, "ASoC: mux %s has no paths\n", w->name);
 		return -EINVAL;
 	}
+
+	path = list_first_entry(&w->sources, struct snd_soc_dapm_path,
+				list_sink);
 
 	ret = dapm_create_or_share_mixmux_kcontrol(w, 0, path);
 	if (ret < 0)

sound/soc/tegra/tegra30_i2s.c

···
 		reg = TEGRA30_I2S_CIF_RX_CTRL;
 	} else {
 		val |= TEGRA30_AUDIOCIF_CTRL_DIRECTION_TX;
-		reg = TEGRA30_I2S_CIF_RX_CTRL;
+		reg = TEGRA30_I2S_CIF_TX_CTRL;
 	}
 
 	regmap_write(i2s->regmap, reg, val);

sound/usb/6fire/midi.c

···
 #include "chip.h"
 #include "comm.h"
 
+enum {
+	MIDI_BUFSIZE = 64
+};
+
 static void usb6fire_midi_out_handler(struct urb *urb)
 {
 	struct midi_runtime *rt = urb->context;
···
 	if (!rt)
 		return -ENOMEM;
 
+	rt->out_buffer = kzalloc(MIDI_BUFSIZE, GFP_KERNEL);
+	if (!rt->out_buffer) {
+		kfree(rt);
+		return -ENOMEM;
+	}
+
 	rt->chip = chip;
 	rt->in_received = usb6fire_midi_in_received;
 	rt->out_buffer[0] = 0x80; /* 'send midi' command */
···
 
 	ret = snd_rawmidi_new(chip->card, "6FireUSB", 0, 1, 1, &rt->instance);
 	if (ret < 0) {
+		kfree(rt->out_buffer);
 		kfree(rt);
 		snd_printk(KERN_ERR PREFIX "unable to create midi.\n");
 		return ret;
···
 
 void usb6fire_midi_destroy(struct sfire_chip *chip)
 {
-	kfree(chip->midi);
+	struct midi_runtime *rt = chip->midi;
+
+	kfree(rt->out_buffer);
+	kfree(rt);
 	chip->midi = NULL;
 }

sound/usb/6fire/midi.h

···
 
 #include "common.h"
 
-enum {
-	MIDI_BUFSIZE = 64
-};
-
 struct midi_runtime {
 	struct sfire_chip *chip;
 	struct snd_rawmidi *instance;
···
 	struct snd_rawmidi_substream *out;
 	struct urb out_urb;
 	u8 out_serial; /* serial number of out packet */
-	u8 out_buffer[MIDI_BUFSIZE];
+	u8 *out_buffer;
 	int buffer_offset;
 
 	void (*in_received)(struct midi_runtime *rt, u8 *data, int length);

sound/usb/6fire/pcm.c

···
 	urb->instance.number_of_packets = PCM_N_PACKETS_PER_URB;
 }
 
+static int usb6fire_pcm_buffers_init(struct pcm_runtime *rt)
+{
+	int i;
+
+	for (i = 0; i < PCM_N_URBS; i++) {
+		rt->out_urbs[i].buffer = kzalloc(PCM_N_PACKETS_PER_URB
+				* PCM_MAX_PACKET_SIZE, GFP_KERNEL);
+		if (!rt->out_urbs[i].buffer)
+			return -ENOMEM;
+		rt->in_urbs[i].buffer = kzalloc(PCM_N_PACKETS_PER_URB
+				* PCM_MAX_PACKET_SIZE, GFP_KERNEL);
+		if (!rt->in_urbs[i].buffer)
+			return -ENOMEM;
+	}
+	return 0;
+}
+
+static void usb6fire_pcm_buffers_destroy(struct pcm_runtime *rt)
+{
+	int i;
+
+	for (i = 0; i < PCM_N_URBS; i++) {
+		kfree(rt->out_urbs[i].buffer);
+		kfree(rt->in_urbs[i].buffer);
+	}
+}
+
 int usb6fire_pcm_init(struct sfire_chip *chip)
 {
 	int i;
···
 
 	if (!rt)
 		return -ENOMEM;
+
+	ret = usb6fire_pcm_buffers_init(rt);
+	if (ret) {
+		usb6fire_pcm_buffers_destroy(rt);
+		kfree(rt);
+		return ret;
+	}
 
 	rt->chip = chip;
 	rt->stream_state = STREAM_DISABLED;
···
 
 	ret = snd_pcm_new(chip->card, "DMX6FireUSB", 0, 1, 1, &pcm);
 	if (ret < 0) {
+		usb6fire_pcm_buffers_destroy(rt);
 		kfree(rt);
 		snd_printk(KERN_ERR PREFIX "cannot create pcm instance.\n");
 		return ret;
···
 	snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &pcm_ops);
 
 	if (ret) {
+		usb6fire_pcm_buffers_destroy(rt);
 		kfree(rt);
 		snd_printk(KERN_ERR PREFIX
 				"error preallocating pcm buffers.\n");
···
 
 void usb6fire_pcm_destroy(struct sfire_chip *chip)
 {
-	kfree(chip->pcm);
+	struct pcm_runtime *rt = chip->pcm;
+
+	usb6fire_pcm_buffers_destroy(rt);
+	kfree(rt);
 	chip->pcm = NULL;
 }

sound/usb/6fire/pcm.h

···
 	struct urb instance;
 	struct usb_iso_packet_descriptor packets[PCM_N_PACKETS_PER_URB];
 	/* END DO NOT SEPARATE */
-	u8 buffer[PCM_N_PACKETS_PER_URB * PCM_MAX_PACKET_SIZE];
+	u8 *buffer;
 
 	struct pcm_urb *peer;
 };

sound/usb/mixer.c

···
 	case USB_ID(0x046d, 0x081b): /* HD Webcam c310 */
 	case USB_ID(0x046d, 0x081d): /* HD Webcam c510 */
 	case USB_ID(0x046d, 0x0825): /* HD Webcam c270 */
+	case USB_ID(0x046d, 0x0826): /* HD Webcam c525 */
 	case USB_ID(0x046d, 0x0991):
 	/* Most audio usb devices lie about volume resolution.
 	 * Most Logitech webcams have res = 384.

sound/usb/quirks.c

···
 	if (altsd->bNumEndpoints < 1)
 		return -ENODEV;
 	epd = get_endpoint(alts, 0);
-	if (!usb_endpoint_xfer_bulk(epd) ||
+	if (!usb_endpoint_xfer_bulk(epd) &&
 	    !usb_endpoint_xfer_int(epd))
 		return -ENODEV;
 
 	switch (USB_ID_VENDOR(chip->usb_id)) {
 	case 0x0499: /* Yamaha */
 		err = create_yamaha_midi_quirk(chip, iface, driver, alts);
-		if (err < 0 && err != -ENODEV)
+		if (err != -ENODEV)
 			return err;
 		break;
 	case 0x0582: /* Roland */
 		err = create_roland_midi_quirk(chip, iface, driver, alts);
-		if (err < 0 && err != -ENODEV)
+		if (err != -ENODEV)
 			return err;
 		break;
 	}