Merge tag 'sound-3.13-rc2' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound fixes from Takashi Iwai:
 "Quite a few HD-Audio fixes, a WUSB audio fix and a fix for FireWire
  audio.  The HD-audio part contains a couple of fixes for the generic
  parser, and these are the only intrusive fixes.  The rest are mostly
  device-specific fixes"

* tag 'sound-3.13-rc2' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
  ALSA: hda - Add LFE chmap to ASUS ET2700
  ALSA: hda - Initialize missing bass speaker pin for ASUS AIO ET2700
  ALSA: hda - limit mic boost on Asus UX31[A,E]
  ALSA: hda - Check leaf nodes to find aamix amps
  ALSA: hda - Fix hp-mic mode without VREF bits
  ALSA: hda - Create Headhpone Mic Jack Mode when really needed
  ALSA: usb: use multiple packets per urb for Wireless USB inbound audio
  ALSA: hda - Enable mute/mic-mute LEDs for more Thinkpads with Conexant codec
  ALSA: hda - Drop bus->avoid_link_reset flag
  ALSA: hda/realtek - Set pcbeep amp for ALC668
  ALSA: hda/realtek - Add support of ALC231 codec
  ALSA: firewire-lib: fix wrong value for FDF field as an empty packet

8 files changed, 138 insertions, 40 deletions

diff --git a/sound/firewire/amdtp.c b/sound/firewire/amdtp.c
index d3226892ad6b..9048777228e2 100644
--- a/sound/firewire/amdtp.c
+++ b/sound/firewire/amdtp.c
@@ -434,17 +434,14 @@ static void queue_out_packet(struct amdtp_out_stream *s, unsigned int cycle)
 		return;
 	index = s->packet_index;
 
+	/* this module generate empty packet for 'no data' */
 	syt = calculate_syt(s, cycle);
-	if (!(s->flags & CIP_BLOCKING)) {
+	if (!(s->flags & CIP_BLOCKING))
 		data_blocks = calculate_data_blocks(s);
-	} else {
-		if (syt != 0xffff) {
-			data_blocks = s->syt_interval;
-		} else {
-			data_blocks = 0;
-			syt = 0xffffff;
-		}
-	}
+	else if (syt != 0xffff)
+		data_blocks = s->syt_interval;
+	else
+		data_blocks = 0;
 
 	buffer = s->buffer.packets[index].buffer;
 	buffer[0] = cpu_to_be32(ACCESS_ONCE(s->source_node_id_field) |
diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h
index 77db69480c19..7aa9870040c1 100644
--- a/sound/pci/hda/hda_codec.h
+++ b/sound/pci/hda/hda_codec.h
@@ -698,7 +698,6 @@ struct hda_bus {
 	unsigned int in_reset:1;	/* during reset operation */
 	unsigned int power_keep_link_on:1; /* don't power off HDA link */
 	unsigned int no_response_fallback:1; /* don't fallback at RIRB error */
-	unsigned int avoid_link_reset:1; /* don't reset link at runtime PM */
 
 	int primary_dig_out_type;	/* primary digital out PCM type */
 };
diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c
index 3067ed4fe3b2..c4671d00babd 100644
--- a/sound/pci/hda/hda_generic.c
+++ b/sound/pci/hda/hda_generic.c
@@ -2506,12 +2506,8 @@ static int create_out_jack_modes(struct hda_codec *codec, int num_pins,
 
 	for (i = 0; i < num_pins; i++) {
 		hda_nid_t pin = pins[i];
-		if (pin == spec->hp_mic_pin) {
-			int ret = create_hp_mic_jack_mode(codec, pin);
-			if (ret < 0)
-				return ret;
+		if (pin == spec->hp_mic_pin)
 			continue;
-		}
 		if (get_out_jack_num_items(codec, pin) > 1) {
 			struct snd_kcontrol_new *knew;
 			char name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN];
@@ -2764,7 +2760,7 @@ static int hp_mic_jack_mode_put(struct snd_kcontrol *kcontrol,
 			val &= ~(AC_PINCTL_VREFEN | PIN_HP);
 			val |= get_vref_idx(vref_caps, idx) | PIN_IN;
 		} else
-			val = snd_hda_get_default_vref(codec, nid);
+			val = snd_hda_get_default_vref(codec, nid) | PIN_IN;
 	}
 	snd_hda_set_pin_ctl_cache(codec, nid, val);
 	call_hp_automute(codec, NULL);
@@ -2784,9 +2780,6 @@ static int create_hp_mic_jack_mode(struct hda_codec *codec, hda_nid_t pin)
 	struct hda_gen_spec *spec = codec->spec;
 	struct snd_kcontrol_new *knew;
 
-	if (get_out_jack_num_items(codec, pin) <= 1 &&
-	    get_in_jack_num_items(codec, pin) <= 1)
-		return 0; /* no need */
 	knew = snd_hda_gen_add_kctl(spec, "Headphone Mic Jack Mode",
 				    &hp_mic_jack_mode_enum);
 	if (!knew)
@@ -2815,6 +2808,42 @@ static int add_loopback_list(struct hda_gen_spec *spec, hda_nid_t mix, int idx)
 	return 0;
 }
 
+/* return true if either a volume or a mute amp is found for the given
+ * aamix path; the amp has to be either in the mixer node or its direct leaf
+ */
+static bool look_for_mix_leaf_ctls(struct hda_codec *codec, hda_nid_t mix_nid,
+				   hda_nid_t pin, unsigned int *mix_val,
+				   unsigned int *mute_val)
+{
+	int idx, num_conns;
+	const hda_nid_t *list;
+	hda_nid_t nid;
+
+	idx = snd_hda_get_conn_index(codec, mix_nid, pin, true);
+	if (idx < 0)
+		return false;
+
+	*mix_val = *mute_val = 0;
+	if (nid_has_volume(codec, mix_nid, HDA_INPUT))
+		*mix_val = HDA_COMPOSE_AMP_VAL(mix_nid, 3, idx, HDA_INPUT);
+	if (nid_has_mute(codec, mix_nid, HDA_INPUT))
+		*mute_val = HDA_COMPOSE_AMP_VAL(mix_nid, 3, idx, HDA_INPUT);
+	if (*mix_val && *mute_val)
+		return true;
+
+	/* check leaf node */
+	num_conns = snd_hda_get_conn_list(codec, mix_nid, &list);
+	if (num_conns < idx)
+		return false;
+	nid = list[idx];
+	if (!*mix_val && nid_has_volume(codec, nid, HDA_OUTPUT))
+		*mix_val = HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT);
+	if (!*mute_val && nid_has_mute(codec, nid, HDA_OUTPUT))
+		*mute_val = HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT);
+
+	return *mix_val || *mute_val;
+}
+
 /* create input playback/capture controls for the given pin */
 static int new_analog_input(struct hda_codec *codec, int input_idx,
 			    hda_nid_t pin, const char *ctlname, int ctlidx,
@@ -2822,12 +2851,11 @@ static int new_analog_input(struct hda_codec *codec, int input_idx,
 {
 	struct hda_gen_spec *spec = codec->spec;
 	struct nid_path *path;
-	unsigned int val;
+	unsigned int mix_val, mute_val;
 	int err, idx;
 
-	if (!nid_has_volume(codec, mix_nid, HDA_INPUT) &&
-	    !nid_has_mute(codec, mix_nid, HDA_INPUT))
-		return 0; /* no need for analog loopback */
+	if (!look_for_mix_leaf_ctls(codec, mix_nid, pin, &mix_val, &mute_val))
+		return 0;
 
 	path = snd_hda_add_new_path(codec, pin, mix_nid, 0);
 	if (!path)
@@ -2836,20 +2864,18 @@ static int new_analog_input(struct hda_codec *codec, int input_idx,
 	spec->loopback_paths[input_idx] = snd_hda_get_path_idx(codec, path);
 
 	idx = path->idx[path->depth - 1];
-	if (nid_has_volume(codec, mix_nid, HDA_INPUT)) {
-		val = HDA_COMPOSE_AMP_VAL(mix_nid, 3, idx, HDA_INPUT);
-		err = __add_pb_vol_ctrl(spec, HDA_CTL_WIDGET_VOL, ctlname, ctlidx, val);
+	if (mix_val) {
+		err = __add_pb_vol_ctrl(spec, HDA_CTL_WIDGET_VOL, ctlname, ctlidx, mix_val);
 		if (err < 0)
 			return err;
-		path->ctls[NID_PATH_VOL_CTL] = val;
+		path->ctls[NID_PATH_VOL_CTL] = mix_val;
 	}
 
-	if (nid_has_mute(codec, mix_nid, HDA_INPUT)) {
-		val = HDA_COMPOSE_AMP_VAL(mix_nid, 3, idx, HDA_INPUT);
-		err = __add_pb_sw_ctrl(spec, HDA_CTL_WIDGET_MUTE, ctlname, ctlidx, val);
+	if (mute_val) {
+		err = __add_pb_sw_ctrl(spec, HDA_CTL_WIDGET_MUTE, ctlname, ctlidx, mute_val);
 		if (err < 0)
 			return err;
-		path->ctls[NID_PATH_MUTE_CTL] = val;
+		path->ctls[NID_PATH_MUTE_CTL] = mute_val;
 	}
 
 	path->active = true;
@@ -4383,6 +4409,17 @@ int snd_hda_gen_parse_auto_config(struct hda_codec *codec,
 	if (err < 0)
 		return err;
 
+	/* create "Headphone Mic Jack Mode" if no input selection is
+	 * available (or user specifies add_jack_modes hint)
+	 */
+	if (spec->hp_mic_pin &&
+	    (spec->auto_mic || spec->input_mux.num_items == 1 ||
+	     spec->add_jack_modes)) {
+		err = create_hp_mic_jack_mode(codec, spec->hp_mic_pin);
+		if (err < 0)
+			return err;
+	}
+
 	if (spec->add_jack_modes) {
 		if (cfg->line_out_type != AUTO_PIN_SPEAKER_OUT) {
 			err = create_out_jack_modes(codec, cfg->line_outs,
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 7a09404579a7..c6d230193da6 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -2994,8 +2994,7 @@ static int azx_runtime_suspend(struct device *dev)
 		  STATESTS_INT_MASK);
 
 	azx_stop_chip(chip);
-	if (!chip->bus->avoid_link_reset)
-		azx_enter_link_reset(chip);
+	azx_enter_link_reset(chip);
 	azx_clear_irq_pending(chip);
 	if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL)
 		hda_display_power(false);
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index c205bb1747fd..1f2717f817a0 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -3244,9 +3244,29 @@ enum {
 #if IS_ENABLED(CONFIG_THINKPAD_ACPI)
 
 #include <linux/thinkpad_acpi.h>
+#include <acpi/acpi.h>
 
 static int (*led_set_func)(int, bool);
 
+static acpi_status acpi_check_cb(acpi_handle handle, u32 lvl, void *context,
+				 void **rv)
+{
+	bool *found = context;
+	*found = true;
+	return AE_OK;
+}
+
+static bool is_thinkpad(struct hda_codec *codec)
+{
+	bool found = false;
+	if (codec->subsystem_id >> 16 != 0x17aa)
+		return false;
+	if (ACPI_SUCCESS(acpi_get_devices("LEN0068", acpi_check_cb, &found, NULL)) && found)
+		return true;
+	found = false;
+	return ACPI_SUCCESS(acpi_get_devices("IBM0068", acpi_check_cb, &found, NULL)) && found;
+}
+
 static void update_tpacpi_mute_led(void *private_data, int enabled)
 {
 	struct hda_codec *codec = private_data;
@@ -3279,6 +3299,8 @@ static void cxt_fixup_thinkpad_acpi(struct hda_codec *codec,
 	bool removefunc = false;
 
 	if (action == HDA_FIXUP_ACT_PROBE) {
+		if (!is_thinkpad(codec))
+			return;
 		if (!led_set_func)
 			led_set_func = symbol_request(tpacpi_led_set);
 		if (!led_set_func) {
@@ -3494,6 +3516,7 @@ static const struct snd_pci_quirk cxt5066_fixups[] = {
 	SND_PCI_QUIRK(0x17aa, 0x3975, "Lenovo U300s", CXT_FIXUP_STEREO_DMIC),
 	SND_PCI_QUIRK(0x17aa, 0x3977, "Lenovo IdeaPad U310", CXT_FIXUP_STEREO_DMIC),
 	SND_PCI_QUIRK(0x17aa, 0x397b, "Lenovo S205", CXT_FIXUP_STEREO_DMIC),
+	SND_PCI_QUIRK_VENDOR(0x17aa, "Thinkpad", CXT_FIXUP_THINKPAD_ACPI),
 	SND_PCI_QUIRK(0x1c06, 0x2011, "Lemote A1004", CXT_PINCFG_LEMOTE_A1004),
 	SND_PCI_QUIRK(0x1c06, 0x2012, "Lemote A1205", CXT_PINCFG_LEMOTE_A1205),
 	{}
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 5e42059f10a1..c770bdba6531 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -1782,6 +1782,8 @@ enum {
 	ALC889_FIXUP_IMAC91_VREF,
 	ALC882_FIXUP_INV_DMIC,
 	ALC882_FIXUP_NO_PRIMARY_HP,
+	ALC887_FIXUP_ASUS_BASS,
+	ALC887_FIXUP_BASS_CHMAP,
 };
 
 static void alc889_fixup_coef(struct hda_codec *codec,
@@ -1915,6 +1917,9 @@ static void alc882_fixup_no_primary_hp(struct hda_codec *codec,
 	}
 }
 
+static void alc_fixup_bass_chmap(struct hda_codec *codec,
+				 const struct hda_fixup *fix, int action);
+
 static const struct hda_fixup alc882_fixups[] = {
 	[ALC882_FIXUP_ABIT_AW9D_MAX] = {
 		.type = HDA_FIXUP_PINS,
@@ -2105,6 +2110,19 @@ static const struct hda_fixup alc882_fixups[] = {
 		.type = HDA_FIXUP_FUNC,
 		.v.func = alc882_fixup_no_primary_hp,
 	},
+	[ALC887_FIXUP_ASUS_BASS] = {
+		.type = HDA_FIXUP_PINS,
+		.v.pins = (const struct hda_pintbl[]) {
+			{0x16, 0x99130130}, /* bass speaker */
+			{}
+		},
+		.chained = true,
+		.chain_id = ALC887_FIXUP_BASS_CHMAP,
+	},
+	[ALC887_FIXUP_BASS_CHMAP] = {
+		.type = HDA_FIXUP_FUNC,
+		.v.func = alc_fixup_bass_chmap,
+	},
 };
 
 static const struct snd_pci_quirk alc882_fixup_tbl[] = {
@@ -2138,6 +2156,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1043, 0x1873, "ASUS W90V", ALC882_FIXUP_ASUS_W90V),
 	SND_PCI_QUIRK(0x1043, 0x1971, "Asus W2JC", ALC882_FIXUP_ASUS_W2JC),
 	SND_PCI_QUIRK(0x1043, 0x835f, "Asus Eee 1601", ALC888_FIXUP_EEE1601),
+	SND_PCI_QUIRK(0x1043, 0x84bc, "ASUS ET2700", ALC887_FIXUP_ASUS_BASS),
 	SND_PCI_QUIRK(0x104d, 0x9047, "Sony Vaio TT", ALC889_FIXUP_VAIO_TT),
 	SND_PCI_QUIRK(0x104d, 0x905a, "Sony Vaio Z", ALC882_FIXUP_NO_PRIMARY_HP),
 	SND_PCI_QUIRK(0x104d, 0x9043, "Sony Vaio VGC-LN51JGB", ALC882_FIXUP_NO_PRIMARY_HP),
@@ -3798,6 +3817,7 @@ enum {
 	ALC271_FIXUP_HP_GATE_MIC_JACK,
 	ALC269_FIXUP_ACER_AC700,
 	ALC269_FIXUP_LIMIT_INT_MIC_BOOST,
+	ALC269VB_FIXUP_ASUS_ZENBOOK,
 	ALC269_FIXUP_LIMIT_INT_MIC_BOOST_MUTE_LED,
 	ALC269VB_FIXUP_ORDISSIMO_EVE2,
 	ALC283_FIXUP_CHROME_BOOK,
@@ -4075,6 +4095,12 @@ static const struct hda_fixup alc269_fixups[] = {
 		.chained = true,
 		.chain_id = ALC269_FIXUP_THINKPAD_ACPI,
 	},
+	[ALC269VB_FIXUP_ASUS_ZENBOOK] = {
+		.type = HDA_FIXUP_FUNC,
+		.v.func = alc269_fixup_limit_int_mic_boost,
+		.chained = true,
+		.chain_id = ALC269VB_FIXUP_DMIC,
+	},
 	[ALC269_FIXUP_LIMIT_INT_MIC_BOOST_MUTE_LED] = {
 		.type = HDA_FIXUP_FUNC,
 		.v.func = alc269_fixup_limit_int_mic_boost,
@@ -4189,8 +4215,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1043, 0x103f, "ASUS TX300", ALC282_FIXUP_ASUS_TX300),
 	SND_PCI_QUIRK(0x1043, 0x106d, "Asus K53BE", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
 	SND_PCI_QUIRK(0x1043, 0x115d, "Asus 1015E", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
-	SND_PCI_QUIRK(0x1043, 0x1427, "Asus Zenbook UX31E", ALC269VB_FIXUP_DMIC),
-	SND_PCI_QUIRK(0x1043, 0x1517, "Asus Zenbook UX31A", ALC269VB_FIXUP_DMIC),
+	SND_PCI_QUIRK(0x1043, 0x1427, "Asus Zenbook UX31E", ALC269VB_FIXUP_ASUS_ZENBOOK),
+	SND_PCI_QUIRK(0x1043, 0x1517, "Asus Zenbook UX31A", ALC269VB_FIXUP_ASUS_ZENBOOK),
 	SND_PCI_QUIRK(0x1043, 0x16e3, "ASUS UX50", ALC269_FIXUP_STEREO_DMIC),
 	SND_PCI_QUIRK(0x1043, 0x1a13, "Asus G73Jw", ALC269_FIXUP_ASUS_G73JW),
 	SND_PCI_QUIRK(0x1043, 0x1b13, "Asus U41SV", ALC269_FIXUP_INV_DMIC),
@@ -4715,7 +4741,7 @@ static const struct snd_pcm_chmap_elem asus_pcm_2_1_chmaps[] = {
 };
 
 /* override the 2.1 chmap */
-static void alc662_fixup_bass_chmap(struct hda_codec *codec,
+static void alc_fixup_bass_chmap(struct hda_codec *codec,
 				    const struct hda_fixup *fix, int action)
 {
 	if (action == HDA_FIXUP_ACT_BUILD) {
@@ -4923,7 +4949,7 @@ static const struct hda_fixup alc662_fixups[] = {
 	},
 	[ALC662_FIXUP_BASS_CHMAP] = {
 		.type = HDA_FIXUP_FUNC,
-		.v.func = alc662_fixup_bass_chmap,
+		.v.func = alc_fixup_bass_chmap,
 		.chained = true,
 		.chain_id = ALC662_FIXUP_ASUS_MODE4
 	},
@@ -4936,7 +4962,7 @@ static const struct hda_fixup alc662_fixups[] = {
 	},
 	[ALC662_FIXUP_BASS_1A_CHMAP] = {
 		.type = HDA_FIXUP_FUNC,
-		.v.func = alc662_fixup_bass_chmap,
+		.v.func = alc_fixup_bass_chmap,
 		.chained = true,
 		.chain_id = ALC662_FIXUP_BASS_1A,
 	},
@@ -5118,6 +5144,7 @@ static int patch_alc662(struct hda_codec *codec)
 		case 0x10ec0272:
 		case 0x10ec0663:
 		case 0x10ec0665:
+		case 0x10ec0668:
 			set_beep_amp(spec, 0x0b, 0x04, HDA_INPUT);
 			break;
 		case 0x10ec0273:
@@ -5175,6 +5202,7 @@ static int patch_alc680(struct hda_codec *codec)
  */
 static const struct hda_codec_preset snd_hda_preset_realtek[] = {
 	{ .id = 0x10ec0221, .name = "ALC221", .patch = patch_alc269 },
+	{ .id = 0x10ec0231, .name = "ALC231", .patch = patch_alc269 },
 	{ .id = 0x10ec0233, .name = "ALC233", .patch = patch_alc269 },
 	{ .id = 0x10ec0255, .name = "ALC255", .patch = patch_alc269 },
 	{ .id = 0x10ec0260, .name = "ALC260", .patch = patch_alc260 },
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index d2cc0041d9d3..088a5afbd1b9 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -2094,7 +2094,8 @@ static void stac92hd83xxx_fixup_hp_mic_led(struct hda_codec *codec,
 
 	if (action == HDA_FIXUP_ACT_PRE_PROBE) {
 		spec->mic_mute_led_gpio = 0x08; /* GPIO3 */
-		codec->bus->avoid_link_reset = 1;
+		/* resetting controller clears GPIO, so we need to keep on */
+		codec->bus->power_keep_link_on = 1;
 	}
 }
 
diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c
index b9ba0fcc45df..83aabea259d7 100644
--- a/sound/usb/endpoint.c
+++ b/sound/usb/endpoint.c
@@ -636,8 +636,22 @@ static int data_ep_set_params(struct snd_usb_endpoint *ep,
 	if (usb_pipein(ep->pipe) ||
 			snd_usb_endpoint_implicit_feedback_sink(ep)) {
 
+		urb_packs = packs_per_ms;
+		/*
+		 * Wireless devices can poll at a max rate of once per 4ms.
+		 * For dataintervals less than 5, increase the packet count to
+		 * allow the host controller to use bursting to fill in the
+		 * gaps.
+		 */
+		if (snd_usb_get_speed(ep->chip->dev) == USB_SPEED_WIRELESS) {
+			int interval = ep->datainterval;
+			while (interval < 5) {
+				urb_packs <<= 1;
+				++interval;
+			}
+		}
 		/* make capture URBs <= 1 ms and smaller than a period */
-		urb_packs = min(max_packs_per_urb, packs_per_ms);
+		urb_packs = min(max_packs_per_urb, urb_packs);
 		while (urb_packs > 1 && urb_packs * maxsize >= period_bytes)
 			urb_packs >>= 1;
 		ep->nurbs = MAX_URBS;