Merge tag 'sound-4.0-rc4' of git://git.kernel.org/pub/scm/linux/kerne…

…l/git/tiwai/sound

Pull sound fixes from Takashi Iwai:
 "This is a round of HD-audio fixes: there are a long-standing
  regression fix and a few more device/codec-specific quirks.

  In addition, a couple of FireWire regression fixes, a USB-audio quirk
  for Roland UA-22 and a sanity check in API for user-defined control
  elements"

* tag 'sound-4.0-rc4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
  ALSA: hda - Don't access stereo amps for mono channel widgets
  ALSA: hda - Add workaround for MacBook Air 5,2 built-in mic
  ALSA: hda - Set single_adc_amp flag for CS420x codecs
  ALSA: snd-usb: add quirks for Roland UA-22
  ALSA: control: Add sanity checks for user ctl id name string
  ALSA: hda - Fix built-in mic on Compaq Presario CQ60
  ALSA: firewire-lib: leave unit reference counting completely
  Revert "ALSA: dice: fix wrong offsets for Dice interface"
  ALSA: hda - Fix regression of HD-audio controller fallback modes

sound/core/control.c

@@ -1170,6 +1170,10 @@ static int snd_ctl_elem_add(struct snd_ctl_file *file,
 
 	if (info->count < 1)
 		return -EINVAL;
+	if (!*info->id.name)
+		return -EINVAL;
+	if (strnlen(info->id.name, sizeof(info->id.name)) >= sizeof(info->id.name))
+		return -EINVAL;
 	access = info->access == 0 ? SNDRV_CTL_ELEM_ACCESS_READWRITE :
 		(info->access & (SNDRV_CTL_ELEM_ACCESS_READWRITE|
 				 SNDRV_CTL_ELEM_ACCESS_INACTIVE|

sound/firewire/dice/dice-interface.h

@@ -298,24 +298,24 @@
  */
 #define RX_ISOCHRONOUS			0x008
 
+/*
+ * Index of first quadlet to be interpreted; read/write.  If > 0, that many
+ * quadlets at the beginning of each data block will be ignored, and all the
+ * audio and MIDI quadlets will follow.
+ */
+#define RX_SEQ_START			0x00c
+
 /*
  * The number of audio channels; read-only.  There will be one quadlet per
  * channel.
  */
-#define RX_NUMBER_AUDIO			0x00c
+#define RX_NUMBER_AUDIO			0x010
 
 /*
  * The number of MIDI ports, 0-8; read-only.  If > 0, there will be one
  * additional quadlet in each data block, following the audio quadlets.
  */
-#define RX_NUMBER_MIDI			0x010
-
-/*
- * Index of first quadlet to be interpreted; read/write.  If > 0, that many
- * quadlets at the beginning of each data block will be ignored, and all the
- * audio and MIDI quadlets will follow.
- */
-#define RX_SEQ_START			0x014
+#define RX_NUMBER_MIDI			0x014
 
 /*
  * Names of all audio channels; read-only.  Quadlets are byte-swapped.  Names

sound/firewire/dice/dice-proc.c

@@ -99,9 +99,9 @@ static void dice_proc_read(struct snd_info_entry *entry,
 		} tx;
 		struct {
 			u32 iso;
+			u32 seq_start;
 			u32 number_audio;
 			u32 number_midi;
-			u32 seq_start;
 			char names[RX_NAMES_SIZE];
 			u32 ac3_caps;
 			u32 ac3_enable;
@@ -204,10 +204,10 @@ static void dice_proc_read(struct snd_info_entry *entry,
 			break;
 		snd_iprintf(buffer, "rx %u:\n", stream);
 		snd_iprintf(buffer, "  iso channel: %d\n", (int)buf.rx.iso);
+		snd_iprintf(buffer, "  sequence start: %u\n", buf.rx.seq_start);
 		snd_iprintf(buffer, "  audio channels: %u\n",
 			    buf.rx.number_audio);
 		snd_iprintf(buffer, "  midi ports: %u\n", buf.rx.number_midi);
-		snd_iprintf(buffer, "  sequence start: %u\n", buf.rx.seq_start);
 		if (quadlets >= 68) {
 			dice_proc_fixup_string(buf.rx.names, RX_NAMES_SIZE);
 			snd_iprintf(buffer, "  names: %s\n", buf.rx.names);

sound/firewire/iso-resources.c

@@ -26,7 +26,7 @@
 int fw_iso_resources_init(struct fw_iso_resources *r, struct fw_unit *unit)
 {
 	r->channels_mask = ~0uLL;
-	r->unit = fw_unit_get(unit);
+	r->unit = unit;
 	mutex_init(&r->mutex);
 	r->allocated = false;
 
@@ -42,7 +42,6 @@ void fw_iso_resources_destroy(struct fw_iso_resources *r)
 {
 	WARN_ON(r->allocated);
 	mutex_destroy(&r->mutex);
-	fw_unit_put(r->unit);
 }
 EXPORT_SYMBOL(fw_iso_resources_destroy);
 

sound/pci/hda/hda_controller.c

@@ -1164,7 +1164,7 @@ static unsigned int azx_rirb_get_response(struct hda_bus *bus,
 		}
 	}
 
-	if (!bus->no_response_fallback)
+	if (bus->no_response_fallback)
 		return -1;
 
 	if (!chip->polling_mode && chip->poll_count < 2) {

sound/pci/hda/hda_generic.c

@@ -692,7 +692,23 @@ static void init_amp(struct hda_codec *codec, hda_nid_t nid, int dir, int idx)
 {
 	unsigned int caps = query_amp_caps(codec, nid, dir);
 	int val = get_amp_val_to_activate(codec, nid, dir, caps, false);
-	snd_hda_codec_amp_init_stereo(codec, nid, dir, idx, 0xff, val);
+
+	if (get_wcaps(codec, nid) & AC_WCAP_STEREO)
+		snd_hda_codec_amp_init_stereo(codec, nid, dir, idx, 0xff, val);
+	else
+		snd_hda_codec_amp_init(codec, nid, 0, dir, idx, 0xff, val);
+}
+
+/* update the amp, doing in stereo or mono depending on NID */
+static int update_amp(struct hda_codec *codec, hda_nid_t nid, int dir, int idx,
+		      unsigned int mask, unsigned int val)
+{
+	if (get_wcaps(codec, nid) & AC_WCAP_STEREO)
+		return snd_hda_codec_amp_stereo(codec, nid, dir, idx,
+						mask, val);
+	else
+		return snd_hda_codec_amp_update(codec, nid, 0, dir, idx,
+						mask, val);
 }
 
 /* calculate amp value mask we can modify;
@@ -732,7 +748,7 @@ static void activate_amp(struct hda_codec *codec, hda_nid_t nid, int dir,
 		return;
 
 	val &= mask;
-	snd_hda_codec_amp_stereo(codec, nid, dir, idx, mask, val);
+	update_amp(codec, nid, dir, idx, mask, val);
 }
 
 static void activate_amp_out(struct hda_codec *codec, struct nid_path *path,
@@ -4424,13 +4440,11 @@ static void mute_all_mixer_nid(struct hda_codec *codec, hda_nid_t mix)
 	has_amp = nid_has_mute(codec, mix, HDA_INPUT);
 	for (i = 0; i < nums; i++) {
 		if (has_amp)
-			snd_hda_codec_amp_stereo(codec, mix,
-						 HDA_INPUT, i,
-						 0xff, HDA_AMP_MUTE);
+			update_amp(codec, mix, HDA_INPUT, i,
+				   0xff, HDA_AMP_MUTE);
 		else if (nid_has_volume(codec, conn[i], HDA_OUTPUT))
-			snd_hda_codec_amp_stereo(codec, conn[i],
-						 HDA_OUTPUT, 0,
-						 0xff, HDA_AMP_MUTE);
+			update_amp(codec, conn[i], HDA_OUTPUT, 0,
+				   0xff, HDA_AMP_MUTE);
 	}
 }
 

sound/pci/hda/patch_cirrus.c

@@ -393,6 +393,7 @@ static const struct snd_pci_quirk cs420x_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x106b, 0x1c00, "MacBookPro 8,1", CS420X_MBP81),
 	SND_PCI_QUIRK(0x106b, 0x2000, "iMac 12,2", CS420X_IMAC27_122),
 	SND_PCI_QUIRK(0x106b, 0x2800, "MacBookPro 10,1", CS420X_MBP101),
+	SND_PCI_QUIRK(0x106b, 0x5600, "MacBookAir 5,2", CS420X_MBP81),
 	SND_PCI_QUIRK(0x106b, 0x5b00, "MacBookAir 4,2", CS420X_MBA42),
 	SND_PCI_QUIRK_VENDOR(0x106b, "Apple", CS420X_APPLE),
 	{} /* terminator */
@@ -584,6 +585,7 @@ static int patch_cs420x(struct hda_codec *codec)
 		return -ENOMEM;
 
 	spec->gen.automute_hook = cs_automute;
+	codec->single_adc_amp = 1;
 
 	snd_hda_pick_fixup(codec, cs420x_models, cs420x_fixup_tbl,
 			   cs420x_fixups);

sound/pci/hda/patch_conexant.c

@@ -223,6 +223,7 @@ enum {
 	CXT_PINCFG_LENOVO_TP410,
 	CXT_PINCFG_LEMOTE_A1004,
 	CXT_PINCFG_LEMOTE_A1205,
+	CXT_PINCFG_COMPAQ_CQ60,
 	CXT_FIXUP_STEREO_DMIC,
 	CXT_FIXUP_INC_MIC_BOOST,
 	CXT_FIXUP_HEADPHONE_MIC_PIN,
@@ -660,6 +661,15 @@ static const struct hda_fixup cxt_fixups[] = {
 		.type = HDA_FIXUP_PINS,
 		.v.pins = cxt_pincfg_lemote,
 	},
+	[CXT_PINCFG_COMPAQ_CQ60] = {
+		.type = HDA_FIXUP_PINS,
+		.v.pins = (const struct hda_pintbl[]) {
+			/* 0x17 was falsely set up as a mic, it should 0x1d */
+			{ 0x17, 0x400001f0 },
+			{ 0x1d, 0x97a70120 },
+			{ }
+		}
+	},
 	[CXT_FIXUP_STEREO_DMIC] = {
 		.type = HDA_FIXUP_FUNC,
 		.v.func = cxt_fixup_stereo_dmic,
@@ -769,6 +779,7 @@ static const struct hda_model_fixup cxt5047_fixup_models[] = {
 };
 
 static const struct snd_pci_quirk cxt5051_fixups[] = {
+	SND_PCI_QUIRK(0x103c, 0x360b, "Compaq CQ60", CXT_PINCFG_COMPAQ_CQ60),
 	SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo X200", CXT_PINCFG_LENOVO_X200),
 	{}
 };

sound/usb/quirks-table.h

@@ -1773,6 +1773,36 @@ YAMAHA_DEVICE(0x7010, "UB99"),
 		}
 	}
 },
+{
+	USB_DEVICE(0x0582, 0x0159),
+	.driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+		/* .vendor_name = "Roland", */
+		/* .product_name = "UA-22", */
+		.ifnum = QUIRK_ANY_INTERFACE,
+		.type = QUIRK_COMPOSITE,
+		.data = (const struct snd_usb_audio_quirk[]) {
+			{
+				.ifnum = 0,
+				.type = QUIRK_AUDIO_STANDARD_INTERFACE
+			},
+			{
+				.ifnum = 1,
+				.type = QUIRK_AUDIO_STANDARD_INTERFACE
+			},
+			{
+				.ifnum = 2,
+				.type = QUIRK_MIDI_FIXED_ENDPOINT,
+				.data = & (const struct snd_usb_midi_endpoint_info) {
+					.out_cables = 0x0001,
+					.in_cables = 0x0001
+				}
+			},
+			{
+				.ifnum = -1
+			}
+		}
+	}
+},
 /* this catches most recent vendor-specific Roland devices */
 {
 	.match_flags = USB_DEVICE_ID_MATCH_VENDOR |