Fixes gstreamer caps and code cleanup.

8 files changed, 113 insertions, 143 deletions

diff --git a/audio/gsta2dpsink.c b/audio/gsta2dpsink.c
index a2b3286c..b5b1c576 100644
--- a/audio/gsta2dpsink.c
+++ b/audio/gsta2dpsink.c
@@ -80,6 +80,17 @@ static void gst_a2dp_sink_finalize(GObject *obj)
 	G_OBJECT_CLASS (parent_class)->finalize (obj);
 }
 
+static GstState gst_a2dp_sink_get_state(GstA2dpSink *self)
+{
+	GstState current, pending;
+
+	gst_element_get_state(GST_ELEMENT(self), &current, &pending, 0);
+	if (pending == GST_STATE_VOID_PENDING)
+		return current;
+
+	return pending;
+}
+
 /*
  * Helper function to create elements, add to the bin and link it
  * to another element.
@@ -89,6 +100,7 @@ static GstElement* gst_a2dp_sink_init_element(GstA2dpSink *self,
 			GstElement *link_to)
 {
 	GstElement *element;
+	GstState state;
 
 	GST_LOG_OBJECT(self, "Initializing %s", elementname);
 
@@ -104,17 +116,19 @@ static GstElement* gst_a2dp_sink_init_element(GstA2dpSink *self,
 		goto cleanup_and_fail;
 	}
 
-	if (gst_element_set_state(element, GST_STATE_PLAYING) ==
+	state = gst_a2dp_sink_get_state(self);
+	if (gst_element_set_state(element, state) ==
 			GST_STATE_CHANGE_FAILURE) {
 		GST_ERROR_OBJECT(self, "%s failed to go to playing",
 						elementname);
 		goto remove_element_and_fail;
 	}
 
-	if (!gst_element_link(link_to, element)) {
-		GST_ERROR_OBJECT(self, "couldn't link %s", elementname);
-		goto remove_element_and_fail;
-	}
+	if (link_to != NULL)
+		if (!gst_element_link(link_to, element)) {
+			GST_ERROR_OBJECT(self, "couldn't link %s", elementname);
+			goto remove_element_and_fail;
+		}
 
 	return element;
 
@@ -241,7 +255,6 @@ static GstStateChangeReturn gst_a2dp_sink_change_state(GstElement *element,
 
 	case GST_STATE_CHANGE_NULL_TO_READY:
 		self->sink_is_in_bin = FALSE;
-
 		self->sink = GST_AVDTP_SINK(gst_element_factory_make(
 				"avdtpsink", "avdtpsink"));
 		if (self->sink == NULL) {
@@ -260,8 +273,10 @@ static GstStateChangeReturn gst_a2dp_sink_change_state(GstElement *element,
 		break;
 	}
 
-	if (ret == GST_STATE_CHANGE_FAILURE)
+	if (ret == GST_STATE_CHANGE_FAILURE) {
+		g_mutex_unlock(self->cb_mutex);
 		return ret;
+	}
 
 	ret = GST_ELEMENT_CLASS(parent_class)->change_state(element,
                         transition);
@@ -295,7 +310,6 @@ static GstStateChangeReturn gst_a2dp_sink_change_state(GstElement *element,
 
 		gst_a2dp_sink_remove_dynamic_elements(self);
 		break;
-
 	default:
 		break;
 	}
@@ -558,16 +572,16 @@ static gboolean gst_a2dp_sink_handle_event(GstPad *pad, GstEvent *event)
 		if (self->newseg_event != NULL)
 			gst_event_unref(self->newseg_event);
 		self->newseg_event = gst_event_ref(event);
+
 	} else if (GST_EVENT_TYPE(event) == GST_EVENT_TAG &&
 			parent != GST_OBJECT_CAST(self)) {
-		if (self->taglist == NULL) {
+		if (self->taglist == NULL)
 			gst_event_parse_tag(event, &self->taglist);
-		} else {
+		else {
 			gst_event_parse_tag(event, &taglist);
 			gst_tag_list_insert(self->taglist, taglist,
 					GST_TAG_MERGE_REPLACE);
 		}
-		/* FIXME handle tag events */
 	}
 
 	if (parent != NULL)
@@ -614,11 +628,15 @@ static gboolean gst_a2dp_sink_init_fakesink(GstA2dpSink *self)
 static gboolean gst_a2dp_sink_remove_fakesink(GstA2dpSink *self)
 {
 	g_mutex_lock(self->cb_mutex);
+
 	if (self->fakesink != NULL) {
+		gst_element_set_locked_state(self->fakesink, TRUE);
 		gst_element_set_state(self->fakesink, GST_STATE_NULL);
+
 		gst_bin_remove(GST_BIN(self), self->fakesink);
 		self->fakesink = NULL;
 	}
+
 	g_mutex_unlock(self->cb_mutex);
 
 	return TRUE;
diff --git a/audio/gstavdtpsink.c b/audio/gstavdtpsink.c
index c956ecbd..25fc1ced 100644
--- a/audio/gstavdtpsink.c
+++ b/audio/gstavdtpsink.c
@@ -68,7 +68,7 @@ struct bluetooth_data {
 };
 
 #define IS_SBC(n) (strcmp((n), "audio/x-sbc") == 0)
-#define IS_MPEG(n) (strcmp((n), "audio/mpeg") == 0)
+#define IS_MPEG_AUDIO(n) (strcmp((n), "audio/mpeg") == 0)
 
 enum {
 	PROP_0,
@@ -87,8 +87,12 @@ static const GstElementDetails avdtp_sink_details =
 static GstStaticPadTemplate avdtp_sink_factory =
 	GST_STATIC_PAD_TEMPLATE("sink", GST_PAD_SINK, GST_PAD_ALWAYS,
 		GST_STATIC_CAPS("application/x-rtp, "
-				"media = (string) \"audio\", "
-				"encoding-name = (string) \"SBC\";"
+				"media = (string) \"audio\","
+				"payload = (int) "
+					GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
+				"clock-rate = (int) { 16000, 32000, "
+					"44100, 48000 }, "
+				"encoding-name = (string) \"SBC\"; "
 				"application/x-rtp, "
 				"media = (string) \"audio\", "
 				"payload = (int) "
@@ -231,7 +235,7 @@ static gint gst_avdtp_sink_bluetooth_recvmsg_fd(GstAvdtpSink *sink)
 	return 0;
 }
 
-static gboolean gst_avdtp_sink_init_pkt_conf(GstAvdtpSink *sink,
+static gboolean gst_avdtp_sink_init_sbc_pkt_conf(GstAvdtpSink *sink,
 					GstCaps *caps,
 					sbc_capabilities_t *pkt)
 {
@@ -242,9 +246,10 @@ static gboolean gst_avdtp_sink_init_pkt_conf(GstAvdtpSink *sink,
 	GstStructure *structure = gst_caps_get_structure(caps, 0);
 
 	name = gst_structure_get_name(structure);
-	/* FIXME only sbc supported here, should suport mp3 */
+
 	if (!(IS_SBC(name))) {
-		GST_ERROR_OBJECT(sink, "Unsupported format %s", name);
+		GST_ERROR_OBJECT(sink, "Unexpected format %s, "
+				"was expecting sbc", name);
 		return FALSE;
 	}
 
@@ -814,7 +819,7 @@ static void gst_avdtp_sink_tag(const GstTagList *taglist,
 
 		if (!gst_tag_list_get_boolean(taglist, tag, &crc)) {
 			GST_WARNING_OBJECT(self, "failed to get crc tag");
-			self->mpeg_stream_changed = TRUE;
+			return;
 		}
 
 		gst_avdtp_sink_set_crc(self, crc);
@@ -824,7 +829,7 @@ static void gst_avdtp_sink_tag(const GstTagList *taglist,
 		if (!gst_tag_list_get_string(taglist, tag, &channel_mode)) {
 			GST_WARNING_OBJECT(self,
 				"failed to get channel-mode tag");
-			self->mpeg_stream_changed = TRUE;
+			return;
 		}
 
 		self->channel_mode = gst_avdtp_sink_get_channel_mode(
@@ -883,7 +888,6 @@ static gboolean gst_avdtp_sink_start(GstBaseSink *basesink)
 	self->stream_caps = NULL;
 	self->mp3_using_crc = -1;
 	self->channel_mode = -1;
-	self->mpeg_stream_changed = FALSE;
 
 	if (!gst_avdtp_sink_get_capabilities(self)) {
 		GST_ERROR_OBJECT(self, "failed to get capabilities "
@@ -957,8 +961,17 @@ static gboolean gst_avdtp_sink_init_mp3_pkt_conf(
 {
 	const GValue *value = NULL;
 	gint rate, layer;
+	const gchar* name;
 	GstStructure *structure = gst_caps_get_structure(caps, 0);
 
+	name = gst_structure_get_name(structure);
+
+	if (!(IS_MPEG_AUDIO(name))) {
+		GST_ERROR_OBJECT(self, "Unexpected format %s, "
+				"was expecting mp3", name);
+		return FALSE;
+	}
+
 	/* layer */
 	value = gst_structure_get_value(structure, "layer");
 	layer = g_value_get_int(value);
@@ -1016,9 +1029,6 @@ static gboolean gst_avdtp_sink_init_mp3_pkt_conf(
 	/* vbr - we always say its vbr, we don't have how to know it */
 	pkt->bitrate = 0x8000;
 
-	/* bitrate - we don't set anything, its vbr */
-	/* FIXME - is this right? */
-
 	return TRUE;
 }
 
@@ -1044,7 +1054,7 @@ static gboolean gst_avdtp_sink_configure(GstAvdtpSink *self,
 	structure = gst_caps_get_structure(caps, 0);
 
 	if (gst_structure_has_name(structure, "audio/x-sbc"))
-		ret = gst_avdtp_sink_init_pkt_conf(self, caps,
+		ret = gst_avdtp_sink_init_sbc_pkt_conf(self, caps,
 				&req->sbc_capabilities);
 	else if (gst_structure_has_name(structure, "audio/mpeg"))
 		ret = gst_avdtp_sink_init_mp3_pkt_conf(self, caps,
@@ -1188,8 +1198,8 @@ static void gst_avdtp_sink_class_init(GstAvdtpSinkClass *klass)
 					"Bluetooth remote device address",
 					NULL, G_PARAM_READWRITE));
 
-	GST_DEBUG_CATEGORY_INIT(avdtp_sink_debug, "a2dpsendersink", 0,
-				"A2DP sink element");
+	GST_DEBUG_CATEGORY_INIT(avdtp_sink_debug, "avdtpsink", 0,
+				"A2DP headset sink element");
 }
 
 static void gst_avdtp_sink_init(GstAvdtpSink *self,
@@ -1203,6 +1213,11 @@ static void gst_avdtp_sink_init(GstAvdtpSink *self,
 	self->dev_caps = NULL;
 
 	self->sink_lock = g_mutex_new();
+
+	/* FIXME this is for not synchronizing with clock, should be tested
+	 * with devices to see the behaviour
+	gst_base_sink_set_sync(GST_BASE_SINK(self), FALSE);
+	*/
 }
 
 static GIOError gst_avdtp_sink_audioservice_send(
@@ -1325,9 +1340,7 @@ void gst_avdtp_sink_set_crc(GstAvdtpSink *self, gboolean crc)
 
 	/* test if we already received a different crc */
 	if (self->mp3_using_crc != -1 && new_crc != self->mp3_using_crc) {
-		GST_ERROR_OBJECT(self, "crc changed during stream");
-		/* FIXME test this, its not being used anywhere */
-		self->mpeg_stream_changed = TRUE;
+		GST_WARNING_OBJECT(self, "crc changed during stream");
 		return;
 	}
 	self->mp3_using_crc = new_crc;
@@ -1342,8 +1355,8 @@ void gst_avdtp_sink_set_channel_mode(GstAvdtpSink *self,
 	new_mode = gst_avdtp_sink_get_channel_mode(mode);
 
 	if (self->channel_mode != -1 && new_mode != self->channel_mode) {
-		GST_ERROR_OBJECT(self, "channel mode changed during stream");
-		self->mpeg_stream_changed = TRUE;
+		GST_WARNING_OBJECT(self, "channel mode changed during stream");
+		return;
 	}
 
 	self->channel_mode = new_mode;
diff --git a/audio/gstavdtpsink.h b/audio/gstavdtpsink.h
index d0380829..237597da 100644
--- a/audio/gstavdtpsink.h
+++ b/audio/gstavdtpsink.h
@@ -57,7 +57,6 @@ struct _GstAvdtpSink {
 	GIOChannel *server;
 
 	/* mp3 stream data (outside caps data)*/
-	gboolean mpeg_stream_changed;
 	gint mp3_using_crc;
 	gint channel_mode;
 
diff --git a/audio/gstrtpsbcpay.c b/audio/gstrtpsbcpay.c
index 25bf70eb..ba2a4741 100644
--- a/audio/gstrtpsbcpay.c
+++ b/audio/gstrtpsbcpay.c
@@ -77,7 +77,7 @@ static const GstElementDetails gst_rtp_sbc_pay_details =
 
 static GstStaticPadTemplate gst_rtp_sbc_pay_sink_factory =
 	GST_STATIC_PAD_TEMPLATE("sink", GST_PAD_SINK, GST_PAD_ALWAYS,
-		GST_STATIC_CAPS("audio/x-sbc, " /* FIXME remove those caps? */
+		GST_STATIC_CAPS("audio/x-sbc, "
 				"rate = (int) { 16000, 32000, 44100, 48000 }, "
 				"channels = (int) [ 1, 2 ], "
 				"mode = (string) { mono, dual, stereo, joint }, "
@@ -89,7 +89,12 @@ static GstStaticPadTemplate gst_rtp_sbc_pay_sink_factory =
 
 static GstStaticPadTemplate gst_rtp_sbc_pay_src_factory =
 	GST_STATIC_PAD_TEMPLATE("src", GST_PAD_SRC, GST_PAD_ALWAYS,
-		GST_STATIC_CAPS("application/x-rtp") /* FIXME put things here */
+		GST_STATIC_CAPS(
+			"application/x-rtp, "
+			"media = (string) \"audio\","
+			"payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
+			"clock-rate = (int) { 16000, 32000, 44100, 48000 },"
+			"encoding-name = (string) \"SBC\"")
 	);
 
 static void gst_rtp_sbc_pay_set_property (GObject * object, guint prop_id,
@@ -148,7 +153,7 @@ static gboolean gst_rtp_sbc_pay_set_caps(GstBaseRTPPayload *payload,
 
 	sbcpay->frame_length = frame_len;
 
-	gst_basertppayload_set_options (payload, "audio", FALSE, "SBC", rate);
+	gst_basertppayload_set_options (payload, "audio", TRUE, "SBC", rate);
 
 	GST_DEBUG_OBJECT(payload, "calculated frame length: %d ", frame_len);
 
@@ -209,6 +214,8 @@ static GstFlowReturn gst_rtp_sbc_pay_handle_buffer(GstBaseRTPPayload *payload,
 	GstRtpSBCPay *sbcpay;
 	guint available;
 
+	/* FIXME check for negotiation */
+
 	sbcpay = GST_RTP_SBC_PAY(payload);
 	gst_adapter_push(sbcpay->adapter, gst_buffer_copy(buffer));
 
diff --git a/audio/gstsbcdec.c b/audio/gstsbcdec.c
index 8c27daba..a686a406 100644
--- a/audio/gstsbcdec.c
+++ b/audio/gstsbcdec.c
@@ -80,26 +80,12 @@ static GstFlowReturn sbc_dec_chain(GstPad *pad, GstBuffer *buffer)
 	while (offset < size) {
 		GstBuffer *output;
 		GstPadTemplate *template;
-		GstCaps *caps, *temp;
+		GstCaps *caps;
 		int consumed;
 
-		caps = gst_caps_new_simple("audio/x-raw-int",
-				"rate", G_TYPE_INT, dec->sbc.rate,
-				"channels", G_TYPE_INT, dec->sbc.channels,
-				NULL);
-
-		template = gst_static_pad_template_get(&sbc_dec_src_factory);
-
-		temp = gst_caps_intersect(caps,
-					gst_pad_template_get_caps(template));
-
-		gst_caps_unref(caps);
-
 		res = gst_pad_alloc_buffer_and_set_caps(dec->srcpad,
 						GST_BUFFER_OFFSET_NONE,
-						codesize, temp, &output);
-
-		gst_caps_unref(temp);
+						codesize, NULL, &output);
 
 		if (res != GST_FLOW_OK)
 			goto done;
@@ -110,7 +96,25 @@ static GstFlowReturn sbc_dec_chain(GstPad *pad, GstBuffer *buffer)
 		if (consumed <= 0)
 			break;
 
-		GST_BUFFER_TIMESTAMP(output) = GST_BUFFER_TIMESTAMP(buffer);
+		/* we will reuse the same caps object */
+		if (dec->outcaps == NULL) {
+			caps = gst_caps_new_simple("audio/x-raw-int",
+					"rate", G_TYPE_INT, dec->sbc.rate,
+					"channels", G_TYPE_INT, dec->sbc.channels,
+					NULL);
+
+			template = gst_static_pad_template_get(&sbc_dec_src_factory);
+
+			dec->outcaps = gst_caps_intersect(caps,
+						gst_pad_template_get_caps(template));
+
+			gst_caps_unref(caps);
+		}
+
+		gst_buffer_set_caps(output, dec->outcaps);
+
+		/* FIXME get a real timestamp */
+		GST_BUFFER_TIMESTAMP(output) = GST_CLOCK_TIME_NONE;
 
 		res = gst_pad_push(dec->srcpad, output);
 		if (res != GST_FLOW_OK)
@@ -143,6 +147,7 @@ static GstStateChangeReturn sbc_dec_change_state(GstElement *element,
 			dec->buffer = NULL;
 		}
 		sbc_init(&dec->sbc, 0);
+		dec->outcaps = NULL;
 		break;
 
 	case GST_STATE_CHANGE_PAUSED_TO_READY:
@@ -152,6 +157,10 @@ static GstStateChangeReturn sbc_dec_change_state(GstElement *element,
 			dec->buffer = NULL;
 		}
 		sbc_finish(&dec->sbc);
+		if (dec->outcaps) {
+			gst_caps_unref(dec->outcaps);
+			dec->outcaps = NULL;
+		}
 		break;
 
 	default:
@@ -197,6 +206,8 @@ static void gst_sbc_dec_init(GstSbcDec *self, GstSbcDecClass *klass)
 	self->srcpad = gst_pad_new_from_static_template(
 			&sbc_dec_src_factory, "src");
 	gst_element_add_pad(GST_ELEMENT(self), self->srcpad);
+
+	self->outcaps = NULL;
 }
 
 gboolean gst_sbc_dec_plugin_init (GstPlugin * plugin)
diff --git a/audio/gstsbcdec.h b/audio/gstsbcdec.h
index 0bb0b57e..a88f8da5 100644
--- a/audio/gstsbcdec.h
+++ b/audio/gstsbcdec.h
@@ -49,6 +49,9 @@ struct _GstSbcDec {
 
 	GstBuffer *buffer;
 
+	/* caps for outgoing buffers */
+	GstCaps *outcaps;
+
 	sbc_t sbc;
 };
 
diff --git a/audio/gstsbcparse.c b/audio/gstsbcparse.c
index 49d0bb6e..1f699620 100644
--- a/audio/gstsbcparse.c
+++ b/audio/gstsbcparse.c
@@ -43,7 +43,8 @@ static const GstElementDetails sbc_parse_details =
 
 static GstStaticPadTemplate sbc_parse_sink_factory =
 	GST_STATIC_PAD_TEMPLATE("sink", GST_PAD_SINK, GST_PAD_ALWAYS,
-		GST_STATIC_CAPS("audio/x-sbc"));
+		GST_STATIC_CAPS("audio/x-sbc,"
+				"parsed = (boolean) false"));
 
 static GstStaticPadTemplate sbc_parse_src_factory =
 	GST_STATIC_PAD_TEMPLATE("src", GST_PAD_SRC, GST_PAD_ALWAYS,
@@ -54,86 +55,8 @@ static GstStaticPadTemplate sbc_parse_src_factory =
 				"blocks = (int) { 4, 8, 12, 16 }, "
 				"subbands = (int) { 4, 8 }, "
 				"allocation = (string) { snr, loudness },"
-				"bitpool = (int) [ 2, 64 ]"));
-
-static gboolean sbc_parse_sink_setcaps(GstPad * pad, GstCaps * caps)
-{
-	GstSbcParse *parse;
-	GstStructure *structure;
-	gint rate, channels;
-
-	parse = GST_SBC_PARSE(GST_PAD_PARENT(pad));
-
-	structure = gst_caps_get_structure(caps, 0);
-
-	if (!gst_structure_get_int(structure, "rate", &rate))
-		return FALSE;
-
-	if (!gst_structure_get_int(structure, "channels", &channels))
-		return FALSE;
-
-	if (!(parse->rate == 0 || rate == parse->rate))
-		return FALSE;
-
-	if (!(parse->channels == 0 || channels == parse->channels))
-		return FALSE;
-
-	parse->rate = rate;
-	parse->channels = channels;
-
-	return gst_sbc_util_fill_sbc_params(&parse->sbc, caps);
-}
-
-static GstCaps* sbc_parse_src_getcaps(GstPad *pad)
-{
-	GstCaps *caps;
-	const GstCaps *allowed_caps;
-	GstStructure *structure;
-	GValue *value;
-	GstSbcParse *parse = GST_SBC_PARSE(GST_PAD_PARENT(pad));
-
-	allowed_caps = gst_pad_get_allowed_caps(pad);
-	if (allowed_caps == NULL)
-		allowed_caps = gst_pad_get_pad_template_caps(pad);
-	caps = gst_caps_copy(allowed_caps);
-
-	value = g_new0(GValue, 1);
-
-	structure = gst_caps_get_structure(caps, 0);
-
-	if (parse->rate != 0)
-		gst_sbc_util_set_structure_int_param(structure, "rate",
-				parse->rate, value);
-	if (parse->channels != 0)
-		gst_sbc_util_set_structure_int_param(structure, "channels",
-				parse->channels, value);
-
-	g_free(value);
-
-	return caps;
-}
-
-static gboolean sbc_parse_src_acceptcaps(GstPad *pad, GstCaps *caps)
-{
-	GstStructure *structure;
-	GstSbcParse *parse;
-	gint rate, channels;
-
-	parse = GST_SBC_PARSE(GST_PAD_PARENT(pad));
-
-	structure = gst_caps_get_structure(caps, 0);
-
-	if (!gst_structure_get_int(structure, "rate", &rate))
-		return FALSE;
-	if (!gst_structure_get_int(structure, "channels", &channels))
-		return FALSE;
-
-	if ((parse->rate == 0 || parse->rate == rate)
-		&& (parse->channels == 0 || parse->channels == channels))
-		return TRUE;
-
-	return FALSE;
-}
+				"bitpool = (int) [ 2, 64 ],"
+				"parsed = (boolean) true"));
 
 static GstFlowReturn sbc_parse_chain(GstPad *pad, GstBuffer *buffer)
 {
@@ -261,17 +184,10 @@ static void gst_sbc_parse_init(GstSbcParse *self, GstSbcParseClass *klass)
 		&sbc_parse_sink_factory, "sink");
 	gst_pad_set_chain_function(self->sinkpad,
 		GST_DEBUG_FUNCPTR(sbc_parse_chain));
-	gst_pad_set_setcaps_function (self->sinkpad,
-		GST_DEBUG_FUNCPTR (sbc_parse_sink_setcaps));
 	gst_element_add_pad(GST_ELEMENT(self), self->sinkpad);
 
 	self->srcpad = gst_pad_new_from_static_template(
 		&sbc_parse_src_factory, "src");
-	gst_pad_set_getcaps_function (self->srcpad,
-		GST_DEBUG_FUNCPTR (sbc_parse_src_getcaps));
-	gst_pad_set_acceptcaps_function (self->srcpad,
-		GST_DEBUG_FUNCPTR (sbc_parse_src_acceptcaps));
-	/* FIXME get encoding parameters on set caps */
 	gst_element_add_pad(GST_ELEMENT(self), self->srcpad);
 }
 
diff --git a/audio/gstsbcutil.c b/audio/gstsbcutil.c
index 78024f7e..e5d21af0 100644
--- a/audio/gstsbcutil.c
+++ b/audio/gstsbcutil.c
@@ -364,9 +364,12 @@ GstCaps* gst_sbc_util_caps_fixate(GstCaps *caps, gchar** error_message)
 		goto error;
 	} else {
 		value = gst_structure_get_value(structure, "mode");
-		if (GST_VALUE_HOLDS_LIST(value))
-			mode = gst_sbc_get_mode_from_list(value);
-		else
+		if (GST_VALUE_HOLDS_LIST(value)) {
+			if (channels == 1)
+				mode = "mono";
+			else
+				mode = gst_sbc_get_mode_from_list(value);
+		} else
 			mode = g_value_get_string(value);
 	}