return ret;
}
+static GstElement *
+_build_fec_encoder (GstWebRTCBin * webrtc, WebRTCTransceiver * trans)
+{
+ GstElement *ret = NULL;
+ GstElement *prev = NULL;
+ guint ulpfec_pt = 0;
+ guint red_pt = 0;
+ GstPad *sinkpad = NULL;
+ GstWebRTCRTPTransceiver *rtp_trans = GST_WEBRTC_RTP_TRANSCEIVER (trans);
+
+ if (trans->stream) {
+ ulpfec_pt =
+ transport_stream_get_pt (trans->stream, "ULPFEC", rtp_trans->mline);
+ red_pt = transport_stream_get_pt (trans->stream, "RED", rtp_trans->mline);
+ }
+
+ if (ulpfec_pt || red_pt)
+ ret = gst_bin_new (NULL);
+
+ if (ulpfec_pt) {
+ GstElement *fecenc = gst_element_factory_make ("rtpulpfecenc", NULL);
+ GstCaps *caps = transport_stream_get_caps_for_pt (trans->stream, ulpfec_pt);
+
+ GST_DEBUG_OBJECT (webrtc,
+ "Creating ULPFEC encoder for mline %u with pt %d", rtp_trans->mline,
+ ulpfec_pt);
+
+ gst_bin_add (GST_BIN (ret), fecenc);
+ sinkpad = gst_element_get_static_pad (fecenc, "sink");
+ g_object_set (fecenc, "pt", ulpfec_pt, "percentage",
+ trans->fec_percentage, NULL);
+
+ if (caps && !gst_caps_is_empty (caps)) {
+ const GstStructure *s = gst_caps_get_structure (caps, 0);
+ const gchar *media = gst_structure_get_string (s, "media");
+
+ if (!g_strcmp0 (media, "video"))
+ g_object_set (fecenc, "multipacket", TRUE, NULL);
+ }
+
+ prev = fecenc;
+ }
+
+ if (red_pt) {
+ GstElement *redenc = gst_element_factory_make ("rtpredenc", NULL);
+
+ GST_DEBUG_OBJECT (webrtc, "Creating RED encoder for mline %u with pt %d",
+ rtp_trans->mline, red_pt);
+
+ gst_bin_add (GST_BIN (ret), redenc);
+ if (prev)
+ gst_element_link (prev, redenc);
+ else
+ sinkpad = gst_element_get_static_pad (redenc, "sink");
+
+ g_object_set (redenc, "pt", red_pt, "allow-no-red-blocks", TRUE, NULL);
+
+ prev = redenc;
+ }
+
+ if (sinkpad) {
+ GstPad *ghost = gst_ghost_pad_new ("sink", sinkpad);
+ gst_object_unref (sinkpad);
+ gst_element_add_pad (ret, ghost);
+ }
+
+ if (prev) {
+ GstPad *srcpad = gst_element_get_static_pad (prev, "src");
+ GstPad *ghost = gst_ghost_pad_new ("src", srcpad);
+ gst_object_unref (srcpad);
+ gst_element_add_pad (ret, ghost);
+ }
+
+ return ret;
+}
+
+
static GstPad *
_connect_input_stream (GstWebRTCBin * webrtc, GstWebRTCBinPad * pad)
{
/*
* Not-bundle case:
*
- * ,--------------------------------------------webrtcbin-------------------------,
- * ; ;
- * ; ,-------rtpbin-------, ,--transport_send_%u--, ;
- * ; ; send_rtp_src_%u o---o rtp_sink ; ;
- * ; ,---clocksync---, ; ; ; ; ;
- * ; ; ; ; send_rtcp_src_%u o---o rtcp_sink ; ;
- * ; sink_%u ; ; ; ; '---------------------' ;
- * o---------o sink src o---o send_rtp_sink_%u ; ;
- * ; '---------------' '--------------------' ;
- * '------------------------------------------------------------------------------'
+ * ,--------------------------------------------webrtcbin--------------------------------------------,
+ * ; ;
+ * ; ,-------rtpbin-------, ,--transport_send_%u--, ;
+ * ; ; send_rtp_src_%u o---o rtp_sink ; ;
+ * ; ,---clocksync---, ; ; ; ; ;
+ * ; ; ; ; send_rtcp_src_%u o---o rtcp_sink ; ;
+ * ; sink_%u ; ; ,---fec encoder---, ; ; '---------------------' ;
+ * o---------o sink src o-o sink src o--o send_rtp_sink_%u ; ;
+ * ; '---------------' ,-----------------, '--------------------' ;
+ * '-------------------------------------------------------------------------------------------------'
*/
/*
* Bundle case:
- * ,-----------------------------------------------------webrtcbin--------------------------------,
- * ; ;
- * ; ,-------rtpbin-------, ,--transport_send_%u--, ;
- * ; ; send_rtp_src_%u o---o rtp_sink ; ;
- * ; ; ; ; ; ;
- * ; sink_%u ,---clocksync---, ,---funnel---, ; send_rtcp_src_%u o---o rtcp_sink ; ;
- * o----------o sink src o---o sink_%u ; ; ; '---------------------' ;
- * ; '---------------' ; ; ; ; ;
- * ; ; src o-o send_rtp_sink_%u ; ;
- * ; sink_%u ,---clocksync---, ; ; ; ; ;
- * o----------o sink src o---o sink%u ; '--------------------' ;
- * ; '---------------' '------------' ;
- * '----------------------------------------------------------------------------------------------'
+ * ,-----------------------------------------------------webrtcbin---------------------------------------------------,
+ * ; ;
+ * ; ,-------rtpbin-------, ,--transport_send_%u--, ;
+ * ; ; send_rtp_src_%u o---o rtp_sink ; ;
+ * ; ; ; ; ; ;
+ * ; sink_%u ,---clocksync---, ,---fec encoder---, ,---funnel---, ; send_rtcp_src_%u o---o rtcp_sink ; ;
+ * o----------o sink src o-o sink src o--o sink_%u ; ; ; '---------------------' ;
+ * ; '---------------' ,-----------------, ; ; ; ; ;
+ * ; ; src o-o send_rtp_sink_%u ; ;
+ * ; sink_%u ,---clocksync---, ,---fec encoder---, ; ; ; ; ;
+ * o----------o sink src o-o sink src o--o sink%u ; '--------------------' ;
+ * ; '---------------' ,-----------------, '------------' ;
+ * '-----------------------------------------------------------------------------------------------------------------'
*/
GstPadTemplate *rtp_templ;
GstPad *rtp_sink, *sinkpad, *srcpad;
gchar *pad_name;
WebRTCTransceiver *trans;
GstElement *clocksync;
+ GstElement *fec_encoder;
g_return_val_if_fail (pad->trans != NULL, NULL);
srcpad = gst_element_get_static_pad (clocksync, "src");
sinkpad = gst_element_get_static_pad (clocksync, "sink");
+ if ((fec_encoder = _build_fec_encoder (webrtc, trans))) {
+ GstPad *fec_sink;
+
+ gst_bin_add (GST_BIN (webrtc), fec_encoder);
+ gst_element_sync_state_with_parent (fec_encoder);
+
+ fec_sink = gst_element_get_static_pad (fec_encoder, "sink");
+ gst_pad_link (srcpad, fec_sink);
+ gst_object_unref (srcpad);
+ gst_object_unref (fec_sink);
+ srcpad = gst_element_get_static_pad (fec_encoder, "src");
+ }
+
if (!webrtc->rtpfunnel) {
rtp_templ =
_find_pad_template (webrtc->rtpbin, GST_PAD_SINK, GST_PAD_REQUEST,
}
item.pt = pt;
+ item.media_idx = media_idx;
gst_caps_unref (outcaps);
g_array_append_val (stream->ptmap, item);
stream = _find_transport_for_session (webrtc, session_id);
if (stream)
- have_rtx = transport_stream_get_pt (stream, "RTX") != 0;
+ have_rtx = transport_stream_get_pt (stream, "RTX", -1) != 0;
GST_LOG_OBJECT (webrtc, "requesting aux sender for stream %" GST_PTR_FORMAT
" with pt map %" GST_PTR_FORMAT, stream, pt_map);
GstElement *prev = NULL;
GstPad *sinkpad = NULL;
TransportStream *stream;
- gint red_pt = 0;
gint rtx_pt = 0;
+ GValue red_pt_array = { 0, };
+ gboolean have_red_pt = FALSE;
+
+ g_value_init (&red_pt_array, GST_TYPE_ARRAY);
stream = _find_transport_for_session (webrtc, session_id);
if (stream) {
- red_pt = transport_stream_get_pt (stream, "RED");
- rtx_pt = transport_stream_get_pt (stream, "RTX");
+ guint i = 0;
+
+ for (i = 0; i < stream->ptmap->len; i++) {
+ PtMapItem *item = &g_array_index (stream->ptmap, PtMapItem, i);
+
+ if (!gst_caps_is_empty (item->caps)) {
+ GstStructure *s = gst_caps_get_structure (item->caps, 0);
+
+ if (!g_strcmp0 (gst_structure_get_string (s, "encoding-name"), "RED")) {
+ GValue ptval = { 0, };
+
+ g_value_init (&ptval, G_TYPE_INT);
+ g_value_set_int (&ptval, item->pt);
+ gst_value_array_append_value (&red_pt_array, &ptval);
+ g_value_unset (&ptval);
+
+ have_red_pt = TRUE;
+ }
+ }
+ }
+
+ rtx_pt = transport_stream_get_pt (stream, "RTX", -1);
}
GST_LOG_OBJECT (webrtc, "requesting aux receiver for stream %" GST_PTR_FORMAT,
stream);
- if (red_pt || rtx_pt)
+ if (have_red_pt || rtx_pt)
ret = gst_bin_new (NULL);
if (rtx_pt) {
prev = gst_object_ref (stream->rtxreceive);
}
- if (red_pt) {
+ if (have_red_pt) {
GstElement *rtpreddec = gst_element_factory_make ("rtpreddec", NULL);
- GST_DEBUG_OBJECT (webrtc, "Creating RED decoder for pt %d in session %u",
- red_pt, session_id);
+ GST_DEBUG_OBJECT (webrtc, "Creating RED decoder in session %u", session_id);
gst_bin_add (GST_BIN (ret), rtpreddec);
- g_object_set (rtpreddec, "pt", red_pt, NULL);
+ g_object_set_property (G_OBJECT (rtpreddec), "payloads", &red_pt_array);
if (prev)
gst_element_link (prev, rtpreddec);
}
out:
+ g_value_unset (&red_pt_array);
return ret;
error:
}
static GstElement *
-on_rtpbin_request_fec_decoder (GstElement * rtpbin, guint session_id,
- GstWebRTCBin * webrtc)
+on_rtpbin_request_fec_decoder_full (GstElement * rtpbin, guint session_id,
+ guint ssrc, guint pt, GstWebRTCBin * webrtc)
{
TransportStream *stream;
GstElement *ret = NULL;
- gint pt = 0;
+ gint fec_pt = 0;
GObject *internal_storage;
stream = _find_transport_for_session (webrtc, session_id);
* we detect it (by connecting to ptdemux:new-payload-type for
* example)
*/
- if (stream)
- pt = transport_stream_get_pt (stream, "ULPFEC");
-
- if (pt) {
- GST_DEBUG_OBJECT (webrtc, "Creating ULPFEC decoder for pt %d in session %u",
- pt, session_id);
- ret = gst_element_factory_make ("rtpulpfecdec", NULL);
- g_signal_emit_by_name (webrtc->rtpbin, "get-internal-storage", session_id,
- &internal_storage);
-
- g_object_set (ret, "pt", pt, "storage", internal_storage, NULL);
- g_object_unref (internal_storage);
- }
-
- return ret;
-}
-
-static GstElement *
-on_rtpbin_request_fec_encoder (GstElement * rtpbin, guint session_id,
- GstWebRTCBin * webrtc)
-{
- GstElement *ret = NULL;
- GstElement *prev = NULL;
- TransportStream *stream;
- guint ulpfec_pt = 0;
- guint red_pt = 0;
- GstPad *sinkpad = NULL;
- GstWebRTCRTPTransceiver *trans;
-
- stream = _find_transport_for_session (webrtc, session_id);
- trans = _find_transceiver (webrtc, &session_id,
- (FindTransceiverFunc) transceiver_match_for_mline);
-
if (stream) {
- ulpfec_pt = transport_stream_get_pt (stream, "ULPFEC");
- red_pt = transport_stream_get_pt (stream, "RED");
- }
-
- if (ulpfec_pt || red_pt)
- ret = gst_bin_new (NULL);
-
- if (ulpfec_pt) {
- GstElement *fecenc = gst_element_factory_make ("rtpulpfecenc", NULL);
- GstCaps *caps = transport_stream_get_caps_for_pt (stream, ulpfec_pt);
-
- GST_DEBUG_OBJECT (webrtc,
- "Creating ULPFEC encoder for session %d with pt %d", session_id,
- ulpfec_pt);
-
- gst_bin_add (GST_BIN (ret), fecenc);
- sinkpad = gst_element_get_static_pad (fecenc, "sink");
- g_object_set (fecenc, "pt", ulpfec_pt, "percentage",
- WEBRTC_TRANSCEIVER (trans)->fec_percentage, NULL);
-
+ guint i;
- if (caps && !gst_caps_is_empty (caps)) {
- const GstStructure *s = gst_caps_get_structure (caps, 0);
- const gchar *media = gst_structure_get_string (s, "media");
+ for (i = 0; i < stream->ptmap->len; i++) {
+ PtMapItem *item = &g_array_index (stream->ptmap, PtMapItem, i);
- if (!g_strcmp0 (media, "video"))
- g_object_set (fecenc, "multipacket", TRUE, NULL);
+ if (item->pt == pt) {
+ fec_pt = transport_stream_get_pt (stream, "ULPFEC", item->media_idx);
+ break;
+ }
}
-
- prev = fecenc;
}
- if (red_pt) {
- GstElement *redenc = gst_element_factory_make ("rtpredenc", NULL);
-
- GST_DEBUG_OBJECT (webrtc, "Creating RED encoder for session %d with pt %d",
- session_id, red_pt);
-
- gst_bin_add (GST_BIN (ret), redenc);
- if (prev)
- gst_element_link (prev, redenc);
- else
- sinkpad = gst_element_get_static_pad (redenc, "sink");
-
- g_object_set (redenc, "pt", red_pt, "allow-no-red-blocks", TRUE, NULL);
-
- prev = redenc;
- }
-
- if (sinkpad) {
- GstPad *ghost = gst_ghost_pad_new ("sink", sinkpad);
- gst_object_unref (sinkpad);
- gst_element_add_pad (ret, ghost);
- }
+ if (fec_pt) {
+ GST_DEBUG_OBJECT (webrtc, "Creating ULPFEC decoder for pt %d in session %u",
+ fec_pt, session_id);
+ ret = gst_element_factory_make ("rtpulpfecdec", NULL);
+ g_signal_emit_by_name (webrtc->rtpbin, "get-internal-storage", session_id,
+ &internal_storage);
- if (prev) {
- GstPad *srcpad = gst_element_get_static_pad (prev, "src");
- GstPad *ghost = gst_ghost_pad_new ("src", srcpad);
- gst_object_unref (srcpad);
- gst_element_add_pad (ret, ghost);
+ g_object_set (ret, "pt", fec_pt, "storage", internal_storage, NULL);
+ g_object_unref (internal_storage);
}
return ret;
G_CALLBACK (on_rtpbin_request_aux_receiver), webrtc);
g_signal_connect (rtpbin, "new-storage",
G_CALLBACK (on_rtpbin_new_storage), webrtc);
- g_signal_connect (rtpbin, "request-fec-decoder",
- G_CALLBACK (on_rtpbin_request_fec_decoder), webrtc);
- g_signal_connect (rtpbin, "request-fec-encoder",
- G_CALLBACK (on_rtpbin_request_fec_encoder), webrtc);
+ g_signal_connect (rtpbin, "request-fec-decoder-full",
+ G_CALLBACK (on_rtpbin_request_fec_decoder_full), webrtc);
g_signal_connect (rtpbin, "on-bye-ssrc",
G_CALLBACK (on_rtpbin_bye_ssrc), webrtc);
g_signal_connect (rtpbin, "on-bye-timeout",
projects / platform / upstream / gstreamer.git / commitdiff