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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2025-01-23 07:38:16 +00:00
webrtc: fix rtx + bundle
If bundle was used in combination with rtx, only the bundled transport stream would have correctly configured rtx parameters. Iterate over the payloads upfront in the bundled case to ensure the correct payload mapping is set for the RTX elements.
This commit is contained in:
Matthew Waters
parent
06b18defc7
commit
2a1176973a
1 changed files with 85 additions and 71 deletions
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@ -2961,6 +2961,75 @@ _filter_sdp_fields (GQuark field_id, const GValue * value,
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return TRUE;
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}
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static void
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_update_transport_ptmap_from_media (GstWebRTCBin * webrtc,
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TransportStream * stream, const GstSDPMessage * sdp, guint media_idx)
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{
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guint i, len;
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const gchar *proto;
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const GstSDPMedia *media = gst_sdp_message_get_media (sdp, media_idx);
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/* get proto */
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proto = gst_sdp_media_get_proto (media);
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if (proto != NULL) {
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/* Parse global SDP attributes once */
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GstCaps *global_caps = gst_caps_new_empty_simple ("application/x-unknown");
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GST_DEBUG_OBJECT (webrtc, "mapping sdp session level attributes to caps");
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gst_sdp_message_attributes_to_caps (sdp, global_caps);
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GST_DEBUG_OBJECT (webrtc, "mapping sdp media level attributes to caps");
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gst_sdp_media_attributes_to_caps (media, global_caps);
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len = gst_sdp_media_formats_len (media);
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for (i = 0; i < len; i++) {
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GstCaps *caps, *outcaps;
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GstStructure *s;
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PtMapItem item;
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gint pt;
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guint j;
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pt = atoi (gst_sdp_media_get_format (media, i));
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GST_DEBUG_OBJECT (webrtc, " looking at %d pt: %d", i, pt);
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/* convert caps */
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caps = gst_sdp_media_get_caps_from_media (media, pt);
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if (caps == NULL) {
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GST_WARNING_OBJECT (webrtc, " skipping pt %d without caps", pt);
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continue;
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}
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/* Merge in global caps */
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/* Intersect will merge in missing fields to the current caps */
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outcaps = gst_caps_intersect (caps, global_caps);
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gst_caps_unref (caps);
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s = gst_caps_get_structure (outcaps, 0);
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gst_structure_set_name (s, "application/x-rtp");
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if (!g_strcmp0 (gst_structure_get_string (s, "encoding-name"), "ULPFEC"))
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gst_structure_set (s, "is-fec", G_TYPE_BOOLEAN, TRUE, NULL);
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item.caps = gst_caps_new_empty ();
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for (j = 0; j < gst_caps_get_size (outcaps); j++) {
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GstStructure *s = gst_caps_get_structure (outcaps, j);
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GstStructure *filtered =
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gst_structure_new_empty (gst_structure_get_name (s));
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gst_structure_foreach (s,
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(GstStructureForeachFunc) _filter_sdp_fields, filtered);
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gst_caps_append_structure (item.caps, filtered);
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}
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item.pt = pt;
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gst_caps_unref (outcaps);
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g_array_append_val (stream->ptmap, item);
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}
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gst_caps_unref (global_caps);
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}
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}
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static void
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_update_transceiver_from_sdp_media (GstWebRTCBin * webrtc,
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const GstSDPMessage * sdp, guint media_idx,
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@ -2990,9 +3059,6 @@ _update_transceiver_from_sdp_media (GstWebRTCBin * webrtc,
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const GstSDPMedia *local_media, *remote_media;
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GstWebRTCRTPTransceiverDirection local_dir, remote_dir;
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GstWebRTCDTLSSetup local_setup, remote_setup;
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guint i, len;
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const gchar *proto;
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GstCaps *global_caps;
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local_media =
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gst_sdp_message_get_media (webrtc->current_local_description->sdp,
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@ -3014,72 +3080,6 @@ _update_transceiver_from_sdp_media (GstWebRTCBin * webrtc,
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if (new_dir == GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_NONE)
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return;
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/* get proto */
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proto = gst_sdp_media_get_proto (media);
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if (proto != NULL) {
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/* Parse global SDP attributes once */
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global_caps = gst_caps_new_empty_simple ("application/x-unknown");
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GST_DEBUG_OBJECT (webrtc, "mapping sdp session level attributes to caps");
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gst_sdp_message_attributes_to_caps (sdp, global_caps);
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GST_DEBUG_OBJECT (webrtc, "mapping sdp media level attributes to caps");
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gst_sdp_media_attributes_to_caps (media, global_caps);
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if (!bundled) {
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/* clear the ptmap */
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g_array_set_size (stream->ptmap, 0);
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}
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len = gst_sdp_media_formats_len (media);
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for (i = 0; i < len; i++) {
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GstCaps *caps, *outcaps;
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GstStructure *s;
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PtMapItem item;
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gint pt;
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guint j;
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pt = atoi (gst_sdp_media_get_format (media, i));
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GST_DEBUG_OBJECT (webrtc, " looking at %d pt: %d", i, pt);
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/* convert caps */
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caps = gst_sdp_media_get_caps_from_media (media, pt);
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if (caps == NULL) {
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GST_WARNING_OBJECT (webrtc, " skipping pt %d without caps", pt);
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continue;
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}
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/* Merge in global caps */
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/* Intersect will merge in missing fields to the current caps */
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outcaps = gst_caps_intersect (caps, global_caps);
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gst_caps_unref (caps);
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s = gst_caps_get_structure (outcaps, 0);
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gst_structure_set_name (s, "application/x-rtp");
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if (!g_strcmp0 (gst_structure_get_string (s, "encoding-name"),
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"ULPFEC"))
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gst_structure_set (s, "is-fec", G_TYPE_BOOLEAN, TRUE, NULL);
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item.caps = gst_caps_new_empty ();
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for (j = 0; j < gst_caps_get_size (outcaps); j++) {
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GstStructure *s = gst_caps_get_structure (outcaps, j);
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GstStructure *filtered =
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gst_structure_new_empty (gst_structure_get_name (s));
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gst_structure_foreach (s,
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(GstStructureForeachFunc) _filter_sdp_fields, filtered);
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gst_caps_append_structure (item.caps, filtered);
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}
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item.pt = pt;
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gst_caps_unref (outcaps);
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g_array_append_val (stream->ptmap, item);
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}
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gst_caps_unref (global_caps);
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}
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if (prev_dir != GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_NONE
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&& prev_dir != new_dir) {
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GST_FIXME_OBJECT (webrtc, "implement transceiver direction changes");
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@ -3381,9 +3381,12 @@ _update_transceivers_from_sdp (GstWebRTCBin * webrtc, SDPSource source,
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bundle_stream = _get_or_create_transport_stream (webrtc, bundle_idx,
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_message_media_is_datachannel (sdp->sdp, bundle_idx));
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g_array_set_size (bundle_stream->ptmap, 0);
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_connect_rtpfunnel (webrtc, bundle_idx);
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g_array_set_size (bundle_stream->ptmap, 0);
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for (i = 0; i < gst_sdp_message_medias_len (sdp->sdp); i++) {
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_update_transport_ptmap_from_media (webrtc, bundle_stream, sdp->sdp, i);
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}
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}
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for (i = 0; i < gst_sdp_message_medias_len (sdp->sdp); i++) {
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@ -3405,6 +3408,10 @@ _update_transceivers_from_sdp (GstWebRTCBin * webrtc, SDPSource source,
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stream = _get_or_create_transport_stream (webrtc, transport_idx,
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_message_media_is_datachannel (sdp->sdp, transport_idx));
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if (!bundled) {
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g_array_set_size (stream->ptmap, 0);
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_update_transport_ptmap_from_media (webrtc, stream, sdp->sdp, i);
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}
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if (trans)
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webrtc_transceiver_set_transport ((WebRTCTransceiver *) trans, stream);
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@ -4277,6 +4284,10 @@ on_rtpbin_request_aux_sender (GstElement * rtpbin, guint session_id,
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}
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}
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GST_LOG_OBJECT (webrtc, "requesting aux sender for stream %" GST_PTR_FORMAT
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" with transport %" GST_PTR_FORMAT " and pt map %" GST_PTR_FORMAT, stream,
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trans, pt_map);
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if (gst_structure_n_fields (pt_map)) {
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GstElement *rtx;
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GstPad *pad;
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@ -4330,6 +4341,9 @@ on_rtpbin_request_aux_receiver (GstElement * rtpbin, guint session_id,
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rtx_pt = transport_stream_get_pt (stream, "RTX");
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}
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GST_LOG_OBJECT (webrtc, "requesting aux receiver for stream %" GST_PTR_FORMAT
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" with pt red:%u rtx:%u", stream, red_pt, rtx_pt);
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if (red_pt || rtx_pt)
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ret = gst_bin_new (NULL);
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