diff options
| -rw-r--r-- | ext/webrtc/gstwebrtcbin.c | 45 |
1 files changed, 32 insertions, 13 deletions
diff --git a/ext/webrtc/gstwebrtcbin.c b/ext/webrtc/gstwebrtcbin.c index 1fcb6bf86..fdc53f52c 100644 --- a/ext/webrtc/gstwebrtcbin.c +++ b/ext/webrtc/gstwebrtcbin.c @@ -6412,28 +6412,47 @@ static void on_rtpbin_new_jitterbuffer (GstElement * rtpbin, GstElement * jitterbuffer, guint session_id, guint ssrc, GstWebRTCBin * webrtc) { - WebRTCTransceiver *trans; + TransportStream *stream; guint i; - trans = (WebRTCTransceiver *) _find_transceiver (webrtc, &session_id, - (FindTransceiverFunc) transceiver_match_for_mline); + PC_LOCK (webrtc); + GST_INFO_OBJECT (webrtc, "new jitterbuffer %" GST_PTR_FORMAT " for " + "session %u ssrc %u", jitterbuffer, session_id, ssrc); - if (trans) { - /* We don't set do-retransmission on rtpbin as we want per-session control */ - g_object_set (jitterbuffer, "do-retransmission", - WEBRTC_TRANSCEIVER (trans)->do_nack, NULL); + if (!(stream = _find_transport_for_session (webrtc, session_id))) { + g_warn_if_reached (); + goto out; + } - for (i = 0; i < trans->stream->remote_ssrcmap->len; i++) { - SsrcMapItem *item = g_ptr_array_index (trans->stream->remote_ssrcmap, i); + /* XXX: this will fail with no ssrc in the remote sdp as used with e.g. simulcast + * newer SDP versions from chrome/firefox */ + for (i = 0; i < stream->remote_ssrcmap->len; i++) { + SsrcMapItem *item = g_ptr_array_index (stream->remote_ssrcmap, i); - if (item->ssrc == ssrc) { - g_weak_ref_set (&item->rtpjitterbuffer, jitterbuffer); + if (item->ssrc == ssrc) { + GstWebRTCRTPTransceiver *trans; + gboolean do_nack; + + trans = _find_transceiver_for_mline (webrtc, item->media_idx); + if (!trans) { + g_warn_if_reached (); break; } + + do_nack = WEBRTC_TRANSCEIVER (trans)->do_nack; + /* We don't set do-retransmission on rtpbin as we want per-session control */ + GST_LOG_OBJECT (webrtc, "setting do-nack=%s for transceiver %" + GST_PTR_FORMAT " with transport %" GST_PTR_FORMAT + " rtp session %u ssrc %u", do_nack ? "true" : "false", trans, stream, + session_id, ssrc); + g_object_set (jitterbuffer, "do-retransmission", do_nack, NULL); + + g_weak_ref_set (&item->rtpjitterbuffer, jitterbuffer); + break; } - } else { - g_assert_not_reached (); } +out: + PC_UNLOCK (webrtc); } static void |