if (source) {
rtp_source_update_caps (source, caps);
g_object_unref (source);
+
+ if (created)
+ on_new_ssrc (sess, source);
}
if (gst_structure_get_uint (s, "rtx-ssrc", &ssrc)) {
source = obtain_internal_source (sess, pinfo.ssrc, &created, current_time);
+ if (created)
+ on_new_ssrc (sess, source);
+
prevsender = RTP_SOURCE_IS_SENDER (source);
oldrate = source->bitrate;
source = obtain_internal_source (sess, sess->suggested_ssrc, &created,
current_time);
+
+ if (created)
+ on_new_ssrc (sess, source);
g_object_unref (source);
}
sess->callbacks.send_rtcp (sess, source, buffer, output->is_bye,
sess->send_rtcp_user_data);
sess->stats.nacks_sent += data.nacked_seqnums;
+
+ RTP_SESSION_LOCK (sess);
+ on_ssrc_active (sess, source);
+ RTP_SESSION_UNLOCK (sess);
} else {
GST_DEBUG ("freeing packet callback: %p"
" do_not_suppress: %d may_suppress: %d",