static GstFlowReturn gst_rtp_session_send_rtp (RTPSession * sess,
RTPSource * src, gpointer data, gpointer user_data);
static GstFlowReturn gst_rtp_session_send_rtcp (RTPSession * sess,
- RTPSource * src, GstBuffer * buffer, gpointer user_data);
+ RTPSource * src, GstBuffer * buffer, gboolean eos, gpointer user_data);
static GstFlowReturn gst_rtp_session_sync_rtcp (RTPSession * sess,
GstBuffer * buffer, gpointer user_data);
static gint gst_rtp_session_clock_rate (RTPSession * sess, guint8 payload,
GST_RTP_SESSION_UNLOCK (rtpsession);
rtp_session_on_timeout (session, current_time, ntpnstime, running_time);
GST_RTP_SESSION_LOCK (rtpsession);
-
- if (!rtp_session_get_num_sources (session)) {
- /* when no sources left in the session, all of the them have went
- * BYE at some point and removed, we can send EOS to the
- * pipeline. */
- GstPad *rtcp_src = rtpsession->send_rtcp_src;
-
- if (rtcp_src) {
- gst_object_ref (rtcp_src);
- GST_LOG_OBJECT (rtpsession, "sending EOS");
- GST_RTP_SESSION_UNLOCK (rtpsession);
- gst_pad_push_event (rtpsession->send_rtcp_src, gst_event_new_eos ());
- GST_RTP_SESSION_LOCK (rtpsession);
- gst_object_unref (rtcp_src);
- }
- }
}
/* mark the thread as stopped now */
rtpsession->priv->thread_stopped = TRUE;
}
/* called when the session manager has an RTCP packet ready for further
- * sending. */
+ * sending. The eos flag is set when an EOS event should be sent downstream as
+ * well. */
static GstFlowReturn
gst_rtp_session_send_rtcp (RTPSession * sess, RTPSource * src,
- GstBuffer * buffer, gpointer user_data)
+ GstBuffer * buffer, gboolean eos, gpointer user_data)
{
GstFlowReturn result;
GstRtpSession *rtpsession;
GST_LOG_OBJECT (rtpsession, "sending RTCP");
result = gst_pad_push (rtcp_src, buffer);
+ /* we have to send EOS after this packet */
+ if (eos) {
+ GST_LOG_OBJECT (rtpsession, "sending EOS");
+ gst_pad_push_event (rtcp_src, gst_event_new_eos ());
+ }
gst_object_unref (rtcp_src);
} else {
GST_RTP_SESSION_UNLOCK (rtpsession);
return ret;
}
+
static gboolean
gst_rtp_session_event_send_rtp_sink (GstPad * pad, GstObject * parent,
GstEvent * event)
typedef struct
{
RTPSource *source;
+ gboolean is_bye;
GstBuffer *buffer;
} ReportOutput;
generate_rtcp (const gchar * key, RTPSource * source, ReportData * data)
{
RTPSession *sess = data->sess;
+ gboolean is_bye = FALSE;
ReportOutput *output;
/* only generate RTCP for active internal sources */
if (source->marked_bye) {
/* send BYE */
make_source_bye (sess, source, data);
+ is_bye = TRUE;
} else if (!data->is_early) {
/* loop over all known sources and add report blocks. If we are early, we
* just make a minimal RTCP packet and skip this step */
output = g_slice_new (ReportOutput);
output->source = g_object_ref (source);
+ output->is_bye = is_bye;
output->buffer = data->rtcp;
/* queue the RTCP packet to push later */
g_queue_push_tail (&data->output, output);
GST_DEBUG ("%p, sending RTCP packet, avg size %u, %u", &sess->stats,
sess->stats.avg_rtcp_packet_size, packet_size);
result =
- sess->callbacks.send_rtcp (sess, source, buffer,
+ sess->callbacks.send_rtcp (sess, source, buffer, output->is_bye,
sess->send_rtcp_user_data);
RTP_SESSION_LOCK (sess);