mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2025-01-26 17:18:15 +00:00
gst/rtpmanager/gstrtpsession.c: Send EOS when the session object instructs us to.
Original commit message from CVS: * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_send_rtcp), (gst_rtp_session_event_send_rtp_sink): Send EOS when the session object instructs us to. * gst/rtpmanager/rtpsession.c: (rtp_session_on_timeout): * gst/rtpmanager/rtpsession.h: Make it possible for the session manager to instruct us to send EOS. We currently will EOS when the session is a sender and when the sender part goes EOS. This is not entirely correct behaviour because the session could still participate as a receiver. Fixes #549409.
This commit is contained in:
parent
62ecaee748
commit
5c89bb2ab3
3 changed files with 22 additions and 6 deletions
|
@ -259,7 +259,7 @@ static GstFlowReturn gst_rtp_session_process_rtp (RTPSession * sess,
|
|||
static GstFlowReturn gst_rtp_session_send_rtp (RTPSession * sess,
|
||||
RTPSource * src, GstBuffer * buffer, 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,
|
||||
RTPSource * src, GstBuffer * buffer, gpointer user_data);
|
||||
static gint gst_rtp_session_clock_rate (RTPSession * sess, guint8 payload,
|
||||
|
@ -1098,10 +1098,11 @@ gst_rtp_session_send_rtp (RTPSession * sess, RTPSource * src,
|
|||
}
|
||||
|
||||
/* 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;
|
||||
|
@ -1122,6 +1123,12 @@ gst_rtp_session_send_rtcp (RTPSession * sess, RTPSource * src,
|
|||
gst_buffer_set_caps (buffer, caps);
|
||||
GST_LOG_OBJECT (rtpsession, "sending RTCP");
|
||||
result = gst_pad_push (rtpsession->send_rtcp_src, buffer);
|
||||
|
||||
/* we have to send EOS after this packet */
|
||||
if (eos) {
|
||||
GST_LOG_OBJECT (rtpsession, "sending EOS");
|
||||
gst_pad_push_event (rtpsession->send_rtcp_src, gst_event_new_eos ());
|
||||
}
|
||||
} else {
|
||||
GST_DEBUG_OBJECT (rtpsession, "not sending RTCP, no output pad");
|
||||
gst_buffer_unref (buffer);
|
||||
|
@ -1557,8 +1564,11 @@ gst_rtp_session_event_send_rtp_sink (GstPad * pad, GstEvent * event)
|
|||
case GST_EVENT_EOS:{
|
||||
GstClockTime current_time;
|
||||
|
||||
/* push downstream FIXME, we are not supposed to leave the session just
|
||||
* because we stop sending. */
|
||||
ret = gst_pad_push_event (rtpsession->send_rtp_src, event);
|
||||
current_time = gst_clock_get_time (rtpsession->priv->sysclock);
|
||||
GST_DEBUG_OBJECT (rtpsession, "scheduling BYE message");
|
||||
rtp_session_send_bye (rtpsession->priv->session, "End of stream",
|
||||
current_time);
|
||||
break;
|
||||
|
|
|
@ -2300,6 +2300,7 @@ rtp_session_on_timeout (RTPSession * sess, GstClockTime current_time,
|
|||
if (is_rtcp_time (sess, current_time, &data)) {
|
||||
if (sess->source->received_bye) {
|
||||
/* generate BYE instead */
|
||||
GST_DEBUG ("generating BYE message");
|
||||
session_bye (sess, &data);
|
||||
sess->sent_bye = TRUE;
|
||||
} else {
|
||||
|
@ -2367,11 +2368,14 @@ rtp_session_on_timeout (RTPSession * sess, GstClockTime current_time,
|
|||
/* close the RTCP packet */
|
||||
gst_rtcp_buffer_end (data.rtcp);
|
||||
|
||||
GST_DEBUG ("sending packet");
|
||||
if (sess->callbacks.send_rtcp)
|
||||
result = sess->callbacks.send_rtcp (sess, sess->source, data.rtcp,
|
||||
sess->send_rtcp_user_data);
|
||||
else
|
||||
sess->sent_bye, sess->send_rtcp_user_data);
|
||||
else {
|
||||
GST_DEBUG ("freeing packet");
|
||||
gst_buffer_unref (data.rtcp);
|
||||
}
|
||||
}
|
||||
|
||||
return result;
|
||||
|
|
|
@ -72,6 +72,7 @@ typedef GstFlowReturn (*RTPSessionSendRTP) (RTPSession *sess, RTPSource *src, Gs
|
|||
* @sess: an #RTPSession
|
||||
* @src: the #RTPSource
|
||||
* @buffer: the RTCP buffer ready for sending
|
||||
* @eos: if an EOS event should be pushed
|
||||
* @user_data: user data specified when registering
|
||||
*
|
||||
* This callback will be called when @sess has @buffer ready for sending to
|
||||
|
@ -79,7 +80,8 @@ typedef GstFlowReturn (*RTPSessionSendRTP) (RTPSession *sess, RTPSource *src, Gs
|
|||
*
|
||||
* Returns: a #GstFlowReturn.
|
||||
*/
|
||||
typedef GstFlowReturn (*RTPSessionSendRTCP) (RTPSession *sess, RTPSource *src, GstBuffer *buffer, gpointer user_data);
|
||||
typedef GstFlowReturn (*RTPSessionSendRTCP) (RTPSession *sess, RTPSource *src, GstBuffer *buffer,
|
||||
gboolean eos, gpointer user_data);
|
||||
|
||||
/**
|
||||
* RTPSessionSyncRTCP:
|
||||
|
|
Loading…
Reference in a new issue