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rtpsession: delay RTCP until first RTP packet

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Delay sending the first RTCP packet until we have sent the first RTP packet.
Otherwise we will send out a Receiver Report instead of a sender report.

See https://bugzilla.gnome.org/show_bug.cgi?id=691400
This commit is contained in:
Wim Taymans 2013-02-08 17:05:27 +01:00
parent 2971ed44ee
commit 2d5319c1fa
1 changed files with 23 additions and 0 deletions
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23
gst/rtpmanager/gstrtpsession.c
View file

@ -226,9 +226,13 @@ enum
#define GST_RTP_SESSION_LOCK(sess) g_mutex_lock (&(sess)->priv->lock)
#define GST_RTP_SESSION_UNLOCK(sess) g_mutex_unlock (&(sess)->priv->lock)
#define GST_RTP_SESSION_WAIT(sess) g_cond_wait (&(sess)->priv->cond, &(sess)->priv->lock)
#define GST_RTP_SESSION_SIGNAL(sess) g_cond_signal (&(sess)->priv->cond)
struct _GstRtpSessionPrivate
{
GMutex lock;
GCond cond;
GstClock *sysclock;
RTPSession *session;
@ -238,6 +242,7 @@ struct _GstRtpSessionPrivate
gboolean stop_thread;
GThread *thread;
gboolean thread_stopped;
gboolean wait_send;
/* caps mapping */
GHashTable *ptmap;
@ -622,6 +627,7 @@ gst_rtp_session_init (GstRtpSession * rtpsession)
{
rtpsession->priv = GST_RTP_SESSION_GET_PRIVATE (rtpsession);
g_mutex_init (&rtpsession->priv->lock);
g_cond_init (&rtpsession->priv->cond);
rtpsession->priv->sysclock = gst_system_clock_obtain ();
rtpsession->priv->session = rtp_session_new ();
rtpsession->priv->use_pipeline_clock = DEFAULT_USE_PIPELINE_CLOCK;
@ -665,6 +671,7 @@ gst_rtp_session_finalize (GObject * object)
g_hash_table_destroy (rtpsession->priv->ptmap);
g_mutex_clear (&rtpsession->priv->lock);
g_cond_clear (&rtpsession->priv->cond);
g_object_unref (rtpsession->priv->sysclock);
g_object_unref (rtpsession->priv->session);
@ -827,6 +834,12 @@ rtcp_thread (GstRtpSession * rtpsession)
GST_RTP_SESSION_LOCK (rtpsession);
while (rtpsession->priv->wait_send) {
GST_LOG_OBJECT (rtpsession, "waiting for RTP thread");
GST_RTP_SESSION_WAIT (rtpsession);
GST_LOG_OBJECT (rtpsession, "signaled...");
}
sysclock = rtpsession->priv->sysclock;
current_time = gst_clock_get_time (sysclock);
@ -926,6 +939,8 @@ stop_rtcp_thread (GstRtpSession * rtpsession)
GST_RTP_SESSION_LOCK (rtpsession);
rtpsession->priv->stop_thread = TRUE;
rtpsession->priv->wait_send = FALSE;
GST_RTP_SESSION_SIGNAL (rtpsession);
if (rtpsession->priv->id)
gst_clock_id_unschedule (rtpsession->priv->id);
GST_RTP_SESSION_UNLOCK (rtpsession);
@ -962,6 +977,9 @@ gst_rtp_session_change_state (GstElement * element, GstStateChange transition)
case GST_STATE_CHANGE_NULL_TO_READY:
break;
case GST_STATE_CHANGE_READY_TO_PAUSED:
GST_RTP_SESSION_LOCK (rtpsession);
rtpsession->priv->wait_send = TRUE;
GST_RTP_SESSION_UNLOCK (rtpsession);
break;
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
break;
@ -1059,6 +1077,11 @@ gst_rtp_session_send_rtp (RTPSession * sess, RTPSource * src,
GST_RTP_SESSION_LOCK (rtpsession);
if ((rtp_src = rtpsession->send_rtp_src))
gst_object_ref (rtp_src);
if (rtpsession->priv->wait_send) {
GST_LOG_OBJECT (rtpsession, "signal RTCP thread");
rtpsession->priv->wait_send = FALSE;
GST_RTP_SESSION_SIGNAL (rtpsession);
}
GST_RTP_SESSION_UNLOCK (rtpsession);
if (rtp_src) {