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-rw-r--r--ChangeLog17
-rw-r--r--gst/rtpmanager/gstrtpjitterbuffer.c33
-rw-r--r--gst/rtpmanager/gstrtpsession.c18
-rw-r--r--gst/rtpmanager/rtpsession.c12
4 files changed, 66 insertions, 14 deletions
diff --git a/ChangeLog b/ChangeLog
index e1ee0b64..a96e06d7 100644
--- a/ChangeLog
+++ b/ChangeLog
@@ -1,5 +1,22 @@
2007-08-16 Wim Taymans <wim.taymans@gmail.com>
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ (gst_rtp_jitter_buffer_sink_event), (gst_rtp_jitter_buffer_chain),
+ (gst_rtp_jitter_buffer_loop):
+ Fix EOS handling.
+ Convert some DEBUG into WARNINGs.
+ Pause task when flushing.
+
+ * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_class_init),
+ (rtcp_thread), (gst_rtp_session_event_recv_rtcp_sink):
+ Use system clock for RTCP session management timeouts.
+
+ * gst/rtpmanager/rtpsession.c: (on_new_ssrc), (on_ssrc_collision),
+ (on_ssrc_validated), (on_bye_ssrc), (on_bye_timeout), (on_timeout):
+ Release the session lock when emiting signals.
+
+2007-08-16 Wim Taymans <wim.taymans@gmail.com>
+
* ext/faad/gstfaad.c: (gst_faad_setcaps),
(gst_faad_chanpos_to_gst):
Add some debug info.
diff --git a/gst/rtpmanager/gstrtpjitterbuffer.c b/gst/rtpmanager/gstrtpjitterbuffer.c
index fe85f87f..e66613bb 100644
--- a/gst/rtpmanager/gstrtpjitterbuffer.c
+++ b/gst/rtpmanager/gstrtpjitterbuffer.c
@@ -696,16 +696,18 @@ gst_rtp_jitter_buffer_sink_event (GstPad * pad, GstEvent * event)
/* check for flushing, we need to discard the event and return FALSE when
* we are flushing */
ret = priv->srcresult == GST_FLOW_OK;
- if (ret) {
+ if (ret && !priv->eos) {
GST_DEBUG_OBJECT (jitterbuffer, "queuing EOS");
priv->eos = TRUE;
JBUF_SIGNAL (priv);
+ } else if (priv->eos) {
+ GST_DEBUG_OBJECT (jitterbuffer, "dropping EOS, we are already EOS");
} else {
GST_DEBUG_OBJECT (jitterbuffer, "dropping EOS, reason %s",
gst_flow_get_name (priv->srcresult));
- gst_event_unref (event);
}
JBUF_UNLOCK (priv);
+ gst_event_unref (event);
break;
}
default:
@@ -863,7 +865,7 @@ invalid_buffer:
}
not_negotiated:
{
- GST_DEBUG_OBJECT (jitterbuffer, "No clock-rate in caps!");
+ GST_WARNING_OBJECT (jitterbuffer, "No clock-rate in caps!");
gst_buffer_unref (buffer);
gst_object_unref (jitterbuffer);
return GST_FLOW_NOT_NEGOTIATED;
@@ -878,13 +880,13 @@ out_flushing:
have_eos:
{
ret = GST_FLOW_UNEXPECTED;
- GST_DEBUG_OBJECT (jitterbuffer, "we are EOS, refusing buffer");
+ GST_WARNING_OBJECT (jitterbuffer, "we are EOS, refusing buffer");
gst_buffer_unref (buffer);
goto finished;
}
too_late:
{
- GST_DEBUG_OBJECT (jitterbuffer, "Packet #%d too late as #%d was already"
+ GST_WARNING_OBJECT (jitterbuffer, "Packet #%d too late as #%d was already"
" popped, dropping", seqnum, priv->last_popped_seqnum);
priv->num_late++;
gst_buffer_unref (buffer);
@@ -892,7 +894,7 @@ too_late:
}
duplicate:
{
- GST_DEBUG_OBJECT (jitterbuffer, "Duplicate packet #%d detected, dropping",
+ GST_WARNING_OBJECT (jitterbuffer, "Duplicate packet #%d detected, dropping",
seqnum);
priv->num_duplicates++;
gst_buffer_unref (buffer);
@@ -923,13 +925,19 @@ gst_rtp_jitter_buffer_loop (GstRTPJitterBuffer * jitterbuffer)
JBUF_LOCK_CHECK (priv, flushing);
again:
GST_DEBUG_OBJECT (jitterbuffer, "Popping item");
- /* wait if we are blocked or don't have a packet and eos */
- while (priv->blocked || !(rtp_jitter_buffer_num_packets (priv->jbuf)
- || priv->eos)) {
+ while (TRUE) {
+
+ /* always wait if we are blocked */
+ if (!priv->blocked) {
+ /* if we have a packet, we can grab it */
+ if (rtp_jitter_buffer_num_packets (priv->jbuf) > 0)
+ break;
+ /* no packets but we are EOS, do eos logic */
+ if (priv->eos)
+ goto do_eos;
+ }
JBUF_WAIT_CHECK (priv, flushing);
}
- if (priv->eos)
- goto do_eos;
/* pop a buffer, we must have a buffer now */
outbuf = rtp_jitter_buffer_pop (priv->jbuf);
@@ -955,7 +963,7 @@ again:
if (priv->next_seqnum != -1) {
/* we expected next_seqnum but received something else, that's a gap */
- GST_DEBUG_OBJECT (jitterbuffer,
+ GST_WARNING_OBJECT (jitterbuffer,
"Sequence number GAP detected -> %d instead of %d", priv->next_seqnum,
seqnum);
} else {
@@ -1092,6 +1100,7 @@ do_eos:
flushing:
{
GST_DEBUG_OBJECT (jitterbuffer, "we are flushing");
+ gst_pad_pause_task (priv->srcpad);
if (outbuf)
gst_buffer_unref (outbuf);
JBUF_UNLOCK (priv);
diff --git a/gst/rtpmanager/gstrtpsession.c b/gst/rtpmanager/gstrtpsession.c
index bb47a29e..01153e23 100644
--- a/gst/rtpmanager/gstrtpsession.c
+++ b/gst/rtpmanager/gstrtpsession.c
@@ -537,9 +537,10 @@ rtcp_thread (GstRTPSession * rtpsession)
GstClockTime current_time;
GstClockTime next_timeout;
- clock = gst_element_get_clock (GST_ELEMENT_CAST (rtpsession));
+ /* RTCP timeouts we use the system clock */
+ clock = gst_system_clock_obtain ();
if (clock == NULL)
- return;
+ goto no_clock;
current_time = gst_clock_get_time (clock);
@@ -590,6 +591,15 @@ rtcp_thread (GstRTPSession * rtpsession)
gst_object_unref (clock);
GST_DEBUG_OBJECT (rtpsession, "leaving RTCP thread");
+ return;
+
+ /* ERRORS */
+no_clock:
+ {
+ GST_ELEMENT_ERROR (rtpsession, CORE, CLOCK, (NULL),
+ ("Could not get system clock"));
+ return;
+ }
}
static gboolean
@@ -900,6 +910,10 @@ gst_rtp_session_event_recv_rtcp_sink (GstPad * pad, GstEvent * event)
switch (GST_EVENT_TYPE (event)) {
default:
+ if (rtpsession->send_rtcp_src) {
+ gst_event_ref (event);
+ ret = gst_pad_push_event (rtpsession->send_rtcp_src, event);
+ }
ret = gst_pad_push_event (rtpsession->sync_src, event);
break;
}
diff --git a/gst/rtpmanager/rtpsession.c b/gst/rtpmanager/rtpsession.c
index 2b3bcb82..9ab3b4a0 100644
--- a/gst/rtpmanager/rtpsession.c
+++ b/gst/rtpmanager/rtpsession.c
@@ -251,39 +251,51 @@ rtp_session_get_property (GObject * object, guint prop_id,
static void
on_new_ssrc (RTPSession * sess, RTPSource * source)
{
+ RTP_SESSION_UNLOCK (sess);
g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_NEW_SSRC], 0, source);
+ RTP_SESSION_LOCK (sess);
}
static void
on_ssrc_collision (RTPSession * sess, RTPSource * source)
{
+ RTP_SESSION_UNLOCK (sess);
g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SSRC_COLLISION], 0,
source);
+ RTP_SESSION_LOCK (sess);
}
static void
on_ssrc_validated (RTPSession * sess, RTPSource * source)
{
+ RTP_SESSION_UNLOCK (sess);
g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SSRC_VALIDATED], 0,
source);
+ RTP_SESSION_LOCK (sess);
}
static void
on_bye_ssrc (RTPSession * sess, RTPSource * source)
{
+ RTP_SESSION_UNLOCK (sess);
g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_BYE_SSRC], 0, source);
+ RTP_SESSION_LOCK (sess);
}
static void
on_bye_timeout (RTPSession * sess, RTPSource * source)
{
+ RTP_SESSION_UNLOCK (sess);
g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_BYE_TIMEOUT], 0, source);
+ RTP_SESSION_LOCK (sess);
}
static void
on_timeout (RTPSession * sess, RTPSource * source)
{
+ RTP_SESSION_UNLOCK (sess);
g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_TIMEOUT], 0, source);
+ RTP_SESSION_LOCK (sess);
}
/**
generated by cgit v1.2.1 (git 2.18.0) at 2024-12-27 12:23:30 +0000