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-rw-r--r--ChangeLog18
-rw-r--r--gst/rtpmanager/gstrtpjitterbuffer.c6
-rw-r--r--gst/rtpmanager/gstrtpsession.c70
-rw-r--r--gst/rtpmanager/rtpsource.c3
4 files changed, 87 insertions, 10 deletions
diff --git a/ChangeLog b/ChangeLog
index d8428b40..e7fe82a7 100644
--- a/ChangeLog
+++ b/ChangeLog
@@ -1,3 +1,21 @@
+2007-08-29 Wim Taymans <wim.taymans@gmail.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ (gst_rtp_jitter_buffer_loop):
+ Improve Comments.
+
+ * gst/rtpmanager/gstrtpsession.c: (stop_rtcp_thread),
+ (gst_rtp_session_change_state), (gst_rtp_session_parse_caps),
+ (gst_rtp_session_clock_rate), (gst_rtp_session_sink_setcaps),
+ (gst_rtp_session_event_send_rtp_sink), (create_recv_rtp_sink),
+ (create_send_rtp_sink):
+ Also parse the sink caps for clock-rate instead of only relying on the
+ result of the signal.
+
+ * gst/rtpmanager/rtpsource.c: (rtp_source_send_rtp):
+ Make sure we fetch the clock rate for payloads we are sending out so
+ that we can use it for SR reports.
+
2007-08-29 Zaheer Abbas Merali <zaheerabbas at merali dot org>
* gst/switch/gstswitch.c (gst_switch_chain, gst_switch_set_property):
diff --git a/gst/rtpmanager/gstrtpjitterbuffer.c b/gst/rtpmanager/gstrtpjitterbuffer.c
index 2e44425a..f4d59bb6 100644
--- a/gst/rtpmanager/gstrtpjitterbuffer.c
+++ b/gst/rtpmanager/gstrtpjitterbuffer.c
@@ -942,6 +942,7 @@ again:
if (priv->eos)
goto do_eos;
}
+ /* wait for packets or flushing now */
JBUF_WAIT_CHECK (priv, flushing);
}
@@ -1004,7 +1005,6 @@ again:
running_time = gst_segment_to_running_time (&priv->segment, GST_FORMAT_TIME,
timestamp);
- /* correct for sync against the gstreamer clock, add latency */
GST_OBJECT_LOCK (jitterbuffer);
clock = GST_ELEMENT_CLOCK (jitterbuffer);
if (!clock) {
@@ -1013,7 +1013,7 @@ again:
goto push_buffer;
}
- /* add latency */
+ /* add latency, this includes our own latency and the peer latency. */
running_time += (priv->latency_ms * GST_MSECOND);
running_time += priv->peer_latency;
@@ -1050,7 +1050,7 @@ again:
if (ret == GST_CLOCK_UNSCHEDULED) {
GST_DEBUG_OBJECT (jitterbuffer,
"Wait got unscheduled, will retry to push with new buffer");
- /* reinserting popped buffer into queue */
+ /* reinsert popped buffer into queue */
if (!rtp_jitter_buffer_insert (priv->jbuf, outbuf)) {
GST_DEBUG_OBJECT (jitterbuffer,
"Duplicate packet #%d detected, dropping", seqnum);
diff --git a/gst/rtpmanager/gstrtpsession.c b/gst/rtpmanager/gstrtpsession.c
index 554422fc..985a3713 100644
--- a/gst/rtpmanager/gstrtpsession.c
+++ b/gst/rtpmanager/gstrtpsession.c
@@ -227,10 +227,15 @@ struct _GstRtpSessionPrivate
{
GMutex *lock;
RTPSession *session;
+
/* thread for sending out RTCP */
GstClockID id;
gboolean stop_thread;
GThread *thread;
+
+ /* caps mapping */
+ guint8 pt;
+ gint clock_rate;
};
/* callbacks to handle actions from the session manager */
@@ -657,6 +662,7 @@ gst_rtp_session_change_state (GstElement * element, GstStateChange transition)
case GST_STATE_CHANGE_NULL_TO_READY:
break;
case GST_STATE_CHANGE_READY_TO_PAUSED:
+ priv->clock_rate = -1;
break;
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
break;
@@ -778,6 +784,31 @@ gst_rtp_session_send_rtcp (RTPSession * sess, RTPSource * src,
return result;
}
+static gboolean
+gst_rtp_session_parse_caps (GstRtpSession * rtpsession, GstCaps * caps)
+{
+ GstRtpSessionPrivate *priv;
+ const GstStructure *caps_struct;
+
+ priv = rtpsession->priv;
+
+ GST_DEBUG_OBJECT (rtpsession, "parsing caps");
+
+ caps_struct = gst_caps_get_structure (caps, 0);
+ if (!gst_structure_get_int (caps_struct, "clock-rate", &priv->clock_rate))
+ goto no_clock_rate;
+
+ GST_DEBUG_OBJECT (rtpsession, "parsed clock-rate %d", priv->clock_rate);
+
+ return TRUE;
+
+ /* ERRORS */
+no_clock_rate:
+ {
+ GST_DEBUG_OBJECT (rtpsession, "No clock-rate in caps!");
+ return FALSE;
+ }
+}
/* called when the session manager needs the clock rate */
static gint
@@ -786,12 +817,17 @@ gst_rtp_session_clock_rate (RTPSession * sess, guint8 payload,
{
gint result = -1;
GstRtpSession *rtpsession;
+ GstRtpSessionPrivate *priv;
GValue ret = { 0 };
GValue args[2] = { {0}, {0} };
GstCaps *caps;
- const GstStructure *caps_struct;
rtpsession = GST_RTP_SESSION_CAST (user_data);
+ priv = rtpsession->priv;
+
+ /* if we have it, return it */
+ if (priv->clock_rate != -1)
+ return priv->clock_rate;
g_value_init (&args[0], GST_TYPE_ELEMENT);
g_value_set_object (&args[0], rtpsession);
@@ -808,11 +844,10 @@ gst_rtp_session_clock_rate (RTPSession * sess, guint8 payload,
if (!caps)
goto no_caps;
- caps_struct = gst_caps_get_structure (caps, 0);
- if (!gst_structure_get_int (caps_struct, "clock-rate", &result))
- goto no_clock_rate;
+ if (!gst_rtp_session_parse_caps (rtpsession, caps))
+ goto parse_failed;
- GST_DEBUG_OBJECT (rtpsession, "parsed clock-rate %d", result);
+ result = priv->clock_rate;
return result;
@@ -822,9 +857,9 @@ no_caps:
GST_DEBUG_OBJECT (rtpsession, "could not get caps");
return -1;
}
-no_clock_rate:
+parse_failed:
{
- GST_DEBUG_OBJECT (rtpsession, "could not clock-rate from caps");
+ GST_DEBUG_OBJECT (rtpsession, "failed to parse caps");
return -1;
}
}
@@ -887,6 +922,23 @@ gst_rtp_session_event_recv_rtp_sink (GstPad * pad, GstEvent * event)
return ret;
}
+static gboolean
+gst_rtp_session_sink_setcaps (GstPad * pad, GstCaps * caps)
+{
+ gboolean res;
+ GstRtpSession *rtpsession;
+ GstRtpSessionPrivate *priv;
+
+ rtpsession = GST_RTP_SESSION (gst_pad_get_parent (pad));
+ priv = rtpsession->priv;
+
+ res = gst_rtp_session_parse_caps (rtpsession, caps);
+
+ gst_object_unref (rtpsession);
+
+ return res;
+}
+
/* receive a packet from a sender, send it to the RTP session manager and
* forward the packet on the rtp_src pad
*/
@@ -1051,6 +1103,8 @@ create_recv_rtp_sink (GstRtpSession * rtpsession)
gst_rtp_session_chain_recv_rtp);
gst_pad_set_event_function (rtpsession->recv_rtp_sink,
gst_rtp_session_event_recv_rtp_sink);
+ gst_pad_set_setcaps_function (rtpsession->recv_rtp_sink,
+ gst_rtp_session_sink_setcaps);
gst_pad_set_active (rtpsession->recv_rtp_sink, TRUE);
gst_element_add_pad (GST_ELEMENT_CAST (rtpsession),
rtpsession->recv_rtp_sink);
@@ -1109,6 +1163,8 @@ create_send_rtp_sink (GstRtpSession * rtpsession)
gst_rtp_session_chain_send_rtp);
gst_pad_set_event_function (rtpsession->send_rtp_sink,
gst_rtp_session_event_send_rtp_sink);
+ gst_pad_set_setcaps_function (rtpsession->send_rtp_sink,
+ gst_rtp_session_sink_setcaps);
gst_pad_set_active (rtpsession->send_rtp_sink, TRUE);
gst_element_add_pad (GST_ELEMENT_CAST (rtpsession),
rtpsession->send_rtp_sink);
diff --git a/gst/rtpmanager/rtpsource.c b/gst/rtpmanager/rtpsource.c
index ad491bd0..24bb8466 100644
--- a/gst/rtpmanager/rtpsource.c
+++ b/gst/rtpmanager/rtpsource.c
@@ -481,6 +481,9 @@ rtp_source_send_rtp (RTPSource * src, GstBuffer * buffer)
if (timestamp != -1)
src->last_timestamp = timestamp;
+ if (src->clock_rate == -1)
+ get_clock_rate (src, gst_rtp_buffer_get_payload_type (buffer));
+
/* push packet */
if (src->callbacks.push_rtp) {
guint32 ssrc;
generated by cgit v1.2.1 (git 2.18.0) at 2024-12-27 13:09:37 +0000