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gst/rtpmanager/gstrtpbin.c: Add some more advanced example pipelines.

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Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_clear_pt_map):
Add some more advanced example pipelines.
* gst/rtpmanager/gstrtpsession.c: (rtcp_thread),
(stop_rtcp_thread), (gst_rtp_session_send_rtcp):
Add some debug and FIXME.
Release LOCK when performing session cleanup.
* gst/rtpmanager/rtpsession.c: (session_report_blocks):
Add some debug.
* gst/rtpmanager/rtpsource.c: (calculate_jitter),
(rtp_source_send_rtp):
Make sure we always send RTP packets with the session SSRC.
This commit is contained in:
Wim Taymans 2007-08-28 20:30:16 +00:00
parent 3db14a2b60
commit c0a64d008a
5 changed files with 85 additions and 7 deletions
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17
ChangeLog
View file

@ -1,3 +1,20 @@
2007-08-28 Wim Taymans <wim.taymans@gmail.com>
* gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_clear_pt_map):
Add some more advanced example pipelines.
* gst/rtpmanager/gstrtpsession.c: (rtcp_thread),
(stop_rtcp_thread), (gst_rtp_session_send_rtcp):
Add some debug and FIXME.
Release LOCK when performing session cleanup.
* gst/rtpmanager/rtpsession.c: (session_report_blocks):
Add some debug.
* gst/rtpmanager/rtpsource.c: (calculate_jitter),
(rtp_source_send_rtp):
Make sure we always send RTP packets with the session SSRC.
2007-08-28 Tim-Philipp Müller <tim at centricular dot net>
* gst/dvdspu/gstdvdspu.c:

48
gst/rtpmanager/gstrtpbin.c
View file

@ -75,9 +75,54 @@
* </programlisting>
* Receive RTP data from port 5000 and send to the session 0 in gstrtpbin.
* </para>
* <para>
* <programlisting>
* gst-launch gstrtpbin name=rtpbin \
* v4l2src ! ffmpegcolorspace ! ffenc_h263 ! rtph263ppay ! rtpbin.send_rtp_sink_0 \
* rtpbin.send_rtp_src_0 ! udpsink port=5000 \
* rtpbin.send_rtcp_src_0 ! udpsink port=5001 sync=false \
* udpsrc port=5005 ! rtpbin.recv_rtcp_sink_0 \
* audiotestsrc ! amrnbenc ! rtpamrpay ! rtpbin.send_rtp_sink_1 \
* rtpbin.send_rtp_src_1 ! udpsink port=5002 \
* rtpbin.send_rtcp_src_1 ! udpsink port=5003 sync=false \
* udpsrc port=5007 ! rtpbin.recv_rtcp_sink_1
* </programlisting>
* Encode and payload H263 video captured from a v4l2src. Encode and payload AMR
* audio generated from audiotestsrc. The video is sent to session 0 in rtpbin
* and the audio is sent to session 1. Video packets are sent on UDP port 5000
* and audio packets on port 5002. The video RTCP packets for session 0 are sent
* on port 5001 and the audio RTCP packets for session 0 are sent on port 5003.
* RTCP packets for session 0 are received on port 5005 and RTCP for session 1
* is received on port 5007. Since RTCP packets from the sender should be sent
* as soon as possible, sync=false is configured on udpsink.
* </para>
* <para>
* <programlisting>
* gst-launch -v gstrtpbin name=rtpbin \
* udpsrc caps="application/x-rtp,media=(string)video,clock-rate=(int)90000,encoding-name=(string)H263-1998" \
* port=5000 ! rtpbin.recv_rtp_sink_0 \
* rtpbin. ! rtph263pdepay ! ffdec_h263 ! xvimagesink \
* udpsrc port=5001 ! rtpbin.recv_rtcp_sink_0 \
* rtpbin.send_rtcp_src_0 ! udpsink port=5005 sync=false \
* udpsrc caps="application/x-rtp,media=(string)audio,clock-rate=(int)8000,encoding-name=(string)AMR,encoding-params=(string)1,octet-align=(string)1" \
* port=5002 ! rtpbin.recv_rtp_sink_1 \
* rtpbin. ! rtpamrdepay ! amrnbdec ! alsasink \
* udpsrc port=5003 ! rtpbin.recv_rtcp_sink_1 \
* rtpbin.send_rtcp_src_1 ! udpsink port=5007 sync=false
* </programlisting>
* Receive H263 on port 5000, send it through rtpbin in session 0, depayload,
* decode and display the video.
* Receive AMR on port 5002, send it through rtpbin in session 1, depayload,
* decode and play the audio.
* Receive server RTCP packets for session 0 on port 5001 and RTCP packets for
* session 1 on port 5003. These packets will be used for session management and
* synchronisation.
* Send RTCP reports for session 0 on port 5005 and RTCP reports for session 1
* on port 5007.
* </para>
* </refsect2>
*
* Last reviewed on 2007-05-28 (0.10.5)
* Last reviewed on 2007-08-28 (0.10.6)
*/
#ifdef HAVE_CONFIG_H
@ -452,6 +497,7 @@ gst_rtp_bin_clear_pt_map (GstRtpBin * bin)
GSList *walk;
GST_RTP_BIN_LOCK (bin);
GST_DEBUG_OBJECT (bin, "clearing pt map");
for (walk = bin->sessions; walk; walk = g_slist_next (walk)) {
GstRtpBinSession *session = (GstRtpBinSession *) walk->data;

9
gst/rtpmanager/gstrtpsession.c
View file

@ -583,8 +583,10 @@ rtcp_thread (GstRtpSession * rtpsession)
GST_DEBUG_OBJECT (rtpsession, "unlocked %d, current %" GST_TIME_FORMAT,
res, GST_TIME_ARGS (current_time));
/* perform actions, we ignore result. */
/* perform actions, we ignore result. Release lock because it might push. */
GST_RTP_SESSION_UNLOCK (rtpsession);
rtp_session_on_timeout (rtpsession->priv->session, current_time);
GST_RTP_SESSION_LOCK (rtpsession);
}
GST_RTP_SESSION_UNLOCK (rtpsession);
@ -637,6 +639,7 @@ stop_rtcp_thread (GstRtpSession * rtpsession)
gst_clock_id_unschedule (rtpsession->priv->id);
GST_RTP_SESSION_UNLOCK (rtpsession);
/* FIXME, can deadlock because the thread might be blocked in a push */
g_thread_join (rtpsession->priv->thread);
}
@ -753,11 +756,11 @@ gst_rtp_session_send_rtcp (RTPSession * sess, RTPSource * src,
rtpsession = GST_RTP_SESSION (user_data);
priv = rtpsession->priv;
GST_DEBUG_OBJECT (rtpsession, "sending RTCP");
if (rtpsession->send_rtcp_src) {
GST_DEBUG_OBJECT (rtpsession, "sending RTCP");
result = gst_pad_push (rtpsession->send_rtcp_src, buffer);
} else {
GST_DEBUG_OBJECT (rtpsession, "not sending RTCP, no output pad");
gst_buffer_unref (buffer);
result = GST_FLOW_OK;
}

1
gst/rtpmanager/rtpsession.c
View file

@ -1659,6 +1659,7 @@ session_report_blocks (const gchar * key, RTPSource * source, ReportData * data)
DLSR = gst_util_uint64_scale_int (diff, 65536, GST_SECOND);
} else {
/* No valid SR received, LSR/DLSR are set to 0 then */
GST_DEBUG ("no valid SR received");
LSR = 0;
DLSR = 0;
}

17
gst/rtpmanager/rtpsource.c
View file

@ -266,8 +266,8 @@ calculate_jitter (RTPSource * src, GstBuffer * buffer,
src->stats.prev_rtptime = src->stats.last_rtptime;
src->stats.last_rtptime = rtparrival;
GST_DEBUG ("rtparrival %u, rtptime %u, clock-rate %d, diff %d, jitter: %u",
rtparrival, rtptime, clock_rate, diff, src->stats.jitter);
GST_DEBUG ("rtparrival %u, rtptime %u, clock-rate %d, diff %d, jitter: %f",
rtparrival, rtptime, clock_rate, diff, (src->stats.jitter) / 16.0);
return;
@ -446,6 +446,8 @@ rtp_source_process_bye (RTPSource * src, const gchar * reason)
* @buffer: an RTP buffer
*
* Send an RTP @buffer originating from @src. This will make @src a sender.
* This function takes ownership of @buffer and modifies the SSRC in the RTP
* packet to that of @src.
*
* Returns: a #GstFlowReturn.
*/
@ -467,9 +469,18 @@ rtp_source_send_rtp (RTPSource * src, GstBuffer * buffer)
src->stats.packets_sent++;
src->stats.octets_sent += len;
/* push packet */
if (src->callbacks.push_rtp) {
guint32 ssrc;
ssrc = gst_rtp_buffer_get_ssrc (buffer);
if (ssrc != src->ssrc) {
GST_DEBUG ("updating SSRC from %u to %u", ssrc, src->ssrc);
buffer = gst_buffer_make_writable (buffer);
gst_rtp_buffer_set_ssrc (buffer, src->ssrc);
}
GST_DEBUG ("pushing RTP packet %" G_GUINT64_FORMAT,
src->stats.packets_sent);
result = src->callbacks.push_rtp (src, buffer, src->user_data);