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author | Wim Taymans <wim.taymans@gmail.com> | 2007-09-17 02:01:41 +0000 |
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committer | Wim Taymans <wim.taymans@gmail.com> | 2007-09-17 02:01:41 +0000 |
commit | c0aa28ca5bfc2eaf6b9cbba1159e9490c7c118c4 (patch) | |
tree | 6cbc87e37fd54f7e593a59656ea09358473c391f /gst/rtpmanager/gstrtpsession.c | |
parent | 04d3b8290698e41034809e8baec11622ca128243 (diff) | |
download | gst-plugins-bad-c0aa28ca5bfc2eaf6b9cbba1159e9490c7c118c4.tar.gz gst-plugins-bad-c0aa28ca5bfc2eaf6b9cbba1159e9490c7c118c4.tar.bz2 gst-plugins-bad-c0aa28ca5bfc2eaf6b9cbba1159e9490c7c118c4.zip |
gst/rtpmanager/gstrtpbin.c: Link to the right pads regardless of which one was created first in the ssrc demuxer.
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (new_ssrc_pad_found):
Link to the right pads regardless of which one was created first in the
ssrc demuxer.
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_chain), (gst_rtp_jitter_buffer_loop):
* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_process_rtp),
(gst_rtp_session_chain_recv_rtp), (gst_rtp_session_chain_send_rtp):
* gst/rtpmanager/rtpsource.c: (calculate_jitter):
Improve debugging.
* gst/rtpmanager/gstrtpssrcdemux.c: (create_demux_pad_for_ssrc),
(gst_rtp_ssrc_demux_init), (gst_rtp_ssrc_demux_finalize),
(gst_rtp_ssrc_demux_sink_event),
(gst_rtp_ssrc_demux_rtcp_sink_event), (gst_rtp_ssrc_demux_chain),
(gst_rtp_ssrc_demux_rtcp_chain),
(gst_rtp_ssrc_demux_internal_links):
* gst/rtpmanager/gstrtpssrcdemux.h:
Fix race in creating the RTP and RTCP pads when a new SSRC is detected.
Diffstat (limited to 'gst/rtpmanager/gstrtpsession.c')
-rw-r--r-- | gst/rtpmanager/gstrtpsession.c | 27 |
1 files changed, 25 insertions, 2 deletions
diff --git a/gst/rtpmanager/gstrtpsession.c b/gst/rtpmanager/gstrtpsession.c index 6d4cf8b0..83210ff8 100644 --- a/gst/rtpmanager/gstrtpsession.c +++ b/gst/rtpmanager/gstrtpsession.c @@ -781,11 +781,11 @@ gst_rtp_session_process_rtp (RTPSession * sess, RTPSource * src, rtpsession = GST_RTP_SESSION (user_data); priv = rtpsession->priv; - GST_DEBUG_OBJECT (rtpsession, "reading receiving RTP packet"); - if (rtpsession->recv_rtp_src) { + GST_DEBUG_OBJECT (rtpsession, "pushing received RTP packet"); result = gst_pad_push (rtpsession->recv_rtp_src, buffer); } else { + GST_DEBUG_OBJECT (rtpsession, "dropping received RTP packet"); gst_buffer_unref (buffer); result = GST_FLOW_OK; } @@ -1114,10 +1114,22 @@ gst_rtp_session_chain_recv_rtp (GstPad * pad, GstBuffer * buffer) } ret = rtp_session_process_rtp (priv->session, buffer, ntpnstime); + if (ret != GST_FLOW_OK) + goto push_error; + +done: gst_object_unref (rtpsession); return ret; + + /* ERRORS */ +push_error: + { + GST_DEBUG_OBJECT (rtpsession, "process returned %s", + gst_flow_get_name (ret)); + goto done; + } } static GstFlowReturn @@ -1286,10 +1298,21 @@ gst_rtp_session_chain_send_rtp (GstPad * pad, GstBuffer * buffer) } ret = rtp_session_send_rtp (priv->session, buffer, ntpnstime); + if (ret != GST_FLOW_OK) + goto push_error; +done: gst_object_unref (rtpsession); return ret; + + /* ERRORS */ +push_error: + { + GST_DEBUG_OBJECT (rtpsession, "process returned %s", + gst_flow_get_name (ret)); + goto done; + } } /* Create sinkpad to receive RTP packets from senders. This will also create a |