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rtpbin: whitespace fixes

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This commit is contained in:
Wim Taymans 2009-08-31 16:33:26 +02:00
parent 4cf513da9b
commit a522a2d4d2
1 changed files with 19 additions and 19 deletions
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38
gst/rtpmanager/gstrtpbin.c
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@ -24,12 +24,12 @@
* RTP bin combines the functions of #GstRtpSession, #GstRtpsSrcDemux,
* #GstRtpJitterBuffer and #GstRtpPtDemux in one element. It allows for multiple
* RTP sessions that will be synchronized together using RTCP SR packets.
*
*
* #GstRtpBin is configured with a number of request pads that define the
* functionality that is activated, similar to the #GstRtpSession element.
*
*
* To use #GstRtpBin as an RTP receiver, request a recv_rtp_sink_%%d pad. The session
* number must be specified in the pad name.
* number must be specified in the pad name.
* Data received on the recv_rtp_sink_%%d pad will be processed in the gstrtpsession
* manager and after being validated forwarded on #GstRtpsSrcDemux element. Each
* RTP stream is demuxed based on the SSRC and send to a #GstRtpJitterBuffer. After
@ -38,26 +38,26 @@
* on the payload type and will create a unique pad recv_rtp_src_%%d_%%d_%%d on
* gstrtpbin with the session number, SSRC and payload type respectively as the pad
* name.
*
*
* To also use #GstRtpBin as an RTCP receiver, request a recv_rtcp_sink_%%d pad. The
* session number must be specified in the pad name.
*
*
* If you want the session manager to generate and send RTCP packets, request
* the send_rtcp_src_%%d pad with the session number in the pad name. Packet pushed
* on this pad contain SR/RR RTCP reports that should be sent to all participants
* in the session.
*
*
* To use #GstRtpBin as a sender, request a send_rtp_sink_%%d pad, which will
* automatically create a send_rtp_src_%%d pad. If the session number is not provided,
* the pad from the lowest available session will be returned. The session manager will modify the
* SSRC in the RTP packets to its own SSRC and wil forward the packets on the
* send_rtp_src_%%d pad after updating its internal state.
*
*
* The session manager needs the clock-rate of the payload types it is handling
* and will signal the #GstRtpSession::request-pt-map signal when it needs such a
* mapping. One can clear the cached values with the #GstRtpSession::clear-pt-map
* signal.
*
*
* <refsect2>
* <title>Example pipelines</title>
* |[
@ -81,7 +81,7 @@
* 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 and do not participate in preroll, sync=false and
* as soon as possible and do not participate in preroll, sync=false and
* async=false is configured on udpsink
* |[
* gst-launch -v gstrtpbin name=rtpbin \
@ -906,7 +906,7 @@ gst_rtp_bin_associate (GstRtpBin * bin, GstRtpBinStream * stream, guint8 len,
/* calculate the min of all deltas, ignoring streams that did not yet have a
* valid unix_delta because we did not yet receive an SR packet for those
* streams.
* streams.
* We calculate the mininum because we would like to only apply positive
* offsets to streams, delaying their playback instead of trying to speed up
* other streams (which might be imposible when we have to create negative
@ -1294,7 +1294,7 @@ gst_rtp_bin_class_init (GstRtpBinClass * klass)
* GstRtpBin::on-new-ssrc:
* @rtpbin: the object which received the signal
* @session: the session
* @ssrc: the SSRC
* @ssrc: the SSRC
*
* Notify of a new SSRC that entered @session.
*/
@ -1307,7 +1307,7 @@ gst_rtp_bin_class_init (GstRtpBinClass * klass)
* GstRtpBin::on-ssrc-collision:
* @rtpbin: the object which received the signal
* @session: the session
* @ssrc: the SSRC
* @ssrc: the SSRC
*
* Notify when we have an SSRC collision
*/
@ -1320,7 +1320,7 @@ gst_rtp_bin_class_init (GstRtpBinClass * klass)
* GstRtpBin::on-ssrc-validated:
* @rtpbin: the object which received the signal
* @session: the session
* @ssrc: the SSRC
* @ssrc: the SSRC
*
* Notify of a new SSRC that became validated.
*/
@ -1360,7 +1360,7 @@ gst_rtp_bin_class_init (GstRtpBinClass * klass)
* GstRtpBin::on-bye-ssrc:
* @rtpbin: the object which received the signal
* @session: the session
* @ssrc: the SSRC
* @ssrc: the SSRC
*
* Notify of an SSRC that became inactive because of a BYE packet.
*/
@ -1373,7 +1373,7 @@ gst_rtp_bin_class_init (GstRtpBinClass * klass)
* GstRtpBin::on-bye-timeout:
* @rtpbin: the object which received the signal
* @session: the session
* @ssrc: the SSRC
* @ssrc: the SSRC
*
* Notify of an SSRC that has timed out because of BYE
*/
@ -1386,7 +1386,7 @@ gst_rtp_bin_class_init (GstRtpBinClass * klass)
* GstRtpBin::on-timeout:
* @rtpbin: the object which received the signal
* @session: the session
* @ssrc: the SSRC
* @ssrc: the SSRC
*
* Notify of an SSRC that has timed out
*/
@ -1399,7 +1399,7 @@ gst_rtp_bin_class_init (GstRtpBinClass * klass)
* GstRtpBin::on-sender-timeout:
* @rtpbin: the object which received the signal
* @session: the session
* @ssrc: the SSRC
* @ssrc: the SSRC
*
* Notify of a sender SSRC that has timed out and became a receiver
*/
@ -1413,7 +1413,7 @@ gst_rtp_bin_class_init (GstRtpBinClass * klass)
* GstRtpBin::on-npt-stop:
* @rtpbin: the object which received the signal
* @session: the session
* @ssrc: the SSRC
* @ssrc: the SSRC
*
* Notify that SSRC sender has sent data up to the configured NPT stop time.
*/
@ -2348,7 +2348,7 @@ gst_rtp_bin_get_free_pad_name (GstElement * element, GstPadTemplate * templ)
return pad_name;
}
/*
/*
*/
static GstPad *
gst_rtp_bin_request_new_pad (GstElement * element,