From a522a2d4d2c9f3ef516a82d59c6db292147cc8e3 Mon Sep 17 00:00:00 2001 From: Wim Taymans Date: Mon, 31 Aug 2009 16:33:26 +0200 Subject: rtpbin: whitespace fixes --- gst/rtpmanager/gstrtpbin.c | 38 +++++++++++++++++++------------------- 1 file changed, 19 insertions(+), 19 deletions(-) (limited to 'gst/rtpmanager') diff --git a/gst/rtpmanager/gstrtpbin.c b/gst/rtpmanager/gstrtpbin.c index ed9fa3ff..0c00e5c8 100644 --- a/gst/rtpmanager/gstrtpbin.c +++ b/gst/rtpmanager/gstrtpbin.c @@ -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. - * + * * * Example pipelines * |[ @@ -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, -- cgit