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gst/rtsp/gstrtspsrc.*: Parse bandwidth modifiers, they are not yet configured in the session manager because we don't...
Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_get_bandwidth), (gst_rtspsrc_collect_bandwidth), (gst_rtspsrc_create_stream), (gst_rtspsrc_media_to_caps), (gst_rtspsrc_loop_interleaved): * gst/rtsp/gstrtspsrc.h: Parse bandwidth modifiers, they are not yet configured in the session manager because we don't have an API for that yet.
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3 changed files with 65 additions and 0 deletions
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@ -1,3 +1,12 @@
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2007-10-01 Wim Taymans <wim.taymans@gmail.com>
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* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_get_bandwidth),
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(gst_rtspsrc_collect_bandwidth), (gst_rtspsrc_create_stream),
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(gst_rtspsrc_media_to_caps), (gst_rtspsrc_loop_interleaved):
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* gst/rtsp/gstrtspsrc.h:
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Parse bandwidth modifiers, they are not yet configured in the session
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manager because we don't have an API for that yet.
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2007-10-01 Wim Taymans <wim.taymans@gmail.com>
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* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_parse_rtpmap),
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@ -537,6 +537,53 @@ find_stream (GstRTSPSrc * src, gconstpointer data, gconstpointer func)
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return NULL;
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}
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static const GstSDPBandwidth *
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gst_rtspsrc_get_bandwidth (GstRTSPSrc * src, const GstSDPMessage * sdp,
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const GstSDPMedia * media, const gchar * type)
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{
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guint i, len;
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/* first look in the media specific section */
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len = gst_sdp_media_bandwidths_len (media);
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for (i = 0; i < len; i++) {
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const GstSDPBandwidth *bw = gst_sdp_media_get_bandwidth (media, i);
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if (strcmp (bw->bwtype, type) == 0)
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return bw;
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}
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/* then look in the message specific section */
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len = gst_sdp_message_bandwidths_len (sdp);
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for (i = 0; i < len; i++) {
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const GstSDPBandwidth *bw = gst_sdp_message_get_bandwidth (sdp, i);
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if (strcmp (bw->bwtype, type) == 0)
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return bw;
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}
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return NULL;
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}
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static void
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gst_rtspsrc_collect_bandwidth (GstRTSPSrc * src, const GstSDPMessage * sdp,
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const GstSDPMedia * media, GstRTSPStream * stream)
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{
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const GstSDPBandwidth *bw;
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if ((bw = gst_rtspsrc_get_bandwidth (src, sdp, media, GST_SDP_BWTYPE_AS)))
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stream->as_bandwidth = bw->bandwidth;
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else
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stream->as_bandwidth = -1;
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if ((bw = gst_rtspsrc_get_bandwidth (src, sdp, media, GST_SDP_BWTYPE_RR)))
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stream->rr_bandwidth = bw->bandwidth;
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else
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stream->rr_bandwidth = -1;
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if ((bw = gst_rtspsrc_get_bandwidth (src, sdp, media, GST_SDP_BWTYPE_RS)))
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stream->rs_bandwidth = bw->bandwidth;
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else
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stream->rs_bandwidth = -1;
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}
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static GstRTSPStream *
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gst_rtspsrc_create_stream (GstRTSPSrc * src, GstSDPMessage * sdp, gint idx)
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{
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@ -561,6 +608,9 @@ gst_rtspsrc_create_stream (GstRTSPSrc * src, GstSDPMessage * sdp, gint idx)
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stream->eos = FALSE;
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stream->discont = TRUE;
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/* collect bandwidth information for this steam */
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gst_rtspsrc_collect_bandwidth (src, sdp, media, stream);
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/* we must have a payload. No payload means we cannot create caps */
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/* FIXME, handle multiple formats. */
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if ((payload = gst_sdp_media_get_format (media, 0))) {
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@ -584,6 +634,7 @@ gst_rtspsrc_create_stream (GstRTSPSrc * src, GstSDPMessage * sdp, gint idx)
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* the RTP-Info header field returned from PLAY. */
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control_url = gst_sdp_media_get_attribute_val (media, "control");
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GST_DEBUG_OBJECT (src, "stream %d, (%p)", stream->id, stream);
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GST_DEBUG_OBJECT (src, " pt: %d", stream->pt);
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GST_DEBUG_OBJECT (src, " container: %d", stream->container);
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@ -120,6 +120,11 @@ struct _GstRTSPStream {
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guint32 ssrc;
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guint32 seqbase;
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guint64 timebase;
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/* bandwidth */
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guint as_bandwidth;
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guint rs_bandwidth;
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guint rr_bandwidth;
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};
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struct _GstRTSPSrc {
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