Commit b7524708 authored by Wim Taymans's avatar Wim Taymans

docs/plugins/gst-plugins-good-plugins-sections.txt: Fix docs.

Original commit message from CVS:
* docs/plugins/gst-plugins-good-plugins-sections.txt:
Fix docs.
* gst/rtsp/URLS:
Add some more example urls.
* gst/rtsp/gstrtpdec.c: (gst_rtp_dec_marshal_BOXED__UINT_UINT),
(gst_rtp_dec_chain_rtp):
Better debugging.
* gst/rtsp/gstrtspsrc.c: (request_pt_map),
(gst_rtspsrc_activate_streams), (gst_rtspsrc_loop_interleaved),
(gst_rtspsrc_parse_rtpinfo):
Remove unused code.
parent 3bf1b5ec
2007-04-13 Wim Taymans <wim@fluendo.com>
* docs/plugins/gst-plugins-good-plugins-sections.txt:
Fix docs.
* gst/rtsp/URLS:
Add some more example urls.
* gst/rtsp/gstrtpdec.c: (gst_rtp_dec_marshal_BOXED__UINT_UINT),
(gst_rtp_dec_chain_rtp):
Better debugging.
* gst/rtsp/gstrtspsrc.c: (request_pt_map),
(gst_rtspsrc_activate_streams), (gst_rtspsrc_loop_interleaved),
(gst_rtspsrc_parse_rtpinfo):
Remove unused code.
2007-04-13 Stefan Kost <ensonic@users.sf.net>
* gst/wavparse/gstwavparse.c: (uint64_ceiling_scale_int),
......
......@@ -475,10 +475,10 @@ gst_progress_report_get_type
<SECTION>
<FILE>element-rtspsrc</FILE>
GstRTSPProto
GstRTSPSrc
<TITLE>rtspsrc</TITLE>
<SUBSECTION Standard>
gst_rtspsrc_send
GstRTSPStream
GstRTSPSrcClass
GST_RTSPSRC
......@@ -487,6 +487,10 @@ GST_TYPE_RTSPSRC
gst_rtspsrc_get_type
GST_RTSPSRC_CLASS
GST_IS_RTSPSRC_CLASS
GST_RTSPSRC_CAST
GST_RTSP_LOOP_GET_COND
GST_RTSP_LOOP_SIGNAL
GST_RTSP_LOOP_WAIT
</SECTION>
<SECTION>
......@@ -494,14 +498,14 @@ GST_IS_RTSPSRC_CLASS
GstRTPDec
<TITLE>rtpdec</TITLE>
<SUBSECTION Standard>
gst_rtpdec_plugin_init
GstRTPDecClass
GST_RTPDEC
GST_IS_RTPDEC
GST_TYPE_RTPDEC
gst_rtpdec_get_type
GST_RTPDEC_CLASS
GST_IS_RTPDEC_CLASS
GstRTPDecSession
GST_RTP_DEC
GST_IS_RTP_DEC
GST_TYPE_RTP_DEC
gst_rtp_dec_get_type
GST_RTP_DEC_CLASS
GST_IS_RTP_DEC_CLASS
</SECTION>
<SECTION>
......
......@@ -14,11 +14,16 @@ MP4V-ES/mpeg4-generic(ACC):
rtsp://vod.nwec.jp/quicktime/505.mov
rtsp://203.140.68.241:554/hirakataeizou9.mp4
rtsp://kmdi.utoronto.ca:555/osconf/2004_may9.1.mp4
X-QT(h264)/mpeg4-generic(ACC):
rtsp://a2047.v1413b.c1413.g.vq.akamaistream.net/5/2047/1413/1_h264_110/1a1a1ae656c632970267e04ebd3196c428970e7ce857b81c4aab1677e445aedc3fae1b4a7bafe013/8848125_1_110.mov
MP4V-ES/MP4A-LATM
rtsp://68.251.168.13/thisislove.3gp
H264/MPA
rtsp://130.192.86.166/ed.mov
REAL:
rtsp://213.254.239.61/farm/*/encoder/tagesschau/live1high.rm
rtsp://64.192.137.105:554/real.amazon-de.eu2/phononet/B/0/0/0/H/W/Y/4/K/S/01.01.rm?cloakport=80,554,7070
......@@ -351,7 +351,7 @@ gst_rtp_dec_chain_rtp (GstPad * pad, GstBuffer * buffer)
ssrc = gst_rtp_buffer_get_ssrc (buffer);
pt = gst_rtp_buffer_get_payload_type (buffer);
GST_DEBUG_OBJECT (rtpdec, "SSRC %d, PT %d", ssrc, pt);
GST_DEBUG_OBJECT (rtpdec, "SSRC %08x, PT %d", ssrc, pt);
/* find session */
session = gst_pad_get_element_private (pad);
......
......@@ -1372,19 +1372,6 @@ start_session_failure:
}
}
static gboolean
gst_rtspsrc_stream_configure_caps (GstRTSPStream * stream)
{
/* configure the caps on the UDP source and the channelpad */
if (stream->udpsrc[0]) {
//g_object_set (G_OBJECT (stream->udpsrc[0]), "caps", stream->caps, NULL);
}
if (stream->channelpad[0]) {
//gst_pad_set_caps (stream->channelpad[0], stream->caps);
}
return TRUE;
}
/* Adds the source pads of all configured streams to the element.
* This code is performed when we detected dataflow.
*
......@@ -1410,7 +1397,6 @@ gst_rtspsrc_activate_streams (GstRTSPSrc * src)
gst_pad_set_active (stream->srcpad, TRUE);
/* add the pad */
if (!stream->added) {
//gst_pad_set_caps (stream->srcpad, stream->caps);
gst_element_add_pad (GST_ELEMENT_CAST (src), stream->srcpad);
stream->added = TRUE;
}
......@@ -1508,7 +1494,6 @@ gst_rtspsrc_loop_interleaved (GstRTSPSrc * src)
guint8 *data;
guint size;
GstFlowReturn ret = GST_FLOW_OK;
GstCaps *caps = NULL;
GstBuffer *buf;
gboolean is_rtcp = FALSE;
......@@ -1531,7 +1516,6 @@ gst_rtspsrc_loop_interleaved (GstRTSPSrc * src)
stream = (GstRTSPStream *) lstream->data;
if (channel == stream->channel[0]) {
outpad = stream->channelpad[0];
caps = stream->caps;
} else if (channel == stream->channel[1]) {
outpad = stream->channelpad[1];
is_rtcp = TRUE;
......@@ -1567,9 +1551,6 @@ gst_rtspsrc_loop_interleaved (GstRTSPSrc * src)
/* don't need message anymore */
rtsp_message_unset (&response);
if (caps)
gst_buffer_set_caps (buf, caps);
GST_DEBUG_OBJECT (src, "pushing data of size %d on channel %d", size,
channel);
......@@ -2806,9 +2787,6 @@ gst_rtspsrc_parse_rtpinfo (GstRTSPSrc * src, gchar * rtpinfo)
/* update caps */
gst_caps_set_simple (caps, "clock-base", G_TYPE_UINT, timebase,
"seqnum-base", G_TYPE_UINT, seqbase, NULL);
/* and configure the stream caps */
gst_rtspsrc_stream_configure_caps (stream);
}
}
}
......
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