diff options
| -rw-r--r-- | ChangeLog | 17 | ||||
| -rw-r--r-- | docs/plugins/gst-plugins-good-plugins-sections.txt | 20 | ||||
| -rw-r--r-- | gst/rtsp/URLS | 5 | ||||
| -rw-r--r-- | gst/rtsp/gstrtpdec.c | 2 | ||||
| -rw-r--r-- | gst/rtsp/gstrtspsrc.c | 22 |
5 files changed, 35 insertions, 31 deletions
@@ -1,3 +1,20 @@ +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), diff --git a/docs/plugins/gst-plugins-good-plugins-sections.txt b/docs/plugins/gst-plugins-good-plugins-sections.txt index a092e485..7a694bbb 100644 --- a/docs/plugins/gst-plugins-good-plugins-sections.txt +++ b/docs/plugins/gst-plugins-good-plugins-sections.txt @@ -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> diff --git a/gst/rtsp/URLS b/gst/rtsp/URLS index 7ac187a5..653fe1dc 100644 --- a/gst/rtsp/URLS +++ b/gst/rtsp/URLS @@ -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 diff --git a/gst/rtsp/gstrtpdec.c b/gst/rtsp/gstrtpdec.c index 8ee03502..a1339994 100644 --- a/gst/rtsp/gstrtpdec.c +++ b/gst/rtsp/gstrtpdec.c @@ -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); diff --git a/gst/rtsp/gstrtspsrc.c b/gst/rtsp/gstrtspsrc.c index 0685adb0..db0c36ae 100644 --- a/gst/rtsp/gstrtspsrc.c +++ b/gst/rtsp/gstrtspsrc.c @@ -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); } } } |