2006-02-09 14:20:14 +00:00
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/* GStreamer
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gst/rtsp/URLS: Added some test URLS.
Original commit message from CVS:
* gst/rtsp/URLS:
Added some test URLS.
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_create_stream),
(gst_rtspsrc_loop), (gst_rtspsrc_open):
* gst/rtsp/gstrtspsrc.h:
When creating streams, give access to the complete SDP.
Fix some leaks.
Collect and merge global stream properties in stream caps.
Preliminary support for WMServer.
* gst/rtsp/rtspconnection.c: (rtsp_connection_create),
(rtsp_connection_connect), (rtsp_connection_read), (read_body),
(rtsp_connection_receive):
* gst/rtsp/rtspconnection.h:
Make connection interruptable.
Refactor to make it reconnectable.
Don't fail on short reads when reading data packets.
* gst/rtsp/rtspurl.c: (rtsp_url_parse), (rtsp_url_set_port),
(rtsp_url_get_port):
* gst/rtsp/rtspurl.h:
Add methods for getting/setting the port.
* gst/rtsp/sdpmessage.c: (sdp_message_get_attribute_val_n),
(sdp_message_get_attribute_val), (sdp_media_get_attribute),
(sdp_media_get_attribute_val_n), (sdp_media_get_attribute_val),
(sdp_media_get_format), (sdp_parse_line),
(sdp_message_parse_buffer):
Fix headers.
Add methods for getting multiple attributes with the same name.
Increase buffer size when parsing.
Fix parsing of a=foo fields.
* gst/rtsp/test.c: (main):
Update to new connection API.
* gst/rtsp/rtspmessage.c: (rtsp_message_new_response),
(rtsp_message_init_response), (rtsp_message_init_data),
(rtsp_message_unset), (rtsp_message_free), (rtsp_message_dump):
* gst/rtsp/rtspmessage.h:
* gst/rtsp/rtsptransport.c: (rtsp_transport_free):
* gst/rtsp/rtsptransport.h:
* gst/rtsp/sdp.h:
* gst/rtsp/sdpmessage.h:
* gst/rtsp/gstrtsp.c:
* gst/rtsp/gstrtsp.h:
* gst/rtsp/gstrtpdec.c:
* gst/rtsp/gstrtpdec.h:
* gst/rtsp/rtsp.h:
* gst/rtsp/rtspdefs.c:
* gst/rtsp/rtspdefs.h:
Dual licensed under MIT and LGPL now.
2006-09-20 16:06:27 +00:00
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* Copyright (C) <2005,2006> Wim Taymans <wim@fluendo.com>
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2006-02-09 14:20:14 +00:00
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
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* Boston, MA 02111-1307, USA.
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*/
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gst/rtsp/URLS: Added some test URLS.
Original commit message from CVS:
* gst/rtsp/URLS:
Added some test URLS.
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_create_stream),
(gst_rtspsrc_loop), (gst_rtspsrc_open):
* gst/rtsp/gstrtspsrc.h:
When creating streams, give access to the complete SDP.
Fix some leaks.
Collect and merge global stream properties in stream caps.
Preliminary support for WMServer.
* gst/rtsp/rtspconnection.c: (rtsp_connection_create),
(rtsp_connection_connect), (rtsp_connection_read), (read_body),
(rtsp_connection_receive):
* gst/rtsp/rtspconnection.h:
Make connection interruptable.
Refactor to make it reconnectable.
Don't fail on short reads when reading data packets.
* gst/rtsp/rtspurl.c: (rtsp_url_parse), (rtsp_url_set_port),
(rtsp_url_get_port):
* gst/rtsp/rtspurl.h:
Add methods for getting/setting the port.
* gst/rtsp/sdpmessage.c: (sdp_message_get_attribute_val_n),
(sdp_message_get_attribute_val), (sdp_media_get_attribute),
(sdp_media_get_attribute_val_n), (sdp_media_get_attribute_val),
(sdp_media_get_format), (sdp_parse_line),
(sdp_message_parse_buffer):
Fix headers.
Add methods for getting multiple attributes with the same name.
Increase buffer size when parsing.
Fix parsing of a=foo fields.
* gst/rtsp/test.c: (main):
Update to new connection API.
* gst/rtsp/rtspmessage.c: (rtsp_message_new_response),
(rtsp_message_init_response), (rtsp_message_init_data),
(rtsp_message_unset), (rtsp_message_free), (rtsp_message_dump):
* gst/rtsp/rtspmessage.h:
* gst/rtsp/rtsptransport.c: (rtsp_transport_free):
* gst/rtsp/rtsptransport.h:
* gst/rtsp/sdp.h:
* gst/rtsp/sdpmessage.h:
* gst/rtsp/gstrtsp.c:
* gst/rtsp/gstrtsp.h:
* gst/rtsp/gstrtpdec.c:
* gst/rtsp/gstrtpdec.h:
* gst/rtsp/rtsp.h:
* gst/rtsp/rtspdefs.c:
* gst/rtsp/rtspdefs.h:
Dual licensed under MIT and LGPL now.
2006-09-20 16:06:27 +00:00
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/*
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* Unless otherwise indicated, Source Code is licensed under MIT license.
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* See further explanation attached in License Statement (distributed in the file
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* LICENSE).
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*
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* Permission is hereby granted, free of charge, to any person obtaining a copy of
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* this software and associated documentation files (the "Software"), to deal in
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* the Software without restriction, including without limitation the rights to
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* use, copy, modify, merge, publish, distribute, sublicense, and/or sell copies
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* of the Software, and to permit persons to whom the Software is furnished to do
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* so, subject to the following conditions:
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*
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* The above copyright notice and this permission notice shall be included in all
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* copies or substantial portions of the Software.
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*
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* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
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* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
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* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
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* AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
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* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
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* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE
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* SOFTWARE.
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*/
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2006-02-09 14:20:14 +00:00
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/* Element-Checklist-Version: 5 */
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2006-06-20 14:57:09 +00:00
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/**
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* SECTION:element-rtpdec
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*
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* <refsect2>
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* <para>
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* A simple RTP session manager used internally by rtspsrc.
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* </para>
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* </refsect2>
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*
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* Last reviewed on 2006-06-20 (0.10.4)
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*/
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2006-02-09 14:20:14 +00:00
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#include "gstrtpdec.h"
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2006-06-22 19:31:04 +00:00
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GST_DEBUG_CATEGORY_STATIC (rtpdec_debug);
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2006-02-09 14:20:14 +00:00
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#define GST_CAT_DEFAULT (rtpdec_debug)
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/* elementfactory information */
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Define GstElementDetails as const and also static (when defined as global)
Original commit message from CVS:
* ext/aalib/gstaasink.c:
* ext/annodex/gstcmmldec.c:
* ext/annodex/gstcmmlenc.c:
* ext/cairo/gsttextoverlay.c:
* ext/cairo/gsttimeoverlay.c:
* ext/cdio/gstcdiocddasrc.c:
* ext/dv/gstdvdec.c:
* ext/dv/gstdvdemux.c:
* ext/esd/esdmon.c:
* ext/esd/esdsink.c:
* ext/flac/gstflacenc.c:
* ext/flac/gstflactag.c:
* ext/gconf/gstgconfaudiosink.c: (gst_gconf_audio_sink_base_init):
* ext/gconf/gstgconfaudiosrc.c: (gst_gconf_audio_src_base_init):
* ext/gconf/gstgconfvideosink.c: (gst_gconf_video_sink_base_init):
* ext/gconf/gstgconfvideosrc.c: (gst_gconf_video_src_base_init):
* ext/gdk_pixbuf/pixbufscale.c:
* ext/hal/gsthalaudiosink.c: (gst_hal_audio_sink_base_init):
* ext/hal/gsthalaudiosrc.c: (gst_hal_audio_src_base_init):
* ext/jpeg/gstjpegdec.c:
* ext/jpeg/gstjpegenc.c:
* ext/jpeg/gstsmokedec.c:
* ext/jpeg/gstsmokeenc.c:
* ext/libcaca/gstcacasink.c:
* ext/libmng/gstmngdec.c:
* ext/libmng/gstmngenc.c:
* ext/libpng/gstpngdec.c:
* ext/libpng/gstpngenc.c:
* ext/mikmod/gstmikmod.c:
* ext/raw1394/gstdv1394src.c:
* ext/shout2/gstshout2.c: (gst_shout2send_init):
* ext/shout2/gstshout2.h:
* ext/speex/gstspeexdec.c:
* ext/speex/gstspeexenc.c:
* gst/alpha/gstalpha.c:
* gst/alpha/gstalphacolor.c:
* gst/apetag/gstapedemux.c:
* gst/auparse/gstauparse.c:
* gst/autodetect/gstautoaudiosink.c:
(gst_auto_audio_sink_base_init):
* gst/autodetect/gstautovideosink.c:
(gst_auto_video_sink_base_init):
* gst/avi/gstavidemux.c: (gst_avi_demux_base_init):
* gst/avi/gstavimux.c: (gst_avimux_base_init):
* gst/cutter/gstcutter.c:
* gst/debug/breakmydata.c:
* gst/debug/efence.c:
* gst/debug/gstnavigationtest.c:
* gst/debug/gstnavseek.c:
* gst/debug/negotiation.c:
* gst/debug/progressreport.c:
* gst/debug/testplugin.c:
* gst/effectv/gstaging.c:
* gst/effectv/gstdice.c:
* gst/effectv/gstedge.c:
* gst/effectv/gstquark.c:
* gst/effectv/gstrev.c:
* gst/effectv/gstshagadelic.c:
* gst/effectv/gstvertigo.c:
* gst/effectv/gstwarp.c:
* gst/flx/gstflxdec.c:
* gst/goom/gstgoom.c:
* gst/icydemux/gsticydemux.c:
* gst/id3demux/gstid3demux.c:
* gst/interleave/deinterleave.c:
* gst/interleave/interleave.c:
* gst/law/alaw-decode.c: (gst_alawdec_base_init):
* gst/law/alaw-encode.c: (gst_alawenc_base_init):
* gst/law/mulaw-decode.c: (gst_mulawdec_base_init):
* gst/law/mulaw-encode.c: (gst_mulawenc_base_init):
* gst/level/gstlevel.c:
* gst/matroska/matroska-demux.c: (gst_matroska_demux_base_init):
* gst/matroska/matroska-mux.c: (gst_matroska_mux_base_init):
* gst/median/gstmedian.c:
* gst/monoscope/gstmonoscope.c:
* gst/multipart/multipartdemux.c:
* gst/multipart/multipartmux.c:
* gst/oldcore/gstaggregator.c:
* gst/oldcore/gstfdsink.c:
* gst/oldcore/gstmd5sink.c:
* gst/oldcore/gstmultifilesrc.c:
* gst/oldcore/gstpipefilter.c:
* gst/oldcore/gstshaper.c:
* gst/oldcore/gststatistics.c:
* gst/rtp/gstasteriskh263.c:
* gst/rtp/gstrtpL16depay.c:
* gst/rtp/gstrtpL16pay.c:
* gst/rtp/gstrtpamrdepay.c:
* gst/rtp/gstrtpamrpay.c:
* gst/rtp/gstrtpdepay.c:
* gst/rtp/gstrtpgsmpay.c:
* gst/rtp/gstrtph263pay.c:
* gst/rtp/gstrtph263pdepay.c:
* gst/rtp/gstrtph263ppay.c:
* gst/rtp/gstrtpilbcdepay.c:
* gst/rtp/gstrtpmp4gpay.c:
* gst/rtp/gstrtpmp4vdepay.c:
* gst/rtp/gstrtpmp4vpay.c:
* gst/rtp/gstrtpmpadepay.c:
* gst/rtp/gstrtpmpapay.c:
* gst/rtp/gstrtppcmadepay.c:
* gst/rtp/gstrtppcmapay.c:
* gst/rtp/gstrtppcmudepay.c:
* gst/rtp/gstrtppcmupay.c:
* gst/rtp/gstrtpspeexdepay.c:
* gst/rtp/gstrtpspeexpay.c:
* gst/rtsp/gstrtpdec.c:
* gst/rtsp/gstrtspsrc.c:
* gst/smpte/gstsmpte.c:
* gst/udp/gstdynudpsink.c:
* gst/udp/gstmultiudpsink.c:
* gst/udp/gstudpsink.c:
* gst/udp/gstudpsrc.c:
* gst/videobox/gstvideobox.c:
* gst/videofilter/gstgamma.c: (gst_gamma_base_init):
* gst/videofilter/gstvideobalance.c:
* gst/videofilter/gstvideoflip.c:
* gst/videofilter/gstvideotemplate.c:
(gst_videotemplate_base_init):
* gst/videomixer/videomixer.c:
* gst/wavparse/gstwavparse.c: (gst_wavparse_base_init),
(gst_wavparse_class_init), (gst_wavparse_dispose),
(gst_wavparse_reset), (gst_wavparse_init),
(gst_wavparse_perform_seek), (gst_wavparse_peek_chunk_info),
(gst_wavparse_peek_chunk), (gst_wavparse_stream_headers),
(gst_wavparse_parse_stream_init), (gst_wavparse_send_event),
(gst_wavparse_add_src_pad), (gst_wavparse_stream_data),
(gst_wavparse_chain), (gst_wavparse_srcpad_event),
(gst_wavparse_sink_activate), (gst_wavparse_sink_activate_pull),
(gst_wavparse_change_state):
* gst/wavparse/gstwavparse.h:
* sys/oss/gstossmixerelement.c:
* sys/oss/gstosssink.c:
* sys/oss/gstosssrc.c:
* sys/osxaudio/gstosxaudioelement.c:
* sys/osxaudio/gstosxaudiosink.c:
* sys/osxaudio/gstosxaudiosrc.c:
* sys/sunaudio/gstsunaudiomixer.c:
* sys/sunaudio/gstsunaudiosink.c:
Define GstElementDetails as const and also static (when defined as
global)
2006-04-25 21:39:46 +00:00
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static const GstElementDetails rtpdec_details =
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GST_ELEMENT_DETAILS ("RTP Decoder",
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2006-02-09 14:20:14 +00:00
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"Codec/Parser/Network",
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"Accepts raw RTP and RTCP packets and sends them forward",
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"Wim Taymans <wim@fluendo.com>");
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/* GstRTPDec signals and args */
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enum
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{
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/* FILL ME */
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LAST_SIGNAL
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};
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enum
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{
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ARG_0,
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ARG_SKIP
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/* FILL ME */
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};
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static GstStaticPadTemplate gst_rtpdec_src_rtp_template =
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GST_STATIC_PAD_TEMPLATE ("srcrtp",
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GST_PAD_SRC,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("application/x-rtp")
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);
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static GstStaticPadTemplate gst_rtpdec_src_rtcp_template =
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GST_STATIC_PAD_TEMPLATE ("srcrtcp",
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GST_PAD_SRC,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("application/x-rtcp")
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);
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static GstStaticPadTemplate gst_rtpdec_sink_rtp_template =
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GST_STATIC_PAD_TEMPLATE ("sinkrtp",
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GST_PAD_SINK,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("application/x-rtp")
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);
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static GstStaticPadTemplate gst_rtpdec_sink_rtcp_template =
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GST_STATIC_PAD_TEMPLATE ("sinkrtcp",
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GST_PAD_SINK,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("application/x-rtcp")
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);
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static void gst_rtpdec_class_init (gpointer g_class);
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static void gst_rtpdec_init (GstRTPDec * rtpdec);
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static GstCaps *gst_rtpdec_getcaps (GstPad * pad);
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static GstFlowReturn gst_rtpdec_chain_rtp (GstPad * pad, GstBuffer * buffer);
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static GstFlowReturn gst_rtpdec_chain_rtcp (GstPad * pad, GstBuffer * buffer);
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static void gst_rtpdec_set_property (GObject * object,
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guint prop_id, const GValue * value, GParamSpec * pspec);
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static void gst_rtpdec_get_property (GObject * object,
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guint prop_id, GValue * value, GParamSpec * pspec);
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static GstStateChangeReturn gst_rtpdec_change_state (GstElement * element,
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GstStateChange transition);
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static GstElementClass *parent_class = NULL;
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/*static guint gst_rtpdec_signals[LAST_SIGNAL] = { 0 };*/
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GType
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gst_rtpdec_get_type (void)
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{
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static GType rtpdec_type = 0;
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if (!rtpdec_type) {
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static const GTypeInfo rtpdec_info = {
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sizeof (GstRTPDecClass), NULL,
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NULL,
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(GClassInitFunc) gst_rtpdec_class_init,
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NULL,
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NULL,
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sizeof (GstRTPDec),
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0,
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(GInstanceInitFunc) gst_rtpdec_init,
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};
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rtpdec_type =
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g_type_register_static (GST_TYPE_ELEMENT, "GstRTPDec", &rtpdec_info, 0);
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}
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return rtpdec_type;
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}
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static void
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gst_rtpdec_class_init (gpointer g_class)
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{
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GObjectClass *gobject_class;
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GstElementClass *gstelement_class;
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GstRTPDecClass *klass;
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klass = (GstRTPDecClass *) g_class;
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gobject_class = (GObjectClass *) klass;
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gstelement_class = (GstElementClass *) klass;
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gst_element_class_add_pad_template (gstelement_class,
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gst_static_pad_template_get (&gst_rtpdec_src_rtp_template));
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gst_element_class_add_pad_template (gstelement_class,
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gst_static_pad_template_get (&gst_rtpdec_src_rtcp_template));
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gst_element_class_add_pad_template (gstelement_class,
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gst_static_pad_template_get (&gst_rtpdec_sink_rtp_template));
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gst_element_class_add_pad_template (gstelement_class,
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gst_static_pad_template_get (&gst_rtpdec_sink_rtcp_template));
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gst_element_class_set_details (gstelement_class, &rtpdec_details);
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gobject_class->set_property = gst_rtpdec_set_property;
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gobject_class->get_property = gst_rtpdec_get_property;
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2006-06-20 14:57:09 +00:00
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/* FIXME, this is unused and probably copied from somewhere */
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g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_SKIP,
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g_param_spec_int ("skip", "Skip", "skip (unused)", G_MININT, G_MAXINT, 0,
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G_PARAM_READWRITE));
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2006-02-09 14:20:14 +00:00
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2006-04-08 21:21:45 +00:00
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parent_class = g_type_class_peek_parent (klass);
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2006-02-09 14:20:14 +00:00
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gstelement_class->change_state = gst_rtpdec_change_state;
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GST_DEBUG_CATEGORY_INIT (rtpdec_debug, "rtpdec", 0, "RTP decoder");
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}
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static void
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gst_rtpdec_init (GstRTPDec * rtpdec)
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{
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/* the input rtp pad */
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rtpdec->sink_rtp =
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2006-03-15 16:17:12 +00:00
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gst_pad_new_from_static_template (&gst_rtpdec_sink_rtp_template,
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"sinkrtp");
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2006-02-09 14:20:14 +00:00
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gst_pad_set_getcaps_function (rtpdec->sink_rtp, gst_rtpdec_getcaps);
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gst_pad_set_chain_function (rtpdec->sink_rtp, gst_rtpdec_chain_rtp);
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gst/rtsp/gstrtpdec.c: Add pads after setting them up.
Original commit message from CVS:
* gst/rtsp/gstrtpdec.c: (gst_rtpdec_init), (gst_rtpdec_getcaps):
Add pads after setting them up.
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_class_init),
(gst_rtspsrc_init), (gst_rtspsrc_finalize),
(gst_rtspsrc_free_stream), (gst_rtspsrc_media_to_caps),
(gst_rtspsrc_stream_setup_rtp),
(gst_rtspsrc_stream_configure_transport),
(gst_rtspsrc_combine_flows), (gst_rtspsrc_loop),
(gst_rtspsrc_open), (gst_rtspsrc_close), (gst_rtspsrc_play),
(gst_rtspsrc_pause):
* gst/rtsp/gstrtspsrc.h:
Fix interleaved mode.
- Protect streaming with lock.
- Combine flows
- set caps on outgoing buffers.
- strip trailing \0 from data packets.
- Configure RTP/RTCP in stream.
Use DEBUG_OBJECT more.
2006-08-16 09:48:26 +00:00
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gst_element_add_pad (GST_ELEMENT (rtpdec), rtpdec->sink_rtp);
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2006-02-09 14:20:14 +00:00
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/* the input rtcp pad */
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rtpdec->sink_rtcp =
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2006-03-15 16:17:12 +00:00
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gst_pad_new_from_static_template (&gst_rtpdec_sink_rtcp_template,
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"sinkrtcp");
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2006-02-09 14:20:14 +00:00
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gst_pad_set_chain_function (rtpdec->sink_rtcp, gst_rtpdec_chain_rtcp);
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gst/rtsp/gstrtpdec.c: Add pads after setting them up.
Original commit message from CVS:
* gst/rtsp/gstrtpdec.c: (gst_rtpdec_init), (gst_rtpdec_getcaps):
Add pads after setting them up.
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_class_init),
(gst_rtspsrc_init), (gst_rtspsrc_finalize),
(gst_rtspsrc_free_stream), (gst_rtspsrc_media_to_caps),
(gst_rtspsrc_stream_setup_rtp),
(gst_rtspsrc_stream_configure_transport),
(gst_rtspsrc_combine_flows), (gst_rtspsrc_loop),
(gst_rtspsrc_open), (gst_rtspsrc_close), (gst_rtspsrc_play),
(gst_rtspsrc_pause):
* gst/rtsp/gstrtspsrc.h:
Fix interleaved mode.
- Protect streaming with lock.
- Combine flows
- set caps on outgoing buffers.
- strip trailing \0 from data packets.
- Configure RTP/RTCP in stream.
Use DEBUG_OBJECT more.
2006-08-16 09:48:26 +00:00
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gst_element_add_pad (GST_ELEMENT (rtpdec), rtpdec->sink_rtcp);
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2006-02-09 14:20:14 +00:00
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/* the output rtp pad */
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rtpdec->src_rtp =
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2006-03-15 16:17:12 +00:00
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gst_pad_new_from_static_template (&gst_rtpdec_src_rtp_template, "srcrtp");
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2006-02-09 14:20:14 +00:00
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gst_pad_set_getcaps_function (rtpdec->src_rtp, gst_rtpdec_getcaps);
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gst_element_add_pad (GST_ELEMENT (rtpdec), rtpdec->src_rtp);
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/* the output rtcp pad */
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rtpdec->src_rtcp =
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2006-03-15 16:17:12 +00:00
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gst_pad_new_from_static_template (&gst_rtpdec_src_rtcp_template,
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"srcrtcp");
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2006-02-09 14:20:14 +00:00
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gst_element_add_pad (GST_ELEMENT (rtpdec), rtpdec->src_rtcp);
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}
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static GstCaps *
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gst_rtpdec_getcaps (GstPad * pad)
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{
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GstRTPDec *src;
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GstPad *other;
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GstCaps *caps;
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gst/rtsp/: Factor out extension in separate module.
Original commit message from CVS:
* gst/rtsp/Makefile.am:
* gst/rtsp/gstrtpdec.c: (gst_rtpdec_getcaps),
(gst_rtpdec_chain_rtp), (gst_rtpdec_chain_rtcp):
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_init),
(gst_rtspsrc_finalize), (gst_rtspsrc_create_stream),
(gst_rtspsrc_parse_rtpmap),
(gst_rtspsrc_stream_configure_transport), (gst_rtspsrc_send),
(gst_rtspsrc_parse_methods), (gst_rtspsrc_open),
(gst_rtspsrc_play), (gst_rtspsrc_handle_message):
* gst/rtsp/gstrtspsrc.h:
* gst/rtsp/rtspdefs.c: (rtsp_strresult):
* gst/rtsp/rtspdefs.h:
* gst/rtsp/rtspext.h:
* gst/rtsp/rtspextwms.c: (rtsp_ext_wms_parse_sdp),
(rtsp_ext_wms_get_context):
* gst/rtsp/rtspextwms.h:
* gst/rtsp/rtsptransport.c: (rtsp_transport_init), (parse_mode),
(rtsp_transport_parse):
* gst/rtsp/rtsptransport.h:
Factor out extension in separate module.
Fix getcaps to filter against the padtemplate.
Use Content-Base if the server gives one.
Rework the transport parsing a bit for future extensions.
Added some Real Header field definitions.
2006-10-04 17:24:40 +00:00
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const GstCaps *templ;
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2006-02-09 14:20:14 +00:00
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src = GST_RTPDEC (GST_PAD_PARENT (pad));
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gst/rtsp/gstrtpdec.c: Add pads after setting them up.
Original commit message from CVS:
* gst/rtsp/gstrtpdec.c: (gst_rtpdec_init), (gst_rtpdec_getcaps):
Add pads after setting them up.
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_class_init),
(gst_rtspsrc_init), (gst_rtspsrc_finalize),
(gst_rtspsrc_free_stream), (gst_rtspsrc_media_to_caps),
(gst_rtspsrc_stream_setup_rtp),
(gst_rtspsrc_stream_configure_transport),
(gst_rtspsrc_combine_flows), (gst_rtspsrc_loop),
(gst_rtspsrc_open), (gst_rtspsrc_close), (gst_rtspsrc_play),
(gst_rtspsrc_pause):
* gst/rtsp/gstrtspsrc.h:
Fix interleaved mode.
- Protect streaming with lock.
- Combine flows
- set caps on outgoing buffers.
- strip trailing \0 from data packets.
- Configure RTP/RTCP in stream.
Use DEBUG_OBJECT more.
2006-08-16 09:48:26 +00:00
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other = (pad == src->src_rtp ? src->sink_rtp : src->src_rtp);
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2006-02-09 14:20:14 +00:00
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caps = gst_pad_peer_get_caps (other);
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gst/rtsp/: Factor out extension in separate module.
Original commit message from CVS:
* gst/rtsp/Makefile.am:
* gst/rtsp/gstrtpdec.c: (gst_rtpdec_getcaps),
(gst_rtpdec_chain_rtp), (gst_rtpdec_chain_rtcp):
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_init),
(gst_rtspsrc_finalize), (gst_rtspsrc_create_stream),
(gst_rtspsrc_parse_rtpmap),
(gst_rtspsrc_stream_configure_transport), (gst_rtspsrc_send),
(gst_rtspsrc_parse_methods), (gst_rtspsrc_open),
(gst_rtspsrc_play), (gst_rtspsrc_handle_message):
* gst/rtsp/gstrtspsrc.h:
* gst/rtsp/rtspdefs.c: (rtsp_strresult):
* gst/rtsp/rtspdefs.h:
* gst/rtsp/rtspext.h:
* gst/rtsp/rtspextwms.c: (rtsp_ext_wms_parse_sdp),
(rtsp_ext_wms_get_context):
* gst/rtsp/rtspextwms.h:
* gst/rtsp/rtsptransport.c: (rtsp_transport_init), (parse_mode),
(rtsp_transport_parse):
* gst/rtsp/rtsptransport.h:
Factor out extension in separate module.
Fix getcaps to filter against the padtemplate.
Use Content-Base if the server gives one.
Rework the transport parsing a bit for future extensions.
Added some Real Header field definitions.
2006-10-04 17:24:40 +00:00
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templ = gst_pad_get_pad_template_caps (pad);
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if (caps == NULL) {
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GST_DEBUG_OBJECT (src, "copy template");
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caps = gst_caps_copy (templ);
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} else {
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GstCaps *intersect;
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GST_DEBUG_OBJECT (src, "intersect with template");
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intersect = gst_caps_intersect (caps, templ);
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gst_caps_unref (caps);
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caps = intersect;
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}
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2006-02-09 14:20:14 +00:00
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return caps;
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}
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static GstFlowReturn
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gst_rtpdec_chain_rtp (GstPad * pad, GstBuffer * buffer)
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{
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GstRTPDec *src;
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|
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src = GST_RTPDEC (GST_PAD_PARENT (pad));
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gst/rtsp/: Factor out extension in separate module.
Original commit message from CVS:
* gst/rtsp/Makefile.am:
* gst/rtsp/gstrtpdec.c: (gst_rtpdec_getcaps),
(gst_rtpdec_chain_rtp), (gst_rtpdec_chain_rtcp):
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_init),
(gst_rtspsrc_finalize), (gst_rtspsrc_create_stream),
(gst_rtspsrc_parse_rtpmap),
(gst_rtspsrc_stream_configure_transport), (gst_rtspsrc_send),
(gst_rtspsrc_parse_methods), (gst_rtspsrc_open),
(gst_rtspsrc_play), (gst_rtspsrc_handle_message):
* gst/rtsp/gstrtspsrc.h:
* gst/rtsp/rtspdefs.c: (rtsp_strresult):
* gst/rtsp/rtspdefs.h:
* gst/rtsp/rtspext.h:
* gst/rtsp/rtspextwms.c: (rtsp_ext_wms_parse_sdp),
(rtsp_ext_wms_get_context):
* gst/rtsp/rtspextwms.h:
* gst/rtsp/rtsptransport.c: (rtsp_transport_init), (parse_mode),
(rtsp_transport_parse):
* gst/rtsp/rtsptransport.h:
Factor out extension in separate module.
Fix getcaps to filter against the padtemplate.
Use Content-Base if the server gives one.
Rework the transport parsing a bit for future extensions.
Added some Real Header field definitions.
2006-10-04 17:24:40 +00:00
|
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GST_DEBUG_OBJECT (src, "got rtp packet");
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2006-02-09 14:20:14 +00:00
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return gst_pad_push (src->src_rtp, buffer);
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}
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|
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static GstFlowReturn
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|
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gst_rtpdec_chain_rtcp (GstPad * pad, GstBuffer * buffer)
|
|
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{
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gst/rtsp/: Factor out extension in separate module.
Original commit message from CVS:
* gst/rtsp/Makefile.am:
* gst/rtsp/gstrtpdec.c: (gst_rtpdec_getcaps),
(gst_rtpdec_chain_rtp), (gst_rtpdec_chain_rtcp):
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_init),
(gst_rtspsrc_finalize), (gst_rtspsrc_create_stream),
(gst_rtspsrc_parse_rtpmap),
(gst_rtspsrc_stream_configure_transport), (gst_rtspsrc_send),
(gst_rtspsrc_parse_methods), (gst_rtspsrc_open),
(gst_rtspsrc_play), (gst_rtspsrc_handle_message):
* gst/rtsp/gstrtspsrc.h:
* gst/rtsp/rtspdefs.c: (rtsp_strresult):
* gst/rtsp/rtspdefs.h:
* gst/rtsp/rtspext.h:
* gst/rtsp/rtspextwms.c: (rtsp_ext_wms_parse_sdp),
(rtsp_ext_wms_get_context):
* gst/rtsp/rtspextwms.h:
* gst/rtsp/rtsptransport.c: (rtsp_transport_init), (parse_mode),
(rtsp_transport_parse):
* gst/rtsp/rtsptransport.h:
Factor out extension in separate module.
Fix getcaps to filter against the padtemplate.
Use Content-Base if the server gives one.
Rework the transport parsing a bit for future extensions.
Added some Real Header field definitions.
2006-10-04 17:24:40 +00:00
|
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GstRTPDec *src;
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src = GST_RTPDEC (GST_PAD_PARENT (pad));
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GST_DEBUG_OBJECT (src, "got rtcp packet");
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2006-02-09 14:20:14 +00:00
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gst_buffer_unref (buffer);
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return GST_FLOW_OK;
|
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}
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static void
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gst_rtpdec_set_property (GObject * object, guint prop_id,
|
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const GValue * value, GParamSpec * pspec)
|
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|
|
{
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GstRTPDec *src;
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src = GST_RTPDEC (object);
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switch (prop_id) {
|
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case ARG_SKIP:
|
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break;
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default:
|
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break;
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}
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}
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static void
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gst_rtpdec_get_property (GObject * object, guint prop_id, GValue * value,
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GParamSpec * pspec)
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{
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GstRTPDec *src;
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|
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src = GST_RTPDEC (object);
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switch (prop_id) {
|
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case ARG_SKIP:
|
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break;
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default:
|
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break;
|
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}
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}
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static GstStateChangeReturn
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gst_rtpdec_change_state (GstElement * element, GstStateChange transition)
|
|
|
|
{
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GstStateChangeReturn ret;
|
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GstRTPDec *rtpdec;
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|
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rtpdec = GST_RTPDEC (element);
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|
|
|
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switch (transition) {
|
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case GST_STATE_CHANGE_PAUSED_TO_READY:
|
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|
break;
|
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default:
|
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|
break;
|
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}
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ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
|
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|
|
|
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switch (transition) {
|
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case GST_STATE_CHANGE_PAUSED_TO_READY:
|
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break;
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default:
|
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break;
|
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}
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|
|
|
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return ret;
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}
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