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* Free leftover udp ports (if any) when a setup request fails.Wim Taymans2009-01-221-0/+2
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* gst/rtsp/gstrtspsrc.c: Prevent further read/write actions taken to the ↵이문형2008-11-271-2/+14
| | | | | | | | | | connect-failed socket by erroring out quickly.... Original commit message from CVS: Patch by: 이문형 <iwings at gmail dot com> * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_loop_udp): Prevent further read/write actions taken to the connect-failed socket by erroring out quickly. See #562258.
* gst/rtsp/gstrtspsrc.c: Add some more debugging.Wim Taymans2008-11-241-1/+9
| | | | | | | | | Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (new_session_pad), (gst_rtspsrc_parse_range): Add some more debugging. Use the reanges received from the server unconditionally. Fixes #561625.
* gst/rtsp/: Remove google extension again, it's not needed anymore because we ↵Wim Taymans2008-11-132-8/+2
| | | | | | | | | | | | never send multiple transports anymore. Original commit message from CVS: * gst/rtsp/Makefile.am: * gst/rtsp/gstrtsp.c: (plugin_init): * gst/rtsp/gstrtspgoogle.c: * gst/rtsp/gstrtspgoogle.h: Remove google extension again, it's not needed anymore because we never send multiple transports anymore.
* gst/rtsp/gstrtspsrc.*: Add property to configure NAT traversal method.Eric Zhang2008-11-132-76/+213
| | | | | | | | | | | | | | | | | | | | | Original commit message from CVS: Based on patch by: Eric Zhang <chao.zhang at access-company dot com> * gst/rtsp/gstrtspsrc.c: (gst_rtsp_nat_method_get_type), (gst_rtspsrc_class_init), (gst_rtspsrc_set_property), (gst_rtspsrc_get_property), (gst_rtspsrc_create_stream), (gst_rtspsrc_stream_free), (gst_rtspsrc_stream_configure_udp_sinks), (gst_rtspsrc_stream_configure_transport), (gst_rtspsrc_send_dummy_packets), (gst_rtspsrc_create_transports_string), (gst_rtspsrc_handle_message), (gst_rtspsrc_change_state): * gst/rtsp/gstrtspsrc.h: Add property to configure NAT traversal method. Ignore EOS from the internal sinks. Implement sending dummy packets as a (simple) method to open up some firewalls. Send PLAY request to the server after we started the udp sources. Fixes #559545.
* gst/rtsp/gstrtspsrc.c: Only send one transport at a time for improved ↵Wim Taymans2008-11-111-4/+3
| | | | | | | | | | compatibility with some broken servers. See #53... Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_create_transports_string), (gst_rtspsrc_change_state): Only send one transport at a time for improved compatibility with some broken servers. See #537832.
* gst/rtsp/gstrtspsrc.c: Only pause/play in the seek handler when the source ↵Wim Taymans2008-11-111-6/+11
| | | | | | | | | | was playing. Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_do_seek), (gst_rtspsrc_perform_seek): Only pause/play in the seek handler when the source was playing. Fixes #529379.
* gst/rtsp/gstrtspsrc.c: Pause the RTSP stream before doing a new play request.Eric Zhang2008-11-101-0/+5
| | | | | | | | | | Original commit message from CVS: Based on patch by: Eric Zhang <chao.zhang at access-company dot com> * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_perform_seek), (gst_rtspsrc_stream_configure_udp_sink): Pause the RTSP stream before doing a new play request. Make sure that adding the udpsinks does not cause the rtspsrc to become a sink. Fixes #559547.
* Don't install static libs for plugins. Fixes #550851 for -good.Stefan Kost2008-11-041-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Original commit message from CVS: * ext/aalib/Makefile.am: * ext/annodex/Makefile.am: * ext/cairo/Makefile.am: * ext/dv/Makefile.am: * ext/esd/Makefile.am: * ext/flac/Makefile.am: * ext/gconf/Makefile.am: * ext/gdk_pixbuf/Makefile.am: * ext/hal/Makefile.am: * ext/jpeg/Makefile.am: * ext/ladspa/Makefile.am: * ext/libcaca/Makefile.am: * ext/libmng/Makefile.am: * ext/libpng/Makefile.am: * ext/mikmod/Makefile.am: * ext/pulse/Makefile.am: * ext/raw1394/Makefile.am: * ext/shout2/Makefile.am: * ext/soup/Makefile.am: * ext/speex/Makefile.am: * ext/taglib/Makefile.am: * ext/wavpack/Makefile.am: * gst/alpha/Makefile.am: * gst/apetag/Makefile.am: * gst/audiofx/Makefile.am: * gst/auparse/Makefile.am: * gst/autodetect/Makefile.am: * gst/avi/Makefile.am: * gst/cutter/Makefile.am: * gst/debug/Makefile.am: * gst/effectv/Makefile.am: * gst/equalizer/Makefile.am: * gst/flx/Makefile.am: * gst/goom/Makefile.am: * gst/goom2k1/Makefile.am: * gst/icydemux/Makefile.am: * gst/id3demux/Makefile.am: * gst/interleave/Makefile.am: * gst/law/Makefile.am: * gst/level/Makefile.am: * gst/matroska/Makefile.am: * gst/median/Makefile.am: * gst/monoscope/Makefile.am: * gst/multifile/Makefile.am: * gst/multipart/Makefile.am: * gst/oldcore/Makefile.am: * gst/qtdemux/Makefile.am: * gst/replaygain/Makefile.am: * gst/rtp/Makefile.am: * gst/rtsp/Makefile.am: * gst/smpte/Makefile.am: * gst/spectrum/Makefile.am: * gst/udp/Makefile.am: * gst/videobox/Makefile.am: * gst/videocrop/Makefile.am: * gst/videofilter/Makefile.am: * gst/videomixer/Makefile.am: * gst/wavenc/Makefile.am: * gst/wavparse/Makefile.am: * sys/directdraw/Makefile.am: * sys/directsound/Makefile.am: * sys/oss/Makefile.am: * sys/osxaudio/Makefile.am: * sys/osxvideo/Makefile.am: * sys/sunaudio/Makefile.am: * sys/v4l2/Makefile.am: * sys/waveform/Makefile.am: * sys/ximage/Makefile.am: Don't install static libs for plugins. Fixes #550851 for -good.
* gst/rtsp/gstrtspsrc.c: Return TRUE instead of FALSE from the event handler ↵Wim Taymans2008-10-091-1/+1
| | | | | | | | | when we swallowed the event. Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_handle_src_event): Return TRUE instead of FALSE from the event handler when we swallowed the event.
* gst/rtsp/gstrtspsrc.c: Don't assume the server supports PAUSE by default. ↵Wim Taymans2008-09-251-3/+3
| | | | | | | | Fixes #551048. Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_parse_methods): Don't assume the server supports PAUSE by default. Fixes #551048.
* gst/rtsp/gstrtspsrc.c: Handle the case where we cannot do desribe or when ↵Wim Taymans2008-09-231-0/+15
| | | | | | | | | the describe result does not contain a vali... Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_open): Handle the case where we cannot do desribe or when the describe result does not contain a valid SDP message.
* gst/rtsp/gstrtspgoogle.c: Things that can happen when your brain is in ↵Wim Taymans2008-08-201-1/+1
| | | | | | | | | | google mode trying to deal with their google r... Original commit message from CVS: * gst/rtsp/gstrtspgoogle.c: Things that can happen when your brain is in google mode trying to deal with their google rtsp server extensions and trying to type your google mail account.
* gst/rtsp/: Add google RTSP extension, it can only handle udp and responds ↵Wim Taymans2008-08-205-27/+316
| | | | | | | | | | | | | | | | | | | | | | | | | with unsupported if we do anything else. Fi... Original commit message from CVS: * gst/rtsp/Makefile.am: * gst/rtsp/gstrtsp.c: (plugin_init): * gst/rtsp/gstrtspgoogle.c: (gst_rtsp_google_before_send), (gst_rtsp_google_after_send), (gst_rtsp_google_get_transports), (_do_init), (gst_rtsp_google_base_init), (gst_rtsp_google_class_init), (gst_rtsp_google_init), (gst_rtsp_google_finalize), (gst_rtsp_google_change_state), (gst_rtsp_google_extension_init): * gst/rtsp/gstrtspgoogle.h: Add google RTSP extension, it can only handle udp and responds with unsupported if we do anything else. Fixes #546465. * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_connection_send), (gst_rtspsrc_connection_receive), (gst_rtspsrc_loop_send_cmd), (gst_rtspsrc_create_transports_string), (gst_rtspsrc_setup_streams), (gst_rtspsrc_open), (gst_rtspsrc_close), (gst_rtspsrc_pause): Make transport setup code a bit better using GString. Add some more debug. Check for closed connections before doing anything on them.
* gst/rtsp/gstrtspsrc.c: Don't try to configure RTCP back to the server when ↵Wim Taymans2008-08-201-0/+10
| | | | | | | | | the server did not give us a valid port nu... Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_stream_configure_udp_sink): Don't try to configure RTCP back to the server when the server did not give us a valid port number.
* gst/rtsp/gstrtspsrc.c: Improve udp port setup. Fixes #545710.Aurelien Grimaud2008-08-051-39/+51
| | | | | | | Original commit message from CVS: Patch by: Aurelien Grimaud <gstelzz at yahoo dot fr> * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_alloc_udp_ports): Improve udp port setup. Fixes #545710.
* gst/rtp/: Add MP1S depayloader.Wim Taymans2008-08-051-0/+3
| | | | | | | | | | | | | | | | | Original commit message from CVS: * gst/rtp/Makefile.am: * gst/rtp/gstrtp.c: (plugin_init): * gst/rtp/gstrtpmp1sdepay.c: (gst_rtp_mp1s_depay_base_init), (gst_rtp_mp1s_depay_class_init), (gst_rtp_mp1s_depay_init), (gst_rtp_mp1s_depay_setcaps), (gst_rtp_mp1s_depay_process), (gst_rtp_mp1s_depay_set_property), (gst_rtp_mp1s_depay_get_property), (gst_rtp_mp1s_depay_change_state), (gst_rtp_mp1s_depay_plugin_init): * gst/rtp/gstrtpmp1sdepay.h: Add MP1S depayloader. * gst/rtsp/URLS: Some more sample rtsp streams.
* gst/rtsp/URLS: Add another URL.Wim Taymans2008-08-051-0/+3
| | | | | | | | | Original commit message from CVS: * gst/rtsp/URLS: Add another URL. * tests/check/elements/id3v2mux.c: (test_taglib_id3mux_with_tags): * tests/check/elements/rglimiter.c: (GST_START_TEST): Add some more debug info.
* gst/rtp/gstrtpvrawdepay.c: Include stdlib.h for atoi().Stefan Kost2008-07-071-1/+1
| | | | | | | | Original commit message from CVS: * gst/rtp/gstrtpvrawdepay.c: Include stdlib.h for atoi(). * gst/rtsp/gstrtspsrc.c: Use floating point math for latencies < 0 sec in log output.
* gst/rtsp/URLS: Some more urls.Wim Taymans2008-06-171-0/+3
| | | | | | | | | | Original commit message from CVS: * gst/rtsp/URLS: Some more urls. * gst/smpte/barboxwipes.c: Add a comment * tests/examples/rtp/server-v4l2-H264-alsasrc-PCMA.sh: Fix typo, add audioresample to the pipeline.
* gst/rtsp/gstrtspsrc.c: Set udpsrc for receiving data from multicast groups ↵Wim Taymans2008-06-121-2/+2
| | | | | | | | | to PAUSED instead of leaving them in READY... Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_stream_configure_mcast): Set udpsrc for receiving data from multicast groups to PAUSED instead of leaving them in READY. Fixes #537832.
* gst/rtsp/gstrtspsrc.c: Use the new gst_rtsp_connection_get_ip() to access ↵Peter Kjellerstedt2008-06-041-2/+3
| | | | | | | | | the IP address of a GstRTSPConnection since... Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_stream_configure_udp_sink): Use the new gst_rtsp_connection_get_ip() to access the IP address of a GstRTSPConnection since it is a private member.
* Don't use gst_element_get_pad(), it's a bad method.Wim Taymans2008-05-211-4/+4
| | | | | | | | | | | | | | | | | | | | | | | | | | | | Original commit message from CVS: * ext/gconf/gstgconfaudiosrc.c: (gst_gconf_audio_src_reset), (do_toggle_element): * ext/gconf/gstgconfvideosink.c: (gst_gconf_video_sink_reset), (do_toggle_element): * ext/gconf/gstgconfvideosrc.c: (gst_gconf_video_src_reset), (do_toggle_element): * ext/gconf/gstswitchsink.c: (gst_switch_commit_new_kid): * ext/hal/gsthalaudiosink.c: (gst_hal_audio_sink_reset), (do_toggle_element): * ext/hal/gsthalaudiosrc.c: (gst_hal_audio_src_reset), (do_toggle_element): * gst/autodetect/gstautoaudiosink.c: (gst_auto_audio_sink_reset), (gst_auto_audio_sink_detect): * gst/autodetect/gstautovideosink.c: (gst_auto_video_sink_reset), (gst_auto_video_sink_detect): * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_init), (gst_rtspsrc_stream_free), (gst_rtspsrc_stream_configure_udp), (gst_rtspsrc_stream_configure_udp_sink), (gst_rtspsrc_skip_lws), (gst_rtspsrc_unskip_lws), (gst_rtspsrc_skip_commas), (gst_rtspsrc_skip_item), (gst_rtsp_decode_quoted_string), (gst_rtspsrc_parse_digest_challenge), (gst_rtspsrc_parse_auth_hdr): * tests/icles/videocrop-test.c: (test_with_caps), (video_crop_get_test_caps): Don't use gst_element_get_pad(), it's a bad method.
* gst/rtsp/gstrtspsrc.c: Support Digest authentication. Fixes #532065.Wouter Cloetens2008-05-081-6/+127
| | | | | | | | | | | | Original commit message from CVS: Based on patch by: Wouter Cloetens <wouter at mind be> * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_init), (gst_rtspsrc_skip_lws), (gst_rtspsrc_unskip_lws), (gst_rtspsrc_skip_commas), (gst_rtspsrc_skip_item), (gst_rtsp_decode_quoted_string), (gst_rtspsrc_parse_digest_challenge), (gst_rtspsrc_parse_auth_hdr), (gst_rtspsrc_setup_auth): Support Digest authentication. Fixes #532065.
* gst/rtsp/gstrtspsrc.c: Don't leak file descriptors on error. Fixes #531532.Sjoerd Simons2008-05-051-0/+4
| | | | | | | Original commit message from CVS: Patch by: Sjoerd Simons <sjoerd at luon dot net> * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_init), (gst_rtspsrc_open): Don't leak file descriptors on error. Fixes #531532.
* gst/rtsp/gstrtspsrc.c: Ref caps as the return value for the request_pt_map ↵Wim Taymans2008-04-211-4/+3
| | | | | | | | | | signal. Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_init), (request_pt_map), (gst_rtspsrc_configure_caps): Ref caps as the return value for the request_pt_map signal. Remove some caps weirdness when configuring a stream. See #528245.
* gst/rtsp/gstrtspsrc.c: Call WSAStartup() and WSACleanup before using the ↵Ole André Vadla Ravnås2008-03-171-0/+16
| | | | | | | | | | | Winsock API. Original commit message from CVS: Patch by: Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com> * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_init), (gst_rtspsrc_finalize): Call WSAStartup() and WSACleanup before using the Winsock API. See #520808.
* fix license file, remove extra line copied over by mistakeChristian Schaller2008-03-141-2/+0
| | | | | Original commit message from CVS: fix license file, remove extra line copied over by mistake
* gst/rtsp/gstrtspsrc.c: Post the server response code in an error message ↵Wim Taymans2008-02-221-2/+10
| | | | | | | | | instead of a generic 'error' message. Fixes ... Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_setup_streams): Post the server response code in an error message instead of a generic 'error' message. Fixes #517237.
* gst/rtsp/gstrtspsrc.c: Init values to -1 instead of the default 0 value.Wim Taymans2008-02-181-0/+2
| | | | | | | Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_create_stream): Init values to -1 instead of the default 0 value. Fixes #516524.
* gst/rtsp/gstrtspsrc.c: Include unistd.h only if HAVE_UNISTD_H is definedSébastien Moutte2008-02-071-0/+2
| | | | | | | | | | | | | | | | | Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: Include unistd.h only if HAVE_UNISTD_H is defined * win32/common/config.h.in: * win32/common/config.h: Define socklen_t as it seems it's not defined in default Visual Studio headers. * win32/vs6/libgstalpha.dsp: * win32/vs6/libgstapetag.dsp: * win32/vs6/libgstavi.dsp: * win32/vs6/libgstrtp.dsp: * win32/vs6/libgstrtsp.dsp: * win32/vs6/libgstvideomixer.dsp: Update project file dependencies and add new source files
* gst/rtsp/gstrtspsrc.c: Use g_ascii_strtoll() instead of atoll, which is only ↵Tim-Philipp Müller2008-01-281-3/+3
| | | | | | | | | available in C99. Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_parse_rtpinfo): Use g_ascii_strtoll() instead of atoll, which is only available in C99.
* As found by: Tommi Myöhänen <ext-tommi.myohanen nokia com>Wim Taymans2008-01-141-1/+1
| | | | | | | Original commit message from CVS: As found by: Tommi Myöhänen <ext-tommi.myohanen nokia com> * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_parse_rtpinfo): Use atoll to parse the rtptime with enough precision. Fixes #509329.
* gst/: Initialise variables to work around (false) 'foo might be used ↵Tim-Philipp Müller2008-01-141-2/+2
| | | | | | | | | | uninitialized in this function' warnings by gcc-... Original commit message from CVS: * gst/avi/gstavisubtitle.c: (gst_avi_subtitle_extract_file): * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_send): Initialise variables to work around (false) 'foo might be used uninitialized in this function' warnings by gcc-3.3.3 (#509298).
* gst/rtsp/gstrtspsrc.*: Implement redirect for the DESCRIBE reply. Fixes #506025.Wim Taymans2007-12-312-6/+57
| | | | | | | Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_send), (gst_rtspsrc_open): * gst/rtsp/gstrtspsrc.h: Implement redirect for the DESCRIBE reply. Fixes #506025.
* gst/rtsp/gstrtspsrc.c: Fix some more leaks. Fixes #497007.Tommi Myöhänen2007-11-151-0/+3
| | | | | | | Original commit message from CVS: Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com> * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_setup_streams): Fix some more leaks. Fixes #497007.
* gst/rtsp/gstrtspsrc.c: Fix 3 pad leaks. Fixes #496983.Tommi Myöhänen2007-11-151-2/+10
| | | | | | | | Original commit message from CVS: Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com> * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_stream_free), (gst_rtspsrc_stream_configure_tcp): Fix 3 pad leaks. Fixes #496983.
* gst/rtsp/gstrtspsrc.c: Don't leak sdp message contents (fixes #496773).Tim-Philipp Müller2007-11-141-0/+2
| | | | | | | | Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_open): Don't leak sdp message contents (fixes #496773). * gst/udp/gstudpsink.c: (gst_udpsink_finalize): Don't leak URI string.
* gst/rtsp/gstrtspsrc.c: Don't leak event, don't leak range (fixes #496752).Tommi Myöhänen2007-11-141-0/+3
| | | | | | | | Original commit message from CVS: Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com> * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_handle_src_event), (gst_rtspsrc_parse_range): Don't leak event, don't leak range (fixes #496752).
* gst/rtsp/gstrtspsrc.c: Fix race when pausing a RTSP stream in interleaved.Tommi Myöhänen2007-10-221-0/+17
| | | | | | | | Original commit message from CVS: Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com> * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_loop_interleaved): Fix race when pausing a RTSP stream in interleaved. Fixes #475784.
* gst/rtsp/gstrtspsrc.c: Use allowed name for the GstStructure.Wim Taymans2007-10-171-1/+1
| | | | | | Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_open): Use allowed name for the GstStructure.
* gst/rtsp/gstrtspsrc.c: Fix compiler warning by using GST_CLOCK_TIME_NONE to ↵Jan Schmidt2007-10-081-1/+1
| | | | | | | | | initialise a GstClockTime. Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_flush): Fix compiler warning by using GST_CLOCK_TIME_NONE to initialise a GstClockTime.
* gst/rtsp/gstrtspsrc.c: More seeking fixes, mostly passing around the new ↵Wim Taymans2007-10-081-38/+45
| | | | | | | | | | | | | | | playback segment in order to configure it pr... Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_flush), (gst_rtspsrc_do_seek), (gst_rtspsrc_perform_seek), (gst_rtspsrc_configure_caps), (gst_rtspsrc_loop_udp), (gst_rtspsrc_parse_range), (gst_rtspsrc_open), (gst_rtspsrc_play), (gst_rtspsrc_change_state): More seeking fixes, mostly passing around the new playback segment in order to configure it properly. Also reset base_time of udp sources when setting them back to PLAYING as a temporary hack until core supports seek in live sources properly.
* gst/rtsp/gstrtspsrc.c: Improve flushing behaviour.Wim Taymans2007-10-051-17/+93
| | | | | | | | | | | | | | Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_flush), (gst_rtspsrc_perform_seek), (gst_rtspsrc_handle_src_event), (gst_rtspsrc_handle_internal_src_query), (gst_rtspsrc_handle_src_query), (new_session_pad), (gst_rtspsrc_stream_configure_tcp), (gst_rtspsrc_stream_configure_transport), (gst_rtspsrc_loop_send_cmd): Improve flushing behaviour. Set state of the udp sources to PAUSE/PLAYING correctly. Handle events and queries for UDP and TCP transport now.
* gst/rtsp/gstrtspsrc.*: Parse bandwidth modifiers, they are not yet ↵Wim Taymans2007-10-012-0/+56
| | | | | | | | | | | | 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.
* gst/rtsp/gstrtspsrc.c: Use shiny new function in -base to get the default ↵Wim Taymans2007-10-011-50/+22
| | | | | | | | | | clock-rate. Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_parse_rtpmap), (gst_rtspsrc_media_to_caps), (gst_rtspsrc_loop_interleaved): Use shiny new function in -base to get the default clock-rate. Update some docs.
* gst/rtsp/gstrtspsrc.*: In TCP mode, only timestamp the first buffer. TCP is ↵Wim Taymans2007-09-282-9/+22
| | | | | | | | | | | | not real time and it does not make sense ... Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_loop_interleaved), (gst_rtspsrc_play): * gst/rtsp/gstrtspsrc.h: In TCP mode, only timestamp the first buffer. TCP is not real time and it does not make sense to try to skew compensate, also some servers send the first batch of data in a burst.
* gst/rtsp/gstrtspsrc.*: Set timestamps on RTP buffers in interleaved mode.Wim Taymans2007-09-262-8/+20
| | | | | | | | | | | Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_create_stream), (gst_rtspsrc_handle_src_query), (gst_rtspsrc_loop_interleaved), (gst_rtspsrc_loop_udp), (gst_rtspsrc_close): * gst/rtsp/gstrtspsrc.h: Set timestamps on RTP buffers in interleaved mode. Mark first buffers with a DISCONT. Remove flush hack now that sync for live sources has been figured out.
* gst/: Fix compiler warnings shown with Forte.Jan Schmidt2007-09-171-7/+16
| | | | | | | | | | Original commit message from CVS: * gst/audiofx/audiodynamic.c: (gst_audio_dynamic_class_init): * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_create_stream), (new_session_pad), (request_pt_map), (gst_rtspsrc_do_stream_eos), (gst_rtspsrc_loop_interleaved), (gst_rtspsrc_parse_rtpinfo), (gst_rtspsrc_handle_message): Fix compiler warnings shown with Forte.
* gst/rtsp/gstrtspsrc.c: Give meaningfull error when all streams failed to ↵Wim Taymans2007-09-171-4/+18
| | | | | | | | | | configure for some reason. Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_setup_streams), (gst_rtspsrc_dup_printf): Give meaningfull error when all streams failed to configure for some reason.
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