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* gst/rtp/gstrtph263ppay.*: Add new fragmentation mode base on GOB headers. ↵Antoine Tremblay2007-05-212-11/+139
| | | | | | | | | | | | | Fixes #438940. Original commit message from CVS: Patch by: Antoine Tremblay <hexa00 at gmail dot com> * gst/rtp/gstrtph263ppay.c: (gst_fragmentation_mode_get_type), (gst_rtp_h263p_pay_class_init), (gst_rtp_h263p_pay_init), (gst_rtp_h263p_pay_set_property), (gst_rtp_h263p_pay_get_property), (gst_rtp_h263p_pay_flush): * gst/rtp/gstrtph263ppay.h: Add new fragmentation mode base on GOB headers. Fixes #438940.
* gst/: Printf format fixes (#439910, #439911).Tim-Philipp Müller2007-05-201-1/+1
| | | | | | | Original commit message from CVS: * gst/qtdemux/qtdemux.c: (gst_qtdemux_prepare_current_sample): * gst/switch/gstswitch.c: (gst_switch_chain): Printf format fixes (#439910, #439911).
* gst/rtsp/gstrtspsrc.c: Printf format fix.Tim-Philipp Müller2007-05-201-1/+1
| | | | | | Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_loop_udp): Printf format fix.
* Add replaygain playback elements (#412710).René Stadler2007-05-1910-152/+1302
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Original commit message from CVS: Patch by: René Stadler <mail at renestadler de> * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-bad-plugins-docs.sgml: * docs/plugins/gst-plugins-bad-plugins-sections.txt: * docs/plugins/inspect/plugin-replaygain.xml: * gst/replaygain/Makefile.am: * gst/replaygain/gstrganalysis.c: (gst_rg_analysis_class_init), (gst_rg_analysis_start), (gst_rg_analysis_set_caps), (gst_rg_analysis_transform_ip), (gst_rg_analysis_event), (gst_rg_analysis_stop), (gst_rg_analysis_handle_tags), (gst_rg_analysis_handle_eos), (gst_rg_analysis_track_result), (gst_rg_analysis_album_result): * gst/replaygain/gstrganalysis.h: * gst/replaygain/gstrglimiter.c: (gst_rg_limiter_base_init), (gst_rg_limiter_class_init), (gst_rg_limiter_init), (gst_rg_limiter_set_property), (gst_rg_limiter_get_property), (gst_rg_limiter_transform_ip): * gst/replaygain/gstrglimiter.h: * gst/replaygain/gstrgvolume.c: (gst_rg_volume_base_init), (gst_rg_volume_class_init), (gst_rg_volume_init), (gst_rg_volume_set_property), (gst_rg_volume_get_property), (gst_rg_volume_dispose), (gst_rg_volume_change_state), (gst_rg_volume_sink_event), (gst_rg_volume_tag_event), (gst_rg_volume_reset), (gst_rg_volume_update_gain), (gst_rg_volume_determine_gain): * gst/replaygain/gstrgvolume.h: * gst/replaygain/replaygain.c: (plugin_init): * gst/replaygain/replaygain.h: * gst/replaygain/rganalysis.h: * tests/check/Makefile.am: * tests/check/elements/.cvsignore: * tests/check/elements/rganalysis.c: (send_eos_event), (GST_START_TEST): * tests/check/elements/rglimiter.c: (setup_rglimiter), (cleanup_rglimiter), (set_playing_state), (create_test_buffer), (verify_test_buffer), (GST_START_TEST), (rglimiter_suite), (main): * tests/check/elements/rgvolume.c: (event_func), (setup_rgvolume), (cleanup_rgvolume), (set_playing_state), (set_null_state), (send_eos_event), (send_tag_event), (test_buffer_new), (fail_unless_target_gain), (fail_unless_result_gain), (fail_unless_gain), (GST_START_TEST), (rgvolume_suite), (main): Add replaygain playback elements (#412710).
* gst/rtsp/gstrtspsrc.c: Don't crash when an unsupported transport error was ↵Wim Taymans2007-05-181-4/+15
| | | | | | | | | returned by the server, just try to config... Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_setup_streams): Don't crash when an unsupported transport error was returned by the server, just try to configure the next stream. Fixes #439255.
* gst/rtsp/gstrtspsrc.*: Add TCP timeout property and use it for all TCP ↵Wim Taymans2007-05-183-13/+147
| | | | | | | | | | | | | | | | | connection. Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_class_init), (gst_rtspsrc_set_property), (gst_rtspsrc_get_property), (gst_rtspsrc_stream_configure_udp), (gst_rtspsrc_loop_interleaved), (gst_rtspsrc_loop_udp), (gst_rtspsrc_try_send), (gst_rtspsrc_send), (gst_rtspsrc_setup_streams), (gst_rtspsrc_open): * gst/rtsp/gstrtspsrc.h: Add TCP timeout property and use it for all TCP connection. * gst/rtsp/rtspconnection.c: (rtsp_connection_connect), (rtsp_connection_write), (rtsp_connection_next_timeout), (rtsp_connection_reset_timeout): Make connect and writes cancelable and make them use the timeout.
* gst/rtsp/gstrtspsrc.c: Refactor timeout handling.Wim Taymans2007-05-183-63/+130
| | | | | | | | | | | | | | | | Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_send_keep_alive), (gst_rtspsrc_loop_interleaved), (gst_rtspsrc_loop_udp), (gst_rtspsrc_try_send), (gst_rtspsrc_send), (gst_rtspsrc_setup_streams): Refactor timeout handling. Also send keep-alive when dealing with TCP transport. * gst/rtsp/rtspconnection.c: (rtsp_connection_create), (rtsp_connection_free), (rtsp_connection_next_timeout), (rtsp_connection_reset_timeout): * gst/rtsp/rtspconnection.h: Use a timer to handle the session timeouts, add some methods to deal with timeouts.
* gst/rtsp/gstrtspsrc.c: Ignore streams that fail the setup command, we will ↵Wim Taymans2007-05-172-3/+17
| | | | | | | | | | | | | retry with a different transport later on. Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_send), (gst_rtspsrc_setup_streams): Ignore streams that fail the setup command, we will retry with a different transport later on. * gst/rtsp/rtspextwms.c: (rtsp_ext_wms_parse_sdp), (rtsp_ext_wms_configure_stream): Fix encoding name case.
* gst/debug/breakmydata.c (gst_break_my_data_init): One more try. This should ↵Stefan Kost2007-05-151-1/+1
| | | | | | | | be the proper fix now. Original commit message from CVS: * gst/debug/breakmydata.c (gst_break_my_data_init): One more try. This should be the proper fix now.
* gst/debug/breakmydata.c: Ooops, no // comments please.Stefan Kost2007-05-151-1/+0
| | | | | | Original commit message from CVS: * gst/debug/breakmydata.c: Ooops, no // comments please.
* gst/debug/breakmydata.c: Fix gst_buffer_is_writable() assertion.Stefan Kost2007-05-151-3/+4
| | | | | | | Original commit message from CVS: * gst/debug/breakmydata.c: (gst_break_my_data_class_init), (gst_break_my_data_init): Fix gst_buffer_is_writable() assertion.
* gst/rtp/: Update theora pay/depayloader in a similar to vorbis.Wim Taymans2007-05-143-101/+236
| | | | | | | | | | | | | Original commit message from CVS: * gst/rtp/gstrtptheoradepay.c: (decode_base64), (gst_rtp_theora_depay_parse_configuration): * gst/rtp/gstrtptheorapay.c: (encode_base64), (gst_rtp_theora_pay_finish_headers), (gst_rtp_theora_pay_handle_buffer): Update theora pay/depayloader in a similar to vorbis. * gst/rtp/gstrtpvorbisdepay.c: (gst_rtp_vorbis_depay_parse_configuration): Update docs.
* gst/rtsp/gstrtspsrc.c: When we try to execute a method that is not supported ↵Wim Taymans2007-05-141-1/+15
| | | | | | | | | | by the server, don't error out but remov... Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_send): When we try to execute a method that is not supported by the server, don't error out but remove the method from the accepted methods so that we never try to perform this method again.
* gst/rtp/gstrtpvorbisdepay.c: Remove annoying _dump_mem.Wim Taymans2007-05-141-2/+0
| | | | | | Original commit message from CVS: * gst/rtp/gstrtpvorbisdepay.c: (gst_rtp_vorbis_depay_process): Remove annoying _dump_mem.
* gst/rtsp/gstrtspsrc.c: Parse range correctly.Wim Taymans2007-05-142-6/+21
| | | | | | | | Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_parse_range): Parse range correctly. * gst/rtsp/rtspurl.c: (rtsp_url_get_request_uri): The baseurl now always has a '/' at the start.
* gst/rtsp/gstrtspsrc.c: Factor out caps configuration and configure more ↵Wim Taymans2007-05-142-40/+74
| | | | | | | | | | | | | stuff such as the time ranges and speed/scale... Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_configure_caps), (gst_rtspsrc_parse_range), (gst_rtspsrc_open), (gst_rtspsrc_parse_rtpinfo), (gst_rtspsrc_play): Factor out caps configuration and configure more stuff such as the time ranges and speed/scale values. * gst/rtsp/rtsptransport.c: Add Copyright after non-trival fixes.
* gst/replaygain/rganalysis.c: Fix wrong ifdef for visual C++. Fixes: #437403.David Schleef2007-05-131-2/+4
| | | | | | | Original commit message from CVS: * gst/replaygain/rganalysis.c: Fix wrong ifdef for visual C++. Fixes: #437403. By Ali Sabil <ali.sabil@gmail.com>.
* gst/level/gstlevel.c: Use guint8 * instead of gpointer then vs6 can build ↵Sébastien Moutte2007-05-131-1/+1
| | | | | | | | | in_data += (filter->width / 8). Original commit message from CVS: * gst/level/gstlevel.c: (gst_level_transform_ip): Use guint8 * instead of gpointer then vs6 can build in_data += (filter->width / 8).
* gst/rtsp/: Make channel guint8 where possible.Peter Kjellerstedt2007-05-127-9/+12
| | | | | | | | | | | | | | | | | | | Original commit message from CVS: Patch by: Peter Kjellerstedt <pkj at axis com> * gst/rtsp/gstrtspsrc.h: * gst/rtsp/rtspconnection.c: (rtsp_connection_receive): * gst/rtsp/rtspmessage.c: (rtsp_message_init_data), (rtsp_message_get_header): * gst/rtsp/rtspmessage.h: Make channel guint8 where possible. Make rtsp_message_init_data() take the channel as a guint8. * gst/rtsp/rtspdefs.c: Fixed a typo: Timout -> Timeout * gst/rtsp/rtspdefs.h: Make RTSP_CHECK() behave as a statement. * gst/rtsp/sdpmessage.c: Avoid a compiler warning in INIT_ARRAY(). Fixes #437692.
* gst/rtsp/rtspurl.*: Add support for query parameters to RTSP URLs.Peter Kjellerstedt2007-05-122-21/+37
| | | | | | | | | Original commit message from CVS: Patch by: Peter Kjellerstedt <pkj at axis com> * gst/rtsp/rtspurl.c: (rtsp_url_parse), (rtsp_url_free), (rtsp_url_get_request_uri): * gst/rtsp/rtspurl.h: Add support for query parameters to RTSP URLs.
* gst/rtsp/rtsptransport.*: Add validation to rtsp_transport_parse().Peter Kjellerstedt2007-05-122-38/+364
| | | | | | | | | | | | | | | | | | Original commit message from CVS: Patch by: Peter Kjellerstedt <pkj at axis com> * gst/rtsp/rtsptransport.c: (rtsp_transport_init), (parse_mode), (parse_range), (range_as_text), (rtsp_transport_mode_as_text), (rtsp_transport_profile_as_text), (rtsp_transport_ltrans_as_text), (rtsp_transport_parse), (rtsp_transport_as_text): * gst/rtsp/rtsptransport.h: Add validation to rtsp_transport_parse(). Add rtsp_transport_as_text() to generate an RTSP header from an RTSPTransport. Change ssrc to guint (was a string) since that is what it is, even though it is sent as a hex string. Correctly identify PLAY|RECORD mode parameters (the syntax in the RFC is incorrect, which can be seen when looking at the examples in the RFC). Fixes #437670.
* gst/wavparse/gstwavparse.c: Skip LIST chunks before the fmt chunk (fixes ↵Tim-Philipp Müller2007-05-111-5/+1
| | | | | | | | | | #437499). Also fix streaming mode regression... Original commit message from CVS: * gst/wavparse/gstwavparse.c: (gst_wavparse_stream_headers): Skip LIST chunks before the fmt chunk (fixes #437499). Also fix streaming mode regression for file from #343837 with 'bext' chunk before the 'fmt' chunk.
* gst/rtsp/: Preliminary seek support.Wim Taymans2007-05-113-16/+218
| | | | | | | | | | | | | | | | Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_flush), (gst_rtspsrc_do_seek), (gst_rtspsrc_perform_seek), (gst_rtspsrc_handle_src_event), (gst_rtspsrc_stream_configure_manager), (gst_rtspsrc_stream_configure_tcp), (gst_rtspsrc_loop_interleaved), (gst_rtspsrc_send_keep_alive), (gst_rtspsrc_open), (gst_rtspsrc_parse_rtpinfo), (gst_rtspsrc_play): * gst/rtsp/gstrtspsrc.h: * gst/rtsp/rtspdefs.h: Preliminary seek support. Activate internal pads so that we can receive events on them. Don't try to parse a range string when it's NULL.
* gst/rtp/README: Update README with new RTP variables that will be used for ↵Wim Taymans2007-05-113-95/+264
| | | | | | | | | | | | | | | | synchronisation. Original commit message from CVS: * gst/rtp/README: Update README with new RTP variables that will be used for synchronisation. * gst/rtp/gstrtpvorbisdepay.c: (decode_base64), (gst_rtp_vorbis_depay_parse_configuration), (gst_rtp_vorbis_depay_process): * gst/rtp/gstrtpvorbispay.c: (encode_base64), (gst_rtp_vorbis_pay_finish_headers), (gst_rtp_vorbis_pay_handle_buffer): Update vorbis pay and depayloader to draft-04.
* gst/rtsp/rtsptransport.c: UDP MCAST is actually the default for RTP/AVP.Wim Taymans2007-05-111-1/+1
| | | | | | Original commit message from CVS: * gst/rtsp/rtsptransport.c: UDP MCAST is actually the default for RTP/AVP.
* gst/rtsp/rtsptransport.c: Make UDP the default transport when not specified.Wim Taymans2007-05-111-1/+1
| | | | | | Original commit message from CVS: * gst/rtsp/rtsptransport.c: Make UDP the default transport when not specified.
* gst/qtdemux/qtdemux.c (gst_qtdemux_move_stream, ↵Stefan Kost2007-05-101-9/+15
| | | | | | | | | | | | | | | | | | | | | gst_qtdemux_loop_state_header, gst_qtdemux_activate_segment, gst_qtde... Original commit message from CVS: * gst/qtdemux/qtdemux.c (gst_qtdemux_move_stream, gst_qtdemux_loop_state_header, gst_qtdemux_activate_segment, gst_qtdemux_prepare_current_sample, gst_qtdemux_combine_flows, gst_qtdemux_loop_state_movie, gst_qtdemux_loop, qtdemux_parse_segments, qtdemux_parse_trak): * gst/rtpmanager/rtpsession.c (rtp_session_get_bandwidth, rtp_session_get_rtcp_bandwidth, rtp_session_get_cname, rtp_session_get_name, rtp_session_get_email, rtp_session_get_phone, rtp_session_get_location, rtp_session_get_tool, rtp_session_process_bye, session_report_blocks): * gst/rtpmanager/rtpsource.c (rtp_source_process_rtp, rtp_source_send_rtp, rtp_source_process_sr, rtp_source_process_rb): More format arg fixing (spotted by Ali Sabil <ali.sabil@gmail.com>). * gst/switch/Makefile.am: Add require libraries(spotted by Ali Sabil <ali.sabil@gmail.com>).
* gst/level/gstlevel.c: Revert last change.David Schleef2007-05-101-3/+3
| | | | | | Original commit message from CVS: * gst/level/gstlevel.c: Revert last change.
* gst/level/gstlevel.c: Use guint8 * instead of gpointer then vs6 know the ↵Sébastien Moutte2007-05-092-6/+6
| | | | | | | | | | | | | | | | size of data pointed when moving the pointer. Original commit message from CVS: * gst/level/gstlevel.c: (gst_level_calculate_##TYPE), (gst_level_transform_ip): Use guint8 * instead of gpointer then vs6 know the size of data pointed when moving the pointer. * gst/rtp/gstrtph264pay.c: (gst_rtp_h264_pay_handle_buffer): Move instructions after variables declaration. * win32/vs6/autogen.dsp: * win32/vs6/libgstrtp.dsp: * win32/vs6/libgstrtsp.dsp: Update vs6 project files.
* gst/rtsp/: Add code to parse time ranges.Wim Taymans2007-05-094-3/+302
| | | | | | | | | | | | | Original commit message from CVS: * gst/rtsp/Makefile.am: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_handle_src_query), (gst_rtspsrc_send_keep_alive), (gst_rtspsrc_open): * gst/rtsp/rtsprange.c: (parse_npt_time), (parse_npt_range), (parse_clock_range), (parse_smpte_range), (rtsp_range_parse), (rtsp_range_free): * gst/rtsp/rtsprange.h: Add code to parse time ranges. Report DURATION on the stream when possible.
* gst/videomixer/videomixer.c: Fix strides calculation for AYUV (it's just ↵Tim-Philipp Müller2007-05-081-19/+9
| | | | | | | | | | width*4) (#436910). Original commit message from CVS: * gst/videomixer/videomixer.c: (gst_videomixer_blend_ayuv_ayuv), (gst_videomixer_fill_checker), (gst_videomixer_fill_color), (gst_videomixer_collected): Fix strides calculation for AYUV (it's just width*4) (#436910).
* gst/audiofx/: Sync the GObject properties before each processing step to ↵Sebastian Dröge2007-05-063-4/+13
| | | | | | | | | | | properly work with the controller. Original commit message from CVS: * gst/audiofx/audioamplify.c: (gst_audio_amplify_transform_ip): * gst/audiofx/audiodynamic.c: (gst_audio_dynamic_transform_ip): * gst/audiofx/audioinvert.c: (gst_audio_invert_transform_ip): Sync the GObject properties before each processing step to properly work with the controller.
* gst/rtsp/gstrtspsrc.c: Let more error state trickle down so that we can ↵Wim Taymans2007-05-041-22/+41
| | | | | | | | | | | | | | | | catch more error cases. Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_send_keep_alive), (gst_rtspsrc_loop_udp), (gst_rtspsrc_try_send), (gst_rtspsrc_send), (gst_rtspsrc_setup_streams), (gst_rtspsrc_open), (gst_rtspsrc_close), (gst_rtspsrc_play), (gst_rtspsrc_pause), (gst_rtspsrc_change_state): Let more error state trickle down so that we can catch more error cases. Handle keep-alive a little smarter by selecting a method the server actually supports. Fix a race in UDP streaming shutdown.
* gst/rtsp/gstrtspsrc.c: Ignore errors when trying to use the keep-alive messages.Wim Taymans2007-05-041-1/+3
| | | | | | Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_send_keep_alive): Ignore errors when trying to use the keep-alive messages.
* gst/rtsp/gstrtspsrc.c: Send RTCP messages back to the server over the TCP ↵Wim Taymans2007-05-043-59/+167
| | | | | | | | | | | | | | | | | | | | | connection. Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_handle_src_event), (gst_rtspsrc_handle_src_query), (gst_rtspsrc_sink_chain), (gst_rtspsrc_stream_configure_manager), (gst_rtspsrc_stream_free_udp), (gst_rtspsrc_stream_configure_tcp), (gst_rtspsrc_stream_configure_mcast), (gst_rtspsrc_stream_configure_udp), (gst_rtspsrc_stream_configure_udp_sink), (gst_rtspsrc_stream_configure_transport): Send RTCP messages back to the server over the TCP connection. * gst/rtsp/rtspconnection.c: (rtsp_connection_write), (rtsp_connection_send), (rtsp_connection_read), (read_body), (rtsp_connection_receive): * gst/rtsp/rtspconnection.h: Factor out and expose lowlevel _write and _read methods. Implement sending data messages to the server.
* gst/multipart/multipartmux.c: Fix timestamps on outgoing buffers.Wim Taymans2007-05-031-4/+10
| | | | | | | Original commit message from CVS: * gst/multipart/multipartmux.c: (gst_multipart_mux_queue_pads), (gst_multipart_mux_collected): Fix timestamps on outgoing buffers.
* gst/multipart/multipartmux.c: Emit NEWSEGMENT events before pushing the ↵Wim Taymans2007-05-031-55/+52
| | | | | | | | | | first buffer. Original commit message from CVS: * gst/multipart/multipartmux.c: (gst_multipart_mux_request_new_pad), (gst_multipart_mux_collected), (gst_multipart_mux_change_state): Emit NEWSEGMENT events before pushing the first buffer.
* gst/rtsp/gstrtspsrc.c: Refactor transport configuration code.Wim Taymans2007-05-031-220/+416
| | | | | | | | | | | | | | | | | | | | | | Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_finalize), (gst_rtspsrc_alloc_udp_ports), (gst_rtspsrc_handle_src_event), (gst_rtspsrc_handle_src_query), (gst_rtspsrc_stream_configure_manager), (gst_rtspsrc_stream_free_udp), (gst_rtspsrc_stream_configure_tcp), (gst_rtspsrc_stream_configure_mcast), (gst_rtspsrc_stream_configure_udp), (gst_rtspsrc_stream_configure_udp_sink), (gst_rtspsrc_stream_configure_transport), (gst_rtspsrc_push_event), (gst_rtspsrc_loop_udp), (gst_rtspsrc_open), (gst_rtspsrc_parse_rtpinfo), (gst_rtspsrc_play), (gst_rtspsrc_pause): Refactor transport configuration code. Create internal pads for TCP transport so that we can implement events and queries. Handle events and queries. Parse range from the SDP. Fix race in pause handler where the connection could still be flushing.
* gst/rtsp/gstrtspsrc.*: Fix race when multiple udp sources post timeouts, ↵Wim Taymans2007-05-023-102/+163
| | | | | | | | | | | | | | | | | | | | | just act on the first received timeout. Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_init), (gst_rtspsrc_finalize), (new_session_pad), (request_pt_map), (gst_rtspsrc_loop_send_cmd), (gst_rtspsrc_try_send), (gst_rtspsrc_send), (gst_rtspsrc_async_open), (gst_rtspsrc_close), (gst_rtspsrc_play), (gst_rtspsrc_handle_message), (gst_rtspsrc_change_state): * gst/rtsp/gstrtspsrc.h: Fix race when multiple udp sources post timeouts, just act on the first received timeout. Protect stream list with a recursive lock to fix some races. Flush connection when we need to do a reconnect or stop. Make state lock recursive. * gst/rtsp/rtspconnection.c: (rtsp_connection_connect), (rtsp_connection_close): Some small cleanups.
* gst/wavparse/gstwavparse.c: Only set DISCONT when there actually is a ↵Wim Taymans2007-05-021-2/+15
| | | | | | | | | | discont or when we just started. Original commit message from CVS: * gst/wavparse/gstwavparse.c: (gst_wavparse_perform_seek), (gst_wavparse_stream_headers), (gst_wavparse_stream_data): Only set DISCONT when there actually is a discont or when we just started.
* gst/wavparse/gstwavparse.*: Be a bit more clever when dealing with VBR files ↵Wim Taymans2007-05-022-21/+36
| | | | | | | | | | | | | | with FACT tags, we don't want to timesta... Original commit message from CVS: * gst/wavparse/gstwavparse.c: (gst_wavparse_perform_seek), (gst_wavparse_stream_headers), (gst_wavparse_add_src_pad), (gst_wavparse_stream_data): * gst/wavparse/gstwavparse.h: Be a bit more clever when dealing with VBR files with FACT tags, we don't want to timestamp buffers in that case but the estimated BPS can be used for seeking. Only send close segment in the streaming thread.
* gst/rtsp/test.c: Fix compilation of deprecated test just because I'm too ↵Wim Taymans2007-05-021-8/+8
| | | | | | | | | lazy to delete it. Original commit message from CVS: * gst/rtsp/test.c: (main): Fix compilation of deprecated test just because I'm too lazy to delete it.
* gst/rtsp/gstrtspsrc.*: Fix sending RTCP to the right place.Wim Taymans2007-05-027-99/+289
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_init), (gst_rtspsrc_finalize), (gst_rtspsrc_stream_configure_transport), (gst_rtspsrc_handle_request), (gst_rtspsrc_loop_interleaved), (gst_rtspsrc_send_keep_alive), (gst_rtspsrc_loop_udp), (gst_rtspsrc_loop_send_cmd), (gst_rtspsrc_try_send), (gst_rtspsrc_open), (gst_rtspsrc_handle_message): * gst/rtsp/gstrtspsrc.h: Fix sending RTCP to the right place. Fix bug in reffing the wrong UDP element. Use new pad names for the session manager. Implement handling server requests in interleaved and UDP modes. Handle session keep-alive in UDP modes. Remove GCond for handling UDP timeouts. * gst/rtsp/rtspconnection.c: (rtsp_connection_connect), (rtsp_connection_send), (rtsp_connection_read), (read_body), (rtsp_connection_receive), (rtsp_connection_close): * gst/rtsp/rtspconnection.h: Store connection IP address for later. Add timeout args to all operations that might block forever. Parse session timeout. Only close sockets when not already closed. * gst/rtsp/rtspdefs.c: * gst/rtsp/rtspdefs.h: Add timeout return value and error string. * gst/rtsp/rtspmessage.c: (rtsp_message_init_response): Add small comment.
* gst/rtp/gstrtpmp4vpay.*: Handle NEWSEGMENT and FLUSH events. Fixes #434824.Sjoerd Simons2007-05-012-0/+44
| | | | | | | | | Original commit message from CVS: Patch by: Sjoerd Simons <sjoerd at luon dot net> * gst/rtp/gstrtpmp4vpay.c: (gst_rtp_mp4v_pay_init), (gst_rtp_mp4v_pay_empty), (gst_rtp_mp4v_pay_event): * gst/rtp/gstrtpmp4vpay.h: Handle NEWSEGMENT and FLUSH events. Fixes #434824.
* gst/udp/gstmultiudpsink.c: Add code to drop membership of a multicast group.Wim Taymans2007-04-293-6/+84
| | | | | | | | | | | | | | Original commit message from CVS: * gst/udp/gstmultiudpsink.c: (leave_multicast), (gst_multiudpsink_add), (gst_multiudpsink_remove): Add code to drop membership of a multicast group. * gst/udp/gstudpsink.c: (gst_udpsink_update_uri), (gst_udpsink_set_uri): Implement URI handler. * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_stream_configure_transport), (gst_rtspsrc_parse_rtpinfo): Use URI handler to make udpsink instace. Improve code to configure port and destination.
* gst/udp/gstmultiudpsink.c: Fix multicast detection.Wim Taymans2007-04-292-3/+6
| | | | | | | | | Original commit message from CVS: * gst/udp/gstmultiudpsink.c: (gst_multiudpsink_add): Fix multicast detection. Don't try to join a multicast group if the address is not multicast. * gst/udp/gstudpsrc.c: (gst_udpsrc_update_uri): Small debug improvement.
* gst/rtsp/gstrtspsrc.c: Ignore ASYNC state messages from the udpsink, it's ↵Wim Taymans2007-04-271-1/+11
| | | | | | | | | | | irrelevant for the parent. Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_stream_configure_transport), (gst_rtspsrc_parse_rtpinfo), (gst_rtspsrc_play), (gst_rtspsrc_handle_message): Ignore ASYNC state messages from the udpsink, it's irrelevant for the parent.
* gst/rtp/gstrtpilbcdepay.h: Fix mode property when specified as an arg.Wim Taymans2007-04-271-2/+2
| | | | | | Original commit message from CVS: * gst/rtp/gstrtpilbcdepay.h: Fix mode property when specified as an arg.
* gst/rtsp/gstrtspsrc.*: Protect state changes with a lock.Wim Taymans2007-04-264-8/+87
| | | | | | | | | | | | | | | Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_init), (gst_rtspsrc_finalize), (gst_rtspsrc_stream_configure_transport), (gst_rtspsrc_open), (gst_rtspsrc_close), (gst_rtspsrc_parse_rtpinfo), (gst_rtspsrc_play), (gst_rtspsrc_pause): * gst/rtsp/gstrtspsrc.h: Protect state changes with a lock. * gst/rtsp/rtspconnection.c: (rtsp_connection_create), (parse_line): * gst/rtsp/rtspconnection.h: Remove some unused stuff.
* gst/udp/gstudpsrc.c: Handle the case where there are exactly 0 bytes to read ↵Wim Taymans2007-04-261-2/+14
| | | | | | | | | and the ioctl did not report an error. F... Original commit message from CVS: * gst/udp/gstudpsrc.c: (gst_udpsrc_create): Handle the case where there are exactly 0 bytes to read and the ioctl did not report an error. Fixes #433530.
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