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* gst/avi/gstavidemux.*: Parse subtitle text streams instead of erroring out ↵Tim-Philipp Müller2007-05-302-2/+22
| | | | | | | | | | | (#442034). Still needs a parser for the su... Original commit message from CVS: * gst/avi/gstavidemux.c: (gst_avi_demux_base_init), (gst_avi_demux_reset), (gst_avi_demux_parse_stream): * gst/avi/gstavidemux.h: Parse subtitle text streams instead of erroring out (#442034). Still needs a parser for the subtitles to actually show up.
* gst/avi/gstavidemux.c: Make _push_event() return TRUE if the event could be ↵Tim-Philipp Müller2007-05-301-8/+9
| | | | | | | | | | | | pushed on at least one pad and not only i... Original commit message from CVS: * gst/avi/gstavidemux.c: (gst_avi_demux_push_event), (gst_avi_demux_loop): Make _push_event() return TRUE if the event could be pushed on at least one pad and not only if it could be pushed on all pads, otherwise we'll end up posting an error message on EOS if one or more source pads are not connected.
* gst/rtsp/rtsptransport.c: Use renamed RTP bin.Wim Taymans2007-05-281-1/+1
| | | | | | Original commit message from CVS: * gst/rtsp/rtsptransport.c: Use renamed RTP bin.
* gst/videobox/gstvideobox.c: Add AYUV->AYUV and AYUV->I420 formats.Dejan Sakelšak2007-05-281-119/+588
| | | | | | | | | | | | | | | | Original commit message from CVS: Based on patch by: Dejan Sakelšak <sakdean at gmail dot com> * gst/videobox/gstvideobox.c: (gst_video_box_class_init), (gst_video_box_set_property), (gst_video_box_transform_caps), (video_box_recalc_transform), (gst_video_box_set_caps), (gst_video_box_get_unit_size), (gst_video_box_apply_alpha), (gst_video_box_ayuv_ayuv), (gst_video_box_clear), (UVfloor), (UVceil), (gst_video_box_ayuv_i420), (gst_video_box_i420_ayuv), (gst_video_box_i420_i420), (gst_video_box_transform), (plugin_init): Add AYUV->AYUV and AYUV->I420 formats. Fix negotiation and I420->AYUV conversion. Fixes #429329.
* gst/id3demux/gstid3demux.c: Don't leak newsegment events.Tim-Philipp Müller2007-05-251-0/+1
| | | | | | Original commit message from CVS: * gst/id3demux/gstid3demux.c: (gst_id3demux_sink_event): Don't leak newsegment events.
* gst/wavparse/Makefile.am: Add '-lm' to LIBS for ceil(), don't assume one of ↵Tim-Philipp Müller2007-05-251-1/+2
| | | | | | | | | our dependencies drags it in. Original commit message from CVS: * gst/wavparse/Makefile.am: Add '-lm' to LIBS for ceil(), don't assume one of our dependencies drags it in.
* gst/: Handle and adjust new-segment events so that downstream really sees a ↵Jan Schmidt2007-05-253-0/+250
| | | | | | | | | | | | | | | | | | stream with the tag pieces stripped off t... Original commit message from CVS: * gst/id3demux/gstid3demux.c: (gst_id3demux_reset), (gst_id3demux_send_new_segment), (gst_id3demux_chain), (gst_id3demux_sink_event): * gst/id3demux/gstid3demux.h: * gst/apetag/gsttagdemux.c: (gst_tag_demux_reset), (gst_tag_demux_chain), (gst_tag_demux_sink_event), (gst_tag_demux_send_new_segment): Handle and adjust new-segment events so that downstream really sees a stream with the tag pieces stripped off the front and back. Fixes strangeness in seeking when mp3 decoders use the new-segment byte position to estimate their current playback position timestamp and then the arriving buffers don't match up.
* gst/autodetect/gstautoaudiosink.c: Don't unnecessarily perform a ↵Jan Schmidt2007-05-251-1/+6
| | | | | | | | | READY->NULL->READY transition on the detected audio ... Original commit message from CVS: * gst/autodetect/gstautoaudiosink.c: (gst_auto_audio_sink_detect): Don't unnecessarily perform a READY->NULL->READY transition on the detected audio sink when starting up. Fixes: #440127
* gst/rtsp/gstrtspsrc.c: Init value to avoid infinte loops.Wim Taymans2007-05-241-1/+2
| | | | | | Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_parse_methods): Init value to avoid infinte loops.
* gst/rtsp/: Fix for new API.Peter Kjellerstedt2007-05-245-113/+167
| | | | | | | | | | | | | | | | | | | | | | | Original commit message from CVS: Patch by: Peter Kjellerstedt <pkj at axis com> * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_setup_auth), (gst_rtspsrc_try_send), (gst_rtspsrc_parse_methods), (gst_rtspsrc_setup_streams), (gst_rtspsrc_open), (gst_rtspsrc_play): (rtsp_connection_send), (rtsp_connection_receive): * gst/rtsp/rtspextwms.c: (rtsp_ext_wms_after_send): Fix for new API. * gst/rtsp/rtspconnection.c: (add_auth_header), Only add authorisation and session headers when sending messages. * gst/rtsp/rtspmessage.c: (key_value_foreach), (rtsp_message_init), (rtsp_message_init_request), (rtsp_message_init_response), (rtsp_message_unset), (rtsp_message_add_header), (rtsp_message_remove_header), (rtsp_message_get_header), (rtsp_message_append_headers), (dump_key_value), (rtsp_message_dump): * gst/rtsp/rtspmessage.h: Add support for multiple headers of the same type by storing the parsed headers in a GArray instaed of a hashtable.
* configure.ac: Depend on gstreamer-0.10.12.1. gst/equalizer/gstiirequalizer.c ↵Stefan Kost2007-05-225-31/+28
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | (ARG_BAND_WIDTH, _do_init, ARG_GAIN, _Gs... Original commit message from CVS: * configure.ac: Depend on gstreamer-0.10.12.1. * gst/equalizer/gstiirequalizer.c (ARG_BAND_WIDTH, _do_init, ARG_GAIN, _GstIirEqualizerBand, object, _GstIirEqualizerBandClass, parent_class, gst_iir_equalizer_band_set_property, gst_iir_equalizer_band_class_init, gst_iir_equalizer_band_get_type, gst_iir_equalizer_child_proxy_get_child_by_index, gst_iir_equalizer_child_proxy_get_children_count, gst_iir_equalizer_child_proxy_interface_init, setup_filter, gst_iir_equalizer_compute_frequencies, gst_iir_equalizer_set_property, gst_iir_equalizer_get_property, plugin_init): * gst/equalizer/gstiirequalizer.h (audiofilter): * gst/equalizer/gstiirequalizernbands.c (ARG_NUM_BANDS, gst_iir_equalizer_nbands_base_init, gst_iir_equalizer_nbands_init, gst_iir_equalizer_nbands_set_property): Use new locking macros. * gst/filter/gstbpwsinc.c (bpwsinc_set_caps): Add fixme. * gst/spectrum/gstspectrum.c (SPECTRUM_WINDOW_BASE, SPECTRUM_WINDOW_LEN, gst_spectrum_init, gst_spectrum_set_property, gst_spectrum_event, gst_spectrum_transform_ip): Use new locking macros. Turn two fixed values into #defines.
* ChangeLog: ChangeLog surgery. gst/equalizer/gstiirequalizer.c ↵Stefan Kost2007-05-211-0/+2
| | | | | | | | | | | | | | | | | | (ARG_BAND_WIDTH, _do_init, ARG_GAIN, _GstIirEqualizerBa... Original commit message from CVS: * ChangeLog: ChangeLog surgery. * gst/equalizer/gstiirequalizer.c (ARG_BAND_WIDTH, _do_init, ARG_GAIN, _GstIirEqualizerBand, object, _GstIirEqualizerBandClass, parent_class, gst_iir_equalizer_band_set_property, gst_iir_equalizer_band_class_init, gst_iir_equalizer_band_get_type, gst_iir_equalizer_child_proxy_get_child_by_index, gst_iir_equalizer_child_proxy_get_children_count, gst_iir_equalizer_child_proxy_interface_init, setup_filter, gst_iir_equalizer_compute_frequencies, plugin_init): * tests/icles/equalizer-test.c: Add fixme and comment for example.
* gst/spectrum/gstspectrum.c (gst_spectrum_set_property, gst_spectrum_event, ↵Stefan Kost2007-05-211-1/+2
| | | | | | | | | gst_spectrum_transform_ip): Original commit message from CVS: * gst/spectrum/gstspectrum.c (gst_spectrum_set_property, gst_spectrum_event, gst_spectrum_transform_ip): Use lock to protect from concurrent access.
* gst/udp/gstudpsrc.c: Since we depend on 0.10.13 -core, override the ↵Wim Taymans2007-05-211-31/+32
| | | | | | | | | | unlock_stop vmethod for safer shutdown. Original commit message from CVS: * gst/udp/gstudpsrc.c: (gst_udpsrc_class_init), (gst_udpsrc_create), (gst_udpsrc_unlock), (gst_udpsrc_unlock_stop): Since we depend on 0.10.13 -core, override the unlock_stop vmethod for safer shutdown.
* gst/rtsp/gstrtpdec.*: Added signal for backwards compat.Wim Taymans2007-05-212-0/+8
| | | | | | | Original commit message from CVS: * gst/rtsp/gstrtpdec.c: (gst_rtp_dec_class_init): * gst/rtsp/gstrtpdec.h: Added signal for backwards compat.
* Use audioconvert for converting from non-native endianness floats in auparse ↵René Stadler2007-05-212-41/+2
| | | | | | | | | | | | | | instead of doing it ourself. Fixes #424527. Original commit message from CVS: Patch by: René Stadler <mail at renestadler dot de> * configure.ac: * gst/auparse/gstauparse.c: (gst_au_parse_reset), (gst_au_parse_parse_header), (gst_au_parse_chain): * gst/auparse/gstauparse.h: Use audioconvert for converting from non-native endianness floats in auparse instead of doing it ourself. Fixes #424527. This needs the audioconvert from plugins-base CVS.
* gst/rtp/gstrtph263ppay.c: Fix enum registration.Wim Taymans2007-05-211-2/+2
| | | | | | | Original commit message from CVS: * gst/rtp/gstrtph263ppay.c: (gst_fragmentation_mode_get_type), (gst_rtp_h263p_pay_flush): Fix enum registration.
* 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.
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