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2007-04-29autogen.sh: Require automake 1.7Thomas Vander Stichele1-0/+17
Original commit message from CVS: * autogen.sh: Require automake 1.7 * ext/alsaspdif/Makefile.am: * ext/divx/Makefile.am: * ext/ivorbis/Makefile.am: * ext/musicbrainz/Makefile.am: * ext/neon/Makefile.am: * ext/sdl/Makefile.am: * ext/swfdec/Makefile.am: * ext/theora/Makefile.am: * ext/wavpack/Makefile.am: * ext/xvid/Makefile.am: * gst/modplug/Makefile.am: Fix up Makefile.am accordingly.
2007-04-29docs/plugins/inspect/: Add jack and update.Thomas Vander Stichele1-0/+32
Original commit message from CVS: * docs/plugins/inspect/plugin-alsaspdif.xml: * docs/plugins/inspect/plugin-bz2.xml: * docs/plugins/inspect/plugin-cdxaparse.xml: * docs/plugins/inspect/plugin-dfbvideosink.xml: * docs/plugins/inspect/plugin-faac.xml: * docs/plugins/inspect/plugin-faad.xml: * docs/plugins/inspect/plugin-filter.xml: * docs/plugins/inspect/plugin-freeze.xml: * docs/plugins/inspect/plugin-glimagesink.xml: * docs/plugins/inspect/plugin-gsm.xml: * docs/plugins/inspect/plugin-h264parse.xml: * docs/plugins/inspect/plugin-jack.xml: * docs/plugins/inspect/plugin-mms.xml: * docs/plugins/inspect/plugin-modplug.xml: * docs/plugins/inspect/plugin-musepack.xml: * docs/plugins/inspect/plugin-musicbrainz.xml: * docs/plugins/inspect/plugin-neon.xml: * docs/plugins/inspect/plugin-nsfdec.xml: * docs/plugins/inspect/plugin-replaygain.xml: * docs/plugins/inspect/plugin-sdl.xml: * docs/plugins/inspect/plugin-soundtouch.xml: * docs/plugins/inspect/plugin-spectrum.xml: * docs/plugins/inspect/plugin-speed.xml: * docs/plugins/inspect/plugin-tta.xml: * docs/plugins/inspect/plugin-videocrop.xml: * docs/plugins/inspect/plugin-wavpack.xml: * docs/plugins/inspect/plugin-xingheader.xml: * docs/plugins/inspect/plugin-xvid.xml: Add jack and update.
2007-04-28configure.ac: Don't build equalizer unless we have core from CVS (it won't ↵Tim-Philipp Müller1-0/+7
work with earlier versions due to GstChild... Original commit message from CVS: * configure.ac: Don't build equalizer unless we have core from CVS (it won't work with earlier versions due to GstChildProxy brokeness). Also up requirements to last released core/base.
2007-04-27gst/rtpmanager/gstrtpsession.c: Move reconsideration code to the rtpsession ↵Wim Taymans1-0/+34
object. Original commit message from CVS: * gst/rtpmanager/gstrtpsession.c: (rtcp_thread), (gst_rtp_session_send_rtcp), (gst_rtp_session_reconsider): Move reconsideration code to the rtpsession object. Simplify timout handling and add reconsideration. * gst/rtpmanager/rtpsession.c: (rtp_session_class_init), (rtp_session_init), (rtp_session_finalize), (on_bye_ssrc), (on_bye_timeout), (on_timeout), (rtp_session_set_callbacks), (obtain_source), (rtp_session_create_source), (update_arrival_stats), (rtp_session_process_rtp), (rtp_session_process_sr), (rtp_session_process_rr), (rtp_session_process_bye), (rtp_session_process_rtcp), (calculate_rtcp_interval), (rtp_session_send_bye), (rtp_session_next_timeout), (session_start_rtcp), (session_report_blocks), (session_cleanup), (session_sdes), (session_bye), (is_rtcp_time), (rtp_session_on_timeout): * gst/rtpmanager/rtpsession.h: Handle timeout of inactive sources and senders. Implement BYE scheduling. * gst/rtpmanager/rtpsource.c: (calculate_jitter), (rtp_source_process_sr), (rtp_source_get_last_sr), (rtp_source_get_last_rb): * gst/rtpmanager/rtpsource.h: Add members to check for timeouts. * gst/rtpmanager/rtpstats.c: (rtp_stats_init_defaults), (rtp_stats_calculate_rtcp_interval), (rtp_stats_add_rtcp_jitter), (rtp_stats_calculate_bye_interval): * gst/rtpmanager/rtpstats.h: Use RFC algorithm for calculating the reporting interval.
2007-04-26docs/plugins/: Add documentation for osxvideoEdward Hervey1-0/+8
Original commit message from CVS: * docs/plugins/gst-plugins-bad-plugins-docs.sgml: * docs/plugins/gst-plugins-bad-plugins-sections.txt: * docs/plugins/gst-plugins-bad-plugins.hierarchy: * docs/plugins/inspect/plugin-osxvideo.xml: Add documentation for osxvideo
2007-04-25gst/rtpmanager/gstrtpsession.c: Implement forward and reverse reconsideration.Wim Taymans1-0/+11
Original commit message from CVS: * gst/rtpmanager/gstrtpsession.c: (rtcp_thread): Implement forward and reverse reconsideration. * gst/rtpmanager/rtpsession.c: (rtp_session_get_num_sources), (rtp_session_get_num_active_sources), (rtp_session_process_sr), (session_report_blocks): * gst/rtpmanager/rtpsession.h: Small cleanups.
2007-04-25gst/rtpmanager/gstrtpbin.*: Make default jitterbuffer latency configurable.Wim Taymans1-0/+17
Original commit message from CVS: reviewed by: <delete if not using a buddy> * gst/rtpmanager/gstrtpbin.c: (create_stream), (gst_rtp_bin_class_init), (gst_rtp_bin_set_property), (gst_rtp_bin_get_property): * gst/rtpmanager/gstrtpbin.h: Make default jitterbuffer latency configurable. * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_rtp_jitter_buffer_class_init), (gst_jitter_buffer_sink_parse_caps), (gst_rtp_jitter_buffer_loop), (gst_rtp_jitter_buffer_set_property), (gst_rtp_jitter_buffer_get_property): Debuging cleanups.
2007-04-25gst/rtpmanager/gstrtpjitterbuffer.c: Report NO_PREROLL when going to PAUSED.Wim Taymans1-0/+22
Original commit message from CVS: * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_rtp_jitter_buffer_change_state): Report NO_PREROLL when going to PAUSED. * gst/rtpmanager/gstrtpsession.c: (rtcp_thread): Don't send RTCP right before we are shutting down. * gst/rtpmanager/rtpsession.c: (rtp_session_process_rtp), (rtp_session_process_sr), (session_report_blocks), (rtp_session_perform_reporting): Improve report blocks. * gst/rtpmanager/rtpsource.c: (calculate_jitter), (init_seq), (rtp_source_process_rtp), (rtp_source_process_sr), (rtp_source_process_rb), (rtp_source_get_last_sr), (rtp_source_get_last_rb): * gst/rtpmanager/rtpsource.h: * gst/rtpmanager/rtpstats.h: Cleanups, add methods to access stats.
2007-04-25gst/rtpmanager/gstrtpbin.c: fix for pad name changeWim Taymans1-0/+40
Original commit message from CVS: * gst/rtpmanager/gstrtpbin.c: (create_rtcp): fix for pad name change * gst/rtpmanager/gstrtpsession.c: (rtcp_thread), (gst_rtp_session_send_rtcp), (gst_rtp_session_clock_rate): Fix for renamed methods. * gst/rtpmanager/rtpsession.c: (rtp_session_init), (rtp_session_finalize), (rtp_session_set_cname), (rtp_session_get_cname), (rtp_session_set_name), (rtp_session_get_name), (rtp_session_set_email), (rtp_session_get_email), (rtp_session_set_phone), (rtp_session_get_phone), (rtp_session_set_location), (rtp_session_get_location), (rtp_session_set_tool), (rtp_session_get_tool), (rtp_session_set_note), (rtp_session_get_note), (source_push_rtp), (obtain_source), (rtp_session_add_source), (rtp_session_get_source_by_ssrc), (rtp_session_create_source), (rtp_session_process_rtp), (rtp_session_process_sr), (rtp_session_process_sdes), (rtp_session_process_rtcp), (rtp_session_send_rtp), (rtp_session_get_reporting_interval), (session_report_blocks), (session_sdes), (rtp_session_perform_reporting): * gst/rtpmanager/rtpsession.h: Prepare for implementing SSRC sampling. Create SSRC for the session. Add methods to set the SDES entries. fix accounting of senders/receivers. Implement SR/RR/SDES RTCP reporting. * gst/rtpmanager/rtpsource.c: (rtp_source_init), (init_seq), (rtp_source_process_rtp), (rtp_source_process_sr): * gst/rtpmanager/rtpsource.h: Implement extended sequence number. * gst/rtpmanager/rtpstats.c: (rtp_stats_calculate_rtcp_interval): * gst/rtpmanager/rtpstats.h: Rename some fields.
2007-04-24Plug some leaks; try to make build bot happy again.Tim-Philipp Müller1-0/+7
Original commit message from CVS: * gst/y4m/gsty4mencode.c: (gst_y4m_encode_init), (gst_y4m_encode_setcaps): * tests/check/elements/y4menc.c: (GST_START_TEST): Plug some leaks; try to make build bot happy again.
2007-04-21gst/Makefile.am: Fix distcheck, hopefully (rtpmanager is already in ↵Tim-Philipp Müller1-0/+5
GST_PLUGINS_ALL). Original commit message from CVS: * gst/Makefile.am: Fix distcheck, hopefully (rtpmanager is already in GST_PLUGINS_ALL).
2007-04-21gst/rtpmanager/rtpsession.c: Don't use GLib-2.10 API, we only require GLib ↵Tim-Philipp Müller1-0/+5
2.8 at the moment. Original commit message from CVS: * gst/rtpmanager/rtpsession.c: (rtp_session_finalize): Don't use GLib-2.10 API, we only require GLib 2.8 at the moment.
2007-04-20ext/faad/gstfaad.c: FAAD fails to decode low (e.g. 8 kHz) sample rate AAC ↵Michael Smith1-0/+8
data in quicktime because of sample rate mi... Original commit message from CVS: * ext/faad/gstfaad.c: (gst_faad_open_decoder): FAAD fails to decode low (e.g. 8 kHz) sample rate AAC data in quicktime because of sample rate mismatches. Reenable overriding the implicit SBR behaviour (accidently changed?) to allow playback of these files.
2007-04-19configure.ac: Change rtpmanager disabling to keep -bad releasable.David Schleef1-0/+5
Original commit message from CVS: * configure.ac: Change rtpmanager disabling to keep -bad releasable.
2007-04-18Fix wtay's hack. rtpmanager is disabled in configure.ac on line 268.David Schleef1-0/+7
Original commit message from CVS: * configure.ac: * gst/Makefile.am: Fix wtay's hack. rtpmanager is disabled in configure.ac on line 268.
2007-04-18gst/Makefile.am: Add rtpmanager dir to dist.Wim Taymans1-0/+5
Original commit message from CVS: * gst/Makefile.am: Add rtpmanager dir to dist.
2007-04-18configure.ac: Disable rtpmanager for now because it depends on CVS -base.Wim Taymans1-0/+78
Original commit message from CVS: * configure.ac: Disable rtpmanager for now because it depends on CVS -base. * gst/rtpmanager/Makefile.am: Added new files for session manager. * gst/rtpmanager/gstrtpjitterbuffer.h: * gst/rtpmanager/gstrtpbin.c: (create_session), (get_pt_map), (create_stream), (pt_map_requested), (new_ssrc_pad_found): Some cleanups. the session manager can now also request a pt-map. * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_base_init), (gst_rtp_session_class_init), (gst_rtp_session_init), (gst_rtp_session_finalize), (rtcp_thread), (start_rtcp_thread), (stop_rtcp_thread), (gst_rtp_session_change_state), (gst_rtp_session_process_rtp), (gst_rtp_session_send_rtp), (gst_rtp_session_send_rtcp), (gst_rtp_session_clock_rate), (gst_rtp_session_get_time), (gst_rtp_session_event_recv_rtp_sink), (gst_rtp_session_chain_recv_rtp), (gst_rtp_session_event_recv_rtcp_sink), (gst_rtp_session_chain_recv_rtcp), (gst_rtp_session_event_send_rtp_sink), (gst_rtp_session_chain_send_rtp), (create_send_rtcp_src), (gst_rtp_session_request_new_pad): * gst/rtpmanager/gstrtpsession.h: We can ask for pt-map now too when the session manager needs it. Hook up to the new session manager, implement the needed callbacks for pushing data, getting clock time and requesting clock-rates. Rename rtcp_src to send_rtcp_src to make it clear that this RTCP is to be send to clients. Add code to start and stop the thread that will schedule RTCP through the session manager. * gst/rtpmanager/rtpsession.c: (rtp_session_class_init), (rtp_session_init), (rtp_session_finalize), (rtp_session_set_property), (rtp_session_get_property), (on_new_ssrc), (on_ssrc_collision), (on_ssrc_validated), (on_bye_ssrc), (rtp_session_new), (rtp_session_set_callbacks), (rtp_session_set_bandwidth), (rtp_session_get_bandwidth), (rtp_session_set_rtcp_bandwidth), (rtp_session_get_rtcp_bandwidth), (source_push_rtp), (source_clock_rate), (check_collision), (obtain_source), (rtp_session_add_source), (rtp_session_get_num_sources), (rtp_session_get_num_active_sources), (rtp_session_get_source_by_ssrc), (rtp_session_get_source_by_cname), (rtp_session_create_source), (update_arrival_stats), (rtp_session_process_rtp), (rtp_session_process_sr), (rtp_session_process_rr), (rtp_session_process_sdes), (rtp_session_process_bye), (rtp_session_process_app), (rtp_session_process_rtcp), (rtp_session_send_rtp), (rtp_session_get_rtcp_interval), (rtp_session_produce_rtcp): * gst/rtpmanager/rtpsession.h: The advanced beginnings of the main session manager that handles the participant database of RTPSources, SSRC probation, SSRC collisions, parse RTCP to update source stats. etc.. * gst/rtpmanager/rtpsource.c: (rtp_source_class_init), (rtp_source_init), (rtp_source_finalize), (rtp_source_new), (rtp_source_set_callbacks), (rtp_source_set_as_csrc), (rtp_source_set_rtp_from), (rtp_source_set_rtcp_from), (push_packet), (get_clock_rate), (calculate_jitter), (rtp_source_process_rtp), (rtp_source_process_bye), (rtp_source_send_rtp), (rtp_source_process_sr), (rtp_source_process_rb): * gst/rtpmanager/rtpsource.h: Object that encapsulates an SSRC and its state in the database. Calculates the jitter and transit times of data packets. * gst/rtpmanager/rtpstats.c: (rtp_stats_init_defaults), (rtp_stats_calculate_rtcp_interval), (rtp_stats_add_rtcp_jitter): * gst/rtpmanager/rtpstats.h: Various stats regarding the session and sources. Used to calculate the RTCP interval.
2007-04-17gst/app/Makefile.am: Fix CFLAGS and hopefully #430594.Tim-Philipp Müller1-0/+5
Original commit message from CVS: * gst/app/Makefile.am: Fix CFLAGS and hopefully #430594.
2007-04-17gst/nsf/types.h: Rename #ifndef header guard symbol to something less ↵Tim-Philipp Müller1-0/+8
generic, so types.h doesn't get skipped over wh... Original commit message from CVS: * gst/nsf/types.h: Rename #ifndef header guard symbol to something less generic, so types.h doesn't get skipped over when compiling on MingW. Include GLib headers and use those to set the endianness and the basic types so that this isn't entirely broken for non-x86 architectures.
2007-04-17gst/mve/gstmvedemux.c: Use G_LITTLE_ENDIAN instead of LITTLE_ENDIAN, so ↵Tim-Philipp Müller1-0/+7
stuff compiles on Original commit message from CVS: * gst/mve/gstmvedemux.c: (gst_mve_audio_init): Use G_LITTLE_ENDIAN instead of LITTLE_ENDIAN, so stuff compiles on MingW (no idea though why we add a BYTE_ORDER endianness field if the audio is compressed).
2007-04-14docs/plugins/inspect/: Add xml doc files for Windows sinksSébastien Moutte1-0/+11
Original commit message from CVS: * docs/plugins/inspect/plugin-directdraw.xml: * docs/plugins/inspect/plugin-directsound.xml: * docs/plugins/inspect/plugin-waveform.xml: Add xml doc files for Windows sinks * win32/vs6/libgstqtdemux.dsp: * win32/vs6/libgstmpegvideoparse.dsp: * win32/vs6/gst_plugins_bad.dsw: Update projects files.
2007-04-13gst/rtpmanager/: Protect lists and structures with locks.Wim Taymans1-0/+16
Original commit message from CVS: * gst/rtpmanager/gstrtpbin.c: (create_session), (get_pt_map), (gst_rtp_bin_init), (gst_rtp_bin_finalize), (new_ssrc_pad_found), (create_recv_rtp), (gst_rtp_bin_request_new_pad): * gst/rtpmanager/gstrtpbin.h: * gst/rtpmanager/gstrtpclient.c: * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_class_init), (gst_rtp_session_init), (gst_rtp_session_finalize), (gst_rtp_session_event_recv_rtp_sink), (gst_rtp_session_event_recv_rtcp_sink), (gst_rtp_session_chain_recv_rtcp), (gst_rtp_session_request_new_pad): Protect lists and structures with locks. Return FLOW_OK from RTCP messages for now.
2007-04-12gst/qtdemux/qtdemux.c: Make timescale 32 bits again so we don't screw up the ↵Wim Taymans1-0/+6
pts_offset calculations. Original commit message from CVS: * gst/qtdemux/qtdemux.c: Make timescale 32 bits again so we don't screw up the pts_offset calculations.
2007-04-12gst/rtpmanager/gstrtpbin.c: Emit pt map requests and cache results.Wim Taymans1-0/+16
Original commit message from CVS: * gst/rtpmanager/gstrtpbin.c: (create_session), (get_pt_map), (create_stream), (gst_rtp_bin_class_init), (pt_map_requested): Emit pt map requests and cache results. * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_rtp_jitter_buffer_class_init), (gst_jitter_buffer_sink_parse_caps), (gst_jitter_buffer_sink_setcaps), (gst_rtp_jitter_buffer_get_clock_rate), (gst_rtp_jitter_buffer_chain), (gst_rtp_jitter_buffer_loop): * gst/rtpmanager/gstrtpjitterbuffer.h: * gst/rtpmanager/gstrtpptdemux.c: (gst_rtp_pt_demux_chain): Emit request-pt-map signals.
2007-04-11gst/rtpmanager/gstrtpbin-marshal.list: Some more custom marshallers.Wim Taymans1-0/+25
Original commit message from CVS: * gst/rtpmanager/gstrtpbin-marshal.list: Some more custom marshallers. * gst/rtpmanager/gstrtpbin.c: (create_session), (get_pt_map), (clock_rate_request), (create_stream), (gst_rtp_bin_class_init), (pt_map_requested), (new_ssrc_pad_found), (create_recv_rtp): * gst/rtpmanager/gstrtpbin.h: Prepare for caching pt maps. Connect to signals to collect pt maps. * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_rtp_jitter_buffer_class_init), (gst_jitter_buffer_sink_setcaps), (gst_rtp_jitter_buffer_loop): * gst/rtpmanager/gstrtpjitterbuffer.h: Add request_clock_rate signal. Use scale insteat of scale_int because the later does not deal with negative numbers. * gst/rtpmanager/gstrtpptdemux.c: (gst_rtp_pt_demux_class_init), (gst_rtp_pt_demux_chain): * gst/rtpmanager/gstrtpptdemux.h: Implement request-pt-map signal.
2007-04-11gst/qtdemux/: Handle version 1 mdhd atoms to get extended precision durations.Wim Taymans1-0/+9
Original commit message from CVS: * gst/qtdemux/qtdemux.c: (gst_qtdemux_get_duration), (qtdemux_parse_samples), (qtdemux_parse_segments), (qtdemux_parse_trak), (qtdemux_parse_tree): * gst/qtdemux/qtdemux_dump.c: (qtdemux_dump_mdhd): Handle version 1 mdhd atoms to get extended precision durations. Fixes #426972.
2007-04-10gst/rtpmanager/: Added custom marshallers for signals.Wim Taymans1-0/+16
Original commit message from CVS: * gst/rtpmanager/.cvsignore: * gst/rtpmanager/Makefile.am: * gst/rtpmanager/gstrtpbin-marshal.list: Added custom marshallers for signals. * gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_class_init): * gst/rtpmanager/gstrtpbin.h: Prepare for emiting pt map signals. * gst/rtpmanager/gstrtpptdemux.c: (gst_rtp_pt_demux_class_init): * gst/rtpmanager/gstrtpssrcdemux.c: (gst_rtp_ssrc_demux_class_init): Fix signals.
2007-04-06gst/rtpmanager/gstrtpbin.*: Provide a clock.Wim Taymans1-0/+7
Original commit message from CVS: * gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_class_init), (gst_rtp_bin_init), (gst_rtp_bin_provide_clock): * gst/rtpmanager/gstrtpbin.h: Provide a clock.
2007-04-06gst/rtpmanager/gstrtpbin.c: Fix pad template name parsing.Wim Taymans1-0/+5
Original commit message from CVS: * gst/rtpmanager/gstrtpbin.c: (create_rtcp): Fix pad template name parsing.
2007-04-05gst/rtpmanager/gstrtpjitterbuffer.c: Add some debug and comments.Wim Taymans1-0/+8
Original commit message from CVS: * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_jitter_buffer_sink_setcaps), (gst_rtp_jitter_buffer_chain), (gst_rtp_jitter_buffer_loop): Add some debug and comments. Fix double unref() in error cases.
2007-04-05gst/qtdemux/gstrtpxqtdepay.*: Try to recover from packet loss a little better.Wim Taymans1-0/+7
Original commit message from CVS: * gst/qtdemux/gstrtpxqtdepay.c: (gst_rtp_xqt_depay_process), (gst_rtp_xqt_depay_change_state): * gst/qtdemux/gstrtpxqtdepay.h: Try to recover from packet loss a little better.
2007-04-05gst/rtpmanager/gstrtpbin.*: Add debugging category.Wim Taymans1-0/+41
Original commit message from CVS: * gst/rtpmanager/gstrtpbin.c: (find_session_by_id), (create_session), (find_stream_by_ssrc), (create_stream), (gst_rtp_bin_class_init), (new_payload_found), (new_ssrc_pad_found), (create_recv_rtp), (create_recv_rtcp), (create_send_rtp), (create_rtcp): * gst/rtpmanager/gstrtpbin.h: Add debugging category. Added RTPStream to manage stream per SSRC, each with its own jitterbuffer and ptdemux. Added SSRCDemux. Connect to various SSRC and PT signals and create ghostpads, link stuff. * gst/rtpmanager/gstrtpmanager.c: (plugin_init): Added rtpbin to elements. * gst/rtpmanager/gstrtpptdemux.c: (gst_rtp_pt_demux_chain): Fix caps and forward GstFlowReturn * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_class_init), (gst_rtp_session_event_recv_rtp_sink), (gst_rtp_session_chain_recv_rtp), (gst_rtp_session_event_recv_rtcp_sink), (gst_rtp_session_chain_recv_rtcp), (gst_rtp_session_event_send_rtp_sink), (gst_rtp_session_chain_send_rtp), (create_recv_rtp_sink), (create_recv_rtcp_sink), (create_send_rtp_sink), (create_rtcp_src), (gst_rtp_session_request_new_pad): Add debug category. Add event handling * gst/rtpmanager/gstrtpssrcdemux.c: (find_rtp_pad_for_ssrc), (create_rtp_pad_for_ssrc), (gst_rtp_ssrc_demux_class_init), (gst_rtp_ssrc_demux_init), (gst_rtp_ssrc_demux_chain), (gst_rtp_ssrc_demux_change_state): * gst/rtpmanager/gstrtpssrcdemux.h: Add debug category. Add new-pt-pad signal.
2007-04-05po/: Added Danish translation.Thomas Vander Stichele1-0/+8
Original commit message from CVS: submitted by: Mogens Jaeger <mogens@jaeger.tf> * po/LINGUAS: * po/da.po: Added Danish translation.
2007-04-04gst/rtpmanager/: Added simple SSRC demuxer.Wim Taymans1-0/+13
Original commit message from CVS: * gst/rtpmanager/Makefile.am: * gst/rtpmanager/gstrtpmanager.c: (plugin_init): * gst/rtpmanager/gstrtpssrcdemux.c: (find_pad_for_ssrc), (create_pad_for_ssrc), (gst_rtp_ssrc_demux_base_init), (gst_rtp_ssrc_demux_class_init), (gst_rtp_ssrc_demux_init), (gst_rtp_ssrc_demux_finalize), (gst_rtp_ssrc_demux_sink_event), (gst_rtp_ssrc_demux_chain), (gst_rtp_ssrc_demux_src_event), (gst_rtp_ssrc_demux_change_state): * gst/rtpmanager/gstrtpssrcdemux.h: Added simple SSRC demuxer.
2007-04-04ext/jack/gstjackaudiosink.c: Try t better name clients. properly handle ↵Stefan Kost1-0/+7
return codes when re- establishing links. Original commit message from CVS: * ext/jack/gstjackaudiosink.c: (gst_jack_ring_buffer_open_device), (gst_jack_ring_buffer_acquire): Try t better name clients. properly handle return codes when re- establishing links.
2007-04-03sys/glsink/glimagesink.c: Fix handling of video/x-raw-yuv. Add overlay ↵David Schleef1-0/+5
handling. Original commit message from CVS: * sys/glsink/glimagesink.c: Fix handling of video/x-raw-yuv. Add overlay handling.
2007-04-03gst/rtpmanager/: Some more ghostpad magic.Wim Taymans1-0/+10
Original commit message from CVS: * gst/rtpmanager/gstrtpbin.c: (find_session_by_id), (create_session), (gst_rtp_bin_base_init), (create_recv_rtp), (create_recv_rtcp), (create_send_rtp), (create_rtcp), (gst_rtp_bin_request_new_pad): * gst/rtpmanager/gstrtpbin.h: * gst/rtpmanager/gstrtpclient.c: Some more ghostpad magic.
2007-04-03gst/rtpmanager/Makefile.am: Add .h file so it can be disted properly.Wim Taymans1-0/+5
Original commit message from CVS: * gst/rtpmanager/Makefile.am: Add .h file so it can be disted properly.
2007-04-03Add RTP session management elements. Still in progress.Wim Taymans1-0/+73
Original commit message from CVS: * configure.ac: * gst/rtpmanager/Makefile.am: * gst/rtpmanager/async_jitter_queue.c: (async_jitter_queue_new), (signal_waiting_threads), (async_jitter_queue_ref), (async_jitter_queue_ref_unlocked), (async_jitter_queue_set_low_threshold), (async_jitter_queue_set_high_threshold), (async_jitter_queue_set_max_queue_length), (async_jitter_queue_get_g_queue), (calculate_ts_diff), (async_jitter_queue_length_ts_units_unlocked), (async_jitter_queue_unref_and_unlock), (async_jitter_queue_unref), (async_jitter_queue_lock), (async_jitter_queue_unlock), (async_jitter_queue_push), (async_jitter_queue_push_unlocked), (async_jitter_queue_push_sorted), (async_jitter_queue_push_sorted_unlocked), (async_jitter_queue_insert_after_unlocked), (async_jitter_queue_pop_intern_unlocked), (async_jitter_queue_pop), (async_jitter_queue_pop_unlocked), (async_jitter_queue_length), (async_jitter_queue_length_unlocked), (async_jitter_queue_set_flushing_unlocked), (async_jitter_queue_unset_flushing_unlocked), (async_jitter_queue_set_blocking_unlocked): * gst/rtpmanager/async_jitter_queue.h: * gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_base_init), (gst_rtp_bin_class_init), (gst_rtp_bin_init), (gst_rtp_bin_finalize), (gst_rtp_bin_set_property), (gst_rtp_bin_get_property), (gst_rtp_bin_change_state), (gst_rtp_bin_request_new_pad), (gst_rtp_bin_release_pad): * gst/rtpmanager/gstrtpbin.h: * gst/rtpmanager/gstrtpclient.c: (new_pad), (create_stream), (free_stream), (find_stream_by_ssrc), (gst_rtp_client_base_init), (gst_rtp_client_class_init), (gst_rtp_client_init), (gst_rtp_client_finalize), (gst_rtp_client_set_property), (gst_rtp_client_get_property), (gst_rtp_client_change_state), (gst_rtp_client_request_new_pad), (gst_rtp_client_release_pad): * gst/rtpmanager/gstrtpclient.h: * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_rtp_jitter_buffer_base_init), (gst_rtp_jitter_buffer_class_init), (gst_rtp_jitter_buffer_init), (gst_rtp_jitter_buffer_dispose), (gst_rtp_jitter_buffer_getcaps), (gst_jitter_buffer_sink_setcaps), (free_func), (gst_rtp_jitter_buffer_flush_start), (gst_rtp_jitter_buffer_flush_stop), (gst_rtp_jitter_buffer_src_activate_push), (gst_rtp_jitter_buffer_change_state), (priv_compare_rtp_seq_lt), (compare_rtp_buffers_seq_num), (gst_rtp_jitter_buffer_sink_event), (gst_rtp_jitter_buffer_chain), (gst_rtp_jitter_buffer_loop), (gst_rtp_jitter_buffer_query), (gst_rtp_jitter_buffer_set_property), (gst_rtp_jitter_buffer_get_property): * gst/rtpmanager/gstrtpjitterbuffer.h: * gst/rtpmanager/gstrtpmanager.c: (plugin_init): * gst/rtpmanager/gstrtpptdemux.c: (gst_rtp_pt_demux_base_init), (gst_rtp_pt_demux_class_init), (gst_rtp_pt_demux_init), (gst_rtp_pt_demux_finalize), (gst_rtp_pt_demux_chain), (gst_rtp_pt_demux_getcaps), (find_pad_for_pt), (gst_rtp_pt_demux_setup), (gst_rtp_pt_demux_release), (gst_rtp_pt_demux_change_state): * gst/rtpmanager/gstrtpptdemux.h: * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_base_init), (gst_rtp_session_class_init), (gst_rtp_session_init), (gst_rtp_session_finalize), (gst_rtp_session_set_property), (gst_rtp_session_get_property), (gst_rtp_session_change_state), (gst_rtp_session_chain_recv_rtp), (gst_rtp_session_chain_recv_rtcp), (gst_rtp_session_chain_send_rtp), (create_recv_rtp_sink), (create_recv_rtcp_sink), (create_send_rtp_sink), (create_rtcp_src), (gst_rtp_session_request_new_pad), (gst_rtp_session_release_pad): * gst/rtpmanager/gstrtpsession.h: Add RTP session management elements. Still in progress.
2007-03-30ext/wavpack/: Don't play audioconvert. As wavpack wants/outputs all samples ↵Sebastian Dröge1-0/+35
with width==32 and depth=[1,32] accept th... Original commit message from CVS: * ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_reset), (gst_wavpack_dec_init), (gst_wavpack_dec_sink_set_caps), (gst_wavpack_dec_clip_outgoing_buffer), (gst_wavpack_dec_post_tags), (gst_wavpack_dec_chain): * ext/wavpack/gstwavpackdec.h: * ext/wavpack/gstwavpackenc.c: (gst_wavpack_enc_reset), (gst_wavpack_enc_sink_set_caps), (gst_wavpack_enc_set_wp_config), (gst_wavpack_enc_chain): * ext/wavpack/gstwavpackenc.h: * ext/wavpack/gstwavpackparse.c: Don't play audioconvert. As wavpack wants/outputs all samples with width==32 and depth=[1,32] accept this and let audioconvert convert to accepted formats instead of doing it in the element for n*8 depths. This also adds support for non-n*8 depths and prevents some useless memory allocations. Fixes #421598 Also add a workaround for bug #421542 in wavpackenc for now... * tests/check/elements/wavpackdec.c: (GST_START_TEST): * tests/check/elements/wavpackenc.c: (GST_START_TEST): * tests/check/elements/wavpackparse.c: (GST_START_TEST): Consider the change above in the unit tests and test if the correct caps are accepted and set. Also check for GST_BUFFER_OFFSET_END in the wavpackparse unit test. * ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_init), (gst_wavpack_dec_sink_set_caps): Set caps on the src pad as soon as possible. * ext/wavpack/gstwavpackdec.h: * ext/wavpack/gstwavpackcommon.h: * ext/wavpack/gstwavpackenc.h: * ext/wavpack/gstwavpackparse.h: Fix indention. gst-indent is now called by cicl.
2007-03-28gst/qtdemux/: Process 'ctts' atoms, which are present in AVC ISO files (.mov ↵Edward Hervey1-0/+14
files with h264 video). Original commit message from CVS: * gst/qtdemux/qtdemux.c: (gst_qtdemux_prepare_current_sample), (gst_qtdemux_chain), (qtdemux_parse_samples): * gst/qtdemux/qtdemux_dump.c: (qtdemux_dump_ctts): * gst/qtdemux/qtdemux_dump.h: * gst/qtdemux/qtdemux_fourcc.h: * gst/qtdemux/qtdemux_types.c: Process 'ctts' atoms, which are present in AVC ISO files (.mov files with h264 video). Use the offset present in 'ctts' to calculate the PTS for each packet and set the PTS on outgoing buffers. Fixes #423283
2007-03-27ext/xvid/gstxviddec.c: Add some debug log and fix a stupid output buffer ↵Julien Moutte1-0/+5
duration bug. Original commit message from CVS: 2007-03-27 Julien MOUTTE <julien@moutte.net> * ext/xvid/gstxviddec.c: (gst_xviddec_chain): Add some debug log and fix a stupid output buffer duration bug.
2007-03-25Add libx264-based h264 encoder plugin (#421110). Probably doesn't handle ↵Michal Benes1-0/+26
'odd' widths and heights correctly yet. Original commit message from CVS: Patch by: Michal Benes <michal.benes at itonis tv> Patch by: Josef Zlomek <josef.zlomek at itonis tv> * configure.ac: * ext/Makefile.am: * ext/x264/Makefile.am: * ext/x264/gstx264enc.c: (gst_x264_enc_me_get_type), (gst_x264_enc_analyse_get_type), (gst_x264_enc_timestamp_queue_init), (gst_x264_enc_timestamp_queue_free), (gst_x264_enc_timestamp_queue_put), (gst_x264_enc_timestamp_queue_get), (gst_x264_enc_header_buf), (gst_x264_enc_set_src_caps), (gst_x264_enc_sink_set_caps), (gst_x264_enc_base_init), (gst_x264_enc_class_init), (gst_x264_enc_init), (gst_x264_enc_init_encoder), (gst_x264_enc_close_encoder), (gst_x264_enc_dispose), (gst_x264_enc_sink_event), (gst_x264_enc_chain), (gst_x264_enc_encode_frame), (gst_x264_enc_change_state), (gst_x264_enc_set_property), (gst_x264_enc_get_property), (plugin_init): * ext/x264/gstx264enc.h: Add libx264-based h264 encoder plugin (#421110). Probably doesn't handle 'odd' widths and heights correctly yet.
2007-03-24gst/interleave/deinterleave.c: Remove 'channel-positions' field when munging ↵Tim-Philipp Müller1-0/+8
input caps into 1-channel output caps (I... Original commit message from CVS: * gst/interleave/deinterleave.c: (gst_deinterleave_sink_setcaps): Remove 'channel-positions' field when munging input caps into 1-channel output caps (I guess technically we should set the position for each channel on the output caps if it's non-NONE, but I'll save that as a task for another day).
2007-03-23gst/vmnc/vmncdec.c: Redesign to include a parser for raw files (no ↵Michael Smith1-0/+16
timestamps in that mode yet, though). Original commit message from CVS: * gst/vmnc/vmncdec.c: (gst_vmnc_dec_class_init), (gst_vmnc_dec_init), (vmnc_dec_finalize), (gst_vmnc_dec_reset), (vmnc_handle_wmvi_rectangle), (render_colour_cursor), (render_cursor), (vmnc_make_buffer), (vmnc_handle_wmvd_rectangle), (vmnc_handle_wmve_rectangle), (vmnc_handle_wmvf_rectangle), (vmnc_handle_wmvg_rectangle), (vmnc_handle_wmvh_rectangle), (vmnc_handle_wmvj_rectangle), (render_raw_tile), (render_subrect), (vmnc_handle_raw_rectangle), (vmnc_handle_copy_rectangle), (vmnc_handle_hextile_rectangle), (vmnc_handle_packet), (vmnc_dec_setcaps), (vmnc_dec_chain_frame), (vmnc_dec_chain), (vmnc_dec_set_property), (vmnc_dec_get_property): Redesign to include a parser for raw files (no timestamps in that mode yet, though).
2007-03-22gst/interleave/deinterleave.c: Don't leak input buffer in chain function; ↵Tim-Philipp Müller1-0/+10
maintain our own list of source pads - ther... Original commit message from CVS: * gst/interleave/deinterleave.c: (gst_deinterleave_add_new_pads), (gst_deinterleave_remove_pads), (gst_deinterleave_process), (gst_deinterleave_chain): Don't leak input buffer in chain function; maintain our own list of source pads - there are no guarantees about the order of the list in the GstElement struct, and we want a very specific order; lastly, some more debugging.
2007-03-22ext/neon/gstneonhttpsrc.c: Alloc user agent string only once.Tim-Philipp Müller1-0/+5
Original commit message from CVS: * ext/neon/gstneonhttpsrc.c: (gst_neonhttp_src_init): Alloc user agent string only once.
2007-03-22ext/wavpack/gstwavpackparse.c: Revert last commit, preventing infinite ↵Sebastian Dröge1-0/+7
plugging loops with ranks is no clean solution... Original commit message from CVS: * ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_plugin_init): Revert last commit, preventing infinite plugging loops with ranks is no clean solution and in general there's no reason why one wants to parse framed wavpack data again.
2007-03-22ext/wavpack/gstwavpackenc.c: Send the new segment event in time format ↵Sebastian Dröge1-0/+12
instead of bytes. This allows "wavpackenc ! wa... Original commit message from CVS: * ext/wavpack/gstwavpackenc.c: (gst_wavpack_enc_push_block): Send the new segment event in time format instead of bytes. This allows "wavpackenc ! wavpackdec ! someaudiosink" pipelines. * ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_plugin_init): Accept framed and non-framed input, wavpackparse doesn't care. To prevent "wavpackparse ! wavpackparse ! ..." pipelines lower the rank of wavpackparse by one. This allows "wavpackenc ! wavpackparse ! ..." pipelines.
2007-03-22gst-libs/gst/app/Makefile.am: Use GST_ALL_LDFLAGS, which actually exists, ↵Thomas Vander Stichele1-0/+6
but maybe David can confirm that was what h... Original commit message from CVS: * gst-libs/gst/app/Makefile.am: Use GST_ALL_LDFLAGS, which actually exists, but maybe David can confirm that was what he wanted.
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