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2007-05-02ext/wavpack/gstwavpack.c: Call bindtextdomain() to get localized strings.Sebastian Dröge5-2/+44
Original commit message from CVS: * ext/wavpack/gstwavpack.c: (plugin_init): Call bindtextdomain() to get localized strings. * ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_chain): * ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_reset), (gst_wavpack_parse_handle_seek_event), (gst_wavpack_parse_push_buffer), (gst_wavpack_parse_chain): * ext/wavpack/gstwavpackparse.h: Handle DISCONT buffers by correctly setting the DISCONT flag on outgoing buffers when necessary. * ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_handle_seek_event) Send newsegment from the streaming thread.
2007-05-02ext/wavpack/gstwavpackparse.c: Remove old workaround that was needed when ↵Sebastian Dröge2-5/+8
seeking after the last sample. With the fix... Original commit message from CVS: * ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_handle_seek_event): Remove old workaround that was needed when seeking after the last sample. With the fixed error handling this works now as expected without pushing the last sample although it wasn't requested.
2007-05-02ext/wavpack/gstwavpackparse.c: Handle segment seeks in the seek event ↵Sebastian Dröge2-1/+21
handler, correctly work with stop position == -... Original commit message from CVS: * ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_handle_seek_event): Handle segment seeks in the seek event handler, correctly work with stop position == -1 and instead of stopping the task on seek just pause it.
2007-05-02ext/wavpack/gstwavpackparse.c: Add handling for segment seeks.Sebastian Dröge2-4/+23
Original commit message from CVS: * ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_loop): Add handling for segment seeks.
2007-05-02ext/wavpack/gstwavpackparse.c: Correctly handle errors, especially in the ↵Sebastian Dröge2-26/+38
loop function. Before it was easy to get th... Original commit message from CVS: * ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_pull_buffer), (gst_wavpack_parse_create_src_pad), (gst_wavpack_parse_resync_loop), (gst_wavpack_parse_loop), (gst_wavpack_parse_chain): Correctly handle errors, especially in the loop function. Before it was easy to get the task paused but no error being posted on the bus.
2007-04-30update specChristian Schaller1-1/+1
Original commit message from CVS: update spec
2007-04-30gst/rtpmanager/async_jitter_queue.c: Fix the case where the buffer underruns ↵Wim Taymans8-43/+151
and does not block. Original commit message from CVS: * gst/rtpmanager/async_jitter_queue.c: (signal_waiting_threads), (async_jitter_queue_pop_intern_unlocked): Fix the case where the buffer underruns and does not block. * gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_base_init), (create_recv_rtcp), (create_send_rtp), (create_rtcp), (gst_rtp_bin_request_new_pad): Rename RTCP send pad, like in the session manager. Allow getting an RTCP pad for receiving even if we don't receive RTP. fix handling of send_rtp_src pad. * gst/rtpmanager/gstrtpptdemux.c: (gst_rtp_pt_demux_chain): When no pt map could be found, fall back to the sinkpad caps. * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_process_rtp), (gst_rtp_session_send_rtp), (create_recv_rtp_sink), (create_recv_rtcp_sink), (create_send_rtp_sink), (create_send_rtcp_src): Fix pad names. * gst/rtpmanager/rtpsession.c: (source_push_rtp), (rtp_session_create_source), (rtp_session_process_sr), (rtp_session_send_rtp), (session_start_rtcp): * gst/rtpmanager/rtpsession.h: Unlock session when performing a callback. Add callbacks for the internal session object. Fix sending of RTP packets. first attempt at adding NTP times in the SR packets. Small debug and doc improvements. * gst/rtpmanager/rtpsource.c: (rtp_source_send_rtp): Update stats for SR reports.
2007-04-29gst/rtpmanager/gstrtpsession.c: Remove debug.Wim Taymans4-11/+26
Original commit message from CVS: * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_send_rtcp): Remove debug. * gst/rtpmanager/rtpsession.c: (rtp_session_process_sr), (rtp_session_process_sdes), (calculate_rtcp_interval), (rtp_session_next_timeout), (session_report_blocks): * gst/rtpmanager/rtpstats.c: (rtp_stats_calculate_rtcp_interval): Improve debugging Fix interval for BYE/RTCP packets.
2007-04-29docs/plugins/gst-plugins-bad-plugins.*: Commit result of running scanobj-updateThomas Vander Stichele3-13/+728
Original commit message from CVS: * docs/plugins/gst-plugins-bad-plugins.args: * docs/plugins/gst-plugins-bad-plugins.signals: Commit result of running scanobj-update
2007-04-2980 char policeThomas Vander Stichele2-6/+6
Original commit message from CVS: 80 char police
2007-04-29autogen.sh: Require automake 1.7Thomas Vander Stichele13-50/+56
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 Stichele30-55/+107
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 Müller2-3/+19
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.
2007-04-27ext/theora/theoradec.c: Calculate buffer duration correctly to generate a ↵Julien Moutte1-3/+1
perfect stream (#433888). Original commit message from CVS: 2007-04-27 Julien MOUTTE <julien@moutte.net> * ext/theora/theoradec.c: (_theora_granule_time), (theora_dec_push_forward), (theora_handle_data_packet), (theora_dec_decode_buffer): Calculate buffer duration correctly to generate a perfect stream (#433888). * gst/audioresample/gstaudioresample.c: (audioresample_check_discont): Glib provides ABS.
2007-04-27gst/rtpmanager/gstrtpsession.c: Move reconsideration code to the rtpsession ↵Wim Taymans7-181/+690
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 Hervey5-0/+39
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 Taymans4-23/+90
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 Taymans4-16/+53
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 Taymans7-42/+183
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 Taymans9-86/+665
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 Müller4-11/+19
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 Müller2-1/+6
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 Müller2-2/+7
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.
2007-04-21gst/audioresample/gstaudioresample.c: Make more functions static, just ↵Tim-Philipp Müller1-9/+9
because we can. Original commit message from CVS: * gst/audioresample/gstaudioresample.c: Make more functions static, just because we can.
2007-04-21tests/check/elements/audioresample.c: Add unit test for audioresample ↵Tim-Philipp Müller1-17/+48
shutdown crasher (#420106). Original commit message from CVS: * tests/check/elements/audioresample.c: Add unit test for audioresample shutdown crasher (#420106).
2007-04-20ext/faad/gstfaad.c: FAAD fails to decode low (e.g. 8 kHz) sample rate AAC ↵Michael Smith2-1/+9
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 Schleef3-3/+11
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 Schleef2-1/+12
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 Taymans2-1/+6
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 Taymans14-36/+2555
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 Müller2-3/+6
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 Müller2-11/+31
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 Müller2-1/+8
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).
2007-04-16ext/pango/gstclockoverlay.c: Fix unused variable warning if HAVE_LOCALTIME_R ↵Vincent Torri1-1/+1
is undefinied Original commit message from CVS: Patch by: Vincent Torri <vtorri at univ-evry dot fr> * ext/pango/gstclockoverlay.c: (gst_clock_overlay_render_time): Fix unused variable warning if HAVE_LOCALTIME_R is undefinied * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps): * gst/audioresample/gstaudioresample.c: (audioresample_do_output): Use the correct format strings for integer formats.
2007-04-14docs/plugins/inspect/: Add xml doc files for Windows sinksSébastien Moutte7-0/+219
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 Taymans5-12/+87
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 Taymans2-1/+7
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 Taymans5-91/+178
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 Taymans8-27/+216
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 Taymans3-21/+61
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 Taymans9-9/+58
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 Taymans3-0/+22
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 Taymans2-1/+6
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 Taymans2-6/+23
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 Taymans3-1/+32
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 Taymans8-56/+487
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.
2007-04-05update dutchThomas Vander Stichele1-208/+17
Original commit message from CVS: update dutch
2007-04-05po/: Added Danish translation.Thomas Vander Stichele3-1/+53
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 Taymans5-0/+373
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 Kost2-7/+22
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.
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