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2007-06-01Add plugin to generate a pattern detectable by videodetect.Wim Taymans4-2/+454
Original commit message from CVS: * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-bad-plugins-docs.sgml: * docs/plugins/gst-plugins-bad-plugins-sections.txt: * docs/plugins/inspect/plugin-videosignal.xml: * gst/videosignal/Makefile.am: * gst/videosignal/gstvideomark.c: (gst_video_mark_set_caps), (gst_video_mark_draw_box), (gst_video_mark_420), (gst_video_mark_transform_ip), (gst_video_mark_set_property), (gst_video_mark_get_property), (gst_video_mark_base_init), (gst_video_mark_class_init), (gst_video_mark_init), (gst_video_mark_get_type): * gst/videosignal/gstvideomark.h: * gst/videosignal/gstvideosignal.c: (plugin_init): Add plugin to generate a pattern detectable by videodetect.
2007-05-31ext/libmms/gstmms.h: No reason to use gpointers instead of typed pointes ↵Tim-Philipp Müller1-2/+1
here as far as I can see. Original commit message from CVS: * ext/libmms/gstmms.h: No reason to use gpointers instead of typed pointes here as far as I can see. * ext/mythtv/gstmythtvsrc.c: * ext/neon/gstneonhttpsrc.c: * gst/switch/gstswitch.c: Don't use gtk-doc magic markers for things that aren't meant to be parsed by gtk-doc. Makes gtk-doc complain a bit less.
2007-05-30Added videosignal plugin with two plugins to analyse video frames.Wim Taymans7-0/+1105
Original commit message from CVS: * configure.ac: * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-bad-plugins-docs.sgml: * docs/plugins/gst-plugins-bad-plugins-sections.txt: * docs/plugins/inspect/plugin-videosignal.xml: * gst/videosignal/Makefile.am: * gst/videosignal/gstvideoanalyse.c: (gst_video_analyse_set_caps), (gst_video_analyse_post_message), (gst_video_analyse_420), (gst_video_analyse_transform_ip), (gst_video_analyse_set_property), (gst_video_analyse_get_property), (gst_video_analyse_base_init), (gst_video_analyse_class_init), (gst_video_analyse_init), (gst_video_analyse_get_type): * gst/videosignal/gstvideoanalyse.h: * gst/videosignal/gstvideodetect.c: (gst_video_detect_set_caps), (gst_video_detect_post_message), (gst_video_detect_calc_brightness), (gst_video_detect_420), (gst_video_detect_transform_ip), (gst_video_detect_set_property), (gst_video_detect_get_property), (gst_video_detect_base_init), (gst_video_detect_class_init), (gst_video_detect_init), (gst_video_detect_get_type): * gst/videosignal/gstvideodetect.h: * gst/videosignal/gstvideosignal.c: (plugin_init): * gst/videosignal/gstvideosignal.h: Added videosignal plugin with two plugins to analyse video frames. Added videoanalyse to report about brightness and variance in video frames. Added videodetect to detect predefined patterns in a video signal.
2007-05-28Rename elements to avoid conflict with farsight elements with the same name. ↵Wim Taymans7-93/+97
Fixes #430664. Original commit message from CVS: * docs/plugins/gst-plugins-bad-plugins-docs.sgml: * docs/plugins/gst-plugins-bad-plugins-sections.txt: * gst/rtpmanager/gstrtpbin.c: (create_session), (create_stream), (gst_rtp_bin_class_init), (create_recv_rtp), (create_recv_rtcp), (create_send_rtp), (create_rtcp), (gst_rtp_bin_request_new_pad): * gst/rtpmanager/gstrtpclient.c: (create_stream), (gst_rtp_client_request_new_pad): * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_rtp_jitter_buffer_clear_pt_map), (gst_rtp_jitter_buffer_loop): * gst/rtpmanager/gstrtpmanager.c: (plugin_init): * gst/rtpmanager/gstrtpptdemux.c: * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_class_init), (gst_rtp_session_request_new_pad): * gst/rtpmanager/gstrtpssrcdemux.c: Rename elements to avoid conflict with farsight elements with the same name. Fixes #430664.
2007-05-23Document stuff.Wim Taymans10-28/+334
Original commit message from CVS: * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-bad-plugins-docs.sgml: * docs/plugins/gst-plugins-bad-plugins-sections.txt: * gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_class_init): * gst/rtpmanager/gstrtpbin.h: * gst/rtpmanager/gstrtpclient.c: * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_rtp_jitter_buffer_class_init), (gst_rtp_jitter_buffer_clear_pt_map), (gst_rtp_jitter_buffer_loop): * gst/rtpmanager/gstrtpjitterbuffer.h: * gst/rtpmanager/gstrtpptdemux.c: (gst_rtp_pt_demux_class_init), (gst_rtp_pt_demux_clear_pt_map): * gst/rtpmanager/gstrtpptdemux.h: * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_class_init), (rtcp_thread), (gst_rtp_session_clear_pt_map): * gst/rtpmanager/gstrtpsession.h: * gst/rtpmanager/gstrtpssrcdemux.c: (gst_rtp_ssrc_demux_class_init): Document stuff. Add clear-pt-map action signal where needed.
2007-05-22configure.ac: Depend on gstreamer-0.10.12.1. gst/equalizer/gstiirequalizer.c ↵Stefan Kost5-31/+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.
2007-05-21ChangeLog: ChangeLog surgery. gst/equalizer/gstiirequalizer.c ↵Stefan Kost1-0/+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.
2007-05-21gst/spectrum/gstspectrum.c (gst_spectrum_set_property, gst_spectrum_event, ↵Stefan Kost1-1/+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.
2007-05-20gst/: Printf format fixes (#439910, #439911).Tim-Philipp Müller2-3/+4
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).
2007-05-19Add replaygain playback elements (#412710).René Stadler10-152/+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).
2007-05-17gst/switch/gstswitch.c (ARG_0, ARG_NB_SOURCES, ARG_ACTIVE_SOURCE,Zaheer Abbas Merali2-36/+146
Original commit message from CVS: * gst/switch/gstswitch.c (ARG_0, ARG_NB_SOURCES, ARG_ACTIVE_SOURCE, ARG_START_VALUE, ARG_STOP_VALUE, ARG_LAST_TS, ARG_QUEUE_BUFFERS, parent_class, gst_switch_release_pad, gst_switch_request_new_pad, gst_switch_chain, gst_switch_event, gst_switch_set_property, gst_switch_get_property, gst_switch_getcaps, gst_switch_dispose, unref_buffer, unref_buffers_and_destroy_list, gst_switch_init, gst_switch_base_init, gst_switch_class_init): * gst/switch/gstswitch.h (need_to_send_newsegment, queue_buffers, stop_value, start_value, current_start, last_ts, stored_buffers): Add handling of application provided stop and start values, allowing A/V sync across 2 switch elements.
2007-05-16gst/real/: Don't crash when we get a buffer and our input caps haven't been ↵Tim-Philipp Müller2-14/+50
set yet; also, don't leak all the input b... Original commit message from CVS: * gst/real/gstrealaudiodec.c: (gst_real_audio_dec_chain), (gst_real_audio_dec_setcaps): * gst/real/gstrealvideodec.c: (gst_real_video_dec_chain): Don't crash when we get a buffer and our input caps haven't been set yet; also, don't leak all the input buffers (realaudiodec only).
2007-05-15ext/x264/gstx264enc.c (gst_x264_enc_init_encoder): This needs a version check.Stefan Kost1-2/+3
Original commit message from CVS: * ext/x264/gstx264enc.c (gst_x264_enc_init_encoder): This needs a version check. * gst/bayer/Makefile.am: Fix the build.
2007-05-15gst/rtpmanager/gstrtpptdemux.c: We always use fixed caps.Wim Taymans1-0/+1
Original commit message from CVS: * gst/rtpmanager/gstrtpptdemux.c: (gst_rtp_pt_demux_chain): We always use fixed caps.
2007-05-15gst/rtpmanager/gstrtpbin.c: g_hash_table_remove_all() only exists in 2.12. ↵David Schleef1-0/+11
Work around. Original commit message from CVS: * gst/rtpmanager/gstrtpbin.c: g_hash_table_remove_all() only exists in 2.12. Work around.
2007-05-15Add a Bayer-to-RGB converter. You know you want one, uh-huh.David Schleef3-0/+422
Original commit message from CVS: * configure.ac: * gst/bayer/Makefile.am: * gst/bayer/gstbayer.c: * gst/bayer/gstbayer2rgb.c: Add a Bayer-to-RGB converter. You know you want one, uh-huh. Partial fix for #314160.
2007-05-14gst/switch/gstswitch.c (ARG_ACTIVE_SOURCE, ARG_STOP_VALUE,Zaheer Abbas Merali2-3/+88
Original commit message from CVS: * gst/switch/gstswitch.c (ARG_ACTIVE_SOURCE, ARG_STOP_VALUE, ARG_LAST_TS, parent_class, gst_switch_release_pad, gst_switch_request_new_pad, gst_switch_chain, gst_switch_event, gst_switch_set_property, gst_switch_get_property, gst_switch_getcaps, gst_switch_dispose, gst_switch_init, gst_switch_class_init): * gst/switch/gstswitch.h (previous_sinkpad, nb_sinkpads, stop_value, current_start, last_ts): Allow application to provide a stop timestamp, so a new segment update can be sent before switching.
2007-05-14gst/rtpmanager/async_jitter_queue.c: Fix leak when flushing.Wim Taymans4-4/+41
Original commit message from CVS: * gst/rtpmanager/async_jitter_queue.c: (async_jitter_queue_set_flushing_unlocked): Fix leak when flushing. * gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_clear_pt_map), (gst_rtp_bin_class_init): * gst/rtpmanager/gstrtpbin.h: Add clear-pt-map signal. * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_rtp_jitter_buffer_flush_stop), (gst_rtp_jitter_buffer_sink_event), (gst_rtp_jitter_buffer_loop): Init clock-rate to -1 to mark unknow clock rate. Fix flushing.
2007-05-13gst/replaygain/rganalysis.c: Fix wrong ifdef for visual C++. Fixes: #437403.David Schleef1-2/+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>.
2007-05-10gst/qtdemux/qtdemux.c (gst_qtdemux_move_stream, ↵Stefan Kost4-29/+40
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>).
2007-05-10gst/rtpmanager/async_jitter_queue.c (tail_buffer_duration, ↵Stefan Kost1-30/+30
async_jitter_queue_ref, async_jitter_queue_ref_unlocked, a... Original commit message from CVS: * gst/rtpmanager/async_jitter_queue.c (tail_buffer_duration, async_jitter_queue_ref, async_jitter_queue_ref_unlocked, async_jitter_queue_set_low_threshold, async_jitter_queue_length_ts_units_unlocked, async_jitter_queue_unref_and_unlock, async_jitter_queue_unref, async_jitter_queue_lock, async_jitter_queue_push, async_jitter_queue_push_unlocked, async_jitter_queue_push_sorted, async_jitter_queue_pop_intern_unlocked, async_jitter_queue_pop, async_jitter_queue_pop_unlocked, async_jitter_queue_length_unlocked, async_jitter_queue_set_flushing_unlocked, async_jitter_queue_unset_flushing_unlocked): Format arg fix (spotted by Ali Sabil <ali.sabil@gmail.com>)
2007-05-09gst/rtpmanager/gstrtpjitterbuffer.c: Pass queries upstream.Wim Taymans1-0/+1
Original commit message from CVS: * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_rtp_jitter_buffer_query): Pass queries upstream.
2007-05-06gst/real/: Use GModule instead of using dlsym() directly. Fixes #430598.Tim-Philipp Müller4-60/+85
Original commit message from CVS: * gst/real/gstrealaudiodec.c: (gst_real_audio_dec_setcaps), (gst_real_audio_dec_finalize): * gst/real/gstrealaudiodec.h: * gst/real/gstrealvideodec.c: (open_library), (close_library): * gst/real/gstrealvideodec.h: Use GModule instead of using dlsym() directly. Fixes #430598.
2007-05-04gst/speed/gstspeed.c: Fix event handling a bit by replacing completely ↵Tim-Philipp Müller1-88/+47
dubious code written by someone else with comp... Original commit message from CVS: * gst/speed/gstspeed.c: (speed_src_event), (speed_sink_event), (speed_chain), (speed_change_state): Fix event handling a bit by replacing completely dubious code written by someone else with completely dubious code written by me. Should at least fix #412077 though.
2007-05-04gst/speed/gstspeed.c: Add debug category; use gst_pad_query_peer_*() utility ↵Tim-Philipp Müller1-29/+51
functions; use gst_util_scale*(); add gt... Original commit message from CVS: * gst/speed/gstspeed.c: (speed_src_query), (speed_chain), (plugin_init): Add debug category; use gst_pad_query_peer_*() utility functions; use gst_util_scale*(); add gtk-doc blurb.
2007-05-04gst/rtpmanager/gstrtpjitterbuffer.c: Add some debug info.Wim Taymans2-2/+6
Original commit message from CVS: * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_rtp_jitter_buffer_query): Add some debug info. * gst/rtpmanager/rtpsession.c: (rtp_session_init), (rtp_session_send_rtp): Store real user name in the session.
2007-05-03examples/switch/switcher.c (loop, my_bus_callback, switch_timer, ↵Zaheer Abbas Merali2-241/+302
last_message_received, main): gst/switch/gstswitch.c... Original commit message from CVS: * configure.ac: * examples/Makefile.am: * examples/switch/switcher.c (loop, my_bus_callback, switch_timer, last_message_received, main): * gst/switch/gstswitch.c (GST_CAT_DEFAULT, gst_switch_details, gst_switch_src_factory, parent_class, gst_switch_release_pad, gst_switch_request_new_pad, gst_switch_chain, gst_switch_event, gst_switch_set_property, gst_switch_get_property, gst_switch_get_linked_pad, gst_switch_getcaps, gst_switch_bufferalloc, gst_switch_get_linked_pads, gst_switch_dispose, gst_switch_init, gst_switch_base_init, gst_switch_class_init): * gst/switch/gstswitch.h (GstSwitch, GstSwitchClass, _GstSwitch, element, active_sinkpad, srcpad, nb_sinkpads, newsegment_events, need_to_send_newsegment): Port switch element and example program to 0.10.
2007-04-30gst/rtpmanager/async_jitter_queue.c: Fix the case where the buffer underruns ↵Wim Taymans7-43/+116
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 Taymans3-11/+14
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.
2007-04-29autogen.sh: Require automake 1.7Thomas Vander Stichele1-3/+2
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.
2007-04-27ext/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 Taymans6-181/+656
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.
2007-04-25gst/rtpmanager/gstrtpsession.c: Implement forward and reverse reconsideration.Wim Taymans3-23/+79
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 Taymans3-16/+36
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 Taymans6-42/+161
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 Taymans8-86/+625
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 Müller1-10/+11
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 Müller1-1/+1
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 Müller1-2/+2
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.
2007-04-18gst/Makefile.am: Add rtpmanager dir to dist.Wim Taymans1-1/+1
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 Taymans11-35/+2477
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 Müller1-3/+1
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 Müller1-11/+23
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 Müller1-1/+1
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.
2007-04-13gst/rtpmanager/: Protect lists and structures with locks.Wim Taymans4-12/+71
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 Taymans1-1/+1
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 Taymans4-91/+162
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 Taymans7-27/+191
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.
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