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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:
* gst/qtdemux/Makefile.am:
* gst/qtdemux/gstrtpxqtdepay.c:
* gst/qtdemux/gstrtpxqtdepay.h:
* gst/qtdemux/qtdemux.c:
* gst/qtdemux/qtdemux.h:
* gst/qtdemux/qtdemux.vcproj:
* gst/qtdemux/qtdemux_dump.c:
* gst/qtdemux/qtdemux_dump.h:
* gst/qtdemux/qtdemux_fourcc.h:
* gst/qtdemux/qtdemux_types.c:
* gst/qtdemux/qtdemux_types.h:
* gst/qtdemux/qtpalette.h:
* gst/qtdemux/quicktime.c:
* win32/MANIFEST:
* win32/vs6/libgstqtdemux.dsp:
Move qtdemux to -good.
* gst-plugins-bad.spec.in:
Update spec file to reflect wavpack & qtdemux moving to -good.
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return from downstream.
Original commit message from CVS:
* ext/timidity/gsttimidity.c: (gst_timidity_loop):
* ext/timidity/gstwildmidi.c: (gst_wildmidi_loop):
* gst/tta/gstttaparse.c: (gst_tta_parse_loop):
When driving the pipeline, also post an error when we get a
not-linked flow return from downstream.
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Original commit message from CVS:
* gst/mpegvideoparse/mpegpacketiser.c:
(mpeg_util_parse_extension_packet), (mpeg_util_parse_sequence_hdr),
(mpeg_util_parse_picture_hdr):
* gst/mpegvideoparse/mpegvideoparse.c:
(mpegvideoparse_handle_sequence), (mpegvideoparse_drain_avail):
Fix some silly bugs with calculating the guard sizes.
Properly compare the old sequence header structure with the new one.
Don't error out on an invalid sequence - just ignore it.
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right number there.
Original commit message from CVS:
* gst/real/gstrealvideodec.c: (gst_real_video_dec_decode):
Printf fix in debug statement; also print the right number there.
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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.
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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.
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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.
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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.
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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.
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(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.
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(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.
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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.
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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).
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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).
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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.
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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).
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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.
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Original commit message from CVS:
* gst/rtpmanager/gstrtpptdemux.c: (gst_rtp_pt_demux_chain):
We always use fixed caps.
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Work around.
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c:
g_hash_table_remove_all() only exists in 2.12. Work around.
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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.
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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.
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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.
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Original commit message from CVS:
* gst/replaygain/rganalysis.c:
Fix wrong ifdef for visual C++. Fixes: #437403.
By Ali Sabil <ali.sabil@gmail.com>.
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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>).
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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>)
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Original commit message from CVS:
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_query):
Pass queries upstream.
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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.
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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.
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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.
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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.
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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.
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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.
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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.
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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.
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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.
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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.
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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.
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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.
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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.
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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.
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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.
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GST_PLUGINS_ALL).
Original commit message from CVS:
* gst/Makefile.am:
Fix distcheck, hopefully (rtpmanager is already in GST_PLUGINS_ALL).
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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.
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because we can.
Original commit message from CVS:
* gst/audioresample/gstaudioresample.c:
Make more functions static, just because we can.
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Original commit message from CVS:
* gst/Makefile.am:
Add rtpmanager dir to dist.
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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.
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Original commit message from CVS:
* gst/app/Makefile.am:
Fix CFLAGS and hopefully #430594.
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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.
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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).
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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.
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