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that we can generate better
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (calc_ntp_ns_base),
(gst_rtp_bin_change_state), (new_payload_found), (create_send_rtp):
Calculate and configure the NTP base time so that we can generate better
NTP times in SR packets.
Set caps on new ghostpad.
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_loop):
Clean debug statement.
* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_class_init),
(gst_rtp_session_init), (gst_rtp_session_set_property),
(gst_rtp_session_get_property), (get_current_ntp_ns_time),
(rtcp_thread), (gst_rtp_session_event_recv_rtp_sink),
(gst_rtp_session_internal_links), (gst_rtp_session_chain_recv_rtp),
(gst_rtp_session_event_send_rtp_sink),
(gst_rtp_session_chain_send_rtp), (create_recv_rtp_sink),
(create_send_rtp_sink):
* gst/rtpmanager/gstrtpsession.h:
Add ntp-ns-base property to convert running_time to NTP time.
Handle NEWSEGMENT events on send and recv RTP pads so that we can
calculate the running time and thus NTP time of the packets.
Simplify getting the current NTP time using the pipeline clock.
Implement internal links functions.
Use the buffer timestamp to calculate the NTP time instead of the clock.
* gst/rtpmanager/gstrtpssrcdemux.c: (create_demux_pad_for_ssrc),
(gst_rtp_ssrc_demux_init), (gst_rtp_ssrc_demux_sink_event),
(gst_rtp_ssrc_demux_chain), (gst_rtp_ssrc_demux_rtcp_chain),
(gst_rtp_ssrc_demux_internal_links),
(gst_rtp_ssrc_demux_src_query):
* gst/rtpmanager/gstrtpssrcdemux.h:
Implement internal links function.
Calculate the diff between different streams, this might be used later
to get the inter stream latency.
* gst/rtpmanager/rtpsession.c: (rtp_session_send_rtp):
Simple cleanup.
* gst/rtpmanager/rtpsource.c: (rtp_source_init),
(calculate_jitter), (rtp_source_send_rtp), (rtp_source_get_new_sr):
Make the clock skew window a little bigger.
Apply the clock skew to all buffers, not just one with a new timestamp.
Calculate and debug sender clock drift.
Use extended last timestamp to interpollate for SR reports.
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Original commit message from CVS:
Patch by: Peter Kjellerstedt <pkj at axis com>
* gst-libs/gst/app/gstappsink.c:
* gst/flv/gstflvdemux.c:
* gst/flv/gstflvparse.c:
* gst/interleave/deinterleave.c:
* gst/switch/gstswitch.c:
Printf format fixes (#476128).
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calls.
Original commit message from CVS:
* gst/spectrum/gstspectrum.c: (gst_spectrum_start),
(gst_spectrum_transform_ip):
Use the correct parameter order for the memset calls.
Thanks to Christian Schaller for noticing.
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Original commit message from CVS:
* gst/mpegtsparse/mpegtsparse.c:
Fix the build (missing stdlib.h).
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Original commit message from CVS:
* gst/spectrum/fix_fft.c:
Remove fixed point FFT as it's not used anymore.
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float and double, use floats for the message...
Original commit message from CVS:
* configure.ac:
* gst/spectrum/Makefile.am:
* gst/spectrum/demo-audiotest.c: (draw_spectrum),
(message_handler), (main):
* gst/spectrum/demo-osssrc.c: (draw_spectrum), (message_handler):
* gst/spectrum/gstspectrum.c: (gst_spectrum_base_init),
(gst_spectrum_class_init), (gst_spectrum_init),
(gst_spectrum_dispose), (gst_spectrum_set_property),
(gst_spectrum_get_property), (gst_spectrum_start),
(gst_spectrum_setup), (gst_spectrum_message_new),
(gst_spectrum_transform_ip):
* gst/spectrum/gstspectrum.h:
Port GstSpectrum to GstAudioFilter and libgstfft, add support
for int32, float and double, use floats for the message contents,
average all FFTs done in one interval for better results, use
a better windowing function, allow posting the phase in the message
and actually do an FFT with the requested number of bands instead
of interpolating.
* tests/check/elements/spectrum.c: (GST_START_TEST),
(spectrum_suite):
Improve the units tests by checking for a 11025Hz sine wave
and add unit tests for all 4 supported sample types.
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Original commit message from CVS:
* gst/real/gstrealvideodec.c: (gst_real_video_dec_chain),
(gst_real_video_dec_setcaps):
Add some more debugging.
Don't set LONG for width/height in caps.
Set correct output buffer size when caps changed.
The custom message sent to the decoder should not include the format and
subformat. Fixes #471554.
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-Wall -Werror (#473562).
Original commit message from CVS:
* gst/rtpmanager/gstrtpsession.c:
Make compiler happy: fix compilation with -Wall -Werror
(#473562).
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Original commit message from CVS:
Nosefart -> NES Sound Format
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Original commit message from CVS:
2007-09-03 Johan Dahlin <johan@gnome.org>
* gst/nsf/gstnsf.c: (gst_nsfdec_finalize), (start_play_tune):
* gst/nsf/gstnsf.h:
Add support for (very) basic tagging.
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Original commit message from CVS:
* gst/rtpmanager/gstrtpbin-marshal.list:
* gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_get_client),
(gst_rtp_bin_associate), (gst_rtp_bin_sync_chain), (create_stream),
(gst_rtp_bin_init), (caps_changed), (new_ssrc_pad_found),
(create_recv_rtp), (create_recv_rtcp), (create_send_rtp):
* gst/rtpmanager/gstrtpbin.h:
Updated example pipelines in docs.
Handle sync_rtcp buffers from the SSRC demuxer to perform lip-sync.
Set the default latency correctly.
Add some more points where we can get caps.
* 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_query),
(gst_rtp_jitter_buffer_set_property),
(gst_rtp_jitter_buffer_get_property):
Add ts-offset property to control timestamping.
* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_class_init),
(gst_rtp_session_init), (gst_rtp_session_set_property),
(gst_rtp_session_get_property), (get_current_ntp_ns_time),
(rtcp_thread), (stop_rtcp_thread), (gst_rtp_session_change_state),
(gst_rtp_session_send_rtcp), (gst_rtp_session_sync_rtcp),
(gst_rtp_session_cache_caps), (gst_rtp_session_clock_rate),
(gst_rtp_session_sink_setcaps), (gst_rtp_session_chain_recv_rtp),
(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_send_rtcp_src):
Various cleanups.
Feed rtpsession manager with NTP time based on pipeline clock when
handling RTP packets and RTCP timeouts.
Perform all RTCP with the system clock.
Set caps on RTCP outgoing buffers.
* gst/rtpmanager/gstrtpssrcdemux.c: (find_demux_pad_for_ssrc),
(create_demux_pad_for_ssrc), (gst_rtp_ssrc_demux_base_init),
(gst_rtp_ssrc_demux_init), (gst_rtp_ssrc_demux_sink_event),
(gst_rtp_ssrc_demux_rtcp_sink_event), (gst_rtp_ssrc_demux_chain),
(gst_rtp_ssrc_demux_rtcp_chain):
* gst/rtpmanager/gstrtpssrcdemux.h:
Also demux RTCP messages.
* gst/rtpmanager/rtpsession.c: (rtp_session_set_callbacks),
(update_arrival_stats), (rtp_session_process_rtp),
(rtp_session_process_rb), (rtp_session_process_sr),
(rtp_session_process_rr), (rtp_session_process_rtcp),
(rtp_session_send_rtp), (rtp_session_send_bye),
(session_start_rtcp), (session_report_blocks), (session_cleanup),
(rtp_session_on_timeout):
* gst/rtpmanager/rtpsession.h:
Remove the get_time callback, the GStreamer part will feed us with
enough timing information.
Split sync timing and RTCP timing information.
Factor out common RB handling for SR and RR.
Send out SR RTCP packets for lip-sync.
Move SR and RR packet info generation to the source.
* gst/rtpmanager/rtpsource.c: (rtp_source_init),
(rtp_source_update_caps), (get_clock_rate), (calculate_jitter),
(rtp_source_process_rtp), (rtp_source_send_rtp),
(rtp_source_process_sr), (rtp_source_process_rb),
(rtp_source_get_new_sr), (rtp_source_get_new_rb),
(rtp_source_get_last_sr):
* gst/rtpmanager/rtpsource.h:
* gst/rtpmanager/rtpstats.h:
Use caps on incomming buffers to get timing information when they are
there.
Calculate clock scew of the receiver compared to the sender and adjust
the rtp timestamps.
Calculate the round trip in sources.
Do SR and RR calculations in the source.
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buffers from the jitterbuffer so that we can h...
Original commit message from CVS:
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_flush_stop),
(gst_rtp_jitter_buffer_change_state), (gst_rtp_jitter_buffer_loop):
Use extended timestamp to release buffers from the jitterbuffer so that
we can handle the rtp wraparound correctly.
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Original commit message from CVS:
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_loop):
Improve Comments.
* gst/rtpmanager/gstrtpsession.c: (stop_rtcp_thread),
(gst_rtp_session_change_state), (gst_rtp_session_parse_caps),
(gst_rtp_session_clock_rate), (gst_rtp_session_sink_setcaps),
(gst_rtp_session_event_send_rtp_sink), (create_recv_rtp_sink),
(create_send_rtp_sink):
Also parse the sink caps for clock-rate instead of only relying on the
result of the signal.
* gst/rtpmanager/rtpsource.c: (rtp_source_send_rtp):
Make sure we fetch the clock rate for payloads we are sending out so
that we can use it for SR reports.
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information is known at time of setting st...
Original commit message from CVS:
* gst/switch/gstswitch.c (gst_switch_chain, gst_switch_set_property):
If all information is known at time of setting start-time
property, send new segments then.
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session manager so that it can generate ...
Original commit message from CVS:
* gst/rtpmanager/gstrtpsession.c: (stop_rtcp_thread),
(gst_rtp_session_change_state),
(gst_rtp_session_event_send_rtp_sink):
* gst/rtpmanager/gstrtpsession.h:
Distribute synchronisation parameters to the session manager so that it
can generate correct SR packets for lip-sync.
* gst/rtpmanager/rtpsession.c: (rtp_session_set_base_time),
(rtp_session_set_timestamp_sync), (session_start_rtcp):
* gst/rtpmanager/rtpsession.h:
Add methods for setting sync parameters.
Set correct RTP time in SR packets using the sync params.
* gst/rtpmanager/rtpsource.c: (rtp_source_send_rtp):
* gst/rtpmanager/rtpsource.h:
Record last RTP <-> GST timestamp so that we can use them to convert NTP
to RTP timestamps in SR packets.
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Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_clear_pt_map):
Add some more advanced example pipelines.
* gst/rtpmanager/gstrtpsession.c: (rtcp_thread),
(stop_rtcp_thread), (gst_rtp_session_send_rtcp):
Add some debug and FIXME.
Release LOCK when performing session cleanup.
* gst/rtpmanager/rtpsession.c: (session_report_blocks):
Add some debug.
* gst/rtpmanager/rtpsource.c: (calculate_jitter),
(rtp_source_send_rtp):
Make sure we always send RTP packets with the session SSRC.
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uninstalled setup).
Original commit message from CVS:
* gst/dvdspu/gstdvdspu.c:
Don't need this include (fixes compilation in uninstalled setup).
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latency into account.
Original commit message from CVS:
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_change_state), (gst_rtp_jitter_buffer_loop),
(gst_rtp_jitter_buffer_query):
When synchronizing buffers, take peer latency into account.
Don't try to add our latency to invalid peer max latency values.
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Original commit message from CVS:
2007-08-27 Julien MOUTTE <julien@moutte.net>
* gst/flv/gstflvdemux.c: (gst_flv_demux_handle_seek_pull):
Make sure we initialize the seek result.
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Original commit message from CVS:
* gst/dvdspu/Makefile.am:
Commit the makefile too.
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Original commit message from CVS:
* configure.ac:
* gst/dvdspu/.cvsignore:
* gst/dvdspu/Notes.txt:
* gst/dvdspu/gstdvdspu-render.c: (dvdspu_recalc_palette),
(dvdspu_update_palettes), (dvdspu_clear_comp_buffers),
(dvdspu_get_nibble), (dvdspu_get_rle_code), (dvdspu_draw_rle_run),
(rle_end_x), (dvdspu_render_line), (dvdspu_update_chgcol),
(dvdspu_render_line_with_chgcol), (dvdspu_blend_comp_buffers),
(gstdvdspu_render_spu):
* gst/dvdspu/gstdvdspu.c: (dvdspu_base_init), (dvdspu_class_init),
(dvdspu_init), (dvdspu_clear), (dvdspu_dispose), (dvdspu_finalize),
(dvdspu_flush_spu_info), (dvdspu_buffer_alloc), (dvdspu_src_event),
(dvdspu_video_set_caps), (dvdspu_video_proxy_getcaps),
(dvdspu_video_event), (dvdspu_video_chain),
(dvspu_handle_vid_buffer), (dvdspu_redraw_still),
(gstdvdspu_parse_chg_colcon), (dvdspu_exec_cmd_blk),
(dvdspu_finish_spu_buf), (dvdspu_setup_cmd_blk),
(dvdspu_handle_new_spu_buf), (dvdspu_handle_dvd_event),
(dvdspu_dump_dcsq), (dvdspu_advance_spu),
(dvdspu_check_still_updates), (dvdspu_subpic_chain),
(dvdspu_subpic_event), (dvdspu_change_state),
(gstdvdspu_plugin_init):
* gst/dvdspu/gstdvdspu.h:
dvdspu element donated by Fluendo. It implements a DVD Sub-Picture
Unit, decoding and overlaying DVD subtitles and menu graphics.
* gst/mpeg2sub/.cvsignore:
* gst/mpeg2sub/Makefile.am:
* gst/mpeg2sub/Notes.txt:
* gst/mpeg2sub/gstmpeg2subt.c:
* gst/mpeg2sub/gstmpeg2subt.h:
* gst/mpeg2sub/mpeg2subt.vcproj:
Delete old and broken mpeg2subt element that was never ported from 0.8
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Original commit message from CVS:
2007-08-24 Julien MOUTTE <julien@moutte.net>
* gst/flv/gstflvdemux.c: (gst_flv_demux_flush),
(gst_flv_demux_chain), (gst_flv_demux_pull_tag),
(gst_flv_demux_find_offset), (gst_flv_demux_handle_seek_push),
(gst_flv_demux_handle_seek_pull), (gst_flv_demux_sink_event),
(gst_flv_demux_src_event): Remove some useless ifdef.
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Original commit message from CVS:
* examples/switch/switcher.c (main):
* gst/switch/gstswitch.c (gst_switch_chain):
Make switch more reliable and also not lock up when
sink pad caps change.
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Original commit message from CVS:
2007-08-24 Julien MOUTTE <julien@moutte.net>
* gst/flv/gstflvdemux.c: (gst_flv_demux_flush),
(gst_flv_demux_cleanup), (gst_flv_demux_chain),
(gst_flv_demux_pull_tag), (gst_flv_demux_find_offset),
(gst_flv_demux_handle_seek_push),
(gst_flv_demux_handle_seek_pull),
(gst_flv_demux_sink_event), (gst_flv_demux_src_event): Implement
seeking in push mode.
* gst/flv/gstflvdemux.h:
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Original commit message from CVS:
* gst/mpegtsparse/flutspatinfo.c:
* gst/mpegtsparse/flutspatinfo.h:
* gst/mpegtsparse/flutspmtinfo.c:
* gst/mpegtsparse/flutspmtinfo.h:
Update licences to reflect LGPL-ness of these files also.
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a GType that's different than the GstRTPF...
Original commit message from CVS:
* docs/plugins/gst-plugins-bad-plugins-sections.txt:
* docs/plugins/gst-plugins-bad-plugins.hierarchy:
* docs/plugins/gst-plugins-bad-plugins.interfaces:
* docs/plugins/gst-plugins-bad-plugins.signals:
* gst/rtpmanager/gstrtpbin.c:
* gst/rtpmanager/gstrtpbin.h:
* gst/rtpmanager/gstrtpclient.c:
* gst/rtpmanager/gstrtpclient.h:
* gst/rtpmanager/gstrtpjitterbuffer.c:
* gst/rtpmanager/gstrtpjitterbuffer.h:
* gst/rtpmanager/gstrtpptdemux.c:
* gst/rtpmanager/gstrtpptdemux.h:
* gst/rtpmanager/gstrtpsession.c:
* gst/rtpmanager/gstrtpsession.h:
* gst/rtpmanager/gstrtpssrcdemux.c:
* gst/rtpmanager/gstrtpssrcdemux.h:
Rename all GstRTPFoo structs to GstRtpFoo so that GST_BOILERPLATE
registers a GType that's different than the GstRTPFoo types that
farsight registers (luckily GType names are case sensitive). Should
finally fix #430664.
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Original commit message from CVS:
Patch by: Jonathan Matthew <jonathan at kaolin wh9 net>
* configure.ac:
* gst/festival/Makefile.am:
* gst/festival/gstfestival.c:
Port festival plugin to GStreamer-0.10 (#461377).
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Fluendo muxer supports this and
Original commit message from CVS:
2007-08-22 Julien MOUTTE <julien@moutte.net>
* gst/flv/gstflvdemux.c: (gst_flv_demux_cleanup),
(gst_flv_demux_pull_tag):
* gst/flv/gstflvdemux.h:
* gst/flv/gstflvparse.c: (gst_flv_parse_metadata_item),
(gst_flv_parse_tag_script), (gst_flv_parse_tag_audio),
(gst_flv_parse_tag_video): Handle pixel aspect ratio through
metadata tags like ASF does. Fluendo muxer supports this and
Flash players can support it as well this way.
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parsed. Fix the way we decide to push ta...
Original commit message from CVS:
2007-08-22 Julien MOUTTE <julien@moutte.net>
* gst/flv/gstflvdemux.c: (gst_flv_demux_pull_tag):
* gst/flv/gstflvparse.c: (gst_flv_parse_metadata_item),
(gst_flv_parse_tag_script), (gst_flv_parse_tag_audio),
(gst_flv_parse_tag_video): Make sure we don't try filling up the
index if no times object was parsed. Fix the way we decide to
push
tags and emit no-more-pads. Fix some printf typing in debugging.
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no latency configured, just push the buf...
Original commit message from CVS:
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_chain),
(gst_rtp_jitter_buffer_set_property):
When drop-on-latency is set but we have no latency configured, just push
the buffer as fast as possible.
Fix typo in comment.
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Original commit message from CVS:
* gst/rtpmanager/rtpjitterbuffer.c:
(rtp_jitter_buffer_get_ts_diff):
* gst/rtpmanager/rtpjitterbuffer.h:
Fix undefined overflow prone ts_diff handling.
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Original commit message from CVS:
* gst/modplug/gstmodplug.cc:
Fix compiler warning.
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Original commit message from CVS:
* gst/modplug/gstmodplug.cc:
Don't use NULL caps for srcpad is not yet connected.
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Original commit message from CVS:
* configure.ac:
* gst/mpegtsparse/Makefile.am:
* gst/mpegtsparse/flutspatinfo.c:
* gst/mpegtsparse/flutspatinfo.h:
* gst/mpegtsparse/flutspmtinfo.c:
* gst/mpegtsparse/flutspmtinfo.h:
* gst/mpegtsparse/flutspmtstreaminfo.c:
* gst/mpegtsparse/flutspmtstreaminfo.h:
* gst/mpegtsparse/mpegtspacketizer.c:
* gst/mpegtsparse/mpegtspacketizer.h:
* gst/mpegtsparse/mpegtsparse.c:
* gst/mpegtsparse/mpegtsparse.h:
* gst/mpegtsparse/mpegtsparsemarshal.list:
Add mpeg transport stream parser written by:
Alessandro Decina. Includes a couple of files from the
Fluendo transport stream demuxer that Fluendo have
kindly allowed to be licenced under LGPL also.
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Original commit message from CVS:
* gst/filter/gstbpwsinc.c:
* gst/filter/gstlpwsinc.c:
Add small comparision with the chebyshev filters in the docs.
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types, add lower upper boundaries for the GOb...
Original commit message from CVS:
* gst/filter/gstbpwsinc.c: (gst_bpwsinc_class_init),
(bpwsinc_set_property), (bpwsinc_get_property):
* gst/filter/gstbpwsinc.h:
* gst/filter/gstlpwsinc.c: (gst_lpwsinc_class_init),
(gst_lpwsinc_init), (lpwsinc_build_kernel), (lpwsinc_set_property),
(lpwsinc_get_property):
* gst/filter/gstlpwsinc.h:
* tests/check/elements/lpwsinc.c: (GST_START_TEST):
Use generator macros for the process functions for the different
sample types, add lower upper boundaries for the GObject properties
so automatically generated UIs can use sliders and change frequency
properties to floats to save a bit of memory, even ints would in
theory be enough. Also rename frequency to cutoff for consistency
reasons.
* docs/plugins/gst-plugins-bad-plugins.args:
* docs/plugins/gst-plugins-bad-plugins.signals:
* docs/plugins/inspect/plugin-gstrtpmanager.xml:
Regenerated for the above changes.
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samples downstream that actually contain the ...
Original commit message from CVS:
* gst/filter/gstbpwsinc.c: (gst_bpwsinc_class_init),
(gst_bpwsinc_init), (process_32), (process_64),
(bpwsinc_build_kernel), (bpwsinc_push_residue),
(bpwsinc_transform), (bpwsinc_start), (bpwsinc_query),
(bpwsinc_query_type), (bpwsinc_event), (bpwsinc_set_property):
* gst/filter/gstbpwsinc.h:
Implement latency query and only forward those samples downstream
that actually contain the data we want, i.e. drop kernel_length/2
in the beginning and append kernel_length/2 (created by convolving
the filter kernel with zeroes) to the end.
* tests/check/elements/bpwsinc.c: (GST_START_TEST):
Adjust the unit test for this slightly changed behaviour.
* gst/filter/gstlpwsinc.c: (lpwsinc_build_kernel):
Reset residue length only when actually creating a residue.
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Original commit message from CVS:
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_sink_event), (gst_rtp_jitter_buffer_chain),
(gst_rtp_jitter_buffer_loop):
Fix EOS handling.
Convert some DEBUG into WARNINGs.
Pause task when flushing.
* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_class_init),
(rtcp_thread), (gst_rtp_session_event_recv_rtcp_sink):
Use system clock for RTCP session management timeouts.
* gst/rtpmanager/rtpsession.c: (on_new_ssrc), (on_ssrc_collision),
(on_ssrc_validated), (on_bye_ssrc), (on_bye_timeout), (on_timeout):
Release the session lock when emiting signals.
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samples downstream that actually contain the ...
Original commit message from CVS:
* gst/filter/gstlpwsinc.c: (gst_lpwsinc_class_init),
(gst_lpwsinc_init), (process_32), (process_64),
(lpwsinc_build_kernel), (lpwsinc_push_residue),
(lpwsinc_transform), (lpwsinc_start), (lpwsinc_query),
(lpwsinc_query_type), (lpwsinc_event), (lpwsinc_set_property):
* gst/filter/gstlpwsinc.h:
Implement latency query and only forward those samples downstream
that actually contain the data we want, i.e. drop kernel_length/2
in the beginning and append kernel_length/2 (created by convolving
the filter kernel with zeroes) to the end.
* tests/check/elements/lpwsinc.c: (GST_START_TEST):
Adjust the unit test for this slightly changed behaviour.
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Original commit message from CVS:
* gst/flv/gstflvdemux.c: (gst_flv_demux_set_index),
(gst_flv_demux_get_index):
Fix locking and refcounting on the index.
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built step by step or coming from metadata.
Original commit message from CVS:
2007-08-14 Julien MOUTTE <julien@moutte.net>
* gst/flv/gstflvdemux.c: (gst_flv_demux_cleanup),
(gst_flv_demux_adapter_flush), (gst_flv_demux_chain),
(gst_flv_demux_pull_tag), (gst_flv_demux_do_seek),
(gst_flv_demux_handle_seek), (gst_flv_demux_sink_event),
(gst_flv_demux_src_event), (gst_flv_demux_query),
(gst_flv_demux_change_state), (gst_flv_demux_set_index),
(gst_flv_demux_get_index), (gst_flv_demux_dispose),
(gst_flv_demux_class_init): First method for seeking in pull
mode using the index built step by step or coming from metadata.
* gst/flv/gstflvdemux.h:
* gst/flv/gstflvparse.c: (FLV_GET_STRING),
(gst_flv_parse_metadata_item), (gst_flv_parse_tag_script),
(gst_flv_parse_tag_audio), (gst_flv_parse_tag_video): Parse
more metadata types and keyframes index.
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While doing that also update all other doc...
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/gst-plugins-bad-plugins.args:
* docs/plugins/gst-plugins-bad-plugins.signals:
* docs/plugins/inspect/plugin-bz2.xml:
* docs/plugins/inspect/plugin-cdxaparse.xml:
* docs/plugins/inspect/plugin-dtsdec.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-gsm.xml:
* docs/plugins/inspect/plugin-gstrtpmanager.xml:
* docs/plugins/inspect/plugin-h264parse.xml:
* docs/plugins/inspect/plugin-modplug.xml:
* docs/plugins/inspect/plugin-mpeg2enc.xml:
* docs/plugins/inspect/plugin-musepack.xml:
* docs/plugins/inspect/plugin-musicbrainz.xml:
* docs/plugins/inspect/plugin-nsfdec.xml:
* docs/plugins/inspect/plugin-replaygain.xml:
* docs/plugins/inspect/plugin-soundtouch.xml:
* docs/plugins/inspect/plugin-spcdec.xml:
* docs/plugins/inspect/plugin-spectrum.xml:
* docs/plugins/inspect/plugin-speed.xml:
* docs/plugins/inspect/plugin-tta.xml:
* docs/plugins/inspect/plugin-videosignal.xml:
* docs/plugins/inspect/plugin-xingheader.xml:
* docs/plugins/inspect/plugin-xvid.xml:
* gst/filter/gstbpwsinc.c:
* gst/filter/gstbpwsinc.h:
* gst/filter/gstlpwsinc.c:
* gst/filter/gstlpwsinc.h:
Add docs for lpwsinc and bpwsinc and integrate them
into the build system. While doing that also update
all other docs via make update in docs/plugins.
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Original commit message from CVS:
* gst/rtpmanager/rtpjitterbuffer.c:
Include stdlib.
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Original commit message from CVS:
* gst/filter/gstbpwsinc.c: (bpwsinc_build_kernel):
* gst/filter/gstlpwsinc.c: (lpwsinc_build_kernel):
Improve debugging a bit.
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residue on stream changes.
Original commit message from CVS:
* gst/filter/gstbpwsinc.c: (gst_bpwsinc_class_init),
(bpwsinc_start):
* gst/filter/gstlpwsinc.c: (gst_lpwsinc_class_init),
(lpwsinc_start):
Reset the residue in BaseTransform::start to get a clean residue
on stream changes.
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filter kernel size.
Original commit message from CVS:
* gst/filter/gstbpwsinc.c: (process_32), (process_64):
* gst/filter/gstlpwsinc.c: (process_32), (process_64):
Fix processing with buffer sizes that are larger than the filter
kernel size.
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jitterbuffer code while also fixing some...
Original commit message from CVS:
* gst/rtpmanager/Makefile.am:
* gst/rtpmanager/async_jitter_queue.c:
* gst/rtpmanager/async_jitter_queue.h:
* gst/rtpmanager/rtpjitterbuffer.c: (rtp_jitter_buffer_class_init),
(rtp_jitter_buffer_init), (rtp_jitter_buffer_finalize),
(rtp_jitter_buffer_new), (compare_seqnum),
(rtp_jitter_buffer_insert), (rtp_jitter_buffer_pop),
(rtp_jitter_buffer_flush), (rtp_jitter_buffer_num_packets),
(rtp_jitter_buffer_get_ts_diff):
* gst/rtpmanager/rtpjitterbuffer.h:
Remove complicated async queue and replace with more simple jitterbuffer
code while also fixing some bugs.
* gst/rtpmanager/gstrtpbin-marshal.list:
* gst/rtpmanager/gstrtpbin.c: (on_new_ssrc), (on_ssrc_collision),
(on_ssrc_validated), (on_bye_ssrc), (on_bye_timeout), (on_timeout),
(create_session), (gst_rtp_bin_class_init), (create_recv_rtp),
(create_send_rtp):
* gst/rtpmanager/gstrtpbin.h:
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_init), (gst_rtp_jitter_buffer_dispose),
(gst_jitter_buffer_sink_parse_caps),
(gst_rtp_jitter_buffer_flush_start),
(gst_rtp_jitter_buffer_flush_stop),
(gst_rtp_jitter_buffer_change_state),
(gst_rtp_jitter_buffer_sink_event), (gst_rtp_jitter_buffer_chain),
(gst_rtp_jitter_buffer_loop), (gst_rtp_jitter_buffer_set_property):
* gst/rtpmanager/gstrtpsession.c: (on_new_ssrc),
(on_ssrc_collision), (on_ssrc_validated), (on_bye_ssrc),
(on_bye_timeout), (on_timeout), (gst_rtp_session_class_init),
(gst_rtp_session_init):
* gst/rtpmanager/gstrtpsession.h:
* gst/rtpmanager/rtpsession.c: (on_bye_ssrc), (session_cleanup):
Use new jitterbuffer code.
Expose some new signals in preparation for handling EOS.
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rebuild the kernel & residue with every ...
Original commit message from CVS:
* gst/filter/gstbpwsinc.c: (bpwsinc_build_kernel):
Fix a segfault with more than one channel and don't rebuild
the kernel & residue with every buffer.
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specifying the window function that should be used.
Original commit message from CVS:
* gst/filter/gstbpwsinc.c: (gst_bpwsinc_mode_get_type),
(gst_bpwsinc_window_get_type), (gst_bpwsinc_class_init),
(gst_bpwsinc_init), (bpwsinc_build_kernel), (bpwsinc_set_property),
(bpwsinc_get_property):
* gst/filter/gstbpwsinc.h:
Add support for a bandreject mode and allow specifying the window
function that should be used.
* gst/filter/gstlpwsinc.c:
And another small formatting fix.
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Original commit message from CVS:
* gst/filter/gstbpwsinc.c: (gst_bpwsinc_class_init),
(gst_bpwsinc_init), (process_32), (process_64),
(bpwsinc_build_kernel), (bpwsinc_setup), (bpwsinc_get_unit_size),
(bpwsinc_transform), (bpwsinc_set_property),
(bpwsinc_get_property):
* gst/filter/gstbpwsinc.h:
Apply the same changes to the bandpass filter:
- Support double input
- Fix processing for input with >1 channels
- Specify frequency in Hz
- Specify actual filter kernel length
- Use transform instead of transform_ip as we're working
out of place anyway
- Factor out filter kernel generation and update the filter
kernel when the properties are set
Fix bandpass filter kernel generation to actually generate
a bandpass filter by creating a highpass instead of a second
lowpass.
* gst/filter/gstlpwsinc.c: (gst_lpwsinc_class_init):
Small formatting fix.
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