| Age | Commit message (Collapse) | Author | Files | Lines |
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alpha channel which needs a different dec...
Original commit message from CVS:
* gst/flv/gstflvparse.c: (gst_flv_parse_audio_negotiate),
(gst_flv_parse_tag_audio), (gst_flv_parse_video_negotiate):
Correct caps for video codec id 5: It's On2 VP6 with alpha channel
which needs a different decoder and has different caps.
Add support for audio codec id 14, which is MP3 with 8kHz sampling
rate.
Fix endianness and signedness for raw audio codec ids.
Add support for alaw and mulaw audio.
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instead of parsing until the GstAdapter is...
Original commit message from CVS:
* gst/flv/gstflvdemux.c: (gst_flv_demux_chain):
Go out of the parse loop as soon as we get an error instead
of parsing until the GstAdapter is empty.
Add some explanations about the header and tag size.
Don't print synchronizing message if everything is fine.
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writing of stream metadata.
Original commit message from CVS:
* gst/flv/Makefile.am:
* gst/flv/gstflvdemux.c: (plugin_init):
* gst/flv/gstflvmux.c: (gst_flv_mux_base_init),
(gst_flv_mux_class_init), (gst_flv_mux_init),
(gst_flv_mux_finalize), (gst_flv_mux_reset),
(gst_flv_mux_handle_src_event), (gst_flv_mux_handle_sink_event),
(gst_flv_mux_video_pad_setcaps), (gst_flv_mux_audio_pad_setcaps),
(gst_flv_mux_request_new_pad), (gst_flv_mux_release_pad),
(gst_flv_mux_write_header), (gst_flv_mux_write_buffer),
(gst_flv_mux_collected), (gst_flv_mux_change_state):
* gst/flv/gstflvmux.h:
Add first version of a FLV muxer. The only missing feature is writing
of stream metadata.
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Original commit message from CVS:
* gst/mpegdemux/gstmpegdemux.c:
* gst/mpegdemux/gstmpegtsdemux.c:
Add Fluendo to the Long Name.
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Original commit message from CVS:
* configure.ac:
* gst-plugins-bad.spec.in:
* gst/mpegdemux/Makefile.am:
* gst/mpegdemux/flumpegdemux.c:
* gst/mpegdemux/gstmpegdesc.c:
* gst/mpegdemux/gstmpegdesc.h:
* gst/mpegdemux/mpegtspacketizer.c:
* gst/mpegdemux/mpegtspacketizer.h:
* gst/mpegdemux/mpegtsparse.c:
* gst/mpegdemux/mpegtsparse.h:
Move of mpegtsparse to mpegdemux.
Fixes #555193.
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Original commit message from CVS:
Move of mpegtsparse to mpegdemux
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was zero.
Original commit message from CVS:
* gst/mpegdemux/gstmpegdemux.c: (gst_flups_demux_send_data),
(gst_flups_demux_parse_pack_start):
Prevent a division by zero if last mux rate was zero.
If we're going to send a NEWSEGMENT event but the segment start
and the current buffer timestamp differ by more than a second we
will start the NEWSEGMENT at the buffer timestamp.
This fixes playback of the tv2-1_25.mpg file, which has 0 as first SCR
but the first PTS are around 1 hour and 40 minutes.
Fixes bug #553755.
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Original commit message from CVS:
Patch by: Håvard Graff <havard dot graff at tandberg dot com>
* gst/rtpmanager/gstrtpbin-marshal.list:
Add marshaller for new action signal.
* gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_get_internal_session),
(gst_rtp_bin_class_init):
* gst/rtpmanager/gstrtpbin.h:
Add action signal to retrieve the internal RTPSession object.
* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_class_init),
(gst_rtp_session_get_property), (gst_rtp_session_release_pad):
Add property to access the internal RTPSession object.
* gst/rtpmanager/rtpsession.c: (rtp_session_class_init),
(check_collision):
* gst/rtpmanager/rtpsession.h:
Add action signal to retrieve an RTPSource object by SSRC.
See #555396.
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#555244
Original commit message from CVS:
* gst/selector/gstoutputselector.c:
Choose right pad for sending events. Fixes #555244
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Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (find_session_by_pad),
(free_session), (gst_rtp_bin_dispose), (remove_recv_rtp),
(remove_recv_rtcp), (remove_send_rtp), (remove_rtcp),
(gst_rtp_bin_release_pad):
Release pads of the session manager.
Start implementing releasing pads of gstrtpbin.
* gst/rtpmanager/gstrtpsession.c: (remove_recv_rtp_sink),
(remove_recv_rtcp_sink), (remove_send_rtp_sink),
(remove_send_rtcp_src), (gst_rtp_session_release_pad):
Implement releasing pads in gstrtpsession.
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not already configured for the streams.
Original commit message from CVS:
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_jitter_buffer_sink_parse_caps):
Only update the seqnum-base when it was not already configured for the
streams.
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assigning it because it gets freed straight...
Original commit message from CVS:
* gst/mpegtsparse/mpegtsparse.c:
Actually copy the structure passed in when assigning it because
it gets freed straight after the function call.
Re: pat_info and pmt_info GstStructures.
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previous optimisation.
Original commit message from CVS:
Patch by: Josep Torra
* gst/mpegdemux/gstmpegtsdemux.c:
Fix wrong firing of critical introduced by previous optimisation.
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freed structure.
Original commit message from CVS:
* gst/mpegtsparse/mpegtsparse.c:
Fix possible crash where pat is pointing to a freed structure.
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Original commit message from CVS:
* gst/flacparse/gstbaseparse.c: (gst_base_parse_finalize),
(gst_base_parse_class_init), (gst_base_parse_push_buffer),
(gst_base_parse_change_state), (gst_base_parse_set_index),
(gst_base_parse_get_index):
Add support for GstIndex.
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whether a frame is inside the segment or not...
Original commit message from CVS:
* gst/flacparse/gstbaseparse.c: (gst_base_parse_class_init),
(gst_base_parse_push_buffer),
(gst_base_parse_update_upstream_durations),
(gst_base_parse_convert), (gst_base_parse_frame_in_segment):
* gst/flacparse/gstbaseparse.h:
Provide a vfunc for the subclass to decide whether a frame is inside
the segment or not and add a default implementation.
Fix approximate bitrate calculations.
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size if possible and add a default conver...
Original commit message from CVS:
* gst/flacparse/gstbaseparse.c: (gst_base_parse_class_init),
(gst_base_parse_init), (gst_base_parse_push_buffer),
(gst_base_parse_update_upstream_durations), (gst_base_parse_chain),
(gst_base_parse_loop), (gst_base_parse_activate),
(gst_base_parse_convert), (gst_base_parse_query):
Approximate the average bitrate, duration and size if possible
and add a default conversion function which uses this for
time<->byte conversions.
* gst/flacparse/gstflacparse.c: (gst_flac_parse_get_frame_size):
Fix parsing if upstream gives -1 as duration.
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session lock when we emit the signals.
Original commit message from CVS:
* gst/rtpmanager/rtpsession.c: (on_new_ssrc), (on_ssrc_collision),
(on_ssrc_validated), (on_ssrc_active), (on_ssrc_sdes),
(on_bye_ssrc), (on_bye_timeout), (on_timeout), (on_sender_timeout):
Ref the rtpsource object before we release the session lock when we emit
the signals.
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that will be added to libgstbase later.
Original commit message from CVS:
* configure.ac:
* gst/flacparse/Makefile.am:
* gst/flacparse/gstbaseparse.c: (gst_base_parse_get_type),
(gst_base_parse_base_init), (gst_base_parse_base_finalize),
(gst_base_parse_finalize), (gst_base_parse_class_init),
(gst_base_parse_init), (gst_base_parse_check_frame),
(gst_base_parse_parse_frame), (gst_base_parse_bytepos_to_time),
(gst_base_parse_sink_event), (gst_base_parse_sink_eventfunc),
(gst_base_parse_src_event), (gst_base_parse_src_eventfunc),
(gst_base_parse_is_seekable), (gst_base_parse_push_buffer),
(gst_base_parse_handle_and_push_buffer), (gst_base_parse_drain),
(gst_base_parse_chain), (gst_base_parse_pull_range),
(gst_base_parse_loop), (gst_base_parse_sink_activate),
(gst_base_parse_activate), (gst_base_parse_sink_activate_push),
(gst_base_parse_sink_activate_pull), (gst_base_parse_set_duration),
(gst_base_parse_set_min_frame_size),
(gst_base_parse_get_querytypes), (gst_base_parse_query),
(gst_base_parse_handle_seek), (gst_base_parse_sink_setcaps):
* gst/flacparse/gstbaseparse.h:
* gst/flacparse/gstbitreader.c: (gst_bit_reader_new),
(gst_bit_reader_new_from_buffer), (gst_bit_reader_free),
(gst_bit_reader_init), (gst_bit_reader_init_from_buffer),
(gst_bit_reader_set_pos), (gst_bit_reader_get_pos),
(gst_bit_reader_get_remaining), (gst_bit_reader_skip),
(gst_bit_reader_skip_to_byte):
* gst/flacparse/gstbitreader.h:
* gst/flacparse/gstbytereader.c: (GDOUBLE_SWAP_LE_BE),
(GFLOAT_SWAP_LE_BE), (gst_byte_reader_new),
(gst_byte_reader_new_from_buffer), (gst_byte_reader_free),
(gst_byte_reader_init), (gst_byte_reader_init_from_buffer),
(gst_byte_reader_set_pos), (gst_byte_reader_get_pos),
(gst_byte_reader_get_remaining), (gst_byte_reader_skip),
(gst_byte_reader_get_uint8), (gst_byte_reader_get_int8),
(gst_byte_reader_peek_uint8), (gst_byte_reader_peek_int8),
(gst_byte_reader_get_uint24_le), (gst_byte_reader_get_uint24_be),
(gst_byte_reader_get_int24_le), (gst_byte_reader_get_int24_be),
(gst_byte_reader_peek_uint24_le), (gst_byte_reader_peek_uint24_be),
(gst_byte_reader_peek_int24_le), (gst_byte_reader_peek_int24_be):
* gst/flacparse/gstbytereader.h:
* gst/flacparse/gstflac.c: (plugin_init):
* gst/flacparse/gstflacparse.c: (gst_flac_parse_base_init),
(gst_flac_parse_class_init), (gst_flac_parse_init),
(gst_flac_parse_finalize), (gst_flac_parse_start),
(gst_flac_parse_stop), (gst_flac_parse_get_frame_size),
(gst_flac_parse_check_valid_frame),
(gst_flac_parse_handle_streaminfo),
(gst_flac_parse_handle_vorbiscomment),
(gst_flac_parse_handle_picture), (_value_array_append_buffer),
(gst_flac_parse_handle_headers), (gst_flac_parse_generate_headers),
(gst_flac_parse_parse_frame):
* gst/flacparse/gstflacparse.h:
Add FLAC parser, based on GstBaseParse. Also add the bit and byte reader
that will be added to libgstbase later.
The FLAC parser is currently not 100% bug free and fails to get the
correct frame size for some frames in some streams.
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Original commit message from CVS:
* gst/rtpmanager/rtpjitterbuffer.c: (rtp_jitter_buffer_insert),
(rtp_jitter_buffer_get_sync):
* gst/rtpmanager/rtpsession.c: (on_sender_timeout),
(session_cleanup):
* gst/rtpmanager/rtpsource.c:
Fix some docs.
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Original commit message from CVS:
Patch from: Josep Torra
* gst/mpegdemux/gstmpegtsdemux.c:
* gst/mpegdemux/gstmpegtsdemux.h:
Use a preallocated buffer per stream for PES packets sent on src pads.
Adaptively adjust buffer size appropriately.
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Original commit message from CVS:
* ext/jack/gstjackaudiosink.c: (jack_process_cb):
* gst/rtpmanager/rtpjitterbuffer.c: (calculate_skew):
Fix compiler warnings on OS/X
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Original commit message from CVS:
* ext/celt/gstceltenc.h:
Help gtk-doc to parse this correctly.
* gst/pcapparse/gstpcapparse.c:
Add missing include.
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which we have not yet seen an SR packet. A...
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (create_session),
(gst_rtp_bin_associate), (gst_rtp_bin_sync_chain):
Do not try to adjust the offset of streams for which we have not yet
seen an SR packet. Avoids large ts-offsets in some cases.
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Original commit message from CVS:
* gst/mpegdemux/flumpegdemux.c: (plugin_init):
* gst/mpegdemux/gstmpegdemux.c: (gst_flups_demux_sync_get_type),
(gst_flups_demux_get_type), (gst_flups_demux_plugin_init):
* gst/mpegdemux/gstmpegtsdemux.c: (gst_fluts_demux_get_type),
(gst_fluts_demux_plugin_init):
Fix conflicting public names in new mpeg demuxers.
Fixes #550468
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and any order so long as we get COMM bef...
Original commit message from CVS:
* gst/aiffparse/aiffparse.c:
Support chunks in AIFF in any order in pull mode, and any order so
long as we get COMM before the actual data (SSND) in push mode.
Fixes playback of AIFC files.
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Original commit message from CVS:
* gst/selector/gstinputselector.c: (gst_selector_pad_reset),
(gst_input_selector_reset), (gst_input_selector_change_state):
Reset the selector state when going to READY.
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becomes a receiver.
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (on_sender_timeout),
(create_session), (gst_rtp_bin_associate),
(gst_rtp_bin_sync_chain), (gst_rtp_bin_class_init),
(gst_rtp_bin_request_new_pad):
* gst/rtpmanager/gstrtpbin.h:
Add signal to notify listeners when a sender becomes a receiver.
Tweak lip-sync code, don't store our own copy of the ts-offset of the
jitterbuffer, don't adjust sync if the change is less than 4msec.
Get the RTP timestamp <-> GStreamer timestamp relation directly from
the jitterbuffer instead of our inaccurate version from the source.
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_chain), (gst_rtp_jitter_buffer_loop),
(gst_rtp_jitter_buffer_get_sync):
* gst/rtpmanager/gstrtpjitterbuffer.h:
Add G_LIKELY macros, use global defines for max packet reorder and
dropouts.
Reset the jitterbuffer clock skew detection when packets seqnums are
changed unexpectedly.
* gst/rtpmanager/gstrtpsession.c: (on_sender_timeout),
(gst_rtp_session_class_init), (gst_rtp_session_init):
* gst/rtpmanager/gstrtpsession.h:
Add sender timeout signal.
* gst/rtpmanager/rtpjitterbuffer.c: (rtp_jitter_buffer_reset_skew),
(calculate_skew), (rtp_jitter_buffer_insert),
(rtp_jitter_buffer_get_sync):
* gst/rtpmanager/rtpjitterbuffer.h:
Add some G_LIKELY macros.
Keep track of the extended RTP timestamp so that we can report the RTP
timestamp <-> GStreamer timestamp relation for lip-sync.
Remove server timestamp gap detection code, the server can sometimes
make a huge gap in timestamps (talk spurts,...) see #549774.
Detect timetamp weirdness instead by observing the sender/receiver
timestamp relation and resync if it changes more than 1 second.
Add method to report about the current rtp <-> gst timestamp relation
which is needed for lip-sync.
* gst/rtpmanager/rtpsession.c: (rtp_session_class_init),
(on_sender_timeout), (check_collision), (rtp_session_process_sr),
(session_cleanup):
* gst/rtpmanager/rtpsession.h:
Add sender timeout signal.
Remove inaccurate rtp <-> gst timestamp relation code, the
jitterbuffer can now do an accurate reporting about this.
* gst/rtpmanager/rtpsource.c: (rtp_source_init),
(rtp_source_update_caps), (calculate_jitter),
(rtp_source_process_rtp):
* gst/rtpmanager/rtpsource.h:
Remove inaccurate rtp <-> gst timestamp relation code.
* gst/rtpmanager/rtpstats.h:
Define global max-reorder and max-dropout constants for use in various
subsystems.
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Original commit message from CVS:
* gst/mpegdemux/gstmpegdemux.c: (gst_flups_demux_parse_pack_start):
* gst/mpegdemux/gstmpegtsdemux.c: (gst_fluts_demux_data_cb):
Fix build on macosx.
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licensed MPL and LGPL.
Original commit message from CVS:
* configure.ac:
* gst/mpegdemux/Makefile.am:
* gst/mpegdemux/flumpegdemux.c:
* gst/mpegdemux/flutspatinfo.c:
* gst/mpegdemux/flutspatinfo.h:
* gst/mpegdemux/flutspmtinfo.c:
* gst/mpegdemux/flutspmtinfo.h:
* gst/mpegdemux/flutspmtstreaminfo.c:
* gst/mpegdemux/flutspmtstreaminfo.h:
* gst/mpegdemux/gstmpegdefs.h:
* gst/mpegdemux/gstmpegdemux.c:
* gst/mpegdemux/gstmpegdemux.h:
* gst/mpegdemux/gstmpegdesc.c:
* gst/mpegdemux/gstmpegdesc.h:
* gst/mpegdemux/gstmpegtsdemux.c:
* gst/mpegdemux/gstmpegtsdemux.h:
* gst/mpegdemux/gstpesfilter.c:
* gst/mpegdemux/gstpesfilter.h:
* gst/mpegdemux/gstsectionfilter.c:
* gst/mpegdemux/gstsectionfilter.h:
Add Fluendo MPEG PS and TS demuxers to gst-plugins-bad. This
is now dual licensed MPL and LGPL.
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Original commit message from CVS:
* gst/mpegtsmux/mpegtsmux.c: (new_packet_cb):
Set caps on outgoing buffers.
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Original commit message from CVS:
* ext/resindvd/plugin.c: (plugin_init):
* ext/resindvd/resindvdsrc.c:
* ext/twolame/gsttwolame.c: (plugin_init):
* gst/aiffparse/aiffparse.c: (plugin_init):
Enable/fix up translations for these plugins.
* po/LINGUAS:
Add 'ca' to LINGUAS.
* po/POTFILES.in:
* po/POTFILES.skip:
Add more files for translation and more files which tools
should skip.
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Original commit message from CVS:
* gst/mpegtsmux/tsmux/tsmux.c: (tsmux_write_ts_header):
Fix build on macosx.
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instead of using malloc().
Original commit message from CVS:
* gst/mpegtsmux/mpegtsmux_aac.c: (mpegtsmux_prepare_aac):
Allocate a fixed size buffer on the stack instead of using malloc().
* gst/mpegtsmux/tsmux/tsmux.c: (tsmux_new), (tsmux_free),
(tsmux_program_new), (tsmux_program_free):
* gst/mpegtsmux/tsmux/tsmuxstream.c: (tsmux_stream_new),
(tsmux_stream_free), (tsmux_stream_consume),
(tsmux_stream_add_data):
Use GSlice.
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Original commit message from CVS:
* gst/mpegtsmux/mpegtsmux.c: (mpegtsmux_create_stream):
Add support for muxing MPEG4 video.
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Original commit message from CVS:
* gst/mpegtsmux/tsmux/tsmux.h:
* gst/mpegtsmux/tsmux/tsmuxstream.h:
Fix build of mpegtsmux.
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to mpegtsmux to prevent conflicts. Also al...
Original commit message from CVS:
* configure.ac:
* gst/mpegtsmux/Makefile.am:
* gst/mpegtsmux/mpegtsmux.c: (mpegtsmux_base_init),
(mpegtsmux_class_init), (mpegtsmux_init), (mpegtsmux_dispose),
(gst_mpegtsmux_set_property), (gst_mpegtsmux_get_property),
(release_buffer_cb), (mpegtsmux_create_stream),
(mpegtsmux_create_streams), (mpegtsmux_choose_best_stream),
(mpegtsmux_collected), (mpegtsmux_request_new_pad),
(mpegtsmux_release_pad), (new_packet_cb),
(mpegtsdemux_prepare_srcpad), (mpegtsmux_change_state),
(plugin_init):
* gst/mpegtsmux/mpegtsmux.h:
* gst/mpegtsmux/mpegtsmux_aac.c: (mpegtsmux_prepare_aac):
* gst/mpegtsmux/mpegtsmux_aac.h:
* gst/mpegtsmux/mpegtsmux_h264.c: (mpegtsmux_prepare_h264):
* gst/mpegtsmux/mpegtsmux_h264.h:
* gst/mpegtsmux/tsmux/Makefile.am:
* gst/mpegtsmux/tsmux/crc.h:
* gst/mpegtsmux/tsmux/tsmux.c: (tsmux_new), (tsmux_set_write_func),
(tsmux_set_pat_frequency), (tsmux_get_pat_frequency), (tsmux_free),
(tsmux_program_new), (tsmux_set_pmt_frequency),
(tsmux_get_pmt_frequency), (tsmux_program_add_stream),
(tsmux_program_set_pcr_stream), (tsmux_get_new_pid),
(tsmux_create_stream), (tsmux_find_stream), (tsmux_packet_out),
(tsmux_write_adaptation_field), (tsmux_write_ts_header),
(tsmux_write_stream_packet), (tsmux_program_free),
(tsmux_write_section), (tsmux_write_section_hdr),
(tsmux_write_pat), (tsmux_write_pmt):
* gst/mpegtsmux/tsmux/tsmux.h:
* gst/mpegtsmux/tsmux/tsmuxcommon.h:
* gst/mpegtsmux/tsmux/tsmuxstream.c: (tsmux_stream_new),
(tsmux_stream_get_pid), (tsmux_stream_free),
(tsmux_stream_set_buffer_release_func), (tsmux_stream_consume),
(tsmux_stream_at_pes_start), (tsmux_stream_bytes_avail),
(tsmux_stream_bytes_in_buffer), (tsmux_stream_get_data),
(tsmux_stream_pes_header_length),
(tsmux_stream_find_pts_dts_within),
(tsmux_stream_write_pes_header), (tsmux_stream_add_data),
(tsmux_stream_get_es_descrs), (tsmux_stream_pcr_ref),
(tsmux_stream_pcr_unref), (tsmux_stream_is_pcr),
(tsmux_stream_get_pts):
* gst/mpegtsmux/tsmux/tsmuxstream.h:
Add Fluendo MPEG-TS muxer and libtsmux to gst-plugins-bad. This
is renamed to mpegtsmux to prevent conflicts. Also all relevant
informations about copyright and license are added to the top of
every file but apart from that no changes compared to the latest
SVN versions happened.
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and sinkpads because they are the same.
Original commit message from CVS:
* gst/selector/gstinputselector.c: (gst_input_selector_init),
(gst_input_selector_event), (gst_input_selector_query):
Reuse the get_linked_pads for both source and sinkpads because they are
the same.
Implement a custum event handler and get the internally linked pad
directly instead of relying on the default (slower) implementation.
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changing the pitch by handling seeks with a r...
Original commit message from CVS:
Patch by: Rov Juvano <rovjuvano at users dot sourceforge dot net>
* 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-scaletempo.xml:
* examples/scaletempo/Makefile.am:
* examples/scaletempo/demo-gui.c: (pop_status_bar),
(status_bar_printf), (demo_gui_seek_bar_format), (update_position),
(demo_gui_seek_bar_change), (demo_gui_do_change_rate),
(demo_gui_do_set_rate), (demo_gui_do_rate_entered),
(demo_gui_do_toggle_advanced), (demo_gui_do_toggle_disabled),
(demo_gui_do_seek), (demo_gui_do_play), (demo_gui_do_pause),
(demo_gui_do_play_pause), (demo_gui_do_open_file),
(demo_gui_do_playlist_prev), (demo_gui_do_playlist_next),
(demo_gui_do_about_dialog), (demo_gui_do_quit),
(demo_gui_request_set_stride), (demo_gui_request_set_overlap),
(demo_gui_request_set_search), (demo_gui_rate_changed),
(demo_gui_playing_started), (demo_gui_playing_paused),
(demo_gui_playing_ended), (demo_gui_player_errored),
(demo_gui_stride_changed), (demo_gui_overlap_changed),
(demo_gui_search_changed), (demo_gui_set_player_func),
(demo_gui_set_playlist_func), (build_gvalue_array),
(create_action), (demo_gui_show_func), (demo_gui_set_player),
(demo_gui_set_playlist), (demo_gui_show), (demo_gui_get_property),
(demo_gui_set_property), (demo_gui_init), (demo_gui_class_init),
(demo_gui_get_type):
* examples/scaletempo/demo-gui.h:
* examples/scaletempo/demo-main.c: (handle_error_message),
(handle_quit), (main):
* examples/scaletempo/demo-player.c: (no_pipeline),
(demo_player_event_listener), (demo_player_state_changed_cb),
(demo_player_eos_cb), (demo_player_build_pipeline), (_set_rate),
(demo_player_scale_rate_func), (demo_player_set_rate_func),
(_set_state_and_wait), (demo_player_load_uri_func),
(demo_player_play_func), (demo_player_pause_func), (_seek_to),
(demo_player_seek_by_func), (demo_player_seek_to_func),
(demo_player_get_position_func), (demo_player_get_duration_func),
(demo_player_scale_rate), (demo_player_set_rate),
(demo_player_load_uri), (demo_player_play), (demo_player_pause),
(demo_player_seek_by), (demo_player_seek_to),
(demo_player_get_position), (demo_player_get_duration),
(demo_player_get_property), (demo_player_set_property),
(demo_player_init), (demo_player_class_init),
(demo_player_get_type):
* examples/scaletempo/demo-player.h:
* gst/scaletempo/Makefile.am:
* gst/scaletempo/gstscaletempo.c: (best_overlap_offset_float),
(best_overlap_offset_s16), (output_overlap_float),
(output_overlap_s16), (fill_queue), (reinit_buffers),
(gst_scaletempo_transform), (gst_scaletempo_transform_size),
(gst_scaletempo_sink_event), (gst_scaletempo_set_caps),
(gst_scaletempo_get_property), (gst_scaletempo_set_property),
(gst_scaletempo_base_init), (gst_scaletempo_class_init),
(gst_scaletempo_init):
* gst/scaletempo/gstscaletempo.h:
* gst/scaletempo/gstscaletempoplugin.c: (plugin_init):
Add scaletempo plugin, which allows to scale the speed of audio without
changing the pitch by handling seeks with a rate!=1.0.
Integrate it into the docs and add the example application for it.
Fixes bug #537700.
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Original commit message from CVS:
* gst/dccp/gstdccp.c:
* gst/dccp/gstdccpclientsrc.c:
Fix compilation on Solaris by including filio.h as needed.
* gst/deinterlace2/tvtime/tomsmocomp/TomsMoCompAll.inc:
* gst/deinterlace2/tvtime/tomsmocomp/TomsMoCompAll2.inc:
Fix compilation with Forte - apparently it hates concatenating a
macro argument that starts with an underscore??
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us to.
Original commit message from CVS:
* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_send_rtcp),
(gst_rtp_session_event_send_rtp_sink):
Send EOS when the session object instructs us to.
* gst/rtpmanager/rtpsession.c: (rtp_session_on_timeout):
* gst/rtpmanager/rtpsession.h:
Make it possible for the session manager to instruct us to send EOS. We
currently will EOS when the session is a sender and when the sender part
goes EOS. This is not entirely correct behaviour because the session
could still participate as a receiver.
Fixes #549409.
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with the correct endianness. Fixes playback ...
Original commit message from CVS:
* gst/aiffparse/aiffparse.c:
Read size of chunks preceeding the audio data with the
correct endianness. Fixes playback of some files.
Fixes #538500
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Original commit message from CVS:
* configure.ac:
* gst/aiffparse/Makefile.am:
* gst/aiffparse/aiffparse.c:
* gst/aiffparse/aiffparse.h:
Add an AIFF parsing element, heavily based on wavparse.
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by taking the latency of all sinkpads an...
Original commit message from CVS:
* gst/selector/gstinputselector.c: (gst_input_selector_init),
(gst_input_selector_query):
Implement the LATENCY query in a better way by taking the latency of all
sinkpads and taking the min/max instead of just taking a random pad.
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once. This should give a small speedup an...
Original commit message from CVS:
* gst/deinterlace2/tvtime/tomsmocomp/SearchLoopBottom.inc:
* gst/deinterlace2/tvtime/tomsmocomp/SearchLoopTop.inc:
* gst/deinterlace2/tvtime/tomsmocomp/StrangeBob.inc:
* gst/deinterlace2/tvtime/tomsmocomp/WierdBob.inc:
Unroll the loop to handle two bytes at once. This should give
a small speedup and makes it possible to handle chroma and luma
different which is needed later.
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in a header !
Original commit message from CVS:
* gst/dccp/gstdccpserversink.c:
* gst/dccp/gstdccpserversink.h:
Don't put globals only used by one '.c' file in a header !
Declare it as static, fixes build on macosx.
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Original commit message from CVS:
* gst/dccp/gstdccp.c: (gst_dccp_send_buffer):
Whoops, that was one fix too much :)
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convenience macro instead. Fixes build on...
Original commit message from CVS:
* gst/dccp/gstdccp.c: (gst_dccp_read_buffer),
(gst_dccp_send_buffer), (gst_dccp_set_sock_windowsize):
size_t's size varies by platform/architecture. Use glib convenience
macro instead. Fixes build on macosx.
Remove ending '\n' in debug statements.
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Original commit message from CVS:
* gst/pcapparse/gstpcapparse.c: (gst_pcap_parse_class_init):
Remove unused code and fix includes.
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deinterlacing algorithm. This only supports s...
Original commit message from CVS:
* gst/deinterlace2/gstdeinterlace2.c:
(gst_deinterlace_method_class_init):
* gst/deinterlace2/gstdeinterlace2.h:
* gst/deinterlace2/tvtime/tomsmocomp.c:
(gst_deinterlace_method_tomsmocomp_class_init):
* gst/deinterlace2/tvtime/tomsmocomp/SearchLoopBottom.inc:
* gst/deinterlace2/tvtime/tomsmocomp/SearchLoopTop.inc:
* gst/deinterlace2/tvtime/tomsmocomp/StrangeBob.inc:
* gst/deinterlace2/tvtime/tomsmocomp/TomsMoCompAll.inc:
* gst/deinterlace2/tvtime/tomsmocomp/TomsMoCompAll2.inc:
* gst/deinterlace2/tvtime/tomsmocomp/WierdBob.inc:
* gst/deinterlace2/tvtime/tomsmocomp/tomsmocompmacros.h:
First part of the C implementation of the tomsmocomp deinterlacing
algorithm. This only supports search-effort=0 currently, is painfully
slow and needs some cleanup later when all search-effort settings
are implemented in C.
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