Original commit message from CVS:
* gst/asfdemux/Makefile.am:
* gst/asfdemux/gstasf.c: (plugin_init):
* gst/asfdemux/gstrtspwms.c: (gst_rtsp_wms_before_send),
(gst_rtsp_wms_after_send), (gst_rtsp_wms_parse_sdp),
(gst_rtsp_wms_configure_stream), (_do_init),
(gst_rtsp_wms_base_init), (gst_rtsp_wms_class_init),
(gst_rtsp_wms_init), (gst_rtsp_wms_finalize),
(gst_rtsp_wms_change_state), (gst_rtsp_wms_extension_init):
* gst/asfdemux/gstrtspwms.h:
Move WMS RTSP extension from -good to here.
Port it to the new pluggable extension interface.
Original commit message from CVS:
* configure.ac:
Sync liboil check with plugins-base. Add libm check.
* gst/synaesthesia/Makefile.am:
Link against libm. We're using sqrt here.
Original commit message from CVS:
* gst/mpegaudioparse/gstmpegaudioparse.c: (gst_mp3parse_reset),
(gst_mp3parse_emit_frame), (mp3parse_handle_seek):
* gst/mpegaudioparse/gstmpegaudioparse.h:
Save some memory for each frame by only saving the start timestamp
and start byte position instead of additionally the stop timestamp
and stop byte position. This requires us to use a doubly-linked list
but still saves 8-12 bytes per frame.
Original commit message from CVS:
* gst/mpegaudioparse/gstmpegaudioparse.c:
(gst_mp3parse_emit_frame):
Fix a calculation that was causing mp3parse to drop every incoming
frame when upstream delivered a segment in TIME format, breaking
playback of all mpeg system streams.
Original commit message from CVS:
* configure.ac:
* ext/mpeg2dec/gstmpeg2dec.c: (crop_buffer):
* gst/asfdemux/gstasfdemux.c: (gst_asf_demux_descramble_buffer):
* gst/dvdlpcmdec/gstdvdlpcmdec.c: (gst_dvdlpcmdec_chain_raw):
Fix build against core CVS by not using deprecated API. Bump
requirements for new API (overdue anyway).
Original commit message from CVS:
* gst/mpegaudioparse/gstmpegaudioparse.c: (gst_mp3parse_base_init),
(gst_mp3parse_init):
Use GST_BOILERPLATE instead of manual GType magic.
* gst/mpegaudioparse/gstmpegaudioparse.c:
(gst_mp3parse_handle_first_frame), (mp3parse_time_to_bytepos),
(mp3parse_bytepos_to_time):
* gst/mpegaudioparse/gstmpegaudioparse.h:
Implement seeking, byte->time, time->byte conversions with the Xing
seek table if available. This allows better at least a bit more
accurate seeks and file position reporting.
Original commit message from CVS:
* gst/mpegaudioparse/gstmpegaudioparse.c:
(gst_mp3parse_handle_first_frame):
Copy the complete Xing seek table in the 100 byte array instead of
copying the first byte 100 times.
* gst/mpegaudioparse/gstmpegaudioparse.c: (mp3parse_total_bytes),
(mp3parse_total_time), (mp3parse_time_to_bytepos):
Add seeking support based on the Xing header but comment it out for
now as it seems to yield worse result than the other method.
Also use gst_pad_query_peer_duration() instead of getting the peer pad
ourself, creating a new GstQuery, etc.
Original commit message from CVS:
* gst/mpegaudioparse/gstmpegaudioparse.c: (mp3_caps_create):
Fix "pad caps are not a real subset of its template caps" warning.
Original commit message from CVS:
* gst/dvdsub/gstdvdsubdec.c:(gst_dvd_sub_dec_parse_subpic):
Use gst_util_guint64_to_gdouble for conversion.
* win32/vs6/libgstasfdemux.dsp:
Add asfpacket.c to the build.
Original commit message from CVS:
* gst/mpegaudioparse/gstmpegaudioparse.c: (gst_mp3parse_reset),
(gst_mp3parse_emit_frame), (gst_mp3parse_handle_first_frame):
* gst/mpegaudioparse/gstmpegaudioparse.h:
If the Xing header provides a total time, use it to calculate the
correct average bitrate immediately, instead of sending updates as
we parse the stream.
Original commit message from CVS:
Patch by by: Mark Nauwelaerts <manauw at skynet dot be>
* gst/dvdsub/gstdvdsubdec.c: (gst_dvd_sub_dec_parse_subpic):
Use GstClockTime instead of guint for a time variable to prevent
overflows on too large subtitle durations. Fixes#444514.
Original commit message from CVS:
Patch by: Mark Nauwelaerts <manauw at skynet be>
* gst/dvdsub/gstdvdsubdec.c: (gst_dvd_sub_dec_sink_event):
Clear state when handling the serialized FLUSH_STOP event instead of
the FLUSH_START event, thereby making sure we don't free buffers the
chain function is still using. Fixes dvdsubdec crashing when flusing
or seeking (#442706).
Original commit message from CVS:
* gst/mpegstream/gstmpegdemux.c: (gst_mpeg_demux_send_subbuffer):
Add sanity check so we don't abort for broken or non-MPEG streams,
but instead error out. Fixes crashes/aborts for when our typefinder
wrongly identifies quicktime files as mpeg (which should be fixed in
-base now too). (#440120).
Original commit message from CVS:
* gst/mpegaudioparse/gstmpegaudioparse.c:
(mp3_type_frame_length_from_header), (gst_mp3parse_reset),
(gst_mp3parse_emit_frame), (gst_mp3parse_handle_first_frame),
(gst_mp3parse_chain), (mp3parse_total_bytes),
(mp3parse_total_time):
* gst/mpegaudioparse/gstmpegaudioparse.h:
Implement parsing of Xing headers from the first frame of the stream,
and use it to report duration correctly where possible.
Original commit message from CVS:
* gst/realmedia/rmdemux.c: (gst_rmdemux_descramble_cook_audio):
After descrambling, push the packets out as individual packets
instead of one big descrambled buffer. Makes cook audio decoding
work with the 'realaudiodec' decoder from gst-plugins-bad.
Original commit message from CVS:
* gst/realmedia/rmdemux.c: (gst_rmdemux_init),
(gst_rmdemux_sink_event), (gst_rmdemux_perform_seek),
(gst_rmdemux_reset), (gst_rmdemux_chain), (gst_rmdemux_add_stream),
(gst_rmdemux_parse_packet):
* gst/realmedia/rmdemux.h:
Remember first timestamp encountered in stream and re-timestamp
stream to start from zero (fixes#397219); only send one newsegment
event, not two; when seeking, send newsegment events from the
streaming thread and not from the seeking thread.
Original commit message from CVS:
Based on patch by: Mark Nauwelaerts <manauw skynet be>
* gst/mpegstream/gstdvddemux.c: (gst_dvd_demux_process_event):
* gst/mpegstream/gstmpegdemux.c: (gst_mpeg_demux_class_init),
(gst_mpeg_demux_process_event), (gst_mpeg_streams_reset_last_flow):
* gst/mpegstream/gstmpegdemux.h:
Reset last_flow values for the various streams after a flushing
seek, otherwise we might aggregate wrong flow returns afterwards
that will make upstream pause silently. This should fix seeking
in DVDs and also fix the Thoggen cropping dialog (#438610).
Original commit message from CVS:
* gst/asfdemux/gstasfdemux.c: (gst_asf_demux_reset),
(gst_asf_demux_chain_headers),
(gst_asf_demux_parse_data_object_start), (all_streams_prerolled),
(gst_asf_demux_have_mutually_exclusive_active_stream),
(gst_asf_demux_check_activate_streams),
(gst_asf_demux_find_stream_with_complete_payload),
(gst_asf_demux_push_complete_payloads), (gst_asf_demux_loop),
(gst_asf_demux_activate_ext_props_streams),
(gst_asf_demux_process_object):
* gst/asfdemux/gstasfdemux.h:
Activate streams (ie. add the pads to the element) depending on
whether we actually get data for those streams within the ASF
preroll value specified. Currently only done in pull-mode though
(this will fix problems with playbin hanging on mms streams once
we use this in push-mode as well).
Original commit message from CVS:
* gst/asfdemux/asfpacket.c: (gst_asf_payload_queue_for_stream):
* gst/asfdemux/gstasfdemux.c: (gst_asf_demux_reset),
(gst_asf_demux_init), (gst_asf_demux_push_complete_payloads),
(gst_asf_demux_process_file):
* gst/asfdemux/gstasfdemux.h:
Make all timestamps start from zero in pull-mode too; some small
clean-ups and FIXMEs here and there.
Original commit message from CVS:
* gst/asfdemux/asfpacket.c: (gst_asf_demux_parse_payload),
(gst_asf_demux_parse_packet):
If packet size is specified within the packet and smaller than
the actual packet size, don't parse beyond the size specified in
the packet (this makes us parse some cases of packets with single
compressed payloads cleanly, see e.g stream from #431318). Also
add a sanity check when parsing compressed single payloads.
Original commit message from CVS:
* gst/asfdemux/asfpacket.c: (gst_asf_payload_queue_for_stream):
* gst/asfdemux/gstasfdemux.c: (gst_asf_demux_seek_index_lookup),
(gst_asf_demux_handle_seek_event),
(gst_asf_demux_push_complete_payloads):
Seeking improvements: honour the KEY_UNIT seek flag; after a seek, only
send data from the keyframe right before the new segment start to
make sure the decoder doesn't have to decode more than absolutely
necessary.
Original commit message from CVS:
* gst/asfdemux/asfheaders.h:
* gst/asfdemux/gstasfdemux.c: (gst_asf_demux_free_stream),
(gst_asf_demux_reset), (gst_asf_demux_parse_data_object_start),
(gst_asf_demux_loop), (gst_asf_demux_setup_pad),
(gst_asf_demux_add_audio_stream), (gst_asf_demux_activate_stream),
(gst_asf_demux_parse_stream_object),
(gst_asf_demux_process_ext_stream_props),
(gst_asf_demux_process_queued_extended_stream_objects),
(gst_asf_demux_activate_ext_props_streams),
(gst_asf_demux_process_object):
* gst/asfdemux/gstasfdemux.h:
Refactor stream parse/activation a bit (stream activation heuristics
are still the same though); some more clean-ups.
Original commit message from CVS:
* gst/asfdemux/gstasf.c: (plugin_init):
* gst/asfdemux/gstasfdemux.c: (gst_asf_demux_base_init):
* gst/asfdemux/gstasfdemux.h:
Init debug category before using it.
Original commit message from CVS:
* gst/asfdemux/gstasfdemux.c: (gst_asf_demux_pull_data),
(gst_asf_demux_push_complete_payloads), (gst_asf_demux_loop):
Fix silly bug when we can't pull as much data as we want; don't
forget to announce pending tags in the new packet parsing code.
Original commit message from CVS:
* gst/asfdemux/Makefile.am:
* gst/asfdemux/asfpacket.c: (asf_packet_read_varlen_int),
(asf_packet_create_payload_buffer),
(asf_payload_find_previous_fragment),
(gst_asf_payload_queue_for_stream), (gst_asf_demux_parse_payload),
(gst_asf_demux_parse_packet):
* gst/asfdemux/asfpacket.h:
* gst/asfdemux/gstasfdemux.c:
(gst_asf_demux_reset_stream_state_after_discont),
(gst_asf_demux_push_complete_payloads), (gst_asf_demux_loop),
(gst_asf_demux_setup_pad), (gst_asf_demux_descramble_buffer),
(gst_asf_demux_process_chunk):
* gst/asfdemux/gstasfdemux.h:
New packet parsing code: should put halfway decent timestamps on
buffers, and might even set the appropriate keyframe/discont buffer
flags from time to time (and even if it doesn't, I'm at least able
to debug this code); only used in pull-mode so far. Still needs
some more work, like payload extensions parsing and proper flow
aggregation, and stream activation based on preroll. Stay tuned.
Original commit message from CVS:
* gst/asfdemux/asfheaders.h:
* gst/asfdemux/gstasfdemux.c: (gst_asf_demux_free_stream),
(gst_asf_demux_seek_index_lookup),
(gst_asf_demux_handle_seek_event), (gst_asf_demux_get_stream),
(gst_asf_demux_setup_pad), (gst_asf_demux_add_video_stream),
(gst_asf_demux_process_ext_content_desc),
(gst_asf_demux_process_file), (gst_asf_demux_descramble_segment),
(gst_asf_demux_push_buffer), (gst_asf_demux_process_chunk),
(gst_asf_demux_process_segment), (gst_asf_demux_handle_data):
* gst/asfdemux/gstasfdemux.h:
Some clean-ups and small fixes: rename asf_stream_context structure to
AsfStream; inline some three-line utility functions that are only used
once anyway and get rid of their associated helper structs; make debug
category global so that it is used by the debug statements in the other
file as well; simplify gst_asf_demux_get_stream(); fix accidental
implicit initialisation of stream->last_buffer_timestamp to 0, which
would lead to missing timestamps on the first buffer; put fourcc format
into video caps to make certain proprietary wmv decoders happy (for the
case of WMVA in particular); play_time is offset by preroll as well, so
fix overreporting of duration for some files.
Original commit message from CVS:
* gst/mpegstream/gstmpegparse.c: (gst_mpeg_parse_process_event),
(gst_mpeg_parse_send_event):
Post an error message if EOS wasn't handled by anything downstream.
This should fix playbin freezing/hanging with small VobSub subtitle
files (background: not-linked flow returns from downstream are
ignored for a while at the beginning, so if the file is small
upstream will never get a not-linked flow return even if nothing
is connected downstream). (#429960).
Original commit message from CVS:
* gst/asfdemux/asfheaders.c:
* gst/asfdemux/asfheaders.h:
* gst/asfdemux/gstasfdemux.c: (gst_asf_demux_free_stream),
(gst_asf_demux_reset), (gst_asf_demux_init),
(gst_asf_demux_activate), (gst_asf_demux_activate_push),
(gst_asf_demux_activate_pull), (gst_asf_demux_sink_event),
(gst_asf_demux_seek_index_lookup),
(gst_asf_demux_reset_stream_state_after_discont),
(gst_asf_demux_handle_seek_event),
(gst_asf_demux_handle_src_event), (gst_asf_demux_chain_headers),
(gst_asf_demux_chain), (gst_asf_demux_pull_data),
(gst_asf_demux_pull_indices),
(gst_asf_demux_parse_data_object_start),
(gst_asf_demux_pull_headers), (gst_asf_demux_loop),
(gst_asf_demux_setup_pad), (gst_asf_demux_process_file),
(gst_asf_demux_process_simple_index),
(gst_asf_demux_process_object),
(gst_asf_demux_send_event_unlocked), (gst_asf_demux_push_buffer),
(gst_asf_demux_handle_data), (gst_asf_demux_change_state):
* gst/asfdemux/gstasfdemux.h:
Make asfdemux work in pull mode where possible. If there's an index
at the end of the file, read it and use it for seeking purposes.
Original commit message from CVS:
Patch by: René Stadler <mail at renestadler dot de>
* gst/realmedia/rmdemux.c: (find_seek_offset_bytes),
(find_seek_offset_time), (gst_rmdemux_reset),
(gst_rmdemux_get_stream_by_id), (gst_rmdemux_send_event),
(gst_rmdemux_add_stream), (gst_rmdemux_combine_flows):
* gst/realmedia/rmdemux.h:
Make rmdemux handle any number of logical streams. Fixes#428698.
Original commit message from CVS:
* gst/mpegaudioparse/gstmpegaudioparse.c:
(mp3_type_frame_length_from_header), (gst_mp3parse_reset),
(gst_mp3parse_init), (gst_mp3parse_sink_event),
(gst_mp3parse_emit_frame), (gst_mp3parse_chain),
(gst_mp3parse_change_state), (mp3parse_time_to_bytepos),
(mp3parse_bytepos_to_time), (mp3parse_total_bytes),
(mp3parse_total_time), (mp3parse_handle_seek),
(mp3parse_src_event), (mp3parse_src_query),
(mp3parse_get_query_types), (plugin_init):
* gst/mpegaudioparse/gstmpegaudioparse.h:
Implement seeking via average bitrate, and position+duration
querying in mp3parse. Later, it will support frame-accurate seeking by
building a seek table as it parses.
Add 'parsed=false' to the sink pad caps, and 'parsed=true' to the src
pad caps. Bump the priority to PRIMARY+1 so that it is autoplugged
before any extant MP3 decoder plugin. This allows us to remove framing
support from the decoders, if we want, and will provide them with
accurate seeking automatically once it is finished.
Fix the handling of MPEG-1 Layer 1 files.
Partially fix timestamping of packets arriving from a demuxer by
queueing the incoming timestamp until the next packet starts, rather
than applying it immediately to the next pushed buffer.
Original commit message from CVS:
* gst/dvdlpcmdec/gstdvdlpcmdec.c: (gst_dvdlpcm_reset),
(update_timestamps), (parse_header), (gst_dvdlpcmdec_chain_dvd),
(gst_dvdlpcmdec_chain_raw), (dvdlpcmdec_sink_event):
* gst/dvdlpcmdec/gstdvdlpcmdec.h:
Implement all sample rates.
Implement sample permutation a little smarter avoiding a memcpy.
Fix timestamps, use segments, fix seeking.
Original commit message from CVS:
* gst/asfdemux/gstasfdemux.c: (gst_asf_demux_process_file),
(gst_asf_demux_process_advanced_mutual_exclusion),
(gst_asf_demux_process_queued_extended_stream_objects),
(gst_asf_demux_process_object), (gst_asf_demux_change_state):
* gst/asfdemux/gstasfdemux.h:
Parse advanced mutual exclusion object and only add pads for
'hidden' streams (those in an extended stream header) that are
mutually exclusive with an already existing 'main stream' if
the broadcasting flag is not set. If the broadcasting flag is set,
assume that data for this stream isn't sent. (This should ideally be
solved better by making playbin more robust against this and/or by
making mmssrc send some information downstream about which streams
will be streamed). Fixes#353116.
Original commit message from CVS:
* gst/synaesthesia/gstsynaesthesia.c:
(gst_synaesthesia_class_init), (gst_synaesthesia_init),
(gst_synaesthesia_finalize), (gst_synaesthesia_chain):
* gst/synaesthesia/synaescope.c: (synaescope_coreGo),
(synaescope32), (synaescope_set_data), (synaesthesia_update),
(synaesthesia_init), (synaesthesia_new), (synaesthesia_close):
* gst/synaesthesia/synaescope.h:
Move all the mutable engine state into a structure so that
multiple element instances can run without interfering.
Original commit message from CVS:
* gst/realmedia/rmdemux.c:(gst_rmdemux_parse_indx_data):
Use gst_guint64_to_gdouble for conversions.
* gst/synaesthesia/synaescope.c:
Define M_PI and do not include <pthread.h> and
<sys/time.h> for G_OS_WIN32
* win32/vs6/libgstrealmedia.dsp:
* win32/vs6/synaesthesia.dsp:
Update projects files.
* win32/common/config.h.in:
Add config.h.in for autogen of config.h
Original commit message from CVS:
* ext/lame/gstlame.c: (gst_lame_sink_event), (gst_lame_chain),
(gst_lame_change_state):
* ext/lame/gstlame.h:
On receiving EOS, we try to push a last buffer with the remaining
samples. Don't do that if we got an unclean flow return on the last
gst_pad_push(), downstream might not handle this very gracefully
(see #403168).
* gst/mpegaudioparse/gstmpegaudioparse.c: (gst_mp3parse_chain):
Pass flow returns upstream (helps #403168).
Original commit message from CVS:
* gst/synaesthesia/gstsynaesthesia.c:
(gst_synaesthesia_class_init), (gst_synaesthesia_init),
(gst_synaesthesia_sink_setcaps), (gst_synaesthesia_src_getcaps),
(gst_synaesthesia_chain), (plugin_init):
check result of gst_pad_push() in _chain.
