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:
* 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/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:
* 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/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:
* 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/mpegaudioparse/gstmpegaudioparse.c: (gst_mp3parse_chain):
All sample-rates < 32khz come from the LSF extensions, which only
use 1 granule. Fixes parsing of 22.05khz, 24khz and 16khz files.
Use gst_util_uint64_scale because we can.
Original commit message from CVS:
* gst/mpegaudioparse/gstmpegaudioparse.c:
(mp3_type_frame_length_from_header), (gst_mp3parse_reset),
(gst_mp3parse_init), (gst_mp3parse_dispose),
(gst_mp3parse_sink_event), (gst_mp3parse_chain), (head_check),
(gst_mp3parse_change_state):
* gst/mpegaudioparse/gstmpegaudioparse.h:
Make timestamp handling in mp3parse saner; now works for at least
simple cases.
Original commit message from CVS:
* gst/mpegaudioparse/Makefile.am:
* gst/mpegaudioparse/gstmpegaudioparse.c:
(mp3_type_frame_length_from_header), (gst_mp3parse_class_init),
(gst_mp3parse_reset), (gst_mp3parse_init), (gst_mp3parse_dispose),
(gst_mp3parse_sink_event), (gst_mp3parse_chain), (head_check),
(gst_mp3parse_change_state), (plugin_init):
* gst/mpegaudioparse/gstmpegaudioparse.h:
Bring mp3parse into the 21st century.
Use its own debug category, use gstadapter, format nicely to 80
columns, and fix incorrect handling of 32 kHz and less files.
Original commit message from CVS:
* ext/lame/Makefile.am:
* ext/mpeg2dec/Makefile.am:
* gst/dvdlpcmdec/Makefile.am:
* gst/dvdsub/Makefile.am:
* gst/mpegaudioparse/Makefile.am:
Add missing GST_LIBS to the link flags
Original commit message from CVS:
2006-03-11 Christophe Fergeau <teuf@gnome.org>
Reviewed by: Tim-Philipp Müller <tim at centricular dot net>
* configure.ac:
* gst/xingheader/Makefile.am:
* gst/xingheader/gstxingmux.c:
* gst/xingheader/gstxingmux.h: added new element to add Xing headers
to MP3 files (this allows decoder to figure out the length of VBR
files)
Original commit message from CVS:
* gst/mpegaudioparse/gstmpegaudioparse.c:
(gst_mp3parse_sink_event), (gst_mp3parse_chain):
Set correct caps on buffers too.
Original commit message from CVS:
* gst/mpegaudioparse/gstmpegaudioparse.c: (gst_mp3parse_init),
(gst_mp3parse_sink_event), (gst_mp3parse_chain):
* gst/mpegaudioparse/gstmpegaudioparse.h:
Put timestamps on buffers.
Original commit message from CVS:
2004-01-25 Ronald Bultje <rbultje@ronald.bitfreak.net>
* gst-libs/gst/riff/riff-read.c: (gst_riff_read_info):
Additional pad usability check.
* gst/mpeg1videoparse/gstmp1videoparse.c: (gst_mp1videoparse_init),
(mp1videoparse_find_next_gop), (gst_mp1videoparse_time_code),
(gst_mp1videoparse_real_chain):
Fix MPEG video stream parsing. The original plugin had several
issues, including not timestamping streams where the source was
not timestamped (this happens with PTS values in mpeg system
streams, but MPEG video is also a valid stream on its own so
that needs timestamps too). We use the display time code for that
for now. Also, if one incoming buffer contains multiple valid
frames, we push them all on correctly now, including proper EOS
handling. Lastly, several potential segfaults were fixed, and we
properly sync on new sequence/gop headers to include them in next,
not previous frames (since they're header for the next frame, not
the previous). Also see #119206.
* gst/mpegaudioparse/gstmpegaudioparse.c: (gst_mp3parse_chain),
(bpf_from_header):
Move caps setting so we only do it after finding several valid
MPEG-1 fraes sequentially, not right after the first one (which
might be coincidental).
* gst/typefind/gsttypefindfunctions.c: (mpeg1_sys_type_find),
(mpeg_video_type_find), (mpeg_video_stream_type_find),
(plugin_init):
Add unsynced MPEG video stream typefinding, and change some
probability values so we detect streams rightly. The idea is as
follows: I can have an unsynced system stream which contains
video. In the current code, I would randomly get a type for either
system or video stream type found, because the probabilities are
being calculated rather randomly. I now use fixed values, so we
always prefer system stream if that was found (and that is how it
should be). If no system stream was found, we can still identity
the stream as video-only.
