Currently, we have a AV_CODEC_ID_SSA, which matches the way the ASS/SSA
markup is muxed in a standalone .ass/.ssa file. This means the AVPacket
data starts with a "Dialogue:" string, followed by a timing information
(start and end of the event as string) and a trailing CRLF after each
line. One packet can contain several lines. We'll refer to this layout
as "SSA" or "SSA lines".
In matroska, this markup is not stored as such: it has no "Dialogue:"
prefix, it contains a ReadOrder field, the timing information is not in
the payload, and it doesn't contain the trailing CRLF. See [1] for more
info. We'll refer to this layout as "ASS".
Since we have only one common codec for both formats, the matroska
demuxer is constructing an AVPacket following the "SSA lines" format.
This causes several problems, so it was decided to change this into
clean ASS packets.
Some insight about what is changed or unchanged in this commit:
CODECS
------
- the decoding process still writes "SSA lines" markup inside the ass
fields of the subtitles rectangles (sub->rects[n]->ass), which is
still the current common way of representing decoded subtitles
markup. It is meant to change later.
- new ASS codec id: AV_CODEC_ID_ASS (which is different from the
legacy AV_CODEC_ID_SSA)
- lavc/assdec: the "ass" decoder is renamed into "ssa" (instead of
"ass") for consistency with the codec id and allows to add a real
ass decoder. This ass decoder receives clean ASS lines (so it starts
with a ReadOrder, is followed by the Layer, etc). We make sure this
is decoded properly in a new ass-line rectangle of the decoded
subtitles (the ssa decoder OTOH is doing a simple straightforward
copy). Using the packet timing instead of data string makes sure the
ass-line now contains the appropriate timing.
- lavc/assenc: just like the ass decoder, the "ssa" encoder is renamed
into "ssa" (instead of "ass") for consistency with the codec id, and
allows to add a real "ass" encoder.
One important thing about this encoder is that it only supports one
ass rectangle: we could have put several dialogue events in the
AVPacket (separated by a \0 for instance) but this would have cause
trouble for the muxer which needs not only the start time, but also
the duration: typically, you have merged events with the same start
time (stored in the AVPacket->pts) but a different duration. At the
moment, only the matroska do the merge with the SSA-line codec.
We will need to make sure all the decoders in the future can't add
more than one rectangle (and only one Dialogue line in it
obviously).
FORMATS
-------
- lavf/assenc: the .ass/.ssa muxer can take both SSA and ASS packets.
In the case of ASS packets as input, it adds the timing based on the
AVPacket pts and duration, and mux it with "Dialogue:", trailing
CRLF, etc.
- lavf/assdec: unchanged; it currently still only outputs SSA-lines
packets.
- lavf/mkv: the demuxer can now output ASS packets without the need of
any "SSA-lines" reconstruction hack. It will become the default at
next libavformat bump, and the SSA support will be dropped from the
demuxer. The muxer can take ASS packets since it's muxed normally,
and still supports the old SSA packets. All the SSA support and
hacks in Matroska code will be dropped at next lavf bump.
[1]: http://www.matroska.org/technical/specs/subtitles/ssa.html
Avoid to write more than one cuepoint per track and PTS in
mkv_write_cues(). This avoids a later assertion failure on "(bytes >=
needed_bytes)" in put_ebml_num() called from end_ebml_master(), in case
there are several cuepoints per track with the same PTS.
This may happen with files containing packets with duplicated PTS in the
same track.
This reverts 312645e :
"Do not set codec_tag property for matroska muxers."
Also adds dummy codec_tag lists with codecs
supported in mkv but not in wav / avi.
Fixes ticket #2169.
This fixes playback in some circumstances (like webm in firefox).
Regression after 2c34367b.
It is also matching the Matroska specifications:
http://matroska.org/technical/specs/notes.html, "The quick eye will
notice that if a Cluster's Timecode is set to zero, it is possible to
have Blocks with a negative Raw Timecode. Blocks with a negative Raw
Timecode are not valid."
* commit 'b94e4acb4874843e914fd3cb8e089aff0756bb4a':
cmdutils_read_file: increment *size after writing the trailing \0
af_resample: unref out_buf when avresample_convert returns 0
af_amix: prevent memory leak on error path
vc1dec: prevent memory leak in error path
vc1dec: prevent memory leak on av_realloc error
af_channelmap: free old extended_data on reallocation
avconv: simplify memory allocation in copy_chapters
matroskaenc: check cue point validity before reallocation
swfenc: error out for more than 1 audio or video stream
build: link test programs only against static libs
Conflicts:
ffmpeg_opt.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '117d8c6d1f1c187ffc6098d9618457e00534e013':
matroska: implement support for ProRes
matroska: implement support for ALAC
Conflicts:
libavformat/matroskaenc.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Support Matroska native formatting.
On demuxing prepend a Frame container atom (32bit big endian encoded
frame size and 'icpf' string).
On muxing remove it.
* commit '3f7fd59d151a2773f0e2e93e56b6b13ec6e5334b':
avformat: fix typo in avformat_close_input
mp3enc: write Xing TOC
mp3enc: support MPEG-2 and MPEG-2.5 in Xing header.
mp3enc: downgrade some errors in writing Xing frame to warnings
lavf: flush the output AVIOContext in av_write_trailer().
lavf: cosmetics, reformat av_write_trailer().
avio: flush the internal buffer in avio_close()
Enhance doc on asyncts audiofilter
cmdutils: avoid setting data pointers to invalid values in alloc_buffer()
libavcodec: remove av_destruct_packet_nofree()
Conflicts:
libavcodec/avpacket.c
libavformat/mp3enc.c
libavformat/nutenc.c
libavformat/utils.c
libavformat/version.h
tests/ref/lavf/voc
tests/ref/lavf/voc_s16
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This is consistent with stdio and is what we want to do in all cases.
Fixes a bug in the voc muxer which didn't flush in write_trailer()
previously. This is the cause of the change in the test results.
After much discussion and back-and-forth, we reached the conclusion
that matroska uses convergence_duration for subtitle duration because
a 32bit value isn't large enough to store the duration if sub-micro-second
timebases are used. Matroska may not be the only one that supports these
timebases, but it's certainly the only one that ffmpeg attempts to support
in this way.
The long term solution that we seemed to reach was that if we encounter
a matroska file with a sub-micro-second timebase, we should internally
scale it up to at least micro-second, and then duration can be used
normally. This suggests that on the encode side, we should not allow
generation of files with sub-micro-second timebases, but that's a separate
issue.
That being a non-trivial change, and the subtitle interoperability breakage
being very real, I'm re-submitting this small change for consideration.
In this diff, we make sure that duration is populated by the matroska
demuxer, and that convergence_duration is respected in matroskaenc and
srtenc, but that duration is used otherwise. This ends up being a strict
improvement - pipelines that use convergence duration are unchanged, and
ones that are currently broken due to the duration mismatch will start
working - except for the ones with the extreme timebases, but those were
already broken.
Signed-off-by: Philip Langdale <philipl@overt.org>
* qatar/master:
avformat: Drop pointless "format" from container long names
swscale: bury one more piece of inline asm under HAVE_INLINE_ASM.
wv: K&R formatting cosmetics
configure: Add missing descriptions to help output
h264_ps: declare array of colorspace strings on its own line.
fate: amix: specify f32 sample format for comparison
tiny_psnr: support 32-bit float samples
eamad/eatgq/eatqi: call special EA IDCT directly
eamad: remove use of MpegEncContext
mpegvideo: remove unnecessary inclusions of faandct.h
af_asyncts: avoid overflow in out_size with large delta values
af_asyncts: add first_pts option
Conflicts:
configure
libavcodec/eamad.c
libavcodec/h264_ps.c
libavformat/crcenc.c
libavformat/ffmdec.c
libavformat/ffmenc.c
libavformat/framecrcenc.c
libavformat/md5enc.c
libavformat/nutdec.c
libavformat/rawenc.c
libavformat/yuv4mpeg.c
tests/tiny_psnr.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
v410dec: Implement explode mode support
zerocodec: fix direct rendering.
wav: init st to NULL to avoid a false-positive warning.
wavpack: set bits_per_raw_sample for S32 samples to properly identify 24-bit
h264: refactor NAL decode loop
RTMPTE protocol support
RTMPE protocol support
rtmp: Add ff_rtmp_calc_digest_pos()
rtmp: Rename rtmp_calc_digest to ff_rtmp_calc_digest and make it global
swscale: add missing HAVE_INLINE_ASM check.
lavfi: place x86 inline assembly under HAVE_INLINE_ASM.
vc1: Add a test for interlaced field pictures
swscale: Mark all init functions as av_cold
swscale: x86: Drop pointless _mmx suffix from filenames
lavf: use conditional notation for default codec in muxer declarations.
swscale: place inline assembly bilinear scaler under HAVE_INLINE_ASM.
dsputil: ppc: cosmetics: pretty-print
dsputil: x86: add SHUFFLE_MASK_W macro
configure: respect CC_O setting in check_cc
Conflicts:
Changelog
configure
libavcodec/v410dec.c
libavcodec/zerocodec.c
libavformat/asfenc.c
libavformat/version.h
libswscale/utils.c
libswscale/x86/swscale.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Also add missing trailing commas, break long codec_tag lines and
add spaces in codec_tag declarations.
Signed-off-by: Martin Storsjö <martin@martin.st>
This avoids problems
where avio_tell() returns 0. I've updated all the checks against
cluster_pos
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Prefix the functions/tables brktimegm, pcm_read_seek,
dv_offset_reset, voc_get_packet, codec_movaudio_tags,
codec_movvideo_tags.
After this, lavf has no global symbols without the proper prefix.
Signed-off-by: Martin Storsjö <martin@martin.st>
* qatar/master:
x86: cabac: replace explicit memory references with "m" operands
avplay: don't request a stereo downmix
wmapro: use av_float2int()
lavc: avoid invalid memcpy() in avcodec_default_release_buffer()
lavu: replace int/float punning functions
lavfi: install libavfilter/vsrc_buffer.h
Remove extraneous semicolons
sdp: Restore the original mp4 format h264 extradata if converted
rtpenc: Add support for mp4 format h264
rtpenc: Simplify code by introducing a separate end pointer
movenc: Use the actual converted sample for RTP hinting
Fix a bunch of common typos.
Conflicts:
doc/developer.texi
doc/eval.texi
doc/filters.texi
doc/protocols.texi
ffmpeg.c
ffplay.c
libavcodec/mpegvideo.h
libavcodec/x86/cabac.h
libavfilter/Makefile
libavformat/avformat.h
libavformat/cafdec.c
libavformat/flvdec.c
libavformat/flvenc.c
libavformat/gxfenc.c
libavformat/img2.c
libavformat/movenc.c
libavformat/mpegts.c
libavformat/rtpenc_h264.c
libavformat/utils.c
libavformat/wtv.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
The existing functions defined in intfloat_readwrite.[ch] are
both slow and incorrect (infinities are not handled).
This introduces a new header with fast, inline conversion
functions using direct union punning assuming an IEEE-754
system, an assumption already made throughout the code.
The one use of Intel/Motorola extended 80-bit format is
replaced by simpler code sufficient under the present
constraints (positive normal values).
The old functions are marked deprecated and retained for
compatibility.
Signed-off-by: Mans Rullgard <mans@mansr.com>
* qatar/master:
aac_latm: reconfigure decoder on audio specific config changes
latmdec: fix audio specific config parsing
Add avcodec_decode_audio4().
avcodec: change number of plane pointers from 4 to 8 at next major bump.
Update developers documentation with coding conventions.
svq1dec: avoid undefined get_bits(0) call
ARM: h264dsp_neon cosmetics
ARM: make some NEON macros reusable
Do not memcpy raw video frames when using null muxer
fate: update asf seektest
vp8: flush buffers on size changes.
doc: improve general documentation for MacOSX
asf: use packet dts as approximation of pts
asf: do not call av_read_frame
rtsp: Initialize the media_type_mask in the rtp guessing demuxer
Cleaned up alacenc.c
Conflicts:
doc/APIchanges
doc/developer.texi
libavcodec/8svx.c
libavcodec/aacdec.c
libavcodec/ac3dec.c
libavcodec/avcodec.h
libavcodec/nellymoserdec.c
libavcodec/tta.c
libavcodec/utils.c
libavcodec/version.h
libavcodec/wmadec.c
libavformat/asfdec.c
tests/ref/seek/lavf_asf
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Pass the correct size in bits to mpeg4audio_get_config and add a flag
to disable parsing of the sync extension when the size is not known.
Latm with AudioMuxVersion 0 does not specify the size of the audio
specific config. Data after the audio specific config can be
misinterpreted as sync extension resulting in random and wrong configs.
