Fixes Ticket3901
the seek test error codes change due to a change in the failure path,
this could be avoided by changing the respective error codes to EINVAL
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Add fate tests that test out the functionality of WebM DASH
Manifest XML generation. This patch contains the vpx.mak file
changes and the reference gold XML files.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'b263f8ffe7599d9cd27ec477a12700da8eb2790d':
lavf: add AVFormatContext.max_ts_probe
Conflicts:
doc/APIchanges
libavformat/avformat.h
libavformat/utils.c
libavformat/version.h
lavf-fate/mp3 changes as the estimated input bitrate changes and that is
copied to the output
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Fixes assertion failure
Fixes Ticket3822
as a side-effect this makes some mkv files a few bytes smaller
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
The arrays are fairly large and could cause problems on some embedded systems
also they are not endian safe as they mix 32 and 8bit
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* commit '706208ef47bffd525c982975d2756f7b2b220b8d':
fate: Split fate-pixdesc tests and dispatch them through Make
Conflicts:
tests/fate-run.sh
tests/ref/fate/filter-pixdesc
Merged-by: Michael Niedermayer <michaelni@gmx.at>
- all of them testing HEVC version 1
cherry picked from commit adcdabb4dd062694fb8de6df0faecaad1c36ba33
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Normally, a Laplace distribution is more typical of the residuals
encoded, but for noisy input, it's both better and simpler to be
safe and use a 1/d^2 distribution. Second hunk could use some
renormalization but it has effectively little impact.
Output size of ffvhuff on various 4:2:0 sequences:
context=0,1/d: 851974 27226 1137281
context=0,1/d²: 619081 25069 1051500
context=0,1/d³: 501983 30454 1290561
context=0,lapl: 500650 31754 1304082
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* commit '194be1f43ea391eb986732707435176e579265aa':
lavf: switch to AVStream.time_base as the hint for the muxer timebase
Conflicts:
doc/APIchanges
libavformat/filmstripenc.c
libavformat/movenc.c
libavformat/mxfenc.c
libavformat/oggenc.c
libavformat/swf.h
libavformat/version.h
tests/ref/lavf/mkv
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This avoids the following libass warning when using the subtitles
filter: "Neither PlayResX nor PlayResY defined. Assuming 384x288"
Subtitles tests change because the output is ASS and the PlayRes[XY]
ends up in the output.
Previously, AVStream.codec.time_base was used for that purpose, which
was quite confusing for the callers. This change also opens the path for
removing AVStream.codec.
The change in the lavf-mkv test is due to the native timebase (1/1000)
being used instead of the default one (1/90000), so the packets are now
sent to the crc muxer in the same order in which they are demuxed
(previously some of them got reordered because of inexact timestamp
conversion).
It has not been properly maintained for years and there is little hope
of that changing in the future.
It appears simpler to write a new replacement from scratch than
unbreaking it.
This very slightly improves compression
Found-by: Christophe Gisquet <christophe.gisquet@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
The actual predictor value, set by the trellis code, never
was written back into the variable that was written into
the block header. This was accidentally removed in b304244b.
This significantly improves the audio quality of the trellis
case, which was plain broken since b304244b.
Encoding IMA QT with trellis still actually gives a slightly
worse quality than without trellis, since the trellis encoder
doesn't use the exact same way of rounding as in
adpcm_ima_qt_compress_sample and adpcm_ima_qt_expand_nibble.
Fixes part of Ticket3701
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
adpcm_ima_qt does not produce reproducible results, so it is temporarily
disabled (see #3701).
Signed-off-by: Timothy Gu <timothygu99@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This results in DefaultDuration not being written when the framerate is
not known, but as this field is purely informative, this should not
break any sane demuxers.
This corrects the bug that caused the checksums to change in
9767d7c092.
It caused the EOS flag to be set incorrectly; the ogg spec does not
allow it to be set in the middle of a logical bitstream.
Signed-off-by: Andrew Kelley <superjoe30@gmail.com>
Signed-off-by: Martin Storsjö <martin@martin.st>
Before, header information for ogg format files was sent with the
first encoded packet.
This patch makes it so that it is possible for API users to
differentiate between headers and encoded audio. This is useful, for
example, when creating an audio stream where you want to send one set
of headers for every client that connects and then the encoded stream
of audio.
Signed-off-by: Martin Storsjö <martin@martin.st>
Based off the srt encoder. The following features are unimplemented:
- fonts, colors, sizes
- alignment and positioning
The rest works well. For example, use ffmpeg to convert subtitles into the .vtt format:
ffmpeg -i input.srt output.vtt
Signed-off-by: Aman Gupta <ffmpeg@tmm1.net>
Signed-off-by: Clément Bœsch <u@pkh.me>
* commit '6656370b858329ca07a60a2de954d5e90daa0206':
avconv: set the "encoder" tag when transcoding
Conflicts:
ffmpeg.c
tests/ref/lavf/mkv
tests/ref/seek/lavf-mkv
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '93afb6c98df876b15e3d911a9450ad55f92080ce':
avconv: set output avg_frame_rate when known
Conflicts:
ffmpeg.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '6072184e702b4b631ac72f1b66b75e5f21e0ce2d':
asfenc: use codec descriptors instead of AVCodecs to write codec info
Conflicts:
tests/ref/lavf/asf
tests/ref/seek/lavf-asf
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Also, stop using AVCodecContext.codec_name as fallback, since it will be
deprecated.
Changes the result of the lavf-asf test (and its associated seektest),
since 'msmpeg4v3' gets written instead of just 'msmpeg4'.
Partially undoes commit 2c4e08d893:
riff: always generate a proper WAVEFORMATEX structure in
ff_put_wav_header
A new flag, FF_PUT_WAV_HEADER_FORCE_WAVEFORMATEX, is added to force the
use of WAVEFORMATEX rather than PCMWAVEFORMAT even for PCM codecs.
This flag is used in the Matroska muxer (the cause of the original
change) and in the ASF muxer, because the specifications for
these formats indicate explicitly that WAVEFORMATEX should be used.
Muxers for other formats will return to the original behavior of writing
PCMWAVEFORMAT when writing a header for raw PCM.
In particular, this causes raw PCM in WAV to generate the canonical
44-byte header expected by some tools.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
The rational for this is another issue that plex has exposed. When it is
conducting a transcode of video to HLS for streaming, my father noticed
artifacts when played on his GoogleTV (NSZ-GT1). He sent me a test file
and I reproduced it on my device of the same model. It is important to
note that the artifacts were not present when streaming to VLC or QuickTime
Player. I copied the command-line that plex used, and conducted all of the
following tests using FFmpeg git.
Transcode to HLS: artifacts on playback
Transcode to TS: playback is fine
Cat HLS segments into a single TS: playback is fine
Segment single TS file to segments: artifacts on playback
Segment single TS file to segments using Apple's HLS segmenter: playback is
fine
At this point I carefully examined the differences between Apple's HLS
segmenter output and FFmpeg's. Among the considerable differences, I
noticed that the video PES packets always had a 0 length. So I continued:
Transcode to HLS using FFmpeg with 0 length PES packets: playback is fine.
Segment single TS to segments with 0 length PES packets: playback is fine.
All failures mentioned are only on the GTV since it is the only player on
which I could reproduce artifacts. I only tested the GTV, VLC, and
QuickTime Player though, so my test case is limited. I do not know if
other players exhibit this issue.
Since it was useful last time, I have uploaded the test file as
hls_pes_packet_length.m4v along with its associated txt file which contains
the transcode command-line that was used.
Reviewed-by: Kieran Kunhya <kierank@obe.tv>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
