Uses a similar approach as vf_yadif to flush the last frame in idet.
Quick test with 50 frames from vsynth1:
./ffmpeg.old -i fate-suite/ffmpeg-synthetic/vsynth1/%02d.pgm -vf idet -f mp4 -y /dev/null 2>&1 | grep Multi
(gives) [Parsed_idet_0 @ 0x261ebb0] Multi frame detection: TFF:0 BFF:0 Progressive:48 Undetermined:1
./ffmpeg -i fate-suite/ffmpeg-synthetic/vsynth1/%02d.pgm -vf idet -f mp4 -y /dev/null 2>&1 | grep Multi
(gives) [Parsed_idet_0 @ 0x35a0bb0] Multi frame detection: TFF:0 BFF:0 Progressive:49 Undetermined:1
Fate tests have been updated.
(In testing, it seems this filter will also need a subsequent patch for single frame input)
Signed-off-by: Neil Birkbeck <neil.birkbeck@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Regression test for the bug from trac ticket #3849 fixed in commit 14e30255
Reviewed-by: Ronald S. Bultje <rsbultje@gmail.com>
Signed-off-by: James Almer <jamrial@gmail.com>
In these cases, only drop dts. Because if we drop both we have no
timestamps at all for some files.
This improves playback of HLS streams from GoPro cameras.
Signed-off-by: Martin Storsjö <martin@martin.st>
Width, Height and Sample Rate should be in the AdaptationSet tag
only if all the contained representations have the same width,
height and sampling rate. Otherwise they should go into the
Representation tag. This patch adds this functionality and a fate
test for the same.
Signed-off-by: Vignesh Venkatasubramanian <vigneshv@google.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Fix an incorrect hard code in cues_end computation. Updating the fate
test reference files related to the fix as well. The earlier computation
was clearly wrong as the cues_end field was greater than the file size
itself in some cases.
Signed-off-by: Vignesh Venkatasubramanian <vigneshv@google.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This also un-does the fate changes from a52f443714,
leaving this fix without even small differences in the output, that is
a sample for which this makes a vissible difference is very welcome
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Fixes CID1194380
There are no vissible differences in the changed fate samples. Only
a tiny number of pixels change by tiny amounts in the frames i checked
If someone has a file that shows a vissible difference, please post it.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'b39ebcddd47daf37659796aaa7d068668086507a':
fate: Add VC-1 interlaced twomv test
Note, this test is not free of artifacts on both sides of the merge
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '28f5cd312c9da9072108edf8b7685d009374ea96':
fate: Switch ra4-288 test from framecrc() to pcm()
Conflicts:
tests/fate/real.mak
The test is kept disabled as it still does not pass on x86-64 due to float
rounding
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This may make fate failures where only the console output is available
easier to analyze
Suggested-by: Andreas Cadhalpun
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
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>
Comming from commit 1013d8dd69
the old Port and BindAddress config vars have been deprecated
in favor of HTTPPort and HTTPBindAddress respectively.
Signed-off-by: Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
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 'd395895cdb2ac8c95bd488549e7f893bd4dcc248':
fate: generate tests/pixfmts.mak for all targets requiring it
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'eee813eec7d3c0b0689f80665d3f796401742935':
fate: Only generate tests/pixfmts.mak if some pixfmts fate test is run
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '23dfa00b88fc927d4c1854ab4fc60f5c6398f3ac':
fate: explicitly set the default THREADS value
Conflicts:
tests/Makefile
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This makes the default of '1' more explicit than defaulting to '1' in
fate-run.sh and regression-funcs.sh if THREADS is not set.
Fixes the reported thread count in fate-cpu if THREADS is not set.
* commit '07d8fa58121be8fe315bd51ab760547fe209a745':
fate: add informative cpu test
Conflicts:
tests/fate/libavutil.mak
Merged-by: Michael Niedermayer <michaelni@gmx.at>
libavutil/cpu-test prints raw and effective cpu flags to STDERR. Detected
cpu flags can be useful for debugging fate errors.
No comparison of the result against a expected result since that would
require fate config specific references.
* 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>
Fixes fate on haiku, where cat dies due to too many arguments
xargs could be used too but we do not use xargs currently so it
would be an additional dependency.
Also the plain cat is left in place as it is faster than the loop
Signed-off-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>
* commit '42eb9154a83e9a7aedb1168b2f1112af765cf2b5':
fate: support testing of release branches
Conflicts:
tests/fate.sh
The communication protocol is left at version 0 as our fate server
hasnt been updated to support this yet
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Adding 'branch=release/10' to the fate config file will check the
release/10 branch instead of master. If no branch is specified it will
use 'master' so that existing config are still valid.
The server side changes are already deployed, see
https://fate.libav.org/v10/ for an example. The server supports only the
release/* branches.
The server enforces that a single slot tests always the same branch.
Please append "-v$RELEASE" to the slot of release branch configs or make
the slot otherwise unique.
A different fate samples dir is needed for each release branch. make
fate-rsync has the correct URL in each branch.
* commit '16b7328058fa600d5158c84d9cc621a134eb88bc':
build: Conditionally build and run DCT test program
Conflicts:
libavcodec/Makefile
libavcodec/dct-test.c
tests/fate/libavcodec.mak
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'bd499d9af668aef979ec9f3f3215b8dd508c7ec1':
build: Conditionally build and test iirfilter
Conflicts:
libavcodec/Makefile
Merged-by: Michael Niedermayer <michaelni@gmx.at>
The order of error codes will be useful in my future fateserver patches.
Signed-off-by: Timothy Gu <timothygu99@gmail.com>
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 prevents all results from being declared whenever the function is called.
Signed-off-by: Timothy Gu <timothygu99@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This only checks that things havnt changed, the values provide little
help in determining if a change is good or bad.
Improvements welcome!
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This is the same as 5a15602a4e, which
accidentally did not get merged.
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.
The official samples are 50% smaller
Avoid having reference samples which are strongly linked to the used resampler
implementation. (which for example would require new samples to be used if this
implementation changes)
Also its more correct to use the official samples instead of the current
decoder output
also enable tests
The tests also fully pass as well with the previous samples.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
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 'b70d7a4ac72d23f3448f3b08b770fdf5f57de222':
lavc: add a native Opus decoder.
Conflicts:
Changelog
configure
libavcodec/version.h
Fate tests pass with both avresample as well as swresample based opus decoder, but
are disabled (reference files are very large so i want to think a day or 2 about
if theres an alternative or if they could be avoided, they also dont match the
official samples)
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Use it instead of checking CODEC_FLAG_BITEXACT in the first stream's
codec context.
Using codec options inside lavf is fragile and can easily break when the
muxing codec context is not the encoding context.
Initial implementation by Andrew D'Addesio <modchipv12@gmail.com> during
GSoC 2012.
Completion by Anton Khirnov <anton@khirnov.net>, sponsored by the
Mozilla Corporation.
Further contributions by:
Christophe Gisquet <christophe.gisquet@gmail.com>
Janne Grunau <janne-libav@jannau.net>
Luca Barbato <lu_zero@gentoo.org>
* 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>
* commit '911fa05b514e1be009e00b79d7004b93717c023b':
mvc: Specify the pixel format for the mv-mvc* tests
Conflicts:
tests/fate/video.mak
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Also set the RGBA pixel format correctly as the native endian format,
which is what it returns.
This fixes the tests on big endian.
Signed-off-by: Martin Storsjö <martin@martin.st>
This should avoid slight differences in the output causes by input
size alignment differences between archs
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Fixes conversion of pal8 to rgb formats with alpha.
Updated references for 2 FATE tests which previously encoded fully
transparent images.
Based on a patch by Baptiste Coudurier <baptiste.coudurier@gmail.com>
If 384k is too high for the samplerate, choose the closest
possible
Idea to increase the bitrate from: 46439e1562
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This causes us to favor RGB8 over PAL8 when FF_LOSS_COLORQUANT is used
It probably makes sense to reinvestigate the exact scoring of pal8 when
our pal8 support improves to be supperior to rgb8
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* commit '60fd7d36c47d62d4c603bf16c213b1a924f5cfcf':
fate: correctly set sample rate for mp2 tests
Conflicts:
tests/fate/acodec.mak
tests/lavf-regression.sh
one hunk has been ommited as it breaks fate
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '92b099daf4b8ef93513e38b43899cb8458a2fde3':
swscale: support converting YVYU422 pixel format
Conflicts:
libswscale/input.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* cus/stable:
mpeg12enc: always set closed gop flag on the first gop
mpeg12enc: always write closed gops for intra only outputs
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '66d04c068a30751750818dcfbb6555ab74eb3f6d':
fate: Explicitly use gray16le in fate-sgi-gray16
Conflicts:
tests/fate/image.mak
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Improves compatibility with XDCAM HD formats. It has been set for a long time
in ffmbc.
Reviewed-by: Michael Niedermayer <michaelni@gmx.at>
Signed-off-by: Marton Balint <cus@passwd.hu>
* commit '3e4e2142d246699a1a3a0045ba7124b18bc34d7a':
fate: Convert the paletted output in the brenderpix tests to rgb24
Merged-by: Michael Niedermayer <michaelni@gmx.at>
