* commit '3ad678a85b96fc5fecd60e3d3a31ca5ffc89d67f':
fate: Update ac3 test to the new request_channel_layout option
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '441e8ae5efd681055e5af6f4317fb60110de9dd0':
FATE: drop the last truncated frame from the wmapro tests
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'd3ea79e8a65ddad4da11813bb43c46701295f68c':
FATE: drop the last truncated frame from the wma lossless test
Conflicts:
tests/fate/lossless-audio.mak
Merged-by: Michael Niedermayer <michaelni@gmx.at>
The old one is the result of the reverse engineering and guesswork.
The new one has been written following the now-available specification.
This work is part of Outreach Program for Women Summer 2014 activities
for the Libav project.
The fate references had to be changed because the old demuxer truncates
the last frame in some cases, the new one handles it properly.
The seek-test reference is changed because seeking works differently
in the new demuxer. When seeking, the packet is not read from the stream
directly, but it is rather constructed by the demuxer. That is why
position is -1 now in the reference.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
The intention of this change is to allow separation of API tests from the
existing tests, and also to have a place for the API test source/executable
files so they're not mixed in with the actual library code.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Result differs in pkt_duration and time_base.den for some reason.
Right now it tests only one example (adjusted to match the output).
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Ideally this should be discarded by the demuxer but this is not
possible without fully parsing which would be then very similar
to this. The current ID3v1 discard code in the demuxer does not work
and will be removed in a subsequent commit
The discard code could be adjusted if needed to also discard tags at
other locations than the end or to limit this possibly to input
from the mp3 demuxer or even to move the discarding to the
decoder.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'c0b105756f61d253bdabcc2bb49453a2557e7c3b':
txd: Use the TextureDSP module for decoding
Conflicts:
configure
libavcodec/s3tc.c
libavcodec/s3tc.h
libavcodec/txd.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Using the internal DXTC routines brings support for non multiple of 4
textures. A new test is added to cover this feature. Hashes differ
since the decoding algorithm is different, though no visual changes
have been spotted.
Signed-off-by: Vittorio Giovara <vittorio.giovara@gmail.com>
thats how the specification defines it, this also improves numerical
accuracy of the integer wavelet implementation. It otherwise should
be equivalent, in case of overflows this can be reverted.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'c060d046aa2f89c0e601a2dcfbce53f0e36cf498':
af_resample: Set the number of samples in the last frame
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '6ec688e1bc76dd93151cbca1c340162ae4b10d77':
mp3: enable packed main_data decoding in MP4
Conflicts:
libavcodec/mpegaudiodec_template.c
Only the parts needed to support the available sample are merged
the remaining error checks are left in place
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Even if the jpeg2000 spec uses a wrong value this does not
make mathematics work this way, also this has been corrected in the 2004
version AFAIK
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
or if no rematrix and no resampling is performed and the input is 16bit
note reampling and rematrix itself always use more than 16bit internally
the "internal" sampling format is the format between these steps
Its unlikely the difference from this commit is audible in any case
unless there is some bug either before or after the change.
but multiple people prefer this and it slightly improves the precission
of computations.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This is almost certainly closer to how the actual Nintendo players work,
and fixes some output pops in files with blank ADPC/SEEK tables (like
those from brawlcustommusic).
14496-3 suggests packing main_data of MP3 that is usually scattered
into multiple frames due to bit reservoir.
However, after packing main_data into a access unit, bitrate index
in the MPEG audio frame header doesn't match with actual frame size.
In order to accept this, this patch removes unnecessary frame size
checking on mp3 decoder.
Also, mov demuxer was changed to use MP3 parser only on special cases
(QT MOV with specific sample description) to avoid re-packetizing.
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
* commit '063f7467e4d14ab7fe01b2845dab60cc75df8b53':
rtmpdh: Add fate test for the DH handshake routine
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Previous version Reviewed-by: tim nicholson <nichot20@yahoo.com>
Previous version Reviewed-by: Tomas Härdin <tomas.hardin@codemill.se>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
The positioning was completely wrong. First, the coordinates are
expressed in ASS playback resolution (which is by default 384x288).
Secondly, the coordinates define a drawing rectangle, not a moving area.
The previous code was making subtitles move from a random position to
another random position.
Here we rescale assuming the video resolution is a DVD one (720x480). We
can't really do anything better so far, but since this positioning
information is often from a DVD rip we can consider them relatively
safe.
No real difference in quality, its a bit slower for the same dia_size as more
vectors are searched for the same dia_size
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
AV_PIX_FMT_GRAY8/16 are considered YUV formats, and the color_range is
not set - so the API user will have to assume limitted range. (Unless
the API user adds a special-case for the PNG decoder.)
Just export the correct range - full range.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'd81fb63d87692765c004c19934b49427df434a07':
fate: Add a PICT test
Conflicts:
tests/fate/image.mak
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Remove the direct profile from UTCTiming element. Per DASH spec,
direct profile value should be the time at which the request was
made to the server and not the time at which the manifest was
written. So ffmpeg cannot write this value. This patch removes
the direct profile and write the UTCTiming element with the http
profile only if a URL is passed as a parameter. Update the fate
test to reflect this change.
Signed-off-by: Vignesh Venkatasubramanian <vigneshv@google.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This will test properly CRLF with make fate, make fate-subtitles and any
make fate-sub-* test. Before this commit, the rawdiff was triggered only
by make fate-subtitles.
Also make sure fate-sub-* only match the tests relying on fmtstdout
command, to at least avoid failing on MingW. See
https://ffmpeg.org/pipermail/ffmpeg-devel/2015-April/172395.html
failure to calculate psnr should not result in tiny_psnr returning success
Reviewed-by: Timothy Gu <timothygu99@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
These could be kept, but they are not overly useful. The only thing they
had over the remaining mp3 gapless test was seeking, which was incorrect
in the toc test, and only by chance correct in the notoc test.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
It's the most useful mode, because it seeks accurately, and does not
break features like gapless audio.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
"-usetoc 2" now invokes the generic seek and indexing mode. This mode
skips data until the seek target is reached, and this is exact. It also
makes gapless audio actually work if a seek past the start of the file
is involved.
Change the fate-gapless-mp3 test to use the new mode, and move the old
one to fate-gapless-mp3-toc (since the test forces use of the Xing TOC).
The new mode has a different result for the seek - this result is
actually correct.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
The mp2 seek test results change. Whether to skip samples if the file
had no LAME gapless tags was inconsistent. When seeking to the start
of the file, 529 samples were skipped, but when playing from start,
nothing was skipped. This commit changes the behavior on seek to skip
nothing.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Some players do not support setting minimumUpdatePeriod to zero.
This patch adds a new parameter that will let the users set any
value to this field. Also updates the test and the documentation.
Signed-off-by: Vignesh Venkatasubramanian <vigneshv@google.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
DASH spec requires the presence of either duration of the period
or the minimumUpdatePeriod element. This patch adds the
minimumUpdatePeriod element hardcoded with the value 0 as the
manifest will never be updated for WebM DASH Live streams. Also
updating the fate test reference file.
Signed-off-by: Vignesh Venkatasubramanian <vigneshv@google.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
With this the returned timestamp should match the packet instead of
the requested timestamp, which may lay between packets
Reviewed-by: wm4 <nfxjfg@googlemail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Seeking to a negative time did not have the desired effect of seeking to
the next valid position (the file start). On the other hand, just
"-ss 0" will normally seek to a position higher than 0, because it adds
the start time of the file. (The start time is not 0 because the gapless
code skips a few samples from the start.)
Fix this by using the "-seek_timestamp 1" option, which makes "-ss 0" do
what you'd expect it would do.
Also put the -ss option at the right place, before -i. This actually
makes it seek, instead of something completely else. The ".out-3" test
is no different in the -usetoc 0/1 cases, because the seeking is
inaccurate (in both cases).
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
The FATE server does not report this information anyway and omitting
it makes the successful run send much less data.
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
MicroDVD has a "hack" for specifying the video framerate the subtitle
was authored against. The demuxer reads this hint correctly, but didn't
skip it correctly.
This was not noticed, because the exported packet has its duration set
to 0, making it invisible (depending on the API user's rendering logic).
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This patch adds support for creating DASH manifests for WebM Live
Streams. It also updates the documentation and adds a fate test to
verify the behavior of the new muxer flag.
Signed-off-by: Vignesh Venkatasubramanian <vigneshv@google.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Previously unset, and some software mishandles files if it is absent
Signed-off-by: Tim Nicholson <tim.nicholson@bbc.co.uk>
Reviewed-by: tomas.hardin@codemill.se
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* commit '2889c5e16711770437f380f1bead5f72c6a0b17a':
movenc: Heuristically set the duration of the last sample in a fragment if not set
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'f01c77157789b8e3a59ed2c9646faf8299e41641':
fate: add explicit support for the toolchain configure option
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'acbe15a99f158dbb0edb837fb6557171dc4376d4':
fate: Add test for DCA XLL
Conflicts:
tests/fate/audio.mak
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '46d4d8575979a24a8d026d9805039b724e0e3e5f':
movenc: Avoid writing separate flags for the first sample if not necessary
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '00d751d4fc20ec88d2cc2c9f39ec8b9e9c8cdeba':
movenc: Set tfhd default sample flags based on actual samples, if possible
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Regression test for the bug from trac ticket #4359 fixed in commit efff3854
Reviewed-by: Michael Niedermayer <michaelni@gmx.at>
Signed-off-by: James Almer <jamrial@gmail.com>
* commit '62139b14e621f096d0f8ed90920d042b92867e40':
fate: Specify the idct to use for the aic-oddsize test
Conflicts:
tests/fate/video.mak
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This is a small change, but it does have a big impact on bit allocation.
all the regressions marked in the report have no audible
difference (I didn't check them all though), but the improvements can
be heard.
This affects mostly high bit rates. It's related to issue #2686.
In the report, A is the patched version, B is unpatched, all
comparisons show deltas in the form (A-B), so a positive pSNR delta
means a better quality in the patched version, and negative a
regression. Regressions are only considered for pSNR deltas below
-1db, they're considered serious below -6db.
All measurements were done with tiny_psnr.
The summary of the report inline for quick reading:
Files: 58
Bitrates: 6
Tests: 347
Serious Regressions: 0 (0%)
Regressions: 10 (2%)
Improvements: 54 (15%)
Big improvements: 26 (7%)
Worst regression - sine_tester.flac - 384k
- StdDev: 1.68 pSNR: -3.05 maxdiff: -178.00
Best improvement - 07 - Bound.flac - 384k
- StdDev: -1700.05 pSNR: 20.64 maxdiff: -29595.00
Average - StdDev: -55.67 pSNR: 1.20 maxdiff: -1593.00
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'ce52869c22738ad584995d48103ce3aa2301736b':
fate: Rename fate-dts test to fate-dca-core
Conflicts:
tests/fate/audio.mak
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'a982c5d74fbc7ff5bd2f2f73af61ae48e9b1bcc6':
tests: drop bc dependency
Conflicts:
tests/fate-run.sh
See: d47eeff274
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Outputting DNxHD into .mov containers 'corrupts' following atoms until end of stsd
ffmpeg and qtdump could not decode pasp/colr atoms in the files made by ffmpeg,
when outputting DNxHD due to the incorrect padding placement. Now we add the
padding in the correct place
Tidy up FATE changes due to padding changes.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'e21d85309943a51b7808f5e01dd258b262e09148':
FATE: add a test for the SVQ1 header byte swapping
Conflicts:
tests/fate/qt.mak
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Avid prefers mpeg range [16-235] by default this change brings
ffmpeg into line with that. To obtain the old behaviour use
'-color_range jpeg' on the command line prior to the ouput
filename.
Signed-off-by: Kevin Wheatley <kevin.j.wheatley@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This is not sufficient to run "make fate-ffprobe" on a remote system:
The ffprobe output contains the relative path to the testfile, it is
necessary to run the test from the build directory.
One solution is to use a script like the following as --target-exec:
ssh target "cd /remote/build/directory; $(printf "%q " "$@")"
This is a bit ugly as it attempts to keep most of the computation
in integers before the double based fps code. The use of integers
is to reduce the chances of rounding differences between platforms
Previously the timestamp was rounded to the encoder timebase
before being converted back to double precision which could cause loss
of precision
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
The tests which use encoders which either use slices or store the encoder thread count
keep a hardcoded value of 1
This will help test more threading code like in filters
Found-by: ubitux
Reviewed-by: Clément Bœsch <u@pkh.me>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* commit '50036c30df83b609bc5a95276f1287f8b9b8bdd6':
fate: Use bitexact conversions in the dpxparser test
Conflicts:
tests/fate/image.mak
Merged-by: Michael Niedermayer <michaelni@gmx.at>
The existing meridian audio test does not test
ff_mlp_rematrix_channel_arm. This sample (first 640k of
https://samples.libav.org/A-codecs/TrueHD/TrueHD.raw) uses
ff_mlp_rematrix_channel_arm. Since this sample has 5.1 channels it also
allows testing the integrated downmixing.
This uses the RIFF header stored size to figure out the expected AVI
file size, instead of the actual file. To work fully it requires handling
failed avio_seek() instead of assuming they always succeed.
Some fate file has been cut off and contains half a frame at the end which
previously was not output during demuxing. This frame is now output to
encoder, thus the fate diff update.
Bug-Id: 261
Signed-off-by: Vittorio Giovara <vittorio.giovara@gmail.com>
According to the DPX file format description found at
http://www.fileformat.info/format/dpx/egff.htm the ImageElement part of
the GenericImageHeader also contains an an offset to the real image data
beside the same member that can be found in the GenericFileHeader.
Libav keeps this member empty (=0) while some applications expects it to
be filled properly. FATE test updated accordingly.
Bug-Id: 742
Signed-off-by: Vittorio Giovara <vittorio.giovara@gmail.com>
Only shift limited range luma, and always only shift chroma
for upconversion.
Based off a patch by Michael Niedermayer.
Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
The file is already present in git and by using it we can perform more tests
without the need of fate samples
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
The new reference.pnm is a freely licensed replacement. The photo has
been taken by Reinhard Tartler on August 28 2014, and is licensed under
the expat license as stated at http://www.jclark.com/xml/copying.txt
* commit '9257692ac15eff7b07540c1f61cebde0d8823fbd':
lavf: Only initialize s->offset once when using avoid_negative_ts make_zero
Conflicts:
libavformat/mux.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This is a generic solution that will not reqiore modifications when new options are added.
This also fixes problem with current implementation when qmin or qmax=-1.
Only 8 bits was sent and read back as 255.
Fixes#1275Fixes#1461
Signed-off-by: Lukasz Marek <lukasz.m.luki2@gmail.com>
Function allows to create string containing object's serialized options.
Such string may be passed back to av_set_options_string() in order to restore options.
Signed-off-by: Lukasz Marek <lukasz.m.luki2@gmail.com>
