Compute individual stream durations in matroska muxer.
Write them as string tags in the same format as mkvmerge tool does.
Signed-off-by: Sasi Inguva <isasi@google.com>
* commit 'a0797950527120c85263c910eb6ba08fddcfdcb3':
fate/mp3: specify the number of output samples instead of filesize
Merged-by: Michael Niedermayer <michael@niedermayer.cc>
* commit 'bf0cef5c3a114df452e5476167634dd8f51eb448':
checkasm: Include io.h for isatty, if available
Merged-by: Michael Niedermayer <michael@niedermayer.cc>
configure does check for isatty, and checkasm properly checks
HAVE_ISATTY, but on some platforms (e.g. WinRT), io.h needs to be
included for isatty to be available.
Signed-off-by: Martin Storsjö <martin@martin.st>
Also replace custom tests for MD5 with those published in RFC 2202
Signed-off-by: James Almer <jamrial@gmail.com>
Signed-off-by: Martin Storsjö <martin@martin.st>
Avoid clipping due to quantization noise to produce audible
artifacts, by detecting near-clipping signals and both attenuating
them a little and encoding escape-encoded bands (usually the
loudest) rounding towards zero instead of nearest, which tends to
decrease overall energy and thus clipping.
Currently fate tests measure numerical error so this change makes
tests using asynth (which are near clipping) report higher error
not less, because of window attenuation. Yet, they sound better,
not worse (albeit subtle, other samples aren't subtle at all).
Only measuring psychoacoustically weighted error would make for
a representative test, so that will be left for a future patch.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
The test file they use needs avdevice to be created
Probably fixes Ticket 4455
Reviewed-by: James Almer <jamrial@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
* commit 'e605bf3b590d295f215fcc9fd58eb11be55b68cb':
checkasm: remove empty array initializer list in h264pred test
Merged-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes Ticket4664
The changed fate tests lack red/blue shades and thus look correct
either way
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
* commit '82e6ac85ff9aa7631b8c01521b3d6b5ca0bc8014':
checkasm: test all architectures with optimisations
Merged-by: Michael Niedermayer <michael@niedermayer.cc>
* commit '6cc4d3e9a982e926494f4b919d9733fe29774acf':
checkasm: exit with status 0 instead of 1 if there are no tests to perform
Merged-by: Michael Niedermayer <michael@niedermayer.cc>
It provides the following features:
* verify correctness by comparing output to the C version.
* detect failure to save and restore clobbered callee-saved registers.
* detect 32-bit parameters being used as if they were 64-bit in x86-64
(the upper halves are not guaranteed to be zero - but in practice
they very often are, which makes those bugs hard to spot otherwise).
* easy benchmarking.
Compile by running 'make checkasm'.
Execute by running 'tests/checkasm/checkasm'.
Optional arguments are '--bench' to run benchmarks for all functions,
'--bench=<pattern>' to run benchmarks for all functions that starts with
<pattern>, and '<integer>' to seed the PRNG for reproducible results.
Contains unit tests for most h264pred functions to get started, more tests
can be added afterwards using those as a reference.
Loosely based on code from x264. Currently only supports x86 and x86-64,
but additional architectures shouldn't be too much of an obstacle to add.
Note that functions with floating point parameters or floating point
return values are not supported. Some compiler-specific features or
preprocessor hacks would likely be required to add support for that.
Signed-off-by: Janne Grunau <janne-libav@jannau.net>
* commit '02b7c630875c0bc63cee5ec597aa33baf9bf4e20':
h261: Signal freeze picture release for intra frames
Conflicts:
tests/ref/vsynth/vsynth1-h261
tests/ref/vsynth/vsynth2-h261
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Freeze picture release should be set to 1 when we're responding to a
fast update request. For simplicity we set it for all intra frames,
including those that starts a GOP.
Fixes issue where Tandberg MXP1700 does not recover from packet loss
state since it's waiting for the freeze picture relase indication.
Bug-Id: 873
CC: libav-stable@libav.org
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
Ref H.261 recommendation section 4.2.1.3, setting the still image flag
to 1 disables still image mode. Some decoders require this in order to
decode the bitstream as normal video.
Fixes H.261 calls to Cisco E20.
Also, reserved (aka spare) bits should be set to 1 unless specified
otherwise.
Bug-Id: 872
CC: libav-stable@libav.org
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
This change fixes a bug where a test that required a sample was being included
in the suite when SAMPLES was not set. It also improves the consistency of
variable names relating to the API tests.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'f91fe24e9bd6912c29bbb03d8afe878e045f9721':
g2meet: force simple idct for identical results over all fate configs
Conflicts:
tests/ref/fate/g2m3
tests/ref/fate/g2m4
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '4d1229dabf7a7e3b6a7b326afd79102256c3b008':
g2meet: Add FATE tests for all three G2M variants
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Most of the fate-dds-* and fate-txd-* tests already
output into the same pixel format regardless of
platform endianness, so there's no need to force
conversion to another format.
This fixes the tests fate-txd-16bpp, fate-txd-odd,
fate-dds-rgb16, fate-dds-rgb24 and fate-dds-xrgb on
big endian, where the tests seem to fail due to issues
with certain conversion codepaths in swscale.
Those conversion codepaths should of course be fixed, but
the individual decoder tests should use as little extra
conversion steps as possible.
Signed-off-by: Martin Storsjö <martin@martin.st>
* 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>
