* 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 '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>
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>
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>
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).
* commit '063f7467e4d14ab7fe01b2845dab60cc75df8b53':
rtmpdh: Add fate test for the DH handshake routine
Merged-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.
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
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>
"-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>
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>
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>
* 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 '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 'e21d85309943a51b7808f5e01dd258b262e09148':
FATE: add a test for the SVQ1 header byte swapping
Conflicts:
tests/fate/qt.mak
Merged-by: Michael Niedermayer <michaelni@gmx.at>
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 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>
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>
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>
According to the DASH spec, Representation IDs should be unique
across all adaptation sets. Fixing that and updating the fate
reference file to reflect this change.
Signed-off-by: Vignesh Venkatasubramanian <vigneshv@google.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* commit '00c67fe1d0bc7c2ce49daac9c80ea39d5a663b73':
movenc: Write a 0 duration in mdhd and tkhd for an empty initial moov
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '600d5ee6b12bad144756b0772319bb04796bc528':
movenc: Signal iso6 in compatible_brands when using tfdt
Merged-by: Michael Niedermayer <michaelni@gmx.at>
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>
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>
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>
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.
This very slightly improves compression
Found-by: Christophe Gisquet <christophe.gisquet@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
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>
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>
