The failures on various architectures and compilers on the RGB(A)
tests seem to have been because of one-off YCbCr->RGB conversion
results. This should make the conversion results match on most if
not all code paths.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
Unsurprisingly, if a timing-less subrip decoder is desireable, an
encoder is as well. With this in place, we can move on to remove
the use of the old encoder/decoder with embedded timing and move
all timing handling the (de)muxer where they belong.
Signed-off-by: Philip Langdale <philipl@overt.org>
Fixes Ticket1627
The fate change is due to ffmpeg no longer pushing audio timestamps
aggressively up (which is what caused the AV sync issues in the ticket)
but leaving them as they are.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
The previous code dependent on the input buffer matching the
buffer that has been provided by yadifs get_buffer.
The API does in now way gurantee this though its often true.
This fixes some out of array reads.
The regression test checksums change due to "out of picture" values
being initialized differently.
There should be no visual difference in the filters output
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This change introduces a basic encoder for 3GPP Timed Text subtitles,
also known as TX3G, Quicktime subtitles, or "movtext" in the existing
code.
This initial change doesn't attempt to write styling information,
and just writes the plain text of the subtitles. I intend to add
support for styles eventually, but it's challenging due to a lack
of existing players that support them.
Note that an additional change is required to the mov/mp4 muxer to
write empty subtitle packets to indicate subtitle duration.
Signed-off-by: Philip Langdale <philipl@overt.org>
Restore functionality to set the samples directory via the
FATE_SAMPLES environment variable . This is broken since commit
63dcd16 was merged.
Additionally the name FATE_EXTERN is more suited as the current
FATE_SAMPLES make file variable does not carry the name of the
FATE samples or the name of the directory they are stored in, but
does contain the names of the FATE targets that need external
samples. That is samples that are not in the repository and are
not generated on the fly.
Signed-off-by: Alexander Strasser <eclipse7@gmx.net>
According to its description, it is supposed to be the LCM of all the
frame durations. The usability of such a thing is vanishingly small,
especially since we cannot determine it with any amount of reliability.
Therefore get rid of it after the next bump.
Replace it with the average framerate where it makes sense.
FATE results for the wtv and xmv demux tests change. In the wtv case
this is caused by the file being corrupted (or possibly badly cut) and
containing invalid timestamps. This results in lavf estimating the
framerate wrong and making up wrong frame durations.
In the xmv case the file contains pts jumps, so again the estimated
framerate is far from anything sane and lavf again makes up different
frame durations.
In some other tests lavf starts making up frame durations from different
frame.
AVPacket.duration is mostly made up and thus completely useless, this is
especially true for video streams.
Therefore use dts difference for framerate estimation and
the max_analyze_duration check.
The asyncts test now needs -analyzeduration, because the default is 5
seconds and the audio stream in the sample appears at ~10 seconds.
MMX-enabled systems by default use some dsputil functions differing
from the C versions. Adding these flags ensures accurate ones are
used everywhere.
Signed-off-by: Mans Rullgard <mans@mansr.com>
there are some technical problems with fate.ffmpeg.org
thus split the subdomain between fate-suite and fate
fate-suite is now (temporary) provided by our main server
until fate-suite.ffmpeg.org is setup to point somewhere
we use fate-suite.avcodec.org
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
By moving it to a later point relative and unknown timestamps
are more likely to have been corrected
similar patch reviewed-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Conflicts:
libavformat/utils.c
commit 20e88d8618
Fix avui stream-copy.
The native decoder and MPlayer's binary decoder only need the
APRG atom, QuickTime at least requires also the ARES atom and
four additional 0 bytes padding at the end of stsd.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
These filters are designed for storing and transmitting video sequences
with alpha using higher-efficiency codecs such as x264 which don't
natively support an alpha channel. 'alphaextract' takes an input stream
with an alpha channel and returns a video containing just the alpha
component as a grayscale value; 'alphamerge' takes an RGB or YUV stream
and adds an alpha channel recovered from a second grayscale stream.
Signed-off-by: Steven Robertson <steven@strobe.cc>
Signed-off-by: Stefano Sabatini <stefasab@gmail.com>
Convert them to zigzag order, as the rest of them are.
When I was adding support for 10-bit DNxHD, I just copy-pasted the
missing quant matrices from the spec. Now it turns out the existing
matrices in dnxhddata.c were in zigzag order. This resulted in wrong
quantization for 10-bit DNxHD. The attached patch fixes the problem by
converting 10-bit quant matrices to zigzag order.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Jan Ekström
Mans Rullgard
Clément Bœsch
Philip Langdale
Boris Maksalov
Nicolas George
Michael Niedermayer
Diego Biurrun
Kostya Shishkov
Janne Grunau
Alexander Strasser
Ronald S. Bultje
Paul B Mahol
Anton Khirnov
Mashiat Sarker Shakkhar
Steven Robertson
Joseph Artsimovich