All versions of MSVC that support C11 (namely >= v19.27)
also support the restrict keyword, therefore av_restrict
is no longer necessary since 75697836b1.
Reviewed-by: Martin Storsjö <martin@martin.st>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
This simplifies the code for checking the output, and can print
the failing output (including a map of matching/mismatching
elements) if checkasm is run with the -v/--verbose option.
Signed-off-by: J. Dekker <jdek@itanimul.li>
Previously it only checked half the output in 8 bit per pixel mode,
as the output actually is 16 bit elements here.
Signed-off-by: J. Dekker <jdek@itanimul.li>
Muxing multiple streams to raw files is allowed but the packets are
interleaved, so the output is dependant of packet size.
Signed-off-by: Marton Balint <cus@passwd.hu>
The samples I found all have 2000 sample packets, and by forcing the packet
size with a bsf we could automagically make muxing work for packets containing
more than 3640 samples.
Signed-off-by: Marton Balint <cus@passwd.hu>
Treat it analogously to stream parameters like format/dimensions/etc.
This is functionally different from previous code in 2 ways:
* for non-CFR video, the frame timebase (set by the decoder) is used
rather than the demuxer timebase
* for sub2video, AV_TIME_BASE_Q is used, which is hardcoded by the
subtitle decoding API
These changes should avoid unnecessary and potentially lossy timestamp
conversions from decoder timebase into the demuxer one.
Changes the timebases used in sub2video tests.
Some encoders, like flac, propagate updated extradata at the end of encoding
as packet side data. Use it to update the relevant codec_config.
Signed-off-by: James Almer <jamrial@gmail.com>
The wav demuxer by default tried to demux 4096-byte packets which caused
packets with very few number of samples for files with high channel count.
This caused a significant overhead especially since the latest ffmpeg.c
threading changes.
So let's use a similar approach for selecting audio frame size which is already
used in the PCM demuxer, which is to read 25 times per second but at most 1024
samples.
Signed-off-by: Marton Balint <cus@passwd.hu>
Since PTS is changed randomly for every audio frame, it matters. Also add some
forgotten filter dependencies.
Signed-off-by: Marton Balint <cus@passwd.hu>
Depending on input chunk size noticable corrpution was hearable, here is an
example command line:
ffplay -f lavfi -i "sine=440:r=8000:samples_per_frame=32,aresample=24000:filter_size=1:phase_shift=0"
Fix this by rounding the fixed point fractions up instead of down.
Signed-off-by: Marton Balint <cus@passwd.hu>
GEN=1 is used to generate reference files in the source tree, not
auto-generated reference samples.
Without this patch GEN=1 could overwrite the auto generated reference files
in each test where they are used causing failures.
Signed-off-by: Marton Balint <cus@passwd.hu>
James Almer
Andreas Rheinhardt
Martin Storsjö
Marton Balint
Kieran Kunhya
Anton Khirnov
Nicolas Gaullier