Both of these two structures were first available with X264_BUILD
163, so make relevant functionality conditional on the version
being at least such.
Keep handle_side_data available in all cases as this way X264_init
does not require additional version based conditions within it.
Finally, add a FATE test which verifies that pass-through of the
MDCV/CLL side data is working during encoding.
These two were added in 28e23d7f348c78d49a726c7469f9d4e38edec341
and 3558c1f2e97455e0b89edef31b9a72ab7fa30550 for version 0.9.0 of
SVT-AV1, which is also our minimum requirement right now.
In other words, no additional version limiting conditions seem
to be required.
Additionally, add a FATE test which verifies that pass-through of
the MDCV/CLL side data is working during encoding.
In particular, test writing tags with odd strlen.
(These tags are zero-padded to even size.)
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Also make use of the av_channel_from_string() function to determine the channel
id. This fixes some parse issues in av_channel_layout_from_string().
Signed-off-by: Marton Balint <cus@passwd.hu>
We lacked tests which supposed to fail, and there are some which should fail
but right now it does not. This will be fixed in a later commit.
Signed-off-by: Marton Balint <cus@passwd.hu>
Deduplicates a lot of code.
Some minor differences (mostly white space and inconsistent use of quotes) are
expected in the fate tests, there was no point aiming for exactly the same
formatting.
Signed-off-by: Marton Balint <cus@passwd.hu>
This makes the wav and pcm demuxer demux bigger packets, which is more
efficient.
As a side effect of the bigger packets, audio durations can become less exact
for command lines such as "ffmpeg -i $INPUT -c:a copy -t 1.0 $OUTPUT".
Signed-off-by: Marton Balint <cus@passwd.hu>
- Remove the 1024 cap on the number of samples, for high sample rate audio it
was suboptimal, calculate the low neighbour power of two for the number of
samples (audio blocks) instead.
- Make the function work correctly also for non-pcm codecs by using the stream
bitrate to estimate the target packet size. A previous version of this patch
used av_get_audio_frame_duration2() the estimate the desired packet size, but
for some codecs that returns the duration of a single audio frame regardless
of frame_bytes.
- Fallback to 4096/block_align*block_align if bitrate is not available.
Signed-off-by: Marton Balint <cus@passwd.hu>
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>
Jan Ekström
Andreas Rheinhardt
Gyan Doshi
Zhu Pengfei
Marton Balint
James Almer
Martin Storsjö