In this case in_channel_idx was never set and the default 0 was used.
Suprisingly no one noticed that the respective fate test output was wrong.
Signed-off-by: Marton Balint <cus@passwd.hu>
This test muxes two streams into a single pcm file, although
the two streams are of course not recoverable from the output
(unless one has extra information). So use the streamhash muxer
instead (which also provides coverage for it; it was surprisingly
unused in FATE so far). This is in preparation for actually
enforcing a limit of one stream for the PCM muxers.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
The newer of these two are the separate integers for content light
level, introduced in 3952bf3e98c76c31594529a3fe34e056d3e3e2ea ,
with X265_BUILD 75. As we already require X265_BUILD of at least
89, no further conditions are required.
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>
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>
There is no MMX DSP code for VVC, so one can use the stricter
declare_func which also tests that we are not in MMX mode
at the end of this function.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Otherwise aacenc.o gets pulled in by the aacencdsp checkasm
test and it in turn pulls the rest of lavc in.
Besides being bad size-wise this also has the downside that
it pulls in avpriv_(cga|vga16)_font from libavutil which are
marked as being imported from another library when building
libavcodec as a DLL and this breaks checkasm because it links
both lavc and lavu statically.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Jpeg2000 decoder is decoding in native endian, so let's use the same workaround
as in fate-mxf-probe-applehdr10.
Fixes ticket #10868.
Signed-off-by: Marton Balint <cus@passwd.hu>
Contrary to the existing "fate-checkasm", this always prints the
tool output, and runs all tests at once instead of splitting it up
per target group. This is more useful when the user expects to
look directly at the tool output, instead of being part of a full
fate run.
(On failure with the regular "make fate-checkasm" targets, none of
the tool output is printed, but stored in files. If run with reporting
set up to the FATE website, the individual failures are uploaded there,
but if it is run in some sort of other CI setup, the intermediate files
might not be available afterwards for inspection.)
Signed-off-by: Martin Storsjö <martin@martin.st>
