The test constructs 66 sine waves of different frequencies and joins
them into a 7th order ambisonic layout with extra non-diegetic
stereo. The channels are then shifted circularly with channelmap and
separated into individual streams for muxing with channelsplit.
Note that the last step is necessary because swr is not capable of
interleaving planar to packed for more than 64 channels, which would be
necessary to mux such a stream as PCM.
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>
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>
Explicitly insert the scale or aresample filter where it would
have been inserted by the negotiation.
Re-enable conversions if it cannot be done easily.
If a conversion is needed in a test, we want to know about it.
If the negotiation changes and makes new conversion necessary,
we want to know about it even more.
The tremolo filter uses floating point internally, and uses
multiplication factors derived from sin(fmod()), neither of
which is bitexact for use with framecrc.
This fixes running this test when built with for mingw/x86_32
with clang.
In this case, a 1 ulp difference in the output from fmod() would
end up in an output from the filter that differs by 1 ulp, but
which makes the lrint() in swresample/audioconvert.c round in a
different direction.
Signed-off-by: Martin Storsjö <martin@martin.st>
The flac parser uses a fifo to buffer its data. Consequently, when
searching for sync codes of flac packets, one needs to take care of
the possibility of wraparound. This is done by using an optimized start
code search that works on each of the continuous buffers separately and
by explicitly checking whether the last pre-wrap byte and the first
post-wrap byte constitute a valid sync code.
Moreover, the last MAX_FRAME_HEADER_SIZE - 1 bytes ought not to be searched
for (the start of) a sync code because a header that might be found in this
region might not be completely available. These bytes ought to be searched
lateron when more data is available or when flushing.
Unfortunately there was an off-by-one error in the calculation of the
length to search of the post-wrap buffer: It was too large, because the
calculation was based on the amount of bytes available in the fifo from
the last pre-wrap byte onwards. This meant that a header might be
parsed twice (once prematurely and once regularly when more data is
available); it could also mean that an invalid header will be treated as
valid (namely if the length of said invalid header is
MAX_FRAME_HEADER_SIZE and the invalid byte that will be treated as the
last byte of this potential header happens to be the right CRC-8).
Should a header be parsed twice, the second instance will be the best child
of the first instance; the first instance's score will be
FLAC_HEADER_BASE_SCORE - FLAC_HEADER_CHANGED_PENALTY ( = 3) higher than
the second instance's score. So the frame belonging to the first
instance will be output and it will be done as a zero length frame (the
difference of the header's offset and the child's offset). This has
serious consequences when flushing, as returning a zero length buffer
signals to the caller that no more data will be output; consequently the
last frames not yet output will be dropped.
Furthermore, a "sample/frame number mismatch in adjacent frames" warning
got output when returning the zero-length frame belonging to the first
header, because the child's sample/frame number of course didn't match
the expected sample frame/number given its parent.
filter/hdcd-mix.flac from the FATE-suite was affected by this (the last
frame was omitted) which is the reason why several FATE-tests needed to
be updated.
Fixes ticket #5937.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Add tests for upmixing and downmixing with audio channel counts that
have a corresponding default layout and also tests where there is no
default layout.
Update the existing "stereo4" test so it actually outputs stereo like
the other stereo tests. Rename the previous "stereo4" test into
"upmix1".
Reviewed-by: Nicolas George <george@nsup.org>
Signed-off-by: Tobias Rapp <t.rapp@noa-archive.com>
* commit '4141a5a240fba44b4b4a1c488c279d7dd8a11ec7':
Use modern avconv syntax for codec selection in documentation and tests
Merged-by: James Almer <jamrial@gmail.com>
Adds another test for asetnsamples filter where padding of the last
frame is switched off. Renames the existing test to make the difference
obvious.
Tested-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: Tobias Rapp <t.rapp@noa-archive.com>
The md5 protocol has no seek support, but some tests use seeks. This changes
the fate tests to actually create the output files and calculate the md5 on the
written files, which also makes the tests independent of the size of the output
buffers and output buffering in general.
A new md5pipe fate test method is also introduced to keep the old functionality
for tests where using a non-seekable output was intentional, and matroska md5
tests are changed to use that.
Signed-off-by: Marton Balint <cus@passwd.hu>
* commit '043b0b9fb1481053b712d06d2c5b772f1845b72b':
Replace leftover uses of -aframes|-dframes|-vframes with -frames:a|d|v
The merge also includes all our own occurences.
Merged-by: Clément Bœsch <u@pkh.me>
Additional/Modified FATE tests improve code coverage from 63.7% to 98.1%.
Changed fate-suite sample files:
* filter/hdcd-mix.flac (958K) added. It is a much better test than
filter/hdcd.flac (910K), which is now unused, but can't be removed.
* filter/hdcd-fake20bit.flac (168K) added. It is the first second of
filter/hdcd.flac, with the 16-bit LSB copied into bit 20 of a 24-bit
stream. There isn't an actual non-16-bit HDCD sample available to test.
Signed-off-by: Burt P <pburt0@gmail.com>
add tests/ref/fate/filter-hls-append for FATE
add hls-list-append fate use filter make audio data and test hls_flags
append options
Signed-off-by: Steven Liu <lingjiujianke@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
If this still doesnt give the same results on all platforms then this should be
disabled
Found-by: James Almer <jamrial@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
