When a Matroska Block is only stored in compressed form, the size of
the uncompressed block is not explicitly coded and therefore not known
before decompressing it. Therefore the demuxer uses a guess for the
uncompressed size: The first guess is three times the compressed size
and if this is not enough, it is repeatedly incremented by a factor of
three. But when this happens with lzo, the decompression is neither
resumed nor started again. Instead when av_lzo1x_decode indicates that x
bytes of input data could not be decoded, because the output buffer is
already full, the first (not the last) x bytes of the input buffer are
resent for decoding in the next try; they overwrite already decoded
data.
This commit fixes this by instead restarting the decompression anew,
just with a bigger buffer.
This seems to be a regression since 935ec5a1.
A FATE-test for this has been added.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: James Almer <jamrial@gmail.com>
This test tests that demuxing ProRes that is muxed like it should be in
Matroska (i.e. with the first header ("icpf") atom stripped away) works;
it also tests bz2 decompression as well as the handling of
unknown-length clusters.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: James Almer <jamrial@gmail.com>
In these cases, we must pass the full path of the file to ffprobe
(as the current working dir on the remote system, e.g. when invoked
with "ssh remote ffprobe ..." isn't the wanted one).
The input filename passed to ffprobe is also included in the output,
which is part of the reference test data. Add a new option to
ffprobe to allow overriding what path is printed, to keep the
original relative path in the tests.
An alternative approach could be an option to allow requesting omitting
the file name from the dumped data, and updating the test references
accordingly.
Signed-off-by: Martin Storsjö <martin@martin.st>
5 cabac states for cbf_cb and cbf_cr are supported according to
Table 9-4.
Add a test for 64x64 4:4:4 8bit HEVC clips with TUDepth = 4, cbf_cr > 0.
Signed-off-by: Xu Guangxin <guangxin.xu@intel.com>
Signed-off-by: Linjie Fu <linjie.fu@intel.com>
Signed-off-by: James Almer <jamrial@gmail.com>
When testing on a memory limited system, these tests consume a
significant amount of memory and can often fail if testing by running
multiple processes in parallel.
Signed-off-by: Martin Storsjö <martin@martin.st>
The IVF muxer autoinserts the av1_metadata filter unconditionally, which is
not desirable for these tests.
Signed-off-by: James Almer <jamrial@gmail.com>
These dependencies are evaluted by make and must be expressed with
the paths as in the local filesystem.
Signed-off-by: Martin Storsjö <martin@martin.st>
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>
contained in Vorbis comments in the CodecPrivate of flac tracks.
Moreover, it also tests header removal compression.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: James Almer <jamrial@gmail.com>
This test contains a track with zlib compressed CodecPrivate in addition
to compressed frames; the former was unchecked before.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: James Almer <jamrial@gmail.com>
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>
Right now, the concat filter does not set the frame_rate value on any of
the out links. As a result, the default ffmpeg behaviour kicks in - to
copy the framerate from the first input to the outputs.
If a later input is higher framerate, this results in dropped frames; if
a later input is lower framerate it might cause judder.
This patch checks if all of the video inputs have the same framerate, and
if not it sets the out link to use '1/0' as the frame rate, the value
meaning "unknown/vfr".
A test is added to verify the VFR behaviour. The existing test for CFR
behaviour passes unchanged.
This fixes make fate issue for frame thread scale in my local testing
Signed-off-by: Limin Wang <lance.lmwang@gmail.com>
Signed-off-by: Marton Balint <cus@passwd.hu>
why change .4 to .25, it's for:
one scenecut(pkt_pts=20040) isn't detected by 0.4 threshold
why not change to 0.3 instead of 0.25:
it will miss the scenecut(pkt_pts=20040) after applying the next
patch which enables yuvj420
for fate testing, it's better to catch all scenecut scenes.
Reviewed-by: Marton Balint <cus@passwd.hu>
Signed-off-by: Limin Wang <lance.lmwang@gmail.com>
The tests previously rounded the timestamps. Its better in a fate test to preserve
the data from the demuxer and decoder.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Up until now, the length field of most level 1 elements has been written
using eight bytes, although it is known in advance how much space the
content of said elements will take up so that it would be possible to
determine the minimal amount of bytes for the length field. This
commit changes this.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: James Almer <jamrial@gmail.com>
Given that in both the seekable as well as the non-seekable mode dynamic
buffers are used to write level 1 elements and that now no seeks are
used in the seekable case any more, the two modes can be combined; as a
consequence, the non-seekable mode automatically inherits the ability to
write CRC-32 elements.
There are no differences in case the output is seekable; when it is not
and writing CRC-32 elements is disabled, there can still be minor
differences because before this commit, the EBML ID and length field
were counted towards the cluster size limit; now they no longer are.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: James Almer <jamrial@gmail.com>
Up until now the EBML Header length field has been written with eight
bytes, although the EBML Header is always so small that only one byte
is needed for it. This patch saves seven bytes for every Matroska/Webm
file.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: James Almer <jamrial@gmail.com>
The transcode() helper function will already prepend the TARGET_PATH to
the sample path, if its a relative path. This avoids an issue on
Windows, where the relative path check could fail.
write_tmcd allows tmcd track to be created with any mode but in
mov_write_header, index for first tmcd track is only set for modes
MP4 or MOV, causing a crash if tmcd creation is attempted with other
modes.
* commit 'f8df5e2f31a5ba7b30a0e1caaaf5a03c753b3f9b':
tests: Add a convenience function for video-only lavf tests
Merged-by: James Almer <jamrial@gmail.com>
* commit 'a70eac7a9b193e8434b5bed90bd72aa4cb688363':
tests: Convert image2pipe tests to non-legacy test scripts
Merged-by: James Almer <jamrial@gmail.com>
init add three test examples:
1. check no endlist at the end
2. check endlist at the end
3. check hls_list_size 0 full list
Tested-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: Steven Liu <lq@chinaffmpeg.org>
Fixes vorbis mp4 audio files, with edit list specified. Since
st->skip_samples is not set in case of vorbis , ffmpeg computes the
start_time as negative.
Signed-off-by: Sasi Inguva <isasi@google.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
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>
verify that the stco atom is upgraded to co64 when the addition of moov
size to the offsets results in an overflow
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
If start_time is not set, ffmpeg takes the duration from the global
movie instead of the per stream duration.
Signed-off-by: Sasi Inguva <isasi@google.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This uses any devices it can find on the host system - on a system with no
hardware device support or in builds with no support included it will do
nothing and pass.
This new optional flag makes it easier to deal with mpegts
samples where the PMT is updated and elementary streams move
to different PIDs in the middle of playback.
Previously, new AVStreams were created per PID, and it was up
to the user to figure out which streams had migrated to a new PID
(by iterating over the list of AVProgram and making guesses), and
switch seamlessly to the new AVStream during playback.
Transcoding or remuxing these streams with ffmpeg on the CLI was
also quite painful, and the user would need to extract each set
of PIDs into a separate file and then stitch them back together.
With this new option, the mpegts demuxer will automatically detect
PMT changes and feed data from the new PID to the original AVStream
that was created for the orignal PID. For mpegts samples with
stream_identifier_descriptor available, the unique ID is used to
merge PIDs together. If the stream id is not available, the demuxer
attempts to map PIDs based on their position within the PMT.
With this change, I am able to playback and transcode/remux these
two samples which previously caused issues:
https://tmm1.s3.amazonaws.com/pmt-version-change.tshttps://kuroko.fushizen.eu/videos/pid_switch_sample.ts
I also have another longer sample in which the PMT changes
repeatedly and ES streams move to different pids three times
during playback:
https://tmm1.s3.amazonaws.com/multiple-pmt-change.ts
Demuxing this sample with the new option shows several new log
messages as the PMT changes are handled:
[mpegts] detected PMT change (program=1, version=3/6, pcr_pid=0xf98/0xfb7)
[mpegts] re-using existing video stream 0 (pid=0xf98) for new pid=0xfb7
[mpegts] re-using existing audio stream 1 (pid=0xf99) for new pid=0xfb8
[mpegts] re-using existing audio stream 2 (pid=0xf9a) for new pid=0xfb9
[mpegts] detected PMT change (program=1, version=6/3, pcr_pid=0xfb7/0xf98)
[mpegts] detected PMT change (program=1, version=3/4, pcr_pid=0xf98/0xf9b)
[mpegts] re-using existing video stream 0 (pid=0xf98) for new pid=0xf9b
[mpegts] re-using existing audio stream 1 (pid=0xf99) for new pid=0xf9c
[mpegts] re-using existing audio stream 2 (pid=0xf9a) for new pid=0xf9d
[mpegts] detected PMT change (program=1, version=4/5, pcr_pid=0xf9b/0xfa9)
[mpegts] re-using existing video stream 0 (pid=0xf98) for new pid=0xfa9
[mpegts] re-using existing audio stream 1 (pid=0xf99) for new pid=0xfaa
[mpegts] re-using existing audio stream 2 (pid=0xf9a) for new pid=0xfab
[mpegts] detected PMT change (program=1, version=5/6, pcr_pid=0xfa9/0xfb7)
Signed-off-by: Aman Gupta <aman@tmm1.net>
Generates color bar test patterns based on EBU PAL recommendations.
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Tobias Rapp <t.rapp@noa-archive.com>
Adds tests for the hue angle and brightness filter parameters.
Renames the existing saturation parameter test for consistency.
Signed-off-by: Tobias Rapp <t.rapp@noa-archive.com>
The artificial sample file sei-1.h264 contains five frames (IDR P B I B)
and the following SEI message types:
* Buffering period
* Picture timing
* Pan-scan rectangle (display as 4:3)
* User data registered, containing A/53 closed captions (captions match
frame content, including reordering)
* Recovery point (at the I frame)
* Display orientation (identity transformation)
* Mastering display (with arbitrary contents)
* Undefined SEI type 1234 (containing ascending bytes)
Uses the same mechanism as other codecs - conformance test files are
passed through the metadata filter (which, with no options, reads the
input and writes it back) and the output verified to match the input.
- Parse schm atom to get different encryption schemes.
- Allow senc atom to appear in track fragments.
- Allow 16-byte IVs.
- Allow constant IVs (specified in tenc).
- Allow only tenc to specify encryption (i.e. no senc/saiz/saio).
- Use sample descriptor to detect clear fragments.
This doesn't support:
- Different sample descriptor holding different encryption info.
- Only first sample descriptor can be encrypted.
- Encrypted sample groups (i.e. seig).
- Non-'cenc' encryption scheme when using -decryption_key.
Signed-off-by: Jacob Trimble <modmaker@google.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Some ADTS streams can have multiple ID3 tags between frames. This
change parses all of them, rather than just the first one.
Signed-off-by: Mattias Amnefelt <mattiasa@avm.se>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
On modern x86 systems its around 2x faster. For systems without
FPUs it'll be slower, but our policy is to prefer floating point
implementations and to let users decide what's best (or just not
compile them on systems without FPUs).
Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
Set relevant filter parameters such that the result can easily be
checked with a waveform editor.
In particular, it makes it clear the silence_start is not accurate in
the current code.
test extract color and alpha
with the three main kind of hap frame :
- no snappy compression
- snappy compression and one chunk
- snappy compression and several chunks (16 here)
like the bsf filter need to be used with vtag and encoder edition
also test the information of the target mov for color and alpha
Fixes seek for files with empty edits and files with negative ctts
(dts_shift > 0). Added fate samples and tests.
Signed-off-by: Sasi Inguva <isasi@isasi.mtv.corp.google.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
These tests cover specific rounding behaviour, to ensure that I don't
introduce any regressions with the rewritten "activate" callback based
fps filter.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
It tests a useless profile which sounds no better than regular aac and which
takes extremely long to encoder something. Also it has been behind experimental
flag for as long as it has been supported.
Should be removed altogether sometime in the future.
The twoloop coder sounds decent at low bitrates, however at higher bitrates
it sounds worse than the fast coder (which used to be the old twoloop coder
before October 2015) and needs quite a lot more CPU.
Change the default to fast. It has been well tested and has had little changes
over the years so its been confirmed to be quite stable.
Also change its description (not valid for more than a year) and the
documentation.
Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
The PERSIST_RPARAM_A_RExt_Sony_1 bitstream has an out-of-range value
and has therefore been superseded.
It is otherwise identical, and decodes the same.
Signed-off-by: James Almer <jamrial@gmail.com>
Resulted in valgrind errors due to uninitialized memory.
Also updates fate and makes it use the tron sample result.
Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
Also change note to say that we compare against the officially decoded
samples rather than our own, this was changed long ago.
Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
After c2a8f0fcbe this can happen on normal edit lists starting on a B-frame.
Signed-off-by: Sasi Inguva <isasi@google.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
* commit '39e16ee2289e4240a82597b97db5541bbbd2b996':
Revert "fate: Skip the checkasm test if CONFIG_STATIC is disabled"
Merged-by: James Almer <jamrial@gmail.com>
Previously alac encoder was used, from a first glance I thought it is bitexact,
but it turns out it is using floating point arithmetic as well, so probably it
is not. Fixes fate failures on mingw32/64.
Signed-off-by: Marton Balint <cus@passwd.hu>
Sets the correct start padding value when an edit list is present.
A new fate test is added, fate-mov-440hz-10ms, to ensure this is
handled correctly.
Signed-off-by: Dale Curtis <dalecurtis@chromium.org>
Reviewed-by: Sasi Inguva <isasi-at-google.com@ffmpeg.org>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Use the appropriate metadata filter for each codec - in the absence of any
options to modify the stream, the output bitstream should be identical to
the input (though the output file may differ in padding).
All tests use conformance bitstreams, the MPEG-2 streams are newly added
from the conformance test streams
<http://standards.iso.org/ittf/PubliclyAvailableStandards/ISO_IEC_13818-4_2004_Conformance_Testing/Video/>
(cherry picked from commit 3cae7f8b9b)
(cherry picked from commit fbd63170bc)
* commit '4141a5a240fba44b4b4a1c488c279d7dd8a11ec7':
Use modern avconv syntax for codec selection in documentation and tests
Merged-by: James Almer <jamrial@gmail.com>
The first frame changes depending on --enable-memory-poisoning being
used to configure ffmpeg or not, even if requesting bitexact decoding.
Disable the test until this is fixed.
Signed-off-by: James Almer <jamrial@gmail.com>
* commit '8e4d4efc67e154fdffd65964a7cfeef740320827':
fate: Add another SVQ3 test to increase coverage
Also included a fix from da8093f712.
The demuxer option "-ignore_editlist 1 " is temporarily added to the
test as well, to workaround a regression in the edit list mov parsing
code.
Merged-by: James Almer <jamrial@gmail.com>
Correctly set the interlaced_frame and top_field_first fields when pic_struct
indicates paired fields.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Metadata filter output is passed through an Awk script comparing floats
against reference values with specified "fuzz" tolerance to account for
architectural differences (e.g. x86-32 vs. x86-64).
Signed-off-by: Tobias Rapp <t.rapp@noa-archive.com>
Use the appropriate metadata filter for each codec - in the absence of any
options to modify the stream, the output bitstream should be identical to
the input (though the output file may differ in padding).
All tests use conformance bitstreams, the MPEG-2 streams are newly added
from the conformance test streams
<http://standards.iso.org/ittf/PubliclyAvailableStandards/ISO_IEC_13818-4_2004_Conformance_Testing/Video/>
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>
When we use dllexport properly for shared libraries on windows,
there's no longer any issue with linking the object files for
e.g. libavcodec statically into checkasm. (It's still not possible
to link the built object files for e.g. libavformat statically to
libavcodec though, since libavformat exepcts to load av_export_*
symbols from a DLL.)
This reverts commit 4e62b57ee0.
Signed-off-by: Martin Storsjö <martin@martin.st>
Adds FATE tests for the previously untested allrgb, allyuv, rgbtestsrc,
smptebars, smptehdbars and yuvtestsrc filters.
Also adds a test for testsrc2 filter with rgb+alpha.
Tested-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: Tobias Rapp <t.rapp@noa-archive.com>
The -map option allows for a trailing ? so that an error is not thrown if
the input stream does not exist.
This capability is extended to the map_channel option.
This allows a ffmpeg command not to break if an input channel does not
exist, which can be of use (for instance, scripts processing audio
channels with sources having unset number of audio channels).
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
When sidx box support is enabled, the code will skip reading all
trun boxes (each containing ctts entries for samples inthat box).
If seeks are attempted before all ctts values are known, the old
code would dump ctts entries into the wrong location. These are
then used to compute pts values which leads to out of order and
incorrectly timestamped packets.
This patch fixes ctts processing by always using the index returned
by av_add_index_entry() as the ctts_data index. When the index gains
new entries old values are reshuffled as appropriate.
This approach makes sense since the mov demuxer is already relying
on the mapping of AVIndex entries to samples for correct demuxing.
As a result of this all ctts entries are now 1-count. A followup
change will be submitted to remove support for > 1 count entries
which will simplify seeking.
Notes for future improvement:
Probably there are other boxes (stts, stsc, etc) that are impacted
by this issue... this patch only attempts to fix ctts since it
completely breaks packet timestamping.
This patch continues using an array for the ctts data, which is not
the most ideal given the rearrangement that needs to happen (via
memmove as new entries are read in). Ideally AVIndex and the ctts
data would be set-type structures so addition is always worst case
O(lg(n)) instead of the O(n^2) that exists now; this slowdown is
noticeable during seeks.
Signed-off-by: Dale Curtis <dalecurtis@chromium.org>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
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>
This test the demuxer discarding non ADTS frames at the beginning and
end of the input.
As a side effect, this commit also enables fate-adts-demux, which was
accidentally disabled in 324f0fbff1.
Tested-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: James Almer <jamrial@gmail.com>
This new FATE test for the scale2ref filter makes use of the recently
added scale2ref-specific variables to maintain the aspect ratio of a
test input.
Filtergraph explanation:
[main] has an AR of 4:3. [ref] has an AR of 16:9.
640 / 4 = 160. So the new width for [main] is 160.
160 / ((320 / 240) * (1 / 1)) = 160 / (4 / 3) = 120. So the new
height for [main] is 120.
160 / 120 = 4 / 3 so [main]'s aspect ratio has been maintained while
using [ref]'s width as a reference point.
[ref] is nullsink'd since it is left unchanged by scale2ref (and so
shouldn't need to be tested).
If we were to use "iw/4:-1" in place of "iw/4:ow/mdar":
640 / 4 = 160. So the new width for [main] would be 160.
360 / 4 = 90. So the new height for [main] would be 90.
160 / 90 = 16 / 9 so [main] now has the same aspect ratio as [ref]
which is probably what you do not want.
This is currently the only test for scale2ref.
Signed-off-by: Kevin Mark <kmark937@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This removes the current API violating behavior of overwritting the stream's
extradata during packet filtering, something that should not happen after the
av_bsf_init() call.
The bitstream filter generated extradata is no longer available during
write_header(), and as such not usable with non seekable output. The FATE
tests are updated to reflect this.
Signed-off-by: James Almer <jamrial@gmail.com>
