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>
Makes the handling of unspecified/unknown color_range values on stream
level consistent to the value used on frame level.
Signed-off-by: Tobias Rapp <t.rapp@noa-archive.com>
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>
Fixes filter-pixfmts-scale test failing on big-endian systems due to
alpSrc not being cast to (const int32_t**).
Also fixes distortions in the output alpha channel values by copying the
alpha channel code from the rgba64 case found elsewhere in output.c.
Fixes ticket 6555.
Signed-off-by: James Cowgill <James.Cowgill@imgtec.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This commit switches off forced correct nesting of tags and only keeps
it for font tags. See long explanations in the code for the rationale.
This results in various FATE changes which I'll explain here:
- various swapping in font attributes, this is mostly noise due to the
old reverse stack way of printing them. The new one is more correct as
the last attribute takes over the previous ones.
- unrecognized tags disappears
- invalid tags that were previously displayed aren't anymore (instead,
we have a warning). This is better for the end user
The main benefit of this commit is to be more tolerant to error, leading
to a better handling of badly nested tags or random wrong formatting for
the end user.
The scale2ref filter will now maintain the DAR of the main input and
not the DAR of the reference input. This previous behavior was deemed
counterintuitive for most (all?) use-cases.
Before:
scale2ref=iw/4:ow/mdar
in w:320 h:240 fmt:rgb24 sar:1/1
ref w:640 h:360 fmt:rgb24 sar:1/1
out w:160 h:120 fmt:rgb24 sar:4/3 flags:0x2
SAR: ((120 * 640) / (160 * 360)) * (1 / 1) = 4 / 3
DAR: (160 / 120) * (4 / 3) = 16 / 9
(main out now same DAR as ref)
Now:
scale2ref=iw/4:ow/mdar
in w:320 h:240 fmt:rgb24 sar:1/1
ref w:640 h:360 fmt:rgb24 sar:1/1
out w:160 h:120 fmt:rgb24 sar:1/1 flags:0x2
SAR: ((120 * 320) / (160 * 240)) * (1 / 1) = 1 / 1
DAR: (160 / 120) * (1 / 1) = 4 / 3
(main out same DAR as main in)
The scale2ref FATE test has also been updated.
Signed-off-by: Kevin Mark <kmark937@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This is actually internal utvideo format.
Allows to make use of SIMD for median prediction for rgb(a) formats,
thus speeding up decoding.
Simplifies code, eases further developement and maintenance.
Update FATE because of pixel format switch.
Signed-off-by: Paul B Mahol <onemda@gmail.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>
If the videos starts with B frame, then the minimum composition time
as computed by stts + ctts will be non-zero. Hence we need to shift
the DTS, so that the first pts is zero. This was the intention of that
code-block. However it was subtracting by the wrong amount.
For example, for one of the videos in the bug nonFormatted.mp4 we have
stts:
sample_count duration
960 1001
ctts:
sample_count duration
1 3003
2 0
1 3003
....
The resulting composition times are : 3003, 1001, 2002, 6006, ...
The minimum composition time or PTS is 1001, which should be used to
offset DTS. However the code block was wrongly using ctts[0] which is
3003. Hence the PTS was negative. This change computes the minimum pts
encountered while fixing the index, and then subtracts it from all the
timestamps after the edit list fixes are applied.
Samples files available from:
https://bugs.chromium.org/p/chromium/issues/detail?id=721451https://bugs.chromium.org/p/chromium/issues/detail?id=723537
fate-suite/h264/twofields_packet.mp4 is a similar file starting with 2
B frames. Before this change the PTS of first two B-frames was -6006
and -3003, and I am guessing one of them got dropped when being decoded
and remuxed to the framecrc before, and now it is not being dropped.
Signed-off-by: Sasi Inguva <isasi@google.com>
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>
* commit '019ab88a95cb31b698506d90e8ce56695a7f1cc5':
lavc: add an option for exporting cropping information to the caller
Merged-by: James Almer <jamrial@gmail.com>
This complex (-1 2 6 2 -1) filter slightly less reduces interlace 'twitter' but better retain detail and subjective sharpness impression compared to the linear (1 2 1) filter.
Signed-off-by: Thomas Mundt <tmundt75@gmail.com>
Signed-off-by: James Almer <jamrial@gmail.com>
the tested sample contain negative value in the red channel
need to be clip to zero, and not set to MAX_RED
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Add an option to webm_dash_manifest demuxer to specify a value for
"bandwidth" field in the DASH manifest. The value is then used by
the muxer. Fixes an existing FIXME in the code.
Signed-off-by: Vignesh Venkatasubramanian <vigneshv@google.com>
Signed-off-by: James Zern <jzern@google.com>
This merges commits 8e2ea69135 and
096a8effa3 by Anton Khirnov, with the
following change:
- extract_extradata_check() is added to know if the codec is supported
by the bsf before trying to initialize it. This behaviour is similar to
the old AVCodecParser.split checks.
The FATE reference changes are due to the filtered out NAL units that
the old AVCodecParser.split implementation left alone.
Decoding is unchanged as the functions that parse extradata simply
ignored said unnecessary NAL units.
Signed-off-by: James Almer <jamrial@gmail.com>
* commit '481ff3cf018811ba3235f1c236e970f32a6300b9':
fate: Add h264 and hevc extradata reload tests
Only the HEVC part is merged, see 00c8079816
Merged-by: Clément Bœsch <u@pkh.me>
* commit 'b90c8a3d08e3f9ad4de1253376d2d1d93abb8b8c':
fate: Add tests for mov display matrix
Adapted to use ffprobe -show_entries
Merged-by: James Almer <jamrial@gmail.com>
This field is of little value, and interferes with testing side data,
since sizes can be different on multiple architectures.
Signed-off-by: Vittorio Giovara <vittorio.giovara@gmail.com>
Allows to get a more realistic total bitrate (and estimated file size)
in avi_write_header. Previously a static default value of 200k was
assumed.
Adds an internal helper function for bitrate guessing.
Signed-off-by: Tobias Rapp <t.rapp@noa-archive.com>
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Preparation for potentially disabling merged side data by default in the
libs. Do this in particular because it affects fate tests.
The changed tests either reflect added packet side data, or the changed
packet size due to merged side data removal reducing the packet size.
The Chen-Shapiro(CS) test was used to test normality for
Lagged Fibonacci PRNG.
Normality Hypothesis Test:
The null hypothesis formally tests if the population
the sample represents is normally-distributed. For
CS, when the normality hypothesis is True, the
distribution of QH will have a mean close to 1.
Information on CS can be found here:
http://www.stata-journal.com/sjpdf.html?articlenum=st0264http://www.originlab.com/doc/Origin-Help/NormalityTest-Algorithm
Signed-off-by: Thomas Turner <thomastdt@googlemail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
The constants used in the decoder used floating point precision,
and this caused different values to be generated on different
architectures. Additionally on big endian machines, the fate test
would output bytes in native order, which is different from the one
hardcoded in the test.
So, eradicate floating point numbers and use fixed point (32.32)
arithmetics everywhere, replacing constants with precomputed integer
values, and force the pixel format output to be the same in the fate
test.
Signed-off-by: Vittorio Giovara <vittorio.giovara@gmail.com>
This makes sure the actual stream parameters are used, which is
important mainly for hardware decoding+filtering cases, which would
previously require various weird workarounds to handle the fact that a
fake software graph has to be constructed, but never used.
This should also improve behaviour in rare cases where
avformat_find_stream_info() does not provide accurate information.
This merges Libav commit a3a0230. It was previously skipped.
The code in flush_encoders() which sets up a "fake" format wasn't in
Libav. I'm not sure if it's a good idea, but it tends to give
behavior closer to the old one in certain corner cases.
The vp8-size-change gives different result, because now the size of
the first frame is used. libavformat reported the size of the largest
frame for some reason.
The exr tests now use the sample aspect ratio of the first frame. For
some reason libavformat determines 0/1 as aspect ratio, while the
decoder returns the correct one.
The ffm and mxf tests change the field_order values. I'm assuming
another libavformat/decoding mismatch.
Signed-off-by: wm4 <nfxjfg@googlemail.com>
This will be useful in the following commit, after which the muxer
timebase is not always available when encoding.
This merges Libav commit 3e265ca. It was previously skipped.
There are some changes with how/when the mux_timebase field is set,
because the Libav approach often causes a too imprecise time base
to be set. This is hard, because the muxer's write_header function
can readjust the timebase, at which point we might already have
encoded packets buffered. (It might be better to buffer them after
the encoder, instead of after all the timestamp handling logic
before muxing.)
The two FATE tests change because the output time base is raised
for subtitles. (Needed to avoid certain rounding issues in other
cases.)
Includes a minor merge fix by Mark Thompson, and
avconv: Move rescale to stream timebase before monotonisation
also by Mark Thompson <sw@jkqxz.net>.
Signed-off-by: wm4 <nfxjfg@googlemail.com>
This should fix the fate failure due to a truncated last frame.
Alternatively the frame could be dropped.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes: 664/clusterfuzz-testcase-4917047475568640
The change to fate is due to a truncated last frames which is now detected as damaged.
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/targets/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Current code returned the number of channels as channel layout in that case,
and if nret is not set then unknown layouts are typically not supported.
Also use the common parsing code. Use a temporary workaround to parse an
unknown channel layout such as '13c', after a 1 year grace period only '13C'
will work.
Signed-off-by: Marton Balint <cus@passwd.hu>
* commit '38efff92f1ef81f3de20ff0460ec7b70c253d714':
FATE: add a test for H.264 with two fields per packet
h264: fix decoding multiple fields per packet with slice threads
This merge includes two commits because the FATE test was useful in
order to make proper testing.
The merge gets rid of the now unused:
- SLICE_SINGLETHREAD and SLICE_SKIPED macros
- max_contexts
- "again" label in decode_nal_units()
This commit also includes the fix from d3e4d406b.
Thanks to wm4 and Michael Niedermayer for their testing.
Merged-by: Clément Bœsch <u@pkh.me>
Merged-by: Matthieu Bouron <matthieu.bouron@gmail.com>
* commit '8d07e941b04d63fc4443dd986e3dc7b69cdcca43':
FATE: add a test of H.264 SEI recovery in an intra refresh stream
Our H264 decoder drops 3 frames from the beginning of the stream, but
all frames after those match, hence the difference in the fate test.
Merged-by: Hendrik Leppkes <h.leppkes@gmail.com>
The test is not supposed to cover audio.
Also, using -vframes along with an audio stream depends on
the exact order the frames are processed by filters, it is
too much constraint to guarantee.
Add keyframe index metadata
Used to facilitate seeking; particularly for HTTP pseudo streaming.
1. read live streaming or file by sequence
2. if use add_keyframe_index option, add a mark flag at the position,
use to insert new context at the last step.
3. add the keyframes *offset* and *timestamp* into a list
4. if use add_keyframe_index option, shift the metadata data from
mark flag offset
5. insert the keyframes *offset* and *timestamp* from the list by
sequence
6. free the list
7. end.
Add FATE test case;
Reviewed-by: Lou Logan <lou@lrcd.com>
Signed-off-by: Steven Liu <liuqi@gosun.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This should be more useful for users since numerical values for channel
layout can be confusing and unintuitive.
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: James Almer <jamrial@gmail.com>
This matrix needs to be applied after all others have (currently only
display matrix from trak), but cannot be handled in movie box, since
streams are not allocated yet. So store it in main context, and apply
it when appropriate, that is after parsing the tkhd one.
Fate tests are updated accordingly.
Signed-off-by: Vittorio Giovara <vittorio.giovara@gmail.com>
The dynamic buffer does not contain the CRC32 element so calls to avio_tell()
don't take it into account. This resulted in CueRelativePosition values being
six bytes short.
This is a regression since 6724525a15
Instead of adding yet another custom check for CRC32 to fix a size or an offset,
remove the existing ones and reserve the six bytes in the dynamic buffer.
Signed-off-by: James Almer <jamrial@gmail.com>
This also fixes a minor bug introduced in the codecpar conversion, where
the termination condition for extracting the extradata does not match
the actual extradata setting code. As a result, the packet durations
made up by lavf go back to their values before the codecpar conversion.
That is of little consequence since that code should eventually be
dropped completely.
We don't currently support values 1 (centimeters), 2 (inches) or 3 (DAR),
only the default value 0 (pixels) which doesn't need to be written.
The fate refs are updated as unknown SAR is now signaled in the output
files with the addition of the new element.
Reviewed-by: Carl Eugen Hoyos <ceffmpeg@gmail.com>
Signed-off-by: James Almer <jamrial@gmail.com>
Using the stream timebase simply overflows
Fix integer overflow in psp framerate computation
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Implements part of ticket #4347
Tested-by: Dave Rice <dave@dericed.com>
Tested-by: Jerome Martinez <jerome@mediaarea.net>
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: James Almer <jamrial@gmail.com>
Implements part of ticket #4347
Tested-by: Dave Rice <dave@dericed.com>
Tested-by: Jerome Martinez <jerome@mediaarea.net>
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: James Almer <jamrial@gmail.com>
Implements part of ticket #4347
Tested-by: Dave Rice <dave@dericed.com>
Tested-by: Jerome Martinez <jerome@mediaarea.net>
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: James Almer <jamrial@gmail.com>
Implements part of ticket #4347
Tested-by: Dave Rice <dave@dericed.com>
Tested-by: Jerome Martinez <jerome@mediaarea.net>
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: James Almer <jamrial@gmail.com>
Implements part of ticket #4347
Tested-by: Dave Rice <dave@dericed.com>
Tested-by: Jerome Martinez <jerome@mediaarea.net>
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: James Almer <jamrial@gmail.com>
The durations are never written in that situation.
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: James Almer <jamrial@gmail.com>
This is a bit messy, mainly due to timestamp handling.
decode_video() relied on the fact that it could set dts on a flush/drain
packet. This is not possible with the old API, and won't be. (I think
doing this was very questionable with the old API. Flush packets should
not contain any information; they just cause a FIFO to be emptied.) This
is replaced with checking the best_effort_timestamp for AV_NOPTS_VALUE,
and using the suggested DTS in the drain case.
The modified tests (fate-cavs and others) still fails due to dropping
the last frame. This happens because the timestamp of the last frame
goes backwards (ffprobe -show_frames shows the same thing). I suspect
that this "worked" due to the best effort timestamp logic picking the
DTS over the decreasing PTS. Since this logic is in libavcodec (where
it probably shouldn't be), this can't be easily fixed. The timestamps
of the cavs samples are weird anyway, so I chose not to fix it.
Another strange thing is the timestamp handling in the video path of
process_input_packet (after the decode_video() call). It looks like
the code to increase next_dts and next_pts should be run every time
a frame is decoded - but it's needed even if output is skipped.
Fixes gapless decoding. Adjust skip_samples field correctly in case of DISCARDed audio frames.
Signed-off-by: Sasi Inguva <isasi@google.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This commit is initially largely based on commit 4426540 from Anton
Khirnov <anton@khirnov.net> and two following fixes (80fb19b and
fe7b21c) which were previously skipped respectively in 98e3153, c9ee36e,
and 7fe7cdc.
mpeg4-bsf-unpack-bframes FATE reference is updated because the bsf
filter now actually fixes the extradata (mpeg4_unpack_bframes_init()
changing one byte is now honored on the output extradata).
The FATE references for remove_extra change because the packet flags
were wrong and the keyframes weren't marked, causing the bsf relying on
these proprieties to not actually work as intended.
The following was fixed by James Almer:
The filter option arguments are now also parsed correctly.
A hack to propagate extradata changed by bitstream filters after the
first av_bsf_receive_packet() call is added to maintain the current
behavior. This was previously done by av_bitstream_filter_filter() and
is needed for the aac_adtstoasc bsf.
The exit_on_error was not being checked anymore, and led to an exit
error in the last frame of h264_mp4toannexb test. Restoring this
behaviour prevents erroring out. The test is still changed as a result
due to the badly filtered frame now not being written after the failure.
Signed-off-by: Clément Bœsch <u@pkh.me>
Signed-off-by: James Almer <jamrial@gmail.com>
This commit is largely based on commit 15e84ed3 from Anton Khirnov
<anton@khirnov.net> which was previously skipped in bbf5ef9d.
There are still a bunch of things raising codecpar related warnings that
need fixing, such as:
- the use of codec->debug in the interactive debug mode
- read_ffserver_streams(): it's probably broken now but there is no test
- lowres stuff
- codec copy apparently required by bitstream filters
The matroska references are updated because they now properly forward
the field_order (previously unknown, now progressive).
Thanks to James Almer for fixing a bunch of FATE issues in this commit.
Signed-off-by: Clément Bœsch <clement@stupeflix.com>
Signed-off-by: James Almer <jamrial@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>
Fixes regressions with stream copy and output timebase/fps being twice as fine as needed
Makes the timebase and ticks per frame handled identical which should make the
code easier to understand and work with. It does not solve the problem without
st->codec access
Suggested-by: Hendrik Leppkes
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Allows testing simple_idct12 correctness/bitexactness, as the sample
was generated using faani as idct.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
As Nvidia has put the most recent Video Codec SDK behind a double
registration wall, of which one needs manual approval of a lenghty
application, bundling this header saves everyone trying to use NVENC
from that headache.
The header is still MIT licensed and thus fine to bundle with ffmpeg.
Not bundling this header would get ffmpeg stuck at SDK v6, which is
still freely available, holding back future development of the NVENC
encoder.
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>
* commit 'a79aafd0b4d37eda6f15dc68e6509d4e815290c9':
movenc: Add a test for VFR with b-frames, with a duration change at a fragment end
Merged-by: Matthieu Bouron <matthieu.bouron@stupeflix.com>
* commit '1982d0cc561912d685a0c2dbe58bc19f50bae231':
fate: Add test for MTS2/MSS4
The timestamps differ because we use a more appropriate timebase.
Merged-by: Clément Bœsch <clement@stupeflix.com>
* commit '5b1409c75563b4a3aca113c34d09e3b5442de47f':
fate: Add test for MSS1
Test was already present, see 849e55e58e.
The merge removes the audio decoding present in our version and
simplifies the rules.
Merged-by: Clément Bœsch <clement@stupeflix.com>
