1
0
mirror of https://github.com/FFmpeg/FFmpeg.git synced 2024-12-18 03:19:31 +02:00
Commit Graph

81 Commits

Author SHA1 Message Date
Anton Khirnov
bab7f91c36 tests/fate/ffmpeg: add a test for the issue fixed in previous commit 2024-01-27 09:24:29 +01:00
Anton Khirnov
c316c4c77b fftools/ffmpeg: deprecate -filter_complex_script
It is equivalent to -/filter_complex.
2024-01-20 10:23:24 +01:00
Anton Khirnov
ae06111d74 fftools/ffmpeg_demux: implement -bsf for input
Previously bitstream filters could only be applied right before muxing,
this allows to apply them right after demuxing.
2024-01-19 17:54:10 +01:00
Zhao Zhili
641f8a71fb fate/h264: move mp4toannexb_ticket5927 test to fate-h264
Signed-off-by: Zhao Zhili <zhilizhao@tencent.com>
2023-11-22 19:42:15 +08:00
Anton Khirnov
a8d9d6b08d tests/fate: replace deprecated -vsync with -fps_mode 2023-11-14 18:18:26 +01:00
Anton Khirnov
23de85d1ec tests/fate/ffmpeg: replace deprecated -vbsf with -bsf:v 2023-11-14 18:18:26 +01:00
Anton Khirnov
a07b2f5f11 tests/fate/ffmpeg: add tests for -force_key_frames source 2023-10-10 12:41:31 +02:00
Andreas Rheinhardt
091c41794d fate/ffmpeg: Add bitexact flag for ffmpeg-input-r test
Fixes the test when the non-bitexact MMXEXT versions of
the hpeldsp functions are used.

Reviewed-by: Anton Khirnov <anton@khirnov.net>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
2023-09-04 14:38:56 +02:00
Anton Khirnov
83b3ff741b tests/fate/ffmpeg: silence the audio for fate-ffmpeg-streamloop-transcode-av
Fixed-point AAC decoder currently does not produce the same output on
all platforms. Until that is fixed, silence the audio stream using the
volume filter.

Also, actually use the aac_fixed decoder as was the original intent.
2023-06-21 10:07:41 +02:00
Anton Khirnov
47a14b542e tests/fate: add a test for -streamloop with transcoding video+audio 2023-06-19 09:48:55 +02:00
Anton Khirnov
90a26e75a4 tests/fate: rename ffmpeg-streamloop to ffmpeg-streamloop-copy
Makes it clear that this tests -streamloop with streamcopy, to
distinguish it from further -streamloop tests added in future commits.
2023-06-19 09:48:55 +02:00
Anton Khirnov
09af34dc91 tests/fate/ffmpeg: add tests for -max_error_rate 2023-06-05 16:15:04 +02:00
Anton Khirnov
4e521e6102 fate/tests/ffmpeg: use -idct simple for fate-ffmpeg-input-r
Makes the test bitexact on non-x86_64.
2023-05-23 13:54:10 +02:00
Anton Khirnov
8c0f516133 tests/fate/ffmpeg: add a test for input -r option 2023-05-22 17:10:44 +02:00
Anton Khirnov
900bb3f8e2 tests/fate/ffmpeg: move a misplaced line 2023-05-22 17:10:44 +02:00
Jan Ekström
9a820ec8b1 ffmpeg: add video heartbeat capability to fix_sub_duration
Splits the currently handled subtitle at random access point
packets that can be configured to follow a specific output stream.
Currently only subtitle streams which are directly mapped into the
same output in which the heartbeat stream resides are affected.

This way the subtitle - which is known to be shown at this time
can be split and passed to muxer before its full duration is
yet known. This is also a drawback, as this essentially outputs
multiple subtitles from a single input subtitle that continues
over multiple random access points. Thus this feature should not
be utilized in cases where subtitle output latency does not matter.

Co-authored-by: Andrzej Nadachowski <andrzej.nadachowski@24i.com>
Co-authored-by: Bernard Boulay <bernard.boulay@24i.com>

Signed-off-by: Jan Ekström <jan.ekstrom@24i.com>
2023-02-03 16:17:29 +02:00
Andreas Rheinhardt
6a8b3e7eb1 fate/ffmpeg: Set max_delay for shortest-sub
The aim of this test is to show the interleavement
of the file generated in the first pass; so make the
interleavement queue in the framecrc muxer in the second
pass as small as possible so that the framecrc muxer does not
fix wrong interleavement of the input file behind our backs.

Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
2022-09-20 18:32:28 +02:00
Andreas Rheinhardt
71364c54d4 fate/ffmpeg: Use transcode instead of enc_dec in shortest-sub test
enc_dec is designed for raw input and output and computes
the PSNR between these two. The input of the shortest-sub
test is the idx file of a vobsub sub+idx combination
and the output is the output of framecrc of said vobsub
subtitle muxed into Matroska together with a synthesized
video. Calculating the PSNR between these two files makes
no sense, therefore switch to a transcode test, where
the ref file file contains the output of framecrc directly,
making the interleavement better visible in the ref file
at the cost of a larger ref file (>400 lines).

Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
2022-09-20 18:32:28 +02:00
Anton Khirnov
4740fea7dd fftools/ffmpeg: rework -shortest implementation
The -shortest option (which finishes the output file at the time the
shortest stream ends) is currently implemented by faking the -t option
when an output stream ends. This approach is fragile, since it depends
on the frames/packets being processed in a specific order. E.g. there
are currently some situations in which the output file length will
depend unpredictably on unrelated factors like encoder delay. More
importantly, the present work aiming at splitting various ffmpeg
components into different threads will make this approach completely
unworkable, since the frames/packets will arrive in effectively random
order.

This commit introduces a "sync queue", which is essentially a collection
of FIFOs, one per stream. Frames/packets are submitted to these FIFOs
and are then released for further processing (encoding or muxing) when
it is ensured that the frame in question will not cause its stream to
get ahead of the other streams (the logic is similar to libavformat's
interleaving queue).

These sync queues are then used for encoding and/or muxing when the
-shortest option is specified.

A new option – -shortest_buf_duration – controls the maximum number of
queued packets, to avoid runaway memory usage.

This commit changes the results of the following tests:
- copy-shortest[12]: the last audio frame is now gone. This is
  correct, since it actually outlasts the last video frame.
- shortest-sub: the video packets following the last subtitle packet are
  now gone. This is also correct.
2022-07-23 11:53:19 +02:00
Anton Khirnov
d55b8dbcff fate/ffmpeg: add a test for interleaving video+subs 2022-07-23 11:53:19 +02:00
Andreas Rheinhardt
fab9130c7a fate/ffmpeg: Fix test requirements
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
2022-05-28 13:04:21 +02:00
Andreas Rheinhardt
e4563c2caf tests/fate-run: Remove temporary fate-lavf files if possible
The temporary fate-lavf files can easily be removed
if they are not needed as inputs for other tests (mainly
fate-seek-tests). This commit implements this.
The size of the remaining files decreases from 260890083B
to 79481793B.

Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
2022-05-06 05:39:25 +02:00
Andreas Rheinhardt
95cbd97cce tests/Makefile: Redo how to keep intermediate FATE-files
Extend the ordinary mechanism to signal KEEP for this.
This also allows to remove the keep-parameter from enc_dec,
transcode and stream_remux, so that several empty parameters
'""' could be removed.

Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
2022-05-06 05:39:25 +02:00
James Almer
c5628ae347 fate: add a setts bsf test
Signed-off-by: James Almer <jamrial@gmail.com>
2022-03-17 13:04:44 -03:00
Jan Ekström
0a83ecbf48 tests: add test for ffmpeg's fix_sub_duration feature
This long-existing feature calculates subtitle durations by keeping
it around until the following subtitle is decoded, and then utilizes
the following subtitle's pts as the end point of the previous one.

Signed-off-by: Jan Ekström <jan.ekstrom@24i.com>
2022-01-24 12:57:03 +02:00
Andreas Rheinhardt
741b5061ea fate/ffmpeg: Add test for autorotating video
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
2022-01-22 17:01:16 +01:00
James Almer
b1ef5882e3 fate/ffmpeg: add missing samples dependency to fate-shortest
Signed-off-by: James Almer <jamrial@gmail.com>
2022-01-16 00:32:52 -03:00
rcombs
0e7684e554 FATE: always pass -nostdin to ffmpeg
This avoids making terminal config changes that may not be reverted properly
during parallel testing.
2021-12-22 18:38:40 -06:00
Limin Wang
0e1f5f8871 fate: use single thread for rawvideo
Signed-off-by: Limin Wang <lance.lmwang@gmail.com>
2021-12-22 09:27:30 +08:00
Andreas Rheinhardt
b94db16bf5 fate/ffmpeg: Fix requirements of shortest tests
Fixes FATE failures if e.g. libavdevice is disabled.

Reviewed-by: James Almer <jamrial@gmail.com>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
2021-12-02 17:44:27 +01:00
Andreas Rheinhardt
4a6aece703 fate/ffmpeg: Fix shortest tests
The mpeg4 encoder is slice-threaded and its output depends upon
the number of threads used. Therefore all tests of this encoder
use a hardcoded number of threads (ENC_OPTS in fate-run.sh contains
"-threads 1"; only the vsynth%-mpeg4-thread tests override this
for the mpeg4 encoder, but they also use a hardcoded value to
be consistent across different systems); only the new shortest
and copy-shortest[12] (implicitly due to the sample used) tests
don't and this leads to FATE-failures.
Fix this by explicitly setting the thread count.

Also switch the shortest test to framecrc, because hashing side data
is itchy even though the side data used here (AV_PKT_DATA_QUALITY_STATS)
has a defined endianness.

Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
2021-12-02 11:33:32 +01:00
James Almer
6507e96e71 fate/ffmpeg: add some more flags to the shortest tests
Signed-off-by: James Almer <jamrial@gmail.com>
2021-12-01 22:22:19 -03:00
James Almer
686c7c132d fate/ffmpeg: add missing bitexact flags to the shortest tests
Should fix fate failures on some targets.

Signed-off-by: James Almer <jamrial@gmail.com>
2021-12-01 20:30:06 -03:00
James Almer
bb0a28560d fate/ffmpeg: add tests for shortest option
Signed-off-by: James Almer <jamrial@gmail.com>
2021-12-01 18:28:44 -03:00
Lynne
2d85e6e723
ac3enc_fixed: convert to 32-bit sample format
The AC3 encoder used to be a separate library called "Aften", which
got merged into libavcodec (literally, SVN commits and all).
The merge preserved as much features from the library as possible.

The code had two versions - a fixed point version and a floating
point version. FFmpeg had floating point DSP code used by other
codecs, the AC3 decoder including, so the floating-point DSP was
simply replaced with FFmpeg's own functions.
However, FFmpeg had no fixed-point audio code at that point. So
the encoder brought along its own fixed-point DSP functions,
including a fixed-point MDCT.

The fixed-point MDCT itself is trivially just a float MDCT with a
different type and each multiply being a fixed-point multiply.
So over time, it got refactored, and the FFT used for all other codecs
was templated.

Due to design decisions at the time, the fixed-point version of the
encoder operates at 16-bits of precision. Although convenient, this,
even at the time, was inadequate and inefficient. The encoder is noisy,
does not produce output comparable to the float encoder, and even
rings at higher frequencies due to the badly approximated winow function.

Enter MIPS (owned by Imagination Technologies at the time). They wanted
quick fixed-point decoding on their FPUless cores. So they contributed
patches to template the AC3 decoder so it had both a fixed-point
and a floating-point version. They also did the same for the AAC decoder.
They however, used 32-bit samples. Not 16-bits. And we did not have
32-bit fixed-point DSP functions, including an MDCT. But instead of
templating our MDCT to output 3 versions (float, 32-bit fixed and 16-bit fixed),
they simply copy-pasted their own MDCT into ours, and completely
ifdeffed our own MDCT code out if a 32-bit fixed point MDCT was selected.

This is also the status quo nowadays - 2 separate MDCTs, one which
produces floating point and 16-bit fixed point versions, and one
sort-of integrated which produces 32-bit MDCT.

MIPS weren't all that interested in encoding, so they left the encoder
as-is, and they didn't care much about the ifdeffery, mess or quality - it's
not their problem.

So the MDCT/FFT code has always been a thorn in anyone looking to clean up
code's eye.

Backstory over. Internally AC3 operates on 25-bit fixed-point coefficients.
So for the floating point version, the encoder simply runs the float MDCT,
and converts the resulting coefficients to 25-bit fixed-point, as AC3 is inherently
a fixed-point codec. For the fixed-point version, the input is 16-bit samples,
so to maximize precision the frame samples are analyzed and the highest set
bit is detected via ac3_max_msb_abs_int16(), and the coefficients are then
scaled up via ac3_lshift_int16(), so the input for the FFT is always at least 14 bits,
computed in normalize_samples(). After FFT, the coefficients are scaled up to 25 bits.

This patch simply changes the encoder to accept 32-bit samples, reusing
the already well-optimized 32-bit MDCT code, allowing us to clean up and drop
a large part of a very messy code of ours, as well as prepare for the future lavu/tx
conversion. The coefficients are simply scaled down to 25 bits during windowing,
skipping 2 separate scalings, as the hacks to extend precision are simply no longer
necessary. There's no point in running the MDCT always at 32 bits when you're
going to drop 6 bits off anyway, the headroom is plenty, and the MDCT rounds
properly.

This also makes the encoder even slightly more accurate over the float version,
as there's no coefficient conversion step necessary.

SIZE SAVINGS:
ARM32:
HARDCODED TABLES:
BASE           - 10709590
DROP  DSP      - 10702872 - diff:   -6.56KiB
DROP  MDCT     - 10667932 - diff:  -34.12KiB - both:   -40.68KiB
DROP  FFT      - 10336652 - diff: -323.52KiB - all:   -364.20KiB
SOFTCODED TABLES:
BASE           -  9685096
DROP  DSP      -  9678378 - diff:   -6.56KiB
DROP  MDCT     -  9643466 - diff:  -34.09KiB - both:   -40.65KiB
DROP  FFT      -  9573918 - diff:  -67.92KiB - all:   -108.57KiB

ARM64:
HARDCODED TABLES:
BASE           - 14641112
DROP  DSP      - 14633806 - diff:   -7.13KiB
DROP  MDCT     - 14604812 - diff:  -28.31KiB - both:   -35.45KiB
DROP  FFT      - 14286826 - diff: -310.53KiB - all:   -345.98KiB
SOFTCODED TABLES:
BASE           - 13636238
DROP  DSP      - 13628932 - diff:   -7.13KiB
DROP  MDCT     - 13599866 - diff:  -28.38KiB - both:   -35.52KiB
DROP  FFT      - 13542080 - diff:  -56.43KiB - all:    -91.95KiB

x86:
HARDCODED TABLES:
BASE           - 12367336
DROP  DSP      - 12354698 - diff:  -12.34KiB
DROP  MDCT     - 12331024 - diff:  -23.12KiB - both:   -35.46KiB
DROP  FFT      - 12029788 - diff: -294.18KiB - all:   -329.64KiB
SOFTCODED TABLES:
BASE           - 11358094
DROP  DSP      - 11345456 - diff:  -12.34KiB
DROP  MDCT     - 11321742 - diff:  -23.16KiB - both:   -35.50KiB
DROP  FFT      - 11276946 - diff:  -43.75KiB - all:    -79.25KiB

PERFORMANCE (10min random s32le):
ARM32 - before -  39.9x - 0m15.046s
ARM32 - after  -  28.2x - 0m21.525s
                       Speed:  -30%

ARM64 - before -  36.1x - 0m16.637s
ARM64 - after  -  36.0x - 0m16.727s
                       Speed: -0.5%

x86   - before - 184x -    0m3.277s
x86   - after  - 190x -    0m3.187s
                       Speed:   +3%
2021-01-14 01:44:12 +01:00
Nicolas George
f08e024ac7 fate: disable automatic conversions on many tests.
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.
2020-09-08 14:16:08 +02:00
Jan Ekström
c149f16db1 fate/ffmpeg: add test for time limited sub2video instance
Utilizes a subpicture sample with one decodable subpicture for the
test.

Based on a failing test case in reported by Michael in
https://ffmpeg.org/pipermail/ffmpeg-devel/2019-February/240398.html
which at the time had no test case for it.

Additionally, this is the first test case for the presentation
graphics format.
2020-03-16 19:35:17 +02:00
Jan Ekström
9c8a5fd57e fate/ffmpeg: add a second, simple sub2video test 2020-03-16 19:35:17 +02:00
Martin Storsjö
b85dcd8586 fate: Fix dependencies to sample files to use local paths
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>
2019-12-12 11:27:55 +02:00
Gyan Doshi
2b66c757d6 fate: add test for stream_loop
Checks that seek to start indeed seeks to start.
2019-09-05 23:23:24 +05:30
Hendrik Leppkes
a87774636b tests: don't include TARGET_PATH in the sample path needlessly
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.
2019-04-19 16:24:14 +02:00
Carl Eugen Hoyos
def18ac43b tests/fate/ffmpeg: Check for apng codec for fate-copy-apng.
The file has to be created first, fixes fate without zlib.
2019-03-15 01:41:22 +01:00
James Almer
ac4b5d8622 Merge commit 'f8df5e2f31a5ba7b30a0e1caaaf5a03c753b3f9b'
* commit 'f8df5e2f31a5ba7b30a0e1caaaf5a03c753b3f9b':
  tests: Add a convenience function for video-only lavf tests

Merged-by: James Almer <jamrial@gmail.com>
2019-03-14 14:59:45 -03:00
Marton Balint
859144fb1f fate: change fate-ffmpeg-attached_pics to encode to pcm_s16le
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>
2017-10-28 21:35:57 +02:00
Marton Balint
273f11969d fate: fix ffmpeg-attached_pics test dependencies
Signed-off-by: Marton Balint <cus@passwd.hu>
2017-10-27 02:48:28 +02:00
Marton Balint
61e25ab0bf fate: add fate test for ticket
Signed-off-by: Marton Balint <cus@passwd.hu>
2017-10-27 00:33:56 +02:00
Marton Balint
cbcb8f5a22 fate: add fate test for ticket
Signed-off-by: Marton Balint <cus@passwd.hu>
2017-10-27 00:33:56 +02:00
James Almer
a2b0602cda Merge commit '4141a5a240fba44b4b4a1c488c279d7dd8a11ec7'
* commit '4141a5a240fba44b4b4a1c488c279d7dd8a11ec7':
  Use modern avconv syntax for codec selection in documentation and tests

Merged-by: James Almer <jamrial@gmail.com>
2017-10-03 21:28:07 -03:00
pkviet
e0436ddaa4 ffmpeg options: Enable trailing ? for map_channel
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>
2017-08-25 23:41:11 +02:00
Michael Niedermayer
0b180d2066 fate: Add fate-copy-trac3074
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2017-06-30 03:03:08 +02:00