aboutsummaryrefslogtreecommitdiffstats
path: root/tests/fate/ffmpeg.mak
Commit message (Collapse)AuthorAgeFilesLines
* tests/fate: replace deprecated -vsync with -fps_modeAnton Khirnov2023-11-141-2/+2
|
* tests/fate/ffmpeg: replace deprecated -vbsf with -bsf:vAnton Khirnov2023-11-141-3/+3
|
* tests/fate/ffmpeg: add tests for -force_key_frames sourceAnton Khirnov2023-10-101-0/+20
|
* fate/ffmpeg: Add bitexact flag for ffmpeg-input-r testAndreas Rheinhardt2023-09-041-1/+1
| | | | | | | | 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>
* tests/fate/ffmpeg: silence the audio for fate-ffmpeg-streamloop-transcode-avAnton Khirnov2023-06-211-2/+5
| | | | | | | | 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.
* tests/fate: add a test for -streamloop with transcoding video+audioAnton Khirnov2023-06-191-0/+4
|
* tests/fate: rename ffmpeg-streamloop to ffmpeg-streamloop-copyAnton Khirnov2023-06-191-2/+2
| | | | | Makes it clear that this tests -streamloop with streamcopy, to distinguish it from further -streamloop tests added in future commits.
* tests/fate/ffmpeg: add tests for -max_error_rateAnton Khirnov2023-06-051-0/+6
|
* fate/tests/ffmpeg: use -idct simple for fate-ffmpeg-input-rAnton Khirnov2023-05-231-1/+1
| | | | Makes the test bitexact on non-x86_64.
* tests/fate/ffmpeg: add a test for input -r optionAnton Khirnov2023-05-221-0/+4
|
* tests/fate/ffmpeg: move a misplaced lineAnton Khirnov2023-05-221-2/+1
|
* ffmpeg: add video heartbeat capability to fix_sub_durationJan Ekström2023-02-031-0/+15
| | | | | | | | | | | | | | | | | | | 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>
* fate/ffmpeg: Set max_delay for shortest-subAndreas Rheinhardt2022-09-201-1/+1
| | | | | | | | | | 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>
* fate/ffmpeg: Use transcode instead of enc_dec in shortest-sub testAndreas Rheinhardt2022-09-201-2/+2
| | | | | | | | | | | | | | | 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>
* fftools/ffmpeg: rework -shortest implementationAnton Khirnov2022-07-231-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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.
* fate/ffmpeg: add a test for interleaving video+subsAnton Khirnov2022-07-231-0/+8
|
* fate/ffmpeg: Fix test requirementsAndreas Rheinhardt2022-05-281-74/+48
| | | | Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
* tests/fate-run: Remove temporary fate-lavf files if possibleAndreas Rheinhardt2022-05-061-0/+1
| | | | | | | | | | 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>
* tests/Makefile: Redo how to keep intermediate FATE-filesAndreas Rheinhardt2022-05-061-2/+2
| | | | | | | | | 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>
* fate: add a setts bsf testJames Almer2022-03-171-0/+2
| | | | Signed-off-by: James Almer <jamrial@gmail.com>
* tests: add test for ffmpeg's fix_sub_duration featureJan Ekström2022-01-241-0/+11
| | | | | | | | 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>
* fate/ffmpeg: Add test for autorotating videoAndreas Rheinhardt2022-01-221-0/+6
| | | | Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
* fate/ffmpeg: add missing samples dependency to fate-shortestJames Almer2022-01-161-1/+1
| | | | Signed-off-by: James Almer <jamrial@gmail.com>
* FATE: always pass -nostdin to ffmpegrcombs2021-12-221-1/+1
| | | | | This avoids making terminal config changes that may not be reverted properly during parallel testing.
* fate: use single thread for rawvideoLimin Wang2021-12-221-5/+5
| | | | Signed-off-by: Limin Wang <lance.lmwang@gmail.com>
* fate/ffmpeg: Fix requirements of shortest testsAndreas Rheinhardt2021-12-021-3/+17
| | | | | | | 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>
* fate/ffmpeg: Fix shortest testsAndreas Rheinhardt2021-12-021-2/+2
| | | | | | | | | | | | | | | | | | 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>
* fate/ffmpeg: add some more flags to the shortest testsJames Almer2021-12-011-5/+5
| | | | Signed-off-by: James Almer <jamrial@gmail.com>
* fate/ffmpeg: add missing bitexact flags to the shortest testsJames Almer2021-12-011-7/+8
| | | | | | Should fix fate failures on some targets. Signed-off-by: James Almer <jamrial@gmail.com>
* fate/ffmpeg: add tests for shortest optionJames Almer2021-12-011-0/+19
| | | | Signed-off-by: James Almer <jamrial@gmail.com>
* ac3enc_fixed: convert to 32-bit sample formatLynne2021-01-141-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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%
* fate: disable automatic conversions on many tests.Nicolas George2020-09-081-10/+10
| | | | | | | | | | 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.
* fate/ffmpeg: add test for time limited sub2video instanceJan Ekström2020-03-161-0/+10
| | | | | | | | | | | | 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.
* fate/ffmpeg: add a second, simple sub2video testJan Ekström2020-03-161-0/+9
|
* fate: Fix dependencies to sample files to use local pathsMartin Storsjö2019-12-121-16/+16
| | | | | | | 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>
* fate: add test for stream_loopGyan Doshi2019-09-051-0/+4
| | | | Checks that seek to start indeed seeks to start.
* tests: don't include TARGET_PATH in the sample path needlesslyHendrik Leppkes2019-04-191-1/+1
| | | | | | 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.
* tests/fate/ffmpeg: Check for apng codec for fate-copy-apng.Carl Eugen Hoyos2019-03-151-1/+1
| | | | The file has to be created first, fixes fate without zlib.
* Merge commit 'f8df5e2f31a5ba7b30a0e1caaaf5a03c753b3f9b'James Almer2019-03-141-0/+4
| | | | | | | * commit 'f8df5e2f31a5ba7b30a0e1caaaf5a03c753b3f9b': tests: Add a convenience function for video-only lavf tests Merged-by: James Almer <jamrial@gmail.com>
* fate: change fate-ffmpeg-attached_pics to encode to pcm_s16leMarton Balint2017-10-281-3/+3
| | | | | | | | 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>
* fate: fix ffmpeg-attached_pics test dependenciesMarton Balint2017-10-271-1/+1
| | | | Signed-off-by: Marton Balint <cus@passwd.hu>
* fate: add fate test for ticket #6375Marton Balint2017-10-271-0/+4
| | | | Signed-off-by: Marton Balint <cus@passwd.hu>
* fate: add fate test for ticket #6603Marton Balint2017-10-271-0/+4
| | | | Signed-off-by: Marton Balint <cus@passwd.hu>
* Merge commit '4141a5a240fba44b4b4a1c488c279d7dd8a11ec7'James Almer2017-10-031-1/+1
| | | | | | | * commit '4141a5a240fba44b4b4a1c488c279d7dd8a11ec7': Use modern avconv syntax for codec selection in documentation and tests Merged-by: James Almer <jamrial@gmail.com>
* ffmpeg options: Enable trailing ? for map_channelpkviet2017-08-251-0/+8
| | | | | | | | | | | 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>
* fate: Add fate-copy-trac3074Michael Niedermayer2017-06-301-0/+6
| | | | Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
* fate: add test for -time_base optionMichael Bradshaw2017-06-101-0/+6
| | | | | Signed-off-by: Michael Bradshaw <mjbshaw@google.com> Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
* Merge commit '043b0b9fb1481053b712d06d2c5b772f1845b72b'Clément Bœsch2017-03-241-2/+2
| | | | | | | | | * 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>
* fate/colorkey: disable audio stream.Nicolas George2016-11-131-1/+1
| | | | | | | 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.
* fate: add bsf tests for ticket 5927James Almer2016-11-041-0/+6
| | | | | Tested-by: Michael Niedermayer <michael@niedermayer.cc> Signed-off-by: James Almer <jamrial@gmail.com>
generated by cgit v1.2.3 (git 2.25.1) at 2025-08-06 17:12:24 +0000