2830 Commits
Author SHA1 Message Date
Ayoub Nabil 5ba2a2c841 avcodec/ac3dec_fixed: fix dither at exponent 24
the fixed decoder discarded fractional coefficient data before the imdct, causing low-level coefficients to be truncated at large exponents. dithered bap 0 coefficients at exponent 24 were consequently quantized with a negative bias.

keep ac-3 coefficients in q2, compensate during windowing, and round the affected dither symmetrically. keep the forced e-ac-3 path in q0.

add a synthetic fate test covering the corrected output.
2026-08-10 22:01:35 +00:00
James Almer afd6b8451f libswresample/tests/rematrix: rewrite into a matrix printing tool
Signed-off-by: James Almer <jamrial@gmail.com>
2026-08-09 13:55:37 -03:00
Andreas Rheinhardt d7ce8c30e0 tests/checkasm/ttadsp: Also test ttaencdsp
Reviewed-by: Lynne <dev@lynne.ee>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
2026-08-09 17:22:57 +02:00
Andreas Rheinhardt ee89beda21 tests/checkasm: Add ttadsp test
Reviewed-by: Lynne <dev@lynne.ee>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
2026-08-09 17:10:52 +02:00
Romain Beauxis 23fccd657c avformat/mp3enc: keep trailing padding spanning several packets
The trailing padding is read from the AV_PKT_DATA_SKIP_SAMPLES side data of
every packet, overwriting the previous value, so only the last packet was
ever accounted for. A single packet holds at most one frame, which caps the
padding that can be written at 1152 + 528 + 1 samples.

LAME regularly reports more than that: gapless/gapless.mp3 carries 1984 and
comes out of a stream copy with 1681, decoding to 303 samples more than the
file it was copied from.

Accumulate instead, and add the decoder delay once the total is known.

Fixes: https://trac.ffmpeg.org/ticket/9755
2026-08-07 21:18:03 +00:00
Ayoub Nabil Boubagrat 5c395992f9 swresample/rematrix: handle top-back channels
top-back channels are currently left unaccounted and can be dropped during downmixing.

when the output retains top-front channels, follow IAMF 1.1.0 and fold top-back into top-front at 0.707. prefer this path over ear-level rear channels to preserve the height layer in x.1.4 to x.1.2 downmixes.

when no matching height output remains, map top-back to back or side channels, then fall back to front or mono outputs. handle top-back center separately and add direct tests for every matrix path.

Signed-off-by: Ayoub Nabil Boubagrat <237098474+ayoubnabil@users.noreply.github.com>
2026-08-07 02:33:33 +00:00
Ayoub Nabil fc02470c62 avfilter/f_ebur128: fix top-back channel weighting
ITU-R BS.1770 assigns a weight of 1.0 to height channels, but the
filter included the top-back channels in the 1.41 surround mask.

remove the top-back channels from that mask and add a FATE test for
the resulting loudness measurement.

fixes #23968.

Signed-off-by: Ayoub Nabil <237098474+ayoubnabil@users.noreply.github.com>
2026-08-06 00:30:00 +00:00
iSold Leo f9ed016d42 fate/filter-audio: add test for pan named input channel id rejection
"pan=stereo|FL=UNK" resolves to AVChannel id 768, which was used as an
index into a 64 element array before the previous commit.

UNK is used because it was the only one of the three reachable high ids
that exited with 0 before the fix; AMBI (1024) aborted and UNSD (512)
failed with an unrelated message, so a test that only checked for a
non-zero exit status would have passed before the fix as well.

For the same reason the test greps the error message on stderr instead of
only looking at the exit status: a crash also exits non-zero.

Signed-off-by: iSold Leo <s@qwqlog.com>
2026-08-05 19:37:11 +00:00
iSold Leo ec2a4105e2 swscale: avoid overflow in fast bilinear edge handling
The final edge clamp computes the source position using int
multiplication before shifting. With sufficiently wide inputs this
overflows, which may suppress the clamp and leave the last output
pixels interpolated with the padding byte.

Promote the multiplication to int64_t in the C, MMXEXT and VSX
implementations.

Add a regression test covering the rightmost pixel of a wide upscale,
which is wrong before this change on both the C and the MMXEXT path.

Fixes: signed integer overflow: 15 * 255918080 cannot be represented in type 'int'
Fixes: #21591
Signed-off-by: iSold Leo <s@qwqlog.com>
2026-08-05 19:34:30 +00:00
张永鹏 d09b0c5e17 fftools/ffmpeg_opt: return an error when refusing overwrite
Return EEXIST when output overwrite is refused so ffmpeg exits with a
non-zero status for -n and interactive no responses.

Signed-off-by: 张永鹏 <roc63@outlook.com>
2026-08-02 11:33:56 -05:00
panboxiaosa b57a3d028d avformat/mov: set video packet durations in presentation order
STTS sample deltas follow decode order, while AVPacket.duration is
defined as the interval to the next PTS in presentation order. Assigning
the deltas directly therefore produces incorrect packet durations for
VFR video with reordered frames.

After index construction and edit-list processing, sort samples by PTS
and set every duration with a following PTS from adjacent presentation
timestamps. This reorders STTS durations where possible and derives
intervals produced by the STTS/CTTS combination when necessary. Keep the
original timing table if allocation fails or timestamps are invalid.

Add FATE coverage for the official VFR H.264 sample and for a generated
MPEG-4 case whose presentation intervals are not a permutation of its
STTS deltas. Update the HEVC dts2pts CRA reference for the corrected
presentation-order durations.

Signed-off-by: panboxiaosa <panboxiaosa@gmail.com>
2026-07-29 23:49:33 +00:00
Kacper Michajłow 84bc576371 avformat/hls: set stream start_time for all playlist types
Rebase the first packet's timestamp onto the start of the segment list
regardless of playlist type, and report it as start_time. Previously
only EVENT playlists did this and live streams took start_time from the
first loaded segment at the live edge. Which is not correct as some
servers provide very deep (hours even) time shift in live playlist. The
evicted segments are already tracked by EXT-X-MEDIA-SEQUENCE. This
allows us to use all available segments, not only the future ones on
live playlist.

Also prefer PTS over DTS, as EXTINF durations and start_time are
presentation time.

Signed-off-by: Kacper Michajłow <kasper93@gmail.com>
2026-07-28 18:01:18 +02:00
Kacper Michajłow b449f86287 tests/fate/pcm: add fate-pcm_dvda test
Input is cut down version of https://samples.mplayerhq.hu/DVD-Audio/ats.AOB

Signed-off-by: Kacper Michajłow <kasper93@gmail.com>
2026-07-28 17:55:33 +02:00
Christopher Decker 94ff335d76 avformat/os_support: fix return value of win32_rename
The return value of MoveFileExW was not being correctly interpreted,
see https://learn.microsoft.com/en-us/windows/win32/api/winbase/nf-winbase-movefileexw.

On Windows a failed rename over the file: protocol now surfaces as failed to rename file %s to %s:
Operation not permitted (ff_rename, libavformat/avio.c:867) plus an AVERROR(EPERM) return, where previously the muxer
reported success and the user was left with a missing or stale output file and no diagnostic. This affects the
write-to-temp-then-rename paths in hlsenc, dashenc, hdsenc, smoothstreamingenc, segment, and img2enc (e.g. HLS/DASH
playlist updates, -write_temp_file, segment list finalization).

Also, added unit tests to exercise the rename function.

Signed-off-by: Christopher Decker <chris.decker08@gmail.com>
2026-07-28 04:59:01 +00:00
Timo Rothenpieler 2be4a694a3 Revert "fate: disable tests that depend on premultipled conversion"
This reverts commit 336b175c9f.
2026-07-27 20:34:37 +02:00
Timo Rothenpieler 7e1f4dc318 Reapply "avfilter/avfiltergraph: always retry format negotiation after auto-filters"
This reverts commit cd9d265344.
2026-07-27 20:13:00 +02:00
Nicolas George 336b175c9f fate: disable tests that depend on premultipled conversion
They can be re-enabled onece automatic conversion of
premultiplied alpha is fixed.
2026-07-27 16:07:38 +02:00
Nicolas George cd9d265344 Revert "avfilter/avfiltergraph: always retry format negotiation after auto-filters"
This reverts commit 978a0821ee.

The stability of the negotiation process has never been studied
with multiple rounds.

It has always been documented to users that scale was the filter
to insert to manage format conversions. A second filter for
a specific type of conversion should never have been added:
the proper fix for the issue that this commit tried to fix is
to give scale the ability to handle premultiplication too.
2026-07-27 10:27:27 +02:00
Nathan Lucas 7002e01c19 tests/fate/spdif: add TrueHD branch padding test
Add a small synthetic TrueHD sample that exercises large MAT padding in
spdifenc. The sample covers both the input_timing path with padding above
MAT_FRAME_SIZE / 2 and the output_timing discontinuity path.

It also verifies queued MAT frame emission when one input packet completes
more than one MAT frame.

Signed-off-by: Nathan Lucas <nlucasgit@gmail.com>
2026-07-25 20:12:16 -06:00
James Almer 9c8fc66ce1 tests/fate/matroska: add a test for Track Timescale handling
Signed-off-by: James Almer <jamrial@gmail.com>
2026-07-22 20:50:14 -03:00
Philip Langdale 2cf3f4d64d avutil/hwcontext_cuda: remove format allowlist, accept any pixel format
Remove the static supported_formats[] allowlist from hwcontext_cuda.
cuda_frames_get_constraints() now iterates all registered pixel formats
and includes every non-hwaccel, non-palette format. cuda_frames_init()
validates with av_pix_fmt_desc_get() instead of checking against the
allowlist.

Palette formats have a special-cased use of plane[1] for the palette itself
and it's not worth the hassle of supporting in an otherwise clean generic
copy loop.

Also add a roundtrip test that uploads a deterministic byte pattern to
CUDA, downloads it back, and verifies the data match. This tests all
non-hwaccel pixel formats. The test is gated on CONFIG_CUDA.
2026-07-22 21:42:40 +00:00
StachowiakDawid c182b1c09c Add new mode to mpdecimate video filter 2026-07-20 17:49:47 +00:00
Lynne 3a8a1a7c6a avcodec/aacpsy: fix the attack lookback
This fixes a 15+ year old bug.

The novelty check read its previous-frame envelope from
prev_energy_subshort after the sub-block loop had already overwritten
it with current-frame values, so early sub-blocks were compared
against the frame's own future instead of the past. That accident
suppressed short-period pitch trains well enough, but also suppressed
genuine onsets, and could not see pulse periods beyond ~12ms at all:
on quiet slow pulse trains (engine-idle buzz and the like) the ratio
test leaks sporadic isolated short excursions, each an audible click -
reported against this encoder since 2015, at every bitrate, and
immune to disabling every coding tool.

Keep a rolling two-frame history of HP sub-block peaks instead and
require an attack to tower over it: within ~12ms unconditionally
(pitch-rate trains), and within ~44ms when coming from steady
long-window state (slow trains, where an isolated short excursion can
only click). Dense irregular transients tower locally and reset
frames_since_short, so their block switching is untouched.
2026-07-19 23:50:55 +08:00
Lynne 9d0a4132ff avcodec/aacenc: rework NMR stereo decisions, retune coded bandwidth
Stereo decisions are made per band before quantization, from the psy
model's spectra, and carry cross-frame memory (EMA-smoothed statistics,
per-grid mode banks, leave-hysteresis) so the image holds instead of
churning:

  * M/S adopts content-driven and rate-free (side under half the mid);
    for mid-dominant bands M/S is simply the better coding at every
    rate, and a wandering L/R fraction reads as image instability.
  * I/S competes with M/S above 6.1 kHz instead of only seeing M/S
    rejects (which are exactly the wide bands it cannot render), and
    engages under SUSTAINED strain only: the pressure ramp gated by the
    lambda floor, so pressure spikes at a comfortable operating point
    cannot flood it onto content where coding the side is affordable
    and strictly better. Unengaged candidates fall back to M/S.
  * Pairs whose joint-tool candidacy fraction stays low decouple:
    block switching goes per-channel and M/S stops, matching how
    independent coding wins on diffuse decorrelated content.
  * PNS in a pair is reserved for clearly-wide bands (it renders
    uncorrelated noise per channel).

Block switching for a pair is decided through the psy window_pair()
hook so common_window survives transients.

The NMR rate-to-bandwidth table is retuned upward at >= 48 kbps/ch to
track the bandwidth strong encoders deliver; both quality metrics
improve on a 16-clip battery and wider regresses. Lower rates are
unchanged.
2026-07-19 20:41:53 +08:00
张永鹏 629a5b4b0f fftools: reject negative -t durations
Reject negative recording durations for input and output -t before they reach streamcopy or trim handling. Keep -t 0 accepted.

Signed-off-by: 张永鹏 <roc63@outlook.com>
2026-07-18 12:04:01 -05:00
Romain Beauxis de038122a3 tests/fate/ogg-vorbis.mak: add test to show packets initial padding and
side data.
2026-07-18 15:48:34 +00:00
Romain Beauxis 9dcb14d982 tests/fate/id3v2.mak: add test for XXX lang in COMM and USLT frames, fix
comment for -sort test.
2026-07-18 09:59:57 -05:00
Vignesh Venkat 5002b55ce7 fate/mov: Add fate test for t35 cdsc track
Signed-off-by: Vignesh Venkat <vigneshv@google.com>
2026-07-14 15:50:30 -03:00
Michael NiedermayerandFable-5 cff4a69294 avfilter: add latticepal filter
Convert packed 24/32-bit RGB/BGR/RGBA/BGRA input to PAL8 using a per-frame palette
whose colors are placed on a face-centered cubic lattice (realized as the
scaled D3/D4 checkerboard lattice), with a user-supplied density controlling
the number of lattice steps spanning one color axis.

Only lattice points actually used by a frame enter its palette; if a frame
needs more than 256 of them, the filter will itearatively drop palettte
entries and reassign affected pixels until 256 color remain

lookup uses the Conway-Sloane rounding algorithm. Supported dithering
modes: none, ordered 8x8 bayer (swscale), Cluster & Void blue noise and
Floyd-Steinberg error diffusion.

Co-Authored-by: Fable-5
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2026-07-13 18:16:18 +00:00
Jun Zhao a6ed750664 avformat/utils: fix ff_mkdir_p() swallowing intermediate mkdir errors
When creating nested directories (e.g. /a/b/c), a genuine mkdir()
failure for an intermediate component was overwritten by later
attempts, making the original failure harder to diagnose.

Stop immediately on intermediate errors other than EEXIST, preserving
errno for the caller. Existing path components remain non-fatal, as
required by mkdir -p semantics. Add a regression test for creating a
child below an existing parent directory.

Signed-off-by: Jun Zhao <barryjzhao@tencent.com>
2026-07-12 00:02:40 +00:00
James Almer c16840b826 avcodec/dca_xll: export downmix matrix coefficients
Only for primary channel sets if they embeed coefficients for now.

$ ./ffmpeg -downmix stereo -i $fate-suite/dts/dcadec-suite/xll_51_24_48_768.dtshd -y native.wav

Before:
$ ./ffmpeg -i $fate-suite/dts/dcadec-suite/xll_51_24_48_768.dtshd -af aresample,aformat=channel_layouts=stereo -y swr.wav
[...]
[Parsed_aresample_0] [SWR] Using fltp internally between filters
[Parsed_aresample_0] [SWR] Matrix coefficients:
[Parsed_aresample_0] [SWR] FL: FL:0.414214 FR:0.000000 FC:0.292893 LFE:0.000000 SL:0.292893 SR:0.000000
[Parsed_aresample_0] [SWR] FR: FL:0.000000 FR:0.414214 FC:0.292893 LFE:0.000000 SL:0.000000 SR:0.292893
[Parsed_aresample_0] ch:6 chl:5.1(side) fmt:s32p r:48000Hz -> ch:2 chl:stereo fmt:s16 r:48000Hz
[...]

$ tests/tiny_psnr.exe native.wav swr.wav
stddev:21178.05 PSNR:  9.81 MAXDIFF:49613 bytes:     8192/     8192

After:
$ ./ffmpeg -i $fate-suite/dts/dcadec-suite/xll_51_24_48_768.dtshd -af aresample,aformat=channel_layouts=stereo -y swr.wav
[...]
[Parsed_aresample_0] [SWR] Using fltp internally between filters
[Parsed_aresample_0] [SWR] Custom matrix coefficients:
[Parsed_aresample_0] [SWR] FL: FL:-0.100006 FR:0.000000 FC:0.000000 LFE:0.000000 SL:-0.707092 SR:0.000000
[Parsed_aresample_0] [SWR] FR: FL:0.000000 FR:-0.100006 FC:0.000000 LFE:0.000000 SL:0.000000 SR:-0.707092
[Parsed_aresample_0] ch:6 chl:5.1(side) fmt:s32p r:48000Hz -> ch:2 chl:stereo fmt:s16 r:48000Hz
[...]

$ tests/tiny_psnr.exe native.wav swr.wav
stddev:    0.71 PSNR: 99.22 MAXDIFF:    1 bytes:     8192/     8192

Signed-off-by: James Almer <jamrial@gmail.com>
2026-07-11 19:32:17 -03:00
James Almer 57708daf61 avcodec/dca_core: export downmix matrix coefficients
$ ./ffmpeg -downmix stereo -i $fate-suite/dts/dcadec-suite/core_51_24_48_768_1.dtshd -y native.wav

Before:
$ ./ffmpeg -i $fate-suite/dts/dcadec-suite/core_51_24_48_768_1.dtshd -af aresample,aformat=channel_layouts=stereo -y swr.wav
[...]
[Parsed_aresample_0] [SWR] Using fltp internally between filters
[Parsed_aresample_0] [SWR] Matrix coefficients:
[Parsed_aresample_0] [SWR] FL: FL:0.414214 FR:0.000000 FC:0.292893 LFE:0.000000 SL:0.292893 SR:0.000000
[Parsed_aresample_0] [SWR] FR: FL:0.000000 FR:0.414214 FC:0.292893 LFE:0.000000 SL:0.000000 SR:0.292893
[Parsed_aresample_0] ch:6 chl:5.1(side) fmt:fltp r:48000Hz -> ch:2 chl:stereo fmt:s16 r:48000Hz
[...]

$ tests/tiny_psnr.exe native.wav swr.wav s16
stddev:18907.06 PSNR: 10.80 MAXDIFF:51590 bytes:     8192/     8192

After:
$ ./ffmpeg -i $fate-suite/dts/dcadec-suite/core_51_24_48_768_1.dtshd -af aresample,aformat=channel_layouts=stereo -y swr.wav
[...]
[Parsed_aresample_0] [SWR] Using fltp internally between filters
[Parsed_aresample_0] [SWR] Custom matrix coefficients:
[Parsed_aresample_0] [SWR] FL: FL:-0.100006 FR:0.000000 FC:0.000000 LFE:0.000000 SL:-0.707092 SR:0.000000
[Parsed_aresample_0] [SWR] FR: FL:0.000000 FR:-0.100006 FC:0.000000 LFE:0.000000 SL:0.000000 SR:-0.707092
[Parsed_aresample_0] ch:6 chl:5.1(side) fmt:fltp r:48000Hz -> ch:2 chl:stereo fmt:s16 r:48000Hz
[...]

$ tests/tiny_psnr.exe native.wav swr.wav
stddev:    0.02 PSNR:129.44 MAXDIFF:    1 bytes:     8192/     8192

Signed-off-by: James Almer <jamrial@gmail.com>
2026-07-11 19:32:17 -03:00
Frank Plowman 265d39e551 lavc/hevc: Always generate missing refs for foll pictures
Since bc1a3bfd2c, missing reference
pictures are not replaced with generated pictures unless `-flags
show_corrupt` is used.  For `ST_FOLL`/`LT_FOLL` reference pictures,
however, generation of missing references is required by the spec per
8.3.3.  We should not require the `show_corrupt` flag to be used in
order to be spec-conformant, so this patch removes the `show_corrupt`
behaviour added in bc1a3bfd2c for
`ST_FOLL`/`LT_FOLL` references and instead unconditionally generates
reference pictures in these sets when unavailable

Fixes: NUT_A_ericsson_5
Fixes: RPS_D_ericsson_6
2026-07-11 20:36:31 +00:00
Romain Beauxis 4641be96e5 fftools/ffmpeg_mux_init.c: delete encoder-specific metadata when re-encoding. 2026-07-11 17:14:43 +00:00
Romain Beauxis c1be9b4a81 Add FATE tests for stale metadata during re-encoding. 2026-07-11 17:14:43 +00:00
James Almer 4eb56054c2 fftools/ffmpeg_demux: merge LCEVC enhancement payloads into the base stream
This is a temporary implementation until support for bitstream filter graphs is
generically added to the scheduler.

The dual track tests are as such changed so their output is not affected by builds
with liblcevc-dec enabled.

Signed-off-by: James Almer <jamrial@gmail.com>
2026-07-06 13:27:29 -03:00
Vladimir Panteleev e058e8c0c5 vf_photosensitivity: fix metadata when duplicating frames
When the blending factor was calculated to be 0, the hot path skipped
updating variables later emitted in metadata.

Fix the flow to ensure they are populated consistently.

Extend the FATE test suite to check metadata output.
2026-07-05 00:41:30 +00:00
Vladimir PanteleevandCodex CLI dd057bd856 fate/filter-video: add photosensitivity blend test
Add a small self-contained lavfi test for the photosensitivity filter's
blend path. The graph produces one black RGB frame followed by two white
RGB frames, then runs:

    photosensitivity=frames=2:threshold=95:blend=0.5

The threshold is chosen so that the first black-to-white transition
barely exceeds the detector limit. A full 8x8 RGB grid transition has
badness 64 * 3 * 255 = 48960, while threshold=95 gives 48640 for the
first checked history window. That forces the filter into the blend
branch without needing any external sample.

With blend=0.5, the runtime factor is 48640 / 48960 * 0.5, or about
0.4967. The fixed-point blender truncates this to an input-frame weight
of 127/256, so blending black toward white produces an RGB component
value of 126. The reference therefore records black, the blended gray
frame, and then the following accepted white frame.

This makes the otherwise cryptic CRCs tie directly to the blend
calculation and history update behavior.

Co-authored-by: Codex CLI <noreply@openai.com>
2026-07-05 00:41:30 +00:00
marcos ashton 2f779272e0 libavformat/img2enc: add update_filemtime option
Add a new boolean option -update_filemtime to the image2 muxer that
sets each output file's modification time based on the creation_time
metadata plus the frame's PTS offset.

This is useful when extracting frames from dashcam or action camera
footage where wall-clock timestamps should be preserved on the output
files, allowing photo management tools to sort frames by capture time
without post-processing.

The option requires creation_time metadata to be set (via -metadata
creation_time=...). If not present, a warning is logged and the
option is silently disabled. When PTS is unavailable, the creation
time is used as-is without frame offset.

Uses utimes() on POSIX and _utime() on Windows to set file timestamps
with microsecond and second precision respectively.

Includes a FATE roundtrip test that writes frames with a known
creation_time, reads them back using the demuxer's -ts_from_file
option, and verifies the PTS values match the expected timestamps.

Closes: https://code.ffmpeg.org/FFmpeg/FFmpeg/issues/22537
Signed-off-by: marcos ashton <marcosashiglesias@gmail.com>
2026-07-03 19:47:10 +00:00
James Almer a37171bed5 tests/fate/hevc: add scale filter dependency to fate-hevc-bsf-mp4toannexb-new-extradata
Should fix failures when said filter is not compiled in.

Signed-off-by: James Almer <jamrial@gmail.com>
2026-07-03 15:43:12 -03:00
James Almer 9011e7083e tests/fate/ffmpeg: remove audio stream from fate-autorotate
It's not relevant to the test, and it's often affected by unrelated changes to the
ac3 encoder.

Signed-off-by: James Almer <jamrial@gmail.com>
2026-07-03 09:53:23 -03:00
James Almer d8c1dd4233 tests/fate/ac3: add a test to check correct encoder sample output
Signed-off-by: James Almer <jamrial@gmail.com>
2026-07-02 17:53:21 -03:00
James Almer b5bfb7716c avcodec/ac3enc: drain the buffered samples at the end of encoding
When encoding a stream with an amount of samples multiple of a block, the last
the last 256 samples would be lost as the encoders were not marked as
AV_CODEC_CAP_DELAY.

This can be easily reproduced with:

ffmpeg -f lavfi -i sine -ac 2 -af atrim=start_sample=0:end_sample=4608 -c:a eac3 -f framecrc -

Signed-off-by: James Almer <jamrial@gmail.com>
2026-07-02 17:53:20 -03:00
Andreas Rheinhardt cd9e6996af swscale/uops: Move stuff for generating uops_macros.h out
It is not used for normal builds and is more an auxiliary
dev tool; move the code into a new file, uops_macros_gen.c
to be built as a DEVPROG.

Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
2026-07-02 20:39:29 +02:00
marcos ashton 784ab15589 tests/fate/libavutil: add FATE test for video_hint
Test av_video_hint_alloc with 0, 1, and 4 rects, and
av_video_hint_create_side_data. Verifies that av_video_hint_rects
and av_video_hint_get_rect return pointers consistent with
rect_offset and rect_size, write/read-back of rect coordinates,
both hint type values, and OOM paths via av_max_alloc.

Coverage for libavutil/video_hint.c: 0.00% -> 82.05%

The remaining uncovered lines are the nb_rects overflow guard and
the av_buffer_create / av_frame_new_side_data_from_buf failure
cleanup paths, which av_max_alloc() cannot reach since it forces
the first allocation to fail.
2026-07-02 13:48:58 +00:00
marcos ashton 7ea44ca6a6 tests/fate/libavutil: add FATE test for mastering_display_metadata
Test all 5 public functions: av_mastering_display_metadata_alloc,
av_mastering_display_metadata_alloc_size, the create_side_data
variant, av_content_light_metadata_alloc, and its create_side_data
variant. Verifies the {0,1} rational defaults set by get_defaults(),
write/read-back of HDR metadata fields, frame side data attachment
for both mastering display and content light metadata, and OOM
paths via av_max_alloc.

Coverage for libavutil/mastering_display_metadata.c: 86.49% -> 100.00%
2026-07-02 13:48:58 +00:00
marcos ashton 324617b26f tests/fate/libavutil: add FATE test for dovi_meta
Test av_dovi_alloc, av_dovi_metadata_alloc, and av_dovi_find_level.
Verifies that the four inline offset-based accessors (get_header,
get_mapping, get_color, get_ext) return pointers consistent with
the offset fields, that find_level returns the first matching ext
block or NULL for a missing level, and OOM paths via av_max_alloc.

Coverage for libavutil/dovi_meta.c: 63.16% -> 100.00%
2026-07-02 13:48:58 +00:00
James Almer a693a4cd80 tests/fate/generic-tags: use codec copy instead of encoding
Using ac3_fixed output was not enough as there's float to int conversion due to
the fact the mp3float decoder is used.
Instead of playing with codec combinations, just remove encoding from the test
altogheter. Mov supports muxing mp3 just fine.

Signed-off-by: James Almer <jamrial@gmail.com>
2026-07-01 14:35:14 -03:00
James Almer d7d2d9d539 tests/fate/generic-tags: use ac3_fixed encoder instead of float aac
There's no guarantee the aac encoder will be bitexact in its output across platforms.
Use ac3_fixed instead of aac_fixed while at it, so the aac encoder can get improvements
without affecting this test.

Signed-off-by: James Almer <jamrial@gmail.com>
2026-07-01 13:52:48 -03:00
Lynne 38161819e7 fate: update AAC encoder tests 2026-07-01 21:05:27 +09:00