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>
Segment starts on the seek timeline are derived from EXTINF durations,
so the first keyframe of the target segment can have a slightly lower
DTS than the snapped seek position, in which case the packet discard
loop would skip it and resume playback one segment late. Make sure we
start at the very beginning of the stream in this case.
Signed-off-by: Kacper Michajłow <kasper93@gmail.com>
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>
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.
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>
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.
Rate control: replace the integral servo with a stateless pressure
offset (exp2(-K*fill/R)) so a drained reservoir cannot wind up and
crater quality after loud stretches; slew-limit the final operating
lambda per frame (bits deviate instead, the reservoir absorbs); seed
the reservoir full at stream start; and track rate strain explicitly:
a long-frame lambda EMA against anchors that scale up when the achieved
distortion/mask ratio flags noise-class content (whose psy masks are
wholesale violated and lambda reads inflated), plus a lambda min-tracker
separating sustained starvation from transient spikes at a comfortable
operating point. The resulting pressure ramp gates every
pressure-adaptive tool from one place.
CPE budget pooling: solve both channels of a pair jointly under one
shared lambda against a pooled budget (NMRSlot defer/solve/commit)
instead of an equal per-channel split, which starved the mid/carrier
while the side gold-plated. Mono and VBR output are unchanged.
Transients: isolated onsets are coded uniformly finer across the short
run and repaid from steady stretches; dense-beat runs get a
starvation-scaled boost; a transition premask clamps START-frame
thresholds toward the previous long frame (an attack cannot mask
backwards).
Decision memory: marginal per-frame re-decisions oscillate audibly, so
every stateful choice now carries hysteresis - band zeroing, PNS
enter/leave with debounce (and near-masked bands staying noise until a
loudness guard), per-grid stereo mode banks that survive window
switches, and the short-TNS accept state.
TNS-covered bands price distortion by the synthesis filter's
re-amplification gain, so the trellis spends where noise will actually
be heard.
Replace the first-page packet scan with a direct use of the first
packet's duration as the timestamp offset (the Vorbis I spec
guarantees it produces no PCM output).
Track last_page_granule to correctly anchor EOS trimming regardless
of any initial granule offset. Track bos_pos to reinitialize parser
state at stream boundaries without resetting at page boundaries.
Restore the granule=0 workaround from ticket #3710 and handle
chained streams by clearing bos_pos when new headers are collected.
Vorbis I spec: https://xiph.org/vorbis/doc/Vorbis_I_spec.html
Rather than having two uops for the C reference code and a separate
uop for the x86 backend, we can just implement the C reference
using the same register-register assignment logic.
This results in only slightly worse codegen, since the compiler is
smart enough to eliminate the unnecessary copies.
The main downside is the fact that we need to allocate extra stack space
to back-fill the remaining (unused) pointers. In theory, we could do this
more efficiently by re-using unused *input* registers directly, but that
would require a lot of reliance on compile-time constant folding teasing
apart the entire register index tracking loop and duplicate check, so this
is a useful middle ground.
On platforms like x86, we still don't care about the PERMUTE vs
COPY distinction, but it's also harmless (2-3 extra lines to
instantiate the underlying MOVE macro on both uop variants).
Finally, we need to add a mask to the move uops to keep track of which
outputs are actually needed (especially for checkasm). Though it does result
in the rather unfortunate downside that it bloats the names of the uop
names.
The op lists change to reflect the new representation, e.g.:
rgb24 16x16 -> bgr24 16x16:
[ u8 +++X] SWS_OP_READ : 3 elem(s) packed >> 0
min: {0 0 0 _}, max: {255 255 255 _}
[ u8 +++X] SWS_OP_SWIZZLE : 2103
min: {0 0 0 _}, max: {255 255 255 _}
[ u8 XXXX] SWS_OP_WRITE : 3 elem(s) packed >> 0
('X' unused, 'z' byteswapped, '=' copied, '$' const, '+' integer, '0' zero)
translated micro-ops:
u8_read_packed_xyz
- u8_permute_zyxw
+ u8_permute_xyzw_txz_xzt
u8_write_packed_xyz
Signed-off-by: Niklas Haas <git@haasn.dev>
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>
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>
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.
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>
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>
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>
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.
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%
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%
This is in line with other encoders, and is needed by the matroska muxer
to properly signal end trimming samples.
gaplessenc-pcm-to-mov-aac changes as the mov muxer now uses this information
when writing the edit list, so the shorter frame duration is not needed.
Signed-off-by: James Almer <jamrial@gmail.com>
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>
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>
When writing edit lists and the last packet has skip samples side data to
report discard padding, for codecs that have a fixed frame size it's best to
always use the full packet duration to calculate track duration (and thus
bitrate in some cases) and let the edit list be the sole source of triming
information.
Signed-off-by: James Almer <jamrial@gmail.com>
A recent change affected what stream_remux() prints, and the CI didn't detect it
because it ran before doing a fast-forward rebase.
Signed-off-by: James Almer <jamrial@gmail.com>
Do not overwrite the output codecpar extradata after init(). Keep the currently active HEVC parameter sets in HEVCBSFContext instead.
When AV_PKT_DATA_NEW_EXTRADATA on an input packet carries hvcC extradata, convert it to Annex B, update the runtime extradata state, and remove the packet side data.
Also add a FATE test covering extradata reloads with hevc_mp4toannexb,hevc_metadata.
Also change the function to return 0 on error, this is a more sensible error
value for aspect ratios anyway.
Signed-off-by: Marton Balint <cus@passwd.hu>
This tests the previously untested alternative ASF demuxer, plus the metadata
is actually a bit different from what is returned by the historical ASF
demuxer.
Signed-off-by: Marton Balint <cus@passwd.hu>
Sometimes the duration itself reflects the remainder samples by being set to a
value lower than the frame size, but in other cases it's done only through
a discard padding value in side data.
Take the latter into account when calculating durations.
Fixes issue #23532.
Signed-off-by: James Almer <jamrial@gmail.com>
panboxiaosa
Kacper Michajłow
Timo Rothenpieler
Nicolas George
Nathan Lucas
James Almer
StachowiakDawid
Lynne
Romain Beauxis
Ramiro Polla
Vignesh Venkat
Niklas Haas
Michael NiedermayerandFable-5
Vladimir Panteleev
marcos ashton
jiangjie
Marton Balint