* craft-audio: a program config element in the shape of a channel configuration gets its speakers
An AAC stream with channel configuration 0 whose program config element lists
a centre, a front pair and then exactly the elements of configuration 3 to 6
decoded as C L R Ls Rs LFE and was reported as an unordered layout, so a
caller reading it in WAV order sent the centre to the left speaker. Such a
stream now has the configuration's layout and WAV order. Any other element
list keeps the decoded order, as before.
Ported from storytold/filmcraft#4280 (aac_stream_order in filmcraft-codecs).
* craft-audio: the ADTS program config oracle expects 5.1 in WAV order
ffmpeg 9 writes its 5.1 program config element centre first, so the stream
now has 5.1's speakers: each decoded channel must match ffmpeg's channel of
the same index. An element list in another order keeps the old check.
---------
Co-authored-by: mpratsaki <110930793+mpratsaki@users.noreply.github.com>
FilmCraft's MPEG audio Layer I/II decoder as `craft_audio::mpa` behind
`mpa-decode`, kept over symphonia's: the same quality where both
decode, and it also decodes the MP2 files ffmpeg writes by default
(mono at 384 kbit/s), which symphonia rejects. `TrackDecoder` and the
native read path take MP1/MP2 in MP4, Matroska and raw streams;
symphonia keeps MP3 only, so its `mp1` and `mp2` features are dropped.
Co-authored-by: Claude Opus 5.5 <noreply@anthropic.com>
Co-authored-by: Dylan M. Taylor <dylan@dylanmtaylor.com>
* craft-audio: AC-3 and E-AC-3 decoder
FilmCraft's AC-3 decoder as `craft_audio::ac3` behind `ac3-decode`:
every AC-3 channel mode and frame size, and E-AC-3 independent
substream 0 with spectral extension, in WAV order without downmix.
`TrackDecoder` and the native read path take AC-3 and E-AC-3 in MP4,
Matroska and raw streams.
Not bit-exact: zero-bit mantissas are filled with decoder-specific
dither (A/52 §7.3.4), so tests compare at 65-94 dB SNR on tones and
27-48 dB on white noise. 7.1 E-AC-3 decodes as its 5.1 core.
Co-Authored-By: Claude Opus 5.5 <noreply@anthropic.com>
* craft-audio: review fixes, round 1 (#447)
An AC-3 / E-AC-3 packet is as long as all the primary syncframes in it.
Before the first decode, TrackDecoder::frames_per_packet gave 1536 for any
AC-3 track, and conceal used the first syncframe's header only. An E-AC-3
stream of one-, two- or three-block syncframes then got the wrong MP4 length
and seek positions and the wrong Matroska positions. A packet that groups
several such syncframes concealed at a fraction of its length.
Seeks in AC-3 and E-AC-3 now have to match the whole-file decode bit for bit,
as every other codec's do. The dither is seeded from each syncframe, so the
old 25 dB bound was looser than the decoder needs.
Co-Authored-By: Claude Opus 5.5 <noreply@anthropic.com>
* craft-audio: review fixes, round 2 (#447)
- AC-3 / E-AC-3 packets that decode to different lengths (in the packets
read ahead, or any decoded since) have no frames_per_packet, so a
Matroska seek finds its packet by timestamp rather than by index times
the first packet's length.
- E-AC-3 syncframes of fewer than six blocks may reuse exponents from any
syncframe before them: once one is seen, a seek decodes from the start
of the stream instead of 1536 frames before the target.
Co-Authored-By: Claude Opus 5.5 <noreply@anthropic.com>
* craft-audio: review fixes, round 3 (#447)
- Matroska E-AC-3: the first seek reads every packet's syncframe headers (only the headers,
header stripping undone) and places packets by the frames their primary syncframes hold.
Packets that started to differ in length after the eight read ahead were placed as the first
packet's length times their index.
- A packet whose later syncframe fails no longer sets the length lost packets are concealed at,
nor marks the packets' lengths as differing: it decoded only in part.
Co-Authored-By: Claude Opus 5.5 <noreply@anthropic.com>
* Merge main; E-AC-3 in Matroska: read packet headers only up to the seek target, and count a packet of dependent syncframes as zero frames
---------
Co-authored-by: Claude Opus 5.5 <noreply@anthropic.com>
Co-authored-by: Dylan M. Taylor <dylan@dylanmtaylor.com>
* craft-audio: Opus decoder
FilmCraft's `filmcraft-opus` decoder as `craft_audio::opus` behind
`opus-decode`: SILK, hybrid and CELT, every frame size, PLC and FEC,
multistream (`Decoder`) and single stream (`StreamDecoder`), 8-48 kHz
output.
- `TrackDecoder` decodes Opus; mono and stereo without an `OpusHead`
(RTP, MPEG-TS) get a default one. Lost packets use Opus's own
concealment.
- The native read path takes Ogg Opus (pre-skip and end trimming per
RFC 7845), and Opus in Matroska/WebM and MP4, where packet starts
after a seek come from the packets' own lengths.
The RFC 8251 conformance test runs all 36 vectors.
`scripts/fetch-opus-vectors.sh` downloads them (SHA-256 checked) and
the CI test job caches them.
Co-Authored-By: Claude Opus 5.5 <noreply@anthropic.com>
* craft-audio: review fixes, round 1 (#445)
- CELT concealment keeps its pitch in its own field. It reused postfilter_period_old, which the
next good frame's post-filter cross-fades from.
- Matroska Opus seeks: a failed packet read no longer caches a wrong start; a one-byte packet
is read as one byte.
- A malformed OpusHead in Ogg is reported as malformed instead of falling back.
- The libopus oracle tests skip through the testkit, so CRAFT_MEDIA_REQUIRE_ORACLES fails them.
- Decoder::reset documents that it re-arms pre-skip trimming.
- fetch-opus-vectors.sh retries its size probe.
- CI: the vectors are fetched after ffmpeg is installed, and apt's timeout runs as root so a
stalled apt-get is killed and frees the dpkg lock.
Co-Authored-By: Claude Opus 5.5 <noreply@anthropic.com>
* craft-audio: review fixes, round 2 (#445)
- CELT: the noise-based concealment sets skip_plc, as libopus does, so
pitch-based concealment waits for two good frames in a row instead of
extending history that holds noise.
- SILK: decode_only_middle is per call, as in libopus. A lost or mono
frame no longer reuses the mid-only flag from an earlier frame.
- A damaged Opus packet is concealed at its own length (from its TOC and
framing) rather than the previous packet's.
Co-Authored-By: Claude Opus 5.5 <noreply@anthropic.com>
---------
Co-authored-by: Claude Opus 5.5 <noreply@anthropic.com>
Co-authored-by: Dylan M. Taylor <dylan@dylanmtaylor.com>
From wanikhawar's review:
- AudioSpecificConfig::parse reads backward-compatible SBR and PS signalling
(a 0x2b7 sync extension after the config), which craft-media's Matroska
demuxer writes for legacy HE-AAC IDs. Without lookahead those streams decoded
as their core.
- An MP4 media edit after one empty edit is applied: its priming skip and its
end. The empty edit's delay is not added as silence.
- A packet lost whole is concealed as long as the last one decoded, so a stream
of multi-frame ADTS packets keeps its timing.
- ADTS-framed MP4 samples keep their sample table's timing, instead of every
sample being read at open to look for several frames in one.
- AdtsHeader::write refuses an object type, sampling frequency index or channel
configuration its fields cannot hold; it masked them into other values.
- AudioSpecificConfig::lc no longer panics for a sampling frequency index with no
rate, and the decoder refuses such a config.
- AudioSpecificConfig::to_bytes refuses channel configuration 0 without a program
config element, which would not parse back.
- Upsample2x starts each channel's history afresh when the channel count
changes.
Co-Authored-By: Claude Opus 5.5 <noreply@anthropic.com>
Reading:
- an element of another type or a tag a program config does not list is
refused, not written to another channel; only an LFE coded as an SCE
is still allowed, as ffmpeg allows it
- an explicit sampling rate of 0 is refused
- a packet's later ADTS frames are searched for SBR
- a multi-frame ADTS packet that fails leaves the decoder as it was, and
one whose frames change the channel count is an error
- the whole ADTS frames before a damaged end of a packet decode
- a failed read of a raw stream's first frames is an I/O error
- an MP4 sample of several ADTS frames past the lookahead keeps its length
- an Ogg file with leading bytes keeps its track order and mask comments
Writing:
- ProgramConfig::write, AudioSpecificConfig::to_bytes and
AdtsHeader::write refuse what their fields cannot hold
Tests:
- the routing test checks one alignment for the ffmpeg version under test
- a default decode past an Ogg stream symphonia has no track for
Co-Authored-By: Claude Opus 5.5 <noreply@anthropic.com>
Two tests from round 1 failed on ffmpeg before 9, including CI's 8.0. Neither was
a decoding fault:
- ffmpeg 9 ends an MP4 with no edit list at its sample durations and earlier
versions present the whole last packet, so the length is checked against the
file's known duration, not ffmpeg's decode.
- The order ffmpeg gives channels with -aac_pce differs by version, so each decoded
channel must match a different one of ffmpeg's, whatever the order.
Co-Authored-By: Claude Opus 5.5 <noreply@anthropic.com>
- TrackDecoder takes an ADTS stream's format from the one most of the lookahead
shares, as the file path did, so one damaged first header does not fail every
packet of an MPEG-TS stream. open_adts now just asks it.
- MP4 samples of several ADTS frames keep the sample table's timing; a packet
count of one frame each truncated them silently.
- Sample durations past what the packets decode to no longer over-report the
length.
- split_adts_upto is not behind aac-decode, as split_adts is not: an aac-encode
build in the next PR did not compile.
Removes an unreachable ID_END check and open_adts's unused core rate.
Co-Authored-By: Claude Opus 5.5 <noreply@anthropic.com>
Corrects round 2:
- ADTS headers are compared with the stream's format only when the first header
gave it. With extradata the format comes from there, so ADTS in MP4 whose
headers say something else decodes again; every packet had been concealed.
- A stream's only program config element is found in its first frames even when
that frame's header is damaged.
- The test of wrong sample durations patched the second run's count, not the
delta, so "durations of one" changed nothing. It now patches the delta, and
checks a file was changed and that seeks land on the same audio.
MP4 seeks place a packet by its index when every packet decodes to the same
length, as decoding does; they used the durations round 2 had stopped trusting
for the length, and returned the wrong audio. A last delta of 0 is overridden too.
Docs: what TrackInfo says for raw ADTS, that opening one reads the whole file, and
which codecs symphonia reads.
Co-Authored-By: Claude Opus 5.5 <noreply@anthropic.com>
Corrects round 1:
- A multi-block ADTS frame after the stream has started is concealed at its
whole length; round 1's filter skipped it, shifting the audio after it.
- Sample durations only trim an MP4's last packet: wrong ones (some count a
HE-AAC packet's core frames) were cutting audio. Durations of 0 with an edit
use the edit.
Raw ADTS packets now keep their headers, so TrackDecoder alone handles format
changes, multi-block frames and concealment, for apps as for files, and nested
ADTS payloads are not split again. Headers come from the frame scan's reads,
not one read per frame. The stream's format is the one most of its first frames
share, so one damaged header does not silence the file. The ID3 walk and a
packet's ADTS split are bounded.
AAC:
- HE-AAC is delayed by the SBR filterbanks' 481 core samples; it played 19.5 ms
early against ffmpeg in every container.
- A failed access unit leaves the decoder's state as it was, so the next packet
decodes as if it had not been there.
- reset restarts the noise generator, so a seek into PNS repeats.
- A signalled rate that is not twice the core's plays the core; it reported the
signalled rate with the core's samples.
The decoder oracle's tolerance tightens from 1e-3 to 1e-5. Tests cover HE-AAC.
Co-Authored-By: Claude Opus 5.5 <noreply@anthropic.com>
Raw ADTS and TrackDecoder:
- Channel configuration 0 (a PCE in the stream, as ffmpeg's -aac_pce and
AudioToolbox's 7.x write) decodes; it was refused.
- Frames with several raw data blocks are refused; only their first block
was decoded, silently losing the rest.
- A frame that changes the sample rate, profile or channel configuration is
concealed; it played at the wrong speed.
- A source that fails during the frame scan is AudioError::Io; it looked
like the end of the file.
- A stream after an ID3 tag longer than the file head is found.
- conceal counts all the ADTS frames of a packet, and a packet of more than
1024 frames is refused before it is decoded into memory.
AAC decoder:
- Indexed channel configurations place elements by arrival order, not tag,
as ffmpeg does; tagged differently, channels were swapped.
- A PCE in an access unit that fails no longer changes the layout, which
left every later packet failing.
- Non-finite samples are zeroed before the output and the overlap.
MP4: probe and decode agree: the samples' durations end the track when no
single edit does, and an edit cannot outlast the media.
The routing oracle fails on a fixture ffmpeg cannot write when oracles are
required. The README's Limits names HE-AAC v2 as mono, multi-block ADTS,
iTunSMPB, Matroska's tick precision and AAC in CAF.
Co-Authored-By: Claude Opus 5.5 <noreply@anthropic.com>
FilmCraft's `filmcraft-aac` decoder as `craft_audio::aac` behind
`aac-decode`, kept over symphonia's: the same quality (about 135 dB
SNR against ffmpeg), and it decodes 5.1, which symphonia refuses.
HE-AAC decodes as its AAC-LC core, upsampled 2x to the stream's rate.
- `TrackDecoder` decodes AAC, including ADTS-framed packets without
extradata (MPEG-TS, raw `.aac`).
- The native read path takes AAC in MP4 and Matroska and raw ADTS
streams (frame sync over the file, with resync after damage).
- AAC errors convert into `AudioError`.
The AAC module's shared parts (Huffman, MDCT, TNS, ICS) arrive whole
as ported; their encoder half is unused until the encoder PR.
Co-Authored-By: Claude Opus 5.5 <noreply@anthropic.com>
* craft-audio: core types, WAV/AIFF read and write, peaks, symphonia
The new crate without any codec module: what every later codec plugs
into.
- Shared types: `AudioBuffer`, `AudioInfo`, `ChannelLayout`/`Speaker`,
`AudioError`, `EncodeOptions` (built with `new` and `with_*`),
`EncodeFormat`, `BitDepth`, `BwfInfo`, `SampleFormat`.
- WAV/BWF/RF64 and AIFF/AIFF-C reading and writing from SoundCraft's
`soundcraft-audio-io`, kept over EffectCraft's and FilmCraft's
smaller writers (only it has BWF, dither, any speaker mask and
streaming without a known length). `encode` and `FileEncoder` behind
`pcm-encode`; `lossless-encode` and `encode` turn it on.
- `decode`, `probe`, `tracks` and the streaming, seekable `Decoder`
(feature `decode-native`). The native path reads WAV, AIFF, and PCM in
MP4/MOV and Matroska through craft-media's demuxers, packet by packet,
with edit lists and `CodecDelay`/`DiscardPadding` applied.
`TrackDecoder` decodes PCM; each codec PR adds its codec to it.
- symphonia 0.6 (MPL-2.0, used unmodified) behind the `symphonia`
feature, part of `decode`: the fallback for every format with no
native decoder yet. Its size fields are checked before it reads, and
its panics are caught where they unwind.
- `peaks`: the min/max/RMS pyramid for waveform drawing.
Workspace: craft-media is a workspace dependency without default
features, so an encode-only craft-audio does not build the MP4
demuxer, and `symphonia-bundle-flac` builds without overflow checks in
dev profiles (its 32-bit LPC restore wraps on valid input).
CI: `audio.yml`, with clippy per feature, rustdoc, WASM, MSRV, and the
tests with ffmpeg oracles.
Co-Authored-By: Claude Opus 5.5 <noreply@anthropic.com>
* craft-audio: review fixes (CodeRabbit on #441)
- Reading: AIFF that declares frames without an SSND chunk is Malformed; a Matroska PCM track's
length counts from its first frame; seek targets saturate; an empty FLAC has one track; a
failed header read is an Io error; deinterleave allocates fallibly; concealed silence is capped
like a decoded packet, and symphonia's uses the damaged packet's own duration when known.
- Peaks sum squares in f64, so loud float samples give a finite RMS.
- Writing: .aifc maps to AIFF; BWF text is cut at a character boundary; a WAV byte rate past
32 bits is an error; the FileEncoder docs say nothing may follow the sink's position.
- README: the Limits list says which codecs go through symphonia and which MP4 edit lists apply.
- Tests for each, alias and dither coverage, and a FLAC case in the routing oracle.
Co-Authored-By: Claude Opus 5.5 <noreply@anthropic.com>
* craft-audio: review fixes, second round (CodeRabbit on #441)
- Every header read is checked: `Native::open`, `native::tracks` and the no-symphonia error read
it with `head_checked`. craft-media demuxer errors that are I/O failures (not a file ending
early) are returned as `AudioError::Io` instead of sending the file to the other path.
- A Matroska track's presented length runs from its first frame plus `CodecDelay` (unclamped,
in timestamp ticks with the delay rounded to one) to its end.
- symphonia's concealment allocates fallibly, and a damaged first packet is concealed when the
demuxer gives its duration. `deinterleave` reserves its channel list fallibly too.
- `TrackDecoder::conceal` documents its cap at the decode limit.
- Tests: an I/O failure past the first 64 KiB, codec delay with the first block at -20, 0 and
50 ms (checked with `probe`), a seek back after an overshoot, and the loud-sample RMS value.
Co-Authored-By: Claude Opus 5.5 <noreply@anthropic.com>
* craft-audio: TrackDecoder::conceal returns a Result
Concealment allocates fallibly and fails with TooLarge when the silence would pass the 2^30-sample
decode limit, instead of returning less than the packet's length. The native path, the README,
the packets example and the doc example pass the error on.
Co-Authored-By: Claude Opus 5.5 <noreply@anthropic.com>
* craft-audio: review fixes, third round (CodeRabbit on #441)
- Concealed silence is held to the decode limit inside pcm::silence, so symphonia's path checks
frames × channels too, and symphonia's decoded audio is copied into fallibly allocated planes.
- TrackDecoder::reset forgets the last packet's length.
- Without symphonia, a damaged MP4, Matroska or Ogg file (truncated, invalid, no moov) is
Malformed instead of Unsupported; a feature the demuxer does not read still names the build.
- Tests: an MP4 with no moov in a native-only build, conceal after a reset and past the limit.
Co-Authored-By: Claude Opus 5.5 <noreply@anthropic.com>
* craft-audio: review fixes, fourth round (CodeRabbit on #441)
- A file the native demuxers find damaged is Malformed with symphonia too, when symphonia calls
it unsupported; before, an ftyp-only MP4 was Unsupported in symphonia builds.
- symphonia's decoded blocks are held to the 2^30-sample decode limit before allocating.
- Symph::seek forgets the last packet's length, so a lost packet after a seek is not concealed
with the length of one before it.
Co-Authored-By: Claude Opus 5.5 <noreply@anthropic.com>
* craft-audio: review fixes, fifth round (CodeRabbit on #441)
- The damaged-file test covers Matroska as well as MP4, and fails when either branch of
native::damage is removed.
- native::damage leaves out Ogg: ogg::open reports no damage (it skips damaged pages and calls a
file with none it can read not Ogg), and symphonia reports damaged Ogg as malformed itself.
Co-Authored-By: Claude Opus 5.5 <noreply@anthropic.com>
* craft-audio: run the ffmpeg oracle tests on ubuntu-26.04
ubuntu-latest's apt ships ffmpeg 6.1.1, which writes MP2 and E-AC-3 in
Matroska with ends a few milliseconds early, so the routing tests on the
later codec PRs cannot pass on it. ubuntu-26.04 ships ffmpeg 8.0.
Co-Authored-By: Claude Opus 5.5 <noreply@anthropic.com>
---------
Co-authored-by: Claude Opus 5.5 <noreply@anthropic.com>
