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>
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>
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>
