Problem: our AAC decoder outputs channels in AAC element order (C, L, R,
Ls, Rs, LFE for channel configuration 6) and PacketDecoder passed that
through unchanged, while the mixer, panner, loudness and export all use
WAV/SMPTE order (L, R, C, LFE, Ls, Rs). A 5.1 AAC import therefore put
centre in L, L in R, R in C, the surrounds one slot early and LFE in Rs.
The export path already reorders for the encoder; the decode path did not.
Fix: PacketDecoder reorders AAC output for channel configurations 3-6
into WAV/SMPTE order, like the AC-3 decoder and Opus mapping family 1
already do. The Opus reorder now shares the same bounds-checked helper.
Co-Authored-By: Claude Opus 5.5 <noreply@anthropic.com>
The project Look-Ahead Time parameter was stored but never applied, so 0 ms and 30 ms mixed the same clip. The limiter now sizes its look-ahead from that parameter and pre-rolls for the chosen window.
Problem: with the device at another rate than the sequence (44.1 kHz
device, 48 kHz sequence), each buffer truncated its source start (phase
lost every block: a buzz), took the nearest sample (aliasing), and asked
the mixer for overlapping spans, so the mixer's graph cache missed every
buffer and rebuilt effect state.
Solution: audio-dsp StreamResampler interpolates each output sample from
its absolute position (exact rational), bit-identical however the stream
is cut, and pulls the source as one contiguous stream. start_audio uses
it when rates differ and restarts it when the rate or layout changes.
Context: tests cover block-size invariance at four rate pairs, a 1 kHz
tone within 3e-3, contiguous pulls, seeks and hostile input. Same lines
as #131; whichever lands second needs a small rebase.
Co-Authored-By: Claude Opus 5.5 <noreply@anthropic.com>
- ac3: LFE channel taken without `expect`.
- audio-dsp: the channel matrix match is exhaustive (no `unreachable!`); an unknown internal
parameter reads as 0 (debug-asserted) instead of `panic!`.
- opus: dual-stereo bands and SILK frames take the optional buffer / range decoder with `if let`.
- mpeg2v: 64-bit peek through `get()` + `first_chunk`; the literal VLC table parser no longer
panics (debug-asserted).
- prores, dnx: IDCT rows through `as_chunks`; ProRes block store and alpha slices without
unwrap/expect; DNx header timecode through `get()`.
- dnx: `Encoder::new` returns `Result` instead of panicking on a raster the profile can't code
(e.g. 0x0), with a regression test; missing CID table is an error.
- Each crate denies unwrap/expect/panic/unimplemented/todo/unreachable outside tests.
Co-Authored-By: Claude Opus 5.5 <noreply@anthropic.com>
5.1 / multichannel:
- audio-dsp::channels: layouts (L R C LFE Ls Rs), ITU-R BS.775 up/downmix, 5.1 Mixdown Types,
equal-power 5.1 panner (puck, Center %, LFE).
- Mixer graph buses are 2 or 6 channels wide; 5.1 tracks/submixes/Mix; strips feeding a 5.1 bus
use the 5.1 panner (lanes pan51.x/y/center/lfe stored in MixerStrip::lanes, no schema change);
5.1 strips fold into stereo buses with BS.775; per-channel meters; clip effects on 6 channels.
- file.newSequence {mix, trackType}, sequence.settings {mix}; playback honours the 5.1 Mixdown
Type preference on stereo devices and plays 6 channels on 5.1 devices.
- Export Audio Channels 5.1: WAV WAVE_FORMAT_EXTENSIBLE (mask 0x3F), QuickTime PCM with a chan
atom, AAC channel configuration 6 (reordered to C L R Ls Rs LFE).
- Track Mixer: 5.1 panner puck with Center/LFE, N-channel meters, Show/Hide Tracks, Meter
Input(s) Only; transport record button runs voice-over.
Voice-over: engine AudioInput trait + SyntheticInput, cpal input in the app, audio.voiceover.*
commands (punch-in, pre/post-roll, countdown cues, WAV next to project / scratch disk, one undo
step), track header mic button and Voice-Over Record Settings dialog.
Remix: beat tracking + chroma/cepstra self-similarity planner in audio-dsp, hidden `remix` clip
effect holding the plan, render::remix playback with equal-power crossfades, clip.remix.*
commands, Remix tool and Remix Properties dialog.
Clip Mixer automation: per-track Latch/Touch/Write (clipMixer.setMode/touch/release) heard live
and written as thinned clip Volume/Panner keyframes at the end of the pass.
Co-Authored-By: Claude Opus 5.5 <noreply@anthropic.com>
Seven generated signals (sine, pink noise at 48/96 kHz, speech-like at
48/44.1 kHz, gated stereo with silence and sub-gate passages, fs/4
inter-sample peak) written as float WAV and measured by LoudnessMeter
and ffmpeg ebur128 (metadata, 3 decimals per 100 ms). Tolerances: M/S
+-0.1 LU every 100 ms, I +-0.1 LU, LRA +-0.5 LU, true peak +-0.2 dB.
Measured: M/S <= 0.0005, I <= 0.007, LRA <= 0.04 LU, TP <= 0.045 dB.
On the fs/4 tone ffmpeg's true peak is +0.6 dB above the analytic
value, so that case is checked against the analytic peak (ours -0.053).
testkit gains a float WAV writer.
Co-Authored-By: Claude Opus 5.5 <noreply@anthropic.com>
Earl Co