Setup > Playback Engine now chooses the audio system (cpal host), output
and input device, device sample rate and buffer size, kept per machine
with the UI preferences and applied by reopening the devices (refused
while recording; the transport position is kept). ui.audio_device does
the same for agents and lists every system with its devices, channels,
rates and buffer range.
Player and recorder open the chosen device, fall back to the defaults
for anything unavailable (unplugged device, unsupported rate), retry
with the device's default buffer when the chosen one is refused, and
convert 32/16-bit integer devices from and to the f32 mix through
scratch buffers allocated up front (no allocation in the callbacks).
ASIO is behind an opt-in 'asio' feature (playback and app): building it
downloads Steinberg's ASIO SDK (proprietary licence or GPLv3) and needs
LLVM, so it stays off by default. Verified on Windows with an SSL 2+:
WASAPI output/input switching and buffer 256, ASIO duplex at 128.
Docs: G-01 shrunk, hardware-parity rows, roadmap log.
Co-Authored-By: Claude Opus 5.5 <noreply@anthropic.com>
The input callback appended every block to one growing Vec: it
reallocated on the audio thread as the take grew, kept every input
channel in RAM, and the take existed only in memory until Stop, so a
crash during a take lost it.
- The input callback copies each block into a preallocated lock-free
ring (atomic slots, no unsafe) and never locks or allocates. Samples
that don't fit are paid back as silence before the next block, so the
take keeps its timing; the loss is reported in the status line.
- A writer thread drains the ring every 10 ms into one 32-bit float WAV
per armed track, holding only that track's input channels. Files go
to the session's Audio Files folder, or to its autosave folder while
the session has never been saved, named like the incumbent's takes
(<track>_<NN>.wav). An existing file is never overwritten.
- The header is rewritten every second, so a file left by a crash plays
up to the last update. Above 4 GiB it switches to RF64 in place.
- If a take file can't be created, Record doesn't start and says why;
a write error ends that take where it failed.
- At Stop the files become the takes' sources. Loop recording keeps one
file per track, one clip per pass (at most 1000) on its own playlist,
in one undo step.
- Saving uses a file already in the target's Audio Files folder in
place instead of rewriting it, so neither Save nor autosave copies a
take. A take recorded at another rate than the session's is saved once
at the session's rate.
soundcraft-audio-io gains wav_float_header (its length doesn't depend
on the frame count, so it can be rewritten in place); encode shares the
header code. Splitting a loop take into passes moved from the UI into
soundcraft_engine::io::add_recording, where it is tested.
Surround:
- main and bus formats up to 9.1.6 and Ambisonics; pairwise constant-power
surround panner (divergence, centre, LFE, height) per track and send;
ITU fold-down; multichannel bounce with correct WAV channel masks;
multichannel device output; surround panner and multichannel meters in the
Mix window; setup.main_format, mix.surround_pan, mix.send_surround_pan
VST3:
- soundcraft-vst3-host: isolated unsafe crate (vst3 bindings, MIT), scans
standard folders, hosts components/processors/controllers, params,
latency, notes; VST3 menu; engine.vst3_plugins; e2e tests with a fixture
Hardening:
- positions from parameters clamped; offline renders capped and bounded to
track material; loop/transport limits; hostile-parameter property test
Co-Authored-By: Claude Opus 5.5 <noreply@anthropic.com>
Workspace for SoundCraft, a clean-room pure-Rust Pro Tools-class DAW:
timebases and tempo maps, WAV/AIFF/FLAC/symphonia audio IO with peaks,
SMF and MIDI ops, 26 original DSP plugins, the session model, the mix
engine, ~180 registered commands with undo, a cpal player, and the
agent rules (AGENTS.md, ATTRIBUTION.md).
Co-Authored-By: Claude Opus 5.5 <noreply@anthropic.com>
