Extends the two-path correlation from rate-only to the full layout, removing the
"channels/bit-depth assumed = device" limitation. correlate_format tries each
candidate de-interleaving (float32 / int16; mono..7.1) of the hook capture, runs
the rate correlation per layout, and keeps whichever aligns with the loopback;
a wrong de-interleaving is noise and won't.
The catch: the hook can't know a guessed stream's real frame size, so its verify
tap pads each render buffer to the device block -- which over-reads stale staging
bytes for a stream with fewer channels/bits, scrambling the audio. So the tap is
now self-describing: it prefixes each buffer with its frame count
([count][count*device_block bytes]), and the host strips the padding per candidate
layout (take the real count*real_block of each chunk) before de-interleaving.
- audio_correlate.hpp: ChunkedCapture + chunk-aware correlate_format + candidate
layouts; absolute-margin confidence gate (the true layout scores ~1.0, a truly
ambiguous alternative within ~0.001 -- 2ch@R == 1ch@2R for identical channels --
is correctly left unconfident).
- audio_hook.cpp: chunked verify tap (free-space-checked so framing can't tear).
- audio_format_verifier: parse chunks; recover_layout path. AudioMirror now corrects
the full format.
- audio_correlation_test: layout recovery from padded chunks (stereo float, 16-bit
PCM, 5.1, mono). audio_verify_test gains scenario (b): 2ch on a multichannel
endpoint with distinct per-channel content (new env-gated ToneSource mode) ->
recovers ch=2/32-bit float end-to-end.
Co-Authored-By: Claude Opus 4.8 <noreply@anthropic.com>
Hooked audio mirroring played back pitch-shifted on games we inject into
that render at a non-device sample rate (e.g. Godot/Brotato render 44100 Hz
on a 48000 Hz endpoint via WASAPI AUTOCONVERTPCM). We attach to an
already-running game, so the render-hook never saw its IAudioClient::
Initialize and assumed the device mix format -- right channels/bits, wrong
rate -- so 44100 audio was rendered as 48000 (+~1.5 semitones).
Fix: treat a pre-existing client's format as a guess and measure its true
sample rate from the render cadence (frames/sec over a steady-state window,
snapped to the nearest standard rate) before publishing it, deferring
capture until verified. Discard the first measurement window so the
buffer-fill burst at attach time doesn't over-count. Streams created after
we inject still carry their exact Initialize format.
Channels/bit-depth genuinely can't be recovered for a pre-existing client:
AUTOCONVERTPCM hands GetBuffer a fixed staging buffer (no buffer stride to
measure -- confirmed empirically) and WASAPI exposes no API for the format.
They stay the device-mix guess, which is correct for the common case
(engines render stereo float, matching the endpoint). To keep a wrong guess
safe, a VirtualQuery clamp stops the capture copy from ever over-reading the
source buffer when the guessed bytes/frame is too large.
Surface all of this: a per-stream AudioFormatState (known / measuring /
measured rate (ch/bits assumed)) in HookStatus, shown in the Audio panel for
the hooked path and as "device endpoint (known)" for loopback; clear hook
logs; and enriched mirror status strings. Documented in README (Limitations
+ Lessons learned). The loopback fallback was always correct (post-mix at
the device format).
Tests: extract a shared, configurable ToneSource (used by coop_tone and the
hook self-test); coop_tone takes rate/channels/bits/format args. Rewrite
audio_hook_test to a format matrix x both code paths -- see-init (exact) and
guess (rate measured) -- plus a byte-incompatible guess that asserts the
clamp keeps capture safe. The matrix caught the attach-burst over-count.
audio_loopback_test now spawns coop_tone at several source formats to
confirm loopback is format-agnostic. 11/11 x64 + 3/3 x86 pass.
Co-Authored-By: Claude Opus 4.8 <noreply@anthropic.com>
