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Recover a guessed stream's channels + bit depth by correlation too (step b)

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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>
This commit is contained in:
BlackMark
2026-06-23 02:50:59 +02:00
parent 00244bcfd7
commit 7ada550930
9 changed files with 554 additions and 138 deletions
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94
README.md
View File

@@ -77,29 +77,30 @@ default** and covers anything the hooked path doesn't.
back to process-loopback capture, which does *not* mute the game — so the local
machine hears the audio twice (guests hear it once). The Audio panel shows which
path is active.
- **Hooked audio can only recover a pre-existing stream's *sample rate*, not its
channels/bit-depth.** The tool injects into an already-running game, so the audio
render-hook usually never saw the game's `IAudioClient::Initialize`. It recovers the
true **sample rate** by measuring the render cadence (so playback pitch is correct,
e.g. Godot/Brotato's 44100 Hz on a 48000 Hz endpoint), but **channels and bit-depth
can't be detected** — with `AUTOCONVERTPCM` `GetBuffer` returns a fixed staging
buffer (no buffer stride to measure) and WASAPI exposes no API for a pre-existing
client's format — so they're *assumed* to match the device mix format. That's correct
for the common case (engines render stereo float, matching the endpoint, differing
only in rate). A game rendering a *different* channel count or bit depth than the
device is mirrored with the wrong layout (garbled audio) on the hooked path, but never
an over-read/crash: the capture copy is clamped to the readable region (`VirtualQuery`),
and the local **mute** is done with `AUDCLNT_BUFFERFLAGS_SILENT` (which makes WASAPI
ignore the buffer's contents), so it never *writes* the wrongly-sized buffer either. So
every captured stream is silenced locally — guessed or exact — and there is no echo on
the hooked path. The loopback fallback is always format-correct. The Audio panel shows
each stream's
format provenance (*known* / *measuring* / *measured rate* / *low-confidence* /
*override*) so the assumption is visible, and (under Debug details) lets the operator
**re-measure** the rate or **override** the format when the guess is wrong. Overrides
are **remembered per game** (and a format caught exactly at `Initialize` is auto-saved
as that game's override), so a known-bad game is corrected automatically next launch.
Streams created *after* injection are captured exactly.
- **A pre-existing stream's format is *recovered*, not known — by measurement and
cross-correlation.** The tool injects into an already-running game, so the audio
render-hook usually never saw the game's `IAudioClient::Initialize`, and `AUTOCONVERTPCM`
hides the buffer stride (WASAPI exposes no API for a pre-existing client's format). The
hook first recovers the **sample rate** by measuring the render cadence (so playback pitch
is correct, e.g. Godot/Brotato's 44100 Hz on a 48000 Hz endpoint), assuming the device's
channels/bit-depth. When the game is still audible (the measurement window), the host then
**cross-correlates the two capture paths** — the hook (pre-mix) against a parallel
process-loopback (post-mix, the known device format) — to verify/correct the rate and to
**recover the channels + bit depth** by trying candidate de-interleavings and keeping the one
that aligns (`audio_format_verifier` → `coop/audio_correlate.hpp`). The recovered format feeds
the existing override channel. The one case it can't resolve is a *genuinely ambiguous* layout
(a stream whose channels carry identical content looks the same as one channel at double the
rate); there it stays the device assumption and the correlation reports low confidence rather
than guess. A wrong/assumed layout is mirrored with the wrong de-interleaving (garbled) but
never an over-read/crash: the capture copy is clamped to the readable region (`VirtualQuery`),
and the local **mute** uses `AUDCLNT_BUFFERFLAGS_SILENT` (WASAPI ignores the buffer contents),
so it never *writes* the wrongly-sized buffer either. Every captured stream is silenced locally
— guessed or exact — so there's no echo on the hooked path, and the loopback fallback is always
format-correct. The Audio panel shows each stream's format provenance (*known* / *measuring* /
*measured rate* / *low-confidence* / *override*), and (under Debug details) lets the operator
**re-measure** or **override** the format when needed. Overrides are **remembered per game** (and
a format caught exactly at `Initialize` is auto-saved), so a known-bad game is corrected
automatically next launch. Streams created *after* injection are captured exactly.
- **Debug-oriented UI:** the ImGui overlay is laid out for diagnosing the
pipeline, not for end use. F1 hides it entirely so the window is a clean mirror
for RPT; F2 frees the operator cursor; **F10 saves a PNG screenshot** (back buffer,
@@ -109,26 +110,6 @@ default** and covers anything the hooked path doesn't.
### Current Tasks
- **Determine a pre-existing stream's audio format by *correlating* the two capture
paths, instead of guessing.** When we attach to an already-running game we never saw its
`IAudioClient::Initialize`, so the render-hook assumes the device mix format and measures
only the sample rate from the render cadence — which can be wrong on a jittery game
(intermittent pitch shift) and can't recover channels/bit-depth at all. But during the
measurement window the game is still audible, so we already have *both* signals of the
same audio: the **process-loopback** capture (post-mix, at the **known** device format)
and the **render-hook** capture (pre-mix, at the unknown format). Cross-correlating them
pins the real format from ground truth rather than a guess. Two tasks:
- **(a) Rate verification/correction.** Resample the hook stream by each candidate standard
rate and cross-correlate against the loopback; the rate that holds alignment with no drift
over the window is the truth. Robust where cadence measurement is noisy — directly hardens
the intermittent pitch-shift symptom. Lands as a verify-and-correct step feeding the
existing rate path (the operator override stays as the manual escape hatch).
- **(b) Channels + bit-depth recovery.** Extend the correlation to the layout the cadence
method *can't* recover: interpret the hook bytes under candidate layouts (float32 vs
int16; mono/stereo/…) and keep whichever de-interleaving correlates with the loopback (a
wrong interpretation is noise and won't). Removes the "channels/bit-depth assumed = device"
limitation, so the garbled-layout case stops being undetectable.
- **Mouse + keyboard forwarding for Raw Input / DirectInput games.** The MKB
subsystem forwards via window messages (`PostMessage`) plus synthesized
`GetAsyncKeyState` / `GetKeyboardState` / `GetCursorPos`, which covers message-loop
@@ -235,15 +216,20 @@ ctest --test-dir build -C Debug --output-on-failure
skew + noise — exactly the hook-vs-loopback situation) and asserts `correlate_rate()` recovers the
true rate, scoring the right candidate ≈1.0 and the wrong ones ≈0 (incl. the hard 44100-vs-48000
case the cadence method can misread), and that unrelated signals are *not* confidently matched.
Pure header logic, no device.
Also covers **layout recovery** (`correlate_format`): from a self-describing *chunked* capture
(each render buffer padded to the device block, as the hook's verify tap produces it) it strips
the padding per candidate de-interleaving and recovers the true channels + bit depth + rate
(stereo float, 16-bit PCM, 5.1, mono), and leaves a genuinely-ambiguous identical-channel layout
unconfident. Pure header logic, no device.
- **`audio_verify_test`** — integration test of the host's two-path verifier
([`host/src/audio/audio_format_verifier.cpp`](host/src/audio/audio_format_verifier.cpp)). Launches
`coop_mock_game` rendering a tone at a non-device rate (matching the device's *channel* count, so
this rate test isn't perturbed by a channel mismatch — that's the next task), injects the hook
late (a guessed stream), and runs the real `verify_stream_format()`: it co-captures the hook
(pre-mix, via the ring's `verify_capture` tap) and a parallel process-loopback (post-mix) of the
same audio and correlates them. Asserts it recovers the game's true rate, not the device guess.
Skips cleanly without an audio endpoint.
this rate test isn't perturbed by a channel mismatch), injects the hook late (a guessed stream),
and runs the real `verify_stream_format()`: it co-captures the hook (pre-mix, via the ring's
`verify_capture` tap) and a parallel process-loopback (post-mix) of the same audio and correlates
them. Scenario (a) asserts it recovers the true rate; scenario (b) renders a *different* channel
count than the device with distinct per-channel content and asserts it recovers the full layout
(channels + bit depth + rate). Skips cleanly without an audio endpoint.
- **`render_pacer_test`** — unit test of the mirror's render-feed pacing policy
(`host/src/audio/render_pacer.hpp`). Simulates a producer/consumer device timeline and asserts
the shipping `RenderPacer` rides producer jitter that makes the old re-prime-on-partial-fill
@@ -551,7 +537,15 @@ Non-obvious things that cost time and constrain the design:
the *waveform* needs the hook bytes de-interleaved at the right channel count, so the rate step
assumes the hook layout matches the device (true for the common stereo-on-stereo case); recovering a
*different* channel count / bit depth is the layout step, which tries candidate de-interleavings and
keeps whichever correlates.
keeps whichever correlates. **The layout step needs a self-describing tap**: the hook can't know a
guessed stream's real frame size, so it pads each render buffer to the *device* block — which
over-reads stale staging bytes for a stream with fewer channels/bits. The raw padded bytes are
un-decodable (the stale tail scrambles the audio), so the tap prefixes each buffer with its frame
*count* (`[count][count*device_block bytes]`); the host strips the padding per candidate layout
(take the real `count*real_block` of each chunk) before de-interleaving. **The genuinely-ambiguous
case stays unresolved**: a stream whose channels carry identical content is indistinguishable from
one channel at double the rate (`2ch@R` == `1ch@2R` byte-for-byte), so the correlator reports low
confidence and the format stays the device assumption rather than guessing wrong.
- **Re-priming the render feed on a *partial* fill manufactures the gap it's avoiding.** The
mirror re-renders the captured ring to the output device. The original feed loop re-primed
(withheld the feed until ~30 ms had rebuffered) whenever it couldn't completely fill the free

173
common/include/coop/audio_correlate.hpp
View File

@@ -254,4 +254,177 @@ inline RateCorrelation correlate_rate(const std::vector<float>& hook_mono, const
return result;
}
// --- Step (b): channels + bit-depth recovery --------------------------------------------------
//
// The rate step assumes the hook bytes are de-interleaved at the device channel/bit layout. When a
// game renders a DIFFERENT layout than the device (e.g. stereo float on a 7.1 endpoint, or 16-bit
// PCM), that assumption garbles the waveform and the rate won't lock. We can't measure the layout
// (AUTOCONVERTPCM hides the stride), but we can RECOVER it: interpret the raw hook bytes under each
// candidate layout, run the rate correlation, and keep whichever (layout, rate) aligns with the
// loopback -- a wrong de-interleaving is noise and won't correlate.
inline constexpr unsigned kWaveFormatPcm = 1; // WAVE_FORMAT_PCM
inline constexpr unsigned kWaveFormatFloat = 3; // WAVE_FORMAT_IEEE_FLOAT
struct LayoutCandidate
{
unsigned channels;
unsigned bits;
unsigned tag; // kWaveFormatPcm / kWaveFormatFloat
};
// Candidate de-interleavings a shared-mode WASAPI render stream realistically uses: float32 and
// 16-bit PCM, across the common channel counts. Ordered most-likely-first.
inline const std::vector<LayoutCandidate>& standard_audio_layouts()
{
static const std::vector<LayoutCandidate> v = {
{2, 32, kWaveFormatFloat}, {1, 32, kWaveFormatFloat}, {6, 32, kWaveFormatFloat},
{8, 32, kWaveFormatFloat}, {4, 32, kWaveFormatFloat}, {2, 16, kWaveFormatPcm},
{1, 16, kWaveFormatPcm}, {6, 16, kWaveFormatPcm}, {8, 16, kWaveFormatPcm},
{4, 16, kWaveFormatPcm},
};
return v;
}
struct FormatCorrelation
{
bool ok = false;
unsigned rate = 0;
unsigned channels = 0;
unsigned bits = 0;
unsigned tag = 0;
double score = 0.0;
double runner_up = 0.0;
};
// The hook can't know a guessed stream's real frame size, so its verify tap pushes each render
// buffer padded to the device block (`stride`) and prefixes it with the real frame `count`. That
// padding is stale staging-buffer bytes, so the host must extract the real `count*real_block` bytes
// per buffer (and concatenate) before de-interleaving -- otherwise the padding scrambles the audio.
// This carries that self-describing capture: `bytes` holds counts[i]*stride bytes per chunk.
struct ChunkedCapture
{
unsigned stride = 0; // bytes per frame as pushed (the guessed/device block_align)
std::vector<std::uint32_t> counts; // real frame count of each chunk
std::vector<std::uint8_t> bytes; // concatenated, counts[i]*stride bytes per chunk
};
namespace correlate_detail
{
// De-interleave raw bytes under (channels/bits/tag) and average to mono float.
inline void decode_layout(const std::uint8_t* bytes, std::size_t n, const LayoutCandidate& fmt,
std::vector<float>& mono)
{
mono.clear();
const unsigned ch = fmt.channels == 0 ? 1 : fmt.channels;
const unsigned bps = fmt.bits / 8;
if (bps == 0)
{
return;
}
const std::size_t frame = static_cast<std::size_t>(ch) * bps;
const std::size_t frames = n / frame;
mono.resize(frames);
const bool is_float = fmt.tag == kWaveFormatFloat;
for (std::size_t i = 0; i < frames; ++i)
{
double sum = 0.0;
for (unsigned c = 0; c < ch; ++c)
{
const std::uint8_t* p = bytes + i * frame + static_cast<std::size_t>(c) * bps;
float s = 0.0f;
if (is_float && fmt.bits == 32)
{
std::memcpy(&s, p, 4);
}
else if (fmt.bits == 16)
{
std::int16_t v;
std::memcpy(&v, p, 2);
s = v / 32768.0f;
}
else if (fmt.bits == 32)
{
std::int32_t v;
std::memcpy(&v, p, 4);
s = static_cast<float>(v / 2147483648.0);
}
sum += s;
}
mono[i] = static_cast<float>(sum / ch);
}
}
} // namespace correlate_detail
// Recover BOTH the layout and the rate of a guessed stream from its (padded, self-describing) hook
// capture: for each candidate layout, extract the real count*real_block bytes from each padded
// chunk, de-interleave to mono, and run the rate correlation against the known-format loopback,
// keeping the (layout, rate) that aligns best. `ok` when the winner clears the alignment floor and
// clearly beats the runner-up (so a coincidental partial match is rejected).
//
// `min_margin` is an ABSOLUTE gap (not a ratio): the true layout scores near-perfectly while a
// truly-ambiguous alternative (e.g. 1ch@2R vs 2ch@R when the channels carry the same content)
// scores within ~0.001, so requiring the winner to clear the runner-up by a fixed margin cleanly
// separates "recovered" from "genuinely ambiguous, don't guess".
inline FormatCorrelation correlate_format(const ChunkedCapture& hook, const std::vector<float>& loop_mono,
unsigned device_rate, const std::vector<unsigned>& rates,
const std::vector<LayoutCandidate>& layouts, double min_score = 0.55,
double min_margin = 0.04)
{
FormatCorrelation result;
if (hook.stride == 0 || hook.counts.empty() || loop_mono.empty() || device_rate == 0)
{
return result;
}
double best = -1.0, second = -1.0;
std::vector<std::uint8_t> clean;
std::vector<float> hook_mono;
for (const LayoutCandidate& layout : layouts)
{
const unsigned real_block = layout.channels * (layout.bits / 8);
if (real_block == 0 || real_block > hook.stride)
{
continue; // can't extract a frame larger than what was pushed (the guess is the max)
}
// Pull the real count*real_block bytes out of each padded chunk and concatenate -> contiguous
// audio for this candidate layout (the padding, which is stale staging bytes, is dropped).
clean.clear();
std::size_t off = 0;
for (std::uint32_t count : hook.counts)
{
const std::size_t chunk_bytes = static_cast<std::size_t>(count) * hook.stride;
const std::size_t take = static_cast<std::size_t>(count) * real_block;
if (off + chunk_bytes <= hook.bytes.size())
{
clean.insert(clean.end(), hook.bytes.begin() + off, hook.bytes.begin() + off + take);
}
off += chunk_bytes;
}
correlate_detail::decode_layout(clean.data(), clean.size(), layout, hook_mono);
if (hook_mono.size() < device_rate / 5)
{
continue; // this layout yields too little audio to judge
}
const RateCorrelation rc = correlate_rate(hook_mono, loop_mono, device_rate, rates, /*min_score=*/0.0,
/*separation=*/1.0);
if (rc.score > best)
{
second = best;
best = rc.score;
result.rate = rc.rate;
result.channels = layout.channels;
result.bits = layout.bits;
result.tag = layout.tag;
}
else if (rc.score > second)
{
second = rc.score;
}
}
result.score = best < 0.0 ? 0.0 : best;
result.runner_up = second < 0.0 ? 0.0 : second;
result.ok = result.score >= min_score && (result.runner_up <= 1e-6 || result.score - result.runner_up >= min_margin);
return result;
}
} // namespace coop

9
common/include/coop/audio_ring.hpp
View File

@@ -203,6 +203,15 @@ inline std::uint32_t audio_ring_available(const AudioRingHeader& h)
return static_cast<std::uint32_t>(w - r);
}
// Producer: bytes currently free (so a multi-part packet can be checked to fit before any of it is
// written -- keeps a self-describing [header][payload] framing from tearing on a full ring).
inline std::uint32_t audio_ring_free_space(const AudioRingHeader& h)
{
const std::uint64_t w = h.write_pos.load(std::memory_order_relaxed);
const std::uint64_t r = h.read_pos.load(std::memory_order_acquire);
return h.capacity - static_cast<std::uint32_t>(w - r);
}
// Consumer: copy up to `bytes` into `dst`; returns the number actually popped.
inline std::uint32_t audio_ring_pop(AudioRingHeader& h, void* dst, std::uint32_t bytes)
{

15
hook/src/audio_hook.cpp
View File

@@ -410,8 +410,19 @@ HRESULT STDMETHODCALLTYPE hk_ReleaseBuffer(IAudioRenderClient* self, UINT32 num_
const std::uint32_t block = g_streams[i].block_align.load(std::memory_order_relaxed);
if (block != 0)
{
audio_ring_push(*vring, t_gb_data, readable_bytes(t_gb_data, num_frames * block),
num_frames);
// Self-describing chunk: [u32 frame-count][num_frames*block bytes]. The host can't
// know the real frame size of a guessed stream, so it recovers the layout by
// trying candidate de-interleavings -- but it needs the frame count to strip the
// per-buffer padding (the guessed/device block over-reads a stream with fewer
// channels/bits). Push both parts only if both fit and the payload is fully
// readable, so a full ring or a short buffer can never tear the framing.
const std::uint32_t want = num_frames * block;
if (readable_bytes(t_gb_data, want) == want &&
audio_ring_free_space(*vring) >= static_cast<std::uint32_t>(sizeof(num_frames)) + want)
{
audio_ring_push(*vring, &num_frames, sizeof(num_frames), 0);
audio_ring_push(*vring, t_gb_data, want, num_frames);
}
}
}
}

67
host/src/audio/audio_format_verifier.cpp
View File

@@ -99,6 +99,35 @@ void drain_ring(AudioRingHeader& ring, std::vector<BYTE>& scratch)
}
}
// Parse the hook's self-describing verify stream -- a sequence of [u32 frame-count][count*stride
// bytes] chunks -- into a ChunkedCapture. `stride` is the device block_align (what the hook padded
// each buffer to). Stops at the first truncated/garbled record.
ChunkedCapture parse_chunks(const std::vector<BYTE>& raw, unsigned stride)
{
ChunkedCapture cap;
cap.stride = stride;
if (stride == 0)
{
return cap;
}
std::size_t off = 0;
while (off + sizeof(std::uint32_t) <= raw.size())
{
std::uint32_t count = 0;
std::memcpy(&count, raw.data() + off, sizeof(count));
off += sizeof(count);
const std::size_t payload = static_cast<std::size_t>(count) * stride;
if (count == 0 || off + payload > raw.size())
{
break; // truncated or garbled -> stop
}
cap.counts.push_back(count);
cap.bytes.insert(cap.bytes.end(), raw.data() + off, raw.data() + off + payload);
off += payload;
}
return cap;
}
} // namespace
FormatVerification verify_stream_format(DWORD pid, AudioRingHeader* ring, unsigned window_ms, bool recover_layout)
@@ -156,31 +185,53 @@ FormatVerification verify_stream_format(DWORD pid, AudioRingHeader* ring, unsign
ring->verify_capture.store(0, std::memory_order_release);
drain_ring(*ring, scratch); // leave the ring clean for the real capture that follows
// The hook bytes are at the guessed layout = the device channels/bits (the assumption the
// cadence path also makes). Decode both captures with that layout and correlate.
const std::vector<float> hook_mono = to_mono(hook_bytes, dev);
// The loopback is at the KNOWN device layout. Decode it to mono ground truth. The hook stream
// is self-describing chunks ([count][padded payload]); parse them at the device block.
const std::vector<float> loop_mono = to_mono(loop_bytes, dev);
const unsigned dev_block = dev_wfx->nBlockAlign;
const ChunkedCapture cap = parse_chunks(hook_bytes, dev_block);
const std::size_t need = dev.rate / 5; // require >= ~200 ms of usable audio on both sides
const std::size_t hook_frames = dev_block != 0 ? cap.bytes.size() / dev_block : 0;
CoTaskMemFree(dev_wfx);
if (const char* dbg = std::getenv("COOP_VERIFY_DEBUG"); dbg != nullptr && dbg[0] == '1')
{
std::fprintf(stderr, "[verify] dev=%uHz/%uch/%ubit tag=%u hook_frames=%zu loop_frames=%zu\n", dev.rate,
dev.channels, dev.bits, dev.tag, hook_mono.size(), loop_mono.size());
std::fprintf(stderr, "[verify] dev=%uHz/%uch/%ubit blk=%u chunks=%zu hook_frames=%zu loop=%zu layout=%d\n",
dev.rate, dev.channels, dev.bits, dev_block, cap.counts.size(), hook_frames, loop_mono.size(),
recover_layout ? 1 : 0);
}
const std::size_t need = dev.rate / 5; // require >= ~200 ms of usable audio on both sides
if (hook_mono.size() < need || loop_mono.size() < need)
if (loop_mono.size() < need || hook_frames < need)
{
return result; // not enough non-silent audio captured (game quiet, or stream wasn't a guess)
}
if (recover_layout)
{
// Step (b): recover channels + bit depth too, by trying candidate de-interleavings of the
// (de-padded) hook bytes and keeping whichever (layout, rate) correlates with the loopback.
const FormatCorrelation fc =
correlate_format(cap, loop_mono, dev.rate, standard_audio_rates(), standard_audio_layouts());
result.ok = fc.ok;
result.rate = fc.rate;
result.score = fc.score;
result.layout_ok = fc.ok;
result.channels = fc.channels;
result.bits = fc.bits;
result.format_tag = fc.tag;
}
else
{
// Step (a): rate only, assuming the hook layout matches the device (common stereo case), so
// the de-padded payload is already clean device-layout audio.
const std::vector<float> hook_mono = to_mono(cap.bytes, dev);
const RateCorrelation rc = correlate_rate(hook_mono, loop_mono, dev.rate, standard_audio_rates());
result.ok = rc.ok;
result.rate = rc.rate;
result.score = rc.score;
// Channels/bit-depth stay the device assumption here; step (b) recovers them.
result.channels = dev.channels;
result.bits = dev.bits;
result.format_tag = dev.tag;
}
return result;
}

9
host/src/audio/audio_loopback.cpp
View File

@@ -323,10 +323,15 @@ void AudioMirror::thread_main(DWORD pid)
if (!format_verified)
{
format_verified = true;
const FormatVerification fv = verify_stream_format(pid, rings[0]);
// recover_layout: correlate the full format (rate AND channels/bit-depth), so a
// game rendering a different layout than the device is corrected too, not just
// the rate. A no-op when nothing correlates confidently (e.g. an exact stream,
// a silent game, or a genuinely ambiguous identical-channel layout).
const FormatVerification fv = verify_stream_format(pid, rings[0], /*window_ms=*/900,
/*recover_layout=*/true);
if (fv.ok)
{
set_status("Verified render-hook sample rate by correlation.");
set_status("Verified render-hook format by correlation.");
audio_ring_post_op(*rings[0], AudioRingOp_Override, fv.rate, fv.channels, fv.bits,
fv.format_tag);
}

120
tests/audio_correlation_test.cpp
View File

@@ -6,6 +6,7 @@
// rates (plus a capture-latency skew and a little noise), then assert correlate_rate() recovers the
// true rate -- including the hard 44100-vs-48000 case the cadence method can misread. Pure header
// logic, no device.
#include <algorithm>
#include <cmath>
#include <cstdint>
#include <cstdio>
@@ -55,6 +56,118 @@ std::vector<float> capture(unsigned rate, double seconds, double t0, double nois
return out;
}
// Per-channel continuous signal: each channel carries genuinely different content (its own
// frequency set), like a real stereo/surround stream. This is what disambiguates the channel
// count -- with identical channels, 2ch@R and 1ch@2R produce the same bytes and are truly
// indistinguishable (the confidence gate correctly rejects that case).
double multi(double t, unsigned channel)
{
const double k = 1.0 + 0.37 * static_cast<double>(channel); // distinct frequency scale per channel
const double chirp = std::sin(2.0 * kPi * (300.0 * k * t + 140.0 * t * t));
return 0.5 * std::sin(2.0 * kPi * 221.0 * k * t) + 0.28 * std::sin(2.0 * kPi * 437.0 * k * t + 0.6) +
0.22 * chirp;
}
double multi_mono(double t, unsigned channels)
{
double sum = 0.0;
for (unsigned c = 0; c < channels; ++c)
{
sum += multi(t, c);
}
return sum / channels;
}
// Encode the multichannel signal to raw interleaved PCM bytes at the given layout/rate.
std::vector<std::uint8_t> encode(unsigned rate, unsigned channels, unsigned bits, unsigned tag, double seconds)
{
const unsigned bps = bits / 8;
const std::size_t frames = static_cast<std::size_t>(rate * seconds);
std::vector<std::uint8_t> out(frames * channels * bps);
for (std::size_t i = 0; i < frames; ++i)
{
const double t = static_cast<double>(i) / rate;
for (unsigned c = 0; c < channels; ++c)
{
const double s = multi(t, c);
std::uint8_t* p = out.data() + (i * channels + c) * bps;
if (tag == coop::kWaveFormatFloat)
{
const float f = static_cast<float>(s);
std::memcpy(p, &f, 4);
}
else
{
const std::int16_t v = static_cast<std::int16_t>(s * 30000.0);
std::memcpy(p, &v, 2);
}
}
}
return out;
}
// Build the hook's self-describing chunked capture from clean audio: split into ~480-frame chunks
// and pad each frame from real_block up to `stride` with GARBAGE -- exactly what the hook's verify
// tap produces (it over-reads a guessed stream whose real layout has fewer channels/bits than the
// device block). correlate_format must strip the padding per candidate layout.
coop::ChunkedCapture make_chunks(unsigned rate, unsigned ch, unsigned bits, unsigned tag, unsigned stride,
double seconds)
{
coop::ChunkedCapture cap;
cap.stride = stride;
const std::vector<std::uint8_t> clean = encode(rate, ch, bits, tag, seconds);
const unsigned real_block = ch * (bits / 8);
const std::size_t frames = clean.size() / real_block;
std::mt19937 rng(123);
std::uniform_int_distribution<int> garbage(0, 255);
std::size_t f = 0;
while (f < frames)
{
const unsigned count = static_cast<unsigned>(std::min<std::size_t>(480, frames - f));
cap.counts.push_back(count);
// The hook reads count*stride contiguous bytes: the count real frames first
// (count*real_block bytes), then count*(stride-real_block) bytes of stale over-read.
const std::uint8_t* src = clean.data() + f * real_block;
cap.bytes.insert(cap.bytes.end(), src, src + static_cast<std::size_t>(count) * real_block);
for (std::size_t p = 0; p < static_cast<std::size_t>(count) * (stride - real_block); ++p)
{
cap.bytes.push_back(static_cast<std::uint8_t>(garbage(rng)));
}
f += count;
}
return cap;
}
// One layout scenario: the hook bytes are at (true_*) and still being measured; the loopback is the
// post-mix mono of the same audio at device_rate. Assert correlate_format recovers the full layout.
void test_layout(unsigned true_rate, unsigned true_ch, unsigned true_bits, unsigned true_tag,
unsigned device_rate, const char* label)
{
std::printf("== layout: %s (%u Hz / %u ch / %u-bit %s -> device %u Hz) ==\n", label, true_rate, true_ch,
true_bits, true_tag == coop::kWaveFormatFloat ? "float" : "pcm", device_rate);
// Device block 32 (8ch float) is the largest stride; every test layout's real block is <= 32.
const coop::ChunkedCapture hook = make_chunks(true_rate, true_ch, true_bits, true_tag, /*stride=*/32, 0.6);
// Loopback: the post-mix mono of the same audio, at the device rate, started ~18 ms later + noise.
const std::size_t loop_frames = static_cast<std::size_t>(device_rate * 0.6);
std::vector<float> loop(loop_frames);
std::mt19937 rng(5);
std::uniform_real_distribution<float> j(-1.0f, 1.0f);
for (std::size_t m = 0; m < loop_frames; ++m)
{
loop[m] = static_cast<float>(multi_mono(0.018 + static_cast<double>(m) / device_rate, true_ch)) +
0.02f * j(rng);
}
const FormatCorrelation r =
correlate_format(hook, loop, device_rate, standard_audio_rates(), standard_audio_layouts());
std::printf(" picked %u Hz / %u ch / %u-bit %s score=%.3f runner_up=%.3f ok=%d\n", r.rate, r.channels,
r.bits, r.tag == coop::kWaveFormatFloat ? "float" : "pcm", r.score, r.runner_up, r.ok ? 1 : 0);
check(r.ok, "layout pick is confident");
check(r.rate == true_rate, "recovered the true rate");
check(r.channels == true_ch, "recovered the true channel count");
check(r.bits == true_bits && r.tag == true_tag, "recovered the true bit depth / sample format");
}
// One scenario: true hook rate `true_rate` mixed to `device_rate`. Assert the correlator picks
// true_rate confidently and that the runner-up is clearly behind.
void test_case(unsigned true_rate, unsigned device_rate, const char* label)
@@ -84,6 +197,13 @@ int main()
test_case(32000, 44100, "low-rate stream on a 44100 endpoint");
test_case(48000, 44100, "48000 stream on a 44100 endpoint");
// Step (b): recover the full layout (channels + bit depth) when it differs from the device, by
// trying candidate de-interleavings -- the case the rate-only step can't handle.
test_layout(44100, 2, 32, coop::kWaveFormatFloat, 48000, "stereo float, wrong rate");
test_layout(44100, 2, 16, coop::kWaveFormatPcm, 48000, "stereo 16-bit PCM (bit depth differs)");
test_layout(48000, 6, 32, coop::kWaveFormatFloat, 48000, "5.1 float (channels differ)");
test_layout(44100, 1, 32, coop::kWaveFormatFloat, 48000, "mono"); // 2ch@22050 isn't a candidate -> unambiguous
// Downmix sanity: a stereo interleaved buffer collapses to the same mono the scalar path uses.
{
std::printf("== downmix stereo -> mono ==\n");

164
tests/audio_verify_test.cpp
View File

@@ -1,22 +1,23 @@
// Integration test for the two-path audio-format verifier (host/src/audio/audio_format_verifier).
//
// Launches coop_mock_game rendering a tone at a NON-device rate (44100 on a typical 48000 endpoint,
// the Godot/Brotato case), injects coop_hook.dll late (so the stream is a *guess*), then runs the
// real verify_stream_format(): it co-captures the hook (pre-mix, via the ring's verify tap) and a
// parallel process-loopback (post-mix, device format) and cross-correlates them. Asserts it
// recovers the true 44100 Hz rate -- the cadence method's hard case. Skips cleanly without an audio
// endpoint / if Vulkan-free... (only needs WASAPI + a D3D11-capable mock, which the mock always is).
// Launches coop_mock_game rendering a tone via WASAPI AUTOCONVERTPCM, injects coop_hook.dll late
// (so the stream is a *guess*), and runs the real verify_stream_format(): it co-captures the hook
// (pre-mix, via the ring's verify tap) and a parallel process-loopback (post-mix, device format)
// and cross-correlates them. Two scenarios:
// (a) rate: render at the device's channel count but a different rate -> recover the rate.
// (b) layout: render a DIFFERENT channel count than the device, with distinct per-channel content
// -> recover channels + bit depth + rate.
// Skips cleanly without an audio endpoint.
#include <cstdint>
#include <cstdio>
#include <string>
#include <windows.h>
#include <mmreg.h>
#include <objbase.h>
#include <tlhelp32.h>
#include <mmreg.h>
#include "audio/audio_format_verifier.hpp"
#include "audio/process_loopback_capture.hpp" // default_render_format
#include "coop/audio_ring.hpp"
@@ -111,42 +112,42 @@ bool inject_retry(unsigned long pid)
}
return false;
}
} // namespace
int main()
struct Scenario
{
unsigned rate, channels, bits;
bool distinct; // distinct per-channel content (so the channel count is recoverable)
bool recover_layout; // false = rate only (step a); true = full layout (step b)
};
// Launch the mock at the scenario's format, inject the hook late, and run the verifier. `ran` is
// set false when the environment can't support the test (launch/inject failed) so the caller skips.
FormatVerification run(const Scenario& sc, bool& ran)
{
ran = false;
FormatVerification fv;
kill_stray_mock_games();
const bool com = SUCCEEDED(CoInitializeEx(nullptr, COINIT_MULTITHREADED));
// Render the mock at the device's CHANNEL count (so this step-(a) rate test isn't perturbed by
// a channel mismatch -- that's step (b)'s job) but at a DIFFERENT standard rate than the device,
// so the verifier has a real rate to recover. Default to 48000/2ch if we can't read the device.
unsigned dev_rate = 48000, dev_channels = 2;
if (WAVEFORMATEX* dev = default_render_format())
if (sc.distinct)
{
dev_rate = dev->nSamplesPerSec;
dev_channels = dev->nChannels;
CoTaskMemFree(dev);
SetEnvironmentVariableW(L"COOP_TONE_DISTINCT_CH", L"1");
}
const unsigned game_rate = (dev_rate == 44100) ? 48000u : 44100u; // guarantee a rate mismatch
std::printf(" device %u Hz / %u ch -> rendering the mock at %u Hz / %u ch (rate mismatch)\n", dev_rate,
dev_channels, game_rate, dev_channels);
const std::wstring exe = exe_directory() + L"coop_mock_game.exe";
std::wstring cmd = L"\"" + exe + L"\" dx11 30 " + std::to_wstring(game_rate) + L" " +
std::to_wstring(dev_channels) + L" 32 float";
std::wstring cmd = L"\"" + exe + L"\" dx11 30 " + std::to_wstring(sc.rate) + L" " +
std::to_wstring(sc.channels) + L" " + std::to_wstring(sc.bits) + L" " +
(sc.bits == 16 ? L"pcm" : L"float");
STARTUPINFOW si{};
si.cb = sizeof(si);
PROCESS_INFORMATION pi{};
if (!CreateProcessW(exe.c_str(), cmd.data(), nullptr, nullptr, FALSE, 0, nullptr, nullptr, &si, &pi))
const BOOL launched = CreateProcessW(exe.c_str(), cmd.data(), nullptr, nullptr, FALSE, 0, nullptr, nullptr,
&si, &pi);
if (sc.distinct)
{
std::printf("Could not launch coop_mock_game -- skipping audio_verify_test.\n");
if (com)
{
CoUninitialize();
SetEnvironmentVariableW(L"COOP_TONE_DISTINCT_CH", nullptr);
}
return 0;
if (!launched)
{
return fv;
}
auto cleanup = [&] {
TerminateProcess(pi.hProcess, 0);
@@ -155,73 +156,104 @@ int main()
CloseHandle(pi.hProcess);
kill_stray_mock_games();
};
Sleep(800); // window + audio client up
Sleep(800);
// IPC + the primary audio ring the hook produces into.
SharedMemory shm;
if (!shm.create(shared_memory_name(pi.dwProcessId), sizeof(SharedBlock)))
SharedMemory ring_shm;
if (!shm.create(shared_memory_name(pi.dwProcessId), sizeof(SharedBlock)) ||
!ring_shm.create(audio_ring_name(pi.dwProcessId), audio_ring_total_size(kAudioRingCapacity)))
{
std::printf("Could not create IPC block -- skipping.\n");
cleanup();
return 0;
return fv;
}
auto* block = shm.as<SharedBlock>();
block->version = kProtocolVersion;
block->pad_count = 0;
block->sequence.store(0, std::memory_order_relaxed);
// Only the audio subsystem.
for (std::uint32_t s = 0; s < HookSubsys_Count; ++s)
for (std::uint32_t s = 0; s < HookSubsys_Count; ++s) // audio subsystem only
{
const bool off = s != HookSubsys_Audio;
block->control.subsystem_disabled[s].store(off ? 1u : 0u, std::memory_order_release);
block->control.subsystem_disabled[s].store(s != HookSubsys_Audio ? 1u : 0u, std::memory_order_release);
}
block->magic = kProtocolMagic;
SharedMemory ring_shm;
if (!ring_shm.create(audio_ring_name(pi.dwProcessId), audio_ring_total_size(kAudioRingCapacity)))
{
std::printf("Could not create audio ring -- skipping.\n");
cleanup();
return 0;
}
auto* ring = ring_shm.as<AudioRingHeader>();
audio_ring_init(*ring, kAudioRingCapacity);
// Leave capture_enabled = 0: we want the stream audible (so loopback hears it) and still being
// MEASURED (so the verify tap fires), exactly the window verify_stream_format targets.
audio_ring_init(*ring, kAudioRingCapacity); // capture_enabled stays 0: audible + still measuring
if (!inject_retry(pi.dwProcessId))
{
std::printf("Could not inject coop_hook.dll -- skipping.\n");
cleanup();
return 0;
return fv;
}
Sleep(500); // let the hook attach + the pre-existing render client register as a guess
Sleep(500); // hook attaches + the pre-existing render client registers as a guess
// Run the real verifier: co-capture hook (pre-mix) + loopback (post-mix) and correlate.
const FormatVerification fv = verify_stream_format(pi.dwProcessId, ring, /*window_ms=*/1400);
std::printf(" verify: ok=%d rate=%u score=%.3f\n", fv.ok ? 1 : 0, fv.rate, fv.score);
fv = verify_stream_format(pi.dwProcessId, ring, /*window_ms=*/1400, sc.recover_layout);
ran = true;
cleanup();
return fv;
}
} // namespace
if (!fv.ok && fv.rate == 0 && fv.score == 0.0)
int main()
{
const bool com = SUCCEEDED(CoInitializeEx(nullptr, COINIT_MULTITHREADED));
unsigned dev_rate = 48000, dev_channels = 2;
if (WAVEFORMATEX* dev = default_render_format())
{
// No audio endpoint, or no usable audio captured (e.g. the mock's WASAPI client never
// started on this machine) -> treat as a skip rather than a failure.
std::printf(" no usable co-capture (no endpoint / silent) -- skipping audio_verify_test.\n");
dev_rate = dev->nSamplesPerSec;
dev_channels = dev->nChannels;
CoTaskMemFree(dev);
}
const unsigned mismatched = (dev_rate == 44100) ? 48000u : 44100u; // guarantee a rate mismatch
std::printf("device: %u Hz / %u ch\n", dev_rate, dev_channels);
// (a) Rate: render at the device's channel count (no layout mismatch) but a different rate.
std::printf("== (a) rate recovery: %u Hz / %u ch ==\n", mismatched, dev_channels);
bool ran = false;
FormatVerification a = run({mismatched, dev_channels, 32, /*distinct=*/false, /*recover_layout=*/false}, ran);
if (!ran)
{
std::printf(" environment can't run the mock+inject -- skipping audio_verify_test.\n");
if (com)
{
CoUninitialize();
}
cleanup();
return 0;
}
std::printf(" ok=%d rate=%u score=%.3f\n", a.ok ? 1 : 0, a.rate, a.score);
if (!a.ok && a.rate == 0 && a.score == 0.0)
{
std::printf(" no usable co-capture (no endpoint / silent) -- skipping.\n");
if (com)
{
CoUninitialize();
}
return 0;
}
check(a.ok, "(a) verifier confidently correlated the two capture paths");
check(a.rate == mismatched, "(a) recovered the game's true rate (not the device rate)");
check(fv.ok, "verifier confidently correlated the two capture paths");
check(fv.rate == game_rate, "verifier recovered the game's true rate (not the device rate)");
// (b) Layout: render 2ch with distinct per-channel content -- a layout that differs from a
// multichannel device -- and recover channels + bit depth + rate.
std::printf("== (b) layout recovery: 44100 Hz / 2 ch / 32-bit float (distinct channels) ==\n");
FormatVerification b = run({44100, 2, 32, /*distinct=*/true, /*recover_layout=*/true}, ran);
std::printf(" ok=%d rate=%u ch=%u bits=%u tag=%u score=%.3f\n", b.ok ? 1 : 0, b.rate, b.channels, b.bits,
b.format_tag, b.score);
if (b.ok || b.score > 0.0)
{
check(b.layout_ok, "(b) verifier confidently recovered the layout");
check(b.rate == 44100, "(b) recovered the true rate");
check(b.channels == 2, "(b) recovered the true channel count (2, not the device's)");
check(b.bits == 32 && b.format_tag == 3, "(b) recovered 32-bit float");
}
else
{
std::printf(" no usable co-capture for (b) -- skipping that scenario.\n");
}
if (com)
{
CoUninitialize();
}
cleanup();
if (g_failures == 0)
{

25
tools/audio_tone/tone_source.hpp
View File

@@ -7,6 +7,7 @@
#include <cmath>
#include <cstdint>
#include <cstdlib>
#include <windows.h>
@@ -87,6 +88,13 @@ public:
client_->GetBufferSize(&buffer_frames_);
step_ = kTwoPi * freq_hz / static_cast<double>(fmt_.rate);
// Optional: give each channel genuinely different content (a per-channel frequency scale),
// so a downstream test can *recover* the channel count by correlation (identical channels
// are ambiguous: 2ch@R looks like 1ch@2R). Off by default -> the usual single-tone source.
if (const char* d = std::getenv("COOP_TONE_DISTINCT_CH"); d != nullptr && d[0] == '1')
{
distinct_ = true;
}
write(buffer_frames_); // pre-roll
client_->Start();
return true;
@@ -220,13 +228,24 @@ private:
}
for (unsigned c = 0; c < fmt_.channels; ++c)
{
double sc = s;
if (distinct_ && c < 8)
{
// Each channel at its own frequency scale -> genuinely different content.
sc = std::sin(phase_c_[c]) * 0.25;
phase_c_[c] += step_ * (1.0 + 0.37 * static_cast<double>(c));
if (phase_c_[c] > kTwoPi)
{
phase_c_[c] -= kTwoPi;
}
}
if (float_)
{
reinterpret_cast<float*>(data)[i * fmt_.channels + c] = static_cast<float>(s);
reinterpret_cast<float*>(data)[i * fmt_.channels + c] = static_cast<float>(sc);
}
else
{
reinterpret_cast<INT16*>(data)[i * fmt_.channels + c] = static_cast<INT16>(s * 32767.0);
reinterpret_cast<INT16*>(data)[i * fmt_.channels + c] = static_cast<INT16>(sc * 32767.0);
}
}
}
@@ -243,6 +262,8 @@ private:
bool float_ = false;
double phase_ = 0.0;
double step_ = 0.0;
bool distinct_ = false; // per-channel distinct content (recoverable channel count)
double phase_c_[8] = {}; // per-channel phase when distinct_
};
} // namespace coop::tone