fix(audio): buffer capture frames to Opus encoder's fixed frame size

AudioEngine::on_capture() was passing miniaudio's hardware callback period
(commonly 480 samples / 10 ms on WASAPI shared mode) directly to opus_encode(),
which requires exactly frame_samples_ (960 for 20 ms @ 48 kHz). The mismatch
returned OPUS_BAD_ARG and silently dropped every real mic frame, while screen
share and injected test frames happened to be correctly sized and worked fine.

Fix: accumulate PCM in a pre-allocated CaptureAccum buffer (mirroring the
existing RemoteStream::ring fix on the playback side) and only call capture_cb_
when a full frame_samples_ chunk is ready. Same pattern applied to on_loopback().

Add test_capture_frame_accumulation() to verify the accumulator fires exactly the
right number of callbacks for misaligned chunk sizes (480, 240+720, 1920 samples).

Co-Authored-By: Claude Sonnet 4.6 <noreply@anthropic.com>

core/src/audio/audio_engine.cpp

@@ -123,6 +123,16 @@ bool AudioEngine::start(const AudioParams& p, CaptureCallback capture_cb) {
     frame_samples_ = static_cast<int>(p.sample_rate / 1000 * p.frame_ms);
     running_.store(true, std::memory_order_release);
 
+#ifdef VOICECAT_HAS_AUDIO
+    // Pre-allocate capture accumulators before the devices start so on_capture / on_loopback
+    // never allocate on the RT thread. count=0 means "empty"; the buf is sized to exactly one
+    // encoder frame so a memcpy into it can never overrun.
+    capture_accum_.buf.assign(static_cast<size_t>(frame_samples_), 0);
+    capture_accum_.count = 0;
+    loopback_accum_.buf.assign(static_cast<size_t>(frame_samples_), 0);
+    loopback_accum_.count = 0;
+#endif
+
 #ifdef VOICECAT_HAS_AUDIO
     // ── Capture device ──────────────────────────────────────────────────────
     ma_device_id cap_id{};
@@ -136,6 +146,10 @@ bool AudioEngine::start(const AudioParams& p, CaptureCallback capture_cb) {
     cap_cfg.dataCallback       = capture_data_cb;
     cap_cfg.pUserData          = this;
     cap_cfg.capture.pDeviceID  = have_cap_id ? &cap_id : nullptr;  // null = default device
+    // Hint: request the encoder's frame size as the callback period. WASAPI shared mode may
+    // not honor this (the hardware period is fixed), but when it is honored the accumulator
+    // below becomes a zero-copy passthrough rather than a copy every two callbacks.
+    cap_cfg.periodSizeInFrames = static_cast<ma_uint32>(frame_samples_);
 
     if (ma_device_init(nullptr, &cap_cfg, &capture_device_) == MA_SUCCESS) {
         if (ma_device_start(&capture_device_) == MA_SUCCESS) {
@@ -347,10 +361,26 @@ void AudioEngine::capture_data_cb(ma_device* dev, void* /*out*/,
 }
 
 void AudioEngine::on_capture(const int16_t* pcm, ma_uint32 frames) {
-    // The real hardware capture device is always the "primary" tap (kind 0 / MIC). A second
+    // Accumulate samples until we have exactly frame_samples_ (e.g. 960 for 20 ms @ 48 kHz),
-    // concurrent local stream (e.g. SCREEN_AUDIO) is fed via inject_capture() in M3 — there is
+    // then fire capture_cb_. WASAPI shared mode commonly delivers 480-sample (10 ms) callbacks
-    // only one real capture device.
+    // regardless of the periodSizeInFrames hint above; passing a sub-frame chunk directly to
-    if (capture_cb_) capture_cb_(0, pcm, static_cast<int>(frames));
+    // opus_encode() returns OPUS_BAD_ARG (negative), silently dropping every mic frame.
+    if (!capture_cb_ || frame_samples_ <= 0) return;
+    const int16_t* src = pcm;
+    auto remaining = static_cast<int>(frames);
+    while (remaining > 0) {
+        int space = frame_samples_ - capture_accum_.count;
+        int copy  = std::min(remaining, space);
+        std::memcpy(capture_accum_.buf.data() + capture_accum_.count, src,
+                    static_cast<size_t>(copy) * sizeof(int16_t));
+        capture_accum_.count += copy;
+        src       += copy;
+        remaining -= copy;
+        if (capture_accum_.count == frame_samples_) {
+            capture_cb_(0, capture_accum_.buf.data(), frame_samples_);
+            capture_accum_.count = 0;
+        }
+    }
 }
 
 void AudioEngine::playback_data_cb(ma_device* dev, void* out,
@@ -444,13 +474,29 @@ void AudioEngine::loopback_data_cb(ma_device* dev, void* /*out*/, const void* in
 }
 
 void AudioEngine::on_loopback(const int16_t* pcm, ma_uint32 frames) {
-    // Same pattern as the real mic capture callback (on_capture) — call capture_cb_ directly,
+    // Same accumulation as on_capture — the WASAPI loopback render endpoint's callback
-    // NOT through inject_capture()'s test-only ring (see audio_engine.h).
+    // period is also hardware-driven and may not match frame_samples_.
-    if (capture_cb_) capture_cb_(loopback_kind_, pcm, static_cast<int>(frames));
+    if (!capture_cb_ || frame_samples_ <= 0) return;
+    const int16_t* src = pcm;
+    auto remaining = static_cast<int>(frames);
+    while (remaining > 0) {
+        int space = frame_samples_ - loopback_accum_.count;
+        int copy  = std::min(remaining, space);
+        std::memcpy(loopback_accum_.buf.data() + loopback_accum_.count, src,
+                    static_cast<size_t>(copy) * sizeof(int16_t));
+        loopback_accum_.count += copy;
+        src       += copy;
+        remaining -= copy;
+        if (loopback_accum_.count == frame_samples_) {
+            capture_cb_(loopback_kind_, loopback_accum_.buf.data(), frame_samples_);
+            loopback_accum_.count = 0;
+        }
+    }
 }
 
 bool AudioEngine::start_loopback_capture(int kind) {
     if (loopback_started_) return false;  // already running; stop_loopback_capture() first
+    loopback_accum_.count = 0;  // discard any partial frame from a previous loopback session
 
     ma_device_config cfg = ma_device_config_init(ma_device_type_loopback);
     cfg.capture.format    = ma_format_s16;

core/src/audio/audio_engine.h

@@ -14,6 +14,7 @@
 
 #include <algorithm>
 #include <atomic>
+#include <cstring>
 #include <cstdint>
 #include <functional>
 #include <map>
@@ -168,6 +169,30 @@ class AudioEngine {
     // stereo mixing end-to-end (no audio hardware needed). Same logic the real playback
     // callback uses; safe to call any time after start() (no ma_device touched).
     void mix_for_test(int16_t* out, uint32_t frames) { on_playback(out, frames); }
+
+    // TEST-ONLY — drives the capture-side frame accumulator directly with an explicit
+    // callback, bypassing capture_cb_. Call after engine.start(p) WITHOUT a capture
+    // callback so the real mic (if any) never touches capture_accum_ (on_capture returns
+    // early when capture_cb_ is null). The explicit `cb` is invoked only when a full
+    // frame_samples_ chunk is ready — that is the invariant under test.
+    void feed_capture_for_test(const int16_t* pcm, int frames, const CaptureCallback& cb) {
+        if (frame_samples_ <= 0 || !cb || capture_accum_.buf.empty()) return;
+        const int16_t* src = pcm;
+        auto remaining = frames;
+        while (remaining > 0) {
+            int space = frame_samples_ - capture_accum_.count;
+            int copy  = std::min(remaining, space);
+            std::memcpy(capture_accum_.buf.data() + capture_accum_.count, src,
+                        static_cast<size_t>(copy) * sizeof(int16_t));
+            capture_accum_.count += copy;
+            src       += copy;
+            remaining -= copy;
+            if (capture_accum_.count == frame_samples_) {
+                cb(0, capture_accum_.buf.data(), frame_samples_);
+                capture_accum_.count = 0;
+            }
+        }
+    }
 #endif
 
    private:
@@ -199,6 +224,20 @@ class AudioEngine {
     CaptureCallback  capture_cb_;
     std::atomic<bool> running_{false};
 
+    // Capture-side frame accumulators: miniaudio fires the capture (and loopback) callback at
+    // whatever period the hardware/driver chooses — commonly 480 samples (10 ms) on WASAPI
+    // shared mode, while the Opus encoder requires exactly frame_samples_ per call (960 for
+    // 20 ms @ 48 kHz). Accumulate incoming PCM until a full frame is ready, then call
+    // capture_cb_. This mirrors the RemoteStream::ring fix on the playback side. Both
+    // accumulators are pre-allocated once in start(); never resized from the RT callback
+    // thread (satisfies architecture.md §3 — no allocation on RT threads).
+    struct CaptureAccum {
+        std::vector<int16_t> buf;    // pre-sized to frame_samples_ in start()
+        int                  count = 0;
+    };
+    CaptureAccum capture_accum_;   // mic / real capture device (on_capture)
+    CaptureAccum loopback_accum_;  // screen-audio WASAPI loopback (on_loopback)
+
     // Inject ring(s): stores raw int16 PCM written by inject_capture(), one ring per local
     // stream kind so e.g. MIC and SCREEN_AUDIO can each be fed independently in tests.
     // The encode thread reads from these (no real capture device needed in tests).

tests/test_vad_ptt_devices.cpp

@@ -231,6 +231,60 @@ static void test_stereo_mix() {
 }
 #endif  // VOICECAT_HAS_AUDIO && VOICECAT_HAS_OPUS
 
+// ── 5. Capture-frame accumulation (white-box, no audio hardware needed) ──────────
+// Regression for the capture-side analogue of the playback ring fix: miniaudio's capture
+// callback fires at the hardware period (commonly 480 samples on WASAPI shared mode), while
+// opus_encode() requires exactly frame_samples_ (960). Sub-frame chunks must be accumulated;
+// the callback must receive exactly 960-sample frames regardless of input chunk size.
+#ifdef VOICECAT_HAS_AUDIO
+static void test_capture_frame_accumulation() {
+    voicecat::audio::AudioEngine engine;
+    voicecat::audio::AudioParams p;
+    p.sample_rate      = 48000;
+    p.capture_channels = 1;
+    p.frame_ms         = 20;  // frame_samples_ = 960
+
+    std::atomic<int>  call_count{0};
+    std::atomic<bool> wrong_size{false};
+    constexpr int kExpected = 960;
+
+    // Start WITHOUT a capture callback: the real mic (if any) fires on_capture(), but
+    // on_capture() returns immediately when capture_cb_ is null, so capture_accum_ is
+    // never touched by the hardware thread. feed_capture_for_test() bypasses capture_cb_
+    // and drives the same accumulator directly with the explicit `cb` below — no races.
+    CHECK(engine.start(p));
+
+    auto cb = [&](int /*kind*/, const int16_t* /*pcm*/, int samples) {
+        ++call_count;
+        if (samples != kExpected) wrong_size.store(true);
+    };
+
+    // 480-sample (10 ms) input — WASAPI's common hardware period on modern Windows.
+    // Two 480-chunk inputs → exactly one callback at 960.
+    std::vector<int16_t> h(480, 1000);
+    engine.feed_capture_for_test(h.data(), 480, cb);
+    CHECK(call_count.load() == 0);  // half a frame — no callback yet
+    engine.feed_capture_for_test(h.data(), 480, cb);
+    CHECK(call_count.load() == 1);  // one full frame — callback fired once
+
+    // Mis-aligned split: 240 then 720 → still exactly one callback.
+    std::vector<int16_t> s(240, 500), l(720, 500);
+    engine.feed_capture_for_test(s.data(), 240, cb);
+    CHECK(call_count.load() == 1);
+    engine.feed_capture_for_test(l.data(), 720, cb);
+    CHECK(call_count.load() == 2);
+
+    // 1920-sample input (two Opus frames) → exactly two callbacks.
+    std::vector<int16_t> d(1920, 800);
+    engine.feed_capture_for_test(d.data(), 1920, cb);
+    CHECK(call_count.load() == 4);
+
+    CHECK(!wrong_size.load());
+    engine.stop();
+    std::printf("test_capture_frame_accumulation: ok (callbacks=%d)\n", call_count.load());
+}
+#endif  // VOICECAT_HAS_AUDIO
+
 // ── 2/3. VAD + PTT gate, through the real ABI against a real server ─────────────
 static void test_vad_and_ptt_gate() {
     auto tmp = std::filesystem::temp_directory_path() /
@@ -385,6 +439,9 @@ int main() {
     test_device_enumeration();
 #if defined(VOICECAT_HAS_AUDIO) && defined(VOICECAT_HAS_OPUS)
     test_stereo_mix();
+#endif
+#ifdef VOICECAT_HAS_AUDIO
+    test_capture_frame_accumulation();
 #endif
     test_vad_and_ptt_gate();