/*
 * test_vad_ptt_devices — closes M3's "explicitly out of scope" gaps (PROGRESS.md): device
 * enumeration, the VAD/PTT send-side input gate, and true stereo playback mixing.
 *
 * Mirrors test_m3_multistream.cpp's approach (real vc_client instances against a real
 * in-process server, not raw sockets) for the ABI-level pieces, plus a white-box AudioEngine
 * test for the stereo mixer (no audio hardware needed — see AudioEngine::mix_for_test).
 *
 *   1. Device enumeration (vc_list_devices) works pre-connect, for both kinds, and tolerates
 *      an empty list (headless CI build agents may have zero audio devices) — VC_OK is the
 *      only thing asserted, never count > 0.
 *   2. VAD gate: under VC_INPUT_VOICE_ACTIVATION (the default), silent PCM never reaches the
 *      peer (no talking edge); loud PCM does.
 *   3. PTT gate: under VC_INPUT_PUSH_TO_TALK, loud PCM is gated closed until
 *      vc_set_push_to_talk(1); then it reaches the peer.
 *   4. Stereo playback mixer: white-box (AudioEngine directly) — a genuinely stereo decoded
 *      stream survives into the mix without being downmixed to mono.
 */
#include <cstdio>

#ifdef VOICECAT_HAS_NET

#include <atomic>
#include <chrono>
#include <cmath>
#include <condition_variable>
#include <cstdlib>
#include <filesystem>
#include <mutex>
#include <string>
#include <thread>
#include <vector>

#include "voicecat.h"
#include "server.h"
#include "db.h"

#ifdef VOICECAT_HAS_AUDIO
#include "audio/audio_engine.h"
#endif
#ifdef VOICECAT_HAS_OPUS
#include "codec/opus_codec.h"
#endif

// ── Event tracking (same shape as test_m3_multistream.cpp) ──────────────────────

struct TalkEvent {
    uint32_t user_id;
    uint32_t stream_id;
    bool     talking;
};

struct EventStore {
    std::mutex              mu;
    std::condition_variable cv;

    bool     auth_ok{false};
    uint32_t self_user_id{0};
    bool     channel_list_received{false};
    bool     saw_stream_started{false};
    std::vector<TalkEvent> talk_events;
    bool     disconnected{false};

    const char* label{nullptr};

    // Set right after vc_client_create, before vc_connect — lets on_event auto-confirm the
    // M4 TOFU gate (VC_EVENT_SERVER_IDENTITY below) for this headless test.
    vc_client* client{nullptr};
};

static void on_event(void* user, const vc_event* ev) {
    auto* s = static_cast<EventStore*>(user);
    std::lock_guard lk(s->mu);
    switch (ev->type) {
        case VC_EVENT_SERVER_IDENTITY:
            // No human to ask in a headless test — trust on first connect unconditionally.
            vc_confirm_server_identity(s->client, 1);
            break;
        case VC_EVENT_AUTH_RESULT:
            s->auth_ok      = (ev->result == VC_OK);
            s->self_user_id = ev->user_id;
            break;
        case VC_EVENT_CHANNEL_LIST:
            s->channel_list_received = true;
            break;
        case VC_EVENT_STREAM_STARTED:
            s->saw_stream_started = true;
            break;
        case VC_EVENT_TALK_STATE:
            s->talk_events.push_back({ev->user_id, ev->stream_id, ev->u32a != 0});
            break;
        case VC_EVENT_DISCONNECTED:
            s->disconnected = true;
            break;
        default:
            break;
    }
    s->cv.notify_all();
}

template <typename Pred>
static bool wait_for(EventStore& s, Pred pred, int timeout_ms) {
    auto deadline = std::chrono::steady_clock::now() + std::chrono::milliseconds(timeout_ms);
    std::unique_lock lk(s.mu);
    return s.cv.wait_until(lk, deadline, [&] { return pred(s); });
}

static std::vector<int16_t> make_sine_frame(int frame_idx, float freq_hz,
                                            int frame_samples = 960) {
    std::vector<int16_t> pcm(frame_samples);
    for (int i = 0; i < frame_samples; ++i) {
        float t = static_cast<float>(frame_idx * frame_samples + i) / 48000.0f;
        pcm[i] = static_cast<int16_t>(std::sin(2.0f * 3.14159265f * freq_hz * t) * 16000.0f);
    }
    return pcm;
}

static std::vector<int16_t> make_silence_frame(int frame_samples = 960) {
    return std::vector<int16_t>(frame_samples, 0);
}

// Did `talking==true` ever fire for (user_id, stream_id) at index >= `from`?
static bool saw_talking_true(EventStore& s, uint32_t user_id, uint32_t stream_id, size_t from) {
    std::lock_guard lk(s.mu);
    for (size_t i = from; i < s.talk_events.size(); ++i) {
        auto& e = s.talk_events[i];
        if (e.user_id == user_id && e.stream_id == stream_id && e.talking) return true;
    }
    return false;
}

static size_t talk_event_count(EventStore& s) {
    std::lock_guard lk(s.mu);
    return s.talk_events.size();
}

// ── Test harness ──────────────────────────────────────────────────────────────

static int g_failures = 0;
#define CHECK(cond)                                                        \
    do {                                                                   \
        if (!(cond)) {                                                     \
            std::printf("FAIL: %s (%s:%d)\n", #cond, __FILE__, __LINE__); \
            ++g_failures;                                                  \
        }                                                                  \
    } while (0)

// ── 1. Device enumeration (no server needed) ────────────────────────────────────
static void test_device_enumeration() {
    vc_config cfg{"test-devices", "0.1", VC_LOG_OFF};
    vc_callbacks cb{};
    vc_client* c = vc_client_create(&cfg, cb);
    CHECK(c != nullptr);

    for (vc_device_kind kind : {VC_DEVICE_INPUT, VC_DEVICE_OUTPUT}) {
        vc_device_list dl{};
        vc_result      r = vc_list_devices(c, kind, &dl);
#ifdef VOICECAT_HAS_AUDIO
        CHECK(r == VC_OK);
        // Headless CI build agents may legitimately report zero devices — never assert
        // count > 0, only that the call itself succeeded and the list is well-formed.
        for (size_t i = 0; i < dl.count; ++i) {
            CHECK(dl.items[i].id != nullptr);
            CHECK(dl.items[i].name != nullptr);
        }
#else
        CHECK(r == VC_ERR_NOT_IMPLEMENTED);
#endif
        vc_free_device_list(&dl);
        vc_free_device_list(&dl);  // idempotent — must not crash on a second call
    }

    vc_client_destroy(c);
    std::printf("test_device_enumeration: ok\n");
}

// ── 4. Stereo playback mixer (white-box, no audio hardware needed) ──────────────
#if defined(VOICECAT_HAS_AUDIO) && defined(VOICECAT_HAS_OPUS)
static void test_stereo_mix() {
    voicecat::audio::AudioEngine engine;
    voicecat::audio::AudioParams p;
    p.sample_rate       = 48000;
    p.capture_channels  = 1;
    p.playback_channels = 2;
    p.frame_ms          = 20;
    CHECK(engine.start(p));  // capture_cb intentionally omitted — not exercised here

    voicecat::codec::OpusParams stereo_params;
    stereo_params.stereo = true;
    int frame_samples = voicecat::codec::opus_frame_samples(stereo_params);

    voicecat::codec::OpusEncoder enc;
    CHECK(enc.init(stereo_params));

    // Loud left channel, silent right channel — a real downmix would average them into a
    // single audible-but-quieter centered sample; true stereo should keep them distinct.
    std::vector<int16_t> interleaved(static_cast<size_t>(frame_samples) * 2);
    for (int i = 0; i < frame_samples; ++i) {
        float t = static_cast<float>(i) / 48000.0f;
        interleaved[i * 2]     = static_cast<int16_t>(std::sin(2.0f * 3.14159265f * 440.0f * t) * 20000.0f);
        interleaved[i * 2 + 1] = 0;
    }

    uint8_t opus_buf[1500];
    int opus_len = enc.encode(interleaved.data(), frame_samples, opus_buf, sizeof(opus_buf));
    CHECK(opus_len > 0);

    engine.init_recv_stream(/*ssrc=*/1, stereo_params, /*user_id=*/0, /*stream_id=*/0);

    voicecat::audio::JitterBuffer::Frame f;
    f.seq         = 0;
    f.timestamp   = 0;
    f.fec_present = false;
    f.payload.assign(opus_buf, opus_buf + opus_len);
    engine.push_recv_frame(1, std::move(f));

    std::vector<int16_t> out(static_cast<size_t>(frame_samples) * 2, 0);
    engine.mix_for_test(out.data(), static_cast<uint32_t>(frame_samples));

    // If the engine downmixed (old M3 behavior), every L/R pair would be identical (the
    // average of a loud sample and 0). True stereo should show a clear, consistent L != R
    // difference across the frame.
    int64_t total_diff = 0;
    for (int i = 0; i < frame_samples; ++i)
        total_diff += std::abs(static_cast<int>(out[i * 2]) - static_cast<int>(out[i * 2 + 1]));
    CHECK(total_diff > static_cast<int64_t>(frame_samples) * 1000);  // well above decode noise

    engine.remove_stream(1);
    engine.stop();
    std::printf("test_stereo_mix: ok (total_diff=%lld)\n", static_cast<long long>(total_diff));
}

// ── 4a-2. Stereo screen-audio loopback capture (white-box, no audio hardware needed) ──
// Regression for the mono-loopback bug: start_loopback_capture used to hardcode channels=1,
// downmixing the system's stereo mix to mono before the encoder ever saw it (and on_capture_frame
// then upmixed L=R to produce a fake-stereo bitstream). Now the loopback device opens in the
// channel's mode (stereo when the channel is stereo), so the encoder receives real interleaved
// L/R PCM and encodes it directly. This test drives feed_loopback_for_test with a loud-L /
// silent-R stereo signal, encodes it (as on_capture_frame now does for channels==2), decodes,
// and mixes — asserting L != R across the frame. A mono-downmixed-then-upmixed bitstream would
// have L == R. Mirrors test_stereo_mix but routes the encode side through the loopback
// accumulator path that the fix touches (feed_loopback_for_test → on_loopback's accumulator).
#if defined(VOICECAT_HAS_LOOPBACK) && defined(VOICECAT_HAS_OPUS)
static void test_loopback_stereo_capture() {
    voicecat::audio::AudioEngine engine;
    voicecat::audio::AudioParams p;
    p.sample_rate       = 48000;
    p.capture_channels  = 1;   // mic path — irrelevant here; loopback has its own channel count
    p.playback_channels = 2;   // stereo mix output (for mix_for_test below)
    p.frame_ms          = 20;
    CHECK(engine.start(p));  // no capture_cb — the real mic (if any) won't touch capture_accum_

    voicecat::codec::OpusParams stereo_params;
    stereo_params.stereo      = true;
    stereo_params.application = voicecat::codec::OpusApplication::Audio;  // screen-audio channel
    stereo_params.bitrate_bps = 128000;  // music/screen-audio channel default
    int frame_samples = voicecat::codec::opus_frame_samples(stereo_params);

    voicecat::codec::OpusEncoder enc;
    CHECK(enc.init(stereo_params));

    // Loud left channel, silent right — a real mono downmix would average them into a single
    // audible-but-quieter centered sample; true stereo keeps them distinct.
    std::vector<int16_t> interleaved(static_cast<size_t>(frame_samples) * 2);
    for (int i = 0; i < frame_samples; ++i) {
        float t = static_cast<float>(i) / 48000.0f;
        interleaved[i * 2]     = static_cast<int16_t>(std::sin(2.0f * 3.14159265f * 440.0f * t) * 20000.0f);
        interleaved[i * 2 + 1] = 0;
    }

    // Encode via the loopback accumulator path: feed_loopback_for_test drives on_loopback's
    // accumulator and invokes the callback with channels=2 (the fix). The callback encodes
    // exactly as on_capture_frame does for real-stereo SCREEN_AUDIO PCM — no upmix.
    uint8_t opus_buf[1500];
    int opus_len = 0;
    int seen_channels = 0;
    auto cb = [&](int /*kind*/, const int16_t* pcm, int /*samples*/, int channels) {
        seen_channels = channels;
        if (channels == 2) {
            // The loopback accumulator must have preserved L/R distinctness pre-encode.
            int64_t pre_diff = 0;
            for (int i = 0; i < frame_samples; ++i)
                pre_diff += std::abs(static_cast<int>(pcm[i * 2]) - static_cast<int>(pcm[i * 2 + 1]));
            CHECK(pre_diff > static_cast<int64_t>(frame_samples) * 1000);
        }
        opus_len = enc.encode(pcm, frame_samples, opus_buf, sizeof(opus_buf));
    };
    engine.feed_loopback_for_test(interleaved.data(), frame_samples, 2, cb);
    CHECK(seen_channels == 2);  // the loopback path reported stereo, not downmixed mono
    CHECK(opus_len > 0);

    // Decode + mix — same recv path as test_stereo_mix. A real stereo bitstream should
    // survive with L != R; a mono-downmixed-then-upmixed bitstream would have L == R.
    engine.init_recv_stream(/*ssrc=*/3, stereo_params, /*user_id=*/0, /*stream_id=*/0);
    voicecat::audio::JitterBuffer::Frame f;
    f.seq         = 0;
    f.timestamp   = 0;
    f.fec_present = false;
    f.payload.assign(opus_buf, opus_buf + opus_len);
    engine.push_recv_frame(3, std::move(f));

    std::vector<int16_t> out(static_cast<size_t>(frame_samples) * 2, 0);
    engine.mix_for_test(out.data(), static_cast<uint32_t>(frame_samples));

    int64_t total_diff = 0;
    for (int i = 0; i < frame_samples; ++i)
        total_diff += std::abs(static_cast<int>(out[i * 2]) - static_cast<int>(out[i * 2 + 1]));
    CHECK(total_diff > static_cast<int64_t>(frame_samples) * 1000);

    engine.remove_stream(3);
    engine.stop();
    std::printf("test_loopback_stereo_capture: ok (total_diff=%lld, seen_channels=%d)\n",
                static_cast<long long>(total_diff), seen_channels);
}
#endif

// ── 4b. Playout-clock re-sync after a late join / silence gap ────────────────────
// Regression for the "talk indicator lit, no audio" bug: the playout clock free-runs (it
// advances every callback via PLC), while the sender's frame timestamps only advance while it
// is actually transmitting. After a silence gap or a late join the clock drifts past the jitter
// buffer's 500 ms late-drop window, so every real frame is dropped-as-late and the stream is
// permanently silent. on_playback must re-seed the clock to the earliest buffered frame.
static void test_playout_resync() {
    voicecat::audio::AudioEngine engine;
    voicecat::audio::AudioParams p;
    p.sample_rate       = 48000;
    p.capture_channels  = 1;
    p.playback_channels = 2;
    p.frame_ms          = 20;
    CHECK(engine.start(p));

    voicecat::codec::OpusParams mono_params;  // mono = the mic path
    mono_params.stereo = false;
    int frame_samples  = voicecat::codec::opus_frame_samples(mono_params);

    voicecat::codec::OpusEncoder enc;
    CHECK(enc.init(mono_params));

    std::vector<int16_t> sine(static_cast<size_t>(frame_samples));
    for (int i = 0; i < frame_samples; ++i) {
        float t = static_cast<float>(i) / 48000.0f;
        sine[i] = static_cast<int16_t>(std::sin(2.0f * 3.14159265f * 440.0f * t) * 20000.0f);
    }
    uint8_t opus_buf[1500];
    int opus_len = enc.encode(sine.data(), frame_samples, opus_buf, sizeof(opus_buf));
    CHECK(opus_len > 0);

    engine.init_recv_stream(/*ssrc=*/2, mono_params, /*user_id=*/0, /*stream_id=*/0);

    std::vector<int16_t> out(static_cast<size_t>(frame_samples) * 2, 0);

    // Free-run the playout clock with an empty jitter buffer (PLC every callback) far past the
    // 500 ms late-drop window — this is what a silence gap / late join does in the field.
    for (int i = 0; i < 100; ++i)  // ~100 frames @ 20 ms = ~2 s, well past 500 ms
        engine.mix_for_test(out.data(), static_cast<uint32_t>(frame_samples));

    // Now a real frame arrives carrying a timestamp far behind the free-run clock. Without the
    // re-sync it is dropped-as-late and playback stays silent; with it the clock snaps back and
    // the frame is decoded and mixed.
    voicecat::audio::JitterBuffer::Frame f;
    f.seq         = 0;
    f.timestamp   = 0;  // stream-relative start, now far behind the drifted playout clock
    f.fec_present = false;
    f.payload.assign(opus_buf, opus_buf + opus_len);
    engine.push_recv_frame(2, std::move(f));

    std::fill(out.begin(), out.end(), 0);
    engine.mix_for_test(out.data(), static_cast<uint32_t>(frame_samples));

    int64_t energy = 0;
    for (int16_t s : out) energy += std::abs(static_cast<int>(s));
    CHECK(energy > static_cast<int64_t>(frame_samples) * 1000);  // audible, not PLC silence

    engine.remove_stream(2);
    engine.stop();
    std::printf("test_playout_resync: ok (energy=%lld)\n", static_cast<long long>(energy));
}
#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, int /*channels*/) {
        ++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() /
               ("vctest_vadptt_" + std::to_string(
                   std::chrono::steady_clock::now().time_since_epoch().count()));
    std::filesystem::create_directories(tmp);
    std::string data_dir = tmp.string();

    std::atomic<uint16_t> bound_port{0};
    std::mutex              ready_mu;
    std::condition_variable ready_cv;
    bool                    ready{false};

    voicecat::server::Config cfg;
    cfg.data_dir     = data_dir;
    cfg.bind_port    = 0;
    cfg.media_port   = 0;
    cfg.server_name  = "VoiceCat-VadPttTest";
    cfg.allow_guests = true;
    cfg.on_ready = [&](uint16_t p) {
        bound_port.store(p);
        { std::lock_guard lk(ready_mu); ready = true; }
        ready_cv.notify_all();
    };

    voicecat::server::Server server(cfg);
    std::thread server_thread([&] { server.run(); });

    {
        std::unique_lock lk(ready_mu);
        bool ok = ready_cv.wait_for(lk, std::chrono::seconds(10), [&] { return ready; });
        if (!ok) {
            std::printf("FAIL: server did not become ready within 10s\n");
            ++g_failures;
            server.stop();
            server_thread.join();
            std::filesystem::remove_all(tmp);
            return;
        }
    }

    uint16_t port = bound_port.load();
    std::printf("test_vad_and_ptt_gate: server ready on :%u\n", port);

    EventStore evA;
    evA.label = "A";
    vc_callbacks cbA{on_event, nullptr, &evA};
    vc_config cfgA{"test-A", "0.1", VC_LOG_OFF};
    vc_client* clientA = vc_client_create(&cfgA, cbA);
    CHECK(clientA != nullptr);
    evA.client = clientA;
    CHECK(vc_connect(clientA, "127.0.0.1", port) == VC_OK);
    CHECK(vc_authenticate_guest(clientA, "VP-A") == VC_OK);
    CHECK(wait_for(evA, [](EventStore& s) { return s.auth_ok; }, 8000));
    CHECK(wait_for(evA, [](EventStore& s) { return s.channel_list_received; }, 3000));

    EventStore evB;
    evB.label = "B";
    vc_callbacks cbB{on_event, nullptr, &evB};
    vc_config cfgB{"test-B", "0.1", VC_LOG_OFF};
    vc_client* clientB = vc_client_create(&cfgB, cbB);
    CHECK(clientB != nullptr);
    evB.client = clientB;
    CHECK(vc_connect(clientB, "127.0.0.1", port) == VC_OK);
    CHECK(vc_authenticate_guest(clientB, "VP-B") == VC_OK);
    CHECK(wait_for(evB, [](EventStore& s) { return s.auth_ok; }, 8000));
    CHECK(wait_for(evB, [](EventStore& s) { return s.channel_list_received; }, 3000));

    uint32_t a_uid = 0;
    { std::lock_guard lk(evA.mu); a_uid = evA.self_user_id; }

    std::this_thread::sleep_for(std::chrono::milliseconds(500));

    vc_stream_desc mic_desc{};
    mic_desc.kind  = VC_STREAM_MIC;
    mic_desc.label = "mic";
    uint32_t mic_sid = 0;
    CHECK(vc_stream_start(clientA, &mic_desc, &mic_sid) == VC_OK);
    CHECK(wait_for(evB, [](EventStore& s) { return s.saw_stream_started; }, 5000));
    CHECK(wait_for(evA, [](EventStore& s) { return s.saw_stream_started; }, 5000));

    // ── 2a. VAD mode (default), silent PCM: must NOT reach B as a talking edge ──
    CHECK(vc_set_input_mode(clientA, VC_INPUT_VOICE_ACTIVATION) == VC_OK);
    for (int i = 0; i < 15; ++i) {
        auto silence = make_silence_frame();
        CHECK(vc_test_inject_capture(clientA, mic_sid, silence.data(), silence.size()) == VC_OK);
        std::this_thread::sleep_for(std::chrono::milliseconds(20));
    }
    CHECK(!saw_talking_true(evB, a_uid, mic_sid, 0));

    // ── 2b. VAD mode, loud PCM: must reach B as a talking edge ──────────────────
    size_t mark = talk_event_count(evB);
    for (int i = 0; i < 20; ++i) {
        auto loud = make_sine_frame(i, 440.0f);
        CHECK(vc_test_inject_capture(clientA, mic_sid, loud.data(), loud.size()) == VC_OK);
        std::this_thread::sleep_for(std::chrono::milliseconds(20));
    }
    CHECK(wait_for(evB, [&](EventStore& s) {
        for (size_t i = mark; i < s.talk_events.size(); ++i) {
            auto& e = s.talk_events[i];
            if (e.user_id == a_uid && e.stream_id == mic_sid && e.talking) return true;
        }
        return false;
    }, 3000));

    // ── 3a. PTT mode, key up: loud PCM must NOT reach B as a new talking edge ───
    CHECK(vc_set_input_mode(clientA, VC_INPUT_PUSH_TO_TALK) == VC_OK);
    CHECK(vc_set_push_to_talk(clientA, 0) == VC_OK);
    // Let any in-flight VAD-driven talking state lapse (hang-time ~300ms) before measuring.
    std::this_thread::sleep_for(std::chrono::milliseconds(500));
    mark = talk_event_count(evB);
    for (int i = 0; i < 20; ++i) {
        auto loud = make_sine_frame(i, 440.0f);
        CHECK(vc_test_inject_capture(clientA, mic_sid, loud.data(), loud.size()) == VC_OK);
        std::this_thread::sleep_for(std::chrono::milliseconds(20));
    }
    std::this_thread::sleep_for(std::chrono::milliseconds(200));
    CHECK(!saw_talking_true(evB, a_uid, mic_sid, mark));

    // ── 3b. PTT mode, key down: loud PCM must reach B as a talking edge ─────────
    CHECK(vc_set_push_to_talk(clientA, 1) == VC_OK);
    mark = talk_event_count(evB);
    for (int i = 0; i < 20; ++i) {
        auto loud = make_sine_frame(i, 440.0f);
        CHECK(vc_test_inject_capture(clientA, mic_sid, loud.data(), loud.size()) == VC_OK);
        std::this_thread::sleep_for(std::chrono::milliseconds(20));
    }
    CHECK(wait_for(evB, [&](EventStore& s) {
        for (size_t i = mark; i < s.talk_events.size(); ++i) {
            auto& e = s.talk_events[i];
            if (e.user_id == a_uid && e.stream_id == mic_sid && e.talking) return true;
        }
        return false;
    }, 3000));

    { std::lock_guard lk(evA.mu); CHECK(!evA.disconnected); }
    { std::lock_guard lk(evB.mu); CHECK(!evB.disconnected); }

    vc_disconnect(clientA);
    vc_disconnect(clientB);
    vc_client_destroy(clientA);
    vc_client_destroy(clientB);

    server.stop();
    server_thread.join();
    std::filesystem::remove_all(tmp);

    std::printf("test_vad_and_ptt_gate: done\n");
}

// ── 5. Stereo mic capture (vc_set_capture_channels) ───────────────────────────
// Verifies that the mic capture accumulator path handles stereo (channels=2) correctly:
// the accumulator is sized to frame_samples_*capture_channels, on_capture forwards the
// correct channel count, and the encoder receives real interleaved L/R PCM (not a mono
// downmix). Mirrors test_loopback_stereo_capture but routes through the mic capture
// accumulator (feed_capture_for_test with channels=2) instead of the loopback path.
// This is the headless CI test for the iOS stereo built-in mic feature (Part D).
#if defined(VOICECAT_HAS_AUDIO) && defined(VOICECAT_HAS_OPUS)
static void test_stereo_mic_capture() {
    voicecat::audio::AudioEngine engine;
    voicecat::audio::AudioParams p;
    p.sample_rate       = 48000;
    p.capture_channels  = 2;   // stereo mic capture (vc_set_capture_channels path)
    p.playback_channels = 2;   // stereo mix output (for mix_for_test below)
    p.frame_ms          = 20;
    CHECK(engine.start(p));  // no capture_cb — the real mic (if any) won't touch capture_accum_

    voicecat::codec::OpusParams stereo_params;
    stereo_params.stereo      = true;
    stereo_params.application = voicecat::codec::OpusApplication::Voip;  // mic stream
    stereo_params.bitrate_bps = 64000;  // mic default
    int frame_samples = voicecat::codec::opus_frame_samples(stereo_params);

    voicecat::codec::OpusEncoder enc;
    CHECK(enc.init(stereo_params));

    // Loud left channel, silent right — a mono downmix would average them; true stereo
    // keeps them distinct (same signal as test_loopback_stereo_capture).
    std::vector<int16_t> interleaved(static_cast<size_t>(frame_samples) * 2);
    for (int i = 0; i < frame_samples; ++i) {
        float t = static_cast<float>(i) / 48000.0f;
        interleaved[i * 2]     = static_cast<int16_t>(std::sin(2.0f * 3.14159265f * 440.0f * t) * 20000.0f);
        interleaved[i * 2 + 1] = 0;
    }

    // Encode via the mic capture accumulator path: feed_capture_for_test with channels=2
    // drives on_capture's accumulator and invokes the callback with channels=2. The callback
    // encodes exactly as on_capture_frame does for channels==2 — direct stereo, no upmix.
    uint8_t opus_buf[1500];
    int opus_len = 0;
    int seen_channels = 0;
    auto cb = [&](int /*kind*/, const int16_t* pcm, int /*samples*/, int channels) {
        seen_channels = channels;
        if (channels == 2) {
            // The capture accumulator must have preserved L/R distinctness pre-encode.
            int64_t pre_diff = 0;
            for (int i = 0; i < frame_samples; ++i)
                pre_diff += std::abs(static_cast<int>(pcm[i * 2]) - static_cast<int>(pcm[i * 2 + 1]));
            CHECK(pre_diff > static_cast<int64_t>(frame_samples) * 1000);
        }
        opus_len = enc.encode(pcm, frame_samples, opus_buf, sizeof(opus_buf));
    };
    engine.feed_capture_for_test(interleaved.data(), frame_samples, 2, cb);
    CHECK(seen_channels == 2);  // the mic capture path reported stereo, not mono
    CHECK(opus_len > 0);

    // Decode + mix — same recv path as test_stereo_mix. A real stereo bitstream should
    // survive with L != R; a mono-downmixed-then-upmixed bitstream would have L == R.
    engine.init_recv_stream(/*ssrc=*/5, stereo_params, /*user_id=*/0, /*stream_id=*/0);
    voicecat::audio::JitterBuffer::Frame f;
    f.seq         = 0;
    f.timestamp   = 0;
    f.fec_present = false;
    f.payload.assign(opus_buf, opus_buf + opus_len);
    engine.push_recv_frame(5, std::move(f));

    std::vector<int16_t> out(static_cast<size_t>(frame_samples) * 2, 0);
    engine.mix_for_test(out.data(), static_cast<uint32_t>(frame_samples));

    int64_t total_diff = 0;
    for (int i = 0; i < frame_samples; ++i)
        total_diff += std::abs(static_cast<int>(out[i * 2]) - static_cast<int>(out[i * 2 + 1]));
    CHECK(total_diff > static_cast<int64_t>(frame_samples) * 1000);

    engine.remove_stream(5);
    engine.stop();
    std::printf("test_stereo_mic_capture: ok (total_diff=%lld, seen_channels=%d)\n",
                static_cast<long long>(total_diff), seen_channels);
}
#endif

// ── 6. Stereo mic capture on a MONO channel (downmix safety) ──────────────────
// A stereo mic (vc_set_capture_channels=2) can be enabled while on a mono channel. The mic
// then delivers interleaved L/R, but the channel's Opus encoder is mono. encode_and_send_frame
// must fold L/R to mono before encoding — handing interleaved pairs straight to a mono
// opus_encode makes it read 2× the samples it should (wrong pitch / garbage). This mirrors that
// fold and proves the result is a valid mono bitstream that decodes to the expected averaged
// signal, rather than half-length junk.
#if defined(VOICECAT_HAS_AUDIO) && defined(VOICECAT_HAS_OPUS)
static void test_stereo_mic_mono_channel() {
    voicecat::codec::OpusParams mono_params;
    mono_params.stereo      = false;  // mono channel — encoder is mono
    mono_params.application = voicecat::codec::OpusApplication::Voip;
    mono_params.bitrate_bps = 64000;
    const int frame_samples = voicecat::codec::opus_frame_samples(mono_params);

    voicecat::codec::OpusEncoder enc;
    CHECK(enc.init(mono_params));

    // Loud left, silent right — folding (L+R)/2 yields a half-amplitude tone on every sample.
    std::vector<int16_t> interleaved(static_cast<size_t>(frame_samples) * 2);
    for (int i = 0; i < frame_samples; ++i) {
        float t = static_cast<float>(i) / 48000.0f;
        interleaved[i * 2]     = static_cast<int16_t>(std::sin(2.0f * 3.14159265f * 440.0f * t) * 20000.0f);
        interleaved[i * 2 + 1] = 0;
    }

    // Fold exactly as encode_and_send_frame does for a stereo frame on a mono channel.
    std::vector<int16_t> folded(frame_samples);
    for (int i = 0; i < frame_samples; ++i)
        folded[i] = static_cast<int16_t>(
            (static_cast<int32_t>(interleaved[i * 2]) + static_cast<int32_t>(interleaved[i * 2 + 1])) / 2);

    uint8_t opus_buf[1500];
    int opus_len = enc.encode(folded.data(), frame_samples, opus_buf, sizeof(opus_buf));
    CHECK(opus_len > 0);

    // Decode mono and verify a full-length frame with real energy survived (a garbage half-read
    // would either fail to decode the full frame_samples or come back near-silent / wrong length).
    voicecat::codec::OpusDecoder dec;
    CHECK(dec.init(mono_params));
    std::vector<int16_t> decoded(frame_samples, 0);
    int dec_samples = dec.decode(opus_buf, opus_len, decoded.data(), frame_samples);
    CHECK(dec_samples == frame_samples);

    int64_t energy = 0;
    for (int i = 0; i < frame_samples; ++i) energy += std::abs(static_cast<int>(decoded[i]));
    CHECK(energy > static_cast<int64_t>(frame_samples) * 500);  // clearly audible, not silence

    std::printf("test_stereo_mic_mono_channel: ok (opus_len=%d, energy=%lld)\n",
                opus_len, static_cast<long long>(energy));
}
#endif

int main() {
    test_device_enumeration();
#if defined(VOICECAT_HAS_AUDIO) && defined(VOICECAT_HAS_OPUS)
    test_stereo_mix();
#if defined(VOICECAT_HAS_LOOPBACK)
    test_loopback_stereo_capture();
#endif
    test_stereo_mic_capture();
    test_stereo_mic_mono_channel();
    test_playout_resync();
#endif
#ifdef VOICECAT_HAS_AUDIO
    test_capture_frame_accumulation();
#endif
    test_vad_and_ptt_gate();

    if (g_failures == 0) {
        std::printf("vad_ptt_devices: all checks passed\n");
        return 0;
    }
    std::printf("vad_ptt_devices: %d failure(s)\n", g_failures);
    return 1;
}

#else  // !VOICECAT_HAS_NET

int main() {
    std::printf("vad_ptt_devices: SKIP (VOICECAT_HAS_NET not defined)\n");
    return 0;
}

#endif  // VOICECAT_HAS_NET