feat(M2): UDP voice/media plane -- SFU relay, Opus, AEAD, jitter buffer
Adds the full voice pipeline: 14-byte binary frame header, ChaCha20-Poly1305 AEAD keyed from the TLS exporter, libopus encode/decode with FEC/PLC/DTX, an adaptive per-ssrc jitter buffer, a miniaudio capture/playback engine, an APM passthrough stub, and the UdpBinding/StreamAnnounce signaling chain wired through ConnSession/SessionRegistry into a new server-side SFU (MediaRelay) that decrypts and re-encrypts frames per channel member. Exit criterion verified: test_m2_voice — two headless clients relay 50 encrypted Opus frames through the server; ctest --preset m1-dev is 9/9 green. Also corrects protocol.md's UdpBinding diagram, which described the UDP-side binding packet as AEAD-sealed when it is in fact a plaintext bootstrap frame (separate from the TCP/TLS UdpBinding ack). Co-Authored-By: Claude Sonnet 4.6 <noreply@anthropic.com>
This commit is contained in:
21
core/src/audio/apm_processor.cpp
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21
core/src/audio/apm_processor.cpp
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@@ -0,0 +1,21 @@
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#include "audio/apm_processor.h"
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namespace voicecat::audio {
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// ── ApmPassthrough ────────────────────────────────────────────────────────────
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// No-op: returns true (VAD always open), does not modify PCM.
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// Replaced by WebrtcApmProcessor when VOICECAT_HAS_APM is defined.
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class ApmPassthrough final : public ApmProcessor {
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public:
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void process_render(const int16_t*, int, int) override {}
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bool process_capture(int16_t*, int, int) override { return true; }
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};
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std::unique_ptr<ApmProcessor> ApmProcessor::create() {
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#ifdef VOICECAT_HAS_APM
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// TODO(M3): return std::make_unique<WebrtcApmProcessor>();
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#endif
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return std::make_unique<ApmPassthrough>();
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}
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} // namespace voicecat::audio
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33
core/src/audio/apm_processor.h
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33
core/src/audio/apm_processor.h
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/*
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* audio/apm_processor.h — Send-side audio processing module (AEC/NS/AGC/VAD).
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*
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* Design: docs/voice.md §11. Uses webrtc-audio-processing when VOICECAT_HAS_APM is defined;
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* falls back to a no-op passthrough (VAD always open, PCM unmodified) otherwise.
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*/
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#ifndef VOICECAT_AUDIO_APM_PROCESSOR_H
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#define VOICECAT_AUDIO_APM_PROCESSOR_H
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#include <cstdint>
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#include <memory>
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namespace voicecat::audio {
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class ApmProcessor {
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public:
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virtual ~ApmProcessor() = default;
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// Feed the most recent playback reference (for AEC). Call before process_capture().
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virtual void process_render(const int16_t* pcm, int samples, int sample_rate) = 0;
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// Process one capture frame in-place (AEC, NS, AGC).
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// Returns true if VAD detects speech (or always true in passthrough mode).
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// Returns false → caller should skip encode/send (silence gate).
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virtual bool process_capture(int16_t* pcm, int samples, int sample_rate) = 0;
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// Factory: returns a real APM if VOICECAT_HAS_APM is defined, else a passthrough.
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static std::unique_ptr<ApmProcessor> create();
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};
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} // namespace voicecat::audio
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#endif // VOICECAT_AUDIO_APM_PROCESSOR_H
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@@ -1,8 +1,261 @@
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#ifdef VOICECAT_HAS_AUDIO
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#define MINIAUDIO_IMPLEMENTATION
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#include <miniaudio.h>
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#endif
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#include "audio/audio_engine.h"
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#include <algorithm>
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#include <cstring>
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namespace voicecat::audio {
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// M0 stub. Capture/playback (miniaudio), APM DSP, jitter buffer, and mixer land in M2/M3.
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// See docs/voice.md §8–11.
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// ── JitterBuffer ─────────────────────────────────────────────────────────────
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void JitterBuffer::push(Frame f) {
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std::lock_guard lk(mu_);
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uint32_t ts = f.timestamp;
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// Jitter estimation (EWMA of inter-arrival gap vs expected gap).
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if (!first_push_) {
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uint32_t arrived_gap = ts - last_push_ts_;
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uint32_t expected_gap = 960; // 20 ms @48k; TODO: derive from params
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uint32_t diff = (arrived_gap > expected_gap) ? (arrived_gap - expected_gap)
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: (expected_gap - arrived_gap);
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jitter_est_ = (jitter_est_ * 7 + diff) / 8;
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uint32_t depth = std::clamp(jitter_est_ * 2 + 960u, 960u, 48000u * 200u / 1000u);
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target_depth_ms_.store(depth * 1000u / 48000u, std::memory_order_relaxed);
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}
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last_push_ts_ = ts;
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first_push_ = false;
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buf_.emplace(ts, std::move(f));
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}
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std::optional<JitterBuffer::Frame> JitterBuffer::pop(uint32_t playout_ts) {
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std::unique_lock lk(mu_, std::try_to_lock);
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if (!lk) return std::nullopt; // contended — caller does PLC
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if (buf_.empty()) return std::nullopt;
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auto it = buf_.begin();
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uint32_t ts = it->first;
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// Drop frames that are too old (> 500 ms late).
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if (static_cast<int32_t>(playout_ts - ts) > static_cast<int32_t>(kLateDropSamples)) {
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lost_.fetch_add(1, std::memory_order_relaxed);
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buf_.erase(it);
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return std::nullopt;
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}
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// Return frame only when it's due.
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if (static_cast<int32_t>(ts - playout_ts) > 0) return std::nullopt;
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Frame f = std::move(it->second);
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buf_.erase(it);
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return f;
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}
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void JitterBuffer::reset() {
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std::lock_guard lk(mu_);
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buf_.clear();
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lost_.store(0);
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first_push_ = true;
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}
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// ── AudioEngine ──────────────────────────────────────────────────────────────
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AudioEngine::AudioEngine() {
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inject_ring_.resize(kInjectCapSamples, 0);
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}
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AudioEngine::~AudioEngine() { stop(); }
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bool AudioEngine::start(const AudioParams& p, CaptureCallback capture_cb) {
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if (running_.load()) return false;
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params_ = p;
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capture_cb_ = std::move(capture_cb);
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frame_samples_ = static_cast<int>(p.sample_rate / 1000 * p.frame_ms);
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running_.store(true, std::memory_order_release);
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#ifdef VOICECAT_HAS_AUDIO
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// ── Capture device ──────────────────────────────────────────────────────
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ma_device_config cap_cfg = ma_device_config_init(ma_device_type_capture);
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cap_cfg.capture.format = ma_format_s16;
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cap_cfg.capture.channels = p.channels;
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cap_cfg.sampleRate = p.sample_rate;
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cap_cfg.dataCallback = capture_data_cb;
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cap_cfg.pUserData = this;
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cap_cfg.capture.pDeviceID = nullptr; // always default for now
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if (ma_device_init(nullptr, &cap_cfg, &capture_device_) == MA_SUCCESS) {
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if (ma_device_start(&capture_device_) == MA_SUCCESS) {
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capture_started_ = true;
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} else {
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ma_device_uninit(&capture_device_);
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}
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}
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// ── Playback device ─────────────────────────────────────────────────────
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ma_device_config pb_cfg = ma_device_config_init(ma_device_type_playback);
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pb_cfg.playback.format = ma_format_s16;
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pb_cfg.playback.channels = p.channels;
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pb_cfg.sampleRate = p.sample_rate;
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pb_cfg.dataCallback = playback_data_cb;
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pb_cfg.pUserData = this;
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pb_cfg.playback.pDeviceID = nullptr;
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if (ma_device_init(nullptr, &pb_cfg, &playback_device_) == MA_SUCCESS) {
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if (ma_device_start(&playback_device_) == MA_SUCCESS) {
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playback_started_ = true;
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} else {
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ma_device_uninit(&playback_device_);
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}
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}
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#endif // VOICECAT_HAS_AUDIO
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return true;
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}
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void AudioEngine::stop() {
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if (!running_.exchange(false)) return;
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#ifdef VOICECAT_HAS_AUDIO
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if (capture_started_) {
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ma_device_stop(&capture_device_);
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ma_device_uninit(&capture_device_);
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capture_started_ = false;
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}
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if (playback_started_) {
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ma_device_stop(&playback_device_);
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ma_device_uninit(&playback_device_);
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playback_started_ = false;
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}
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#endif
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}
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void AudioEngine::inject_capture(const int16_t* pcm, size_t n) {
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std::lock_guard lk(inject_mu_);
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size_t w = inject_write_.load(std::memory_order_relaxed);
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for (size_t i = 0; i < n; ++i)
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inject_ring_[(w + i) % kInjectCapSamples] = pcm[i];
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inject_write_.store(w + n, std::memory_order_release);
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// Fire capture_cb_ for each complete frame now available.
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while (true) {
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size_t r = inject_read_.load(std::memory_order_relaxed);
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size_t avail = inject_write_.load(std::memory_order_acquire) - r;
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if (avail < static_cast<size_t>(frame_samples_)) break;
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std::vector<int16_t> frame(frame_samples_);
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for (int i = 0; i < frame_samples_; ++i)
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frame[i] = inject_ring_[(r + i) % kInjectCapSamples];
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inject_read_.store(r + frame_samples_, std::memory_order_release);
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if (capture_cb_) capture_cb_(frame.data(), frame_samples_);
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}
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}
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void AudioEngine::push_recv_frame(uint32_t ssrc, JitterBuffer::Frame f) {
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std::lock_guard lk(streams_mu_);
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streams_[ssrc].jitter.push(std::move(f));
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}
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void AudioEngine::set_stream_gain(uint32_t ssrc, float gain) {
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std::lock_guard lk(streams_mu_);
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streams_[ssrc].gain = gain;
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}
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void AudioEngine::set_stream_mute(uint32_t ssrc, bool mute) {
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std::lock_guard lk(streams_mu_);
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streams_[ssrc].mute = mute;
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}
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void AudioEngine::remove_stream(uint32_t ssrc) {
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std::lock_guard lk(streams_mu_);
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streams_.erase(ssrc);
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}
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uint32_t AudioEngine::stream_packets_lost(uint32_t ssrc) const {
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std::lock_guard lk(streams_mu_);
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auto it = streams_.find(ssrc);
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return (it != streams_.end()) ? it->second.jitter.packets_lost() : 0;
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}
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uint32_t AudioEngine::stream_target_depth_ms(uint32_t ssrc) const {
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std::lock_guard lk(streams_mu_);
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auto it = streams_.find(ssrc);
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return (it != streams_.end()) ? it->second.jitter.target_depth_ms() : 40;
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}
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#ifdef VOICECAT_HAS_OPUS
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void AudioEngine::init_recv_stream(uint32_t ssrc, const codec::OpusParams& p) {
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std::lock_guard lk(streams_mu_);
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streams_[ssrc].decoder.init(p);
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}
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#endif
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#ifdef VOICECAT_HAS_AUDIO
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void AudioEngine::capture_data_cb(ma_device* dev, void* /*out*/,
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const void* in, ma_uint32 frame_count) {
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auto* self = static_cast<AudioEngine*>(dev->pUserData);
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self->on_capture(static_cast<const int16_t*>(in), frame_count);
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}
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void AudioEngine::on_capture(const int16_t* pcm, ma_uint32 frames) {
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if (capture_cb_) capture_cb_(pcm, static_cast<int>(frames));
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}
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void AudioEngine::playback_data_cb(ma_device* dev, void* out,
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const void* /*in*/, ma_uint32 frame_count) {
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auto* self = static_cast<AudioEngine*>(dev->pUserData);
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self->on_playback(static_cast<int16_t*>(out), frame_count);
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}
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void AudioEngine::on_playback(int16_t* out, ma_uint32 frames) {
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std::memset(out, 0, frames * params_.channels * sizeof(int16_t));
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std::unique_lock lk(streams_mu_, std::try_to_lock);
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if (!lk) return; // contended: emit silence this period
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#ifdef VOICECAT_HAS_OPUS
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std::vector<int32_t> mix(frames * params_.channels, 0);
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for (auto& [ssrc, stream] : streams_) {
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if (stream.mute || !stream.decoder.valid()) continue;
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auto maybe_frame = stream.jitter.pop(stream.playout_ts);
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std::vector<int16_t> pcm(frames * params_.channels);
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int n;
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if (maybe_frame) {
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n = stream.decoder.decode(
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maybe_frame->payload.data(),
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static_cast<int>(maybe_frame->payload.size()),
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pcm.data(), static_cast<int>(pcm.size()));
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} else {
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n = stream.decoder.decode(nullptr, 0, pcm.data(),
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static_cast<int>(pcm.size()));
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}
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if (n > 0) {
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float g = stream.gain;
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for (int i = 0; i < n * static_cast<int>(params_.channels); ++i)
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mix[i] += static_cast<int32_t>(static_cast<float>(pcm[i]) * g);
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}
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stream.playout_ts += frames;
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}
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for (ma_uint32 i = 0; i < frames * params_.channels; ++i)
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out[i] = static_cast<int16_t>(std::clamp(mix[i], -32768, 32767));
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#else
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(void)out;
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(void)frames;
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#endif
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}
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#endif // VOICECAT_HAS_AUDIO
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} // namespace voicecat::audio
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@@ -6,33 +6,160 @@
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* ... → Opus decode → per-user recv NS (listener-chosen) → gain/mute → mix → playback
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*
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* REAL-TIME RULE: audio-callback threads never allocate, lock, or block (architecture.md §3).
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*
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* STATUS: M0 stub.
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* The JitterBuffer and per-stream maps are accessed only under a try_lock; a failed lock
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* causes PLC for that period (acceptable for M2; lock-free ring buffer is the M3 upgrade).
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*/
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#ifndef VOICECAT_AUDIO_AUDIO_ENGINE_H
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#define VOICECAT_AUDIO_AUDIO_ENGINE_H
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#include <atomic>
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#include <cstdint>
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#include <functional>
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#include <map>
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#include <memory>
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#include <mutex>
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#include <optional>
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#include <unordered_map>
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#include <vector>
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#ifdef VOICECAT_HAS_AUDIO
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// miniaudio single-header — MINIAUDIO_IMPLEMENTATION defined in audio_engine.cpp
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#include <miniaudio.h>
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#endif
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#ifdef VOICECAT_HAS_OPUS
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#include "codec/opus_codec.h"
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#endif
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namespace voicecat::audio {
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// Adaptive per-ssrc jitter buffer (voice.md §5). TODO(M2).
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// ── JitterBuffer ─────────────────────────────────────────────────────────────
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// Per-ssrc adaptive jitter buffer. Thread-safe via internal mutex.
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class JitterBuffer {
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public:
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uint32_t target_depth_ms() const { return target_depth_ms_; }
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struct Frame {
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uint16_t seq;
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uint32_t timestamp;
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bool fec_present;
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std::vector<uint8_t> payload;
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};
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// Insert an incoming frame. Thread-safe.
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void push(Frame f);
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// Return the next frame whose timestamp <= playout_ts, or nullopt (caller should PLC).
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// Drops frames that are too old (more than kLateDropSamples late).
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std::optional<Frame> pop(uint32_t playout_ts);
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uint32_t target_depth_ms()const { return target_depth_ms_.load(); }
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uint32_t packets_lost() const { return lost_.load(); }
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void reset();
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private:
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uint32_t target_depth_ms_ = 40;
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static constexpr uint32_t kLateDropSamples = 48000 / 2; // 500 ms @48 kHz
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mutable std::mutex mu_;
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std::map<uint32_t, Frame> buf_; // keyed by timestamp (u32 wraps are handled below)
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std::atomic<uint32_t> target_depth_ms_{40};
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std::atomic<uint32_t> lost_{0};
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// Jitter estimation (EWMA).
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uint32_t last_push_ts_ = 0; // local clock estimate on last push
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uint32_t jitter_est_ = 0; // EWMA jitter in samples
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bool first_push_ = true;
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};
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// Owns miniaudio capture/playback, the APM instances, codecs, jitter buffers, and the mixer.
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// ── AudioParams ──────────────────────────────────────────────────────────────
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struct AudioParams {
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uint32_t sample_rate = 48000;
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uint32_t channels = 1;
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uint32_t frame_ms = 20;
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std::string capture_device_id; // "" = default
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std::string playback_device_id; // "" = default
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};
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// ── AudioEngine ──────────────────────────────────────────────────────────────
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// Owns miniaudio capture/playback, per-ssrc jitter buffers + Opus decoders, and the mixer.
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class AudioEngine {
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public:
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// TODO(M2): start/stop capture+playback; push/pull frames via lock-free ring buffers.
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bool running() const { return running_; }
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// Callback type for encoded capture frames ready to be sent.
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using CaptureCallback = std::function<void(const int16_t* pcm, int samples)>;
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AudioEngine();
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~AudioEngine();
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AudioEngine(const AudioEngine&) = delete;
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AudioEngine& operator=(const AudioEngine&) = delete;
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// Start capture+playback devices. capture_cb is called on the encode thread
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// for each capture frame (not on the audio callback thread).
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bool start(const AudioParams& p, CaptureCallback capture_cb = nullptr);
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void stop();
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||||
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bool running() const { return running_.load(std::memory_order_acquire); }
|
||||
|
||||
// Inject synthetic PCM directly into the capture pipeline (bypasses real device).
|
||||
// Thread-safe; can be called from any thread including tests.
|
||||
void inject_capture(const int16_t* pcm, size_t n);
|
||||
|
||||
// Called by the net thread when a decoded voice frame arrives for a remote stream.
|
||||
void push_recv_frame(uint32_t ssrc, JitterBuffer::Frame f);
|
||||
|
||||
// Per-stream receive-side controls (safe from any thread).
|
||||
void set_stream_gain(uint32_t ssrc, float gain); // 0.0–2.0, default 1.0
|
||||
void set_stream_mute(uint32_t ssrc, bool mute);
|
||||
void remove_stream(uint32_t ssrc);
|
||||
|
||||
// Get stats for a remote stream's jitter buffer.
|
||||
uint32_t stream_packets_lost(uint32_t ssrc) const;
|
||||
uint32_t stream_target_depth_ms(uint32_t ssrc) const;
|
||||
|
||||
#ifdef VOICECAT_HAS_OPUS
|
||||
// Configure the Opus decoder for an incoming ssrc (must be called before
|
||||
// push_recv_frame for that ssrc). Thread-safe.
|
||||
void init_recv_stream(uint32_t ssrc, const codec::OpusParams& p);
|
||||
#endif
|
||||
|
||||
private:
|
||||
bool running_ = false;
|
||||
#ifdef VOICECAT_HAS_AUDIO
|
||||
static void capture_data_cb(ma_device*, void*, const void*, ma_uint32);
|
||||
static void playback_data_cb(ma_device*, void*, const void*, ma_uint32);
|
||||
void on_capture(const int16_t* pcm, ma_uint32 frames);
|
||||
void on_playback(int16_t* out, ma_uint32 frames);
|
||||
|
||||
ma_device capture_device_{};
|
||||
ma_device playback_device_{};
|
||||
bool capture_started_ = false;
|
||||
bool playback_started_ = false;
|
||||
#endif
|
||||
|
||||
AudioParams params_{};
|
||||
CaptureCallback capture_cb_;
|
||||
std::atomic<bool> running_{false};
|
||||
|
||||
// Inject ring: stores raw int16 PCM written by inject_capture().
|
||||
// The encode thread reads from this (no real capture device needed in tests).
|
||||
std::mutex inject_mu_;
|
||||
std::vector<int16_t> inject_ring_; // circular, size = frame_samples_
|
||||
std::atomic<size_t> inject_write_{0};
|
||||
std::atomic<size_t> inject_read_{0};
|
||||
static constexpr size_t kInjectCapSamples = 48000 * 2; // 2 s @48 kHz mono
|
||||
|
||||
// Per remote stream (protected by streams_mu_).
|
||||
struct RemoteStream {
|
||||
JitterBuffer jitter;
|
||||
#ifdef VOICECAT_HAS_OPUS
|
||||
codec::OpusDecoder decoder;
|
||||
#endif
|
||||
float gain = 1.0f;
|
||||
bool mute = false;
|
||||
uint32_t playout_ts = 0;
|
||||
};
|
||||
mutable std::mutex streams_mu_;
|
||||
std::unordered_map<uint32_t, RemoteStream> streams_;
|
||||
|
||||
int frame_samples_ = 960; // 20 ms @48 kHz
|
||||
};
|
||||
|
||||
} // namespace voicecat::audio
|
||||
|
||||
Reference in New Issue
Block a user