Implements docs/roadmap.md M3: multiple concurrent streams per user (MIC + SCREEN_AUDIO + AUX_DEVICE), independent per-stream receiver gain/mute/noise- reduction, talk indicators, and enforced per-channel Opus configurability (mono/stereo, bitrate, frame size, FEC/DTX, application). Bugs fixed along the way (found while implementing, not pre-existing scope): - Server hard-coded stream_id=1 for every announce, so a second stream from the same user silently overwrote the first in SessionRegistry::set_user_stream. Now a per-session counter (ConnSession::next_stream_id_); handle_stream_stop validates against announced_stream_ids_ before clearing. - Client dropped mode/dtx/complexity/application from effective_audio even for the single M2 stream -- only sample_rate/bitrate_bps/frame_ms/fec were ever applied to OpusParams. Fixed on both the send (handle_stream_announce_result) and receive (sync_remote_streams) paths via a shared opus_params_from_audio_config() helper. - OpusEncoder always used OPUS_APPLICATION_VOIP; added OpusParams::application and wired it through. - on_playback's per-stream decode passed the wrong frame_size to opus_decode (total samples instead of samples-per-channel), which would have overflowed the decode buffer for any stereo stream. - teardown_voice() raced when called concurrently from run_io()'s own cleanup and from disconnect() on a different thread -- both could see udp_thread_/talk_timer_thread_ as joinable() at once and race to join() the same std::thread (intermittent std::system_error under ctest). Fixed with a teardown_mu_ guard instead of carrying the flake forward. New: - Per-channel AudioConfig: SessionRegistry now seeds Lobby (mono/24kbps/VOIP/ FEC+DTX) and a new "Music Room" channel (stereo/128kbps/AUDIO/no DTX); handle_stream_announce enforces the channel's config, clamping (not overriding) bitrate_bps to its ceiling. - core/src/core/client.h/.cpp: local-stream state is now a std::unordered_map<int, LocalStream> keyed by vc_stream_kind, with request_id-correlated announce/result handling (request_id already round-tripped on the wire; just wasn't read before). on_capture_frame is kind-aware and upmixes mono capture to stereo when a stream's config calls for it. set_self_mute's mic_muted now only gates the MIC kind. NS is wired through set_remote_stream. New run_talk_timer() thread emits VC_EVENT_TALK_STATE from both remote and local edge detection. - core/src/audio/audio_engine.h/.cpp: kind-keyed injection taps (inject_capture), stereo-to-mono downmix at the decode/mix boundary, RemoteStream gains recv_ns (lazy ApmProcessor) + noise_reduction_enabled and last_voice_ms/talking; new set_stream_noise_reduction() and poll_talk_transitions(). - core/src/session/session.h/.cpp: Stream now carries the full AudioConfig, not just sample_rate/frame_ms. - New additive C ABI (core/include/voicecat.h): vc_audio_config + vc_get_stream_audio_config (effective Opus config for any stream you own or a peer's); vc_test_inject_capture (test-only synthetic PCM injection, clearly marked, mirrors AudioEngine::inject_capture). - tests/test_m3_multistream.cpp: the M3 exit criterion through the real ABI (mirrors test_voice_client_abi.cpp's approach, not raw sockets) -- two concurrent local streams, independent gain/mute/NS control, per-channel config divergence via vc_get_stream_audio_config, talk indicators. Explicitly out of scope for this pass (tracked in PROGRESS.md, not silently dropped): VAD/PTT input gate + device enumeration; real WASAPI loopback capture for SCREEN_AUDIO (synthetic injection only); true stereo playback output (AudioEngine's mixer/output device stays mono -- Opus itself is fully stereo-correct on the wire). ctest --test-dir build/m1-dev: 11/11 green, verified across 3 consecutive full-suite runs plus 8 standalone runs of the new test. Co-Authored-By: Claude Sonnet 4.6 <noreply@anthropic.com>
119 lines
4.0 KiB
C++
119 lines
4.0 KiB
C++
/*
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* codec/opus_codec.h — Opus encode/decode (libopus 1.6).
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*
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* Design: docs/voice.md §3–4. Per-channel AudioConfig (mono/stereo, bitrate, frame size,
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* FEC, DTX, complexity). The server relays Opus payloads unmodified (no transcode).
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*/
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#ifndef VOICECAT_CODEC_OPUS_CODEC_H
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#define VOICECAT_CODEC_OPUS_CODEC_H
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#include <cstdint>
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#include <vector>
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#ifdef VOICECAT_HAS_OPUS
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#include <opus/opus.h>
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#endif
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namespace voicecat::codec {
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// Mirrors voicecat::v1::OpusApplication (core/proto/voicecat.proto) without depending on
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// generated protobuf headers from this low-level codec module.
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enum class OpusApplication {
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Voip = 0, // speech, optimized for low-rate intelligibility
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Audio = 1, // music/screen-audio, optimized for fidelity
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LowDelay = 2, // monitoring, minimal algorithmic delay
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};
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struct OpusParams {
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uint32_t sample_rate = 48000;
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uint32_t bitrate_bps = 24000;
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uint32_t frame_ms = 20;
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bool stereo = false;
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bool fec = true;
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bool dtx = false;
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uint32_t complexity = 10;
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uint32_t expected_packet_loss = 0; // % 0..100
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OpusApplication application = OpusApplication::Voip;
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};
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// Returns frame_samples for a given sample_rate + frame_ms.
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inline int opus_frame_samples(const OpusParams& p) {
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return static_cast<int>(p.sample_rate / 1000 * p.frame_ms);
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}
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class OpusEncoder {
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public:
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OpusEncoder() = default;
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~OpusEncoder() { destroy(); }
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OpusEncoder(const OpusEncoder&) = delete;
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OpusEncoder& operator=(const OpusEncoder&) = delete;
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// Initialise with the given params. Must be called before encode().
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// Returns true on success; check error_string() on failure.
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bool init(const OpusParams& p);
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// Encode one frame of PCM (frame_ms ms @ sample_rate Hz, mono or stereo).
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// pcm: interleaved int16 samples (frame_samples * channels samples).
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// out_buf: caller-allocated output buffer (recommend >= 4000 bytes).
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// Returns number of bytes written to out_buf, or -1 on error.
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int encode(const int16_t* pcm, int frame_samples, uint8_t* out_buf, int out_cap);
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void destroy();
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bool valid() const { return enc_ != nullptr; }
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int frame_samples()const { return frame_samples_; }
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int channels() const { return channels_; }
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const char* error_string() const { return err_; }
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private:
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#ifdef VOICECAT_HAS_OPUS
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::OpusEncoder* enc_ = nullptr;
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#else
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void* enc_ = nullptr;
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#endif
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int frame_samples_ = 0;
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int channels_ = 1;
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const char* err_ = nullptr;
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};
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class OpusDecoder {
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public:
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OpusDecoder() = default;
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~OpusDecoder() { destroy(); }
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OpusDecoder(const OpusDecoder&) = delete;
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OpusDecoder& operator=(const OpusDecoder&) = delete;
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// Initialise. Must be called before decode().
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bool init(const OpusParams& p);
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// Decode one Opus packet into out_pcm (frame_samples * channels int16 samples).
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// opus_data=nullptr, len=0 → PLC (free, always enabled by libopus).
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// fec=true, next valid packet in opus_data → FEC recovery from previous loss.
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// Returns number of samples decoded (= frame_samples), or -1 on error.
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int decode(const uint8_t* opus_data, int len, int16_t* out_pcm, int max_samples,
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bool fec = false);
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void destroy();
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bool valid() const { return dec_ != nullptr; }
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int frame_samples()const { return frame_samples_; }
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int channels() const { return channels_; }
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const char* error_string() const { return err_; }
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private:
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#ifdef VOICECAT_HAS_OPUS
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::OpusDecoder* dec_ = nullptr;
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#else
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void* dec_ = nullptr;
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#endif
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int frame_samples_ = 0;
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int channels_ = 1;
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const char* err_ = nullptr;
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};
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} // namespace voicecat::codec
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#endif // VOICECAT_CODEC_OPUS_CODEC_H
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